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[PATCH 0/7] ASoC: mediatek: mt8188: revise AFE driver
by Trevor Wu 20 Apr '23

20 Apr '23

The series of patches consists of three major changes. First, remove redundant supply for ADDA DAI dirver. Second, revise ETDM control including APLL dynamic switch via DAPM, so APLL can be enabled when it is really required. Finally, AFE probe function is updated. Bus protection change was dropped at the previous patch because the dependent change was not accepted. Trevor Wu (7): ASoC: mediatek: mt8188: remove supply AUDIO_HIRES ASoC: mediatek: mt8188: complete set_tdm_slot function ASoC: mediatek: mt8188: revise ETDM control flow ASoC: mediatek: mt8188: refine APLL control ASoC: mediatek: mt8188: combine afe component registration ASoC: mediatek: mt8188: add bus protection ASoC: dt-bindings: mediatek,mt8188-afe: add audio properties .../bindings/sound/mediatek,mt8188-afe.yaml | 18 + sound/soc/mediatek/mt8188/mt8188-afe-clk.c | 138 ++- sound/soc/mediatek/mt8188/mt8188-afe-clk.h | 11 + sound/soc/mediatek/mt8188/mt8188-afe-pcm.c | 113 ++- sound/soc/mediatek/mt8188/mt8188-dai-adda.c | 37 - sound/soc/mediatek/mt8188/mt8188-dai-etdm.c | 922 ++++++++++-------- sound/soc/mediatek/mt8188/mt8188-reg.h | 2 + 7 files changed, 759 insertions(+), 482 deletions(-) -- 2.18.0

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[GIT PULL] ASoC fixes for v6.3-rc7
by Mark Brown 20 Apr '23

20 Apr '23

The following changes since commit e3720f92e0237921da537e47a0b24e27899203f8: ASoC: SOF: avoid a NULL dereference with unsupported widgets (2023-03-29 14:40:18 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.3-rc7 for you to fetch changes up to 86a24e99c97234f87d9f70b528a691150e145197: ASoC: fsl_asrc_dma: fix potential null-ptr-deref (2023-04-19 13:42:56 +0100) ---------------------------------------------------------------- ASoC: Fixes for v6.3 A few remaining small fixes for v6.3, all small driver specific ones. ---------------------------------------------------------------- Chancel Liu (1): ASoC: fsl_sai: Fix pins setting for i.MX8QM platform Daniel Baluta (1): ASoC: SOF: pm: Tear down pipelines only if DSP was active Long Wang (1): ASoC: max98373: change power down sequence for smart amp Nikita Zhandarovich (1): ASoC: fsl_asrc_dma: fix potential null-ptr-deref Peter Ujfalusi (1): ASoC: SOF: ipc4-topology: Clarify bind failure caused by missing fw_module sound/soc/codecs/max98373.c | 4 ++-- sound/soc/fsl/fsl_asrc_dma.c | 11 ++++++++--- sound/soc/fsl/fsl_sai.c | 2 +- sound/soc/sof/ipc4-topology.c | 10 ++++++---- sound/soc/sof/pm.c | 8 +++++++- 5 files changed, 24 insertions(+), 11 deletions(-)

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[PATCH] ALSA: hda/realtek: fix mute/micmute LEDs for a HP ProBook
by Andy Chi 20 Apr '23

20 Apr '23

There is a HP ProBook 455 G10 which using ALC236 codec and need the ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF quirk to make mute LED and micmute LED work. Signed-off-by: Andy Chi <andy.chi(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6a6c72b5ea26..f70d6a33421d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9468,6 +9468,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8b47, "HP", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8b5d, "HP", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), SND_PCI_QUIRK(0x103c, 0x8b5e, "HP", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), + SND_PCI_QUIRK(0x103c, 0x8b65, "HP ProBook 455 15.6 inch G10 Notebook PC", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), SND_PCI_QUIRK(0x103c, 0x8b66, "HP", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), SND_PCI_QUIRK(0x103c, 0x8b7a, "HP", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8b7d, "HP", ALC236_FIXUP_HP_GPIO_LED), -- 2.34.1

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[PATCH] Revert "ASoC: fsl: remove unnecessary dai_link->platform"
by Shengjiu Wang 20 Apr '23

20 Apr '23

This reverts commit 33683cbf49b5412061cb1e4c876063fdef86def4. dai_link->platform is needed. The platform component is "snd_dmaengine_pcm", which is registered from cpu driver, If dai_link->platform is not assigned, then platform component will not be probed, then there will be issue: aplay: main:831: audio open error: Invalid argument Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/imx-audmix.c | 14 ++++++++++---- sound/soc/fsl/imx-spdif.c | 5 ++++- 2 files changed, 14 insertions(+), 5 deletions(-) diff --git a/sound/soc/fsl/imx-audmix.c b/sound/soc/fsl/imx-audmix.c index 2c57fe9d2d08..1292a845c424 100644 --- a/sound/soc/fsl/imx-audmix.c +++ b/sound/soc/fsl/imx-audmix.c @@ -207,8 +207,8 @@ static int imx_audmix_probe(struct platform_device *pdev) for (i = 0; i < num_dai; i++) { struct snd_soc_dai_link_component *dlc; - /* for CPU/Codec x 2 */ - dlc = devm_kcalloc(&pdev->dev, 4, sizeof(*dlc), GFP_KERNEL); + /* for CPU/Codec/Platform x 2 */ + dlc = devm_kcalloc(&pdev->dev, 6, sizeof(*dlc), GFP_KERNEL); if (!dlc) return -ENOMEM; @@ -240,9 +240,11 @@ static int imx_audmix_probe(struct platform_device *pdev) priv->dai[i].cpus = &dlc[0]; priv->dai[i].codecs = &dlc[1]; + priv->dai[i].platforms = &dlc[2]; priv->dai[i].num_cpus = 1; priv->dai[i].num_codecs = 1; + priv->dai[i].num_platforms = 1; priv->dai[i].name = dai_name; priv->dai[i].stream_name = "HiFi-AUDMIX-FE"; @@ -250,6 +252,7 @@ static int imx_audmix_probe(struct platform_device *pdev) priv->dai[i].codecs->name = "snd-soc-dummy"; priv->dai[i].cpus->of_node = args.np; priv->dai[i].cpus->dai_name = dev_name(&cpu_pdev->dev); + priv->dai[i].platforms->of_node = args.np; priv->dai[i].dynamic = 1; priv->dai[i].dpcm_playback = 1; priv->dai[i].dpcm_capture = (i == 0 ? 1 : 0); @@ -264,17 +267,20 @@ static int imx_audmix_probe(struct platform_device *pdev) be_cp = devm_kasprintf(&pdev->dev, GFP_KERNEL, "AUDMIX-Capture-%d", i); - priv->dai[num_dai + i].cpus = &dlc[2]; - priv->dai[num_dai + i].codecs = &dlc[3]; + priv->dai[num_dai + i].cpus = &dlc[3]; + priv->dai[num_dai + i].codecs = &dlc[4]; + priv->dai[num_dai + i].platforms = &dlc[5]; priv->dai[num_dai + i].num_cpus = 1; priv->dai[num_dai + i].num_codecs = 1; + priv->dai[num_dai + i].num_platforms = 1; priv->dai[num_dai + i].name = be_name; priv->dai[num_dai + i].codecs->dai_name = "snd-soc-dummy-dai"; priv->dai[num_dai + i].codecs->name = "snd-soc-dummy"; priv->dai[num_dai + i].cpus->of_node = audmix_np; priv->dai[num_dai + i].cpus->dai_name = be_name; + priv->dai[num_dai + i].platforms->name = "snd-soc-dummy"; priv->dai[num_dai + i].no_pcm = 1; priv->dai[num_dai + i].dpcm_playback = 1; priv->dai[num_dai + i].dpcm_capture = 1; diff --git a/sound/soc/fsl/imx-spdif.c b/sound/soc/fsl/imx-spdif.c index 114b49660193..4446fba755b9 100644 --- a/sound/soc/fsl/imx-spdif.c +++ b/sound/soc/fsl/imx-spdif.c @@ -26,7 +26,7 @@ static int imx_spdif_audio_probe(struct platform_device *pdev) } data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL); - comp = devm_kzalloc(&pdev->dev, 2 * sizeof(*comp), GFP_KERNEL); + comp = devm_kzalloc(&pdev->dev, 3 * sizeof(*comp), GFP_KERNEL); if (!data || !comp) { ret = -ENOMEM; goto end; @@ -34,15 +34,18 @@ static int imx_spdif_audio_probe(struct platform_device *pdev) data->dai.cpus = &comp[0]; data->dai.codecs = &comp[1]; + data->dai.platforms = &comp[2]; data->dai.num_cpus = 1; data->dai.num_codecs = 1; + data->dai.num_platforms = 1; data->dai.name = "S/PDIF PCM"; data->dai.stream_name = "S/PDIF PCM"; data->dai.codecs->dai_name = "snd-soc-dummy-dai"; data->dai.codecs->name = "snd-soc-dummy"; data->dai.cpus->of_node = spdif_np; + data->dai.platforms->of_node = spdif_np; data->dai.playback_only = true; data->dai.capture_only = true; -- 2.34.1

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[PATCH] ASoC: nau8825: fix delay time range check
by Arnd Bergmann 19 Apr '23

19 Apr '23

From: Arnd Bergmann <arnd(a)arndb.de> clang points out that the recently added range check is nonsensical: sound/soc/codecs/nau8825.c:2826:31: error: overlapping comparisons always evaluate to false [-Werror,-Wtautological-overlap-compare] if (nau8825->adc_delay < 125 && nau8825->adc_delay > 500) ~~~~~~~~~~~~~~~~~~~~~~~~~^~~~~~~~~~~~~~~~~~~~~~~~~~~ The DT binding document confirms that the intention is to warn if it's outside of the 125..500 range, so do that instead. Fixes: fc0b096c9291 ("ASoC: nau8825: Add delay control for input path") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/codecs/nau8825.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index c4389f5fe603..f4eb999761a4 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -2823,7 +2823,7 @@ static int nau8825_read_device_properties(struct device *dev, ret = device_property_read_u32(dev, "nuvoton,adc-delay-ms", &nau8825->adc_delay); if (ret) nau8825->adc_delay = 125; - if (nau8825->adc_delay < 125 && nau8825->adc_delay > 500) + if (nau8825->adc_delay < 125 || nau8825->adc_delay > 500) dev_warn(dev, "Please set the suitable delay time!\n"); nau8825->mclk = devm_clk_get(dev, "mclk"); -- 2.39.2

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[PATCH] ASoC: fsl: Simplify an error message
by Christophe JAILLET 19 Apr '23

19 Apr '23

dev_err_probe() already display the error code. There is no need to duplicate it explicitly in the error message. Signed-off-by: Christophe JAILLET <christophe.jaillet(a)wanadoo.fr> --- sound/soc/fsl/fsl-asoc-card.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index bffa1048d31e..40870668ee24 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -858,7 +858,7 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) ret = devm_snd_soc_register_card(&pdev->dev, &priv->card); if (ret) { - dev_err_probe(&pdev->dev, ret, "snd_soc_register_card failed: %d\n", ret); + dev_err_probe(&pdev->dev, ret, "snd_soc_register_card failed\n"); goto asrc_fail; } -- 2.34.1

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[PATCH] ASoC: fsl_asrc_dma: fix potential null-ptr-deref
by Nikita Zhandarovich 19 Apr '23

19 Apr '23

dma_request_slave_channel() may return NULL which will lead to NULL pointer dereference error in 'tmp_chan->private'. Correct this behaviour by, first, switching from deprecated function dma_request_slave_channel() to dma_request_chan(). Secondly, enable sanity check for the resuling value of dma_request_chan(). Also, fix description that follows the enacted changes and that concerns the use of dma_request_slave_channel(). Fixes: 706e2c881158 ("ASoC: fsl_asrc_dma: Reuse the dma channel if available in Back-End") Co-developed-by: Natalia Petrova <n.petrova(a)fintech.ru> Signed-off-by: Nikita Zhandarovich <n.zhandarovich(a)fintech.ru> --- sound/soc/fsl/fsl_asrc_dma.c | 11 ++++++++--- 1 file changed, 8 insertions(+), 3 deletions(-) diff --git a/sound/soc/fsl/fsl_asrc_dma.c b/sound/soc/fsl/fsl_asrc_dma.c index 3b81a465814a..05a7d1588d20 100644 --- a/sound/soc/fsl/fsl_asrc_dma.c +++ b/sound/soc/fsl/fsl_asrc_dma.c @@ -209,14 +209,19 @@ static int fsl_asrc_dma_hw_params(struct snd_soc_component *component, be_chan = soc_component_to_pcm(component_be)->chan[substream->stream]; tmp_chan = be_chan; } - if (!tmp_chan) - tmp_chan = dma_request_slave_channel(dev_be, tx ? "tx" : "rx"); + if (!tmp_chan) { + tmp_chan = dma_request_chan(dev_be, tx ? "tx" : "rx"); + if (IS_ERR(tmp_chan)) { + dev_err(dev, "failed to request DMA channel for Back-End\n"); + return -EINVAL; + } + } /* * An EDMA DEV_TO_DEV channel is fixed and bound with DMA event of each * peripheral, unlike SDMA channel that is allocated dynamically. So no * need to configure dma_request and dma_request2, but get dma_chan of - * Back-End device directly via dma_request_slave_channel. + * Back-End device directly via dma_request_chan. */ if (!asrc->use_edma) { /* Get DMA request of Back-End */

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[PATCH] ASoC: fsl_sai: Fix pins setting for i.MX8QM platform
by Chancel Liu 19 Apr '23

19 Apr '23

SAI on i.MX8QM platform supports the data lines up to 4. So the pins setting should be corrected to 4. Fixes: eba0f0077519 ("ASoC: fsl_sai: Enable combine mode soft") Signed-off-by: Chancel Liu <chancel.liu(a)nxp.com> --- sound/soc/fsl/fsl_sai.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 07d13dca852e..abdaffb00fbd 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -1544,7 +1544,7 @@ static const struct fsl_sai_soc_data fsl_sai_imx8qm_data = { .use_imx_pcm = true, .use_edma = true, .fifo_depth = 64, - .pins = 1, + .pins = 4, .reg_offset = 0, .mclk0_is_mclk1 = false, .flags = 0, -- 2.25.1

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snd-firewire-ctl-services 0.1.0 released
by Takashi Sakamoto 19 Apr '23

19 Apr '23

Hi all, This is a short announcement about a new release of snd-firewire-ctl-services project. After development for mostly three years[1], version 0.1.0 is released today. If you have FireWire audio device and interests in ALSA support, the release would be worth to check out. * https://github.com/alsa-project/snd-firewire-ctl-services/releases/tag/snd-… The project provides user space service programs to operate digital signal processing function in devices supported by ALSA firewire stack. If using such device, you can configure DSP functions such as the volumes of device. For some device, effects such as dynamics and equalizers are also configurable. Available devices are listed in README.rst of the project. The service programs work as 'message broaker'. ALSA control applications are available for end-user interface to communicate to the service programs[2]. The device connected to IEEE 1394 bus is the peer. The service programs receives messages from both ends, then translates and convert the messages to send to the opposite end. Against your expectation, the project provides neither GUI programs nor end-user interfaces. I know that the user experience is heavily on look-and-feel on the GUI programs, while it is out of the aim of project. I daily use quashctl in QasTools[3] and it is pretty good to me for such ALSA control application. If encountering any issue, please file it to github repository[4]. README should be helpful to new users. Thanks for your support and long patience to ALSA firewire stack. [1] https://lists.linuxaudio.org/hyperkitty/list/linux-audio-dev@lists.linuxaud… [2] For TASCAM FireWire series, ALSA sequencer application is such end as well to operate control surface. [3] https://gitlab.com/sebholt/qastools [4] https://github.com/alsa-project/snd-firewire-ctl-services Regards Takashi Sakamoto

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[PATCH] ALSA: hda/realtek: Remove specific patch for Dell Precision 3260
by Jaroslav Kysela 19 Apr '23

19 Apr '23

Unfortunately, the tester gave a weak feedback (working/non-working) for this case. After the double confirmation, this change is not really required. The standard code with alc269_fallback_pin_fixup_tbl should work on this hardware. Fixes: 5911d78fabbb ("ALSA: hda/realtek: Improve support for Dell Precision 3260") Fixes: 5f4efc9dfcfd ("ALSA: hda/realtek: Fix support for Dell Precision 3260") Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> --- sound/pci/hda/patch_realtek.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 26187f5d56b5..50b8573b5206 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9263,7 +9263,6 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0a62, "Dell Precision 5560", ALC289_FIXUP_DUAL_SPK), SND_PCI_QUIRK(0x1028, 0x0a9d, "Dell Latitude 5430", ALC269_FIXUP_DELL4_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0a9e, "Dell Latitude 5430", ALC269_FIXUP_DELL4_MIC_NO_PRESENCE), - SND_PCI_QUIRK(0x1028, 0x0ac9, "Dell Precision 3260", ALC283_FIXUP_CHROME_BOOK), SND_PCI_QUIRK(0x1028, 0x0b19, "Dell XPS 15 9520", ALC289_FIXUP_DUAL_SPK), SND_PCI_QUIRK(0x1028, 0x0b1a, "Dell Precision 5570", ALC289_FIXUP_DUAL_SPK), SND_PCI_QUIRK(0x1028, 0x0b37, "Dell Inspiron 16 Plus 7620 2-in-1", ALC295_FIXUP_DELL_INSPIRON_TOP_SPEAKERS), -- 2.39.2

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