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

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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Question about buildroot vs aplay/arecord with -d
by Kuninori Morimoto 09 Oct '20

09 Oct '20

Hi ALSA ML I'm now using buildroot, and want to use aplay/arecord in the same time with -d option. This works for me aplay -d 3 xxx And this works for me, too arecord -d 3 yyy But, this doesn't work, and system will hung up aplay -d 3 xxx & arecord -d 3 yyy Sometimes it will indicates this message rcu: INFO: rcu_preempt detected stalls on CPUs/tasks: rcu: 0-....: (1 GPs behind) idle=496/0/0x1 softirq=228/228 fqs=2172 (detected by 1, t=6502 jiffies, g=-103, q=17) Task dump for CPU 0: task:swapper/0 state:R running task stack: 0 pid: 0 ppid: 0 flags:0x0000002a Call trace: __switch_to+0x144/0x1a0 init_task+0x0/0x1780 ... Do you know why this happen ? Thank you for your help !! Best regards --- Kuninori Morimoto

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[PATCH][next] ASoC: mchp-spdifrx: fix spelling mistake "overrrun" -> "overrun"
by Colin King 09 Oct '20

09 Oct '20

From: Colin Ian King <colin.king(a)canonical.com> There is a spelling mistake in a dev_warn message. Fix it. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/atmel/mchp-spdifrx.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/atmel/mchp-spdifrx.c b/sound/soc/atmel/mchp-spdifrx.c index 726e4951d9a5..e6ded6f8453f 100644 --- a/sound/soc/atmel/mchp-spdifrx.c +++ b/sound/soc/atmel/mchp-spdifrx.c @@ -338,7 +338,7 @@ static irqreturn_t mchp_spdif_interrupt(int irq, void *dev_id) } if (pending & SPDIFRX_IR_OVERRUN) { - dev_warn(dev->dev, "Overrrun detected\n"); + dev_warn(dev->dev, "Overrun detected\n"); ret = IRQ_HANDLED; } -- 2.27.0

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[PATCH 0/2] ASoC: stm32: dfsdm: change rate limits
by Olivier Moysan 09 Oct '20

09 Oct '20

Widening of the supported rate range in the STM32 DFSDM driver. The rates were previously limited to 8kHz, 16kHz and 32kHz. Allow rate capture in the whole range 8kHz-48kHz as there is no hardware limitation to support it. Actual sample resolution is dependent on audio rate and DFSDM configuration. Add a trace to allow simple check of sample resolution. Olivier Moysan (2): ASoC: stm32: dfsdm: change rate limits ASoC: stm32: dfsdm: add actual resolution trace drivers/iio/adc/stm32-dfsdm-adc.c | 4 ++++ drivers/iio/adc/stm32-dfsdm.h | 2 ++ sound/soc/stm/stm32_adfsdm.c | 8 +++----- 3 files changed, 9 insertions(+), 5 deletions(-) -- 2.17.1

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[PATCH v2 1/2] dt-bindings: tas2764: Add the TAS2764 binding doc
by Dan Murphy 09 Oct '20

09 Oct '20

Add the binding for the TAS2764 Smart Amplifier. Signed-off-by: Dan Murphy <dmurphy(a)ti.com> --- .../devicetree/bindings/sound/tas2764.yaml | 74 +++++++++++++++++++ 1 file changed, 74 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/tas2764.yaml diff --git a/Documentation/devicetree/bindings/sound/tas2764.yaml b/Documentation/devicetree/bindings/sound/tas2764.yaml new file mode 100644 index 000000000000..d2c90c82b13f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tas2764.yaml @@ -0,0 +1,74 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +# Copyright (C) 2020 Texas Instruments Incorporated +%YAML 1.2 +--- +$id: "http://devicetree.org/schemas/sound/tas2764.yaml#" +$schema: "http://devicetree.org/meta-schemas/core.yaml#" + +title: Texas Instruments TAS2764 Smart PA + +maintainers: + - Dan Murphy <dmurphy(a)ti.com> + +description: | + The TAS2764 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,tas2764 + + reg: + maxItems: 1 + description: | + I2C address of the device can be between 0x38 to 0x45. + + reset-gpios: + description: GPIO used to reset the device. + + 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. + + ti,vmon-slot-no: + $ref: /schemas/types.yaml#/definitions/uint32 + description: TDM TX voltage 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@38 { + compatible = "ti,tas2764"; + reg = <0x38>; + #sound-dai-cells = <1>; + interrupt-parent = <&gpio1>; + interrupts = <14>; + reset-gpio = <&gpio1 15 0>; + shutdown-gpios = <&gpio1 15 0>; + ti,imon-slot-no = <0>; + ti,vmon-slot-no = <2>; + }; + }; + +... -- 2.28.0.585.ge1cfff676549

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