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Re: Rockchip I2S commit possibly ignored
by sugar zhang 21 Aug '22

21 Aug '22

Hi Geraldo, Mark, 在 2022/8/17 1:54, Geraldo Nascimento 写道: > On Tue, Aug 16, 2022 at 04:22:37PM +0100, Mark Brown wrote: >> On Tue, Aug 16, 2022 at 10:37:18AM -0300, Geraldo Nascimento wrote: >> >>> This isn't my patch, it's a patch by Sugar Zhang from Rockchip that was >>> supposed to have been applied one year ago, give or take 10 days. I >>> can't resend a patch that wasn't authored by me. >> There's absolutely no problem with reposting someone else's patch - just >> add your Signed-off-by at the end of the signoff chain. > Mark, I'm in no position to lecture anyone, least of who, you, hard-working > ASoC maintainer that you are. But there are *tons* of problems with > reposting someone else's patch, even if they had been previously given > the green-light but misteriously vanished. > > You are assuming responsibility for someone else's work! Let's see in > this case what would happen: > > In this case, the Rockchip I2S missing reset affair, I have tested > the proposed changes, and they are OK, *if* you add the reset properties > in the device tree. > > For example, I was playing with HDMI sound on the RK3399pro Radxa Rock Pi > N10 and I added the following properties to my private rk3399.dtsi: > > --- a/arch/arm64/boot/dts/rockchip/rk3399.dtsi > +++ b/arch/arm64/boot/dts/rockchip/rk3399.dtsi > @@ -1695,6 +1695,8 @@ i2s2: i2s@ff8a0000 { > dma-names = "tx", "rx"; > clock-names = "i2s_clk", "i2s_hclk"; > clocks = <&cru SCLK_I2S2_8CH>, <&cru HCLK_I2S2_8CH>; > + reset-names = "reset-m", "reset-h"; > + resets = <&cru SRST_I2S2_8CH>, <&cru SRST_H_I2S2_8CH>; > power-domains = <&power RK3399_PD_SDIOAUDIO>; > #sound-dai-cells = <0>; > status = "disabled"; > > --- > > > And once I okay &i2s2 and &hdmi_sound inside rk3399pro-vmarc-som.dtsi I > get kind of working HDMI sound, very clipped, I have to software adjust > the volume to 30% to get clean, not speaker-blowing sound. But I > digress, the point that it kinda works is not the point at all. > > My main point is that without adding "resets" and "reset-names" to *at > least* every Rockchip device tree that enables sound over HDMI, just an > example, you get systems with non-working HDMI. I just tested it, I > omitted "resets" and "reset-names" from my device tree and then had > to SSH into the black screen to revert the changes to my boot partition. Actually, the reset is used for i2s SLAVE mode which may fail to clear if the MASTER side stop the clk before i2s done. in this situation, we do reset the controller and bring it back to normal state. But, for HDMI sound, the i2s works as MASTER mode and should not be fail to clear. could you share your I2S and HDMI register dump to me to dig out the root cause. OTOH，I have noticed that some patchs ignored on upstream, I will check and repost those patchs. > So it's not trivial to RESEND this. It amounts to device tree overhaul > of arch/arm64/boot/dts/rockchip > > This would require community effort of say, equivalent to LibreELEC > resources, which are not many, but they have enough patience to test > every proposed change to Rockchip device trees, and could help upstream > this. > > Thank you, > Geraldo Nascimento -- Best Regards! 张学广/Sugar 瑞芯微电子股份有限公司 Rockchip Electronics Co., Ltd.

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Rockchip/rk3399-gru-sound: Use 48KHz sample rate for all devices
by GitHub pull_request - opened 20 Aug '22

20 Aug '22

alsa-project/alsa-ucm-conf pull request #201 was opened from alpernebbi: The rk3399-gru-kevin board has problems when trying to simultaneously playback and record audio with the default PulseAudio sample rate of 44.1KHz. When the following command is run, the playback starts a few seconds after the recording finishes. arecord -vvv -d 4 /dev/null & sleep 0.2; speaker-test -l 1 -p 100000 -t sine When the sample rates are set to 48KHz either with PA configuration or in the UCM, playback and recording both start immediately. Another example is when a music player is running and we start `arecord`, the music starts stuttering but `arecord` can't record anything either. Apparently, this is a hardware limitation due to the I2S lines being shared between the devices. Set all the device rates to 48KHz like Chrome OS does to make things work smoothly. [1] https://chromium-review.googlesource.com/389695 [2] https://chromium-review.googlesource.com/898682 Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/201 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/201.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[PATCH RESEND 0/3] ASoC: mediatek: support SOF for mt8186
by Chunxu Li 20 Aug '22

20 Aug '22

From: "chunxu.li" <chunxu.li(a)mediatek.com> In these patches, we add SOF support for mt8186 which have HiFi5 DSP inside. Chunxu Li (3): ASoC: mediatek: mt8186: support DSP downlink ASoC: mediatek: mt8186: add SOF support on mt8186-mt6366-rt1019-rt5682s ASoC: mediatek: mt8186: add SOF support on mt8186-mt6366-da7219-max98357 sound/soc/mediatek/mt8186/mt8186-afe-pcm.c | 3 + sound/soc/mediatek/mt8186/mt8186-dai-i2s.c | 12 ++ .../mt8186/mt8186-mt6366-da7219-max98357.c | 168 ++++++++++++++++-- .../mt8186/mt8186-mt6366-rt1019-rt5682s.c | 168 ++++++++++++++++-- 4 files changed, 325 insertions(+), 26 deletions(-) -- 2.25.1

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[RESEND PATCH v2 1/2] dt-bindings: dsp: fsl: Add SOF compatile string for i.MX8ULP
by Shengjiu Wang 20 Aug '22

20 Aug '22

Add SOF compatile string "fsl,imx8ulp-dsp" for supporting DSP device on i.MX8ULP platform. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> Acked-by: Rob Herring <robh(a)kernel.org> --- changes in v2 - add acked-by Rob Documentation/devicetree/bindings/dsp/fsl,dsp.yaml | 1 + 1 file changed, 1 insertion(+) diff --git a/Documentation/devicetree/bindings/dsp/fsl,dsp.yaml b/Documentation/devicetree/bindings/dsp/fsl,dsp.yaml index e66ef2da7879..9af40da5688e 100644 --- a/Documentation/devicetree/bindings/dsp/fsl,dsp.yaml +++ b/Documentation/devicetree/bindings/dsp/fsl,dsp.yaml @@ -20,6 +20,7 @@ properties: - fsl,imx8qxp-dsp - fsl,imx8qm-dsp - fsl,imx8mp-dsp + - fsl,imx8ulp-dsp - fsl,imx8qxp-hifi4 - fsl,imx8qm-hifi4 - fsl,imx8mp-hifi4 -- 2.34.1

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[PATCH -next 1/3] ASoC: amd: acp: add missing platform_device_unregister() in acp_pci_probe()
by Yang Yingliang 20 Aug '22

20 Aug '22

Add missing platform_device_unregister() in error path in acp_pci_probe(). Fixes: c49f5e74a11e ("ASoC: amd: acp: Add error handling cases") Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/soc/amd/acp/acp-pci.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/amd/acp/acp-pci.c b/sound/soc/amd/acp/acp-pci.c index 2c8e960cc9a6..5bb23ebe1216 100644 --- a/sound/soc/amd/acp/acp-pci.c +++ b/sound/soc/amd/acp/acp-pci.c @@ -104,6 +104,7 @@ static int acp_pci_probe(struct pci_dev *pci, const struct pci_device_id *pci_id addr = pci_resource_start(pci, 0); chip->base = devm_ioremap(&pci->dev, addr, pci_resource_len(pci, 0)); if (!chip->base) { + platform_device_unregister(dmic_dev); ret = -ENOMEM; goto release_regions; } -- 2.25.1

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[PATCH] ASoC: amd: acp: remove unnecessary NULL checks
by Dan Carpenter 20 Aug '22

20 Aug '22

The list iterator can never be NULL. Delete the bogus NULL checks. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/soc/amd/acp/acp-platform.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/amd/acp/acp-platform.c b/sound/soc/amd/acp/acp-platform.c index beee53aedeaf..85a81add4ef9 100644 --- a/sound/soc/amd/acp/acp-platform.c +++ b/sound/soc/amd/acp/acp-platform.c @@ -106,14 +106,14 @@ static irqreturn_t i2s_irq_handler(int irq, void *data) spin_lock(&adata->acp_lock); list_for_each_entry(stream, &adata->stream_list, list) { - if (stream && (ext_intr_stat & stream->irq_bit)) { + if (ext_intr_stat & stream->irq_bit) { writel(stream->irq_bit, ACP_EXTERNAL_INTR_STAT(adata, rsrc->irqp_used)); snd_pcm_period_elapsed(stream->substream); i2s_flag = 1; } if (adata->rsrc->no_of_ctrls == 2) { - if (stream && (ext_intr_stat1 & stream->irq_bit)) { + if (ext_intr_stat1 & stream->irq_bit) { writel(stream->irq_bit, ACP_EXTERNAL_INTR_STAT(adata, (rsrc->irqp_used - 1))); snd_pcm_period_elapsed(stream->substream); -- 2.35.1

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[PATCH] ASoC: Change handling of unimplemented set_bclk_ratio
by Martin Povišer 20 Aug '22

20 Aug '22

If a 'set_bclk_ratio' call is attempted on a DAI not implementing the method, make it an -ENOSUPP error instead of -EINVAL. Assume the DAI can still be okay with the ratio, just does not care to register a handler. No current in-tree users of snd_soc_dai_set_bclk_ratio seem to inspect the return value, but -ENOSUPP disables an error print from within the common soc_dai_ret return filter. With the new behavior a machine driver can do a blanket 'set_bclk_ratio' on all DAIs on a bus, some of which may care about the ratio, some of which may not. Signed-off-by: Martin Povišer <povik+lin(a)cutebit.org> --- sound/soc/soc-dai.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c index d530e8c2b77b..49752af0e205 100644 --- a/sound/soc/soc-dai.c +++ b/sound/soc/soc-dai.c @@ -124,7 +124,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_pll); */ int snd_soc_dai_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio) { - int ret = -EINVAL; + int ret = -ENOTSUPP; if (dai->driver->ops && dai->driver->ops->set_bclk_ratio) -- 2.33.0

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[PATCH] soundwire: qcom: remove unneeded check
by Steev Klimaszewski 19 Aug '22

19 Aug '22

commit 1fd0d85affe4d6 ("soundwire: qcom: Add flag for software clock gating check") added a flag for software clock gating check, however in commit 33ba01788889666 ("soundwire: qcom: Add support for controlling audio CGCR from HLOS") the same check was added without the flag, so we would still end up failing the software clock gating check. Originally reported by Amit Pundir on DB845c, I also saw it on the Lenovo Yoga C630, right before the splat, we would see qcom-soundwire wcd934x-soundwire.6.auto: Failed to get audio_cgcr reset required for soundwire-v1.6.0 however, SDM845 has a soundwire-v1.3.0 Since the flag was added in 1fd0d85affe, lets just remove this one. Fixes: 33ba01788889 ("soundwire: qcom: Add support for controlling audio CGCR from HLOS") Signed-off-by: Steev Klimaszewski <steev(a)kali.org> Reported-by: Amit Pundir <amit.pundir(a)linaro.org> --- drivers/soundwire/qcom.c | 4 ---- 1 file changed, 4 deletions(-) diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c index 709a7c1e0704..b621f7fb866f 100644 --- a/drivers/soundwire/qcom.c +++ b/drivers/soundwire/qcom.c @@ -1355,10 +1355,6 @@ static int qcom_swrm_probe(struct platform_device *pdev) ctrl->bus.compute_params = &qcom_swrm_compute_params; ctrl->bus.clk_stop_timeout = 300; - ctrl->audio_cgcr = devm_reset_control_get_exclusive(dev, "swr_audio_cgcr"); - if (IS_ERR(ctrl->audio_cgcr)) - dev_err(dev, "Failed to get audio_cgcr reset required for soundwire-v1.6.0\n"); - ret = qcom_swrm_get_port_config(ctrl); if (ret) goto err_clk; -- 2.35.1

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[PATCH v2] ASoC: SOF: Fix compilation when HDA_AUDIO_CODEC config is disabled
by Cezary Rojewski 19 Aug '22

19 Aug '22

hda_codec_device_init() expects three parameters, not two. Fixes: 3fd63658caed ("ASoC: Intel: Drop hdac_ext usage for codec device creation") Reported-by: kernel test robot <lkp(a)intel.com> Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> --- Changes in v2: - adjusted SHA-1 provided in Fixes tag sound/soc/sof/intel/hda-codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/intel/hda-codec.c b/sound/soc/sof/intel/hda-codec.c index 73336648cd25..1e9afc48394c 100644 --- a/sound/soc/sof/intel/hda-codec.c +++ b/sound/soc/sof/intel/hda-codec.c @@ -213,7 +213,7 @@ static int hda_codec_probe(struct snd_sof_dev *sdev, int address, put_device(&codec->core.dev); } #else - codec = hda_codec_device_init(&hbus->core, address); + codec = hda_codec_device_init(&hbus->core, address, HDA_DEV_ASOC); ret = PTR_ERR_OR_ZERO(codec); #endif -- 2.25.1

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[PATCH] ASoC: SOF: Fix compilation when HDA_AUDIO_CODEC config is disabled
by Cezary Rojewski 19 Aug '22

19 Aug '22

hda_codec_device_init() expects three parameters, not two. Fixes: f2183b08c81a ("ASoC: Intel: Drop hdac_ext usage for codec device creation") Reported-by: kernel test robot <lkp(a)intel.com> Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> --- sound/soc/sof/intel/hda-codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/intel/hda-codec.c b/sound/soc/sof/intel/hda-codec.c index 73336648cd25..1e9afc48394c 100644 --- a/sound/soc/sof/intel/hda-codec.c +++ b/sound/soc/sof/intel/hda-codec.c @@ -213,7 +213,7 @@ static int hda_codec_probe(struct snd_sof_dev *sdev, int address, put_device(&codec->core.dev); } #else - codec = hda_codec_device_init(&hbus->core, address); + codec = hda_codec_device_init(&hbus->core, address, HDA_DEV_ASOC); ret = PTR_ERR_OR_ZERO(codec); #endif -- 2.25.1

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