From 19441e35a43b616ea6afad91ed0d9e77268d8f6a Mon Sep 17 00:00:00 2001 From: Olivier Moysan Date: Mon, 4 Mar 2019 15:52:43 +0100 Subject: ASoC: stm32: dfsdm: manage multiple prepare The DFSDM must be stopped when a new setting is applied. restart systematically DFSDM on multiple prepare calls, to apply changes. Signed-off-by: Olivier Moysan Signed-off-by: Mark Brown --- sound/soc/stm/stm32_adfsdm.c | 17 ++++++++++++++++- 1 file changed, 16 insertions(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/stm/stm32_adfsdm.c b/sound/soc/stm/stm32_adfsdm.c index 47901983a6ff..d77f0421e4f0 100644 --- a/sound/soc/stm/stm32_adfsdm.c +++ b/sound/soc/stm/stm32_adfsdm.c @@ -9,6 +9,7 @@ #include #include +#include #include #include @@ -37,6 +38,8 @@ struct stm32_adfsdm_priv { /* PCM buffer */ unsigned char *pcm_buff; unsigned int pos; + + struct mutex lock; /* protect against race condition on iio state */ }; static const struct snd_pcm_hardware stm32_adfsdm_pcm_hw = { @@ -62,10 +65,12 @@ static void stm32_adfsdm_shutdown(struct snd_pcm_substream *substream, { struct stm32_adfsdm_priv *priv = snd_soc_dai_get_drvdata(dai); + mutex_lock(&priv->lock); if (priv->iio_active) { iio_channel_stop_all_cb(priv->iio_cb); priv->iio_active = false; } + mutex_unlock(&priv->lock); } static int stm32_adfsdm_dai_prepare(struct snd_pcm_substream *substream, @@ -74,13 +79,19 @@ static int stm32_adfsdm_dai_prepare(struct snd_pcm_substream *substream, struct stm32_adfsdm_priv *priv = snd_soc_dai_get_drvdata(dai); int ret; + mutex_lock(&priv->lock); + if (priv->iio_active) { + iio_channel_stop_all_cb(priv->iio_cb); + priv->iio_active = false; + } + ret = iio_write_channel_attribute(priv->iio_ch, substream->runtime->rate, 0, IIO_CHAN_INFO_SAMP_FREQ); if (ret < 0) { dev_err(dai->dev, "%s: Failed to set %d sampling rate\n", __func__, substream->runtime->rate); - return ret; + goto out; } if (!priv->iio_active) { @@ -92,6 +103,9 @@ static int stm32_adfsdm_dai_prepare(struct snd_pcm_substream *substream, __func__, ret); } +out: + mutex_unlock(&priv->lock); + return ret; } @@ -299,6 +313,7 @@ static int stm32_adfsdm_probe(struct platform_device *pdev) priv->dev = &pdev->dev; priv->dai_drv = stm32_adfsdm_dai; + mutex_init(&priv->lock); dev_set_drvdata(&pdev->dev, priv); -- cgit v1.2.3 From c47255b61129857b74b0d86eaf59335348be05e0 Mon Sep 17 00:00:00 2001 From: Olivier Moysan Date: Mon, 4 Mar 2019 15:52:44 +0100 Subject: ASoC: stm32: dfsdm: fix debugfs warnings on entry creation Register platform component with a prefix, to avoid warnings on debugfs entries creation, due to component name redundancy. Signed-off-by: Olivier Moysan Signed-off-by: Mark Brown --- sound/soc/stm/stm32_adfsdm.c | 21 ++++++++++++++++++--- 1 file changed, 18 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/stm/stm32_adfsdm.c b/sound/soc/stm/stm32_adfsdm.c index d77f0421e4f0..78bed9734713 100644 --- a/sound/soc/stm/stm32_adfsdm.c +++ b/sound/soc/stm/stm32_adfsdm.c @@ -305,6 +305,7 @@ MODULE_DEVICE_TABLE(of, stm32_adfsdm_of_match); static int stm32_adfsdm_probe(struct platform_device *pdev) { struct stm32_adfsdm_priv *priv; + struct snd_soc_component *component; int ret; priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); @@ -332,9 +333,15 @@ static int stm32_adfsdm_probe(struct platform_device *pdev) if (IS_ERR(priv->iio_cb)) return PTR_ERR(priv->iio_cb); - ret = devm_snd_soc_register_component(&pdev->dev, - &stm32_adfsdm_soc_platform, - NULL, 0); + component = devm_kzalloc(&pdev->dev, sizeof(*component), GFP_KERNEL); + if (!component) + return -ENOMEM; +#ifdef CONFIG_DEBUG_FS + component->debugfs_prefix = "pcm"; +#endif + + ret = snd_soc_add_component(&pdev->dev, component, + &stm32_adfsdm_soc_platform, NULL, 0); if (ret < 0) dev_err(&pdev->dev, "%s: Failed to register PCM platform\n", __func__); @@ -342,12 +349,20 @@ static int stm32_adfsdm_probe(struct platform_device *pdev) return ret; } +static int stm32_adfsdm_remove(struct platform_device *pdev) +{ + snd_soc_unregister_component(&pdev->dev); + + return 0; +} + static struct platform_driver stm32_adfsdm_driver = { .driver = { .name = STM32_ADFSDM_DRV_NAME, .of_match_table = stm32_adfsdm_of_match, }, .probe = stm32_adfsdm_probe, + .remove = stm32_adfsdm_remove, }; module_platform_driver(stm32_adfsdm_driver); -- cgit v1.2.3 From f0f2338a9cfaf71db895fa989ea7234e8a9b471d Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Wed, 20 Mar 2019 22:41:56 +0100 Subject: ASoC: cs4270: Set auto-increment bit for register writes The CS4270 does not by default increment the register address on consecutive writes. During normal operation it doesn't matter as all register accesses are done individually. At resume time after suspend, however, the regcache code gathers the biggest possible block of registers to sync and sends them one on one go. To fix this, set the INCR bit in all cases. Signed-off-by: Daniel Mack Signed-off-by: Mark Brown --- sound/soc/codecs/cs4270.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 33d74f163bd7..793a14d58667 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -642,6 +642,7 @@ static const struct regmap_config cs4270_regmap = { .reg_defaults = cs4270_reg_defaults, .num_reg_defaults = ARRAY_SIZE(cs4270_reg_defaults), .cache_type = REGCACHE_RBTREE, + .write_flag_mask = CS4270_I2C_INCR, .readable_reg = cs4270_reg_is_readable, .volatile_reg = cs4270_reg_is_volatile, -- cgit v1.2.3 From 53f67a78663811968f426d480bc55887d787bd94 Mon Sep 17 00:00:00 2001 From: "S.j. Wang" Date: Sat, 2 Mar 2019 05:52:19 +0000 Subject: ASoC: fsl_asrc: add constraint for the asrc of older version There is a constraint for the channel number setting on the asrc of older version (e.g. imx35), the channel number should be even, odd number isn't valid. So add this constraint when the asrc of older version is used. Acked-by: Nicolin Chen Signed-off-by: Shengjiu Wang Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_asrc.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c index 528e8b108422..0b937924d2e4 100644 --- a/sound/soc/fsl/fsl_asrc.c +++ b/sound/soc/fsl/fsl_asrc.c @@ -445,6 +445,19 @@ struct dma_chan *fsl_asrc_get_dma_channel(struct fsl_asrc_pair *pair, bool dir) } EXPORT_SYMBOL_GPL(fsl_asrc_get_dma_channel); +static int fsl_asrc_dai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct fsl_asrc *asrc_priv = snd_soc_dai_get_drvdata(dai); + + /* Odd channel number is not valid for older ASRC (channel_bits==3) */ + if (asrc_priv->channel_bits == 3) + snd_pcm_hw_constraint_step(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_CHANNELS, 2); + + return 0; +} + static int fsl_asrc_dai_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -539,6 +552,7 @@ static int fsl_asrc_dai_trigger(struct snd_pcm_substream *substream, int cmd, } static const struct snd_soc_dai_ops fsl_asrc_dai_ops = { + .startup = fsl_asrc_dai_startup, .hw_params = fsl_asrc_dai_hw_params, .hw_free = fsl_asrc_dai_hw_free, .trigger = fsl_asrc_dai_trigger, -- cgit v1.2.3 From 0ff4e8c61b794a4bf6c854ab071a1abaaa80f358 Mon Sep 17 00:00:00 2001 From: "S.j. Wang" Date: Wed, 27 Feb 2019 06:31:12 +0000 Subject: ASoC: fsl_esai: fix channel swap issue when stream starts There is very low possibility ( < 0.1% ) that channel swap happened in beginning when multi output/input pin is enabled. The issue is that hardware can't send data to correct pin in the beginning with the normal enable flow. This is hardware issue, but there is no errata, the workaround flow is that: Each time playback/recording, firstly clear the xSMA/xSMB, then enable TE/RE, then enable xSMB and xSMA (xSMB must be enabled before xSMA). Which is to use the xSMA as the trigger start register, previously the xCR_TE or xCR_RE is the bit for starting. Fixes commit 43d24e76b698 ("ASoC: fsl_esai: Add ESAI CPU DAI driver") Cc: Reviewed-by: Fabio Estevam Acked-by: Nicolin Chen Signed-off-by: Shengjiu Wang Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_esai.c | 47 +++++++++++++++++++++++++++++++++++++---------- 1 file changed, 37 insertions(+), 10 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index afe67c865330..3623aa9a6f2e 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -54,6 +54,8 @@ struct fsl_esai { u32 fifo_depth; u32 slot_width; u32 slots; + u32 tx_mask; + u32 rx_mask; u32 hck_rate[2]; u32 sck_rate[2]; bool hck_dir[2]; @@ -361,21 +363,13 @@ static int fsl_esai_set_dai_tdm_slot(struct snd_soc_dai *dai, u32 tx_mask, regmap_update_bits(esai_priv->regmap, REG_ESAI_TCCR, ESAI_xCCR_xDC_MASK, ESAI_xCCR_xDC(slots)); - regmap_update_bits(esai_priv->regmap, REG_ESAI_TSMA, - ESAI_xSMA_xS_MASK, ESAI_xSMA_xS(tx_mask)); - regmap_update_bits(esai_priv->regmap, REG_ESAI_TSMB, - ESAI_xSMB_xS_MASK, ESAI_xSMB_xS(tx_mask)); - regmap_update_bits(esai_priv->regmap, REG_ESAI_RCCR, ESAI_xCCR_xDC_MASK, ESAI_xCCR_xDC(slots)); - regmap_update_bits(esai_priv->regmap, REG_ESAI_RSMA, - ESAI_xSMA_xS_MASK, ESAI_xSMA_xS(rx_mask)); - regmap_update_bits(esai_priv->regmap, REG_ESAI_RSMB, - ESAI_xSMB_xS_MASK, ESAI_xSMB_xS(rx_mask)); - esai_priv->slot_width = slot_width; esai_priv->slots = slots; + esai_priv->tx_mask = tx_mask; + esai_priv->rx_mask = rx_mask; return 0; } @@ -596,6 +590,7 @@ static int fsl_esai_trigger(struct snd_pcm_substream *substream, int cmd, bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; u8 i, channels = substream->runtime->channels; u32 pins = DIV_ROUND_UP(channels, esai_priv->slots); + u32 mask; switch (cmd) { case SNDRV_PCM_TRIGGER_START: @@ -608,15 +603,38 @@ static int fsl_esai_trigger(struct snd_pcm_substream *substream, int cmd, for (i = 0; tx && i < channels; i++) regmap_write(esai_priv->regmap, REG_ESAI_ETDR, 0x0); + /* + * When set the TE/RE in the end of enablement flow, there + * will be channel swap issue for multi data line case. + * In order to workaround this issue, we switch the bit + * enablement sequence to below sequence + * 1) clear the xSMB & xSMA: which is done in probe and + * stop state. + * 2) set TE/RE + * 3) set xSMB + * 4) set xSMA: xSMA is the last one in this flow, which + * will trigger esai to start. + */ regmap_update_bits(esai_priv->regmap, REG_ESAI_xCR(tx), tx ? ESAI_xCR_TE_MASK : ESAI_xCR_RE_MASK, tx ? ESAI_xCR_TE(pins) : ESAI_xCR_RE(pins)); + mask = tx ? esai_priv->tx_mask : esai_priv->rx_mask; + + regmap_update_bits(esai_priv->regmap, REG_ESAI_xSMB(tx), + ESAI_xSMB_xS_MASK, ESAI_xSMB_xS(mask)); + regmap_update_bits(esai_priv->regmap, REG_ESAI_xSMA(tx), + ESAI_xSMA_xS_MASK, ESAI_xSMA_xS(mask)); + break; case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: regmap_update_bits(esai_priv->regmap, REG_ESAI_xCR(tx), tx ? ESAI_xCR_TE_MASK : ESAI_xCR_RE_MASK, 0); + regmap_update_bits(esai_priv->regmap, REG_ESAI_xSMA(tx), + ESAI_xSMA_xS_MASK, 0); + regmap_update_bits(esai_priv->regmap, REG_ESAI_xSMB(tx), + ESAI_xSMB_xS_MASK, 0); /* Disable and reset FIFO */ regmap_update_bits(esai_priv->regmap, REG_ESAI_xFCR(tx), @@ -906,6 +924,15 @@ static int fsl_esai_probe(struct platform_device *pdev) return ret; } + esai_priv->tx_mask = 0xFFFFFFFF; + esai_priv->rx_mask = 0xFFFFFFFF; + + /* Clear the TSMA, TSMB, RSMA, RSMB */ + regmap_write(esai_priv->regmap, REG_ESAI_TSMA, 0); + regmap_write(esai_priv->regmap, REG_ESAI_TSMB, 0); + regmap_write(esai_priv->regmap, REG_ESAI_RSMA, 0); + regmap_write(esai_priv->regmap, REG_ESAI_RSMB, 0); + ret = devm_snd_soc_register_component(&pdev->dev, &fsl_esai_component, &fsl_esai_dai, 1); if (ret) { -- cgit v1.2.3 From 8f71370f4b02730e8c27faf460af7a3586e24e1f Mon Sep 17 00:00:00 2001 From: Guenter Roeck Date: Fri, 22 Mar 2019 15:39:48 -0700 Subject: ASoC: intel: Fix crash at suspend/resume after failed codec registration If codec registration fails after the ASoC Intel SST driver has been probed, the kernel will Oops and crash at suspend/resume. general protection fault: 0000 [#1] PREEMPT SMP KASAN PTI CPU: 1 PID: 2811 Comm: cat Tainted: G W 4.19.30 #15 Hardware name: GOOGLE Clapper, BIOS Google_Clapper.5216.199.7 08/22/2014 RIP: 0010:snd_soc_suspend+0x5a/0xd21 Code: 03 80 3c 10 00 49 89 d7 74 0b 48 89 df e8 71 72 c4 fe 4c 89 fa 48 8b 03 48 89 45 d0 48 8d 98 a0 01 00 00 48 89 d8 48 c1 e8 03 <8a> 04 10 84 c0 0f 85 85 0c 00 00 80 3b 00 0f 84 6b 0c 00 00 48 8b RSP: 0018:ffff888035407750 EFLAGS: 00010202 RAX: 0000000000000034 RBX: 00000000000001a0 RCX: 0000000000000000 RDX: dffffc0000000000 RSI: 0000000000000008 RDI: ffff88805c417098 RBP: ffff8880354077b0 R08: dffffc0000000000 R09: ffffed100b975718 R10: 0000000000000001 R11: ffffffff949ea4a3 R12: 1ffff1100b975746 R13: dffffc0000000000 R14: ffff88805cba4588 R15: dffffc0000000000 FS: 0000794a78e91b80(0000) GS:ffff888068d00000(0000) knlGS:0000000000000000 CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 CR2: 00007bd5283ccf58 CR3: 000000004b7aa000 CR4: 00000000001006e0 Call Trace: ? dpm_complete+0x67b/0x67b ? i915_gem_suspend+0x14d/0x1ad sst_soc_prepare+0x91/0x1dd ? sst_be_hw_params+0x7e/0x7e dpm_prepare+0x39a/0x88b dpm_suspend_start+0x13/0x9d suspend_devices_and_enter+0x18f/0xbd7 ? arch_suspend_enable_irqs+0x11/0x11 ? printk+0xd9/0x12d ? lock_release+0x95f/0x95f ? log_buf_vmcoreinfo_setup+0x131/0x131 ? rcu_read_lock_sched_held+0x140/0x22a ? __bpf_trace_rcu_utilization+0xa/0xa ? __pm_pr_dbg+0x186/0x190 ? pm_notifier_call_chain+0x39/0x39 ? suspend_test+0x9d/0x9d pm_suspend+0x2f4/0x728 ? trace_suspend_resume+0x3da/0x3da ? lock_release+0x95f/0x95f ? kernfs_fop_write+0x19f/0x32d state_store+0xd8/0x147 ? sysfs_kf_read+0x155/0x155 kernfs_fop_write+0x23e/0x32d __vfs_write+0x108/0x608 ? vfs_read+0x2e9/0x2e9 ? rcu_read_lock_sched_held+0x140/0x22a ? __bpf_trace_rcu_utilization+0xa/0xa ? debug_smp_processor_id+0x10/0x10 ? selinux_file_permission+0x1c5/0x3c8 ? rcu_sync_lockdep_assert+0x6a/0xad ? __sb_start_write+0x129/0x2ac vfs_write+0x1aa/0x434 ksys_write+0xfe/0x1be ? __ia32_sys_read+0x82/0x82 do_syscall_64+0xcd/0x120 entry_SYSCALL_64_after_hwframe+0x49/0xbe In the observed situation, the problem is seen because the codec driver failed to probe due to a hardware problem. max98090 i2c-193C9890:00: Failed to read device revision: -1 max98090 i2c-193C9890:00: ASoC: failed to probe component -1 cht-bsw-max98090 cht-bsw-max98090: ASoC: failed to instantiate card -1 cht-bsw-max98090 cht-bsw-max98090: snd_soc_register_card failed -1 cht-bsw-max98090: probe of cht-bsw-max98090 failed with error -1 The problem is similar to the problem solved with commit 2fc995a87f2e ("ASoC: intel: Fix crash at suspend/resume without card registration"), but codec registration fails at a later point. At that time, the pointer checked with the above mentioned commit is already set, but it is not cleared if the device is subsequently removed. Adding a remove function to clear the pointer fixes the problem. Cc: stable@vger.kernel.org Cc: Jarkko Nikula Cc: Curtis Malainey Signed-off-by: Guenter Roeck Acked-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 8 ++++++++ 1 file changed, 8 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c index 08cea5b5cda9..0e8b1c5eec88 100644 --- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c @@ -706,9 +706,17 @@ static int sst_soc_probe(struct snd_soc_component *component) return sst_dsp_init_v2_dpcm(component); } +static void sst_soc_remove(struct snd_soc_component *component) +{ + struct sst_data *drv = dev_get_drvdata(component->dev); + + drv->soc_card = NULL; +} + static const struct snd_soc_component_driver sst_soc_platform_drv = { .name = DRV_NAME, .probe = sst_soc_probe, + .remove = sst_soc_remove, .ops = &sst_platform_ops, .compr_ops = &sst_platform_compr_ops, .pcm_new = sst_pcm_new, -- cgit v1.2.3 From cacea3a90e211f0c111975535508d446a4a928d2 Mon Sep 17 00:00:00 2001 From: Pankaj Bharadiya Date: Fri, 22 Mar 2019 18:00:09 +0530 Subject: ASoC: dapm: Fix NULL pointer dereference in snd_soc_dapm_free_kcontrol w_text_param can be NULL and it is being dereferenced without checking. Add the missing sanity check to prevent NULL pointer dereference. Signed-off-by: Pankaj Bharadiya Acked-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 1ec06ef6d161..67b032ca1601 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3957,6 +3957,10 @@ snd_soc_dapm_free_kcontrol(struct snd_soc_card *card, int count; devm_kfree(card->dev, (void *)*private_value); + + if (!w_param_text) + return; + for (count = 0 ; count < num_params; count++) devm_kfree(card->dev, (void *)w_param_text[count]); devm_kfree(card->dev, w_param_text); -- cgit v1.2.3 From 36e075ce74ec4e261a638bf09d10b3348ca4d883 Mon Sep 17 00:00:00 2001 From: Jenny TC Date: Sat, 23 Mar 2019 18:40:10 +0530 Subject: ASoC: Intel: Skylake: enable S24_LE format support To enable S24_LE format, sample_type in topology fw has to be set to 1. But sample_type defined in topology firmware configuration is not getting reflected in the dsp param. This patch sets sample_type in base config so that the sample type defined in the topology firmware is reflected in the dsp params. This issues was uncovered while debugging the S24_LE format which require the MSB byte in 32 bit word to be skipped. Setting sample_type in topology firmware to 1 helps to skip MSB byte word. Signed-off-by: Jenny TC Acked-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/skylake/skl-messages.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/soc') diff --git a/sound/soc/intel/skylake/skl-messages.c b/sound/soc/intel/skylake/skl-messages.c index 28c4806b196a..4bf70b4429f0 100644 --- a/sound/soc/intel/skylake/skl-messages.c +++ b/sound/soc/intel/skylake/skl-messages.c @@ -483,6 +483,7 @@ static void skl_set_base_module_format(struct skl_sst *ctx, base_cfg->audio_fmt.bit_depth = format->bit_depth; base_cfg->audio_fmt.valid_bit_depth = format->valid_bit_depth; base_cfg->audio_fmt.ch_cfg = format->ch_cfg; + base_cfg->audio_fmt.sample_type = format->sample_type; dev_dbg(ctx->dev, "bit_depth=%x valid_bd=%x ch_config=%x\n", format->bit_depth, format->valid_bit_depth, -- cgit v1.2.3 From c63adb28f6d913310430f14c69f0a2ea55eed0cc Mon Sep 17 00:00:00 2001 From: Annaliese McDermond Date: Sat, 30 Mar 2019 09:02:02 -0700 Subject: ASoC: tlv320aic32x4: Fix Common Pins The common pins were mistakenly not added to the DAPM graph. Adding these pins will allow valid graphs to be created. Signed-off-by: Annaliese McDermond Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic32x4.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index 96f1526cb258..5520044929f4 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -490,6 +490,8 @@ static const struct snd_soc_dapm_widget aic32x4_dapm_widgets[] = { SND_SOC_DAPM_INPUT("IN2_R"), SND_SOC_DAPM_INPUT("IN3_L"), SND_SOC_DAPM_INPUT("IN3_R"), + SND_SOC_DAPM_INPUT("CM_L"), + SND_SOC_DAPM_INPUT("CM_R"), }; static const struct snd_soc_dapm_route aic32x4_dapm_routes[] = { -- cgit v1.2.3 From 6246f283d5e02ac757bd8d9bacde8fdc54c4582d Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Mon, 1 Apr 2019 15:03:54 +0200 Subject: ASoC: dpcm: skip missing substream while applying symmetry If for any reason, the backend does not have the requested substream (like capture on a playback only backend), the BE will be skipped in dpcm_be_dai_startup(). However, dpcm_apply_symmetry() does not skip those BE and will dereference the be_substream (NULL) pointer anyway. Like in dpcm_be_dai_startup(), just skip those BE. Fixes: 906c7d690c3b ("ASoC: dpcm: Apply symmetry for DPCM") Signed-off-by: Jerome Brunet Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 7fe5321000e8..2d5d5cac4ba6 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1911,10 +1911,15 @@ static int dpcm_apply_symmetry(struct snd_pcm_substream *fe_substream, struct snd_soc_pcm_runtime *be = dpcm->be; struct snd_pcm_substream *be_substream = snd_soc_dpcm_get_substream(be, stream); - struct snd_soc_pcm_runtime *rtd = be_substream->private_data; + struct snd_soc_pcm_runtime *rtd; struct snd_soc_dai *codec_dai; int i; + /* A backend may not have the requested substream */ + if (!be_substream) + continue; + + rtd = be_substream->private_data; if (rtd->dai_link->be_hw_params_fixup) continue; -- cgit v1.2.3 From 6e3bfcff191ec9476ca5ef9b2ad85a15ba829374 Mon Sep 17 00:00:00 2001 From: Ranjani Sridharan Date: Fri, 1 Mar 2019 19:08:53 -0600 Subject: ASoC: dapm: set power_check callback for widgets that shouldnt be always on Currently, buffers, schedulers, src's, encoders, decoders and effect type dapm widgets remain always on as their power_check method is not set. Setting this callback allows these widgets in the audio path to be powered managed properly. Signed-off-by: Ranjani Sridharan Signed-off-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 7 +++++++ 1 file changed, 7 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 67b032ca1601..0382a47b30bd 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3650,6 +3650,13 @@ snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm, case snd_soc_dapm_dac: case snd_soc_dapm_aif_in: case snd_soc_dapm_pga: + case snd_soc_dapm_buffer: + case snd_soc_dapm_scheduler: + case snd_soc_dapm_effect: + case snd_soc_dapm_src: + case snd_soc_dapm_asrc: + case snd_soc_dapm_encoder: + case snd_soc_dapm_decoder: case snd_soc_dapm_out_drv: case snd_soc_dapm_micbias: case snd_soc_dapm_line: -- cgit v1.2.3 From 43d147be5738a9ed6cfb25c285ac50d6dd5793be Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 2 Apr 2019 13:49:14 +0100 Subject: ASoC: wm_adsp: Check for buffer in trigger stop Trigger stop can be called in situations where trigger start failed and as such it can't be assumed the buffer is already attached to the compressed stream or a NULL pointer may be dereferenced. Fixes: 639e5eb3c7d6 ("ASoC: wm_adsp: Correct handling of compressed streams that restart") Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 5608ed5decca..b0b48eb9c7c9 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -3587,7 +3587,8 @@ int wm_adsp_compr_trigger(struct snd_compr_stream *stream, int cmd) } break; case SNDRV_PCM_TRIGGER_STOP: - wm_adsp_buffer_clear(compr->buf); + if (wm_adsp_compr_attached(compr)) + wm_adsp_buffer_clear(compr->buf); break; default: ret = -EINVAL; -- cgit v1.2.3 From 4bcdec39c454c4e8f9512115bdcc3efec1ba5f55 Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Tue, 2 Apr 2019 12:20:49 +0200 Subject: ASoC: Intel: cht_bsw_max98090_ti: Enable codec clock once and keep it enabled Users have been seeing sound stability issues with max98090 codecs since: commit 648e921888ad ("clk: x86: Stop marking clocks as CLK_IS_CRITICAL") At first that commit broke sound for Chromebook Swanky and Clapper models, the problem was that the machine-driver has been controlling the wrong clock on those models since support for them was added. This was hidden by clk-pmc-atom.c keeping the actual clk on unconditionally. With the machine-driver controlling the proper clock, sound works again but we are seeing bug reports describing it as: low volume, "sounds like played at 10x speed" and instable. When these issues are hit the following message is seen in dmesg: "max98090 i2c-193C9890:00: PLL unlocked". Attempts have been made to fix this by inserting a delay between enabling the clk and enabling and checking the pll, but this has not helped. It seems that at least on boards which use pmc_plt_clk_0 as clock, if we ever disable the clk, the pll looses its lock and after that we get various issues. This commit fixes this by enabling the clock once at probe time on these boards. In essence this restores the old behavior of clk-pmc-atom.c always keeping the clk on on these boards. Fixes: 648e921888ad ("clk: x86: Stop marking clocks as CLK_IS_CRITICAL") Reported-by: Mogens Jensen Reported-by: Dean Wallace Signed-off-by: Hans de Goede Acked-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 47 ++++++++++++++++++++++++---- 1 file changed, 41 insertions(+), 6 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/intel/boards/cht_bsw_max98090_ti.c b/sound/soc/intel/boards/cht_bsw_max98090_ti.c index 3263b0495853..c0e0844f75b9 100644 --- a/sound/soc/intel/boards/cht_bsw_max98090_ti.c +++ b/sound/soc/intel/boards/cht_bsw_max98090_ti.c @@ -43,6 +43,7 @@ struct cht_mc_private { struct clk *mclk; struct snd_soc_jack jack; bool ts3a227e_present; + int quirks; }; static int platform_clock_control(struct snd_soc_dapm_widget *w, @@ -54,6 +55,10 @@ static int platform_clock_control(struct snd_soc_dapm_widget *w, struct cht_mc_private *ctx = snd_soc_card_get_drvdata(card); int ret; + /* See the comment in snd_cht_mc_probe() */ + if (ctx->quirks & QUIRK_PMC_PLT_CLK_0) + return 0; + codec_dai = snd_soc_card_get_codec_dai(card, CHT_CODEC_DAI); if (!codec_dai) { dev_err(card->dev, "Codec dai not found; Unable to set platform clock\n"); @@ -223,6 +228,10 @@ static int cht_codec_init(struct snd_soc_pcm_runtime *runtime) "jack detection gpios not added, error %d\n", ret); } + /* See the comment in snd_cht_mc_probe() */ + if (ctx->quirks & QUIRK_PMC_PLT_CLK_0) + return 0; + /* * The firmware might enable the clock at * boot (this information may or may not @@ -423,16 +432,15 @@ static int snd_cht_mc_probe(struct platform_device *pdev) const char *mclk_name; struct snd_soc_acpi_mach *mach; const char *platform_name; - int quirks = 0; - - dmi_id = dmi_first_match(cht_max98090_quirk_table); - if (dmi_id) - quirks = (unsigned long)dmi_id->driver_data; drv = devm_kzalloc(&pdev->dev, sizeof(*drv), GFP_KERNEL); if (!drv) return -ENOMEM; + dmi_id = dmi_first_match(cht_max98090_quirk_table); + if (dmi_id) + drv->quirks = (unsigned long)dmi_id->driver_data; + drv->ts3a227e_present = acpi_dev_found("104C227E"); if (!drv->ts3a227e_present) { /* no need probe TI jack detection chip */ @@ -458,7 +466,7 @@ static int snd_cht_mc_probe(struct platform_device *pdev) snd_soc_card_cht.dev = &pdev->dev; snd_soc_card_set_drvdata(&snd_soc_card_cht, drv); - if (quirks & QUIRK_PMC_PLT_CLK_0) + if (drv->quirks & QUIRK_PMC_PLT_CLK_0) mclk_name = "pmc_plt_clk_0"; else mclk_name = "pmc_plt_clk_3"; @@ -471,6 +479,21 @@ static int snd_cht_mc_probe(struct platform_device *pdev) return PTR_ERR(drv->mclk); } + /* + * Boards which have the MAX98090's clk connected to clk_0 do not seem + * to like it if we muck with the clock. If we disable the clock when + * it is unused we get "max98090 i2c-193C9890:00: PLL unlocked" errors + * and the PLL never seems to lock again. + * So for these boards we enable it here once and leave it at that. + */ + if (drv->quirks & QUIRK_PMC_PLT_CLK_0) { + ret_val = clk_prepare_enable(drv->mclk); + if (ret_val < 0) { + dev_err(&pdev->dev, "MCLK enable error: %d\n", ret_val); + return ret_val; + } + } + ret_val = devm_snd_soc_register_card(&pdev->dev, &snd_soc_card_cht); if (ret_val) { dev_err(&pdev->dev, @@ -481,11 +504,23 @@ static int snd_cht_mc_probe(struct platform_device *pdev) return ret_val; } +static int snd_cht_mc_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + struct cht_mc_private *ctx = snd_soc_card_get_drvdata(card); + + if (ctx->quirks & QUIRK_PMC_PLT_CLK_0) + clk_disable_unprepare(ctx->mclk); + + return 0; +} + static struct platform_driver snd_cht_mc_driver = { .driver = { .name = "cht-bsw-max98090", }, .probe = snd_cht_mc_probe, + .remove = snd_cht_mc_remove, }; module_platform_driver(snd_cht_mc_driver) -- cgit v1.2.3 From 7297ba6c74c5b9e78d8e936af82eecfcf7d32dfb Mon Sep 17 00:00:00 2001 From: Annaliese McDermond Date: Wed, 3 Apr 2019 21:17:15 -0700 Subject: ASoC: tlv320aic32x4: Change author's name The author of these files has changed her name. Update instances in the code of her dead name to current legal name. Signed-off-by: Annaliese McDermond Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic32x4-i2c.c | 4 ++-- sound/soc/codecs/tlv320aic32x4-spi.c | 4 ++-- 2 files changed, 4 insertions(+), 4 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/tlv320aic32x4-i2c.c b/sound/soc/codecs/tlv320aic32x4-i2c.c index 385fa2e9525a..22c3a6bc0b6c 100644 --- a/sound/soc/codecs/tlv320aic32x4-i2c.c +++ b/sound/soc/codecs/tlv320aic32x4-i2c.c @@ -3,7 +3,7 @@ * * Copyright 2011 NW Digital Radio * - * Author: Jeremy McDermond + * Author: Annaliese McDermond * * Based on sound/soc/codecs/wm8974 and TI driver for kernel 2.6.27. * @@ -72,5 +72,5 @@ static struct i2c_driver aic32x4_i2c_driver = { module_i2c_driver(aic32x4_i2c_driver); MODULE_DESCRIPTION("ASoC TLV320AIC32x4 codec driver I2C"); -MODULE_AUTHOR("Jeremy McDermond "); +MODULE_AUTHOR("Annaliese McDermond "); MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tlv320aic32x4-spi.c b/sound/soc/codecs/tlv320aic32x4-spi.c index 07d78ae51e05..aa5b7ba0254b 100644 --- a/sound/soc/codecs/tlv320aic32x4-spi.c +++ b/sound/soc/codecs/tlv320aic32x4-spi.c @@ -3,7 +3,7 @@ * * Copyright 2011 NW Digital Radio * - * Author: Jeremy McDermond + * Author: Annaliese McDermond * * Based on sound/soc/codecs/wm8974 and TI driver for kernel 2.6.27. * @@ -74,5 +74,5 @@ static struct spi_driver aic32x4_spi_driver = { module_spi_driver(aic32x4_spi_driver); MODULE_DESCRIPTION("ASoC TLV320AIC32x4 codec driver SPI"); -MODULE_AUTHOR("Jeremy McDermond "); +MODULE_AUTHOR("Annaliese McDermond "); MODULE_LICENSE("GPL"); -- cgit v1.2.3 From c2c616021d64d952dc9d37793924ce57833d7754 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 4 Apr 2019 09:52:15 +0900 Subject: ASoC: audio-graph-card: don't select DPCM via audio-graph-card commit ae3cb5790906b ("ASoC: audio-graph-card: merge audio-graph-scu-card") merged audio-graph-scu-card which can handle DPCM into audio-graph-card. By this patch, the judgement to select "normal sound card" or "DPCM sound card" is based on its OF-graph endpoint connection. But, because of it, existing "audio-graph-card" user who is assuming "normal sound card" might select DPCM unintentionally. To solve this issue, this patch allows "audio-graph-card" user can select "normal sound card", and "audio-graph-scu-card" user can select both "normal sound card" and "DPCM sound card". This keeps compatibility collectry. Fixes: ae3cb5790906b ("ASoC: audio-graph-card: merge audio-graph-scu-card") Reported-by: Arnaud Pouliquen Signed-off-by: Kuninori Morimoto Acked-by: Arnaud Pouliquen Signed-off-by: Mark Brown --- sound/soc/generic/audio-graph-card.c | 11 ++++++++--- 1 file changed, 8 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index bb12351330e8..69bc4848d787 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -20,6 +20,8 @@ #include #include +#define DPCM_SELECTABLE 1 + struct graph_priv { struct snd_soc_card snd_card; struct graph_dai_props { @@ -440,6 +442,7 @@ static int graph_for_each_link(struct graph_priv *priv, struct device_node *codec_port; struct device_node *codec_port_old = NULL; struct asoc_simple_card_data adata; + uintptr_t dpcm_selectable = (uintptr_t)of_device_get_match_data(dev); int rc, ret; /* loop for all listed CPU port */ @@ -470,8 +473,9 @@ static int graph_for_each_link(struct graph_priv *priv, * if Codec port has many endpoints, * or has convert-xxx property */ - if ((of_get_child_count(codec_port) > 1) || - adata.convert_rate || adata.convert_channels) + if (dpcm_selectable && + ((of_get_child_count(codec_port) > 1) || + adata.convert_rate || adata.convert_channels)) ret = func_dpcm(priv, cpu_ep, codec_ep, li, (codec_port_old == codec_port)); /* else normal sound */ @@ -732,7 +736,8 @@ static int graph_remove(struct platform_device *pdev) static const struct of_device_id graph_of_match[] = { { .compatible = "audio-graph-card", }, - { .compatible = "audio-graph-scu-card", }, + { .compatible = "audio-graph-scu-card", + .data = (void *)DPCM_SELECTABLE }, {}, }; MODULE_DEVICE_TABLE(of, graph_of_match); -- cgit v1.2.3 From 42bf029a55a9bb8036f1d738a28dba2f7ec1e79d Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 4 Apr 2019 09:52:52 +0900 Subject: ASoC: simple-card: don't select DPCM via simple-audio-card commit da215354eb55c ("ASoC: simple-card: merge simple-scu-card") merged simple-scu-audio-card which can handle DPCM into simple-audio-card. By this patch, the judgement to select "normal sound card" or "DPCM sound card" is based on its CPU/Codec DAI count. But, because of it, existing "simple-audio-card" user who is assuming "normal sound card" might select DPCM unintentionally. To solve this issue, this patch allows "simple-audio-card" user can select "normal sound card", and "simple-scu-audio-card" user can select both "normal sound card" and "DPCM sound card". This keeps compatibility collectry. Fixes: da215354eb55c ("ASoC: simple-card: merge simple-scu-card") Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/generic/simple-card.c | 12 +++++++++--- 1 file changed, 9 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 7147bba45a2a..34de32efc4c4 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -9,12 +9,15 @@ #include #include #include +#include #include #include #include #include #include +#define DPCM_SELECTABLE 1 + struct simple_priv { struct snd_soc_card snd_card; struct simple_dai_props { @@ -441,6 +444,7 @@ static int simple_for_each_link(struct simple_priv *priv, struct device *dev = simple_priv_to_dev(priv); struct device_node *top = dev->of_node; struct device_node *node; + uintptr_t dpcm_selectable = (uintptr_t)of_device_get_match_data(dev); bool is_top = 0; int ret = 0; @@ -480,8 +484,9 @@ static int simple_for_each_link(struct simple_priv *priv, * if it has many CPUs, * or has convert-xxx property */ - if (num > 2 || - adata.convert_rate || adata.convert_channels) + if (dpcm_selectable && + (num > 2 || + adata.convert_rate || adata.convert_channels)) ret = func_dpcm(priv, np, codec, li, is_top); /* else normal sound */ else @@ -822,7 +827,8 @@ static int simple_remove(struct platform_device *pdev) static const struct of_device_id simple_of_match[] = { { .compatible = "simple-audio-card", }, - { .compatible = "simple-scu-audio-card", }, + { .compatible = "simple-scu-audio-card", + .data = (void *)DPCM_SELECTABLE }, {}, }; MODULE_DEVICE_TABLE(of, simple_of_match); -- cgit v1.2.3 From c85064435fe7a216ec0f0238ef2b8f7cd850a450 Mon Sep 17 00:00:00 2001 From: Sugar Zhang Date: Wed, 3 Apr 2019 21:40:45 +0800 Subject: ASoC: rockchip: pdm: fix regmap_ops hang issue This is because set_fmt ops maybe called when PD is off, and in such case, regmap_ops will lead system hang. enale PD before doing regmap_ops. Signed-off-by: Sugar Zhang Signed-off-by: Mark Brown --- sound/soc/rockchip/rockchip_pdm.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/rockchip/rockchip_pdm.c b/sound/soc/rockchip/rockchip_pdm.c index 400e29edb1c9..8a2e3bbce3a1 100644 --- a/sound/soc/rockchip/rockchip_pdm.c +++ b/sound/soc/rockchip/rockchip_pdm.c @@ -208,7 +208,9 @@ static int rockchip_pdm_set_fmt(struct snd_soc_dai *cpu_dai, return -EINVAL; } + pm_runtime_get_sync(cpu_dai->dev); regmap_update_bits(pdm->regmap, PDM_CLK_CTRL, mask, val); + pm_runtime_put(cpu_dai->dev); return 0; } -- cgit v1.2.3 From 86a7b6ffd90095d81d9fa0d8b48955b7c83b2e2f Mon Sep 17 00:00:00 2001 From: Sugar Zhang Date: Thu, 4 Apr 2019 11:48:11 +0800 Subject: ASoC: rockchip: pdm: change dma burst to 8 This patch decreases the transfer bursts to avoid the fifo overrun. Signed-off-by: Sugar Zhang Signed-off-by: Mark Brown --- sound/soc/rockchip/rockchip_pdm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/rockchip/rockchip_pdm.c b/sound/soc/rockchip/rockchip_pdm.c index 8a2e3bbce3a1..d0b403a0e27b 100644 --- a/sound/soc/rockchip/rockchip_pdm.c +++ b/sound/soc/rockchip/rockchip_pdm.c @@ -24,7 +24,7 @@ #include "rockchip_pdm.h" -#define PDM_DMA_BURST_SIZE (16) /* size * width: 16*4 = 64 bytes */ +#define PDM_DMA_BURST_SIZE (8) /* size * width: 8*4 = 32 bytes */ struct rk_pdm_dev { struct device *dev; -- cgit v1.2.3