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[PATCHv4 2/9] ASoC: fsl-asoc-card: add second dai link component for codecs
by Elinor Montmasson 15 May '24

15 May '24

Add a second dai link component for codecs that will be used for the generic codec use case. It will use spdif_receiver and spdif_transmitter drivers as dummy codec drivers, needing 2 codecs slots for the links. To prevent deferring in use cases using only one codec, also set by default the number of codecs to 1 for the relevant dai links. Signed-off-by: Elinor Montmasson <elinor.montmasson(a)savoirfairelinux.com> Co-authored-by: Philip-Dylan Gleonec <philip-dylan.gleonec(a)savoirfairelinux.com> --- sound/soc/fsl/fsl-asoc-card.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 8a2a6e5461dc..c83492e7cec2 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -296,7 +296,7 @@ static int be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, SND_SOC_DAILINK_DEFS(hifi, DAILINK_COMP_ARRAY(COMP_EMPTY()), - DAILINK_COMP_ARRAY(COMP_EMPTY()), + DAILINK_COMP_ARRAY(COMP_EMPTY(), COMP_EMPTY()), DAILINK_COMP_ARRAY(COMP_EMPTY())); SND_SOC_DAILINK_DEFS(hifi_fe, @@ -306,7 +306,7 @@ SND_SOC_DAILINK_DEFS(hifi_fe, SND_SOC_DAILINK_DEFS(hifi_be, DAILINK_COMP_ARRAY(COMP_EMPTY()), - DAILINK_COMP_ARRAY(COMP_EMPTY())); + DAILINK_COMP_ARRAY(COMP_EMPTY(), COMP_EMPTY())); static const struct snd_soc_dai_link fsl_asoc_card_dai[] = { /* Default ASoC DAI Link*/ @@ -618,6 +618,8 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) memcpy(priv->dai_link, fsl_asoc_card_dai, sizeof(struct snd_soc_dai_link) * ARRAY_SIZE(priv->dai_link)); + priv->dai_link[0].num_codecs = 1; + priv->dai_link[2].num_codecs = 1; priv->card.dapm_routes = audio_map; priv->card.num_dapm_routes = ARRAY_SIZE(audio_map); -- 2.34.1

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[PATCHv4 1/9] ASoC: fsl-asoc-card: add support for dai links with multiple codecs
by Elinor Montmasson 15 May '24

15 May '24

Add support for dai links using multiple codecs for multi-codec use cases. Signed-off-by: Elinor Montmasson <elinor.montmasson(a)savoirfairelinux.com> Co-authored-by: Philip-Dylan Gleonec <philip-dylan.gleonec(a)savoirfairelinux.com> --- sound/soc/fsl/fsl-asoc-card.c | 22 ++++++++++++++-------- 1 file changed, 14 insertions(+), 8 deletions(-) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 5ddc0c2fe53f..8a2a6e5461dc 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -815,10 +815,10 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) /* Normal DAI Link */ priv->dai_link[0].cpus->of_node = cpu_np; - priv->dai_link[0].codecs->dai_name = codec_dai_name; + priv->dai_link[0].codecs[0].dai_name = codec_dai_name; if (!fsl_asoc_card_is_ac97(priv)) - priv->dai_link[0].codecs->of_node = codec_np; + priv->dai_link[0].codecs[0].of_node = codec_np; else { u32 idx; @@ -829,11 +829,11 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) goto asrc_fail; } - priv->dai_link[0].codecs->name = + priv->dai_link[0].codecs[0].name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "ac97-codec.%u", (unsigned int)idx); - if (!priv->dai_link[0].codecs->name) { + if (!priv->dai_link[0].codecs[0].name) { ret = -ENOMEM; goto asrc_fail; } @@ -844,13 +844,19 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->card.num_links = 1; if (asrc_pdev) { + int i; + struct snd_soc_dai_link_component *codec; + struct snd_soc_dai_link *link; + /* DPCM DAI Links only if ASRC exists */ priv->dai_link[1].cpus->of_node = asrc_np; priv->dai_link[1].platforms->of_node = asrc_np; - priv->dai_link[2].codecs->dai_name = codec_dai_name; - priv->dai_link[2].codecs->of_node = codec_np; - priv->dai_link[2].codecs->name = - priv->dai_link[0].codecs->name; + link = &(priv->dai_link[2]); + for_each_link_codecs(link, i, codec) { + codec->dai_name = priv->dai_link[0].codecs[i].dai_name; + codec->of_node = priv->dai_link[0].codecs[i].of_node; + codec->name = priv->dai_link[0].codecs[i].name; + } priv->dai_link[2].cpus->of_node = cpu_np; priv->dai_link[2].dai_fmt = priv->dai_fmt; priv->card.num_links = 3; -- 2.34.1

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[PATCH v2 0/3] ASoC: grace time for DPCM cleanup
by Kuninori Morimoto 15 May '24

15 May '24

Hi Mark, Pierre-Louis, Jerome Cc each ASoC driver maintainer As we discussed in [1], we don't need to use dpcm_playback/capture flag, so we remove it. But we have been using it for 10 years, some driver might get damage. The most likely case is that the device/driver can use both playback/capture, but have only one flag, and not using xxx_only flag. [1/3] patch indicates warning in such case. And because of its history, DPCM has been checking CPU side only. But it should check Codec side too same as non-DPCM. Some device/driver has been bypassed this check. It should be error. [2/3] patch indicates warning in such case. Because dpcm_xxx flag is no longer used by [1/3] patch, snd_soc_dai_link_set_capabilities() is no longer needed. [3/3] patch remove it. These adds grace time for DPCM cleanup. I'm not sure when dpcm_xxx will be removed, and Codec check bypass will be error, but maybe v6.11 or v6.12 ? Please check each driver by that time. [1] https://lore.kernel.org/r/87edaym2cg.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87edaym2cg.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87wmo6dyxg.wl-kuninori.morimoto.gx@renesas.com v1 -> v2 - tidyup Codec check warning output condition Kuninori Morimoto (3): ASoC: soc-pcm: Indicate warning if dpcm_playback/capture were used for availability limition ASoC: soc-pcm: Indicate warning if CPU / Codec availability mismatch ASoC: remove snd_soc_dai_link_set_capabilities() include/sound/soc-dai.h | 1 - include/sound/soc.h | 1 + sound/soc/fsl/imx-card.c | 3 - sound/soc/generic/audio-graph-card.c | 2 - sound/soc/generic/audio-graph-card2.c | 2 - sound/soc/generic/simple-card.c | 2 - sound/soc/meson/axg-card.c | 1 - sound/soc/meson/gx-card.c | 1 - sound/soc/qcom/common.c | 1 - sound/soc/soc-dai.c | 38 --------- sound/soc/soc-pcm.c | 110 +++++++++++++++++++------- 11 files changed, 81 insertions(+), 81 deletions(-) -- 2.25.1

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[PATCH] ASoC: rt711-sdca: add GE selected mode control
by shumingf＠realtek.com 15 May '24

15 May '24

From: Shuming Fan <shumingf(a)realtek.com> The SDCA spec defines a 'selected_mode' control which can override the 'detected_mode' reported by hardware. This is useful for platform integration as well as in cases where the hardware is not able to accurately detect the jack type. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt711-sdca.c | 79 +++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt711-sdca.h | 1 + 2 files changed, 80 insertions(+) diff --git a/sound/soc/codecs/rt711-sdca.c b/sound/soc/codecs/rt711-sdca.c index 1e8dbfc3ecd9..6ff07170d770 100644 --- a/sound/soc/codecs/rt711-sdca.c +++ b/sound/soc/codecs/rt711-sdca.c @@ -81,6 +81,24 @@ static void rt711_sdca_reset(struct rt711_sdca_priv *rt711) RT711_HDA_LEGACY_RESET_CTL, 0x1, 0x1); } +static void rt711_sdca_ge_force_jack_type(struct rt711_sdca_priv *rt711, unsigned int det_mode) +{ + switch (det_mode) { + case 0x00: + rt711_sdca_index_update_bits(rt711, RT711_VENDOR_REG, RT711_COMBO_JACK_AUTO_CTL1, 0x8000, 0x0000); + rt711_sdca_index_update_bits(rt711, RT711_VENDOR_HDA_CTL, RT711_PUSH_BTN_INT_CTL0, 0x10, 0x00); + break; + case 0x03: + rt711_sdca_index_update_bits(rt711, RT711_VENDOR_REG, RT711_COMBO_JACK_AUTO_CTL1, 0x8000, 0x8000); + rt711_sdca_index_update_bits(rt711, RT711_VENDOR_HDA_CTL, RT711_PUSH_BTN_INT_CTL0, 0x17, 0x13); + break; + case 0x05: + rt711_sdca_index_update_bits(rt711, RT711_VENDOR_REG, RT711_COMBO_JACK_AUTO_CTL1, 0x8000, 0x8000); + rt711_sdca_index_update_bits(rt711, RT711_VENDOR_HDA_CTL, RT711_PUSH_BTN_INT_CTL0, 0x17, 0x15); + break; + } +} + static int rt711_sdca_calibration(struct rt711_sdca_priv *rt711) { unsigned int val, loop_rc = 0, loop_dc = 0; @@ -267,6 +285,15 @@ static int rt711_sdca_headset_detect(struct rt711_sdca_priv *rt711) break; } + if (rt711->ge_mode_override) { + if ((det_mode != rt711->ge_mode_override) && det_mode) { + det_mode = rt711->ge_mode_override; + rt711->jack_type = + (rt711->jack_type == SND_JACK_HEADPHONE) ? SND_JACK_HEADSET : SND_JACK_HEADPHONE; + } + rt711_sdca_ge_force_jack_type(rt711, det_mode); + } + /* write selected_mode */ if (det_mode) { ret = regmap_write(rt711->regmap, @@ -790,6 +817,56 @@ static int rt711_sdca_fu0f_capture_put(struct snd_kcontrol *kcontrol, return changed; } +static int rt711_sdca_ge_select_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct rt711_sdca_priv *rt711 = snd_soc_component_get_drvdata(component); + unsigned int val, item; + + val = (rt711->ge_mode_override >> e->shift_l) & e->mask; + item = snd_soc_enum_val_to_item(e, val); + ucontrol->value.enumerated.item[0] = item; + return 0; +} + +static int rt711_sdca_ge_select_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + unsigned int *item = ucontrol->value.enumerated.item; + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct rt711_sdca_priv *rt711 = snd_soc_component_get_drvdata(component); + unsigned int val, change = 0; + + if (item[0] >= e->items) + return -EINVAL; + + val = snd_soc_enum_item_to_val(e, item[0]) << e->shift_l; + if (rt711->ge_mode_override != val) { + rt711->ge_mode_override = val; + change = 1; + } + + return change; +} + +static const char * const rt711_sdca_ge_select[] = { + "None", + "Headphone", + "Headset", +}; + +static int rt711_sdca_ge_select_values[] = { + 0, + 3, + 5, +}; + +static SOC_VALUE_ENUM_SINGLE_DECL(rt711_sdca_ge_mode_enum, SND_SOC_NOPM, + 0, 0x7, rt711_sdca_ge_select, rt711_sdca_ge_select_values); + static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -6525, 75, 0); static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -1725, 75, 0); static const DECLARE_TLV_DB_SCALE(mic_vol_tlv, 0, 1000, 0); @@ -824,6 +901,8 @@ static const struct snd_kcontrol_new rt711_sdca_snd_controls[] = { SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT711_SDCA_ENT_PLATFORM_FU15, RT711_SDCA_CTL_FU_CH_GAIN, CH_R), 8, 3, 0, rt711_sdca_set_gain_get, rt711_sdca_set_gain_put, mic_vol_tlv), + SOC_ENUM_EXT("GE49 Selected Mode", rt711_sdca_ge_mode_enum, + rt711_sdca_ge_select_get, rt711_sdca_ge_select_put), }; static int rt711_sdca_mux_get(struct snd_kcontrol *kcontrol, diff --git a/sound/soc/codecs/rt711-sdca.h b/sound/soc/codecs/rt711-sdca.h index 11d421e8ab2b..15263dcb0314 100644 --- a/sound/soc/codecs/rt711-sdca.h +++ b/sound/soc/codecs/rt711-sdca.h @@ -33,6 +33,7 @@ struct rt711_sdca_priv { int hw_ver; bool fu0f_dapm_mute, fu0f_mixer_l_mute, fu0f_mixer_r_mute; bool fu1e_dapm_mute, fu1e_mixer_l_mute, fu1e_mixer_r_mute; + unsigned int ge_mode_override; }; /* NID */ -- 2.34.1

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HCtl event callback gives "Fatal Python error: drop_gil: drop_gil: GIL is not locked"
by GitHub issues - edited 14 May '24

14 May '24

alsa-project/alsa-python issue #11 was edited from HEnquist: I'm running the [hctltest2.py](https://github.com/alsa-project/alsa-python/blob/master/test/… test. Then change one of the controls, I tried the "Z Test Volume" that gets numid 4: `amixer -D hw:0 cset numid=4 5` This generates and event, which runs the callback function. That results in the test script crashing, with: ``` Fatal Python error: drop_gil: drop_gil: GIL is not locked Python runtime state: initialized Thread 0x00007cae1adbc740 (most recent call first): File "/home/henrik/repos/camilladsp-controller/hctltest2.py", line 25 in event_callback Current thread 0x00007cae1adbc740 (most recent call first): File "/home/henrik/repos/camilladsp-controller/hctltest2.py", line 63 in <module> Extension modules: pyalsa.alsahcontrol (total: 1) Aborted (core dumped) ``` Tested on manjaro with python 3.11, and latest alsa-python. Also tested on Python 3.8.19, using the latest alsa-python code directly from this repo, with the same result. Issue URL : https://github.com/alsa-project/alsa-python/issues/11 Repository URL: https://github.com/alsa-project/alsa-python

1 0

HCtl event callback gives "Fatal Python error: drop_gil: drop_gil: GIL is not locked"
by GitHub issues - opened 14 May '24

14 May '24

alsa-project/alsa-python issue #11 was opened from HEnquist: I'm running the [hctltest2.py](https://github.com/alsa-project/alsa-python/blob/master/test/… test. Then change one of the controls, I tried the "Z Test Volume" that gets numid 4: `amixer -D hw:0 cset numid=4 5` This generates and event, which runs the callback function. That results in the test script crashing, with: ``` Fatal Python error: drop_gil: drop_gil: GIL is not locked Python runtime state: initialized Thread 0x00007cae1adbc740 (most recent call first): File "/home/henrik/repos/camilladsp-controller/hctltest2.py", line 25 in event_callback Current thread 0x00007cae1adbc740 (most recent call first): File "/home/henrik/repos/camilladsp-controller/hctltest2.py", line 63 in <module> Extension modules: pyalsa.alsahcontrol (total: 1) Aborted (core dumped) ``` Tested on manjaro with python 3.11, and latest alsa-python. Issue URL : https://github.com/alsa-project/alsa-python/issues/11 Repository URL: https://github.com/alsa-project/alsa-python

1 0

[PATCH] ASoC: rt5645: mic-in detection threshold modification
by Jack Yu 14 May '24

14 May '24

Modify mic-in detection threshold for better performance. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- sound/soc/codecs/rt5645.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 05f574bf8b8f..cdb7ff7020e9 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -81,6 +81,7 @@ static const struct reg_sequence init_list[] = { static const struct reg_sequence rt5650_init_list[] = { {0xf6, 0x0100}, {RT5645_PWR_ANLG1, 0x02}, + {RT5645_IL_CMD3, 0x0018}, }; static const struct reg_default rt5645_reg[] = { -- 2.34.1

2 1

[DNM] common: pcm/split: add support up to 32 / 8 channels
by GitHub pull_request - opened 14 May '24

14 May '24

alsa-project/alsa-ucm-conf pull request #417 was opened from perexg: **DO NOT MERGE - after 1.2.12!** MOTU 828 has up to 32 hardware channels. Add support for this configuration to common/pcm/split.conf. Also, add support for up to 8 channels for the abstact devices (like 7.1 surround). Note that this change requires recent alsa-lib (Syntax 7). Link: https://github.com/alsa-project/alsa-ucm-conf/pull/416 Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/417 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/417.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH 0/3] ASoC: grace time for DPCM cleanup
by Kuninori Morimoto 14 May '24

14 May '24

Hi Mark, Pierre-Louis Cc each ASoC driver maintainer As we discussed in [1], we don't need to use dpcm_playback/capture flag, so we remove it. But we have been using it for 10 years, some driver might get damage. The most likely case is that the device/driver can use both playback/capture, but have only one flag, and not using xxx_only flag. [1/3] patch indicates warning in such case. And because of its history, DPCM has been checking CPU side only. But it should check Codec side too same as non-DPCM. Some device/driver has been bypassed this check. It should be error. [2/3] patch indicates warning in such case. Because dpcm_xxx flag is no longer used by [1/3] patch, snd_soc_dai_link_set_capabilities() is no longer needed. [3/3] patch remove it. These adds grace time for DPCM cleanup. I'm not sure when dpcm_xxx will be removed, and Codec check bypass will be error, but maybe v6.11 or v6.12 ? Please check each driver by that time. [1] https://lore.kernel.org/r/87edaym2cg.wl-kuninori.morimoto.gx@renesas.com Kuninori Morimoto (3): ASoC: soc-pcm: Indicate warning if dpcm_playback/capture were used for availability limition ASoC: soc-pcm: Indicate warning if CPU / Codec availability mismatch ASoC: remove snd_soc_dai_link_set_capabilities() include/sound/soc-dai.h | 1 - include/sound/soc.h | 1 + sound/soc/fsl/imx-card.c | 3 - sound/soc/generic/audio-graph-card.c | 2 - sound/soc/generic/audio-graph-card2.c | 2 - sound/soc/generic/simple-card.c | 2 - sound/soc/meson/axg-card.c | 1 - sound/soc/meson/gx-card.c | 1 - sound/soc/qcom/common.c | 1 - sound/soc/soc-dai.c | 38 ----------- sound/soc/soc-pcm.c | 96 ++++++++++++++++++--------- 11 files changed, 67 insertions(+), 81 deletions(-) -- 2.25.1

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[PATCH v5 1/3] ASoC: tas2781: Fix wrong loading calibrated data sequence
by Shenghao Ding 14 May '24

14 May '24

Remove declaration of unused API which load calibrated data in wrong sequence, changed the copyright year and correct file name in license header. Fixes: ef3bcde75d06 ("ASoc: tas2781: Add tas2781 driver") Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- include/sound/tas2781-dsp.h | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) diff --git a/include/sound/tas2781-dsp.h b/include/sound/tas2781-dsp.h index ea9af2726a53..7fba7ea26a4b 100644 --- a/include/sound/tas2781-dsp.h +++ b/include/sound/tas2781-dsp.h @@ -2,7 +2,7 @@ // // ALSA SoC Texas Instruments TAS2781 Audio Smart Amplifier // -// Copyright (C) 2022 - 2023 Texas Instruments Incorporated +// Copyright (C) 2022 - 2024 Texas Instruments Incorporated // https://www.ti.com // // The TAS2781 driver implements a flexible and configurable @@ -13,8 +13,8 @@ // Author: Kevin Lu <kevin-lu(a)ti.com> // -#ifndef __TASDEVICE_DSP_H__ -#define __TASDEVICE_DSP_H__ +#ifndef __TAS2781_DSP_H__ +#define __TAS2781_DSP_H__ #define MAIN_ALL_DEVICES 0x0d #define MAIN_DEVICE_A 0x01 @@ -180,7 +180,6 @@ void tasdevice_calbin_remove(void *context); int tasdevice_select_tuningprm_cfg(void *context, int prm, int cfg_no, int rca_conf_no); int tasdevice_prmg_load(void *context, int prm_no); -int tasdevice_prmg_calibdata_load(void *context, int prm_no); void tasdevice_tuning_switch(void *context, int state); int tas2781_load_calibration(void *context, char *file_name, unsigned short i); -- 2.34.1

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