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[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

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[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

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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

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[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

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[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

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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

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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

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[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

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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

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/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…

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