From a0abacd82c553d18f8e359f7296246b3009cc207 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 4 Mar 2012 14:06:36 +0000 Subject: ASoC: ak4641: Convert to module_i2c_driver() Signed-off-by: Mark Brown --- sound/soc/codecs/ak4641.c | 18 +----------------- 1 file changed, 1 insertion(+), 17 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/ak4641.c b/sound/soc/codecs/ak4641.c index c4d165a4bddf..611e8f05d87e 100644 --- a/sound/soc/codecs/ak4641.c +++ b/sound/soc/codecs/ak4641.c @@ -641,23 +641,7 @@ static struct i2c_driver ak4641_i2c_driver = { .id_table = ak4641_i2c_id, }; -static int __init ak4641_modinit(void) -{ - int ret; - - ret = i2c_add_driver(&ak4641_i2c_driver); - if (ret != 0) - pr_err("Failed to register AK4641 I2C driver: %d\n", ret); - - return ret; -} -module_init(ak4641_modinit); - -static void __exit ak4641_exit(void) -{ - i2c_del_driver(&ak4641_i2c_driver); -} -module_exit(ak4641_exit); +module_i2c_driver(ak4641_i2c_driver); MODULE_DESCRIPTION("SoC AK4641 driver"); MODULE_AUTHOR("Harald Welte "); -- cgit v1.2.3 From 253322c18830965331e54ee33c5e8064a2f15717 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 4 Mar 2012 14:20:18 +0000 Subject: ASoC: ak4641: Push GPIO allocation out into the I2C probe It's more idiomatic to do this and it means we don't try to bring up the card if the CODEC didn't manage to bind successfully. Signed-off-by: Mark Brown --- sound/soc/codecs/ak4641.c | 92 ++++++++++++++++++++++++----------------------- 1 file changed, 48 insertions(+), 44 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/ak4641.c b/sound/soc/codecs/ak4641.c index 611e8f05d87e..7f42d4a673e3 100644 --- a/sound/soc/codecs/ak4641.c +++ b/sound/soc/codecs/ak4641.c @@ -517,67 +517,24 @@ static int ak4641_resume(struct snd_soc_codec *codec) static int ak4641_probe(struct snd_soc_codec *codec) { - struct ak4641_platform_data *pdata = codec->dev->platform_data; int ret; - - if (pdata) { - if (gpio_is_valid(pdata->gpio_power)) { - ret = gpio_request_one(pdata->gpio_power, - GPIOF_OUT_INIT_LOW, "ak4641 power"); - if (ret) - goto err_out; - } - if (gpio_is_valid(pdata->gpio_npdn)) { - ret = gpio_request_one(pdata->gpio_npdn, - GPIOF_OUT_INIT_LOW, "ak4641 npdn"); - if (ret) - goto err_gpio; - - udelay(1); /* > 150 ns */ - gpio_set_value(pdata->gpio_npdn, 1); - } - } - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - goto err_register; + return ret; } /* power on device */ ak4641_set_bias_level(codec, SND_SOC_BIAS_STANDBY); return 0; - -err_register: - if (pdata) { - if (gpio_is_valid(pdata->gpio_power)) - gpio_set_value(pdata->gpio_power, 0); - if (gpio_is_valid(pdata->gpio_npdn)) - gpio_free(pdata->gpio_npdn); - } -err_gpio: - if (pdata && gpio_is_valid(pdata->gpio_power)) - gpio_free(pdata->gpio_power); -err_out: - return ret; } static int ak4641_remove(struct snd_soc_codec *codec) { - struct ak4641_platform_data *pdata = codec->dev->platform_data; - ak4641_set_bias_level(codec, SND_SOC_BIAS_OFF); - if (pdata) { - if (gpio_is_valid(pdata->gpio_power)) { - gpio_set_value(pdata->gpio_power, 0); - gpio_free(pdata->gpio_power); - } - if (gpio_is_valid(pdata->gpio_npdn)) - gpio_free(pdata->gpio_npdn); - } return 0; } @@ -604,6 +561,7 @@ static struct snd_soc_codec_driver soc_codec_dev_ak4641 = { static int __devinit ak4641_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { + struct ak4641_platform_data *pdata = i2c->dev.platform_data; struct ak4641_priv *ak4641; int ret; @@ -612,16 +570,62 @@ static int __devinit ak4641_i2c_probe(struct i2c_client *i2c, if (!ak4641) return -ENOMEM; + if (pdata) { + if (gpio_is_valid(pdata->gpio_power)) { + ret = gpio_request_one(pdata->gpio_power, + GPIOF_OUT_INIT_LOW, "ak4641 power"); + if (ret) + goto err_out; + } + if (gpio_is_valid(pdata->gpio_npdn)) { + ret = gpio_request_one(pdata->gpio_npdn, + GPIOF_OUT_INIT_LOW, "ak4641 npdn"); + if (ret) + goto err_gpio; + + udelay(1); /* > 150 ns */ + gpio_set_value(pdata->gpio_npdn, 1); + } + } + i2c_set_clientdata(i2c, ak4641); ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_ak4641, ak4641_dai, ARRAY_SIZE(ak4641_dai)); + if (ret != 0) + goto err_gpio2; + + return 0; + +err_gpio2: + if (pdata) { + if (gpio_is_valid(pdata->gpio_power)) + gpio_set_value(pdata->gpio_power, 0); + if (gpio_is_valid(pdata->gpio_npdn)) + gpio_free(pdata->gpio_npdn); + } +err_gpio: + if (pdata && gpio_is_valid(pdata->gpio_power)) + gpio_free(pdata->gpio_power); +err_out: return ret; } static int __devexit ak4641_i2c_remove(struct i2c_client *i2c) { + struct ak4641_platform_data *pdata = i2c->dev.platform_data; + snd_soc_unregister_codec(&i2c->dev); + + if (pdata) { + if (gpio_is_valid(pdata->gpio_power)) { + gpio_set_value(pdata->gpio_power, 0); + gpio_free(pdata->gpio_power); + } + if (gpio_is_valid(pdata->gpio_npdn)) + gpio_free(pdata->gpio_npdn); + } + return 0; } -- cgit v1.2.3 From 01b9d99a1f45befa604543ead29f44fdb0878844 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Wed, 7 Mar 2012 10:38:25 +0000 Subject: ASoC: core: Add card mutex locking subclasses This is the first part of a change that is intended to improve ASoC locking protection for DAPM and PCM operations. This part of the series adds a mutex class for the soc_card mutex. The SND_SOC_CARD_CLASS_INIT class is used for card initialisation only whilst the SND_SOC_CARD_CLASS_PCM class is used for the forth coming Dynamic PCM operations. The new mutex classes are required otherwise we will see a false positive mutex deadlock warning between the card initialisation and the PCM operations (something that would never deadlock in real life). Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown --- include/sound/soc.h | 5 +++++ sound/soc/soc-core.c | 2 +- 2 files changed, 6 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/include/sound/soc.h b/include/sound/soc.h index 2ebf7877c148..70de2f82e872 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -288,6 +288,11 @@ enum snd_soc_pcm_subclass { SND_SOC_PCM_CLASS_BE = 1, }; +enum snd_soc_card_subclass { + SND_SOC_CARD_CLASS_INIT = 0, + SND_SOC_CARD_CLASS_PCM = 1, +}; + int snd_soc_codec_set_sysclk(struct snd_soc_codec *codec, int clk_id, int source, unsigned int freq, int dir); int snd_soc_codec_set_pll(struct snd_soc_codec *codec, int pll_id, int source, diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index a4deebc0801a..a6da20a72d43 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1416,7 +1416,7 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) struct snd_soc_dai_link *dai_link; int ret, i, order; - mutex_lock(&card->mutex); + mutex_lock_nested(&card->mutex, SND_SOC_CARD_CLASS_INIT); if (card->instantiated) { mutex_unlock(&card->mutex); -- cgit v1.2.3 From a73fb2df01866b772a48fab93401fe3edbe0b38d Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Wed, 7 Mar 2012 10:38:26 +0000 Subject: ASoC: dapm: Use DAPM mutex for DAPM ops instead of codec mutex It has now become necessary to use a DAPM mutex instead of the codec mutex to lock the DAPM operations. This is due to the recent multi component support and forth coming Dynamic PCM updates. Currently we lock DAPM operations with the codec mutex of the calling RTD context. However, DAPM operations can span the whole card context and all components. This patch updates the DAPM operations that use the codec mutex to now use the DAPM mutex PCM subclass for all DAPM ops. We also add a mutex subclass for DAPM init and PCM operations. Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown --- include/sound/soc-dapm.h | 5 ++++ include/sound/soc.h | 1 + sound/soc/soc-core.c | 1 + sound/soc/soc-dapm.c | 59 ++++++++++++++++++++++++++++++++---------------- 4 files changed, 46 insertions(+), 20 deletions(-) (limited to 'sound') diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 8da3c2409060..055242e69dc3 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -432,6 +432,11 @@ enum snd_soc_dapm_type { snd_soc_dapm_dai, /* link to DAI structure */ }; +enum snd_soc_dapm_subclass { + SND_SOC_DAPM_CLASS_INIT = 0, + SND_SOC_DAPM_CLASS_PCM = 1, +}; + /* * DAPM audio route definition. * diff --git a/include/sound/soc.h b/include/sound/soc.h index 70de2f82e872..66fd9bc9789d 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -805,6 +805,7 @@ struct snd_soc_card { struct list_head list; struct mutex mutex; + struct mutex dapm_mutex; bool instantiated; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index a6da20a72d43..4a145cb43018 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3126,6 +3126,7 @@ int snd_soc_register_card(struct snd_soc_card *card) INIT_LIST_HEAD(&card->dapm_dirty); card->instantiated = 0; mutex_init(&card->mutex); + mutex_init(&card->dapm_mutex); mutex_lock(&client_mutex); list_add(&card->list, &card_list); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 6241490fff30..78aa19257143 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1949,6 +1949,8 @@ static int snd_soc_dapm_set_pin(struct snd_soc_dapm_context *dapm, */ int snd_soc_dapm_sync(struct snd_soc_dapm_context *dapm) { + int ret; + /* * Suppress early reports (eg, jacks syncing their state) to avoid * silly DAPM runs during card startup. @@ -1956,7 +1958,10 @@ int snd_soc_dapm_sync(struct snd_soc_dapm_context *dapm) if (!dapm->card || !dapm->card->instantiated) return 0; - return dapm_power_widgets(dapm, SND_SOC_DAPM_STREAM_NOP); + mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_PCM); + ret = dapm_power_widgets(dapm, SND_SOC_DAPM_STREAM_NOP); + mutex_unlock(&dapm->card->dapm_mutex); + return ret; } EXPORT_SYMBOL_GPL(snd_soc_dapm_sync); @@ -2122,6 +2127,7 @@ int snd_soc_dapm_add_routes(struct snd_soc_dapm_context *dapm, { int i, ret; + mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_INIT); for (i = 0; i < num; i++) { ret = snd_soc_dapm_add_route(dapm, route); if (ret < 0) { @@ -2131,6 +2137,7 @@ int snd_soc_dapm_add_routes(struct snd_soc_dapm_context *dapm, } route++; } + mutex_unlock(&dapm->card->dapm_mutex); return 0; } @@ -2203,12 +2210,14 @@ int snd_soc_dapm_weak_routes(struct snd_soc_dapm_context *dapm, int i, err; int ret = 0; + mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_INIT); for (i = 0; i < num; i++) { err = snd_soc_dapm_weak_route(dapm, route); if (err) ret = err; route++; } + mutex_unlock(&dapm->card->dapm_mutex); return ret; } @@ -2227,6 +2236,8 @@ int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm) struct snd_soc_dapm_widget *w; unsigned int val; + mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_INIT); + list_for_each_entry(w, &dapm->card->widgets, list) { if (w->new) @@ -2236,8 +2247,10 @@ int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm) w->kcontrols = kzalloc(w->num_kcontrols * sizeof(struct snd_kcontrol *), GFP_KERNEL); - if (!w->kcontrols) + if (!w->kcontrols) { + mutex_unlock(&dapm->card->dapm_mutex); return -ENOMEM; + } } switch(w->id) { @@ -2277,6 +2290,7 @@ int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm) } dapm_power_widgets(dapm, SND_SOC_DAPM_STREAM_NOP); + mutex_unlock(&dapm->card->dapm_mutex); return 0; } EXPORT_SYMBOL_GPL(snd_soc_dapm_new_widgets); @@ -2336,6 +2350,7 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); struct snd_soc_dapm_widget *widget = wlist->widgets[0]; struct snd_soc_codec *codec = widget->codec; + struct snd_soc_card *card = codec->card; struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; unsigned int reg = mc->reg; @@ -2362,7 +2377,7 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, /* old connection must be powered down */ connect = invert ? 1 : 0; - mutex_lock(&codec->mutex); + mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_PCM); change = snd_soc_test_bits(widget->codec, reg, mask, val); if (change) { @@ -2384,7 +2399,7 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, } } - mutex_unlock(&codec->mutex); + mutex_unlock(&card->dapm_mutex); return 0; } EXPORT_SYMBOL_GPL(snd_soc_dapm_put_volsw); @@ -2433,6 +2448,7 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); struct snd_soc_dapm_widget *widget = wlist->widgets[0]; struct snd_soc_codec *codec = widget->codec; + struct snd_soc_card *card = codec->card; struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; unsigned int val, mux, change; unsigned int mask, bitmask; @@ -2453,7 +2469,7 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, mask |= (bitmask - 1) << e->shift_r; } - mutex_lock(&codec->mutex); + mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_PCM); change = snd_soc_test_bits(widget->codec, e->reg, mask, val); if (change) { @@ -2475,7 +2491,7 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, } } - mutex_unlock(&codec->mutex); + mutex_unlock(&card->dapm_mutex); return change; } EXPORT_SYMBOL_GPL(snd_soc_dapm_put_enum_double); @@ -2512,6 +2528,7 @@ int snd_soc_dapm_put_enum_virt(struct snd_kcontrol *kcontrol, struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); struct snd_soc_dapm_widget *widget = wlist->widgets[0]; struct snd_soc_codec *codec = widget->codec; + struct snd_soc_card *card = codec->card; struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; int change; @@ -2521,7 +2538,7 @@ int snd_soc_dapm_put_enum_virt(struct snd_kcontrol *kcontrol, if (ucontrol->value.enumerated.item[0] >= e->max) return -EINVAL; - mutex_lock(&codec->mutex); + mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_PCM); change = widget->value != ucontrol->value.enumerated.item[0]; if (change) { @@ -2534,7 +2551,7 @@ int snd_soc_dapm_put_enum_virt(struct snd_kcontrol *kcontrol, } } - mutex_unlock(&codec->mutex); + mutex_unlock(&card->dapm_mutex); return ret; } EXPORT_SYMBOL_GPL(snd_soc_dapm_put_enum_virt); @@ -2599,6 +2616,7 @@ int snd_soc_dapm_put_value_enum_double(struct snd_kcontrol *kcontrol, struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); struct snd_soc_dapm_widget *widget = wlist->widgets[0]; struct snd_soc_codec *codec = widget->codec; + struct snd_soc_card *card = codec->card; struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; unsigned int val, mux, change; unsigned int mask; @@ -2617,7 +2635,7 @@ int snd_soc_dapm_put_value_enum_double(struct snd_kcontrol *kcontrol, mask |= e->mask << e->shift_r; } - mutex_lock(&codec->mutex); + mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_PCM); change = snd_soc_test_bits(widget->codec, e->reg, mask, val); if (change) { @@ -2639,7 +2657,7 @@ int snd_soc_dapm_put_value_enum_double(struct snd_kcontrol *kcontrol, } } - mutex_unlock(&codec->mutex); + mutex_unlock(&card->dapm_mutex); return change; } EXPORT_SYMBOL_GPL(snd_soc_dapm_put_value_enum_double); @@ -2676,12 +2694,12 @@ int snd_soc_dapm_get_pin_switch(struct snd_kcontrol *kcontrol, struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); const char *pin = (const char *)kcontrol->private_value; - mutex_lock(&card->mutex); + mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_PCM); ucontrol->value.integer.value[0] = snd_soc_dapm_get_pin_status(&card->dapm, pin); - mutex_unlock(&card->mutex); + mutex_unlock(&card->dapm_mutex); return 0; } @@ -2699,17 +2717,16 @@ int snd_soc_dapm_put_pin_switch(struct snd_kcontrol *kcontrol, struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); const char *pin = (const char *)kcontrol->private_value; - mutex_lock(&card->mutex); + mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_PCM); if (ucontrol->value.integer.value[0]) snd_soc_dapm_enable_pin(&card->dapm, pin); else snd_soc_dapm_disable_pin(&card->dapm, pin); - snd_soc_dapm_sync(&card->dapm); - - mutex_unlock(&card->mutex); + mutex_unlock(&card->dapm_mutex); + snd_soc_dapm_sync(&card->dapm); return 0; } EXPORT_SYMBOL_GPL(snd_soc_dapm_put_pin_switch); @@ -2827,6 +2844,7 @@ int snd_soc_dapm_new_controls(struct snd_soc_dapm_context *dapm, struct snd_soc_dapm_widget *w; int i; + mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_INIT); for (i = 0; i < num; i++) { w = snd_soc_dapm_new_control(dapm, widget); if (!w) { @@ -2837,6 +2855,7 @@ int snd_soc_dapm_new_controls(struct snd_soc_dapm_context *dapm, } widget++; } + mutex_unlock(&dapm->card->dapm_mutex); return 0; } EXPORT_SYMBOL_GPL(snd_soc_dapm_new_controls); @@ -2991,11 +3010,11 @@ static void soc_dapm_stream_event(struct snd_soc_dapm_context *dapm, int snd_soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, int stream, struct snd_soc_dai *dai, int event) { - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_card *card = rtd->card; - mutex_lock(&codec->mutex); - soc_dapm_stream_event(&codec->dapm, stream, dai, event); - mutex_unlock(&codec->mutex); + mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_PCM); + soc_dapm_stream_event(&card->dapm, stream, dai, event); + mutex_unlock(&card->dapm_mutex); return 0; } -- cgit v1.2.3 From 4edbb34577c98297f958f131e093a150b9f3226f Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Wed, 7 Mar 2012 10:38:27 +0000 Subject: ASoC: dapm: lock mixer & mux update power with DAPM mutex Both snd_soc_dapm_mux_update_power() and snd_soc_dapm_mixer_update_power() can be called internally within DAPM core (with DAPM mutex held) and externally. Provide some wrappers so that external users of both functions do not have to remember to hold the DAPM mutex. Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 36 ++++++++++++++++++++++++++++++------ 1 file changed, 30 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 78aa19257143..e1863d7e8012 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1719,7 +1719,7 @@ static inline void dapm_debugfs_cleanup(struct snd_soc_dapm_context *dapm) #endif /* test and update the power status of a mux widget */ -int snd_soc_dapm_mux_update_power(struct snd_soc_dapm_widget *widget, +static int soc_dapm_mux_update_power(struct snd_soc_dapm_widget *widget, struct snd_kcontrol *kcontrol, int mux, struct soc_enum *e) { struct snd_soc_dapm_path *path; @@ -1758,10 +1758,22 @@ int snd_soc_dapm_mux_update_power(struct snd_soc_dapm_widget *widget, return 0; } + +int snd_soc_dapm_mux_update_power(struct snd_soc_dapm_widget *widget, + struct snd_kcontrol *kcontrol, int mux, struct soc_enum *e) +{ + struct snd_soc_card *card = widget->dapm->card; + int ret; + + mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_PCM); + ret = soc_dapm_mux_update_power(widget, kcontrol, mux, e); + mutex_unlock(&card->dapm_mutex); + return ret; +} EXPORT_SYMBOL_GPL(snd_soc_dapm_mux_update_power); /* test and update the power status of a mixer or switch widget */ -int snd_soc_dapm_mixer_update_power(struct snd_soc_dapm_widget *widget, +static int soc_dapm_mixer_update_power(struct snd_soc_dapm_widget *widget, struct snd_kcontrol *kcontrol, int connect) { struct snd_soc_dapm_path *path; @@ -1790,6 +1802,18 @@ int snd_soc_dapm_mixer_update_power(struct snd_soc_dapm_widget *widget, return 0; } + +int snd_soc_dapm_mixer_update_power(struct snd_soc_dapm_widget *widget, + struct snd_kcontrol *kcontrol, int connect) +{ + struct snd_soc_card *card = widget->dapm->card; + int ret; + + mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_PCM); + ret = soc_dapm_mixer_update_power(widget, kcontrol, connect); + mutex_unlock(&card->dapm_mutex); + return ret; +} EXPORT_SYMBOL_GPL(snd_soc_dapm_mixer_update_power); /* show dapm widget status in sys fs */ @@ -2393,7 +2417,7 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, update.val = val; widget->dapm->update = &update; - snd_soc_dapm_mixer_update_power(widget, kcontrol, connect); + soc_dapm_mixer_update_power(widget, kcontrol, connect); widget->dapm->update = NULL; } @@ -2485,7 +2509,7 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, update.val = val; widget->dapm->update = &update; - snd_soc_dapm_mux_update_power(widget, kcontrol, mux, e); + soc_dapm_mux_update_power(widget, kcontrol, mux, e); widget->dapm->update = NULL; } @@ -2547,7 +2571,7 @@ int snd_soc_dapm_put_enum_virt(struct snd_kcontrol *kcontrol, widget->value = ucontrol->value.enumerated.item[0]; - snd_soc_dapm_mux_update_power(widget, kcontrol, widget->value, e); + soc_dapm_mux_update_power(widget, kcontrol, widget->value, e); } } @@ -2651,7 +2675,7 @@ int snd_soc_dapm_put_value_enum_double(struct snd_kcontrol *kcontrol, update.val = val; widget->dapm->update = &update; - snd_soc_dapm_mux_update_power(widget, kcontrol, mux, e); + soc_dapm_mux_update_power(widget, kcontrol, mux, e); widget->dapm->update = NULL; } -- cgit v1.2.3 From be09ad90e17b79fdb0d513a31e814ff4d42e3dff Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Wed, 7 Mar 2012 11:47:41 +0000 Subject: ASoC: core: Add platform DAI widget mapping Add platform driver support for CPU DAI DAPM widgets. Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown --- include/sound/soc-dai.h | 1 + sound/soc/soc-core.c | 14 ++++++++++++++ 2 files changed, 15 insertions(+) (limited to 'sound') diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index c429f248cf4e..3248fbc3326a 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -241,6 +241,7 @@ struct snd_soc_dai { struct snd_soc_dapm_widget *playback_widget; struct snd_soc_dapm_widget *capture_widget; + struct snd_soc_dapm_context dapm; /* DAI DMA data */ void *playback_dma_data; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4a145cb43018..42ce14485b92 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1074,6 +1074,7 @@ static int soc_probe_platform(struct snd_soc_card *card, { int ret = 0; const struct snd_soc_platform_driver *driver = platform->driver; + struct snd_soc_dai *dai; platform->card = card; platform->dapm.card = card; @@ -1087,6 +1088,14 @@ static int soc_probe_platform(struct snd_soc_card *card, snd_soc_dapm_new_controls(&platform->dapm, driver->dapm_widgets, driver->num_dapm_widgets); + /* Create DAPM widgets for each DAI stream */ + list_for_each_entry(dai, &dai_list, list) { + if (dai->dev != platform->dev) + continue; + + snd_soc_dapm_new_dai_widgets(&platform->dapm, dai); + } + if (driver->probe) { ret = driver->probe(platform); if (ret < 0) { @@ -1222,9 +1231,12 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num, int order) /* probe the cpu_dai */ if (!cpu_dai->probed && cpu_dai->driver->probe_order == order) { + cpu_dai->dapm.card = card; if (!try_module_get(cpu_dai->dev->driver->owner)) return -ENODEV; + snd_soc_dapm_new_dai_widgets(&cpu_dai->dapm, cpu_dai); + if (cpu_dai->driver->probe) { ret = cpu_dai->driver->probe(cpu_dai); if (ret < 0) { @@ -3242,6 +3254,7 @@ int snd_soc_register_dai(struct device *dev, dai->dev = dev; dai->driver = dai_drv; + dai->dapm.dev = dev; if (!dai->driver->ops) dai->driver->ops = &null_dai_ops; @@ -3318,6 +3331,7 @@ int snd_soc_register_dais(struct device *dev, dai->id = dai->driver->id; else dai->id = i; + dai->dapm.dev = dev; if (!dai->driver->ops) dai->driver->ops = &null_dai_ops; -- cgit v1.2.3 From d9b0951b96e4ee0d22fae0a30f0b53354ca541cd Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Wed, 7 Mar 2012 16:32:59 +0000 Subject: ASoC: dapm: Add platform stream event support Currently stream events are only perfomed on codec stream widgets only. There is now a need to be able to perform stream events on platform widgets too. e.g. we have the ABE platform driver with several DAI links to dummy codecs. We need to be able to perform stream events on any of the dummy codec DAI links. This patch also removes the snd_soc_dai * parameter since it's already contained within the rtd * parameter. Finally makle stream event return void since no one checks it anyway. Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown --- include/sound/soc-dapm.h | 4 +-- sound/soc/soc-core.c | 8 ++--- sound/soc/soc-dapm.c | 79 +++++++++++++++++++++++++++++++----------------- sound/soc/soc-pcm.c | 9 +++--- 4 files changed, 59 insertions(+), 41 deletions(-) (limited to 'sound') diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 055242e69dc3..6c64dbeb28ba 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -369,8 +369,8 @@ int snd_soc_dapm_weak_routes(struct snd_soc_dapm_context *dapm, const struct snd_soc_dapm_route *route, int num); /* dapm events */ -int snd_soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, int stream, - struct snd_soc_dai *dai, int event); +void snd_soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, int stream, + int event); void snd_soc_dapm_shutdown(struct snd_soc_card *card); /* external DAPM widget events */ diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 42ce14485b92..61b51b673d49 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -573,19 +573,16 @@ int snd_soc_suspend(struct device *dev) } for (i = 0; i < card->num_rtd; i++) { - struct snd_soc_dai *codec_dai = card->rtd[i].codec_dai; if (card->rtd[i].dai_link->ignore_suspend) continue; snd_soc_dapm_stream_event(&card->rtd[i], SNDRV_PCM_STREAM_PLAYBACK, - codec_dai, SND_SOC_DAPM_STREAM_SUSPEND); snd_soc_dapm_stream_event(&card->rtd[i], SNDRV_PCM_STREAM_CAPTURE, - codec_dai, SND_SOC_DAPM_STREAM_SUSPEND); } @@ -689,17 +686,16 @@ static void soc_resume_deferred(struct work_struct *work) } for (i = 0; i < card->num_rtd; i++) { - struct snd_soc_dai *codec_dai = card->rtd[i].codec_dai; if (card->rtd[i].dai_link->ignore_suspend) continue; snd_soc_dapm_stream_event(&card->rtd[i], - SNDRV_PCM_STREAM_PLAYBACK, codec_dai, + SNDRV_PCM_STREAM_PLAYBACK, SND_SOC_DAPM_STREAM_RESUME); snd_soc_dapm_stream_event(&card->rtd[i], - SNDRV_PCM_STREAM_CAPTURE, codec_dai, + SNDRV_PCM_STREAM_CAPTURE, SND_SOC_DAPM_STREAM_RESUME); } diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index e1863d7e8012..3fcefd1060a2 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2987,37 +2987,61 @@ int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card) return 0; } -static void soc_dapm_stream_event(struct snd_soc_dapm_context *dapm, - int stream, struct snd_soc_dai *dai, - int event) +static void soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, int stream, + int event) { - struct snd_soc_dapm_widget *w; - if (stream == SNDRV_PCM_STREAM_PLAYBACK) - w = dai->playback_widget; - else - w = dai->capture_widget; + struct snd_soc_dapm_widget *w_cpu, *w_codec; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; - if (!w) - return; + if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + w_cpu = cpu_dai->playback_widget; + w_codec = codec_dai->playback_widget; + } else { + w_cpu = cpu_dai->capture_widget; + w_codec = codec_dai->capture_widget; + } - dapm_mark_dirty(w, "stream event"); + if (w_cpu) { - switch (event) { - case SND_SOC_DAPM_STREAM_START: - w->active = 1; - break; - case SND_SOC_DAPM_STREAM_STOP: - w->active = 0; - break; - case SND_SOC_DAPM_STREAM_SUSPEND: - case SND_SOC_DAPM_STREAM_RESUME: - case SND_SOC_DAPM_STREAM_PAUSE_PUSH: - case SND_SOC_DAPM_STREAM_PAUSE_RELEASE: - break; + dapm_mark_dirty(w_cpu, "stream event"); + + switch (event) { + case SND_SOC_DAPM_STREAM_START: + w_cpu->active = 1; + break; + case SND_SOC_DAPM_STREAM_STOP: + w_cpu->active = 0; + break; + case SND_SOC_DAPM_STREAM_SUSPEND: + case SND_SOC_DAPM_STREAM_RESUME: + case SND_SOC_DAPM_STREAM_PAUSE_PUSH: + case SND_SOC_DAPM_STREAM_PAUSE_RELEASE: + break; + } + } + + if (w_codec) { + + dapm_mark_dirty(w_codec, "stream event"); + + switch (event) { + case SND_SOC_DAPM_STREAM_START: + w_codec->active = 1; + break; + case SND_SOC_DAPM_STREAM_STOP: + w_codec->active = 0; + break; + case SND_SOC_DAPM_STREAM_SUSPEND: + case SND_SOC_DAPM_STREAM_RESUME: + case SND_SOC_DAPM_STREAM_PAUSE_PUSH: + case SND_SOC_DAPM_STREAM_PAUSE_RELEASE: + break; + } } - dapm_power_widgets(dapm, event); + dapm_power_widgets(&rtd->card->dapm, event); } /** @@ -3031,15 +3055,14 @@ static void soc_dapm_stream_event(struct snd_soc_dapm_context *dapm, * * Returns 0 for success else error. */ -int snd_soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, int stream, - struct snd_soc_dai *dai, int event) +void snd_soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, int stream, + int event) { struct snd_soc_card *card = rtd->card; mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_PCM); - soc_dapm_stream_event(&card->dapm, stream, dai, event); + soc_dapm_stream_event(rtd, stream, event); mutex_unlock(&card->dapm_mutex); - return 0; } /** diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 0ad8dcacd2f3..26a60b4061cf 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -308,7 +308,7 @@ static void close_delayed_work(struct work_struct *work) if (codec_dai->pop_wait == 1) { codec_dai->pop_wait = 0; snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_PLAYBACK, - codec_dai, SND_SOC_DAPM_STREAM_STOP); + SND_SOC_DAPM_STREAM_STOP); } mutex_unlock(&rtd->pcm_mutex); @@ -373,7 +373,6 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) /* powered down playback stream now */ snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_PLAYBACK, - codec_dai, SND_SOC_DAPM_STREAM_STOP); } else { /* start delayed pop wq here for playback streams */ @@ -384,7 +383,7 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) } else { /* capture streams can be powered down now */ snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_CAPTURE, - codec_dai, SND_SOC_DAPM_STREAM_STOP); + SND_SOC_DAPM_STREAM_STOP); } mutex_unlock(&rtd->pcm_mutex); @@ -453,8 +452,8 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) cancel_delayed_work(&rtd->delayed_work); } - snd_soc_dapm_stream_event(rtd, substream->stream, codec_dai, - SND_SOC_DAPM_STREAM_START); + snd_soc_dapm_stream_event(rtd, substream->stream, + SND_SOC_DAPM_STREAM_START); snd_soc_dai_digital_mute(codec_dai, 0); -- cgit v1.2.3 From 3cd043436c2d5d6f8e9a5395d02ba966f0dfdf84 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Fri, 9 Mar 2012 12:02:08 +0000 Subject: ASoC: dapm: Rename dapm mutex subclass to better match usage Rename SND_SOC_DAPM_CLASS_PCM to SND_SOC_DAPM_CLASS_RUNTIME to better match the usage and align with card mutex too. Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown --- include/sound/soc-dapm.h | 4 ++-- sound/soc/soc-dapm.c | 20 ++++++++++---------- 2 files changed, 12 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 6c64dbeb28ba..64302384dea3 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -433,8 +433,8 @@ enum snd_soc_dapm_type { }; enum snd_soc_dapm_subclass { - SND_SOC_DAPM_CLASS_INIT = 0, - SND_SOC_DAPM_CLASS_PCM = 1, + SND_SOC_DAPM_CLASS_INIT = 0, + SND_SOC_DAPM_CLASS_RUNTIME = 1, }; /* diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 3fcefd1060a2..de001698825d 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1765,7 +1765,7 @@ int snd_soc_dapm_mux_update_power(struct snd_soc_dapm_widget *widget, struct snd_soc_card *card = widget->dapm->card; int ret; - mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_PCM); + mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); ret = soc_dapm_mux_update_power(widget, kcontrol, mux, e); mutex_unlock(&card->dapm_mutex); return ret; @@ -1809,7 +1809,7 @@ int snd_soc_dapm_mixer_update_power(struct snd_soc_dapm_widget *widget, struct snd_soc_card *card = widget->dapm->card; int ret; - mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_PCM); + mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); ret = soc_dapm_mixer_update_power(widget, kcontrol, connect); mutex_unlock(&card->dapm_mutex); return ret; @@ -1982,7 +1982,7 @@ int snd_soc_dapm_sync(struct snd_soc_dapm_context *dapm) if (!dapm->card || !dapm->card->instantiated) return 0; - mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_PCM); + mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); ret = dapm_power_widgets(dapm, SND_SOC_DAPM_STREAM_NOP); mutex_unlock(&dapm->card->dapm_mutex); return ret; @@ -2401,7 +2401,7 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, /* old connection must be powered down */ connect = invert ? 1 : 0; - mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_PCM); + mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); change = snd_soc_test_bits(widget->codec, reg, mask, val); if (change) { @@ -2493,7 +2493,7 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, mask |= (bitmask - 1) << e->shift_r; } - mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_PCM); + mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); change = snd_soc_test_bits(widget->codec, e->reg, mask, val); if (change) { @@ -2562,7 +2562,7 @@ int snd_soc_dapm_put_enum_virt(struct snd_kcontrol *kcontrol, if (ucontrol->value.enumerated.item[0] >= e->max) return -EINVAL; - mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_PCM); + mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); change = widget->value != ucontrol->value.enumerated.item[0]; if (change) { @@ -2659,7 +2659,7 @@ int snd_soc_dapm_put_value_enum_double(struct snd_kcontrol *kcontrol, mask |= e->mask << e->shift_r; } - mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_PCM); + mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); change = snd_soc_test_bits(widget->codec, e->reg, mask, val); if (change) { @@ -2718,7 +2718,7 @@ int snd_soc_dapm_get_pin_switch(struct snd_kcontrol *kcontrol, struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); const char *pin = (const char *)kcontrol->private_value; - mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_PCM); + mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); ucontrol->value.integer.value[0] = snd_soc_dapm_get_pin_status(&card->dapm, pin); @@ -2741,7 +2741,7 @@ int snd_soc_dapm_put_pin_switch(struct snd_kcontrol *kcontrol, struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); const char *pin = (const char *)kcontrol->private_value; - mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_PCM); + mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); if (ucontrol->value.integer.value[0]) snd_soc_dapm_enable_pin(&card->dapm, pin); @@ -3060,7 +3060,7 @@ void snd_soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, int stream, { struct snd_soc_card *card = rtd->card; - mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_PCM); + mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); soc_dapm_stream_event(rtd, stream, event); mutex_unlock(&card->dapm_mutex); } -- cgit v1.2.3 From a3cc056b64065efaf98d3e3fe8a6b9d508121492 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Fri, 9 Mar 2012 17:20:16 +0000 Subject: ASoC: dapm: Add regulator member to struct dapm_widget Currently DAPM widgets use the private data for their regulator. Add a regulator * for widgets to use instead of private data. Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown --- include/sound/soc-dapm.h | 2 ++ sound/soc/soc-dapm.c | 10 +++++----- 2 files changed, 7 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 64302384dea3..7562b8fb6974 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -324,6 +324,7 @@ struct snd_soc_dapm_path; struct snd_soc_dapm_pin; struct snd_soc_dapm_route; struct snd_soc_dapm_context; +struct regulator; int dapm_reg_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); @@ -487,6 +488,7 @@ struct snd_soc_dapm_widget { struct snd_soc_dapm_context *dapm; void *priv; /* widget specific data */ + struct regulator *regulator; /* attached regulator */ /* dapm control */ short reg; /* negative reg = no direct dapm */ diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index de001698825d..42602ddea243 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -861,9 +861,9 @@ int dapm_regulator_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { if (SND_SOC_DAPM_EVENT_ON(event)) - return regulator_enable(w->priv); + return regulator_enable(w->regulator); else - return regulator_disable_deferred(w->priv, w->shift); + return regulator_disable_deferred(w->regulator, w->shift); } EXPORT_SYMBOL_GPL(dapm_regulator_event); @@ -2768,9 +2768,9 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, switch (w->id) { case snd_soc_dapm_regulator_supply: - w->priv = devm_regulator_get(dapm->dev, w->name); - if (IS_ERR(w->priv)) { - ret = PTR_ERR(w->priv); + w->regulator = devm_regulator_get(dapm->dev, w->name); + if (IS_ERR(w->regulator)) { + ret = PTR_ERR(w->regulator); dev_err(dapm->dev, "Failed to request %s: %d\n", w->name, ret); return NULL; -- cgit v1.2.3 From 49575fb52bf76bf48e2d29ff034e8dad8d7ba638 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Tue, 6 Mar 2012 18:16:19 +0000 Subject: ASoC: DAPM: Make sure DAPM widget IO ops hold the component mutex Currently not all DAPM widget IO ops are holding their component mutex (codec or platform). Make sure this is now held for DAPM widget IO operations. Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 34 ++++++++++++++++++++++++++++------ 1 file changed, 28 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 42602ddea243..1e449f375d98 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -206,7 +206,23 @@ static int soc_widget_write(struct snd_soc_dapm_widget *w, int reg, int val) return -1; } -static int soc_widget_update_bits(struct snd_soc_dapm_widget *w, +static inline void soc_widget_lock(struct snd_soc_dapm_widget *w) +{ + if (w->codec) + mutex_lock(&w->codec->mutex); + else if (w->platform) + mutex_lock(&w->platform->mutex); +} + +static inline void soc_widget_unlock(struct snd_soc_dapm_widget *w) +{ + if (w->codec) + mutex_unlock(&w->codec->mutex); + else if (w->platform) + mutex_unlock(&w->platform->mutex); +} + +static int soc_widget_update_bits_locked(struct snd_soc_dapm_widget *w, unsigned short reg, unsigned int mask, unsigned int value) { bool change; @@ -219,18 +235,24 @@ static int soc_widget_update_bits(struct snd_soc_dapm_widget *w, if (ret != 0) return ret; } else { + soc_widget_lock(w); ret = soc_widget_read(w, reg); - if (ret < 0) + if (ret < 0) { + soc_widget_unlock(w); return ret; + } old = ret; new = (old & ~mask) | (value & mask); change = old != new; if (change) { ret = soc_widget_write(w, reg, new); - if (ret < 0) + if (ret < 0) { + soc_widget_unlock(w); return ret; + } } + soc_widget_unlock(w); } return change; @@ -847,7 +869,7 @@ int dapm_reg_event(struct snd_soc_dapm_widget *w, else val = w->off_val; - soc_widget_update_bits(w, -(w->reg + 1), + soc_widget_update_bits_locked(w, -(w->reg + 1), w->mask << w->shift, val << w->shift); return 0; @@ -1105,7 +1127,7 @@ static void dapm_seq_run_coalesced(struct snd_soc_dapm_context *dapm, "pop test : Applying 0x%x/0x%x to %x in %dms\n", value, mask, reg, card->pop_time); pop_wait(card->pop_time); - soc_widget_update_bits(w, reg, mask, value); + soc_widget_update_bits_locked(w, reg, mask, value); } list_for_each_entry(w, pending, power_list) { @@ -1235,7 +1257,7 @@ static void dapm_widget_update(struct snd_soc_dapm_context *dapm) w->name, ret); } - ret = snd_soc_update_bits(w->codec, update->reg, update->mask, + ret = soc_widget_update_bits_locked(w, update->reg, update->mask, update->val); if (ret < 0) pr_err("%s DAPM update failed: %d\n", w->name, ret); -- cgit v1.2.3 From e06ab3b8e82c89a8ada25f61fbd9f02d6ca3103f Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 6 Mar 2012 23:58:22 +0000 Subject: ASoC: dapm: Only lock CODEC for I/O if not using regmap If we do use regmap then regmap will take care of things for us. We actually already have this check at a higher level for the current users but this makes sure we do the right thing in the future too if we need to. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/soc-dapm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 1e449f375d98..dd603fa38097 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -208,7 +208,7 @@ static int soc_widget_write(struct snd_soc_dapm_widget *w, int reg, int val) static inline void soc_widget_lock(struct snd_soc_dapm_widget *w) { - if (w->codec) + if (w->codec && !w->codec->using_regmap) mutex_lock(&w->codec->mutex); else if (w->platform) mutex_lock(&w->platform->mutex); @@ -216,7 +216,7 @@ static inline void soc_widget_lock(struct snd_soc_dapm_widget *w) static inline void soc_widget_unlock(struct snd_soc_dapm_widget *w) { - if (w->codec) + if (w->codec && !w->codec->using_regmap) mutex_unlock(&w->codec->mutex); else if (w->platform) mutex_unlock(&w->platform->mutex); -- cgit v1.2.3 From 77caabaa74524dce5f83b93b8f0a6d0c1c5e860f Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 30 Jan 2012 20:01:53 +0000 Subject: ASoC: wm5100: Convert to devm_regmap_init_i2c() Signed-off-by: Mark Brown --- sound/soc/codecs/wm5100.c | 9 +++------ 1 file changed, 3 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index b9c185ce64e4..0594636b9107 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -2454,7 +2454,7 @@ static __devinit int wm5100_i2c_probe(struct i2c_client *i2c, wm5100->dev = &i2c->dev; - wm5100->regmap = regmap_init_i2c(i2c, &wm5100_regmap); + wm5100->regmap = devm_regmap_init_i2c(i2c, &wm5100_regmap); if (IS_ERR(wm5100->regmap)) { ret = PTR_ERR(wm5100->regmap); dev_err(&i2c->dev, "Failed to allocate register map: %d\n", @@ -2479,7 +2479,7 @@ static __devinit int wm5100_i2c_probe(struct i2c_client *i2c, if (ret != 0) { dev_err(&i2c->dev, "Failed to request core supplies: %d\n", ret); - goto err_regmap; + goto err; } ret = regulator_bulk_enable(ARRAY_SIZE(wm5100->core_supplies), @@ -2487,7 +2487,7 @@ static __devinit int wm5100_i2c_probe(struct i2c_client *i2c, if (ret != 0) { dev_err(&i2c->dev, "Failed to enable core supplies: %d\n", ret); - goto err_regmap; + goto err; } if (wm5100->pdata.ldo_ena) { @@ -2660,8 +2660,6 @@ err_ldo: err_enable: regulator_bulk_disable(ARRAY_SIZE(wm5100->core_supplies), wm5100->core_supplies); -err_regmap: - regmap_exit(wm5100->regmap); err: return ret; } @@ -2682,7 +2680,6 @@ static __devexit int wm5100_i2c_remove(struct i2c_client *i2c) gpio_set_value_cansleep(wm5100->pdata.ldo_ena, 0); gpio_free(wm5100->pdata.ldo_ena); } - regmap_exit(wm5100->regmap); return 0; } -- cgit v1.2.3 From ecd1732f0118f3bc47429ceffa01593ec16c364d Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 12 Mar 2012 16:34:35 +0000 Subject: ASoC: wm8994: Don't lock CODEC mutex to do DAPM sync DAPM now has a DAPM-level lock which it manages itself so we don't need to take the CODEC mutex to call DAPM any more. Also remove a redundant call to snd_soc_dapm_sync(), jack reporting also triggers a DAPM sync. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 17 ++++------------- 1 file changed, 4 insertions(+), 13 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 7c49642af052..fbcaf499b0ff 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3247,9 +3247,6 @@ static void wm8958_default_micdet(u16 status, void *data) wm8958_micd_set_rate(codec); - snd_soc_jack_report(wm8994->micdet[0].jack, SND_JACK_HEADPHONE, - SND_JACK_HEADSET); - /* If we have jackdet that will detect removal */ if (wm8994->jackdet) { mutex_lock(&wm8994->accdet_lock); @@ -3262,14 +3259,13 @@ static void wm8958_default_micdet(u16 status, void *data) mutex_unlock(&wm8994->accdet_lock); - if (wm8994->pdata->jd_ext_cap) { - mutex_lock(&codec->mutex); + if (wm8994->pdata->jd_ext_cap) snd_soc_dapm_disable_pin(&codec->dapm, "MICBIAS2"); - snd_soc_dapm_sync(&codec->dapm); - mutex_unlock(&codec->mutex); - } } + + snd_soc_jack_report(wm8994->micdet[0].jack, SND_JACK_HEADPHONE, + SND_JACK_HEADSET); } /* Report short circuit as a button */ @@ -3358,16 +3354,11 @@ static irqreturn_t wm1811_jackdet_irq(int irq, void *data) /* If required for an external cap force MICBIAS on */ if (wm8994->pdata->jd_ext_cap) { - mutex_lock(&codec->mutex); - if (present) snd_soc_dapm_force_enable_pin(&codec->dapm, "MICBIAS2"); else snd_soc_dapm_disable_pin(&codec->dapm, "MICBIAS2"); - - snd_soc_dapm_sync(&codec->dapm); - mutex_unlock(&codec->mutex); } if (present) -- cgit v1.2.3 From 2667b4b8bef8598917adb1b4af46ed2b7d4fa0d7 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 12 Mar 2012 14:07:49 +0000 Subject: ASoC: jack: Push locking for jacks down to the jack Currently operations on jack reporting take the CODEC mutex both to protect the current jack status and also to protect the DAPM run which is triggered on status updates. Since the addition of a DAPM-specific lock we no longer need to worry about locking DAPM as it has its own finer grained lock so create a per jack lock to take care of the jack status. This is both cleaner where the jack isn't specifically associated with a CODEC and clearer as it's much more obvious what the lock is protecting. Signed-off-by: Mark Brown --- include/sound/soc.h | 1 + sound/soc/soc-jack.c | 5 +++-- 2 files changed, 4 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/include/sound/soc.h b/include/sound/soc.h index 098998743969..b8163ddf94d2 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -518,6 +518,7 @@ struct snd_soc_jack_gpio { #endif struct snd_soc_jack { + struct mutex mutex; struct snd_jack *jack; struct snd_soc_codec *codec; struct list_head pins; diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index ee4353f843ea..7f8b3b7428bb 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -36,6 +36,7 @@ int snd_soc_jack_new(struct snd_soc_codec *codec, const char *id, int type, struct snd_soc_jack *jack) { + mutex_init(&jack->mutex); jack->codec = codec; INIT_LIST_HEAD(&jack->pins); INIT_LIST_HEAD(&jack->jack_zones); @@ -75,7 +76,7 @@ void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask) codec = jack->codec; dapm = &codec->dapm; - mutex_lock(&codec->mutex); + mutex_lock(&jack->mutex); oldstatus = jack->status; @@ -109,7 +110,7 @@ void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask) snd_jack_report(jack->jack, jack->status); out: - mutex_unlock(&codec->mutex); + mutex_unlock(&jack->mutex); } EXPORT_SYMBOL_GPL(snd_soc_jack_report); -- cgit v1.2.3 From b19e6e7b763c7144bfe2ceccf988b64d66d6dd0a Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 14 Mar 2012 21:18:39 +0000 Subject: ASoC: core: Use driver core probe deferral In version 3.4 the driver core acquired probe deferral which is a core way of doing essentially the same thing as ASoC has been doing since forever to make sure that all the devices needed to make up the card are present without needing open coding in the subsystem. Make basic use of this probe deferral mechanism for the cards, removing the need to handle partially instantiated cards. We should be able to remove even more code than this, though some of the checks we're currently doing should stay since they're about things like suppressing unneeded DAPM runs rather than deferring probes. In order to avoid robustness issues with our teardown paths (which do need quite a bit of TLC) add a check for aux_devs prior to attempting to set things up, this means that we've got a reasonable idea that everything will be there before we start. As with the removal of partial instantiation support more work will be needed to make this work neatly. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- include/sound/soc.h | 1 - sound/soc/soc-core.c | 147 ++++++++++++++++++++++----------------------------- 2 files changed, 62 insertions(+), 86 deletions(-) (limited to 'sound') diff --git a/include/sound/soc.h b/include/sound/soc.h index b8163ddf94d2..9e238fa2eb17 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -896,7 +896,6 @@ struct snd_soc_pcm_runtime { enum snd_soc_pcm_subclass pcm_subclass; struct snd_pcm_ops ops; - unsigned int complete:1; unsigned int dev_registered:1; long pmdown_time; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 61b51b673d49..cab72f87c194 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -54,7 +54,6 @@ EXPORT_SYMBOL_GPL(snd_soc_debugfs_root); #endif static DEFINE_MUTEX(client_mutex); -static LIST_HEAD(card_list); static LIST_HEAD(dai_list); static LIST_HEAD(platform_list); static LIST_HEAD(codec_list); @@ -785,15 +784,9 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num) struct snd_soc_dai *codec_dai, *cpu_dai; const char *platform_name; - if (rtd->complete) - return 1; dev_dbg(card->dev, "binding %s at idx %d\n", dai_link->name, num); - /* do we already have the CPU DAI for this link ? */ - if (rtd->cpu_dai) { - goto find_codec; - } - /* no, then find CPU DAI from registered DAIs*/ + /* Find CPU DAI from registered DAIs*/ list_for_each_entry(cpu_dai, &dai_list, list) { if (dai_link->cpu_dai_of_node) { if (cpu_dai->dev->of_node != dai_link->cpu_dai_of_node) @@ -804,18 +797,15 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num) } rtd->cpu_dai = cpu_dai; - goto find_codec; } - dev_dbg(card->dev, "CPU DAI %s not registered\n", - dai_link->cpu_dai_name); -find_codec: - /* do we already have the CODEC for this link ? */ - if (rtd->codec) { - goto find_platform; + if (!rtd->cpu_dai) { + dev_dbg(card->dev, "CPU DAI %s not registered\n", + dai_link->cpu_dai_name); + return -EPROBE_DEFER; } - /* no, then find CODEC from registered CODECs*/ + /* Find CODEC from registered CODECs */ list_for_each_entry(codec, &codec_list, list) { if (dai_link->codec_of_node) { if (codec->dev->of_node != dai_link->codec_of_node) @@ -837,28 +827,28 @@ find_codec: dai_link->codec_dai_name)) { rtd->codec_dai = codec_dai; - goto find_platform; } } - dev_dbg(card->dev, "CODEC DAI %s not registered\n", - dai_link->codec_dai_name); - goto find_platform; + if (!rtd->codec_dai) { + dev_dbg(card->dev, "CODEC DAI %s not registered\n", + dai_link->codec_dai_name); + return -EPROBE_DEFER; + } } - dev_dbg(card->dev, "CODEC %s not registered\n", - dai_link->codec_name); -find_platform: - /* do we need a platform? */ - if (rtd->platform) - goto out; + if (!rtd->codec) { + dev_dbg(card->dev, "CODEC %s not registered\n", + dai_link->codec_name); + return -EPROBE_DEFER; + } /* if there's no platform we match on the empty platform */ platform_name = dai_link->platform_name; if (!platform_name && !dai_link->platform_of_node) platform_name = "snd-soc-dummy"; - /* no, then find one from the set of registered platforms */ + /* find one from the set of registered platforms */ list_for_each_entry(platform, &platform_list, list) { if (dai_link->platform_of_node) { if (platform->dev->of_node != @@ -870,20 +860,16 @@ find_platform: } rtd->platform = platform; - goto out; } - - dev_dbg(card->dev, "platform %s not registered\n", + if (!rtd->platform) { + dev_dbg(card->dev, "platform %s not registered\n", dai_link->platform_name); - return 0; - -out: - /* mark rtd as complete if we found all 4 of our client devices */ - if (rtd->codec && rtd->codec_dai && rtd->platform && rtd->cpu_dai) { - rtd->complete = 1; - card->num_rtd++; + return -EPROBE_DEFER; } - return 1; + + card->num_rtd++; + + return 0; } static void soc_remove_codec(struct snd_soc_codec *codec) @@ -1346,6 +1332,20 @@ static void soc_unregister_ac97_dai_link(struct snd_soc_codec *codec) } #endif +static int soc_check_aux_dev(struct snd_soc_card *card, int num) +{ + struct snd_soc_aux_dev *aux_dev = &card->aux_dev[num]; + struct snd_soc_codec *codec; + + /* find CODEC from registered CODECs*/ + list_for_each_entry(codec, &codec_list, list) { + if (!strcmp(codec->name, aux_dev->codec_name)) + return 0; + } + + return -EPROBE_DEFER; +} + static int soc_probe_aux_dev(struct snd_soc_card *card, int num) { struct snd_soc_aux_dev *aux_dev = &card->aux_dev[num]; @@ -1366,7 +1366,7 @@ static int soc_probe_aux_dev(struct snd_soc_card *card, int num) } /* codec not found */ dev_err(card->dev, "asoc: codec %s not found", aux_dev->codec_name); - goto out; + return -EPROBE_DEFER; found: ret = soc_probe_codec(card, codec); @@ -1416,7 +1416,7 @@ static int snd_soc_init_codec_cache(struct snd_soc_codec *codec, return 0; } -static void snd_soc_instantiate_card(struct snd_soc_card *card) +static int snd_soc_instantiate_card(struct snd_soc_card *card) { struct snd_soc_codec *codec; struct snd_soc_codec_conf *codec_conf; @@ -1426,19 +1426,18 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) mutex_lock_nested(&card->mutex, SND_SOC_CARD_CLASS_INIT); - if (card->instantiated) { - mutex_unlock(&card->mutex); - return; - } - /* bind DAIs */ - for (i = 0; i < card->num_links; i++) - soc_bind_dai_link(card, i); + for (i = 0; i < card->num_links; i++) { + ret = soc_bind_dai_link(card, i); + if (ret != 0) + goto base_error; + } - /* bind completed ? */ - if (card->num_rtd != card->num_links) { - mutex_unlock(&card->mutex); - return; + /* check aux_devs too */ + for (i = 0; i < card->num_aux_devs; i++) { + ret = soc_check_aux_dev(card, i); + if (ret != 0) + goto base_error; } /* initialize the register cache for each available codec */ @@ -1458,10 +1457,8 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) } } ret = snd_soc_init_codec_cache(codec, compress_type); - if (ret < 0) { - mutex_unlock(&card->mutex); - return; - } + if (ret < 0) + goto base_error; } /* card bind complete so register a sound card */ @@ -1470,8 +1467,7 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) if (ret < 0) { pr_err("asoc: can't create sound card for card %s: %d\n", card->name, ret); - mutex_unlock(&card->mutex); - return; + goto base_error; } card->snd_card->dev = card->dev; @@ -1611,7 +1607,8 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) card->instantiated = 1; snd_soc_dapm_sync(&card->dapm); mutex_unlock(&card->mutex); - return; + + return 0; probe_aux_dev_err: for (i = 0; i < card->num_aux_devs; i++) @@ -1626,18 +1623,10 @@ card_probe_error: snd_card_free(card->snd_card); +base_error: mutex_unlock(&card->mutex); -} -/* - * Attempt to initialise any uninitialised cards. Must be called with - * client_mutex. - */ -static void snd_soc_instantiate_cards(void) -{ - struct snd_soc_card *card; - list_for_each_entry(card, &card_list, list) - snd_soc_instantiate_card(card); + return ret; } /* probes a new socdev */ @@ -3072,7 +3061,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_digital_mute); */ int snd_soc_register_card(struct snd_soc_card *card) { - int i; + int i, ret; if (!card->name || !card->dev) return -EINVAL; @@ -3136,14 +3125,11 @@ int snd_soc_register_card(struct snd_soc_card *card) mutex_init(&card->mutex); mutex_init(&card->dapm_mutex); - mutex_lock(&client_mutex); - list_add(&card->list, &card_list); - snd_soc_instantiate_cards(); - mutex_unlock(&client_mutex); + ret = snd_soc_instantiate_card(card); + if (ret != 0) + soc_cleanup_card_debugfs(card); - dev_dbg(card->dev, "Registered card '%s'\n", card->name); - - return 0; + return ret; } EXPORT_SYMBOL_GPL(snd_soc_register_card); @@ -3157,9 +3143,6 @@ int snd_soc_unregister_card(struct snd_soc_card *card) { if (card->instantiated) soc_cleanup_card_resources(card); - mutex_lock(&client_mutex); - list_del(&card->list); - mutex_unlock(&client_mutex); dev_dbg(card->dev, "Unregistered card '%s'\n", card->name); return 0; @@ -3256,7 +3239,6 @@ int snd_soc_register_dai(struct device *dev, mutex_lock(&client_mutex); list_add(&dai->list, &dai_list); - snd_soc_instantiate_cards(); mutex_unlock(&client_mutex); pr_debug("Registered DAI '%s'\n", dai->name); @@ -3338,9 +3320,6 @@ int snd_soc_register_dais(struct device *dev, pr_debug("Registered DAI '%s'\n", dai->name); } - mutex_lock(&client_mutex); - snd_soc_instantiate_cards(); - mutex_unlock(&client_mutex); return 0; err: @@ -3398,7 +3377,6 @@ int snd_soc_register_platform(struct device *dev, mutex_lock(&client_mutex); list_add(&platform->list, &platform_list); - snd_soc_instantiate_cards(); mutex_unlock(&client_mutex); pr_debug("Registered platform '%s'\n", platform->name); @@ -3557,7 +3535,6 @@ int snd_soc_register_codec(struct device *dev, mutex_lock(&client_mutex); list_add(&codec->list, &codec_list); - snd_soc_instantiate_cards(); mutex_unlock(&client_mutex); pr_debug("Registered codec '%s'\n", codec->name); -- cgit v1.2.3 From 5f1cba63a3a65b01a70ac09914176bb3719725d6 Mon Sep 17 00:00:00 2001 From: Philipp Zabel Date: Thu, 15 Mar 2012 19:16:12 +0100 Subject: ASoC: pxa2xx-i2s: Add clk_prepare/clk_unprepare calls This patch adds clk_prepare/clk_unprepare calls to the pxa2xx-i2s driver by using the helper functions clk_prepare_enable and clk_disable_unprepare. Signed-off-by: Philipp Zabel Signed-off-by: Mark Brown --- sound/soc/pxa/pxa2xx-i2s.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index 609abd51e55f..453b0925a0b5 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -165,7 +165,7 @@ static int pxa2xx_i2s_hw_params(struct snd_pcm_substream *substream, struct pxa2xx_pcm_dma_params *dma_data; BUG_ON(IS_ERR(clk_i2s)); - clk_enable(clk_i2s); + clk_prepare_enable(clk_i2s); clk_ena = 1; pxa_i2s_wait(); @@ -258,7 +258,7 @@ static void pxa2xx_i2s_shutdown(struct snd_pcm_substream *substream, SACR0 &= ~SACR0_ENB; pxa_i2s_wait(); if (clk_ena) { - clk_disable(clk_i2s); + clk_disable_unprepare(clk_i2s); clk_ena = 0; } } -- cgit v1.2.3 From 9dd90c5db0401061009183e6407feff3724ebc8b Mon Sep 17 00:00:00 2001 From: Rhyland Klein Date: Thu, 15 Mar 2012 15:07:47 -0700 Subject: ASoC: max98095: add jack detection This change adds the logic to support using the jack detect mechanism built in to the codec to detect both when a jack was inserted and what type of jack is present. This change also supports the use of an external mechanism for headphone detection. If this mechanism exists, when the max98095_jack_detect function is called, the hp_jack is simply passed NULL. This change supports both simple headphones, powered headphones, microphones and headsets with both headphones and a mic. Signed-off-by: Rhyland Klein Signed-off-by: Mark Brown --- include/sound/max98095.h | 12 ++++ sound/soc/codecs/max98095.c | 158 +++++++++++++++++++++++++++++++++++++++++++- sound/soc/codecs/max98095.h | 22 ++++++ 3 files changed, 191 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/include/sound/max98095.h b/include/sound/max98095.h index 7513a42dd4aa..65b1c90f78ab 100644 --- a/include/sound/max98095.h +++ b/include/sound/max98095.h @@ -49,6 +49,18 @@ struct max98095_pdata { */ unsigned int digmic_left_mode:1; unsigned int digmic_right_mode:1; + + /* Pin5 is the mechanical method of sensing jack insertion + * but it is something that might not be supported. + * 0 = PIN5 not supported + * 1 = PIN5 supported + */ + int jack_detect_pin5en:1; + + /* Slew amount for jack detection. Calculated as 4 * (delay + 1). + * Default delay is 24 to get a time of 100ms. + */ + unsigned int jack_detect_delay; }; #endif diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c index 0bb511a0388d..0752840e01c2 100644 --- a/sound/soc/codecs/max98095.c +++ b/sound/soc/codecs/max98095.c @@ -24,6 +24,7 @@ #include #include #include +#include #include "max98095.h" enum max98095_type { @@ -51,6 +52,8 @@ struct max98095_priv { u8 lin_state; unsigned int mic1pre; unsigned int mic2pre; + struct snd_soc_jack *headphone_jack; + struct snd_soc_jack *mic_jack; }; static const u8 max98095_reg_def[M98095_REG_CNT] = { @@ -2173,9 +2176,125 @@ static void max98095_handle_pdata(struct snd_soc_codec *codec) max98095_handle_bq_pdata(codec); } +static irqreturn_t max98095_report_jack(int irq, void *data) +{ + struct snd_soc_codec *codec = data; + struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec); + unsigned int value; + int hp_report = 0; + int mic_report = 0; + + /* Read the Jack Status Register */ + value = snd_soc_read(codec, M98095_007_JACK_AUTO_STS); + + /* If ddone is not set, then detection isn't finished yet */ + if ((value & M98095_DDONE) == 0) + return IRQ_NONE; + + /* if hp, check its bit, and if set, clear it */ + if ((value & M98095_HP_IN || value & M98095_LO_IN) && + max98095->headphone_jack) + hp_report |= SND_JACK_HEADPHONE; + + /* if mic, check its bit, and if set, clear it */ + if ((value & M98095_MIC_IN) && max98095->mic_jack) + mic_report |= SND_JACK_MICROPHONE; + + if (max98095->headphone_jack == max98095->mic_jack) { + snd_soc_jack_report(max98095->headphone_jack, + hp_report | mic_report, + SND_JACK_HEADSET); + } else { + if (max98095->headphone_jack) + snd_soc_jack_report(max98095->headphone_jack, + hp_report, SND_JACK_HEADPHONE); + if (max98095->mic_jack) + snd_soc_jack_report(max98095->mic_jack, + mic_report, SND_JACK_MICROPHONE); + } + + return IRQ_HANDLED; +} + +int max98095_jack_detect_enable(struct snd_soc_codec *codec) +{ + struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec); + int ret = 0; + int detect_enable = M98095_JDEN; + unsigned int slew = M98095_DEFAULT_SLEW_DELAY; + + if (max98095->pdata->jack_detect_pin5en) + detect_enable |= M98095_PIN5EN; + + if (max98095->jack_detect_delay) + slew = max98095->jack_detect_delay; + + ret = snd_soc_write(codec, M98095_08E_JACK_DC_SLEW, slew); + if (ret < 0) { + dev_err(codec->dev, "Failed to cfg auto detect %d\n", ret); + return ret; + } + + /* configure auto detection to be enabled */ + ret = snd_soc_write(codec, M98095_089_JACK_DET_AUTO, detect_enable); + if (ret < 0) { + dev_err(codec->dev, "Failed to cfg auto detect %d\n", ret); + return ret; + } + + return ret; +} + +int max98095_jack_detect_disable(struct snd_soc_codec *codec) +{ + int ret = 0; + + /* configure auto detection to be disabled */ + ret = snd_soc_write(codec, M98095_089_JACK_DET_AUTO, 0x0); + if (ret < 0) { + dev_err(codec->dev, "Failed to cfg auto detect %d\n", ret); + return ret; + } + + return ret; +} + +int max98095_jack_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *hp_jack, struct snd_soc_jack *mic_jack) +{ + struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec); + struct i2c_client *client = to_i2c_client(codec->dev); + int ret = 0; + + max98095->headphone_jack = hp_jack; + max98095->mic_jack = mic_jack; + + /* only progress if we have at least 1 jack pointer */ + if (!hp_jack && !mic_jack) + return -EINVAL; + + max98095_jack_detect_enable(codec); + + /* enable interrupts for headphone jack detection */ + ret = snd_soc_update_bits(codec, M98095_013_JACK_INT_EN, + M98095_IDDONE, M98095_IDDONE); + if (ret < 0) { + dev_err(codec->dev, "Failed to cfg jack irqs %d\n", ret); + return ret; + } + + max98095_report_jack(client->irq, codec); + return 0; +} + #ifdef CONFIG_PM static int max98095_suspend(struct snd_soc_codec *codec) { + struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec); + + if (max98095->headphone_jack || max98095->mic_jack) + max98095_jack_detect_disable(codec); + max98095_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; @@ -2183,8 +2302,16 @@ static int max98095_suspend(struct snd_soc_codec *codec) static int max98095_resume(struct snd_soc_codec *codec) { + struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec); + struct i2c_client *client = to_i2c_client(codec->dev); + max98095_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + if (max98095->headphone_jack || max98095->mic_jack) { + max98095_jack_detect_enable(codec); + max98095_report_jack(client->irq, codec); + } + return 0; } #else @@ -2227,6 +2354,7 @@ static int max98095_probe(struct snd_soc_codec *codec) { struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec); struct max98095_cdata *cdata; + struct i2c_client *client; int ret = 0; ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C); @@ -2238,6 +2366,8 @@ static int max98095_probe(struct snd_soc_codec *codec) /* reset the codec, the DSP core, and disable all interrupts */ max98095_reset(codec); + client = to_i2c_client(codec->dev); + /* initialize private data */ max98095->sysclk = (unsigned)-1; @@ -2266,11 +2396,23 @@ static int max98095_probe(struct snd_soc_codec *codec) max98095->mic1pre = 0; max98095->mic2pre = 0; + if (client->irq) { + /* register an audio interrupt */ + ret = request_threaded_irq(client->irq, NULL, + max98095_report_jack, + IRQF_TRIGGER_FALLING | IRQF_TRIGGER_RISING, + "max98095", codec); + if (ret) { + dev_err(codec->dev, "Failed to request IRQ: %d\n", ret); + goto err_access; + } + } + ret = snd_soc_read(codec, M98095_0FF_REV_ID); if (ret < 0) { dev_err(codec->dev, "Failure reading hardware revision: %d\n", ret); - goto err_access; + goto err_irq; } dev_info(codec->dev, "Hardware revision: %c\n", ret - 0x40 + 'A'); @@ -2306,14 +2448,28 @@ static int max98095_probe(struct snd_soc_codec *codec) max98095_add_widgets(codec); + return 0; + +err_irq: + if (client->irq) + free_irq(client->irq, codec); err_access: return ret; } static int max98095_remove(struct snd_soc_codec *codec) { + struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec); + struct i2c_client *client = to_i2c_client(codec->dev); + max98095_set_bias_level(codec, SND_SOC_BIAS_OFF); + if (max98095->headphone_jack || max98095->mic_jack) + max98095_jack_detect_disable(codec); + + if (client->irq) + free_irq(client->irq, codec); + return 0; } diff --git a/sound/soc/codecs/max98095.h b/sound/soc/codecs/max98095.h index 891584a0eb03..2ebbe4e894bf 100644 --- a/sound/soc/codecs/max98095.h +++ b/sound/soc/codecs/max98095.h @@ -175,11 +175,23 @@ /* MAX98095 Registers Bit Fields */ +/* M98095_007_JACK_AUTO_STS */ + #define M98095_MIC_IN (1<<3) + #define M98095_LO_IN (1<<5) + #define M98095_HP_IN (1<<6) + #define M98095_DDONE (1<<7) + /* M98095_00F_HOST_CFG */ #define M98095_SEG (1<<0) #define M98095_XTEN (1<<1) #define M98095_MDLLEN (1<<2) +/* M98095_013_JACK_INT_EN */ + #define M98095_IMIC_IN (1<<3) + #define M98095_ILO_IN (1<<5) + #define M98095_IHP_IN (1<<6) + #define M98095_IDDONE (1<<7) + /* M98095_027_DAI1_CLKMODE, M98095_031_DAI2_CLKMODE, M98095_03B_DAI3_CLKMODE */ #define M98095_CLKMODE_MASK 0xFF @@ -255,6 +267,10 @@ #define M98095_EQ2EN (1<<1) #define M98095_EQ1EN (1<<0) +/* M98095_089_JACK_DET_AUTO */ + #define M98095_PIN5EN (1<<2) + #define M98095_JDEN (1<<7) + /* M98095_090_PWR_EN_IN */ #define M98095_INEN (1<<7) #define M98095_MB2EN (1<<3) @@ -296,4 +312,10 @@ #define M98095_174_DAI1_BQ_BASE 0x74 #define M98095_17E_DAI2_BQ_BASE 0x7E +/* Default Delay used in Slew Rate Calculation for Jack detection */ +#define M98095_DEFAULT_SLEW_DELAY 0x18 + +extern int max98095_jack_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *hp_jack, struct snd_soc_jack *mic_jack); + #endif -- cgit v1.2.3 From 26b427a701f81d4092869682c386a3e317983c9d Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 23 Feb 2012 20:19:47 +0000 Subject: ASoC: wm8962: Implement DSP2 configuration initialisation We can simply use the register cache code to synchronise the current configuration down to the device when bringing up the DSP. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8962.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 15d467ff91b4..640001cf479b 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -1478,7 +1478,8 @@ static const DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0); static int wm8962_dsp2_write_config(struct snd_soc_codec *codec) { - return 0; + return regcache_sync_region(codec->control_data, + WM8962_HDBASS_AI_1, WM8962_MAX_REGISTER); } static int wm8962_dsp2_set_enable(struct snd_soc_codec *codec, u16 val) -- cgit v1.2.3 From 69e5a39f39c371abc288f89be0fd1edd29be851a Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 21 Feb 2012 23:21:17 +0000 Subject: ASoC: wm8962: Add 3D enhancement support Signed-off-by: Mark Brown --- sound/soc/codecs/wm8962.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 640001cf479b..7579df1e6e20 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -1756,6 +1756,9 @@ SOC_DOUBLE_R_TLV("EQ4 Volume", WM8962_EQ3, WM8962_EQ23, SOC_DOUBLE_R_TLV("EQ5 Volume", WM8962_EQ3, WM8962_EQ23, WM8962_EQL_B5_GAIN_SHIFT, 31, 0, eq_tlv), +SOC_SINGLE("3D Switch", WM8962_THREED1, 0, 1, 0), +SND_SOC_BYTES_MASK("3D Coefficients", WM8962_THREED1, 4, WM8962_THREED_ENA), + WM8962_DSP2_ENABLE("VSS Switch", WM8962_VSS_ENA_SHIFT), WM8962_DSP2_ENABLE("HPF1 Switch", WM8962_HPF1_ENA_SHIFT), WM8962_DSP2_ENABLE("HPF2 Switch", WM8962_HPF2_ENA_SHIFT), -- cgit v1.2.3 From acf31d43928161fb11743dfa43921d1c6fb6d024 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 21 Feb 2012 23:24:46 +0000 Subject: ASoC: wm8962: Add Direct-Form 1 filter support Signed-off-by: Mark Brown --- sound/soc/codecs/wm8962.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 7579df1e6e20..bee6378e9e01 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -1759,6 +1759,9 @@ SOC_DOUBLE_R_TLV("EQ5 Volume", WM8962_EQ3, WM8962_EQ23, SOC_SINGLE("3D Switch", WM8962_THREED1, 0, 1, 0), SND_SOC_BYTES_MASK("3D Coefficients", WM8962_THREED1, 4, WM8962_THREED_ENA), +SOC_SINGLE("DF1 Switch", WM8962_DF1, 0, 1, 0), +SND_SOC_BYTES_MASK("DF1 Coefficients", WM8962_DF1, 7, WM8962_DF1_ENA), + WM8962_DSP2_ENABLE("VSS Switch", WM8962_VSS_ENA_SHIFT), WM8962_DSP2_ENABLE("HPF1 Switch", WM8962_HPF1_ENA_SHIFT), WM8962_DSP2_ENABLE("HPF2 Switch", WM8962_HPF2_ENA_SHIFT), -- cgit v1.2.3 From fd0ca45bef3c05a36cf7d9d0b0ca7eda66daf932 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 21 Feb 2012 23:25:05 +0000 Subject: ASoC: wm8962: Add Dynamic Range Control support Signed-off-by: Mark Brown --- sound/soc/codecs/wm8962.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index bee6378e9e01..cb2856a08afa 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -1762,6 +1762,9 @@ SND_SOC_BYTES_MASK("3D Coefficients", WM8962_THREED1, 4, WM8962_THREED_ENA), SOC_SINGLE("DF1 Switch", WM8962_DF1, 0, 1, 0), SND_SOC_BYTES_MASK("DF1 Coefficients", WM8962_DF1, 7, WM8962_DF1_ENA), +SOC_SINGLE("DRC Switch", WM8962_DRC_1, 0, 1, 0), +SND_SOC_BYTES_MASK("DRC Coefficients", WM8962_DRC_1, 5, WM8962_DRC_ENA), + WM8962_DSP2_ENABLE("VSS Switch", WM8962_VSS_ENA_SHIFT), WM8962_DSP2_ENABLE("HPF1 Switch", WM8962_HPF1_ENA_SHIFT), WM8962_DSP2_ENABLE("HPF2 Switch", WM8962_HPF2_ENA_SHIFT), -- cgit v1.2.3 From 5462fccde54735e3f101f20ff1c42cb8cbf161e6 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 21 Feb 2012 23:33:26 +0000 Subject: ASoC: wm8962: Add HD Bass and VSS coefficient configuration Signed-off-by: Mark Brown --- sound/soc/codecs/wm8962.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index cb2856a08afa..69006270f960 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -1766,9 +1766,11 @@ SOC_SINGLE("DRC Switch", WM8962_DRC_1, 0, 1, 0), SND_SOC_BYTES_MASK("DRC Coefficients", WM8962_DRC_1, 5, WM8962_DRC_ENA), WM8962_DSP2_ENABLE("VSS Switch", WM8962_VSS_ENA_SHIFT), +SND_SOC_BYTES("VSS Coefficients", WM8962_VSS_XHD2_1, 148), WM8962_DSP2_ENABLE("HPF1 Switch", WM8962_HPF1_ENA_SHIFT), WM8962_DSP2_ENABLE("HPF2 Switch", WM8962_HPF2_ENA_SHIFT), WM8962_DSP2_ENABLE("HD Bass Switch", WM8962_HDBASS_ENA_SHIFT), +SND_SOC_BYTES("HD Bass Coefficients", WM8962_HDBASS_AI_1, 30), }; static const struct snd_kcontrol_new wm8962_spk_mono_controls[] = { -- cgit v1.2.3 From 93a86bea26637c2dd4db6d736e010789d339a96d Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 6 Mar 2012 00:29:37 +0000 Subject: ASoC: wm8962: Add HPF coefficient configuration support Signed-off-by: Mark Brown --- sound/soc/codecs/wm8962.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 69006270f960..507f479f4387 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -1769,6 +1769,7 @@ WM8962_DSP2_ENABLE("VSS Switch", WM8962_VSS_ENA_SHIFT), SND_SOC_BYTES("VSS Coefficients", WM8962_VSS_XHD2_1, 148), WM8962_DSP2_ENABLE("HPF1 Switch", WM8962_HPF1_ENA_SHIFT), WM8962_DSP2_ENABLE("HPF2 Switch", WM8962_HPF2_ENA_SHIFT), +SND_SOC_BYTES("HPF Coefficients", WM8962_LHPF2, 1), WM8962_DSP2_ENABLE("HD Bass Switch", WM8962_HDBASS_ENA_SHIFT), SND_SOC_BYTES("HD Bass Coefficients", WM8962_HDBASS_AI_1, 30), }; -- cgit v1.2.3 From d61e11260016f3589d60f94286c89ef823a26605 Mon Sep 17 00:00:00 2001 From: Shawn Guo Date: Fri, 16 Mar 2012 16:56:37 +0800 Subject: ASoC: fsl: separate SSI and DMA Kconfig options The fsl_ssi driver will possibly be shared between Freescale PowerPC and ARM/IMX families, so give it a separate Kconfig option. Then fsl_ssi driver can possibly be selected independently from selecting PowerPC DMA based PCM driver. Signed-off-by: Shawn Guo Acked-by: Sascha Hauer Acked-by: Timur Tabi Signed-off-by: Mark Brown --- sound/soc/fsl/Kconfig | 15 +++++++++------ sound/soc/fsl/Makefile | 3 ++- 2 files changed, 11 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index d754d34d68a6..ca693b2ed838 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -1,10 +1,11 @@ config SND_MPC52xx_DMA tristate -# ASoC platform support for the Freescale PowerPC SOCs that have an SSI and -# an Elo DMA controller, such as the MPC8610 and P1022. You will still need to -# select a platform driver and a codec driver. -config SND_SOC_POWERPC_SSI +config SND_SOC_FSL_SSI + tristate + depends on FSL_SOC + +config SND_SOC_POWERPC_DMA tristate depends on FSL_SOC @@ -12,7 +13,8 @@ config SND_SOC_MPC8610_HPCD tristate "ALSA SoC support for the Freescale MPC8610 HPCD board" # I2C is necessary for the CS4270 driver depends on MPC8610_HPCD && I2C - select SND_SOC_POWERPC_SSI + select SND_SOC_FSL_SSI + select SND_SOC_POWERPC_DMA select SND_SOC_CS4270 select SND_SOC_CS4270_VD33_ERRATA default y if MPC8610_HPCD @@ -23,7 +25,8 @@ config SND_SOC_P1022_DS tristate "ALSA SoC support for the Freescale P1022 DS board" # I2C is necessary for the WM8776 driver depends on P1022_DS && I2C - select SND_SOC_POWERPC_SSI + select SND_SOC_FSL_SSI + select SND_SOC_POWERPC_DMA select SND_SOC_WM8776 default y if P1022_DS help diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index b4a38c0ac58c..95d483f6af7f 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -9,7 +9,8 @@ obj-$(CONFIG_SND_SOC_P1022_DS) += snd-soc-p1022-ds.o # Freescale PowerPC SSI/DMA Platform Support snd-soc-fsl-ssi-objs := fsl_ssi.o snd-soc-fsl-dma-objs := fsl_dma.o -obj-$(CONFIG_SND_SOC_POWERPC_SSI) += snd-soc-fsl-ssi.o snd-soc-fsl-dma.o +obj-$(CONFIG_SND_SOC_FSL_SSI) += snd-soc-fsl-ssi.o +obj-$(CONFIG_SND_SOC_POWERPC_DMA) += snd-soc-fsl-dma.o # MPC5200 Platform Support obj-$(CONFIG_SND_MPC52xx_DMA) += mpc5200_dma.o -- cgit v1.2.3 From a23dc694828e3de96bf18e20459ba885ba91cb29 Mon Sep 17 00:00:00 2001 From: Shawn Guo Date: Fri, 16 Mar 2012 16:56:38 +0800 Subject: ASoC: imx: merge sound/soc/imx into sound/soc/fsl Freescale PowerPC and ARM/IMX families share the same SSI IP block. The patch merges sound/soc/imx into sound/soc/fsl, so that the possible code sharing and consolidation can happen. This is a plain merge, except that menuconfig SND_POWERPC_SOC is added in Kconfig for PowerPC platform as a correspondence to SND_IMX_SOC for IMX platform. Signed-off-by: Shawn Guo Acked-by: Sascha Hauer Acked-by: Timur Tabi Signed-off-by: Mark Brown --- arch/powerpc/configs/86xx/mpc8610_hpcd_defconfig | 1 + arch/powerpc/configs/mpc85xx_defconfig | 1 + arch/powerpc/configs/mpc85xx_smp_defconfig | 1 + sound/soc/Kconfig | 1 - sound/soc/Makefile | 1 - sound/soc/fsl/Kconfig | 92 +++ sound/soc/fsl/Makefile | 22 + sound/soc/fsl/eukrea-tlv320.c | 164 ++++++ sound/soc/fsl/imx-audmux.c | 314 +++++++++++ sound/soc/fsl/imx-audmux.h | 60 ++ sound/soc/fsl/imx-pcm-dma-mx2.c | 175 ++++++ sound/soc/fsl/imx-pcm-fiq.c | 336 +++++++++++ sound/soc/fsl/imx-pcm.c | 105 ++++ sound/soc/fsl/imx-pcm.h | 32 ++ sound/soc/fsl/imx-ssi.c | 690 +++++++++++++++++++++++ sound/soc/fsl/imx-ssi.h | 216 +++++++ sound/soc/fsl/mx27vis-aic32x4.c | 245 ++++++++ sound/soc/fsl/phycore-ac97.c | 125 ++++ sound/soc/fsl/wm1133-ev1.c | 304 ++++++++++ sound/soc/imx/Kconfig | 79 --- sound/soc/imx/Makefile | 22 - sound/soc/imx/eukrea-tlv320.c | 164 ------ sound/soc/imx/imx-audmux.c | 314 ----------- sound/soc/imx/imx-audmux.h | 60 -- sound/soc/imx/imx-pcm-dma-mx2.c | 175 ------ sound/soc/imx/imx-pcm-fiq.c | 336 ----------- sound/soc/imx/imx-pcm.c | 105 ---- sound/soc/imx/imx-pcm.h | 32 -- sound/soc/imx/imx-ssi.c | 690 ----------------------- sound/soc/imx/imx-ssi.h | 216 ------- sound/soc/imx/mx27vis-aic32x4.c | 245 -------- sound/soc/imx/phycore-ac97.c | 125 ---- sound/soc/imx/wm1133-ev1.c | 304 ---------- 33 files changed, 2883 insertions(+), 2869 deletions(-) create mode 100644 sound/soc/fsl/eukrea-tlv320.c create mode 100644 sound/soc/fsl/imx-audmux.c create mode 100644 sound/soc/fsl/imx-audmux.h create mode 100644 sound/soc/fsl/imx-pcm-dma-mx2.c create mode 100644 sound/soc/fsl/imx-pcm-fiq.c create mode 100644 sound/soc/fsl/imx-pcm.c create mode 100644 sound/soc/fsl/imx-pcm.h create mode 100644 sound/soc/fsl/imx-ssi.c create mode 100644 sound/soc/fsl/imx-ssi.h create mode 100644 sound/soc/fsl/mx27vis-aic32x4.c create mode 100644 sound/soc/fsl/phycore-ac97.c create mode 100644 sound/soc/fsl/wm1133-ev1.c delete mode 100644 sound/soc/imx/Kconfig delete mode 100644 sound/soc/imx/Makefile delete mode 100644 sound/soc/imx/eukrea-tlv320.c delete mode 100644 sound/soc/imx/imx-audmux.c delete mode 100644 sound/soc/imx/imx-audmux.h delete mode 100644 sound/soc/imx/imx-pcm-dma-mx2.c delete mode 100644 sound/soc/imx/imx-pcm-fiq.c delete mode 100644 sound/soc/imx/imx-pcm.c delete mode 100644 sound/soc/imx/imx-pcm.h delete mode 100644 sound/soc/imx/imx-ssi.c delete mode 100644 sound/soc/imx/imx-ssi.h delete mode 100644 sound/soc/imx/mx27vis-aic32x4.c delete mode 100644 sound/soc/imx/phycore-ac97.c delete mode 100644 sound/soc/imx/wm1133-ev1.c (limited to 'sound') diff --git a/arch/powerpc/configs/86xx/mpc8610_hpcd_defconfig b/arch/powerpc/configs/86xx/mpc8610_hpcd_defconfig index 0db9ba0423ff..c09598b31de1 100644 --- a/arch/powerpc/configs/86xx/mpc8610_hpcd_defconfig +++ b/arch/powerpc/configs/86xx/mpc8610_hpcd_defconfig @@ -100,6 +100,7 @@ CONFIG_SND_MIXER_OSS=y CONFIG_SND_PCM_OSS=y # CONFIG_SND_SUPPORT_OLD_API is not set CONFIG_SND_SOC=y +CONFIG_SND_POWERPC_SOC=y CONFIG_RTC_CLASS=y CONFIG_RTC_DRV_CMOS=y CONFIG_EXT2_FS=y diff --git a/arch/powerpc/configs/mpc85xx_defconfig b/arch/powerpc/configs/mpc85xx_defconfig index cc87a8441566..b7ac92a8b28f 100644 --- a/arch/powerpc/configs/mpc85xx_defconfig +++ b/arch/powerpc/configs/mpc85xx_defconfig @@ -140,6 +140,7 @@ CONFIG_SND_INTEL8X0=y # CONFIG_SND_PPC is not set # CONFIG_SND_USB is not set CONFIG_SND_SOC=y +CONFIG_SND_POWERPC_SOC=y CONFIG_HID_A4TECH=y CONFIG_HID_APPLE=y CONFIG_HID_BELKIN=y diff --git a/arch/powerpc/configs/mpc85xx_smp_defconfig b/arch/powerpc/configs/mpc85xx_smp_defconfig index 48d6682f2434..41d5e4709a08 100644 --- a/arch/powerpc/configs/mpc85xx_smp_defconfig +++ b/arch/powerpc/configs/mpc85xx_smp_defconfig @@ -142,6 +142,7 @@ CONFIG_SND_INTEL8X0=y # CONFIG_SND_PPC is not set # CONFIG_SND_USB is not set CONFIG_SND_SOC=y +CONFIG_SND_POWERPC_SOC=y CONFIG_HID_A4TECH=y CONFIG_HID_APPLE=y CONFIG_HID_BELKIN=y diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 91c985599d32..0f85f6d526e0 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -35,7 +35,6 @@ source "sound/soc/blackfin/Kconfig" source "sound/soc/davinci/Kconfig" source "sound/soc/ep93xx/Kconfig" source "sound/soc/fsl/Kconfig" -source "sound/soc/imx/Kconfig" source "sound/soc/jz4740/Kconfig" source "sound/soc/nuc900/Kconfig" source "sound/soc/omap/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 2feaf376e94b..363dfd6cffe7 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -12,7 +12,6 @@ obj-$(CONFIG_SND_SOC) += blackfin/ obj-$(CONFIG_SND_SOC) += davinci/ obj-$(CONFIG_SND_SOC) += ep93xx/ obj-$(CONFIG_SND_SOC) += fsl/ -obj-$(CONFIG_SND_SOC) += imx/ obj-$(CONFIG_SND_SOC) += jz4740/ obj-$(CONFIG_SND_SOC) += mid-x86/ obj-$(CONFIG_SND_SOC) += mxs/ diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index ca693b2ed838..19856a05208c 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -1,3 +1,15 @@ +config SND_SOC_FSL_SSI + tristate + +menuconfig SND_POWERPC_SOC + tristate "SoC Audio for Freescale PowerPC CPUs" + depends on FSL_SOC + help + Say Y or M if you want to add support for codecs attached to + the PowerPC CPUs. + +if SND_POWERPC_SOC + config SND_MPC52xx_DMA tristate @@ -68,3 +80,83 @@ config SND_MPC52xx_SOC_EFIKA help Say Y if you want to add support for sound on the Efika. +endif # SND_POWERPC_SOC + +menuconfig SND_IMX_SOC + tristate "SoC Audio for Freescale i.MX CPUs" + depends on ARCH_MXC + help + Say Y or M if you want to add support for codecs attached to + the i.MX CPUs. + +if SND_IMX_SOC + +config SND_SOC_IMX_SSI + tristate + +config SND_SOC_IMX_PCM + tristate + +config SND_MXC_SOC_FIQ + tristate + select FIQ + select SND_SOC_IMX_PCM + +config SND_MXC_SOC_MX2 + tristate + select SND_SOC_DMAENGINE_PCM + select SND_SOC_IMX_PCM + +config SND_SOC_IMX_AUDMUX + tristate + +config SND_MXC_SOC_WM1133_EV1 + tristate "Audio on the the i.MX31ADS with WM1133-EV1 fitted" + depends on MACH_MX31ADS_WM1133_EV1 && EXPERIMENTAL + select SND_SOC_WM8350 + select SND_MXC_SOC_FIQ + select SND_SOC_IMX_AUDMUX + select SND_SOC_IMX_SSI + help + Enable support for audio on the i.MX31ADS with the WM1133-EV1 + PMIC board with WM8835x fitted. + +config SND_SOC_MX27VIS_AIC32X4 + tristate "SoC audio support for Visstrim M10 boards" + depends on MACH_IMX27_VISSTRIM_M10 && I2C + select SND_SOC_TLV320AIC32X4 + select SND_MXC_SOC_MX2 + select SND_SOC_IMX_AUDMUX + select SND_SOC_IMX_SSI + help + Say Y if you want to add support for SoC audio on Visstrim SM10 + board with TLV320AIC32X4 codec. + +config SND_SOC_PHYCORE_AC97 + tristate "SoC Audio support for Phytec phyCORE (and phyCARD) boards" + depends on MACH_PCM043 || MACH_PCA100 + select SND_SOC_AC97_BUS + select SND_SOC_WM9712 + select SND_MXC_SOC_FIQ + select SND_SOC_IMX_AUDMUX + select SND_SOC_IMX_SSI + help + Say Y if you want to add support for SoC audio on Phytec phyCORE + and phyCARD boards in AC97 mode + +config SND_SOC_EUKREA_TLV320 + tristate "Eukrea TLV320" + depends on MACH_EUKREA_MBIMX27_BASEBOARD \ + || MACH_EUKREA_MBIMXSD25_BASEBOARD \ + || MACH_EUKREA_MBIMXSD35_BASEBOARD \ + || MACH_EUKREA_MBIMXSD51_BASEBOARD + depends on I2C + select SND_SOC_TLV320AIC23 + select SND_MXC_SOC_FIQ + select SND_SOC_IMX_AUDMUX + select SND_SOC_IMX_SSI + help + Enable I2S based access to the TLV320AIC23B codec attached + to the SSI interface + +endif # SND_IMX_SOC diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index 95d483f6af7f..36c257fd35bb 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -21,3 +21,25 @@ obj-$(CONFIG_SND_SOC_MPC5200_AC97) += mpc5200_psc_ac97.o obj-$(CONFIG_SND_MPC52xx_SOC_PCM030) += pcm030-audio-fabric.o obj-$(CONFIG_SND_MPC52xx_SOC_EFIKA) += efika-audio-fabric.o +# i.MX Platform Support +snd-soc-imx-ssi-objs := imx-ssi.o +snd-soc-imx-audmux-objs := imx-audmux.o + +obj-$(CONFIG_SND_SOC_IMX_SSI) += snd-soc-imx-ssi.o +obj-$(CONFIG_SND_SOC_IMX_AUDMUX) += snd-soc-imx-audmux.o + +obj-$(CONFIG_SND_SOC_IMX_PCM) += snd-soc-imx-pcm.o +snd-soc-imx-pcm-y := imx-pcm.o +snd-soc-imx-pcm-$(CONFIG_SND_MXC_SOC_FIQ) += imx-pcm-fiq.o +snd-soc-imx-pcm-$(CONFIG_SND_MXC_SOC_MX2) += imx-pcm-dma-mx2.o + +# i.MX Machine Support +snd-soc-eukrea-tlv320-objs := eukrea-tlv320.o +snd-soc-phycore-ac97-objs := phycore-ac97.o +snd-soc-mx27vis-aic32x4-objs := mx27vis-aic32x4.o +snd-soc-wm1133-ev1-objs := wm1133-ev1.o + +obj-$(CONFIG_SND_SOC_EUKREA_TLV320) += snd-soc-eukrea-tlv320.o +obj-$(CONFIG_SND_SOC_PHYCORE_AC97) += snd-soc-phycore-ac97.o +obj-$(CONFIG_SND_SOC_MX27VIS_AIC32X4) += snd-soc-mx27vis-aic32x4.o +obj-$(CONFIG_SND_MXC_SOC_WM1133_EV1) += snd-soc-wm1133-ev1.o diff --git a/sound/soc/fsl/eukrea-tlv320.c b/sound/soc/fsl/eukrea-tlv320.c new file mode 100644 index 000000000000..efb9ede01208 --- /dev/null +++ b/sound/soc/fsl/eukrea-tlv320.c @@ -0,0 +1,164 @@ +/* + * eukrea-tlv320.c -- SoC audio for eukrea_cpuimxXX in I2S mode + * + * Copyright 2010 Eric Bénard, Eukréa Electromatique + * + * based on sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c + * which is Copyright 2009 Simtec Electronics + * and on sound/soc/imx/phycore-ac97.c which is + * Copyright 2009 Sascha Hauer, Pengutronix + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + */ + +#include +#include +#include +#include +#include +#include +#include +#include + +#include "../codecs/tlv320aic23.h" +#include "imx-ssi.h" +#include "imx-audmux.h" + +#define CODEC_CLOCK 12000000 + +static int eukrea_tlv320_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + int ret; + + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret) { + pr_err("%s: failed set cpu dai format\n", __func__); + return ret; + } + + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret) { + pr_err("%s: failed set codec dai format\n", __func__); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, 0, + CODEC_CLOCK, SND_SOC_CLOCK_OUT); + if (ret) { + pr_err("%s: failed setting codec sysclk\n", __func__); + return ret; + } + snd_soc_dai_set_tdm_slot(cpu_dai, 0xffffffc, 0xffffffc, 2, 0); + + ret = snd_soc_dai_set_sysclk(cpu_dai, IMX_SSP_SYS_CLK, 0, + SND_SOC_CLOCK_IN); + if (ret) { + pr_err("can't set CPU system clock IMX_SSP_SYS_CLK\n"); + return ret; + } + + return 0; +} + +static struct snd_soc_ops eukrea_tlv320_snd_ops = { + .hw_params = eukrea_tlv320_hw_params, +}; + +static struct snd_soc_dai_link eukrea_tlv320_dai = { + .name = "tlv320aic23", + .stream_name = "TLV320AIC23", + .codec_dai_name = "tlv320aic23-hifi", + .platform_name = "imx-fiq-pcm-audio.0", + .codec_name = "tlv320aic23-codec.0-001a", + .cpu_dai_name = "imx-ssi.0", + .ops = &eukrea_tlv320_snd_ops, +}; + +static struct snd_soc_card eukrea_tlv320 = { + .name = "cpuimx-audio", + .owner = THIS_MODULE, + .dai_link = &eukrea_tlv320_dai, + .num_links = 1, +}; + +static struct platform_device *eukrea_tlv320_snd_device; + +static int __init eukrea_tlv320_init(void) +{ + int ret; + int int_port = 0, ext_port; + + if (machine_is_eukrea_cpuimx27()) { + imx_audmux_v1_configure_port(MX27_AUDMUX_HPCR1_SSI0, + IMX_AUDMUX_V1_PCR_SYN | + IMX_AUDMUX_V1_PCR_TFSDIR | + IMX_AUDMUX_V1_PCR_TCLKDIR | + IMX_AUDMUX_V1_PCR_RFSDIR | + IMX_AUDMUX_V1_PCR_RCLKDIR | + IMX_AUDMUX_V1_PCR_TFCSEL(MX27_AUDMUX_HPCR3_SSI_PINS_4) | + IMX_AUDMUX_V1_PCR_RFCSEL(MX27_AUDMUX_HPCR3_SSI_PINS_4) | + IMX_AUDMUX_V1_PCR_RXDSEL(MX27_AUDMUX_HPCR3_SSI_PINS_4) + ); + imx_audmux_v1_configure_port(MX27_AUDMUX_HPCR3_SSI_PINS_4, + IMX_AUDMUX_V1_PCR_SYN | + IMX_AUDMUX_V1_PCR_RXDSEL(MX27_AUDMUX_HPCR1_SSI0) + ); + } else if (machine_is_eukrea_cpuimx25sd() || + machine_is_eukrea_cpuimx35sd() || + machine_is_eukrea_cpuimx51sd()) { + ext_port = machine_is_eukrea_cpuimx25sd() ? 4 : 3; + imx_audmux_v2_configure_port(int_port, + IMX_AUDMUX_V2_PTCR_SYN | + IMX_AUDMUX_V2_PTCR_TFSDIR | + IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) | + IMX_AUDMUX_V2_PTCR_TCLKDIR | + IMX_AUDMUX_V2_PTCR_TCSEL(ext_port), + IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port) + ); + imx_audmux_v2_configure_port(ext_port, + IMX_AUDMUX_V2_PTCR_SYN, + IMX_AUDMUX_V2_PDCR_RXDSEL(int_port) + ); + } else { + /* return happy. We might run on a totally different machine */ + return 0; + } + + eukrea_tlv320_snd_device = platform_device_alloc("soc-audio", -1); + if (!eukrea_tlv320_snd_device) + return -ENOMEM; + + platform_set_drvdata(eukrea_tlv320_snd_device, &eukrea_tlv320); + ret = platform_device_add(eukrea_tlv320_snd_device); + + if (ret) { + printk(KERN_ERR "ASoC: Platform device allocation failed\n"); + platform_device_put(eukrea_tlv320_snd_device); + } + + return ret; +} + +static void __exit eukrea_tlv320_exit(void) +{ + platform_device_unregister(eukrea_tlv320_snd_device); +} + +module_init(eukrea_tlv320_init); +module_exit(eukrea_tlv320_exit); + +MODULE_AUTHOR("Eric Bénard "); +MODULE_DESCRIPTION("CPUIMX ALSA SoC driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c new file mode 100644 index 000000000000..601df809a26a --- /dev/null +++ b/sound/soc/fsl/imx-audmux.c @@ -0,0 +1,314 @@ +/* + * Copyright 2012 Freescale Semiconductor, Inc. + * Copyright 2012 Linaro Ltd. + * Copyright 2009 Pengutronix, Sascha Hauer + * + * Initial development of this code was funded by + * Phytec Messtechnik GmbH, http://www.phytec.de + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "imx-audmux.h" + +#define DRIVER_NAME "imx-audmux" + +static struct clk *audmux_clk; +static void __iomem *audmux_base; + +#define IMX_AUDMUX_V2_PTCR(x) ((x) * 8) +#define IMX_AUDMUX_V2_PDCR(x) ((x) * 8 + 4) + +#ifdef CONFIG_DEBUG_FS +static struct dentry *audmux_debugfs_root; + +static int audmux_open_file(struct inode *inode, struct file *file) +{ + file->private_data = inode->i_private; + return 0; +} + +/* There is an annoying discontinuity in the SSI numbering with regard + * to the Linux number of the devices */ +static const char *audmux_port_string(int port) +{ + switch (port) { + case MX31_AUDMUX_PORT1_SSI0: + return "imx-ssi.0"; + case MX31_AUDMUX_PORT2_SSI1: + return "imx-ssi.1"; + case MX31_AUDMUX_PORT3_SSI_PINS_3: + return "SSI3"; + case MX31_AUDMUX_PORT4_SSI_PINS_4: + return "SSI4"; + case MX31_AUDMUX_PORT5_SSI_PINS_5: + return "SSI5"; + case MX31_AUDMUX_PORT6_SSI_PINS_6: + return "SSI6"; + default: + return "UNKNOWN"; + } +} + +static ssize_t audmux_read_file(struct file *file, char __user *user_buf, + size_t count, loff_t *ppos) +{ + ssize_t ret; + char *buf = kmalloc(PAGE_SIZE, GFP_KERNEL); + int port = (int)file->private_data; + u32 pdcr, ptcr; + + if (!buf) + return -ENOMEM; + + if (audmux_clk) + clk_prepare_enable(audmux_clk); + + ptcr = readl(audmux_base + IMX_AUDMUX_V2_PTCR(port)); + pdcr = readl(audmux_base + IMX_AUDMUX_V2_PDCR(port)); + + if (audmux_clk) + clk_disable_unprepare(audmux_clk); + + ret = snprintf(buf, PAGE_SIZE, "PDCR: %08x\nPTCR: %08x\n", + pdcr, ptcr); + + if (ptcr & IMX_AUDMUX_V2_PTCR_TFSDIR) + ret += snprintf(buf + ret, PAGE_SIZE - ret, + "TxFS output from %s, ", + audmux_port_string((ptcr >> 27) & 0x7)); + else + ret += snprintf(buf + ret, PAGE_SIZE - ret, + "TxFS input, "); + + if (ptcr & IMX_AUDMUX_V2_PTCR_TCLKDIR) + ret += snprintf(buf + ret, PAGE_SIZE - ret, + "TxClk output from %s", + audmux_port_string((ptcr >> 22) & 0x7)); + else + ret += snprintf(buf + ret, PAGE_SIZE - ret, + "TxClk input"); + + ret += snprintf(buf + ret, PAGE_SIZE - ret, "\n"); + + if (ptcr & IMX_AUDMUX_V2_PTCR_SYN) { + ret += snprintf(buf + ret, PAGE_SIZE - ret, + "Port is symmetric"); + } else { + if (ptcr & IMX_AUDMUX_V2_PTCR_RFSDIR) + ret += snprintf(buf + ret, PAGE_SIZE - ret, + "RxFS output from %s, ", + audmux_port_string((ptcr >> 17) & 0x7)); + else + ret += snprintf(buf + ret, PAGE_SIZE - ret, + "RxFS input, "); + + if (ptcr & IMX_AUDMUX_V2_PTCR_RCLKDIR) + ret += snprintf(buf + ret, PAGE_SIZE - ret, + "RxClk output from %s", + audmux_port_string((ptcr >> 12) & 0x7)); + else + ret += snprintf(buf + ret, PAGE_SIZE - ret, + "RxClk input"); + } + + ret += snprintf(buf + ret, PAGE_SIZE - ret, + "\nData received from %s\n", + audmux_port_string((pdcr >> 13) & 0x7)); + + ret = simple_read_from_buffer(user_buf, count, ppos, buf, ret); + + kfree(buf); + + return ret; +} + +static const struct file_operations audmux_debugfs_fops = { + .open = audmux_open_file, + .read = audmux_read_file, + .llseek = default_llseek, +}; + +static void __init audmux_debugfs_init(void) +{ + int i; + char buf[20]; + + audmux_debugfs_root = debugfs_create_dir("audmux", NULL); + if (!audmux_debugfs_root) { + pr_warning("Failed to create AUDMUX debugfs root\n"); + return; + } + + for (i = 1; i < 8; i++) { + snprintf(buf, sizeof(buf), "ssi%d", i); + if (!debugfs_create_file(buf, 0444, audmux_debugfs_root, + (void *)i, &audmux_debugfs_fops)) + pr_warning("Failed to create AUDMUX port %d debugfs file\n", + i); + } +} + +static void __devexit audmux_debugfs_remove(void) +{ + debugfs_remove_recursive(audmux_debugfs_root); +} +#else +static inline void audmux_debugfs_init(void) +{ +} + +static inline void audmux_debugfs_remove(void) +{ +} +#endif + +enum imx_audmux_type { + IMX21_AUDMUX, + IMX31_AUDMUX, +} audmux_type; + +static struct platform_device_id imx_audmux_ids[] = { + { + .name = "imx21-audmux", + .driver_data = IMX21_AUDMUX, + }, { + .name = "imx31-audmux", + .driver_data = IMX31_AUDMUX, + }, { + /* sentinel */ + } +}; +MODULE_DEVICE_TABLE(platform, imx_audmux_ids); + +static const struct of_device_id imx_audmux_dt_ids[] = { + { .compatible = "fsl,imx21-audmux", .data = &imx_audmux_ids[0], }, + { .compatible = "fsl,imx31-audmux", .data = &imx_audmux_ids[1], }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, imx_audmux_dt_ids); + +static const uint8_t port_mapping[] = { + 0x0, 0x4, 0x8, 0x10, 0x14, 0x1c, +}; + +int imx_audmux_v1_configure_port(unsigned int port, unsigned int pcr) +{ + if (audmux_type != IMX21_AUDMUX) + return -EINVAL; + + if (!audmux_base) + return -ENOSYS; + + if (port >= ARRAY_SIZE(port_mapping)) + return -EINVAL; + + writel(pcr, audmux_base + port_mapping[port]); + + return 0; +} +EXPORT_SYMBOL_GPL(imx_audmux_v1_configure_port); + +int imx_audmux_v2_configure_port(unsigned int port, unsigned int ptcr, + unsigned int pdcr) +{ + if (audmux_type != IMX31_AUDMUX) + return -EINVAL; + + if (!audmux_base) + return -ENOSYS; + + if (audmux_clk) + clk_prepare_enable(audmux_clk); + + writel(ptcr, audmux_base + IMX_AUDMUX_V2_PTCR(port)); + writel(pdcr, audmux_base + IMX_AUDMUX_V2_PDCR(port)); + + if (audmux_clk) + clk_disable_unprepare(audmux_clk); + + return 0; +} +EXPORT_SYMBOL_GPL(imx_audmux_v2_configure_port); + +static int __devinit imx_audmux_probe(struct platform_device *pdev) +{ + struct resource *res; + const struct of_device_id *of_id = + of_match_device(imx_audmux_dt_ids, &pdev->dev); + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + audmux_base = devm_request_and_ioremap(&pdev->dev, res); + if (!audmux_base) + return -EADDRNOTAVAIL; + + audmux_clk = clk_get(&pdev->dev, "audmux"); + if (IS_ERR(audmux_clk)) { + dev_dbg(&pdev->dev, "cannot get clock: %ld\n", + PTR_ERR(audmux_clk)); + audmux_clk = NULL; + } + + if (of_id) + pdev->id_entry = of_id->data; + audmux_type = pdev->id_entry->driver_data; + if (audmux_type == IMX31_AUDMUX) + audmux_debugfs_init(); + + return 0; +} + +static int __devexit imx_audmux_remove(struct platform_device *pdev) +{ + if (audmux_type == IMX31_AUDMUX) + audmux_debugfs_remove(); + clk_put(audmux_clk); + + return 0; +} + +static struct platform_driver imx_audmux_driver = { + .probe = imx_audmux_probe, + .remove = __devexit_p(imx_audmux_remove), + .id_table = imx_audmux_ids, + .driver = { + .name = DRIVER_NAME, + .owner = THIS_MODULE, + .of_match_table = imx_audmux_dt_ids, + } +}; + +static int __init imx_audmux_init(void) +{ + return platform_driver_register(&imx_audmux_driver); +} +subsys_initcall(imx_audmux_init); + +static void __exit imx_audmux_exit(void) +{ + platform_driver_unregister(&imx_audmux_driver); +} +module_exit(imx_audmux_exit); + +MODULE_DESCRIPTION("Freescale i.MX AUDMUX driver"); +MODULE_AUTHOR("Sascha Hauer "); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:" DRIVER_NAME); diff --git a/sound/soc/fsl/imx-audmux.h b/sound/soc/fsl/imx-audmux.h new file mode 100644 index 000000000000..04ebbab8d7b9 --- /dev/null +++ b/sound/soc/fsl/imx-audmux.h @@ -0,0 +1,60 @@ +#ifndef __IMX_AUDMUX_H +#define __IMX_AUDMUX_H + +#define MX27_AUDMUX_HPCR1_SSI0 0 +#define MX27_AUDMUX_HPCR2_SSI1 1 +#define MX27_AUDMUX_HPCR3_SSI_PINS_4 2 +#define MX27_AUDMUX_PPCR1_SSI_PINS_1 3 +#define MX27_AUDMUX_PPCR2_SSI_PINS_2 4 +#define MX27_AUDMUX_PPCR3_SSI_PINS_3 5 + +#define MX31_AUDMUX_PORT1_SSI0 0 +#define MX31_AUDMUX_PORT2_SSI1 1 +#define MX31_AUDMUX_PORT3_SSI_PINS_3 2 +#define MX31_AUDMUX_PORT4_SSI_PINS_4 3 +#define MX31_AUDMUX_PORT5_SSI_PINS_5 4 +#define MX31_AUDMUX_PORT6_SSI_PINS_6 5 + +#define MX51_AUDMUX_PORT1_SSI0 0 +#define MX51_AUDMUX_PORT2_SSI1 1 +#define MX51_AUDMUX_PORT3 2 +#define MX51_AUDMUX_PORT4 3 +#define MX51_AUDMUX_PORT5 4 +#define MX51_AUDMUX_PORT6 5 +#define MX51_AUDMUX_PORT7 6 + +/* Register definitions for the i.MX21/27 Digital Audio Multiplexer */ +#define IMX_AUDMUX_V1_PCR_INMMASK(x) ((x) & 0xff) +#define IMX_AUDMUX_V1_PCR_INMEN (1 << 8) +#define IMX_AUDMUX_V1_PCR_TXRXEN (1 << 10) +#define IMX_AUDMUX_V1_PCR_SYN (1 << 12) +#define IMX_AUDMUX_V1_PCR_RXDSEL(x) (((x) & 0x7) << 13) +#define IMX_AUDMUX_V1_PCR_RFCSEL(x) (((x) & 0xf) << 20) +#define IMX_AUDMUX_V1_PCR_RCLKDIR (1 << 24) +#define IMX_AUDMUX_V1_PCR_RFSDIR (1 << 25) +#define IMX_AUDMUX_V1_PCR_TFCSEL(x) (((x) & 0xf) << 26) +#define IMX_AUDMUX_V1_PCR_TCLKDIR (1 << 30) +#define IMX_AUDMUX_V1_PCR_TFSDIR (1 << 31) + +/* Register definitions for the i.MX25/31/35/51 Digital Audio Multiplexer */ +#define IMX_AUDMUX_V2_PTCR_TFSDIR (1 << 31) +#define IMX_AUDMUX_V2_PTCR_TFSEL(x) (((x) & 0xf) << 27) +#define IMX_AUDMUX_V2_PTCR_TCLKDIR (1 << 26) +#define IMX_AUDMUX_V2_PTCR_TCSEL(x) (((x) & 0xf) << 22) +#define IMX_AUDMUX_V2_PTCR_RFSDIR (1 << 21) +#define IMX_AUDMUX_V2_PTCR_RFSEL(x) (((x) & 0xf) << 17) +#define IMX_AUDMUX_V2_PTCR_RCLKDIR (1 << 16) +#define IMX_AUDMUX_V2_PTCR_RCSEL(x) (((x) & 0xf) << 12) +#define IMX_AUDMUX_V2_PTCR_SYN (1 << 11) + +#define IMX_AUDMUX_V2_PDCR_RXDSEL(x) (((x) & 0x7) << 13) +#define IMX_AUDMUX_V2_PDCR_TXRXEN (1 << 12) +#define IMX_AUDMUX_V2_PDCR_MODE(x) (((x) & 0x3) << 8) +#define IMX_AUDMUX_V2_PDCR_INMMASK(x) ((x) & 0xff) + +int imx_audmux_v1_configure_port(unsigned int port, unsigned int pcr); + +int imx_audmux_v2_configure_port(unsigned int port, unsigned int ptcr, + unsigned int pdcr); + +#endif /* __IMX_AUDMUX_H */ diff --git a/sound/soc/fsl/imx-pcm-dma-mx2.c b/sound/soc/fsl/imx-pcm-dma-mx2.c new file mode 100644 index 000000000000..6b818de2fc03 --- /dev/null +++ b/sound/soc/fsl/imx-pcm-dma-mx2.c @@ -0,0 +1,175 @@ +/* + * imx-pcm-dma-mx2.c -- ALSA Soc Audio Layer + * + * Copyright 2009 Sascha Hauer + * + * This code is based on code copyrighted by Freescale, + * Liam Girdwood, Javier Martin and probably others. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include +#include +#include +#include +#include +#include + +#include + +#include "imx-pcm.h" + +static bool filter(struct dma_chan *chan, void *param) +{ + if (!imx_dma_is_general_purpose(chan)) + return false; + + chan->private = param; + + return true; +} + +static int snd_imx_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct dma_chan *chan = snd_dmaengine_pcm_get_chan(substream); + struct imx_pcm_dma_params *dma_params; + struct dma_slave_config slave_config; + int ret; + + dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + + ret = snd_hwparams_to_dma_slave_config(substream, params, &slave_config); + if (ret) + return ret; + + slave_config.device_fc = false; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + slave_config.dst_addr = dma_params->dma_addr; + slave_config.dst_maxburst = dma_params->burstsize; + } else { + slave_config.src_addr = dma_params->dma_addr; + slave_config.src_maxburst = dma_params->burstsize; + } + + ret = dmaengine_slave_config(chan, &slave_config); + if (ret) + return ret; + + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); + + return 0; +} + +static struct snd_pcm_hardware snd_imx_hardware = { + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_RESUME, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .rate_min = 8000, + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = IMX_SSI_DMABUF_SIZE, + .period_bytes_min = 128, + .period_bytes_max = 65535, /* Limited by SDMA engine */ + .periods_min = 2, + .periods_max = 255, + .fifo_size = 0, +}; + +static int snd_imx_open(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct imx_pcm_dma_params *dma_params; + struct imx_dma_data *dma_data; + int ret; + + snd_soc_set_runtime_hwparams(substream, &snd_imx_hardware); + + dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + + dma_data = kzalloc(sizeof(*dma_data), GFP_KERNEL); + dma_data->peripheral_type = IMX_DMATYPE_SSI; + dma_data->priority = DMA_PRIO_HIGH; + dma_data->dma_request = dma_params->dma; + + ret = snd_dmaengine_pcm_open(substream, filter, dma_data); + if (ret) { + kfree(dma_data); + return 0; + } + + snd_dmaengine_pcm_set_data(substream, dma_data); + + return 0; +} + +static int snd_imx_close(struct snd_pcm_substream *substream) +{ + struct imx_dma_data *dma_data = snd_dmaengine_pcm_get_data(substream); + + snd_dmaengine_pcm_close(substream); + kfree(dma_data); + + return 0; +} + +static struct snd_pcm_ops imx_pcm_ops = { + .open = snd_imx_open, + .close = snd_imx_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_imx_pcm_hw_params, + .trigger = snd_dmaengine_pcm_trigger, + .pointer = snd_dmaengine_pcm_pointer, + .mmap = snd_imx_pcm_mmap, +}; + +static struct snd_soc_platform_driver imx_soc_platform_mx2 = { + .ops = &imx_pcm_ops, + .pcm_new = imx_pcm_new, + .pcm_free = imx_pcm_free, +}; + +static int __devinit imx_soc_platform_probe(struct platform_device *pdev) +{ + return snd_soc_register_platform(&pdev->dev, &imx_soc_platform_mx2); +} + +static int __devexit imx_soc_platform_remove(struct platform_device *pdev) +{ + snd_soc_unregister_platform(&pdev->dev); + return 0; +} + +static struct platform_driver imx_pcm_driver = { + .driver = { + .name = "imx-pcm-audio", + .owner = THIS_MODULE, + }, + .probe = imx_soc_platform_probe, + .remove = __devexit_p(imx_soc_platform_remove), +}; + +module_platform_driver(imx_pcm_driver); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:imx-pcm-audio"); diff --git a/sound/soc/fsl/imx-pcm-fiq.c b/sound/soc/fsl/imx-pcm-fiq.c new file mode 100644 index 000000000000..456b7d723d66 --- /dev/null +++ b/sound/soc/fsl/imx-pcm-fiq.c @@ -0,0 +1,336 @@ +/* + * imx-pcm-fiq.c -- ALSA Soc Audio Layer + * + * Copyright 2009 Sascha Hauer + * + * This code is based on code copyrighted by Freescale, + * Liam Girdwood, Javier Martin and probably others. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include +#include +#include +#include +#include + +#include + +#include + +#include "imx-ssi.h" + +struct imx_pcm_runtime_data { + int period; + int periods; + unsigned long offset; + unsigned long last_offset; + unsigned long size; + struct hrtimer hrt; + int poll_time_ns; + struct snd_pcm_substream *substream; + atomic_t running; +}; + +static enum hrtimer_restart snd_hrtimer_callback(struct hrtimer *hrt) +{ + struct imx_pcm_runtime_data *iprtd = + container_of(hrt, struct imx_pcm_runtime_data, hrt); + struct snd_pcm_substream *substream = iprtd->substream; + struct snd_pcm_runtime *runtime = substream->runtime; + struct pt_regs regs; + unsigned long delta; + + if (!atomic_read(&iprtd->running)) + return HRTIMER_NORESTART; + + get_fiq_regs(®s); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + iprtd->offset = regs.ARM_r8 & 0xffff; + else + iprtd->offset = regs.ARM_r9 & 0xffff; + + /* How much data have we transferred since the last period report? */ + if (iprtd->offset >= iprtd->last_offset) + delta = iprtd->offset - iprtd->last_offset; + else + delta = runtime->buffer_size + iprtd->offset + - iprtd->last_offset; + + /* If we've transferred at least a period then report it and + * reset our poll time */ + if (delta >= iprtd->period) { + snd_pcm_period_elapsed(substream); + iprtd->last_offset = iprtd->offset; + } + + hrtimer_forward_now(hrt, ns_to_ktime(iprtd->poll_time_ns)); + + return HRTIMER_RESTART; +} + +static struct fiq_handler fh = { + .name = DRV_NAME, +}; + +static int snd_imx_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct imx_pcm_runtime_data *iprtd = runtime->private_data; + + iprtd->size = params_buffer_bytes(params); + iprtd->periods = params_periods(params); + iprtd->period = params_period_bytes(params) ; + iprtd->offset = 0; + iprtd->last_offset = 0; + iprtd->poll_time_ns = 1000000000 / params_rate(params) * + params_period_size(params); + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); + + return 0; +} + +static int snd_imx_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct imx_pcm_runtime_data *iprtd = runtime->private_data; + struct pt_regs regs; + + get_fiq_regs(®s); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + regs.ARM_r8 = (iprtd->period * iprtd->periods - 1) << 16; + else + regs.ARM_r9 = (iprtd->period * iprtd->periods - 1) << 16; + + set_fiq_regs(®s); + + return 0; +} + +static int fiq_enable; +static int imx_pcm_fiq; + +static int snd_imx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct imx_pcm_runtime_data *iprtd = runtime->private_data; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + atomic_set(&iprtd->running, 1); + hrtimer_start(&iprtd->hrt, ns_to_ktime(iprtd->poll_time_ns), + HRTIMER_MODE_REL); + if (++fiq_enable == 1) + enable_fiq(imx_pcm_fiq); + + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + atomic_set(&iprtd->running, 0); + + if (--fiq_enable == 0) + disable_fiq(imx_pcm_fiq); + + break; + default: + return -EINVAL; + } + + return 0; +} + +static snd_pcm_uframes_t snd_imx_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct imx_pcm_runtime_data *iprtd = runtime->private_data; + + return bytes_to_frames(substream->runtime, iprtd->offset); +} + +static struct snd_pcm_hardware snd_imx_hardware = { + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_RESUME, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .rate_min = 8000, + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = IMX_SSI_DMABUF_SIZE, + .period_bytes_min = 128, + .period_bytes_max = 16 * 1024, + .periods_min = 4, + .periods_max = 255, + .fifo_size = 0, +}; + +static int snd_imx_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct imx_pcm_runtime_data *iprtd; + int ret; + + iprtd = kzalloc(sizeof(*iprtd), GFP_KERNEL); + if (iprtd == NULL) + return -ENOMEM; + runtime->private_data = iprtd; + + iprtd->substream = substream; + + atomic_set(&iprtd->running, 0); + hrtimer_init(&iprtd->hrt, CLOCK_MONOTONIC, HRTIMER_MODE_REL); + iprtd->hrt.function = snd_hrtimer_callback; + + ret = snd_pcm_hw_constraint_integer(substream->runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (ret < 0) { + kfree(iprtd); + return ret; + } + + snd_soc_set_runtime_hwparams(substream, &snd_imx_hardware); + return 0; +} + +static int snd_imx_close(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct imx_pcm_runtime_data *iprtd = runtime->private_data; + + hrtimer_cancel(&iprtd->hrt); + + kfree(iprtd); + + return 0; +} + +static struct snd_pcm_ops imx_pcm_ops = { + .open = snd_imx_open, + .close = snd_imx_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_imx_pcm_hw_params, + .prepare = snd_imx_pcm_prepare, + .trigger = snd_imx_pcm_trigger, + .pointer = snd_imx_pcm_pointer, + .mmap = snd_imx_pcm_mmap, +}; + +static int ssi_irq = 0; + +static int imx_pcm_fiq_new(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_pcm *pcm = rtd->pcm; + struct snd_pcm_substream *substream; + int ret; + + ret = imx_pcm_new(rtd); + if (ret) + return ret; + + substream = pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream; + if (substream) { + struct snd_dma_buffer *buf = &substream->dma_buffer; + + imx_ssi_fiq_tx_buffer = (unsigned long)buf->area; + } + + substream = pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream; + if (substream) { + struct snd_dma_buffer *buf = &substream->dma_buffer; + + imx_ssi_fiq_rx_buffer = (unsigned long)buf->area; + } + + set_fiq_handler(&imx_ssi_fiq_start, + &imx_ssi_fiq_end - &imx_ssi_fiq_start); + + return 0; +} + +static void imx_pcm_fiq_free(struct snd_pcm *pcm) +{ + mxc_set_irq_fiq(ssi_irq, 0); + release_fiq(&fh); + imx_pcm_free(pcm); +} + +static struct snd_soc_platform_driver imx_soc_platform_fiq = { + .ops = &imx_pcm_ops, + .pcm_new = imx_pcm_fiq_new, + .pcm_free = imx_pcm_fiq_free, +}; + +static int __devinit imx_soc_platform_probe(struct platform_device *pdev) +{ + struct imx_ssi *ssi = platform_get_drvdata(pdev); + int ret; + + ret = claim_fiq(&fh); + if (ret) { + dev_err(&pdev->dev, "failed to claim fiq: %d", ret); + return ret; + } + + mxc_set_irq_fiq(ssi->irq, 1); + ssi_irq = ssi->irq; + + imx_pcm_fiq = ssi->irq; + + imx_ssi_fiq_base = (unsigned long)ssi->base; + + ssi->dma_params_tx.burstsize = 4; + ssi->dma_params_rx.burstsize = 6; + + ret = snd_soc_register_platform(&pdev->dev, &imx_soc_platform_fiq); + if (ret) + goto failed_register; + + return 0; + +failed_register: + mxc_set_irq_fiq(ssi_irq, 0); + release_fiq(&fh); + + return ret; +} + +static int __devexit imx_soc_platform_remove(struct platform_device *pdev) +{ + snd_soc_unregister_platform(&pdev->dev); + return 0; +} + +static struct platform_driver imx_pcm_driver = { + .driver = { + .name = "imx-fiq-pcm-audio", + .owner = THIS_MODULE, + }, + + .probe = imx_soc_platform_probe, + .remove = __devexit_p(imx_soc_platform_remove), +}; + +module_platform_driver(imx_pcm_driver); + +MODULE_LICENSE("GPL"); diff --git a/sound/soc/fsl/imx-pcm.c b/sound/soc/fsl/imx-pcm.c new file mode 100644 index 000000000000..93dc360b1777 --- /dev/null +++ b/sound/soc/fsl/imx-pcm.c @@ -0,0 +1,105 @@ +/* + * Copyright 2009 Sascha Hauer + * + * This code is based on code copyrighted by Freescale, + * Liam Girdwood, Javier Martin and probably others. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#include +#include +#include +#include +#include "imx-pcm.h" + +int snd_imx_pcm_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + int ret; + + ret = dma_mmap_writecombine(substream->pcm->card->dev, vma, + runtime->dma_area, runtime->dma_addr, runtime->dma_bytes); + + pr_debug("%s: ret: %d %p 0x%08x 0x%08x\n", __func__, ret, + runtime->dma_area, + runtime->dma_addr, + runtime->dma_bytes); + return ret; +} +EXPORT_SYMBOL_GPL(snd_imx_pcm_mmap); + +static int imx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) +{ + struct snd_pcm_substream *substream = pcm->streams[stream].substream; + struct snd_dma_buffer *buf = &substream->dma_buffer; + size_t size = IMX_SSI_DMABUF_SIZE; + + buf->dev.type = SNDRV_DMA_TYPE_DEV; + buf->dev.dev = pcm->card->dev; + buf->private_data = NULL; + buf->area = dma_alloc_writecombine(pcm->card->dev, size, + &buf->addr, GFP_KERNEL); + if (!buf->area) + return -ENOMEM; + buf->bytes = size; + + return 0; +} + +static u64 imx_pcm_dmamask = DMA_BIT_MASK(32); + +int imx_pcm_new(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_card *card = rtd->card->snd_card; + struct snd_pcm *pcm = rtd->pcm; + int ret = 0; + + if (!card->dev->dma_mask) + card->dev->dma_mask = &imx_pcm_dmamask; + if (!card->dev->coherent_dma_mask) + card->dev->coherent_dma_mask = DMA_BIT_MASK(32); + if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { + ret = imx_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_PLAYBACK); + if (ret) + goto out; + } + + if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { + ret = imx_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_CAPTURE); + if (ret) + goto out; + } + +out: + return ret; +} +EXPORT_SYMBOL_GPL(imx_pcm_new); + +void imx_pcm_free(struct snd_pcm *pcm) +{ + struct snd_pcm_substream *substream; + struct snd_dma_buffer *buf; + int stream; + + for (stream = 0; stream < 2; stream++) { + substream = pcm->streams[stream].substream; + if (!substream) + continue; + + buf = &substream->dma_buffer; + if (!buf->area) + continue; + + dma_free_writecombine(pcm->card->dev, buf->bytes, + buf->area, buf->addr); + buf->area = NULL; + } +} +EXPORT_SYMBOL_GPL(imx_pcm_free); diff --git a/sound/soc/fsl/imx-pcm.h b/sound/soc/fsl/imx-pcm.h new file mode 100644 index 000000000000..b5f5c3acf34d --- /dev/null +++ b/sound/soc/fsl/imx-pcm.h @@ -0,0 +1,32 @@ +/* + * Copyright 2009 Sascha Hauer + * + * This code is based on code copyrighted by Freescale, + * Liam Girdwood, Javier Martin and probably others. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#ifndef _IMX_PCM_H +#define _IMX_PCM_H + +/* + * Do not change this as the FIQ handler depends on this size + */ +#define IMX_SSI_DMABUF_SIZE (64 * 1024) + +struct imx_pcm_dma_params { + int dma; + unsigned long dma_addr; + int burstsize; +}; + +int snd_imx_pcm_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma); +int imx_pcm_new(struct snd_soc_pcm_runtime *rtd); +void imx_pcm_free(struct snd_pcm *pcm); + +#endif /* _IMX_PCM_H */ diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c new file mode 100644 index 000000000000..cf3ed0362c9c --- /dev/null +++ b/sound/soc/fsl/imx-ssi.c @@ -0,0 +1,690 @@ +/* + * imx-ssi.c -- ALSA Soc Audio Layer + * + * Copyright 2009 Sascha Hauer + * + * This code is based on code copyrighted by Freescale, + * Liam Girdwood, Javier Martin and probably others. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + * + * The i.MX SSI core has some nasty limitations in AC97 mode. While most + * sane processor vendors have a FIFO per AC97 slot, the i.MX has only + * one FIFO which combines all valid receive slots. We cannot even select + * which slots we want to receive. The WM9712 with which this driver + * was developed with always sends GPIO status data in slot 12 which + * we receive in our (PCM-) data stream. The only chance we have is to + * manually skip this data in the FIQ handler. With sampling rates different + * from 48000Hz not every frame has valid receive data, so the ratio + * between pcm data and GPIO status data changes. Our FIQ handler is not + * able to handle this, hence this driver only works with 48000Hz sampling + * rate. + * Reading and writing AC97 registers is another challenge. The core + * provides us status bits when the read register is updated with *another* + * value. When we read the same register two times (and the register still + * contains the same value) these status bits are not set. We work + * around this by not polling these bits but only wait a fixed delay. + * + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include +#include +#include +#include +#include + +#include +#include + +#include "imx-ssi.h" + +#define SSI_SACNT_DEFAULT (SSI_SACNT_AC97EN | SSI_SACNT_FV) + +/* + * SSI Network Mode or TDM slots configuration. + * Should only be called when port is inactive (i.e. SSIEN = 0). + */ +static int imx_ssi_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai, + unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width) +{ + struct imx_ssi *ssi = snd_soc_dai_get_drvdata(cpu_dai); + u32 sccr; + + sccr = readl(ssi->base + SSI_STCCR); + sccr &= ~SSI_STCCR_DC_MASK; + sccr |= SSI_STCCR_DC(slots - 1); + writel(sccr, ssi->base + SSI_STCCR); + + sccr = readl(ssi->base + SSI_SRCCR); + sccr &= ~SSI_STCCR_DC_MASK; + sccr |= SSI_STCCR_DC(slots - 1); + writel(sccr, ssi->base + SSI_SRCCR); + + writel(tx_mask, ssi->base + SSI_STMSK); + writel(rx_mask, ssi->base + SSI_SRMSK); + + return 0; +} + +/* + * SSI DAI format configuration. + * Should only be called when port is inactive (i.e. SSIEN = 0). + */ +static int imx_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) +{ + struct imx_ssi *ssi = snd_soc_dai_get_drvdata(cpu_dai); + u32 strcr = 0, scr; + + scr = readl(ssi->base + SSI_SCR) & ~(SSI_SCR_SYN | SSI_SCR_NET); + + /* DAI mode */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + /* data on rising edge of bclk, frame low 1clk before data */ + strcr |= SSI_STCR_TFSI | SSI_STCR_TEFS | SSI_STCR_TXBIT0; + scr |= SSI_SCR_NET; + if (ssi->flags & IMX_SSI_USE_I2S_SLAVE) { + scr &= ~SSI_I2S_MODE_MASK; + scr |= SSI_SCR_I2S_MODE_SLAVE; + } + break; + case SND_SOC_DAIFMT_LEFT_J: + /* data on rising edge of bclk, frame high with data */ + strcr |= SSI_STCR_TXBIT0; + break; + case SND_SOC_DAIFMT_DSP_B: + /* data on rising edge of bclk, frame high with data */ + strcr |= SSI_STCR_TFSL | SSI_STCR_TXBIT0; + break; + case SND_SOC_DAIFMT_DSP_A: + /* data on rising edge of bclk, frame high 1clk before data */ + strcr |= SSI_STCR_TFSL | SSI_STCR_TXBIT0 | SSI_STCR_TEFS; + break; + } + + /* DAI clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_IB_IF: + strcr |= SSI_STCR_TFSI; + strcr &= ~SSI_STCR_TSCKP; + break; + case SND_SOC_DAIFMT_IB_NF: + strcr &= ~(SSI_STCR_TSCKP | SSI_STCR_TFSI); + break; + case SND_SOC_DAIFMT_NB_IF: + strcr |= SSI_STCR_TFSI | SSI_STCR_TSCKP; + break; + case SND_SOC_DAIFMT_NB_NF: + strcr &= ~SSI_STCR_TFSI; + strcr |= SSI_STCR_TSCKP; + break; + } + + /* DAI clock master masks */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + break; + default: + /* Master mode not implemented, needs handling of clocks. */ + return -EINVAL; + } + + strcr |= SSI_STCR_TFEN0; + + if (ssi->flags & IMX_SSI_NET) + scr |= SSI_SCR_NET; + if (ssi->flags & IMX_SSI_SYN) + scr |= SSI_SCR_SYN; + + writel(strcr, ssi->base + SSI_STCR); + writel(strcr, ssi->base + SSI_SRCR); + writel(scr, ssi->base + SSI_SCR); + + return 0; +} + +/* + * SSI system clock configuration. + * Should only be called when port is inactive (i.e. SSIEN = 0). + */ +static int imx_ssi_set_dai_sysclk(struct snd_soc_dai *cpu_dai, + int clk_id, unsigned int freq, int dir) +{ + struct imx_ssi *ssi = snd_soc_dai_get_drvdata(cpu_dai); + u32 scr; + + scr = readl(ssi->base + SSI_SCR); + + switch (clk_id) { + case IMX_SSP_SYS_CLK: + if (dir == SND_SOC_CLOCK_OUT) + scr |= SSI_SCR_SYS_CLK_EN; + else + scr &= ~SSI_SCR_SYS_CLK_EN; + break; + default: + return -EINVAL; + } + + writel(scr, ssi->base + SSI_SCR); + + return 0; +} + +/* + * SSI Clock dividers + * Should only be called when port is inactive (i.e. SSIEN = 0). + */ +static int imx_ssi_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, + int div_id, int div) +{ + struct imx_ssi *ssi = snd_soc_dai_get_drvdata(cpu_dai); + u32 stccr, srccr; + + stccr = readl(ssi->base + SSI_STCCR); + srccr = readl(ssi->base + SSI_SRCCR); + + switch (div_id) { + case IMX_SSI_TX_DIV_2: + stccr &= ~SSI_STCCR_DIV2; + stccr |= div; + break; + case IMX_SSI_TX_DIV_PSR: + stccr &= ~SSI_STCCR_PSR; + stccr |= div; + break; + case IMX_SSI_TX_DIV_PM: + stccr &= ~0xff; + stccr |= SSI_STCCR_PM(div); + break; + case IMX_SSI_RX_DIV_2: + stccr &= ~SSI_STCCR_DIV2; + stccr |= div; + break; + case IMX_SSI_RX_DIV_PSR: + stccr &= ~SSI_STCCR_PSR; + stccr |= div; + break; + case IMX_SSI_RX_DIV_PM: + stccr &= ~0xff; + stccr |= SSI_STCCR_PM(div); + break; + default: + return -EINVAL; + } + + writel(stccr, ssi->base + SSI_STCCR); + writel(srccr, ssi->base + SSI_SRCCR); + + return 0; +} + +static int imx_ssi_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) +{ + struct imx_ssi *ssi = snd_soc_dai_get_drvdata(cpu_dai); + struct imx_pcm_dma_params *dma_data; + + /* Tx/Rx config */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + dma_data = &ssi->dma_params_tx; + else + dma_data = &ssi->dma_params_rx; + + snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data); + + return 0; +} + +/* + * Should only be called when port is inactive (i.e. SSIEN = 0), + * although can be called multiple times by upper layers. + */ +static int imx_ssi_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *cpu_dai) +{ + struct imx_ssi *ssi = snd_soc_dai_get_drvdata(cpu_dai); + u32 reg, sccr; + + /* Tx/Rx config */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + reg = SSI_STCCR; + else + reg = SSI_SRCCR; + + if (ssi->flags & IMX_SSI_SYN) + reg = SSI_STCCR; + + sccr = readl(ssi->base + reg) & ~SSI_STCCR_WL_MASK; + + /* DAI data (word) size */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + sccr |= SSI_SRCCR_WL(16); + break; + case SNDRV_PCM_FORMAT_S20_3LE: + sccr |= SSI_SRCCR_WL(20); + break; + case SNDRV_PCM_FORMAT_S24_LE: + sccr |= SSI_SRCCR_WL(24); + break; + } + + writel(sccr, ssi->base + reg); + + return 0; +} + +static int imx_ssi_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct imx_ssi *ssi = snd_soc_dai_get_drvdata(dai); + unsigned int sier_bits, sier; + unsigned int scr; + + scr = readl(ssi->base + SSI_SCR); + sier = readl(ssi->base + SSI_SIER); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (ssi->flags & IMX_SSI_DMA) + sier_bits = SSI_SIER_TDMAE; + else + sier_bits = SSI_SIER_TIE | SSI_SIER_TFE0_EN; + } else { + if (ssi->flags & IMX_SSI_DMA) + sier_bits = SSI_SIER_RDMAE; + else + sier_bits = SSI_SIER_RIE | SSI_SIER_RFF0_EN; + } + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + scr |= SSI_SCR_TE; + else + scr |= SSI_SCR_RE; + sier |= sier_bits; + + if (++ssi->enabled == 1) + scr |= SSI_SCR_SSIEN; + + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + scr &= ~SSI_SCR_TE; + else + scr &= ~SSI_SCR_RE; + sier &= ~sier_bits; + + if (--ssi->enabled == 0) + scr &= ~SSI_SCR_SSIEN; + + break; + default: + return -EINVAL; + } + + if (!(ssi->flags & IMX_SSI_USE_AC97)) + /* rx/tx are always enabled to access ac97 registers */ + writel(scr, ssi->base + SSI_SCR); + + writel(sier, ssi->base + SSI_SIER); + + return 0; +} + +static const struct snd_soc_dai_ops imx_ssi_pcm_dai_ops = { + .startup = imx_ssi_startup, + .hw_params = imx_ssi_hw_params, + .set_fmt = imx_ssi_set_dai_fmt, + .set_clkdiv = imx_ssi_set_dai_clkdiv, + .set_sysclk = imx_ssi_set_dai_sysclk, + .set_tdm_slot = imx_ssi_set_dai_tdm_slot, + .trigger = imx_ssi_trigger, +}; + +static int imx_ssi_dai_probe(struct snd_soc_dai *dai) +{ + struct imx_ssi *ssi = dev_get_drvdata(dai->dev); + uint32_t val; + + snd_soc_dai_set_drvdata(dai, ssi); + + val = SSI_SFCSR_TFWM0(ssi->dma_params_tx.burstsize) | + SSI_SFCSR_RFWM0(ssi->dma_params_rx.burstsize); + writel(val, ssi->base + SSI_SFCSR); + + return 0; +} + +static struct snd_soc_dai_driver imx_ssi_dai = { + .probe = imx_ssi_dai_probe, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .ops = &imx_ssi_pcm_dai_ops, +}; + +static struct snd_soc_dai_driver imx_ac97_dai = { + .probe = imx_ssi_dai_probe, + .ac97_control = 1, + .playback = { + .stream_name = "AC97 Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .capture = { + .stream_name = "AC97 Capture", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .ops = &imx_ssi_pcm_dai_ops, +}; + +static void setup_channel_to_ac97(struct imx_ssi *imx_ssi) +{ + void __iomem *base = imx_ssi->base; + + writel(0x0, base + SSI_SCR); + writel(0x0, base + SSI_STCR); + writel(0x0, base + SSI_SRCR); + + writel(SSI_SCR_SYN | SSI_SCR_NET, base + SSI_SCR); + + writel(SSI_SFCSR_RFWM0(8) | + SSI_SFCSR_TFWM0(8) | + SSI_SFCSR_RFWM1(8) | + SSI_SFCSR_TFWM1(8), base + SSI_SFCSR); + + writel(SSI_STCCR_WL(16) | SSI_STCCR_DC(12), base + SSI_STCCR); + writel(SSI_STCCR_WL(16) | SSI_STCCR_DC(12), base + SSI_SRCCR); + + writel(SSI_SCR_SYN | SSI_SCR_NET | SSI_SCR_SSIEN, base + SSI_SCR); + writel(SSI_SOR_WAIT(3), base + SSI_SOR); + + writel(SSI_SCR_SYN | SSI_SCR_NET | SSI_SCR_SSIEN | + SSI_SCR_TE | SSI_SCR_RE, + base + SSI_SCR); + + writel(SSI_SACNT_DEFAULT, base + SSI_SACNT); + writel(0xff, base + SSI_SACCDIS); + writel(0x300, base + SSI_SACCEN); +} + +static struct imx_ssi *ac97_ssi; + +static void imx_ssi_ac97_write(struct snd_ac97 *ac97, unsigned short reg, + unsigned short val) +{ + struct imx_ssi *imx_ssi = ac97_ssi; + void __iomem *base = imx_ssi->base; + unsigned int lreg; + unsigned int lval; + + if (reg > 0x7f) + return; + + pr_debug("%s: 0x%02x 0x%04x\n", __func__, reg, val); + + lreg = reg << 12; + writel(lreg, base + SSI_SACADD); + + lval = val << 4; + writel(lval , base + SSI_SACDAT); + + writel(SSI_SACNT_DEFAULT | SSI_SACNT_WR, base + SSI_SACNT); + udelay(100); +} + +static unsigned short imx_ssi_ac97_read(struct snd_ac97 *ac97, + unsigned short reg) +{ + struct imx_ssi *imx_ssi = ac97_ssi; + void __iomem *base = imx_ssi->base; + + unsigned short val = -1; + unsigned int lreg; + + lreg = (reg & 0x7f) << 12 ; + writel(lreg, base + SSI_SACADD); + writel(SSI_SACNT_DEFAULT | SSI_SACNT_RD, base + SSI_SACNT); + + udelay(100); + + val = (readl(base + SSI_SACDAT) >> 4) & 0xffff; + + pr_debug("%s: 0x%02x 0x%04x\n", __func__, reg, val); + + return val; +} + +static void imx_ssi_ac97_reset(struct snd_ac97 *ac97) +{ + struct imx_ssi *imx_ssi = ac97_ssi; + + if (imx_ssi->ac97_reset) + imx_ssi->ac97_reset(ac97); +} + +static void imx_ssi_ac97_warm_reset(struct snd_ac97 *ac97) +{ + struct imx_ssi *imx_ssi = ac97_ssi; + + if (imx_ssi->ac97_warm_reset) + imx_ssi->ac97_warm_reset(ac97); +} + +struct snd_ac97_bus_ops soc_ac97_ops = { + .read = imx_ssi_ac97_read, + .write = imx_ssi_ac97_write, + .reset = imx_ssi_ac97_reset, + .warm_reset = imx_ssi_ac97_warm_reset +}; +EXPORT_SYMBOL_GPL(soc_ac97_ops); + +static int imx_ssi_probe(struct platform_device *pdev) +{ + struct resource *res; + struct imx_ssi *ssi; + struct imx_ssi_platform_data *pdata = pdev->dev.platform_data; + int ret = 0; + struct snd_soc_dai_driver *dai; + + ssi = kzalloc(sizeof(*ssi), GFP_KERNEL); + if (!ssi) + return -ENOMEM; + dev_set_drvdata(&pdev->dev, ssi); + + if (pdata) { + ssi->ac97_reset = pdata->ac97_reset; + ssi->ac97_warm_reset = pdata->ac97_warm_reset; + ssi->flags = pdata->flags; + } + + ssi->irq = platform_get_irq(pdev, 0); + + ssi->clk = clk_get(&pdev->dev, NULL); + if (IS_ERR(ssi->clk)) { + ret = PTR_ERR(ssi->clk); + dev_err(&pdev->dev, "Cannot get the clock: %d\n", + ret); + goto failed_clk; + } + clk_enable(ssi->clk); + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!res) { + ret = -ENODEV; + goto failed_get_resource; + } + + if (!request_mem_region(res->start, resource_size(res), DRV_NAME)) { + dev_err(&pdev->dev, "request_mem_region failed\n"); + ret = -EBUSY; + goto failed_get_resource; + } + + ssi->base = ioremap(res->start, resource_size(res)); + if (!ssi->base) { + dev_err(&pdev->dev, "ioremap failed\n"); + ret = -ENODEV; + goto failed_ioremap; + } + + if (ssi->flags & IMX_SSI_USE_AC97) { + if (ac97_ssi) { + ret = -EBUSY; + goto failed_ac97; + } + ac97_ssi = ssi; + setup_channel_to_ac97(ssi); + dai = &imx_ac97_dai; + } else + dai = &imx_ssi_dai; + + writel(0x0, ssi->base + SSI_SIER); + + ssi->dma_params_rx.dma_addr = res->start + SSI_SRX0; + ssi->dma_params_tx.dma_addr = res->start + SSI_STX0; + + ssi->dma_params_tx.burstsize = 6; + ssi->dma_params_rx.burstsize = 4; + + res = platform_get_resource_byname(pdev, IORESOURCE_DMA, "tx0"); + if (res) + ssi->dma_params_tx.dma = res->start; + + res = platform_get_resource_byname(pdev, IORESOURCE_DMA, "rx0"); + if (res) + ssi->dma_params_rx.dma = res->start; + + platform_set_drvdata(pdev, ssi); + + ret = snd_soc_register_dai(&pdev->dev, dai); + if (ret) { + dev_err(&pdev->dev, "register DAI failed\n"); + goto failed_register; + } + + ssi->soc_platform_pdev_fiq = platform_device_alloc("imx-fiq-pcm-audio", pdev->id); + if (!ssi->soc_platform_pdev_fiq) { + ret = -ENOMEM; + goto failed_pdev_fiq_alloc; + } + + platform_set_drvdata(ssi->soc_platform_pdev_fiq, ssi); + ret = platform_device_add(ssi->soc_platform_pdev_fiq); + if (ret) { + dev_err(&pdev->dev, "failed to add platform device\n"); + goto failed_pdev_fiq_add; + } + + ssi->soc_platform_pdev = platform_device_alloc("imx-pcm-audio", pdev->id); + if (!ssi->soc_platform_pdev) { + ret = -ENOMEM; + goto failed_pdev_alloc; + } + + platform_set_drvdata(ssi->soc_platform_pdev, ssi); + ret = platform_device_add(ssi->soc_platform_pdev); + if (ret) { + dev_err(&pdev->dev, "failed to add platform device\n"); + goto failed_pdev_add; + } + + return 0; + +failed_pdev_add: + platform_device_put(ssi->soc_platform_pdev); +failed_pdev_alloc: + platform_device_del(ssi->soc_platform_pdev_fiq); +failed_pdev_fiq_add: + platform_device_put(ssi->soc_platform_pdev_fiq); +failed_pdev_fiq_alloc: + snd_soc_unregister_dai(&pdev->dev); +failed_register: +failed_ac97: + iounmap(ssi->base); +failed_ioremap: + release_mem_region(res->start, resource_size(res)); +failed_get_resource: + clk_disable(ssi->clk); + clk_put(ssi->clk); +failed_clk: + kfree(ssi); + + return ret; +} + +static int __devexit imx_ssi_remove(struct platform_device *pdev) +{ + struct resource *res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + struct imx_ssi *ssi = platform_get_drvdata(pdev); + + platform_device_unregister(ssi->soc_platform_pdev); + platform_device_unregister(ssi->soc_platform_pdev_fiq); + + snd_soc_unregister_dai(&pdev->dev); + + if (ssi->flags & IMX_SSI_USE_AC97) + ac97_ssi = NULL; + + iounmap(ssi->base); + release_mem_region(res->start, resource_size(res)); + clk_disable(ssi->clk); + clk_put(ssi->clk); + kfree(ssi); + + return 0; +} + +static struct platform_driver imx_ssi_driver = { + .probe = imx_ssi_probe, + .remove = __devexit_p(imx_ssi_remove), + + .driver = { + .name = "imx-ssi", + .owner = THIS_MODULE, + }, +}; + +module_platform_driver(imx_ssi_driver); + +/* Module information */ +MODULE_AUTHOR("Sascha Hauer, "); +MODULE_DESCRIPTION("i.MX I2S/ac97 SoC Interface"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:imx-ssi"); diff --git a/sound/soc/fsl/imx-ssi.h b/sound/soc/fsl/imx-ssi.h new file mode 100644 index 000000000000..5744e86ca878 --- /dev/null +++ b/sound/soc/fsl/imx-ssi.h @@ -0,0 +1,216 @@ +/* + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _IMX_SSI_H +#define _IMX_SSI_H + +#define SSI_STX0 0x00 +#define SSI_STX1 0x04 +#define SSI_SRX0 0x08 +#define SSI_SRX1 0x0c + +#define SSI_SCR 0x10 +#define SSI_SCR_CLK_IST (1 << 9) +#define SSI_SCR_CLK_IST_SHIFT 9 +#define SSI_SCR_TCH_EN (1 << 8) +#define SSI_SCR_SYS_CLK_EN (1 << 7) +#define SSI_SCR_I2S_MODE_NORM (0 << 5) +#define SSI_SCR_I2S_MODE_MSTR (1 << 5) +#define SSI_SCR_I2S_MODE_SLAVE (2 << 5) +#define SSI_I2S_MODE_MASK (3 << 5) +#define SSI_SCR_SYN (1 << 4) +#define SSI_SCR_NET (1 << 3) +#define SSI_SCR_RE (1 << 2) +#define SSI_SCR_TE (1 << 1) +#define SSI_SCR_SSIEN (1 << 0) + +#define SSI_SISR 0x14 +#define SSI_SISR_MASK ((1 << 19) - 1) +#define SSI_SISR_CMDAU (1 << 18) +#define SSI_SISR_CMDDU (1 << 17) +#define SSI_SISR_RXT (1 << 16) +#define SSI_SISR_RDR1 (1 << 15) +#define SSI_SISR_RDR0 (1 << 14) +#define SSI_SISR_TDE1 (1 << 13) +#define SSI_SISR_TDE0 (1 << 12) +#define SSI_SISR_ROE1 (1 << 11) +#define SSI_SISR_ROE0 (1 << 10) +#define SSI_SISR_TUE1 (1 << 9) +#define SSI_SISR_TUE0 (1 << 8) +#define SSI_SISR_TFS (1 << 7) +#define SSI_SISR_RFS (1 << 6) +#define SSI_SISR_TLS (1 << 5) +#define SSI_SISR_RLS (1 << 4) +#define SSI_SISR_RFF1 (1 << 3) +#define SSI_SISR_RFF0 (1 << 2) +#define SSI_SISR_TFE1 (1 << 1) +#define SSI_SISR_TFE0 (1 << 0) + +#define SSI_SIER 0x18 +#define SSI_SIER_RDMAE (1 << 22) +#define SSI_SIER_RIE (1 << 21) +#define SSI_SIER_TDMAE (1 << 20) +#define SSI_SIER_TIE (1 << 19) +#define SSI_SIER_CMDAU_EN (1 << 18) +#define SSI_SIER_CMDDU_EN (1 << 17) +#define SSI_SIER_RXT_EN (1 << 16) +#define SSI_SIER_RDR1_EN (1 << 15) +#define SSI_SIER_RDR0_EN (1 << 14) +#define SSI_SIER_TDE1_EN (1 << 13) +#define SSI_SIER_TDE0_EN (1 << 12) +#define SSI_SIER_ROE1_EN (1 << 11) +#define SSI_SIER_ROE0_EN (1 << 10) +#define SSI_SIER_TUE1_EN (1 << 9) +#define SSI_SIER_TUE0_EN (1 << 8) +#define SSI_SIER_TFS_EN (1 << 7) +#define SSI_SIER_RFS_EN (1 << 6) +#define SSI_SIER_TLS_EN (1 << 5) +#define SSI_SIER_RLS_EN (1 << 4) +#define SSI_SIER_RFF1_EN (1 << 3) +#define SSI_SIER_RFF0_EN (1 << 2) +#define SSI_SIER_TFE1_EN (1 << 1) +#define SSI_SIER_TFE0_EN (1 << 0) + +#define SSI_STCR 0x1c +#define SSI_STCR_TXBIT0 (1 << 9) +#define SSI_STCR_TFEN1 (1 << 8) +#define SSI_STCR_TFEN0 (1 << 7) +#define SSI_FIFO_ENABLE_0_SHIFT 7 +#define SSI_STCR_TFDIR (1 << 6) +#define SSI_STCR_TXDIR (1 << 5) +#define SSI_STCR_TSHFD (1 << 4) +#define SSI_STCR_TSCKP (1 << 3) +#define SSI_STCR_TFSI (1 << 2) +#define SSI_STCR_TFSL (1 << 1) +#define SSI_STCR_TEFS (1 << 0) + +#define SSI_SRCR 0x20 +#define SSI_SRCR_RXBIT0 (1 << 9) +#define SSI_SRCR_RFEN1 (1 << 8) +#define SSI_SRCR_RFEN0 (1 << 7) +#define SSI_FIFO_ENABLE_0_SHIFT 7 +#define SSI_SRCR_RFDIR (1 << 6) +#define SSI_SRCR_RXDIR (1 << 5) +#define SSI_SRCR_RSHFD (1 << 4) +#define SSI_SRCR_RSCKP (1 << 3) +#define SSI_SRCR_RFSI (1 << 2) +#define SSI_SRCR_RFSL (1 << 1) +#define SSI_SRCR_REFS (1 << 0) + +#define SSI_SRCCR 0x28 +#define SSI_SRCCR_DIV2 (1 << 18) +#define SSI_SRCCR_PSR (1 << 17) +#define SSI_SRCCR_WL(x) ((((x) - 2) >> 1) << 13) +#define SSI_SRCCR_DC(x) (((x) & 0x1f) << 8) +#define SSI_SRCCR_PM(x) (((x) & 0xff) << 0) +#define SSI_SRCCR_WL_MASK (0xf << 13) +#define SSI_SRCCR_DC_MASK (0x1f << 8) +#define SSI_SRCCR_PM_MASK (0xff << 0) + +#define SSI_STCCR 0x24 +#define SSI_STCCR_DIV2 (1 << 18) +#define SSI_STCCR_PSR (1 << 17) +#define SSI_STCCR_WL(x) ((((x) - 2) >> 1) << 13) +#define SSI_STCCR_DC(x) (((x) & 0x1f) << 8) +#define SSI_STCCR_PM(x) (((x) & 0xff) << 0) +#define SSI_STCCR_WL_MASK (0xf << 13) +#define SSI_STCCR_DC_MASK (0x1f << 8) +#define SSI_STCCR_PM_MASK (0xff << 0) + +#define SSI_SFCSR 0x2c +#define SSI_SFCSR_RFCNT1(x) (((x) & 0xf) << 28) +#define SSI_RX_FIFO_1_COUNT_SHIFT 28 +#define SSI_SFCSR_TFCNT1(x) (((x) & 0xf) << 24) +#define SSI_TX_FIFO_1_COUNT_SHIFT 24 +#define SSI_SFCSR_RFWM1(x) (((x) & 0xf) << 20) +#define SSI_SFCSR_TFWM1(x) (((x) & 0xf) << 16) +#define SSI_SFCSR_RFCNT0(x) (((x) & 0xf) << 12) +#define SSI_RX_FIFO_0_COUNT_SHIFT 12 +#define SSI_SFCSR_TFCNT0(x) (((x) & 0xf) << 8) +#define SSI_TX_FIFO_0_COUNT_SHIFT 8 +#define SSI_SFCSR_RFWM0(x) (((x) & 0xf) << 4) +#define SSI_SFCSR_TFWM0(x) (((x) & 0xf) << 0) +#define SSI_SFCSR_RFWM0_MASK (0xf << 4) +#define SSI_SFCSR_TFWM0_MASK (0xf << 0) + +#define SSI_STR 0x30 +#define SSI_STR_TEST (1 << 15) +#define SSI_STR_RCK2TCK (1 << 14) +#define SSI_STR_RFS2TFS (1 << 13) +#define SSI_STR_RXSTATE(x) (((x) & 0xf) << 8) +#define SSI_STR_TXD2RXD (1 << 7) +#define SSI_STR_TCK2RCK (1 << 6) +#define SSI_STR_TFS2RFS (1 << 5) +#define SSI_STR_TXSTATE(x) (((x) & 0xf) << 0) + +#define SSI_SOR 0x34 +#define SSI_SOR_CLKOFF (1 << 6) +#define SSI_SOR_RX_CLR (1 << 5) +#define SSI_SOR_TX_CLR (1 << 4) +#define SSI_SOR_INIT (1 << 3) +#define SSI_SOR_WAIT(x) (((x) & 0x3) << 1) +#define SSI_SOR_WAIT_MASK (0x3 << 1) +#define SSI_SOR_SYNRST (1 << 0) + +#define SSI_SACNT 0x38 +#define SSI_SACNT_FRDIV(x) (((x) & 0x3f) << 5) +#define SSI_SACNT_WR (1 << 4) +#define SSI_SACNT_RD (1 << 3) +#define SSI_SACNT_TIF (1 << 2) +#define SSI_SACNT_FV (1 << 1) +#define SSI_SACNT_AC97EN (1 << 0) + +#define SSI_SACADD 0x3c +#define SSI_SACDAT 0x40 +#define SSI_SATAG 0x44 +#define SSI_STMSK 0x48 +#define SSI_SRMSK 0x4c +#define SSI_SACCST 0x50 +#define SSI_SACCEN 0x54 +#define SSI_SACCDIS 0x58 + +/* SSI clock sources */ +#define IMX_SSP_SYS_CLK 0 + +/* SSI audio dividers */ +#define IMX_SSI_TX_DIV_2 0 +#define IMX_SSI_TX_DIV_PSR 1 +#define IMX_SSI_TX_DIV_PM 2 +#define IMX_SSI_RX_DIV_2 3 +#define IMX_SSI_RX_DIV_PSR 4 +#define IMX_SSI_RX_DIV_PM 5 + +#define DRV_NAME "imx-ssi" + +#include +#include +#include "imx-pcm.h" + +struct imx_ssi { + struct platform_device *ac97_dev; + + struct snd_soc_dai *imx_ac97; + struct clk *clk; + void __iomem *base; + int irq; + int fiq_enable; + unsigned int offset; + + unsigned int flags; + + void (*ac97_reset) (struct snd_ac97 *ac97); + void (*ac97_warm_reset)(struct snd_ac97 *ac97); + + struct imx_pcm_dma_params dma_params_rx; + struct imx_pcm_dma_params dma_params_tx; + + int enabled; + + struct platform_device *soc_platform_pdev; + struct platform_device *soc_platform_pdev_fiq; +}; + +#endif /* _IMX_SSI_H */ diff --git a/sound/soc/fsl/mx27vis-aic32x4.c b/sound/soc/fsl/mx27vis-aic32x4.c new file mode 100644 index 000000000000..f6d04ad4bb39 --- /dev/null +++ b/sound/soc/fsl/mx27vis-aic32x4.c @@ -0,0 +1,245 @@ +/* + * mx27vis-aic32x4.c + * + * Copyright 2011 Vista Silicon S.L. + * + * Author: Javier Martin + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, + * MA 02110-1301, USA. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "../codecs/tlv320aic32x4.h" +#include "imx-ssi.h" +#include "imx-audmux.h" + +#define MX27VIS_AMP_GAIN 0 +#define MX27VIS_AMP_MUTE 1 + +#define MX27VIS_PIN_G0 (GPIO_PORTF + 9) +#define MX27VIS_PIN_G1 (GPIO_PORTF + 8) +#define MX27VIS_PIN_SDL (GPIO_PORTE + 5) +#define MX27VIS_PIN_SDR (GPIO_PORTF + 7) + +static int mx27vis_amp_gain; +static int mx27vis_amp_mute; + +static const int mx27vis_amp_pins[] = { + MX27VIS_PIN_G0 | GPIO_GPIO | GPIO_OUT, + MX27VIS_PIN_G1 | GPIO_GPIO | GPIO_OUT, + MX27VIS_PIN_SDL | GPIO_GPIO | GPIO_OUT, + MX27VIS_PIN_SDR | GPIO_GPIO | GPIO_OUT, +}; + +static int mx27vis_aic32x4_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + int ret; + u32 dai_format; + + dai_format = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM; + + /* set codec DAI configuration */ + snd_soc_dai_set_fmt(codec_dai, dai_format); + + /* set cpu DAI configuration */ + snd_soc_dai_set_fmt(cpu_dai, dai_format); + + ret = snd_soc_dai_set_sysclk(codec_dai, 0, + 25000000, SND_SOC_CLOCK_OUT); + if (ret) { + pr_err("%s: failed setting codec sysclk\n", __func__); + return ret; + } + + ret = snd_soc_dai_set_sysclk(cpu_dai, IMX_SSP_SYS_CLK, 0, + SND_SOC_CLOCK_IN); + if (ret) { + pr_err("can't set CPU system clock IMX_SSP_SYS_CLK\n"); + return ret; + } + + return 0; +} + +static struct snd_soc_ops mx27vis_aic32x4_snd_ops = { + .hw_params = mx27vis_aic32x4_hw_params, +}; + +static int mx27vis_amp_set(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + int value = ucontrol->value.integer.value[0]; + unsigned int reg = mc->reg; + int max = mc->max; + + if (value > max) + return -EINVAL; + + switch (reg) { + case MX27VIS_AMP_GAIN: + gpio_set_value(MX27VIS_PIN_G0, value & 1); + gpio_set_value(MX27VIS_PIN_G1, value >> 1); + mx27vis_amp_gain = value; + break; + case MX27VIS_AMP_MUTE: + gpio_set_value(MX27VIS_PIN_SDL, value & 1); + gpio_set_value(MX27VIS_PIN_SDR, value >> 1); + mx27vis_amp_mute = value; + break; + } + return 0; +} + +static int mx27vis_amp_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + unsigned int reg = mc->reg; + + switch (reg) { + case MX27VIS_AMP_GAIN: + ucontrol->value.integer.value[0] = mx27vis_amp_gain; + break; + case MX27VIS_AMP_MUTE: + ucontrol->value.integer.value[0] = mx27vis_amp_mute; + break; + } + return 0; +} + +/* From 6dB to 24dB in steps of 6dB */ +static const DECLARE_TLV_DB_SCALE(mx27vis_amp_tlv, 600, 600, 0); + +static const struct snd_kcontrol_new mx27vis_aic32x4_controls[] = { + SOC_DAPM_PIN_SWITCH("External Mic"), + SOC_SINGLE_EXT_TLV("LO Ext Boost", MX27VIS_AMP_GAIN, 0, 3, 0, + mx27vis_amp_get, mx27vis_amp_set, mx27vis_amp_tlv), + SOC_DOUBLE_EXT("LO Ext Mute Switch", MX27VIS_AMP_MUTE, 0, 1, 1, 0, + mx27vis_amp_get, mx27vis_amp_set), +}; + +static const struct snd_soc_dapm_widget aic32x4_dapm_widgets[] = { + SND_SOC_DAPM_MIC("External Mic", NULL), +}; + +static const struct snd_soc_dapm_route aic32x4_dapm_routes[] = { + {"Mic Bias", NULL, "External Mic"}, + {"IN1_R", NULL, "Mic Bias"}, + {"IN2_R", NULL, "Mic Bias"}, + {"IN3_R", NULL, "Mic Bias"}, + {"IN1_L", NULL, "Mic Bias"}, + {"IN2_L", NULL, "Mic Bias"}, + {"IN3_L", NULL, "Mic Bias"}, +}; + +static struct snd_soc_dai_link mx27vis_aic32x4_dai = { + .name = "tlv320aic32x4", + .stream_name = "TLV320AIC32X4", + .codec_dai_name = "tlv320aic32x4-hifi", + .platform_name = "imx-pcm-audio.0", + .codec_name = "tlv320aic32x4.0-0018", + .cpu_dai_name = "imx-ssi.0", + .ops = &mx27vis_aic32x4_snd_ops, +}; + +static struct snd_soc_card mx27vis_aic32x4 = { + .name = "visstrim_m10-audio", + .owner = THIS_MODULE, + .dai_link = &mx27vis_aic32x4_dai, + .num_links = 1, + .controls = mx27vis_aic32x4_controls, + .num_controls = ARRAY_SIZE(mx27vis_aic32x4_controls), + .dapm_widgets = aic32x4_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(aic32x4_dapm_widgets), + .dapm_routes = aic32x4_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(aic32x4_dapm_routes), +}; + +static int __devinit mx27vis_aic32x4_probe(struct platform_device *pdev) +{ + int ret; + + mx27vis_aic32x4.dev = &pdev->dev; + ret = snd_soc_register_card(&mx27vis_aic32x4); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", + ret); + return ret; + } + + /* Connect SSI0 as clock slave to SSI1 external pins */ + imx_audmux_v1_configure_port(MX27_AUDMUX_HPCR1_SSI0, + IMX_AUDMUX_V1_PCR_SYN | + IMX_AUDMUX_V1_PCR_TFSDIR | + IMX_AUDMUX_V1_PCR_TCLKDIR | + IMX_AUDMUX_V1_PCR_TFCSEL(MX27_AUDMUX_PPCR1_SSI_PINS_1) | + IMX_AUDMUX_V1_PCR_RXDSEL(MX27_AUDMUX_PPCR1_SSI_PINS_1) + ); + imx_audmux_v1_configure_port(MX27_AUDMUX_PPCR1_SSI_PINS_1, + IMX_AUDMUX_V1_PCR_SYN | + IMX_AUDMUX_V1_PCR_RXDSEL(MX27_AUDMUX_HPCR1_SSI0) + ); + + ret = mxc_gpio_setup_multiple_pins(mx27vis_amp_pins, + ARRAY_SIZE(mx27vis_amp_pins), "MX27VIS_AMP"); + if (ret) + printk(KERN_ERR "ASoC: unable to setup gpios\n"); + + return ret; +} + +static int __devexit mx27vis_aic32x4_remove(struct platform_device *pdev) +{ + snd_soc_unregister_card(&mx27vis_aic32x4); + + return 0; +} + +static struct platform_driver mx27vis_aic32x4_audio_driver = { + .driver = { + .name = "mx27vis", + .owner = THIS_MODULE, + }, + .probe = mx27vis_aic32x4_probe, + .remove = __devexit_p(mx27vis_aic32x4_remove), +}; + +module_platform_driver(mx27vis_aic32x4_audio_driver); + +MODULE_AUTHOR("Javier Martin "); +MODULE_DESCRIPTION("ALSA SoC AIC32X4 mx27 visstrim"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:mx27vis"); diff --git a/sound/soc/fsl/phycore-ac97.c b/sound/soc/fsl/phycore-ac97.c new file mode 100644 index 000000000000..f8da6dd115ed --- /dev/null +++ b/sound/soc/fsl/phycore-ac97.c @@ -0,0 +1,125 @@ +/* + * phycore-ac97.c -- SoC audio for imx_phycore in AC97 mode + * + * Copyright 2009 Sascha Hauer, Pengutronix + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + */ + +#include +#include +#include +#include +#include +#include +#include +#include + +#include "imx-audmux.h" + +static struct snd_soc_card imx_phycore; + +static struct snd_soc_ops imx_phycore_hifi_ops = { +}; + +static struct snd_soc_dai_link imx_phycore_dai_ac97[] = { + { + .name = "HiFi", + .stream_name = "HiFi", + .codec_dai_name = "wm9712-hifi", + .codec_name = "wm9712-codec", + .cpu_dai_name = "imx-ssi.0", + .platform_name = "imx-fiq-pcm-audio.0", + .ops = &imx_phycore_hifi_ops, + }, +}; + +static struct snd_soc_card imx_phycore = { + .name = "PhyCORE-ac97-audio", + .owner = THIS_MODULE, + .dai_link = imx_phycore_dai_ac97, + .num_links = ARRAY_SIZE(imx_phycore_dai_ac97), +}; + +static struct platform_device *imx_phycore_snd_ac97_device; +static struct platform_device *imx_phycore_snd_device; + +static int __init imx_phycore_init(void) +{ + int ret; + + if (machine_is_pca100()) { + imx_audmux_v1_configure_port(MX27_AUDMUX_HPCR1_SSI0, + IMX_AUDMUX_V1_PCR_SYN | /* 4wire mode */ + IMX_AUDMUX_V1_PCR_TFCSEL(3) | + IMX_AUDMUX_V1_PCR_TCLKDIR | /* clock is output */ + IMX_AUDMUX_V1_PCR_RXDSEL(3)); + imx_audmux_v1_configure_port(3, + IMX_AUDMUX_V1_PCR_SYN | /* 4wire mode */ + IMX_AUDMUX_V1_PCR_TFCSEL(0) | + IMX_AUDMUX_V1_PCR_TFSDIR | + IMX_AUDMUX_V1_PCR_RXDSEL(0)); + } else if (machine_is_pcm043()) { + imx_audmux_v2_configure_port(3, + IMX_AUDMUX_V2_PTCR_SYN | /* 4wire mode */ + IMX_AUDMUX_V2_PTCR_TFSEL(0) | + IMX_AUDMUX_V2_PTCR_TFSDIR, + IMX_AUDMUX_V2_PDCR_RXDSEL(0)); + imx_audmux_v2_configure_port(0, + IMX_AUDMUX_V2_PTCR_SYN | /* 4wire mode */ + IMX_AUDMUX_V2_PTCR_TCSEL(3) | + IMX_AUDMUX_V2_PTCR_TCLKDIR, /* clock is output */ + IMX_AUDMUX_V2_PDCR_RXDSEL(3)); + } else { + /* return happy. We might run on a totally different machine */ + return 0; + } + + imx_phycore_snd_ac97_device = platform_device_alloc("soc-audio", -1); + if (!imx_phycore_snd_ac97_device) + return -ENOMEM; + + platform_set_drvdata(imx_phycore_snd_ac97_device, &imx_phycore); + ret = platform_device_add(imx_phycore_snd_ac97_device); + if (ret) + goto fail1; + + imx_phycore_snd_device = platform_device_alloc("wm9712-codec", -1); + if (!imx_phycore_snd_device) { + ret = -ENOMEM; + goto fail2; + } + ret = platform_device_add(imx_phycore_snd_device); + + if (ret) { + printk(KERN_ERR "ASoC: Platform device allocation failed\n"); + goto fail3; + } + + return 0; + +fail3: + platform_device_put(imx_phycore_snd_device); +fail2: + platform_device_del(imx_phycore_snd_ac97_device); +fail1: + platform_device_put(imx_phycore_snd_ac97_device); + return ret; +} + +static void __exit imx_phycore_exit(void) +{ + platform_device_unregister(imx_phycore_snd_device); + platform_device_unregister(imx_phycore_snd_ac97_device); +} + +late_initcall(imx_phycore_init); +module_exit(imx_phycore_exit); + +MODULE_AUTHOR("Sascha Hauer "); +MODULE_DESCRIPTION("PhyCORE ALSA SoC driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/fsl/wm1133-ev1.c b/sound/soc/fsl/wm1133-ev1.c new file mode 100644 index 000000000000..fe54a69073e5 --- /dev/null +++ b/sound/soc/fsl/wm1133-ev1.c @@ -0,0 +1,304 @@ +/* + * wm1133-ev1.c - Audio for WM1133-EV1 on i.MX31ADS + * + * Copyright (c) 2010 Wolfson Microelectronics plc + * Author: Mark Brown + * + * Based on an earlier driver for the same hardware by Liam Girdwood. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#include +#include +#include +#include +#include +#include +#include +#include + +#include "imx-ssi.h" +#include "../codecs/wm8350.h" +#include "imx-audmux.h" + +/* There is a silicon mic on the board optionally connected via a solder pad + * SP1. Define this to enable it. + */ +#undef USE_SIMIC + +struct _wm8350_audio { + unsigned int channels; + snd_pcm_format_t format; + unsigned int rate; + unsigned int sysclk; + unsigned int bclkdiv; + unsigned int clkdiv; + unsigned int lr_rate; +}; + +/* in order of power consumption per rate (lowest first) */ +static const struct _wm8350_audio wm8350_audio[] = { + /* 16bit mono modes */ + {1, SNDRV_PCM_FORMAT_S16_LE, 8000, 12288000 >> 1, + WM8350_BCLK_DIV_48, WM8350_DACDIV_3, 16,}, + + /* 16 bit stereo modes */ + {2, SNDRV_PCM_FORMAT_S16_LE, 8000, 12288000, + WM8350_BCLK_DIV_48, WM8350_DACDIV_6, 32,}, + {2, SNDRV_PCM_FORMAT_S16_LE, 16000, 12288000, + WM8350_BCLK_DIV_24, WM8350_DACDIV_3, 32,}, + {2, SNDRV_PCM_FORMAT_S16_LE, 32000, 12288000, + WM8350_BCLK_DIV_12, WM8350_DACDIV_1_5, 32,}, + {2, SNDRV_PCM_FORMAT_S16_LE, 48000, 12288000, + WM8350_BCLK_DIV_8, WM8350_DACDIV_1, 32,}, + {2, SNDRV_PCM_FORMAT_S16_LE, 96000, 24576000, + WM8350_BCLK_DIV_8, WM8350_DACDIV_1, 32,}, + {2, SNDRV_PCM_FORMAT_S16_LE, 11025, 11289600, + WM8350_BCLK_DIV_32, WM8350_DACDIV_4, 32,}, + {2, SNDRV_PCM_FORMAT_S16_LE, 22050, 11289600, + WM8350_BCLK_DIV_16, WM8350_DACDIV_2, 32,}, + {2, SNDRV_PCM_FORMAT_S16_LE, 44100, 11289600, + WM8350_BCLK_DIV_8, WM8350_DACDIV_1, 32,}, + {2, SNDRV_PCM_FORMAT_S16_LE, 88200, 22579200, + WM8350_BCLK_DIV_8, WM8350_DACDIV_1, 32,}, + + /* 24bit stereo modes */ + {2, SNDRV_PCM_FORMAT_S24_LE, 48000, 12288000, + WM8350_BCLK_DIV_4, WM8350_DACDIV_1, 64,}, + {2, SNDRV_PCM_FORMAT_S24_LE, 96000, 24576000, + WM8350_BCLK_DIV_4, WM8350_DACDIV_1, 64,}, + {2, SNDRV_PCM_FORMAT_S24_LE, 44100, 11289600, + WM8350_BCLK_DIV_4, WM8350_DACDIV_1, 64,}, + {2, SNDRV_PCM_FORMAT_S24_LE, 88200, 22579200, + WM8350_BCLK_DIV_4, WM8350_DACDIV_1, 64,}, +}; + +static int wm1133_ev1_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + int i, found = 0; + snd_pcm_format_t format = params_format(params); + unsigned int rate = params_rate(params); + unsigned int channels = params_channels(params); + u32 dai_format; + + /* find the correct audio parameters */ + for (i = 0; i < ARRAY_SIZE(wm8350_audio); i++) { + if (rate == wm8350_audio[i].rate && + format == wm8350_audio[i].format && + channels == wm8350_audio[i].channels) { + found = 1; + break; + } + } + if (!found) + return -EINVAL; + + /* codec FLL input is 14.75 MHz from MCLK */ + snd_soc_dai_set_pll(codec_dai, 0, 0, 14750000, wm8350_audio[i].sysclk); + + dai_format = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM; + + /* set codec DAI configuration */ + snd_soc_dai_set_fmt(codec_dai, dai_format); + + /* set cpu DAI configuration */ + snd_soc_dai_set_fmt(cpu_dai, dai_format); + + /* TODO: The SSI driver should figure this out for us */ + switch (channels) { + case 2: + snd_soc_dai_set_tdm_slot(cpu_dai, 0xffffffc, 0xffffffc, 2, 0); + break; + case 1: + snd_soc_dai_set_tdm_slot(cpu_dai, 0xffffffe, 0xffffffe, 1, 0); + break; + default: + return -EINVAL; + } + + /* set MCLK as the codec system clock for DAC and ADC */ + snd_soc_dai_set_sysclk(codec_dai, WM8350_MCLK_SEL_PLL_MCLK, + wm8350_audio[i].sysclk, SND_SOC_CLOCK_IN); + + /* set codec BCLK division for sample rate */ + snd_soc_dai_set_clkdiv(codec_dai, WM8350_BCLK_CLKDIV, + wm8350_audio[i].bclkdiv); + + /* DAI is synchronous and clocked with DAC LRCLK & ADC LRC */ + snd_soc_dai_set_clkdiv(codec_dai, + WM8350_DACLR_CLKDIV, wm8350_audio[i].lr_rate); + snd_soc_dai_set_clkdiv(codec_dai, + WM8350_ADCLR_CLKDIV, wm8350_audio[i].lr_rate); + + /* now configure DAC and ADC clocks */ + snd_soc_dai_set_clkdiv(codec_dai, + WM8350_DAC_CLKDIV, wm8350_audio[i].clkdiv); + + snd_soc_dai_set_clkdiv(codec_dai, + WM8350_ADC_CLKDIV, wm8350_audio[i].clkdiv); + + return 0; +} + +static struct snd_soc_ops wm1133_ev1_ops = { + .hw_params = wm1133_ev1_hw_params, +}; + +static const struct snd_soc_dapm_widget wm1133_ev1_widgets[] = { +#ifdef USE_SIMIC + SND_SOC_DAPM_MIC("SiMIC", NULL), +#endif + SND_SOC_DAPM_MIC("Mic1 Jack", NULL), + SND_SOC_DAPM_MIC("Mic2 Jack", NULL), + SND_SOC_DAPM_LINE("Line In Jack", NULL), + SND_SOC_DAPM_LINE("Line Out Jack", NULL), + SND_SOC_DAPM_HP("Headphone Jack", NULL), +}; + +/* imx32ads soc_card audio map */ +static const struct snd_soc_dapm_route wm1133_ev1_map[] = { + +#ifdef USE_SIMIC + /* SiMIC --> IN1LN (with automatic bias) via SP1 */ + { "IN1LN", NULL, "Mic Bias" }, + { "Mic Bias", NULL, "SiMIC" }, +#endif + + /* Mic 1 Jack --> IN1LN and IN1LP (with automatic bias) */ + { "IN1LN", NULL, "Mic Bias" }, + { "IN1LP", NULL, "Mic1 Jack" }, + { "Mic Bias", NULL, "Mic1 Jack" }, + + /* Mic 2 Jack --> IN1RN and IN1RP (with automatic bias) */ + { "IN1RN", NULL, "Mic Bias" }, + { "IN1RP", NULL, "Mic2 Jack" }, + { "Mic Bias", NULL, "Mic2 Jack" }, + + /* Line in Jack --> AUX (L+R) */ + { "IN3R", NULL, "Line In Jack" }, + { "IN3L", NULL, "Line In Jack" }, + + /* Out1 --> Headphone Jack */ + { "Headphone Jack", NULL, "OUT1R" }, + { "Headphone Jack", NULL, "OUT1L" }, + + /* Out1 --> Line Out Jack */ + { "Line Out Jack", NULL, "OUT2R" }, + { "Line Out Jack", NULL, "OUT2L" }, +}; + +static struct snd_soc_jack hp_jack; + +static struct snd_soc_jack_pin hp_jack_pins[] = { + { .pin = "Headphone Jack", .mask = SND_JACK_HEADPHONE }, +}; + +static struct snd_soc_jack mic_jack; + +static struct snd_soc_jack_pin mic_jack_pins[] = { + { .pin = "Mic1 Jack", .mask = SND_JACK_MICROPHONE }, + { .pin = "Mic2 Jack", .mask = SND_JACK_MICROPHONE }, +}; + +static int wm1133_ev1_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; + + snd_soc_dapm_new_controls(dapm, wm1133_ev1_widgets, + ARRAY_SIZE(wm1133_ev1_widgets)); + + snd_soc_dapm_add_routes(dapm, wm1133_ev1_map, + ARRAY_SIZE(wm1133_ev1_map)); + + /* Headphone jack detection */ + snd_soc_jack_new(codec, "Headphone", SND_JACK_HEADPHONE, &hp_jack); + snd_soc_jack_add_pins(&hp_jack, ARRAY_SIZE(hp_jack_pins), + hp_jack_pins); + wm8350_hp_jack_detect(codec, WM8350_JDR, &hp_jack, SND_JACK_HEADPHONE); + + /* Microphone jack detection */ + snd_soc_jack_new(codec, "Microphone", + SND_JACK_MICROPHONE | SND_JACK_BTN_0, &mic_jack); + snd_soc_jack_add_pins(&mic_jack, ARRAY_SIZE(mic_jack_pins), + mic_jack_pins); + wm8350_mic_jack_detect(codec, &mic_jack, SND_JACK_MICROPHONE, + SND_JACK_BTN_0); + + snd_soc_dapm_force_enable_pin(dapm, "Mic Bias"); + + return 0; +} + + +static struct snd_soc_dai_link wm1133_ev1_dai = { + .name = "WM1133-EV1", + .stream_name = "Audio", + .cpu_dai_name = "imx-ssi.0", + .codec_dai_name = "wm8350-hifi", + .platform_name = "imx-fiq-pcm-audio.0", + .codec_name = "wm8350-codec.0-0x1a", + .init = wm1133_ev1_init, + .ops = &wm1133_ev1_ops, + .symmetric_rates = 1, +}; + +static struct snd_soc_card wm1133_ev1 = { + .name = "WM1133-EV1", + .owner = THIS_MODULE, + .dai_link = &wm1133_ev1_dai, + .num_links = 1, +}; + +static struct platform_device *wm1133_ev1_snd_device; + +static int __init wm1133_ev1_audio_init(void) +{ + int ret; + unsigned int ptcr, pdcr; + + /* SSI0 mastered by port 5 */ + ptcr = IMX_AUDMUX_V2_PTCR_SYN | + IMX_AUDMUX_V2_PTCR_TFSDIR | + IMX_AUDMUX_V2_PTCR_TFSEL(MX31_AUDMUX_PORT5_SSI_PINS_5) | + IMX_AUDMUX_V2_PTCR_TCLKDIR | + IMX_AUDMUX_V2_PTCR_TCSEL(MX31_AUDMUX_PORT5_SSI_PINS_5); + pdcr = IMX_AUDMUX_V2_PDCR_RXDSEL(MX31_AUDMUX_PORT5_SSI_PINS_5); + imx_audmux_v2_configure_port(MX31_AUDMUX_PORT1_SSI0, ptcr, pdcr); + + ptcr = IMX_AUDMUX_V2_PTCR_SYN; + pdcr = IMX_AUDMUX_V2_PDCR_RXDSEL(MX31_AUDMUX_PORT1_SSI0); + imx_audmux_v2_configure_port(MX31_AUDMUX_PORT5_SSI_PINS_5, ptcr, pdcr); + + wm1133_ev1_snd_device = platform_device_alloc("soc-audio", -1); + if (!wm1133_ev1_snd_device) + return -ENOMEM; + + platform_set_drvdata(wm1133_ev1_snd_device, &wm1133_ev1); + ret = platform_device_add(wm1133_ev1_snd_device); + + if (ret) + platform_device_put(wm1133_ev1_snd_device); + + return ret; +} +module_init(wm1133_ev1_audio_init); + +static void __exit wm1133_ev1_audio_exit(void) +{ + platform_device_unregister(wm1133_ev1_snd_device); +} +module_exit(wm1133_ev1_audio_exit); + +MODULE_AUTHOR("Mark Brown "); +MODULE_DESCRIPTION("Audio for WM1133-EV1 on i.MX31ADS"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/imx/Kconfig b/sound/soc/imx/Kconfig deleted file mode 100644 index 810acaa09009..000000000000 --- a/sound/soc/imx/Kconfig +++ /dev/null @@ -1,79 +0,0 @@ -menuconfig SND_IMX_SOC - tristate "SoC Audio for Freescale i.MX CPUs" - depends on ARCH_MXC - help - Say Y or M if you want to add support for codecs attached to - the i.MX SSI interface. - - -if SND_IMX_SOC - -config SND_SOC_IMX_SSI - tristate - -config SND_SOC_IMX_PCM - tristate - -config SND_MXC_SOC_FIQ - tristate - select FIQ - select SND_SOC_IMX_PCM - -config SND_MXC_SOC_MX2 - select SND_SOC_DMAENGINE_PCM - tristate - select SND_SOC_IMX_PCM - -config SND_SOC_IMX_AUDMUX - tristate - -config SND_MXC_SOC_WM1133_EV1 - tristate "Audio on the the i.MX31ADS with WM1133-EV1 fitted" - depends on MACH_MX31ADS_WM1133_EV1 && EXPERIMENTAL - select SND_SOC_WM8350 - select SND_MXC_SOC_FIQ - select SND_SOC_IMX_AUDMUX - select SND_SOC_IMX_SSI - help - Enable support for audio on the i.MX31ADS with the WM1133-EV1 - PMIC board with WM8835x fitted. - -config SND_SOC_MX27VIS_AIC32X4 - tristate "SoC audio support for Visstrim M10 boards" - depends on MACH_IMX27_VISSTRIM_M10 && I2C - select SND_SOC_TLV320AIC32X4 - select SND_MXC_SOC_MX2 - select SND_SOC_IMX_AUDMUX - select SND_SOC_IMX_SSI - help - Say Y if you want to add support for SoC audio on Visstrim SM10 - board with TLV320AIC32X4 codec. - -config SND_SOC_PHYCORE_AC97 - tristate "SoC Audio support for Phytec phyCORE (and phyCARD) boards" - depends on MACH_PCM043 || MACH_PCA100 - select SND_SOC_AC97_BUS - select SND_SOC_WM9712 - select SND_MXC_SOC_FIQ - select SND_SOC_IMX_AUDMUX - select SND_SOC_IMX_SSI - help - Say Y if you want to add support for SoC audio on Phytec phyCORE - and phyCARD boards in AC97 mode - -config SND_SOC_EUKREA_TLV320 - tristate "Eukrea TLV320" - depends on MACH_EUKREA_MBIMX27_BASEBOARD \ - || MACH_EUKREA_MBIMXSD25_BASEBOARD \ - || MACH_EUKREA_MBIMXSD35_BASEBOARD \ - || MACH_EUKREA_MBIMXSD51_BASEBOARD - depends on I2C - select SND_SOC_TLV320AIC23 - select SND_MXC_SOC_FIQ - select SND_SOC_IMX_AUDMUX - select SND_SOC_IMX_SSI - help - Enable I2S based access to the TLV320AIC23B codec attached - to the SSI interface - -endif # SND_IMX_SOC diff --git a/sound/soc/imx/Makefile b/sound/soc/imx/Makefile deleted file mode 100644 index f5db3e92d0d1..000000000000 --- a/sound/soc/imx/Makefile +++ /dev/null @@ -1,22 +0,0 @@ -# i.MX Platform Support -snd-soc-imx-ssi-objs := imx-ssi.o -snd-soc-imx-audmux-objs := imx-audmux.o - -obj-$(CONFIG_SND_SOC_IMX_SSI) += snd-soc-imx-ssi.o -obj-$(CONFIG_SND_SOC_IMX_AUDMUX) += snd-soc-imx-audmux.o - -obj-$(CONFIG_SND_SOC_IMX_PCM) += snd-soc-imx-pcm.o -snd-soc-imx-pcm-y := imx-pcm.o -snd-soc-imx-pcm-$(CONFIG_SND_MXC_SOC_FIQ) += imx-pcm-fiq.o -snd-soc-imx-pcm-$(CONFIG_SND_MXC_SOC_MX2) += imx-pcm-dma-mx2.o - -# i.MX Machine Support -snd-soc-eukrea-tlv320-objs := eukrea-tlv320.o -snd-soc-phycore-ac97-objs := phycore-ac97.o -snd-soc-mx27vis-aic32x4-objs := mx27vis-aic32x4.o -snd-soc-wm1133-ev1-objs := wm1133-ev1.o - -obj-$(CONFIG_SND_SOC_EUKREA_TLV320) += snd-soc-eukrea-tlv320.o -obj-$(CONFIG_SND_SOC_PHYCORE_AC97) += snd-soc-phycore-ac97.o -obj-$(CONFIG_SND_SOC_MX27VIS_AIC32X4) += snd-soc-mx27vis-aic32x4.o -obj-$(CONFIG_SND_MXC_SOC_WM1133_EV1) += snd-soc-wm1133-ev1.o diff --git a/sound/soc/imx/eukrea-tlv320.c b/sound/soc/imx/eukrea-tlv320.c deleted file mode 100644 index 7d4475cfdb24..000000000000 --- a/sound/soc/imx/eukrea-tlv320.c +++ /dev/null @@ -1,164 +0,0 @@ -/* - * eukrea-tlv320.c -- SoC audio for eukrea_cpuimxXX in I2S mode - * - * Copyright 2010 Eric Bénard, Eukréa Electromatique - * - * based on sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c - * which is Copyright 2009 Simtec Electronics - * and on sound/soc/imx/phycore-ac97.c which is - * Copyright 2009 Sascha Hauer, Pengutronix - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - * - */ - -#include -#include -#include -#include -#include -#include -#include -#include - -#include "../codecs/tlv320aic23.h" -#include "imx-ssi.h" -#include "imx-audmux.h" - -#define CODEC_CLOCK 12000000 - -static int eukrea_tlv320_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - int ret; - - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM); - if (ret) { - pr_err("%s: failed set cpu dai format\n", __func__); - return ret; - } - - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM); - if (ret) { - pr_err("%s: failed set codec dai format\n", __func__); - return ret; - } - - ret = snd_soc_dai_set_sysclk(codec_dai, 0, - CODEC_CLOCK, SND_SOC_CLOCK_OUT); - if (ret) { - pr_err("%s: failed setting codec sysclk\n", __func__); - return ret; - } - snd_soc_dai_set_tdm_slot(cpu_dai, 0xffffffc, 0xffffffc, 2, 0); - - ret = snd_soc_dai_set_sysclk(cpu_dai, IMX_SSP_SYS_CLK, 0, - SND_SOC_CLOCK_IN); - if (ret) { - pr_err("can't set CPU system clock IMX_SSP_SYS_CLK\n"); - return ret; - } - - return 0; -} - -static struct snd_soc_ops eukrea_tlv320_snd_ops = { - .hw_params = eukrea_tlv320_hw_params, -}; - -static struct snd_soc_dai_link eukrea_tlv320_dai = { - .name = "tlv320aic23", - .stream_name = "TLV320AIC23", - .codec_dai_name = "tlv320aic23-hifi", - .platform_name = "imx-fiq-pcm-audio.0", - .codec_name = "tlv320aic23-codec.0-001a", - .cpu_dai_name = "imx-ssi.0", - .ops = &eukrea_tlv320_snd_ops, -}; - -static struct snd_soc_card eukrea_tlv320 = { - .name = "cpuimx-audio", - .owner = THIS_MODULE, - .dai_link = &eukrea_tlv320_dai, - .num_links = 1, -}; - -static struct platform_device *eukrea_tlv320_snd_device; - -static int __init eukrea_tlv320_init(void) -{ - int ret; - int int_port = 0, ext_port; - - if (machine_is_eukrea_cpuimx27()) { - imx_audmux_v1_configure_port(MX27_AUDMUX_HPCR1_SSI0, - IMX_AUDMUX_V1_PCR_SYN | - IMX_AUDMUX_V1_PCR_TFSDIR | - IMX_AUDMUX_V1_PCR_TCLKDIR | - IMX_AUDMUX_V1_PCR_RFSDIR | - IMX_AUDMUX_V1_PCR_RCLKDIR | - IMX_AUDMUX_V1_PCR_TFCSEL(MX27_AUDMUX_HPCR3_SSI_PINS_4) | - IMX_AUDMUX_V1_PCR_RFCSEL(MX27_AUDMUX_HPCR3_SSI_PINS_4) | - IMX_AUDMUX_V1_PCR_RXDSEL(MX27_AUDMUX_HPCR3_SSI_PINS_4) - ); - imx_audmux_v1_configure_port(MX27_AUDMUX_HPCR3_SSI_PINS_4, - IMX_AUDMUX_V1_PCR_SYN | - IMX_AUDMUX_V1_PCR_RXDSEL(MX27_AUDMUX_HPCR1_SSI0) - ); - } else if (machine_is_eukrea_cpuimx25sd() || - machine_is_eukrea_cpuimx35sd() || - machine_is_eukrea_cpuimx51sd()) { - ext_port = machine_is_eukrea_cpuimx25sd() ? 4 : 3; - imx_audmux_v2_configure_port(int_port, - IMX_AUDMUX_V2_PTCR_SYN | - IMX_AUDMUX_V2_PTCR_TFSDIR | - IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) | - IMX_AUDMUX_V2_PTCR_TCLKDIR | - IMX_AUDMUX_V2_PTCR_TCSEL(ext_port), - IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port) - ); - imx_audmux_v2_configure_port(ext_port, - IMX_AUDMUX_V2_PTCR_SYN, - IMX_AUDMUX_V2_PDCR_RXDSEL(int_port) - ); - } else { - /* return happy. We might run on a totally different machine */ - return 0; - } - - eukrea_tlv320_snd_device = platform_device_alloc("soc-audio", -1); - if (!eukrea_tlv320_snd_device) - return -ENOMEM; - - platform_set_drvdata(eukrea_tlv320_snd_device, &eukrea_tlv320); - ret = platform_device_add(eukrea_tlv320_snd_device); - - if (ret) { - printk(KERN_ERR "ASoC: Platform device allocation failed\n"); - platform_device_put(eukrea_tlv320_snd_device); - } - - return ret; -} - -static void __exit eukrea_tlv320_exit(void) -{ - platform_device_unregister(eukrea_tlv320_snd_device); -} - -module_init(eukrea_tlv320_init); -module_exit(eukrea_tlv320_exit); - -MODULE_AUTHOR("Eric Bénard "); -MODULE_DESCRIPTION("CPUIMX ALSA SoC driver"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/imx/imx-audmux.c b/sound/soc/imx/imx-audmux.c deleted file mode 100644 index 601df809a26a..000000000000 --- a/sound/soc/imx/imx-audmux.c +++ /dev/null @@ -1,314 +0,0 @@ -/* - * Copyright 2012 Freescale Semiconductor, Inc. - * Copyright 2012 Linaro Ltd. - * Copyright 2009 Pengutronix, Sascha Hauer - * - * Initial development of this code was funded by - * Phytec Messtechnik GmbH, http://www.phytec.de - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - */ - -#include -#include -#include -#include -#include -#include -#include -#include -#include - -#include "imx-audmux.h" - -#define DRIVER_NAME "imx-audmux" - -static struct clk *audmux_clk; -static void __iomem *audmux_base; - -#define IMX_AUDMUX_V2_PTCR(x) ((x) * 8) -#define IMX_AUDMUX_V2_PDCR(x) ((x) * 8 + 4) - -#ifdef CONFIG_DEBUG_FS -static struct dentry *audmux_debugfs_root; - -static int audmux_open_file(struct inode *inode, struct file *file) -{ - file->private_data = inode->i_private; - return 0; -} - -/* There is an annoying discontinuity in the SSI numbering with regard - * to the Linux number of the devices */ -static const char *audmux_port_string(int port) -{ - switch (port) { - case MX31_AUDMUX_PORT1_SSI0: - return "imx-ssi.0"; - case MX31_AUDMUX_PORT2_SSI1: - return "imx-ssi.1"; - case MX31_AUDMUX_PORT3_SSI_PINS_3: - return "SSI3"; - case MX31_AUDMUX_PORT4_SSI_PINS_4: - return "SSI4"; - case MX31_AUDMUX_PORT5_SSI_PINS_5: - return "SSI5"; - case MX31_AUDMUX_PORT6_SSI_PINS_6: - return "SSI6"; - default: - return "UNKNOWN"; - } -} - -static ssize_t audmux_read_file(struct file *file, char __user *user_buf, - size_t count, loff_t *ppos) -{ - ssize_t ret; - char *buf = kmalloc(PAGE_SIZE, GFP_KERNEL); - int port = (int)file->private_data; - u32 pdcr, ptcr; - - if (!buf) - return -ENOMEM; - - if (audmux_clk) - clk_prepare_enable(audmux_clk); - - ptcr = readl(audmux_base + IMX_AUDMUX_V2_PTCR(port)); - pdcr = readl(audmux_base + IMX_AUDMUX_V2_PDCR(port)); - - if (audmux_clk) - clk_disable_unprepare(audmux_clk); - - ret = snprintf(buf, PAGE_SIZE, "PDCR: %08x\nPTCR: %08x\n", - pdcr, ptcr); - - if (ptcr & IMX_AUDMUX_V2_PTCR_TFSDIR) - ret += snprintf(buf + ret, PAGE_SIZE - ret, - "TxFS output from %s, ", - audmux_port_string((ptcr >> 27) & 0x7)); - else - ret += snprintf(buf + ret, PAGE_SIZE - ret, - "TxFS input, "); - - if (ptcr & IMX_AUDMUX_V2_PTCR_TCLKDIR) - ret += snprintf(buf + ret, PAGE_SIZE - ret, - "TxClk output from %s", - audmux_port_string((ptcr >> 22) & 0x7)); - else - ret += snprintf(buf + ret, PAGE_SIZE - ret, - "TxClk input"); - - ret += snprintf(buf + ret, PAGE_SIZE - ret, "\n"); - - if (ptcr & IMX_AUDMUX_V2_PTCR_SYN) { - ret += snprintf(buf + ret, PAGE_SIZE - ret, - "Port is symmetric"); - } else { - if (ptcr & IMX_AUDMUX_V2_PTCR_RFSDIR) - ret += snprintf(buf + ret, PAGE_SIZE - ret, - "RxFS output from %s, ", - audmux_port_string((ptcr >> 17) & 0x7)); - else - ret += snprintf(buf + ret, PAGE_SIZE - ret, - "RxFS input, "); - - if (ptcr & IMX_AUDMUX_V2_PTCR_RCLKDIR) - ret += snprintf(buf + ret, PAGE_SIZE - ret, - "RxClk output from %s", - audmux_port_string((ptcr >> 12) & 0x7)); - else - ret += snprintf(buf + ret, PAGE_SIZE - ret, - "RxClk input"); - } - - ret += snprintf(buf + ret, PAGE_SIZE - ret, - "\nData received from %s\n", - audmux_port_string((pdcr >> 13) & 0x7)); - - ret = simple_read_from_buffer(user_buf, count, ppos, buf, ret); - - kfree(buf); - - return ret; -} - -static const struct file_operations audmux_debugfs_fops = { - .open = audmux_open_file, - .read = audmux_read_file, - .llseek = default_llseek, -}; - -static void __init audmux_debugfs_init(void) -{ - int i; - char buf[20]; - - audmux_debugfs_root = debugfs_create_dir("audmux", NULL); - if (!audmux_debugfs_root) { - pr_warning("Failed to create AUDMUX debugfs root\n"); - return; - } - - for (i = 1; i < 8; i++) { - snprintf(buf, sizeof(buf), "ssi%d", i); - if (!debugfs_create_file(buf, 0444, audmux_debugfs_root, - (void *)i, &audmux_debugfs_fops)) - pr_warning("Failed to create AUDMUX port %d debugfs file\n", - i); - } -} - -static void __devexit audmux_debugfs_remove(void) -{ - debugfs_remove_recursive(audmux_debugfs_root); -} -#else -static inline void audmux_debugfs_init(void) -{ -} - -static inline void audmux_debugfs_remove(void) -{ -} -#endif - -enum imx_audmux_type { - IMX21_AUDMUX, - IMX31_AUDMUX, -} audmux_type; - -static struct platform_device_id imx_audmux_ids[] = { - { - .name = "imx21-audmux", - .driver_data = IMX21_AUDMUX, - }, { - .name = "imx31-audmux", - .driver_data = IMX31_AUDMUX, - }, { - /* sentinel */ - } -}; -MODULE_DEVICE_TABLE(platform, imx_audmux_ids); - -static const struct of_device_id imx_audmux_dt_ids[] = { - { .compatible = "fsl,imx21-audmux", .data = &imx_audmux_ids[0], }, - { .compatible = "fsl,imx31-audmux", .data = &imx_audmux_ids[1], }, - { /* sentinel */ } -}; -MODULE_DEVICE_TABLE(of, imx_audmux_dt_ids); - -static const uint8_t port_mapping[] = { - 0x0, 0x4, 0x8, 0x10, 0x14, 0x1c, -}; - -int imx_audmux_v1_configure_port(unsigned int port, unsigned int pcr) -{ - if (audmux_type != IMX21_AUDMUX) - return -EINVAL; - - if (!audmux_base) - return -ENOSYS; - - if (port >= ARRAY_SIZE(port_mapping)) - return -EINVAL; - - writel(pcr, audmux_base + port_mapping[port]); - - return 0; -} -EXPORT_SYMBOL_GPL(imx_audmux_v1_configure_port); - -int imx_audmux_v2_configure_port(unsigned int port, unsigned int ptcr, - unsigned int pdcr) -{ - if (audmux_type != IMX31_AUDMUX) - return -EINVAL; - - if (!audmux_base) - return -ENOSYS; - - if (audmux_clk) - clk_prepare_enable(audmux_clk); - - writel(ptcr, audmux_base + IMX_AUDMUX_V2_PTCR(port)); - writel(pdcr, audmux_base + IMX_AUDMUX_V2_PDCR(port)); - - if (audmux_clk) - clk_disable_unprepare(audmux_clk); - - return 0; -} -EXPORT_SYMBOL_GPL(imx_audmux_v2_configure_port); - -static int __devinit imx_audmux_probe(struct platform_device *pdev) -{ - struct resource *res; - const struct of_device_id *of_id = - of_match_device(imx_audmux_dt_ids, &pdev->dev); - - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - audmux_base = devm_request_and_ioremap(&pdev->dev, res); - if (!audmux_base) - return -EADDRNOTAVAIL; - - audmux_clk = clk_get(&pdev->dev, "audmux"); - if (IS_ERR(audmux_clk)) { - dev_dbg(&pdev->dev, "cannot get clock: %ld\n", - PTR_ERR(audmux_clk)); - audmux_clk = NULL; - } - - if (of_id) - pdev->id_entry = of_id->data; - audmux_type = pdev->id_entry->driver_data; - if (audmux_type == IMX31_AUDMUX) - audmux_debugfs_init(); - - return 0; -} - -static int __devexit imx_audmux_remove(struct platform_device *pdev) -{ - if (audmux_type == IMX31_AUDMUX) - audmux_debugfs_remove(); - clk_put(audmux_clk); - - return 0; -} - -static struct platform_driver imx_audmux_driver = { - .probe = imx_audmux_probe, - .remove = __devexit_p(imx_audmux_remove), - .id_table = imx_audmux_ids, - .driver = { - .name = DRIVER_NAME, - .owner = THIS_MODULE, - .of_match_table = imx_audmux_dt_ids, - } -}; - -static int __init imx_audmux_init(void) -{ - return platform_driver_register(&imx_audmux_driver); -} -subsys_initcall(imx_audmux_init); - -static void __exit imx_audmux_exit(void) -{ - platform_driver_unregister(&imx_audmux_driver); -} -module_exit(imx_audmux_exit); - -MODULE_DESCRIPTION("Freescale i.MX AUDMUX driver"); -MODULE_AUTHOR("Sascha Hauer "); -MODULE_LICENSE("GPL v2"); -MODULE_ALIAS("platform:" DRIVER_NAME); diff --git a/sound/soc/imx/imx-audmux.h b/sound/soc/imx/imx-audmux.h deleted file mode 100644 index 04ebbab8d7b9..000000000000 --- a/sound/soc/imx/imx-audmux.h +++ /dev/null @@ -1,60 +0,0 @@ -#ifndef __IMX_AUDMUX_H -#define __IMX_AUDMUX_H - -#define MX27_AUDMUX_HPCR1_SSI0 0 -#define MX27_AUDMUX_HPCR2_SSI1 1 -#define MX27_AUDMUX_HPCR3_SSI_PINS_4 2 -#define MX27_AUDMUX_PPCR1_SSI_PINS_1 3 -#define MX27_AUDMUX_PPCR2_SSI_PINS_2 4 -#define MX27_AUDMUX_PPCR3_SSI_PINS_3 5 - -#define MX31_AUDMUX_PORT1_SSI0 0 -#define MX31_AUDMUX_PORT2_SSI1 1 -#define MX31_AUDMUX_PORT3_SSI_PINS_3 2 -#define MX31_AUDMUX_PORT4_SSI_PINS_4 3 -#define MX31_AUDMUX_PORT5_SSI_PINS_5 4 -#define MX31_AUDMUX_PORT6_SSI_PINS_6 5 - -#define MX51_AUDMUX_PORT1_SSI0 0 -#define MX51_AUDMUX_PORT2_SSI1 1 -#define MX51_AUDMUX_PORT3 2 -#define MX51_AUDMUX_PORT4 3 -#define MX51_AUDMUX_PORT5 4 -#define MX51_AUDMUX_PORT6 5 -#define MX51_AUDMUX_PORT7 6 - -/* Register definitions for the i.MX21/27 Digital Audio Multiplexer */ -#define IMX_AUDMUX_V1_PCR_INMMASK(x) ((x) & 0xff) -#define IMX_AUDMUX_V1_PCR_INMEN (1 << 8) -#define IMX_AUDMUX_V1_PCR_TXRXEN (1 << 10) -#define IMX_AUDMUX_V1_PCR_SYN (1 << 12) -#define IMX_AUDMUX_V1_PCR_RXDSEL(x) (((x) & 0x7) << 13) -#define IMX_AUDMUX_V1_PCR_RFCSEL(x) (((x) & 0xf) << 20) -#define IMX_AUDMUX_V1_PCR_RCLKDIR (1 << 24) -#define IMX_AUDMUX_V1_PCR_RFSDIR (1 << 25) -#define IMX_AUDMUX_V1_PCR_TFCSEL(x) (((x) & 0xf) << 26) -#define IMX_AUDMUX_V1_PCR_TCLKDIR (1 << 30) -#define IMX_AUDMUX_V1_PCR_TFSDIR (1 << 31) - -/* Register definitions for the i.MX25/31/35/51 Digital Audio Multiplexer */ -#define IMX_AUDMUX_V2_PTCR_TFSDIR (1 << 31) -#define IMX_AUDMUX_V2_PTCR_TFSEL(x) (((x) & 0xf) << 27) -#define IMX_AUDMUX_V2_PTCR_TCLKDIR (1 << 26) -#define IMX_AUDMUX_V2_PTCR_TCSEL(x) (((x) & 0xf) << 22) -#define IMX_AUDMUX_V2_PTCR_RFSDIR (1 << 21) -#define IMX_AUDMUX_V2_PTCR_RFSEL(x) (((x) & 0xf) << 17) -#define IMX_AUDMUX_V2_PTCR_RCLKDIR (1 << 16) -#define IMX_AUDMUX_V2_PTCR_RCSEL(x) (((x) & 0xf) << 12) -#define IMX_AUDMUX_V2_PTCR_SYN (1 << 11) - -#define IMX_AUDMUX_V2_PDCR_RXDSEL(x) (((x) & 0x7) << 13) -#define IMX_AUDMUX_V2_PDCR_TXRXEN (1 << 12) -#define IMX_AUDMUX_V2_PDCR_MODE(x) (((x) & 0x3) << 8) -#define IMX_AUDMUX_V2_PDCR_INMMASK(x) ((x) & 0xff) - -int imx_audmux_v1_configure_port(unsigned int port, unsigned int pcr); - -int imx_audmux_v2_configure_port(unsigned int port, unsigned int ptcr, - unsigned int pdcr); - -#endif /* __IMX_AUDMUX_H */ diff --git a/sound/soc/imx/imx-pcm-dma-mx2.c b/sound/soc/imx/imx-pcm-dma-mx2.c deleted file mode 100644 index 6b818de2fc03..000000000000 --- a/sound/soc/imx/imx-pcm-dma-mx2.c +++ /dev/null @@ -1,175 +0,0 @@ -/* - * imx-pcm-dma-mx2.c -- ALSA Soc Audio Layer - * - * Copyright 2009 Sascha Hauer - * - * This code is based on code copyrighted by Freescale, - * Liam Girdwood, Javier Martin and probably others. - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - */ -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include - -#include -#include -#include -#include -#include -#include - -#include - -#include "imx-pcm.h" - -static bool filter(struct dma_chan *chan, void *param) -{ - if (!imx_dma_is_general_purpose(chan)) - return false; - - chan->private = param; - - return true; -} - -static int snd_imx_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct dma_chan *chan = snd_dmaengine_pcm_get_chan(substream); - struct imx_pcm_dma_params *dma_params; - struct dma_slave_config slave_config; - int ret; - - dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); - - ret = snd_hwparams_to_dma_slave_config(substream, params, &slave_config); - if (ret) - return ret; - - slave_config.device_fc = false; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - slave_config.dst_addr = dma_params->dma_addr; - slave_config.dst_maxburst = dma_params->burstsize; - } else { - slave_config.src_addr = dma_params->dma_addr; - slave_config.src_maxburst = dma_params->burstsize; - } - - ret = dmaengine_slave_config(chan, &slave_config); - if (ret) - return ret; - - snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); - - return 0; -} - -static struct snd_pcm_hardware snd_imx_hardware = { - .info = SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_PAUSE | - SNDRV_PCM_INFO_RESUME, - .formats = SNDRV_PCM_FMTBIT_S16_LE, - .rate_min = 8000, - .channels_min = 2, - .channels_max = 2, - .buffer_bytes_max = IMX_SSI_DMABUF_SIZE, - .period_bytes_min = 128, - .period_bytes_max = 65535, /* Limited by SDMA engine */ - .periods_min = 2, - .periods_max = 255, - .fifo_size = 0, -}; - -static int snd_imx_open(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct imx_pcm_dma_params *dma_params; - struct imx_dma_data *dma_data; - int ret; - - snd_soc_set_runtime_hwparams(substream, &snd_imx_hardware); - - dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); - - dma_data = kzalloc(sizeof(*dma_data), GFP_KERNEL); - dma_data->peripheral_type = IMX_DMATYPE_SSI; - dma_data->priority = DMA_PRIO_HIGH; - dma_data->dma_request = dma_params->dma; - - ret = snd_dmaengine_pcm_open(substream, filter, dma_data); - if (ret) { - kfree(dma_data); - return 0; - } - - snd_dmaengine_pcm_set_data(substream, dma_data); - - return 0; -} - -static int snd_imx_close(struct snd_pcm_substream *substream) -{ - struct imx_dma_data *dma_data = snd_dmaengine_pcm_get_data(substream); - - snd_dmaengine_pcm_close(substream); - kfree(dma_data); - - return 0; -} - -static struct snd_pcm_ops imx_pcm_ops = { - .open = snd_imx_open, - .close = snd_imx_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = snd_imx_pcm_hw_params, - .trigger = snd_dmaengine_pcm_trigger, - .pointer = snd_dmaengine_pcm_pointer, - .mmap = snd_imx_pcm_mmap, -}; - -static struct snd_soc_platform_driver imx_soc_platform_mx2 = { - .ops = &imx_pcm_ops, - .pcm_new = imx_pcm_new, - .pcm_free = imx_pcm_free, -}; - -static int __devinit imx_soc_platform_probe(struct platform_device *pdev) -{ - return snd_soc_register_platform(&pdev->dev, &imx_soc_platform_mx2); -} - -static int __devexit imx_soc_platform_remove(struct platform_device *pdev) -{ - snd_soc_unregister_platform(&pdev->dev); - return 0; -} - -static struct platform_driver imx_pcm_driver = { - .driver = { - .name = "imx-pcm-audio", - .owner = THIS_MODULE, - }, - .probe = imx_soc_platform_probe, - .remove = __devexit_p(imx_soc_platform_remove), -}; - -module_platform_driver(imx_pcm_driver); -MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:imx-pcm-audio"); diff --git a/sound/soc/imx/imx-pcm-fiq.c b/sound/soc/imx/imx-pcm-fiq.c deleted file mode 100644 index 456b7d723d66..000000000000 --- a/sound/soc/imx/imx-pcm-fiq.c +++ /dev/null @@ -1,336 +0,0 @@ -/* - * imx-pcm-fiq.c -- ALSA Soc Audio Layer - * - * Copyright 2009 Sascha Hauer - * - * This code is based on code copyrighted by Freescale, - * Liam Girdwood, Javier Martin and probably others. - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - */ -#include -#include -#include -#include -#include -#include -#include -#include -#include - -#include -#include -#include -#include -#include - -#include - -#include - -#include "imx-ssi.h" - -struct imx_pcm_runtime_data { - int period; - int periods; - unsigned long offset; - unsigned long last_offset; - unsigned long size; - struct hrtimer hrt; - int poll_time_ns; - struct snd_pcm_substream *substream; - atomic_t running; -}; - -static enum hrtimer_restart snd_hrtimer_callback(struct hrtimer *hrt) -{ - struct imx_pcm_runtime_data *iprtd = - container_of(hrt, struct imx_pcm_runtime_data, hrt); - struct snd_pcm_substream *substream = iprtd->substream; - struct snd_pcm_runtime *runtime = substream->runtime; - struct pt_regs regs; - unsigned long delta; - - if (!atomic_read(&iprtd->running)) - return HRTIMER_NORESTART; - - get_fiq_regs(®s); - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - iprtd->offset = regs.ARM_r8 & 0xffff; - else - iprtd->offset = regs.ARM_r9 & 0xffff; - - /* How much data have we transferred since the last period report? */ - if (iprtd->offset >= iprtd->last_offset) - delta = iprtd->offset - iprtd->last_offset; - else - delta = runtime->buffer_size + iprtd->offset - - iprtd->last_offset; - - /* If we've transferred at least a period then report it and - * reset our poll time */ - if (delta >= iprtd->period) { - snd_pcm_period_elapsed(substream); - iprtd->last_offset = iprtd->offset; - } - - hrtimer_forward_now(hrt, ns_to_ktime(iprtd->poll_time_ns)); - - return HRTIMER_RESTART; -} - -static struct fiq_handler fh = { - .name = DRV_NAME, -}; - -static int snd_imx_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct imx_pcm_runtime_data *iprtd = runtime->private_data; - - iprtd->size = params_buffer_bytes(params); - iprtd->periods = params_periods(params); - iprtd->period = params_period_bytes(params) ; - iprtd->offset = 0; - iprtd->last_offset = 0; - iprtd->poll_time_ns = 1000000000 / params_rate(params) * - params_period_size(params); - snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); - - return 0; -} - -static int snd_imx_pcm_prepare(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct imx_pcm_runtime_data *iprtd = runtime->private_data; - struct pt_regs regs; - - get_fiq_regs(®s); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - regs.ARM_r8 = (iprtd->period * iprtd->periods - 1) << 16; - else - regs.ARM_r9 = (iprtd->period * iprtd->periods - 1) << 16; - - set_fiq_regs(®s); - - return 0; -} - -static int fiq_enable; -static int imx_pcm_fiq; - -static int snd_imx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct imx_pcm_runtime_data *iprtd = runtime->private_data; - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_RESUME: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - atomic_set(&iprtd->running, 1); - hrtimer_start(&iprtd->hrt, ns_to_ktime(iprtd->poll_time_ns), - HRTIMER_MODE_REL); - if (++fiq_enable == 1) - enable_fiq(imx_pcm_fiq); - - break; - - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - atomic_set(&iprtd->running, 0); - - if (--fiq_enable == 0) - disable_fiq(imx_pcm_fiq); - - break; - default: - return -EINVAL; - } - - return 0; -} - -static snd_pcm_uframes_t snd_imx_pcm_pointer(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct imx_pcm_runtime_data *iprtd = runtime->private_data; - - return bytes_to_frames(substream->runtime, iprtd->offset); -} - -static struct snd_pcm_hardware snd_imx_hardware = { - .info = SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_PAUSE | - SNDRV_PCM_INFO_RESUME, - .formats = SNDRV_PCM_FMTBIT_S16_LE, - .rate_min = 8000, - .channels_min = 2, - .channels_max = 2, - .buffer_bytes_max = IMX_SSI_DMABUF_SIZE, - .period_bytes_min = 128, - .period_bytes_max = 16 * 1024, - .periods_min = 4, - .periods_max = 255, - .fifo_size = 0, -}; - -static int snd_imx_open(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct imx_pcm_runtime_data *iprtd; - int ret; - - iprtd = kzalloc(sizeof(*iprtd), GFP_KERNEL); - if (iprtd == NULL) - return -ENOMEM; - runtime->private_data = iprtd; - - iprtd->substream = substream; - - atomic_set(&iprtd->running, 0); - hrtimer_init(&iprtd->hrt, CLOCK_MONOTONIC, HRTIMER_MODE_REL); - iprtd->hrt.function = snd_hrtimer_callback; - - ret = snd_pcm_hw_constraint_integer(substream->runtime, - SNDRV_PCM_HW_PARAM_PERIODS); - if (ret < 0) { - kfree(iprtd); - return ret; - } - - snd_soc_set_runtime_hwparams(substream, &snd_imx_hardware); - return 0; -} - -static int snd_imx_close(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct imx_pcm_runtime_data *iprtd = runtime->private_data; - - hrtimer_cancel(&iprtd->hrt); - - kfree(iprtd); - - return 0; -} - -static struct snd_pcm_ops imx_pcm_ops = { - .open = snd_imx_open, - .close = snd_imx_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = snd_imx_pcm_hw_params, - .prepare = snd_imx_pcm_prepare, - .trigger = snd_imx_pcm_trigger, - .pointer = snd_imx_pcm_pointer, - .mmap = snd_imx_pcm_mmap, -}; - -static int ssi_irq = 0; - -static int imx_pcm_fiq_new(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_pcm *pcm = rtd->pcm; - struct snd_pcm_substream *substream; - int ret; - - ret = imx_pcm_new(rtd); - if (ret) - return ret; - - substream = pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream; - if (substream) { - struct snd_dma_buffer *buf = &substream->dma_buffer; - - imx_ssi_fiq_tx_buffer = (unsigned long)buf->area; - } - - substream = pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream; - if (substream) { - struct snd_dma_buffer *buf = &substream->dma_buffer; - - imx_ssi_fiq_rx_buffer = (unsigned long)buf->area; - } - - set_fiq_handler(&imx_ssi_fiq_start, - &imx_ssi_fiq_end - &imx_ssi_fiq_start); - - return 0; -} - -static void imx_pcm_fiq_free(struct snd_pcm *pcm) -{ - mxc_set_irq_fiq(ssi_irq, 0); - release_fiq(&fh); - imx_pcm_free(pcm); -} - -static struct snd_soc_platform_driver imx_soc_platform_fiq = { - .ops = &imx_pcm_ops, - .pcm_new = imx_pcm_fiq_new, - .pcm_free = imx_pcm_fiq_free, -}; - -static int __devinit imx_soc_platform_probe(struct platform_device *pdev) -{ - struct imx_ssi *ssi = platform_get_drvdata(pdev); - int ret; - - ret = claim_fiq(&fh); - if (ret) { - dev_err(&pdev->dev, "failed to claim fiq: %d", ret); - return ret; - } - - mxc_set_irq_fiq(ssi->irq, 1); - ssi_irq = ssi->irq; - - imx_pcm_fiq = ssi->irq; - - imx_ssi_fiq_base = (unsigned long)ssi->base; - - ssi->dma_params_tx.burstsize = 4; - ssi->dma_params_rx.burstsize = 6; - - ret = snd_soc_register_platform(&pdev->dev, &imx_soc_platform_fiq); - if (ret) - goto failed_register; - - return 0; - -failed_register: - mxc_set_irq_fiq(ssi_irq, 0); - release_fiq(&fh); - - return ret; -} - -static int __devexit imx_soc_platform_remove(struct platform_device *pdev) -{ - snd_soc_unregister_platform(&pdev->dev); - return 0; -} - -static struct platform_driver imx_pcm_driver = { - .driver = { - .name = "imx-fiq-pcm-audio", - .owner = THIS_MODULE, - }, - - .probe = imx_soc_platform_probe, - .remove = __devexit_p(imx_soc_platform_remove), -}; - -module_platform_driver(imx_pcm_driver); - -MODULE_LICENSE("GPL"); diff --git a/sound/soc/imx/imx-pcm.c b/sound/soc/imx/imx-pcm.c deleted file mode 100644 index 93dc360b1777..000000000000 --- a/sound/soc/imx/imx-pcm.c +++ /dev/null @@ -1,105 +0,0 @@ -/* - * Copyright 2009 Sascha Hauer - * - * This code is based on code copyrighted by Freescale, - * Liam Girdwood, Javier Martin and probably others. - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - */ - -#include -#include -#include -#include -#include "imx-pcm.h" - -int snd_imx_pcm_mmap(struct snd_pcm_substream *substream, - struct vm_area_struct *vma) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - int ret; - - ret = dma_mmap_writecombine(substream->pcm->card->dev, vma, - runtime->dma_area, runtime->dma_addr, runtime->dma_bytes); - - pr_debug("%s: ret: %d %p 0x%08x 0x%08x\n", __func__, ret, - runtime->dma_area, - runtime->dma_addr, - runtime->dma_bytes); - return ret; -} -EXPORT_SYMBOL_GPL(snd_imx_pcm_mmap); - -static int imx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) -{ - struct snd_pcm_substream *substream = pcm->streams[stream].substream; - struct snd_dma_buffer *buf = &substream->dma_buffer; - size_t size = IMX_SSI_DMABUF_SIZE; - - buf->dev.type = SNDRV_DMA_TYPE_DEV; - buf->dev.dev = pcm->card->dev; - buf->private_data = NULL; - buf->area = dma_alloc_writecombine(pcm->card->dev, size, - &buf->addr, GFP_KERNEL); - if (!buf->area) - return -ENOMEM; - buf->bytes = size; - - return 0; -} - -static u64 imx_pcm_dmamask = DMA_BIT_MASK(32); - -int imx_pcm_new(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_card *card = rtd->card->snd_card; - struct snd_pcm *pcm = rtd->pcm; - int ret = 0; - - if (!card->dev->dma_mask) - card->dev->dma_mask = &imx_pcm_dmamask; - if (!card->dev->coherent_dma_mask) - card->dev->coherent_dma_mask = DMA_BIT_MASK(32); - if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { - ret = imx_pcm_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_PLAYBACK); - if (ret) - goto out; - } - - if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { - ret = imx_pcm_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_CAPTURE); - if (ret) - goto out; - } - -out: - return ret; -} -EXPORT_SYMBOL_GPL(imx_pcm_new); - -void imx_pcm_free(struct snd_pcm *pcm) -{ - struct snd_pcm_substream *substream; - struct snd_dma_buffer *buf; - int stream; - - for (stream = 0; stream < 2; stream++) { - substream = pcm->streams[stream].substream; - if (!substream) - continue; - - buf = &substream->dma_buffer; - if (!buf->area) - continue; - - dma_free_writecombine(pcm->card->dev, buf->bytes, - buf->area, buf->addr); - buf->area = NULL; - } -} -EXPORT_SYMBOL_GPL(imx_pcm_free); diff --git a/sound/soc/imx/imx-pcm.h b/sound/soc/imx/imx-pcm.h deleted file mode 100644 index b5f5c3acf34d..000000000000 --- a/sound/soc/imx/imx-pcm.h +++ /dev/null @@ -1,32 +0,0 @@ -/* - * Copyright 2009 Sascha Hauer - * - * This code is based on code copyrighted by Freescale, - * Liam Girdwood, Javier Martin and probably others. - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - */ - -#ifndef _IMX_PCM_H -#define _IMX_PCM_H - -/* - * Do not change this as the FIQ handler depends on this size - */ -#define IMX_SSI_DMABUF_SIZE (64 * 1024) - -struct imx_pcm_dma_params { - int dma; - unsigned long dma_addr; - int burstsize; -}; - -int snd_imx_pcm_mmap(struct snd_pcm_substream *substream, - struct vm_area_struct *vma); -int imx_pcm_new(struct snd_soc_pcm_runtime *rtd); -void imx_pcm_free(struct snd_pcm *pcm); - -#endif /* _IMX_PCM_H */ diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/imx/imx-ssi.c deleted file mode 100644 index 4f81ed456325..000000000000 --- a/sound/soc/imx/imx-ssi.c +++ /dev/null @@ -1,690 +0,0 @@ -/* - * imx-ssi.c -- ALSA Soc Audio Layer - * - * Copyright 2009 Sascha Hauer - * - * This code is based on code copyrighted by Freescale, - * Liam Girdwood, Javier Martin and probably others. - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - * - * - * The i.MX SSI core has some nasty limitations in AC97 mode. While most - * sane processor vendors have a FIFO per AC97 slot, the i.MX has only - * one FIFO which combines all valid receive slots. We cannot even select - * which slots we want to receive. The WM9712 with which this driver - * was developed with always sends GPIO status data in slot 12 which - * we receive in our (PCM-) data stream. The only chance we have is to - * manually skip this data in the FIQ handler. With sampling rates different - * from 48000Hz not every frame has valid receive data, so the ratio - * between pcm data and GPIO status data changes. Our FIQ handler is not - * able to handle this, hence this driver only works with 48000Hz sampling - * rate. - * Reading and writing AC97 registers is another challenge. The core - * provides us status bits when the read register is updated with *another* - * value. When we read the same register two times (and the register still - * contains the same value) these status bits are not set. We work - * around this by not polling these bits but only wait a fixed delay. - * - */ - -#include -#include -#include -#include -#include -#include -#include -#include -#include - -#include -#include -#include -#include -#include - -#include -#include - -#include "imx-ssi.h" - -#define SSI_SACNT_DEFAULT (SSI_SACNT_AC97EN | SSI_SACNT_FV) - -/* - * SSI Network Mode or TDM slots configuration. - * Should only be called when port is inactive (i.e. SSIEN = 0). - */ -static int imx_ssi_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai, - unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width) -{ - struct imx_ssi *ssi = snd_soc_dai_get_drvdata(cpu_dai); - u32 sccr; - - sccr = readl(ssi->base + SSI_STCCR); - sccr &= ~SSI_STCCR_DC_MASK; - sccr |= SSI_STCCR_DC(slots - 1); - writel(sccr, ssi->base + SSI_STCCR); - - sccr = readl(ssi->base + SSI_SRCCR); - sccr &= ~SSI_STCCR_DC_MASK; - sccr |= SSI_STCCR_DC(slots - 1); - writel(sccr, ssi->base + SSI_SRCCR); - - writel(tx_mask, ssi->base + SSI_STMSK); - writel(rx_mask, ssi->base + SSI_SRMSK); - - return 0; -} - -/* - * SSI DAI format configuration. - * Should only be called when port is inactive (i.e. SSIEN = 0). - */ -static int imx_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) -{ - struct imx_ssi *ssi = snd_soc_dai_get_drvdata(cpu_dai); - u32 strcr = 0, scr; - - scr = readl(ssi->base + SSI_SCR) & ~(SSI_SCR_SYN | SSI_SCR_NET); - - /* DAI mode */ - switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { - case SND_SOC_DAIFMT_I2S: - /* data on rising edge of bclk, frame low 1clk before data */ - strcr |= SSI_STCR_TFSI | SSI_STCR_TEFS | SSI_STCR_TXBIT0; - scr |= SSI_SCR_NET; - if (ssi->flags & IMX_SSI_USE_I2S_SLAVE) { - scr &= ~SSI_I2S_MODE_MASK; - scr |= SSI_SCR_I2S_MODE_SLAVE; - } - break; - case SND_SOC_DAIFMT_LEFT_J: - /* data on rising edge of bclk, frame high with data */ - strcr |= SSI_STCR_TXBIT0; - break; - case SND_SOC_DAIFMT_DSP_B: - /* data on rising edge of bclk, frame high with data */ - strcr |= SSI_STCR_TFSL | SSI_STCR_TXBIT0; - break; - case SND_SOC_DAIFMT_DSP_A: - /* data on rising edge of bclk, frame high 1clk before data */ - strcr |= SSI_STCR_TFSL | SSI_STCR_TXBIT0 | SSI_STCR_TEFS; - break; - } - - /* DAI clock inversion */ - switch (fmt & SND_SOC_DAIFMT_INV_MASK) { - case SND_SOC_DAIFMT_IB_IF: - strcr |= SSI_STCR_TFSI; - strcr &= ~SSI_STCR_TSCKP; - break; - case SND_SOC_DAIFMT_IB_NF: - strcr &= ~(SSI_STCR_TSCKP | SSI_STCR_TFSI); - break; - case SND_SOC_DAIFMT_NB_IF: - strcr |= SSI_STCR_TFSI | SSI_STCR_TSCKP; - break; - case SND_SOC_DAIFMT_NB_NF: - strcr &= ~SSI_STCR_TFSI; - strcr |= SSI_STCR_TSCKP; - break; - } - - /* DAI clock master masks */ - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: - break; - default: - /* Master mode not implemented, needs handling of clocks. */ - return -EINVAL; - } - - strcr |= SSI_STCR_TFEN0; - - if (ssi->flags & IMX_SSI_NET) - scr |= SSI_SCR_NET; - if (ssi->flags & IMX_SSI_SYN) - scr |= SSI_SCR_SYN; - - writel(strcr, ssi->base + SSI_STCR); - writel(strcr, ssi->base + SSI_SRCR); - writel(scr, ssi->base + SSI_SCR); - - return 0; -} - -/* - * SSI system clock configuration. - * Should only be called when port is inactive (i.e. SSIEN = 0). - */ -static int imx_ssi_set_dai_sysclk(struct snd_soc_dai *cpu_dai, - int clk_id, unsigned int freq, int dir) -{ - struct imx_ssi *ssi = snd_soc_dai_get_drvdata(cpu_dai); - u32 scr; - - scr = readl(ssi->base + SSI_SCR); - - switch (clk_id) { - case IMX_SSP_SYS_CLK: - if (dir == SND_SOC_CLOCK_OUT) - scr |= SSI_SCR_SYS_CLK_EN; - else - scr &= ~SSI_SCR_SYS_CLK_EN; - break; - default: - return -EINVAL; - } - - writel(scr, ssi->base + SSI_SCR); - - return 0; -} - -/* - * SSI Clock dividers - * Should only be called when port is inactive (i.e. SSIEN = 0). - */ -static int imx_ssi_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, - int div_id, int div) -{ - struct imx_ssi *ssi = snd_soc_dai_get_drvdata(cpu_dai); - u32 stccr, srccr; - - stccr = readl(ssi->base + SSI_STCCR); - srccr = readl(ssi->base + SSI_SRCCR); - - switch (div_id) { - case IMX_SSI_TX_DIV_2: - stccr &= ~SSI_STCCR_DIV2; - stccr |= div; - break; - case IMX_SSI_TX_DIV_PSR: - stccr &= ~SSI_STCCR_PSR; - stccr |= div; - break; - case IMX_SSI_TX_DIV_PM: - stccr &= ~0xff; - stccr |= SSI_STCCR_PM(div); - break; - case IMX_SSI_RX_DIV_2: - stccr &= ~SSI_STCCR_DIV2; - stccr |= div; - break; - case IMX_SSI_RX_DIV_PSR: - stccr &= ~SSI_STCCR_PSR; - stccr |= div; - break; - case IMX_SSI_RX_DIV_PM: - stccr &= ~0xff; - stccr |= SSI_STCCR_PM(div); - break; - default: - return -EINVAL; - } - - writel(stccr, ssi->base + SSI_STCCR); - writel(srccr, ssi->base + SSI_SRCCR); - - return 0; -} - -static int imx_ssi_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *cpu_dai) -{ - struct imx_ssi *ssi = snd_soc_dai_get_drvdata(cpu_dai); - struct imx_pcm_dma_params *dma_data; - - /* Tx/Rx config */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - dma_data = &ssi->dma_params_tx; - else - dma_data = &ssi->dma_params_rx; - - snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data); - - return 0; -} - -/* - * Should only be called when port is inactive (i.e. SSIEN = 0), - * although can be called multiple times by upper layers. - */ -static int imx_ssi_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *cpu_dai) -{ - struct imx_ssi *ssi = snd_soc_dai_get_drvdata(cpu_dai); - u32 reg, sccr; - - /* Tx/Rx config */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - reg = SSI_STCCR; - else - reg = SSI_SRCCR; - - if (ssi->flags & IMX_SSI_SYN) - reg = SSI_STCCR; - - sccr = readl(ssi->base + reg) & ~SSI_STCCR_WL_MASK; - - /* DAI data (word) size */ - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: - sccr |= SSI_SRCCR_WL(16); - break; - case SNDRV_PCM_FORMAT_S20_3LE: - sccr |= SSI_SRCCR_WL(20); - break; - case SNDRV_PCM_FORMAT_S24_LE: - sccr |= SSI_SRCCR_WL(24); - break; - } - - writel(sccr, ssi->base + reg); - - return 0; -} - -static int imx_ssi_trigger(struct snd_pcm_substream *substream, int cmd, - struct snd_soc_dai *dai) -{ - struct imx_ssi *ssi = snd_soc_dai_get_drvdata(dai); - unsigned int sier_bits, sier; - unsigned int scr; - - scr = readl(ssi->base + SSI_SCR); - sier = readl(ssi->base + SSI_SIER); - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - if (ssi->flags & IMX_SSI_DMA) - sier_bits = SSI_SIER_TDMAE; - else - sier_bits = SSI_SIER_TIE | SSI_SIER_TFE0_EN; - } else { - if (ssi->flags & IMX_SSI_DMA) - sier_bits = SSI_SIER_RDMAE; - else - sier_bits = SSI_SIER_RIE | SSI_SIER_RFF0_EN; - } - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_RESUME: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - scr |= SSI_SCR_TE; - else - scr |= SSI_SCR_RE; - sier |= sier_bits; - - if (++ssi->enabled == 1) - scr |= SSI_SCR_SSIEN; - - break; - - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - scr &= ~SSI_SCR_TE; - else - scr &= ~SSI_SCR_RE; - sier &= ~sier_bits; - - if (--ssi->enabled == 0) - scr &= ~SSI_SCR_SSIEN; - - break; - default: - return -EINVAL; - } - - if (!(ssi->flags & IMX_SSI_USE_AC97)) - /* rx/tx are always enabled to access ac97 registers */ - writel(scr, ssi->base + SSI_SCR); - - writel(sier, ssi->base + SSI_SIER); - - return 0; -} - -static const struct snd_soc_dai_ops imx_ssi_pcm_dai_ops = { - .startup = imx_ssi_startup, - .hw_params = imx_ssi_hw_params, - .set_fmt = imx_ssi_set_dai_fmt, - .set_clkdiv = imx_ssi_set_dai_clkdiv, - .set_sysclk = imx_ssi_set_dai_sysclk, - .set_tdm_slot = imx_ssi_set_dai_tdm_slot, - .trigger = imx_ssi_trigger, -}; - -static int imx_ssi_dai_probe(struct snd_soc_dai *dai) -{ - struct imx_ssi *ssi = dev_get_drvdata(dai->dev); - uint32_t val; - - snd_soc_dai_set_drvdata(dai, ssi); - - val = SSI_SFCSR_TFWM0(ssi->dma_params_tx.burstsize) | - SSI_SFCSR_RFWM0(ssi->dma_params_rx.burstsize); - writel(val, ssi->base + SSI_SFCSR); - - return 0; -} - -static struct snd_soc_dai_driver imx_ssi_dai = { - .probe = imx_ssi_dai_probe, - .playback = { - .channels_min = 1, - .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_96000, - .formats = SNDRV_PCM_FMTBIT_S16_LE, - }, - .capture = { - .channels_min = 1, - .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_96000, - .formats = SNDRV_PCM_FMTBIT_S16_LE, - }, - .ops = &imx_ssi_pcm_dai_ops, -}; - -static struct snd_soc_dai_driver imx_ac97_dai = { - .probe = imx_ssi_dai_probe, - .ac97_control = 1, - .playback = { - .stream_name = "AC97 Playback", - .channels_min = 2, - .channels_max = 2, - .rates = SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S16_LE, - }, - .capture = { - .stream_name = "AC97 Capture", - .channels_min = 2, - .channels_max = 2, - .rates = SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S16_LE, - }, - .ops = &imx_ssi_pcm_dai_ops, -}; - -static void setup_channel_to_ac97(struct imx_ssi *imx_ssi) -{ - void __iomem *base = imx_ssi->base; - - writel(0x0, base + SSI_SCR); - writel(0x0, base + SSI_STCR); - writel(0x0, base + SSI_SRCR); - - writel(SSI_SCR_SYN | SSI_SCR_NET, base + SSI_SCR); - - writel(SSI_SFCSR_RFWM0(8) | - SSI_SFCSR_TFWM0(8) | - SSI_SFCSR_RFWM1(8) | - SSI_SFCSR_TFWM1(8), base + SSI_SFCSR); - - writel(SSI_STCCR_WL(16) | SSI_STCCR_DC(12), base + SSI_STCCR); - writel(SSI_STCCR_WL(16) | SSI_STCCR_DC(12), base + SSI_SRCCR); - - writel(SSI_SCR_SYN | SSI_SCR_NET | SSI_SCR_SSIEN, base + SSI_SCR); - writel(SSI_SOR_WAIT(3), base + SSI_SOR); - - writel(SSI_SCR_SYN | SSI_SCR_NET | SSI_SCR_SSIEN | - SSI_SCR_TE | SSI_SCR_RE, - base + SSI_SCR); - - writel(SSI_SACNT_DEFAULT, base + SSI_SACNT); - writel(0xff, base + SSI_SACCDIS); - writel(0x300, base + SSI_SACCEN); -} - -static struct imx_ssi *ac97_ssi; - -static void imx_ssi_ac97_write(struct snd_ac97 *ac97, unsigned short reg, - unsigned short val) -{ - struct imx_ssi *imx_ssi = ac97_ssi; - void __iomem *base = imx_ssi->base; - unsigned int lreg; - unsigned int lval; - - if (reg > 0x7f) - return; - - pr_debug("%s: 0x%02x 0x%04x\n", __func__, reg, val); - - lreg = reg << 12; - writel(lreg, base + SSI_SACADD); - - lval = val << 4; - writel(lval , base + SSI_SACDAT); - - writel(SSI_SACNT_DEFAULT | SSI_SACNT_WR, base + SSI_SACNT); - udelay(100); -} - -static unsigned short imx_ssi_ac97_read(struct snd_ac97 *ac97, - unsigned short reg) -{ - struct imx_ssi *imx_ssi = ac97_ssi; - void __iomem *base = imx_ssi->base; - - unsigned short val = -1; - unsigned int lreg; - - lreg = (reg & 0x7f) << 12 ; - writel(lreg, base + SSI_SACADD); - writel(SSI_SACNT_DEFAULT | SSI_SACNT_RD, base + SSI_SACNT); - - udelay(100); - - val = (readl(base + SSI_SACDAT) >> 4) & 0xffff; - - pr_debug("%s: 0x%02x 0x%04x\n", __func__, reg, val); - - return val; -} - -static void imx_ssi_ac97_reset(struct snd_ac97 *ac97) -{ - struct imx_ssi *imx_ssi = ac97_ssi; - - if (imx_ssi->ac97_reset) - imx_ssi->ac97_reset(ac97); -} - -static void imx_ssi_ac97_warm_reset(struct snd_ac97 *ac97) -{ - struct imx_ssi *imx_ssi = ac97_ssi; - - if (imx_ssi->ac97_warm_reset) - imx_ssi->ac97_warm_reset(ac97); -} - -struct snd_ac97_bus_ops soc_ac97_ops = { - .read = imx_ssi_ac97_read, - .write = imx_ssi_ac97_write, - .reset = imx_ssi_ac97_reset, - .warm_reset = imx_ssi_ac97_warm_reset -}; -EXPORT_SYMBOL_GPL(soc_ac97_ops); - -static int imx_ssi_probe(struct platform_device *pdev) -{ - struct resource *res; - struct imx_ssi *ssi; - struct imx_ssi_platform_data *pdata = pdev->dev.platform_data; - int ret = 0; - struct snd_soc_dai_driver *dai; - - ssi = kzalloc(sizeof(*ssi), GFP_KERNEL); - if (!ssi) - return -ENOMEM; - dev_set_drvdata(&pdev->dev, ssi); - - if (pdata) { - ssi->ac97_reset = pdata->ac97_reset; - ssi->ac97_warm_reset = pdata->ac97_warm_reset; - ssi->flags = pdata->flags; - } - - ssi->irq = platform_get_irq(pdev, 0); - - ssi->clk = clk_get(&pdev->dev, NULL); - if (IS_ERR(ssi->clk)) { - ret = PTR_ERR(ssi->clk); - dev_err(&pdev->dev, "Cannot get the clock: %d\n", - ret); - goto failed_clk; - } - clk_enable(ssi->clk); - - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!res) { - ret = -ENODEV; - goto failed_get_resource; - } - - if (!request_mem_region(res->start, resource_size(res), DRV_NAME)) { - dev_err(&pdev->dev, "request_mem_region failed\n"); - ret = -EBUSY; - goto failed_get_resource; - } - - ssi->base = ioremap(res->start, resource_size(res)); - if (!ssi->base) { - dev_err(&pdev->dev, "ioremap failed\n"); - ret = -ENODEV; - goto failed_ioremap; - } - - if (ssi->flags & IMX_SSI_USE_AC97) { - if (ac97_ssi) { - ret = -EBUSY; - goto failed_ac97; - } - ac97_ssi = ssi; - setup_channel_to_ac97(ssi); - dai = &imx_ac97_dai; - } else - dai = &imx_ssi_dai; - - writel(0x0, ssi->base + SSI_SIER); - - ssi->dma_params_rx.dma_addr = res->start + SSI_SRX0; - ssi->dma_params_tx.dma_addr = res->start + SSI_STX0; - - ssi->dma_params_tx.burstsize = 6; - ssi->dma_params_rx.burstsize = 4; - - res = platform_get_resource_byname(pdev, IORESOURCE_DMA, "tx0"); - if (res) - ssi->dma_params_tx.dma = res->start; - - res = platform_get_resource_byname(pdev, IORESOURCE_DMA, "rx0"); - if (res) - ssi->dma_params_rx.dma = res->start; - - platform_set_drvdata(pdev, ssi); - - ret = snd_soc_register_dai(&pdev->dev, dai); - if (ret) { - dev_err(&pdev->dev, "register DAI failed\n"); - goto failed_register; - } - - ssi->soc_platform_pdev_fiq = platform_device_alloc("imx-fiq-pcm-audio", pdev->id); - if (!ssi->soc_platform_pdev_fiq) { - ret = -ENOMEM; - goto failed_pdev_fiq_alloc; - } - - platform_set_drvdata(ssi->soc_platform_pdev_fiq, ssi); - ret = platform_device_add(ssi->soc_platform_pdev_fiq); - if (ret) { - dev_err(&pdev->dev, "failed to add platform device\n"); - goto failed_pdev_fiq_add; - } - - ssi->soc_platform_pdev = platform_device_alloc("imx-pcm-audio", pdev->id); - if (!ssi->soc_platform_pdev) { - ret = -ENOMEM; - goto failed_pdev_alloc; - } - - platform_set_drvdata(ssi->soc_platform_pdev, ssi); - ret = platform_device_add(ssi->soc_platform_pdev); - if (ret) { - dev_err(&pdev->dev, "failed to add platform device\n"); - goto failed_pdev_add; - } - - return 0; - -failed_pdev_add: - platform_device_put(ssi->soc_platform_pdev); -failed_pdev_alloc: - platform_device_del(ssi->soc_platform_pdev_fiq); -failed_pdev_fiq_add: - platform_device_put(ssi->soc_platform_pdev_fiq); -failed_pdev_fiq_alloc: - snd_soc_unregister_dai(&pdev->dev); -failed_register: -failed_ac97: - iounmap(ssi->base); -failed_ioremap: - release_mem_region(res->start, resource_size(res)); -failed_get_resource: - clk_disable(ssi->clk); - clk_put(ssi->clk); -failed_clk: - kfree(ssi); - - return ret; -} - -static int __devexit imx_ssi_remove(struct platform_device *pdev) -{ - struct resource *res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - struct imx_ssi *ssi = platform_get_drvdata(pdev); - - platform_device_unregister(ssi->soc_platform_pdev); - platform_device_unregister(ssi->soc_platform_pdev_fiq); - - snd_soc_unregister_dai(&pdev->dev); - - if (ssi->flags & IMX_SSI_USE_AC97) - ac97_ssi = NULL; - - iounmap(ssi->base); - release_mem_region(res->start, resource_size(res)); - clk_disable(ssi->clk); - clk_put(ssi->clk); - kfree(ssi); - - return 0; -} - -static struct platform_driver imx_ssi_driver = { - .probe = imx_ssi_probe, - .remove = __devexit_p(imx_ssi_remove), - - .driver = { - .name = "imx-ssi", - .owner = THIS_MODULE, - }, -}; - -module_platform_driver(imx_ssi_driver); - -/* Module information */ -MODULE_AUTHOR("Sascha Hauer, "); -MODULE_DESCRIPTION("i.MX I2S/ac97 SoC Interface"); -MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:imx-ssi"); diff --git a/sound/soc/imx/imx-ssi.h b/sound/soc/imx/imx-ssi.h deleted file mode 100644 index 5744e86ca878..000000000000 --- a/sound/soc/imx/imx-ssi.h +++ /dev/null @@ -1,216 +0,0 @@ -/* - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#ifndef _IMX_SSI_H -#define _IMX_SSI_H - -#define SSI_STX0 0x00 -#define SSI_STX1 0x04 -#define SSI_SRX0 0x08 -#define SSI_SRX1 0x0c - -#define SSI_SCR 0x10 -#define SSI_SCR_CLK_IST (1 << 9) -#define SSI_SCR_CLK_IST_SHIFT 9 -#define SSI_SCR_TCH_EN (1 << 8) -#define SSI_SCR_SYS_CLK_EN (1 << 7) -#define SSI_SCR_I2S_MODE_NORM (0 << 5) -#define SSI_SCR_I2S_MODE_MSTR (1 << 5) -#define SSI_SCR_I2S_MODE_SLAVE (2 << 5) -#define SSI_I2S_MODE_MASK (3 << 5) -#define SSI_SCR_SYN (1 << 4) -#define SSI_SCR_NET (1 << 3) -#define SSI_SCR_RE (1 << 2) -#define SSI_SCR_TE (1 << 1) -#define SSI_SCR_SSIEN (1 << 0) - -#define SSI_SISR 0x14 -#define SSI_SISR_MASK ((1 << 19) - 1) -#define SSI_SISR_CMDAU (1 << 18) -#define SSI_SISR_CMDDU (1 << 17) -#define SSI_SISR_RXT (1 << 16) -#define SSI_SISR_RDR1 (1 << 15) -#define SSI_SISR_RDR0 (1 << 14) -#define SSI_SISR_TDE1 (1 << 13) -#define SSI_SISR_TDE0 (1 << 12) -#define SSI_SISR_ROE1 (1 << 11) -#define SSI_SISR_ROE0 (1 << 10) -#define SSI_SISR_TUE1 (1 << 9) -#define SSI_SISR_TUE0 (1 << 8) -#define SSI_SISR_TFS (1 << 7) -#define SSI_SISR_RFS (1 << 6) -#define SSI_SISR_TLS (1 << 5) -#define SSI_SISR_RLS (1 << 4) -#define SSI_SISR_RFF1 (1 << 3) -#define SSI_SISR_RFF0 (1 << 2) -#define SSI_SISR_TFE1 (1 << 1) -#define SSI_SISR_TFE0 (1 << 0) - -#define SSI_SIER 0x18 -#define SSI_SIER_RDMAE (1 << 22) -#define SSI_SIER_RIE (1 << 21) -#define SSI_SIER_TDMAE (1 << 20) -#define SSI_SIER_TIE (1 << 19) -#define SSI_SIER_CMDAU_EN (1 << 18) -#define SSI_SIER_CMDDU_EN (1 << 17) -#define SSI_SIER_RXT_EN (1 << 16) -#define SSI_SIER_RDR1_EN (1 << 15) -#define SSI_SIER_RDR0_EN (1 << 14) -#define SSI_SIER_TDE1_EN (1 << 13) -#define SSI_SIER_TDE0_EN (1 << 12) -#define SSI_SIER_ROE1_EN (1 << 11) -#define SSI_SIER_ROE0_EN (1 << 10) -#define SSI_SIER_TUE1_EN (1 << 9) -#define SSI_SIER_TUE0_EN (1 << 8) -#define SSI_SIER_TFS_EN (1 << 7) -#define SSI_SIER_RFS_EN (1 << 6) -#define SSI_SIER_TLS_EN (1 << 5) -#define SSI_SIER_RLS_EN (1 << 4) -#define SSI_SIER_RFF1_EN (1 << 3) -#define SSI_SIER_RFF0_EN (1 << 2) -#define SSI_SIER_TFE1_EN (1 << 1) -#define SSI_SIER_TFE0_EN (1 << 0) - -#define SSI_STCR 0x1c -#define SSI_STCR_TXBIT0 (1 << 9) -#define SSI_STCR_TFEN1 (1 << 8) -#define SSI_STCR_TFEN0 (1 << 7) -#define SSI_FIFO_ENABLE_0_SHIFT 7 -#define SSI_STCR_TFDIR (1 << 6) -#define SSI_STCR_TXDIR (1 << 5) -#define SSI_STCR_TSHFD (1 << 4) -#define SSI_STCR_TSCKP (1 << 3) -#define SSI_STCR_TFSI (1 << 2) -#define SSI_STCR_TFSL (1 << 1) -#define SSI_STCR_TEFS (1 << 0) - -#define SSI_SRCR 0x20 -#define SSI_SRCR_RXBIT0 (1 << 9) -#define SSI_SRCR_RFEN1 (1 << 8) -#define SSI_SRCR_RFEN0 (1 << 7) -#define SSI_FIFO_ENABLE_0_SHIFT 7 -#define SSI_SRCR_RFDIR (1 << 6) -#define SSI_SRCR_RXDIR (1 << 5) -#define SSI_SRCR_RSHFD (1 << 4) -#define SSI_SRCR_RSCKP (1 << 3) -#define SSI_SRCR_RFSI (1 << 2) -#define SSI_SRCR_RFSL (1 << 1) -#define SSI_SRCR_REFS (1 << 0) - -#define SSI_SRCCR 0x28 -#define SSI_SRCCR_DIV2 (1 << 18) -#define SSI_SRCCR_PSR (1 << 17) -#define SSI_SRCCR_WL(x) ((((x) - 2) >> 1) << 13) -#define SSI_SRCCR_DC(x) (((x) & 0x1f) << 8) -#define SSI_SRCCR_PM(x) (((x) & 0xff) << 0) -#define SSI_SRCCR_WL_MASK (0xf << 13) -#define SSI_SRCCR_DC_MASK (0x1f << 8) -#define SSI_SRCCR_PM_MASK (0xff << 0) - -#define SSI_STCCR 0x24 -#define SSI_STCCR_DIV2 (1 << 18) -#define SSI_STCCR_PSR (1 << 17) -#define SSI_STCCR_WL(x) ((((x) - 2) >> 1) << 13) -#define SSI_STCCR_DC(x) (((x) & 0x1f) << 8) -#define SSI_STCCR_PM(x) (((x) & 0xff) << 0) -#define SSI_STCCR_WL_MASK (0xf << 13) -#define SSI_STCCR_DC_MASK (0x1f << 8) -#define SSI_STCCR_PM_MASK (0xff << 0) - -#define SSI_SFCSR 0x2c -#define SSI_SFCSR_RFCNT1(x) (((x) & 0xf) << 28) -#define SSI_RX_FIFO_1_COUNT_SHIFT 28 -#define SSI_SFCSR_TFCNT1(x) (((x) & 0xf) << 24) -#define SSI_TX_FIFO_1_COUNT_SHIFT 24 -#define SSI_SFCSR_RFWM1(x) (((x) & 0xf) << 20) -#define SSI_SFCSR_TFWM1(x) (((x) & 0xf) << 16) -#define SSI_SFCSR_RFCNT0(x) (((x) & 0xf) << 12) -#define SSI_RX_FIFO_0_COUNT_SHIFT 12 -#define SSI_SFCSR_TFCNT0(x) (((x) & 0xf) << 8) -#define SSI_TX_FIFO_0_COUNT_SHIFT 8 -#define SSI_SFCSR_RFWM0(x) (((x) & 0xf) << 4) -#define SSI_SFCSR_TFWM0(x) (((x) & 0xf) << 0) -#define SSI_SFCSR_RFWM0_MASK (0xf << 4) -#define SSI_SFCSR_TFWM0_MASK (0xf << 0) - -#define SSI_STR 0x30 -#define SSI_STR_TEST (1 << 15) -#define SSI_STR_RCK2TCK (1 << 14) -#define SSI_STR_RFS2TFS (1 << 13) -#define SSI_STR_RXSTATE(x) (((x) & 0xf) << 8) -#define SSI_STR_TXD2RXD (1 << 7) -#define SSI_STR_TCK2RCK (1 << 6) -#define SSI_STR_TFS2RFS (1 << 5) -#define SSI_STR_TXSTATE(x) (((x) & 0xf) << 0) - -#define SSI_SOR 0x34 -#define SSI_SOR_CLKOFF (1 << 6) -#define SSI_SOR_RX_CLR (1 << 5) -#define SSI_SOR_TX_CLR (1 << 4) -#define SSI_SOR_INIT (1 << 3) -#define SSI_SOR_WAIT(x) (((x) & 0x3) << 1) -#define SSI_SOR_WAIT_MASK (0x3 << 1) -#define SSI_SOR_SYNRST (1 << 0) - -#define SSI_SACNT 0x38 -#define SSI_SACNT_FRDIV(x) (((x) & 0x3f) << 5) -#define SSI_SACNT_WR (1 << 4) -#define SSI_SACNT_RD (1 << 3) -#define SSI_SACNT_TIF (1 << 2) -#define SSI_SACNT_FV (1 << 1) -#define SSI_SACNT_AC97EN (1 << 0) - -#define SSI_SACADD 0x3c -#define SSI_SACDAT 0x40 -#define SSI_SATAG 0x44 -#define SSI_STMSK 0x48 -#define SSI_SRMSK 0x4c -#define SSI_SACCST 0x50 -#define SSI_SACCEN 0x54 -#define SSI_SACCDIS 0x58 - -/* SSI clock sources */ -#define IMX_SSP_SYS_CLK 0 - -/* SSI audio dividers */ -#define IMX_SSI_TX_DIV_2 0 -#define IMX_SSI_TX_DIV_PSR 1 -#define IMX_SSI_TX_DIV_PM 2 -#define IMX_SSI_RX_DIV_2 3 -#define IMX_SSI_RX_DIV_PSR 4 -#define IMX_SSI_RX_DIV_PM 5 - -#define DRV_NAME "imx-ssi" - -#include -#include -#include "imx-pcm.h" - -struct imx_ssi { - struct platform_device *ac97_dev; - - struct snd_soc_dai *imx_ac97; - struct clk *clk; - void __iomem *base; - int irq; - int fiq_enable; - unsigned int offset; - - unsigned int flags; - - void (*ac97_reset) (struct snd_ac97 *ac97); - void (*ac97_warm_reset)(struct snd_ac97 *ac97); - - struct imx_pcm_dma_params dma_params_rx; - struct imx_pcm_dma_params dma_params_tx; - - int enabled; - - struct platform_device *soc_platform_pdev; - struct platform_device *soc_platform_pdev_fiq; -}; - -#endif /* _IMX_SSI_H */ diff --git a/sound/soc/imx/mx27vis-aic32x4.c b/sound/soc/imx/mx27vis-aic32x4.c deleted file mode 100644 index f6d04ad4bb39..000000000000 --- a/sound/soc/imx/mx27vis-aic32x4.c +++ /dev/null @@ -1,245 +0,0 @@ -/* - * mx27vis-aic32x4.c - * - * Copyright 2011 Vista Silicon S.L. - * - * Author: Javier Martin - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, - * MA 02110-1301, USA. - */ - -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include - -#include "../codecs/tlv320aic32x4.h" -#include "imx-ssi.h" -#include "imx-audmux.h" - -#define MX27VIS_AMP_GAIN 0 -#define MX27VIS_AMP_MUTE 1 - -#define MX27VIS_PIN_G0 (GPIO_PORTF + 9) -#define MX27VIS_PIN_G1 (GPIO_PORTF + 8) -#define MX27VIS_PIN_SDL (GPIO_PORTE + 5) -#define MX27VIS_PIN_SDR (GPIO_PORTF + 7) - -static int mx27vis_amp_gain; -static int mx27vis_amp_mute; - -static const int mx27vis_amp_pins[] = { - MX27VIS_PIN_G0 | GPIO_GPIO | GPIO_OUT, - MX27VIS_PIN_G1 | GPIO_GPIO | GPIO_OUT, - MX27VIS_PIN_SDL | GPIO_GPIO | GPIO_OUT, - MX27VIS_PIN_SDR | GPIO_GPIO | GPIO_OUT, -}; - -static int mx27vis_aic32x4_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - int ret; - u32 dai_format; - - dai_format = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM; - - /* set codec DAI configuration */ - snd_soc_dai_set_fmt(codec_dai, dai_format); - - /* set cpu DAI configuration */ - snd_soc_dai_set_fmt(cpu_dai, dai_format); - - ret = snd_soc_dai_set_sysclk(codec_dai, 0, - 25000000, SND_SOC_CLOCK_OUT); - if (ret) { - pr_err("%s: failed setting codec sysclk\n", __func__); - return ret; - } - - ret = snd_soc_dai_set_sysclk(cpu_dai, IMX_SSP_SYS_CLK, 0, - SND_SOC_CLOCK_IN); - if (ret) { - pr_err("can't set CPU system clock IMX_SSP_SYS_CLK\n"); - return ret; - } - - return 0; -} - -static struct snd_soc_ops mx27vis_aic32x4_snd_ops = { - .hw_params = mx27vis_aic32x4_hw_params, -}; - -static int mx27vis_amp_set(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - int value = ucontrol->value.integer.value[0]; - unsigned int reg = mc->reg; - int max = mc->max; - - if (value > max) - return -EINVAL; - - switch (reg) { - case MX27VIS_AMP_GAIN: - gpio_set_value(MX27VIS_PIN_G0, value & 1); - gpio_set_value(MX27VIS_PIN_G1, value >> 1); - mx27vis_amp_gain = value; - break; - case MX27VIS_AMP_MUTE: - gpio_set_value(MX27VIS_PIN_SDL, value & 1); - gpio_set_value(MX27VIS_PIN_SDR, value >> 1); - mx27vis_amp_mute = value; - break; - } - return 0; -} - -static int mx27vis_amp_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - unsigned int reg = mc->reg; - - switch (reg) { - case MX27VIS_AMP_GAIN: - ucontrol->value.integer.value[0] = mx27vis_amp_gain; - break; - case MX27VIS_AMP_MUTE: - ucontrol->value.integer.value[0] = mx27vis_amp_mute; - break; - } - return 0; -} - -/* From 6dB to 24dB in steps of 6dB */ -static const DECLARE_TLV_DB_SCALE(mx27vis_amp_tlv, 600, 600, 0); - -static const struct snd_kcontrol_new mx27vis_aic32x4_controls[] = { - SOC_DAPM_PIN_SWITCH("External Mic"), - SOC_SINGLE_EXT_TLV("LO Ext Boost", MX27VIS_AMP_GAIN, 0, 3, 0, - mx27vis_amp_get, mx27vis_amp_set, mx27vis_amp_tlv), - SOC_DOUBLE_EXT("LO Ext Mute Switch", MX27VIS_AMP_MUTE, 0, 1, 1, 0, - mx27vis_amp_get, mx27vis_amp_set), -}; - -static const struct snd_soc_dapm_widget aic32x4_dapm_widgets[] = { - SND_SOC_DAPM_MIC("External Mic", NULL), -}; - -static const struct snd_soc_dapm_route aic32x4_dapm_routes[] = { - {"Mic Bias", NULL, "External Mic"}, - {"IN1_R", NULL, "Mic Bias"}, - {"IN2_R", NULL, "Mic Bias"}, - {"IN3_R", NULL, "Mic Bias"}, - {"IN1_L", NULL, "Mic Bias"}, - {"IN2_L", NULL, "Mic Bias"}, - {"IN3_L", NULL, "Mic Bias"}, -}; - -static struct snd_soc_dai_link mx27vis_aic32x4_dai = { - .name = "tlv320aic32x4", - .stream_name = "TLV320AIC32X4", - .codec_dai_name = "tlv320aic32x4-hifi", - .platform_name = "imx-pcm-audio.0", - .codec_name = "tlv320aic32x4.0-0018", - .cpu_dai_name = "imx-ssi.0", - .ops = &mx27vis_aic32x4_snd_ops, -}; - -static struct snd_soc_card mx27vis_aic32x4 = { - .name = "visstrim_m10-audio", - .owner = THIS_MODULE, - .dai_link = &mx27vis_aic32x4_dai, - .num_links = 1, - .controls = mx27vis_aic32x4_controls, - .num_controls = ARRAY_SIZE(mx27vis_aic32x4_controls), - .dapm_widgets = aic32x4_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(aic32x4_dapm_widgets), - .dapm_routes = aic32x4_dapm_routes, - .num_dapm_routes = ARRAY_SIZE(aic32x4_dapm_routes), -}; - -static int __devinit mx27vis_aic32x4_probe(struct platform_device *pdev) -{ - int ret; - - mx27vis_aic32x4.dev = &pdev->dev; - ret = snd_soc_register_card(&mx27vis_aic32x4); - if (ret) { - dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", - ret); - return ret; - } - - /* Connect SSI0 as clock slave to SSI1 external pins */ - imx_audmux_v1_configure_port(MX27_AUDMUX_HPCR1_SSI0, - IMX_AUDMUX_V1_PCR_SYN | - IMX_AUDMUX_V1_PCR_TFSDIR | - IMX_AUDMUX_V1_PCR_TCLKDIR | - IMX_AUDMUX_V1_PCR_TFCSEL(MX27_AUDMUX_PPCR1_SSI_PINS_1) | - IMX_AUDMUX_V1_PCR_RXDSEL(MX27_AUDMUX_PPCR1_SSI_PINS_1) - ); - imx_audmux_v1_configure_port(MX27_AUDMUX_PPCR1_SSI_PINS_1, - IMX_AUDMUX_V1_PCR_SYN | - IMX_AUDMUX_V1_PCR_RXDSEL(MX27_AUDMUX_HPCR1_SSI0) - ); - - ret = mxc_gpio_setup_multiple_pins(mx27vis_amp_pins, - ARRAY_SIZE(mx27vis_amp_pins), "MX27VIS_AMP"); - if (ret) - printk(KERN_ERR "ASoC: unable to setup gpios\n"); - - return ret; -} - -static int __devexit mx27vis_aic32x4_remove(struct platform_device *pdev) -{ - snd_soc_unregister_card(&mx27vis_aic32x4); - - return 0; -} - -static struct platform_driver mx27vis_aic32x4_audio_driver = { - .driver = { - .name = "mx27vis", - .owner = THIS_MODULE, - }, - .probe = mx27vis_aic32x4_probe, - .remove = __devexit_p(mx27vis_aic32x4_remove), -}; - -module_platform_driver(mx27vis_aic32x4_audio_driver); - -MODULE_AUTHOR("Javier Martin "); -MODULE_DESCRIPTION("ALSA SoC AIC32X4 mx27 visstrim"); -MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:mx27vis"); diff --git a/sound/soc/imx/phycore-ac97.c b/sound/soc/imx/phycore-ac97.c deleted file mode 100644 index f8da6dd115ed..000000000000 --- a/sound/soc/imx/phycore-ac97.c +++ /dev/null @@ -1,125 +0,0 @@ -/* - * phycore-ac97.c -- SoC audio for imx_phycore in AC97 mode - * - * Copyright 2009 Sascha Hauer, Pengutronix - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - * - */ - -#include -#include -#include -#include -#include -#include -#include -#include - -#include "imx-audmux.h" - -static struct snd_soc_card imx_phycore; - -static struct snd_soc_ops imx_phycore_hifi_ops = { -}; - -static struct snd_soc_dai_link imx_phycore_dai_ac97[] = { - { - .name = "HiFi", - .stream_name = "HiFi", - .codec_dai_name = "wm9712-hifi", - .codec_name = "wm9712-codec", - .cpu_dai_name = "imx-ssi.0", - .platform_name = "imx-fiq-pcm-audio.0", - .ops = &imx_phycore_hifi_ops, - }, -}; - -static struct snd_soc_card imx_phycore = { - .name = "PhyCORE-ac97-audio", - .owner = THIS_MODULE, - .dai_link = imx_phycore_dai_ac97, - .num_links = ARRAY_SIZE(imx_phycore_dai_ac97), -}; - -static struct platform_device *imx_phycore_snd_ac97_device; -static struct platform_device *imx_phycore_snd_device; - -static int __init imx_phycore_init(void) -{ - int ret; - - if (machine_is_pca100()) { - imx_audmux_v1_configure_port(MX27_AUDMUX_HPCR1_SSI0, - IMX_AUDMUX_V1_PCR_SYN | /* 4wire mode */ - IMX_AUDMUX_V1_PCR_TFCSEL(3) | - IMX_AUDMUX_V1_PCR_TCLKDIR | /* clock is output */ - IMX_AUDMUX_V1_PCR_RXDSEL(3)); - imx_audmux_v1_configure_port(3, - IMX_AUDMUX_V1_PCR_SYN | /* 4wire mode */ - IMX_AUDMUX_V1_PCR_TFCSEL(0) | - IMX_AUDMUX_V1_PCR_TFSDIR | - IMX_AUDMUX_V1_PCR_RXDSEL(0)); - } else if (machine_is_pcm043()) { - imx_audmux_v2_configure_port(3, - IMX_AUDMUX_V2_PTCR_SYN | /* 4wire mode */ - IMX_AUDMUX_V2_PTCR_TFSEL(0) | - IMX_AUDMUX_V2_PTCR_TFSDIR, - IMX_AUDMUX_V2_PDCR_RXDSEL(0)); - imx_audmux_v2_configure_port(0, - IMX_AUDMUX_V2_PTCR_SYN | /* 4wire mode */ - IMX_AUDMUX_V2_PTCR_TCSEL(3) | - IMX_AUDMUX_V2_PTCR_TCLKDIR, /* clock is output */ - IMX_AUDMUX_V2_PDCR_RXDSEL(3)); - } else { - /* return happy. We might run on a totally different machine */ - return 0; - } - - imx_phycore_snd_ac97_device = platform_device_alloc("soc-audio", -1); - if (!imx_phycore_snd_ac97_device) - return -ENOMEM; - - platform_set_drvdata(imx_phycore_snd_ac97_device, &imx_phycore); - ret = platform_device_add(imx_phycore_snd_ac97_device); - if (ret) - goto fail1; - - imx_phycore_snd_device = platform_device_alloc("wm9712-codec", -1); - if (!imx_phycore_snd_device) { - ret = -ENOMEM; - goto fail2; - } - ret = platform_device_add(imx_phycore_snd_device); - - if (ret) { - printk(KERN_ERR "ASoC: Platform device allocation failed\n"); - goto fail3; - } - - return 0; - -fail3: - platform_device_put(imx_phycore_snd_device); -fail2: - platform_device_del(imx_phycore_snd_ac97_device); -fail1: - platform_device_put(imx_phycore_snd_ac97_device); - return ret; -} - -static void __exit imx_phycore_exit(void) -{ - platform_device_unregister(imx_phycore_snd_device); - platform_device_unregister(imx_phycore_snd_ac97_device); -} - -late_initcall(imx_phycore_init); -module_exit(imx_phycore_exit); - -MODULE_AUTHOR("Sascha Hauer "); -MODULE_DESCRIPTION("PhyCORE ALSA SoC driver"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/imx/wm1133-ev1.c b/sound/soc/imx/wm1133-ev1.c deleted file mode 100644 index fe54a69073e5..000000000000 --- a/sound/soc/imx/wm1133-ev1.c +++ /dev/null @@ -1,304 +0,0 @@ -/* - * wm1133-ev1.c - Audio for WM1133-EV1 on i.MX31ADS - * - * Copyright (c) 2010 Wolfson Microelectronics plc - * Author: Mark Brown - * - * Based on an earlier driver for the same hardware by Liam Girdwood. - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - */ - -#include -#include -#include -#include -#include -#include -#include -#include - -#include "imx-ssi.h" -#include "../codecs/wm8350.h" -#include "imx-audmux.h" - -/* There is a silicon mic on the board optionally connected via a solder pad - * SP1. Define this to enable it. - */ -#undef USE_SIMIC - -struct _wm8350_audio { - unsigned int channels; - snd_pcm_format_t format; - unsigned int rate; - unsigned int sysclk; - unsigned int bclkdiv; - unsigned int clkdiv; - unsigned int lr_rate; -}; - -/* in order of power consumption per rate (lowest first) */ -static const struct _wm8350_audio wm8350_audio[] = { - /* 16bit mono modes */ - {1, SNDRV_PCM_FORMAT_S16_LE, 8000, 12288000 >> 1, - WM8350_BCLK_DIV_48, WM8350_DACDIV_3, 16,}, - - /* 16 bit stereo modes */ - {2, SNDRV_PCM_FORMAT_S16_LE, 8000, 12288000, - WM8350_BCLK_DIV_48, WM8350_DACDIV_6, 32,}, - {2, SNDRV_PCM_FORMAT_S16_LE, 16000, 12288000, - WM8350_BCLK_DIV_24, WM8350_DACDIV_3, 32,}, - {2, SNDRV_PCM_FORMAT_S16_LE, 32000, 12288000, - WM8350_BCLK_DIV_12, WM8350_DACDIV_1_5, 32,}, - {2, SNDRV_PCM_FORMAT_S16_LE, 48000, 12288000, - WM8350_BCLK_DIV_8, WM8350_DACDIV_1, 32,}, - {2, SNDRV_PCM_FORMAT_S16_LE, 96000, 24576000, - WM8350_BCLK_DIV_8, WM8350_DACDIV_1, 32,}, - {2, SNDRV_PCM_FORMAT_S16_LE, 11025, 11289600, - WM8350_BCLK_DIV_32, WM8350_DACDIV_4, 32,}, - {2, SNDRV_PCM_FORMAT_S16_LE, 22050, 11289600, - WM8350_BCLK_DIV_16, WM8350_DACDIV_2, 32,}, - {2, SNDRV_PCM_FORMAT_S16_LE, 44100, 11289600, - WM8350_BCLK_DIV_8, WM8350_DACDIV_1, 32,}, - {2, SNDRV_PCM_FORMAT_S16_LE, 88200, 22579200, - WM8350_BCLK_DIV_8, WM8350_DACDIV_1, 32,}, - - /* 24bit stereo modes */ - {2, SNDRV_PCM_FORMAT_S24_LE, 48000, 12288000, - WM8350_BCLK_DIV_4, WM8350_DACDIV_1, 64,}, - {2, SNDRV_PCM_FORMAT_S24_LE, 96000, 24576000, - WM8350_BCLK_DIV_4, WM8350_DACDIV_1, 64,}, - {2, SNDRV_PCM_FORMAT_S24_LE, 44100, 11289600, - WM8350_BCLK_DIV_4, WM8350_DACDIV_1, 64,}, - {2, SNDRV_PCM_FORMAT_S24_LE, 88200, 22579200, - WM8350_BCLK_DIV_4, WM8350_DACDIV_1, 64,}, -}; - -static int wm1133_ev1_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - int i, found = 0; - snd_pcm_format_t format = params_format(params); - unsigned int rate = params_rate(params); - unsigned int channels = params_channels(params); - u32 dai_format; - - /* find the correct audio parameters */ - for (i = 0; i < ARRAY_SIZE(wm8350_audio); i++) { - if (rate == wm8350_audio[i].rate && - format == wm8350_audio[i].format && - channels == wm8350_audio[i].channels) { - found = 1; - break; - } - } - if (!found) - return -EINVAL; - - /* codec FLL input is 14.75 MHz from MCLK */ - snd_soc_dai_set_pll(codec_dai, 0, 0, 14750000, wm8350_audio[i].sysclk); - - dai_format = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM; - - /* set codec DAI configuration */ - snd_soc_dai_set_fmt(codec_dai, dai_format); - - /* set cpu DAI configuration */ - snd_soc_dai_set_fmt(cpu_dai, dai_format); - - /* TODO: The SSI driver should figure this out for us */ - switch (channels) { - case 2: - snd_soc_dai_set_tdm_slot(cpu_dai, 0xffffffc, 0xffffffc, 2, 0); - break; - case 1: - snd_soc_dai_set_tdm_slot(cpu_dai, 0xffffffe, 0xffffffe, 1, 0); - break; - default: - return -EINVAL; - } - - /* set MCLK as the codec system clock for DAC and ADC */ - snd_soc_dai_set_sysclk(codec_dai, WM8350_MCLK_SEL_PLL_MCLK, - wm8350_audio[i].sysclk, SND_SOC_CLOCK_IN); - - /* set codec BCLK division for sample rate */ - snd_soc_dai_set_clkdiv(codec_dai, WM8350_BCLK_CLKDIV, - wm8350_audio[i].bclkdiv); - - /* DAI is synchronous and clocked with DAC LRCLK & ADC LRC */ - snd_soc_dai_set_clkdiv(codec_dai, - WM8350_DACLR_CLKDIV, wm8350_audio[i].lr_rate); - snd_soc_dai_set_clkdiv(codec_dai, - WM8350_ADCLR_CLKDIV, wm8350_audio[i].lr_rate); - - /* now configure DAC and ADC clocks */ - snd_soc_dai_set_clkdiv(codec_dai, - WM8350_DAC_CLKDIV, wm8350_audio[i].clkdiv); - - snd_soc_dai_set_clkdiv(codec_dai, - WM8350_ADC_CLKDIV, wm8350_audio[i].clkdiv); - - return 0; -} - -static struct snd_soc_ops wm1133_ev1_ops = { - .hw_params = wm1133_ev1_hw_params, -}; - -static const struct snd_soc_dapm_widget wm1133_ev1_widgets[] = { -#ifdef USE_SIMIC - SND_SOC_DAPM_MIC("SiMIC", NULL), -#endif - SND_SOC_DAPM_MIC("Mic1 Jack", NULL), - SND_SOC_DAPM_MIC("Mic2 Jack", NULL), - SND_SOC_DAPM_LINE("Line In Jack", NULL), - SND_SOC_DAPM_LINE("Line Out Jack", NULL), - SND_SOC_DAPM_HP("Headphone Jack", NULL), -}; - -/* imx32ads soc_card audio map */ -static const struct snd_soc_dapm_route wm1133_ev1_map[] = { - -#ifdef USE_SIMIC - /* SiMIC --> IN1LN (with automatic bias) via SP1 */ - { "IN1LN", NULL, "Mic Bias" }, - { "Mic Bias", NULL, "SiMIC" }, -#endif - - /* Mic 1 Jack --> IN1LN and IN1LP (with automatic bias) */ - { "IN1LN", NULL, "Mic Bias" }, - { "IN1LP", NULL, "Mic1 Jack" }, - { "Mic Bias", NULL, "Mic1 Jack" }, - - /* Mic 2 Jack --> IN1RN and IN1RP (with automatic bias) */ - { "IN1RN", NULL, "Mic Bias" }, - { "IN1RP", NULL, "Mic2 Jack" }, - { "Mic Bias", NULL, "Mic2 Jack" }, - - /* Line in Jack --> AUX (L+R) */ - { "IN3R", NULL, "Line In Jack" }, - { "IN3L", NULL, "Line In Jack" }, - - /* Out1 --> Headphone Jack */ - { "Headphone Jack", NULL, "OUT1R" }, - { "Headphone Jack", NULL, "OUT1L" }, - - /* Out1 --> Line Out Jack */ - { "Line Out Jack", NULL, "OUT2R" }, - { "Line Out Jack", NULL, "OUT2L" }, -}; - -static struct snd_soc_jack hp_jack; - -static struct snd_soc_jack_pin hp_jack_pins[] = { - { .pin = "Headphone Jack", .mask = SND_JACK_HEADPHONE }, -}; - -static struct snd_soc_jack mic_jack; - -static struct snd_soc_jack_pin mic_jack_pins[] = { - { .pin = "Mic1 Jack", .mask = SND_JACK_MICROPHONE }, - { .pin = "Mic2 Jack", .mask = SND_JACK_MICROPHONE }, -}; - -static int wm1133_ev1_init(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_codec *codec = rtd->codec; - struct snd_soc_dapm_context *dapm = &codec->dapm; - - snd_soc_dapm_new_controls(dapm, wm1133_ev1_widgets, - ARRAY_SIZE(wm1133_ev1_widgets)); - - snd_soc_dapm_add_routes(dapm, wm1133_ev1_map, - ARRAY_SIZE(wm1133_ev1_map)); - - /* Headphone jack detection */ - snd_soc_jack_new(codec, "Headphone", SND_JACK_HEADPHONE, &hp_jack); - snd_soc_jack_add_pins(&hp_jack, ARRAY_SIZE(hp_jack_pins), - hp_jack_pins); - wm8350_hp_jack_detect(codec, WM8350_JDR, &hp_jack, SND_JACK_HEADPHONE); - - /* Microphone jack detection */ - snd_soc_jack_new(codec, "Microphone", - SND_JACK_MICROPHONE | SND_JACK_BTN_0, &mic_jack); - snd_soc_jack_add_pins(&mic_jack, ARRAY_SIZE(mic_jack_pins), - mic_jack_pins); - wm8350_mic_jack_detect(codec, &mic_jack, SND_JACK_MICROPHONE, - SND_JACK_BTN_0); - - snd_soc_dapm_force_enable_pin(dapm, "Mic Bias"); - - return 0; -} - - -static struct snd_soc_dai_link wm1133_ev1_dai = { - .name = "WM1133-EV1", - .stream_name = "Audio", - .cpu_dai_name = "imx-ssi.0", - .codec_dai_name = "wm8350-hifi", - .platform_name = "imx-fiq-pcm-audio.0", - .codec_name = "wm8350-codec.0-0x1a", - .init = wm1133_ev1_init, - .ops = &wm1133_ev1_ops, - .symmetric_rates = 1, -}; - -static struct snd_soc_card wm1133_ev1 = { - .name = "WM1133-EV1", - .owner = THIS_MODULE, - .dai_link = &wm1133_ev1_dai, - .num_links = 1, -}; - -static struct platform_device *wm1133_ev1_snd_device; - -static int __init wm1133_ev1_audio_init(void) -{ - int ret; - unsigned int ptcr, pdcr; - - /* SSI0 mastered by port 5 */ - ptcr = IMX_AUDMUX_V2_PTCR_SYN | - IMX_AUDMUX_V2_PTCR_TFSDIR | - IMX_AUDMUX_V2_PTCR_TFSEL(MX31_AUDMUX_PORT5_SSI_PINS_5) | - IMX_AUDMUX_V2_PTCR_TCLKDIR | - IMX_AUDMUX_V2_PTCR_TCSEL(MX31_AUDMUX_PORT5_SSI_PINS_5); - pdcr = IMX_AUDMUX_V2_PDCR_RXDSEL(MX31_AUDMUX_PORT5_SSI_PINS_5); - imx_audmux_v2_configure_port(MX31_AUDMUX_PORT1_SSI0, ptcr, pdcr); - - ptcr = IMX_AUDMUX_V2_PTCR_SYN; - pdcr = IMX_AUDMUX_V2_PDCR_RXDSEL(MX31_AUDMUX_PORT1_SSI0); - imx_audmux_v2_configure_port(MX31_AUDMUX_PORT5_SSI_PINS_5, ptcr, pdcr); - - wm1133_ev1_snd_device = platform_device_alloc("soc-audio", -1); - if (!wm1133_ev1_snd_device) - return -ENOMEM; - - platform_set_drvdata(wm1133_ev1_snd_device, &wm1133_ev1); - ret = platform_device_add(wm1133_ev1_snd_device); - - if (ret) - platform_device_put(wm1133_ev1_snd_device); - - return ret; -} -module_init(wm1133_ev1_audio_init); - -static void __exit wm1133_ev1_audio_exit(void) -{ - platform_device_unregister(wm1133_ev1_snd_device); -} -module_exit(wm1133_ev1_audio_exit); - -MODULE_AUTHOR("Mark Brown "); -MODULE_DESCRIPTION("Audio for WM1133-EV1 on i.MX31ADS"); -MODULE_LICENSE("GPL"); -- cgit v1.2.3 From f19493a3d25ddf3ac7f27d846d54e95fb91af119 Mon Sep 17 00:00:00 2001 From: Shawn Guo Date: Fri, 16 Mar 2012 16:56:39 +0800 Subject: ASoC: fsl: rename imx-pcm Kconfig options and filename Rename a couple of imx-pcm Kconfig options and filename to get them well named and less confusing. Signed-off-by: Shawn Guo Acked-by: Sascha Hauer Acked-by: Timur Tabi Signed-off-by: Mark Brown --- sound/soc/fsl/Kconfig | 12 +-- sound/soc/fsl/Makefile | 4 +- sound/soc/fsl/imx-pcm-dma-mx2.c | 175 ---------------------------------------- sound/soc/fsl/imx-pcm-dma.c | 175 ++++++++++++++++++++++++++++++++++++++++ 4 files changed, 183 insertions(+), 183 deletions(-) delete mode 100644 sound/soc/fsl/imx-pcm-dma-mx2.c create mode 100644 sound/soc/fsl/imx-pcm-dma.c (limited to 'sound') diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 19856a05208c..e3f75093d197 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -97,12 +97,12 @@ config SND_SOC_IMX_SSI config SND_SOC_IMX_PCM tristate -config SND_MXC_SOC_FIQ +config SND_SOC_IMX_PCM_FIQ tristate select FIQ select SND_SOC_IMX_PCM -config SND_MXC_SOC_MX2 +config SND_SOC_IMX_PCM_DMA tristate select SND_SOC_DMAENGINE_PCM select SND_SOC_IMX_PCM @@ -114,7 +114,7 @@ config SND_MXC_SOC_WM1133_EV1 tristate "Audio on the the i.MX31ADS with WM1133-EV1 fitted" depends on MACH_MX31ADS_WM1133_EV1 && EXPERIMENTAL select SND_SOC_WM8350 - select SND_MXC_SOC_FIQ + select SND_SOC_IMX_PCM_FIQ select SND_SOC_IMX_AUDMUX select SND_SOC_IMX_SSI help @@ -125,7 +125,7 @@ config SND_SOC_MX27VIS_AIC32X4 tristate "SoC audio support for Visstrim M10 boards" depends on MACH_IMX27_VISSTRIM_M10 && I2C select SND_SOC_TLV320AIC32X4 - select SND_MXC_SOC_MX2 + select SND_SOC_IMX_PCM_DMA select SND_SOC_IMX_AUDMUX select SND_SOC_IMX_SSI help @@ -137,7 +137,7 @@ config SND_SOC_PHYCORE_AC97 depends on MACH_PCM043 || MACH_PCA100 select SND_SOC_AC97_BUS select SND_SOC_WM9712 - select SND_MXC_SOC_FIQ + select SND_SOC_IMX_PCM_FIQ select SND_SOC_IMX_AUDMUX select SND_SOC_IMX_SSI help @@ -152,7 +152,7 @@ config SND_SOC_EUKREA_TLV320 || MACH_EUKREA_MBIMXSD51_BASEBOARD depends on I2C select SND_SOC_TLV320AIC23 - select SND_MXC_SOC_FIQ + select SND_SOC_IMX_PCM_FIQ select SND_SOC_IMX_AUDMUX select SND_SOC_IMX_SSI help diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index 36c257fd35bb..f031409a3929 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -30,8 +30,8 @@ obj-$(CONFIG_SND_SOC_IMX_AUDMUX) += snd-soc-imx-audmux.o obj-$(CONFIG_SND_SOC_IMX_PCM) += snd-soc-imx-pcm.o snd-soc-imx-pcm-y := imx-pcm.o -snd-soc-imx-pcm-$(CONFIG_SND_MXC_SOC_FIQ) += imx-pcm-fiq.o -snd-soc-imx-pcm-$(CONFIG_SND_MXC_SOC_MX2) += imx-pcm-dma-mx2.o +snd-soc-imx-pcm-$(CONFIG_SND_SOC_IMX_PCM_FIQ) += imx-pcm-fiq.o +snd-soc-imx-pcm-$(CONFIG_SND_SOC_IMX_PCM_DMA) += imx-pcm-dma.o # i.MX Machine Support snd-soc-eukrea-tlv320-objs := eukrea-tlv320.o diff --git a/sound/soc/fsl/imx-pcm-dma-mx2.c b/sound/soc/fsl/imx-pcm-dma-mx2.c deleted file mode 100644 index 6b818de2fc03..000000000000 --- a/sound/soc/fsl/imx-pcm-dma-mx2.c +++ /dev/null @@ -1,175 +0,0 @@ -/* - * imx-pcm-dma-mx2.c -- ALSA Soc Audio Layer - * - * Copyright 2009 Sascha Hauer - * - * This code is based on code copyrighted by Freescale, - * Liam Girdwood, Javier Martin and probably others. - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - */ -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include - -#include -#include -#include -#include -#include -#include - -#include - -#include "imx-pcm.h" - -static bool filter(struct dma_chan *chan, void *param) -{ - if (!imx_dma_is_general_purpose(chan)) - return false; - - chan->private = param; - - return true; -} - -static int snd_imx_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct dma_chan *chan = snd_dmaengine_pcm_get_chan(substream); - struct imx_pcm_dma_params *dma_params; - struct dma_slave_config slave_config; - int ret; - - dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); - - ret = snd_hwparams_to_dma_slave_config(substream, params, &slave_config); - if (ret) - return ret; - - slave_config.device_fc = false; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - slave_config.dst_addr = dma_params->dma_addr; - slave_config.dst_maxburst = dma_params->burstsize; - } else { - slave_config.src_addr = dma_params->dma_addr; - slave_config.src_maxburst = dma_params->burstsize; - } - - ret = dmaengine_slave_config(chan, &slave_config); - if (ret) - return ret; - - snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); - - return 0; -} - -static struct snd_pcm_hardware snd_imx_hardware = { - .info = SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_PAUSE | - SNDRV_PCM_INFO_RESUME, - .formats = SNDRV_PCM_FMTBIT_S16_LE, - .rate_min = 8000, - .channels_min = 2, - .channels_max = 2, - .buffer_bytes_max = IMX_SSI_DMABUF_SIZE, - .period_bytes_min = 128, - .period_bytes_max = 65535, /* Limited by SDMA engine */ - .periods_min = 2, - .periods_max = 255, - .fifo_size = 0, -}; - -static int snd_imx_open(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct imx_pcm_dma_params *dma_params; - struct imx_dma_data *dma_data; - int ret; - - snd_soc_set_runtime_hwparams(substream, &snd_imx_hardware); - - dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); - - dma_data = kzalloc(sizeof(*dma_data), GFP_KERNEL); - dma_data->peripheral_type = IMX_DMATYPE_SSI; - dma_data->priority = DMA_PRIO_HIGH; - dma_data->dma_request = dma_params->dma; - - ret = snd_dmaengine_pcm_open(substream, filter, dma_data); - if (ret) { - kfree(dma_data); - return 0; - } - - snd_dmaengine_pcm_set_data(substream, dma_data); - - return 0; -} - -static int snd_imx_close(struct snd_pcm_substream *substream) -{ - struct imx_dma_data *dma_data = snd_dmaengine_pcm_get_data(substream); - - snd_dmaengine_pcm_close(substream); - kfree(dma_data); - - return 0; -} - -static struct snd_pcm_ops imx_pcm_ops = { - .open = snd_imx_open, - .close = snd_imx_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = snd_imx_pcm_hw_params, - .trigger = snd_dmaengine_pcm_trigger, - .pointer = snd_dmaengine_pcm_pointer, - .mmap = snd_imx_pcm_mmap, -}; - -static struct snd_soc_platform_driver imx_soc_platform_mx2 = { - .ops = &imx_pcm_ops, - .pcm_new = imx_pcm_new, - .pcm_free = imx_pcm_free, -}; - -static int __devinit imx_soc_platform_probe(struct platform_device *pdev) -{ - return snd_soc_register_platform(&pdev->dev, &imx_soc_platform_mx2); -} - -static int __devexit imx_soc_platform_remove(struct platform_device *pdev) -{ - snd_soc_unregister_platform(&pdev->dev); - return 0; -} - -static struct platform_driver imx_pcm_driver = { - .driver = { - .name = "imx-pcm-audio", - .owner = THIS_MODULE, - }, - .probe = imx_soc_platform_probe, - .remove = __devexit_p(imx_soc_platform_remove), -}; - -module_platform_driver(imx_pcm_driver); -MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:imx-pcm-audio"); diff --git a/sound/soc/fsl/imx-pcm-dma.c b/sound/soc/fsl/imx-pcm-dma.c new file mode 100644 index 000000000000..6b818de2fc03 --- /dev/null +++ b/sound/soc/fsl/imx-pcm-dma.c @@ -0,0 +1,175 @@ +/* + * imx-pcm-dma-mx2.c -- ALSA Soc Audio Layer + * + * Copyright 2009 Sascha Hauer + * + * This code is based on code copyrighted by Freescale, + * Liam Girdwood, Javier Martin and probably others. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include +#include +#include +#include +#include +#include + +#include + +#include "imx-pcm.h" + +static bool filter(struct dma_chan *chan, void *param) +{ + if (!imx_dma_is_general_purpose(chan)) + return false; + + chan->private = param; + + return true; +} + +static int snd_imx_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct dma_chan *chan = snd_dmaengine_pcm_get_chan(substream); + struct imx_pcm_dma_params *dma_params; + struct dma_slave_config slave_config; + int ret; + + dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + + ret = snd_hwparams_to_dma_slave_config(substream, params, &slave_config); + if (ret) + return ret; + + slave_config.device_fc = false; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + slave_config.dst_addr = dma_params->dma_addr; + slave_config.dst_maxburst = dma_params->burstsize; + } else { + slave_config.src_addr = dma_params->dma_addr; + slave_config.src_maxburst = dma_params->burstsize; + } + + ret = dmaengine_slave_config(chan, &slave_config); + if (ret) + return ret; + + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); + + return 0; +} + +static struct snd_pcm_hardware snd_imx_hardware = { + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_RESUME, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .rate_min = 8000, + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = IMX_SSI_DMABUF_SIZE, + .period_bytes_min = 128, + .period_bytes_max = 65535, /* Limited by SDMA engine */ + .periods_min = 2, + .periods_max = 255, + .fifo_size = 0, +}; + +static int snd_imx_open(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct imx_pcm_dma_params *dma_params; + struct imx_dma_data *dma_data; + int ret; + + snd_soc_set_runtime_hwparams(substream, &snd_imx_hardware); + + dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + + dma_data = kzalloc(sizeof(*dma_data), GFP_KERNEL); + dma_data->peripheral_type = IMX_DMATYPE_SSI; + dma_data->priority = DMA_PRIO_HIGH; + dma_data->dma_request = dma_params->dma; + + ret = snd_dmaengine_pcm_open(substream, filter, dma_data); + if (ret) { + kfree(dma_data); + return 0; + } + + snd_dmaengine_pcm_set_data(substream, dma_data); + + return 0; +} + +static int snd_imx_close(struct snd_pcm_substream *substream) +{ + struct imx_dma_data *dma_data = snd_dmaengine_pcm_get_data(substream); + + snd_dmaengine_pcm_close(substream); + kfree(dma_data); + + return 0; +} + +static struct snd_pcm_ops imx_pcm_ops = { + .open = snd_imx_open, + .close = snd_imx_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_imx_pcm_hw_params, + .trigger = snd_dmaengine_pcm_trigger, + .pointer = snd_dmaengine_pcm_pointer, + .mmap = snd_imx_pcm_mmap, +}; + +static struct snd_soc_platform_driver imx_soc_platform_mx2 = { + .ops = &imx_pcm_ops, + .pcm_new = imx_pcm_new, + .pcm_free = imx_pcm_free, +}; + +static int __devinit imx_soc_platform_probe(struct platform_device *pdev) +{ + return snd_soc_register_platform(&pdev->dev, &imx_soc_platform_mx2); +} + +static int __devexit imx_soc_platform_remove(struct platform_device *pdev) +{ + snd_soc_unregister_platform(&pdev->dev); + return 0; +} + +static struct platform_driver imx_pcm_driver = { + .driver = { + .name = "imx-pcm-audio", + .owner = THIS_MODULE, + }, + .probe = imx_soc_platform_probe, + .remove = __devexit_p(imx_soc_platform_remove), +}; + +module_platform_driver(imx_pcm_driver); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:imx-pcm-audio"); -- cgit v1.2.3 From 60aae8da298e3ac0af07c8cdb6a98e47e8deab35 Mon Sep 17 00:00:00 2001 From: Shawn Guo Date: Fri, 16 Mar 2012 16:56:40 +0800 Subject: ASoC: fsl: create fsl_utils to accommodate the common functions There is some amount of code duplication between mpc8610_hpcd and p1022_ds machine drivers, and the same code will be duplicated again when another new machine driver is added. The patch creates fsl_utils to accommodate the common functions to stop the code duplication. Signed-off-by: Shawn Guo Acked-by: Sascha Hauer Acked-by: Timur Tabi Signed-off-by: Mark Brown --- sound/soc/fsl/Kconfig | 5 ++ sound/soc/fsl/Makefile | 2 + sound/soc/fsl/fsl_utils.c | 135 +++++++++++++++++++++++++++++++++++++ sound/soc/fsl/fsl_utils.h | 27 ++++++++ sound/soc/fsl/mpc8610_hpcd.c | 157 +++---------------------------------------- sound/soc/fsl/p1022_ds.c | 149 +++------------------------------------- 6 files changed, 189 insertions(+), 286 deletions(-) create mode 100644 sound/soc/fsl/fsl_utils.c create mode 100644 sound/soc/fsl/fsl_utils.h (limited to 'sound') diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index e3f75093d197..535ee73ecf15 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -1,6 +1,9 @@ config SND_SOC_FSL_SSI tristate +config SND_SOC_FSL_UTILS + tristate + menuconfig SND_POWERPC_SOC tristate "SoC Audio for Freescale PowerPC CPUs" depends on FSL_SOC @@ -26,6 +29,7 @@ config SND_SOC_MPC8610_HPCD # I2C is necessary for the CS4270 driver depends on MPC8610_HPCD && I2C select SND_SOC_FSL_SSI + select SND_SOC_FSL_UTILS select SND_SOC_POWERPC_DMA select SND_SOC_CS4270 select SND_SOC_CS4270_VD33_ERRATA @@ -38,6 +42,7 @@ config SND_SOC_P1022_DS # I2C is necessary for the WM8776 driver depends on P1022_DS && I2C select SND_SOC_FSL_SSI + select SND_SOC_FSL_UTILS select SND_SOC_POWERPC_DMA select SND_SOC_WM8776 default y if P1022_DS diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index f031409a3929..fbdbed0af909 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -8,8 +8,10 @@ obj-$(CONFIG_SND_SOC_P1022_DS) += snd-soc-p1022-ds.o # Freescale PowerPC SSI/DMA Platform Support snd-soc-fsl-ssi-objs := fsl_ssi.o +snd-soc-fsl-utils-objs := fsl_utils.o snd-soc-fsl-dma-objs := fsl_dma.o obj-$(CONFIG_SND_SOC_FSL_SSI) += snd-soc-fsl-ssi.o +obj-$(CONFIG_SND_SOC_FSL_UTILS) += snd-soc-fsl-utils.o obj-$(CONFIG_SND_SOC_POWERPC_DMA) += snd-soc-fsl-dma.o # MPC5200 Platform Support diff --git a/sound/soc/fsl/fsl_utils.c b/sound/soc/fsl/fsl_utils.c new file mode 100644 index 000000000000..4370c28e5362 --- /dev/null +++ b/sound/soc/fsl/fsl_utils.c @@ -0,0 +1,135 @@ +/** + * Freescale ALSA SoC Machine driver utility + * + * Author: Timur Tabi + * + * Copyright 2010 Freescale Semiconductor, Inc. + * + * This file is licensed under the terms of the GNU General Public License + * version 2. This program is licensed "as is" without any warranty of any + * kind, whether express or implied. + */ + +#include +#include +#include +#include + +#include "fsl_utils.h" + +/** + * fsl_asoc_get_codec_dev_name - determine the dev_name for a codec node + * + * @np: pointer to the I2C device tree node + * @buf: buffer to be filled with the dev_name of the I2C device + * @len: the length of the buffer + * + * This function determines the dev_name for an I2C node. This is the name + * that would be returned by dev_name() if this device_node were part of a + * 'struct device' It's ugly and hackish, but it works. + * + * The dev_name for such devices include the bus number and I2C address. For + * example, "cs4270.0-004f". + */ +int fsl_asoc_get_codec_dev_name(struct device_node *np, char *buf, size_t len) +{ + const u32 *iprop; + u32 addr; + char temp[DAI_NAME_SIZE]; + struct i2c_client *i2c; + + of_modalias_node(np, temp, DAI_NAME_SIZE); + + iprop = of_get_property(np, "reg", NULL); + if (!iprop) + return -EINVAL; + + addr = be32_to_cpup(iprop); + + /* We need the adapter number */ + i2c = of_find_i2c_device_by_node(np); + if (!i2c) { + put_device(&i2c->dev); + return -ENODEV; + } + + snprintf(buf, len, "%s.%u-%04x", temp, i2c->adapter->nr, addr); + put_device(&i2c->dev); + + return 0; +} +EXPORT_SYMBOL(fsl_asoc_get_codec_dev_name); + +/** + * fsl_asoc_get_dma_channel - determine the dma channel for a SSI node + * + * @ssi_np: pointer to the SSI device tree node + * @name: name of the phandle pointing to the dma channel + * @dai: ASoC DAI link pointer to be filled with platform_name + * @dma_channel_id: dma channel id to be returned + * @dma_id: dma id to be returned + * + * This function determines the dma and channel id for given SSI node. It + * also discovers the platform_name for the ASoC DAI link. + */ +int fsl_asoc_get_dma_channel(struct device_node *ssi_np, + const char *name, + struct snd_soc_dai_link *dai, + unsigned int *dma_channel_id, + unsigned int *dma_id) +{ + struct resource res; + struct device_node *dma_channel_np, *dma_np; + const u32 *iprop; + int ret; + + dma_channel_np = of_parse_phandle(ssi_np, name, 0); + if (!dma_channel_np) + return -EINVAL; + + if (!of_device_is_compatible(dma_channel_np, "fsl,ssi-dma-channel")) { + of_node_put(dma_channel_np); + return -EINVAL; + } + + /* Determine the dev_name for the device_node. This code mimics the + * behavior of of_device_make_bus_id(). We need this because ASoC uses + * the dev_name() of the device to match the platform (DMA) device with + * the CPU (SSI) device. It's all ugly and hackish, but it works (for + * now). + * + * dai->platform name should already point to an allocated buffer. + */ + ret = of_address_to_resource(dma_channel_np, 0, &res); + if (ret) { + of_node_put(dma_channel_np); + return ret; + } + snprintf((char *)dai->platform_name, DAI_NAME_SIZE, "%llx.%s", + (unsigned long long) res.start, dma_channel_np->name); + + iprop = of_get_property(dma_channel_np, "cell-index", NULL); + if (!iprop) { + of_node_put(dma_channel_np); + return -EINVAL; + } + *dma_channel_id = be32_to_cpup(iprop); + + dma_np = of_get_parent(dma_channel_np); + iprop = of_get_property(dma_np, "cell-index", NULL); + if (!iprop) { + of_node_put(dma_np); + return -EINVAL; + } + *dma_id = be32_to_cpup(iprop); + + of_node_put(dma_np); + of_node_put(dma_channel_np); + + return 0; +} +EXPORT_SYMBOL(fsl_asoc_get_dma_channel); + +MODULE_AUTHOR("Timur Tabi "); +MODULE_DESCRIPTION("Freescale ASoC utility code"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/fsl/fsl_utils.h b/sound/soc/fsl/fsl_utils.h new file mode 100644 index 000000000000..44d143670c37 --- /dev/null +++ b/sound/soc/fsl/fsl_utils.h @@ -0,0 +1,27 @@ +/** + * Freescale ALSA SoC Machine driver utility + * + * Author: Timur Tabi + * + * Copyright 2010 Freescale Semiconductor, Inc. + * + * This file is licensed under the terms of the GNU General Public License + * version 2. This program is licensed "as is" without any warranty of any + * kind, whether express or implied. + */ + +#ifndef _FSL_UTILS_H +#define _FSL_UTILS_H + +#define DAI_NAME_SIZE 32 + +struct snd_soc_dai_link; +struct device_node; + +int fsl_asoc_get_codec_dev_name(struct device_node *np, char *buf, size_t len); +int fsl_asoc_get_dma_channel(struct device_node *ssi_np, const char *name, + struct snd_soc_dai_link *dai, + unsigned int *dma_channel_id, + unsigned int *dma_id); + +#endif /* _FSL_UTILS_H */ diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c index afbabf427f27..41951820b50e 100644 --- a/sound/soc/fsl/mpc8610_hpcd.c +++ b/sound/soc/fsl/mpc8610_hpcd.c @@ -14,18 +14,16 @@ #include #include #include -#include #include #include #include "fsl_dma.h" #include "fsl_ssi.h" +#include "fsl_utils.h" /* There's only one global utilities register */ static phys_addr_t guts_phys; -#define DAI_NAME_SIZE 32 - /** * mpc8610_hpcd_data: machine-specific ASoC device data * @@ -180,141 +178,6 @@ static struct snd_soc_ops mpc8610_hpcd_ops = { .startup = mpc8610_hpcd_startup, }; -/** - * get_node_by_phandle_name - get a node by its phandle name - * - * This function takes a node, the name of a property in that node, and a - * compatible string. Assuming the property is a phandle to another node, - * it returns that node, (optionally) if that node is compatible. - * - * If the property is not a phandle, or the node it points to is not compatible - * with the specific string, then NULL is returned. - */ -static struct device_node *get_node_by_phandle_name(struct device_node *np, - const char *name, - const char *compatible) -{ - const phandle *ph; - int len; - - ph = of_get_property(np, name, &len); - if (!ph || (len != sizeof(phandle))) - return NULL; - - np = of_find_node_by_phandle(*ph); - if (!np) - return NULL; - - if (compatible && !of_device_is_compatible(np, compatible)) { - of_node_put(np); - return NULL; - } - - return np; -} - -/** - * get_parent_cell_index -- return the cell-index of the parent of a node - * - * Return the value of the cell-index property of the parent of the given - * node. This is used for DMA channel nodes that need to know the DMA ID - * of the controller they are on. - */ -static int get_parent_cell_index(struct device_node *np) -{ - struct device_node *parent = of_get_parent(np); - const u32 *iprop; - - if (!parent) - return -1; - - iprop = of_get_property(parent, "cell-index", NULL); - of_node_put(parent); - - if (!iprop) - return -1; - - return be32_to_cpup(iprop); -} - -/** - * codec_node_dev_name - determine the dev_name for a codec node - * - * This function determines the dev_name for an I2C node. This is the name - * that would be returned by dev_name() if this device_node were part of a - * 'struct device' It's ugly and hackish, but it works. - * - * The dev_name for such devices include the bus number and I2C address. For - * example, "cs4270.0-004f". - */ -static int codec_node_dev_name(struct device_node *np, char *buf, size_t len) -{ - const u32 *iprop; - int addr; - char temp[DAI_NAME_SIZE]; - struct i2c_client *i2c; - - of_modalias_node(np, temp, DAI_NAME_SIZE); - - iprop = of_get_property(np, "reg", NULL); - if (!iprop) - return -EINVAL; - - addr = be32_to_cpup(iprop); - - /* We need the adapter number */ - i2c = of_find_i2c_device_by_node(np); - if (!i2c) - return -ENODEV; - - snprintf(buf, len, "%s.%u-%04x", temp, i2c->adapter->nr, addr); - - return 0; -} - -static int get_dma_channel(struct device_node *ssi_np, - const char *name, - struct snd_soc_dai_link *dai, - unsigned int *dma_channel_id, - unsigned int *dma_id) -{ - struct resource res; - struct device_node *dma_channel_np; - const u32 *iprop; - int ret; - - dma_channel_np = get_node_by_phandle_name(ssi_np, name, - "fsl,ssi-dma-channel"); - if (!dma_channel_np) - return -EINVAL; - - /* Determine the dev_name for the device_node. This code mimics the - * behavior of of_device_make_bus_id(). We need this because ASoC uses - * the dev_name() of the device to match the platform (DMA) device with - * the CPU (SSI) device. It's all ugly and hackish, but it works (for - * now). - * - * dai->platform name should already point to an allocated buffer. - */ - ret = of_address_to_resource(dma_channel_np, 0, &res); - if (ret) - return ret; - snprintf((char *)dai->platform_name, DAI_NAME_SIZE, "%llx.%s", - (unsigned long long) res.start, dma_channel_np->name); - - iprop = of_get_property(dma_channel_np, "cell-index", NULL); - if (!iprop) { - of_node_put(dma_channel_np); - return -EINVAL; - } - - *dma_channel_id = be32_to_cpup(iprop); - *dma_id = get_parent_cell_index(dma_channel_np); - of_node_put(dma_channel_np); - - return 0; -} - /** * mpc8610_hpcd_probe: platform probe function for the machine driver * @@ -353,8 +216,8 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev) machine_data->dai[0].ops = &mpc8610_hpcd_ops; /* Determine the codec name, it will be used as the codec DAI name */ - ret = codec_node_dev_name(codec_np, machine_data->codec_name, - DAI_NAME_SIZE); + ret = fsl_asoc_get_codec_dev_name(codec_np, machine_data->codec_name, + DAI_NAME_SIZE); if (ret) { dev_err(&pdev->dev, "invalid codec node %s\n", codec_np->full_name); @@ -458,9 +321,10 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev) /* Find the playback DMA channel to use. */ machine_data->dai[0].platform_name = machine_data->platform_name[0]; - ret = get_dma_channel(np, "fsl,playback-dma", &machine_data->dai[0], - &machine_data->dma_channel_id[0], - &machine_data->dma_id[0]); + ret = fsl_asoc_get_dma_channel(np, "fsl,playback-dma", + &machine_data->dai[0], + &machine_data->dma_channel_id[0], + &machine_data->dma_id[0]); if (ret) { dev_err(&pdev->dev, "missing/invalid playback DMA phandle\n"); goto error; @@ -468,9 +332,10 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev) /* Find the capture DMA channel to use. */ machine_data->dai[1].platform_name = machine_data->platform_name[1]; - ret = get_dma_channel(np, "fsl,capture-dma", &machine_data->dai[1], - &machine_data->dma_channel_id[1], - &machine_data->dma_id[1]); + ret = fsl_asoc_get_dma_channel(np, "fsl,capture-dma", + &machine_data->dai[1], + &machine_data->dma_channel_id[1], + &machine_data->dma_id[1]); if (ret) { dev_err(&pdev->dev, "missing/invalid capture DMA phandle\n"); goto error; diff --git a/sound/soc/fsl/p1022_ds.c b/sound/soc/fsl/p1022_ds.c index 46623405a2ce..be79e7388c0f 100644 --- a/sound/soc/fsl/p1022_ds.c +++ b/sound/soc/fsl/p1022_ds.c @@ -14,12 +14,12 @@ #include #include #include -#include #include #include #include "fsl_dma.h" #include "fsl_ssi.h" +#include "fsl_utils.h" /* P1022-specific PMUXCR and DMUXCR bit definitions */ @@ -57,8 +57,6 @@ static inline void guts_set_dmuxcr(struct ccsr_guts_85xx __iomem *guts, /* There's only one global utilities register */ static phys_addr_t guts_phys; -#define DAI_NAME_SIZE 32 - /** * machine_data: machine-specific ASoC device data * @@ -190,136 +188,6 @@ static struct snd_soc_ops p1022_ds_ops = { .startup = p1022_ds_startup, }; -/** - * get_node_by_phandle_name - get a node by its phandle name - * - * This function takes a node, the name of a property in that node, and a - * compatible string. Assuming the property is a phandle to another node, - * it returns that node, (optionally) if that node is compatible. - * - * If the property is not a phandle, or the node it points to is not compatible - * with the specific string, then NULL is returned. - */ -static struct device_node *get_node_by_phandle_name(struct device_node *np, - const char *name, const char *compatible) -{ - np = of_parse_phandle(np, name, 0); - if (!np) - return NULL; - - if (!of_device_is_compatible(np, compatible)) { - of_node_put(np); - return NULL; - } - - return np; -} - -/** - * get_parent_cell_index -- return the cell-index of the parent of a node - * - * Return the value of the cell-index property of the parent of the given - * node. This is used for DMA channel nodes that need to know the DMA ID - * of the controller they are on. - */ -static int get_parent_cell_index(struct device_node *np) -{ - struct device_node *parent = of_get_parent(np); - const u32 *iprop; - int ret = -1; - - if (!parent) - return -1; - - iprop = of_get_property(parent, "cell-index", NULL); - if (iprop) - ret = be32_to_cpup(iprop); - - of_node_put(parent); - - return ret; -} - -/** - * codec_node_dev_name - determine the dev_name for a codec node - * - * This function determines the dev_name for an I2C node. This is the name - * that would be returned by dev_name() if this device_node were part of a - * 'struct device' It's ugly and hackish, but it works. - * - * The dev_name for such devices include the bus number and I2C address. For - * example, "cs4270-codec.0-004f". - */ -static int codec_node_dev_name(struct device_node *np, char *buf, size_t len) -{ - const u32 *iprop; - int addr; - char temp[DAI_NAME_SIZE]; - struct i2c_client *i2c; - - of_modalias_node(np, temp, DAI_NAME_SIZE); - - iprop = of_get_property(np, "reg", NULL); - if (!iprop) - return -EINVAL; - - addr = be32_to_cpup(iprop); - - /* We need the adapter number */ - i2c = of_find_i2c_device_by_node(np); - if (!i2c) - return -ENODEV; - - snprintf(buf, len, "%s.%u-%04x", temp, i2c->adapter->nr, addr); - - return 0; -} - -static int get_dma_channel(struct device_node *ssi_np, - const char *name, - struct snd_soc_dai_link *dai, - unsigned int *dma_channel_id, - unsigned int *dma_id) -{ - struct resource res; - struct device_node *dma_channel_np; - const u32 *iprop; - int ret; - - dma_channel_np = get_node_by_phandle_name(ssi_np, name, - "fsl,ssi-dma-channel"); - if (!dma_channel_np) - return -EINVAL; - - /* Determine the dev_name for the device_node. This code mimics the - * behavior of of_device_make_bus_id(). We need this because ASoC uses - * the dev_name() of the device to match the platform (DMA) device with - * the CPU (SSI) device. It's all ugly and hackish, but it works (for - * now). - * - * dai->platform name should already point to an allocated buffer. - */ - ret = of_address_to_resource(dma_channel_np, 0, &res); - if (ret) { - of_node_put(dma_channel_np); - return ret; - } - snprintf((char *)dai->platform_name, DAI_NAME_SIZE, "%llx.%s", - (unsigned long long) res.start, dma_channel_np->name); - - iprop = of_get_property(dma_channel_np, "cell-index", NULL); - if (!iprop) { - of_node_put(dma_channel_np); - return -EINVAL; - } - - *dma_channel_id = be32_to_cpup(iprop); - *dma_id = get_parent_cell_index(dma_channel_np); - of_node_put(dma_channel_np); - - return 0; -} - /** * p1022_ds_probe: platform probe function for the machine driver * @@ -358,7 +226,8 @@ static int p1022_ds_probe(struct platform_device *pdev) mdata->dai[0].ops = &p1022_ds_ops; /* Determine the codec name, it will be used as the codec DAI name */ - ret = codec_node_dev_name(codec_np, mdata->codec_name, DAI_NAME_SIZE); + ret = fsl_asoc_get_codec_dev_name(codec_np, mdata->codec_name, + DAI_NAME_SIZE); if (ret) { dev_err(&pdev->dev, "invalid codec node %s\n", codec_np->full_name); @@ -462,9 +331,9 @@ static int p1022_ds_probe(struct platform_device *pdev) /* Find the playback DMA channel to use. */ mdata->dai[0].platform_name = mdata->platform_name[0]; - ret = get_dma_channel(np, "fsl,playback-dma", &mdata->dai[0], - &mdata->dma_channel_id[0], - &mdata->dma_id[0]); + ret = fsl_asoc_get_dma_channel(np, "fsl,playback-dma", &mdata->dai[0], + &mdata->dma_channel_id[0], + &mdata->dma_id[0]); if (ret) { dev_err(&pdev->dev, "missing/invalid playback DMA phandle\n"); goto error; @@ -472,9 +341,9 @@ static int p1022_ds_probe(struct platform_device *pdev) /* Find the capture DMA channel to use. */ mdata->dai[1].platform_name = mdata->platform_name[1]; - ret = get_dma_channel(np, "fsl,capture-dma", &mdata->dai[1], - &mdata->dma_channel_id[1], - &mdata->dma_id[1]); + ret = fsl_asoc_get_dma_channel(np, "fsl,capture-dma", &mdata->dai[1], + &mdata->dma_channel_id[1], + &mdata->dma_id[1]); if (ret) { dev_err(&pdev->dev, "missing/invalid capture DMA phandle\n"); goto error; -- cgit v1.2.3 From 8f549d7e7795e5e07ff871a79708bf2e387104dd Mon Sep 17 00:00:00 2001 From: Shawn Guo Date: Fri, 16 Mar 2012 16:56:41 +0800 Subject: ASoC: fsl: remove helper fsl_asoc_get_codec_dev_name The ASoC core now can support matching codec with device node besides name, so we can save helper function fsl_asoc_get_codec_dev_name. Signed-off-by: Shawn Guo Acked-by: Sascha Hauer Acked-by: Timur Tabi Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_utils.c | 44 -------------------------------------------- sound/soc/fsl/fsl_utils.h | 1 - sound/soc/fsl/mpc8610_hpcd.c | 13 ++----------- sound/soc/fsl/p1022_ds.c | 13 ++----------- 4 files changed, 4 insertions(+), 67 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_utils.c b/sound/soc/fsl/fsl_utils.c index 4370c28e5362..b9e42b503a37 100644 --- a/sound/soc/fsl/fsl_utils.c +++ b/sound/soc/fsl/fsl_utils.c @@ -12,54 +12,10 @@ #include #include -#include #include #include "fsl_utils.h" -/** - * fsl_asoc_get_codec_dev_name - determine the dev_name for a codec node - * - * @np: pointer to the I2C device tree node - * @buf: buffer to be filled with the dev_name of the I2C device - * @len: the length of the buffer - * - * This function determines the dev_name for an I2C node. This is the name - * that would be returned by dev_name() if this device_node were part of a - * 'struct device' It's ugly and hackish, but it works. - * - * The dev_name for such devices include the bus number and I2C address. For - * example, "cs4270.0-004f". - */ -int fsl_asoc_get_codec_dev_name(struct device_node *np, char *buf, size_t len) -{ - const u32 *iprop; - u32 addr; - char temp[DAI_NAME_SIZE]; - struct i2c_client *i2c; - - of_modalias_node(np, temp, DAI_NAME_SIZE); - - iprop = of_get_property(np, "reg", NULL); - if (!iprop) - return -EINVAL; - - addr = be32_to_cpup(iprop); - - /* We need the adapter number */ - i2c = of_find_i2c_device_by_node(np); - if (!i2c) { - put_device(&i2c->dev); - return -ENODEV; - } - - snprintf(buf, len, "%s.%u-%04x", temp, i2c->adapter->nr, addr); - put_device(&i2c->dev); - - return 0; -} -EXPORT_SYMBOL(fsl_asoc_get_codec_dev_name); - /** * fsl_asoc_get_dma_channel - determine the dma channel for a SSI node * diff --git a/sound/soc/fsl/fsl_utils.h b/sound/soc/fsl/fsl_utils.h index 44d143670c37..b2951126527c 100644 --- a/sound/soc/fsl/fsl_utils.h +++ b/sound/soc/fsl/fsl_utils.h @@ -18,7 +18,6 @@ struct snd_soc_dai_link; struct device_node; -int fsl_asoc_get_codec_dev_name(struct device_node *np, char *buf, size_t len); int fsl_asoc_get_dma_channel(struct device_node *ssi_np, const char *name, struct snd_soc_dai_link *dai, unsigned int *dma_channel_id, diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c index 41951820b50e..8fdc430a4f68 100644 --- a/sound/soc/fsl/mpc8610_hpcd.c +++ b/sound/soc/fsl/mpc8610_hpcd.c @@ -41,7 +41,6 @@ struct mpc8610_hpcd_data { unsigned int dma_id[2]; /* 0 = DMA1, 1 = DMA2, etc */ unsigned int dma_channel_id[2]; /* 0 = ch 0, 1 = ch 1, etc*/ char codec_dai_name[DAI_NAME_SIZE]; - char codec_name[DAI_NAME_SIZE]; char platform_name[2][DAI_NAME_SIZE]; /* One for each DMA channel */ }; @@ -215,16 +214,8 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev) machine_data->dai[0].cpu_dai_name = dev_name(&ssi_pdev->dev); machine_data->dai[0].ops = &mpc8610_hpcd_ops; - /* Determine the codec name, it will be used as the codec DAI name */ - ret = fsl_asoc_get_codec_dev_name(codec_np, machine_data->codec_name, - DAI_NAME_SIZE); - if (ret) { - dev_err(&pdev->dev, "invalid codec node %s\n", - codec_np->full_name); - ret = -EINVAL; - goto error; - } - machine_data->dai[0].codec_name = machine_data->codec_name; + /* ASoC core can match codec with device node */ + machine_data->dai[0].codec_of_node = codec_np; /* The DAI name from the codec (snd_soc_dai_driver.name) */ machine_data->dai[0].codec_dai_name = "cs4270-hifi"; diff --git a/sound/soc/fsl/p1022_ds.c b/sound/soc/fsl/p1022_ds.c index be79e7388c0f..202298556ba7 100644 --- a/sound/soc/fsl/p1022_ds.c +++ b/sound/soc/fsl/p1022_ds.c @@ -73,7 +73,6 @@ struct machine_data { unsigned int ssi_id; /* 0 = SSI1, 1 = SSI2, etc */ unsigned int dma_id[2]; /* 0 = DMA1, 1 = DMA2, etc */ unsigned int dma_channel_id[2]; /* 0 = ch 0, 1 = ch 1, etc*/ - char codec_name[DAI_NAME_SIZE]; char platform_name[2][DAI_NAME_SIZE]; /* One for each DMA channel */ }; @@ -225,16 +224,8 @@ static int p1022_ds_probe(struct platform_device *pdev) mdata->dai[0].cpu_dai_name = dev_name(&ssi_pdev->dev); mdata->dai[0].ops = &p1022_ds_ops; - /* Determine the codec name, it will be used as the codec DAI name */ - ret = fsl_asoc_get_codec_dev_name(codec_np, mdata->codec_name, - DAI_NAME_SIZE); - if (ret) { - dev_err(&pdev->dev, "invalid codec node %s\n", - codec_np->full_name); - ret = -EINVAL; - goto error; - } - mdata->dai[0].codec_name = mdata->codec_name; + /* ASoC core can match codec with device node */ + mdata->dai[0].codec_of_node = codec_np; /* We register two DAIs per SSI, one for playback and the other for * capture. We support codecs that have separate DAIs for both playback -- cgit v1.2.3 From dfa1a10785cf861c725aed0492f0ab69662bffea Mon Sep 17 00:00:00 2001 From: Shawn Guo Date: Fri, 16 Mar 2012 16:56:42 +0800 Subject: ASoC: fsl: make fsl_ssi driver compilable on ARM/IMX Provide different pair of accessors for accessing SSI registers on PowerPC and ARM/IMX, so that fsl_ssi driver can be built on both architectures. Signed-off-by: Shawn Guo Acked-by: Sascha Hauer Acked-by: Timur Tabi Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_ssi.c | 64 ++++++++++++++++++++++++++++++++----------------- 1 file changed, 42 insertions(+), 22 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 2eb407fa3b48..db9a73404385 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -11,11 +11,14 @@ */ #include +#include #include #include #include #include #include +#include +#include #include #include @@ -26,6 +29,25 @@ #include "fsl_ssi.h" +#ifdef PPC +#define read_ssi(addr) in_be32(addr) +#define write_ssi(val, addr) out_be32(addr, val) +#define write_ssi_mask(addr, clear, set) clrsetbits_be32(addr, clear, set) +#elif defined ARM +#define read_ssi(addr) readl(addr) +#define write_ssi(val, addr) writel(val, addr) +/* + * FIXME: Proper locking should be added at write_ssi_mask caller level + * to ensure this register read/modify/write sequence is race free. + */ +static inline void write_ssi_mask(u32 __iomem *addr, u32 clear, u32 set) +{ + u32 val = readl(addr); + val = (val & ~clear) | set; + writel(val, addr); +} +#endif + /** * FSLSSI_I2S_RATES: sample rates supported by the I2S * @@ -145,7 +167,7 @@ static irqreturn_t fsl_ssi_isr(int irq, void *dev_id) were interrupted for. We mask it with the Interrupt Enable register so that we only check for events that we're interested in. */ - sisr = in_be32(&ssi->sisr) & SIER_FLAGS; + sisr = read_ssi(&ssi->sisr) & SIER_FLAGS; if (sisr & CCSR_SSI_SISR_RFRC) { ssi_private->stats.rfrc++; @@ -260,7 +282,7 @@ static irqreturn_t fsl_ssi_isr(int irq, void *dev_id) /* Clear the bits that we set */ if (sisr2) - out_be32(&ssi->sisr, sisr2); + write_ssi(sisr2, &ssi->sisr); return ret; } @@ -295,7 +317,7 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, * SSI needs to be disabled before updating the registers we set * here. */ - clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN); + write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_SSIEN, 0); /* * Program the SSI into I2S Slave Non-Network Synchronous mode. @@ -303,20 +325,18 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, * * FIXME: Little-endian samples require a different shift dir */ - clrsetbits_be32(&ssi->scr, + write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_I2S_MODE_MASK | CCSR_SSI_SCR_SYN, CCSR_SSI_SCR_TFR_CLK_DIS | CCSR_SSI_SCR_I2S_MODE_SLAVE | (synchronous ? CCSR_SSI_SCR_SYN : 0)); - out_be32(&ssi->stcr, - CCSR_SSI_STCR_TXBIT0 | CCSR_SSI_STCR_TFEN0 | + write_ssi(CCSR_SSI_STCR_TXBIT0 | CCSR_SSI_STCR_TFEN0 | CCSR_SSI_STCR_TFSI | CCSR_SSI_STCR_TEFS | - CCSR_SSI_STCR_TSCKP); + CCSR_SSI_STCR_TSCKP, &ssi->stcr); - out_be32(&ssi->srcr, - CCSR_SSI_SRCR_RXBIT0 | CCSR_SSI_SRCR_RFEN0 | + write_ssi(CCSR_SSI_SRCR_RXBIT0 | CCSR_SSI_SRCR_RFEN0 | CCSR_SSI_SRCR_RFSI | CCSR_SSI_SRCR_REFS | - CCSR_SSI_SRCR_RSCKP); + CCSR_SSI_SRCR_RSCKP, &ssi->srcr); /* * The DC and PM bits are only used if the SSI is the clock @@ -324,7 +344,7 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, */ /* Enable the interrupts and DMA requests */ - out_be32(&ssi->sier, SIER_FLAGS); + write_ssi(SIER_FLAGS, &ssi->sier); /* * Set the watermark for transmit FIFI 0 and receive FIFO 0. We @@ -339,9 +359,9 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, * make this value larger (and maybe we should), but this way * data will be written to memory as soon as it's available. */ - out_be32(&ssi->sfcsr, - CCSR_SSI_SFCSR_TFWM0(ssi_private->fifo_depth - 2) | - CCSR_SSI_SFCSR_RFWM0(ssi_private->fifo_depth - 2)); + write_ssi(CCSR_SSI_SFCSR_TFWM0(ssi_private->fifo_depth - 2) | + CCSR_SSI_SFCSR_RFWM0(ssi_private->fifo_depth - 2), + &ssi->sfcsr); /* * We keep the SSI disabled because if we enable it, then the @@ -417,7 +437,7 @@ static int fsl_ssi_hw_params(struct snd_pcm_substream *substream, unsigned int sample_size = snd_pcm_format_width(params_format(hw_params)); u32 wl = CCSR_SSI_SxCCR_WL(sample_size); - int enabled = in_be32(&ssi->scr) & CCSR_SSI_SCR_SSIEN; + int enabled = read_ssi(&ssi->scr) & CCSR_SSI_SCR_SSIEN; /* * If we're in synchronous mode, and the SSI is already enabled, @@ -439,9 +459,9 @@ static int fsl_ssi_hw_params(struct snd_pcm_substream *substream, /* In synchronous mode, the SSI uses STCCR for capture */ if ((substream->stream == SNDRV_PCM_STREAM_PLAYBACK) || ssi_private->cpu_dai_drv.symmetric_rates) - clrsetbits_be32(&ssi->stccr, CCSR_SSI_SxCCR_WL_MASK, wl); + write_ssi_mask(&ssi->stccr, CCSR_SSI_SxCCR_WL_MASK, wl); else - clrsetbits_be32(&ssi->srccr, CCSR_SSI_SxCCR_WL_MASK, wl); + write_ssi_mask(&ssi->srccr, CCSR_SSI_SxCCR_WL_MASK, wl); return 0; } @@ -466,19 +486,19 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - setbits32(&ssi->scr, + write_ssi_mask(&ssi->scr, 0, CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_TE); else - setbits32(&ssi->scr, + write_ssi_mask(&ssi->scr, 0, CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_RE); break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - clrbits32(&ssi->scr, CCSR_SSI_SCR_TE); + write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_TE, 0); else - clrbits32(&ssi->scr, CCSR_SSI_SCR_RE); + write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_RE, 0); break; default: @@ -510,7 +530,7 @@ static void fsl_ssi_shutdown(struct snd_pcm_substream *substream, if (!ssi_private->first_stream) { struct ccsr_ssi __iomem *ssi = ssi_private->ssi; - clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN); + write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_SSIEN, 0); } } -- cgit v1.2.3 From 09ce1111f3893106463559ed62f27fe999ace5d6 Mon Sep 17 00:00:00 2001 From: Shawn Guo Date: Fri, 16 Mar 2012 16:56:43 +0800 Subject: ASoC: fsl: let fsl_ssi work with imx pcm and machine drivers Makes necessary changes on fsl_ssi to let it work with imx pcm and machine drivers. Signed-off-by: Shawn Guo Acked-by: Sascha Hauer Acked-by: Timur Tabi Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_ssi.c | 77 ++++++++++++++++++++++++++++++++++++++++++++----- 1 file changed, 70 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index db9a73404385..30ff605144c6 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -28,6 +28,7 @@ #include #include "fsl_ssi.h" +#include "imx-pcm.h" #ifdef PPC #define read_ssi(addr) in_be32(addr) @@ -116,6 +117,12 @@ struct fsl_ssi_private { struct device_attribute dev_attr; struct platform_device *pdev; + bool new_binding; + bool ssi_on_imx; + struct platform_device *imx_pcm_pdev; + struct imx_pcm_dma_params dma_params_tx; + struct imx_pcm_dma_params dma_params_rx; + struct { unsigned int rfrc; unsigned int tfrc; @@ -413,6 +420,12 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, ssi_private->second_stream = substream; } + if (ssi_private->ssi_on_imx) + snd_soc_dai_set_dma_data(dai, substream, + (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? + &ssi_private->dma_params_tx : + &ssi_private->dma_params_rx); + return 0; } @@ -642,12 +655,6 @@ static int __devinit fsl_ssi_probe(struct platform_device *pdev) if (!of_device_is_available(np)) return -ENODEV; - /* Check for a codec-handle property. */ - if (!of_get_property(np, "codec-handle", NULL)) { - dev_err(&pdev->dev, "missing codec-handle property\n"); - return -ENODEV; - } - /* We only support the SSI in "I2S Slave" mode */ sprop = of_get_property(np, "fsl,mode", NULL); if (!sprop || strcmp(sprop, "i2s-slave")) { @@ -712,6 +719,35 @@ static int __devinit fsl_ssi_probe(struct platform_device *pdev) /* Older 8610 DTs didn't have the fifo-depth property */ ssi_private->fifo_depth = 8; + if (of_device_is_compatible(pdev->dev.of_node, "fsl,imx21-ssi")) { + u32 dma_events[2]; + ssi_private->ssi_on_imx = true; + /* + * We have burstsize be "fifo_depth - 2" to match the SSI + * watermark setting in fsl_ssi_startup(). + */ + ssi_private->dma_params_tx.burstsize = + ssi_private->fifo_depth - 2; + ssi_private->dma_params_rx.burstsize = + ssi_private->fifo_depth - 2; + ssi_private->dma_params_tx.dma_addr = + ssi_private->ssi_phys + offsetof(struct ccsr_ssi, stx0); + ssi_private->dma_params_rx.dma_addr = + ssi_private->ssi_phys + offsetof(struct ccsr_ssi, srx0); + /* + * TODO: This is a temporary solution and should be changed + * to use generic DMA binding later when the helplers get in. + */ + ret = of_property_read_u32_array(pdev->dev.of_node, + "fsl,ssi-dma-events", dma_events, 2); + if (ret) { + dev_err(&pdev->dev, "could not get dma events\n"); + goto error_irq; + } + ssi_private->dma_params_tx.dma = dma_events[0]; + ssi_private->dma_params_rx.dma = dma_events[1]; + } + /* Initialize the the device_attribute structure */ dev_attr = &ssi_private->dev_attr; sysfs_attr_init(&dev_attr->attr); @@ -735,6 +771,26 @@ static int __devinit fsl_ssi_probe(struct platform_device *pdev) goto error_dev; } + if (ssi_private->ssi_on_imx) { + ssi_private->imx_pcm_pdev = + platform_device_register_simple("imx-pcm-audio", + -1, NULL, 0); + if (IS_ERR(ssi_private->imx_pcm_pdev)) { + ret = PTR_ERR(ssi_private->imx_pcm_pdev); + goto error_dev; + } + } + + /* + * If codec-handle property is missing from SSI node, we assume + * that the machine driver uses new binding which does not require + * SSI driver to trigger machine driver's probe. + */ + if (!of_get_property(np, "codec-handle", NULL)) { + ssi_private->new_binding = true; + goto done; + } + /* Trigger the machine driver's probe function. The platform driver * name of the machine driver is taken from /compatible property of the * device tree. We also pass the address of the CPU DAI driver @@ -756,9 +812,12 @@ static int __devinit fsl_ssi_probe(struct platform_device *pdev) goto error_dai; } +done: return 0; error_dai: + if (ssi_private->ssi_on_imx) + platform_device_unregister(ssi_private->imx_pcm_pdev); snd_soc_unregister_dai(&pdev->dev); error_dev: @@ -784,7 +843,10 @@ static int fsl_ssi_remove(struct platform_device *pdev) { struct fsl_ssi_private *ssi_private = dev_get_drvdata(&pdev->dev); - platform_device_unregister(ssi_private->pdev); + if (!ssi_private->new_binding) + platform_device_unregister(ssi_private->pdev); + if (ssi_private->ssi_on_imx) + platform_device_unregister(ssi_private->imx_pcm_pdev); snd_soc_unregister_dai(&pdev->dev); device_remove_file(&pdev->dev, &ssi_private->dev_attr); @@ -799,6 +861,7 @@ static int fsl_ssi_remove(struct platform_device *pdev) static const struct of_device_id fsl_ssi_ids[] = { { .compatible = "fsl,mpc8610-ssi", }, + { .compatible = "fsl,imx21-ssi", }, {} }; MODULE_DEVICE_TABLE(of, fsl_ssi_ids); -- cgit v1.2.3 From c448303e86c970cca4833bd9480c08f09b948b40 Mon Sep 17 00:00:00 2001 From: Shawn Guo Date: Fri, 16 Mar 2012 16:56:44 +0800 Subject: ASoC: fsl: add imx-sgtl5000 machine driver This is the initial imx-sgtl5000 machine driver support with only playback dai link implemented. More features can be added on top of it later. It's a device tree only machine driver working with fsl_ssi driver. Signed-off-by: Shawn Guo Acked-by: Sascha Hauer Acked-by: Timur Tabi Signed-off-by: Mark Brown --- .../bindings/sound/imx-audio-sgtl5000.txt | 24 +++ sound/soc/fsl/Kconfig | 12 ++ sound/soc/fsl/Makefile | 2 + sound/soc/fsl/imx-sgtl5000.c | 177 +++++++++++++++++++++ 4 files changed, 215 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-sgtl5000.txt create mode 100644 sound/soc/fsl/imx-sgtl5000.c (limited to 'sound') diff --git a/Documentation/devicetree/bindings/sound/imx-audio-sgtl5000.txt b/Documentation/devicetree/bindings/sound/imx-audio-sgtl5000.txt new file mode 100644 index 000000000000..421a374790b3 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/imx-audio-sgtl5000.txt @@ -0,0 +1,24 @@ +Freescale i.MX audio complex with SGTL5000 codec + +Required properties: +- compatible : "fsl,imx-audio-sgtl5000" +- model : The user-visible name of this sound complex +- ssi-controller : The phandle of the i.MX SSI controller +- audio-codec : The phandle of the SGTL5000 audio codec +- mux-int-port : The internal port of the i.MX audio muxer (AUDMUX) +- mux-ext-port : The external port of the i.MX audio muxer + +Note: The AUDMUX port numbering should start at 1, which is consistent with +hardware manual. + +Example: + +sound { + compatible = "fsl,imx51-babbage-sgtl5000", + "fsl,imx-audio-sgtl5000"; + model = "imx51-babbage-sgtl5000"; + ssi-controller = <&ssi1>; + audio-codec = <&sgtl5000>; + mux-int-port = <1>; + mux-ext-port = <3>; +}; diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 535ee73ecf15..aa14fc9744cc 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -164,4 +164,16 @@ config SND_SOC_EUKREA_TLV320 Enable I2S based access to the TLV320AIC23B codec attached to the SSI interface +config SND_SOC_IMX_SGTL5000 + tristate "SoC Audio support for i.MX boards with sgtl5000" + depends on OF && I2C + select SND_SOC_SGTL5000 + select SND_SOC_IMX_PCM_DMA + select SND_SOC_IMX_AUDMUX + select SND_SOC_FSL_SSI + select SND_SOC_FSL_UTILS + help + Say Y if you want to add support for SoC audio on an i.MX board with + a sgtl5000 codec. + endif # SND_IMX_SOC diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index fbdbed0af909..638d3852601a 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -40,8 +40,10 @@ snd-soc-eukrea-tlv320-objs := eukrea-tlv320.o snd-soc-phycore-ac97-objs := phycore-ac97.o snd-soc-mx27vis-aic32x4-objs := mx27vis-aic32x4.o snd-soc-wm1133-ev1-objs := wm1133-ev1.o +snd-soc-imx-sgtl5000-objs := imx-sgtl5000.o obj-$(CONFIG_SND_SOC_EUKREA_TLV320) += snd-soc-eukrea-tlv320.o obj-$(CONFIG_SND_SOC_PHYCORE_AC97) += snd-soc-phycore-ac97.o obj-$(CONFIG_SND_SOC_MX27VIS_AIC32X4) += snd-soc-mx27vis-aic32x4.o obj-$(CONFIG_SND_MXC_SOC_WM1133_EV1) += snd-soc-wm1133-ev1.o +obj-$(CONFIG_SND_SOC_IMX_SGTL5000) += snd-soc-imx-sgtl5000.o diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c new file mode 100644 index 000000000000..3786b61b4fea --- /dev/null +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -0,0 +1,177 @@ +/* + * Copyright 2012 Freescale Semiconductor, Inc. + * Copyright 2012 Linaro Ltd. + * + * The code contained herein is licensed under the GNU General Public + * License. You may obtain a copy of the GNU General Public License + * Version 2 or later at the following locations: + * + * http://www.opensource.org/licenses/gpl-license.html + * http://www.gnu.org/copyleft/gpl.html + */ + +#include +#include +#include +#include + +#include "../codecs/sgtl5000.h" +#include "imx-audmux.h" + +#define DAI_NAME_SIZE 32 + +struct imx_sgtl5000_data { + struct snd_soc_dai_link dai; + struct snd_soc_card card; + char codec_dai_name[DAI_NAME_SIZE]; + char platform_name[DAI_NAME_SIZE]; + unsigned int clk_frequency; +}; + +static int imx_sgtl5000_dai_init(struct snd_soc_pcm_runtime *rtd) +{ + struct imx_sgtl5000_data *data = container_of(rtd->card, + struct imx_sgtl5000_data, card); + struct device *dev = rtd->card->dev; + int ret; + + ret = snd_soc_dai_set_sysclk(rtd->codec_dai, SGTL5000_SYSCLK, + data->clk_frequency, SND_SOC_CLOCK_IN); + if (ret) { + dev_err(dev, "could not set codec driver clock params\n"); + return ret; + } + + return 0; +} + +static int __devinit imx_sgtl5000_probe(struct platform_device *pdev) +{ + struct device_node *np = pdev->dev.of_node; + struct device_node *ssi_np, *codec_np; + struct platform_device *ssi_pdev; + struct imx_sgtl5000_data *data; + int int_port, ext_port; + int ret; + + ret = of_property_read_u32(np, "mux-int-port", &int_port); + if (ret) { + dev_err(&pdev->dev, "mux-int-port missing or invalid\n"); + return ret; + } + ret = of_property_read_u32(np, "mux-ext-port", &ext_port); + if (ret) { + dev_err(&pdev->dev, "mux-ext-port missing or invalid\n"); + return ret; + } + + /* + * The port numbering in the hardware manual starts at 1, while + * the audmux API expects it starts at 0. + */ + int_port--; + ext_port--; + ret = imx_audmux_v2_configure_port(int_port, + IMX_AUDMUX_V2_PTCR_SYN | + IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) | + IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) | + IMX_AUDMUX_V2_PTCR_TFSDIR | + IMX_AUDMUX_V2_PTCR_TCLKDIR, + IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port)); + if (ret) { + dev_err(&pdev->dev, "audmux internal port setup failed\n"); + return ret; + } + imx_audmux_v2_configure_port(ext_port, + IMX_AUDMUX_V2_PTCR_SYN | + IMX_AUDMUX_V2_PTCR_TCSEL(int_port), + IMX_AUDMUX_V2_PDCR_RXDSEL(int_port)); + if (ret) { + dev_err(&pdev->dev, "audmux external port setup failed\n"); + return ret; + } + + ssi_np = of_parse_phandle(pdev->dev.of_node, "ssi-controller", 0); + codec_np = of_parse_phandle(pdev->dev.of_node, "audio-codec", 0); + if (!ssi_np || !codec_np) { + dev_err(&pdev->dev, "phandle missing or invalid\n"); + return -EINVAL; + } + + ssi_pdev = of_find_device_by_node(ssi_np); + if (!ssi_pdev) { + dev_err(&pdev->dev, "failed to find SSI platform device\n"); + return -EINVAL; + } + + data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL); + if (!data) + return -ENOMEM; + + ret = of_property_read_u32(codec_np, "clock-frequency", + &data->clk_frequency); + if (ret) { + dev_err(&pdev->dev, "clock-frequency missing or invalid\n"); + return ret; + } + + data->dai.name = "HiFi"; + data->dai.stream_name = "HiFi"; + data->dai.codec_dai_name = "sgtl5000"; + data->dai.codec_of_node = codec_np; + data->dai.cpu_dai_name = dev_name(&ssi_pdev->dev); + data->dai.platform_name = "imx-pcm-audio"; + data->dai.init = &imx_sgtl5000_dai_init; + data->dai.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM; + + data->card.dev = &pdev->dev; + ret = snd_soc_of_parse_card_name(&data->card, "model"); + if (ret) + return ret; + data->card.num_links = 1; + data->card.dai_link = &data->dai; + + ret = snd_soc_register_card(&data->card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); + return ret; + } + + platform_set_drvdata(pdev, data); + of_node_put(ssi_np); + of_node_put(codec_np); + + return 0; +} + +static int __devexit imx_sgtl5000_remove(struct platform_device *pdev) +{ + struct imx_sgtl5000_data *data = platform_get_drvdata(pdev); + + snd_soc_unregister_card(&data->card); + + return 0; +} + +static const struct of_device_id imx_sgtl5000_dt_ids[] = { + { .compatible = "fsl,imx-audio-sgtl5000", }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, imx_sgtl5000_dt_ids); + +static struct platform_driver imx_sgtl5000_driver = { + .driver = { + .name = "imx-sgtl5000", + .owner = THIS_MODULE, + .of_match_table = imx_sgtl5000_dt_ids, + }, + .probe = imx_sgtl5000_probe, + .remove = __devexit_p(imx_sgtl5000_remove), +}; +module_platform_driver(imx_sgtl5000_driver); + +MODULE_AUTHOR("Shawn Guo "); +MODULE_DESCRIPTION("Freescale i.MX SGTL5000 ASoC machine driver"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:imx-sgtl5000"); -- cgit v1.2.3 From 497098beffaa898ea9fa0076e626f055ef5c832e Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 8 Mar 2012 15:06:09 +0000 Subject: ASoC: dapm: Remove bodges for no-widget CODECs Now that we're creating widgets for all DAIs there should be no more need for the bodges we've been carrying for non-DAPM CODEC drivers so remove them. Signed-off-by: Mark Brown Acked-by: Timur Tabi Acked-by: Liam Girdwood --- sound/soc/soc-dapm.c | 36 ++++-------------------------------- 1 file changed, 4 insertions(+), 32 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index dd603fa38097..69e9452386e6 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1441,12 +1441,10 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) trace_snd_soc_dapm_start(card); list_for_each_entry(d, &card->dapm_list, list) { - if (d->n_widgets || d->codec == NULL) { - if (d->idle_bias_off) - d->target_bias_level = SND_SOC_BIAS_OFF; - else - d->target_bias_level = SND_SOC_BIAS_STANDBY; - } + if (d->idle_bias_off) + d->target_bias_level = SND_SOC_BIAS_OFF; + else + d->target_bias_level = SND_SOC_BIAS_STANDBY; } dapm_reset(card); @@ -1491,32 +1489,6 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) } - /* If there are no DAPM widgets then try to figure out power from the - * event type. - */ - if (!dapm->n_widgets) { - switch (event) { - case SND_SOC_DAPM_STREAM_START: - case SND_SOC_DAPM_STREAM_RESUME: - dapm->target_bias_level = SND_SOC_BIAS_ON; - break; - case SND_SOC_DAPM_STREAM_STOP: - if (dapm->codec && dapm->codec->active) - dapm->target_bias_level = SND_SOC_BIAS_ON; - else - dapm->target_bias_level = SND_SOC_BIAS_STANDBY; - break; - case SND_SOC_DAPM_STREAM_SUSPEND: - dapm->target_bias_level = SND_SOC_BIAS_STANDBY; - break; - case SND_SOC_DAPM_STREAM_NOP: - dapm->target_bias_level = dapm->bias_level; - break; - default: - break; - } - } - /* Force all contexts in the card to the same bias state if * they're not ground referenced. */ -- cgit v1.2.3 From ab92d09d1306c738b751b839d81e867af1039d14 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 19 Mar 2012 16:15:43 +0000 Subject: ASoC: cs4270: Check that we can enable regulators on resume It's possible that the regulator enable will fail and if it does we may as well just give up with trying to bring the rest of the device up and report the original error. Signed-off-by: Mark Brown Acked-by: Timur Tabi --- sound/soc/codecs/cs4270.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 1d672f528662..df8fe387a66f 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -600,10 +600,12 @@ static int cs4270_soc_suspend(struct snd_soc_codec *codec) static int cs4270_soc_resume(struct snd_soc_codec *codec) { struct cs4270_private *cs4270 = snd_soc_codec_get_drvdata(codec); - int reg; + int reg, ret; - regulator_bulk_enable(ARRAY_SIZE(cs4270->supplies), - cs4270->supplies); + ret = regulator_bulk_enable(ARRAY_SIZE(cs4270->supplies), + cs4270->supplies); + if (ret != 0) + return ret; /* In case the device was put to hard reset during sleep, we need to * wait 500ns here before any I2C communication. */ -- cgit v1.2.3 From d808fe9f3e7f4092580c3294692bb801369b9c9f Mon Sep 17 00:00:00 2001 From: Tomoya MORINAGA Date: Mon, 19 Mar 2012 20:59:28 +0900 Subject: ASoC: Add LAPIS Semiconductor ML26124 driver ML26124-01HB/ML26124-02GD is 16bit monaural audio CODEC which has high resistance to voltage noise. On chip regulator realizes power supply rejection ratio be over 90dB so more than 50dB is improved than ever. ML26124-01HB/ ML26124-02GD can deliver stable audio performance without being affected by noise from the power supply circuit and peripheral components. The chip also includes a composite video signal output, which can be applied to various portable device requirements. The ML26124 is realized these functions into very small package the size is only 2.56mm x 2.46mm therefore can be construct high quality sound system easily. ML26124-01HB is 25pin WCSP package; ML26124-02GD is 32pin WQFN package. Signed-off-by: Tomoya MORINAGA Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ml26124.c | 681 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/ml26124.h | 184 ++++++++++++ 4 files changed, 871 insertions(+) create mode 100644 sound/soc/codecs/ml26124.c create mode 100644 sound/soc/codecs/ml26124.h (limited to 'sound') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 6508e8b790bb..e314a66b30cd 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -42,6 +42,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MAX9850 if I2C select SND_SOC_MAX9768 if I2C select SND_SOC_MAX9877 if I2C + select SND_SOC_ML26124 if I2C select SND_SOC_PCM3008 select SND_SOC_RT5631 if I2C select SND_SOC_SGTL5000 if I2C @@ -436,5 +437,8 @@ config SND_SOC_MAX9768 config SND_SOC_MAX9877 tristate +config SND_SOC_ML26124 + tristate + config SND_SOC_TPA6130A2 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 6662eb0cdcc0..9108ee992395 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -29,6 +29,7 @@ snd-soc-max9768-objs := max9768.o snd-soc-max98088-objs := max98088.o snd-soc-max98095-objs := max98095.o snd-soc-max9850-objs := max9850.o +snd-soc-ml26124-objs := ml26124.o snd-soc-pcm3008-objs := pcm3008.o snd-soc-rt5631-objs := rt5631.o snd-soc-sgtl5000-objs := sgtl5000.o @@ -135,6 +136,7 @@ obj-$(CONFIG_SND_SOC_MAX9768) += snd-soc-max9768.o obj-$(CONFIG_SND_SOC_MAX98088) += snd-soc-max98088.o obj-$(CONFIG_SND_SOC_MAX98095) += snd-soc-max98095.o obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o +obj-$(CONFIG_SND_SOC_ML26124) += snd-soc-ml26124.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o obj-$(CONFIG_SND_SOC_RT5631) += snd-soc-rt5631.o obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o diff --git a/sound/soc/codecs/ml26124.c b/sound/soc/codecs/ml26124.c new file mode 100644 index 000000000000..22cb5bf59273 --- /dev/null +++ b/sound/soc/codecs/ml26124.c @@ -0,0 +1,681 @@ +/* + * Copyright (C) 2011 LAPIS Semiconductor Co., Ltd. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307, USA. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include "ml26124.h" + +#define DVOL_CTL_DVMUTE_ON BIT(4) /* Digital volume MUTE On */ +#define DVOL_CTL_DVMUTE_OFF 0 /* Digital volume MUTE Off */ +#define ML26124_SAI_NO_DELAY BIT(1) +#define ML26124_SAI_FRAME_SYNC (BIT(5) | BIT(0)) /* For mono (Telecodec) */ +#define ML26134_CACHESIZE 212 +#define ML26124_VMID BIT(1) +#define ML26124_RATES (SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_32000 |\ + SNDRV_PCM_RATE_48000) +#define ML26124_FORMATS (SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S32_LE) +#define ML26124_NUM_REGISTER ML26134_CACHESIZE + +struct ml26124_priv { + u32 mclk; + u32 rate; + struct regmap *regmap; + int clk_in; + struct snd_pcm_substream *substream; +}; + +struct clk_coeff { + u32 mclk; + u32 rate; + u8 pllnl; + u8 pllnh; + u8 pllml; + u8 pllmh; + u8 plldiv; +}; + +/* ML26124 configuration */ +static const DECLARE_TLV_DB_SCALE(digital_tlv, -7150, 50, 0); + +static const DECLARE_TLV_DB_SCALE(alclvl, -2250, 150, 0); +static const DECLARE_TLV_DB_SCALE(mingain, -1200, 600, 0); +static const DECLARE_TLV_DB_SCALE(maxgain, -675, 600, 0); +static const DECLARE_TLV_DB_SCALE(boost_vol, -1200, 75, 0); +static const DECLARE_TLV_DB_SCALE(ngth, -7650, 150, 0); + +static const char * const ml26124_companding[] = {"16bit PCM", "u-law", + "A-law"}; + +static const struct soc_enum ml26124_adc_companding_enum + = SOC_ENUM_SINGLE(ML26124_SAI_TRANS_CTL, 6, 3, ml26124_companding); + +static const struct soc_enum ml26124_dac_companding_enum + = SOC_ENUM_SINGLE(ML26124_SAI_RCV_CTL, 6, 3, ml26124_companding); + +static const struct snd_kcontrol_new ml26124_snd_controls[] = { + SOC_SINGLE_TLV("Capture Digital Volume", ML26124_RECORD_DIG_VOL, 0, + 0xff, 1, digital_tlv), + SOC_SINGLE_TLV("Playback Digital Volume", ML26124_PLBAK_DIG_VOL, 0, + 0xff, 1, digital_tlv), + SOC_SINGLE_TLV("Digital Boost Volume", ML26124_DIGI_BOOST_VOL, 0, + 0x3f, 0, boost_vol), + SOC_SINGLE_TLV("EQ Band0 Volume", ML26124_EQ_GAIN_BRAND0, 0, + 0xff, 1, digital_tlv), + SOC_SINGLE_TLV("EQ Band1 Volume", ML26124_EQ_GAIN_BRAND1, 0, + 0xff, 1, digital_tlv), + SOC_SINGLE_TLV("EQ Band2 Volume", ML26124_EQ_GAIN_BRAND2, 0, + 0xff, 1, digital_tlv), + SOC_SINGLE_TLV("EQ Band3 Volume", ML26124_EQ_GAIN_BRAND3, 0, + 0xff, 1, digital_tlv), + SOC_SINGLE_TLV("EQ Band4 Volume", ML26124_EQ_GAIN_BRAND4, 0, + 0xff, 1, digital_tlv), + SOC_SINGLE_TLV("ALC Target Level", ML26124_ALC_TARGET_LEV, 0, + 0xf, 1, alclvl), + SOC_SINGLE_TLV("ALC Min Input Volume", ML26124_ALC_MAXMIN_GAIN, 0, + 7, 0, mingain), + SOC_SINGLE_TLV("ALC Max Input Volume", ML26124_ALC_MAXMIN_GAIN, 4, + 7, 1, maxgain), + SOC_SINGLE_TLV("Playback Limiter Min Input Volume", + ML26124_PL_MAXMIN_GAIN, 0, 7, 0, mingain), + SOC_SINGLE_TLV("Playback Limiter Max Input Volume", + ML26124_PL_MAXMIN_GAIN, 4, 7, 1, maxgain), + SOC_SINGLE_TLV("Playback Boost Volume", ML26124_PLYBAK_BOST_VOL, 0, + 0x3f, 0, boost_vol), + SOC_SINGLE("DC High Pass Filter Switch", ML26124_FILTER_EN, 0, 1, 0), + SOC_SINGLE("Noise High Pass Filter Switch", ML26124_FILTER_EN, 1, 1, 0), + SOC_SINGLE("ZC Switch", ML26124_PW_ZCCMP_PW_MNG, 1, + 1, 0), + SOC_SINGLE("EQ Band0 Switch", ML26124_FILTER_EN, 2, 1, 0), + SOC_SINGLE("EQ Band1 Switch", ML26124_FILTER_EN, 3, 1, 0), + SOC_SINGLE("EQ Band2 Switch", ML26124_FILTER_EN, 4, 1, 0), + SOC_SINGLE("EQ Band3 Switch", ML26124_FILTER_EN, 5, 1, 0), + SOC_SINGLE("EQ Band4 Switch", ML26124_FILTER_EN, 6, 1, 0), + SOC_SINGLE("Play Limiter", ML26124_DVOL_CTL, 0, 1, 0), + SOC_SINGLE("Capture Limiter", ML26124_DVOL_CTL, 1, 1, 0), + SOC_SINGLE("Digital Volume Fade Switch", ML26124_DVOL_CTL, 3, 1, 0), + SOC_SINGLE("Digital Switch", ML26124_DVOL_CTL, 4, 1, 0), + SOC_ENUM("DAC Companding", ml26124_dac_companding_enum), + SOC_ENUM("ADC Companding", ml26124_adc_companding_enum), +}; + +static const struct snd_kcontrol_new ml26124_output_mixer_controls[] = { + SOC_DAPM_SINGLE("DAC Switch", ML26124_SPK_AMP_OUT, 1, 1, 0), + SOC_DAPM_SINGLE("Line in loopback Switch", ML26124_SPK_AMP_OUT, 3, 1, + 0), + SOC_DAPM_SINGLE("PGA Switch", ML26124_SPK_AMP_OUT, 5, 1, 0), +}; + +/* Input mux */ +static const char * const ml26124_input_select[] = {"Analog MIC SingleEnded in", + "Digital MIC in", "Analog MIC Differential in"}; + +static const struct soc_enum ml26124_insel_enum = + SOC_ENUM_SINGLE(ML26124_MIC_IF_CTL, 0, 3, ml26124_input_select); + +static const struct snd_kcontrol_new ml26124_input_mux_controls = + SOC_DAPM_ENUM("Input Select", ml26124_insel_enum); + +static const struct snd_kcontrol_new ml26124_line_control = + SOC_DAPM_SINGLE("Switch", ML26124_PW_LOUT_PW_MNG, 1, 1, 0); + +static const struct snd_soc_dapm_widget ml26124_dapm_widgets[] = { + SND_SOC_DAPM_SUPPLY("MCLKEN", ML26124_CLK_EN, 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("PLLEN", ML26124_CLK_EN, 1, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("PLLOE", ML26124_CLK_EN, 2, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("MICBIAS", ML26124_PW_REF_PW_MNG, 2, 0, NULL, 0), + SND_SOC_DAPM_MIXER("Output Mixer", SND_SOC_NOPM, 0, 0, + &ml26124_output_mixer_controls[0], + ARRAY_SIZE(ml26124_output_mixer_controls)), + SND_SOC_DAPM_DAC("DAC", "Playback", ML26124_PW_DAC_PW_MNG, 1, 0), + SND_SOC_DAPM_ADC("ADC", "Capture", ML26124_PW_IN_PW_MNG, 1, 0), + SND_SOC_DAPM_PGA("PGA", ML26124_PW_IN_PW_MNG, 3, 0, NULL, 0), + SND_SOC_DAPM_MUX("Input Mux", SND_SOC_NOPM, 0, 0, + &ml26124_input_mux_controls), + SND_SOC_DAPM_SWITCH("Line Out Enable", SND_SOC_NOPM, 0, 0, + &ml26124_line_control), + SND_SOC_DAPM_INPUT("MDIN"), + SND_SOC_DAPM_INPUT("MIN"), + SND_SOC_DAPM_INPUT("LIN"), + SND_SOC_DAPM_OUTPUT("SPOUT"), + SND_SOC_DAPM_OUTPUT("LOUT"), +}; + +static const struct snd_soc_dapm_route ml26124_intercon[] = { + /* Supply */ + {"DAC", NULL, "MCLKEN"}, + {"ADC", NULL, "MCLKEN"}, + {"DAC", NULL, "PLLEN"}, + {"ADC", NULL, "PLLEN"}, + {"DAC", NULL, "PLLOE"}, + {"ADC", NULL, "PLLOE"}, + + /* output mixer */ + {"Output Mixer", "DAC Switch", "DAC"}, + {"Output Mixer", "Line in loopback Switch", "LIN"}, + + /* outputs */ + {"LOUT", NULL, "Output Mixer"}, + {"SPOUT", NULL, "Output Mixer"}, + {"Line Out Enable", NULL, "LOUT"}, + + /* input */ + {"ADC", NULL, "Input Mux"}, + {"Input Mux", "Analog MIC SingleEnded in", "PGA"}, + {"Input Mux", "Analog MIC Differential in", "PGA"}, + {"PGA", NULL, "MIN"}, +}; + +/* PLLOutputFreq(Hz) = InputMclkFreq(Hz) * PLLM / (PLLN * PLLDIV) */ +static const struct clk_coeff coeff_div[] = { + {12288000, 16000, 0xc, 0x0, 0x20, 0x0, 0x4}, + {12288000, 32000, 0xc, 0x0, 0x20, 0x0, 0x4}, + {12288000, 48000, 0xc, 0x0, 0x30, 0x0, 0x4}, +}; + +static struct reg_default ml26124_reg[] = { + /* CLOCK control Register */ + {0x00, 0x00 }, /* Sampling Rate */ + {0x02, 0x00}, /* PLL NL */ + {0x04, 0x00}, /* PLLNH */ + {0x06, 0x00}, /* PLLML */ + {0x08, 0x00}, /* MLLMH */ + {0x0a, 0x00}, /* PLLDIV */ + {0x0c, 0x00}, /* Clock Enable */ + {0x0e, 0x00}, /* CLK Input/Output Control */ + + /* System Control Register */ + {0x10, 0x00}, /* Software RESET */ + {0x12, 0x00}, /* Record/Playback Run */ + {0x14, 0x00}, /* Mic Input/Output control */ + + /* Power Management Register */ + {0x20, 0x00}, /* Reference Power Management */ + {0x22, 0x00}, /* Input Power Management */ + {0x24, 0x00}, /* DAC Power Management */ + {0x26, 0x00}, /* SP-AMP Power Management */ + {0x28, 0x00}, /* LINEOUT Power Management */ + {0x2a, 0x00}, /* VIDEO Power Management */ + {0x2e, 0x00}, /* AC-CMP Power Management */ + + /* Analog reference Control Register */ + {0x30, 0x04}, /* MICBIAS Voltage Control */ + + /* Input/Output Amplifier Control Register */ + {0x32, 0x10}, /* MIC Input Volume */ + {0x38, 0x00}, /* Mic Boost Volume */ + {0x3a, 0x33}, /* Speaker AMP Volume */ + {0x48, 0x00}, /* AMP Volume Control Function Enable */ + {0x4a, 0x00}, /* Amplifier Volume Fader Control */ + + /* Analog Path Control Register */ + {0x54, 0x00}, /* Speaker AMP Output Control */ + {0x5a, 0x00}, /* Mic IF Control */ + {0xe8, 0x01}, /* Mic Select Control */ + + /* Audio Interface Control Register */ + {0x60, 0x00}, /* SAI-Trans Control */ + {0x62, 0x00}, /* SAI-Receive Control */ + {0x64, 0x00}, /* SAI Mode select */ + + /* DSP Control Register */ + {0x66, 0x01}, /* Filter Func Enable */ + {0x68, 0x00}, /* Volume Control Func Enable */ + {0x6A, 0x00}, /* Mixer & Volume Control*/ + {0x6C, 0xff}, /* Record Digital Volume */ + {0x70, 0xff}, /* Playback Digital Volume */ + {0x72, 0x10}, /* Digital Boost Volume */ + {0x74, 0xe7}, /* EQ gain Band0 */ + {0x76, 0xe7}, /* EQ gain Band1 */ + {0x78, 0xe7}, /* EQ gain Band2 */ + {0x7A, 0xe7}, /* EQ gain Band3 */ + {0x7C, 0xe7}, /* EQ gain Band4 */ + {0x7E, 0x00}, /* HPF2 CutOff*/ + {0x80, 0x00}, /* EQ Band0 Coef0L */ + {0x82, 0x00}, /* EQ Band0 Coef0H */ + {0x84, 0x00}, /* EQ Band0 Coef0L */ + {0x86, 0x00}, /* EQ Band0 Coef0H */ + {0x88, 0x00}, /* EQ Band1 Coef0L */ + {0x8A, 0x00}, /* EQ Band1 Coef0H */ + {0x8C, 0x00}, /* EQ Band1 Coef0L */ + {0x8E, 0x00}, /* EQ Band1 Coef0H */ + {0x90, 0x00}, /* EQ Band2 Coef0L */ + {0x92, 0x00}, /* EQ Band2 Coef0H */ + {0x94, 0x00}, /* EQ Band2 Coef0L */ + {0x96, 0x00}, /* EQ Band2 Coef0H */ + {0x98, 0x00}, /* EQ Band3 Coef0L */ + {0x9A, 0x00}, /* EQ Band3 Coef0H */ + {0x9C, 0x00}, /* EQ Band3 Coef0L */ + {0x9E, 0x00}, /* EQ Band3 Coef0H */ + {0xA0, 0x00}, /* EQ Band4 Coef0L */ + {0xA2, 0x00}, /* EQ Band4 Coef0H */ + {0xA4, 0x00}, /* EQ Band4 Coef0L */ + {0xA6, 0x00}, /* EQ Band4 Coef0H */ + + /* ALC Control Register */ + {0xb0, 0x00}, /* ALC Mode */ + {0xb2, 0x02}, /* ALC Attack Time */ + {0xb4, 0x03}, /* ALC Decay Time */ + {0xb6, 0x00}, /* ALC Hold Time */ + {0xb8, 0x0b}, /* ALC Target Level */ + {0xba, 0x70}, /* ALC Max/Min Gain */ + {0xbc, 0x00}, /* Noise Gate Threshold */ + {0xbe, 0x00}, /* ALC ZeroCross TimeOut */ + + /* Playback Limiter Control Register */ + {0xc0, 0x04}, /* PL Attack Time */ + {0xc2, 0x05}, /* PL Decay Time */ + {0xc4, 0x0d}, /* PL Target Level */ + {0xc6, 0x70}, /* PL Max/Min Gain */ + {0xc8, 0x10}, /* Playback Boost Volume */ + {0xca, 0x00}, /* PL ZeroCross TimeOut */ + + /* Video Amplifier Control Register */ + {0xd0, 0x01}, /* VIDEO AMP Gain Control */ + {0xd2, 0x01}, /* VIDEO AMP Setup 1 */ + {0xd4, 0x01}, /* VIDEO AMP Control2 */ +}; + +/* Get sampling rate value of sampling rate setting register (0x0) */ +static inline int get_srate(int rate) +{ + int srate; + + switch (rate) { + case 16000: + srate = 3; + break; + case 32000: + srate = 6; + break; + case 48000: + srate = 8; + break; + default: + return -EINVAL; + } + return srate; +} + +static inline int get_coeff(int mclk, int rate) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(coeff_div); i++) { + if (coeff_div[i].rate == rate && coeff_div[i].mclk == mclk) + return i; + } + return -EINVAL; +} + +static int ml26124_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct ml26124_priv *priv = snd_soc_codec_get_drvdata(codec); + int i = get_coeff(priv->mclk, params_rate(hw_params)); + + priv->substream = substream; + priv->rate = params_rate(hw_params); + + if (priv->clk_in) { + switch (priv->mclk / params_rate(hw_params)) { + case 256: + snd_soc_update_bits(codec, ML26124_CLK_CTL, + BIT(0) | BIT(1), 1); + break; + case 512: + snd_soc_update_bits(codec, ML26124_CLK_CTL, + BIT(0) | BIT(1), 2); + break; + case 1024: + snd_soc_update_bits(codec, ML26124_CLK_CTL, + BIT(0) | BIT(1), 3); + break; + default: + dev_err(codec->dev, "Unsupported MCLKI\n"); + break; + } + } else { + snd_soc_update_bits(codec, ML26124_CLK_CTL, + BIT(0) | BIT(1), 0); + } + + switch (params_rate(hw_params)) { + case 16000: + snd_soc_update_bits(codec, ML26124_SMPLING_RATE, 0xf, + get_srate(params_rate(hw_params))); + snd_soc_update_bits(codec, ML26124_PLLNL, 0xff, + coeff_div[i].pllnl); + snd_soc_update_bits(codec, ML26124_PLLNH, 0x1, + coeff_div[i].pllnh); + snd_soc_update_bits(codec, ML26124_PLLML, 0xff, + coeff_div[i].pllml); + snd_soc_update_bits(codec, ML26124_PLLMH, 0x3f, + coeff_div[i].pllmh); + snd_soc_update_bits(codec, ML26124_PLLDIV, 0x1f, + coeff_div[i].plldiv); + break; + case 32000: + snd_soc_update_bits(codec, ML26124_SMPLING_RATE, 0xf, + get_srate(params_rate(hw_params))); + snd_soc_update_bits(codec, ML26124_PLLNL, 0xff, + coeff_div[i].pllnl); + snd_soc_update_bits(codec, ML26124_PLLNH, 0x1, + coeff_div[i].pllnh); + snd_soc_update_bits(codec, ML26124_PLLML, 0xff, + coeff_div[i].pllml); + snd_soc_update_bits(codec, ML26124_PLLMH, 0x3f, + coeff_div[i].pllmh); + snd_soc_update_bits(codec, ML26124_PLLDIV, 0x1f, + coeff_div[i].plldiv); + break; + case 48000: + snd_soc_update_bits(codec, ML26124_SMPLING_RATE, 0xf, + get_srate(params_rate(hw_params))); + snd_soc_update_bits(codec, ML26124_PLLNL, 0xff, + coeff_div[i].pllnl); + snd_soc_update_bits(codec, ML26124_PLLNH, 0x1, + coeff_div[i].pllnh); + snd_soc_update_bits(codec, ML26124_PLLML, 0xff, + coeff_div[i].pllml); + snd_soc_update_bits(codec, ML26124_PLLMH, 0x3f, + coeff_div[i].pllmh); + snd_soc_update_bits(codec, ML26124_PLLDIV, 0x1f, + coeff_div[i].plldiv); + break; + default: + pr_err("%s:this rate is no support for ml26124\n", __func__); + return -EINVAL; + } + + return 0; +} + +static int ml26124_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + struct ml26124_priv *priv = snd_soc_codec_get_drvdata(codec); + + switch (priv->substream->stream) { + case SNDRV_PCM_STREAM_CAPTURE: + snd_soc_update_bits(codec, ML26124_REC_PLYBAK_RUN, BIT(0), 1); + break; + case SNDRV_PCM_STREAM_PLAYBACK: + snd_soc_update_bits(codec, ML26124_REC_PLYBAK_RUN, BIT(1), 2); + break; + } + + if (mute) + snd_soc_update_bits(codec, ML26124_DVOL_CTL, BIT(4), + DVOL_CTL_DVMUTE_ON); + else + snd_soc_update_bits(codec, ML26124_DVOL_CTL, BIT(4), + DVOL_CTL_DVMUTE_OFF); + + return 0; +} + +static int ml26124_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + unsigned char mode; + struct snd_soc_codec *codec = codec_dai->codec; + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + mode = 1; + break; + case SND_SOC_DAIFMT_CBS_CFS: + mode = 0; + break; + default: + return -EINVAL; + } + snd_soc_update_bits(codec, ML26124_SAI_MODE_SEL, BIT(0), mode); + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + default: + return -EINVAL; + } + + /* clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + default: + return -EINVAL; + } + + return 0; +} + +static int ml26124_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct ml26124_priv *priv = snd_soc_codec_get_drvdata(codec); + + switch (clk_id) { + case ML26124_USE_PLLOUT: + priv->clk_in = ML26124_USE_PLLOUT; + break; + case ML26124_USE_MCLKI: + priv->clk_in = ML26124_USE_MCLKI; + break; + default: + return -EINVAL; + } + + priv->mclk = freq; + + return 0; +} + +static int ml26124_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct ml26124_priv *priv = snd_soc_codec_get_drvdata(codec); + + switch (level) { + case SND_SOC_BIAS_ON: + snd_soc_update_bits(codec, ML26124_PW_SPAMP_PW_MNG, + ML26124_R26_MASK, ML26124_BLT_PREAMP_ON); + msleep(100); + snd_soc_update_bits(codec, ML26124_PW_SPAMP_PW_MNG, + ML26124_R26_MASK, + ML26124_MICBEN_ON | ML26124_BLT_ALL_ON); + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + /* VMID ON */ + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + snd_soc_update_bits(codec, ML26124_PW_REF_PW_MNG, + ML26124_VMID, ML26124_VMID); + msleep(500); + regcache_sync(priv->regmap); + } + break; + case SND_SOC_BIAS_OFF: + /* VMID OFF */ + snd_soc_update_bits(codec, ML26124_PW_REF_PW_MNG, + ML26124_VMID, 0); + break; + } + codec->dapm.bias_level = level; + return 0; +} + +static const struct snd_soc_dai_ops ml26124_dai_ops = { + .hw_params = ml26124_hw_params, + .digital_mute = ml26124_mute, + .set_fmt = ml26124_set_dai_fmt, + .set_sysclk = ml26124_set_dai_sysclk, +}; + +static struct snd_soc_dai_driver ml26124_dai = { + .name = "ml26124-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = ML26124_RATES, + .formats = ML26124_FORMATS,}, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = ML26124_RATES, + .formats = ML26124_FORMATS,}, + .ops = &ml26124_dai_ops, + .symmetric_rates = 1, +}; + +#ifdef CONFIG_PM +static int ml26124_suspend(struct snd_soc_codec *codec) +{ + ml26124_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int ml26124_resume(struct snd_soc_codec *codec) +{ + ml26124_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return 0; +} +#else +#define ml26124_suspend NULL +#define ml26124_resume NULL +#endif + +static int ml26124_probe(struct snd_soc_codec *codec) +{ + int ret; + struct ml26124_priv *priv = snd_soc_codec_get_drvdata(codec); + codec->control_data = priv->regmap; + + ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + /* Software Reset */ + snd_soc_update_bits(codec, ML26124_SW_RST, 0x01, 1); + snd_soc_update_bits(codec, ML26124_SW_RST, 0x01, 0); + + ml26124_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_ml26124 = { + .probe = ml26124_probe, + .suspend = ml26124_suspend, + .resume = ml26124_resume, + .set_bias_level = ml26124_set_bias_level, + .dapm_widgets = ml26124_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ml26124_dapm_widgets), + .dapm_routes = ml26124_intercon, + .num_dapm_routes = ARRAY_SIZE(ml26124_intercon), + .controls = ml26124_snd_controls, + .num_controls = ARRAY_SIZE(ml26124_snd_controls), +}; + +static const struct regmap_config ml26124_i2c_regmap = { + .val_bits = 8, + .reg_bits = 8, + .max_register = ML26124_NUM_REGISTER, + .reg_defaults = ml26124_reg, + .num_reg_defaults = ARRAY_SIZE(ml26124_reg), + .cache_type = REGCACHE_RBTREE, + .write_flag_mask = 0x01, +}; + +static __devinit int ml26124_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct ml26124_priv *priv; + int ret; + + priv = devm_kzalloc(&i2c->dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + i2c_set_clientdata(i2c, priv); + + priv->regmap = regmap_init_i2c(i2c, &ml26124_i2c_regmap); + if (IS_ERR(priv->regmap)) { + ret = PTR_ERR(priv->regmap); + dev_err(&i2c->dev, "regmap_init_i2c() failed: %d\n", ret); + return ret; + } + + return snd_soc_register_codec(&i2c->dev, + &soc_codec_dev_ml26124, &ml26124_dai, 1); +} + +static __devexit int ml26124_i2c_remove(struct i2c_client *client) +{ + struct ml26124_priv *priv = i2c_get_clientdata(client); + + snd_soc_unregister_codec(&client->dev); + regmap_exit(priv->regmap); + return 0; +} + +static const struct i2c_device_id ml26124_i2c_id[] = { + { "ml26124", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, ml26124_i2c_id); + +static struct i2c_driver ml26124_i2c_driver = { + .driver = { + .name = "ml26124", + .owner = THIS_MODULE, + }, + .probe = ml26124_i2c_probe, + .remove = __devexit_p(ml26124_i2c_remove), + .id_table = ml26124_i2c_id, +}; + +module_i2c_driver(ml26124_i2c_driver); + +MODULE_AUTHOR("Tomoya MORINAGA "); +MODULE_DESCRIPTION("LAPIS Semiconductor ML26124 ALSA SoC codec driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/ml26124.h b/sound/soc/codecs/ml26124.h new file mode 100644 index 000000000000..5ea0cbb8c46c --- /dev/null +++ b/sound/soc/codecs/ml26124.h @@ -0,0 +1,184 @@ +/* + * Copyright (C) 2011 LAPIS Semiconductor Co., Ltd. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307, USA. + */ + +#ifndef ML26124_H +#define ML26124_H + +/* Clock Control Register */ +#define ML26124_SMPLING_RATE 0x00 +#define ML26124_PLLNL 0x02 +#define ML26124_PLLNH 0x04 +#define ML26124_PLLML 0x06 +#define ML26124_PLLMH 0x08 +#define ML26124_PLLDIV 0x0a +#define ML26124_CLK_EN 0x0c +#define ML26124_CLK_CTL 0x0e + +/* System Control Register */ +#define ML26124_SW_RST 0x10 +#define ML26124_REC_PLYBAK_RUN 0x12 +#define ML26124_MIC_TIM 0x14 + +/* Power Mnagement Register */ +#define ML26124_PW_REF_PW_MNG 0x20 +#define ML26124_PW_IN_PW_MNG 0x22 +#define ML26124_PW_DAC_PW_MNG 0x24 +#define ML26124_PW_SPAMP_PW_MNG 0x26 +#define ML26124_PW_LOUT_PW_MNG 0x28 +#define ML26124_PW_VOUT_PW_MNG 0x2a +#define ML26124_PW_ZCCMP_PW_MNG 0x2e + +/* Analog Reference Control Register */ +#define ML26124_PW_MICBIAS_VOL 0x30 + +/* Input/Output Amplifier Control Register */ +#define ML26124_PW_MIC_IN_VOL 0x32 +#define ML26124_PW_MIC_BOST_VOL 0x38 +#define ML26124_PW_SPK_AMP_VOL 0x3a +#define ML26124_PW_AMP_VOL_FUNC 0x48 +#define ML26124_PW_AMP_VOL_FADE 0x4a + +/* Analog Path Control Register */ +#define ML26124_SPK_AMP_OUT 0x54 +#define ML26124_MIC_IF_CTL 0x5a +#define ML26124_MIC_SELECT 0xe8 + +/* Audio Interface Control Register */ +#define ML26124_SAI_TRANS_CTL 0x60 +#define ML26124_SAI_RCV_CTL 0x62 +#define ML26124_SAI_MODE_SEL 0x64 + +/* DSP Control Register */ +#define ML26124_FILTER_EN 0x66 +#define ML26124_DVOL_CTL 0x68 +#define ML26124_MIXER_VOL_CTL 0x6a +#define ML26124_RECORD_DIG_VOL 0x6c +#define ML26124_PLBAK_DIG_VOL 0x70 +#define ML26124_DIGI_BOOST_VOL 0x72 +#define ML26124_EQ_GAIN_BRAND0 0x74 +#define ML26124_EQ_GAIN_BRAND1 0x76 +#define ML26124_EQ_GAIN_BRAND2 0x78 +#define ML26124_EQ_GAIN_BRAND3 0x7a +#define ML26124_EQ_GAIN_BRAND4 0x7c +#define ML26124_HPF2_CUTOFF 0x7e +#define ML26124_EQBRAND0_F0L 0x80 +#define ML26124_EQBRAND0_F0H 0x82 +#define ML26124_EQBRAND0_F1L 0x84 +#define ML26124_EQBRAND0_F1H 0x86 +#define ML26124_EQBRAND1_F0L 0x88 +#define ML26124_EQBRAND1_F0H 0x8a +#define ML26124_EQBRAND1_F1L 0x8c +#define ML26124_EQBRAND1_F1H 0x8e +#define ML26124_EQBRAND2_F0L 0x90 +#define ML26124_EQBRAND2_F0H 0x92 +#define ML26124_EQBRAND2_F1L 0x94 +#define ML26124_EQBRAND2_F1H 0x96 +#define ML26124_EQBRAND3_F0L 0x98 +#define ML26124_EQBRAND3_F0H 0x9a +#define ML26124_EQBRAND3_F1L 0x9c +#define ML26124_EQBRAND3_F1H 0x9e +#define ML26124_EQBRAND4_F0L 0xa0 +#define ML26124_EQBRAND4_F0H 0xa2 +#define ML26124_EQBRAND4_F1L 0xa4 +#define ML26124_EQBRAND4_F1H 0xa6 + +/* ALC Control Register */ +#define ML26124_ALC_MODE 0xb0 +#define ML26124_ALC_ATTACK_TIM 0xb2 +#define ML26124_ALC_DECAY_TIM 0xb4 +#define ML26124_ALC_HOLD_TIM 0xb6 +#define ML26124_ALC_TARGET_LEV 0xb8 +#define ML26124_ALC_MAXMIN_GAIN 0xba +#define ML26124_NOIS_GATE_THRSH 0xbc +#define ML26124_ALC_ZERO_TIMOUT 0xbe + +/* Playback Limiter Control Register */ +#define ML26124_PL_ATTACKTIME 0xc0 +#define ML26124_PL_DECAYTIME 0xc2 +#define ML26124_PL_TARGETTIME 0xc4 +#define ML26124_PL_MAXMIN_GAIN 0xc6 +#define ML26124_PLYBAK_BOST_VOL 0xc8 +#define ML26124_PL_0CROSS_TIMOUT 0xca + +/* Video Amplifer Control Register */ +#define ML26124_VIDEO_AMP_GAIN_CTL 0xd0 +#define ML26124_VIDEO_AMP_SETUP1 0xd2 +#define ML26124_VIDEO_AMP_CTL2 0xd4 + +/* Clock select for machine driver */ +#define ML26124_USE_PLL 0 +#define ML26124_USE_MCLKI_256FS 1 +#define ML26124_USE_MCLKI_512FS 2 +#define ML26124_USE_MCLKI_1024FS 3 + +/* Register Mask */ +#define ML26124_R0_MASK 0xf +#define ML26124_R2_MASK 0xff +#define ML26124_R4_MASK 0x1 +#define ML26124_R6_MASK 0xf +#define ML26124_R8_MASK 0x3f +#define ML26124_Ra_MASK 0x1f +#define ML26124_Rc_MASK 0x1f +#define ML26124_Re_MASK 0x7 +#define ML26124_R10_MASK 0x1 +#define ML26124_R12_MASK 0x17 +#define ML26124_R14_MASK 0x3f +#define ML26124_R20_MASK 0x47 +#define ML26124_R22_MASK 0xa +#define ML26124_R24_MASK 0x2 +#define ML26124_R26_MASK 0x1f +#define ML26124_R28_MASK 0x2 +#define ML26124_R2a_MASK 0x2 +#define ML26124_R2e_MASK 0x2 +#define ML26124_R30_MASK 0x7 +#define ML26124_R32_MASK 0x3f +#define ML26124_R38_MASK 0x38 +#define ML26124_R3a_MASK 0x3f +#define ML26124_R48_MASK 0x3 +#define ML26124_R4a_MASK 0x7 +#define ML26124_R54_MASK 0x2a +#define ML26124_R5a_MASK 0x3 +#define ML26124_Re8_MASK 0x3 +#define ML26124_R60_MASK 0xff +#define ML26124_R62_MASK 0xff +#define ML26124_R64_MASK 0x1 +#define ML26124_R66_MASK 0xff +#define ML26124_R68_MASK 0x3b +#define ML26124_R6a_MASK 0xf3 +#define ML26124_R6c_MASK 0xff +#define ML26124_R70_MASK 0xff + +#define ML26124_MCLKEN BIT(0) +#define ML26124_PLLEN BIT(1) +#define ML26124_PLLOE BIT(2) +#define ML26124_MCLKOE BIT(3) + +#define ML26124_BLT_ALL_ON 0x1f +#define ML26124_BLT_PREAMP_ON 0x13 + +#define ML26124_MICBEN_ON BIT(2) + +enum ml26124_regs { + ML26124_MCLK = 0, +}; + +enum ml26124_clk_in { + ML26124_USE_PLLOUT = 0, + ML26124_USE_MCLKI, +}; + +#endif -- cgit v1.2.3 From 1ae93b9d34c26494eea6c127c179b4c88c78bab7 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Tue, 20 Mar 2012 14:55:48 -0600 Subject: ASoC: tegra: fix comment indentation in ALC5632 machine Fix comment indentation to clear checkpatch errors in a later patch. Signed-off-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/tegra/tegra_alc5632.c | 24 ++++++++++++------------ 1 file changed, 12 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c index e45ccd851f6a..03ae0952ccd4 100644 --- a/sound/soc/tegra/tegra_alc5632.c +++ b/sound/soc/tegra/tegra_alc5632.c @@ -1,16 +1,16 @@ /* -* tegra_alc5632.c -- Toshiba AC100(PAZ00) machine ASoC driver -* -* Copyright (C) 2011 The AC100 Kernel Team -* -* Authors: Leon Romanovsky -* Andrey Danin -* Marc Dietrich -* -* This program is free software; you can redistribute it and/or modify -* it under the terms of the GNU General Public License version 2 as -* published by the Free Software Foundation. -*/ + * tegra_alc5632.c -- Toshiba AC100(PAZ00) machine ASoC driver + * + * Copyright (C) 2011 The AC100 Kernel Team + * + * Authors: Leon Romanovsky + * Andrey Danin + * Marc Dietrich + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ #include -- cgit v1.2.3 From 518de86ba106185212ec30fea501be024a12f5db Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Tue, 20 Mar 2012 14:55:49 -0600 Subject: ASoC: tegra: register 'platform' from DAIs, get rid of pdev Previously, the ASoC 'platform' (PCM/DMA) object was instantiated via a platform_device. This didn't represent the hardware well, since there was no separate hardware associated with this platform_device; it was a virtual device with sole purpose to call snd_soc_register_platform(). This mechanism required all board files to register this device, and all ASoC machine drivers to create and register this device when booting using device tree. This change removes the platform_device completely. Each Tegra DAI now registers the ASoC 'platform' itself. Machine drivers are adjusted for the new 'platform' name. Signed-off-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/tegra/tegra_alc5632.c | 21 +++------------------ sound/soc/tegra/tegra_i2s.c | 11 ++++++++++- sound/soc/tegra/tegra_pcm.c | 26 +++++++------------------- sound/soc/tegra/tegra_pcm.h | 5 ++++- sound/soc/tegra/tegra_spdif.c | 11 ++++++++++- sound/soc/tegra/tegra_wm8903.c | 24 ++++++------------------ sound/soc/tegra/trimslice.c | 2 +- 7 files changed, 41 insertions(+), 59 deletions(-) (limited to 'sound') diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c index 03ae0952ccd4..396181cd2c78 100644 --- a/sound/soc/tegra/tegra_alc5632.c +++ b/sound/soc/tegra/tegra_alc5632.c @@ -2,6 +2,7 @@ * tegra_alc5632.c -- Toshiba AC100(PAZ00) machine ASoC driver * * Copyright (C) 2011 The AC100 Kernel Team + * Copyright (C) 2012 - NVIDIA, Inc. * * Authors: Leon Romanovsky * Andrey Danin @@ -39,7 +40,6 @@ struct tegra_alc5632 { struct tegra_asoc_utils_data util_data; - struct platform_device *pcm_dev; int gpio_requested; int gpio_hp_det; }; @@ -140,7 +140,6 @@ static int tegra_alc5632_asoc_init(struct snd_soc_pcm_runtime *rtd) static struct snd_soc_dai_link tegra_alc5632_dai = { .name = "ALC5632", .stream_name = "ALC5632 PCM", - .platform_name = "tegra-pcm-audio", .codec_dai_name = "alc5632-hifi", .init = tegra_alc5632_asoc_init, .ops = &tegra_alc5632_asoc_ops, @@ -179,8 +178,6 @@ static __devinit int tegra_alc5632_probe(struct platform_device *pdev) platform_set_drvdata(pdev, card); snd_soc_card_set_drvdata(card, alc5632); - alc5632->pcm_dev = ERR_PTR(-EINVAL); - if (!(pdev->dev.of_node)) { dev_err(&pdev->dev, "Must be instantiated using device tree\n"); ret = -EINVAL; @@ -214,18 +211,11 @@ static __devinit int tegra_alc5632_probe(struct platform_device *pdev) goto err; } - alc5632->pcm_dev = platform_device_register_simple( - "tegra-pcm-audio", -1, NULL, 0); - if (IS_ERR(alc5632->pcm_dev)) { - dev_err(&pdev->dev, - "Can't instantiate tegra-pcm-audio\n"); - ret = PTR_ERR(alc5632->pcm_dev); - goto err; - } + tegra_alc5632_dai.platform_of_node = tegra_alc5632_dai.cpu_dai_of_node; ret = tegra_asoc_utils_init(&alc5632->util_data, &pdev->dev); if (ret) - goto err_unregister; + goto err; ret = snd_soc_register_card(card); if (ret) { @@ -238,9 +228,6 @@ static __devinit int tegra_alc5632_probe(struct platform_device *pdev) err_fini_utils: tegra_asoc_utils_fini(&alc5632->util_data); -err_unregister: - if (!IS_ERR(alc5632->pcm_dev)) - platform_device_unregister(alc5632->pcm_dev); err: return ret; } @@ -259,8 +246,6 @@ static int __devexit tegra_alc5632_remove(struct platform_device *pdev) snd_soc_unregister_card(card); tegra_asoc_utils_fini(&machine->util_data); - if (!IS_ERR(machine->pcm_dev)) - platform_device_unregister(machine->pcm_dev); return 0; } diff --git a/sound/soc/tegra/tegra_i2s.c b/sound/soc/tegra/tegra_i2s.c index 33509de52540..546e29be8a24 100644 --- a/sound/soc/tegra/tegra_i2s.c +++ b/sound/soc/tegra/tegra_i2s.c @@ -2,7 +2,7 @@ * tegra_i2s.c - Tegra I2S driver * * Author: Stephen Warren - * Copyright (C) 2010 - NVIDIA, Inc. + * Copyright (C) 2010,2012 - NVIDIA, Inc. * * Based on code copyright/by: * @@ -409,10 +409,18 @@ static __devinit int tegra_i2s_platform_probe(struct platform_device *pdev) goto err_clk_put; } + ret = tegra_pcm_platform_register(&pdev->dev); + if (ret) { + dev_err(&pdev->dev, "Could not register PCM: %d\n", ret); + goto err_unregister_dai; + } + tegra_i2s_debug_add(i2s); return 0; +err_unregister_dai: + snd_soc_unregister_dai(&pdev->dev); err_clk_put: clk_put(i2s->clk_i2s); err: @@ -423,6 +431,7 @@ static int __devexit tegra_i2s_platform_remove(struct platform_device *pdev) { struct tegra_i2s *i2s = dev_get_drvdata(&pdev->dev); + tegra_pcm_platform_unregister(&pdev->dev); snd_soc_unregister_dai(&pdev->dev); tegra_i2s_debug_remove(i2s); diff --git a/sound/soc/tegra/tegra_pcm.c b/sound/soc/tegra/tegra_pcm.c index 8b4457137c7c..476b8ac3a24d 100644 --- a/sound/soc/tegra/tegra_pcm.c +++ b/sound/soc/tegra/tegra_pcm.c @@ -2,7 +2,7 @@ * tegra_pcm.c - Tegra PCM driver * * Author: Stephen Warren - * Copyright (C) 2010 - NVIDIA, Inc. + * Copyright (C) 2010,2012 - NVIDIA, Inc. * * Based on code copyright/by: * @@ -39,8 +39,6 @@ #include "tegra_pcm.h" -#define DRV_NAME "tegra-pcm-audio" - static const struct snd_pcm_hardware tegra_pcm_hardware = { .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | @@ -372,28 +370,18 @@ static struct snd_soc_platform_driver tegra_pcm_platform = { .pcm_free = tegra_pcm_free, }; -static int __devinit tegra_pcm_platform_probe(struct platform_device *pdev) +int __devinit tegra_pcm_platform_register(struct device *dev) { - return snd_soc_register_platform(&pdev->dev, &tegra_pcm_platform); + return snd_soc_register_platform(dev, &tegra_pcm_platform); } +EXPORT_SYMBOL_GPL(tegra_pcm_platform_register); -static int __devexit tegra_pcm_platform_remove(struct platform_device *pdev) +void __devexit tegra_pcm_platform_unregister(struct device *dev) { - snd_soc_unregister_platform(&pdev->dev); - return 0; + snd_soc_unregister_platform(dev); } - -static struct platform_driver tegra_pcm_driver = { - .driver = { - .name = DRV_NAME, - .owner = THIS_MODULE, - }, - .probe = tegra_pcm_platform_probe, - .remove = __devexit_p(tegra_pcm_platform_remove), -}; -module_platform_driver(tegra_pcm_driver); +EXPORT_SYMBOL_GPL(tegra_pcm_platform_unregister); MODULE_AUTHOR("Stephen Warren "); MODULE_DESCRIPTION("Tegra PCM ASoC driver"); MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:" DRV_NAME); diff --git a/sound/soc/tegra/tegra_pcm.h b/sound/soc/tegra/tegra_pcm.h index dbb90339fe0d..985d418a35e7 100644 --- a/sound/soc/tegra/tegra_pcm.h +++ b/sound/soc/tegra/tegra_pcm.h @@ -2,7 +2,7 @@ * tegra_pcm.h - Definitions for Tegra PCM driver * * Author: Stephen Warren - * Copyright (C) 2010 - NVIDIA, Inc. + * Copyright (C) 2010,2012 - NVIDIA, Inc. * * Based on code copyright/by: * @@ -52,4 +52,7 @@ struct tegra_runtime_data { struct tegra_dma_channel *dma_chan; }; +int tegra_pcm_platform_register(struct device *dev); +void tegra_pcm_platform_unregister(struct device *dev); + #endif diff --git a/sound/soc/tegra/tegra_spdif.c b/sound/soc/tegra/tegra_spdif.c index 475428cf270e..cd836cbcb0c9 100644 --- a/sound/soc/tegra/tegra_spdif.c +++ b/sound/soc/tegra/tegra_spdif.c @@ -2,7 +2,7 @@ * tegra_spdif.c - Tegra SPDIF driver * * Author: Stephen Warren - * Copyright (C) 2011 - NVIDIA, Inc. + * Copyright (C) 2011-2012 - NVIDIA, Inc. * * This program is free software; you can redistribute it and/or * modify it under the terms of the GNU General Public License @@ -306,10 +306,18 @@ static __devinit int tegra_spdif_platform_probe(struct platform_device *pdev) goto err_unmap; } + ret = tegra_pcm_platform_register(&pdev->dev); + if (ret) { + dev_err(&pdev->dev, "Could not register PCM: %d\n", ret); + goto err_unregister_dai; + } + tegra_spdif_debug_add(spdif); return 0; +err_unregister_dai: + snd_soc_unregister_dai(&pdev->dev); err_unmap: iounmap(spdif->regs); err_release: @@ -327,6 +335,7 @@ static int __devexit tegra_spdif_platform_remove(struct platform_device *pdev) struct tegra_spdif *spdif = dev_get_drvdata(&pdev->dev); struct resource *res; + tegra_pcm_platform_unregister(&pdev->dev); snd_soc_unregister_dai(&pdev->dev); tegra_spdif_debug_remove(spdif); diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c index 566655e23b7d..fbd13357e1fd 100644 --- a/sound/soc/tegra/tegra_wm8903.c +++ b/sound/soc/tegra/tegra_wm8903.c @@ -2,7 +2,7 @@ * tegra_wm8903.c - Tegra machine ASoC driver for boards using WM8903 codec. * * Author: Stephen Warren - * Copyright (C) 2010-2011 - NVIDIA, Inc. + * Copyright (C) 2010-2012 - NVIDIA, Inc. * * Based on code copyright/by: * @@ -61,7 +61,6 @@ struct tegra_wm8903 { struct tegra_wm8903_platform_data pdata; - struct platform_device *pcm_dev; struct tegra_asoc_utils_data util_data; int gpio_requested; }; @@ -354,7 +353,7 @@ static struct snd_soc_dai_link tegra_wm8903_dai = { .name = "WM8903", .stream_name = "WM8903 PCM", .codec_name = "wm8903.0-001a", - .platform_name = "tegra-pcm-audio", + .platform_name = "tegra-i2s.0", .cpu_dai_name = "tegra-i2s.0", .codec_dai_name = "wm8903-hifi", .init = tegra_wm8903_init, @@ -392,7 +391,6 @@ static __devinit int tegra_wm8903_driver_probe(struct platform_device *pdev) ret = -ENOMEM; goto err; } - machine->pcm_dev = ERR_PTR(-EINVAL); card->dev = &pdev->dev; platform_set_drvdata(pdev, card); @@ -428,14 +426,9 @@ static __devinit int tegra_wm8903_driver_probe(struct platform_device *pdev) goto err; } - machine->pcm_dev = platform_device_register_simple( - "tegra-pcm-audio", -1, NULL, 0); - if (IS_ERR(machine->pcm_dev)) { - dev_err(&pdev->dev, - "Can't instantiate tegra-pcm-audio\n"); - ret = PTR_ERR(machine->pcm_dev); - goto err; - } + tegra_wm8903_dai.platform_name = NULL; + tegra_wm8903_dai.platform_of_node = + tegra_wm8903_dai.cpu_dai_of_node; } else { if (machine_is_harmony()) { card->dapm_routes = harmony_audio_map; @@ -454,7 +447,7 @@ static __devinit int tegra_wm8903_driver_probe(struct platform_device *pdev) ret = tegra_asoc_utils_init(&machine->util_data, &pdev->dev); if (ret) - goto err_unregister; + goto err; ret = snd_soc_register_card(card); if (ret) { @@ -467,9 +460,6 @@ static __devinit int tegra_wm8903_driver_probe(struct platform_device *pdev) err_fini_utils: tegra_asoc_utils_fini(&machine->util_data); -err_unregister: - if (!IS_ERR(machine->pcm_dev)) - platform_device_unregister(machine->pcm_dev); err: return ret; } @@ -497,8 +487,6 @@ static int __devexit tegra_wm8903_driver_remove(struct platform_device *pdev) snd_soc_unregister_card(card); tegra_asoc_utils_fini(&machine->util_data); - if (!IS_ERR(machine->pcm_dev)) - platform_device_unregister(machine->pcm_dev); return 0; } diff --git a/sound/soc/tegra/trimslice.c b/sound/soc/tegra/trimslice.c index 2bdfc550cff8..6be9e0f747c9 100644 --- a/sound/soc/tegra/trimslice.c +++ b/sound/soc/tegra/trimslice.c @@ -119,7 +119,7 @@ static struct snd_soc_dai_link trimslice_tlv320aic23_dai = { .name = "TLV320AIC23", .stream_name = "AIC23", .codec_name = "tlv320aic23-codec.2-001a", - .platform_name = "tegra-pcm-audio", + .platform_name = "tegra-i2s.0", .cpu_dai_name = "tegra-i2s.0", .codec_dai_name = "tlv320aic23-hifi", .ops = &trimslice_asoc_ops, -- cgit v1.2.3 From b46b373f4084cc02d4d41a5a42199fe8462c2f13 Mon Sep 17 00:00:00 2001 From: Shawn Guo Date: Wed, 28 Mar 2012 15:34:56 +0800 Subject: ASoC: fsl: assign dma peripheral type according to bus topology The dma peripheral_type for SSI should be IMX_DMATYPE_SSI_SP if the SSI is on SPBA bus. Signed-off-by: Shawn Guo Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_ssi.c | 6 ++++++ sound/soc/fsl/imx-pcm-dma.c | 3 ++- sound/soc/fsl/imx-pcm.h | 1 + 3 files changed, 9 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 30ff605144c6..4f76b5df0ce4 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -746,6 +746,12 @@ static int __devinit fsl_ssi_probe(struct platform_device *pdev) } ssi_private->dma_params_tx.dma = dma_events[0]; ssi_private->dma_params_rx.dma = dma_events[1]; + + ssi_private->dma_params_tx.shared_peripheral = + of_device_is_compatible(of_get_parent(np), + "fsl,spba-bus"); + ssi_private->dma_params_rx.shared_peripheral = + ssi_private->dma_params_tx.shared_peripheral; } /* Initialize the the device_attribute structure */ diff --git a/sound/soc/fsl/imx-pcm-dma.c b/sound/soc/fsl/imx-pcm-dma.c index 6b818de2fc03..f3c0a5ef35c8 100644 --- a/sound/soc/fsl/imx-pcm-dma.c +++ b/sound/soc/fsl/imx-pcm-dma.c @@ -109,7 +109,8 @@ static int snd_imx_open(struct snd_pcm_substream *substream) dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); dma_data = kzalloc(sizeof(*dma_data), GFP_KERNEL); - dma_data->peripheral_type = IMX_DMATYPE_SSI; + dma_data->peripheral_type = dma_params->shared_peripheral ? + IMX_DMATYPE_SSI_SP : IMX_DMATYPE_SSI; dma_data->priority = DMA_PRIO_HIGH; dma_data->dma_request = dma_params->dma; diff --git a/sound/soc/fsl/imx-pcm.h b/sound/soc/fsl/imx-pcm.h index b5f5c3acf34d..83c0ed7d55c9 100644 --- a/sound/soc/fsl/imx-pcm.h +++ b/sound/soc/fsl/imx-pcm.h @@ -22,6 +22,7 @@ struct imx_pcm_dma_params { int dma; unsigned long dma_addr; int burstsize; + bool shared_peripheral; /* The peripheral is on SPBA bus */ }; int snd_imx_pcm_mmap(struct snd_pcm_substream *substream, -- cgit v1.2.3 From e7cff0abf99a0895bc2094e08809bd5e0c4f6a4a Mon Sep 17 00:00:00 2001 From: H Hartley Sweeten Date: Mon, 26 Mar 2012 16:00:08 -0700 Subject: ASoC: ep93xx-ac97: use devm_* helpers to cleanup probe Use the devm_* helpers to cleanup the probe routine. This also eliminates having to carry the mem and irq values in the private data for the remove. Signed-off-by: H Hartley Sweeten Tested-by: Mika Westerberg Acked-by: Mika Westerberg Signed-off-by: Mark Brown --- sound/soc/ep93xx/ep93xx-ac97.c | 74 +++++++++++++++--------------------------- 1 file changed, 26 insertions(+), 48 deletions(-) (limited to 'sound') diff --git a/sound/soc/ep93xx/ep93xx-ac97.c b/sound/soc/ep93xx/ep93xx-ac97.c index 0678637abd66..bdffab33e160 100644 --- a/sound/soc/ep93xx/ep93xx-ac97.c +++ b/sound/soc/ep93xx/ep93xx-ac97.c @@ -87,17 +87,13 @@ * struct ep93xx_ac97_info - EP93xx AC97 controller info structure * @lock: mutex serializing access to the bus (slot 1 & 2 ops) * @dev: pointer to the platform device dev structure - * @mem: physical memory resource for the registers * @regs: mapped AC97 controller registers - * @irq: AC97 interrupt number * @done: bus ops wait here for an interrupt */ struct ep93xx_ac97_info { struct mutex lock; struct device *dev; - struct resource *mem; void __iomem *regs; - int irq; struct completion done; }; @@ -359,66 +355,50 @@ static struct snd_soc_dai_driver ep93xx_ac97_dai = { static int __devinit ep93xx_ac97_probe(struct platform_device *pdev) { struct ep93xx_ac97_info *info; + struct resource *res; + unsigned int irq; int ret; - info = kzalloc(sizeof(struct ep93xx_ac97_info), GFP_KERNEL); + info = devm_kzalloc(&pdev->dev, sizeof(*info), GFP_KERNEL); if (!info) return -ENOMEM; - dev_set_drvdata(&pdev->dev, info); - - mutex_init(&info->lock); - init_completion(&info->done); - info->dev = &pdev->dev; + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!res) + return -ENODEV; - info->mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!info->mem) { - ret = -ENXIO; - goto fail_free_info; - } + info->regs = devm_request_and_ioremap(&pdev->dev, res); + if (!info->regs) + return -ENXIO; - info->irq = platform_get_irq(pdev, 0); - if (!info->irq) { - ret = -ENXIO; - goto fail_free_info; - } + irq = platform_get_irq(pdev, 0); + if (!irq) + return -ENODEV; - if (!request_mem_region(info->mem->start, resource_size(info->mem), - pdev->name)) { - ret = -EBUSY; - goto fail_free_info; - } + ret = devm_request_irq(&pdev->dev, irq, ep93xx_ac97_interrupt, + IRQF_TRIGGER_HIGH, pdev->name, info); + if (ret) + goto fail; - info->regs = ioremap(info->mem->start, resource_size(info->mem)); - if (!info->regs) { - ret = -ENOMEM; - goto fail_release_mem; - } + dev_set_drvdata(&pdev->dev, info); - ret = request_irq(info->irq, ep93xx_ac97_interrupt, IRQF_TRIGGER_HIGH, - pdev->name, info); - if (ret) - goto fail_unmap_mem; + mutex_init(&info->lock); + init_completion(&info->done); + info->dev = &pdev->dev; ep93xx_ac97_info = info; platform_set_drvdata(pdev, info); ret = snd_soc_register_dai(&pdev->dev, &ep93xx_ac97_dai); if (ret) - goto fail_free_irq; + goto fail; return 0; -fail_free_irq: +fail: platform_set_drvdata(pdev, NULL); - free_irq(info->irq, info); -fail_unmap_mem: - iounmap(info->regs); -fail_release_mem: - release_mem_region(info->mem->start, resource_size(info->mem)); -fail_free_info: - kfree(info); - + ep93xx_ac97_info = NULL; + dev_set_drvdata(&pdev->dev, NULL); return ret; } @@ -431,11 +411,9 @@ static int __devexit ep93xx_ac97_remove(struct platform_device *pdev) /* disable the AC97 controller */ ep93xx_ac97_write_reg(info, AC97GCR, 0); - free_irq(info->irq, info); - iounmap(info->regs); - release_mem_region(info->mem->start, resource_size(info->mem)); platform_set_drvdata(pdev, NULL); - kfree(info); + ep93xx_ac97_info = NULL; + dev_set_drvdata(&pdev->dev, NULL); return 0; } -- cgit v1.2.3 From 01651babb66fb7b51719ff3389a7bfc3ad25fe13 Mon Sep 17 00:00:00 2001 From: H Hartley Sweeten Date: Mon, 26 Mar 2012 16:00:17 -0700 Subject: ASoC: ep93xx-i2s: use devm_* helpers to cleanup probe Use the devm_* helpers to cleanup the probe routine. This also eliminates having to carry the mem value in the private data for the remove. Signed-off-by: H Hartley Sweeten Tested-by: Mika Westerberg Acked-by: Mika Westerberg Signed-off-by: Mark Brown --- sound/soc/ep93xx/ep93xx-i2s.c | 49 +++++++++++++------------------------------ 1 file changed, 14 insertions(+), 35 deletions(-) (limited to 'sound') diff --git a/sound/soc/ep93xx/ep93xx-i2s.c b/sound/soc/ep93xx/ep93xx-i2s.c index f7a62348e3fe..8df8f6dc474f 100644 --- a/sound/soc/ep93xx/ep93xx-i2s.c +++ b/sound/soc/ep93xx/ep93xx-i2s.c @@ -63,7 +63,6 @@ struct ep93xx_i2s_info { struct clk *sclk; struct clk *lrclk; struct ep93xx_pcm_dma_params *dma_params; - struct resource *mem; void __iomem *regs; }; @@ -373,38 +372,22 @@ static int ep93xx_i2s_probe(struct platform_device *pdev) struct resource *res; int err; - info = kzalloc(sizeof(struct ep93xx_i2s_info), GFP_KERNEL); - if (!info) { - err = -ENOMEM; - goto fail; - } - - dev_set_drvdata(&pdev->dev, info); - info->dma_params = ep93xx_i2s_dma_params; + info = devm_kzalloc(&pdev->dev, sizeof(*info), GFP_KERNEL); + if (!info) + return -ENOMEM; res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!res) { - err = -ENODEV; - goto fail_free_info; - } + if (!res) + return -ENODEV; - info->mem = request_mem_region(res->start, resource_size(res), - pdev->name); - if (!info->mem) { - err = -EBUSY; - goto fail_free_info; - } - - info->regs = ioremap(info->mem->start, resource_size(info->mem)); - if (!info->regs) { - err = -ENXIO; - goto fail_release_mem; - } + info->regs = devm_request_and_ioremap(&pdev->dev, res); + if (!info->regs) + return -ENXIO; info->mclk = clk_get(&pdev->dev, "mclk"); if (IS_ERR(info->mclk)) { err = PTR_ERR(info->mclk); - goto fail_unmap_mem; + goto fail; } info->sclk = clk_get(&pdev->dev, "sclk"); @@ -419,6 +402,9 @@ static int ep93xx_i2s_probe(struct platform_device *pdev) goto fail_put_sclk; } + dev_set_drvdata(&pdev->dev, info); + info->dma_params = ep93xx_i2s_dma_params; + err = snd_soc_register_dai(&pdev->dev, &ep93xx_i2s_dai); if (err) goto fail_put_lrclk; @@ -426,17 +412,12 @@ static int ep93xx_i2s_probe(struct platform_device *pdev) return 0; fail_put_lrclk: + dev_set_drvdata(&pdev->dev, NULL); clk_put(info->lrclk); fail_put_sclk: clk_put(info->sclk); fail_put_mclk: clk_put(info->mclk); -fail_unmap_mem: - iounmap(info->regs); -fail_release_mem: - release_mem_region(info->mem->start, resource_size(info->mem)); -fail_free_info: - kfree(info); fail: return err; } @@ -446,12 +427,10 @@ static int __devexit ep93xx_i2s_remove(struct platform_device *pdev) struct ep93xx_i2s_info *info = dev_get_drvdata(&pdev->dev); snd_soc_unregister_dai(&pdev->dev); + dev_set_drvdata(&pdev->dev, NULL); clk_put(info->lrclk); clk_put(info->sclk); clk_put(info->mclk); - iounmap(info->regs); - release_mem_region(info->mem->start, resource_size(info->mem)); - kfree(info); return 0; } -- cgit v1.2.3 From 95cd98f9a6dcf112d2abf724ac07c56ec745180f Mon Sep 17 00:00:00 2001 From: Shawn Guo Date: Thu, 29 Mar 2012 10:53:41 +0800 Subject: ASoC: fsl: enable ssi clock in probe function For power saving, most IMX platform initilization code turns off modules' clock, and expects driver turn on clock as needed. Signed-off-by: Shawn Guo Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_ssi.c | 24 ++++++++++++++++++++++-- 1 file changed, 22 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 4f76b5df0ce4..4ed2afd47782 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -14,6 +14,7 @@ #include #include #include +#include #include #include #include @@ -119,6 +120,7 @@ struct fsl_ssi_private { bool new_binding; bool ssi_on_imx; + struct clk *clk; struct platform_device *imx_pcm_pdev; struct imx_pcm_dma_params dma_params_tx; struct imx_pcm_dma_params dma_params_rx; @@ -722,6 +724,15 @@ static int __devinit fsl_ssi_probe(struct platform_device *pdev) if (of_device_is_compatible(pdev->dev.of_node, "fsl,imx21-ssi")) { u32 dma_events[2]; ssi_private->ssi_on_imx = true; + + ssi_private->clk = clk_get(&pdev->dev, NULL); + if (IS_ERR(ssi_private->clk)) { + ret = PTR_ERR(ssi_private->clk); + dev_err(&pdev->dev, "could not get clock: %d\n", ret); + goto error_irq; + } + clk_prepare_enable(ssi_private->clk); + /* * We have burstsize be "fifo_depth - 2" to match the SSI * watermark setting in fsl_ssi_startup(). @@ -742,7 +753,7 @@ static int __devinit fsl_ssi_probe(struct platform_device *pdev) "fsl,ssi-dma-events", dma_events, 2); if (ret) { dev_err(&pdev->dev, "could not get dma events\n"); - goto error_irq; + goto error_clk; } ssi_private->dma_params_tx.dma = dma_events[0]; ssi_private->dma_params_rx.dma = dma_events[1]; @@ -830,6 +841,12 @@ error_dev: dev_set_drvdata(&pdev->dev, NULL); device_remove_file(&pdev->dev, dev_attr); +error_clk: + if (ssi_private->ssi_on_imx) { + clk_disable_unprepare(ssi_private->clk); + clk_put(ssi_private->clk); + } + error_irq: free_irq(ssi_private->irq, ssi_private); @@ -851,8 +868,11 @@ static int fsl_ssi_remove(struct platform_device *pdev) if (!ssi_private->new_binding) platform_device_unregister(ssi_private->pdev); - if (ssi_private->ssi_on_imx) + if (ssi_private->ssi_on_imx) { platform_device_unregister(ssi_private->imx_pcm_pdev); + clk_disable_unprepare(ssi_private->clk); + clk_put(ssi_private->clk); + } snd_soc_unregister_dai(&pdev->dev); device_remove_file(&pdev->dev, &ssi_private->dev_attr); -- cgit v1.2.3 From e413ba88044db34b3fc9aa1b432a4579db9072b3 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 29 Mar 2012 14:49:27 +0100 Subject: ASoC: wm8994: Don't allow reconfiguration of FLL when it provides SYSCLK Rather than trying to work around machine drivers which try to reprogram the FLL while it is providing SYSCLK just return an error if they try. This will avoid audio glitches during FLL reconfiguration, or at least move the introduction of the glitches to the machine driver. Since disabling the source for an active SYSCLK is not supported in the first place systems shouldn't be doing this in the first place. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 35 +++++++++++++++++------------------ 1 file changed, 17 insertions(+), 18 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index fbcaf499b0ff..317159eec44c 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -1900,24 +1900,20 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src, struct wm8994 *control = wm8994->wm8994; int reg_offset, ret; struct fll_div fll; - u16 reg, aif1, aif2; + u16 reg, clk1, aif_reg, aif_src; unsigned long timeout; bool was_enabled; - aif1 = snd_soc_read(codec, WM8994_AIF1_CLOCKING_1) - & WM8994_AIF1CLK_ENA; - - aif2 = snd_soc_read(codec, WM8994_AIF2_CLOCKING_1) - & WM8994_AIF2CLK_ENA; - switch (id) { case WM8994_FLL1: reg_offset = 0; id = 0; + aif_src = 0x10; break; case WM8994_FLL2: reg_offset = 0x20; id = 1; + aif_src = 0x18; break; default: return -EINVAL; @@ -1959,11 +1955,20 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src, if (ret < 0) return ret; - /* Gate the AIF clocks while we reclock */ - snd_soc_update_bits(codec, WM8994_AIF1_CLOCKING_1, - WM8994_AIF1CLK_ENA, 0); - snd_soc_update_bits(codec, WM8994_AIF2_CLOCKING_1, - WM8994_AIF2CLK_ENA, 0); + /* Make sure that we're not providing SYSCLK right now */ + clk1 = snd_soc_read(codec, WM8994_CLOCKING_1); + if (clk1 & WM8994_SYSCLK_SRC) + aif_reg = WM8994_AIF2_CLOCKING_1; + else + aif_reg = WM8994_AIF1_CLOCKING_1; + reg = snd_soc_read(codec, aif_reg); + + if ((reg & WM8994_AIF1CLK_ENA) && + (reg & WM8994_AIF1CLK_SRC_MASK) == aif_src) { + dev_err(codec->dev, "FLL%d is currently providing SYSCLK\n", + id + 1); + return -EBUSY; + } /* We always need to disable the FLL while reconfiguring */ snd_soc_update_bits(codec, WM8994_FLL1_CONTROL_1 + reg_offset, @@ -2049,12 +2054,6 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src, wm8994->fll[id].out = freq_out; wm8994->fll[id].src = src; - /* Enable any gated AIF clocks */ - snd_soc_update_bits(codec, WM8994_AIF1_CLOCKING_1, - WM8994_AIF1CLK_ENA, aif1); - snd_soc_update_bits(codec, WM8994_AIF2_CLOCKING_1, - WM8994_AIF2CLK_ENA, aif2); - configure_clock(codec); return 0; -- cgit v1.2.3 From 8fc8ec92a5db47cdf3526adc5717041c611e5516 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 28 Mar 2012 20:51:43 +0100 Subject: ASoC: sgtl5000: Convert mic bias to a supply widget No current users and it's the last user of MICBIAS_E(). Signed-off-by: Mark Brown Acked-by: Dong Aisheng Acked-by: Zeng Zhaoming Tested-by: Shawn Guo --- sound/soc/codecs/sgtl5000.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index d1926266fe00..77beb6d7d269 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -197,9 +197,9 @@ static const struct snd_soc_dapm_widget sgtl5000_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("HP_OUT"), SND_SOC_DAPM_OUTPUT("LINE_OUT"), - SND_SOC_DAPM_MICBIAS_E("Mic Bias", SGTL5000_CHIP_MIC_CTRL, 8, 0, - mic_bias_event, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_SUPPLY("Mic Bias", SGTL5000_CHIP_MIC_CTRL, 8, 0, + mic_bias_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), SND_SOC_DAPM_PGA_E("HP", SGTL5000_CHIP_ANA_POWER, 4, 0, NULL, 0, small_pop_event, -- cgit v1.2.3 From aa0e25caafb7950e839db930649a65e8b7f70e1a Mon Sep 17 00:00:00 2001 From: Ashish Chavan Date: Thu, 29 Mar 2012 19:06:29 +0530 Subject: ASoC: da7210: Add support for spi regmap This patch adds support for spi regmap feature to existing da7210 driver. Signed-off-by: Ashish Chavan Signed-off-by: David Dajun Chen Signed-off-by: Mark Brown --- sound/soc/codecs/da7210.c | 154 +++++++++++++++++++++++++++++++++++++++------- 1 file changed, 133 insertions(+), 21 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 7843711729bc..8e45aa9cb7b7 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -17,6 +17,7 @@ #include #include +#include #include #include #include @@ -27,6 +28,7 @@ #include /* DA7210 register space */ +#define DA7210_PAGE_CONTROL 0x00 #define DA7210_CONTROL 0x01 #define DA7210_STATUS 0x02 #define DA7210_STARTUP1 0x03 @@ -631,6 +633,7 @@ struct da7210_priv { }; static struct reg_default da7210_reg_defaults[] = { + { 0x00, 0x00 }, { 0x01, 0x11 }, { 0x03, 0x00 }, { 0x04, 0x00 }, @@ -927,13 +930,6 @@ static int da7210_probe(struct snd_soc_codec *codec) * DA7210_PLL_DIV3 :: DA7210_MCLK_RANGExxx */ - /* - * make sure that DA7210 use bypass mode before start up - */ - snd_soc_write(codec, DA7210_STARTUP1, 0); - snd_soc_write(codec, DA7210_PLL_DIV3, - DA7210_MCLK_RANGE_10_20_MHZ | DA7210_PLL_BYP); - /* * ADC settings */ @@ -1025,16 +1021,6 @@ static int da7210_probe(struct snd_soc_codec *codec) DA7210_MCLK_RANGE_10_20_MHZ | DA7210_PLL_BYP); snd_soc_update_bits(codec, DA7210_PLL, DA7210_PLL_EN, DA7210_PLL_EN); - /* As suggested by Dialog */ - /* unlock */ - regmap_write(da7210->regmap, DA7210_A_HID_UNLOCK, 0x8B); - regmap_write(da7210->regmap, DA7210_A_TEST_UNLOCK, 0xB4); - regmap_write(da7210->regmap, DA7210_A_PLL1, 0x01); - regmap_write(da7210->regmap, DA7210_A_CP_MODE, 0x7C); - /* re-lock */ - regmap_write(da7210->regmap, DA7210_A_HID_UNLOCK, 0x00); - regmap_write(da7210->regmap, DA7210_A_TEST_UNLOCK, 0x00); - /* Activate all enabled subsystem */ snd_soc_write(codec, DA7210_STARTUP1, DA7210_SC_MST_EN); @@ -1055,7 +1041,26 @@ static struct snd_soc_codec_driver soc_codec_dev_da7210 = { .num_dapm_routes = ARRAY_SIZE(da7210_audio_map), }; -static struct regmap_config da7210_regmap = { +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + +static struct reg_default da7210_regmap_i2c_patch[] = { + + /* System controller master disable */ + { DA7210_STARTUP1, 0x00 }, + /* make sure that DA7210 use bypass mode before start up */ + { DA7210_PLL_DIV3, DA7210_MCLK_RANGE_10_20_MHZ | DA7210_PLL_BYP }, + + /* to unlock */ + { DA7210_A_HID_UNLOCK, 0x8B}, + { DA7210_A_TEST_UNLOCK, 0xB4}, + { DA7210_A_PLL1, 0x01}, + { DA7210_A_CP_MODE, 0x7C}, + /* to re-lock */ + { DA7210_A_HID_UNLOCK, 0x00}, + { DA7210_A_TEST_UNLOCK, 0x00}, +}; + +static const struct regmap_config da7210_regmap_config_i2c = { .reg_bits = 8, .val_bits = 8, @@ -1066,7 +1071,6 @@ static struct regmap_config da7210_regmap = { .cache_type = REGCACHE_RBTREE, }; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) static int __devinit da7210_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -1080,13 +1084,18 @@ static int __devinit da7210_i2c_probe(struct i2c_client *i2c, i2c_set_clientdata(i2c, da7210); - da7210->regmap = regmap_init_i2c(i2c, &da7210_regmap); + da7210->regmap = regmap_init_i2c(i2c, &da7210_regmap_config_i2c); if (IS_ERR(da7210->regmap)) { ret = PTR_ERR(da7210->regmap); dev_err(&i2c->dev, "regmap_init() failed: %d\n", ret); return ret; } + ret = regmap_register_patch(da7210->regmap, da7210_regmap_i2c_patch, + ARRAY_SIZE(da7210_regmap_i2c_patch)); + if (ret != 0) + dev_warn(&i2c->dev, "Failed to apply regmap patch: %d\n", ret); + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_da7210, &da7210_dai, 1); if (ret < 0) { @@ -1119,7 +1128,7 @@ MODULE_DEVICE_TABLE(i2c, da7210_i2c_id); /* I2C codec control layer */ static struct i2c_driver da7210_i2c_driver = { .driver = { - .name = "da7210-codec", + .name = "da7210", .owner = THIS_MODULE, }, .probe = da7210_i2c_probe, @@ -1128,11 +1137,111 @@ static struct i2c_driver da7210_i2c_driver = { }; #endif +#if defined(CONFIG_SPI_MASTER) + +static struct reg_default da7210_regmap_spi_patch[] = { + /* Dummy read to give two pulses over nCS for SPI */ + { DA7210_AUX2, 0x00 }, + { DA7210_AUX2, 0x00 }, + + /* System controller master disable */ + { DA7210_STARTUP1, 0x00 }, + /* make sure that DA7210 use bypass mode before start up */ + { DA7210_PLL_DIV3, DA7210_MCLK_RANGE_10_20_MHZ | DA7210_PLL_BYP }, + + /* to set PAGE1 of SPI register space */ + { DA7210_PAGE_CONTROL, 0x80 }, + /* to unlock */ + { DA7210_A_HID_UNLOCK, 0x8B}, + { DA7210_A_TEST_UNLOCK, 0xB4}, + { DA7210_A_PLL1, 0x01}, + { DA7210_A_CP_MODE, 0x7C}, + /* to re-lock */ + { DA7210_A_HID_UNLOCK, 0x00}, + { DA7210_A_TEST_UNLOCK, 0x00}, + /* to set back PAGE0 of SPI register space */ + { DA7210_PAGE_CONTROL, 0x00 }, +}; + +static const struct regmap_config da7210_regmap_config_spi = { + .reg_bits = 8, + .val_bits = 8, + .read_flag_mask = 0x01, + .write_flag_mask = 0x00, + + .reg_defaults = da7210_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(da7210_reg_defaults), + .volatile_reg = da7210_volatile_register, + .readable_reg = da7210_readable_register, + .cache_type = REGCACHE_RBTREE, +}; + +static int __devinit da7210_spi_probe(struct spi_device *spi) +{ + struct da7210_priv *da7210; + int ret; + + da7210 = devm_kzalloc(&spi->dev, sizeof(struct da7210_priv), + GFP_KERNEL); + if (!da7210) + return -ENOMEM; + + spi_set_drvdata(spi, da7210); + da7210->regmap = devm_regmap_init_spi(spi, &da7210_regmap_config_spi); + if (IS_ERR(da7210->regmap)) { + ret = PTR_ERR(da7210->regmap); + dev_err(&spi->dev, "Failed to register regmap: %d\n", ret); + return ret; + } + + ret = regmap_register_patch(da7210->regmap, da7210_regmap_spi_patch, + ARRAY_SIZE(da7210_regmap_spi_patch)); + if (ret != 0) + dev_warn(&spi->dev, "Failed to apply regmap patch: %d\n", ret); + + ret = snd_soc_register_codec(&spi->dev, + &soc_codec_dev_da7210, &da7210_dai, 1); + if (ret < 0) + goto err_regmap; + + return ret; + +err_regmap: + regmap_exit(da7210->regmap); + + return ret; +} + +static int __devexit da7210_spi_remove(struct spi_device *spi) +{ + struct da7210_priv *da7210 = spi_get_drvdata(spi); + snd_soc_unregister_codec(&spi->dev); + regmap_exit(da7210->regmap); + return 0; +} + +static struct spi_driver da7210_spi_driver = { + .driver = { + .name = "da7210", + .owner = THIS_MODULE, + }, + .probe = da7210_spi_probe, + .remove = __devexit_p(da7210_spi_remove) +}; +#endif + static int __init da7210_modinit(void) { int ret = 0; #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) ret = i2c_add_driver(&da7210_i2c_driver); +#endif +#if defined(CONFIG_SPI_MASTER) + ret = spi_register_driver(&da7210_spi_driver); + if (ret) { + printk(KERN_ERR "Failed to register da7210 SPI driver: %d\n", + ret); + } #endif return ret; } @@ -1143,6 +1252,9 @@ static void __exit da7210_exit(void) #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) i2c_del_driver(&da7210_i2c_driver); #endif +#if defined(CONFIG_SPI_MASTER) + spi_unregister_driver(&da7210_spi_driver); +#endif } module_exit(da7210_exit); -- cgit v1.2.3 From 172b4c5c8afdb7471d9b03fc96a6b6455a49e19d Mon Sep 17 00:00:00 2001 From: Shawn Guo Date: Fri, 30 Mar 2012 00:13:03 +0800 Subject: ASoC: fsl: add audio routing for imx-sgtl5000 Add DAPM widgets and audio routing support for imx-sgtl5000 machine driver. Signed-off-by: Shawn Guo Signed-off-by: Mark Brown --- .../bindings/sound/imx-audio-sgtl5000.txt | 25 ++++++++++++++++++++++ sound/soc/fsl/imx-sgtl5000.c | 13 +++++++++++ 2 files changed, 38 insertions(+) (limited to 'sound') diff --git a/Documentation/devicetree/bindings/sound/imx-audio-sgtl5000.txt b/Documentation/devicetree/bindings/sound/imx-audio-sgtl5000.txt index 421a374790b3..d09b4e31461c 100644 --- a/Documentation/devicetree/bindings/sound/imx-audio-sgtl5000.txt +++ b/Documentation/devicetree/bindings/sound/imx-audio-sgtl5000.txt @@ -5,6 +5,27 @@ Required properties: - model : The user-visible name of this sound complex - ssi-controller : The phandle of the i.MX SSI controller - audio-codec : The phandle of the SGTL5000 audio codec +- audio-routing : A list of the connections between audio components. + Each entry is a pair of strings, the first being the connection's sink, + the second being the connection's source. Valid names could be power + supplies, SGTL5000 pins, and the jacks on the board: + + Power supplies: + * Mic Bias + + SGTL5000 pins: + * MIC_IN + * LINE_IN + * HP_OUT + * LINE_OUT + + Board connectors: + * Mic Jack + * Line In Jack + * Headphone Jack + * Line Out Jack + * Ext Spk + - mux-int-port : The internal port of the i.MX audio muxer (AUDMUX) - mux-ext-port : The external port of the i.MX audio muxer @@ -19,6 +40,10 @@ sound { model = "imx51-babbage-sgtl5000"; ssi-controller = <&ssi1>; audio-codec = <&sgtl5000>; + audio-routing = + "Mic Jack", "Mic Bias", + "MIC_IN", "Mic Bias", + "Headphone Jack", "HP_OUT"; mux-int-port = <1>; mux-ext-port = <3>; }; diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index 3786b61b4fea..e1a7441ec133 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -45,6 +45,14 @@ static int imx_sgtl5000_dai_init(struct snd_soc_pcm_runtime *rtd) return 0; } +static const struct snd_soc_dapm_widget imx_sgtl5000_dapm_widgets[] = { + SND_SOC_DAPM_MIC("Mic Jack", NULL), + SND_SOC_DAPM_LINE("Line In Jack", NULL), + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_SPK("Line Out Jack", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), +}; + static int __devinit imx_sgtl5000_probe(struct platform_device *pdev) { struct device_node *np = pdev->dev.of_node; @@ -127,10 +135,15 @@ static int __devinit imx_sgtl5000_probe(struct platform_device *pdev) data->card.dev = &pdev->dev; ret = snd_soc_of_parse_card_name(&data->card, "model"); + if (ret) + return ret; + ret = snd_soc_of_parse_audio_routing(&data->card, "audio-routing"); if (ret) return ret; data->card.num_links = 1; data->card.dai_link = &data->dai; + data->card.dapm_widgets = imx_sgtl5000_dapm_widgets; + data->card.num_dapm_widgets = ARRAY_SIZE(imx_sgtl5000_dapm_widgets); ret = snd_soc_register_card(&data->card); if (ret) { -- cgit v1.2.3 From 1947dadf2a2d64b6f7db8a6547f46b9bbdd79dc3 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 30 Mar 2012 11:21:45 +0100 Subject: ASoC: wm8994: Don't bother lowering clock dividers inside idle AIFs This increases the chances we'll manage to hit a partially configured state on restart and the power savings are extremely small. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 29 ----------------------------- 1 file changed, 29 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 317159eec44c..c9af13f14331 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2623,33 +2623,6 @@ static int wm8994_aif3_hw_params(struct snd_pcm_substream *substream, return snd_soc_update_bits(codec, aif1_reg, WM8994_AIF1_WL_MASK, aif1); } -static void wm8994_aif_shutdown(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct snd_soc_codec *codec = dai->codec; - int rate_reg = 0; - - switch (dai->id) { - case 1: - rate_reg = WM8994_AIF1_RATE; - break; - case 2: - rate_reg = WM8994_AIF2_RATE; - break; - default: - break; - } - - /* If the DAI is idle then configure the divider tree for the - * lowest output rate to save a little power if the clock is - * still active (eg, because it is system clock). - */ - if (rate_reg && !dai->playback_active && !dai->capture_active) - snd_soc_update_bits(codec, rate_reg, - WM8994_AIF1_SR_MASK | - WM8994_AIF1CLK_RATE_MASK, 0x9); -} - static int wm8994_aif_mute(struct snd_soc_dai *codec_dai, int mute) { struct snd_soc_codec *codec = codec_dai->codec; @@ -2731,7 +2704,6 @@ static const struct snd_soc_dai_ops wm8994_aif1_dai_ops = { .set_sysclk = wm8994_set_dai_sysclk, .set_fmt = wm8994_set_dai_fmt, .hw_params = wm8994_hw_params, - .shutdown = wm8994_aif_shutdown, .digital_mute = wm8994_aif_mute, .set_pll = wm8994_set_fll, .set_tristate = wm8994_set_tristate, @@ -2741,7 +2713,6 @@ static const struct snd_soc_dai_ops wm8994_aif2_dai_ops = { .set_sysclk = wm8994_set_dai_sysclk, .set_fmt = wm8994_set_dai_fmt, .hw_params = wm8994_hw_params, - .shutdown = wm8994_aif_shutdown, .digital_mute = wm8994_aif_mute, .set_pll = wm8994_set_fll, .set_tristate = wm8994_set_tristate, -- cgit v1.2.3 From fcff5f99742e0d0e036ea2ce80a21bfec434bc88 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Fri, 30 Mar 2012 17:07:18 -0600 Subject: ASoC: tegra: remove unnecessary includes These include aren't needed, and some of the files are about to be renamed. Signed-off-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/tegra/tegra_alc5632.c | 3 --- sound/soc/tegra/tegra_wm8903.c | 3 --- sound/soc/tegra/trimslice.c | 3 --- 3 files changed, 9 deletions(-) (limited to 'sound') diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c index 396181cd2c78..32de7006daf0 100644 --- a/sound/soc/tegra/tegra_alc5632.c +++ b/sound/soc/tegra/tegra_alc5632.c @@ -29,9 +29,6 @@ #include "../codecs/alc5632.h" -#include "tegra_das.h" -#include "tegra_i2s.h" -#include "tegra_pcm.h" #include "tegra_asoc_utils.h" #define DRV_NAME "tegra-alc5632" diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c index fbd13357e1fd..2f9e9ff771c8 100644 --- a/sound/soc/tegra/tegra_wm8903.c +++ b/sound/soc/tegra/tegra_wm8903.c @@ -46,9 +46,6 @@ #include "../codecs/wm8903.h" -#include "tegra_das.h" -#include "tegra_i2s.h" -#include "tegra_pcm.h" #include "tegra_asoc_utils.h" #define DRV_NAME "tegra-snd-wm8903" diff --git a/sound/soc/tegra/trimslice.c b/sound/soc/tegra/trimslice.c index 6be9e0f747c9..8884667aff43 100644 --- a/sound/soc/tegra/trimslice.c +++ b/sound/soc/tegra/trimslice.c @@ -38,9 +38,6 @@ #include "../codecs/tlv320aic23.h" -#include "tegra_das.h" -#include "tegra_i2s.h" -#include "tegra_pcm.h" #include "tegra_asoc_utils.h" #define DRV_NAME "tegra-snd-trimslice" -- cgit v1.2.3 From 4df8271e1ffcc4302a3c5326de15eb6737697001 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Fri, 30 Mar 2012 17:07:20 -0600 Subject: ASoC: tegra: fix Kconfig SND_SOC_TEGRA_ALC5632 indentation Indent with TABs to be consistent with the rest of the file. Signed-off-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/tegra/Kconfig | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig index ce1b773c351f..f8fcda33208e 100644 --- a/sound/soc/tegra/Kconfig +++ b/sound/soc/tegra/Kconfig @@ -49,10 +49,10 @@ config SND_SOC_TEGRA_TRIMSLICE TrimSlice platform. config SND_SOC_TEGRA_ALC5632 - tristate "SoC Audio support for Tegra boards using an ALC5632 codec" - depends on SND_SOC_TEGRA && I2C - select SND_SOC_TEGRA_I2S - select SND_SOC_ALC5632 - help - Say Y or M here if you want to add support for SoC audio on the - Toshiba AC100 netbook. + tristate "SoC Audio support for Tegra boards using an ALC5632 codec" + depends on SND_SOC_TEGRA && I2C + select SND_SOC_TEGRA_I2S + select SND_SOC_ALC5632 + help + Say Y or M here if you want to add support for SoC audio on the + Toshiba AC100 netbook. -- cgit v1.2.3 From 7deb2b450df9975ab05deb885e386d59a16213a9 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Fri, 30 Mar 2012 17:07:21 -0600 Subject: ASoC: tegra: fix some checkpatch warnings ERROR: trailing whitespace ERROR: code indent should use tabs where possible WARNING: please, no spaces at the start of a line ERROR: "foo * bar" should be "foo *bar" Signed-off-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/tegra/tegra_das.h | 2 +- sound/soc/tegra/tegra_i2s.c | 8 ++++---- 2 files changed, 5 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/tegra/tegra_das.h b/sound/soc/tegra/tegra_das.h index 2c96c7b3c459..1810cd1e7f0c 100644 --- a/sound/soc/tegra/tegra_das.h +++ b/sound/soc/tegra/tegra_das.h @@ -94,7 +94,7 @@ struct tegra_das { * DAS: Digital audio switch (HW module controlled by this driver) * DAP: Digital audio port (port/pins on Tegra device) * DAC: Digital audio controller (e.g. I2S or AC97 controller elsewhere) - * + * * The Tegra DAS is a mux/cross-bar which can connect each DAP to a specific * DAC, or another DAP. When DAPs are connected, one must be the master and * one the slave. Each DAC allows selection of a specific DAP for input, to diff --git a/sound/soc/tegra/tegra_i2s.c b/sound/soc/tegra/tegra_i2s.c index 546e29be8a24..d66e5095363a 100644 --- a/sound/soc/tegra/tegra_i2s.c +++ b/sound/soc/tegra/tegra_i2s.c @@ -144,7 +144,7 @@ static int tegra_i2s_set_fmt(struct snd_soc_dai *dai, return -EINVAL; } - i2s->reg_ctrl &= ~(TEGRA_I2S_CTRL_BIT_FORMAT_MASK | + i2s->reg_ctrl &= ~(TEGRA_I2S_CTRL_BIT_FORMAT_MASK | TEGRA_I2S_CTRL_LRCK_MASK); switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_DSP_A: @@ -178,7 +178,7 @@ static int tegra_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct device *dev = substream->pcm->card->dev; + struct device *dev = substream->pcm->card->dev; struct tegra_i2s *i2s = snd_soc_dai_get_drvdata(dai); u32 reg; int ret, sample_size, srate, i2sclock, bitcnt; @@ -296,7 +296,7 @@ static int tegra_i2s_trigger(struct snd_pcm_substream *substream, int cmd, static int tegra_i2s_probe(struct snd_soc_dai *dai) { - struct tegra_i2s * i2s = snd_soc_dai_get_drvdata(dai); + struct tegra_i2s *i2s = snd_soc_dai_get_drvdata(dai); dai->capture_dma_data = &i2s->capture_dma_data; dai->playback_dma_data = &i2s->playback_dma_data; @@ -330,7 +330,7 @@ static const struct snd_soc_dai_driver tegra_i2s_dai_template = { static __devinit int tegra_i2s_platform_probe(struct platform_device *pdev) { - struct tegra_i2s * i2s; + struct tegra_i2s *i2s; struct resource *mem, *memregion, *dmareq; u32 of_dma[2]; u32 dma_ch; -- cgit v1.2.3 From d9bba496d47085555f4b011be6d716308cf4de03 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Fri, 30 Mar 2012 17:07:22 -0600 Subject: ASoC: tegra: introduce separate Kconfig variable for DAS driver This is mainly for symmetry with a future Tegra30 driver, where the equivalent of the DAS (the AHUB) is useful separately from the I2S driver. Signed-off-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/tegra/Kconfig | 9 +++++++++ sound/soc/tegra/Makefile | 2 +- 2 files changed, 10 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig index f8fcda33208e..23ab00d5b192 100644 --- a/sound/soc/tegra/Kconfig +++ b/sound/soc/tegra/Kconfig @@ -4,9 +4,18 @@ config SND_SOC_TEGRA help Say Y or M here if you want support for SoC audio on Tegra. +config SND_SOC_TEGRA_DAS + tristate "Tegra Digital Audio Switch driver" + depends on SND_SOC_TEGRA + help + Say Y or M if you want to add support for the Tegra DAS module. + You will also need to select the individual machine drivers to + support below. + config SND_SOC_TEGRA_I2S tristate depends on SND_SOC_TEGRA + select SND_SOC_TEGRA_DAS help Say Y or M if you want to add support for codecs attached to the Tegra I2S interface. You will also need to select the individual diff --git a/sound/soc/tegra/Makefile b/sound/soc/tegra/Makefile index 8e584b8fcfba..6d5d81e188a2 100644 --- a/sound/soc/tegra/Makefile +++ b/sound/soc/tegra/Makefile @@ -6,7 +6,7 @@ snd-soc-tegra-spdif-objs := tegra_spdif.o snd-soc-tegra-utils-objs += tegra_asoc_utils.o obj-$(CONFIG_SND_SOC_TEGRA) += snd-soc-tegra-utils.o -obj-$(CONFIG_SND_SOC_TEGRA) += snd-soc-tegra-das.o +obj-$(CONFIG_SND_SOC_TEGRA_DAS) += snd-soc-tegra-das.o obj-$(CONFIG_SND_SOC_TEGRA) += snd-soc-tegra-pcm.o obj-$(CONFIG_SND_SOC_TEGRA_I2S) += snd-soc-tegra-i2s.o obj-$(CONFIG_SND_SOC_TEGRA_SPDIF) += snd-soc-tegra-spdif.o -- cgit v1.2.3 From c0d5a47ca86047aca1616b744ab3ef31b3448994 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Fri, 30 Mar 2012 17:07:24 -0600 Subject: ASoC: tegra: sort Makefile into common and per-SoC files The DAS, I2S, and SPDIF drivers are Tegra20-specific. Group these together so that when Tegra30-specific equivalents are added later, the file ordering makes sense. Signed-off-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/tegra/Makefile | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/tegra/Makefile b/sound/soc/tegra/Makefile index 6d5d81e188a2..714d36070b86 100644 --- a/sound/soc/tegra/Makefile +++ b/sound/soc/tegra/Makefile @@ -1,13 +1,13 @@ # Tegra platform Support -snd-soc-tegra-das-objs := tegra_das.o snd-soc-tegra-pcm-objs := tegra_pcm.o +snd-soc-tegra-utils-objs += tegra_asoc_utils.o +snd-soc-tegra-das-objs := tegra_das.o snd-soc-tegra-i2s-objs := tegra_i2s.o snd-soc-tegra-spdif-objs := tegra_spdif.o -snd-soc-tegra-utils-objs += tegra_asoc_utils.o +obj-$(CONFIG_SND_SOC_TEGRA) += snd-soc-tegra-pcm.o obj-$(CONFIG_SND_SOC_TEGRA) += snd-soc-tegra-utils.o obj-$(CONFIG_SND_SOC_TEGRA_DAS) += snd-soc-tegra-das.o -obj-$(CONFIG_SND_SOC_TEGRA) += snd-soc-tegra-pcm.o obj-$(CONFIG_SND_SOC_TEGRA_I2S) += snd-soc-tegra-i2s.o obj-$(CONFIG_SND_SOC_TEGRA_SPDIF) += snd-soc-tegra-spdif.o -- cgit v1.2.3 From 30d436a64415e6d01b8696d6288abe7ad0b383b5 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Fri, 30 Mar 2012 17:07:16 -0600 Subject: ASoC: tegra: remove open-coded clk reference counting clk_enable/disable() already reference count the enable calls, so there's no need for the callers to do the same. Signed-off-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/tegra/tegra_i2s.c | 14 ++++---------- sound/soc/tegra/tegra_i2s.h | 1 - sound/soc/tegra/tegra_spdif.c | 8 ++------ sound/soc/tegra/tegra_spdif.h | 1 - 4 files changed, 6 insertions(+), 18 deletions(-) (limited to 'sound') diff --git a/sound/soc/tegra/tegra_i2s.c b/sound/soc/tegra/tegra_i2s.c index d66e5095363a..2d32b8ce5f2e 100644 --- a/sound/soc/tegra/tegra_i2s.c +++ b/sound/soc/tegra/tegra_i2s.c @@ -220,8 +220,7 @@ static int tegra_i2s_hw_params(struct snd_pcm_substream *substream, if (i2sclock % (2 * srate)) reg |= TEGRA_I2S_TIMING_NON_SYM_ENABLE; - if (!i2s->clk_refs) - clk_enable(i2s->clk_i2s); + clk_enable(i2s->clk_i2s); tegra_i2s_write(i2s, TEGRA_I2S_TIMING, reg); @@ -229,8 +228,7 @@ static int tegra_i2s_hw_params(struct snd_pcm_substream *substream, TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_FOUR_SLOTS | TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_FOUR_SLOTS); - if (!i2s->clk_refs) - clk_disable(i2s->clk_i2s); + clk_disable(i2s->clk_i2s); return 0; } @@ -268,9 +266,7 @@ static int tegra_i2s_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: case SNDRV_PCM_TRIGGER_RESUME: - if (!i2s->clk_refs) - clk_enable(i2s->clk_i2s); - i2s->clk_refs++; + clk_enable(i2s->clk_i2s); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) tegra_i2s_start_playback(i2s); else @@ -283,9 +279,7 @@ static int tegra_i2s_trigger(struct snd_pcm_substream *substream, int cmd, tegra_i2s_stop_playback(i2s); else tegra_i2s_stop_capture(i2s); - i2s->clk_refs--; - if (!i2s->clk_refs) - clk_disable(i2s->clk_i2s); + clk_disable(i2s->clk_i2s); break; default: return -EINVAL; diff --git a/sound/soc/tegra/tegra_i2s.h b/sound/soc/tegra/tegra_i2s.h index 15ce1e2e8bde..c08e019c0260 100644 --- a/sound/soc/tegra/tegra_i2s.h +++ b/sound/soc/tegra/tegra_i2s.h @@ -155,7 +155,6 @@ struct tegra_i2s { struct snd_soc_dai_driver dai; struct clk *clk_i2s; - int clk_refs; struct tegra_pcm_dma_params capture_dma_data; struct tegra_pcm_dma_params playback_dma_data; void __iomem *regs; diff --git a/sound/soc/tegra/tegra_spdif.c b/sound/soc/tegra/tegra_spdif.c index cd836cbcb0c9..3426633e9147 100644 --- a/sound/soc/tegra/tegra_spdif.c +++ b/sound/soc/tegra/tegra_spdif.c @@ -196,18 +196,14 @@ static int tegra_spdif_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: case SNDRV_PCM_TRIGGER_RESUME: - if (!spdif->clk_refs) - clk_enable(spdif->clk_spdif_out); - spdif->clk_refs++; + clk_enable(spdif->clk_spdif_out); tegra_spdif_start_playback(spdif); break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: case SNDRV_PCM_TRIGGER_SUSPEND: tegra_spdif_stop_playback(spdif); - spdif->clk_refs--; - if (!spdif->clk_refs) - clk_disable(spdif->clk_spdif_out); + clk_disable(spdif->clk_spdif_out); break; default: return -EINVAL; diff --git a/sound/soc/tegra/tegra_spdif.h b/sound/soc/tegra/tegra_spdif.h index 2e03db430279..f5fc7124aa7b 100644 --- a/sound/soc/tegra/tegra_spdif.h +++ b/sound/soc/tegra/tegra_spdif.h @@ -462,7 +462,6 @@ struct tegra_spdif { struct clk *clk_spdif_out; - int clk_refs; struct tegra_pcm_dma_params capture_dma_data; struct tegra_pcm_dma_params playback_dma_data; void __iomem *regs; -- cgit v1.2.3 From dbf7a733f5fb9da9de750716ec7c7615c30cbfb8 Mon Sep 17 00:00:00 2001 From: M R Swami Reddy Date: Fri, 30 Mar 2012 16:03:43 +0530 Subject: ASoC: Support TI LM49453 Audio driver Signed-off-by: M R Swami Reddy Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 4 +- sound/soc/codecs/lm49453.c | 1554 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/lm49453.h | 380 +++++++++++ 4 files changed, 1941 insertions(+), 1 deletion(-) create mode 100644 sound/soc/codecs/lm49453.c create mode 100644 sound/soc/codecs/lm49453.h (limited to 'sound') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index e314a66b30cd..2e51eb08b303 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -37,6 +37,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_DFBMCS320 select SND_SOC_JZ4740_CODEC select SND_SOC_LM4857 if I2C + select SND_SOC_LM49453 if I2C select SND_SOC_MAX98088 if I2C select SND_SOC_MAX98095 if I2C select SND_SOC_MAX9850 if I2C @@ -218,6 +219,9 @@ config SND_SOC_DFBMCS320 config SND_SOC_DMIC tristate +config SND_SOC_LM49453 + tristate + config SND_SOC_MAX98088 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 9108ee992395..db61c4499715 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -25,6 +25,7 @@ snd-soc-dmic-objs := dmic.o snd-soc-jz4740-codec-objs := jz4740.o snd-soc-l3-objs := l3.o snd-soc-lm4857-objs := lm4857.o +snd-soc-lm49453-objs := lm49453.o snd-soc-max9768-objs := max9768.o snd-soc-max98088-objs := max98088.o snd-soc-max98095-objs := max98095.o @@ -129,9 +130,10 @@ obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o obj-$(CONFIG_SND_SOC_DA7210) += snd-soc-da7210.o obj-$(CONFIG_SND_SOC_DFBMCS320) += snd-soc-dfbmcs320.o obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o +obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o obj-$(CONFIG_SND_SOC_LM4857) += snd-soc-lm4857.o -obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o +obj-$(CONFIG_SND_SOC_LM49453) += snd-soc-lm49453.o obj-$(CONFIG_SND_SOC_MAX9768) += snd-soc-max9768.o obj-$(CONFIG_SND_SOC_MAX98088) += snd-soc-max98088.o obj-$(CONFIG_SND_SOC_MAX98095) += snd-soc-max98095.o diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c new file mode 100644 index 000000000000..744063da40d5 --- /dev/null +++ b/sound/soc/codecs/lm49453.c @@ -0,0 +1,1554 @@ +/* + * lm49453.c - LM49453 ALSA Soc Audio driver + * + * Copyright (c) 2012 Texas Instruments, Inc + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * Initially based on sound/soc/codecs/wm8350.c + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include "lm49453.h" + +static struct reg_default lm49453_reg_defs[] = { + { 0, 0x00 }, + { 1, 0x00 }, + { 2, 0x00 }, + { 3, 0x00 }, + { 4, 0x00 }, + { 5, 0x00 }, + { 6, 0x00 }, + { 7, 0x00 }, + { 8, 0x00 }, + { 9, 0x00 }, + { 10, 0x00 }, + { 11, 0x00 }, + { 12, 0x00 }, + { 13, 0x00 }, + { 14, 0x00 }, + { 15, 0x00 }, + { 16, 0x00 }, + { 17, 0x00 }, + { 18, 0x00 }, + { 19, 0x00 }, + { 20, 0x00 }, + { 21, 0x00 }, + { 22, 0x00 }, + { 23, 0x00 }, + { 32, 0x00 }, + { 33, 0x00 }, + { 35, 0x00 }, + { 36, 0x00 }, + { 37, 0x00 }, + { 46, 0x00 }, + { 48, 0x00 }, + { 49, 0x00 }, + { 51, 0x00 }, + { 56, 0x00 }, + { 58, 0x00 }, + { 59, 0x00 }, + { 60, 0x00 }, + { 61, 0x00 }, + { 62, 0x00 }, + { 63, 0x00 }, + { 64, 0x00 }, + { 65, 0x00 }, + { 66, 0x00 }, + { 67, 0x00 }, + { 68, 0x00 }, + { 69, 0x00 }, + { 70, 0x00 }, + { 71, 0x00 }, + { 72, 0x00 }, + { 73, 0x00 }, + { 74, 0x00 }, + { 75, 0x00 }, + { 76, 0x00 }, + { 77, 0x00 }, + { 78, 0x00 }, + { 79, 0x00 }, + { 80, 0x00 }, + { 81, 0x00 }, + { 82, 0x00 }, + { 83, 0x00 }, + { 85, 0x00 }, + { 85, 0x00 }, + { 86, 0x00 }, + { 87, 0x00 }, + { 88, 0x00 }, + { 89, 0x00 }, + { 90, 0x00 }, + { 91, 0x00 }, + { 92, 0x00 }, + { 93, 0x00 }, + { 94, 0x00 }, + { 95, 0x00 }, + { 96, 0x01 }, + { 97, 0x00 }, + { 98, 0x00 }, + { 99, 0x00 }, + { 100, 0x00 }, + { 101, 0x00 }, + { 102, 0x00 }, + { 103, 0x01 }, + { 105, 0x01 }, + { 106, 0x00 }, + { 107, 0x01 }, + { 107, 0x00 }, + { 108, 0x00 }, + { 109, 0x00 }, + { 110, 0x00 }, + { 111, 0x02 }, + { 112, 0x02 }, + { 113, 0x00 }, + { 121, 0x80 }, + { 122, 0xBB }, + { 123, 0x80 }, + { 124, 0xBB }, + { 128, 0x00 }, + { 130, 0x00 }, + { 131, 0x00 }, + { 132, 0x00 }, + { 133, 0x0A }, + { 134, 0x0A }, + { 135, 0x0A }, + { 136, 0x0F }, + { 137, 0x00 }, + { 138, 0x73 }, + { 139, 0x33 }, + { 140, 0x73 }, + { 141, 0x33 }, + { 142, 0x73 }, + { 143, 0x33 }, + { 144, 0x73 }, + { 145, 0x33 }, + { 146, 0x73 }, + { 147, 0x33 }, + { 148, 0x73 }, + { 149, 0x33 }, + { 150, 0x73 }, + { 151, 0x33 }, + { 152, 0x00 }, + { 153, 0x00 }, + { 154, 0x00 }, + { 155, 0x00 }, + { 176, 0x00 }, + { 177, 0x00 }, + { 178, 0x00 }, + { 179, 0x00 }, + { 180, 0x00 }, + { 181, 0x00 }, + { 182, 0x00 }, + { 183, 0x00 }, + { 184, 0x00 }, + { 185, 0x00 }, + { 186, 0x00 }, + { 189, 0x00 }, + { 188, 0x00 }, + { 194, 0x00 }, + { 195, 0x00 }, + { 196, 0x00 }, + { 197, 0x00 }, + { 200, 0x00 }, + { 201, 0x00 }, + { 202, 0x00 }, + { 203, 0x00 }, + { 204, 0x00 }, + { 205, 0x00 }, + { 208, 0x00 }, + { 209, 0x00 }, + { 210, 0x00 }, + { 211, 0x00 }, + { 213, 0x00 }, + { 214, 0x00 }, + { 215, 0x00 }, + { 216, 0x00 }, + { 217, 0x00 }, + { 218, 0x00 }, + { 219, 0x00 }, + { 221, 0x00 }, + { 222, 0x00 }, + { 224, 0x00 }, + { 225, 0x00 }, + { 226, 0x00 }, + { 227, 0x00 }, + { 228, 0x00 }, + { 229, 0x00 }, + { 230, 0x13 }, + { 231, 0x00 }, + { 232, 0x80 }, + { 233, 0x0C }, + { 234, 0xDD }, + { 235, 0x00 }, + { 236, 0x04 }, + { 237, 0x00 }, + { 238, 0x00 }, + { 239, 0x00 }, + { 240, 0x00 }, + { 241, 0x00 }, + { 242, 0x00 }, + { 243, 0x00 }, + { 244, 0x00 }, + { 245, 0x00 }, + { 248, 0x00 }, + { 249, 0x00 }, + { 254, 0x00 }, + { 255, 0x00 }, +}; + +/* codec private data */ +struct lm49453_priv { + struct regmap *regmap; + int fs_rate; +}; + +/* capture path controls */ + +static const char *lm49453_mic2mode_text[] = {"Single Ended", "Differential"}; + +static const SOC_ENUM_SINGLE_DECL(lm49453_mic2mode_enum, LM49453_P0_MICR_REG, 5, + lm49453_mic2mode_text); + +static const char *lm49453_dmic_cfg_text[] = {"DMICDAT1", "DMICDAT2"}; + +static const SOC_ENUM_SINGLE_DECL(lm49453_dmic12_cfg_enum, + LM49453_P0_DIGITAL_MIC1_CONFIG_REG, + 7, lm49453_dmic_cfg_text); + +static const SOC_ENUM_SINGLE_DECL(lm49453_dmic34_cfg_enum, + LM49453_P0_DIGITAL_MIC2_CONFIG_REG, + 7, lm49453_dmic_cfg_text); + +/* MUX Controls */ +static const char *lm49453_adcl_mux_text[] = { "MIC1", "Aux_L" }; + +static const char *lm49453_adcr_mux_text[] = { "MIC2", "Aux_R" }; + +static const struct soc_enum lm49453_adcl_enum = + SOC_ENUM_SINGLE(LM49453_P0_ANALOG_MIXER_ADC_REG, 0, + ARRAY_SIZE(lm49453_adcl_mux_text), + lm49453_adcl_mux_text); + +static const struct soc_enum lm49453_adcr_enum = + SOC_ENUM_SINGLE(LM49453_P0_ANALOG_MIXER_ADC_REG, 1, + ARRAY_SIZE(lm49453_adcr_mux_text), + lm49453_adcr_mux_text); + +static const struct snd_kcontrol_new lm49453_adcl_mux_control = + SOC_DAPM_ENUM("ADC Left Mux", lm49453_adcl_enum); + +static const struct snd_kcontrol_new lm49453_adcr_mux_control = + SOC_DAPM_ENUM("ADC Right Mux", lm49453_adcr_enum); + +static const struct snd_kcontrol_new lm49453_headset_left_mixer[] = { +SOC_DAPM_SINGLE("Port1_1 Switch", LM49453_P0_DACHPL1_REG, 0, 1, 0), +SOC_DAPM_SINGLE("Port1_2 Switch", LM49453_P0_DACHPL1_REG, 1, 1, 0), +SOC_DAPM_SINGLE("Port1_3 Switch", LM49453_P0_DACHPL1_REG, 2, 1, 0), +SOC_DAPM_SINGLE("Port1_4 Switch", LM49453_P0_DACHPL1_REG, 3, 1, 0), +SOC_DAPM_SINGLE("Port1_5 Switch", LM49453_P0_DACHPL1_REG, 4, 1, 0), +SOC_DAPM_SINGLE("Port1_6 Switch", LM49453_P0_DACHPL1_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_7 Switch", LM49453_P0_DACHPL1_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port1_8 Switch", LM49453_P0_DACHPL1_REG, 7, 1, 0), +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_DACHPL2_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_DACHPL2_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_DACHPL2_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_DACHPL2_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_DACHPL2_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_DACHPL2_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port2_1 Switch", LM49453_P0_DACHPL2_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_DACHPL2_REG, 7, 1, 0), +SOC_DAPM_SINGLE("Sidetone Switch", LM49453_P0_STN_SEL_REG, 0, 0, 0), +}; + +static const struct snd_kcontrol_new lm49453_headset_right_mixer[] = { +SOC_DAPM_SINGLE("Port1_1 Switch", LM49453_P0_DACHPR1_REG, 0, 1, 0), +SOC_DAPM_SINGLE("Port1_2 Switch", LM49453_P0_DACHPR1_REG, 1, 1, 0), +SOC_DAPM_SINGLE("Port1_3 Switch", LM49453_P0_DACHPR1_REG, 2, 1, 0), +SOC_DAPM_SINGLE("Port1_4 Switch", LM49453_P0_DACHPR1_REG, 3, 1, 0), +SOC_DAPM_SINGLE("Port1_5 Switch", LM49453_P0_DACHPR1_REG, 4, 1, 0), +SOC_DAPM_SINGLE("Port1_6 Switch", LM49453_P0_DACHPR1_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_7 Switch", LM49453_P0_DACHPR1_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port1_8 Switch", LM49453_P0_DACHPR1_REG, 7, 1, 0), +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_DACHPR2_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_DACHPR2_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_DACHPR2_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_DACHPR2_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_DACHPR2_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_DACHPR2_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port2_1 Switch", LM49453_P0_DACHPR2_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_DACHPR2_REG, 7, 1, 0), +SOC_DAPM_SINGLE("Sidetone Switch", LM49453_P0_STN_SEL_REG, 1, 0, 0), +}; + +static const struct snd_kcontrol_new lm49453_speaker_left_mixer[] = { +SOC_DAPM_SINGLE("Port1_1 Switch", LM49453_P0_DACLSL1_REG, 0, 1, 0), +SOC_DAPM_SINGLE("Port1_2 Switch", LM49453_P0_DACLSL1_REG, 1, 1, 0), +SOC_DAPM_SINGLE("Port1_3 Switch", LM49453_P0_DACLSL1_REG, 2, 1, 0), +SOC_DAPM_SINGLE("Port1_4 Switch", LM49453_P0_DACLSL1_REG, 3, 1, 0), +SOC_DAPM_SINGLE("Port1_5 Switch", LM49453_P0_DACLSL1_REG, 4, 1, 0), +SOC_DAPM_SINGLE("Port1_6 Switch", LM49453_P0_DACLSL1_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_7 Switch", LM49453_P0_DACLSL1_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port1_8 Switch", LM49453_P0_DACLSL1_REG, 7, 1, 0), +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_DACLSL2_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_DACLSL2_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_DACLSL2_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_DACLSL2_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_DACLSL2_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_DACLSL2_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port2_1 Switch", LM49453_P0_DACLSL2_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_DACLSL2_REG, 7, 1, 0), +SOC_DAPM_SINGLE("Sidetone Switch", LM49453_P0_STN_SEL_REG, 2, 0, 0), +}; + +static const struct snd_kcontrol_new lm49453_speaker_right_mixer[] = { +SOC_DAPM_SINGLE("Port1_1 Switch", LM49453_P0_DACLSR1_REG, 0, 1, 0), +SOC_DAPM_SINGLE("Port1_2 Switch", LM49453_P0_DACLSR1_REG, 1, 1, 0), +SOC_DAPM_SINGLE("Port1_3 Switch", LM49453_P0_DACLSR1_REG, 2, 1, 0), +SOC_DAPM_SINGLE("Port1_4 Switch", LM49453_P0_DACLSR1_REG, 3, 1, 0), +SOC_DAPM_SINGLE("Port1_5 Switch", LM49453_P0_DACLSR1_REG, 4, 1, 0), +SOC_DAPM_SINGLE("Port1_6 Switch", LM49453_P0_DACLSR1_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_7 Switch", LM49453_P0_DACLSR1_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port1_8 Switch", LM49453_P0_DACLSR1_REG, 7, 1, 0), +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_DACLSR2_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_DACLSR2_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_DACLSR2_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_DACLSR2_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_DACLSR2_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_DACLSR2_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port2_1 Switch", LM49453_P0_DACLSR2_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_DACLSR2_REG, 7, 1, 0), +SOC_DAPM_SINGLE("Sidetone Switch", LM49453_P0_STN_SEL_REG, 3, 0, 0), +}; + +static const struct snd_kcontrol_new lm49453_haptic_left_mixer[] = { +SOC_DAPM_SINGLE("Port1_1 Switch", LM49453_P0_DACHAL1_REG, 0, 1, 0), +SOC_DAPM_SINGLE("Port1_2 Switch", LM49453_P0_DACHAL1_REG, 1, 1, 0), +SOC_DAPM_SINGLE("Port1_3 Switch", LM49453_P0_DACHAL1_REG, 2, 1, 0), +SOC_DAPM_SINGLE("Port1_4 Switch", LM49453_P0_DACHAL1_REG, 3, 1, 0), +SOC_DAPM_SINGLE("Port1_5 Switch", LM49453_P0_DACHAL1_REG, 4, 1, 0), +SOC_DAPM_SINGLE("Port1_6 Switch", LM49453_P0_DACHAL1_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_7 Switch", LM49453_P0_DACHAL1_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port1_8 Switch", LM49453_P0_DACHAL1_REG, 7, 1, 0), +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_DACHAL2_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_DACHAL2_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_DACHAL2_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_DACHAL2_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_DACHAL2_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_DACHAL2_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port2_1 Switch", LM49453_P0_DACHAL2_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_DACHAL2_REG, 7, 1, 0), +SOC_DAPM_SINGLE("Sidetone Switch", LM49453_P0_STN_SEL_REG, 4, 0, 0), +}; + +static const struct snd_kcontrol_new lm49453_haptic_right_mixer[] = { +SOC_DAPM_SINGLE("Port1_1 Switch", LM49453_P0_DACHAR1_REG, 0, 1, 0), +SOC_DAPM_SINGLE("Port1_2 Switch", LM49453_P0_DACHAR1_REG, 1, 1, 0), +SOC_DAPM_SINGLE("Port1_3 Switch", LM49453_P0_DACHAR1_REG, 2, 1, 0), +SOC_DAPM_SINGLE("Port1_4 Switch", LM49453_P0_DACHAR1_REG, 3, 1, 0), +SOC_DAPM_SINGLE("Port1_5 Switch", LM49453_P0_DACHAR1_REG, 4, 1, 0), +SOC_DAPM_SINGLE("Port1_6 Switch", LM49453_P0_DACHAR1_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_7 Switch", LM49453_P0_DACHAR1_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port1_8 Switch", LM49453_P0_DACHAR1_REG, 7, 1, 0), +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_DACHAR2_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_DACHAR2_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_DACHAR2_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_DACHAR2_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_DACHAR2_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_DACHAR2_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port2_1 Switch", LM49453_P0_DACHAR2_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_DACHAR2_REG, 7, 1, 0), +SOC_DAPM_SINGLE("Sidetone Switch", LM49453_P0_STN_SEL_REG, 5, 0, 0), +}; + +static const struct snd_kcontrol_new lm49453_lineout_left_mixer[] = { +SOC_DAPM_SINGLE("Port1_1 Switch", LM49453_P0_DACLOL1_REG, 0, 1, 0), +SOC_DAPM_SINGLE("Port1_2 Switch", LM49453_P0_DACLOL1_REG, 1, 1, 0), +SOC_DAPM_SINGLE("Port1_3 Switch", LM49453_P0_DACLOL1_REG, 2, 1, 0), +SOC_DAPM_SINGLE("Port1_4 Switch", LM49453_P0_DACLOL1_REG, 3, 1, 0), +SOC_DAPM_SINGLE("Port1_5 Switch", LM49453_P0_DACLOL1_REG, 4, 1, 0), +SOC_DAPM_SINGLE("Port1_6 Switch", LM49453_P0_DACLOL1_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_7 Switch", LM49453_P0_DACLOL1_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port1_8 Switch", LM49453_P0_DACLOL1_REG, 7, 1, 0), +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_DACLOL2_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_DACLOL2_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_DACLOL2_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_DACLOL2_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_DACLOL2_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_DACLOL2_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port2_1 Switch", LM49453_P0_DACLOL2_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_DACLOL2_REG, 7, 1, 0), +SOC_DAPM_SINGLE("Sidetone Switch", LM49453_P0_STN_SEL_REG, 6, 0, 0), +}; + +static const struct snd_kcontrol_new lm49453_lineout_right_mixer[] = { +SOC_DAPM_SINGLE("Port1_1 Switch", LM49453_P0_DACLOR1_REG, 0, 1, 0), +SOC_DAPM_SINGLE("Port1_2 Switch", LM49453_P0_DACLOR1_REG, 1, 1, 0), +SOC_DAPM_SINGLE("Port1_3 Switch", LM49453_P0_DACLOR1_REG, 2, 1, 0), +SOC_DAPM_SINGLE("Port1_4 Switch", LM49453_P0_DACLOR1_REG, 3, 1, 0), +SOC_DAPM_SINGLE("Port1_5 Switch", LM49453_P0_DACLOR1_REG, 4, 1, 0), +SOC_DAPM_SINGLE("Port1_6 Switch", LM49453_P0_DACLOR1_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_7 Switch", LM49453_P0_DACLOR1_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port1_8 Switch", LM49453_P0_DACLOR1_REG, 7, 1, 0), +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_DACLOR2_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_DACLOR2_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_DACLOR2_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_DACLOR2_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_DACLOR2_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_DACLOR2_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port2_1 Switch", LM49453_P0_DACLOR2_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_DACLOR2_REG, 7, 1, 0), +SOC_DAPM_SINGLE("Sidetone Switch", LM49453_P0_STN_SEL_REG, 7, 0, 0), +}; + +static const struct snd_kcontrol_new lm49453_port1_tx1_mixer[] = { +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_PORT1_TX1_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_PORT1_TX1_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_PORT1_TX1_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_PORT1_TX1_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_PORT1_TX1_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_PORT1_TX1_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_1 Switch", LM49453_P0_PORT1_TX1_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port2_1 Switch", LM49453_P0_PORT1_TX1_REG, 7, 1, 0), +}; + +static const struct snd_kcontrol_new lm49453_port1_tx2_mixer[] = { +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_PORT1_TX2_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_PORT1_TX2_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_PORT1_TX2_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_PORT1_TX2_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_PORT1_TX2_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_PORT1_TX2_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_2 Switch", LM49453_P0_PORT1_TX2_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_PORT1_TX2_REG, 7, 1, 0), +}; + +static const struct snd_kcontrol_new lm49453_port1_tx3_mixer[] = { +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_PORT1_TX3_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_PORT1_TX3_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_PORT1_TX3_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_PORT1_TX3_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_PORT1_TX3_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_PORT1_TX3_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_3 Switch", LM49453_P0_PORT1_TX3_REG, 6, 1, 0), +}; + +static const struct snd_kcontrol_new lm49453_port1_tx4_mixer[] = { +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_PORT1_TX4_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_PORT1_TX4_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_PORT1_TX4_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_PORT1_TX4_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_PORT1_TX4_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_PORT1_TX4_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_4 Switch", LM49453_P0_PORT1_TX4_REG, 6, 1, 0), +}; + +static const struct snd_kcontrol_new lm49453_port1_tx5_mixer[] = { +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_PORT1_TX5_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_PORT1_TX5_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_PORT1_TX5_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_PORT1_TX5_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_PORT1_TX5_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_PORT1_TX5_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_5 Switch", LM49453_P0_PORT1_TX5_REG, 6, 1, 0), +}; + +static const struct snd_kcontrol_new lm49453_port1_tx6_mixer[] = { +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_PORT1_TX6_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_PORT1_TX6_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_PORT1_TX6_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_PORT1_TX6_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_PORT1_TX6_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_PORT1_TX6_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_6 Switch", LM49453_P0_PORT1_TX6_REG, 6, 1, 0), +}; + +static const struct snd_kcontrol_new lm49453_port1_tx7_mixer[] = { +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_PORT1_TX7_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_PORT1_TX7_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_PORT1_TX7_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_PORT1_TX7_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_PORT1_TX7_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_PORT1_TX7_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_7 Switch", LM49453_P0_PORT1_TX7_REG, 6, 1, 0), +}; + +static const struct snd_kcontrol_new lm49453_port1_tx8_mixer[] = { +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_PORT1_TX8_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_PORT1_TX8_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_PORT1_TX8_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_PORT1_TX8_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_PORT1_TX8_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_PORT1_TX8_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_8 Switch", LM49453_P0_PORT1_TX8_REG, 6, 1, 0), +}; + +static const struct snd_kcontrol_new lm49453_port2_tx1_mixer[] = { +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_PORT2_TX1_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_PORT2_TX1_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_PORT2_TX1_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_PORT2_TX1_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_PORT2_TX1_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_PORT2_TX1_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_1 Switch", LM49453_P0_PORT2_TX1_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port2_1 Switch", LM49453_P0_PORT2_TX1_REG, 7, 1, 0), +}; + +static const struct snd_kcontrol_new lm49453_port2_tx2_mixer[] = { +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_PORT2_TX2_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_PORT2_TX2_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_PORT2_TX2_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_PORT2_TX2_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_PORT2_TX2_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_PORT2_TX2_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_2 Switch", LM49453_P0_PORT2_TX2_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_PORT2_TX2_REG, 7, 1, 0), +}; + +/* TLV Declarations */ +static const DECLARE_TLV_DB_SCALE(digital_tlv, -7650, 150, 1); +static const DECLARE_TLV_DB_SCALE(port_tlv, 0, 600, 0); + +static const struct snd_kcontrol_new lm49453_sidetone_mixer_controls[] = { +/* Sidetone supports mono only */ +SOC_DAPM_SINGLE_TLV("Sidetone ADCL Volume", LM49453_P0_STN_VOL_ADCL_REG, + 0, 0x3F, 0, digital_tlv), +SOC_DAPM_SINGLE_TLV("Sidetone ADCR Volume", LM49453_P0_STN_VOL_ADCR_REG, + 0, 0x3F, 0, digital_tlv), +SOC_DAPM_SINGLE_TLV("Sidetone DMIC1L Volume", LM49453_P0_STN_VOL_DMIC1L_REG, + 0, 0x3F, 0, digital_tlv), +SOC_DAPM_SINGLE_TLV("Sidetone DMIC1R Volume", LM49453_P0_STN_VOL_DMIC1R_REG, + 0, 0x3F, 0, digital_tlv), +SOC_DAPM_SINGLE_TLV("Sidetone DMIC2L Volume", LM49453_P0_STN_VOL_DMIC2L_REG, + 0, 0x3F, 0, digital_tlv), +SOC_DAPM_SINGLE_TLV("Sidetone DMIC2R Volume", LM49453_P0_STN_VOL_DMIC2R_REG, + 0, 0x3F, 0, digital_tlv), +}; + +static const struct snd_kcontrol_new lm49453_snd_controls[] = { + /* mic1 and mic2 supports mono only */ + SOC_SINGLE_TLV("Mic1 Volume", LM49453_P0_ADC_LEVELL_REG, 0, 6, + 0, digital_tlv), + SOC_SINGLE_TLV("Mic2 Volume", LM49453_P0_ADC_LEVELR_REG, 0, 6, + 0, digital_tlv), + + SOC_DOUBLE_R_TLV("DMIC1 Volume", LM49453_P0_DMIC1_LEVELL_REG, + LM49453_P0_DMIC1_LEVELR_REG, 0, 6, 0, digital_tlv), + SOC_DOUBLE_R_TLV("DMIC2 Volume", LM49453_P0_DMIC2_LEVELL_REG, + LM49453_P0_DMIC2_LEVELR_REG, 0, 6, 0, digital_tlv), + + SOC_DAPM_ENUM("Mic2Mode", lm49453_mic2mode_enum), + SOC_DAPM_ENUM("DMIC12 SRC", lm49453_dmic12_cfg_enum), + SOC_DAPM_ENUM("DMIC34 SRC", lm49453_dmic34_cfg_enum), + + /* Capture path filter enable */ + SOC_SINGLE("DMIC1 HPFilter Switch", LM49453_P0_ADC_FX_ENABLES_REG, + 0, 1, 0), + SOC_SINGLE("DMIC2 HPFilter Switch", LM49453_P0_ADC_FX_ENABLES_REG, + 1, 1, 0), + SOC_SINGLE("ADC HPFilter Switch", LM49453_P0_ADC_FX_ENABLES_REG, + 2, 1, 0), + + SOC_DOUBLE_R_TLV("DAC HP Volume", LM49453_P0_DAC_HP_LEVELL_REG, + LM49453_P0_DAC_HP_LEVELR_REG, 0, 6, 0, digital_tlv), + SOC_DOUBLE_R_TLV("DAC LO Volume", LM49453_P0_DAC_LO_LEVELL_REG, + LM49453_P0_DAC_LO_LEVELR_REG, 0, 6, 0, digital_tlv), + SOC_DOUBLE_R_TLV("DAC LS Volume", LM49453_P0_DAC_LS_LEVELL_REG, + LM49453_P0_DAC_LS_LEVELR_REG, 0, 6, 0, digital_tlv), + SOC_DOUBLE_R_TLV("DAC HA Volume", LM49453_P0_DAC_HA_LEVELL_REG, + LM49453_P0_DAC_HA_LEVELR_REG, 0, 6, 0, digital_tlv), + + SOC_SINGLE_TLV("EP Volume", LM49453_P0_DAC_LS_LEVELL_REG, + 0, 6, 0, digital_tlv), + + SOC_SINGLE_TLV("PORT1_1_RX_LVL Volume", LM49453_P0_PORT1_RX_LVL1_REG, + 0, 3, 0, port_tlv), + SOC_SINGLE_TLV("PORT1_2_RX_LVL Volume", LM49453_P0_PORT1_RX_LVL1_REG, + 2, 3, 0, port_tlv), + SOC_SINGLE_TLV("PORT1_3_RX_LVL Volume", LM49453_P0_PORT1_RX_LVL1_REG, + 4, 3, 0, port_tlv), + SOC_SINGLE_TLV("PORT1_4_RX_LVL Volume", LM49453_P0_PORT1_RX_LVL1_REG, + 6, 3, 0, port_tlv), + SOC_SINGLE_TLV("PORT1_5_RX_LVL Volume", LM49453_P0_PORT1_RX_LVL2_REG, + 0, 3, 0, port_tlv), + SOC_SINGLE_TLV("PORT1_6_RX_LVL Volume", LM49453_P0_PORT1_RX_LVL2_REG, + 2, 3, 0, port_tlv), + SOC_SINGLE_TLV("PORT1_7_RX_LVL Volume", LM49453_P0_PORT1_RX_LVL2_REG, + 4, 3, 0, port_tlv), + SOC_SINGLE_TLV("PORT1_8_RX_LVL Volume", LM49453_P0_PORT1_RX_LVL2_REG, + 6, 3, 0, port_tlv), + + SOC_SINGLE_TLV("PORT2_1_RX_LVL Volume", LM49453_P0_PORT2_RX_LVL_REG, + 0, 3, 0, port_tlv), + SOC_SINGLE_TLV("PORT2_2_RX_LVL Volume", LM49453_P0_PORT2_RX_LVL_REG, + 2, 3, 0, port_tlv), + + SOC_SINGLE("Port1 Playback Switch", LM49453_P0_AUDIO_PORT1_BASIC_REG, + 1, 1, 0), + SOC_SINGLE("Port2 Playback Switch", LM49453_P0_AUDIO_PORT2_BASIC_REG, + 1, 1, 0), + SOC_SINGLE("Port1 Capture Switch", LM49453_P0_AUDIO_PORT1_BASIC_REG, + 2, 1, 0), + SOC_SINGLE("Port2 Capture Switch", LM49453_P0_AUDIO_PORT2_BASIC_REG, + 2, 1, 0) + +}; + +/* DAPM widgets */ +static const struct snd_soc_dapm_widget lm49453_dapm_widgets[] = { + + /* All end points HP,EP, LS, Lineout and Haptic */ + SND_SOC_DAPM_OUTPUT("HPOUTL"), + SND_SOC_DAPM_OUTPUT("HPOUTR"), + SND_SOC_DAPM_OUTPUT("EPOUT"), + SND_SOC_DAPM_OUTPUT("LSOUTL"), + SND_SOC_DAPM_OUTPUT("LSOUTR"), + SND_SOC_DAPM_OUTPUT("LOOUTR"), + SND_SOC_DAPM_OUTPUT("LOOUTL"), + SND_SOC_DAPM_OUTPUT("HAOUTL"), + SND_SOC_DAPM_OUTPUT("HAOUTR"), + + SND_SOC_DAPM_INPUT("AMIC1"), + SND_SOC_DAPM_INPUT("AMIC2"), + SND_SOC_DAPM_INPUT("DMIC1DAT"), + SND_SOC_DAPM_INPUT("DMIC2DAT"), + SND_SOC_DAPM_INPUT("AUXL"), + SND_SOC_DAPM_INPUT("AUXR"), + + SND_SOC_DAPM_PGA("PORT1_1_RX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("PORT1_2_RX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("PORT1_3_RX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("PORT1_4_RX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("PORT1_5_RX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("PORT1_6_RX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("PORT1_7_RX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("PORT1_8_RX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("PORT2_1_RX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("PORT2_2_RX", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_SUPPLY("AMIC1Bias", LM49453_P0_MICL_REG, 6, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("AMIC2Bias", LM49453_P0_MICR_REG, 6, 0, NULL, 0), + + /* playback path driver enables */ + SND_SOC_DAPM_OUT_DRV("Headset Switch", + LM49453_P0_PMC_SETUP_REG, 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("Earpiece Switch", + LM49453_P0_EP_REG, 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("Speaker Left Switch", + LM49453_P0_DIS_PKVL_FB_REG, 0, 1, NULL, 0), + SND_SOC_DAPM_OUT_DRV("Speaker Right Switch", + LM49453_P0_DIS_PKVL_FB_REG, 1, 1, NULL, 0), + SND_SOC_DAPM_OUT_DRV("Haptic Left Switch", + LM49453_P0_DIS_PKVL_FB_REG, 2, 1, NULL, 0), + SND_SOC_DAPM_OUT_DRV("Haptic Right Switch", + LM49453_P0_DIS_PKVL_FB_REG, 3, 1, NULL, 0), + + /* DAC */ + SND_SOC_DAPM_DAC("HPL DAC", "Headset", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("HPR DAC", "Headset", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("LSL DAC", "Speaker", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("LSR DAC", "Speaker", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("HAL DAC", "Haptic", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("HAR DAC", "Haptic", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("LOL DAC", "Lineout", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("LOR DAC", "Lineout", SND_SOC_NOPM, 0, 0), + + + SND_SOC_DAPM_PGA("AUXL Input", + LM49453_P0_ANALOG_MIXER_ADC_REG, 2, 0, NULL, 0), + SND_SOC_DAPM_PGA("AUXR Input", + LM49453_P0_ANALOG_MIXER_ADC_REG, 3, 0, NULL, 0), + + SND_SOC_DAPM_PGA("Sidetone", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* ADC */ + SND_SOC_DAPM_ADC("DMIC1 Left", "Capture", SND_SOC_NOPM, 1, 0), + SND_SOC_DAPM_ADC("DMIC1 Right", "Capture", SND_SOC_NOPM, 1, 0), + SND_SOC_DAPM_ADC("DMIC2 Left", "Capture", SND_SOC_NOPM, 1, 0), + SND_SOC_DAPM_ADC("DMIC2 Right", "Capture", SND_SOC_NOPM, 1, 0), + + SND_SOC_DAPM_ADC("ADC Left", "Capture", SND_SOC_NOPM, 1, 0), + SND_SOC_DAPM_ADC("ADC Right", "Capture", SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_MUX("ADCL Mux", SND_SOC_NOPM, 0, 0, + &lm49453_adcl_mux_control), + SND_SOC_DAPM_MUX("ADCR Mux", SND_SOC_NOPM, 0, 0, + &lm49453_adcr_mux_control), + + SND_SOC_DAPM_MUX("Mic1 Input", + SND_SOC_NOPM, 0, 0, &lm49453_adcl_mux_control), + + SND_SOC_DAPM_MUX("Mic2 Input", + SND_SOC_NOPM, 0, 0, &lm49453_adcr_mux_control), + + /* AIF */ + SND_SOC_DAPM_AIF_IN("PORT1_SDI", NULL, 0, + LM49453_P0_PULL_CONFIG1_REG, 2, 0), + SND_SOC_DAPM_AIF_IN("PORT2_SDI", NULL, 0, + LM49453_P0_PULL_CONFIG1_REG, 6, 0), + + SND_SOC_DAPM_AIF_OUT("PORT1_SDO", NULL, 0, + LM49453_P0_PULL_CONFIG1_REG, 3, 0), + SND_SOC_DAPM_AIF_OUT("PORT2_SDO", NULL, 0, + LM49453_P0_PULL_CONFIG1_REG, 7, 0), + + /* Port1 TX controls */ + SND_SOC_DAPM_OUT_DRV("P1_1_TX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("P1_2_TX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("P1_3_TX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("P1_4_TX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("P1_5_TX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("P1_6_TX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("P1_7_TX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("P1_8_TX", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* Port2 TX controls */ + SND_SOC_DAPM_OUT_DRV("P2_1_TX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("P2_2_TX", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* Sidetone Mixer */ + SND_SOC_DAPM_MIXER("Sidetone Mixer", SND_SOC_NOPM, 0, 0, + lm49453_sidetone_mixer_controls, + ARRAY_SIZE(lm49453_sidetone_mixer_controls)), + + /* DAC MIXERS */ + SND_SOC_DAPM_MIXER("HPL Mixer", SND_SOC_NOPM, 0, 0, + lm49453_headset_left_mixer, + ARRAY_SIZE(lm49453_headset_left_mixer)), + SND_SOC_DAPM_MIXER("HPR Mixer", SND_SOC_NOPM, 0, 0, + lm49453_headset_right_mixer, + ARRAY_SIZE(lm49453_headset_right_mixer)), + SND_SOC_DAPM_MIXER("LOL Mixer", SND_SOC_NOPM, 0, 0, + lm49453_lineout_left_mixer, + ARRAY_SIZE(lm49453_lineout_left_mixer)), + SND_SOC_DAPM_MIXER("LOR Mixer", SND_SOC_NOPM, 0, 0, + lm49453_lineout_right_mixer, + ARRAY_SIZE(lm49453_lineout_right_mixer)), + SND_SOC_DAPM_MIXER("LSL Mixer", SND_SOC_NOPM, 0, 0, + lm49453_speaker_left_mixer, + ARRAY_SIZE(lm49453_speaker_left_mixer)), + SND_SOC_DAPM_MIXER("LSR Mixer", SND_SOC_NOPM, 0, 0, + lm49453_speaker_right_mixer, + ARRAY_SIZE(lm49453_speaker_right_mixer)), + SND_SOC_DAPM_MIXER("HAL Mixer", SND_SOC_NOPM, 0, 0, + lm49453_haptic_left_mixer, + ARRAY_SIZE(lm49453_haptic_left_mixer)), + SND_SOC_DAPM_MIXER("HAR Mixer", SND_SOC_NOPM, 0, 0, + lm49453_haptic_right_mixer, + ARRAY_SIZE(lm49453_haptic_right_mixer)), + + /* Capture Mixer */ + SND_SOC_DAPM_MIXER("Port1_1 Mixer", SND_SOC_NOPM, 0, 0, + lm49453_port1_tx1_mixer, + ARRAY_SIZE(lm49453_port1_tx1_mixer)), + SND_SOC_DAPM_MIXER("Port1_2 Mixer", SND_SOC_NOPM, 0, 0, + lm49453_port1_tx2_mixer, + ARRAY_SIZE(lm49453_port1_tx2_mixer)), + SND_SOC_DAPM_MIXER("Port1_3 Mixer", SND_SOC_NOPM, 0, 0, + lm49453_port1_tx3_mixer, + ARRAY_SIZE(lm49453_port1_tx3_mixer)), + SND_SOC_DAPM_MIXER("Port1_4 Mixer", SND_SOC_NOPM, 0, 0, + lm49453_port1_tx4_mixer, + ARRAY_SIZE(lm49453_port1_tx4_mixer)), + SND_SOC_DAPM_MIXER("Port1_5 Mixer", SND_SOC_NOPM, 0, 0, + lm49453_port1_tx5_mixer, + ARRAY_SIZE(lm49453_port1_tx5_mixer)), + SND_SOC_DAPM_MIXER("Port1_6 Mixer", SND_SOC_NOPM, 0, 0, + lm49453_port1_tx6_mixer, + ARRAY_SIZE(lm49453_port1_tx6_mixer)), + SND_SOC_DAPM_MIXER("Port1_7 Mixer", SND_SOC_NOPM, 0, 0, + lm49453_port1_tx7_mixer, + ARRAY_SIZE(lm49453_port1_tx7_mixer)), + SND_SOC_DAPM_MIXER("Port1_8 Mixer", SND_SOC_NOPM, 0, 0, + lm49453_port1_tx8_mixer, + ARRAY_SIZE(lm49453_port1_tx8_mixer)), + + SND_SOC_DAPM_MIXER("Port2_1 Mixer", SND_SOC_NOPM, 0, 0, + lm49453_port2_tx1_mixer, + ARRAY_SIZE(lm49453_port2_tx1_mixer)), + SND_SOC_DAPM_MIXER("Port2_2 Mixer", SND_SOC_NOPM, 0, 0, + lm49453_port2_tx2_mixer, + ARRAY_SIZE(lm49453_port2_tx2_mixer)), +}; + +static const struct snd_soc_dapm_route lm49453_audio_map[] = { + /* Port SDI mapping */ + { "PORT1_1_RX", "Port1 Playback Switch", "PORT1_SDI" }, + { "PORT1_2_RX", "Port1 Playback Switch", "PORT1_SDI" }, + { "PORT1_3_RX", "Port1 Playback Switch", "PORT1_SDI" }, + { "PORT1_4_RX", "Port1 Playback Switch", "PORT1_SDI" }, + { "PORT1_5_RX", "Port1 Playback Switch", "PORT1_SDI" }, + { "PORT1_6_RX", "Port1 Playback Switch", "PORT1_SDI" }, + { "PORT1_7_RX", "Port1 Playback Switch", "PORT1_SDI" }, + { "PORT1_8_RX", "Port1 Playback Switch", "PORT1_SDI" }, + + { "PORT2_1_RX", "Port2 Playback Switch", "PORT2_SDI" }, + { "PORT2_2_RX", "Port2 Playback Switch", "PORT2_SDI" }, + + /* HP mapping */ + { "HPL Mixer", "Port1_1 Switch", "PORT1_1_RX" }, + { "HPL Mixer", "Port1_2 Switch", "PORT1_2_RX" }, + { "HPL Mixer", "Port1_3 Switch", "PORT1_3_RX" }, + { "HPL Mixer", "Port1_4 Switch", "PORT1_4_RX" }, + { "HPL Mixer", "Port1_5 Switch", "PORT1_5_RX" }, + { "HPL Mixer", "Port1_6 Switch", "PORT1_6_RX" }, + { "HPL Mixer", "Port1_7 Switch", "PORT1_7_RX" }, + { "HPL Mixer", "Port1_8 Switch", "PORT1_8_RX" }, + + { "HPL Mixer", "Port2_1 Switch", "PORT2_1_RX" }, + { "HPL Mixer", "Port2_2 Switch", "PORT2_2_RX" }, + + { "HPL Mixer", "ADCL Switch", "ADC Left" }, + { "HPL Mixer", "ADCR Switch", "ADC Right" }, + { "HPL Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "HPL Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "HPL Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "HPL Mixer", "DMIC2R Switch", "DMIC2 Right" }, + { "HPL Mixer", "Sidetone Switch", "Sidetone" }, + + { "HPL DAC", NULL, "HPL Mixer" }, + + { "HPR Mixer", "Port1_1 Switch", "PORT1_1_RX" }, + { "HPR Mixer", "Port1_2 Switch", "PORT1_2_RX" }, + { "HPR Mixer", "Port1_3 Switch", "PORT1_3_RX" }, + { "HPR Mixer", "Port1_4 Switch", "PORT1_4_RX" }, + { "HPR Mixer", "Port1_5 Switch", "PORT1_5_RX" }, + { "HPR Mixer", "Port1_6 Switch", "PORT1_6_RX" }, + { "HPR Mixer", "Port1_7 Switch", "PORT1_7_RX" }, + { "HPR Mixer", "Port1_8 Switch", "PORT1_8_RX" }, + + /* Port 2 */ + { "HPR Mixer", "Port2_1 Switch", "PORT2_1_RX" }, + { "HPR Mixer", "Port2_2 Switch", "PORT2_2_RX" }, + + { "HPR Mixer", "ADCL Switch", "ADC Left" }, + { "HPR Mixer", "ADCR Switch", "ADC Right" }, + { "HPR Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "HPR Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "HPR Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "HPR Mixer", "DMIC2L Switch", "DMIC2 Right" }, + { "HPR Mixer", "Sidetone Switch", "Sidetone" }, + + { "HPR DAC", NULL, "HPR Mixer" }, + + { "HPOUTL", "Headset Switch", "HPL DAC"}, + { "HPOUTR", "Headset Switch", "HPR DAC"}, + + /* EP map */ + { "EPOUT", "Earpiece Switch", "HPL DAC" }, + + /* Speaker map */ + { "LSL Mixer", "Port1_1 Switch", "PORT1_1_RX" }, + { "LSL Mixer", "Port1_2 Switch", "PORT1_2_RX" }, + { "LSL Mixer", "Port1_3 Switch", "PORT1_3_RX" }, + { "LSL Mixer", "Port1_4 Switch", "PORT1_4_RX" }, + { "LSL Mixer", "Port1_5 Switch", "PORT1_5_RX" }, + { "LSL Mixer", "Port1_6 Switch", "PORT1_6_RX" }, + { "LSL Mixer", "Port1_7 Switch", "PORT1_7_RX" }, + { "LSL Mixer", "Port1_8 Switch", "PORT1_8_RX" }, + + /* Port 2 */ + { "LSL Mixer", "Port2_1 Switch", "PORT2_1_RX" }, + { "LSL Mixer", "Port2_2 Switch", "PORT2_2_RX" }, + + { "LSL Mixer", "ADCL Switch", "ADC Left" }, + { "LSL Mixer", "ADCR Switch", "ADC Right" }, + { "LSL Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "LSL Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "LSL Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "LSL Mixer", "DMIC2R Switch", "DMIC2 Right" }, + { "LSL Mixer", "Sidetone Switch", "Sidetone" }, + + { "LSL DAC", NULL, "LSL Mixer" }, + + { "LSR Mixer", "Port1_1 Switch", "PORT1_1_RX" }, + { "LSR Mixer", "Port1_2 Switch", "PORT1_2_RX" }, + { "LSR Mixer", "Port1_3 Switch", "PORT1_3_RX" }, + { "LSR Mixer", "Port1_4 Switch", "PORT1_4_RX" }, + { "LSR Mixer", "Port1_5 Switch", "PORT1_5_RX" }, + { "LSR Mixer", "Port1_6 Switch", "PORT1_6_RX" }, + { "LSR Mixer", "Port1_7 Switch", "PORT1_7_RX" }, + { "LSR Mixer", "Port1_8 Switch", "PORT1_8_RX" }, + + /* Port 2 */ + { "LSR Mixer", "Port2_1 Switch", "PORT2_1_RX" }, + { "LSR Mixer", "Port2_2 Switch", "PORT2_2_RX" }, + + { "LSR Mixer", "ADCL Switch", "ADC Left" }, + { "LSR Mixer", "ADCR Switch", "ADC Right" }, + { "LSR Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "LSR Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "LSR Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "LSR Mixer", "DMIC2R Switch", "DMIC2 Right" }, + { "LSR Mixer", "Sidetone Switch", "Sidetone" }, + + { "LSR DAC", NULL, "LSR Mixer" }, + + { "LSOUTL", "Speaker Left Switch", "LSL DAC"}, + { "LSOUTR", "Speaker Left Switch", "LSR DAC"}, + + /* Haptic map */ + { "HAL Mixer", "Port1_1 Switch", "PORT1_1_RX" }, + { "HAL Mixer", "Port1_2 Switch", "PORT1_2_RX" }, + { "HAL Mixer", "Port1_3 Switch", "PORT1_3_RX" }, + { "HAL Mixer", "Port1_4 Switch", "PORT1_4_RX" }, + { "HAL Mixer", "Port1_5 Switch", "PORT1_5_RX" }, + { "HAL Mixer", "Port1_6 Switch", "PORT1_6_RX" }, + { "HAL Mixer", "Port1_7 Switch", "PORT1_7_RX" }, + { "HAL Mixer", "Port1_8 Switch", "PORT1_8_RX" }, + + /* Port 2 */ + { "HAL Mixer", "Port2_1 Switch", "PORT2_1_RX" }, + { "HAL Mixer", "Port2_2 Switch", "PORT2_2_RX" }, + + { "HAL Mixer", "ADCL Switch", "ADC Left" }, + { "HAL Mixer", "ADCR Switch", "ADC Right" }, + { "HAL Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "HAL Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "HAL Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "HAL Mixer", "DMIC2R Switch", "DMIC2 Right" }, + { "HAL Mixer", "Sidetone Switch", "Sidetone" }, + + { "HAL DAC", NULL, "HAL Mixer" }, + + { "HAR Mixer", "Port1_1 Switch", "PORT1_1_RX" }, + { "HAR Mixer", "Port1_2 Switch", "PORT1_2_RX" }, + { "HAR Mixer", "Port1_3 Switch", "PORT1_3_RX" }, + { "HAR Mixer", "Port1_4 Switch", "PORT1_4_RX" }, + { "HAR Mixer", "Port1_5 Switch", "PORT1_5_RX" }, + { "HAR Mixer", "Port1_6 Switch", "PORT1_6_RX" }, + { "HAR Mixer", "Port1_7 Switch", "PORT1_7_RX" }, + { "HAR Mixer", "Port1_8 Switch", "PORT1_8_RX" }, + + /* Port 2 */ + { "HAR Mixer", "Port2_1 Switch", "PORT2_1_RX" }, + { "HAR Mixer", "Port2_2 Switch", "PORT2_2_RX" }, + + { "HAR Mixer", "ADCL Switch", "ADC Left" }, + { "HAR Mixer", "ADCR Switch", "ADC Right" }, + { "HAR Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "HAR Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "HAR Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "HAR Mixer", "DMIC2R Switch", "DMIC2 Right" }, + { "HAR Mixer", "Sideton Switch", "Sidetone" }, + + { "HAR DAC", NULL, "HAR Mixer" }, + + { "HAOUTL", "Haptic Left Switch", "HAL DAC" }, + { "HAOUTR", "Haptic Right Switch", "HAR DAC" }, + + /* Lineout map */ + { "LOL Mixer", "Port1_1 Switch", "PORT1_1_RX" }, + { "LOL Mixer", "Port1_2 Switch", "PORT1_2_RX" }, + { "LOL Mixer", "Port1_3 Switch", "PORT1_3_RX" }, + { "LOL Mixer", "Port1_4 Switch", "PORT1_4_RX" }, + { "LOL Mixer", "Port1_5 Switch", "PORT1_5_RX" }, + { "LOL Mixer", "Port1_6 Switch", "PORT1_6_RX" }, + { "LOL Mixer", "Port1_7 Switch", "PORT1_7_RX" }, + { "LOL Mixer", "Port1_8 Switch", "PORT1_8_RX" }, + + /* Port 2 */ + { "LOL Mixer", "Port2_1 Switch", "PORT2_1_RX" }, + { "LOL Mixer", "Port2_2 Switch", "PORT2_2_RX" }, + + { "LOL Mixer", "ADCL Switch", "ADC Left" }, + { "LOL Mixer", "ADCR Switch", "ADC Right" }, + { "LOL Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "LOL Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "LOL Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "LOL Mixer", "DMIC2R Switch", "DMIC2 Right" }, + { "LOL Mixer", "Sidetone Switch", "Sidetone" }, + + { "LOL DAC", NULL, "LOL Mixer" }, + + { "LOR Mixer", "Port1_1 Switch", "PORT1_1_RX" }, + { "LOR Mixer", "Port1_2 Switch", "PORT1_2_RX" }, + { "LOR Mixer", "Port1_3 Switch", "PORT1_3_RX" }, + { "LOR Mixer", "Port1_4 Switch", "PORT1_4_RX" }, + { "LOR Mixer", "Port1_5 Switch", "PORT1_5_RX" }, + { "LOR Mixer", "Port1_6 Switch", "PORT1_6_RX" }, + { "LOR Mixer", "Port1_7 Switch", "PORT1_7_RX" }, + { "LOR Mixer", "Port1_8 Switch", "PORT1_8_RX" }, + + /* Port 2 */ + { "LOR Mixer", "Port2_1 Switch", "PORT2_1_RX" }, + { "LOR Mixer", "Port2_2 Switch", "PORT2_2_RX" }, + + { "LOR Mixer", "ADCL Switch", "ADC Left" }, + { "LOR Mixer", "ADCR Switch", "ADC Right" }, + { "LOR Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "LOR Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "LOR Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "LOR Mixer", "DMIC2R Switch", "DMIC2 Right" }, + { "LOR Mixer", "Sidetone Switch", "Sidetone" }, + + { "LOR DAC", NULL, "LOR Mixer" }, + + { "LOOUTL", NULL, "LOL DAC" }, + { "LOOUTR", NULL, "LOR DAC" }, + + /* TX map */ + /* Port1 mappings */ + { "Port1_1 Mixer", "ADCL Switch", "ADC Left" }, + { "Port1_1 Mixer", "ADCR Switch", "ADC Right" }, + { "Port1_1 Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "Port1_1 Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "Port1_1 Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "Port1_1 Mixer", "DMIC2R Switch", "DMIC2 Right" }, + + { "Port1_2 Mixer", "ADCL Switch", "ADC Left" }, + { "Port1_2 Mixer", "ADCR Switch", "ADC Right" }, + { "Port1_2 Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "Port1_2 Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "Port1_2 Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "Port1_2 Mixer", "DMIC2R Switch", "DMIC2 Right" }, + + { "Port1_3 Mixer", "ADCL Switch", "ADC Left" }, + { "Port1_3 Mixer", "ADCR Switch", "ADC Right" }, + { "Port1_3 Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "Port1_3 Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "Port1_3 Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "Port1_3 Mixer", "DMIC2R Switch", "DMIC2 Right" }, + + { "Port1_4 Mixer", "ADCL Switch", "ADC Left" }, + { "Port1_4 Mixer", "ADCR Switch", "ADC Right" }, + { "Port1_4 Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "Port1_4 Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "Port1_4 Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "Port1_4 Mixer", "DMIC2R Switch", "DMIC2 Right" }, + + { "Port1_5 Mixer", "ADCL Switch", "ADC Left" }, + { "Port1_5 Mixer", "ADCR Switch", "ADC Right" }, + { "Port1_5 Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "Port1_5 Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "Port1_5 Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "Port1_5 Mixer", "DMIC2R Switch", "DMIC2 Right" }, + + { "Port1_6 Mixer", "ADCL Switch", "ADC Left" }, + { "Port1_6 Mixer", "ADCR Switch", "ADC Right" }, + { "Port1_6 Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "Port1_6 Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "Port1_6 Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "Port1_6 Mixer", "DMIC2R Switch", "DMIC2 Right" }, + + { "Port1_7 Mixer", "ADCL Switch", "ADC Left" }, + { "Port1_7 Mixer", "ADCR Switch", "ADC Right" }, + { "Port1_7 Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "Port1_7 Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "Port1_7 Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "Port1_7 Mixer", "DMIC2R Switch", "DMIC2 Right" }, + + { "Port1_8 Mixer", "ADCL Switch", "ADC Left" }, + { "Port1_8 Mixer", "ADCR Switch", "ADC Right" }, + { "Port1_8 Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "Port1_8 Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "Port1_8 Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "Port1_8 Mixer", "DMIC2R Switch", "DMIC2 Right" }, + + { "Port2_1 Mixer", "ADCL Switch", "ADC Left" }, + { "Port2_1 Mixer", "ADCR Switch", "ADC Right" }, + { "Port2_1 Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "Port2_1 Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "Port2_1 Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "Port2_1 Mixer", "DMIC2R Switch", "DMIC2 Right" }, + + { "Port2_2 Mixer", "ADCL Switch", "ADC Left" }, + { "Port2_2 Mixer", "ADCR Switch", "ADC Right" }, + { "Port2_2 Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "Port2_2 Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "Port2_2 Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "Port2_2 Mixer", "DMIC2R Switch", "DMIC2 Right" }, + + { "P1_1_TX", NULL, "Port1_1 Mixer" }, + { "P1_2_TX", NULL, "Port1_2 Mixer" }, + { "P1_3_TX", NULL, "Port1_3 Mixer" }, + { "P1_4_TX", NULL, "Port1_4 Mixer" }, + { "P1_5_TX", NULL, "Port1_5 Mixer" }, + { "P1_6_TX", NULL, "Port1_6 Mixer" }, + { "P1_7_TX", NULL, "Port1_7 Mixer" }, + { "P1_8_TX", NULL, "Port1_8 Mixer" }, + + { "P2_1_TX", NULL, "Port2_1 Mixer" }, + { "P2_2_TX", NULL, "Port2_2 Mixer" }, + + { "PORT1_SDO", "Port1 Capture Switch", "P1_1_TX"}, + { "PORT1_SDO", "Port1 Capture Switch", "P1_2_TX"}, + { "PORT1_SDO", "Port1 Capture Switch", "P1_3_TX"}, + { "PORT1_SDO", "Port1 Capture Switch", "P1_4_TX"}, + { "PORT1_SDO", "Port1 Capture Switch", "P1_5_TX"}, + { "PORT1_SDO", "Port1 Capture Switch", "P1_6_TX"}, + { "PORT1_SDO", "Port1 Capture Switch", "P1_7_TX"}, + { "PORT1_SDO", "Port1 Capture Switch", "P1_8_TX"}, + + { "PORT2_SDO", "Port2 Capture Switch", "P2_1_TX"}, + { "PORT2_SDO", "Port2 Capture Switch", "P2_2_TX"}, + + { "Mic1 Input", NULL, "AMIC1" }, + { "Mic2 Input", NULL, "AMIC2" }, + + { "AUXL Input", NULL, "AUXL" }, + { "AUXR Input", NULL, "AUXR" }, + + /* AUX connections */ + { "ADCL Mux", "Aux_L", "AUXL Input" }, + { "ADCL Mux", "MIC1", "Mic1 Input" }, + + { "ADCR Mux", "Aux_R", "AUXR Input" }, + { "ADCR Mux", "MIC2", "Mic2 Input" }, + + /* ADC connection */ + { "ADC Left", NULL, "ADCL Mux"}, + { "ADC Right", NULL, "ADCR Mux"}, + + { "DMIC1 Left", NULL, "DMIC1DAT"}, + { "DMIC1 Right", NULL, "DMIC1DAT"}, + { "DMIC2 Left", NULL, "DMIC2DAT"}, + { "DMIC2 Right", NULL, "DMIC2DAT"}, + + /* Sidetone map */ + { "Sidetone Mixer", NULL, "ADC Left" }, + { "Sidetone Mixer", NULL, "ADC Right" }, + { "Sidetone Mixer", NULL, "DMIC1 Left" }, + { "Sidetone Mixer", NULL, "DMIC1 Right" }, + { "Sidetone Mixer", NULL, "DMIC2 Left" }, + { "Sidetone Mixer", NULL, "DMIC2 Right" }, + + { "Sidetone", "Sidetone Switch", "Sidetone Mixer" }, +}; + +static int lm49453_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + struct lm49453_priv *lm49453 = snd_soc_codec_get_drvdata(codec); + u16 clk_div = 0; + + lm49453->fs_rate = params_rate(params); + + /* Setting DAC clock dividers based on substream sample rate. */ + switch (lm49453->fs_rate) { + case 8000: + case 16000: + case 32000: + case 24000: + case 48000: + clk_div = 256; + break; + case 11025: + case 22050: + case 44100: + clk_div = 216; + break; + case 96000: + clk_div = 127; + break; + default: + return -EINVAL; + } + + snd_soc_write(codec, LM49453_P0_ADC_CLK_DIV_REG, clk_div); + snd_soc_write(codec, LM49453_P0_DAC_HP_CLK_DIV_REG, clk_div); + + return 0; +} + +static int lm49453_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + + int aif_val = 0; + int mode = 0; + int clk_phase = 0; + int clk_shift = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + aif_val = ~LM49453_AUDIO_PORT1_BASIC_CLK_MS | + ~LM49453_AUDIO_PORT1_BASIC_SYNC_MS; + break; + case SND_SOC_DAIFMT_CBS_CFM: + aif_val = ~LM49453_AUDIO_PORT1_BASIC_CLK_MS | + LM49453_AUDIO_PORT1_BASIC_SYNC_MS; + break; + case SND_SOC_DAIFMT_CBM_CFS: + aif_val = LM49453_AUDIO_PORT1_BASIC_CLK_MS | + ~LM49453_AUDIO_PORT1_BASIC_SYNC_MS; + break; + case SND_SOC_DAIFMT_CBM_CFM: + aif_val = LM49453_AUDIO_PORT1_BASIC_CLK_MS | + LM49453_AUDIO_PORT1_BASIC_SYNC_MS; + break; + default: + return -EINVAL; + } + + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + case SND_SOC_DAIFMT_DSP_A: + mode = 1; + clk_phase = (1 << 5); + clk_shift = 1; + break; + case SND_SOC_DAIFMT_DSP_B: + mode = 1; + clk_phase = (1 << 5); + clk_shift = 0; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, LM49453_P0_AUDIO_PORT1_BASIC_REG, + LM49453_AUDIO_PORT1_BASIC_FMT_MASK|BIT(1)|BIT(5), + (aif_val | mode | clk_phase)); + + snd_soc_write(codec, LM49453_P0_AUDIO_PORT1_RX_MSB_REG, clk_shift); + + return 0; +} + +static int lm49453_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + u16 pll_clk = 0; + + switch (freq) { + case 12288000: + case 26000000: + case 19200000: + /* pll clk slection */ + pll_clk = 0; + break; + case 48000: + case 32576: + /* fll clk slection */ + pll_clk = BIT(4); + return 0; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, LM49453_P0_PMC_SETUP_REG, BIT(4), pll_clk); + + return 0; +} + +static int lm49453_hp_mute(struct snd_soc_dai *dai, int mute) +{ + snd_soc_update_bits(dai->codec, LM49453_P0_DAC_DSP_REG, BIT(1)|BIT(0), + (mute ? (BIT(1)|BIT(0)) : 0)); + return 0; +} + +static int lm49453_lo_mute(struct snd_soc_dai *dai, int mute) +{ + snd_soc_update_bits(dai->codec, LM49453_P0_DAC_DSP_REG, BIT(3)|BIT(2), + (mute ? (BIT(3)|BIT(2)) : 0)); + return 0; +} + +static int lm49453_ls_mute(struct snd_soc_dai *dai, int mute) +{ + snd_soc_update_bits(dai->codec, LM49453_P0_DAC_DSP_REG, BIT(5)|BIT(4), + (mute ? (BIT(5)|BIT(4)) : 0)); + return 0; +} + +static int lm49453_ep_mute(struct snd_soc_dai *dai, int mute) +{ + snd_soc_update_bits(dai->codec, LM49453_P0_DAC_DSP_REG, BIT(4), + (mute ? BIT(4) : 0)); + return 0; +} + +static int lm49453_ha_mute(struct snd_soc_dai *dai, int mute) +{ + snd_soc_update_bits(dai->codec, LM49453_P0_DAC_DSP_REG, BIT(7)|BIT(6), + (mute ? (BIT(7)|BIT(6)) : 0)); + return 0; +} + +static int lm49453_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct lm49453_priv *lm49453 = snd_soc_codec_get_drvdata(codec); + + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + break; + + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) + regcache_sync(lm49453->regmap); + + snd_soc_update_bits(codec, LM49453_P0_PMC_SETUP_REG, + LM49453_PMC_SETUP_CHIP_EN, LM49453_CHIP_EN); + break; + + case SND_SOC_BIAS_OFF: + snd_soc_update_bits(codec, LM49453_P0_PMC_SETUP_REG, + LM49453_PMC_SETUP_CHIP_EN, 0); + break; + } + + codec->dapm.bias_level = level; + + return 0; +} + +/* Formates supported by LM49453 driver. */ +#define LM49453_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_ops lm49453_headset_dai_ops = { + .hw_params = lm49453_hw_params, + .set_sysclk = lm49453_set_dai_sysclk, + .set_fmt = lm49453_set_dai_fmt, + .digital_mute = lm49453_hp_mute, +}; + +static struct snd_soc_dai_ops lm49453_speaker_dai_ops = { + .hw_params = lm49453_hw_params, + .set_sysclk = lm49453_set_dai_sysclk, + .set_fmt = lm49453_set_dai_fmt, + .digital_mute = lm49453_ls_mute, +}; + +static struct snd_soc_dai_ops lm49453_haptic_dai_ops = { + .hw_params = lm49453_hw_params, + .set_sysclk = lm49453_set_dai_sysclk, + .set_fmt = lm49453_set_dai_fmt, + .digital_mute = lm49453_ha_mute, +}; + +static struct snd_soc_dai_ops lm49453_ep_dai_ops = { + .hw_params = lm49453_hw_params, + .set_sysclk = lm49453_set_dai_sysclk, + .set_fmt = lm49453_set_dai_fmt, + .digital_mute = lm49453_ep_mute, +}; + +static struct snd_soc_dai_ops lm49453_lineout_dai_ops = { + .hw_params = lm49453_hw_params, + .set_sysclk = lm49453_set_dai_sysclk, + .set_fmt = lm49453_set_dai_fmt, + .digital_mute = lm49453_lo_mute, +}; + +/* LM49453 dai structure. */ +struct snd_soc_dai_driver lm49453_dai[] = { + { + .name = "LM49453 Headset", + .playback = { + .stream_name = "Headset", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = LM49453_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 5, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = LM49453_FORMATS, + }, + .ops = &lm49453_headset_dai_ops, + .symmetric_rates = 1, + }, + { + .name = "LM49453 Speaker", + .playback = { + .stream_name = "Speaker", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = LM49453_FORMATS, + }, + .ops = &lm49453_speaker_dai_ops, + }, + { + .name = "LM49453 Haptic", + .playback = { + .stream_name = "Haptic", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = LM49453_FORMATS, + }, + .ops = &lm49453_haptic_dai_ops, + }, + { + .name = "LM49453 Earpiece", + .playback = { + .stream_name = "Earpiece", + .channels_min = 1, + .channels_max = 1, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = LM49453_FORMATS, + }, + .ops = &lm49453_ep_dai_ops, + }, + { + .name = "LM49453 line out", + .playback = { + .stream_name = "Lineout", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = LM49453_FORMATS, + }, + .ops = &lm49453_lineout_dai_ops, + }, +}; + +static int lm49453_suspend(struct snd_soc_codec *codec) +{ + lm49453_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int lm49453_resume(struct snd_soc_codec *codec) +{ + lm49453_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + return 0; +} + +static int lm49453_probe(struct snd_soc_codec *codec) +{ + struct lm49453_priv *lm49453 = snd_soc_codec_get_drvdata(codec); + int ret = 0; + + codec->control_data = lm49453->regmap; + + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + return 0; +} + +/* power down chip */ +static int lm49453_remove(struct snd_soc_codec *codec) +{ + lm49453_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_lm49453 = { + .probe = lm49453_probe, + .remove = lm49453_remove, + .suspend = lm49453_suspend, + .resume = lm49453_resume, + .set_bias_level = lm49453_set_bias_level, + .controls = lm49453_snd_controls, + .num_controls = ARRAY_SIZE(lm49453_snd_controls), + .dapm_widgets = lm49453_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(lm49453_dapm_widgets), + .dapm_routes = lm49453_audio_map, + .num_dapm_routes = ARRAY_SIZE(lm49453_audio_map), + .idle_bias_off = true, +}; + +static const struct regmap_config lm49453_regmap_config = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = LM49453_MAX_REGISTER, + .reg_defaults = lm49453_reg_defs, + .num_reg_defaults = ARRAY_SIZE(lm49453_reg_defs), + .cache_type = REGCACHE_RBTREE, +}; + +static __devinit int lm49453_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct lm49453_priv *lm49453; + int ret = 0; + + lm49453 = devm_kzalloc(&i2c->dev, sizeof(struct lm49453_priv), + GFP_KERNEL); + + if (lm49453 == NULL) + return -ENOMEM; + + i2c_set_clientdata(i2c, lm49453); + + lm49453->regmap = regmap_init_i2c(i2c, &lm49453_regmap_config); + if (IS_ERR(lm49453->regmap)) { + ret = PTR_ERR(lm49453->regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", + ret); + return ret; + } + + ret = snd_soc_register_codec(&i2c->dev, + &soc_codec_dev_lm49453, + lm49453_dai, ARRAY_SIZE(lm49453_dai)); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to register codec: %d\n", ret); + regmap_exit(lm49453->regmap); + return ret; + } + + return ret; +} + +static int __devexit lm49453_i2c_remove(struct i2c_client *client) +{ + struct lm49453_priv *lm49453 = i2c_get_clientdata(client); + + snd_soc_unregister_codec(&client->dev); + regmap_exit(lm49453->regmap); + return 0; +} + +static const struct i2c_device_id lm49453_i2c_id[] = { + { "lm49453", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, lm49453_i2c_id); + +static struct i2c_driver lm49453_i2c_driver = { + .driver = { + .name = "lm49453", + .owner = THIS_MODULE, + }, + .probe = lm49453_i2c_probe, + .remove = __devexit_p(lm49453_i2c_remove), + .id_table = lm49453_i2c_id, +}; + +module_i2c_driver(lm49453_i2c_driver); + +MODULE_DESCRIPTION("ASoC LM49453 driver"); +MODULE_AUTHOR("M R Swami Reddy + +/* LM49453_P0 register space for page0 */ +#define LM49453_P0_PMC_SETUP_REG 0x00 +#define LM49453_P0_PLL_CLK_SEL1_REG 0x01 +#define LM49453_P0_PLL_CLK_SEL2_REG 0x02 +#define LM49453_P0_PMC_CLK_DIV_REG 0x03 +#define LM49453_P0_HSDET_CLK_DIV_REG 0x04 +#define LM49453_P0_DMIC_CLK_DIV_REG 0x05 +#define LM49453_P0_ADC_CLK_DIV_REG 0x06 +#define LM49453_P0_DAC_OT_CLK_DIV_REG 0x07 +#define LM49453_P0_PLL_HF_M_REG 0x08 +#define LM49453_P0_PLL_LF_M_REG 0x09 +#define LM49453_P0_PLL_NL_REG 0x0A +#define LM49453_P0_PLL_N_MODL_REG 0x0B +#define LM49453_P0_PLL_N_MODH_REG 0x0C +#define LM49453_P0_PLL_P1_REG 0x0D +#define LM49453_P0_PLL_P2_REG 0x0E +#define LM49453_P0_FLL_REF_FREQL_REG 0x0F +#define LM49453_P0_FLL_REF_FREQH_REG 0x10 +#define LM49453_P0_VCO_TARGETLL_REG 0x11 +#define LM49453_P0_VCO_TARGETLH_REG 0x12 +#define LM49453_P0_VCO_TARGETHL_REG 0x13 +#define LM49453_P0_VCO_TARGETHH_REG 0x14 +#define LM49453_P0_PLL_CONFIG_REG 0x15 +#define LM49453_P0_DAC_CLK_SEL_REG 0x16 +#define LM49453_P0_DAC_HP_CLK_DIV_REG 0x17 + +/* Analog Mixer Input Stages */ +#define LM49453_P0_MICL_REG 0x20 +#define LM49453_P0_MICR_REG 0x21 +#define LM49453_P0_EP_REG 0x24 +#define LM49453_P0_DIS_PKVL_FB_REG 0x25 + +/* Analog Mixer Output Stages */ +#define LM49453_P0_ANALOG_MIXER_ADC_REG 0x2E + +/*ADC or DAC */ +#define LM49453_P0_ADC_DSP_REG 0x30 +#define LM49453_P0_DAC_DSP_REG 0x31 + +/* EFFECTS ENABLES */ +#define LM49453_P0_ADC_FX_ENABLES_REG 0x33 + +/* GPIO */ +#define LM49453_P0_GPIO1_REG 0x38 +#define LM49453_P0_GPIO2_REG 0x39 +#define LM49453_P0_GPIO3_REG 0x3A +#define LM49453_P0_HAP_CTL_REG 0x3B +#define LM49453_P0_HAP_FREQ_PROG_LEFTL_REG 0x3C +#define LM49453_P0_HAP_FREQ_PROG_LEFTH_REG 0x3D +#define LM49453_P0_HAP_FREQ_PROG_RIGHTL_REG 0x3E +#define LM49453_P0_HAP_FREQ_PROG_RIGHTH_REG 0x3F + +/* DIGITAL MIXER */ +#define LM49453_P0_DMIX_CLK_SEL_REG 0x40 +#define LM49453_P0_PORT1_RX_LVL1_REG 0x41 +#define LM49453_P0_PORT1_RX_LVL2_REG 0x42 +#define LM49453_P0_PORT2_RX_LVL_REG 0x43 +#define LM49453_P0_PORT1_TX1_REG 0x44 +#define LM49453_P0_PORT1_TX2_REG 0x45 +#define LM49453_P0_PORT1_TX3_REG 0x46 +#define LM49453_P0_PORT1_TX4_REG 0x47 +#define LM49453_P0_PORT1_TX5_REG 0x48 +#define LM49453_P0_PORT1_TX6_REG 0x49 +#define LM49453_P0_PORT1_TX7_REG 0x4A +#define LM49453_P0_PORT1_TX8_REG 0x4B +#define LM49453_P0_PORT2_TX1_REG 0x4C +#define LM49453_P0_PORT2_TX2_REG 0x4D +#define LM49453_P0_STN_SEL_REG 0x4F +#define LM49453_P0_DACHPL1_REG 0x50 +#define LM49453_P0_DACHPL2_REG 0x51 +#define LM49453_P0_DACHPR1_REG 0x52 +#define LM49453_P0_DACHPR2_REG 0x53 +#define LM49453_P0_DACLOL1_REG 0x54 +#define LM49453_P0_DACLOL2_REG 0x55 +#define LM49453_P0_DACLOR1_REG 0x56 +#define LM49453_P0_DACLOR2_REG 0x57 +#define LM49453_P0_DACLSL1_REG 0x58 +#define LM49453_P0_DACLSL2_REG 0x59 +#define LM49453_P0_DACLSR1_REG 0x5A +#define LM49453_P0_DACLSR2_REG 0x5B +#define LM49453_P0_DACHAL1_REG 0x5C +#define LM49453_P0_DACHAL2_REG 0x5D +#define LM49453_P0_DACHAR1_REG 0x5E +#define LM49453_P0_DACHAR2_REG 0x5F + +/* AUDIO PORT 1 (TDM) */ +#define LM49453_P0_AUDIO_PORT1_BASIC_REG 0x60 +#define LM49453_P0_AUDIO_PORT1_CLK_GEN1_REG 0x61 +#define LM49453_P0_AUDIO_PORT1_CLK_GEN2_REG 0x62 +#define LM49453_P0_AUDIO_PORT1_CLK_GEN3_REG 0x63 +#define LM49453_P0_AUDIO_PORT1_SYNC_RATE_REG 0x64 +#define LM49453_P0_AUDIO_PORT1_SYNC_SDO_SETUP_REG 0x65 +#define LM49453_P0_AUDIO_PORT1_DATA_WIDTH_REG 0x66 +#define LM49453_P0_AUDIO_PORT1_RX_MSB_REG 0x67 +#define LM49453_P0_AUDIO_PORT1_TX_MSB_REG 0x68 +#define LM49453_P0_AUDIO_PORT1_TDM_CHANNELS_REG 0x69 + +/* AUDIO PORT 2 */ +#define LM49453_P0_AUDIO_PORT2_BASIC_REG 0x6A +#define LM49453_P0_AUDIO_PORT2_CLK_GEN1_REG 0x6B +#define LM49453_P0_AUDIO_PORT2_CLK_GEN2_REG 0x6C +#define LM49453_P0_AUDIO_PORT2_SYNC_GEN_REG 0x6D +#define LM49453_P0_AUDIO_PORT2_DATA_WIDTH_REG 0x6E +#define LM49453_P0_AUDIO_PORT2_RX_MODE_REG 0x6F +#define LM49453_P0_AUDIO_PORT2_TX_MODE_REG 0x70 + +/* SAMPLE RATE */ +#define LM49453_P0_PORT1_SR_LSB_REG 0x79 +#define LM49453_P0_PORT1_SR_MSB_REG 0x7A +#define LM49453_P0_PORT2_SR_LSB_REG 0x7B +#define LM49453_P0_PORT2_SR_MSB_REG 0x7C + +/* EFFECTS - HPFs */ +#define LM49453_P0_HPF_REG 0x80 + +/* EFFECTS ADC ALC */ +#define LM49453_P0_ADC_ALC1_REG 0x82 +#define LM49453_P0_ADC_ALC2_REG 0x83 +#define LM49453_P0_ADC_ALC3_REG 0x84 +#define LM49453_P0_ADC_ALC4_REG 0x85 +#define LM49453_P0_ADC_ALC5_REG 0x86 +#define LM49453_P0_ADC_ALC6_REG 0x87 +#define LM49453_P0_ADC_ALC7_REG 0x88 +#define LM49453_P0_ADC_ALC8_REG 0x89 +#define LM49453_P0_DMIC1_LEVELL_REG 0x8A +#define LM49453_P0_DMIC1_LEVELR_REG 0x8B +#define LM49453_P0_DMIC2_LEVELL_REG 0x8C +#define LM49453_P0_DMIC2_LEVELR_REG 0x8D +#define LM49453_P0_ADC_LEVELL_REG 0x8E +#define LM49453_P0_ADC_LEVELR_REG 0x8F +#define LM49453_P0_DAC_HP_LEVELL_REG 0x90 +#define LM49453_P0_DAC_HP_LEVELR_REG 0x91 +#define LM49453_P0_DAC_LO_LEVELL_REG 0x92 +#define LM49453_P0_DAC_LO_LEVELR_REG 0x93 +#define LM49453_P0_DAC_LS_LEVELL_REG 0x94 +#define LM49453_P0_DAC_LS_LEVELR_REG 0x95 +#define LM49453_P0_DAC_HA_LEVELL_REG 0x96 +#define LM49453_P0_DAC_HA_LEVELR_REG 0x97 +#define LM49453_P0_SOFT_MUTE_REG 0x98 +#define LM49453_P0_DMIC_MUTE_CFG_REG 0x99 +#define LM49453_P0_ADC_MUTE_CFG_REG 0x9A +#define LM49453_P0_DAC_MUTE_CFG_REG 0x9B + +/*DIGITAL MIC1 */ +#define LM49453_P0_DIGITAL_MIC1_CONFIG_REG 0xB0 +#define LM49453_P0_DIGITAL_MIC1_DATA_DELAYL_REG 0xB1 +#define LM49453_P0_DIGITAL_MIC1_DATA_DELAYR_REG 0xB2 + +/*DIGITAL MIC2 */ +#define LM49453_P0_DIGITAL_MIC2_CONFIG_REG 0xB3 +#define LM49453_P0_DIGITAL_MIC2_DATA_DELAYL_REG 0xB4 +#define LM49453_P0_DIGITAL_MIC2_DATA_DELAYR_REG 0xB5 + +/* ADC DECIMATOR */ +#define LM49453_P0_ADC_DECIMATOR_REG 0xB6 + +/* DAC CONFIGURE */ +#define LM49453_P0_DAC_CONFIG_REG 0xB7 + +/* SIDETONE */ +#define LM49453_P0_STN_VOL_ADCL_REG 0xB8 +#define LM49453_P0_STN_VOL_ADCR_REG 0xB9 +#define LM49453_P0_STN_VOL_DMIC1L_REG 0xBA +#define LM49453_P0_STN_VOL_DMIC1R_REG 0xBB +#define LM49453_P0_STN_VOL_DMIC2L_REG 0xBC +#define LM49453_P0_STN_VOL_DMIC2R_REG 0xBD + +/* ADC/DAC CLIPPING MONITORS (Read Only/Write to Clear) */ +#define LM49453_P0_ADC_DEC_CLIP_REG 0xC2 +#define LM49453_P0_ADC_HPF_CLIP_REG 0xC3 +#define LM49453_P0_ADC_LVL_CLIP_REG 0xC4 +#define LM49453_P0_DAC_LVL_CLIP_REG 0xC5 + +/* ADC ALC EFFECT MONITORS (Read Only) */ +#define LM49453_P0_ADC_LVLMONL_REG 0xC8 +#define LM49453_P0_ADC_LVLMONR_REG 0xC9 +#define LM49453_P0_ADC_ALCMONL_REG 0xCA +#define LM49453_P0_ADC_ALCMONR_REG 0xCB +#define LM49453_P0_ADC_MUTED_REG 0xCC +#define LM49453_P0_DAC_MUTED_REG 0xCD + +/* HEADSET DETECT */ +#define LM49453_P0_HSD_PPB_LONG_CNT_LIMITL_REG 0xD0 +#define LM49453_P0_HSD_PPB_LONG_CNT_LIMITR_REG 0xD1 +#define LM49453_P0_HSD_PIN3_4_EX_LOOP_CNT_LIMITL_REG 0xD2 +#define LM49453_P0_HSD_PIN3_4_EX_LOOP_CNT_LIMITH_REG 0xD3 +#define LM49453_P0_HSD_TIMEOUT1_REG 0xD4 +#define LM49453_P0_HSD_TIMEOUT2_REG 0xD5 +#define LM49453_P0_HSD_TIMEOUT3_REG 0xD6 +#define LM49453_P0_HSD_PIN3_4_CFG_REG 0xD7 +#define LM49453_P0_HSD_IRQ1_REG 0xD8 +#define LM49453_P0_HSD_IRQ2_REG 0xD9 +#define LM49453_P0_HSD_IRQ3_REG 0xDA +#define LM49453_P0_HSD_IRQ4_REG 0xDB +#define LM49453_P0_HSD_IRQ_MASK1_REG 0xDC +#define LM49453_P0_HSD_IRQ_MASK2_REG 0xDD +#define LM49453_P0_HSD_IRQ_MASK3_REG 0xDE +#define LM49453_P0_HSD_R_HPLL_REG 0xE0 +#define LM49453_P0_HSD_R_HPLH_REG 0xE1 +#define LM49453_P0_HSD_R_HPLU_REG 0xE2 +#define LM49453_P0_HSD_R_HPRL_REG 0xE3 +#define LM49453_P0_HSD_R_HPRH_REG 0xE4 +#define LM49453_P0_HSD_R_HPRU_REG 0xE5 +#define LM49453_P0_HSD_VEL_L_FINALL_REG 0xE6 +#define LM49453_P0_HSD_VEL_L_FINALH_REG 0xE7 +#define LM49453_P0_HSD_VEL_L_FINALU_REG 0xE8 +#define LM49453_P0_HSD_RO_FINALL_REG 0xE9 +#define LM49453_P0_HSD_RO_FINALH_REG 0xEA +#define LM49453_P0_HSD_RO_FINALU_REG 0xEB +#define LM49453_P0_HSD_VMIC_BIAS_FINALL_REG 0xEC +#define LM49453_P0_HSD_VMIC_BIAS_FINALH_REG 0xED +#define LM49453_P0_HSD_VMIC_BIAS_FINALU_REG 0xEE +#define LM49453_P0_HSD_PIN_CONFIG_REG 0xEF +#define LM49453_P0_HSD_PLUG_DETECT_BB_IRQ_STATUS1_REG 0xF1 +#define LM49453_P0_HSD_PLUG_DETECT_BB_IRQ_STATUS2_REG 0xF2 +#define LM49453_P0_HSD_PLUG_DETECT_BB_IRQ_STATUS3_REG 0xF3 +#define LM49453_P0_HSD_PLUG_DETECT_BB_IRQ_STATEL_REG 0xF4 +#define LM49453_P0_HSD_PLUG_DETECT_BB_IRQ_STATEH_REG 0xF5 + +/* I/O PULLDOWN CONFIG */ +#define LM49453_P0_PULL_CONFIG1_REG 0xF8 +#define LM49453_P0_PULL_CONFIG2_REG 0xF9 +#define LM49453_P0_PULL_CONFIG3_REG 0xFA + +/* RESET */ +#define LM49453_P0_RESET_REG 0xFE + +/* PAGE */ +#define LM49453_PAGE_REG 0xFF + +#define LM49453_MAX_REGISTER (0xFF+1) + +/* LM49453_P0_PMC_SETUP_REG (0x00h) */ +#define LM49453_PMC_SETUP_CHIP_EN (BIT(1)|BIT(0)) +#define LM49453_PMC_SETUP_PLL_EN BIT(2) +#define LM49453_PMC_SETUP_PLL_P2_EN BIT(3) +#define LM49453_PMC_SETUP_PLL_FLL BIT(4) +#define LM49453_PMC_SETUP_MCLK_OVER BIT(5) +#define LM49453_PMC_SETUP_RTC_CLK_OVER BIT(6) +#define LM49453_PMC_SETUP_CHIP_ACTIVE BIT(7) + +/* Chip Enable bits */ +#define LM49453_CHIP_EN_SHUTDOWN 0x00 +#define LM49453_CHIP_EN 0x01 +#define LM49453_CHIP_EN_HSD_DETECT 0x02 +#define LM49453_CHIP_EN_INVALID_HSD 0x03 + +/* LM49453_P0_PLL_CLK_SEL1_REG (0x01h) */ +#define LM49453_CLK_SEL1_MCLK_SEL 0x11 +#define LM49453_CLK_SEL1_RTC_SEL 0x11 +#define LM49453_CLK_SEL1_PORT1_SEL 0x10 +#define LM49453_CLK_SEL1_PORT2_SEL 0x11 + +/* LM49453_P0_PLL_CLK_SEL2_REG (0x02h) */ +#define LM49453_CLK_SEL2_ADC_CLK_SEL 0x38 + +/* LM49453_P0_FLL_REF_FREQL_REG (0x0F) */ +#define LM49453_FLL_REF_FREQ_VAL 0x8ca0001 + +/* LM49453_P0_VCO_TARGETLL_REG (0x11) */ +#define LM49453_VCO_TARGET_VAL 0x8ca0001 + +/* LM49453_P0_ADC_DSP_REG (0x30h) */ +#define LM49453_ADC_DSP_ADC_MUTEL BIT(0) +#define LM49453_ADC_DSP_ADC_MUTER BIT(1) +#define LM49453_ADC_DSP_DMIC1_MUTEL BIT(2) +#define LM49453_ADC_DSP_DMIC1_MUTER BIT(3) +#define LM49453_ADC_DSP_DMIC2_MUTEL BIT(4) +#define LM49453_ADC_DSP_DMIC2_MUTER BIT(5) +#define LM49453_ADC_DSP_MUTE_ALL 0x3F + +/* LM49453_P0_DAC_DSP_REG (0x31h) */ +#define LM49453_DAC_DSP_MUTE_ALL 0xFF + +/* LM49453_P0_AUDIO_PORT1_BASIC_REG (0x60h) */ +#define LM49453_AUDIO_PORT1_BASIC_FMT_MASK (BIT(4)|BIT(3)) +#define LM49453_AUDIO_PORT1_BASIC_CLK_MS BIT(3) +#define LM49453_AUDIO_PORT1_BASIC_SYNC_MS BIT(4) + +/* LM49453_P0_RESET_REG (0xFEh) */ +#define LM49453_RESET_REG_RST BIT(0) + +/* Page select register bits (0xFF) */ +#define LM49453_PAGE0_SELECT 0x0 +#define LM49453_PAGE1_SELECT 0x1 + +/* LM49453_P0_HSD_PIN3_4_CFG_REG (Jack Pin config - 0xD7) */ +#define LM49453_JACK_DISABLE 0x00 +#define LM49453_JACK_CONFIG1 0x01 +#define LM49453_JACK_CONFIG2 0x02 +#define LM49453_JACK_CONFIG3 0x03 +#define LM49453_JACK_CONFIG4 0x04 +#define LM49453_JACK_CONFIG5 0x05 + +/* Page 1 REGISTERS */ + +/* SIDETONE */ +#define LM49453_P1_SIDETONE_SA0L_REG 0x80 +#define LM49453_P1_SIDETONE_SA0H_REG 0x81 +#define LM49453_P1_SIDETONE_SAB0U_REG 0x82 +#define LM49453_P1_SIDETONE_SB0L_REG 0x83 +#define LM49453_P1_SIDETONE_SB0H_REG 0x84 +#define LM49453_P1_SIDETONE_SH0L_REG 0x85 +#define LM49453_P1_SIDETONE_SH0H_REG 0x86 +#define LM49453_P1_SIDETONE_SH0U_REG 0x87 +#define LM49453_P1_SIDETONE_SA1L_REG 0x88 +#define LM49453_P1_SIDETONE_SA1H_REG 0x89 +#define LM49453_P1_SIDETONE_SAB1U_REG 0x8A +#define LM49453_P1_SIDETONE_SB1L_REG 0x8B +#define LM49453_P1_SIDETONE_SB1H_REG 0x8C +#define LM49453_P1_SIDETONE_SH1L_REG 0x8D +#define LM49453_P1_SIDETONE_SH1H_REG 0x8E +#define LM49453_P1_SIDETONE_SH1U_REG 0x8F +#define LM49453_P1_SIDETONE_SA2L_REG 0x90 +#define LM49453_P1_SIDETONE_SA2H_REG 0x91 +#define LM49453_P1_SIDETONE_SAB2U_REG 0x92 +#define LM49453_P1_SIDETONE_SB2L_REG 0x93 +#define LM49453_P1_SIDETONE_SB2H_REG 0x94 +#define LM49453_P1_SIDETONE_SH2L_REG 0x95 +#define LM49453_P1_SIDETONE_SH2H_REG 0x96 +#define LM49453_P1_SIDETONE_SH2U_REG 0x97 +#define LM49453_P1_SIDETONE_SA3L_REG 0x98 +#define LM49453_P1_SIDETONE_SA3H_REG 0x99 +#define LM49453_P1_SIDETONE_SAB3U_REG 0x9A +#define LM49453_P1_SIDETONE_SB3L_REG 0x9B +#define LM49453_P1_SIDETONE_SB3H_REG 0x9C +#define LM49453_P1_SIDETONE_SH3L_REG 0x9D +#define LM49453_P1_SIDETONE_SH3H_REG 0x9E +#define LM49453_P1_SIDETONE_SH3U_REG 0x9F +#define LM49453_P1_SIDETONE_SA4L_REG 0xA0 +#define LM49453_P1_SIDETONE_SA4H_REG 0xA1 +#define LM49453_P1_SIDETONE_SAB4U_REG 0xA2 +#define LM49453_P1_SIDETONE_SB4L_REG 0xA3 +#define LM49453_P1_SIDETONE_SB4H_REG 0xA4 +#define LM49453_P1_SIDETONE_SH4L_REG 0xA5 +#define LM49453_P1_SIDETONE_SH4H_REG 0xA6 +#define LM49453_P1_SIDETONE_SH4U_REG 0xA7 +#define LM49453_P1_SIDETONE_SA5L_REG 0xA8 +#define LM49453_P1_SIDETONE_SA5H_REG 0xA9 +#define LM49453_P1_SIDETONE_SAB5U_REG 0xAA +#define LM49453_P1_SIDETONE_SB5L_REG 0xAB +#define LM49453_P1_SIDETONE_SB5H_REG 0xAC +#define LM49453_P1_SIDETONE_SH5L_REG 0xAD +#define LM49453_P1_SIDETONE_SH5H_REG 0xAE +#define LM49453_P1_SIDETONE_SH5U_REG 0xAF + +/* CHARGE PUMP CONFIG */ +#define LM49453_P1_CP_CONFIG1_REG 0xB0 +#define LM49453_P1_CP_CONFIG2_REG 0xB1 +#define LM49453_P1_CP_CONFIG3_REG 0xB2 +#define LM49453_P1_CP_CONFIG4_REG 0xB3 +#define LM49453_P1_CP_LA_VTH1L_REG 0xB4 +#define LM49453_P1_CP_LA_VTH1M_REG 0xB5 +#define LM49453_P1_CP_LA_VTH2L_REG 0xB6 +#define LM49453_P1_CP_LA_VTH2M_REG 0xB7 +#define LM49453_P1_CP_LA_VTH3L_REG 0xB8 +#define LM49453_P1_CP_LA_VTH3H_REG 0xB9 +#define LM49453_P1_CP_CLK_DIV_REG 0xBA + +/* DAC */ +#define LM49453_P1_DAC_CHOP_REG 0xC0 + +#define LM49453_CLK_SRC_MCLK 1 +#endif -- cgit v1.2.3 From 0841b04a5ffba5bd5a7f5b49f014adaf639bf3a4 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 2 Apr 2012 14:53:13 +0100 Subject: ASoC: max98095: Fix build failure sound/soc/codecs/max98095.c: In function 'max98095_jack_detect_enable': sound/soc/codecs/max98095.c:2229:14: error: 'struct max98095_priv' has no member named 'jack_detect_delay' sound/soc/codecs/max98095.c:2230:18: error: 'struct max98095_priv' has no member named 'jack_detect_delay' Reported-by: Stephen Rothwell Signed-off-by: Mark Brown --- sound/soc/codecs/max98095.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c index 0752840e01c2..35179e2c23c9 100644 --- a/sound/soc/codecs/max98095.c +++ b/sound/soc/codecs/max98095.c @@ -2226,8 +2226,8 @@ int max98095_jack_detect_enable(struct snd_soc_codec *codec) if (max98095->pdata->jack_detect_pin5en) detect_enable |= M98095_PIN5EN; - if (max98095->jack_detect_delay) - slew = max98095->jack_detect_delay; + if (max98095->pdata->jack_detect_delay) + slew = max98095->pdata->jack_detect_delay; ret = snd_soc_write(codec, M98095_08E_JACK_DC_SLEW, slew); if (ret < 0) { -- cgit v1.2.3 From 1d99f2436d0d1c7741d6dfd9d27b5376cdbbca40 Mon Sep 17 00:00:00 2001 From: Brian Austin Date: Fri, 30 Mar 2012 10:43:55 -0500 Subject: ASoC: core: Rework SOC_DOUBLE_R_SX_TLV add SOC_SINGLE_SX_TLV Some codecs namely Cirrus Logic Codecs have a way of wrapping the dB scale around 0dB without 0dB being in the middle. Rework of SOC_DOUBLE_R_SX_TLV to be more consistent with other asoc tlv macros. Add single register macro : SOC_SINGLE_SX_TLV. Use snd_soc_info_volsw for .info Use snd_soc_get_volsw_sx, snd_soc_put_volsw_sx for single and double. kcontrols for CS42L51 and CS42L73 are adjusted to these new TLV Macros. The max value is determined by: (number of steps) +1 for 0dB +max from codec datasheet. Signed-off-by: Brian Austin Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- include/sound/soc.h | 48 +++++++------ sound/soc/codecs/cs42l51.c | 6 +- sound/soc/codecs/cs42l73.c | 26 +++---- sound/soc/soc-core.c | 174 +++++++++++++++++++++------------------------ 4 files changed, 125 insertions(+), 129 deletions(-) (limited to 'sound') diff --git a/include/sound/soc.h b/include/sound/soc.h index 9e238fa2eb17..acb57b834e58 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -55,6 +55,18 @@ .info = snd_soc_info_volsw, .get = snd_soc_get_volsw,\ .put = snd_soc_put_volsw, \ .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) } +#define SOC_SINGLE_SX_TLV(xname, xreg, xshift, xmin, xmax, tlv_array) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ + SNDRV_CTL_ELEM_ACCESS_READWRITE, \ + .tlv.p = (tlv_array),\ + .info = snd_soc_info_volsw, \ + .get = snd_soc_get_volsw_sx,\ + .put = snd_soc_put_volsw_sx, \ + .private_value = (unsigned long)&(struct soc_mixer_control) \ + {.reg = xreg, .rreg = xreg, \ + .shift = xshift, .rshift = xshift, \ + .max = xmax, .min = xmin} } #define SOC_DOUBLE(xname, reg, shift_left, shift_right, max, invert) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\ .info = snd_soc_info_volsw, .get = snd_soc_get_volsw, \ @@ -85,6 +97,18 @@ .get = snd_soc_get_volsw, .put = snd_soc_put_volsw, \ .private_value = SOC_DOUBLE_R_VALUE(reg_left, reg_right, xshift, \ xmax, xinvert) } +#define SOC_DOUBLE_R_SX_TLV(xname, xreg, xrreg, xshift, xmin, xmax, tlv_array) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ + SNDRV_CTL_ELEM_ACCESS_READWRITE, \ + .tlv.p = (tlv_array), \ + .info = snd_soc_info_volsw, \ + .get = snd_soc_get_volsw_sx, \ + .put = snd_soc_put_volsw_sx, \ + .private_value = (unsigned long)&(struct soc_mixer_control) \ + {.reg = xreg, .rreg = xrreg, \ + .shift = xshift, .rshift = xshift, \ + .max = xmax, .min = xmin} } #define SOC_DOUBLE_S8_TLV(xname, xreg, xmin, xmax, tlv_array) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ @@ -171,20 +195,6 @@ .get = xhandler_get, .put = xhandler_put, \ .private_value = (unsigned long)&xenum } -#define SOC_DOUBLE_R_SX_TLV(xname, xreg_left, xreg_right, xshift,\ - xmin, xmax, tlv_array) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ - .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ - SNDRV_CTL_ELEM_ACCESS_READWRITE, \ - .tlv.p = (tlv_array), \ - .info = snd_soc_info_volsw_2r_sx, \ - .get = snd_soc_get_volsw_2r_sx, \ - .put = snd_soc_put_volsw_2r_sx, \ - .private_value = (unsigned long)&(struct soc_mixer_control) \ - {.reg = xreg_left, \ - .rreg = xreg_right, .shift = xshift, \ - .min = xmin, .max = xmax} } - #define SND_SOC_BYTES(xname, xbase, xregs) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .info = snd_soc_bytes_info, .get = snd_soc_bytes_get, \ @@ -418,6 +428,10 @@ int snd_soc_put_volsw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); #define snd_soc_get_volsw_2r snd_soc_get_volsw #define snd_soc_put_volsw_2r snd_soc_put_volsw +int snd_soc_get_volsw_sx(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); +int snd_soc_put_volsw_sx(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); int snd_soc_info_volsw_s8(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo); int snd_soc_get_volsw_s8(struct snd_kcontrol *kcontrol, @@ -426,12 +440,6 @@ int snd_soc_put_volsw_s8(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); int snd_soc_limit_volume(struct snd_soc_codec *codec, const char *name, int max); -int snd_soc_info_volsw_2r_sx(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo); -int snd_soc_get_volsw_2r_sx(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol); -int snd_soc_put_volsw_2r_sx(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol); int snd_soc_bytes_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo); int snd_soc_bytes_get(struct snd_kcontrol *kcontrol, diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index a8bf588e8740..85d60695c210 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -141,15 +141,15 @@ static const struct soc_enum cs42l51_chan_mix = static const struct snd_kcontrol_new cs42l51_snd_controls[] = { SOC_DOUBLE_R_SX_TLV("PCM Playback Volume", CS42L51_PCMA_VOL, CS42L51_PCMB_VOL, - 7, 0xffffff99, 0x18, adc_pcm_tlv), + 6, 0x19, 0x7F, adc_pcm_tlv), SOC_DOUBLE_R("PCM Playback Switch", CS42L51_PCMA_VOL, CS42L51_PCMB_VOL, 7, 1, 1), SOC_DOUBLE_R_SX_TLV("Analog Playback Volume", CS42L51_AOUTA_VOL, CS42L51_AOUTB_VOL, - 8, 0xffffff19, 0x18, aout_tlv), + 0, 0x34, 0xE4, aout_tlv), SOC_DOUBLE_R_SX_TLV("ADC Mixer Volume", CS42L51_ADCA_VOL, CS42L51_ADCB_VOL, - 7, 0xffffff99, 0x18, adc_pcm_tlv), + 6, 0x19, 0x7F, adc_pcm_tlv), SOC_DOUBLE_R("ADC Mixer Switch", CS42L51_ADCA_VOL, CS42L51_ADCB_VOL, 7, 1, 1), SOC_SINGLE("Playback Deemphasis Switch", CS42L51_DAC_CTL, 3, 1, 0), diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index 78979b3e0e95..d39b3e78703b 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -402,37 +402,37 @@ static const struct snd_kcontrol_new ear_amp_ctl = static const struct snd_kcontrol_new cs42l73_snd_controls[] = { SOC_DOUBLE_R_SX_TLV("Headphone Analog Playback Volume", - CS42L73_HPAAVOL, CS42L73_HPBAVOL, 7, - 0xffffffC1, 0x0C, hpaloa_tlv), + CS42L73_HPAAVOL, CS42L73_HPBAVOL, 0, + 0x41, 0x4B, hpaloa_tlv), SOC_DOUBLE_R_SX_TLV("LineOut Analog Playback Volume", CS42L73_LOAAVOL, - CS42L73_LOBAVOL, 7, 0xffffffC1, 0x0C, hpaloa_tlv), + CS42L73_LOBAVOL, 0, 0x41, 0x4B, hpaloa_tlv), SOC_DOUBLE_R_SX_TLV("Input PGA Analog Volume", CS42L73_MICAPREPGAAVOL, - CS42L73_MICBPREPGABVOL, 5, 0xffffff35, - 0x34, micpga_tlv), + CS42L73_MICBPREPGABVOL, 5, 0x34, + 0x24, micpga_tlv), SOC_DOUBLE_R("MIC Preamp Switch", CS42L73_MICAPREPGAAVOL, CS42L73_MICBPREPGABVOL, 6, 1, 1), SOC_DOUBLE_R_SX_TLV("Input Path Digital Volume", CS42L73_IPADVOL, - CS42L73_IPBDVOL, 7, 0xffffffA0, 0xA0, ipd_tlv), + CS42L73_IPBDVOL, 0, 0xA0, 0x6C, ipd_tlv), SOC_DOUBLE_R_SX_TLV("HL Digital Playback Volume", - CS42L73_HLADVOL, CS42L73_HLBDVOL, 7, 0xffffffE5, - 0xE4, hl_tlv), + CS42L73_HLADVOL, CS42L73_HLBDVOL, + 0, 0x34, 0xE4, hl_tlv), SOC_SINGLE_TLV("ADC A Boost Volume", CS42L73_ADCIPC, 2, 0x01, 1, adc_boost_tlv), SOC_SINGLE_TLV("ADC B Boost Volume", - CS42L73_ADCIPC, 6, 0x01, 1, adc_boost_tlv), + CS42L73_ADCIPC, 6, 0x01, 1, adc_boost_tlv), - SOC_SINGLE_TLV("Speakerphone Digital Playback Volume", - CS42L73_SPKDVOL, 0, 0xE4, 1, hl_tlv), + SOC_SINGLE_SX_TLV("Speakerphone Digital Volume", + CS42L73_SPKDVOL, 0, 0x34, 0xE4, hl_tlv), - SOC_SINGLE_TLV("Ear Speaker Digital Playback Volume", - CS42L73_ESLDVOL, 0, 0xE4, 1, hl_tlv), + SOC_SINGLE_SX_TLV("Ear Speaker Digital Volume", + CS42L73_ESLDVOL, 0, 0x34, 0xE4, hl_tlv), SOC_DOUBLE_R("Headphone Analog Playback Switch", CS42L73_HPAAVOL, CS42L73_HPBAVOL, 7, 1, 1), diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index cab72f87c194..7b1a4fd932d7 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2527,6 +2527,87 @@ int snd_soc_put_volsw(struct snd_kcontrol *kcontrol, } EXPORT_SYMBOL_GPL(snd_soc_put_volsw); +/** + * snd_soc_get_volsw_sx - single mixer get callback + * @kcontrol: mixer control + * @ucontrol: control element information + * + * Callback to get the value of a single mixer control, or a double mixer + * control that spans 2 registers. + * + * Returns 0 for success. + */ +int snd_soc_get_volsw_sx(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + + unsigned int reg = mc->reg; + unsigned int reg2 = mc->rreg; + unsigned int shift = mc->shift; + unsigned int rshift = mc->rshift; + int max = mc->max; + int min = mc->min; + int mask = (1 << (fls(min + max) - 1)) - 1; + + ucontrol->value.integer.value[0] = + ((snd_soc_read(codec, reg) >> shift) - min) & mask; + + if (snd_soc_volsw_is_stereo(mc)) + ucontrol->value.integer.value[1] = + ((snd_soc_read(codec, reg2) >> rshift) - min) & mask; + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_get_volsw_sx); + +/** + * snd_soc_put_volsw_sx - double mixer set callback + * @kcontrol: mixer control + * @uinfo: control element information + * + * Callback to set the value of a double mixer control that spans 2 registers. + * + * Returns 0 for success. + */ +int snd_soc_put_volsw_sx(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + + unsigned int reg = mc->reg; + unsigned int reg2 = mc->rreg; + unsigned int shift = mc->shift; + unsigned int rshift = mc->rshift; + int max = mc->max; + int min = mc->min; + int mask = (1 << (fls(min + max) - 1)) - 1; + int err; + unsigned short val, val_mask, val2 = 0; + + val_mask = mask << shift; + val = (ucontrol->value.integer.value[0] + min) & mask; + val = val << shift; + + if (snd_soc_update_bits_locked(codec, reg, val_mask, val)) + return err; + + if (snd_soc_volsw_is_stereo(mc)) { + val_mask = mask << rshift; + val2 = (ucontrol->value.integer.value[1] + min) & mask; + val2 = val2 << rshift; + + if (snd_soc_update_bits_locked(codec, reg2, val_mask, val2)) + return err; + } + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_put_volsw_sx); + /** * snd_soc_info_volsw_s8 - signed mixer info callback * @kcontrol: mixer control @@ -2648,99 +2729,6 @@ int snd_soc_limit_volume(struct snd_soc_codec *codec, } EXPORT_SYMBOL_GPL(snd_soc_limit_volume); -/** - * snd_soc_info_volsw_2r_sx - double with tlv and variable data size - * mixer info callback - * @kcontrol: mixer control - * @uinfo: control element information - * - * Returns 0 for success. - */ -int snd_soc_info_volsw_2r_sx(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - int max = mc->max; - int min = mc->min; - - uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; - uinfo->count = 2; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = max-min; - - return 0; -} -EXPORT_SYMBOL_GPL(snd_soc_info_volsw_2r_sx); - -/** - * snd_soc_get_volsw_2r_sx - double with tlv and variable data size - * mixer get callback - * @kcontrol: mixer control - * @uinfo: control element information - * - * Returns 0 for success. - */ -int snd_soc_get_volsw_2r_sx(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - unsigned int mask = (1<shift)-1; - int min = mc->min; - int val = snd_soc_read(codec, mc->reg) & mask; - int valr = snd_soc_read(codec, mc->rreg) & mask; - - ucontrol->value.integer.value[0] = ((val & 0xff)-min) & mask; - ucontrol->value.integer.value[1] = ((valr & 0xff)-min) & mask; - return 0; -} -EXPORT_SYMBOL_GPL(snd_soc_get_volsw_2r_sx); - -/** - * snd_soc_put_volsw_2r_sx - double with tlv and variable data size - * mixer put callback - * @kcontrol: mixer control - * @uinfo: control element information - * - * Returns 0 for success. - */ -int snd_soc_put_volsw_2r_sx(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - unsigned int mask = (1<shift)-1; - int min = mc->min; - int ret; - unsigned int val, valr, oval, ovalr; - - val = ((ucontrol->value.integer.value[0]+min) & 0xff); - val &= mask; - valr = ((ucontrol->value.integer.value[1]+min) & 0xff); - valr &= mask; - - oval = snd_soc_read(codec, mc->reg) & mask; - ovalr = snd_soc_read(codec, mc->rreg) & mask; - - ret = 0; - if (oval != val) { - ret = snd_soc_write(codec, mc->reg, val); - if (ret < 0) - return ret; - } - if (ovalr != valr) { - ret = snd_soc_write(codec, mc->rreg, valr); - if (ret < 0) - return ret; - } - - return 0; -} -EXPORT_SYMBOL_GPL(snd_soc_put_volsw_2r_sx); - int snd_soc_bytes_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { -- cgit v1.2.3 From 27f1d75921e7273e484cc2fab132d9fcbe9f845d Mon Sep 17 00:00:00 2001 From: Brian Austin Date: Tue, 3 Apr 2012 11:33:50 -0500 Subject: ASoC: core: Initialize err for snd_soc_put_volsw_sx MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit sound/soc/soc-core.c: In function ‘snd_soc_put_volsw_sx’: sound/soc/soc-core.c:2600: warning: ‘err’ may be used uninitialized in this function Signed-off-by: Brian Austin Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 7b1a4fd932d7..a6922bc96184 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2586,7 +2586,7 @@ int snd_soc_put_volsw_sx(struct snd_kcontrol *kcontrol, int max = mc->max; int min = mc->min; int mask = (1 << (fls(min + max) - 1)) - 1; - int err; + int err = 0; unsigned short val, val_mask, val2 = 0; val_mask = mask << shift; -- cgit v1.2.3 From cd041f642c706fdda679877cdabf3dc8a6a8e58f Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Tue, 3 Apr 2012 18:05:20 -0300 Subject: ASoC: sgtl5000: Fix warning due to the lack of REGULATOR_CHANGE_VOLTAGE Fix the following warning during kernel boot: 0-000a: 850 <--> 1600 mV at 1200 mV normal 0-000a: Voltage range but no REGULATOR_CHANGE_VOLTAGE Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/codecs/sgtl5000.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 77beb6d7d269..977be8c456ff 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -84,8 +84,8 @@ static struct regulator_consumer_supply ldo_consumer[] = { static struct regulator_init_data ldo_init_data = { .constraints = { - .min_uV = 850000, - .max_uV = 1600000, + .min_uV = 1200000, + .max_uV = 1200000, .valid_modes_mask = REGULATOR_MODE_NORMAL, .valid_ops_mask = REGULATOR_CHANGE_STATUS, }, -- cgit v1.2.3 From 67d45090e6154d401e50c3e0f4a2844cfea404c4 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 3 Apr 2012 22:35:18 +0100 Subject: ASoC: sgtl5000: Convert to module_i2c_driver() Signed-off-by: Mark Brown --- sound/soc/codecs/sgtl5000.c | 12 +----------- 1 file changed, 1 insertion(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 977be8c456ff..84077aae7c4a 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1450,17 +1450,7 @@ static struct i2c_driver sgtl5000_i2c_driver = { .id_table = sgtl5000_id, }; -static int __init sgtl5000_modinit(void) -{ - return i2c_add_driver(&sgtl5000_i2c_driver); -} -module_init(sgtl5000_modinit); - -static void __exit sgtl5000_exit(void) -{ - i2c_del_driver(&sgtl5000_i2c_driver); -} -module_exit(sgtl5000_exit); +module_i2c_driver(sgtl5000_i2c_driver); MODULE_DESCRIPTION("Freescale SGTL5000 ALSA SoC Codec Driver"); MODULE_AUTHOR("Zeng Zhaoming "); -- cgit v1.2.3 From 149c53b514d0a42abbb2c9611ffc9fa2d94857e8 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 3 Mar 2012 00:10:02 +0000 Subject: ASoC: wm8994: Don't bother updating the jackdet mode needlessly If we're not doing jackdet it's not needed. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index c9af13f14331..44f72dc27711 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -686,6 +686,9 @@ static void wm1811_jackdet_set_mode(struct snd_soc_codec *codec, u16 mode) { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + if (!wm8994->jackdet || !wm8994->jack_cb) + return; + if (!wm8994->jackdet || !wm8994->jack_cb) return; -- cgit v1.2.3 From 20731c759a7c6877d1fe619c81b8ab9caebc8f96 Mon Sep 17 00:00:00 2001 From: Shawn Guo Date: Fri, 30 Mar 2012 12:31:31 +0800 Subject: ASoC: fsl: remove redundant Kconfig option SND_SOC_FSL_SSI While commit 606d620 (ASoC: imx: merge sound/soc/imx into sound/soc/fsl) adds SND_SOC_FSL_SSI outside "menuconfig SND_POWERPC_SOC" to make it visible for both PowerPC and ARM/IMX, it forgot removing the one inside "menuconfig SND_POWERPC_SOC". Signed-off-by: Shawn Guo Signed-off-by: Mark Brown --- sound/soc/fsl/Kconfig | 4 ---- 1 file changed, 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index aa14fc9744cc..d23d612381e8 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -16,10 +16,6 @@ if SND_POWERPC_SOC config SND_MPC52xx_DMA tristate -config SND_SOC_FSL_SSI - tristate - depends on FSL_SOC - config SND_SOC_POWERPC_DMA tristate depends on FSL_SOC -- cgit v1.2.3 From 5c7b4a08b7f3ce49756b79386dcdad5a2a21ccb6 Mon Sep 17 00:00:00 2001 From: Shawn Guo Date: Fri, 30 Mar 2012 12:31:32 +0800 Subject: ASoC: fsl: remove redundant Kconfig dependency on SND_SOC_POWERPC_DMA Kconfig option SND_SOC_POWERPC_DMA is under menuconfig SND_POWERPC_SOC. Since SND_POWERPC_SOC already depends on FSL_SOC, there is no need for SND_SOC_POWERPC_DMA to do the same. Signed-off-by: Shawn Guo Signed-off-by: Mark Brown --- sound/soc/fsl/Kconfig | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index d23d612381e8..0094789c43ff 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -18,7 +18,6 @@ config SND_MPC52xx_DMA config SND_SOC_POWERPC_DMA tristate - depends on FSL_SOC config SND_SOC_MPC8610_HPCD tristate "ALSA SoC support for the Freescale MPC8610 HPCD board" -- cgit v1.2.3 From fc9a30e85e4a9df7e692eda45b8484fc028238f0 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 4 Apr 2012 15:33:45 +0100 Subject: ASoC: tlv320aic23: Remove driver-specific version number It's never been updated since the driver was merged. Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic23.c | 4 ---- 1 file changed, 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index 16d55f91a653..b0cafd32c684 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -34,8 +34,6 @@ #include "tlv320aic23.h" -#define AIC23_VERSION "0.1" - /* * AIC23 register cache */ @@ -548,8 +546,6 @@ static int tlv320aic23_probe(struct snd_soc_codec *codec) struct aic23 *aic23 = snd_soc_codec_get_drvdata(codec); int ret; - printk(KERN_INFO "AIC23 Audio Codec %s\n", AIC23_VERSION); - ret = snd_soc_codec_set_cache_io(codec, 7, 9, aic23->control_type); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); -- cgit v1.2.3 From e6968a1719a88afa4708ff43696d6615f0be90be Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 4 Apr 2012 15:58:16 +0100 Subject: ASoC: codecs: Remove rtd->codec usage from CODEC drivers In order to support CODEC<->CODEC links remove the assumption that there is only a single CODEC on a DAI link by removing the use of the CODEC pointer in the rtd from the CODEC drivers. They are already being passed their DAI whenever they are passed an rtd and can get the CODEC from there. Signed-off-by: Mark Brown --- sound/soc/codecs/ac97.c | 4 +--- sound/soc/codecs/ad1836.c | 4 +--- sound/soc/codecs/ad193x.c | 4 +--- sound/soc/codecs/adau1701.c | 3 +-- sound/soc/codecs/ak4104.c | 3 +-- sound/soc/codecs/ak4535.c | 3 +-- sound/soc/codecs/ak4641.c | 3 +-- sound/soc/codecs/alc5623.c | 3 +-- sound/soc/codecs/alc5632.c | 3 +-- sound/soc/codecs/cs4270.c | 3 +-- sound/soc/codecs/cs4271.c | 3 +-- sound/soc/codecs/cs42l51.c | 3 +-- sound/soc/codecs/cs42l73.c | 3 +-- sound/soc/codecs/da7210.c | 3 +-- sound/soc/codecs/jz4740.c | 3 +-- sound/soc/codecs/lm49453.c | 3 +-- sound/soc/codecs/rt5631.c | 3 +-- sound/soc/codecs/sgtl5000.c | 3 +-- sound/soc/codecs/ssm2602.c | 9 +++------ sound/soc/codecs/sta32x.c | 3 +-- sound/soc/codecs/tlv320aic23.c | 9 +++------ sound/soc/codecs/tlv320aic26.c | 3 +-- sound/soc/codecs/tlv320aic3x.c | 3 +-- sound/soc/codecs/tlv320dac33.c | 35 +++++++++++++++-------------------- sound/soc/codecs/twl4030.c | 18 ++++++------------ sound/soc/codecs/twl6040.c | 9 +++------ sound/soc/codecs/uda134x.c | 6 ++---- sound/soc/codecs/uda1380.c | 6 ++---- sound/soc/codecs/wl1273.c | 6 ++---- sound/soc/codecs/wm8400.c | 3 +-- sound/soc/codecs/wm8510.c | 3 +-- sound/soc/codecs/wm8523.c | 3 +-- sound/soc/codecs/wm8728.c | 3 +-- sound/soc/codecs/wm8737.c | 3 +-- sound/soc/codecs/wm8741.c | 3 +-- sound/soc/codecs/wm8750.c | 3 +-- sound/soc/codecs/wm8753.c | 6 ++---- sound/soc/codecs/wm8900.c | 3 +-- sound/soc/codecs/wm8903.c | 3 +-- sound/soc/codecs/wm8940.c | 3 +-- sound/soc/codecs/wm8960.c | 3 +-- sound/soc/codecs/wm8962.c | 3 +-- sound/soc/codecs/wm8971.c | 3 +-- sound/soc/codecs/wm8978.c | 3 +-- sound/soc/codecs/wm8988.c | 3 +-- sound/soc/codecs/wm8990.c | 3 +-- sound/soc/codecs/wm9705.c | 4 +--- sound/soc/codecs/wm9712.c | 8 ++------ 48 files changed, 78 insertions(+), 152 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c index 1bbad4c16d28..a99a1b304e1c 100644 --- a/sound/soc/codecs/ac97.c +++ b/sound/soc/codecs/ac97.c @@ -26,9 +26,7 @@ static int ac97_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; int reg = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? AC97_PCM_FRONT_DAC_RATE : AC97_PCM_LR_ADC_RATE; diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c index 12e3b4118557..c67b50d8b317 100644 --- a/sound/soc/codecs/ad1836.c +++ b/sound/soc/codecs/ad1836.c @@ -162,9 +162,7 @@ static int ad1836_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { int word_len = 0; - - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; /* bit size */ switch (params_format(params)) { diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c index a4a6bef2c0bb..13e62be4f990 100644 --- a/sound/soc/codecs/ad193x.c +++ b/sound/soc/codecs/ad193x.c @@ -245,9 +245,7 @@ static int ad193x_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { int word_len = 0, master_rate = 0; - - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct ad193x_priv *ad193x = snd_soc_codec_get_drvdata(codec); /* bit size */ diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c index 78e9ce48bb99..3d50fc8646b6 100644 --- a/sound/soc/codecs/adau1701.c +++ b/sound/soc/codecs/adau1701.c @@ -258,8 +258,7 @@ static int adau1701_set_playback_pcm_format(struct snd_soc_codec *codec, static int adau1701_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; snd_pcm_format_t format; unsigned int val; diff --git a/sound/soc/codecs/ak4104.c b/sound/soc/codecs/ak4104.c index ceb96ecf5588..31d4483245d0 100644 --- a/sound/soc/codecs/ak4104.c +++ b/sound/soc/codecs/ak4104.c @@ -88,8 +88,7 @@ static int ak4104_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; int val = 0; /* set the IEC958 bits: consumer mode, no copyright bit */ diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c index 838ae8b22b50..618fdc30f73e 100644 --- a/sound/soc/codecs/ak4535.c +++ b/sound/soc/codecs/ak4535.c @@ -262,8 +262,7 @@ static int ak4535_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct ak4535_priv *ak4535 = snd_soc_codec_get_drvdata(codec); u8 mode2 = snd_soc_read(codec, AK4535_MODE2) & ~(0x3 << 5); int rate = params_rate(params), fs = 256; diff --git a/sound/soc/codecs/ak4641.c b/sound/soc/codecs/ak4641.c index 7f42d4a673e3..543a12f471be 100644 --- a/sound/soc/codecs/ak4641.c +++ b/sound/soc/codecs/ak4641.c @@ -296,8 +296,7 @@ static int ak4641_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct ak4641_priv *ak4641 = snd_soc_codec_get_drvdata(codec); int rate = params_rate(params), fs = 256; u8 mode2; diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c index d47b62ddb210..3061d35e9f5c 100644 --- a/sound/soc/codecs/alc5623.c +++ b/sound/soc/codecs/alc5623.c @@ -705,8 +705,7 @@ static int alc5623_set_dai_fmt(struct snd_soc_dai *codec_dai, static int alc5623_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec); int coeff, rate; u16 iface; diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c index e2111e0ccad7..93a5909fcc49 100644 --- a/sound/soc/codecs/alc5632.c +++ b/sound/soc/codecs/alc5632.c @@ -861,8 +861,7 @@ static int alc5632_set_dai_fmt(struct snd_soc_dai *codec_dai, static int alc5632_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; int coeff, rate; u16 iface; diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index df8fe387a66f..047917f0b8ae 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -307,8 +307,7 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct cs4270_private *cs4270 = snd_soc_codec_get_drvdata(codec); int ret; unsigned int i; diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index bf7141280a74..9eb01d7d58a3 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -318,8 +318,7 @@ static int cs4271_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); int i, ret; unsigned int ratio, val; diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index 85d60695c210..091d0193f507 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -356,8 +356,7 @@ static int cs42l51_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct cs42l51_private *cs42l51 = snd_soc_codec_get_drvdata(codec); int ret; unsigned int i; diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index d39b3e78703b..2cc50b379a0e 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -1089,8 +1089,7 @@ static int cs42l73_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct cs42l73_private *priv = snd_soc_codec_get_drvdata(codec); int id = dai->id; int mclk_coeff; diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 8e45aa9cb7b7..5dfdf6e7a39a 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -716,8 +716,7 @@ static int da7210_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; u32 dai_cfg1; u32 fs, bypass; diff --git a/sound/soc/codecs/jz4740.c b/sound/soc/codecs/jz4740.c index 4624e752a188..85d9cabe6d55 100644 --- a/sound/soc/codecs/jz4740.c +++ b/sound/soc/codecs/jz4740.c @@ -164,8 +164,7 @@ static int jz4740_codec_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { uint32_t val; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec =rtd->codec; + struct snd_soc_codec *codec = dai->codec; switch (params_rate(params)) { case 8000: diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c index 744063da40d5..89bbe037f81f 100644 --- a/sound/soc/codecs/lm49453.c +++ b/sound/soc/codecs/lm49453.c @@ -1140,8 +1140,7 @@ static int lm49453_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct lm49453_priv *lm49453 = snd_soc_codec_get_drvdata(codec); u16 clk_div = 0; diff --git a/sound/soc/codecs/rt5631.c b/sound/soc/codecs/rt5631.c index 20c324c7c349..95147e451982 100644 --- a/sound/soc/codecs/rt5631.c +++ b/sound/soc/codecs/rt5631.c @@ -1361,8 +1361,7 @@ static int get_coeff(int mclk, int rate, int timesofbclk) static int rt5631_hifi_pcm_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct rt5631_priv *rt5631 = snd_soc_codec_get_drvdata(codec); int timesofbclk = 32, coeff; unsigned int iface = 0; diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 84077aae7c4a..9538d41c1e5c 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -664,8 +664,7 @@ static int sgtl5000_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); int channels = params_channels(params); int i2s_ctl = 0; diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index de2b20544ceb..4c94fd211fb0 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -254,8 +254,7 @@ static int ssm2602_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec); u16 iface = snd_soc_read(codec, SSM2602_IFACE) & 0xfff3; int srate = ssm2602_get_coeff(ssm2602->sysclk, params_rate(params)); @@ -291,8 +290,7 @@ static int ssm2602_hw_params(struct snd_pcm_substream *substream, static int ssm2602_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec); struct snd_pcm_runtime *master_runtime; @@ -328,8 +326,7 @@ static int ssm2602_startup(struct snd_pcm_substream *substream, static void ssm2602_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec); if (ssm2602->master_substream == substream) diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index 7db6fa515028..8d717f4b5a87 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -609,8 +609,7 @@ static int sta32x_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec); unsigned int rate; int i, mcs = -1, ir = -1; diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index b0cafd32c684..8c758b214a25 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -323,8 +323,7 @@ static int tlv320aic23_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; u16 iface_reg; int ret; struct aic23 *aic23 = snd_soc_codec_get_drvdata(codec); @@ -369,8 +368,7 @@ static int tlv320aic23_hw_params(struct snd_pcm_substream *substream, static int tlv320aic23_pcm_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; /* set active */ snd_soc_write(codec, TLV320AIC23_ACTIVE, 0x0001); @@ -381,8 +379,7 @@ static int tlv320aic23_pcm_prepare(struct snd_pcm_substream *substream, static void tlv320aic23_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct aic23 *aic23 = snd_soc_codec_get_drvdata(codec); /* deactivate */ diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c index 802064b5030d..85944e953578 100644 --- a/sound/soc/codecs/tlv320aic26.c +++ b/sound/soc/codecs/tlv320aic26.c @@ -126,8 +126,7 @@ static int aic26_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct aic26 *aic26 = snd_soc_codec_get_drvdata(codec); int fsref, divisor, wlen, pval, jval, dval, qval; u16 reg; diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 8d20f6ec20f3..bde2d1ade0c6 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -802,8 +802,7 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec =rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec); int codec_clk = 0, bypass_pll = 0, fsref, last_clk = 0; u8 data, j, r, p, pll_q, pll_p = 1, pll_r = 1, pll_j = 1; diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index 4587ddd0fbf8..0dd41077ab79 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -62,8 +62,10 @@ #define UTHR_FROM_PERIOD_SIZE(samples, playrate, burstrate) \ (((samples)*5000) / (((burstrate)*5000) / ((burstrate) - (playrate)))) -static void dac33_calculate_times(struct snd_pcm_substream *substream); -static int dac33_prepare_chip(struct snd_pcm_substream *substream); +static void dac33_calculate_times(struct snd_pcm_substream *substream, + struct snd_soc_codec *codec); +static int dac33_prepare_chip(struct snd_pcm_substream *substream, + struct snd_soc_codec *codec); enum dac33_state { DAC33_IDLE = 0, @@ -427,8 +429,8 @@ static int dac33_playback_event(struct snd_soc_dapm_widget *w, switch (event) { case SND_SOC_DAPM_PRE_PMU: if (likely(dac33->substream)) { - dac33_calculate_times(dac33->substream); - dac33_prepare_chip(dac33->substream); + dac33_calculate_times(dac33->substream, w->codec); + dac33_prepare_chip(dac33->substream, w->codec); } break; case SND_SOC_DAPM_POST_PMD: @@ -799,8 +801,7 @@ static void dac33_oscwait(struct snd_soc_codec *codec) static int dac33_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec); /* Stream started, save the substream pointer */ @@ -812,8 +813,7 @@ static int dac33_startup(struct snd_pcm_substream *substream, static void dac33_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec); dac33->substream = NULL; @@ -825,8 +825,7 @@ static int dac33_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec); /* Check parameters for validity */ @@ -868,10 +867,9 @@ static int dac33_hw_params(struct snd_pcm_substream *substream, * writes happens in different order, than dac33 might end up in unknown state. * Use the known, working sequence of register writes to initialize the dac33. */ -static int dac33_prepare_chip(struct snd_pcm_substream *substream) +static int dac33_prepare_chip(struct snd_pcm_substream *substream, + struct snd_soc_codec *codec) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec); unsigned int oscset, ratioset, pwr_ctrl, reg_tmp; u8 aictrl_a, aictrl_b, fifoctrl_a; @@ -1067,10 +1065,9 @@ static int dac33_prepare_chip(struct snd_pcm_substream *substream) return 0; } -static void dac33_calculate_times(struct snd_pcm_substream *substream) +static void dac33_calculate_times(struct snd_pcm_substream *substream, + struct snd_soc_codec *codec) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec); unsigned int period_size = substream->runtime->period_size; unsigned int rate = substream->runtime->rate; @@ -1128,8 +1125,7 @@ static void dac33_calculate_times(struct snd_pcm_substream *substream) static int dac33_pcm_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec); int ret = 0; @@ -1161,8 +1157,7 @@ static snd_pcm_sframes_t dac33_dai_delay( struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec); unsigned long long t0, t1, t_now; unsigned int time_delta, uthr; diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 170cf9a8fc79..391fcfc7b63b 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -1685,8 +1685,7 @@ static void twl4030_tdm_enable(struct snd_soc_codec *codec, int direction, static int twl4030_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec); if (twl4030->master_substream) { @@ -1715,8 +1714,7 @@ static int twl4030_startup(struct snd_pcm_substream *substream, static void twl4030_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec); if (twl4030->master_substream == substream) @@ -1740,8 +1738,7 @@ static int twl4030_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec); u8 mode, old_mode, format, old_format; @@ -1974,8 +1971,7 @@ static void twl4030_voice_enable(struct snd_soc_codec *codec, int direction, static int twl4030_voice_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec); u8 mode; @@ -2007,8 +2003,7 @@ static int twl4030_voice_startup(struct snd_pcm_substream *substream, static void twl4030_voice_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; /* Enable voice digital filters */ twl4030_voice_enable(codec, substream->stream, 0); @@ -2017,8 +2012,7 @@ static void twl4030_voice_shutdown(struct snd_pcm_substream *substream, static int twl4030_voice_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec); u8 old_mode, mode; diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 2d8c6b825e57..ccbc88aa8497 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -1340,8 +1340,7 @@ static int twl6040_set_bias_level(struct snd_soc_codec *codec, static int twl6040_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); snd_pcm_hw_constraint_list(substream->runtime, 0, @@ -1355,8 +1354,7 @@ static int twl6040_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); int rate; @@ -1392,8 +1390,7 @@ static int twl6040_hw_params(struct snd_pcm_substream *substream, static int twl6040_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct twl6040 *twl6040 = codec->control_data; struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); int ret; diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c index 797b0dde2c68..6c3d43b8ee85 100644 --- a/sound/soc/codecs/uda134x.c +++ b/sound/soc/codecs/uda134x.c @@ -159,8 +159,7 @@ static int uda134x_mute(struct snd_soc_dai *dai, int mute) static int uda134x_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec =rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct uda134x_priv *uda134x = snd_soc_codec_get_drvdata(codec); struct snd_pcm_runtime *master_runtime; @@ -191,8 +190,7 @@ static int uda134x_startup(struct snd_pcm_substream *substream, static void uda134x_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct uda134x_priv *uda134x = snd_soc_codec_get_drvdata(codec); if (uda134x->master_substream == substream) diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c index 4f1b23d7e404..2502214b84ab 100644 --- a/sound/soc/codecs/uda1380.c +++ b/sound/soc/codecs/uda1380.c @@ -502,8 +502,7 @@ static int uda1380_set_dai_fmt_capture(struct snd_soc_dai *codec_dai, static int uda1380_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct uda1380_priv *uda1380 = snd_soc_codec_get_drvdata(codec); int mixer = uda1380_read_reg_cache(codec, UDA1380_MIXER); @@ -528,8 +527,7 @@ static int uda1380_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; u16 clk = uda1380_read_reg_cache(codec, UDA1380_CLK); /* set WSPLL power and divider if running from this clock */ diff --git a/sound/soc/codecs/wl1273.c b/sound/soc/codecs/wl1273.c index 3d868dc40092..7b24d6d192e1 100644 --- a/sound/soc/codecs/wl1273.c +++ b/sound/soc/codecs/wl1273.c @@ -293,8 +293,7 @@ static const struct snd_kcontrol_new wl1273_controls[] = { static int wl1273_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct wl1273_priv *wl1273 = snd_soc_codec_get_drvdata(codec); switch (wl1273->mode) { @@ -329,8 +328,7 @@ static int wl1273_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct wl1273_priv *wl1273 = snd_soc_codec_get_drvdata(rtd->codec); + struct wl1273_priv *wl1273 = snd_soc_codec_get_drvdata(dai->codec); struct wl1273_core *core = wl1273->core; unsigned int rate, width, r; diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c index 898979d23010..fcc0cac9ccd4 100644 --- a/sound/soc/codecs/wm8400.c +++ b/sound/soc/codecs/wm8400.c @@ -1145,8 +1145,7 @@ static int wm8400_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; u16 audio1 = wm8400_read(codec, WM8400_AUDIO_INTERFACE_1); audio1 &= ~WM8400_AIF_WL_MASK; diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index 9166126bd312..56a049555e2c 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -392,8 +392,7 @@ static int wm8510_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; u16 iface = snd_soc_read(codec, WM8510_IFACE) & 0x19f; u16 adn = snd_soc_read(codec, WM8510_ADD) & 0x1f1; diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c index 7fea2c3bf7e7..1c3ffb290cdc 100644 --- a/sound/soc/codecs/wm8523.c +++ b/sound/soc/codecs/wm8523.c @@ -145,8 +145,7 @@ static int wm8523_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct wm8523_priv *wm8523 = snd_soc_codec_get_drvdata(codec); int i; u16 aifctrl1 = snd_soc_read(codec, WM8523_AIF_CTRL1); diff --git a/sound/soc/codecs/wm8728.c b/sound/soc/codecs/wm8728.c index fc3d59e49084..1467f97dce21 100644 --- a/sound/soc/codecs/wm8728.c +++ b/sound/soc/codecs/wm8728.c @@ -88,8 +88,7 @@ static int wm8728_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; u16 dac = snd_soc_read(codec, WM8728_DACCTL); dac &= ~0x18; diff --git a/sound/soc/codecs/wm8737.c b/sound/soc/codecs/wm8737.c index 4fe9d191e277..d0520124616d 100644 --- a/sound/soc/codecs/wm8737.c +++ b/sound/soc/codecs/wm8737.c @@ -329,8 +329,7 @@ static int wm8737_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct wm8737_priv *wm8737 = snd_soc_codec_get_drvdata(codec); int i; u16 clocking = 0; diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c index 3941f50bf187..6e849cb04243 100644 --- a/sound/soc/codecs/wm8741.c +++ b/sound/soc/codecs/wm8741.c @@ -203,8 +203,7 @@ static int wm8741_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct wm8741_priv *wm8741 = snd_soc_codec_get_drvdata(codec); u16 iface = snd_soc_read(codec, WM8741_FORMAT_CONTROL) & 0x1FC; int i; diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index e4c50ce7d9c0..89151ca5e776 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -547,8 +547,7 @@ static int wm8750_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct wm8750_priv *wm8750 = snd_soc_codec_get_drvdata(codec); u16 iface = snd_soc_read(codec, WM8750_IFACE) & 0x1f3; u16 srate = snd_soc_read(codec, WM8750_SRATE) & 0x1c0; diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index e27e7b62b365..a26482cd7654 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -931,8 +931,7 @@ static int wm8753_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct wm8753_priv *wm8753 = snd_soc_codec_get_drvdata(codec); u16 voice = snd_soc_read(codec, WM8753_PCM) & 0x01f3; u16 srate = snd_soc_read(codec, WM8753_SRATE1) & 0x017f; @@ -1161,8 +1160,7 @@ static int wm8753_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct wm8753_priv *wm8753 = snd_soc_codec_get_drvdata(codec); u16 srate = snd_soc_read(codec, WM8753_SRATE1) & 0x01c0; u16 hifi = snd_soc_read(codec, WM8753_HIFI) & 0x01f3; diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index f18c554efc98..077c9628c70d 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -610,8 +610,7 @@ static int wm8900_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; u16 reg; reg = snd_soc_read(codec, WM8900_REG_AUDIO1) & ~0x60; diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index c91fb2f99c13..86b8a2926591 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -1432,8 +1432,7 @@ static int wm8903_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec =rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec); int fs = params_rate(params); int bclk; diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c index d2883affea3b..481a3d9cfe48 100644 --- a/sound/soc/codecs/wm8940.c +++ b/sound/soc/codecs/wm8940.c @@ -371,8 +371,7 @@ static int wm8940_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; u16 iface = snd_soc_read(codec, WM8940_IFACE) & 0xFD9F; u16 addcntrl = snd_soc_read(codec, WM8940_ADDCNTRL) & 0xFFF1; u16 companding = snd_soc_read(codec, diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 840d72086d04..8bc659d8dd2e 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -505,8 +505,7 @@ static int wm8960_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); u16 iface = snd_soc_read(codec, WM8960_IFACE1) & 0xfff3; int i; diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 507f479f4387..0cfce9999c89 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2532,8 +2532,7 @@ static int wm8962_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec); int i; int aif0 = 0; diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c index 28fe59e3ce01..eef783f6b6d6 100644 --- a/sound/soc/codecs/wm8971.c +++ b/sound/soc/codecs/wm8971.c @@ -478,8 +478,7 @@ static int wm8971_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct wm8971_priv *wm8971 = snd_soc_codec_get_drvdata(codec); u16 iface = snd_soc_read(codec, WM8971_IFACE) & 0x1f3; u16 srate = snd_soc_read(codec, WM8971_SRATE) & 0x1c0; diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c index 72d5fdcd3cc2..a5be3adecf75 100644 --- a/sound/soc/codecs/wm8978.c +++ b/sound/soc/codecs/wm8978.c @@ -723,8 +723,7 @@ static int wm8978_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct wm8978_priv *wm8978 = snd_soc_codec_get_drvdata(codec); /* Word length mask = 0x60 */ u16 iface_ctl = snd_soc_read(codec, WM8978_AUDIO_INTERFACE) & ~0x60; diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c index 6cdf6a2bc283..1d4c5cf47b06 100644 --- a/sound/soc/codecs/wm8988.c +++ b/sound/soc/codecs/wm8988.c @@ -668,8 +668,7 @@ static int wm8988_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct wm8988_priv *wm8988 = snd_soc_codec_get_drvdata(codec); u16 iface = snd_soc_read(codec, WM8988_IFACE) & 0x1f3; u16 srate = snd_soc_read(codec, WM8988_SRATE) & 0x180; diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 9d242351e6e8..db63c97ddf51 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -1112,8 +1112,7 @@ static int wm8990_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; u16 audio1 = snd_soc_read(codec, WM8990_AUDIO_INTERFACE_1); audio1 &= ~WM8990_AIF_WL_MASK; diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c index cacc6a86b46f..7c095939c179 100644 --- a/sound/soc/codecs/wm9705.c +++ b/sound/soc/codecs/wm9705.c @@ -236,9 +236,7 @@ static int ac97_write(struct snd_soc_codec *codec, unsigned int reg, static int ac97_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; int reg; u16 vra; diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index b342ae50bcd6..260386365d58 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -467,9 +467,7 @@ static int ac97_write(struct snd_soc_codec *codec, unsigned int reg, static int ac97_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec =rtd->codec; + struct snd_soc_codec *codec = dai->codec; int reg; u16 vra; @@ -487,9 +485,7 @@ static int ac97_prepare(struct snd_pcm_substream *substream, static int ac97_aux_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; u16 vra, xsle; vra = ac97_read(codec, AC97_EXTENDED_STATUS); -- cgit v1.2.3 From 18dcd3044e4c4b3ab6341c98e8d0e81e0d58d5e3 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Mon, 2 Apr 2012 15:40:27 +0200 Subject: ALSA: hda - Fix internal mic for Lenovo Ideapad U300s The internal mic input is phase inverted on one channel. To avoid people in userspace summing the channels together and get zero result, use a separate mixer control for the inverted channel. BugLink: https://bugs.launchpad.net/bugs/903853 Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 88 +++++++++++++++++++++++++++++++++++------- 1 file changed, 75 insertions(+), 13 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 8c6523bbc797..213fb80c11f5 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -142,6 +142,7 @@ struct conexant_spec { unsigned int asus:1; unsigned int pin_eapd_ctrls:1; unsigned int single_adc_amp:1; + unsigned int fixup_stereo_dmic:1; unsigned int adc_switching:1; @@ -4107,9 +4108,9 @@ static int cx_auto_init(struct hda_codec *codec) static int cx_auto_add_volume_idx(struct hda_codec *codec, const char *basename, const char *dir, int cidx, - hda_nid_t nid, int hda_dir, int amp_idx) + hda_nid_t nid, int hda_dir, int amp_idx, int chs) { - static char name[32]; + static char name[44]; static struct snd_kcontrol_new knew[] = { HDA_CODEC_VOLUME(name, 0, 0, 0), HDA_CODEC_MUTE(name, 0, 0, 0), @@ -4119,7 +4120,7 @@ static int cx_auto_add_volume_idx(struct hda_codec *codec, const char *basename, for (i = 0; i < 2; i++) { struct snd_kcontrol *kctl; - knew[i].private_value = HDA_COMPOSE_AMP_VAL(nid, 3, amp_idx, + knew[i].private_value = HDA_COMPOSE_AMP_VAL(nid, chs, amp_idx, hda_dir); knew[i].subdevice = HDA_SUBDEV_AMP_FLAG; knew[i].index = cidx; @@ -4138,7 +4139,7 @@ static int cx_auto_add_volume_idx(struct hda_codec *codec, const char *basename, } #define cx_auto_add_volume(codec, str, dir, cidx, nid, hda_dir) \ - cx_auto_add_volume_idx(codec, str, dir, cidx, nid, hda_dir, 0) + cx_auto_add_volume_idx(codec, str, dir, cidx, nid, hda_dir, 0, 3) #define cx_auto_add_pb_volume(codec, nid, str, idx) \ cx_auto_add_volume(codec, str, " Playback", idx, nid, HDA_OUTPUT) @@ -4208,6 +4209,36 @@ static int cx_auto_build_output_controls(struct hda_codec *codec) return 0; } +/* Returns zero if this is a normal stereo channel, and non-zero if it should + be split in two independent channels. + dest_label must be at least 44 characters. */ +static int cx_auto_get_rightch_label(struct hda_codec *codec, const char *label, + char *dest_label, int nid) +{ + struct conexant_spec *spec = codec->spec; + int i; + + if (!spec->fixup_stereo_dmic) + return 0; + + for (i = 0; i < AUTO_CFG_MAX_INS; i++) { + int def_conf; + if (spec->autocfg.inputs[i].pin != nid) + continue; + + if (spec->autocfg.inputs[i].type != AUTO_PIN_MIC) + return 0; + def_conf = snd_hda_codec_get_pincfg(codec, nid); + if (snd_hda_get_input_pin_attr(def_conf) != INPUT_PIN_ATTR_INT) + return 0; + + /* Finally found the inverted internal mic! */ + snprintf(dest_label, 44, "Inverted %s", label); + return 1; + } + return 0; +} + static int cx_auto_add_capture_volume(struct hda_codec *codec, hda_nid_t nid, const char *label, const char *pfx, int cidx) @@ -4216,14 +4247,25 @@ static int cx_auto_add_capture_volume(struct hda_codec *codec, hda_nid_t nid, int i; for (i = 0; i < spec->num_adc_nids; i++) { + char rightch_label[44]; hda_nid_t adc_nid = spec->adc_nids[i]; int idx = get_input_connection(codec, adc_nid, nid); if (idx < 0) continue; if (spec->single_adc_amp) idx = 0; + + if (cx_auto_get_rightch_label(codec, label, rightch_label, nid)) { + /* Make two independent kcontrols for left and right */ + int err = cx_auto_add_volume_idx(codec, label, pfx, + cidx, adc_nid, HDA_INPUT, idx, 1); + if (err < 0) + return err; + return cx_auto_add_volume_idx(codec, rightch_label, pfx, + cidx, adc_nid, HDA_INPUT, idx, 2); + } return cx_auto_add_volume_idx(codec, label, pfx, - cidx, adc_nid, HDA_INPUT, idx); + cidx, adc_nid, HDA_INPUT, idx, 3); } return 0; } @@ -4236,9 +4278,19 @@ static int cx_auto_add_boost_volume(struct hda_codec *codec, int idx, int i, con; nid = spec->imux_info[idx].pin; - if (get_wcaps(codec, nid) & AC_WCAP_IN_AMP) + if (get_wcaps(codec, nid) & AC_WCAP_IN_AMP) { + char rightch_label[44]; + if (cx_auto_get_rightch_label(codec, label, rightch_label, nid)) { + int err = cx_auto_add_volume_idx(codec, label, " Boost", + cidx, nid, HDA_INPUT, 0, 1); + if (err < 0) + return err; + return cx_auto_add_volume_idx(codec, rightch_label, " Boost", + cidx, nid, HDA_INPUT, 0, 2); + } return cx_auto_add_volume(codec, label, " Boost", cidx, nid, HDA_INPUT); + } con = __select_input_connection(codec, spec->imux_info[idx].adc, nid, &mux, false, 0); if (con < 0) @@ -4405,22 +4457,30 @@ static void apply_pincfg(struct hda_codec *codec, const struct cxt_pincfg *cfg) } -static void apply_pin_fixup(struct hda_codec *codec, +enum { + CXT_PINCFG_LENOVO_X200, + CXT_FIXUP_STEREO_DMIC, +}; + +static void apply_fixup(struct hda_codec *codec, const struct snd_pci_quirk *quirk, const struct cxt_pincfg **table) { + struct conexant_spec *spec = codec->spec; + quirk = snd_pci_quirk_lookup(codec->bus->pci, quirk); - if (quirk) { + if (quirk && table[quirk->value]) { snd_printdd(KERN_INFO "hda_codec: applying pincfg for %s\n", quirk->name); apply_pincfg(codec, table[quirk->value]); } + if (quirk->value == CXT_FIXUP_STEREO_DMIC) { + snd_printdd(KERN_INFO "hda_codec: applying internal mic workaround for %s\n", + quirk->name); + spec->fixup_stereo_dmic = 1; + } } -enum { - CXT_PINCFG_LENOVO_X200, -}; - static const struct cxt_pincfg cxt_pincfg_lenovo_x200[] = { { 0x16, 0x042140ff }, /* HP (seq# overridden) */ { 0x17, 0x21a11000 }, /* dock-mic */ @@ -4431,10 +4491,12 @@ static const struct cxt_pincfg cxt_pincfg_lenovo_x200[] = { static const struct cxt_pincfg *cxt_pincfg_tbl[] = { [CXT_PINCFG_LENOVO_X200] = cxt_pincfg_lenovo_x200, + [CXT_FIXUP_STEREO_DMIC] = NULL, }; static const struct snd_pci_quirk cxt_fixups[] = { SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo X200", CXT_PINCFG_LENOVO_X200), + SND_PCI_QUIRK(0x17aa, 0x3975, "Lenovo U300s", CXT_FIXUP_STEREO_DMIC), {} }; @@ -4477,7 +4539,7 @@ static int patch_conexant_auto(struct hda_codec *codec) break; } - apply_pin_fixup(codec, cxt_fixups, cxt_pincfg_tbl); + apply_fixup(codec, cxt_fixups, cxt_pincfg_tbl); /* Show mute-led control only on HP laptops * This is a sort of white-list: on HP laptops, EAPD corresponds -- cgit v1.2.3 From 063dd9d4488184f35c4598fb68f46fbba959d58e Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Thu, 5 Apr 2012 13:13:49 -0600 Subject: ASoC: tegra: drop Kconfig description for SND_SOC_TEGRA_DAS The DAS, I2S, and SPDIF Kconfig options are intended to be selected by the Kconfig options for ASoC machine drivers. As such, they don't need to be user-visible themselves. Drop the description from the DAS variable to achieve this. I2S and SPDIF already don't have a description. Signed-off-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/tegra/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig index 23ab00d5b192..16632b1be150 100644 --- a/sound/soc/tegra/Kconfig +++ b/sound/soc/tegra/Kconfig @@ -5,7 +5,7 @@ config SND_SOC_TEGRA Say Y or M here if you want support for SoC audio on Tegra. config SND_SOC_TEGRA_DAS - tristate "Tegra Digital Audio Switch driver" + tristate depends on SND_SOC_TEGRA help Say Y or M if you want to add support for the Tegra DAS module. -- cgit v1.2.3 From a7fda2ba82b7531158c68f700fa806e645ff3b7c Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Thu, 5 Apr 2012 13:14:52 -0600 Subject: ASoC: tegra: make Tegra20 drivers depend on Tegra20 support Without this, the Tegra20 drivers can be built into a kernel that's built only for Tegra30. Signed-off-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/tegra/Kconfig | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig index 16632b1be150..d51945e78017 100644 --- a/sound/soc/tegra/Kconfig +++ b/sound/soc/tegra/Kconfig @@ -6,7 +6,7 @@ config SND_SOC_TEGRA config SND_SOC_TEGRA_DAS tristate - depends on SND_SOC_TEGRA + depends on SND_SOC_TEGRA && ARCH_TEGRA_2x_SOC help Say Y or M if you want to add support for the Tegra DAS module. You will also need to select the individual machine drivers to @@ -14,7 +14,7 @@ config SND_SOC_TEGRA_DAS config SND_SOC_TEGRA_I2S tristate - depends on SND_SOC_TEGRA + depends on SND_SOC_TEGRA && ARCH_TEGRA_2x_SOC select SND_SOC_TEGRA_DAS help Say Y or M if you want to add support for codecs attached to the @@ -23,7 +23,7 @@ config SND_SOC_TEGRA_I2S config SND_SOC_TEGRA_SPDIF tristate - depends on SND_SOC_TEGRA + depends on SND_SOC_TEGRA && ARCH_TEGRA_2x_SOC default m help Say Y or M if you want to add support for the SPDIF interface. @@ -41,7 +41,7 @@ config SND_SOC_TEGRA_WM8903 tristate "SoC Audio support for Tegra boards using a WM8903 codec" depends on SND_SOC_TEGRA && I2C depends on MACH_HAS_SND_SOC_TEGRA_WM8903 - select SND_SOC_TEGRA_I2S + select SND_SOC_TEGRA_I2S if ARCH_TEGRA_2x_SOC select SND_SOC_WM8903 help Say Y or M here if you want to add support for SoC audio on Tegra @@ -51,7 +51,7 @@ config SND_SOC_TEGRA_WM8903 config SND_SOC_TEGRA_TRIMSLICE tristate "SoC Audio support for TrimSlice board" depends on SND_SOC_TEGRA && MACH_TRIMSLICE && I2C - select SND_SOC_TEGRA_I2S + select SND_SOC_TEGRA_I2S if ARCH_TEGRA_2x_SOC select SND_SOC_TLV320AIC23 help Say Y or M here if you want to add support for SoC audio on the @@ -60,7 +60,7 @@ config SND_SOC_TEGRA_TRIMSLICE config SND_SOC_TEGRA_ALC5632 tristate "SoC Audio support for Tegra boards using an ALC5632 codec" depends on SND_SOC_TEGRA && I2C - select SND_SOC_TEGRA_I2S + select SND_SOC_TEGRA_I2S if ARCH_TEGRA_2x_SOC select SND_SOC_ALC5632 help Say Y or M here if you want to add support for SoC audio on the -- cgit v1.2.3 From 7e811ae74b12392fd2f6d6dc9bdee020814e2a0e Mon Sep 17 00:00:00 2001 From: M R Swami Reddy Date: Thu, 5 Apr 2012 20:54:09 +0530 Subject: ASoC: lm49453: fix build warnings sound/soc/codecs/lm49453.c: In function 'lm49453_set_dai_fmt': sound/soc/codecs/lm49453.c:1189:4: warning: overflow in implicit constant conversion [-Woverflow] sound/soc/codecs/lm49453.c:1193:4: warning: overflow in implicit constant conversion [-Woverflow] sound/soc/codecs/lm49453.c:1197:4: warning: overflow in implicit constant conversion [-Woverflow] Reported-by: Stephen Rothwell Signed-off-by: M R Swami Reddy Signed-off-by: Mark Brown --- sound/soc/codecs/lm49453.c | 13 +++++-------- 1 file changed, 5 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c index 89bbe037f81f..5eb726b3ff1b 100644 --- a/sound/soc/codecs/lm49453.c +++ b/sound/soc/codecs/lm49453.c @@ -1177,27 +1177,24 @@ static int lm49453_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; - int aif_val = 0; + u16 aif_val; int mode = 0; int clk_phase = 0; int clk_shift = 0; switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBS_CFS: - aif_val = ~LM49453_AUDIO_PORT1_BASIC_CLK_MS | - ~LM49453_AUDIO_PORT1_BASIC_SYNC_MS; + aif_val = 0; break; case SND_SOC_DAIFMT_CBS_CFM: - aif_val = ~LM49453_AUDIO_PORT1_BASIC_CLK_MS | - LM49453_AUDIO_PORT1_BASIC_SYNC_MS; + aif_val = LM49453_AUDIO_PORT1_BASIC_SYNC_MS; break; case SND_SOC_DAIFMT_CBM_CFS: - aif_val = LM49453_AUDIO_PORT1_BASIC_CLK_MS | - ~LM49453_AUDIO_PORT1_BASIC_SYNC_MS; + aif_val = LM49453_AUDIO_PORT1_BASIC_CLK_MS; break; case SND_SOC_DAIFMT_CBM_CFM: aif_val = LM49453_AUDIO_PORT1_BASIC_CLK_MS | - LM49453_AUDIO_PORT1_BASIC_SYNC_MS; + LM49453_AUDIO_PORT1_BASIC_SYNC_MS; break; default: return -EINVAL; -- cgit v1.2.3 From 5fa87d34846e347b62bebf40edf51167e7ffb081 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 5 Apr 2012 22:05:18 +0100 Subject: ASoC: wm8400: Use snd_soc_write() and snd_soc_read() Signed-off-by: Mark Brown --- sound/soc/codecs/wm8400.c | 132 +++++++++++++++++++++++----------------------- 1 file changed, 66 insertions(+), 66 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c index fcc0cac9ccd4..5dc31ebcd0e7 100644 --- a/sound/soc/codecs/wm8400.c +++ b/sound/soc/codecs/wm8400.c @@ -138,8 +138,8 @@ static int wm8400_outpga_put_volsw_vu(struct snd_kcontrol *kcontrol, return ret; /* now hit the volume update bits (always bit 8) */ - val = wm8400_read(codec, reg); - return wm8400_write(codec, reg, val | 0x0100); + val = snd_soc_read(codec, reg); + return snd_soc_write(codec, reg, val | 0x0100); } #define WM8400_OUTPGA_SINGLE_R_TLV(xname, reg, shift, max, invert, tlv_array) \ @@ -362,8 +362,8 @@ static int inmixer_event (struct snd_soc_dapm_widget *w, { u16 reg, fakepower; - reg = wm8400_read(w->codec, WM8400_POWER_MANAGEMENT_2); - fakepower = wm8400_read(w->codec, WM8400_INTDRIVBITS); + reg = snd_soc_read(w->codec, WM8400_POWER_MANAGEMENT_2); + fakepower = snd_soc_read(w->codec, WM8400_INTDRIVBITS); if (fakepower & ((1 << WM8400_INMIXL_PWR) | (1 << WM8400_AINLMUX_PWR))) { @@ -378,7 +378,7 @@ static int inmixer_event (struct snd_soc_dapm_widget *w, } else { reg &= ~WM8400_AINR_ENA; } - wm8400_write(w->codec, WM8400_POWER_MANAGEMENT_2, reg); + snd_soc_write(w->codec, WM8400_POWER_MANAGEMENT_2, reg); return 0; } @@ -394,7 +394,7 @@ static int outmixer_event (struct snd_soc_dapm_widget *w, switch (reg_shift) { case WM8400_SPEAKER_MIXER | (WM8400_LDSPK << 8) : - reg = wm8400_read(w->codec, WM8400_OUTPUT_MIXER1); + reg = snd_soc_read(w->codec, WM8400_OUTPUT_MIXER1); if (reg & WM8400_LDLO) { printk(KERN_WARNING "Cannot set as Output Mixer 1 LDLO Set\n"); @@ -402,7 +402,7 @@ static int outmixer_event (struct snd_soc_dapm_widget *w, } break; case WM8400_SPEAKER_MIXER | (WM8400_RDSPK << 8): - reg = wm8400_read(w->codec, WM8400_OUTPUT_MIXER2); + reg = snd_soc_read(w->codec, WM8400_OUTPUT_MIXER2); if (reg & WM8400_RDRO) { printk(KERN_WARNING "Cannot set as Output Mixer 2 RDRO Set\n"); @@ -410,7 +410,7 @@ static int outmixer_event (struct snd_soc_dapm_widget *w, } break; case WM8400_OUTPUT_MIXER1 | (WM8400_LDLO << 8): - reg = wm8400_read(w->codec, WM8400_SPEAKER_MIXER); + reg = snd_soc_read(w->codec, WM8400_SPEAKER_MIXER); if (reg & WM8400_LDSPK) { printk(KERN_WARNING "Cannot set as Speaker Mixer LDSPK Set\n"); @@ -418,7 +418,7 @@ static int outmixer_event (struct snd_soc_dapm_widget *w, } break; case WM8400_OUTPUT_MIXER2 | (WM8400_RDRO << 8): - reg = wm8400_read(w->codec, WM8400_SPEAKER_MIXER); + reg = snd_soc_read(w->codec, WM8400_SPEAKER_MIXER); if (reg & WM8400_RDSPK) { printk(KERN_WARNING "Cannot set as Speaker Mixer RDSPK Set\n"); @@ -1021,13 +1021,13 @@ static int wm8400_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, wm8400->fll_in = freq_in; /* We *must* disable the FLL before any changes */ - reg = wm8400_read(codec, WM8400_POWER_MANAGEMENT_2); + reg = snd_soc_read(codec, WM8400_POWER_MANAGEMENT_2); reg &= ~WM8400_FLL_ENA; - wm8400_write(codec, WM8400_POWER_MANAGEMENT_2, reg); + snd_soc_write(codec, WM8400_POWER_MANAGEMENT_2, reg); - reg = wm8400_read(codec, WM8400_FLL_CONTROL_1); + reg = snd_soc_read(codec, WM8400_FLL_CONTROL_1); reg &= ~WM8400_FLL_OSC_ENA; - wm8400_write(codec, WM8400_FLL_CONTROL_1, reg); + snd_soc_write(codec, WM8400_FLL_CONTROL_1, reg); if (!freq_out) return 0; @@ -1035,15 +1035,15 @@ static int wm8400_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, reg &= ~(WM8400_FLL_REF_FREQ | WM8400_FLL_FRATIO_MASK); reg |= WM8400_FLL_FRAC | factors.fratio; reg |= factors.freq_ref << WM8400_FLL_REF_FREQ_SHIFT; - wm8400_write(codec, WM8400_FLL_CONTROL_1, reg); + snd_soc_write(codec, WM8400_FLL_CONTROL_1, reg); - wm8400_write(codec, WM8400_FLL_CONTROL_2, factors.k); - wm8400_write(codec, WM8400_FLL_CONTROL_3, factors.n); + snd_soc_write(codec, WM8400_FLL_CONTROL_2, factors.k); + snd_soc_write(codec, WM8400_FLL_CONTROL_3, factors.n); - reg = wm8400_read(codec, WM8400_FLL_CONTROL_4); + reg = snd_soc_read(codec, WM8400_FLL_CONTROL_4); reg &= ~WM8400_FLL_OUTDIV_MASK; reg |= factors.outdiv; - wm8400_write(codec, WM8400_FLL_CONTROL_4, reg); + snd_soc_write(codec, WM8400_FLL_CONTROL_4, reg); return 0; } @@ -1057,8 +1057,8 @@ static int wm8400_set_dai_fmt(struct snd_soc_dai *codec_dai, struct snd_soc_codec *codec = codec_dai->codec; u16 audio1, audio3; - audio1 = wm8400_read(codec, WM8400_AUDIO_INTERFACE_1); - audio3 = wm8400_read(codec, WM8400_AUDIO_INTERFACE_3); + audio1 = snd_soc_read(codec, WM8400_AUDIO_INTERFACE_1); + audio3 = snd_soc_read(codec, WM8400_AUDIO_INTERFACE_3); /* set master/slave audio interface */ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { @@ -1099,8 +1099,8 @@ static int wm8400_set_dai_fmt(struct snd_soc_dai *codec_dai, return -EINVAL; } - wm8400_write(codec, WM8400_AUDIO_INTERFACE_1, audio1); - wm8400_write(codec, WM8400_AUDIO_INTERFACE_3, audio3); + snd_soc_write(codec, WM8400_AUDIO_INTERFACE_1, audio1); + snd_soc_write(codec, WM8400_AUDIO_INTERFACE_3, audio3); return 0; } @@ -1112,24 +1112,24 @@ static int wm8400_set_dai_clkdiv(struct snd_soc_dai *codec_dai, switch (div_id) { case WM8400_MCLK_DIV: - reg = wm8400_read(codec, WM8400_CLOCKING_2) & + reg = snd_soc_read(codec, WM8400_CLOCKING_2) & ~WM8400_MCLK_DIV_MASK; - wm8400_write(codec, WM8400_CLOCKING_2, reg | div); + snd_soc_write(codec, WM8400_CLOCKING_2, reg | div); break; case WM8400_DACCLK_DIV: - reg = wm8400_read(codec, WM8400_CLOCKING_2) & + reg = snd_soc_read(codec, WM8400_CLOCKING_2) & ~WM8400_DAC_CLKDIV_MASK; - wm8400_write(codec, WM8400_CLOCKING_2, reg | div); + snd_soc_write(codec, WM8400_CLOCKING_2, reg | div); break; case WM8400_ADCCLK_DIV: - reg = wm8400_read(codec, WM8400_CLOCKING_2) & + reg = snd_soc_read(codec, WM8400_CLOCKING_2) & ~WM8400_ADC_CLKDIV_MASK; - wm8400_write(codec, WM8400_CLOCKING_2, reg | div); + snd_soc_write(codec, WM8400_CLOCKING_2, reg | div); break; case WM8400_BCLK_DIV: - reg = wm8400_read(codec, WM8400_CLOCKING_1) & + reg = snd_soc_read(codec, WM8400_CLOCKING_1) & ~WM8400_BCLK_DIV_MASK; - wm8400_write(codec, WM8400_CLOCKING_1, reg | div); + snd_soc_write(codec, WM8400_CLOCKING_1, reg | div); break; default: return -EINVAL; @@ -1146,7 +1146,7 @@ static int wm8400_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_codec *codec = dai->codec; - u16 audio1 = wm8400_read(codec, WM8400_AUDIO_INTERFACE_1); + u16 audio1 = snd_soc_read(codec, WM8400_AUDIO_INTERFACE_1); audio1 &= ~WM8400_AIF_WL_MASK; /* bit size */ @@ -1164,19 +1164,19 @@ static int wm8400_hw_params(struct snd_pcm_substream *substream, break; } - wm8400_write(codec, WM8400_AUDIO_INTERFACE_1, audio1); + snd_soc_write(codec, WM8400_AUDIO_INTERFACE_1, audio1); return 0; } static int wm8400_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; - u16 val = wm8400_read(codec, WM8400_DAC_CTRL) & ~WM8400_DAC_MUTE; + u16 val = snd_soc_read(codec, WM8400_DAC_CTRL) & ~WM8400_DAC_MUTE; if (mute) - wm8400_write(codec, WM8400_DAC_CTRL, val | WM8400_DAC_MUTE); + snd_soc_write(codec, WM8400_DAC_CTRL, val | WM8400_DAC_MUTE); else - wm8400_write(codec, WM8400_DAC_CTRL, val); + snd_soc_write(codec, WM8400_DAC_CTRL, val); return 0; } @@ -1195,9 +1195,9 @@ static int wm8400_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_PREPARE: /* VMID=2*50k */ - val = wm8400_read(codec, WM8400_POWER_MANAGEMENT_1) & + val = snd_soc_read(codec, WM8400_POWER_MANAGEMENT_1) & ~WM8400_VMID_MODE_MASK; - wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, val | 0x2); + snd_soc_write(codec, WM8400_POWER_MANAGEMENT_1, val | 0x2); break; case SND_SOC_BIAS_STANDBY: @@ -1211,74 +1211,74 @@ static int wm8400_set_bias_level(struct snd_soc_codec *codec, return ret; } - wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, + snd_soc_write(codec, WM8400_POWER_MANAGEMENT_1, WM8400_CODEC_ENA | WM8400_SYSCLK_ENA); /* Enable POBCTRL, SOFT_ST, VMIDTOG and BUFDCOPEN */ - wm8400_write(codec, WM8400_ANTIPOP2, WM8400_SOFTST | + snd_soc_write(codec, WM8400_ANTIPOP2, WM8400_SOFTST | WM8400_BUFDCOPEN | WM8400_POBCTRL); msleep(50); /* Enable VREF & VMID at 2x50k */ - val = wm8400_read(codec, WM8400_POWER_MANAGEMENT_1); + val = snd_soc_read(codec, WM8400_POWER_MANAGEMENT_1); val |= 0x2 | WM8400_VREF_ENA; - wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, val); + snd_soc_write(codec, WM8400_POWER_MANAGEMENT_1, val); /* Enable BUFIOEN */ - wm8400_write(codec, WM8400_ANTIPOP2, WM8400_SOFTST | + snd_soc_write(codec, WM8400_ANTIPOP2, WM8400_SOFTST | WM8400_BUFDCOPEN | WM8400_POBCTRL | WM8400_BUFIOEN); /* disable POBCTRL, SOFT_ST and BUFDCOPEN */ - wm8400_write(codec, WM8400_ANTIPOP2, WM8400_BUFIOEN); + snd_soc_write(codec, WM8400_ANTIPOP2, WM8400_BUFIOEN); } /* VMID=2*300k */ - val = wm8400_read(codec, WM8400_POWER_MANAGEMENT_1) & + val = snd_soc_read(codec, WM8400_POWER_MANAGEMENT_1) & ~WM8400_VMID_MODE_MASK; - wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, val | 0x4); + snd_soc_write(codec, WM8400_POWER_MANAGEMENT_1, val | 0x4); break; case SND_SOC_BIAS_OFF: /* Enable POBCTRL and SOFT_ST */ - wm8400_write(codec, WM8400_ANTIPOP2, WM8400_SOFTST | + snd_soc_write(codec, WM8400_ANTIPOP2, WM8400_SOFTST | WM8400_POBCTRL | WM8400_BUFIOEN); /* Enable POBCTRL, SOFT_ST and BUFDCOPEN */ - wm8400_write(codec, WM8400_ANTIPOP2, WM8400_SOFTST | + snd_soc_write(codec, WM8400_ANTIPOP2, WM8400_SOFTST | WM8400_BUFDCOPEN | WM8400_POBCTRL | WM8400_BUFIOEN); /* mute DAC */ - val = wm8400_read(codec, WM8400_DAC_CTRL); - wm8400_write(codec, WM8400_DAC_CTRL, val | WM8400_DAC_MUTE); + val = snd_soc_read(codec, WM8400_DAC_CTRL); + snd_soc_write(codec, WM8400_DAC_CTRL, val | WM8400_DAC_MUTE); /* Enable any disabled outputs */ - val = wm8400_read(codec, WM8400_POWER_MANAGEMENT_1); + val = snd_soc_read(codec, WM8400_POWER_MANAGEMENT_1); val |= WM8400_SPK_ENA | WM8400_OUT3_ENA | WM8400_OUT4_ENA | WM8400_LOUT_ENA | WM8400_ROUT_ENA; - wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, val); + snd_soc_write(codec, WM8400_POWER_MANAGEMENT_1, val); /* Disable VMID */ val &= ~WM8400_VMID_MODE_MASK; - wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, val); + snd_soc_write(codec, WM8400_POWER_MANAGEMENT_1, val); msleep(300); /* Enable all output discharge bits */ - wm8400_write(codec, WM8400_ANTIPOP1, WM8400_DIS_LLINE | + snd_soc_write(codec, WM8400_ANTIPOP1, WM8400_DIS_LLINE | WM8400_DIS_RLINE | WM8400_DIS_OUT3 | WM8400_DIS_OUT4 | WM8400_DIS_LOUT | WM8400_DIS_ROUT); /* Disable VREF */ val &= ~WM8400_VREF_ENA; - wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, val); + snd_soc_write(codec, WM8400_POWER_MANAGEMENT_1, val); /* disable POBCTRL, SOFT_ST and BUFDCOPEN */ - wm8400_write(codec, WM8400_ANTIPOP2, 0x0); + snd_soc_write(codec, WM8400_ANTIPOP2, 0x0); ret = regulator_bulk_disable(ARRAY_SIZE(power), &power[0]); @@ -1384,19 +1384,19 @@ static int wm8400_codec_probe(struct snd_soc_codec *codec) wm8400_codec_reset(codec); - reg = wm8400_read(codec, WM8400_POWER_MANAGEMENT_1); - wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, reg | WM8400_CODEC_ENA); + reg = snd_soc_read(codec, WM8400_POWER_MANAGEMENT_1); + snd_soc_write(codec, WM8400_POWER_MANAGEMENT_1, reg | WM8400_CODEC_ENA); /* Latch volume update bits */ - reg = wm8400_read(codec, WM8400_LEFT_LINE_INPUT_1_2_VOLUME); - wm8400_write(codec, WM8400_LEFT_LINE_INPUT_1_2_VOLUME, + reg = snd_soc_read(codec, WM8400_LEFT_LINE_INPUT_1_2_VOLUME); + snd_soc_write(codec, WM8400_LEFT_LINE_INPUT_1_2_VOLUME, reg & WM8400_IPVU); - reg = wm8400_read(codec, WM8400_RIGHT_LINE_INPUT_1_2_VOLUME); - wm8400_write(codec, WM8400_RIGHT_LINE_INPUT_1_2_VOLUME, + reg = snd_soc_read(codec, WM8400_RIGHT_LINE_INPUT_1_2_VOLUME); + snd_soc_write(codec, WM8400_RIGHT_LINE_INPUT_1_2_VOLUME, reg & WM8400_IPVU); - wm8400_write(codec, WM8400_LEFT_OUTPUT_VOLUME, 0x50 | (1<<8)); - wm8400_write(codec, WM8400_RIGHT_OUTPUT_VOLUME, 0x50 | (1<<8)); + snd_soc_write(codec, WM8400_LEFT_OUTPUT_VOLUME, 0x50 | (1<<8)); + snd_soc_write(codec, WM8400_RIGHT_OUTPUT_VOLUME, 0x50 | (1<<8)); if (!schedule_work(&priv->work)) { ret = -EINVAL; @@ -1413,8 +1413,8 @@ static int wm8400_codec_remove(struct snd_soc_codec *codec) { u16 reg; - reg = wm8400_read(codec, WM8400_POWER_MANAGEMENT_1); - wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, + reg = snd_soc_read(codec, WM8400_POWER_MANAGEMENT_1); + snd_soc_write(codec, WM8400_POWER_MANAGEMENT_1, reg & (~WM8400_CODEC_ENA)); regulator_bulk_free(ARRAY_SIZE(power), power); @@ -1427,7 +1427,7 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8400 = { .remove = wm8400_codec_remove, .suspend = wm8400_suspend, .resume = wm8400_resume, - .read = wm8400_read, + .read = snd_soc_read, .write = wm8400_write, .set_bias_level = wm8400_set_bias_level, -- cgit v1.2.3 From ef280d3907cea21b6093802398bbe4193e221a64 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Thu, 5 Apr 2012 15:54:53 -0600 Subject: ASoC: tegra: rename Tegra20-specific driver files Rename these files so they include a specific hardware version in their filenames. The contents is only touched minimally so that git's rename tracking operates correctly; renaming all symbols in the files results in a diff so large that the rename detection fails. Signed-off-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/tegra/Makefile | 12 +- sound/soc/tegra/tegra20_das.c | 261 ++++++++++++++++++++++ sound/soc/tegra/tegra20_das.h | 135 ++++++++++++ sound/soc/tegra/tegra20_i2s.c | 458 ++++++++++++++++++++++++++++++++++++++ sound/soc/tegra/tegra20_i2s.h | 165 ++++++++++++++ sound/soc/tegra/tegra20_spdif.c | 365 +++++++++++++++++++++++++++++++ sound/soc/tegra/tegra20_spdif.h | 472 ++++++++++++++++++++++++++++++++++++++++ sound/soc/tegra/tegra_das.c | 261 ---------------------- sound/soc/tegra/tegra_das.h | 135 ------------ sound/soc/tegra/tegra_i2s.c | 458 -------------------------------------- sound/soc/tegra/tegra_i2s.h | 165 -------------- sound/soc/tegra/tegra_spdif.c | 365 ------------------------------- sound/soc/tegra/tegra_spdif.h | 472 ---------------------------------------- 13 files changed, 1862 insertions(+), 1862 deletions(-) create mode 100644 sound/soc/tegra/tegra20_das.c create mode 100644 sound/soc/tegra/tegra20_das.h create mode 100644 sound/soc/tegra/tegra20_i2s.c create mode 100644 sound/soc/tegra/tegra20_i2s.h create mode 100644 sound/soc/tegra/tegra20_spdif.c create mode 100644 sound/soc/tegra/tegra20_spdif.h delete mode 100644 sound/soc/tegra/tegra_das.c delete mode 100644 sound/soc/tegra/tegra_das.h delete mode 100644 sound/soc/tegra/tegra_i2s.c delete mode 100644 sound/soc/tegra/tegra_i2s.h delete mode 100644 sound/soc/tegra/tegra_spdif.c delete mode 100644 sound/soc/tegra/tegra_spdif.h (limited to 'sound') diff --git a/sound/soc/tegra/Makefile b/sound/soc/tegra/Makefile index 714d36070b86..4a33884115f8 100644 --- a/sound/soc/tegra/Makefile +++ b/sound/soc/tegra/Makefile @@ -1,15 +1,15 @@ # Tegra platform Support snd-soc-tegra-pcm-objs := tegra_pcm.o snd-soc-tegra-utils-objs += tegra_asoc_utils.o -snd-soc-tegra-das-objs := tegra_das.o -snd-soc-tegra-i2s-objs := tegra_i2s.o -snd-soc-tegra-spdif-objs := tegra_spdif.o +snd-soc-tegra20-das-objs := tegra20_das.o +snd-soc-tegra20-i2s-objs := tegra20_i2s.o +snd-soc-tegra20-spdif-objs := tegra20_spdif.o obj-$(CONFIG_SND_SOC_TEGRA) += snd-soc-tegra-pcm.o obj-$(CONFIG_SND_SOC_TEGRA) += snd-soc-tegra-utils.o -obj-$(CONFIG_SND_SOC_TEGRA_DAS) += snd-soc-tegra-das.o -obj-$(CONFIG_SND_SOC_TEGRA_I2S) += snd-soc-tegra-i2s.o -obj-$(CONFIG_SND_SOC_TEGRA_SPDIF) += snd-soc-tegra-spdif.o +obj-$(CONFIG_SND_SOC_TEGRA_DAS) += snd-soc-tegra20-das.o +obj-$(CONFIG_SND_SOC_TEGRA_I2S) += snd-soc-tegra20-i2s.o +obj-$(CONFIG_SND_SOC_TEGRA_SPDIF) += snd-soc-tegra20-spdif.o # Tegra machine Support snd-soc-tegra-wm8903-objs := tegra_wm8903.o diff --git a/sound/soc/tegra/tegra20_das.c b/sound/soc/tegra/tegra20_das.c new file mode 100644 index 000000000000..b5e369820cb3 --- /dev/null +++ b/sound/soc/tegra/tegra20_das.c @@ -0,0 +1,261 @@ +/* + * tegra20_das.c - Tegra20 DAS driver + * + * Author: Stephen Warren + * Copyright (C) 2010 - NVIDIA, Inc. + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include "tegra20_das.h" + +#define DRV_NAME "tegra-das" + +static struct tegra_das *das; + +static inline void tegra_das_write(u32 reg, u32 val) +{ + __raw_writel(val, das->regs + reg); +} + +static inline u32 tegra_das_read(u32 reg) +{ + return __raw_readl(das->regs + reg); +} + +int tegra_das_connect_dap_to_dac(int dap, int dac) +{ + u32 addr; + u32 reg; + + if (!das) + return -ENODEV; + + addr = TEGRA_DAS_DAP_CTRL_SEL + + (dap * TEGRA_DAS_DAP_CTRL_SEL_STRIDE); + reg = dac << TEGRA_DAS_DAP_CTRL_SEL_DAP_CTRL_SEL_P; + + tegra_das_write(addr, reg); + + return 0; +} +EXPORT_SYMBOL_GPL(tegra_das_connect_dap_to_dac); + +int tegra_das_connect_dap_to_dap(int dap, int otherdap, int master, + int sdata1rx, int sdata2rx) +{ + u32 addr; + u32 reg; + + if (!das) + return -ENODEV; + + addr = TEGRA_DAS_DAP_CTRL_SEL + + (dap * TEGRA_DAS_DAP_CTRL_SEL_STRIDE); + reg = otherdap << TEGRA_DAS_DAP_CTRL_SEL_DAP_CTRL_SEL_P | + !!sdata2rx << TEGRA_DAS_DAP_CTRL_SEL_DAP_SDATA2_TX_RX_P | + !!sdata1rx << TEGRA_DAS_DAP_CTRL_SEL_DAP_SDATA1_TX_RX_P | + !!master << TEGRA_DAS_DAP_CTRL_SEL_DAP_MS_SEL_P; + + tegra_das_write(addr, reg); + + return 0; +} +EXPORT_SYMBOL_GPL(tegra_das_connect_dap_to_dap); + +int tegra_das_connect_dac_to_dap(int dac, int dap) +{ + u32 addr; + u32 reg; + + if (!das) + return -ENODEV; + + addr = TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL + + (dac * TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_STRIDE); + reg = dap << TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_CLK_SEL_P | + dap << TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA1_SEL_P | + dap << TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA2_SEL_P; + + tegra_das_write(addr, reg); + + return 0; +} +EXPORT_SYMBOL_GPL(tegra_das_connect_dac_to_dap); + +#ifdef CONFIG_DEBUG_FS +static int tegra_das_show(struct seq_file *s, void *unused) +{ + int i; + u32 addr; + u32 reg; + + for (i = 0; i < TEGRA_DAS_DAP_CTRL_SEL_COUNT; i++) { + addr = TEGRA_DAS_DAP_CTRL_SEL + + (i * TEGRA_DAS_DAP_CTRL_SEL_STRIDE); + reg = tegra_das_read(addr); + seq_printf(s, "TEGRA_DAS_DAP_CTRL_SEL[%d] = %08x\n", i, reg); + } + + for (i = 0; i < TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_COUNT; i++) { + addr = TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL + + (i * TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_STRIDE); + reg = tegra_das_read(addr); + seq_printf(s, "TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL[%d] = %08x\n", + i, reg); + } + + return 0; +} + +static int tegra_das_debug_open(struct inode *inode, struct file *file) +{ + return single_open(file, tegra_das_show, inode->i_private); +} + +static const struct file_operations tegra_das_debug_fops = { + .open = tegra_das_debug_open, + .read = seq_read, + .llseek = seq_lseek, + .release = single_release, +}; + +static void tegra_das_debug_add(struct tegra_das *das) +{ + das->debug = debugfs_create_file(DRV_NAME, S_IRUGO, + snd_soc_debugfs_root, das, + &tegra_das_debug_fops); +} + +static void tegra_das_debug_remove(struct tegra_das *das) +{ + if (das->debug) + debugfs_remove(das->debug); +} +#else +static inline void tegra_das_debug_add(struct tegra_das *das) +{ +} + +static inline void tegra_das_debug_remove(struct tegra_das *das) +{ +} +#endif + +static int __devinit tegra_das_probe(struct platform_device *pdev) +{ + struct resource *res, *region; + int ret = 0; + + if (das) + return -ENODEV; + + das = devm_kzalloc(&pdev->dev, sizeof(struct tegra_das), GFP_KERNEL); + if (!das) { + dev_err(&pdev->dev, "Can't allocate tegra_das\n"); + ret = -ENOMEM; + goto err; + } + das->dev = &pdev->dev; + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!res) { + dev_err(&pdev->dev, "No memory resource\n"); + ret = -ENODEV; + goto err; + } + + region = devm_request_mem_region(&pdev->dev, res->start, + resource_size(res), pdev->name); + if (!region) { + dev_err(&pdev->dev, "Memory region already claimed\n"); + ret = -EBUSY; + goto err; + } + + das->regs = devm_ioremap(&pdev->dev, res->start, resource_size(res)); + if (!das->regs) { + dev_err(&pdev->dev, "ioremap failed\n"); + ret = -ENOMEM; + goto err; + } + + ret = tegra_das_connect_dap_to_dac(TEGRA_DAS_DAP_ID_1, + TEGRA_DAS_DAP_SEL_DAC1); + if (ret) { + dev_err(&pdev->dev, "Can't set up DAS DAP connection\n"); + goto err; + } + ret = tegra_das_connect_dac_to_dap(TEGRA_DAS_DAC_ID_1, + TEGRA_DAS_DAC_SEL_DAP1); + if (ret) { + dev_err(&pdev->dev, "Can't set up DAS DAC connection\n"); + goto err; + } + + tegra_das_debug_add(das); + + platform_set_drvdata(pdev, das); + + return 0; + +err: + das = NULL; + return ret; +} + +static int __devexit tegra_das_remove(struct platform_device *pdev) +{ + if (!das) + return -ENODEV; + + tegra_das_debug_remove(das); + + das = NULL; + + return 0; +} + +static const struct of_device_id tegra_das_of_match[] __devinitconst = { + { .compatible = "nvidia,tegra20-das", }, + {}, +}; + +static struct platform_driver tegra_das_driver = { + .probe = tegra_das_probe, + .remove = __devexit_p(tegra_das_remove), + .driver = { + .name = DRV_NAME, + .owner = THIS_MODULE, + .of_match_table = tegra_das_of_match, + }, +}; +module_platform_driver(tegra_das_driver); + +MODULE_AUTHOR("Stephen Warren "); +MODULE_DESCRIPTION("Tegra DAS driver"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:" DRV_NAME); +MODULE_DEVICE_TABLE(of, tegra_das_of_match); diff --git a/sound/soc/tegra/tegra20_das.h b/sound/soc/tegra/tegra20_das.h new file mode 100644 index 000000000000..896bf031a801 --- /dev/null +++ b/sound/soc/tegra/tegra20_das.h @@ -0,0 +1,135 @@ +/* + * tegra20_das.h - Definitions for Tegra20 DAS driver + * + * Author: Stephen Warren + * Copyright (C) 2010 - NVIDIA, Inc. + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#ifndef __TEGRA_DAS_H__ +#define __TEGRA_DAS_H__ + +/* Register TEGRA_DAS_DAP_CTRL_SEL */ +#define TEGRA_DAS_DAP_CTRL_SEL 0x00 +#define TEGRA_DAS_DAP_CTRL_SEL_COUNT 5 +#define TEGRA_DAS_DAP_CTRL_SEL_STRIDE 4 +#define TEGRA_DAS_DAP_CTRL_SEL_DAP_MS_SEL_P 31 +#define TEGRA_DAS_DAP_CTRL_SEL_DAP_MS_SEL_S 1 +#define TEGRA_DAS_DAP_CTRL_SEL_DAP_SDATA1_TX_RX_P 30 +#define TEGRA_DAS_DAP_CTRL_SEL_DAP_SDATA1_TX_RX_S 1 +#define TEGRA_DAS_DAP_CTRL_SEL_DAP_SDATA2_TX_RX_P 29 +#define TEGRA_DAS_DAP_CTRL_SEL_DAP_SDATA2_TX_RX_S 1 +#define TEGRA_DAS_DAP_CTRL_SEL_DAP_CTRL_SEL_P 0 +#define TEGRA_DAS_DAP_CTRL_SEL_DAP_CTRL_SEL_S 5 + +/* Values for field TEGRA_DAS_DAP_CTRL_SEL_DAP_CTRL_SEL */ +#define TEGRA_DAS_DAP_SEL_DAC1 0 +#define TEGRA_DAS_DAP_SEL_DAC2 1 +#define TEGRA_DAS_DAP_SEL_DAC3 2 +#define TEGRA_DAS_DAP_SEL_DAP1 16 +#define TEGRA_DAS_DAP_SEL_DAP2 17 +#define TEGRA_DAS_DAP_SEL_DAP3 18 +#define TEGRA_DAS_DAP_SEL_DAP4 19 +#define TEGRA_DAS_DAP_SEL_DAP5 20 + +/* Register TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL */ +#define TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL 0x40 +#define TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_COUNT 3 +#define TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_STRIDE 4 +#define TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA2_SEL_P 28 +#define TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA2_SEL_S 4 +#define TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA1_SEL_P 24 +#define TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA1_SEL_S 4 +#define TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_CLK_SEL_P 0 +#define TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_CLK_SEL_S 4 + +/* + * Values for: + * TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA2_SEL + * TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA1_SEL + * TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_CLK_SEL + */ +#define TEGRA_DAS_DAC_SEL_DAP1 0 +#define TEGRA_DAS_DAC_SEL_DAP2 1 +#define TEGRA_DAS_DAC_SEL_DAP3 2 +#define TEGRA_DAS_DAC_SEL_DAP4 3 +#define TEGRA_DAS_DAC_SEL_DAP5 4 + +/* + * Names/IDs of the DACs/DAPs. + */ + +#define TEGRA_DAS_DAP_ID_1 0 +#define TEGRA_DAS_DAP_ID_2 1 +#define TEGRA_DAS_DAP_ID_3 2 +#define TEGRA_DAS_DAP_ID_4 3 +#define TEGRA_DAS_DAP_ID_5 4 + +#define TEGRA_DAS_DAC_ID_1 0 +#define TEGRA_DAS_DAC_ID_2 1 +#define TEGRA_DAS_DAC_ID_3 2 + +struct tegra_das { + struct device *dev; + void __iomem *regs; + struct dentry *debug; +}; + +/* + * Terminology: + * DAS: Digital audio switch (HW module controlled by this driver) + * DAP: Digital audio port (port/pins on Tegra device) + * DAC: Digital audio controller (e.g. I2S or AC97 controller elsewhere) + * + * The Tegra DAS is a mux/cross-bar which can connect each DAP to a specific + * DAC, or another DAP. When DAPs are connected, one must be the master and + * one the slave. Each DAC allows selection of a specific DAP for input, to + * cater for the case where N DAPs are connected to 1 DAC for broadcast + * output. + * + * This driver is dumb; no attempt is made to ensure that a valid routing + * configuration is programmed. + */ + +/* + * Connect a DAP to to a DAC + * dap_id: DAP to connect: TEGRA_DAS_DAP_ID_* + * dac_sel: DAC to connect to: TEGRA_DAS_DAP_SEL_DAC* + */ +extern int tegra_das_connect_dap_to_dac(int dap_id, int dac_sel); + +/* + * Connect a DAP to to another DAP + * dap_id: DAP to connect: TEGRA_DAS_DAP_ID_* + * other_dap_sel: DAP to connect to: TEGRA_DAS_DAP_SEL_DAP* + * master: Is this DAP the master (1) or slave (0) + * sdata1rx: Is this DAP's SDATA1 pin RX (1) or TX (0) + * sdata2rx: Is this DAP's SDATA2 pin RX (1) or TX (0) + */ +extern int tegra_das_connect_dap_to_dap(int dap_id, int other_dap_sel, + int master, int sdata1rx, + int sdata2rx); + +/* + * Connect a DAC's input to a DAP + * (DAC outputs are selected by the DAP) + * dac_id: DAC ID to connect: TEGRA_DAS_DAC_ID_* + * dap_sel: DAP to receive input from: TEGRA_DAS_DAC_SEL_DAP* + */ +extern int tegra_das_connect_dac_to_dap(int dac_id, int dap_sel); + +#endif diff --git a/sound/soc/tegra/tegra20_i2s.c b/sound/soc/tegra/tegra20_i2s.c new file mode 100644 index 000000000000..e24759ae291b --- /dev/null +++ b/sound/soc/tegra/tegra20_i2s.c @@ -0,0 +1,458 @@ +/* + * tegra20_i2s.c - Tegra20 I2S driver + * + * Author: Stephen Warren + * Copyright (C) 2010,2012 - NVIDIA, Inc. + * + * Based on code copyright/by: + * + * Copyright (c) 2009-2010, NVIDIA Corporation. + * Scott Peterson + * + * Copyright (C) 2010 Google, Inc. + * Iliyan Malchev + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "tegra20_i2s.h" + +#define DRV_NAME "tegra-i2s" + +static inline void tegra_i2s_write(struct tegra_i2s *i2s, u32 reg, u32 val) +{ + __raw_writel(val, i2s->regs + reg); +} + +static inline u32 tegra_i2s_read(struct tegra_i2s *i2s, u32 reg) +{ + return __raw_readl(i2s->regs + reg); +} + +#ifdef CONFIG_DEBUG_FS +static int tegra_i2s_show(struct seq_file *s, void *unused) +{ +#define REG(r) { r, #r } + static const struct { + int offset; + const char *name; + } regs[] = { + REG(TEGRA_I2S_CTRL), + REG(TEGRA_I2S_STATUS), + REG(TEGRA_I2S_TIMING), + REG(TEGRA_I2S_FIFO_SCR), + REG(TEGRA_I2S_PCM_CTRL), + REG(TEGRA_I2S_NW_CTRL), + REG(TEGRA_I2S_TDM_CTRL), + REG(TEGRA_I2S_TDM_TX_RX_CTRL), + }; +#undef REG + + struct tegra_i2s *i2s = s->private; + int i; + + for (i = 0; i < ARRAY_SIZE(regs); i++) { + u32 val = tegra_i2s_read(i2s, regs[i].offset); + seq_printf(s, "%s = %08x\n", regs[i].name, val); + } + + return 0; +} + +static int tegra_i2s_debug_open(struct inode *inode, struct file *file) +{ + return single_open(file, tegra_i2s_show, inode->i_private); +} + +static const struct file_operations tegra_i2s_debug_fops = { + .open = tegra_i2s_debug_open, + .read = seq_read, + .llseek = seq_lseek, + .release = single_release, +}; + +static void tegra_i2s_debug_add(struct tegra_i2s *i2s) +{ + i2s->debug = debugfs_create_file(i2s->dai.name, S_IRUGO, + snd_soc_debugfs_root, i2s, + &tegra_i2s_debug_fops); +} + +static void tegra_i2s_debug_remove(struct tegra_i2s *i2s) +{ + if (i2s->debug) + debugfs_remove(i2s->debug); +} +#else +static inline void tegra_i2s_debug_add(struct tegra_i2s *i2s, int id) +{ +} + +static inline void tegra_i2s_debug_remove(struct tegra_i2s *i2s) +{ +} +#endif + +static int tegra_i2s_set_fmt(struct snd_soc_dai *dai, + unsigned int fmt) +{ + struct tegra_i2s *i2s = snd_soc_dai_get_drvdata(dai); + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + default: + return -EINVAL; + } + + i2s->reg_ctrl &= ~TEGRA_I2S_CTRL_MASTER_ENABLE; + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + i2s->reg_ctrl |= TEGRA_I2S_CTRL_MASTER_ENABLE; + break; + case SND_SOC_DAIFMT_CBM_CFM: + break; + default: + return -EINVAL; + } + + i2s->reg_ctrl &= ~(TEGRA_I2S_CTRL_BIT_FORMAT_MASK | + TEGRA_I2S_CTRL_LRCK_MASK); + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_A: + i2s->reg_ctrl |= TEGRA_I2S_CTRL_BIT_FORMAT_DSP; + i2s->reg_ctrl |= TEGRA_I2S_CTRL_LRCK_L_LOW; + break; + case SND_SOC_DAIFMT_DSP_B: + i2s->reg_ctrl |= TEGRA_I2S_CTRL_BIT_FORMAT_DSP; + i2s->reg_ctrl |= TEGRA_I2S_CTRL_LRCK_R_LOW; + break; + case SND_SOC_DAIFMT_I2S: + i2s->reg_ctrl |= TEGRA_I2S_CTRL_BIT_FORMAT_I2S; + i2s->reg_ctrl |= TEGRA_I2S_CTRL_LRCK_L_LOW; + break; + case SND_SOC_DAIFMT_RIGHT_J: + i2s->reg_ctrl |= TEGRA_I2S_CTRL_BIT_FORMAT_RJM; + i2s->reg_ctrl |= TEGRA_I2S_CTRL_LRCK_L_LOW; + break; + case SND_SOC_DAIFMT_LEFT_J: + i2s->reg_ctrl |= TEGRA_I2S_CTRL_BIT_FORMAT_LJM; + i2s->reg_ctrl |= TEGRA_I2S_CTRL_LRCK_L_LOW; + break; + default: + return -EINVAL; + } + + return 0; +} + +static int tegra_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct device *dev = substream->pcm->card->dev; + struct tegra_i2s *i2s = snd_soc_dai_get_drvdata(dai); + u32 reg; + int ret, sample_size, srate, i2sclock, bitcnt; + + i2s->reg_ctrl &= ~TEGRA_I2S_CTRL_BIT_SIZE_MASK; + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + i2s->reg_ctrl |= TEGRA_I2S_CTRL_BIT_SIZE_16; + sample_size = 16; + break; + case SNDRV_PCM_FORMAT_S24_LE: + i2s->reg_ctrl |= TEGRA_I2S_CTRL_BIT_SIZE_24; + sample_size = 24; + break; + case SNDRV_PCM_FORMAT_S32_LE: + i2s->reg_ctrl |= TEGRA_I2S_CTRL_BIT_SIZE_32; + sample_size = 32; + break; + default: + return -EINVAL; + } + + srate = params_rate(params); + + /* Final "* 2" required by Tegra hardware */ + i2sclock = srate * params_channels(params) * sample_size * 2; + + ret = clk_set_rate(i2s->clk_i2s, i2sclock); + if (ret) { + dev_err(dev, "Can't set I2S clock rate: %d\n", ret); + return ret; + } + + bitcnt = (i2sclock / (2 * srate)) - 1; + if (bitcnt < 0 || bitcnt > TEGRA_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US) + return -EINVAL; + reg = bitcnt << TEGRA_I2S_TIMING_CHANNEL_BIT_COUNT_SHIFT; + + if (i2sclock % (2 * srate)) + reg |= TEGRA_I2S_TIMING_NON_SYM_ENABLE; + + clk_enable(i2s->clk_i2s); + + tegra_i2s_write(i2s, TEGRA_I2S_TIMING, reg); + + tegra_i2s_write(i2s, TEGRA_I2S_FIFO_SCR, + TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_FOUR_SLOTS | + TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_FOUR_SLOTS); + + clk_disable(i2s->clk_i2s); + + return 0; +} + +static void tegra_i2s_start_playback(struct tegra_i2s *i2s) +{ + i2s->reg_ctrl |= TEGRA_I2S_CTRL_FIFO1_ENABLE; + tegra_i2s_write(i2s, TEGRA_I2S_CTRL, i2s->reg_ctrl); +} + +static void tegra_i2s_stop_playback(struct tegra_i2s *i2s) +{ + i2s->reg_ctrl &= ~TEGRA_I2S_CTRL_FIFO1_ENABLE; + tegra_i2s_write(i2s, TEGRA_I2S_CTRL, i2s->reg_ctrl); +} + +static void tegra_i2s_start_capture(struct tegra_i2s *i2s) +{ + i2s->reg_ctrl |= TEGRA_I2S_CTRL_FIFO2_ENABLE; + tegra_i2s_write(i2s, TEGRA_I2S_CTRL, i2s->reg_ctrl); +} + +static void tegra_i2s_stop_capture(struct tegra_i2s *i2s) +{ + i2s->reg_ctrl &= ~TEGRA_I2S_CTRL_FIFO2_ENABLE; + tegra_i2s_write(i2s, TEGRA_I2S_CTRL, i2s->reg_ctrl); +} + +static int tegra_i2s_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct tegra_i2s *i2s = snd_soc_dai_get_drvdata(dai); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + case SNDRV_PCM_TRIGGER_RESUME: + clk_enable(i2s->clk_i2s); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + tegra_i2s_start_playback(i2s); + else + tegra_i2s_start_capture(i2s); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + case SNDRV_PCM_TRIGGER_SUSPEND: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + tegra_i2s_stop_playback(i2s); + else + tegra_i2s_stop_capture(i2s); + clk_disable(i2s->clk_i2s); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int tegra_i2s_probe(struct snd_soc_dai *dai) +{ + struct tegra_i2s *i2s = snd_soc_dai_get_drvdata(dai); + + dai->capture_dma_data = &i2s->capture_dma_data; + dai->playback_dma_data = &i2s->playback_dma_data; + + return 0; +} + +static const struct snd_soc_dai_ops tegra_i2s_dai_ops = { + .set_fmt = tegra_i2s_set_fmt, + .hw_params = tegra_i2s_hw_params, + .trigger = tegra_i2s_trigger, +}; + +static const struct snd_soc_dai_driver tegra_i2s_dai_template = { + .probe = tegra_i2s_probe, + .playback = { + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .capture = { + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .ops = &tegra_i2s_dai_ops, + .symmetric_rates = 1, +}; + +static __devinit int tegra_i2s_platform_probe(struct platform_device *pdev) +{ + struct tegra_i2s *i2s; + struct resource *mem, *memregion, *dmareq; + u32 of_dma[2]; + u32 dma_ch; + int ret; + + i2s = devm_kzalloc(&pdev->dev, sizeof(struct tegra_i2s), GFP_KERNEL); + if (!i2s) { + dev_err(&pdev->dev, "Can't allocate tegra_i2s\n"); + ret = -ENOMEM; + goto err; + } + dev_set_drvdata(&pdev->dev, i2s); + + i2s->dai = tegra_i2s_dai_template; + i2s->dai.name = dev_name(&pdev->dev); + + i2s->clk_i2s = clk_get(&pdev->dev, NULL); + if (IS_ERR(i2s->clk_i2s)) { + dev_err(&pdev->dev, "Can't retrieve i2s clock\n"); + ret = PTR_ERR(i2s->clk_i2s); + goto err; + } + + mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!mem) { + dev_err(&pdev->dev, "No memory resource\n"); + ret = -ENODEV; + goto err_clk_put; + } + + dmareq = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!dmareq) { + if (of_property_read_u32_array(pdev->dev.of_node, + "nvidia,dma-request-selector", + of_dma, 2) < 0) { + dev_err(&pdev->dev, "No DMA resource\n"); + ret = -ENODEV; + goto err_clk_put; + } + dma_ch = of_dma[1]; + } else { + dma_ch = dmareq->start; + } + + memregion = devm_request_mem_region(&pdev->dev, mem->start, + resource_size(mem), DRV_NAME); + if (!memregion) { + dev_err(&pdev->dev, "Memory region already claimed\n"); + ret = -EBUSY; + goto err_clk_put; + } + + i2s->regs = devm_ioremap(&pdev->dev, mem->start, resource_size(mem)); + if (!i2s->regs) { + dev_err(&pdev->dev, "ioremap failed\n"); + ret = -ENOMEM; + goto err_clk_put; + } + + i2s->capture_dma_data.addr = mem->start + TEGRA_I2S_FIFO2; + i2s->capture_dma_data.wrap = 4; + i2s->capture_dma_data.width = 32; + i2s->capture_dma_data.req_sel = dma_ch; + + i2s->playback_dma_data.addr = mem->start + TEGRA_I2S_FIFO1; + i2s->playback_dma_data.wrap = 4; + i2s->playback_dma_data.width = 32; + i2s->playback_dma_data.req_sel = dma_ch; + + i2s->reg_ctrl = TEGRA_I2S_CTRL_FIFO_FORMAT_PACKED; + + ret = snd_soc_register_dai(&pdev->dev, &i2s->dai); + if (ret) { + dev_err(&pdev->dev, "Could not register DAI: %d\n", ret); + ret = -ENOMEM; + goto err_clk_put; + } + + ret = tegra_pcm_platform_register(&pdev->dev); + if (ret) { + dev_err(&pdev->dev, "Could not register PCM: %d\n", ret); + goto err_unregister_dai; + } + + tegra_i2s_debug_add(i2s); + + return 0; + +err_unregister_dai: + snd_soc_unregister_dai(&pdev->dev); +err_clk_put: + clk_put(i2s->clk_i2s); +err: + return ret; +} + +static int __devexit tegra_i2s_platform_remove(struct platform_device *pdev) +{ + struct tegra_i2s *i2s = dev_get_drvdata(&pdev->dev); + + tegra_pcm_platform_unregister(&pdev->dev); + snd_soc_unregister_dai(&pdev->dev); + + tegra_i2s_debug_remove(i2s); + + clk_put(i2s->clk_i2s); + + return 0; +} + +static const struct of_device_id tegra_i2s_of_match[] __devinitconst = { + { .compatible = "nvidia,tegra20-i2s", }, + {}, +}; + +static struct platform_driver tegra_i2s_driver = { + .driver = { + .name = DRV_NAME, + .owner = THIS_MODULE, + .of_match_table = tegra_i2s_of_match, + }, + .probe = tegra_i2s_platform_probe, + .remove = __devexit_p(tegra_i2s_platform_remove), +}; +module_platform_driver(tegra_i2s_driver); + +MODULE_AUTHOR("Stephen Warren "); +MODULE_DESCRIPTION("Tegra I2S ASoC driver"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:" DRV_NAME); +MODULE_DEVICE_TABLE(of, tegra_i2s_of_match); diff --git a/sound/soc/tegra/tegra20_i2s.h b/sound/soc/tegra/tegra20_i2s.h new file mode 100644 index 000000000000..56f1e0f87c0b --- /dev/null +++ b/sound/soc/tegra/tegra20_i2s.h @@ -0,0 +1,165 @@ +/* + * tegra20_i2s.h - Definitions for Tegra20 I2S driver + * + * Author: Stephen Warren + * Copyright (C) 2010 - NVIDIA, Inc. + * + * Based on code copyright/by: + * + * Copyright (c) 2009-2010, NVIDIA Corporation. + * Scott Peterson + * + * Copyright (C) 2010 Google, Inc. + * Iliyan Malchev + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#ifndef __TEGRA_I2S_H__ +#define __TEGRA_I2S_H__ + +#include "tegra_pcm.h" + +/* Register offsets from TEGRA_I2S1_BASE and TEGRA_I2S2_BASE */ + +#define TEGRA_I2S_CTRL 0x00 +#define TEGRA_I2S_STATUS 0x04 +#define TEGRA_I2S_TIMING 0x08 +#define TEGRA_I2S_FIFO_SCR 0x0c +#define TEGRA_I2S_PCM_CTRL 0x10 +#define TEGRA_I2S_NW_CTRL 0x14 +#define TEGRA_I2S_TDM_CTRL 0x20 +#define TEGRA_I2S_TDM_TX_RX_CTRL 0x24 +#define TEGRA_I2S_FIFO1 0x40 +#define TEGRA_I2S_FIFO2 0x80 + +/* Fields in TEGRA_I2S_CTRL */ + +#define TEGRA_I2S_CTRL_FIFO2_TX_ENABLE (1 << 30) +#define TEGRA_I2S_CTRL_FIFO1_ENABLE (1 << 29) +#define TEGRA_I2S_CTRL_FIFO2_ENABLE (1 << 28) +#define TEGRA_I2S_CTRL_FIFO1_RX_ENABLE (1 << 27) +#define TEGRA_I2S_CTRL_FIFO_LPBK_ENABLE (1 << 26) +#define TEGRA_I2S_CTRL_MASTER_ENABLE (1 << 25) + +#define TEGRA_I2S_LRCK_LEFT_LOW 0 +#define TEGRA_I2S_LRCK_RIGHT_LOW 1 + +#define TEGRA_I2S_CTRL_LRCK_SHIFT 24 +#define TEGRA_I2S_CTRL_LRCK_MASK (1 << TEGRA_I2S_CTRL_LRCK_SHIFT) +#define TEGRA_I2S_CTRL_LRCK_L_LOW (TEGRA_I2S_LRCK_LEFT_LOW << TEGRA_I2S_CTRL_LRCK_SHIFT) +#define TEGRA_I2S_CTRL_LRCK_R_LOW (TEGRA_I2S_LRCK_RIGHT_LOW << TEGRA_I2S_CTRL_LRCK_SHIFT) + +#define TEGRA_I2S_BIT_FORMAT_I2S 0 +#define TEGRA_I2S_BIT_FORMAT_RJM 1 +#define TEGRA_I2S_BIT_FORMAT_LJM 2 +#define TEGRA_I2S_BIT_FORMAT_DSP 3 + +#define TEGRA_I2S_CTRL_BIT_FORMAT_SHIFT 10 +#define TEGRA_I2S_CTRL_BIT_FORMAT_MASK (3 << TEGRA_I2S_CTRL_BIT_FORMAT_SHIFT) +#define TEGRA_I2S_CTRL_BIT_FORMAT_I2S (TEGRA_I2S_BIT_FORMAT_I2S << TEGRA_I2S_CTRL_BIT_FORMAT_SHIFT) +#define TEGRA_I2S_CTRL_BIT_FORMAT_RJM (TEGRA_I2S_BIT_FORMAT_RJM << TEGRA_I2S_CTRL_BIT_FORMAT_SHIFT) +#define TEGRA_I2S_CTRL_BIT_FORMAT_LJM (TEGRA_I2S_BIT_FORMAT_LJM << TEGRA_I2S_CTRL_BIT_FORMAT_SHIFT) +#define TEGRA_I2S_CTRL_BIT_FORMAT_DSP (TEGRA_I2S_BIT_FORMAT_DSP << TEGRA_I2S_CTRL_BIT_FORMAT_SHIFT) + +#define TEGRA_I2S_BIT_SIZE_16 0 +#define TEGRA_I2S_BIT_SIZE_20 1 +#define TEGRA_I2S_BIT_SIZE_24 2 +#define TEGRA_I2S_BIT_SIZE_32 3 + +#define TEGRA_I2S_CTRL_BIT_SIZE_SHIFT 8 +#define TEGRA_I2S_CTRL_BIT_SIZE_MASK (3 << TEGRA_I2S_CTRL_BIT_SIZE_SHIFT) +#define TEGRA_I2S_CTRL_BIT_SIZE_16 (TEGRA_I2S_BIT_SIZE_16 << TEGRA_I2S_CTRL_BIT_SIZE_SHIFT) +#define TEGRA_I2S_CTRL_BIT_SIZE_20 (TEGRA_I2S_BIT_SIZE_20 << TEGRA_I2S_CTRL_BIT_SIZE_SHIFT) +#define TEGRA_I2S_CTRL_BIT_SIZE_24 (TEGRA_I2S_BIT_SIZE_24 << TEGRA_I2S_CTRL_BIT_SIZE_SHIFT) +#define TEGRA_I2S_CTRL_BIT_SIZE_32 (TEGRA_I2S_BIT_SIZE_32 << TEGRA_I2S_CTRL_BIT_SIZE_SHIFT) + +#define TEGRA_I2S_FIFO_16_LSB 0 +#define TEGRA_I2S_FIFO_20_LSB 1 +#define TEGRA_I2S_FIFO_24_LSB 2 +#define TEGRA_I2S_FIFO_32 3 +#define TEGRA_I2S_FIFO_PACKED 7 + +#define TEGRA_I2S_CTRL_FIFO_FORMAT_SHIFT 4 +#define TEGRA_I2S_CTRL_FIFO_FORMAT_MASK (7 << TEGRA_I2S_CTRL_FIFO_FORMAT_SHIFT) +#define TEGRA_I2S_CTRL_FIFO_FORMAT_16_LSB (TEGRA_I2S_FIFO_16_LSB << TEGRA_I2S_CTRL_FIFO_FORMAT_SHIFT) +#define TEGRA_I2S_CTRL_FIFO_FORMAT_20_LSB (TEGRA_I2S_FIFO_20_LSB << TEGRA_I2S_CTRL_FIFO_FORMAT_SHIFT) +#define TEGRA_I2S_CTRL_FIFO_FORMAT_24_LSB (TEGRA_I2S_FIFO_24_LSB << TEGRA_I2S_CTRL_FIFO_FORMAT_SHIFT) +#define TEGRA_I2S_CTRL_FIFO_FORMAT_32 (TEGRA_I2S_FIFO_32 << TEGRA_I2S_CTRL_FIFO_FORMAT_SHIFT) +#define TEGRA_I2S_CTRL_FIFO_FORMAT_PACKED (TEGRA_I2S_FIFO_PACKED << TEGRA_I2S_CTRL_FIFO_FORMAT_SHIFT) + +#define TEGRA_I2S_CTRL_IE_FIFO1_ERR (1 << 3) +#define TEGRA_I2S_CTRL_IE_FIFO2_ERR (1 << 2) +#define TEGRA_I2S_CTRL_QE_FIFO1 (1 << 1) +#define TEGRA_I2S_CTRL_QE_FIFO2 (1 << 0) + +/* Fields in TEGRA_I2S_STATUS */ + +#define TEGRA_I2S_STATUS_FIFO1_RDY (1 << 31) +#define TEGRA_I2S_STATUS_FIFO2_RDY (1 << 30) +#define TEGRA_I2S_STATUS_FIFO1_BSY (1 << 29) +#define TEGRA_I2S_STATUS_FIFO2_BSY (1 << 28) +#define TEGRA_I2S_STATUS_FIFO1_ERR (1 << 3) +#define TEGRA_I2S_STATUS_FIFO2_ERR (1 << 2) +#define TEGRA_I2S_STATUS_QS_FIFO1 (1 << 1) +#define TEGRA_I2S_STATUS_QS_FIFO2 (1 << 0) + +/* Fields in TEGRA_I2S_TIMING */ + +#define TEGRA_I2S_TIMING_NON_SYM_ENABLE (1 << 12) +#define TEGRA_I2S_TIMING_CHANNEL_BIT_COUNT_SHIFT 0 +#define TEGRA_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US 0x7fff +#define TEGRA_I2S_TIMING_CHANNEL_BIT_COUNT_MASK (TEGRA_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US << TEGRA_I2S_TIMING_CHANNEL_BIT_COUNT_SHIFT) + +/* Fields in TEGRA_I2S_FIFO_SCR */ + +#define TEGRA_I2S_FIFO_SCR_FIFO2_FULL_EMPTY_COUNT_SHIFT 24 +#define TEGRA_I2S_FIFO_SCR_FIFO1_FULL_EMPTY_COUNT_SHIFT 16 +#define TEGRA_I2S_FIFO_SCR_FIFO_FULL_EMPTY_COUNT_MASK 0x3f + +#define TEGRA_I2S_FIFO_SCR_FIFO2_CLR (1 << 12) +#define TEGRA_I2S_FIFO_SCR_FIFO1_CLR (1 << 8) + +#define TEGRA_I2S_FIFO_ATN_LVL_ONE_SLOT 0 +#define TEGRA_I2S_FIFO_ATN_LVL_FOUR_SLOTS 1 +#define TEGRA_I2S_FIFO_ATN_LVL_EIGHT_SLOTS 2 +#define TEGRA_I2S_FIFO_ATN_LVL_TWELVE_SLOTS 3 + +#define TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_SHIFT 4 +#define TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_MASK (3 << TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_SHIFT) +#define TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_ONE_SLOT (TEGRA_I2S_FIFO_ATN_LVL_ONE_SLOT << TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_SHIFT) +#define TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_FOUR_SLOTS (TEGRA_I2S_FIFO_ATN_LVL_FOUR_SLOTS << TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_SHIFT) +#define TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_EIGHT_SLOTS (TEGRA_I2S_FIFO_ATN_LVL_EIGHT_SLOTS << TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_SHIFT) +#define TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_TWELVE_SLOTS (TEGRA_I2S_FIFO_ATN_LVL_TWELVE_SLOTS << TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_SHIFT) + +#define TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_SHIFT 0 +#define TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_MASK (3 << TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_SHIFT) +#define TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_ONE_SLOT (TEGRA_I2S_FIFO_ATN_LVL_ONE_SLOT << TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_SHIFT) +#define TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_FOUR_SLOTS (TEGRA_I2S_FIFO_ATN_LVL_FOUR_SLOTS << TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_SHIFT) +#define TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_EIGHT_SLOTS (TEGRA_I2S_FIFO_ATN_LVL_EIGHT_SLOTS << TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_SHIFT) +#define TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_TWELVE_SLOTS (TEGRA_I2S_FIFO_ATN_LVL_TWELVE_SLOTS << TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_SHIFT) + +struct tegra_i2s { + struct snd_soc_dai_driver dai; + struct clk *clk_i2s; + struct tegra_pcm_dma_params capture_dma_data; + struct tegra_pcm_dma_params playback_dma_data; + void __iomem *regs; + struct dentry *debug; + u32 reg_ctrl; +}; + +#endif diff --git a/sound/soc/tegra/tegra20_spdif.c b/sound/soc/tegra/tegra20_spdif.c new file mode 100644 index 000000000000..ed1fd502ef8d --- /dev/null +++ b/sound/soc/tegra/tegra20_spdif.c @@ -0,0 +1,365 @@ +/* + * tegra20_spdif.c - Tegra20 SPDIF driver + * + * Author: Stephen Warren + * Copyright (C) 2011-2012 - NVIDIA, Inc. + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "tegra20_spdif.h" + +#define DRV_NAME "tegra-spdif" + +static inline void tegra_spdif_write(struct tegra_spdif *spdif, u32 reg, + u32 val) +{ + __raw_writel(val, spdif->regs + reg); +} + +static inline u32 tegra_spdif_read(struct tegra_spdif *spdif, u32 reg) +{ + return __raw_readl(spdif->regs + reg); +} + +#ifdef CONFIG_DEBUG_FS +static int tegra_spdif_show(struct seq_file *s, void *unused) +{ +#define REG(r) { r, #r } + static const struct { + int offset; + const char *name; + } regs[] = { + REG(TEGRA_SPDIF_CTRL), + REG(TEGRA_SPDIF_STATUS), + REG(TEGRA_SPDIF_STROBE_CTRL), + REG(TEGRA_SPDIF_DATA_FIFO_CSR), + REG(TEGRA_SPDIF_CH_STA_RX_A), + REG(TEGRA_SPDIF_CH_STA_RX_B), + REG(TEGRA_SPDIF_CH_STA_RX_C), + REG(TEGRA_SPDIF_CH_STA_RX_D), + REG(TEGRA_SPDIF_CH_STA_RX_E), + REG(TEGRA_SPDIF_CH_STA_RX_F), + REG(TEGRA_SPDIF_CH_STA_TX_A), + REG(TEGRA_SPDIF_CH_STA_TX_B), + REG(TEGRA_SPDIF_CH_STA_TX_C), + REG(TEGRA_SPDIF_CH_STA_TX_D), + REG(TEGRA_SPDIF_CH_STA_TX_E), + REG(TEGRA_SPDIF_CH_STA_TX_F), + }; +#undef REG + + struct tegra_spdif *spdif = s->private; + int i; + + for (i = 0; i < ARRAY_SIZE(regs); i++) { + u32 val = tegra_spdif_read(spdif, regs[i].offset); + seq_printf(s, "%s = %08x\n", regs[i].name, val); + } + + return 0; +} + +static int tegra_spdif_debug_open(struct inode *inode, struct file *file) +{ + return single_open(file, tegra_spdif_show, inode->i_private); +} + +static const struct file_operations tegra_spdif_debug_fops = { + .open = tegra_spdif_debug_open, + .read = seq_read, + .llseek = seq_lseek, + .release = single_release, +}; + +static void tegra_spdif_debug_add(struct tegra_spdif *spdif) +{ + spdif->debug = debugfs_create_file(DRV_NAME, S_IRUGO, + snd_soc_debugfs_root, spdif, + &tegra_spdif_debug_fops); +} + +static void tegra_spdif_debug_remove(struct tegra_spdif *spdif) +{ + if (spdif->debug) + debugfs_remove(spdif->debug); +} +#else +static inline void tegra_spdif_debug_add(struct tegra_spdif *spdif) +{ +} + +static inline void tegra_spdif_debug_remove(struct tegra_spdif *spdif) +{ +} +#endif + +static int tegra_spdif_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct device *dev = substream->pcm->card->dev; + struct tegra_spdif *spdif = snd_soc_dai_get_drvdata(dai); + int ret, spdifclock; + + spdif->reg_ctrl &= ~TEGRA_SPDIF_CTRL_PACK; + spdif->reg_ctrl &= ~TEGRA_SPDIF_CTRL_BIT_MODE_MASK; + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + spdif->reg_ctrl |= TEGRA_SPDIF_CTRL_PACK; + spdif->reg_ctrl |= TEGRA_SPDIF_CTRL_BIT_MODE_16BIT; + break; + default: + return -EINVAL; + } + + switch (params_rate(params)) { + case 32000: + spdifclock = 4096000; + break; + case 44100: + spdifclock = 5644800; + break; + case 48000: + spdifclock = 6144000; + break; + case 88200: + spdifclock = 11289600; + break; + case 96000: + spdifclock = 12288000; + break; + case 176400: + spdifclock = 22579200; + break; + case 192000: + spdifclock = 24576000; + break; + default: + return -EINVAL; + } + + ret = clk_set_rate(spdif->clk_spdif_out, spdifclock); + if (ret) { + dev_err(dev, "Can't set SPDIF clock rate: %d\n", ret); + return ret; + } + + return 0; +} + +static void tegra_spdif_start_playback(struct tegra_spdif *spdif) +{ + spdif->reg_ctrl |= TEGRA_SPDIF_CTRL_TX_EN; + tegra_spdif_write(spdif, TEGRA_SPDIF_CTRL, spdif->reg_ctrl); +} + +static void tegra_spdif_stop_playback(struct tegra_spdif *spdif) +{ + spdif->reg_ctrl &= ~TEGRA_SPDIF_CTRL_TX_EN; + tegra_spdif_write(spdif, TEGRA_SPDIF_CTRL, spdif->reg_ctrl); +} + +static int tegra_spdif_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct tegra_spdif *spdif = snd_soc_dai_get_drvdata(dai); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + case SNDRV_PCM_TRIGGER_RESUME: + clk_enable(spdif->clk_spdif_out); + tegra_spdif_start_playback(spdif); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + case SNDRV_PCM_TRIGGER_SUSPEND: + tegra_spdif_stop_playback(spdif); + clk_disable(spdif->clk_spdif_out); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int tegra_spdif_probe(struct snd_soc_dai *dai) +{ + struct tegra_spdif *spdif = snd_soc_dai_get_drvdata(dai); + + dai->capture_dma_data = NULL; + dai->playback_dma_data = &spdif->playback_dma_data; + + return 0; +} + +static const struct snd_soc_dai_ops tegra_spdif_dai_ops = { + .hw_params = tegra_spdif_hw_params, + .trigger = tegra_spdif_trigger, +}; + +static struct snd_soc_dai_driver tegra_spdif_dai = { + .name = DRV_NAME, + .probe = tegra_spdif_probe, + .playback = { + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .ops = &tegra_spdif_dai_ops, +}; + +static __devinit int tegra_spdif_platform_probe(struct platform_device *pdev) +{ + struct tegra_spdif *spdif; + struct resource *mem, *memregion, *dmareq; + int ret; + + spdif = kzalloc(sizeof(struct tegra_spdif), GFP_KERNEL); + if (!spdif) { + dev_err(&pdev->dev, "Can't allocate tegra_spdif\n"); + ret = -ENOMEM; + goto exit; + } + dev_set_drvdata(&pdev->dev, spdif); + + spdif->clk_spdif_out = clk_get(&pdev->dev, "spdif_out"); + if (IS_ERR(spdif->clk_spdif_out)) { + pr_err("Can't retrieve spdif clock\n"); + ret = PTR_ERR(spdif->clk_spdif_out); + goto err_free; + } + + mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!mem) { + dev_err(&pdev->dev, "No memory resource\n"); + ret = -ENODEV; + goto err_clk_put; + } + + dmareq = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!dmareq) { + dev_err(&pdev->dev, "No DMA resource\n"); + ret = -ENODEV; + goto err_clk_put; + } + + memregion = request_mem_region(mem->start, resource_size(mem), + DRV_NAME); + if (!memregion) { + dev_err(&pdev->dev, "Memory region already claimed\n"); + ret = -EBUSY; + goto err_clk_put; + } + + spdif->regs = ioremap(mem->start, resource_size(mem)); + if (!spdif->regs) { + dev_err(&pdev->dev, "ioremap failed\n"); + ret = -ENOMEM; + goto err_release; + } + + spdif->playback_dma_data.addr = mem->start + TEGRA_SPDIF_DATA_OUT; + spdif->playback_dma_data.wrap = 4; + spdif->playback_dma_data.width = 32; + spdif->playback_dma_data.req_sel = dmareq->start; + + ret = snd_soc_register_dai(&pdev->dev, &tegra_spdif_dai); + if (ret) { + dev_err(&pdev->dev, "Could not register DAI: %d\n", ret); + ret = -ENOMEM; + goto err_unmap; + } + + ret = tegra_pcm_platform_register(&pdev->dev); + if (ret) { + dev_err(&pdev->dev, "Could not register PCM: %d\n", ret); + goto err_unregister_dai; + } + + tegra_spdif_debug_add(spdif); + + return 0; + +err_unregister_dai: + snd_soc_unregister_dai(&pdev->dev); +err_unmap: + iounmap(spdif->regs); +err_release: + release_mem_region(mem->start, resource_size(mem)); +err_clk_put: + clk_put(spdif->clk_spdif_out); +err_free: + kfree(spdif); +exit: + return ret; +} + +static int __devexit tegra_spdif_platform_remove(struct platform_device *pdev) +{ + struct tegra_spdif *spdif = dev_get_drvdata(&pdev->dev); + struct resource *res; + + tegra_pcm_platform_unregister(&pdev->dev); + snd_soc_unregister_dai(&pdev->dev); + + tegra_spdif_debug_remove(spdif); + + iounmap(spdif->regs); + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + release_mem_region(res->start, resource_size(res)); + + clk_put(spdif->clk_spdif_out); + + kfree(spdif); + + return 0; +} + +static struct platform_driver tegra_spdif_driver = { + .driver = { + .name = DRV_NAME, + .owner = THIS_MODULE, + }, + .probe = tegra_spdif_platform_probe, + .remove = __devexit_p(tegra_spdif_platform_remove), +}; + +module_platform_driver(tegra_spdif_driver); + +MODULE_AUTHOR("Stephen Warren "); +MODULE_DESCRIPTION("Tegra SPDIF ASoC driver"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:" DRV_NAME); diff --git a/sound/soc/tegra/tegra20_spdif.h b/sound/soc/tegra/tegra20_spdif.h new file mode 100644 index 000000000000..001f9dcd2033 --- /dev/null +++ b/sound/soc/tegra/tegra20_spdif.h @@ -0,0 +1,472 @@ +/* + * tegra20_spdif.h - Definitions for Tegra20 SPDIF driver + * + * Author: Stephen Warren + * Copyright (C) 2011 - NVIDIA, Inc. + * + * Based on code copyright/by: + * Copyright (c) 2008-2009, NVIDIA Corporation + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#ifndef __TEGRA_SPDIF_H__ +#define __TEGRA_SPDIF_H__ + +#include "tegra_pcm.h" + +/* Offsets from TEGRA_SPDIF_BASE */ + +#define TEGRA_SPDIF_CTRL 0x0 +#define TEGRA_SPDIF_STATUS 0x4 +#define TEGRA_SPDIF_STROBE_CTRL 0x8 +#define TEGRA_SPDIF_DATA_FIFO_CSR 0x0C +#define TEGRA_SPDIF_DATA_OUT 0x40 +#define TEGRA_SPDIF_DATA_IN 0x80 +#define TEGRA_SPDIF_CH_STA_RX_A 0x100 +#define TEGRA_SPDIF_CH_STA_RX_B 0x104 +#define TEGRA_SPDIF_CH_STA_RX_C 0x108 +#define TEGRA_SPDIF_CH_STA_RX_D 0x10C +#define TEGRA_SPDIF_CH_STA_RX_E 0x110 +#define TEGRA_SPDIF_CH_STA_RX_F 0x114 +#define TEGRA_SPDIF_CH_STA_TX_A 0x140 +#define TEGRA_SPDIF_CH_STA_TX_B 0x144 +#define TEGRA_SPDIF_CH_STA_TX_C 0x148 +#define TEGRA_SPDIF_CH_STA_TX_D 0x14C +#define TEGRA_SPDIF_CH_STA_TX_E 0x150 +#define TEGRA_SPDIF_CH_STA_TX_F 0x154 +#define TEGRA_SPDIF_USR_STA_RX_A 0x180 +#define TEGRA_SPDIF_USR_DAT_TX_A 0x1C0 + +/* Fields in TEGRA_SPDIF_CTRL */ + +/* Start capturing from 0=right, 1=left channel */ +#define TEGRA_SPDIF_CTRL_CAP_LC (1 << 30) + +/* SPDIF receiver(RX) enable */ +#define TEGRA_SPDIF_CTRL_RX_EN (1 << 29) + +/* SPDIF Transmitter(TX) enable */ +#define TEGRA_SPDIF_CTRL_TX_EN (1 << 28) + +/* Transmit Channel status */ +#define TEGRA_SPDIF_CTRL_TC_EN (1 << 27) + +/* Transmit user Data */ +#define TEGRA_SPDIF_CTRL_TU_EN (1 << 26) + +/* Interrupt on transmit error */ +#define TEGRA_SPDIF_CTRL_IE_TXE (1 << 25) + +/* Interrupt on receive error */ +#define TEGRA_SPDIF_CTRL_IE_RXE (1 << 24) + +/* Interrupt on invalid preamble */ +#define TEGRA_SPDIF_CTRL_IE_P (1 << 23) + +/* Interrupt on "B" preamble */ +#define TEGRA_SPDIF_CTRL_IE_B (1 << 22) + +/* Interrupt when block of channel status received */ +#define TEGRA_SPDIF_CTRL_IE_C (1 << 21) + +/* Interrupt when a valid information unit (IU) is received */ +#define TEGRA_SPDIF_CTRL_IE_U (1 << 20) + +/* Interrupt when RX user FIFO attention level is reached */ +#define TEGRA_SPDIF_CTRL_QE_RU (1 << 19) + +/* Interrupt when TX user FIFO attention level is reached */ +#define TEGRA_SPDIF_CTRL_QE_TU (1 << 18) + +/* Interrupt when RX data FIFO attention level is reached */ +#define TEGRA_SPDIF_CTRL_QE_RX (1 << 17) + +/* Interrupt when TX data FIFO attention level is reached */ +#define TEGRA_SPDIF_CTRL_QE_TX (1 << 16) + +/* Loopback test mode enable */ +#define TEGRA_SPDIF_CTRL_LBK_EN (1 << 15) + +/* + * Pack data mode: + * 0 = Single data (16 bit needs to be padded to match the + * interface data bit size). + * 1 = Packeted left/right channel data into a single word. + */ +#define TEGRA_SPDIF_CTRL_PACK (1 << 14) + +/* + * 00 = 16bit data + * 01 = 20bit data + * 10 = 24bit data + * 11 = raw data + */ +#define TEGRA_SPDIF_BIT_MODE_16BIT 0 +#define TEGRA_SPDIF_BIT_MODE_20BIT 1 +#define TEGRA_SPDIF_BIT_MODE_24BIT 2 +#define TEGRA_SPDIF_BIT_MODE_RAW 3 + +#define TEGRA_SPDIF_CTRL_BIT_MODE_SHIFT 12 +#define TEGRA_SPDIF_CTRL_BIT_MODE_MASK (3 << TEGRA_SPDIF_CTRL_BIT_MODE_SHIFT) +#define TEGRA_SPDIF_CTRL_BIT_MODE_16BIT (TEGRA_SPDIF_BIT_MODE_16BIT << TEGRA_SPDIF_CTRL_BIT_MODE_SHIFT) +#define TEGRA_SPDIF_CTRL_BIT_MODE_20BIT (TEGRA_SPDIF_BIT_MODE_20BIT << TEGRA_SPDIF_CTRL_BIT_MODE_SHIFT) +#define TEGRA_SPDIF_CTRL_BIT_MODE_24BIT (TEGRA_SPDIF_BIT_MODE_24BIT << TEGRA_SPDIF_CTRL_BIT_MODE_SHIFT) +#define TEGRA_SPDIF_CTRL_BIT_MODE_RAW (TEGRA_SPDIF_BIT_MODE_RAW << TEGRA_SPDIF_CTRL_BIT_MODE_SHIFT) + +/* Fields in TEGRA_SPDIF_STATUS */ + +/* + * Note: IS_P, IS_B, IS_C, and IS_U are sticky bits. Software must + * write a 1 to the corresponding bit location to clear the status. + */ + +/* + * Receiver(RX) shifter is busy receiving data. + * This bit is asserted when the receiver first locked onto the + * preamble of the data stream after RX_EN is asserted. This bit is + * deasserted when either, + * (a) the end of a frame is reached after RX_EN is deeasserted, or + * (b) the SPDIF data stream becomes inactive. + */ +#define TEGRA_SPDIF_STATUS_RX_BSY (1 << 29) + +/* + * Transmitter(TX) shifter is busy transmitting data. + * This bit is asserted when TX_EN is asserted. + * This bit is deasserted when the end of a frame is reached after + * TX_EN is deasserted. + */ +#define TEGRA_SPDIF_STATUS_TX_BSY (1 << 28) + +/* + * TX is busy shifting out channel status. + * This bit is asserted when both TX_EN and TC_EN are asserted and + * data from CH_STA_TX_A register is loaded into the internal shifter. + * This bit is deasserted when either, + * (a) the end of a frame is reached after TX_EN is deasserted, or + * (b) CH_STA_TX_F register is loaded into the internal shifter. + */ +#define TEGRA_SPDIF_STATUS_TC_BSY (1 << 27) + +/* + * TX User data FIFO busy. + * This bit is asserted when TX_EN and TXU_EN are asserted and + * there's data in the TX user FIFO. This bit is deassert when either, + * (a) the end of a frame is reached after TX_EN is deasserted, or + * (b) there's no data left in the TX user FIFO. + */ +#define TEGRA_SPDIF_STATUS_TU_BSY (1 << 26) + +/* TX FIFO Underrun error status */ +#define TEGRA_SPDIF_STATUS_TX_ERR (1 << 25) + +/* RX FIFO Overrun error status */ +#define TEGRA_SPDIF_STATUS_RX_ERR (1 << 24) + +/* Preamble status: 0=Preamble OK, 1=bad/missing preamble */ +#define TEGRA_SPDIF_STATUS_IS_P (1 << 23) + +/* B-preamble detection status: 0=not detected, 1=B-preamble detected */ +#define TEGRA_SPDIF_STATUS_IS_B (1 << 22) + +/* + * RX channel block data receive status: + * 0=entire block not recieved yet. + * 1=received entire block of channel status, + */ +#define TEGRA_SPDIF_STATUS_IS_C (1 << 21) + +/* RX User Data Valid flag: 1=valid IU detected, 0 = no IU detected. */ +#define TEGRA_SPDIF_STATUS_IS_U (1 << 20) + +/* + * RX User FIFO Status: + * 1=attention level reached, 0=attention level not reached. + */ +#define TEGRA_SPDIF_STATUS_QS_RU (1 << 19) + +/* + * TX User FIFO Status: + * 1=attention level reached, 0=attention level not reached. + */ +#define TEGRA_SPDIF_STATUS_QS_TU (1 << 18) + +/* + * RX Data FIFO Status: + * 1=attention level reached, 0=attention level not reached. + */ +#define TEGRA_SPDIF_STATUS_QS_RX (1 << 17) + +/* + * TX Data FIFO Status: + * 1=attention level reached, 0=attention level not reached. + */ +#define TEGRA_SPDIF_STATUS_QS_TX (1 << 16) + +/* Fields in TEGRA_SPDIF_STROBE_CTRL */ + +/* + * Indicates the approximate number of detected SPDIFIN clocks within a + * bi-phase period. + */ +#define TEGRA_SPDIF_STROBE_CTRL_PERIOD_SHIFT 16 +#define TEGRA_SPDIF_STROBE_CTRL_PERIOD_MASK (0xff << TEGRA_SPDIF_STROBE_CTRL_PERIOD_SHIFT) + +/* Data strobe mode: 0=Auto-locked 1=Manual locked */ +#define TEGRA_SPDIF_STROBE_CTRL_STROBE (1 << 15) + +/* + * Manual data strobe time within the bi-phase clock period (in terms of + * the number of over-sampling clocks). + */ +#define TEGRA_SPDIF_STROBE_CTRL_DATA_STROBES_SHIFT 8 +#define TEGRA_SPDIF_STROBE_CTRL_DATA_STROBES_MASK (0x1f << TEGRA_SPDIF_STROBE_CTRL_DATA_STROBES_SHIFT) + +/* + * Manual SPDIFIN bi-phase clock period (in terms of the number of + * over-sampling clocks). + */ +#define TEGRA_SPDIF_STROBE_CTRL_CLOCK_PERIOD_SHIFT 0 +#define TEGRA_SPDIF_STROBE_CTRL_CLOCK_PERIOD_MASK (0x3f << TEGRA_SPDIF_STROBE_CTRL_CLOCK_PERIOD_SHIFT) + +/* Fields in SPDIF_DATA_FIFO_CSR */ + +/* Clear Receiver User FIFO (RX USR.FIFO) */ +#define TEGRA_SPDIF_DATA_FIFO_CSR_RU_CLR (1 << 31) + +#define TEGRA_SPDIF_FIFO_ATN_LVL_U_ONE_SLOT 0 +#define TEGRA_SPDIF_FIFO_ATN_LVL_U_TWO_SLOTS 1 +#define TEGRA_SPDIF_FIFO_ATN_LVL_U_THREE_SLOTS 2 +#define TEGRA_SPDIF_FIFO_ATN_LVL_U_FOUR_SLOTS 3 + +/* RU FIFO attention level */ +#define TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_SHIFT 29 +#define TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_MASK \ + (0x3 << TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_SHIFT) +#define TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_RU1_WORD_FULL \ + (TEGRA_SPDIF_FIFO_ATN_LVL_U_ONE_SLOT << TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_SHIFT) +#define TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_RU2_WORD_FULL \ + (TEGRA_SPDIF_FIFO_ATN_LVL_U_TWO_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_SHIFT) +#define TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_RU3_WORD_FULL \ + (TEGRA_SPDIF_FIFO_ATN_LVL_U_THREE_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_SHIFT) +#define TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_RU4_WORD_FULL \ + (TEGRA_SPDIF_FIFO_ATN_LVL_U_FOUR_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_SHIFT) + +/* Number of RX USR.FIFO levels with valid data. */ +#define TEGRA_SPDIF_DATA_FIFO_CSR_RU_FULL_COUNT_SHIFT 24 +#define TEGRA_SPDIF_DATA_FIFO_CSR_RU_FULL_COUNT_MASK (0x1f << TEGRA_SPDIF_DATA_FIFO_CSR_RU_FULL_COUNT_SHIFT) + +/* Clear Transmitter User FIFO (TX USR.FIFO) */ +#define TEGRA_SPDIF_DATA_FIFO_CSR_TU_CLR (1 << 23) + +/* TU FIFO attention level */ +#define TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_SHIFT 21 +#define TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_MASK \ + (0x3 << TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_SHIFT) +#define TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_TU1_WORD_FULL \ + (TEGRA_SPDIF_FIFO_ATN_LVL_U_ONE_SLOT << TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_SHIFT) +#define TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_TU2_WORD_FULL \ + (TEGRA_SPDIF_FIFO_ATN_LVL_U_TWO_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_SHIFT) +#define TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_TU3_WORD_FULL \ + (TEGRA_SPDIF_FIFO_ATN_LVL_U_THREE_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_SHIFT) +#define TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_TU4_WORD_FULL \ + (TEGRA_SPDIF_FIFO_ATN_LVL_U_FOUR_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_SHIFT) + +/* Number of TX USR.FIFO levels that could be filled. */ +#define TEGRA_SPDIF_DATA_FIFO_CSR_TU_EMPTY_COUNT_SHIFT 16 +#define TEGRA_SPDIF_DATA_FIFO_CSR_TU_EMPTY_COUNT_MASK (0x1f << SPDIF_DATA_FIFO_CSR_TU_EMPTY_COUNT_SHIFT) + +/* Clear Receiver Data FIFO (RX DATA.FIFO) */ +#define TEGRA_SPDIF_DATA_FIFO_CSR_RX_CLR (1 << 15) + +#define TEGRA_SPDIF_FIFO_ATN_LVL_D_ONE_SLOT 0 +#define TEGRA_SPDIF_FIFO_ATN_LVL_D_FOUR_SLOTS 1 +#define TEGRA_SPDIF_FIFO_ATN_LVL_D_EIGHT_SLOTS 2 +#define TEGRA_SPDIF_FIFO_ATN_LVL_D_TWELVE_SLOTS 3 + +/* RU FIFO attention level */ +#define TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_SHIFT 13 +#define TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_MASK \ + (0x3 << TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_SHIFT) +#define TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_RU1_WORD_FULL \ + (TEGRA_SPDIF_FIFO_ATN_LVL_D_ONE_SLOT << TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_SHIFT) +#define TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_RU4_WORD_FULL \ + (TEGRA_SPDIF_FIFO_ATN_LVL_D_FOUR_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_SHIFT) +#define TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_RU8_WORD_FULL \ + (TEGRA_SPDIF_FIFO_ATN_LVL_D_EIGHT_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_SHIFT) +#define TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_RU12_WORD_FULL \ + (TEGRA_SPDIF_FIFO_ATN_LVL_D_TWELVE_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_SHIFT) + +/* Number of RX DATA.FIFO levels with valid data. */ +#define TEGRA_SPDIF_DATA_FIFO_CSR_RX_FULL_COUNT_SHIFT 8 +#define TEGRA_SPDIF_DATA_FIFO_CSR_RX_FULL_COUNT_MASK (0x1f << TEGRA_SPDIF_DATA_FIFO_CSR_RX_FULL_COUNT_SHIFT) + +/* Clear Transmitter Data FIFO (TX DATA.FIFO) */ +#define TEGRA_SPDIF_DATA_FIFO_CSR_TX_CLR (1 << 7) + +/* TU FIFO attention level */ +#define TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_SHIFT 5 +#define TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_MASK \ + (0x3 << TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_SHIFT) +#define TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_TU1_WORD_FULL \ + (TEGRA_SPDIF_FIFO_ATN_LVL_D_ONE_SLOT << TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_SHIFT) +#define TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_TU4_WORD_FULL \ + (TEGRA_SPDIF_FIFO_ATN_LVL_D_FOUR_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_SHIFT) +#define TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_TU8_WORD_FULL \ + (TEGRA_SPDIF_FIFO_ATN_LVL_D_EIGHT_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_SHIFT) +#define TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_TU12_WORD_FULL \ + (TEGRA_SPDIF_FIFO_ATN_LVL_D_TWELVE_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_SHIFT) + +/* Number of TX DATA.FIFO levels that could be filled. */ +#define TEGRA_SPDIF_DATA_FIFO_CSR_TX_EMPTY_COUNT_SHIFT 0 +#define TEGRA_SPDIF_DATA_FIFO_CSR_TX_EMPTY_COUNT_MASK (0x1f << SPDIF_DATA_FIFO_CSR_TX_EMPTY_COUNT_SHIFT) + +/* Fields in TEGRA_SPDIF_DATA_OUT */ + +/* + * This register has 5 different formats: + * 16-bit (BIT_MODE=00, PACK=0) + * 20-bit (BIT_MODE=01, PACK=0) + * 24-bit (BIT_MODE=10, PACK=0) + * raw (BIT_MODE=11, PACK=0) + * 16-bit packed (BIT_MODE=00, PACK=1) + */ + +#define TEGRA_SPDIF_DATA_OUT_DATA_16_SHIFT 0 +#define TEGRA_SPDIF_DATA_OUT_DATA_16_MASK (0xffff << TEGRA_SPDIF_DATA_OUT_DATA_16_SHIFT) + +#define TEGRA_SPDIF_DATA_OUT_DATA_20_SHIFT 0 +#define TEGRA_SPDIF_DATA_OUT_DATA_20_MASK (0xfffff << TEGRA_SPDIF_DATA_OUT_DATA_20_SHIFT) + +#define TEGRA_SPDIF_DATA_OUT_DATA_24_SHIFT 0 +#define TEGRA_SPDIF_DATA_OUT_DATA_24_MASK (0xffffff << TEGRA_SPDIF_DATA_OUT_DATA_24_SHIFT) + +#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_P (1 << 31) +#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_C (1 << 30) +#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_U (1 << 29) +#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_V (1 << 28) + +#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_DATA_SHIFT 8 +#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_DATA_MASK (0xfffff << TEGRA_SPDIF_DATA_OUT_DATA_RAW_DATA_SHIFT) + +#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_AUX_SHIFT 4 +#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_AUX_MASK (0xf << TEGRA_SPDIF_DATA_OUT_DATA_RAW_AUX_SHIFT) + +#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_PREAMBLE_SHIFT 0 +#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_PREAMBLE_MASK (0xf << TEGRA_SPDIF_DATA_OUT_DATA_RAW_PREAMBLE_SHIFT) + +#define TEGRA_SPDIF_DATA_OUT_DATA_16_PACKED_RIGHT_SHIFT 16 +#define TEGRA_SPDIF_DATA_OUT_DATA_16_PACKED_RIGHT_MASK (0xffff << TEGRA_SPDIF_DATA_OUT_DATA_16_PACKED_RIGHT_SHIFT) + +#define TEGRA_SPDIF_DATA_OUT_DATA_16_PACKED_LEFT_SHIFT 0 +#define TEGRA_SPDIF_DATA_OUT_DATA_16_PACKED_LEFT_MASK (0xffff << TEGRA_SPDIF_DATA_OUT_DATA_16_PACKED_LEFT_SHIFT) + +/* Fields in TEGRA_SPDIF_DATA_IN */ + +/* + * This register has 5 different formats: + * 16-bit (BIT_MODE=00, PACK=0) + * 20-bit (BIT_MODE=01, PACK=0) + * 24-bit (BIT_MODE=10, PACK=0) + * raw (BIT_MODE=11, PACK=0) + * 16-bit packed (BIT_MODE=00, PACK=1) + * + * Bits 31:24 are common to all modes except 16-bit packed + */ + +#define TEGRA_SPDIF_DATA_IN_DATA_P (1 << 31) +#define TEGRA_SPDIF_DATA_IN_DATA_C (1 << 30) +#define TEGRA_SPDIF_DATA_IN_DATA_U (1 << 29) +#define TEGRA_SPDIF_DATA_IN_DATA_V (1 << 28) + +#define TEGRA_SPDIF_DATA_IN_DATA_PREAMBLE_SHIFT 24 +#define TEGRA_SPDIF_DATA_IN_DATA_PREAMBLE_MASK (0xf << TEGRA_SPDIF_DATA_IN_DATA_PREAMBLE_SHIFT) + +#define TEGRA_SPDIF_DATA_IN_DATA_16_SHIFT 0 +#define TEGRA_SPDIF_DATA_IN_DATA_16_MASK (0xffff << TEGRA_SPDIF_DATA_IN_DATA_16_SHIFT) + +#define TEGRA_SPDIF_DATA_IN_DATA_20_SHIFT 0 +#define TEGRA_SPDIF_DATA_IN_DATA_20_MASK (0xfffff << TEGRA_SPDIF_DATA_IN_DATA_20_SHIFT) + +#define TEGRA_SPDIF_DATA_IN_DATA_24_SHIFT 0 +#define TEGRA_SPDIF_DATA_IN_DATA_24_MASK (0xffffff << TEGRA_SPDIF_DATA_IN_DATA_24_SHIFT) + +#define TEGRA_SPDIF_DATA_IN_DATA_RAW_DATA_SHIFT 8 +#define TEGRA_SPDIF_DATA_IN_DATA_RAW_DATA_MASK (0xfffff << TEGRA_SPDIF_DATA_IN_DATA_RAW_DATA_SHIFT) + +#define TEGRA_SPDIF_DATA_IN_DATA_RAW_AUX_SHIFT 4 +#define TEGRA_SPDIF_DATA_IN_DATA_RAW_AUX_MASK (0xf << TEGRA_SPDIF_DATA_IN_DATA_RAW_AUX_SHIFT) + +#define TEGRA_SPDIF_DATA_IN_DATA_RAW_PREAMBLE_SHIFT 0 +#define TEGRA_SPDIF_DATA_IN_DATA_RAW_PREAMBLE_MASK (0xf << TEGRA_SPDIF_DATA_IN_DATA_RAW_PREAMBLE_SHIFT) + +#define TEGRA_SPDIF_DATA_IN_DATA_16_PACKED_RIGHT_SHIFT 16 +#define TEGRA_SPDIF_DATA_IN_DATA_16_PACKED_RIGHT_MASK (0xffff << TEGRA_SPDIF_DATA_IN_DATA_16_PACKED_RIGHT_SHIFT) + +#define TEGRA_SPDIF_DATA_IN_DATA_16_PACKED_LEFT_SHIFT 0 +#define TEGRA_SPDIF_DATA_IN_DATA_16_PACKED_LEFT_MASK (0xffff << TEGRA_SPDIF_DATA_IN_DATA_16_PACKED_LEFT_SHIFT) + +/* Fields in TEGRA_SPDIF_CH_STA_RX_A */ +/* Fields in TEGRA_SPDIF_CH_STA_RX_B */ +/* Fields in TEGRA_SPDIF_CH_STA_RX_C */ +/* Fields in TEGRA_SPDIF_CH_STA_RX_D */ +/* Fields in TEGRA_SPDIF_CH_STA_RX_E */ +/* Fields in TEGRA_SPDIF_CH_STA_RX_F */ + +/* + * The 6-word receive channel data page buffer holds a block (192 frames) of + * channel status information. The order of receive is from LSB to MSB + * bit, and from CH_STA_RX_A to CH_STA_RX_F then back to CH_STA_RX_A. + */ + +/* Fields in TEGRA_SPDIF_CH_STA_TX_A */ +/* Fields in TEGRA_SPDIF_CH_STA_TX_B */ +/* Fields in TEGRA_SPDIF_CH_STA_TX_C */ +/* Fields in TEGRA_SPDIF_CH_STA_TX_D */ +/* Fields in TEGRA_SPDIF_CH_STA_TX_E */ +/* Fields in TEGRA_SPDIF_CH_STA_TX_F */ + +/* + * The 6-word transmit channel data page buffer holds a block (192 frames) of + * channel status information. The order of transmission is from LSB to MSB + * bit, and from CH_STA_TX_A to CH_STA_TX_F then back to CH_STA_TX_A. + */ + +/* Fields in TEGRA_SPDIF_USR_STA_RX_A */ + +/* + * This 4-word deep FIFO receives user FIFO field information. The order of + * receive is from LSB to MSB bit. + */ + +/* Fields in TEGRA_SPDIF_USR_DAT_TX_A */ + +/* + * This 4-word deep FIFO transmits user FIFO field information. The order of + * transmission is from LSB to MSB bit. + */ + +struct tegra_spdif { + struct clk *clk_spdif_out; + struct tegra_pcm_dma_params capture_dma_data; + struct tegra_pcm_dma_params playback_dma_data; + void __iomem *regs; + struct dentry *debug; + u32 reg_ctrl; +}; + +#endif diff --git a/sound/soc/tegra/tegra_das.c b/sound/soc/tegra/tegra_das.c deleted file mode 100644 index 3b3c1ba4d235..000000000000 --- a/sound/soc/tegra/tegra_das.c +++ /dev/null @@ -1,261 +0,0 @@ -/* - * tegra_das.c - Tegra DAS driver - * - * Author: Stephen Warren - * Copyright (C) 2010 - NVIDIA, Inc. - * - * This program is free software; you can redistribute it and/or - * modify it under the terms of the GNU General Public License - * version 2 as published by the Free Software Foundation. - * - * This program is distributed in the hope that it will be useful, but - * WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA - * 02110-1301 USA - * - */ - -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include "tegra_das.h" - -#define DRV_NAME "tegra-das" - -static struct tegra_das *das; - -static inline void tegra_das_write(u32 reg, u32 val) -{ - __raw_writel(val, das->regs + reg); -} - -static inline u32 tegra_das_read(u32 reg) -{ - return __raw_readl(das->regs + reg); -} - -int tegra_das_connect_dap_to_dac(int dap, int dac) -{ - u32 addr; - u32 reg; - - if (!das) - return -ENODEV; - - addr = TEGRA_DAS_DAP_CTRL_SEL + - (dap * TEGRA_DAS_DAP_CTRL_SEL_STRIDE); - reg = dac << TEGRA_DAS_DAP_CTRL_SEL_DAP_CTRL_SEL_P; - - tegra_das_write(addr, reg); - - return 0; -} -EXPORT_SYMBOL_GPL(tegra_das_connect_dap_to_dac); - -int tegra_das_connect_dap_to_dap(int dap, int otherdap, int master, - int sdata1rx, int sdata2rx) -{ - u32 addr; - u32 reg; - - if (!das) - return -ENODEV; - - addr = TEGRA_DAS_DAP_CTRL_SEL + - (dap * TEGRA_DAS_DAP_CTRL_SEL_STRIDE); - reg = otherdap << TEGRA_DAS_DAP_CTRL_SEL_DAP_CTRL_SEL_P | - !!sdata2rx << TEGRA_DAS_DAP_CTRL_SEL_DAP_SDATA2_TX_RX_P | - !!sdata1rx << TEGRA_DAS_DAP_CTRL_SEL_DAP_SDATA1_TX_RX_P | - !!master << TEGRA_DAS_DAP_CTRL_SEL_DAP_MS_SEL_P; - - tegra_das_write(addr, reg); - - return 0; -} -EXPORT_SYMBOL_GPL(tegra_das_connect_dap_to_dap); - -int tegra_das_connect_dac_to_dap(int dac, int dap) -{ - u32 addr; - u32 reg; - - if (!das) - return -ENODEV; - - addr = TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL + - (dac * TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_STRIDE); - reg = dap << TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_CLK_SEL_P | - dap << TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA1_SEL_P | - dap << TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA2_SEL_P; - - tegra_das_write(addr, reg); - - return 0; -} -EXPORT_SYMBOL_GPL(tegra_das_connect_dac_to_dap); - -#ifdef CONFIG_DEBUG_FS -static int tegra_das_show(struct seq_file *s, void *unused) -{ - int i; - u32 addr; - u32 reg; - - for (i = 0; i < TEGRA_DAS_DAP_CTRL_SEL_COUNT; i++) { - addr = TEGRA_DAS_DAP_CTRL_SEL + - (i * TEGRA_DAS_DAP_CTRL_SEL_STRIDE); - reg = tegra_das_read(addr); - seq_printf(s, "TEGRA_DAS_DAP_CTRL_SEL[%d] = %08x\n", i, reg); - } - - for (i = 0; i < TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_COUNT; i++) { - addr = TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL + - (i * TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_STRIDE); - reg = tegra_das_read(addr); - seq_printf(s, "TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL[%d] = %08x\n", - i, reg); - } - - return 0; -} - -static int tegra_das_debug_open(struct inode *inode, struct file *file) -{ - return single_open(file, tegra_das_show, inode->i_private); -} - -static const struct file_operations tegra_das_debug_fops = { - .open = tegra_das_debug_open, - .read = seq_read, - .llseek = seq_lseek, - .release = single_release, -}; - -static void tegra_das_debug_add(struct tegra_das *das) -{ - das->debug = debugfs_create_file(DRV_NAME, S_IRUGO, - snd_soc_debugfs_root, das, - &tegra_das_debug_fops); -} - -static void tegra_das_debug_remove(struct tegra_das *das) -{ - if (das->debug) - debugfs_remove(das->debug); -} -#else -static inline void tegra_das_debug_add(struct tegra_das *das) -{ -} - -static inline void tegra_das_debug_remove(struct tegra_das *das) -{ -} -#endif - -static int __devinit tegra_das_probe(struct platform_device *pdev) -{ - struct resource *res, *region; - int ret = 0; - - if (das) - return -ENODEV; - - das = devm_kzalloc(&pdev->dev, sizeof(struct tegra_das), GFP_KERNEL); - if (!das) { - dev_err(&pdev->dev, "Can't allocate tegra_das\n"); - ret = -ENOMEM; - goto err; - } - das->dev = &pdev->dev; - - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!res) { - dev_err(&pdev->dev, "No memory resource\n"); - ret = -ENODEV; - goto err; - } - - region = devm_request_mem_region(&pdev->dev, res->start, - resource_size(res), pdev->name); - if (!region) { - dev_err(&pdev->dev, "Memory region already claimed\n"); - ret = -EBUSY; - goto err; - } - - das->regs = devm_ioremap(&pdev->dev, res->start, resource_size(res)); - if (!das->regs) { - dev_err(&pdev->dev, "ioremap failed\n"); - ret = -ENOMEM; - goto err; - } - - ret = tegra_das_connect_dap_to_dac(TEGRA_DAS_DAP_ID_1, - TEGRA_DAS_DAP_SEL_DAC1); - if (ret) { - dev_err(&pdev->dev, "Can't set up DAS DAP connection\n"); - goto err; - } - ret = tegra_das_connect_dac_to_dap(TEGRA_DAS_DAC_ID_1, - TEGRA_DAS_DAC_SEL_DAP1); - if (ret) { - dev_err(&pdev->dev, "Can't set up DAS DAC connection\n"); - goto err; - } - - tegra_das_debug_add(das); - - platform_set_drvdata(pdev, das); - - return 0; - -err: - das = NULL; - return ret; -} - -static int __devexit tegra_das_remove(struct platform_device *pdev) -{ - if (!das) - return -ENODEV; - - tegra_das_debug_remove(das); - - das = NULL; - - return 0; -} - -static const struct of_device_id tegra_das_of_match[] __devinitconst = { - { .compatible = "nvidia,tegra20-das", }, - {}, -}; - -static struct platform_driver tegra_das_driver = { - .probe = tegra_das_probe, - .remove = __devexit_p(tegra_das_remove), - .driver = { - .name = DRV_NAME, - .owner = THIS_MODULE, - .of_match_table = tegra_das_of_match, - }, -}; -module_platform_driver(tegra_das_driver); - -MODULE_AUTHOR("Stephen Warren "); -MODULE_DESCRIPTION("Tegra DAS driver"); -MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:" DRV_NAME); -MODULE_DEVICE_TABLE(of, tegra_das_of_match); diff --git a/sound/soc/tegra/tegra_das.h b/sound/soc/tegra/tegra_das.h deleted file mode 100644 index 1810cd1e7f0c..000000000000 --- a/sound/soc/tegra/tegra_das.h +++ /dev/null @@ -1,135 +0,0 @@ -/* - * tegra_das.h - Definitions for Tegra DAS driver - * - * Author: Stephen Warren - * Copyright (C) 2010 - NVIDIA, Inc. - * - * This program is free software; you can redistribute it and/or - * modify it under the terms of the GNU General Public License - * version 2 as published by the Free Software Foundation. - * - * This program is distributed in the hope that it will be useful, but - * WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA - * 02110-1301 USA - * - */ - -#ifndef __TEGRA_DAS_H__ -#define __TEGRA_DAS_H__ - -/* Register TEGRA_DAS_DAP_CTRL_SEL */ -#define TEGRA_DAS_DAP_CTRL_SEL 0x00 -#define TEGRA_DAS_DAP_CTRL_SEL_COUNT 5 -#define TEGRA_DAS_DAP_CTRL_SEL_STRIDE 4 -#define TEGRA_DAS_DAP_CTRL_SEL_DAP_MS_SEL_P 31 -#define TEGRA_DAS_DAP_CTRL_SEL_DAP_MS_SEL_S 1 -#define TEGRA_DAS_DAP_CTRL_SEL_DAP_SDATA1_TX_RX_P 30 -#define TEGRA_DAS_DAP_CTRL_SEL_DAP_SDATA1_TX_RX_S 1 -#define TEGRA_DAS_DAP_CTRL_SEL_DAP_SDATA2_TX_RX_P 29 -#define TEGRA_DAS_DAP_CTRL_SEL_DAP_SDATA2_TX_RX_S 1 -#define TEGRA_DAS_DAP_CTRL_SEL_DAP_CTRL_SEL_P 0 -#define TEGRA_DAS_DAP_CTRL_SEL_DAP_CTRL_SEL_S 5 - -/* Values for field TEGRA_DAS_DAP_CTRL_SEL_DAP_CTRL_SEL */ -#define TEGRA_DAS_DAP_SEL_DAC1 0 -#define TEGRA_DAS_DAP_SEL_DAC2 1 -#define TEGRA_DAS_DAP_SEL_DAC3 2 -#define TEGRA_DAS_DAP_SEL_DAP1 16 -#define TEGRA_DAS_DAP_SEL_DAP2 17 -#define TEGRA_DAS_DAP_SEL_DAP3 18 -#define TEGRA_DAS_DAP_SEL_DAP4 19 -#define TEGRA_DAS_DAP_SEL_DAP5 20 - -/* Register TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL */ -#define TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL 0x40 -#define TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_COUNT 3 -#define TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_STRIDE 4 -#define TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA2_SEL_P 28 -#define TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA2_SEL_S 4 -#define TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA1_SEL_P 24 -#define TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA1_SEL_S 4 -#define TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_CLK_SEL_P 0 -#define TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_CLK_SEL_S 4 - -/* - * Values for: - * TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA2_SEL - * TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA1_SEL - * TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_CLK_SEL - */ -#define TEGRA_DAS_DAC_SEL_DAP1 0 -#define TEGRA_DAS_DAC_SEL_DAP2 1 -#define TEGRA_DAS_DAC_SEL_DAP3 2 -#define TEGRA_DAS_DAC_SEL_DAP4 3 -#define TEGRA_DAS_DAC_SEL_DAP5 4 - -/* - * Names/IDs of the DACs/DAPs. - */ - -#define TEGRA_DAS_DAP_ID_1 0 -#define TEGRA_DAS_DAP_ID_2 1 -#define TEGRA_DAS_DAP_ID_3 2 -#define TEGRA_DAS_DAP_ID_4 3 -#define TEGRA_DAS_DAP_ID_5 4 - -#define TEGRA_DAS_DAC_ID_1 0 -#define TEGRA_DAS_DAC_ID_2 1 -#define TEGRA_DAS_DAC_ID_3 2 - -struct tegra_das { - struct device *dev; - void __iomem *regs; - struct dentry *debug; -}; - -/* - * Terminology: - * DAS: Digital audio switch (HW module controlled by this driver) - * DAP: Digital audio port (port/pins on Tegra device) - * DAC: Digital audio controller (e.g. I2S or AC97 controller elsewhere) - * - * The Tegra DAS is a mux/cross-bar which can connect each DAP to a specific - * DAC, or another DAP. When DAPs are connected, one must be the master and - * one the slave. Each DAC allows selection of a specific DAP for input, to - * cater for the case where N DAPs are connected to 1 DAC for broadcast - * output. - * - * This driver is dumb; no attempt is made to ensure that a valid routing - * configuration is programmed. - */ - -/* - * Connect a DAP to to a DAC - * dap_id: DAP to connect: TEGRA_DAS_DAP_ID_* - * dac_sel: DAC to connect to: TEGRA_DAS_DAP_SEL_DAC* - */ -extern int tegra_das_connect_dap_to_dac(int dap_id, int dac_sel); - -/* - * Connect a DAP to to another DAP - * dap_id: DAP to connect: TEGRA_DAS_DAP_ID_* - * other_dap_sel: DAP to connect to: TEGRA_DAS_DAP_SEL_DAP* - * master: Is this DAP the master (1) or slave (0) - * sdata1rx: Is this DAP's SDATA1 pin RX (1) or TX (0) - * sdata2rx: Is this DAP's SDATA2 pin RX (1) or TX (0) - */ -extern int tegra_das_connect_dap_to_dap(int dap_id, int other_dap_sel, - int master, int sdata1rx, - int sdata2rx); - -/* - * Connect a DAC's input to a DAP - * (DAC outputs are selected by the DAP) - * dac_id: DAC ID to connect: TEGRA_DAS_DAC_ID_* - * dap_sel: DAP to receive input from: TEGRA_DAS_DAC_SEL_DAP* - */ -extern int tegra_das_connect_dac_to_dap(int dac_id, int dap_sel); - -#endif diff --git a/sound/soc/tegra/tegra_i2s.c b/sound/soc/tegra/tegra_i2s.c deleted file mode 100644 index 2d32b8ce5f2e..000000000000 --- a/sound/soc/tegra/tegra_i2s.c +++ /dev/null @@ -1,458 +0,0 @@ -/* - * tegra_i2s.c - Tegra I2S driver - * - * Author: Stephen Warren - * Copyright (C) 2010,2012 - NVIDIA, Inc. - * - * Based on code copyright/by: - * - * Copyright (c) 2009-2010, NVIDIA Corporation. - * Scott Peterson - * - * Copyright (C) 2010 Google, Inc. - * Iliyan Malchev - * - * This program is free software; you can redistribute it and/or - * modify it under the terms of the GNU General Public License - * version 2 as published by the Free Software Foundation. - * - * This program is distributed in the hope that it will be useful, but - * WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA - * 02110-1301 USA - * - */ - -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include - -#include "tegra_i2s.h" - -#define DRV_NAME "tegra-i2s" - -static inline void tegra_i2s_write(struct tegra_i2s *i2s, u32 reg, u32 val) -{ - __raw_writel(val, i2s->regs + reg); -} - -static inline u32 tegra_i2s_read(struct tegra_i2s *i2s, u32 reg) -{ - return __raw_readl(i2s->regs + reg); -} - -#ifdef CONFIG_DEBUG_FS -static int tegra_i2s_show(struct seq_file *s, void *unused) -{ -#define REG(r) { r, #r } - static const struct { - int offset; - const char *name; - } regs[] = { - REG(TEGRA_I2S_CTRL), - REG(TEGRA_I2S_STATUS), - REG(TEGRA_I2S_TIMING), - REG(TEGRA_I2S_FIFO_SCR), - REG(TEGRA_I2S_PCM_CTRL), - REG(TEGRA_I2S_NW_CTRL), - REG(TEGRA_I2S_TDM_CTRL), - REG(TEGRA_I2S_TDM_TX_RX_CTRL), - }; -#undef REG - - struct tegra_i2s *i2s = s->private; - int i; - - for (i = 0; i < ARRAY_SIZE(regs); i++) { - u32 val = tegra_i2s_read(i2s, regs[i].offset); - seq_printf(s, "%s = %08x\n", regs[i].name, val); - } - - return 0; -} - -static int tegra_i2s_debug_open(struct inode *inode, struct file *file) -{ - return single_open(file, tegra_i2s_show, inode->i_private); -} - -static const struct file_operations tegra_i2s_debug_fops = { - .open = tegra_i2s_debug_open, - .read = seq_read, - .llseek = seq_lseek, - .release = single_release, -}; - -static void tegra_i2s_debug_add(struct tegra_i2s *i2s) -{ - i2s->debug = debugfs_create_file(i2s->dai.name, S_IRUGO, - snd_soc_debugfs_root, i2s, - &tegra_i2s_debug_fops); -} - -static void tegra_i2s_debug_remove(struct tegra_i2s *i2s) -{ - if (i2s->debug) - debugfs_remove(i2s->debug); -} -#else -static inline void tegra_i2s_debug_add(struct tegra_i2s *i2s, int id) -{ -} - -static inline void tegra_i2s_debug_remove(struct tegra_i2s *i2s) -{ -} -#endif - -static int tegra_i2s_set_fmt(struct snd_soc_dai *dai, - unsigned int fmt) -{ - struct tegra_i2s *i2s = snd_soc_dai_get_drvdata(dai); - - switch (fmt & SND_SOC_DAIFMT_INV_MASK) { - case SND_SOC_DAIFMT_NB_NF: - break; - default: - return -EINVAL; - } - - i2s->reg_ctrl &= ~TEGRA_I2S_CTRL_MASTER_ENABLE; - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBS_CFS: - i2s->reg_ctrl |= TEGRA_I2S_CTRL_MASTER_ENABLE; - break; - case SND_SOC_DAIFMT_CBM_CFM: - break; - default: - return -EINVAL; - } - - i2s->reg_ctrl &= ~(TEGRA_I2S_CTRL_BIT_FORMAT_MASK | - TEGRA_I2S_CTRL_LRCK_MASK); - switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { - case SND_SOC_DAIFMT_DSP_A: - i2s->reg_ctrl |= TEGRA_I2S_CTRL_BIT_FORMAT_DSP; - i2s->reg_ctrl |= TEGRA_I2S_CTRL_LRCK_L_LOW; - break; - case SND_SOC_DAIFMT_DSP_B: - i2s->reg_ctrl |= TEGRA_I2S_CTRL_BIT_FORMAT_DSP; - i2s->reg_ctrl |= TEGRA_I2S_CTRL_LRCK_R_LOW; - break; - case SND_SOC_DAIFMT_I2S: - i2s->reg_ctrl |= TEGRA_I2S_CTRL_BIT_FORMAT_I2S; - i2s->reg_ctrl |= TEGRA_I2S_CTRL_LRCK_L_LOW; - break; - case SND_SOC_DAIFMT_RIGHT_J: - i2s->reg_ctrl |= TEGRA_I2S_CTRL_BIT_FORMAT_RJM; - i2s->reg_ctrl |= TEGRA_I2S_CTRL_LRCK_L_LOW; - break; - case SND_SOC_DAIFMT_LEFT_J: - i2s->reg_ctrl |= TEGRA_I2S_CTRL_BIT_FORMAT_LJM; - i2s->reg_ctrl |= TEGRA_I2S_CTRL_LRCK_L_LOW; - break; - default: - return -EINVAL; - } - - return 0; -} - -static int tegra_i2s_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) -{ - struct device *dev = substream->pcm->card->dev; - struct tegra_i2s *i2s = snd_soc_dai_get_drvdata(dai); - u32 reg; - int ret, sample_size, srate, i2sclock, bitcnt; - - i2s->reg_ctrl &= ~TEGRA_I2S_CTRL_BIT_SIZE_MASK; - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: - i2s->reg_ctrl |= TEGRA_I2S_CTRL_BIT_SIZE_16; - sample_size = 16; - break; - case SNDRV_PCM_FORMAT_S24_LE: - i2s->reg_ctrl |= TEGRA_I2S_CTRL_BIT_SIZE_24; - sample_size = 24; - break; - case SNDRV_PCM_FORMAT_S32_LE: - i2s->reg_ctrl |= TEGRA_I2S_CTRL_BIT_SIZE_32; - sample_size = 32; - break; - default: - return -EINVAL; - } - - srate = params_rate(params); - - /* Final "* 2" required by Tegra hardware */ - i2sclock = srate * params_channels(params) * sample_size * 2; - - ret = clk_set_rate(i2s->clk_i2s, i2sclock); - if (ret) { - dev_err(dev, "Can't set I2S clock rate: %d\n", ret); - return ret; - } - - bitcnt = (i2sclock / (2 * srate)) - 1; - if (bitcnt < 0 || bitcnt > TEGRA_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US) - return -EINVAL; - reg = bitcnt << TEGRA_I2S_TIMING_CHANNEL_BIT_COUNT_SHIFT; - - if (i2sclock % (2 * srate)) - reg |= TEGRA_I2S_TIMING_NON_SYM_ENABLE; - - clk_enable(i2s->clk_i2s); - - tegra_i2s_write(i2s, TEGRA_I2S_TIMING, reg); - - tegra_i2s_write(i2s, TEGRA_I2S_FIFO_SCR, - TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_FOUR_SLOTS | - TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_FOUR_SLOTS); - - clk_disable(i2s->clk_i2s); - - return 0; -} - -static void tegra_i2s_start_playback(struct tegra_i2s *i2s) -{ - i2s->reg_ctrl |= TEGRA_I2S_CTRL_FIFO1_ENABLE; - tegra_i2s_write(i2s, TEGRA_I2S_CTRL, i2s->reg_ctrl); -} - -static void tegra_i2s_stop_playback(struct tegra_i2s *i2s) -{ - i2s->reg_ctrl &= ~TEGRA_I2S_CTRL_FIFO1_ENABLE; - tegra_i2s_write(i2s, TEGRA_I2S_CTRL, i2s->reg_ctrl); -} - -static void tegra_i2s_start_capture(struct tegra_i2s *i2s) -{ - i2s->reg_ctrl |= TEGRA_I2S_CTRL_FIFO2_ENABLE; - tegra_i2s_write(i2s, TEGRA_I2S_CTRL, i2s->reg_ctrl); -} - -static void tegra_i2s_stop_capture(struct tegra_i2s *i2s) -{ - i2s->reg_ctrl &= ~TEGRA_I2S_CTRL_FIFO2_ENABLE; - tegra_i2s_write(i2s, TEGRA_I2S_CTRL, i2s->reg_ctrl); -} - -static int tegra_i2s_trigger(struct snd_pcm_substream *substream, int cmd, - struct snd_soc_dai *dai) -{ - struct tegra_i2s *i2s = snd_soc_dai_get_drvdata(dai); - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - case SNDRV_PCM_TRIGGER_RESUME: - clk_enable(i2s->clk_i2s); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - tegra_i2s_start_playback(i2s); - else - tegra_i2s_start_capture(i2s); - break; - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - case SNDRV_PCM_TRIGGER_SUSPEND: - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - tegra_i2s_stop_playback(i2s); - else - tegra_i2s_stop_capture(i2s); - clk_disable(i2s->clk_i2s); - break; - default: - return -EINVAL; - } - - return 0; -} - -static int tegra_i2s_probe(struct snd_soc_dai *dai) -{ - struct tegra_i2s *i2s = snd_soc_dai_get_drvdata(dai); - - dai->capture_dma_data = &i2s->capture_dma_data; - dai->playback_dma_data = &i2s->playback_dma_data; - - return 0; -} - -static const struct snd_soc_dai_ops tegra_i2s_dai_ops = { - .set_fmt = tegra_i2s_set_fmt, - .hw_params = tegra_i2s_hw_params, - .trigger = tegra_i2s_trigger, -}; - -static const struct snd_soc_dai_driver tegra_i2s_dai_template = { - .probe = tegra_i2s_probe, - .playback = { - .channels_min = 2, - .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_96000, - .formats = SNDRV_PCM_FMTBIT_S16_LE, - }, - .capture = { - .channels_min = 2, - .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_96000, - .formats = SNDRV_PCM_FMTBIT_S16_LE, - }, - .ops = &tegra_i2s_dai_ops, - .symmetric_rates = 1, -}; - -static __devinit int tegra_i2s_platform_probe(struct platform_device *pdev) -{ - struct tegra_i2s *i2s; - struct resource *mem, *memregion, *dmareq; - u32 of_dma[2]; - u32 dma_ch; - int ret; - - i2s = devm_kzalloc(&pdev->dev, sizeof(struct tegra_i2s), GFP_KERNEL); - if (!i2s) { - dev_err(&pdev->dev, "Can't allocate tegra_i2s\n"); - ret = -ENOMEM; - goto err; - } - dev_set_drvdata(&pdev->dev, i2s); - - i2s->dai = tegra_i2s_dai_template; - i2s->dai.name = dev_name(&pdev->dev); - - i2s->clk_i2s = clk_get(&pdev->dev, NULL); - if (IS_ERR(i2s->clk_i2s)) { - dev_err(&pdev->dev, "Can't retrieve i2s clock\n"); - ret = PTR_ERR(i2s->clk_i2s); - goto err; - } - - mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!mem) { - dev_err(&pdev->dev, "No memory resource\n"); - ret = -ENODEV; - goto err_clk_put; - } - - dmareq = platform_get_resource(pdev, IORESOURCE_DMA, 0); - if (!dmareq) { - if (of_property_read_u32_array(pdev->dev.of_node, - "nvidia,dma-request-selector", - of_dma, 2) < 0) { - dev_err(&pdev->dev, "No DMA resource\n"); - ret = -ENODEV; - goto err_clk_put; - } - dma_ch = of_dma[1]; - } else { - dma_ch = dmareq->start; - } - - memregion = devm_request_mem_region(&pdev->dev, mem->start, - resource_size(mem), DRV_NAME); - if (!memregion) { - dev_err(&pdev->dev, "Memory region already claimed\n"); - ret = -EBUSY; - goto err_clk_put; - } - - i2s->regs = devm_ioremap(&pdev->dev, mem->start, resource_size(mem)); - if (!i2s->regs) { - dev_err(&pdev->dev, "ioremap failed\n"); - ret = -ENOMEM; - goto err_clk_put; - } - - i2s->capture_dma_data.addr = mem->start + TEGRA_I2S_FIFO2; - i2s->capture_dma_data.wrap = 4; - i2s->capture_dma_data.width = 32; - i2s->capture_dma_data.req_sel = dma_ch; - - i2s->playback_dma_data.addr = mem->start + TEGRA_I2S_FIFO1; - i2s->playback_dma_data.wrap = 4; - i2s->playback_dma_data.width = 32; - i2s->playback_dma_data.req_sel = dma_ch; - - i2s->reg_ctrl = TEGRA_I2S_CTRL_FIFO_FORMAT_PACKED; - - ret = snd_soc_register_dai(&pdev->dev, &i2s->dai); - if (ret) { - dev_err(&pdev->dev, "Could not register DAI: %d\n", ret); - ret = -ENOMEM; - goto err_clk_put; - } - - ret = tegra_pcm_platform_register(&pdev->dev); - if (ret) { - dev_err(&pdev->dev, "Could not register PCM: %d\n", ret); - goto err_unregister_dai; - } - - tegra_i2s_debug_add(i2s); - - return 0; - -err_unregister_dai: - snd_soc_unregister_dai(&pdev->dev); -err_clk_put: - clk_put(i2s->clk_i2s); -err: - return ret; -} - -static int __devexit tegra_i2s_platform_remove(struct platform_device *pdev) -{ - struct tegra_i2s *i2s = dev_get_drvdata(&pdev->dev); - - tegra_pcm_platform_unregister(&pdev->dev); - snd_soc_unregister_dai(&pdev->dev); - - tegra_i2s_debug_remove(i2s); - - clk_put(i2s->clk_i2s); - - return 0; -} - -static const struct of_device_id tegra_i2s_of_match[] __devinitconst = { - { .compatible = "nvidia,tegra20-i2s", }, - {}, -}; - -static struct platform_driver tegra_i2s_driver = { - .driver = { - .name = DRV_NAME, - .owner = THIS_MODULE, - .of_match_table = tegra_i2s_of_match, - }, - .probe = tegra_i2s_platform_probe, - .remove = __devexit_p(tegra_i2s_platform_remove), -}; -module_platform_driver(tegra_i2s_driver); - -MODULE_AUTHOR("Stephen Warren "); -MODULE_DESCRIPTION("Tegra I2S ASoC driver"); -MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:" DRV_NAME); -MODULE_DEVICE_TABLE(of, tegra_i2s_of_match); diff --git a/sound/soc/tegra/tegra_i2s.h b/sound/soc/tegra/tegra_i2s.h deleted file mode 100644 index c08e019c0260..000000000000 --- a/sound/soc/tegra/tegra_i2s.h +++ /dev/null @@ -1,165 +0,0 @@ -/* - * tegra_i2s.h - Definitions for Tegra I2S driver - * - * Author: Stephen Warren - * Copyright (C) 2010 - NVIDIA, Inc. - * - * Based on code copyright/by: - * - * Copyright (c) 2009-2010, NVIDIA Corporation. - * Scott Peterson - * - * Copyright (C) 2010 Google, Inc. - * Iliyan Malchev - * - * This program is free software; you can redistribute it and/or - * modify it under the terms of the GNU General Public License - * version 2 as published by the Free Software Foundation. - * - * This program is distributed in the hope that it will be useful, but - * WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA - * 02110-1301 USA - * - */ - -#ifndef __TEGRA_I2S_H__ -#define __TEGRA_I2S_H__ - -#include "tegra_pcm.h" - -/* Register offsets from TEGRA_I2S1_BASE and TEGRA_I2S2_BASE */ - -#define TEGRA_I2S_CTRL 0x00 -#define TEGRA_I2S_STATUS 0x04 -#define TEGRA_I2S_TIMING 0x08 -#define TEGRA_I2S_FIFO_SCR 0x0c -#define TEGRA_I2S_PCM_CTRL 0x10 -#define TEGRA_I2S_NW_CTRL 0x14 -#define TEGRA_I2S_TDM_CTRL 0x20 -#define TEGRA_I2S_TDM_TX_RX_CTRL 0x24 -#define TEGRA_I2S_FIFO1 0x40 -#define TEGRA_I2S_FIFO2 0x80 - -/* Fields in TEGRA_I2S_CTRL */ - -#define TEGRA_I2S_CTRL_FIFO2_TX_ENABLE (1 << 30) -#define TEGRA_I2S_CTRL_FIFO1_ENABLE (1 << 29) -#define TEGRA_I2S_CTRL_FIFO2_ENABLE (1 << 28) -#define TEGRA_I2S_CTRL_FIFO1_RX_ENABLE (1 << 27) -#define TEGRA_I2S_CTRL_FIFO_LPBK_ENABLE (1 << 26) -#define TEGRA_I2S_CTRL_MASTER_ENABLE (1 << 25) - -#define TEGRA_I2S_LRCK_LEFT_LOW 0 -#define TEGRA_I2S_LRCK_RIGHT_LOW 1 - -#define TEGRA_I2S_CTRL_LRCK_SHIFT 24 -#define TEGRA_I2S_CTRL_LRCK_MASK (1 << TEGRA_I2S_CTRL_LRCK_SHIFT) -#define TEGRA_I2S_CTRL_LRCK_L_LOW (TEGRA_I2S_LRCK_LEFT_LOW << TEGRA_I2S_CTRL_LRCK_SHIFT) -#define TEGRA_I2S_CTRL_LRCK_R_LOW (TEGRA_I2S_LRCK_RIGHT_LOW << TEGRA_I2S_CTRL_LRCK_SHIFT) - -#define TEGRA_I2S_BIT_FORMAT_I2S 0 -#define TEGRA_I2S_BIT_FORMAT_RJM 1 -#define TEGRA_I2S_BIT_FORMAT_LJM 2 -#define TEGRA_I2S_BIT_FORMAT_DSP 3 - -#define TEGRA_I2S_CTRL_BIT_FORMAT_SHIFT 10 -#define TEGRA_I2S_CTRL_BIT_FORMAT_MASK (3 << TEGRA_I2S_CTRL_BIT_FORMAT_SHIFT) -#define TEGRA_I2S_CTRL_BIT_FORMAT_I2S (TEGRA_I2S_BIT_FORMAT_I2S << TEGRA_I2S_CTRL_BIT_FORMAT_SHIFT) -#define TEGRA_I2S_CTRL_BIT_FORMAT_RJM (TEGRA_I2S_BIT_FORMAT_RJM << TEGRA_I2S_CTRL_BIT_FORMAT_SHIFT) -#define TEGRA_I2S_CTRL_BIT_FORMAT_LJM (TEGRA_I2S_BIT_FORMAT_LJM << TEGRA_I2S_CTRL_BIT_FORMAT_SHIFT) -#define TEGRA_I2S_CTRL_BIT_FORMAT_DSP (TEGRA_I2S_BIT_FORMAT_DSP << TEGRA_I2S_CTRL_BIT_FORMAT_SHIFT) - -#define TEGRA_I2S_BIT_SIZE_16 0 -#define TEGRA_I2S_BIT_SIZE_20 1 -#define TEGRA_I2S_BIT_SIZE_24 2 -#define TEGRA_I2S_BIT_SIZE_32 3 - -#define TEGRA_I2S_CTRL_BIT_SIZE_SHIFT 8 -#define TEGRA_I2S_CTRL_BIT_SIZE_MASK (3 << TEGRA_I2S_CTRL_BIT_SIZE_SHIFT) -#define TEGRA_I2S_CTRL_BIT_SIZE_16 (TEGRA_I2S_BIT_SIZE_16 << TEGRA_I2S_CTRL_BIT_SIZE_SHIFT) -#define TEGRA_I2S_CTRL_BIT_SIZE_20 (TEGRA_I2S_BIT_SIZE_20 << TEGRA_I2S_CTRL_BIT_SIZE_SHIFT) -#define TEGRA_I2S_CTRL_BIT_SIZE_24 (TEGRA_I2S_BIT_SIZE_24 << TEGRA_I2S_CTRL_BIT_SIZE_SHIFT) -#define TEGRA_I2S_CTRL_BIT_SIZE_32 (TEGRA_I2S_BIT_SIZE_32 << TEGRA_I2S_CTRL_BIT_SIZE_SHIFT) - -#define TEGRA_I2S_FIFO_16_LSB 0 -#define TEGRA_I2S_FIFO_20_LSB 1 -#define TEGRA_I2S_FIFO_24_LSB 2 -#define TEGRA_I2S_FIFO_32 3 -#define TEGRA_I2S_FIFO_PACKED 7 - -#define TEGRA_I2S_CTRL_FIFO_FORMAT_SHIFT 4 -#define TEGRA_I2S_CTRL_FIFO_FORMAT_MASK (7 << TEGRA_I2S_CTRL_FIFO_FORMAT_SHIFT) -#define TEGRA_I2S_CTRL_FIFO_FORMAT_16_LSB (TEGRA_I2S_FIFO_16_LSB << TEGRA_I2S_CTRL_FIFO_FORMAT_SHIFT) -#define TEGRA_I2S_CTRL_FIFO_FORMAT_20_LSB (TEGRA_I2S_FIFO_20_LSB << TEGRA_I2S_CTRL_FIFO_FORMAT_SHIFT) -#define TEGRA_I2S_CTRL_FIFO_FORMAT_24_LSB (TEGRA_I2S_FIFO_24_LSB << TEGRA_I2S_CTRL_FIFO_FORMAT_SHIFT) -#define TEGRA_I2S_CTRL_FIFO_FORMAT_32 (TEGRA_I2S_FIFO_32 << TEGRA_I2S_CTRL_FIFO_FORMAT_SHIFT) -#define TEGRA_I2S_CTRL_FIFO_FORMAT_PACKED (TEGRA_I2S_FIFO_PACKED << TEGRA_I2S_CTRL_FIFO_FORMAT_SHIFT) - -#define TEGRA_I2S_CTRL_IE_FIFO1_ERR (1 << 3) -#define TEGRA_I2S_CTRL_IE_FIFO2_ERR (1 << 2) -#define TEGRA_I2S_CTRL_QE_FIFO1 (1 << 1) -#define TEGRA_I2S_CTRL_QE_FIFO2 (1 << 0) - -/* Fields in TEGRA_I2S_STATUS */ - -#define TEGRA_I2S_STATUS_FIFO1_RDY (1 << 31) -#define TEGRA_I2S_STATUS_FIFO2_RDY (1 << 30) -#define TEGRA_I2S_STATUS_FIFO1_BSY (1 << 29) -#define TEGRA_I2S_STATUS_FIFO2_BSY (1 << 28) -#define TEGRA_I2S_STATUS_FIFO1_ERR (1 << 3) -#define TEGRA_I2S_STATUS_FIFO2_ERR (1 << 2) -#define TEGRA_I2S_STATUS_QS_FIFO1 (1 << 1) -#define TEGRA_I2S_STATUS_QS_FIFO2 (1 << 0) - -/* Fields in TEGRA_I2S_TIMING */ - -#define TEGRA_I2S_TIMING_NON_SYM_ENABLE (1 << 12) -#define TEGRA_I2S_TIMING_CHANNEL_BIT_COUNT_SHIFT 0 -#define TEGRA_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US 0x7fff -#define TEGRA_I2S_TIMING_CHANNEL_BIT_COUNT_MASK (TEGRA_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US << TEGRA_I2S_TIMING_CHANNEL_BIT_COUNT_SHIFT) - -/* Fields in TEGRA_I2S_FIFO_SCR */ - -#define TEGRA_I2S_FIFO_SCR_FIFO2_FULL_EMPTY_COUNT_SHIFT 24 -#define TEGRA_I2S_FIFO_SCR_FIFO1_FULL_EMPTY_COUNT_SHIFT 16 -#define TEGRA_I2S_FIFO_SCR_FIFO_FULL_EMPTY_COUNT_MASK 0x3f - -#define TEGRA_I2S_FIFO_SCR_FIFO2_CLR (1 << 12) -#define TEGRA_I2S_FIFO_SCR_FIFO1_CLR (1 << 8) - -#define TEGRA_I2S_FIFO_ATN_LVL_ONE_SLOT 0 -#define TEGRA_I2S_FIFO_ATN_LVL_FOUR_SLOTS 1 -#define TEGRA_I2S_FIFO_ATN_LVL_EIGHT_SLOTS 2 -#define TEGRA_I2S_FIFO_ATN_LVL_TWELVE_SLOTS 3 - -#define TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_SHIFT 4 -#define TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_MASK (3 << TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_SHIFT) -#define TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_ONE_SLOT (TEGRA_I2S_FIFO_ATN_LVL_ONE_SLOT << TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_SHIFT) -#define TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_FOUR_SLOTS (TEGRA_I2S_FIFO_ATN_LVL_FOUR_SLOTS << TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_SHIFT) -#define TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_EIGHT_SLOTS (TEGRA_I2S_FIFO_ATN_LVL_EIGHT_SLOTS << TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_SHIFT) -#define TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_TWELVE_SLOTS (TEGRA_I2S_FIFO_ATN_LVL_TWELVE_SLOTS << TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_SHIFT) - -#define TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_SHIFT 0 -#define TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_MASK (3 << TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_SHIFT) -#define TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_ONE_SLOT (TEGRA_I2S_FIFO_ATN_LVL_ONE_SLOT << TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_SHIFT) -#define TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_FOUR_SLOTS (TEGRA_I2S_FIFO_ATN_LVL_FOUR_SLOTS << TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_SHIFT) -#define TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_EIGHT_SLOTS (TEGRA_I2S_FIFO_ATN_LVL_EIGHT_SLOTS << TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_SHIFT) -#define TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_TWELVE_SLOTS (TEGRA_I2S_FIFO_ATN_LVL_TWELVE_SLOTS << TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_SHIFT) - -struct tegra_i2s { - struct snd_soc_dai_driver dai; - struct clk *clk_i2s; - struct tegra_pcm_dma_params capture_dma_data; - struct tegra_pcm_dma_params playback_dma_data; - void __iomem *regs; - struct dentry *debug; - u32 reg_ctrl; -}; - -#endif diff --git a/sound/soc/tegra/tegra_spdif.c b/sound/soc/tegra/tegra_spdif.c deleted file mode 100644 index 3426633e9147..000000000000 --- a/sound/soc/tegra/tegra_spdif.c +++ /dev/null @@ -1,365 +0,0 @@ -/* - * tegra_spdif.c - Tegra SPDIF driver - * - * Author: Stephen Warren - * Copyright (C) 2011-2012 - NVIDIA, Inc. - * - * This program is free software; you can redistribute it and/or - * modify it under the terms of the GNU General Public License - * version 2 as published by the Free Software Foundation. - * - * This program is distributed in the hope that it will be useful, but - * WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA - * 02110-1301 USA - * - */ - -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include - -#include "tegra_spdif.h" - -#define DRV_NAME "tegra-spdif" - -static inline void tegra_spdif_write(struct tegra_spdif *spdif, u32 reg, - u32 val) -{ - __raw_writel(val, spdif->regs + reg); -} - -static inline u32 tegra_spdif_read(struct tegra_spdif *spdif, u32 reg) -{ - return __raw_readl(spdif->regs + reg); -} - -#ifdef CONFIG_DEBUG_FS -static int tegra_spdif_show(struct seq_file *s, void *unused) -{ -#define REG(r) { r, #r } - static const struct { - int offset; - const char *name; - } regs[] = { - REG(TEGRA_SPDIF_CTRL), - REG(TEGRA_SPDIF_STATUS), - REG(TEGRA_SPDIF_STROBE_CTRL), - REG(TEGRA_SPDIF_DATA_FIFO_CSR), - REG(TEGRA_SPDIF_CH_STA_RX_A), - REG(TEGRA_SPDIF_CH_STA_RX_B), - REG(TEGRA_SPDIF_CH_STA_RX_C), - REG(TEGRA_SPDIF_CH_STA_RX_D), - REG(TEGRA_SPDIF_CH_STA_RX_E), - REG(TEGRA_SPDIF_CH_STA_RX_F), - REG(TEGRA_SPDIF_CH_STA_TX_A), - REG(TEGRA_SPDIF_CH_STA_TX_B), - REG(TEGRA_SPDIF_CH_STA_TX_C), - REG(TEGRA_SPDIF_CH_STA_TX_D), - REG(TEGRA_SPDIF_CH_STA_TX_E), - REG(TEGRA_SPDIF_CH_STA_TX_F), - }; -#undef REG - - struct tegra_spdif *spdif = s->private; - int i; - - for (i = 0; i < ARRAY_SIZE(regs); i++) { - u32 val = tegra_spdif_read(spdif, regs[i].offset); - seq_printf(s, "%s = %08x\n", regs[i].name, val); - } - - return 0; -} - -static int tegra_spdif_debug_open(struct inode *inode, struct file *file) -{ - return single_open(file, tegra_spdif_show, inode->i_private); -} - -static const struct file_operations tegra_spdif_debug_fops = { - .open = tegra_spdif_debug_open, - .read = seq_read, - .llseek = seq_lseek, - .release = single_release, -}; - -static void tegra_spdif_debug_add(struct tegra_spdif *spdif) -{ - spdif->debug = debugfs_create_file(DRV_NAME, S_IRUGO, - snd_soc_debugfs_root, spdif, - &tegra_spdif_debug_fops); -} - -static void tegra_spdif_debug_remove(struct tegra_spdif *spdif) -{ - if (spdif->debug) - debugfs_remove(spdif->debug); -} -#else -static inline void tegra_spdif_debug_add(struct tegra_spdif *spdif) -{ -} - -static inline void tegra_spdif_debug_remove(struct tegra_spdif *spdif) -{ -} -#endif - -static int tegra_spdif_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) -{ - struct device *dev = substream->pcm->card->dev; - struct tegra_spdif *spdif = snd_soc_dai_get_drvdata(dai); - int ret, spdifclock; - - spdif->reg_ctrl &= ~TEGRA_SPDIF_CTRL_PACK; - spdif->reg_ctrl &= ~TEGRA_SPDIF_CTRL_BIT_MODE_MASK; - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: - spdif->reg_ctrl |= TEGRA_SPDIF_CTRL_PACK; - spdif->reg_ctrl |= TEGRA_SPDIF_CTRL_BIT_MODE_16BIT; - break; - default: - return -EINVAL; - } - - switch (params_rate(params)) { - case 32000: - spdifclock = 4096000; - break; - case 44100: - spdifclock = 5644800; - break; - case 48000: - spdifclock = 6144000; - break; - case 88200: - spdifclock = 11289600; - break; - case 96000: - spdifclock = 12288000; - break; - case 176400: - spdifclock = 22579200; - break; - case 192000: - spdifclock = 24576000; - break; - default: - return -EINVAL; - } - - ret = clk_set_rate(spdif->clk_spdif_out, spdifclock); - if (ret) { - dev_err(dev, "Can't set SPDIF clock rate: %d\n", ret); - return ret; - } - - return 0; -} - -static void tegra_spdif_start_playback(struct tegra_spdif *spdif) -{ - spdif->reg_ctrl |= TEGRA_SPDIF_CTRL_TX_EN; - tegra_spdif_write(spdif, TEGRA_SPDIF_CTRL, spdif->reg_ctrl); -} - -static void tegra_spdif_stop_playback(struct tegra_spdif *spdif) -{ - spdif->reg_ctrl &= ~TEGRA_SPDIF_CTRL_TX_EN; - tegra_spdif_write(spdif, TEGRA_SPDIF_CTRL, spdif->reg_ctrl); -} - -static int tegra_spdif_trigger(struct snd_pcm_substream *substream, int cmd, - struct snd_soc_dai *dai) -{ - struct tegra_spdif *spdif = snd_soc_dai_get_drvdata(dai); - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - case SNDRV_PCM_TRIGGER_RESUME: - clk_enable(spdif->clk_spdif_out); - tegra_spdif_start_playback(spdif); - break; - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - case SNDRV_PCM_TRIGGER_SUSPEND: - tegra_spdif_stop_playback(spdif); - clk_disable(spdif->clk_spdif_out); - break; - default: - return -EINVAL; - } - - return 0; -} - -static int tegra_spdif_probe(struct snd_soc_dai *dai) -{ - struct tegra_spdif *spdif = snd_soc_dai_get_drvdata(dai); - - dai->capture_dma_data = NULL; - dai->playback_dma_data = &spdif->playback_dma_data; - - return 0; -} - -static const struct snd_soc_dai_ops tegra_spdif_dai_ops = { - .hw_params = tegra_spdif_hw_params, - .trigger = tegra_spdif_trigger, -}; - -static struct snd_soc_dai_driver tegra_spdif_dai = { - .name = DRV_NAME, - .probe = tegra_spdif_probe, - .playback = { - .channels_min = 2, - .channels_max = 2, - .rates = SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | - SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S16_LE, - }, - .ops = &tegra_spdif_dai_ops, -}; - -static __devinit int tegra_spdif_platform_probe(struct platform_device *pdev) -{ - struct tegra_spdif *spdif; - struct resource *mem, *memregion, *dmareq; - int ret; - - spdif = kzalloc(sizeof(struct tegra_spdif), GFP_KERNEL); - if (!spdif) { - dev_err(&pdev->dev, "Can't allocate tegra_spdif\n"); - ret = -ENOMEM; - goto exit; - } - dev_set_drvdata(&pdev->dev, spdif); - - spdif->clk_spdif_out = clk_get(&pdev->dev, "spdif_out"); - if (IS_ERR(spdif->clk_spdif_out)) { - pr_err("Can't retrieve spdif clock\n"); - ret = PTR_ERR(spdif->clk_spdif_out); - goto err_free; - } - - mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!mem) { - dev_err(&pdev->dev, "No memory resource\n"); - ret = -ENODEV; - goto err_clk_put; - } - - dmareq = platform_get_resource(pdev, IORESOURCE_DMA, 0); - if (!dmareq) { - dev_err(&pdev->dev, "No DMA resource\n"); - ret = -ENODEV; - goto err_clk_put; - } - - memregion = request_mem_region(mem->start, resource_size(mem), - DRV_NAME); - if (!memregion) { - dev_err(&pdev->dev, "Memory region already claimed\n"); - ret = -EBUSY; - goto err_clk_put; - } - - spdif->regs = ioremap(mem->start, resource_size(mem)); - if (!spdif->regs) { - dev_err(&pdev->dev, "ioremap failed\n"); - ret = -ENOMEM; - goto err_release; - } - - spdif->playback_dma_data.addr = mem->start + TEGRA_SPDIF_DATA_OUT; - spdif->playback_dma_data.wrap = 4; - spdif->playback_dma_data.width = 32; - spdif->playback_dma_data.req_sel = dmareq->start; - - ret = snd_soc_register_dai(&pdev->dev, &tegra_spdif_dai); - if (ret) { - dev_err(&pdev->dev, "Could not register DAI: %d\n", ret); - ret = -ENOMEM; - goto err_unmap; - } - - ret = tegra_pcm_platform_register(&pdev->dev); - if (ret) { - dev_err(&pdev->dev, "Could not register PCM: %d\n", ret); - goto err_unregister_dai; - } - - tegra_spdif_debug_add(spdif); - - return 0; - -err_unregister_dai: - snd_soc_unregister_dai(&pdev->dev); -err_unmap: - iounmap(spdif->regs); -err_release: - release_mem_region(mem->start, resource_size(mem)); -err_clk_put: - clk_put(spdif->clk_spdif_out); -err_free: - kfree(spdif); -exit: - return ret; -} - -static int __devexit tegra_spdif_platform_remove(struct platform_device *pdev) -{ - struct tegra_spdif *spdif = dev_get_drvdata(&pdev->dev); - struct resource *res; - - tegra_pcm_platform_unregister(&pdev->dev); - snd_soc_unregister_dai(&pdev->dev); - - tegra_spdif_debug_remove(spdif); - - iounmap(spdif->regs); - - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - release_mem_region(res->start, resource_size(res)); - - clk_put(spdif->clk_spdif_out); - - kfree(spdif); - - return 0; -} - -static struct platform_driver tegra_spdif_driver = { - .driver = { - .name = DRV_NAME, - .owner = THIS_MODULE, - }, - .probe = tegra_spdif_platform_probe, - .remove = __devexit_p(tegra_spdif_platform_remove), -}; - -module_platform_driver(tegra_spdif_driver); - -MODULE_AUTHOR("Stephen Warren "); -MODULE_DESCRIPTION("Tegra SPDIF ASoC driver"); -MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:" DRV_NAME); diff --git a/sound/soc/tegra/tegra_spdif.h b/sound/soc/tegra/tegra_spdif.h deleted file mode 100644 index f5fc7124aa7b..000000000000 --- a/sound/soc/tegra/tegra_spdif.h +++ /dev/null @@ -1,472 +0,0 @@ -/* - * tegra_spdif.h - Definitions for Tegra SPDIF driver - * - * Author: Stephen Warren - * Copyright (C) 2011 - NVIDIA, Inc. - * - * Based on code copyright/by: - * Copyright (c) 2008-2009, NVIDIA Corporation - * - * This program is free software; you can redistribute it and/or - * modify it under the terms of the GNU General Public License - * version 2 as published by the Free Software Foundation. - * - * This program is distributed in the hope that it will be useful, but - * WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA - * 02110-1301 USA - * - */ - -#ifndef __TEGRA_SPDIF_H__ -#define __TEGRA_SPDIF_H__ - -#include "tegra_pcm.h" - -/* Offsets from TEGRA_SPDIF_BASE */ - -#define TEGRA_SPDIF_CTRL 0x0 -#define TEGRA_SPDIF_STATUS 0x4 -#define TEGRA_SPDIF_STROBE_CTRL 0x8 -#define TEGRA_SPDIF_DATA_FIFO_CSR 0x0C -#define TEGRA_SPDIF_DATA_OUT 0x40 -#define TEGRA_SPDIF_DATA_IN 0x80 -#define TEGRA_SPDIF_CH_STA_RX_A 0x100 -#define TEGRA_SPDIF_CH_STA_RX_B 0x104 -#define TEGRA_SPDIF_CH_STA_RX_C 0x108 -#define TEGRA_SPDIF_CH_STA_RX_D 0x10C -#define TEGRA_SPDIF_CH_STA_RX_E 0x110 -#define TEGRA_SPDIF_CH_STA_RX_F 0x114 -#define TEGRA_SPDIF_CH_STA_TX_A 0x140 -#define TEGRA_SPDIF_CH_STA_TX_B 0x144 -#define TEGRA_SPDIF_CH_STA_TX_C 0x148 -#define TEGRA_SPDIF_CH_STA_TX_D 0x14C -#define TEGRA_SPDIF_CH_STA_TX_E 0x150 -#define TEGRA_SPDIF_CH_STA_TX_F 0x154 -#define TEGRA_SPDIF_USR_STA_RX_A 0x180 -#define TEGRA_SPDIF_USR_DAT_TX_A 0x1C0 - -/* Fields in TEGRA_SPDIF_CTRL */ - -/* Start capturing from 0=right, 1=left channel */ -#define TEGRA_SPDIF_CTRL_CAP_LC (1 << 30) - -/* SPDIF receiver(RX) enable */ -#define TEGRA_SPDIF_CTRL_RX_EN (1 << 29) - -/* SPDIF Transmitter(TX) enable */ -#define TEGRA_SPDIF_CTRL_TX_EN (1 << 28) - -/* Transmit Channel status */ -#define TEGRA_SPDIF_CTRL_TC_EN (1 << 27) - -/* Transmit user Data */ -#define TEGRA_SPDIF_CTRL_TU_EN (1 << 26) - -/* Interrupt on transmit error */ -#define TEGRA_SPDIF_CTRL_IE_TXE (1 << 25) - -/* Interrupt on receive error */ -#define TEGRA_SPDIF_CTRL_IE_RXE (1 << 24) - -/* Interrupt on invalid preamble */ -#define TEGRA_SPDIF_CTRL_IE_P (1 << 23) - -/* Interrupt on "B" preamble */ -#define TEGRA_SPDIF_CTRL_IE_B (1 << 22) - -/* Interrupt when block of channel status received */ -#define TEGRA_SPDIF_CTRL_IE_C (1 << 21) - -/* Interrupt when a valid information unit (IU) is received */ -#define TEGRA_SPDIF_CTRL_IE_U (1 << 20) - -/* Interrupt when RX user FIFO attention level is reached */ -#define TEGRA_SPDIF_CTRL_QE_RU (1 << 19) - -/* Interrupt when TX user FIFO attention level is reached */ -#define TEGRA_SPDIF_CTRL_QE_TU (1 << 18) - -/* Interrupt when RX data FIFO attention level is reached */ -#define TEGRA_SPDIF_CTRL_QE_RX (1 << 17) - -/* Interrupt when TX data FIFO attention level is reached */ -#define TEGRA_SPDIF_CTRL_QE_TX (1 << 16) - -/* Loopback test mode enable */ -#define TEGRA_SPDIF_CTRL_LBK_EN (1 << 15) - -/* - * Pack data mode: - * 0 = Single data (16 bit needs to be padded to match the - * interface data bit size). - * 1 = Packeted left/right channel data into a single word. - */ -#define TEGRA_SPDIF_CTRL_PACK (1 << 14) - -/* - * 00 = 16bit data - * 01 = 20bit data - * 10 = 24bit data - * 11 = raw data - */ -#define TEGRA_SPDIF_BIT_MODE_16BIT 0 -#define TEGRA_SPDIF_BIT_MODE_20BIT 1 -#define TEGRA_SPDIF_BIT_MODE_24BIT 2 -#define TEGRA_SPDIF_BIT_MODE_RAW 3 - -#define TEGRA_SPDIF_CTRL_BIT_MODE_SHIFT 12 -#define TEGRA_SPDIF_CTRL_BIT_MODE_MASK (3 << TEGRA_SPDIF_CTRL_BIT_MODE_SHIFT) -#define TEGRA_SPDIF_CTRL_BIT_MODE_16BIT (TEGRA_SPDIF_BIT_MODE_16BIT << TEGRA_SPDIF_CTRL_BIT_MODE_SHIFT) -#define TEGRA_SPDIF_CTRL_BIT_MODE_20BIT (TEGRA_SPDIF_BIT_MODE_20BIT << TEGRA_SPDIF_CTRL_BIT_MODE_SHIFT) -#define TEGRA_SPDIF_CTRL_BIT_MODE_24BIT (TEGRA_SPDIF_BIT_MODE_24BIT << TEGRA_SPDIF_CTRL_BIT_MODE_SHIFT) -#define TEGRA_SPDIF_CTRL_BIT_MODE_RAW (TEGRA_SPDIF_BIT_MODE_RAW << TEGRA_SPDIF_CTRL_BIT_MODE_SHIFT) - -/* Fields in TEGRA_SPDIF_STATUS */ - -/* - * Note: IS_P, IS_B, IS_C, and IS_U are sticky bits. Software must - * write a 1 to the corresponding bit location to clear the status. - */ - -/* - * Receiver(RX) shifter is busy receiving data. - * This bit is asserted when the receiver first locked onto the - * preamble of the data stream after RX_EN is asserted. This bit is - * deasserted when either, - * (a) the end of a frame is reached after RX_EN is deeasserted, or - * (b) the SPDIF data stream becomes inactive. - */ -#define TEGRA_SPDIF_STATUS_RX_BSY (1 << 29) - -/* - * Transmitter(TX) shifter is busy transmitting data. - * This bit is asserted when TX_EN is asserted. - * This bit is deasserted when the end of a frame is reached after - * TX_EN is deasserted. - */ -#define TEGRA_SPDIF_STATUS_TX_BSY (1 << 28) - -/* - * TX is busy shifting out channel status. - * This bit is asserted when both TX_EN and TC_EN are asserted and - * data from CH_STA_TX_A register is loaded into the internal shifter. - * This bit is deasserted when either, - * (a) the end of a frame is reached after TX_EN is deasserted, or - * (b) CH_STA_TX_F register is loaded into the internal shifter. - */ -#define TEGRA_SPDIF_STATUS_TC_BSY (1 << 27) - -/* - * TX User data FIFO busy. - * This bit is asserted when TX_EN and TXU_EN are asserted and - * there's data in the TX user FIFO. This bit is deassert when either, - * (a) the end of a frame is reached after TX_EN is deasserted, or - * (b) there's no data left in the TX user FIFO. - */ -#define TEGRA_SPDIF_STATUS_TU_BSY (1 << 26) - -/* TX FIFO Underrun error status */ -#define TEGRA_SPDIF_STATUS_TX_ERR (1 << 25) - -/* RX FIFO Overrun error status */ -#define TEGRA_SPDIF_STATUS_RX_ERR (1 << 24) - -/* Preamble status: 0=Preamble OK, 1=bad/missing preamble */ -#define TEGRA_SPDIF_STATUS_IS_P (1 << 23) - -/* B-preamble detection status: 0=not detected, 1=B-preamble detected */ -#define TEGRA_SPDIF_STATUS_IS_B (1 << 22) - -/* - * RX channel block data receive status: - * 0=entire block not recieved yet. - * 1=received entire block of channel status, - */ -#define TEGRA_SPDIF_STATUS_IS_C (1 << 21) - -/* RX User Data Valid flag: 1=valid IU detected, 0 = no IU detected. */ -#define TEGRA_SPDIF_STATUS_IS_U (1 << 20) - -/* - * RX User FIFO Status: - * 1=attention level reached, 0=attention level not reached. - */ -#define TEGRA_SPDIF_STATUS_QS_RU (1 << 19) - -/* - * TX User FIFO Status: - * 1=attention level reached, 0=attention level not reached. - */ -#define TEGRA_SPDIF_STATUS_QS_TU (1 << 18) - -/* - * RX Data FIFO Status: - * 1=attention level reached, 0=attention level not reached. - */ -#define TEGRA_SPDIF_STATUS_QS_RX (1 << 17) - -/* - * TX Data FIFO Status: - * 1=attention level reached, 0=attention level not reached. - */ -#define TEGRA_SPDIF_STATUS_QS_TX (1 << 16) - -/* Fields in TEGRA_SPDIF_STROBE_CTRL */ - -/* - * Indicates the approximate number of detected SPDIFIN clocks within a - * bi-phase period. - */ -#define TEGRA_SPDIF_STROBE_CTRL_PERIOD_SHIFT 16 -#define TEGRA_SPDIF_STROBE_CTRL_PERIOD_MASK (0xff << TEGRA_SPDIF_STROBE_CTRL_PERIOD_SHIFT) - -/* Data strobe mode: 0=Auto-locked 1=Manual locked */ -#define TEGRA_SPDIF_STROBE_CTRL_STROBE (1 << 15) - -/* - * Manual data strobe time within the bi-phase clock period (in terms of - * the number of over-sampling clocks). - */ -#define TEGRA_SPDIF_STROBE_CTRL_DATA_STROBES_SHIFT 8 -#define TEGRA_SPDIF_STROBE_CTRL_DATA_STROBES_MASK (0x1f << TEGRA_SPDIF_STROBE_CTRL_DATA_STROBES_SHIFT) - -/* - * Manual SPDIFIN bi-phase clock period (in terms of the number of - * over-sampling clocks). - */ -#define TEGRA_SPDIF_STROBE_CTRL_CLOCK_PERIOD_SHIFT 0 -#define TEGRA_SPDIF_STROBE_CTRL_CLOCK_PERIOD_MASK (0x3f << TEGRA_SPDIF_STROBE_CTRL_CLOCK_PERIOD_SHIFT) - -/* Fields in SPDIF_DATA_FIFO_CSR */ - -/* Clear Receiver User FIFO (RX USR.FIFO) */ -#define TEGRA_SPDIF_DATA_FIFO_CSR_RU_CLR (1 << 31) - -#define TEGRA_SPDIF_FIFO_ATN_LVL_U_ONE_SLOT 0 -#define TEGRA_SPDIF_FIFO_ATN_LVL_U_TWO_SLOTS 1 -#define TEGRA_SPDIF_FIFO_ATN_LVL_U_THREE_SLOTS 2 -#define TEGRA_SPDIF_FIFO_ATN_LVL_U_FOUR_SLOTS 3 - -/* RU FIFO attention level */ -#define TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_SHIFT 29 -#define TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_MASK \ - (0x3 << TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_SHIFT) -#define TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_RU1_WORD_FULL \ - (TEGRA_SPDIF_FIFO_ATN_LVL_U_ONE_SLOT << TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_SHIFT) -#define TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_RU2_WORD_FULL \ - (TEGRA_SPDIF_FIFO_ATN_LVL_U_TWO_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_SHIFT) -#define TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_RU3_WORD_FULL \ - (TEGRA_SPDIF_FIFO_ATN_LVL_U_THREE_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_SHIFT) -#define TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_RU4_WORD_FULL \ - (TEGRA_SPDIF_FIFO_ATN_LVL_U_FOUR_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_SHIFT) - -/* Number of RX USR.FIFO levels with valid data. */ -#define TEGRA_SPDIF_DATA_FIFO_CSR_RU_FULL_COUNT_SHIFT 24 -#define TEGRA_SPDIF_DATA_FIFO_CSR_RU_FULL_COUNT_MASK (0x1f << TEGRA_SPDIF_DATA_FIFO_CSR_RU_FULL_COUNT_SHIFT) - -/* Clear Transmitter User FIFO (TX USR.FIFO) */ -#define TEGRA_SPDIF_DATA_FIFO_CSR_TU_CLR (1 << 23) - -/* TU FIFO attention level */ -#define TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_SHIFT 21 -#define TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_MASK \ - (0x3 << TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_SHIFT) -#define TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_TU1_WORD_FULL \ - (TEGRA_SPDIF_FIFO_ATN_LVL_U_ONE_SLOT << TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_SHIFT) -#define TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_TU2_WORD_FULL \ - (TEGRA_SPDIF_FIFO_ATN_LVL_U_TWO_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_SHIFT) -#define TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_TU3_WORD_FULL \ - (TEGRA_SPDIF_FIFO_ATN_LVL_U_THREE_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_SHIFT) -#define TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_TU4_WORD_FULL \ - (TEGRA_SPDIF_FIFO_ATN_LVL_U_FOUR_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_SHIFT) - -/* Number of TX USR.FIFO levels that could be filled. */ -#define TEGRA_SPDIF_DATA_FIFO_CSR_TU_EMPTY_COUNT_SHIFT 16 -#define TEGRA_SPDIF_DATA_FIFO_CSR_TU_EMPTY_COUNT_MASK (0x1f << SPDIF_DATA_FIFO_CSR_TU_EMPTY_COUNT_SHIFT) - -/* Clear Receiver Data FIFO (RX DATA.FIFO) */ -#define TEGRA_SPDIF_DATA_FIFO_CSR_RX_CLR (1 << 15) - -#define TEGRA_SPDIF_FIFO_ATN_LVL_D_ONE_SLOT 0 -#define TEGRA_SPDIF_FIFO_ATN_LVL_D_FOUR_SLOTS 1 -#define TEGRA_SPDIF_FIFO_ATN_LVL_D_EIGHT_SLOTS 2 -#define TEGRA_SPDIF_FIFO_ATN_LVL_D_TWELVE_SLOTS 3 - -/* RU FIFO attention level */ -#define TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_SHIFT 13 -#define TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_MASK \ - (0x3 << TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_SHIFT) -#define TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_RU1_WORD_FULL \ - (TEGRA_SPDIF_FIFO_ATN_LVL_D_ONE_SLOT << TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_SHIFT) -#define TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_RU4_WORD_FULL \ - (TEGRA_SPDIF_FIFO_ATN_LVL_D_FOUR_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_SHIFT) -#define TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_RU8_WORD_FULL \ - (TEGRA_SPDIF_FIFO_ATN_LVL_D_EIGHT_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_SHIFT) -#define TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_RU12_WORD_FULL \ - (TEGRA_SPDIF_FIFO_ATN_LVL_D_TWELVE_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_SHIFT) - -/* Number of RX DATA.FIFO levels with valid data. */ -#define TEGRA_SPDIF_DATA_FIFO_CSR_RX_FULL_COUNT_SHIFT 8 -#define TEGRA_SPDIF_DATA_FIFO_CSR_RX_FULL_COUNT_MASK (0x1f << TEGRA_SPDIF_DATA_FIFO_CSR_RX_FULL_COUNT_SHIFT) - -/* Clear Transmitter Data FIFO (TX DATA.FIFO) */ -#define TEGRA_SPDIF_DATA_FIFO_CSR_TX_CLR (1 << 7) - -/* TU FIFO attention level */ -#define TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_SHIFT 5 -#define TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_MASK \ - (0x3 << TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_SHIFT) -#define TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_TU1_WORD_FULL \ - (TEGRA_SPDIF_FIFO_ATN_LVL_D_ONE_SLOT << TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_SHIFT) -#define TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_TU4_WORD_FULL \ - (TEGRA_SPDIF_FIFO_ATN_LVL_D_FOUR_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_SHIFT) -#define TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_TU8_WORD_FULL \ - (TEGRA_SPDIF_FIFO_ATN_LVL_D_EIGHT_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_SHIFT) -#define TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_TU12_WORD_FULL \ - (TEGRA_SPDIF_FIFO_ATN_LVL_D_TWELVE_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_SHIFT) - -/* Number of TX DATA.FIFO levels that could be filled. */ -#define TEGRA_SPDIF_DATA_FIFO_CSR_TX_EMPTY_COUNT_SHIFT 0 -#define TEGRA_SPDIF_DATA_FIFO_CSR_TX_EMPTY_COUNT_MASK (0x1f << SPDIF_DATA_FIFO_CSR_TX_EMPTY_COUNT_SHIFT) - -/* Fields in TEGRA_SPDIF_DATA_OUT */ - -/* - * This register has 5 different formats: - * 16-bit (BIT_MODE=00, PACK=0) - * 20-bit (BIT_MODE=01, PACK=0) - * 24-bit (BIT_MODE=10, PACK=0) - * raw (BIT_MODE=11, PACK=0) - * 16-bit packed (BIT_MODE=00, PACK=1) - */ - -#define TEGRA_SPDIF_DATA_OUT_DATA_16_SHIFT 0 -#define TEGRA_SPDIF_DATA_OUT_DATA_16_MASK (0xffff << TEGRA_SPDIF_DATA_OUT_DATA_16_SHIFT) - -#define TEGRA_SPDIF_DATA_OUT_DATA_20_SHIFT 0 -#define TEGRA_SPDIF_DATA_OUT_DATA_20_MASK (0xfffff << TEGRA_SPDIF_DATA_OUT_DATA_20_SHIFT) - -#define TEGRA_SPDIF_DATA_OUT_DATA_24_SHIFT 0 -#define TEGRA_SPDIF_DATA_OUT_DATA_24_MASK (0xffffff << TEGRA_SPDIF_DATA_OUT_DATA_24_SHIFT) - -#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_P (1 << 31) -#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_C (1 << 30) -#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_U (1 << 29) -#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_V (1 << 28) - -#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_DATA_SHIFT 8 -#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_DATA_MASK (0xfffff << TEGRA_SPDIF_DATA_OUT_DATA_RAW_DATA_SHIFT) - -#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_AUX_SHIFT 4 -#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_AUX_MASK (0xf << TEGRA_SPDIF_DATA_OUT_DATA_RAW_AUX_SHIFT) - -#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_PREAMBLE_SHIFT 0 -#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_PREAMBLE_MASK (0xf << TEGRA_SPDIF_DATA_OUT_DATA_RAW_PREAMBLE_SHIFT) - -#define TEGRA_SPDIF_DATA_OUT_DATA_16_PACKED_RIGHT_SHIFT 16 -#define TEGRA_SPDIF_DATA_OUT_DATA_16_PACKED_RIGHT_MASK (0xffff << TEGRA_SPDIF_DATA_OUT_DATA_16_PACKED_RIGHT_SHIFT) - -#define TEGRA_SPDIF_DATA_OUT_DATA_16_PACKED_LEFT_SHIFT 0 -#define TEGRA_SPDIF_DATA_OUT_DATA_16_PACKED_LEFT_MASK (0xffff << TEGRA_SPDIF_DATA_OUT_DATA_16_PACKED_LEFT_SHIFT) - -/* Fields in TEGRA_SPDIF_DATA_IN */ - -/* - * This register has 5 different formats: - * 16-bit (BIT_MODE=00, PACK=0) - * 20-bit (BIT_MODE=01, PACK=0) - * 24-bit (BIT_MODE=10, PACK=0) - * raw (BIT_MODE=11, PACK=0) - * 16-bit packed (BIT_MODE=00, PACK=1) - * - * Bits 31:24 are common to all modes except 16-bit packed - */ - -#define TEGRA_SPDIF_DATA_IN_DATA_P (1 << 31) -#define TEGRA_SPDIF_DATA_IN_DATA_C (1 << 30) -#define TEGRA_SPDIF_DATA_IN_DATA_U (1 << 29) -#define TEGRA_SPDIF_DATA_IN_DATA_V (1 << 28) - -#define TEGRA_SPDIF_DATA_IN_DATA_PREAMBLE_SHIFT 24 -#define TEGRA_SPDIF_DATA_IN_DATA_PREAMBLE_MASK (0xf << TEGRA_SPDIF_DATA_IN_DATA_PREAMBLE_SHIFT) - -#define TEGRA_SPDIF_DATA_IN_DATA_16_SHIFT 0 -#define TEGRA_SPDIF_DATA_IN_DATA_16_MASK (0xffff << TEGRA_SPDIF_DATA_IN_DATA_16_SHIFT) - -#define TEGRA_SPDIF_DATA_IN_DATA_20_SHIFT 0 -#define TEGRA_SPDIF_DATA_IN_DATA_20_MASK (0xfffff << TEGRA_SPDIF_DATA_IN_DATA_20_SHIFT) - -#define TEGRA_SPDIF_DATA_IN_DATA_24_SHIFT 0 -#define TEGRA_SPDIF_DATA_IN_DATA_24_MASK (0xffffff << TEGRA_SPDIF_DATA_IN_DATA_24_SHIFT) - -#define TEGRA_SPDIF_DATA_IN_DATA_RAW_DATA_SHIFT 8 -#define TEGRA_SPDIF_DATA_IN_DATA_RAW_DATA_MASK (0xfffff << TEGRA_SPDIF_DATA_IN_DATA_RAW_DATA_SHIFT) - -#define TEGRA_SPDIF_DATA_IN_DATA_RAW_AUX_SHIFT 4 -#define TEGRA_SPDIF_DATA_IN_DATA_RAW_AUX_MASK (0xf << TEGRA_SPDIF_DATA_IN_DATA_RAW_AUX_SHIFT) - -#define TEGRA_SPDIF_DATA_IN_DATA_RAW_PREAMBLE_SHIFT 0 -#define TEGRA_SPDIF_DATA_IN_DATA_RAW_PREAMBLE_MASK (0xf << TEGRA_SPDIF_DATA_IN_DATA_RAW_PREAMBLE_SHIFT) - -#define TEGRA_SPDIF_DATA_IN_DATA_16_PACKED_RIGHT_SHIFT 16 -#define TEGRA_SPDIF_DATA_IN_DATA_16_PACKED_RIGHT_MASK (0xffff << TEGRA_SPDIF_DATA_IN_DATA_16_PACKED_RIGHT_SHIFT) - -#define TEGRA_SPDIF_DATA_IN_DATA_16_PACKED_LEFT_SHIFT 0 -#define TEGRA_SPDIF_DATA_IN_DATA_16_PACKED_LEFT_MASK (0xffff << TEGRA_SPDIF_DATA_IN_DATA_16_PACKED_LEFT_SHIFT) - -/* Fields in TEGRA_SPDIF_CH_STA_RX_A */ -/* Fields in TEGRA_SPDIF_CH_STA_RX_B */ -/* Fields in TEGRA_SPDIF_CH_STA_RX_C */ -/* Fields in TEGRA_SPDIF_CH_STA_RX_D */ -/* Fields in TEGRA_SPDIF_CH_STA_RX_E */ -/* Fields in TEGRA_SPDIF_CH_STA_RX_F */ - -/* - * The 6-word receive channel data page buffer holds a block (192 frames) of - * channel status information. The order of receive is from LSB to MSB - * bit, and from CH_STA_RX_A to CH_STA_RX_F then back to CH_STA_RX_A. - */ - -/* Fields in TEGRA_SPDIF_CH_STA_TX_A */ -/* Fields in TEGRA_SPDIF_CH_STA_TX_B */ -/* Fields in TEGRA_SPDIF_CH_STA_TX_C */ -/* Fields in TEGRA_SPDIF_CH_STA_TX_D */ -/* Fields in TEGRA_SPDIF_CH_STA_TX_E */ -/* Fields in TEGRA_SPDIF_CH_STA_TX_F */ - -/* - * The 6-word transmit channel data page buffer holds a block (192 frames) of - * channel status information. The order of transmission is from LSB to MSB - * bit, and from CH_STA_TX_A to CH_STA_TX_F then back to CH_STA_TX_A. - */ - -/* Fields in TEGRA_SPDIF_USR_STA_RX_A */ - -/* - * This 4-word deep FIFO receives user FIFO field information. The order of - * receive is from LSB to MSB bit. - */ - -/* Fields in TEGRA_SPDIF_USR_DAT_TX_A */ - -/* - * This 4-word deep FIFO transmits user FIFO field information. The order of - * transmission is from LSB to MSB bit. - */ - -struct tegra_spdif { - struct clk *clk_spdif_out; - struct tegra_pcm_dma_params capture_dma_data; - struct tegra_pcm_dma_params playback_dma_data; - void __iomem *regs; - struct dentry *debug; - u32 reg_ctrl; -}; - -#endif -- cgit v1.2.3 From 896637ac1be95a239b68dbe61c12a8a9bc00a9a3 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Fri, 6 Apr 2012 10:30:52 -0600 Subject: ASoC: tegra: complete Tegra->Tegra20 renaming Rename Tegra20-specific Kconfig variables, module filenames, all internal symbol names, clocks, and platform devices, to reflect the fact the DAS and I2S drivers are for a specific HW version. Signed-off-by: Stephen Warren Signed-off-by: Mark Brown --- arch/arm/mach-tegra/board-dt-tegra20.c | 6 +- arch/arm/mach-tegra/devices.c | 6 +- arch/arm/mach-tegra/tegra2_clocks.c | 4 +- sound/soc/tegra/Kconfig | 20 +- sound/soc/tegra/Makefile | 6 +- sound/soc/tegra/tegra20_das.c | 134 +++++------ sound/soc/tegra/tegra20_das.h | 124 +++++----- sound/soc/tegra/tegra20_i2s.c | 204 ++++++++--------- sound/soc/tegra/tegra20_i2s.h | 246 ++++++++++---------- sound/soc/tegra/tegra20_spdif.c | 140 ++++++------ sound/soc/tegra/tegra20_spdif.h | 398 ++++++++++++++++----------------- sound/soc/tegra/tegra_wm8903.c | 4 +- sound/soc/tegra/trimslice.c | 4 +- 13 files changed, 648 insertions(+), 648 deletions(-) (limited to 'sound') diff --git a/arch/arm/mach-tegra/board-dt-tegra20.c b/arch/arm/mach-tegra/board-dt-tegra20.c index 0952494f481a..86abcf0d2be9 100644 --- a/arch/arm/mach-tegra/board-dt-tegra20.c +++ b/arch/arm/mach-tegra/board-dt-tegra20.c @@ -64,9 +64,9 @@ struct of_dev_auxdata tegra20_auxdata_lookup[] __initdata = { OF_DEV_AUXDATA("nvidia,tegra20-i2c", TEGRA_I2C2_BASE, "tegra-i2c.1", NULL), OF_DEV_AUXDATA("nvidia,tegra20-i2c", TEGRA_I2C3_BASE, "tegra-i2c.2", NULL), OF_DEV_AUXDATA("nvidia,tegra20-i2c-dvc", TEGRA_DVC_BASE, "tegra-i2c.3", NULL), - OF_DEV_AUXDATA("nvidia,tegra20-i2s", TEGRA_I2S1_BASE, "tegra-i2s.0", NULL), - OF_DEV_AUXDATA("nvidia,tegra20-i2s", TEGRA_I2S2_BASE, "tegra-i2s.1", NULL), - OF_DEV_AUXDATA("nvidia,tegra20-das", TEGRA_APB_MISC_DAS_BASE, "tegra-das", NULL), + OF_DEV_AUXDATA("nvidia,tegra20-i2s", TEGRA_I2S1_BASE, "tegra20-i2s.0", NULL), + OF_DEV_AUXDATA("nvidia,tegra20-i2s", TEGRA_I2S2_BASE, "tegra20-i2s.1", NULL), + OF_DEV_AUXDATA("nvidia,tegra20-das", TEGRA_APB_MISC_DAS_BASE, "tegra20-das", NULL), OF_DEV_AUXDATA("nvidia,tegra20-ehci", TEGRA_USB_BASE, "tegra-ehci.0", &tegra_ehci1_pdata), OF_DEV_AUXDATA("nvidia,tegra20-ehci", TEGRA_USB2_BASE, "tegra-ehci.1", diff --git a/arch/arm/mach-tegra/devices.c b/arch/arm/mach-tegra/devices.c index c3b990097f28..d6c9b2624d85 100644 --- a/arch/arm/mach-tegra/devices.c +++ b/arch/arm/mach-tegra/devices.c @@ -671,14 +671,14 @@ static struct resource i2s_resource2[] = { }; struct platform_device tegra_i2s_device1 = { - .name = "tegra-i2s", + .name = "tegra20-i2s", .id = 0, .resource = i2s_resource1, .num_resources = ARRAY_SIZE(i2s_resource1), }; struct platform_device tegra_i2s_device2 = { - .name = "tegra-i2s", + .name = "tegra20-i2s", .id = 1, .resource = i2s_resource2, .num_resources = ARRAY_SIZE(i2s_resource2), @@ -693,7 +693,7 @@ static struct resource tegra_das_resources[] = { }; struct platform_device tegra_das_device = { - .name = "tegra-das", + .name = "tegra20-das", .id = -1, .num_resources = ARRAY_SIZE(tegra_das_resources), .resource = tegra_das_resources, diff --git a/arch/arm/mach-tegra/tegra2_clocks.c b/arch/arm/mach-tegra/tegra2_clocks.c index 592a4eeb5328..cf4999bcd4d8 100644 --- a/arch/arm/mach-tegra/tegra2_clocks.c +++ b/arch/arm/mach-tegra/tegra2_clocks.c @@ -2142,8 +2142,8 @@ static struct clk tegra_list_clks[] = { PERIPH_CLK("apbdma", "tegra-dma", NULL, 34, 0, 108000000, mux_pclk, 0), PERIPH_CLK("rtc", "rtc-tegra", NULL, 4, 0, 32768, mux_clk_32k, PERIPH_NO_RESET), PERIPH_CLK("timer", "timer", NULL, 5, 0, 26000000, mux_clk_m, 0), - PERIPH_CLK("i2s1", "tegra-i2s.0", NULL, 11, 0x100, 26000000, mux_pllaout0_audio2x_pllp_clkm, MUX | DIV_U71), - PERIPH_CLK("i2s2", "tegra-i2s.1", NULL, 18, 0x104, 26000000, mux_pllaout0_audio2x_pllp_clkm, MUX | DIV_U71), + PERIPH_CLK("i2s1", "tegra20-i2s.0", NULL, 11, 0x100, 26000000, mux_pllaout0_audio2x_pllp_clkm, MUX | DIV_U71), + PERIPH_CLK("i2s2", "tegra20-i2s.1", NULL, 18, 0x104, 26000000, mux_pllaout0_audio2x_pllp_clkm, MUX | DIV_U71), PERIPH_CLK("spdif_out", "spdif_out", NULL, 10, 0x108, 100000000, mux_pllaout0_audio2x_pllp_clkm, MUX | DIV_U71), PERIPH_CLK("spdif_in", "spdif_in", NULL, 10, 0x10c, 100000000, mux_pllp_pllc_pllm, MUX | DIV_U71), PERIPH_CLK("pwm", "pwm", NULL, 17, 0x110, 432000000, mux_pllp_pllc_audio_clkm_clk32, MUX | DIV_U71), diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig index d51945e78017..556cac29ea15 100644 --- a/sound/soc/tegra/Kconfig +++ b/sound/soc/tegra/Kconfig @@ -4,29 +4,29 @@ config SND_SOC_TEGRA help Say Y or M here if you want support for SoC audio on Tegra. -config SND_SOC_TEGRA_DAS +config SND_SOC_TEGRA20_DAS tristate depends on SND_SOC_TEGRA && ARCH_TEGRA_2x_SOC help - Say Y or M if you want to add support for the Tegra DAS module. + Say Y or M if you want to add support for the Tegra20 DAS module. You will also need to select the individual machine drivers to support below. -config SND_SOC_TEGRA_I2S +config SND_SOC_TEGRA20_I2S tristate depends on SND_SOC_TEGRA && ARCH_TEGRA_2x_SOC - select SND_SOC_TEGRA_DAS + select SND_SOC_TEGRA20_DAS help Say Y or M if you want to add support for codecs attached to the - Tegra I2S interface. You will also need to select the individual + Tegra20 I2S interface. You will also need to select the individual machine drivers to support below. -config SND_SOC_TEGRA_SPDIF +config SND_SOC_TEGRA20_SPDIF tristate depends on SND_SOC_TEGRA && ARCH_TEGRA_2x_SOC default m help - Say Y or M if you want to add support for the SPDIF interface. + Say Y or M if you want to add support for the Tegra20 SPDIF interface. You will also need to select the individual machine drivers to support below. @@ -41,7 +41,7 @@ config SND_SOC_TEGRA_WM8903 tristate "SoC Audio support for Tegra boards using a WM8903 codec" depends on SND_SOC_TEGRA && I2C depends on MACH_HAS_SND_SOC_TEGRA_WM8903 - select SND_SOC_TEGRA_I2S if ARCH_TEGRA_2x_SOC + select SND_SOC_TEGRA20_I2S if ARCH_TEGRA_2x_SOC select SND_SOC_WM8903 help Say Y or M here if you want to add support for SoC audio on Tegra @@ -51,7 +51,7 @@ config SND_SOC_TEGRA_WM8903 config SND_SOC_TEGRA_TRIMSLICE tristate "SoC Audio support for TrimSlice board" depends on SND_SOC_TEGRA && MACH_TRIMSLICE && I2C - select SND_SOC_TEGRA_I2S if ARCH_TEGRA_2x_SOC + select SND_SOC_TEGRA20_I2S if ARCH_TEGRA_2x_SOC select SND_SOC_TLV320AIC23 help Say Y or M here if you want to add support for SoC audio on the @@ -60,7 +60,7 @@ config SND_SOC_TEGRA_TRIMSLICE config SND_SOC_TEGRA_ALC5632 tristate "SoC Audio support for Tegra boards using an ALC5632 codec" depends on SND_SOC_TEGRA && I2C - select SND_SOC_TEGRA_I2S if ARCH_TEGRA_2x_SOC + select SND_SOC_TEGRA20_I2S if ARCH_TEGRA_2x_SOC select SND_SOC_ALC5632 help Say Y or M here if you want to add support for SoC audio on the diff --git a/sound/soc/tegra/Makefile b/sound/soc/tegra/Makefile index 4a33884115f8..4726b909664d 100644 --- a/sound/soc/tegra/Makefile +++ b/sound/soc/tegra/Makefile @@ -7,9 +7,9 @@ snd-soc-tegra20-spdif-objs := tegra20_spdif.o obj-$(CONFIG_SND_SOC_TEGRA) += snd-soc-tegra-pcm.o obj-$(CONFIG_SND_SOC_TEGRA) += snd-soc-tegra-utils.o -obj-$(CONFIG_SND_SOC_TEGRA_DAS) += snd-soc-tegra20-das.o -obj-$(CONFIG_SND_SOC_TEGRA_I2S) += snd-soc-tegra20-i2s.o -obj-$(CONFIG_SND_SOC_TEGRA_SPDIF) += snd-soc-tegra20-spdif.o +obj-$(CONFIG_SND_SOC_TEGRA20_DAS) += snd-soc-tegra20-das.o +obj-$(CONFIG_SND_SOC_TEGRA20_I2S) += snd-soc-tegra20-i2s.o +obj-$(CONFIG_SND_SOC_TEGRA20_SPDIF) += snd-soc-tegra20-spdif.o # Tegra machine Support snd-soc-tegra-wm8903-objs := tegra_wm8903.o diff --git a/sound/soc/tegra/tegra20_das.c b/sound/soc/tegra/tegra20_das.c index b5e369820cb3..486d7b245cf4 100644 --- a/sound/soc/tegra/tegra20_das.c +++ b/sound/soc/tegra/tegra20_das.c @@ -31,21 +31,21 @@ #include #include "tegra20_das.h" -#define DRV_NAME "tegra-das" +#define DRV_NAME "tegra20-das" -static struct tegra_das *das; +static struct tegra20_das *das; -static inline void tegra_das_write(u32 reg, u32 val) +static inline void tegra20_das_write(u32 reg, u32 val) { __raw_writel(val, das->regs + reg); } -static inline u32 tegra_das_read(u32 reg) +static inline u32 tegra20_das_read(u32 reg) { return __raw_readl(das->regs + reg); } -int tegra_das_connect_dap_to_dac(int dap, int dac) +int tegra20_das_connect_dap_to_dac(int dap, int dac) { u32 addr; u32 reg; @@ -53,18 +53,18 @@ int tegra_das_connect_dap_to_dac(int dap, int dac) if (!das) return -ENODEV; - addr = TEGRA_DAS_DAP_CTRL_SEL + - (dap * TEGRA_DAS_DAP_CTRL_SEL_STRIDE); - reg = dac << TEGRA_DAS_DAP_CTRL_SEL_DAP_CTRL_SEL_P; + addr = TEGRA20_DAS_DAP_CTRL_SEL + + (dap * TEGRA20_DAS_DAP_CTRL_SEL_STRIDE); + reg = dac << TEGRA20_DAS_DAP_CTRL_SEL_DAP_CTRL_SEL_P; - tegra_das_write(addr, reg); + tegra20_das_write(addr, reg); return 0; } -EXPORT_SYMBOL_GPL(tegra_das_connect_dap_to_dac); +EXPORT_SYMBOL_GPL(tegra20_das_connect_dap_to_dac); -int tegra_das_connect_dap_to_dap(int dap, int otherdap, int master, - int sdata1rx, int sdata2rx) +int tegra20_das_connect_dap_to_dap(int dap, int otherdap, int master, + int sdata1rx, int sdata2rx) { u32 addr; u32 reg; @@ -72,20 +72,20 @@ int tegra_das_connect_dap_to_dap(int dap, int otherdap, int master, if (!das) return -ENODEV; - addr = TEGRA_DAS_DAP_CTRL_SEL + - (dap * TEGRA_DAS_DAP_CTRL_SEL_STRIDE); - reg = otherdap << TEGRA_DAS_DAP_CTRL_SEL_DAP_CTRL_SEL_P | - !!sdata2rx << TEGRA_DAS_DAP_CTRL_SEL_DAP_SDATA2_TX_RX_P | - !!sdata1rx << TEGRA_DAS_DAP_CTRL_SEL_DAP_SDATA1_TX_RX_P | - !!master << TEGRA_DAS_DAP_CTRL_SEL_DAP_MS_SEL_P; + addr = TEGRA20_DAS_DAP_CTRL_SEL + + (dap * TEGRA20_DAS_DAP_CTRL_SEL_STRIDE); + reg = otherdap << TEGRA20_DAS_DAP_CTRL_SEL_DAP_CTRL_SEL_P | + !!sdata2rx << TEGRA20_DAS_DAP_CTRL_SEL_DAP_SDATA2_TX_RX_P | + !!sdata1rx << TEGRA20_DAS_DAP_CTRL_SEL_DAP_SDATA1_TX_RX_P | + !!master << TEGRA20_DAS_DAP_CTRL_SEL_DAP_MS_SEL_P; - tegra_das_write(addr, reg); + tegra20_das_write(addr, reg); return 0; } -EXPORT_SYMBOL_GPL(tegra_das_connect_dap_to_dap); +EXPORT_SYMBOL_GPL(tegra20_das_connect_dap_to_dap); -int tegra_das_connect_dac_to_dap(int dac, int dap) +int tegra20_das_connect_dac_to_dap(int dac, int dap) { u32 addr; u32 reg; @@ -93,78 +93,78 @@ int tegra_das_connect_dac_to_dap(int dac, int dap) if (!das) return -ENODEV; - addr = TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL + - (dac * TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_STRIDE); - reg = dap << TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_CLK_SEL_P | - dap << TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA1_SEL_P | - dap << TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA2_SEL_P; + addr = TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL + + (dac * TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL_STRIDE); + reg = dap << TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_CLK_SEL_P | + dap << TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA1_SEL_P | + dap << TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA2_SEL_P; - tegra_das_write(addr, reg); + tegra20_das_write(addr, reg); return 0; } -EXPORT_SYMBOL_GPL(tegra_das_connect_dac_to_dap); +EXPORT_SYMBOL_GPL(tegra20_das_connect_dac_to_dap); #ifdef CONFIG_DEBUG_FS -static int tegra_das_show(struct seq_file *s, void *unused) +static int tegra20_das_show(struct seq_file *s, void *unused) { int i; u32 addr; u32 reg; - for (i = 0; i < TEGRA_DAS_DAP_CTRL_SEL_COUNT; i++) { - addr = TEGRA_DAS_DAP_CTRL_SEL + - (i * TEGRA_DAS_DAP_CTRL_SEL_STRIDE); - reg = tegra_das_read(addr); - seq_printf(s, "TEGRA_DAS_DAP_CTRL_SEL[%d] = %08x\n", i, reg); + for (i = 0; i < TEGRA20_DAS_DAP_CTRL_SEL_COUNT; i++) { + addr = TEGRA20_DAS_DAP_CTRL_SEL + + (i * TEGRA20_DAS_DAP_CTRL_SEL_STRIDE); + reg = tegra20_das_read(addr); + seq_printf(s, "TEGRA20_DAS_DAP_CTRL_SEL[%d] = %08x\n", i, reg); } - for (i = 0; i < TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_COUNT; i++) { - addr = TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL + - (i * TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_STRIDE); - reg = tegra_das_read(addr); - seq_printf(s, "TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL[%d] = %08x\n", - i, reg); + for (i = 0; i < TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL_COUNT; i++) { + addr = TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL + + (i * TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL_STRIDE); + reg = tegra20_das_read(addr); + seq_printf(s, "TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL[%d] = %08x\n", + i, reg); } return 0; } -static int tegra_das_debug_open(struct inode *inode, struct file *file) +static int tegra20_das_debug_open(struct inode *inode, struct file *file) { - return single_open(file, tegra_das_show, inode->i_private); + return single_open(file, tegra20_das_show, inode->i_private); } -static const struct file_operations tegra_das_debug_fops = { - .open = tegra_das_debug_open, +static const struct file_operations tegra20_das_debug_fops = { + .open = tegra20_das_debug_open, .read = seq_read, .llseek = seq_lseek, .release = single_release, }; -static void tegra_das_debug_add(struct tegra_das *das) +static void tegra20_das_debug_add(struct tegra20_das *das) { das->debug = debugfs_create_file(DRV_NAME, S_IRUGO, snd_soc_debugfs_root, das, - &tegra_das_debug_fops); + &tegra20_das_debug_fops); } -static void tegra_das_debug_remove(struct tegra_das *das) +static void tegra20_das_debug_remove(struct tegra20_das *das) { if (das->debug) debugfs_remove(das->debug); } #else -static inline void tegra_das_debug_add(struct tegra_das *das) +static inline void tegra20_das_debug_add(struct tegra20_das *das) { } -static inline void tegra_das_debug_remove(struct tegra_das *das) +static inline void tegra20_das_debug_remove(struct tegra20_das *das) { } #endif -static int __devinit tegra_das_probe(struct platform_device *pdev) +static int __devinit tegra20_das_probe(struct platform_device *pdev) { struct resource *res, *region; int ret = 0; @@ -172,9 +172,9 @@ static int __devinit tegra_das_probe(struct platform_device *pdev) if (das) return -ENODEV; - das = devm_kzalloc(&pdev->dev, sizeof(struct tegra_das), GFP_KERNEL); + das = devm_kzalloc(&pdev->dev, sizeof(struct tegra20_das), GFP_KERNEL); if (!das) { - dev_err(&pdev->dev, "Can't allocate tegra_das\n"); + dev_err(&pdev->dev, "Can't allocate tegra20_das\n"); ret = -ENOMEM; goto err; } @@ -202,20 +202,20 @@ static int __devinit tegra_das_probe(struct platform_device *pdev) goto err; } - ret = tegra_das_connect_dap_to_dac(TEGRA_DAS_DAP_ID_1, - TEGRA_DAS_DAP_SEL_DAC1); + ret = tegra20_das_connect_dap_to_dac(TEGRA20_DAS_DAP_ID_1, + TEGRA20_DAS_DAP_SEL_DAC1); if (ret) { dev_err(&pdev->dev, "Can't set up DAS DAP connection\n"); goto err; } - ret = tegra_das_connect_dac_to_dap(TEGRA_DAS_DAC_ID_1, - TEGRA_DAS_DAC_SEL_DAP1); + ret = tegra20_das_connect_dac_to_dap(TEGRA20_DAS_DAC_ID_1, + TEGRA20_DAS_DAC_SEL_DAP1); if (ret) { dev_err(&pdev->dev, "Can't set up DAS DAC connection\n"); goto err; } - tegra_das_debug_add(das); + tegra20_das_debug_add(das); platform_set_drvdata(pdev, das); @@ -226,36 +226,36 @@ err: return ret; } -static int __devexit tegra_das_remove(struct platform_device *pdev) +static int __devexit tegra20_das_remove(struct platform_device *pdev) { if (!das) return -ENODEV; - tegra_das_debug_remove(das); + tegra20_das_debug_remove(das); das = NULL; return 0; } -static const struct of_device_id tegra_das_of_match[] __devinitconst = { +static const struct of_device_id tegra20_das_of_match[] __devinitconst = { { .compatible = "nvidia,tegra20-das", }, {}, }; -static struct platform_driver tegra_das_driver = { - .probe = tegra_das_probe, - .remove = __devexit_p(tegra_das_remove), +static struct platform_driver tegra20_das_driver = { + .probe = tegra20_das_probe, + .remove = __devexit_p(tegra20_das_remove), .driver = { .name = DRV_NAME, .owner = THIS_MODULE, - .of_match_table = tegra_das_of_match, + .of_match_table = tegra20_das_of_match, }, }; -module_platform_driver(tegra_das_driver); +module_platform_driver(tegra20_das_driver); MODULE_AUTHOR("Stephen Warren "); -MODULE_DESCRIPTION("Tegra DAS driver"); +MODULE_DESCRIPTION("Tegra20 DAS driver"); MODULE_LICENSE("GPL"); MODULE_ALIAS("platform:" DRV_NAME); -MODULE_DEVICE_TABLE(of, tegra_das_of_match); +MODULE_DEVICE_TABLE(of, tegra20_das_of_match); diff --git a/sound/soc/tegra/tegra20_das.h b/sound/soc/tegra/tegra20_das.h index 896bf031a801..ade4fe080c45 100644 --- a/sound/soc/tegra/tegra20_das.h +++ b/sound/soc/tegra/tegra20_das.h @@ -2,7 +2,7 @@ * tegra20_das.h - Definitions for Tegra20 DAS driver * * Author: Stephen Warren - * Copyright (C) 2010 - NVIDIA, Inc. + * Copyright (C) 2010,2012 - NVIDIA, Inc. * * This program is free software; you can redistribute it and/or * modify it under the terms of the GNU General Public License @@ -20,70 +20,70 @@ * */ -#ifndef __TEGRA_DAS_H__ -#define __TEGRA_DAS_H__ +#ifndef __TEGRA20_DAS_H__ +#define __TEGRA20_DAS_H__ -/* Register TEGRA_DAS_DAP_CTRL_SEL */ -#define TEGRA_DAS_DAP_CTRL_SEL 0x00 -#define TEGRA_DAS_DAP_CTRL_SEL_COUNT 5 -#define TEGRA_DAS_DAP_CTRL_SEL_STRIDE 4 -#define TEGRA_DAS_DAP_CTRL_SEL_DAP_MS_SEL_P 31 -#define TEGRA_DAS_DAP_CTRL_SEL_DAP_MS_SEL_S 1 -#define TEGRA_DAS_DAP_CTRL_SEL_DAP_SDATA1_TX_RX_P 30 -#define TEGRA_DAS_DAP_CTRL_SEL_DAP_SDATA1_TX_RX_S 1 -#define TEGRA_DAS_DAP_CTRL_SEL_DAP_SDATA2_TX_RX_P 29 -#define TEGRA_DAS_DAP_CTRL_SEL_DAP_SDATA2_TX_RX_S 1 -#define TEGRA_DAS_DAP_CTRL_SEL_DAP_CTRL_SEL_P 0 -#define TEGRA_DAS_DAP_CTRL_SEL_DAP_CTRL_SEL_S 5 +/* Register TEGRA20_DAS_DAP_CTRL_SEL */ +#define TEGRA20_DAS_DAP_CTRL_SEL 0x00 +#define TEGRA20_DAS_DAP_CTRL_SEL_COUNT 5 +#define TEGRA20_DAS_DAP_CTRL_SEL_STRIDE 4 +#define TEGRA20_DAS_DAP_CTRL_SEL_DAP_MS_SEL_P 31 +#define TEGRA20_DAS_DAP_CTRL_SEL_DAP_MS_SEL_S 1 +#define TEGRA20_DAS_DAP_CTRL_SEL_DAP_SDATA1_TX_RX_P 30 +#define TEGRA20_DAS_DAP_CTRL_SEL_DAP_SDATA1_TX_RX_S 1 +#define TEGRA20_DAS_DAP_CTRL_SEL_DAP_SDATA2_TX_RX_P 29 +#define TEGRA20_DAS_DAP_CTRL_SEL_DAP_SDATA2_TX_RX_S 1 +#define TEGRA20_DAS_DAP_CTRL_SEL_DAP_CTRL_SEL_P 0 +#define TEGRA20_DAS_DAP_CTRL_SEL_DAP_CTRL_SEL_S 5 -/* Values for field TEGRA_DAS_DAP_CTRL_SEL_DAP_CTRL_SEL */ -#define TEGRA_DAS_DAP_SEL_DAC1 0 -#define TEGRA_DAS_DAP_SEL_DAC2 1 -#define TEGRA_DAS_DAP_SEL_DAC3 2 -#define TEGRA_DAS_DAP_SEL_DAP1 16 -#define TEGRA_DAS_DAP_SEL_DAP2 17 -#define TEGRA_DAS_DAP_SEL_DAP3 18 -#define TEGRA_DAS_DAP_SEL_DAP4 19 -#define TEGRA_DAS_DAP_SEL_DAP5 20 +/* Values for field TEGRA20_DAS_DAP_CTRL_SEL_DAP_CTRL_SEL */ +#define TEGRA20_DAS_DAP_SEL_DAC1 0 +#define TEGRA20_DAS_DAP_SEL_DAC2 1 +#define TEGRA20_DAS_DAP_SEL_DAC3 2 +#define TEGRA20_DAS_DAP_SEL_DAP1 16 +#define TEGRA20_DAS_DAP_SEL_DAP2 17 +#define TEGRA20_DAS_DAP_SEL_DAP3 18 +#define TEGRA20_DAS_DAP_SEL_DAP4 19 +#define TEGRA20_DAS_DAP_SEL_DAP5 20 -/* Register TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL */ -#define TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL 0x40 -#define TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_COUNT 3 -#define TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_STRIDE 4 -#define TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA2_SEL_P 28 -#define TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA2_SEL_S 4 -#define TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA1_SEL_P 24 -#define TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA1_SEL_S 4 -#define TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_CLK_SEL_P 0 -#define TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_CLK_SEL_S 4 +/* Register TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL */ +#define TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL 0x40 +#define TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL_COUNT 3 +#define TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL_STRIDE 4 +#define TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA2_SEL_P 28 +#define TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA2_SEL_S 4 +#define TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA1_SEL_P 24 +#define TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA1_SEL_S 4 +#define TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_CLK_SEL_P 0 +#define TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_CLK_SEL_S 4 /* * Values for: - * TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA2_SEL - * TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA1_SEL - * TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_CLK_SEL + * TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA2_SEL + * TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA1_SEL + * TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_CLK_SEL */ -#define TEGRA_DAS_DAC_SEL_DAP1 0 -#define TEGRA_DAS_DAC_SEL_DAP2 1 -#define TEGRA_DAS_DAC_SEL_DAP3 2 -#define TEGRA_DAS_DAC_SEL_DAP4 3 -#define TEGRA_DAS_DAC_SEL_DAP5 4 +#define TEGRA20_DAS_DAC_SEL_DAP1 0 +#define TEGRA20_DAS_DAC_SEL_DAP2 1 +#define TEGRA20_DAS_DAC_SEL_DAP3 2 +#define TEGRA20_DAS_DAC_SEL_DAP4 3 +#define TEGRA20_DAS_DAC_SEL_DAP5 4 /* * Names/IDs of the DACs/DAPs. */ -#define TEGRA_DAS_DAP_ID_1 0 -#define TEGRA_DAS_DAP_ID_2 1 -#define TEGRA_DAS_DAP_ID_3 2 -#define TEGRA_DAS_DAP_ID_4 3 -#define TEGRA_DAS_DAP_ID_5 4 +#define TEGRA20_DAS_DAP_ID_1 0 +#define TEGRA20_DAS_DAP_ID_2 1 +#define TEGRA20_DAS_DAP_ID_3 2 +#define TEGRA20_DAS_DAP_ID_4 3 +#define TEGRA20_DAS_DAP_ID_5 4 -#define TEGRA_DAS_DAC_ID_1 0 -#define TEGRA_DAS_DAC_ID_2 1 -#define TEGRA_DAS_DAC_ID_3 2 +#define TEGRA20_DAS_DAC_ID_1 0 +#define TEGRA20_DAS_DAC_ID_2 1 +#define TEGRA20_DAS_DAC_ID_3 2 -struct tegra_das { +struct tegra20_das { struct device *dev; void __iomem *regs; struct dentry *debug; @@ -107,29 +107,29 @@ struct tegra_das { /* * Connect a DAP to to a DAC - * dap_id: DAP to connect: TEGRA_DAS_DAP_ID_* - * dac_sel: DAC to connect to: TEGRA_DAS_DAP_SEL_DAC* + * dap_id: DAP to connect: TEGRA20_DAS_DAP_ID_* + * dac_sel: DAC to connect to: TEGRA20_DAS_DAP_SEL_DAC* */ -extern int tegra_das_connect_dap_to_dac(int dap_id, int dac_sel); +extern int tegra20_das_connect_dap_to_dac(int dap_id, int dac_sel); /* * Connect a DAP to to another DAP - * dap_id: DAP to connect: TEGRA_DAS_DAP_ID_* - * other_dap_sel: DAP to connect to: TEGRA_DAS_DAP_SEL_DAP* + * dap_id: DAP to connect: TEGRA20_DAS_DAP_ID_* + * other_dap_sel: DAP to connect to: TEGRA20_DAS_DAP_SEL_DAP* * master: Is this DAP the master (1) or slave (0) * sdata1rx: Is this DAP's SDATA1 pin RX (1) or TX (0) * sdata2rx: Is this DAP's SDATA2 pin RX (1) or TX (0) */ -extern int tegra_das_connect_dap_to_dap(int dap_id, int other_dap_sel, - int master, int sdata1rx, - int sdata2rx); +extern int tegra20_das_connect_dap_to_dap(int dap_id, int other_dap_sel, + int master, int sdata1rx, + int sdata2rx); /* * Connect a DAC's input to a DAP * (DAC outputs are selected by the DAP) - * dac_id: DAC ID to connect: TEGRA_DAS_DAC_ID_* - * dap_sel: DAP to receive input from: TEGRA_DAS_DAC_SEL_DAP* + * dac_id: DAC ID to connect: TEGRA20_DAS_DAC_ID_* + * dap_sel: DAP to receive input from: TEGRA20_DAS_DAC_SEL_DAP* */ -extern int tegra_das_connect_dac_to_dap(int dac_id, int dap_sel); +extern int tegra20_das_connect_dac_to_dap(int dac_id, int dap_sel); #endif diff --git a/sound/soc/tegra/tegra20_i2s.c b/sound/soc/tegra/tegra20_i2s.c index e24759ae291b..878798c89e49 100644 --- a/sound/soc/tegra/tegra20_i2s.c +++ b/sound/soc/tegra/tegra20_i2s.c @@ -45,86 +45,86 @@ #include "tegra20_i2s.h" -#define DRV_NAME "tegra-i2s" +#define DRV_NAME "tegra20-i2s" -static inline void tegra_i2s_write(struct tegra_i2s *i2s, u32 reg, u32 val) +static inline void tegra20_i2s_write(struct tegra20_i2s *i2s, u32 reg, u32 val) { __raw_writel(val, i2s->regs + reg); } -static inline u32 tegra_i2s_read(struct tegra_i2s *i2s, u32 reg) +static inline u32 tegra20_i2s_read(struct tegra20_i2s *i2s, u32 reg) { return __raw_readl(i2s->regs + reg); } #ifdef CONFIG_DEBUG_FS -static int tegra_i2s_show(struct seq_file *s, void *unused) +static int tegra20_i2s_show(struct seq_file *s, void *unused) { #define REG(r) { r, #r } static const struct { int offset; const char *name; } regs[] = { - REG(TEGRA_I2S_CTRL), - REG(TEGRA_I2S_STATUS), - REG(TEGRA_I2S_TIMING), - REG(TEGRA_I2S_FIFO_SCR), - REG(TEGRA_I2S_PCM_CTRL), - REG(TEGRA_I2S_NW_CTRL), - REG(TEGRA_I2S_TDM_CTRL), - REG(TEGRA_I2S_TDM_TX_RX_CTRL), + REG(TEGRA20_I2S_CTRL), + REG(TEGRA20_I2S_STATUS), + REG(TEGRA20_I2S_TIMING), + REG(TEGRA20_I2S_FIFO_SCR), + REG(TEGRA20_I2S_PCM_CTRL), + REG(TEGRA20_I2S_NW_CTRL), + REG(TEGRA20_I2S_TDM_CTRL), + REG(TEGRA20_I2S_TDM_TX_RX_CTRL), }; #undef REG - struct tegra_i2s *i2s = s->private; + struct tegra20_i2s *i2s = s->private; int i; for (i = 0; i < ARRAY_SIZE(regs); i++) { - u32 val = tegra_i2s_read(i2s, regs[i].offset); + u32 val = tegra20_i2s_read(i2s, regs[i].offset); seq_printf(s, "%s = %08x\n", regs[i].name, val); } return 0; } -static int tegra_i2s_debug_open(struct inode *inode, struct file *file) +static int tegra20_i2s_debug_open(struct inode *inode, struct file *file) { - return single_open(file, tegra_i2s_show, inode->i_private); + return single_open(file, tegra20_i2s_show, inode->i_private); } -static const struct file_operations tegra_i2s_debug_fops = { - .open = tegra_i2s_debug_open, +static const struct file_operations tegra20_i2s_debug_fops = { + .open = tegra20_i2s_debug_open, .read = seq_read, .llseek = seq_lseek, .release = single_release, }; -static void tegra_i2s_debug_add(struct tegra_i2s *i2s) +static void tegra20_i2s_debug_add(struct tegra20_i2s *i2s) { i2s->debug = debugfs_create_file(i2s->dai.name, S_IRUGO, snd_soc_debugfs_root, i2s, - &tegra_i2s_debug_fops); + &tegra20_i2s_debug_fops); } -static void tegra_i2s_debug_remove(struct tegra_i2s *i2s) +static void tegra20_i2s_debug_remove(struct tegra20_i2s *i2s) { if (i2s->debug) debugfs_remove(i2s->debug); } #else -static inline void tegra_i2s_debug_add(struct tegra_i2s *i2s, int id) +static inline void tegra20_i2s_debug_add(struct tegra20_i2s *i2s, int id) { } -static inline void tegra_i2s_debug_remove(struct tegra_i2s *i2s) +static inline void tegra20_i2s_debug_remove(struct tegra20_i2s *i2s) { } #endif -static int tegra_i2s_set_fmt(struct snd_soc_dai *dai, +static int tegra20_i2s_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) { - struct tegra_i2s *i2s = snd_soc_dai_get_drvdata(dai); + struct tegra20_i2s *i2s = snd_soc_dai_get_drvdata(dai); switch (fmt & SND_SOC_DAIFMT_INV_MASK) { case SND_SOC_DAIFMT_NB_NF: @@ -133,10 +133,10 @@ static int tegra_i2s_set_fmt(struct snd_soc_dai *dai, return -EINVAL; } - i2s->reg_ctrl &= ~TEGRA_I2S_CTRL_MASTER_ENABLE; + i2s->reg_ctrl &= ~TEGRA20_I2S_CTRL_MASTER_ENABLE; switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBS_CFS: - i2s->reg_ctrl |= TEGRA_I2S_CTRL_MASTER_ENABLE; + i2s->reg_ctrl |= TEGRA20_I2S_CTRL_MASTER_ENABLE; break; case SND_SOC_DAIFMT_CBM_CFM: break; @@ -144,28 +144,28 @@ static int tegra_i2s_set_fmt(struct snd_soc_dai *dai, return -EINVAL; } - i2s->reg_ctrl &= ~(TEGRA_I2S_CTRL_BIT_FORMAT_MASK | - TEGRA_I2S_CTRL_LRCK_MASK); + i2s->reg_ctrl &= ~(TEGRA20_I2S_CTRL_BIT_FORMAT_MASK | + TEGRA20_I2S_CTRL_LRCK_MASK); switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_DSP_A: - i2s->reg_ctrl |= TEGRA_I2S_CTRL_BIT_FORMAT_DSP; - i2s->reg_ctrl |= TEGRA_I2S_CTRL_LRCK_L_LOW; + i2s->reg_ctrl |= TEGRA20_I2S_CTRL_BIT_FORMAT_DSP; + i2s->reg_ctrl |= TEGRA20_I2S_CTRL_LRCK_L_LOW; break; case SND_SOC_DAIFMT_DSP_B: - i2s->reg_ctrl |= TEGRA_I2S_CTRL_BIT_FORMAT_DSP; - i2s->reg_ctrl |= TEGRA_I2S_CTRL_LRCK_R_LOW; + i2s->reg_ctrl |= TEGRA20_I2S_CTRL_BIT_FORMAT_DSP; + i2s->reg_ctrl |= TEGRA20_I2S_CTRL_LRCK_R_LOW; break; case SND_SOC_DAIFMT_I2S: - i2s->reg_ctrl |= TEGRA_I2S_CTRL_BIT_FORMAT_I2S; - i2s->reg_ctrl |= TEGRA_I2S_CTRL_LRCK_L_LOW; + i2s->reg_ctrl |= TEGRA20_I2S_CTRL_BIT_FORMAT_I2S; + i2s->reg_ctrl |= TEGRA20_I2S_CTRL_LRCK_L_LOW; break; case SND_SOC_DAIFMT_RIGHT_J: - i2s->reg_ctrl |= TEGRA_I2S_CTRL_BIT_FORMAT_RJM; - i2s->reg_ctrl |= TEGRA_I2S_CTRL_LRCK_L_LOW; + i2s->reg_ctrl |= TEGRA20_I2S_CTRL_BIT_FORMAT_RJM; + i2s->reg_ctrl |= TEGRA20_I2S_CTRL_LRCK_L_LOW; break; case SND_SOC_DAIFMT_LEFT_J: - i2s->reg_ctrl |= TEGRA_I2S_CTRL_BIT_FORMAT_LJM; - i2s->reg_ctrl |= TEGRA_I2S_CTRL_LRCK_L_LOW; + i2s->reg_ctrl |= TEGRA20_I2S_CTRL_BIT_FORMAT_LJM; + i2s->reg_ctrl |= TEGRA20_I2S_CTRL_LRCK_L_LOW; break; default: return -EINVAL; @@ -174,27 +174,27 @@ static int tegra_i2s_set_fmt(struct snd_soc_dai *dai, return 0; } -static int tegra_i2s_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) +static int tegra20_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct device *dev = substream->pcm->card->dev; - struct tegra_i2s *i2s = snd_soc_dai_get_drvdata(dai); + struct tegra20_i2s *i2s = snd_soc_dai_get_drvdata(dai); u32 reg; int ret, sample_size, srate, i2sclock, bitcnt; - i2s->reg_ctrl &= ~TEGRA_I2S_CTRL_BIT_SIZE_MASK; + i2s->reg_ctrl &= ~TEGRA20_I2S_CTRL_BIT_SIZE_MASK; switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: - i2s->reg_ctrl |= TEGRA_I2S_CTRL_BIT_SIZE_16; + i2s->reg_ctrl |= TEGRA20_I2S_CTRL_BIT_SIZE_16; sample_size = 16; break; case SNDRV_PCM_FORMAT_S24_LE: - i2s->reg_ctrl |= TEGRA_I2S_CTRL_BIT_SIZE_24; + i2s->reg_ctrl |= TEGRA20_I2S_CTRL_BIT_SIZE_24; sample_size = 24; break; case SNDRV_PCM_FORMAT_S32_LE: - i2s->reg_ctrl |= TEGRA_I2S_CTRL_BIT_SIZE_32; + i2s->reg_ctrl |= TEGRA20_I2S_CTRL_BIT_SIZE_32; sample_size = 32; break; default: @@ -213,54 +213,54 @@ static int tegra_i2s_hw_params(struct snd_pcm_substream *substream, } bitcnt = (i2sclock / (2 * srate)) - 1; - if (bitcnt < 0 || bitcnt > TEGRA_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US) + if (bitcnt < 0 || bitcnt > TEGRA20_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US) return -EINVAL; - reg = bitcnt << TEGRA_I2S_TIMING_CHANNEL_BIT_COUNT_SHIFT; + reg = bitcnt << TEGRA20_I2S_TIMING_CHANNEL_BIT_COUNT_SHIFT; if (i2sclock % (2 * srate)) - reg |= TEGRA_I2S_TIMING_NON_SYM_ENABLE; + reg |= TEGRA20_I2S_TIMING_NON_SYM_ENABLE; clk_enable(i2s->clk_i2s); - tegra_i2s_write(i2s, TEGRA_I2S_TIMING, reg); + tegra20_i2s_write(i2s, TEGRA20_I2S_TIMING, reg); - tegra_i2s_write(i2s, TEGRA_I2S_FIFO_SCR, - TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_FOUR_SLOTS | - TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_FOUR_SLOTS); + tegra20_i2s_write(i2s, TEGRA20_I2S_FIFO_SCR, + TEGRA20_I2S_FIFO_SCR_FIFO2_ATN_LVL_FOUR_SLOTS | + TEGRA20_I2S_FIFO_SCR_FIFO1_ATN_LVL_FOUR_SLOTS); clk_disable(i2s->clk_i2s); return 0; } -static void tegra_i2s_start_playback(struct tegra_i2s *i2s) +static void tegra20_i2s_start_playback(struct tegra20_i2s *i2s) { - i2s->reg_ctrl |= TEGRA_I2S_CTRL_FIFO1_ENABLE; - tegra_i2s_write(i2s, TEGRA_I2S_CTRL, i2s->reg_ctrl); + i2s->reg_ctrl |= TEGRA20_I2S_CTRL_FIFO1_ENABLE; + tegra20_i2s_write(i2s, TEGRA20_I2S_CTRL, i2s->reg_ctrl); } -static void tegra_i2s_stop_playback(struct tegra_i2s *i2s) +static void tegra20_i2s_stop_playback(struct tegra20_i2s *i2s) { - i2s->reg_ctrl &= ~TEGRA_I2S_CTRL_FIFO1_ENABLE; - tegra_i2s_write(i2s, TEGRA_I2S_CTRL, i2s->reg_ctrl); + i2s->reg_ctrl &= ~TEGRA20_I2S_CTRL_FIFO1_ENABLE; + tegra20_i2s_write(i2s, TEGRA20_I2S_CTRL, i2s->reg_ctrl); } -static void tegra_i2s_start_capture(struct tegra_i2s *i2s) +static void tegra20_i2s_start_capture(struct tegra20_i2s *i2s) { - i2s->reg_ctrl |= TEGRA_I2S_CTRL_FIFO2_ENABLE; - tegra_i2s_write(i2s, TEGRA_I2S_CTRL, i2s->reg_ctrl); + i2s->reg_ctrl |= TEGRA20_I2S_CTRL_FIFO2_ENABLE; + tegra20_i2s_write(i2s, TEGRA20_I2S_CTRL, i2s->reg_ctrl); } -static void tegra_i2s_stop_capture(struct tegra_i2s *i2s) +static void tegra20_i2s_stop_capture(struct tegra20_i2s *i2s) { - i2s->reg_ctrl &= ~TEGRA_I2S_CTRL_FIFO2_ENABLE; - tegra_i2s_write(i2s, TEGRA_I2S_CTRL, i2s->reg_ctrl); + i2s->reg_ctrl &= ~TEGRA20_I2S_CTRL_FIFO2_ENABLE; + tegra20_i2s_write(i2s, TEGRA20_I2S_CTRL, i2s->reg_ctrl); } -static int tegra_i2s_trigger(struct snd_pcm_substream *substream, int cmd, - struct snd_soc_dai *dai) +static int tegra20_i2s_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) { - struct tegra_i2s *i2s = snd_soc_dai_get_drvdata(dai); + struct tegra20_i2s *i2s = snd_soc_dai_get_drvdata(dai); switch (cmd) { case SNDRV_PCM_TRIGGER_START: @@ -268,17 +268,17 @@ static int tegra_i2s_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_RESUME: clk_enable(i2s->clk_i2s); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - tegra_i2s_start_playback(i2s); + tegra20_i2s_start_playback(i2s); else - tegra_i2s_start_capture(i2s); + tegra20_i2s_start_capture(i2s); break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: case SNDRV_PCM_TRIGGER_SUSPEND: if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - tegra_i2s_stop_playback(i2s); + tegra20_i2s_stop_playback(i2s); else - tegra_i2s_stop_capture(i2s); + tegra20_i2s_stop_capture(i2s); clk_disable(i2s->clk_i2s); break; default: @@ -288,9 +288,9 @@ static int tegra_i2s_trigger(struct snd_pcm_substream *substream, int cmd, return 0; } -static int tegra_i2s_probe(struct snd_soc_dai *dai) +static int tegra20_i2s_probe(struct snd_soc_dai *dai) { - struct tegra_i2s *i2s = snd_soc_dai_get_drvdata(dai); + struct tegra20_i2s *i2s = snd_soc_dai_get_drvdata(dai); dai->capture_dma_data = &i2s->capture_dma_data; dai->playback_dma_data = &i2s->playback_dma_data; @@ -298,14 +298,14 @@ static int tegra_i2s_probe(struct snd_soc_dai *dai) return 0; } -static const struct snd_soc_dai_ops tegra_i2s_dai_ops = { - .set_fmt = tegra_i2s_set_fmt, - .hw_params = tegra_i2s_hw_params, - .trigger = tegra_i2s_trigger, +static const struct snd_soc_dai_ops tegra20_i2s_dai_ops = { + .set_fmt = tegra20_i2s_set_fmt, + .hw_params = tegra20_i2s_hw_params, + .trigger = tegra20_i2s_trigger, }; -static const struct snd_soc_dai_driver tegra_i2s_dai_template = { - .probe = tegra_i2s_probe, +static const struct snd_soc_dai_driver tegra20_i2s_dai_template = { + .probe = tegra20_i2s_probe, .playback = { .channels_min = 2, .channels_max = 2, @@ -318,27 +318,27 @@ static const struct snd_soc_dai_driver tegra_i2s_dai_template = { .rates = SNDRV_PCM_RATE_8000_96000, .formats = SNDRV_PCM_FMTBIT_S16_LE, }, - .ops = &tegra_i2s_dai_ops, + .ops = &tegra20_i2s_dai_ops, .symmetric_rates = 1, }; -static __devinit int tegra_i2s_platform_probe(struct platform_device *pdev) +static __devinit int tegra20_i2s_platform_probe(struct platform_device *pdev) { - struct tegra_i2s *i2s; + struct tegra20_i2s *i2s; struct resource *mem, *memregion, *dmareq; u32 of_dma[2]; u32 dma_ch; int ret; - i2s = devm_kzalloc(&pdev->dev, sizeof(struct tegra_i2s), GFP_KERNEL); + i2s = devm_kzalloc(&pdev->dev, sizeof(struct tegra20_i2s), GFP_KERNEL); if (!i2s) { - dev_err(&pdev->dev, "Can't allocate tegra_i2s\n"); + dev_err(&pdev->dev, "Can't allocate tegra20_i2s\n"); ret = -ENOMEM; goto err; } dev_set_drvdata(&pdev->dev, i2s); - i2s->dai = tegra_i2s_dai_template; + i2s->dai = tegra20_i2s_dai_template; i2s->dai.name = dev_name(&pdev->dev); i2s->clk_i2s = clk_get(&pdev->dev, NULL); @@ -384,17 +384,17 @@ static __devinit int tegra_i2s_platform_probe(struct platform_device *pdev) goto err_clk_put; } - i2s->capture_dma_data.addr = mem->start + TEGRA_I2S_FIFO2; + i2s->capture_dma_data.addr = mem->start + TEGRA20_I2S_FIFO2; i2s->capture_dma_data.wrap = 4; i2s->capture_dma_data.width = 32; i2s->capture_dma_data.req_sel = dma_ch; - i2s->playback_dma_data.addr = mem->start + TEGRA_I2S_FIFO1; + i2s->playback_dma_data.addr = mem->start + TEGRA20_I2S_FIFO1; i2s->playback_dma_data.wrap = 4; i2s->playback_dma_data.width = 32; i2s->playback_dma_data.req_sel = dma_ch; - i2s->reg_ctrl = TEGRA_I2S_CTRL_FIFO_FORMAT_PACKED; + i2s->reg_ctrl = TEGRA20_I2S_CTRL_FIFO_FORMAT_PACKED; ret = snd_soc_register_dai(&pdev->dev, &i2s->dai); if (ret) { @@ -409,7 +409,7 @@ static __devinit int tegra_i2s_platform_probe(struct platform_device *pdev) goto err_unregister_dai; } - tegra_i2s_debug_add(i2s); + tegra20_i2s_debug_add(i2s); return 0; @@ -421,38 +421,38 @@ err: return ret; } -static int __devexit tegra_i2s_platform_remove(struct platform_device *pdev) +static int __devexit tegra20_i2s_platform_remove(struct platform_device *pdev) { - struct tegra_i2s *i2s = dev_get_drvdata(&pdev->dev); + struct tegra20_i2s *i2s = dev_get_drvdata(&pdev->dev); tegra_pcm_platform_unregister(&pdev->dev); snd_soc_unregister_dai(&pdev->dev); - tegra_i2s_debug_remove(i2s); + tegra20_i2s_debug_remove(i2s); clk_put(i2s->clk_i2s); return 0; } -static const struct of_device_id tegra_i2s_of_match[] __devinitconst = { +static const struct of_device_id tegra20_i2s_of_match[] __devinitconst = { { .compatible = "nvidia,tegra20-i2s", }, {}, }; -static struct platform_driver tegra_i2s_driver = { +static struct platform_driver tegra20_i2s_driver = { .driver = { .name = DRV_NAME, .owner = THIS_MODULE, - .of_match_table = tegra_i2s_of_match, + .of_match_table = tegra20_i2s_of_match, }, - .probe = tegra_i2s_platform_probe, - .remove = __devexit_p(tegra_i2s_platform_remove), + .probe = tegra20_i2s_platform_probe, + .remove = __devexit_p(tegra20_i2s_platform_remove), }; -module_platform_driver(tegra_i2s_driver); +module_platform_driver(tegra20_i2s_driver); MODULE_AUTHOR("Stephen Warren "); -MODULE_DESCRIPTION("Tegra I2S ASoC driver"); +MODULE_DESCRIPTION("Tegra20 I2S ASoC driver"); MODULE_LICENSE("GPL"); MODULE_ALIAS("platform:" DRV_NAME); -MODULE_DEVICE_TABLE(of, tegra_i2s_of_match); +MODULE_DEVICE_TABLE(of, tegra20_i2s_of_match); diff --git a/sound/soc/tegra/tegra20_i2s.h b/sound/soc/tegra/tegra20_i2s.h index 56f1e0f87c0b..86ab327dbd26 100644 --- a/sound/soc/tegra/tegra20_i2s.h +++ b/sound/soc/tegra/tegra20_i2s.h @@ -2,7 +2,7 @@ * tegra20_i2s.h - Definitions for Tegra20 I2S driver * * Author: Stephen Warren - * Copyright (C) 2010 - NVIDIA, Inc. + * Copyright (C) 2010,2012 - NVIDIA, Inc. * * Based on code copyright/by: * @@ -28,131 +28,131 @@ * */ -#ifndef __TEGRA_I2S_H__ -#define __TEGRA_I2S_H__ +#ifndef __TEGRA20_I2S_H__ +#define __TEGRA20_I2S_H__ #include "tegra_pcm.h" -/* Register offsets from TEGRA_I2S1_BASE and TEGRA_I2S2_BASE */ - -#define TEGRA_I2S_CTRL 0x00 -#define TEGRA_I2S_STATUS 0x04 -#define TEGRA_I2S_TIMING 0x08 -#define TEGRA_I2S_FIFO_SCR 0x0c -#define TEGRA_I2S_PCM_CTRL 0x10 -#define TEGRA_I2S_NW_CTRL 0x14 -#define TEGRA_I2S_TDM_CTRL 0x20 -#define TEGRA_I2S_TDM_TX_RX_CTRL 0x24 -#define TEGRA_I2S_FIFO1 0x40 -#define TEGRA_I2S_FIFO2 0x80 - -/* Fields in TEGRA_I2S_CTRL */ - -#define TEGRA_I2S_CTRL_FIFO2_TX_ENABLE (1 << 30) -#define TEGRA_I2S_CTRL_FIFO1_ENABLE (1 << 29) -#define TEGRA_I2S_CTRL_FIFO2_ENABLE (1 << 28) -#define TEGRA_I2S_CTRL_FIFO1_RX_ENABLE (1 << 27) -#define TEGRA_I2S_CTRL_FIFO_LPBK_ENABLE (1 << 26) -#define TEGRA_I2S_CTRL_MASTER_ENABLE (1 << 25) - -#define TEGRA_I2S_LRCK_LEFT_LOW 0 -#define TEGRA_I2S_LRCK_RIGHT_LOW 1 - -#define TEGRA_I2S_CTRL_LRCK_SHIFT 24 -#define TEGRA_I2S_CTRL_LRCK_MASK (1 << TEGRA_I2S_CTRL_LRCK_SHIFT) -#define TEGRA_I2S_CTRL_LRCK_L_LOW (TEGRA_I2S_LRCK_LEFT_LOW << TEGRA_I2S_CTRL_LRCK_SHIFT) -#define TEGRA_I2S_CTRL_LRCK_R_LOW (TEGRA_I2S_LRCK_RIGHT_LOW << TEGRA_I2S_CTRL_LRCK_SHIFT) - -#define TEGRA_I2S_BIT_FORMAT_I2S 0 -#define TEGRA_I2S_BIT_FORMAT_RJM 1 -#define TEGRA_I2S_BIT_FORMAT_LJM 2 -#define TEGRA_I2S_BIT_FORMAT_DSP 3 - -#define TEGRA_I2S_CTRL_BIT_FORMAT_SHIFT 10 -#define TEGRA_I2S_CTRL_BIT_FORMAT_MASK (3 << TEGRA_I2S_CTRL_BIT_FORMAT_SHIFT) -#define TEGRA_I2S_CTRL_BIT_FORMAT_I2S (TEGRA_I2S_BIT_FORMAT_I2S << TEGRA_I2S_CTRL_BIT_FORMAT_SHIFT) -#define TEGRA_I2S_CTRL_BIT_FORMAT_RJM (TEGRA_I2S_BIT_FORMAT_RJM << TEGRA_I2S_CTRL_BIT_FORMAT_SHIFT) -#define TEGRA_I2S_CTRL_BIT_FORMAT_LJM (TEGRA_I2S_BIT_FORMAT_LJM << TEGRA_I2S_CTRL_BIT_FORMAT_SHIFT) -#define TEGRA_I2S_CTRL_BIT_FORMAT_DSP (TEGRA_I2S_BIT_FORMAT_DSP << TEGRA_I2S_CTRL_BIT_FORMAT_SHIFT) - -#define TEGRA_I2S_BIT_SIZE_16 0 -#define TEGRA_I2S_BIT_SIZE_20 1 -#define TEGRA_I2S_BIT_SIZE_24 2 -#define TEGRA_I2S_BIT_SIZE_32 3 - -#define TEGRA_I2S_CTRL_BIT_SIZE_SHIFT 8 -#define TEGRA_I2S_CTRL_BIT_SIZE_MASK (3 << TEGRA_I2S_CTRL_BIT_SIZE_SHIFT) -#define TEGRA_I2S_CTRL_BIT_SIZE_16 (TEGRA_I2S_BIT_SIZE_16 << TEGRA_I2S_CTRL_BIT_SIZE_SHIFT) -#define TEGRA_I2S_CTRL_BIT_SIZE_20 (TEGRA_I2S_BIT_SIZE_20 << TEGRA_I2S_CTRL_BIT_SIZE_SHIFT) -#define TEGRA_I2S_CTRL_BIT_SIZE_24 (TEGRA_I2S_BIT_SIZE_24 << TEGRA_I2S_CTRL_BIT_SIZE_SHIFT) -#define TEGRA_I2S_CTRL_BIT_SIZE_32 (TEGRA_I2S_BIT_SIZE_32 << TEGRA_I2S_CTRL_BIT_SIZE_SHIFT) - -#define TEGRA_I2S_FIFO_16_LSB 0 -#define TEGRA_I2S_FIFO_20_LSB 1 -#define TEGRA_I2S_FIFO_24_LSB 2 -#define TEGRA_I2S_FIFO_32 3 -#define TEGRA_I2S_FIFO_PACKED 7 - -#define TEGRA_I2S_CTRL_FIFO_FORMAT_SHIFT 4 -#define TEGRA_I2S_CTRL_FIFO_FORMAT_MASK (7 << TEGRA_I2S_CTRL_FIFO_FORMAT_SHIFT) -#define TEGRA_I2S_CTRL_FIFO_FORMAT_16_LSB (TEGRA_I2S_FIFO_16_LSB << TEGRA_I2S_CTRL_FIFO_FORMAT_SHIFT) -#define TEGRA_I2S_CTRL_FIFO_FORMAT_20_LSB (TEGRA_I2S_FIFO_20_LSB << TEGRA_I2S_CTRL_FIFO_FORMAT_SHIFT) -#define TEGRA_I2S_CTRL_FIFO_FORMAT_24_LSB (TEGRA_I2S_FIFO_24_LSB << TEGRA_I2S_CTRL_FIFO_FORMAT_SHIFT) -#define TEGRA_I2S_CTRL_FIFO_FORMAT_32 (TEGRA_I2S_FIFO_32 << TEGRA_I2S_CTRL_FIFO_FORMAT_SHIFT) -#define TEGRA_I2S_CTRL_FIFO_FORMAT_PACKED (TEGRA_I2S_FIFO_PACKED << TEGRA_I2S_CTRL_FIFO_FORMAT_SHIFT) - -#define TEGRA_I2S_CTRL_IE_FIFO1_ERR (1 << 3) -#define TEGRA_I2S_CTRL_IE_FIFO2_ERR (1 << 2) -#define TEGRA_I2S_CTRL_QE_FIFO1 (1 << 1) -#define TEGRA_I2S_CTRL_QE_FIFO2 (1 << 0) - -/* Fields in TEGRA_I2S_STATUS */ - -#define TEGRA_I2S_STATUS_FIFO1_RDY (1 << 31) -#define TEGRA_I2S_STATUS_FIFO2_RDY (1 << 30) -#define TEGRA_I2S_STATUS_FIFO1_BSY (1 << 29) -#define TEGRA_I2S_STATUS_FIFO2_BSY (1 << 28) -#define TEGRA_I2S_STATUS_FIFO1_ERR (1 << 3) -#define TEGRA_I2S_STATUS_FIFO2_ERR (1 << 2) -#define TEGRA_I2S_STATUS_QS_FIFO1 (1 << 1) -#define TEGRA_I2S_STATUS_QS_FIFO2 (1 << 0) - -/* Fields in TEGRA_I2S_TIMING */ - -#define TEGRA_I2S_TIMING_NON_SYM_ENABLE (1 << 12) -#define TEGRA_I2S_TIMING_CHANNEL_BIT_COUNT_SHIFT 0 -#define TEGRA_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US 0x7fff -#define TEGRA_I2S_TIMING_CHANNEL_BIT_COUNT_MASK (TEGRA_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US << TEGRA_I2S_TIMING_CHANNEL_BIT_COUNT_SHIFT) - -/* Fields in TEGRA_I2S_FIFO_SCR */ - -#define TEGRA_I2S_FIFO_SCR_FIFO2_FULL_EMPTY_COUNT_SHIFT 24 -#define TEGRA_I2S_FIFO_SCR_FIFO1_FULL_EMPTY_COUNT_SHIFT 16 -#define TEGRA_I2S_FIFO_SCR_FIFO_FULL_EMPTY_COUNT_MASK 0x3f - -#define TEGRA_I2S_FIFO_SCR_FIFO2_CLR (1 << 12) -#define TEGRA_I2S_FIFO_SCR_FIFO1_CLR (1 << 8) - -#define TEGRA_I2S_FIFO_ATN_LVL_ONE_SLOT 0 -#define TEGRA_I2S_FIFO_ATN_LVL_FOUR_SLOTS 1 -#define TEGRA_I2S_FIFO_ATN_LVL_EIGHT_SLOTS 2 -#define TEGRA_I2S_FIFO_ATN_LVL_TWELVE_SLOTS 3 - -#define TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_SHIFT 4 -#define TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_MASK (3 << TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_SHIFT) -#define TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_ONE_SLOT (TEGRA_I2S_FIFO_ATN_LVL_ONE_SLOT << TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_SHIFT) -#define TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_FOUR_SLOTS (TEGRA_I2S_FIFO_ATN_LVL_FOUR_SLOTS << TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_SHIFT) -#define TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_EIGHT_SLOTS (TEGRA_I2S_FIFO_ATN_LVL_EIGHT_SLOTS << TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_SHIFT) -#define TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_TWELVE_SLOTS (TEGRA_I2S_FIFO_ATN_LVL_TWELVE_SLOTS << TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_SHIFT) - -#define TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_SHIFT 0 -#define TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_MASK (3 << TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_SHIFT) -#define TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_ONE_SLOT (TEGRA_I2S_FIFO_ATN_LVL_ONE_SLOT << TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_SHIFT) -#define TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_FOUR_SLOTS (TEGRA_I2S_FIFO_ATN_LVL_FOUR_SLOTS << TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_SHIFT) -#define TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_EIGHT_SLOTS (TEGRA_I2S_FIFO_ATN_LVL_EIGHT_SLOTS << TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_SHIFT) -#define TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_TWELVE_SLOTS (TEGRA_I2S_FIFO_ATN_LVL_TWELVE_SLOTS << TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_SHIFT) - -struct tegra_i2s { +/* Register offsets from TEGRA20_I2S1_BASE and TEGRA20_I2S2_BASE */ + +#define TEGRA20_I2S_CTRL 0x00 +#define TEGRA20_I2S_STATUS 0x04 +#define TEGRA20_I2S_TIMING 0x08 +#define TEGRA20_I2S_FIFO_SCR 0x0c +#define TEGRA20_I2S_PCM_CTRL 0x10 +#define TEGRA20_I2S_NW_CTRL 0x14 +#define TEGRA20_I2S_TDM_CTRL 0x20 +#define TEGRA20_I2S_TDM_TX_RX_CTRL 0x24 +#define TEGRA20_I2S_FIFO1 0x40 +#define TEGRA20_I2S_FIFO2 0x80 + +/* Fields in TEGRA20_I2S_CTRL */ + +#define TEGRA20_I2S_CTRL_FIFO2_TX_ENABLE (1 << 30) +#define TEGRA20_I2S_CTRL_FIFO1_ENABLE (1 << 29) +#define TEGRA20_I2S_CTRL_FIFO2_ENABLE (1 << 28) +#define TEGRA20_I2S_CTRL_FIFO1_RX_ENABLE (1 << 27) +#define TEGRA20_I2S_CTRL_FIFO_LPBK_ENABLE (1 << 26) +#define TEGRA20_I2S_CTRL_MASTER_ENABLE (1 << 25) + +#define TEGRA20_I2S_LRCK_LEFT_LOW 0 +#define TEGRA20_I2S_LRCK_RIGHT_LOW 1 + +#define TEGRA20_I2S_CTRL_LRCK_SHIFT 24 +#define TEGRA20_I2S_CTRL_LRCK_MASK (1 << TEGRA20_I2S_CTRL_LRCK_SHIFT) +#define TEGRA20_I2S_CTRL_LRCK_L_LOW (TEGRA20_I2S_LRCK_LEFT_LOW << TEGRA20_I2S_CTRL_LRCK_SHIFT) +#define TEGRA20_I2S_CTRL_LRCK_R_LOW (TEGRA20_I2S_LRCK_RIGHT_LOW << TEGRA20_I2S_CTRL_LRCK_SHIFT) + +#define TEGRA20_I2S_BIT_FORMAT_I2S 0 +#define TEGRA20_I2S_BIT_FORMAT_RJM 1 +#define TEGRA20_I2S_BIT_FORMAT_LJM 2 +#define TEGRA20_I2S_BIT_FORMAT_DSP 3 + +#define TEGRA20_I2S_CTRL_BIT_FORMAT_SHIFT 10 +#define TEGRA20_I2S_CTRL_BIT_FORMAT_MASK (3 << TEGRA20_I2S_CTRL_BIT_FORMAT_SHIFT) +#define TEGRA20_I2S_CTRL_BIT_FORMAT_I2S (TEGRA20_I2S_BIT_FORMAT_I2S << TEGRA20_I2S_CTRL_BIT_FORMAT_SHIFT) +#define TEGRA20_I2S_CTRL_BIT_FORMAT_RJM (TEGRA20_I2S_BIT_FORMAT_RJM << TEGRA20_I2S_CTRL_BIT_FORMAT_SHIFT) +#define TEGRA20_I2S_CTRL_BIT_FORMAT_LJM (TEGRA20_I2S_BIT_FORMAT_LJM << TEGRA20_I2S_CTRL_BIT_FORMAT_SHIFT) +#define TEGRA20_I2S_CTRL_BIT_FORMAT_DSP (TEGRA20_I2S_BIT_FORMAT_DSP << TEGRA20_I2S_CTRL_BIT_FORMAT_SHIFT) + +#define TEGRA20_I2S_BIT_SIZE_16 0 +#define TEGRA20_I2S_BIT_SIZE_20 1 +#define TEGRA20_I2S_BIT_SIZE_24 2 +#define TEGRA20_I2S_BIT_SIZE_32 3 + +#define TEGRA20_I2S_CTRL_BIT_SIZE_SHIFT 8 +#define TEGRA20_I2S_CTRL_BIT_SIZE_MASK (3 << TEGRA20_I2S_CTRL_BIT_SIZE_SHIFT) +#define TEGRA20_I2S_CTRL_BIT_SIZE_16 (TEGRA20_I2S_BIT_SIZE_16 << TEGRA20_I2S_CTRL_BIT_SIZE_SHIFT) +#define TEGRA20_I2S_CTRL_BIT_SIZE_20 (TEGRA20_I2S_BIT_SIZE_20 << TEGRA20_I2S_CTRL_BIT_SIZE_SHIFT) +#define TEGRA20_I2S_CTRL_BIT_SIZE_24 (TEGRA20_I2S_BIT_SIZE_24 << TEGRA20_I2S_CTRL_BIT_SIZE_SHIFT) +#define TEGRA20_I2S_CTRL_BIT_SIZE_32 (TEGRA20_I2S_BIT_SIZE_32 << TEGRA20_I2S_CTRL_BIT_SIZE_SHIFT) + +#define TEGRA20_I2S_FIFO_16_LSB 0 +#define TEGRA20_I2S_FIFO_20_LSB 1 +#define TEGRA20_I2S_FIFO_24_LSB 2 +#define TEGRA20_I2S_FIFO_32 3 +#define TEGRA20_I2S_FIFO_PACKED 7 + +#define TEGRA20_I2S_CTRL_FIFO_FORMAT_SHIFT 4 +#define TEGRA20_I2S_CTRL_FIFO_FORMAT_MASK (7 << TEGRA20_I2S_CTRL_FIFO_FORMAT_SHIFT) +#define TEGRA20_I2S_CTRL_FIFO_FORMAT_16_LSB (TEGRA20_I2S_FIFO_16_LSB << TEGRA20_I2S_CTRL_FIFO_FORMAT_SHIFT) +#define TEGRA20_I2S_CTRL_FIFO_FORMAT_20_LSB (TEGRA20_I2S_FIFO_20_LSB << TEGRA20_I2S_CTRL_FIFO_FORMAT_SHIFT) +#define TEGRA20_I2S_CTRL_FIFO_FORMAT_24_LSB (TEGRA20_I2S_FIFO_24_LSB << TEGRA20_I2S_CTRL_FIFO_FORMAT_SHIFT) +#define TEGRA20_I2S_CTRL_FIFO_FORMAT_32 (TEGRA20_I2S_FIFO_32 << TEGRA20_I2S_CTRL_FIFO_FORMAT_SHIFT) +#define TEGRA20_I2S_CTRL_FIFO_FORMAT_PACKED (TEGRA20_I2S_FIFO_PACKED << TEGRA20_I2S_CTRL_FIFO_FORMAT_SHIFT) + +#define TEGRA20_I2S_CTRL_IE_FIFO1_ERR (1 << 3) +#define TEGRA20_I2S_CTRL_IE_FIFO2_ERR (1 << 2) +#define TEGRA20_I2S_CTRL_QE_FIFO1 (1 << 1) +#define TEGRA20_I2S_CTRL_QE_FIFO2 (1 << 0) + +/* Fields in TEGRA20_I2S_STATUS */ + +#define TEGRA20_I2S_STATUS_FIFO1_RDY (1 << 31) +#define TEGRA20_I2S_STATUS_FIFO2_RDY (1 << 30) +#define TEGRA20_I2S_STATUS_FIFO1_BSY (1 << 29) +#define TEGRA20_I2S_STATUS_FIFO2_BSY (1 << 28) +#define TEGRA20_I2S_STATUS_FIFO1_ERR (1 << 3) +#define TEGRA20_I2S_STATUS_FIFO2_ERR (1 << 2) +#define TEGRA20_I2S_STATUS_QS_FIFO1 (1 << 1) +#define TEGRA20_I2S_STATUS_QS_FIFO2 (1 << 0) + +/* Fields in TEGRA20_I2S_TIMING */ + +#define TEGRA20_I2S_TIMING_NON_SYM_ENABLE (1 << 12) +#define TEGRA20_I2S_TIMING_CHANNEL_BIT_COUNT_SHIFT 0 +#define TEGRA20_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US 0x7fff +#define TEGRA20_I2S_TIMING_CHANNEL_BIT_COUNT_MASK (TEGRA20_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US << TEGRA20_I2S_TIMING_CHANNEL_BIT_COUNT_SHIFT) + +/* Fields in TEGRA20_I2S_FIFO_SCR */ + +#define TEGRA20_I2S_FIFO_SCR_FIFO2_FULL_EMPTY_COUNT_SHIFT 24 +#define TEGRA20_I2S_FIFO_SCR_FIFO1_FULL_EMPTY_COUNT_SHIFT 16 +#define TEGRA20_I2S_FIFO_SCR_FIFO_FULL_EMPTY_COUNT_MASK 0x3f + +#define TEGRA20_I2S_FIFO_SCR_FIFO2_CLR (1 << 12) +#define TEGRA20_I2S_FIFO_SCR_FIFO1_CLR (1 << 8) + +#define TEGRA20_I2S_FIFO_ATN_LVL_ONE_SLOT 0 +#define TEGRA20_I2S_FIFO_ATN_LVL_FOUR_SLOTS 1 +#define TEGRA20_I2S_FIFO_ATN_LVL_EIGHT_SLOTS 2 +#define TEGRA20_I2S_FIFO_ATN_LVL_TWELVE_SLOTS 3 + +#define TEGRA20_I2S_FIFO_SCR_FIFO2_ATN_LVL_SHIFT 4 +#define TEGRA20_I2S_FIFO_SCR_FIFO2_ATN_LVL_MASK (3 << TEGRA20_I2S_FIFO_SCR_FIFO2_ATN_LVL_SHIFT) +#define TEGRA20_I2S_FIFO_SCR_FIFO2_ATN_LVL_ONE_SLOT (TEGRA20_I2S_FIFO_ATN_LVL_ONE_SLOT << TEGRA20_I2S_FIFO_SCR_FIFO2_ATN_LVL_SHIFT) +#define TEGRA20_I2S_FIFO_SCR_FIFO2_ATN_LVL_FOUR_SLOTS (TEGRA20_I2S_FIFO_ATN_LVL_FOUR_SLOTS << TEGRA20_I2S_FIFO_SCR_FIFO2_ATN_LVL_SHIFT) +#define TEGRA20_I2S_FIFO_SCR_FIFO2_ATN_LVL_EIGHT_SLOTS (TEGRA20_I2S_FIFO_ATN_LVL_EIGHT_SLOTS << TEGRA20_I2S_FIFO_SCR_FIFO2_ATN_LVL_SHIFT) +#define TEGRA20_I2S_FIFO_SCR_FIFO2_ATN_LVL_TWELVE_SLOTS (TEGRA20_I2S_FIFO_ATN_LVL_TWELVE_SLOTS << TEGRA20_I2S_FIFO_SCR_FIFO2_ATN_LVL_SHIFT) + +#define TEGRA20_I2S_FIFO_SCR_FIFO1_ATN_LVL_SHIFT 0 +#define TEGRA20_I2S_FIFO_SCR_FIFO1_ATN_LVL_MASK (3 << TEGRA20_I2S_FIFO_SCR_FIFO1_ATN_LVL_SHIFT) +#define TEGRA20_I2S_FIFO_SCR_FIFO1_ATN_LVL_ONE_SLOT (TEGRA20_I2S_FIFO_ATN_LVL_ONE_SLOT << TEGRA20_I2S_FIFO_SCR_FIFO1_ATN_LVL_SHIFT) +#define TEGRA20_I2S_FIFO_SCR_FIFO1_ATN_LVL_FOUR_SLOTS (TEGRA20_I2S_FIFO_ATN_LVL_FOUR_SLOTS << TEGRA20_I2S_FIFO_SCR_FIFO1_ATN_LVL_SHIFT) +#define TEGRA20_I2S_FIFO_SCR_FIFO1_ATN_LVL_EIGHT_SLOTS (TEGRA20_I2S_FIFO_ATN_LVL_EIGHT_SLOTS << TEGRA20_I2S_FIFO_SCR_FIFO1_ATN_LVL_SHIFT) +#define TEGRA20_I2S_FIFO_SCR_FIFO1_ATN_LVL_TWELVE_SLOTS (TEGRA20_I2S_FIFO_ATN_LVL_TWELVE_SLOTS << TEGRA20_I2S_FIFO_SCR_FIFO1_ATN_LVL_SHIFT) + +struct tegra20_i2s { struct snd_soc_dai_driver dai; struct clk *clk_i2s; struct tegra_pcm_dma_params capture_dma_data; diff --git a/sound/soc/tegra/tegra20_spdif.c b/sound/soc/tegra/tegra20_spdif.c index ed1fd502ef8d..052bff816c4d 100644 --- a/sound/soc/tegra/tegra20_spdif.c +++ b/sound/soc/tegra/tegra20_spdif.c @@ -36,105 +36,105 @@ #include "tegra20_spdif.h" -#define DRV_NAME "tegra-spdif" +#define DRV_NAME "tegra20-spdif" -static inline void tegra_spdif_write(struct tegra_spdif *spdif, u32 reg, +static inline void tegra20_spdif_write(struct tegra20_spdif *spdif, u32 reg, u32 val) { __raw_writel(val, spdif->regs + reg); } -static inline u32 tegra_spdif_read(struct tegra_spdif *spdif, u32 reg) +static inline u32 tegra20_spdif_read(struct tegra20_spdif *spdif, u32 reg) { return __raw_readl(spdif->regs + reg); } #ifdef CONFIG_DEBUG_FS -static int tegra_spdif_show(struct seq_file *s, void *unused) +static int tegra20_spdif_show(struct seq_file *s, void *unused) { #define REG(r) { r, #r } static const struct { int offset; const char *name; } regs[] = { - REG(TEGRA_SPDIF_CTRL), - REG(TEGRA_SPDIF_STATUS), - REG(TEGRA_SPDIF_STROBE_CTRL), - REG(TEGRA_SPDIF_DATA_FIFO_CSR), - REG(TEGRA_SPDIF_CH_STA_RX_A), - REG(TEGRA_SPDIF_CH_STA_RX_B), - REG(TEGRA_SPDIF_CH_STA_RX_C), - REG(TEGRA_SPDIF_CH_STA_RX_D), - REG(TEGRA_SPDIF_CH_STA_RX_E), - REG(TEGRA_SPDIF_CH_STA_RX_F), - REG(TEGRA_SPDIF_CH_STA_TX_A), - REG(TEGRA_SPDIF_CH_STA_TX_B), - REG(TEGRA_SPDIF_CH_STA_TX_C), - REG(TEGRA_SPDIF_CH_STA_TX_D), - REG(TEGRA_SPDIF_CH_STA_TX_E), - REG(TEGRA_SPDIF_CH_STA_TX_F), + REG(TEGRA20_SPDIF_CTRL), + REG(TEGRA20_SPDIF_STATUS), + REG(TEGRA20_SPDIF_STROBE_CTRL), + REG(TEGRA20_SPDIF_DATA_FIFO_CSR), + REG(TEGRA20_SPDIF_CH_STA_RX_A), + REG(TEGRA20_SPDIF_CH_STA_RX_B), + REG(TEGRA20_SPDIF_CH_STA_RX_C), + REG(TEGRA20_SPDIF_CH_STA_RX_D), + REG(TEGRA20_SPDIF_CH_STA_RX_E), + REG(TEGRA20_SPDIF_CH_STA_RX_F), + REG(TEGRA20_SPDIF_CH_STA_TX_A), + REG(TEGRA20_SPDIF_CH_STA_TX_B), + REG(TEGRA20_SPDIF_CH_STA_TX_C), + REG(TEGRA20_SPDIF_CH_STA_TX_D), + REG(TEGRA20_SPDIF_CH_STA_TX_E), + REG(TEGRA20_SPDIF_CH_STA_TX_F), }; #undef REG - struct tegra_spdif *spdif = s->private; + struct tegra20_spdif *spdif = s->private; int i; for (i = 0; i < ARRAY_SIZE(regs); i++) { - u32 val = tegra_spdif_read(spdif, regs[i].offset); + u32 val = tegra20_spdif_read(spdif, regs[i].offset); seq_printf(s, "%s = %08x\n", regs[i].name, val); } return 0; } -static int tegra_spdif_debug_open(struct inode *inode, struct file *file) +static int tegra20_spdif_debug_open(struct inode *inode, struct file *file) { - return single_open(file, tegra_spdif_show, inode->i_private); + return single_open(file, tegra20_spdif_show, inode->i_private); } -static const struct file_operations tegra_spdif_debug_fops = { - .open = tegra_spdif_debug_open, +static const struct file_operations tegra20_spdif_debug_fops = { + .open = tegra20_spdif_debug_open, .read = seq_read, .llseek = seq_lseek, .release = single_release, }; -static void tegra_spdif_debug_add(struct tegra_spdif *spdif) +static void tegra20_spdif_debug_add(struct tegra20_spdif *spdif) { spdif->debug = debugfs_create_file(DRV_NAME, S_IRUGO, snd_soc_debugfs_root, spdif, - &tegra_spdif_debug_fops); + &tegra20_spdif_debug_fops); } -static void tegra_spdif_debug_remove(struct tegra_spdif *spdif) +static void tegra20_spdif_debug_remove(struct tegra20_spdif *spdif) { if (spdif->debug) debugfs_remove(spdif->debug); } #else -static inline void tegra_spdif_debug_add(struct tegra_spdif *spdif) +static inline void tegra20_spdif_debug_add(struct tegra20_spdif *spdif) { } -static inline void tegra_spdif_debug_remove(struct tegra_spdif *spdif) +static inline void tegra20_spdif_debug_remove(struct tegra20_spdif *spdif) { } #endif -static int tegra_spdif_hw_params(struct snd_pcm_substream *substream, +static int tegra20_spdif_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { struct device *dev = substream->pcm->card->dev; - struct tegra_spdif *spdif = snd_soc_dai_get_drvdata(dai); + struct tegra20_spdif *spdif = snd_soc_dai_get_drvdata(dai); int ret, spdifclock; - spdif->reg_ctrl &= ~TEGRA_SPDIF_CTRL_PACK; - spdif->reg_ctrl &= ~TEGRA_SPDIF_CTRL_BIT_MODE_MASK; + spdif->reg_ctrl &= ~TEGRA20_SPDIF_CTRL_PACK; + spdif->reg_ctrl &= ~TEGRA20_SPDIF_CTRL_BIT_MODE_MASK; switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: - spdif->reg_ctrl |= TEGRA_SPDIF_CTRL_PACK; - spdif->reg_ctrl |= TEGRA_SPDIF_CTRL_BIT_MODE_16BIT; + spdif->reg_ctrl |= TEGRA20_SPDIF_CTRL_PACK; + spdif->reg_ctrl |= TEGRA20_SPDIF_CTRL_BIT_MODE_16BIT; break; default: return -EINVAL; @@ -175,34 +175,34 @@ static int tegra_spdif_hw_params(struct snd_pcm_substream *substream, return 0; } -static void tegra_spdif_start_playback(struct tegra_spdif *spdif) +static void tegra20_spdif_start_playback(struct tegra20_spdif *spdif) { - spdif->reg_ctrl |= TEGRA_SPDIF_CTRL_TX_EN; - tegra_spdif_write(spdif, TEGRA_SPDIF_CTRL, spdif->reg_ctrl); + spdif->reg_ctrl |= TEGRA20_SPDIF_CTRL_TX_EN; + tegra20_spdif_write(spdif, TEGRA20_SPDIF_CTRL, spdif->reg_ctrl); } -static void tegra_spdif_stop_playback(struct tegra_spdif *spdif) +static void tegra20_spdif_stop_playback(struct tegra20_spdif *spdif) { - spdif->reg_ctrl &= ~TEGRA_SPDIF_CTRL_TX_EN; - tegra_spdif_write(spdif, TEGRA_SPDIF_CTRL, spdif->reg_ctrl); + spdif->reg_ctrl &= ~TEGRA20_SPDIF_CTRL_TX_EN; + tegra20_spdif_write(spdif, TEGRA20_SPDIF_CTRL, spdif->reg_ctrl); } -static int tegra_spdif_trigger(struct snd_pcm_substream *substream, int cmd, +static int tegra20_spdif_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { - struct tegra_spdif *spdif = snd_soc_dai_get_drvdata(dai); + struct tegra20_spdif *spdif = snd_soc_dai_get_drvdata(dai); switch (cmd) { case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: case SNDRV_PCM_TRIGGER_RESUME: clk_enable(spdif->clk_spdif_out); - tegra_spdif_start_playback(spdif); + tegra20_spdif_start_playback(spdif); break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: case SNDRV_PCM_TRIGGER_SUSPEND: - tegra_spdif_stop_playback(spdif); + tegra20_spdif_stop_playback(spdif); clk_disable(spdif->clk_spdif_out); break; default: @@ -212,9 +212,9 @@ static int tegra_spdif_trigger(struct snd_pcm_substream *substream, int cmd, return 0; } -static int tegra_spdif_probe(struct snd_soc_dai *dai) +static int tegra20_spdif_probe(struct snd_soc_dai *dai) { - struct tegra_spdif *spdif = snd_soc_dai_get_drvdata(dai); + struct tegra20_spdif *spdif = snd_soc_dai_get_drvdata(dai); dai->capture_dma_data = NULL; dai->playback_dma_data = &spdif->playback_dma_data; @@ -222,14 +222,14 @@ static int tegra_spdif_probe(struct snd_soc_dai *dai) return 0; } -static const struct snd_soc_dai_ops tegra_spdif_dai_ops = { - .hw_params = tegra_spdif_hw_params, - .trigger = tegra_spdif_trigger, +static const struct snd_soc_dai_ops tegra20_spdif_dai_ops = { + .hw_params = tegra20_spdif_hw_params, + .trigger = tegra20_spdif_trigger, }; -static struct snd_soc_dai_driver tegra_spdif_dai = { +static struct snd_soc_dai_driver tegra20_spdif_dai = { .name = DRV_NAME, - .probe = tegra_spdif_probe, + .probe = tegra20_spdif_probe, .playback = { .channels_min = 2, .channels_max = 2, @@ -237,18 +237,18 @@ static struct snd_soc_dai_driver tegra_spdif_dai = { SNDRV_PCM_RATE_48000, .formats = SNDRV_PCM_FMTBIT_S16_LE, }, - .ops = &tegra_spdif_dai_ops, + .ops = &tegra20_spdif_dai_ops, }; -static __devinit int tegra_spdif_platform_probe(struct platform_device *pdev) +static __devinit int tegra20_spdif_platform_probe(struct platform_device *pdev) { - struct tegra_spdif *spdif; + struct tegra20_spdif *spdif; struct resource *mem, *memregion, *dmareq; int ret; - spdif = kzalloc(sizeof(struct tegra_spdif), GFP_KERNEL); + spdif = kzalloc(sizeof(struct tegra20_spdif), GFP_KERNEL); if (!spdif) { - dev_err(&pdev->dev, "Can't allocate tegra_spdif\n"); + dev_err(&pdev->dev, "Can't allocate tegra20_spdif\n"); ret = -ENOMEM; goto exit; } @@ -290,12 +290,12 @@ static __devinit int tegra_spdif_platform_probe(struct platform_device *pdev) goto err_release; } - spdif->playback_dma_data.addr = mem->start + TEGRA_SPDIF_DATA_OUT; + spdif->playback_dma_data.addr = mem->start + TEGRA20_SPDIF_DATA_OUT; spdif->playback_dma_data.wrap = 4; spdif->playback_dma_data.width = 32; spdif->playback_dma_data.req_sel = dmareq->start; - ret = snd_soc_register_dai(&pdev->dev, &tegra_spdif_dai); + ret = snd_soc_register_dai(&pdev->dev, &tegra20_spdif_dai); if (ret) { dev_err(&pdev->dev, "Could not register DAI: %d\n", ret); ret = -ENOMEM; @@ -308,7 +308,7 @@ static __devinit int tegra_spdif_platform_probe(struct platform_device *pdev) goto err_unregister_dai; } - tegra_spdif_debug_add(spdif); + tegra20_spdif_debug_add(spdif); return 0; @@ -326,15 +326,15 @@ exit: return ret; } -static int __devexit tegra_spdif_platform_remove(struct platform_device *pdev) +static int __devexit tegra20_spdif_platform_remove(struct platform_device *pdev) { - struct tegra_spdif *spdif = dev_get_drvdata(&pdev->dev); + struct tegra20_spdif *spdif = dev_get_drvdata(&pdev->dev); struct resource *res; tegra_pcm_platform_unregister(&pdev->dev); snd_soc_unregister_dai(&pdev->dev); - tegra_spdif_debug_remove(spdif); + tegra20_spdif_debug_remove(spdif); iounmap(spdif->regs); @@ -348,18 +348,18 @@ static int __devexit tegra_spdif_platform_remove(struct platform_device *pdev) return 0; } -static struct platform_driver tegra_spdif_driver = { +static struct platform_driver tegra20_spdif_driver = { .driver = { .name = DRV_NAME, .owner = THIS_MODULE, }, - .probe = tegra_spdif_platform_probe, - .remove = __devexit_p(tegra_spdif_platform_remove), + .probe = tegra20_spdif_platform_probe, + .remove = __devexit_p(tegra20_spdif_platform_remove), }; -module_platform_driver(tegra_spdif_driver); +module_platform_driver(tegra20_spdif_driver); MODULE_AUTHOR("Stephen Warren "); -MODULE_DESCRIPTION("Tegra SPDIF ASoC driver"); +MODULE_DESCRIPTION("Tegra20 SPDIF ASoC driver"); MODULE_LICENSE("GPL"); MODULE_ALIAS("platform:" DRV_NAME); diff --git a/sound/soc/tegra/tegra20_spdif.h b/sound/soc/tegra/tegra20_spdif.h index 001f9dcd2033..823af4c6bb7c 100644 --- a/sound/soc/tegra/tegra20_spdif.h +++ b/sound/soc/tegra/tegra20_spdif.h @@ -23,83 +23,83 @@ * */ -#ifndef __TEGRA_SPDIF_H__ -#define __TEGRA_SPDIF_H__ +#ifndef __TEGRA20_SPDIF_H__ +#define __TEGRA20_SPDIF_H__ #include "tegra_pcm.h" -/* Offsets from TEGRA_SPDIF_BASE */ - -#define TEGRA_SPDIF_CTRL 0x0 -#define TEGRA_SPDIF_STATUS 0x4 -#define TEGRA_SPDIF_STROBE_CTRL 0x8 -#define TEGRA_SPDIF_DATA_FIFO_CSR 0x0C -#define TEGRA_SPDIF_DATA_OUT 0x40 -#define TEGRA_SPDIF_DATA_IN 0x80 -#define TEGRA_SPDIF_CH_STA_RX_A 0x100 -#define TEGRA_SPDIF_CH_STA_RX_B 0x104 -#define TEGRA_SPDIF_CH_STA_RX_C 0x108 -#define TEGRA_SPDIF_CH_STA_RX_D 0x10C -#define TEGRA_SPDIF_CH_STA_RX_E 0x110 -#define TEGRA_SPDIF_CH_STA_RX_F 0x114 -#define TEGRA_SPDIF_CH_STA_TX_A 0x140 -#define TEGRA_SPDIF_CH_STA_TX_B 0x144 -#define TEGRA_SPDIF_CH_STA_TX_C 0x148 -#define TEGRA_SPDIF_CH_STA_TX_D 0x14C -#define TEGRA_SPDIF_CH_STA_TX_E 0x150 -#define TEGRA_SPDIF_CH_STA_TX_F 0x154 -#define TEGRA_SPDIF_USR_STA_RX_A 0x180 -#define TEGRA_SPDIF_USR_DAT_TX_A 0x1C0 - -/* Fields in TEGRA_SPDIF_CTRL */ +/* Offsets from TEGRA20_SPDIF_BASE */ + +#define TEGRA20_SPDIF_CTRL 0x0 +#define TEGRA20_SPDIF_STATUS 0x4 +#define TEGRA20_SPDIF_STROBE_CTRL 0x8 +#define TEGRA20_SPDIF_DATA_FIFO_CSR 0x0C +#define TEGRA20_SPDIF_DATA_OUT 0x40 +#define TEGRA20_SPDIF_DATA_IN 0x80 +#define TEGRA20_SPDIF_CH_STA_RX_A 0x100 +#define TEGRA20_SPDIF_CH_STA_RX_B 0x104 +#define TEGRA20_SPDIF_CH_STA_RX_C 0x108 +#define TEGRA20_SPDIF_CH_STA_RX_D 0x10C +#define TEGRA20_SPDIF_CH_STA_RX_E 0x110 +#define TEGRA20_SPDIF_CH_STA_RX_F 0x114 +#define TEGRA20_SPDIF_CH_STA_TX_A 0x140 +#define TEGRA20_SPDIF_CH_STA_TX_B 0x144 +#define TEGRA20_SPDIF_CH_STA_TX_C 0x148 +#define TEGRA20_SPDIF_CH_STA_TX_D 0x14C +#define TEGRA20_SPDIF_CH_STA_TX_E 0x150 +#define TEGRA20_SPDIF_CH_STA_TX_F 0x154 +#define TEGRA20_SPDIF_USR_STA_RX_A 0x180 +#define TEGRA20_SPDIF_USR_DAT_TX_A 0x1C0 + +/* Fields in TEGRA20_SPDIF_CTRL */ /* Start capturing from 0=right, 1=left channel */ -#define TEGRA_SPDIF_CTRL_CAP_LC (1 << 30) +#define TEGRA20_SPDIF_CTRL_CAP_LC (1 << 30) /* SPDIF receiver(RX) enable */ -#define TEGRA_SPDIF_CTRL_RX_EN (1 << 29) +#define TEGRA20_SPDIF_CTRL_RX_EN (1 << 29) /* SPDIF Transmitter(TX) enable */ -#define TEGRA_SPDIF_CTRL_TX_EN (1 << 28) +#define TEGRA20_SPDIF_CTRL_TX_EN (1 << 28) /* Transmit Channel status */ -#define TEGRA_SPDIF_CTRL_TC_EN (1 << 27) +#define TEGRA20_SPDIF_CTRL_TC_EN (1 << 27) /* Transmit user Data */ -#define TEGRA_SPDIF_CTRL_TU_EN (1 << 26) +#define TEGRA20_SPDIF_CTRL_TU_EN (1 << 26) /* Interrupt on transmit error */ -#define TEGRA_SPDIF_CTRL_IE_TXE (1 << 25) +#define TEGRA20_SPDIF_CTRL_IE_TXE (1 << 25) /* Interrupt on receive error */ -#define TEGRA_SPDIF_CTRL_IE_RXE (1 << 24) +#define TEGRA20_SPDIF_CTRL_IE_RXE (1 << 24) /* Interrupt on invalid preamble */ -#define TEGRA_SPDIF_CTRL_IE_P (1 << 23) +#define TEGRA20_SPDIF_CTRL_IE_P (1 << 23) /* Interrupt on "B" preamble */ -#define TEGRA_SPDIF_CTRL_IE_B (1 << 22) +#define TEGRA20_SPDIF_CTRL_IE_B (1 << 22) /* Interrupt when block of channel status received */ -#define TEGRA_SPDIF_CTRL_IE_C (1 << 21) +#define TEGRA20_SPDIF_CTRL_IE_C (1 << 21) /* Interrupt when a valid information unit (IU) is received */ -#define TEGRA_SPDIF_CTRL_IE_U (1 << 20) +#define TEGRA20_SPDIF_CTRL_IE_U (1 << 20) /* Interrupt when RX user FIFO attention level is reached */ -#define TEGRA_SPDIF_CTRL_QE_RU (1 << 19) +#define TEGRA20_SPDIF_CTRL_QE_RU (1 << 19) /* Interrupt when TX user FIFO attention level is reached */ -#define TEGRA_SPDIF_CTRL_QE_TU (1 << 18) +#define TEGRA20_SPDIF_CTRL_QE_TU (1 << 18) /* Interrupt when RX data FIFO attention level is reached */ -#define TEGRA_SPDIF_CTRL_QE_RX (1 << 17) +#define TEGRA20_SPDIF_CTRL_QE_RX (1 << 17) /* Interrupt when TX data FIFO attention level is reached */ -#define TEGRA_SPDIF_CTRL_QE_TX (1 << 16) +#define TEGRA20_SPDIF_CTRL_QE_TX (1 << 16) /* Loopback test mode enable */ -#define TEGRA_SPDIF_CTRL_LBK_EN (1 << 15) +#define TEGRA20_SPDIF_CTRL_LBK_EN (1 << 15) /* * Pack data mode: @@ -107,7 +107,7 @@ * interface data bit size). * 1 = Packeted left/right channel data into a single word. */ -#define TEGRA_SPDIF_CTRL_PACK (1 << 14) +#define TEGRA20_SPDIF_CTRL_PACK (1 << 14) /* * 00 = 16bit data @@ -115,19 +115,19 @@ * 10 = 24bit data * 11 = raw data */ -#define TEGRA_SPDIF_BIT_MODE_16BIT 0 -#define TEGRA_SPDIF_BIT_MODE_20BIT 1 -#define TEGRA_SPDIF_BIT_MODE_24BIT 2 -#define TEGRA_SPDIF_BIT_MODE_RAW 3 +#define TEGRA20_SPDIF_BIT_MODE_16BIT 0 +#define TEGRA20_SPDIF_BIT_MODE_20BIT 1 +#define TEGRA20_SPDIF_BIT_MODE_24BIT 2 +#define TEGRA20_SPDIF_BIT_MODE_RAW 3 -#define TEGRA_SPDIF_CTRL_BIT_MODE_SHIFT 12 -#define TEGRA_SPDIF_CTRL_BIT_MODE_MASK (3 << TEGRA_SPDIF_CTRL_BIT_MODE_SHIFT) -#define TEGRA_SPDIF_CTRL_BIT_MODE_16BIT (TEGRA_SPDIF_BIT_MODE_16BIT << TEGRA_SPDIF_CTRL_BIT_MODE_SHIFT) -#define TEGRA_SPDIF_CTRL_BIT_MODE_20BIT (TEGRA_SPDIF_BIT_MODE_20BIT << TEGRA_SPDIF_CTRL_BIT_MODE_SHIFT) -#define TEGRA_SPDIF_CTRL_BIT_MODE_24BIT (TEGRA_SPDIF_BIT_MODE_24BIT << TEGRA_SPDIF_CTRL_BIT_MODE_SHIFT) -#define TEGRA_SPDIF_CTRL_BIT_MODE_RAW (TEGRA_SPDIF_BIT_MODE_RAW << TEGRA_SPDIF_CTRL_BIT_MODE_SHIFT) +#define TEGRA20_SPDIF_CTRL_BIT_MODE_SHIFT 12 +#define TEGRA20_SPDIF_CTRL_BIT_MODE_MASK (3 << TEGRA20_SPDIF_CTRL_BIT_MODE_SHIFT) +#define TEGRA20_SPDIF_CTRL_BIT_MODE_16BIT (TEGRA20_SPDIF_BIT_MODE_16BIT << TEGRA20_SPDIF_CTRL_BIT_MODE_SHIFT) +#define TEGRA20_SPDIF_CTRL_BIT_MODE_20BIT (TEGRA20_SPDIF_BIT_MODE_20BIT << TEGRA20_SPDIF_CTRL_BIT_MODE_SHIFT) +#define TEGRA20_SPDIF_CTRL_BIT_MODE_24BIT (TEGRA20_SPDIF_BIT_MODE_24BIT << TEGRA20_SPDIF_CTRL_BIT_MODE_SHIFT) +#define TEGRA20_SPDIF_CTRL_BIT_MODE_RAW (TEGRA20_SPDIF_BIT_MODE_RAW << TEGRA20_SPDIF_CTRL_BIT_MODE_SHIFT) -/* Fields in TEGRA_SPDIF_STATUS */ +/* Fields in TEGRA20_SPDIF_STATUS */ /* * Note: IS_P, IS_B, IS_C, and IS_U are sticky bits. Software must @@ -142,7 +142,7 @@ * (a) the end of a frame is reached after RX_EN is deeasserted, or * (b) the SPDIF data stream becomes inactive. */ -#define TEGRA_SPDIF_STATUS_RX_BSY (1 << 29) +#define TEGRA20_SPDIF_STATUS_RX_BSY (1 << 29) /* * Transmitter(TX) shifter is busy transmitting data. @@ -150,7 +150,7 @@ * This bit is deasserted when the end of a frame is reached after * TX_EN is deasserted. */ -#define TEGRA_SPDIF_STATUS_TX_BSY (1 << 28) +#define TEGRA20_SPDIF_STATUS_TX_BSY (1 << 28) /* * TX is busy shifting out channel status. @@ -160,7 +160,7 @@ * (a) the end of a frame is reached after TX_EN is deasserted, or * (b) CH_STA_TX_F register is loaded into the internal shifter. */ -#define TEGRA_SPDIF_STATUS_TC_BSY (1 << 27) +#define TEGRA20_SPDIF_STATUS_TC_BSY (1 << 27) /* * TX User data FIFO busy. @@ -169,173 +169,173 @@ * (a) the end of a frame is reached after TX_EN is deasserted, or * (b) there's no data left in the TX user FIFO. */ -#define TEGRA_SPDIF_STATUS_TU_BSY (1 << 26) +#define TEGRA20_SPDIF_STATUS_TU_BSY (1 << 26) /* TX FIFO Underrun error status */ -#define TEGRA_SPDIF_STATUS_TX_ERR (1 << 25) +#define TEGRA20_SPDIF_STATUS_TX_ERR (1 << 25) /* RX FIFO Overrun error status */ -#define TEGRA_SPDIF_STATUS_RX_ERR (1 << 24) +#define TEGRA20_SPDIF_STATUS_RX_ERR (1 << 24) /* Preamble status: 0=Preamble OK, 1=bad/missing preamble */ -#define TEGRA_SPDIF_STATUS_IS_P (1 << 23) +#define TEGRA20_SPDIF_STATUS_IS_P (1 << 23) /* B-preamble detection status: 0=not detected, 1=B-preamble detected */ -#define TEGRA_SPDIF_STATUS_IS_B (1 << 22) +#define TEGRA20_SPDIF_STATUS_IS_B (1 << 22) /* * RX channel block data receive status: * 0=entire block not recieved yet. * 1=received entire block of channel status, */ -#define TEGRA_SPDIF_STATUS_IS_C (1 << 21) +#define TEGRA20_SPDIF_STATUS_IS_C (1 << 21) /* RX User Data Valid flag: 1=valid IU detected, 0 = no IU detected. */ -#define TEGRA_SPDIF_STATUS_IS_U (1 << 20) +#define TEGRA20_SPDIF_STATUS_IS_U (1 << 20) /* * RX User FIFO Status: * 1=attention level reached, 0=attention level not reached. */ -#define TEGRA_SPDIF_STATUS_QS_RU (1 << 19) +#define TEGRA20_SPDIF_STATUS_QS_RU (1 << 19) /* * TX User FIFO Status: * 1=attention level reached, 0=attention level not reached. */ -#define TEGRA_SPDIF_STATUS_QS_TU (1 << 18) +#define TEGRA20_SPDIF_STATUS_QS_TU (1 << 18) /* * RX Data FIFO Status: * 1=attention level reached, 0=attention level not reached. */ -#define TEGRA_SPDIF_STATUS_QS_RX (1 << 17) +#define TEGRA20_SPDIF_STATUS_QS_RX (1 << 17) /* * TX Data FIFO Status: * 1=attention level reached, 0=attention level not reached. */ -#define TEGRA_SPDIF_STATUS_QS_TX (1 << 16) +#define TEGRA20_SPDIF_STATUS_QS_TX (1 << 16) -/* Fields in TEGRA_SPDIF_STROBE_CTRL */ +/* Fields in TEGRA20_SPDIF_STROBE_CTRL */ /* * Indicates the approximate number of detected SPDIFIN clocks within a * bi-phase period. */ -#define TEGRA_SPDIF_STROBE_CTRL_PERIOD_SHIFT 16 -#define TEGRA_SPDIF_STROBE_CTRL_PERIOD_MASK (0xff << TEGRA_SPDIF_STROBE_CTRL_PERIOD_SHIFT) +#define TEGRA20_SPDIF_STROBE_CTRL_PERIOD_SHIFT 16 +#define TEGRA20_SPDIF_STROBE_CTRL_PERIOD_MASK (0xff << TEGRA20_SPDIF_STROBE_CTRL_PERIOD_SHIFT) /* Data strobe mode: 0=Auto-locked 1=Manual locked */ -#define TEGRA_SPDIF_STROBE_CTRL_STROBE (1 << 15) +#define TEGRA20_SPDIF_STROBE_CTRL_STROBE (1 << 15) /* * Manual data strobe time within the bi-phase clock period (in terms of * the number of over-sampling clocks). */ -#define TEGRA_SPDIF_STROBE_CTRL_DATA_STROBES_SHIFT 8 -#define TEGRA_SPDIF_STROBE_CTRL_DATA_STROBES_MASK (0x1f << TEGRA_SPDIF_STROBE_CTRL_DATA_STROBES_SHIFT) +#define TEGRA20_SPDIF_STROBE_CTRL_DATA_STROBES_SHIFT 8 +#define TEGRA20_SPDIF_STROBE_CTRL_DATA_STROBES_MASK (0x1f << TEGRA20_SPDIF_STROBE_CTRL_DATA_STROBES_SHIFT) /* * Manual SPDIFIN bi-phase clock period (in terms of the number of * over-sampling clocks). */ -#define TEGRA_SPDIF_STROBE_CTRL_CLOCK_PERIOD_SHIFT 0 -#define TEGRA_SPDIF_STROBE_CTRL_CLOCK_PERIOD_MASK (0x3f << TEGRA_SPDIF_STROBE_CTRL_CLOCK_PERIOD_SHIFT) +#define TEGRA20_SPDIF_STROBE_CTRL_CLOCK_PERIOD_SHIFT 0 +#define TEGRA20_SPDIF_STROBE_CTRL_CLOCK_PERIOD_MASK (0x3f << TEGRA20_SPDIF_STROBE_CTRL_CLOCK_PERIOD_SHIFT) /* Fields in SPDIF_DATA_FIFO_CSR */ /* Clear Receiver User FIFO (RX USR.FIFO) */ -#define TEGRA_SPDIF_DATA_FIFO_CSR_RU_CLR (1 << 31) +#define TEGRA20_SPDIF_DATA_FIFO_CSR_RU_CLR (1 << 31) -#define TEGRA_SPDIF_FIFO_ATN_LVL_U_ONE_SLOT 0 -#define TEGRA_SPDIF_FIFO_ATN_LVL_U_TWO_SLOTS 1 -#define TEGRA_SPDIF_FIFO_ATN_LVL_U_THREE_SLOTS 2 -#define TEGRA_SPDIF_FIFO_ATN_LVL_U_FOUR_SLOTS 3 +#define TEGRA20_SPDIF_FIFO_ATN_LVL_U_ONE_SLOT 0 +#define TEGRA20_SPDIF_FIFO_ATN_LVL_U_TWO_SLOTS 1 +#define TEGRA20_SPDIF_FIFO_ATN_LVL_U_THREE_SLOTS 2 +#define TEGRA20_SPDIF_FIFO_ATN_LVL_U_FOUR_SLOTS 3 /* RU FIFO attention level */ -#define TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_SHIFT 29 -#define TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_MASK \ - (0x3 << TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_SHIFT) -#define TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_RU1_WORD_FULL \ - (TEGRA_SPDIF_FIFO_ATN_LVL_U_ONE_SLOT << TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_SHIFT) -#define TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_RU2_WORD_FULL \ - (TEGRA_SPDIF_FIFO_ATN_LVL_U_TWO_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_SHIFT) -#define TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_RU3_WORD_FULL \ - (TEGRA_SPDIF_FIFO_ATN_LVL_U_THREE_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_SHIFT) -#define TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_RU4_WORD_FULL \ - (TEGRA_SPDIF_FIFO_ATN_LVL_U_FOUR_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_SHIFT) +#define TEGRA20_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_SHIFT 29 +#define TEGRA20_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_MASK \ + (0x3 << TEGRA20_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_SHIFT) +#define TEGRA20_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_RU1_WORD_FULL \ + (TEGRA20_SPDIF_FIFO_ATN_LVL_U_ONE_SLOT << TEGRA20_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_SHIFT) +#define TEGRA20_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_RU2_WORD_FULL \ + (TEGRA20_SPDIF_FIFO_ATN_LVL_U_TWO_SLOTS << TEGRA20_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_SHIFT) +#define TEGRA20_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_RU3_WORD_FULL \ + (TEGRA20_SPDIF_FIFO_ATN_LVL_U_THREE_SLOTS << TEGRA20_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_SHIFT) +#define TEGRA20_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_RU4_WORD_FULL \ + (TEGRA20_SPDIF_FIFO_ATN_LVL_U_FOUR_SLOTS << TEGRA20_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_SHIFT) /* Number of RX USR.FIFO levels with valid data. */ -#define TEGRA_SPDIF_DATA_FIFO_CSR_RU_FULL_COUNT_SHIFT 24 -#define TEGRA_SPDIF_DATA_FIFO_CSR_RU_FULL_COUNT_MASK (0x1f << TEGRA_SPDIF_DATA_FIFO_CSR_RU_FULL_COUNT_SHIFT) +#define TEGRA20_SPDIF_DATA_FIFO_CSR_RU_FULL_COUNT_SHIFT 24 +#define TEGRA20_SPDIF_DATA_FIFO_CSR_RU_FULL_COUNT_MASK (0x1f << TEGRA20_SPDIF_DATA_FIFO_CSR_RU_FULL_COUNT_SHIFT) /* Clear Transmitter User FIFO (TX USR.FIFO) */ -#define TEGRA_SPDIF_DATA_FIFO_CSR_TU_CLR (1 << 23) +#define TEGRA20_SPDIF_DATA_FIFO_CSR_TU_CLR (1 << 23) /* TU FIFO attention level */ -#define TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_SHIFT 21 -#define TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_MASK \ - (0x3 << TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_SHIFT) -#define TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_TU1_WORD_FULL \ - (TEGRA_SPDIF_FIFO_ATN_LVL_U_ONE_SLOT << TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_SHIFT) -#define TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_TU2_WORD_FULL \ - (TEGRA_SPDIF_FIFO_ATN_LVL_U_TWO_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_SHIFT) -#define TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_TU3_WORD_FULL \ - (TEGRA_SPDIF_FIFO_ATN_LVL_U_THREE_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_SHIFT) -#define TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_TU4_WORD_FULL \ - (TEGRA_SPDIF_FIFO_ATN_LVL_U_FOUR_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_SHIFT) +#define TEGRA20_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_SHIFT 21 +#define TEGRA20_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_MASK \ + (0x3 << TEGRA20_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_SHIFT) +#define TEGRA20_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_TU1_WORD_FULL \ + (TEGRA20_SPDIF_FIFO_ATN_LVL_U_ONE_SLOT << TEGRA20_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_SHIFT) +#define TEGRA20_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_TU2_WORD_FULL \ + (TEGRA20_SPDIF_FIFO_ATN_LVL_U_TWO_SLOTS << TEGRA20_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_SHIFT) +#define TEGRA20_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_TU3_WORD_FULL \ + (TEGRA20_SPDIF_FIFO_ATN_LVL_U_THREE_SLOTS << TEGRA20_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_SHIFT) +#define TEGRA20_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_TU4_WORD_FULL \ + (TEGRA20_SPDIF_FIFO_ATN_LVL_U_FOUR_SLOTS << TEGRA20_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_SHIFT) /* Number of TX USR.FIFO levels that could be filled. */ -#define TEGRA_SPDIF_DATA_FIFO_CSR_TU_EMPTY_COUNT_SHIFT 16 -#define TEGRA_SPDIF_DATA_FIFO_CSR_TU_EMPTY_COUNT_MASK (0x1f << SPDIF_DATA_FIFO_CSR_TU_EMPTY_COUNT_SHIFT) +#define TEGRA20_SPDIF_DATA_FIFO_CSR_TU_EMPTY_COUNT_SHIFT 16 +#define TEGRA20_SPDIF_DATA_FIFO_CSR_TU_EMPTY_COUNT_MASK (0x1f << SPDIF_DATA_FIFO_CSR_TU_EMPTY_COUNT_SHIFT) /* Clear Receiver Data FIFO (RX DATA.FIFO) */ -#define TEGRA_SPDIF_DATA_FIFO_CSR_RX_CLR (1 << 15) +#define TEGRA20_SPDIF_DATA_FIFO_CSR_RX_CLR (1 << 15) -#define TEGRA_SPDIF_FIFO_ATN_LVL_D_ONE_SLOT 0 -#define TEGRA_SPDIF_FIFO_ATN_LVL_D_FOUR_SLOTS 1 -#define TEGRA_SPDIF_FIFO_ATN_LVL_D_EIGHT_SLOTS 2 -#define TEGRA_SPDIF_FIFO_ATN_LVL_D_TWELVE_SLOTS 3 +#define TEGRA20_SPDIF_FIFO_ATN_LVL_D_ONE_SLOT 0 +#define TEGRA20_SPDIF_FIFO_ATN_LVL_D_FOUR_SLOTS 1 +#define TEGRA20_SPDIF_FIFO_ATN_LVL_D_EIGHT_SLOTS 2 +#define TEGRA20_SPDIF_FIFO_ATN_LVL_D_TWELVE_SLOTS 3 /* RU FIFO attention level */ -#define TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_SHIFT 13 -#define TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_MASK \ - (0x3 << TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_SHIFT) -#define TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_RU1_WORD_FULL \ - (TEGRA_SPDIF_FIFO_ATN_LVL_D_ONE_SLOT << TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_SHIFT) -#define TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_RU4_WORD_FULL \ - (TEGRA_SPDIF_FIFO_ATN_LVL_D_FOUR_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_SHIFT) -#define TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_RU8_WORD_FULL \ - (TEGRA_SPDIF_FIFO_ATN_LVL_D_EIGHT_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_SHIFT) -#define TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_RU12_WORD_FULL \ - (TEGRA_SPDIF_FIFO_ATN_LVL_D_TWELVE_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_SHIFT) +#define TEGRA20_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_SHIFT 13 +#define TEGRA20_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_MASK \ + (0x3 << TEGRA20_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_SHIFT) +#define TEGRA20_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_RU1_WORD_FULL \ + (TEGRA20_SPDIF_FIFO_ATN_LVL_D_ONE_SLOT << TEGRA20_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_SHIFT) +#define TEGRA20_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_RU4_WORD_FULL \ + (TEGRA20_SPDIF_FIFO_ATN_LVL_D_FOUR_SLOTS << TEGRA20_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_SHIFT) +#define TEGRA20_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_RU8_WORD_FULL \ + (TEGRA20_SPDIF_FIFO_ATN_LVL_D_EIGHT_SLOTS << TEGRA20_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_SHIFT) +#define TEGRA20_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_RU12_WORD_FULL \ + (TEGRA20_SPDIF_FIFO_ATN_LVL_D_TWELVE_SLOTS << TEGRA20_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_SHIFT) /* Number of RX DATA.FIFO levels with valid data. */ -#define TEGRA_SPDIF_DATA_FIFO_CSR_RX_FULL_COUNT_SHIFT 8 -#define TEGRA_SPDIF_DATA_FIFO_CSR_RX_FULL_COUNT_MASK (0x1f << TEGRA_SPDIF_DATA_FIFO_CSR_RX_FULL_COUNT_SHIFT) +#define TEGRA20_SPDIF_DATA_FIFO_CSR_RX_FULL_COUNT_SHIFT 8 +#define TEGRA20_SPDIF_DATA_FIFO_CSR_RX_FULL_COUNT_MASK (0x1f << TEGRA20_SPDIF_DATA_FIFO_CSR_RX_FULL_COUNT_SHIFT) /* Clear Transmitter Data FIFO (TX DATA.FIFO) */ -#define TEGRA_SPDIF_DATA_FIFO_CSR_TX_CLR (1 << 7) +#define TEGRA20_SPDIF_DATA_FIFO_CSR_TX_CLR (1 << 7) /* TU FIFO attention level */ -#define TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_SHIFT 5 -#define TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_MASK \ - (0x3 << TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_SHIFT) -#define TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_TU1_WORD_FULL \ - (TEGRA_SPDIF_FIFO_ATN_LVL_D_ONE_SLOT << TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_SHIFT) -#define TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_TU4_WORD_FULL \ - (TEGRA_SPDIF_FIFO_ATN_LVL_D_FOUR_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_SHIFT) -#define TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_TU8_WORD_FULL \ - (TEGRA_SPDIF_FIFO_ATN_LVL_D_EIGHT_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_SHIFT) -#define TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_TU12_WORD_FULL \ - (TEGRA_SPDIF_FIFO_ATN_LVL_D_TWELVE_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_SHIFT) +#define TEGRA20_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_SHIFT 5 +#define TEGRA20_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_MASK \ + (0x3 << TEGRA20_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_SHIFT) +#define TEGRA20_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_TU1_WORD_FULL \ + (TEGRA20_SPDIF_FIFO_ATN_LVL_D_ONE_SLOT << TEGRA20_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_SHIFT) +#define TEGRA20_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_TU4_WORD_FULL \ + (TEGRA20_SPDIF_FIFO_ATN_LVL_D_FOUR_SLOTS << TEGRA20_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_SHIFT) +#define TEGRA20_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_TU8_WORD_FULL \ + (TEGRA20_SPDIF_FIFO_ATN_LVL_D_EIGHT_SLOTS << TEGRA20_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_SHIFT) +#define TEGRA20_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_TU12_WORD_FULL \ + (TEGRA20_SPDIF_FIFO_ATN_LVL_D_TWELVE_SLOTS << TEGRA20_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_SHIFT) /* Number of TX DATA.FIFO levels that could be filled. */ -#define TEGRA_SPDIF_DATA_FIFO_CSR_TX_EMPTY_COUNT_SHIFT 0 -#define TEGRA_SPDIF_DATA_FIFO_CSR_TX_EMPTY_COUNT_MASK (0x1f << SPDIF_DATA_FIFO_CSR_TX_EMPTY_COUNT_SHIFT) +#define TEGRA20_SPDIF_DATA_FIFO_CSR_TX_EMPTY_COUNT_SHIFT 0 +#define TEGRA20_SPDIF_DATA_FIFO_CSR_TX_EMPTY_COUNT_MASK (0x1f << SPDIF_DATA_FIFO_CSR_TX_EMPTY_COUNT_SHIFT) -/* Fields in TEGRA_SPDIF_DATA_OUT */ +/* Fields in TEGRA20_SPDIF_DATA_OUT */ /* * This register has 5 different formats: @@ -346,36 +346,36 @@ * 16-bit packed (BIT_MODE=00, PACK=1) */ -#define TEGRA_SPDIF_DATA_OUT_DATA_16_SHIFT 0 -#define TEGRA_SPDIF_DATA_OUT_DATA_16_MASK (0xffff << TEGRA_SPDIF_DATA_OUT_DATA_16_SHIFT) +#define TEGRA20_SPDIF_DATA_OUT_DATA_16_SHIFT 0 +#define TEGRA20_SPDIF_DATA_OUT_DATA_16_MASK (0xffff << TEGRA20_SPDIF_DATA_OUT_DATA_16_SHIFT) -#define TEGRA_SPDIF_DATA_OUT_DATA_20_SHIFT 0 -#define TEGRA_SPDIF_DATA_OUT_DATA_20_MASK (0xfffff << TEGRA_SPDIF_DATA_OUT_DATA_20_SHIFT) +#define TEGRA20_SPDIF_DATA_OUT_DATA_20_SHIFT 0 +#define TEGRA20_SPDIF_DATA_OUT_DATA_20_MASK (0xfffff << TEGRA20_SPDIF_DATA_OUT_DATA_20_SHIFT) -#define TEGRA_SPDIF_DATA_OUT_DATA_24_SHIFT 0 -#define TEGRA_SPDIF_DATA_OUT_DATA_24_MASK (0xffffff << TEGRA_SPDIF_DATA_OUT_DATA_24_SHIFT) +#define TEGRA20_SPDIF_DATA_OUT_DATA_24_SHIFT 0 +#define TEGRA20_SPDIF_DATA_OUT_DATA_24_MASK (0xffffff << TEGRA20_SPDIF_DATA_OUT_DATA_24_SHIFT) -#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_P (1 << 31) -#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_C (1 << 30) -#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_U (1 << 29) -#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_V (1 << 28) +#define TEGRA20_SPDIF_DATA_OUT_DATA_RAW_P (1 << 31) +#define TEGRA20_SPDIF_DATA_OUT_DATA_RAW_C (1 << 30) +#define TEGRA20_SPDIF_DATA_OUT_DATA_RAW_U (1 << 29) +#define TEGRA20_SPDIF_DATA_OUT_DATA_RAW_V (1 << 28) -#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_DATA_SHIFT 8 -#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_DATA_MASK (0xfffff << TEGRA_SPDIF_DATA_OUT_DATA_RAW_DATA_SHIFT) +#define TEGRA20_SPDIF_DATA_OUT_DATA_RAW_DATA_SHIFT 8 +#define TEGRA20_SPDIF_DATA_OUT_DATA_RAW_DATA_MASK (0xfffff << TEGRA20_SPDIF_DATA_OUT_DATA_RAW_DATA_SHIFT) -#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_AUX_SHIFT 4 -#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_AUX_MASK (0xf << TEGRA_SPDIF_DATA_OUT_DATA_RAW_AUX_SHIFT) +#define TEGRA20_SPDIF_DATA_OUT_DATA_RAW_AUX_SHIFT 4 +#define TEGRA20_SPDIF_DATA_OUT_DATA_RAW_AUX_MASK (0xf << TEGRA20_SPDIF_DATA_OUT_DATA_RAW_AUX_SHIFT) -#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_PREAMBLE_SHIFT 0 -#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_PREAMBLE_MASK (0xf << TEGRA_SPDIF_DATA_OUT_DATA_RAW_PREAMBLE_SHIFT) +#define TEGRA20_SPDIF_DATA_OUT_DATA_RAW_PREAMBLE_SHIFT 0 +#define TEGRA20_SPDIF_DATA_OUT_DATA_RAW_PREAMBLE_MASK (0xf << TEGRA20_SPDIF_DATA_OUT_DATA_RAW_PREAMBLE_SHIFT) -#define TEGRA_SPDIF_DATA_OUT_DATA_16_PACKED_RIGHT_SHIFT 16 -#define TEGRA_SPDIF_DATA_OUT_DATA_16_PACKED_RIGHT_MASK (0xffff << TEGRA_SPDIF_DATA_OUT_DATA_16_PACKED_RIGHT_SHIFT) +#define TEGRA20_SPDIF_DATA_OUT_DATA_16_PACKED_RIGHT_SHIFT 16 +#define TEGRA20_SPDIF_DATA_OUT_DATA_16_PACKED_RIGHT_MASK (0xffff << TEGRA20_SPDIF_DATA_OUT_DATA_16_PACKED_RIGHT_SHIFT) -#define TEGRA_SPDIF_DATA_OUT_DATA_16_PACKED_LEFT_SHIFT 0 -#define TEGRA_SPDIF_DATA_OUT_DATA_16_PACKED_LEFT_MASK (0xffff << TEGRA_SPDIF_DATA_OUT_DATA_16_PACKED_LEFT_SHIFT) +#define TEGRA20_SPDIF_DATA_OUT_DATA_16_PACKED_LEFT_SHIFT 0 +#define TEGRA20_SPDIF_DATA_OUT_DATA_16_PACKED_LEFT_MASK (0xffff << TEGRA20_SPDIF_DATA_OUT_DATA_16_PACKED_LEFT_SHIFT) -/* Fields in TEGRA_SPDIF_DATA_IN */ +/* Fields in TEGRA20_SPDIF_DATA_IN */ /* * This register has 5 different formats: @@ -388,44 +388,44 @@ * Bits 31:24 are common to all modes except 16-bit packed */ -#define TEGRA_SPDIF_DATA_IN_DATA_P (1 << 31) -#define TEGRA_SPDIF_DATA_IN_DATA_C (1 << 30) -#define TEGRA_SPDIF_DATA_IN_DATA_U (1 << 29) -#define TEGRA_SPDIF_DATA_IN_DATA_V (1 << 28) +#define TEGRA20_SPDIF_DATA_IN_DATA_P (1 << 31) +#define TEGRA20_SPDIF_DATA_IN_DATA_C (1 << 30) +#define TEGRA20_SPDIF_DATA_IN_DATA_U (1 << 29) +#define TEGRA20_SPDIF_DATA_IN_DATA_V (1 << 28) -#define TEGRA_SPDIF_DATA_IN_DATA_PREAMBLE_SHIFT 24 -#define TEGRA_SPDIF_DATA_IN_DATA_PREAMBLE_MASK (0xf << TEGRA_SPDIF_DATA_IN_DATA_PREAMBLE_SHIFT) +#define TEGRA20_SPDIF_DATA_IN_DATA_PREAMBLE_SHIFT 24 +#define TEGRA20_SPDIF_DATA_IN_DATA_PREAMBLE_MASK (0xf << TEGRA20_SPDIF_DATA_IN_DATA_PREAMBLE_SHIFT) -#define TEGRA_SPDIF_DATA_IN_DATA_16_SHIFT 0 -#define TEGRA_SPDIF_DATA_IN_DATA_16_MASK (0xffff << TEGRA_SPDIF_DATA_IN_DATA_16_SHIFT) +#define TEGRA20_SPDIF_DATA_IN_DATA_16_SHIFT 0 +#define TEGRA20_SPDIF_DATA_IN_DATA_16_MASK (0xffff << TEGRA20_SPDIF_DATA_IN_DATA_16_SHIFT) -#define TEGRA_SPDIF_DATA_IN_DATA_20_SHIFT 0 -#define TEGRA_SPDIF_DATA_IN_DATA_20_MASK (0xfffff << TEGRA_SPDIF_DATA_IN_DATA_20_SHIFT) +#define TEGRA20_SPDIF_DATA_IN_DATA_20_SHIFT 0 +#define TEGRA20_SPDIF_DATA_IN_DATA_20_MASK (0xfffff << TEGRA20_SPDIF_DATA_IN_DATA_20_SHIFT) -#define TEGRA_SPDIF_DATA_IN_DATA_24_SHIFT 0 -#define TEGRA_SPDIF_DATA_IN_DATA_24_MASK (0xffffff << TEGRA_SPDIF_DATA_IN_DATA_24_SHIFT) +#define TEGRA20_SPDIF_DATA_IN_DATA_24_SHIFT 0 +#define TEGRA20_SPDIF_DATA_IN_DATA_24_MASK (0xffffff << TEGRA20_SPDIF_DATA_IN_DATA_24_SHIFT) -#define TEGRA_SPDIF_DATA_IN_DATA_RAW_DATA_SHIFT 8 -#define TEGRA_SPDIF_DATA_IN_DATA_RAW_DATA_MASK (0xfffff << TEGRA_SPDIF_DATA_IN_DATA_RAW_DATA_SHIFT) +#define TEGRA20_SPDIF_DATA_IN_DATA_RAW_DATA_SHIFT 8 +#define TEGRA20_SPDIF_DATA_IN_DATA_RAW_DATA_MASK (0xfffff << TEGRA20_SPDIF_DATA_IN_DATA_RAW_DATA_SHIFT) -#define TEGRA_SPDIF_DATA_IN_DATA_RAW_AUX_SHIFT 4 -#define TEGRA_SPDIF_DATA_IN_DATA_RAW_AUX_MASK (0xf << TEGRA_SPDIF_DATA_IN_DATA_RAW_AUX_SHIFT) +#define TEGRA20_SPDIF_DATA_IN_DATA_RAW_AUX_SHIFT 4 +#define TEGRA20_SPDIF_DATA_IN_DATA_RAW_AUX_MASK (0xf << TEGRA20_SPDIF_DATA_IN_DATA_RAW_AUX_SHIFT) -#define TEGRA_SPDIF_DATA_IN_DATA_RAW_PREAMBLE_SHIFT 0 -#define TEGRA_SPDIF_DATA_IN_DATA_RAW_PREAMBLE_MASK (0xf << TEGRA_SPDIF_DATA_IN_DATA_RAW_PREAMBLE_SHIFT) +#define TEGRA20_SPDIF_DATA_IN_DATA_RAW_PREAMBLE_SHIFT 0 +#define TEGRA20_SPDIF_DATA_IN_DATA_RAW_PREAMBLE_MASK (0xf << TEGRA20_SPDIF_DATA_IN_DATA_RAW_PREAMBLE_SHIFT) -#define TEGRA_SPDIF_DATA_IN_DATA_16_PACKED_RIGHT_SHIFT 16 -#define TEGRA_SPDIF_DATA_IN_DATA_16_PACKED_RIGHT_MASK (0xffff << TEGRA_SPDIF_DATA_IN_DATA_16_PACKED_RIGHT_SHIFT) +#define TEGRA20_SPDIF_DATA_IN_DATA_16_PACKED_RIGHT_SHIFT 16 +#define TEGRA20_SPDIF_DATA_IN_DATA_16_PACKED_RIGHT_MASK (0xffff << TEGRA20_SPDIF_DATA_IN_DATA_16_PACKED_RIGHT_SHIFT) -#define TEGRA_SPDIF_DATA_IN_DATA_16_PACKED_LEFT_SHIFT 0 -#define TEGRA_SPDIF_DATA_IN_DATA_16_PACKED_LEFT_MASK (0xffff << TEGRA_SPDIF_DATA_IN_DATA_16_PACKED_LEFT_SHIFT) +#define TEGRA20_SPDIF_DATA_IN_DATA_16_PACKED_LEFT_SHIFT 0 +#define TEGRA20_SPDIF_DATA_IN_DATA_16_PACKED_LEFT_MASK (0xffff << TEGRA20_SPDIF_DATA_IN_DATA_16_PACKED_LEFT_SHIFT) -/* Fields in TEGRA_SPDIF_CH_STA_RX_A */ -/* Fields in TEGRA_SPDIF_CH_STA_RX_B */ -/* Fields in TEGRA_SPDIF_CH_STA_RX_C */ -/* Fields in TEGRA_SPDIF_CH_STA_RX_D */ -/* Fields in TEGRA_SPDIF_CH_STA_RX_E */ -/* Fields in TEGRA_SPDIF_CH_STA_RX_F */ +/* Fields in TEGRA20_SPDIF_CH_STA_RX_A */ +/* Fields in TEGRA20_SPDIF_CH_STA_RX_B */ +/* Fields in TEGRA20_SPDIF_CH_STA_RX_C */ +/* Fields in TEGRA20_SPDIF_CH_STA_RX_D */ +/* Fields in TEGRA20_SPDIF_CH_STA_RX_E */ +/* Fields in TEGRA20_SPDIF_CH_STA_RX_F */ /* * The 6-word receive channel data page buffer holds a block (192 frames) of @@ -433,12 +433,12 @@ * bit, and from CH_STA_RX_A to CH_STA_RX_F then back to CH_STA_RX_A. */ -/* Fields in TEGRA_SPDIF_CH_STA_TX_A */ -/* Fields in TEGRA_SPDIF_CH_STA_TX_B */ -/* Fields in TEGRA_SPDIF_CH_STA_TX_C */ -/* Fields in TEGRA_SPDIF_CH_STA_TX_D */ -/* Fields in TEGRA_SPDIF_CH_STA_TX_E */ -/* Fields in TEGRA_SPDIF_CH_STA_TX_F */ +/* Fields in TEGRA20_SPDIF_CH_STA_TX_A */ +/* Fields in TEGRA20_SPDIF_CH_STA_TX_B */ +/* Fields in TEGRA20_SPDIF_CH_STA_TX_C */ +/* Fields in TEGRA20_SPDIF_CH_STA_TX_D */ +/* Fields in TEGRA20_SPDIF_CH_STA_TX_E */ +/* Fields in TEGRA20_SPDIF_CH_STA_TX_F */ /* * The 6-word transmit channel data page buffer holds a block (192 frames) of @@ -446,21 +446,21 @@ * bit, and from CH_STA_TX_A to CH_STA_TX_F then back to CH_STA_TX_A. */ -/* Fields in TEGRA_SPDIF_USR_STA_RX_A */ +/* Fields in TEGRA20_SPDIF_USR_STA_RX_A */ /* * This 4-word deep FIFO receives user FIFO field information. The order of * receive is from LSB to MSB bit. */ -/* Fields in TEGRA_SPDIF_USR_DAT_TX_A */ +/* Fields in TEGRA20_SPDIF_USR_DAT_TX_A */ /* * This 4-word deep FIFO transmits user FIFO field information. The order of * transmission is from LSB to MSB bit. */ -struct tegra_spdif { +struct tegra20_spdif { struct clk *clk_spdif_out; struct tegra_pcm_dma_params capture_dma_data; struct tegra_pcm_dma_params playback_dma_data; diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c index 2f9e9ff771c8..0b0df49d9d33 100644 --- a/sound/soc/tegra/tegra_wm8903.c +++ b/sound/soc/tegra/tegra_wm8903.c @@ -350,8 +350,8 @@ static struct snd_soc_dai_link tegra_wm8903_dai = { .name = "WM8903", .stream_name = "WM8903 PCM", .codec_name = "wm8903.0-001a", - .platform_name = "tegra-i2s.0", - .cpu_dai_name = "tegra-i2s.0", + .platform_name = "tegra20-i2s.0", + .cpu_dai_name = "tegra20-i2s.0", .codec_dai_name = "wm8903-hifi", .init = tegra_wm8903_init, .ops = &tegra_wm8903_ops, diff --git a/sound/soc/tegra/trimslice.c b/sound/soc/tegra/trimslice.c index 8884667aff43..0fd115e69a8a 100644 --- a/sound/soc/tegra/trimslice.c +++ b/sound/soc/tegra/trimslice.c @@ -116,8 +116,8 @@ static struct snd_soc_dai_link trimslice_tlv320aic23_dai = { .name = "TLV320AIC23", .stream_name = "AIC23", .codec_name = "tlv320aic23-codec.2-001a", - .platform_name = "tegra-i2s.0", - .cpu_dai_name = "tegra-i2s.0", + .platform_name = "tegra20-i2s.0", + .cpu_dai_name = "tegra20-i2s.0", .codec_dai_name = "tlv320aic23-hifi", .ops = &trimslice_asoc_ops, }; -- cgit v1.2.3 From cd115367424ebaba93d24abdf74e0d0fe498e41b Mon Sep 17 00:00:00 2001 From: Maarten ter Huurne Date: Fri, 6 Apr 2012 19:16:27 +0200 Subject: ASoC: JZ4740: Replaced comma operators with semicolons. They were harmless but also unnecessary, probably a leftover from earlier code. Signed-off-by: Maarten ter Huurne Signed-off-by: Mark Brown --- sound/soc/jz4740/jz4740-i2s.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/jz4740/jz4740-i2s.c b/sound/soc/jz4740/jz4740-i2s.c index a5af7c42e62b..41349670adab 100644 --- a/sound/soc/jz4740/jz4740-i2s.c +++ b/sound/soc/jz4740/jz4740-i2s.c @@ -346,7 +346,7 @@ static void jz4740_i2c_init_pcm_config(struct jz4740_i2s *i2s) /* Playback */ dma_config = &i2s->pcm_config_playback.dma_config; - dma_config->src_width = JZ4740_DMA_WIDTH_32BIT, + dma_config->src_width = JZ4740_DMA_WIDTH_32BIT; dma_config->transfer_size = JZ4740_DMA_TRANSFER_SIZE_16BYTE; dma_config->request_type = JZ4740_DMA_TYPE_AIC_TRANSMIT; dma_config->flags = JZ4740_DMA_SRC_AUTOINC; @@ -355,7 +355,7 @@ static void jz4740_i2c_init_pcm_config(struct jz4740_i2s *i2s) /* Capture */ dma_config = &i2s->pcm_config_capture.dma_config; - dma_config->dst_width = JZ4740_DMA_WIDTH_32BIT, + dma_config->dst_width = JZ4740_DMA_WIDTH_32BIT; dma_config->transfer_size = JZ4740_DMA_TRANSFER_SIZE_16BYTE; dma_config->request_type = JZ4740_DMA_TYPE_AIC_RECEIVE; dma_config->flags = JZ4740_DMA_DST_AUTOINC; -- cgit v1.2.3 From 7613c508dfd7880cb7e17ad5eb64b9dfb0b80292 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Fri, 6 Apr 2012 11:12:25 -0600 Subject: ASoC: tegra: sort includes, remove mach/iomap.h Signed-off-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/tegra/tegra20_das.c | 5 ++--- sound/soc/tegra/tegra20_i2s.c | 7 +++---- sound/soc/tegra/tegra20_spdif.c | 5 ++--- sound/soc/tegra/tegra_pcm.c | 2 +- 4 files changed, 8 insertions(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/soc/tegra/tegra20_das.c b/sound/soc/tegra/tegra20_das.c index 486d7b245cf4..812696d9c863 100644 --- a/sound/soc/tegra/tegra20_das.c +++ b/sound/soc/tegra/tegra20_das.c @@ -20,14 +20,13 @@ * */ -#include #include #include +#include +#include #include #include #include -#include -#include #include #include "tegra20_das.h" diff --git a/sound/soc/tegra/tegra20_i2s.c b/sound/soc/tegra/tegra20_i2s.c index 878798c89e49..9427f36e6a25 100644 --- a/sound/soc/tegra/tegra20_i2s.c +++ b/sound/soc/tegra/tegra20_i2s.c @@ -29,15 +29,14 @@ */ #include -#include #include #include +#include +#include +#include #include #include #include -#include -#include -#include #include #include #include diff --git a/sound/soc/tegra/tegra20_spdif.c b/sound/soc/tegra/tegra20_spdif.c index 052bff816c4d..00a2d33352f4 100644 --- a/sound/soc/tegra/tegra20_spdif.c +++ b/sound/soc/tegra/tegra20_spdif.c @@ -21,14 +21,13 @@ */ #include -#include #include #include +#include +#include #include #include #include -#include -#include #include #include #include diff --git a/sound/soc/tegra/tegra_pcm.c b/sound/soc/tegra/tegra_pcm.c index 476b8ac3a24d..127348dc09b1 100644 --- a/sound/soc/tegra/tegra_pcm.c +++ b/sound/soc/tegra/tegra_pcm.c @@ -29,8 +29,8 @@ * */ -#include #include +#include #include #include #include -- cgit v1.2.3 From 17933db2e4758b0cefdf23124087b488bce24b82 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Fri, 6 Apr 2012 11:14:04 -0600 Subject: ASoC: tegra: use devm_ APIs in SPDIF driver The devm_ APIs remove the need to manually clean up allocations, thus removing some code. Signed-off-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/tegra/tegra20_spdif.c | 33 ++++++++++----------------------- 1 file changed, 10 insertions(+), 23 deletions(-) (limited to 'sound') diff --git a/sound/soc/tegra/tegra20_spdif.c b/sound/soc/tegra/tegra20_spdif.c index 00a2d33352f4..ef5d49ed5658 100644 --- a/sound/soc/tegra/tegra20_spdif.c +++ b/sound/soc/tegra/tegra20_spdif.c @@ -245,11 +245,12 @@ static __devinit int tegra20_spdif_platform_probe(struct platform_device *pdev) struct resource *mem, *memregion, *dmareq; int ret; - spdif = kzalloc(sizeof(struct tegra20_spdif), GFP_KERNEL); + spdif = devm_kzalloc(&pdev->dev, sizeof(struct tegra20_spdif), + GFP_KERNEL); if (!spdif) { dev_err(&pdev->dev, "Can't allocate tegra20_spdif\n"); ret = -ENOMEM; - goto exit; + goto err; } dev_set_drvdata(&pdev->dev, spdif); @@ -257,7 +258,7 @@ static __devinit int tegra20_spdif_platform_probe(struct platform_device *pdev) if (IS_ERR(spdif->clk_spdif_out)) { pr_err("Can't retrieve spdif clock\n"); ret = PTR_ERR(spdif->clk_spdif_out); - goto err_free; + goto err; } mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); @@ -274,19 +275,19 @@ static __devinit int tegra20_spdif_platform_probe(struct platform_device *pdev) goto err_clk_put; } - memregion = request_mem_region(mem->start, resource_size(mem), - DRV_NAME); + memregion = devm_request_mem_region(&pdev->dev, mem->start, + resource_size(mem), DRV_NAME); if (!memregion) { dev_err(&pdev->dev, "Memory region already claimed\n"); ret = -EBUSY; goto err_clk_put; } - spdif->regs = ioremap(mem->start, resource_size(mem)); + spdif->regs = devm_ioremap(&pdev->dev, mem->start, resource_size(mem)); if (!spdif->regs) { dev_err(&pdev->dev, "ioremap failed\n"); ret = -ENOMEM; - goto err_release; + goto err_clk_put; } spdif->playback_dma_data.addr = mem->start + TEGRA20_SPDIF_DATA_OUT; @@ -298,7 +299,7 @@ static __devinit int tegra20_spdif_platform_probe(struct platform_device *pdev) if (ret) { dev_err(&pdev->dev, "Could not register DAI: %d\n", ret); ret = -ENOMEM; - goto err_unmap; + goto err_clk_put; } ret = tegra_pcm_platform_register(&pdev->dev); @@ -313,37 +314,23 @@ static __devinit int tegra20_spdif_platform_probe(struct platform_device *pdev) err_unregister_dai: snd_soc_unregister_dai(&pdev->dev); -err_unmap: - iounmap(spdif->regs); -err_release: - release_mem_region(mem->start, resource_size(mem)); err_clk_put: clk_put(spdif->clk_spdif_out); -err_free: - kfree(spdif); -exit: +err: return ret; } static int __devexit tegra20_spdif_platform_remove(struct platform_device *pdev) { struct tegra20_spdif *spdif = dev_get_drvdata(&pdev->dev); - struct resource *res; tegra_pcm_platform_unregister(&pdev->dev); snd_soc_unregister_dai(&pdev->dev); tegra20_spdif_debug_remove(spdif); - iounmap(spdif->regs); - - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - release_mem_region(res->start, resource_size(res)); - clk_put(spdif->clk_spdif_out); - kfree(spdif); - return 0; } -- cgit v1.2.3 From a9005b67b3a2103b2b7e32bf602d3f023076fe06 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Fri, 6 Apr 2012 11:18:16 -0600 Subject: ASoC: tegra: set a sensible initial clock rate Initialize the audio clock tree appropriately for some reasonable rate. This makes sure the PLLs etc. are actually programmed to something reasonable when the audio driver is loaded. Signed-off-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/tegra/tegra_asoc_utils.c | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'sound') diff --git a/sound/soc/tegra/tegra_asoc_utils.c b/sound/soc/tegra/tegra_asoc_utils.c index f8428e410e05..30424e157f69 100644 --- a/sound/soc/tegra/tegra_asoc_utils.c +++ b/sound/soc/tegra/tegra_asoc_utils.c @@ -133,8 +133,14 @@ int tegra_asoc_utils_init(struct tegra_asoc_utils_data *data, goto err_put_pll_a_out0; } + ret = tegra_asoc_utils_set_rate(data, 44100, 256 * 44100); + if (ret) + goto err_put_cdev1; + return 0; +err_put_cdev1: + clk_put(data->clk_cdev1); err_put_pll_a_out0: clk_put(data->clk_pll_a_out0); err_put_pll_a: -- cgit v1.2.3 From c2f6702d318e43bf841da9c0ba5b6f1695661bbc Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Fri, 6 Apr 2012 11:15:55 -0600 Subject: ASoC: tegra: utils: add support for Tegra30 devices Tegra30 has some additional clocks that need to be manipulated, names some clocks differently, runs PLLs at different base rates, etc. The utility code needs to handle this. Signed-off-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/tegra/tegra_asoc_utils.c | 29 ++++++++++++++++++++++++----- sound/soc/tegra/tegra_asoc_utils.h | 9 +++++++-- 2 files changed, 31 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/tegra/tegra_asoc_utils.c b/sound/soc/tegra/tegra_asoc_utils.c index 30424e157f69..266189d4ff13 100644 --- a/sound/soc/tegra/tegra_asoc_utils.c +++ b/sound/soc/tegra/tegra_asoc_utils.c @@ -2,7 +2,7 @@ * tegra_asoc_utils.c - Harmony machine ASoC driver * * Author: Stephen Warren - * Copyright (C) 2010 - NVIDIA, Inc. + * Copyright (C) 2010,2012 - NVIDIA, Inc. * * This program is free software; you can redistribute it and/or * modify it under the terms of the GNU General Public License @@ -25,6 +25,7 @@ #include #include #include +#include #include "tegra_asoc_utils.h" @@ -40,7 +41,10 @@ int tegra_asoc_utils_set_rate(struct tegra_asoc_utils_data *data, int srate, case 22050: case 44100: case 88200: - new_baseclock = 56448000; + if (data->soc == TEGRA_ASOC_UTILS_SOC_TEGRA20) + new_baseclock = 56448000; + else + new_baseclock = 564480000; break; case 8000: case 16000: @@ -48,7 +52,10 @@ int tegra_asoc_utils_set_rate(struct tegra_asoc_utils_data *data, int srate, case 48000: case 64000: case 96000: - new_baseclock = 73728000; + if (data->soc == TEGRA_ASOC_UTILS_SOC_TEGRA20) + new_baseclock = 73728000; + else + new_baseclock = 552960000; break; default: return -EINVAL; @@ -78,7 +85,7 @@ int tegra_asoc_utils_set_rate(struct tegra_asoc_utils_data *data, int srate, return err; } - /* Don't set cdev1 rate; its locked to pll_a_out0 */ + /* Don't set cdev1/extern1 rate; it's locked to pll_a_out0 */ err = clk_enable(data->clk_pll_a); if (err) { @@ -112,6 +119,15 @@ int tegra_asoc_utils_init(struct tegra_asoc_utils_data *data, data->dev = dev; + if (!of_have_populated_dt()) + data->soc = TEGRA_ASOC_UTILS_SOC_TEGRA20; + else if (of_machine_is_compatible("nvidia,tegra20")) + data->soc = TEGRA_ASOC_UTILS_SOC_TEGRA20; + else if (of_machine_is_compatible("nvidia,tegra30")) + data->soc = TEGRA_ASOC_UTILS_SOC_TEGRA30; + else + return -EINVAL; + data->clk_pll_a = clk_get_sys(NULL, "pll_a"); if (IS_ERR(data->clk_pll_a)) { dev_err(data->dev, "Can't retrieve clk pll_a\n"); @@ -126,7 +142,10 @@ int tegra_asoc_utils_init(struct tegra_asoc_utils_data *data, goto err_put_pll_a; } - data->clk_cdev1 = clk_get_sys(NULL, "cdev1"); + if (data->soc == TEGRA_ASOC_UTILS_SOC_TEGRA20) + data->clk_cdev1 = clk_get_sys(NULL, "cdev1"); + else + data->clk_cdev1 = clk_get_sys("extern1", NULL); if (IS_ERR(data->clk_cdev1)) { dev_err(data->dev, "Can't retrieve clk cdev1\n"); ret = PTR_ERR(data->clk_cdev1); diff --git a/sound/soc/tegra/tegra_asoc_utils.h b/sound/soc/tegra/tegra_asoc_utils.h index 4818195da25c..44db1dbb8f21 100644 --- a/sound/soc/tegra/tegra_asoc_utils.h +++ b/sound/soc/tegra/tegra_asoc_utils.h @@ -2,7 +2,7 @@ * tegra_asoc_utils.h - Definitions for Tegra DAS driver * * Author: Stephen Warren - * Copyright (C) 2010 - NVIDIA, Inc. + * Copyright (C) 2010,2012 - NVIDIA, Inc. * * This program is free software; you can redistribute it and/or * modify it under the terms of the GNU General Public License @@ -26,8 +26,14 @@ struct clk; struct device; +enum tegra_asoc_utils_soc { + TEGRA_ASOC_UTILS_SOC_TEGRA20, + TEGRA_ASOC_UTILS_SOC_TEGRA30, +}; + struct tegra_asoc_utils_data { struct device *dev; + enum tegra_asoc_utils_soc soc; struct clk *clk_pll_a; struct clk *clk_pll_a_out0; struct clk *clk_cdev1; @@ -42,4 +48,3 @@ int tegra_asoc_utils_init(struct tegra_asoc_utils_data *data, void tegra_asoc_utils_fini(struct tegra_asoc_utils_data *data); #endif - -- cgit v1.2.3 From 82ef0ae46b8614f052cc3ee856c5624eff614063 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Mon, 9 Apr 2012 09:52:22 -0600 Subject: ASoC: tegra: add runtime PM support To the Tegra I2S and SPDIF drivers Signed-off-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/tegra/tegra20_i2s.c | 54 +++++++++++++++++++++++++++++++++++------ sound/soc/tegra/tegra20_spdif.c | 50 +++++++++++++++++++++++++++++++++++--- 2 files changed, 94 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/soc/tegra/tegra20_i2s.c b/sound/soc/tegra/tegra20_i2s.c index 9427f36e6a25..b598ebdefdee 100644 --- a/sound/soc/tegra/tegra20_i2s.c +++ b/sound/soc/tegra/tegra20_i2s.c @@ -35,6 +35,7 @@ #include #include #include +#include #include #include #include @@ -56,6 +57,29 @@ static inline u32 tegra20_i2s_read(struct tegra20_i2s *i2s, u32 reg) return __raw_readl(i2s->regs + reg); } +static int tegra20_i2s_runtime_suspend(struct device *dev) +{ + struct tegra20_i2s *i2s = dev_get_drvdata(dev); + + clk_disable(i2s->clk_i2s); + + return 0; +} + +static int tegra20_i2s_runtime_resume(struct device *dev) +{ + struct tegra20_i2s *i2s = dev_get_drvdata(dev); + int ret; + + ret = clk_enable(i2s->clk_i2s); + if (ret) { + dev_err(dev, "clk_enable failed: %d\n", ret); + return ret; + } + + return 0; +} + #ifdef CONFIG_DEBUG_FS static int tegra20_i2s_show(struct seq_file *s, void *unused) { @@ -219,16 +243,12 @@ static int tegra20_i2s_hw_params(struct snd_pcm_substream *substream, if (i2sclock % (2 * srate)) reg |= TEGRA20_I2S_TIMING_NON_SYM_ENABLE; - clk_enable(i2s->clk_i2s); - tegra20_i2s_write(i2s, TEGRA20_I2S_TIMING, reg); tegra20_i2s_write(i2s, TEGRA20_I2S_FIFO_SCR, TEGRA20_I2S_FIFO_SCR_FIFO2_ATN_LVL_FOUR_SLOTS | TEGRA20_I2S_FIFO_SCR_FIFO1_ATN_LVL_FOUR_SLOTS); - clk_disable(i2s->clk_i2s); - return 0; } @@ -265,7 +285,6 @@ static int tegra20_i2s_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: case SNDRV_PCM_TRIGGER_RESUME: - clk_enable(i2s->clk_i2s); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) tegra20_i2s_start_playback(i2s); else @@ -278,7 +297,6 @@ static int tegra20_i2s_trigger(struct snd_pcm_substream *substream, int cmd, tegra20_i2s_stop_playback(i2s); else tegra20_i2s_stop_capture(i2s); - clk_disable(i2s->clk_i2s); break; default: return -EINVAL; @@ -395,11 +413,18 @@ static __devinit int tegra20_i2s_platform_probe(struct platform_device *pdev) i2s->reg_ctrl = TEGRA20_I2S_CTRL_FIFO_FORMAT_PACKED; + pm_runtime_enable(&pdev->dev); + if (!pm_runtime_enabled(&pdev->dev)) { + ret = tegra20_i2s_runtime_resume(&pdev->dev); + if (ret) + goto err_pm_disable; + } + ret = snd_soc_register_dai(&pdev->dev, &i2s->dai); if (ret) { dev_err(&pdev->dev, "Could not register DAI: %d\n", ret); ret = -ENOMEM; - goto err_clk_put; + goto err_suspend; } ret = tegra_pcm_platform_register(&pdev->dev); @@ -414,6 +439,11 @@ static __devinit int tegra20_i2s_platform_probe(struct platform_device *pdev) err_unregister_dai: snd_soc_unregister_dai(&pdev->dev); +err_suspend: + if (!pm_runtime_status_suspended(&pdev->dev)) + tegra20_i2s_runtime_suspend(&pdev->dev); +err_pm_disable: + pm_runtime_disable(&pdev->dev); err_clk_put: clk_put(i2s->clk_i2s); err: @@ -424,6 +454,10 @@ static int __devexit tegra20_i2s_platform_remove(struct platform_device *pdev) { struct tegra20_i2s *i2s = dev_get_drvdata(&pdev->dev); + pm_runtime_disable(&pdev->dev); + if (!pm_runtime_status_suspended(&pdev->dev)) + tegra20_i2s_runtime_suspend(&pdev->dev); + tegra_pcm_platform_unregister(&pdev->dev); snd_soc_unregister_dai(&pdev->dev); @@ -439,11 +473,17 @@ static const struct of_device_id tegra20_i2s_of_match[] __devinitconst = { {}, }; +static const struct dev_pm_ops tegra20_i2s_pm_ops __devinitconst = { + SET_RUNTIME_PM_OPS(tegra20_i2s_runtime_suspend, + tegra20_i2s_runtime_resume, NULL) +}; + static struct platform_driver tegra20_i2s_driver = { .driver = { .name = DRV_NAME, .owner = THIS_MODULE, .of_match_table = tegra20_i2s_of_match, + .pm = &tegra20_i2s_pm_ops, }, .probe = tegra20_i2s_platform_probe, .remove = __devexit_p(tegra20_i2s_platform_remove), diff --git a/sound/soc/tegra/tegra20_spdif.c b/sound/soc/tegra/tegra20_spdif.c index ef5d49ed5658..9efd71e13a0a 100644 --- a/sound/soc/tegra/tegra20_spdif.c +++ b/sound/soc/tegra/tegra20_spdif.c @@ -26,6 +26,7 @@ #include #include #include +#include #include #include #include @@ -48,6 +49,29 @@ static inline u32 tegra20_spdif_read(struct tegra20_spdif *spdif, u32 reg) return __raw_readl(spdif->regs + reg); } +static int tegra20_spdif_runtime_suspend(struct device *dev) +{ + struct tegra20_spdif *spdif = dev_get_drvdata(dev); + + clk_disable(spdif->clk_spdif_out); + + return 0; +} + +static int tegra20_spdif_runtime_resume(struct device *dev) +{ + struct tegra20_spdif *spdif = dev_get_drvdata(dev); + int ret; + + ret = clk_enable(spdif->clk_spdif_out); + if (ret) { + dev_err(dev, "clk_enable failed: %d\n", ret); + return ret; + } + + return 0; +} + #ifdef CONFIG_DEBUG_FS static int tegra20_spdif_show(struct seq_file *s, void *unused) { @@ -195,14 +219,12 @@ static int tegra20_spdif_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: case SNDRV_PCM_TRIGGER_RESUME: - clk_enable(spdif->clk_spdif_out); tegra20_spdif_start_playback(spdif); break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: case SNDRV_PCM_TRIGGER_SUSPEND: tegra20_spdif_stop_playback(spdif); - clk_disable(spdif->clk_spdif_out); break; default: return -EINVAL; @@ -295,11 +317,18 @@ static __devinit int tegra20_spdif_platform_probe(struct platform_device *pdev) spdif->playback_dma_data.width = 32; spdif->playback_dma_data.req_sel = dmareq->start; + pm_runtime_enable(&pdev->dev); + if (!pm_runtime_enabled(&pdev->dev)) { + ret = tegra20_spdif_runtime_resume(&pdev->dev); + if (ret) + goto err_pm_disable; + } + ret = snd_soc_register_dai(&pdev->dev, &tegra20_spdif_dai); if (ret) { dev_err(&pdev->dev, "Could not register DAI: %d\n", ret); ret = -ENOMEM; - goto err_clk_put; + goto err_suspend; } ret = tegra_pcm_platform_register(&pdev->dev); @@ -314,6 +343,11 @@ static __devinit int tegra20_spdif_platform_probe(struct platform_device *pdev) err_unregister_dai: snd_soc_unregister_dai(&pdev->dev); +err_suspend: + if (!pm_runtime_status_suspended(&pdev->dev)) + tegra20_spdif_runtime_suspend(&pdev->dev); +err_pm_disable: + pm_runtime_disable(&pdev->dev); err_clk_put: clk_put(spdif->clk_spdif_out); err: @@ -324,6 +358,10 @@ static int __devexit tegra20_spdif_platform_remove(struct platform_device *pdev) { struct tegra20_spdif *spdif = dev_get_drvdata(&pdev->dev); + pm_runtime_disable(&pdev->dev); + if (!pm_runtime_status_suspended(&pdev->dev)) + tegra20_spdif_runtime_suspend(&pdev->dev); + tegra_pcm_platform_unregister(&pdev->dev); snd_soc_unregister_dai(&pdev->dev); @@ -334,10 +372,16 @@ static int __devexit tegra20_spdif_platform_remove(struct platform_device *pdev) return 0; } +static const struct dev_pm_ops tegra20_spdif_pm_ops __devinitconst = { + SET_RUNTIME_PM_OPS(tegra20_spdif_runtime_suspend, + tegra20_spdif_runtime_resume, NULL) +}; + static struct platform_driver tegra20_spdif_driver = { .driver = { .name = DRV_NAME, .owner = THIS_MODULE, + .pm = &tegra20_spdif_pm_ops, }, .probe = tegra20_spdif_platform_probe, .remove = __devexit_p(tegra20_spdif_platform_remove), -- cgit v1.2.3 From 507230c9997ef47147818508108b97b3e189826f Mon Sep 17 00:00:00 2001 From: Jesper Juhl Date: Mon, 9 Apr 2012 22:52:10 +0200 Subject: ALSA: riptide: remove redundant NULL test before release_firmware() release_firmware() deals gracefully with NULL pointers, no need to check first. Signed-off-by: Jesper Juhl Signed-off-by: Takashi Iwai --- sound/pci/riptide/riptide.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c index 0481d94aac9b..cbeb3f77350c 100644 --- a/sound/pci/riptide/riptide.c +++ b/sound/pci/riptide/riptide.c @@ -1837,8 +1837,7 @@ static int snd_riptide_free(struct snd_riptide *chip) } if (chip->irq >= 0) free_irq(chip->irq, chip); - if (chip->fw_entry) - release_firmware(chip->fw_entry); + release_firmware(chip->fw_entry); release_and_free_resource(chip->res_port); kfree(chip); return 0; -- cgit v1.2.3 From bbdd39155682e444941cc70f991154f2936a522b Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 9 Apr 2012 19:37:25 +0100 Subject: ASoC: wm1250-ev1: Convert to module_i2c_driver Signed-off-by: Mark Brown --- sound/soc/codecs/wm1250-ev1.c | 18 +----------------- 1 file changed, 1 insertion(+), 17 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm1250-ev1.c b/sound/soc/codecs/wm1250-ev1.c index aefb4f89be0e..e7a599a78170 100644 --- a/sound/soc/codecs/wm1250-ev1.c +++ b/sound/soc/codecs/wm1250-ev1.c @@ -215,23 +215,7 @@ static struct i2c_driver wm1250_ev1_i2c_driver = { .id_table = wm1250_ev1_i2c_id, }; -static int __init wm1250_ev1_modinit(void) -{ - int ret = 0; - - ret = i2c_add_driver(&wm1250_ev1_i2c_driver); - if (ret != 0) - pr_err("Failed to register WM1250-EV1 I2C driver: %d\n", ret); - - return ret; -} -module_init(wm1250_ev1_modinit); - -static void __exit wm1250_ev1_exit(void) -{ - i2c_del_driver(&wm1250_ev1_i2c_driver); -} -module_exit(wm1250_ev1_exit); +module_i2c_driver(wm1250_ev1_i2c_driver); MODULE_AUTHOR("Mark Brown "); MODULE_DESCRIPTION("WM1250-EV1 audio I/O module driver"); -- cgit v1.2.3 From 34ff0f95b1d7afc707f121ea3ae6b211fc176fbd Mon Sep 17 00:00:00 2001 From: Jesper Juhl Date: Mon, 9 Apr 2012 22:52:19 +0200 Subject: ASoC: wm8994: Don't test for NULL before release_firmware() release_firmware() does its own NULL ptr testing, it's redundant to also test before calling it. Signed-off-by: Jesper Juhl Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 12 ++++-------- 1 file changed, 4 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 44f72dc27711..8b05e78b11c9 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3936,7 +3936,7 @@ err_irq: return ret; } -static int wm8994_codec_remove(struct snd_soc_codec *codec) +static int wm8994_codec_remove(struct snd_soc_codec *codec) { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); struct wm8994 *control = wm8994->wm8994; @@ -3977,14 +3977,10 @@ static int wm8994_codec_remove(struct snd_soc_codec *codec) free_irq(wm8994->micdet_irq, wm8994); break; } - if (wm8994->mbc) - release_firmware(wm8994->mbc); - if (wm8994->mbc_vss) - release_firmware(wm8994->mbc_vss); - if (wm8994->enh_eq) - release_firmware(wm8994->enh_eq); + release_firmware(wm8994->mbc); + release_firmware(wm8994->mbc_vss); + release_firmware(wm8994->enh_eq); kfree(wm8994->retune_mobile_texts); - return 0; } -- cgit v1.2.3 From 20dc24a951f4792070803d8f1838c8ed3f4e5d57 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 5 Apr 2012 12:55:20 +0100 Subject: ASoC: wm8994: Implement FLL bypass support Later WM8994 class devices can bypass the FLL from BCLK. Do this automatically when the FLL input and output frequencies match up. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 19 +++++++++++++++++++ sound/soc/codecs/wm8994.h | 1 + 2 files changed, 20 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 8b05e78b11c9..01ecdb56ecdc 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -1977,6 +1977,14 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src, snd_soc_update_bits(codec, WM8994_FLL1_CONTROL_1 + reg_offset, WM8994_FLL1_ENA, 0); + if (wm8994->fll_byp && src == WM8994_FLL_SRC_BCLK && + freq_in == freq_out) { + dev_dbg(codec->dev, "Bypassing FLL%d\n", id + 1); + snd_soc_update_bits(codec, WM8994_FLL1_CONTROL_5 + reg_offset, + WM8958_FLL1_BYP, WM8958_FLL1_BYP); + goto out; + } + reg = (fll.outdiv << WM8994_FLL1_OUTDIV_SHIFT) | (fll.fll_fratio << WM8994_FLL1_FRATIO_SHIFT); snd_soc_update_bits(codec, WM8994_FLL1_CONTROL_2 + reg_offset, @@ -1991,6 +1999,7 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src, fll.n << WM8994_FLL1_N_SHIFT); snd_soc_update_bits(codec, WM8994_FLL1_CONTROL_5 + reg_offset, + WM8958_FLL1_BYP | WM8994_FLL1_REFCLK_DIV_MASK | WM8994_FLL1_REFCLK_SRC_MASK, (fll.clk_ref_div << WM8994_FLL1_REFCLK_DIV_SHIFT) | @@ -2053,6 +2062,7 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src, } } +out: wm8994->fll[id].in = freq_in; wm8994->fll[id].out = freq_out; wm8994->fll[id].src = src; @@ -3579,6 +3589,14 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) case WM8958: wm8994->hubs.dcs_readback_mode = 1; wm8994->hubs.hp_startup_mode = 1; + + switch (wm8994->revision) { + case 0: + break; + default: + wm8994->fll_byp = true; + break; + } break; case WM1811: @@ -3586,6 +3604,7 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) wm8994->hubs.no_series_update = 1; wm8994->hubs.hp_startup_mode = 1; wm8994->hubs.no_cache_class_w = true; + wm8994->fll_byp = true; switch (wm8994->revision) { case 0: diff --git a/sound/soc/codecs/wm8994.h b/sound/soc/codecs/wm8994.h index c724112998d8..91650bba205e 100644 --- a/sound/soc/codecs/wm8994.h +++ b/sound/soc/codecs/wm8994.h @@ -79,6 +79,7 @@ struct wm8994_priv { struct wm8994_fll_config fll[2], fll_suspend[2]; struct completion fll_locked[2]; bool fll_locked_irq; + bool fll_byp; int vmid_refcount; int active_refcount; -- cgit v1.2.3 From 83b0c6ba999643ee8ad6329f26e1cdc870e1a920 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Tue, 10 Apr 2012 13:05:29 +0200 Subject: ALSA: hda - Fix oops caused by recent commit "Fix internal mic for Lenovo Ideapad U300s" Make sure we don't dereference the "quirk" pointer when it is null. Reported-by: Dan Carpenter Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index cbe115b6c80c..abb59f472d48 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -4441,7 +4441,9 @@ static void apply_fixup(struct hda_codec *codec, struct conexant_spec *spec = codec->spec; quirk = snd_pci_quirk_lookup(codec->bus->pci, quirk); - if (quirk && table[quirk->value]) { + if (!quirk) + return; + if (table[quirk->value]) { snd_printdd(KERN_INFO "hda_codec: applying pincfg for %s\n", quirk->name); apply_pincfg(codec, table[quirk->value]); -- cgit v1.2.3 From 8127bf5529f6a42d20e9e3613643d149e4dbb697 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Tue, 10 Apr 2012 13:11:17 -0600 Subject: ASoC: tegra: utils: Don't use of_have_populated_dt() Recent list discussions concluded that drivers should not be calling of_have_populated_dt(), and hence of_have_populated_dt() should not be exported. Use a different mechanism to detect DT vs. non-DT boot. Signed-off-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/tegra/tegra_asoc_utils.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/tegra/tegra_asoc_utils.c b/sound/soc/tegra/tegra_asoc_utils.c index 266189d4ff13..9515ce58ea02 100644 --- a/sound/soc/tegra/tegra_asoc_utils.c +++ b/sound/soc/tegra/tegra_asoc_utils.c @@ -119,13 +119,15 @@ int tegra_asoc_utils_init(struct tegra_asoc_utils_data *data, data->dev = dev; - if (!of_have_populated_dt()) - data->soc = TEGRA_ASOC_UTILS_SOC_TEGRA20; - else if (of_machine_is_compatible("nvidia,tegra20")) + if (of_machine_is_compatible("nvidia,tegra20")) data->soc = TEGRA_ASOC_UTILS_SOC_TEGRA20; else if (of_machine_is_compatible("nvidia,tegra30")) data->soc = TEGRA_ASOC_UTILS_SOC_TEGRA30; + else if (!dev->of_node) + /* non-DT is always Tegra20 */ + data->soc = TEGRA_ASOC_UTILS_SOC_TEGRA20; else + /* DT boot, but unknown SoC */ return -EINVAL; data->clk_pll_a = clk_get_sys(NULL, "pll_a"); -- cgit v1.2.3 From b46ac308bf4f675cac9caf63c2de22f2a18f9347 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Tue, 10 Apr 2012 18:33:07 -0300 Subject: ASoC: wm9712: Fix build due to missing definition of "runtime" Fix the following build error: sound/soc/codecs/wm9712.c:482:32: error: 'runtime' undeclared (first use in this function) sound/soc/codecs/wm9712.c:499:33: error: 'runtime' undeclared (first use in this function) Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/codecs/wm9712.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 260386365d58..a1541414d904 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -470,6 +470,7 @@ static int ac97_prepare(struct snd_pcm_substream *substream, struct snd_soc_codec *codec = dai->codec; int reg; u16 vra; + struct snd_pcm_runtime *runtime = substream->runtime; vra = ac97_read(codec, AC97_EXTENDED_STATUS); ac97_write(codec, AC97_EXTENDED_STATUS, vra | 0x1); @@ -487,6 +488,7 @@ static int ac97_aux_prepare(struct snd_pcm_substream *substream, { struct snd_soc_codec *codec = dai->codec; u16 vra, xsle; + struct snd_pcm_runtime *runtime = substream->runtime; vra = ac97_read(codec, AC97_EXTENDED_STATUS); ac97_write(codec, AC97_EXTENDED_STATUS, vra | 0x1); -- cgit v1.2.3 From 7a0a289c5f4aa7547e4b2630c7b18da8b0fb8b4f Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Tue, 10 Apr 2012 19:38:23 -0300 Subject: ASoC: ac97: Fix build due to removal of 'runtime' definition Fix the following build error: sound/soc/codecs/ac97.c: In function 'ac97_prepare': sound/soc/codecs/ac97.c:33: error: 'runtime' undeclared (first use in this function) This was caused by commit e6968a (ASoC: codecs: Remove rtd->codec usage from CODEC drivers), which removed the 'struct snd_pcm_runtime *runtime = substream->runtime' definition. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/codecs/ac97.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c index a99a1b304e1c..2023c749f232 100644 --- a/sound/soc/codecs/ac97.c +++ b/sound/soc/codecs/ac97.c @@ -30,7 +30,7 @@ static int ac97_prepare(struct snd_pcm_substream *substream, int reg = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? AC97_PCM_FRONT_DAC_RATE : AC97_PCM_LR_ADC_RATE; - return snd_ac97_set_rate(codec->ac97, reg, runtime->rate); + return snd_ac97_set_rate(codec->ac97, reg, substream->runtime->rate); } #define STD_AC97_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ -- cgit v1.2.3 From 019ec5059b771cc36fb302a61f0eb08a88beb88b Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Tue, 10 Apr 2012 19:38:24 -0300 Subject: ASoC: wm9705: Fix build due to removal of 'runtime' definition sound/soc/codecs/wm9705.c: In function 'ac97_prepare': sound/soc/codecs/wm9705.c:251: error: 'runtime' undeclared (first use in this function) This was caused by commit e6968a (ASoC: codecs: Remove rtd->codec usage from CODEC drivers), which removed the 'struct snd_pcm_runtime *runtime = substream->runtime' definition. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/codecs/wm9705.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c index 7c095939c179..e8e782a0c78d 100644 --- a/sound/soc/codecs/wm9705.c +++ b/sound/soc/codecs/wm9705.c @@ -248,7 +248,7 @@ static int ac97_prepare(struct snd_pcm_substream *substream, else reg = AC97_PCM_LR_ADC_RATE; - return ac97_write(codec, reg, runtime->rate); + return ac97_write(codec, reg, substream->runtime->rate); } #define WM9705_AC97_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | \ -- cgit v1.2.3 From 596580d0ee1d17af70920a7bb06c963418014dd1 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Thu, 12 Apr 2012 13:51:10 +0200 Subject: ALSA: snd-usb: add snd_usb_audio-wide mutex This is needed for new card-wide list operations. Signed-off-by: Daniel Mack Signed-off-by: Takashi Iwai --- sound/usb/card.c | 2 ++ sound/usb/usbaudio.h | 1 + 2 files changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/usb/card.c b/sound/usb/card.c index 4a7be7b98331..6bc88b7ce4fd 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -276,6 +276,7 @@ static int snd_usb_create_streams(struct snd_usb_audio *chip, int ctrlif) static int snd_usb_audio_free(struct snd_usb_audio *chip) { + mutex_destroy(&chip->mutex); kfree(chip); return 0; } @@ -336,6 +337,7 @@ static int snd_usb_audio_create(struct usb_device *dev, int idx, return -ENOMEM; } + mutex_init(&chip->mutex); mutex_init(&chip->shutdown_mutex); chip->index = idx; chip->dev = dev; diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index 3e2b03577936..a16c21d2d7ed 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -36,6 +36,7 @@ struct snd_usb_audio { struct snd_card *card; struct usb_interface *pm_intf; u32 usb_id; + struct mutex mutex; struct mutex shutdown_mutex; unsigned int shutdown:1; unsigned int probing:1; -- cgit v1.2.3 From 8fdff6a319e7dac757c558bd283dc4577e68cde7 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Thu, 12 Apr 2012 13:51:11 +0200 Subject: ALSA: snd-usb: implement new endpoint streaming model This patch adds a new generic streaming logic for audio over USB. It defines a model (snd_usb_endpoint) that handles everything that is related to an USB endpoint and its streaming. There are functions to activate and deactivate an endpoint (which call usb_set_interface()), and to start and stop its URBs. It also has function pointers to be called when data was received or is about to be sent, and pointer to a sync slave (another snd_usb_endpoint) that is informed when data has been received. A snd_usb_endpoint knows about its state and implements a refcounting, so only the first user will actually start the URBs and only the last one to stop it will tear them down again. With this sort of abstraction, the actual streaming is decoupled from the pcm handling, which makes the "implicit feedback" mechanisms easy to implement. In order to split changes properly, this patch only adds the new implementation but leaves the old one around, so the the driver doesn't change its behaviour. The switch to actually use the new code is submitted separately. Signed-off-by: Daniel Mack Signed-off-by: Takashi Iwai --- sound/usb/card.h | 58 ++++ sound/usb/endpoint.c | 928 ++++++++++++++++++++++++++++++++++++++++++++++++++- sound/usb/endpoint.h | 26 ++ sound/usb/usbaudio.h | 1 + 4 files changed, 1002 insertions(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/usb/card.h b/sound/usb/card.h index da5fa1ac4eda..9acbd4a1228d 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -30,13 +30,17 @@ struct audioformat { }; struct snd_usb_substream; +struct snd_usb_endpoint; struct snd_urb_ctx { struct urb *urb; unsigned int buffer_size; /* size of data buffer, if data URB */ struct snd_usb_substream *subs; + struct snd_usb_endpoint *ep; int index; /* index for urb array */ int packets; /* number of packets per urb */ + int packet_size[MAX_PACKS_HS]; /* size of packets for next submission */ + struct list_head ready_list; }; struct snd_urb_ops { @@ -46,6 +50,60 @@ struct snd_urb_ops { int (*retire_sync)(struct snd_usb_substream *subs, struct snd_pcm_runtime *runtime, struct urb *u); }; +struct snd_usb_endpoint { + struct snd_usb_audio *chip; + + int use_count; + int ep_num; /* the referenced endpoint number */ + int type; /* SND_USB_ENDPOINT_TYPE_* */ + unsigned long flags; + + void (*prepare_data_urb) (struct snd_usb_substream *subs, + struct urb *urb); + void (*retire_data_urb) (struct snd_usb_substream *subs, + struct urb *urb); + + struct snd_usb_substream *data_subs; + struct snd_usb_endpoint *sync_master; + struct snd_usb_endpoint *sync_slave; + + struct snd_urb_ctx urb[MAX_URBS]; + + struct snd_usb_packet_info { + uint32_t packet_size[MAX_PACKS_HS]; + int packets; + } next_packet[MAX_URBS]; + int next_packet_read_pos, next_packet_write_pos; + struct list_head ready_playback_urbs; + + unsigned int nurbs; /* # urbs */ + unsigned long active_mask; /* bitmask of active urbs */ + unsigned long unlink_mask; /* bitmask of unlinked urbs */ + char *syncbuf; /* sync buffer for all sync URBs */ + dma_addr_t sync_dma; /* DMA address of syncbuf */ + + unsigned int pipe; /* the data i/o pipe */ + unsigned int freqn; /* nominal sampling rate in fs/fps in Q16.16 format */ + unsigned int freqm; /* momentary sampling rate in fs/fps in Q16.16 format */ + int freqshift; /* how much to shift the feedback value to get Q16.16 */ + unsigned int freqmax; /* maximum sampling rate, used for buffer management */ + unsigned int phase; /* phase accumulator */ + unsigned int maxpacksize; /* max packet size in bytes */ + unsigned int maxframesize; /* max packet size in frames */ + unsigned int curpacksize; /* current packet size in bytes (for capture) */ + unsigned int curframesize; /* current packet size in frames (for capture) */ + unsigned int syncmaxsize; /* sync endpoint packet size */ + unsigned int fill_max:1; /* fill max packet size always */ + unsigned int datainterval; /* log_2 of data packet interval */ + unsigned int syncinterval; /* P for adaptive mode, 0 otherwise */ + unsigned char silence_value; + unsigned int stride; + int iface, alt_idx; + + spinlock_t lock; + struct list_head list; +}; + struct snd_usb_substream { struct snd_usb_stream *stream; struct usb_device *dev; diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 08dcce53720b..ea25265427ad 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -20,9 +20,11 @@ #include #include #include +#include #include #include +#include #include "usbaudio.h" #include "helper.h" @@ -30,6 +32,9 @@ #include "endpoint.h" #include "pcm.h" +#define EP_FLAG_ACTIVATED 0 +#define EP_FLAG_RUNNING 1 + /* * convert a sampling rate into our full speed format (fs/1000 in Q16.16) * this will overflow at approx 524 kHz @@ -51,7 +56,7 @@ static inline unsigned get_usb_high_speed_rate(unsigned int rate) /* * unlink active urbs. */ -static int deactivate_urbs(struct snd_usb_substream *subs, int force, int can_sleep) +static int deactivate_urbs_old(struct snd_usb_substream *subs, int force, int can_sleep) { struct snd_usb_audio *chip = subs->stream->chip; unsigned int i; @@ -113,7 +118,7 @@ static void release_urb_ctx(struct snd_urb_ctx *u) /* * wait until all urbs are processed. */ -static int wait_clear_urbs(struct snd_usb_substream *subs) +static int wait_clear_urbs_old(struct snd_usb_substream *subs) { unsigned long end_time = jiffies + msecs_to_jiffies(1000); unsigned int i; @@ -148,8 +153,8 @@ void snd_usb_release_substream_urbs(struct snd_usb_substream *subs, int force) int i; /* stop urbs (to be sure) */ - deactivate_urbs(subs, force, 1); - wait_clear_urbs(subs); + deactivate_urbs_old(subs, force, 1); + wait_clear_urbs_old(subs); for (i = 0; i < MAX_URBS; i++) release_urb_ctx(&subs->dataurb[i]); @@ -164,7 +169,7 @@ void snd_usb_release_substream_urbs(struct snd_usb_substream *subs, int force) /* * complete callback from data urb */ -static void snd_complete_urb(struct urb *urb) +static void snd_complete_urb_old(struct urb *urb) { struct snd_urb_ctx *ctx = urb->context; struct snd_usb_substream *subs = ctx->subs; @@ -318,7 +323,7 @@ int snd_usb_init_substream_urbs(struct snd_usb_substream *subs, u->urb->transfer_flags = URB_ISO_ASAP | URB_NO_TRANSFER_DMA_MAP; u->urb->interval = 1 << subs->datainterval; u->urb->context = u; - u->urb->complete = snd_complete_urb; + u->urb->complete = snd_complete_urb_old; } if (subs->syncpipe) { @@ -856,7 +861,7 @@ static int start_urbs(struct snd_usb_substream *subs, struct snd_pcm_runtime *ru __error: // snd_pcm_stop(subs->pcm_substream, SNDRV_PCM_STATE_XRUN); - deactivate_urbs(subs, 0, 0); + deactivate_urbs_old(subs, 0, 0); return -EPIPE; } @@ -917,7 +922,7 @@ int snd_usb_substream_playback_trigger(struct snd_pcm_substream *substream, int subs->ops.prepare = prepare_playback_urb; return 0; case SNDRV_PCM_TRIGGER_STOP: - return deactivate_urbs(subs, 0, 0); + return deactivate_urbs_old(subs, 0, 0); case SNDRV_PCM_TRIGGER_PAUSE_PUSH: subs->ops.prepare = prepare_nodata_playback_urb; return 0; @@ -935,7 +940,7 @@ int snd_usb_substream_capture_trigger(struct snd_pcm_substream *substream, int c subs->ops.retire = retire_capture_urb; return start_urbs(subs, substream->runtime); case SNDRV_PCM_TRIGGER_STOP: - return deactivate_urbs(subs, 0, 0); + return deactivate_urbs_old(subs, 0, 0); case SNDRV_PCM_TRIGGER_PAUSE_PUSH: subs->ops.retire = retire_paused_capture_urb; return 0; @@ -951,8 +956,8 @@ int snd_usb_substream_prepare(struct snd_usb_substream *subs, struct snd_pcm_runtime *runtime) { /* clear urbs (to be sure) */ - deactivate_urbs(subs, 0, 1); - wait_clear_urbs(subs); + deactivate_urbs_old(subs, 0, 1); + wait_clear_urbs_old(subs); /* for playback, submit the URBs now; otherwise, the first hwptr_done * updates for all URBs would happen at the same time when starting */ @@ -964,3 +969,904 @@ int snd_usb_substream_prepare(struct snd_usb_substream *subs, return 0; } +int snd_usb_endpoint_implict_feedback_sink(struct snd_usb_endpoint *ep) +{ + return ep->sync_master && + ep->sync_master->type == SND_USB_ENDPOINT_TYPE_DATA && + ep->type == SND_USB_ENDPOINT_TYPE_DATA && + usb_pipeout(ep->pipe); +} + +/* determine the number of frames in the next packet */ +static int next_packet_size(struct snd_usb_endpoint *ep) +{ + unsigned long flags; + int ret; + + if (ep->fill_max) + return ep->maxframesize; + + spin_lock_irqsave(&ep->lock, flags); + ep->phase = (ep->phase & 0xffff) + + (ep->freqm << ep->datainterval); + ret = min(ep->phase >> 16, ep->maxframesize); + spin_unlock_irqrestore(&ep->lock, flags); + + return ret; +} + +static void retire_outbound_urb(struct snd_usb_endpoint *ep, + struct snd_urb_ctx *urb_ctx) +{ + if (ep->retire_data_urb) + ep->retire_data_urb(ep->data_subs, urb_ctx->urb); +} + +static void retire_inbound_urb(struct snd_usb_endpoint *ep, + struct snd_urb_ctx *urb_ctx) +{ + struct urb *urb = urb_ctx->urb; + + if (ep->sync_slave) + snd_usb_handle_sync_urb(ep->sync_slave, ep, urb); + + if (ep->retire_data_urb) + ep->retire_data_urb(ep->data_subs, urb); +} + +static void prepare_outbound_urb_sizes(struct snd_usb_endpoint *ep, + struct snd_urb_ctx *ctx) +{ + int i; + + for (i = 0; i < ctx->packets; ++i) + ctx->packet_size[i] = next_packet_size(ep); +} + +/* + * Prepare a PLAYBACK urb for submission to the bus. + */ +static void prepare_outbound_urb(struct snd_usb_endpoint *ep, + struct snd_urb_ctx *ctx) +{ + int i; + struct urb *urb = ctx->urb; + unsigned char *cp = urb->transfer_buffer; + + urb->dev = ep->chip->dev; /* we need to set this at each time */ + + switch (ep->type) { + case SND_USB_ENDPOINT_TYPE_DATA: + if (ep->prepare_data_urb) { + ep->prepare_data_urb(ep->data_subs, urb); + } else { + /* no data provider, so send silence */ + unsigned int offs = 0; + for (i = 0; i < ctx->packets; ++i) { + int counts = ctx->packet_size[i]; + urb->iso_frame_desc[i].offset = offs * ep->stride; + urb->iso_frame_desc[i].length = counts * ep->stride; + offs += counts; + } + + urb->number_of_packets = ctx->packets; + urb->transfer_buffer_length = offs * ep->stride; + memset(urb->transfer_buffer, ep->silence_value, + offs * ep->stride); + } + break; + + case SND_USB_ENDPOINT_TYPE_SYNC: + if (snd_usb_get_speed(ep->chip->dev) >= USB_SPEED_HIGH) { + /* + * fill the length and offset of each urb descriptor. + * the fixed 12.13 frequency is passed as 16.16 through the pipe. + */ + urb->iso_frame_desc[0].length = 4; + urb->iso_frame_desc[0].offset = 0; + cp[0] = ep->freqn; + cp[1] = ep->freqn >> 8; + cp[2] = ep->freqn >> 16; + cp[3] = ep->freqn >> 24; + } else { + /* + * fill the length and offset of each urb descriptor. + * the fixed 10.14 frequency is passed through the pipe. + */ + urb->iso_frame_desc[0].length = 3; + urb->iso_frame_desc[0].offset = 0; + cp[0] = ep->freqn >> 2; + cp[1] = ep->freqn >> 10; + cp[2] = ep->freqn >> 18; + } + + break; + } +} + +/* + * Prepare a CAPTURE or SYNC urb for submission to the bus. + */ +static inline void prepare_inbound_urb(struct snd_usb_endpoint *ep, + struct snd_urb_ctx *urb_ctx) +{ + int i, offs; + struct urb *urb = urb_ctx->urb; + + urb->dev = ep->chip->dev; /* we need to set this at each time */ + + switch (ep->type) { + case SND_USB_ENDPOINT_TYPE_DATA: + offs = 0; + for (i = 0; i < urb_ctx->packets; i++) { + urb->iso_frame_desc[i].offset = offs; + urb->iso_frame_desc[i].length = ep->curpacksize; + offs += ep->curpacksize; + } + + urb->transfer_buffer_length = offs; + urb->number_of_packets = urb_ctx->packets; + break; + + case SND_USB_ENDPOINT_TYPE_SYNC: + urb->iso_frame_desc[0].length = min(4u, ep->syncmaxsize); + urb->iso_frame_desc[0].offset = 0; + break; + } +} + +static void queue_pending_output_urbs(struct snd_usb_endpoint *ep) +{ + while (test_bit(EP_FLAG_RUNNING, &ep->flags)) { + + unsigned long flags; + struct snd_usb_packet_info *packet; + struct snd_urb_ctx *ctx = NULL; + struct urb *urb; + int err, i; + + spin_lock_irqsave(&ep->lock, flags); + if (ep->next_packet_read_pos != ep->next_packet_write_pos) { + packet = ep->next_packet + ep->next_packet_read_pos; + ep->next_packet_read_pos++; + ep->next_packet_read_pos %= MAX_URBS; + + /* take URB out of FIFO */ + if (!list_empty(&ep->ready_playback_urbs)) + ctx = list_first_entry(&ep->ready_playback_urbs, + struct snd_urb_ctx, ready_list); + } + spin_unlock_irqrestore(&ep->lock, flags); + + if (ctx == NULL) + return; + + list_del_init(&ctx->ready_list); + urb = ctx->urb; + + /* copy over the length information */ + for (i = 0; i < packet->packets; i++) + ctx->packet_size[i] = packet->packet_size[i]; + + prepare_outbound_urb(ep, ctx); + + err = usb_submit_urb(ctx->urb, GFP_ATOMIC); + if (err < 0) + snd_printk(KERN_ERR "Unable to submit urb #%d: %d (urb %p)\n", + ctx->index, err, ctx->urb); + else + set_bit(ctx->index, &ep->active_mask); + } +} + +/* + * complete callback for urbs + */ +static void snd_complete_urb(struct urb *urb) +{ + struct snd_urb_ctx *ctx = urb->context; + struct snd_usb_endpoint *ep = ctx->ep; + int err; + + if (unlikely(urb->status == -ENOENT || /* unlinked */ + urb->status == -ENODEV || /* device removed */ + urb->status == -ECONNRESET || /* unlinked */ + urb->status == -ESHUTDOWN || /* device disabled */ + ep->chip->shutdown)) /* device disconnected */ + goto exit_clear; + + if (usb_pipeout(ep->pipe)) { + retire_outbound_urb(ep, ctx); + /* can be stopped during retire callback */ + if (unlikely(!test_bit(EP_FLAG_RUNNING, &ep->flags))) + goto exit_clear; + + if (snd_usb_endpoint_implict_feedback_sink(ep)) { + unsigned long flags; + + spin_lock_irqsave(&ep->lock, flags); + list_add_tail(&ctx->ready_list, &ep->ready_playback_urbs); + spin_unlock_irqrestore(&ep->lock, flags); + queue_pending_output_urbs(ep); + + goto exit_clear; + } + + prepare_outbound_urb_sizes(ep, ctx); + prepare_outbound_urb(ep, ctx); + } else { + retire_inbound_urb(ep, ctx); + /* can be stopped during retire callback */ + if (unlikely(!test_bit(EP_FLAG_RUNNING, &ep->flags))) + goto exit_clear; + + prepare_inbound_urb(ep, ctx); + } + + err = usb_submit_urb(urb, GFP_ATOMIC); + if (err == 0) + return; + + snd_printk(KERN_ERR "cannot submit urb (err = %d)\n", err); + //snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); + +exit_clear: + clear_bit(ctx->index, &ep->active_mask); +} + +struct snd_usb_endpoint *snd_usb_add_endpoint(struct snd_usb_audio *chip, + struct usb_host_interface *alts, + int ep_num, int direction, int type) +{ + struct list_head *p; + struct snd_usb_endpoint *ep; + int ret, is_playback = direction == SNDRV_PCM_STREAM_PLAYBACK; + + mutex_lock(&chip->mutex); + + list_for_each(p, &chip->ep_list) { + ep = list_entry(p, struct snd_usb_endpoint, list); + if (ep->ep_num == ep_num && + ep->iface == alts->desc.bInterfaceNumber && + ep->alt_idx == alts->desc.bAlternateSetting) { + snd_printdd(KERN_DEBUG "Re-using EP %x in iface %d,%d @%p\n", + ep_num, ep->iface, ep->alt_idx, ep); + goto __exit_unlock; + } + } + + snd_printdd(KERN_DEBUG "Creating new %s %s endpoint #%x\n", + is_playback ? "playback" : "capture", + type == SND_USB_ENDPOINT_TYPE_DATA ? "data" : "sync", + ep_num); + + /* select the alt setting once so the endpoints become valid */ + ret = usb_set_interface(chip->dev, alts->desc.bInterfaceNumber, + alts->desc.bAlternateSetting); + if (ret < 0) { + snd_printk(KERN_ERR "%s(): usb_set_interface() failed, ret = %d\n", + __func__, ret); + ep = NULL; + goto __exit_unlock; + } + + ep = kzalloc(sizeof(*ep), GFP_KERNEL); + if (!ep) + goto __exit_unlock; + + ep->chip = chip; + spin_lock_init(&ep->lock); + ep->type = type; + ep->ep_num = ep_num; + ep->iface = alts->desc.bInterfaceNumber; + ep->alt_idx = alts->desc.bAlternateSetting; + INIT_LIST_HEAD(&ep->ready_playback_urbs); + ep_num &= USB_ENDPOINT_NUMBER_MASK; + + if (is_playback) + ep->pipe = usb_sndisocpipe(chip->dev, ep_num); + else + ep->pipe = usb_rcvisocpipe(chip->dev, ep_num); + + if (type == SND_USB_ENDPOINT_TYPE_SYNC) { + if (get_endpoint(alts, 1)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE && + get_endpoint(alts, 1)->bRefresh >= 1 && + get_endpoint(alts, 1)->bRefresh <= 9) + ep->syncinterval = get_endpoint(alts, 1)->bRefresh; + else if (snd_usb_get_speed(chip->dev) == USB_SPEED_FULL) + ep->syncinterval = 1; + else if (get_endpoint(alts, 1)->bInterval >= 1 && + get_endpoint(alts, 1)->bInterval <= 16) + ep->syncinterval = get_endpoint(alts, 1)->bInterval - 1; + else + ep->syncinterval = 3; + + ep->syncmaxsize = le16_to_cpu(get_endpoint(alts, 1)->wMaxPacketSize); + } + + list_add_tail(&ep->list, &chip->ep_list); + +__exit_unlock: + mutex_unlock(&chip->mutex); + + return ep; +} + +/* + * wait until all urbs are processed. + */ +static int wait_clear_urbs(struct snd_usb_endpoint *ep) +{ + unsigned long end_time = jiffies + msecs_to_jiffies(1000); + unsigned int i; + int alive; + + do { + alive = 0; + for (i = 0; i < ep->nurbs; i++) + if (test_bit(i, &ep->active_mask)) + alive++; + + if (!alive) + break; + + schedule_timeout_uninterruptible(1); + } while (time_before(jiffies, end_time)); + + if (alive) + snd_printk(KERN_ERR "timeout: still %d active urbs on EP #%x\n", + alive, ep->ep_num); + + return 0; +} + +/* + * unlink active urbs. + */ +static int deactivate_urbs(struct snd_usb_endpoint *ep, int force, int can_sleep) +{ + unsigned long flags; + unsigned int i; + int async; + + if (!force && ep->chip->shutdown) /* to be sure... */ + return -EBADFD; + + async = !can_sleep && ep->chip->async_unlink; + + clear_bit(EP_FLAG_RUNNING, &ep->flags); + + INIT_LIST_HEAD(&ep->ready_playback_urbs); + ep->next_packet_read_pos = 0; + ep->next_packet_write_pos = 0; + + if (!async && in_interrupt()) + return 0; + + for (i = 0; i < ep->nurbs; i++) { + if (test_bit(i, &ep->active_mask)) { + if (!test_and_set_bit(i, &ep->unlink_mask)) { + struct urb *u = ep->urb[i].urb; + if (async) + usb_unlink_urb(u); + else + usb_kill_urb(u); + } + } + } + + return 0; +} + +/* + * release an endpoint's urbs + */ +static void release_urbs(struct snd_usb_endpoint *ep, int force) +{ + int i; + + /* route incoming urbs to nirvana */ + ep->retire_data_urb = NULL; + ep->prepare_data_urb = NULL; + + /* stop urbs */ + deactivate_urbs(ep, force, 1); + wait_clear_urbs(ep); + + for (i = 0; i < ep->nurbs; i++) + release_urb_ctx(&ep->urb[i]); + + if (ep->syncbuf) + usb_free_coherent(ep->chip->dev, SYNC_URBS * 4, + ep->syncbuf, ep->sync_dma); + + ep->syncbuf = NULL; + ep->nurbs = 0; +} + +static int data_ep_set_params(struct snd_usb_endpoint *ep, + struct snd_pcm_hw_params *hw_params, + struct audioformat *fmt, + struct snd_usb_endpoint *sync_ep) +{ + unsigned int maxsize, i, urb_packs, total_packs, packs_per_ms; + int period_bytes = params_period_bytes(hw_params); + int format = params_format(hw_params); + int is_playback = usb_pipeout(ep->pipe); + int frame_bits = snd_pcm_format_physical_width(params_format(hw_params)) * + params_channels(hw_params); + + ep->datainterval = fmt->datainterval; + ep->stride = frame_bits >> 3; + ep->silence_value = format == SNDRV_PCM_FORMAT_U8 ? 0x80 : 0; + + /* calculate max. frequency */ + if (ep->maxpacksize) { + /* whatever fits into a max. size packet */ + maxsize = ep->maxpacksize; + ep->freqmax = (maxsize / (frame_bits >> 3)) + << (16 - ep->datainterval); + } else { + /* no max. packet size: just take 25% higher than nominal */ + ep->freqmax = ep->freqn + (ep->freqn >> 2); + maxsize = ((ep->freqmax + 0xffff) * (frame_bits >> 3)) + >> (16 - ep->datainterval); + } + + if (ep->fill_max) + ep->curpacksize = ep->maxpacksize; + else + ep->curpacksize = maxsize; + + if (snd_usb_get_speed(ep->chip->dev) != USB_SPEED_FULL) + packs_per_ms = 8 >> ep->datainterval; + else + packs_per_ms = 1; + + if (is_playback && !snd_usb_endpoint_implict_feedback_sink(ep)) { + urb_packs = max(ep->chip->nrpacks, 1); + urb_packs = min(urb_packs, (unsigned int) MAX_PACKS); + } else { + urb_packs = 1; + } + + urb_packs *= packs_per_ms; + + if (sync_ep && !snd_usb_endpoint_implict_feedback_sink(ep)) + urb_packs = min(urb_packs, 1U << sync_ep->syncinterval); + + /* decide how many packets to be used */ + if (is_playback && !snd_usb_endpoint_implict_feedback_sink(ep)) { + unsigned int minsize, maxpacks; + /* determine how small a packet can be */ + minsize = (ep->freqn >> (16 - ep->datainterval)) + * (frame_bits >> 3); + /* with sync from device, assume it can be 12% lower */ + if (sync_ep) + minsize -= minsize >> 3; + minsize = max(minsize, 1u); + total_packs = (period_bytes + minsize - 1) / minsize; + /* we need at least two URBs for queueing */ + if (total_packs < 2) { + total_packs = 2; + } else { + /* and we don't want too long a queue either */ + maxpacks = max(MAX_QUEUE * packs_per_ms, urb_packs * 2); + total_packs = min(total_packs, maxpacks); + } + } else { + while (urb_packs > 1 && urb_packs * maxsize >= period_bytes) + urb_packs >>= 1; + total_packs = MAX_URBS * urb_packs; + } + + ep->nurbs = (total_packs + urb_packs - 1) / urb_packs; + if (ep->nurbs > MAX_URBS) { + /* too much... */ + ep->nurbs = MAX_URBS; + total_packs = MAX_URBS * urb_packs; + } else if (ep->nurbs < 2) { + /* too little - we need at least two packets + * to ensure contiguous playback/capture + */ + ep->nurbs = 2; + } + + /* allocate and initialize data urbs */ + for (i = 0; i < ep->nurbs; i++) { + struct snd_urb_ctx *u = &ep->urb[i]; + u->index = i; + u->ep = ep; + u->packets = (i + 1) * total_packs / ep->nurbs + - i * total_packs / ep->nurbs; + u->buffer_size = maxsize * u->packets; + + if (fmt->fmt_type == UAC_FORMAT_TYPE_II) + u->packets++; /* for transfer delimiter */ + u->urb = usb_alloc_urb(u->packets, GFP_KERNEL); + if (!u->urb) + goto out_of_memory; + + u->urb->transfer_buffer = + usb_alloc_coherent(ep->chip->dev, u->buffer_size, + GFP_KERNEL, &u->urb->transfer_dma); + if (!u->urb->transfer_buffer) + goto out_of_memory; + u->urb->pipe = ep->pipe; + u->urb->transfer_flags = URB_ISO_ASAP | URB_NO_TRANSFER_DMA_MAP; + u->urb->interval = 1 << ep->datainterval; + u->urb->context = u; + u->urb->complete = snd_complete_urb; + INIT_LIST_HEAD(&u->ready_list); + } + + return 0; + +out_of_memory: + release_urbs(ep, 0); + return -ENOMEM; +} + +static int sync_ep_set_params(struct snd_usb_endpoint *ep, + struct snd_pcm_hw_params *hw_params, + struct audioformat *fmt) +{ + int i; + + ep->syncbuf = usb_alloc_coherent(ep->chip->dev, SYNC_URBS * 4, + GFP_KERNEL, &ep->sync_dma); + if (!ep->syncbuf) + return -ENOMEM; + + for (i = 0; i < SYNC_URBS; i++) { + struct snd_urb_ctx *u = &ep->urb[i]; + u->index = i; + u->ep = ep; + u->packets = 1; + u->urb = usb_alloc_urb(1, GFP_KERNEL); + if (!u->urb) + goto out_of_memory; + u->urb->transfer_buffer = ep->syncbuf + i * 4; + u->urb->transfer_dma = ep->sync_dma + i * 4; + u->urb->transfer_buffer_length = 4; + u->urb->pipe = ep->pipe; + u->urb->transfer_flags = URB_ISO_ASAP | + URB_NO_TRANSFER_DMA_MAP; + u->urb->number_of_packets = 1; + u->urb->interval = 1 << ep->syncinterval; + u->urb->context = u; + u->urb->complete = snd_complete_urb; + } + + ep->nurbs = SYNC_URBS; + + return 0; + +out_of_memory: + release_urbs(ep, 0); + return -ENOMEM; +} + +int snd_usb_endpoint_set_params(struct snd_usb_endpoint *ep, + struct snd_pcm_hw_params *hw_params, + struct audioformat *fmt, + struct snd_usb_endpoint *sync_ep) +{ + int err; + + if (ep->use_count != 0) { + snd_printk(KERN_WARNING "Unable to change format on ep #%x: already in use\n", + ep->ep_num); + return -EBUSY; + } + + /* release old buffers, if any */ + release_urbs(ep, 0); + + ep->datainterval = fmt->datainterval; + ep->maxpacksize = fmt->maxpacksize; + ep->fill_max = fmt->attributes & UAC_EP_CS_ATTR_FILL_MAX; + + if (snd_usb_get_speed(ep->chip->dev) == USB_SPEED_FULL) + ep->freqn = get_usb_full_speed_rate(params_rate(hw_params)); + else + ep->freqn = get_usb_high_speed_rate(params_rate(hw_params)); + + /* calculate the frequency in 16.16 format */ + ep->freqm = ep->freqn; + ep->freqshift = INT_MIN; + + ep->phase = 0; + + switch (ep->type) { + case SND_USB_ENDPOINT_TYPE_DATA: + err = data_ep_set_params(ep, hw_params, fmt, sync_ep); + break; + case SND_USB_ENDPOINT_TYPE_SYNC: + err = sync_ep_set_params(ep, hw_params, fmt); + break; + default: + err = -EINVAL; + } + + snd_printdd(KERN_DEBUG "Setting params for ep #%x (type %d, %d urbs), ret=%d\n", + ep->ep_num, ep->type, ep->nurbs, err); + + return err; +} + +int snd_usb_endpoint_start(struct snd_usb_endpoint *ep) +{ + int err; + unsigned int i; + + if (ep->chip->shutdown) + return -EBADFD; + + /* already running? */ + if (++ep->use_count != 1) + return 0; + + if (snd_BUG_ON(!test_bit(EP_FLAG_ACTIVATED, &ep->flags))) + return -EINVAL; + + /* just to be sure */ + deactivate_urbs(ep, 0, 1); + wait_clear_urbs(ep); + + ep->active_mask = 0; + ep->unlink_mask = 0; + ep->phase = 0; + + /* + * If this endpoint has a data endpoint as implicit feedback source, + * don't start the urbs here. Instead, mark them all as available, + * wait for the record urbs to arrive and queue from that context. + */ + + set_bit(EP_FLAG_RUNNING, &ep->flags); + + if (snd_usb_endpoint_implict_feedback_sink(ep)) { + for (i = 0; i < ep->nurbs; i++) { + struct snd_urb_ctx *ctx = ep->urb + i; + list_add_tail(&ctx->ready_list, &ep->ready_playback_urbs); + } + + return 0; + } + + for (i = 0; i < ep->nurbs; i++) { + struct urb *urb = ep->urb[i].urb; + + if (snd_BUG_ON(!urb)) + goto __error; + + if (usb_pipeout(ep->pipe)) { + prepare_outbound_urb_sizes(ep, urb->context); + prepare_outbound_urb(ep, urb->context); + } else { + prepare_inbound_urb(ep, urb->context); + } + + err = usb_submit_urb(urb, GFP_ATOMIC); + if (err < 0) { + snd_printk(KERN_ERR "cannot submit urb %d, error %d: %s\n", + i, err, usb_error_string(err)); + goto __error; + } + set_bit(i, &ep->active_mask); + } + + return 0; + +__error: + clear_bit(EP_FLAG_RUNNING, &ep->flags); + ep->use_count--; + deactivate_urbs(ep, 0, 0); + return -EPIPE; +} + +void snd_usb_endpoint_stop(struct snd_usb_endpoint *ep, + int force, int can_sleep, int wait) +{ + if (!ep) + return; + + if (snd_BUG_ON(ep->use_count == 0)) + return; + + if (snd_BUG_ON(!test_bit(EP_FLAG_ACTIVATED, &ep->flags))) + return; + + if (--ep->use_count == 0) { + deactivate_urbs(ep, force, can_sleep); + ep->data_subs = NULL; + ep->sync_slave = NULL; + ep->retire_data_urb = NULL; + ep->prepare_data_urb = NULL; + + if (wait) + wait_clear_urbs(ep); + } +} + +int snd_usb_endpoint_activate(struct snd_usb_endpoint *ep) +{ + if (ep->use_count != 0) + return 0; + + if (!ep->chip->shutdown && + !test_and_set_bit(EP_FLAG_ACTIVATED, &ep->flags)) { + int ret; + + ret = usb_set_interface(ep->chip->dev, ep->iface, ep->alt_idx); + if (ret < 0) { + snd_printk(KERN_ERR "%s() usb_set_interface() failed, ret = %d\n", + __func__, ret); + clear_bit(EP_FLAG_ACTIVATED, &ep->flags); + return ret; + } + + return 0; + } + + return -EBUSY; +} + +int snd_usb_endpoint_deactivate(struct snd_usb_endpoint *ep) +{ + if (!ep) + return -EINVAL; + + if (ep->use_count != 0) + return 0; + + if (!ep->chip->shutdown && + test_and_clear_bit(EP_FLAG_ACTIVATED, &ep->flags)) { + int ret; + + ret = usb_set_interface(ep->chip->dev, ep->iface, 0); + if (ret < 0) { + snd_printk(KERN_ERR "%s(): usb_set_interface() failed, ret = %d\n", + __func__, ret); + return ret; + } + + return 0; + } + + return -EBUSY; +} + +void snd_usb_endpoint_free(struct list_head *head) +{ + struct snd_usb_endpoint *ep; + + ep = list_entry(head, struct snd_usb_endpoint, list); + release_urbs(ep, 1); + kfree(ep); +} + +/* + * process after playback sync complete + * + * Full speed devices report feedback values in 10.14 format as samples per + * frame, high speed devices in 16.16 format as samples per microframe. + * Because the Audio Class 1 spec was written before USB 2.0, many high speed + * devices use a wrong interpretation, some others use an entirely different + * format. Therefore, we cannot predict what format any particular device uses + * and must detect it automatically. + */ +void snd_usb_handle_sync_urb(struct snd_usb_endpoint *ep, + struct snd_usb_endpoint *sender, + const struct urb *urb) +{ + int shift; + unsigned int f; + unsigned long flags; + + snd_BUG_ON(ep == sender); + + if (snd_usb_endpoint_implict_feedback_sink(ep) && + ep->use_count != 0) { + + /* implicit feedback case */ + int i, bytes = 0; + struct snd_urb_ctx *in_ctx; + struct snd_usb_packet_info *out_packet; + + in_ctx = urb->context; + + /* Count overall packet size */ + for (i = 0; i < in_ctx->packets; i++) + if (urb->iso_frame_desc[i].status == 0) + bytes += urb->iso_frame_desc[i].actual_length; + + /* + * skip empty packets. At least M-Audio's Fast Track Ultra stops + * streaming once it received a 0-byte OUT URB + */ + if (bytes == 0) + return; + + spin_lock_irqsave(&ep->lock, flags); + out_packet = ep->next_packet + ep->next_packet_write_pos; + + /* + * Iterate through the inbound packet and prepare the lengths + * for the output packet. The OUT packet we are about to send + * will have the same amount of payload than the IN packet we + * just received. + */ + + out_packet->packets = in_ctx->packets; + for (i = 0; i < in_ctx->packets; i++) { + if (urb->iso_frame_desc[i].status == 0) + out_packet->packet_size[i] = + urb->iso_frame_desc[i].actual_length / ep->stride; + else + out_packet->packet_size[i] = 0; + } + + ep->next_packet_write_pos++; + ep->next_packet_write_pos %= MAX_URBS; + spin_unlock_irqrestore(&ep->lock, flags); + queue_pending_output_urbs(ep); + + return; + } + + /* parse sync endpoint packet */ + + if (urb->iso_frame_desc[0].status != 0 || + urb->iso_frame_desc[0].actual_length < 3) + return; + + f = le32_to_cpup(urb->transfer_buffer); + if (urb->iso_frame_desc[0].actual_length == 3) + f &= 0x00ffffff; + else + f &= 0x0fffffff; + + if (f == 0) + return; + + if (unlikely(ep->freqshift == INT_MIN)) { + /* + * The first time we see a feedback value, determine its format + * by shifting it left or right until it matches the nominal + * frequency value. This assumes that the feedback does not + * differ from the nominal value more than +50% or -25%. + */ + shift = 0; + while (f < ep->freqn - ep->freqn / 4) { + f <<= 1; + shift++; + } + while (f > ep->freqn + ep->freqn / 2) { + f >>= 1; + shift--; + } + ep->freqshift = shift; + } else if (ep->freqshift >= 0) + f <<= ep->freqshift; + else + f >>= -ep->freqshift; + + if (likely(f >= ep->freqn - ep->freqn / 8 && f <= ep->freqmax)) { + /* + * If the frequency looks valid, set it. + * This value is referred to in prepare_playback_urb(). + */ + spin_lock_irqsave(&ep->lock, flags); + ep->freqm = f; + spin_unlock_irqrestore(&ep->lock, flags); + } else { + /* + * Out of range; maybe the shift value is wrong. + * Reset it so that we autodetect again the next time. + */ + ep->freqshift = INT_MIN; + } +} + diff --git a/sound/usb/endpoint.h b/sound/usb/endpoint.h index 88eb63a636eb..9f083d7d6cf1 100644 --- a/sound/usb/endpoint.h +++ b/sound/usb/endpoint.h @@ -18,4 +18,30 @@ int snd_usb_substream_prepare(struct snd_usb_substream *subs, int snd_usb_substream_playback_trigger(struct snd_pcm_substream *substream, int cmd); int snd_usb_substream_capture_trigger(struct snd_pcm_substream *substream, int cmd); + +#define SND_USB_ENDPOINT_TYPE_DATA 0 +#define SND_USB_ENDPOINT_TYPE_SYNC 1 + +struct snd_usb_endpoint *snd_usb_add_endpoint(struct snd_usb_audio *chip, + struct usb_host_interface *alts, + int ep_num, int direction, int type); + +int snd_usb_endpoint_set_params(struct snd_usb_endpoint *ep, + struct snd_pcm_hw_params *hw_params, + struct audioformat *fmt, + struct snd_usb_endpoint *sync_ep); + +int snd_usb_endpoint_start(struct snd_usb_endpoint *ep); +void snd_usb_endpoint_stop(struct snd_usb_endpoint *ep, + int force, int can_sleep, int wait); +int snd_usb_endpoint_activate(struct snd_usb_endpoint *ep); +int snd_usb_endpoint_deactivate(struct snd_usb_endpoint *ep); +void snd_usb_endpoint_free(struct list_head *head); + +int snd_usb_endpoint_implict_feedback_sink(struct snd_usb_endpoint *ep); + +void snd_usb_handle_sync_urb(struct snd_usb_endpoint *ep, + struct snd_usb_endpoint *sender, + const struct urb *urb); + #endif /* __USBAUDIO_ENDPOINT_H */ diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index a16c21d2d7ed..b8233ebe250f 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -47,6 +47,7 @@ struct snd_usb_audio { int num_suspended_intf; struct list_head pcm_list; /* list of pcm streams */ + struct list_head ep_list; /* list of audio-related endpoints */ int pcm_devs; struct list_head midi_list; /* list of midi interfaces */ -- cgit v1.2.3 From edcd3633e72a1590c4cf46befe5e6cd03b5aec3e Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Thu, 12 Apr 2012 13:51:12 +0200 Subject: ALSA: snd-usb: switch over to new endpoint streaming logic With the previous commit that added the new streaming model, all endpoint and streaming related code is now in endpoint.c, and pcm.c only acts as a wrapper for handling the packet's payload. Signed-off-by: Daniel Mack Signed-off-by: Takashi Iwai --- sound/usb/card.c | 8 +- sound/usb/card.h | 4 + sound/usb/endpoint.c | 40 ----- sound/usb/endpoint.h | 3 - sound/usb/pcm.c | 418 ++++++++++++++++++++++++++++++++++++++++++--------- sound/usb/stream.c | 31 +++- 6 files changed, 385 insertions(+), 119 deletions(-) (limited to 'sound') diff --git a/sound/usb/card.c b/sound/usb/card.c index 6bc88b7ce4fd..d5b5c3388e28 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -131,8 +131,9 @@ static void snd_usb_stream_disconnect(struct list_head *head) subs = &as->substream[idx]; if (!subs->num_formats) continue; - snd_usb_release_substream_urbs(subs, 1); subs->interface = -1; + subs->data_endpoint = NULL; + subs->sync_endpoint = NULL; } } @@ -350,6 +351,7 @@ static int snd_usb_audio_create(struct usb_device *dev, int idx, chip->usb_id = USB_ID(le16_to_cpu(dev->descriptor.idVendor), le16_to_cpu(dev->descriptor.idProduct)); INIT_LIST_HEAD(&chip->pcm_list); + INIT_LIST_HEAD(&chip->ep_list); INIT_LIST_HEAD(&chip->midi_list); INIT_LIST_HEAD(&chip->mixer_list); @@ -567,6 +569,10 @@ static void snd_usb_audio_disconnect(struct usb_device *dev, list_for_each(p, &chip->pcm_list) { snd_usb_stream_disconnect(p); } + /* release the endpoint resources */ + list_for_each(p, &chip->ep_list) { + snd_usb_endpoint_free(p); + } /* release the midi resources */ list_for_each(p, &chip->midi_list) { snd_usbmidi_disconnect(p); diff --git a/sound/usb/card.h b/sound/usb/card.h index 9acbd4a1228d..8a08687e5bc0 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -145,6 +145,10 @@ struct snd_usb_substream { struct snd_urb_ctx syncurb[SYNC_URBS]; /* sync urb table */ char *syncbuf; /* sync buffer for all sync URBs */ dma_addr_t sync_dma; /* DMA address of syncbuf */ + /* data and sync endpoints for this stream */ + struct snd_usb_endpoint *data_endpoint; + struct snd_usb_endpoint *sync_endpoint; + unsigned long flags; u64 formats; /* format bitmasks (all or'ed) */ unsigned int num_formats; /* number of supported audio formats (list) */ diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index ea25265427ad..1b0ed22cff73 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -912,46 +912,6 @@ void snd_usb_init_substream(struct snd_usb_stream *as, subs->fmt_type = fp->fmt_type; } -int snd_usb_substream_playback_trigger(struct snd_pcm_substream *substream, int cmd) -{ - struct snd_usb_substream *subs = substream->runtime->private_data; - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - subs->ops.prepare = prepare_playback_urb; - return 0; - case SNDRV_PCM_TRIGGER_STOP: - return deactivate_urbs_old(subs, 0, 0); - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - subs->ops.prepare = prepare_nodata_playback_urb; - return 0; - } - - return -EINVAL; -} - -int snd_usb_substream_capture_trigger(struct snd_pcm_substream *substream, int cmd) -{ - struct snd_usb_substream *subs = substream->runtime->private_data; - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - subs->ops.retire = retire_capture_urb; - return start_urbs(subs, substream->runtime); - case SNDRV_PCM_TRIGGER_STOP: - return deactivate_urbs_old(subs, 0, 0); - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - subs->ops.retire = retire_paused_capture_urb; - return 0; - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - subs->ops.retire = retire_capture_urb; - return 0; - } - - return -EINVAL; -} - int snd_usb_substream_prepare(struct snd_usb_substream *subs, struct snd_pcm_runtime *runtime) { diff --git a/sound/usb/endpoint.h b/sound/usb/endpoint.h index 9f083d7d6cf1..e540768de3d5 100644 --- a/sound/usb/endpoint.h +++ b/sound/usb/endpoint.h @@ -15,9 +15,6 @@ void snd_usb_release_substream_urbs(struct snd_usb_substream *subs, int force); int snd_usb_substream_prepare(struct snd_usb_substream *subs, struct snd_pcm_runtime *runtime); -int snd_usb_substream_playback_trigger(struct snd_pcm_substream *substream, int cmd); -int snd_usb_substream_capture_trigger(struct snd_pcm_substream *substream, int cmd); - #define SND_USB_ENDPOINT_TYPE_DATA 0 #define SND_USB_ENDPOINT_TYPE_SYNC 1 diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 0eed6115c2d4..0f107834c100 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -16,6 +16,7 @@ #include #include +#include #include #include #include @@ -34,6 +35,9 @@ #include "clock.h" #include "power.h" +#define SUBSTREAM_FLAG_DATA_EP_STARTED 0 +#define SUBSTREAM_FLAG_SYNC_EP_STARTED 1 + /* return the estimated delay based on USB frame counters */ snd_pcm_uframes_t snd_usb_pcm_delay(struct snd_usb_substream *subs, unsigned int rate) @@ -208,6 +212,84 @@ int snd_usb_init_pitch(struct snd_usb_audio *chip, int iface, } } +static int start_endpoints(struct snd_usb_substream *subs) +{ + int err; + + if (!subs->data_endpoint) + return -EINVAL; + + if (!test_and_set_bit(SUBSTREAM_FLAG_DATA_EP_STARTED, &subs->flags)) { + struct snd_usb_endpoint *ep = subs->data_endpoint; + + snd_printdd(KERN_DEBUG "Starting data EP @%p\n", ep); + + ep->data_subs = subs; + err = snd_usb_endpoint_start(ep); + if (err < 0) { + clear_bit(SUBSTREAM_FLAG_DATA_EP_STARTED, &subs->flags); + return err; + } + } + + if (subs->sync_endpoint && + !test_and_set_bit(SUBSTREAM_FLAG_SYNC_EP_STARTED, &subs->flags)) { + struct snd_usb_endpoint *ep = subs->sync_endpoint; + + snd_printdd(KERN_DEBUG "Starting sync EP @%p\n", ep); + + ep->sync_slave = subs->data_endpoint; + err = snd_usb_endpoint_start(ep); + if (err < 0) { + clear_bit(SUBSTREAM_FLAG_SYNC_EP_STARTED, &subs->flags); + return err; + } + } + + return 0; +} + +static void stop_endpoints(struct snd_usb_substream *subs, + int force, int can_sleep, int wait) +{ + if (test_and_clear_bit(SUBSTREAM_FLAG_SYNC_EP_STARTED, &subs->flags)) + snd_usb_endpoint_stop(subs->sync_endpoint, + force, can_sleep, wait); + + if (test_and_clear_bit(SUBSTREAM_FLAG_DATA_EP_STARTED, &subs->flags)) + snd_usb_endpoint_stop(subs->data_endpoint, + force, can_sleep, wait); +} + +static int activate_endpoints(struct snd_usb_substream *subs) +{ + if (subs->sync_endpoint) { + int ret; + + ret = snd_usb_endpoint_activate(subs->sync_endpoint); + if (ret < 0) + return ret; + } + + return snd_usb_endpoint_activate(subs->data_endpoint); +} + +static int deactivate_endpoints(struct snd_usb_substream *subs) +{ + int reta, retb; + + reta = snd_usb_endpoint_deactivate(subs->sync_endpoint); + retb = snd_usb_endpoint_deactivate(subs->data_endpoint); + + if (reta < 0) + return reta; + + if (retb < 0) + return retb; + + return 0; +} + /* * find a matching format and set up the interface */ @@ -232,40 +314,11 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt) if (fmt == subs->cur_audiofmt) return 0; - /* close the old interface */ - if (subs->interface >= 0 && subs->interface != fmt->iface) { - if (usb_set_interface(subs->dev, subs->interface, 0) < 0) { - snd_printk(KERN_ERR "%d:%d:%d: return to setting 0 failed\n", - dev->devnum, fmt->iface, fmt->altsetting); - return -EIO; - } - subs->interface = -1; - subs->altset_idx = 0; - } - - /* set interface */ - if (subs->interface != fmt->iface || subs->altset_idx != fmt->altset_idx) { - if (usb_set_interface(dev, fmt->iface, fmt->altsetting) < 0) { - snd_printk(KERN_ERR "%d:%d:%d: usb_set_interface failed\n", - dev->devnum, fmt->iface, fmt->altsetting); - return -EIO; - } - snd_printdd(KERN_INFO "setting usb interface %d:%d\n", fmt->iface, fmt->altsetting); - subs->interface = fmt->iface; - subs->altset_idx = fmt->altset_idx; - } - - /* create a data pipe */ - ep = fmt->endpoint & USB_ENDPOINT_NUMBER_MASK; - if (is_playback) - subs->datapipe = usb_sndisocpipe(dev, ep); - else - subs->datapipe = usb_rcvisocpipe(dev, ep); - subs->datainterval = fmt->datainterval; - subs->syncpipe = subs->syncinterval = 0; - subs->maxpacksize = fmt->maxpacksize; - subs->syncmaxsize = 0; - subs->fill_max = 0; + subs->data_endpoint = snd_usb_add_endpoint(subs->stream->chip, + alts, fmt->endpoint, subs->direction, + SND_USB_ENDPOINT_TYPE_DATA); + if (!subs->data_endpoint) + return -EINVAL; /* we need a sync pipe in async OUT or adaptive IN mode */ /* check the number of EP, since some devices have broken @@ -276,6 +329,15 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt) if (((is_playback && attr == USB_ENDPOINT_SYNC_ASYNC) || (! is_playback && attr == USB_ENDPOINT_SYNC_ADAPTIVE)) && altsd->bNumEndpoints >= 2) { + switch (subs->stream->chip->usb_id) { + case USB_ID(0x0763, 0x2080): /* M-Audio FastTrack Ultra */ + case USB_ID(0x0763, 0x2081): + ep = 0x81; + iface = usb_ifnum_to_if(dev, 2); + alts = &iface->altsetting[1]; + goto add_sync_ep; + } + /* check sync-pipe endpoint */ /* ... and check descriptor size before accessing bSynchAddress because there is a version of the SB Audigy 2 NX firmware lacking @@ -295,28 +357,16 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt) dev->devnum, fmt->iface, fmt->altsetting); return -EINVAL; } - ep &= USB_ENDPOINT_NUMBER_MASK; - if (is_playback) - subs->syncpipe = usb_rcvisocpipe(dev, ep); - else - subs->syncpipe = usb_sndisocpipe(dev, ep); - if (get_endpoint(alts, 1)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE && - get_endpoint(alts, 1)->bRefresh >= 1 && - get_endpoint(alts, 1)->bRefresh <= 9) - subs->syncinterval = get_endpoint(alts, 1)->bRefresh; - else if (snd_usb_get_speed(subs->dev) == USB_SPEED_FULL) - subs->syncinterval = 1; - else if (get_endpoint(alts, 1)->bInterval >= 1 && - get_endpoint(alts, 1)->bInterval <= 16) - subs->syncinterval = get_endpoint(alts, 1)->bInterval - 1; - else - subs->syncinterval = 3; - subs->syncmaxsize = le16_to_cpu(get_endpoint(alts, 1)->wMaxPacketSize); - } - - /* always fill max packet size */ - if (fmt->attributes & UAC_EP_CS_ATTR_FILL_MAX) - subs->fill_max = 1; +add_sync_ep: + subs->sync_endpoint = snd_usb_add_endpoint(subs->stream->chip, + alts, ep, !subs->direction, + SND_USB_ENDPOINT_TYPE_SYNC); + + if (!subs->sync_endpoint) + return -EINVAL; + + subs->data_endpoint->sync_master = subs->sync_endpoint; + } if ((err = snd_usb_init_pitch(subs->stream->chip, subs->interface, alts, fmt)) < 0) return err; @@ -390,12 +440,22 @@ static int snd_usb_hw_params(struct snd_pcm_substream *substream, if (changed) { mutex_lock(&subs->stream->chip->shutdown_mutex); /* format changed */ - snd_usb_release_substream_urbs(subs, 0); - /* influenced: period_bytes, channels, rate, format, */ - ret = snd_usb_init_substream_urbs(subs, params_period_bytes(hw_params), - params_rate(hw_params), - snd_pcm_format_physical_width(params_format(hw_params)) * - params_channels(hw_params)); + stop_endpoints(subs, 0, 0, 0); + deactivate_endpoints(subs); + + ret = activate_endpoints(subs); + if (ret < 0) + goto unlock; + + ret = snd_usb_endpoint_set_params(subs->data_endpoint, hw_params, fmt, + subs->sync_endpoint); + if (ret < 0) + goto unlock; + + if (subs->sync_endpoint) + ret = snd_usb_endpoint_set_params(subs->sync_endpoint, + hw_params, fmt, NULL); +unlock: mutex_unlock(&subs->stream->chip->shutdown_mutex); } @@ -415,7 +475,7 @@ static int snd_usb_hw_free(struct snd_pcm_substream *substream) subs->cur_rate = 0; subs->period_bytes = 0; mutex_lock(&subs->stream->chip->shutdown_mutex); - snd_usb_release_substream_urbs(subs, 0); + stop_endpoints(subs, 0, 1, 1); mutex_unlock(&subs->stream->chip->shutdown_mutex); return snd_pcm_lib_free_vmalloc_buffer(substream); } @@ -435,19 +495,28 @@ static int snd_usb_pcm_prepare(struct snd_pcm_substream *substream) return -ENXIO; } + if (snd_BUG_ON(!subs->data_endpoint)) + return -EIO; + /* some unit conversions in runtime */ - subs->maxframesize = bytes_to_frames(runtime, subs->maxpacksize); - subs->curframesize = bytes_to_frames(runtime, subs->curpacksize); + subs->data_endpoint->maxframesize = + bytes_to_frames(runtime, subs->data_endpoint->maxpacksize); + subs->data_endpoint->curframesize = + bytes_to_frames(runtime, subs->data_endpoint->curpacksize); /* reset the pointer */ subs->hwptr_done = 0; subs->transfer_done = 0; - subs->phase = 0; subs->last_delay = 0; subs->last_frame_number = 0; runtime->delay = 0; - return snd_usb_substream_prepare(subs, runtime); + /* for playback, submit the URBs now; otherwise, the first hwptr_done + * updates for all URBs would happen at the same time when starting */ + if (subs->direction == SNDRV_PCM_STREAM_PLAYBACK) + return start_endpoints(subs); + + return 0; } static struct snd_pcm_hardware snd_usb_hardware = @@ -842,16 +911,171 @@ static int snd_usb_pcm_open(struct snd_pcm_substream *substream, int direction) static int snd_usb_pcm_close(struct snd_pcm_substream *substream, int direction) { + int ret; struct snd_usb_stream *as = snd_pcm_substream_chip(substream); struct snd_usb_substream *subs = &as->substream[direction]; - if (!as->chip->shutdown && subs->interface >= 0) { - usb_set_interface(subs->dev, subs->interface, 0); - subs->interface = -1; - } + stop_endpoints(subs, 0, 0, 0); + ret = deactivate_endpoints(subs); subs->pcm_substream = NULL; snd_usb_autosuspend(subs->stream->chip); - return 0; + + return ret; +} + +/* Since a URB can handle only a single linear buffer, we must use double + * buffering when the data to be transferred overflows the buffer boundary. + * To avoid inconsistencies when updating hwptr_done, we use double buffering + * for all URBs. + */ +static void retire_capture_urb(struct snd_usb_substream *subs, + struct urb *urb) +{ + struct snd_pcm_runtime *runtime = subs->pcm_substream->runtime; + unsigned int stride, frames, bytes, oldptr; + int i, period_elapsed = 0; + unsigned long flags; + unsigned char *cp; + + stride = runtime->frame_bits >> 3; + + for (i = 0; i < urb->number_of_packets; i++) { + cp = (unsigned char *)urb->transfer_buffer + urb->iso_frame_desc[i].offset; + if (urb->iso_frame_desc[i].status && printk_ratelimit()) { + snd_printdd(KERN_ERR "frame %d active: %d\n", i, urb->iso_frame_desc[i].status); + // continue; + } + bytes = urb->iso_frame_desc[i].actual_length; + frames = bytes / stride; + if (!subs->txfr_quirk) + bytes = frames * stride; + if (bytes % (runtime->sample_bits >> 3) != 0) { +#ifdef CONFIG_SND_DEBUG_VERBOSE + int oldbytes = bytes; +#endif + bytes = frames * stride; + snd_printdd(KERN_ERR "Corrected urb data len. %d->%d\n", + oldbytes, bytes); + } + /* update the current pointer */ + spin_lock_irqsave(&subs->lock, flags); + oldptr = subs->hwptr_done; + subs->hwptr_done += bytes; + if (subs->hwptr_done >= runtime->buffer_size * stride) + subs->hwptr_done -= runtime->buffer_size * stride; + frames = (bytes + (oldptr % stride)) / stride; + subs->transfer_done += frames; + if (subs->transfer_done >= runtime->period_size) { + subs->transfer_done -= runtime->period_size; + period_elapsed = 1; + } + spin_unlock_irqrestore(&subs->lock, flags); + /* copy a data chunk */ + if (oldptr + bytes > runtime->buffer_size * stride) { + unsigned int bytes1 = + runtime->buffer_size * stride - oldptr; + memcpy(runtime->dma_area + oldptr, cp, bytes1); + memcpy(runtime->dma_area, cp + bytes1, bytes - bytes1); + } else { + memcpy(runtime->dma_area + oldptr, cp, bytes); + } + } + + if (period_elapsed) + snd_pcm_period_elapsed(subs->pcm_substream); +} + +static void prepare_playback_urb(struct snd_usb_substream *subs, + struct urb *urb) +{ + struct snd_pcm_runtime *runtime = subs->pcm_substream->runtime; + struct snd_urb_ctx *ctx = urb->context; + unsigned int counts, frames, bytes; + int i, stride, period_elapsed = 0; + unsigned long flags; + + stride = runtime->frame_bits >> 3; + + frames = 0; + urb->number_of_packets = 0; + spin_lock_irqsave(&subs->lock, flags); + for (i = 0; i < ctx->packets; i++) { + counts = ctx->packet_size[i]; + /* set up descriptor */ + urb->iso_frame_desc[i].offset = frames * stride; + urb->iso_frame_desc[i].length = counts * stride; + frames += counts; + urb->number_of_packets++; + subs->transfer_done += counts; + if (subs->transfer_done >= runtime->period_size) { + subs->transfer_done -= runtime->period_size; + period_elapsed = 1; + if (subs->fmt_type == UAC_FORMAT_TYPE_II) { + if (subs->transfer_done > 0) { + /* FIXME: fill-max mode is not + * supported yet */ + frames -= subs->transfer_done; + counts -= subs->transfer_done; + urb->iso_frame_desc[i].length = + counts * stride; + subs->transfer_done = 0; + } + i++; + if (i < ctx->packets) { + /* add a transfer delimiter */ + urb->iso_frame_desc[i].offset = + frames * stride; + urb->iso_frame_desc[i].length = 0; + urb->number_of_packets++; + } + break; + } + } + if (period_elapsed && + !snd_usb_endpoint_implict_feedback_sink(subs->data_endpoint)) /* finish at the period boundary */ + break; + } + bytes = frames * stride; + if (subs->hwptr_done + bytes > runtime->buffer_size * stride) { + /* err, the transferred area goes over buffer boundary. */ + unsigned int bytes1 = + runtime->buffer_size * stride - subs->hwptr_done; + memcpy(urb->transfer_buffer, + runtime->dma_area + subs->hwptr_done, bytes1); + memcpy(urb->transfer_buffer + bytes1, + runtime->dma_area, bytes - bytes1); + } else { + memcpy(urb->transfer_buffer, + runtime->dma_area + subs->hwptr_done, bytes); + } + subs->hwptr_done += bytes; + if (subs->hwptr_done >= runtime->buffer_size * stride) + subs->hwptr_done -= runtime->buffer_size * stride; + runtime->delay += frames; + spin_unlock_irqrestore(&subs->lock, flags); + urb->transfer_buffer_length = bytes; + if (period_elapsed) + snd_pcm_period_elapsed(subs->pcm_substream); +} + +/* + * process after playback data complete + * - decrease the delay count again + */ +static void retire_playback_urb(struct snd_usb_substream *subs, + struct urb *urb) +{ + unsigned long flags; + struct snd_pcm_runtime *runtime = subs->pcm_substream->runtime; + int stride = runtime->frame_bits >> 3; + int processed = urb->transfer_buffer_length / stride; + + spin_lock_irqsave(&subs->lock, flags); + if (processed > runtime->delay) + runtime->delay = 0; + else + runtime->delay -= processed; + spin_unlock_irqrestore(&subs->lock, flags); } static int snd_usb_playback_open(struct snd_pcm_substream *substream) @@ -874,6 +1098,56 @@ static int snd_usb_capture_close(struct snd_pcm_substream *substream) return snd_usb_pcm_close(substream, SNDRV_PCM_STREAM_CAPTURE); } +static int snd_usb_substream_playback_trigger(struct snd_pcm_substream *substream, + int cmd) +{ + struct snd_usb_substream *subs = substream->runtime->private_data; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + subs->data_endpoint->prepare_data_urb = prepare_playback_urb; + subs->data_endpoint->retire_data_urb = retire_playback_urb; + return 0; + case SNDRV_PCM_TRIGGER_STOP: + stop_endpoints(subs, 0, 0, 0); + return 0; + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + subs->data_endpoint->prepare_data_urb = NULL; + subs->data_endpoint->retire_data_urb = NULL; + return 0; + } + + return -EINVAL; +} + +int snd_usb_substream_capture_trigger(struct snd_pcm_substream *substream, int cmd) +{ + int err; + struct snd_usb_substream *subs = substream->runtime->private_data; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + err = start_endpoints(subs); + if (err < 0) + return err; + + subs->data_endpoint->retire_data_urb = retire_capture_urb; + return 0; + case SNDRV_PCM_TRIGGER_STOP: + stop_endpoints(subs, 0, 0, 0); + return 0; + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + subs->data_endpoint->retire_data_urb = NULL; + return 0; + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + subs->data_endpoint->retire_data_urb = retire_capture_urb; + return 0; + } + + return -EINVAL; +} + static struct snd_pcm_ops snd_usb_playback_ops = { .open = snd_usb_playback_open, .close = snd_usb_playback_close, diff --git a/sound/usb/stream.c b/sound/usb/stream.c index 5ff8010b2d6f..6b7d7a2b7baa 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -73,6 +73,31 @@ static void snd_usb_audio_pcm_free(struct snd_pcm *pcm) } } +/* + * initialize the substream instance. + */ + +static void snd_usb_init_substream(struct snd_usb_stream *as, + int stream, + struct audioformat *fp) +{ + struct snd_usb_substream *subs = &as->substream[stream]; + + INIT_LIST_HEAD(&subs->fmt_list); + spin_lock_init(&subs->lock); + + subs->stream = as; + subs->direction = stream; + subs->dev = as->chip->dev; + subs->txfr_quirk = as->chip->txfr_quirk; + + snd_usb_set_pcm_ops(as->pcm, stream); + + list_add_tail(&fp->list, &subs->fmt_list); + subs->formats |= fp->formats; + subs->num_formats++; + subs->fmt_type = fp->fmt_type; +} /* * add this endpoint to the chip instance. @@ -94,9 +119,9 @@ int snd_usb_add_audio_stream(struct snd_usb_audio *chip, if (as->fmt_type != fp->fmt_type) continue; subs = &as->substream[stream]; - if (!subs->endpoint) + if (!subs->data_endpoint) continue; - if (subs->endpoint == fp->endpoint) { + if (subs->data_endpoint->ep_num == fp->endpoint) { list_add_tail(&fp->list, &subs->fmt_list); subs->num_formats++; subs->formats |= fp->formats; @@ -109,7 +134,7 @@ int snd_usb_add_audio_stream(struct snd_usb_audio *chip, if (as->fmt_type != fp->fmt_type) continue; subs = &as->substream[stream]; - if (subs->endpoint) + if (subs->data_endpoint) continue; err = snd_pcm_new_stream(as->pcm, stream, 1); if (err < 0) -- cgit v1.2.3 From d399ff9593e088d33fb38f5206c6427825892baa Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Thu, 12 Apr 2012 13:51:13 +0200 Subject: ALSA: snd-usb: remove old streaming logic Signed-off-by: Daniel Mack Signed-off-by: Takashi Iwai --- sound/usb/endpoint.c | 850 +-------------------------------------------------- sound/usb/endpoint.h | 15 - 2 files changed, 6 insertions(+), 859 deletions(-) (limited to 'sound') diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 1b0ed22cff73..64853f76c10c 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -53,728 +53,17 @@ static inline unsigned get_usb_high_speed_rate(unsigned int rate) return ((rate << 10) + 62) / 125; } -/* - * unlink active urbs. - */ -static int deactivate_urbs_old(struct snd_usb_substream *subs, int force, int can_sleep) -{ - struct snd_usb_audio *chip = subs->stream->chip; - unsigned int i; - int async; - - subs->running = 0; - - if (!force && subs->stream->chip->shutdown) /* to be sure... */ - return -EBADFD; - - async = !can_sleep && chip->async_unlink; - - if (!async && in_interrupt()) - return 0; - - for (i = 0; i < subs->nurbs; i++) { - if (test_bit(i, &subs->active_mask)) { - if (!test_and_set_bit(i, &subs->unlink_mask)) { - struct urb *u = subs->dataurb[i].urb; - if (async) - usb_unlink_urb(u); - else - usb_kill_urb(u); - } - } - } - if (subs->syncpipe) { - for (i = 0; i < SYNC_URBS; i++) { - if (test_bit(i+16, &subs->active_mask)) { - if (!test_and_set_bit(i+16, &subs->unlink_mask)) { - struct urb *u = subs->syncurb[i].urb; - if (async) - usb_unlink_urb(u); - else - usb_kill_urb(u); - } - } - } - } - return 0; -} - - /* * release a urb data */ static void release_urb_ctx(struct snd_urb_ctx *u) { - if (u->urb) { - if (u->buffer_size) - usb_free_coherent(u->subs->dev, u->buffer_size, - u->urb->transfer_buffer, - u->urb->transfer_dma); - usb_free_urb(u->urb); - u->urb = NULL; - } -} - -/* - * wait until all urbs are processed. - */ -static int wait_clear_urbs_old(struct snd_usb_substream *subs) -{ - unsigned long end_time = jiffies + msecs_to_jiffies(1000); - unsigned int i; - int alive; - - do { - alive = 0; - for (i = 0; i < subs->nurbs; i++) { - if (test_bit(i, &subs->active_mask)) - alive++; - } - if (subs->syncpipe) { - for (i = 0; i < SYNC_URBS; i++) { - if (test_bit(i + 16, &subs->active_mask)) - alive++; - } - } - if (! alive) - break; - schedule_timeout_uninterruptible(1); - } while (time_before(jiffies, end_time)); - if (alive) - snd_printk(KERN_ERR "timeout: still %d active urbs..\n", alive); - return 0; -} - -/* - * release a substream - */ -void snd_usb_release_substream_urbs(struct snd_usb_substream *subs, int force) -{ - int i; - - /* stop urbs (to be sure) */ - deactivate_urbs_old(subs, force, 1); - wait_clear_urbs_old(subs); - - for (i = 0; i < MAX_URBS; i++) - release_urb_ctx(&subs->dataurb[i]); - for (i = 0; i < SYNC_URBS; i++) - release_urb_ctx(&subs->syncurb[i]); - usb_free_coherent(subs->dev, SYNC_URBS * 4, - subs->syncbuf, subs->sync_dma); - subs->syncbuf = NULL; - subs->nurbs = 0; -} - -/* - * complete callback from data urb - */ -static void snd_complete_urb_old(struct urb *urb) -{ - struct snd_urb_ctx *ctx = urb->context; - struct snd_usb_substream *subs = ctx->subs; - struct snd_pcm_substream *substream = ctx->subs->pcm_substream; - int err = 0; - - if ((subs->running && subs->ops.retire(subs, substream->runtime, urb)) || - !subs->running || /* can be stopped during retire callback */ - (err = subs->ops.prepare(subs, substream->runtime, urb)) < 0 || - (err = usb_submit_urb(urb, GFP_ATOMIC)) < 0) { - clear_bit(ctx->index, &subs->active_mask); - if (err < 0) { - snd_printd(KERN_ERR "cannot submit urb (err = %d)\n", err); - snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); - } - } -} - - -/* - * complete callback from sync urb - */ -static void snd_complete_sync_urb(struct urb *urb) -{ - struct snd_urb_ctx *ctx = urb->context; - struct snd_usb_substream *subs = ctx->subs; - struct snd_pcm_substream *substream = ctx->subs->pcm_substream; - int err = 0; - - if ((subs->running && subs->ops.retire_sync(subs, substream->runtime, urb)) || - !subs->running || /* can be stopped during retire callback */ - (err = subs->ops.prepare_sync(subs, substream->runtime, urb)) < 0 || - (err = usb_submit_urb(urb, GFP_ATOMIC)) < 0) { - clear_bit(ctx->index + 16, &subs->active_mask); - if (err < 0) { - snd_printd(KERN_ERR "cannot submit sync urb (err = %d)\n", err); - snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); - } - } -} - - -/* - * initialize a substream for plaback/capture - */ -int snd_usb_init_substream_urbs(struct snd_usb_substream *subs, - unsigned int period_bytes, - unsigned int rate, - unsigned int frame_bits) -{ - unsigned int maxsize, i; - int is_playback = subs->direction == SNDRV_PCM_STREAM_PLAYBACK; - unsigned int urb_packs, total_packs, packs_per_ms; - struct snd_usb_audio *chip = subs->stream->chip; - - /* calculate the frequency in 16.16 format */ - if (snd_usb_get_speed(subs->dev) == USB_SPEED_FULL) - subs->freqn = get_usb_full_speed_rate(rate); - else - subs->freqn = get_usb_high_speed_rate(rate); - subs->freqm = subs->freqn; - subs->freqshift = INT_MIN; - /* calculate max. frequency */ - if (subs->maxpacksize) { - /* whatever fits into a max. size packet */ - maxsize = subs->maxpacksize; - subs->freqmax = (maxsize / (frame_bits >> 3)) - << (16 - subs->datainterval); - } else { - /* no max. packet size: just take 25% higher than nominal */ - subs->freqmax = subs->freqn + (subs->freqn >> 2); - maxsize = ((subs->freqmax + 0xffff) * (frame_bits >> 3)) - >> (16 - subs->datainterval); - } - subs->phase = 0; - - if (subs->fill_max) - subs->curpacksize = subs->maxpacksize; - else - subs->curpacksize = maxsize; - - if (snd_usb_get_speed(subs->dev) != USB_SPEED_FULL) - packs_per_ms = 8 >> subs->datainterval; - else - packs_per_ms = 1; - - if (is_playback) { - urb_packs = max(chip->nrpacks, 1); - urb_packs = min(urb_packs, (unsigned int)MAX_PACKS); - } else - urb_packs = 1; - urb_packs *= packs_per_ms; - if (subs->syncpipe) - urb_packs = min(urb_packs, 1U << subs->syncinterval); - - /* decide how many packets to be used */ - if (is_playback) { - unsigned int minsize, maxpacks; - /* determine how small a packet can be */ - minsize = (subs->freqn >> (16 - subs->datainterval)) - * (frame_bits >> 3); - /* with sync from device, assume it can be 12% lower */ - if (subs->syncpipe) - minsize -= minsize >> 3; - minsize = max(minsize, 1u); - total_packs = (period_bytes + minsize - 1) / minsize; - /* we need at least two URBs for queueing */ - if (total_packs < 2) { - total_packs = 2; - } else { - /* and we don't want too long a queue either */ - maxpacks = max(MAX_QUEUE * packs_per_ms, urb_packs * 2); - total_packs = min(total_packs, maxpacks); - } - } else { - while (urb_packs > 1 && urb_packs * maxsize >= period_bytes) - urb_packs >>= 1; - total_packs = MAX_URBS * urb_packs; - } - subs->nurbs = (total_packs + urb_packs - 1) / urb_packs; - if (subs->nurbs > MAX_URBS) { - /* too much... */ - subs->nurbs = MAX_URBS; - total_packs = MAX_URBS * urb_packs; - } else if (subs->nurbs < 2) { - /* too little - we need at least two packets - * to ensure contiguous playback/capture - */ - subs->nurbs = 2; - } - - /* allocate and initialize data urbs */ - for (i = 0; i < subs->nurbs; i++) { - struct snd_urb_ctx *u = &subs->dataurb[i]; - u->index = i; - u->subs = subs; - u->packets = (i + 1) * total_packs / subs->nurbs - - i * total_packs / subs->nurbs; - u->buffer_size = maxsize * u->packets; - if (subs->fmt_type == UAC_FORMAT_TYPE_II) - u->packets++; /* for transfer delimiter */ - u->urb = usb_alloc_urb(u->packets, GFP_KERNEL); - if (!u->urb) - goto out_of_memory; - u->urb->transfer_buffer = - usb_alloc_coherent(subs->dev, u->buffer_size, - GFP_KERNEL, &u->urb->transfer_dma); - if (!u->urb->transfer_buffer) - goto out_of_memory; - u->urb->pipe = subs->datapipe; - u->urb->transfer_flags = URB_ISO_ASAP | URB_NO_TRANSFER_DMA_MAP; - u->urb->interval = 1 << subs->datainterval; - u->urb->context = u; - u->urb->complete = snd_complete_urb_old; - } - - if (subs->syncpipe) { - /* allocate and initialize sync urbs */ - subs->syncbuf = usb_alloc_coherent(subs->dev, SYNC_URBS * 4, - GFP_KERNEL, &subs->sync_dma); - if (!subs->syncbuf) - goto out_of_memory; - for (i = 0; i < SYNC_URBS; i++) { - struct snd_urb_ctx *u = &subs->syncurb[i]; - u->index = i; - u->subs = subs; - u->packets = 1; - u->urb = usb_alloc_urb(1, GFP_KERNEL); - if (!u->urb) - goto out_of_memory; - u->urb->transfer_buffer = subs->syncbuf + i * 4; - u->urb->transfer_dma = subs->sync_dma + i * 4; - u->urb->transfer_buffer_length = 4; - u->urb->pipe = subs->syncpipe; - u->urb->transfer_flags = URB_ISO_ASAP | - URB_NO_TRANSFER_DMA_MAP; - u->urb->number_of_packets = 1; - u->urb->interval = 1 << subs->syncinterval; - u->urb->context = u; - u->urb->complete = snd_complete_sync_urb; - } - } - return 0; - -out_of_memory: - snd_usb_release_substream_urbs(subs, 0); - return -ENOMEM; -} - -/* - * prepare urb for full speed capture sync pipe - * - * fill the length and offset of each urb descriptor. - * the fixed 10.14 frequency is passed through the pipe. - */ -static int prepare_capture_sync_urb(struct snd_usb_substream *subs, - struct snd_pcm_runtime *runtime, - struct urb *urb) -{ - unsigned char *cp = urb->transfer_buffer; - struct snd_urb_ctx *ctx = urb->context; - - urb->dev = ctx->subs->dev; /* we need to set this at each time */ - urb->iso_frame_desc[0].length = 3; - urb->iso_frame_desc[0].offset = 0; - cp[0] = subs->freqn >> 2; - cp[1] = subs->freqn >> 10; - cp[2] = subs->freqn >> 18; - return 0; -} - -/* - * prepare urb for high speed capture sync pipe - * - * fill the length and offset of each urb descriptor. - * the fixed 12.13 frequency is passed as 16.16 through the pipe. - */ -static int prepare_capture_sync_urb_hs(struct snd_usb_substream *subs, - struct snd_pcm_runtime *runtime, - struct urb *urb) -{ - unsigned char *cp = urb->transfer_buffer; - struct snd_urb_ctx *ctx = urb->context; - - urb->dev = ctx->subs->dev; /* we need to set this at each time */ - urb->iso_frame_desc[0].length = 4; - urb->iso_frame_desc[0].offset = 0; - cp[0] = subs->freqn; - cp[1] = subs->freqn >> 8; - cp[2] = subs->freqn >> 16; - cp[3] = subs->freqn >> 24; - return 0; -} - -/* - * process after capture sync complete - * - nothing to do - */ -static int retire_capture_sync_urb(struct snd_usb_substream *subs, - struct snd_pcm_runtime *runtime, - struct urb *urb) -{ - return 0; -} - -/* - * prepare urb for capture data pipe - * - * fill the offset and length of each descriptor. - * - * we use a temporary buffer to write the captured data. - * since the length of written data is determined by host, we cannot - * write onto the pcm buffer directly... the data is thus copied - * later at complete callback to the global buffer. - */ -static int prepare_capture_urb(struct snd_usb_substream *subs, - struct snd_pcm_runtime *runtime, - struct urb *urb) -{ - int i, offs; - struct snd_urb_ctx *ctx = urb->context; - - offs = 0; - urb->dev = ctx->subs->dev; /* we need to set this at each time */ - for (i = 0; i < ctx->packets; i++) { - urb->iso_frame_desc[i].offset = offs; - urb->iso_frame_desc[i].length = subs->curpacksize; - offs += subs->curpacksize; - } - urb->transfer_buffer_length = offs; - urb->number_of_packets = ctx->packets; - return 0; -} - -/* - * process after capture complete - * - * copy the data from each desctiptor to the pcm buffer, and - * update the current position. - */ -static int retire_capture_urb(struct snd_usb_substream *subs, - struct snd_pcm_runtime *runtime, - struct urb *urb) -{ - unsigned long flags; - unsigned char *cp; - int i; - unsigned int stride, frames, bytes, oldptr; - int period_elapsed = 0; - - stride = runtime->frame_bits >> 3; - - for (i = 0; i < urb->number_of_packets; i++) { - cp = (unsigned char *)urb->transfer_buffer + urb->iso_frame_desc[i].offset; - if (urb->iso_frame_desc[i].status && printk_ratelimit()) { - snd_printdd("frame %d active: %d\n", i, urb->iso_frame_desc[i].status); - // continue; - } - bytes = urb->iso_frame_desc[i].actual_length; - frames = bytes / stride; - if (!subs->txfr_quirk) - bytes = frames * stride; - if (bytes % (runtime->sample_bits >> 3) != 0) { -#ifdef CONFIG_SND_DEBUG_VERBOSE - int oldbytes = bytes; -#endif - bytes = frames * stride; - snd_printdd(KERN_ERR "Corrected urb data len. %d->%d\n", - oldbytes, bytes); - } - /* update the current pointer */ - spin_lock_irqsave(&subs->lock, flags); - oldptr = subs->hwptr_done; - subs->hwptr_done += bytes; - if (subs->hwptr_done >= runtime->buffer_size * stride) - subs->hwptr_done -= runtime->buffer_size * stride; - frames = (bytes + (oldptr % stride)) / stride; - subs->transfer_done += frames; - if (subs->transfer_done >= runtime->period_size) { - subs->transfer_done -= runtime->period_size; - period_elapsed = 1; - } - spin_unlock_irqrestore(&subs->lock, flags); - /* copy a data chunk */ - if (oldptr + bytes > runtime->buffer_size * stride) { - unsigned int bytes1 = - runtime->buffer_size * stride - oldptr; - memcpy(runtime->dma_area + oldptr, cp, bytes1); - memcpy(runtime->dma_area, cp + bytes1, bytes - bytes1); - } else { - memcpy(runtime->dma_area + oldptr, cp, bytes); - } - } - if (period_elapsed) - snd_pcm_period_elapsed(subs->pcm_substream); - return 0; -} - -/* - * Process after capture complete when paused. Nothing to do. - */ -static int retire_paused_capture_urb(struct snd_usb_substream *subs, - struct snd_pcm_runtime *runtime, - struct urb *urb) -{ - return 0; -} - - -/* - * prepare urb for playback sync pipe - * - * set up the offset and length to receive the current frequency. - */ -static int prepare_playback_sync_urb(struct snd_usb_substream *subs, - struct snd_pcm_runtime *runtime, - struct urb *urb) -{ - struct snd_urb_ctx *ctx = urb->context; - - urb->dev = ctx->subs->dev; /* we need to set this at each time */ - urb->iso_frame_desc[0].length = min(4u, ctx->subs->syncmaxsize); - urb->iso_frame_desc[0].offset = 0; - return 0; -} - -/* - * process after playback sync complete - * - * Full speed devices report feedback values in 10.14 format as samples per - * frame, high speed devices in 16.16 format as samples per microframe. - * Because the Audio Class 1 spec was written before USB 2.0, many high speed - * devices use a wrong interpretation, some others use an entirely different - * format. Therefore, we cannot predict what format any particular device uses - * and must detect it automatically. - */ -static int retire_playback_sync_urb(struct snd_usb_substream *subs, - struct snd_pcm_runtime *runtime, - struct urb *urb) -{ - unsigned int f; - int shift; - unsigned long flags; - - if (urb->iso_frame_desc[0].status != 0 || - urb->iso_frame_desc[0].actual_length < 3) - return 0; - - f = le32_to_cpup(urb->transfer_buffer); - if (urb->iso_frame_desc[0].actual_length == 3) - f &= 0x00ffffff; - else - f &= 0x0fffffff; - if (f == 0) - return 0; - - if (unlikely(subs->freqshift == INT_MIN)) { - /* - * The first time we see a feedback value, determine its format - * by shifting it left or right until it matches the nominal - * frequency value. This assumes that the feedback does not - * differ from the nominal value more than +50% or -25%. - */ - shift = 0; - while (f < subs->freqn - subs->freqn / 4) { - f <<= 1; - shift++; - } - while (f > subs->freqn + subs->freqn / 2) { - f >>= 1; - shift--; - } - subs->freqshift = shift; - } - else if (subs->freqshift >= 0) - f <<= subs->freqshift; - else - f >>= -subs->freqshift; - - if (likely(f >= subs->freqn - subs->freqn / 8 && f <= subs->freqmax)) { - /* - * If the frequency looks valid, set it. - * This value is referred to in prepare_playback_urb(). - */ - spin_lock_irqsave(&subs->lock, flags); - subs->freqm = f; - spin_unlock_irqrestore(&subs->lock, flags); - } else { - /* - * Out of range; maybe the shift value is wrong. - * Reset it so that we autodetect again the next time. - */ - subs->freqshift = INT_MIN; - } - - return 0; -} - -/* determine the number of frames in the next packet */ -static int snd_usb_audio_next_packet_size(struct snd_usb_substream *subs) -{ - if (subs->fill_max) - return subs->maxframesize; - else { - subs->phase = (subs->phase & 0xffff) - + (subs->freqm << subs->datainterval); - return min(subs->phase >> 16, subs->maxframesize); - } -} - -/* - * Prepare urb for streaming before playback starts or when paused. - * - * We don't have any data, so we send silence. - */ -static int prepare_nodata_playback_urb(struct snd_usb_substream *subs, - struct snd_pcm_runtime *runtime, - struct urb *urb) -{ - unsigned int i, offs, counts; - struct snd_urb_ctx *ctx = urb->context; - int stride = runtime->frame_bits >> 3; - - offs = 0; - urb->dev = ctx->subs->dev; - for (i = 0; i < ctx->packets; ++i) { - counts = snd_usb_audio_next_packet_size(subs); - urb->iso_frame_desc[i].offset = offs * stride; - urb->iso_frame_desc[i].length = counts * stride; - offs += counts; - } - urb->number_of_packets = ctx->packets; - urb->transfer_buffer_length = offs * stride; - memset(urb->transfer_buffer, - runtime->format == SNDRV_PCM_FORMAT_U8 ? 0x80 : 0, - offs * stride); - return 0; -} - -/* - * prepare urb for playback data pipe - * - * Since a URB can handle only a single linear buffer, we must use double - * buffering when the data to be transferred overflows the buffer boundary. - * To avoid inconsistencies when updating hwptr_done, we use double buffering - * for all URBs. - */ -static int prepare_playback_urb(struct snd_usb_substream *subs, - struct snd_pcm_runtime *runtime, - struct urb *urb) -{ - int i, stride; - unsigned int counts, frames, bytes; - unsigned long flags; - int period_elapsed = 0; - struct snd_urb_ctx *ctx = urb->context; - - stride = runtime->frame_bits >> 3; - - frames = 0; - urb->dev = ctx->subs->dev; /* we need to set this at each time */ - urb->number_of_packets = 0; - spin_lock_irqsave(&subs->lock, flags); - for (i = 0; i < ctx->packets; i++) { - counts = snd_usb_audio_next_packet_size(subs); - /* set up descriptor */ - urb->iso_frame_desc[i].offset = frames * stride; - urb->iso_frame_desc[i].length = counts * stride; - frames += counts; - urb->number_of_packets++; - subs->transfer_done += counts; - if (subs->transfer_done >= runtime->period_size) { - subs->transfer_done -= runtime->period_size; - period_elapsed = 1; - if (subs->fmt_type == UAC_FORMAT_TYPE_II) { - if (subs->transfer_done > 0) { - /* FIXME: fill-max mode is not - * supported yet */ - frames -= subs->transfer_done; - counts -= subs->transfer_done; - urb->iso_frame_desc[i].length = - counts * stride; - subs->transfer_done = 0; - } - i++; - if (i < ctx->packets) { - /* add a transfer delimiter */ - urb->iso_frame_desc[i].offset = - frames * stride; - urb->iso_frame_desc[i].length = 0; - urb->number_of_packets++; - } - break; - } - } - if (period_elapsed) /* finish at the period boundary */ - break; - } - bytes = frames * stride; - if (subs->hwptr_done + bytes > runtime->buffer_size * stride) { - /* err, the transferred area goes over buffer boundary. */ - unsigned int bytes1 = - runtime->buffer_size * stride - subs->hwptr_done; - memcpy(urb->transfer_buffer, - runtime->dma_area + subs->hwptr_done, bytes1); - memcpy(urb->transfer_buffer + bytes1, - runtime->dma_area, bytes - bytes1); - } else { - memcpy(urb->transfer_buffer, - runtime->dma_area + subs->hwptr_done, bytes); - } - subs->hwptr_done += bytes; - if (subs->hwptr_done >= runtime->buffer_size * stride) - subs->hwptr_done -= runtime->buffer_size * stride; - - /* update delay with exact number of samples queued */ - runtime->delay = subs->last_delay; - runtime->delay += frames; - subs->last_delay = runtime->delay; - - /* realign last_frame_number */ - subs->last_frame_number = usb_get_current_frame_number(subs->dev); - subs->last_frame_number &= 0xFF; /* keep 8 LSBs */ - - spin_unlock_irqrestore(&subs->lock, flags); - urb->transfer_buffer_length = bytes; - if (period_elapsed) - snd_pcm_period_elapsed(subs->pcm_substream); - return 0; -} - -/* - * process after playback data complete - * - decrease the delay count again - */ -static int retire_playback_urb(struct snd_usb_substream *subs, - struct snd_pcm_runtime *runtime, - struct urb *urb) -{ - unsigned long flags; - int stride = runtime->frame_bits >> 3; - int processed = urb->transfer_buffer_length / stride; - int est_delay; - - spin_lock_irqsave(&subs->lock, flags); - - est_delay = snd_usb_pcm_delay(subs, runtime->rate); - /* update delay with exact number of samples played */ - if (processed > subs->last_delay) - subs->last_delay = 0; - else - subs->last_delay -= processed; - runtime->delay = subs->last_delay; - - /* - * Report when delay estimate is off by more than 2ms. - * The error should be lower than 2ms since the estimate relies - * on two reads of a counter updated every ms. - */ - if (abs(est_delay - subs->last_delay) * 1000 > runtime->rate * 2) - snd_printk(KERN_DEBUG "delay: estimated %d, actual %d\n", - est_delay, subs->last_delay); - - spin_unlock_irqrestore(&subs->lock, flags); - return 0; + if (u->buffer_size) + usb_free_coherent(u->ep->chip->dev, u->buffer_size, + u->urb->transfer_buffer, + u->urb->transfer_dma); + usb_free_urb(u->urb); + u->urb = NULL; } static const char *usb_error_string(int err) @@ -802,133 +91,6 @@ static const char *usb_error_string(int err) } } -/* - * set up and start data/sync urbs - */ -static int start_urbs(struct snd_usb_substream *subs, struct snd_pcm_runtime *runtime) -{ - unsigned int i; - int err; - - if (subs->stream->chip->shutdown) - return -EBADFD; - - for (i = 0; i < subs->nurbs; i++) { - if (snd_BUG_ON(!subs->dataurb[i].urb)) - return -EINVAL; - if (subs->ops.prepare(subs, runtime, subs->dataurb[i].urb) < 0) { - snd_printk(KERN_ERR "cannot prepare datapipe for urb %d\n", i); - goto __error; - } - } - if (subs->syncpipe) { - for (i = 0; i < SYNC_URBS; i++) { - if (snd_BUG_ON(!subs->syncurb[i].urb)) - return -EINVAL; - if (subs->ops.prepare_sync(subs, runtime, subs->syncurb[i].urb) < 0) { - snd_printk(KERN_ERR "cannot prepare syncpipe for urb %d\n", i); - goto __error; - } - } - } - - subs->active_mask = 0; - subs->unlink_mask = 0; - subs->running = 1; - for (i = 0; i < subs->nurbs; i++) { - err = usb_submit_urb(subs->dataurb[i].urb, GFP_ATOMIC); - if (err < 0) { - snd_printk(KERN_ERR "cannot submit datapipe " - "for urb %d, error %d: %s\n", - i, err, usb_error_string(err)); - goto __error; - } - set_bit(i, &subs->active_mask); - } - if (subs->syncpipe) { - for (i = 0; i < SYNC_URBS; i++) { - err = usb_submit_urb(subs->syncurb[i].urb, GFP_ATOMIC); - if (err < 0) { - snd_printk(KERN_ERR "cannot submit syncpipe " - "for urb %d, error %d: %s\n", - i, err, usb_error_string(err)); - goto __error; - } - set_bit(i + 16, &subs->active_mask); - } - } - return 0; - - __error: - // snd_pcm_stop(subs->pcm_substream, SNDRV_PCM_STATE_XRUN); - deactivate_urbs_old(subs, 0, 0); - return -EPIPE; -} - - -/* - */ -static struct snd_urb_ops audio_urb_ops[2] = { - { - .prepare = prepare_nodata_playback_urb, - .retire = retire_playback_urb, - .prepare_sync = prepare_playback_sync_urb, - .retire_sync = retire_playback_sync_urb, - }, - { - .prepare = prepare_capture_urb, - .retire = retire_capture_urb, - .prepare_sync = prepare_capture_sync_urb, - .retire_sync = retire_capture_sync_urb, - }, -}; - -/* - * initialize the substream instance. - */ - -void snd_usb_init_substream(struct snd_usb_stream *as, - int stream, struct audioformat *fp) -{ - struct snd_usb_substream *subs = &as->substream[stream]; - - INIT_LIST_HEAD(&subs->fmt_list); - spin_lock_init(&subs->lock); - - subs->stream = as; - subs->direction = stream; - subs->dev = as->chip->dev; - subs->txfr_quirk = as->chip->txfr_quirk; - subs->ops = audio_urb_ops[stream]; - if (snd_usb_get_speed(subs->dev) >= USB_SPEED_HIGH) - subs->ops.prepare_sync = prepare_capture_sync_urb_hs; - - snd_usb_set_pcm_ops(as->pcm, stream); - - list_add_tail(&fp->list, &subs->fmt_list); - subs->formats |= fp->formats; - subs->endpoint = fp->endpoint; - subs->num_formats++; - subs->fmt_type = fp->fmt_type; -} - -int snd_usb_substream_prepare(struct snd_usb_substream *subs, - struct snd_pcm_runtime *runtime) -{ - /* clear urbs (to be sure) */ - deactivate_urbs_old(subs, 0, 1); - wait_clear_urbs_old(subs); - - /* for playback, submit the URBs now; otherwise, the first hwptr_done - * updates for all URBs would happen at the same time when starting */ - if (subs->direction == SNDRV_PCM_STREAM_PLAYBACK) { - subs->ops.prepare = prepare_nodata_playback_urb; - return start_urbs(subs, runtime); - } - - return 0; -} - int snd_usb_endpoint_implict_feedback_sink(struct snd_usb_endpoint *ep) { return ep->sync_master && diff --git a/sound/usb/endpoint.h b/sound/usb/endpoint.h index e540768de3d5..ee2723fb174f 100644 --- a/sound/usb/endpoint.h +++ b/sound/usb/endpoint.h @@ -1,21 +1,6 @@ #ifndef __USBAUDIO_ENDPOINT_H #define __USBAUDIO_ENDPOINT_H -void snd_usb_init_substream(struct snd_usb_stream *as, - int stream, - struct audioformat *fp); - -int snd_usb_init_substream_urbs(struct snd_usb_substream *subs, - unsigned int period_bytes, - unsigned int rate, - unsigned int frame_bits); - -void snd_usb_release_substream_urbs(struct snd_usb_substream *subs, int force); - -int snd_usb_substream_prepare(struct snd_usb_substream *subs, - struct snd_pcm_runtime *runtime); - - #define SND_USB_ENDPOINT_TYPE_DATA 0 #define SND_USB_ENDPOINT_TYPE_SYNC 1 -- cgit v1.2.3 From c75a8a7ae565d7cd9baa87a504ba9162e355b4b0 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Thu, 12 Apr 2012 13:51:14 +0200 Subject: ALSA: snd-usb: add support for implicit feedback Implicit feedback is a streaming mode that does not rely on dedicated sync endpoints but uses the information provided by record streams to clock output streams. Now that the streaming logic is decoupled from the PCM streams, this is easy to implement. Signed-off-by: Daniel Mack Signed-off-by: Takashi Iwai --- sound/usb/pcm.c | 37 ++++++++++++++++++++++++++----------- 1 file changed, 26 insertions(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 0f107834c100..2d3a04d829b7 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -301,7 +301,7 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt) struct usb_interface *iface; unsigned int ep, attr; int is_playback = subs->direction == SNDRV_PCM_STREAM_PLAYBACK; - int err; + int err, implicit_fb = 0; iface = usb_ifnum_to_if(dev, fmt->iface); if (WARN_ON(!iface)) @@ -326,25 +326,34 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt) * assume it as adaptive-out or sync-in. */ attr = fmt->ep_attr & USB_ENDPOINT_SYNCTYPE; - if (((is_playback && attr == USB_ENDPOINT_SYNC_ASYNC) || - (! is_playback && attr == USB_ENDPOINT_SYNC_ADAPTIVE)) && - altsd->bNumEndpoints >= 2) { - switch (subs->stream->chip->usb_id) { - case USB_ID(0x0763, 0x2080): /* M-Audio FastTrack Ultra */ - case USB_ID(0x0763, 0x2081): + + switch (subs->stream->chip->usb_id) { + case USB_ID(0x0763, 0x2080): /* M-Audio FastTrack Ultra */ + case USB_ID(0x0763, 0x2081): + if (is_playback) { + implicit_fb = 1; ep = 0x81; iface = usb_ifnum_to_if(dev, 2); + + if (!iface || iface->num_altsetting == 0) + return -EINVAL; + alts = &iface->altsetting[1]; goto add_sync_ep; } + } + if (((is_playback && attr == USB_ENDPOINT_SYNC_ASYNC) || + (!is_playback && attr == USB_ENDPOINT_SYNC_ADAPTIVE)) && + altsd->bNumEndpoints >= 2) { /* check sync-pipe endpoint */ /* ... and check descriptor size before accessing bSynchAddress because there is a version of the SB Audigy 2 NX firmware lacking the audio fields in the endpoint descriptors */ if ((get_endpoint(alts, 1)->bmAttributes & USB_ENDPOINT_XFERTYPE_MASK) != 0x01 || (get_endpoint(alts, 1)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE && - get_endpoint(alts, 1)->bSynchAddress != 0)) { + get_endpoint(alts, 1)->bSynchAddress != 0 && + !implicit_fb)) { snd_printk(KERN_ERR "%d:%d:%d : invalid synch pipe\n", dev->devnum, fmt->iface, fmt->altsetting); return -EINVAL; @@ -352,16 +361,22 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt) ep = get_endpoint(alts, 1)->bEndpointAddress; if (get_endpoint(alts, 0)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE && (( is_playback && ep != (unsigned int)(get_endpoint(alts, 0)->bSynchAddress | USB_DIR_IN)) || - (!is_playback && ep != (unsigned int)(get_endpoint(alts, 0)->bSynchAddress & ~USB_DIR_IN)))) { + (!is_playback && ep != (unsigned int)(get_endpoint(alts, 0)->bSynchAddress & ~USB_DIR_IN)) || + ( is_playback && !implicit_fb))) { snd_printk(KERN_ERR "%d:%d:%d : invalid synch pipe\n", dev->devnum, fmt->iface, fmt->altsetting); return -EINVAL; } + + implicit_fb = (get_endpoint(alts, 1)->bmAttributes & USB_ENDPOINT_USAGE_MASK) + == USB_ENDPOINT_USAGE_IMPLICIT_FB; + add_sync_ep: subs->sync_endpoint = snd_usb_add_endpoint(subs->stream->chip, alts, ep, !subs->direction, - SND_USB_ENDPOINT_TYPE_SYNC); - + implicit_fb ? + SND_USB_ENDPOINT_TYPE_DATA : + SND_USB_ENDPOINT_TYPE_SYNC); if (!subs->sync_endpoint) return -EINVAL; -- cgit v1.2.3 From 94c27215bc5b7941a52d3d3a02d904bfdcd5fe5e Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Thu, 12 Apr 2012 13:51:15 +0200 Subject: ALSA: snd-usb: add some documentation Document the new streaming code and some of the functions so that contributers can catch up easier. Signed-off-by: Daniel Mack Signed-off-by: Takashi Iwai --- sound/usb/endpoint.c | 182 +++++++++++++++++++++++++++++++++++++++++++++++---- 1 file changed, 171 insertions(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 64853f76c10c..d1ef3048dbba 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -35,6 +35,32 @@ #define EP_FLAG_ACTIVATED 0 #define EP_FLAG_RUNNING 1 +/* + * snd_usb_endpoint is a model that abstracts everything related to an + * USB endpoint and its streaming. + * + * There are functions to activate and deactivate the streaming URBs and + * optinal callbacks to let the pcm logic handle the actual content of the + * packets for playback and record. Thus, the bus streaming and the audio + * handlers are fully decoupled. + * + * There are two different types of endpoints in for audio applications. + * + * SND_USB_ENDPOINT_TYPE_DATA handles full audio data payload for both + * inbound and outbound traffic. + * + * SND_USB_ENDPOINT_TYPE_SYNC are for inbound traffic only and expect the + * payload to carry Q16.16 formatted sync information (3 or 4 bytes). + * + * Each endpoint has to be configured (by calling + * snd_usb_endpoint_set_params()) before it can be used. + * + * The model incorporates a reference counting, so that multiple users + * can call snd_usb_endpoint_start() and snd_usb_endpoint_stop(), and + * only the first user will effectively start the URBs, and only the last + * one will tear them down again. + */ + /* * convert a sampling rate into our full speed format (fs/1000 in Q16.16) * this will overflow at approx 524 kHz @@ -91,6 +117,14 @@ static const char *usb_error_string(int err) } } +/** + * snd_usb_endpoint_implicit_feedback_sink: Report endpoint usage type + * + * @ep: The endpoint + * + * Determine whether an endpoint is driven by an implicit feedback + * data endpoint source. + */ int snd_usb_endpoint_implict_feedback_sink(struct snd_usb_endpoint *ep) { return ep->sync_master && @@ -99,7 +133,13 @@ int snd_usb_endpoint_implict_feedback_sink(struct snd_usb_endpoint *ep) usb_pipeout(ep->pipe); } -/* determine the number of frames in the next packet */ +/* + * For streaming based on information derived from sync endpoints, + * prepare_outbound_urb_sizes() will call next_packet_size() to + * determine the number of samples to be sent in the next packet. + * + * For implicit feedback, next_packet_size() is unused. + */ static int next_packet_size(struct snd_usb_endpoint *ep) { unsigned long flags; @@ -237,6 +277,19 @@ static inline void prepare_inbound_urb(struct snd_usb_endpoint *ep, } } +/* + * Send output urbs that have been prepared previously. Urbs are dequeued + * from ep->ready_playback_urbs and in case there there aren't any available + * or there are no packets that have been prepared, this function does + * nothing. + * + * The reason why the functionality of sending and preparing urbs is separated + * is that host controllers don't guarantee an ordering in returing inbound + * and outbound packets to their submitters. + * + * This function is only used for implicit feedback endpoints. For endpoints + * driven by sync endpoints, urbs are submitted from their completion handler. + */ static void queue_pending_output_urbs(struct snd_usb_endpoint *ep) { while (test_bit(EP_FLAG_RUNNING, &ep->flags)) { @@ -270,6 +323,7 @@ static void queue_pending_output_urbs(struct snd_usb_endpoint *ep) for (i = 0; i < packet->packets; i++) ctx->packet_size[i] = packet->packet_size[i]; + /* call the data handler to fill in playback data */ prepare_outbound_urb(ep, ctx); err = usb_submit_urb(ctx->urb, GFP_ATOMIC); @@ -336,6 +390,22 @@ exit_clear: clear_bit(ctx->index, &ep->active_mask); } +/** + * snd_usb_add_endpoint: Add an endpoint to an audio chip + * + * @chip: The chip + * @alts: The USB host interface + * @ep_num: The number of the endpoint to use + * @direction: SNDRV_PCM_STREAM_PLAYBACK or SNDRV_PCM_STREAM_CAPTURE + * @type: SND_USB_ENDPOINT_TYPE_DATA or SND_USB_ENDPOINT_TYPE_SYNC + * + * If the requested endpoint has not been added to the given chip before, + * a new instance is created. Otherwise, a pointer to the previoulsy + * created instance is returned. In case of any error, NULL is returned. + * + * New endpoints will be added to chip->ep_list and must be freed by + * calling snd_usb_endpoint_free(). + */ struct snd_usb_endpoint *snd_usb_add_endpoint(struct snd_usb_audio *chip, struct usb_host_interface *alts, int ep_num, int direction, int type) @@ -506,6 +576,9 @@ static void release_urbs(struct snd_usb_endpoint *ep, int force) ep->nurbs = 0; } +/* + * configure a data endpoint + */ static int data_ep_set_params(struct snd_usb_endpoint *ep, struct snd_pcm_hw_params *hw_params, struct audioformat *fmt, @@ -629,6 +702,9 @@ out_of_memory: return -ENOMEM; } +/* + * configure a sync endpoint + */ static int sync_ep_set_params(struct snd_usb_endpoint *ep, struct snd_pcm_hw_params *hw_params, struct audioformat *fmt) @@ -669,6 +745,15 @@ out_of_memory: return -ENOMEM; } +/** + * snd_usb_endpoint_set_params: configure an snd_endpoint + * + * @ep: the endpoint to configure + * + * Determine the number of of URBs to be used on this endpoint. + * An endpoint must be configured before it can be started. + * An endpoint that is already running can not be reconfigured. + */ int snd_usb_endpoint_set_params(struct snd_usb_endpoint *ep, struct snd_pcm_hw_params *hw_params, struct audioformat *fmt, @@ -717,6 +802,19 @@ int snd_usb_endpoint_set_params(struct snd_usb_endpoint *ep, return err; } +/** + * snd_usb_endpoint_start: start an snd_usb_endpoint + * + * @ep: the endpoint to start + * + * A call to this function will increment the use count of the endpoint. + * In case this not already running, the URBs for this endpoint will be + * submitted. Otherwise, this function does nothing. + * + * Must be balanced to calls of snd_usb_endpoint_stop(). + * + * Returns an error if the URB submission failed, 0 in all other cases. + */ int snd_usb_endpoint_start(struct snd_usb_endpoint *ep) { int err; @@ -788,6 +886,17 @@ __error: return -EPIPE; } +/** + * snd_usb_endpoint_stop: stop an snd_usb_endpoint + * + * @ep: the endpoint to stop (may be NULL) + * + * A call to this function will decrement the use count of the endpoint. + * In case the last user has requested the endpoint stop, the URBs will + * actually deactivated. + * + * Must be balanced to calls of snd_usb_endpoint_start(). + */ void snd_usb_endpoint_stop(struct snd_usb_endpoint *ep, int force, int can_sleep, int wait) { @@ -812,6 +921,19 @@ void snd_usb_endpoint_stop(struct snd_usb_endpoint *ep, } } +/** + * snd_usb_endpoint_activate: activate an snd_usb_endpoint + * + * @ep: the endpoint to activate + * + * If the endpoint is not currently in use, this functions will select the + * correct alternate interface setting for the interface of this endpoint. + * + * In case of any active users, this functions does nothing. + * + * Returns an error if usb_set_interface() failed, 0 in all other + * cases. + */ int snd_usb_endpoint_activate(struct snd_usb_endpoint *ep) { if (ep->use_count != 0) @@ -835,6 +957,19 @@ int snd_usb_endpoint_activate(struct snd_usb_endpoint *ep) return -EBUSY; } +/** + * snd_usb_endpoint_deactivate: deactivate an snd_usb_endpoint + * + * @ep: the endpoint to deactivate + * + * If the endpoint is not currently in use, this functions will select the + * alternate interface setting 0 for the interface of this endpoint. + * + * In case of any active users, this functions does nothing. + * + * Returns an error if usb_set_interface() failed, 0 in all other + * cases. + */ int snd_usb_endpoint_deactivate(struct snd_usb_endpoint *ep) { if (!ep) @@ -860,6 +995,13 @@ int snd_usb_endpoint_deactivate(struct snd_usb_endpoint *ep) return -EBUSY; } +/** snd_usb_endpoint_free: Free the resources of an snd_usb_endpoint + * + * @ep: the list header of the endpoint to free + * + * This function does not care for the endpoint's use count but will tear + * down all the streaming URBs immediately and free all resources. + */ void snd_usb_endpoint_free(struct list_head *head) { struct snd_usb_endpoint *ep; @@ -869,15 +1011,15 @@ void snd_usb_endpoint_free(struct list_head *head) kfree(ep); } -/* - * process after playback sync complete - * - * Full speed devices report feedback values in 10.14 format as samples per - * frame, high speed devices in 16.16 format as samples per microframe. - * Because the Audio Class 1 spec was written before USB 2.0, many high speed - * devices use a wrong interpretation, some others use an entirely different - * format. Therefore, we cannot predict what format any particular device uses - * and must detect it automatically. +/** + * snd_usb_handle_sync_urb: parse an USB sync packet + * + * @ep: the endpoint to handle the packet + * @sender: the sending endpoint + * @urb: the received packet + * + * This function is called from the context of an endpoint that received + * the packet and is used to let another endpoint object handle the payload. */ void snd_usb_handle_sync_urb(struct snd_usb_endpoint *ep, struct snd_usb_endpoint *sender, @@ -889,6 +1031,11 @@ void snd_usb_handle_sync_urb(struct snd_usb_endpoint *ep, snd_BUG_ON(ep == sender); + /* + * In case the endpoint is operating in implicit feedback mode, prepare + * and a new outbound URB that has the same layout as the received + * packet and add it to the list of pending urbs. + */ if (snd_usb_endpoint_implict_feedback_sink(ep) && ep->use_count != 0) { @@ -938,7 +1085,20 @@ void snd_usb_handle_sync_urb(struct snd_usb_endpoint *ep, return; } - /* parse sync endpoint packet */ + /* + * process after playback sync complete + * + * Full speed devices report feedback values in 10.14 format as samples + * per frame, high speed devices in 16.16 format as samples per + * microframe. + * + * Because the Audio Class 1 spec was written before USB 2.0, many high + * speed devices use a wrong interpretation, some others use an + * entirely different format. + * + * Therefore, we cannot predict what format any particular device uses + * and must detect it automatically. + */ if (urb->iso_frame_desc[0].status != 0 || urb->iso_frame_desc[0].actual_length < 3) -- cgit v1.2.3 From c5ee4ec82801d11c67836565539acd3b47100208 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 13 Apr 2012 10:27:28 +0200 Subject: ALSA: usb: Remove unused variable MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit sound/usb/endpoint.c: In function ‘deactivate_urbs’: sound/usb/endpoint.c:520:16: warning: unused variable ‘flags’ [-Wunused-variable] Signed-off-by: Takashi Iwai --- sound/usb/endpoint.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index d1ef3048dbba..8b695d5562e6 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -517,7 +517,6 @@ static int wait_clear_urbs(struct snd_usb_endpoint *ep) */ static int deactivate_urbs(struct snd_usb_endpoint *ep, int force, int can_sleep) { - unsigned long flags; unsigned int i; int async; -- cgit v1.2.3 From be944d42ccc125f1b200e7a4185af5bb87865190 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Tue, 10 Apr 2012 16:31:59 -0600 Subject: ASoC: tegra: add tegra30-ahub driver The AHUB (Audio Hub) is a mux/crossbar which links all audio-related devices except the HDA controller on Tegra30. The devices include the DMA FIFOs, DAM (Digital Audio Mixers), I2S controllers, and SPDIF controller. Audio data may be routed between these devices in various combinations as required by board design/application. Includes a squashed bugfix from Nikesh Oswal Includes squashed bugfixes from Sumit Bhattacharya Signed-off-by: Stephen Warren Signed-off-by: Mark Brown --- .../bindings/sound/nvidia,tegra30-ahub.txt | 32 ++ sound/soc/tegra/tegra30_ahub.c | 631 +++++++++++++++++++++ sound/soc/tegra/tegra30_ahub.h | 483 ++++++++++++++++ 3 files changed, 1146 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra30-ahub.txt create mode 100644 sound/soc/tegra/tegra30_ahub.c create mode 100644 sound/soc/tegra/tegra30_ahub.h (limited to 'sound') diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra30-ahub.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra30-ahub.txt new file mode 100644 index 000000000000..1ac7b1642186 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra30-ahub.txt @@ -0,0 +1,32 @@ +NVIDIA Tegra30 AHUB (Audio Hub) + +Required properties: +- compatible : "nvidia,tegra30-ahub" +- reg : Should contain the register physical address and length for each of + the AHUB's APBIF registers and the AHUB's own registers. +- interrupts : Should contain AHUB interrupt +- nvidia,dma-request-selector : The Tegra DMA controller's phandle and + request selector for the first APBIF channel. +- ranges : The bus address mapping for the configlink register bus. + Can be empty since the mapping is 1:1. +- #address-cells : For the configlink bus. Should be <1>; +- #size-cells : For the configlink bus. Should be <1>. + +AHUB client modules need to specify the IDs of their CIFs (Client InterFaces). +For RX CIFs, the numbers indicate the register number within AHUB routing +register space (APBIF 0..3 RX, I2S 0..5 RX, DAM 0..2 RX 0..1, SPDIF RX 0..1). +For TX CIFs, the numbers indicate the bit position within the AHUB routing +registers (APBIF 0..3 TX, I2S 0..5 TX, DAM 0..2 TX, SPDIF TX 0..1). + +Example: + +ahub@70080000 { + compatible = "nvidia,tegra30-ahub"; + reg = <0x70080000 0x200 0x70080200 0x100>; + interrupts = < 0 103 0x04 >; + nvidia,dma-request-selector = <&apbdma 1>; + + ranges; + #address-cells = <1>; + #size-cells = <1>; +}; diff --git a/sound/soc/tegra/tegra30_ahub.c b/sound/soc/tegra/tegra30_ahub.c new file mode 100644 index 000000000000..57cd419f743e --- /dev/null +++ b/sound/soc/tegra/tegra30_ahub.c @@ -0,0 +1,631 @@ +/* + * tegra30_ahub.c - Tegra30 AHUB driver + * + * Copyright (c) 2011,2012, NVIDIA CORPORATION. All rights reserved. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms and conditions of the GNU General Public License, + * version 2, as published by the Free Software Foundation. + * + * This program is distributed in the hope it will be useful, but WITHOUT + * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or + * FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for + * more details. + * + * You should have received a copy of the GNU General Public License + * along with this program. If not, see . + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include "tegra30_ahub.h" + +#define DRV_NAME "tegra30-ahub" + +static struct tegra30_ahub *ahub; + +static inline void tegra30_apbif_write(u32 reg, u32 val) +{ + regmap_write(ahub->regmap_apbif, reg, val); +} + +static inline u32 tegra30_apbif_read(u32 reg) +{ + u32 val; + regmap_read(ahub->regmap_apbif, reg, &val); + return val; +} + +static inline void tegra30_audio_write(u32 reg, u32 val) +{ + regmap_write(ahub->regmap_ahub, reg, val); +} + +static int tegra30_ahub_runtime_suspend(struct device *dev) +{ + regcache_cache_only(ahub->regmap_apbif, true); + regcache_cache_only(ahub->regmap_ahub, true); + + clk_disable(ahub->clk_apbif); + clk_disable(ahub->clk_d_audio); + + return 0; +} + +/* + * clk_apbif isn't required for an I2S<->I2S configuration where no PCM data + * is read from or sent to memory. However, that's not something the rest of + * the driver supports right now, so we'll just treat the two clocks as one + * for now. + * + * These functions should not be a plain ref-count. Instead, each active stream + * contributes some requirement to the minimum clock rate, so starting or + * stopping streams should dynamically adjust the clock as required. However, + * this is not yet implemented. + */ +static int tegra30_ahub_runtime_resume(struct device *dev) +{ + int ret; + + ret = clk_enable(ahub->clk_d_audio); + if (ret) { + dev_err(dev, "clk_enable d_audio failed: %d\n", ret); + return ret; + } + ret = clk_enable(ahub->clk_apbif); + if (ret) { + dev_err(dev, "clk_enable apbif failed: %d\n", ret); + clk_disable(ahub->clk_d_audio); + return ret; + } + + regcache_cache_only(ahub->regmap_apbif, false); + regcache_cache_only(ahub->regmap_ahub, false); + + return 0; +} + +int tegra30_ahub_allocate_rx_fifo(enum tegra30_ahub_rxcif *rxcif, + unsigned long *fiforeg, + unsigned long *reqsel) +{ + int channel; + u32 reg, val; + + channel = find_first_zero_bit(ahub->rx_usage, + TEGRA30_AHUB_CHANNEL_CTRL_COUNT); + if (channel >= TEGRA30_AHUB_CHANNEL_CTRL_COUNT) + return -EBUSY; + + __set_bit(channel, ahub->rx_usage); + + *rxcif = TEGRA30_AHUB_RXCIF_APBIF_RX0 + channel; + *fiforeg = ahub->apbif_addr + TEGRA30_AHUB_CHANNEL_RXFIFO + + (channel * TEGRA30_AHUB_CHANNEL_RXFIFO_STRIDE); + *reqsel = ahub->dma_sel + channel; + + reg = TEGRA30_AHUB_CHANNEL_CTRL + + (channel * TEGRA30_AHUB_CHANNEL_CTRL_STRIDE); + val = tegra30_apbif_read(reg); + val &= ~(TEGRA30_AHUB_CHANNEL_CTRL_RX_THRESHOLD_MASK | + TEGRA30_AHUB_CHANNEL_CTRL_RX_PACK_MASK); + val |= (7 << TEGRA30_AHUB_CHANNEL_CTRL_RX_THRESHOLD_SHIFT) | + TEGRA30_AHUB_CHANNEL_CTRL_RX_PACK_EN | + TEGRA30_AHUB_CHANNEL_CTRL_RX_PACK_16; + tegra30_apbif_write(reg, val); + + reg = TEGRA30_AHUB_CIF_RX_CTRL + + (channel * TEGRA30_AHUB_CIF_RX_CTRL_STRIDE); + val = (0 << TEGRA30_AUDIOCIF_CTRL_FIFO_THRESHOLD_SHIFT) | + (1 << TEGRA30_AUDIOCIF_CTRL_AUDIO_CHANNELS_SHIFT) | + (1 << TEGRA30_AUDIOCIF_CTRL_CLIENT_CHANNELS_SHIFT) | + TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_16 | + TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_16 | + TEGRA30_AUDIOCIF_CTRL_DIRECTION_RX; + tegra30_apbif_write(reg, val); + + return 0; +} +EXPORT_SYMBOL_GPL(tegra30_ahub_allocate_rx_fifo); + +int tegra30_ahub_enable_rx_fifo(enum tegra30_ahub_rxcif rxcif) +{ + int channel = rxcif - TEGRA30_AHUB_RXCIF_APBIF_RX0; + int reg, val; + + reg = TEGRA30_AHUB_CHANNEL_CTRL + + (channel * TEGRA30_AHUB_CHANNEL_CTRL_STRIDE); + val = tegra30_apbif_read(reg); + val |= TEGRA30_AHUB_CHANNEL_CTRL_RX_EN; + tegra30_apbif_write(reg, val); + + return 0; +} +EXPORT_SYMBOL_GPL(tegra30_ahub_enable_rx_fifo); + +int tegra30_ahub_disable_rx_fifo(enum tegra30_ahub_rxcif rxcif) +{ + int channel = rxcif - TEGRA30_AHUB_RXCIF_APBIF_RX0; + int reg, val; + + reg = TEGRA30_AHUB_CHANNEL_CTRL + + (channel * TEGRA30_AHUB_CHANNEL_CTRL_STRIDE); + val = tegra30_apbif_read(reg); + val &= ~TEGRA30_AHUB_CHANNEL_CTRL_RX_EN; + tegra30_apbif_write(reg, val); + + return 0; +} +EXPORT_SYMBOL_GPL(tegra30_ahub_disable_rx_fifo); + +int tegra30_ahub_free_rx_fifo(enum tegra30_ahub_rxcif rxcif) +{ + int channel = rxcif - TEGRA30_AHUB_RXCIF_APBIF_RX0; + + __clear_bit(channel, ahub->rx_usage); + + return 0; +} +EXPORT_SYMBOL_GPL(tegra30_ahub_free_rx_fifo); + +int tegra30_ahub_allocate_tx_fifo(enum tegra30_ahub_txcif *txcif, + unsigned long *fiforeg, + unsigned long *reqsel) +{ + int channel; + u32 reg, val; + + channel = find_first_zero_bit(ahub->tx_usage, + TEGRA30_AHUB_CHANNEL_CTRL_COUNT); + if (channel >= TEGRA30_AHUB_CHANNEL_CTRL_COUNT) + return -EBUSY; + + __set_bit(channel, ahub->tx_usage); + + *txcif = TEGRA30_AHUB_TXCIF_APBIF_TX0 + channel; + *fiforeg = ahub->apbif_addr + TEGRA30_AHUB_CHANNEL_TXFIFO + + (channel * TEGRA30_AHUB_CHANNEL_TXFIFO_STRIDE); + *reqsel = ahub->dma_sel + channel; + + reg = TEGRA30_AHUB_CHANNEL_CTRL + + (channel * TEGRA30_AHUB_CHANNEL_CTRL_STRIDE); + val = tegra30_apbif_read(reg); + val &= ~(TEGRA30_AHUB_CHANNEL_CTRL_TX_THRESHOLD_MASK | + TEGRA30_AHUB_CHANNEL_CTRL_TX_PACK_MASK); + val |= (7 << TEGRA30_AHUB_CHANNEL_CTRL_TX_THRESHOLD_SHIFT) | + TEGRA30_AHUB_CHANNEL_CTRL_TX_PACK_EN | + TEGRA30_AHUB_CHANNEL_CTRL_TX_PACK_16; + tegra30_apbif_write(reg, val); + + reg = TEGRA30_AHUB_CIF_TX_CTRL + + (channel * TEGRA30_AHUB_CIF_TX_CTRL_STRIDE); + val = (0 << TEGRA30_AUDIOCIF_CTRL_FIFO_THRESHOLD_SHIFT) | + (1 << TEGRA30_AUDIOCIF_CTRL_AUDIO_CHANNELS_SHIFT) | + (1 << TEGRA30_AUDIOCIF_CTRL_CLIENT_CHANNELS_SHIFT) | + TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_16 | + TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_16 | + TEGRA30_AUDIOCIF_CTRL_DIRECTION_TX; + tegra30_apbif_write(reg, val); + + return 0; +} +EXPORT_SYMBOL_GPL(tegra30_ahub_allocate_tx_fifo); + +int tegra30_ahub_enable_tx_fifo(enum tegra30_ahub_txcif txcif) +{ + int channel = txcif - TEGRA30_AHUB_TXCIF_APBIF_TX0; + int reg, val; + + reg = TEGRA30_AHUB_CHANNEL_CTRL + + (channel * TEGRA30_AHUB_CHANNEL_CTRL_STRIDE); + val = tegra30_apbif_read(reg); + val |= TEGRA30_AHUB_CHANNEL_CTRL_TX_EN; + tegra30_apbif_write(reg, val); + + return 0; +} +EXPORT_SYMBOL_GPL(tegra30_ahub_enable_tx_fifo); + +int tegra30_ahub_disable_tx_fifo(enum tegra30_ahub_txcif txcif) +{ + int channel = txcif - TEGRA30_AHUB_TXCIF_APBIF_TX0; + int reg, val; + + reg = TEGRA30_AHUB_CHANNEL_CTRL + + (channel * TEGRA30_AHUB_CHANNEL_CTRL_STRIDE); + val = tegra30_apbif_read(reg); + val &= ~TEGRA30_AHUB_CHANNEL_CTRL_TX_EN; + tegra30_apbif_write(reg, val); + + return 0; +} +EXPORT_SYMBOL_GPL(tegra30_ahub_disable_tx_fifo); + +int tegra30_ahub_free_tx_fifo(enum tegra30_ahub_txcif txcif) +{ + int channel = txcif - TEGRA30_AHUB_TXCIF_APBIF_TX0; + + __clear_bit(channel, ahub->tx_usage); + + return 0; +} +EXPORT_SYMBOL_GPL(tegra30_ahub_free_tx_fifo); + +int tegra30_ahub_set_rx_cif_source(enum tegra30_ahub_rxcif rxcif, + enum tegra30_ahub_txcif txcif) +{ + int channel = rxcif - TEGRA30_AHUB_RXCIF_APBIF_RX0; + int reg; + + reg = TEGRA30_AHUB_AUDIO_RX + + (channel * TEGRA30_AHUB_AUDIO_RX_STRIDE); + tegra30_audio_write(reg, 1 << txcif); + + return 0; +} +EXPORT_SYMBOL_GPL(tegra30_ahub_set_rx_cif_source); + +int tegra30_ahub_unset_rx_cif_source(enum tegra30_ahub_rxcif rxcif) +{ + int channel = rxcif - TEGRA30_AHUB_RXCIF_APBIF_RX0; + int reg; + + reg = TEGRA30_AHUB_AUDIO_RX + + (channel * TEGRA30_AHUB_AUDIO_RX_STRIDE); + tegra30_audio_write(reg, 0); + + return 0; +} +EXPORT_SYMBOL_GPL(tegra30_ahub_unset_rx_cif_source); + +static const char * const configlink_clocks[] __devinitconst = { + "i2s0", + "i2s1", + "i2s2", + "i2s3", + "i2s4", + "dam0", + "dam1", + "dam2", + "spdif_in", +}; + +struct of_dev_auxdata ahub_auxdata[] __devinitdata = { + OF_DEV_AUXDATA("nvidia,tegra30-i2s", 0x70080300, "tegra30-i2s.0", NULL), + OF_DEV_AUXDATA("nvidia,tegra30-i2s", 0x70080400, "tegra30-i2s.1", NULL), + OF_DEV_AUXDATA("nvidia,tegra30-i2s", 0x70080500, "tegra30-i2s.2", NULL), + OF_DEV_AUXDATA("nvidia,tegra30-i2s", 0x70080600, "tegra30-i2s.3", NULL), + OF_DEV_AUXDATA("nvidia,tegra30-i2s", 0x70080700, "tegra30-i2s.4", NULL), + {} +}; + +#define LAST_REG(name) \ + (TEGRA30_AHUB_##name + \ + (TEGRA30_AHUB_##name##_STRIDE * TEGRA30_AHUB_##name##_COUNT) - 4) + +#define REG_IN_ARRAY(reg, name) \ + ((reg >= TEGRA30_AHUB_##name) && \ + (reg <= LAST_REG(name) && \ + (!((reg - TEGRA30_AHUB_##name) % TEGRA30_AHUB_##name##_STRIDE)))) + +static bool tegra30_ahub_apbif_wr_rd_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case TEGRA30_AHUB_CONFIG_LINK_CTRL: + case TEGRA30_AHUB_MISC_CTRL: + case TEGRA30_AHUB_APBDMA_LIVE_STATUS: + case TEGRA30_AHUB_I2S_LIVE_STATUS: + case TEGRA30_AHUB_SPDIF_LIVE_STATUS: + case TEGRA30_AHUB_I2S_INT_MASK: + case TEGRA30_AHUB_DAM_INT_MASK: + case TEGRA30_AHUB_SPDIF_INT_MASK: + case TEGRA30_AHUB_APBIF_INT_MASK: + case TEGRA30_AHUB_I2S_INT_STATUS: + case TEGRA30_AHUB_DAM_INT_STATUS: + case TEGRA30_AHUB_SPDIF_INT_STATUS: + case TEGRA30_AHUB_APBIF_INT_STATUS: + case TEGRA30_AHUB_I2S_INT_SOURCE: + case TEGRA30_AHUB_DAM_INT_SOURCE: + case TEGRA30_AHUB_SPDIF_INT_SOURCE: + case TEGRA30_AHUB_APBIF_INT_SOURCE: + case TEGRA30_AHUB_I2S_INT_SET: + case TEGRA30_AHUB_DAM_INT_SET: + case TEGRA30_AHUB_SPDIF_INT_SET: + case TEGRA30_AHUB_APBIF_INT_SET: + return true; + default: + break; + }; + + if (REG_IN_ARRAY(reg, CHANNEL_CTRL) || + REG_IN_ARRAY(reg, CHANNEL_CLEAR) || + REG_IN_ARRAY(reg, CHANNEL_STATUS) || + REG_IN_ARRAY(reg, CHANNEL_TXFIFO) || + REG_IN_ARRAY(reg, CHANNEL_RXFIFO) || + REG_IN_ARRAY(reg, CIF_TX_CTRL) || + REG_IN_ARRAY(reg, CIF_RX_CTRL) || + REG_IN_ARRAY(reg, DAM_LIVE_STATUS)) + return true; + + return false; +} + +static bool tegra30_ahub_apbif_volatile_reg(struct device *dev, + unsigned int reg) +{ + switch (reg) { + case TEGRA30_AHUB_CONFIG_LINK_CTRL: + case TEGRA30_AHUB_MISC_CTRL: + case TEGRA30_AHUB_APBDMA_LIVE_STATUS: + case TEGRA30_AHUB_I2S_LIVE_STATUS: + case TEGRA30_AHUB_SPDIF_LIVE_STATUS: + case TEGRA30_AHUB_I2S_INT_STATUS: + case TEGRA30_AHUB_DAM_INT_STATUS: + case TEGRA30_AHUB_SPDIF_INT_STATUS: + case TEGRA30_AHUB_APBIF_INT_STATUS: + case TEGRA30_AHUB_I2S_INT_SET: + case TEGRA30_AHUB_DAM_INT_SET: + case TEGRA30_AHUB_SPDIF_INT_SET: + case TEGRA30_AHUB_APBIF_INT_SET: + return true; + default: + break; + }; + + if (REG_IN_ARRAY(reg, CHANNEL_CLEAR) || + REG_IN_ARRAY(reg, CHANNEL_STATUS) || + REG_IN_ARRAY(reg, CHANNEL_TXFIFO) || + REG_IN_ARRAY(reg, CHANNEL_RXFIFO) || + REG_IN_ARRAY(reg, DAM_LIVE_STATUS)) + return true; + + return false; +} + +static bool tegra30_ahub_apbif_precious_reg(struct device *dev, + unsigned int reg) +{ + if (REG_IN_ARRAY(reg, CHANNEL_TXFIFO) || + REG_IN_ARRAY(reg, CHANNEL_RXFIFO)) + return true; + + return false; +} + +static const struct regmap_config tegra30_ahub_apbif_regmap_config = { + .name = "apbif", + .reg_bits = 32, + .val_bits = 32, + .reg_stride = 4, + .max_register = TEGRA30_AHUB_APBIF_INT_SET, + .writeable_reg = tegra30_ahub_apbif_wr_rd_reg, + .readable_reg = tegra30_ahub_apbif_wr_rd_reg, + .volatile_reg = tegra30_ahub_apbif_volatile_reg, + .precious_reg = tegra30_ahub_apbif_precious_reg, + .cache_type = REGCACHE_RBTREE, +}; + +static bool tegra30_ahub_ahub_wr_rd_reg(struct device *dev, unsigned int reg) +{ + if (REG_IN_ARRAY(reg, AUDIO_RX)) + return true; + + return false; +} + +static const struct regmap_config tegra30_ahub_ahub_regmap_config = { + .name = "ahub", + .reg_bits = 32, + .val_bits = 32, + .reg_stride = 4, + .max_register = LAST_REG(AUDIO_RX), + .writeable_reg = tegra30_ahub_ahub_wr_rd_reg, + .readable_reg = tegra30_ahub_ahub_wr_rd_reg, + .cache_type = REGCACHE_RBTREE, +}; + +static int __devinit tegra30_ahub_probe(struct platform_device *pdev) +{ + struct clk *clk; + int i; + struct resource *res0, *res1, *region; + u32 of_dma[2]; + void __iomem *regs_apbif, *regs_ahub; + int ret = 0; + + if (ahub) + return -ENODEV; + + /* + * The AHUB hosts a register bus: the "configlink". For this to + * operate correctly, all devices on this bus must be out of reset. + * Ensure that here. + */ + for (i = 0; i < ARRAY_SIZE(configlink_clocks); i++) { + clk = clk_get_sys(NULL, configlink_clocks[i]); + if (IS_ERR(clk)) { + dev_err(&pdev->dev, "Can't get clock %s\n", + configlink_clocks[i]); + ret = PTR_ERR(clk); + goto err; + } + tegra_periph_reset_deassert(clk); + clk_put(clk); + } + + ahub = devm_kzalloc(&pdev->dev, sizeof(struct tegra30_ahub), + GFP_KERNEL); + if (!ahub) { + dev_err(&pdev->dev, "Can't allocate tegra30_ahub\n"); + ret = -ENOMEM; + goto err; + } + dev_set_drvdata(&pdev->dev, ahub); + + ahub->dev = &pdev->dev; + + ahub->clk_d_audio = clk_get(&pdev->dev, "d_audio"); + if (IS_ERR(ahub->clk_d_audio)) { + dev_err(&pdev->dev, "Can't retrieve ahub d_audio clock\n"); + ret = PTR_ERR(ahub->clk_d_audio); + goto err; + } + + ahub->clk_apbif = clk_get(&pdev->dev, "apbif"); + if (IS_ERR(ahub->clk_apbif)) { + dev_err(&pdev->dev, "Can't retrieve ahub apbif clock\n"); + ret = PTR_ERR(ahub->clk_apbif); + goto err_clk_put_d_audio; + } + + if (of_property_read_u32_array(pdev->dev.of_node, + "nvidia,dma-request-selector", + of_dma, 2) < 0) { + dev_err(&pdev->dev, + "Missing property nvidia,dma-request-selector\n"); + ret = -ENODEV; + goto err_clk_put_d_audio; + } + ahub->dma_sel = of_dma[1]; + + res0 = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!res0) { + dev_err(&pdev->dev, "No apbif memory resource\n"); + ret = -ENODEV; + goto err_clk_put_apbif; + } + + region = devm_request_mem_region(&pdev->dev, res0->start, + resource_size(res0), DRV_NAME); + if (!region) { + dev_err(&pdev->dev, "request region apbif failed\n"); + ret = -EBUSY; + goto err_clk_put_apbif; + } + ahub->apbif_addr = res0->start; + + regs_apbif = devm_ioremap(&pdev->dev, res0->start, + resource_size(res0)); + if (!regs_apbif) { + dev_err(&pdev->dev, "ioremap apbif failed\n"); + ret = -ENOMEM; + goto err_clk_put_apbif; + } + + ahub->regmap_apbif = devm_regmap_init_mmio(&pdev->dev, regs_apbif, + &tegra30_ahub_apbif_regmap_config); + if (IS_ERR(ahub->regmap_apbif)) { + dev_err(&pdev->dev, "apbif regmap init failed\n"); + ret = PTR_ERR(ahub->regmap_apbif); + goto err_clk_put_apbif; + } + regcache_cache_only(ahub->regmap_apbif, true); + + res1 = platform_get_resource(pdev, IORESOURCE_MEM, 1); + if (!res1) { + dev_err(&pdev->dev, "No ahub memory resource\n"); + ret = -ENODEV; + goto err_clk_put_apbif; + } + + region = devm_request_mem_region(&pdev->dev, res1->start, + resource_size(res1), DRV_NAME); + if (!region) { + dev_err(&pdev->dev, "request region ahub failed\n"); + ret = -EBUSY; + goto err_clk_put_apbif; + } + + regs_ahub = devm_ioremap(&pdev->dev, res1->start, + resource_size(res1)); + if (!regs_ahub) { + dev_err(&pdev->dev, "ioremap ahub failed\n"); + ret = -ENOMEM; + goto err_clk_put_apbif; + } + + ahub->regmap_ahub = devm_regmap_init_mmio(&pdev->dev, regs_ahub, + &tegra30_ahub_ahub_regmap_config); + if (IS_ERR(ahub->regmap_ahub)) { + dev_err(&pdev->dev, "ahub regmap init failed\n"); + ret = PTR_ERR(ahub->regmap_ahub); + goto err_clk_put_apbif; + } + regcache_cache_only(ahub->regmap_ahub, true); + + pm_runtime_enable(&pdev->dev); + if (!pm_runtime_enabled(&pdev->dev)) { + ret = tegra30_ahub_runtime_resume(&pdev->dev); + if (ret) + goto err_pm_disable; + } + + of_platform_populate(pdev->dev.of_node, NULL, ahub_auxdata, + &pdev->dev); + + return 0; + +err_pm_disable: + pm_runtime_disable(&pdev->dev); +err_clk_put_apbif: + clk_put(ahub->clk_apbif); +err_clk_put_d_audio: + clk_put(ahub->clk_d_audio); + ahub = 0; +err: + return ret; +} + +static int __devexit tegra30_ahub_remove(struct platform_device *pdev) +{ + if (!ahub) + return -ENODEV; + + pm_runtime_disable(&pdev->dev); + if (!pm_runtime_status_suspended(&pdev->dev)) + tegra30_ahub_runtime_suspend(&pdev->dev); + + clk_put(ahub->clk_apbif); + clk_put(ahub->clk_d_audio); + + ahub = 0; + + return 0; +} + +static const struct of_device_id tegra30_ahub_of_match[] __devinitconst = { + { .compatible = "nvidia,tegra30-ahub", }, + {}, +}; + +static const struct dev_pm_ops tegra30_ahub_pm_ops __devinitconst = { + SET_RUNTIME_PM_OPS(tegra30_ahub_runtime_suspend, + tegra30_ahub_runtime_resume, NULL) +}; + +static struct platform_driver tegra30_ahub_driver = { + .probe = tegra30_ahub_probe, + .remove = __devexit_p(tegra30_ahub_remove), + .driver = { + .name = DRV_NAME, + .owner = THIS_MODULE, + .of_match_table = tegra30_ahub_of_match, + .pm = &tegra30_ahub_pm_ops, + }, +}; +module_platform_driver(tegra30_ahub_driver); + +MODULE_AUTHOR("Stephen Warren "); +MODULE_DESCRIPTION("Tegra30 AHUB driver"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:" DRV_NAME); diff --git a/sound/soc/tegra/tegra30_ahub.h b/sound/soc/tegra/tegra30_ahub.h new file mode 100644 index 000000000000..e690e2eecc92 --- /dev/null +++ b/sound/soc/tegra/tegra30_ahub.h @@ -0,0 +1,483 @@ +/* + * tegra30_ahub.h - Definitions for Tegra30 AHUB driver + * + * Copyright (c) 2011,2012, NVIDIA CORPORATION. All rights reserved. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms and conditions of the GNU General Public License, + * version 2, as published by the Free Software Foundation. + * + * This program is distributed in the hope it will be useful, but WITHOUT + * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or + * FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for + * more details. + * + * You should have received a copy of the GNU General Public License + * along with this program. If not, see . + */ + +#ifndef __TEGRA30_AHUB_H__ +#define __TEGRA30_AHUB_H__ + +/* Fields in *_CIF_RX/TX_CTRL; used by AHUB FIFOs, and all other audio modules */ + +#define TEGRA30_AUDIOCIF_CTRL_FIFO_THRESHOLD_SHIFT 28 +#define TEGRA30_AUDIOCIF_CTRL_FIFO_THRESHOLD_MASK_US 0xf +#define TEGRA30_AUDIOCIF_CTRL_FIFO_THRESHOLD_MASK (TEGRA30_AUDIOCIF_CTRL_FIFO_THRESHOLD_MASK_US << TEGRA30_AUDIOCIF_CTRL_FIFO_THRESHOLD_SHIFT) + +/* Channel count minus 1 */ +#define TEGRA30_AUDIOCIF_CTRL_AUDIO_CHANNELS_SHIFT 24 +#define TEGRA30_AUDIOCIF_CTRL_AUDIO_CHANNELS_MASK_US 7 +#define TEGRA30_AUDIOCIF_CTRL_AUDIO_CHANNELS_MASK (TEGRA30_AUDIOCIF_CTRL_AUDIO_CHANNELS_MASK_US << TEGRA30_AUDIOCIF_CTRL_AUDIO_CHANNELS_SHIFT) + +/* Channel count minus 1 */ +#define TEGRA30_AUDIOCIF_CTRL_CLIENT_CHANNELS_SHIFT 16 +#define TEGRA30_AUDIOCIF_CTRL_CLIENT_CHANNELS_MASK_US 7 +#define TEGRA30_AUDIOCIF_CTRL_CLIENT_CHANNELS_MASK (TEGRA30_AUDIOCIF_CTRL_CLIENT_CHANNELS_MASK_US << TEGRA30_AUDIOCIF_CTRL_CLIENT_CHANNELS_SHIFT) + +#define TEGRA30_AUDIOCIF_BITS_4 0 +#define TEGRA30_AUDIOCIF_BITS_8 1 +#define TEGRA30_AUDIOCIF_BITS_12 2 +#define TEGRA30_AUDIOCIF_BITS_16 3 +#define TEGRA30_AUDIOCIF_BITS_20 4 +#define TEGRA30_AUDIOCIF_BITS_24 5 +#define TEGRA30_AUDIOCIF_BITS_28 6 +#define TEGRA30_AUDIOCIF_BITS_32 7 + +#define TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_SHIFT 12 +#define TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_MASK (7 << TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_4 (TEGRA30_AUDIOCIF_BITS_4 << TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_8 (TEGRA30_AUDIOCIF_BITS_8 << TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_12 (TEGRA30_AUDIOCIF_BITS_12 << TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_16 (TEGRA30_AUDIOCIF_BITS_16 << TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_20 (TEGRA30_AUDIOCIF_BITS_20 << TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_24 (TEGRA30_AUDIOCIF_BITS_24 << TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_28 (TEGRA30_AUDIOCIF_BITS_28 << TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_32 (TEGRA30_AUDIOCIF_BITS_32 << TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_SHIFT) + +#define TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_SHIFT 8 +#define TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_MASK (7 << TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_4 (TEGRA30_AUDIOCIF_BITS_4 << TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_8 (TEGRA30_AUDIOCIF_BITS_8 << TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_12 (TEGRA30_AUDIOCIF_BITS_12 << TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_16 (TEGRA30_AUDIOCIF_BITS_16 << TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_20 (TEGRA30_AUDIOCIF_BITS_20 << TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_24 (TEGRA30_AUDIOCIF_BITS_24 << TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_28 (TEGRA30_AUDIOCIF_BITS_28 << TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_32 (TEGRA30_AUDIOCIF_BITS_32 << TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_SHIFT) + +#define TEGRA30_AUDIOCIF_EXPAND_ZERO 0 +#define TEGRA30_AUDIOCIF_EXPAND_ONE 1 +#define TEGRA30_AUDIOCIF_EXPAND_LFSR 2 + +#define TEGRA30_AUDIOCIF_CTRL_EXPAND_SHIFT 6 +#define TEGRA30_AUDIOCIF_CTRL_EXPAND_MASK (3 << TEGRA30_AUDIOCIF_CTRL_EXPAND_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_EXPAND_ZERO (TEGRA30_AUDIOCIF_EXPAND_ZERO << TEGRA30_AUDIOCIF_CTRL_EXPAND_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_EXPAND_ONE (TEGRA30_AUDIOCIF_EXPAND_ONE << TEGRA30_AUDIOCIF_CTRL_EXPAND_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_EXPAND_LFSR (TEGRA30_AUDIOCIF_EXPAND_LFSR << TEGRA30_AUDIOCIF_CTRL_EXPAND_SHIFT) + +#define TEGRA30_AUDIOCIF_STEREO_CONV_CH0 0 +#define TEGRA30_AUDIOCIF_STEREO_CONV_CH1 1 +#define TEGRA30_AUDIOCIF_STEREO_CONV_AVG 2 + +#define TEGRA30_AUDIOCIF_CTRL_STEREO_CONV_SHIFT 4 +#define TEGRA30_AUDIOCIF_CTRL_STEREO_CONV_MASK (3 << TEGRA30_AUDIOCIF_CTRL_STEREO_CONV_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_STEREO_CONV_CH0 (TEGRA30_AUDIOCIF_STEREO_CONV_CH0 << TEGRA30_AUDIOCIF_CTRL_STEREO_CONV_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_STEREO_CONV_CH1 (TEGRA30_AUDIOCIF_STEREO_CONV_CH1 << TEGRA30_AUDIOCIF_CTRL_STEREO_CONV_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_STEREO_CONV_AVG (TEGRA30_AUDIOCIF_STEREO_CONV_AVG << TEGRA30_AUDIOCIF_CTRL_STEREO_CONV_SHIFT) + +#define TEGRA30_AUDIOCIF_CTRL_REPLICATE 3 + +#define TEGRA30_AUDIOCIF_DIRECTION_TX 0 +#define TEGRA30_AUDIOCIF_DIRECTION_RX 1 + +#define TEGRA30_AUDIOCIF_CTRL_DIRECTION_SHIFT 2 +#define TEGRA30_AUDIOCIF_CTRL_DIRECTION_MASK (1 << TEGRA30_AUDIOCIF_CTRL_DIRECTION_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_DIRECTION_TX (TEGRA30_AUDIOCIF_DIRECTION_TX << TEGRA30_AUDIOCIF_CTRL_DIRECTION_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_DIRECTION_RX (TEGRA30_AUDIOCIF_DIRECTION_RX << TEGRA30_AUDIOCIF_CTRL_DIRECTION_SHIFT) + +#define TEGRA30_AUDIOCIF_TRUNCATE_ROUND 0 +#define TEGRA30_AUDIOCIF_TRUNCATE_CHOP 1 + +#define TEGRA30_AUDIOCIF_CTRL_TRUNCATE_SHIFT 1 +#define TEGRA30_AUDIOCIF_CTRL_TRUNCATE_MASK (1 << TEGRA30_AUDIOCIF_CTRL_TRUNCATE_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_TRUNCATE_ROUND (TEGRA30_AUDIOCIF_TRUNCATE_ROUND << TEGRA30_AUDIOCIF_CTRL_TRUNCATE_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_TRUNCATE_CHOP (TEGRA30_AUDIOCIF_TRUNCATE_CHOP << TEGRA30_AUDIOCIF_CTRL_TRUNCATE_SHIFT) + +#define TEGRA30_AUDIOCIF_MONO_CONV_ZERO 0 +#define TEGRA30_AUDIOCIF_MONO_CONV_COPY 1 + +#define TEGRA30_AUDIOCIF_CTRL_MONO_CONV_SHIFT 0 +#define TEGRA30_AUDIOCIF_CTRL_MONO_CONV_MASK (1 << TEGRA30_AUDIOCIF_CTRL_MONO_CONV_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_MONO_CONV_ZERO (TEGRA30_AUDIOCIF_MONO_CONV_ZERO << TEGRA30_AUDIOCIF_CTRL_MONO_CONV_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_MONO_CONV_COPY (TEGRA30_AUDIOCIF_MONO_CONV_COPY << TEGRA30_AUDIOCIF_CTRL_MONO_CONV_SHIFT) + +/* Registers within TEGRA30_AUDIO_CLUSTER_BASE */ + +/* TEGRA30_AHUB_CHANNEL_CTRL */ + +#define TEGRA30_AHUB_CHANNEL_CTRL 0x0 +#define TEGRA30_AHUB_CHANNEL_CTRL_STRIDE 0x20 +#define TEGRA30_AHUB_CHANNEL_CTRL_COUNT 4 +#define TEGRA30_AHUB_CHANNEL_CTRL_TX_EN (1 << 31) +#define TEGRA30_AHUB_CHANNEL_CTRL_RX_EN (1 << 30) +#define TEGRA30_AHUB_CHANNEL_CTRL_LOOPBACK (1 << 29) + +#define TEGRA30_AHUB_CHANNEL_CTRL_TX_THRESHOLD_SHIFT 16 +#define TEGRA30_AHUB_CHANNEL_CTRL_TX_THRESHOLD_MASK_US 0xff +#define TEGRA30_AHUB_CHANNEL_CTRL_TX_THRESHOLD_MASK (TEGRA30_AHUB_CHANNEL_CTRL_TX_THRESHOLD_MASK_US << TEGRA30_AHUB_CHANNEL_CTRL_TX_THRESHOLD_SHIFT) + +#define TEGRA30_AHUB_CHANNEL_CTRL_RX_THRESHOLD_SHIFT 8 +#define TEGRA30_AHUB_CHANNEL_CTRL_RX_THRESHOLD_MASK_US 0xff +#define TEGRA30_AHUB_CHANNEL_CTRL_RX_THRESHOLD_MASK (TEGRA30_AHUB_CHANNEL_CTRL_RX_THRESHOLD_MASK_US << TEGRA30_AHUB_CHANNEL_CTRL_RX_THRESHOLD_SHIFT) + +#define TEGRA30_AHUB_CHANNEL_CTRL_TX_PACK_EN (1 << 6) + +#define TEGRA30_PACK_8_4 2 +#define TEGRA30_PACK_16 3 + +#define TEGRA30_AHUB_CHANNEL_CTRL_TX_PACK_SHIFT 4 +#define TEGRA30_AHUB_CHANNEL_CTRL_TX_PACK_MASK_US 3 +#define TEGRA30_AHUB_CHANNEL_CTRL_TX_PACK_MASK (TEGRA30_AHUB_CHANNEL_CTRL_TX_PACK_MASK_US << TEGRA30_AHUB_CHANNEL_CTRL_TX_PACK_SHIFT) +#define TEGRA30_AHUB_CHANNEL_CTRL_TX_PACK_8_4 (TEGRA30_PACK_8_4 << TEGRA30_AHUB_CHANNEL_CTRL_TX_PACK_SHIFT) +#define TEGRA30_AHUB_CHANNEL_CTRL_TX_PACK_16 (TEGRA30_PACK_16 << TEGRA30_AHUB_CHANNEL_CTRL_TX_PACK_SHIFT) + +#define TEGRA30_AHUB_CHANNEL_CTRL_RX_PACK_EN (1 << 2) + +#define TEGRA30_AHUB_CHANNEL_CTRL_RX_PACK_SHIFT 0 +#define TEGRA30_AHUB_CHANNEL_CTRL_RX_PACK_MASK_US 3 +#define TEGRA30_AHUB_CHANNEL_CTRL_RX_PACK_MASK (TEGRA30_AHUB_CHANNEL_CTRL_RX_PACK_MASK_US << TEGRA30_AHUB_CHANNEL_CTRL_RX_PACK_SHIFT) +#define TEGRA30_AHUB_CHANNEL_CTRL_RX_PACK_8_4 (TEGRA30_PACK_8_4 << TEGRA30_AHUB_CHANNEL_CTRL_RX_PACK_SHIFT) +#define TEGRA30_AHUB_CHANNEL_CTRL_RX_PACK_16 (TEGRA30_PACK_16 << TEGRA30_AHUB_CHANNEL_CTRL_RX_PACK_SHIFT) + +/* TEGRA30_AHUB_CHANNEL_CLEAR */ + +#define TEGRA30_AHUB_CHANNEL_CLEAR 0x4 +#define TEGRA30_AHUB_CHANNEL_CLEAR_STRIDE 0x20 +#define TEGRA30_AHUB_CHANNEL_CLEAR_COUNT 4 +#define TEGRA30_AHUB_CHANNEL_CLEAR_TX_SOFT_RESET (1 << 31) +#define TEGRA30_AHUB_CHANNEL_CLEAR_RX_SOFT_RESET (1 << 30) + +/* TEGRA30_AHUB_CHANNEL_STATUS */ + +#define TEGRA30_AHUB_CHANNEL_STATUS 0x8 +#define TEGRA30_AHUB_CHANNEL_STATUS_STRIDE 0x20 +#define TEGRA30_AHUB_CHANNEL_STATUS_COUNT 4 +#define TEGRA30_AHUB_CHANNEL_STATUS_TX_FREE_SHIFT 24 +#define TEGRA30_AHUB_CHANNEL_STATUS_TX_FREE_MASK_US 0xff +#define TEGRA30_AHUB_CHANNEL_STATUS_TX_FREE_MASK (TEGRA30_AHUB_CHANNEL_STATUS_TX_FREE_MASK_US << TEGRA30_AHUB_CHANNEL_STATUS_TX_FREE_SHIFT) +#define TEGRA30_AHUB_CHANNEL_STATUS_RX_FREE_SHIFT 16 +#define TEGRA30_AHUB_CHANNEL_STATUS_RX_FREE_MASK_US 0xff +#define TEGRA30_AHUB_CHANNEL_STATUS_RX_FREE_MASK (TEGRA30_AHUB_CHANNEL_STATUS_RX_FREE_MASK_US << TEGRA30_AHUB_CHANNEL_STATUS_RX_FREE_SHIFT) +#define TEGRA30_AHUB_CHANNEL_STATUS_TX_TRIG (1 << 1) +#define TEGRA30_AHUB_CHANNEL_STATUS_RX_TRIG (1 << 0) + +/* TEGRA30_AHUB_CHANNEL_TXFIFO */ + +#define TEGRA30_AHUB_CHANNEL_TXFIFO 0xc +#define TEGRA30_AHUB_CHANNEL_TXFIFO_STRIDE 0x20 +#define TEGRA30_AHUB_CHANNEL_TXFIFO_COUNT 4 + +/* TEGRA30_AHUB_CHANNEL_RXFIFO */ + +#define TEGRA30_AHUB_CHANNEL_RXFIFO 0x10 +#define TEGRA30_AHUB_CHANNEL_RXFIFO_STRIDE 0x20 +#define TEGRA30_AHUB_CHANNEL_RXFIFO_COUNT 4 + +/* TEGRA30_AHUB_CIF_TX_CTRL */ + +#define TEGRA30_AHUB_CIF_TX_CTRL 0x14 +#define TEGRA30_AHUB_CIF_TX_CTRL_STRIDE 0x20 +#define TEGRA30_AHUB_CIF_TX_CTRL_COUNT 4 +/* Uses field from TEGRA30_AUDIOCIF_CTRL_* */ + +/* TEGRA30_AHUB_CIF_RX_CTRL */ + +#define TEGRA30_AHUB_CIF_RX_CTRL 0x18 +#define TEGRA30_AHUB_CIF_RX_CTRL_STRIDE 0x20 +#define TEGRA30_AHUB_CIF_RX_CTRL_COUNT 4 +/* Uses field from TEGRA30_AUDIOCIF_CTRL_* */ + +/* TEGRA30_AHUB_CONFIG_LINK_CTRL */ + +#define TEGRA30_AHUB_CONFIG_LINK_CTRL 0x80 +#define TEGRA30_AHUB_CONFIG_LINK_CTRL_MASTER_FIFO_FULL_CNT_SHIFT 28 +#define TEGRA30_AHUB_CONFIG_LINK_CTRL_MASTER_FIFO_FULL_CNT_MASK_US 0xf +#define TEGRA30_AHUB_CONFIG_LINK_CTRL_MASTER_FIFO_FULL_CNT_MASK (TEGRA30_AHUB_CONFIG_LINK_CTRL_MASTER_FIFO_FULL_CNT_MASK_US << TEGRA30_AHUB_CONFIG_LINK_CTRL_MASTER_FIFO_FULL_CNT_SHIFT) +#define TEGRA30_AHUB_CONFIG_LINK_CTRL_TIMEOUT_CNT_SHIFT 16 +#define TEGRA30_AHUB_CONFIG_LINK_CTRL_TIMEOUT_CNT_MASK_US 0xfff +#define TEGRA30_AHUB_CONFIG_LINK_CTRL_TIMEOUT_CNT_MASK (TEGRA30_AHUB_CONFIG_LINK_CTRL_TIMEOUT_CNT_MASK_US << TEGRA30_AHUB_CONFIG_LINK_CTRL_TIMEOUT_CNT_SHIFT) +#define TEGRA30_AHUB_CONFIG_LINK_CTRL_IDLE_CNT_SHIFT 4 +#define TEGRA30_AHUB_CONFIG_LINK_CTRL_IDLE_CNT_MASK_US 0xfff +#define TEGRA30_AHUB_CONFIG_LINK_CTRL_IDLE_CNT_MASK (TEGRA30_AHUB_CONFIG_LINK_CTRL_IDLE_CNT_MASK_US << TEGRA30_AHUB_CONFIG_LINK_CTRL_IDLE_CNT_SHIFT) +#define TEGRA30_AHUB_CONFIG_LINK_CTRL_CG_EN (1 << 2) +#define TEGRA30_AHUB_CONFIG_LINK_CTRL_CLEAR_TIMEOUT_CNTR (1 << 1) +#define TEGRA30_AHUB_CONFIG_LINK_CTRL_SOFT_RESET (1 << 0) + +/* TEGRA30_AHUB_MISC_CTRL */ + +#define TEGRA30_AHUB_MISC_CTRL 0x84 +#define TEGRA30_AHUB_MISC_CTRL_AUDIO_ACTIVE (1 << 31) +#define TEGRA30_AHUB_MISC_CTRL_AUDIO_CG_EN (1 << 8) +#define TEGRA30_AHUB_MISC_CTRL_AUDIO_OBS_SEL_SHIFT 0 +#define TEGRA30_AHUB_MISC_CTRL_AUDIO_OBS_SEL_MASK (0x1f << TEGRA30_AHUB_MISC_CTRL_AUDIO_OBS_SEL_SHIFT) + +/* TEGRA30_AHUB_APBDMA_LIVE_STATUS */ + +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS 0x88 +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH3_RX_CIF_FIFO_FULL (1 << 31) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH3_TX_CIF_FIFO_FULL (1 << 30) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH2_RX_CIF_FIFO_FULL (1 << 29) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH2_TX_CIF_FIFO_FULL (1 << 28) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH1_RX_CIF_FIFO_FULL (1 << 27) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH1_TX_CIF_FIFO_FULL (1 << 26) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH0_RX_CIF_FIFO_FULL (1 << 25) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH0_TX_CIF_FIFO_FULL (1 << 24) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH3_RX_CIF_FIFO_EMPTY (1 << 23) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH3_TX_CIF_FIFO_EMPTY (1 << 22) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH2_RX_CIF_FIFO_EMPTY (1 << 21) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH2_TX_CIF_FIFO_EMPTY (1 << 20) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH1_RX_CIF_FIFO_EMPTY (1 << 19) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH1_TX_CIF_FIFO_EMPTY (1 << 18) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH0_RX_CIF_FIFO_EMPTY (1 << 17) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH0_TX_CIF_FIFO_EMPTY (1 << 16) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH3_RX_DMA_FIFO_FULL (1 << 15) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH3_TX_DMA_FIFO_FULL (1 << 14) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH2_RX_DMA_FIFO_FULL (1 << 13) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH2_TX_DMA_FIFO_FULL (1 << 12) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH1_RX_DMA_FIFO_FULL (1 << 11) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH1_TX_DMA_FIFO_FULL (1 << 10) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH0_RX_DMA_FIFO_FULL (1 << 9) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH0_TX_DMA_FIFO_FULL (1 << 8) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH3_RX_DMA_FIFO_EMPTY (1 << 7) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH3_TX_DMA_FIFO_EMPTY (1 << 6) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH2_RX_DMA_FIFO_EMPTY (1 << 5) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH2_TX_DMA_FIFO_EMPTY (1 << 4) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH1_RX_DMA_FIFO_EMPTY (1 << 3) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH1_TX_DMA_FIFO_EMPTY (1 << 2) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH0_RX_DMA_FIFO_EMPTY (1 << 1) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH0_TX_DMA_FIFO_EMPTY (1 << 0) + +/* TEGRA30_AHUB_I2S_LIVE_STATUS */ + +#define TEGRA30_AHUB_I2S_LIVE_STATUS 0x8c +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S4_RX_FIFO_FULL (1 << 29) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S4_TX_FIFO_FULL (1 << 28) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S3_RX_FIFO_FULL (1 << 27) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S3_TX_FIFO_FULL (1 << 26) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S2_RX_FIFO_FULL (1 << 25) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S2_TX_FIFO_FULL (1 << 24) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S1_RX_FIFO_FULL (1 << 23) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S1_TX_FIFO_FULL (1 << 22) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S0_RX_FIFO_FULL (1 << 21) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S0_TX_FIFO_FULL (1 << 20) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S4_RX_FIFO_ENABLED (1 << 19) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S4_TX_FIFO_ENABLED (1 << 18) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S3_RX_FIFO_ENABLED (1 << 17) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S3_TX_FIFO_ENABLED (1 << 16) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S2_RX_FIFO_ENABLED (1 << 15) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S2_TX_FIFO_ENABLED (1 << 14) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S1_RX_FIFO_ENABLED (1 << 13) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S1_TX_FIFO_ENABLED (1 << 12) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S0_RX_FIFO_ENABLED (1 << 11) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S0_TX_FIFO_ENABLED (1 << 10) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S4_RX_FIFO_EMPTY (1 << 9) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S4_TX_FIFO_EMPTY (1 << 8) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S3_RX_FIFO_EMPTY (1 << 7) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S3_TX_FIFO_EMPTY (1 << 6) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S2_RX_FIFO_EMPTY (1 << 5) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S2_TX_FIFO_EMPTY (1 << 4) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S1_RX_FIFO_EMPTY (1 << 3) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S1_TX_FIFO_EMPTY (1 << 2) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S0_RX_FIFO_EMPTY (1 << 1) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S0_TX_FIFO_EMPTY (1 << 0) + +/* TEGRA30_AHUB_DAM0_LIVE_STATUS */ + +#define TEGRA30_AHUB_DAM_LIVE_STATUS 0x90 +#define TEGRA30_AHUB_DAM_LIVE_STATUS_STRIDE 0x8 +#define TEGRA30_AHUB_DAM_LIVE_STATUS_COUNT 3 +#define TEGRA30_AHUB_DAM_LIVE_STATUS_TX_ENABLED (1 << 26) +#define TEGRA30_AHUB_DAM_LIVE_STATUS_RX1_ENABLED (1 << 25) +#define TEGRA30_AHUB_DAM_LIVE_STATUS_RX0_ENABLED (1 << 24) +#define TEGRA30_AHUB_DAM_LIVE_STATUS_TXFIFO_FULL (1 << 15) +#define TEGRA30_AHUB_DAM_LIVE_STATUS_RX1FIFO_FULL (1 << 9) +#define TEGRA30_AHUB_DAM_LIVE_STATUS_RX0FIFO_FULL (1 << 8) +#define TEGRA30_AHUB_DAM_LIVE_STATUS_TXFIFO_EMPTY (1 << 7) +#define TEGRA30_AHUB_DAM_LIVE_STATUS_RX1FIFO_EMPTY (1 << 1) +#define TEGRA30_AHUB_DAM_LIVE_STATUS_RX0FIFO_EMPTY (1 << 0) + +/* TEGRA30_AHUB_SPDIF_LIVE_STATUS */ + +#define TEGRA30_AHUB_SPDIF_LIVE_STATUS 0xa8 +#define TEGRA30_AHUB_SPDIF_LIVE_STATUS_USER_TX_ENABLED (1 << 11) +#define TEGRA30_AHUB_SPDIF_LIVE_STATUS_USER_RX_ENABLED (1 << 10) +#define TEGRA30_AHUB_SPDIF_LIVE_STATUS_DATA_TX_ENABLED (1 << 9) +#define TEGRA30_AHUB_SPDIF_LIVE_STATUS_DATA_RX_ENABLED (1 << 8) +#define TEGRA30_AHUB_SPDIF_LIVE_STATUS_USER_TXFIFO_FULL (1 << 7) +#define TEGRA30_AHUB_SPDIF_LIVE_STATUS_USER_RXFIFO_FULL (1 << 6) +#define TEGRA30_AHUB_SPDIF_LIVE_STATUS_DATA_TXFIFO_FULL (1 << 5) +#define TEGRA30_AHUB_SPDIF_LIVE_STATUS_DATA_RXFIFO_FULL (1 << 4) +#define TEGRA30_AHUB_SPDIF_LIVE_STATUS_USER_TXFIFO_EMPTY (1 << 3) +#define TEGRA30_AHUB_SPDIF_LIVE_STATUS_USER_RXFIFO_EMPTY (1 << 2) +#define TEGRA30_AHUB_SPDIF_LIVE_STATUS_DATA_TXFIFO_EMPTY (1 << 1) +#define TEGRA30_AHUB_SPDIF_LIVE_STATUS_DATA_RXFIFO_EMPTY (1 << 0) + +/* TEGRA30_AHUB_I2S_INT_MASK */ + +#define TEGRA30_AHUB_I2S_INT_MASK 0xb0 + +/* TEGRA30_AHUB_DAM_INT_MASK */ + +#define TEGRA30_AHUB_DAM_INT_MASK 0xb4 + +/* TEGRA30_AHUB_SPDIF_INT_MASK */ + +#define TEGRA30_AHUB_SPDIF_INT_MASK 0xbc + +/* TEGRA30_AHUB_APBIF_INT_MASK */ + +#define TEGRA30_AHUB_APBIF_INT_MASK 0xc0 + +/* TEGRA30_AHUB_I2S_INT_STATUS */ + +#define TEGRA30_AHUB_I2S_INT_STATUS 0xc8 + +/* TEGRA30_AHUB_DAM_INT_STATUS */ + +#define TEGRA30_AHUB_DAM_INT_STATUS 0xcc + +/* TEGRA30_AHUB_SPDIF_INT_STATUS */ + +#define TEGRA30_AHUB_SPDIF_INT_STATUS 0xd4 + +/* TEGRA30_AHUB_APBIF_INT_STATUS */ + +#define TEGRA30_AHUB_APBIF_INT_STATUS 0xd8 + +/* TEGRA30_AHUB_I2S_INT_SOURCE */ + +#define TEGRA30_AHUB_I2S_INT_SOURCE 0xe0 + +/* TEGRA30_AHUB_DAM_INT_SOURCE */ + +#define TEGRA30_AHUB_DAM_INT_SOURCE 0xe4 + +/* TEGRA30_AHUB_SPDIF_INT_SOURCE */ + +#define TEGRA30_AHUB_SPDIF_INT_SOURCE 0xec + +/* TEGRA30_AHUB_APBIF_INT_SOURCE */ + +#define TEGRA30_AHUB_APBIF_INT_SOURCE 0xf0 + +/* TEGRA30_AHUB_I2S_INT_SET */ + +#define TEGRA30_AHUB_I2S_INT_SET 0xf8 + +/* TEGRA30_AHUB_DAM_INT_SET */ + +#define TEGRA30_AHUB_DAM_INT_SET 0xfc + +/* TEGRA30_AHUB_SPDIF_INT_SET */ + +#define TEGRA30_AHUB_SPDIF_INT_SET 0x100 + +/* TEGRA30_AHUB_APBIF_INT_SET */ + +#define TEGRA30_AHUB_APBIF_INT_SET 0x104 + +/* Registers within TEGRA30_AHUB_BASE */ + +#define TEGRA30_AHUB_AUDIO_RX 0x0 +#define TEGRA30_AHUB_AUDIO_RX_STRIDE 0x4 +#define TEGRA30_AHUB_AUDIO_RX_COUNT 17 +/* This register repeats once for each entry in enum tegra30_ahub_rxcif */ +/* The fields in this register are 1 bit per entry in tegra30_ahub_txcif */ + +/* + * Terminology: + * AHUB: Audio Hub; a cross-bar switch between the audio devices: DMA FIFOs, + * I2S controllers, SPDIF controllers, and DAMs. + * XBAR: The core cross-bar component of the AHUB. + * CIF: Client Interface; the HW module connecting an audio device to the + * XBAR. + * DAM: Digital Audio Mixer: A HW module that mixes multiple audio streams, + * possibly including sample-rate conversion. + * + * Each TX CIF transmits data into the XBAR. Each RX CIF can receive audio + * transmitted by a particular TX CIF. + * + * This driver is currently very simplistic; many HW features are not + * exposed; DAMs are not supported, only 16-bit stereo audio is supported, + * etc. + */ + +enum tegra30_ahub_txcif { + TEGRA30_AHUB_TXCIF_APBIF_TX0, + TEGRA30_AHUB_TXCIF_APBIF_TX1, + TEGRA30_AHUB_TXCIF_APBIF_TX2, + TEGRA30_AHUB_TXCIF_APBIF_TX3, + TEGRA30_AHUB_TXCIF_I2S0_TX0, + TEGRA30_AHUB_TXCIF_I2S1_TX0, + TEGRA30_AHUB_TXCIF_I2S2_TX0, + TEGRA30_AHUB_TXCIF_I2S3_TX0, + TEGRA30_AHUB_TXCIF_I2S4_TX0, + TEGRA30_AHUB_TXCIF_DAM0_TX0, + TEGRA30_AHUB_TXCIF_DAM1_TX0, + TEGRA30_AHUB_TXCIF_DAM2_TX0, + TEGRA30_AHUB_TXCIF_SPDIF_TX0, + TEGRA30_AHUB_TXCIF_SPDIF_TX1, +}; + +enum tegra30_ahub_rxcif { + TEGRA30_AHUB_RXCIF_APBIF_RX0, + TEGRA30_AHUB_RXCIF_APBIF_RX1, + TEGRA30_AHUB_RXcIF_APBIF_RX2, + TEGRA30_AHUB_RXCIF_APBIF_RX3, + TEGRA30_AHUB_RXCIF_I2S0_RX0, + TEGRA30_AHUB_RXCIF_I2S1_RX0, + TEGRA30_AHUB_RXCIF_I2S2_RX0, + TEGRA30_AHUB_RXCIF_I2S3_RX0, + TEGRA30_AHUB_RXCIF_I2S4_RX0, + TEGRA30_AHUB_RXCIF_DAM0_RX0, + TEGRA30_AHUB_RXCIF_DAM0_RX1, + TEGRA30_AHUB_RXCIF_DAM1_RX0, + TEGRA30_AHUB_RXCIF_DAM2_RX1, + TEGRA30_AHUB_RXCIF_DAM3_RX0, + TEGRA30_AHUB_RXCIF_DAM3_RX1, + TEGRA30_AHUB_RXCIF_SPDIF_RX0, + TEGRA30_AHUB_RXCIF_SPDIF_RX1, +}; + +extern int tegra30_ahub_allocate_rx_fifo(enum tegra30_ahub_rxcif *rxcif, + unsigned long *fiforeg, + unsigned long *reqsel); +extern int tegra30_ahub_enable_rx_fifo(enum tegra30_ahub_rxcif rxcif); +extern int tegra30_ahub_disable_rx_fifo(enum tegra30_ahub_rxcif rxcif); +extern int tegra30_ahub_free_rx_fifo(enum tegra30_ahub_rxcif rxcif); + +extern int tegra30_ahub_allocate_tx_fifo(enum tegra30_ahub_txcif *txcif, + unsigned long *fiforeg, + unsigned long *reqsel); +extern int tegra30_ahub_enable_tx_fifo(enum tegra30_ahub_txcif txcif); +extern int tegra30_ahub_disable_tx_fifo(enum tegra30_ahub_txcif txcif); +extern int tegra30_ahub_free_tx_fifo(enum tegra30_ahub_txcif txcif); + +extern int tegra30_ahub_set_rx_cif_source(enum tegra30_ahub_rxcif rxcif, + enum tegra30_ahub_txcif txcif); +extern int tegra30_ahub_unset_rx_cif_source(enum tegra30_ahub_rxcif rxcif); + +struct tegra30_ahub { + struct device *dev; + struct clk *clk_d_audio; + struct clk *clk_apbif; + int dma_sel; + resource_size_t apbif_addr; + struct regmap *regmap_apbif; + struct regmap *regmap_ahub; + DECLARE_BITMAP(rx_usage, TEGRA30_AHUB_CHANNEL_CTRL_COUNT); + DECLARE_BITMAP(tx_usage, TEGRA30_AHUB_CHANNEL_CTRL_COUNT); +}; + +#endif -- cgit v1.2.3 From 4fb0384f3dc68da10cf3f134c45efc6ab14f71df Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Tue, 10 Apr 2012 16:32:00 -0600 Subject: ASoC: tegra: add tegra30-i2s driver This provides an ASoC DAI interface for Tegra 30's I2S controller. Includes a squashed bugfix from Sumit Bhattacharya Signed-off-by: Stephen Warren Signed-off-by: Mark Brown --- .../bindings/sound/nvidia,tegra30-i2s.txt | 15 + sound/soc/tegra/tegra30_i2s.c | 536 +++++++++++++++++++++ sound/soc/tegra/tegra30_i2s.h | 242 ++++++++++ 3 files changed, 793 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra30-i2s.txt create mode 100644 sound/soc/tegra/tegra30_i2s.c create mode 100644 sound/soc/tegra/tegra30_i2s.h (limited to 'sound') diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra30-i2s.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra30-i2s.txt new file mode 100644 index 000000000000..dfa6c037124a --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra30-i2s.txt @@ -0,0 +1,15 @@ +NVIDIA Tegra30 I2S controller + +Required properties: +- compatible : "nvidia,tegra30-i2s" +- reg : Should contain I2S registers location and length +- nvidia,ahub-cif-ids : The list of AHUB CIF IDs for this port, rx (playback) + first, tx (capture) second. See nvidia,tegra30-ahub.txt for values. + +Example: + +i2s@70002800 { + compatible = "nvidia,tegra30-i2s"; + reg = <0x70080300 0x100>; + nvidia,ahub-cif-ids = <4 4>; +}; diff --git a/sound/soc/tegra/tegra30_i2s.c b/sound/soc/tegra/tegra30_i2s.c new file mode 100644 index 000000000000..8596032985dc --- /dev/null +++ b/sound/soc/tegra/tegra30_i2s.c @@ -0,0 +1,536 @@ +/* + * tegra30_i2s.c - Tegra30 I2S driver + * + * Author: Stephen Warren + * Copyright (c) 2010-2012, NVIDIA CORPORATION. All rights reserved. + * + * Based on code copyright/by: + * + * Copyright (c) 2009-2010, NVIDIA Corporation. + * Scott Peterson + * + * Copyright (C) 2010 Google, Inc. + * Iliyan Malchev + * + * This program is free software; you can redistribute it and/or modify it + * under the terms and conditions of the GNU General Public License, + * version 2, as published by the Free Software Foundation. + * + * This program is distributed in the hope it will be useful, but WITHOUT + * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or + * FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for + * more details. + * + * You should have received a copy of the GNU General Public License + * along with this program. If not, see . + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "tegra30_ahub.h" +#include "tegra30_i2s.h" + +#define DRV_NAME "tegra30-i2s" + +static inline void tegra30_i2s_write(struct tegra30_i2s *i2s, u32 reg, u32 val) +{ + regmap_write(i2s->regmap, reg, val); +} + +static inline u32 tegra30_i2s_read(struct tegra30_i2s *i2s, u32 reg) +{ + u32 val; + regmap_read(i2s->regmap, reg, &val); + return val; +} + +static int tegra30_i2s_runtime_suspend(struct device *dev) +{ + struct tegra30_i2s *i2s = dev_get_drvdata(dev); + + regcache_cache_only(i2s->regmap, true); + + clk_disable(i2s->clk_i2s); + + return 0; +} + +static int tegra30_i2s_runtime_resume(struct device *dev) +{ + struct tegra30_i2s *i2s = dev_get_drvdata(dev); + int ret; + + ret = clk_enable(i2s->clk_i2s); + if (ret) { + dev_err(dev, "clk_enable failed: %d\n", ret); + return ret; + } + + regcache_cache_only(i2s->regmap, false); + + return 0; +} + +int tegra30_i2s_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct tegra30_i2s *i2s = snd_soc_dai_get_drvdata(dai); + int ret; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + ret = tegra30_ahub_allocate_tx_fifo(&i2s->playback_fifo_cif, + &i2s->playback_dma_data.addr, + &i2s->playback_dma_data.req_sel); + i2s->playback_dma_data.wrap = 4; + i2s->playback_dma_data.width = 32; + tegra30_ahub_set_rx_cif_source(i2s->playback_i2s_cif, + i2s->playback_fifo_cif); + } else { + ret = tegra30_ahub_allocate_rx_fifo(&i2s->capture_fifo_cif, + &i2s->capture_dma_data.addr, + &i2s->capture_dma_data.req_sel); + i2s->capture_dma_data.wrap = 4; + i2s->capture_dma_data.width = 32; + tegra30_ahub_set_rx_cif_source(i2s->capture_fifo_cif, + i2s->capture_i2s_cif); + } + + return ret; +} + +void tegra30_i2s_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct tegra30_i2s *i2s = snd_soc_dai_get_drvdata(dai); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + tegra30_ahub_unset_rx_cif_source(i2s->playback_i2s_cif); + tegra30_ahub_free_tx_fifo(i2s->playback_fifo_cif); + } else { + tegra30_ahub_unset_rx_cif_source(i2s->capture_fifo_cif); + tegra30_ahub_free_rx_fifo(i2s->capture_fifo_cif); + } +} + +static int tegra30_i2s_set_fmt(struct snd_soc_dai *dai, + unsigned int fmt) +{ + struct tegra30_i2s *i2s = snd_soc_dai_get_drvdata(dai); + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + default: + return -EINVAL; + } + + i2s->reg_ctrl &= ~TEGRA30_I2S_CTRL_MASTER_ENABLE; + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + i2s->reg_ctrl |= TEGRA30_I2S_CTRL_MASTER_ENABLE; + break; + case SND_SOC_DAIFMT_CBM_CFM: + break; + default: + return -EINVAL; + } + + i2s->reg_ctrl &= ~(TEGRA30_I2S_CTRL_FRAME_FORMAT_MASK | + TEGRA30_I2S_CTRL_LRCK_MASK); + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_A: + i2s->reg_ctrl |= TEGRA30_I2S_CTRL_FRAME_FORMAT_FSYNC; + i2s->reg_ctrl |= TEGRA30_I2S_CTRL_LRCK_L_LOW; + break; + case SND_SOC_DAIFMT_DSP_B: + i2s->reg_ctrl |= TEGRA30_I2S_CTRL_FRAME_FORMAT_FSYNC; + i2s->reg_ctrl |= TEGRA30_I2S_CTRL_LRCK_R_LOW; + break; + case SND_SOC_DAIFMT_I2S: + i2s->reg_ctrl |= TEGRA30_I2S_CTRL_FRAME_FORMAT_LRCK; + i2s->reg_ctrl |= TEGRA30_I2S_CTRL_LRCK_L_LOW; + break; + case SND_SOC_DAIFMT_RIGHT_J: + i2s->reg_ctrl |= TEGRA30_I2S_CTRL_FRAME_FORMAT_LRCK; + i2s->reg_ctrl |= TEGRA30_I2S_CTRL_LRCK_L_LOW; + break; + case SND_SOC_DAIFMT_LEFT_J: + i2s->reg_ctrl |= TEGRA30_I2S_CTRL_FRAME_FORMAT_LRCK; + i2s->reg_ctrl |= TEGRA30_I2S_CTRL_LRCK_L_LOW; + break; + default: + return -EINVAL; + } + + return 0; +} + +static int tegra30_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct device *dev = substream->pcm->card->dev; + struct tegra30_i2s *i2s = snd_soc_dai_get_drvdata(dai); + u32 val; + int ret, sample_size, srate, i2sclock, bitcnt; + + if (params_channels(params) != 2) + return -EINVAL; + + i2s->reg_ctrl &= ~TEGRA30_I2S_CTRL_BIT_SIZE_MASK; + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + i2s->reg_ctrl |= TEGRA30_I2S_CTRL_BIT_SIZE_16; + sample_size = 16; + break; + default: + return -EINVAL; + } + + srate = params_rate(params); + + /* Final "* 2" required by Tegra hardware */ + i2sclock = srate * params_channels(params) * sample_size * 2; + + bitcnt = (i2sclock / (2 * srate)) - 1; + if (bitcnt < 0 || bitcnt > TEGRA30_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US) + return -EINVAL; + + ret = clk_set_rate(i2s->clk_i2s, i2sclock); + if (ret) { + dev_err(dev, "Can't set I2S clock rate: %d\n", ret); + return ret; + } + + val = bitcnt << TEGRA30_I2S_TIMING_CHANNEL_BIT_COUNT_SHIFT; + + if (i2sclock % (2 * srate)) + val |= TEGRA30_I2S_TIMING_NON_SYM_ENABLE; + + tegra30_i2s_write(i2s, TEGRA30_I2S_TIMING, val); + + val = (0 << TEGRA30_AUDIOCIF_CTRL_FIFO_THRESHOLD_SHIFT) | + (1 << TEGRA30_AUDIOCIF_CTRL_AUDIO_CHANNELS_SHIFT) | + (1 << TEGRA30_AUDIOCIF_CTRL_CLIENT_CHANNELS_SHIFT) | + TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_16 | + TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_16; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + val |= TEGRA30_AUDIOCIF_CTRL_DIRECTION_RX; + tegra30_i2s_write(i2s, TEGRA30_I2S_CIF_RX_CTRL, val); + } else { + val |= TEGRA30_AUDIOCIF_CTRL_DIRECTION_TX; + tegra30_i2s_write(i2s, TEGRA30_I2S_CIF_TX_CTRL, val); + } + + val = (1 << TEGRA30_I2S_OFFSET_RX_DATA_OFFSET_SHIFT) | + (1 << TEGRA30_I2S_OFFSET_TX_DATA_OFFSET_SHIFT); + tegra30_i2s_write(i2s, TEGRA30_I2S_OFFSET, val); + + return 0; +} + +static void tegra30_i2s_start_playback(struct tegra30_i2s *i2s) +{ + tegra30_ahub_enable_tx_fifo(i2s->playback_fifo_cif); + i2s->reg_ctrl |= TEGRA30_I2S_CTRL_XFER_EN_TX; + tegra30_i2s_write(i2s, TEGRA30_I2S_CTRL, i2s->reg_ctrl); +} + +static void tegra30_i2s_stop_playback(struct tegra30_i2s *i2s) +{ + tegra30_ahub_disable_tx_fifo(i2s->playback_fifo_cif); + i2s->reg_ctrl &= ~TEGRA30_I2S_CTRL_XFER_EN_TX; + tegra30_i2s_write(i2s, TEGRA30_I2S_CTRL, i2s->reg_ctrl); +} + +static void tegra30_i2s_start_capture(struct tegra30_i2s *i2s) +{ + tegra30_ahub_enable_rx_fifo(i2s->capture_fifo_cif); + i2s->reg_ctrl |= TEGRA30_I2S_CTRL_XFER_EN_RX; + tegra30_i2s_write(i2s, TEGRA30_I2S_CTRL, i2s->reg_ctrl); +} + +static void tegra30_i2s_stop_capture(struct tegra30_i2s *i2s) +{ + tegra30_ahub_disable_rx_fifo(i2s->capture_fifo_cif); + i2s->reg_ctrl &= ~TEGRA30_I2S_CTRL_XFER_EN_RX; + tegra30_i2s_write(i2s, TEGRA30_I2S_CTRL, i2s->reg_ctrl); +} + +static int tegra30_i2s_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct tegra30_i2s *i2s = snd_soc_dai_get_drvdata(dai); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + case SNDRV_PCM_TRIGGER_RESUME: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + tegra30_i2s_start_playback(i2s); + else + tegra30_i2s_start_capture(i2s); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + case SNDRV_PCM_TRIGGER_SUSPEND: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + tegra30_i2s_stop_playback(i2s); + else + tegra30_i2s_stop_capture(i2s); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int tegra30_i2s_probe(struct snd_soc_dai *dai) +{ + struct tegra30_i2s *i2s = snd_soc_dai_get_drvdata(dai); + + dai->capture_dma_data = &i2s->capture_dma_data; + dai->playback_dma_data = &i2s->playback_dma_data; + + return 0; +} + +static struct snd_soc_dai_ops tegra30_i2s_dai_ops = { + .startup = tegra30_i2s_startup, + .shutdown = tegra30_i2s_shutdown, + .set_fmt = tegra30_i2s_set_fmt, + .hw_params = tegra30_i2s_hw_params, + .trigger = tegra30_i2s_trigger, +}; + +static const struct snd_soc_dai_driver tegra30_i2s_dai_template = { + .probe = tegra30_i2s_probe, + .playback = { + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .capture = { + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .ops = &tegra30_i2s_dai_ops, + .symmetric_rates = 1, +}; + +static bool tegra30_i2s_wr_rd_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case TEGRA30_I2S_CTRL: + case TEGRA30_I2S_TIMING: + case TEGRA30_I2S_OFFSET: + case TEGRA30_I2S_CH_CTRL: + case TEGRA30_I2S_SLOT_CTRL: + case TEGRA30_I2S_CIF_RX_CTRL: + case TEGRA30_I2S_CIF_TX_CTRL: + case TEGRA30_I2S_FLOWCTL: + case TEGRA30_I2S_TX_STEP: + case TEGRA30_I2S_FLOW_STATUS: + case TEGRA30_I2S_FLOW_TOTAL: + case TEGRA30_I2S_FLOW_OVER: + case TEGRA30_I2S_FLOW_UNDER: + case TEGRA30_I2S_LCOEF_1_4_0: + case TEGRA30_I2S_LCOEF_1_4_1: + case TEGRA30_I2S_LCOEF_1_4_2: + case TEGRA30_I2S_LCOEF_1_4_3: + case TEGRA30_I2S_LCOEF_1_4_4: + case TEGRA30_I2S_LCOEF_1_4_5: + case TEGRA30_I2S_LCOEF_2_4_0: + case TEGRA30_I2S_LCOEF_2_4_1: + case TEGRA30_I2S_LCOEF_2_4_2: + return true; + default: + return false; + }; +} + +static bool tegra30_i2s_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case TEGRA30_I2S_FLOW_STATUS: + case TEGRA30_I2S_FLOW_TOTAL: + case TEGRA30_I2S_FLOW_OVER: + case TEGRA30_I2S_FLOW_UNDER: + return true; + default: + return false; + }; +} + +static const struct regmap_config tegra30_i2s_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = TEGRA30_I2S_LCOEF_2_4_2, + .writeable_reg = tegra30_i2s_wr_rd_reg, + .readable_reg = tegra30_i2s_wr_rd_reg, + .volatile_reg = tegra30_i2s_volatile_reg, + .cache_type = REGCACHE_RBTREE, +}; + +static __devinit int tegra30_i2s_platform_probe(struct platform_device *pdev) +{ + struct tegra30_i2s *i2s; + u32 cif_ids[2]; + struct resource *mem, *memregion; + void __iomem *regs; + int ret; + + i2s = devm_kzalloc(&pdev->dev, sizeof(struct tegra30_i2s), GFP_KERNEL); + if (!i2s) { + dev_err(&pdev->dev, "Can't allocate tegra30_i2s\n"); + ret = -ENOMEM; + goto err; + } + dev_set_drvdata(&pdev->dev, i2s); + + i2s->dai = tegra30_i2s_dai_template; + i2s->dai.name = dev_name(&pdev->dev); + + ret = of_property_read_u32_array(pdev->dev.of_node, + "nvidia,ahub-cif-ids", cif_ids, + ARRAY_SIZE(cif_ids)); + if (ret < 0) + goto err; + + i2s->playback_i2s_cif = cif_ids[0]; + i2s->capture_i2s_cif = cif_ids[1]; + + i2s->clk_i2s = clk_get(&pdev->dev, NULL); + if (IS_ERR(i2s->clk_i2s)) { + dev_err(&pdev->dev, "Can't retrieve i2s clock\n"); + ret = PTR_ERR(i2s->clk_i2s); + goto err; + } + + mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!mem) { + dev_err(&pdev->dev, "No memory resource\n"); + ret = -ENODEV; + goto err_clk_put; + } + + memregion = devm_request_mem_region(&pdev->dev, mem->start, + resource_size(mem), DRV_NAME); + if (!memregion) { + dev_err(&pdev->dev, "Memory region already claimed\n"); + ret = -EBUSY; + goto err_clk_put; + } + + regs = devm_ioremap(&pdev->dev, mem->start, resource_size(mem)); + if (!regs) { + dev_err(&pdev->dev, "ioremap failed\n"); + ret = -ENOMEM; + goto err_clk_put; + } + + i2s->regmap = devm_regmap_init_mmio(&pdev->dev, regs, + &tegra30_i2s_regmap_config); + if (IS_ERR(i2s->regmap)) { + dev_err(&pdev->dev, "regmap init failed\n"); + ret = PTR_ERR(i2s->regmap); + goto err_clk_put; + } + regcache_cache_only(i2s->regmap, true); + + pm_runtime_enable(&pdev->dev); + if (!pm_runtime_enabled(&pdev->dev)) { + ret = tegra30_i2s_runtime_resume(&pdev->dev); + if (ret) + goto err_pm_disable; + } + + ret = snd_soc_register_dai(&pdev->dev, &i2s->dai); + if (ret) { + dev_err(&pdev->dev, "Could not register DAI: %d\n", ret); + ret = -ENOMEM; + goto err_suspend; + } + + ret = tegra_pcm_platform_register(&pdev->dev); + if (ret) { + dev_err(&pdev->dev, "Could not register PCM: %d\n", ret); + goto err_unregister_dai; + } + + return 0; + +err_unregister_dai: + snd_soc_unregister_dai(&pdev->dev); +err_suspend: + if (!pm_runtime_status_suspended(&pdev->dev)) + tegra30_i2s_runtime_suspend(&pdev->dev); +err_pm_disable: + pm_runtime_disable(&pdev->dev); +err_clk_put: + clk_put(i2s->clk_i2s); +err: + return ret; +} + +static int __devexit tegra30_i2s_platform_remove(struct platform_device *pdev) +{ + struct tegra30_i2s *i2s = dev_get_drvdata(&pdev->dev); + + pm_runtime_disable(&pdev->dev); + if (!pm_runtime_status_suspended(&pdev->dev)) + tegra30_i2s_runtime_suspend(&pdev->dev); + + tegra_pcm_platform_unregister(&pdev->dev); + snd_soc_unregister_dai(&pdev->dev); + + clk_put(i2s->clk_i2s); + + return 0; +} + +static const struct of_device_id tegra30_i2s_of_match[] __devinitconst = { + { .compatible = "nvidia,tegra30-i2s", }, + {}, +}; + +static const struct dev_pm_ops tegra30_i2s_pm_ops __devinitconst = { + SET_RUNTIME_PM_OPS(tegra30_i2s_runtime_suspend, + tegra30_i2s_runtime_resume, NULL) +}; + +static struct platform_driver tegra30_i2s_driver = { + .driver = { + .name = DRV_NAME, + .owner = THIS_MODULE, + .of_match_table = tegra30_i2s_of_match, + .pm = &tegra30_i2s_pm_ops, + }, + .probe = tegra30_i2s_platform_probe, + .remove = __devexit_p(tegra30_i2s_platform_remove), +}; +module_platform_driver(tegra30_i2s_driver); + +MODULE_AUTHOR("Stephen Warren "); +MODULE_DESCRIPTION("Tegra30 I2S ASoC driver"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:" DRV_NAME); +MODULE_DEVICE_TABLE(of, tegra30_i2s_of_match); diff --git a/sound/soc/tegra/tegra30_i2s.h b/sound/soc/tegra/tegra30_i2s.h new file mode 100644 index 000000000000..91adf29c7a87 --- /dev/null +++ b/sound/soc/tegra/tegra30_i2s.h @@ -0,0 +1,242 @@ +/* + * tegra30_i2s.h - Definitions for Tegra30 I2S driver + * + * Copyright (c) 2011,2012, NVIDIA CORPORATION. All rights reserved. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms and conditions of the GNU General Public License, + * version 2, as published by the Free Software Foundation. + * + * This program is distributed in the hope it will be useful, but WITHOUT + * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or + * FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for + * more details. + * + * You should have received a copy of the GNU General Public License + * along with this program. If not, see . + */ + +#ifndef __TEGRA30_I2S_H__ +#define __TEGRA30_I2S_H__ + +#include "tegra_pcm.h" + +/* Register offsets from TEGRA30_I2S*_BASE */ + +#define TEGRA30_I2S_CTRL 0x0 +#define TEGRA30_I2S_TIMING 0x4 +#define TEGRA30_I2S_OFFSET 0x08 +#define TEGRA30_I2S_CH_CTRL 0x0c +#define TEGRA30_I2S_SLOT_CTRL 0x10 +#define TEGRA30_I2S_CIF_RX_CTRL 0x14 +#define TEGRA30_I2S_CIF_TX_CTRL 0x18 +#define TEGRA30_I2S_FLOWCTL 0x1c +#define TEGRA30_I2S_TX_STEP 0x20 +#define TEGRA30_I2S_FLOW_STATUS 0x24 +#define TEGRA30_I2S_FLOW_TOTAL 0x28 +#define TEGRA30_I2S_FLOW_OVER 0x2c +#define TEGRA30_I2S_FLOW_UNDER 0x30 +#define TEGRA30_I2S_LCOEF_1_4_0 0x34 +#define TEGRA30_I2S_LCOEF_1_4_1 0x38 +#define TEGRA30_I2S_LCOEF_1_4_2 0x3c +#define TEGRA30_I2S_LCOEF_1_4_3 0x40 +#define TEGRA30_I2S_LCOEF_1_4_4 0x44 +#define TEGRA30_I2S_LCOEF_1_4_5 0x48 +#define TEGRA30_I2S_LCOEF_2_4_0 0x4c +#define TEGRA30_I2S_LCOEF_2_4_1 0x50 +#define TEGRA30_I2S_LCOEF_2_4_2 0x54 + +/* Fields in TEGRA30_I2S_CTRL */ + +#define TEGRA30_I2S_CTRL_XFER_EN_TX (1 << 31) +#define TEGRA30_I2S_CTRL_XFER_EN_RX (1 << 30) +#define TEGRA30_I2S_CTRL_CG_EN (1 << 29) +#define TEGRA30_I2S_CTRL_SOFT_RESET (1 << 28) +#define TEGRA30_I2S_CTRL_TX_FLOWCTL_EN (1 << 27) + +#define TEGRA30_I2S_CTRL_OBS_SEL_SHIFT 24 +#define TEGRA30_I2S_CTRL_OBS_SEL_MASK (7 << TEGRA30_I2S_CTRL_OBS_SEL_SHIFT) + +#define TEGRA30_I2S_FRAME_FORMAT_LRCK 0 +#define TEGRA30_I2S_FRAME_FORMAT_FSYNC 1 + +#define TEGRA30_I2S_CTRL_FRAME_FORMAT_SHIFT 12 +#define TEGRA30_I2S_CTRL_FRAME_FORMAT_MASK (7 << TEGRA30_I2S_CTRL_FRAME_FORMAT_SHIFT) +#define TEGRA30_I2S_CTRL_FRAME_FORMAT_LRCK (TEGRA30_I2S_FRAME_FORMAT_LRCK << TEGRA30_I2S_CTRL_FRAME_FORMAT_SHIFT) +#define TEGRA30_I2S_CTRL_FRAME_FORMAT_FSYNC (TEGRA30_I2S_FRAME_FORMAT_FSYNC << TEGRA30_I2S_CTRL_FRAME_FORMAT_SHIFT) + +#define TEGRA30_I2S_CTRL_MASTER_ENABLE (1 << 10) + +#define TEGRA30_I2S_LRCK_LEFT_LOW 0 +#define TEGRA30_I2S_LRCK_RIGHT_LOW 1 + +#define TEGRA30_I2S_CTRL_LRCK_SHIFT 9 +#define TEGRA30_I2S_CTRL_LRCK_MASK (1 << TEGRA30_I2S_CTRL_LRCK_SHIFT) +#define TEGRA30_I2S_CTRL_LRCK_L_LOW (TEGRA30_I2S_LRCK_LEFT_LOW << TEGRA30_I2S_CTRL_LRCK_SHIFT) +#define TEGRA30_I2S_CTRL_LRCK_R_LOW (TEGRA30_I2S_LRCK_RIGHT_LOW << TEGRA30_I2S_CTRL_LRCK_SHIFT) + +#define TEGRA30_I2S_CTRL_LPBK_ENABLE (1 << 8) + +#define TEGRA30_I2S_BIT_CODE_LINEAR 0 +#define TEGRA30_I2S_BIT_CODE_ULAW 1 +#define TEGRA30_I2S_BIT_CODE_ALAW 2 + +#define TEGRA30_I2S_CTRL_BIT_CODE_SHIFT 4 +#define TEGRA30_I2S_CTRL_BIT_CODE_MASK (3 << TEGRA30_I2S_CTRL_BIT_CODE_SHIFT) +#define TEGRA30_I2S_CTRL_BIT_CODE_LINEAR (TEGRA30_I2S_BIT_CODE_LINEAR << TEGRA30_I2S_CTRL_BIT_CODE_SHIFT) +#define TEGRA30_I2S_CTRL_BIT_CODE_ULAW (TEGRA30_I2S_BIT_CODE_ULAW << TEGRA30_I2S_CTRL_BIT_CODE_SHIFT) +#define TEGRA30_I2S_CTRL_BIT_CODE_ALAW (TEGRA30_I2S_BIT_CODE_ALAW << TEGRA30_I2S_CTRL_BIT_CODE_SHIFT) + +#define TEGRA30_I2S_BITS_8 1 +#define TEGRA30_I2S_BITS_12 2 +#define TEGRA30_I2S_BITS_16 3 +#define TEGRA30_I2S_BITS_20 4 +#define TEGRA30_I2S_BITS_24 5 +#define TEGRA30_I2S_BITS_28 6 +#define TEGRA30_I2S_BITS_32 7 + +/* Sample container size; see {RX,TX}_MASK field in CH_CTRL below */ +#define TEGRA30_I2S_CTRL_BIT_SIZE_SHIFT 0 +#define TEGRA30_I2S_CTRL_BIT_SIZE_MASK (7 << TEGRA30_I2S_CTRL_BIT_SIZE_SHIFT) +#define TEGRA30_I2S_CTRL_BIT_SIZE_8 (TEGRA30_I2S_BITS_8 << TEGRA30_I2S_CTRL_BIT_SIZE_SHIFT) +#define TEGRA30_I2S_CTRL_BIT_SIZE_12 (TEGRA30_I2S_BITS_12 << TEGRA30_I2S_CTRL_BIT_SIZE_SHIFT) +#define TEGRA30_I2S_CTRL_BIT_SIZE_16 (TEGRA30_I2S_BITS_16 << TEGRA30_I2S_CTRL_BIT_SIZE_SHIFT) +#define TEGRA30_I2S_CTRL_BIT_SIZE_20 (TEGRA30_I2S_BITS_20 << TEGRA30_I2S_CTRL_BIT_SIZE_SHIFT) +#define TEGRA30_I2S_CTRL_BIT_SIZE_24 (TEGRA30_I2S_BITS_24 << TEGRA30_I2S_CTRL_BIT_SIZE_SHIFT) +#define TEGRA30_I2S_CTRL_BIT_SIZE_28 (TEGRA30_I2S_BITS_28 << TEGRA30_I2S_CTRL_BIT_SIZE_SHIFT) +#define TEGRA30_I2S_CTRL_BIT_SIZE_32 (TEGRA30_I2S_BITS_32 << TEGRA30_I2S_CTRL_BIT_SIZE_SHIFT) + +/* Fields in TEGRA30_I2S_TIMING */ + +#define TEGRA30_I2S_TIMING_NON_SYM_ENABLE (1 << 12) +#define TEGRA30_I2S_TIMING_CHANNEL_BIT_COUNT_SHIFT 0 +#define TEGRA30_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US 0x7fff +#define TEGRA30_I2S_TIMING_CHANNEL_BIT_COUNT_MASK (TEGRA30_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US << TEGRA30_I2S_TIMING_CHANNEL_BIT_COUNT_SHIFT) + +/* Fields in TEGRA30_I2S_OFFSET */ + +#define TEGRA30_I2S_OFFSET_RX_DATA_OFFSET_SHIFT 16 +#define TEGRA30_I2S_OFFSET_RX_DATA_OFFSET_MASK_US 0x7ff +#define TEGRA30_I2S_OFFSET_RX_DATA_OFFSET_MASK (TEGRA30_I2S_OFFSET_RX_DATA_OFFSET_MASK_US << TEGRA30_I2S_OFFSET_RX_DATA_OFFSET_SHIFT) +#define TEGRA30_I2S_OFFSET_TX_DATA_OFFSET_SHIFT 0 +#define TEGRA30_I2S_OFFSET_TX_DATA_OFFSET_MASK_US 0x7ff +#define TEGRA30_I2S_OFFSET_TX_DATA_OFFSET_MASK (TEGRA30_I2S_OFFSET_TX_DATA_OFFSET_MASK_US << TEGRA30_I2S_OFFSET_TX_DATA_OFFSET_SHIFT) + +/* Fields in TEGRA30_I2S_CH_CTRL */ + +/* (FSYNC width - 1) in bit clocks */ +#define TEGRA30_I2S_CH_CTRL_FSYNC_WIDTH_SHIFT 24 +#define TEGRA30_I2S_CH_CTRL_FSYNC_WIDTH_MASK_US 0xff +#define TEGRA30_I2S_CH_CTRL_FSYNC_WIDTH_MASK (TEGRA30_I2S_CH_CTRL_FSYNC_WIDTH_MASK_US << TEGRA30_I2S_CH_CTRL_FSYNC_WIDTH_SHIFT) + +#define TEGRA30_I2S_HIGHZ_NO 0 +#define TEGRA30_I2S_HIGHZ_YES 1 +#define TEGRA30_I2S_HIGHZ_ON_HALF_BIT_CLK 2 + +#define TEGRA30_I2S_CH_CTRL_HIGHZ_CTRL_SHIFT 12 +#define TEGRA30_I2S_CH_CTRL_HIGHZ_CTRL_MASK (3 << TEGRA30_I2S_CH_CTRL_HIGHZ_CTRL_SHIFT) +#define TEGRA30_I2S_CH_CTRL_HIGHZ_CTRL_NO (TEGRA30_I2S_HIGHZ_NO << TEGRA30_I2S_CH_CTRL_HIGHZ_CTRL_SHIFT) +#define TEGRA30_I2S_CH_CTRL_HIGHZ_CTRL_YES (TEGRA30_I2S_HIGHZ_YES << TEGRA30_I2S_CH_CTRL_HIGHZ_CTRL_SHIFT) +#define TEGRA30_I2S_CH_CTRL_HIGHZ_CTRL_ON_HALF_BIT_CLK (TEGRA30_I2S_HIGHZ_ON_HALF_BIT_CLK << TEGRA30_I2S_CH_CTRL_HIGHZ_CTRL_SHIFT) + +#define TEGRA30_I2S_MSB_FIRST 0 +#define TEGRA30_I2S_LSB_FIRST 1 + +#define TEGRA30_I2S_CH_CTRL_RX_BIT_ORDER_SHIFT 10 +#define TEGRA30_I2S_CH_CTRL_RX_BIT_ORDER_MASK (1 << TEGRA30_I2S_CH_CTRL_RX_BIT_ORDER_SHIFT) +#define TEGRA30_I2S_CH_CTRL_RX_BIT_ORDER_MSB_FIRST (TEGRA30_I2S_MSB_FIRST << TEGRA30_I2S_CH_CTRL_RX_BIT_ORDER_SHIFT) +#define TEGRA30_I2S_CH_CTRL_RX_BIT_ORDER_LSB_FIRST (TEGRA30_I2S_LSB_FIRST << TEGRA30_I2S_CH_CTRL_RX_BIT_ORDER_SHIFT) +#define TEGRA30_I2S_CH_CTRL_TX_BIT_ORDER_SHIFT 9 +#define TEGRA30_I2S_CH_CTRL_TX_BIT_ORDER_MASK (1 << TEGRA30_I2S_CH_CTRL_TX_BIT_ORDER_SHIFT) +#define TEGRA30_I2S_CH_CTRL_TX_BIT_ORDER_MSB_FIRST (TEGRA30_I2S_MSB_FIRST << TEGRA30_I2S_CH_CTRL_TX_BIT_ORDER_SHIFT) +#define TEGRA30_I2S_CH_CTRL_TX_BIT_ORDER_LSB_FIRST (TEGRA30_I2S_LSB_FIRST << TEGRA30_I2S_CH_CTRL_TX_BIT_ORDER_SHIFT) + +#define TEGRA30_I2S_POS_EDGE 0 +#define TEGRA30_I2S_NEG_EDGE 1 + +#define TEGRA30_I2S_CH_CTRL_EGDE_CTRL_SHIFT 8 +#define TEGRA30_I2S_CH_CTRL_EGDE_CTRL_MASK (1 << TEGRA30_I2S_CH_CTRL_EGDE_CTRL_SHIFT) +#define TEGRA30_I2S_CH_CTRL_EGDE_CTRL_POS_EDGE (TEGRA30_I2S_POS_EDGE << TEGRA30_I2S_CH_CTRL_EGDE_CTRL_SHIFT) +#define TEGRA30_I2S_CH_CTRL_EGDE_CTRL_NEG_EDGE (TEGRA30_I2S_NEG_EDGE << TEGRA30_I2S_CH_CTRL_EGDE_CTRL_SHIFT) + +/* Sample size is # bits from BIT_SIZE minus this field */ +#define TEGRA30_I2S_CH_CTRL_RX_MASK_BITS_SHIFT 4 +#define TEGRA30_I2S_CH_CTRL_RX_MASK_BITS_MASK_US 7 +#define TEGRA30_I2S_CH_CTRL_RX_MASK_BITS_MASK (TEGRA30_I2S_CH_CTRL_RX_MASK_BITS_MASK_US << TEGRA30_I2S_CH_CTRL_RX_MASK_BITS_SHIFT) + +#define TEGRA30_I2S_CH_CTRL_TX_MASK_BITS_SHIFT 0 +#define TEGRA30_I2S_CH_CTRL_TX_MASK_BITS_MASK_US 7 +#define TEGRA30_I2S_CH_CTRL_TX_MASK_BITS_MASK (TEGRA30_I2S_CH_CTRL_TX_MASK_BITS_MASK_US << TEGRA30_I2S_CH_CTRL_TX_MASK_BITS_SHIFT) + +/* Fields in TEGRA30_I2S_SLOT_CTRL */ + +/* Number of slots in frame, minus 1 */ +#define TEGRA30_I2S_SLOT_CTRL_TOTAL_SLOTS_SHIFT 16 +#define TEGRA30_I2S_SLOT_CTRL_TOTAL_SLOTS_MASK_US 7 +#define TEGRA30_I2S_SLOT_CTRL_TOTAL_SLOTS_MASK (TEGRA30_I2S_SLOT_CTRL_TOTAL_SLOT_MASK_US << TEGRA30_I2S_SLOT_CTRL_TOTAL_SLOT_SHIFT) + +/* TDM mode slot enable bitmask */ +#define TEGRA30_I2S_SLOT_CTRL_RX_SLOT_ENABLES_SHIFT 8 +#define TEGRA30_I2S_SLOT_CTRL_RX_SLOT_ENABLES_MASK (0xff << TEGRA30_I2S_SLOT_CTRL_RX_SLOT_ENABLES_SHIFT) + +#define TEGRA30_I2S_SLOT_CTRL_TX_SLOT_ENABLES_SHIFT 0 +#define TEGRA30_I2S_SLOT_CTRL_TX_SLOT_ENABLES_MASK (0xff << TEGRA30_I2S_SLOT_CTRL_TX_SLOT_ENABLES_SHIFT) + +/* Fields in TEGRA30_I2S_CIF_RX_CTRL */ +/* Uses field from TEGRA30_AUDIOCIF_CTRL_* in tegra30_ahub.h */ + +/* Fields in TEGRA30_I2S_CIF_TX_CTRL */ +/* Uses field from TEGRA30_AUDIOCIF_CTRL_* in tegra30_ahub.h */ + +/* Fields in TEGRA30_I2S_FLOWCTL */ + +#define TEGRA30_I2S_FILTER_LINEAR 0 +#define TEGRA30_I2S_FILTER_QUAD 1 + +#define TEGRA30_I2S_FLOWCTL_FILTER_SHIFT 31 +#define TEGRA30_I2S_FLOWCTL_FILTER_MASK (1 << TEGRA30_I2S_FLOWCTL_FILTER_SHIFT) +#define TEGRA30_I2S_FLOWCTL_FILTER_LINEAR (TEGRA30_I2S_FILTER_LINEAR << TEGRA30_I2S_FLOWCTL_FILTER_SHIFT) +#define TEGRA30_I2S_FLOWCTL_FILTER_QUAD (TEGRA30_I2S_FILTER_QUAD << TEGRA30_I2S_FLOWCTL_FILTER_SHIFT) + +/* Fields in TEGRA30_I2S_TX_STEP */ + +#define TEGRA30_I2S_TX_STEP_SHIFT 0 +#define TEGRA30_I2S_TX_STEP_MASK_US 0xffff +#define TEGRA30_I2S_TX_STEP_MASK (TEGRA30_I2S_TX_STEP_MASK_US << TEGRA30_I2S_TX_STEP_SHIFT) + +/* Fields in TEGRA30_I2S_FLOW_STATUS */ + +#define TEGRA30_I2S_FLOW_STATUS_UNDERFLOW (1 << 31) +#define TEGRA30_I2S_FLOW_STATUS_OVERFLOW (1 << 30) +#define TEGRA30_I2S_FLOW_STATUS_MONITOR_INT_EN (1 << 4) +#define TEGRA30_I2S_FLOW_STATUS_COUNTER_CLR (1 << 3) +#define TEGRA30_I2S_FLOW_STATUS_MONITOR_CLR (1 << 2) +#define TEGRA30_I2S_FLOW_STATUS_COUNTER_EN (1 << 1) +#define TEGRA30_I2S_FLOW_STATUS_MONITOR_EN (1 << 0) + +/* + * There are no fields in TEGRA30_I2S_FLOW_TOTAL, TEGRA30_I2S_FLOW_OVER, + * TEGRA30_I2S_FLOW_UNDER; they are counters taking the whole register. + */ + +/* Fields in TEGRA30_I2S_LCOEF_* */ + +#define TEGRA30_I2S_LCOEF_COEF_SHIFT 0 +#define TEGRA30_I2S_LCOEF_COEF_MASK_US 0xffff +#define TEGRA30_I2S_LCOEF_COEF_MASK (TEGRA30_I2S_LCOEF_COEF_MASK_US << TEGRA30_I2S_LCOEF_COEF_SHIFT) + +struct tegra30_i2s { + struct snd_soc_dai_driver dai; + int cif_id; + struct clk *clk_i2s; + enum tegra30_ahub_txcif capture_i2s_cif; + enum tegra30_ahub_rxcif capture_fifo_cif; + struct tegra_pcm_dma_params capture_dma_data; + enum tegra30_ahub_rxcif playback_i2s_cif; + enum tegra30_ahub_txcif playback_fifo_cif; + struct tegra_pcm_dma_params playback_dma_data; + struct regmap *regmap; + u32 reg_ctrl; +}; + +#endif -- cgit v1.2.3 From cdc04fd1e982e91936cbcf3dec59a576517d67a1 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Tue, 10 Apr 2012 16:32:01 -0600 Subject: ASoC: tegra: add Kconfig and Makefile support for Tegra30 This adds Kconfig options for the Tegra30 AHUB and I2S controller, and updates the Tegra+WM8903 machine driver Kconfig to select those. Includes a squashed bugfix from Sumit Bhattacharya Signed-off-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/tegra/Kconfig | 18 ++++++++++++++++++ sound/soc/tegra/Makefile | 4 ++++ 2 files changed, 22 insertions(+) (limited to 'sound') diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig index 556cac29ea15..441c317252f9 100644 --- a/sound/soc/tegra/Kconfig +++ b/sound/soc/tegra/Kconfig @@ -30,6 +30,23 @@ config SND_SOC_TEGRA20_SPDIF You will also need to select the individual machine drivers to support below. +config SND_SOC_TEGRA30_AHUB + tristate + depends on SND_SOC_TEGRA && ARCH_TEGRA_3x_SOC + help + Say Y or M if you want to add support for the Tegra20 AHUB module. + You will also need to select the individual machine drivers to + support below. + +config SND_SOC_TEGRA30_I2S + tristate + depends on SND_SOC_TEGRA && ARCH_TEGRA_3x_SOC + select SND_SOC_TEGRA30_AHUB + help + Say Y or M if you want to add support for codecs attached to the + Tegra30 I2S interface. You will also need to select the individual + machine drivers to support below. + config MACH_HAS_SND_SOC_TEGRA_WM8903 bool help @@ -42,6 +59,7 @@ config SND_SOC_TEGRA_WM8903 depends on SND_SOC_TEGRA && I2C depends on MACH_HAS_SND_SOC_TEGRA_WM8903 select SND_SOC_TEGRA20_I2S if ARCH_TEGRA_2x_SOC + select SND_SOC_TEGRA30_I2S if ARCH_TEGRA_3x_SOC select SND_SOC_WM8903 help Say Y or M here if you want to add support for SoC audio on Tegra diff --git a/sound/soc/tegra/Makefile b/sound/soc/tegra/Makefile index 4726b909664d..98704b48b62a 100644 --- a/sound/soc/tegra/Makefile +++ b/sound/soc/tegra/Makefile @@ -4,12 +4,16 @@ snd-soc-tegra-utils-objs += tegra_asoc_utils.o snd-soc-tegra20-das-objs := tegra20_das.o snd-soc-tegra20-i2s-objs := tegra20_i2s.o snd-soc-tegra20-spdif-objs := tegra20_spdif.o +snd-soc-tegra30-ahub-objs := tegra30_ahub.o +snd-soc-tegra30-i2s-objs := tegra30_i2s.o obj-$(CONFIG_SND_SOC_TEGRA) += snd-soc-tegra-pcm.o obj-$(CONFIG_SND_SOC_TEGRA) += snd-soc-tegra-utils.o obj-$(CONFIG_SND_SOC_TEGRA20_DAS) += snd-soc-tegra20-das.o obj-$(CONFIG_SND_SOC_TEGRA20_I2S) += snd-soc-tegra20-i2s.o obj-$(CONFIG_SND_SOC_TEGRA20_SPDIF) += snd-soc-tegra20-spdif.o +obj-$(CONFIG_SND_SOC_TEGRA30_AHUB) += snd-soc-tegra30-ahub.o +obj-$(CONFIG_SND_SOC_TEGRA30_I2S) += snd-soc-tegra30-i2s.o # Tegra machine Support snd-soc-tegra-wm8903-objs := tegra_wm8903.o -- cgit v1.2.3 From f2390880ec0264a0ed26b32c23bc23435b4297da Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Sun, 8 Apr 2012 21:17:50 -0700 Subject: ASoC: add generic simple-card support Current ASoC requires card.c file to each platforms in order to specifies its CPU and Codecs pair. But the differences between these were only value/strings of setting. In order to reduce duplicate driver, this patch adds generic/simple-card. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/simple_card.h | 38 ++++++++++++++ sound/soc/Kconfig | 3 ++ sound/soc/Makefile | 1 + sound/soc/generic/Kconfig | 4 ++ sound/soc/generic/Makefile | 3 ++ sound/soc/generic/simple-card.c | 114 ++++++++++++++++++++++++++++++++++++++++ 6 files changed, 163 insertions(+) create mode 100644 include/sound/simple_card.h create mode 100644 sound/soc/generic/Kconfig create mode 100644 sound/soc/generic/Makefile create mode 100644 sound/soc/generic/simple-card.c (limited to 'sound') diff --git a/include/sound/simple_card.h b/include/sound/simple_card.h new file mode 100644 index 000000000000..4b62b8dc6a4f --- /dev/null +++ b/include/sound/simple_card.h @@ -0,0 +1,38 @@ +/* + * ASoC simple sound card support + * + * Copyright (C) 2012 Renesas Solutions Corp. + * Kuninori Morimoto + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __SIMPLE_CARD_H +#define __SIMPLE_CARD_H + +#include + +struct asoc_simple_dai_init_info { + unsigned int fmt; + unsigned int cpu_daifmt; + unsigned int codec_daifmt; + unsigned int sysclk; +}; + +struct asoc_simple_card_info { + const char *name; + const char *card; + const char *cpu_dai; + const char *codec; + const char *platform; + const char *codec_dai; + struct asoc_simple_dai_init_info *init; /* for snd_link.init */ + + /* used in simple-card.c */ + struct snd_soc_dai_link snd_link; + struct snd_soc_card snd_card; +}; + +#endif /* __SIMPLE_CARD_H */ diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 0f85f6d526e0..a7df779894d5 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -51,5 +51,8 @@ source "sound/soc/txx9/Kconfig" # Supported codecs source "sound/soc/codecs/Kconfig" +# generic frame-work +source "sound/soc/generic/Kconfig" + endif # SND_SOC diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 363dfd6cffe7..57d3463ce9d6 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -6,6 +6,7 @@ obj-$(CONFIG_SND_SOC_DMAENGINE_PCM) += snd-soc-dmaengine-pcm.o obj-$(CONFIG_SND_SOC) += snd-soc-core.o obj-$(CONFIG_SND_SOC) += codecs/ +obj-$(CONFIG_SND_SOC) += generic/ obj-$(CONFIG_SND_SOC) += atmel/ obj-$(CONFIG_SND_SOC) += au1x/ obj-$(CONFIG_SND_SOC) += blackfin/ diff --git a/sound/soc/generic/Kconfig b/sound/soc/generic/Kconfig new file mode 100644 index 000000000000..610f61251640 --- /dev/null +++ b/sound/soc/generic/Kconfig @@ -0,0 +1,4 @@ +config SND_SIMPLE_CARD + tristate "ASoC Simple sound card support" + help + This option enables generic simple sound card support diff --git a/sound/soc/generic/Makefile b/sound/soc/generic/Makefile new file mode 100644 index 000000000000..9c3b246792bf --- /dev/null +++ b/sound/soc/generic/Makefile @@ -0,0 +1,3 @@ +snd-soc-simple-card-objs := simple-card.o + +obj-$(CONFIG_SND_SIMPLE_CARD) += snd-soc-simple-card.o diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c new file mode 100644 index 000000000000..b4b4cab30232 --- /dev/null +++ b/sound/soc/generic/simple-card.c @@ -0,0 +1,114 @@ +/* + * ASoC simple sound card support + * + * Copyright (C) 2012 Renesas Solutions Corp. + * Kuninori Morimoto + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include +#include +#include + +#define asoc_simple_get_card_info(p) \ + container_of(p->dai_link, struct asoc_simple_card_info, snd_link) + +static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd) +{ + struct asoc_simple_card_info *cinfo = asoc_simple_get_card_info(rtd); + struct asoc_simple_dai_init_info *iinfo = cinfo->init; + struct snd_soc_dai *codec = rtd->codec_dai; + struct snd_soc_dai *cpu = rtd->cpu_dai; + unsigned int cpu_daifmt = iinfo->fmt | iinfo->cpu_daifmt; + unsigned int codec_daifmt = iinfo->fmt | iinfo->codec_daifmt; + int ret; + + if (codec_daifmt) { + ret = snd_soc_dai_set_fmt(codec, codec_daifmt); + if (ret < 0) + return ret; + } + + if (iinfo->sysclk) { + ret = snd_soc_dai_set_sysclk(codec, 0, iinfo->sysclk, 0); + if (ret < 0) + return ret; + } + + if (cpu_daifmt) { + ret = snd_soc_dai_set_fmt(cpu, cpu_daifmt); + if (ret < 0) + return ret; + } + + return 0; +} + +static int asoc_simple_card_probe(struct platform_device *pdev) +{ + struct asoc_simple_card_info *cinfo = pdev->dev.platform_data; + + if (!cinfo) { + dev_err(&pdev->dev, "no info for asoc-simple-card\n"); + return -EINVAL; + } + + if (!cinfo->name || + !cinfo->card || + !cinfo->cpu_dai || + !cinfo->codec || + !cinfo->platform || + !cinfo->codec_dai) { + dev_err(&pdev->dev, "insufficient asoc_simple_card_info settings\n"); + return -EINVAL; + } + + /* + * init snd_soc_dai_link + */ + cinfo->snd_link.name = cinfo->name; + cinfo->snd_link.stream_name = cinfo->name; + cinfo->snd_link.cpu_dai_name = cinfo->cpu_dai; + cinfo->snd_link.platform_name = cinfo->platform; + cinfo->snd_link.codec_name = cinfo->codec; + cinfo->snd_link.codec_dai_name = cinfo->codec_dai; + + /* enable snd_link.init if cinfo has settings */ + if (cinfo->init) + cinfo->snd_link.init = asoc_simple_card_dai_init; + + /* + * init snd_soc_card + */ + cinfo->snd_card.name = cinfo->card; + cinfo->snd_card.owner = THIS_MODULE; + cinfo->snd_card.dai_link = &cinfo->snd_link; + cinfo->snd_card.num_links = 1; + cinfo->snd_card.dev = &pdev->dev; + + return snd_soc_register_card(&cinfo->snd_card); +} + +static int asoc_simple_card_remove(struct platform_device *pdev) +{ + struct asoc_simple_card_info *cinfo = pdev->dev.platform_data; + + return snd_soc_unregister_card(&cinfo->snd_card); +} + +static struct platform_driver asoc_simple_card = { + .driver = { + .name = "asoc-simple-card", + }, + .probe = asoc_simple_card_probe, + .remove = asoc_simple_card_remove, +}; + +module_platform_driver(asoc_simple_card); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("ASoC Simple Sound Card"); +MODULE_AUTHOR("Kuninori Morimoto "); -- cgit v1.2.3 From af8a2fe12fae1b59178dc96e396e5665bcbea7da Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Sun, 8 Apr 2012 21:18:28 -0700 Subject: ASoC: sh: fsi: use simple-card instead of fsi-ak4642 This patch uses simple-card driver instead of fsi-ak4642 on each board. To select AK4642 driver, each boards select it on Kconfig. This patch removes fsi-ak4642 driver which is no longer needed Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- arch/arm/mach-shmobile/Kconfig | 2 + arch/arm/mach-shmobile/board-ap4evb.c | 15 ++++- arch/arm/mach-shmobile/board-mackerel.c | 15 ++++- arch/sh/boards/Kconfig | 1 + arch/sh/boards/mach-se/7724/setup.c | 15 ++++- include/sound/sh_fsi.h | 12 ---- sound/soc/sh/Kconfig | 8 --- sound/soc/sh/Makefile | 2 - sound/soc/sh/fsi-ak4642.c | 108 -------------------------------- 9 files changed, 39 insertions(+), 139 deletions(-) delete mode 100644 sound/soc/sh/fsi-ak4642.c (limited to 'sound') diff --git a/arch/arm/mach-shmobile/Kconfig b/arch/arm/mach-shmobile/Kconfig index 34560cab45d9..2cda0c2af230 100644 --- a/arch/arm/mach-shmobile/Kconfig +++ b/arch/arm/mach-shmobile/Kconfig @@ -58,6 +58,7 @@ config MACH_AP4EVB depends on ARCH_SH7372 select ARCH_REQUIRE_GPIOLIB select SH_LCD_MIPI_DSI + select SND_SOC_AK4642 if SND_SIMPLE_CARD choice prompt "AP4EVB LCD panel selection" @@ -82,6 +83,7 @@ config MACH_MACKEREL bool "mackerel board" depends on ARCH_SH7372 select ARCH_REQUIRE_GPIOLIB + select SND_SOC_AK4642 if SND_SIMPLE_CARD config MACH_KOTA2 bool "KOTA2 board" diff --git a/arch/arm/mach-shmobile/board-ap4evb.c b/arch/arm/mach-shmobile/board-ap4evb.c index b56dde2732bb..b39751244daa 100644 --- a/arch/arm/mach-shmobile/board-ap4evb.c +++ b/arch/arm/mach-shmobile/board-ap4evb.c @@ -50,6 +50,7 @@ #include #include +#include #include