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

[PATCH 0/2] Enable DMA mode on Intel Keem Bay platform
by Michael Sit Wei Hong 13 Jan '21

13 Jan '21

v1: Initial patch version, to enable DMA mode on Intel Keembay platform. Michael Sit Wei Hong (2): dt-bindings: sound: intel, keembay-i2s: Add info for device to use DMA ASoC: Intel: KMB: Enable DMA transfer mode .../bindings/sound/intel,keembay-i2s.yaml | 14 ++ sound/soc/intel/Kconfig | 2 + sound/soc/intel/keembay/kmb_platform.c | 157 ++++++++++++++++-- sound/soc/intel/keembay/kmb_platform.h | 9 + 4 files changed, 167 insertions(+), 15 deletions(-) -- 2.17.1

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[PATCH] ASoC: qcom: qdsp6: Move frontend AIFs to q6asm-dai
by Stephan Gerhold 13 Jan '21

13 Jan '21

At the moment it is necessary to set up the DAPM routes between front-end AIF<->DAI explicitly in the device tree, e.g. using audio-routing = "MM_DL1", "MultiMedia1 Playback", "MM_DL3", "MultiMedia3 Playback", "MM_DL4", "MultiMedia4 Playback", "MultiMedia2 Capture", "MM_UL2"; This is prone to mistakes and (sadly) there is no clear error if one of these routes is missing. :( Actually, this should not be necessary because the ASoC core normally automatically links AIF<->DAI within snd_soc_dapm_link_dai_widgets(). This is done using the "stname" parameter of SND_SOC_DAPM_AIF_IN/OUT. For SND_SOC_DAPM_AIF_IN("MM_DL1", "MultiMedia1 Playback", 0, 0, 0, 0), it should create the route from above: MM_DL1 <-> MultiMedia1 Playback. This does not work at the moment because the AIF widget (MM_DL1) and the DAI widget (MultiMedia1 Playback) belong to different DAPM contexts (q6routing / q6asm-dai). Fix this by declaring the AIF widgets in the same driver as the DAIs (q6asm-dai). Now the routes above are created automatically and no longer need to be specified in the device tree. This is also more consistent with the back-end AIFs which are already declared in q6afe-dais instead of q6routing. q6routing should only link the components together using mixers. Cc: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Fixes: 2a9e92d371db ("ASoC: qdsp6: q6asm: Add q6asm dai driver") Signed-off-by: Stephan Gerhold <stephan(a)gerhold.net> --- sound/soc/qcom/qdsp6/q6asm-dai.c | 21 +++++++++++++++++++++ sound/soc/qcom/qdsp6/q6routing.c | 18 ------------------ 2 files changed, 21 insertions(+), 18 deletions(-) diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c index c9ac9c1d26c4..9766725c2916 100644 --- a/sound/soc/qcom/qdsp6/q6asm-dai.c +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c @@ -1233,6 +1233,25 @@ static void q6asm_dai_pcm_free(struct snd_soc_component *component, } } +static const struct snd_soc_dapm_widget q6asm_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN("MM_DL1", "MultiMedia1 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("MM_DL2", "MultiMedia2 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("MM_DL3", "MultiMedia3 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("MM_DL4", "MultiMedia4 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("MM_DL5", "MultiMedia5 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("MM_DL6", "MultiMedia6 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("MM_DL7", "MultiMedia7 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("MM_DL8", "MultiMedia8 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("MM_UL1", "MultiMedia1 Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("MM_UL2", "MultiMedia2 Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("MM_UL3", "MultiMedia3 Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("MM_UL4", "MultiMedia4 Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("MM_UL5", "MultiMedia5 Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("MM_UL6", "MultiMedia6 Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("MM_UL7", "MultiMedia7 Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("MM_UL8", "MultiMedia8 Capture", 0, SND_SOC_NOPM, 0, 0), +}; + static const struct snd_soc_component_driver q6asm_fe_dai_component = { .name = DRV_NAME, .open = q6asm_dai_open, @@ -1245,6 +1264,8 @@ static const struct snd_soc_component_driver q6asm_fe_dai_component = { .pcm_construct = q6asm_dai_pcm_new, .pcm_destruct = q6asm_dai_pcm_free, .compress_ops = &q6asm_dai_compress_ops, + .dapm_widgets = q6asm_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(q6asm_dapm_widgets), }; static struct snd_soc_dai_driver q6asm_fe_dais_template[] = { diff --git a/sound/soc/qcom/qdsp6/q6routing.c b/sound/soc/qcom/qdsp6/q6routing.c index 53185e26fea1..0a6b9433f6ac 100644 --- a/sound/soc/qcom/qdsp6/q6routing.c +++ b/sound/soc/qcom/qdsp6/q6routing.c @@ -713,24 +713,6 @@ static const struct snd_kcontrol_new mmul8_mixer_controls[] = { Q6ROUTING_TX_MIXERS(MSM_FRONTEND_DAI_MULTIMEDIA8) }; static const struct snd_soc_dapm_widget msm_qdsp6_widgets[] = { - /* Frontend AIF */ - SND_SOC_DAPM_AIF_IN("MM_DL1", "MultiMedia1 Playback", 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("MM_DL2", "MultiMedia2 Playback", 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("MM_DL3", "MultiMedia3 Playback", 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("MM_DL4", "MultiMedia4 Playback", 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("MM_DL5", "MultiMedia5 Playback", 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("MM_DL6", "MultiMedia6 Playback", 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("MM_DL7", "MultiMedia7 Playback", 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("MM_DL8", "MultiMedia8 Playback", 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("MM_UL1", "MultiMedia1 Capture", 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("MM_UL2", "MultiMedia2 Capture", 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("MM_UL3", "MultiMedia3 Capture", 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("MM_UL4", "MultiMedia4 Capture", 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("MM_UL5", "MultiMedia5 Capture", 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("MM_UL6", "MultiMedia6 Capture", 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("MM_UL7", "MultiMedia7 Capture", 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("MM_UL8", "MultiMedia8 Capture", 0, 0, 0, 0), - /* Mixer definitions */ SND_SOC_DAPM_MIXER("HDMI Mixer", SND_SOC_NOPM, 0, 0, hdmi_mixer_controls, -- 2.29.2

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[PATCH] ASoC: cs42l56: fix up error handling in probe
by Dan Carpenter 13 Jan '21

13 Jan '21

There are two issues with this code. The first error path forgot to set the error code and instead returns success. The second error path doesn't clean up. Fixes: 272b5edd3b8f ("ASoC: Add support for CS42L56 CODEC") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/soc/codecs/cs42l56.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs42l56.c b/sound/soc/codecs/cs42l56.c index bb9599cc832b..c44a5cdb796e 100644 --- a/sound/soc/codecs/cs42l56.c +++ b/sound/soc/codecs/cs42l56.c @@ -1250,6 +1250,7 @@ static int cs42l56_i2c_probe(struct i2c_client *i2c_client, dev_err(&i2c_client->dev, "CS42L56 Device ID (%X). Expected %X\n", devid, CS42L56_DEVID); + ret = -EINVAL; goto err_enable; } alpha_rev = reg & CS42L56_AREV_MASK; @@ -1307,7 +1308,7 @@ static int cs42l56_i2c_probe(struct i2c_client *i2c_client, ret = devm_snd_soc_register_component(&i2c_client->dev, &soc_component_dev_cs42l56, &cs42l56_dai, 1); if (ret < 0) - return ret; + goto err_enable; return 0; -- 2.29.2

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[PATCH 0/6] ASoC: soc-pcm: cleanup code
by Kuninori Morimoto 13 Jan '21

13 Jan '21

Hi Mark These are not so important, but for soc-pcm cleanup patches. Kuninori Morimoto (6): ASoC: soc-pcm: move dpcm_set_fe_update_state() ASoC: soc-pcm: add dpcm_set_be_update_state() ASoC: soc-pcm: add soc_pcm_set_dai_params() ASoC: soc-pcm: cleanup soc_pcm_apply_symmetry() ASoC: soc-pcm: cleanup soc_pcm_params_symmetry() ASoC: soc-pcm: setup pcm at one place in soc_new_pcm() sound/soc/soc-pcm.c | 231 +++++++++++++++++--------------------------- 1 file changed, 90 insertions(+), 141 deletions(-) -- 2.25.1 Thank you for your help !! Best regards --- Kuninori Morimoto

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[PATCH v2 2/2] ASoC: rt715:add Mic Mute LED control support
by Perry Yuan 13 Jan '21

13 Jan '21

From: Perry Yuan <perry_yuan(a)dell.com> Some new Dell system is going to support audio internal micphone privacy setting from hardware level with micmute led state changing When micmute hotkey pressed by user, soft mute will need to be enabled firstly in case of pop noise, and codec driver need to react to mic mute event to EC notifying that SW mute is completed. Then EC will do the HW mute physically. This patch allow codec rt715 driver to ack EC when micmute key pressed through this micphone led control interface like hda_generic provided. Signed-off-by: Perry Yuan <perry_yuan(a)dell.com> Signed-off-by: Limonciello Mario <mario_limonciello(a)dell.com> --- sound/soc/codecs/rt715-sdca.c | 19 ++++++++++++++++++- sound/soc/codecs/rt715-sdca.h | 1 + sound/soc/codecs/rt715.c | 19 ++++++++++++++++++- sound/soc/codecs/rt715.h | 1 + 4 files changed, 38 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt715-sdca.c b/sound/soc/codecs/rt715-sdca.c index b43ac8559e45..e168ef6efcf5 100644 --- a/sound/soc/codecs/rt715-sdca.c +++ b/sound/soc/codecs/rt715-sdca.c @@ -12,6 +12,7 @@ #include <linux/version.h> #include <linux/kernel.h> #include <linux/init.h> +#include <linux/leds.h> #include <linux/pm_runtime.h> #include <linux/pm.h> #include <linux/soundwire/sdw.h> @@ -268,6 +269,7 @@ static int rt715_sdca_put_volsw(struct snd_kcontrol *kcontrol, unsigned int reg2 = mc->rreg; unsigned int reg = mc->reg; unsigned int max = mc->max; + unsigned int val0, val1; int err; val = ucontrol->value.integer.value[0]; @@ -286,7 +288,22 @@ static int rt715_sdca_put_volsw(struct snd_kcontrol *kcontrol, if (err < 0) return err; } - +#if IS_ENABLED(CONFIG_DELL_PRIVACY) + /* Privacy LED Trigger State Changed by muted/unmute switch */ + if (mc->invert) { + val0 = ucontrol->value.integer.value[0]; + val1 = ucontrol->value.integer.value[1]; + if (val0 == 1 && val1 == 1) { + rt715->micmute_led = LED_OFF; + ledtrig_audio_set(LED_AUDIO_MICMUTE, + rt715->micmute_led ? LED_ON : LED_OFF); + } else if (val0 == 0 && val1 == 0) { + rt715->micmute_led = LED_ON; + ledtrig_audio_set(LED_AUDIO_MICMUTE, + rt715->micmute_led ? LED_ON : LED_OFF); + } + } +#endif return 0; } diff --git a/sound/soc/codecs/rt715-sdca.h b/sound/soc/codecs/rt715-sdca.h index 840c237895dd..2ab8724ae50b 100644 --- a/sound/soc/codecs/rt715-sdca.h +++ b/sound/soc/codecs/rt715-sdca.h @@ -31,6 +31,7 @@ struct rt715_sdca_priv { int l_is_unmute; int r_is_unmute; int hw_sdw_ver; + int micmute_led; }; struct rt715_sdw_stream_data { diff --git a/sound/soc/codecs/rt715.c b/sound/soc/codecs/rt715.c index cdcba70146da..faa4dee6b39a 100644 --- a/sound/soc/codecs/rt715.c +++ b/sound/soc/codecs/rt715.c @@ -13,6 +13,7 @@ #include <linux/init.h> #include <linux/delay.h> #include <linux/i2c.h> +#include <linux/leds.h> #include <linux/pm_runtime.h> #include <linux/pm.h> #include <linux/soundwire/sdw.h> @@ -88,13 +89,29 @@ static int rt715_set_amp_gain_put(struct snd_kcontrol *kcontrol, RT715_SET_GAIN_MIX_ADC2_L}; unsigned int addr_h, addr_l, val_h, val_ll, val_lr; unsigned int read_ll, read_rl, i, j, loop_cnt; + unsigned int val0, val1; if (strstr(ucontrol->id.name, "Main Capture Switch") || strstr(ucontrol->id.name, "Main Capture Volume")) loop_cnt = 4; else loop_cnt = 1; - +#if IS_ENABLED(CONFIG_DELL_PRIVACY) + /* Privacy micmute led trigger for muted/unmute switch */ + if (mc->invert) { + val0 = ucontrol->value.integer.value[0]; + val1 = ucontrol->value.integer.value[1]; + if (val0 == 1 && val1 == 1) { + rt715->micmute_led = LED_OFF; + ledtrig_audio_set(LED_AUDIO_MICMUTE, + rt715->micmute_led ? LED_ON : LED_OFF); + } else if (val0 == 0 && val1 == 0) { + rt715->micmute_led = LED_ON; + ledtrig_audio_set(LED_AUDIO_MICMUTE, + rt715->micmute_led ? LED_ON : LED_OFF); + } + } +#endif for (j = 0; j < loop_cnt; j++) { /* Can't use update bit function, so read the original value first */ if (loop_cnt == 1) { diff --git a/sound/soc/codecs/rt715.h b/sound/soc/codecs/rt715.h index 009a8266f606..2d3b5b299514 100644 --- a/sound/soc/codecs/rt715.h +++ b/sound/soc/codecs/rt715.h @@ -22,6 +22,7 @@ struct rt715_priv { struct sdw_bus_params params; bool hw_init; bool first_hw_init; + int micmute_led; }; struct sdw_stream_data { -- 2.25.1

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[PATCH] ASoC: imx-hdmi: Fix warning of the uninitialized variable ret
by Shengjiu Wang 12 Jan '21

12 Jan '21

From: shengjiu wang <shengjiu.wang(a)nxp.com> When condition ((hdmi_out && hdmi_in) || (!hdmi_out && !hdmi_in)) is true, then goto fail, the uninitialized variable ret will be returned. Signed-off-by: shengjiu wang <shengjiu.wang(a)nxp.com> Reported-by: kernel test robot <lkp(a)intel.com> --- sound/soc/fsl/imx-hdmi.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/imx-hdmi.c b/sound/soc/fsl/imx-hdmi.c index 2c2a76a71940..ede4a9ad1054 100644 --- a/sound/soc/fsl/imx-hdmi.c +++ b/sound/soc/fsl/imx-hdmi.c @@ -164,6 +164,7 @@ static int imx_hdmi_probe(struct platform_device *pdev) if ((hdmi_out && hdmi_in) || (!hdmi_out && !hdmi_in)) { dev_err(&pdev->dev, "Invalid HDMI DAI link\n"); + ret = -EINVAL; goto fail; } -- 2.17.1

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SoF support for 8086:9d71?
by Kai-Heng Feng 12 Jan '21

12 Jan '21

Hi, On more recent kernels legacy HDA stops working for device 8086:9d71: [ 47.211381] snd_hda_intel 0000:00:1f.3: DSP detected with PCI class/subclass/prog-if info 0x040100 [ 47.211383] snd_hda_intel 0000:00:1f.3: Digital mics found on Skylake+ platform, using SST driver However, only legacy HDA supports it: 00:1f.3 Multimedia audio controller [0401]: Intel Corporation Sunrise Point-LP HD Audio [8086:9d71] (rev 21) .. Kernel modules: snd_hda_intel Ubuntu kernel doesn't select SND_SOC_INTEL_SKYLAKE, which builds snd-soc-skl, because based on the Kconfig description SOF should support it. I wonder if SND_SOC_INTEL_SKYLAKE should be select again? BugLink: <https://bugs.launchpad.net/bugs/1902457>

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[PATCH] ASoC: Intel: bytcr_rt5640: Add quirks for DeeQ X5-Z8300
by Chris Chiu 11 Jan '21

11 Jan '21

DeeQ X5-Z8300 is a laptop empowered by Intel Atom Cherry Trail. Add quirks to select the correct input map, jack-detect options to enable jack sensing and internal/headset microphones. Signed-off-by: Chris Chiu <chiu(a)endlessos.org> --- sound/soc/intel/boards/bytcr_rt5651.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index 688b5e0a49e3..9df09e26b05a 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -507,6 +507,14 @@ static const struct dmi_system_id byt_rt5651_quirk_table[] = { BYT_RT5651_SSP0_AIF1 | BYT_RT5651_MONO_SPEAKER), }, + { + .callback = byt_rt5651_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "DeeQ"), + DMI_MATCH(DMI_PRODUCT_NAME, "X5-Z8300"), + }, + .driver_data = (void *)(BYT_RT5651_IN1_IN2_MAP | BYT_RT5651_JD1_1), + }, {} }; -- 2.20.1

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[PATCH] ASoC: audio-graph-card: Drop remote-endpoint as required property
by Sameer Pujar 11 Jan '21

11 Jan '21

The remote-endpoint may not be available if it is part of some pluggable module. One such example would be an audio card, the Codec endpoint will not be available until it is plugged in. Hence drop 'remote-endpoint' as a required property. Cc: Rob Herring <robh(a)kernel.org> Cc: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Sameer Pujar <spujar(a)nvidia.com> --- Documentation/devicetree/bindings/sound/audio-graph-port.yaml | 3 --- 1 file changed, 3 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/audio-graph-port.yaml b/Documentation/devicetree/bindings/sound/audio-graph-port.yaml index 2005014..766e910 100644 --- a/Documentation/devicetree/bindings/sound/audio-graph-port.yaml +++ b/Documentation/devicetree/bindings/sound/audio-graph-port.yaml @@ -71,9 +71,6 @@ properties: description: CPU to Codec rate channels. $ref: /schemas/types.yaml#/definitions/uint32 - required: - - remote-endpoint - ports: description: multi OF-Graph subnode type: object -- 2.7.4

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[PATCH 0/2] fix tegra-hda on tegra30 devices
by Peter Geis 08 Jan '21

08 Jan '21

The following patches fix tegra-hda on legacy tegra devices. Two issues were discovered preventing tegra-hda from functioning: The hda clocks on tegra30 were assigned to clk_m and running at too low of a rate to function. The tegra-hda encounters an input/output error when opening a stream. Since the only mainline device that used tegra-hda previously was the t124, it is unknown exactly when this was broken. Fortunately a recent patch was submitted that fixed the issue only on t194 devices. We can apply it universally to the tegra-hda device to resolve the issues across the board. Note that downstream devices used the spdif device instead of hda for hdmi audio. The spdif device lacks a driver on mainline. Peter Geis (2): clk: tegra30: Add hda clock default rates to clock driver ALSA: hda/tegra: fix tegra-hda on tegra30 soc drivers/clk/tegra/clk-tegra30.c | 2 ++ sound/pci/hda/hda_tegra.c | 3 +-- 2 files changed, 3 insertions(+), 2 deletions(-) -- 2.25.1

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