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[PATCH] ASoC: ak5558: Use modern ASoC DAI format terminology
by Mark Brown 27 Sep '21

27 Sep '21

As part of moving to remove the old style defines for the bus clocks update the ak5558 driver to use more modern terminology for clocking. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/ak5558.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/ak5558.c b/sound/soc/codecs/ak5558.c index 37d4600b6f2c..c94cfde3e4a8 100644 --- a/sound/soc/codecs/ak5558.c +++ b/sound/soc/codecs/ak5558.c @@ -198,13 +198,13 @@ static int ak5558_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) struct snd_soc_component *component = dai->component; u8 format; - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBS_CFS: + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBC_CFC: break; - case SND_SOC_DAIFMT_CBM_CFM: + case SND_SOC_DAIFMT_CBP_CFP: break; - case SND_SOC_DAIFMT_CBS_CFM: - case SND_SOC_DAIFMT_CBM_CFS: + case SND_SOC_DAIFMT_CBC_CFP: + case SND_SOC_DAIFMT_CBP_CFC: default: dev_err(dai->dev, "Clock mode unsupported"); return -EINVAL; -- 2.20.1

1 1

[PATCH] ASoC: cs42l42: Add HP Volume Scale control
by Vitaly Rodionov 27 Sep '21

27 Sep '21

From: Richard Fitzgerald <rf(a)opensource.cirrus.com> cs42l42 has a configurable scaling of the maximum volume. Add an ALSA control for this. Note that the datasheet name is "full scale volume" but this conflicts with ALSA naming convention so the control is named "HP Volume Scale". Before this change the FULL_SCALE_VOLUME was set based on the value in RLA_STAT, but as there isn't any load detection result this always set the scaling to -6dB instead of the default 0dB. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> Signed-off-by: Vitaly Rodionov <vitalyr(a)opensource.cirrus.com> --- sound/soc/codecs/cs42l42.c | 25 ++++++++++++------------- 1 file changed, 12 insertions(+), 13 deletions(-) diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index d5e1e5228b5f..78f8c89410b6 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -425,6 +425,14 @@ static SOC_ENUM_SINGLE_DECL(cs42l42_wnf3_freq_enum, CS42L42_ADC_WNF_HPF_CTL, CS42L42_ADC_WNF_CF_SHIFT, cs42l42_wnf3_freq_text); +static const char * const cs42l42_full_scale_vol_text[] = { + "0dB", "-6dB" +}; + +static SOC_ENUM_SINGLE_DECL(cs42l42_full_scale_vol_enum, CS42L42_HP_CTL, + CS42L42_HP_FULL_SCALE_VOL_SHIFT, + cs42l42_full_scale_vol_text); + static const struct snd_kcontrol_new cs42l42_snd_controls[] = { /* ADC Volume and Filter Controls */ SOC_SINGLE("ADC Notch Switch", CS42L42_ADC_CTL, @@ -450,6 +458,7 @@ static const struct snd_kcontrol_new cs42l42_snd_controls[] = { CS42L42_DACB_INV_SHIFT, true, false), SOC_SINGLE("DAC HPF Switch", CS42L42_DAC_CTL2, CS42L42_DAC_HPF_EN_SHIFT, true, false), + SOC_ENUM("HP Volume Scale", cs42l42_full_scale_vol_enum), SOC_DOUBLE_R_TLV("Mixer Volume", CS42L42_MIXER_CHA_VOL, CS42L42_MIXER_CHB_VOL, CS42L42_MIXER_CH_VOL_SHIFT, 0x3f, 1, mixer_tlv) @@ -922,7 +931,6 @@ static int cs42l42_mute_stream(struct snd_soc_dai *dai, int mute, int stream) struct snd_soc_component *component = dai->component; struct cs42l42_private *cs42l42 = snd_soc_component_get_drvdata(component); unsigned int regval; - u8 fullScaleVol; int ret; if (mute) { @@ -993,20 +1001,11 @@ static int cs42l42_mute_stream(struct snd_soc_dai *dai, int mute, int stream) cs42l42->stream_use |= 1 << stream; if (stream == SNDRV_PCM_STREAM_PLAYBACK) { - /* Read the headphone load */ - regval = snd_soc_component_read(component, CS42L42_LOAD_DET_RCSTAT); - if (((regval & CS42L42_RLA_STAT_MASK) >> CS42L42_RLA_STAT_SHIFT) == - CS42L42_RLA_STAT_15_OHM) { - fullScaleVol = CS42L42_HP_FULL_SCALE_VOL_MASK; - } else { - fullScaleVol = 0; - } - - /* Un-mute the headphone, set the full scale volume flag */ + /* Un-mute the headphone */ snd_soc_component_update_bits(component, CS42L42_HP_CTL, CS42L42_HP_ANA_AMUTE_MASK | - CS42L42_HP_ANA_BMUTE_MASK | - CS42L42_HP_FULL_SCALE_VOL_MASK, fullScaleVol); + CS42L42_HP_ANA_BMUTE_MASK, + 0); } } -- 2.25.1

2 1

Broken Pipe with SNDRV_PCM_IOCTL_WRITEI_FRAMES
by Ryan McClue 27 Sep '21

27 Sep '21

Hello again. I'm using alsa-kernel. My configuration: SNDRV_PCM_HW_PARAM_ACCESS = SNDRV_PCM_ACCESS_RW_INTERLEAVED SNDRV_PCM_HW_PARAM_FORMAT = SNDRV_PCM_FORMAT_S16_LE SNDRV_PCM_HW_PARAM_SUBFORMAT = SNDRV_PCM_SUBFORMAT_STD SNDRV_PCM_HW_PARAM_CHANNELS = 2 SNDRV_PCM_HW_PARAM_RATE = 48000 I have a vsynced game loop that is running at 60fps: > int num_base_samples = 48000 * (1 / 60); > int num_samples = num_base_samples * 2; > int16_t buffer[num_samples] = {}; > > while (true) { > int16_t *samples = buffer; > for (int sample_i = 0; sample_i < num_base_samples; sample_i++) { > *samples++ = 0x33; > *samples++ = 0x33; > } > > struct snd_xferi transfer = {}; > transfer.buf = buffer; > transfer.frames = num_base_samples; > ioctl(fd, SNDRV_PCM_IOCTL_WRITEI_FRAMES, &transfer); > > // ioctl(fd, SNDRV_PCM_IOCTL_DRAIN); --> NECESSARY??? > // ioctl(fd, SNDRV_PCM_IOCTL_PREPARE); --> NECESSARY??? > } On the first iteration of SNDRV_PCM_IOCTL_WRITEI_FRAMES I get no error. All subsequent iterations, I get Broken Pipe error. So, to counteract this at the end of each frame I call SNDRV_PCM_IOCTL_DRAIN and SNDRV_PCM_IOCTL_PREPARE. This removes the Broken Pipe error however slows the program down by half and no sound is heard. How best to solve this issue? Thanks again -- Ryan McClue, Sydney

2 3

[PATCH v2 00/12] Add support for on demand pipeline setup/destroy
by Daniel Baluta 27 Sep '21

27 Sep '21

From: Daniel Baluta <daniel.baluta(a)nxp.com> This patchseries implements the new feature to setup/teardown pipeline as needed when a PCM is open/closed. This only only support single cores, with support for multicore arriving with a future patchseries. Review with SOF community at https://github.com/thesofproject/linux/pull/2794 Changes since v1: - added my own Signed-off-by tag. Ranjani Sridharan (12): ASoC: topology: change the complete op in snd_soc_tplg_ops to return int ASoC: SOF: control: Add access field in struct snd_sof_control ASoC: SOF: topology: Add new token for dynamic pipeline ASoC: SOF: sof-audio: add helpers for widgets, kcontrols and dai config set up AsoC: dapm: export a couple of functions ASoC: SOF: Add new fields to snd_sof_route ASoC: SOF: restore kcontrols for widget during set up ASoC: SOF: Don't set up widgets during topology parsing ASoC: SOF: Introduce widget use_count ASoC: SOF: Intel: hda: make sure DAI widget is set up before IPC ASoC: SOF: Add support for dynamic pipelines ASoC: SOF: topology: Add kernel parameter for topology verification include/sound/soc-dpcm.h | 1 + include/sound/soc-topology.h | 2 +- include/uapi/sound/sof/tokens.h | 1 + sound/soc/intel/skylake/skl-topology.c | 6 +- sound/soc/soc-dapm.c | 2 + sound/soc/soc-pcm.c | 4 +- sound/soc/soc-topology.c | 10 +- sound/soc/sof/intel/hda-dai.c | 176 +++--- sound/soc/sof/intel/hda.c | 177 ++++-- sound/soc/sof/intel/hda.h | 5 + sound/soc/sof/ipc.c | 22 + sound/soc/sof/pcm.c | 58 +- sound/soc/sof/pm.c | 4 +- sound/soc/sof/sof-audio.c | 709 +++++++++++++++++++------ sound/soc/sof/sof-audio.h | 32 +- sound/soc/sof/sof-priv.h | 1 + sound/soc/sof/topology.c | 362 +++++-------- 17 files changed, 1034 insertions(+), 538 deletions(-) -- 2.27.0

4 17

[PATCH v3] ASoC: Intel: boards: add max98390 2/4 speakers support
by Mac Chiang 24 Sep '21

24 Sep '21

support 2 hw boards. 1. SSP2 connects max98390, 2 speakers. 2. SSP1 connects max98390, 2/4 speakers. 2 or 4 speakers playback add echo reference capture add bt offload support add DMI_OEM_STRING for board varients add ALC5682I-VS support 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> Signed-off-by: Brent Lu <brent.lu(a)intel.com> --- v3: - add acpi_dev_present() for additional tweeter condition, otherwise stereo speakers. - add ALC5682I-VS headphone codec support --- sound/soc/intel/boards/Kconfig | 1 + sound/soc/intel/boards/sof_maxim_common.c | 181 ++++++++++++++++++ sound/soc/intel/boards/sof_maxim_common.h | 16 ++ sound/soc/intel/boards/sof_rt5682.c | 84 ++++++++ .../intel/common/soc-acpi-intel-adl-match.c | 21 ++ sound/soc/sof/sof-pci-dev.c | 9 + 6 files changed, 312 insertions(+) diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 3e20c697b569..760d845ac6a1 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..ea5058a5b331 100644 --- a/sound/soc/intel/boards/sof_maxim_common.c +++ b/sound/soc/intel/boards/sof_maxim_common.c @@ -5,6 +5,7 @@ #include <linux/string.h> #include <sound/pcm.h> #include <sound/soc.h> +#include <sound/soc-acpi.h> #include <sound/soc-dai.h> #include <sound/soc-dapm.h> #include <uapi/sound/asound.h> @@ -133,6 +134,186 @@ 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 + */ +const struct snd_soc_dapm_route max_98390_dapm_routes[] = { + /* speaker */ + { "Left Spk", NULL, "Left BE_OUT" }, + { "Right Spk", NULL, "Right BE_OUT" }, +}; + +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" }, +}; + +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; + + /* add regular speakers dapm route */ + ret = snd_soc_dapm_add_routes(&card->dapm, max_98390_dapm_routes, + ARRAY_SIZE(max_98390_dapm_routes)); + if (ret) { + dev_err(rtd->dev, "unable to add Left/Right Speaker dapm, ret %d\n", ret); + return ret; + } + + /* add widgets/controls/dapm for tweeter speakers */ + if (acpi_dev_present("MX98390", "3", -1)) { + 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 tweeter 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 tweeter card controls, ret %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, + "unable to add Tweeter Left/Right Speaker dapm, ret %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..9dd69f5d8721 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; + +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 3f6f19d9b19e..774e4dd9539c 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -59,6 +59,9 @@ (((quirk) << SOF_BT_OFFLOAD_SSP_SHIFT) & SOF_BT_OFFLOAD_SSP_MASK) #define SOF_SSP_BT_OFFLOAD_PRESENT BIT(22) #define SOF_RT5682S_HEADPHONE_CODEC_PRESENT BIT(23) +#define SOF_MAX98390_SPEAKER_AMP_PRESENT BIT(24) +#define SOF_MAX98390_TWEETER_SPEAKER_PRESENT BIT(25) + /* Default: MCLK on, MCLK 19.2M, SSP0 */ static unsigned long sof_rt5682_quirk = SOF_RT5682_MCLK_EN | @@ -179,6 +182,36 @@ 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), + + }, {} }; @@ -486,6 +519,7 @@ 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); } @@ -784,6 +818,20 @@ 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); + } else { + links[id].codecs = max_98390_components; + links[id].num_codecs = ARRAY_SIZE(max_98390_components); + } + links[id].init = max_98390_spk_codec_init; + links[id].ops = &max_98390_ops; + links[id].dpcm_capture = 1; + } else { max_98357a_dai_link(&links[id]); } @@ -864,6 +912,10 @@ static int sof_audio_probe(struct platform_device *pdev) if ((sof_rt5682_quirk & SOF_SPEAKER_AMP_PRESENT) && !mach->quirk_data) sof_rt5682_quirk &= ~SOF_SPEAKER_AMP_PRESENT; + /* Detect the headset codec variant to support machines in DMI quirk */ + if (acpi_dev_present("RTL5682", NULL, -1)) + sof_rt5682_quirk |= SOF_RT5682S_HEADPHONE_CODEC_PRESENT; + if (soc_intel_is_byt() || soc_intel_is_cht()) { is_legacy_cpu = 1; dmic_be_num = 0; @@ -920,6 +972,14 @@ 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++; @@ -1046,6 +1106,29 @@ 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), + }, + { + .name = "adl_mx98390_rt5682s", + .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | + SOF_RT5682_SSP_CODEC(0) | + SOF_RT5682S_HEADPHONE_CODEC_PRESENT | + 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), + }, { .name = "jsl_rt5682s_rt1015", .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | @@ -1095,6 +1178,7 @@ 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..a62a78edc855 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 +302,22 @@ 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", + }, + { + .id = "RTL5682", + .drv_name = "adl_mx98390_rt5682s", + .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

2 1

include a sample `alsamixer.rc` file?
by GitHub issues - opened 24 Sep '21

24 Sep '21

alsa-project/alsa-utils issue #112 was opened from int-e: I think it would be useful to include an example `alsamixer.rc` file; see https://gist.github.com/int-e/25bd77d4e59531fad7d48bfaf1fc1046 for a candidate. In particular this example replicates the default color theme, making it easier to adjust (say, replace `default` by `black` to get sane colors back in terminals with bright backgrounds). The default keybindings are also replicated but that's probably less useful. Issue URL : https://github.com/alsa-project/alsa-utils/issues/112 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH v2] ALSA: rawmidi: introduce SNDRV_RAWMIDI_IOCTL_USER_PVERSION
by Jaroslav Kysela 23 Sep '21

23 Sep '21

The new framing mode causes the user space regression, because the alsa-lib code does not initialize the reserved space in the params structure when the device is opened. This change adds SNDRV_RAWMIDI_IOCTL_USER_PVERSION like we do for the PCM interface for the protocol acknowledgment. Cc: David Henningsson <coding(a)diwic.se> Cc: <stable(a)vger.kernel.org> Fixes: 08fdced60ca0 ("ALSA: rawmidi: Add framing mode") BugLink: https://github.com/alsa-project/alsa-lib/issues/178 Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> --- include/sound/rawmidi.h | 1 + include/uapi/sound/asound.h | 1 + sound/core/rawmidi.c | 9 +++++++++ 3 files changed, 11 insertions(+) diff --git a/include/sound/rawmidi.h b/include/sound/rawmidi.h index 989e1517332d..7a08ed2acd60 100644 --- a/include/sound/rawmidi.h +++ b/include/sound/rawmidi.h @@ -98,6 +98,7 @@ struct snd_rawmidi_file { struct snd_rawmidi *rmidi; struct snd_rawmidi_substream *input; struct snd_rawmidi_substream *output; + unsigned int user_pversion; /* supported protocol version */ }; struct snd_rawmidi_str { diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index 1d84ec9db93b..5859ca0a1439 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -784,6 +784,7 @@ struct snd_rawmidi_status { #define SNDRV_RAWMIDI_IOCTL_PVERSION _IOR('W', 0x00, int) #define SNDRV_RAWMIDI_IOCTL_INFO _IOR('W', 0x01, struct snd_rawmidi_info) +#define SNDRV_RAWMIDI_IOCTL_USER_PVERSION _IOW('W', 0x02, int) #define SNDRV_RAWMIDI_IOCTL_PARAMS _IOWR('W', 0x10, struct snd_rawmidi_params) #define SNDRV_RAWMIDI_IOCTL_STATUS _IOWR('W', 0x20, struct snd_rawmidi_status) #define SNDRV_RAWMIDI_IOCTL_DROP _IOW('W', 0x30, int) diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index 6c0a4a67ad2e..6f30231bdb88 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -873,12 +873,21 @@ static long snd_rawmidi_ioctl(struct file *file, unsigned int cmd, unsigned long return -EINVAL; } } + case SNDRV_RAWMIDI_IOCTL_USER_PVERSION: + if (get_user(rfile->user_pversion, (unsigned int __user *)arg)) + return -EFAULT; + return 0; + case SNDRV_RAWMIDI_IOCTL_PARAMS: { struct snd_rawmidi_params params; if (copy_from_user(&params, argp, sizeof(struct snd_rawmidi_params))) return -EFAULT; + if (rfile->user_pversion < SNDRV_PROTOCOL_VERSION(2, 0, 2)) { + params.mode = 0; + memset(params.reserved, 0, sizeof(params.reserved)); + } switch (params.stream) { case SNDRV_RAWMIDI_STREAM_OUTPUT: if (rfile->output == NULL) -- 2.31.1

2 1

[PATCH v7 00/22] ASoC: qcom: Add AudioReach support
by Srinivas Kandagatla 22 Sep '21

22 Sep '21

Many thanks for reviewing v6 This version addresses all the comments raised as part of v7 review. This patchset adds ASoC driver support to configure signal processing framework ("AudioReach") which is integral part of Qualcomm next generation audio SDK and will be deployed on upcoming Qualcomm chipsets. It makes use of ASoC Topology to load graphs on to the DSP which is then managed by APM (Audio Processing Manager) service to prepare/start/stop. Here is simplified high-level block diagram of AudioReach: ___________________________________________________________ | CPU (Application Processor) | | +---------+ +---------+ +----------+ | | | q6apm | | q6apm | | q6apm | | | | dais | <------> | | <-----> |lpass-dais| | | +---------+ +---------+ +----------+ | | ^ ^ | | | | +---------+ | | +---------+ v +---------->|topology | | | | q6prm | +---------+ | | | | | |<-------->| GPR | +---------+ | | +---------+ +---------+ | | ^ ^ | | | | | | +----------+ | | | | q6prm | | | | |lpass-clks| | | | +----------+ | | |____________________________|______________________________| | | RPMSG (IPC over GLINK) ____________________________|______________________________ | | | | +-----------------------+ | | | | | | v v q6 (Audio DSP) | |+-----+ +----------------------------------+ | || PRM | | APM (Audio Processing Manager) | | |+-----+ | . Graph Management | | | | . Command Handing | | | | . Event Management | | | | ... | | | +----------------------------------+ | | ^ | |____________________________|______________________________| | | LPASS AIF ____________________________|______________________________ | | Audio I/O | | v | | +--------------------------------------------------+ | | | Audio devices | | | | CODEC | HDMI-TX | PCM | SLIMBUS | I2S |MI2S |...| | | | | | | +--------------------------------------------------+ | |___________________________________________________________| AudioReach has constructs of sub-graph, container and modules. Each sub-graph can have N containers and each Container can have N Modules and connections between them can be linear or non-linear. An audio function can be realized with one or many connected sub-graphs. There are also control/event paths between modules that can be wired up while building graph to achieve various control mechanism between modules. These concepts of Sub-Graph, Containers and Modules are represented in ASoC topology. Here is simple I2S graph with a Write Shared Memory and a Volume control module within a single Subgraph (1) with one Container (1) and 5 modules. ____________________________________________________________ | Sub-Graph [1] | | _______________________________________________________ | | | Container [1] | | | | [WR_SH] -> [PCM DEC] -> [PCM CONV] -> [VOL]-> [I2S-EP]| | | |_______________________________________________________| | |____________________________________________________________| For now this graph is split into two subgraphs to achieve dpcm like below: ________________________________________________ _________________ | Sub-Graph [1] | | Sub-Graph [2] | | ____________________________________________ | | _____________ | | | Container [1] | | | |Container [2]| | | | [WR_SH] -> [PCM DEC] -> [PCM CONV] -> [VOL]| | | | [I2S-EP] | | | |____________________________________________| | | |_____________| | |________________________________________________| |_________________| _________________ | Sub-Graph [3] | | _____________ | | |Container [3]| | | | [DMA-EP] | | | |_____________| | |_________________| This patchset adds very minimal support for AudioReach which includes supporting sub-graphs containing CODEC DMA ports and simple PCM Decoder/Encoder and Logger Modules. Additional capabilities will be built over time to expose features offered by AudioReach. This patchset is Tested on SM8250 SoC based Qualcomm Robotics Platform RB5 and SM9250 MTP with WSA881X Smart Speaker Amplifiers, DMICs connected via VA Macro and WCD938x Codec connected via TX and RX Macro and HDMI audio via I2S. First 10 Patches are mostly reorganization existing Old QDSP Audio Framework code and bindings so that we could reuse them on AudioReach. ASoC topology graphs for DragonBoard RB5 and SM8250 MTP are available at https://git.linaro.org/people/srinivas.kandagatla/audioreach-topology.git/ and Qualcomm AudioReach DSP headers are available at: https://source.codeaurora.org/quic/la/platform/vendor/opensource/arspf-head… Note: There is one false positive warning in this patchset: audioreach.c:80:45: warning: array of flexible structures Thanks, srini Changes since v6: - fixed last usage of flexible array in apr - removed leftover comments - added support to derive topology firmware name from soundcard to support different variants, as suggested by Pierre. Srinivas Kandagatla (22): soc: dt-bindings: qcom: apr: convert to yaml soc: dt-bindings: qcom: apr: deprecate qcom,apr-domain property soc: qcom: apr: make code more reuseable soc: dt-bindings: qcom: add gpr bindings soc: qcom: apr: Add GPR support ASoC: dt-bindings: move LPASS dai related bindings out of q6afe ASoC: dt-bindings: move LPASS clocks related bindings out of q6afe ASoC: dt-bindings: rename q6afe.h to q6dsp-lpass-ports.h ASoC: qdsp6: q6afe-dai: move lpass audio ports to common file ASoC: qdsp6: q6afe-clocks: move audio-clocks to common file ASoC: dt-bindings: q6dsp: add q6apm-lpass-dai compatible ASoC: dt-bindings: lpass-clocks: add q6prm clocks compatible ASoC: dt-bindings: add q6apm digital audio stream bindings ASoC: qdsp6: audioreach: add basic pkt alloc support ASoC: qdsp6: audioreach: add q6apm support ASoC: qdsp6: audioreach: add module configuration command helpers ASoC: qdsp6: audioreach: add Kconfig and Makefile ASoC: qdsp6: audioreach: add topology support ASoC: qdsp6: audioreach: add q6apm-dai support ASoC: qdsp6: audioreach: add q6apm lpass dai support ASoC: qdsp6: audioreach: add q6prm support ASoC: qdsp6: audioreach: add support for q6prm-clocks .../devicetree/bindings/soc/qcom/qcom,apr.txt | 134 -- .../bindings/soc/qcom/qcom,apr.yaml | 177 +++ .../devicetree/bindings/sound/qcom,q6afe.txt | 181 --- .../bindings/sound/qcom,q6apm-dai.yaml | 53 + .../sound/qcom,q6dsp-lpass-clocks.yaml | 77 ++ .../sound/qcom,q6dsp-lpass-ports.yaml | 205 +++ drivers/soc/qcom/Kconfig | 2 +- drivers/soc/qcom/apr.c | 287 ++++- include/dt-bindings/soc/qcom,gpr.h | 19 + include/dt-bindings/sound/qcom,q6afe.h | 203 +-- .../sound/qcom,q6dsp-lpass-ports.h | 208 +++ include/linux/soc/qcom/apr.h | 70 +- include/uapi/sound/snd_ar_tokens.h | 208 +++ sound/soc/qcom/Kconfig | 22 + sound/soc/qcom/qdsp6/Makefile | 11 +- sound/soc/qcom/qdsp6/audioreach.c | 1135 +++++++++++++++++ sound/soc/qcom/qdsp6/audioreach.h | 726 +++++++++++ sound/soc/qcom/qdsp6/q6afe-clocks.c | 187 +-- sound/soc/qcom/qdsp6/q6afe-dai.c | 687 +--------- sound/soc/qcom/qdsp6/q6apm-dai.c | 415 ++++++ sound/soc/qcom/qdsp6/q6apm-lpass-dais.c | 260 ++++ sound/soc/qcom/qdsp6/q6apm.c | 836 ++++++++++++ sound/soc/qcom/qdsp6/q6apm.h | 152 +++ sound/soc/qcom/qdsp6/q6dsp-lpass-clocks.c | 186 +++ sound/soc/qcom/qdsp6/q6dsp-lpass-clocks.h | 30 + sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c | 627 +++++++++ sound/soc/qcom/qdsp6/q6dsp-lpass-ports.h | 22 + sound/soc/qcom/qdsp6/q6prm-clocks.c | 85 ++ sound/soc/qcom/qdsp6/q6prm.c | 202 +++ sound/soc/qcom/qdsp6/q6prm.h | 78 ++ sound/soc/qcom/qdsp6/topology.c | 1133 ++++++++++++++++ 31 files changed, 7203 insertions(+), 1415 deletions(-) delete mode 100644 Documentation/devicetree/bindings/soc/qcom/qcom,apr.txt create mode 100644 Documentation/devicetree/bindings/soc/qcom/qcom,apr.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6apm-dai.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6dsp-lpass-clocks.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6dsp-lpass-ports.yaml create mode 100644 include/dt-bindings/soc/qcom,gpr.h create mode 100644 include/dt-bindings/sound/qcom,q6dsp-lpass-ports.h create mode 100644 include/uapi/sound/snd_ar_tokens.h create mode 100644 sound/soc/qcom/qdsp6/audioreach.c create mode 100644 sound/soc/qcom/qdsp6/audioreach.h create mode 100644 sound/soc/qcom/qdsp6/q6apm-dai.c create mode 100644 sound/soc/qcom/qdsp6/q6apm-lpass-dais.c create mode 100644 sound/soc/qcom/qdsp6/q6apm.c create mode 100644 sound/soc/qcom/qdsp6/q6apm.h create mode 100644 sound/soc/qcom/qdsp6/q6dsp-lpass-clocks.c create mode 100644 sound/soc/qcom/qdsp6/q6dsp-lpass-clocks.h create mode 100644 sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c create mode 100644 sound/soc/qcom/qdsp6/q6dsp-lpass-ports.h create mode 100644 sound/soc/qcom/qdsp6/q6prm-clocks.c create mode 100644 sound/soc/qcom/qdsp6/q6prm.c create mode 100644 sound/soc/qcom/qdsp6/q6prm.h create mode 100644 sound/soc/qcom/qdsp6/topology.c -- 2.21.0

2 31

[PATCH 01/16] ASoC: eureka-tlv320: Update to modern clocking terminology
by Mark Brown 22 Sep '21

22 Sep '21

As part of moving to remove the old style defines for the bus clocks update the eureka-tlv320 driver to use more modern terminology for clocking. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/fsl/eukrea-tlv320.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/eukrea-tlv320.c b/sound/soc/fsl/eukrea-tlv320.c index e13271ea84de..8b61582753c8 100644 --- a/sound/soc/fsl/eukrea-tlv320.c +++ b/sound/soc/fsl/eukrea-tlv320.c @@ -70,7 +70,7 @@ static struct snd_soc_dai_link eukrea_tlv320_dai = { .name = "tlv320aic23", .stream_name = "TLV320AIC23", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM, + SND_SOC_DAIFMT_CBP_CFP, .ops = &eukrea_tlv320_snd_ops, SND_SOC_DAILINK_REG(hifi), }; -- 2.20.1

2 18

[PATCH] ASoC: cros_ec_codec: Use modern ASoC DAI format terminology
by Mark Brown 22 Sep '21

22 Sep '21

As part of moving to remove the old style defines for the bus clocks update the cros_ec_codec driver to use more modern terminology for clocking. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/cros_ec_codec.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cros_ec_codec.c b/sound/soc/codecs/cros_ec_codec.c index a201d652aca2..9b92e1a0d1a3 100644 --- a/sound/soc/codecs/cros_ec_codec.c +++ b/sound/soc/codecs/cros_ec_codec.c @@ -283,8 +283,8 @@ static int i2s_rx_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) struct ec_param_ec_codec_i2s_rx p; enum ec_codec_i2s_rx_daifmt daifmt; - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBS_CFS: + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBC_CFC: break; default: return -EINVAL; -- 2.20.1

2 2