October 2020 - Alsa-devel - mailman.alsa-project.org

[PATCH] dt-bindings: sound: sun8i-a33-codec: Add Allwinner A64 codec compatible fallback
by Clément Péron 12 Oct '20

12 Oct '20

make dtbs_check report a warning because the documentation for the A64 codec compatible is missing. The A64 codec compatible is actually a simple fallback to the A33. Reflect this in the dt-bindings Documentation. Signed-off-by: Clément Péron <peron.clem(a)gmail.com> --- .../bindings/sound/allwinner,sun8i-a33-codec.yaml | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/allwinner,sun8i-a33-codec.yaml b/Documentation/devicetree/bindings/sound/allwinner,sun8i-a33-codec.yaml index 55d28268d2f4..e78a22496e7c 100644 --- a/Documentation/devicetree/bindings/sound/allwinner,sun8i-a33-codec.yaml +++ b/Documentation/devicetree/bindings/sound/allwinner,sun8i-a33-codec.yaml @@ -15,7 +15,11 @@ properties: const: 0 compatible: - const: allwinner,sun8i-a33-codec + oneOf: + - const: allwinner,sun8i-a33-codec + - items: + - const: allwinner,sun50i-a64-codec + - const: allwinner,sun8i-a33-codec reg: maxItems: 1 -- 2.25.1

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Can't aplay/arecord simultaneously inside and outside of linux container
by GitHub issues - opened 12 Oct '20

12 Oct '20

alsa-project/alsa-utils issue #62 was opened from fearless-fan: Hi, I run into a problem that can't aplay/arecord simultaneously inside and outside of linux container. Steps to reproduce: 1. go into container's shell and exec a record command. ~ # lxc-attach -n Default -P /tmp/run/lxc / # arecord -d 60 /tmp/test.wav & / # Recording WAVE '/tmp/test.wav' : Unsigned 8 bit, Rate 8000 Hz, Mono / # ps PID USER TIME COMMAND 1 root 0:00 /sbin/init 3 root 0:00 /bin/sh 1519 root 0:00 arecord -d 60 /tmp/test.wav 1520 root 0:00 /bin/ash 1521 root 0:00 ps 2. exit and exec another record command outside of container / # exit ~ # arecord -d 60 /tmp/test.wav & ~ # ALSA lib pcm_dsnoop.c:641:(snd_pcm_dsnoop_open) unable to open slave arecord: main:830: audio open error: Device or resource busy [1]+ Done(1) arecord -d 60 /tmp/test.wav 3. playback has same problem as below ALSA lib pcm_dmix.c:1089:(snd_pcm_dmix_open) unable to open slave aplay: main:830: audio open error: Device or resource busy Expect: It should be allowed to play/record simultaneously inside and outside of linux container. Here's my asound.conf: ~ # cat etc/asound.conf pcm.asymed { type asym playback.pcm "dmix:0,0" capture.pcm "dsnoop:0,0" } pcm.!default { type plug slave.pcm "asymed" } ctl.!default { type hw card 0 } Is there anyone has experience of how to address it? Thank you. Issue URL : https://github.com/alsa-project/alsa-utils/issues/62 Repository URL: https://github.com/alsa-project/alsa-utils

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[PATCH] ASoC: Intel: Do not load legacy SST driver on BYT when SND_SOC_SOF_BAYTRAIL is enabled
by Hans de Goede 12 Oct '20

12 Oct '20

The legacy 80860F28 / sst_acpi_baytrail_desc match in sst_acpi_match is already conditional on the the newer SND_SST_IPC_ACPI driver not being enabled. But now that we have an even newer driver in the form of SOF support for BYT devices, we also need to take this into account, so we also must not include the sst_acpi_baytrail_desc match when SND_SOC_SOF_BAYTRAIL is enabled. This fixes snd-soc-sst-acpi binding to the 80860F28 platform device, blocking snd-sof-acpi from binding, which breaks audio support. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/soc/intel/common/sst-acpi.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/common/sst-acpi.c b/sound/soc/intel/common/sst-acpi.c index 5854868650b9..b4b2e1e0e845 100644 --- a/sound/soc/intel/common/sst-acpi.c +++ b/sound/soc/intel/common/sst-acpi.c @@ -198,7 +198,7 @@ static struct sst_acpi_desc sst_acpi_broadwell_desc = { .dma_size = SST_LPT_DSP_DMA_SIZE, }; -#if !IS_ENABLED(CONFIG_SND_SST_IPC_ACPI) +#if !IS_ENABLED(CONFIG_SND_SST_IPC_ACPI) && !IS_ENABLED(CONFIG_SND_SOC_SOF_BAYTRAIL) static struct sst_acpi_desc sst_acpi_baytrail_desc = { .drv_name = "baytrail-pcm-audio", .machines = snd_soc_acpi_intel_baytrail_legacy_machines, @@ -214,7 +214,7 @@ static struct sst_acpi_desc sst_acpi_baytrail_desc = { static const struct acpi_device_id sst_acpi_match[] = { { "INT33C8", (unsigned long)&sst_acpi_haswell_desc }, { "INT3438", (unsigned long)&sst_acpi_broadwell_desc }, -#if !IS_ENABLED(CONFIG_SND_SST_IPC_ACPI) +#if !IS_ENABLED(CONFIG_SND_SST_IPC_ACPI) && !IS_ENABLED(CONFIG_SND_SOC_SOF_BAYTRAIL) { "80860F28", (unsigned long)&sst_acpi_baytrail_desc }, #endif { } -- 2.28.0

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[PATCH 0/5] ASoC: Mediatek: Add support for MT8192 SoC
by Jiaxin Yu 12 Oct '20

12 Oct '20

This series of patches adds support for Mediatek AFE for MT8192 SoC. At the same time, the calibration function of MT6359 is completed with real machine driver. The patch is based on broonie tree "for-next" branch. Jiaxin Yu (5): ASoC: mediatek: mt6359: add the calibration functions ASoC: mediatek: mt8192: add platform driver dt-bindings: mediatek: mt8192: add audio afe document ASoC: mediatek: mt8192: add machine driver with mt6359, rt1015 and rt5682 dt-bindings: mediatek: mt8192: add mt8192-mt6358-rt1015-rt5682 document .../bindings/sound/mt8192-afe-pcm.yaml | 98 + .../sound/mt8192-mt6359-rt1015-rt5682.yaml | 44 + sound/soc/codecs/mt6359.c | 110 + sound/soc/codecs/mt6359.h | 7 + sound/soc/mediatek/Kconfig | 23 + sound/soc/mediatek/Makefile | 1 + sound/soc/mediatek/common/mtk-afe-fe-dai.c | 13 +- sound/soc/mediatek/common/mtk-base-afe.h | 1 + sound/soc/mediatek/mt8192/Makefile | 16 + sound/soc/mediatek/mt8192/mt8192-afe-clk.c | 669 ++++ sound/soc/mediatek/mt8192/mt8192-afe-clk.h | 244 ++ sound/soc/mediatek/mt8192/mt8192-afe-common.h | 170 + .../soc/mediatek/mt8192/mt8192-afe-control.c | 163 + sound/soc/mediatek/mt8192/mt8192-afe-gpio.c | 306 ++ sound/soc/mediatek/mt8192/mt8192-afe-gpio.h | 19 + sound/soc/mediatek/mt8192/mt8192-afe-pcm.c | 2389 +++++++++++++ sound/soc/mediatek/mt8192/mt8192-dai-adda.c | 1489 ++++++++ sound/soc/mediatek/mt8192/mt8192-dai-i2s.c | 2139 +++++++++++ sound/soc/mediatek/mt8192/mt8192-dai-pcm.c | 409 +++ sound/soc/mediatek/mt8192/mt8192-dai-tdm.c | 778 ++++ .../mediatek/mt8192/mt8192-interconnection.h | 65 + .../mt8192/mt8192-mt6359-rt1015-rt5682.c | 1058 ++++++ sound/soc/mediatek/mt8192/mt8192-reg.h | 3131 +++++++++++++++++ 23 files changed, 13338 insertions(+), 4 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/mt8192-afe-pcm.yaml create mode 100644 Documentation/devicetree/bindings/sound/mt8192-mt6359-rt1015-rt5682.yaml create mode 100644 sound/soc/mediatek/mt8192/Makefile create mode 100644 sound/soc/mediatek/mt8192/mt8192-afe-clk.c create mode 100644 sound/soc/mediatek/mt8192/mt8192-afe-clk.h create mode 100644 sound/soc/mediatek/mt8192/mt8192-afe-common.h create mode 100644 sound/soc/mediatek/mt8192/mt8192-afe-control.c create mode 100644 sound/soc/mediatek/mt8192/mt8192-afe-gpio.c create mode 100644 sound/soc/mediatek/mt8192/mt8192-afe-gpio.h create mode 100644 sound/soc/mediatek/mt8192/mt8192-afe-pcm.c create mode 100644 sound/soc/mediatek/mt8192/mt8192-dai-adda.c create mode 100644 sound/soc/mediatek/mt8192/mt8192-dai-i2s.c create mode 100644 sound/soc/mediatek/mt8192/mt8192-dai-pcm.c create mode 100644 sound/soc/mediatek/mt8192/mt8192-dai-tdm.c create mode 100644 sound/soc/mediatek/mt8192/mt8192-interconnection.h create mode 100644 sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c create mode 100644 sound/soc/mediatek/mt8192/mt8192-reg.h -- 2.18.0

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Re: [PATCH 07/10] ALSA: firewire-lib: code refactoring for data block calculation
by MICHAEL Mosier 11 Oct '20

11 Oct '20

👋 hello I see our work has brought us together once again

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Buggy hardware mixing
by Kevin Maciel 11 Oct '20

11 Oct '20

Hi all, I was referred here from the #alsa irc channel on freenode. I've been having issues with hardware mixing in ALSA since I started using Linux at least 10 years ago. Sometimes when I am trying to play multiple audio sources at once, one source will stop producing sound or all sound will all of a sudden turn into a mash of loud, jumbled noise. This has occurred across every combination of hardware and distribution of Linux I have used for my personal computers. I was trying to set up a custom dmix configuration to work around the problem, but I could not get multiple audio sources to play with my custom dmix definition. It seems to work if I use dmix:CARD=xxx, but I would like to set my own period_size and other such settings. I was told it is a better idea to try to figure out what the problem is with hardware mixing than to try to work around it with dmix. So here we are. I always thought this was a known problem and that hardware mixing by default was considered "new" or experimental. Thanks

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snd_pcm_status() does not update status->avail
by Jonas Holmberg 09 Oct '20

09 Oct '20

Hi, I have a problem with status->avail not being updated when using multiple extplug plugins and softvol. I managed to get it to work with the following patch: diff --git a/src/pcm/pcm_plugin.c b/src/pcm/pcm_plugin.c index ea60eb98..89e819d6 100644 --- a/src/pcm/pcm_plugin.c +++ b/src/pcm/pcm_plugin.c @@ -551,6 +551,8 @@ static int snd_pcm_plugin_status(snd_pcm_t *pcm, snd_pcm_status_t * status) return err; status->appl_ptr = *pcm->appl.ptr; status->hw_ptr = *pcm->hw.ptr; + status->avail = snd_pcm_mmap_capture_avail(pcm); + status->delay = snd_pcm_mmap_capture_delay(pcm); return 0; } My question is if this is the correct solution? BR /Jonas

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[PATCH 1/1] ALSA: hda: fix jack detection with Realtek codecs when in D3
by Kai Vehmanen 09 Oct '20

09 Oct '20

In case HDA controller is active, but codec is runtime suspended, jack detection is not successful and no interrupt is raised. This has been observed with multiple Realtek codecs and HDA controllers from different vendors. Bug does not occur if both codec and controller are active, or both are in suspend. Bug can be easily hit on desktop systems with no built-in speaker. The problem can be fixed by powering up the codec once after every controller runtime resume. Even if codec goes back to suspend, the jack detection will now work. Add a flag to 'hda_codec' to describe codecs that require this flow from the controller driver. Mark all Realtek codecs with this flag. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=209379 Cc: Kailang Yang <kailang(a)realtek.com> Co-developed-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- include/sound/hda_codec.h | 1 + sound/pci/hda/hda_intel.c | 8 ++++++-- sound/pci/hda/patch_realtek.c | 6 ++++++ 3 files changed, 13 insertions(+), 2 deletions(-) diff --git a/include/sound/hda_codec.h b/include/sound/hda_codec.h index 0fea49bfc5e8..73827b7d17e0 100644 --- a/include/sound/hda_codec.h +++ b/include/sound/hda_codec.h @@ -253,6 +253,7 @@ struct hda_codec { unsigned int force_pin_prefix:1; /* Add location prefix */ unsigned int link_down_at_suspend:1; /* link down at runtime suspend */ unsigned int relaxed_resume:1; /* don't resume forcibly for jack */ + unsigned int forced_resume:1; /* forced resume for jack */ unsigned int mst_no_extra_pcms:1; /* no backup PCMs for DP-MST */ #ifdef CONFIG_PM diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 61e495187b1a..cfc073c992e7 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1002,12 +1002,16 @@ static void __azx_runtime_resume(struct azx *chip, bool from_rt) azx_init_pci(chip); hda_intel_init_chip(chip, true); - if (status && from_rt) { - list_for_each_codec(codec, &chip->bus) + if (from_rt) { + list_for_each_codec(codec, &chip->bus) { + if (codec->forced_resume && pm_runtime_suspended(hda_codec_dev(codec))) + pm_request_resume(hda_codec_dev(codec)); + if (!codec->relaxed_resume && (status & (1 << codec->addr))) schedule_delayed_work(&codec->jackpoll_work, codec->jackpoll_interval); + } } /* power down again for link-controlled chips */ diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f89506dffd5b..e4ab483db72f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -849,6 +849,12 @@ static int alc_build_controls(struct hda_codec *codec) static void alc_pre_init(struct hda_codec *codec) { alc_fill_eapd_coef(codec); + + /* + * if controller is resumed from suspend, while codec remains in D3, codec + * needs to be woken up once or otherwise jack detection does not work + */ + codec->forced_resume = 1; } #define is_s3_resume(codec) \ -- 2.28.0

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[PATCH] ASoC: mchp-spdiftx: remove 'TX' from playback stream name
by Codrin Ciubotariu 09 Oct '20

09 Oct '20

Do not include the 'TX' in the stream name since it's obvious for playback. Signed-off-by: Codrin Ciubotariu <codrin.ciubotariu(a)microchip.com> --- sound/soc/atmel/mchp-spdiftx.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/atmel/mchp-spdiftx.c b/sound/soc/atmel/mchp-spdiftx.c index 36c23eb3a5ad..82c1eecd2528 100644 --- a/sound/soc/atmel/mchp-spdiftx.c +++ b/sound/soc/atmel/mchp-spdiftx.c @@ -744,7 +744,7 @@ static struct snd_soc_dai_driver mchp_spdiftx_dai = { .probe = mchp_spdiftx_dai_probe, .remove = mchp_spdiftx_dai_remove, .playback = { - .stream_name = "S/PDIF TX Playback", + .stream_name = "S/PDIF Playback", .channels_min = 1, .channels_max = 2, .rates = MCHP_SPDIFTX_RATES, -- 2.25.1

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[PATCH] ALSA: hda/i915 - fix list corruption with concurrent probes
by Kai Vehmanen 09 Oct '20

09 Oct '20

From: Takashi Iwai <tiwai(a)suse.de> Current hdac_i915 uses a static completion instance to wait for i915 driver to complete the component bind. This design is not safe if multiple HDA controllers are active and communicating with different i915 instances, and can lead to list corruption and failed audio driver probe. Fix the design by moving completion mechanism to common acomp code and remove the related code from hdac_i915. Co-developed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/drm/drm_audio_component.h | 4 ++++ sound/hda/hdac_component.c | 3 +++ sound/hda/hdac_i915.c | 23 +++-------------------- 3 files changed, 10 insertions(+), 20 deletions(-) diff --git a/include/drm/drm_audio_component.h b/include/drm/drm_audio_component.h index a45f93487039..0d36bfd1a4cd 100644 --- a/include/drm/drm_audio_component.h +++ b/include/drm/drm_audio_component.h @@ -117,6 +117,10 @@ struct drm_audio_component { * @audio_ops: Ops implemented by hda driver, called by DRM driver */ const struct drm_audio_component_audio_ops *audio_ops; + /** + * @master_bind_complete: completion held during component master binding + */ + struct completion master_bind_complete; }; #endif /* _DRM_AUDIO_COMPONENT_H_ */ diff --git a/sound/hda/hdac_component.c b/sound/hda/hdac_component.c index 89126c6fd216..bb37e7e0bd79 100644 --- a/sound/hda/hdac_component.c +++ b/sound/hda/hdac_component.c @@ -210,12 +210,14 @@ static int hdac_component_master_bind(struct device *dev) goto module_put; } + complete_all(&acomp->master_bind_complete); return 0; module_put: module_put(acomp->ops->owner); out_unbind: component_unbind_all(dev, acomp); + complete_all(&acomp->master_bind_complete); return ret; } @@ -296,6 +298,7 @@ int snd_hdac_acomp_init(struct hdac_bus *bus, if (!acomp) return -ENOMEM; acomp->audio_ops = aops; + init_completion(&acomp->master_bind_complete); bus->audio_component = acomp; devres_add(dev, acomp); diff --git a/sound/hda/hdac_i915.c b/sound/hda/hdac_i915.c index 5f0a1aa6ad84..454474ac5716 100644 --- a/sound/hda/hdac_i915.c +++ b/sound/hda/hdac_i915.c @@ -11,8 +11,6 @@ #include <sound/hda_i915.h> #include <sound/hda_register.h> -static struct completion bind_complete; - #define IS_HSW_CONTROLLER(pci) (((pci)->device == 0x0a0c) || \ ((pci)->device == 0x0c0c) || \ ((pci)->device == 0x0d0c) || \ @@ -130,19 +128,6 @@ static bool i915_gfx_present(void) return pci_dev_present(ids); } -static int i915_master_bind(struct device *dev, - struct drm_audio_component *acomp) -{ - complete_all(&bind_complete); - /* clear audio_ops here as it was needed only for completion call */ - acomp->audio_ops = NULL; - return 0; -} - -static const struct drm_audio_component_audio_ops i915_init_ops = { - .master_bind = i915_master_bind -}; - /** * snd_hdac_i915_init - Initialize i915 audio component * @bus: HDA core bus @@ -163,9 +148,7 @@ int snd_hdac_i915_init(struct hdac_bus *bus) if (!i915_gfx_present()) return -ENODEV; - init_completion(&bind_complete); - - err = snd_hdac_acomp_init(bus, &i915_init_ops, + err = snd_hdac_acomp_init(bus, NULL, i915_component_master_match, sizeof(struct i915_audio_component) - sizeof(*acomp)); if (err < 0) @@ -177,8 +160,8 @@ int snd_hdac_i915_init(struct hdac_bus *bus) if (!IS_ENABLED(CONFIG_MODULES) || !request_module("i915")) { /* 60s timeout */ - wait_for_completion_timeout(&bind_complete, - msecs_to_jiffies(60 * 1000)); + wait_for_completion_timeout(&acomp->master_bind_complete, + msecs_to_jiffies(60 * 1000)); } } if (!acomp->ops) { -- 2.28.0

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