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

[PATCH] ALSA: seq: Use bool for snd_seq_queue internal flags
by Takashi Iwai 06 Dec '20

06 Dec '20

The snd_seq_queue struct contains various flags in the bit fields. Those are categorized to two different use cases, both of which are protected by different spinlocks. That implies that there are still potential risks of the bad operations for bit fields by concurrent accesses. For addressing the problem, this patch rearranges those flags to be a standard bool instead of a bit field. Reported-by: syzbot+63cbe31877bb80ef58f5(a)syzkaller.appspotmail.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/seq/seq_queue.h | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/core/seq/seq_queue.h b/sound/core/seq/seq_queue.h index 1c3a8d3254d9..c69105dc1a10 100644 --- a/sound/core/seq/seq_queue.h +++ b/sound/core/seq/seq_queue.h @@ -26,10 +26,10 @@ struct snd_seq_queue { struct snd_seq_timer *timer; /* time keeper for this queue */ int owner; /* client that 'owns' the timer */ - unsigned int locked:1, /* timer is only accesibble by owner if set */ - klocked:1, /* kernel lock (after START) */ - check_again:1, - check_blocked:1; + bool locked; /* timer is only accesibble by owner if set */ + bool klocked; /* kernel lock (after START) */ + bool check_again; /* concurrent access happened during check */ + bool check_blocked; /* queue being checked */ unsigned int flags; /* status flags */ unsigned int info_flags; /* info for sync */ -- 2.26.2

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[PATCH 0/1] dt-bindings: eliminate yamllint warnings
by Zhen Lei 06 Dec '20

06 Dec '20

There're too many people, I just send to the maintainer, reviewer, supporter. Eliminate below warnings: ./Documentation/devicetree/bindings/clock/imx8qxp-lpcg.yaml:32:13: [warning] wrong indentation: expected 14 but found 12 (indentation) ./Documentation/devicetree/bindings/clock/imx8qxp-lpcg.yaml:35:9: [warning] wrong indentation: expected 10 but found 8 (indentation) ./Documentation/devicetree/bindings/display/intel,keembay-msscam.yaml:21:6: [warning] wrong indentation: expected 6 but found 5 (indentation) ./Documentation/devicetree/bindings/display/bridge/analogix,anx7625.yaml:52:9: [warning] wrong indentation: expected 6 but found 8 (indentation) ./Documentation/devicetree/bindings/display/bridge/intel,keembay-dsi.yaml:42:8: [warning] wrong indentation: expected 8 but found 7 (indentation) ./Documentation/devicetree/bindings/display/bridge/intel,keembay-dsi.yaml:45:8: [warning] wrong indentation: expected 8 but found 7 (indentation) ./Documentation/devicetree/bindings/display/panel/novatek,nt36672a.yaml:25:10: [warning] wrong indentation: expected 10 but found 9 (indentation) ./Documentation/devicetree/bindings/media/i2c/adv7604.yaml:24:9: [warning] wrong indentation: expected 10 but found 8 (indentation) ./Documentation/devicetree/bindings/media/i2c/mipi-ccs.yaml:4:1: [error] missing document start "---" (document-start) ./Documentation/devicetree/bindings/media/i2c/mipi-ccs.yaml:29:9: [warning] wrong indentation: expected 10 but found 8 (indentation) ./Documentation/devicetree/bindings/media/i2c/mipi-ccs.yaml:32:9: [warning] wrong indentation: expected 10 but found 8 (indentation) ./Documentation/devicetree/bindings/media/i2c/ovti,ov772x.yaml:79:17: [warning] wrong indentation: expected 14 but found 16 (indentation) ./Documentation/devicetree/bindings/media/i2c/ovti,ov772x.yaml:88:17: [warning] wrong indentation: expected 14 but found 16 (indentation) ./Documentation/devicetree/bindings/media/i2c/sony,imx214.yaml:72:17: [warning] wrong indentation: expected 18 but found 16 (indentation) ./Documentation/devicetree/bindings/media/i2c/sony,imx214.yaml:75:17: [warning] wrong indentation: expected 18 but found 16 (indentation) ./Documentation/devicetree/bindings/mmc/mtk-sd.yaml:20:9: [warning] wrong indentation: expected 10 but found 8 (indentation) ./Documentation/devicetree/bindings/mmc/mtk-sd.yaml:30:9: [warning] wrong indentation: expected 10 but found 8 (indentation) ./Documentation/devicetree/bindings/mmc/mtk-sd.yaml:33:9: [warning] wrong indentation: expected 10 but found 8 (indentation) ./Documentation/devicetree/bindings/sound/mt8192-mt6359-rt1015-rt5682.yaml:10:4: [warning] wrong indentation: expected 2 but found 3 (indentation) Zhen Lei (1): dt-bindings: eliminate yamllint warnings .../devicetree/bindings/clock/imx8qxp-lpcg.yaml | 20 ++++++++--------- .../bindings/display/bridge/analogix,anx7625.yaml | 4 ++-- .../bindings/display/bridge/intel,keembay-dsi.yaml | 4 ++-- .../bindings/display/intel,keembay-msscam.yaml | 4 ++-- .../bindings/display/panel/novatek,nt36672a.yaml | 2 +- .../devicetree/bindings/media/i2c/adv7604.yaml | 4 ++-- .../devicetree/bindings/media/i2c/mipi-ccs.yaml | 11 ++++----- .../devicetree/bindings/media/i2c/ovti,ov772x.yaml | 12 +++++----- .../devicetree/bindings/media/i2c/sony,imx214.yaml | 12 +++++----- Documentation/devicetree/bindings/mmc/mtk-sd.yaml | 26 +++++++++++----------- .../sound/mt8192-mt6359-rt1015-rt5682.yaml | 4 ++-- 11 files changed, 52 insertions(+), 51 deletions(-) -- 1.8.3

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thinkpad t14s amd: unable to control built-in mic and headphones mic independently
by GitHub issues - opened 05 Dec '20

05 Dec '20

alsa-project/alsa-ucm-conf issue #64 was opened from tanuk: I did some investigation on a [pulseaudio bug report](https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/1067), and found that the UCM configuration for the internal mic and headset mic both use "Capture" as the CaptureMixerElem. This means that muting one device mutes also the other (and I guess volume control is shared too). In amixer output I don't see a separate control for the headset mic, but for the internal mic volume I guess "Digital" can be used (there's no switch, though). So it seems that these changes are needed: - don't set CaptureMixerElem, CaptureSwitch or CaptureVolume for the headset mic - set the internal mic CaptureMixerElem and CaptureVolume to Digital - you may want to also set CaptureMasterElem for both both mics to Capture Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/64 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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Re: Behringer WING usb audio - cyclic xruns dependent on periods/buffers
by Takashi Iwai 05 Dec '20

05 Dec '20

On Thu, 03 Dec 2020 21:06:24 +0100, Ben Bell wrote: > > On Sat, Nov 28, 2020 at 09:36:00AM +0000, Ben Bell wrote: > > > In general you should avoid 44.1kHz if you want a small period size > > > for a realtime process on USB-audio. With 44.1kHz, the packet size > > > can't be fixed in integer, and the ISO transfer requires variable > > > packet sizes. OTOH, ALSA API requires the fixed period size, hence > > > it'll lead to inconsistencies occasionally. > > So changing to 48kHz had no appreciable effect, but I'm working at that > rate for now to eliminate the 44.1kHz weirdness from investigations. > Using a USB2 card, also had no effect. Testing with a different USB audio > interface (albeit a simple stereo one) didn't exhibit the behaviour, even > when I took buffer sizes right down. > > I'm not sure where to go to get more information or where's the best place > to ask for help. I'm happy to do the leg work, but I don't know enough > about the kernel, alsa or USB to figure it out without some help. > > Current question: what is the delay in /proc/asound/card1/pcm0c/sub0/status > actually measuring? I'm assuming it's measured in samples? I've written > something to scrape the stats out in a tight loop and report. > > What I see is a cycle where the delay rises and then a chunk equal to the > frames per period (or sometimes, earlier on fpp-48) is removed. That feels > like chunks being read out of a buffer. All fine. > > After a while though, the maximum delay we reach with each cycle is creeping > up. It increases by one every few cycles (usually two or three, or three or > four -- always oscillating between two values) but of it's still only being > emptied by a full period's worth of samples each cycle. So the overall effect > is the delay creeps up and up until it hits the buffer size and then we get > an xrun. > > Like I said in the initial email, it feels like some sort of clock drift > problem, where we're managing very slowly to collect more samples than > we're reading -- to the tune of about 1 extra every few cycles -- and > nothing on the consumer side is ever managing to compensate for that. > I'm not even sure how that sort of drift would be possible though. Seems > surprising. > > Does any of this sound suspicious, or for that matter completely normal? > Any suggestions where should I be looking next? At least you can try the latest patch set destined for 5.11, which should improve the cases for the implicit feedback. The patches are either in linux-next tree or the topic/usb-audio-refactoring branch of my sound.git tree. It's based on 5.10-rc, so should be cleanly mergeable to the latest Linus tree. Takashi

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[PATCH] slimbus: qcom-ngd-ctrl: Fix error handling to keep reference balanced
by Zhang Qilong 05 Dec '20

05 Dec '20

In the error branch, we should call pm_runtime_put_noidle and pm_runtime_disable to decrease usage counter and increase the power disable depth to avoid reference leaks. Fixes: 917809e2280bb ("slimbus: ngd: Add qcom SLIMBus NGD driver") Signed-off-by: Zhang Qilong <zhangqilong3(a)huawei.com> --- drivers/slimbus/qcom-ngd-ctrl.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/drivers/slimbus/qcom-ngd-ctrl.c b/drivers/slimbus/qcom-ngd-ctrl.c index c054e83ab636..a9b4a1d95045 100644 --- a/drivers/slimbus/qcom-ngd-ctrl.c +++ b/drivers/slimbus/qcom-ngd-ctrl.c @@ -1483,7 +1483,7 @@ static int qcom_slim_ngd_probe(struct platform_device *pdev) ret = qcom_slim_ngd_qmi_svc_event_init(ctrl); if (ret) { dev_err(&pdev->dev, "QMI service registration failed:%d", ret); - return ret; + goto pm_err; } INIT_WORK(&ctrl->m_work, qcom_slim_ngd_master_worker); @@ -1500,6 +1500,9 @@ static int qcom_slim_ngd_probe(struct platform_device *pdev) qcom_slim_ngd_qmi_svc_event_deinit(&ctrl->qmi); if (ctrl->mwq) destroy_workqueue(ctrl->mwq); +pm_err: + pm_runtime_put_noidle(dev); + pm_runtime_disable(dev); return ret; } -- 2.25.4

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[PATCH] soundwire: intel: fix another unused-function warning
by Arnd Bergmann 05 Dec '20

05 Dec '20

From: Arnd Bergmann <arnd(a)arndb.de> Without CONFIG_PM, there is another warning about an unused function: drivers/soundwire/intel.c:530:12: error: 'intel_link_power_down' defined but not used [-Werror=unused-function] After a previous fix, the driver already uses both an #ifdef and a __maybe_unused annotation but still gets it wrong. Remove the ifdef and instead use __maybe_unused consistently to avoid the problem for good. Fixes: f046b2334083 ("soundwire: intel: fix intel_suspend/resume defined but not used warning") Fixes: ebf878eddbb4 ("soundwire: intel: add pm_runtime support") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- drivers/soundwire/intel.c | 8 ++------ 1 file changed, 2 insertions(+), 6 deletions(-) diff --git a/drivers/soundwire/intel.c b/drivers/soundwire/intel.c index 6a1e862b16c3..66adb258a425 100644 --- a/drivers/soundwire/intel.c +++ b/drivers/soundwire/intel.c @@ -1585,8 +1585,6 @@ int intel_master_process_wakeen_event(struct platform_device *pdev) * PM calls */ -#ifdef CONFIG_PM - static int __maybe_unused intel_suspend(struct device *dev) { struct sdw_cdns *cdns = dev_get_drvdata(dev); @@ -1641,7 +1639,7 @@ static int __maybe_unused intel_suspend(struct device *dev) return 0; } -static int intel_suspend_runtime(struct device *dev) +static int __maybe_unused intel_suspend_runtime(struct device *dev) { struct sdw_cdns *cdns = dev_get_drvdata(dev); struct sdw_intel *sdw = cdns_to_intel(cdns); @@ -1796,7 +1794,7 @@ static int __maybe_unused intel_resume(struct device *dev) return ret; } -static int intel_resume_runtime(struct device *dev) +static int __maybe_unused intel_resume_runtime(struct device *dev) { struct sdw_cdns *cdns = dev_get_drvdata(dev); struct sdw_intel *sdw = cdns_to_intel(cdns); @@ -1969,8 +1967,6 @@ static int intel_resume_runtime(struct device *dev) return ret; } -#endif - static const struct dev_pm_ops intel_pm = { SET_SYSTEM_SLEEP_PM_OPS(intel_suspend, intel_resume) SET_RUNTIME_PM_OPS(intel_suspend_runtime, intel_resume_runtime, NULL) -- 2.27.0

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[PATCH v2] ALSA: hda/realtek: make bass spk volume adjustable on a yoga laptop
by Hui Wang 05 Dec '20

05 Dec '20

This change could fix 2 issues on this machine: - the bass speaker's output volume can't be adjusted, that is because the bass speaker is routed to the DAC (Nid 0x6) which has no volume control. - after plugging a headset with vol+, vol- and pause buttons on it, press those buttons, nothing happens, this means those buttons don't work at all. This machine has alc287 codec, need to add the codec id to the disable/enable_headset_jack_key(), then the headset button could work. The quirk of ALC285_FIXUP_THINKPAD_HEADSET_JACK could fix both of these 2 issues. Cc: <stable(a)vger.kernel.org> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8616c5624870..5a905fa1b33a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3104,6 +3104,7 @@ static void alc_disable_headset_jack_key(struct hda_codec *codec) case 0x10ec0215: case 0x10ec0225: case 0x10ec0285: + case 0x10ec0287: case 0x10ec0295: case 0x10ec0289: case 0x10ec0299: @@ -3130,6 +3131,7 @@ static void alc_enable_headset_jack_key(struct hda_codec *codec) case 0x10ec0215: case 0x10ec0225: case 0x10ec0285: + case 0x10ec0287: case 0x10ec0295: case 0x10ec0289: case 0x10ec0299: @@ -8578,6 +8580,11 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x14, 0x90170110}, {0x19, 0x04a11040}, {0x21, 0x04211020}), + SND_HDA_PIN_QUIRK(0x10ec0287, 0x17aa, "Lenovo", ALC285_FIXUP_THINKPAD_HEADSET_JACK, + {0x14, 0x90170110}, + {0x17, 0x90170111}, + {0x19, 0x03a11030}, + {0x21, 0x03211020}), SND_HDA_PIN_QUIRK(0x10ec0286, 0x1025, "Acer", ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE, {0x12, 0x90a60130}, {0x17, 0x90170110}, -- 2.25.1

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[PATCH] ALSA: hda/realtek: make bass spk volume adjustable on a yoga laptop
by Hui Wang 05 Dec '20

05 Dec '20

This also make the headset button work on this machine. Cc: <stable(a)vger.kernel.org> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8616c5624870..5a905fa1b33a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3104,6 +3104,7 @@ static void alc_disable_headset_jack_key(struct hda_codec *codec) case 0x10ec0215: case 0x10ec0225: case 0x10ec0285: + case 0x10ec0287: case 0x10ec0295: case 0x10ec0289: case 0x10ec0299: @@ -3130,6 +3131,7 @@ static void alc_enable_headset_jack_key(struct hda_codec *codec) case 0x10ec0215: case 0x10ec0225: case 0x10ec0285: + case 0x10ec0287: case 0x10ec0295: case 0x10ec0289: case 0x10ec0299: @@ -8578,6 +8580,11 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x14, 0x90170110}, {0x19, 0x04a11040}, {0x21, 0x04211020}), + SND_HDA_PIN_QUIRK(0x10ec0287, 0x17aa, "Lenovo", ALC285_FIXUP_THINKPAD_HEADSET_JACK, + {0x14, 0x90170110}, + {0x17, 0x90170111}, + {0x19, 0x03a11030}, + {0x21, 0x03211020}), SND_HDA_PIN_QUIRK(0x10ec0286, 0x1025, "Acer", ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE, {0x12, 0x90a60130}, {0x17, 0x90170110}, -- 2.25.1

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[PATCH] ASoC: SOF: control: fix cppcheck warning in snd_sof_volume_info()
by Kai Vehmanen 05 Dec '20

05 Dec '20

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Fix cppcheck warning: sound/soc/sof/control.c:117:82: style:inconclusive: Function 'snd_sof_volume_info' argument 2 names different: declaration 'ucontrol' definition 'uinfo'. [funcArgNamesDifferent] Fixes: fca18e62984a ("ASoC: SOF: control: override volume info callback") Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/soc/sof/sof-audio.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/sof-audio.h b/sound/soc/sof/sof-audio.h index d00a918b6f39..dc930fc2f4b5 100644 --- a/sound/soc/sof/sof-audio.h +++ b/sound/soc/sof/sof-audio.h @@ -125,7 +125,7 @@ int snd_sof_volume_get(struct snd_kcontrol *kcontrol, int snd_sof_volume_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); int snd_sof_volume_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *ucontrol); + struct snd_ctl_elem_info *uinfo); int snd_sof_switch_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); int snd_sof_switch_put(struct snd_kcontrol *kcontrol, -- 2.28.0

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[PATCH] ASoC: q6afe-clocks: Add missing parent clock rate
by Srinivas Kandagatla 05 Dec '20

05 Dec '20

setting clock rate on child clocks without a parent clock rate will result in zero clk rate for child. This also means that when audio is started dsp will attempt to access registers without enabling clock resulting in board boot up. Fix this by adding the missing parent clock rate. Fixes: 520a1c396d196 ("ASoC: q6afe-clocks: add q6afe clock controller") Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/qcom/qdsp6/q6afe-clocks.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/qcom/qdsp6/q6afe-clocks.c b/sound/soc/qcom/qdsp6/q6afe-clocks.c index 87e4633afe2c..f0362f061652 100644 --- a/sound/soc/qcom/qdsp6/q6afe-clocks.c +++ b/sound/soc/qcom/qdsp6/q6afe-clocks.c @@ -16,6 +16,7 @@ .afe_clk_id = Q6AFE_##id, \ .name = #id, \ .attributes = LPASS_CLK_ATTRIBUTE_COUPLE_NO, \ + .rate = 19200000, \ .hw.init = &(struct clk_init_data) { \ .ops = &clk_q6afe_ops, \ .name = #id, \ -- 2.21.0

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