- Alsa-devel - mailman.alsa-project.org

[PATCH v2 4/5] ASoC: amd: Adding DAI LINK for rt1015 codec
by Ravulapati Vishnu vardhan rao 28 Jul '20

28 Jul '20

DAI link support for RTK 1015 and providing the codec details depending on the snd_soc_card selected by ACPI ID. Signed-off-by: Ravulapati Vishnu vardhan rao <Vishnuvardhanrao.Ravulapati(a)amd.com> --- sound/soc/amd/acp3x-rt5682-max9836.c | 35 ++++++++++++++++++++++++---- 1 file changed, 30 insertions(+), 5 deletions(-) diff --git a/sound/soc/amd/acp3x-rt5682-max9836.c b/sound/soc/amd/acp3x-rt5682-max9836.c index 015b6a3312f5..e0c767716b9a 100644 --- a/sound/soc/amd/acp3x-rt5682-max9836.c +++ b/sound/soc/amd/acp3x-rt5682-max9836.c @@ -33,6 +33,12 @@ static struct clk *rt5682_dai_bclk; static struct gpio_desc *dmic_sel; void *soc_is_rltk_max(struct device *dev); +enum { + RT5682 = 0, + MAX, + EC, +}; + static int acp3x_5682_init(struct snd_soc_pcm_runtime *rtd) { int ret; @@ -242,6 +248,9 @@ SND_SOC_DAILINK_DEF(rt5682, DAILINK_COMP_ARRAY(COMP_CODEC("i2c-10EC5682:00", "rt5682-aif1"))); SND_SOC_DAILINK_DEF(max, DAILINK_COMP_ARRAY(COMP_CODEC("MX98357A:00", "HiFi"))); +SND_SOC_DAILINK_DEF(rt1015, + DAILINK_COMP_ARRAY(COMP_CODEC("i2c-10EC1015:00", "rt1015-aif"), + COMP_CODEC("i2c-10EC1015:01", "rt1015-aif"))); SND_SOC_DAILINK_DEF(cros_ec, DAILINK_COMP_ARRAY(COMP_CODEC("GOOG0013:00", "EC Codec I2S RX"))); @@ -260,7 +269,7 @@ static struct snd_soc_codec_conf rt1015_conf[] = { }; static struct snd_soc_dai_link acp3x_dai[] = { - { + [RT5682] = { .name = "acp3x-5682-play", .stream_name = "Playback", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF @@ -271,16 +280,19 @@ static struct snd_soc_dai_link acp3x_dai[] = { .ops = &acp3x_5682_ops, SND_SOC_DAILINK_REG(acp3x_i2s, rt5682, platform), }, - { + [MAX] = { .name = "acp3x-max98357-play", .stream_name = "HiFi Playback", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM, + | SND_SOC_DAIFMT_CBS_CFS, .dpcm_playback = 1, .ops = &acp3x_max_play_ops, - SND_SOC_DAILINK_REG(acp3x_bt, max, platform), + .cpus = acp3x_bt, + .num_cpus = ARRAY_SIZE(acp3x_bt), + .platforms = platform, + .num_platforms = ARRAY_SIZE(platform), }, - { + [EC] = { .name = "acp3x-ec-dmic0-capture", .stream_name = "Capture DMIC0", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF @@ -390,6 +402,18 @@ void *soc_is_rltk_max(struct device *dev) return (void *)match->driver_data; } +static void card_spk_dai_link_present(struct snd_soc_dai_link *links, + const char *card_name) +{ + if (!strcmp(card_name, "acp3xalc56821015")) { + links[1].codecs = rt1015; + links[1].num_codecs = ARRAY_SIZE(rt1015); + } else { + links[1].codecs = max; + links[1].num_codecs = ARRAY_SIZE(max); + } +} + static int acp3x_probe(struct platform_device *pdev) { int ret; @@ -405,6 +429,7 @@ static int acp3x_probe(struct platform_device *pdev) if (!machine) return -ENOMEM; + card_spk_dai_link_present(card->dai_link, card->name); card->dev = &pdev->dev; platform_set_drvdata(pdev, card); snd_soc_card_set_drvdata(card, machine); -- 2.17.1

1 0

[PATCH v2 3/5] ASoC: amd: Adding support for ALC1015 codec in machine driver
by Ravulapati Vishnu vardhan rao 28 Jul '20

28 Jul '20

Adding support for ALC1015 RTK codec in machine driver. Passing specific card structure based on its ACPI ID. Signed-off-by: Ravulapati Vishnu vardhan rao <Vishnuvardhanrao.Ravulapati(a)amd.com> --- sound/soc/amd/Kconfig | 1 + sound/soc/amd/acp3x-rt5682-max9836.c | 63 ++++++++++++++++++++++++++-- 2 files changed, 60 insertions(+), 4 deletions(-) diff --git a/sound/soc/amd/Kconfig b/sound/soc/amd/Kconfig index e37cf72f2bab..a6ce000fac3f 100644 --- a/sound/soc/amd/Kconfig +++ b/sound/soc/amd/Kconfig @@ -33,6 +33,7 @@ config SND_SOC_AMD_RV_RT5682_MACH select SND_SOC_MAX98357A select SND_SOC_CROS_EC_CODEC select I2C_CROS_EC_TUNNEL + select SND_SOC_RT1015 depends on SND_SOC_AMD_ACP3x && I2C && CROS_EC help This option enables machine driver for RT5682 and MAX9835. diff --git a/sound/soc/amd/acp3x-rt5682-max9836.c b/sound/soc/amd/acp3x-rt5682-max9836.c index 2c672657d139..015b6a3312f5 100644 --- a/sound/soc/amd/acp3x-rt5682-max9836.c +++ b/sound/soc/amd/acp3x-rt5682-max9836.c @@ -21,6 +21,7 @@ #include "raven/acp3x.h" #include "../codecs/rt5682.h" +#include "../codecs/rt1015.h" #define PCO_PLAT_CLK 48000000 #define RT5682_PLL_FREQ (48000 * 512) @@ -247,7 +248,18 @@ SND_SOC_DAILINK_DEF(cros_ec, SND_SOC_DAILINK_DEF(platform, DAILINK_COMP_ARRAY(COMP_PLATFORM("acp3x_rv_i2s_dma.0"))); -static struct snd_soc_dai_link acp3x_dai_5682_98357[] = { +static struct snd_soc_codec_conf rt1015_conf[] = { + { + .dlc = COMP_CODEC_CONF("i2c-10EC1015:00"), + .name_prefix = "Left", + }, + { + .dlc = COMP_CODEC_CONF("i2c-10EC1015:01"), + .name_prefix = "Right", + }, +}; + +static struct snd_soc_dai_link acp3x_dai[] = { { .name = "acp3x-5682-play", .stream_name = "Playback", @@ -317,8 +329,8 @@ static const struct snd_kcontrol_new acp3x_5682_mc_controls[] = { static struct snd_soc_card acp3x_5682 = { .name = "acp3xalc5682m98357", .owner = THIS_MODULE, - .dai_link = acp3x_dai_5682_98357, - .num_links = ARRAY_SIZE(acp3x_dai_5682_98357), + .dai_link = acp3x_dai, + .num_links = ARRAY_SIZE(acp3x_dai), .dapm_widgets = acp3x_5682_widgets, .num_dapm_widgets = ARRAY_SIZE(acp3x_5682_widgets), .dapm_routes = acp3x_5682_audio_route, @@ -327,6 +339,47 @@ static struct snd_soc_card acp3x_5682 = { .num_controls = ARRAY_SIZE(acp3x_5682_mc_controls), }; +static const struct snd_soc_dapm_widget acp3x_1015_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MUX("Dmic Mux", SND_SOC_NOPM, 0, 0, + &acp3x_dmic_mux_control), + SND_SOC_DAPM_SPK("Left Spk", NULL), + SND_SOC_DAPM_SPK("Right Spk", NULL), +}; + +static const struct snd_soc_dapm_route acp3x_1015_route[] = { + {"Headphone Jack", NULL, "HPOL"}, + {"Headphone Jack", NULL, "HPOR"}, + {"IN1P", NULL, "Headset Mic"}, + {"Dmic Mux", "Front Mic", "DMIC"}, + {"Dmic Mux", "Rear Mic", "DMIC"}, + {"Left Spk", NULL, "Left SPO"}, + {"Right Spk", NULL, "Right SPO"}, +}; + +static const struct snd_kcontrol_new acp3x_mc_1015_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphone Jack"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), + SOC_DAPM_PIN_SWITCH("Left Spk"), + SOC_DAPM_PIN_SWITCH("Right Spk"), +}; + +static struct snd_soc_card acp3x_1015 = { + .name = "acp3xalc56821015", + .owner = THIS_MODULE, + .dai_link = acp3x_dai, + .num_links = ARRAY_SIZE(acp3x_dai), + .dapm_widgets = acp3x_1015_widgets, + .num_dapm_widgets = ARRAY_SIZE(acp3x_1015_widgets), + .dapm_routes = acp3x_1015_route, + .num_dapm_routes = ARRAY_SIZE(acp3x_1015_route), + .codec_conf = rt1015_conf, + .num_configs = ARRAY_SIZE(rt1015_conf), + .controls = acp3x_mc_1015_controls, + .num_controls = ARRAY_SIZE(acp3x_mc_1015_controls), +}; + void *soc_is_rltk_max(struct device *dev) { const struct acpi_device_id *match; @@ -375,6 +428,7 @@ static int acp3x_probe(struct platform_device *pdev) static const struct acpi_device_id acp3x_audio_acpi_match[] = { { "AMDI5682", (unsigned long)&acp3x_5682}, + { "AMDI1015", (unsigned long)&acp3x_1015}, {}, }; MODULE_DEVICE_TABLE(acpi, acp3x_audio_acpi_match); @@ -391,5 +445,6 @@ static struct platform_driver acp3x_audio = { module_platform_driver(acp3x_audio); MODULE_AUTHOR("akshu.agrawal(a)amd.com"); -MODULE_DESCRIPTION("ALC5682 & MAX98357 audio support"); +MODULE_AUTHOR("Vishnuvardhanrao.Ravulapati(a)amd.com"); +MODULE_DESCRIPTION("ALC5682 ALC1015 & MAX98357 audio support"); MODULE_LICENSE("GPL v2"); -- 2.17.1

1 0

[PATCH v2 2/5] ASoC: amd: Passing card structure based on codec
by Ravulapati Vishnu vardhan rao 28 Jul '20

28 Jul '20

Passing specific snd_soc_card structure depending on the ACPI ID. In future we can add other IDs in the ACPI table and pass the structure. Signed-off-by: Ravulapati Vishnu vardhan rao <Vishnuvardhanrao.Ravulapati(a)amd.com> --- sound/soc/amd/acp3x-rt5682-max9836.c | 25 ++++++++++++++++++++----- 1 file changed, 20 insertions(+), 5 deletions(-) diff --git a/sound/soc/amd/acp3x-rt5682-max9836.c b/sound/soc/amd/acp3x-rt5682-max9836.c index 8b5af064864f..2c672657d139 100644 --- a/sound/soc/amd/acp3x-rt5682-max9836.c +++ b/sound/soc/amd/acp3x-rt5682-max9836.c @@ -30,6 +30,7 @@ static struct snd_soc_jack pco_jack; static struct clk *rt5682_dai_wclk; static struct clk *rt5682_dai_bclk; static struct gpio_desc *dmic_sel; +void *soc_is_rltk_max(struct device *dev); static int acp3x_5682_init(struct snd_soc_pcm_runtime *rtd) { @@ -326,18 +327,32 @@ static struct snd_soc_card acp3x_5682 = { .num_controls = ARRAY_SIZE(acp3x_5682_mc_controls), }; +void *soc_is_rltk_max(struct device *dev) +{ + const struct acpi_device_id *match; + + match = acpi_match_device(dev->driver->acpi_match_table, dev); + if (!match) + return NULL; + return (void *)match->driver_data; +} + static int acp3x_probe(struct platform_device *pdev) { int ret; struct snd_soc_card *card; struct acp3x_platform_info *machine; + struct device *dev = &pdev->dev; + + card = (struct snd_soc_card *)soc_is_rltk_max(dev); + if (!card) + return -ENODEV; machine = devm_kzalloc(&pdev->dev, sizeof(*machine), GFP_KERNEL); if (!machine) return -ENOMEM; - card = &acp3x_5682; - acp3x_5682.dev = &pdev->dev; + card->dev = &pdev->dev; platform_set_drvdata(pdev, card); snd_soc_card_set_drvdata(card, machine); @@ -348,18 +363,18 @@ static int acp3x_probe(struct platform_device *pdev) return PTR_ERR(dmic_sel); } - ret = devm_snd_soc_register_card(&pdev->dev, &acp3x_5682); + ret = devm_snd_soc_register_card(&pdev->dev, card); if (ret) { dev_err(&pdev->dev, "devm_snd_soc_register_card(%s) failed: %d\n", - acp3x_5682.name, ret); + card->name, ret); return ret; } return 0; } static const struct acpi_device_id acp3x_audio_acpi_match[] = { - { "AMDI5682", 0 }, + { "AMDI5682", (unsigned long)&acp3x_5682}, {}, }; MODULE_DEVICE_TABLE(acpi, acp3x_audio_acpi_match); -- 2.17.1

1 0

[PATCH 1/6] ASoC: amd: Renaming snd-soc-card structure and fields.
by Ravulapati Vishnu vardhan rao 28 Jul '20

28 Jul '20

As in future our machine driver supports multiple codecs So changing naming convention of snd_soc_card struct and its fields. Signed-off-by: Ravulapati Vishnu vardhan rao <Vishnuvardhanrao.Ravulapati(a)amd.com> --- sound/soc/amd/acp3x-rt5682-max9836.c | 28 ++++++++++++++-------------- 1 file changed, 14 insertions(+), 14 deletions(-) diff --git a/sound/soc/amd/acp3x-rt5682-max9836.c b/sound/soc/amd/acp3x-rt5682-max9836.c index 6009e444b858..8b5af064864f 100644 --- a/sound/soc/amd/acp3x-rt5682-max9836.c +++ b/sound/soc/amd/acp3x-rt5682-max9836.c @@ -290,7 +290,7 @@ static const struct snd_kcontrol_new acp3x_dmic_mux_control = SOC_DAPM_ENUM_EXT("DMIC Select Mux", acp3x_dmic_enum, dmic_get, dmic_set); -static const struct snd_soc_dapm_widget acp3x_widgets[] = { +static const struct snd_soc_dapm_widget acp3x_5682_widgets[] = { SND_SOC_DAPM_HP("Headphone Jack", NULL), SND_SOC_DAPM_SPK("Spk", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), @@ -298,7 +298,7 @@ static const struct snd_soc_dapm_widget acp3x_widgets[] = { &acp3x_dmic_mux_control), }; -static const struct snd_soc_dapm_route acp3x_audio_route[] = { +static const struct snd_soc_dapm_route acp3x_5682_audio_route[] = { {"Headphone Jack", NULL, "HPOL"}, {"Headphone Jack", NULL, "HPOR"}, {"IN1P", NULL, "Headset Mic"}, @@ -307,23 +307,23 @@ static const struct snd_soc_dapm_route acp3x_audio_route[] = { {"Dmic Mux", "Rear Mic", "DMIC"}, }; -static const struct snd_kcontrol_new acp3x_mc_controls[] = { +static const struct snd_kcontrol_new acp3x_5682_mc_controls[] = { SOC_DAPM_PIN_SWITCH("Headphone Jack"), SOC_DAPM_PIN_SWITCH("Spk"), SOC_DAPM_PIN_SWITCH("Headset Mic"), }; -static struct snd_soc_card acp3x_card = { +static struct snd_soc_card acp3x_5682 = { .name = "acp3xalc5682m98357", .owner = THIS_MODULE, .dai_link = acp3x_dai_5682_98357, .num_links = ARRAY_SIZE(acp3x_dai_5682_98357), - .dapm_widgets = acp3x_widgets, - .num_dapm_widgets = ARRAY_SIZE(acp3x_widgets), - .dapm_routes = acp3x_audio_route, - .num_dapm_routes = ARRAY_SIZE(acp3x_audio_route), - .controls = acp3x_mc_controls, - .num_controls = ARRAY_SIZE(acp3x_mc_controls), + .dapm_widgets = acp3x_5682_widgets, + .num_dapm_widgets = ARRAY_SIZE(acp3x_5682_widgets), + .dapm_routes = acp3x_5682_audio_route, + .num_dapm_routes = ARRAY_SIZE(acp3x_5682_audio_route), + .controls = acp3x_5682_mc_controls, + .num_controls = ARRAY_SIZE(acp3x_5682_mc_controls), }; static int acp3x_probe(struct platform_device *pdev) @@ -336,8 +336,8 @@ static int acp3x_probe(struct platform_device *pdev) if (!machine) return -ENOMEM; - card = &acp3x_card; - acp3x_card.dev = &pdev->dev; + card = &acp3x_5682; + acp3x_5682.dev = &pdev->dev; platform_set_drvdata(pdev, card); snd_soc_card_set_drvdata(card, machine); @@ -348,11 +348,11 @@ static int acp3x_probe(struct platform_device *pdev) return PTR_ERR(dmic_sel); } - ret = devm_snd_soc_register_card(&pdev->dev, &acp3x_card); + ret = devm_snd_soc_register_card(&pdev->dev, &acp3x_5682); if (ret) { dev_err(&pdev->dev, "devm_snd_soc_register_card(%s) failed: %d\n", - acp3x_card.name, ret); + acp3x_5682.name, ret); return ret; } return 0; -- 2.17.1

6 16

arecord command stop when output file set /dev/null
by GitHub issues - opened 28 Jul '20

28 Jul '20

alsa-project/alsa-utils issue #48 was opened from garyfcg: How to reproduce it? 1. Input the below command, then record go on arecord -r 96000 -c 4 -f S32_LE 1> /dev/null 2. Wait for some minutes, it depended on system memory size. For example,system memory had 2G, (2*1024*1024*1024)/(96k*4*32/8)=1400s The results: 1. this command quit auto. Expect: 1. This command should not be interrupted. Issue URL : https://github.com/alsa-project/alsa-utils/issues/48 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH] ALSA: hda/hdmi: Fix keep_power assignment for non-component devices
by Takashi Iwai 28 Jul '20

28 Jul '20

It's been reported that, when neither nouveau nor Nvidia graphics driver is used, the screen starts flickering. And, after comparing between the working case (stable 4.4.x) and the broken case, it turned out that the problem comes from the audio component binding. The Nvidia and AMD audio binding code clears the bus->keep_power flag whenever snd_hdac_acomp_init() succeeds. But this doesn't mean that the component is actually bound, but it merely indicates that it's ready for binding. So, when both nouveau and Nvidia are blacklisted or not ready, the driver keeps running without the audio component but also with bus->keep_power = false. This made the driver runtime PM kicked in and powering down when unused, which results in flickering in the graphics side, as it seems. For fixing the bug, this patch moves the bus->keep_power flag change into generic_acomp_notifier_set() that is the function called from the master_bind callback of component ops; i.e. it's guaranteed that the binding succeeded. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=208609 Fixes: 5a858e79c911 ("ALSA: hda - Disable audio component for legacy Nvidia HDMI codecs") Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_hdmi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 41eaa89660c3..cd46247988e4 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2440,6 +2440,7 @@ static void generic_acomp_notifier_set(struct drm_audio_component *acomp, mutex_lock(&spec->bind_lock); spec->use_acomp_notifier = use_acomp; spec->codec->relaxed_resume = use_acomp; + spec->codec->bus->keep_power = 0; /* reprogram each jack detection logic depending on the notifier */ for (i = 0; i < spec->num_pins; i++) reprogram_jack_detect(spec->codec, @@ -2534,7 +2535,6 @@ static void generic_acomp_init(struct hda_codec *codec, if (!snd_hdac_acomp_init(&codec->bus->core, &spec->drm_audio_ops, match_bound_vga, 0)) { spec->acomp_registered = true; - codec->bus->keep_power = 0; } } -- 2.16.4

1 0

[PATCH 3/3] ASoC: max98390: update dsm param bin max size
by Steve Lee 28 Jul '20

28 Jul '20

MAX98390_DSM_PARAM_MAX_SIZE is changed to support extended register update. Signed-off-by: Steve Lee <steves.lee(a)maximintegrated.com> --- sound/soc/codecs/max98390.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/max98390.h b/sound/soc/codecs/max98390.h index 5f444e7779b0..dff884f68e3e 100644 --- a/sound/soc/codecs/max98390.h +++ b/sound/soc/codecs/max98390.h @@ -650,7 +650,7 @@ /* DSM register offset */ #define MAX98390_DSM_PAYLOAD_OFFSET 16 -#define MAX98390_DSM_PARAM_MAX_SIZE 770 +#define MAX98390_DSM_PARAM_MAX_SIZE 1024 #define MAX98390_DSM_PARAM_MIN_SIZE 670 struct max98390_priv { -- 2.17.1

2 4

USB interface: Mixer controls out-of-sync?
by Tanjeff Moos 27 Jul '20

27 Jul '20

Hi all, after connecting my USB interface (Focusrite Scarlett 6i6 1st Gen), the alsa control element values reported by libalsa are not in sync with the hardware. For example, the "Master 1 (Monitor)" control is shown as muted, but a signal can be heard. Unmuting changes nothing, and muting again disables the signal. In practice I often have to go through many controls and alter them a little to synchronize alsamixer with the actual hardware settings. Is this a known behavior? Kind regards, Tanjeff

1 0

headphones detected but don't show up in alsa mixer, no audio on ubuntu 18.04
by GitHub issues - edited 27 Jul '20

27 Jul '20

alsa-project/alsa-lib issue #75 was edited from rbavery: Here is my ALSA info: http://alsa-project.org/db/?f=dd3d1d8a9fa0020900a0adc03bc6adfaf1f77ee5 I tried following these remedies (including reinstalling alsa, reloading, restarting computer) but nothing has panned out: https://support.system76.com/articles/audio/ Any help or suggestions are appreciated Issue URL : https://github.com/alsa-project/alsa-lib/issues/75 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH] ALSA: hda: Workaround for spurious wakeups on some Intel platforms
by Takashi Iwai 27 Jul '20

27 Jul '20

We've received a regression report on Intel HD-audio controller that wakes up immediately after S3 suspend. The bisection leads to the commit c4c8dd6ef807 ("ALSA: hda: Skip controller resume if not needed"). This commit replaces the system-suspend to use pm_runtime_force_suspend() instead of the direct call of __azx_runtime_suspend(). However, by some really mysterious reason, pm_runtime_force_suspend() causes a spurious wakeup (although it calls the same __azx_runtime_suspend() internally). As an ugly workaround for now, revert the behavior to call __azx_runtime_suspend() and __azx_runtime_resume() for those old Intel platforms that may exhibit such a problem, while keeping the new standard pm_runtime_force_suspend() and pm_runtime_force_resume() pair for the remaining chips. Fixes: c4c8dd6ef807 ("ALSA: hda: Skip controller resume if not needed") BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=208649 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_controller.h | 2 +- sound/pci/hda/hda_intel.c | 17 ++++++++++++++--- 2 files changed, 15 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/hda_controller.h b/sound/pci/hda/hda_controller.h index fe171685492d..be63ead8161f 100644 --- a/sound/pci/hda/hda_controller.h +++ b/sound/pci/hda/hda_controller.h @@ -41,7 +41,7 @@ /* 24 unused */ #define AZX_DCAPS_COUNT_LPIB_DELAY (1 << 25) /* Take LPIB as delay */ #define AZX_DCAPS_PM_RUNTIME (1 << 26) /* runtime PM support */ -/* 27 unused */ +#define AZX_DCAPS_SUSPEND_SPURIOUS_WAKEUP (1 << 27) /* Workaround for spurious wakeups after suspend */ #define AZX_DCAPS_CORBRP_SELF_CLEAR (1 << 28) /* CORBRP clears itself after reset */ #define AZX_DCAPS_NO_MSI64 (1 << 29) /* Stick to 32-bit MSIs */ #define AZX_DCAPS_SEPARATE_STREAM_TAG (1 << 30) /* capture and playback use separate stream tag */ diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 9ba1fb8f0b7f..fb65450d8de1 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -297,7 +297,8 @@ enum { /* PCH for HSW/BDW; with runtime PM */ /* no i915 binding for this as HSW/BDW has another controller for HDMI */ #define AZX_DCAPS_INTEL_PCH \ - (AZX_DCAPS_INTEL_PCH_BASE | AZX_DCAPS_PM_RUNTIME) + (AZX_DCAPS_INTEL_PCH_BASE | AZX_DCAPS_PM_RUNTIME |\ + AZX_DCAPS_SUSPEND_SPURIOUS_WAKEUP) /* HSW HDMI */ #define AZX_DCAPS_INTEL_HASWELL \ @@ -1026,7 +1027,14 @@ static int azx_suspend(struct device *dev) chip = card->private_data; bus = azx_bus(chip); snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); - pm_runtime_force_suspend(dev); + /* An ugly workaround: direct call of __azx_runtime_suspend() and + * __azx_runtime_resume() for old Intel platforms that suffer from + * spurious wakeups after S3 suspend + */ + if (chip->driver_caps & AZX_DCAPS_SUSPEND_SPURIOUS_WAKEUP) + __azx_runtime_suspend(chip); + else + pm_runtime_force_suspend(dev); if (bus->irq >= 0) { free_irq(bus->irq, chip); bus->irq = -1; @@ -1055,7 +1063,10 @@ static int azx_resume(struct device *dev) if (azx_acquire_irq(chip, 1) < 0) return -EIO; - pm_runtime_force_resume(dev); + if (chip->driver_caps & AZX_DCAPS_SUSPEND_SPURIOUS_WAKEUP) + __azx_runtime_resume(chip, false); + else + pm_runtime_force_resume(dev); snd_power_change_state(card, SNDRV_CTL_POWER_D0); trace_azx_resume(chip); -- 2.26.2

2 2

Lenovo X270 with HDA ALC298
by Yasushi SHOJI 27 Jul '20

27 Jul '20

Hi all, I noticed that my 3 years old Lenovo X270 with Intel HDA & RealTek ALC298 doesn't work with 3.5 mm audio combo jack with 4 pole CTIA. A bit of googling showed me that `snd-hda-intel` module can take some options like `model` to run some fix ups. and I have found that `model=alc298-dell1` seems to at least work. But its gain is way too low. It's mostly just noise. It sounds like my mic is on the line-in. So, I opened `patch_realtek.c` to see what's going on. I found the following but not sure what the values for `.v.pins` are for. ``` [ALC298_FIXUP_DELL1_MIC_NO_PRESENCE] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { { 0x18, 0x01a1913c }, /* use as headset mic, without its own jack detect */ { 0x1a, 0x01a1913d }, /* use as headphone mic, without its own jack detect */ { } }, .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MODE }, ``` In `apply_fix(), these `.v.pins` values are passed to `snd_hda_apply_pincfgs()` but I can't find what values mean. Would someone please enlighten me? My questions are 1) How can I found out which pin goes where. There is x270 schematics on the net but not sure it's correct. 2) What does the value, ie, 0x01a1913d, represent? Thanks, -- yashi

2 1

/dev/sequencer descriptions
by Andrey Konovalov 27 Jul '20

27 Jul '20

Hi Takashi, We're working on syzkaller descriptions for /dev/sequencer [1], but have some troubles understanding its functionality. The main question is: is sound/synth/emux/ code reachable when tested in a VM (we're using QEMU with -soundhw all), or does it require some specific hardware to be present? In our setup (with Debian Stretch userspace image) we have /dev/sequencer with SNDRV_MINOR_OSS_SEQUENCER and /dev/sequencer2 with SNDRV_MINOR_OSS_MUSIC. For the former, there are no synth devices as reported by SNDCTL_SEQ_NRSYNTHS. For the latter, synth devices are there, but the load_patch() callback is not reachable as this check [2] fails. Is that code impossible to test in a VM at all? Or are there some specific kernel options/drivers that need to be enabled? Thanks! [1] https://github.com/google/syzkaller/pull/1952 [2] https://elixir.bootlin.com/linux/v5.7.8/source/sound/core/seq/oss/seq_oss_s…

2 2

[PATCH v2 1/3] ASoC: samsung: Document DT bindings for Midas sound subsystem
by Sylwester Nawrocki 27 Jul '20

27 Jul '20

This patch adds documentation of DT biding for the Midas sound complex. Partially based on the *txt version by Simon Shields <simon(a)lineageos.org>. Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- Changes for v2: - fix wrong *-gpios entries in the example, - mark sound-dai properties as required. --- .../bindings/sound/samsung,midas-audio.yaml | 108 +++++++++++++++++++++ 1 file changed, 108 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/samsung,midas-audio.yaml diff --git a/Documentation/devicetree/bindings/sound/samsung,midas-audio.yaml b/Documentation/devicetree/bindings/sound/samsung,midas-audio.yaml new file mode 100644 index 0000000..fb685e51 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/samsung,midas-audio.yaml @@ -0,0 +1,108 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/samsung,midas-audio.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Samsung Midas audio complex with WM1811 codec + +maintainers: + - Sylwester Nawrocki <s.nawrocki(a)samsung.com> + +properties: + compatible: + const: samsung,midas-audio + + model: + $ref: /schemas/types.yaml#/definitions/string + description: The user-visible name of this sound complex. + + cpu: + type: object + properties: + sound-dai: + $ref: /schemas/types.yaml#/definitions/phandle-array + description: phandle to the I2S controller + required: + - sound-dai + + codec: + type: object + properties: + sound-dai: + $ref: /schemas/types.yaml#/definitions/phandle-array + description: phandle to the WM1811 CODEC node + required: + - sound-dai + + samsung,audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: | + List of the connections between audio components; each entry is + a pair of strings, the first being the connection's sink, the second + being the connection's source; valid names for sources and sinks are + the WM1811's pins (as documented in its binding), and the jacks + on the board: HP, SPK, Main Mic, Sub Mic, Headset Mic. + + mic-bias-supply: + description: Supply for the micbias on the Main microphone + + submic-bias-supply: + description: Supply for the micbias on the Sub microphone + + fm-sel-gpios: + description: GPIO pin for FM selection + + lineout-sel-gpios: + description: GPIO pin for line out selection + +required: + - compatible + - model + - cpu + - codec + - samsung,audio-routing + - mic-bias-supply + - submic-bias-supply + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + + sound { + compatible = "samsung,midas-audio"; + model = "Midas"; + + fm-sel-gpios = <&gpaa0 3 GPIO_ACTIVE_HIGH>; + + mic-bias-supply = <&mic_bias_reg>; + submic-bias-supply = <&submic_bias_reg>; + + samsung,audio-routing = + "HP", "HPOUT1L", + "HP", "HPOUT1R", + + "SPK", "SPKOUTLN", + "SPK", "SPKOUTLP", + "SPK", "SPKOUTRN", + "SPK", "SPKOUTRP", + + "RCV", "HPOUT2N", + "RCV", "HPOUT2P", + + "IN1LP", "Main Mic", + "IN1LN", "Main Mic", + "IN1RP", "Sub Mic", + "IN1LP", "Sub Mic"; + + cpu { + sound-dai = <&i2s0>; + }; + + codec { + sound-dai = <&wm1811>; + }; + + }; -- 2.7.4

2 5

[PATCH] usb: endpoint : remove needless check before usb_free_coherent()
by Xu Wang 27 Jul '20

27 Jul '20

usb_free_coherent() is safe with NULL addr and this check is not required. Signed-off-by: Xu Wang <vulab(a)iscas.ac.cn> --- sound/usb/endpoint.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 88760268fb55..3a2b2a309a71 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -615,9 +615,8 @@ static void release_urbs(struct snd_usb_endpoint *ep, int force) for (i = 0; i < ep->nurbs; i++) release_urb_ctx(&ep->urb[i]); - if (ep->syncbuf) - usb_free_coherent(ep->chip->dev, SYNC_URBS * 4, - ep->syncbuf, ep->sync_dma); + usb_free_coherent(ep->chip->dev, SYNC_URBS * 4, + ep->syncbuf, ep->sync_dma); ep->syncbuf = NULL; ep->nurbs = 0; -- 2.17.1

2 1

[PATCH 1/3] ASoC: samsung: Document DT bindings for Midas sound subsystem
by Sylwester Nawrocki 27 Jul '20

27 Jul '20

This patch adds documentation of DT biding for the Midas sound complex. Partially based on the *txt version by Simon Shields <simon(a)lineageos.org>. Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- .../bindings/sound/samsung,midas-audio.yaml | 105 +++++++++++++++++++++ 1 file changed, 105 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/samsung,midas-audio.yaml diff --git a/Documentation/devicetree/bindings/sound/samsung,midas-audio.yaml b/Documentation/devicetree/bindings/sound/samsung,midas-audio.yaml new file mode 100644 index 0000000..4a82829 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/samsung,midas-audio.yaml @@ -0,0 +1,105 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/samsung,midas-audio.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Samsung Midas audio complex with WM1811 codec + +maintainers: + - Sylwester Nawrocki <s.nawrocki(a)samsung.com> + +properties: + compatible: + const: samsung,midas-audio + + model: + $ref: /schemas/types.yaml#/definitions/string + description: The user-visible name of this sound complex. + + cpu: + type: object + properties: + sound-dai: + $ref: /schemas/types.yaml#/definitions/phandle-array + description: phandle to the I2S controller + + codec: + type: object + properties: + sound-dai: + $ref: /schemas/types.yaml#/definitions/phandle-array + description: phandle to the WM1811 CODEC node + + samsung,audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: | + List of the connections between audio components; each entry is + a pair of strings, the first being the connection's sink, the second + being the connection's source; valid names for sources and sinks are + the WM1811's pins (as documented in its binding), and the jacks + on the board: HP, SPK, Main Mic, Sub Mic, Headset Mic. + + mic-bias-supply: + description: Supply for the micbias on the Main microphone + + submic-bias-supply: + description: Supply for the micbias on the Sub microphone + + fm-sel-gpios: + description: GPIO pin for FM selection + + lineout-sel-gpios: + description: GPIO pin for line out selection + +required: + - compatible + - model + - cpu + - codec + - samsung,audio-routing + - mic-bias-supply + - submic-bias-supply + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + + sound { + compatible = "samsung,midas-audio"; + model = "Midas"; + + mic-bias-gpios = <&gpf1 7 GPIO_ACTIVE_HIGH>; + submic-bias-gpios = <&gpf2 0 GPIO_ACTIVE_HIGH>; + + mic-bias-supply = <&mic_bias_reg>; + submic-bias-supply = <&submic_bias_reg>; + + samsung,audio-routing = + "HP", "HPOUT1L", + "HP", "HPOUT1R", + + "SPK", "SPKOUTLN", + "SPK", "SPKOUTLP", + "SPK", "SPKOUTRN", + "SPK", "SPKOUTRP", + + "RCV", "HPOUT2N", + "RCV", "HPOUT2P", + + "IN1LP", "Main Mic", + "IN1LN", "Main Mic", + "IN1RP", "Sub Mic", + "IN1LP", "Sub Mic"; + + cpu { + sound-dai = <&i2s0>; + }; + + codec { + sound-dai = <&wm1811>; + }; + + }; -- 2.7.4

3 6

[PATCH 0/2] ASoC: intel: use asoc_substream_to_rtd()
by Kuninori Morimoto 27 Jul '20

27 Jul '20

Hi Mark I have posted "ASoC: add asoc_substream_to_rtd() macro" patch-set to ALSA SoC ML (= see Link), and then Pierre-Louis wanted that Intel patch was separated for boards. The patches which are not for Intel were already accepted. This is for Intel, and Intel boards. Link: https://lore.kernel.org/r/87y2nf0yw2.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/877duz0ysw.wl-kuninori.morimoto.gx@renesas.com Kuninori Morimoto (2): ASoC: intel/boards: use asoc_substream_to_rtd() ASoC: intel: use asoc_substream_to_rtd() sound/soc/intel/atom/sst-mfld-platform-pcm.c | 6 +++--- sound/soc/intel/baytrail/sst-baytrail-pcm.c | 16 ++++++++-------- sound/soc/intel/boards/bdw-rt5650.c | 2 +- sound/soc/intel/boards/bdw-rt5677.c | 4 ++-- sound/soc/intel/boards/broadwell.c | 2 +- sound/soc/intel/boards/bxt_rt298.c | 2 +- sound/soc/intel/boards/byt-rt5640.c | 2 +- sound/soc/intel/boards/bytcht_da7213.c | 4 ++-- sound/soc/intel/boards/bytcr_rt5640.c | 2 +- sound/soc/intel/boards/bytcr_rt5651.c | 2 +- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 2 +- sound/soc/intel/boards/cht_bsw_nau8824.c | 2 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 2 +- sound/soc/intel/boards/cht_bsw_rt5672.c | 2 +- sound/soc/intel/boards/cml_rt1011_rt5682.c | 4 ++-- sound/soc/intel/boards/ehl_rt5660.c | 2 +- sound/soc/intel/boards/glk_rt5682_max98357a.c | 2 +- sound/soc/intel/boards/haswell.c | 2 +- sound/soc/intel/boards/kbl_da7219_max98927.c | 8 ++++---- sound/soc/intel/boards/kbl_rt5660.c | 2 +- sound/soc/intel/boards/kbl_rt5663_max98927.c | 4 ++-- .../intel/boards/kbl_rt5663_rt5514_max98927.c | 4 ++-- sound/soc/intel/boards/skl_nau88l25_max98357a.c | 2 +- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 2 +- sound/soc/intel/boards/skl_rt286.c | 2 +- sound/soc/intel/boards/sof_da7219_max98373.c | 2 +- sound/soc/intel/boards/sof_maxim_common.c | 4 ++-- sound/soc/intel/boards/sof_pcm512x.c | 4 ++-- sound/soc/intel/boards/sof_rt5682.c | 4 ++-- sound/soc/intel/boards/sof_sdw_rt1308.c | 2 +- sound/soc/intel/boards/sof_wm8804.c | 2 +- sound/soc/intel/haswell/sst-haswell-pcm.c | 12 ++++++------ sound/soc/intel/keembay/kmb_platform.c | 2 +- sound/soc/intel/skylake/skl-pcm.c | 8 ++++---- 34 files changed, 62 insertions(+), 62 deletions(-) -- 2.25.1

2 3

[PATCH] ASoC: dt-bindings: q6asm: Add Q6ASM_DAI_{TX_RX, TX, RX} defines
by Stephan Gerhold 27 Jul '20

27 Jul '20

Right now the direction of a DAI has to be specified as a literal number in the device tree, e.g.: dai@0 { reg = <0>; direction = <2>; }; but this does not make it immediately clear that this is a playback/RX-only DAI. Actually, q6asm-dai.c has useful defines for this. Move them to the dt-bindings header to allow using them in the dts(i) files. The example above then becomes: dai@0 { reg = <0>; direction = <Q6ASM_DAI_RX>; }; which is immediately recognizable as playback/RX-only DAI. Cc: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Signed-off-by: Stephan Gerhold <stephan(a)gerhold.net> --- Documentation/devicetree/bindings/sound/qcom,q6asm.txt | 9 +++++---- include/dt-bindings/sound/qcom,q6asm.h | 4 ++++ sound/soc/qcom/qdsp6/q6asm-dai.c | 3 --- 3 files changed, 9 insertions(+), 7 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/qcom,q6asm.txt b/Documentation/devicetree/bindings/sound/qcom,q6asm.txt index 6b9a88d0ea3f..8c4883becae9 100644 --- a/Documentation/devicetree/bindings/sound/qcom,q6asm.txt +++ b/Documentation/devicetree/bindings/sound/qcom,q6asm.txt @@ -39,9 +39,9 @@ configuration of each dai. Must contain the following properties. Usage: Required for Compress offload dais Value type: <u32> Definition: Specifies the direction of the dai stream - 0 for both tx and rx - 1 for only tx (Capture/Encode) - 2 for only rx (Playback/Decode) + Q6ASM_DAI_TX_RX (0) for both tx and rx + Q6ASM_DAI_TX (1) for only tx (Capture/Encode) + Q6ASM_DAI_RX (2) for only rx (Playback/Decode) - is-compress-dai: Usage: Required for Compress offload dais @@ -50,6 +50,7 @@ configuration of each dai. Must contain the following properties. = EXAMPLE +#include <dt-bindings/sound/qcom,q6asm.h> apr-service@7 { compatible = "qcom,q6asm"; @@ -62,7 +63,7 @@ apr-service@7 { dai@0 { reg = <0>; - direction = <2>; + direction = <Q6ASM_DAI_RX>; is-compress-dai; }; }; diff --git a/include/dt-bindings/sound/qcom,q6asm.h b/include/dt-bindings/sound/qcom,q6asm.h index 1eb77d87c2e8..f59d74f14395 100644 --- a/include/dt-bindings/sound/qcom,q6asm.h +++ b/include/dt-bindings/sound/qcom,q6asm.h @@ -19,4 +19,8 @@ #define MSM_FRONTEND_DAI_MULTIMEDIA15 14 #define MSM_FRONTEND_DAI_MULTIMEDIA16 15 +#define Q6ASM_DAI_TX_RX 0 +#define Q6ASM_DAI_TX 1 +#define Q6ASM_DAI_RX 2 + #endif /* __DT_BINDINGS_Q6_ASM_H__ */ diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c index a2acb7564eb8..9b7b218f2a20 100644 --- a/sound/soc/qcom/qdsp6/q6asm-dai.c +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c @@ -37,9 +37,6 @@ #define COMPR_PLAYBACK_MAX_FRAGMENT_SIZE (128 * 1024) #define COMPR_PLAYBACK_MIN_NUM_FRAGMENTS (4) #define COMPR_PLAYBACK_MAX_NUM_FRAGMENTS (16 * 4) -#define Q6ASM_DAI_TX_RX 0 -#define Q6ASM_DAI_TX 1 -#define Q6ASM_DAI_RX 2 #define ALAC_CH_LAYOUT_MONO ((101 << 16) | 1) #define ALAC_CH_LAYOUT_STEREO ((101 << 16) | 2) -- 2.27.0

3 2

[PATCH] ASoC: AMD: Restore PME_EN state at Power On
by Akshu Agrawal 27 Jul '20

27 Jul '20

PME_EN state needs to restored to the value set by fmw. For the devices which are not using I2S wake event which gets enabled by PME_EN bit, keeping PME_EN enabled burns considerable amount of power as it blocks low power state. For the devices using I2S wake event, PME_EN gets enabled in fmw and the state should be maintained after ACP Power On. Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- sound/soc/amd/raven/pci-acp3x.c | 21 +++++++++++++-------- 1 file changed, 13 insertions(+), 8 deletions(-) diff --git a/sound/soc/amd/raven/pci-acp3x.c b/sound/soc/amd/raven/pci-acp3x.c index ebf4388b6262..31b797c8bfe6 100644 --- a/sound/soc/amd/raven/pci-acp3x.c +++ b/sound/soc/amd/raven/pci-acp3x.c @@ -19,10 +19,12 @@ struct acp3x_dev_data { bool acp3x_audio_mode; struct resource *res; struct platform_device *pdev[ACP3x_DEVS]; + u32 pme_en; }; -static int acp3x_power_on(void __iomem *acp3x_base) +static int acp3x_power_on(struct acp3x_dev_data *adata) { + void __iomem *acp3x_base = adata->acp3x_base; u32 val; int timeout; @@ -39,10 +41,10 @@ static int acp3x_power_on(void __iomem *acp3x_base) while (++timeout < 500) { val = rv_readl(acp3x_base + mmACP_PGFSM_STATUS); if (!val) { - /* Set PME_EN as after ACP power On, - * PME_EN gets cleared + /* ACP power On clears PME_EN. + * Restore the value to its prior state */ - rv_writel(0x1, acp3x_base + mmACP_PME_EN); + rv_writel(adata->pme_en, acp3x_base + mmACP_PME_EN); return 0; } udelay(1); @@ -74,12 +76,13 @@ static int acp3x_reset(void __iomem *acp3x_base) return -ETIMEDOUT; } -static int acp3x_init(void __iomem *acp3x_base) +static int acp3x_init(struct acp3x_dev_data *adata) { + void __iomem *acp3x_base = adata->acp3x_base; int ret; /* power on */ - ret = acp3x_power_on(acp3x_base); + ret = acp3x_power_on(adata); if (ret) { pr_err("ACP3x power on failed\n"); return ret; @@ -151,7 +154,9 @@ static int snd_acp3x_probe(struct pci_dev *pci, } pci_set_master(pci); pci_set_drvdata(pci, adata); - ret = acp3x_init(adata->acp3x_base); + /* Save ACP_PME_EN state */ + adata->pme_en = rv_readl(adata->acp3x_base + mmACP_PME_EN); + ret = acp3x_init(adata); if (ret) goto disable_msi; @@ -274,7 +279,7 @@ static int snd_acp3x_resume(struct device *dev) struct acp3x_dev_data *adata; adata = dev_get_drvdata(dev); - ret = acp3x_init(adata->acp3x_base); + ret = acp3x_init(adata); if (ret) { dev_err(dev, "ACP init failed\n"); return ret; -- 2.20.1

2 1

[PATCH] ALSA: hda/realtek: Fix add a "ultra_low_power" function for intel reference board (alc256)
by Takashi Iwai 27 Jul '20

27 Jul '20

From: PeiSen Hou <pshou(a)realtek.com> Intel requires to enable power saving mode for intel reference board (alc256) Signed-off-by: PeiSen Hou <pshou(a)realtek.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3f2512942daf..0f640d99a396 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7590,7 +7590,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x10cf, 0x1629, "Lifebook U7x7", ALC255_FIXUP_LIFEBOOK_U7x7_HEADSET_MIC), SND_PCI_QUIRK(0x10cf, 0x1845, "Lifebook U904", ALC269_FIXUP_LIFEBOOK_EXTMIC), SND_PCI_QUIRK(0x10ec, 0x10f2, "Intel Reference board", ALC700_FIXUP_INTEL_REFERENCE), - SND_PCI_QUIRK(0x10ec, 0x1230, "Intel Reference board", ALC225_FIXUP_HEADSET_JACK), + SND_PCI_QUIRK(0x10ec, 0x1230, "Intel Reference board", ALC295_FIXUP_CHROME_BOOK), SND_PCI_QUIRK(0x10f7, 0x8338, "Panasonic CF-SZ6", ALC269_FIXUP_HEADSET_MODE), SND_PCI_QUIRK(0x144d, 0xc109, "Samsung Ativ book 9 (NP900X3G)", ALC269_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x144d, 0xc169, "Samsung Notebook 9 Pen (NP930SBE-K01US)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), -- 2.16.4

1 0

[PATCH] alsaucm: Fix ending with quotes commands
by Cheng Yueh 27 Jul '20

27 Jul '20

If we use 'alsaucm -n -b -' with 'get "CapturePCM/Internal Mic"' then the alsaucm will report error and stop immediately. The reason is that the parse_line in usecase.c appends an empty argument if a command ends with quotes. This change adds a patch to fix the parse_line function in usecase.c. Signed-off-by: Cheng Yueh <cyueh(a)chromium.org> --- alsaucm/usecase.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/alsaucm/usecase.c b/alsaucm/usecase.c index 1ed91ea..d39a159 100644 --- a/alsaucm/usecase.c +++ b/alsaucm/usecase.c @@ -130,6 +130,8 @@ static int parse_line(struct context *context, char *line) *line == '\n')) line++; c = *line; + if (c == '\0') + return 0; if (c == '\"' || c == '\'') { start = ++line; while (*line && *line != c) -- 2.26.2

2 1

snd_soc_component_driver substream ops
by Daniel Baluta 27 Jul '20

27 Jul '20

Hi Morimoto-san, Looking at snd_soc_component_driver I see there are some operations like: open, close, hw_params, hw_free. (1) Now, snd_soc_component_driver has snd_compress_ops. Do you think it is worth it to group operations from (1) in a similar structure maybe snd_<xyz>_ops. Daniel.

2 2

Acer TravelMate 5735Z (Realtek ALC272X): Periodic crackling noises
by Paul Menzel 27 Jul '20

27 Jul '20

Dear Linux folks, On Acer TravelMate 5735Z with Debian Sid/unstable and Linux 5.7.6, there are recurring crackling noises on the internal speaker. I ran `alsa-info` from the Debian package, and the logs are uploaded [1]. It’d be awesome, if you took a look, and told me, what I can try to solve this. Kind regards, Paul [1]: https://alsa-project.org/db/?f=67bde144f2e2a4f06da89a8cc8abf93da8c91d5b

1 0

No sound with Realtek ALC287
by GitHub issues - opened 27 Jul '20

27 Jul '20

alsa-project/alsa-lib issue #76 was opened from BLuFeNiX: I get no sound via the speakers of my Lenovo Legion 7i laptop, which alsamixer tells me is using a Realtek ALC287. Here is my alsa-info.sh output: http://alsa-project.org/db/?f=4272343a3590cc08f192f98113dedfc0418afe52 I made sure to disable Auto-Mute in alsamixer, and turn all volume levels to maximum. In all cases, I get no sounds from the speakers (running speaker-test, playing music, etc.). I *am* able to get sound via headphones and HDMI (though I believe HDMI is via a different sound card). I previously reported this here: https://bugzilla.kernel.org/show_bug.cgi?id=208555 but I received no response in almost 3 weeks, so I think maybe I have reported it to the wrong place. Please let me know if I can provide any assistance or additional information. Issue URL : https://github.com/alsa-project/alsa-lib/issues/76 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[RESEND 3/3] ASoC: max98390: update dsm param bin max size
by Steve Lee 27 Jul '20

27 Jul '20

MAX98390_DSM_PARAM_MAX_SIZE is changed to support extended register update. Signed-off-by: Steve Lee <steves.lee(a)maximintegrated.com> --- sound/soc/codecs/max98390.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/max98390.h b/sound/soc/codecs/max98390.h index 5f444e7779b0..dff884f68e3e 100644 --- a/sound/soc/codecs/max98390.h +++ b/sound/soc/codecs/max98390.h @@ -650,7 +650,7 @@ /* DSM register offset */ #define MAX98390_DSM_PAYLOAD_OFFSET 16 -#define MAX98390_DSM_PARAM_MAX_SIZE 770 +#define MAX98390_DSM_PARAM_MAX_SIZE 1024 #define MAX98390_DSM_PARAM_MIN_SIZE 670 struct max98390_priv { -- 2.17.1

1 0

[RESEND 2/3] ASoC: max98390: Update dsm init sequence and condition.
by Steve Lee 27 Jul '20

27 Jul '20

Modify dsm_init sequence and dsm param bin check condition. - Move dsm_init() to after amp init setting to make sure dsm init is last setting. - dsm param bin check condition changed for extended register setting. Signed-off-by: Steve Lee <steves.lee(a)maximintegrated.com> --- sound/soc/codecs/max98390.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/max98390.c b/sound/soc/codecs/max98390.c index 44ffebac15ad..ff5cc9bbec29 100644 --- a/sound/soc/codecs/max98390.c +++ b/sound/soc/codecs/max98390.c @@ -790,7 +790,7 @@ static int max98390_dsm_init(struct snd_soc_component *component) param_start_addr = (dsm_param[0] & 0xff) | (dsm_param[1] & 0xff) << 8; param_size = (dsm_param[2] & 0xff) | (dsm_param[3] & 0xff) << 8; if (param_size > MAX98390_DSM_PARAM_MAX_SIZE || - param_start_addr < DSM_STBASS_HPF_B0_BYTE0 || + param_start_addr < MAX98390_IRQ_CTRL || fw->size < param_size + MAX98390_DSM_PAYLOAD_OFFSET) { dev_err(component->dev, "param fw is invalid.\n"); @@ -864,11 +864,11 @@ static int max98390_probe(struct snd_soc_component *component) regmap_write(max98390->regmap, MAX98390_SOFTWARE_RESET, 0x01); /* Sleep reset settle time */ msleep(20); - /* Update dsm bin param */ - max98390_dsm_init(component); /* Amp init setting */ max98390_init_regs(component); + /* Update dsm bin param */ + max98390_dsm_init(component); /* Dsm Setting */ if (max98390->ref_rdc_value) { -- 2.17.1

1 0

[RESEND 1/3] ASoC: max98390: Fix dac event dapm mixer.
by Steve Lee 27 Jul '20

27 Jul '20

Global EN register guide to off before AMP_EN register when amp disable sequence. - remove AMP_EN control before max98390_dac_event call Signed-off-by: Steve Lee <steves.lee(a)maximintegrated.com> --- sound/soc/codecs/max98390.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/max98390.c b/sound/soc/codecs/max98390.c index 3e8094241645..44ffebac15ad 100644 --- a/sound/soc/codecs/max98390.c +++ b/sound/soc/codecs/max98390.c @@ -678,7 +678,7 @@ static const struct snd_kcontrol_new max98390_dai_controls = static const struct snd_soc_dapm_widget max98390_dapm_widgets[] = { SND_SOC_DAPM_DAC_E("Amp Enable", "HiFi Playback", - MAX98390_R203A_AMP_EN, 0, 0, max98390_dac_event, + SND_SOC_NOPM, 0, 0, max98390_dac_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_MUX("DAI Sel Mux", SND_SOC_NOPM, 0, 0, &max98390_dai_controls), -- 2.17.1

1 0

headphones detected but don't show up in alsa mixer, no audio on ubuntu 18.04 with gigabyte z370 gaming wifi motherboard
by GitHub issues - opened 27 Jul '20

27 Jul '20

alsa-project/alsa-lib issue #75 was opened from rbavery: Here is my ALSA info: http://alsa-project.org/db/?f=dd3d1d8a9fa0020900a0adc03bc6adfaf1f77ee5 I tried following these remedies (including reinstalling alsa, reloading, restarting computer) but nothing has panned out: https://support.system76.com/articles/audio/ Any help or suggestions are appreciated Issue URL : https://github.com/alsa-project/alsa-lib/issues/75 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH 14216/14216] ALSA: usb-audio: This patch for prevent auto wakeup from s3 trig by usb disconnect signal from Lenovo Thinkcentre TI024Gen3 USB-audio.
by penghao 25 Jul '20

25 Jul '20

From: "penghao(a)deepin.com" <penghao(a)deepin.com> TI024Gen3 USB-audio is controlled by TI024Gen3,when TI024Gens enter sleep mode, USB-audio will disconnect from USB bus port, so disabled the /sys/bus/usb/*/power/wakeup Fixesimmediately wakup form s3 state Signed-off-by: penghao(a)uniontech.com <penghao(a)uniontech.com> Signed-off-by: penghao(a)deepin.com <penghao(a)deepin.com> --- sound/usb/card.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/usb/card.c b/sound/usb/card.c index 2644a5ae2b75..969c3809e051 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -640,6 +640,12 @@ static int usb_audio_probe(struct usb_interface *intf, } } dev_set_drvdata(&dev->dev, chip); + /* + * ALSA: usb-audio: Add prevent wakeup from s3 state trig by lenovo + * ThinkCentre TI024Gen3 USB-audio + */ + if ((usb_id->idVendor == 0x17ef) && (usb_id->idProduct == 0xa012)) + device_set_wakeup_enable(&dev->dev, 0); /* * For devices with more than one control interface, we assume the @@ -914,3 +920,4 @@ static struct usb_driver usb_audio_driver = { }; module_usb_driver(usb_audio_driver); + -- 2.11.0

3 3

[PATCH] ASoC: dapm: don't call pm_runtime_* on card device
by Tzung-Bi Shih 24 Jul '20

24 Jul '20

runtime_usage of sound card has been observed to grow without bound. For example: $ cat /sys/devices/platform/sound/power/runtime_usage 46 $ sox -n -t s16 -r 48000 -c 2 - synth 1 sine 440 vol 0.1 | \ aplay -q -D hw:0,0 -f S16_LE -r 48000 -c 2 $ cat /sys/devices/platform/sound/power/runtime_usage 52 Commit 4e872a46823c ("ASoC: dapm: Don't force card bias level to be updated") stops to force update bias_level on card. If card doesn't provide set_bias_level callback, the snd_soc_dapm_set_bias_level() is equivalent to NOP for card device. As a result, dapm_pre_sequence_async() doesn't change the bias_level of card device correctly. Thus, pm_runtime_get_sync() would be called in dapm_pre_sequence_async() without symmetric pm_runtime_put() in dapm_post_sequence_async(). Don't call pm_runtime_* on card device. Signed-off-by: Tzung-Bi Shih <tzungbi(a)google.com> --- sound/soc/soc-dapm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) Hi, I'm not very sure if the patch makes sense. I was considering: (1) update the bias_level of card device correctly (2) skip pm_runtime_* for card device Per commit message on 4e872a46823c, I believe there is some context I'm not aware of it. Thus, the patch adopts (2). diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 5076299abf37..3273161e2787 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1798,7 +1798,7 @@ static void dapm_pre_sequence_async(void *data, async_cookie_t cookie) /* If we're off and we're not supposed to go into STANDBY */ if (d->bias_level == SND_SOC_BIAS_OFF && d->target_bias_level != SND_SOC_BIAS_OFF) { - if (d->dev) + if (d->dev && cookie) pm_runtime_get_sync(d->dev); ret = snd_soc_dapm_set_bias_level(d, SND_SOC_BIAS_STANDBY); @@ -1845,7 +1845,7 @@ static void dapm_post_sequence_async(void *data, async_cookie_t cookie) dev_err(d->dev, "ASoC: Failed to turn off bias: %d\n", ret); - if (d->dev) + if (d->dev && cookie) pm_runtime_put(d->dev); } -- 2.28.0.rc0.142.g3c755180ce-goog

2 6

[PATCH v1] ALSA: hda/realtek: typo_fix: enable headset mic of ASUS ROG Zephyrus G14(GA401) series with ALC289
by Armas Spann 24 Jul '20

24 Jul '20

This patch fixes a small typo I accidently submitted with the initial patch. The board should be named GA401 not G401. Signed-off-by: Armas Spann <zappel(a)retarded.farm> --- sound/pci/hda/patch_realtek.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1b2d8e56390a..2dd7150ff6da 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6152,7 +6152,7 @@ enum { ALC269VC_FIXUP_ACER_VCOPPERBOX_PINS, ALC269VC_FIXUP_ACER_HEADSET_MIC, ALC269VC_FIXUP_ACER_MIC_NO_PRESENCE, - ALC289_FIXUP_ASUS_G401, + ALC289_FIXUP_ASUS_GA401, ALC256_FIXUP_ACER_MIC_NO_PRESENCE, }; @@ -7363,7 +7363,7 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MIC }, - [ALC289_FIXUP_ASUS_G401] = { + [ALC289_FIXUP_ASUS_GA401] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { { 0x19, 0x03a11020 }, /* headset mic with jack detect */ @@ -7561,7 +7561,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1bbd, "ASUS Z550MA", ALC255_FIXUP_ASUS_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x1c23, "Asus X55U", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x1043, 0x1ccd, "ASUS X555UB", ALC256_FIXUP_ASUS_MIC), - SND_PCI_QUIRK(0x1043, 0x1f11, "ASUS Zephyrus G14", ALC289_FIXUP_ASUS_G401), + SND_PCI_QUIRK(0x1043, 0x1f11, "ASUS Zephyrus G14", ALC289_FIXUP_ASUS_GA401), SND_PCI_QUIRK(0x1043, 0x3030, "ASUS ZN270IE", ALC256_FIXUP_ASUS_AIO_GPIO2), SND_PCI_QUIRK(0x1043, 0x831a, "ASUS P901", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x834a, "ASUS S101", ALC269_FIXUP_STEREO_DMIC), -- 2.27.0

2 1

[PATCH v1] ALSA: hda/realtek: enable headset mic of ASUS ROG Zephyrus G15(GA502) series with ALC289
by Armas Spann 24 Jul '20

24 Jul '20

This patch adds support for headset mic to the ASUS ROG Zephyrus G15(GA502) notebook series by adding the corresponding vendor/pci_device id, as well as adding a new fixup for the used realtek ALC289. The fixup stets the correct pin to get the headset mic correctly recognized on audio-jack. Signed-off-by: Armas Spann <zappel(a)retarded.farm> --- sound/pci/hda/patch_realtek.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1b2d8e56390a..a2c48a7f4594 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6153,6 +6153,7 @@ enum { ALC269VC_FIXUP_ACER_HEADSET_MIC, ALC269VC_FIXUP_ACER_MIC_NO_PRESENCE, ALC289_FIXUP_ASUS_G401, + ALC289_FIXUP_ASUS_GA502, ALC256_FIXUP_ACER_MIC_NO_PRESENCE, }; @@ -7370,6 +7371,13 @@ static const struct hda_fixup alc269_fixups[] = { { } }, }, + [ALC289_FIXUP_ASUS_GA502] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x03a11020 }, /* headset mic with jack detect */ + { } + }, + }, [ALC256_FIXUP_ACER_MIC_NO_PRESENCE] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -7561,6 +7569,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1bbd, "ASUS Z550MA", ALC255_FIXUP_ASUS_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x1c23, "Asus X55U", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x1043, 0x1ccd, "ASUS X555UB", ALC256_FIXUP_ASUS_MIC), + SND_PCI_QUIRK(0x1043, 0x1e11, "ASUS Zephyrus G15", ALC289_FIXUP_ASUS_GA502), SND_PCI_QUIRK(0x1043, 0x1f11, "ASUS Zephyrus G14", ALC289_FIXUP_ASUS_G401), SND_PCI_QUIRK(0x1043, 0x3030, "ASUS ZN270IE", ALC256_FIXUP_ASUS_AIO_GPIO2), SND_PCI_QUIRK(0x1043, 0x831a, "ASUS P901", ALC269_FIXUP_STEREO_DMIC), -- 2.27.0

2 1

[PATCH] dt-bindings: sound: convert Everest ES8316 binding to yaml
by Katsuhiro Suzuki 24 Jul '20

24 Jul '20

This patch converts Everest Semiconductor ES8316 low power audio CODEC binding to DT schema. Signed-off-by: Katsuhiro Suzuki <katsuhiro(a)katsuster.net> --- .../bindings/sound/everest,es8316.txt | 23 --------- .../bindings/sound/everest,es8316.yaml | 49 +++++++++++++++++++ 2 files changed, 49 insertions(+), 23 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/everest,es8316.txt create mode 100644 Documentation/devicetree/bindings/sound/everest,es8316.yaml diff --git a/Documentation/devicetree/bindings/sound/everest,es8316.txt b/Documentation/devicetree/bindings/sound/everest,es8316.txt deleted file mode 100644 index 1bf03c5f2af4..000000000000 --- a/Documentation/devicetree/bindings/sound/everest,es8316.txt +++ /dev/null @@ -1,23 +0,0 @@ -Everest ES8316 audio CODEC - -This device supports both I2C and SPI. - -Required properties: - - - compatible : should be "everest,es8316" - - reg : the I2C address of the device for I2C - -Optional properties: - - - clocks : a list of phandle, should contain entries for clock-names - - clock-names : should include as follows: - "mclk" : master clock (MCLK) of the device - -Example: - -es8316: codec@11 { - compatible = "everest,es8316"; - reg = <0x11>; - clocks = <&clks 10>; - clock-names = "mclk"; -}; diff --git a/Documentation/devicetree/bindings/sound/everest,es8316.yaml b/Documentation/devicetree/bindings/sound/everest,es8316.yaml new file mode 100644 index 000000000000..b713404dac4f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/everest,es8316.yaml @@ -0,0 +1,49 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/everest,es8316.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Everest ES8316 audio CODEC + +maintainers: + - Mark Brown <broonie(a)kernel.org> + +properties: + compatible: + const: everest,es8316 + + reg: + maxItems: 1 + + clocks: + items: + - description: clock for master clock (MCLK) + + clock-names: + items: + - const: mclk + + "#sound-dai-cells": + const: 0 + +required: + - compatible + - reg + - "#sound-dai-cells" + +additionalProperties: false + +examples: + - | + i2c0 { + #address-cells = <1>; + #size-cells = <0>; + es8316: codec@11 { + compatible = "everest,es8316"; + reg = <0x11>; + clocks = <&clks 10>; + clock-names = "mclk"; + #sound-dai-cells = <0>; + }; + }; -- 2.27.0

2 2

[PATCH] ASoC: soc-component: don't report of_xlate_dai_name failures
by Jerome Brunet 24 Jul '20

24 Jul '20

With commit e2329eeba45f ("ASoC: soc-component: add soc_component_err()") every error different for ENOTSUPP or EPROBE_DEFER will log an error. However, as explained in snd_soc_get_dai_name(), this callback may error to indicate that the DAI is not matched by the component tested. If the device provides other components, those may still match. Logging an error in this case is misleading. Don't use soc_component_ret() in snd_soc_component_of_xlate_dai_name() to avoid spamming the log. Fixes: e2329eeba45f ("ASoC: soc-component: add soc_component_err()") Cc: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> --- If we wish to report an error when no name is found for dai phandle, I think it would be more approriate to it with with "struct device *" of the card, which is the one actually failing to get what it needs. These user of snd_soc_get_dai_name() are already reporting the error: * sound/soc/samsung/tm2_wm5110.c * sound/soc/rockchip/rk3288_hdmi_analog.c These could be silently failing: * Simple cards * Amlogic's cards * Samsumg odroid card sound/soc/soc-component.c | 14 ++++++++------ 1 file changed, 8 insertions(+), 6 deletions(-) diff --git a/sound/soc/soc-component.c b/sound/soc/soc-component.c index 9565a0dd7cb6..fb6479b22d26 100644 --- a/sound/soc/soc-component.c +++ b/sound/soc/soc-component.c @@ -344,13 +344,15 @@ int snd_soc_component_of_xlate_dai_name(struct snd_soc_component *component, struct of_phandle_args *args, const char **dai_name) { - int ret = -ENOTSUPP; - if (component->driver->of_xlate_dai_name) - ret = component->driver->of_xlate_dai_name(component, - args, dai_name); - - return soc_component_ret(component, ret); + return component->driver->of_xlate_dai_name(component, + args, dai_name); + /* + * Don't use soc_component_ret here because we may not want to report + * the error just yet. If a device has more than one component, the + * first may not match and we don't want spam the log with this. + */ + return -ENOTSUPP; } void snd_soc_component_setup_regmap(struct snd_soc_component *component) -- 2.25.4

2 1

[PATCH 0/7] ASoC: qcom: Use qcom_snd_parse_of() for apq8016_sbc
by Stephan Gerhold 24 Jul '20

24 Jul '20

At the moment we have two separate functions to parse the sound card properties from the device tree: qcom_snd_parse_of() for DPCM and apq8016_sbc_parse_of() without DPCM. These functions are almost identical except for a few minor differences. This patch set extends qcom_snd_parse_of() to handle links without DPCM, so that we can use one common function for all (qcom) machine drivers. Stephan Gerhold (7): ASoC: qcom: Use devm for resource management ASoC: qcom: common: Use snd_soc_dai_link_set_capabilities() ASoC: q6afe: Remove unused q6afe_is_rx_port() function ASoC: qcom: common: Support parsing links without DPCM ASoC: qcom: common: Parse properties with "qcom," prefix ASoC: qcom: apq8016_sbc: Use qcom_snd_parse_of() ASoC: qcom: common: Avoid printing errors for -EPROBE_DEFER sound/soc/qcom/Kconfig | 1 + sound/soc/qcom/apq8016_sbc.c | 120 ++++------------------------------- sound/soc/qcom/apq8096.c | 28 +------- sound/soc/qcom/common.c | 58 ++++++++++------- sound/soc/qcom/qdsp6/q6afe.c | 8 --- sound/soc/qcom/qdsp6/q6afe.h | 1 - sound/soc/qcom/sdm845.c | 40 ++---------- 7 files changed, 59 insertions(+), 197 deletions(-) -- 2.27.0

3 9

[PATCH] soundwire: master: enable pm runtime
by Bard Liao 24 Jul '20

24 Jul '20

We should enable pm runtime. Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- drivers/soundwire/master.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/drivers/soundwire/master.c b/drivers/soundwire/master.c index 5f0b2189defe..3488bb824e84 100644 --- a/drivers/soundwire/master.c +++ b/drivers/soundwire/master.c @@ -154,6 +154,7 @@ int sdw_master_device_add(struct sdw_bus *bus, struct device *parent, bus->dev = &md->dev; bus->md = md; + pm_runtime_enable(&bus->md->dev); device_register_err: return ret; } @@ -166,6 +167,7 @@ int sdw_master_device_add(struct sdw_bus *bus, struct device *parent, */ int sdw_master_device_del(struct sdw_bus *bus) { + pm_runtime_disable(&bus->md->dev); device_unregister(bus->dev); return 0; -- 2.17.1

3 2

[PATCH 2/2] ASoC: max98390: Update dsm init sequence and condition.
by Steve Lee 24 Jul '20

24 Jul '20

Modify dsm_init sequence and dsm param bin check condition. - Move dsm_init() to after amp init setting to make sure dsm init is last setting. - dsm param bin check condition changed for extended register setting. Signed-off-by: Steve Lee <steves.lee(a)maximintegrated.com> --- sound/soc/codecs/max98390.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/max98390.c b/sound/soc/codecs/max98390.c index 44ffebac15ad..ff5cc9bbec29 100644 --- a/sound/soc/codecs/max98390.c +++ b/sound/soc/codecs/max98390.c @@ -790,7 +790,7 @@ static int max98390_dsm_init(struct snd_soc_component *component) param_start_addr = (dsm_param[0] & 0xff) | (dsm_param[1] & 0xff) << 8; param_size = (dsm_param[2] & 0xff) | (dsm_param[3] & 0xff) << 8; if (param_size > MAX98390_DSM_PARAM_MAX_SIZE || - param_start_addr < DSM_STBASS_HPF_B0_BYTE0 || + param_start_addr < MAX98390_IRQ_CTRL || fw->size < param_size + MAX98390_DSM_PAYLOAD_OFFSET) { dev_err(component->dev, "param fw is invalid.\n"); @@ -864,11 +864,11 @@ static int max98390_probe(struct snd_soc_component *component) regmap_write(max98390->regmap, MAX98390_SOFTWARE_RESET, 0x01); /* Sleep reset settle time */ msleep(20); - /* Update dsm bin param */ - max98390_dsm_init(component); /* Amp init setting */ max98390_init_regs(component); + /* Update dsm bin param */ + max98390_dsm_init(component); /* Dsm Setting */ if (max98390->ref_rdc_value) { -- 2.17.1

1 0

[PATCH 1/2] ASoC: max98390: Fix dac event dapm mixer.
by Steve Lee 24 Jul '20

24 Jul '20

Global EN register guide to off before AMP_EN register when amp disable sequence. - remove AMP_EN control before max98390_dac_event call Signed-off-by: Steve Lee <steves.lee(a)maximintegrated.com> --- sound/soc/codecs/max98390.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/max98390.c b/sound/soc/codecs/max98390.c index 3e8094241645..44ffebac15ad 100644 --- a/sound/soc/codecs/max98390.c +++ b/sound/soc/codecs/max98390.c @@ -678,7 +678,7 @@ static const struct snd_kcontrol_new max98390_dai_controls = static const struct snd_soc_dapm_widget max98390_dapm_widgets[] = { SND_SOC_DAPM_DAC_E("Amp Enable", "HiFi Playback", - MAX98390_R203A_AMP_EN, 0, 0, max98390_dac_event, + SND_SOC_NOPM, 0, 0, max98390_dac_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_MUX("DAI Sel Mux", SND_SOC_NOPM, 0, 0, &max98390_dai_controls), -- 2.17.1

1 0

[PATCH 1/2] ALSA: hda/realtek: Fix headset mic on Loongson platform
by Kaige Li 24 Jul '20

24 Jul '20

Add pin quirks to enable use of the headset mic on Loongson platform. Signed-off-by: Kaige Li <likaige(a)loongson.cn> --- sound/pci/hda/patch_realtek.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 4c7e191..b227be3 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6041,6 +6041,7 @@ enum { ALC290_FIXUP_SUBWOOFER_HSJACK, ALC269_FIXUP_THINKPAD_ACPI, ALC269_FIXUP_DMIC_THINKPAD_ACPI, + ALC269_FIXUP_LOONGSON_HDA, ALC255_FIXUP_ACER_MIC_NO_PRESENCE, ALC255_FIXUP_ASUS_MIC_NO_PRESENCE, ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, @@ -6381,6 +6382,14 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC }, + [ALC269_FIXUP_LOONGSON_HDA] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x1b, 0x02214c40 }, /* Front Mic */ + { 0x15, 0x01014030 }, /* Rear Mic */ + { } + }, + }, [ALC269_FIXUP_DELL4_MIC_NO_PRESENCE] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -7654,6 +7663,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K), SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD), SND_PCI_QUIRK(0x19e5, 0x3204, "Huawei MACH-WX9", ALC256_FIXUP_HUAWEI_MACH_WX9_PINS), + SND_PCI_QUIRK(0x10ec, 0x0269, "Loongson HDA", ALC269_FIXUP_LOONGSON_HDA), SND_PCI_QUIRK(0x1b7d, 0xa831, "Ordissimo EVE2 ", ALC269VB_FIXUP_ORDISSIMO_EVE2), /* Also known as Malata PC-B1303 */ SND_PCI_QUIRK(0x1d72, 0x1901, "RedmiBook 14", ALC256_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x10ec, 0x118c, "Medion EE4254 MD62100", ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE), -- 2.1.0

2 5

[PATCH] ASoC: Intel: boards: eve: Fix DMIC records zero
by vamshi.krishna.gopal＠intel.com 24 Jul '20

24 Jul '20

From: Brent Lu <brent.lu(a)intel.com> Add a dapm route to provide ssp mclk/sclk early for DMIC on SSP0(rt5514) and Headset on SSP1(rt5663) since sclk for both codecs are different the struct now defines SSP0 and SSP1 mclk , sclk separately This change ensures the DMIC PCM port will not return all-zero data Signed-off-by: Brent Lu <brent.lu(a)intel.com> Signed-off-by: Vamshi Krishna Gopal <vamshi.krishna.gopal(a)intel.com> --- .../intel/boards/kbl_rt5663_rt5514_max98927.c | 150 ++++++++++++------ 1 file changed, 102 insertions(+), 48 deletions(-) diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index b34cf6cf1139..584e4f9cedc2 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -53,8 +53,10 @@ struct kbl_codec_private { struct snd_soc_jack kabylake_headset; struct list_head hdmi_pcm_list; struct snd_soc_jack kabylake_hdmi[2]; - struct clk *mclk; - struct clk *sclk; + struct clk *ssp0_mclk; + struct clk *ssp0_sclk; + struct clk *ssp1_mclk; + struct clk *ssp1_sclk; }; enum { @@ -77,13 +79,31 @@ static const struct snd_kcontrol_new kabylake_controls[] = { }; static int platform_clock_control(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *k, int event) + struct snd_kcontrol *k, int event, int ssp_num) { struct snd_soc_dapm_context *dapm = w->dapm; struct snd_soc_card *card = dapm->card; struct kbl_codec_private *priv = snd_soc_card_get_drvdata(card); + struct clk *mclk, *sclk; + unsigned long sclk_rate; int ret = 0; + switch (ssp_num) { + case 0: + mclk = priv->ssp0_mclk; + sclk = priv->ssp0_sclk; + sclk_rate = 6144000; + break; + case 1: + mclk = priv->ssp1_mclk; + sclk = priv->ssp1_sclk; + sclk_rate = 3072000; + break; + default: + dev_err(card->dev, "Invalid ssp_num %d\n", ssp_num); + return -EINVAL; + } + /* * MCLK/SCLK need to be ON early for a successful synchronization of * codec internal clock. And the clocks are turned off during @@ -91,38 +111,46 @@ static int platform_clock_control(struct snd_soc_dapm_widget *w, */ switch (event) { case SND_SOC_DAPM_PRE_PMU: + if (__clk_is_enabled(mclk)) + return 0; + /* Enable MCLK */ - ret = clk_set_rate(priv->mclk, 24000000); + ret = clk_set_rate(mclk, 24000000); if (ret < 0) { - dev_err(card->dev, "Can't set rate for mclk, err: %d\n", - ret); + dev_err(card->dev, "Can't set rate for ssp%d_mclk, err: %d\n", + ssp_num, ret); return ret; } - ret = clk_prepare_enable(priv->mclk); + ret = clk_prepare_enable(mclk); if (ret < 0) { - dev_err(card->dev, "Can't enable mclk, err: %d\n", ret); + dev_err(card->dev, "Can't enable ssp%d_mclk, err: %d\n", + ssp_num, ret); return ret; } /* Enable SCLK */ - ret = clk_set_rate(priv->sclk, 3072000); + ret = clk_set_rate(sclk, sclk_rate); if (ret < 0) { - dev_err(card->dev, "Can't set rate for sclk, err: %d\n", - ret); - clk_disable_unprepare(priv->mclk); + dev_err(card->dev, "Can't set rate for ssp%d_sclk, err: %d\n", + ssp_num, ret); + clk_disable_unprepare(mclk); return ret; } - ret = clk_prepare_enable(priv->sclk); + ret = clk_prepare_enable(sclk); if (ret < 0) { - dev_err(card->dev, "Can't enable sclk, err: %d\n", ret); - clk_disable_unprepare(priv->mclk); + dev_err(card->dev, "Can't enable ssp%d_sclk, err: %d\n", + ssp_num, ret); + clk_disable_unprepare(mclk); } break; case SND_SOC_DAPM_POST_PMD: - clk_disable_unprepare(priv->mclk); - clk_disable_unprepare(priv->sclk); + if (!__clk_is_enabled(mclk)) + return 0; + + clk_disable_unprepare(mclk); + clk_disable_unprepare(sclk); break; default: return 0; @@ -131,6 +159,18 @@ static int platform_clock_control(struct snd_soc_dapm_widget *w, return 0; } +static int platform_clock_control_ssp0(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + return platform_clock_control(w, k, event, 0); +} + +static int platform_clock_control_ssp1(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + return platform_clock_control(w, k, event, 1); +} + static const struct snd_soc_dapm_widget kabylake_widgets[] = { SND_SOC_DAPM_HP("Headphone Jack", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), @@ -139,15 +179,17 @@ static const struct snd_soc_dapm_widget kabylake_widgets[] = { SND_SOC_DAPM_MIC("DMIC", NULL), SND_SOC_DAPM_SPK("HDMI1", NULL), SND_SOC_DAPM_SPK("HDMI2", NULL), - SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, - platform_clock_control, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_SUPPLY("Platform Clock SSP0", SND_SOC_NOPM, 0, 0, + platform_clock_control_ssp0, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SUPPLY("Platform Clock SSP1", SND_SOC_NOPM, 0, 0, + platform_clock_control_ssp1, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), - }; static const struct snd_soc_dapm_route kabylake_map[] = { /* Headphones */ - { "Headphone Jack", NULL, "Platform Clock" }, + { "Headphone Jack", NULL, "Platform Clock SSP1" }, { "Headphone Jack", NULL, "HPOL" }, { "Headphone Jack", NULL, "HPOR" }, @@ -156,7 +198,7 @@ static const struct snd_soc_dapm_route kabylake_map[] = { { "Right Spk", NULL, "Right BE_OUT" }, /* other jacks */ - { "Headset Mic", NULL, "Platform Clock" }, + { "Headset Mic", NULL, "Platform Clock SSP1" }, { "IN1P", NULL, "Headset Mic" }, { "IN1N", NULL, "Headset Mic" }, @@ -180,6 +222,7 @@ static const struct snd_soc_dapm_route kabylake_map[] = { { "ssp0 Rx", NULL, "Right HiFi Capture" }, /* DMIC */ + { "DMIC", NULL, "Platform Clock SSP0" }, { "DMIC1L", NULL, "DMIC" }, { "DMIC1R", NULL, "DMIC" }, { "DMIC2L", NULL, "DMIC" }, @@ -666,7 +709,7 @@ static int kabylake_set_bias_level(struct snd_soc_card *card, if (!component || strcmp(component->name, RT5514_DEV_NAME)) return 0; - if (IS_ERR(priv->mclk)) + if (IS_ERR(priv->ssp0_mclk)) return 0; /* @@ -678,17 +721,17 @@ static int kabylake_set_bias_level(struct snd_soc_card *card, case SND_SOC_BIAS_PREPARE: if (dapm->bias_level == SND_SOC_BIAS_ON) { dev_dbg(card->dev, "Disable mclk"); - clk_disable_unprepare(priv->mclk); + clk_disable_unprepare(priv->ssp0_mclk); } else { dev_dbg(card->dev, "Enable mclk"); - ret = clk_set_rate(priv->mclk, 24000000); + ret = clk_set_rate(priv->ssp0_mclk, 24000000); if (ret) { dev_err(card->dev, "Can't set rate for mclk, err: %d\n", ret); return ret; } - ret = clk_prepare_enable(priv->mclk); + ret = clk_prepare_enable(priv->ssp0_mclk); if (ret) { dev_err(card->dev, "Can't enable mclk, err: %d\n", ret); @@ -757,6 +800,29 @@ static struct snd_soc_card kabylake_audio_card = { .late_probe = kabylake_card_late_probe, }; +static int kabylake_audio_clk_get(struct device *dev, const char *id, + struct clk **clk) +{ + int ret = 0; + + if (!clk) + return -EINVAL; + + *clk = devm_clk_get(dev, id); + if (IS_ERR(*clk)) { + ret = PTR_ERR(*clk); + if (ret == -ENOENT) { + dev_info(dev, "Failed to get %s, defer probe\n", id); + return -EPROBE_DEFER; + } + + dev_err(dev, "Failed to get %s with err:%d\n", id, ret); + return ret; + } + + return ret; +} + static int kabylake_audio_probe(struct platform_device *pdev) { struct kbl_codec_private *ctx; @@ -777,33 +843,21 @@ static int kabylake_audio_probe(struct platform_device *pdev) dmic_constraints = mach->mach_params.dmic_num == 2 ? &constraints_dmic_2ch : &constraints_dmic_channels; - ctx->mclk = devm_clk_get(&pdev->dev, "ssp1_mclk"); - if (IS_ERR(ctx->mclk)) { - ret = PTR_ERR(ctx->mclk); - if (ret == -ENOENT) { - dev_info(&pdev->dev, - "Failed to get ssp1_mclk, defer probe\n"); - return -EPROBE_DEFER; - } + ret = kabylake_audio_clk_get(&pdev->dev, "ssp0_mclk", &ctx->ssp0_mclk); + if (ret != 0) + return ret; - dev_err(&pdev->dev, "Failed to get ssp1_mclk with err:%d\n", - ret); + ret = kabylake_audio_clk_get(&pdev->dev, "ssp0_sclk", &ctx->ssp0_sclk); + if (ret != 0) return ret; - } - ctx->sclk = devm_clk_get(&pdev->dev, "ssp1_sclk"); - if (IS_ERR(ctx->sclk)) { - ret = PTR_ERR(ctx->sclk); - if (ret == -ENOENT) { - dev_info(&pdev->dev, - "Failed to get ssp1_sclk, defer probe\n"); - return -EPROBE_DEFER; - } + ret = kabylake_audio_clk_get(&pdev->dev, "ssp1_mclk", &ctx->ssp1_mclk); + if (ret != 0) + return ret; - dev_err(&pdev->dev, "Failed to get ssp1_sclk with err:%d\n", - ret); + ret = kabylake_audio_clk_get(&pdev->dev, "ssp1_sclk", &ctx->ssp1_sclk); + if (ret != 0) return ret; - } return devm_snd_soc_register_card(&pdev->dev, &kabylake_audio_card); } -- 2.17.1

3 2

Alsa or alsamixer muting itself
by GitHub issues - opened 24 Jul '20

24 Jul '20

alsa-project/alsa-lib issue #74 was opened from allelopath: Alsa or alsamixer mutes the master track, after 10 or so minutes or on restart Dell Optiplex 7020 Desktop journalctrl attached [journalctl.txt](https://github.com/alsa-project/alsa-lib/files/4969120/jour… $ lspci 00:00.0 Host bridge: Intel Corporation 4th Gen Core Processor DRAM Controller (rev 06) 00:02.0 VGA compatible controller: Intel Corporation Xeon E3-1200 v3/4th Gen Core Processor Integrated Graphics Controller (rev 06) 00:03.0 Audio device: Intel Corporation Xeon E3-1200 v3/4th Gen Core Processor HD Audio Controller (rev 06) 00:14.0 USB controller: Intel Corporation 8 Series/C220 Series Chipset Family USB xHCI (rev 04) 00:16.0 Communication controller: Intel Corporation 8 Series/C220 Series Chipset Family MEI Controller #1 (rev 04) 00:16.3 Serial controller: Intel Corporation 8 Series/C220 Series Chipset Family KT Controller (rev 04) 00:19.0 Ethernet controller: Intel Corporation Ethernet Connection I217-LM (rev 04) 00:1a.0 USB controller: Intel Corporation 8 Series/C220 Series Chipset Family USB EHCI #2 (rev 04) 00:1b.0 Audio device: Intel Corporation 8 Series/C220 Series Chipset High Definition Audio Controller (rev 04) 00:1d.0 USB controller: Intel Corporation 8 Series/C220 Series Chipset Family USB EHCI #1 (rev 04) 00:1f.0 ISA bridge: Intel Corporation Q87 Express LPC Controller (rev 04) 00:1f.2 SATA controller: Intel Corporation 8 Series/C220 Series Chipset Family 6-port SATA Controller 1 [AHCI mode] (rev 04) 00:1f.3 SMBus: Intel Corporation 8 Series/C220 Series Chipset Family SMBus Controller (rev 04) Issue URL : https://github.com/alsa-project/alsa-lib/issues/74 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH] ASoC: tegra: Fix build error due to 64-by-32 division
by Sameer Pujar 23 Jul '20

23 Jul '20

Build errors are seen on 32-bit platforms because of a plain 64-by-32 division. For example, following build erros were reported. "ERROR: modpost: "__udivdi3" [sound/soc/tegra/snd-soc-tegra210-dmic.ko] undefined!" "ERROR: modpost: "__divdi3" [sound/soc/tegra/snd-soc-tegra210-dmic.ko] undefined!" This can be fixed by using div_u64() helper from 'math64.h' header. Fixes: 8c8ff982e9e2 ("ASoC: tegra: Add Tegra210 based DMIC driver") Reported-by: Geert Uytterhoeven <geert(a)linux-m68k.org> Reported-by: Randy Dunlap <rdunlap(a)infradead.org> Signed-off-by: Sameer Pujar <spujar(a)nvidia.com> --- sound/soc/tegra/tegra210_dmic.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/tegra/tegra210_dmic.c b/sound/soc/tegra/tegra210_dmic.c index ff6fd65..d682414 100644 --- a/sound/soc/tegra/tegra210_dmic.c +++ b/sound/soc/tegra/tegra210_dmic.c @@ -6,6 +6,7 @@ #include <linux/clk.h> #include <linux/device.h> +#include <linux/math64.h> #include <linux/module.h> #include <linux/of_device.h> #include <linux/platform_device.h> @@ -129,7 +130,7 @@ static int tegra210_dmic_hw_params(struct snd_pcm_substream *substream, * Boost Gain Volume control has 100x factor. */ if (dmic->boost_gain) - gain_q23 = (gain_q23 * dmic->boost_gain) / 100; + gain_q23 = div_u64(gain_q23 * dmic->boost_gain, 100); regmap_write(dmic->regmap, TEGRA210_DMIC_LP_FILTER_GAIN, (unsigned int)gain_q23); -- 2.7.4

3 2

[PATCH v2 1/2] dt-bindings: tas2562: Convert the tas2562 binding to yaml
by Dan Murphy 23 Jul '20

23 Jul '20

Convert the TAS2562 text file to yaml format. Signed-off-by: Dan Murphy <dmurphy(a)ti.com> --- v2 - Updated the shutdown-gpio to shutdown-gpios and fixed licensing to be GPL-2.0-only. .../devicetree/bindings/sound/tas2562.yaml | 69 +++++++++++++++++++ 1 file changed, 69 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/tas2562.yaml diff --git a/Documentation/devicetree/bindings/sound/tas2562.yaml b/Documentation/devicetree/bindings/sound/tas2562.yaml new file mode 100644 index 000000000000..8d75a798740b --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tas2562.yaml @@ -0,0 +1,69 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +# Copyright (C) 2019 Texas Instruments Incorporated +%YAML 1.2 +--- +$id: "http://devicetree.org/schemas/sound/tas2562.yaml#" +$schema: "http://devicetree.org/meta-schemas/core.yaml#" + +title: Texas Instruments TAS2562 Smart PA + +maintainers: + - Dan Murphy <dmurphy(a)ti.com> + +description: | + The TAS2562 is a mono, digital input Class-D audio amplifier optimized for + efficiently driving high peak power into small loudspeakers. + Integrated speaker voltage and current sense provides for + real time monitoring of loudspeaker behavior. + +properties: + compatible: + enum: + - ti,tas2562 + - ti,tas2563 + + reg: + maxItems: 1 + description: | + I2C address of the device can be one of these 0x4c, 0x4d, 0x4e or 0x4f + + shut-down-gpios: + description: GPIO used to control the state of the device. + deprecated: true + + shutdown-gpios: + description: GPIO used to control the state of the device. + + interrupts: + maxItems: 1 + + ti,imon-slot-no: + $ref: /schemas/types.yaml#/definitions/uint32 + description: TDM TX current sense time slot. + + '#sound-dai-cells': + const: 1 + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + i2c0 { + #address-cells = <1>; + #size-cells = <0>; + codec: codec@4c { + compatible = "ti,tas2562"; + reg = <0x4c>; + #sound-dai-cells = <1>; + interrupt-parent = <&gpio1>; + interrupts = <14>; + shutdown-gpios = <&gpio1 15 0>; + ti,imon-slot-no = <0>; + }; + }; + -- 2.27.0

2 4

[PATCH] ASoC: fsl_esai: add IRQF_SHARED for devm_request_irq
by Shengjiu Wang 23 Jul '20

23 Jul '20

ESAI interfaces may share same interrupt line with EDMA on some platforms (e.g. i.MX8QXP, i.MX8QM). Add IRQF_SHARED flag to allow sharing the irq among several devices Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> Signed-off-by: Viorel Suman <viorel.suman(a)nxp.com> --- sound/soc/fsl/fsl_esai.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index b8fbd7ba94af..4ae36099ae82 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -1012,7 +1012,7 @@ static int fsl_esai_probe(struct platform_device *pdev) if (irq < 0) return irq; - ret = devm_request_irq(&pdev->dev, irq, esai_isr, 0, + ret = devm_request_irq(&pdev->dev, irq, esai_isr, IRQF_SHARED, esai_priv->name, esai_priv); if (ret) { dev_err(&pdev->dev, "failed to claim irq %u\n", irq); -- 2.27.0

3 2

[PATCH] ASoC: omap: Replace HTTP links with HTTPS ones
by Alexander A. Klimov 23 Jul '20

23 Jul '20

Rationale: Reduces attack surface on kernel devs opening the links for MITM as HTTPS traffic is much harder to manipulate. Deterministic algorithm: For each file: If not .svg: For each line: If doesn't contain `\bxmlns\b`: For each link, `\bhttp://[^# \t\r\n]*(?:\w|/)`: If neither `\bgnu\.org/license`, nor `\bmozilla\.org/MPL\b`: If both the HTTP and HTTPS versions return 200 OK and serve the same content: Replace HTTP with HTTPS. Signed-off-by: Alexander A. Klimov <grandmaster(a)al2klimov.de> --- Continuing my work started at 93431e0607e5. See also: git log --oneline '--author=Alexander A. Klimov <grandmaster(a)al2klimov.de>' v5.7..master If there are any URLs to be removed completely or at least not (just) HTTPSified: Just clearly say so and I'll *undo my change*. See also: https://lkml.org/lkml/2020/6/27/64 If there are any valid, but yet not changed URLs: See: https://lkml.org/lkml/2020/6/26/837 If you apply the patch, please let me know. sound/soc/ti/omap-hdmi.c | 2 +- sound/soc/ti/omap-twl4030.c | 2 +- sound/soc/ti/sdma-pcm.c | 2 +- sound/soc/ti/sdma-pcm.h | 2 +- 4 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/ti/omap-hdmi.c b/sound/soc/ti/omap-hdmi.c index def2a0ce8886..3328c02f93c7 100644 --- a/sound/soc/ti/omap-hdmi.c +++ b/sound/soc/ti/omap-hdmi.c @@ -2,7 +2,7 @@ /* * omap-hdmi-audio.c -- OMAP4+ DSS HDMI audio support library * - * Copyright (C) 2014 Texas Instruments Incorporated - http://www.ti.com + * Copyright (C) 2014 Texas Instruments Incorporated - https://www.ti.com * * Author: Jyri Sarha <jsarha(a)ti.com> */ diff --git a/sound/soc/ti/omap-twl4030.c b/sound/soc/ti/omap-twl4030.c index 92dbe2c67290..4b408cec7a90 100644 --- a/sound/soc/ti/omap-twl4030.c +++ b/sound/soc/ti/omap-twl4030.c @@ -2,7 +2,7 @@ /* * omap-twl4030.c -- SoC audio for TI SoC based boards with twl4030 codec * - * Copyright (C) 2012 Texas Instruments Incorporated - http://www.ti.com + * Copyright (C) 2012 Texas Instruments Incorporated - https://www.ti.com * All rights reserved. * * Author: Peter Ujfalusi <peter.ujfalusi(a)ti.com> diff --git a/sound/soc/ti/sdma-pcm.c b/sound/soc/ti/sdma-pcm.c index 2b0bc234e1b6..9e7691103f05 100644 --- a/sound/soc/ti/sdma-pcm.c +++ b/sound/soc/ti/sdma-pcm.c @@ -1,6 +1,6 @@ // SPDX-License-Identifier: GPL-2.0 /* - * Copyright (C) 2018 Texas Instruments Incorporated - http://www.ti.com + * Copyright (C) 2018 Texas Instruments Incorporated - https://www.ti.com * Author: Peter Ujfalusi <peter.ujfalusi(a)ti.com> */ diff --git a/sound/soc/ti/sdma-pcm.h b/sound/soc/ti/sdma-pcm.h index cb0627c8dd34..c19efb4c043d 100644 --- a/sound/soc/ti/sdma-pcm.h +++ b/sound/soc/ti/sdma-pcm.h @@ -1,6 +1,6 @@ /* SPDX-License-Identifier: GPL-2.0 */ /* - * Copyright (C) 2018 Texas Instruments Incorporated - http://www.ti.com + * Copyright (C) 2018 Texas Instruments Incorporated - https://www.ti.com * Author: Peter Ujfalusi <peter.ujfalusi(a)ti.com> */ -- 2.27.0

2 1

[PATCH v2] dt-bindings: sound: convert ROHM BD28623 amplifier binding to yaml
by Katsuhiro Suzuki 23 Jul '20

23 Jul '20

This patch converts ROHM BD28623UMV class D speaker amplifier binding to DT schema. Signed-off-by: Katsuhiro Suzuki <katsuhiro(a)katsuster.net> Reviewed-by: Rob Herring <robh(a)kernel.org> --- Changes in v2: - add additionalProperties:false - add Rob's Reviewed-by --- .../bindings/sound/rohm,bd28623.txt | 29 -------- .../bindings/sound/rohm,bd28623.yaml | 67 +++++++++++++++++++ 2 files changed, 67 insertions(+), 29 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/rohm,bd28623.txt create mode 100644 Documentation/devicetree/bindings/sound/rohm,bd28623.yaml diff --git a/Documentation/devicetree/bindings/sound/rohm,bd28623.txt b/Documentation/devicetree/bindings/sound/rohm,bd28623.txt deleted file mode 100644 index d84557c2686e..000000000000 --- a/Documentation/devicetree/bindings/sound/rohm,bd28623.txt +++ /dev/null @@ -1,29 +0,0 @@ -ROHM BD28623MUV Class D speaker amplifier for digital input - -This codec does not have any control buses such as I2C, it detect format and -rate of I2S signal automatically. It has two signals that can be connected -to GPIOs: reset and mute. - -Required properties: -- compatible : should be "rohm,bd28623" -- #sound-dai-cells: should be 0. -- VCCA-supply : regulator phandle for the VCCA supply -- VCCP1-supply : regulator phandle for the VCCP1 supply -- VCCP2-supply : regulator phandle for the VCCP2 supply - -Optional properties: -- reset-gpios : GPIO specifier for the active low reset line -- mute-gpios : GPIO specifier for the active low mute line - -Example: - - codec { - compatible = "rohm,bd28623"; - #sound-dai-cells = <0>; - - VCCA-supply = <&vcc_reg>; - VCCP1-supply = <&vcc_reg>; - VCCP2-supply = <&vcc_reg>; - reset-gpios = <&gpio 0 GPIO_ACTIVE_LOW>; - mute-gpios = <&gpio 1 GPIO_ACTIVE_LOW>; - }; diff --git a/Documentation/devicetree/bindings/sound/rohm,bd28623.yaml b/Documentation/devicetree/bindings/sound/rohm,bd28623.yaml new file mode 100644 index 000000000000..859ce64da152 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rohm,bd28623.yaml @@ -0,0 +1,67 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/rohm,bd28623.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: ROHM BD28623MUV Class D speaker amplifier for digital input + +description: + This codec does not have any control buses such as I2C, it detect + format and rate of I2S signal automatically. It has two signals + that can be connected to GPIOs reset and mute. + +maintainers: + - Katsuhiro Suzuki <katsuhiro(a)katsuster.net> + +properties: + compatible: + const: rohm,bd28623 + + "#sound-dai-cells": + const: 0 + + VCCA-supply: + description: + regulator phandle for the VCCA (for analog) power supply + + VCCP1-supply: + description: + regulator phandle for the VCCP1 (for ch1) power supply + + VCCP2-supply: + description: + regulator phandle for the VCCP2 (for ch2) power supply + + reset-gpios: + maxItems: 1 + description: + GPIO specifier for the active low reset line + + mute-gpios: + maxItems: 1 + description: + GPIO specifier for the active low mute line + +required: + - compatible + - VCCA-supply + - VCCP1-supply + - VCCP2-supply + - "#sound-dai-cells" + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + codec { + compatible = "rohm,bd28623"; + #sound-dai-cells = <0>; + + VCCA-supply = <&vcc_reg>; + VCCP1-supply = <&vcc_reg>; + VCCP2-supply = <&vcc_reg>; + reset-gpios = <&gpio 0 GPIO_ACTIVE_LOW>; + mute-gpios = <&gpio 1 GPIO_ACTIVE_LOW>; + }; -- 2.27.0

2 1

[PATCH 1/2] ASoC: fsl-asoc-card: Support configuring dai fmt from DT
by Shengjiu Wang 23 Jul '20

23 Jul '20

Support same propeties as simple card for configuring fmt from DT. In order to make this change compatible with old DT, these properties are optional. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl-asoc-card.c | 28 ++++++++++++++++++++++++++++ 1 file changed, 28 insertions(+) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index ee80d02b56c6..4848ba61d083 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -531,11 +531,14 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) struct device_node *cpu_np, *codec_np, *asrc_np; struct device_node *np = pdev->dev.of_node; struct platform_device *asrc_pdev = NULL; + struct device_node *bitclkmaster = NULL; + struct device_node *framemaster = NULL; struct platform_device *cpu_pdev; struct fsl_asoc_card_priv *priv; struct device *codec_dev = NULL; const char *codec_dai_name; const char *codec_dev_name; + unsigned int daifmt; u32 width; int ret; @@ -667,6 +670,31 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) goto asrc_fail; } + /* Format info from DT is optional. */ + daifmt = snd_soc_of_parse_daifmt(np, NULL, + &bitclkmaster, &framemaster); + daifmt &= ~SND_SOC_DAIFMT_MASTER_MASK; + if (bitclkmaster || framemaster) { + if (codec_np == bitclkmaster) + daifmt |= (codec_np == framemaster) ? + SND_SOC_DAIFMT_CBM_CFM : SND_SOC_DAIFMT_CBM_CFS; + else + daifmt |= (codec_np == framemaster) ? + SND_SOC_DAIFMT_CBS_CFM : SND_SOC_DAIFMT_CBS_CFS; + + /* Override dai_fmt with value from DT */ + priv->dai_fmt = daifmt; + } + + /* Change direction according to format */ + if (priv->dai_fmt & SND_SOC_DAIFMT_CBM_CFM) { + priv->cpu_priv.sysclk_dir[TX] = SND_SOC_CLOCK_IN; + priv->cpu_priv.sysclk_dir[RX] = SND_SOC_CLOCK_IN; + } + + of_node_put(bitclkmaster); + of_node_put(framemaster); + if (!fsl_asoc_card_is_ac97(priv) && !codec_dev) { dev_err(&pdev->dev, "failed to find codec device\n"); ret = -EPROBE_DEFER; -- 2.27.0

3 3

[PATCH 00/29] ASoC: add asoc_substream_to_rtd() macro
by Kuninori Morimoto 23 Jul '20

23 Jul '20

Hi Mark Many ASoC drivers are getting rtd from substream by rtd = substream->private_data OTOH, we have snd_pcm_substream_chip() macro for it. #define snd_pcm_substream_chip(substream) ((substream)->private_data) But, both are not understandable for reader. This patch adds new asoc_substream_to_rtd() which is easy to understand. These are not important, but for readable code. Kuninori Morimoto (29): ASoC: soc-xxx: add asoc_substream_to_rtd() ASoC: ux500: use asoc_substream_to_rtd() ASoC: ti: use asoc_substream_to_rtd() ASoC: tegra: use asoc_substream_to_rtd() ASoC: sunxi: use asoc_substream_to_rtd() ASoC: stm: use asoc_substream_to_rtd() ASoC: sof: use asoc_substream_to_rtd() ASoC: sh: use asoc_substream_to_rtd() ASoC: samsung: use asoc_substream_to_rtd() ASoC: pxa: use asoc_substream_to_rtd() ASoC: cirrus: use asoc_substream_to_rtd() ASoC: rockchip: use asoc_substream_to_rtd() ASoC: amd: use asoc_substream_to_rtd() ASoC: fsl: use asoc_substream_to_rtd() ASoC: mediatek: use asoc_substream_to_rtd() ASoC: atmel: use asoc_substream_to_rtd() ASoC: qcom: use asoc_substream_to_rtd() ASoC: dwc: use asoc_substream_to_rtd() ASoC: intel: use asoc_substream_to_rtd() ASoC: meson: use asoc_substream_to_rtd() ASoC: au1x: use asoc_substream_to_rtd() ASoC: bcm: use asoc_substream_to_rtd() ASoC: codecs: use asoc_substream_to_rtd() ASoC: generic: use asoc_substream_to_rtd() ASoC: sprd: use asoc_substream_to_rtd() ASoC: kirkwood: use asoc_substream_to_rtd() ASoC: xtensa: use asoc_substream_to_rtd() ASoC: mxs: use asoc_substream_to_rtd() ASoC: uniphier: use asoc_substream_to_rtd() include/sound/soc.h | 2 + sound/soc/amd/acp-da7219-max98357a.c | 12 ++-- sound/soc/amd/acp-pcm-dma.c | 2 +- sound/soc/amd/acp-rt5645.c | 2 +- sound/soc/amd/acp3x-rt5682-max9836.c | 8 +-- sound/soc/amd/raven/acp3x-i2s.c | 2 +- sound/soc/amd/raven/acp3x-pcm-dma.c | 6 +- sound/soc/atmel/atmel-classd.c | 8 +-- sound/soc/atmel/atmel-pcm-dma.c | 4 +- sound/soc/atmel/atmel-pcm-pdc.c | 2 +- sound/soc/atmel/atmel-pdmic.c | 10 +-- sound/soc/atmel/atmel_wm8904.c | 2 +- sound/soc/au1x/db1200.c | 2 +- sound/soc/au1x/dbdma2.c | 2 +- sound/soc/au1x/dma.c | 2 +- sound/soc/bcm/bcm63xx-pcm-whistler.c | 12 ++-- sound/soc/bcm/cygnus-pcm.c | 16 ++--- sound/soc/cirrus/edb93xx.c | 2 +- sound/soc/cirrus/snappercl15.c | 2 +- sound/soc/codecs/rt5677-spi.c | 4 +- sound/soc/dwc/dwc-pcm.c | 2 +- sound/soc/fsl/eukrea-tlv320.c | 2 +- sound/soc/fsl/fsl-asoc-card.c | 2 +- sound/soc/fsl/fsl_asrc_dma.c | 4 +- sound/soc/fsl/fsl_dma.c | 2 +- sound/soc/fsl/fsl_spdif.c | 10 +-- sound/soc/fsl/fsl_ssi.c | 8 +-- sound/soc/fsl/imx-audmix.c | 6 +- sound/soc/fsl/imx-mc13783.c | 2 +- sound/soc/fsl/mpc5200_dma.c | 8 +-- sound/soc/fsl/mpc5200_psc_i2s.c | 2 +- sound/soc/fsl/mpc8610_hpcd.c | 2 +- sound/soc/fsl/mx27vis-aic32x4.c | 2 +- sound/soc/fsl/p1022_ds.c | 2 +- sound/soc/fsl/p1022_rdk.c | 2 +- sound/soc/fsl/wm1133-ev1.c | 2 +- sound/soc/generic/simple-card-utils.c | 6 +- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 6 +- sound/soc/intel/baytrail/sst-baytrail-pcm.c | 16 ++--- sound/soc/intel/boards/bdw-rt5650.c | 2 +- sound/soc/intel/boards/bdw-rt5677.c | 4 +- sound/soc/intel/boards/broadwell.c | 2 +- sound/soc/intel/boards/bxt_rt298.c | 2 +- sound/soc/intel/boards/byt-rt5640.c | 2 +- sound/soc/intel/boards/bytcht_da7213.c | 4 +- sound/soc/intel/boards/bytcr_rt5640.c | 2 +- sound/soc/intel/boards/bytcr_rt5651.c | 2 +- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 2 +- sound/soc/intel/boards/cht_bsw_nau8824.c | 2 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 2 +- sound/soc/intel/boards/cht_bsw_rt5672.c | 2 +- sound/soc/intel/boards/cml_rt1011_rt5682.c | 4 +- sound/soc/intel/boards/ehl_rt5660.c | 2 +- sound/soc/intel/boards/glk_rt5682_max98357a.c | 2 +- sound/soc/intel/boards/haswell.c | 2 +- sound/soc/intel/boards/kbl_da7219_max98927.c | 8 +-- sound/soc/intel/boards/kbl_rt5660.c | 2 +- sound/soc/intel/boards/kbl_rt5663_max98927.c | 4 +- .../intel/boards/kbl_rt5663_rt5514_max98927.c | 4 +- .../soc/intel/boards/skl_nau88l25_max98357a.c | 2 +- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 2 +- sound/soc/intel/boards/skl_rt286.c | 2 +- sound/soc/intel/boards/sof_da7219_max98373.c | 2 +- sound/soc/intel/boards/sof_maxim_common.c | 4 +- sound/soc/intel/boards/sof_pcm512x.c | 4 +- sound/soc/intel/boards/sof_rt5682.c | 4 +- sound/soc/intel/boards/sof_sdw_rt1308.c | 2 +- sound/soc/intel/boards/sof_wm8804.c | 2 +- sound/soc/intel/haswell/sst-haswell-pcm.c | 12 ++-- sound/soc/intel/keembay/kmb_platform.c | 2 +- sound/soc/intel/skylake/skl-pcm.c | 8 +-- sound/soc/kirkwood/armada-370-db.c | 2 +- sound/soc/mediatek/common/mtk-afe-fe-dai.c | 12 ++-- .../mediatek/common/mtk-afe-platform-driver.c | 2 +- sound/soc/mediatek/mt2701/mt2701-afe-pcm.c | 2 +- sound/soc/mediatek/mt2701/mt2701-cs42448.c | 2 +- sound/soc/mediatek/mt2701/mt2701-wm8960.c | 2 +- sound/soc/mediatek/mt6797/mt6797-afe-pcm.c | 4 +- sound/soc/mediatek/mt8173/mt8173-afe-pcm.c | 2 +- sound/soc/mediatek/mt8173/mt8173-max98090.c | 2 +- .../mediatek/mt8173/mt8173-rt5650-rt5514.c | 2 +- .../mediatek/mt8173/mt8173-rt5650-rt5676.c | 2 +- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 2 +- sound/soc/mediatek/mt8183/mt8183-afe-pcm.c | 4 +- .../mediatek/mt8183/mt8183-da7219-max98357.c | 8 +-- .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 12 ++-- sound/soc/meson/axg-card.c | 2 +- sound/soc/meson/gx-card.c | 2 +- sound/soc/meson/meson-card-utils.c | 2 +- sound/soc/meson/meson-codec-glue.c | 2 +- sound/soc/mxs/mxs-sgtl5000.c | 2 +- sound/soc/pxa/brownstone.c | 2 +- sound/soc/pxa/corgi.c | 4 +- sound/soc/pxa/hx4700.c | 2 +- sound/soc/pxa/imote2.c | 2 +- sound/soc/pxa/magician.c | 6 +- sound/soc/pxa/mmp-pcm.c | 2 +- sound/soc/pxa/poodle.c | 4 +- sound/soc/pxa/pxa2xx-i2s.c | 2 +- sound/soc/pxa/spitz.c | 4 +- sound/soc/pxa/tosa.c | 2 +- sound/soc/pxa/z2.c | 2 +- sound/soc/pxa/zylonite.c | 2 +- sound/soc/qcom/apq8096.c | 2 +- sound/soc/qcom/lpass-platform.c | 14 ++--- sound/soc/qcom/qdsp6/q6asm-dai.c | 6 +- sound/soc/qcom/qdsp6/q6routing.c | 2 +- sound/soc/qcom/sdm845.c | 14 ++--- sound/soc/qcom/storm.c | 2 +- sound/soc/rockchip/rk3288_hdmi_analog.c | 2 +- sound/soc/rockchip/rk3399_gru_sound.c | 8 +-- sound/soc/rockchip/rockchip_i2s.c | 2 +- sound/soc/rockchip/rockchip_max98090.c | 2 +- sound/soc/rockchip/rockchip_rt5645.c | 2 +- sound/soc/samsung/aries_wm8994.c | 4 +- sound/soc/samsung/arndale.c | 4 +- sound/soc/samsung/h1940_uda1380.c | 2 +- sound/soc/samsung/i2s.c | 2 +- sound/soc/samsung/jive_wm8750.c | 2 +- sound/soc/samsung/littlemill.c | 2 +- sound/soc/samsung/neo1973_wm8753.c | 8 +-- sound/soc/samsung/odroid.c | 6 +- sound/soc/samsung/pcm.c | 4 +- sound/soc/samsung/rx1950_uda1380.c | 2 +- sound/soc/samsung/s3c-i2s-v2.c | 2 +- sound/soc/samsung/s3c24xx_simtec.c | 2 +- sound/soc/samsung/s3c24xx_uda134x.c | 6 +- sound/soc/samsung/smartq_wm8987.c | 2 +- sound/soc/samsung/smdk_spdif.c | 2 +- sound/soc/samsung/smdk_wm8580.c | 2 +- sound/soc/samsung/smdk_wm8994.c | 2 +- sound/soc/samsung/smdk_wm8994pcm.c | 2 +- sound/soc/samsung/snow.c | 2 +- sound/soc/samsung/spdif.c | 6 +- sound/soc/samsung/tm2_wm5110.c | 8 +-- sound/soc/sh/dma-sh7760.c | 12 ++-- sound/soc/sh/fsi.c | 2 +- sound/soc/sh/migor.c | 4 +- sound/soc/sh/rcar/core.c | 4 +- sound/soc/soc-component.c | 20 +++--- sound/soc/soc-dai.c | 8 +-- sound/soc/soc-dapm.c | 6 +- sound/soc/soc-generic-dmaengine-pcm.c | 4 +- sound/soc/soc-link.c | 12 ++-- sound/soc/soc-pcm.c | 62 +++++++++---------- sound/soc/soc-utils.c | 2 +- sound/soc/sof/intel/hda-dai.c | 10 +-- sound/soc/sof/intel/hda-dsp.c | 2 +- sound/soc/sof/intel/hda-pcm.c | 2 +- sound/soc/sof/pcm.c | 18 +++--- sound/soc/sprd/sprd-pcm-dma.c | 2 +- sound/soc/stm/stm32_adfsdm.c | 12 ++-- sound/soc/stm/stm32_sai_sub.c | 2 +- sound/soc/sunxi/sun4i-codec.c | 12 ++-- sound/soc/sunxi/sun4i-spdif.c | 2 +- sound/soc/tegra/tegra_alc5632.c | 2 +- sound/soc/tegra/tegra_max98090.c | 2 +- sound/soc/tegra/tegra_rt5640.c | 2 +- sound/soc/tegra/tegra_rt5677.c | 2 +- sound/soc/tegra/tegra_sgtl5000.c | 2 +- sound/soc/tegra/tegra_wm8753.c | 2 +- sound/soc/tegra/tegra_wm8903.c | 2 +- sound/soc/tegra/trimslice.c | 2 +- sound/soc/ti/davinci-evm.c | 6 +- sound/soc/ti/davinci-vcif.c | 4 +- sound/soc/ti/j721e-evm.c | 6 +- sound/soc/ti/n810.c | 4 +- sound/soc/ti/omap-abe-twl6040.c | 4 +- sound/soc/ti/omap-mcbsp.c | 4 +- sound/soc/ti/omap-twl4030.c | 2 +- sound/soc/ti/omap3pandora.c | 2 +- sound/soc/ti/osk5912.c | 2 +- sound/soc/ti/rx51.c | 4 +- sound/soc/uniphier/aio-dma.c | 6 +- sound/soc/ux500/mop500_ab8500.c | 8 +-- sound/soc/ux500/ux500_pcm.c | 2 +- sound/soc/xtensa/xtfpga-i2s.c | 2 +- 177 files changed, 397 insertions(+), 395 deletions(-) -- 2.25.1

3 34

[PATCH] ASoC: ti: Replace HTTP links with HTTPS ones
by Alexander A. Klimov 23 Jul '20

23 Jul '20

Rationale: Reduces attack surface on kernel devs opening the links for MITM as HTTPS traffic is much harder to manipulate. Deterministic algorithm: For each file: If not .svg: For each line: If doesn't contain `\bxmlns\b`: For each link, `\bhttp://[^# \t\r\n]*(?:\w|/)`: If neither `\bgnu\.org/license`, nor `\bmozilla\.org/MPL\b`: If both the HTTP and HTTPS versions return 200 OK and serve the same content: Replace HTTP with HTTPS. Signed-off-by: Alexander A. Klimov <grandmaster(a)al2klimov.de> --- Continuing my work started at 93431e0607e5. See also: git log --oneline '--author=Alexander A. Klimov <grandmaster(a)al2klimov.de>' v5.7..master If there are any URLs to be removed completely or at least not (just) HTTPSified: Just clearly say so and I'll *undo my change*. See also: https://lkml.org/lkml/2020/6/27/64 If there are any valid, but yet not changed URLs: See: https://lkml.org/lkml/2020/6/26/837 If you apply the patch, please let me know. sound/soc/ti/udma-pcm.c | 2 +- sound/soc/ti/udma-pcm.h | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/ti/udma-pcm.c b/sound/soc/ti/udma-pcm.c index 39830caaaf7c..2ff0f518aba5 100644 --- a/sound/soc/ti/udma-pcm.c +++ b/sound/soc/ti/udma-pcm.c @@ -1,6 +1,6 @@ // SPDX-License-Identifier: GPL-2.0 /* - * Copyright (C) 2020 Texas Instruments Incorporated - http://www.ti.com + * Copyright (C) 2020 Texas Instruments Incorporated - https://www.ti.com * Author: Peter Ujfalusi <peter.ujfalusi(a)ti.com> */ diff --git a/sound/soc/ti/udma-pcm.h b/sound/soc/ti/udma-pcm.h index 54111e7312c1..9ed588fd79b9 100644 --- a/sound/soc/ti/udma-pcm.h +++ b/sound/soc/ti/udma-pcm.h @@ -1,6 +1,6 @@ /* SPDX-License-Identifier: GPL-2.0 */ /* - * Copyright (C) 2018 Texas Instruments Incorporated - http://www.ti.com + * Copyright (C) 2018 Texas Instruments Incorporated - https://www.ti.com */ #ifndef __UDMA_PCM_H__ -- 2.27.0

2 1

[PATCH] ASoC: wm_adsp: Support new metadata block ID's
by Charles Keepax 23 Jul '20

23 Jul '20

From: James Schulman <james.schulman(a)cirrus.com> Coefficient files now support additional metadata blocks, these contain machine parsable text strings describing the parameters contained in the coefficient file. Signed-off-by: James Schulman <james.schulman(a)cirrus.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/codecs/wm_adsp.c | 1 + sound/soc/codecs/wmfw.h | 1 + 2 files changed, 2 insertions(+) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 88c397c700ee9..410cca57da52d 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -2621,6 +2621,7 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) switch (type) { case (WMFW_NAME_TEXT << 8): case (WMFW_INFO_TEXT << 8): + case (WMFW_METADATA << 8): break; case (WMFW_ABSOLUTE << 8): /* diff --git a/sound/soc/codecs/wmfw.h b/sound/soc/codecs/wmfw.h index 4278aa6aeb01b..7423272c30e94 100644 --- a/sound/soc/codecs/wmfw.h +++ b/sound/soc/codecs/wmfw.h @@ -180,6 +180,7 @@ struct wmfw_coeff_item { #define WMFW_ABSOLUTE 0xf0 #define WMFW_ALGORITHM_DATA 0xf2 +#define WMFW_METADATA 0xfc #define WMFW_NAME_TEXT 0xfe #define WMFW_INFO_TEXT 0xff -- 2.11.0

2 1

[PATCH] ASoC: fsl: Replace HTTP links with HTTPS ones
by Alexander A. Klimov 23 Jul '20

23 Jul '20

Rationale: Reduces attack surface on kernel devs opening the links for MITM as HTTPS traffic is much harder to manipulate. Deterministic algorithm: For each file: If not .svg: For each line: If doesn't contain `\bxmlns\b`: For each link, `\bhttp://[^# \t\r\n]*(?:\w|/)`: If neither `\bgnu\.org/license`, nor `\bmozilla\.org/MPL\b`: If both the HTTP and HTTPS versions return 200 OK and serve the same content: Replace HTTP with HTTPS. Signed-off-by: Alexander A. Klimov <grandmaster(a)al2klimov.de> --- Continuing my work started at 93431e0607e5. See also: git log --oneline '--author=Alexander A. Klimov <grandmaster(a)al2klimov.de>' v5.7..master If there are any URLs to be removed completely or at least not (just) HTTPSified: Just clearly say so and I'll *undo my change*. See also: https://lkml.org/lkml/2020/6/27/64 If there are any valid, but yet not changed URLs: See: https://lkml.org/lkml/2020/6/26/837 If you apply the patch, please let me know. sound/soc/fsl/imx-audmix.c | 4 ++-- sound/soc/fsl/imx-audmux.c | 2 +- 2 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/fsl/imx-audmix.c b/sound/soc/fsl/imx-audmix.c index e09b45de0efd..96980cb0497f 100644 --- a/sound/soc/fsl/imx-audmix.c +++ b/sound/soc/fsl/imx-audmix.c @@ -6,8 +6,8 @@ * License. You may obtain a copy of the GNU General Public License * Version 2 or later at the following locations: * - * http://www.opensource.org/licenses/gpl-license.html - * http://www.gnu.org/copyleft/gpl.html + * https://www.opensource.org/licenses/gpl-license.html + * https://www.gnu.org/copyleft/gpl.html */ #include <linux/module.h> diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c index 3ce85a43e08f..25c18b9e348f 100644 --- a/sound/soc/fsl/imx-audmux.c +++ b/sound/soc/fsl/imx-audmux.c @@ -5,7 +5,7 @@ // Copyright 2009 Pengutronix, Sascha Hauer <s.hauer(a)pengutronix.de> // // Initial development of this code was funded by -// Phytec Messtechnik GmbH, http://www.phytec.de +// Phytec Messtechnik GmbH, https://www.phytec.de #include <linux/clk.h> #include <linux/debugfs.h> -- 2.27.0

2 1

[PATCH -next] ASoC: ti: fix SND_SOC_J721E_EVM warnings & errors
by Randy Dunlap 23 Jul '20

23 Jul '20

From: Randy Dunlap <rdunlap(a)infradead.org> SND_SOC_J721E_EVM should not select SND_SOC_PCM3168A_I2C when I2C is not enabled. That causes build errors, so make this driver's symbol depend on I2C. WARNING: unmet direct dependencies detected for SND_SOC_PCM3168A_I2C Depends on [n]: SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && I2C [=n] Selected by [m]: - SND_SOC_J721E_EVM [=m] && SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && (DMA_OMAP [=y] || TI_EDMA [=m] || TI_K3_UDMA [=n] || COMPILE_TEST [=y]) && (ARCH_K3_J721E_SOC [=n] || COMPILE_TEST [=y]) ../sound/soc/codecs/pcm3168a-i2c.c:59:1: warning: data definition has no type or storage class module_i2c_driver(pcm3168a_i2c_driver); ^~~~~~~~~~~~~~~~~ ../sound/soc/codecs/pcm3168a-i2c.c:59:1: error: type defaults to ‘int’ in declaration of ‘module_i2c_driver’ [-Werror=implicit-int] ../sound/soc/codecs/pcm3168a-i2c.c:59:1: warning: parameter names (without types) in function declaration ../sound/soc/codecs/pcm3168a-i2c.c:49:26: warning: ‘pcm3168a_i2c_driver’ defined but not used [-Wunused-variable] static struct i2c_driver pcm3168a_i2c_driver = { ^~~~~~~~~~~~~~~~~~~ cc1: some warnings being treated as errors Fixes: 6748d0559059 ("ASoC: ti: Add custom machine driver for j721e EVM (CPB and IVI)") Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Cc: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Cc: Mark Brown <broonie(a)kernel.org> --- sound/soc/ti/Kconfig | 1 + 1 file changed, 1 insertion(+) --- linux-next-20200720.orig/sound/soc/ti/Kconfig +++ linux-next-20200720/sound/soc/ti/Kconfig @@ -222,6 +222,7 @@ config SND_SOC_DM365_VOICE_CODEC_MODULE config SND_SOC_J721E_EVM tristate "SoC Audio support for j721e EVM" depends on ARCH_K3_J721E_SOC || COMPILE_TEST + depends on I2C select SND_SOC_PCM3168A_I2C select SND_SOC_DAVINCI_MCASP help

3 2

[PATCH 1/4] ASoC: tas2562: Add right and left channel slot programming
by Dan Murphy 23 Jul '20

23 Jul '20

Add programming for the tdm slots for the right and left. This also requires configuring the RX/TX offsets for the DAI format type. Signed-off-by: Dan Murphy <dmurphy(a)ti.com> --- sound/soc/codecs/tas2562.c | 71 +++++++++++++++++++++++++++++--------- sound/soc/codecs/tas2562.h | 5 ++- 2 files changed, 59 insertions(+), 17 deletions(-) diff --git a/sound/soc/codecs/tas2562.c b/sound/soc/codecs/tas2562.c index d26e30a2948c..1d3c381aeefe 100644 --- a/sound/soc/codecs/tas2562.c +++ b/sound/soc/codecs/tas2562.c @@ -175,7 +175,37 @@ static int tas2562_set_dai_tdm_slot(struct snd_soc_dai *dai, { struct snd_soc_component *component = dai->component; struct tas2562_data *tas2562 = snd_soc_component_get_drvdata(component); - int ret = 0; + int left_slot, right_slot; + int slots_cfg; + int ret; + + if (!tx_mask) { + dev_err(component->dev, "tx masks must not be 0\n"); + return -EINVAL; + } + + if (slots == 1) { + if (tx_mask != 1) + return -EINVAL; + + left_slot = 0; + right_slot = 0; + } else { + left_slot = __ffs(tx_mask); + tx_mask &= ~(1 << left_slot); + if (tx_mask == 0) { + right_slot = left_slot; + } else { + right_slot = __ffs(tx_mask); + tx_mask &= ~(1 << right_slot); + } + } + + slots_cfg = (right_slot << TAS2562_RIGHT_SLOT_SHIFT) | left_slot; + + ret = snd_soc_component_write(component, TAS2562_TDM_CFG3, slots_cfg); + if (ret < 0) + return ret; switch (slot_width) { case 16: @@ -208,6 +238,18 @@ static int tas2562_set_dai_tdm_slot(struct snd_soc_dai *dai, if (ret < 0) return ret; + ret = snd_soc_component_update_bits(component, TAS2562_TDM_CFG5, + TAS2562_TDM_CFG5_VSNS_SLOT_MASK, + tas2562->v_sense_slot); + if (ret < 0) + return ret; + + ret = snd_soc_component_update_bits(component, TAS2562_TDM_CFG6, + TAS2562_TDM_CFG6_ISNS_SLOT_MASK, + tas2562->i_sense_slot); + if (ret < 0) + return ret; + return 0; } @@ -285,7 +327,8 @@ static int tas2562_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) { struct snd_soc_component *component = dai->component; struct tas2562_data *tas2562 = snd_soc_component_get_drvdata(component); - u8 tdm_rx_start_slot = 0, asi_cfg_1 = 0; + u8 asi_cfg_1 = 0; + u8 tdm_rx_start_slot = 0; int ret; switch (fmt & SND_SOC_DAIFMT_INV_MASK) { @@ -307,27 +350,23 @@ static int tas2562_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) dev_err(tas2562->dev, "Failed to set RX edge\n"); return ret; } - switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { - case (SND_SOC_DAIFMT_I2S): - case (SND_SOC_DAIFMT_DSP_A): - case (SND_SOC_DAIFMT_DSP_B): - tdm_rx_start_slot = BIT(1); - break; - case (SND_SOC_DAIFMT_LEFT_J): + case SND_SOC_DAIFMT_LEFT_J: + case SND_SOC_DAIFMT_DSP_B: tdm_rx_start_slot = 0; break; - default: - dev_err(tas2562->dev, "DAI Format is not found, fmt=0x%x\n", - fmt); - ret = -EINVAL; + case SND_SOC_DAIFMT_I2S: + case SND_SOC_DAIFMT_DSP_A: + tdm_rx_start_slot = 1; break; + default: + dev_err(tas2562->dev, + "DAI Format is not found, fmt=0x%x\n", fmt); + return -EINVAL; } ret = snd_soc_component_update_bits(component, TAS2562_TDM_CFG1, - TAS2562_TDM_CFG1_RX_OFFSET_MASK, - tdm_rx_start_slot); - + TAS2562_RX_OFF_MASK, (tdm_rx_start_slot << 1)); if (ret < 0) return ret; diff --git a/sound/soc/codecs/tas2562.h b/sound/soc/codecs/tas2562.h index 28e75fc431d0..18209f397921 100644 --- a/sound/soc/codecs/tas2562.h +++ b/sound/soc/codecs/tas2562.h @@ -34,6 +34,10 @@ #define TAS2562_TDM_DET TAS2562_REG(0, 0x11) #define TAS2562_REV_ID TAS2562_REG(0, 0x7d) +#define TAS2562_RX_OFF_MASK GENMASK(5, 1) +#define TAS2562_TX_OFF_MASK GENMASK(3, 1) +#define TAS2562_RIGHT_SLOT_SHIFT 4 + /* Page 2 */ #define TAS2562_DVC_CFG1 TAS2562_REG(2, 0x0c) #define TAS2562_DVC_CFG2 TAS2562_REG(2, 0x0d) @@ -49,7 +53,6 @@ #define TAS2562_TDM_CFG1_RX_EDGE_MASK BIT(0) #define TAS2562_TDM_CFG1_RX_FALLING 1 -#define TAS2562_TDM_CFG1_RX_OFFSET_MASK GENMASK(4, 0) #define TAS2562_TDM_CFG0_RAMPRATE_MASK BIT(5) #define TAS2562_TDM_CFG0_RAMPRATE_44_1 BIT(5) -- 2.26.2

3 7

[PATCH v8] ASoC: Intel: kbl_rt5663_rt5514_max98927: Fix kabylake_ssp_fixup function
by Harsha Priya 23 Jul '20

23 Jul '20

kabylake_ssp_fixup function uses snd_soc_dpcm to identify the codecs DAIs. The HW parameters are changed based on the codec DAI of the stream. The earlier approach to get snd_soc_dpcm was using container_of() macro on snd_pcm_hw_params. The structures have been modified over time and snd_soc_dpcm does not have snd_pcm_hw_params as a reference but as a copy. This causes the current driver to crash when used. This patch changes the way snd_soc_dpcm is extracted. snd_soc_pcm_runtime holds 2 dpcm instances (one for playback and one for capture). 2 codecs on the SSP are dmic (capture) and speakers (playback). Based on the stream direction, snd_soc_dpcm is extracted from snd_soc_pcm_runtime. Tested for all use cases of the driver. Signed-off-by: Harsha Priya <harshapriya.n(a)intel.com> Signed-off-by: Vamshi Krishna Gopal <vamshi.krishna.gopal(a)intel.com> Tested-by: Lukasz Majczak <lma(a)semihalf.com> --- v1 -> v2: - Extract dmic from SSP0 as every BE should have own fixup function. v2 -> v3: - Restore naming in the dapm route table to not confuse with other drivers - Fixed indentations v3 -> v4: - Updated code and commit description according to solution proposed by Harsha v4 -> v5: - Cosmetic Changes v5 -> v6: - Dmic regression seen with v4 fixed - Using available routines for obtaining dpcm information v6 -> v7: - Updated comments - initilize rtd_dpcm variable - added break statement in the loop v7 -> v8: - Updated commit message - return on rtd_dpcm being NULL - typo fixed --- --- .../intel/boards/kbl_rt5663_rt5514_max98927.c | 41 +++++++++++++++---- 1 file changed, 32 insertions(+), 9 deletions(-) diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index 584e4f9cedc2..21808fe13481 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -379,22 +379,45 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_interval *chan = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); - struct snd_soc_dpcm *dpcm = container_of( - params, struct snd_soc_dpcm, hw_params); - struct snd_soc_dai_link *fe_dai_link = dpcm->fe->dai_link; - struct snd_soc_dai_link *be_dai_link = dpcm->be->dai_link; + struct snd_soc_dpcm *dpcm, *rtd_dpcm = NULL; + + /* + * The following loop will be called only for playback stream + * In this platform, there is only one playback device on every SSP + */ + for_each_dpcm_fe(rtd, SNDRV_PCM_STREAM_PLAYBACK, dpcm) { + rtd_dpcm = dpcm; + break; + } + + /* + * This following loop will be called only for capture stream + * In this platform, there is only one capture device on every SSP + */ + for_each_dpcm_fe(rtd, SNDRV_PCM_STREAM_CAPTURE, dpcm) { + rtd_dpcm = dpcm; + break; + } + + if (!rtd_dpcm) + return -EINVAL; + + /* + * The above 2 loops are mutually exclusive based on the stream direction, + * thus rtd_dpcm variable will never be overwritten + */ /* * The ADSP will convert the FE rate to 48k, stereo, 24 bit */ - if (!strcmp(fe_dai_link->name, "Kbl Audio Port") || - !strcmp(fe_dai_link->name, "Kbl Audio Headset Playback") || - !strcmp(fe_dai_link->name, "Kbl Audio Capture Port")) { + if (!strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio Port") || + !strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio Headset Playback") || + !strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio Capture Port")) { rate->min = rate->max = 48000; chan->min = chan->max = 2; snd_mask_none(fmt); snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE); - } else if (!strcmp(fe_dai_link->name, "Kbl Audio DMIC cap")) { + } else if (!strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio DMIC cap")) { if (params_channels(params) == 2 || DMIC_CH(dmic_constraints) == 2) chan->min = chan->max = 2; @@ -405,7 +428,7 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, * The speaker on the SSP0 supports S16_LE and not S24_LE. * thus changing the mask here */ - if (!strcmp(be_dai_link->name, "SSP0-Codec")) + if (!strcmp(rtd_dpcm->be->dai_link->name, "SSP0-Codec")) snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S16_LE); return 0; -- 2.17.1

3 2

[PATCH 0/4] ASoC: fsl_asrc: allow selecting arbitrary clocks
by Arnaud Ferraris 23 Jul '20

23 Jul '20

The current ASRC driver hardcodes the input and output clocks used for sample rate conversions. In order to allow greater flexibility and to cover more use cases, it would be preferable to select the clocks using device-tree properties. This series also fix register configuration and clock assignment so conversion can be conducted effectively in both directions with a good quality. Arnaud Ferraris (4): dt-bindings: sound: fsl,asrc: add properties to select in/out clocks ASoC: fsl_asrc: allow using arbitrary input and output clocks ASoC: fsl_asrc: always use ratio for conversion ASoC: fsl_asrc: swap input and output clocks in capture mode Documentation/devicetree/bindings/sound/fsl,asrc.txt | 8 ++++++++ sound/soc/fsl/fsl_asrc.c | 69 ++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++----- sound/soc/fsl/fsl_asrc_common.h | 3 +++ 3 files changed, 75 insertions(+), 5 deletions(-)

3 21

Re: linux-next: Tree for Jul 22 (sound/soc/tegra/snd-soc-tegra210-dmic)
by Randy Dunlap 22 Jul '20

22 Jul '20

On 7/22/20 6:16 AM, Stephen Rothwell wrote: > Hi all, > > Changes since 20200721: > on i386: ERROR: modpost: "__udivdi3" [sound/soc/tegra/snd-soc-tegra210-dmic.ko] undefined! ERROR: modpost: "__divdi3" [sound/soc/tegra/snd-soc-tegra210-dmic.ko] undefined! -- ~Randy Reported-by: Randy Dunlap <rdunlap(a)infradead.org>

1 0

[PATCH] dt-bindings: sound: convert ROHM BD28623 amplifier binding to yaml
by Katsuhiro Suzuki 22 Jul '20

22 Jul '20

This patch converts ROHM BD28623UMV class D speaker amplifier binding to DT schema. Signed-off-by: Katsuhiro Suzuki <katsuhiro(a)katsuster.net> --- .../bindings/sound/rohm,bd28623.txt | 29 --------- .../bindings/sound/rohm,bd28623.yaml | 65 +++++++++++++++++++ 2 files changed, 65 insertions(+), 29 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/rohm,bd28623.txt create mode 100644 Documentation/devicetree/bindings/sound/rohm,bd28623.yaml diff --git a/Documentation/devicetree/bindings/sound/rohm,bd28623.txt b/Documentation/devicetree/bindings/sound/rohm,bd28623.txt deleted file mode 100644 index d84557c2686e..000000000000 --- a/Documentation/devicetree/bindings/sound/rohm,bd28623.txt +++ /dev/null @@ -1,29 +0,0 @@ -ROHM BD28623MUV Class D speaker amplifier for digital input - -This codec does not have any control buses such as I2C, it detect format and -rate of I2S signal automatically. It has two signals that can be connected -to GPIOs: reset and mute. - -Required properties: -- compatible : should be "rohm,bd28623" -- #sound-dai-cells: should be 0. -- VCCA-supply : regulator phandle for the VCCA supply -- VCCP1-supply : regulator phandle for the VCCP1 supply -- VCCP2-supply : regulator phandle for the VCCP2 supply - -Optional properties: -- reset-gpios : GPIO specifier for the active low reset line -- mute-gpios : GPIO specifier for the active low mute line - -Example: - - codec { - compatible = "rohm,bd28623"; - #sound-dai-cells = <0>; - - VCCA-supply = <&vcc_reg>; - VCCP1-supply = <&vcc_reg>; - VCCP2-supply = <&vcc_reg>; - reset-gpios = <&gpio 0 GPIO_ACTIVE_LOW>; - mute-gpios = <&gpio 1 GPIO_ACTIVE_LOW>; - }; diff --git a/Documentation/devicetree/bindings/sound/rohm,bd28623.yaml b/Documentation/devicetree/bindings/sound/rohm,bd28623.yaml new file mode 100644 index 000000000000..acd8609252b4 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rohm,bd28623.yaml @@ -0,0 +1,65 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/rohm,bd28623.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: ROHM BD28623MUV Class D speaker amplifier for digital input + +description: + This codec does not have any control buses such as I2C, it detect + format and rate of I2S signal automatically. It has two signals + that can be connected to GPIOs reset and mute. + +maintainers: + - Katsuhiro Suzuki <katsuhiro(a)katsuster.net> + +properties: + compatible: + const: rohm,bd28623 + + "#sound-dai-cells": + const: 0 + + VCCA-supply: + description: + regulator phandle for the VCCA (for analog) power supply + + VCCP1-supply: + description: + regulator phandle for the VCCP1 (for ch1) power supply + + VCCP2-supply: + description: + regulator phandle for the VCCP2 (for ch2) power supply + + reset-gpios: + maxItems: 1 + description: + GPIO specifier for the active low reset line + + mute-gpios: + maxItems: 1 + description: + GPIO specifier for the active low mute line + +required: + - compatible + - VCCA-supply + - VCCP1-supply + - VCCP2-supply + - "#sound-dai-cells" + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + codec { + compatible = "rohm,bd28623"; + #sound-dai-cells = <0>; + + VCCA-supply = <&vcc_reg>; + VCCP1-supply = <&vcc_reg>; + VCCP2-supply = <&vcc_reg>; + reset-gpios = <&gpio 0 GPIO_ACTIVE_LOW>; + mute-gpios = <&gpio 1 GPIO_ACTIVE_LOW>; + }; -- 2.27.0

2 2

Several issues with HP OMEN dc-0019-ur
by GitHub issues - edited 22 Jul '20

22 Jul '20

alsa-project/alsa-lib issue #73 was edited from UnicodeMan: Hi. I am using HP OMEN dc0019ur laptop with linux 5.7-xanmod kernel and KDE neon (Ubuntu-based) system. Laptop has combined 3.5mm jack and a seperate stereo microphone jack. Tried various quirks by creating /etc/modprobe.d/mute.conf file and adding line "options snd-hda-intel model=XX", where XX is the name of the quirk from documentation here: https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/tree/Doc… With no modprobe changes: mute led does not work, headphones work, but microphone is not detected, and if i plug anything into microphone jack, it switches to microphone in combined jack. With hp-line1-mic1 quirk: mute led works fine, headphones are detected, except microphone, though it works anyway. Seperate microphone jack does not work at all. With headset-mic quirk: mute led does not work, headphones jack works, microphone (from headphones jack) is not detected when present but works when switched to anyway Microphone jack works perfectly (detects a plugged in mic, works in stereo). --- alsa-info.sh logs --- no quirks: http://alsa-project.org/db/?f=a73a83ba0610e3fd88310999b197fdde4372249a hp-line1-mic1: http://alsa-project.org/db/?f=f4d08dec7918c79a391fb1898d50c57a2448d872 headset-mic: https://alsa-project.org/db/?f=4b1eeef821f39eea28071ac2f2c0babcf515ef14 Not sure if it matters, but I want to add that I'm using nvidia render offload with reverse PRIME support, the gpu has one DP and one HDMI port wired directly to it. aaand, I hope I am writing this issue in correct place :D Issue URL : https://github.com/alsa-project/alsa-lib/issues/73 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH v2 0/9] ASoC: qdsp6: add gapless compressed audio support
by Srinivas Kandagatla 22 Jul '20

22 Jul '20

This patchset adds gapless compressed audio support on q6asm. Gapless on q6asm is implemented using 2 streams in a single q6asm session. First few patches such as stream id per each command, gapless flags and silence meta data are for preparedness for adding gapless support. Last patch implements copy callback to allow finer control over buffer offsets, specially in partial drain cases. This patchset is tested on RB3 aka DB845c platform. This patchset as it is will support gapless however QDSP can also support switching decoders on a single stream. Patches to support such feature are send in different patchset which involves adding generic interfaces. Thanks, srini Changes since v1: - Fixed all the comments to use "q6asm" wording correctly. - dropped patches that are already applied Srinivas Kandagatla (9): ASoC: q6asm: rename misleading session id variable ASoC: q6asm: make commands specific to streams ASoC: q6asm: use flags directly from q6asm-dai ASoC: q6asm: add length to write command token ASoC: q6asm: add support to remove intial and trailing silence ASoC: q6asm: add support to gapless flag in q6asm open ASoC: q6asm-dai: add next track metadata support ASoC: qdsp6-dai: add gapless support ASoC: q6asm-dai: add support to copy callback sound/soc/qcom/qdsp6/q6asm-dai.c | 413 +++++++++++++++++++++++-------- sound/soc/qcom/qdsp6/q6asm.c | 169 +++++++++---- sound/soc/qcom/qdsp6/q6asm.h | 48 ++-- 3 files changed, 467 insertions(+), 163 deletions(-) -- 2.21.0

2 15

[PATCH AUTOSEL 4.4 7/9] ALSA: hda/hdmi: fix failures at PCM open on Intel ICL and later
by Sasha Levin 22 Jul '20

22 Jul '20

From: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> [ Upstream commit 56275036d8185f92eceac7479d48b858ee3dab84 ] When HDMI PCM devices are opened in a specific order, with at least one HDMI/DP receiver connected, ALSA PCM open fails to -EBUSY on the connected monitor, on recent Intel platforms (ICL/JSL and newer). While this is not a typical sequence, at least Pulseaudio does this every time when it is started, to discover the available PCMs. The rootcause is an invalid assumption in hdmi_add_pin(), where the total number of converters is assumed to be known at the time the function is called. On older Intel platforms this held true, but after ICL/JSL, the order how pins and converters are in the subnode list as returned by snd_hda_get_sub_nodes(), was changed. As a result, information for some converters was not stored to per_pin->mux_nids. And this means some pins cannot be connected to all converters, and application instead gets -EBUSY instead at open. The assumption that converters are always before pins in the subnode list, is not really a valid one. Fix the problem in hdmi_parse_codec() by introducing separate loops for discovering converters and pins. BugLink: https://github.com/thesofproject/linux/issues/1978 BugLink: https://github.com/thesofproject/linux/issues/2216 BugLink: https://github.com/thesofproject/linux/issues/2217 Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Link: https://lore.kernel.org/r/20200703153818.2808592-1-kai.vehmanen@linux.intel… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_hdmi.c | 36 +++++++++++++++++++++++------------- 1 file changed, 23 insertions(+), 13 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index b249b1b857464..6aa7403ad80ac 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1735,33 +1735,43 @@ static int hdmi_add_cvt(struct hda_codec *codec, hda_nid_t cvt_nid) static int hdmi_parse_codec(struct hda_codec *codec) { - hda_nid_t nid; + hda_nid_t start_nid; + unsigned int caps; int i, nodes; - nodes = snd_hda_get_sub_nodes(codec, codec->core.afg, &nid); - if (!nid || nodes < 0) { + nodes = snd_hda_get_sub_nodes(codec, codec->core.afg, &start_nid); + if (!start_nid || nodes < 0) { codec_warn(codec, "HDMI: failed to get afg sub nodes\n"); return -EINVAL; } - for (i = 0; i < nodes; i++, nid++) { - unsigned int caps; - unsigned int type; + /* + * hdmi_add_pin() assumes total amount of converters to + * be known, so first discover all converters + */ + for (i = 0; i < nodes; i++) { + hda_nid_t nid = start_nid + i; caps = get_wcaps(codec, nid); - type = get_wcaps_type(caps); if (!(caps & AC_WCAP_DIGITAL)) continue; - switch (type) { - case AC_WID_AUD_OUT: + if (get_wcaps_type(caps) == AC_WID_AUD_OUT) hdmi_add_cvt(codec, nid); - break; - case AC_WID_PIN: + } + + /* discover audio pins */ + for (i = 0; i < nodes; i++) { + hda_nid_t nid = start_nid + i; + + caps = get_wcaps(codec, nid); + + if (!(caps & AC_WCAP_DIGITAL)) + continue; + + if (get_wcaps_type(caps) == AC_WID_PIN) hdmi_add_pin(codec, nid); - break; - } } return 0; -- 2.25.1

2 2

[PATCH 0/3] ASoC: mediatek: mt8183: support machine driver for max98357b
by Tzung-Bi Shih 22 Jul '20

22 Jul '20

The series re-uses mt8183-mt6358-ts3a227-max98357.c to support machine driver with max98357b. The 1st patch enables left justified format from mt8183 audio platform. The 2nd patch adds document for the new proposed compatible string for max98357b. The 3rd patch supports machine driver with max98357b and uses left justified format for it. Tzung-Bi Shih (3): ASoC: mediatek: mt8183: support left justified format for I2S ASoC: dt-bindings: mt8183: add compatible string for using max98357b ASoC: mediatek: mt8183: support machine driver with max98357b .../sound/mt8183-mt6358-ts3a227-max98357.txt | 1 + sound/soc/mediatek/mt8183/mt8183-dai-i2s.c | 59 ++++++++++++++++--- .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 22 ++++++- 3 files changed, 73 insertions(+), 9 deletions(-) -- 2.28.0.rc0.105.gf9edc3c819-goog

2 4

[PATCH 0/7] SOF IMX fixes
by Daniel Baluta 22 Jul '20

22 Jul '20

From: Daniel Baluta <daniel.baluta(a)nxp.com> This patchseries contains a couple of SOF IMX fixes found during our first IMX SOF release. Daniel Baluta (7): ASoC: SOF: define INFO_ flags in dsp_ops for imx8 ASoC: SOF: imx: Use ARRAY_SIZE instead of hardcoded value ASoC: SOF: imx8: Fix ESAI DAI driver name for i.MX8/iMX8X ASoC: SOF: imx8m: Fix SAI DAI driver for i.MX8M ASoC: SOF: imx8: Add SAI dai driver for i.MX/i.MX8X ASoC: SOF: topology: Update SAI config bclk/fsync rate ASoC: SOF: pcm: Update rate/channels for SAI/ESAI DAIs sound/soc/sof/imx/imx8.c | 24 +++++++++++++++++++++--- sound/soc/sof/imx/imx8m.c | 4 ++-- sound/soc/sof/pcm.c | 8 ++++++++ sound/soc/sof/topology.c | 2 ++ 4 files changed, 33 insertions(+), 5 deletions(-) -- 2.17.1

2 8

[PATCH 1/3] ASoC: hdac_hda: call patch_ops.free() on probe error
by Kai Vehmanen 22 Jul '20

22 Jul '20

Add error handling for patch_ops in hdac_hda_codec_probe(). Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> --- sound/soc/codecs/hdac_hda.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/hdac_hda.c b/sound/soc/codecs/hdac_hda.c index 473efe9ef998..72bd779bf942 100644 --- a/sound/soc/codecs/hdac_hda.c +++ b/sound/soc/codecs/hdac_hda.c @@ -467,7 +467,7 @@ static int hdac_hda_codec_probe(struct snd_soc_component *component) ret = snd_hda_codec_parse_pcms(hcodec); if (ret < 0) { dev_err(&hdev->dev, "unable to map pcms to dai %d\n", ret); - goto error_regmap; + goto error_patch; } /* HDMI controls need to be created in machine drivers */ @@ -476,7 +476,7 @@ static int hdac_hda_codec_probe(struct snd_soc_component *component) if (ret < 0) { dev_err(&hdev->dev, "unable to create controls %d\n", ret); - goto error_regmap; + goto error_patch; } } @@ -496,6 +496,9 @@ static int hdac_hda_codec_probe(struct snd_soc_component *component) return 0; +error_patch: + if (hcodec->patch_ops.free) + hcodec->patch_ops.free(hcodec); error_regmap: snd_hdac_regmap_exit(hdev); error_pm: -- 2.27.0

2 3

[PATCH for v5.9] ASoC: Replace HTTP links with HTTPS ones
by Alexander A. Klimov 22 Jul '20

22 Jul '20

Rationale: Reduces attack surface on kernel devs opening the links for MITM as HTTPS traffic is much harder to manipulate. Deterministic algorithm: For each file: If not .svg: For each line: If doesn't contain `\bxmlns\b`: For each link, `\bhttp://[^# \t\r\n]*(?:\w|/)`: If neither `\bgnu\.org/license`, nor `\bmozilla\.org/MPL\b`: If both the HTTP and HTTPS versions return 200 OK and serve the same content: Replace HTTP with HTTPS. Signed-off-by: Alexander A. Klimov <grandmaster(a)al2klimov.de> --- Continuing my work started at 93431e0607e5. See also: git log --oneline '--author=Alexander A. Klimov <grandmaster(a)al2klimov.de>' v5.7..master (Actually letting a shell for loop submit all this stuff for me.) If there are any URLs to be removed completely or at least not (just) HTTPSified: Just clearly say so and I'll *undo my change*. See also: https://lkml.org/lkml/2020/6/27/64 If there are any valid, but yet not changed URLs: See: https://lkml.org/lkml/2020/6/26/837 If you apply the patch, please let me know. Sorry again to all maintainers who complained about subject lines. Now I realized that you want an actually perfect prefixes, not just subsystem ones. I tried my best... And yes, *I could* (at least half-)automate it. Impossible is nothing! :) Documentation/devicetree/bindings/sound/adi,adau1977.txt | 6 +++--- Documentation/devicetree/bindings/sound/tas2552.txt | 2 +- Documentation/devicetree/bindings/sound/tas5720.txt | 6 +++--- Documentation/devicetree/bindings/sound/ti,tas6424.txt | 2 +- Documentation/devicetree/bindings/sound/tlv320adcx140.yaml | 6 +++--- Documentation/sound/soc/dai.rst | 2 +- sound/soc/cirrus/ep93xx-ac97.c | 2 +- sound/soc/codecs/hdmi-codec.c | 2 +- sound/soc/codecs/max9850.c | 2 +- sound/soc/codecs/mc13783.c | 2 +- sound/soc/codecs/pcm186x-i2c.c | 2 +- sound/soc/codecs/pcm186x-spi.c | 2 +- sound/soc/codecs/pcm186x.c | 2 +- sound/soc/codecs/pcm186x.h | 2 +- sound/soc/codecs/tas2552.c | 2 +- sound/soc/codecs/tas2552.h | 2 +- sound/soc/codecs/tas2562.h | 2 +- sound/soc/codecs/tas2770.c | 2 +- sound/soc/codecs/tas2770.h | 2 +- sound/soc/codecs/tas5720.c | 2 +- sound/soc/codecs/tas5720.h | 2 +- sound/soc/codecs/tas6424.c | 2 +- sound/soc/codecs/tas6424.h | 2 +- sound/soc/codecs/tlv320adcx140.c | 2 +- sound/soc/codecs/tlv320adcx140.h | 2 +- sound/soc/codecs/tlv320aic31xx.c | 4 ++-- sound/soc/codecs/tlv320aic31xx.h | 2 +- 27 files changed, 34 insertions(+), 34 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/adi,adau1977.txt b/Documentation/devicetree/bindings/sound/adi,adau1977.txt index 9225472c80b4..37f8aad01203 100644 --- a/Documentation/devicetree/bindings/sound/adi,adau1977.txt +++ b/Documentation/devicetree/bindings/sound/adi,adau1977.txt @@ -1,9 +1,9 @@ Analog Devices ADAU1977/ADAU1978/ADAU1979 Datasheets: -http://www.analog.com/media/en/technical-documentation/data-sheets/ADAU1977.pdf -http://www.analog.com/media/en/technical-documentation/data-sheets/ADAU1978.pdf -http://www.analog.com/media/en/technical-documentation/data-sheets/ADAU1979.pdf +https://www.analog.com/media/en/technical-documentation/data-sheets/ADAU1977.pdf +https://www.analog.com/media/en/technical-documentation/data-sheets/ADAU1978.pdf +https://www.analog.com/media/en/technical-documentation/data-sheets/ADAU1979.pdf This driver supports both the I2C and SPI bus. diff --git a/Documentation/devicetree/bindings/sound/tas2552.txt b/Documentation/devicetree/bindings/sound/tas2552.txt index 2d71eb05c1d3..a7eecad83db1 100644 --- a/Documentation/devicetree/bindings/sound/tas2552.txt +++ b/Documentation/devicetree/bindings/sound/tas2552.txt @@ -33,4 +33,4 @@ tas2552: tas2552@41 { }; For more product information please see the link below: -http://www.ti.com/product/TAS2552 +https://www.ti.com/product/TAS2552 diff --git a/Documentation/devicetree/bindings/sound/tas5720.txt b/Documentation/devicetree/bindings/sound/tas5720.txt index 7481653fe8e3..df99ca9451b0 100644 --- a/Documentation/devicetree/bindings/sound/tas5720.txt +++ b/Documentation/devicetree/bindings/sound/tas5720.txt @@ -4,9 +4,9 @@ The TAS5720 serial control bus communicates through the I2C protocol only. The serial bus is also used for periodic codec fault checking/reporting during audio playback. For more product information please see the links below: -http://www.ti.com/product/TAS5720L -http://www.ti.com/product/TAS5720M -http://www.ti.com/product/TAS5722L +https://www.ti.com/product/TAS5720L +https://www.ti.com/product/TAS5720M +https://www.ti.com/product/TAS5722L Required properties: diff --git a/Documentation/devicetree/bindings/sound/ti,tas6424.txt b/Documentation/devicetree/bindings/sound/ti,tas6424.txt index eacb54f34188..00940c489299 100644 --- a/Documentation/devicetree/bindings/sound/ti,tas6424.txt +++ b/Documentation/devicetree/bindings/sound/ti,tas6424.txt @@ -19,4 +19,4 @@ tas6424: tas6424@6a { }; For more product information please see the link below: -http://www.ti.com/product/TAS6424-Q1 +https://www.ti.com/product/TAS6424-Q1 diff --git a/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml b/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml index 2e6ac5d2ee96..8e008b7cf926 100644 --- a/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml +++ b/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml @@ -18,9 +18,9 @@ description: | microphone bias or supply voltage generation. Specifications can be found at: - http://www.ti.com/lit/ds/symlink/tlv320adc3140.pdf - http://www.ti.com/lit/ds/symlink/tlv320adc5140.pdf - http://www.ti.com/lit/ds/symlink/tlv320adc6140.pdf + https://www.ti.com/lit/ds/symlink/tlv320adc3140.pdf + https://www.ti.com/lit/ds/symlink/tlv320adc5140.pdf + https://www.ti.com/lit/ds/symlink/tlv320adc6140.pdf properties: compatible: diff --git a/Documentation/sound/soc/dai.rst b/Documentation/sound/soc/dai.rst index 2e99183a7a47..009b07e5a0f3 100644 --- a/Documentation/sound/soc/dai.rst +++ b/Documentation/sound/soc/dai.rst @@ -17,7 +17,7 @@ frame (FRAME) (usually 48kHz) is always driven by the controller. Each AC97 frame is 21uS long and is divided into 13 time slots. The AC97 specification can be found at : -http://www.intel.com/p/en_US/business/design +https://www.intel.com/p/en_US/business/design I2S diff --git a/sound/soc/cirrus/ep93xx-ac97.c b/sound/soc/cirrus/ep93xx-ac97.c index 1c45fb9ff990..16f9bb283b5c 100644 --- a/sound/soc/cirrus/ep93xx-ac97.c +++ b/sound/soc/cirrus/ep93xx-ac97.c @@ -285,7 +285,7 @@ static int ep93xx_ac97_trigger(struct snd_pcm_substream *substream, /* * As per Cirrus EP93xx errata described below: * - * http://www.cirrus.com/en/pubs/errata/ER667E2B.pdf + * https://www.cirrus.com/en/pubs/errata/ER667E2B.pdf * * we will wait for the TX FIFO to be empty before * clearing the TEN bit. diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index f005751da2cc..86c636ee312a 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -1,7 +1,7 @@ // SPDX-License-Identifier: GPL-2.0-only /* * ALSA SoC codec for HDMI encoder drivers - * Copyright (C) 2015 Texas Instruments Incorporated - http://www.ti.com/ + * Copyright (C) 2015 Texas Instruments Incorporated - https://www.ti.com/ * Author: Jyri Sarha <jsarha(a)ti.com> */ #include <linux/module.h> diff --git a/sound/soc/codecs/max9850.c b/sound/soc/codecs/max9850.c index 6f43748f9239..4659b8c6e746 100644 --- a/sound/soc/codecs/max9850.c +++ b/sound/soc/codecs/max9850.c @@ -7,7 +7,7 @@ * Author: Christian Glindkamp <christian.glindkamp(a)taskit.de> * * Initial development of this code was funded by - * MICRONIC Computer Systeme GmbH, http://www.mcsberlin.de/ + * MICRONIC Computer Systeme GmbH, https://www.mcsberlin.de/ */ #include <linux/module.h> diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c index f9830bd3da18..9e6a0cda43d0 100644 --- a/sound/soc/codecs/mc13783.c +++ b/sound/soc/codecs/mc13783.c @@ -5,7 +5,7 @@ * Copyright 2012 Philippe Retornaz, philippe.retornaz(a)epfl.ch * * Initial development of this code was funded by - * Phytec Messtechnik GmbH, http://www.phytec.de + * Phytec Messtechnik GmbH, https://www.phytec.de */ #include <linux/module.h> #include <linux/device.h> diff --git a/sound/soc/codecs/pcm186x-i2c.c b/sound/soc/codecs/pcm186x-i2c.c index 0214dc6d84d0..f8382b74391d 100644 --- a/sound/soc/codecs/pcm186x-i2c.c +++ b/sound/soc/codecs/pcm186x-i2c.c @@ -2,7 +2,7 @@ /* * Texas Instruments PCM186x Universal Audio ADC - I2C * - * Copyright (C) 2015-2017 Texas Instruments Incorporated - http://www.ti.com + * Copyright (C) 2015-2017 Texas Instruments Incorporated - https://www.ti.com * Andreas Dannenberg <dannenberg(a)ti.com> * Andrew F. Davis <afd(a)ti.com> */ diff --git a/sound/soc/codecs/pcm186x-spi.c b/sound/soc/codecs/pcm186x-spi.c index b56e19827497..bc1b0f0698ed 100644 --- a/sound/soc/codecs/pcm186x-spi.c +++ b/sound/soc/codecs/pcm186x-spi.c @@ -2,7 +2,7 @@ /* * Texas Instruments PCM186x Universal Audio ADC - SPI * - * Copyright (C) 2015-2017 Texas Instruments Incorporated - http://www.ti.com + * Copyright (C) 2015-2017 Texas Instruments Incorporated - https://www.ti.com * Andreas Dannenberg <dannenberg(a)ti.com> * Andrew F. Davis <afd(a)ti.com> */ diff --git a/sound/soc/codecs/pcm186x.c b/sound/soc/codecs/pcm186x.c index c5fcc632f670..f0da55901dcb 100644 --- a/sound/soc/codecs/pcm186x.c +++ b/sound/soc/codecs/pcm186x.c @@ -2,7 +2,7 @@ /* * Texas Instruments PCM186x Universal Audio ADC * - * Copyright (C) 2015-2017 Texas Instruments Incorporated - http://www.ti.com + * Copyright (C) 2015-2017 Texas Instruments Incorporated - https://www.ti.com * Andreas Dannenberg <dannenberg(a)ti.com> * Andrew F. Davis <afd(a)ti.com> */ diff --git a/sound/soc/codecs/pcm186x.h b/sound/soc/codecs/pcm186x.h index bb3f0c42a1cd..4d493754a3e2 100644 --- a/sound/soc/codecs/pcm186x.h +++ b/sound/soc/codecs/pcm186x.h @@ -2,7 +2,7 @@ /* * Texas Instruments PCM186x Universal Audio ADC * - * Copyright (C) 2015-2017 Texas Instruments Incorporated - http://www.ti.com + * Copyright (C) 2015-2017 Texas Instruments Incorporated - https://www.ti.com * Andreas Dannenberg <dannenberg(a)ti.com> * Andrew F. Davis <afd(a)ti.com> */ diff --git a/sound/soc/codecs/tas2552.c b/sound/soc/codecs/tas2552.c index d90e5f2b6f27..a94ad2580729 100644 --- a/sound/soc/codecs/tas2552.c +++ b/sound/soc/codecs/tas2552.c @@ -2,7 +2,7 @@ /* * tas2552.c - ALSA SoC Texas Instruments TAS2552 Mono Audio Amplifier * - * Copyright (C) 2014 Texas Instruments Incorporated - http://www.ti.com + * Copyright (C) 2014 Texas Instruments Incorporated - https://www.ti.com * * Author: Dan Murphy <dmurphy(a)ti.com> */ diff --git a/sound/soc/codecs/tas2552.h b/sound/soc/codecs/tas2552.h index d0958315d6a2..b9c2e70df57e 100644 --- a/sound/soc/codecs/tas2552.h +++ b/sound/soc/codecs/tas2552.h @@ -2,7 +2,7 @@ /* * tas2552.h - ALSA SoC Texas Instruments TAS2552 Mono Audio Amplifier * - * Copyright (C) 2014 Texas Instruments Incorporated - http://www.ti.com + * Copyright (C) 2014 Texas Instruments Incorporated - https://www.ti.com * * Author: Dan Murphy <dmurphy(a)ti.com> */ diff --git a/sound/soc/codecs/tas2562.h b/sound/soc/codecs/tas2562.h index 28e75fc431d0..61f22b45fe1b 100644 --- a/sound/soc/codecs/tas2562.h +++ b/sound/soc/codecs/tas2562.h @@ -2,7 +2,7 @@ /* * tas2562.h - ALSA SoC Texas Instruments TAS2562 Mono Audio Amplifier * - * Copyright (C) 2019 Texas Instruments Incorporated - http://www.ti.com + * Copyright (C) 2019 Texas Instruments Incorporated - https://www.ti.com * * Author: Dan Murphy <dmurphy(a)ti.com> */ diff --git a/sound/soc/codecs/tas2770.c b/sound/soc/codecs/tas2770.c index 54c8135fe43c..ad660d1a5ed9 100644 --- a/sound/soc/codecs/tas2770.c +++ b/sound/soc/codecs/tas2770.c @@ -3,7 +3,7 @@ // ALSA SoC Texas Instruments TAS2770 20-W Digital Input Mono Class-D // Audio Amplifier with Speaker I/V Sense // -// Copyright (C) 2016-2017 Texas Instruments Incorporated - http://www.ti.com/ +// Copyright (C) 2016-2017 Texas Instruments Incorporated - https://www.ti.com/ // Author: Tracy Yi <tracy-yi(a)ti.com> // Frank Shi <shifu0704(a)thundersoft.com> diff --git a/sound/soc/codecs/tas2770.h b/sound/soc/codecs/tas2770.h index cbb858369fe6..96683971ee9b 100644 --- a/sound/soc/codecs/tas2770.h +++ b/sound/soc/codecs/tas2770.h @@ -2,7 +2,7 @@ * * ALSA SoC TAS2770 codec driver * - * Copyright (C) 2016-2017 Texas Instruments Incorporated - http://www.ti.com/ + * Copyright (C) 2016-2017 Texas Instruments Incorporated - https://www.ti.com/ */ #ifndef __TAS2770__ #define __TAS2770__ diff --git a/sound/soc/codecs/tas5720.c b/sound/soc/codecs/tas5720.c index 37fab8f22800..616c8ab89c68 100644 --- a/sound/soc/codecs/tas5720.c +++ b/sound/soc/codecs/tas5720.c @@ -2,7 +2,7 @@ /* * tas5720.c - ALSA SoC Texas Instruments TAS5720 Mono Audio Amplifier * - * Copyright (C)2015-2016 Texas Instruments Incorporated - http://www.ti.com + * Copyright (C)2015-2016 Texas Instruments Incorporated - https://www.ti.com * * Author: Andreas Dannenberg <dannenberg(a)ti.com> */ diff --git a/sound/soc/codecs/tas5720.h b/sound/soc/codecs/tas5720.h index 93079f954f09..223858f0de71 100644 --- a/sound/soc/codecs/tas5720.h +++ b/sound/soc/codecs/tas5720.h @@ -2,7 +2,7 @@ /* * tas5720.h - ALSA SoC Texas Instruments TAS5720 Mono Audio Amplifier * - * Copyright (C)2015-2016 Texas Instruments Incorporated - http://www.ti.com + * Copyright (C)2015-2016 Texas Instruments Incorporated - https://www.ti.com * * Author: Andreas Dannenberg <dannenberg(a)ti.com> */ diff --git a/sound/soc/codecs/tas6424.c b/sound/soc/codecs/tas6424.c index aaba39295079..3e6a77d60a2f 100644 --- a/sound/soc/codecs/tas6424.c +++ b/sound/soc/codecs/tas6424.c @@ -2,7 +2,7 @@ /* * ALSA SoC Texas Instruments TAS6424 Quad-Channel Audio Amplifier * - * Copyright (C) 2016-2017 Texas Instruments Incorporated - http://www.ti.com/ + * Copyright (C) 2016-2017 Texas Instruments Incorporated - https://www.ti.com/ * Author: Andreas Dannenberg <dannenberg(a)ti.com> * Andrew F. Davis <afd(a)ti.com> */ diff --git a/sound/soc/codecs/tas6424.h b/sound/soc/codecs/tas6424.h index c67a7835ca66..a6a0d00e5190 100644 --- a/sound/soc/codecs/tas6424.h +++ b/sound/soc/codecs/tas6424.h @@ -2,7 +2,7 @@ /* * ALSA SoC Texas Instruments TAS6424 Quad-Channel Audio Amplifier * - * Copyright (C) 2016-2017 Texas Instruments Incorporated - http://www.ti.com/ + * Copyright (C) 2016-2017 Texas Instruments Incorporated - https://www.ti.com/ * Author: Andreas Dannenberg <dannenberg(a)ti.com> * Andrew F. Davis <afd(a)ti.com> */ diff --git a/sound/soc/codecs/tlv320adcx140.c b/sound/soc/codecs/tlv320adcx140.c index 35fe8ee5bce9..0c3b08d1d749 100644 --- a/sound/soc/codecs/tlv320adcx140.c +++ b/sound/soc/codecs/tlv320adcx140.c @@ -1,6 +1,6 @@ // SPDX-License-Identifier: GPL-2.0 // TLV320ADCX140 Sound driver -// Copyright (C) 2020 Texas Instruments Incorporated - http://www.ti.com/ +// Copyright (C) 2020 Texas Instruments Incorporated - https://www.ti.com/ #include <linux/module.h> #include <linux/moduleparam.h> diff --git a/sound/soc/codecs/tlv320adcx140.h b/sound/soc/codecs/tlv320adcx140.h index 39206bf1af12..ab3fec866ae9 100644 --- a/sound/soc/codecs/tlv320adcx140.h +++ b/sound/soc/codecs/tlv320adcx140.h @@ -1,6 +1,6 @@ // SPDX-License-Identifier: GPL-2.0 // TLV320ADCX104 Sound driver -// Copyright (C) 2020 Texas Instruments Incorporated - http://www.ti.com/ +// Copyright (C) 2020 Texas Instruments Incorporated - https://www.ti.com/ #ifndef _TLV320ADCX140_H #define _TLV320ADCX140_H diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index 31daa60695bd..9d4063138e22 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -2,7 +2,7 @@ /* * ALSA SoC TLV320AIC31xx CODEC Driver * - * Copyright (C) 2014-2017 Texas Instruments Incorporated - http://www.ti.com/ + * Copyright (C) 2014-2017 Texas Instruments Incorporated - https://www.ti.com/ * Jyri Sarha <jsarha(a)ti.com> * * Based on ground work by: Ajit Kulkarni <x0175765(a)ti.com> @@ -877,7 +877,7 @@ static int aic31xx_setup_pll(struct snd_soc_component *component, there may be trouble. To fix the issue edit the aic31xx_divs table for your mclk and sample rate. Details can be found from: - http://www.ti.com/lit/ds/symlink/tlv320aic3100.pdf + https://www.ti.com/lit/ds/symlink/tlv320aic3100.pdf Section: 5.6 CLOCK Generation and PLL */ } diff --git a/sound/soc/codecs/tlv320aic31xx.h b/sound/soc/codecs/tlv320aic31xx.h index 0523884cee74..81952984613d 100644 --- a/sound/soc/codecs/tlv320aic31xx.h +++ b/sound/soc/codecs/tlv320aic31xx.h @@ -2,7 +2,7 @@ /* * ALSA SoC TLV320AIC31xx CODEC Driver Definitions * - * Copyright (C) 2014-2017 Texas Instruments Incorporated - http://www.ti.com/ + * Copyright (C) 2014-2017 Texas Instruments Incorporated - https://www.ti.com/ */ #ifndef _TLV320AIC31XX_H -- 2.27.0

3 2

Several issues with HP OMEN dc-0019-ur
by GitHub issues - edited 22 Jul '20

22 Jul '20

alsa-project/alsa-lib issue #73 was edited from UnicodeMan: Hi. I am using HP OMEN dc0019ur laptop with linux 5.7-xanmod kernel and KDE neon system. Laptop has combined 3.5mm jack and a seperate stereo microphone jack. Tried various quirks by creating /etc/modprobe.d/mute.conf file and adding line "options snd-hda-intel model=XX", where XX is the name of the quirk from documentation here: https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/tree/Doc… With no modprobe changes: mute led does not work, headphones work, but microphone is not detected, and if i plug anything into microphone jack, it switches to microphone in combined jack. With hp-line1-mic1 quirk: mute led works fine, headphones are detected, except microphone, though it works anyway. Seperate microphone jack does not work at all. With headset-mic quirk: mute led does not work, headphones jack works, microphone (from headphones jack) is not detected when present but works when switched to anyway Microphone jack works perfectly (detects a plugged in mic, works in stereo). --- alsa-info.sh logs --- no quirks: http://alsa-project.org/db/?f=a73a83ba0610e3fd88310999b197fdde4372249a hp-line1-mic1: http://alsa-project.org/db/?f=f4d08dec7918c79a391fb1898d50c57a2448d872 headset-mic: https://alsa-project.org/db/?f=4b1eeef821f39eea28071ac2f2c0babcf515ef14 Not sure if it matters, but I want to add that I'm using nvidia render offload with reverse PRIME support, the gpu has one DP and one HDMI port wired directly to it. aaand, I hope I am writing this issue in correct place :D Issue URL : https://github.com/alsa-project/alsa-lib/issues/73 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Several issues with HP OMEN dc-0019-ur
by GitHub issues - edited 22 Jul '20

22 Jul '20

alsa-project/alsa-lib issue #73 was edited from UnicodeMan: Hi. I am using HP OMEN dc0019ur laptop with linux 5.7-xanmod kernel and KDE neon system. Laptop has combined 3.5mm jack and a seperate stereo microphone jack. Tried various quirks by creating /etc/modprobe.d/mute.conf file and adding line "options snd-hda-intel model=XX", where XX is the name of the quirk from documentation here: https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/tree/Doc… With no modprobe changes: mute led does not work, headphones work, but microphone is not detected, and if i plug anything into microphone jack, it switches to microphone in combined jack. With hp-line1-mic1 quirk: mute led works fine, headphones are detected, except microphone, though it works anyway. Seperate microphone jack does not work at all. With headset-mic quirk: mute led does not work, headphones jack works, microphone (from headphones jack) is not detected when present but works when switched to anyway Microphone jack works perfectly (detects a plugged in mic, works in stereo). --- alsa-info.sh logs --- no quirks: http://alsa-project.org/db/?f=a73a83ba0610e3fd88310999b197fdde4372249a hp-line1-mic1: http://alsa-project.org/db/?f=f4d08dec7918c79a391fb1898d50c57a2448d872 headset-mic: https://alsa-project.org/db/?f=4b1eeef821f39eea28071ac2f2c0babcf515ef14 aaand, I hope I am writing this issue in correct place :D Issue URL : https://github.com/alsa-project/alsa-lib/issues/73 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Several issues with HP OMEN dc-0019-ur
by GitHub issues - opened 22 Jul '20

22 Jul '20

alsa-project/alsa-lib issue #73 was opened from UnicodeMan: Hi. I am using HP OMEN dc0019ur laptop with linux 5.7-xanmod kernel and KDE neon system. Laptop has combined 3.5mm jack and a seperate stereo microphone jack. With no modprobe changes: mute led does not work, headphones work, but microphone is not detected, and if i plug anything into microphone jack, it switches to microphone in combined jack. With hp-line1-mic1 quirk: mute led works fine, headphones are detected, except microphone, though it works anyway. Seperate microphone jack does not work at all. With headset-mic quirk: mute led does not work, headphones jack works, microphone (from headphones jack) is not detected when present but works when switched to anyway Microphone jack works perfectly (detects a plugged in mic, works in stereo). no quirks: http://alsa-project.org/db/?f=a73a83ba0610e3fd88310999b197fdde4372249a hp-line1-mic1: http://alsa-project.org/db/?f=f4d08dec7918c79a391fb1898d50c57a2448d872 headset-mic: https://alsa-project.org/db/?f=4b1eeef821f39eea28071ac2f2c0babcf515ef14 aaand, I hope I am writing this issue in correct place :D Issue URL : https://github.com/alsa-project/alsa-lib/issues/73 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH v3 0/8] ASoC: qcom: Add support for SC7180 lpass variant
by Rohit kumar 22 Jul '20

22 Jul '20

This patch chain add audio support for SC7180 soc by doing the required modification in existing common lpass-cpu/lpass-platform driver. Below is a brief summary of patch series: PATCH v3 0001 ... 0005: Update lpass-cpu, lpass-platform drivers to make it more generic and support newer soc registers configuration. This also updates existing lpass-apq8096.c and lpass-ipq806x.c. PATCH v2 0005 ... 0007: Add documentation and platform driver for newer SC7180 SOC variant. Changes since v2: - Moved yaml conversion of Documentation to the end of patch series - Used REG_FIELD_ID instead of REG_FIELD for DMACTL and I2SCTL registers. Move reg_fields to struct lpass_variant as suggested by Srinivas. Ajit Pandey (5): ASoC: qcom: Add common array to initialize soc based core clocks include: dt-bindings: sound: Add sc7180-lpass bindings header ASoC: qcom: lpass-platform: Replace card->dev with component->dev ASoC: qcom: lpass-sc7180: Add platform driver for lpass audio dt-bindings: sound: lpass-cpu: Move to yaml format Rohit kumar (3): ASoC: qcom: lpass-cpu: Move ahbix clk to platform specific function ASoC: qcom: lpass: Use regmap_field for i2sctl and dmactl registers dt-bindings: sound: lpass-cpu: Add sc7180 lpass cpu node .../devicetree/bindings/sound/qcom,lpass-cpu.txt | 79 -------- .../devicetree/bindings/sound/qcom,lpass-cpu.yaml | 154 +++++++++++++++ include/dt-bindings/sound/sc7180-lpass.h | 10 + sound/soc/qcom/Kconfig | 5 + sound/soc/qcom/Makefile | 2 + sound/soc/qcom/lpass-apq8016.c | 86 ++++++-- sound/soc/qcom/lpass-cpu.c | 193 +++++++++--------- sound/soc/qcom/lpass-ipq806x.c | 67 +++++++ sound/soc/qcom/lpass-lpaif-reg.h | 157 ++++++++------- sound/soc/qcom/lpass-platform.c | 156 +++++++++++---- sound/soc/qcom/lpass-sc7180.c | 216 +++++++++++++++++++++ sound/soc/qcom/lpass.h | 63 +++++- 12 files changed, 886 insertions(+), 302 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/qcom,lpass-cpu.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,lpass-cpu.yaml create mode 100644 include/dt-bindings/sound/sc7180-lpass.h create mode 100644 sound/soc/qcom/lpass-sc7180.c -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

4 28

Fwd: Fwd: [PATCH] Add implicit feedback quirk for SSL2. (from laurie@tratt.net)
by Dmitry Pavlushko 22 Jul '20

22 Jul '20

Hi devs, Sorry for the interruption. Any chance you can push the line for SSL2 usb id? Thanks, Dmitry -------- Forwarded Message -------- Subject: Fwd: [PATCH] Add implicit feedback quirk for SSL2. (from laurie(a)tratt.net) Date: Mon, 15 Jun 2020 10:36:39 +0100 From: Laurence Tratt <laurie(a)tratt.net> To: dpavlushko(a)gmail.com I forgot to CC you. Sorry! Laurie ----- Forwarded message from Laurence Tratt <laurie(a)tratt.net> ----- From: Laurence Tratt <laurie(a)tratt.net> To: alsa-devel(a)alsa-project.org Date: Mon, 15 Jun 2020 08:30:23 +0100 Subject: [PATCH] Add implicit feedback quirk for SSL2. As expected, this requires the same quirk as the SSL2+ in order for the clock to sync. This was suggested by, and tested on an SSL2, by Dmitry. Signed-off-by: Dmitry <dpavlushko(a)gmail.com> Signed-off-by: Laurence Tratt <laurie(a)tratt.net> --- sound/usb/pcm.c | 1 + 1 file changed, 1 insertion(+) diff --git sound/usb/pcm.c sound/usb/pcm.c index 84c0ae4319..dc1608bdf6 100644 --- sound/usb/pcm.c +++ sound/usb/pcm.c @@ -367,6 +367,7 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs, ifnum = 0; goto add_sync_ep_from_ifnum; case USB_ID(0x07fd, 0x0008): /* MOTU M Series */ + case USB_ID(0x31e9, 0x0001): /* Solid State Logic SSL2 */ case USB_ID(0x31e9, 0x0002): /* Solid State Logic SSL2+ */ ep = 0x81; ifnum = 2; -- 2.27.0 ----- End forwarded message -----

2 1

Re: 14216-add-prevent-wakeup-from-s3-by-lenovo-Thinkcentre-TI
by gregkh 22 Jul '20

22 Jul '20

> 此电子邮件消息仅供预期收件人使用，其中可能包含保密或特权使用信息。如果您不是预 > 期收件人，请勿使用、传播、分发或复制此电子邮件或信赖此邮件采取任何行动。如果您 > 误收了此邮件，请立即回复邮件通知统信软件技术有限公司发件人，并删除误收电子邮件 > 及其相关附件。感谢配合！ > > This email message is intended only for the use of the individual or entity who > /which is the intended recipient and may contain information that is privileged > or confidential. If you are not the intended recipient, you are hereby notified > that any use, dissemination, distribution or copying of, or taking any action > in reliance on, this e-mail is strictly prohibited. If you have received this > email in error, please notify UnionTech Software Technology immediately by > replying to this e-mail and immediately delete and discard all copies of the > e-mail and the attachment thereto (if any). Thank you. This is not compatible with kernel development, sorry, now deleted. greg k-h

1 0

[PATCH] ASoC: max98357a: move control of SD_MODE back to DAI ops
by Tzung-Bi Shih 22 Jul '20

22 Jul '20

Partially reverts commit 128f825aeab7 ("ASoC: max98357a: move control of SD_MODE to DAPM"). In order to have mute control of max98357 from machine drivers, commit 128f825aeab7 ("ASoC: max98357a: move control of SD_MODE to DAPM") moves the control of SD_MODE from DAI ops to DAPM events. However, pop noise has been observed on rk3399-gru-kevin boards due to this commit. The commit 128f825aeab7 caused sequence of DAI clocks and SD_MODE changed on rk3399-gru-kevin boards. With the commit 128f825aeab7: - SD_MODE will be set to 1 before DAI clocks start. - SD_MODE will be set to 0 after DAI clocks stop. As a result, pop noise. Moves the control of SD_MODE back to DAI ops. In the meantime, uses an additional flag in DAPM event to provide chance of mute control for machine drivers. Signed-off-by: Tzung-Bi Shih <tzungbi(a)google.com> --- This patch addresses the reported issue in the thread https://lkml.org/lkml/2020/7/16/427 sound/soc/codecs/max98357a.c | 50 ++++++++++++++++++++++++++++-------- 1 file changed, 40 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/max98357a.c b/sound/soc/codecs/max98357a.c index 4f431133d0bb..918812763884 100644 --- a/sound/soc/codecs/max98357a.c +++ b/sound/soc/codecs/max98357a.c @@ -23,36 +23,61 @@ struct max98357a_priv { struct gpio_desc *sdmode; unsigned int sdmode_delay; + int sdmode_switch; }; -static int max98357a_sdmode_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) +static int max98357a_daiops_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *dai) { - struct snd_soc_component *component = - snd_soc_dapm_to_component(w->dapm); + struct snd_soc_component *component = dai->component; struct max98357a_priv *max98357a = snd_soc_component_get_drvdata(component); if (!max98357a->sdmode) return 0; - if (event & SND_SOC_DAPM_POST_PMU) { - msleep(max98357a->sdmode_delay); - gpiod_set_value(max98357a->sdmode, 1); - dev_dbg(component->dev, "set sdmode to 1"); - } else if (event & SND_SOC_DAPM_PRE_PMD) { + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + mdelay(max98357a->sdmode_delay); + if (max98357a->sdmode_switch) { + gpiod_set_value(max98357a->sdmode, 1); + dev_dbg(component->dev, "set sdmode to 1"); + } + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: gpiod_set_value(max98357a->sdmode, 0); dev_dbg(component->dev, "set sdmode to 0"); + break; } return 0; } +static int max98357a_sdmode_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct max98357a_priv *max98357a = + snd_soc_component_get_drvdata(component); + + if (event & SND_SOC_DAPM_POST_PMU) + max98357a->sdmode_switch = 1; + else if (event & SND_SOC_DAPM_POST_PMD) + max98357a->sdmode_switch = 0; + + return 0; +} + static const struct snd_soc_dapm_widget max98357a_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("Speaker"), SND_SOC_DAPM_OUT_DRV_E("SD_MODE", SND_SOC_NOPM, 0, 0, NULL, 0, max98357a_sdmode_event, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), }; static const struct snd_soc_dapm_route max98357a_dapm_routes[] = { @@ -71,6 +96,10 @@ static const struct snd_soc_component_driver max98357a_component_driver = { .non_legacy_dai_naming = 1, }; +static const struct snd_soc_dai_ops max98357a_dai_ops = { + .trigger = max98357a_daiops_trigger, +}; + static struct snd_soc_dai_driver max98357a_dai_driver = { .name = "HiFi", .playback = { @@ -90,6 +119,7 @@ static struct snd_soc_dai_driver max98357a_dai_driver = { .channels_min = 1, .channels_max = 2, }, + .ops = &max98357a_dai_ops, }; static int max98357a_platform_probe(struct platform_device *pdev) -- 2.28.0.rc0.105.gf9edc3c819-goog

3 2

[PATCH] dt-bindings: sound: sgtl5000: Convert to json-schema
by Thierry Reding 22 Jul '20

22 Jul '20

From: Thierry Reding <treding(a)nvidia.com> Convert the Freescale SGTL5000 device tree bindings from free-form text format to json-schema. Signed-off-by: Thierry Reding <treding(a)nvidia.com> --- .../devicetree/bindings/sound/sgtl5000.txt | 60 ---------- .../devicetree/bindings/sound/sgtl5000.yaml | 103 ++++++++++++++++++ .../devicetree/bindings/trivial-devices.yaml | 2 - 3 files changed, 103 insertions(+), 62 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/sgtl5000.txt create mode 100644 Documentation/devicetree/bindings/sound/sgtl5000.yaml diff --git a/Documentation/devicetree/bindings/sound/sgtl5000.txt b/Documentation/devicetree/bindings/sound/sgtl5000.txt deleted file mode 100644 index 9d9ff5184939..000000000000 --- a/Documentation/devicetree/bindings/sound/sgtl5000.txt +++ /dev/null @@ -1,60 +0,0 @@ -* Freescale SGTL5000 Stereo Codec - -Required properties: -- compatible : "fsl,sgtl5000". - -- reg : the I2C address of the device - -- #sound-dai-cells: must be equal to 0 - -- clocks : the clock provider of SYS_MCLK - -- VDDA-supply : the regulator provider of VDDA - -- VDDIO-supply: the regulator provider of VDDIO - -Optional properties: - -- VDDD-supply : the regulator provider of VDDD - -- micbias-resistor-k-ohms : the bias resistor to be used in kOhms - The resistor can take values of 2k, 4k or 8k. - If set to 0 it will be off. - If this node is not mentioned or if the value is unknown, then - micbias resistor is set to 4K. - -- micbias-voltage-m-volts : the bias voltage to be used in mVolts - The voltage can take values from 1.25V to 3V by 250mV steps - If this node is not mentioned or the value is unknown, then - the value is set to 1.25V. - -- lrclk-strength: the LRCLK pad strength. Possible values are: -0, 1, 2 and 3 as per the table below: - -VDDIO 1.8V 2.5V 3.3V -0 = Disable -1 = 1.66 mA 2.87 mA 4.02 mA -2 = 3.33 mA 5.74 mA 8.03 mA -3 = 4.99 mA 8.61 mA 12.05 mA - -- sclk-strength: the SCLK pad strength. Possible values are: -0, 1, 2 and 3 as per the table below: - -VDDIO 1.8V 2.5V 3.3V -0 = Disable -1 = 1.66 mA 2.87 mA 4.02 mA -2 = 3.33 mA 5.74 mA 8.03 mA -3 = 4.99 mA 8.61 mA 12.05 mA - -Example: - -sgtl5000: codec@a { - compatible = "fsl,sgtl5000"; - reg = <0x0a>; - #sound-dai-cells = <0>; - clocks = <&clks 150>; - micbias-resistor-k-ohms = <2>; - micbias-voltage-m-volts = <2250>; - VDDA-supply = <&reg_3p3v>; - VDDIO-supply = <&reg_3p3v>; -}; diff --git a/Documentation/devicetree/bindings/sound/sgtl5000.yaml b/Documentation/devicetree/bindings/sound/sgtl5000.yaml new file mode 100644 index 000000000000..4f29b63c54d3 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/sgtl5000.yaml @@ -0,0 +1,103 @@ +# SPDX-License-Identifier: GPL-2.0-only +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/sgtl5000.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Freescale SGTL5000 Stereo Codec + +maintainers: + - Fabio Estevam <festevam(a)gmail.com> + +properties: + compatible: + const: fsl,sgtl5000 + + reg: + maxItems: 1 + + "#sound-dai-cells": + const: 0 + + clocks: + items: + - description: the clock provider of SYS_MCLK + + VDDA-supply: + description: the regulator provider of VDDA + + VDDIO-supply: + description: the regulator provider of VDDIO + + VDDD-supply: + description: the regulator provider of VDDD + + micbias-resistor-k-ohms: + description: The bias resistor to be used in kOhms. The resistor can take + values of 2k, 4k or 8k. If set to 0 it will be off. If this node is not + mentioned or if the value is unknown, then micbias resistor is set to + 4k. + $ref: "/schemas/types.yaml#/definitions/uint32" + enum: [ 0, 2, 4, 8 ] + + micbias-voltage-m-volts: + description: The bias voltage to be used in mVolts. The voltage can take + values from 1.25V to 3V by 250mV steps. If this node is not mentioned + or the value is unknown, then the value is set to 1.25V. + $ref: "/schemas/types.yaml#/definitions/uint32" + enum: [ 1250, 1500, 1750, 2000, 2250, 2500, 2750, 3000 ] + + lrclk-strength: + description: | + The LRCLK pad strength. Possible values are: 0, 1, 2 and 3 as per the + table below: + + VDDIO 1.8V 2.5V 3.3V + 0 = Disable + 1 = 1.66 mA 2.87 mA 4.02 mA + 2 = 3.33 mA 5.74 mA 8.03 mA + 3 = 4.99 mA 8.61 mA 12.05 mA + $ref: "/schemas/types.yaml#/definitions/uint32" + enum: [ 0, 1, 2, 3 ] + + sclk-strength: + description: | + The SCLK pad strength. Possible values are: 0, 1, 2 and 3 as per the + table below: + + VDDIO 1.8V 2.5V 3.3V + 0 = Disable + 1 = 1.66 mA 2.87 mA 4.02 mA + 2 = 3.33 mA 5.74 mA 8.03 mA + 3 = 4.99 mA 8.61 mA 12.05 mA + $ref: "/schemas/types.yaml#/definitions/uint32" + enum: [ 0, 1, 2, 3 ] + +required: + - compatible + - reg + - "#sound-dai-cells" + - clocks + - VDDA-supply + - VDDIO-supply + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + codec@a { + compatible = "fsl,sgtl5000"; + reg = <0x0a>; + #sound-dai-cells = <0>; + clocks = <&clks 150>; + micbias-resistor-k-ohms = <2>; + micbias-voltage-m-volts = <2250>; + VDDA-supply = <&reg_3p3v>; + VDDIO-supply = <&reg_3p3v>; + }; + }; +... diff --git a/Documentation/devicetree/bindings/trivial-devices.yaml b/Documentation/devicetree/bindings/trivial-devices.yaml index 4165352a590a..b7e94fe8643f 100644 --- a/Documentation/devicetree/bindings/trivial-devices.yaml +++ b/Documentation/devicetree/bindings/trivial-devices.yaml @@ -80,8 +80,6 @@ properties: - fsl,mpl3115 # MPR121: Proximity Capacitive Touch Sensor Controller - fsl,mpr121 - # SGTL5000: Ultra Low-Power Audio Codec - - fsl,sgtl5000 # G751: Digital Temperature Sensor and Thermal Watchdog with Two-Wire Interface - gmt,g751 # Infineon IR38064 Voltage Regulator -- 2.24.1

2 1

[PATCH v3 1/2] dt-bindings: tas2770: Convert tas2770 binding to yaml
by Dan Murphy 22 Jul '20

22 Jul '20

Convert the tas2770 binding to yaml format. Add in the reset-gpio to the binding as it is in the code but not documented in the binding. Reviewed-by: Rob Herring <robh(a)kernel.org> Signed-off-by: Dan Murphy <dmurphy(a)ti.com> --- .../devicetree/bindings/sound/tas2770.txt | 37 --------- .../devicetree/bindings/sound/tas2770.yaml | 76 +++++++++++++++++++ 2 files changed, 76 insertions(+), 37 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/tas2770.txt create mode 100644 Documentation/devicetree/bindings/sound/tas2770.yaml diff --git a/Documentation/devicetree/bindings/sound/tas2770.txt b/Documentation/devicetree/bindings/sound/tas2770.txt deleted file mode 100644 index ede6bb3d9637..000000000000 --- a/Documentation/devicetree/bindings/sound/tas2770.txt +++ /dev/null @@ -1,37 +0,0 @@ -Texas Instruments TAS2770 Smart PA - -The TAS2770 is a mono, digital input Class-D audio amplifier optimized for -efficiently driving high peak power into small loudspeakers. -Integrated speaker voltage and current sense provides for -real time monitoring of loudspeaker behavior. - -Required properties: - - - compatible: - Should contain "ti,tas2770". - - reg: - The i2c address. Should contain <0x4c>, <0x4d>,<0x4e>, or <0x4f>. - - #address-cells - Should be <1>. - - #size-cells - Should be <0>. - - ti,asi-format: - Sets TDM RX capture edge. 0->Rising; 1->Falling. - - ti,imon-slot-no:- TDM TX current sense time slot. - - ti,vmon-slot-no:- TDM TX voltage sense time slot. - -Optional properties: - -- interrupt-parent: the phandle to the interrupt controller which provides - the interrupt. -- interrupts: interrupt specification for data-ready. - -Examples: - - tas2770@4c { - compatible = "ti,tas2770"; - reg = <0x4c>; - #address-cells = <1>; - #size-cells = <0>; - interrupt-parent = <&msm_gpio>; - interrupts = <97 0>; - ti,asi-format = <0>; - ti,imon-slot-no = <0>; - ti,vmon-slot-no = <2>; - }; - diff --git a/Documentation/devicetree/bindings/sound/tas2770.yaml b/Documentation/devicetree/bindings/sound/tas2770.yaml new file mode 100644 index 000000000000..8192450d72dc --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tas2770.yaml @@ -0,0 +1,76 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +# Copyright (C) 2019-20 Texas Instruments Incorporated +%YAML 1.2 +--- +$id: "http://devicetree.org/schemas/sound/tas2770.yaml#" +$schema: "http://devicetree.org/meta-schemas/core.yaml#" + +title: Texas Instruments TAS2770 Smart PA + +maintainers: + - Shi Fu <shifu0704(a)thundersoft.com> + +description: | + The TAS2770 is a mono, digital input Class-D audio amplifier optimized for + efficiently driving high peak power into small loudspeakers. + Integrated speaker voltage and current sense provides for + real time monitoring of loudspeaker behavior. + +properties: + compatible: + enum: + - ti,tas2770 + + reg: + maxItems: 1 + description: | + I2C address of the device can be one of these 0x4c, 0x4d, 0x4e or 0x4f + + reset-gpio: + description: GPIO used to reset the device. + + interrupts: + maxItems: 1 + + ti,imon-slot-no: + $ref: /schemas/types.yaml#/definitions/uint32 + description: TDM TX current sense time slot. + + ti,vmon-slot-no: + $ref: /schemas/types.yaml#/definitions/uint32 + description: TDM TX voltage sense time slot. + + ti,asi-format: + $ref: /schemas/types.yaml#/definitions/uint32 + description: Sets TDM RX capture edge. + enum: + - 0 # Rising edge + - 1 # Falling edge + + '#sound-dai-cells': + const: 1 + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + i2c0 { + #address-cells = <1>; + #size-cells = <0>; + codec: codec@4c { + compatible = "ti,tas2770"; + reg = <0x4c>; + #sound-dai-cells = <1>; + interrupt-parent = <&gpio1>; + interrupts = <14>; + reset-gpio = <&gpio1 15 0>; + ti,imon-slot-no = <0>; + ti,vmon-slot-no = <2>; + }; + }; + -- 2.27.0

2 4

[PATCH 1/2] dt-bindings: tas2770: Convert tas2770 binding to yaml
by Dan Murphy 22 Jul '20

22 Jul '20

Convert the tas2770 binding to yaml format. Add in the reset-gpio to the binding as it is in the code but not documented in the binding. Signed-off-by: Dan Murphy <dmurphy(a)ti.com> --- .../devicetree/bindings/sound/tas2770.txt | 37 --------- .../devicetree/bindings/sound/tas2770.yaml | 76 +++++++++++++++++++ 2 files changed, 76 insertions(+), 37 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/tas2770.txt create mode 100644 Documentation/devicetree/bindings/sound/tas2770.yaml diff --git a/Documentation/devicetree/bindings/sound/tas2770.txt b/Documentation/devicetree/bindings/sound/tas2770.txt deleted file mode 100644 index ede6bb3d9637..000000000000 --- a/Documentation/devicetree/bindings/sound/tas2770.txt +++ /dev/null @@ -1,37 +0,0 @@ -Texas Instruments TAS2770 Smart PA - -The TAS2770 is a mono, digital input Class-D audio amplifier optimized for -efficiently driving high peak power into small loudspeakers. -Integrated speaker voltage and current sense provides for -real time monitoring of loudspeaker behavior. - -Required properties: - - - compatible: - Should contain "ti,tas2770". - - reg: - The i2c address. Should contain <0x4c>, <0x4d>,<0x4e>, or <0x4f>. - - #address-cells - Should be <1>. - - #size-cells - Should be <0>. - - ti,asi-format: - Sets TDM RX capture edge. 0->Rising; 1->Falling. - - ti,imon-slot-no:- TDM TX current sense time slot. - - ti,vmon-slot-no:- TDM TX voltage sense time slot. - -Optional properties: - -- interrupt-parent: the phandle to the interrupt controller which provides - the interrupt. -- interrupts: interrupt specification for data-ready. - -Examples: - - tas2770@4c { - compatible = "ti,tas2770"; - reg = <0x4c>; - #address-cells = <1>; - #size-cells = <0>; - interrupt-parent = <&msm_gpio>; - interrupts = <97 0>; - ti,asi-format = <0>; - ti,imon-slot-no = <0>; - ti,vmon-slot-no = <2>; - }; - diff --git a/Documentation/devicetree/bindings/sound/tas2770.yaml b/Documentation/devicetree/bindings/sound/tas2770.yaml new file mode 100644 index 000000000000..8c667fd37a57 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tas2770.yaml @@ -0,0 +1,76 @@ +# SPDX-License-Identifier: (GPL-2.0+ OR BSD-2-Clause) +# Copyright (C) 2019-20 Texas Instruments Incorporated +%YAML 1.2 +--- +$id: "http://devicetree.org/schemas/sound/tas2770.yaml#" +$schema: "http://devicetree.org/meta-schemas/core.yaml#" + +title: Texas Instruments TAS2770 Smart PA + +maintainers: + - Shi Fu <shifu0704(a)thundersoft.com> + +description: | + The TAS2770 is a mono, digital input Class-D audio amplifier optimized for + efficiently driving high peak power into small loudspeakers. + Integrated speaker voltage and current sense provides for + real time monitoring of loudspeaker behavior. + +properties: + compatible: + enum: + - ti,tas2770 + + reg: + maxItems: 1 + description: | + I2C address of the device can be one of these 0x4c, 0x4d, 0x4e or 0x4f + + reset-gpio: + description: GPIO used to reset the device. + + interrupts: + maxItems: 1 + + ti,imon-slot-no: + $ref: /schemas/types.yaml#/definitions/uint32 + description: TDM TX current sense time slot. + + ti,vmon-slot-no: + $ref: /schemas/types.yaml#/definitions/uint32 + description: TDM TX voltage sense time slot. + + ti,asi-format: + $ref: /schemas/types.yaml#/definitions/uint32 + description: Sets TDM RX capture edge. + enum: + - 0 # Rising edge + - 1 # Falling edge + + '#sound-dai-cells': + const: 1 + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + i2c0 { + #address-cells = <1>; + #size-cells = <0>; + codec: codec@4c { + compatible = "ti,tas2770"; + reg = <0x4c>; + #sound-dai-cells = <1>; + interrupt-parent = <&gpio1>; + interrupts = <14>; + reset-gpio = <&gpio1 15 0>; + ti,imon-slot-no = <0>; + ti,vmon-slot-no = <2>; + }; + }; + -- 2.26.2

3 3

[PATCH for-next v2 1/2] ASoC: tas2770: Fix reset gpio property name
by Dan Murphy 22 Jul '20

22 Jul '20

Fix the reset property name when allocating the GPIO descriptor. The gpiod_get_optional appends either the -gpio or -gpios suffix to the name. Fixes: 1a476abc723e6 ("tas2770: add tas2770 smart PA kernel driver") Signed-off-by: Dan Murphy <dmurphy(a)ti.com> --- sound/soc/codecs/tas2770.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/codecs/tas2770.c b/sound/soc/codecs/tas2770.c index 54c8135fe43c..cf071121c839 100644 --- a/sound/soc/codecs/tas2770.c +++ b/sound/soc/codecs/tas2770.c @@ -758,8 +758,7 @@ static int tas2770_i2c_probe(struct i2c_client *client, } } - tas2770->reset_gpio = devm_gpiod_get_optional(tas2770->dev, - "reset-gpio", + tas2770->reset_gpio = devm_gpiod_get_optional(tas2770->dev, "reset", GPIOD_OUT_HIGH); if (IS_ERR(tas2770->reset_gpio)) { if (PTR_ERR(tas2770->reset_gpio) == -EPROBE_DEFER) { -- 2.27.0

2 2

[PATCH v7] ASoC: Intel: kbl_rt5663_rt5514_max98927: Fix kabylake_ssp_fixup function
by Harsha Priya 22 Jul '20

22 Jul '20

kabylake_ssp_fixup function uses snd_soc_dpcm to identify the codecs DAIs. The hw parameters are changed based on the codec DAI, the stream is intended for. The earlier approach to get snd_soc_dpcm was using container_of() macro on snd_pcm_hw_params. The structures have been modified over time and snd_soc_dpcm does not have snd_pcm_hw_params as a reference but as a copy. This causes the current driver to crash when used. This patch changes the way snd_soc_dpcm is extracted. The snd_soc_pcm_runtime holds 2 dpcm instances (one for playback and one for capture). The 2 codecs on this SSP are dmic and speakers. One is for capture and one is for playback respectively. Based on the direction of the stream, the snd_soc_dpcm is extracted from the snd_soc_pcm_runtime structure. Tested for all use cases of the driver. Signed-off-by: Harsha Priya <harshapriya.n(a)intel.com> Signed-off-by: Vamshi Krishna Gopal <vamshi.krishna.gopal(a)intel.com> Tested-by: Lukasz Majczak <lma(a)semihalf.com> --- v1 -> v2: - Extract dmic from SSP0 as every BE should have own fixup function. v2 -> v3: - Restore naming in the dapm route table to not confuse with other drivers - Fixed indentations v3 -> v4: - Updated code and commit description according to solution proposed by Harsha v4 -> v5: - Cosmetic Changes v5 -> v6: - Dmic regression seen with v4 fixed - Using available routines for obtaining dpcm information v6 -> v7: - Updated comments - initilize rtd_dpcm variable - added break statement in the loop --- --- .../intel/boards/kbl_rt5663_rt5514_max98927.c | 38 ++++++++++++++----- 1 file changed, 29 insertions(+), 9 deletions(-) diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index 584e4f9cedc2..9f4b949cc39c 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -379,22 +379,42 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_interval *chan = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); - struct snd_soc_dpcm *dpcm = container_of( - params, struct snd_soc_dpcm, hw_params); - struct snd_soc_dai_link *fe_dai_link = dpcm->fe->dai_link; - struct snd_soc_dai_link *be_dai_link = dpcm->be->dai_link; + struct snd_soc_dpcm *dpcm, *rtd_dpcm = NULL; + + /* + * The following loop will be called only for playback stream + * In this platform, there is only one playback device on every SSP + */ + for_each_dpcm_fe(rtd, SNDRV_PCM_STREAM_PLAYBACK, dpcm) { + rtd_dpcm = dpcm; + break; + } + + /* + * This following loop will be called only for capture stream + * In this platform, there is only one capture device on every SSP + */ + for_each_dpcm_fe(rtd, SNDRV_PCM_STREAM_CAPTURE, dpcm) { + rtd_dpcm = dpcm; + break; + } + + /* + * The above 2 loops are mutually exclusive based on the strem direction, + * thus rtd_dpcm variable will never be overwritten + */ /* * The ADSP will convert the FE rate to 48k, stereo, 24 bit */ - if (!strcmp(fe_dai_link->name, "Kbl Audio Port") || - !strcmp(fe_dai_link->name, "Kbl Audio Headset Playback") || - !strcmp(fe_dai_link->name, "Kbl Audio Capture Port")) { + if (!strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio Port") || + !strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio Headset Playback") || + !strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio Capture Port")) { rate->min = rate->max = 48000; chan->min = chan->max = 2; snd_mask_none(fmt); snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE); - } else if (!strcmp(fe_dai_link->name, "Kbl Audio DMIC cap")) { + } else if (!strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio DMIC cap")) { if (params_channels(params) == 2 || DMIC_CH(dmic_constraints) == 2) chan->min = chan->max = 2; @@ -405,7 +425,7 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, * The speaker on the SSP0 supports S16_LE and not S24_LE. * thus changing the mask here */ - if (!strcmp(be_dai_link->name, "SSP0-Codec")) + if (!strcmp(rtd_dpcm->be->dai_link->name, "SSP0-Codec")) snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S16_LE); return 0; -- 2.17.1

2 1

Speaker pops with max98357a on rk3399-gru-kevin since v5.7
by Alper Nebi Yasak 21 Jul '20

21 Jul '20

Hi, I have been getting "pop" sounds from the speaker on my rk3399-gru-kevin for a while, and bisected it to 128f825aeab7 ("ASoC: max98357a: move control of SD_MODE to DAPM"), but looks like the pops were somewhat expected: On 12 Feb 2020 13:55:15 +0800, Tzung-Bi Shih wrote [1]: > Possible drawback: may generate pop noise when BCLK=on but LRCLK=off. > As the datasheet[2] mentioned: > > Do not remove LRCLK while BCLK is present. > > Removing LRCLK while BCLK is present can cause unexpected output behavior > > including a large DC output voltage. [1] https://lore.kernel.org/alsa-devel/20200212055517.122728-1-tzungbi@google.c… As of v5.8-rc5 I'm still getting the speaker pops. More info below, but not all pops coincide with "set sdmode" messages, and vice versa. Reverting that commit stops the pops, but then the "Speakers Switch" can no longer mute the speakers. I don't really know much about these things, could anyone have a look at it? I'll try to read relevant documentation and code to make sense of it, but wanted to send an email in case there is an immediately obvious solution (besides reverting) and maybe just nobody had the time to implement it yet. Thanks in advance. --- I've experimented a bit on a Debian userspace (with #define DEBUG 1 to get sdmode messages), here's what happens without PulseAudio: - When playback starts, speaker pops with "set sdmode to 1". - During playback, toggling "Speakers Switch" sets sdmode to 0/1 and mutes/unmutes speakers without any pops. - Pausing/resuming playback (like seeking on a media player) make pops without changing sdmode. Also pops when playback ends (e.g. when "speaker-test -l 1" quits). - A few seconds after playback ends, speaker pops for the last time with "set sdmode to 0". - Within that few seconds, toggling "Speakers Switch" sets sdmode to 0/1 and makes a pop each time. It's mostly the same with PulseAudio but: - Pause/resume during playback doesn't make any pops. - The penultimate pop happens a few seconds after playback stops, when PA says it's suspending the max98357a device (no sdmode changes). - The final pop comes a few seconds after that (still with "set sdmode to 0"). Even without any playback it makes several pops during boot when e.g. PulseAudio starts/stops running in the display manager or in the user session.

2 4

[PATCH 1/2] ASoC: qcom: dt-bindings: Add sc7180 machine bindings
by Cheng-Yi Chiang 21 Jul '20

21 Jul '20

Add devicetree bindings documentation file for sc7180 sound card. Signed-off-by: Cheng-Yi Chiang <cychiang(a)chromium.org> --- .../bindings/sound/qcom,sc7180.yaml | 123 ++++++++++++++++++ 1 file changed, 123 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/qcom,sc7180.yaml diff --git a/Documentation/devicetree/bindings/sound/qcom,sc7180.yaml b/Documentation/devicetree/bindings/sound/qcom,sc7180.yaml new file mode 100644 index 000000000000..d60d2880d991 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,sc7180.yaml @@ -0,0 +1,123 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/qcom,sc7180.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Qualcomm Technologies Inc. SC7180 ASoC sound card driver + +maintainers: + - Rohit kumar <rohitkr(a)codeaurora.org> + - Cheng-Yi Chiang <cychiang(a)chromium.org> + +description: | + This binding describes the SC7180 sound card, which uses LPASS for audio. + +definitions: + + link-name: + description: Indicates dai-link name and PCM stream name. + $ref: /schemas/types.yaml#/definitions/string + maxItems: 1 + + dai: + type: object + properties: + sound-dai: + maxItems: 1 + $ref: /schemas/types.yaml#/definitions/phandle-array + description: phandle array of the codec or CPU DAI + + required: + - sound-dai + + unidirectional: + description: Specify direction of unidirectional dai link. + 0 for playback only. 1 for capture only. + $ref: /schemas/types.yaml#/definitions/uint32 + +properties: + compatible: + contains: + enum: + - qcom,sc7180-sndcard + + audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: |- + A list of the connections between audio components. Each entry is a + pair of strings, the first being the connection's sink, the second + being the connection's source. + + model: + $ref: /schemas/types.yaml#/definitions/string + description: User specified audio sound card name + +patternProperties: + "^dai-link-[0-9]+$": + description: | + Each subnode represents a dai link. Subnodes of each dai links would be + cpu/codec dais. + + type: object + + properties: + link-name: + $ref: "#/definitions/link-name" + + unidirectional: + $ref: "#/definitions/unidirectional" + + cpu: + $ref: "#/definitions/dai" + + codec: + $ref: "#/definitions/dai" + + required: + - link-name + - cpu + - codec + + additionalProperties: false + +examples: + + - | + snd { + compatible = "qcom,sc7180-sndcard"; + model = "sc7180-snd-card"; + + pinctrl-names = "default"; + pinctrl-0 = <&sec_mi2s_active &sec_mi2s_dout_active + &sec_mi2s_ws_active &pri_mi2s_active + &pri_mi2s_dout_active &pri_mi2s_ws_active + &pri_mi2s_din_active &pri_mi2s_mclk_active>; + + audio-routing = + "Headphone Jack", "HPOL", + "Headphone Jack", "HPOR"; + + dai-link-0 { + link-name = "MultiMedia0"; + cpu { + sound-dai = <&lpass_cpu 0>; + }; + + codec { + sound-dai = <&alc5682 0>; + }; + }; + + dai-link-1 { + link-name = "MultiMedia1"; + unidirectional = <0>; + cpu { + sound-dai = <&lpass_cpu 1>; + }; + + codec { + sound-dai = <&max98357a>; + }; + }; + }; -- 2.28.0.rc0.105.gf9edc3c819-goog

3 5

[PATCH v2 0/9] soundwire: intel: revisit SHIM programming
by Bard Liao 21 Jul '20

21 Jul '20

This series does some cleanup, revisits SHIM programming sequences, and merges Soundwire interrupt handlers/threads. changes in v2: - Resume M device instead of S device in intel_master_process_wakeen_event function. See the comment in intel_master_process_wakeen_event() in detail. Bard Liao (2): soundwire: intel/cadence: merge Soundwire interrupt handlers/threads Soundwire: intel_init: save Slave(s) _ADR info in sdw_intel_ctx Pierre-Louis Bossart (6): soundwire: intel: reuse code for wait loops to set/clear bits soundwire: intel: revisit SHIM programming sequences. soundwire: intel: introduce a helper to arm link synchronization soundwire: intel: introduce helper for link synchronization soundwire: intel_init: add implementation of sdw_intel_enable_irq() soundwire: intel_init: use EXPORT_SYMBOL_NS Rander Wang (1): soundwire: intel: add wake interrupt support drivers/soundwire/cadence_master.c | 18 +- drivers/soundwire/cadence_master.h | 4 + drivers/soundwire/intel.c | 381 +++++++++++++++++++++------- drivers/soundwire/intel.h | 9 + drivers/soundwire/intel_init.c | 101 +++++++- include/linux/soundwire/sdw_intel.h | 2 + 6 files changed, 417 insertions(+), 98 deletions(-) -- 2.17.1

2 10

[PATCH v2 0/2] snd-usb-6fire: firmware load and pulseaudio assumption
by René Herman 21 Jul '20

21 Jul '20

Hi Takashi. v2 of the earlier today sent out "set" for snd-usb-6fire, although now only two patches left, incorportaing the requested changes. The snd-usb-6fire driver for the TerraTec DMX 6Fire USB soundcard has been failing its firmware upload due to a non DMA-capable buffer on the stack. First of the patches kmallocs said bufffer instead and fixes the firmware upload; also makes it more alsa-generic by using the "goto out" structure. Note that it only looks a bit ungeneric as a patch due to also needing to at the same time unify its failure path: it's obvious when looked at post-apply. After that first patch the driver nominally works again but still has Pulseaudio crap out due to struct snd_pcm_hardware.channels_min=1 causing it to recognize it as a mono device only. Comparing with e.g. the TerraTec Aureon 7.1 Universe driver it seems that the solution is to simply set channels_min=2 as per the second patch. With these changes the card works again. Driver author Torsten Schenk has seen these and is fine with them: maintains an external driver with more options. I or he might time permitting start integrating more into the kernel driver over time. Regards, René René Herman (2): Move DMA-buffer off the stack Pulseaudio needs snd_pcm_hardware.channels_min > 1 sound/usb/6fire/firmware.c | 95 ++++++++++++++++++-------------------- sound/usb/6fire/pcm.c | 2 +- 2 files changed, 47 insertions(+), 50 deletions(-) -- 2.17.1

2 4

[PATCH 0/3] snd-usb-6fire: firmware load and pulseaudio assumption
by René Herman 21 Jul '20

21 Jul '20

Hi Takashi. I sent these out a week ago to alsa-devel (only) but am not sure they got anywhere, perhaps due to at least the alsa-devel archive scrubbing the text/x-patch attachments as "non-text". The snd-usb-6fire driver for the TerraTec DMX 6Fire USB soundcard has been failing its firmware upload due to a non DMA-capable buffer on the stack. First of the patches kmallocs said bufffer instead and fixes the firmware upload. After that first patch the driver nominally works again but still has Pulseaudio crap out due to struct snd_pcm_hardware.channels_min=1 causing it to recognize it as a mono device only. Comparing with e.g. the TerraTec Aureon 7.1 Universe driver it seems that the solution is to simply set channels_min=2 as per the second patch. Third patch unmarks the snd_pcm_hardware struct const as it it is in fact changed in usb6fire_pcm_open(), even though through a pointer, and is supposedly trivial. With these changes the card works again. Driver author Torsten Schenk has seen these and is fine with them: maintains an external driver with more options. I or he might time permitting start integrating more into the kernel driver over time. Regards, René René Herman (3): snd-usb-6fire: Move DMA-buffer off of the stack snd-usb-6fire: Pulseaudio needs snd_pcm_hardware.channels_min > 1 snd-usb-6fire: Unmark struct snd_pcm_hardware const firmware.c | 95 ++++++++++++++++++++++++++---------------------------- pcm.c | 4 +-- 2 files changed, 48 insertions(+), 51 deletions(-) -- 2.17.1

2 9

amd/renoir/acp: use the machine driver's name for top-level's conf
by GitHub pull_request - opened 21 Jul '20

21 Jul '20

alsa-project/alsa-ucm-conf pull request #41 was opened from jason77-wang: On the machines with amd renoir audio, the /sys/class/sound/card2/ device/driver links to /sys/bus/platform/drivers/acp_pdm_mach. Signed-off-by: Hui Wang <hui.wang(a)canonical.com> Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/41 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/41.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Reference USB card driver?
by René Herman 21 Jul '20

21 Jul '20

Good day. I recently acquired a TerraTec DMX 6Fire USB card to play with, which works after one/two just to alsa-devel sent out patches. Not very good though; it's for example rare to not have my music skip when Pulseaudio mixes in the "there's new mail" sound from Thunderbird. I haven't been around here for a very long time now but have some ALSA-experience with ISA and PCI hardware from long ago. None with USB whatsoever though. Is there a reference/best-practices USB card driver that I can/should look at? Regards, Rene

2 2

Realtek ALC3254 is not recongised
by GitHub issues - opened 21 Jul '20

21 Jul '20

alsa-project/alsa-lib issue #72 was opened from daleclack: I upgraded the newest alsa,but I found my sound card can't found,and sound output is unable to use. My Device:DELL G3 3500 Sound Card:Realtek ALC3254 Terminal Output: [dale-g33500 dale]# lspci | grep -i audio 00:1f.3 Multimedia audio controller: Intel Corporation Device 06c8 01:00.1 Audio device: NVIDIA Corporation Device 10fa (rev a1) [dale-g33500 dale]# cat /proc/asound/cards 0 [NVidia ]: HDA-Intel - HDA NVidia HDA NVidia at 0x83000000 irq 17 [dale-g33500 dale]# cat /proc/asound/version Advanced Linux Sound Architecture Driver Version k5.7.9-1-MANJARO. [dale-g33500 dale]# head -5 /proc/asound/card0/codec#0 Codec: Nvidia GPU 94 HDMI/DP Address: 0 AFG Function Id: 0x1 (unsol 0) Vendor Id: 0x10de0094 Subsystem Id: 0x102809e1 [dale-g33500 dale]# pacman -Q alsa-lib alsa-lib 1.2.3.2-1 The Realtek Sound Card is disappeared! Issue URL : https://github.com/alsa-project/alsa-lib/issues/72 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

omap-mcbsp 49022000.mcbsp: TX Buffer Overflow!
by Dave Young 21 Jul '20

21 Jul '20

Hi, I'm trying to use g_audio on my Nokia N900 with mainline kernel. Seems it does not work. No sound when I play from a laptop, and also see a lot of error like below: [ 4729.557647] omap-mcbsp 49022000.mcbsp: TX Buffer Overflow! ... Is this a know issue? Thanks Dave

3 4

[PATCH AUTOSEL 5.4 27/34] ALSA: hda/realtek - fixup for yet another Intel reference board
by Sasha Levin 21 Jul '20

21 Jul '20

From: PeiSen Hou <pshou(a)realtek.com.tw> [ Upstream commit 5734e509d5d515c187f642937ef2de1e58d7715d ] Add headset_jack for the intel reference board support with 10ec:1230. Signed-off-by: PeiSen Hou <pshou(a)realtek.com.tw> Link: https://lore.kernel.org/r/20200716090134.9811-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 160e00e8007aa..c46651207a0ca 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7523,6 +7523,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x10cf, 0x1629, "Lifebook U7x7", ALC255_FIXUP_LIFEBOOK_U7x7_HEADSET_MIC), SND_PCI_QUIRK(0x10cf, 0x1845, "Lifebook U904", ALC269_FIXUP_LIFEBOOK_EXTMIC), SND_PCI_QUIRK(0x10ec, 0x10f2, "Intel Reference board", ALC700_FIXUP_INTEL_REFERENCE), + SND_PCI_QUIRK(0x10ec, 0x1230, "Intel Reference board", ALC225_FIXUP_HEADSET_JACK), SND_PCI_QUIRK(0x10f7, 0x8338, "Panasonic CF-SZ6", ALC269_FIXUP_HEADSET_MODE), SND_PCI_QUIRK(0x144d, 0xc109, "Samsung Ativ book 9 (NP900X3G)", ALC269_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x144d, 0xc169, "Samsung Notebook 9 Pen (NP930SBE-K01US)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), -- 2.25.1

1 0

[PATCH AUTOSEL 5.7 32/40] ALSA: hda/realtek - fixup for yet another Intel reference board
by Sasha Levin 21 Jul '20

21 Jul '20

From: PeiSen Hou <pshou(a)realtek.com.tw> [ Upstream commit 5734e509d5d515c187f642937ef2de1e58d7715d ] Add headset_jack for the intel reference board support with 10ec:1230. Signed-off-by: PeiSen Hou <pshou(a)realtek.com.tw> Link: https://lore.kernel.org/r/20200716090134.9811-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 16ecc8515db8e..1470321bf31ef 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7523,6 +7523,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x10cf, 0x1629, "Lifebook U7x7", ALC255_FIXUP_LIFEBOOK_U7x7_HEADSET_MIC), SND_PCI_QUIRK(0x10cf, 0x1845, "Lifebook U904", ALC269_FIXUP_LIFEBOOK_EXTMIC), SND_PCI_QUIRK(0x10ec, 0x10f2, "Intel Reference board", ALC700_FIXUP_INTEL_REFERENCE), + SND_PCI_QUIRK(0x10ec, 0x1230, "Intel Reference board", ALC225_FIXUP_HEADSET_JACK), SND_PCI_QUIRK(0x10f7, 0x8338, "Panasonic CF-SZ6", ALC269_FIXUP_HEADSET_MODE), SND_PCI_QUIRK(0x144d, 0xc109, "Samsung Ativ book 9 (NP900X3G)", ALC269_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x144d, 0xc169, "Samsung Notebook 9 Pen (NP930SBE-K01US)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), -- 2.25.1

1 0

Compile with debugging symbols
by GitHub issues - opened 20 Jul '20

20 Jul '20

alsa-project/alsa-lib issue #71 was opened from project-owner: Hi, how can I compile the alsa-lib with debugging symbols enabled? Thank you! Issue URL : https://github.com/alsa-project/alsa-lib/issues/71 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

S/PDIF fix for Asus Xonar SE Output
by GitHub pull_request - opened 20 Jul '20

20 Jul '20

alsa-project/alsa-lib pull request #70 was opened from Odzinic: The ASUS Xonar SE soundcard (ID 0b05:189d ASUSTek Computer, Inc. Xonar SoundCard) defaults to the first device for S/PDIF when it should actually use the third for its digital output. This fix was discussed in this AskUbuntu post: https://askubuntu.com/a/1210401/117741. Signed-off by: Omar Dzinic <odzinic(a)gmail.com> Request URL : https://github.com/alsa-project/alsa-lib/pull/70 Patch URL : https://github.com/alsa-project/alsa-lib/pull/70.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH 0/6] ASoC: Intel: machine driver updates for 5.9
by Pierre-Louis Bossart 20 Jul '20

20 Jul '20

Small patchset to harden the SoundWire machine driver, change bad HIDs, update PLL settings and avoid memory leaks. Given that the SoundWire core parts are not upstream it's probably not necessary to provide the patches to stable branches. Bard Liao (1): ASoC: Intel: sof_sdw_rt711: remove hard-coded codec name Kai Vehmanen (2): ASoC: Intel: sof_sdw: add support for systems without i915 audio ASoC: Intel: sof_sdw: avoid crash if invalid DSP topology loaded Libin Yang (1): ASoC: Intel: common: change match table ehl-rt5660 Pierre-Louis Bossart (1): ASoC: Intel: sof_sdw_rt711: remove properties in card remove Yong Zhi (1): ASoC: intel: board: sof_rt5682: Update rt1015 pll input clk freq sound/soc/intel/boards/sof_rt5682.c | 9 +++++- sound/soc/intel/boards/sof_sdw.c | 31 +++++++++++++------ sound/soc/intel/boards/sof_sdw_common.h | 2 ++ sound/soc/intel/boards/sof_sdw_hdmi.c | 6 ++++ sound/soc/intel/boards/sof_sdw_rt711.c | 17 +++++++++- .../intel/common/soc-acpi-intel-ehl-match.c | 2 +- 6 files changed, 54 insertions(+), 13 deletions(-) base-commit: 22e9b54307987787efa0ee534aa9e31982ec1161 -- 2.25.1

2 9

[PATCH][next] ASoC: Intel: Skylake: Avoid the use of one-element array
by Gustavo A. R. Silva 20 Jul '20

20 Jul '20

One-element arrays are being deprecated[1]. Replace the one-element array with a simple value type 'u8 reserved'[2], once it seems this is just a placeholder for alignment. [1] https://github.com/KSPP/linux/issues/79 [2] https://github.com/KSPP/linux/issues/86 Tested-by: kernel test robot <lkp(a)intel.com> Link: https://github.com/GustavoARSilva/linux-hardening/blob/master/cii/0-day/sky… Signed-off-by: Gustavo A. R. Silva <gustavoars(a)kernel.org> --- sound/soc/intel/skylake/skl-topology.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/skylake/skl-topology.h b/sound/soc/intel/skylake/skl-topology.h index 9889f728752c..5e93ad85e06d 100644 --- a/sound/soc/intel/skylake/skl-topology.h +++ b/sound/soc/intel/skylake/skl-topology.h @@ -97,7 +97,7 @@ struct skl_audio_data_format { u8 number_of_channels; u8 valid_bit_depth; u8 sample_type; - u8 reserved[1]; + u8 reserved; } __packed; struct skl_base_cfg { -- 2.27.0

3 2

[PATCH] ASoC: wcd9335.h: fix duplicated word
by Randy Dunlap 20 Jul '20

20 Jul '20

Fix the doubled word "in" in a comment by adding punctuation in 3 places and capitalization. Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Cc: Mark Brown <broonie(a)kernel.org> Cc: alsa-devel(a)alsa-project.org --- sound/soc/codecs/wcd9335.h | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) --- linux-next-20200717.orig/sound/soc/codecs/wcd9335.h +++ linux-next-20200717/sound/soc/codecs/wcd9335.h @@ -4,9 +4,9 @@ #define __WCD9335_H__ /* - * WCD9335 register base can change according to the mode it works in - * in slimbus mode the reg base starts from 0x800 - * in i2s/i2c mode the reg base is 0x0 + * WCD9335 register base can change according to the mode it works in. + * In slimbus mode the reg base starts from 0x800. + * In i2s/i2c mode the reg base is 0x0. */ #define WCD9335_REG(pg, r) ((pg << 8) | (r)) #define WCD9335_REG_OFFSET(r) (r & 0xFF)

2 1

[PATCH] ASoC: soc-dai.h: drop a duplicated word
by Randy Dunlap 20 Jul '20

20 Jul '20

Drop the repeated word "be" in a comment. Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: alsa-devel(a)alsa-project.org --- include/sound/soc-dai.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) --- linux-next-20200717.orig/include/sound/soc-dai.h +++ linux-next-20200717/include/sound/soc-dai.h @@ -39,7 +39,7 @@ struct snd_compr_stream; /* * DAI Clock gating. * - * DAI bit clocks can be be gated (disabled) when the DAI is not + * DAI bit clocks can be gated (disabled) when the DAI is not * sending or receiving PCM data in a frame. This can be used to save power. */ #define SND_SOC_DAIFMT_CONT (1 << 4) /* continuous clock */

2 1

[PATCH] ASoC: tegra20_das.h: delete duplicated words
by Randy Dunlap 20 Jul '20

20 Jul '20

Delete the doubled word "to" in two comments. Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Cc: Stephen Warren <swarren(a)nvidia.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: alsa-devel(a)alsa-project.org --- sound/soc/tegra/tegra20_das.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) --- linux-next-20200717.orig/sound/soc/tegra/tegra20_das.h +++ linux-next-20200717/sound/soc/tegra/tegra20_das.h @@ -91,14 +91,14 @@ struct tegra20_das { */ /* - * Connect a DAP to to a DAC + * Connect a DAP to a DAC * dap_id: DAP to connect: TEGRA20_DAS_DAP_ID_* * dac_sel: DAC to connect to: TEGRA20_DAS_DAP_SEL_DAC* */ extern int tegra20_das_connect_dap_to_dac(int dap_id, int dac_sel); /* - * Connect a DAP to to another DAP + * Connect a DAP to another DAP * dap_id: DAP to connect: TEGRA20_DAS_DAP_ID_* * other_dap_sel: DAP to connect to: TEGRA20_DAS_DAP_SEL_DAP* * master: Is this DAP the master (1) or slave (0)

2 1

[PATCH 0/2] ASoC: fsl_asrc: improve clock selection and quality
by Arnaud Ferraris 20 Jul '20

20 Jul '20

This series fixes the automatic clock selection and enables internal ratio in order to improve audio quality. The clock selection patches have been set aside for now, as the discussion is still ongoing regarding that matter. Arnaud Ferraris(2): ASoC: fsl_asrc: make sure the input and output clocks are different ASoC: fsl_asrc: always use internal ratio sound/soc/fsl/fsl_asrc.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-)

4 17

[PATCH v4 0/3] ASoC: merge .digital_mute() into .mute_stream()
by Kuninori Morimoto 20 Jul '20

20 Jul '20

Hi Mark These are v4 digital_mute() patch which adjusts to atmel which had conflict on v3. v3 -> v4 - tidyup for atmel which had conflict v2 -> v3 - uses "xxx_mute_stream" for .mute_stream naming if it was better - removed verbose Cc email address v1 -> v2 - return -ENOTSUPP at hdmi-codec - add new .no_capture_mute flag and emulate .digital_mute() by .mute_stream() Link: https://lore.kernel.org/r/87h7uhxxk6.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/874kqy2y5t.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87ftam37ko.wl-kuninori.morimoto.gx@renesas.com Kuninori Morimoto (3): ASoC: soc-dai: remove .digital_mute ASoC: soc-core: snd_soc_dai_digital_mute() for both CPU/Codec ASoC: soc-xxx: add asoc_substream_to_rtd() include/sound/soc-dai.h | 1 - include/sound/soc.h | 2 + sound/soc/soc-component.c | 20 ++++----- sound/soc/soc-core.c | 4 +- sound/soc/soc-dai.c | 12 ++---- sound/soc/soc-dapm.c | 6 +-- sound/soc/soc-generic-dmaengine-pcm.c | 4 +- sound/soc/soc-link.c | 12 +++--- sound/soc/soc-pcm.c | 62 +++++++++++++-------------- sound/soc/soc-utils.c | 2 +- 10 files changed, 61 insertions(+), 64 deletions(-) -- 2.25.1

1 5

[PATCH v2] ASoC: cros_ec_codec: Reset I2S RX when probing
by Yu-Hsuan Hsu 19 Jul '20

19 Jul '20

It is not guaranteed that I2S RX is disabled when the kernel booting. For example, if the kernel crashes while it is enabled, it will keep enabled until the next time EC reboots. Reset I2S RX when probing to fix this issue. Signed-off-by: Yu-Hsuan Hsu <yuhsuan(a)chromium.org> --- drivers/platform/chrome/cros_ec_proto.c | 7 ++++++- include/linux/platform_data/cros_ec_commands.h | 1 + sound/soc/codecs/cros_ec_codec.c | 9 +++++++++ 3 files changed, 16 insertions(+), 1 deletion(-) diff --git a/drivers/platform/chrome/cros_ec_proto.c b/drivers/platform/chrome/cros_ec_proto.c index 3e745e0fe092c..2c60690d7147c 100644 --- a/drivers/platform/chrome/cros_ec_proto.c +++ b/drivers/platform/chrome/cros_ec_proto.c @@ -572,7 +572,12 @@ int cros_ec_cmd_xfer_status(struct cros_ec_device *ec_dev, return -ENOTSUPP; } else if (msg->result != EC_RES_SUCCESS) { dev_dbg(ec_dev->dev, "Command result (err: %d)\n", msg->result); - return -EPROTO; + switch (msg->result) { + case EC_RES_INVALID_PARAM: + return -EINVAL; + default: + return -EPROTO; + } } return ret; diff --git a/include/linux/platform_data/cros_ec_commands.h b/include/linux/platform_data/cros_ec_commands.h index 69210881ebac8..11ce917ca924c 100644 --- a/include/linux/platform_data/cros_ec_commands.h +++ b/include/linux/platform_data/cros_ec_commands.h @@ -4598,6 +4598,7 @@ enum ec_codec_i2s_rx_subcmd { EC_CODEC_I2S_RX_SET_SAMPLE_DEPTH = 0x2, EC_CODEC_I2S_RX_SET_DAIFMT = 0x3, EC_CODEC_I2S_RX_SET_BCLK = 0x4, + EC_CODEC_I2S_RX_RESET = 0x5, EC_CODEC_I2S_RX_SUBCMD_COUNT, }; diff --git a/sound/soc/codecs/cros_ec_codec.c b/sound/soc/codecs/cros_ec_codec.c index f23956cf4ed84..b5ff30b7f1aa8 100644 --- a/sound/soc/codecs/cros_ec_codec.c +++ b/sound/soc/codecs/cros_ec_codec.c @@ -1034,6 +1034,15 @@ static int cros_ec_codec_platform_probe(struct platform_device *pdev) } priv->ec_capabilities = r.capabilities; + /* Reset EC codec I2S RX. */ + p.cmd = EC_CODEC_I2S_RX_RESET; + ret = send_ec_host_command(priv->ec_device, EC_CMD_EC_CODEC_I2S_RX, + (uint8_t *)&p, sizeof(p), NULL, 0); + if (ret == -EINVAL) + dev_info(dev, "Missing reset command. Please update your EC firmware.\n"); + else if (ret) + dev_err(dev, "failed to EC_CODEC_I2S_RESET: %d\n", ret); + platform_set_drvdata(pdev, priv); ret = devm_snd_soc_register_component(dev, &i2s_rx_component_driver, -- 2.27.0.389.gc38d7665816-goog

3 5

[PATCH] sound: asihpi: delete duplicated word
by Randy Dunlap 19 Jul '20

19 Jul '20

Delete the doubled word "of" in a comment. Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: alsa-devel(a)alsa-project.org --- sound/pci/asihpi/hpi_internal.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) --- linux-next-20200717.orig/sound/pci/asihpi/hpi_internal.h +++ linux-next-20200717/sound/pci/asihpi/hpi_internal.h @@ -53,7 +53,7 @@ If handle is invalid *pPhysicalAddr is s u16 hpios_locked_mem_get_phys_addr(struct consistent_dma_area *locked_mem_handle, u32 *p_physical_addr); -/** Get the CPU address of of memory represented by LockedMemHandle. +/** Get the CPU address of memory represented by LockedMemHandle. If handle is NULL *ppvVirtualAddr is set to NULL and return 1 */

2 1

[PATCH for v5.9] ALSA: Replace HTTP links with HTTPS ones
by Alexander A. Klimov 19 Jul '20

19 Jul '20

Rationale: Reduces attack surface on kernel devs opening the links for MITM as HTTPS traffic is much harder to manipulate. Deterministic algorithm: For each file: If not .svg: For each line: If doesn't contain `\bxmlns\b`: For each link, `\bhttp://[^# \t\r\n]*(?:\w|/)`: If neither `\bgnu\.org/license`, nor `\bmozilla\.org/MPL\b`: If both the HTTP and HTTPS versions return 200 OK and serve the same content: Replace HTTP with HTTPS. Signed-off-by: Alexander A. Klimov <grandmaster(a)al2klimov.de> --- Continuing my work started at 93431e0607e5. See also: git log --oneline '--author=Alexander A. Klimov <grandmaster(a)al2klimov.de>' v5.7..master (Actually letting a shell for loop submit all this stuff for me.) If there are any URLs to be removed completely or at least not (just) HTTPSified: Just clearly say so and I'll *undo my change*. See also: https://lkml.org/lkml/2020/6/27/64 If there are any valid, but yet not changed URLs: See: https://lkml.org/lkml/2020/6/26/837 If you apply the patch, please let me know. Sorry again to all maintainers who complained about subject lines. Now I realized that you want an actually perfect prefixes, not just subsystem ones. I tried my best... And yes, *I could* (at least half-)automate it. Impossible is nothing! :) Documentation/sound/alsa-configuration.rst | 6 +++--- Documentation/sound/cards/audigy-mixer.rst | 2 +- Documentation/sound/cards/sb-live-mixer.rst | 2 +- Documentation/sound/hd-audio/notes.rst | 6 +++--- include/sound/hdmi-codec.h | 2 +- include/sound/omap-hdmi-audio.h | 2 +- sound/isa/Kconfig | 2 +- sound/sparc/dbri.c | 2 +- sound/usb/mixer_maps.c | 2 +- 9 files changed, 13 insertions(+), 13 deletions(-) diff --git a/Documentation/sound/alsa-configuration.rst b/Documentation/sound/alsa-configuration.rst index 72f97d4b01a7..c755b1c5e16f 100644 --- a/Documentation/sound/alsa-configuration.rst +++ b/Documentation/sound/alsa-configuration.rst @@ -309,7 +309,7 @@ pcifix This module supports all ADB PCM channels, ac97 mixer, SPDIF, hardware EQ, mpu401, gameport. A3D and wavetable support are still in development. Development and reverse engineering work is being coordinated at -http://savannah.nongnu.org/projects/openvortex/ +https://savannah.nongnu.org/projects/openvortex/ SPDIF output has a copy of the AC97 codec output, unless you use the ``spdif`` pcm device, which allows raw data passthru. The hardware EQ hardware and SPDIF is only present in the Vortex2 and @@ -1575,7 +1575,7 @@ See Documentation/sound/cards/multisound.sh for important information about this driver. Note that it has been discontinued, but the Voyetra Turtle Beach knowledge base entry for it is still available at -http://www.turtlebeach.com +https://www.turtlebeach.com Module snd-msnd-pinnacle ------------------------ @@ -2703,4 +2703,4 @@ Kernel Bugzilla ALSA Developers ML mailto:alsa-devel@alsa-project.org alsa-info.sh script - http://www.alsa-project.org/alsa-info.sh + https://www.alsa-project.org/alsa-info.sh diff --git a/Documentation/sound/cards/audigy-mixer.rst b/Documentation/sound/cards/audigy-mixer.rst index 86213234435f..998f76e19cdd 100644 --- a/Documentation/sound/cards/audigy-mixer.rst +++ b/Documentation/sound/cards/audigy-mixer.rst @@ -331,7 +331,7 @@ WO 9901953 (A1) Execution and Audio Data Sequencing (Jan. 14, 1999) -US Patents (http://www.uspto.gov/) +US Patents (https://www.uspto.gov/) ---------------------------------- US 5925841 diff --git a/Documentation/sound/cards/sb-live-mixer.rst b/Documentation/sound/cards/sb-live-mixer.rst index bcb62fc99bbb..eccb0f0ffd0f 100644 --- a/Documentation/sound/cards/sb-live-mixer.rst +++ b/Documentation/sound/cards/sb-live-mixer.rst @@ -336,7 +336,7 @@ WO 9901953 (A1) Execution and Audio Data Sequencing (Jan. 14, 1999) -US Patents (http://www.uspto.gov/) +US Patents (https://www.uspto.gov/) ---------------------------------- US 5925841 diff --git a/Documentation/sound/hd-audio/notes.rst b/Documentation/sound/hd-audio/notes.rst index 0f3109d9abc8..cf4d7158af78 100644 --- a/Documentation/sound/hd-audio/notes.rst +++ b/Documentation/sound/hd-audio/notes.rst @@ -42,7 +42,7 @@ If you are interested in the deep debugging of HD-audio, read the HD-audio specification at first. The specification is found on Intel's web page, for example: -* http://www.intel.com/standards/hdaudio/ +* https://www.intel.com/standards/hdaudio/ HD-Audio Controller @@ -728,7 +728,7 @@ version can be found on git repository: The script can be fetched directly from the following URL, too: -* http://www.alsa-project.org/alsa-info.sh +* https://www.alsa-project.org/alsa-info.sh Run this script as root, and it will gather the important information such as the module lists, module parameters, proc file contents @@ -818,7 +818,7 @@ proc-compatible output. The hda-analyzer: -* http://git.alsa-project.org/?p=alsa.git;a=tree;f=hda-analyzer +* https://git.alsa-project.org/?p=alsa.git;a=tree;f=hda-analyzer is a part of alsa.git repository in alsa-project.org: diff --git a/include/sound/hdmi-codec.h b/include/sound/hdmi-codec.h index 83b17682e01c..cc0b29bbcde3 100644 --- a/include/sound/hdmi-codec.h +++ b/include/sound/hdmi-codec.h @@ -2,7 +2,7 @@ /* * hdmi-codec.h - HDMI Codec driver API * - * Copyright (C) 2014 Texas Instruments Incorporated - http://www.ti.com + * Copyright (C) 2014 Texas Instruments Incorporated - https://www.ti.com * * Author: Jyri Sarha <jsarha(a)ti.com> */ diff --git a/include/sound/omap-hdmi-audio.h b/include/sound/omap-hdmi-audio.h index 16c007b651f4..e5f82044a404 100644 --- a/include/sound/omap-hdmi-audio.h +++ b/include/sound/omap-hdmi-audio.h @@ -2,7 +2,7 @@ /* * hdmi-audio.c -- OMAP4+ DSS HDMI audio support library * - * Copyright (C) 2014 Texas Instruments Incorporated - http://www.ti.com + * Copyright (C) 2014 Texas Instruments Incorporated - https://www.ti.com * * Author: Jyri Sarha <jsarha(a)ti.com> */ diff --git a/sound/sparc/dbri.c b/sound/sparc/dbri.c index cf7049999261..2d0144562173 100644 --- a/sound/sparc/dbri.c +++ b/sound/sparc/dbri.c @@ -22,7 +22,7 @@ * - Data sheet of the T7903, a newer but very similar ISA bus equivalent * available from the Lucent (formerly AT&T microelectronics) home * page. - * - http://www.freesoft.org/Linux/DBRI/ + * - https://www.freesoft.org/Linux/DBRI/ * - MMCODEC: Crystal Semiconductor CS4215 16 bit Multimedia Audio Codec * Interfaces: CHI, Audio In & Out, 2 bits parallel * Documentation: from the Crystal Semiconductor home page. diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c index 9af7aa93f6fa..2ec484655201 100644 --- a/sound/usb/mixer_maps.c +++ b/sound/usb/mixer_maps.c @@ -233,7 +233,7 @@ static const struct usbmix_name_map maya44_map[] = { }; /* Section "justlink_map" below added by James Courtier-Dutton <James(a)superbug.demon.co.uk> - * sourced from Maplin Electronics (http://www.maplin.co.uk), part number A56AK + * sourced from Maplin Electronics (https://www.maplin.co.uk), part number A56AK * Part has 2 connectors that act as a single output. (TOSLINK Optical for digital out, and 3.5mm Jack for Analogue out.) * The USB Mixer publishes a Microphone and extra Volume controls for it, but none exist on the device, * so this map removes all unwanted sliders from alsamixer -- 2.27.0

2 1

[PATCH] ASoC: soc-jack: calling snd_soc_jack_report causes a null pointer access
by mnlife qiao 19 Jul '20

19 Jul '20

From: mnlife <mnlife(a)foxmail.com> When snd_soc_card_jack_new is not called or the call fails, calling this function causes a null pointer access Signed-off-by: mnlife <mnlife(a)foxmail.com> --- sound/soc/soc-jack.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index 0f1820f36b4d..51b799ee98b7 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -44,7 +44,7 @@ void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask) unsigned int sync = 0; int enable; - if (!jack) + if (!jack || !jack->jack) return; trace_snd_soc_jack_report(jack, mask, status); -- 2.17.1

1 0

[PATCH] ASoC: soc-jack: calling snd_soc_jack_report causes a null pointer access
by mnlife qiao 19 Jul '20

19 Jul '20

> From: mnlife <mnlife(a)foxmail.com> > > When snd_soc_card_jack_new is not called or the call fails, > calling this function causes a null pointer access > > Signed-off-by: mnlife <mnlife(a)foxmail.com> > --- > sound/soc/soc-jack.c | 2 +- > 1 file changed, 1 insertion(+), 1 deletion(-) > > diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c > index 0f1820f36b4d..51b799ee98b7 100644 > --- a/sound/soc/soc-jack.c > +++ b/sound/soc/soc-jack.c > @@ -44,7 +44,7 @@ void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask) > unsigned int sync = 0; > int enable; > > - if (!jack) > + if (!jack || !jack->jack) > return; > trace_snd_soc_jack_report(jack, mask, status); > > -- > 2.17.1 > >

1 0

Realtek ALC892 produces low volume garbled output.
by GitHub issues - opened 19 Jul '20

19 Jul '20

alsa-project/alsa-lib issue #69 was opened from RoryO: I'm sorry if this isn't the correct project for this. I don't know if this problem is relevant to the Alsa project or the Linux kernel directly and I don't know how to find that out. Personal background === Programmer, experienced. First used Linux with Red Hat Linux 6 about 20 years ago. Mostly using Linux on the server side of things since. I'm not afraid to try things so have at it. Problem === PCM audio to device heavily distorted and extremely low volume. Happens with all applications. Works flawlessly with Windows. Bypassing PulseAudio using aplay or speaker-test does not help. Interestingly, playing a raw sine wave with speaker-test works fine! Therefore it's only PCM based output causing an issue. Even using something like VLC which has it's own codecs and outputs directly to the sound buffer exhibits this issue, so it's not a system codec issue. For example, this produces a perfectly fine sounding sine wave speaker-test -l 3 -t sine -c 2 And this produces nothing speaker-test -l 3 -t wav -c 2 If I play something on Spotify or Youtube and max out the volume I can barely hear some heavily distorted, clipped noise vaguely sounding like it should. I only have powered headphones for testing, no powered external speakers. Relevant details === Latest KDE neon, which is Ubuntu 18.04 base with the GUI switched from Gnome to KDE Plasma. Hardware is an Asus z170 motherboard, integrated Realtek/Intel audio. Using the rear green line out jack. I think there are headphone jack headers on the board as well, they are unused. lspci output 00:1f.3 Audio device: Intel Corporation 100 Series/C230 Series Chipset Family HD Audio Controller (rev 31) 01:00.1 Audio device: NVIDIA Corporation GP104 High Definition Audio Controller (rev a1) kernel messages regarding snd snd_hda_intel 0000:01:00.1: Disabling MSI snd_hda_intel 0000:01:00.1: Handle vga_switcheroo audio client snd_hda_codec_realtek hdaudioC0D0: autoconfig for ALC892: line_outs=3 (0x14/0x15/0x16/0x0/0x0) type:line snd_hda_codec_realtek hdaudioC0D0: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) snd_hda_codec_realtek hdaudioC0D0: hp_outs=1 (0x1b/0x0/0x0/0x0/0x0) snd_hda_codec_realtek hdaudioC0D0: mono: mono_out=0x0 snd_hda_codec_realtek hdaudioC0D0: dig-out=0x11/0x1e snd_hda_codec_realtek hdaudioC0D0: inputs: snd_hda_codec_realtek hdaudioC0D0: Front Mic=0x19 snd_hda_codec_realtek hdaudioC0D0: Rear Mic=0x18 snd_hda_codec_realtek hdaudioC0D0: Line=0x1a Output of alsa-info dumping everything it knows about https://pastebin.com/wCif6Qzv I'm stuck and don't know where to go from here. Since utilities like aplay and sound-test bypass PulseAudio I think it's a problem with Alsa or the device driver for this card. That's as far as I can discover on my own. Things I have tried === - Using alsamixer, confirming all volume sliders at max, none are muted. Maxing out all sliders is when I can barely make out garbled output through powered headphones. - Using Ubuntu Mainline Kernel Installer, signing it properly, and upgrading the kernel from 5.3.0 to 5.7.9 - Using `hdajackretask` and overriding jack plugin detection - Modifying PulseAudio config to not resample and force sampling to 44.1khz - Playing around with `pavucontrol` and kde equivalents, disabling the unused nvidia output and trying different jack configurations I also found some modprobe settings for Realtek devices, however usually that information was usually about 6-8 years old and didn't look relevant. Issue URL : https://github.com/alsa-project/alsa-lib/issues/69 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH] ASoC: soc-jack: calling snd_soc_jack_report causes a null pointer access
by mnlife 18 Jul '20

18 Jul '20

From: mnlife <mnlife(a)foxmail.com> When snd_soc_card_jack_new is not called or the call fails, calling this function causes a null pointer access Signed-off-by: mnlife <mnlife(a)foxmail.com> --- sound/soc/soc-jack.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index 0f1820f36b4d..51b799ee98b7 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -44,7 +44,7 @@ void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask) unsigned int sync = 0; int enable; - if (!jack) + if (!jack || !jack->jack) return; trace_snd_soc_jack_report(jack, mask, status); -- 2.17.1 .

1 0

[PATCH] ALSA: hda/realtek: Fixed ALC298 sound bug by adding quirk for Samsung Notebook Pen S
by Junar Wohn 18 Jul '20

18 Jul '20

Fixed no headphone sound bug on laptop Samsung Notebook Pen S (950SBE-951SBE), by using existing patch in Linus' tree, commit 14425f1f521f (ALSA: hda/realtek: Add quirk for Samsung Notebook). This laptop uses the same ALC298 but different subsystem id 0x144dc812. I added SND_PCI_QUIRK at sound/pci/hda/patch_realtek.c Signed-off-by: Joonho Wohn <doomsheart(a)gmail.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 737ef82a75fd..ebedc6093642 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7528,6 +7528,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x144d, 0xc169, "Samsung Notebook 9 Pen (NP930SBE-K01US)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), SND_PCI_QUIRK(0x144d, 0xc176, "Samsung Notebook 9 Pro (NP930MBE-K04US)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), SND_PCI_QUIRK(0x144d, 0xc740, "Samsung Ativ book 8 (NP870Z5G)", ALC269_FIXUP_ATIV_BOOK_8), + SND_PCI_QUIRK(0x144d, 0xc812, "Samsung Notebook Pen S (NT950SBE-X58)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), SND_PCI_QUIRK(0x1458, 0xfa53, "Gigabyte BXBT-2807", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1462, 0xb120, "MSI Cubi MS-B120", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1462, 0xb171, "Cubi N 8GL (MS-B171)", ALC283_FIXUP_HEADSET_MIC), --

2 1

snd_pcm_avail() returned a value that is exceptionally large
by GitHub issues - opened 18 Jul '20

18 Jul '20

alsa-project/alsa-lib issue #68 was opened from reezer: I hope this is the right place to report this. This occurred in relation to a crash. http://alsa-project.org/db/?f=52e8505d884682fc06adac4ab8ce92a4ca3e0870 ``` E: [alsa-sink-ALC887-VD Analog] alsa-util.c: snd_pcm_avail() returned a value that is exceptionally large: 177312 bytes (923 ms). E: [alsa-sink-ALC887-VD Analog] alsa-util.c: Most likely this is a bug in the ALSA driver 'snd_hda_intel'. Please report this issue to the ALSA developers. E: [alsa-sink-ALC887-VD Analog] alsa-util.c: snd_pcm_dump(): E: [alsa-sink-ALC887-VD Analog] alsa-util.c: Soft volume PCM E: [alsa-sink-ALC887-VD Analog] alsa-util.c: Control: PCM Playback Volume E: [alsa-sink-ALC887-VD Analog] alsa-util.c: min_dB: -51 E: [alsa-sink-ALC887-VD Analog] alsa-util.c: max_dB: 0 E: [alsa-sink-ALC887-VD Analog] alsa-util.c: resolution: 256 E: [alsa-sink-ALC887-VD Analog] alsa-util.c: Its setup is: E: [alsa-sink-ALC887-VD Analog] alsa-util.c: stream : PLAYBACK E: [alsa-sink-ALC887-VD Analog] alsa-util.c: access : MMAP_INTERLEAVED E: [alsa-sink-ALC887-VD Analog] alsa-util.c: format : S16_LE E: [alsa-sink-ALC887-VD Analog] alsa-util.c: subformat : STD E: [alsa-sink-ALC887-VD Analog] alsa-util.c: channels : 2 E: [alsa-sink-ALC887-VD Analog] alsa-util.c: rate : 48000 E: [alsa-sink-ALC887-VD Analog] alsa-util.c: exact rate : 48000 (48000/1) E: [alsa-sink-ALC887-VD Analog] alsa-util.c: msbits : 16 E: [alsa-sink-ALC887-VD Analog] alsa-util.c: buffer_size : 4800 ``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/68 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[RFT] ALSA control service programs for Fireworks board module
by Takashi Sakamoto 18 Jul '20

18 Jul '20

Hi, This is request-for-test (RFT) to the ALSA control service programs for devices with Fireworks board module. The program is available in 'topic/service-for-efw' of below repository, and the RFT is corresponding to Pull Request #2: * https://github.com/alsa-project/snd-firewire-ctl-services/ If you have listed devices and handle them by ALSA fireworks driver, please test the service program according to the below instruction. * Mackie (Loud) Onyx 1200F * Mackie (Loud) Onyx 400F * Echo Audio Audiofire 12 (till Jul 2009) * Echo Audio Audiofire 8 (till Jul 2009) * Echo Audio Audiofire 12 (since Jul 2009) * Echo Audio Audiofire 8 (since Jul 2009) * Echo Audio Audiofire 2 * Echo Audio Audiofire 4 * Echo Audio Audiofire Pre8 * Gibson Robot Interface Pack (RIP) for Robot Guitar series Background ---------- The models with Fireworks board module are already supported by ALSA fireworks driver for their isochronous packet stream functionality and vendor-specific transaction. Although the driver allows userspace applications to process PCM frames and MIDI messages in the packet streaming, it doesn't support the other functionalities such as volume control since it can be achieved by any usespace application which executes ioctl(2) to ALSA hwdep character device for the transaction. The service program is to satisfy the demand to control the models. ALSA control applications can request the control by IPC to the service program. In detail of mechanism, please read README.rst in the repository[1]. You can see some illustrations. How to build ------------- The project depends on several libraries in below repositories: * glib[2] * alsa-gobject v0.1.0[3] * libhinawa v2.0.0[4] Before building it, please install them with gobject-introspection support. The project is written by Rust programming language[5] and packaged by cargo[6]. To build, just run below commands in the environment to connect to internet: ``` $ git clone https://github.com/alsa-project/snd-firewire-ctl-services.git -b topic/service-for-efw $ cd snd-firewire-ctl-services $ cargo build ``` All of required crates are downloaded automatically, then start build. Executables ----------- After building, `snd-fireworks-ctl-service` executable is available. When running, the program adds and maintains control elements, then enter event loop to dispatch events. The added elements are visible and manipulable for any userspace applications. The program dispatch any manipulation event for transactions to control actual device. The executable has an option for the numerical ID of sound card. ``` Usage: snd-fireworks-ctl-service CARD_ID where: CARD_ID: The numerical ID of sound card. ``` It's easy to run the executable by cargo, like: ``` $ cargo run --bin snd-fireworks-ctl-service 2 ``` Receiving SIGTERM signal terminates the event loop, then the program finishes. During runtime, any ALSA control applications can enumerate and manipulate the added control elements. This is an example of Echo Audio AudioFire 2: ``` $ amixer -c 2 controls numid=33,iface=MIXER,name='clock-detect' numid=2,iface=MIXER,name='clock-rate' numid=1,iface=MIXER,name='clock-source' numid=36,iface=MIXER,name='input-meter' numid=34,iface=MIXER,name='midi-in-detect' numid=35,iface=MIXER,name='midi-out-detect' numid=6,iface=MIXER,name='monitor-gain' numid=7,iface=MIXER,name='monitor-gain',index=1 numid=8,iface=MIXER,name='monitor-gain',index=2 numid=9,iface=MIXER,name='monitor-gain',index=3 numid=10,iface=MIXER,name='monitor-gain',index=4 numid=11,iface=MIXER,name='monitor-gain',index=5 numid=12,iface=MIXER,name='monitor-mute' numid=13,iface=MIXER,name='monitor-mute',index=1 numid=14,iface=MIXER,name='monitor-mute',index=2 numid=15,iface=MIXER,name='monitor-mute',index=3 numid=16,iface=MIXER,name='monitor-mute',index=4 numid=17,iface=MIXER,name='monitor-mute',index=5 numid=24,iface=MIXER,name='monitor-pan' numid=25,iface=MIXER,name='monitor-pan',index=1 numid=26,iface=MIXER,name='monitor-pan',index=2 numid=27,iface=MIXER,name='monitor-pan',index=3 numid=28,iface=MIXER,name='monitor-pan',index=4 numid=29,iface=MIXER,name='monitor-pan',index=5 numid=18,iface=MIXER,name='monitor-solo' numid=19,iface=MIXER,name='monitor-solo',index=1 numid=20,iface=MIXER,name='monitor-solo',index=2 numid=21,iface=MIXER,name='monitor-solo',index=3 numid=22,iface=MIXER,name='monitor-solo',index=4 numid=23,iface=MIXER,name='monitor-solo',index=5 numid=38,iface=MIXER,name='monitoring' numid=37,iface=MIXER,name='output-meter' numid=31,iface=MIXER,name='output-mute' numid=30,iface=MIXER,name='output-volume' numid=4,iface=MIXER,name='playback-mute' numid=5,iface=MIXER,name='playback-solo' numid=3,iface=MIXER,name='playback-volume' numid=32,iface=MIXER,name='stream-playback-routing' ``` Access permissions for relevant character devices -------------------------------------------------- The executable manipulates below character devices to dispatch, and emit events from end to end. The '%u' is the numerical number of instance in each subsystem: * ALSA control character device (/dev/snd/controlC%u) * ALSA hwdep character device (/dev/snd/hwC%uD%u) * Linux firewire character device (/dev/fw%u) Feedback -------- Any feedback is welcome. For questions, please use mailing list of alsa-devel or alsa-users[7]. For issues, please use service of github.com[8]. Issues ------ The name of control elements are not fixed yet since the convention of name for element set is not clear yet for recording equipment. Users may feel inconvenience to which channel corresponds to which physical port. Furthermore, in the case that element set includes several elements, it's unclear that which element corresponds to which physical port as well. ALSA control core has no feature for the above issues at present. [1] https://github.com/alsa-project/snd-firewire-ctl-services [2] https://gitlab.gnome.org/GNOME/glib [3] https://github.com/alsa-project/alsa-gobject/ [4] https://github.com/alsa-project/libhinawa [5] https://www.rust-lang.org/ [6] https://doc.rust-lang.org/cargo/ [7] https://www.alsa-project.org/wiki/Mailing-lists [8] https://github.com/alsa-project/snd-firewire-ctl-services/issues Regards Takashi Sakamoto

1 0

[PATCH 0/4] ASoC: Fix dailink checks for DPCM
by Pierre-Louis Bossart 17 Jul '20

17 Jul '20

We've had a couple of changes that introduce regressions with the multi-cpu DAI solutions, and while trying to fix them we found additional inconsistencies that should also go to stable branches. Bard Liao (1): ASoC: core: only convert non DPCM link to DPCM link Pierre-Louis Bossart (3): ASoC: soc-pcm: dpcm: fix playback/capture checks ASoC: Intel: boards: replace capture_only by dpcm_capture ASoC: SOF: nocodec: conditionally set dpcm_capture/dpcm_playback flags sound/soc/intel/boards/glk_rt5682_max98357a.c | 2 +- sound/soc/intel/boards/kbl_da7219_max98927.c | 4 +- sound/soc/intel/boards/kbl_rt5663_max98927.c | 2 +- .../intel/boards/kbl_rt5663_rt5514_max98927.c | 2 +- sound/soc/soc-core.c | 22 ++++++++-- sound/soc/soc-pcm.c | 44 ++++++++++++++----- sound/soc/sof/nocodec.c | 6 ++- 7 files changed, 62 insertions(+), 20 deletions(-) base-commit: 8a9144c1cf523221b37dd3393827253c91fcbf54 -- 2.20.1

4 25

[GIT PULL] ASoC fixes for v5.8-rc5
by Mark Brown 17 Jul '20

17 Jul '20

The following changes since commit bc5c7f55f5ea91e137fc7939435ed2e2bb6e5a15: AsoC: amd: add missing snd- module prefix to the acp3x-rn driver kernel module (2020-06-25 15:06:56 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v5.8-rc5 for you to fetch changes up to fe0a53044b4bce947045eadd7fa1adbc4685afab: ASoC: rt5682: Report the button event in the headset type only (2020-07-16 20:29:09 +0100) ---------------------------------------------------------------- ASoC: Fixes for v5.8 An awful lot of mostly small fixes here, mainly for x86 based platforms and the CODEC drivers mainly used on them. For the most part this is either minor device specific stuff which seems to come from detailed testing or robustness against errors which comes from people having done some fuzzing runs aginst the topology code. ---------------------------------------------------------------- Daniel Baluta (1): ASoC: SOF: imx: add min/max channels for SAI/ESAI on i.MX8/i.MX8M Fabio Estevam (1): ASoC: dt-bindings: simple-card: Fix 'make dt_binding_check' warnings Geert Uytterhoeven (1): ASoC: qcom: Drop HAS_DMA dependency to fix link failure Hans de Goede (4): ASoC: Intel: cht_bsw_rt5672: Change bus format to I2S 2 channel ASoC: rt5670: Correct RT5670_LDO_SEL_MASK ASoC: rt5670: Add new gpio1_is_ext_spk_en quirk and enable it on the Lenovo Miix 2 10 ASoC: rt5670: Fix dac- and adc- vol-tlv values being off by a factor of 10 Jing Xiangfeng (1): ASoC: Intel: bytcht_es8316: Add missed put_device() Mark Brown (3): Merge series "ASoC: codecs: add MAX98373 Soundwire driver" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: Merge series "ASoC: topology: fix error handling flow" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: Merge series "ASoC: more fixes for dpcm checks" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: Maxime Ripard (1): ASoC: core: Remove only the registered component in devm functions Nicolas Ferre (1): MAINTAINERS: Change Maintainer for some at91 drivers Nicolin Chen (1): MAINTAINERS: Add Shengjiu to reviewer list of sound/soc/fsl Oder Chiou (2): ASoC: rt5682: cancel jack_detect_work if hs_jack is set to null even soundwire mode ASoC: rt5682: Report the button event in the headset type only Pierre-Louis Bossart (4): ASoC: soc-dai: set dai_link dpcm_ flags with a helper ASoC: Intel: bdw-rt5677: fix non BE conversion ASoC: topology: fix kernel oops on route addition error ASoC: topology: fix tlvs in error handling for widget_dmixer Puyou Lu (2): ASoC: wm8974: fix Boost Mixer Aux Switch ASoC: wm8974: remove unsupported clock mode Ranjani Sridharan (1): ASoC: SOF: core: fix null-ptr-deref bug during device removal Ravulapati Vishnu vardhan rao (1): ASoC: amd: Rectifying Unbalanced pm_runtime_enable! issue Ryan Lee (1): ASoC: codecs: max98373: Removed superfluous volume control from chip default Shuming Fan (1): ASoC: rt286: fix unexpected interrupt happens Yu-Hsuan Hsu (1): ASoC: rockchip: add format and rate constraints on rk3399 derek.fang (1): ASoC: rt5682: Enable Vref2 under using PLL2 randerwang (1): ASoc: codecs: max98373: remove Idle_bias_on to let codec suspend .../devicetree/bindings/sound/simple-card.yaml | 17 +++-- MAINTAINERS | 7 +- include/sound/rt5670.h | 1 + include/sound/soc-dai.h | 1 + include/sound/soc.h | 2 + sound/soc/amd/raven/pci-acp3x.c | 4 +- sound/soc/codecs/max98373.c | 8 --- sound/soc/codecs/rt286.c | 8 +-- sound/soc/codecs/rt5670.c | 75 +++++++++++++++++----- sound/soc/codecs/rt5670.h | 2 +- sound/soc/codecs/rt5682.c | 46 ++++++++----- sound/soc/codecs/wm8974.c | 6 +- sound/soc/generic/audio-graph-card.c | 4 +- sound/soc/generic/simple-card.c | 4 +- sound/soc/intel/boards/bdw-rt5677.c | 1 + sound/soc/intel/boards/bytcht_es8316.c | 4 +- sound/soc/intel/boards/cht_bsw_rt5672.c | 23 ++++--- sound/soc/qcom/Kconfig | 2 +- sound/soc/rockchip/rk3399_gru_sound.c | 13 ++++ sound/soc/soc-core.c | 27 ++++++++ sound/soc/soc-dai.c | 38 +++++++++++ sound/soc/soc-devres.c | 8 ++- sound/soc/soc-generic-dmaengine-pcm.c | 2 +- sound/soc/soc-topology.c | 24 +++++-- sound/soc/sof/core.c | 10 +-- sound/soc/sof/imx/imx8.c | 8 +++ sound/soc/sof/imx/imx8m.c | 8 +++ 27 files changed, 262 insertions(+), 91 deletions(-)

2 1

[PATCH v2] ALSA: Replace the word "slave" in vmaster API
by Takashi Iwai 17 Jul '20

17 Jul '20

Follow the recent inclusive terminology guidelines and replace the word "slave" in vmaster API. I chose the word "follower" at this time since it seems fitting for the purpose. Note that the word "master" is kept in API, since it refers rather to audio master volume control. Also, while we're at it, a typo in comments is corrected, too. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- v1->v2: Use "follower" instead of "replica" include/sound/control.h | 45 +++---- include/sound/hda_codec.h | 2 +- sound/core/vmaster.c | 263 ++++++++++++++++++++-------------------- sound/pci/ac97/ac97_patch.c | 20 +-- sound/pci/ca0106/ca0106_mixer.c | 18 +-- sound/pci/hda/hda_codec.c | 96 +++++++-------- sound/pci/hda/hda_generic.c | 20 +-- sound/pci/hda/hda_generic.h | 2 +- sound/pci/hda/hda_local.h | 10 +- sound/pci/hda/patch_ca0132.c | 10 +- sound/pci/hda/patch_sigmatel.c | 2 +- sound/pci/ice1712/juli.c | 20 +-- sound/pci/ice1712/quartet.c | 14 +-- sound/ppc/awacs.c | 12 +- sound/usb/6fire/control.c | 2 +- 15 files changed, 269 insertions(+), 267 deletions(-) diff --git a/include/sound/control.h b/include/sound/control.h index aeaed2a05bae..e128cff10dfa 100644 --- a/include/sound/control.h +++ b/include/sound/control.h @@ -188,20 +188,21 @@ int snd_ctl_enum_info(struct snd_ctl_elem_info *info, unsigned int channels, */ struct snd_kcontrol *snd_ctl_make_virtual_master(char *name, const unsigned int *tlv); -int _snd_ctl_add_slave(struct snd_kcontrol *master, struct snd_kcontrol *slave, - unsigned int flags); -/* optional flags for slave */ -#define SND_CTL_SLAVE_NEED_UPDATE (1 << 0) +int _snd_ctl_add_follower(struct snd_kcontrol *master, + struct snd_kcontrol *follower, + unsigned int flags); +/* optional flags for follower */ +#define SND_CTL_FOLLOWER_NEED_UPDATE (1 << 0) /** - * snd_ctl_add_slave - Add a virtual slave control + * snd_ctl_add_follower - Add a virtual follower control * @master: vmaster element - * @slave: slave element to add + * @follower: follower element to add * - * Add a virtual slave control to the given master element created via + * Add a virtual follower control to the given master element created via * snd_ctl_create_virtual_master() beforehand. * - * All slaves must be the same type (returning the same information + * All followers must be the same type (returning the same information * via info callback). The function doesn't check it, so it's your * responsibility. * @@ -213,18 +214,18 @@ int _snd_ctl_add_slave(struct snd_kcontrol *master, struct snd_kcontrol *slave, * Return: Zero if successful or a negative error code. */ static inline int -snd_ctl_add_slave(struct snd_kcontrol *master, struct snd_kcontrol *slave) +snd_ctl_add_follower(struct snd_kcontrol *master, struct snd_kcontrol *follower) { - return _snd_ctl_add_slave(master, slave, 0); + return _snd_ctl_add_follower(master, follower, 0); } /** - * snd_ctl_add_slave_uncached - Add a virtual slave control + * snd_ctl_add_follower_uncached - Add a virtual follower control * @master: vmaster element - * @slave: slave element to add + * @follower: follower element to add * - * Add a virtual slave control to the given master. - * Unlike snd_ctl_add_slave(), the element added via this function + * Add a virtual follower control to the given master. + * Unlike snd_ctl_add_follower(), the element added via this function * is supposed to have volatile values, and get callback is called * at each time queried from the master. * @@ -235,10 +236,10 @@ snd_ctl_add_slave(struct snd_kcontrol *master, struct snd_kcontrol *slave) * Return: Zero if successful or a negative error code. */ static inline int -snd_ctl_add_slave_uncached(struct snd_kcontrol *master, - struct snd_kcontrol *slave) +snd_ctl_add_follower_uncached(struct snd_kcontrol *master, + struct snd_kcontrol *follower) { - return _snd_ctl_add_slave(master, slave, SND_CTL_SLAVE_NEED_UPDATE); + return _snd_ctl_add_follower(master, follower, SND_CTL_FOLLOWER_NEED_UPDATE); } int snd_ctl_add_vmaster_hook(struct snd_kcontrol *kctl, @@ -246,11 +247,11 @@ int snd_ctl_add_vmaster_hook(struct snd_kcontrol *kctl, void *private_data); void snd_ctl_sync_vmaster(struct snd_kcontrol *kctl, bool hook_only); #define snd_ctl_sync_vmaster_hook(kctl) snd_ctl_sync_vmaster(kctl, true) -int snd_ctl_apply_vmaster_slaves(struct snd_kcontrol *kctl, - int (*func)(struct snd_kcontrol *vslave, - struct snd_kcontrol *slave, - void *arg), - void *arg); +int snd_ctl_apply_vmaster_followers(struct snd_kcontrol *kctl, + int (*func)(struct snd_kcontrol *vfollower, + struct snd_kcontrol *follower, + void *arg), + void *arg); /* * Helper functions for jack-detection controls diff --git a/include/sound/hda_codec.h b/include/sound/hda_codec.h index d16a4229209b..f4cc364d837f 100644 --- a/include/sound/hda_codec.h +++ b/include/sound/hda_codec.h @@ -208,7 +208,7 @@ struct hda_codec { struct mutex control_mutex; struct snd_array spdif_out; unsigned int spdif_in_enable; /* SPDIF input enable? */ - const hda_nid_t *slave_dig_outs; /* optional digital out slave widgets */ + const hda_nid_t *follower_dig_outs; /* optional digital out follower widgets */ struct snd_array init_pins; /* initial (BIOS) pin configurations */ struct snd_array driver_pins; /* pin configs set by codec parser */ struct snd_array cvt_setups; /* audio convert setups */ diff --git a/sound/core/vmaster.c b/sound/core/vmaster.c index ab54d79654c9..ab36f9898711 100644 --- a/sound/core/vmaster.c +++ b/sound/core/vmaster.c @@ -1,6 +1,6 @@ // SPDX-License-Identifier: GPL-2.0-only /* - * Virtual master and slave controls + * Virtual master and follower controls * * Copyright (c) 2008 by Takashi Iwai <tiwai(a)suse.de> */ @@ -21,15 +21,15 @@ struct link_ctl_info { }; /* - * link master - this contains a list of slave controls that are + * link master - this contains a list of follower controls that are * identical types, i.e. info returns the same value type and value * ranges, but may have different number of counts. * * The master control is so far only mono volume/switch for simplicity. - * The same value will be applied to all slaves. + * The same value will be applied to all followers. */ struct link_master { - struct list_head slaves; + struct list_head followers; struct link_ctl_info info; int val; /* the master value */ unsigned int tlv[4]; @@ -38,23 +38,23 @@ struct link_master { }; /* - * link slave - this contains a slave control element + * link follower - this contains a follower control element * - * It fakes the control callbacsk with additional attenuation by the - * master control. A slave may have either one or two channels. + * It fakes the control callbacks with additional attenuation by the + * master control. A follower may have either one or two channels. */ -struct link_slave { +struct link_follower { struct list_head list; struct link_master *master; struct link_ctl_info info; int vals[2]; /* current values */ unsigned int flags; struct snd_kcontrol *kctl; /* original kcontrol pointer */ - struct snd_kcontrol slave; /* the copy of original control entry */ + struct snd_kcontrol follower; /* the copy of original control entry */ }; -static int slave_update(struct link_slave *slave) +static int follower_update(struct link_follower *follower) { struct snd_ctl_elem_value *uctl; int err, ch; @@ -62,68 +62,68 @@ static int slave_update(struct link_slave *slave) uctl = kzalloc(sizeof(*uctl), GFP_KERNEL); if (!uctl) return -ENOMEM; - uctl->id = slave->slave.id; - err = slave->slave.get(&slave->slave, uctl); + uctl->id = follower->follower.id; + err = follower->follower.get(&follower->follower, uctl); if (err < 0) goto error; - for (ch = 0; ch < slave->info.count; ch++) - slave->vals[ch] = uctl->value.integer.value[ch]; + for (ch = 0; ch < follower->info.count; ch++) + follower->vals[ch] = uctl->value.integer.value[ch]; error: kfree(uctl); return err < 0 ? err : 0; } -/* get the slave ctl info and save the initial values */ -static int slave_init(struct link_slave *slave) +/* get the follower ctl info and save the initial values */ +static int follower_init(struct link_follower *follower) { struct snd_ctl_elem_info *uinfo; int err; - if (slave->info.count) { + if (follower->info.count) { /* already initialized */ - if (slave->flags & SND_CTL_SLAVE_NEED_UPDATE) - return slave_update(slave); + if (follower->flags & SND_CTL_FOLLOWER_NEED_UPDATE) + return follower_update(follower); return 0; } uinfo = kmalloc(sizeof(*uinfo), GFP_KERNEL); if (!uinfo) return -ENOMEM; - uinfo->id = slave->slave.id; - err = slave->slave.info(&slave->slave, uinfo); + uinfo->id = follower->follower.id; + err = follower->follower.info(&follower->follower, uinfo); if (err < 0) { kfree(uinfo); return err; } - slave->info.type = uinfo->type; - slave->info.count = uinfo->count; - if (slave->info.count > 2 || - (slave->info.type != SNDRV_CTL_ELEM_TYPE_INTEGER && - slave->info.type != SNDRV_CTL_ELEM_TYPE_BOOLEAN)) { - pr_err("ALSA: vmaster: invalid slave element\n"); + follower->info.type = uinfo->type; + follower->info.count = uinfo->count; + if (follower->info.count > 2 || + (follower->info.type != SNDRV_CTL_ELEM_TYPE_INTEGER && + follower->info.type != SNDRV_CTL_ELEM_TYPE_BOOLEAN)) { + pr_err("ALSA: vmaster: invalid follower element\n"); kfree(uinfo); return -EINVAL; } - slave->info.min_val = uinfo->value.integer.min; - slave->info.max_val = uinfo->value.integer.max; + follower->info.min_val = uinfo->value.integer.min; + follower->info.max_val = uinfo->value.integer.max; kfree(uinfo); - return slave_update(slave); + return follower_update(follower); } /* initialize master volume */ static int master_init(struct link_master *master) { - struct link_slave *slave; + struct link_follower *follower; if (master->info.count) return 0; /* already initialized */ - list_for_each_entry(slave, &master->slaves, list) { - int err = slave_init(slave); + list_for_each_entry(follower, &master->followers, list) { + int err = follower_init(follower); if (err < 0) return err; - master->info = slave->info; + master->info = follower->info; master->info.count = 1; /* always mono */ /* set full volume as default (= no attenuation) */ master->val = master->info.max_val; @@ -134,113 +134,113 @@ static int master_init(struct link_master *master) return -ENOENT; } -static int slave_get_val(struct link_slave *slave, - struct snd_ctl_elem_value *ucontrol) +static int follower_get_val(struct link_follower *follower, + struct snd_ctl_elem_value *ucontrol) { int err, ch; - err = slave_init(slave); + err = follower_init(follower); if (err < 0) return err; - for (ch = 0; ch < slave->info.count; ch++) - ucontrol->value.integer.value[ch] = slave->vals[ch]; + for (ch = 0; ch < follower->info.count; ch++) + ucontrol->value.integer.value[ch] = follower->vals[ch]; return 0; } -static int slave_put_val(struct link_slave *slave, - struct snd_ctl_elem_value *ucontrol) +static int follower_put_val(struct link_follower *follower, + struct snd_ctl_elem_value *ucontrol) { int err, ch, vol; - err = master_init(slave->master); + err = master_init(follower->master); if (err < 0) return err; - switch (slave->info.type) { + switch (follower->info.type) { case SNDRV_CTL_ELEM_TYPE_BOOLEAN: - for (ch = 0; ch < slave->info.count; ch++) + for (ch = 0; ch < follower->info.count; ch++) ucontrol->value.integer.value[ch] &= - !!slave->master->val; + !!follower->master->val; break; case SNDRV_CTL_ELEM_TYPE_INTEGER: - for (ch = 0; ch < slave->info.count; ch++) { + for (ch = 0; ch < follower->info.count; ch++) { /* max master volume is supposed to be 0 dB */ vol = ucontrol->value.integer.value[ch]; - vol += slave->master->val - slave->master->info.max_val; - if (vol < slave->info.min_val) - vol = slave->info.min_val; - else if (vol > slave->info.max_val) - vol = slave->info.max_val; + vol += follower->master->val - follower->master->info.max_val; + if (vol < follower->info.min_val) + vol = follower->info.min_val; + else if (vol > follower->info.max_val) + vol = follower->info.max_val; ucontrol->value.integer.value[ch] = vol; } break; } - return slave->slave.put(&slave->slave, ucontrol); + return follower->follower.put(&follower->follower, ucontrol); } /* - * ctl callbacks for slaves + * ctl callbacks for followers */ -static int slave_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) +static int follower_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) { - struct link_slave *slave = snd_kcontrol_chip(kcontrol); - return slave->slave.info(&slave->slave, uinfo); + struct link_follower *follower = snd_kcontrol_chip(kcontrol); + return follower->follower.info(&follower->follower, uinfo); } -static int slave_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) +static int follower_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { - struct link_slave *slave = snd_kcontrol_chip(kcontrol); - return slave_get_val(slave, ucontrol); + struct link_follower *follower = snd_kcontrol_chip(kcontrol); + return follower_get_val(follower, ucontrol); } -static int slave_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) +static int follower_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { - struct link_slave *slave = snd_kcontrol_chip(kcontrol); + struct link_follower *follower = snd_kcontrol_chip(kcontrol); int err, ch, changed = 0; - err = slave_init(slave); + err = follower_init(follower); if (err < 0) return err; - for (ch = 0; ch < slave->info.count; ch++) { - if (slave->vals[ch] != ucontrol->value.integer.value[ch]) { + for (ch = 0; ch < follower->info.count; ch++) { + if (follower->vals[ch] != ucontrol->value.integer.value[ch]) { changed = 1; - slave->vals[ch] = ucontrol->value.integer.value[ch]; + follower->vals[ch] = ucontrol->value.integer.value[ch]; } } if (!changed) return 0; - err = slave_put_val(slave, ucontrol); + err = follower_put_val(follower, ucontrol); if (err < 0) return err; return 1; } -static int slave_tlv_cmd(struct snd_kcontrol *kcontrol, - int op_flag, unsigned int size, - unsigned int __user *tlv) +static int follower_tlv_cmd(struct snd_kcontrol *kcontrol, + int op_flag, unsigned int size, + unsigned int __user *tlv) { - struct link_slave *slave = snd_kcontrol_chip(kcontrol); + struct link_follower *follower = snd_kcontrol_chip(kcontrol); /* FIXME: this assumes that the max volume is 0 dB */ - return slave->slave.tlv.c(&slave->slave, op_flag, size, tlv); + return follower->follower.tlv.c(&follower->follower, op_flag, size, tlv); } -static void slave_free(struct snd_kcontrol *kcontrol) +static void follower_free(struct snd_kcontrol *kcontrol) { - struct link_slave *slave = snd_kcontrol_chip(kcontrol); - if (slave->slave.private_free) - slave->slave.private_free(&slave->slave); - if (slave->master) - list_del(&slave->list); - kfree(slave); + struct link_follower *follower = snd_kcontrol_chip(kcontrol); + if (follower->follower.private_free) + follower->follower.private_free(&follower->follower); + if (follower->master) + list_del(&follower->list); + kfree(follower); } /* - * Add a slave control to the group with the given master control + * Add a follower control to the group with the given master control * - * All slaves must be the same type (returning the same information + * All followers must be the same type (returning the same information * via info callback). The function doesn't check it, so it's your * responsibility. * @@ -249,35 +249,36 @@ static void slave_free(struct snd_kcontrol *kcontrol) * - logarithmic volume control (dB level), no linear volume * - master can only attenuate the volume, no gain */ -int _snd_ctl_add_slave(struct snd_kcontrol *master, struct snd_kcontrol *slave, - unsigned int flags) +int _snd_ctl_add_follower(struct snd_kcontrol *master, + struct snd_kcontrol *follower, + unsigned int flags) { struct link_master *master_link = snd_kcontrol_chip(master); - struct link_slave *srec; + struct link_follower *srec; - srec = kzalloc(struct_size(srec, slave.vd, slave->count), + srec = kzalloc(struct_size(srec, follower.vd, follower->count), GFP_KERNEL); if (!srec) return -ENOMEM; - srec->kctl = slave; - srec->slave = *slave; - memcpy(srec->slave.vd, slave->vd, slave->count * sizeof(*slave->vd)); + srec->kctl = follower; + srec->follower = *follower; + memcpy(srec->follower.vd, follower->vd, follower->count * sizeof(*follower->vd)); srec->master = master_link; srec->flags = flags; /* override callbacks */ - slave->info = slave_info; - slave->get = slave_get; - slave->put = slave_put; - if (slave->vd[0].access & SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK) - slave->tlv.c = slave_tlv_cmd; - slave->private_data = srec; - slave->private_free = slave_free; - - list_add_tail(&srec->list, &master_link->slaves); + follower->info = follower_info; + follower->get = follower_get; + follower->put = follower_put; + if (follower->vd[0].access & SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK) + follower->tlv.c = follower_tlv_cmd; + follower->private_data = srec; + follower->private_free = follower_free; + + list_add_tail(&srec->list, &master_link->followers); return 0; } -EXPORT_SYMBOL(_snd_ctl_add_slave); +EXPORT_SYMBOL(_snd_ctl_add_follower); /* * ctl callbacks for master controls @@ -309,20 +310,20 @@ static int master_get(struct snd_kcontrol *kcontrol, return 0; } -static int sync_slaves(struct link_master *master, int old_val, int new_val) +static int sync_followers(struct link_master *master, int old_val, int new_val) { - struct link_slave *slave; + struct link_follower *follower; struct snd_ctl_elem_value *uval; uval = kmalloc(sizeof(*uval), GFP_KERNEL); if (!uval) return -ENOMEM; - list_for_each_entry(slave, &master->slaves, list) { + list_for_each_entry(follower, &master->followers, list) { master->val = old_val; - uval->id = slave->slave.id; - slave_get_val(slave, uval); + uval->id = follower->follower.id; + follower_get_val(follower, uval); master->val = new_val; - slave_put_val(slave, uval); + follower_put_val(follower, uval); } kfree(uval); return 0; @@ -344,7 +345,7 @@ static int master_put(struct snd_kcontrol *kcontrol, if (new_val == old_val) return 0; - err = sync_slaves(master, old_val, new_val); + err = sync_followers(master, old_val, new_val); if (err < 0) return err; if (master->hook && !first_init) @@ -355,17 +356,17 @@ static int master_put(struct snd_kcontrol *kcontrol, static void master_free(struct snd_kcontrol *kcontrol) { struct link_master *master = snd_kcontrol_chip(kcontrol); - struct link_slave *slave, *n; + struct link_follower *follower, *n; - /* free all slave links and retore the original slave kctls */ - list_for_each_entry_safe(slave, n, &master->slaves, list) { - struct snd_kcontrol *sctl = slave->kctl; + /* free all follower links and retore the original follower kctls */ + list_for_each_entry_safe(follower, n, &master->followers, list) { + struct snd_kcontrol *sctl = follower->kctl; struct list_head olist = sctl->list; - memcpy(sctl, &slave->slave, sizeof(*sctl)); - memcpy(sctl->vd, slave->slave.vd, + memcpy(sctl, &follower->follower, sizeof(*sctl)); + memcpy(sctl->vd, follower->follower.vd, sctl->count * sizeof(*sctl->vd)); sctl->list = olist; /* keep the current linked-list */ - kfree(slave); + kfree(follower); } kfree(master); } @@ -378,8 +379,8 @@ static void master_free(struct snd_kcontrol *kcontrol) * * Creates a virtual master control with the given name string. * - * After creating a vmaster element, you can add the slave controls - * via snd_ctl_add_slave() or snd_ctl_add_slave_uncached(). + * After creating a vmaster element, you can add the follower controls + * via snd_ctl_add_follower() or snd_ctl_add_follower_uncached(). * * The optional argument @tlv can be used to specify the TLV information * for dB scale of the master control. It should be a single element @@ -403,7 +404,7 @@ struct snd_kcontrol *snd_ctl_make_virtual_master(char *name, master = kzalloc(sizeof(*master), GFP_KERNEL); if (!master) return NULL; - INIT_LIST_HEAD(&master->slaves); + INIT_LIST_HEAD(&master->followers); kctl = snd_ctl_new1(&knew, master); if (!kctl) { @@ -455,11 +456,11 @@ int snd_ctl_add_vmaster_hook(struct snd_kcontrol *kcontrol, EXPORT_SYMBOL_GPL(snd_ctl_add_vmaster_hook); /** - * snd_ctl_sync_vmaster - Sync the vmaster slaves and hook + * snd_ctl_sync_vmaster - Sync the vmaster followers and hook * @kcontrol: vmaster kctl element * @hook_only: sync only the hook * - * Forcibly call the put callback of each slave and call the hook function + * Forcibly call the put callback of each follower and call the hook function * to synchronize with the current value of the given vmaster element. * NOP when NULL is passed to @kcontrol. */ @@ -476,7 +477,7 @@ void snd_ctl_sync_vmaster(struct snd_kcontrol *kcontrol, bool hook_only) if (err < 0) return; first_init = err; - err = sync_slaves(master, master->val, master->val); + err = sync_followers(master, master->val, master->val); if (err < 0) return; } @@ -487,34 +488,34 @@ void snd_ctl_sync_vmaster(struct snd_kcontrol *kcontrol, bool hook_only) EXPORT_SYMBOL_GPL(snd_ctl_sync_vmaster); /** - * snd_ctl_apply_vmaster_slaves - Apply function to each vmaster slave + * snd_ctl_apply_vmaster_followers - Apply function to each vmaster follower * @kctl: vmaster kctl element * @func: function to apply * @arg: optional function argument * - * Apply the function @func to each slave kctl of the given vmaster kctl. + * Apply the function @func to each follower kctl of the given vmaster kctl. * Returns 0 if successful, or a negative error code. */ -int snd_ctl_apply_vmaster_slaves(struct snd_kcontrol *kctl, - int (*func)(struct snd_kcontrol *vslave, - struct snd_kcontrol *slave, - void *arg), - void *arg) +int snd_ctl_apply_vmaster_followers(struct snd_kcontrol *kctl, + int (*func)(struct snd_kcontrol *vfollower, + struct snd_kcontrol *follower, + void *arg), + void *arg) { struct link_master *master; - struct link_slave *slave; + struct link_follower *follower; int err; master = snd_kcontrol_chip(kctl); err = master_init(master); if (err < 0) return err; - list_for_each_entry(slave, &master->slaves, list) { - err = func(slave->kctl, &slave->slave, arg); + list_for_each_entry(follower, &master->followers, list) { + err = func(follower->kctl, &follower->follower, arg); if (err < 0) return err; } return 0; } -EXPORT_SYMBOL_GPL(snd_ctl_apply_vmaster_slaves); +EXPORT_SYMBOL_GPL(snd_ctl_apply_vmaster_followers); diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c index 40967d9cc631..1627a74baf3c 100644 --- a/sound/pci/ac97/ac97_patch.c +++ b/sound/pci/ac97/ac97_patch.c @@ -19,7 +19,7 @@ static struct snd_kcontrol *snd_ac97_find_mixer_ctl(struct snd_ac97 *ac97, const char *name); static int snd_ac97_add_vmaster(struct snd_ac97 *ac97, char *name, const unsigned int *tlv, - const char * const *slaves); + const char * const *followers); /* * Chip specific initialization @@ -3373,7 +3373,7 @@ AC97_SINGLE("Downmix LFE and Center to Front", 0x5a, 12, 1, 0), AC97_SINGLE("Downmix Surround to Front", 0x5a, 11, 1, 0), }; -static const char * const slave_vols_vt1616[] = { +static const char * const follower_vols_vt1616[] = { "Front Playback Volume", "Surround Playback Volume", "Center Playback Volume", @@ -3381,7 +3381,7 @@ static const char * const slave_vols_vt1616[] = { NULL }; -static const char * const slave_sws_vt1616[] = { +static const char * const follower_sws_vt1616[] = { "Front Playback Switch", "Surround Playback Switch", "Center Playback Switch", @@ -3400,10 +3400,10 @@ static struct snd_kcontrol *snd_ac97_find_mixer_ctl(struct snd_ac97 *ac97, return snd_ctl_find_id(ac97->bus->card, &id); } -/* create a virtual master control and add slaves */ +/* create a virtual master control and add followers */ static int snd_ac97_add_vmaster(struct snd_ac97 *ac97, char *name, const unsigned int *tlv, - const char * const *slaves) + const char * const *followers) { struct snd_kcontrol *kctl; const char * const *s; @@ -3416,16 +3416,16 @@ static int snd_ac97_add_vmaster(struct snd_ac97 *ac97, char *name, if (err < 0) return err; - for (s = slaves; *s; s++) { + for (s = followers; *s; s++) { struct snd_kcontrol *sctl; sctl = snd_ac97_find_mixer_ctl(ac97, *s); if (!sctl) { dev_dbg(ac97->bus->card->dev, - "Cannot find slave %s, skipped\n", *s); + "Cannot find follower %s, skipped\n", *s); continue; } - err = snd_ctl_add_slave(kctl, sctl); + err = snd_ctl_add_follower(kctl, sctl); if (err < 0) return err; } @@ -3451,12 +3451,12 @@ static int patch_vt1616_specific(struct snd_ac97 * ac97) snd_ac97_rename_vol_ctl(ac97, "Master Playback", "Front Playback"); err = snd_ac97_add_vmaster(ac97, "Master Playback Volume", - kctl->tlv.p, slave_vols_vt1616); + kctl->tlv.p, follower_vols_vt1616); if (err < 0) return err; err = snd_ac97_add_vmaster(ac97, "Master Playback Switch", - NULL, slave_sws_vt1616); + NULL, follower_sws_vt1616); if (err < 0) return err; diff --git a/sound/pci/ca0106/ca0106_mixer.c b/sound/pci/ca0106/ca0106_mixer.c index 3b8ec673dc0a..c852c6a75b91 100644 --- a/sound/pci/ca0106/ca0106_mixer.c +++ b/sound/pci/ca0106/ca0106_mixer.c @@ -739,7 +739,7 @@ static int rename_ctl(struct snd_card *card, const char *src, const char *dst) static DECLARE_TLV_DB_SCALE(snd_ca0106_master_db_scale, -6375, 25, 1); -static const char * const slave_vols[] = { +static const char * const follower_vols[] = { "Analog Front Playback Volume", "Analog Rear Playback Volume", "Analog Center/LFE Playback Volume", @@ -752,7 +752,7 @@ static const char * const slave_vols[] = { NULL }; -static const char * const slave_sws[] = { +static const char * const follower_sws[] = { "Analog Front Playback Switch", "Analog Rear Playback Switch", "Analog Center/LFE Playback Switch", @@ -761,13 +761,13 @@ static const char * const slave_sws[] = { NULL }; -static void add_slaves(struct snd_card *card, - struct snd_kcontrol *master, const char * const *list) +static void add_followers(struct snd_card *card, + struct snd_kcontrol *master, const char * const *list) { for (; *list; list++) { - struct snd_kcontrol *slave = ctl_find(card, *list); - if (slave) - snd_ctl_add_slave(master, slave); + struct snd_kcontrol *follower = ctl_find(card, *list); + if (follower) + snd_ctl_add_follower(master, follower); } } @@ -852,7 +852,7 @@ int snd_ca0106_mixer(struct snd_ca0106 *emu) err = snd_ctl_add(card, vmaster); if (err < 0) return err; - add_slaves(card, vmaster, slave_vols); + add_followers(card, vmaster, follower_vols); if (emu->details->spi_dac) { vmaster = snd_ctl_make_virtual_master("Master Playback Switch", @@ -862,7 +862,7 @@ int snd_ca0106_mixer(struct snd_ca0106 *emu) err = snd_ctl_add(card, vmaster); if (err < 0) return err; - add_slaves(card, vmaster, slave_sws); + add_followers(card, vmaster, follower_sws); } strcpy(card->mixername, "CA0106"); diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 7e3ae4534df9..40f3c175954d 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -785,7 +785,7 @@ void snd_hda_codec_cleanup_for_unbind(struct hda_codec *codec) snd_array_free(&codec->spdif_out); snd_array_free(&codec->verbs); codec->preset = NULL; - codec->slave_dig_outs = NULL; + codec->follower_dig_outs = NULL; codec->spdif_status_reset = 0; snd_array_free(&codec->mixers); snd_array_free(&codec->nids); @@ -1806,11 +1806,11 @@ int snd_hda_codec_reset(struct hda_codec *codec) return 0; } -typedef int (*map_slave_func_t)(struct hda_codec *, void *, struct snd_kcontrol *); +typedef int (*map_follower_func_t)(struct hda_codec *, void *, struct snd_kcontrol *); -/* apply the function to all matching slave ctls in the mixer list */ -static int map_slaves(struct hda_codec *codec, const char * const *slaves, - const char *suffix, map_slave_func_t func, void *data) +/* apply the function to all matching follower ctls in the mixer list */ +static int map_followers(struct hda_codec *codec, const char * const *followers, + const char *suffix, map_follower_func_t func, void *data) { struct hda_nid_item *items; const char * const *s; @@ -1821,7 +1821,7 @@ static int map_slaves(struct hda_codec *codec, const char * const *slaves, struct snd_kcontrol *sctl = items[i].kctl; if (!sctl || sctl->id.iface != SNDRV_CTL_ELEM_IFACE_MIXER) continue; - for (s = slaves; *s; s++) { + for (s = followers; *s; s++) { char tmpname[sizeof(sctl->id.name)]; const char *name = *s; if (suffix) { @@ -1840,8 +1840,8 @@ static int map_slaves(struct hda_codec *codec, const char * const *slaves, return 0; } -static int check_slave_present(struct hda_codec *codec, - void *data, struct snd_kcontrol *sctl) +static int check_follower_present(struct hda_codec *codec, + void *data, struct snd_kcontrol *sctl) { return 1; } @@ -1860,17 +1860,17 @@ static int put_kctl_with_value(struct snd_kcontrol *kctl, int val) return 0; } -struct slave_init_arg { +struct follower_init_arg { struct hda_codec *codec; int step; }; -/* initialize the slave volume with 0dB via snd_ctl_apply_vmaster_slaves() */ -static int init_slave_0dB(struct snd_kcontrol *slave, - struct snd_kcontrol *kctl, - void *_arg) +/* initialize the follower volume with 0dB via snd_ctl_apply_vmaster_followers() */ +static int init_follower_0dB(struct snd_kcontrol *follower, + struct snd_kcontrol *kctl, + void *_arg) { - struct slave_init_arg *arg = _arg; + struct follower_init_arg *arg = _arg; int _tlv[4]; const int *tlv = NULL; int step; @@ -1879,7 +1879,7 @@ static int init_slave_0dB(struct snd_kcontrol *slave, if (kctl->vd[0].access & SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK) { if (kctl->tlv.c != snd_hda_mixer_amp_tlv) { codec_err(arg->codec, - "Unexpected TLV callback for slave %s:%d\n", + "Unexpected TLV callback for follower %s:%d\n", kctl->id.name, kctl->id.index); return 0; /* ignore */ } @@ -1897,7 +1897,7 @@ static int init_slave_0dB(struct snd_kcontrol *slave, return 0; if (arg->step && arg->step != step) { codec_err(arg->codec, - "Mismatching dB step for vmaster slave (%d!=%d)\n", + "Mismatching dB step for vmaster follower (%d!=%d)\n", arg->step, step); return 0; } @@ -1905,49 +1905,49 @@ static int init_slave_0dB(struct snd_kcontrol *slave, arg->step = step; val = -tlv[SNDRV_CTL_TLVO_DB_SCALE_MIN] / step; if (val > 0) { - put_kctl_with_value(slave, val); + put_kctl_with_value(follower, val); return val; } return 0; } -/* unmute the slave via snd_ctl_apply_vmaster_slaves() */ -static int init_slave_unmute(struct snd_kcontrol *slave, - struct snd_kcontrol *kctl, - void *_arg) +/* unmute the follower via snd_ctl_apply_vmaster_followers() */ +static int init_follower_unmute(struct snd_kcontrol *follower, + struct snd_kcontrol *kctl, + void *_arg) { - return put_kctl_with_value(slave, 1); + return put_kctl_with_value(follower, 1); } -static int add_slave(struct hda_codec *codec, - void *data, struct snd_kcontrol *slave) +static int add_follower(struct hda_codec *codec, + void *data, struct snd_kcontrol *follower) { - return snd_ctl_add_slave(data, slave); + return snd_ctl_add_follower(data, follower); } /** - * __snd_hda_add_vmaster - create a virtual master control and add slaves + * __snd_hda_add_vmaster - create a virtual master control and add followers * @codec: HD-audio codec * @name: vmaster control name * @tlv: TLV data (optional) - * @slaves: slave control names (optional) - * @suffix: suffix string to each slave name (optional) - * @init_slave_vol: initialize slaves to unmute/0dB + * @followers: follower control names (optional) + * @suffix: suffix string to each follower name (optional) + * @init_follower_vol: initialize followers to unmute/0dB * @ctl_ret: store the vmaster kcontrol in return * * Create a virtual master control with the given name. The TLV data * must be either NULL or a valid data. * - * @slaves is a NULL-terminated array of strings, each of which is a - * slave control name. All controls with these names are assigned to + * @followers is a NULL-terminated array of strings, each of which is a + * follower control name. All controls with these names are assigned to * the new virtual master control. * * This function returns zero if successful or a negative error code. */ int __snd_hda_add_vmaster(struct hda_codec *codec, char *name, - unsigned int *tlv, const char * const *slaves, - const char *suffix, bool init_slave_vol, + unsigned int *tlv, const char * const *followers, + const char *suffix, bool init_follower_vol, struct snd_kcontrol **ctl_ret) { struct snd_kcontrol *kctl; @@ -1956,9 +1956,9 @@ int __snd_hda_add_vmaster(struct hda_codec *codec, char *name, if (ctl_ret) *ctl_ret = NULL; - err = map_slaves(codec, slaves, suffix, check_slave_present, NULL); + err = map_followers(codec, followers, suffix, check_follower_present, NULL); if (err != 1) { - codec_dbg(codec, "No slave found for %s\n", name); + codec_dbg(codec, "No follower found for %s\n", name); return 0; } kctl = snd_ctl_make_virtual_master(name, tlv); @@ -1968,20 +1968,20 @@ int __snd_hda_add_vmaster(struct hda_codec *codec, char *name, if (err < 0) return err; - err = map_slaves(codec, slaves, suffix, add_slave, kctl); + err = map_followers(codec, followers, suffix, add_follower, kctl); if (err < 0) return err; /* init with master mute & zero volume */ put_kctl_with_value(kctl, 0); - if (init_slave_vol) { - struct slave_init_arg arg = { + if (init_follower_vol) { + struct follower_init_arg arg = { .codec = codec, .step = 0, }; - snd_ctl_apply_vmaster_slaves(kctl, - tlv ? init_slave_0dB : init_slave_unmute, - &arg); + snd_ctl_apply_vmaster_followers(kctl, + tlv ? init_follower_0dB : init_follower_unmute, + &arg); } if (ctl_ret) @@ -2284,7 +2284,7 @@ static unsigned int convert_to_spdif_status(unsigned short val) return sbits; } -/* set digital convert verbs both for the given NID and its slaves */ +/* set digital convert verbs both for the given NID and its followers */ static void set_dig_out(struct hda_codec *codec, hda_nid_t nid, int mask, int val) { @@ -2292,7 +2292,7 @@ static void set_dig_out(struct hda_codec *codec, hda_nid_t nid, snd_hdac_regmap_update(&codec->core, nid, AC_VERB_SET_DIGI_CONVERT_1, mask, val); - d = codec->slave_dig_outs; + d = codec->follower_dig_outs; if (!d) return; for (; *d; d++) @@ -3580,9 +3580,9 @@ static void setup_dig_out_stream(struct hda_codec *codec, hda_nid_t nid, spdif->ctls & ~AC_DIG1_ENABLE & 0xff, -1); snd_hda_codec_setup_stream(codec, nid, stream_tag, 0, format); - if (codec->slave_dig_outs) { + if (codec->follower_dig_outs) { const hda_nid_t *d; - for (d = codec->slave_dig_outs; *d; d++) + for (d = codec->follower_dig_outs; *d; d++) snd_hda_codec_setup_stream(codec, *d, stream_tag, 0, format); } @@ -3595,9 +3595,9 @@ static void setup_dig_out_stream(struct hda_codec *codec, hda_nid_t nid, static void cleanup_dig_out_stream(struct hda_codec *codec, hda_nid_t nid) { snd_hda_codec_cleanup_stream(codec, nid); - if (codec->slave_dig_outs) { + if (codec->follower_dig_outs) { const hda_nid_t *d; - for (d = codec->slave_dig_outs; *d; d++) + for (d = codec->follower_dig_outs; *d; d++) snd_hda_codec_cleanup_stream(codec, *d); } } @@ -3679,7 +3679,7 @@ EXPORT_SYMBOL_GPL(snd_hda_multi_out_dig_close); * @hinfo: PCM information to assign * * Open analog outputs and set up the hw-constraints. - * If the digital outputs can be opened as slave, open the digital + * If the digital outputs can be opened as follower, open the digital * outputs, too. */ int snd_hda_multi_out_analog_open(struct hda_codec *codec, diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 9074265d934a..d07947a73ac2 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -4112,7 +4112,7 @@ static void parse_digital(struct hda_codec *codec) int i, nums; hda_nid_t dig_nid, pin; - /* support multiple SPDIFs; the secondary is set up as a slave */ + /* support multiple SPDIFs; the secondary is set up as a follower */ nums = 0; for (i = 0; i < spec->autocfg.dig_outs; i++) { pin = spec->autocfg.dig_out_pins[i]; @@ -4131,10 +4131,10 @@ static void parse_digital(struct hda_codec *codec) spec->multiout.dig_out_nid = dig_nid; spec->dig_out_type = spec->autocfg.dig_out_type[0]; } else { - spec->multiout.slave_dig_outs = spec->slave_dig_outs; - if (nums >= ARRAY_SIZE(spec->slave_dig_outs) - 1) + spec->multiout.follower_dig_outs = spec->follower_dig_outs; + if (nums >= ARRAY_SIZE(spec->follower_dig_outs) - 1) break; - spec->slave_dig_outs[nums - 1] = dig_nid; + spec->follower_dig_outs[nums - 1] = dig_nid; } nums++; } @@ -4589,7 +4589,7 @@ static void call_update_outputs(struct hda_codec *codec) else snd_hda_gen_update_outputs(codec); - /* sync the whole vmaster slaves to reflect the new auto-mute status */ + /* sync the whole vmaster followers to reflect the new auto-mute status */ if (spec->auto_mute_via_amp && !codec->bus->shutdown) snd_ctl_sync_vmaster(spec->vmaster_mute.sw_kctl, false); } @@ -5233,8 +5233,8 @@ EXPORT_SYMBOL_GPL(snd_hda_gen_parse_auto_config); * Build control elements */ -/* slave controls for virtual master */ -static const char * const slave_pfxs[] = { +/* follower controls for virtual master */ +static const char * const follower_pfxs[] = { "Front", "Surround", "Center", "LFE", "Side", "Headphone", "Speaker", "Mono", "Line Out", "CLFE", "Bass Speaker", "PCM", @@ -5286,7 +5286,7 @@ int snd_hda_gen_build_controls(struct hda_codec *codec) if (!spec->no_analog && !spec->suppress_vmaster && !snd_hda_find_mixer_ctl(codec, "Master Playback Volume")) { err = snd_hda_add_vmaster(codec, "Master Playback Volume", - spec->vmaster_tlv, slave_pfxs, + spec->vmaster_tlv, follower_pfxs, "Playback Volume"); if (err < 0) return err; @@ -5294,7 +5294,7 @@ int snd_hda_gen_build_controls(struct hda_codec *codec) if (!spec->no_analog && !spec->suppress_vmaster && !snd_hda_find_mixer_ctl(codec, "Master Playback Switch")) { err = __snd_hda_add_vmaster(codec, "Master Playback Switch", - NULL, slave_pfxs, + NULL, follower_pfxs, "Playback Switch", true, &spec->vmaster_mute.sw_kctl); if (err < 0) @@ -5809,7 +5809,7 @@ int snd_hda_gen_build_pcms(struct hda_codec *codec) spec->stream_name_digital); if (!info) return -ENOMEM; - codec->slave_dig_outs = spec->multiout.slave_dig_outs; + codec->follower_dig_outs = spec->multiout.follower_dig_outs; spec->pcm_rec[1] = info; if (spec->dig_out_type) info->pcm_type = spec->dig_out_type; diff --git a/sound/pci/hda/hda_generic.h b/sound/pci/hda/hda_generic.h index bbd6d2b741f2..a43f0bb77dae 100644 --- a/sound/pci/hda/hda_generic.h +++ b/sound/pci/hda/hda_generic.h @@ -116,7 +116,7 @@ struct hda_gen_spec { * dig_out_nid and hp_nid are optional */ hda_nid_t alt_dac_nid; - hda_nid_t slave_dig_outs[3]; /* optional - for auto-parsing */ + hda_nid_t follower_dig_outs[3]; /* optional - for auto-parsing */ int dig_out_type; /* capture */ diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 3dca65d79b02..8c28b1022f49 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -129,11 +129,11 @@ void snd_hda_set_vmaster_tlv(struct hda_codec *codec, hda_nid_t nid, int dir, struct snd_kcontrol *snd_hda_find_mixer_ctl(struct hda_codec *codec, const char *name); int __snd_hda_add_vmaster(struct hda_codec *codec, char *name, - unsigned int *tlv, const char * const *slaves, - const char *suffix, bool init_slave_vol, + unsigned int *tlv, const char * const *followers, + const char *suffix, bool init_follower_vol, struct snd_kcontrol **ctl_ret); -#define snd_hda_add_vmaster(codec, name, tlv, slaves, suffix) \ - __snd_hda_add_vmaster(codec, name, tlv, slaves, suffix, true, NULL) +#define snd_hda_add_vmaster(codec, name, tlv, followers, suffix) \ + __snd_hda_add_vmaster(codec, name, tlv, followers, suffix, true, NULL) int snd_hda_codec_reset(struct hda_codec *codec); void snd_hda_codec_register(struct hda_codec *codec); void snd_hda_codec_cleanup_for_unbind(struct hda_codec *codec); @@ -216,7 +216,7 @@ struct hda_multi_out { hda_nid_t hp_out_nid[HDA_MAX_OUTS]; /* DACs for multiple HPs */ hda_nid_t extra_out_nid[HDA_MAX_OUTS]; /* other (e.g. speaker) DACs */ hda_nid_t dig_out_nid; /* digital out audio widget */ - const hda_nid_t *slave_dig_outs; + const hda_nid_t *follower_dig_outs; int max_channels; /* currently supported analog channels */ int dig_out_used; /* current usage of digital out (HDA_DIG_XXX) */ int no_share_stream; /* don't share a stream with multiple pins */ diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 34fe753a46fb..19c575fd28a1 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -6245,10 +6245,10 @@ static int zxr_add_headphone_gain_switch(struct hda_codec *codec) } /* - * Need to create slave controls for the alternate codecs that have surround + * Need to create follower controls for the alternate codecs that have surround * capabilities. */ -static const char * const ca0132_alt_slave_pfxs[] = { +static const char * const ca0132_alt_follower_pfxs[] = { "Front", "Surround", "Center", "LFE", NULL, }; @@ -6376,15 +6376,15 @@ static int ca0132_build_controls(struct hda_codec *codec) if (err < 0) return err; } - /* Setup vmaster with surround slaves for desktop ca0132 devices */ + /* Setup vmaster with surround followers for desktop ca0132 devices */ if (ca0132_use_alt_functions(spec)) { snd_hda_set_vmaster_tlv(codec, spec->dacs[0], HDA_OUTPUT, spec->tlv); snd_hda_add_vmaster(codec, "Master Playback Volume", - spec->tlv, ca0132_alt_slave_pfxs, + spec->tlv, ca0132_alt_follower_pfxs, "Playback Volume"); err = __snd_hda_add_vmaster(codec, "Master Playback Switch", - NULL, ca0132_alt_slave_pfxs, + NULL, ca0132_alt_follower_pfxs, "Playback Switch", true, &spec->vmaster_mute.sw_kctl); if (err < 0) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index d8ed69cb9df1..7f5ed5805d3b 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -3135,7 +3135,7 @@ static void fixup_hp_headphone(struct hda_codec *codec, hda_nid_t pin) unsigned int pin_cfg = snd_hda_codec_get_pincfg(codec, pin); /* It was changed in the BIOS to just satisfy MS DTM. - * Lets turn it back into slaved HP + * Lets turn it back into follower HP */ pin_cfg = (pin_cfg & (~AC_DEFCFG_DEVICE)) | (AC_JACK_HP_OUT << AC_DEFCFG_DEVICE_SHIFT); diff --git a/sound/pci/ice1712/juli.c b/sound/pci/ice1712/juli.c index 7be4eb42f05e..e57a55cebc5a 100644 --- a/sound/pci/ice1712/juli.c +++ b/sound/pci/ice1712/juli.c @@ -397,7 +397,7 @@ static const struct snd_kcontrol_new juli_mute_controls[] = { }, }; -static const char * const slave_vols[] = { +static const char * const follower_vols[] = { PCM_VOLUME, MONITOR_AN_IN_VOLUME, MONITOR_DIG_IN_VOLUME, @@ -418,16 +418,16 @@ static struct snd_kcontrol *ctl_find(struct snd_card *card, return snd_ctl_find_id(card, &sid); } -static void add_slaves(struct snd_card *card, - struct snd_kcontrol *master, - const char * const *list) +static void add_followers(struct snd_card *card, + struct snd_kcontrol *master, + const char * const *list) { for (; *list; list++) { - struct snd_kcontrol *slave = ctl_find(card, *list); - /* dev_dbg(card->dev, "add_slaves - %s\n", *list); */ - if (slave) { - /* dev_dbg(card->dev, "slave %s found\n", *list); */ - snd_ctl_add_slave(master, slave); + struct snd_kcontrol *follower = ctl_find(card, *list); + /* dev_dbg(card->dev, "add_followers - %s\n", *list); */ + if (follower) { + /* dev_dbg(card->dev, "follower %s found\n", *list); */ + snd_ctl_add_follower(master, follower); } } } @@ -454,7 +454,7 @@ static int juli_add_controls(struct snd_ice1712 *ice) juli_master_db_scale); if (!vmaster) return -ENOMEM; - add_slaves(ice->card, vmaster, slave_vols); + add_followers(ice->card, vmaster, follower_vols); err = snd_ctl_add(ice->card, vmaster); if (err < 0) return err; diff --git a/sound/pci/ice1712/quartet.c b/sound/pci/ice1712/quartet.c index 866596205710..0e3e04aa9faf 100644 --- a/sound/pci/ice1712/quartet.c +++ b/sound/pci/ice1712/quartet.c @@ -757,7 +757,7 @@ static const struct snd_kcontrol_new qtet_controls[] = { QTET_CONTROL("Output 3/4 to Monitor 1/2", sw, OUT34_MON12), }; -static const char * const slave_vols[] = { +static const char * const follower_vols[] = { PCM_12_PLAYBACK_VOLUME, PCM_34_PLAYBACK_VOLUME, NULL @@ -776,13 +776,13 @@ static struct snd_kcontrol *ctl_find(struct snd_card *card, return snd_ctl_find_id(card, &sid); } -static void add_slaves(struct snd_card *card, - struct snd_kcontrol *master, const char * const *list) +static void add_followers(struct snd_card *card, + struct snd_kcontrol *master, const char * const *list) { for (; *list; list++) { - struct snd_kcontrol *slave = ctl_find(card, *list); - if (slave) - snd_ctl_add_slave(master, slave); + struct snd_kcontrol *follower = ctl_find(card, *list); + if (follower) + snd_ctl_add_follower(master, follower); } } @@ -806,7 +806,7 @@ static int qtet_add_controls(struct snd_ice1712 *ice) qtet_master_db_scale); if (!vmaster) return -ENOMEM; - add_slaves(ice->card, vmaster, slave_vols); + add_followers(ice->card, vmaster, follower_vols); err = snd_ctl_add(ice->card, vmaster); if (err < 0) return err; diff --git a/sound/ppc/awacs.c b/sound/ppc/awacs.c index 73c0fd7277e6..53d558b2806c 100644 --- a/sound/ppc/awacs.c +++ b/sound/ppc/awacs.c @@ -1063,12 +1063,12 @@ snd_pmac_awacs_init(struct snd_pmac *chip) if (pm5500 || imac || lombard) { vmaster_sw = snd_ctl_make_virtual_master( "Master Playback Switch", (unsigned int *) NULL); - err = snd_ctl_add_slave_uncached(vmaster_sw, - chip->master_sw_ctl); + err = snd_ctl_add_follower_uncached(vmaster_sw, + chip->master_sw_ctl); if (err < 0) return err; - err = snd_ctl_add_slave_uncached(vmaster_sw, - chip->speaker_sw_ctl); + err = snd_ctl_add_follower_uncached(vmaster_sw, + chip->speaker_sw_ctl); if (err < 0) return err; err = snd_ctl_add(chip->card, vmaster_sw); @@ -1076,10 +1076,10 @@ snd_pmac_awacs_init(struct snd_pmac *chip) return err; vmaster_vol = snd_ctl_make_virtual_master( "Master Playback Volume", (unsigned int *) NULL); - err = snd_ctl_add_slave(vmaster_vol, master_vol); + err = snd_ctl_add_follower(vmaster_vol, master_vol); if (err < 0) return err; - err = snd_ctl_add_slave(vmaster_vol, speaker_vol); + err = snd_ctl_add_follower(vmaster_vol, speaker_vol); if (err < 0) return err; err = snd_ctl_add(chip->card, vmaster_vol); diff --git a/sound/usb/6fire/control.c b/sound/usb/6fire/control.c index 20f34d2ace5f..9bd8dcbb68e4 100644 --- a/sound/usb/6fire/control.c +++ b/sound/usb/6fire/control.c @@ -539,7 +539,7 @@ static int usb6fire_control_add_virtual( ret = snd_ctl_add(card, control); if (ret < 0) return ret; - ret = snd_ctl_add_slave(vmaster, control); + ret = snd_ctl_add_follower(vmaster, control); if (ret < 0) return ret; i++; -- 2.16.4

1 0

[PATCH] ASoC: qcom: qdsp6: Use the defined variable to clean code
by Tang Bin 17 Jul '20

17 Jul '20

Use the defined variable "dev" to make the code cleaner. Signed-off-by: Zhang Shengju <zhangshengju(a)cmss.chinamobile.com> Signed-off-by: Tang Bin <tangbin(a)cmss.chinamobile.com> --- sound/soc/qcom/qdsp6/q6adm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/qcom/qdsp6/q6adm.c b/sound/soc/qcom/qdsp6/q6adm.c index da242515e..2ec868b2d 100644 --- a/sound/soc/qcom/qdsp6/q6adm.c +++ b/sound/soc/qcom/qdsp6/q6adm.c @@ -588,12 +588,12 @@ static int q6adm_probe(struct apr_device *adev) struct device *dev = &adev->dev; struct q6adm *adm; - adm = devm_kzalloc(&adev->dev, sizeof(*adm), GFP_KERNEL); + adm = devm_kzalloc(dev, sizeof(*adm), GFP_KERNEL); if (!adm) return -ENOMEM; adm->apr = adev; - dev_set_drvdata(&adev->dev, adm); + dev_set_drvdata(dev, adm); adm->dev = dev; q6core_get_svc_api_info(adev->svc_id, &adm->ainfo); mutex_init(&adm->lock); -- 2.20.1.windows.1

2 1

[PATCH v2] ASoC: meson: fixes the missed kfree() for axg_card_add_tdm_loopback
by Jing Xiangfeng 17 Jul '20

17 Jul '20

axg_card_add_tdm_loopback() misses to call kfree() in an error path. We can use devm_kasprintf() to fix the issue, also improve maintainability. So use it instead. Fixes: c84836d7f650 ("ASoC: meson: axg-card: use modern dai_link style") Signed-off-by: Jing Xiangfeng <jingxiangfeng(a)huawei.com> --- sound/soc/meson/axg-card.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/meson/axg-card.c b/sound/soc/meson/axg-card.c index 89f7f64747cd..47f2d93224fe 100644 --- a/sound/soc/meson/axg-card.c +++ b/sound/soc/meson/axg-card.c @@ -116,7 +116,7 @@ static int axg_card_add_tdm_loopback(struct snd_soc_card *card, lb = &card->dai_link[*index + 1]; - lb->name = kasprintf(GFP_KERNEL, "%s-lb", pad->name); + lb->name = devm_kasprintf(card->dev, GFP_KERNEL, "%s-lb", pad->name); if (!lb->name) return -ENOMEM; -- 2.17.1

3 2

[PATCH] ASoC: soc-dai.h: don't use discriminatory terms for comment
by Kuninori Morimoto 17 Jul '20

17 Jul '20

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> soc-dai is using discriminatory terms for comment. This patch renames "slave" to "secondary", thus we can keep M/S initials. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- include/sound/soc-dai.h | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 6cb25e8f9b20..09ab2d542639 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -76,12 +76,12 @@ struct snd_compr_stream; * * This is wrt the codec, the inverse is true for the interface * i.e. if the codec is clk and FRM master then the interface is - * clk and frame slave. + * clk and frame secondary. */ #define SND_SOC_DAIFMT_CBM_CFM (1 << 12) /* codec clk & FRM master */ -#define SND_SOC_DAIFMT_CBS_CFM (2 << 12) /* codec clk slave & FRM master */ -#define SND_SOC_DAIFMT_CBM_CFS (3 << 12) /* codec clk master & frame slave */ -#define SND_SOC_DAIFMT_CBS_CFS (4 << 12) /* codec clk & FRM slave */ +#define SND_SOC_DAIFMT_CBS_CFM (2 << 12) /* codec clk secondary & FRM master */ +#define SND_SOC_DAIFMT_CBM_CFS (3 << 12) /* codec clk master & frame secondary */ +#define SND_SOC_DAIFMT_CBS_CFS (4 << 12) /* codec clk & FRM secondary */ #define SND_SOC_DAIFMT_FORMAT_MASK 0x000f #define SND_SOC_DAIFMT_CLOCK_MASK 0x00f0 -- 2.25.1

2 1

[PATCH] ASoC: qcom: qdsp6: Use IS_ERR() instead of IS_ERR_OR_NULL()
by Tang Bin 17 Jul '20

17 Jul '20

In the function q6adm_open(), q6adm_alloc_copp() doesn't return NULL. Thus use IS_ERR() to validate the returned value instead of IS_ERR_OR_NULL(). And delete the extra line. Signed-off-by: Zhang Shengju <zhangshengju(a)cmss.chinamobile.com> Signed-off-by: Tang Bin <tangbin(a)cmss.chinamobile.com> --- sound/soc/qcom/qdsp6/q6adm.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/qcom/qdsp6/q6adm.c b/sound/soc/qcom/qdsp6/q6adm.c index d8069864b..44ac0728f 100644 --- a/sound/soc/qcom/qdsp6/q6adm.c +++ b/sound/soc/qcom/qdsp6/q6adm.c @@ -403,7 +403,7 @@ struct q6copp *q6adm_open(struct device *dev, int port_id, int path, int rate, spin_lock_irqsave(&adm->copps_list_lock, flags); copp = q6adm_alloc_copp(adm, port_id); - if (IS_ERR_OR_NULL(copp)) { + if (IS_ERR(copp)) { spin_unlock_irqrestore(&adm->copps_list_lock, flags); return ERR_CAST(copp); } @@ -419,7 +419,6 @@ struct q6copp *q6adm_open(struct device *dev, int port_id, int path, int rate, copp->bit_width = bit_width; copp->app_type = app_type; - ret = q6adm_device_open(adm, copp, port_id, path, topology, channel_mode, bit_width, rate); if (ret < 0) { -- 2.20.1.windows.1

2 1

wm8962: error at soc_component_read_no_lock
by Fabio Estevam 17 Jul '20

17 Jul '20

Hi, Running linux-next 20200714 on a imx6q sabresd I see the following errors: wm8962 0-001a: ASoC: error at soc_component_read_no_lock on wm8962.0-001a: -16 wm8962 0-001a: ASoC: error at soc_component_read_no_lock on wm8962.0-001a: -16 input: WM8962 Beep Generator as /devices/platform/soc/2100000.bus/21a0000.i2c/i2c-0/0-001a/input/input3 wm8962 0-001a: ASoC: error at soc_component_read_no_lock on wm8962.0-001a: -16 wm8962 0-001a: ASoC: error at soc_component_read_no_lock on wm8962.0-001a: -16 Despite these errors, the codec probes fine. What is the correct way to avoid such errors? Thanks, Fabio Estevam

3 6

[PATCH v4 00/23] Add support for Tegra210 Audio
by Sameer Pujar 17 Jul '20

17 Jul '20

Overview ======== Audio Processing Engine (APE) comprises of Audio DMA (ADMA) and Audio Hub (AHUB) unit. AHUB is a collection of hardware accelerators for audio pre-processing and post-processing. It also includes a programmable full crossbar for routing audio data across these accelerators. This series exposes some of these below mentioned HW devices as ASoC components for Tegra platforms from Tegra210 onwards. * ADMAIF : The interface between ADMA and AHUB * XBAR : Crossbar for routing audio samples across various modules * I2S : Inter-IC Sound Controller * DMIC : Digital Microphone * DSPK : Digital Speaker Following is the summary of current series. * Add YAML DT binding documentation for above mentioned modules. * Helper function for ACIF programming is exposed for Tegra210 and later. * Add ASoC driver components for each of the above modules. * Add DT entries for above components for Tegra210, Tegra186 and Tegra194. * Enable these components for Jetson Nano/TX1/TX2/Xavier * Enhance simple-card DPCM driver to suit Tegra Audio applications with few changes in core. * To begin with, enable sound card support for Tegra210 based platforms like Jetson Nano/TX1. Sound card support for platforms based on Tegra186 and later will be subsequently added which can re-use all of the above components. Changelog ========= v3 -> v4 -------- * [1/23] "ASoC: dt-bindings: tegra: Add DT bindings for Tegra210" - Removed multiple examples and retained one example per doc - Fixed as per inputs on the previous series - Tested bindings with 'make dt_binding_check/dtbs_check' * [2/23] "ASoC: tegra: Add support for CIF programming" - No change * Common changes (for patch [3/10] to [7/10]) - Mixer control overrides, for PCM parameters (rate, channel, bits), in each driver are dropped. - Updated routing as per DPCM usage - Minor changes related to formatting * New changes (patch [8/23] to [18/23] and patch [23/23]) - Based on discussions in following threads DPCM is used for Tegra Audio. https://lkml.org/lkml/2020/2/20/91 https://lkml.org/lkml/2020/4/30/519 - The simple-card driver is used for Tegra Audio and accordingly some enhancements are made in simple-card and core drivers. - Patch [8/23] to [18/23] are related to simple-card and core changes. - Patch [23/23] adds sound card support to realize complete audio path. This is based on simple-card driver with proposed enhancements. - Re-ordered patches depending on above v2 -> v3 -------- * [1/10] "dt-bindings: sound: tegra: add DT binding for AHUB - Updated licence - Removed redundancy w.r.t items/const/enum - Added constraints wherever needed with "pattern" property * [2/10] "ASoC: tegra: add support for CIF programming" - Removed tegra_cif.c - Instead added inline helper function in tegra_cif.h * common changes (for patch [3/10] to [7/10]) - Replace LATE system calls with Normal sleep - Remove explicit RPM suspend in driver remove() call - Use devm_kzalloc() instead of devm_kcalloc() for single element - Replace 'ret' with 'err' for better reading - Consistent error printing style across drivers - Minor formating fixes * [8/10] "arm64: tegra: add AHUB components for few Tegra chips" - no change * [9/10] "arm64: tegra: enable AHUB modules for few Tegra chips" - no change * [10/10] "arm64: defconfig: enable AHUB components for Tegra210 and later" (New patch) - Enables ACONNECT and AHUB components. With this AHUB and components are registered with ASoC core. v1 -> v2 -------- * [1/9] "dt-bindings: sound: tegra: add DT binding for AHUB" - no changes * [2/9] "ASoC: tegra: add support for CIF programming" - removed CIF programming changes for legacy chips. - this patch now exposes helper function for CIF programming, which can be used on Tegra210 later. - later tegra_cif.c can be extended for legacy chips as well. - updated commit message accordingly * [3/9] "ASoC: tegra: add Tegra210 based DMIC driver" - removed unnecessary initialization of 'ret' in probe() * [4/9] "ASoC: tegra: add Tegra210 based I2S driver" - removed unnecessary initialization of 'ret' in probe() - fixed indentation - added consistent bracing for if-else clauses - updated 'rx_fifo_th' type to 'unsigned int' - used BIT() macro for defines like '1 << {x}' in tegra210_i2s.h * [5/9] "ASoC: tegra: add Tegra210 based AHUB driver" - used of_device_get_match_data() to get 'soc_data' and removed explicit of_match_device() - used devm_platform_ioremap_resource() and removed explicit platform_get_resource() - fixed indentation for devm_snd_soc_register_component() - updated commit message - updated commit message to reflect compatible binding for Tegra186 and Tegra194. * [6/9] "ASoC: tegra: add Tegra186 based DSPK driver" - removed unnecessary initialization of 'ret' in probe() - updated 'max_th' to 'unsigned int' - shortened lengthy macro names to avoid wrapping in tegra186_dspk_wr_reg() and to be consistent * [7/9] "ASoC: tegra: add Tegra210 based ADMAIF driver" - used of_device_get_match_data() and removed explicit of_match_device() - used BIT() macro for defines like '1 << {x}' in tegra210_admaif.h - updated commit message to reflect compatible binding for Tegra186 and Tegra194. * [8/9] "arm64: tegra: add AHUB components for few Tegra chips" - no change * [9/9] "arm64: tegra: enable AHUB modules for few Tegra chips" - no change * common changes for patch [3/9] to [7/9] - sorted headers in alphabetical order - moved MODULE_DEVICE_TABLE() right below *_of_match table - removed macro DRV_NAME - removed explicit 'owner' field from platform_driver structure - added 'const' to snd_soc_dai_ops structure ================== Sameer Pujar (23): ASoC: dt-bindings: tegra: Add DT bindings for Tegra210 ASoC: tegra: Add support for CIF programming ASoC: tegra: Add Tegra210 based DMIC driver ASoC: tegra: Add Tegra210 based I2S driver ASoC: tegra: Add Tegra210 based AHUB driver ASoC: tegra: Add Tegra186 based DSPK driver ASoC: tegra: Add Tegra210 based ADMAIF driver ASoC: soc-core: Fix component name_prefix parsing ASoC: simple-card: Use of_node and DAI names for DAI link names ASoC: simple-card: Wrong daifmt for CPU end of DPCM DAI link ASoC: simple-card: Loop over all children for 'mclk-fs' ASoC: simple-card: Support DPCM DAI link with multiple Codecs ASoC: simple-card: DPCM DAI link direction as per DAI capability ASoC: soc-core: Probe auxiliary component before others ASoC: soc-core: Identify 'no_pcm' DAI links for DPCM ASoC: soc-pcm: Get all BEs along DAPM path ASoC: dt-bindings: simple-card: Add compatible for component chaining ASoC: simple-card: Add support for component chaining arm64: defconfig: Build AHUB component drivers arm64: defconfig: Enable CONFIG_TEGRA210_ADMA arm64: tegra: Add DT binding for AHUB components arm64: tegra: Enable AHUB components on few Tegra platforms arm64: tegra: Add support for APE sound card on Jetson Nano and TX1 .../bindings/sound/nvidia,tegra186-dspk.yaml | 88 +++ .../bindings/sound/nvidia,tegra210-admaif.yaml | 116 +++ .../bindings/sound/nvidia,tegra210-ahub.yaml | 144 ++++ .../bindings/sound/nvidia,tegra210-dmic.yaml | 88 +++ .../bindings/sound/nvidia,tegra210-i2s.yaml | 106 +++ .../devicetree/bindings/sound/simple-card.yaml | 1 + arch/arm64/boot/dts/nvidia/tegra186-p2771-0000.dts | 48 ++ arch/arm64/boot/dts/nvidia/tegra186.dtsi | 231 +++++- arch/arm64/boot/dts/nvidia/tegra194-p2972-0000.dts | 36 + arch/arm64/boot/dts/nvidia/tegra194.dtsi | 239 +++++- arch/arm64/boot/dts/nvidia/tegra210-p2371-2180.dts | 136 +++- arch/arm64/boot/dts/nvidia/tegra210-p3450-0000.dts | 85 +++ arch/arm64/boot/dts/nvidia/tegra210.dtsi | 219 +++++- arch/arm64/configs/defconfig | 8 + include/sound/simple_card.h | 5 + include/sound/soc.h | 3 + sound/soc/generic/simple-card.c | 96 ++- sound/soc/soc-core.c | 43 +- sound/soc/soc-dai.c | 1 + sound/soc/soc-pcm.c | 3 +- sound/soc/tegra/Kconfig | 56 ++ sound/soc/tegra/Makefile | 10 + sound/soc/tegra/tegra186_dspk.c | 427 +++++++++++ sound/soc/tegra/tegra186_dspk.h | 70 ++ sound/soc/tegra/tegra210_admaif.c | 845 +++++++++++++++++++++ sound/soc/tegra/tegra210_admaif.h | 162 ++++ sound/soc/tegra/tegra210_ahub.c | 578 ++++++++++++++ sound/soc/tegra/tegra210_ahub.h | 100 +++ sound/soc/tegra/tegra210_dmic.c | 440 +++++++++++ sound/soc/tegra/tegra210_dmic.h | 82 ++ sound/soc/tegra/tegra210_i2s.c | 780 +++++++++++++++++++ sound/soc/tegra/tegra210_i2s.h | 126 +++ sound/soc/tegra/tegra_cif.h | 65 ++ sound/soc/tegra/tegra_pcm.c | 235 +++++- sound/soc/tegra/tegra_pcm.h | 21 +- 35 files changed, 5657 insertions(+), 36 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra186-dspk.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-admaif.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-ahub.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-dmic.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-i2s.yaml create mode 100644 sound/soc/tegra/tegra186_dspk.c create mode 100644 sound/soc/tegra/tegra186_dspk.h create mode 100644 sound/soc/tegra/tegra210_admaif.c create mode 100644 sound/soc/tegra/tegra210_admaif.h create mode 100644 sound/soc/tegra/tegra210_ahub.c create mode 100644 sound/soc/tegra/tegra210_ahub.h create mode 100644 sound/soc/tegra/tegra210_dmic.c create mode 100644 sound/soc/tegra/tegra210_dmic.h create mode 100644 sound/soc/tegra/tegra210_i2s.c create mode 100644 sound/soc/tegra/tegra210_i2s.h create mode 100644 sound/soc/tegra/tegra_cif.h -- 2.7.4

3 62

[bug report] ASoC: soc-dai: return proper error for get_sdw_stream()
by dan.carpenter＠oracle.com 17 Jul '20

17 Jul '20

Hello Srinivas Kandagatla, The patch 308811a327c3: "ASoC: soc-dai: return proper error for get_sdw_stream()" from Mar 16, 2020, leads to the following static checker warning: drivers/soundwire/stream.c:1920 sdw_shutdown_stream() error: 'sdw_stream' dereferencing possible ERR_PTR() drivers/soundwire/stream.c 1902 void sdw_shutdown_stream(void *sdw_substream) 1903 { 1904 struct snd_pcm_substream *substream = sdw_substream; 1905 struct snd_soc_pcm_runtime *rtd = substream->private_data; 1906 struct sdw_stream_runtime *sdw_stream; 1907 struct snd_soc_dai *dai; 1908 1909 /* Find stream from first CPU DAI */ 1910 dai = asoc_rtd_to_cpu(rtd, 0); 1911 1912 sdw_stream = snd_soc_dai_get_sdw_stream(dai, substream->stream); ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ We updated this to return error pointers as well as NULL but forgot to check the caller. 1913 1914 if (!sdw_stream) { 1915 dev_err(rtd->dev, "no stream found for DAI %s", dai->name); 1916 return; 1917 } 1918 1919 /* release memory */ 1920 kfree(sdw_stream->name); 1921 sdw_release_stream(sdw_stream); 1922 regards, dan carpenter

1 0

[PATCH -next] ALSA: line6: Use kmemdup in podhd_set_monitor_level()
by YueHaibing 17 Jul '20

17 Jul '20

Use kmemdup rather than duplicating its implementation. Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/usb/line6/podhd.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/usb/line6/podhd.c b/sound/usb/line6/podhd.c index 1557483ec657..eef45f7fef0d 100644 --- a/sound/usb/line6/podhd.c +++ b/sound/usb/line6/podhd.c @@ -293,12 +293,10 @@ static void podhd_set_monitor_level(struct usb_line6_podhd *podhd, int value) }; unsigned char *buf; - buf = kmalloc(sizeof(msg), GFP_KERNEL); + buf = kmemdup(msg, sizeof(msg), GFP_KERNEL); if (!buf) return; - memcpy(buf, msg, sizeof(msg)); - if (value < 0) value = 0; -- 2.17.1

2 1

[PATCH] ALSA: info: Drop WARN_ON() from buffer NULL sanity check
by Takashi Iwai 17 Jul '20

17 Jul '20

snd_info_get_line() has a sanity check of NULL buffer -- both buffer itself being NULL and buffer->buffer being NULL. Basically both checks are valid and necessary, but the problem is that it's with snd_BUG_ON() macro that triggers WARN_ON(). The latter condition (NULL buffer->buffer) can be met arbitrarily by user since the buffer is allocated at the first write, so it means that user can trigger WARN_ON() at will. This patch addresses it by simply moving buffer->buffer NULL check out of snd_BUG_ON() so that spurious WARNING is no longer triggered. Reported-by: syzbot+e42d0746c3c3699b6061(a)syzkaller.appspotmail.com Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/info.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/core/info.c b/sound/core/info.c index 8c6bc5241df5..9fec3070f8ba 100644 --- a/sound/core/info.c +++ b/sound/core/info.c @@ -606,7 +606,9 @@ int snd_info_get_line(struct snd_info_buffer *buffer, char *line, int len) { int c; - if (snd_BUG_ON(!buffer || !buffer->buffer)) + if (snd_BUG_ON(!buffer)) + return 1; + if (!buffer->buffer) return 1; if (len <= 0 || buffer->stop || buffer->error) return 1; -- 2.16.4

1 0

WARNING in snd_info_get_line
by syzbot 17 Jul '20

17 Jul '20

Hello, syzbot found the following issue on: HEAD commit: f8456690 Merge tag 'clk-fixes-for-linus' of git://git.kern.. git tree: upstream console output: https://syzkaller.appspot.com/x/log.txt?x=11539507100000 kernel config: https://syzkaller.appspot.com/x/.config?x=e944500a36bc4d55 dashboard link: https://syzkaller.appspot.com/bug?extid=e42d0746c3c3699b6061 compiler: clang version 10.0.0 (https://github.com/llvm/llvm-project/ c2443155a0fb245c8f17f2c1c72b6ea391e86e81) syz repro: https://syzkaller.appspot.com/x/repro.syz?x=1531c5ab100000 C reproducer: https://syzkaller.appspot.com/x/repro.c?x=17c399c7100000 IMPORTANT: if you fix the issue, please add the following tag to the commit: Reported-by: syzbot+e42d0746c3c3699b6061(a)syzkaller.appspotmail.com ------------[ cut here ]------------ WARNING: CPU: 1 PID: 6804 at sound/core/info.c:609 snd_info_get_line+0x3d0/0x450 sound/core/info.c:609 Kernel panic - not syncing: panic_on_warn set ... CPU: 1 PID: 6804 Comm: syz-executor373 Not tainted 5.8.0-rc5-syzkaller #0 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 Call Trace: __dump_stack lib/dump_stack.c:77 [inline] dump_stack+0x1f0/0x31e lib/dump_stack.c:118 panic+0x264/0x7a0 kernel/panic.c:231 __warn+0x227/0x250 kernel/panic.c:600 report_bug+0x1b1/0x2e0 lib/bug.c:198 handle_bug+0x42/0x80 arch/x86/kernel/traps.c:235 exc_invalid_op+0x16/0x40 arch/x86/kernel/traps.c:255 asm_exc_invalid_op+0x12/0x20 arch/x86/include/asm/idtentry.h:542 RIP: 0010:snd_info_get_line+0x3d0/0x450 sound/core/info.c:609 Code: 0f 85 84 00 00 00 41 c6 45 00 00 31 ed 89 e8 48 83 c4 40 5b 41 5c 41 5d 41 5e 41 5f 5d c3 e8 57 8a 3b fb eb 05 e8 50 8a 3b fb <0f> 0b bd 01 00 00 00 eb da 89 d9 80 e1 07 80 c1 03 38 c1 0f 8c a3 RSP: 0018:ffffc90001077a38 EFLAGS: 00010293 RAX: ffffffff86390470 RBX: 0000000000000080 RCX: ffff888096156500 RDX: 0000000000000000 RSI: ffffc90001077b10 RDI: ffff8880a1e30600 RBP: ffffc90001077bc8 R08: dffffc0000000000 R09: ffffed10429a6594 R10: ffffed10429a6594 R11: 0000000000000000 R12: ffff8880a2527908 R13: ffffc90001077b10 R14: 1ffff110143c60c0 R15: ffff8880a1e30600 snd_mixer_oss_proc_write+0x6d/0x670 sound/core/oss/mixer_oss.c:1179 snd_info_text_entry_release+0xdf/0x1c0 sound/core/info.c:409 close_pdeo+0x16c/0x310 fs/proc/inode.c:244 proc_reg_release+0x17c/0x1c0 fs/proc/inode.c:561 __fput+0x2f0/0x750 fs/file_table.c:281 task_work_run+0x137/0x1c0 kernel/task_work.c:135 exit_task_work include/linux/task_work.h:25 [inline] do_exit+0x601/0x1f80 kernel/exit.c:805 do_group_exit+0x161/0x2d0 kernel/exit.c:903 __do_sys_exit_group+0x13/0x20 kernel/exit.c:914 __se_sys_exit_group+0x10/0x10 kernel/exit.c:912 __x64_sys_exit_group+0x37/0x40 kernel/exit.c:912 do_syscall_64+0x73/0xe0 arch/x86/entry/common.c:384 entry_SYSCALL_64_after_hwframe+0x44/0xa9 RIP: 0033:0x43eeb8 Code: Bad RIP value. RSP: 002b:00007ffce5a0e308 EFLAGS: 00000246 ORIG_RAX: 00000000000000e7 RAX: ffffffffffffffda RBX: 0000000000000000 RCX: 000000000043eeb8 RDX: 0000000000000000 RSI: 000000000000003c RDI: 0000000000000000 RBP: 00000000004be6c8 R08: 00000000000000e7 R09: ffffffffffffffd0 R10: 0000000000000000 R11: 0000000000000246 R12: 0000000000000001 R13: 00000000006d0180 R14: 0000000000000000 R15: 0000000000000000 Kernel Offset: disabled Rebooting in 86400 seconds.. --- This report is generated by a bot. It may contain errors. See https://goo.gl/tpsmEJ for more information about syzbot. syzbot engineers can be reached at syzkaller(a)googlegroups.com. syzbot will keep track of this issue. See: https://goo.gl/tpsmEJ#status for how to communicate with syzbot. syzbot can test patches for this issue, for details see: https://goo.gl/tpsmEJ#testing-patches

1 0

Adding USB device quirk - packet format?
by Olivia Mackintosh 17 Jul '20

17 Jul '20

Hello, I am trying to add a device quirk for the Pioneer DJM-750 mixer. I have attached a patch to illustrate what I'm trying to achieve. It is nearly identical to https://patchwork.kernel.org/patch/11521019/ except for the differences in channels. I have verified the settings against valid USB transfers. While observing behaviour with a Windows 10 VM, the 8 pieces of data appear to be 8 isochronous packets within a single frame as opposed to 8 microframes (each with their own transfer). Should this have any impact on the ability for the device to function? If so, is there a setting I am missing to account for this? Or, am I simply clutching at straws and completely on the wrong path. This is rather device specific so I would not expect anyone to help troubleshoot this specific device but what I suspect is the problem is the way the audio data is arranged in the transfer. Any suggestions and pointers on how to proceed are very much appreciated. Kind regards, Olivia Mackintosh

2 1

[PATCH 2/2] ASoC: rt5682: disable MICBIAS and Vref2 widget in default
by shumingf＠realtek.com 17 Jul '20

17 Jul '20

From: Shuming Fan <shumingf(a)realtek.com> The pin status of the widget was connected after the sound card registered. The rt5682_headset_detect function will use the pin status of these two widgets to decide the certain register setting on/off. Therefore this patch disables the pin of these two widgets in the codec probe. This patch could avoid the misjudgment. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt5682.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index c06f5763573c..a4713bd6508d 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -2847,6 +2847,7 @@ static int rt5682_probe(struct snd_soc_component *component) struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); struct sdw_slave *slave; unsigned long time; + struct snd_soc_dapm_context *dapm = &component->dapm; #ifdef CONFIG_COMMON_CLK int ret; @@ -2884,6 +2885,9 @@ static int rt5682_probe(struct snd_soc_component *component) #endif } + snd_soc_dapm_disable_pin(dapm, "MICBIAS"); + snd_soc_dapm_disable_pin(dapm, "Vref2"); + snd_soc_dapm_sync(dapm); return 0; } -- 2.27.0

1 0

[PATCH 1/2] ASoC: rt5682: optimize the power consumption
by shumingf＠realtek.com 17 Jul '20

17 Jul '20

From: Shuming Fan <shumingf(a)realtek.com> Some settings should set to default value after the calibration. This patch also disables the 25MHz and 1MHz clock power when the jack unplugged. The JD is triggered by JDH, therefore this patch removes JDL setting. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt5682-i2c.c | 2 +- sound/soc/codecs/rt5682-sdw.c | 2 +- sound/soc/codecs/rt5682.c | 12 +++++++++--- 3 files changed, 11 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/rt5682-i2c.c b/sound/soc/codecs/rt5682-i2c.c index b24f93ff0e55..85aba311bdc8 100644 --- a/sound/soc/codecs/rt5682-i2c.c +++ b/sound/soc/codecs/rt5682-i2c.c @@ -232,7 +232,7 @@ static int rt5682_i2c_probe(struct i2c_client *i2c, regmap_update_bits(rt5682->regmap, RT5682_PWR_ANLG_1, RT5682_LDO1_DVO_MASK | RT5682_HP_DRIVER_MASK, RT5682_LDO1_DVO_12 | RT5682_HP_DRIVER_5X); - regmap_write(rt5682->regmap, RT5682_MICBIAS_2, 0x0380); + regmap_write(rt5682->regmap, RT5682_MICBIAS_2, 0x0080); regmap_update_bits(rt5682->regmap, RT5682_GPIO_CTRL_1, RT5682_GP4_PIN_MASK | RT5682_GP5_PIN_MASK, RT5682_GP4_PIN_ADCDAT1 | RT5682_GP5_PIN_DACDAT1); diff --git a/sound/soc/codecs/rt5682-sdw.c b/sound/soc/codecs/rt5682-sdw.c index 4cecc5ce545c..94bf6bee78e6 100644 --- a/sound/soc/codecs/rt5682-sdw.c +++ b/sound/soc/codecs/rt5682-sdw.c @@ -431,7 +431,7 @@ static int rt5682_io_init(struct device *dev, struct sdw_slave *slave) regmap_update_bits(rt5682->regmap, RT5682_PWR_ANLG_1, RT5682_LDO1_DVO_MASK | RT5682_HP_DRIVER_MASK, RT5682_LDO1_DVO_12 | RT5682_HP_DRIVER_5X); - regmap_write(rt5682->regmap, RT5682_MICBIAS_2, 0x0380); + regmap_write(rt5682->regmap, RT5682_MICBIAS_2, 0x0080); regmap_write(rt5682->regmap, RT5682_TEST_MODE_CTRL_1, 0x0000); regmap_update_bits(rt5682->regmap, RT5682_BIAS_CUR_CTRL_8, RT5682_HPA_CP_BIAS_CTRL_MASK, RT5682_HPA_CP_BIAS_3UA); diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index d7302f968a75..c06f5763573c 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -963,6 +963,9 @@ int rt5682_headset_detect(struct snd_soc_component *component, int jack_insert) RT5682_HP_CHARGE_PUMP_1, RT5682_OSW_L_MASK | RT5682_OSW_R_MASK, RT5682_OSW_L_EN | RT5682_OSW_R_EN); + snd_soc_component_update_bits(component, RT5682_MICBIAS_2, + RT5682_PWR_CLK25M_MASK | RT5682_PWR_CLK1M_MASK, + RT5682_PWR_CLK25M_PU | RT5682_PWR_CLK1M_PU); } else { rt5682_enable_push_button_irq(component, false); snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_1, @@ -975,6 +978,9 @@ int rt5682_headset_detect(struct snd_soc_component *component, int jack_insert) RT5682_PWR_ANLG_1, RT5682_PWR_VREF2, 0); snd_soc_component_update_bits(component, RT5682_PWR_ANLG_3, RT5682_PWR_CBJ, 0); + snd_soc_component_update_bits(component, RT5682_MICBIAS_2, + RT5682_PWR_CLK25M_MASK | RT5682_PWR_CLK1M_MASK, + RT5682_PWR_CLK25M_PD | RT5682_PWR_CLK1M_PD); rt5682->jack_type = 0; } @@ -1022,8 +1028,7 @@ static int rt5682_set_jack_detect(struct snd_soc_component *component, RT5682_POW_ANA, RT5682_POW_IRQ | RT5682_POW_JDH | RT5682_POW_ANA); regmap_update_bits(rt5682->regmap, RT5682_PWR_ANLG_2, - RT5682_PWR_JDH | RT5682_PWR_JDL, - RT5682_PWR_JDH | RT5682_PWR_JDL); + RT5682_PWR_JDH, RT5682_PWR_JDH); regmap_update_bits(rt5682->regmap, RT5682_IRQ_CTRL_2, RT5682_JD1_EN_MASK | RT5682_JD1_POL_MASK, RT5682_JD1_EN | RT5682_JD1_POL_NOR); @@ -3031,13 +3036,14 @@ void rt5682_calibrate(struct rt5682_priv *rt5682) dev_err(rt5682->component->dev, "HP Calibration Failure\n"); /* restore settings */ - regmap_write(rt5682->regmap, RT5682_PWR_ANLG_1, 0x02af); + regmap_write(rt5682->regmap, RT5682_PWR_ANLG_1, 0x002f); regmap_write(rt5682->regmap, RT5682_MICBIAS_2, 0x0080); regmap_write(rt5682->regmap, RT5682_GLB_CLK, 0x0000); regmap_write(rt5682->regmap, RT5682_PWR_DIG_1, 0x0000); regmap_write(rt5682->regmap, RT5682_CHOP_DAC, 0x2000); regmap_write(rt5682->regmap, RT5682_CALIB_ADC_CTRL, 0x2005); regmap_write(rt5682->regmap, RT5682_STO1_ADC_MIXER, 0xc0c4); + regmap_write(rt5682->regmap, RT5682_CAL_REC, 0x0c0c); mutex_unlock(&rt5682->calibrate_mutex); } -- 2.27.0

1 0

[PATCH] Revert "ALSA: pcm: Use SG-buffer only when direct DMA is available"
by Takashi Iwai 17 Jul '20

17 Jul '20

This reverts commit 3ad796cbc36a ("ALSA: pcm: Use SG-buffer only when direct DMA is available") also the modification commit 467fd0e82b62 ("ALSA: pcm: Fix build error on m68k and others"). Poking the DMA internal helper is a layer violation, so we should avoid that. Meanwhile the actual bug has been addressed by the Kconfig fix in commit dbed452a078d ("dma-pool: decouple DMA_REMAP from DMA_COHERENT_POOL"), so we can live without this hack. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/pcm_memory.c | 14 -------------- 1 file changed, 14 deletions(-) diff --git a/sound/core/pcm_memory.c b/sound/core/pcm_memory.c index 795af1b88051..1bf6a3d9e0c2 100644 --- a/sound/core/pcm_memory.c +++ b/sound/core/pcm_memory.c @@ -11,7 +11,6 @@ #include <linux/moduleparam.h> #include <linux/vmalloc.h> #include <linux/export.h> -#include <linux/dma-mapping.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/info.h> @@ -41,19 +40,6 @@ static int do_alloc_pages(struct snd_card *card, int type, struct device *dev, card->total_pcm_alloc_bytes + size > max_alloc_per_card) return -ENOMEM; - -#ifdef CONFIG_SND_DMA_SGBUF - if ((type == SNDRV_DMA_TYPE_DEV_SG || type == SNDRV_DMA_TYPE_DEV_UC_SG) && - !dma_is_direct(get_dma_ops(dev))) { - /* mutate to continuous page allocation */ - dev_dbg(dev, "Use continuous page allocator\n"); - if (type == SNDRV_DMA_TYPE_DEV_SG) - type = SNDRV_DMA_TYPE_DEV; - else - type = SNDRV_DMA_TYPE_DEV_UC; - } -#endif /* CONFIG_SND_DMA_SGBUF */ - err = snd_dma_alloc_pages(type, dev, size, dmab); if (!err) { mutex_lock(&card->memory_mutex); -- 2.16.4

1 0

[PATCH] ASoC: meson: add the missed kfree() for axg_card_add_tdm_loopback
by Jing Xiangfeng 17 Jul '20

17 Jul '20

axg_card_add_tdm_loopback() misses to call kfree() in an error path. Add the missed function call to fix it. Fixes: c84836d7f650 ("ASoC: meson: axg-card: use modern dai_link style") Signed-off-by: Jing Xiangfeng <jingxiangfeng(a)huawei.com> --- sound/soc/meson/axg-card.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/meson/axg-card.c b/sound/soc/meson/axg-card.c index 89f7f64747cd..6eac22ba8b99 100644 --- a/sound/soc/meson/axg-card.c +++ b/sound/soc/meson/axg-card.c @@ -121,8 +121,10 @@ static int axg_card_add_tdm_loopback(struct snd_soc_card *card, return -ENOMEM; dlc = devm_kzalloc(card->dev, 2 * sizeof(*dlc), GFP_KERNEL); - if (!dlc) + if (!dlc) { + kfree(lb->name); return -ENOMEM; + } lb->cpus = &dlc[0]; lb->codecs = &dlc[1]; -- 2.17.1

3 3

[PATCH v3 00/21] ASoC: merge .digital_mute() into .mute_stream()
by Kuninori Morimoto 17 Jul '20

17 Jul '20

Hi Mark These are v3 patch-set. ALSA SoC has 2 mute callbacks (= .digital_mute(), .mute_stream()). But the difference between these 2 are very small. .digital_mute() is for Playback .mute_stream() is for Playback/Capture This patch-set adds new .no_capture_mute flag and emulate .digital_mute() by .mute_stream(). v2 -> v3 - uses "xxx_mute_stream" for .mute_stream naming if it was better - removed verbose Cc email address v1 -> v2 - return -ENOTSUPP at hdmi-codec - add new .no_capture_mute flag and emulate .digital_mute() by .mute_stream() Link: https://lore.kernel.org/r/874kqy2y5t.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87ftam37ko.wl-kuninori.morimoto.gx@renesas.com Kuninori Morimoto (21): ASoC: hdmi-codec: return -ENOTSUPP for digital_mute ASoC: soc-dai.c: add .no_capture_mute support ASoC: hdmi-codec: merge .digital_mute() into .mute_stream() ASoC: ti: merge .digital_mute() into .mute_stream() ASoC: spear: merge .digital_mute() into .mute_stream() ASoC: meson: merge .digital_mute() into .mute_stream() ASoC: atmel: merge .digital_mute() into .mute_stream() ASoC: codecs: merge .digital_mute() into .mute_stream() ASoC: codecs: tlv*: merge .digital_mute() into .mute_stream() ASoC: codecs: tas*: merge .digital_mute() into .mute_stream() ASoC: codecs: ssm*: merge .digital_mute() into .mute_stream() ASoC: codecs: pcm*: merge .digital_mute() into .mute_stream() ASoC: codecs: max*: merge .digital_mute() into .mute_stream() ASoC: codecs: alc*: merge .digital_mute() into .mute_stream() ASoC: codecs: wm*: merge .digital_mute() into .mute_stream() ASoC: codecs: es*: merge .digital_mute() into .mute_stream() ASoC: codecs: da*: merge .digital_mute() into .mute_stream() ASoC: codecs: cs*: merge .digital_mute() into .mute_stream() ASoC: codecs: ak*: merge .digital_mute() into .mute_stream() ASoC: soc-dai: remove .digital_mute ASoC: soc-core: snd_soc_dai_digital_mute() for both CPU/Codec drivers/gpu/drm/bridge/sii902x.c | 7 ++++--- drivers/gpu/drm/exynos/exynos_hdmi.c | 6 ++++-- drivers/gpu/drm/i2c/tda998x_drv.c | 7 ++++--- drivers/gpu/drm/mediatek/mtk_hdmi.c | 6 ++++-- drivers/gpu/drm/rockchip/cdn-dp-core.c | 7 ++++--- drivers/gpu/drm/sti/sti_hdmi.c | 6 ++++-- drivers/gpu/drm/zte/zx_hdmi.c | 7 ++++--- include/sound/hdmi-codec.h | 6 +++++- include/sound/soc-dai.h | 4 +++- sound/soc/atmel/atmel-classd.c | 7 ++++--- sound/soc/codecs/88pm860x-codec.c | 8 +++++--- sound/soc/codecs/ad193x.c | 5 +++-- sound/soc/codecs/adau1701.c | 5 +++-- sound/soc/codecs/ak4458.c | 5 +++-- sound/soc/codecs/ak4535.c | 6 ++++-- sound/soc/codecs/ak4641.c | 8 +++++--- sound/soc/codecs/alc5623.c | 5 +++-- sound/soc/codecs/alc5632.c | 5 +++-- sound/soc/codecs/cpcap.c | 11 +++++++---- sound/soc/codecs/cq93vc.c | 5 +++-- sound/soc/codecs/cs4265.c | 5 +++-- sound/soc/codecs/cs4270.c | 5 +++-- sound/soc/codecs/cs42l42.c | 5 +++-- sound/soc/codecs/cs42l51.c | 5 +++-- sound/soc/codecs/cs42l52.c | 5 +++-- sound/soc/codecs/cs42l56.c | 5 +++-- sound/soc/codecs/cs42xx8.c | 5 +++-- sound/soc/codecs/cs4341.c | 5 +++-- sound/soc/codecs/cs4349.c | 5 +++-- sound/soc/codecs/da7210.c | 5 +++-- sound/soc/codecs/da7213.c | 5 +++-- sound/soc/codecs/da9055.c | 5 +++-- sound/soc/codecs/es8316.c | 5 +++-- sound/soc/codecs/es8328.c | 5 +++-- sound/soc/codecs/hdmi-codec.c | 25 +++++++++++++++++-------- sound/soc/codecs/isabelle.c | 15 +++++++++------ sound/soc/codecs/jz4770.c | 5 +++-- sound/soc/codecs/lm49453.c | 25 +++++++++++++++---------- sound/soc/codecs/max98088.c | 12 ++++++++---- sound/soc/codecs/max98090.c | 6 ++++-- sound/soc/codecs/max9867.c | 5 +++-- sound/soc/codecs/ml26124.c | 5 +++-- sound/soc/codecs/nau8822.c | 5 +++-- sound/soc/codecs/pcm1681.c | 5 +++-- sound/soc/codecs/pcm1789.c | 5 +++-- sound/soc/codecs/pcm179x.c | 5 +++-- sound/soc/codecs/pcm3168a.c | 5 +++-- sound/soc/codecs/pcm512x.c | 5 +++-- sound/soc/codecs/rk3328_codec.c | 5 +++-- sound/soc/codecs/sgtl5000.c | 5 +++-- sound/soc/codecs/ssm2518.c | 5 +++-- sound/soc/codecs/ssm2602.c | 5 +++-- sound/soc/codecs/ssm4567.c | 5 +++-- sound/soc/codecs/sta529.c | 5 +++-- sound/soc/codecs/tas2552.c | 5 +++-- sound/soc/codecs/tas2562.c | 5 +++-- sound/soc/codecs/tas2770.c | 5 +++-- sound/soc/codecs/tas571x.c | 5 +++-- sound/soc/codecs/tas5720.c | 5 +++-- sound/soc/codecs/tas6424.c | 5 +++-- sound/soc/codecs/tfa9879.c | 5 +++-- sound/soc/codecs/tlv320aic23.c | 5 +++-- sound/soc/codecs/tlv320aic26.c | 5 +++-- sound/soc/codecs/tlv320aic31xx.c | 6 ++++-- sound/soc/codecs/tlv320aic32x4.c | 5 +++-- sound/soc/codecs/tlv320aic3x.c | 5 +++-- sound/soc/codecs/twl6040.c | 5 +++-- sound/soc/codecs/uda134x.c | 5 +++-- sound/soc/codecs/wm8350.c | 5 +++-- sound/soc/codecs/wm8400.c | 5 +++-- sound/soc/codecs/wm8510.c | 5 +++-- sound/soc/codecs/wm8580.c | 5 +++-- sound/soc/codecs/wm8711.c | 5 +++-- sound/soc/codecs/wm8728.c | 5 +++-- sound/soc/codecs/wm8731.c | 5 +++-- sound/soc/codecs/wm8741.c | 5 +++-- sound/soc/codecs/wm8750.c | 5 +++-- sound/soc/codecs/wm8753.c | 8 +++++--- sound/soc/codecs/wm8770.c | 5 +++-- sound/soc/codecs/wm8776.c | 5 +++-- sound/soc/codecs/wm8900.c | 5 +++-- sound/soc/codecs/wm8903.c | 5 +++-- sound/soc/codecs/wm8904.c | 5 +++-- sound/soc/codecs/wm8940.c | 5 +++-- sound/soc/codecs/wm8955.c | 5 +++-- sound/soc/codecs/wm8960.c | 5 +++-- sound/soc/codecs/wm8961.c | 5 +++-- sound/soc/codecs/wm8962.c | 5 +++-- sound/soc/codecs/wm8971.c | 5 +++-- sound/soc/codecs/wm8974.c | 5 +++-- sound/soc/codecs/wm8978.c | 5 +++-- sound/soc/codecs/wm8983.c | 7 ++++--- sound/soc/codecs/wm8985.c | 7 ++++--- sound/soc/codecs/wm8988.c | 5 +++-- sound/soc/codecs/wm8990.c | 5 +++-- sound/soc/codecs/wm8991.c | 7 ++++--- sound/soc/codecs/wm8993.c | 5 +++-- sound/soc/codecs/wm8994.c | 9 ++++++--- sound/soc/codecs/wm8995.c | 8 +++++--- sound/soc/codecs/wm9081.c | 5 +++-- sound/soc/meson/axg-spdifout.c | 5 +++-- sound/soc/soc-core.c | 4 ++-- sound/soc/soc-dai.c | 12 +++++++----- sound/soc/spear/spdif_out.c | 8 +++++--- sound/soc/ti/ams-delta.c | 9 +++++---- 105 files changed, 390 insertions(+), 249 deletions(-) -- 2.25.1

2 25

[PATCH] ASoC: Intel: bytcht_es8316: Add missed put_device()
by Jing Xiangfeng 17 Jul '20

17 Jul '20

snd_byt_cht_es8316_mc_probe() misses to call put_device() in an error path. Add the missed function call to fix it. Fixes: ba49cf6f8e4a ("ASoC: Intel: bytcht_es8316: Add quirk for inverted jack detect") Signed-off-by: Jing Xiangfeng <jingxiangfeng(a)huawei.com> --- sound/soc/intel/boards/bytcht_es8316.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/bytcht_es8316.c b/sound/soc/intel/boards/bytcht_es8316.c index 9e5fc9430628..ecbc58e8a37f 100644 --- a/sound/soc/intel/boards/bytcht_es8316.c +++ b/sound/soc/intel/boards/bytcht_es8316.c @@ -543,8 +543,10 @@ static int snd_byt_cht_es8316_mc_probe(struct platform_device *pdev) if (cnt) { ret = device_add_properties(codec_dev, props); - if (ret) + if (ret) { + put_device(codec_dev); return ret; + } } devm_acpi_dev_add_driver_gpios(codec_dev, byt_cht_es8316_gpios); -- 2.17.1

4 5

[PATCH v3] ASoC: dt-bindings: ak4642: switch to yaml base Documentation
by Kuninori Morimoto 17 Jul '20

17 Jul '20

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> This patch switches from .txt base to .yaml base Document. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Reviewed-by: Rob Herring <robh(a)kernel.org> Link: https://lore.kernel.org/r/87lfjwxlna.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87mu4z3904.wl-kuninori.morimoto.gx@renesas.com --- v2 -> v3 - add Reviewd-by - add Link - use generic node name on examples .../devicetree/bindings/sound/ak4642.txt | 37 ------------ .../devicetree/bindings/sound/ak4642.yaml | 58 +++++++++++++++++++ 2 files changed, 58 insertions(+), 37 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/ak4642.txt create mode 100644 Documentation/devicetree/bindings/sound/ak4642.yaml diff --git a/Documentation/devicetree/bindings/sound/ak4642.txt b/Documentation/devicetree/bindings/sound/ak4642.txt deleted file mode 100644 index 58e48ee97175..000000000000 --- a/Documentation/devicetree/bindings/sound/ak4642.txt +++ /dev/null @@ -1,37 +0,0 @@ -AK4642 I2C transmitter - -This device supports I2C mode only. - -Required properties: - - - compatible : "asahi-kasei,ak4642" or "asahi-kasei,ak4643" or "asahi-kasei,ak4648" - - reg : The chip select number on the I2C bus - -Optional properties: - - - #clock-cells : common clock binding; shall be set to 0 - - clocks : common clock binding; MCKI clock - - clock-frequency : common clock binding; frequency of MCKO - - clock-output-names : common clock binding; MCKO clock name - -Example 1: - -&i2c { - ak4648: ak4648@12 { - compatible = "asahi-kasei,ak4642"; - reg = <0x12>; - }; -}; - -Example 2: - -&i2c { - ak4643: codec@12 { - compatible = "asahi-kasei,ak4643"; - reg = <0x12>; - #clock-cells = <0>; - clocks = <&audio_clock>; - clock-frequency = <12288000>; - clock-output-names = "ak4643_mcko"; - }; -}; diff --git a/Documentation/devicetree/bindings/sound/ak4642.yaml b/Documentation/devicetree/bindings/sound/ak4642.yaml new file mode 100644 index 000000000000..6cd213be2266 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ak4642.yaml @@ -0,0 +1,58 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ak4642.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: AK4642 I2C transmitter Device Tree Bindings + +maintainers: + - Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> + +properties: + compatible: + enum: + - asahi-kasei,ak4642 + - asahi-kasei,ak4643 + - asahi-kasei,ak4648 + + reg: + maxItems: 1 + + "#clock-cells": + const: 0 + "#sound-dai-cells": + const: 0 + + clocks: + maxItems: 1 + + clock-frequency: + description: common clock binding; frequency of MCKO + $ref: /schemas/types.yaml#/definitions/uint32 + + clock-output-names: + description: common clock name + $ref: /schemas/types.yaml#/definitions/string + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + ak4643: codec@12 { + compatible = "asahi-kasei,ak4643"; + #sound-dai-cells = <0>; + reg = <0x12>; + #clock-cells = <0>; + clocks = <&audio_clock>; + clock-frequency = <12288000>; + clock-output-names = "ak4643_mcko"; + }; + }; -- 2.25.1

2 1

[PATCH][resend] ASoC: dt-bindings: ak4642: switch to yaml base Documentation
by Kuninori Morimoto 17 Jul '20

17 Jul '20

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> This patch switches from .txt base to .yaml base Document. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- - 2weeks passed, but nothing happen .../devicetree/bindings/sound/ak4642.txt | 37 ------------ .../devicetree/bindings/sound/ak4642.yaml | 57 +++++++++++++++++++ 2 files changed, 57 insertions(+), 37 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/ak4642.txt create mode 100644 Documentation/devicetree/bindings/sound/ak4642.yaml diff --git a/Documentation/devicetree/bindings/sound/ak4642.txt b/Documentation/devicetree/bindings/sound/ak4642.txt deleted file mode 100644 index 58e48ee97175..000000000000 --- a/Documentation/devicetree/bindings/sound/ak4642.txt +++ /dev/null @@ -1,37 +0,0 @@ -AK4642 I2C transmitter - -This device supports I2C mode only. - -Required properties: - - - compatible : "asahi-kasei,ak4642" or "asahi-kasei,ak4643" or "asahi-kasei,ak4648" - - reg : The chip select number on the I2C bus - -Optional properties: - - - #clock-cells : common clock binding; shall be set to 0 - - clocks : common clock binding; MCKI clock - - clock-frequency : common clock binding; frequency of MCKO - - clock-output-names : common clock binding; MCKO clock name - -Example 1: - -&i2c { - ak4648: ak4648@12 { - compatible = "asahi-kasei,ak4642"; - reg = <0x12>; - }; -}; - -Example 2: - -&i2c { - ak4643: codec@12 { - compatible = "asahi-kasei,ak4643"; - reg = <0x12>; - #clock-cells = <0>; - clocks = <&audio_clock>; - clock-frequency = <12288000>; - clock-output-names = "ak4643_mcko"; - }; -}; diff --git a/Documentation/devicetree/bindings/sound/ak4642.yaml b/Documentation/devicetree/bindings/sound/ak4642.yaml new file mode 100644 index 000000000000..c8ffc86fd2d3 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ak4642.yaml @@ -0,0 +1,57 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ak4642.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: AK4642 I2C transmitter Device Tree Bindings + +maintainers: + - Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> + +properties: + compatible: + enum: + - asahi-kasei,ak4642 + - asahi-kasei,ak4643 + - asahi-kasei,ak4648 + + reg: + maxItems: 1 + + "#clock-cells": + const: 0 + + "#sound-dai-cells": + const: 0 + + clocks: + maxItems: 1 + + clock-frequency: + description: frequency of MCKO + + clock-output-names: + maxItems: 1 + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + ak4643: codec@12 { + compatible = "asahi-kasei,ak4643"; + #sound-dai-cells = <0>; + reg = <0x12>; + #clock-cells = <0>; + clocks = <&audio_clock>; + clock-frequency = <12288000>; + clock-output-names = "ak4643_mcko"; + }; + }; -- 2.25.1

3 2

[PATCH 1/1] ASoC: fsl: fsl-asoc-card: Trivial: Fix misspelling of 'exists'
by Lee Jones 17 Jul '20

17 Jul '20

Cc: Timur Tabi <timur(a)kernel.org> Cc: Nicolin Chen <nicoleotsuka(a)gmail.com> Cc: Xiubo Li <Xiubo.Lee(a)gmail.com> Cc: Fabio Estevam <festevam(a)gmail.com> Cc: linuxppc-dev(a)lists.ozlabs.org Signed-off-by: Lee Jones <lee.jones(a)linaro.org> --- sound/soc/fsl/fsl-asoc-card.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index faac6ce9a82cb..399c9aad48b1d 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -92,8 +92,7 @@ struct fsl_asoc_card_priv { }; /* - * This dapm route map exits for DPCM link only. + * This dapm route map exists for DPCM link only. * The other routes shall go through Device Tree. * * Note: keep all ASRC routes in the second half -- 2.25.1

2 4

[PATCH v2 0/3] ASoC: fsl-asoc-card: Support hp and mic detection
by Shengjiu Wang 17 Jul '20

17 Jul '20

Support hp and mic detection. Add a parameter for asoc_simple_init_jack. Shengjiu Wang (3): ASoC: simple-card-utils: Support configure pin_name for asoc_simple_init_jack ASoC: bindings: fsl-asoc-card: Support hp-det-gpio and mic-det-gpio ASoC: fsl-asoc-card: Support Headphone and Microphone Jack detection changes in v2: - Add more comments in third commit - Add Acked-by Nicolin. .../bindings/sound/fsl-asoc-card.txt | 3 + include/sound/simple_card_utils.h | 6 +- sound/soc/fsl/Kconfig | 1 + sound/soc/fsl/fsl-asoc-card.c | 77 ++++++++++++++++++- sound/soc/generic/simple-card-utils.c | 7 +- 5 files changed, 86 insertions(+), 8 deletions(-) -- 2.27.0

3 5

[PATCH v2] ASoC: soc-component: Add missed return for calling soc_component_ret
by Shengjiu Wang 17 Jul '20

17 Jul '20

Add missed return for calling soc_component_ret, otherwise the return value is wrong. Fixes: e2329eeba45f ("ASoC: soc-component: add soc_component_err()") Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- changes in v2 - add missed return in snd_soc_pcm_component_sync_stop and - snd_soc_pcm_component_new sound/soc/soc-component.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-component.c b/sound/soc/soc-component.c index af9909c5492f..9565a0dd7cb6 100644 --- a/sound/soc/soc-component.c +++ b/sound/soc/soc-component.c @@ -647,7 +647,7 @@ int snd_soc_pcm_component_sync_stop(struct snd_pcm_substream *substream) ret = component->driver->sync_stop(component, substream); if (ret < 0) - soc_component_ret(component, ret); + return soc_component_ret(component, ret); } } @@ -705,7 +705,7 @@ int snd_soc_pcm_component_mmap(struct snd_pcm_substream *substream, /* FIXME. it returns 1st mmap now */ for_each_rtd_components(rtd, i, component) if (component->driver->mmap) - soc_component_ret( + return soc_component_ret( component, component->driver->mmap(component, substream, vma)); @@ -723,7 +723,7 @@ int snd_soc_pcm_component_new(struct snd_soc_pcm_runtime *rtd) if (component->driver->pcm_construct) { ret = component->driver->pcm_construct(component, rtd); if (ret < 0) - soc_component_ret(component, ret); + return soc_component_ret(component, ret); } } -- 2.27.0

3 2

[PATCH v3] ASoC: dt-bindings: renesas, fsi: use patternProperties for FSI-A/B
by Kuninori Morimoto 17 Jul '20

17 Jul '20

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> FSI has FSI-A and FSI-B, and has fsia-xxx/fsib-xxx properties. This patch uses patternProperties, and reduce verbose settings. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Reviewed-by: Rob Herring <robh(a)kernel.org> Link: https://lore.kernel.org/r/87o8osxlow.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87pn9v39eo.wl-kuninori.morimoto.gx@renesas.com --- v2 -> v3 - add Reviewd-by - add Link .../bindings/sound/renesas,fsi.yaml | 19 ++++--------------- 1 file changed, 4 insertions(+), 15 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/renesas,fsi.yaml b/Documentation/devicetree/bindings/sound/renesas,fsi.yaml index 8a4406be387a..0dd3f7361399 100644 --- a/Documentation/devicetree/bindings/sound/renesas,fsi.yaml +++ b/Documentation/devicetree/bindings/sound/renesas,fsi.yaml @@ -43,30 +43,19 @@ properties: '#sound-dai-cells': const: 1 - fsia,spdif-connection: +patternProperties: + "^fsi(a|b),spdif-connection$": $ref: /schemas/types.yaml#/definitions/flag description: FSI is connected by S/PDIF - fsia,stream-mode-support: + "^fsi(a|b),stream-mode-support$": $ref: /schemas/types.yaml#/definitions/flag description: FSI supports 16bit stream mode - fsia,use-internal-clock: + "^fsi(a|b),use-internal-clock$": $ref: /schemas/types.yaml#/definitions/flag description: FSI uses internal clock when master mode - fsib,spdif-connection: - $ref: /schemas/types.yaml#/definitions/flag - description: same as fsia - - fsib,stream-mode-support: - $ref: /schemas/types.yaml#/definitions/flag - description: same as fsia - - fsib,use-internal-clock: - $ref: /schemas/types.yaml#/definitions/flag - description: same as fsia - required: - compatible - reg -- 2.25.1

2 2

[PATCH] ASoC: rt5682: Report the button event in the headset type only
by Oder Chiou 17 Jul '20

17 Jul '20

The irq work will be manipulated by resume function, and it will report the wrong jack type while the jack type is headphone in the button event. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5682.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index b634e582ef7e..d7302f968a75 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -1082,7 +1082,8 @@ void rt5682_jack_detect_handler(struct work_struct *work) /* jack was out, report jack type */ rt5682->jack_type = rt5682_headset_detect(rt5682->component, 1); - } else { + } else if ((rt5682->jack_type & SND_JACK_HEADSET) == + SND_JACK_HEADSET) { /* jack is already in, report button event */ rt5682->jack_type = SND_JACK_HEADSET; btn_type = rt5682_button_detect(rt5682->component); -- 2.27.0

2 1

[PATCH][resend] ASoC: dt-bindings: renesas, fsi: use patternProperties for FSI-A/B
by Kuninori Morimoto 17 Jul '20

17 Jul '20

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> FSI has FSI-A and FSI-B, and has fsia-xxx/fsib-xxx properties. This patch uses patternProperties, and reduce verbose settings. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- - 2weeks passed, but nothing happen .../bindings/sound/renesas,fsi.yaml | 19 ++++--------------- 1 file changed, 4 insertions(+), 15 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/renesas,fsi.yaml b/Documentation/devicetree/bindings/sound/renesas,fsi.yaml index 8a4406be387a..0dd3f7361399 100644 --- a/Documentation/devicetree/bindings/sound/renesas,fsi.yaml +++ b/Documentation/devicetree/bindings/sound/renesas,fsi.yaml @@ -43,30 +43,19 @@ properties: '#sound-dai-cells': const: 1 - fsia,spdif-connection: +patternProperties: + "^fsi(a|b),spdif-connection$": $ref: /schemas/types.yaml#/definitions/flag description: FSI is connected by S/PDIF - fsia,stream-mode-support: + "^fsi(a|b),stream-mode-support$": $ref: /schemas/types.yaml#/definitions/flag description: FSI supports 16bit stream mode - fsia,use-internal-clock: + "^fsi(a|b),use-internal-clock$": $ref: /schemas/types.yaml#/definitions/flag description: FSI uses internal clock when master mode - fsib,spdif-connection: - $ref: /schemas/types.yaml#/definitions/flag - description: same as fsia - - fsib,stream-mode-support: - $ref: /schemas/types.yaml#/definitions/flag - description: same as fsia - - fsib,use-internal-clock: - $ref: /schemas/types.yaml#/definitions/flag - description: same as fsia - required: - compatible - reg -- 2.25.1

3 2

[PATCH 0/2] ALSA: hda: export snd_hda_codec_cleanup()
by Kai Vehmanen 17 Jul '20

17 Jul '20

Hi, this small series is preparation for a set of bugfix ASoC patches addressing a memleak at module unload for the HDA codec wrapper. Instead of duplicating HDA code in ASoC tree, I chose to export more functionality from hda_codec.c so it can be (re)used in ASoC's hdac_hda.c. Full series: https://github.com/thesofproject/linux/pull/2252 Takashi and Mark, feedback is welcome on how to best handle this kind of series where I have dependent patches both in sound/pci/hda and in ASoC. For this series, I'm sending the patches separately and when/if first set is merged by Takashi, I'll route the ASoC patches via our usually SOF set to Mark. Kai Vehmanen (2): ALSA: hda: export snd_hda_codec_cleanup_for_unbind() ALSA: hda: fix snd_hda_codec_cleanup() documentation include/sound/hda_codec.h | 2 ++ sound/pci/hda/hda_codec.c | 3 ++- 2 files changed, 4 insertions(+), 1 deletion(-) -- 2.27.0

3 5

[PATCH v2 1/1] ASoC: fsl: fsl-asoc-card: Trivial: Fix misspelling of 'exists'
by Lee Jones 17 Jul '20

17 Jul '20

Cc: Timur Tabi <timur(a)kernel.org> Cc: Nicolin Chen <nicoleotsuka(a)gmail.com> Cc: Xiubo Li <Xiubo.Lee(a)gmail.com> Cc: Fabio Estevam <festevam(a)gmail.com> Cc: linuxppc-dev(a)lists.ozlabs.org Signed-off-by: Lee Jones <lee.jones(a)linaro.org> --- sound/soc/fsl/fsl-asoc-card.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index faac6ce9a82cb..dbacdd25dfe76 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -92,7 +92,7 @@ struct fsl_asoc_card_priv { }; /* - * This dapm route map exits for DPCM link only. + * This dapm route map exists for DPCM link only. * The other routes shall go through Device Tree. * * Note: keep all ASRC routes in the second half -- 2.25.1

3 2

[PATCH] ASoC: soc-jack: calling snd_soc_jack_report causes a null pointer access
by mnlife qiao 17 Jul '20

17 Jul '20

When snd_soc_card_jack_new is not called or the call fails, calling this function causes a null pointer access Signed-off-by: mnlife <mnlife(a)foxmail.com> --- sound/soc/soc-jack.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index 0f1820f36b4d..51b799ee98b7 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -44,7 +44,7 @@ void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask) unsigned int sync = 0; int enable; - if (!jack) + if (!jack || !jack->jack) return; trace_snd_soc_jack_report(jack, mask, status); -- 2.17.1

2 1

[PATCH v6] ASoC: Intel: kbl_rt5663_rt5514_max98927: Fix kabylake_ssp_fixup function
by Harsha Priya 16 Jul '20

16 Jul '20

From: Vamshi Krishna Gopal <vamshi.krishna.gopal(a)intel.com> kabylake_ssp_fixup function uses snd_soc_dpcm to identify the codecs DAIs. The hw parameters are changed based on the codec DAI, the stream is intended for. The earlier approach to get snd_soc_dpcm was using container_of() macro on snd_pcm_hw_params. The structures have been modified over time and snd_soc_dpcm does not have snd_pcm_hw_params as a reference but as a copy. This causes the current driver to crash when used. This patch changes the way snd_soc_dpcm is extracted. The snd_soc_pcm_runtime holds 2 dpcm instances (one for playback and one for capture). The 2 codecs on this SSP are dmic and speakers. One is for capture and one is for playback respectively. Based on the direction of the stream, the snd_soc_dpcm is extracted from the snd_soc_pcm_runtime structure. Tested for all use cases of the driver. Signed-off-by: Harsha Priya <harshapriya.n(a)intel.com> Signed-off-by: Vamshi Krishna Gopal <vamshi.krishna.gopal(a)intel.com> Tested-by: Lukasz Majczak <lma(a)semihalf.com> --- v1 -> v2: - Extract dmic from SSP0 as every BE should have own fixup function. v2 -> v3: - Restore naming in the dapm route table to not confuse with other drivers - Fixed indentations v3 -> v4: - Updated code and commit description according to solution proposed by Harsha v4 -> v5: - Cosmetic Changes v5 -> v6: - Dmic regression seen with v4 fixed - Using available routines for obtaining dpcm information --- --- .../intel/boards/kbl_rt5663_rt5514_max98927.c | 26 ++++++++++++------- 1 file changed, 17 insertions(+), 9 deletions(-) diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index 584e4f9cedc2..b261b1c466a8 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -379,22 +379,30 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_interval *chan = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); - struct snd_soc_dpcm *dpcm = container_of( - params, struct snd_soc_dpcm, hw_params); - struct snd_soc_dai_link *fe_dai_link = dpcm->fe->dai_link; - struct snd_soc_dai_link *be_dai_link = dpcm->be->dai_link; + struct snd_soc_dpcm *dpcm, *rtd_dpcm; + + /* + * This macro will be called for playback stream + */ + for_each_dpcm_fe(rtd, SNDRV_PCM_STREAM_PLAYBACK, dpcm) + rtd_dpcm = dpcm; + /* + * This macro will be called for capture stream + */ + for_each_dpcm_fe(rtd, SNDRV_PCM_STREAM_CAPTURE, dpcm) + rtd_dpcm = dpcm; /* * The ADSP will convert the FE rate to 48k, stereo, 24 bit */ - if (!strcmp(fe_dai_link->name, "Kbl Audio Port") || - !strcmp(fe_dai_link->name, "Kbl Audio Headset Playback") || - !strcmp(fe_dai_link->name, "Kbl Audio Capture Port")) { + if (!strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio Port") || + !strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio Headset Playback") || + !strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio Capture Port")) { rate->min = rate->max = 48000; chan->min = chan->max = 2; snd_mask_none(fmt); snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE); - } else if (!strcmp(fe_dai_link->name, "Kbl Audio DMIC cap")) { + } else if (!strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio DMIC cap")) { if (params_channels(params) == 2 || DMIC_CH(dmic_constraints) == 2) chan->min = chan->max = 2; @@ -405,7 +413,7 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, * The speaker on the SSP0 supports S16_LE and not S24_LE. * thus changing the mask here */ - if (!strcmp(be_dai_link->name, "SSP0-Codec")) + if (!strcmp(rtd_dpcm->be->dai_link->name, "SSP0-Codec")) snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S16_LE); return 0; -- 2.17.1

3 2

[PATCH v3] ASoC: Intel: kbl_rt5663_rt5514_max98927: Split be_hw_params_fixup function
by Lukasz Majczak 16 Jul '20

16 Jul '20

Split be_hw_params_fixup function for different codecs as current common function, leads to crash while trying to get snd_soc_dpcm with container_of() macro in kabylake_ssp_fixup(). The crash call path looks as below: soc_pcm_hw_params() snd_soc_dai_hw_params(codec_dai, substream, &codec_params); rtd->dai_link->be_hw_params_fixup(rtd, params) kabylake_ssp_fixup() In this case, codec_params is just a copy of an internal structure and is not embedded into struct snd_soc_dpcm thus we cannot use container_of() on it. v1 -> v2: - Extract dmic from SSP0 as every BE should have own fixup function. v2 -> v3: - Restore naming in the dapm route table to not confuse with other drivers - Fixed indentations Signed-off-by: Lukasz Majczak <lma(a)semihalf.com> --- .../intel/boards/kbl_rt5663_rt5514_max98927.c | 130 +++++++++++------- 1 file changed, 84 insertions(+), 46 deletions(-) diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index 1b1f8d7a4ea3..b9596cf3a6c4 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -328,42 +328,51 @@ static const struct snd_soc_ops kabylake_rt5663_fe_ops = { .startup = kbl_fe_startup, }; -static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, - struct snd_pcm_hw_params *params) +static void kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params, snd_pcm_format_t pcm_fmt) { struct snd_interval *rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); - struct snd_interval *chan = hw_param_interval(params, + struct snd_interval *channels = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); - struct snd_soc_dpcm *dpcm = container_of( - params, struct snd_soc_dpcm, hw_params); - struct snd_soc_dai_link *fe_dai_link = dpcm->fe->dai_link; - struct snd_soc_dai_link *be_dai_link = dpcm->be->dai_link; /* * The ADSP will convert the FE rate to 48k, stereo, 24 bit */ - if (!strcmp(fe_dai_link->name, "Kbl Audio Port") || - !strcmp(fe_dai_link->name, "Kbl Audio Headset Playback") || - !strcmp(fe_dai_link->name, "Kbl Audio Capture Port")) { - rate->min = rate->max = 48000; - chan->min = chan->max = 2; - snd_mask_none(fmt); - snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE); - } else if (!strcmp(fe_dai_link->name, "Kbl Audio DMIC cap")) { - if (params_channels(params) == 2 || - DMIC_CH(dmic_constraints) == 2) - chan->min = chan->max = 2; - else - chan->min = chan->max = 4; - } - /* - * The speaker on the SSP0 supports S16_LE and not S24_LE. - * thus changing the mask here - */ - if (!strcmp(be_dai_link->name, "SSP0-Codec")) - snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S16_LE); + + rate->min = rate->max = 48000; + channels->min = channels->max = 2; + + snd_mask_none(fmt); + snd_mask_set_format(fmt, pcm_fmt); +} + +static int kabylake_ssp0_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + kabylake_ssp_fixup(rtd, params, SNDRV_PCM_FORMAT_S16_LE); + return 0; +} + +static int kabylake_ssp1_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + + kabylake_ssp_fixup(rtd, params, SNDRV_PCM_FORMAT_S24_LE); + return 0; +} + +static int kabylake_dmic_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + + if (params_channels(params) == 2 || DMIC_CH(dmic_constraints) == 2) + channels->min = channels->max = 2; + else + channels->min = channels->max = 4; return 0; } @@ -400,20 +409,6 @@ static int kabylake_ssp0_hw_params(struct snd_pcm_substream *substream, int ret = 0, j; for_each_rtd_codec_dais(rtd, j, codec_dai) { - if (!strcmp(codec_dai->component->name, RT5514_DEV_NAME)) { - ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xF, 0, 8, 16); - if (ret < 0) { - dev_err(rtd->dev, "set TDM slot err:%d\n", ret); - return ret; - } - - ret = snd_soc_dai_set_sysclk(codec_dai, - RT5514_SCLK_S_MCLK, 24576000, SND_SOC_CLOCK_IN); - if (ret < 0) { - dev_err(rtd->dev, "set sysclk err: %d\n", ret); - return ret; - } - } if (!strcmp(codec_dai->component->name, MAXIM_DEV0_NAME)) { ret = snd_soc_dai_set_tdm_slot(codec_dai, 0x30, 3, 8, 16); if (ret < 0) { @@ -433,10 +428,36 @@ static int kabylake_ssp0_hw_params(struct snd_pcm_substream *substream, return ret; } +static int kabylake_dmic01_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + int ret = 0; + + ret = snd_soc_dai_set_tdm_slot(rtd->codec_dai, 0xF, 0, 8, 16); + if (ret < 0) { + dev_err(rtd->dev, "set TDM slot err:%d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(rtd->codec_dai, + RT5514_SCLK_S_MCLK, 24576000, SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(rtd->dev, "set sysclk err: %d\n", ret); + return ret; + } + + return ret; +} + static struct snd_soc_ops kabylake_ssp0_ops = { .hw_params = kabylake_ssp0_hw_params, }; +static struct snd_soc_ops kabylake_dmic01_ops = { + .hw_params = kabylake_dmic01_hw_params, +}; + static const unsigned int channels_dmic[] = { 4, }; @@ -507,14 +528,19 @@ SND_SOC_DAILINK_DEF(ssp0_pin, SND_SOC_DAILINK_DEF(ssp0_codec, DAILINK_COMP_ARRAY( /* Left */ COMP_CODEC(MAXIM_DEV0_NAME, KBL_MAXIM_CODEC_DAI), - /* Right */COMP_CODEC(MAXIM_DEV1_NAME, KBL_MAXIM_CODEC_DAI), - /* dmic */ COMP_CODEC(RT5514_DEV_NAME, KBL_REALTEK_DMIC_CODEC_DAI))); + /* Right */COMP_CODEC(MAXIM_DEV1_NAME, KBL_MAXIM_CODEC_DAI))); SND_SOC_DAILINK_DEF(ssp1_pin, DAILINK_COMP_ARRAY(COMP_CPU("SSP1 Pin"))); SND_SOC_DAILINK_DEF(ssp1_codec, DAILINK_COMP_ARRAY(COMP_CODEC(RT5663_DEV_NAME, KBL_REALTEK_CODEC_DAI))); +SND_SOC_DAILINK_DEF(dmic01_pin, + DAILINK_COMP_ARRAY(COMP_CPU("DMIC01 Pin"))); +SND_SOC_DAILINK_DEF(dmic01_codec, + DAILINK_COMP_ARRAY( + COMP_CODEC(RT5514_DEV_NAME, KBL_REALTEK_DMIC_CODEC_DAI))); + SND_SOC_DAILINK_DEF(idisp1_pin, DAILINK_COMP_ARRAY(COMP_CPU("iDisp1 Pin"))); SND_SOC_DAILINK_DEF(idisp1_codec, @@ -618,9 +644,8 @@ static struct snd_soc_dai_link kabylake_dais[] = { SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS, .ignore_pmdown_time = 1, - .be_hw_params_fixup = kabylake_ssp_fixup, + .be_hw_params_fixup = kabylake_ssp0_fixup, .dpcm_playback = 1, - .dpcm_capture = 1, .ops = &kabylake_ssp0_ops, SND_SOC_DAILINK_REG(ssp0_pin, ssp0_codec, platform), }, @@ -632,12 +657,25 @@ static struct snd_soc_dai_link kabylake_dais[] = { .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS, .ignore_pmdown_time = 1, - .be_hw_params_fixup = kabylake_ssp_fixup, + .be_hw_params_fixup = kabylake_ssp1_fixup, .ops = &kabylake_rt5663_ops, .dpcm_playback = 1, .dpcm_capture = 1, SND_SOC_DAILINK_REG(ssp1_pin, ssp1_codec, platform), }, + { + .name = "dmic01", + .id = 2, + .no_pcm = 1, + .dai_fmt = SND_SOC_DAIFMT_DSP_B | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + .ignore_pmdown_time = 1, + .be_hw_params_fixup = kabylake_dmic_fixup, + .dpcm_capture = 1, + .ops = &kabylake_dmic01_ops, + SND_SOC_DAILINK_REG(dmic01_pin, dmic01_codec, platform), + }, { .name = "iDisp1", .id = 3, base-commit: a4f6fc98cd2fa1774bcaeb248c67156ef9402a56 -- 2.25.1

4 19

[PATCH] topology: hda-dsp: Change processing domain of srcintc module to LL
by Mateusz Gorski 16 Jul '20

16 Jul '20

In order to make this topology universal for all FW branches, change the processing domain of srcintc module from DP to LL. Signed-off-by: Mateusz Gorski <mateusz.gorski(a)linux.intel.com> --- topology/hda-dsp/skl_hda_dsp_generic-tplg.conf | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/topology/hda-dsp/skl_hda_dsp_generic-tplg.conf b/topology/hda-dsp/skl_hda_dsp_generic-tplg.conf index 32ba9a7..ed2ea80 100644 --- a/topology/hda-dsp/skl_hda_dsp_generic-tplg.conf +++ b/topology/hda-dsp/skl_hda_dsp_generic-tplg.conf @@ -1684,7 +1684,7 @@ SectionVendorTuples."media0_in srcintc 0" { SKL_TKN_U32_PIPE_PRIORITY "0" SKL_TKN_U32_PMODE "0" SKL_TKN_U32_D0I3_CAPS "1" - SKL_TKN_U32_PROC_DOMAIN "1" + SKL_TKN_U32_PROC_DOMAIN "0" SKL_TKN_U32_PIPE_DIRECTION "0" SKL_TKN_U32_NUM_CONFIGS "1" SKL_TKN_U32_DMA_BUF_SIZE "2" -- 2.17.1

1 0

[PATCH v2 0/5] crypto: add sha256() function
by Eric Biggers 16 Jul '20

16 Jul '20

This series adds a function sha256() to the sha256 library so that users who want to compute a hash in one step can just call sha256() instead of sha256_init() + sha256_update() + sha256_final(). Patches 3-5 then convert some users to use it. Changed v1 => v2: - Added sparc patch to fix a build breakage caused by a static variable already named "sha256". - Added Reviewed-by, Acked-by, and Tested-by tags. Eric Biggers (5): crypto: sparc - rename sha256 to sha256_alg crypto: lib/sha256 - add sha256() function efi: use sha256() instead of open coding mptcp: use sha256() instead of open coding ASoC: cros_ec_codec: use sha256() instead of open coding arch/sparc/crypto/sha256_glue.c | 14 ++++++------ drivers/firmware/efi/embedded-firmware.c | 9 +++----- include/crypto/sha.h | 1 + lib/crypto/sha256.c | 10 +++++++++ net/mptcp/crypto.c | 15 +++---------- sound/soc/codecs/cros_ec_codec.c | 27 ++---------------------- 6 files changed, 26 insertions(+), 50 deletions(-) base-commit: 57c8aa43b9f272c382c253573c82be5cb68fe22d -- 2.27.0

2 3

[PATCH] ALSA: hda/realtek - fixup for yet another Intel reference board
by Takashi Iwai 16 Jul '20

16 Jul '20

From: PeiSen Hou <pshou(a)realtek.com.tw> Add headset_jack for the intel reference board support with 10ec:1230. Signed-off-by: PeiSen Hou <pshou(a)realtek.com.tw> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 45df33d1bd1a..1b06c4261248 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7581,6 +7581,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x10cf, 0x1629, "Lifebook U7x7", ALC255_FIXUP_LIFEBOOK_U7x7_HEADSET_MIC), SND_PCI_QUIRK(0x10cf, 0x1845, "Lifebook U904", ALC269_FIXUP_LIFEBOOK_EXTMIC), SND_PCI_QUIRK(0x10ec, 0x10f2, "Intel Reference board", ALC700_FIXUP_INTEL_REFERENCE), + SND_PCI_QUIRK(0x10ec, 0x1230, "Intel Reference board", ALC225_FIXUP_HEADSET_JACK), SND_PCI_QUIRK(0x10f7, 0x8338, "Panasonic CF-SZ6", ALC269_FIXUP_HEADSET_MODE), SND_PCI_QUIRK(0x144d, 0xc109, "Samsung Ativ book 9 (NP900X3G)", ALC269_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x144d, 0xc169, "Samsung Notebook 9 Pen (NP930SBE-K01US)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), -- 2.16.4

1 0

[PATCH] sound/pci/bt87x: Replaces 'blacklist' with 'no_audio_list'
by Paul Schulz 16 Jul '20

16 Jul '20

This patch removes 'blacklist' and replaces it with the more descriptive 'no_audio_list' This is a functionally trivial patch and has no other effect. Signed-off-by: Paul Schulz <paul(a)mawsonlakes.org> --- sound/pci/bt87x.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/pci/bt87x.c b/sound/pci/bt87x.c index 6567504665b9..b74b55bdb146 100644 --- a/sound/pci/bt87x.c +++ b/sound/pci/bt87x.c @@ -801,7 +801,7 @@ MODULE_DEVICE_TABLE(pci, snd_bt87x_ids); * (DVB cards use the audio function to transfer MPEG data) */ static struct { unsigned short subvendor, subdevice; -} blacklist[] = { +} no_audio_list[] = { {0x0071, 0x0101}, /* Nebula Electronics DigiTV */ {0x11bd, 0x001c}, /* Pinnacle PCTV Sat */ {0x11bd, 0x0026}, /* Pinnacle PCTV SAT CI */ @@ -817,7 +817,7 @@ static struct { static struct pci_driver driver; -/* return the id of the card, or a negative value if it's blacklisted */ +/* return the id of the card, or a negative value if it's in the no_audio_list */ static int snd_bt87x_detect_card(struct pci_dev *pci) { int i; @@ -827,9 +827,9 @@ static int snd_bt87x_detect_card(struct pci_dev *pci) if (supported && supported->driver_data > 0) return supported->driver_data; - for (i = 0; i < ARRAY_SIZE(blacklist); ++i) - if (blacklist[i].subvendor == pci->subsystem_vendor && - blacklist[i].subdevice == pci->subsystem_device) { + for (i = 0; i < ARRAY_SIZE(no_audio_list); ++i) + if (no_audio_list[i].subvendor == pci->subsystem_vendor && + no_audio_list[i].subdevice == pci->subsystem_device) { dev_dbg(&pci->dev, "card %#04x-%#04x:%#04x has no audio\n", pci->device, pci->subsystem_vendor, pci->subsystem_device); -- 2.25.1

2 1

ASUS UX563 speaker no ouput
by Kailang 16 Jul '20

16 Jul '20

Hi Takashi, ASUS UX563 speaker also no sound. Add quirk to model ALC294_FIXUP_ASUS_HPE will solve it. BR, Kailang > -----Original Message----- > From: Takashi Iwai <tiwai(a)suse.de> > Sent: Thursday, July 16, 2020 3:19 PM > To: Kailang <kailang(a)realtek.com> > Cc: (alsa-devel(a)alsa-project.org) <alsa-devel(a)alsa-project.org> > Subject: Re: ASUS UX533 and UX534 speaker no ouput > > On Thu, 16 Jul 2020 09:12:26 +0200, > Kailang wrote: > > > > Hi Takashi, > > > > Attach patch will solve speaker no output issue for ASUS platform. > > Many Thanks. > > Pulled now. Thanks. > > > Takashi > > ------Please consider the environment before printing this e-mail.

2 1

ASUS UX533 and UX534 speaker no ouput
by Kailang 16 Jul '20

16 Jul '20

Hi Takashi, Attach patch will solve speaker no output issue for ASUS platform. Many Thanks. BR, Kailang

2 1

[PATCH] ASoC: soc-component: Add missed return for snd_soc_pcm_component_mmap
by Shengjiu Wang 16 Jul '20

16 Jul '20

Add missed return for snd_soc_pcm_component_mmap, otherwise it always return -EINVAL. Fixes: e2329eeba45f ("ASoC: soc-component: add soc_component_err()") Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/soc-component.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-component.c b/sound/soc/soc-component.c index af9909c5492f..cde7b9c609bb 100644 --- a/sound/soc/soc-component.c +++ b/sound/soc/soc-component.c @@ -705,7 +705,7 @@ int snd_soc_pcm_component_mmap(struct snd_pcm_substream *substream, /* FIXME. it returns 1st mmap now */ for_each_rtd_components(rtd, i, component) if (component->driver->mmap) - soc_component_ret( + return soc_component_ret( component, component->driver->mmap(component, substream, vma)); -- 2.27.0

3 2

[PATCH] soundwire: fix the kernel-doc comment
by Vinod Koul 16 Jul '20

16 Jul '20

sdw_startup_stream() and sdw_shutdown_stream() argument has been updated but not the comments, so update these as well to fix warning with W=1 drivers/soundwire/stream.c:1859: warning: Function parameter or member 'sdw_substream' not described in 'sdw_startup_stream' drivers/soundwire/stream.c:1859: warning: Excess function parameter 'stream' description in 'sdw_startup_stream' drivers/soundwire/stream.c:1903: warning: Function parameter or member 'sdw_substream' not described in 'sdw_shutdown_stream' drivers/soundwire/stream.c:1903: warning: Excess function parameter 'stream' description in 'sdw_shutdown_stream' Signed-off-by: Vinod Koul <vkoul(a)kernel.org> --- drivers/soundwire/stream.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/drivers/soundwire/stream.c b/drivers/soundwire/stream.c index 6bc2ff29c202..37290a799023 100644 --- a/drivers/soundwire/stream.c +++ b/drivers/soundwire/stream.c @@ -1851,7 +1851,7 @@ static int set_stream(struct snd_pcm_substream *substream, /** * sdw_startup_stream() - Startup SoundWire stream * - * @stream: Soundwire stream + * @sdw_substream: Soundwire stream * * Documentation/driver-api/soundwire/stream.rst explains this API in detail */ @@ -1895,7 +1895,7 @@ EXPORT_SYMBOL(sdw_startup_stream); /** * sdw_shutdown_stream() - Shutdown SoundWire stream * - * @stream: Soundwire stream + * @sdw_substream: Soundwire stream * * Documentation/driver-api/soundwire/stream.rst explains this API in detail */ -- 2.26.2

2 2

[PATCH][resend] ASoC: dt-bindings: ak4613: switch to yaml base Documentation
by Kuninori Morimoto 16 Jul '20

16 Jul '20

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> This patch switches from .txt base to .yaml base Document. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- - 2weeks passed, but nothing happen .../devicetree/bindings/sound/ak4613.txt | 27 ---------- .../devicetree/bindings/sound/ak4613.yaml | 49 +++++++++++++++++++ 2 files changed, 49 insertions(+), 27 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/ak4613.txt create mode 100644 Documentation/devicetree/bindings/sound/ak4613.yaml diff --git a/Documentation/devicetree/bindings/sound/ak4613.txt b/Documentation/devicetree/bindings/sound/ak4613.txt deleted file mode 100644 index 49a2e74fd9cb..000000000000 --- a/Documentation/devicetree/bindings/sound/ak4613.txt +++ /dev/null @@ -1,27 +0,0 @@ -AK4613 I2C transmitter - -This device supports I2C mode only. - -Required properties: - -- compatible : "asahi-kasei,ak4613" -- reg : The chip select number on the I2C bus - -Optional properties: -- asahi-kasei,in1-single-end : Boolean. Indicate input / output pins are single-ended. -- asahi-kasei,in2-single-end rather than differential. -- asahi-kasei,out1-single-end -- asahi-kasei,out2-single-end -- asahi-kasei,out3-single-end -- asahi-kasei,out4-single-end -- asahi-kasei,out5-single-end -- asahi-kasei,out6-single-end - -Example: - -&i2c { - ak4613: ak4613@10 { - compatible = "asahi-kasei,ak4613"; - reg = <0x10>; - }; -}; diff --git a/Documentation/devicetree/bindings/sound/ak4613.yaml b/Documentation/devicetree/bindings/sound/ak4613.yaml new file mode 100644 index 000000000000..eb915377bc5d --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ak4613.yaml @@ -0,0 +1,49 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ak4613.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: AK4613 I2C transmitter Device Tree Bindings + +maintainers: + - Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> + +properties: + compatible: + const: asahi-kasei,ak4613 + + reg: + maxItems: 1 + + clocks: + maxItems: 1 + + "#sound-dai-cells": + const: 0 + +patternProperties: + "^asahi-kasei,in[1-2]-single-end$": + description: Input Pin 1 - 2. + $ref: /schemas/types.yaml#/definitions/flag + + "^asahi-kasei,out[1-6]-single-end$": + description: Output Pin 1 - 6. + $ref: /schemas/types.yaml#/definitions/flag + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + ak4613: ak4613@10 { + compatible = "asahi-kasei,ak4613"; + reg = <0x10>; + }; + }; -- 2.25.1

3 3

[PATCH] ASoC: cros_ec_codec: Reset I2S RX when probing
by Yu-Hsuan Hsu 15 Jul '20

15 Jul '20

It is not guaranteed that I2S RX is disabled when the kernel booting. For example, if the kernel crashes while it is enabled, it will keep enabled until the next time EC reboots. Reset I2S RX when probing to fix this issue. Signed-off-by: Yu-Hsuan Hsu <yuhsuan(a)chromium.org> --- include/linux/platform_data/cros_ec_commands.h | 1 + sound/soc/codecs/cros_ec_codec.c | 7 +++++++ 2 files changed, 8 insertions(+) diff --git a/include/linux/platform_data/cros_ec_commands.h b/include/linux/platform_data/cros_ec_commands.h index 69210881ebac8..11ce917ca924c 100644 --- a/include/linux/platform_data/cros_ec_commands.h +++ b/include/linux/platform_data/cros_ec_commands.h @@ -4598,6 +4598,7 @@ enum ec_codec_i2s_rx_subcmd { EC_CODEC_I2S_RX_SET_SAMPLE_DEPTH = 0x2, EC_CODEC_I2S_RX_SET_DAIFMT = 0x3, EC_CODEC_I2S_RX_SET_BCLK = 0x4, + EC_CODEC_I2S_RX_RESET = 0x5, EC_CODEC_I2S_RX_SUBCMD_COUNT, }; diff --git a/sound/soc/codecs/cros_ec_codec.c b/sound/soc/codecs/cros_ec_codec.c index 8d45c628e988e..5495214e73e68 100644 --- a/sound/soc/codecs/cros_ec_codec.c +++ b/sound/soc/codecs/cros_ec_codec.c @@ -1034,6 +1034,13 @@ static int cros_ec_codec_platform_probe(struct platform_device *pdev) } priv->ec_capabilities = r.capabilities; + /* Reset EC codec I2S RX. */ + p.cmd = EC_CODEC_I2S_RX_RESET; + ret = send_ec_host_command(priv->ec_device, EC_CMD_EC_CODEC_I2S_RX, + (uint8_t *)&p, sizeof(p), NULL, 0); + if (ret) + dev_err(dev, "failed to EC_CODEC_I2S_RESET: %d\n", ret); + platform_set_drvdata(pdev, priv); ret = devm_snd_soc_register_component(dev, &i2s_rx_component_driver, -- 2.27.0.383.g050319c2ae-goog

3 6

[PATCH 00/10] ALSA: Replace words for inclusive terminalogy guideline
by Takashi Iwai 15 Jul '20

15 Jul '20

Hi, this is a patch set to replace some words and sentences for adapting the recent inclusive terminology guidelines. Takashi === Takashi Iwai (10): ALSA: Replace the word "slave" in vmaster API ALSA: ac97: Replace the words white/blacklist ALSA: bt87x: Replace the words white/blacklist ALSA: ctxfi: Replace the word blacklist ALSA: es1968: Replace the words white/blacklist ALSA: hda: Replace the words white/blacklist ALSA: intel8x0: Replace the words white/blacklist ALSA: nm256: Replace the words blacklist ALSA: via82xx: Replace the words whitelist ALSA: usb-audio: Replace blacklist word include/sound/control.h | 45 +++---- include/sound/hda_codec.h | 2 +- sound/core/vmaster.c | 261 ++++++++++++++++++++-------------------- sound/pci/ac97/ac97_patch.c | 34 +++--- sound/pci/bt87x.c | 14 +-- sound/pci/ca0106/ca0106_mixer.c | 18 +-- sound/pci/ctxfi/ctatc.c | 2 +- sound/pci/es1968.c | 20 +-- sound/pci/hda/hda_codec.c | 94 +++++++-------- sound/pci/hda/hda_generic.c | 20 +-- sound/pci/hda/hda_generic.h | 2 +- sound/pci/hda/hda_intel.c | 26 ++-- sound/pci/hda/hda_local.h | 10 +- sound/pci/hda/patch_ca0132.c | 10 +- sound/pci/hda/patch_realtek.c | 6 +- sound/pci/hda/patch_sigmatel.c | 2 +- sound/pci/ice1712/juli.c | 16 +-- sound/pci/ice1712/quartet.c | 12 +- sound/pci/intel8x0.c | 6 +- sound/pci/nm256/nm256.c | 8 +- sound/pci/via82xx.c | 6 +- sound/ppc/awacs.c | 12 +- sound/usb/6fire/control.c | 2 +- sound/usb/quirks-table.h | 2 +- 24 files changed, 316 insertions(+), 314 deletions(-) -- 2.16.4

3 16

[PATCH] sound/soc/soc-core: Removed instances of 'blacklist'
by Paul Schulz 15 Jul '20

15 Jul '20

This patch removes 'blacklist' and replaces it with 'invalid list' in sound/soc/soc-core.c This is a functionally trivial patch and has no other effect. Signed-off-by: Paul Schulz <paul(a)mawsonlakes.org> --- sound/soc/soc-core.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 0f30f5aabaa8..6bc56c4cb207 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1457,11 +1457,11 @@ EXPORT_SYMBOL_GPL(snd_soc_runtime_set_dai_fmt); #ifdef CONFIG_DMI /* - * If a DMI filed contain strings in this blacklist (e.g. + * If a DMI filed contain strings in this invalid list (e.g. * "Type2 - Board Manufacturer" or "Type1 - TBD by OEM"), it will be taken * as invalid and dropped when setting the card long name from DMI info. */ -static const char * const dmi_blacklist[] = { +static const char * const dmi_invalid_list[] = { "To be filled by OEM", "TBD by OEM", "Default String", @@ -1493,14 +1493,14 @@ static void cleanup_dmi_name(char *name) /* * Check if a DMI field is valid, i.e. not containing any string - * in the black list. + * in the invalid list. */ static int is_dmi_valid(const char *field) { int i = 0; - while (dmi_blacklist[i]) { - if (strstr(field, dmi_blacklist[i])) + while (dmi_invalid_list[i]) { + if (strstr(field, dmi_invalid_list[i])) return 0; i++; } -- 2.25.1

1 0

[PATCH v3 00/10] topology: decode: Various fixes
by Piotr Maziarz 15 Jul '20

15 Jul '20

This series fixes various problems with topology decoding mechanism. Some of the problems were critical like improper memory management or infinite loops that were causing undefined behaviour or program crashes, while other resulted in losing some data during conversion. Bugs found while testing with Intel SST topologies. Changelog: v2: -Divide into more patches, critical fixes are in separate patches now -More specific descriptions -Fix a typo UML to UCM -Add error reporting in topology: decode: fix channel map memory allocation -Remove goto again in topology: Make buffer for saving dynamic size for better readability v3: -No functional changes -Changed UCM to more descriptive standard ALSA configuration files -removed Gerrit's Change-Id -Added missing signed-offs Piotr Maziarz (10): topology: decode: Fix channel map memory allocation topology: decode: Fix infinite loop in decoding enum control topology: decode: Remove decoding values for enum control topology: decode: Add enum control texts as separate element topology: decode: Fix printing texts section topology: decode: Change declaration of enum decoding function topology: decode: Fix decoding PCM formats and rates topology: decode: Print sig_bits field in PCM capabilities section topology: decode: Add DAI name printing topology: Make buffer for saving dynamic size src/topology/ctl.c | 51 ++++++++++++++++++++++------------------------- src/topology/dapm.c | 3 +-- src/topology/pcm.c | 11 +++++++--- src/topology/save.c | 34 ++++++++++++++++++++++++++----- src/topology/text.c | 2 +- src/topology/tplg_local.h | 2 +- 6 files changed, 64 insertions(+), 39 deletions(-) -- 2.7.4

3 16

[PATCH 0/9] soundwire: intel: revisit SHIM programming
by Bard Liao 15 Jul '20

15 Jul '20

This series does some cleanup, revisits SHIM programming sequences, and merges Soundwire interrupt handlers/threads. Bard Liao (2): soundwire: intel/cadence: merge Soundwire interrupt handlers/threads Soundwire: intel_init: save Slave(s) _ADR info in sdw_intel_ctx Pierre-Louis Bossart (6): soundwire: intel: reuse code for wait loops to set/clear bits soundwire: intel: revisit SHIM programming sequences. soundwire: intel: introduce a helper to arm link synchronization soundwire: intel: introduce helper for link synchronization soundwire: intel_init: add implementation of sdw_intel_enable_irq() soundwire: intel_init: use EXPORT_SYMBOL_NS Rander Wang (1): soundwire: intel: add wake interrupt support drivers/soundwire/cadence_master.c | 18 +- drivers/soundwire/cadence_master.h | 4 + drivers/soundwire/intel.c | 389 ++++++++++++++++++++++------ drivers/soundwire/intel.h | 9 + drivers/soundwire/intel_init.c | 101 +++++++- include/linux/soundwire/sdw_intel.h | 2 + 6 files changed, 425 insertions(+), 98 deletions(-) -- 2.17.1

4 22

[DMIX Plugin] Using Different bindings for same Alsa card
by Vijay Palaniswamy (RBEI/ECF3) 15 Jul '20

15 Jul '20

Hi All, I have two virtual device pointing to same Alsa card ,with different bindings When audio played on both device [dmix_4_channels and dmix_2_channels] in parallel. Audio played on dmix_2_channels is getting corrupted. Validated with alsa versio : 1.2.3 Please let me know how to make it work with below configuration. Alsa configuration used for testing: ************************************************* pcm.dmix_4_channels { type dmix ipc_key 5678293 ipc_perm 0660 ipc_gid audio bindings [0 1 2 3] slave { pcm "hardware" channels 4 periods 4 period_time 40000 } } pcm.dmix_2_channels { type dmix ipc_key 5678293 ipc_perm 0660 ipc_gid audio bindings [0 1] slave { pcm "hardware" channels 2 periods 4 period_time 40000 } } pcm.hardware { type hw card 0 channels 4 rate 48000 format S16_LE } ************************************************* Best regards Palaniswamy Vijay

1 0

[PATCH 0/2] soundwire: sdw.h: fix PRBS/Static_1 swapped
by Bard Liao 15 Jul '20

15 Jul '20

Fix PRBS/Static_1 swapped definitions and indentation in sdw.h Pierre-Louis Bossart (2): soundwire: sdw.h: fix PRBS/Static_1 swapped definitions soundwire: sdw.h: fix indentation include/linux/soundwire/sdw.h | 20 ++++++++++---------- 1 file changed, 10 insertions(+), 10 deletions(-) -- 2.17.1

2 3

[PATCH 0/3] ASoC: fsl-asoc-card: Support hp and mic detection
by Shengjiu Wang 15 Jul '20

15 Jul '20

Support hp and mic detection. Add a parameter for asoc_simple_init_jack. Shengjiu Wang (3): ASoC: simple-card-utils: Support configure pin_name for asoc_simple_init_jack ASoC: bindings: fsl-asoc-card: Support hp-det-gpio and mic-det-gpio ASoC: fsl-asoc-card: Support Headphone and Microphone Jack detection .../bindings/sound/fsl-asoc-card.txt | 3 + include/sound/simple_card_utils.h | 6 +- sound/soc/fsl/Kconfig | 1 + sound/soc/fsl/fsl-asoc-card.c | 69 ++++++++++++++++++- sound/soc/generic/simple-card-utils.c | 7 +- 5 files changed, 78 insertions(+), 8 deletions(-) -- 2.27.0

3 6

[PATCH v2 0/5] soundwire: handle stream at the dailink level
by Bard Liao 15 Jul '20

15 Jul '20

Currently, stream is handled at the dai level. But we have to handle stream at the dailink level in the multi-cpu dailink usage. changes in v2: - Add kernel doc - Use single line trace log Pierre-Louis Bossart (5): soundwire: intel: implement get_sdw_stream() operations soundwire: stream: add helper to startup/shutdown streams soundwire: intel: remove stream allocation/free soundwire: cadence: allocate/free dma_data in set_sdw_stream soundwire: intel: don't free dma_data in DAI shutdown Documentation/driver-api/soundwire/stream.rst | 11 ++- drivers/soundwire/cadence_master.c | 52 +++++++--- drivers/soundwire/intel.c | 82 ++++------------ drivers/soundwire/stream.c | 98 +++++++++++++++++++ include/linux/soundwire/sdw.h | 2 + 5 files changed, 167 insertions(+), 78 deletions(-) -- 2.17.1

2 6

[PATCH 1/1] MAINTAINERS: change SoundWire maintainer
by Sanyog Kale 15 Jul '20

15 Jul '20

Add Bard as SoundWire maintainer from Intel and change Sanyog's role as reviewer. Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Signed-off-by: Sanyog Kale <sanyog.r.kale(a)intel.com> --- MAINTAINERS | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/MAINTAINERS b/MAINTAINERS index 1c1527a34d99..9559e947bf67 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -16012,8 +16012,9 @@ F: sound/soc/sof/ SOUNDWIRE SUBSYSTEM M: Vinod Koul <vkoul(a)kernel.org> -M: Sanyog Kale <sanyog.r.kale(a)intel.com> +M: Bard Liao <yung-chuan.liao(a)linux.intel.com> R: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> +R: Sanyog Kale <sanyog.r.kale(a)intel.com> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) S: Supported F: Documentation/driver-api/soundwire/ -- 2.17.1

3 4

[RFT] ALSA control service programs for Digidesign Digi 002/003 family and Tascam FireWire series
by Takashi Sakamoto 15 Jul '20

15 Jul '20

Hi, This is request-for-test (RFT) to the ALSA control service programs for Digidesign Digi 002/003 family and Tascam FireWire series. The programs are available in 'topic/services-for-dg00x-tascam' of below repository, and the RFT is corresponding to Pull Request #1: * https://github.com/alsa-project/snd-firewire-ctl-services/ Background ---------- The models in above family and series are already supported by drivers in ALSA firewire stack for their isochronous packet stream functionality. Although the drivers allow userspace applications to process PCM frames and MIDI messages in the packet streaming, they doesn't support the other functionalities such as volume control since it can be achieved by any usespace applications of Linux firewire subsystem. The service programs added by this patchset are for the supplemental functionality to control the models by asynchronous packet transaction or to process a part of data transferred by the isochronous packet stream. In detail, please read README.rst in the repository[1]. You can see some illustrations. How to build ------------- The project depends on several libraries in below repositories: * glib[2] * alsa-gobject v0.1.0[3] * libhinawa v2.0.0[4] Before building it, please install them with gobject-introspection support. The project is written by Rust programming language[5] and packaged by cargo[6]. To build, just run below commands in the environment to connect to internet: ``` $ git clone https://github.com/alsa-project/snd-firewire-ctl-services.git -b topic/services-for-dg00x-tascam $ cd snd-firewire-ctl-services $ cargo build ``` All of required crates are downloaded automatically, then start build. Executables ----------- The added executables are listed below: * snd-firewire-digi00x-ctl-service * snd-firewire-tascam-ctl-service When running, the programs add and maintain control elements, then enter event loop to dispatch events. Thes elements are visible and manipulable for any userspace applications. The program dispatch any manipulation event for transactions to control actual device. For console model and FE-8 in Tascam FireWire series, the program adds and maintains port in ALSA Sequencer. They have command line options. For all models of Digi 002/003 family, the executable has an option for the numerical ID of sound card. ``` Usage: snd-firewire-digi00x-ctl-service CARD_ID where: CARD_ID: The numerical ID of sound card. ``` For models of Tascam FireWire series except for FE-8, the executable should run with 'snd' as its first option and the numerical ID of sound card as second option. Tascam FE-8 is supported by the service program but not supported by ALSA firewire stack since it doesn't handle isochronous packet streaming. For FE-8, the executable should run with 'fw' as its first option and the numerical ID of firewire character device. ``` Usage: snd-firewire-tascam-ctl-service SUBSYSTEM ID where: SUBSYSTEM: The name of subsystem; 'snd' or 'fw' ID: The numerical ID of sound card or fw character device ``` It's easy to run the executables by cargo, like: ``` $ cargo run --bin snd-firewire-digi00x-ctl-service 2 ``` Receiving SIGTERM signal terminates the event loop, then the program finishes. Access permissions for relevant character devices -------------------------------------------------- The executables manipulate below character devices to dispatch, convert, and emit event from end to end. The '%u' is the numerical number in each subsystem: * ALSA control character device (/dev/snd/controlC%u) * ALSA hwdep character device (/dev/snd/hwC%uD%u) * Linux firewire character device (/dev/fw%u) Additionally, the service program for Tascam FireWire series manipulates ALSA sequencer character device (/dev/snd/seq) to handle events of control surface for console model. Users should pay enough attention to access permission for the above character devices when running the service program. Feedback -------- Any feedback is welcome. For questions, please use mailing list in alsa-devel or alsa-users[7]. For issues, please use service of github.com[8]. Issues ------ The name of control elements are not fixed yet since the convention of name for element set is not clear yet for recording equipment. Users may feel inconvenience to which channel corresponds to which physical port. Furthermore, in the case that element set includes several elements, it's unclear that which element corresponds to which physical port as well. ALSA control core has no feature for the above issues at present. [1] https://github.com/alsa-project/snd-firewire-ctl-services [2] https://gitlab.gnome.org/GNOME/glib [3] https://github.com/alsa-project/alsa-gobject/ [4] https://github.com/alsa-project/libhinawa [5] https://www.rust-lang.org/ [6] https://doc.rust-lang.org/cargo/ [7] https://www.alsa-project.org/wiki/Mailing-lists [8] https://github.com/alsa-project/snd-firewire-ctl-services/issues Regards Takashi Sakamoto

2 4

[PATCH v2] dt-bindings: ASoC: Convert UniPhier EVEA codec to json-schema
by Masahiro Yamada 14 Jul '20

14 Jul '20

Convert the UniPhier EVEA sound codec binding to DT schema format. Signed-off-by: Masahiro Yamada <yamada.masahiro(a)socionext.com> --- Changes in v2: - Add schema for subnode 'port' .../sound/socionext,uniphier-evea.yaml | 70 +++++++++++++++++++ .../bindings/sound/uniphier,evea.txt | 26 ------- 2 files changed, 70 insertions(+), 26 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/socionext,uniphier-evea.yaml delete mode 100644 Documentation/devicetree/bindings/sound/uniphier,evea.txt diff --git a/Documentation/devicetree/bindings/sound/socionext,uniphier-evea.yaml b/Documentation/devicetree/bindings/sound/socionext,uniphier-evea.yaml new file mode 100644 index 000000000000..228168f685cf --- /dev/null +++ b/Documentation/devicetree/bindings/sound/socionext,uniphier-evea.yaml @@ -0,0 +1,70 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/socionext,uniphier-evea.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: UniPhier EVEA SoC-internal sound codec + +maintainers: + - <alsa-devel(a)alsa-project.org> + +properties: + compatible: + const: socionext,uniphier-evea + + reg: + maxItems: 1 + + clock-names: + items: + - const: evea + - const: exiv + + clocks: + minItems: 2 + maxItems: 2 + + reset-names: + items: + - const: evea + - const: exiv + - const: adamv + + resets: + minItems: 3 + maxItems: 3 + + "#sound-dai-cells": + const: 1 + +patternProperties: + "^port@[0-9]$": + type: object + properties: + endpoint: true + required: + - endpoint + +additionalProperties: false + +required: + - compatible + - reg + - clock-names + - clocks + - reset-names + - resets + - "#sound-dai-cells" + +examples: + - | + codec@57900000 { + compatible = "socionext,uniphier-evea"; + reg = <0x57900000 0x1000>; + clock-names = "evea", "exiv"; + clocks = <&sys_clk 41>, <&sys_clk 42>; + reset-names = "evea", "exiv", "adamv"; + resets = <&sys_rst 41>, <&sys_rst 42>, <&adamv_rst 0>; + #sound-dai-cells = <1>; + }; diff --git a/Documentation/devicetree/bindings/sound/uniphier,evea.txt b/Documentation/devicetree/bindings/sound/uniphier,evea.txt deleted file mode 100644 index 3f31b235f18b..000000000000 --- a/Documentation/devicetree/bindings/sound/uniphier,evea.txt +++ /dev/null @@ -1,26 +0,0 @@ -Socionext EVEA - UniPhier SoC internal codec driver - -Required properties: -- compatible : should be "socionext,uniphier-evea". -- reg : offset and length of the register set for the device. -- clock-names : should include following entries: - "evea", "exiv" -- clocks : a list of phandle, should contain an entry for each - entries in clock-names. -- reset-names : should include following entries: - "evea", "exiv", "adamv" -- resets : a list of phandle, should contain reset entries of - reset-names. -- #sound-dai-cells: should be 1. - -Example: - - codec { - compatible = "socionext,uniphier-evea"; - reg = <0x57900000 0x1000>; - clock-names = "evea", "exiv"; - clocks = <&sys_clk 41>, <&sys_clk 42>; - reset-names = "evea", "exiv", "adamv"; - resets = <&sys_rst 41>, <&sys_rst 42>, <&adamv_rst 0>; - #sound-dai-cells = <1>; - }; -- 2.25.1

4 7

[PATCH v2 0/5] ASoC: sh: remove discriminatory terms
by Kuninori Morimoto 14 Jul '20

14 Jul '20

Renesas SH drivers are using discriminatory terms. This patch-set removes or changes it as much as possible. But, because DMA related API function name, it still exists. I hope all these are removed someday. v1 -> v2 - use "secondary" instead of "follower" - care siu/ssi drivers - tidyup git-log Link: https://lore.kernel.org/r/87r1tg3swv.wl-kuninori.morimoto.gx@renesas.com Kuninori Morimoto (5): ASoC: rsnd: don't use discriminatory terms for function names ASoC: rsnd: don't use discriminatory terms for comment ASoC: fsi: don't use discriminatory terms for comment ASoC: siu: don't use discriminatory terms for parameter ASoC: ssi: don't use discriminatory terms for debug log sound/soc/sh/fsi.c | 4 ++-- sound/soc/sh/rcar/core.c | 4 ++-- sound/soc/sh/rcar/rsnd.h | 2 +- sound/soc/sh/rcar/ssi.c | 28 ++++++++++++++-------------- sound/soc/sh/rcar/ssiu.c | 6 +++--- sound/soc/sh/siu_pcm.c | 6 +++--- sound/soc/sh/ssi.c | 2 +- 7 files changed, 26 insertions(+), 26 deletions(-) -- 2.25.1

2 6

[PATCH 0/3] ASoC: sh: remove discriminatory terms
by Kuninori Morimoto 14 Jul '20

14 Jul '20

rsnd / fsi are using discriminatory terms. This patch-set removes it as much as possible. But, because DMA related API function name, it still exists. I hope all these are removed someday. Kuninori Morimoto (3): ASoC: rsnd: don't use discriminatory terms for function names ASoC: rsnd: don't use discriminatory terms for comment ASoC: fsi: don't use discriminatory terms for comment sound/soc/sh/fsi.c | 4 ++-- sound/soc/sh/rcar/core.c | 4 ++-- sound/soc/sh/rcar/rsnd.h | 2 +- sound/soc/sh/rcar/ssi.c | 28 ++++++++++++++-------------- sound/soc/sh/rcar/ssiu.c | 6 +++--- 5 files changed, 22 insertions(+), 22 deletions(-) -- 2.25.1

3 7

[PATCH RESEND] ASoC: rt5682: Enable Vref2 under using PLL2
by derek.fang＠realtek.com 14 Jul '20

14 Jul '20

From: "derek.fang" <derek.fang(a)realtek.com> Enable Vref2 under long term using PLL2 to avoid clock unstable. Signed-off-by: derek.fang <derek.fang(a)realtek.com> --- sound/soc/codecs/rt5682.c | 24 +++++++++++++++++------- 1 file changed, 17 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index d324512..ca8e1f2 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -960,13 +960,12 @@ int rt5682_headset_detect(struct snd_soc_component *component, int jack_insert) 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); - if (snd_soc_dapm_get_pin_status(dapm, "MICBIAS")) + if (!snd_soc_dapm_get_pin_status(dapm, "MICBIAS")) snd_soc_component_update_bits(component, - RT5682_PWR_ANLG_1, RT5682_PWR_VREF2, 0); - else + RT5682_PWR_ANLG_1, RT5682_PWR_MB, 0); + if (!snd_soc_dapm_get_pin_status(dapm, "Vref2")) snd_soc_component_update_bits(component, - RT5682_PWR_ANLG_1, - RT5682_PWR_VREF2 | RT5682_PWR_MB, 0); + RT5682_PWR_ANLG_1, RT5682_PWR_VREF2, 0); snd_soc_component_update_bits(component, RT5682_PWR_ANLG_3, RT5682_PWR_CBJ, 0); @@ -1601,8 +1600,7 @@ static const struct snd_soc_dapm_widget rt5682_dapm_widgets[] = { 0, set_filter_clk, SND_SOC_DAPM_PRE_PMU), SND_SOC_DAPM_SUPPLY("Vref1", RT5682_PWR_ANLG_1, RT5682_PWR_VREF1_BIT, 0, rt5682_set_verf, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), - SND_SOC_DAPM_SUPPLY("Vref2", RT5682_PWR_ANLG_1, RT5682_PWR_VREF2_BIT, 0, - NULL, 0), + SND_SOC_DAPM_SUPPLY("Vref2", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("MICBIAS", SND_SOC_NOPM, 0, 0, NULL, 0), /* ASRC */ @@ -2485,6 +2483,15 @@ static int rt5682_wclk_prepare(struct clk_hw *hw) snd_soc_dapm_force_enable_pin_unlocked(dapm, "MICBIAS"); snd_soc_component_update_bits(component, RT5682_PWR_ANLG_1, RT5682_PWR_MB, RT5682_PWR_MB); + + snd_soc_dapm_force_enable_pin_unlocked(dapm, "Vref2"); + snd_soc_component_update_bits(component, RT5682_PWR_ANLG_1, + RT5682_PWR_VREF2 | RT5682_PWR_FV2, + RT5682_PWR_VREF2); + usleep_range(55000, 60000); + snd_soc_component_update_bits(component, RT5682_PWR_ANLG_1, + RT5682_PWR_FV2, RT5682_PWR_FV2); + snd_soc_dapm_force_enable_pin_unlocked(dapm, "I2S1"); snd_soc_dapm_force_enable_pin_unlocked(dapm, "PLL2F"); snd_soc_dapm_force_enable_pin_unlocked(dapm, "PLL2B"); @@ -2510,9 +2517,12 @@ static void rt5682_wclk_unprepare(struct clk_hw *hw) snd_soc_dapm_mutex_lock(dapm); snd_soc_dapm_disable_pin_unlocked(dapm, "MICBIAS"); + snd_soc_dapm_disable_pin_unlocked(dapm, "Vref2"); if (!rt5682->jack_type) snd_soc_component_update_bits(component, RT5682_PWR_ANLG_1, + RT5682_PWR_VREF2 | RT5682_PWR_FV2 | RT5682_PWR_MB, 0); + snd_soc_dapm_disable_pin_unlocked(dapm, "I2S1"); snd_soc_dapm_disable_pin_unlocked(dapm, "PLL2F"); snd_soc_dapm_disable_pin_unlocked(dapm, "PLL2B"); -- 2.7.4

2 1

[PATCH v3] dt-bindings: sound: convert rk3328 codec binding to yaml
by Katsuhiro Suzuki 14 Jul '20

14 Jul '20

This patch converts Rockchip rk3328 audio codec binding to DT schema. And adds description about "mclk" clock and fixes some errors in original example. Signed-off-by: Katsuhiro Suzuki <katsuhiro(a)katsuster.net> Reviewed-by: Rob Herring <robh(a)kernel.org> --- .../bindings/sound/rockchip,rk3328-codec.txt | 28 -------- .../bindings/sound/rockchip,rk3328-codec.yaml | 69 +++++++++++++++++++ 2 files changed, 69 insertions(+), 28 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.txt create mode 100644 Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.yaml diff --git a/Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.txt b/Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.txt deleted file mode 100644 index 1ecd75d2032a..000000000000 --- a/Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.txt +++ /dev/null @@ -1,28 +0,0 @@ -* Rockchip Rk3328 internal codec - -Required properties: - -- compatible: "rockchip,rk3328-codec" -- reg: physical base address of the controller and length of memory mapped - region. -- rockchip,grf: the phandle of the syscon node for GRF register. -- clocks: a list of phandle + clock-specifer pairs, one for each entry in clock-names. -- clock-names: should be "pclk". -- spk-depop-time-ms: speak depop time msec. - -Optional properties: - -- mute-gpios: GPIO specifier for external line driver control (typically the - dedicated GPIO_MUTE pin) - -Example for rk3328 internal codec: - -codec: codec@ff410000 { - compatible = "rockchip,rk3328-codec"; - reg = <0x0 0xff410000 0x0 0x1000>; - rockchip,grf = <&grf>; - clocks = <&cru PCLK_ACODEC>; - clock-names = "pclk"; - mute-gpios = <&grf_gpio 0 GPIO_ACTIVE_LOW>; - spk-depop-time-ms = 100; -}; diff --git a/Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.yaml b/Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.yaml new file mode 100644 index 000000000000..5b85ad5e4834 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.yaml @@ -0,0 +1,69 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/rockchip,rk3328-codec.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Rockchip rk3328 internal codec + +maintainers: + - Heiko Stuebner <heiko(a)sntech.de> + +properties: + compatible: + const: rockchip,rk3328-codec + + reg: + maxItems: 1 + + clocks: + items: + - description: clock for audio codec + - description: clock for I2S master clock + + clock-names: + items: + - const: pclk + - const: mclk + + rockchip,grf: + $ref: /schemas/types.yaml#/definitions/phandle + description: + The phandle of the syscon node for the GRF register. + + spk-depop-time-ms: + default: 200 + description: + Speaker depop time in msec. + + mute-gpios: + maxItems: 1 + description: + GPIO specifier for external line driver control (typically the + dedicated GPIO_MUTE pin) + + "#sound-dai-cells": + const: 0 + +required: + - compatible + - reg + - clocks + - clock-names + - rockchip,grf + - "#sound-dai-cells" + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + #include <dt-bindings/clock/rk3328-cru.h> + codec: codec@ff410000 { + compatible = "rockchip,rk3328-codec"; + reg = <0xff410000 0x1000>; + clocks = <&cru PCLK_ACODECPHY>, <&cru SCLK_I2S1>; + clock-names = "pclk", "mclk"; + rockchip,grf = <&grf>; + mute-gpios = <&grf_gpio 0 GPIO_ACTIVE_LOW>; + spk-depop-time-ms = <100>; + #sound-dai-cells = <0>; + }; -- 2.27.0

2 1

[PATCH v2] dt-bindings: ASoC: Convert UniPhier AIO audio system to json-schema
by Masahiro Yamada 14 Jul '20

14 Jul '20

Convert the UniPhier AIO audio system binding to DT schema format. Signed-off-by: Masahiro Yamada <yamada.masahiro(a)socionext.com> --- Changes in v2: - Add schema for subnode 'port' .../sound/socionext,uniphier-aio.yaml | 81 +++++++++++++++++++ .../bindings/sound/uniphier,aio.txt | 45 ----------- 2 files changed, 81 insertions(+), 45 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/socionext,uniphier-aio.yaml delete mode 100644 Documentation/devicetree/bindings/sound/uniphier,aio.txt diff --git a/Documentation/devicetree/bindings/sound/socionext,uniphier-aio.yaml b/Documentation/devicetree/bindings/sound/socionext,uniphier-aio.yaml new file mode 100644 index 000000000000..4987eb91f2ab --- /dev/null +++ b/Documentation/devicetree/bindings/sound/socionext,uniphier-aio.yaml @@ -0,0 +1,81 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/socionext,uniphier-aio.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: UniPhier AIO audio system + +maintainers: + - <alsa-devel(a)alsa-project.org> + +properties: + compatible: + enum: + - socionext,uniphier-ld11-aio + - socionext,uniphier-ld20-aio + - socionext,uniphier-pxs2-aio + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clock-names: + const: aio + + clocks: + maxItems: 1 + + reset-names: + const: aio + + resets: + maxItems: 1 + + socionext,syscon: + description: | + Specifies a phandle to soc-glue, which is used for changing mode of S/PDIF + signal pin to output from Hi-Z. This property is optional if you use I2S + signal pins only. + $ref: "/schemas/types.yaml#/definitions/phandle" + + "#sound-dai-cells": + const: 1 + +patternProperties: + "^port@[0-9]$": + type: object + properties: + endpoint: true + required: + - endpoint + +additionalProperties: false + +required: + - compatible + - reg + - interrupts + - clock-names + - clocks + - reset-names + - resets + - "#sound-dai-cells" + +examples: + - | + audio@56000000 { + compatible = "socionext,uniphier-ld20-aio"; + reg = <0x56000000 0x80000>; + interrupts = <0 144 4>; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_aout>; + clock-names = "aio"; + clocks = <&sys_clk 40>; + reset-names = "aio"; + resets = <&sys_rst 40>; + #sound-dai-cells = <1>; + socionext,syscon = <&soc_glue>; + }; diff --git a/Documentation/devicetree/bindings/sound/uniphier,aio.txt b/Documentation/devicetree/bindings/sound/uniphier,aio.txt deleted file mode 100644 index 4ce68ed6f2f2..000000000000 --- a/Documentation/devicetree/bindings/sound/uniphier,aio.txt +++ /dev/null @@ -1,45 +0,0 @@ -Socionext UniPhier SoC audio driver - -The Socionext UniPhier audio subsystem consists of I2S and S/PDIF blocks in -the same register space. - -Required properties: -- compatible : should be one of the following: - "socionext,uniphier-ld11-aio" - "socionext,uniphier-ld20-aio" - "socionext,uniphier-pxs2-aio" -- reg : offset and length of the register set for the device. -- interrupts : should contain I2S or S/PDIF interrupt. -- pinctrl-names : should be "default". -- pinctrl-0 : defined I2S signal pins for an external codec chip. -- clock-names : should include following entries: - "aio" -- clocks : a list of phandle, should contain an entry for each - entry in clock-names. -- reset-names : should include following entries: - "aio" -- resets : a list of phandle, should contain an entry for each - entry in reset-names. -- #sound-dai-cells: should be 1. - -Optional properties: -- socionext,syscon: a phandle, should contain soc-glue. - The soc-glue is used for changing mode of S/PDIF signal pin - to Output from Hi-Z. This property is optional if you use - I2S signal pins only. - -Example: - audio { - compatible = "socionext,uniphier-ld20-aio"; - reg = <0x56000000 0x80000>; - interrupts = <0 144 4>; - pinctrl-names = "default"; - pinctrl-0 = <&pinctrl_aout>; - clock-names = "aio"; - clocks = <&sys_clk 40>; - reset-names = "aio"; - resets = <&sys_rst 40>; - #sound-dai-cells = <1>; - - socionext,syscon = <&sg>; - }; -- 2.25.1

3 2

ASoC: codecs: wm9712: Mux no paths errors
by Primoz Fiser 14 Jul '20

14 Jul '20

Hi all, I have a sound setup based on fsl_ssi + wm9712 codec. On every drivers load I get the following console error: >> [ 5666.776770] wm9712-codec wm9712-codec: ASoC: mux Capture Phone Mux has no paths >> [ 5666.784281] wm9712-codec wm9712-codec: ASoC: mux Differential Source has no paths What is the proper way to get rid of this errors? As a temporary solution I have ifdef-ed the following muxes from the driver like so: > @@ -364,8 +365,10 @@ SND_SOC_DAPM_MUX("Out3 Mux", SND_SOC_NOPM, 0, 0, > &wm9712_out3_mux_controls), > SND_SOC_DAPM_MUX("Speaker Mux", SND_SOC_NOPM, 0, 0, > &wm9712_spk_mux_controls), > +#if 0 > SND_SOC_DAPM_MUX("Capture Phone Mux", SND_SOC_NOPM, 0, 0, > &wm9712_capture_phone_mux_controls), > +#endif > SND_SOC_DAPM_MUX("Left Capture Select", SND_SOC_NOPM, 0, 0, > &wm9712_capture_selectl_controls), > SND_SOC_DAPM_MUX("Right Capture Select", SND_SOC_NOPM, 0, 0, > @@ -374,8 +377,10 @@ SND_SOC_DAPM_MUX("Left Mic Select Source", SND_SOC_NOPM, 0, 0, > &wm9712_mic_src_controls), > SND_SOC_DAPM_MUX("Right Mic Select Source", SND_SOC_NOPM, 0, 0, > &wm9712_mic_src_controls), > +#ifdef 0 > SND_SOC_DAPM_MUX("Differential Source", SND_SOC_NOPM, 0, 0, > &wm9712_diff_sel_controls), > +#endif > SND_SOC_DAPM_MIXER("AC97 Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), > SND_SOC_DAPM_MIXER("Left HP Mixer", AC97_INT_PAGING, 9, 1, > &wm9712_hpl_mixer_controls[0], ARRAY_SIZE(wm9712_hpl_mixer_controls)), which works but feels like a dirty workaround, right? How can this be properly fixed in the wm9712 driver or somewhere else? BR, Primoz

2 1

[PATCH] ALSA: hda/tegra: Disable sync-write operation
by Jon Hunter 14 Jul '20

14 Jul '20

Commit f34a4c9dd4ca ("ALSA: hda: Enable sync-write operation as default for all controllers") enabled sync-write for all controllers and this is causing audio playback on the Tegra186 HDA device to fail. For now, disable sync-write support for Tegra to fix this. Signed-off-by: Jon Hunter <jonathanh(a)nvidia.com> --- sound/pci/hda/hda_tegra.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c index 0cc5fad1af8a..5637f0129932 100644 --- a/sound/pci/hda/hda_tegra.c +++ b/sound/pci/hda/hda_tegra.c @@ -443,6 +443,7 @@ static int hda_tegra_create(struct snd_card *card, if (err < 0) return err; + chip->bus.core.sync_write = 0; chip->bus.core.needs_damn_long_delay = 1; chip->bus.core.aligned_mmio = 1; -- 2.17.1

2 1

[snd-usb-6fire] Unmark struct snd_pcm_hardware const
by René Herman 14 Jul '20

14 Jul '20

struct snd_pcm_hardware pcm_hw is in fact changed in usb6fire_pcm_open() Signed-off-by: René Herman <rene.herman(a)gmail.com>

1 0

[snd-usb-6fire] Pulseaudio needs snd_pcm_hardware.channels_min > 1
by René Herman 14 Jul '20

14 Jul '20

Pulseaudio with snd_pcm_hardware.channels_min=1 creates a mono device only. Signed-off-by: René Herman <rene.herman(a)gmail.com>

1 0

[snd-usb-6fire] Move DMA-buffer off of the stack
by René Herman 14 Jul '20

14 Jul '20

snd-usb-fire currently fails its firmware load with "transfer buffer not dma capable". Move said buffer off of the stack (has been coordinated with the original driver author Torsten Schenk) Signed-off-by: René Herman <rene.herman(a)gmail.com>

1 0

[PATCH] When snd_soc_card_jack_new is not called or the call fails, calling this function causes a null pointer access
by xingxing qiao 14 Jul '20

14 Jul '20

Signed-off-by: mnlife <mnlife(a)foxmail.com> ---  sound/soc/soc-jack.c | 2 +-  1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index 0f1820f36b4d..51b799ee98b7 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -44,7 +44,7 @@ void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask)  unsigned int sync = 0;  int enable;   -if (!jack) +if (!jack || !jack->jack)  return;  trace_snd_soc_jack_report(jack, mask, status);   -- 2.17.1

2 1

[PATCH AUTOSEL 4.9 07/10] ALSA: hda/hdmi: fix failures at PCM open on Intel ICL and later
by Sasha Levin 14 Jul '20

14 Jul '20

From: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> [ Upstream commit 56275036d8185f92eceac7479d48b858ee3dab84 ] When HDMI PCM devices are opened in a specific order, with at least one HDMI/DP receiver connected, ALSA PCM open fails to -EBUSY on the connected monitor, on recent Intel platforms (ICL/JSL and newer). While this is not a typical sequence, at least Pulseaudio does this every time when it is started, to discover the available PCMs. The rootcause is an invalid assumption in hdmi_add_pin(), where the total number of converters is assumed to be known at the time the function is called. On older Intel platforms this held true, but after ICL/JSL, the order how pins and converters are in the subnode list as returned by snd_hda_get_sub_nodes(), was changed. As a result, information for some converters was not stored to per_pin->mux_nids. And this means some pins cannot be connected to all converters, and application instead gets -EBUSY instead at open. The assumption that converters are always before pins in the subnode list, is not really a valid one. Fix the problem in hdmi_parse_codec() by introducing separate loops for discovering converters and pins. BugLink: https://github.com/thesofproject/linux/issues/1978 BugLink: https://github.com/thesofproject/linux/issues/2216 BugLink: https://github.com/thesofproject/linux/issues/2217 Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Link: https://lore.kernel.org/r/20200703153818.2808592-1-kai.vehmanen@linux.intel… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_hdmi.c | 36 +++++++++++++++++++++++------------- 1 file changed, 23 insertions(+), 13 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 2def4ad579ccf..f7bd91b713e83 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1655,33 +1655,43 @@ static int hdmi_add_cvt(struct hda_codec *codec, hda_nid_t cvt_nid) static int hdmi_parse_codec(struct hda_codec *codec) { - hda_nid_t nid; + hda_nid_t start_nid; + unsigned int caps; int i, nodes; - nodes = snd_hda_get_sub_nodes(codec, codec->core.afg, &nid); - if (!nid || nodes < 0) { + nodes = snd_hda_get_sub_nodes(codec, codec->core.afg, &start_nid); + if (!start_nid || nodes < 0) { codec_warn(codec, "HDMI: failed to get afg sub nodes\n"); return -EINVAL; } - for (i = 0; i < nodes; i++, nid++) { - unsigned int caps; - unsigned int type; + /* + * hdmi_add_pin() assumes total amount of converters to + * be known, so first discover all converters + */ + for (i = 0; i < nodes; i++) { + hda_nid_t nid = start_nid + i; caps = get_wcaps(codec, nid); - type = get_wcaps_type(caps); if (!(caps & AC_WCAP_DIGITAL)) continue; - switch (type) { - case AC_WID_AUD_OUT: + if (get_wcaps_type(caps) == AC_WID_AUD_OUT) hdmi_add_cvt(codec, nid); - break; - case AC_WID_PIN: + } + + /* discover audio pins */ + for (i = 0; i < nodes; i++) { + hda_nid_t nid = start_nid + i; + + caps = get_wcaps(codec, nid); + + if (!(caps & AC_WCAP_DIGITAL)) + continue; + + if (get_wcaps_type(caps) == AC_WID_PIN) hdmi_add_pin(codec, nid); - break; - } } return 0; -- 2.25.1

1 0

[PATCH AUTOSEL 4.14 09/12] ALSA: hda/hdmi: fix failures at PCM open on Intel ICL and later
by Sasha Levin 14 Jul '20

14 Jul '20

From: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> [ Upstream commit 56275036d8185f92eceac7479d48b858ee3dab84 ] When HDMI PCM devices are opened in a specific order, with at least one HDMI/DP receiver connected, ALSA PCM open fails to -EBUSY on the connected monitor, on recent Intel platforms (ICL/JSL and newer). While this is not a typical sequence, at least Pulseaudio does this every time when it is started, to discover the available PCMs. The rootcause is an invalid assumption in hdmi_add_pin(), where the total number of converters is assumed to be known at the time the function is called. On older Intel platforms this held true, but after ICL/JSL, the order how pins and converters are in the subnode list as returned by snd_hda_get_sub_nodes(), was changed. As a result, information for some converters was not stored to per_pin->mux_nids. And this means some pins cannot be connected to all converters, and application instead gets -EBUSY instead at open. The assumption that converters are always before pins in the subnode list, is not really a valid one. Fix the problem in hdmi_parse_codec() by introducing separate loops for discovering converters and pins. BugLink: https://github.com/thesofproject/linux/issues/1978 BugLink: https://github.com/thesofproject/linux/issues/2216 BugLink: https://github.com/thesofproject/linux/issues/2217 Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Link: https://lore.kernel.org/r/20200703153818.2808592-1-kai.vehmanen@linux.intel… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_hdmi.c | 36 +++++++++++++++++++++++------------- 1 file changed, 23 insertions(+), 13 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 9e8cfc409b4b3..04420a3bf5eb0 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1805,33 +1805,43 @@ static int hdmi_add_cvt(struct hda_codec *codec, hda_nid_t cvt_nid) static int hdmi_parse_codec(struct hda_codec *codec) { - hda_nid_t nid; + hda_nid_t start_nid; + unsigned int caps; int i, nodes; - nodes = snd_hda_get_sub_nodes(codec, codec->core.afg, &nid); - if (!nid || nodes < 0) { + nodes = snd_hda_get_sub_nodes(codec, codec->core.afg, &start_nid); + if (!start_nid || nodes < 0) { codec_warn(codec, "HDMI: failed to get afg sub nodes\n"); return -EINVAL; } - for (i = 0; i < nodes; i++, nid++) { - unsigned int caps; - unsigned int type; + /* + * hdmi_add_pin() assumes total amount of converters to + * be known, so first discover all converters + */ + for (i = 0; i < nodes; i++) { + hda_nid_t nid = start_nid + i; caps = get_wcaps(codec, nid); - type = get_wcaps_type(caps); if (!(caps & AC_WCAP_DIGITAL)) continue; - switch (type) { - case AC_WID_AUD_OUT: + if (get_wcaps_type(caps) == AC_WID_AUD_OUT) hdmi_add_cvt(codec, nid); - break; - case AC_WID_PIN: + } + + /* discover audio pins */ + for (i = 0; i < nodes; i++) { + hda_nid_t nid = start_nid + i; + + caps = get_wcaps(codec, nid); + + if (!(caps & AC_WCAP_DIGITAL)) + continue; + + if (get_wcaps_type(caps) == AC_WID_PIN) hdmi_add_pin(codec, nid); - break; - } } return 0; -- 2.25.1

1 0

[PATCH AUTOSEL 4.19 09/13] ALSA: hda/hdmi: fix failures at PCM open on Intel ICL and later
by Sasha Levin 14 Jul '20

14 Jul '20

From: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> [ Upstream commit 56275036d8185f92eceac7479d48b858ee3dab84 ] When HDMI PCM devices are opened in a specific order, with at least one HDMI/DP receiver connected, ALSA PCM open fails to -EBUSY on the connected monitor, on recent Intel platforms (ICL/JSL and newer). While this is not a typical sequence, at least Pulseaudio does this every time when it is started, to discover the available PCMs. The rootcause is an invalid assumption in hdmi_add_pin(), where the total number of converters is assumed to be known at the time the function is called. On older Intel platforms this held true, but after ICL/JSL, the order how pins and converters are in the subnode list as returned by snd_hda_get_sub_nodes(), was changed. As a result, information for some converters was not stored to per_pin->mux_nids. And this means some pins cannot be connected to all converters, and application instead gets -EBUSY instead at open. The assumption that converters are always before pins in the subnode list, is not really a valid one. Fix the problem in hdmi_parse_codec() by introducing separate loops for discovering converters and pins. BugLink: https://github.com/thesofproject/linux/issues/1978 BugLink: https://github.com/thesofproject/linux/issues/2216 BugLink: https://github.com/thesofproject/linux/issues/2217 Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Link: https://lore.kernel.org/r/20200703153818.2808592-1-kai.vehmanen@linux.intel… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_hdmi.c | 36 +++++++++++++++++++++++------------- 1 file changed, 23 insertions(+), 13 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 419d099b5582b..1fc12e680f311 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1804,33 +1804,43 @@ static int hdmi_add_cvt(struct hda_codec *codec, hda_nid_t cvt_nid) static int hdmi_parse_codec(struct hda_codec *codec) { - hda_nid_t nid; + hda_nid_t start_nid; + unsigned int caps; int i, nodes; - nodes = snd_hda_get_sub_nodes(codec, codec->core.afg, &nid); - if (!nid || nodes < 0) { + nodes = snd_hda_get_sub_nodes(codec, codec->core.afg, &start_nid); + if (!start_nid || nodes < 0) { codec_warn(codec, "HDMI: failed to get afg sub nodes\n"); return -EINVAL; } - for (i = 0; i < nodes; i++, nid++) { - unsigned int caps; - unsigned int type; + /* + * hdmi_add_pin() assumes total amount of converters to + * be known, so first discover all converters + */ + for (i = 0; i < nodes; i++) { + hda_nid_t nid = start_nid + i; caps = get_wcaps(codec, nid); - type = get_wcaps_type(caps); if (!(caps & AC_WCAP_DIGITAL)) continue; - switch (type) { - case AC_WID_AUD_OUT: + if (get_wcaps_type(caps) == AC_WID_AUD_OUT) hdmi_add_cvt(codec, nid); - break; - case AC_WID_PIN: + } + + /* discover audio pins */ + for (i = 0; i < nodes; i++) { + hda_nid_t nid = start_nid + i; + + caps = get_wcaps(codec, nid); + + if (!(caps & AC_WCAP_DIGITAL)) + continue; + + if (get_wcaps_type(caps) == AC_WID_PIN) hdmi_add_pin(codec, nid); - break; - } } return 0; -- 2.25.1

1 0

[PATCH AUTOSEL 5.4 11/18] ALSA: hda/hdmi: fix failures at PCM open on Intel ICL and later
by Sasha Levin 14 Jul '20

14 Jul '20

From: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> [ Upstream commit 56275036d8185f92eceac7479d48b858ee3dab84 ] When HDMI PCM devices are opened in a specific order, with at least one HDMI/DP receiver connected, ALSA PCM open fails to -EBUSY on the connected monitor, on recent Intel platforms (ICL/JSL and newer). While this is not a typical sequence, at least Pulseaudio does this every time when it is started, to discover the available PCMs. The rootcause is an invalid assumption in hdmi_add_pin(), where the total number of converters is assumed to be known at the time the function is called. On older Intel platforms this held true, but after ICL/JSL, the order how pins and converters are in the subnode list as returned by snd_hda_get_sub_nodes(), was changed. As a result, information for some converters was not stored to per_pin->mux_nids. And this means some pins cannot be connected to all converters, and application instead gets -EBUSY instead at open. The assumption that converters are always before pins in the subnode list, is not really a valid one. Fix the problem in hdmi_parse_codec() by introducing separate loops for discovering converters and pins. BugLink: https://github.com/thesofproject/linux/issues/1978 BugLink: https://github.com/thesofproject/linux/issues/2216 BugLink: https://github.com/thesofproject/linux/issues/2217 Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Link: https://lore.kernel.org/r/20200703153818.2808592-1-kai.vehmanen@linux.intel… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_hdmi.c | 36 +++++++++++++++++++++++------------- 1 file changed, 23 insertions(+), 13 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index e78c4367b6c86..820f534a67b1c 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1817,33 +1817,43 @@ static int hdmi_add_cvt(struct hda_codec *codec, hda_nid_t cvt_nid) static int hdmi_parse_codec(struct hda_codec *codec) { - hda_nid_t nid; + hda_nid_t start_nid; + unsigned int caps; int i, nodes; - nodes = snd_hda_get_sub_nodes(codec, codec->core.afg, &nid); - if (!nid || nodes < 0) { + nodes = snd_hda_get_sub_nodes(codec, codec->core.afg, &start_nid); + if (!start_nid || nodes < 0) { codec_warn(codec, "HDMI: failed to get afg sub nodes\n"); return -EINVAL; } - for (i = 0; i < nodes; i++, nid++) { - unsigned int caps; - unsigned int type; + /* + * hdmi_add_pin() assumes total amount of converters to + * be known, so first discover all converters + */ + for (i = 0; i < nodes; i++) { + hda_nid_t nid = start_nid + i; caps = get_wcaps(codec, nid); - type = get_wcaps_type(caps); if (!(caps & AC_WCAP_DIGITAL)) continue; - switch (type) { - case AC_WID_AUD_OUT: + if (get_wcaps_type(caps) == AC_WID_AUD_OUT) hdmi_add_cvt(codec, nid); - break; - case AC_WID_PIN: + } + + /* discover audio pins */ + for (i = 0; i < nodes; i++) { + hda_nid_t nid = start_nid + i; + + caps = get_wcaps(codec, nid); + + if (!(caps & AC_WCAP_DIGITAL)) + continue; + + if (get_wcaps_type(caps) == AC_WID_PIN) hdmi_add_pin(codec, nid); - break; - } } return 0; -- 2.25.1

1 0

[PATCH AUTOSEL 5.7 12/19] ALSA: hda/hdmi: fix failures at PCM open on Intel ICL and later
by Sasha Levin 14 Jul '20

14 Jul '20

From: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> [ Upstream commit 56275036d8185f92eceac7479d48b858ee3dab84 ] When HDMI PCM devices are opened in a specific order, with at least one HDMI/DP receiver connected, ALSA PCM open fails to -EBUSY on the connected monitor, on recent Intel platforms (ICL/JSL and newer). While this is not a typical sequence, at least Pulseaudio does this every time when it is started, to discover the available PCMs. The rootcause is an invalid assumption in hdmi_add_pin(), where the total number of converters is assumed to be known at the time the function is called. On older Intel platforms this held true, but after ICL/JSL, the order how pins and converters are in the subnode list as returned by snd_hda_get_sub_nodes(), was changed. As a result, information for some converters was not stored to per_pin->mux_nids. And this means some pins cannot be connected to all converters, and application instead gets -EBUSY instead at open. The assumption that converters are always before pins in the subnode list, is not really a valid one. Fix the problem in hdmi_parse_codec() by introducing separate loops for discovering converters and pins. BugLink: https://github.com/thesofproject/linux/issues/1978 BugLink: https://github.com/thesofproject/linux/issues/2216 BugLink: https://github.com/thesofproject/linux/issues/2217 Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Link: https://lore.kernel.org/r/20200703153818.2808592-1-kai.vehmanen@linux.intel… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_hdmi.c | 36 +++++++++++++++++++++++------------- 1 file changed, 23 insertions(+), 13 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 137d655fed8f8..e821c9df81070 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1804,33 +1804,43 @@ static int hdmi_add_cvt(struct hda_codec *codec, hda_nid_t cvt_nid) static int hdmi_parse_codec(struct hda_codec *codec) { - hda_nid_t nid; + hda_nid_t start_nid; + unsigned int caps; int i, nodes; - nodes = snd_hda_get_sub_nodes(codec, codec->core.afg, &nid); - if (!nid || nodes < 0) { + nodes = snd_hda_get_sub_nodes(codec, codec->core.afg, &start_nid); + if (!start_nid || nodes < 0) { codec_warn(codec, "HDMI: failed to get afg sub nodes\n"); return -EINVAL; } - for (i = 0; i < nodes; i++, nid++) { - unsigned int caps; - unsigned int type; + /* + * hdmi_add_pin() assumes total amount of converters to + * be known, so first discover all converters + */ + for (i = 0; i < nodes; i++) { + hda_nid_t nid = start_nid + i; caps = get_wcaps(codec, nid); - type = get_wcaps_type(caps); if (!(caps & AC_WCAP_DIGITAL)) continue; - switch (type) { - case AC_WID_AUD_OUT: + if (get_wcaps_type(caps) == AC_WID_AUD_OUT) hdmi_add_cvt(codec, nid); - break; - case AC_WID_PIN: + } + + /* discover audio pins */ + for (i = 0; i < nodes; i++) { + hda_nid_t nid = start_nid + i; + + caps = get_wcaps(codec, nid); + + if (!(caps & AC_WCAP_DIGITAL)) + continue; + + if (get_wcaps_type(caps) == AC_WID_PIN) hdmi_add_pin(codec, nid); - break; - } } return 0; -- 2.25.1

1 0

[PATCH v2 00/10] topology: decode: Various fixes
by Piotr Maziarz 14 Jul '20

14 Jul '20

This series fixes various problems with topology decoding mechanism. Some of the problems were critical like improper memory management or infinite loops that were causing undefined behaviour or program crashes, while other resulted in losing some data during conversion. Bugs found while testing with Intel SST topologies. Changelog: v2: -Divide into more patches, critical fixes are in separate patches now -More specific descriptions -Fix a typo UML to UCM -Add error reporting in topology: decode: fix channel map memory allocation -Remove goto again in topology: Make buffer for saving dynamic size for better readability Piotr Maziarz (10): topology: decode: Fix channel map memory allocation topology: decode: Fix infinite loop in decoding enum control topology: decode: Remove decoding values for enum control topology: decode: Add enum control texts as separate element topology: decode: Fix printing texts section topology: decode: Change declaration of enum decoding function topology: decode: Fix decoding PCM formats and rates topology: decode: Print sig_bits field in PCM capabilities section topology: decode: Add DAI name printing topology: Make buffer for saving dynamic size src/topology/ctl.c | 51 ++++++++++++++++++++++------------------------- src/topology/dapm.c | 3 +-- src/topology/pcm.c | 11 +++++++--- src/topology/save.c | 34 ++++++++++++++++++++++++++----- src/topology/text.c | 2 +- src/topology/tplg_local.h | 2 +- 6 files changed, 64 insertions(+), 39 deletions(-) -- 2.7.4

2 14

Remove cset value quotation
by GitHub pull_request - opened 14 Jul '20

14 Jul '20

alsa-project/alsa-ucm-conf pull request #40 was opened from libinyang: This patchset removes the quotation marks of the cset values in the ucm2 config files. In ucm2 files, the cset values should not use quotation marks. Otherwise, the cset won't work. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/40 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/40.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH 0/2] iec958 plugin enhancements for HDMI
by Matthias Reichl 14 Jul '20

14 Jul '20

Testing with vc4-hdmi / Raspberry Pi 4 showed that the iec958 plugin was missing two features required for HDMI compatibility: HD audio bitstreams like DTS HD MA or Dolby True HD are usually packed into 8-channel 192kHz 16-bit frames but are in fact a single IEC 61937 stream at 768kHz frame rate. So the data in the 8-channel frame has to be formatted as 4 contiguous IEC 60958 frames. If channel status bits weren't set, iec958 indicated a sample rate of 48kHz in the channel status bits which the HDMI analyzer complained about and caused issues with some HDMI devices if it didn't match the real rate. That was a long outstanding FIXME since the iec958 plugin was added. With this series HD audio passthrough and PCM audio works fine with the downstream 5.4 RPi kernel. Matthias Reichl (2): pcm: iec958: implement HDMI HBR audio formatting pcm: iec958: set channel status bits according to rate and format src/pcm/pcm_iec958.c | 117 ++++++++++++++++++++++++++++++++++++++++--- 1 file changed, 109 insertions(+), 8 deletions(-) -- 2.20.1

2 3

Re: [PATCH] soc jack: When snd_soc_card_jack_new is not called or the call fails, calling this function causes a null pointer access
by Mark Brown 14 Jul '20

14 Jul '20

On Fri, Jul 10, 2020 at 02:56:03PM +0000, qiao mnlife wrote: > ________________________________ > From: mnlife <mnlife(a)foxmail.com> > Sent: Friday, July 10, 2020 10:45:42 PM > To: mnlife.qiao(a)gmail.com <mnlife.qiao(a)gmail.com> > Cc: mnlife <mnlife(a)foxmail.com> > Subject: [PATCH] When snd_soc_card_jack_new is not called or the call fails, calling this function causes a null pointer access > > Signed-off-by: mnlife <mnlife(a)foxmail.com> This looks good but it's a bit mangled in submission - can you please try to resend using something closer to the format in submitting-patches.rst? The changelog should be in the body of the mail and there should be a patch subject in a similar format to to the other patches in git.

1 0

[PATCH] ALSA: hda: Enable sync-write operation as default for all controllers
by Takashi Iwai 14 Jul '20

14 Jul '20

In the end we already enabled the sync-write mode for most of HD-audio controllers including Intel, and it's no big merit to keep the async write mode for the rest. Let's make it as default and drop the superfluous AZX_DCAPS_SYNC_WRITE bit flag. Also, avoid to set the allow_bus_reset flag, which is a quite unstable and hackish behavior that was needed only for some early platforms (decades ago). The straight fallback to the single cmd mode is more robust. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_controller.c | 11 ++--------- sound/pci/hda/hda_controller.h | 2 +- sound/pci/hda/hda_intel.c | 16 +++++++--------- 3 files changed, 10 insertions(+), 19 deletions(-) diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index 9765652a73d7..80016b7b6849 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -1202,15 +1202,8 @@ int azx_bus_init(struct azx *chip, const char *model) if (chip->driver_caps & AZX_DCAPS_4K_BDLE_BOUNDARY) bus->core.align_bdle_4k = true; - /* AMD chipsets often cause the communication stalls upon certain - * sequence like the pin-detection. It seems that forcing the synced - * access works around the stall. Grrr... - */ - if (chip->driver_caps & AZX_DCAPS_SYNC_WRITE) { - dev_dbg(chip->card->dev, "Enable sync_write for stable communication\n"); - bus->core.sync_write = 1; - bus->allow_bus_reset = 1; - } + /* enable sync_write flag for stable communication as default */ + bus->core.sync_write = 1; return 0; } diff --git a/sound/pci/hda/hda_controller.h b/sound/pci/hda/hda_controller.h index 82e26442724b..fe171685492d 100644 --- a/sound/pci/hda/hda_controller.h +++ b/sound/pci/hda/hda_controller.h @@ -33,7 +33,7 @@ #define AZX_DCAPS_POSFIX_LPIB (1 << 16) /* Use LPIB as default */ #define AZX_DCAPS_AMD_WORKAROUND (1 << 17) /* AMD-specific workaround */ #define AZX_DCAPS_NO_64BIT (1 << 18) /* No 64bit address */ -#define AZX_DCAPS_SYNC_WRITE (1 << 19) /* sync each cmd write */ +/* 19 unused */ #define AZX_DCAPS_OLD_SSYNC (1 << 20) /* Old SSYNC reg for ICH */ #define AZX_DCAPS_NO_ALIGN_BUFSIZE (1 << 21) /* no buffer size alignment */ /* 22 unused */ diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 3565e2ab0965..b2bd21f5a52e 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -283,13 +283,12 @@ enum { /* quirks for old Intel chipsets */ #define AZX_DCAPS_INTEL_ICH \ - (AZX_DCAPS_OLD_SSYNC | AZX_DCAPS_NO_ALIGN_BUFSIZE |\ - AZX_DCAPS_SYNC_WRITE) + (AZX_DCAPS_OLD_SSYNC | AZX_DCAPS_NO_ALIGN_BUFSIZE) /* quirks for Intel PCH */ #define AZX_DCAPS_INTEL_PCH_BASE \ (AZX_DCAPS_NO_ALIGN_BUFSIZE | AZX_DCAPS_COUNT_LPIB_DELAY |\ - AZX_DCAPS_SNOOP_TYPE(SCH) | AZX_DCAPS_SYNC_WRITE) + AZX_DCAPS_SNOOP_TYPE(SCH)) /* PCH up to IVB; no runtime PM; bind with i915 gfx */ #define AZX_DCAPS_INTEL_PCH_NOPM \ @@ -304,13 +303,13 @@ enum { #define AZX_DCAPS_INTEL_HASWELL \ (/*AZX_DCAPS_ALIGN_BUFSIZE |*/ AZX_DCAPS_COUNT_LPIB_DELAY |\ AZX_DCAPS_PM_RUNTIME | AZX_DCAPS_I915_COMPONENT |\ - AZX_DCAPS_SNOOP_TYPE(SCH) | AZX_DCAPS_SYNC_WRITE) + AZX_DCAPS_SNOOP_TYPE(SCH)) /* Broadwell HDMI can't use position buffer reliably, force to use LPIB */ #define AZX_DCAPS_INTEL_BROADWELL \ (/*AZX_DCAPS_ALIGN_BUFSIZE |*/ AZX_DCAPS_POSFIX_LPIB |\ AZX_DCAPS_PM_RUNTIME | AZX_DCAPS_I915_COMPONENT |\ - AZX_DCAPS_SNOOP_TYPE(SCH) | AZX_DCAPS_SYNC_WRITE) + AZX_DCAPS_SNOOP_TYPE(SCH)) #define AZX_DCAPS_INTEL_BAYTRAIL \ (AZX_DCAPS_INTEL_PCH_BASE | AZX_DCAPS_I915_COMPONENT) @@ -321,19 +320,18 @@ enum { #define AZX_DCAPS_INTEL_SKYLAKE \ (AZX_DCAPS_INTEL_PCH_BASE | AZX_DCAPS_PM_RUNTIME |\ - AZX_DCAPS_SYNC_WRITE |\ AZX_DCAPS_SEPARATE_STREAM_TAG | AZX_DCAPS_I915_COMPONENT) #define AZX_DCAPS_INTEL_BROXTON AZX_DCAPS_INTEL_SKYLAKE /* quirks for ATI SB / AMD Hudson */ #define AZX_DCAPS_PRESET_ATI_SB \ - (AZX_DCAPS_NO_TCSEL | AZX_DCAPS_SYNC_WRITE | AZX_DCAPS_POSFIX_LPIB |\ + (AZX_DCAPS_NO_TCSEL | AZX_DCAPS_POSFIX_LPIB |\ AZX_DCAPS_SNOOP_TYPE(ATI)) /* quirks for ATI/AMD HDMI */ #define AZX_DCAPS_PRESET_ATI_HDMI \ - (AZX_DCAPS_NO_TCSEL | AZX_DCAPS_SYNC_WRITE | AZX_DCAPS_POSFIX_LPIB|\ + (AZX_DCAPS_NO_TCSEL | AZX_DCAPS_POSFIX_LPIB|\ AZX_DCAPS_NO_MSI64) /* quirks for ATI HDMI with snoop off */ @@ -342,7 +340,7 @@ enum { /* quirks for AMD SB */ #define AZX_DCAPS_PRESET_AMD_SB \ - (AZX_DCAPS_NO_TCSEL | AZX_DCAPS_SYNC_WRITE | AZX_DCAPS_AMD_WORKAROUND |\ + (AZX_DCAPS_NO_TCSEL | AZX_DCAPS_AMD_WORKAROUND |\ AZX_DCAPS_SNOOP_TYPE(ATI) | AZX_DCAPS_PM_RUNTIME) /* quirks for Nvidia */ -- 2.25.0

2 6

[PATCH v2] dt-bindings: sound: convert rk3328 codec binding to yaml
by Katsuhiro Suzuki 14 Jul '20

14 Jul '20

This patch converts Rockchip rk3328 audio codec binding to DT schema. And adds description about "mclk" clock and fixes some errors in original example. Signed-off-by: Katsuhiro Suzuki <katsuhiro(a)katsuster.net> --- .../bindings/sound/rockchip,rk3328-codec.txt | 28 -------- .../bindings/sound/rockchip,rk3328-codec.yaml | 70 +++++++++++++++++++ 2 files changed, 70 insertions(+), 28 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.txt create mode 100644 Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.yaml diff --git a/Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.txt b/Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.txt deleted file mode 100644 index 1ecd75d2032a..000000000000 --- a/Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.txt +++ /dev/null @@ -1,28 +0,0 @@ -* Rockchip Rk3328 internal codec - -Required properties: - -- compatible: "rockchip,rk3328-codec" -- reg: physical base address of the controller and length of memory mapped - region. -- rockchip,grf: the phandle of the syscon node for GRF register. -- clocks: a list of phandle + clock-specifer pairs, one for each entry in clock-names. -- clock-names: should be "pclk". -- spk-depop-time-ms: speak depop time msec. - -Optional properties: - -- mute-gpios: GPIO specifier for external line driver control (typically the - dedicated GPIO_MUTE pin) - -Example for rk3328 internal codec: - -codec: codec@ff410000 { - compatible = "rockchip,rk3328-codec"; - reg = <0x0 0xff410000 0x0 0x1000>; - rockchip,grf = <&grf>; - clocks = <&cru PCLK_ACODEC>; - clock-names = "pclk"; - mute-gpios = <&grf_gpio 0 GPIO_ACTIVE_LOW>; - spk-depop-time-ms = 100; -}; diff --git a/Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.yaml b/Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.yaml new file mode 100644 index 000000000000..525b48c2f5de --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.yaml @@ -0,0 +1,70 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/rockchip,rk3328-codec.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Rockchip rk3328 internal codec + +maintainers: + - Heiko Stuebner <heiko(a)sntech.de> + +properties: + compatible: + const: rockchip,rk3328-codec + + reg: + maxItems: 1 + + clocks: + items: + - description: clock for audio codec + - description: clock for I2S master clock + + clock-names: + items: + - const: pclk + - const: mclk + + rockchip,grf: + $ref: /schemas/types.yaml#/definitions/phandle + description: + The phandle of the syscon node for the GRF register. + + spk-depop-time-ms: + $ref: /schemas/types.yaml#/definitions/uint32 + default: 200 + description: + Speaker depop time in msec. + + mute-gpios: + maxItems: 1 + description: + GPIO specifier for external line driver control (typically the + dedicated GPIO_MUTE pin) + + "#sound-dai-cells": + const: 0 + +required: + - compatible + - reg + - clocks + - clock-names + - rockchip,grf + - "#sound-dai-cells" + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + #include <dt-bindings/clock/rk3328-cru.h> + codec: codec@ff410000 { + compatible = "rockchip,rk3328-codec"; + reg = <0xff410000 0x1000>; + clocks = <&cru PCLK_ACODECPHY>, <&cru SCLK_I2S1>; + clock-names = "pclk", "mclk"; + rockchip,grf = <&grf>; + mute-gpios = <&grf_gpio 0 GPIO_ACTIVE_LOW>; + spk-depop-time-ms = <100>; + #sound-dai-cells = <0>; + }; -- 2.27.0

2 2

[PATCH] ASoC: rockchip: spdif: Handle clk by pm runtime
by Sugar Zhang 13 Jul '20

13 Jul '20

This patch handle the clk by pm runtime mechanism to simplify the clk management. Signed-off-by: Sugar Zhang <sugar.zhang(a)rock-chips.com> --- sound/soc/rockchip/rockchip_spdif.c | 59 +++++++++++-------------------------- 1 file changed, 17 insertions(+), 42 deletions(-) diff --git a/sound/soc/rockchip/rockchip_spdif.c b/sound/soc/rockchip/rockchip_spdif.c index 6635145..6748108 100644 --- a/sound/soc/rockchip/rockchip_spdif.c +++ b/sound/soc/rockchip/rockchip_spdif.c @@ -306,44 +306,22 @@ static int rk_spdif_probe(struct platform_device *pdev) return -ENOMEM; spdif->hclk = devm_clk_get(&pdev->dev, "hclk"); - if (IS_ERR(spdif->hclk)) { - dev_err(&pdev->dev, "Can't retrieve rk_spdif bus clock\n"); + if (IS_ERR(spdif->hclk)) return PTR_ERR(spdif->hclk); - } - ret = clk_prepare_enable(spdif->hclk); - if (ret) { - dev_err(spdif->dev, "hclock enable failed %d\n", ret); - return ret; - } spdif->mclk = devm_clk_get(&pdev->dev, "mclk"); - if (IS_ERR(spdif->mclk)) { - dev_err(&pdev->dev, "Can't retrieve rk_spdif master clock\n"); - ret = PTR_ERR(spdif->mclk); - goto err_disable_hclk; - } - - ret = clk_prepare_enable(spdif->mclk); - if (ret) { - dev_err(spdif->dev, "clock enable failed %d\n", ret); - goto err_disable_clocks; - } + if (IS_ERR(spdif->mclk)) + return PTR_ERR(spdif->mclk); res = platform_get_resource(pdev, IORESOURCE_MEM, 0); regs = devm_ioremap_resource(&pdev->dev, res); - if (IS_ERR(regs)) { - ret = PTR_ERR(regs); - goto err_disable_clocks; - } + if (IS_ERR(regs)) + return PTR_ERR(regs); spdif->regmap = devm_regmap_init_mmio_clk(&pdev->dev, "hclk", regs, &rk_spdif_regmap_config); - if (IS_ERR(spdif->regmap)) { - dev_err(&pdev->dev, - "Failed to initialise managed register map\n"); - ret = PTR_ERR(spdif->regmap); - goto err_disable_clocks; - } + if (IS_ERR(spdif->regmap)) + return PTR_ERR(spdif->regmap); spdif->playback_dma_data.addr = res->start + SPDIF_SMPDR; spdif->playback_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; @@ -352,47 +330,44 @@ static int rk_spdif_probe(struct platform_device *pdev) spdif->dev = &pdev->dev; dev_set_drvdata(&pdev->dev, spdif); - pm_runtime_set_active(&pdev->dev); pm_runtime_enable(&pdev->dev); - pm_request_idle(&pdev->dev); + if (!pm_runtime_enabled(&pdev->dev)) { + ret = rk_spdif_runtime_resume(&pdev->dev); + if (ret) + goto err_pm_runtime; + } ret = devm_snd_soc_register_component(&pdev->dev, &rk_spdif_component, &rk_spdif_dai, 1); if (ret) { dev_err(&pdev->dev, "Could not register DAI\n"); - goto err_pm_runtime; + goto err_pm_suspend; } ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); if (ret) { dev_err(&pdev->dev, "Could not register PCM\n"); - goto err_pm_runtime; + goto err_pm_suspend; } return 0; +err_pm_suspend: + if (!pm_runtime_status_suspended(&pdev->dev)) + rk_spdif_runtime_suspend(&pdev->dev); err_pm_runtime: pm_runtime_disable(&pdev->dev); -err_disable_clocks: - clk_disable_unprepare(spdif->mclk); -err_disable_hclk: - clk_disable_unprepare(spdif->hclk); return ret; } static int rk_spdif_remove(struct platform_device *pdev) { - struct rk_spdif_dev *spdif = dev_get_drvdata(&pdev->dev); - pm_runtime_disable(&pdev->dev); if (!pm_runtime_status_suspended(&pdev->dev)) rk_spdif_runtime_suspend(&pdev->dev); - clk_disable_unprepare(spdif->mclk); - clk_disable_unprepare(spdif->hclk); - return 0; } -- 2.7.4

2 1

[PATCH v4] ALSA: line6: add hw monitor volume control for POD HD500
by Vasily Khoruzhick 13 Jul '20

13 Jul '20

Add hw monitor volume control for POD HD500. The same change may work for HD500X but I don't have it to test. Signed-off-by: Vasily Khoruzhick <anarsoul(a)gmail.com> --- v4: add NULL check for kmalloc-ed buffer v3: - use EXPORT_SYMBOL_GPL instead of EXPORT_SYMBOL - use GFP_KERNEL instead of GFP_ATOMIC for allocating a message - use lower case for float lookup table v2: clamp volume value to [0, ARRAY_SIZE() -1] in podhd_set_monitor_level() sound/usb/line6/driver.c | 3 +- sound/usb/line6/driver.h | 4 ++ sound/usb/line6/podhd.c | 127 ++++++++++++++++++++++++++++++++++++++- 3 files changed, 132 insertions(+), 2 deletions(-) diff --git a/sound/usb/line6/driver.c b/sound/usb/line6/driver.c index 7629116f570e..3e07251b80e3 100644 --- a/sound/usb/line6/driver.c +++ b/sound/usb/line6/driver.c @@ -97,7 +97,7 @@ static void line6_stop_listen(struct usb_line6 *line6) /* Send raw message in pieces of wMaxPacketSize bytes. */ -static int line6_send_raw_message(struct usb_line6 *line6, const char *buffer, +int line6_send_raw_message(struct usb_line6 *line6, const char *buffer, int size) { int i, done = 0; @@ -132,6 +132,7 @@ static int line6_send_raw_message(struct usb_line6 *line6, const char *buffer, return done; } +EXPORT_SYMBOL_GPL(line6_send_raw_message); /* Notification of completion of asynchronous request transmission. diff --git a/sound/usb/line6/driver.h b/sound/usb/line6/driver.h index 1a4e3700c80c..62c686bed0ca 100644 --- a/sound/usb/line6/driver.h +++ b/sound/usb/line6/driver.h @@ -108,6 +108,8 @@ enum { LINE6_CAP_CONTROL_MIDI = 1 << 4, /* device provides low-level information */ LINE6_CAP_CONTROL_INFO = 1 << 5, + /* device provides hardware monitoring volume control */ + LINE6_CAP_HWMON_CTL = 1 << 6, }; /* @@ -185,6 +187,8 @@ extern int line6_read_data(struct usb_line6 *line6, unsigned address, void *data, unsigned datalen); extern int line6_read_serial_number(struct usb_line6 *line6, u32 *serial_number); +extern int line6_send_raw_message(struct usb_line6 *line6, + const char *buffer, int size); extern int line6_send_raw_message_async(struct usb_line6 *line6, const char *buffer, int size); extern int line6_send_sysex_message(struct usb_line6 *line6, diff --git a/sound/usb/line6/podhd.c b/sound/usb/line6/podhd.c index e39dc85c355a..1557483ec657 100644 --- a/sound/usb/line6/podhd.c +++ b/sound/usb/line6/podhd.c @@ -11,6 +11,7 @@ #include <linux/slab.h> #include <linux/module.h> #include <sound/core.h> +#include <sound/control.h> #include <sound/pcm.h> #include "driver.h" @@ -37,6 +38,9 @@ struct usb_line6_podhd { /* Firmware version */ int firmware_version; + + /* Monitor level */ + int monitor_level; }; #define line6_to_podhd(x) container_of(x, struct usb_line6_podhd, line6) @@ -250,6 +254,118 @@ static void podhd_disconnect(struct usb_line6 *line6) } } +static const unsigned int float_zero_to_one_lookup[] = { +0x00000000, 0x3c23d70a, 0x3ca3d70a, 0x3cf5c28f, 0x3d23d70a, 0x3d4ccccd, +0x3d75c28f, 0x3d8f5c29, 0x3da3d70a, 0x3db851ec, 0x3dcccccd, 0x3de147ae, +0x3df5c28f, 0x3e051eb8, 0x3e0f5c29, 0x3e19999a, 0x3e23d70a, 0x3e2e147b, +0x3e3851ec, 0x3e428f5c, 0x3e4ccccd, 0x3e570a3d, 0x3e6147ae, 0x3e6b851f, +0x3e75c28f, 0x3e800000, 0x3e851eb8, 0x3e8a3d71, 0x3e8f5c29, 0x3e947ae1, +0x3e99999a, 0x3e9eb852, 0x3ea3d70a, 0x3ea8f5c3, 0x3eae147b, 0x3eb33333, +0x3eb851ec, 0x3ebd70a4, 0x3ec28f5c, 0x3ec7ae14, 0x3ecccccd, 0x3ed1eb85, +0x3ed70a3d, 0x3edc28f6, 0x3ee147ae, 0x3ee66666, 0x3eeb851f, 0x3ef0a3d7, +0x3ef5c28f, 0x3efae148, 0x3f000000, 0x3f028f5c, 0x3f051eb8, 0x3f07ae14, +0x3f0a3d71, 0x3f0ccccd, 0x3f0f5c29, 0x3f11eb85, 0x3f147ae1, 0x3f170a3d, +0x3f19999a, 0x3f1c28f6, 0x3f1eb852, 0x3f2147ae, 0x3f23d70a, 0x3f266666, +0x3f28f5c3, 0x3f2b851f, 0x3f2e147b, 0x3f30a3d7, 0x3f333333, 0x3f35c28f, +0x3f3851ec, 0x3f3ae148, 0x3f3d70a4, 0x3f400000, 0x3f428f5c, 0x3f451eb8, +0x3f47ae14, 0x3f4a3d71, 0x3f4ccccd, 0x3f4f5c29, 0x3f51eb85, 0x3f547ae1, +0x3f570a3d, 0x3f59999a, 0x3f5c28f6, 0x3f5eb852, 0x3f6147ae, 0x3f63d70a, +0x3f666666, 0x3f68f5c3, 0x3f6b851f, 0x3f6e147b, 0x3f70a3d7, 0x3f733333, +0x3f75c28f, 0x3f7851ec, 0x3f7ae148, 0x3f7d70a4, 0x3f800000 +}; + +static void podhd_set_monitor_level(struct usb_line6_podhd *podhd, int value) +{ + unsigned int fl; + static const unsigned char msg[16] = { + /* Chunk is 0xc bytes (without first word) */ + 0x0c, 0x00, + /* First chunk in the message */ + 0x01, 0x00, + /* Message size is 2 4-byte words */ + 0x02, 0x00, + /* Unknown */ + 0x04, 0x41, + /* Unknown */ + 0x04, 0x00, 0x13, 0x00, + /* Volume, LE float32, 0.0 - 1.0 */ + 0x00, 0x00, 0x00, 0x00 + }; + unsigned char *buf; + + buf = kmalloc(sizeof(msg), GFP_KERNEL); + if (!buf) + return; + + memcpy(buf, msg, sizeof(msg)); + + if (value < 0) + value = 0; + + if (value >= ARRAY_SIZE(float_zero_to_one_lookup)) + value = ARRAY_SIZE(float_zero_to_one_lookup) - 1; + + fl = float_zero_to_one_lookup[value]; + + buf[12] = (fl >> 0) & 0xff; + buf[13] = (fl >> 8) & 0xff; + buf[14] = (fl >> 16) & 0xff; + buf[15] = (fl >> 24) & 0xff; + + line6_send_raw_message(&podhd->line6, buf, sizeof(msg)); + kfree(buf); + + podhd->monitor_level = value; +} + +/* control info callback */ +static int snd_podhd_control_monitor_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 100; + uinfo->value.integer.step = 1; + return 0; +} + +/* control get callback */ +static int snd_podhd_control_monitor_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_line6_pcm *line6pcm = snd_kcontrol_chip(kcontrol); + struct usb_line6_podhd *podhd = line6_to_podhd(line6pcm->line6); + + ucontrol->value.integer.value[0] = podhd->monitor_level; + return 0; +} + +/* control put callback */ +static int snd_podhd_control_monitor_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_line6_pcm *line6pcm = snd_kcontrol_chip(kcontrol); + struct usb_line6_podhd *podhd = line6_to_podhd(line6pcm->line6); + + if (ucontrol->value.integer.value[0] == podhd->monitor_level) + return 0; + + podhd_set_monitor_level(podhd, ucontrol->value.integer.value[0]); + return 1; +} + +/* control definition */ +static const struct snd_kcontrol_new podhd_control_monitor = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Monitor Playback Volume", + .index = 0, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_podhd_control_monitor_info, + .get = snd_podhd_control_monitor_get, + .put = snd_podhd_control_monitor_put +}; + /* Try to init POD HD device. */ @@ -298,6 +414,15 @@ static int podhd_init(struct usb_line6 *line6, return err; } + if (pod->line6.properties->capabilities & LINE6_CAP_HWMON_CTL) { + podhd_set_monitor_level(pod, 100); + err = snd_ctl_add(line6->card, + snd_ctl_new1(&podhd_control_monitor, + line6->line6pcm)); + if (err < 0) + return err; + } + if (!(pod->line6.properties->capabilities & LINE6_CAP_CONTROL_INFO)) { /* register USB audio system directly */ return snd_card_register(line6->card); @@ -354,7 +479,7 @@ static const struct line6_properties podhd_properties_table[] = { .id = "PODHD500", .name = "POD HD500", .capabilities = LINE6_CAP_PCM | LINE6_CAP_CONTROL - | LINE6_CAP_HWMON, + | LINE6_CAP_HWMON | LINE6_CAP_HWMON_CTL, .altsetting = 1, .ctrl_if = 1, .ep_ctrl_r = 0x81, -- 2.27.0

2 1

[PATCH] ALSA: echoaudio: re-enable IRQs on failure path
by Dan Carpenter 13 Jul '20

13 Jul '20

This should be spin_unlock_irq() instead of spin_lock(). Fixes: 6c3312544873 ("ALSA: echoaudio: Prevent races in calls to set_audio_format()") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/pci/echoaudio/echoaudio.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index e81f42811f45..6aeb99aa2414 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -721,7 +721,7 @@ static int pcm_prepare(struct snd_pcm_substream *substream) spin_lock_irq(&chip->lock); if (snd_BUG_ON(!is_pipe_allocated(chip, pipe_index))) { - spin_unlock(&chip->lock); + spin_unlock_irq(&chip->lock); return -EINVAL; } -- 2.27.0

2 1

[PATCH] ASoC: rt5682: Add fmw property to get name of mclk
by Akshu Agrawal 13 Jul '20

13 Jul '20

Non-dts based systems can use ACPI DSDT to pass on the mclk. Thus add fmw property mclk-name to get the name of the system clk and link it to rt5682 mclk. Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- include/sound/rt5682.h | 1 + sound/soc/codecs/rt5682.c | 7 ++++++- 2 files changed, 7 insertions(+), 1 deletion(-) diff --git a/include/sound/rt5682.h b/include/sound/rt5682.h index e1f790561ac1..8717c02f89b8 100644 --- a/include/sound/rt5682.h +++ b/include/sound/rt5682.h @@ -42,6 +42,7 @@ struct rt5682_platform_data { unsigned int dmic_delay; const char *dai_clk_names[RT5682_DAI_NUM_CLKS]; + const char *mclk_name; }; #endif diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index de40b6cd16cf..8b223bd0929e 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -2849,7 +2849,10 @@ static int rt5682_probe(struct snd_soc_component *component) } else { #ifdef CONFIG_COMMON_CLK /* Check if MCLK provided */ - rt5682->mclk = devm_clk_get(component->dev, "mclk"); + if (rt5682->pdata.mclk_name) + rt5682->mclk = clk_get(NULL, rt5682->pdata.mclk_name); + if (!rt5682->mclk) + rt5682->mclk = devm_clk_get(component->dev, "mclk"); if (IS_ERR(rt5682->mclk)) { if (PTR_ERR(rt5682->mclk) != -ENOENT) { ret = PTR_ERR(rt5682->mclk); @@ -2976,6 +2979,8 @@ int rt5682_parse_dt(struct rt5682_priv *rt5682, struct device *dev) rt5682->pdata.dai_clk_names[RT5682_DAI_WCLK_IDX], rt5682->pdata.dai_clk_names[RT5682_DAI_BCLK_IDX]); + device_property_read_string(dev, "realtek,mclk-name", &rt5682->pdata.mclk_name); + return 0; } EXPORT_SYMBOL_GPL(rt5682_parse_dt); -- 2.20.1

3 3

[PATCH] ALSA: hda/realtek: Enable headset mic of Acer TravelMate B311R-31 with ALC256
by Jian-Hong Pan 13 Jul '20

13 Jul '20

The Acer TravelMate B311R-31 laptop's audio (1025:1430) with ALC256 cannot detect the headset microphone until ALC256_FIXUP_ACER_MIC_NO_PRESENCE quirk maps the NID 0x19 as the headset mic pin. Signed-off-by: Jian-Hong Pan <jian-hong(a)endlessm.com> --- Re-send for wrong email address sound/pci/hda/patch_realtek.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 194ffa8c66ce..0585ac97faf4 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6152,6 +6152,7 @@ enum { ALC269VC_FIXUP_ACER_VCOPPERBOX_PINS, ALC269VC_FIXUP_ACER_HEADSET_MIC, ALC269VC_FIXUP_ACER_MIC_NO_PRESENCE, + ALC256_FIXUP_ACER_MIC_NO_PRESENCE, }; static const struct hda_fixup alc269_fixups[] = { @@ -7359,6 +7360,15 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MIC }, + [ALC256_FIXUP_ACER_MIC_NO_PRESENCE] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x02a11120 }, /* use as headset mic, without its own jack detect */ + { } + }, + .chained = true, + .chain_id = ALC256_FIXUP_ASUS_HEADSET_MODE + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -7387,6 +7397,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x1308, "Acer Aspire Z24-890", ALC286_FIXUP_ACER_AIO_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x132a, "Acer TravelMate B114-21", ALC233_FIXUP_ACER_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x1330, "Acer TravelMate X514-51T", ALC255_FIXUP_ACER_HEADSET_MIC), + SND_PCI_QUIRK(0x1025, 0x1430, "Acer TravelMate B311R-31", ALC256_FIXUP_ACER_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z), SND_PCI_QUIRK(0x1028, 0x054b, "Dell XPS one 2710", ALC275_FIXUP_DELL_XPS), SND_PCI_QUIRK(0x1028, 0x05bd, "Dell Latitude E6440", ALC292_FIXUP_DELL_E7X), -- 2.27.0

2 1

[PATCH v3] ALSA: line6: add hw monitor volume control for POD HD500
by Vasily Khoruzhick 13 Jul '20

13 Jul '20

Add hw monitor volume control for POD HD500. The same change may work for HD500X but I don't have it to test. Signed-off-by: Vasily Khoruzhick <anarsoul(a)gmail.com> --- v3: - use EXPORT_SYMBOL_GPL instead of EXPORT_SYMBOL - use GFP_KERNEL instead of GFP_ATOMIC for allocating a message - use lower case for float lookup table v2: clamp volume value to [0, ARRAY_SIZE() -1] in podhd_set_monitor_level() sound/usb/line6/driver.c | 3 +- sound/usb/line6/driver.h | 4 ++ sound/usb/line6/podhd.c | 124 ++++++++++++++++++++++++++++++++++++++- 3 files changed, 129 insertions(+), 2 deletions(-) diff --git a/sound/usb/line6/driver.c b/sound/usb/line6/driver.c index 7629116f570e..3e07251b80e3 100644 --- a/sound/usb/line6/driver.c +++ b/sound/usb/line6/driver.c @@ -97,7 +97,7 @@ static void line6_stop_listen(struct usb_line6 *line6) /* Send raw message in pieces of wMaxPacketSize bytes. */ -static int line6_send_raw_message(struct usb_line6 *line6, const char *buffer, +int line6_send_raw_message(struct usb_line6 *line6, const char *buffer, int size) { int i, done = 0; @@ -132,6 +132,7 @@ static int line6_send_raw_message(struct usb_line6 *line6, const char *buffer, return done; } +EXPORT_SYMBOL_GPL(line6_send_raw_message); /* Notification of completion of asynchronous request transmission. diff --git a/sound/usb/line6/driver.h b/sound/usb/line6/driver.h index 1a4e3700c80c..62c686bed0ca 100644 --- a/sound/usb/line6/driver.h +++ b/sound/usb/line6/driver.h @@ -108,6 +108,8 @@ enum { LINE6_CAP_CONTROL_MIDI = 1 << 4, /* device provides low-level information */ LINE6_CAP_CONTROL_INFO = 1 << 5, + /* device provides hardware monitoring volume control */ + LINE6_CAP_HWMON_CTL = 1 << 6, }; /* @@ -185,6 +187,8 @@ extern int line6_read_data(struct usb_line6 *line6, unsigned address, void *data, unsigned datalen); extern int line6_read_serial_number(struct usb_line6 *line6, u32 *serial_number); +extern int line6_send_raw_message(struct usb_line6 *line6, + const char *buffer, int size); extern int line6_send_raw_message_async(struct usb_line6 *line6, const char *buffer, int size); extern int line6_send_sysex_message(struct usb_line6 *line6, diff --git a/sound/usb/line6/podhd.c b/sound/usb/line6/podhd.c index e39dc85c355a..0410d256aa50 100644 --- a/sound/usb/line6/podhd.c +++ b/sound/usb/line6/podhd.c @@ -11,6 +11,7 @@ #include <linux/slab.h> #include <linux/module.h> #include <sound/core.h> +#include <sound/control.h> #include <sound/pcm.h> #include "driver.h" @@ -37,6 +38,9 @@ struct usb_line6_podhd { /* Firmware version */ int firmware_version; + + /* Monitor level */ + int monitor_level; }; #define line6_to_podhd(x) container_of(x, struct usb_line6_podhd, line6) @@ -250,6 +254,115 @@ static void podhd_disconnect(struct usb_line6 *line6) } } +static const unsigned int float_zero_to_one_lookup[] = { +0x00000000, 0x3c23d70a, 0x3ca3d70a, 0x3cf5c28f, 0x3d23d70a, 0x3d4ccccd, +0x3d75c28f, 0x3d8f5c29, 0x3da3d70a, 0x3db851ec, 0x3dcccccd, 0x3de147ae, +0x3df5c28f, 0x3e051eb8, 0x3e0f5c29, 0x3e19999a, 0x3e23d70a, 0x3e2e147b, +0x3e3851ec, 0x3e428f5c, 0x3e4ccccd, 0x3e570a3d, 0x3e6147ae, 0x3e6b851f, +0x3e75c28f, 0x3e800000, 0x3e851eb8, 0x3e8a3d71, 0x3e8f5c29, 0x3e947ae1, +0x3e99999a, 0x3e9eb852, 0x3ea3d70a, 0x3ea8f5c3, 0x3eae147b, 0x3eb33333, +0x3eb851ec, 0x3ebd70a4, 0x3ec28f5c, 0x3ec7ae14, 0x3ecccccd, 0x3ed1eb85, +0x3ed70a3d, 0x3edc28f6, 0x3ee147ae, 0x3ee66666, 0x3eeb851f, 0x3ef0a3d7, +0x3ef5c28f, 0x3efae148, 0x3f000000, 0x3f028f5c, 0x3f051eb8, 0x3f07ae14, +0x3f0a3d71, 0x3f0ccccd, 0x3f0f5c29, 0x3f11eb85, 0x3f147ae1, 0x3f170a3d, +0x3f19999a, 0x3f1c28f6, 0x3f1eb852, 0x3f2147ae, 0x3f23d70a, 0x3f266666, +0x3f28f5c3, 0x3f2b851f, 0x3f2e147b, 0x3f30a3d7, 0x3f333333, 0x3f35c28f, +0x3f3851ec, 0x3f3ae148, 0x3f3d70a4, 0x3f400000, 0x3f428f5c, 0x3f451eb8, +0x3f47ae14, 0x3f4a3d71, 0x3f4ccccd, 0x3f4f5c29, 0x3f51eb85, 0x3f547ae1, +0x3f570a3d, 0x3f59999a, 0x3f5c28f6, 0x3f5eb852, 0x3f6147ae, 0x3f63d70a, +0x3f666666, 0x3f68f5c3, 0x3f6b851f, 0x3f6e147b, 0x3f70a3d7, 0x3f733333, +0x3f75c28f, 0x3f7851ec, 0x3f7ae148, 0x3f7d70a4, 0x3f800000 +}; + +static void podhd_set_monitor_level(struct usb_line6_podhd *podhd, int value) +{ + unsigned int fl; + static const unsigned char msg[16] = { + /* Chunk is 0xc bytes (without first word) */ + 0x0c, 0x00, + /* First chunk in the message */ + 0x01, 0x00, + /* Message size is 2 4-byte words */ + 0x02, 0x00, + /* Unknown */ + 0x04, 0x41, + /* Unknown */ + 0x04, 0x00, 0x13, 0x00, + /* Volume, LE float32, 0.0 - 1.0 */ + 0x00, 0x00, 0x00, 0x00 + }; + unsigned char *buf; + + buf = kmalloc(sizeof(msg), GFP_KERNEL); + memcpy(buf, msg, sizeof(msg)); + + if (value < 0) + value = 0; + + if (value >= ARRAY_SIZE(float_zero_to_one_lookup)) + value = ARRAY_SIZE(float_zero_to_one_lookup) - 1; + + fl = float_zero_to_one_lookup[value]; + + buf[12] = (fl >> 0) & 0xff; + buf[13] = (fl >> 8) & 0xff; + buf[14] = (fl >> 16) & 0xff; + buf[15] = (fl >> 24) & 0xff; + + line6_send_raw_message(&podhd->line6, buf, sizeof(msg)); + kfree(buf); + + podhd->monitor_level = value; +} + +/* control info callback */ +static int snd_podhd_control_monitor_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 100; + uinfo->value.integer.step = 1; + return 0; +} + +/* control get callback */ +static int snd_podhd_control_monitor_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_line6_pcm *line6pcm = snd_kcontrol_chip(kcontrol); + struct usb_line6_podhd *podhd = line6_to_podhd(line6pcm->line6); + + ucontrol->value.integer.value[0] = podhd->monitor_level; + return 0; +} + +/* control put callback */ +static int snd_podhd_control_monitor_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_line6_pcm *line6pcm = snd_kcontrol_chip(kcontrol); + struct usb_line6_podhd *podhd = line6_to_podhd(line6pcm->line6); + + if (ucontrol->value.integer.value[0] == podhd->monitor_level) + return 0; + + podhd_set_monitor_level(podhd, ucontrol->value.integer.value[0]); + return 1; +} + +/* control definition */ +static const struct snd_kcontrol_new podhd_control_monitor = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Monitor Playback Volume", + .index = 0, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_podhd_control_monitor_info, + .get = snd_podhd_control_monitor_get, + .put = snd_podhd_control_monitor_put +}; + /* Try to init POD HD device. */ @@ -298,6 +411,15 @@ static int podhd_init(struct usb_line6 *line6, return err; } + if (pod->line6.properties->capabilities & LINE6_CAP_HWMON_CTL) { + podhd_set_monitor_level(pod, 100); + err = snd_ctl_add(line6->card, + snd_ctl_new1(&podhd_control_monitor, + line6->line6pcm)); + if (err < 0) + return err; + } + if (!(pod->line6.properties->capabilities & LINE6_CAP_CONTROL_INFO)) { /* register USB audio system directly */ return snd_card_register(line6->card); @@ -354,7 +476,7 @@ static const struct line6_properties podhd_properties_table[] = { .id = "PODHD500", .name = "POD HD500", .capabilities = LINE6_CAP_PCM | LINE6_CAP_CONTROL - | LINE6_CAP_HWMON, + | LINE6_CAP_HWMON | LINE6_CAP_HWMON_CTL, .altsetting = 1, .ctrl_if = 1, .ep_ctrl_r = 0x81, -- 2.27.0

2 1

[PATCH] ALSA: hda/realtek: Enable headset mic of Acer TravelMate B311R-31 with ALC256
by Jian-Hong Pan 13 Jul '20

13 Jul '20

The Acer TravelMate B311R-31 laptop's audio (1025:1430) with ALC256 cannot detect the headset microphone until ALC256_FIXUP_ACER_MIC_NO_PRESENCE quirk maps the NID 0x19 as the headset mic pin. Signed-off-by: Jian-Hong Pan <jian-hong(a)endlessm.com> --- sound/pci/hda/patch_realtek.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 194ffa8c66ce..0585ac97faf4 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6152,6 +6152,7 @@ enum { ALC269VC_FIXUP_ACER_VCOPPERBOX_PINS, ALC269VC_FIXUP_ACER_HEADSET_MIC, ALC269VC_FIXUP_ACER_MIC_NO_PRESENCE, + ALC256_FIXUP_ACER_MIC_NO_PRESENCE, }; static const struct hda_fixup alc269_fixups[] = { @@ -7359,6 +7360,15 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MIC }, + [ALC256_FIXUP_ACER_MIC_NO_PRESENCE] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x02a11120 }, /* use as headset mic, without its own jack detect */ + { } + }, + .chained = true, + .chain_id = ALC256_FIXUP_ASUS_HEADSET_MODE + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -7387,6 +7397,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x1308, "Acer Aspire Z24-890", ALC286_FIXUP_ACER_AIO_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x132a, "Acer TravelMate B114-21", ALC233_FIXUP_ACER_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x1330, "Acer TravelMate X514-51T", ALC255_FIXUP_ACER_HEADSET_MIC), + SND_PCI_QUIRK(0x1025, 0x1430, "Acer TravelMate B311R-31", ALC256_FIXUP_ACER_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z), SND_PCI_QUIRK(0x1028, 0x054b, "Dell XPS one 2710", ALC275_FIXUP_DELL_XPS), SND_PCI_QUIRK(0x1028, 0x05bd, "Dell Latitude E6440", ALC292_FIXUP_DELL_E7X), -- 2.27.0

1 0

Beginning 5.8rc1 kernel USB headsets (ASUS ROG Delta and HyperX Cloud Orbit S) play sound as if in slow-motion.
by Mikhail Gavrilov 13 Jul '20

13 Jul '20

Beginning 5.8rc1 (git 69119673bd50) kernel USB headsets (ASUS ROG Delta and HyperX Cloud Orbit S) play sound as if in slow-motion. And in 5.8rc4 (git dcde237b9b0e) this still not fixed yet. The bisecting is problematic because rc1 also has another issue https://lkml.org/lkml/2020/6/22/21 which completely breaks the sound subsystem. If anyone said how to fix https://lkml.org/lkml/2020/6/22/21 I can bisect this issue. -- Best Regards, Mike Gavrilov.

2 2

alsactl do not save all the parameters
by GitHub issues - opened 12 Jul '20

12 Jul '20

alsa-project/alsa-utils issue #46 was opened from alemirandap: Hello everyone, my laptop need to mute the speaker for the full sound experience, when I configure alsamixer with all the levels and mute the speaker I execute "sudo alsactl store" but after rebooted all the levels and configuration are saved except for muting the speaker. I made some tests muting the master or the headphones and the executed "sudo alsactl store" and these are saved. I have to manually in every boot mute the speaker, I 'm under arch linux, with others distros doesn't happen, can you give me a hand with this?, thank you [alsa-info.txt.](https://github.com/alsa-project/alsa-utils/files/4908789/al… Issue URL : https://github.com/alsa-project/alsa-utils/issues/46 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH] ALSA: trident: Fix a memory leak in snd_trident_create
by Navid Emamdoost 11 Jul '20

11 Jul '20

In the implementation of snd_trident_create(), the allocated trident is leaked if snd_trident_mixer() fails. Release via snd_trident_free(). Fixes: 1da177e4c3f4 ("Linux-2.6.12-rc2") Signed-off-by: Navid Emamdoost <navid.emamdoost(a)gmail.com> --- sound/pci/trident/trident_main.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/pci/trident/trident_main.c b/sound/pci/trident/trident_main.c index 6e50376163a2..e98c692f6aa9 100644 --- a/sound/pci/trident/trident_main.c +++ b/sound/pci/trident/trident_main.c @@ -3582,8 +3582,11 @@ int snd_trident_create(struct snd_card *card, return err; } - if ((err = snd_trident_mixer(trident, pcm_spdif_device)) < 0) + err = snd_trident_mixer(trident, pcm_spdif_device); + if (err < 0) { + snd_trident_free(trident); return err; + } /* initialise synth voices */ for (i = 0; i < 64; i++) { -- 2.17.1

2 2

[PATCH v3] ALSA: hda/realtek: enable headset mic of ASUS ROG Zephyrus G14(G401) series with ALC289
by Armas Spann 11 Jul '20

11 Jul '20

Signed-off-by: Armas Spann <zappel(a)retarded.farm> --- Description: This patch adds support for headset mic to the ASUS ROG Zephyrus G14(GA401) notebook series by adding the corresponding vendor/pci_device id, as well as adding a new fixup for the used realtek ALC289. The fixup stets the correct pin to get the headset mic correctly recognized on audio-jack. Changelog: v1: initial v2: added missing vendor and device(pci) id v3: added description sound/pci/hda/patch_realtek.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 194ffa8c66ce..ff4f3eeedfd0 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6152,6 +6152,7 @@ enum { ALC269VC_FIXUP_ACER_VCOPPERBOX_PINS, ALC269VC_FIXUP_ACER_HEADSET_MIC, ALC269VC_FIXUP_ACER_MIC_NO_PRESENCE, + ALC289_FIXUP_ASUS_G401, }; static const struct hda_fixup alc269_fixups[] = { @@ -7359,6 +7360,13 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MIC }, + [ALC289_FIXUP_ASUS_G401] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x03a11020 }, /* headset mic with jack detect */ + { } + }, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -7539,6 +7547,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1bbd, "ASUS Z550MA", ALC255_FIXUP_ASUS_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x1c23, "Asus X55U", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x1043, 0x1ccd, "ASUS X555UB", ALC256_FIXUP_ASUS_MIC), + SND_PCI_QUIRK(0x1043, 0x1f11, "ASUS Zephyrus G14", ALC289_FIXUP_ASUS_G401), SND_PCI_QUIRK(0x1043, 0x3030, "ASUS ZN270IE", ALC256_FIXUP_ASUS_AIO_GPIO2), SND_PCI_QUIRK(0x1043, 0x831a, "ASUS P901", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x834a, "ASUS S101", ALC269_FIXUP_STEREO_DMIC), -- 2.27.0

2 2

[PATCH v2] ALSA: hda/realtek: enable headset mic of ASUS ROG Zephyrus G14(G401) series with ALC289
by Armas Spann 11 Jul '20

11 Jul '20

Signed-off-by: Armas Spann <zappel(a)retarded.farm> --- sound/pci/hda/patch_realtek.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 194ffa8c66ce..ff4f3eeedfd0 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6152,6 +6152,7 @@ enum { ALC269VC_FIXUP_ACER_VCOPPERBOX_PINS, ALC269VC_FIXUP_ACER_HEADSET_MIC, ALC269VC_FIXUP_ACER_MIC_NO_PRESENCE, + ALC289_FIXUP_ASUS_G401, }; static const struct hda_fixup alc269_fixups[] = { @@ -7359,6 +7360,13 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MIC }, + [ALC289_FIXUP_ASUS_G401] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x03a11020 }, /* headset mic with jack detect */ + { } + }, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -7539,6 +7547,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1bbd, "ASUS Z550MA", ALC255_FIXUP_ASUS_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x1c23, "Asus X55U", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x1043, 0x1ccd, "ASUS X555UB", ALC256_FIXUP_ASUS_MIC), + SND_PCI_QUIRK(0x1043, 0x1f11, "ASUS Zephyrus G14", ALC289_FIXUP_ASUS_G401), SND_PCI_QUIRK(0x1043, 0x3030, "ASUS ZN270IE", ALC256_FIXUP_ASUS_AIO_GPIO2), SND_PCI_QUIRK(0x1043, 0x831a, "ASUS P901", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x834a, "ASUS S101", ALC269_FIXUP_STEREO_DMIC), -- 2.27.0

2 4

[PATCH v1] ALSA: hda/realtek: enable headset mic of ASUS ROG Zephyrus G14(G401) series with ALC289
by Armas Spann 11 Jul '20

11 Jul '20

Signed-off-by: Armas Spann <zappel(a)retarded.farm> --- ALSA: hda/realtek: enable headset mic of ASUS ROG Zephyrus G14(G401) series with ALC289 sound/pci/hda/patch_realtek.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 194ffa8c66ce..2e7d4ebf8ce7 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6152,6 +6152,7 @@ enum { ALC269VC_FIXUP_ACER_VCOPPERBOX_PINS, ALC269VC_FIXUP_ACER_HEADSET_MIC, ALC269VC_FIXUP_ACER_MIC_NO_PRESENCE, + ALC289_FIXUP_ASUS_G401, }; static const struct hda_fixup alc269_fixups[] = { @@ -7359,6 +7360,13 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MIC }, + [ALC289_FIXUP_ASUS_G401] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x03a11020 }, /* headset mic with jack detect */ + { } + }, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { -- 2.27.0

1 0

Change model for ASUS platform
by Kailang 11 Jul '20

11 Jul '20

Hi Takashi, ASUS platform couldn't need to use Headset Mode model. Attach patch will change it. BR, Kailang

2 1

[PATCH] ALSA: usb-audio: Fix race against the error recovery URB submission
by Takashi Iwai 10 Jul '20

10 Jul '20

USB MIDI driver has an error recovery mechanism to resubmit the URB in the delayed timer handler, and this may race with the standard start / stop operations. Although both start and stop operations themselves don't race with each other due to the umidi->mutex protection, but this isn't applied to the timer handler. For fixing this potential race, the following changes are applied: - Since the timer handler can't use the mutex, we apply the umidi->disc_lock protection at each input stream URB submission; this also needs to change the GFP flag to GFP_ATOMIC - Add a check of the URB refcount and skip if already submitted - Move the timer cancel call at disconnection to the beginning of the procedure; this assures the in-flight timer handler is gone properly before killing all pending URBs Reported-by: syzbot+0f4ecfe6a2c322c81728(a)syzkaller.appspotmail.com Reported-by: syzbot+5f1d24c49c1d2c427497(a)syzkaller.appspotmail.com Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/midi.c | 17 ++++++++++++----- 1 file changed, 12 insertions(+), 5 deletions(-) diff --git a/sound/usb/midi.c b/sound/usb/midi.c index 047b90595d65..354f57692938 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -1499,6 +1499,8 @@ void snd_usbmidi_disconnect(struct list_head *p) spin_unlock_irq(&umidi->disc_lock); up_write(&umidi->disc_rwsem); + del_timer_sync(&umidi->error_timer); + for (i = 0; i < MIDI_MAX_ENDPOINTS; ++i) { struct snd_usb_midi_endpoint *ep = &umidi->endpoints[i]; if (ep->out) @@ -1525,7 +1527,6 @@ void snd_usbmidi_disconnect(struct list_head *p) ep->in = NULL; } } - del_timer_sync(&umidi->error_timer); } EXPORT_SYMBOL(snd_usbmidi_disconnect); @@ -2301,16 +2302,22 @@ void snd_usbmidi_input_stop(struct list_head *p) } EXPORT_SYMBOL(snd_usbmidi_input_stop); -static void snd_usbmidi_input_start_ep(struct snd_usb_midi_in_endpoint *ep) +static void snd_usbmidi_input_start_ep(struct snd_usb_midi *umidi, + struct snd_usb_midi_in_endpoint *ep) { unsigned int i; + unsigned long flags; if (!ep) return; for (i = 0; i < INPUT_URBS; ++i) { struct urb *urb = ep->urbs[i]; - urb->dev = ep->umidi->dev; - snd_usbmidi_submit_urb(urb, GFP_KERNEL); + spin_lock_irqsave(&umidi->disc_lock, flags); + if (!atomic_read(&urb->use_count)) { + urb->dev = ep->umidi->dev; + snd_usbmidi_submit_urb(urb, GFP_ATOMIC); + } + spin_unlock_irqrestore(&umidi->disc_lock, flags); } } @@ -2326,7 +2333,7 @@ void snd_usbmidi_input_start(struct list_head *p) if (umidi->input_running || !umidi->opened[1]) return; for (i = 0; i < MIDI_MAX_ENDPOINTS; ++i) - snd_usbmidi_input_start_ep(umidi->endpoints[i].in); + snd_usbmidi_input_start_ep(umidi, umidi->endpoints[i].in); umidi->input_running = 1; } EXPORT_SYMBOL(snd_usbmidi_input_start); -- 2.16.4

1 0

[PATCH 0/6] ASoC: mediatek: mt8183: support DP audio
by Tzung-Bi Shih 10 Jul '20

10 Jul '20

This series is a follow up for a long time ago series (https://patchwork.kernel.org/cover/11204303/). The old series bound too much on the patches of DRM bridge and ASoC machine driver. And unluckily, the dependencies (https://lore.kernel.org/patchwork/patch/1126819/) have not applied. Revewing the ASoC patches in the old series, I found that they could be decoupled from the DRM bridge patches. And they are harmless as it is an optional attribute ("hdmi-codec") in DTS. This series arranges and rebases the harmless ASoC patches for mt8183-mt6358-ts3a227-max98357 and mt8183-da7219-max98357. The 1st and 4th patch add an optional DT property. The 1st patch was acked long time ago (https://patchwork.kernel.org/patch/11204321/). The 2nd and 5th patch add DAI link for using hdmi-codec. The 3rd and 6th patch support the HDMI jack reporting. Tzung-Bi Shih (6): ASoC: dt-bindings: mt8183: add a property "mediatek,hdmi-codec" ASoC: mediatek: mt8183: use hdmi-codec ASoC: mediatek: mt8183: support HDMI jack reporting ASoC: dt-bindings: mt8183-da7219: add a property "mediatek,hdmi-codec" ASoC: mediatek: mt8183-da7219: use hdmi-codec ASoC: mediatek: mt8183-da7219: support HDMI jack reporting .../bindings/sound/mt8183-da7219-max98357.txt | 4 +++ .../sound/mt8183-mt6358-ts3a227-max98357.txt | 2 ++ sound/soc/mediatek/Kconfig | 2 ++ .../mediatek/mt8183/mt8183-da7219-max98357.c | 29 +++++++++++++++++-- .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 29 +++++++++++++++++-- 5 files changed, 60 insertions(+), 6 deletions(-) -- 2.27.0.383.g050319c2ae-goog

2 7

[PATCH 0/2] ASoC: fsl_spdif: Clear the validity bit for TX
by Shengjiu Wang 10 Jul '20

10 Jul '20

Clear the validity bit for TX Add kctl for configuring TX validity bit Shengjiu Wang (2): ASoC: fsl_spdif: Clear the validity bit for TX ASoC: fsl_spdif: Add kctl for configuring TX validity bit sound/soc/fsl/fsl_spdif.c | 51 ++++++++++++++++++++++++++++++++++++--- 1 file changed, 47 insertions(+), 4 deletions(-) -- 2.21.0

2 3

[PATCH] ASoC: Intel: cht_bsw_rt5672: Improve dai-set-fmt comment in cht_codec_fixup()
by Hans de Goede 10 Jul '20

10 Jul '20

As Pierre-Louis Bossart pointed out, saying that the default mode for the SSP is TDM 4 slot is not entirely accurate. There really are 2 default modes: The default mode for the SSP configuration is TDM 4 slot for the cpu-dai (hard-coded in DSP firmware), The default mode for the SSP configuration is I2S for the codec-dai (hard-coded in the 'SSP2-Codec" .dai_fmt masks, so far unused). This commit updates the comment in cht_codec_fixup() to properly reflect this. Suggested-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/soc/intel/boards/cht_bsw_rt5672.c | 18 +++++++++++------- 1 file changed, 11 insertions(+), 7 deletions(-) diff --git a/sound/soc/intel/boards/cht_bsw_rt5672.c b/sound/soc/intel/boards/cht_bsw_rt5672.c index 1470c3de7895..1812b786d33b 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5672.c +++ b/sound/soc/intel/boards/cht_bsw_rt5672.c @@ -253,13 +253,17 @@ static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd, params_set_format(params, SNDRV_PCM_FORMAT_S24_LE); /* - * Default mode for SSP configuration is TDM 4 slot. One board/design, - * the Lenovo Miix 2 10 uses not 1 but 2 codecs connected to SSP2. The - * second piggy-backed, output-only codec is inside the keyboard-dock - * (which has extra speakers). Unlike the main rt5672 codec, we cannot - * configure this codec, it is hard coded to use 2 channel 24 bit I2S. - * Since we only support 2 channels anyways, there is no need for TDM - * on any cht-bsw-rt5672 designs. So we simply use I2S 2ch everywhere. + * The default mode for the cpu-dai is TDM 4 slot. The default mode + * for the codec-dai is I2S. So we need to either set the cpu-dai to + * I2S mode to match the codec-dai, or set the codec-dai to TDM 4 slot + * (or program both to yet another mode). + * One board, the Lenovo Miix 2 10, uses not 1 but 2 codecs connected + * to SSP2. The second piggy-backed, output-only codec is inside the + * keyboard-dock (which has extra speakers). Unlike the main rt5672 + * codec, we cannot configure this codec, it is hard coded to use + * 2 channel 24 bit I2S. For this to work we must use I2S mode on this + * board. Since we only support 2 channels anyways, there is no need + * for TDM on any cht-bsw-rt5672 designs. So we use I2S 2ch everywhere. */ ret = snd_soc_dai_set_fmt(asoc_rtd_to_cpu(rtd, 0), SND_SOC_DAIFMT_I2S | -- 2.26.2

2 1

[PATCH v4 00/10] ASoC: Clean-up W=1 build warnings - part2
by Pierre-Louis Bossart 10 Jul '20

10 Jul '20

Both Lee Jones and I submitted separate series, this is the second part of the merged result, for which no feedback was provided. I picked Lee's patches for rt5659 and ak4458 and added the pxa and ux500 that I didn't fix. The rest is largely identical between our respective series, with the exception of the sunxi which I documented and Lee removed. I don't have any specific preference and will go with the flow on this. Changes since v3: Improved commit subjects from 'fix kernel-doc' as suggested by Lee Jones. In a couple of cases I just reverted to Lee's patches when the code was identical. Added a couple of CC: tags from Lee's patches. Added Arnaud Pouliquen's Acked-by tag in first patch. Lee Jones (6): ASoC: sunxi: sun4i-spdif: Fix misspelling of 'reg_dac_txdata' in kernel-doc ASoC: pxa: pxa-ssp: Demote seemingly unintentional kerneldoc header ASoC: ux500: ux500_msp_i2s: Remove unused variables 'reg_val_DR' and 'reg_val_TSTDR' ASoC: codecs: rt5659: Remove many unused const variables ASoC: codecs: tlv320aic26: Demote seemingly unintentional kerneldoc header ASoC: codecs: ak4458: Remove set but never checked variable 'ret' Pierre-Louis Bossart (4): ASoC: sti: uniperif: fix 'defined by not used' warning ASoC: qcom: qdsp6: q6asm: Provide documentation for 'codec_profile' ASoC: sunxi: sun4i-i2s: add missing clock and format arguments in kernel-doc ASoC: codecs: rt5631: fix spurious kernel-doc start and missing arguments sound/soc/codecs/ak4458.c | 6 +++--- sound/soc/codecs/rt5631.c | 8 +++++-- sound/soc/codecs/rt5659.c | 37 --------------------------------- sound/soc/codecs/tlv320aic26.c | 2 +- sound/soc/pxa/pxa-ssp.c | 2 +- sound/soc/qcom/qdsp6/q6asm.c | 2 +- sound/soc/sti/uniperif.h | 2 +- sound/soc/sunxi/sun4i-i2s.c | 10 ++++++++- sound/soc/sunxi/sun4i-spdif.c | 2 +- sound/soc/ux500/ux500_msp_i2s.c | 8 +++---- 10 files changed, 27 insertions(+), 52 deletions(-) base-commit: 6940701c715e193282cf8b31f970b2ea6eb26341 -- 2.25.1

2 11