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[PATCH v4] ASoC: dt-bindings: mediatek: mt8192: re-add audio afe document
by Tzung-Bi Shih 15 Sep '21

15 Sep '21

The document was merged as commit 1afc60e00de3 ("dt-bindings: mediatek: mt8192: add audio afe document"). However, [1] revealed that the commit 1afc60e00de3 breaks dt_binding_check due to dt-bindings/clock/mt8192-clk.h doesn't exist. As a temporary fix, commit 7d94ca3c8acd ("ASoC: mt8192: revert add audio afe document") reverted commit 1afc60e00de3. dt-bindings/clock/mt8192-clk.h is in mainline per commit f35f1a23e0e1 ("clk: mediatek: Add dt-bindings of MT8192 clocks"). Re-adds the document back. [1]: https://mailman.alsa-project.org/pipermail/alsa-devel/2020-November/176873.… Reviewed-by: Rob Herring <robh(a)kernel.org> Signed-off-by: Tzung-Bi Shih <tzungbi(a)google.com> --- Changes from v3 (https://mailman.alsa-project.org/pipermail/alsa-devel/2021-September/189705…): - Fix the indent errors. Changes from v2 (https://mailman.alsa-project.org/pipermail/alsa-devel/2021-August/189058.ht…): - Simplify the commit message. Changes from v1 (https://mailman.alsa-project.org/pipermail/alsa-devel/2021-August/189048.ht…): - Add more context to the commit message. .../bindings/sound/mt8192-afe-pcm.yaml | 100 ++++++++++++++++++ 1 file changed, 100 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/mt8192-afe-pcm.yaml diff --git a/Documentation/devicetree/bindings/sound/mt8192-afe-pcm.yaml b/Documentation/devicetree/bindings/sound/mt8192-afe-pcm.yaml new file mode 100644 index 000000000000..5e9fe067f440 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mt8192-afe-pcm.yaml @@ -0,0 +1,100 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/mt8192-afe-pcm.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Mediatek AFE PCM controller for mt8192 + +maintainers: + - Jiaxin Yu <jiaxin.yu(a)mediatek.com> + - Shane Chien <shane.chien(a)mediatek.com> + +properties: + compatible: + const: mediatek,mt8192-audio + + interrupts: + maxItems: 1 + + resets: + maxItems: 1 + + reset-names: + const: audiosys + + mediatek,apmixedsys: + $ref: "/schemas/types.yaml#/definitions/phandle" + description: The phandle of the mediatek apmixedsys controller + + mediatek,infracfg: + $ref: "/schemas/types.yaml#/definitions/phandle" + description: The phandle of the mediatek infracfg controller + + mediatek,topckgen: + $ref: "/schemas/types.yaml#/definitions/phandle" + description: The phandle of the mediatek topckgen controller + + power-domains: + maxItems: 1 + + clocks: + items: + - description: AFE clock + - description: ADDA DAC clock + - description: ADDA DAC pre-distortion clock + - description: audio infra sys clock + - description: audio infra 26M clock + + clock-names: + items: + - const: aud_afe_clk + - const: aud_dac_clk + - const: aud_dac_predis_clk + - const: aud_infra_clk + - const: aud_infra_26m_clk + +required: + - compatible + - interrupts + - resets + - reset-names + - mediatek,apmixedsys + - mediatek,infracfg + - mediatek,topckgen + - power-domains + - clocks + - clock-names + +additionalProperties: false + +examples: + - | + #include <dt-bindings/clock/mt8192-clk.h> + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/interrupt-controller/irq.h> + #include <dt-bindings/power/mt8192-power.h> + #include <dt-bindings/reset-controller/mt8192-resets.h> + + afe: mt8192-afe-pcm { + compatible = "mediatek,mt8192-audio"; + interrupts = <GIC_SPI 202 IRQ_TYPE_LEVEL_HIGH>; + resets = <&watchdog MT8192_TOPRGU_AUDIO_SW_RST>; + reset-names = "audiosys"; + mediatek,apmixedsys = <&apmixedsys>; + mediatek,infracfg = <&infracfg>; + mediatek,topckgen = <&topckgen>; + power-domains = <&scpsys MT8192_POWER_DOMAIN_AUDIO>; + clocks = <&audsys CLK_AUD_AFE>, + <&audsys CLK_AUD_DAC>, + <&audsys CLK_AUD_DAC_PREDIS>, + <&infracfg CLK_INFRA_AUDIO>, + <&infracfg CLK_INFRA_AUDIO_26M_B>; + clock-names = "aud_afe_clk", + "aud_dac_clk", + "aud_dac_predis_clk", + "aud_infra_clk", + "aud_infra_26m_clk"; + }; + +... -- 2.33.0.309.g3052b89438-goog

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[PATCH v2] ASoC: Intel: boards: add max98390 2/4 speakers support
by Mac Chiang 14 Sep '21

14 Sep '21

These support 3 hw boards. 1. SSP2 connects max98390, 2 speakers. 2. SSP1 connects max98390, 2 or 4 speakers. Left/Right speakers and 4 speakers tdm playback add echo reference capture add BT_OFFLOAD support add DMI_OEM_STRING for board varients Signed-off-by: Mark Hsieh <mark_hsieh(a)wistron.corp-partner.google.com> Signed-off-by: Mac Chiang <mac.chiang(a)intel.com> Signed-off-by: Kieth Tzeng <keith.tzeng(a)quantatw.com> --- Changes logs: -- add clearer hw boards info -- move Kconfig MAX98390 up, correct SSP2 wrong quirk -- modify snd_soc_dai_set_tdm_slot() supporting 2/4 dai codecs -- aligning topology name: sof-adl-rt5682-ssp0-max98390-ssp2.tplg sound/soc/intel/boards/Kconfig | 1 + sound/soc/intel/boards/sof_maxim_common.c | 170 ++++++++++++++++++ sound/soc/intel/boards/sof_maxim_common.h | 16 ++ sound/soc/intel/boards/sof_rt5682.c | 69 +++++++ .../intel/common/soc-acpi-intel-adl-match.c | 13 ++ sound/soc/sof/sof-pci-dev.c | 9 + 6 files changed, 278 insertions(+) diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 61b71d6c44cf..9723d04cbe74 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -466,6 +466,7 @@ config SND_SOC_INTEL_SOF_RT5682_MACH (MFD_INTEL_LPSS || COMPILE_TEST)) ||\ (SND_SOC_SOF_BAYTRAIL && (X86_INTEL_LPSS || COMPILE_TEST)) select SND_SOC_MAX98373_I2C + select SND_SOC_MAX98390 select SND_SOC_RT1011 select SND_SOC_RT1015 select SND_SOC_RT1015P diff --git a/sound/soc/intel/boards/sof_maxim_common.c b/sound/soc/intel/boards/sof_maxim_common.c index e66dfe666915..eeaa96648c5a 100644 --- a/sound/soc/intel/boards/sof_maxim_common.c +++ b/sound/soc/intel/boards/sof_maxim_common.c @@ -133,6 +133,176 @@ void max_98373_set_codec_conf(struct snd_soc_card *card) } EXPORT_SYMBOL_NS(max_98373_set_codec_conf, SND_SOC_INTEL_SOF_MAXIM_COMMON); +/* + * Maxim MAX98390 + */ +static const struct snd_kcontrol_new max_98390_tt_kcontrols[] = { + SOC_DAPM_PIN_SWITCH("TL Spk"), + SOC_DAPM_PIN_SWITCH("TR Spk"), +}; + +static const struct snd_soc_dapm_widget max_98390_tt_dapm_widgets[] = { + SND_SOC_DAPM_SPK("TL Spk", NULL), + SND_SOC_DAPM_SPK("TR Spk", NULL), +}; + +const struct snd_soc_dapm_route max_98390_tt_dapm_routes[] = { + /* Tweeter speaker */ + { "TL Spk", NULL, "Tweeter Left BE_OUT" }, + { "TR Spk", NULL, "Tweeter Right BE_OUT" }, +}; +EXPORT_SYMBOL_NS(max_98390_tt_dapm_routes, SND_SOC_INTEL_SOF_MAXIM_COMMON); + +static struct snd_soc_codec_conf max_98390_codec_conf[] = { + { + .dlc = COMP_CODEC_CONF(MAX_98390_DEV0_NAME), + .name_prefix = "Right", + }, + { + .dlc = COMP_CODEC_CONF(MAX_98390_DEV1_NAME), + .name_prefix = "Left", + }, +}; + +static struct snd_soc_codec_conf max_98390_4spk_codec_conf[] = { + { + .dlc = COMP_CODEC_CONF(MAX_98390_DEV0_NAME), + .name_prefix = "Right", + }, + { + .dlc = COMP_CODEC_CONF(MAX_98390_DEV1_NAME), + .name_prefix = "Left", + }, + { + .dlc = COMP_CODEC_CONF(MAX_98390_DEV2_NAME), + .name_prefix = "Tweeter Right", + }, + { + .dlc = COMP_CODEC_CONF(MAX_98390_DEV3_NAME), + .name_prefix = "Tweeter Left", + }, +}; + +struct snd_soc_dai_link_component max_98390_components[] = { + { + .name = MAX_98390_DEV0_NAME, + .dai_name = MAX_98390_CODEC_DAI, + }, + { + .name = MAX_98390_DEV1_NAME, + .dai_name = MAX_98390_CODEC_DAI, + }, +}; +EXPORT_SYMBOL_NS(max_98390_components, SND_SOC_INTEL_SOF_MAXIM_COMMON); + +struct snd_soc_dai_link_component max_98390_4spk_components[] = { + { + .name = MAX_98390_DEV0_NAME, + .dai_name = MAX_98390_CODEC_DAI, + }, + { + .name = MAX_98390_DEV1_NAME, + .dai_name = MAX_98390_CODEC_DAI, + }, + { + .name = MAX_98390_DEV2_NAME, + .dai_name = MAX_98390_CODEC_DAI, + }, + { + .name = MAX_98390_DEV3_NAME, + .dai_name = MAX_98390_CODEC_DAI, + }, +}; +EXPORT_SYMBOL_NS(max_98390_4spk_components, SND_SOC_INTEL_SOF_MAXIM_COMMON); + +static int max_98390_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + struct snd_soc_dai *codec_dai; + int i; + + for_each_rtd_codec_dais(rtd, i, codec_dai) { + + if (i >= ARRAY_SIZE(max_98390_4spk_components)) { + dev_err(codec_dai->dev, "invalid codec index %d\n", i); + return -ENODEV; + } + + if (!strcmp(codec_dai->component->name, MAX_98390_DEV0_NAME)) { + /* DEV0 tdm slot configuration Right */ + snd_soc_dai_set_tdm_slot(codec_dai, 0x01, 3, 4, 32); + } + if (!strcmp(codec_dai->component->name, MAX_98390_DEV1_NAME)) { + /* DEV1 tdm slot configuration Left */ + snd_soc_dai_set_tdm_slot(codec_dai, 0x02, 3, 4, 32); + } + + if (!strcmp(codec_dai->component->name, MAX_98390_DEV2_NAME)) { + /* DEVi2 tdm slot configuration Tweeter Right */ + snd_soc_dai_set_tdm_slot(codec_dai, 0x04, 3, 4, 32); + } + if (!strcmp(codec_dai->component->name, MAX_98390_DEV3_NAME)) { + /* DEV3 tdm slot configuration Tweeter Left */ + snd_soc_dai_set_tdm_slot(codec_dai, 0x08, 3, 4, 32); + } + } + return 0; +} + +int max_98390_spk_codec_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_card *card = rtd->card; + int ret; + + ret = snd_soc_dapm_new_controls(&card->dapm, max_98390_tt_dapm_widgets, + ARRAY_SIZE(max_98390_tt_dapm_widgets)); + if (ret) { + dev_err(rtd->dev, "unable to add dapm controls, ret %d\n", ret); + /* Don't need to add routes if widget addition failed */ + return ret; + } + + ret = snd_soc_add_card_controls(card, max_98390_tt_kcontrols, + ARRAY_SIZE(max_98390_tt_kcontrols)); + if (ret) { + dev_err(rtd->dev, "unable to add card controls, ret %d\n", ret); + return ret; + } + + ret = snd_soc_dapm_add_routes(&card->dapm, max_98373_dapm_routes, + ARRAY_SIZE(max_98373_dapm_routes)); + if (ret) { + dev_err(rtd->dev, "Speaker Left, Right map addition failed: %d\n", ret); + return ret; + } + + ret = snd_soc_dapm_add_routes(&card->dapm, max_98390_tt_dapm_routes, + ARRAY_SIZE(max_98390_tt_dapm_routes)); + if (ret) + dev_err(rtd->dev, "Tweeter Speaker Left, Right map addition failed: %d\n", ret); + + return ret; +} +EXPORT_SYMBOL_NS(max_98390_spk_codec_init, SND_SOC_INTEL_SOF_MAXIM_COMMON); + +struct snd_soc_ops max_98390_ops = { + .hw_params = max_98390_hw_params, +}; +EXPORT_SYMBOL_NS(max_98390_ops, SND_SOC_INTEL_SOF_MAXIM_COMMON); + +void max_98390_set_codec_conf(struct snd_soc_card *card, int ch) +{ + if (ch == ARRAY_SIZE(max_98390_4spk_codec_conf)) { + card->codec_conf = max_98390_4spk_codec_conf; + card->num_configs = ARRAY_SIZE(max_98390_4spk_codec_conf); + } else { + card->codec_conf = max_98390_codec_conf; + card->num_configs = ARRAY_SIZE(max_98390_codec_conf); + } +} +EXPORT_SYMBOL_NS(max_98390_set_codec_conf, SND_SOC_INTEL_SOF_MAXIM_COMMON); + /* * Maxim MAX98357A/MAX98360A */ diff --git a/sound/soc/intel/boards/sof_maxim_common.h b/sound/soc/intel/boards/sof_maxim_common.h index 3ff5e8fec4de..412b9dd06bad 100644 --- a/sound/soc/intel/boards/sof_maxim_common.h +++ b/sound/soc/intel/boards/sof_maxim_common.h @@ -24,6 +24,22 @@ int max_98373_spk_codec_init(struct snd_soc_pcm_runtime *rtd); void max_98373_set_codec_conf(struct snd_soc_card *card); int max_98373_trigger(struct snd_pcm_substream *substream, int cmd); +/* + * Maxim MAX98390 + */ +#define MAX_98390_CODEC_DAI "max98390-aif1" +#define MAX_98390_DEV0_NAME "i2c-MX98390:00" +#define MAX_98390_DEV1_NAME "i2c-MX98390:01" +#define MAX_98390_DEV2_NAME "i2c-MX98390:02" +#define MAX_98390_DEV3_NAME "i2c-MX98390:03" + +extern struct snd_soc_dai_link_component max_98390_components[2]; +extern struct snd_soc_dai_link_component max_98390_4spk_components[4]; +extern struct snd_soc_ops max_98390_ops; +extern const struct snd_soc_dapm_route max_98390_dapm_routes[]; +void max_98390_set_codec_conf(struct snd_soc_card *card, int ch); +int max_98390_spk_codec_init(struct snd_soc_pcm_runtime *rtd); + /* * Maxim MAX98357A/MAX98360A */ diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index f096bd6d69be..9acadfe03ef3 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -49,6 +49,8 @@ #define SOF_RT1015P_SPEAKER_AMP_PRESENT BIT(16) #define SOF_MAX98373_SPEAKER_AMP_PRESENT BIT(17) #define SOF_MAX98360A_SPEAKER_AMP_PRESENT BIT(18) +#define SOF_MAX98390_SPEAKER_AMP_PRESENT BIT(23) +#define SOF_MAX98390_TWEETER_SPEAKER_PRESENT BIT(24) /* BT audio offload: reserve 3 bits for future */ #define SOF_BT_OFFLOAD_SSP_SHIFT 19 @@ -176,6 +178,36 @@ static const struct dmi_system_id sof_rt5682_quirk_table[] = { SOF_RT5682_SSP_AMP(2) | SOF_RT5682_NUM_HDMIDEV(4)), }, + { + .callback = sof_rt5682_quirk_cb, + .matches = { + DMI_MATCH(DMI_PRODUCT_FAMILY, "Google_Brya"), + DMI_MATCH(DMI_OEM_STRING, "AUDIO-MAX98390_ALC5682I_I2S"), + }, + .driver_data = (void *)(SOF_RT5682_MCLK_EN | + SOF_RT5682_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_MAX98390_SPEAKER_AMP_PRESENT | + SOF_RT5682_SSP_AMP(2) | + SOF_RT5682_NUM_HDMIDEV(4)), + }, + { + .callback = sof_rt5682_quirk_cb, + .matches = { + DMI_MATCH(DMI_PRODUCT_FAMILY, "Google_Brya"), + DMI_MATCH(DMI_OEM_STRING, "AUDIO-MAX98390_ALC5682I_I2S_4SPK"), + }, + .driver_data = (void *)(SOF_RT5682_MCLK_EN | + SOF_RT5682_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_MAX98390_SPEAKER_AMP_PRESENT | + SOF_MAX98390_TWEETER_SPEAKER_PRESENT | + SOF_RT5682_SSP_AMP(1) | + SOF_RT5682_NUM_HDMIDEV(4) | + SOF_BT_OFFLOAD_SSP(2) | + SOF_SSP_BT_OFFLOAD_PRESENT), + + }, {} }; @@ -459,6 +491,7 @@ static int sof_card_late_probe(struct snd_soc_card *card) if (err < 0) return err; } + return hdac_hdmi_jack_port_init(component, &card->dapm); } @@ -745,6 +778,21 @@ static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, } else if (sof_rt5682_quirk & SOF_RT1011_SPEAKER_AMP_PRESENT) { sof_rt1011_dai_link(&links[id]); + } else if (sof_rt5682_quirk & + SOF_MAX98390_SPEAKER_AMP_PRESENT) { + if (sof_rt5682_quirk & + SOF_MAX98390_TWEETER_SPEAKER_PRESENT) { + links[id].codecs = max_98390_4spk_components; + links[id].num_codecs = ARRAY_SIZE(max_98390_4spk_components); + links[id].init = max_98390_spk_codec_init; + } else { + links[id].codecs = max_98390_components; + links[id].num_codecs = ARRAY_SIZE(max_98390_components); + links[id].init = max_98373_spk_codec_init; + } + links[id].ops = &max_98390_ops; + links[id].dpcm_capture = 1; + } else { max_98357a_dai_link(&links[id]); } @@ -881,6 +929,14 @@ static int sof_audio_probe(struct platform_device *pdev) sof_rt1011_codec_conf(&sof_audio_card_rt5682); else if (sof_rt5682_quirk & SOF_RT1015P_SPEAKER_AMP_PRESENT) sof_rt1015p_codec_conf(&sof_audio_card_rt5682); + else if (sof_rt5682_quirk & SOF_MAX98390_SPEAKER_AMP_PRESENT) { + if (sof_rt5682_quirk & SOF_MAX98390_TWEETER_SPEAKER_PRESENT) + max_98390_set_codec_conf(&sof_audio_card_rt5682, + ARRAY_SIZE(max_98390_4spk_components)); + else + max_98390_set_codec_conf(&sof_audio_card_rt5682, + ARRAY_SIZE(max_98390_components)); + } if (sof_rt5682_quirk & SOF_SSP_BT_OFFLOAD_PRESENT) sof_audio_card_rt5682.num_links++; @@ -1007,6 +1063,18 @@ static const struct platform_device_id board_ids[] = { SOF_RT5682_SSP_AMP(2) | SOF_RT5682_NUM_HDMIDEV(4)), }, + { + .name = "adl_max98390_rt5682", + .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | + SOF_RT5682_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_MAX98390_SPEAKER_AMP_PRESENT | + SOF_RT5682_SSP_AMP(1) | + SOF_RT5682_NUM_HDMIDEV(4) | + SOF_BT_OFFLOAD_SSP(2) | + SOF_SSP_BT_OFFLOAD_PRESENT), + }, + { } }; MODULE_DEVICE_TABLE(platform, board_ids); @@ -1026,6 +1094,7 @@ MODULE_DESCRIPTION("SOF Audio Machine driver"); MODULE_AUTHOR("Bard Liao <bard.liao(a)intel.com>"); MODULE_AUTHOR("Sathya Prakash M R <sathya.prakash.m.r(a)intel.com>"); MODULE_AUTHOR("Brent Lu <brent.lu(a)intel.com>"); +MODULE_AUTHOR("Mac Chiang <mac.chiang(a)intel.com>"); MODULE_LICENSE("GPL v2"); MODULE_IMPORT_NS(SND_SOC_INTEL_HDA_DSP_COMMON); MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_MAXIM_COMMON); diff --git a/sound/soc/intel/common/soc-acpi-intel-adl-match.c b/sound/soc/intel/common/soc-acpi-intel-adl-match.c index a0f6a69c7038..2db152998e4a 100644 --- a/sound/soc/intel/common/soc-acpi-intel-adl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-adl-match.c @@ -280,6 +280,11 @@ static const struct snd_soc_acpi_codecs adl_max98357a_amp = { .codecs = {"MX98357A"} }; +static const struct snd_soc_acpi_codecs adl_max98390_amp = { + .num_codecs = 1, + .codecs = {"MX98390"} +}; + struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = { { .id = "10EC5682", @@ -297,6 +302,14 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = { .sof_fw_filename = "sof-adl.ri", .sof_tplg_filename = "sof-adl-max98357a-rt5682.tplg", }, + { + .id = "10EC5682", + .drv_name = "adl_max98390_rt5682", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &adl_max98390_amp, + .sof_fw_filename = "sof-adl.ri", + .sof_tplg_filename = "sof-adl-max98390-rt5682.tplg", + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_adl_machines); diff --git a/sound/soc/sof/sof-pci-dev.c b/sound/soc/sof/sof-pci-dev.c index bc9e70765678..be90a5f79766 100644 --- a/sound/soc/sof/sof-pci-dev.c +++ b/sound/soc/sof/sof-pci-dev.c @@ -59,6 +59,15 @@ static const struct dmi_system_id sof_tplg_table[] = { }, .driver_data = "sof-adl-rt5682-ssp0-max98373-ssp2.tplg", }, + { + .callback = sof_tplg_cb, + .matches = { + DMI_MATCH(DMI_PRODUCT_FAMILY, "Google_Brya"), + DMI_MATCH(DMI_OEM_STRING, "AUDIO-MAX98390_ALC5682I_I2S"), + }, + .driver_data = "sof-adl-max98390-ssp2-rt5682-ssp0.tplg", + }, + {} }; -- 2.20.1

3 2

git clone failed
by GitHub issues - edited 14 Sep '21

14 Sep '21

alsa-project/tinycompress issue #12 was edited from BalajiMalar: i was building yocto image which uses tinycomrpess as part of NXP's recipe-multimedia. the yocto build failed since it wasnt able to clone the package. i tried to clone from CLI using the command "**git clone git://git.alsa-project.org/tinycompress.git**" and still it failed Issue URL : https://github.com/alsa-project/tinycompress/issues/12 Repository URL: https://github.com/alsa-project/tinycompress

1 0

git clone failed
by GitHub issues - opened 14 Sep '21

14 Sep '21

alsa-project/tinycompress issue #12 was opened from BalajiMalar: i was building yocto image which uses tinycomrpess as part of NXP's recipe-multimedia. the yocto build failed since it wasnt able to clone the package. i tried to clone from CLI using the command "**git clone git clone git://git.alsa-project.org/tinycompress.git**" and still it failed Issue URL : https://github.com/alsa-project/tinycompress/issues/12 Repository URL: https://github.com/alsa-project/tinycompress

1 0

[PATCH v2] ALSA: hda/realtek: Quirks to enable speaker output for Lenovo Legion 7i 15IMHG05, Yoga 7i 14ITL5/15ITL5, and 13s Gen2 laptops.
by Cameron Berkenpas 14 Sep '21

14 Sep '21

This patch initializes and enables speaker output on the Lenovo Legion 7i 15IMHG05, Yoga 7i 14ITL5/15ITL5, and 13s Gen2 series of laptops using the HDA verb sequence specific to each model. Speaker automute is suppressed for the Lenovo Legion 7i 15IMHG05 to avoid breaking speaker output on resume and when devices are unplugged from its headphone jack. Thanks to: Andreas Holzer, Vincent Morel, sycxyc, Max Christian Pohle and all others that helped. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=208555 Signed-off-by: Cameron Berkenpas <cam(a)neo-zeon.de> --- sound/pci/hda/patch_realtek.c | 129 ++++++++++++++++++++++++++++++++++ 1 file changed, 129 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8b7a389b6aed..89a6ed26b7b0 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6429,6 +6429,20 @@ static void alc_fixup_thinkpad_acpi(struct hda_codec *codec, hda_fixup_thinkpad_acpi(codec, fix, action); } +/* Fixup for Lenovo Legion 15IMHg05 speaker output on headset removal. */ +static void alc287_fixup_legion_15imhg05_speakers(struct hda_codec *codec, + const struct hda_fixup *fix, + int action) +{ + struct alc_spec *spec = codec->spec; + + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + spec->gen.suppress_auto_mute = 1; + break; + } +} + /* for alc295_fixup_hp_top_speakers */ #include "hp_x360_helper.c" @@ -6646,6 +6660,10 @@ enum { ALC623_FIXUP_LENOVO_THINKSTATION_P340, ALC255_FIXUP_ACER_HEADPHONE_AND_MIC, ALC236_FIXUP_HP_LIMIT_INT_MIC_BOOST, + ALC287_FIXUP_LEGION_15IMHG05_SPEAKERS, + ALC287_FIXUP_LEGION_15IMHG05_AUTOMUTE, + ALC287_FIXUP_YOGA7_14ITL_SPEAKERS, + ALC287_FIXUP_13S_GEN2_SPEAKERS }; static const struct hda_fixup alc269_fixups[] = { @@ -8236,6 +8254,113 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF, }, + [ALC287_FIXUP_LEGION_15IMHG05_SPEAKERS] = { + .type = HDA_FIXUP_VERBS, + //.v.verbs = legion_15imhg05_coefs, + .v.verbs = (const struct hda_verb[]) { + // set left speaker Legion 7i. + {0x20, AC_VERB_SET_COEF_INDEX, 0x24 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x41 }, + + {0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, + {0x20, AC_VERB_SET_PROC_COEF, 0xc }, + {0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x1a }, + {0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, + + {0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x2 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + {0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, + + // set right speaker Legion 7i. + {0x20, AC_VERB_SET_COEF_INDEX, 0x24 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x42 }, + + {0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, + {0x20, AC_VERB_SET_PROC_COEF, 0xc }, + {0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x2a }, + {0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, + + {0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x2 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + {0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, + {} + }, + .chained = true, + .chain_id = ALC287_FIXUP_LEGION_15IMHG05_AUTOMUTE, + }, + [ALC287_FIXUP_LEGION_15IMHG05_AUTOMUTE] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc287_fixup_legion_15imhg05_speakers, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MODE, + }, + [ALC287_FIXUP_YOGA7_14ITL_SPEAKERS] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + // set left speaker Yoga 7i. + {0x20, AC_VERB_SET_COEF_INDEX, 0x24 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x41 }, + + {0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, + {0x20, AC_VERB_SET_PROC_COEF, 0xc }, + {0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x1a }, + {0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, + + {0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x2 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + {0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, + + // set right speaker Yoga 7i. + {0x20, AC_VERB_SET_COEF_INDEX, 0x24 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x46 }, + + {0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, + {0x20, AC_VERB_SET_PROC_COEF, 0xc }, + {0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x2a }, + {0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, + + {0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x2 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + {0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, + {} + }, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MODE, + }, + [ALC287_FIXUP_13S_GEN2_SPEAKERS] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + {0x20, AC_VERB_SET_COEF_INDEX, 0x24 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x41 }, + {0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x2 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + {0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, + {0x20, AC_VERB_SET_COEF_INDEX, 0x24 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x42 }, + {0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x2 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + {0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, + {} + }, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MODE, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -8630,6 +8755,10 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3818, "Lenovo C940", ALC298_FIXUP_LENOVO_SPK_VOLUME), SND_PCI_QUIRK(0x17aa, 0x3827, "Ideapad S740", ALC285_FIXUP_IDEAPAD_S740_COEF), SND_PCI_QUIRK(0x17aa, 0x3843, "Yoga 9i", ALC287_FIXUP_IDEAPAD_BASS_SPK_AMP), + SND_PCI_QUIRK(0x17aa, 0x3813, "Legion 7i 15IMHG05", ALC287_FIXUP_LEGION_15IMHG05_SPEAKERS), + SND_PCI_QUIRK(0x17aa, 0x3852, "Lenovo Yoga 7 14ITL5", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), + SND_PCI_QUIRK(0x17aa, 0x3853, "Lenovo Yoga 7 15ITL5", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), + SND_PCI_QUIRK(0x17aa, 0x3819, "Lenovo 13s Gen2 ITL", ALC287_FIXUP_13S_GEN2_SPEAKERS), SND_PCI_QUIRK(0x17aa, 0x3902, "Lenovo E50-80", ALC269_FIXUP_DMIC_THINKPAD_ACPI), SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC), SND_PCI_QUIRK(0x17aa, 0x3978, "Lenovo B50-70", ALC269_FIXUP_DMIC_THINKPAD_ACPI), -- 2.30.2

2 1

[PATCH] soundwire: export sdw_nwrite_no_pm and sdw_nread_no_pm functions
by Simon Trimmer 14 Sep '21

14 Sep '21

The patch "soundwire: export sdw_write/read_no_pm functions" exposed the single byte no_pm versions of the IO functions that can be used without touching PM, export the multi byte no_pm versions for the same reason. Signed-off-by: Simon Trimmer <simont(a)opensource.cirrus.com> --- drivers/soundwire/bus.c | 8 ++++---- include/linux/soundwire/sdw.h | 2 ++ 2 files changed, 6 insertions(+), 4 deletions(-) diff --git a/drivers/soundwire/bus.c b/drivers/soundwire/bus.c index 1b115734a8f6..859de302dee2 100644 --- a/drivers/soundwire/bus.c +++ b/drivers/soundwire/bus.c @@ -379,8 +379,7 @@ int sdw_fill_msg(struct sdw_msg *msg, struct sdw_slave *slave, * all clients need to use the pm versions */ -static int -sdw_nread_no_pm(struct sdw_slave *slave, u32 addr, size_t count, u8 *val) +int sdw_nread_no_pm(struct sdw_slave *slave, u32 addr, size_t count, u8 *val) { struct sdw_msg msg; int ret; @@ -395,9 +394,9 @@ sdw_nread_no_pm(struct sdw_slave *slave, u32 addr, size_t count, u8 *val) ret = 0; return ret; } +EXPORT_SYMBOL(sdw_nread_no_pm); -static int -sdw_nwrite_no_pm(struct sdw_slave *slave, u32 addr, size_t count, const u8 *val) +int sdw_nwrite_no_pm(struct sdw_slave *slave, u32 addr, size_t count, const u8 *val) { struct sdw_msg msg; int ret; @@ -412,6 +411,7 @@ sdw_nwrite_no_pm(struct sdw_slave *slave, u32 addr, size_t count, const u8 *val) ret = 0; return ret; } +EXPORT_SYMBOL(sdw_nwrite_no_pm); int sdw_write_no_pm(struct sdw_slave *slave, u32 addr, u8 value) { diff --git a/include/linux/soundwire/sdw.h b/include/linux/soundwire/sdw.h index 76ce3f3ac0f2..2a5395f0dcf1 100644 --- a/include/linux/soundwire/sdw.h +++ b/include/linux/soundwire/sdw.h @@ -1042,7 +1042,9 @@ int sdw_write(struct sdw_slave *slave, u32 addr, u8 value); int sdw_write_no_pm(struct sdw_slave *slave, u32 addr, u8 value); int sdw_read_no_pm(struct sdw_slave *slave, u32 addr); int sdw_nread(struct sdw_slave *slave, u32 addr, size_t count, u8 *val); +int sdw_nread_no_pm(struct sdw_slave *slave, u32 addr, size_t count, u8 *val); int sdw_nwrite(struct sdw_slave *slave, u32 addr, size_t count, const u8 *val); +int sdw_nwrite_no_pm(struct sdw_slave *slave, u32 addr, size_t count, const u8 *val); int sdw_update(struct sdw_slave *slave, u32 addr, u8 mask, u8 val); int sdw_update_no_pm(struct sdw_slave *slave, u32 addr, u8 mask, u8 val); -- 2.33.0

4 3

[PATCH AUTOSEL 5.13 07/19] ASoC: audio-graph: respawn Platform Support
by Sasha Levin 14 Sep '21

14 Sep '21

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> [ Upstream commit 5f939f49771002f347039edf984aca42f30fc31a ] commit 63f2f9cceb09f8 ("ASoC: audio-graph: remove Platform support") removed Platform support from audio-graph, because it doesn't have "plat" support on DT (simple-card has). But, Platform support is needed if user is using snd_dmaengine_pcm_register() which adds generic DMA as Platform. And this Platform dev is using CPU dev. Without this patch, at least STM32MP15 audio sound card is no more functional (v5.13 or later). This patch respawn Platform Support on audio-graph again. Reported-by: Olivier MOYSAN <olivier.moysan(a)foss.st.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Tested-by: Olivier MOYSAN <olivier.moysan(a)foss.st.com> Link: https://lore.kernel.org/r/878s0jzrpf.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/generic/audio-graph-card.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index 5e71382467e8..546f6fd0609e 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -285,6 +285,7 @@ static int graph_dai_link_of_dpcm(struct asoc_simple_priv *priv, if (li->cpu) { struct snd_soc_card *card = simple_priv_to_card(priv); struct snd_soc_dai_link_component *cpus = asoc_link_to_cpu(dai_link, 0); + struct snd_soc_dai_link_component *platforms = asoc_link_to_platform(dai_link, 0); int is_single_links = 0; /* Codec is dummy */ @@ -313,6 +314,7 @@ static int graph_dai_link_of_dpcm(struct asoc_simple_priv *priv, dai_link->no_pcm = 1; asoc_simple_canonicalize_cpu(cpus, is_single_links); + asoc_simple_canonicalize_platform(platforms, cpus); } else { struct snd_soc_codec_conf *cconf = simple_props_to_codec_conf(dai_props, 0); struct snd_soc_dai_link_component *codecs = asoc_link_to_codec(dai_link, 0); @@ -366,6 +368,7 @@ static int graph_dai_link_of(struct asoc_simple_priv *priv, struct snd_soc_dai_link *dai_link = simple_priv_to_link(priv, li->link); struct snd_soc_dai_link_component *cpus = asoc_link_to_cpu(dai_link, 0); struct snd_soc_dai_link_component *codecs = asoc_link_to_codec(dai_link, 0); + struct snd_soc_dai_link_component *platforms = asoc_link_to_platform(dai_link, 0); char dai_name[64]; int ret, is_single_links = 0; @@ -383,6 +386,7 @@ static int graph_dai_link_of(struct asoc_simple_priv *priv, "%s-%s", cpus->dai_name, codecs->dai_name); asoc_simple_canonicalize_cpu(cpus, is_single_links); + asoc_simple_canonicalize_platform(platforms, cpus); ret = graph_link_init(priv, cpu_ep, codec_ep, li, dai_name); if (ret < 0) @@ -608,6 +612,7 @@ static int graph_count_noml(struct asoc_simple_priv *priv, li->num[li->link].cpus = 1; li->num[li->link].codecs = 1; + li->num[li->link].platforms = 1; li->link += 1; /* 1xCPU-Codec */ @@ -630,6 +635,7 @@ static int graph_count_dpcm(struct asoc_simple_priv *priv, if (li->cpu) { li->num[li->link].cpus = 1; + li->num[li->link].platforms = 1; li->link++; /* 1xCPU-dummy */ } else { -- 2.30.2

1 0

[PATCH AUTOSEL 5.14 09/25] ASoC: audio-graph: respawn Platform Support
by Sasha Levin 14 Sep '21

14 Sep '21

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> [ Upstream commit 5f939f49771002f347039edf984aca42f30fc31a ] commit 63f2f9cceb09f8 ("ASoC: audio-graph: remove Platform support") removed Platform support from audio-graph, because it doesn't have "plat" support on DT (simple-card has). But, Platform support is needed if user is using snd_dmaengine_pcm_register() which adds generic DMA as Platform. And this Platform dev is using CPU dev. Without this patch, at least STM32MP15 audio sound card is no more functional (v5.13 or later). This patch respawn Platform Support on audio-graph again. Reported-by: Olivier MOYSAN <olivier.moysan(a)foss.st.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Tested-by: Olivier MOYSAN <olivier.moysan(a)foss.st.com> Link: https://lore.kernel.org/r/878s0jzrpf.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/generic/audio-graph-card.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index 5e71382467e8..546f6fd0609e 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -285,6 +285,7 @@ static int graph_dai_link_of_dpcm(struct asoc_simple_priv *priv, if (li->cpu) { struct snd_soc_card *card = simple_priv_to_card(priv); struct snd_soc_dai_link_component *cpus = asoc_link_to_cpu(dai_link, 0); + struct snd_soc_dai_link_component *platforms = asoc_link_to_platform(dai_link, 0); int is_single_links = 0; /* Codec is dummy */ @@ -313,6 +314,7 @@ static int graph_dai_link_of_dpcm(struct asoc_simple_priv *priv, dai_link->no_pcm = 1; asoc_simple_canonicalize_cpu(cpus, is_single_links); + asoc_simple_canonicalize_platform(platforms, cpus); } else { struct snd_soc_codec_conf *cconf = simple_props_to_codec_conf(dai_props, 0); struct snd_soc_dai_link_component *codecs = asoc_link_to_codec(dai_link, 0); @@ -366,6 +368,7 @@ static int graph_dai_link_of(struct asoc_simple_priv *priv, struct snd_soc_dai_link *dai_link = simple_priv_to_link(priv, li->link); struct snd_soc_dai_link_component *cpus = asoc_link_to_cpu(dai_link, 0); struct snd_soc_dai_link_component *codecs = asoc_link_to_codec(dai_link, 0); + struct snd_soc_dai_link_component *platforms = asoc_link_to_platform(dai_link, 0); char dai_name[64]; int ret, is_single_links = 0; @@ -383,6 +386,7 @@ static int graph_dai_link_of(struct asoc_simple_priv *priv, "%s-%s", cpus->dai_name, codecs->dai_name); asoc_simple_canonicalize_cpu(cpus, is_single_links); + asoc_simple_canonicalize_platform(platforms, cpus); ret = graph_link_init(priv, cpu_ep, codec_ep, li, dai_name); if (ret < 0) @@ -608,6 +612,7 @@ static int graph_count_noml(struct asoc_simple_priv *priv, li->num[li->link].cpus = 1; li->num[li->link].codecs = 1; + li->num[li->link].platforms = 1; li->link += 1; /* 1xCPU-Codec */ @@ -630,6 +635,7 @@ static int graph_count_dpcm(struct asoc_simple_priv *priv, if (li->cpu) { li->num[li->link].cpus = 1; + li->num[li->link].platforms = 1; li->link++; /* 1xCPU-dummy */ } else { -- 2.30.2

1 0

[PATCH] ALSA: hda/realtek: Quirks to enable speaker output for Lenovo Legion 7i 15IMHG05, Yoga 7i 14ITL5/15ITL5, and 13s Gen2 laptops.
by Cameron Berkenpas 14 Sep '21

14 Sep '21

This patch initializes and enables speaker output on the Lenovo Legion 7i 15IMHG05, Yoga 7i 14ITL5/15ITL5, and 13s Gen2 series of laptops using the HDA verb sequence specific to each model. Speaker automute is disabled for the Lenovo Legion 7i 15IMHG05 to avoid breaking speaker output on resume and when devices are unplugged from its headphone jack. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=208555 Signed-off-by: Cameron Berkenpas <cam(a)neo-zeon.de> --- sound/pci/hda/patch_realtek.c | 129 ++++++++++++++++++++++++++++++++++ 1 file changed, 129 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8b7a389b6aed..7e94107e8e59 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6429,6 +6429,20 @@ static void alc_fixup_thinkpad_acpi(struct hda_codec *codec, hda_fixup_thinkpad_acpi(codec, fix, action); } +/* Fixup for Lenovo Legion 15IMHg05 speaker output on headset removal. */ +static void alc287_fixup_legion_15imhg05_speakers(struct hda_codec *codec, + const struct hda_fixup *fix, + int action) +{ + struct alc_spec *spec = codec->spec; + + switch (action) { + case HDA_FIXUP_ACT_INIT: + spec->gen.automute_speaker = 0; + break; + } +} + /* for alc295_fixup_hp_top_speakers */ #include "hp_x360_helper.c" @@ -6646,6 +6660,10 @@ enum { ALC623_FIXUP_LENOVO_THINKSTATION_P340, ALC255_FIXUP_ACER_HEADPHONE_AND_MIC, ALC236_FIXUP_HP_LIMIT_INT_MIC_BOOST, + ALC287_FIXUP_LEGION_15IMHG05_SPEAKERS, + ALC287_FIXUP_LEGION_15IMHG05_AUTOMUTE, + ALC287_FIXUP_YOGA7_14ITL_SPEAKERS, + ALC287_FIXUP_13S_GEN2_SPEAKERS }; static const struct hda_fixup alc269_fixups[] = { @@ -8236,6 +8254,113 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF, }, + [ALC287_FIXUP_LEGION_15IMHG05_SPEAKERS] = { + .type = HDA_FIXUP_VERBS, + //.v.verbs = legion_15imhg05_coefs, + .v.verbs = (const struct hda_verb[]) { + // set left speaker Legion 7i. + {0x20, AC_VERB_SET_COEF_INDEX, 0x24 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x41 }, + + {0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, + {0x20, AC_VERB_SET_PROC_COEF, 0xc }, + {0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x1a }, + {0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, + + {0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x2 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + {0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, + + // set right speaker Legion 7i. + {0x20, AC_VERB_SET_COEF_INDEX, 0x24 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x42 }, + + {0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, + {0x20, AC_VERB_SET_PROC_COEF, 0xc }, + {0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x2a }, + {0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, + + {0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x2 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + {0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, + {} + }, + .chained = true, + .chain_id = ALC287_FIXUP_LEGION_15IMHG05_AUTOMUTE, + }, + [ALC287_FIXUP_LEGION_15IMHG05_AUTOMUTE] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc287_fixup_legion_15imhg05_speakers, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MODE, + }, + [ALC287_FIXUP_YOGA7_14ITL_SPEAKERS] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + // set left speaker Yoga 7i. + {0x20, AC_VERB_SET_COEF_INDEX, 0x24 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x41 }, + + {0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, + {0x20, AC_VERB_SET_PROC_COEF, 0xc }, + {0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x1a }, + {0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, + + {0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x2 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + {0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, + + // set right speaker Yoga 7i. + {0x20, AC_VERB_SET_COEF_INDEX, 0x24 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x46 }, + + {0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, + {0x20, AC_VERB_SET_PROC_COEF, 0xc }, + {0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x2a }, + {0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, + + {0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x2 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + {0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, + {} + }, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MODE, + }, + [ALC287_FIXUP_13S_GEN2_SPEAKERS] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + {0x20, AC_VERB_SET_COEF_INDEX, 0x24 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x41 }, + {0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x2 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + {0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, + {0x20, AC_VERB_SET_COEF_INDEX, 0x24 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x42 }, + {0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x2 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + {0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + {0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, + {} + }, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MODE, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -8630,6 +8755,10 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3818, "Lenovo C940", ALC298_FIXUP_LENOVO_SPK_VOLUME), SND_PCI_QUIRK(0x17aa, 0x3827, "Ideapad S740", ALC285_FIXUP_IDEAPAD_S740_COEF), SND_PCI_QUIRK(0x17aa, 0x3843, "Yoga 9i", ALC287_FIXUP_IDEAPAD_BASS_SPK_AMP), + SND_PCI_QUIRK(0x17aa, 0x3813, "Legion 7i 15IMHG05", ALC287_FIXUP_LEGION_15IMHG05_SPEAKERS), + SND_PCI_QUIRK(0x17aa, 0x3852, "Lenovo Yoga 7 14ITL5", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), + SND_PCI_QUIRK(0x17aa, 0x3853, "Lenovo Yoga 7 15ITL5", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), + SND_PCI_QUIRK(0x17aa, 0x3819, "Lenovo 13s Gen2 ITL", ALC287_FIXUP_13S_GEN2_SPEAKERS), SND_PCI_QUIRK(0x17aa, 0x3902, "Lenovo E50-80", ALC269_FIXUP_DMIC_THINKPAD_ACPI), SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC), SND_PCI_QUIRK(0x17aa, 0x3978, "Lenovo B50-70", ALC269_FIXUP_DMIC_THINKPAD_ACPI), -- 2.30.2

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kernel threads preventing wakeup of thread using snd_pcm_readi?
by Bert Schiettecatte 13 Sep '21

13 Sep '21

Hi all I am working on a realtime audio application on the rk3288 platform. I am using stable kernel 5.13.13. I'm using the open source tracy profiler to profile my application. My development PC receives trace data via ethernet from my platform. I have custom alsa drivers on my platform for my DACs and ADCs. My application has a thread in which I use snd_pcm_readi which blocks until it can read enough samples. I have another thread in which I write the samples to disk (sandisk extreme SD card). I use a ring-buffer so the code is lock-free. I've noticed that over a timespan of a few minutes I get a few clicks, which seem to come from the thread blocking on snd_pcm_readi not waking up in time. I am observing this in the traces I collect via tracy. Right before this happens, I can see kworker/kblockd being scheduled on one of the cores many times and rapidly, or ocasionally ksoftirqd, mmc_complete, or events_highpri. From the trace I collect, i have the impression nothing is being scheduled on the other cores, so I do not see a reason why my thread should not be woken up and scheduled. Why is the thread blocking on snd_pcm_readi not waking up in time, and is it possible that the kworker threads are preventing this, and why? Any suggestions on how to debug this further are greatly appreciated. Thanks Bert

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