January 2022 - Alsa-devel - mailman.alsa-project.org

[PATCH 2/2] ALSA: virmidi: Remove duplicated code
by Stefan Sauer 06 Jan '22

06 Jan '22

seq_virmidi.c: snd_virmidi_new() is already setting seq_mode to SNDRV_VIRMIDI_SEQ_DISPATCH. Signed-off-by: Stefan Sauer <st_kost(a)gmx.de> --- sound/drivers/virmidi.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/drivers/virmidi.c b/sound/drivers/virmidi.c index 7f7eed6faaae..58012de90c38 100644 --- a/sound/drivers/virmidi.c +++ b/sound/drivers/virmidi.c @@ -90,15 +90,12 @@ static int snd_virmidi_probe(struct platform_device *devptr) } for (idx = 0; idx < midi_devs[dev]; idx++) { struct snd_rawmidi *rmidi; - struct snd_virmidi_dev *rdev; err = snd_virmidi_new(card, idx, &rmidi); if (err < 0) return err; - rdev = rmidi->private_data; vmidi->midi[idx] = rmidi; strcpy(rmidi->name, "Virtual Raw MIDI"); - rdev->seq_mode = SNDRV_VIRMIDI_SEQ_DISPATCH; } strcpy(card->driver, "VirMIDI"); -- 2.34.1

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[PATCH 1/2] ALSA: seq: virmidi: Add a drain operation
by Stefan Sauer 06 Jan '22

06 Jan '22

If a driver does not supply a drain operation for outputs, a default code path will execute msleep(50). Especially for a virtual midi device this severely limmits the throughput. This implementation for the virtual midi driver simply flushes the output workqueue. Signed-off-by: Stefan Sauer <st_kost(a)gmx.de> --- sound/core/seq/seq_virmidi.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/core/seq/seq_virmidi.c b/sound/core/seq/seq_virmidi.c index 4abc38c70cae..f5cae49500c8 100644 --- a/sound/core/seq/seq_virmidi.c +++ b/sound/core/seq/seq_virmidi.c @@ -262,6 +262,16 @@ static int snd_virmidi_output_close(struct snd_rawmidi_substream *substream) return 0; } +/* + * drain output work queue + */ +static void snd_virmidi_output_drain(struct snd_rawmidi_substream *substream) +{ + struct snd_virmidi *vmidi = substream->runtime->private_data; + + flush_work(&vmidi->output_work); +} + /* * subscribe callback - allow output to rawmidi device */ @@ -336,6 +346,7 @@ static const struct snd_rawmidi_ops snd_virmidi_output_ops = { .open = snd_virmidi_output_open, .close = snd_virmidi_output_close, .trigger = snd_virmidi_output_trigger, + .drain = snd_virmidi_output_drain, }; /* -- 2.34.1

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[PATCH v2] ALSA: intel_hdmi: Check for error num after setting mask
by Jiasheng Jiang 06 Jan '22

06 Jan '22

To maintain the consistency of the code, it should be better to add the sanity check after calling dma_set_mask_and_coherent(), like tegra_pcm_dma_allocate() in `sound/soc/tegra/tegra_pcm.c`. Signed-off-by: Jiasheng Jiang <jiasheng(a)iscas.ac.cn> --- Changelog v1 -> v2 * Change 1. Remove the fixes tag and change the message. --- sound/x86/intel_hdmi_audio.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/x86/intel_hdmi_audio.c b/sound/x86/intel_hdmi_audio.c index 33b12aa67cf5..6caea517f07f 100644 --- a/sound/x86/intel_hdmi_audio.c +++ b/sound/x86/intel_hdmi_audio.c @@ -1770,7 +1770,9 @@ static int hdmi_lpe_audio_probe(struct platform_device *pdev) card_ctx->irq = irq; /* only 32bit addressable */ - dma_set_mask_and_coherent(&pdev->dev, DMA_BIT_MASK(32)); + ret = dma_set_mask_and_coherent(&pdev->dev, DMA_BIT_MASK(32)); + if (ret) + goto err; init_channel_allocations(); -- 2.25.1

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Re: Re: [PATCH] ALSA: intel_hdmi: Check for error num after setting mask
by Jiasheng Jiang 06 Jan '22

06 Jan '22

On Thu, Jan 06, 2022 at 04:18:01PM +0800, Takashi Iwai wrote: > Well, 32bit DMA mask practically never fails on x86 (and other > architectures, IIRC). It's fine to add a sanity check, but it's > better to be mentioned that it never fails. Actually, I have already found many place check the 32bit DMA mask. For example: snd_ad1889_create() in `sound/pci/ad1889.c`, snd_vortex_create() in `sound/pci/au88x0/au88x0.c` tegra_pcm_dma_allocate() in `sound/soc/tegra/tegra_pcm.c`. Therefore, I think there must be some reason that 32bit may fail. So, to make the system more robust, it should be better to add the sanity check. Sincerely thanks, Jiang

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[PATCH] ALSA: intel_hdmi: Check for error num after setting mask
by Jiasheng Jiang 06 Jan '22

06 Jan '22

As the dma_supported() may fail, the dma_set_mask_and_coherent() may fail too. Therefore, it should be better to check it and return error if fails. Fixes: da8648097497 ("ALSA: x86: Flatten two abstraction layers") Signed-off-by: Jiasheng Jiang <jiasheng(a)iscas.ac.cn> --- sound/x86/intel_hdmi_audio.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/x86/intel_hdmi_audio.c b/sound/x86/intel_hdmi_audio.c index 33b12aa67cf5..6caea517f07f 100644 --- a/sound/x86/intel_hdmi_audio.c +++ b/sound/x86/intel_hdmi_audio.c @@ -1770,7 +1770,9 @@ static int hdmi_lpe_audio_probe(struct platform_device *pdev) card_ctx->irq = irq; /* only 32bit addressable */ - dma_set_mask_and_coherent(&pdev->dev, DMA_BIT_MASK(32)); + ret = dma_set_mask_and_coherent(&pdev->dev, DMA_BIT_MASK(32)); + if (ret) + goto err; init_channel_allocations(); -- 2.25.1

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hda-analyzer porting to GTK3
by GitHub issues - opened 06 Jan '22

06 Jan '22

alsa-project/alsa-utils issue #134 was opened from kosiu: I know that hda-analyzer is not a part of alsa-utils. I'm trying to port it into GTK3 at first but I encounter problems with scrolling of the graph. Source is here: https://github.com/kosiu/alsa/tree/gtk3 Issue URL : https://github.com/alsa-project/alsa-utils/issues/134 Repository URL: https://github.com/alsa-project/alsa-utils

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UAC2: are UAC2 latency controls included in snd_pcm_status_get_delay?
by Pavel Hofman 06 Jan '22

06 Jan '22

Hi, The UAC2 ch. "5.2.4.8 Latency Control" specifies latencies optionally reported by the device for individual entities. Especially the overall terminal latency TE_LATENCY_CONTROL is interesting. Some video players include the delay reported by snd_pcm_status_get_delay into their AV sync calculation. A linux USB audio gadget can perform some advanced DSP, which including the output device latency can result in a non-negligible overall latency. The gadget userspace software could configure the actual value to be reported by the gadget to the host in the selector UAC2_TE_LATENCY (audio_v2.h). The UAC2-specced max value of over 4 secs is more than sufficient for standard use cases. But I could not find any read of the UAC2_TE_LATENCY control selector in the snd-usb-audio driver ( https://elixir.bootlin.com/linux/latest/C/ident/UAC2_TE_LATENCY ). Please does the driver include this value into the overall delay reported through alsa-lib? If not, would it make sense to add such support? Thanks a lot, Pavel.

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[PATCH v3] dt-bindings: dsp: mediatek: add mt8195 dsp document
by YC Hung 06 Jan '22

06 Jan '22

From: "YC Hung" <yc.hung(a)mediatek.com> This patch adds mt8195 dsp document. The dsp is used for Sound Open Firmware driver node. It includes registers, clocks, memory regions, and mailbox for dsp. Signed-off-by: yc.hung <yc.hung(a)mediatek.com> --- Changes since v2: Remove useless watchdog interrupt. Add commit message more detail description. Changes since v1: Rename yaml file name as mediatek,mt8195-dsp.yaml Refine descriptions for mailbox, memory-region and drop unused labels in examples. --- --- .../devicetree/bindings/dsp/mediatek,mt8195-dsp.yaml | 12 ------------ 1 file changed, 12 deletions(-) diff --git a/Documentation/devicetree/bindings/dsp/mediatek,mt8195-dsp.yaml b/Documentation/devicetree/bindings/dsp/mediatek,mt8195-dsp.yaml index bde763191d86..779daa786739 100644 --- a/Documentation/devicetree/bindings/dsp/mediatek,mt8195-dsp.yaml +++ b/Documentation/devicetree/bindings/dsp/mediatek,mt8195-dsp.yaml @@ -27,14 +27,6 @@ properties: - const: cfg - const: sram - interrupts: - items: - - description: watchdog interrupt - - interrupt-names: - items: - - const: wdt - clocks: items: - description: mux for audio dsp clock @@ -75,8 +67,6 @@ required: - compatible - reg - reg-names - - interrupts - - interrupt-names - clocks - clock-names - memory-region @@ -95,8 +85,6 @@ examples: reg = <0x10803000 0x1000>, <0x10840000 0x40000>; reg-names = "cfg", "sram"; - interrupts = <GIC_SPI 694 IRQ_TYPE_LEVEL_HIGH 0>; - interrupt-names = "wdt"; clocks = <&topckgen 10>, //CLK_TOP_ADSP <&clk26m>, <&topckgen 107>, //CLK_TOP_AUDIO_LOCAL_BUS -- 2.18.0

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[PATCH] ALSA: hda: ALC287: Add Lenovo IdeaPad Slim 9i 14ITL5 speaker quirk
by Bart Kroon 05 Jan '22

05 Jan '22

Hello, The speaker fixup that is used for the Yoga 7 14ITL5 also applies to the IdeaPad Slim 9i 14ITL5. The attached patch applies the quirk to initialise the amplifier on the IdeaPad Slim 9i as well. This is validated to work on my laptop. Signed-off-by: Bart Kroon <bart(a)tarmack.eu> Regards, Bart

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[PATCH] ASoC: ak4375: Fix unused function error
by Takashi Iwai 05 Jan '22

05 Jan '22

A randconfig caught a compile warning that is now treated as a fatal error: sound/soc/codecs/ak4375.c:415:13: error: ‘ak4375_power_off’ defined but not used [-Werror=unused-function] where ak4375_power_off() is used only from the PM handler. As both suspend and resumes are already marked with __maybe_unused, let's rip off the superfluous ifdef CONFIG_PM, so that the error above can be avoided. Fixes: 53778b8292b5 ("ASoC: Add AK4375 support") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/codecs/ak4375.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/codecs/ak4375.c b/sound/soc/codecs/ak4375.c index 22cda0699341..9a7b662016b9 100644 --- a/sound/soc/codecs/ak4375.c +++ b/sound/soc/codecs/ak4375.c @@ -438,7 +438,6 @@ static int ak4375_power_on(struct ak4375_priv *ak4375) return 0; } -#ifdef CONFIG_PM static int __maybe_unused ak4375_runtime_suspend(struct device *dev) { struct ak4375_priv *ak4375 = dev_get_drvdata(dev); @@ -463,7 +462,6 @@ static int __maybe_unused ak4375_runtime_resume(struct device *dev) return regcache_sync(ak4375->regmap); } -#endif /* CONFIG_PM */ static const struct snd_soc_component_driver soc_codec_dev_ak4375 = { .controls = ak4375_snd_controls, -- 2.31.1

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