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

[PATCH] ASoC: Intel: boards: add stereo playback by woofer speaker
by mac.chiang＠intel.com 16 Mar '20

16 Mar '20

From: Mac Chiang <mac.chiang(a)intel.com> Separate woofer stereo speakers settings. Signed-off-by: Mac Chiang <mac.chiang(a)intel.com> --- sound/soc/intel/boards/cml_rt1011_rt5682.c | 221 +++++++++++++++++++++++------ 1 file changed, 180 insertions(+), 41 deletions(-) diff --git a/sound/soc/intel/boards/cml_rt1011_rt5682.c b/sound/soc/intel/boards/cml_rt1011_rt5682.c index dd80d01..fc1bd94 100644 --- a/sound/soc/intel/boards/cml_rt1011_rt5682.c +++ b/sound/soc/intel/boards/cml_rt1011_rt5682.c @@ -30,6 +30,36 @@ #define CML_RT5682_CODEC_DAI "rt5682-aif1" #define NAME_SIZE 32 +#define SOF_RT1011_SPEAKER_WL BIT(0) +#define SOF_RT1011_SPEAKER_WR BIT(1) +#define SOF_RT1011_SPEAKER_TL BIT(2) +#define SOF_RT1011_SPEAKER_TR BIT(3) +#define SPK_CH 4 + +/* Default: Woofer+Tweeter speakers */ +static unsigned long sof_rt1011_quirk = SOF_RT1011_SPEAKER_WL | SOF_RT1011_SPEAKER_WR | + SOF_RT1011_SPEAKER_TL | SOF_RT1011_SPEAKER_TR; + +static int sof_rt1011_quirk_cb(const struct dmi_system_id *id) +{ + sof_rt1011_quirk = (unsigned long)id->driver_data; + return 1; +} + +static const struct dmi_system_id sof_rt1011_quirk_table[] = { + { + .callback = sof_rt1011_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Google"), + DMI_MATCH(DMI_PRODUCT_NAME, "Palkia"), + }, + .driver_data = (void *)(SOF_RT1011_SPEAKER_WL | + SOF_RT1011_SPEAKER_WR), + }, + { + } +}; + static struct snd_soc_jack hdmi_jack[3]; struct hdmi_pcm { @@ -48,15 +78,16 @@ struct card_private { static const struct snd_kcontrol_new cml_controls[] = { SOC_DAPM_PIN_SWITCH("Headphone Jack"), SOC_DAPM_PIN_SWITCH("Headset Mic"), - SOC_DAPM_PIN_SWITCH("TL Ext Spk"), - SOC_DAPM_PIN_SWITCH("TR Ext Spk"), SOC_DAPM_PIN_SWITCH("WL Ext Spk"), SOC_DAPM_PIN_SWITCH("WR Ext Spk"), }; +static const struct snd_kcontrol_new cml_rt1011_tt_controls[] = { + SOC_DAPM_PIN_SWITCH("TL Ext Spk"), + SOC_DAPM_PIN_SWITCH("TR Ext Spk"), +}; + static const struct snd_soc_dapm_widget cml_rt1011_rt5682_widgets[] = { - SND_SOC_DAPM_SPK("TL Ext Spk", NULL), - SND_SOC_DAPM_SPK("TR Ext Spk", NULL), SND_SOC_DAPM_SPK("WL Ext Spk", NULL), SND_SOC_DAPM_SPK("WR Ext Spk", NULL), SND_SOC_DAPM_HP("Headphone Jack", NULL), @@ -64,6 +95,11 @@ static const struct snd_soc_dapm_widget cml_rt1011_rt5682_widgets[] = { SND_SOC_DAPM_MIC("SoC DMIC", NULL), }; +static const struct snd_soc_dapm_widget cml_rt1011_tt_widgets[] = { + SND_SOC_DAPM_SPK("TL Ext Spk", NULL), + SND_SOC_DAPM_SPK("TR Ext Spk", NULL), +}; + static const struct snd_soc_dapm_route cml_rt1011_rt5682_map[] = { /*speaker*/ {"TL Ext Spk", NULL, "TL SPO"}, @@ -82,6 +118,12 @@ static const struct snd_soc_dapm_route cml_rt1011_rt5682_map[] = { {"DMic", NULL, "SoC DMIC"}, }; +static const struct snd_soc_dapm_route cml_rt1011_tt_map[] = { + /*TL/TR speaker*/ + {"TL Ext Spk", NULL, "TL SPO" }, + {"TR Ext Spk", NULL, "TR SPO" }, +}; + static int cml_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd) { struct card_private *ctx = snd_soc_card_get_drvdata(rtd->card); @@ -121,6 +163,35 @@ static int cml_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd) return ret; }; +static int cml_rt1011_spk_init(struct snd_soc_pcm_runtime *rtd) +{ + int ret = 0; + struct snd_soc_card *card = rtd->card; + + if (sof_rt1011_quirk & (SOF_RT1011_SPEAKER_TL | + SOF_RT1011_SPEAKER_TR)) { + + ret = snd_soc_add_card_controls(card, cml_rt1011_tt_controls, + ARRAY_SIZE(cml_rt1011_tt_controls)); + if (ret) + return ret; + + ret = snd_soc_dapm_new_controls(&card->dapm, + cml_rt1011_tt_widgets, + ARRAY_SIZE(cml_rt1011_tt_widgets)); + if (ret) + return ret; + + ret = snd_soc_dapm_add_routes(&card->dapm, cml_rt1011_tt_map, + ARRAY_SIZE(cml_rt1011_tt_map)); + + if (ret) + return ret; + } + + return ret; +} + static int cml_rt5682_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -192,31 +263,52 @@ static int cml_rt1011_hw_params(struct snd_pcm_substream *substream, * The feedback is captured for each codec individually. * Hence all 4 codecs use 1 Tx slot each for feedback. */ - if (!strcmp(codec_dai->component->name, "i2c-10EC1011:00")) { - ret = snd_soc_dai_set_tdm_slot(codec_dai, - 0x4, 0x1, 4, 24); - if (ret < 0) - break; - } - if (!strcmp(codec_dai->component->name, "i2c-10EC1011:02")) { - ret = snd_soc_dai_set_tdm_slot(codec_dai, - 0x1, 0x1, 4, 24); - if (ret < 0) - break; - } - /* TDM Rx slot 2 is used for Right Woofer & Tweeters pair */ - if (!strcmp(codec_dai->component->name, "i2c-10EC1011:01")) { - ret = snd_soc_dai_set_tdm_slot(codec_dai, - 0x8, 0x2, 4, 24); - if (ret < 0) - break; - } - if (!strcmp(codec_dai->component->name, "i2c-10EC1011:03")) { - ret = snd_soc_dai_set_tdm_slot(codec_dai, - 0x2, 0x2, 4, 24); - if (ret < 0) - break; + if (sof_rt1011_quirk & (SOF_RT1011_SPEAKER_TL | + SOF_RT1011_SPEAKER_TR)) { + if (!strcmp(codec_dai->component->name, "i2c-10EC1011:00")) { + ret = snd_soc_dai_set_tdm_slot(codec_dai, + 0x4, 0x1, 4, 24); + if (ret < 0) + break; + } + + if (!strcmp(codec_dai->component->name, "i2c-10EC1011:02")) { + ret = snd_soc_dai_set_tdm_slot(codec_dai, + 0x1, 0x1, 4, 24); + if (ret < 0) + break; + } + + /* TDM Rx slot 2 is used for Right Woofer & Tweeters pair */ + if (!strcmp(codec_dai->component->name, "i2c-10EC1011:01")) { + ret = snd_soc_dai_set_tdm_slot(codec_dai, + 0x8, 0x2, 4, 24); + if (ret < 0) + break; + } + + if (!strcmp(codec_dai->component->name, "i2c-10EC1011:03")) { + ret = snd_soc_dai_set_tdm_slot(codec_dai, + 0x2, 0x2, 4, 24); + if (ret < 0) + break; + } + } else { + if (!strcmp(codec_dai->component->name, "i2c-10EC1011:00")) { + ret = snd_soc_dai_set_tdm_slot(codec_dai, + 0x4, 0x1, 4, 24); + if (ret < 0) + break; + } + + if (!strcmp(codec_dai->component->name, "i2c-10EC1011:01")) { + ret = snd_soc_dai_set_tdm_slot(codec_dai, + 0x8, 0x2, 4, 24); + if (ret < 0) + break; + } } + } if (ret < 0) dev_err(rtd->dev, @@ -302,10 +394,8 @@ SND_SOC_DAILINK_DEF(ssp1_pin, DAILINK_COMP_ARRAY(COMP_CPU("SSP1 Pin"))); SND_SOC_DAILINK_DEF(ssp1_codec, DAILINK_COMP_ARRAY( - /* WL */ COMP_CODEC("i2c-10EC1011:00", CML_RT1011_CODEC_DAI), - /* WR */ COMP_CODEC("i2c-10EC1011:01", CML_RT1011_CODEC_DAI), - /* TL */ COMP_CODEC("i2c-10EC1011:02", CML_RT1011_CODEC_DAI), - /* TR */ COMP_CODEC("i2c-10EC1011:03", CML_RT1011_CODEC_DAI))); + /* WL */ COMP_CODEC("i2c-10EC1011:00", CML_RT1011_CODEC_DAI), + /* WR */ COMP_CODEC("i2c-10EC1011:01", CML_RT1011_CODEC_DAI))); SND_SOC_DAILINK_DEF(dmic_pin, DAILINK_COMP_ARRAY(COMP_CPU("DMIC01 Pin"))); @@ -399,6 +489,7 @@ static struct snd_soc_dai_link cml_rt1011_rt5682_dailink[] = { .dpcm_playback = 1, .dpcm_capture = 1, /* Capture stream provides Feedback */ .no_pcm = 1, + .init = cml_rt1011_spk_init, .ops = &cml_rt1011_ops, SND_SOC_DAILINK_REG(ssp1_pin, ssp1_codec, platform), }, @@ -413,14 +504,6 @@ static struct snd_soc_codec_conf rt1011_conf[] = { .dlc = COMP_CODEC_CONF("i2c-10EC1011:01"), .name_prefix = "WR", }, - { - .dlc = COMP_CODEC_CONF("i2c-10EC1011:02"), - .name_prefix = "TL", - }, - { - .dlc = COMP_CODEC_CONF("i2c-10EC1011:03"), - .name_prefix = "TR", - }, }; /* Cometlake audio machine driver for RT1011 and RT5682 */ @@ -442,10 +525,12 @@ static struct snd_soc_card snd_soc_card_cml = { static int snd_cml_rt1011_probe(struct platform_device *pdev) { + struct snd_soc_dai_link_component *rt1011_dais_components; + struct snd_soc_codec_conf *rt1011_dais_confs; struct card_private *ctx; struct snd_soc_acpi_mach *mach; const char *platform_name; - int ret; + int ret, i; ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_ATOMIC); if (!ctx) @@ -456,6 +541,59 @@ static int snd_cml_rt1011_probe(struct platform_device *pdev) snd_soc_card_cml.dev = &pdev->dev; platform_name = mach->mach_params.platform; + dmi_check_system(sof_rt1011_quirk_table); + + dev_info(&pdev->dev, "sof_rt1011_quirk = %lx\n", sof_rt1011_quirk); + + if (sof_rt1011_quirk & (SOF_RT1011_SPEAKER_TL | + SOF_RT1011_SPEAKER_TR)) { + rt1011_dais_confs = devm_kzalloc(&pdev->dev, + sizeof(struct snd_soc_codec_conf) * + SPK_CH, GFP_KERNEL); + + rt1011_dais_components = devm_kzalloc(&pdev->dev, + sizeof(struct snd_soc_dai_link_component) * + SPK_CH, GFP_KERNEL); + + for (i = 0; i < SPK_CH; i++) { + rt1011_dais_confs[i].dlc.name = devm_kasprintf(&pdev->dev, + GFP_KERNEL, + "i2c-10EC1011:0%d", + i); + switch (i) { + case 0: + rt1011_dais_confs[i].name_prefix = "WL"; + break; + + case 1: + rt1011_dais_confs[i].name_prefix = "WR"; + break; + case 2: + rt1011_dais_confs[i].name_prefix = "TL"; + break; + case 3: + rt1011_dais_confs[i].name_prefix = "TR"; + break; + } + rt1011_dais_components[i].name = devm_kasprintf(&pdev->dev, + GFP_KERNEL, + "i2c-10EC1011:0%d", + i); + rt1011_dais_components[i].dai_name = CML_RT1011_CODEC_DAI; + } + + snd_soc_card_cml.codec_conf = rt1011_dais_confs; + snd_soc_card_cml.num_configs = SPK_CH; + + for (i = 0; i < ARRAY_SIZE(cml_rt1011_rt5682_dailink); i++) { + if (!strcmp(cml_rt1011_rt5682_dailink[i].codecs->dai_name, + CML_RT1011_CODEC_DAI)) { + cml_rt1011_rt5682_dailink[i].codecs = rt1011_dais_components; + cml_rt1011_rt5682_dailink[i].num_codecs = SPK_CH; + } + } + } + /* set platform name for each dailink */ ret = snd_soc_fixup_dai_links_platform_name(&snd_soc_card_cml, platform_name); @@ -483,5 +621,6 @@ MODULE_DESCRIPTION("Cometlake Audio Machine driver - RT1011 and RT5682 in I2S mo MODULE_AUTHOR("Naveen Manohar <naveen.m(a)intel.com>"); MODULE_AUTHOR("Sathya Prakash M R <sathya.prakash.m.r(a)intel.com>"); MODULE_AUTHOR("Shuming Fan <shumingf(a)realtek.com>"); +MODULE_AUTHOR("Mac Chiang <mac.chiang(a)intel.com>"); MODULE_LICENSE("GPL v2"); MODULE_ALIAS("platform:cml_rt1011_rt5682"); -- 2.7.4

3 2

[PATCH AUTOSEL 5.4 25/35] ASoC: stm32: sai: manage rebind issue
by Sasha Levin 16 Mar '20

16 Mar '20

From: Olivier Moysan <olivier.moysan(a)st.com> [ Upstream commit 0d6defc7e0e437a9fd53622f7fd85740f38d5693 ] The commit e894efef9ac7 ("ASoC: core: add support to card rebind") allows to rebind the sound card after a rebind of one of its component. With this commit, the sound card is actually rebound, but may be no more functional. The following problems have been seen with STM32 SAI driver. 1) DMA channel is not requested: With the sound card rebind the simplified call sequence is: stm32_sai_sub_probe snd_soc_register_component snd_soc_try_rebind_card snd_soc_instantiate_card devm_snd_dmaengine_pcm_register The problem occurs because the pcm must be registered, before snd_soc_instantiate_card() is called. Modify SAI driver, to change the call sequence as follows: stm32_sai_sub_probe devm_snd_dmaengine_pcm_register snd_soc_register_component snd_soc_try_rebind_card 2) DMA channel is not released: dma_release_channel() is not called when devm_dmaengine_pcm_release() is executed. This occurs because SND_DMAENGINE_PCM_DRV_NAME component, has already been released through devm_component_release(). devm_dmaengine_pcm_release() should be called before devm_component_release() to avoid this problem. Call snd_dmaengine_pcm_unregister() and snd_soc_unregister_component() explicitly from SAI driver, to have the right sequence. Signed-off-by: Olivier Moysan <olivier.moysan(a)st.com> Message-Id: <20200304102406.8093-1-olivier.moysan(a)st.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/stm/stm32_sai_sub.c | 18 ++++++++++-------- 1 file changed, 10 insertions(+), 8 deletions(-) diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index 30bcd5d3a32a8..10eb4b8e8e7ee 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -1543,20 +1543,20 @@ static int stm32_sai_sub_probe(struct platform_device *pdev) return ret; } - ret = devm_snd_soc_register_component(&pdev->dev, &stm32_component, - &sai->cpu_dai_drv, 1); + ret = snd_dmaengine_pcm_register(&pdev->dev, conf, 0); + if (ret) { + dev_err(&pdev->dev, "Could not register pcm dma\n"); + return ret; + } + + ret = snd_soc_register_component(&pdev->dev, &stm32_component, + &sai->cpu_dai_drv, 1); if (ret) return ret; if (STM_SAI_PROTOCOL_IS_SPDIF(sai)) conf = &stm32_sai_pcm_config_spdif; - ret = devm_snd_dmaengine_pcm_register(&pdev->dev, conf, 0); - if (ret) { - dev_err(&pdev->dev, "Could not register pcm dma\n"); - return ret; - } - return 0; } @@ -1565,6 +1565,8 @@ static int stm32_sai_sub_remove(struct platform_device *pdev) struct stm32_sai_sub_data *sai = dev_get_drvdata(&pdev->dev); clk_unprepare(sai->pdata->pclk); + snd_dmaengine_pcm_unregister(&pdev->dev); + snd_soc_unregister_component(&pdev->dev); return 0; } -- 2.20.1

1 0

[PATCH AUTOSEL 5.5 29/41] ASoC: stm32: sai: manage rebind issue
by Sasha Levin 16 Mar '20

16 Mar '20

From: Olivier Moysan <olivier.moysan(a)st.com> [ Upstream commit 0d6defc7e0e437a9fd53622f7fd85740f38d5693 ] The commit e894efef9ac7 ("ASoC: core: add support to card rebind") allows to rebind the sound card after a rebind of one of its component. With this commit, the sound card is actually rebound, but may be no more functional. The following problems have been seen with STM32 SAI driver. 1) DMA channel is not requested: With the sound card rebind the simplified call sequence is: stm32_sai_sub_probe snd_soc_register_component snd_soc_try_rebind_card snd_soc_instantiate_card devm_snd_dmaengine_pcm_register The problem occurs because the pcm must be registered, before snd_soc_instantiate_card() is called. Modify SAI driver, to change the call sequence as follows: stm32_sai_sub_probe devm_snd_dmaengine_pcm_register snd_soc_register_component snd_soc_try_rebind_card 2) DMA channel is not released: dma_release_channel() is not called when devm_dmaengine_pcm_release() is executed. This occurs because SND_DMAENGINE_PCM_DRV_NAME component, has already been released through devm_component_release(). devm_dmaengine_pcm_release() should be called before devm_component_release() to avoid this problem. Call snd_dmaengine_pcm_unregister() and snd_soc_unregister_component() explicitly from SAI driver, to have the right sequence. Signed-off-by: Olivier Moysan <olivier.moysan(a)st.com> Message-Id: <20200304102406.8093-1-olivier.moysan(a)st.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/stm/stm32_sai_sub.c | 18 ++++++++++-------- 1 file changed, 10 insertions(+), 8 deletions(-) diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index 30bcd5d3a32a8..10eb4b8e8e7ee 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -1543,20 +1543,20 @@ static int stm32_sai_sub_probe(struct platform_device *pdev) return ret; } - ret = devm_snd_soc_register_component(&pdev->dev, &stm32_component, - &sai->cpu_dai_drv, 1); + ret = snd_dmaengine_pcm_register(&pdev->dev, conf, 0); + if (ret) { + dev_err(&pdev->dev, "Could not register pcm dma\n"); + return ret; + } + + ret = snd_soc_register_component(&pdev->dev, &stm32_component, + &sai->cpu_dai_drv, 1); if (ret) return ret; if (STM_SAI_PROTOCOL_IS_SPDIF(sai)) conf = &stm32_sai_pcm_config_spdif; - ret = devm_snd_dmaengine_pcm_register(&pdev->dev, conf, 0); - if (ret) { - dev_err(&pdev->dev, "Could not register pcm dma\n"); - return ret; - } - return 0; } @@ -1565,6 +1565,8 @@ static int stm32_sai_sub_remove(struct platform_device *pdev) struct stm32_sai_sub_data *sai = dev_get_drvdata(&pdev->dev); clk_unprepare(sai->pdata->pclk); + snd_dmaengine_pcm_unregister(&pdev->dev); + snd_soc_unregister_component(&pdev->dev); return 0; } -- 2.20.1

1 0

[PATCH AUTOSEL 5.5 09/41] ASoC: meson: g12a: add tohdmitx reset
by Sasha Levin 16 Mar '20

16 Mar '20

From: Jerome Brunet <jbrunet(a)baylibre.com> [ Upstream commit 22946f37557e27697aabc8e4f62642bfe4a17fd8 ] Reset the g12a hdmi codec glue on probe. This ensure a sane startup state. Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> Link: https://lore.kernel.org/r/20200221121146.1498427-1-jbrunet@baylibre.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/meson/g12a-tohdmitx.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/meson/g12a-tohdmitx.c b/sound/soc/meson/g12a-tohdmitx.c index 9cfbd343a00c8..8a0db28a6a406 100644 --- a/sound/soc/meson/g12a-tohdmitx.c +++ b/sound/soc/meson/g12a-tohdmitx.c @@ -8,6 +8,7 @@ #include <linux/module.h> #include <sound/pcm_params.h> #include <linux/regmap.h> +#include <linux/reset.h> #include <sound/soc.h> #include <sound/soc-dai.h> @@ -378,6 +379,11 @@ static int g12a_tohdmitx_probe(struct platform_device *pdev) struct device *dev = &pdev->dev; void __iomem *regs; struct regmap *map; + int ret; + + ret = device_reset(dev); + if (ret) + return ret; regs = devm_platform_ioremap_resource(pdev, 0); if (IS_ERR(regs)) -- 2.20.1

1 0

[PATCH v2 00/10] ASoC: Intel: machine driver updates for 5.7
by Pierre-Louis Bossart 15 Mar '20

15 Mar '20

Fix HDaudio/HDMI Kconfig dependencies, cleanup code, and add support for 2 new configs (DA7219+Maxim 98360A and RT5692+RT1015) Note that the last patch is the first case where we added the Reviewed-by tags automagically from the list of GitHub approvers. Feedback/comments welcome on the format, and thanks to Ranjani for the nice script! Changes since v1: Change base to broonie/for-5.7 Rebase and fix for_each_codec_dais() macro in Patch 9. Add new patch to remove GFP_ATOMIC Guennadi Liakhovetski (3): ASoC: codecs: hdac_hdmi: (cosmetic) remove redundant variable initialisations ASoC: Intel: skylake: (cosmetic) remove redundant variable initialisations ASoC: Intel: (cosmetic) simplify structure member access Kai Vehmanen (4): ASoC: Intel: boards: drop reverse deps for SND_HDA_CODEC_HDMI ASoC: Intel: sof_pcm512x: drop reverse deps for SND_HDA_CODEC_HDMI ASoC: Intel: sof_pcm512x: make HDMI optional for all platforms ASoC: SOF: Intel: hda: remove SND_SOC_SOF_HDA_COMMON_HDMI_CODEC Pierre-Louis Bossart (1): ASoC: Intel: don't use GFP_ATOMIC for machine driver contexts Yong Zhi (2): ASoC: Intel: sof_da7219_max98373: Add support for max98360a speaker amp ASoC: Intel: sof_rt5682: Add rt1015 speaker amp support sound/soc/codecs/hdac_hdmi.c | 6 +- sound/soc/intel/boards/Kconfig | 20 ++-- sound/soc/intel/boards/bdw-rt5650.c | 2 +- sound/soc/intel/boards/bdw-rt5677.c | 2 +- sound/soc/intel/boards/broadwell.c | 2 +- sound/soc/intel/boards/bxt_da7219_max98357a.c | 2 +- sound/soc/intel/boards/bxt_rt298.c | 2 +- sound/soc/intel/boards/bytcht_da7213.c | 2 +- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 2 +- sound/soc/intel/boards/cht_bsw_nau8824.c | 2 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 2 +- sound/soc/intel/boards/cml_rt1011_rt5682.c | 4 +- sound/soc/intel/boards/glk_rt5682_max98357a.c | 2 +- sound/soc/intel/boards/haswell.c | 2 +- sound/soc/intel/boards/kbl_rt5663_max98927.c | 2 +- .../intel/boards/kbl_rt5663_rt5514_max98927.c | 2 +- sound/soc/intel/boards/skl_hda_dsp_generic.c | 2 +- .../soc/intel/boards/skl_nau88l25_max98357a.c | 2 +- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 2 +- sound/soc/intel/boards/sof_da7219_max98373.c | 78 +++++++++++-- sound/soc/intel/boards/sof_pcm512x.c | 38 ++++-- sound/soc/intel/boards/sof_rt5682.c | 110 +++++++++++++++++- .../intel/common/soc-acpi-intel-jsl-match.c | 34 +++++- sound/soc/intel/skylake/skl.c | 4 +- sound/soc/sof/intel/Kconfig | 11 -- sound/soc/sof/intel/hda.c | 3 +- 26 files changed, 269 insertions(+), 71 deletions(-) base-commit: 101247a3b86e1cc0e382b7e887a56176290fc957 -- 2.20.1

3 14

[PATCH Fix for Kingston HyperX Amp (0951:16d8) 0/2 (Try #2)]
by crwulff＠gmail.com 14 Mar '20

14 Mar '20

From: Chris Wulff <crwulff(a)gmail.com> This patch set is modified to use an inline function for the mixer control interface lookup. The attempt at preventing modifications to a PCM that has previously been registered has been changed to defer registration instead, and has been split into a separate patch. This approach now works with pulseaudio.

2 3

Re: Re: [alsa-devel] [PATCH 4/7] ALSA: intel8x0: constify snd_pcm_ops structures
by sandra luna 14 Mar '20

14 Mar '20

1 0

[PATCH 00/16] SoundWire: cadence: add clock stop and fix programming sequences
by Pierre-Louis Bossart 14 Mar '20

14 Mar '20

To make progress with SoundWire support, this patchset provides the missing support for clock stop modes, and revisits all Cadence Master register settings. The current code is for some reason not aligned with internal documentation and hardware recommended flows, specifically for multi-link operation. Pierre-Louis Bossart (12): soundwire: cadence: s/update_config/config_update soundwire: cadence: handle error cases with CONFIG_UPDATE soundwire: cadence: mask Slave interrupt before stopping clock soundwire: cadence: merge routines to clear/set bits soundwire: cadence: move clock/SSP related inits to dedicated function soundwire: cadence: make SSP interval programmable soundwire: cadence: reorder MCP_CONFIG settings soundwire: cadence: enable NORMAL operation in cdns_init() soundwire: cadence: remove PREQ_DELAY assignment soundwire: cadence: remove automatic command retries soundwire: cadence: commit changes in the exit_reset() sequence soundwire: cadence: multi-link support Rander Wang (4): soundwire: cadence: simplifiy cdns_init() soundwire: cadence: add interface to check clock status soundwire: cadence: add clock_stop/restart routines soundwire: cadence: fix a io timeout issue in S3 test drivers/soundwire/cadence_master.c | 297 ++++++++++++++++++++++++----- drivers/soundwire/cadence_master.h | 9 +- drivers/soundwire/intel.c | 2 +- 3 files changed, 261 insertions(+), 47 deletions(-) base-commit: 5de79ba865d7770c3bdde7c266ed425832764aac -- 2.20.1

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[alsa-devel] [PATCH 0/6] ASoC: soc-acpi: add support for SoundWire-based machines
by Pierre-Louis Bossart 14 Mar '20

14 Mar '20

SoundWire support on Intel platforms relies on ACPI _ADR fields exposed in DSDT tables, with a 64-bit value defining which link a Slave device is connected to and its partId/manufacturerID/uniqueID. The existing definitions based on a globally-visible _HID used for I2C/I2S are no longer sufficient and need to be extended to detect during the probe steps which machine driver should be selected. This patchset adds the structure changes, and a set of static tables already used by Intel developers and customers. These tables are provided ahead of time, but they will only be used once all SoundWire-related parts are merged upstream and SoundWire is enabled in the builds. For reference, the complete set of 100+ patches required for SoundWire on Intel platforms is available here: https://github.com/thesofproject/linux/pull/1692 Bard Liao (1): ASoC: Intel: common: add match tables for ICL w/ SoundWire Pierre-Louis Bossart (4): ASoC: soc-acpi: add _ADR-based link descriptors ASoC: Intel: common: soc-acpi: declare new tables for SoundWire ASoC: Intel: common: add match tables for TGL w/ SoundWire ASoC: SOF: Intel: reference SoundWire machine lists Rander Wang (1): ASoC: Intel: common: add match tables for CNL/CFL/CML w/ SoundWire include/sound/soc-acpi-intel-match.h | 6 ++ include/sound/soc-acpi.h | 21 ++++ .../intel/common/soc-acpi-intel-cfl-match.c | 5 + .../intel/common/soc-acpi-intel-cml-match.c | 89 +++++++++++++++++ .../intel/common/soc-acpi-intel-cnl-match.c | 5 + .../intel/common/soc-acpi-intel-icl-match.c | 98 +++++++++++++++++++ .../intel/common/soc-acpi-intel-tgl-match.c | 51 +++++++++- sound/soc/sof/sof-pci-dev.c | 5 + 8 files changed, 278 insertions(+), 2 deletions(-) base-commit: b2e2a13796889e10ba0390ab338949ba63d89462 -- 2.20.1

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[PATCH 3/3] ALSA USB MIDI: Make amount of MIDI events per output URB selectable by user
by Andreas Steinmetz 14 Mar '20

14 Mar '20

This patch introduces a new module parameter "events" which allows userspace to limit the amount of MIDI events per output URB. An "events" value of 0 which is the default means "use wMaxPacketSize". Out of range values for "events" are clamped to the nearest valid value. It is assumed that the previous patches are applied and "outqueue=1" is used. Looking then at a four port USB MIDI interface with a wMaxPacketSize of 64 and assuming an optimal output rate of 320us/byte the following is true: 16 MIDI events PER URB Up to 3 bytes per MIDI event (3 bytes are assumed below) A driver queue of 1 URB which is completely filled. Resulting latency is 16*3*1*320us=15.36ms which still is in the audible range (a latency of less than 5ms is required, think playing a keyboard to MIDI in and using MIDI out to control a piano module and using headphones, while using another port for sysex transfers). After applying this patch and selecting the valid minimum event value of 4 (there must be at least space for one event per port to assert proper output balancing) tests show that the throughput isn't really affected, the worst case latency however drops to 3.84ms which is acceptable. In case of the largest class compliant devices which do have 16 ports the worst case latency is then 15.36ms which is tolerable. If this is still to long it could be fixed by introducing additional complexity to the balancing of snd_usbmidi_standard_output() at some later stage. As the device default is used by default no implications are expected for existing users. By making the parameter writable via sysfs it is possible to do selective output URB event count management from userspace via a script that manages USB authorize and the module option. Signed-off-by: Andreas Steinmetz <ast(a)domdv.de> --- a/sound/usb/midi.c 2020-03-13 20:18:33.443265194 +0100 +++ b/sound/usb/midi.c 2020-03-13 20:42:49.445546326 +0100 @@ -79,9 +79,12 @@ MODULE_LICENSE("Dual BSD/GPL"); static int outqueue = OUTPUT_URBS; +static int events; module_param(outqueue, int, 0644); -MODULE_PARM_DESC(outqueue, "Outgoing queue size, 1-7 (default: 7)."); +MODULE_PARM_DESC(outqueue, "Outgoing MIDI queue size, 1-7 (default: 7)."); +module_param(events, int, 0644); +MODULE_PARM_DESC(events, "MIDI events per queue element, 0=device default (default: 0)."); struct usb_ms_header_descriptor { @@ -1426,6 +1429,17 @@ switch (umidi->usb_id) { default: ep->max_transfer = usb_maxpacket(umidi->dev, pipe, 1); + if (events) { + int ev = 0; + + for (i = 0; i < 0x10; ++i) + if (ep_info->out_cables & (1 << i)) + ev++; + if (events > ev) + ev = events; + if (ev < ep->max_transfer >> 2) + ep->max_transfer = ev << 2; + } break; /* * Various chips declare a packet size larger than 4 bytes, but

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