From a2697972b9369c41afea8a928c30ac5b7f28d292 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Fri, 29 Oct 2021 16:40:27 -0500 Subject: ASoC: cs35l41: Change monitor widgets to siggens Currently the internal monitor sources are input widgets, which means if the card is set to fully routed these will not enable unless connected to something in the machine driver. However, all these are internal monitor signals so it makes no sense to connect them to something in the machine driver. As such switch them to siggen widgets which will have the same behaviour except not require external linkage on a fully routed card. Signed-off-by: Charles Keepax Signed-off-by: David Rhodes Link: https://lore.kernel.org/r/20211029214028.401284-1-drhodes@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs35l41.c | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs35l41.c b/sound/soc/codecs/cs35l41.c index 94ed21d7676f..9d0530dde996 100644 --- a/sound/soc/codecs/cs35l41.c +++ b/sound/soc/codecs/cs35l41.c @@ -612,6 +612,12 @@ static const struct snd_soc_dapm_widget cs35l41_dapm_widgets[] = { SND_SOC_DAPM_AIF_OUT("ASPTX3", NULL, 0, CS35L41_SP_ENABLES, 2, 0), SND_SOC_DAPM_AIF_OUT("ASPTX4", NULL, 0, CS35L41_SP_ENABLES, 3, 0), + SND_SOC_DAPM_SIGGEN("VSENSE"), + SND_SOC_DAPM_SIGGEN("ISENSE"), + SND_SOC_DAPM_SIGGEN("VP"), + SND_SOC_DAPM_SIGGEN("VBST"), + SND_SOC_DAPM_SIGGEN("TEMP"), + SND_SOC_DAPM_ADC("VMON ADC", NULL, CS35L41_PWR_CTRL2, 12, 0), SND_SOC_DAPM_ADC("IMON ADC", NULL, CS35L41_PWR_CTRL2, 13, 0), SND_SOC_DAPM_ADC("VPMON ADC", NULL, CS35L41_PWR_CTRL2, 8, 0), @@ -623,12 +629,6 @@ static const struct snd_soc_dapm_widget cs35l41_dapm_widgets[] = { cs35l41_main_amp_event, SND_SOC_DAPM_POST_PMD | SND_SOC_DAPM_POST_PMU), - SND_SOC_DAPM_INPUT("VP"), - SND_SOC_DAPM_INPUT("VBST"), - SND_SOC_DAPM_INPUT("ISENSE"), - SND_SOC_DAPM_INPUT("VSENSE"), - SND_SOC_DAPM_INPUT("TEMP"), - SND_SOC_DAPM_MUX("ASP TX1 Source", SND_SOC_NOPM, 0, 0, &asp_tx1_mux), SND_SOC_DAPM_MUX("ASP TX2 Source", SND_SOC_NOPM, 0, 0, &asp_tx2_mux), SND_SOC_DAPM_MUX("ASP TX3 Source", SND_SOC_NOPM, 0, 0, &asp_tx3_mux), @@ -674,8 +674,8 @@ static const struct snd_soc_dapm_route cs35l41_audio_map[] = { {"VMON ADC", NULL, "VSENSE"}, {"IMON ADC", NULL, "ISENSE"}, {"VPMON ADC", NULL, "VP"}, - {"TEMPMON ADC", NULL, "TEMP"}, {"VBSTMON ADC", NULL, "VBST"}, + {"TEMPMON ADC", NULL, "TEMP"}, {"ASPRX1", NULL, "AMP Playback"}, {"ASPRX2", NULL, "AMP Playback"}, -- cgit v1.2.3 From d9835eaa3e9fb4770745294fef3f8416446178c0 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 3 Nov 2021 10:27:10 +0200 Subject: ASoC: SOF:control: Fix variable type in snd_sof_refresh_control() The second parameter for snd_sof_ipc_set_get_comp_data() is ipc_cmd, not ipc_ctrl_type and the type is u32. Fixes: 756bbe4205bc6 ("ASoC: SOF: Handle control change notification from firmware") Signed-off-by: Peter Ujfalusi Acked-by: Ranjani Sridharan Link: https://lore.kernel.org/r/20211103082710.17165-1-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/control.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/control.c b/sound/soc/sof/control.c index 58bb89af4de1..bb1dfe4f6d40 100644 --- a/sound/soc/sof/control.c +++ b/sound/soc/sof/control.c @@ -69,7 +69,7 @@ static void snd_sof_refresh_control(struct snd_sof_control *scontrol) { struct sof_ipc_ctrl_data *cdata = scontrol->control_data; struct snd_soc_component *scomp = scontrol->scomp; - enum sof_ipc_ctrl_type ctrl_type; + u32 ipc_cmd; int ret; if (!scontrol->comp_data_dirty) @@ -79,9 +79,9 @@ static void snd_sof_refresh_control(struct snd_sof_control *scontrol) return; if (scontrol->cmd == SOF_CTRL_CMD_BINARY) - ctrl_type = SOF_IPC_COMP_GET_DATA; + ipc_cmd = SOF_IPC_COMP_GET_DATA; else - ctrl_type = SOF_IPC_COMP_GET_VALUE; + ipc_cmd = SOF_IPC_COMP_GET_VALUE; /* set the ABI header values */ cdata->data->magic = SOF_ABI_MAGIC; @@ -89,7 +89,7 @@ static void snd_sof_refresh_control(struct snd_sof_control *scontrol) /* refresh the component data from DSP */ scontrol->comp_data_dirty = false; - ret = snd_sof_ipc_set_get_comp_data(scontrol, ctrl_type, + ret = snd_sof_ipc_set_get_comp_data(scontrol, ipc_cmd, SOF_CTRL_TYPE_VALUE_CHAN_GET, scontrol->cmd, false); if (ret < 0) { -- cgit v1.2.3 From fd572393baf0350835e8d822db588f679dc7bcb8 Mon Sep 17 00:00:00 2001 From: Kai Vehmanen Date: Fri, 5 Nov 2021 13:16:55 +0200 Subject: ASoC: SOF: Intel: hda: fix hotplug when only codec is suspended MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit If codec is in runtime suspend, but controller is not, hotplug events are missed as the codec has no way to alert the controller. Problem does not occur if both controller and codec are active, or when both are suspended. An easy way to reproduce is to play an audio stream on one codec (e.g. to HDMI/DP display codec), wait for other HDA codec to go to runtime suspend, and then plug in a headset to the suspended codec. The jack event is not reported correctly in this case. Another way to reproduce is to force controller to stay active with "snd_sof_pci.sof_pci_debug=0x1" Fix the issue by reconfiguring the WAKEEN register when powering up/down individual links, and handling control events in the interrupt handler. Fixes: 87fc20e4a0cb ("ASoC: SOF: Intel: hda: use hdac_ext fine-grained link management") Reported-by: Hui Wang Signed-off-by: Kai Vehmanen Reviewed-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Reviewed-by: Péter Ujfalusi Link: https://lore.kernel.org/r/20211105111655.668777-1-kai.vehmanen@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-bus.c | 17 +++++++++++++++++ sound/soc/sof/intel/hda-dsp.c | 3 +-- sound/soc/sof/intel/hda.c | 16 ++++++++++++++++ 3 files changed, 34 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/intel/hda-bus.c b/sound/soc/sof/intel/hda-bus.c index 30025d3c16b6..0862ff8b6627 100644 --- a/sound/soc/sof/intel/hda-bus.c +++ b/sound/soc/sof/intel/hda-bus.c @@ -10,6 +10,8 @@ #include #include #include +#include +#include #include "../sof-priv.h" #include "hda.h" @@ -21,6 +23,18 @@ #endif #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) +static void update_codec_wake_enable(struct hdac_bus *bus, unsigned int addr, bool link_power) +{ + unsigned int mask = snd_hdac_chip_readw(bus, WAKEEN); + + if (link_power) + mask &= ~BIT(addr); + else + mask |= BIT(addr); + + snd_hdac_chip_updatew(bus, WAKEEN, STATESTS_INT_MASK, mask); +} + static void sof_hda_bus_link_power(struct hdac_device *codec, bool enable) { struct hdac_bus *bus = codec->bus; @@ -41,6 +55,9 @@ static void sof_hda_bus_link_power(struct hdac_device *codec, bool enable) */ if (codec->addr == HDA_IDISP_ADDR && !enable) snd_hdac_display_power(bus, HDA_CODEC_IDX_CONTROLLER, false); + + /* WAKEEN needs to be set for disabled links */ + update_codec_wake_enable(bus, codec->addr, enable); } static const struct hdac_bus_ops bus_core_ops = { diff --git a/sound/soc/sof/intel/hda-dsp.c b/sound/soc/sof/intel/hda-dsp.c index 058baca2cd0e..287dc0eb6686 100644 --- a/sound/soc/sof/intel/hda-dsp.c +++ b/sound/soc/sof/intel/hda-dsp.c @@ -622,8 +622,7 @@ static int hda_suspend(struct snd_sof_dev *sdev, bool runtime_suspend) hda_dsp_ipc_int_disable(sdev); #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) - if (runtime_suspend) - hda_codec_jack_wake_enable(sdev, true); + hda_codec_jack_wake_enable(sdev, runtime_suspend); /* power down all hda link */ snd_hdac_ext_bus_link_power_down_all(bus); diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index 883d78dd01b5..568d351b7a4e 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -810,6 +810,20 @@ skip_soundwire: return 0; } +static void hda_check_for_state_change(struct snd_sof_dev *sdev) +{ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) + struct hdac_bus *bus = sof_to_bus(sdev); + unsigned int codec_mask; + + codec_mask = snd_hdac_chip_readw(bus, STATESTS); + if (codec_mask) { + hda_codec_jack_check(sdev); + snd_hdac_chip_writew(bus, STATESTS, codec_mask); + } +#endif +} + static irqreturn_t hda_dsp_interrupt_handler(int irq, void *context) { struct snd_sof_dev *sdev = context; @@ -851,6 +865,8 @@ static irqreturn_t hda_dsp_interrupt_thread(int irq, void *context) if (hda_sdw_check_wakeen_irq(sdev)) hda_sdw_process_wakeen(sdev); + hda_check_for_state_change(sdev); + /* enable GIE interrupt */ snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, SOF_HDA_INTCTL, -- cgit v1.2.3 From 827b0913a9d9d07a0c3e559dbb20ca4d6d285a54 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 5 Nov 2021 10:09:25 +0100 Subject: ASoC: DAPM: Cover regression by kctl change notification fix The recent fix for DAPM to correct the kctl change notification by the commit 5af82c81b2c4 ("ASoC: DAPM: Fix missing kctl change notifications") caused other regressions since it changed the behavior of snd_soc_dapm_set_pin() that is called from several API functions. Formerly it returned always 0 for success, but now it returns 0 or 1. This patch addresses it, restoring the old behavior of snd_soc_dapm_set_pin() while keeping the fix in snd_soc_dapm_put_pin_switch(). Fixes: 5af82c81b2c4 ("ASoC: DAPM: Fix missing kctl change notifications") Reported-by: Yu-Hsuan Hsu Cc: Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20211105090925.20575-1-tiwai@suse.de Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 29 +++++++++++++++++++++++------ 1 file changed, 23 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 2892b0aba151..b06c5682445c 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2559,8 +2559,13 @@ static struct snd_soc_dapm_widget *dapm_find_widget( return NULL; } -static int snd_soc_dapm_set_pin(struct snd_soc_dapm_context *dapm, - const char *pin, int status) +/* + * set the DAPM pin status: + * returns 1 when the value has been updated, 0 when unchanged, or a negative + * error code; called from kcontrol put callback + */ +static int __snd_soc_dapm_set_pin(struct snd_soc_dapm_context *dapm, + const char *pin, int status) { struct snd_soc_dapm_widget *w = dapm_find_widget(dapm, pin, true); int ret = 0; @@ -2586,6 +2591,18 @@ static int snd_soc_dapm_set_pin(struct snd_soc_dapm_context *dapm, return ret; } +/* + * similar as __snd_soc_dapm_set_pin(), but returns 0 when successful; + * called from several API functions below + */ +static int snd_soc_dapm_set_pin(struct snd_soc_dapm_context *dapm, + const char *pin, int status) +{ + int ret = __snd_soc_dapm_set_pin(dapm, pin, status); + + return ret < 0 ? ret : 0; +} + /** * snd_soc_dapm_sync_unlocked - scan and power dapm paths * @dapm: DAPM context @@ -3589,10 +3606,10 @@ int snd_soc_dapm_put_pin_switch(struct snd_kcontrol *kcontrol, const char *pin = (const char *)kcontrol->private_value; int ret; - if (ucontrol->value.integer.value[0]) - ret = snd_soc_dapm_enable_pin(&card->dapm, pin); - else - ret = snd_soc_dapm_disable_pin(&card->dapm, pin); + mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); + ret = __snd_soc_dapm_set_pin(&card->dapm, pin, + !!ucontrol->value.integer.value[0]); + mutex_unlock(&card->dapm_mutex); snd_soc_dapm_sync(&card->dapm); return ret; -- cgit v1.2.3 From 0a8facac0d1e38dc8b86ade6d3f0d8b33dae7c58 Mon Sep 17 00:00:00 2001 From: AngeloGioacchino Del Regno Date: Fri, 5 Nov 2021 16:20:13 +0100 Subject: ASoC: mediatek: mt8173-rt5650: Rename Speaker control to Ext Spk Some RT5645 and RT5650 powered platforms are using "Ext Spk" instead of "Speaker", and this is also reflected in alsa-lib configurations for the generic RT5645 usecase manager configs. Rename the "Speaker" control to "Ext Spk" in order to be able to make the userspace reuse/inherit the same configurations also for this machine, along with the others. Signed-off-by: AngeloGioacchino Del Regno Link: https://lore.kernel.org/r/20211105152013.75252-1-angelogioacchino.delregno@collabora.com Signed-off-by: Mark Brown --- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650.c b/sound/soc/mediatek/mt8173/mt8173-rt5650.c index c28ebf891cb0..2cbf679f5c74 100644 --- a/sound/soc/mediatek/mt8173/mt8173-rt5650.c +++ b/sound/soc/mediatek/mt8173/mt8173-rt5650.c @@ -30,15 +30,15 @@ static struct mt8173_rt5650_platform_data mt8173_rt5650_priv = { }; static const struct snd_soc_dapm_widget mt8173_rt5650_widgets[] = { - SND_SOC_DAPM_SPK("Speaker", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), SND_SOC_DAPM_MIC("Int Mic", NULL), SND_SOC_DAPM_HP("Headphone", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), }; static const struct snd_soc_dapm_route mt8173_rt5650_routes[] = { - {"Speaker", NULL, "SPOL"}, - {"Speaker", NULL, "SPOR"}, + {"Ext Spk", NULL, "SPOL"}, + {"Ext Spk", NULL, "SPOR"}, {"DMIC L1", NULL, "Int Mic"}, {"DMIC R1", NULL, "Int Mic"}, {"Headphone", NULL, "HPOL"}, @@ -48,7 +48,7 @@ static const struct snd_soc_dapm_route mt8173_rt5650_routes[] = { }; static const struct snd_kcontrol_new mt8173_rt5650_controls[] = { - SOC_DAPM_PIN_SWITCH("Speaker"), + SOC_DAPM_PIN_SWITCH("Ext Spk"), SOC_DAPM_PIN_SWITCH("Int Mic"), SOC_DAPM_PIN_SWITCH("Headphone"), SOC_DAPM_PIN_SWITCH("Headset Mic"), -- cgit v1.2.3 From 8f4fa45982b3f2daf5b3626ca0f12bde735f31ff Mon Sep 17 00:00:00 2001 From: Gongjun Song Date: Fri, 5 Nov 2021 10:26:38 +0800 Subject: ASoC: Intel: sof_sdw: Add support for SKU 0AF3 product This product supports SoundWire capture from local microphones and two SoundWire amplifiers(no headset codec). Signed-off-by: Gongjun Song Reviewed-by: Pierre-Louis Bossart Reviewed-by: Rander Wang Signed-off-by: Bard Liao Link: https://lore.kernel.org/r/20211105022646.26305-2-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_sdw.c | 10 ++++++++++ 1 file changed, 10 insertions(+) (limited to 'sound') diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index f10496206cee..584f9f2db207 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -248,6 +248,16 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { SOF_BT_OFFLOAD_SSP(2) | SOF_SSP_BT_OFFLOAD_PRESENT), }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0AF3"), + }, + /* No Jack */ + .driver_data = (void *)(SOF_SDW_TGL_HDMI | + SOF_SDW_FOUR_SPK), + }, {} }; -- cgit v1.2.3 From a1797d61cb35848432867a5bc294ce43058b5ead Mon Sep 17 00:00:00 2001 From: Gongjun Song Date: Fri, 5 Nov 2021 10:26:39 +0800 Subject: ASoC: Intel: soc-acpi: add SKU 0AF3 SoundWire configuration New product audio hardware configuration is rt714 on link0, two rt1316s on link1 and link2 Signed-off-by: Gongjun Song Reviewed-by: Pierre-Louis Bossart Reviewed-by: Rander Wang Signed-off-by: Bard Liao Link: https://lore.kernel.org/r/20211105022646.26305-3-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/common/soc-acpi-intel-adl-match.c | 26 +++++++++++++++++++++++ 1 file changed, 26 insertions(+) (limited to 'sound') diff --git a/sound/soc/intel/common/soc-acpi-intel-adl-match.c b/sound/soc/intel/common/soc-acpi-intel-adl-match.c index 06f503452aa5..d8ae94d39d57 100644 --- a/sound/soc/intel/common/soc-acpi-intel-adl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-adl-match.c @@ -209,6 +209,25 @@ static const struct snd_soc_acpi_link_adr adl_sdca_3_in_1[] = { {} }; +static const struct snd_soc_acpi_link_adr adl_sdw_rt1316_link12_rt714_link0[] = { + { + .mask = BIT(1), + .num_adr = ARRAY_SIZE(rt1316_1_group1_adr), + .adr_d = rt1316_1_group1_adr, + }, + { + .mask = BIT(2), + .num_adr = ARRAY_SIZE(rt1316_2_group1_adr), + .adr_d = rt1316_2_group1_adr, + }, + { + .mask = BIT(0), + .num_adr = ARRAY_SIZE(rt714_0_adr), + .adr_d = rt714_0_adr, + }, + {} +}; + static const struct snd_soc_acpi_link_adr adl_sdw_rt1316_link2_rt714_link0[] = { { .mask = BIT(2), @@ -339,6 +358,13 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_sdw_machines[] = { .drv_name = "sof_sdw", .sof_tplg_filename = "sof-adl-rt711-l0-rt1316-l13-rt714-l2.tplg", }, + { + .link_mask = 0x7, /* rt714 on link0 & two rt1316s on link1 and link2 */ + .links = adl_sdw_rt1316_link12_rt714_link0, + .drv_name = "sof_sdw", + .sof_fw_filename = "sof-adl.ri", + .sof_tplg_filename = "sof-adl-rt1316-l12-rt714-l0.tplg", + }, { .link_mask = 0x5, /* 2 active links required */ .links = adl_sdw_rt1316_link2_rt714_link0, -- cgit v1.2.3 From cf304329e4afb97ffabce232eadaba94f025641d Mon Sep 17 00:00:00 2001 From: Gongjun Song Date: Fri, 5 Nov 2021 10:26:40 +0800 Subject: ASoC: Intel: sof_sdw: Add support for SKU 0B00 and 0B01 products Both products support a SoundWire headset codec, SoundWire capture from local microphones and two SoundWire amplifiers. Signed-off-by: Gongjun Song Reviewed-by: Pierre-Louis Bossart Reviewed-by: Rander Wang Signed-off-by: Bard Liao Link: https://lore.kernel.org/r/20211105022646.26305-4-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_sdw.c | 20 ++++++++++++++++++++ 1 file changed, 20 insertions(+) (limited to 'sound') diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 584f9f2db207..55c3e5935585 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -258,6 +258,26 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { .driver_data = (void *)(SOF_SDW_TGL_HDMI | SOF_SDW_FOUR_SPK), }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0B00") + }, + .driver_data = (void *)(SOF_SDW_TGL_HDMI | + RT711_JD2 | + SOF_SDW_FOUR_SPK), + }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0B01") + }, + .driver_data = (void *)(SOF_SDW_TGL_HDMI | + RT711_JD2 | + SOF_SDW_FOUR_SPK), + }, {} }; -- cgit v1.2.3 From 6fef4c2f458680399b7c512cb810c1e1784d7444 Mon Sep 17 00:00:00 2001 From: Gongjun Song Date: Fri, 5 Nov 2021 10:26:41 +0800 Subject: ASoC: Intel: sof_sdw: Add support for SKU 0B11 product This product supports a SoundWire headset codec, SoundWire capture from local microphones and two SoundWire amplifiers. Signed-off-by: Gongjun Song Reviewed-by: Pierre-Louis Bossart Reviewed-by: Rander Wang Signed-off-by: Bard Liao Link: https://lore.kernel.org/r/20211105022646.26305-5-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_sdw.c | 10 ++++++++++ 1 file changed, 10 insertions(+) (limited to 'sound') diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 55c3e5935585..d0bea028b9b7 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -278,6 +278,16 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { RT711_JD2 | SOF_SDW_FOUR_SPK), }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0B11") + }, + .driver_data = (void *)(SOF_SDW_TGL_HDMI | + RT711_JD2 | + SOF_SDW_FOUR_SPK), + }, {} }; -- cgit v1.2.3 From 6448d0596e48dbc16a910f04ffc248c3f3c0a65c Mon Sep 17 00:00:00 2001 From: Gongjun Song Date: Fri, 5 Nov 2021 10:26:42 +0800 Subject: ASoC: Intel: sof_sdw: Add support for SKU 0B13 product This product supports SoundWire capture from local microphones and one SoundWire amplifier(no headset codec). Signed-off-by: Gongjun Song Reviewed-by: Pierre-Louis Bossart Reviewed-by: Rander Wang Signed-off-by: Bard Liao Link: https://lore.kernel.org/r/20211105022646.26305-6-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_sdw.c | 9 +++++++++ 1 file changed, 9 insertions(+) (limited to 'sound') diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index d0bea028b9b7..25cdd61f09a8 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -288,6 +288,15 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { RT711_JD2 | SOF_SDW_FOUR_SPK), }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0B13"), + }, + /* No Jack */ + .driver_data = (void *)SOF_SDW_TGL_HDMI, + }, {} }; -- cgit v1.2.3 From 11e18f582c14fdf08f52d99d439d2b82d98ac37d Mon Sep 17 00:00:00 2001 From: Gongjun Song Date: Fri, 5 Nov 2021 10:26:43 +0800 Subject: ASoC: Intel: soc-acpi: add SKU 0B13 SoundWire configuration Product audio hardware configuration is rt1316 on link2, rt714 on link 3. Signed-off-by: Gongjun Song Reviewed-by: Pierre-Louis Bossart Reviewed-by: Rander Wang Signed-off-by: Bard Liao Link: https://lore.kernel.org/r/20211105022646.26305-7-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/common/soc-acpi-intel-adl-match.c | 21 +++++++++++++++++++++ 1 file changed, 21 insertions(+) (limited to 'sound') diff --git a/sound/soc/intel/common/soc-acpi-intel-adl-match.c b/sound/soc/intel/common/soc-acpi-intel-adl-match.c index d8ae94d39d57..3440c0fa31fa 100644 --- a/sound/soc/intel/common/soc-acpi-intel-adl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-adl-match.c @@ -228,6 +228,20 @@ static const struct snd_soc_acpi_link_adr adl_sdw_rt1316_link12_rt714_link0[] = {} }; +static const struct snd_soc_acpi_link_adr adl_sdw_rt1316_link2_rt714_link3[] = { + { + .mask = BIT(2), + .num_adr = ARRAY_SIZE(rt1316_2_single_adr), + .adr_d = rt1316_2_single_adr, + }, + { + .mask = BIT(3), + .num_adr = ARRAY_SIZE(rt714_3_adr), + .adr_d = rt714_3_adr, + }, + {} +}; + static const struct snd_soc_acpi_link_adr adl_sdw_rt1316_link2_rt714_link0[] = { { .mask = BIT(2), @@ -358,6 +372,13 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_sdw_machines[] = { .drv_name = "sof_sdw", .sof_tplg_filename = "sof-adl-rt711-l0-rt1316-l13-rt714-l2.tplg", }, + { + .link_mask = 0xC, /* rt1316 on link2 & rt714 on link3 */ + .links = adl_sdw_rt1316_link2_rt714_link3, + .drv_name = "sof_sdw", + .sof_fw_filename = "sof-adl.ri", + .sof_tplg_filename = "sof-adl-rt1316-l2-mono-rt714-l3.tplg", + }, { .link_mask = 0x7, /* rt714 on link0 & two rt1316s on link1 and link2 */ .links = adl_sdw_rt1316_link12_rt714_link0, -- cgit v1.2.3 From 0c2ed4f03f0bfe2be34efbabbebe9875c3aa9ca9 Mon Sep 17 00:00:00 2001 From: Gongjun Song Date: Fri, 5 Nov 2021 10:26:44 +0800 Subject: ASoC: Intel: sof_sdw: Add support for SKU 0B29 product This product supports a SoundWire headset codec, SoundWire capture from local microphones and two SoundWire amplifiers. Signed-off-by: Gongjun Song Reviewed-by: Pierre-Louis Bossart Reviewed-by: Rander Wang Signed-off-by: Bard Liao Link: https://lore.kernel.org/r/20211105022646.26305-8-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_sdw.c | 10 ++++++++++ 1 file changed, 10 insertions(+) (limited to 'sound') diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 25cdd61f09a8..bfbdda323b87 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -297,6 +297,16 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { /* No Jack */ .driver_data = (void *)SOF_SDW_TGL_HDMI, }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0B29"), + }, + .driver_data = (void *)(SOF_SDW_TGL_HDMI | + RT711_JD2 | + SOF_SDW_FOUR_SPK), + }, {} }; -- cgit v1.2.3 From 359ace2b9a411c3bd4b89fdc56f8b60e0f6696d2 Mon Sep 17 00:00:00 2001 From: Gongjun Song Date: Fri, 5 Nov 2021 10:26:45 +0800 Subject: ASoC: Intel: soc-acpi: add SKU 0B29 SoundWire configuration Product audio hardware configuration is rt711 on link2, two rt1316s on link0 and link1, rt714 on link 3. Signed-off-by: Gongjun Song Reviewed-by: Pierre-Louis Bossart Reviewed-by: Rander Wang Signed-off-by: Bard Liao Link: https://lore.kernel.org/r/20211105022646.26305-9-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/common/soc-acpi-intel-adl-match.c | 58 +++++++++++++++++++++++ 1 file changed, 58 insertions(+) (limited to 'sound') diff --git a/sound/soc/intel/common/soc-acpi-intel-adl-match.c b/sound/soc/intel/common/soc-acpi-intel-adl-match.c index 3440c0fa31fa..b61a778a9d26 100644 --- a/sound/soc/intel/common/soc-acpi-intel-adl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-adl-match.c @@ -74,6 +74,15 @@ static const struct snd_soc_acpi_adr_device rt711_sdca_0_adr[] = { } }; +static const struct snd_soc_acpi_adr_device rt711_sdca_2_adr[] = { + { + .adr = 0x000230025D071101ull, + .num_endpoints = 1, + .endpoints = &single_endpoint, + .name_prefix = "rt711" + } +}; + static const struct snd_soc_acpi_adr_device rt1316_1_group1_adr[] = { { .adr = 0x000131025D131601ull, /* unique ID is set for some reason */ @@ -101,6 +110,24 @@ static const struct snd_soc_acpi_adr_device rt1316_3_group1_adr[] = { } }; +static const struct snd_soc_acpi_adr_device rt1316_0_group2_adr[] = { + { + .adr = 0x000031025D131601ull, + .num_endpoints = 1, + .endpoints = &spk_l_endpoint, + .name_prefix = "rt1316-1" + } +}; + +static const struct snd_soc_acpi_adr_device rt1316_1_group2_adr[] = { + { + .adr = 0x000130025D131601ull, + .num_endpoints = 1, + .endpoints = &spk_r_endpoint, + .name_prefix = "rt1316-2" + } +}; + static const struct snd_soc_acpi_adr_device rt1316_2_single_adr[] = { { .adr = 0x000230025D131601ull, @@ -209,6 +236,30 @@ static const struct snd_soc_acpi_link_adr adl_sdca_3_in_1[] = { {} }; +static const struct snd_soc_acpi_link_adr adl_sdw_rt711_link2_rt1316_link01_rt714_link3[] = { + { + .mask = BIT(2), + .num_adr = ARRAY_SIZE(rt711_sdca_2_adr), + .adr_d = rt711_sdca_2_adr, + }, + { + .mask = BIT(0), + .num_adr = ARRAY_SIZE(rt1316_0_group2_adr), + .adr_d = rt1316_0_group2_adr, + }, + { + .mask = BIT(1), + .num_adr = ARRAY_SIZE(rt1316_1_group2_adr), + .adr_d = rt1316_1_group2_adr, + }, + { + .mask = BIT(3), + .num_adr = ARRAY_SIZE(rt714_3_adr), + .adr_d = rt714_3_adr, + }, + {} +}; + static const struct snd_soc_acpi_link_adr adl_sdw_rt1316_link12_rt714_link0[] = { { .mask = BIT(1), @@ -372,6 +423,13 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_sdw_machines[] = { .drv_name = "sof_sdw", .sof_tplg_filename = "sof-adl-rt711-l0-rt1316-l13-rt714-l2.tplg", }, + { + .link_mask = 0xF, /* 4 active links required */ + .links = adl_sdw_rt711_link2_rt1316_link01_rt714_link3, + .drv_name = "sof_sdw", + .sof_fw_filename = "sof-adl.ri", + .sof_tplg_filename = "sof-adl-rt711-l2-rt1316-l01-rt714-l3.tplg", + }, { .link_mask = 0xC, /* rt1316 on link2 & rt714 on link3 */ .links = adl_sdw_rt1316_link2_rt714_link3, -- cgit v1.2.3 From f55af7055cd465f6b767a0c1126977d4529c63c8 Mon Sep 17 00:00:00 2001 From: Gongjun Song Date: Fri, 5 Nov 2021 10:26:46 +0800 Subject: ASoC: Intel: sof_sdw: Add support for SKU 0B12 product This product supports a SoundWire headset codec, SoundWire capture from local microphones and two SoundWire amplifiers. Signed-off-by: Libin Yang Signed-off-by: Gongjun Song Reviewed-by: Pierre-Louis Bossart Reviewed-by: Rander Wang Signed-off-by: Bard Liao Link: https://lore.kernel.org/r/20211105022646.26305-10-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_sdw.c | 10 ++++++++++ 1 file changed, 10 insertions(+) (limited to 'sound') diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index bfbdda323b87..77219c3f8766 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -288,6 +288,16 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { RT711_JD2 | SOF_SDW_FOUR_SPK), }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0B12") + }, + .driver_data = (void *)(SOF_SDW_TGL_HDMI | + RT711_JD2 | + SOF_SDW_FOUR_SPK), + }, { .callback = sof_sdw_quirk_cb, .matches = { -- cgit v1.2.3 From 9bb4e4bae5a19ca68527392e85ad5ee88fc4b786 Mon Sep 17 00:00:00 2001 From: ChiYuan Huang Date: Wed, 10 Nov 2021 11:45:19 +0800 Subject: ASoC: rt9120: Update internal ocp level to the correct value Update internal ocp level to correct value. Even the wrong ocp setting can also make the sound output, but the power cannot match the IC capability. Signed-off-by: ChiYuan Huang Link: https://lore.kernel.org/r/1636515921-31694-2-git-send-email-u0084500@gmail.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt9120.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt9120.c b/sound/soc/codecs/rt9120.c index f9574980a407..032c520aca0c 100644 --- a/sound/soc/codecs/rt9120.c +++ b/sound/soc/codecs/rt9120.c @@ -149,8 +149,7 @@ static int rt9120_codec_probe(struct snd_soc_component *comp) snd_soc_component_init_regmap(comp, data->regmap); /* Internal setting */ - snd_soc_component_write(comp, RT9120_REG_INTERNAL1, 0x03); - snd_soc_component_write(comp, RT9120_REG_INTERNAL0, 0x69); + snd_soc_component_write(comp, RT9120_REG_INTERNAL0, 0x04); return 0; } -- cgit v1.2.3 From 8f1f1846d78a318c7cdb8268b47a964a3dbc0075 Mon Sep 17 00:00:00 2001 From: ChiYuan Huang Date: Wed, 10 Nov 2021 11:45:20 +0800 Subject: ASoC: rt9120: Fix clock auto sync issue when fs is the multiple of 48 If fs is divided by 48, to make audio clock sync rate correct, internal sync function have be disabled. Signed-off-by: ChiYuan Huang Link: https://lore.kernel.org/r/1636515921-31694-3-git-send-email-u0084500@gmail.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt9120.c | 24 +++++++++++++++++++----- 1 file changed, 19 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt9120.c b/sound/soc/codecs/rt9120.c index 032c520aca0c..981aead83814 100644 --- a/sound/soc/codecs/rt9120.c +++ b/sound/soc/codecs/rt9120.c @@ -26,6 +26,7 @@ #define RT9120_REG_INTERNAL0 0x65 #define RT9120_REG_INTERNAL1 0x69 #define RT9120_REG_UVPOPT 0x6C +#define RT9120_REG_DIGCFG 0xF8 #define RT9120_VID_MASK GENMASK(15, 8) #define RT9120_SWRST_MASK BIT(7) @@ -46,6 +47,7 @@ #define RT9120_CFG_WORDLEN_24 24 #define RT9120_CFG_WORDLEN_32 32 #define RT9120_DVDD_UVSEL_MASK GENMASK(5, 4) +#define RT9120_AUTOSYNC_MASK BIT(6) #define RT9120_VENDOR_ID 0x4200 #define RT9120_RESET_WAITMS 20 @@ -200,8 +202,8 @@ static int rt9120_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_component *comp = dai->component; - unsigned int param_width, param_slot_width; - int width; + unsigned int param_width, param_slot_width, auto_sync; + int width, fs; switch (width = params_width(param)) { case 16: @@ -239,6 +241,16 @@ static int rt9120_hw_params(struct snd_pcm_substream *substream, snd_soc_component_update_bits(comp, RT9120_REG_I2SWL, RT9120_AUDWL_MASK, param_slot_width); + + fs = width * params_channels(param); + /* If fs is divided by 48, disable auto sync */ + if (fs % 48 == 0) + auto_sync = 0; + else + auto_sync = RT9120_AUTOSYNC_MASK; + + snd_soc_component_update_bits(comp, RT9120_REG_DIGCFG, + RT9120_AUTOSYNC_MASK, auto_sync); return 0; } @@ -280,7 +292,8 @@ static const struct regmap_range rt9120_rd_yes_ranges[] = { regmap_reg_range(0x3A, 0x40), regmap_reg_range(0x65, 0x65), regmap_reg_range(0x69, 0x69), - regmap_reg_range(0x6C, 0x6C) + regmap_reg_range(0x6C, 0x6C), + regmap_reg_range(0xF8, 0xF8) }; static const struct regmap_access_table rt9120_rd_table = { @@ -298,7 +311,8 @@ static const struct regmap_range rt9120_wr_yes_ranges[] = { regmap_reg_range(0x40, 0x40), regmap_reg_range(0x65, 0x65), regmap_reg_range(0x69, 0x69), - regmap_reg_range(0x6C, 0x6C) + regmap_reg_range(0x6C, 0x6C), + regmap_reg_range(0xF8, 0xF8) }; static const struct regmap_access_table rt9120_wr_table = { @@ -369,7 +383,7 @@ static int rt9120_reg_write(void *context, unsigned int reg, unsigned int val) static const struct regmap_config rt9120_regmap_config = { .reg_bits = 8, .val_bits = 32, - .max_register = RT9120_REG_UVPOPT, + .max_register = RT9120_REG_DIGCFG, .reg_read = rt9120_reg_read, .reg_write = rt9120_reg_write, -- cgit v1.2.3 From dbe638f71eaed5c7b5fbbf03fb044e429c4a2d48 Mon Sep 17 00:00:00 2001 From: ChiYuan Huang Date: Wed, 10 Nov 2021 11:45:21 +0800 Subject: ASoC: rt9120: Add the compatibility with rt9120s Use device id reg to be compatible with rt9120 and rt9120s. Signed-off-by: ChiYuan Huang Link: https://lore.kernel.org/r/1636515921-31694-4-git-send-email-u0084500@gmail.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt9120.c | 33 +++++++++++++++++++++++++++++---- 1 file changed, 29 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt9120.c b/sound/soc/codecs/rt9120.c index 981aead83814..7aa1772a915f 100644 --- a/sound/soc/codecs/rt9120.c +++ b/sound/soc/codecs/rt9120.c @@ -1,6 +1,7 @@ // SPDX-License-Identifier: GPL-2.0 #include +#include #include #include #include @@ -23,6 +24,7 @@ #define RT9120_REG_ERRRPT 0x10 #define RT9120_REG_MSVOL 0x20 #define RT9120_REG_SWRESET 0x40 +#define RT9120_REG_INTERCFG 0x63 #define RT9120_REG_INTERNAL0 0x65 #define RT9120_REG_INTERNAL1 0x69 #define RT9120_REG_UVPOPT 0x6C @@ -49,7 +51,8 @@ #define RT9120_DVDD_UVSEL_MASK GENMASK(5, 4) #define RT9120_AUTOSYNC_MASK BIT(6) -#define RT9120_VENDOR_ID 0x4200 +#define RT9120_VENDOR_ID 0x42 +#define RT9120S_VENDOR_ID 0x43 #define RT9120_RESET_WAITMS 20 #define RT9120_CHIPON_WAITMS 20 #define RT9120_AMPON_WAITMS 50 @@ -63,9 +66,16 @@ SNDRV_PCM_FMTBIT_S24_LE |\ SNDRV_PCM_FMTBIT_S32_LE) +enum { + CHIP_IDX_RT9120 = 0, + CHIP_IDX_RT9120S, + CHIP_IDX_MAX +}; + struct rt9120_data { struct device *dev; struct regmap *regmap; + int chip_idx; }; /* 11bit [min,max,step] = [-103.9375dB, 24dB, 0.0625dB] */ @@ -151,7 +161,12 @@ static int rt9120_codec_probe(struct snd_soc_component *comp) snd_soc_component_init_regmap(comp, data->regmap); /* Internal setting */ - snd_soc_component_write(comp, RT9120_REG_INTERNAL0, 0x04); + if (data->chip_idx == CHIP_IDX_RT9120S) { + snd_soc_component_write(comp, RT9120_REG_INTERCFG, 0xde); + snd_soc_component_write(comp, RT9120_REG_INTERNAL0, 0x66); + } else + snd_soc_component_write(comp, RT9120_REG_INTERNAL0, 0x04); + return 0; } @@ -290,6 +305,7 @@ static const struct regmap_range rt9120_rd_yes_ranges[] = { regmap_reg_range(0x20, 0x27), regmap_reg_range(0x30, 0x38), regmap_reg_range(0x3A, 0x40), + regmap_reg_range(0x63, 0x63), regmap_reg_range(0x65, 0x65), regmap_reg_range(0x69, 0x69), regmap_reg_range(0x6C, 0x6C), @@ -309,6 +325,7 @@ static const struct regmap_range rt9120_wr_yes_ranges[] = { regmap_reg_range(0x30, 0x38), regmap_reg_range(0x3A, 0x3D), regmap_reg_range(0x40, 0x40), + regmap_reg_range(0x63, 0x63), regmap_reg_range(0x65, 0x65), regmap_reg_range(0x69, 0x69), regmap_reg_range(0x6C, 0x6C), @@ -401,8 +418,16 @@ static int rt9120_check_vendor_info(struct rt9120_data *data) if (ret) return ret; - if ((devid & RT9120_VID_MASK) != RT9120_VENDOR_ID) { - dev_err(data->dev, "DEVID not correct [0x%04x]\n", devid); + devid = FIELD_GET(RT9120_VID_MASK, devid); + switch (devid) { + case RT9120_VENDOR_ID: + data->chip_idx = CHIP_IDX_RT9120; + break; + case RT9120S_VENDOR_ID: + data->chip_idx = CHIP_IDX_RT9120S; + break; + default: + dev_err(data->dev, "DEVID not correct [0x%0x]\n", devid); return -ENODEV; } -- cgit v1.2.3 From a382285b6feda8db56955e5897453405c198048d Mon Sep 17 00:00:00 2001 From: Jack Yu Date: Thu, 11 Nov 2021 17:17:05 +0800 Subject: ASoC: rt1011: revert 'I2S Reference' to SOC_ENUM_EXT Revert 'I2S Reference' to SOC_ENUM_EXT because the settings are specific for some platforms, the default setting for 'I2S Reference' does nothing, only some SoC platform need to configure it. Previous 'I2S Reference' in SOC_ENUM format only toggles one bit of RT1011_TDM1_SET_1 register, which isn't enough for specific platform. Signed-off-by: Jack Yu Link: https://lore.kernel.org/r/20211111091705.20879-1-jack.yu@realtek.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt1011.c | 55 +++++++++++++++++++++++++++++++++++++++++------ sound/soc/codecs/rt1011.h | 7 ++++++ 2 files changed, 56 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt1011.c b/sound/soc/codecs/rt1011.c index 297af7ff824c..b62301a6281f 100644 --- a/sound/soc/codecs/rt1011.c +++ b/sound/soc/codecs/rt1011.c @@ -1311,13 +1311,54 @@ static int rt1011_r0_load_info(struct snd_kcontrol *kcontrol, .put = rt1011_r0_load_mode_put \ } -static const char * const rt1011_i2s_ref_texts[] = { - "Left Channel", "Right Channel" +static const char * const rt1011_i2s_ref[] = { + "None", "Left Channel", "Right Channel" }; -static SOC_ENUM_SINGLE_DECL(rt1011_i2s_ref_enum, - RT1011_TDM1_SET_1, 7, - rt1011_i2s_ref_texts); +static SOC_ENUM_SINGLE_DECL(rt1011_i2s_ref_enum, 0, 0, + rt1011_i2s_ref); + +static int rt1011_i2s_ref_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct rt1011_priv *rt1011 = + snd_soc_component_get_drvdata(component); + + rt1011->i2s_ref = ucontrol->value.enumerated.item[0]; + switch (rt1011->i2s_ref) { + case RT1011_I2S_REF_LEFT_CH: + regmap_write(rt1011->regmap, RT1011_TDM_TOTAL_SET, 0x0240); + regmap_write(rt1011->regmap, RT1011_TDM1_SET_2, 0x8); + regmap_write(rt1011->regmap, RT1011_TDM1_SET_1, 0x1022); + regmap_write(rt1011->regmap, RT1011_ADCDAT_OUT_SOURCE, 0x4); + break; + case RT1011_I2S_REF_RIGHT_CH: + regmap_write(rt1011->regmap, RT1011_TDM_TOTAL_SET, 0x0240); + regmap_write(rt1011->regmap, RT1011_TDM1_SET_2, 0x8); + regmap_write(rt1011->regmap, RT1011_TDM1_SET_1, 0x10a2); + regmap_write(rt1011->regmap, RT1011_ADCDAT_OUT_SOURCE, 0x4); + break; + default: + dev_info(component->dev, "I2S Reference: Do nothing\n"); + } + + return 0; +} + +static int rt1011_i2s_ref_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct rt1011_priv *rt1011 = + snd_soc_component_get_drvdata(component); + + ucontrol->value.enumerated.item[0] = rt1011->i2s_ref; + + return 0; +} static const struct snd_kcontrol_new rt1011_snd_controls[] = { /* I2S Data In Selection */ @@ -1358,7 +1399,8 @@ static const struct snd_kcontrol_new rt1011_snd_controls[] = { SOC_SINGLE("R0 Temperature", RT1011_STP_INITIAL_RESISTANCE_TEMP, 2, 255, 0), /* I2S Reference */ - SOC_ENUM("I2S Reference", rt1011_i2s_ref_enum), + SOC_ENUM_EXT("I2S Reference", rt1011_i2s_ref_enum, + rt1011_i2s_ref_get, rt1011_i2s_ref_put), }; static int rt1011_is_sys_clk_from_pll(struct snd_soc_dapm_widget *source, @@ -2017,6 +2059,7 @@ static int rt1011_probe(struct snd_soc_component *component) schedule_work(&rt1011->cali_work); + rt1011->i2s_ref = 0; rt1011->bq_drc_params = devm_kcalloc(component->dev, RT1011_ADVMODE_NUM, sizeof(struct rt1011_bq_drc_params *), GFP_KERNEL); diff --git a/sound/soc/codecs/rt1011.h b/sound/soc/codecs/rt1011.h index 68fadc15fa8c..4d6e7492d99c 100644 --- a/sound/soc/codecs/rt1011.h +++ b/sound/soc/codecs/rt1011.h @@ -654,6 +654,12 @@ enum { RT1011_AIFS }; +enum { + RT1011_I2S_REF_NONE, + RT1011_I2S_REF_LEFT_CH, + RT1011_I2S_REF_RIGHT_CH, +}; + /* BiQual & DRC related settings */ #define RT1011_BQ_DRC_NUM 128 struct rt1011_bq_drc_params { @@ -692,6 +698,7 @@ struct rt1011_priv { unsigned int r0_reg, cali_done; unsigned int r0_calib, temperature_calib; int recv_spk_mode; + int i2s_ref; }; #endif /* end of _RT1011_H_ */ -- cgit v1.2.3 From a3774a2a6544a7a4a85186e768afc07044aa507f Mon Sep 17 00:00:00 2001 From: Derek Fang Date: Tue, 9 Nov 2021 17:54:49 +0800 Subject: ASoC: rt5682: Avoid the unexpected IRQ event during going to suspend When the system suspends, the codec driver will set SAR to power saving mode if a headset is plugged in. There is a chance to generate an unexpected IRQ, and leads to issues after resuming such as noise from OMTP type headsets. Signed-off-by: Derek Fang Link: https://lore.kernel.org/r/20211109095450.12950-1-derek.fang@realtek.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt5682.c | 15 +++++++++------ 1 file changed, 9 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 78b4cb5fb6c8..a486ac268c33 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -48,6 +48,7 @@ static const struct reg_sequence patch_list[] = { {RT5682_SAR_IL_CMD_6, 0x0110}, {RT5682_CHARGE_PUMP_1, 0x0210}, {RT5682_HP_LOGIC_CTRL_2, 0x0007}, + {RT5682_SAR_IL_CMD_2, 0xac00}, }; void rt5682_apply_patch_list(struct rt5682_priv *rt5682, struct device *dev) @@ -2943,9 +2944,6 @@ static int rt5682_suspend(struct snd_soc_component *component) cancel_delayed_work_sync(&rt5682->jack_detect_work); cancel_delayed_work_sync(&rt5682->jd_check_work); if (rt5682->hs_jack && rt5682->jack_type == SND_JACK_HEADSET) { - snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_1, - RT5682_MB1_PATH_MASK | RT5682_MB2_PATH_MASK, - RT5682_CTRL_MB1_REG | RT5682_CTRL_MB2_REG); val = snd_soc_component_read(component, RT5682_CBJ_CTRL_2) & RT5682_JACK_TYPE_MASK; @@ -2967,10 +2965,15 @@ static int rt5682_suspend(struct snd_soc_component *component) /* enter SAR ADC power saving mode */ snd_soc_component_update_bits(component, RT5682_SAR_IL_CMD_1, RT5682_SAR_BUTT_DET_MASK | RT5682_SAR_BUTDET_MODE_MASK | - RT5682_SAR_BUTDET_RST_MASK | RT5682_SAR_SEL_MB1_MB2_MASK, 0); + RT5682_SAR_SEL_MB1_MB2_MASK, 0); + usleep_range(5000, 6000); + snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_1, + RT5682_MB1_PATH_MASK | RT5682_MB2_PATH_MASK, + RT5682_CTRL_MB1_REG | RT5682_CTRL_MB2_REG); + usleep_range(10000, 12000); snd_soc_component_update_bits(component, RT5682_SAR_IL_CMD_1, - RT5682_SAR_BUTT_DET_MASK | RT5682_SAR_BUTDET_MODE_MASK | RT5682_SAR_BUTDET_RST_MASK, - RT5682_SAR_BUTT_DET_EN | RT5682_SAR_BUTDET_POW_SAV | RT5682_SAR_BUTDET_RST_NORMAL); + RT5682_SAR_BUTT_DET_MASK | RT5682_SAR_BUTDET_MODE_MASK, + RT5682_SAR_BUTT_DET_EN | RT5682_SAR_BUTDET_POW_SAV); } regcache_cache_only(rt5682->regmap, true); -- cgit v1.2.3 From 2cd9b0ef82d936623d789bb3fbb6fcf52c500367 Mon Sep 17 00:00:00 2001 From: Derek Fang Date: Tue, 9 Nov 2021 17:54:50 +0800 Subject: ASoC: rt5682: Re-detect the combo jack after resuming Sometimes, end-users change the jack type under suspending, so it needs to re-detect the combo jack type after resuming to avoid any unexpected behaviors. Signed-off-by: Derek Fang Link: https://lore.kernel.org/r/20211109095450.12950-2-derek.fang@realtek.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt5682-i2c.c | 1 + sound/soc/codecs/rt5682.c | 23 ++++++++++++++++++++--- sound/soc/codecs/rt5682.h | 1 + 3 files changed, 22 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt5682-i2c.c b/sound/soc/codecs/rt5682-i2c.c index 983347b65127..20e0f90ea498 100644 --- a/sound/soc/codecs/rt5682-i2c.c +++ b/sound/soc/codecs/rt5682-i2c.c @@ -198,6 +198,7 @@ static int rt5682_i2c_probe(struct i2c_client *i2c, } mutex_init(&rt5682->calibrate_mutex); + mutex_init(&rt5682->jdet_mutex); rt5682_calibrate(rt5682); rt5682_apply_patch_list(rt5682, &i2c->dev); diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index a486ac268c33..04cb747c2b12 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -49,6 +49,7 @@ static const struct reg_sequence patch_list[] = { {RT5682_CHARGE_PUMP_1, 0x0210}, {RT5682_HP_LOGIC_CTRL_2, 0x0007}, {RT5682_SAR_IL_CMD_2, 0xac00}, + {RT5682_CBJ_CTRL_7, 0x0104}, }; void rt5682_apply_patch_list(struct rt5682_priv *rt5682, struct device *dev) @@ -941,6 +942,10 @@ int rt5682_headset_detect(struct snd_soc_component *component, int jack_insert) snd_soc_component_update_bits(component, RT5682_HP_CHARGE_PUMP_1, RT5682_OSW_L_MASK | RT5682_OSW_R_MASK, 0); + rt5682_enable_push_button_irq(component, false); + snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_1, + RT5682_TRIG_JD_MASK, RT5682_TRIG_JD_LOW); + usleep_range(55000, 60000); snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_1, RT5682_TRIG_JD_MASK, RT5682_TRIG_JD_HIGH); @@ -1093,6 +1098,7 @@ void rt5682_jack_detect_handler(struct work_struct *work) while (!rt5682->component->card->instantiated) usleep_range(10000, 15000); + mutex_lock(&rt5682->jdet_mutex); mutex_lock(&rt5682->calibrate_mutex); val = snd_soc_component_read(rt5682->component, RT5682_AJD1_CTRL) @@ -1166,6 +1172,7 @@ void rt5682_jack_detect_handler(struct work_struct *work) } mutex_unlock(&rt5682->calibrate_mutex); + mutex_unlock(&rt5682->jdet_mutex); } EXPORT_SYMBOL_GPL(rt5682_jack_detect_handler); @@ -1515,6 +1522,7 @@ static int rt5682_hp_event(struct snd_soc_dapm_widget *w, { struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); switch (event) { case SND_SOC_DAPM_PRE_PMU: @@ -1526,12 +1534,17 @@ static int rt5682_hp_event(struct snd_soc_dapm_widget *w, RT5682_DEPOP_1, 0x60, 0x60); snd_soc_component_update_bits(component, RT5682_DAC_ADC_DIG_VOL1, 0x00c0, 0x0080); + + mutex_lock(&rt5682->jdet_mutex); + snd_soc_component_update_bits(component, RT5682_HP_CTRL_2, RT5682_HP_C2_DAC_L_EN | RT5682_HP_C2_DAC_R_EN, RT5682_HP_C2_DAC_L_EN | RT5682_HP_C2_DAC_R_EN); usleep_range(5000, 10000); snd_soc_component_update_bits(component, RT5682_CHARGE_PUMP_1, RT5682_CP_SW_SIZE_MASK, RT5682_CP_SW_SIZE_L); + + mutex_unlock(&rt5682->jdet_mutex); break; case SND_SOC_DAPM_POST_PMD: @@ -2943,7 +2956,7 @@ static int rt5682_suspend(struct snd_soc_component *component) cancel_delayed_work_sync(&rt5682->jack_detect_work); cancel_delayed_work_sync(&rt5682->jd_check_work); - if (rt5682->hs_jack && rt5682->jack_type == SND_JACK_HEADSET) { + if (rt5682->hs_jack && (rt5682->jack_type & SND_JACK_HEADSET) == SND_JACK_HEADSET) { val = snd_soc_component_read(component, RT5682_CBJ_CTRL_2) & RT5682_JACK_TYPE_MASK; @@ -2974,6 +2987,8 @@ static int rt5682_suspend(struct snd_soc_component *component) snd_soc_component_update_bits(component, RT5682_SAR_IL_CMD_1, RT5682_SAR_BUTT_DET_MASK | RT5682_SAR_BUTDET_MODE_MASK, RT5682_SAR_BUTT_DET_EN | RT5682_SAR_BUTDET_POW_SAV); + snd_soc_component_update_bits(component, RT5682_HP_CHARGE_PUMP_1, + RT5682_OSW_L_MASK | RT5682_OSW_R_MASK, 0); } regcache_cache_only(rt5682->regmap, true); @@ -2991,10 +3006,11 @@ static int rt5682_resume(struct snd_soc_component *component) regcache_cache_only(rt5682->regmap, false); regcache_sync(rt5682->regmap); - if (rt5682->hs_jack && rt5682->jack_type == SND_JACK_HEADSET) { + if (rt5682->hs_jack && (rt5682->jack_type & SND_JACK_HEADSET) == SND_JACK_HEADSET) { snd_soc_component_update_bits(component, RT5682_SAR_IL_CMD_1, RT5682_SAR_BUTDET_MODE_MASK | RT5682_SAR_SEL_MB1_MB2_MASK, RT5682_SAR_BUTDET_POW_NORM | RT5682_SAR_SEL_MB1_MB2_AUTO); + usleep_range(5000, 6000); snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_1, RT5682_MB1_PATH_MASK | RT5682_MB2_PATH_MASK, RT5682_CTRL_MB1_FSM | RT5682_CTRL_MB2_FSM); @@ -3002,8 +3018,9 @@ static int rt5682_resume(struct snd_soc_component *component) RT5682_PWR_CBJ, RT5682_PWR_CBJ); } + rt5682->jack_type = 0; mod_delayed_work(system_power_efficient_wq, - &rt5682->jack_detect_work, msecs_to_jiffies(250)); + &rt5682->jack_detect_work, msecs_to_jiffies(0)); return 0; } diff --git a/sound/soc/codecs/rt5682.h b/sound/soc/codecs/rt5682.h index d93829c35585..c917c76200ea 100644 --- a/sound/soc/codecs/rt5682.h +++ b/sound/soc/codecs/rt5682.h @@ -1463,6 +1463,7 @@ struct rt5682_priv { int jack_type; int irq_work_delay_time; + struct mutex jdet_mutex; }; extern const char *rt5682_supply_names[RT5682_NUM_SUPPLIES]; -- cgit v1.2.3 From 8c32984bc7da29828260ac514d5d4967f7e8f62d Mon Sep 17 00:00:00 2001 From: AngeloGioacchino Del Regno Date: Thu, 11 Nov 2021 17:11:08 +0100 Subject: ASoC: mediatek: mt8173: Fix debugfs registration for components When registering the mt8173-afe-pcm driver, we are also adding two components: one is for the PCM DAIs and one is for the HDMI DAIs, but when debugfs is enabled, we're getting the following issue: [ 17.279176] debugfs: Directory '11220000.audio-controller' with parent 'mtk-rt5650' already present! [ 17.288345] debugfs: Directory '11220000.audio-controller' with parent 'mtk-rt5650' already present! To overcome to that without any potentially big rewrite of this driver, similarly to what was done in mt8195-afe-pcm, add a debugfs_prefix to the components before actually adding them. Signed-off-by: AngeloGioacchino Del Regno Link: https://lore.kernel.org/r/20211111161108.502344-1-angelogioacchino.delregno@collabora.com Signed-off-by: Mark Brown --- sound/soc/mediatek/mt8173/mt8173-afe-pcm.c | 51 +++++++++++++++++++++++++----- 1 file changed, 43 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c b/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c index 6350390414d4..31494930433f 100644 --- a/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c +++ b/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c @@ -1054,6 +1054,7 @@ static int mt8173_afe_pcm_dev_probe(struct platform_device *pdev) int irq_id; struct mtk_base_afe *afe; struct mt8173_afe_private *afe_priv; + struct snd_soc_component *comp_pcm, *comp_hdmi; ret = dma_set_mask_and_coherent(&pdev->dev, DMA_BIT_MASK(33)); if (ret) @@ -1142,23 +1143,55 @@ static int mt8173_afe_pcm_dev_probe(struct platform_device *pdev) if (ret) goto err_pm_disable; - ret = devm_snd_soc_register_component(&pdev->dev, - &mt8173_afe_pcm_dai_component, - mt8173_afe_pcm_dais, - ARRAY_SIZE(mt8173_afe_pcm_dais)); + comp_pcm = devm_kzalloc(&pdev->dev, sizeof(*comp_pcm), GFP_KERNEL); + if (!comp_pcm) { + ret = -ENOMEM; + goto err_pm_disable; + } + + ret = snd_soc_component_initialize(comp_pcm, + &mt8173_afe_pcm_dai_component, + &pdev->dev); if (ret) goto err_pm_disable; - ret = devm_snd_soc_register_component(&pdev->dev, - &mt8173_afe_hdmi_dai_component, - mt8173_afe_hdmi_dais, - ARRAY_SIZE(mt8173_afe_hdmi_dais)); +#ifdef CONFIG_DEBUG_FS + comp_pcm->debugfs_prefix = "pcm"; +#endif + + ret = snd_soc_add_component(comp_pcm, + mt8173_afe_pcm_dais, + ARRAY_SIZE(mt8173_afe_pcm_dais)); + if (ret) + goto err_pm_disable; + + comp_hdmi = devm_kzalloc(&pdev->dev, sizeof(*comp_hdmi), GFP_KERNEL); + if (!comp_hdmi) { + ret = -ENOMEM; + goto err_pm_disable; + } + + ret = snd_soc_component_initialize(comp_hdmi, + &mt8173_afe_hdmi_dai_component, + &pdev->dev); if (ret) goto err_pm_disable; +#ifdef CONFIG_DEBUG_FS + comp_hdmi->debugfs_prefix = "hdmi"; +#endif + + ret = snd_soc_add_component(comp_hdmi, + mt8173_afe_hdmi_dais, + ARRAY_SIZE(mt8173_afe_hdmi_dais)); + if (ret) + goto err_cleanup_components; + dev_info(&pdev->dev, "MT8173 AFE driver initialized.\n"); return 0; +err_cleanup_components: + snd_soc_unregister_component(&pdev->dev); err_pm_disable: pm_runtime_disable(&pdev->dev); return ret; @@ -1166,6 +1199,8 @@ err_pm_disable: static int mt8173_afe_pcm_dev_remove(struct platform_device *pdev) { + snd_soc_unregister_component(&pdev->dev); + pm_runtime_disable(&pdev->dev); if (!pm_runtime_status_suspended(&pdev->dev)) mt8173_afe_runtime_suspend(&pdev->dev); -- cgit v1.2.3 From 1218f06cb3c6e2c51699998bc17c0d9a41ab37a6 Mon Sep 17 00:00:00 2001 From: Arnd Bergmann Date: Mon, 8 Nov 2021 12:11:14 +0100 Subject: ASoC: SOF: build compression interface into snd_sof.ko With CONFIG_SND_SOC_SOF_COMPRESS=m, the compression code is not built into a the main SOF driver when that is built-in: x86_64-linux-ld: sound/soc/sof/ipc.o: in function `ipc_stream_message': ipc.c:(.text+0x5a2): undefined reference to `snd_sof_compr_fragment_elapsed' x86_64-linux-ld: sound/soc/sof/topology.o: in function `sof_dai_load': topology.c:(.text+0x32d1): undefined reference to `snd_sof_compr_init_elapsed_work' x86_64-linux-ld: topology.c:(.text+0x32e1): undefined reference to `snd_sof_compr_init_elapsed_work' Make this a 'bool' symbol so it just decides whether the code gets built at all. Fixes: 858f7a5c45ca ("ASoC: SOF: Introduce fragment elapsed notification API") Signed-off-by: Arnd Bergmann Tested-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20211108111132.3800548-1-arnd@kernel.org Signed-off-by: Mark Brown --- sound/soc/sof/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/sof/Kconfig b/sound/soc/sof/Kconfig index 6bb4db87af03..041c54639c4d 100644 --- a/sound/soc/sof/Kconfig +++ b/sound/soc/sof/Kconfig @@ -47,7 +47,7 @@ config SND_SOC_SOF_OF Say Y if you need this option. If unsure select "N". config SND_SOC_SOF_COMPRESS - tristate + bool select SND_SOC_COMPRESS config SND_SOC_SOF_DEBUG_PROBES -- cgit v1.2.3 From 2ce1b21cb3326e12af3c72c47e1d294b19d73947 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 8 Nov 2021 13:22:55 +0900 Subject: ASoC: rsnd: fixup DMAEngine API commit d5bb69dc54ec1 ("ASoC: sh: rcar: dma: : use proper DMAENGINE API for termination") updated DMAEngine API _all() to _sync(), but it should be _async(). _all() and _async() are almost same, the difference is only return error code. _sync() will call dmaengine_synchronize() and will be kernel panic. This patch is needed for v5.15 or later. [ 27.293264] BUG: scheduling while atomic: irq/130-ec70000/131/0x00000003 [ 27.300084] 2 locks held by irq/130-ec70000/131: [ 27.304743] #0: ffff0004c274d908 (&group->lock){....}-{2:2}, at: _snd_pcm_stream_lock_irqsave+0x48/0x54 [ 27.314344] #1: ffff0004c1788c60 (&priv->lock#2){....}-{2:2}, at: rsnd_soc_dai_trigger+0x70/0x7bc [ 27.323409] irq event stamp: 206 [ 27.326664] hardirqs last enabled at (205): [] _raw_spin_unlock_irq+0x50/0xa0 [ 27.335529] hardirqs last disabled at (206): [] _raw_spin_lock_irqsave+0xc4/0xd0 [ 27.344564] softirqs last enabled at (0): [] copy_process+0x644/0x1b10 [ 27.352819] softirqs last disabled at (0): [<0000000000000000>] 0x0 [ 27.359142] CPU: 0 PID: 131 Comm: irq/130-ec70000 Not tainted 5.14.0-rc1+ #918 [ 27.366429] Hardware name: Renesas H3ULCB Kingfisher board based on r8a77950 (DT) [ 27.373975] Call trace: [ 27.376442] dump_backtrace+0x0/0x1b4 [ 27.380141] show_stack+0x24/0x30 [ 27.383488] dump_stack_lvl+0x8c/0xb8 [ 27.387184] dump_stack+0x18/0x34 [ 27.390528] __schedule_bug+0x8c/0x9c [ 27.394224] __schedule+0x790/0x8dc [ 27.397746] schedule+0x7c/0x110 [ 27.401003] synchronize_irq+0x94/0xd0 [ 27.404786] rcar_dmac_device_synchronize+0x20/0x2c [ 27.409710] rsnd_dmaen_stop+0x50/0x64 [ 27.413495] rsnd_soc_dai_trigger+0x554/0x7bc [ 27.417890] snd_soc_pcm_dai_trigger+0xe8/0x264 Cc: Fixes: commit d5bb69dc54ec1 ("ASoC: sh: rcar: dma: : use proper DMAENGINE API for termination") Link: https://lore.kernel.org/r/TY2PR01MB3692889E1A7476C4322CC296D8AE9@TY2PR01MB3692.jpnprd01.prod.outlook.com Reported-by: Yoshihiro Shimoda Acked-by: Wolfram Sang Signed-off-by: Kuninori Morimoto Link: https://lore.kernel.org/r/87mtmfz36o.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/sh/rcar/dma.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c index 16c6e0265749..03e0d4eca781 100644 --- a/sound/soc/sh/rcar/dma.c +++ b/sound/soc/sh/rcar/dma.c @@ -102,7 +102,7 @@ static int rsnd_dmaen_stop(struct rsnd_mod *mod, struct rsnd_dmaen *dmaen = rsnd_dma_to_dmaen(dma); if (dmaen->chan) - dmaengine_terminate_sync(dmaen->chan); + dmaengine_terminate_async(dmaen->chan); return 0; } -- cgit v1.2.3 From 174a7fb3859ae75b0f0e35ef852459d8882b55b5 Mon Sep 17 00:00:00 2001 From: Werner Sembach Date: Fri, 12 Nov 2021 12:07:04 +0100 Subject: ALSA: hda/realtek: Add quirk for ASRock NUC Box 1100 This applies a SND_PCI_QUIRK(...) to the ASRock NUC Box 1100 series. This fixes the issue of the headphone jack not being detected unless warm rebooted from a certain other OS. When booting a certain other OS some coeff settings are changed that enable the audio jack. These settings are preserved on a warm reboot and can be easily dumped. The relevant indexes and values where gathered by naively diff-ing and reading a working and a non-working coeff dump. Signed-off-by: Werner Sembach Cc: Link: https://lore.kernel.org/r/20211112110704.1022501-1-wse@tuxedocomputers.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 27 +++++++++++++++++++++++++++ 1 file changed, 27 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 2f1727faec69..701c80ed83dc 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6521,6 +6521,27 @@ static void alc256_fixup_tongfang_reset_persistent_settings(struct hda_codec *co alc_write_coef_idx(codec, 0x45, 0x5089); } +static const struct coef_fw alc233_fixup_no_audio_jack_coefs[] = { + WRITE_COEF(0x1a, 0x9003), WRITE_COEF(0x1b, 0x0e2b), WRITE_COEF(0x37, 0xfe06), + WRITE_COEF(0x38, 0x4981), WRITE_COEF(0x45, 0xd489), WRITE_COEF(0x46, 0x0074), + WRITE_COEF(0x49, 0x0149), + {} +}; + +static void alc233_fixup_no_audio_jack(struct hda_codec *codec, + const struct hda_fixup *fix, + int action) +{ + /* + * The audio jack input and output is not detected on the ASRock NUC Box + * 1100 series when cold booting without this fix. Warm rebooting from a + * certain other OS makes the audio functional, as COEF settings are + * preserved in this case. This fix sets these altered COEF values as + * the default. + */ + alc_process_coef_fw(codec, alc233_fixup_no_audio_jack_coefs); +} + enum { ALC269_FIXUP_GPIO2, ALC269_FIXUP_SONY_VAIO, @@ -6740,6 +6761,7 @@ enum { ALC287_FIXUP_13S_GEN2_SPEAKERS, ALC256_FIXUP_TONGFANG_RESET_PERSISTENT_SETTINGS, ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE, + ALC233_FIXUP_NO_AUDIO_JACK, }; static const struct hda_fixup alc269_fixups[] = { @@ -8460,6 +8482,10 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC, }, + [ALC233_FIXUP_NO_AUDIO_JACK] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc233_fixup_no_audio_jack, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -8894,6 +8920,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x511e, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), SND_PCI_QUIRK(0x17aa, 0x511f, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD), + SND_PCI_QUIRK(0x1849, 0x1233, "ASRock NUC Box 1100", ALC233_FIXUP_NO_AUDIO_JACK), SND_PCI_QUIRK(0x19e5, 0x3204, "Huawei MACH-WX9", ALC256_FIXUP_HUAWEI_MACH_WX9_PINS), SND_PCI_QUIRK(0x1b35, 0x1235, "CZC B20", ALC269_FIXUP_CZC_B20), SND_PCI_QUIRK(0x1b35, 0x1236, "CZC TMI", ALC269_FIXUP_CZC_TMI), -- cgit v1.2.3 From bd5e2c22a9cfe7c3735d71920dc4a286348c61d2 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 12 Nov 2021 11:31:37 +0100 Subject: ALSA: cmipci: Drop stale variable assignment Since the recent code refactoring using devres, the variable cm in snd_cmipci_probe() is no longer referred. Fixes: 87e082ad84a7 ("ALSA: cmipci: Allocate resources with device-managed APIs") Reported-by: kernel test robot Link: https://lore.kernel.org/r/cc6383a2-cafb-ffe7-0b4f-27a310a1005c@intel.com Link: https://lore.kernel.org/r/20211112103137.9504-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/cmipci.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c index ea20236f35db..9a678b5cf285 100644 --- a/sound/pci/cmipci.c +++ b/sound/pci/cmipci.c @@ -3218,7 +3218,6 @@ static int snd_cmipci_probe(struct pci_dev *pci, { static int dev; struct snd_card *card; - struct cmipci *cm; int err; if (dev >= SNDRV_CARDS) @@ -3229,10 +3228,9 @@ static int snd_cmipci_probe(struct pci_dev *pci, } err = snd_devm_card_new(&pci->dev, index[dev], id[dev], THIS_MODULE, - sizeof(*cm), &card); + sizeof(struct cmipci), &card); if (err < 0) return err; - cm = card->private_data; switch (pci->device) { case PCI_DEVICE_ID_CMEDIA_CM8738: -- cgit v1.2.3 From a6e849d0007b374fc7fbb18d55941c77aa7c3923 Mon Sep 17 00:00:00 2001 From: Simon Trimmer Date: Mon, 15 Nov 2021 12:01:54 +0000 Subject: ASoC: wm_adsp: wm_adsp_control_add() error: uninitialized symbol 'ret' This patch fixes the static analysis warning as it is correctly indicating a possible code path, it cannot know that for the affected firmware versions subname would always be NULL. Reported-by: kernel test robot Reported-by: Dan Carpenter Signed-off-by: Simon Trimmer Acked-by: Charles Keepax Link: https://lore.kernel.org/r/20211115120154.56782-1-simont@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index d4f0d72cbcc8..6cb01a8e08fb 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -617,8 +617,9 @@ static int wm_adsp_control_add(struct cs_dsp_coeff_ctl *cs_ctl) switch (cs_dsp->fw_ver) { case 0: case 1: - snprintf(name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, "%s %s %x", - cs_dsp->name, region_name, cs_ctl->alg_region.alg); + ret = scnprintf(name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, + "%s %s %x", cs_dsp->name, region_name, + cs_ctl->alg_region.alg); break; case 2: ret = scnprintf(name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, -- cgit v1.2.3 From 2f20640491edda3c03eb6b899d0b92630d3d4c63 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Tue, 16 Nov 2021 11:47:17 +0000 Subject: ASoC: qdsp6: qdsp6: q6prm: handle clk disable correctly Q6PRM clks need to be disabled using PRM_CMD_RELEASE_HW_RSC dsp command rather then using PRM_CMD_RSP_REQUEST_HW_RSC cmd with rate set to zero. DSP will throw errors if we try to disable the clock using existing code. Fix this by properly handling the clk release. Fixes: 9a0e5d6fb16f ("ASoC: qdsp6: audioreach: add q6prm support") Signed-off-by: Srinivas Kandagatla Link: https://lore.kernel.org/r/20211116114721.12517-2-srinivas.kandagatla@linaro.org Signed-off-by: Mark Brown --- sound/soc/qcom/qdsp6/audioreach.h | 4 +++ sound/soc/qcom/qdsp6/q6prm.c | 53 +++++++++++++++++++++++++++++++++++++-- 2 files changed, 55 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/qcom/qdsp6/audioreach.h b/sound/soc/qcom/qdsp6/audioreach.h index 4f693a2660b5..3ee8bfcd0121 100644 --- a/sound/soc/qcom/qdsp6/audioreach.h +++ b/sound/soc/qcom/qdsp6/audioreach.h @@ -550,6 +550,10 @@ struct audio_hw_clk_cfg { uint32_t clock_root; } __packed; +struct audio_hw_clk_rel_cfg { + uint32_t clock_id; +} __packed; + #define PARAM_ID_HW_EP_POWER_MODE_CFG 0x8001176 #define AR_HW_EP_POWER_MODE_0 0 /* default */ #define AR_HW_EP_POWER_MODE_1 1 /* XO Shutdown allowed */ diff --git a/sound/soc/qcom/qdsp6/q6prm.c b/sound/soc/qcom/qdsp6/q6prm.c index 82c40f2d4e1d..cda33ded29be 100644 --- a/sound/soc/qcom/qdsp6/q6prm.c +++ b/sound/soc/qcom/qdsp6/q6prm.c @@ -42,6 +42,12 @@ struct prm_cmd_request_rsc { struct audio_hw_clk_cfg clock_id; } __packed; +struct prm_cmd_release_rsc { + struct apm_module_param_data param_data; + uint32_t num_clk_id; + struct audio_hw_clk_rel_cfg clock_id; +} __packed; + static int q6prm_send_cmd_sync(struct q6prm *prm, struct gpr_pkt *pkt, uint32_t rsp_opcode) { return audioreach_send_cmd_sync(prm->dev, prm->gdev, &prm->result, &prm->lock, @@ -102,8 +108,8 @@ int q6prm_unvote_lpass_core_hw(struct device *dev, uint32_t hw_block_id, uint32_ } EXPORT_SYMBOL_GPL(q6prm_unvote_lpass_core_hw); -int q6prm_set_lpass_clock(struct device *dev, int clk_id, int clk_attr, int clk_root, - unsigned int freq) +static int q6prm_request_lpass_clock(struct device *dev, int clk_id, int clk_attr, int clk_root, + unsigned int freq) { struct q6prm *prm = dev_get_drvdata(dev->parent); struct apm_module_param_data *param_data; @@ -138,6 +144,49 @@ int q6prm_set_lpass_clock(struct device *dev, int clk_id, int clk_attr, int clk_ return rc; } + +static int q6prm_release_lpass_clock(struct device *dev, int clk_id, int clk_attr, int clk_root, + unsigned int freq) +{ + struct q6prm *prm = dev_get_drvdata(dev->parent); + struct apm_module_param_data *param_data; + struct prm_cmd_release_rsc *rel; + gpr_device_t *gdev = prm->gdev; + struct gpr_pkt *pkt; + int rc; + + pkt = audioreach_alloc_cmd_pkt(sizeof(*rel), PRM_CMD_RELEASE_HW_RSC, 0, gdev->svc.id, + GPR_PRM_MODULE_IID); + if (IS_ERR(pkt)) + return PTR_ERR(pkt); + + rel = (void *)pkt + GPR_HDR_SIZE + APM_CMD_HDR_SIZE; + + param_data = &rel->param_data; + + param_data->module_instance_id = GPR_PRM_MODULE_IID; + param_data->error_code = 0; + param_data->param_id = PARAM_ID_RSC_AUDIO_HW_CLK; + param_data->param_size = sizeof(*rel) - APM_MODULE_PARAM_DATA_SIZE; + + rel->num_clk_id = 1; + rel->clock_id.clock_id = clk_id; + + rc = q6prm_send_cmd_sync(prm, pkt, PRM_CMD_RSP_RELEASE_HW_RSC); + + kfree(pkt); + + return rc; +} + +int q6prm_set_lpass_clock(struct device *dev, int clk_id, int clk_attr, int clk_root, + unsigned int freq) +{ + if (freq) + return q6prm_request_lpass_clock(dev, clk_id, clk_attr, clk_attr, freq); + + return q6prm_release_lpass_clock(dev, clk_id, clk_attr, clk_attr, freq); +} EXPORT_SYMBOL_GPL(q6prm_set_lpass_clock); static int prm_callback(struct gpr_resp_pkt *data, void *priv, int op) -- cgit v1.2.3 From 861afeac7990587588d057b2c0b3222331c3da29 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Tue, 16 Nov 2021 11:47:18 +0000 Subject: ASoC: qdsp6: q6routing: Conditionally reset FrontEnd Mixer Stream IDs are reused across multiple BackEnd mixers, do not reset the stream mixers if they are not already set for that particular FrontEnd. Ex: amixer cset iface=MIXER,name='SLIMBUS_0_RX Audio Mixer MultiMedia1' 1 would set the MultiMedia1 steam for SLIMBUS_0_RX, however doing below command will reset previously setup MultiMedia1 stream, because both of them are using MultiMedia1 PCM stream. amixer cset iface=MIXER,name='SLIMBUS_2_RX Audio Mixer MultiMedia1' 0 reset the FrontEnd Mixers conditionally to fix this issue. This is more noticeable in desktop setup, where in alsactl tries to restore the alsa state and overwriting the previous mixer settings. Fixes: e3a33673e845 ("ASoC: qdsp6: q6routing: Add q6routing driver") Signed-off-by: Srinivas Kandagatla Link: https://lore.kernel.org/r/20211116114721.12517-3-srinivas.kandagatla@linaro.org Signed-off-by: Mark Brown --- sound/soc/qcom/qdsp6/q6routing.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/qcom/qdsp6/q6routing.c b/sound/soc/qcom/qdsp6/q6routing.c index 3390ebef9549..243b8179e59d 100644 --- a/sound/soc/qcom/qdsp6/q6routing.c +++ b/sound/soc/qcom/qdsp6/q6routing.c @@ -495,7 +495,11 @@ static int msm_routing_put_audio_mixer(struct snd_kcontrol *kcontrol, session->port_id = be_id; snd_soc_dapm_mixer_update_power(dapm, kcontrol, 1, update); } else { - session->port_id = -1; + if (session->port_id == be_id) { + session->port_id = -1; + return 0; + } + snd_soc_dapm_mixer_update_power(dapm, kcontrol, 0, update); } -- cgit v1.2.3 From 721a94b4352dc8e47bff90b549a0118c39776756 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Tue, 16 Nov 2021 11:47:19 +0000 Subject: ASoC: qdsp6: q6asm: fix q6asm_dai_prepare error handling Error handling in q6asm_dai_prepare() seems to be completely broken, Fix this by handling it properly. Fixes: 2a9e92d371db ("ASoC: qdsp6: q6asm: Add q6asm dai driver") Signed-off-by: Srinivas Kandagatla Link: https://lore.kernel.org/r/20211116114721.12517-4-srinivas.kandagatla@linaro.org Signed-off-by: Mark Brown --- sound/soc/qcom/qdsp6/q6asm-dai.c | 19 +++++++++++++------ 1 file changed, 13 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c index 46f365528d50..b74b67720ef4 100644 --- a/sound/soc/qcom/qdsp6/q6asm-dai.c +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c @@ -269,9 +269,7 @@ static int q6asm_dai_prepare(struct snd_soc_component *component, if (ret < 0) { dev_err(dev, "%s: q6asm_open_write failed\n", __func__); - q6asm_audio_client_free(prtd->audio_client); - prtd->audio_client = NULL; - return -ENOMEM; + goto open_err; } prtd->session_id = q6asm_get_session_id(prtd->audio_client); @@ -279,7 +277,7 @@ static int q6asm_dai_prepare(struct snd_soc_component *component, prtd->session_id, substream->stream); if (ret) { dev_err(dev, "%s: stream reg failed ret:%d\n", __func__, ret); - return ret; + goto routing_err; } if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { @@ -301,10 +299,19 @@ static int q6asm_dai_prepare(struct snd_soc_component *component, } if (ret < 0) dev_info(dev, "%s: CMD Format block failed\n", __func__); + else + prtd->state = Q6ASM_STREAM_RUNNING; - prtd->state = Q6ASM_STREAM_RUNNING; + return ret; - return 0; +routing_err: + q6asm_cmd(prtd->audio_client, prtd->stream_id, CMD_CLOSE); +open_err: + q6asm_unmap_memory_regions(substream->stream, prtd->audio_client); + q6asm_audio_client_free(prtd->audio_client); + prtd->audio_client = NULL; + + return ret; } static int q6asm_dai_trigger(struct snd_soc_component *component, -- cgit v1.2.3 From 0a270471d68533f59c5cfd631a3fce31a3b17144 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Tue, 16 Nov 2021 11:47:20 +0000 Subject: ASoC: qdsp6: q6adm: improve error reporting reset value for port is -1 so printing an hex would not give us very useful debug information, so use %d instead. Signed-off-by: Srinivas Kandagatla Link: https://lore.kernel.org/r/20211116114721.12517-5-srinivas.kandagatla@linaro.org Signed-off-by: Mark Brown --- sound/soc/qcom/qdsp6/q6adm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/qcom/qdsp6/q6adm.c b/sound/soc/qcom/qdsp6/q6adm.c index 3d831b635524..72c5719f1d25 100644 --- a/sound/soc/qcom/qdsp6/q6adm.c +++ b/sound/soc/qcom/qdsp6/q6adm.c @@ -390,7 +390,7 @@ struct q6copp *q6adm_open(struct device *dev, int port_id, int path, int rate, int ret = 0; if (port_id < 0) { - dev_err(dev, "Invalid port_id 0x%x\n", port_id); + dev_err(dev, "Invalid port_id %d\n", port_id); return ERR_PTR(-EINVAL); } @@ -508,7 +508,7 @@ int q6adm_matrix_map(struct device *dev, int path, int port_idx = payload_map.port_id[i]; if (port_idx < 0) { - dev_err(dev, "Invalid port_id 0x%x\n", + dev_err(dev, "Invalid port_id %d\n", payload_map.port_id[i]); kfree(pkt); return -EINVAL; -- cgit v1.2.3 From 6712c2e18c06b0976559fd4bd47774b243038e9c Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Tue, 16 Nov 2021 11:47:21 +0000 Subject: ASoC: qdsp6: q6routing: validate port id before setting up route Validate port id before it starts sending commands to dsp this would make error handling simpler. Signed-off-by: Srinivas Kandagatla Link: https://lore.kernel.org/r/20211116114721.12517-6-srinivas.kandagatla@linaro.org Signed-off-by: Mark Brown --- sound/soc/qcom/qdsp6/q6routing.c | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'sound') diff --git a/sound/soc/qcom/qdsp6/q6routing.c b/sound/soc/qcom/qdsp6/q6routing.c index 243b8179e59d..cd74681e811e 100644 --- a/sound/soc/qcom/qdsp6/q6routing.c +++ b/sound/soc/qcom/qdsp6/q6routing.c @@ -372,6 +372,12 @@ int q6routing_stream_open(int fedai_id, int perf_mode, } session = &routing_data->sessions[stream_id - 1]; + if (session->port_id < 0) { + dev_err(routing_data->dev, "Routing not setup for MultiMedia%d Session\n", + session->fedai_id); + return -EINVAL; + } + pdata = &routing_data->port_data[session->port_id]; mutex_lock(&routing_data->lock); -- cgit v1.2.3 From 7e567b5ae06315ef2d70666b149962e2bb4b97af Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 16 Nov 2021 08:18:12 +0100 Subject: ASoC: topology: Add missing rwsem around snd_ctl_remove() calls snd_ctl_remove() has to be called with card->controls_rwsem held (when called after the card instantiation). This patch add the missing rwsem calls around it. Fixes: 8a9782346dcc ("ASoC: topology: Add topology core") Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20211116071812.18109-1-tiwai@suse.de Signed-off-by: Mark Brown --- sound/soc/soc-topology.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 557e22c5254c..f5b9e66ac3b8 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -2700,6 +2700,7 @@ EXPORT_SYMBOL_GPL(snd_soc_tplg_component_load); /* remove dynamic controls from the component driver */ int snd_soc_tplg_component_remove(struct snd_soc_component *comp) { + struct snd_card *card = comp->card->snd_card; struct snd_soc_dobj *dobj, *next_dobj; int pass = SOC_TPLG_PASS_END; @@ -2707,6 +2708,7 @@ int snd_soc_tplg_component_remove(struct snd_soc_component *comp) while (pass >= SOC_TPLG_PASS_START) { /* remove mixer controls */ + down_write(&card->controls_rwsem); list_for_each_entry_safe(dobj, next_dobj, &comp->dobj_list, list) { @@ -2745,6 +2747,7 @@ int snd_soc_tplg_component_remove(struct snd_soc_component *comp) break; } } + up_write(&card->controls_rwsem); pass--; } -- cgit v1.2.3 From ea157c2ba821dab789a544cd9fbe44dc07036ff8 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Tue, 16 Nov 2021 11:46:21 +0000 Subject: ASoC: codecs: wcd938x: fix volatile register range Interrupt Clear registers WCD938X_INTR_CLEAR_0 - WCD938X_INTR_CLEAR_2 are not marked as volatile. This has resulted in a missing interrupt bug while performing runtime pm. regcache_sync() during runtime pm resume path will write to Interrupt clear registers with previous values which basically clears the pending interrupt and actual interrupt handler never sees this interrupt. This issue is more visible with headset plug-in plug-out case compared to headset button. Fix this by adding the Interrupt clear registers to volatile range Fixes: 8d78602aa87a ("ASoC: codecs: wcd938x: add basic driver") Signed-off-by: Srinivas Kandagatla Link: https://lore.kernel.org/r/20211116114623.11891-2-srinivas.kandagatla@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/wcd938x.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wcd938x.c b/sound/soc/codecs/wcd938x.c index 52de7d14b139..67151c7770c6 100644 --- a/sound/soc/codecs/wcd938x.c +++ b/sound/soc/codecs/wcd938x.c @@ -1174,6 +1174,9 @@ static bool wcd938x_readonly_register(struct device *dev, unsigned int reg) case WCD938X_DIGITAL_INTR_STATUS_0: case WCD938X_DIGITAL_INTR_STATUS_1: case WCD938X_DIGITAL_INTR_STATUS_2: + case WCD938X_DIGITAL_INTR_CLEAR_0: + case WCD938X_DIGITAL_INTR_CLEAR_1: + case WCD938X_DIGITAL_INTR_CLEAR_2: case WCD938X_DIGITAL_SWR_HM_TEST_0: case WCD938X_DIGITAL_SWR_HM_TEST_1: case WCD938X_DIGITAL_EFUSE_T_DATA_0: -- cgit v1.2.3 From 006ea27c4e7037369085755c7b5389effa508c04 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Tue, 16 Nov 2021 11:46:22 +0000 Subject: ASoC: codecs: wcd934x: return error code correctly from hw_params Error returned from wcd934x_slim_set_hw_params() are not passed to upper layer, this could be misleading to the user which can start sending stream leading to unnecessary errors. Fix this by properly returning the errors. Fixes: a61f3b4f476e ("ASoC: wcd934x: add support to wcd9340/wcd9341 codec") Signed-off-by: Srinivas Kandagatla Link: https://lore.kernel.org/r/20211116114623.11891-3-srinivas.kandagatla@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/wcd934x.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wcd934x.c b/sound/soc/codecs/wcd934x.c index c496b359f2f4..4f568abd59e2 100644 --- a/sound/soc/codecs/wcd934x.c +++ b/sound/soc/codecs/wcd934x.c @@ -1896,9 +1896,8 @@ static int wcd934x_hw_params(struct snd_pcm_substream *substream, } wcd->dai[dai->id].sconfig.rate = params_rate(params); - wcd934x_slim_set_hw_params(wcd, &wcd->dai[dai->id], substream->stream); - return 0; + return wcd934x_slim_set_hw_params(wcd, &wcd->dai[dai->id], substream->stream); } static int wcd934x_hw_free(struct snd_pcm_substream *substream, -- cgit v1.2.3 From cb04d8cd0bb0b82acc34cc73cb33ae77cbfb020d Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Tue, 16 Nov 2021 11:46:23 +0000 Subject: ASoC: codecs: lpass-rx-macro: fix HPHR setting CLSH mask For some reason we ended up using snd_soc_component_write_field for HPHL and snd_soc_component_update_bits for HPHR, so fix this. Fixes: af3d54b99764 ("ASoC: codecs: lpass-rx-macro: add support for lpass rx macro") Signed-off-by: Srinivas Kandagatla Link: https://lore.kernel.org/r/20211116114623.11891-4-srinivas.kandagatla@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/lpass-rx-macro.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/lpass-rx-macro.c b/sound/soc/codecs/lpass-rx-macro.c index 2bed5cf229be..aec5127260fd 100644 --- a/sound/soc/codecs/lpass-rx-macro.c +++ b/sound/soc/codecs/lpass-rx-macro.c @@ -2188,7 +2188,7 @@ static int rx_macro_config_classh(struct snd_soc_component *component, snd_soc_component_update_bits(component, CDC_RX_CLSH_DECAY_CTRL, CDC_RX_CLSH_DECAY_RATE_MASK, 0x0); - snd_soc_component_update_bits(component, + snd_soc_component_write_field(component, CDC_RX_RX1_RX_PATH_CFG0, CDC_RX_RXn_CLSH_EN_MASK, 0x1); break; -- cgit v1.2.3 From 424fe7edbed18d47f7b97f7e1322a6f8969b77ae Mon Sep 17 00:00:00 2001 From: Olivier Moysan Date: Wed, 17 Nov 2021 11:44:04 +0100 Subject: ASoC: stm32: i2s: fix 32 bits channel length without mclk Fix divider calculation in the case of 32 bits channel configuration, when no master clock is used. Fixes: e4e6ec7b127c ("ASoC: stm32: Add I2S driver") Signed-off-by: Olivier Moysan Link: https://lore.kernel.org/r/20211117104404.3832-1-olivier.moysan@foss.st.com Signed-off-by: Mark Brown --- sound/soc/stm/stm32_i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/stm/stm32_i2s.c b/sound/soc/stm/stm32_i2s.c index 6254bacad6eb..717f45a83445 100644 --- a/sound/soc/stm/stm32_i2s.c +++ b/sound/soc/stm/stm32_i2s.c @@ -700,7 +700,7 @@ static int stm32_i2s_configure_clock(struct snd_soc_dai *cpu_dai, if (ret < 0) return ret; - nb_bits = frame_len * ((cgfr & I2S_CGFR_CHLEN) + 1); + nb_bits = frame_len * (FIELD_GET(I2S_CGFR_CHLEN, cgfr) + 1); ret = stm32_i2s_calc_clk_div(i2s, i2s_clock_rate, (nb_bits * rate)); if (ret) -- cgit v1.2.3 From 05ec7161084565365ecf267e9909a897a95f243a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 18 Nov 2021 08:16:36 +0100 Subject: ALSA: hda/realtek: Fix LED on HP ProBook 435 G7 HP ProBook 435 G7 (SSID 103c:8735) needs the similar quirk as another HP ProBook for enabling the mute and the mic-mute LEDs. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=215021 Cc: Link: https://lore.kernel.org/r/20211118071636.14738-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 701c80ed83dc..9ce7457533c9 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8665,6 +8665,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8728, "HP EliteBook 840 G7", ALC285_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8729, "HP", ALC285_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8730, "HP ProBook 445 G7", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), + SND_PCI_QUIRK(0x103c, 0x8735, "HP ProBook 435 G7", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), SND_PCI_QUIRK(0x103c, 0x8736, "HP", ALC285_FIXUP_HP_GPIO_AMP_INIT), SND_PCI_QUIRK(0x103c, 0x8760, "HP", ALC285_FIXUP_HP_MUTE_LED), SND_PCI_QUIRK(0x103c, 0x877a, "HP", ALC285_FIXUP_HP_MUTE_LED), -- cgit v1.2.3 From 884c6cb3b7030f75c46e55b9e625d2372708c306 Mon Sep 17 00:00:00 2001 From: Sameer Pujar Date: Thu, 18 Nov 2021 12:36:56 +0530 Subject: ASoC: tegra: Fix wrong value type in ADMAIF The enum controls are expected to use enumerated value type. Update relevant references in control get/put callbacks. Fixes: f74028e159bb ("ASoC: tegra: Add Tegra210 based ADMAIF driver") Suggested-by: Takashi Iwai Signed-off-by: Sameer Pujar Reviewed-by: Takashi Iwai Link: https://lore.kernel.org/r/1637219231-406-2-git-send-email-spujar@nvidia.com Signed-off-by: Mark Brown --- sound/soc/tegra/tegra210_admaif.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/tegra/tegra210_admaif.c b/sound/soc/tegra/tegra210_admaif.c index bcccdf3ddc52..6febe80cfa6f 100644 --- a/sound/soc/tegra/tegra210_admaif.c +++ b/sound/soc/tegra/tegra210_admaif.c @@ -430,7 +430,7 @@ static int tegra_admaif_get_control(struct snd_kcontrol *kcontrol, struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); struct soc_enum *ec = (struct soc_enum *)kcontrol->private_value; struct tegra_admaif *admaif = snd_soc_component_get_drvdata(cmpnt); - long *uctl_val = &ucontrol->value.integer.value[0]; + unsigned int *uctl_val = &ucontrol->value.enumerated.item[0]; if (strstr(kcontrol->id.name, "Playback Mono To Stereo")) *uctl_val = admaif->mono_to_stereo[ADMAIF_TX_PATH][ec->reg]; @@ -450,7 +450,7 @@ static int tegra_admaif_put_control(struct snd_kcontrol *kcontrol, struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); struct soc_enum *ec = (struct soc_enum *)kcontrol->private_value; struct tegra_admaif *admaif = snd_soc_component_get_drvdata(cmpnt); - int value = ucontrol->value.integer.value[0]; + unsigned int value = ucontrol->value.enumerated.item[0]; if (strstr(kcontrol->id.name, "Playback Mono To Stereo")) admaif->mono_to_stereo[ADMAIF_TX_PATH][ec->reg] = value; -- cgit v1.2.3 From 8a2c2fa0c5331445c801e9241f2bb4e0e2a895a8 Mon Sep 17 00:00:00 2001 From: Sameer Pujar Date: Thu, 18 Nov 2021 12:36:57 +0530 Subject: ASoC: tegra: Fix wrong value type in I2S The enum controls are expected to use enumerated value type. Update relevant references in control get/put callbacks. Fixes: c0bfa98349d1 ("ASoC: tegra: Add Tegra210 based I2S driver") Suggested-by: Takashi Iwai Signed-off-by: Sameer Pujar Reviewed-by: Takashi Iwai Link: https://lore.kernel.org/r/1637219231-406-3-git-send-email-spujar@nvidia.com Signed-off-by: Mark Brown --- sound/soc/tegra/tegra210_i2s.c | 42 ++++++++++++++++++++++++------------------ 1 file changed, 24 insertions(+), 18 deletions(-) (limited to 'sound') diff --git a/sound/soc/tegra/tegra210_i2s.c b/sound/soc/tegra/tegra210_i2s.c index 45f31ccb49d8..5c304612769f 100644 --- a/sound/soc/tegra/tegra210_i2s.c +++ b/sound/soc/tegra/tegra210_i2s.c @@ -317,24 +317,27 @@ static int tegra210_i2s_get_control(struct snd_kcontrol *kcontrol, { struct snd_soc_component *compnt = snd_soc_kcontrol_component(kcontrol); struct tegra210_i2s *i2s = snd_soc_component_get_drvdata(compnt); - long *uctl_val = &ucontrol->value.integer.value[0]; if (strstr(kcontrol->id.name, "Loopback")) - *uctl_val = i2s->loopback; + ucontrol->value.integer.value[0] = i2s->loopback; else if (strstr(kcontrol->id.name, "FSYNC Width")) - *uctl_val = i2s->fsync_width; + ucontrol->value.integer.value[0] = i2s->fsync_width; else if (strstr(kcontrol->id.name, "Capture Stereo To Mono")) - *uctl_val = i2s->stereo_to_mono[I2S_TX_PATH]; + ucontrol->value.enumerated.item[0] = + i2s->stereo_to_mono[I2S_TX_PATH]; else if (strstr(kcontrol->id.name, "Capture Mono To Stereo")) - *uctl_val = i2s->mono_to_stereo[I2S_TX_PATH]; + ucontrol->value.enumerated.item[0] = + i2s->mono_to_stereo[I2S_TX_PATH]; else if (strstr(kcontrol->id.name, "Playback Stereo To Mono")) - *uctl_val = i2s->stereo_to_mono[I2S_RX_PATH]; + ucontrol->value.enumerated.item[0] = + i2s->stereo_to_mono[I2S_RX_PATH]; else if (strstr(kcontrol->id.name, "Playback Mono To Stereo")) - *uctl_val = i2s->mono_to_stereo[I2S_RX_PATH]; + ucontrol->value.enumerated.item[0] = + i2s->mono_to_stereo[I2S_RX_PATH]; else if (strstr(kcontrol->id.name, "Playback FIFO Threshold")) - *uctl_val = i2s->rx_fifo_th; + ucontrol->value.integer.value[0] = i2s->rx_fifo_th; else if (strstr(kcontrol->id.name, "BCLK Ratio")) - *uctl_val = i2s->bclk_ratio; + ucontrol->value.integer.value[0] = i2s->bclk_ratio; return 0; } @@ -344,10 +347,9 @@ static int tegra210_i2s_put_control(struct snd_kcontrol *kcontrol, { struct snd_soc_component *compnt = snd_soc_kcontrol_component(kcontrol); struct tegra210_i2s *i2s = snd_soc_component_get_drvdata(compnt); - int value = ucontrol->value.integer.value[0]; if (strstr(kcontrol->id.name, "Loopback")) { - i2s->loopback = value; + i2s->loopback = ucontrol->value.integer.value[0]; regmap_update_bits(i2s->regmap, TEGRA210_I2S_CTRL, I2S_CTRL_LPBK_MASK, @@ -362,24 +364,28 @@ static int tegra210_i2s_put_control(struct snd_kcontrol *kcontrol, * cases mixer control is used to update custom values. A value * of "N" here means, width is "N + 1" bit clock wide. */ - i2s->fsync_width = value; + i2s->fsync_width = ucontrol->value.integer.value[0]; regmap_update_bits(i2s->regmap, TEGRA210_I2S_CTRL, I2S_CTRL_FSYNC_WIDTH_MASK, i2s->fsync_width << I2S_FSYNC_WIDTH_SHIFT); } else if (strstr(kcontrol->id.name, "Capture Stereo To Mono")) { - i2s->stereo_to_mono[I2S_TX_PATH] = value; + i2s->stereo_to_mono[I2S_TX_PATH] = + ucontrol->value.enumerated.item[0]; } else if (strstr(kcontrol->id.name, "Capture Mono To Stereo")) { - i2s->mono_to_stereo[I2S_TX_PATH] = value; + i2s->mono_to_stereo[I2S_TX_PATH] = + ucontrol->value.enumerated.item[0]; } else if (strstr(kcontrol->id.name, "Playback Stereo To Mono")) { - i2s->stereo_to_mono[I2S_RX_PATH] = value; + i2s->stereo_to_mono[I2S_RX_PATH] = + ucontrol->value.enumerated.item[0]; } else if (strstr(kcontrol->id.name, "Playback Mono To Stereo")) { - i2s->mono_to_stereo[I2S_RX_PATH] = value; + i2s->mono_to_stereo[I2S_RX_PATH] = + ucontrol->value.enumerated.item[0]; } else if (strstr(kcontrol->id.name, "Playback FIFO Threshold")) { - i2s->rx_fifo_th = value; + i2s->rx_fifo_th = ucontrol->value.integer.value[0]; } else if (strstr(kcontrol->id.name, "BCLK Ratio")) { - i2s->bclk_ratio = value; + i2s->bclk_ratio = ucontrol->value.integer.value[0]; } return 0; -- cgit v1.2.3 From 559d234569a998a4004de1bd1f12da5487fb826e Mon Sep 17 00:00:00 2001 From: Sameer Pujar Date: Thu, 18 Nov 2021 12:36:58 +0530 Subject: ASoC: tegra: Fix wrong value type in DMIC The enum controls are expected to use enumerated value type. Update relevant references in control get/put callbacks. Fixes: 8c8ff982e9e2 ("ASoC: tegra: Add Tegra210 based DMIC driver") Suggested-by: Takashi Iwai Signed-off-by: Sameer Pujar Reviewed-by: Takashi Iwai Link: https://lore.kernel.org/r/1637219231-406-4-git-send-email-spujar@nvidia.com Signed-off-by: Mark Brown --- sound/soc/tegra/tegra210_dmic.c | 23 +++++++++++------------ 1 file changed, 11 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/soc/tegra/tegra210_dmic.c b/sound/soc/tegra/tegra210_dmic.c index b096478cd2ef..ee2aedb0440f 100644 --- a/sound/soc/tegra/tegra210_dmic.c +++ b/sound/soc/tegra/tegra210_dmic.c @@ -165,15 +165,15 @@ static int tegra210_dmic_get_control(struct snd_kcontrol *kcontrol, if (strstr(kcontrol->id.name, "Boost Gain Volume")) ucontrol->value.integer.value[0] = dmic->boost_gain; else if (strstr(kcontrol->id.name, "Channel Select")) - ucontrol->value.integer.value[0] = dmic->ch_select; + ucontrol->value.enumerated.item[0] = dmic->ch_select; else if (strstr(kcontrol->id.name, "Mono To Stereo")) - ucontrol->value.integer.value[0] = dmic->mono_to_stereo; + ucontrol->value.enumerated.item[0] = dmic->mono_to_stereo; else if (strstr(kcontrol->id.name, "Stereo To Mono")) - ucontrol->value.integer.value[0] = dmic->stereo_to_mono; + ucontrol->value.enumerated.item[0] = dmic->stereo_to_mono; else if (strstr(kcontrol->id.name, "OSR Value")) - ucontrol->value.integer.value[0] = dmic->osr_val; + ucontrol->value.enumerated.item[0] = dmic->osr_val; else if (strstr(kcontrol->id.name, "LR Polarity Select")) - ucontrol->value.integer.value[0] = dmic->lrsel; + ucontrol->value.enumerated.item[0] = dmic->lrsel; return 0; } @@ -183,20 +183,19 @@ static int tegra210_dmic_put_control(struct snd_kcontrol *kcontrol, { struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); struct tegra210_dmic *dmic = snd_soc_component_get_drvdata(comp); - int value = ucontrol->value.integer.value[0]; if (strstr(kcontrol->id.name, "Boost Gain Volume")) - dmic->boost_gain = value; + dmic->boost_gain = ucontrol->value.integer.value[0]; else if (strstr(kcontrol->id.name, "Channel Select")) - dmic->ch_select = ucontrol->value.integer.value[0]; + dmic->ch_select = ucontrol->value.enumerated.item[0]; else if (strstr(kcontrol->id.name, "Mono To Stereo")) - dmic->mono_to_stereo = value; + dmic->mono_to_stereo = ucontrol->value.enumerated.item[0]; else if (strstr(kcontrol->id.name, "Stereo To Mono")) - dmic->stereo_to_mono = value; + dmic->stereo_to_mono = ucontrol->value.enumerated.item[0]; else if (strstr(kcontrol->id.name, "OSR Value")) - dmic->osr_val = value; + dmic->osr_val = ucontrol->value.enumerated.item[0]; else if (strstr(kcontrol->id.name, "LR Polarity Select")) - dmic->lrsel = value; + dmic->lrsel = ucontrol->value.enumerated.item[0]; return 0; } -- cgit v1.2.3 From 3aa0d5c8bb3f5ef622ec2764823f551a1f630711 Mon Sep 17 00:00:00 2001 From: Sameer Pujar Date: Thu, 18 Nov 2021 12:36:59 +0530 Subject: ASoC: tegra: Fix wrong value type in DSPK The enum controls are expected to use enumerated value type. Update relevant references in control get/put callbacks. Fixes: 327ef6470266 ("ASoC: tegra: Add Tegra186 based DSPK driver") Suggested-by: Takashi Iwai Signed-off-by: Sameer Pujar Reviewed-by: Takashi Iwai Link: https://lore.kernel.org/r/1637219231-406-5-git-send-email-spujar@nvidia.com Signed-off-by: Mark Brown --- sound/soc/tegra/tegra186_dspk.c | 23 +++++++++++------------ 1 file changed, 11 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/soc/tegra/tegra186_dspk.c b/sound/soc/tegra/tegra186_dspk.c index 8ee9a77bd83d..67269e77d6e8 100644 --- a/sound/soc/tegra/tegra186_dspk.c +++ b/sound/soc/tegra/tegra186_dspk.c @@ -35,15 +35,15 @@ static int tegra186_dspk_get_control(struct snd_kcontrol *kcontrol, if (strstr(kcontrol->id.name, "FIFO Threshold")) ucontrol->value.integer.value[0] = dspk->rx_fifo_th; else if (strstr(kcontrol->id.name, "OSR Value")) - ucontrol->value.integer.value[0] = dspk->osr_val; + ucontrol->value.enumerated.item[0] = dspk->osr_val; else if (strstr(kcontrol->id.name, "LR Polarity Select")) - ucontrol->value.integer.value[0] = dspk->lrsel; + ucontrol->value.enumerated.item[0] = dspk->lrsel; else if (strstr(kcontrol->id.name, "Channel Select")) - ucontrol->value.integer.value[0] = dspk->ch_sel; + ucontrol->value.enumerated.item[0] = dspk->ch_sel; else if (strstr(kcontrol->id.name, "Mono To Stereo")) - ucontrol->value.integer.value[0] = dspk->mono_to_stereo; + ucontrol->value.enumerated.item[0] = dspk->mono_to_stereo; else if (strstr(kcontrol->id.name, "Stereo To Mono")) - ucontrol->value.integer.value[0] = dspk->stereo_to_mono; + ucontrol->value.enumerated.item[0] = dspk->stereo_to_mono; return 0; } @@ -53,20 +53,19 @@ static int tegra186_dspk_put_control(struct snd_kcontrol *kcontrol, { struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); struct tegra186_dspk *dspk = snd_soc_component_get_drvdata(codec); - int val = ucontrol->value.integer.value[0]; if (strstr(kcontrol->id.name, "FIFO Threshold")) - dspk->rx_fifo_th = val; + dspk->rx_fifo_th = ucontrol->value.integer.value[0]; else if (strstr(kcontrol->id.name, "OSR Value")) - dspk->osr_val = val; + dspk->osr_val = ucontrol->value.enumerated.item[0]; else if (strstr(kcontrol->id.name, "LR Polarity Select")) - dspk->lrsel = val; + dspk->lrsel = ucontrol->value.enumerated.item[0]; else if (strstr(kcontrol->id.name, "Channel Select")) - dspk->ch_sel = val; + dspk->ch_sel = ucontrol->value.enumerated.item[0]; else if (strstr(kcontrol->id.name, "Mono To Stereo")) - dspk->mono_to_stereo = val; + dspk->mono_to_stereo = ucontrol->value.enumerated.item[0]; else if (strstr(kcontrol->id.name, "Stereo To Mono")) - dspk->stereo_to_mono = val; + dspk->stereo_to_mono = ucontrol->value.enumerated.item[0]; return 0; } -- cgit v1.2.3 From 42afca1a65661935cdd54d2e0c5d0cc2426db7af Mon Sep 17 00:00:00 2001 From: Sameer Pujar Date: Thu, 18 Nov 2021 12:37:00 +0530 Subject: ASoC: tegra: Fix wrong value type in SFC The enum controls are expected to use enumerated value type. Update relevant references in control get/put callbacks. Fixes: b2f74ec53a6c ("ASoC: tegra: Add Tegra210 based SFC driver") Suggested-by: Takashi Iwai Signed-off-by: Sameer Pujar Reviewed-by: Takashi Iwai Link: https://lore.kernel.org/r/1637219231-406-6-git-send-email-spujar@nvidia.com Signed-off-by: Mark Brown --- sound/soc/tegra/tegra210_sfc.c | 21 ++++++++++++--------- 1 file changed, 12 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/soc/tegra/tegra210_sfc.c b/sound/soc/tegra/tegra210_sfc.c index dc477ee1b82c..cb592ef55bd3 100644 --- a/sound/soc/tegra/tegra210_sfc.c +++ b/sound/soc/tegra/tegra210_sfc.c @@ -3251,16 +3251,16 @@ static int tegra210_sfc_get_control(struct snd_kcontrol *kcontrol, struct tegra210_sfc *sfc = snd_soc_component_get_drvdata(cmpnt); if (strstr(kcontrol->id.name, "Input Stereo To Mono")) - ucontrol->value.integer.value[0] = + ucontrol->value.enumerated.item[0] = sfc->stereo_to_mono[SFC_RX_PATH]; else if (strstr(kcontrol->id.name, "Input Mono To Stereo")) - ucontrol->value.integer.value[0] = + ucontrol->value.enumerated.item[0] = sfc->mono_to_stereo[SFC_RX_PATH]; else if (strstr(kcontrol->id.name, "Output Stereo To Mono")) - ucontrol->value.integer.value[0] = + ucontrol->value.enumerated.item[0] = sfc->stereo_to_mono[SFC_TX_PATH]; else if (strstr(kcontrol->id.name, "Output Mono To Stereo")) - ucontrol->value.integer.value[0] = + ucontrol->value.enumerated.item[0] = sfc->mono_to_stereo[SFC_TX_PATH]; return 0; @@ -3271,16 +3271,19 @@ static int tegra210_sfc_put_control(struct snd_kcontrol *kcontrol, { struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); struct tegra210_sfc *sfc = snd_soc_component_get_drvdata(cmpnt); - int value = ucontrol->value.integer.value[0]; if (strstr(kcontrol->id.name, "Input Stereo To Mono")) - sfc->stereo_to_mono[SFC_RX_PATH] = value; + sfc->stereo_to_mono[SFC_RX_PATH] = + ucontrol->value.enumerated.item[0]; else if (strstr(kcontrol->id.name, "Input Mono To Stereo")) - sfc->mono_to_stereo[SFC_RX_PATH] = value; + sfc->mono_to_stereo[SFC_RX_PATH] = + ucontrol->value.enumerated.item[0]; else if (strstr(kcontrol->id.name, "Output Stereo To Mono")) - sfc->stereo_to_mono[SFC_TX_PATH] = value; + sfc->stereo_to_mono[SFC_TX_PATH] = + ucontrol->value.enumerated.item[0]; else if (strstr(kcontrol->id.name, "Output Mono To Stereo")) - sfc->mono_to_stereo[SFC_TX_PATH] = value; + sfc->mono_to_stereo[SFC_TX_PATH] = + ucontrol->value.enumerated.item[0]; else return 0; -- cgit v1.2.3 From 6762965d0214df474e3a58e1d4d3ab004c5da0ea Mon Sep 17 00:00:00 2001 From: Sameer Pujar Date: Thu, 18 Nov 2021 12:37:01 +0530 Subject: ASoC: tegra: Fix wrong value type in MVC The enum controls are expected to use enumerated value type. Update relevant references in control get/put callbacks. Fixes: e539891f9687 ("ASoC: tegra: Add Tegra210 based MVC driver") Suggested-by: Takashi Iwai Signed-off-by: Sameer Pujar Reviewed-by: Takashi Iwai Link: https://lore.kernel.org/r/1637219231-406-7-git-send-email-spujar@nvidia.com Signed-off-by: Mark Brown --- sound/soc/tegra/tegra210_mvc.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/tegra/tegra210_mvc.c b/sound/soc/tegra/tegra210_mvc.c index 7b9c7006e419..b7e317065251 100644 --- a/sound/soc/tegra/tegra210_mvc.c +++ b/sound/soc/tegra/tegra210_mvc.c @@ -275,7 +275,7 @@ static int tegra210_mvc_get_curve_type(struct snd_kcontrol *kcontrol, struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); struct tegra210_mvc *mvc = snd_soc_component_get_drvdata(cmpnt); - ucontrol->value.integer.value[0] = mvc->curve_type; + ucontrol->value.enumerated.item[0] = mvc->curve_type; return 0; } @@ -285,7 +285,7 @@ static int tegra210_mvc_put_curve_type(struct snd_kcontrol *kcontrol, { struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); struct tegra210_mvc *mvc = snd_soc_component_get_drvdata(cmpnt); - int value; + unsigned int value; regmap_read(mvc->regmap, TEGRA210_MVC_ENABLE, &value); if (value & TEGRA210_MVC_EN) { @@ -294,10 +294,10 @@ static int tegra210_mvc_put_curve_type(struct snd_kcontrol *kcontrol, return -EINVAL; } - if (mvc->curve_type == ucontrol->value.integer.value[0]) + if (mvc->curve_type == ucontrol->value.enumerated.item[0]) return 0; - mvc->curve_type = ucontrol->value.integer.value[0]; + mvc->curve_type = ucontrol->value.enumerated.item[0]; tegra210_mvc_reset_vol_settings(mvc, cmpnt->dev); -- cgit v1.2.3 From e2b87a18a60c02d0dcd1de801d669587e516cc4d Mon Sep 17 00:00:00 2001 From: Sameer Pujar Date: Thu, 18 Nov 2021 12:37:02 +0530 Subject: ASoC: tegra: Fix kcontrol put callback in ADMAIF The kcontrol put callback is expected to return 1 when there is change in HW or when the update is acknowledged by driver. This would ensure that change notifications are sent to subscribed applications. Update the ADMAIF driver accordingly. Fixes: f74028e159bb ("ASoC: tegra: Add Tegra210 based ADMAIF driver") Suggested-by: Jaroslav Kysela Suggested-by: Mark Brown Signed-off-by: Sameer Pujar Reviewed-by: Takashi Iwai Link: https://lore.kernel.org/r/1637219231-406-8-git-send-email-spujar@nvidia.com Signed-off-by: Mark Brown --- sound/soc/tegra/tegra210_admaif.c | 138 ++++++++++++++++++++++++++++++-------- 1 file changed, 109 insertions(+), 29 deletions(-) (limited to 'sound') diff --git a/sound/soc/tegra/tegra210_admaif.c b/sound/soc/tegra/tegra210_admaif.c index 6febe80cfa6f..1a2e868a6220 100644 --- a/sound/soc/tegra/tegra210_admaif.c +++ b/sound/soc/tegra/tegra210_admaif.c @@ -424,46 +424,122 @@ static const struct snd_soc_dai_ops tegra_admaif_dai_ops = { .trigger = tegra_admaif_trigger, }; -static int tegra_admaif_get_control(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) +static int tegra210_admaif_pget_mono_to_stereo(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); + struct tegra_admaif *admaif = snd_soc_component_get_drvdata(cmpnt); struct soc_enum *ec = (struct soc_enum *)kcontrol->private_value; + + ucontrol->value.enumerated.item[0] = + admaif->mono_to_stereo[ADMAIF_TX_PATH][ec->reg]; + + return 0; +} + +static int tegra210_admaif_pput_mono_to_stereo(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); struct tegra_admaif *admaif = snd_soc_component_get_drvdata(cmpnt); - unsigned int *uctl_val = &ucontrol->value.enumerated.item[0]; + struct soc_enum *ec = (struct soc_enum *)kcontrol->private_value; + unsigned int value = ucontrol->value.enumerated.item[0]; - if (strstr(kcontrol->id.name, "Playback Mono To Stereo")) - *uctl_val = admaif->mono_to_stereo[ADMAIF_TX_PATH][ec->reg]; - else if (strstr(kcontrol->id.name, "Capture Mono To Stereo")) - *uctl_val = admaif->mono_to_stereo[ADMAIF_RX_PATH][ec->reg]; - else if (strstr(kcontrol->id.name, "Playback Stereo To Mono")) - *uctl_val = admaif->stereo_to_mono[ADMAIF_TX_PATH][ec->reg]; - else if (strstr(kcontrol->id.name, "Capture Stereo To Mono")) - *uctl_val = admaif->stereo_to_mono[ADMAIF_RX_PATH][ec->reg]; + if (value == admaif->mono_to_stereo[ADMAIF_TX_PATH][ec->reg]) + return 0; + + admaif->mono_to_stereo[ADMAIF_TX_PATH][ec->reg] = value; + + return 1; +} + +static int tegra210_admaif_cget_mono_to_stereo(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); + struct tegra_admaif *admaif = snd_soc_component_get_drvdata(cmpnt); + struct soc_enum *ec = (struct soc_enum *)kcontrol->private_value; + + ucontrol->value.enumerated.item[0] = + admaif->mono_to_stereo[ADMAIF_RX_PATH][ec->reg]; return 0; } -static int tegra_admaif_put_control(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) +static int tegra210_admaif_cput_mono_to_stereo(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); + struct tegra_admaif *admaif = snd_soc_component_get_drvdata(cmpnt); struct soc_enum *ec = (struct soc_enum *)kcontrol->private_value; + unsigned int value = ucontrol->value.enumerated.item[0]; + + if (value == admaif->mono_to_stereo[ADMAIF_RX_PATH][ec->reg]) + return 0; + + admaif->mono_to_stereo[ADMAIF_RX_PATH][ec->reg] = value; + + return 1; +} + +static int tegra210_admaif_pget_stereo_to_mono(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); struct tegra_admaif *admaif = snd_soc_component_get_drvdata(cmpnt); + struct soc_enum *ec = (struct soc_enum *)kcontrol->private_value; + + ucontrol->value.enumerated.item[0] = + admaif->stereo_to_mono[ADMAIF_TX_PATH][ec->reg]; + + return 0; +} + +static int tegra210_admaif_pput_stereo_to_mono(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); + struct tegra_admaif *admaif = snd_soc_component_get_drvdata(cmpnt); + struct soc_enum *ec = (struct soc_enum *)kcontrol->private_value; unsigned int value = ucontrol->value.enumerated.item[0]; - if (strstr(kcontrol->id.name, "Playback Mono To Stereo")) - admaif->mono_to_stereo[ADMAIF_TX_PATH][ec->reg] = value; - else if (strstr(kcontrol->id.name, "Capture Mono To Stereo")) - admaif->mono_to_stereo[ADMAIF_RX_PATH][ec->reg] = value; - else if (strstr(kcontrol->id.name, "Playback Stereo To Mono")) - admaif->stereo_to_mono[ADMAIF_TX_PATH][ec->reg] = value; - else if (strstr(kcontrol->id.name, "Capture Stereo To Mono")) - admaif->stereo_to_mono[ADMAIF_RX_PATH][ec->reg] = value; + if (value == admaif->stereo_to_mono[ADMAIF_TX_PATH][ec->reg]) + return 0; + + admaif->stereo_to_mono[ADMAIF_TX_PATH][ec->reg] = value; + + return 1; +} + +static int tegra210_admaif_cget_stereo_to_mono(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); + struct tegra_admaif *admaif = snd_soc_component_get_drvdata(cmpnt); + struct soc_enum *ec = (struct soc_enum *)kcontrol->private_value; + + ucontrol->value.enumerated.item[0] = + admaif->stereo_to_mono[ADMAIF_RX_PATH][ec->reg]; return 0; } +static int tegra210_admaif_cput_stereo_to_mono(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); + struct tegra_admaif *admaif = snd_soc_component_get_drvdata(cmpnt); + struct soc_enum *ec = (struct soc_enum *)kcontrol->private_value; + unsigned int value = ucontrol->value.enumerated.item[0]; + + if (value == admaif->stereo_to_mono[ADMAIF_RX_PATH][ec->reg]) + return 0; + + admaif->stereo_to_mono[ADMAIF_RX_PATH][ec->reg] = value; + + return 1; +} + static int tegra_admaif_dai_probe(struct snd_soc_dai *dai) { struct tegra_admaif *admaif = snd_soc_dai_get_drvdata(dai); @@ -559,17 +635,21 @@ static const char * const tegra_admaif_mono_conv_text[] = { } #define TEGRA_ADMAIF_CIF_CTRL(reg) \ - NV_SOC_ENUM_EXT("ADMAIF" #reg " Playback Mono To Stereo", reg - 1,\ - tegra_admaif_get_control, tegra_admaif_put_control, \ + NV_SOC_ENUM_EXT("ADMAIF" #reg " Playback Mono To Stereo", reg - 1, \ + tegra210_admaif_pget_mono_to_stereo, \ + tegra210_admaif_pput_mono_to_stereo, \ tegra_admaif_mono_conv_text), \ - NV_SOC_ENUM_EXT("ADMAIF" #reg " Playback Stereo To Mono", reg - 1,\ - tegra_admaif_get_control, tegra_admaif_put_control, \ + NV_SOC_ENUM_EXT("ADMAIF" #reg " Playback Stereo To Mono", reg - 1, \ + tegra210_admaif_pget_stereo_to_mono, \ + tegra210_admaif_pput_stereo_to_mono, \ tegra_admaif_stereo_conv_text), \ - NV_SOC_ENUM_EXT("ADMAIF" #reg " Capture Mono To Stereo", reg - 1, \ - tegra_admaif_get_control, tegra_admaif_put_control, \ + NV_SOC_ENUM_EXT("ADMAIF" #reg " Capture Mono To Stereo", reg - 1, \ + tegra210_admaif_cget_mono_to_stereo, \ + tegra210_admaif_cput_mono_to_stereo, \ tegra_admaif_mono_conv_text), \ - NV_SOC_ENUM_EXT("ADMAIF" #reg " Capture Stereo To Mono", reg - 1, \ - tegra_admaif_get_control, tegra_admaif_put_control, \ + NV_SOC_ENUM_EXT("ADMAIF" #reg " Capture Stereo To Mono", reg - 1, \ + tegra210_admaif_cget_stereo_to_mono, \ + tegra210_admaif_cput_stereo_to_mono, \ tegra_admaif_stereo_conv_text) static struct snd_kcontrol_new tegra210_admaif_controls[] = { -- cgit v1.2.3 From f21a9df3f7cb0005947679d7b9237c90574e229a Mon Sep 17 00:00:00 2001 From: Sameer Pujar Date: Thu, 18 Nov 2021 12:37:03 +0530 Subject: ASoC: tegra: Fix kcontrol put callback in I2S The kcontrol put callback is expected to return 1 when there is change in HW or when the update is acknowledged by driver. This would ensure that change notifications are sent to subscribed applications. Update the I2S driver accordingly. Fixes: c0bfa98349d1 ("ASoC: tegra: Add Tegra210 based I2S driver") Suggested-by: Jaroslav Kysela Suggested-by: Mark Brown Signed-off-by: Sameer Pujar Reviewed-by: Takashi Iwai Link: https://lore.kernel.org/r/1637219231-406-9-git-send-email-spujar@nvidia.com Signed-off-by: Mark Brown --- sound/soc/tegra/tegra210_i2s.c | 302 ++++++++++++++++++++++++++++++----------- 1 file changed, 226 insertions(+), 76 deletions(-) (limited to 'sound') diff --git a/sound/soc/tegra/tegra210_i2s.c b/sound/soc/tegra/tegra210_i2s.c index 5c304612769f..9552bbb939dd 100644 --- a/sound/soc/tegra/tegra210_i2s.c +++ b/sound/soc/tegra/tegra210_i2s.c @@ -302,91 +302,235 @@ static int tegra210_i2s_set_tdm_slot(struct snd_soc_dai *dai, return 0; } -static int tegra210_i2s_set_dai_bclk_ratio(struct snd_soc_dai *dai, - unsigned int ratio) +static int tegra210_i2s_get_loopback(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { - struct tegra210_i2s *i2s = snd_soc_dai_get_drvdata(dai); + struct snd_soc_component *compnt = snd_soc_kcontrol_component(kcontrol); + struct tegra210_i2s *i2s = snd_soc_component_get_drvdata(compnt); - i2s->bclk_ratio = ratio; + ucontrol->value.integer.value[0] = i2s->loopback; return 0; } -static int tegra210_i2s_get_control(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) +static int tegra210_i2s_put_loopback(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { struct snd_soc_component *compnt = snd_soc_kcontrol_component(kcontrol); struct tegra210_i2s *i2s = snd_soc_component_get_drvdata(compnt); + int value = ucontrol->value.integer.value[0]; + + if (value == i2s->loopback) + return 0; + + i2s->loopback = value; - if (strstr(kcontrol->id.name, "Loopback")) - ucontrol->value.integer.value[0] = i2s->loopback; - else if (strstr(kcontrol->id.name, "FSYNC Width")) - ucontrol->value.integer.value[0] = i2s->fsync_width; - else if (strstr(kcontrol->id.name, "Capture Stereo To Mono")) - ucontrol->value.enumerated.item[0] = - i2s->stereo_to_mono[I2S_TX_PATH]; - else if (strstr(kcontrol->id.name, "Capture Mono To Stereo")) - ucontrol->value.enumerated.item[0] = - i2s->mono_to_stereo[I2S_TX_PATH]; - else if (strstr(kcontrol->id.name, "Playback Stereo To Mono")) - ucontrol->value.enumerated.item[0] = - i2s->stereo_to_mono[I2S_RX_PATH]; - else if (strstr(kcontrol->id.name, "Playback Mono To Stereo")) - ucontrol->value.enumerated.item[0] = - i2s->mono_to_stereo[I2S_RX_PATH]; - else if (strstr(kcontrol->id.name, "Playback FIFO Threshold")) - ucontrol->value.integer.value[0] = i2s->rx_fifo_th; - else if (strstr(kcontrol->id.name, "BCLK Ratio")) - ucontrol->value.integer.value[0] = i2s->bclk_ratio; + regmap_update_bits(i2s->regmap, TEGRA210_I2S_CTRL, I2S_CTRL_LPBK_MASK, + i2s->loopback << I2S_CTRL_LPBK_SHIFT); + + return 1; +} + +static int tegra210_i2s_get_fsync_width(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *compnt = snd_soc_kcontrol_component(kcontrol); + struct tegra210_i2s *i2s = snd_soc_component_get_drvdata(compnt); + + ucontrol->value.integer.value[0] = i2s->fsync_width; return 0; } -static int tegra210_i2s_put_control(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) +static int tegra210_i2s_put_fsync_width(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *compnt = snd_soc_kcontrol_component(kcontrol); + struct tegra210_i2s *i2s = snd_soc_component_get_drvdata(compnt); + int value = ucontrol->value.integer.value[0]; + + if (value == i2s->fsync_width) + return 0; + + i2s->fsync_width = value; + + /* + * Frame sync width is used only for FSYNC modes and not + * applicable for LRCK modes. Reset value for this field is "0", + * which means the width is one bit clock wide. + * The width requirement may depend on the codec and in such + * cases mixer control is used to update custom values. A value + * of "N" here means, width is "N + 1" bit clock wide. + */ + regmap_update_bits(i2s->regmap, TEGRA210_I2S_CTRL, + I2S_CTRL_FSYNC_WIDTH_MASK, + i2s->fsync_width << I2S_FSYNC_WIDTH_SHIFT); + + return 1; +} + +static int tegra210_i2s_cget_stereo_to_mono(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { struct snd_soc_component *compnt = snd_soc_kcontrol_component(kcontrol); struct tegra210_i2s *i2s = snd_soc_component_get_drvdata(compnt); - if (strstr(kcontrol->id.name, "Loopback")) { - i2s->loopback = ucontrol->value.integer.value[0]; + ucontrol->value.enumerated.item[0] = i2s->stereo_to_mono[I2S_TX_PATH]; - regmap_update_bits(i2s->regmap, TEGRA210_I2S_CTRL, - I2S_CTRL_LPBK_MASK, - i2s->loopback << I2S_CTRL_LPBK_SHIFT); + return 0; +} - } else if (strstr(kcontrol->id.name, "FSYNC Width")) { - /* - * Frame sync width is used only for FSYNC modes and not - * applicable for LRCK modes. Reset value for this field is "0", - * which means the width is one bit clock wide. - * The width requirement may depend on the codec and in such - * cases mixer control is used to update custom values. A value - * of "N" here means, width is "N + 1" bit clock wide. - */ - i2s->fsync_width = ucontrol->value.integer.value[0]; - - regmap_update_bits(i2s->regmap, TEGRA210_I2S_CTRL, - I2S_CTRL_FSYNC_WIDTH_MASK, - i2s->fsync_width << I2S_FSYNC_WIDTH_SHIFT); - - } else if (strstr(kcontrol->id.name, "Capture Stereo To Mono")) { - i2s->stereo_to_mono[I2S_TX_PATH] = - ucontrol->value.enumerated.item[0]; - } else if (strstr(kcontrol->id.name, "Capture Mono To Stereo")) { - i2s->mono_to_stereo[I2S_TX_PATH] = - ucontrol->value.enumerated.item[0]; - } else if (strstr(kcontrol->id.name, "Playback Stereo To Mono")) { - i2s->stereo_to_mono[I2S_RX_PATH] = - ucontrol->value.enumerated.item[0]; - } else if (strstr(kcontrol->id.name, "Playback Mono To Stereo")) { - i2s->mono_to_stereo[I2S_RX_PATH] = - ucontrol->value.enumerated.item[0]; - } else if (strstr(kcontrol->id.name, "Playback FIFO Threshold")) { - i2s->rx_fifo_th = ucontrol->value.integer.value[0]; - } else if (strstr(kcontrol->id.name, "BCLK Ratio")) { - i2s->bclk_ratio = ucontrol->value.integer.value[0]; - } +static int tegra210_i2s_cput_stereo_to_mono(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *compnt = snd_soc_kcontrol_component(kcontrol); + struct tegra210_i2s *i2s = snd_soc_component_get_drvdata(compnt); + unsigned int value = ucontrol->value.enumerated.item[0]; + + if (value == i2s->stereo_to_mono[I2S_TX_PATH]) + return 0; + + i2s->stereo_to_mono[I2S_TX_PATH] = value; + + return 1; +} + +static int tegra210_i2s_cget_mono_to_stereo(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *compnt = snd_soc_kcontrol_component(kcontrol); + struct tegra210_i2s *i2s = snd_soc_component_get_drvdata(compnt); + + ucontrol->value.enumerated.item[0] = i2s->mono_to_stereo[I2S_TX_PATH]; + + return 0; +} + +static int tegra210_i2s_cput_mono_to_stereo(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *compnt = snd_soc_kcontrol_component(kcontrol); + struct tegra210_i2s *i2s = snd_soc_component_get_drvdata(compnt); + unsigned int value = ucontrol->value.enumerated.item[0]; + + if (value == i2s->mono_to_stereo[I2S_TX_PATH]) + return 0; + + i2s->mono_to_stereo[I2S_TX_PATH] = value; + + return 1; +} + +static int tegra210_i2s_pget_stereo_to_mono(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *compnt = snd_soc_kcontrol_component(kcontrol); + struct tegra210_i2s *i2s = snd_soc_component_get_drvdata(compnt); + + ucontrol->value.enumerated.item[0] = i2s->stereo_to_mono[I2S_RX_PATH]; + + return 0; +} + +static int tegra210_i2s_pput_stereo_to_mono(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *compnt = snd_soc_kcontrol_component(kcontrol); + struct tegra210_i2s *i2s = snd_soc_component_get_drvdata(compnt); + unsigned int value = ucontrol->value.enumerated.item[0]; + + if (value == i2s->stereo_to_mono[I2S_RX_PATH]) + return 0; + + i2s->stereo_to_mono[I2S_RX_PATH] = value; + + return 1; +} + +static int tegra210_i2s_pget_mono_to_stereo(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *compnt = snd_soc_kcontrol_component(kcontrol); + struct tegra210_i2s *i2s = snd_soc_component_get_drvdata(compnt); + + ucontrol->value.enumerated.item[0] = i2s->mono_to_stereo[I2S_RX_PATH]; + + return 0; +} + +static int tegra210_i2s_pput_mono_to_stereo(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *compnt = snd_soc_kcontrol_component(kcontrol); + struct tegra210_i2s *i2s = snd_soc_component_get_drvdata(compnt); + unsigned int value = ucontrol->value.enumerated.item[0]; + + if (value == i2s->mono_to_stereo[I2S_RX_PATH]) + return 0; + + i2s->mono_to_stereo[I2S_RX_PATH] = value; + + return 1; +} + +static int tegra210_i2s_pget_fifo_th(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *compnt = snd_soc_kcontrol_component(kcontrol); + struct tegra210_i2s *i2s = snd_soc_component_get_drvdata(compnt); + + ucontrol->value.integer.value[0] = i2s->rx_fifo_th; + + return 0; +} + +static int tegra210_i2s_pput_fifo_th(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *compnt = snd_soc_kcontrol_component(kcontrol); + struct tegra210_i2s *i2s = snd_soc_component_get_drvdata(compnt); + int value = ucontrol->value.integer.value[0]; + + if (value == i2s->rx_fifo_th) + return 0; + + i2s->rx_fifo_th = value; + + return 1; +} + +static int tegra210_i2s_get_bclk_ratio(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *compnt = snd_soc_kcontrol_component(kcontrol); + struct tegra210_i2s *i2s = snd_soc_component_get_drvdata(compnt); + + ucontrol->value.integer.value[0] = i2s->bclk_ratio; + + return 0; +} + +static int tegra210_i2s_put_bclk_ratio(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *compnt = snd_soc_kcontrol_component(kcontrol); + struct tegra210_i2s *i2s = snd_soc_component_get_drvdata(compnt); + int value = ucontrol->value.integer.value[0]; + + if (value == i2s->bclk_ratio) + return 0; + + i2s->bclk_ratio = value; + + return 1; +} + +static int tegra210_i2s_set_dai_bclk_ratio(struct snd_soc_dai *dai, + unsigned int ratio) +{ + struct tegra210_i2s *i2s = snd_soc_dai_get_drvdata(dai); + + i2s->bclk_ratio = ratio; return 0; } @@ -604,22 +748,28 @@ static const struct soc_enum tegra210_i2s_stereo_conv_enum = tegra210_i2s_stereo_conv_text); static const struct snd_kcontrol_new tegra210_i2s_controls[] = { - SOC_SINGLE_EXT("Loopback", 0, 0, 1, 0, tegra210_i2s_get_control, - tegra210_i2s_put_control), - SOC_SINGLE_EXT("FSYNC Width", 0, 0, 255, 0, tegra210_i2s_get_control, - tegra210_i2s_put_control), + SOC_SINGLE_EXT("Loopback", 0, 0, 1, 0, tegra210_i2s_get_loopback, + tegra210_i2s_put_loopback), + SOC_SINGLE_EXT("FSYNC Width", 0, 0, 255, 0, + tegra210_i2s_get_fsync_width, + tegra210_i2s_put_fsync_width), SOC_ENUM_EXT("Capture Stereo To Mono", tegra210_i2s_stereo_conv_enum, - tegra210_i2s_get_control, tegra210_i2s_put_control), + tegra210_i2s_cget_stereo_to_mono, + tegra210_i2s_cput_stereo_to_mono), SOC_ENUM_EXT("Capture Mono To Stereo", tegra210_i2s_mono_conv_enum, - tegra210_i2s_get_control, tegra210_i2s_put_control), + tegra210_i2s_cget_mono_to_stereo, + tegra210_i2s_cput_mono_to_stereo), SOC_ENUM_EXT("Playback Stereo To Mono", tegra210_i2s_stereo_conv_enum, - tegra210_i2s_get_control, tegra210_i2s_put_control), + tegra210_i2s_pget_mono_to_stereo, + tegra210_i2s_pput_mono_to_stereo), SOC_ENUM_EXT("Playback Mono To Stereo", tegra210_i2s_mono_conv_enum, - tegra210_i2s_get_control, tegra210_i2s_put_control), + tegra210_i2s_pget_stereo_to_mono, + tegra210_i2s_pput_stereo_to_mono), SOC_SINGLE_EXT("Playback FIFO Threshold", 0, 0, I2S_RX_FIFO_DEPTH - 1, - 0, tegra210_i2s_get_control, tegra210_i2s_put_control), - SOC_SINGLE_EXT("BCLK Ratio", 0, 0, INT_MAX, 0, tegra210_i2s_get_control, - tegra210_i2s_put_control), + 0, tegra210_i2s_pget_fifo_th, tegra210_i2s_pput_fifo_th), + SOC_SINGLE_EXT("BCLK Ratio", 0, 0, INT_MAX, 0, + tegra210_i2s_get_bclk_ratio, + tegra210_i2s_put_bclk_ratio), }; static const struct snd_soc_dapm_widget tegra210_i2s_widgets[] = { -- cgit v1.2.3 From a347dfa10262fa0a10e2b1970ea0194e3d4a3251 Mon Sep 17 00:00:00 2001 From: Sameer Pujar Date: Thu, 18 Nov 2021 12:37:04 +0530 Subject: ASoC: tegra: Fix kcontrol put callback in DMIC The kcontrol put callback is expected to return 1 when there is change in HW or when the update is acknowledged by driver. This would ensure that change notifications are sent to subscribed applications. Update the DMIC driver accordingly. Fixes: 8c8ff982e9e2 ("ASoC: tegra: Add Tegra210 based DMIC driver") Suggested-by: Jaroslav Kysela Suggested-by: Mark Brown Signed-off-by: Sameer Pujar Reviewed-by: Takashi Iwai Link: https://lore.kernel.org/r/1637219231-406-10-git-send-email-spujar@nvidia.com Signed-off-by: Mark Brown --- sound/soc/tegra/tegra210_dmic.c | 183 ++++++++++++++++++++++++++++++++-------- 1 file changed, 149 insertions(+), 34 deletions(-) (limited to 'sound') diff --git a/sound/soc/tegra/tegra210_dmic.c b/sound/soc/tegra/tegra210_dmic.c index ee2aedb0440f..db95794530f4 100644 --- a/sound/soc/tegra/tegra210_dmic.c +++ b/sound/soc/tegra/tegra210_dmic.c @@ -156,50 +156,162 @@ static int tegra210_dmic_hw_params(struct snd_pcm_substream *substream, return 0; } -static int tegra210_dmic_get_control(struct snd_kcontrol *kcontrol, +static int tegra210_dmic_get_boost_gain(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tegra210_dmic *dmic = snd_soc_component_get_drvdata(comp); + + ucontrol->value.integer.value[0] = dmic->boost_gain; + + return 0; +} + +static int tegra210_dmic_put_boost_gain(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tegra210_dmic *dmic = snd_soc_component_get_drvdata(comp); + int value = ucontrol->value.integer.value[0]; + + if (value == dmic->boost_gain) + return 0; + + dmic->boost_gain = value; + + return 1; +} + +static int tegra210_dmic_get_ch_select(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tegra210_dmic *dmic = snd_soc_component_get_drvdata(comp); + + ucontrol->value.enumerated.item[0] = dmic->ch_select; + + return 0; +} + +static int tegra210_dmic_put_ch_select(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tegra210_dmic *dmic = snd_soc_component_get_drvdata(comp); + unsigned int value = ucontrol->value.enumerated.item[0]; + + if (value == dmic->ch_select) + return 0; + + dmic->ch_select = value; + + return 1; +} + +static int tegra210_dmic_get_mono_to_stereo(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tegra210_dmic *dmic = snd_soc_component_get_drvdata(comp); + + ucontrol->value.enumerated.item[0] = dmic->mono_to_stereo; + + return 0; +} + +static int tegra210_dmic_put_mono_to_stereo(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tegra210_dmic *dmic = snd_soc_component_get_drvdata(comp); + unsigned int value = ucontrol->value.enumerated.item[0]; + + if (value == dmic->mono_to_stereo) + return 0; + + dmic->mono_to_stereo = value; + + return 1; +} + +static int tegra210_dmic_get_stereo_to_mono(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tegra210_dmic *dmic = snd_soc_component_get_drvdata(comp); + + ucontrol->value.enumerated.item[0] = dmic->stereo_to_mono; + + return 0; +} + +static int tegra210_dmic_put_stereo_to_mono(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tegra210_dmic *dmic = snd_soc_component_get_drvdata(comp); + unsigned int value = ucontrol->value.enumerated.item[0]; + + if (value == dmic->stereo_to_mono) + return 0; + + dmic->stereo_to_mono = value; + + return 1; +} + +static int tegra210_dmic_get_osr_val(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); struct tegra210_dmic *dmic = snd_soc_component_get_drvdata(comp); - if (strstr(kcontrol->id.name, "Boost Gain Volume")) - ucontrol->value.integer.value[0] = dmic->boost_gain; - else if (strstr(kcontrol->id.name, "Channel Select")) - ucontrol->value.enumerated.item[0] = dmic->ch_select; - else if (strstr(kcontrol->id.name, "Mono To Stereo")) - ucontrol->value.enumerated.item[0] = dmic->mono_to_stereo; - else if (strstr(kcontrol->id.name, "Stereo To Mono")) - ucontrol->value.enumerated.item[0] = dmic->stereo_to_mono; - else if (strstr(kcontrol->id.name, "OSR Value")) - ucontrol->value.enumerated.item[0] = dmic->osr_val; - else if (strstr(kcontrol->id.name, "LR Polarity Select")) - ucontrol->value.enumerated.item[0] = dmic->lrsel; + ucontrol->value.enumerated.item[0] = dmic->osr_val; return 0; } -static int tegra210_dmic_put_control(struct snd_kcontrol *kcontrol, +static int tegra210_dmic_put_osr_val(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tegra210_dmic *dmic = snd_soc_component_get_drvdata(comp); + unsigned int value = ucontrol->value.enumerated.item[0]; + + if (value == dmic->osr_val) + return 0; + + dmic->osr_val = value; + + return 1; +} + +static int tegra210_dmic_get_pol_sel(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); struct tegra210_dmic *dmic = snd_soc_component_get_drvdata(comp); - if (strstr(kcontrol->id.name, "Boost Gain Volume")) - dmic->boost_gain = ucontrol->value.integer.value[0]; - else if (strstr(kcontrol->id.name, "Channel Select")) - dmic->ch_select = ucontrol->value.enumerated.item[0]; - else if (strstr(kcontrol->id.name, "Mono To Stereo")) - dmic->mono_to_stereo = ucontrol->value.enumerated.item[0]; - else if (strstr(kcontrol->id.name, "Stereo To Mono")) - dmic->stereo_to_mono = ucontrol->value.enumerated.item[0]; - else if (strstr(kcontrol->id.name, "OSR Value")) - dmic->osr_val = ucontrol->value.enumerated.item[0]; - else if (strstr(kcontrol->id.name, "LR Polarity Select")) - dmic->lrsel = ucontrol->value.enumerated.item[0]; + ucontrol->value.enumerated.item[0] = dmic->lrsel; return 0; } +static int tegra210_dmic_put_pol_sel(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tegra210_dmic *dmic = snd_soc_component_get_drvdata(comp); + unsigned int value = ucontrol->value.enumerated.item[0]; + + if (value == dmic->lrsel) + return 0; + + dmic->lrsel = value; + + return 1; +} + static const struct snd_soc_dai_ops tegra210_dmic_dai_ops = { .hw_params = tegra210_dmic_hw_params, }; @@ -286,19 +398,22 @@ static const struct soc_enum tegra210_dmic_lrsel_enum = static const struct snd_kcontrol_new tegra210_dmic_controls[] = { SOC_SINGLE_EXT("Boost Gain Volume", 0, 0, MAX_BOOST_GAIN, 0, - tegra210_dmic_get_control, tegra210_dmic_put_control), + tegra210_dmic_get_boost_gain, + tegra210_dmic_put_boost_gain), SOC_ENUM_EXT("Channel Select", tegra210_dmic_ch_enum, - tegra210_dmic_get_control, tegra210_dmic_put_control), + tegra210_dmic_get_ch_select, tegra210_dmic_put_ch_select), SOC_ENUM_EXT("Mono To Stereo", - tegra210_dmic_mono_conv_enum, tegra210_dmic_get_control, - tegra210_dmic_put_control), + tegra210_dmic_mono_conv_enum, + tegra210_dmic_get_mono_to_stereo, + tegra210_dmic_put_mono_to_stereo), SOC_ENUM_EXT("Stereo To Mono", - tegra210_dmic_stereo_conv_enum, tegra210_dmic_get_control, - tegra210_dmic_put_control), + tegra210_dmic_stereo_conv_enum, + tegra210_dmic_get_stereo_to_mono, + tegra210_dmic_put_stereo_to_mono), SOC_ENUM_EXT("OSR Value", tegra210_dmic_osr_enum, - tegra210_dmic_get_control, tegra210_dmic_put_control), + tegra210_dmic_get_osr_val, tegra210_dmic_put_osr_val), SOC_ENUM_EXT("LR Polarity Select", tegra210_dmic_lrsel_enum, - tegra210_dmic_get_control, tegra210_dmic_put_control), + tegra210_dmic_get_pol_sel, tegra210_dmic_put_pol_sel), }; static const struct snd_soc_component_driver tegra210_dmic_compnt = { -- cgit v1.2.3 From d6202a57e79d102271d38c34481fedc9d4c79694 Mon Sep 17 00:00:00 2001 From: Sameer Pujar Date: Thu, 18 Nov 2021 12:37:05 +0530 Subject: ASoC: tegra: Fix kcontrol put callback in DSPK The kcontrol put callback is expected to return 1 when there is change in HW or when the update is acknowledged by driver. This would ensure that change notifications are sent to subscribed applications. Update the DSPK driver accordingly. Fixes: 327ef6470266 ("ASoC: tegra: Add Tegra186 based DSPK driver") Suggested-by: Jaroslav Kysela Suggested-by: Mark Brown Signed-off-by: Sameer Pujar Reviewed-by: Takashi Iwai Link: https://lore.kernel.org/r/1637219231-406-11-git-send-email-spujar@nvidia.com Signed-off-by: Mark Brown --- sound/soc/tegra/tegra186_dspk.c | 178 ++++++++++++++++++++++++++++++++-------- 1 file changed, 146 insertions(+), 32 deletions(-) (limited to 'sound') diff --git a/sound/soc/tegra/tegra186_dspk.c b/sound/soc/tegra/tegra186_dspk.c index 67269e77d6e8..a74c980ee775 100644 --- a/sound/soc/tegra/tegra186_dspk.c +++ b/sound/soc/tegra/tegra186_dspk.c @@ -26,50 +26,162 @@ static const struct reg_default tegra186_dspk_reg_defaults[] = { { TEGRA186_DSPK_CODEC_CTRL, 0x03000000 }, }; -static int tegra186_dspk_get_control(struct snd_kcontrol *kcontrol, +static int tegra186_dspk_get_fifo_th(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); struct tegra186_dspk *dspk = snd_soc_component_get_drvdata(codec); - if (strstr(kcontrol->id.name, "FIFO Threshold")) - ucontrol->value.integer.value[0] = dspk->rx_fifo_th; - else if (strstr(kcontrol->id.name, "OSR Value")) - ucontrol->value.enumerated.item[0] = dspk->osr_val; - else if (strstr(kcontrol->id.name, "LR Polarity Select")) - ucontrol->value.enumerated.item[0] = dspk->lrsel; - else if (strstr(kcontrol->id.name, "Channel Select")) - ucontrol->value.enumerated.item[0] = dspk->ch_sel; - else if (strstr(kcontrol->id.name, "Mono To Stereo")) - ucontrol->value.enumerated.item[0] = dspk->mono_to_stereo; - else if (strstr(kcontrol->id.name, "Stereo To Mono")) - ucontrol->value.enumerated.item[0] = dspk->stereo_to_mono; + ucontrol->value.integer.value[0] = dspk->rx_fifo_th; return 0; } -static int tegra186_dspk_put_control(struct snd_kcontrol *kcontrol, +static int tegra186_dspk_put_fifo_th(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); struct tegra186_dspk *dspk = snd_soc_component_get_drvdata(codec); + int value = ucontrol->value.integer.value[0]; - if (strstr(kcontrol->id.name, "FIFO Threshold")) - dspk->rx_fifo_th = ucontrol->value.integer.value[0]; - else if (strstr(kcontrol->id.name, "OSR Value")) - dspk->osr_val = ucontrol->value.enumerated.item[0]; - else if (strstr(kcontrol->id.name, "LR Polarity Select")) - dspk->lrsel = ucontrol->value.enumerated.item[0]; - else if (strstr(kcontrol->id.name, "Channel Select")) - dspk->ch_sel = ucontrol->value.enumerated.item[0]; - else if (strstr(kcontrol->id.name, "Mono To Stereo")) - dspk->mono_to_stereo = ucontrol->value.enumerated.item[0]; - else if (strstr(kcontrol->id.name, "Stereo To Mono")) - dspk->stereo_to_mono = ucontrol->value.enumerated.item[0]; + if (value == dspk->rx_fifo_th) + return 0; + + dspk->rx_fifo_th = value; + + return 1; +} + +static int tegra186_dspk_get_osr_val(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct tegra186_dspk *dspk = snd_soc_component_get_drvdata(codec); + + ucontrol->value.enumerated.item[0] = dspk->osr_val; + + return 0; +} + +static int tegra186_dspk_put_osr_val(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct tegra186_dspk *dspk = snd_soc_component_get_drvdata(codec); + unsigned int value = ucontrol->value.enumerated.item[0]; + + if (value == dspk->osr_val) + return 0; + + dspk->osr_val = value; + + return 1; +} + +static int tegra186_dspk_get_pol_sel(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct tegra186_dspk *dspk = snd_soc_component_get_drvdata(codec); + + ucontrol->value.enumerated.item[0] = dspk->lrsel; + + return 0; +} + +static int tegra186_dspk_put_pol_sel(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct tegra186_dspk *dspk = snd_soc_component_get_drvdata(codec); + unsigned int value = ucontrol->value.enumerated.item[0]; + + if (value == dspk->lrsel) + return 0; + + dspk->lrsel = value; + + return 1; +} + +static int tegra186_dspk_get_ch_sel(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct tegra186_dspk *dspk = snd_soc_component_get_drvdata(codec); + + ucontrol->value.enumerated.item[0] = dspk->ch_sel; + + return 0; +} + +static int tegra186_dspk_put_ch_sel(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct tegra186_dspk *dspk = snd_soc_component_get_drvdata(codec); + unsigned int value = ucontrol->value.enumerated.item[0]; + + if (value == dspk->ch_sel) + return 0; + + dspk->ch_sel = value; + + return 1; +} + +static int tegra186_dspk_get_mono_to_stereo(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct tegra186_dspk *dspk = snd_soc_component_get_drvdata(codec); + + ucontrol->value.enumerated.item[0] = dspk->mono_to_stereo; + + return 0; +} + +static int tegra186_dspk_put_mono_to_stereo(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct tegra186_dspk *dspk = snd_soc_component_get_drvdata(codec); + unsigned int value = ucontrol->value.enumerated.item[0]; + + if (value == dspk->mono_to_stereo) + return 0; + + dspk->mono_to_stereo = value; + + return 1; +} + +static int tegra186_dspk_get_stereo_to_mono(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct tegra186_dspk *dspk = snd_soc_component_get_drvdata(codec); + + ucontrol->value.enumerated.item[0] = dspk->stereo_to_mono; return 0; } +static int tegra186_dspk_put_stereo_to_mono(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct tegra186_dspk *dspk = snd_soc_component_get_drvdata(codec); + unsigned int value = ucontrol->value.enumerated.item[0]; + + if (value == dspk->stereo_to_mono) + return 0; + + dspk->stereo_to_mono = value; + + return 1; +} + static int __maybe_unused tegra186_dspk_runtime_suspend(struct device *dev) { struct tegra186_dspk *dspk = dev_get_drvdata(dev); @@ -278,17 +390,19 @@ static const struct soc_enum tegra186_dspk_lrsel_enum = static const struct snd_kcontrol_new tegrat186_dspk_controls[] = { SOC_SINGLE_EXT("FIFO Threshold", SND_SOC_NOPM, 0, TEGRA186_DSPK_RX_FIFO_DEPTH - 1, 0, - tegra186_dspk_get_control, tegra186_dspk_put_control), + tegra186_dspk_get_fifo_th, tegra186_dspk_put_fifo_th), SOC_ENUM_EXT("OSR Value", tegra186_dspk_osr_enum, - tegra186_dspk_get_control, tegra186_dspk_put_control), + tegra186_dspk_get_osr_val, tegra186_dspk_put_osr_val), SOC_ENUM_EXT("LR Polarity Select", tegra186_dspk_lrsel_enum, - tegra186_dspk_get_control, tegra186_dspk_put_control), + tegra186_dspk_get_pol_sel, tegra186_dspk_put_pol_sel), SOC_ENUM_EXT("Channel Select", tegra186_dspk_ch_sel_enum, - tegra186_dspk_get_control, tegra186_dspk_put_control), + tegra186_dspk_get_ch_sel, tegra186_dspk_put_ch_sel), SOC_ENUM_EXT("Mono To Stereo", tegra186_dspk_mono_conv_enum, - tegra186_dspk_get_control, tegra186_dspk_put_control), + tegra186_dspk_get_mono_to_stereo, + tegra186_dspk_put_mono_to_stereo), SOC_ENUM_EXT("Stereo To Mono", tegra186_dspk_stereo_conv_enum, - tegra186_dspk_get_control, tegra186_dspk_put_control), + tegra186_dspk_get_stereo_to_mono, + tegra186_dspk_put_stereo_to_mono), }; static const struct snd_soc_component_driver tegra186_dspk_cmpnt = { -- cgit v1.2.3 From a4e37950c9e9b126f9cbee79b8ab94a94646dcf1 Mon Sep 17 00:00:00 2001 From: Sameer Pujar Date: Thu, 18 Nov 2021 12:37:06 +0530 Subject: ASoC: tegra: Fix kcontrol put callback in AHUB The kcontrol put callback is expected to return 1 when there is change in HW or when the update is acknowledged by driver. This would ensure that change notifications are sent to subscribed applications. Update the AHUB driver accordingly. Fixes: 16e1bcc2caf4 ("ASoC: tegra: Add Tegra210 based AHUB driver") Signed-off-by: Sameer Pujar Suggested-by: Jaroslav Kysela Suggested-by: Mark Brown Reviewed-by: Takashi Iwai Link: https://lore.kernel.org/r/1637219231-406-12-git-send-email-spujar@nvidia.com Signed-off-by: Mark Brown --- sound/soc/tegra/tegra210_ahub.c | 11 +++++++---- 1 file changed, 7 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/tegra/tegra210_ahub.c b/sound/soc/tegra/tegra210_ahub.c index a1989eae2b52..388b815443c7 100644 --- a/sound/soc/tegra/tegra210_ahub.c +++ b/sound/soc/tegra/tegra210_ahub.c @@ -62,6 +62,7 @@ static int tegra_ahub_put_value_enum(struct snd_kcontrol *kctl, unsigned int *item = uctl->value.enumerated.item; unsigned int value = e->values[item[0]]; unsigned int i, bit_pos, reg_idx = 0, reg_val = 0; + int change = 0; if (item[0] >= e->items) return -EINVAL; @@ -86,12 +87,14 @@ static int tegra_ahub_put_value_enum(struct snd_kcontrol *kctl, /* Update widget power if state has changed */ if (snd_soc_component_test_bits(cmpnt, update[i].reg, - update[i].mask, update[i].val)) - snd_soc_dapm_mux_update_power(dapm, kctl, item[0], e, - &update[i]); + update[i].mask, + update[i].val)) + change |= snd_soc_dapm_mux_update_power(dapm, kctl, + item[0], e, + &update[i]); } - return 0; + return change; } static struct snd_soc_dai_driver tegra210_ahub_dais[] = { -- cgit v1.2.3 From c7b34b51bbac6ab64e873f6c9bd43564a7442e33 Mon Sep 17 00:00:00 2001 From: Sameer Pujar Date: Thu, 18 Nov 2021 12:37:07 +0530 Subject: ASoC: tegra: Fix kcontrol put callback in MVC The kcontrol put callback is expected to return 1 when there is change in HW or when the update is acknowledged by driver. This would ensure that change notifications are sent to subscribed applications. Filter out duplicate updates in MVC driver. Fixes: e539891f9687 ("ASoC: tegra: Add Tegra210 based MVC driver") Signed-off-by: Sameer Pujar Suggested-by: Jaroslav Kysela Suggested-by: Mark Brown Reviewed-by: Takashi Iwai Link: https://lore.kernel.org/r/1637219231-406-13-git-send-email-spujar@nvidia.com Signed-off-by: Mark Brown --- sound/soc/tegra/tegra210_mvc.c | 22 ++++++++++++++++++---- 1 file changed, 18 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/tegra/tegra210_mvc.c b/sound/soc/tegra/tegra210_mvc.c index b7e317065251..85b155887ec2 100644 --- a/sound/soc/tegra/tegra210_mvc.c +++ b/sound/soc/tegra/tegra210_mvc.c @@ -136,7 +136,7 @@ static int tegra210_mvc_put_mute(struct snd_kcontrol *kcontrol, struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); struct tegra210_mvc *mvc = snd_soc_component_get_drvdata(cmpnt); unsigned int value; - u8 mute_mask; + u8 new_mask, old_mask; int err; pm_runtime_get_sync(cmpnt->dev); @@ -148,11 +148,19 @@ static int tegra210_mvc_put_mute(struct snd_kcontrol *kcontrol, if (err < 0) goto end; - mute_mask = ucontrol->value.integer.value[0]; + regmap_read(mvc->regmap, TEGRA210_MVC_CTRL, &value); + + old_mask = (value >> TEGRA210_MVC_MUTE_SHIFT) & TEGRA210_MUTE_MASK_EN; + new_mask = ucontrol->value.integer.value[0]; + + if (new_mask == old_mask) { + err = 0; + goto end; + } err = regmap_update_bits(mvc->regmap, mc->reg, TEGRA210_MVC_MUTE_MASK, - mute_mask << TEGRA210_MVC_MUTE_SHIFT); + new_mask << TEGRA210_MVC_MUTE_SHIFT); if (err < 0) goto end; @@ -195,7 +203,7 @@ static int tegra210_mvc_put_vol(struct snd_kcontrol *kcontrol, unsigned int reg = mc->reg; unsigned int value; u8 chan; - int err; + int err, old_volume; pm_runtime_get_sync(cmpnt->dev); @@ -207,10 +215,16 @@ static int tegra210_mvc_put_vol(struct snd_kcontrol *kcontrol, goto end; chan = (reg - TEGRA210_MVC_TARGET_VOL) / REG_SIZE; + old_volume = mvc->volume[chan]; tegra210_mvc_conv_vol(mvc, chan, ucontrol->value.integer.value[0]); + if (mvc->volume[chan] == old_volume) { + err = 0; + goto end; + } + /* Configure init volume same as target volume */ regmap_write(mvc->regmap, TEGRA210_MVC_REG_OFFSET(TEGRA210_MVC_INIT_VOL, chan), -- cgit v1.2.3 From b31f8febd1850bbe74aba184779ec54552d92752 Mon Sep 17 00:00:00 2001 From: Sameer Pujar Date: Thu, 18 Nov 2021 12:37:08 +0530 Subject: ASoC: tegra: Fix kcontrol put callback in SFC The kcontrol put callback is expected to return 1 when there is change in HW or when the update is acknowledged by driver. This would ensure that change notifications are sent to subscribed applications. Filter out duplicate updates in SFC driver. Fixes: b2f74ec53a6c ("ASoC: tegra: Add Tegra210 based SFC driver") Signed-off-by: Sameer Pujar Suggested-by: Jaroslav Kysela Suggested-by: Mark Brown Reviewed-by: Takashi Iwai Link: https://lore.kernel.org/r/1637219231-406-14-git-send-email-spujar@nvidia.com Signed-off-by: Mark Brown --- sound/soc/tegra/tegra210_sfc.c | 124 ++++++++++++++++++++++++++++++----------- 1 file changed, 93 insertions(+), 31 deletions(-) (limited to 'sound') diff --git a/sound/soc/tegra/tegra210_sfc.c b/sound/soc/tegra/tegra210_sfc.c index cb592ef55bd3..7a2227ed3df6 100644 --- a/sound/soc/tegra/tegra210_sfc.c +++ b/sound/soc/tegra/tegra210_sfc.c @@ -3244,49 +3244,107 @@ static int tegra210_sfc_init(struct snd_soc_dapm_widget *w, return tegra210_sfc_write_coeff_ram(cmpnt); } -static int tegra210_sfc_get_control(struct snd_kcontrol *kcontrol, +static int tegra210_sfc_iget_stereo_to_mono(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); struct tegra210_sfc *sfc = snd_soc_component_get_drvdata(cmpnt); - if (strstr(kcontrol->id.name, "Input Stereo To Mono")) - ucontrol->value.enumerated.item[0] = - sfc->stereo_to_mono[SFC_RX_PATH]; - else if (strstr(kcontrol->id.name, "Input Mono To Stereo")) - ucontrol->value.enumerated.item[0] = - sfc->mono_to_stereo[SFC_RX_PATH]; - else if (strstr(kcontrol->id.name, "Output Stereo To Mono")) - ucontrol->value.enumerated.item[0] = - sfc->stereo_to_mono[SFC_TX_PATH]; - else if (strstr(kcontrol->id.name, "Output Mono To Stereo")) - ucontrol->value.enumerated.item[0] = - sfc->mono_to_stereo[SFC_TX_PATH]; + ucontrol->value.enumerated.item[0] = sfc->stereo_to_mono[SFC_RX_PATH]; return 0; } -static int tegra210_sfc_put_control(struct snd_kcontrol *kcontrol, +static int tegra210_sfc_iput_stereo_to_mono(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); struct tegra210_sfc *sfc = snd_soc_component_get_drvdata(cmpnt); + unsigned int value = ucontrol->value.enumerated.item[0]; - if (strstr(kcontrol->id.name, "Input Stereo To Mono")) - sfc->stereo_to_mono[SFC_RX_PATH] = - ucontrol->value.enumerated.item[0]; - else if (strstr(kcontrol->id.name, "Input Mono To Stereo")) - sfc->mono_to_stereo[SFC_RX_PATH] = - ucontrol->value.enumerated.item[0]; - else if (strstr(kcontrol->id.name, "Output Stereo To Mono")) - sfc->stereo_to_mono[SFC_TX_PATH] = - ucontrol->value.enumerated.item[0]; - else if (strstr(kcontrol->id.name, "Output Mono To Stereo")) - sfc->mono_to_stereo[SFC_TX_PATH] = - ucontrol->value.enumerated.item[0]; - else + if (value == sfc->stereo_to_mono[SFC_RX_PATH]) + return 0; + + sfc->stereo_to_mono[SFC_RX_PATH] = value; + + return 1; +} + +static int tegra210_sfc_iget_mono_to_stereo(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); + struct tegra210_sfc *sfc = snd_soc_component_get_drvdata(cmpnt); + + ucontrol->value.enumerated.item[0] = sfc->mono_to_stereo[SFC_RX_PATH]; + + return 0; +} + +static int tegra210_sfc_iput_mono_to_stereo(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); + struct tegra210_sfc *sfc = snd_soc_component_get_drvdata(cmpnt); + unsigned int value = ucontrol->value.enumerated.item[0]; + + if (value == sfc->mono_to_stereo[SFC_RX_PATH]) + return 0; + + sfc->mono_to_stereo[SFC_RX_PATH] = value; + + return 1; +} + +static int tegra210_sfc_oget_stereo_to_mono(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); + struct tegra210_sfc *sfc = snd_soc_component_get_drvdata(cmpnt); + + ucontrol->value.enumerated.item[0] = sfc->stereo_to_mono[SFC_TX_PATH]; + + return 0; +} + +static int tegra210_sfc_oput_stereo_to_mono(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); + struct tegra210_sfc *sfc = snd_soc_component_get_drvdata(cmpnt); + unsigned int value = ucontrol->value.enumerated.item[0]; + + if (value == sfc->stereo_to_mono[SFC_TX_PATH]) return 0; + sfc->stereo_to_mono[SFC_TX_PATH] = value; + + return 1; +} + +static int tegra210_sfc_oget_mono_to_stereo(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); + struct tegra210_sfc *sfc = snd_soc_component_get_drvdata(cmpnt); + + ucontrol->value.enumerated.item[0] = sfc->mono_to_stereo[SFC_TX_PATH]; + + return 0; +} + +static int tegra210_sfc_oput_mono_to_stereo(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); + struct tegra210_sfc *sfc = snd_soc_component_get_drvdata(cmpnt); + unsigned int value = ucontrol->value.enumerated.item[0]; + + if (value == sfc->mono_to_stereo[SFC_TX_PATH]) + return 0; + + sfc->mono_to_stereo[SFC_TX_PATH] = value; + return 1; } @@ -3387,13 +3445,17 @@ static const struct soc_enum tegra210_sfc_mono_conv_enum = static const struct snd_kcontrol_new tegra210_sfc_controls[] = { SOC_ENUM_EXT("Input Stereo To Mono", tegra210_sfc_stereo_conv_enum, - tegra210_sfc_get_control, tegra210_sfc_put_control), + tegra210_sfc_iget_stereo_to_mono, + tegra210_sfc_iput_stereo_to_mono), SOC_ENUM_EXT("Input Mono To Stereo", tegra210_sfc_mono_conv_enum, - tegra210_sfc_get_control, tegra210_sfc_put_control), + tegra210_sfc_iget_mono_to_stereo, + tegra210_sfc_iput_mono_to_stereo), SOC_ENUM_EXT("Output Stereo To Mono", tegra210_sfc_stereo_conv_enum, - tegra210_sfc_get_control, tegra210_sfc_put_control), + tegra210_sfc_oget_stereo_to_mono, + tegra210_sfc_oput_stereo_to_mono), SOC_ENUM_EXT("Output Mono To Stereo", tegra210_sfc_mono_conv_enum, - tegra210_sfc_get_control, tegra210_sfc_put_control), + tegra210_sfc_oget_mono_to_stereo, + tegra210_sfc_oput_mono_to_stereo), }; static const struct snd_soc_component_driver tegra210_sfc_cmpnt = { -- cgit v1.2.3 From 8db78ace1ba897302131422ce15c5eb04510cef8 Mon Sep 17 00:00:00 2001 From: Sameer Pujar Date: Thu, 18 Nov 2021 12:37:09 +0530 Subject: ASoC: tegra: Fix kcontrol put callback in AMX The kcontrol put callback is expected to return 1 when there is change in HW or when the update is acknowledged by driver. This would ensure that change notifications are sent to subscribed applications. Filter out duplicate updates in AMX driver. Fixes: 77f7df346c45 ("ASoC: tegra: Add Tegra210 based AMX driver") Signed-off-by: Sameer Pujar Suggested-by: Jaroslav Kysela Suggested-by: Mark Brown Reviewed-by: Takashi Iwai Link: https://lore.kernel.org/r/1637219231-406-15-git-send-email-spujar@nvidia.com Signed-off-by: Mark Brown --- sound/soc/tegra/tegra210_amx.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/soc/tegra/tegra210_amx.c b/sound/soc/tegra/tegra210_amx.c index af9bddfc3120..689576302ede 100644 --- a/sound/soc/tegra/tegra210_amx.c +++ b/sound/soc/tegra/tegra210_amx.c @@ -222,6 +222,9 @@ static int tegra210_amx_put_byte_map(struct snd_kcontrol *kcontrol, int reg = mc->reg; int value = ucontrol->value.integer.value[0]; + if (value == bytes_map[reg]) + return 0; + if (value >= 0 && value <= 255) { /* Update byte map and enable slot */ bytes_map[reg] = value; -- cgit v1.2.3 From 3c97881b8c8a2aa8afd4d7a379b7ff03884c9e4a Mon Sep 17 00:00:00 2001 From: Sameer Pujar Date: Thu, 18 Nov 2021 12:37:10 +0530 Subject: ASoC: tegra: Fix kcontrol put callback in ADX The kcontrol put callback is expected to return 1 when there is change in HW or when the update is acknowledged by driver. This would ensure that change notifications are sent to subscribed applications. Filter out duplicate updates in ADX driver. Fixes: a99ab6f395a9 ("ASoC: tegra: Add Tegra210 based ADX driver") Signed-off-by: Sameer Pujar Suggested-by: Jaroslav Kysela Suggested-by: Mark Brown Reviewed-by: Takashi Iwai Link: https://lore.kernel.org/r/1637219231-406-16-git-send-email-spujar@nvidia.com Signed-off-by: Mark Brown --- sound/soc/tegra/tegra210_adx.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/soc/tegra/tegra210_adx.c b/sound/soc/tegra/tegra210_adx.c index d7c7849c2f92..933c4503fe50 100644 --- a/sound/soc/tegra/tegra210_adx.c +++ b/sound/soc/tegra/tegra210_adx.c @@ -193,6 +193,9 @@ static int tegra210_adx_put_byte_map(struct snd_kcontrol *kcontrol, struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value;; + if (value == bytes_map[mc->reg]) + return 0; + if (value >= 0 && value <= 255) { /* update byte map and enable slot */ bytes_map[mc->reg] = value; -- cgit v1.2.3 From 8cf72c4e75a0265135d34a8e29224b4c1e92b51c Mon Sep 17 00:00:00 2001 From: Sameer Pujar Date: Thu, 18 Nov 2021 12:37:11 +0530 Subject: ASoC: tegra: Fix kcontrol put callback in Mixer The kcontrol put callback is expected to return 1 when there is change in HW or when the update is acknowledged by driver. This would ensure that change notifications are sent to subscribed applications. Filter out duplicate updates in Mixer driver. Fixes: 05bb3d5ec64a ("ASoC: tegra: Add Tegra210 based Mixer driver") Signed-off-by: Sameer Pujar Suggested-by: Jaroslav Kysela Suggested-by: Mark Brown Reviewed-by: Takashi Iwai Link: https://lore.kernel.org/r/1637219231-406-17-git-send-email-spujar@nvidia.com Signed-off-by: Mark Brown --- sound/soc/tegra/tegra210_mixer.c | 26 +++++++++++++++++++------- 1 file changed, 19 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/tegra/tegra210_mixer.c b/sound/soc/tegra/tegra210_mixer.c index 55e61776c565..51d375573cfa 100644 --- a/sound/soc/tegra/tegra210_mixer.c +++ b/sound/soc/tegra/tegra210_mixer.c @@ -192,24 +192,24 @@ static int tegra210_mixer_get_gain(struct snd_kcontrol *kcontrol, return 0; } -static int tegra210_mixer_put_gain(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) +static int tegra210_mixer_apply_gain(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol, + bool instant_gain) { struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); struct tegra210_mixer *mixer = snd_soc_component_get_drvdata(cmpnt); unsigned int reg = mc->reg, id; - bool instant_gain = false; int err; - if (strstr(kcontrol->id.name, "Instant Gain Volume")) - instant_gain = true; - /* Save gain value for specific MIXER input */ id = (reg - TEGRA210_MIXER_GAIN_CFG_RAM_ADDR_0) / TEGRA210_MIXER_GAIN_CFG_RAM_ADDR_STRIDE; + if (mixer->gain_value[id] == ucontrol->value.integer.value[0]) + return 0; + mixer->gain_value[id] = ucontrol->value.integer.value[0]; err = tegra210_mixer_configure_gain(cmpnt, id, instant_gain); @@ -221,6 +221,18 @@ static int tegra210_mixer_put_gain(struct snd_kcontrol *kcontrol, return 1; } +static int tegra210_mixer_put_gain(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return tegra210_mixer_apply_gain(kcontrol, ucontrol, false); +} + +static int tegra210_mixer_put_instant_gain(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return tegra210_mixer_apply_gain(kcontrol, ucontrol, true); +} + static int tegra210_mixer_set_audio_cif(struct tegra210_mixer *mixer, struct snd_pcm_hw_params *params, unsigned int reg, @@ -388,7 +400,7 @@ ADDER_CTRL_DECL(adder5, TEGRA210_MIXER_TX5_ADDER_CONFIG); SOC_SINGLE_EXT("RX" #id " Instant Gain Volume", \ MIXER_GAIN_CFG_RAM_ADDR((id) - 1), 0, \ 0x20000, 0, tegra210_mixer_get_gain, \ - tegra210_mixer_put_gain), + tegra210_mixer_put_instant_gain), /* Volume controls for all MIXER inputs */ static const struct snd_kcontrol_new tegra210_mixer_gain_ctls[] = { -- cgit v1.2.3 From 76c47183224c86e4011048b80f0e2d0d166f01c2 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 18 Nov 2021 22:57:29 +0100 Subject: ALSA: ctxfi: Fix out-of-range access The master and next_conj of rcs_ops are used for iterating the resource list entries, and currently those are supposed to return the current value. The problem is that next_conf may go over the last entry before the loop abort condition is evaluated, and it may return the "current" value that is beyond the array size. It was caught recently as a GPF, for example. Those return values are, however, never actually evaluated, hence basically we don't have to consider the current value as the return at all. By dropping those return values, the potential out-of-range access above is also fixed automatically. This patch changes the return type of master and next_conj callbacks to void and drop the superfluous code accordingly. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=214985 Cc: Link: https://lore.kernel.org/r/20211118215729.26257-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/ctxfi/ctamixer.c | 14 ++++++-------- sound/pci/ctxfi/ctdaio.c | 16 ++++++++-------- sound/pci/ctxfi/ctresource.c | 7 +++---- sound/pci/ctxfi/ctresource.h | 4 ++-- sound/pci/ctxfi/ctsrc.c | 7 +++---- 5 files changed, 22 insertions(+), 26 deletions(-) (limited to 'sound') diff --git a/sound/pci/ctxfi/ctamixer.c b/sound/pci/ctxfi/ctamixer.c index da6e6350ceaf..d074727c3e21 100644 --- a/sound/pci/ctxfi/ctamixer.c +++ b/sound/pci/ctxfi/ctamixer.c @@ -23,16 +23,15 @@ #define BLANK_SLOT 4094 -static int amixer_master(struct rsc *rsc) +static void amixer_master(struct rsc *rsc) { rsc->conj = 0; - return rsc->idx = container_of(rsc, struct amixer, rsc)->idx[0]; + rsc->idx = container_of(rsc, struct amixer, rsc)->idx[0]; } -static int amixer_next_conj(struct rsc *rsc) +static void amixer_next_conj(struct rsc *rsc) { rsc->conj++; - return container_of(rsc, struct amixer, rsc)->idx[rsc->conj]; } static int amixer_index(const struct rsc *rsc) @@ -331,16 +330,15 @@ int amixer_mgr_destroy(struct amixer_mgr *amixer_mgr) /* SUM resource management */ -static int sum_master(struct rsc *rsc) +static void sum_master(struct rsc *rsc) { rsc->conj = 0; - return rsc->idx = container_of(rsc, struct sum, rsc)->idx[0]; + rsc->idx = container_of(rsc, struct sum, rsc)->idx[0]; } -static int sum_next_conj(struct rsc *rsc) +static void sum_next_conj(struct rsc *rsc) { rsc->conj++; - return container_of(rsc, struct sum, rsc)->idx[rsc->conj]; } static int sum_index(const struct rsc *rsc) diff --git a/sound/pci/ctxfi/ctdaio.c b/sound/pci/ctxfi/ctdaio.c index f589da045342..7fc720046ce2 100644 --- a/sound/pci/ctxfi/ctdaio.c +++ b/sound/pci/ctxfi/ctdaio.c @@ -51,12 +51,12 @@ static const struct daio_rsc_idx idx_20k2[NUM_DAIOTYP] = { [SPDIFIO] = {.left = 0x05, .right = 0x85}, }; -static int daio_master(struct rsc *rsc) +static void daio_master(struct rsc *rsc) { /* Actually, this is not the resource index of DAIO. * For DAO, it is the input mapper index. And, for DAI, * it is the output time-slot index. */ - return rsc->conj = rsc->idx; + rsc->conj = rsc->idx; } static int daio_index(const struct rsc *rsc) @@ -64,19 +64,19 @@ static int daio_index(const struct rsc *rsc) return rsc->conj; } -static int daio_out_next_conj(struct rsc *rsc) +static void daio_out_next_conj(struct rsc *rsc) { - return rsc->conj += 2; + rsc->conj += 2; } -static int daio_in_next_conj_20k1(struct rsc *rsc) +static void daio_in_next_conj_20k1(struct rsc *rsc) { - return rsc->conj += 0x200; + rsc->conj += 0x200; } -static int daio_in_next_conj_20k2(struct rsc *rsc) +static void daio_in_next_conj_20k2(struct rsc *rsc) { - return rsc->conj += 0x100; + rsc->conj += 0x100; } static const struct rsc_ops daio_out_rsc_ops = { diff --git a/sound/pci/ctxfi/ctresource.c b/sound/pci/ctxfi/ctresource.c index 81ad26934518..be1d3e61309c 100644 --- a/sound/pci/ctxfi/ctresource.c +++ b/sound/pci/ctxfi/ctresource.c @@ -109,18 +109,17 @@ static int audio_ring_slot(const struct rsc *rsc) return (rsc->conj << 4) + offset_in_audio_slot_block[rsc->type]; } -static int rsc_next_conj(struct rsc *rsc) +static void rsc_next_conj(struct rsc *rsc) { unsigned int i; for (i = 0; (i < 8) && (!(rsc->msr & (0x1 << i))); ) i++; rsc->conj += (AUDIO_SLOT_BLOCK_NUM >> i); - return rsc->conj; } -static int rsc_master(struct rsc *rsc) +static void rsc_master(struct rsc *rsc) { - return rsc->conj = rsc->idx; + rsc->conj = rsc->idx; } static const struct rsc_ops rsc_generic_ops = { diff --git a/sound/pci/ctxfi/ctresource.h b/sound/pci/ctxfi/ctresource.h index fdbfd808816d..58553bda44f4 100644 --- a/sound/pci/ctxfi/ctresource.h +++ b/sound/pci/ctxfi/ctresource.h @@ -39,8 +39,8 @@ struct rsc { }; struct rsc_ops { - int (*master)(struct rsc *rsc); /* Move to master resource */ - int (*next_conj)(struct rsc *rsc); /* Move to next conjugate resource */ + void (*master)(struct rsc *rsc); /* Move to master resource */ + void (*next_conj)(struct rsc *rsc); /* Move to next conjugate resource */ int (*index)(const struct rsc *rsc); /* Return the index of resource */ /* Return the output slot number */ int (*output_slot)(const struct rsc *rsc); diff --git a/sound/pci/ctxfi/ctsrc.c b/sound/pci/ctxfi/ctsrc.c index bd4697b44233..4a94b4708a77 100644 --- a/sound/pci/ctxfi/ctsrc.c +++ b/sound/pci/ctxfi/ctsrc.c @@ -590,16 +590,15 @@ int src_mgr_destroy(struct src_mgr *src_mgr) /* SRCIMP resource manager operations */ -static int srcimp_master(struct rsc *rsc) +static void srcimp_master(struct rsc *rsc) { rsc->conj = 0; - return rsc->idx = container_of(rsc, struct srcimp, rsc)->idx[0]; + rsc->idx = container_of(rsc, struct srcimp, rsc)->idx[0]; } -static int srcimp_next_conj(struct rsc *rsc) +static void srcimp_next_conj(struct rsc *rsc) { rsc->conj++; - return container_of(rsc, struct srcimp, rsc)->idx[rsc->conj]; } static int srcimp_index(const struct rsc *rsc) -- cgit v1.2.3 From eee5d6f1356a016105a974fb176b491288439efa Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 19 Nov 2021 11:24:59 +0100 Subject: ALSA: usb-audio: Switch back to non-latency mode at a later point The recent regression report revealed that the judgment of the low-latency playback mode based on the runtime->stop_threshold cannot work reliably at the prepare stage, as sw_params call may happen at any time, and PCM dmix actually sets it up after the prepare call. This ended up with the stall of the stream as PCM ack won't be issued at all. For addressing this, check the free-wheeling mode again at the PCM trigger right before starting the stream again, and allow switching to the non-LL mode at a late stage. Fixes: d5f871f89e21 ("ALSA: usb-audio: Improved lowlatency playback support") Reported-and-tested-by: Kirill A. Shutemov Link: https://lore.kernel.org/r/20211117161855.m45mxcqszkfcetai@box.shutemov.name Link: https://lore.kernel.org/r/20211119102459.7055-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/pcm.c | 11 +++++++++-- 1 file changed, 9 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 95ec8eec1bb0..57b046e73bfe 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -581,6 +581,12 @@ static int snd_usb_hw_free(struct snd_pcm_substream *substream) return 0; } +/* free-wheeling mode? (e.g. dmix) */ +static int in_free_wheeling_mode(struct snd_pcm_runtime *runtime) +{ + return runtime->stop_threshold > runtime->buffer_size; +} + /* check whether early start is needed for playback stream */ static int lowlatency_playback_available(struct snd_pcm_runtime *runtime, struct snd_usb_substream *subs) @@ -592,8 +598,7 @@ static int lowlatency_playback_available(struct snd_pcm_runtime *runtime, /* disabled via module option? */ if (!chip->lowlatency) return false; - /* free-wheeling mode? (e.g. dmix) */ - if (runtime->stop_threshold > runtime->buffer_size) + if (in_free_wheeling_mode(runtime)) return false; /* implicit feedback mode has own operation mode */ if (snd_usb_endpoint_implicit_feedback_sink(subs->data_endpoint)) @@ -1552,6 +1557,8 @@ static int snd_usb_substream_playback_trigger(struct snd_pcm_substream *substrea subs); if (subs->lowlatency_playback && cmd == SNDRV_PCM_TRIGGER_START) { + if (in_free_wheeling_mode(substream->runtime)) + subs->lowlatency_playback = false; err = start_endpoints(subs); if (err < 0) { snd_usb_endpoint_set_callback(subs->data_endpoint, -- cgit v1.2.3 From 83de8f83816e8e15227dac985163e3d433a2bf9d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 19 Nov 2021 11:26:29 +0100 Subject: ALSA: usb-audio: Don't start stream for capture at prepare The recent change made mistakenly the stream for capture started at prepare stage. Add the stream direction check to avoid it. Fixes: 9c9a3b9da891 ("ALSA: usb-audio: Rename early_playback_start flag with lowlatency_playback") Link: https://lore.kernel.org/r/20211119102629.7476-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/pcm.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 57b046e73bfe..cec6e91afea2 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -640,7 +640,8 @@ static int snd_usb_pcm_prepare(struct snd_pcm_substream *substream) runtime->delay = 0; subs->lowlatency_playback = lowlatency_playback_available(runtime, subs); - if (!subs->lowlatency_playback) + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && + !subs->lowlatency_playback) ret = start_endpoints(subs); unlock: -- cgit v1.2.3 From fa9730b4f28b7bd183d28a0bf636ab7108de35d7 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Wed, 27 Oct 2021 10:32:54 +0800 Subject: ALSA: intel-dsp-config: add quirk for JSL devices based on ES8336 codec These devices are based on an I2C/I2S device, we need to force the use of the SOF driver otherwise the legacy HDaudio driver will be loaded - only HDMI will be supported. We previously added support for other Intel platforms but missed JasperLake. BugLink: https://github.com/thesofproject/linux/issues/3210 Fixes: 9d36ceab9415 ('ALSA: intel-dsp-config: add quirk for APL/GLK/TGL devices based on ES8336 codec') Signed-off-by: Pierre-Louis Bossart Reviewed-by: Kai Vehmanen Signed-off-by: Bard Liao Link: https://lore.kernel.org/r/20211027023254.24955-1-yung-chuan.liao@linux.intel.com Signed-off-by: Takashi Iwai --- sound/hda/intel-dsp-config.c | 9 +++++++++ 1 file changed, 9 insertions(+) (limited to 'sound') diff --git a/sound/hda/intel-dsp-config.c b/sound/hda/intel-dsp-config.c index b9ac9e9e45a4..10a0bffc3cf6 100644 --- a/sound/hda/intel-dsp-config.c +++ b/sound/hda/intel-dsp-config.c @@ -299,6 +299,15 @@ static const struct config_entry config_table[] = { }, #endif +/* JasperLake */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_JASPERLAKE) + { + .flags = FLAG_SOF, + .device = 0x4dc8, + .codec_hid = "ESSX8336", + }, +#endif + /* Tigerlake */ #if IS_ENABLED(CONFIG_SND_SOC_SOF_TIGERLAKE) { -- cgit v1.2.3 From 28c916ade1bd4205958f74bb817fd3a05dbb7afc Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Thu, 18 Nov 2021 16:30:14 +0100 Subject: ASoC: soc-acpi: Set mach->id field on comp_ids matches Commit dac7cbd55dca ("ASoC: Intel: soc-acpi-byt: shrink tables using compatible IDs") and commit 959ae8215a9e ("ASoC: Intel: soc-acpi-cht: shrink tables using compatible IDs") simplified the match tables in soc-acpi-intel-byt-match.c and soc-acpi-intel-cht-match.c by merging identical entries using the new .comp_ids snd_soc_acpi_mach field to point a single entry to multiple ACPI HIDs and clearing the previously unique per entry .id field. But various machine drivers from sound/soc/intel/boards rely on mach->id in one or more ways, e.g. some drivers contain the following snippets: adev = acpi_dev_get_first_match_dev(mach->id, NULL, -1); pkg_found = snd_soc_acpi_find_package_from_hid(mach->id, ... if (!strncmp(snd_soc_cards[i].codec_id, mach->id, 8)) { ... All of which are broken by the match table shrinking. Make the snd_soc_acpi_mach.id field non const (the storage for the tables already is non const) and on a comps_ids match copy the matching HID to the id field to fix this. Fixes: dac7cbd55dca ("ASoC: Intel: soc-acpi-byt: shrink tables using compatible IDs") Fixes: 959ae8215a9e ("ASoC: Intel: soc-acpi-cht: shrink tables using compatible IDs") Suggested-by: Pierre-Louis Bossart Cc: Pierre-Louis Bossart Cc: Brent Lu Signed-off-by: Hans de Goede Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20211118153014.349222-1-hdegoede@redhat.com Signed-off-by: Mark Brown --- include/sound/soc-acpi.h | 2 +- sound/soc/soc-acpi.c | 4 +++- 2 files changed, 4 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/include/sound/soc-acpi.h b/include/sound/soc-acpi.h index 31f4c4f9aeea..ac0893df9c76 100644 --- a/include/sound/soc-acpi.h +++ b/include/sound/soc-acpi.h @@ -147,7 +147,7 @@ struct snd_soc_acpi_link_adr { */ /* Descriptor for SST ASoC machine driver */ struct snd_soc_acpi_mach { - const u8 id[ACPI_ID_LEN]; + u8 id[ACPI_ID_LEN]; const struct snd_soc_acpi_codecs *comp_ids; const u32 link_mask; const struct snd_soc_acpi_link_adr *links; diff --git a/sound/soc/soc-acpi.c b/sound/soc/soc-acpi.c index 2ae99b49d3f5..cbd7ea48837b 100644 --- a/sound/soc/soc-acpi.c +++ b/sound/soc/soc-acpi.c @@ -20,8 +20,10 @@ static bool snd_soc_acpi_id_present(struct snd_soc_acpi_mach *machine) if (comp_ids) { for (i = 0; i < comp_ids->num_codecs; i++) { - if (acpi_dev_present(comp_ids->codecs[i], NULL, -1)) + if (acpi_dev_present(comp_ids->codecs[i], NULL, -1)) { + strscpy(machine->id, comp_ids->codecs[i], ACPI_ID_LEN); return true; + } } } -- cgit v1.2.3 From 428ee30a05cd1362c8aa86a4c909b0d1c6bc48a4 Mon Sep 17 00:00:00 2001 From: Nicolas Frattaroli Date: Sun, 21 Nov 2021 16:05:20 +0100 Subject: ASoC: rk817: Add module alias for rk817-codec Without a module alias, autoloading the driver does not occurr when it is built as a module. By adding a module alias, the driver now probes fine automatically and therefore analog audio output works as it should. Fixes: 0d6a04da9b25 ("ASoC: Add Rockchip rk817 audio CODEC support") Signed-off-by: Nicolas Frattaroli Link: https://lore.kernel.org/r/20211121150521.159543-1-frattaroli.nicolas@gmail.com Signed-off-by: Mark Brown --- sound/soc/codecs/rk817_codec.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/codecs/rk817_codec.c b/sound/soc/codecs/rk817_codec.c index 943d7d933e81..03f24edfe4f6 100644 --- a/sound/soc/codecs/rk817_codec.c +++ b/sound/soc/codecs/rk817_codec.c @@ -539,3 +539,4 @@ module_platform_driver(rk817_codec_driver); MODULE_DESCRIPTION("ASoC RK817 codec driver"); MODULE_AUTHOR("binyuan "); MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:rk817-codec"); -- cgit v1.2.3 From 8a6cc0ded6d942e4a506c421c4d87a634bda6e75 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Mon, 22 Nov 2021 17:23:56 -0600 Subject: ASoC: Intel: soc-acpi: add entry for ESSX8336 on CML We have configurations for this codec on APL, GLK, JSL and TGL, somehow the information that some designs rely on CometLake was not shared. BugLink: https://github.com/thesofproject/linux/issues/3248 Fixes: 790049fb6623 ("ASoC: Intel: soc-acpi: apl/glk/tgl: add entry for devices based on ES8336 codec") Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20211122232356.23505-1-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/common/soc-acpi-intel-cml-match.c | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'sound') diff --git a/sound/soc/intel/common/soc-acpi-intel-cml-match.c b/sound/soc/intel/common/soc-acpi-intel-cml-match.c index b4eb0c97edf1..4eebc79d4b48 100644 --- a/sound/soc/intel/common/soc-acpi-intel-cml-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-cml-match.c @@ -81,6 +81,12 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cml_machines[] = { .sof_fw_filename = "sof-cml.ri", .sof_tplg_filename = "sof-cml-da7219-max98390.tplg", }, + { + .id = "ESSX8336", + .drv_name = "sof-essx8336", + .sof_fw_filename = "sof-cml.ri", + .sof_tplg_filename = "sof-cml-es8336.tplg", + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_cml_machines); -- cgit v1.2.3 From ae26c08e6c8071ba8febb0c7c0829da96c75248c Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Mon, 22 Nov 2021 17:22:54 -0600 Subject: ALSA: intel-dsp-config: add quirk for CML devices based on ES8336 codec We've added quirks for ESS8336 but missed CML, add quirks for both LP and H versions. BugLink: https://github.com/thesofproject/linux/issues/3248 Fixes: 9d36ceab9415 ("ALSA: intel-dsp-config: add quirk for APL/GLK/TGL devices based on ES8336 codec") Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20211122232254.23362-1-pierre-louis.bossart@linux.intel.com Signed-off-by: Takashi Iwai --- sound/hda/intel-dsp-config.c | 10 ++++++++++ 1 file changed, 10 insertions(+) (limited to 'sound') diff --git a/sound/hda/intel-dsp-config.c b/sound/hda/intel-dsp-config.c index 10a0bffc3cf6..4208fa8a4db5 100644 --- a/sound/hda/intel-dsp-config.c +++ b/sound/hda/intel-dsp-config.c @@ -252,6 +252,11 @@ static const struct config_entry config_table[] = { .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE, .device = 0x02c8, }, + { + .flags = FLAG_SOF, + .device = 0x02c8, + .codec_hid = "ESSX8336", + }, /* Cometlake-H */ { .flags = FLAG_SOF, @@ -276,6 +281,11 @@ static const struct config_entry config_table[] = { .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE, .device = 0x06c8, }, + { + .flags = FLAG_SOF, + .device = 0x06c8, + .codec_hid = "ESSX8336", + }, #endif /* Icelake */ -- cgit v1.2.3 From de6da33e6cb79abd4a5721b65b9a7dbed24378f8 Mon Sep 17 00:00:00 2001 From: Juergen Gross Date: Fri, 22 Oct 2021 08:48:00 +0200 Subject: xen: flag xen_snd_front to be not essential for system boot The Xen pv sound driver is not essential for booting. Set the respective flag. [boris: replace semicolon with comma] Signed-off-by: Juergen Gross Reviewed-by: Oleksandr Andrushchenko Reviewed-by: Boris Ostrovsky Link: https://lore.kernel.org/r/20211022064800.14978-6-jgross@suse.com Signed-off-by: Boris Ostrovsky --- sound/xen/xen_snd_front.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/xen/xen_snd_front.c b/sound/xen/xen_snd_front.c index 2cb0a19be2b8..4041748c12e5 100644 --- a/sound/xen/xen_snd_front.c +++ b/sound/xen/xen_snd_front.c @@ -358,6 +358,7 @@ static struct xenbus_driver xen_driver = { .probe = xen_drv_probe, .remove = xen_drv_remove, .otherend_changed = sndback_changed, + .not_essential = true, }; static int __init xen_drv_init(void) -- cgit v1.2.3 From 872fc0b6bde8b2dd6891c740cd792d214255dca3 Mon Sep 17 00:00:00 2001 From: Lucas Tanure Date: Tue, 23 Nov 2021 16:31:39 +0000 Subject: ASoC: cs35l41: Set the max SPI speed for the whole device Higher speeds are only supported when PLL is enabled, but the current driver doesn't enable PLL outside of stream use cases, so better to set the lowest SPI speed accepted by the entire device. Move the current frequency set to the spi sub-driver so the whole device can benefit from that speed. spi-max-frequency property could be used, but ACPI systems don't support it, so by setting it in the spi sub-driver probe both Device Trees and ACPI systems are supported. Signed-off-by: Lucas Tanure Reviewed-by: Charles Keepax Link: https://lore.kernel.org/r/20211123163149.1530535-2-tanureal@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs35l41-spi.c | 32 +++----------------------------- sound/soc/codecs/cs35l41.c | 7 ------- sound/soc/codecs/cs35l41.h | 4 +--- 3 files changed, 4 insertions(+), 39 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs35l41-spi.c b/sound/soc/codecs/cs35l41-spi.c index 90a921f726c3..3fa99741779a 100644 --- a/sound/soc/codecs/cs35l41-spi.c +++ b/sound/soc/codecs/cs35l41-spi.c @@ -42,34 +42,6 @@ static const struct spi_device_id cs35l41_id_spi[] = { MODULE_DEVICE_TABLE(spi, cs35l41_id_spi); -static void cs35l41_spi_otp_setup(struct cs35l41_private *cs35l41, - bool is_pre_setup, unsigned int *freq) -{ - struct spi_device *spi; - u32 orig_spi_freq; - - spi = to_spi_device(cs35l41->dev); - - if (!spi) { - dev_err(cs35l41->dev, "%s: No SPI device\n", __func__); - return; - } - - if (is_pre_setup) { - orig_spi_freq = spi->max_speed_hz; - if (orig_spi_freq > CS35L41_SPI_MAX_FREQ_OTP) { - spi->max_speed_hz = CS35L41_SPI_MAX_FREQ_OTP; - spi_setup(spi); - } - *freq = orig_spi_freq; - } else { - if (spi->max_speed_hz != *freq) { - spi->max_speed_hz = *freq; - spi_setup(spi); - } - } -} - static int cs35l41_spi_probe(struct spi_device *spi) { const struct regmap_config *regmap_config = &cs35l41_regmap_spi; @@ -81,6 +53,9 @@ static int cs35l41_spi_probe(struct spi_device *spi) if (!cs35l41) return -ENOMEM; + spi->max_speed_hz = CS35L41_SPI_MAX_FREQ; + spi_setup(spi); + spi_set_drvdata(spi, cs35l41); cs35l41->regmap = devm_regmap_init_spi(spi, regmap_config); if (IS_ERR(cs35l41->regmap)) { @@ -91,7 +66,6 @@ static int cs35l41_spi_probe(struct spi_device *spi) cs35l41->dev = &spi->dev; cs35l41->irq = spi->irq; - cs35l41->otp_setup = cs35l41_spi_otp_setup; return cs35l41_probe(cs35l41, pdata); } diff --git a/sound/soc/codecs/cs35l41.c b/sound/soc/codecs/cs35l41.c index 9d0530dde996..9c4d481f7614 100644 --- a/sound/soc/codecs/cs35l41.c +++ b/sound/soc/codecs/cs35l41.c @@ -302,7 +302,6 @@ static int cs35l41_otp_unpack(void *data) const struct cs35l41_otp_packed_element_t *otp_map; struct cs35l41_private *cs35l41 = data; int bit_offset, word_offset, ret, i; - unsigned int orig_spi_freq; unsigned int bit_sum = 8; u32 otp_val, otp_id_reg; u32 *otp_mem; @@ -326,9 +325,6 @@ static int cs35l41_otp_unpack(void *data) goto err_otp_unpack; } - if (cs35l41->otp_setup) - cs35l41->otp_setup(cs35l41, true, &orig_spi_freq); - ret = regmap_bulk_read(cs35l41->regmap, CS35L41_OTP_MEM0, otp_mem, CS35L41_OTP_SIZE_WORDS); if (ret < 0) { @@ -336,9 +332,6 @@ static int cs35l41_otp_unpack(void *data) goto err_otp_unpack; } - if (cs35l41->otp_setup) - cs35l41->otp_setup(cs35l41, false, &orig_spi_freq); - otp_map = otp_map_match->map; bit_offset = otp_map_match->bit_offset; diff --git a/sound/soc/codecs/cs35l41.h b/sound/soc/codecs/cs35l41.h index 6cffe8a55beb..48485b08a6f1 100644 --- a/sound/soc/codecs/cs35l41.h +++ b/sound/soc/codecs/cs35l41.h @@ -726,7 +726,7 @@ #define CS35L41_FS2_WINDOW_MASK 0x00FFF800 #define CS35L41_FS2_WINDOW_SHIFT 12 -#define CS35L41_SPI_MAX_FREQ_OTP 4000000 +#define CS35L41_SPI_MAX_FREQ 4000000 #define CS35L41_RX_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE) #define CS35L41_TX_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE) @@ -764,8 +764,6 @@ struct cs35l41_private { int irq; /* GPIO for /RST */ struct gpio_desc *reset_gpio; - void (*otp_setup)(struct cs35l41_private *cs35l41, bool is_pre_setup, - unsigned int *freq); }; int cs35l41_probe(struct cs35l41_private *cs35l41, -- cgit v1.2.3 From 86f74ba3fef56dd1cee19b7a15ae27fc0da5bb61 Mon Sep 17 00:00:00 2001 From: Ranjani Sridharan Date: Tue, 23 Nov 2021 18:57:59 +0200 Subject: ASoC: SOF: hda: reset DAI widget before reconfiguring it It is not unusual for ALSA/ASoC hw_params callbacks to be invoked multiple times. Reset and free the DAI widget before reconfiguring it to keep the DAI widget use_count balanced. Fixes: 0acb48dd31e3 ("ASoC: SOF: Intel: hda: make sure DAI widget is set up before IPC") Signed-off-by: Ranjani Sridharan Reviewed-by: Paul Olaru Reviewed-by: Bard Liao Reviewed-by: Pierre-Louis Bossart Signed-off-by: Kai Vehmanen Link: https://lore.kernel.org/r/20211123165759.127884-1-kai.vehmanen@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda.c | 7 +++++++ 1 file changed, 7 insertions(+) (limited to 'sound') diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index 568d351b7a4e..2c0d4d06ab36 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -58,6 +58,13 @@ int hda_ctrl_dai_widget_setup(struct snd_soc_dapm_widget *w) return -EINVAL; } + /* DAI already configured, reset it before reconfiguring it */ + if (sof_dai->configured) { + ret = hda_ctrl_dai_widget_free(w); + if (ret < 0) + return ret; + } + config = &sof_dai->dai_config[sof_dai->current_config]; /* -- cgit v1.2.3 From 70408f755f589f67957b9ec6852e6b01f858d0a2 Mon Sep 17 00:00:00 2001 From: Sameer Pujar Date: Tue, 23 Nov 2021 19:37:34 +0530 Subject: ASoC: tegra: Balance runtime PM count After successful application of volume/mute settings via mixer control put calls, the control returns without balancing the runtime PM count. This makes device to be always runtime active. Fix this by allowing control to reach pm_runtime_put() call. Fixes: e539891f9687 ("ASoC: tegra: Add Tegra210 based MVC driver") Cc: stable@vger.kernel.org Signed-off-by: Sameer Pujar Link: https://lore.kernel.org/r/1637676459-31191-2-git-send-email-spujar@nvidia.com Signed-off-by: Mark Brown --- sound/soc/tegra/tegra210_mvc.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/tegra/tegra210_mvc.c b/sound/soc/tegra/tegra210_mvc.c index 85b155887ec2..436bb151fed1 100644 --- a/sound/soc/tegra/tegra210_mvc.c +++ b/sound/soc/tegra/tegra210_mvc.c @@ -164,7 +164,7 @@ static int tegra210_mvc_put_mute(struct snd_kcontrol *kcontrol, if (err < 0) goto end; - return 1; + err = 1; end: pm_runtime_put(cmpnt->dev); @@ -236,7 +236,7 @@ static int tegra210_mvc_put_vol(struct snd_kcontrol *kcontrol, TEGRA210_MVC_VOLUME_SWITCH_MASK, TEGRA210_MVC_VOLUME_SWITCH_TRIGGER); - return 1; + err = 1; end: pm_runtime_put(cmpnt->dev); -- cgit v1.2.3 From af120d07bbb0721708b10204beed66ed2cb0cb62 Mon Sep 17 00:00:00 2001 From: Sameer Pujar Date: Tue, 23 Nov 2021 19:37:35 +0530 Subject: ASoC: tegra: Use normal system sleep for SFC The driver currently subscribes for a late system sleep call. The initcall_debug log shows that suspend call for SFC device happens after the parent device (AHUB). This seems to cause suspend failure on Jetson TX2 platform. Also there is no use of having late system sleep specifically for SFC device. Fix the order by using normal system sleep. Fixes: b2f74ec53a6c ("ASoC: tegra: Add Tegra210 based SFC driver") Cc: stable@vger.kernel.org Signed-off-by: Sameer Pujar Link: https://lore.kernel.org/r/1637676459-31191-3-git-send-email-spujar@nvidia.com Signed-off-by: Mark Brown --- sound/soc/tegra/tegra210_sfc.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/tegra/tegra210_sfc.c b/sound/soc/tegra/tegra210_sfc.c index 7a2227ed3df6..368f077e7bee 100644 --- a/sound/soc/tegra/tegra210_sfc.c +++ b/sound/soc/tegra/tegra210_sfc.c @@ -3594,8 +3594,8 @@ static int tegra210_sfc_platform_remove(struct platform_device *pdev) static const struct dev_pm_ops tegra210_sfc_pm_ops = { SET_RUNTIME_PM_OPS(tegra210_sfc_runtime_suspend, tegra210_sfc_runtime_resume, NULL) - SET_LATE_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend, - pm_runtime_force_resume) + SET_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend, + pm_runtime_force_resume) }; static struct platform_driver tegra210_sfc_driver = { -- cgit v1.2.3 From c83d263a89f30d1c0274827c475f3583cf8e477f Mon Sep 17 00:00:00 2001 From: Sameer Pujar Date: Tue, 23 Nov 2021 19:37:36 +0530 Subject: ASoC: tegra: Use normal system sleep for MVC The driver currently subscribes for a late system sleep call. The initcall_debug log shows that suspend call for MVC device happens after the parent device (AHUB). This seems to cause suspend failure on Jetson TX2 platform. Also there is no use of having late system sleep specifically for MVC device. Fix the order by using normal system sleep. Fixes: e539891f9687 ("ASoC: tegra: Add Tegra210 based MVC driver") Cc: stable@vger.kernel.org Signed-off-by: Sameer Pujar Link: https://lore.kernel.org/r/1637676459-31191-4-git-send-email-spujar@nvidia.com Signed-off-by: Mark Brown --- sound/soc/tegra/tegra210_mvc.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/tegra/tegra210_mvc.c b/sound/soc/tegra/tegra210_mvc.c index 436bb151fed1..acf59328dcb6 100644 --- a/sound/soc/tegra/tegra210_mvc.c +++ b/sound/soc/tegra/tegra210_mvc.c @@ -639,8 +639,8 @@ static int tegra210_mvc_platform_remove(struct platform_device *pdev) static const struct dev_pm_ops tegra210_mvc_pm_ops = { SET_RUNTIME_PM_OPS(tegra210_mvc_runtime_suspend, tegra210_mvc_runtime_resume, NULL) - SET_LATE_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend, - pm_runtime_force_resume) + SET_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend, + pm_runtime_force_resume) }; static struct platform_driver tegra210_mvc_driver = { -- cgit v1.2.3 From b78400e41653b3a752a4cd17d2fcbd4a96bb4bc2 Mon Sep 17 00:00:00 2001 From: Sameer Pujar Date: Tue, 23 Nov 2021 19:37:37 +0530 Subject: ASoC: tegra: Use normal system sleep for Mixer The driver currently subscribes for a late system sleep call. The initcall_debug log shows that suspend call for Mixer device happens after the parent device (AHUB). This seems to cause suspend failure on Jetson TX2 platform. Also there is no use of having late system sleep specifically for Mixer device. Fix the order by using normal system sleep. Fixes: 05bb3d5ec64a ("ASoC: tegra: Add Tegra210 based Mixer driver") Cc: stable@vger.kernel.org Signed-off-by: Sameer Pujar Link: https://lore.kernel.org/r/1637676459-31191-5-git-send-email-spujar@nvidia.com Signed-off-by: Mark Brown --- sound/soc/tegra/tegra210_mixer.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/tegra/tegra210_mixer.c b/sound/soc/tegra/tegra210_mixer.c index 51d375573cfa..16e679a95658 100644 --- a/sound/soc/tegra/tegra210_mixer.c +++ b/sound/soc/tegra/tegra210_mixer.c @@ -666,8 +666,8 @@ static int tegra210_mixer_platform_remove(struct platform_device *pdev) static const struct dev_pm_ops tegra210_mixer_pm_ops = { SET_RUNTIME_PM_OPS(tegra210_mixer_runtime_suspend, tegra210_mixer_runtime_resume, NULL) - SET_LATE_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend, - pm_runtime_force_resume) + SET_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend, + pm_runtime_force_resume) }; static struct platform_driver tegra210_mixer_driver = { -- cgit v1.2.3 From 638c31d542a576714a52bb6a9a7dedff98e32a1d Mon Sep 17 00:00:00 2001 From: Sameer Pujar Date: Tue, 23 Nov 2021 19:37:38 +0530 Subject: ASoC: tegra: Use normal system sleep for AMX The driver currently subscribes for a late system sleep call. The initcall_debug log shows that suspend call for AMX device happens after the parent device (AHUB). This seems to cause suspend failure on Jetson TX2 platform. Also there is no use of having late system sleep specifically for AMX device. Fix the order by using normal system sleep. Fixes: 77f7df346c45 ("ASoC: tegra: Add Tegra210 based AMX driver") Cc: stable@vger.kernel.org Signed-off-by: Sameer Pujar Link: https://lore.kernel.org/r/1637676459-31191-6-git-send-email-spujar@nvidia.com Signed-off-by: Mark Brown --- sound/soc/tegra/tegra210_amx.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/tegra/tegra210_amx.c b/sound/soc/tegra/tegra210_amx.c index 689576302ede..d064cc67fea6 100644 --- a/sound/soc/tegra/tegra210_amx.c +++ b/sound/soc/tegra/tegra210_amx.c @@ -583,8 +583,8 @@ static int tegra210_amx_platform_remove(struct platform_device *pdev) static const struct dev_pm_ops tegra210_amx_pm_ops = { SET_RUNTIME_PM_OPS(tegra210_amx_runtime_suspend, tegra210_amx_runtime_resume, NULL) - SET_LATE_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend, - pm_runtime_force_resume) + SET_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend, + pm_runtime_force_resume) }; static struct platform_driver tegra210_amx_driver = { -- cgit v1.2.3 From cf36de4fc5ce5502ce5070a793addd9d49df4113 Mon Sep 17 00:00:00 2001 From: Sameer Pujar Date: Tue, 23 Nov 2021 19:37:39 +0530 Subject: ASoC: tegra: Use normal system sleep for ADX The driver currently subscribes for a late system sleep call. The initcall_debug log shows that suspend call for ADX device happens after the parent device (AHUB). This seems to cause suspend failure on Jetson TX2 platform. Also there is no use of having late system sleep specifically for ADX device. Fix the order by using normal system sleep. Fixes: a99ab6f395a9 ("ASoC: tegra: Add Tegra210 based ADX driver") Cc: stable@vger.kernel.org Signed-off-by: Sameer Pujar Link: https://lore.kernel.org/r/1637676459-31191-7-git-send-email-spujar@nvidia.com Signed-off-by: Mark Brown --- sound/soc/tegra/tegra210_adx.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/tegra/tegra210_adx.c b/sound/soc/tegra/tegra210_adx.c index 933c4503fe50..3785cade2d9a 100644 --- a/sound/soc/tegra/tegra210_adx.c +++ b/sound/soc/tegra/tegra210_adx.c @@ -514,8 +514,8 @@ static int tegra210_adx_platform_remove(struct platform_device *pdev) static const struct dev_pm_ops tegra210_adx_pm_ops = { SET_RUNTIME_PM_OPS(tegra210_adx_runtime_suspend, tegra210_adx_runtime_resume, NULL) - SET_LATE_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend, - pm_runtime_force_resume) + SET_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend, + pm_runtime_force_resume) }; static struct platform_driver tegra210_adx_driver = { -- cgit v1.2.3 From 4999d703c0e66f9f196b6edc0b8fdeca8846b8b6 Mon Sep 17 00:00:00 2001 From: Rob Clark Date: Wed, 17 Nov 2021 17:04:52 -0800 Subject: ASoC: rt5682: Fix crash due to out of scope stack vars Move the declaration of temporary arrays to somewhere that won't go out of scope before the devm_clk_hw_register() call, lest we be at the whim of the compiler for whether those stack variables get overwritten. Fixes a crash seen with gcc version 11.2.1 20210728 (Red Hat 11.2.1-1) Fixes: edbd24ea1e5c ("ASoC: rt5682: Drop usage of __clk_get_name()") Signed-off-by: Rob Clark Reviewed-by: Stephen Boyd Link: https://lore.kernel.org/r/20211118010453.843286-1-robdclark@gmail.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt5682.c | 10 ++++++---- 1 file changed, 6 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 04cb747c2b12..5224123d0d3b 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -2858,6 +2858,8 @@ int rt5682_register_dai_clks(struct rt5682_priv *rt5682) for (i = 0; i < RT5682_DAI_NUM_CLKS; ++i) { struct clk_init_data init = { }; + struct clk_parent_data parent_data; + const struct clk_hw *parent; dai_clk_hw = &rt5682->dai_clks_hw[i]; @@ -2865,17 +2867,17 @@ int rt5682_register_dai_clks(struct rt5682_priv *rt5682) case RT5682_DAI_WCLK_IDX: /* Make MCLK the parent of WCLK */ if (rt5682->mclk) { - init.parent_data = &(struct clk_parent_data){ + parent_data = (struct clk_parent_data){ .fw_name = "mclk", }; + init.parent_data = &parent_data; init.num_parents = 1; } break; case RT5682_DAI_BCLK_IDX: /* Make WCLK the parent of BCLK */ - init.parent_hws = &(const struct clk_hw *){ - &rt5682->dai_clks_hw[RT5682_DAI_WCLK_IDX] - }; + parent = &rt5682->dai_clks_hw[RT5682_DAI_WCLK_IDX]; + init.parent_hws = &parent; init.num_parents = 1; break; default: -- cgit v1.2.3 From 750dc2f622192c08664a15413bc9746d9cbc4361 Mon Sep 17 00:00:00 2001 From: Rob Clark Date: Wed, 17 Nov 2021 17:04:53 -0800 Subject: ASoC: rt5682s: Fix crash due to out of scope stack vars Move the declaration of temporary arrays to somewhere that won't go out of scope before the devm_clk_hw_register() call, lest we be at the whim of the compiler for whether those stack variables get overwritten. Fixes a crash seen with gcc version 11.2.1 20210728 (Red Hat 11.2.1-1) Fixes: bdd229ab26be ("ASoC: rt5682s: Add driver for ALC5682I-VS codec") Signed-off-by: Rob Clark Reviewed-by: Stephen Boyd Link: https://lore.kernel.org/r/20211118010453.843286-2-robdclark@gmail.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt5682s.c | 10 ++++++---- 1 file changed, 6 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt5682s.c b/sound/soc/codecs/rt5682s.c index 470957fcad6b..d49a4f68566d 100644 --- a/sound/soc/codecs/rt5682s.c +++ b/sound/soc/codecs/rt5682s.c @@ -2693,6 +2693,8 @@ static int rt5682s_register_dai_clks(struct snd_soc_component *component) for (i = 0; i < RT5682S_DAI_NUM_CLKS; ++i) { struct clk_init_data init = { }; + struct clk_parent_data parent_data; + const struct clk_hw *parent; dai_clk_hw = &rt5682s->dai_clks_hw[i]; @@ -2700,17 +2702,17 @@ static int rt5682s_register_dai_clks(struct snd_soc_component *component) case RT5682S_DAI_WCLK_IDX: /* Make MCLK the parent of WCLK */ if (rt5682s->mclk) { - init.parent_data = &(struct clk_parent_data){ + parent_data = (struct clk_parent_data){ .fw_name = "mclk", }; + init.parent_data = &parent_data; init.num_parents = 1; } break; case RT5682S_DAI_BCLK_IDX: /* Make WCLK the parent of BCLK */ - init.parent_hws = &(const struct clk_hw *){ - &rt5682s->dai_clks_hw[RT5682S_DAI_WCLK_IDX] - }; + parent = &rt5682s->dai_clks_hw[RT5682S_DAI_WCLK_IDX]; + init.parent_hws = &parent; init.num_parents = 1; break; default: -- cgit v1.2.3 From 65cc4ad62a9ed47c0b4fcd7af667d97d7c29f19d Mon Sep 17 00:00:00 2001 From: Stefan Binding Date: Sun, 28 Nov 2021 11:55:58 +0000 Subject: ALSA: hda/cs8409: Set PMSG_ON earlier inside cs8409 driver For cs8409, it is required to run Jack Detect on resume. Jack Detect on cs8409+cs42l42 requires an interrupt from cs42l42 to be sent to cs8409 which is propogated to the driver via an unsolicited event. However, the hda_codec drops unsolicited events if the power_state is not set to PMSG_ON. Which is set at the end of the resume call. This means there is a race condition between setting power_state to PMSG_ON and receiving the interrupt. To solve this, we can add an API to set the power_state earlier and call that before we start Jack Detect. This does not cause issues, since we know inside our driver that we are already initialized, and ready to handle the unsolicited events. Signed-off-by: Stefan Binding Signed-off-by: Vitaly Rodionov Cc: # v5.15+ Link: https://lore.kernel.org/r/20211128115558.71683-1-vitalyr@opensource.cirrus.com Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_local.h | 9 +++++++++ sound/pci/hda/patch_cs8409.c | 5 +++++ 2 files changed, 14 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index ea8ab8b43337..d22c96eb2f8f 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -438,6 +438,15 @@ int snd_hda_codec_set_pin_target(struct hda_codec *codec, hda_nid_t nid, #define for_each_hda_codec_node(nid, codec) \ for ((nid) = (codec)->core.start_nid; (nid) < (codec)->core.end_nid; (nid)++) +/* Set the codec power_state flag to indicate to allow unsol event handling; + * see hda_codec_unsol_event() in hda_bind.c. Calling this might confuse the + * state tracking, so use with care. + */ +static inline void snd_hda_codec_allow_unsol_events(struct hda_codec *codec) +{ + codec->core.dev.power.power_state = PMSG_ON; +} + /* * get widget capabilities */ diff --git a/sound/pci/hda/patch_cs8409.c b/sound/pci/hda/patch_cs8409.c index 31ff11ab868e..039b9f2f8e94 100644 --- a/sound/pci/hda/patch_cs8409.c +++ b/sound/pci/hda/patch_cs8409.c @@ -750,6 +750,11 @@ static void cs42l42_resume(struct sub_codec *cs42l42) if (cs42l42->full_scale_vol) cs8409_i2c_write(cs42l42, 0x2001, 0x01); + /* we have to explicitly allow unsol event handling even during the + * resume phase so that the jack event is processed properly + */ + snd_hda_codec_allow_unsol_events(cs42l42->codec); + cs42l42_enable_jack_detect(cs42l42); } -- cgit v1.2.3 From 53689f7f91a2ab0079422e1d1b6e096cf68d58f4 Mon Sep 17 00:00:00 2001 From: Nicolas Frattaroli Date: Thu, 25 Nov 2021 09:48:59 +0100 Subject: ASoC: rockchip: i2s_tdm: Dup static DAI template Previously, the DAI template was used directly, which lead to fun bugs such as "why is my channels_max changing?" when one instantiated more than one i2s_tdm IP block in a device tree. This change makes it so that we instead duplicate the template struct, and then use that. Fixes: 081068fd6414 ("ASoC: rockchip: add support for i2s-tdm controller") Signed-off-by: Nicolas Frattaroli Link: https://lore.kernel.org/r/20211125084900.417102-1-frattaroli.nicolas@gmail.com Signed-off-by: Mark Brown --- sound/soc/rockchip/rockchip_i2s_tdm.c | 52 +++++++++++++++++++++-------------- 1 file changed, 31 insertions(+), 21 deletions(-) (limited to 'sound') diff --git a/sound/soc/rockchip/rockchip_i2s_tdm.c b/sound/soc/rockchip/rockchip_i2s_tdm.c index 17b9b287853a..5f9cb5c4c7f0 100644 --- a/sound/soc/rockchip/rockchip_i2s_tdm.c +++ b/sound/soc/rockchip/rockchip_i2s_tdm.c @@ -95,6 +95,7 @@ struct rk_i2s_tdm_dev { spinlock_t lock; /* xfer lock */ bool has_playback; bool has_capture; + struct snd_soc_dai_driver *dai; }; static int to_ch_num(unsigned int val) @@ -1310,19 +1311,14 @@ static const struct of_device_id rockchip_i2s_tdm_match[] = { {}, }; -static struct snd_soc_dai_driver i2s_tdm_dai = { +static const struct snd_soc_dai_driver i2s_tdm_dai = { .probe = rockchip_i2s_tdm_dai_probe, - .playback = { - .stream_name = "Playback", - }, - .capture = { - .stream_name = "Capture", - }, .ops = &rockchip_i2s_tdm_dai_ops, }; -static void rockchip_i2s_tdm_init_dai(struct rk_i2s_tdm_dev *i2s_tdm) +static int rockchip_i2s_tdm_init_dai(struct rk_i2s_tdm_dev *i2s_tdm) { + struct snd_soc_dai_driver *dai; struct property *dma_names; const char *dma_name; u64 formats = (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE | @@ -1337,19 +1333,33 @@ static void rockchip_i2s_tdm_init_dai(struct rk_i2s_tdm_dev *i2s_tdm) i2s_tdm->has_capture = true; } + dai = devm_kmemdup(i2s_tdm->dev, &i2s_tdm_dai, + sizeof(*dai), GFP_KERNEL); + if (!dai) + return -ENOMEM; + if (i2s_tdm->has_playback) { - i2s_tdm_dai.playback.channels_min = 2; - i2s_tdm_dai.playback.channels_max = 8; - i2s_tdm_dai.playback.rates = SNDRV_PCM_RATE_8000_192000; - i2s_tdm_dai.playback.formats = formats; + dai->playback.stream_name = "Playback"; + dai->playback.channels_min = 2; + dai->playback.channels_max = 8; + dai->playback.rates = SNDRV_PCM_RATE_8000_192000; + dai->playback.formats = formats; } if (i2s_tdm->has_capture) { - i2s_tdm_dai.capture.channels_min = 2; - i2s_tdm_dai.capture.channels_max = 8; - i2s_tdm_dai.capture.rates = SNDRV_PCM_RATE_8000_192000; - i2s_tdm_dai.capture.formats = formats; + dai->capture.stream_name = "Capture"; + dai->capture.channels_min = 2; + dai->capture.channels_max = 8; + dai->capture.rates = SNDRV_PCM_RATE_8000_192000; + dai->capture.formats = formats; } + + if (i2s_tdm->clk_trcm != TRCM_TXRX) + dai->symmetric_rate = 1; + + i2s_tdm->dai = dai; + + return 0; } static int rockchip_i2s_tdm_path_check(struct rk_i2s_tdm_dev *i2s_tdm, @@ -1541,8 +1551,6 @@ static int rockchip_i2s_tdm_probe(struct platform_device *pdev) spin_lock_init(&i2s_tdm->lock); i2s_tdm->soc_data = (struct rk_i2s_soc_data *)of_id->data; - rockchip_i2s_tdm_init_dai(i2s_tdm); - i2s_tdm->frame_width = 64; i2s_tdm->clk_trcm = TRCM_TXRX; @@ -1555,8 +1563,10 @@ static int rockchip_i2s_tdm_probe(struct platform_device *pdev) } i2s_tdm->clk_trcm = TRCM_RX; } - if (i2s_tdm->clk_trcm != TRCM_TXRX) - i2s_tdm_dai.symmetric_rate = 1; + + ret = rockchip_i2s_tdm_init_dai(i2s_tdm); + if (ret) + return ret; i2s_tdm->grf = syscon_regmap_lookup_by_phandle(node, "rockchip,grf"); if (IS_ERR(i2s_tdm->grf)) @@ -1678,7 +1688,7 @@ static int rockchip_i2s_tdm_probe(struct platform_device *pdev) ret = devm_snd_soc_register_component(&pdev->dev, &rockchip_i2s_tdm_component, - &i2s_tdm_dai, 1); + i2s_tdm->dai, 1); if (ret) { dev_err(&pdev->dev, "Could not register DAI\n"); -- cgit v1.2.3 From d5c137f41352e8dd864522c417b45d8d1aebca68 Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Tue, 30 Nov 2021 15:56:33 +0300 Subject: ASoC: amd: fix uninitialized variable in snd_acp6x_probe() The "index" is potentially used without being initialized on the error path. Fixes: fc329c1de498 ("ASoC: amd: add platform devices for acp6x pdm driver and dmic driver") Signed-off-by: Dan Carpenter Link: https://lore.kernel.org/r/20211130125633.GA24941@kili Signed-off-by: Mark Brown --- sound/soc/amd/yc/pci-acp6x.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/amd/yc/pci-acp6x.c b/sound/soc/amd/yc/pci-acp6x.c index 957eeb6fb8e3..7e9a9a9d8ddd 100644 --- a/sound/soc/amd/yc/pci-acp6x.c +++ b/sound/soc/amd/yc/pci-acp6x.c @@ -146,10 +146,11 @@ static int snd_acp6x_probe(struct pci_dev *pci, { struct acp6x_dev_data *adata; struct platform_device_info pdevinfo[ACP6x_DEVS]; - int ret, index; + int index = 0; int val = 0x00; u32 addr; unsigned int irqflags; + int ret; irqflags = IRQF_SHARED; /* Yellow Carp device check */ -- cgit v1.2.3 From 046aede2f847676f93a2ea4f48b77909c51dba40 Mon Sep 17 00:00:00 2001 From: Hui Wang Date: Tue, 30 Nov 2021 11:06:06 +0200 Subject: ASoC: SOF: Intel: Retry codec probing if it fails On the latest Lenovo Thinkstation laptops, we often experience the speaker failure after rebooting, check the dmesg, we could see: sof-audio-pci-intel-tgl 0000:00:1f.3: codec #0 probe error, ret: -5 The analogue codec on the machine is ALC287, then we designed a testcase to reboot and check the codec probing result repeatedly, we found the analogue codec probing always failed at least once within several minutes to several hours (roughly 1 reboot per min). This issue happens on all laptops of this Thinkstation model, but with legacy HDA driver, we couldn't reproduce this issue on those laptops. And so far, this issue is not reproduced on machines which don't belong to this model. We tried to make the hda_dsp_ctrl_init_chip() same as hda_intel_init_chip() which is the controller init routine in the legacy HDA driver, but it didn't help. We found when issue happens, the resp is -1, and if we let driver re-run send_cmd() and get_response(), it will get the correct response 10ec0287, then driver continues the rest work, finally boot to the desktop and all audio function work well. Here adding codec probing retries to 3 times, it could fix the issue on this Thinkstation model, and it doesn't bring impact to other machines. Reviewed-by: Bard Liao Reviewed-by: Pierre-Louis Bossart Signed-off-by: Hui Wang Signed-off-by: Kai Vehmanen Link: https://lore.kernel.org/r/20211130090606.529348-1-kai.vehmanen@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-codec.c | 14 +++++++++----- 1 file changed, 9 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/intel/hda-codec.c b/sound/soc/sof/intel/hda-codec.c index 6744318de612..13cd96e6724a 100644 --- a/sound/soc/sof/intel/hda-codec.c +++ b/sound/soc/sof/intel/hda-codec.c @@ -22,6 +22,7 @@ #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_AUDIO_CODEC) #define IDISP_VID_INTEL 0x80860000 +#define CODEC_PROBE_RETRIES 3 /* load the legacy HDA codec driver */ static int request_codec_module(struct hda_codec *codec) @@ -121,12 +122,15 @@ static int hda_codec_probe(struct snd_sof_dev *sdev, int address, u32 hda_cmd = (address << 28) | (AC_NODE_ROOT << 20) | (AC_VERB_PARAMETERS << 8) | AC_PAR_VENDOR_ID; u32 resp = -1; - int ret; + int ret, retry = 0; + + do { + mutex_lock(&hbus->core.cmd_mutex); + snd_hdac_bus_send_cmd(&hbus->core, hda_cmd); + snd_hdac_bus_get_response(&hbus->core, address, &resp); + mutex_unlock(&hbus->core.cmd_mutex); + } while (resp == -1 && retry++ < CODEC_PROBE_RETRIES); - mutex_lock(&hbus->core.cmd_mutex); - snd_hdac_bus_send_cmd(&hbus->core, hda_cmd); - snd_hdac_bus_get_response(&hbus->core, address, &resp); - mutex_unlock(&hbus->core.cmd_mutex); if (resp == -1) return -EIO; dev_dbg(sdev->dev, "HDA codec #%d probed OK: response: %x\n", -- cgit v1.2.3 From d85ffff5302b1509efc482e8877c253b0a668b33 Mon Sep 17 00:00:00 2001 From: Kai Vehmanen Date: Tue, 30 Nov 2021 14:47:31 +0200 Subject: ALSA: hda: Add Intel DG2 PCI ID and HDMI codec vid Add HD Audio PCI ID and HDMI codec vendor ID for Intel DG2. Reviewed-by: Uma Shankar Signed-off-by: Kai Vehmanen Link: https://lore.kernel.org/r/20211130124732.696896-1-kai.vehmanen@linux.intel.com Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 12 +++++++++++- sound/pci/hda/patch_hdmi.c | 1 + 2 files changed, 12 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index fe51163f2d82..1b46b599a5cf 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -335,7 +335,10 @@ enum { ((pci)->device == 0x0c0c) || \ ((pci)->device == 0x0d0c) || \ ((pci)->device == 0x160c) || \ - ((pci)->device == 0x490d)) + ((pci)->device == 0x490d) || \ + ((pci)->device == 0x4f90) || \ + ((pci)->device == 0x4f91) || \ + ((pci)->device == 0x4f92)) #define IS_BXT(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0x5a98) @@ -2473,6 +2476,13 @@ static const struct pci_device_id azx_ids[] = { /* DG1 */ { PCI_DEVICE(0x8086, 0x490d), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, + /* DG2 */ + { PCI_DEVICE(0x8086, 0x4f90), + .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, + { PCI_DEVICE(0x8086, 0x4f91), + .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, + { PCI_DEVICE(0x8086, 0x4f92), + .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, /* Alderlake-S */ { PCI_DEVICE(0x8086, 0x7ad0), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 65d2c5539919..98633d2684de 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -4382,6 +4382,7 @@ HDA_CODEC_ENTRY(0x80862814, "DG1 HDMI", patch_i915_tgl_hdmi), HDA_CODEC_ENTRY(0x80862815, "Alderlake HDMI", patch_i915_tgl_hdmi), HDA_CODEC_ENTRY(0x8086281c, "Alderlake-P HDMI", patch_i915_tgl_hdmi), HDA_CODEC_ENTRY(0x80862816, "Rocketlake HDMI", patch_i915_tgl_hdmi), +HDA_CODEC_ENTRY(0x80862819, "DG2 HDMI", patch_i915_tgl_hdmi), HDA_CODEC_ENTRY(0x8086281a, "Jasperlake HDMI", patch_i915_icl_hdmi), HDA_CODEC_ENTRY(0x8086281b, "Elkhartlake HDMI", patch_i915_icl_hdmi), HDA_CODEC_ENTRY(0x80862880, "CedarTrail HDMI", patch_generic_hdmi), -- cgit v1.2.3 From 289047db1143c42c81820352f195a393ff639a52 Mon Sep 17 00:00:00 2001 From: Kai Vehmanen Date: Tue, 30 Nov 2021 14:47:32 +0200 Subject: ALSA: hda/hdmi: fix HDA codec entry table order for ADL-P Keep the HDA_CODEC_ENTRY entries sorted by the codec VID. ADL-P is the only misplaced Intel HDMI codec. Signed-off-by: Kai Vehmanen Link: https://lore.kernel.org/r/20211130124732.696896-2-kai.vehmanen@linux.intel.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 98633d2684de..415701bd10ac 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -4380,11 +4380,11 @@ HDA_CODEC_ENTRY(0x8086280f, "Icelake HDMI", patch_i915_icl_hdmi), HDA_CODEC_ENTRY(0x80862812, "Tigerlake HDMI", patch_i915_tgl_hdmi), HDA_CODEC_ENTRY(0x80862814, "DG1 HDMI", patch_i915_tgl_hdmi), HDA_CODEC_ENTRY(0x80862815, "Alderlake HDMI", patch_i915_tgl_hdmi), -HDA_CODEC_ENTRY(0x8086281c, "Alderlake-P HDMI", patch_i915_tgl_hdmi), HDA_CODEC_ENTRY(0x80862816, "Rocketlake HDMI", patch_i915_tgl_hdmi), HDA_CODEC_ENTRY(0x80862819, "DG2 HDMI", patch_i915_tgl_hdmi), HDA_CODEC_ENTRY(0x8086281a, "Jasperlake HDMI", patch_i915_icl_hdmi), HDA_CODEC_ENTRY(0x8086281b, "Elkhartlake HDMI", patch_i915_icl_hdmi), +HDA_CODEC_ENTRY(0x8086281c, "Alderlake-P HDMI", patch_i915_tgl_hdmi), HDA_CODEC_ENTRY(0x80862880, "CedarTrail HDMI", patch_generic_hdmi), HDA_CODEC_ENTRY(0x80862882, "Valleyview2 HDMI", patch_i915_byt_hdmi), HDA_CODEC_ENTRY(0x80862883, "Braswell HDMI", patch_i915_byt_hdmi), -- cgit v1.2.3 From 4739d88ad8e1900f809f8a5c98f3c1b65bf76220 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Tue, 30 Nov 2021 16:31:10 +0000 Subject: ASoC: qdsp6: q6routing: Fix return value from msm_routing_put_audio_mixer msm_routing_put_audio_mixer() can return incorrect value in various scenarios. scenario 1: amixer cset iface=MIXER,name='SLIMBUS_0_RX Audio Mixer MultiMedia1' 1 amixer cset iface=MIXER,name='SLIMBUS_0_RX Audio Mixer MultiMedia1' 0 return value is 0 instead of 1 eventhough value was changed scenario 2: amixer cset iface=MIXER,name='SLIMBUS_0_RX Audio Mixer MultiMedia1' 1 amixer cset iface=MIXER,name='SLIMBUS_0_RX Audio Mixer MultiMedia1' 1 return value is 1 instead of 0 eventhough the value was not changed scenario 3: amixer cset iface=MIXER,name='SLIMBUS_0_RX Audio Mixer MultiMedia1' 0 return value is 1 instead of 0 eventhough the value was not changed Fix this by adding checks, so that change notifications are sent correctly. Fixes: e3a33673e845 ("ASoC: qdsp6: q6routing: Add q6routing driver") Signed-off-by: Srinivas Kandagatla Link: https://lore.kernel.org/r/20211130163110.5628-1-srinivas.kandagatla@linaro.org Signed-off-by: Mark Brown --- sound/soc/qcom/qdsp6/q6routing.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/qcom/qdsp6/q6routing.c b/sound/soc/qcom/qdsp6/q6routing.c index cd74681e811e..928fd23e2c27 100644 --- a/sound/soc/qcom/qdsp6/q6routing.c +++ b/sound/soc/qcom/qdsp6/q6routing.c @@ -498,14 +498,16 @@ static int msm_routing_put_audio_mixer(struct snd_kcontrol *kcontrol, struct session_data *session = &data->sessions[session_id]; if (ucontrol->value.integer.value[0]) { + if (session->port_id == be_id) + return 0; + session->port_id = be_id; snd_soc_dapm_mixer_update_power(dapm, kcontrol, 1, update); } else { - if (session->port_id == be_id) { - session->port_id = -1; + if (session->port_id == -1 || session->port_id != be_id) return 0; - } + session->port_id = -1; snd_soc_dapm_mixer_update_power(dapm, kcontrol, 0, update); } -- cgit v1.2.3 From 23ba28616d3063bd4c4953598ed5e439ca891101 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Tue, 30 Nov 2021 16:05:04 +0000 Subject: ASoC: codecs: wcd934x: handle channel mappping list correctly Currently each channel is added as list to dai channel list, however there is danger of adding same channel to multiple dai channel list which endups corrupting the other list where its already added. This patch ensures that the channel is actually free before adding to the dai channel list and also ensures that the channel is on the list before deleting it. This check was missing previously, and we did not hit this issue as we were testing very simple usecases with sequence of amixer commands. Fixes: a70d9245759a ("ASoC: wcd934x: add capture dapm widgets") Fixes: dd9eb19b5673 ("ASoC: wcd934x: add playback dapm widgets") Signed-off-by: Srinivas Kandagatla Link: https://lore.kernel.org/r/20211130160507.22180-2-srinivas.kandagatla@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/wcd934x.c | 119 +++++++++++++++++++++++++++++++++------------ 1 file changed, 88 insertions(+), 31 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wcd934x.c b/sound/soc/codecs/wcd934x.c index 4f568abd59e2..eb4e2f2a24ae 100644 --- a/sound/soc/codecs/wcd934x.c +++ b/sound/soc/codecs/wcd934x.c @@ -3326,6 +3326,31 @@ static int slim_rx_mux_get(struct snd_kcontrol *kc, return 0; } +static int slim_rx_mux_to_dai_id(int mux) +{ + int aif_id; + + switch (mux) { + case 1: + aif_id = AIF1_PB; + break; + case 2: + aif_id = AIF2_PB; + break; + case 3: + aif_id = AIF3_PB; + break; + case 4: + aif_id = AIF4_PB; + break; + default: + aif_id = -1; + break; + } + + return aif_id; +} + static int slim_rx_mux_put(struct snd_kcontrol *kc, struct snd_ctl_elem_value *ucontrol) { @@ -3333,43 +3358,59 @@ static int slim_rx_mux_put(struct snd_kcontrol *kc, struct wcd934x_codec *wcd = dev_get_drvdata(w->dapm->dev); struct soc_enum *e = (struct soc_enum *)kc->private_value; struct snd_soc_dapm_update *update = NULL; + struct wcd934x_slim_ch *ch, *c; u32 port_id = w->shift; + bool found = false; + int mux_idx; + int prev_mux_idx = wcd->rx_port_value[port_id]; + int aif_id; - if (wcd->rx_port_value[port_id] == ucontrol->value.enumerated.item[0]) - return 0; + mux_idx = ucontrol->value.enumerated.item[0]; - wcd->rx_port_value[port_id] = ucontrol->value.enumerated.item[0]; + if (mux_idx == prev_mux_idx) + return 0; - switch (wcd->rx_port_value[port_id]) { + switch(mux_idx) { case 0: - list_del_init(&wcd->rx_chs[port_id].list); - break; - case 1: - list_add_tail(&wcd->rx_chs[port_id].list, - &wcd->dai[AIF1_PB].slim_ch_list); - break; - case 2: - list_add_tail(&wcd->rx_chs[port_id].list, - &wcd->dai[AIF2_PB].slim_ch_list); - break; - case 3: - list_add_tail(&wcd->rx_chs[port_id].list, - &wcd->dai[AIF3_PB].slim_ch_list); + aif_id = slim_rx_mux_to_dai_id(prev_mux_idx); + if (aif_id < 0) + return 0; + + list_for_each_entry_safe(ch, c, &wcd->dai[aif_id].slim_ch_list, list) { + if (ch->port == port_id + WCD934X_RX_START) { + found = true; + list_del_init(&ch->list); + break; + } + } + if (!found) + return 0; + break; - case 4: - list_add_tail(&wcd->rx_chs[port_id].list, - &wcd->dai[AIF4_PB].slim_ch_list); + case 1 ... 4: + aif_id = slim_rx_mux_to_dai_id(mux_idx); + if (aif_id < 0) + return 0; + + if (list_empty(&wcd->rx_chs[port_id].list)) { + list_add_tail(&wcd->rx_chs[port_id].list, + &wcd->dai[aif_id].slim_ch_list); + } else { + dev_err(wcd->dev ,"SLIM_RX%d PORT is busy\n", port_id); + return 0; + } break; + default: - dev_err(wcd->dev, "Unknown AIF %d\n", - wcd->rx_port_value[port_id]); + dev_err(wcd->dev, "Unknown AIF %d\n", mux_idx); goto err; } + wcd->rx_port_value[port_id] = mux_idx; snd_soc_dapm_mux_update_power(w->dapm, kc, wcd->rx_port_value[port_id], e, update); - return 0; + return 1; err: return -EINVAL; } @@ -3815,6 +3856,7 @@ static int slim_tx_mixer_put(struct snd_kcontrol *kc, struct soc_mixer_control *mixer = (struct soc_mixer_control *)kc->private_value; int enable = ucontrol->value.integer.value[0]; + struct wcd934x_slim_ch *ch, *c; int dai_id = widget->shift; int port_id = mixer->shift; @@ -3822,17 +3864,32 @@ static int slim_tx_mixer_put(struct snd_kcontrol *kc, if (enable == wcd->tx_port_value[port_id]) return 0; - wcd->tx_port_value[port_id] = enable; - - if (enable) - list_add_tail(&wcd->tx_chs[port_id].list, - &wcd->dai[dai_id].slim_ch_list); - else - list_del_init(&wcd->tx_chs[port_id].list); + if (enable) { + if (list_empty(&wcd->tx_chs[port_id].list)) { + list_add_tail(&wcd->tx_chs[port_id].list, + &wcd->dai[dai_id].slim_ch_list); + } else { + dev_err(wcd->dev ,"SLIM_TX%d PORT is busy\n", port_id); + return 0; + } + } else { + bool found = false; + + list_for_each_entry_safe(ch, c, &wcd->dai[dai_id].slim_ch_list, list) { + if (ch->port == port_id) { + found = true; + list_del_init(&wcd->tx_chs[port_id].list); + break; + } + } + if (!found) + return 0; + } + wcd->tx_port_value[port_id] = enable; snd_soc_dapm_mixer_update_power(widget->dapm, kc, enable, update); - return 0; + return 1; } static const struct snd_kcontrol_new aif1_slim_cap_mixer[] = { -- cgit v1.2.3 From d9be0ff4796d1b6f5ee391c1b7e3653a43cedfab Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Tue, 30 Nov 2021 16:05:06 +0000 Subject: ASoC: codecs: wcd934x: return correct value from mixer put wcd934x_compander_set() currently returns zero eventhough it changes the value. Fix this, so that change notifications are sent correctly. Fixes: 1cde8b822332 ("ASoC: wcd934x: add basic controls") Signed-off-by: Srinivas Kandagatla Link: https://lore.kernel.org/r/20211130160507.22180-4-srinivas.kandagatla@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/wcd934x.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wcd934x.c b/sound/soc/codecs/wcd934x.c index eb4e2f2a24ae..e63c6b723d76 100644 --- a/sound/soc/codecs/wcd934x.c +++ b/sound/soc/codecs/wcd934x.c @@ -3256,6 +3256,9 @@ static int wcd934x_compander_set(struct snd_kcontrol *kc, int value = ucontrol->value.integer.value[0]; int sel; + if (wcd->comp_enabled[comp] == value) + return 0; + wcd->comp_enabled[comp] = value; sel = value ? WCD934X_HPH_GAIN_SRC_SEL_COMPANDER : WCD934X_HPH_GAIN_SRC_SEL_REGISTER; @@ -3279,10 +3282,10 @@ static int wcd934x_compander_set(struct snd_kcontrol *kc, case COMPANDER_8: break; default: - break; + return 0; } - return 0; + return 1; } static int wcd934x_rx_hph_mode_get(struct snd_kcontrol *kc, -- cgit v1.2.3 From 3fc27e9a1f619b50700f020e6cd270c1b74755f0 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Tue, 30 Nov 2021 16:05:07 +0000 Subject: ASoC: codecs: wsa881x: fix return values from kcontrol put wsa881x_set_port() and wsa881x_put_pa_gain() currently returns zero eventhough it changes the value. Fix this, so that change notifications are sent correctly. Fixes: a0aab9e1404a ("ASoC: codecs: add wsa881x amplifier support") Signed-off-by: Srinivas Kandagatla Link: https://lore.kernel.org/r/20211130160507.22180-5-srinivas.kandagatla@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/wsa881x.c | 16 ++++++++++++---- 1 file changed, 12 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wsa881x.c b/sound/soc/codecs/wsa881x.c index 2da4a5fa7a18..564b78f3cdd0 100644 --- a/sound/soc/codecs/wsa881x.c +++ b/sound/soc/codecs/wsa881x.c @@ -772,7 +772,8 @@ static int wsa881x_put_pa_gain(struct snd_kcontrol *kc, usleep_range(1000, 1010); } - return 0; + + return 1; } static int wsa881x_get_port(struct snd_kcontrol *kcontrol, @@ -816,15 +817,22 @@ static int wsa881x_set_port(struct snd_kcontrol *kcontrol, (struct soc_mixer_control *)kcontrol->private_value; int portidx = mixer->reg; - if (ucontrol->value.integer.value[0]) + if (ucontrol->value.integer.value[0]) { + if (data->port_enable[portidx]) + return 0; + data->port_enable[portidx] = true; - else + } else { + if (!data->port_enable[portidx]) + return 0; + data->port_enable[portidx] = false; + } if (portidx == WSA881X_PORT_BOOST) /* Boost Switch */ wsa881x_boost_ctrl(comp, data->port_enable[portidx]); - return 0; + return 1; } static const char * const smart_boost_lvl_text[] = { -- cgit v1.2.3 From 9d2479c960875ca1239bcb899f386970c13d9cfe Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 1 Dec 2021 08:36:04 +0100 Subject: ALSA: pcm: oss: Fix negative period/buffer sizes The period size calculation in OSS layer may receive a negative value as an error, but the code there assumes only the positive values and handle them with size_t. Due to that, a too big value may be passed to the lower layers. This patch changes the code to handle with ssize_t and adds the proper error checks appropriately. Reported-by: syzbot+bb348e9f9a954d42746f@syzkaller.appspotmail.com Reported-by: Bixuan Cui Cc: Link: https://lore.kernel.org/r/1638270978-42412-1-git-send-email-cuibixuan@linux.alibaba.com Link: https://lore.kernel.org/r/20211201073606.11660-2-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/oss/pcm_oss.c | 24 +++++++++++++++--------- 1 file changed, 15 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index 82a818734a5f..bec7590bc84b 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -147,7 +147,7 @@ snd_pcm_hw_param_value_min(const struct snd_pcm_hw_params *params, * * Return the maximum value for field PAR. */ -static unsigned int +static int snd_pcm_hw_param_value_max(const struct snd_pcm_hw_params *params, snd_pcm_hw_param_t var, int *dir) { @@ -682,18 +682,24 @@ static int snd_pcm_oss_period_size(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *oss_params, struct snd_pcm_hw_params *slave_params) { - size_t s; - size_t oss_buffer_size, oss_period_size, oss_periods; - size_t min_period_size, max_period_size; + ssize_t s; + ssize_t oss_buffer_size; + ssize_t oss_period_size, oss_periods; + ssize_t min_period_size, max_period_size; struct snd_pcm_runtime *runtime = substream->runtime; size_t oss_frame_size; oss_frame_size = snd_pcm_format_physical_width(params_format(oss_params)) * params_channels(oss_params) / 8; + oss_buffer_size = snd_pcm_hw_param_value_max(slave_params, + SNDRV_PCM_HW_PARAM_BUFFER_SIZE, + NULL); + if (oss_buffer_size <= 0) + return -EINVAL; oss_buffer_size = snd_pcm_plug_client_size(substream, - snd_pcm_hw_param_value_max(slave_params, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, NULL)) * oss_frame_size; - if (!oss_buffer_size) + oss_buffer_size * oss_frame_size); + if (oss_buffer_size <= 0) return -EINVAL; oss_buffer_size = rounddown_pow_of_two(oss_buffer_size); if (atomic_read(&substream->mmap_count)) { @@ -730,7 +736,7 @@ static int snd_pcm_oss_period_size(struct snd_pcm_substream *substream, min_period_size = snd_pcm_plug_client_size(substream, snd_pcm_hw_param_value_min(slave_params, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, NULL)); - if (min_period_size) { + if (min_period_size > 0) { min_period_size *= oss_frame_size; min_period_size = roundup_pow_of_two(min_period_size); if (oss_period_size < min_period_size) @@ -739,7 +745,7 @@ static int snd_pcm_oss_period_size(struct snd_pcm_substream *substream, max_period_size = snd_pcm_plug_client_size(substream, snd_pcm_hw_param_value_max(slave_params, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, NULL)); - if (max_period_size) { + if (max_period_size > 0) { max_period_size *= oss_frame_size; max_period_size = rounddown_pow_of_two(max_period_size); if (oss_period_size > max_period_size) @@ -752,7 +758,7 @@ static int snd_pcm_oss_period_size(struct snd_pcm_substream *substream, oss_periods = substream->oss.setup.periods; s = snd_pcm_hw_param_value_max(slave_params, SNDRV_PCM_HW_PARAM_PERIODS, NULL); - if (runtime->oss.maxfrags && s > runtime->oss.maxfrags) + if (s > 0 && runtime->oss.maxfrags && s > runtime->oss.maxfrags) s = runtime->oss.maxfrags; if (oss_periods > s) oss_periods = s; -- cgit v1.2.3 From 8839c8c0f77ab8fc0463f4ab8b37fca3f70677c2 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 1 Dec 2021 08:36:05 +0100 Subject: ALSA: pcm: oss: Limit the period size to 16MB Set the practical limit to the period size (the fragment shift in OSS) instead of a full 31bit; a too large value could lead to the exhaust of memory as we allocate temporary buffers of the period size, too. As of this patch, we set to 16MB limit, which should cover all use cases. Reported-by: syzbot+bb348e9f9a954d42746f@syzkaller.appspotmail.com Reported-by: Bixuan Cui Cc: Link: https://lore.kernel.org/r/1638270978-42412-1-git-send-email-cuibixuan@linux.alibaba.com Link: https://lore.kernel.org/r/20211201073606.11660-3-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/oss/pcm_oss.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index bec7590bc84b..89c4910daf02 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -1962,7 +1962,7 @@ static int snd_pcm_oss_set_fragment1(struct snd_pcm_substream *substream, unsign if (runtime->oss.subdivision || runtime->oss.fragshift) return -EINVAL; fragshift = val & 0xffff; - if (fragshift >= 31) + if (fragshift >= 25) /* should be large enough */ return -EINVAL; runtime->oss.fragshift = fragshift; runtime->oss.maxfrags = (val >> 16) & 0xffff; -- cgit v1.2.3 From 6665bb30a6b1a4a853d52557c05482ee50e71391 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 1 Dec 2021 08:36:06 +0100 Subject: ALSA: pcm: oss: Handle missing errors in snd_pcm_oss_change_params*() A couple of calls in snd_pcm_oss_change_params_locked() ignore the possible errors. Catch those errors and abort the operation for avoiding further problems. Cc: Link: https://lore.kernel.org/r/20211201073606.11660-4-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/oss/pcm_oss.c | 11 +++++++++-- 1 file changed, 9 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index 89c4910daf02..20a0a4771b9a 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -884,8 +884,15 @@ static int snd_pcm_oss_change_params_locked(struct snd_pcm_substream *substream) err = -EINVAL; goto failure; } - choose_rate(substream, sparams, runtime->oss.rate); - snd_pcm_hw_param_near(substream, sparams, SNDRV_PCM_HW_PARAM_CHANNELS, runtime->oss.channels, NULL); + + err = choose_rate(substream, sparams, runtime->oss.rate); + if (err < 0) + goto failure; + err = snd_pcm_hw_param_near(substream, sparams, + SNDRV_PCM_HW_PARAM_CHANNELS, + runtime->oss.channels, NULL); + if (err < 0) + goto failure; format = snd_pcm_oss_format_from(runtime->oss.format); -- cgit v1.2.3 From b6409dd6bdc03aa178bbff0d80db2a30d29b63ac Mon Sep 17 00:00:00 2001 From: Alan Young Date: Thu, 2 Dec 2021 15:06:07 +0000 Subject: ALSA: ctl: Fix copy of updated id with element read/write When control_compat.c:copy_ctl_value_to_user() is used, by ctl_elem_read_user() & ctl_elem_write_user(), it must also copy back the snd_ctl_elem_id value that may have been updated (filled in) by the call to snd_ctl_elem_read/snd_ctl_elem_write(). This matches the functionality provided by snd_ctl_elem_read_user() and snd_ctl_elem_write_user(), via snd_ctl_build_ioff(). Without this, and without making additional calls to snd_ctl_info() which are unnecessary when using the non-compat calls, a userspace application will not know the numid value for the element and consequently will not be able to use the poll/read interface on the control file to determine which elements have updates. Signed-off-by: Alan Young Cc: Link: https://lore.kernel.org/r/20211202150607.543389-1-consult.awy@gmail.com Signed-off-by: Takashi Iwai --- sound/core/control_compat.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/core/control_compat.c b/sound/core/control_compat.c index 470dabc60aa0..edff063e088d 100644 --- a/sound/core/control_compat.c +++ b/sound/core/control_compat.c @@ -264,6 +264,7 @@ static int copy_ctl_value_to_user(void __user *userdata, struct snd_ctl_elem_value *data, int type, int count) { + struct snd_ctl_elem_value32 __user *data32 = userdata; int i, size; if (type == SNDRV_CTL_ELEM_TYPE_BOOLEAN || @@ -280,6 +281,8 @@ static int copy_ctl_value_to_user(void __user *userdata, if (copy_to_user(valuep, data->value.bytes.data, size)) return -EFAULT; } + if (copy_to_user(&data32->id, &data->id, sizeof(data32->id))) + return -EFAULT; return 0; } -- cgit v1.2.3 From 619764cc2ec9ce1283a8bbcd89a1376a7c68293b Mon Sep 17 00:00:00 2001 From: Werner Sembach Date: Thu, 2 Dec 2021 17:50:10 +0100 Subject: ALSA: hda/realtek: Fix quirk for TongFang PHxTxX1 This fixes the SND_PCI_QUIRK(...) of the TongFang PHxTxX1 barebone. This fixes the issue of sound not working after s3 suspend. When waking up from s3 suspend the Coef 0x10 is set to 0x0220 instead of 0x0020. Setting the value manually makes the sound work again. This patch does this automatically. While being on it, I also fixed the comment formatting of the quirk and shortened variable and function names. Signed-off-by: Werner Sembach Fixes: dd6dd6e3c791 ("ALSA: hda/realtek: Add quirk for TongFang PHxTxX1") Cc: Link: https://lore.kernel.org/r/20211202165010.876431-1-wse@tuxedocomputers.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 40 ++++++++++++++++++++++------------------ 1 file changed, 22 insertions(+), 18 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 9ce7457533c9..d361a1260d5a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6503,22 +6503,26 @@ static void alc287_fixup_legion_15imhg05_speakers(struct hda_codec *codec, /* for alc285_fixup_ideapad_s740_coef() */ #include "ideapad_s740_helper.c" -static void alc256_fixup_tongfang_reset_persistent_settings(struct hda_codec *codec, - const struct hda_fixup *fix, - int action) +static const struct coef_fw alc256_fixup_set_coef_defaults_coefs[] = { + WRITE_COEF(0x10, 0x0020), WRITE_COEF(0x24, 0x0000), + WRITE_COEF(0x26, 0x0000), WRITE_COEF(0x29, 0x3000), + WRITE_COEF(0x37, 0xfe05), WRITE_COEF(0x45, 0x5089), + {} +}; + +static void alc256_fixup_set_coef_defaults(struct hda_codec *codec, + const struct hda_fixup *fix, + int action) { /* - * A certain other OS sets these coeffs to different values. On at least one TongFang - * barebone these settings might survive even a cold reboot. So to restore a clean slate the - * values are explicitly reset to default here. Without this, the external microphone is - * always in a plugged-in state, while the internal microphone is always in an unplugged - * state, breaking the ability to use the internal microphone. - */ - alc_write_coef_idx(codec, 0x24, 0x0000); - alc_write_coef_idx(codec, 0x26, 0x0000); - alc_write_coef_idx(codec, 0x29, 0x3000); - alc_write_coef_idx(codec, 0x37, 0xfe05); - alc_write_coef_idx(codec, 0x45, 0x5089); + * A certain other OS sets these coeffs to different values. On at least + * one TongFang barebone these settings might survive even a cold + * reboot. So to restore a clean slate the values are explicitly reset + * to default here. Without this, the external microphone is always in a + * plugged-in state, while the internal microphone is always in an + * unplugged state, breaking the ability to use the internal microphone. + */ + alc_process_coef_fw(codec, alc256_fixup_set_coef_defaults_coefs); } static const struct coef_fw alc233_fixup_no_audio_jack_coefs[] = { @@ -6759,7 +6763,7 @@ enum { ALC287_FIXUP_LEGION_15IMHG05_AUTOMUTE, ALC287_FIXUP_YOGA7_14ITL_SPEAKERS, ALC287_FIXUP_13S_GEN2_SPEAKERS, - ALC256_FIXUP_TONGFANG_RESET_PERSISTENT_SETTINGS, + ALC256_FIXUP_SET_COEF_DEFAULTS, ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE, ALC233_FIXUP_NO_AUDIO_JACK, }; @@ -8465,9 +8469,9 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MODE, }, - [ALC256_FIXUP_TONGFANG_RESET_PERSISTENT_SETTINGS] = { + [ALC256_FIXUP_SET_COEF_DEFAULTS] = { .type = HDA_FIXUP_FUNC, - .v.func = alc256_fixup_tongfang_reset_persistent_settings, + .v.func = alc256_fixup_set_coef_defaults, }, [ALC245_FIXUP_HP_GPIO_LED] = { .type = HDA_FIXUP_FUNC, @@ -8929,7 +8933,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1b7d, 0xa831, "Ordissimo EVE2 ", ALC269VB_FIXUP_ORDISSIMO_EVE2), /* Also known as Malata PC-B1303 */ SND_PCI_QUIRK(0x1c06, 0x2013, "Lemote A1802", ALC269_FIXUP_LEMOTE_A1802), SND_PCI_QUIRK(0x1c06, 0x2015, "Lemote A190X", ALC269_FIXUP_LEMOTE_A190X), - SND_PCI_QUIRK(0x1d05, 0x1132, "TongFang PHxTxX1", ALC256_FIXUP_TONGFANG_RESET_PERSISTENT_SETTINGS), + SND_PCI_QUIRK(0x1d05, 0x1132, "TongFang PHxTxX1", ALC256_FIXUP_SET_COEF_DEFAULTS), SND_PCI_QUIRK(0x1d72, 0x1602, "RedmiBook", ALC255_FIXUP_XIAOMI_HEADSET_MIC), SND_PCI_QUIRK(0x1d72, 0x1701, "XiaomiNotebook Pro", ALC298_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1d72, 0x1901, "RedmiBook 14", ALC256_FIXUP_ASUS_HEADSET_MIC), -- cgit v1.2.3 From fb1af5bea4670c835e42fc0c14c49d3499468774 Mon Sep 17 00:00:00 2001 From: Geraldo Nascimento Date: Sat, 4 Dec 2021 15:52:24 -0300 Subject: ALSA: usb-audio: Reorder snd_djm_devices[] entries Olivia Mackintosh has posted to alsa-devel reporting that there's a potential bug that could break mixer quirks for Pioneer devices introduced by 6d27788160362a7ee6c0d317636fe4b1ddbe59a7 "ALSA: usb-audio: Add support for the Pioneer DJM 750MK2 Mixer/Soundcard". This happened because the DJM 750 MK2 was added last to the Pioneer DJM device table index and defined as 0x4 but was added to snd_djm_devices[] just after the DJM 750 (MK1) entry instead of last, after the DJM 900 NXS2. This escaped review. To prevent that from ever happening again, Takashi Iwai suggested to use C99 array designators in snd_djm_devices[] instead of simply reordering the entries. Fixes: 6d2778816036 ("ALSA: usb-audio: Add support for the Pioneer DJM 750MK2") Reported-by: Olivia Mackintosh Suggested-by: Takashi Iwai Signed-off-by: Geraldo Nascimento Link: https://lore.kernel.org/r/Yau46FDzoql0SNnW@geday Signed-off-by: Takashi Iwai --- sound/usb/mixer_quirks.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index d489c1de3bae..823b6b8de942 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -3016,11 +3016,11 @@ static const struct snd_djm_ctl snd_djm_ctls_750mk2[] = { static const struct snd_djm_device snd_djm_devices[] = { - SND_DJM_DEVICE(250mk2), - SND_DJM_DEVICE(750), - SND_DJM_DEVICE(750mk2), - SND_DJM_DEVICE(850), - SND_DJM_DEVICE(900nxs2) + [SND_DJM_250MK2_IDX] = SND_DJM_DEVICE(250mk2), + [SND_DJM_750_IDX] = SND_DJM_DEVICE(750), + [SND_DJM_850_IDX] = SND_DJM_DEVICE(850), + [SND_DJM_900NXS2_IDX] = SND_DJM_DEVICE(900nxs2), + [SND_DJM_750MK2_IDX] = SND_DJM_DEVICE(750mk2), }; -- cgit v1.2.3 From d7f32791a9fcf0dae8b073cdea9b79e29098c5f4 Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Tue, 23 Nov 2021 16:32:44 +0800 Subject: ALSA: hda/realtek - Add headset Mic support for Lenovo ALC897 platform Lenovo ALC897 platform had headset Mic. This patch enable supported headset Mic. Signed-off-by: Kailang Yang Cc: Link: https://lore.kernel.org/r/baab2c2536cb4cc18677a862c6f6d840@realtek.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 40 ++++++++++++++++++++++++++++++++++++++++ 1 file changed, 40 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d361a1260d5a..3599f4c85ebf 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10235,6 +10235,27 @@ static void alc671_fixup_hp_headset_mic2(struct hda_codec *codec, } } +static void alc897_hp_automute_hook(struct hda_codec *codec, + struct hda_jack_callback *jack) +{ + struct alc_spec *spec = codec->spec; + int vref; + + snd_hda_gen_hp_automute(codec, jack); + vref = spec->gen.hp_jack_present ? (PIN_HP | AC_PINCTL_VREF_100) : PIN_HP; + snd_hda_codec_write(codec, 0x1b, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, + vref); +} + +static void alc897_fixup_lenovo_headset_mic(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + spec->gen.hp_automute_hook = alc897_hp_automute_hook; + } +} + static const struct coef_fw alc668_coefs[] = { WRITE_COEF(0x01, 0xbebe), WRITE_COEF(0x02, 0xaaaa), WRITE_COEF(0x03, 0x0), WRITE_COEF(0x04, 0x0180), WRITE_COEF(0x06, 0x0), WRITE_COEF(0x07, 0x0f80), @@ -10315,6 +10336,8 @@ enum { ALC668_FIXUP_ASUS_NO_HEADSET_MIC, ALC668_FIXUP_HEADSET_MIC, ALC668_FIXUP_MIC_DET_COEF, + ALC897_FIXUP_LENOVO_HEADSET_MIC, + ALC897_FIXUP_HEADSET_MIC_PIN, }; static const struct hda_fixup alc662_fixups[] = { @@ -10721,6 +10744,19 @@ static const struct hda_fixup alc662_fixups[] = { {} }, }, + [ALC897_FIXUP_LENOVO_HEADSET_MIC] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc897_fixup_lenovo_headset_mic, + }, + [ALC897_FIXUP_HEADSET_MIC_PIN] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x1a, 0x03a11050 }, + { } + }, + .chained = true, + .chain_id = ALC897_FIXUP_LENOVO_HEADSET_MIC + }, }; static const struct snd_pci_quirk alc662_fixup_tbl[] = { @@ -10765,6 +10801,10 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x144d, 0xc051, "Samsung R720", ALC662_FIXUP_IDEAPAD), SND_PCI_QUIRK(0x14cd, 0x5003, "USI", ALC662_FIXUP_USI_HEADSET_MODE), SND_PCI_QUIRK(0x17aa, 0x1036, "Lenovo P520", ALC662_FIXUP_LENOVO_MULTI_CODECS), + SND_PCI_QUIRK(0x17aa, 0x32ca, "Lenovo ThinkCentre M80", ALC897_FIXUP_HEADSET_MIC_PIN), + SND_PCI_QUIRK(0x17aa, 0x32cb, "Lenovo ThinkCentre M70", ALC897_FIXUP_HEADSET_MIC_PIN), + SND_PCI_QUIRK(0x17aa, 0x32cf, "Lenovo ThinkCentre M950", ALC897_FIXUP_HEADSET_MIC_PIN), + SND_PCI_QUIRK(0x17aa, 0x32f7, "Lenovo ThinkCentre M90", ALC897_FIXUP_HEADSET_MIC_PIN), SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo Ideapad Y550P", ALC662_FIXUP_IDEAPAD), SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo Ideapad Y550", ALC662_FIXUP_IDEAPAD), SND_PCI_QUIRK(0x1849, 0x5892, "ASRock B150M", ALC892_FIXUP_ASROCK_MOBO), -- cgit v1.2.3