October 2021 - Alsa-devel - mailman.alsa-project.org

[PATCH v4 0/6] Multiple headphone codec driver support
by Brent Lu 29 Oct '21

29 Oct '21

Support multiple headphone drivers in same machine driver. In this case, both rt5682 and rt5682s are supported and enumerated by different ACPI HID "10EC5682" and "RTL5682". V2 Changes: - remove useless 'NULL', 'false' in if-condition - can use 'comp_ids' field alone to enumerate driver - add comma to the end of entry in structure initialization - keep the table of byt/cht/cml/icl untouched V3 Changes: - upstreamd from SOF github, PR#3200 - use new compatiable IDs to shrink the enumerate table of BYT and CHT - add 'const' to snd_soc_acpi_codecs structures V4 Changes: - add signoff to patch 4~6 Brent Lu (3): ASoC: soc-acpi: add comp_ids field for machine driver matching ASoC: Intel: sof_rt5682: detect codec variant in probe function ASoC: Intel: sof_rt5682: use comp_ids to enumerate rt5682s Pierre-Louis Bossart (3): ASoC: Intel: soc-acpi-byt: shrink tables using compatible IDs ASoC: Intel: soc-acpi-cht: shrink tables using compatible IDs ASoC: Intel: soc-acpi: use const for all uses of snd_soc_acpi_codecs include/sound/soc-acpi.h | 3 + sound/soc/intel/boards/sof_rt5682.c | 34 ++------- .../intel/common/soc-acpi-intel-adl-match.c | 11 ++- .../intel/common/soc-acpi-intel-bxt-match.c | 2 +- .../intel/common/soc-acpi-intel-byt-match.c | 68 +++++++----------- .../intel/common/soc-acpi-intel-cht-match.c | 69 +++++++------------ .../intel/common/soc-acpi-intel-cml-match.c | 8 +-- .../intel/common/soc-acpi-intel-glk-match.c | 2 +- .../intel/common/soc-acpi-intel-jsl-match.c | 43 ++++-------- .../intel/common/soc-acpi-intel-kbl-match.c | 12 ++-- .../intel/common/soc-acpi-intel-skl-match.c | 2 +- .../intel/common/soc-acpi-intel-tgl-match.c | 11 ++- sound/soc/soc-acpi.c | 24 ++++++- 13 files changed, 119 insertions(+), 170 deletions(-) -- 2.25.1

2 8

[PATCH 0/6] ASoC: Sanity checks and soc-topology updates
by Cezary Rojewski 29 Oct '21

29 Oct '21

Couple of soc-topology related changes and a use-after-free fix. Said fix and two sanity checks for soc-topology lead the way. While the use-after-free is quite obvious, the sanity checks are here to cover for cases where user malformed the topology file -or- access to filesystem somehow got interrupted during copy operation. We shouldn't be reading outside the file boundary. Afterward a change to soc_tplg_add_kcontrol(): device being passed to soc_tplg_add_dcontrol() from comp->dev to tplg->dev which corrects dev_xxx() invoked later on. Also, device used for topology memory allocations from component->dev to component->card->dev so memory gets freed each time card device (usually platform device) is removed rather than the component device what may happen less frequently. Dummy component gets smarter and no longer overrides hw_params if there are other components accociated with related struct snd_soc_pcm_runtime instance. Amadeusz Sławiński (5): ASoC: core: Remove invalid snd_soc_component_set_jack call ASoC: topology: Check for dapm widget completeness ASoC: topology: Use correct device for prints ASoC: topology: Change topology device to card device ASoC: Stop dummy from overriding hwparams Cezary Rojewski (1): ASoC: topology: Add header payload_size verification sound/soc/soc-core.c | 3 --- sound/soc/soc-topology.c | 34 ++++++++++++++++++++++++++++++---- sound/soc/soc-utils.c | 13 +++++++++++++ 3 files changed, 43 insertions(+), 7 deletions(-) -- 2.25.1

2 7

[RESEND, v2 0/2] ASoC: mediatek: Add support for MT8195 sound card with rt1011 and rt5682
by Trevor Wu 29 Oct '21

29 Oct '21

This series of patches adds support for mt8195 board with mt6359, rt1011 and rt5682. Patches are based on broonie tree "for-next" branch. Changes since v1: - refined code based on Pierre-Louis's suggestions - added missing of_node_put - fixed wrong fixup assignment on HDMITX Trevor Wu (2): ASoC: mediatek: mt8195: add machine driver with mt6359, rt1011 and rt5682 dt-bindings: mediatek: mt8195: add mt8195-mt6359-rt1011-rt5682 document .../sound/mt8195-mt6359-rt1011-rt5682.yaml | 47 + sound/soc/mediatek/Kconfig | 15 + sound/soc/mediatek/mt8195/Makefile | 1 + .../mt8195/mt8195-mt6359-rt1011-rt5682.c | 1155 +++++++++++++++++ 4 files changed, 1218 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/mt8195-mt6359-rt1011-rt5682.yaml create mode 100644 sound/soc/mediatek/mt8195/mt8195-mt6359-rt1011-rt5682.c -- 2.18.0

2 3

[PATCH 0/2] ASoC: mediatek: Add support for MT8195 sound card with rt1011 and rt5682
by Trevor Wu 29 Oct '21

29 Oct '21

This series of patches adds support for mt8195 board with mt6359, rt1011 and rt5682. Patches are based on broonie tree "for-next" branch. Trevor Wu (2): ASoC: mediatek: mt8195: add machine driver with mt6359, rt1011 and rt5682 dt-bindings: mediatek: mt8195: add mt8195-mt6359-rt1011-rt5682 document .../sound/mt8195-mt6359-rt1011-rt5682.yaml | 47 + sound/soc/mediatek/Kconfig | 15 + sound/soc/mediatek/mt8195/Makefile | 1 + .../mt8195/mt8195-mt6359-rt1011-rt5682.c | 1140 +++++++++++++++++ 4 files changed, 1203 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/mt8195-mt6359-rt1011-rt5682.yaml create mode 100644 sound/soc/mediatek/mt8195/mt8195-mt6359-rt1011-rt5682.c -- 2.18.0

3 8

[PATCH] ASoC: cs42l42: Add control for audio slow-start switch
by Richard Fitzgerald 29 Oct '21

29 Oct '21

This adds an ALSA control so that the slow-start audio ramp feature can be disabled. This is useful for high-definition audio applications. The register field is unusual in that it is a 3-bit field with only two valid values, 000=off and 111=on. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> --- sound/soc/codecs/cs42l42.c | 32 +++++++++++++++++++++++++++++++- sound/soc/codecs/cs42l42.h | 3 +++ 2 files changed, 34 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index 27a1c4c73074..0d562c274292 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -42,6 +42,7 @@ static const struct reg_default cs42l42_reg_defaults[] = { { CS42L42_SRC_CTL, 0x10 }, { CS42L42_MCLK_CTL, 0x02 }, { CS42L42_SFTRAMP_RATE, 0xA4 }, + { CS42L42_SLOW_START_ENABLE, 0x70 }, { CS42L42_I2C_DEBOUNCE, 0x88 }, { CS42L42_I2C_STRETCH, 0x03 }, { CS42L42_I2C_TIMEOUT, 0xB7 }, @@ -177,6 +178,7 @@ static bool cs42l42_readable_register(struct device *dev, unsigned int reg) case CS42L42_MCLK_STATUS: case CS42L42_MCLK_CTL: case CS42L42_SFTRAMP_RATE: + case CS42L42_SLOW_START_ENABLE: case CS42L42_I2C_DEBOUNCE: case CS42L42_I2C_STRETCH: case CS42L42_I2C_TIMEOUT: @@ -387,6 +389,30 @@ static const struct regmap_config cs42l42_regmap = { static DECLARE_TLV_DB_SCALE(adc_tlv, -9700, 100, true); static DECLARE_TLV_DB_SCALE(mixer_tlv, -6300, 100, true); +static int cs42l42_slow_start_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + u8 val; + + /* all bits of SLOW_START_EN much change together */ + switch (ucontrol->value.integer.value[0]) { + case 0: + val = 0; + break; + case 1: + val = CS42L42_SLOW_START_EN_MASK; + break; + default: + return -EINVAL; + } + + snd_soc_component_update_bits(component, CS42L42_SLOW_START_ENABLE, + CS42L42_SLOW_START_EN_MASK, val); + + return 0; +} + static const char * const cs42l42_hpf_freq_text[] = { "1.86Hz", "120Hz", "235Hz", "466Hz" }; @@ -431,7 +457,11 @@ static const struct snd_kcontrol_new cs42l42_snd_controls[] = { CS42L42_DAC_HPF_EN_SHIFT, true, false), SOC_DOUBLE_R_TLV("Mixer Volume", CS42L42_MIXER_CHA_VOL, CS42L42_MIXER_CHB_VOL, CS42L42_MIXER_CH_VOL_SHIFT, - 0x3f, 1, mixer_tlv) + 0x3f, 1, mixer_tlv), + + SOC_SINGLE_EXT("Slow Start Switch", CS42L42_SLOW_START_ENABLE, + CS42L42_SLOW_START_EN_SHIFT, true, false, + snd_soc_get_volsw, cs42l42_slow_start_put), }; static int cs42l42_hp_adc_ev(struct snd_soc_dapm_widget *w, diff --git a/sound/soc/codecs/cs42l42.h b/sound/soc/codecs/cs42l42.h index f45bcc9a3a62..c8b3267a318b 100644 --- a/sound/soc/codecs/cs42l42.h +++ b/sound/soc/codecs/cs42l42.h @@ -62,6 +62,9 @@ #define CS42L42_INTERNAL_FS_MASK (1 << CS42L42_INTERNAL_FS_SHIFT) #define CS42L42_SFTRAMP_RATE (CS42L42_PAGE_10 + 0x0A) +#define CS42L42_SLOW_START_ENABLE (CS42L42_PAGE_10 + 0x0B) +#define CS42L42_SLOW_START_EN_MASK GENMASK(6, 4) +#define CS42L42_SLOW_START_EN_SHIFT 4 #define CS42L42_I2C_DEBOUNCE (CS42L42_PAGE_10 + 0x0E) #define CS42L42_I2C_STRETCH (CS42L42_PAGE_10 + 0x0F) #define CS42L42_I2C_TIMEOUT (CS42L42_PAGE_10 + 0x10) -- 2.11.0

2 1

AM_PATH_ALSA. error
by GitHub issues - edited 29 Oct '21

29 Oct '21

alsa-project/alsa-utils issue #116 was edited from steeley: Hi, I'm installing on latest raspios buster lite 32bit: (Linux cm432 5.10.63-v7l+ #1459 SMP Wed Oct 6 16:41:57 BST 2021 armv7l GNU/Linux) my build process is: `git clone https://github.com/alsa-project/alsa-utils.git cd alsa-utils/ git checkout v1.1.9 autoreconf -i ./configure --prefix=/usr/local make sudo make install` I have built and installed alsa-libs, and alsa-plugins correctly, but get this error with alsa-utils when I autoreconf: `configure.ac:22: warning: macro 'AM_PATH_ALSA' not found in library configure.ac:22: error: possibly undefined macro: AM_PATH_ALSA If this token and others are legitimate, please use m4_pattern_allow. See the Autoconf documentation. autoreconf: /usr/bin/autoconf failed with exit status: 1` How to resolve? TIA Issue URL : https://github.com/alsa-project/alsa-utils/issues/116 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

AM_PATH_ALSA. error
by GitHub issues - edited 29 Oct '21

29 Oct '21

alsa-project/alsa-utils issue #116 was edited from steeley: Hi, I'm installing on latest raspios buster lite 32bit: (Linux cm432 5.10.63-v7l+ #1459 SMP Wed Oct 6 16:41:57 BST 2021 armv7l GNU/Linux) my build process is: `git clone https://github.com/alsa-project/alsa-utils.git cd alsa-utils/ git checkout v1.1.9 autoreconf -i ./configure --prefix=/usr/local make sudo make install` I have built and installed alsa-libs, and alia-plugins correctly, but get this error with alsa-utils when I autoreconf: `configure.ac:22: warning: macro 'AM_PATH_ALSA' not found in library configure.ac:22: error: possibly undefined macro: AM_PATH_ALSA If this token and others are legitimate, please use m4_pattern_allow. See the Autoconf documentation. autoreconf: /usr/bin/autoconf failed with exit status: 1` How to resolve? TIA Issue URL : https://github.com/alsa-project/alsa-utils/issues/116 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

AM_PATH_ALSA. error
by GitHub issues - opened 29 Oct '21

29 Oct '21

alsa-project/alsa-utils issue #116 was opened from steeley: Hi, I'm installing on latest raspios buster lite 32bit: (Linux cm432 5.10.63-v7l+ #1459 SMP Wed Oct 6 16:41:57 BST 2021 armv7l GNU/Linux) my build process is: `git clone https://github.com/alsa-project/alsa-utils.git cd alsa-utils/ git checkout v1.1.9 autoreconf -i ./configure --prefix=/usr/local make sudo make install` I have built and installed alsa-libs correctly, but get this error with alas-utils when I autoreconf: `configure.ac:22: warning: macro 'AM_PATH_ALSA' not found in library configure.ac:22: error: possibly undefined macro: AM_PATH_ALSA If this token and others are legitimate, please use m4_pattern_allow. See the Autoconf documentation. autoreconf: /usr/bin/autoconf failed with exit status: 1` How to resolve? TIA Issue URL : https://github.com/alsa-project/alsa-utils/issues/116 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH] ASoC: amd: acp: select CONFIG_SND_SOC_ACPI
by Arnd Bergmann 29 Oct '21

29 Oct '21

From: Arnd Bergmann <arnd(a)arndb.de> The acp-platform driver now needs the ACPI helpers: ld.lld: error: undefined symbol: snd_soc_acpi_find_machine >>> referenced by acp-platform.c >>> soc/amd/acp/acp-platform.o:(acp_machine_select) in archive sound/built-in.a ld.lld: error: undefined symbol: snd_soc_acpi_codec_list >>> referenced by acp-renoir.c >>> soc/amd/acp/acp-renoir.o:(snd_soc_acpi_amd_acp_machines) in archive sound/built-in.a Other drivers using this interface, select SND_SOC_ACPI, so do the same thing here. Fixes: e646b51f5dd5 ("ASoC: amd: acp: Add callback for machine driver on ACP") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/amd/acp/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/amd/acp/Kconfig b/sound/soc/amd/acp/Kconfig index e3d5cf993f5f..52a1371f9e61 100644 --- a/sound/soc/amd/acp/Kconfig +++ b/sound/soc/amd/acp/Kconfig @@ -20,6 +20,7 @@ config SND_SOC_AMD_ACP_I2S config SND_SOC_AMD_ACP_PCM tristate + select SND_SOC_ACPI if ACPI config SND_AMD_ASOC_RENOIR tristate "AMD ACP ASOC Renoir Support" -- 2.29.2

2 1

[PATCH] ASoC: es8316: add support for ESSX8336 ACPI _HID
by Bard Liao 29 Oct '21

29 Oct '21

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> The same codec seems to have different personalities. ESSX8316 was used for Baytrail/CherryTrail, ESSX8336 seems to be used for AppoloLake, GeminiLake, JasperLake and TigerLake devices. BugLink: https://github.com/thesofproject/linux/issues/2955 Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>-e Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- sound/soc/codecs/es8316.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c index 5fb02635c140..8f30a3ea8bfe 100644 --- a/sound/soc/codecs/es8316.c +++ b/sound/soc/codecs/es8316.c @@ -840,6 +840,7 @@ MODULE_DEVICE_TABLE(of, es8316_of_match); #ifdef CONFIG_ACPI static const struct acpi_device_id es8316_acpi_match[] = { {"ESSX8316", 0}, + {"ESSX8336", 0}, {}, }; MODULE_DEVICE_TABLE(acpi, es8316_acpi_match); -- 2.17.1

2 1

[PATCH] ASoC: fix unmet dependencies on GPIOLIB for SND_SOC_RT1015P
by Julian Braha 29 Oct '21

29 Oct '21

When SND_SOC_MT8192_MT6359_RT1015_RT5682, SND_SOC_MT8192_MT6359_RT1015_RT5682, SND_SOC_MT8183_DA7219_MAX98357A, or SND_SOC_MT8183_MT6358_TS3A227E_MAX98357A is selected, and GPIOLIB is not selected, Kbuild gives the following warnings, respectively: WARNING: unmet direct dependencies detected for SND_SOC_DMIC Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && GPIOLIB [=n] Selected by [y]: - SND_SOC_MT8192_MT6359_RT1015_RT5682 [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && I2C [=y] && SND_SOC_MT8192 [=y] && MTK_PMIC_WRAP [=y] WARNING: unmet direct dependencies detected for SND_SOC_RT1015P Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && GPIOLIB [=n] Selected by [y]: - SND_SOC_MT8192_MT6359_RT1015_RT5682 [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && I2C [=y] && SND_SOC_MT8192 [=y] && MTK_PMIC_WRAP [=y] WARNING: unmet direct dependencies detected for SND_SOC_RT1015P Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && GPIOLIB [=n] Selected by [y]: - SND_SOC_MT8183_DA7219_MAX98357A [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_SOC_MT8183 [=y] && I2C [=y] WARNING: unmet direct dependencies detected for SND_SOC_RT1015P Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && GPIOLIB [=n] Selected by [y]: - SND_SOC_MT8183_MT6358_TS3A227E_MAX98357A [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && I2C [=y] && SND_SOC_MT8183 [=y] This is because these config options select SND_SOC_RT1015P without selecting or depending on GPIOLIB, despite SND_SOC_RT1015P depending on GPIOLIB. These unmet dependency bugs were detected by Kismet, a static analysis tool for Kconfig. Please advise if this is not the appropriate solution. Signed-off-by: Julian Braha <julianbraha(a)gmail.com> --- sound/soc/mediatek/Kconfig | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/mediatek/Kconfig b/sound/soc/mediatek/Kconfig index 81ad2dcee9eb..d10d6a3bf5eb 100644 --- a/sound/soc/mediatek/Kconfig +++ b/sound/soc/mediatek/Kconfig @@ -120,7 +120,7 @@ config SND_SOC_MT8183 config SND_SOC_MT8183_MT6358_TS3A227E_MAX98357A tristate "ASoC Audio driver for MT8183 with MT6358 TS3A227E MAX98357A RT1015 codec" - depends on I2C + depends on I2C && GPIOLIB depends on SND_SOC_MT8183 select SND_SOC_MT6358 select SND_SOC_MAX98357A @@ -138,7 +138,7 @@ config SND_SOC_MT8183_MT6358_TS3A227E_MAX98357A config SND_SOC_MT8183_DA7219_MAX98357A tristate "ASoC Audio driver for MT8183 with DA7219 MAX98357A RT1015 codec" - depends on SND_SOC_MT8183 && I2C + depends on SND_SOC_MT8183 && I2C && GPIOLIB select SND_SOC_MT6358 select SND_SOC_MAX98357A select SND_SOC_RT1015 @@ -173,7 +173,7 @@ config SND_SOC_MT8192 config SND_SOC_MT8192_MT6359_RT1015_RT5682 tristate "ASoC Audio driver for MT8192 with MT6359 RT1015 RT5682 codec" - depends on I2C + depends on I2C && GPIOLIB depends on SND_SOC_MT8192 && MTK_PMIC_WRAP select SND_SOC_MT6359 select SND_SOC_RT1015 @@ -200,7 +200,7 @@ config SND_SOC_MT8195 config SND_SOC_MT8195_MT6359_RT1019_RT5682 tristate "ASoC Audio driver for MT8195 with MT6359 RT1019 RT5682 codec" - depends on I2C + depends on I2C && GPIOLIB depends on SND_SOC_MT8195 && MTK_PMIC_WRAP select SND_SOC_MT6359 select SND_SOC_RT1015P -- 2.30.2

3 2

[PATCH v2] ASoc: Intel: glk_rt5682_max98357a: support ALC5682I-VS codec
by Brent Lu 29 Oct '21

29 Oct '21

Detect the codec variant in probe function and update DAI link accordingly. Also add an new entry in enumeration table for machine driver enumeration. Signed-off-by: Brent Lu <brent.lu(a)intel.com> --- sound/soc/intel/boards/Kconfig | 1 + sound/soc/intel/boards/glk_rt5682_max98357a.c | 52 +++++++++++++++---- .../intel/common/soc-acpi-intel-glk-match.c | 8 +++ 3 files changed, 52 insertions(+), 9 deletions(-) diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index f693383eb6e3..2dd5ff7e35ce 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -427,6 +427,7 @@ config SND_SOC_INTEL_GLK_RT5682_MAX98357A_MACH depends on MFD_INTEL_LPSS || COMPILE_TEST depends on SND_HDA_CODEC_HDMI && SND_SOC_SOF_HDA_AUDIO_CODEC select SND_SOC_RT5682_I2C + select SND_SOC_RT5682S select SND_SOC_MAX98357A select SND_SOC_DMIC select SND_SOC_HDAC_HDMI diff --git a/sound/soc/intel/boards/glk_rt5682_max98357a.c b/sound/soc/intel/boards/glk_rt5682_max98357a.c index 9d75beec09d1..bad3829e52ca 100644 --- a/sound/soc/intel/boards/glk_rt5682_max98357a.c +++ b/sound/soc/intel/boards/glk_rt5682_max98357a.c @@ -18,14 +18,18 @@ #include <sound/soc.h> #include <sound/soc-acpi.h> #include "../../codecs/rt5682.h" +#include "../../codecs/rt5682s.h" #include "../../codecs/hdac_hdmi.h" #include "hda_dsp_common.h" /* The platform clock outputs 19.2Mhz clock to codec as I2S MCLK */ #define GLK_PLAT_CLK_FREQ 19200000 #define RT5682_PLL_FREQ (48000 * 512) -#define GLK_REALTEK_CODEC_DAI "rt5682-aif1" +#define RT5682_DAI_NAME "rt5682-aif1" +#define RT5682S_DAI_NAME "rt5682s-aif1" #define GLK_MAXIM_CODEC_DAI "HiFi" +#define RT5682_DEV0_NAME "i2c-10EC5682:00" +#define RT5682S_DEV0_NAME "i2c-RTL5682:00" #define MAXIM_DEV0_NAME "MX98357A:00" #define DUAL_CHANNEL 2 #define QUAD_CHANNEL 4 @@ -43,6 +47,7 @@ struct glk_card_private { struct snd_soc_jack geminilake_headset; struct list_head hdmi_pcm_list; bool common_hdmi_codec_drv; + int is_rt5682s; }; enum { @@ -139,9 +144,19 @@ static int geminilake_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component; struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); struct snd_soc_jack *jack; - int ret; + int pll_id, pll_source, clk_id, ret; + + if (ctx->is_rt5682s) { + pll_id = RT5682S_PLL2; + pll_source = RT5682S_PLL_S_MCLK; + clk_id = RT5682S_SCLK_S_PLL2; + } else { + pll_id = RT5682_PLL1; + pll_source = RT5682_PLL1_S_MCLK; + clk_id = RT5682_SCLK_S_PLL1; + } - ret = snd_soc_dai_set_pll(codec_dai, 0, RT5682_PLL1_S_MCLK, + ret = snd_soc_dai_set_pll(codec_dai, pll_id, pll_source, GLK_PLAT_CLK_FREQ, RT5682_PLL_FREQ); if (ret < 0) { dev_err(rtd->dev, "can't set codec pll: %d\n", ret); @@ -149,7 +164,7 @@ static int geminilake_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd) } /* Configure sysclk for codec */ - ret = snd_soc_dai_set_sysclk(codec_dai, RT5682_SCLK_S_PLL1, + ret = snd_soc_dai_set_sysclk(codec_dai, clk_id, RT5682_PLL_FREQ, SND_SOC_CLOCK_IN); if (ret < 0) dev_err(rtd->dev, "snd_soc_dai_set_sysclk err = %d\n", ret); @@ -344,9 +359,12 @@ SND_SOC_DAILINK_DEF(ssp1_codec, SND_SOC_DAILINK_DEF(ssp2_pin, DAILINK_COMP_ARRAY(COMP_CPU("SSP2 Pin"))); -SND_SOC_DAILINK_DEF(ssp2_codec, - DAILINK_COMP_ARRAY(COMP_CODEC("i2c-10EC5682:00", - GLK_REALTEK_CODEC_DAI))); +SND_SOC_DAILINK_DEF(ssp2_codec_5682, + DAILINK_COMP_ARRAY(COMP_CODEC(RT5682_DEV0_NAME, + RT5682_DAI_NAME))); +SND_SOC_DAILINK_DEF(ssp2_codec_5682s, + DAILINK_COMP_ARRAY(COMP_CODEC(RT5682S_DEV0_NAME, + RT5682S_DAI_NAME))); SND_SOC_DAILINK_DEF(dmic_pin, DAILINK_COMP_ARRAY(COMP_CPU("DMIC01 Pin"))); @@ -492,7 +510,7 @@ static struct snd_soc_dai_link geminilake_dais[] = { .ops = &geminilake_rt5682_ops, .dpcm_playback = 1, .dpcm_capture = 1, - SND_SOC_DAILINK_REG(ssp2_pin, ssp2_codec, platform), + SND_SOC_DAILINK_REG(ssp2_pin, ssp2_codec_5682, platform), }, { .name = "dmic01", @@ -592,12 +610,28 @@ static int geminilake_audio_probe(struct platform_device *pdev) struct snd_soc_acpi_mach *mach; const char *platform_name; struct snd_soc_card *card; - int ret; + int ret, i; ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); if (!ctx) return -ENOMEM; + /* Detect the headset codec variant */ + if (acpi_dev_present("RTL5682", NULL, -1)) { + /* ALC5682I-VS is detected */ + ctx->is_rt5682s = 1; + + for (i = 0; i < glk_audio_card_rt5682_m98357a.num_links; i++) { + if (strcmp(geminilake_dais[i].name, "SSP2-Codec")) + continue; + + /* update the dai link to use rt5682s codec */ + geminilake_dais[i].codecs = ssp2_codec_5682s; + geminilake_dais[i].num_codecs = ARRAY_SIZE(ssp2_codec_5682s); + break; + } + } + INIT_LIST_HEAD(&ctx->hdmi_pcm_list); card = &glk_audio_card_rt5682_m98357a; diff --git a/sound/soc/intel/common/soc-acpi-intel-glk-match.c b/sound/soc/intel/common/soc-acpi-intel-glk-match.c index 32fff9389eb3..3900e3dbae30 100644 --- a/sound/soc/intel/common/soc-acpi-intel-glk-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-glk-match.c @@ -40,6 +40,14 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_glk_machines[] = { .sof_fw_filename = "sof-glk.ri", .sof_tplg_filename = "sof-glk-rt5682.tplg", }, + { + .id = "RTL5682", + .drv_name = "glk_rt5682_max98357a", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &glk_codecs, + .sof_fw_filename = "sof-glk.ri", + .sof_tplg_filename = "sof-glk-rt5682.tplg", + }, { .id = "10134242", .drv_name = "glk_cs4242_mx98357a", -- 2.25.1

3 2

[PATCH] ALSA: oxfw: fix functional regression for Mackie Onyx 1640i in v5.14 or later
by Takashi Sakamoto 29 Oct '21

29 Oct '21

A user reports functional regression for Mackie Onyx 1640i that the device generates slow sound with ALSA oxfw driver which supports media clock recovery. Although the device is based on OXFW971 ASIC, it does not transfer isochronous packet with own event frequency as expected. The device seems to adjust event frequency according to events in received isochronous packets in the beginning of packet streaming. This is unknown quirk. This commit fixes the regression to turn the recovery off in driver side. As a result, nominal frequency is used in duplex packet streaming between device and driver. For stability of sampling rate in events of transferred isochronous packet, 4,000 isochronous packets are skipped in the beginning of packet streaming. Reference: https://github.com/takaswie/snd-firewire-improve/issues/38 Fixes: 029ffc429440 ("ALSA: oxfw: perform sequence replay for media clock recovery") Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/oxfw/oxfw-stream.c | 7 ++++++- sound/firewire/oxfw/oxfw.c | 8 ++++++++ sound/firewire/oxfw/oxfw.h | 5 +++++ 3 files changed, 19 insertions(+), 1 deletion(-) diff --git a/sound/firewire/oxfw/oxfw-stream.c b/sound/firewire/oxfw/oxfw-stream.c index fff18b5d4e05..f4a702def397 100644 --- a/sound/firewire/oxfw/oxfw-stream.c +++ b/sound/firewire/oxfw/oxfw-stream.c @@ -9,7 +9,7 @@ #include <linux/delay.h> #define AVC_GENERIC_FRAME_MAXIMUM_BYTES 512 -#define READY_TIMEOUT_MS 200 +#define READY_TIMEOUT_MS 600 /* * According to datasheet of Oxford Semiconductor: @@ -367,6 +367,11 @@ int snd_oxfw_stream_start_duplex(struct snd_oxfw *oxfw) // Just after changing sampling transfer frequency, many cycles are // skipped for packet transmission. tx_init_skip_cycles = 400; + } else if (oxfw->quirks & SND_OXFW_QUIRK_VOLUNTARY_RECOVERY) { + // It takes a bit time for target device to adjust event frequency + // according to nominal event frequency in isochronous packets from + // ALSA oxfw driver. + tx_init_skip_cycles = 4000; } else { replay_seq = true; } diff --git a/sound/firewire/oxfw/oxfw.c b/sound/firewire/oxfw/oxfw.c index daf731364695..b496f87841ae 100644 --- a/sound/firewire/oxfw/oxfw.c +++ b/sound/firewire/oxfw/oxfw.c @@ -25,6 +25,7 @@ #define MODEL_SATELLITE 0x00200f #define MODEL_SCS1M 0x001000 #define MODEL_DUET_FW 0x01dddd +#define MODEL_ONYX_1640I 0x001640 #define SPECIFIER_1394TA 0x00a02d #define VERSION_AVC 0x010001 @@ -192,6 +193,13 @@ static int detect_quirks(struct snd_oxfw *oxfw, const struct ieee1394_device_id // OXFW971-based models may transfer events by blocking method. if (!(oxfw->quirks & SND_OXFW_QUIRK_JUMBO_PAYLOAD)) oxfw->quirks |= SND_OXFW_QUIRK_BLOCKING_TRANSMISSION; + + if (model == MODEL_ONYX_1640I) { + //Unless receiving packets without NOINFO packet, the device transfers + //mostly half of events in packets than expected. + oxfw->quirks |= SND_OXFW_QUIRK_IGNORE_NO_INFO_PACKET | + SND_OXFW_QUIRK_VOLUNTARY_RECOVERY; + } } return 0; diff --git a/sound/firewire/oxfw/oxfw.h b/sound/firewire/oxfw/oxfw.h index c13034f6c2ca..d728e451a25c 100644 --- a/sound/firewire/oxfw/oxfw.h +++ b/sound/firewire/oxfw/oxfw.h @@ -47,6 +47,11 @@ enum snd_oxfw_quirk { // the device to process audio data even if the value is invalid in a point of // IEC 61883-1/6. SND_OXFW_QUIRK_IGNORE_NO_INFO_PACKET = 0x10, + // Loud Technologies Mackie Onyx 1640i seems to configure OXFW971 ASIC so that it decides + // event frequency according to events in received isochronous packets. The device looks to + // performs media clock recovery voluntarily. In the recovery, the packets with NO_INFO + // are ignored, thus driver should transfer packets with timestamp. + SND_OXFW_QUIRK_VOLUNTARY_RECOVERY = 0x20, }; /* This is an arbitrary number for convinience. */ -- 2.30.2

2 4

[PATCH] ALSA: firewire-motu: remove TODO for interaction with userspace about control message
by Takashi Sakamoto 29 Oct '21

29 Oct '21

Now UAPI of ALSA firewire stack got enough functions to interact with userspace for protocol of MOTU FireWire series. Let's remove the relevant TODO. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/motu/amdtp-motu.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/firewire/motu/amdtp-motu.c b/sound/firewire/motu/amdtp-motu.c index 3ea91e281147..2fb52f481d12 100644 --- a/sound/firewire/motu/amdtp-motu.c +++ b/sound/firewire/motu/amdtp-motu.c @@ -424,8 +424,6 @@ static unsigned int process_it_ctx_payloads(struct amdtp_stream *s, if (p->midi_ports) write_midi_messages(s, buf, data_blocks); - // TODO: how to interact control messages between userspace? - write_sph(p->cache, buf, data_blocks, s->data_block_quadlets); } -- 2.30.2

2 1

[PATCH 0/3] Add pin control support for lpass sc7280
by Srinivasa Rao Mandadapu 29 Oct '21

29 Oct '21

This patch series is to make lpass variant independent pin control functions common and to add lpass sc7280 pincontrol support. It also includes dt-bindings for lpass sc7280 lpi compatible. Srinivasa Rao Mandadapu (3): pinctrl: qcom: Update lpass variant independent functions as generic dt-bindings: pinctrl: qcom: Add sc7280 lpass lpi pinctrl compatible pinctrl: qcom: Add SC7280 lpass pin configuration .../bindings/pinctrl/qcom,lpass-lpi-pinctrl.yaml | 4 +- drivers/pinctrl/qcom/pinctrl-lpass-lpi.c | 57 +++++++++++++++++++--- 2 files changed, 52 insertions(+), 9 deletions(-) -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

3 11

[PATCH RESEND] ASoC: fix unmet dependencies on GPIOLIB for SND_SOC_DMIC
by Julian Braha 28 Oct '21

28 Oct '21

When SND_SOC_AMD_RENOIR_MACH or SND_SOC_AMD_RV_RT5682_MACH are selected, and GPIOLIB is not selected, Kbuild gives the following warnings, respectively: WARNING: unmet direct dependencies detected for SND_SOC_DMIC Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && GPIOLIB [=n] Selected by [y]: - SND_SOC_AMD_RENOIR_MACH [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_SOC_AMD_RENOIR [=y] and WARNING: unmet direct dependencies detected for SND_SOC_MAX98357A Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && GPIOLIB [=n] Selected by [y]: - SND_SOC_AMD_RV_RT5682_MACH [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_SOC_AMD_ACP3x [=y] && I2C [=y] && CROS_EC [=y] This is because SND_SOC_DMIC and SND_SOC_MAX98357A are selected by SND_SOC_AMD_RV_RT5682_MACH and SND_SOC_AMD_RENOIR_MACH, respectively. However, neither of the selectors depend on or select GPIOLIB, despite their selectees depending on GPIOLIB. These unmet dependency bugs were detected by Kismet, a static analysis tool for Kconfig. Please advise if this is not the appropriate solution. Signed-off-by: Julian Braha <julianbraha(a)gmail.com> --- sound/soc/amd/Kconfig | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/amd/Kconfig b/sound/soc/amd/Kconfig index 56e7c079deac..2c6af3f8f296 100644 --- a/sound/soc/amd/Kconfig +++ b/sound/soc/amd/Kconfig @@ -38,7 +38,7 @@ config SND_SOC_AMD_RV_RT5682_MACH select I2C_CROS_EC_TUNNEL select SND_SOC_RT1015 select SND_SOC_RT1015P - depends on SND_SOC_AMD_ACP3x && I2C && CROS_EC + depends on SND_SOC_AMD_ACP3x && I2C && CROS_EC && GPIOLIB help This option enables machine driver for RT5682 and MAX9835. @@ -51,7 +51,7 @@ config SND_SOC_AMD_RENOIR config SND_SOC_AMD_RENOIR_MACH tristate "AMD Renoir support for DMIC" select SND_SOC_DMIC - depends on SND_SOC_AMD_RENOIR + depends on SND_SOC_AMD_RENOIR && GPIOLIB help This option enables machine driver for DMIC -- 2.30.2

3 5

[PATCH] ASoC: Intel: add sof-nau8825 machine driver
by Mac Chiang 28 Oct '21

28 Oct '21

From: David Lin <CTLIN0(a)nuvoton.com> The machine driver is a generic machine driver for SOF with nau8825 codec w or w/o speaker additionally. Depending on the SOC HDMI, DMIC, Bluetooth offload support are added dynamically. Only add information related to SOF since the machine driver was only tested with SOF. There are currently 4 i2s machine variants of ADL. This supports the headphone NUA8825(SSP0) alone or with smart or dumb speakers. Board 2,3,4 use SSP2 for Bluetooth offload support except board 1. Board 1 : NAU8825 + RT1019P(SSP2) Board 2 : NAU8825 + MAX98373(SSP1) Board 3 : NAU8825 + MAX98360A(SSP1) Board 4 : NAU8825 Signed-off-by: David Lin <CTLIN0(a)nuvoton.com> Co-developed-by: Mac Chiang <mac.chiang(a)intel.com> Signed-off-by: Mac Chiang <mac.chiang(a)intel.com> --- sound/soc/intel/boards/Kconfig | 19 + sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/sof_nau8825.c | 683 ++++++++++++++++++ .../intel/common/soc-acpi-intel-adl-match.c | 35 + 4 files changed, 739 insertions(+) create mode 100644 sound/soc/intel/boards/sof_nau8825.c diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 5a225a4ce604..0944934119f5 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -526,6 +526,25 @@ config SND_SOC_INTEL_SOF_ES8336_MACH Say Y if you have such a device. If unsure select "N". +config SND_SOC_INTEL_SOF_NAU8825_MACH + tristate "SOF with nau8825 codec in I2S Mode" + depends on I2C && ACPI && GPIOLIB + depends on ((SND_HDA_CODEC_HDMI && SND_SOC_SOF_HDA_AUDIO_CODEC) &&\ + (MFD_INTEL_LPSS || COMPILE_TEST)) + select SND_SOC_NAU8825 + select SND_SOC_RT1019 + select SND_SOC_MAX98373_I2C + select SND_SOC_MAX98357A + select SND_SOC_DMIC + select SND_SOC_HDAC_HDMI + select SND_SOC_INTEL_HDA_DSP_COMMON + select SND_SOC_INTEL_SOF_MAXIM_COMMON + help + This adds support for ASoC machine driver for SOF platforms + with nau8825 codec. + Say Y if you have such a device. + If unsure select "N". + endif ## SND_SOC_SOF_HDA_LINK || SND_SOC_SOF_BAYTRAIL if (SND_SOC_SOF_COMETLAKE && SND_SOC_SOF_HDA_LINK) diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index 9ee8ed864f5d..3ea273d27168 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -22,6 +22,7 @@ snd-soc-sst-byt-cht-nocodec-objs := bytcht_nocodec.o snd-soc-sof_rt5682-objs := sof_rt5682.o sof_realtek_common.o snd-soc-sof_cs42l42-objs := sof_cs42l42.o snd-soc-sof_es8336-objs := sof_es8336.o +snd-soc-sof_nau8825-objs := sof_nau8825.o sof_realtek_common.o snd-soc-cml_rt1011_rt5682-objs := cml_rt1011_rt5682.o snd-soc-kbl_da7219_max98357a-objs := kbl_da7219_max98357a.o snd-soc-kbl_da7219_max98927-objs := kbl_da7219_max98927.o @@ -44,6 +45,7 @@ snd-soc-sof-sdw-objs += sof_sdw.o \ obj-$(CONFIG_SND_SOC_INTEL_SOF_RT5682_MACH) += snd-soc-sof_rt5682.o obj-$(CONFIG_SND_SOC_INTEL_SOF_CS42L42_MACH) += snd-soc-sof_cs42l42.o obj-$(CONFIG_SND_SOC_INTEL_SOF_ES8336_MACH) += snd-soc-sof_es8336.o +obj-$(CONFIG_SND_SOC_INTEL_SOF_NAU8825_MACH) += snd-soc-sof_nau8825.o obj-$(CONFIG_SND_SOC_INTEL_HASWELL_MACH) += snd-soc-sst-haswell.o obj-$(CONFIG_SND_SOC_INTEL_BXT_DA7219_MAX98357A_COMMON) += snd-soc-sst-bxt-da7219_max98357a.o obj-$(CONFIG_SND_SOC_INTEL_BXT_RT298_MACH) += snd-soc-sst-bxt-rt298.o diff --git a/sound/soc/intel/boards/sof_nau8825.c b/sound/soc/intel/boards/sof_nau8825.c new file mode 100644 index 000000000000..04f676922018 --- /dev/null +++ b/sound/soc/intel/boards/sof_nau8825.c @@ -0,0 +1,683 @@ +// SPDX-License-Identifier: GPL-2.0-only +// Copyright(c) 2021 Intel Corporation. +// Copyright(c) 2021 Nuvoton Corporation. + +/* + * Intel SOF Machine Driver with Nuvoton headphone codec NAU8825 + * and speaker codec RT1019P MAX98360a or MAX98373 + */ +#include <linux/i2c.h> +#include <linux/input.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/dmi.h> +#include <sound/core.h> +#include <sound/jack.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/sof.h> +#include <sound/soc-acpi.h> +#include "../../codecs/nau8825.h" +#include "../../codecs/hdac_hdmi.h" +#include "../common/soc-intel-quirks.h" +#include "hda_dsp_common.h" +#include "sof_realtek_common.h" +#include "sof_maxim_common.h" + +#define NAME_SIZE 32 + +#define SOF_NAU8825_SSP_CODEC(quirk) ((quirk) & GENMASK(2, 0)) +#define SOF_NAU8825_SSP_CODEC_MASK (GENMASK(2, 0)) +#define SOF_SPEAKER_AMP_PRESENT BIT(3) +#define SOF_NAU8825_SSP_AMP_SHIFT 4 +#define SOF_NAU8825_SSP_AMP_MASK (GENMASK(6, 4)) +#define SOF_NAU8825_SSP_AMP(quirk) \ + (((quirk) << SOF_NAU8825_SSP_AMP_SHIFT) & SOF_NAU8825_SSP_AMP_MASK) +#define SOF_NAU8825_NUM_HDMIDEV_SHIFT 7 +#define SOF_NAU8825_NUM_HDMIDEV_MASK (GENMASK(9, 7)) +#define SOF_NAU8825_NUM_HDMIDEV(quirk) \ + (((quirk) << SOF_NAU8825_NUM_HDMIDEV_SHIFT) & SOF_NAU8825_NUM_HDMIDEV_MASK) + +/* BT audio offload: reserve 3 bits for future */ +#define SOF_BT_OFFLOAD_SSP_SHIFT 10 +#define SOF_BT_OFFLOAD_SSP_MASK (GENMASK(12, 10)) +#define SOF_BT_OFFLOAD_SSP(quirk) \ + (((quirk) << SOF_BT_OFFLOAD_SSP_SHIFT) & SOF_BT_OFFLOAD_SSP_MASK) +#define SOF_SSP_BT_OFFLOAD_PRESENT BIT(13) +#define SOF_RT1019P_SPEAKER_AMP_PRESENT BIT(14) +#define SOF_MAX98373_SPEAKER_AMP_PRESENT BIT(15) +#define SOF_MAX98360A_SPEAKER_AMP_PRESENT BIT(16) + +static unsigned long sof_nau8825_quirk = SOF_NAU8825_SSP_CODEC(0); + +static struct snd_soc_jack sof_hdmi[3]; + +struct sof_hdmi_pcm { + struct list_head head; + struct snd_soc_dai *codec_dai; + int device; +}; + +struct sof_card_private { + struct clk *mclk; + struct snd_soc_jack sof_headset; + struct list_head hdmi_pcm_list; + bool common_hdmi_codec_drv; +}; + +static int sof_hdmi_init(struct snd_soc_pcm_runtime *rtd) +{ + struct sof_card_private *ctx = snd_soc_card_get_drvdata(rtd->card); + struct snd_soc_dai *dai = asoc_rtd_to_codec(rtd, 0); + struct sof_hdmi_pcm *pcm; + + pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); + if (!pcm) + return -ENOMEM; + + /* dai_link id is 1:1 mapped to the PCM device */ + pcm->device = rtd->dai_link->id; + pcm->codec_dai = dai; + + list_add_tail(&pcm->head, &ctx->hdmi_pcm_list); + + return 0; +} + +static int sof_nau8825_codec_init(struct snd_soc_pcm_runtime *rtd) +{ + struct sof_card_private *ctx = snd_soc_card_get_drvdata(rtd->card); + struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component; + + struct snd_soc_jack *jack; + int ret; + + /* + * Headset buttons map to the google Reference headset. + * These can be configured by userspace. + */ + ret = snd_soc_card_jack_new(rtd->card, "Headset Jack", + SND_JACK_HEADSET | SND_JACK_BTN_0 | + SND_JACK_BTN_1 | SND_JACK_BTN_2 | + SND_JACK_BTN_3, + &ctx->sof_headset, NULL, 0); + if (ret) { + dev_err(rtd->dev, "Headset Jack creation failed: %d\n", ret); + return ret; + } + + jack = &ctx->sof_headset; + + snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); + snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOICECOMMAND); + snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEUP); + snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN); + ret = snd_soc_component_set_jack(component, jack, NULL); + + if (ret) { + dev_err(rtd->dev, "Headset Jack call-back failed: %d\n", ret); + return ret; + } + + return ret; +}; + +static void sof_nau8825_codec_exit(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component; + + snd_soc_component_set_jack(component, NULL, NULL); +} + +static int sof_nau8825_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 = asoc_rtd_to_codec(rtd, 0); + int clk_freq, ret; + + clk_freq = sof_dai_get_bclk(rtd); /* BCLK freq */ + + if (clk_freq <= 0) { + dev_err(rtd->dev, "get bclk freq failed: %d\n", clk_freq); + return -EINVAL; + } + + /* Configure clock for codec */ + ret = snd_soc_dai_set_sysclk(codec_dai, NAU8825_CLK_FLL_BLK, 0, + SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(codec_dai->dev, "can't set BCLK clock %d\n", ret); + return ret; + } + + /* Configure pll for codec */ + ret = snd_soc_dai_set_pll(codec_dai, 0, 0, clk_freq, + params_rate(params) * 256); + if (ret < 0) { + dev_err(codec_dai->dev, "can't set BCLK: %d\n", ret); + return ret; + } + + return ret; +} + +static struct snd_soc_ops sof_nau8825_ops = { + .hw_params = sof_nau8825_hw_params, +}; + +static struct snd_soc_dai_link_component platform_component[] = { + { + /* name might be overridden during probe */ + .name = "0000:00:1f.3" + } +}; + +static int sof_card_late_probe(struct snd_soc_card *card) +{ + struct sof_card_private *ctx = snd_soc_card_get_drvdata(card); + struct snd_soc_component *component = NULL; + struct snd_soc_dapm_context *dapm = &card->dapm; + char jack_name[NAME_SIZE]; + struct sof_hdmi_pcm *pcm; + int err; + int i = 0; + + if (list_empty(&ctx->hdmi_pcm_list)) + return -EINVAL; + + if (ctx->common_hdmi_codec_drv) { + pcm = list_first_entry(&ctx->hdmi_pcm_list, struct sof_hdmi_pcm, + head); + component = pcm->codec_dai->component; + return hda_dsp_hdmi_build_controls(card, component); + } + + list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { + component = pcm->codec_dai->component; + snprintf(jack_name, sizeof(jack_name), + "HDMI/DP, pcm=%d Jack", pcm->device); + err = snd_soc_card_jack_new(card, jack_name, + SND_JACK_AVOUT, &sof_hdmi[i], + NULL, 0); + + if (err) + return err; + + err = hdac_hdmi_jack_init(pcm->codec_dai, pcm->device, + &sof_hdmi[i]); + if (err < 0) + return err; + + i++; + } + + if (sof_nau8825_quirk & SOF_MAX98373_SPEAKER_AMP_PRESENT) { + /* Disable Left and Right Spk pin after boot */ + snd_soc_dapm_disable_pin(dapm, "Left Spk"); + snd_soc_dapm_disable_pin(dapm, "Right Spk"); + err = snd_soc_dapm_sync(dapm); + if (err < 0) + return err; + } + return hdac_hdmi_jack_port_init(component, &card->dapm); +} + +static const struct snd_kcontrol_new sof_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphone Jack"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), + SOC_DAPM_PIN_SWITCH("Left Spk"), + SOC_DAPM_PIN_SWITCH("Right Spk"), +}; + +static const struct snd_kcontrol_new speaker_controls[] = { + SOC_DAPM_PIN_SWITCH("Spk"), +}; + +static const struct snd_soc_dapm_widget sof_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_SPK("Left Spk", NULL), + SND_SOC_DAPM_SPK("Right Spk", NULL), +}; + +static const struct snd_soc_dapm_widget speaker_widgets[] = { + SND_SOC_DAPM_SPK("Spk", NULL), +}; + +static const struct snd_soc_dapm_widget dmic_widgets[] = { + SND_SOC_DAPM_MIC("SoC DMIC", NULL), +}; + +static const struct snd_soc_dapm_route sof_map[] = { + /* HP jack connectors - unknown if we have jack detection */ + { "Headphone Jack", NULL, "HPOL" }, + { "Headphone Jack", NULL, "HPOR" }, + + /* other jacks */ + { "MIC", NULL, "Headset Mic" }, +}; + +static const struct snd_soc_dapm_route speaker_map[] = { + /* speaker */ + { "Spk", NULL, "Speaker" }, +}; + +static const struct snd_soc_dapm_route dmic_map[] = { + /* digital mics */ + {"DMic", NULL, "SoC DMIC"}, +}; + +static int speaker_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, speaker_widgets, + ARRAY_SIZE(speaker_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, speaker_controls, + ARRAY_SIZE(speaker_controls)); + 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, speaker_map, + ARRAY_SIZE(speaker_map)); + + if (ret) + dev_err(rtd->dev, "Speaker map addition failed: %d\n", ret); + return ret; +} + +static int dmic_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_card *card = rtd->card; + int ret; + + ret = snd_soc_dapm_new_controls(&card->dapm, dmic_widgets, + ARRAY_SIZE(dmic_widgets)); + if (ret) { + dev_err(card->dev, "DMic widget addition failed: %d\n", ret); + /* Don't need to add routes if widget addition failed */ + return ret; + } + + ret = snd_soc_dapm_add_routes(&card->dapm, dmic_map, + ARRAY_SIZE(dmic_map)); + + if (ret) + dev_err(card->dev, "DMic map addition failed: %d\n", ret); + + return ret; +} + +/* sof audio machine driver for nau8825 codec */ +static struct snd_soc_card sof_audio_card_nau8825 = { + .name = "nau8825", /* the sof- prefix is added by the core */ + .owner = THIS_MODULE, + .controls = sof_controls, + .num_controls = ARRAY_SIZE(sof_controls), + .dapm_widgets = sof_widgets, + .num_dapm_widgets = ARRAY_SIZE(sof_widgets), + .dapm_routes = sof_map, + .num_dapm_routes = ARRAY_SIZE(sof_map), + .fully_routed = true, + .late_probe = sof_card_late_probe, +}; + +static struct snd_soc_dai_link_component nau8825_component[] = { + { + .name = "i2c-10508825:00", + .dai_name = "nau8825-hifi", + } +}; + +static struct snd_soc_dai_link_component dmic_component[] = { + { + .name = "dmic-codec", + .dai_name = "dmic-hifi", + } +}; + +static struct snd_soc_dai_link_component rt1019p_component[] = { + { + .name = "RTL1019:00", + .dai_name = "HiFi", + } +}; + +static struct snd_soc_dai_link_component dummy_component[] = { + { + .name = "snd-soc-dummy", + .dai_name = "snd-soc-dummy-dai", + } +}; + +static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, + int ssp_codec, + int ssp_amp, + int dmic_be_num, + int hdmi_num) +{ + struct snd_soc_dai_link_component *idisp_components; + struct snd_soc_dai_link_component *cpus; + struct snd_soc_dai_link *links; + int i, id = 0; + + links = devm_kzalloc(dev, sizeof(struct snd_soc_dai_link) * + sof_audio_card_nau8825.num_links, GFP_KERNEL); + cpus = devm_kzalloc(dev, sizeof(struct snd_soc_dai_link_component) * + sof_audio_card_nau8825.num_links, GFP_KERNEL); + if (!links || !cpus) + goto devm_err; + + /* codec SSP */ + links[id].name = devm_kasprintf(dev, GFP_KERNEL, + "SSP%d-Codec", ssp_codec); + if (!links[id].name) + goto devm_err; + + links[id].id = id; + links[id].codecs = nau8825_component; + links[id].num_codecs = ARRAY_SIZE(nau8825_component); + links[id].platforms = platform_component; + links[id].num_platforms = ARRAY_SIZE(platform_component); + links[id].init = sof_nau8825_codec_init; + links[id].exit = sof_nau8825_codec_exit; + links[id].ops = &sof_nau8825_ops; + links[id].dpcm_playback = 1; + links[id].dpcm_capture = 1; + links[id].no_pcm = 1; + links[id].cpus = &cpus[id]; + links[id].num_cpus = 1; + + links[id].cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, + "SSP%d Pin", + ssp_codec); + if (!links[id].cpus->dai_name) + goto devm_err; + + id++; + + /* dmic */ + if (dmic_be_num > 0) { + /* at least we have dmic01 */ + links[id].name = "dmic01"; + links[id].cpus = &cpus[id]; + links[id].cpus->dai_name = "DMIC01 Pin"; + links[id].init = dmic_init; + if (dmic_be_num > 1) { + /* set up 2 BE links at most */ + links[id + 1].name = "dmic16k"; + links[id + 1].cpus = &cpus[id + 1]; + links[id + 1].cpus->dai_name = "DMIC16k Pin"; + dmic_be_num = 2; + } + } + + for (i = 0; i < dmic_be_num; i++) { + links[id].id = id; + links[id].num_cpus = 1; + links[id].codecs = dmic_component; + links[id].num_codecs = ARRAY_SIZE(dmic_component); + links[id].platforms = platform_component; + links[id].num_platforms = ARRAY_SIZE(platform_component); + links[id].ignore_suspend = 1; + links[id].dpcm_capture = 1; + links[id].no_pcm = 1; + id++; + } + + /* HDMI */ + if (hdmi_num > 0) { + idisp_components = devm_kzalloc(dev, + sizeof(struct snd_soc_dai_link_component) * + hdmi_num, GFP_KERNEL); + if (!idisp_components) + goto devm_err; + } + for (i = 1; i <= hdmi_num; i++) { + links[id].name = devm_kasprintf(dev, GFP_KERNEL, + "iDisp%d", i); + if (!links[id].name) + goto devm_err; + + links[id].id = id; + links[id].cpus = &cpus[id]; + links[id].num_cpus = 1; + links[id].cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, + "iDisp%d Pin", i); + if (!links[id].cpus->dai_name) + goto devm_err; + + idisp_components[i - 1].name = "ehdaudio0D2"; + idisp_components[i - 1].dai_name = devm_kasprintf(dev, + GFP_KERNEL, + "intel-hdmi-hifi%d", + i); + if (!idisp_components[i - 1].dai_name) + goto devm_err; + + links[id].codecs = &idisp_components[i - 1]; + links[id].num_codecs = 1; + links[id].platforms = platform_component; + links[id].num_platforms = ARRAY_SIZE(platform_component); + links[id].init = sof_hdmi_init; + links[id].dpcm_playback = 1; + links[id].no_pcm = 1; + id++; + } + + /* speaker amp */ + if (sof_nau8825_quirk & SOF_SPEAKER_AMP_PRESENT) { + links[id].name = devm_kasprintf(dev, GFP_KERNEL, + "SSP%d-Codec", ssp_amp); + if (!links[id].name) + goto devm_err; + + links[id].id = id; + if (sof_nau8825_quirk & SOF_RT1019P_SPEAKER_AMP_PRESENT) { + links[id].codecs = rt1019p_component; + links[id].num_codecs = ARRAY_SIZE(rt1019p_component); + links[id].init = speaker_codec_init; + } else if (sof_nau8825_quirk & + SOF_MAX98373_SPEAKER_AMP_PRESENT) { + links[id].codecs = max_98373_components; + links[id].num_codecs = ARRAY_SIZE(max_98373_components); + links[id].init = max_98373_spk_codec_init; + links[id].ops = &max_98373_ops; + /* feedback stream */ + links[id].dpcm_capture = 1; + } else if (sof_nau8825_quirk & + SOF_MAX98360A_SPEAKER_AMP_PRESENT) { + max_98360a_dai_link(&links[id]); + } else { + max_98360a_dai_link(&links[id]); + } + + links[id].platforms = platform_component; + links[id].num_platforms = ARRAY_SIZE(platform_component); + links[id].dpcm_playback = 1; + links[id].no_pcm = 1; + links[id].cpus = &cpus[id]; + links[id].num_cpus = 1; + links[id].cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, + "SSP%d Pin", + ssp_amp); + if (!links[id].cpus->dai_name) + goto devm_err; + id++; + } + + /* BT audio offload */ + if (sof_nau8825_quirk & SOF_SSP_BT_OFFLOAD_PRESENT) { + int port = (sof_nau8825_quirk & SOF_BT_OFFLOAD_SSP_MASK) >> + SOF_BT_OFFLOAD_SSP_SHIFT; + + links[id].id = id; + links[id].cpus = &cpus[id]; + links[id].cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, + "SSP%d Pin", port); + if (!links[id].cpus->dai_name) + goto devm_err; + links[id].name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d-BT", port); + if (!links[id].name) + goto devm_err; + links[id].codecs = dummy_component; + links[id].num_codecs = ARRAY_SIZE(dummy_component); + links[id].platforms = platform_component; + links[id].num_platforms = ARRAY_SIZE(platform_component); + links[id].dpcm_playback = 1; + links[id].dpcm_capture = 1; + links[id].no_pcm = 1; + links[id].num_cpus = 1; + } + + return links; +devm_err: + return NULL; +} + +static int sof_audio_probe(struct platform_device *pdev) +{ + struct snd_soc_dai_link *dai_links; + struct snd_soc_acpi_mach *mach; + struct sof_card_private *ctx; + int dmic_be_num, hdmi_num; + int ret, ssp_amp, ssp_codec; + + ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); + if (!ctx) + return -ENOMEM; + + if (pdev->id_entry && pdev->id_entry->driver_data) + sof_nau8825_quirk = (unsigned long)pdev->id_entry->driver_data; + + mach = pdev->dev.platform_data; + + /* A speaker amp might not be present when the quirk claims one is. + * Detect this via whether the machine driver match includes quirk_data. + */ + if ((sof_nau8825_quirk & SOF_SPEAKER_AMP_PRESENT) && !mach->quirk_data) + sof_nau8825_quirk &= ~SOF_SPEAKER_AMP_PRESENT; + + dev_dbg(&pdev->dev, "sof_nau8825_quirk = %lx\n", sof_nau8825_quirk); + + /* default number of DMIC DAI's */ + dmic_be_num = 2; + hdmi_num = (sof_nau8825_quirk & SOF_NAU8825_NUM_HDMIDEV_MASK) >> + SOF_NAU8825_NUM_HDMIDEV_SHIFT; + /* default number of HDMI DAI's */ + if (!hdmi_num) + hdmi_num = 3; + + ssp_amp = (sof_nau8825_quirk & SOF_NAU8825_SSP_AMP_MASK) >> + SOF_NAU8825_SSP_AMP_SHIFT; + + ssp_codec = sof_nau8825_quirk & SOF_NAU8825_SSP_CODEC_MASK; + + /* compute number of dai links */ + sof_audio_card_nau8825.num_links = 1 + dmic_be_num + hdmi_num; + + if (sof_nau8825_quirk & SOF_SPEAKER_AMP_PRESENT) + sof_audio_card_nau8825.num_links++; + + if (sof_nau8825_quirk & SOF_MAX98373_SPEAKER_AMP_PRESENT) + max_98373_set_codec_conf(&sof_audio_card_nau8825); + + if (sof_nau8825_quirk & SOF_SSP_BT_OFFLOAD_PRESENT) + sof_audio_card_nau8825.num_links++; + + dai_links = sof_card_dai_links_create(&pdev->dev, ssp_codec, ssp_amp, + dmic_be_num, hdmi_num); + if (!dai_links) + return -ENOMEM; + + sof_audio_card_nau8825.dai_link = dai_links; + + INIT_LIST_HEAD(&ctx->hdmi_pcm_list); + + sof_audio_card_nau8825.dev = &pdev->dev; + + /* set platform name for each dailink */ + ret = snd_soc_fixup_dai_links_platform_name(&sof_audio_card_nau8825, + mach->mach_params.platform); + if (ret) + return ret; + + ctx->common_hdmi_codec_drv = mach->mach_params.common_hdmi_codec_drv; + + snd_soc_card_set_drvdata(&sof_audio_card_nau8825, ctx); + + return devm_snd_soc_register_card(&pdev->dev, + &sof_audio_card_nau8825); +} + +static const struct platform_device_id board_ids[] = { + { + .name = "sof_nau8825", + .driver_data = (kernel_ulong_t)(SOF_NAU8825_SSP_CODEC(0) | + SOF_NAU8825_NUM_HDMIDEV(4) | + SOF_BT_OFFLOAD_SSP(2) | + SOF_SSP_BT_OFFLOAD_PRESENT), + + }, + { + .name = "adl_rt1019p_nau8825", + .driver_data = (kernel_ulong_t)(SOF_NAU8825_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_RT1019P_SPEAKER_AMP_PRESENT | + SOF_NAU8825_SSP_AMP(2) | + SOF_NAU8825_NUM_HDMIDEV(4)), + }, + { + .name = "adl_max98373_nau8825", + .driver_data = (kernel_ulong_t)(SOF_NAU8825_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_MAX98373_SPEAKER_AMP_PRESENT | + SOF_NAU8825_SSP_AMP(1) | + SOF_NAU8825_NUM_HDMIDEV(4) | + SOF_BT_OFFLOAD_SSP(2) | + SOF_SSP_BT_OFFLOAD_PRESENT), + }, + { + /* The limitation of length of char array, shorten the name */ + .name = "adl_mx98360a_nau8825", + .driver_data = (kernel_ulong_t)(SOF_NAU8825_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_MAX98360A_SPEAKER_AMP_PRESENT | + SOF_NAU8825_SSP_AMP(1) | + SOF_NAU8825_NUM_HDMIDEV(4) | + SOF_BT_OFFLOAD_SSP(2) | + SOF_SSP_BT_OFFLOAD_PRESENT), + + }, + { } +}; +MODULE_DEVICE_TABLE(platform, board_ids); + +static struct platform_driver sof_audio = { + .probe = sof_audio_probe, + .driver = { + .name = "sof_nau8825", + .pm = &snd_soc_pm_ops, + }, + .id_table = board_ids, +}; +module_platform_driver(sof_audio) + +/* Module information */ +MODULE_DESCRIPTION("SOF Audio Machine driver for NAU8825"); +MODULE_AUTHOR("David Lin <ctlin0(a)nuvoton.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 f5b21a95d222..98dc41b682cf 100644 --- a/sound/soc/intel/common/soc-acpi-intel-adl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-adl-match.c @@ -285,6 +285,11 @@ static const struct snd_soc_acpi_codecs adl_max98360a_amp = { .codecs = {"MX98360A"} }; +static const struct snd_soc_acpi_codecs adl_rt1019p_amp = { + .num_codecs = 1, + .codecs = {"RTL1019"} +}; + struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = { { .id = "10EC5682", @@ -310,6 +315,36 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = { .sof_fw_filename = "sof-adl.ri", .sof_tplg_filename = "sof-adl-max98360a-rt5682.tplg", }, + { + .id = "10508825", + .drv_name = "adl_rt1019p_nau8825", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &adl_rt1019p_amp, + .sof_fw_filename = "sof-adl.ri", + .sof_tplg_filename = "sof-adl-rt1019-nau8825.tplg", + }, + { + .id = "10508825", + .drv_name = "adl_max98373_nau8825", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &adl_max98373_amp, + .sof_fw_filename = "sof-adl.ri", + .sof_tplg_filename = "sof-adl-max98373-nau8825.tplg", + }, + { + .id = "10508825", + .drv_name = "adl_mx98360a_nau8825", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &adl_max98360a_amp, + .sof_fw_filename = "sof-adl.ri", + .sof_tplg_filename = "sof-adl-mx98360a-nau8825.tplg", + }, + { + .id = "10508825", + .drv_name = "sof_nau8825", + .sof_fw_filename = "sof-adl.ri", + .sof_tplg_filename = "sof-adl-nau8825.tplg", + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_adl_machines); -- 2.20.1

2 1

[PATCH] ASoc: Intel: glk_rt5682_max98357a: support ALC5682I-VS codec
by Brent Lu 28 Oct '21

28 Oct '21

Detect the codec variant in probe function and update DAI link accordingly. Also add an new entry in enumeration table for machine driver enumeration. Signed-off-by: Brent Lu <brent.lu(a)intel.com> --- sound/soc/intel/boards/Kconfig | 1 + sound/soc/intel/boards/glk_rt5682_max98357a.c | 53 +++++++++++++++---- .../intel/common/soc-acpi-intel-glk-match.c | 9 ++++ 3 files changed, 54 insertions(+), 9 deletions(-) diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index f693383eb6e3..2dd5ff7e35ce 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -427,6 +427,7 @@ config SND_SOC_INTEL_GLK_RT5682_MAX98357A_MACH depends on MFD_INTEL_LPSS || COMPILE_TEST depends on SND_HDA_CODEC_HDMI && SND_SOC_SOF_HDA_AUDIO_CODEC select SND_SOC_RT5682_I2C + select SND_SOC_RT5682S select SND_SOC_MAX98357A select SND_SOC_DMIC select SND_SOC_HDAC_HDMI diff --git a/sound/soc/intel/boards/glk_rt5682_max98357a.c b/sound/soc/intel/boards/glk_rt5682_max98357a.c index 9d75beec09d1..058aa7cb899a 100644 --- a/sound/soc/intel/boards/glk_rt5682_max98357a.c +++ b/sound/soc/intel/boards/glk_rt5682_max98357a.c @@ -18,14 +18,18 @@ #include <sound/soc.h> #include <sound/soc-acpi.h> #include "../../codecs/rt5682.h" +#include "../../codecs/rt5682s.h" #include "../../codecs/hdac_hdmi.h" #include "hda_dsp_common.h" /* The platform clock outputs 19.2Mhz clock to codec as I2S MCLK */ #define GLK_PLAT_CLK_FREQ 19200000 #define RT5682_PLL_FREQ (48000 * 512) -#define GLK_REALTEK_CODEC_DAI "rt5682-aif1" +#define RT5682_DAI_NAME "rt5682-aif1" +#define RT5682S_DAI_NAME "rt5682s-aif1" #define GLK_MAXIM_CODEC_DAI "HiFi" +#define RT5682_DEV0_NAME "i2c-10EC5682:00" +#define RT5682S_DEV0_NAME "i2c-RTL5682:00" #define MAXIM_DEV0_NAME "MX98357A:00" #define DUAL_CHANNEL 2 #define QUAD_CHANNEL 4 @@ -43,6 +47,7 @@ struct glk_card_private { struct snd_soc_jack geminilake_headset; struct list_head hdmi_pcm_list; bool common_hdmi_codec_drv; + int is_rt5682s; }; enum { @@ -139,9 +144,19 @@ static int geminilake_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component; struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); struct snd_soc_jack *jack; - int ret; + int pll_id, pll_source, clk_id, ret; + + if (ctx->is_rt5682s) { + pll_id = RT5682S_PLL2; + pll_source = RT5682S_PLL_S_MCLK; + clk_id = RT5682S_SCLK_S_PLL2; + } else { + pll_id = RT5682_PLL1; + pll_source = RT5682_PLL1_S_MCLK; + clk_id = RT5682_SCLK_S_PLL1; + } - ret = snd_soc_dai_set_pll(codec_dai, 0, RT5682_PLL1_S_MCLK, + ret = snd_soc_dai_set_pll(codec_dai, pll_id, pll_source, GLK_PLAT_CLK_FREQ, RT5682_PLL_FREQ); if (ret < 0) { dev_err(rtd->dev, "can't set codec pll: %d\n", ret); @@ -149,7 +164,7 @@ static int geminilake_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd) } /* Configure sysclk for codec */ - ret = snd_soc_dai_set_sysclk(codec_dai, RT5682_SCLK_S_PLL1, + ret = snd_soc_dai_set_sysclk(codec_dai, clk_id, RT5682_PLL_FREQ, SND_SOC_CLOCK_IN); if (ret < 0) dev_err(rtd->dev, "snd_soc_dai_set_sysclk err = %d\n", ret); @@ -344,9 +359,12 @@ SND_SOC_DAILINK_DEF(ssp1_codec, SND_SOC_DAILINK_DEF(ssp2_pin, DAILINK_COMP_ARRAY(COMP_CPU("SSP2 Pin"))); -SND_SOC_DAILINK_DEF(ssp2_codec, - DAILINK_COMP_ARRAY(COMP_CODEC("i2c-10EC5682:00", - GLK_REALTEK_CODEC_DAI))); +SND_SOC_DAILINK_DEF(ssp2_codec_5682, + DAILINK_COMP_ARRAY(COMP_CODEC(RT5682_DEV0_NAME, + RT5682_DAI_NAME))); +SND_SOC_DAILINK_DEF(ssp2_codec_5682s, + DAILINK_COMP_ARRAY(COMP_CODEC(RT5682S_DEV0_NAME, + RT5682S_DAI_NAME))); SND_SOC_DAILINK_DEF(dmic_pin, DAILINK_COMP_ARRAY(COMP_CPU("DMIC01 Pin"))); @@ -492,7 +510,7 @@ static struct snd_soc_dai_link geminilake_dais[] = { .ops = &geminilake_rt5682_ops, .dpcm_playback = 1, .dpcm_capture = 1, - SND_SOC_DAILINK_REG(ssp2_pin, ssp2_codec, platform), + SND_SOC_DAILINK_REG(ssp2_pin, ssp2_codec_5682, platform), }, { .name = "dmic01", @@ -592,12 +610,29 @@ static int geminilake_audio_probe(struct platform_device *pdev) struct snd_soc_acpi_mach *mach; const char *platform_name; struct snd_soc_card *card; - int ret; + int ret, i; ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); if (!ctx) return -ENOMEM; + /* Detect the headset codec variant */ + if (acpi_dev_present("RTL5682", NULL, -1)) { + /* ALC5682I-VS is detected */ + ctx->is_rt5682s = 1; + + for (i = 0; i < glk_audio_card_rt5682_m98357a.num_links; i++) { + if (strcmp(geminilake_dais[i].name, "SSP2-Codec")) + continue; + + /* update the dai link to use rt5682s codec */ + geminilake_dais[i].codecs = ssp2_codec_5682s; + geminilake_dais[i].num_codecs = ARRAY_SIZE(ssp2_codec_5682s); + break; + } + } else + ctx->is_rt5682s = 0; + INIT_LIST_HEAD(&ctx->hdmi_pcm_list); card = &glk_audio_card_rt5682_m98357a; diff --git a/sound/soc/intel/common/soc-acpi-intel-glk-match.c b/sound/soc/intel/common/soc-acpi-intel-glk-match.c index 32fff9389eb3..4de4add74443 100644 --- a/sound/soc/intel/common/soc-acpi-intel-glk-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-glk-match.c @@ -40,6 +40,15 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_glk_machines[] = { .sof_fw_filename = "sof-glk.ri", .sof_tplg_filename = "sof-glk-rt5682.tplg", }, + { + .id = "RTL5682", + .drv_name = "glk_rt5682_max98357a", + .fw_filename = "intel/dsp_fw_glk.bin", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &glk_codecs, + .sof_fw_filename = "sof-glk.ri", + .sof_tplg_filename = "sof-glk-rt5682.tplg", + }, { .id = "10134242", .drv_name = "glk_cs4242_mx98357a", -- 2.25.1

3 2

[PATCH RFC] ALSA: control: Use xarray for faster lookups
by Takashi Iwai 28 Oct '21

28 Oct '21

The control elements are managed in a single linked list and we traverse the whole list for matching each numid or ctl id per every inquiry of a control element. This is OK-ish for a small number of elements but obviously it doesn't scale. Especially the matching with the ctl id takes time because it checks each field of the snd_ctl_id element, e.g. the name string is matched with strcmp(). This patch adds the hash tables with Xarray in addition to the linked list for improving the lookup speed of a control element. There are two xarray tables added to the card; one for numid and another for ctl id. For the numid, we use the numid as the index, while for the ctl id, we calculate a hash key. The lookup is done via a single xa_load(). When nothing hits, we look through the full list entries as a fallback. So, the inquiry for a non-existing element would take (even slightly longer) time with this patch. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- This is not really meant to be merged now, but just shown as a reference implementation, as the issue doesn't surface unless you have a large number of elements and query the late element frequently. In my test, 1,000,000 queries of the 1000th element took over 12 seconds, while the patched kernel took 0.2 second. But for the most cases (at most 20 or 30 elements), it doesn't matter much. include/sound/core.h | 3 + sound/core/control.c | 142 +++++++++++++++++++++++++++++++++---------- sound/core/init.c | 2 + 3 files changed, 115 insertions(+), 32 deletions(-) diff --git a/include/sound/core.h b/include/sound/core.h index b7e9b58d3c78..e8bb6d943f3f 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -14,6 +14,7 @@ #include <linux/pm.h> /* pm_message_t */ #include <linux/stringify.h> #include <linux/printk.h> +#include <linux/xarray.h> /* number of supported soundcards */ #ifdef CONFIG_SND_DYNAMIC_MINORS @@ -103,6 +104,8 @@ struct snd_card { size_t user_ctl_alloc_size; // current memory allocation by user controls. struct list_head controls; /* all controls for this card */ struct list_head ctl_files; /* active control files */ + struct xarray ctl_numids; /* hash table for numids */ + struct xarray ctl_hash; /* hash table for ctl id matching */ struct snd_info_entry *proc_root; /* root for soundcard specific files */ struct proc_dir_entry *proc_root_link; /* number link to real id */ diff --git a/sound/core/control.c b/sound/core/control.c index a25c0d64d104..d0253cc4b22d 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -364,6 +364,78 @@ static int snd_ctl_find_hole(struct snd_card *card, unsigned int count) return 0; } +/* Compute a hash key for the corresponding ctl id + * It's for the name lookup, hence the numid is excluded. + * The hash key is bound in LONG_MAX to be used for Xarray key. + */ +#define MULTIPLIER 37 +static unsigned long get_ctl_id_hash(const struct snd_ctl_elem_id *id) +{ + unsigned long h; + const unsigned char *p; + + h = id->iface; + h = MULTIPLIER * h + id->device; + h = MULTIPLIER * h + id->subdevice; + for (p = id->name; *p; p++) + h = MULTIPLIER * h + *p; + h = MULTIPLIER * h + id->index; + h &= LONG_MAX; + return h; +} + +/* check whether the given id is contained in the given kctl */ +static bool elem_id_matches(const struct snd_kcontrol *kctl, + const struct snd_ctl_elem_id *id) +{ + return kctl->id.iface == id->iface && + kctl->id.device == id->device && + kctl->id.subdevice == id->subdevice && + !strncmp(kctl->id.name, id->name, sizeof(kctl->id.name)) && + kctl->id.index <= id->index && + kctl->id.index + kctl->count > id->index; +} + +/* add hash entries to numid and ctl xarray tables */ +static void add_hash_entries(struct snd_card *card, + struct snd_kcontrol *kcontrol) +{ + struct snd_ctl_elem_id id = kcontrol->id; + int i, err; + + xa_store_range(&card->ctl_numids, kcontrol->id.numid, + kcontrol->id.numid + kcontrol->count - 1, + kcontrol, GFP_KERNEL); + + for (i = 0; i < kcontrol->count; i++) { + id.index = kcontrol->id.index + i; + err = xa_insert(&card->ctl_hash, get_ctl_id_hash(&id), + kcontrol, GFP_KERNEL); + /* ignore errors - it's a best effort */ + } +} + +/* remove hash entries that have been added */ +static void remove_hash_entries(struct snd_card *card, + struct snd_kcontrol *kcontrol) +{ + struct snd_ctl_elem_id id = kcontrol->id; + struct snd_kcontrol *matched; + unsigned long h; + int i; + + for (i = 0; i < kcontrol->count; i++) { + xa_erase(&card->ctl_numids, id.numid); + h = get_ctl_id_hash(&id); + matched = xa_load(&card->ctl_hash, h); + if (matched && (matched == kcontrol || + elem_id_matches(matched, &id))) + xa_erase(&card->ctl_hash, h); + id.index++; + id.numid++; + } +} + enum snd_ctl_add_mode { CTL_ADD_EXCLUSIVE, CTL_REPLACE, CTL_ADD_ON_REPLACE, }; @@ -408,6 +480,8 @@ static int __snd_ctl_add_replace(struct snd_card *card, kcontrol->id.numid = card->last_numid + 1; card->last_numid += kcontrol->count; + add_hash_entries(card, kcontrol); + for (idx = 0; idx < kcontrol->count; idx++) snd_ctl_notify_one(card, SNDRV_CTL_EVENT_MASK_ADD, kcontrol, idx); @@ -479,6 +553,26 @@ int snd_ctl_replace(struct snd_card *card, struct snd_kcontrol *kcontrol, } EXPORT_SYMBOL(snd_ctl_replace); +static int __snd_ctl_remove(struct snd_card *card, + struct snd_kcontrol *kcontrol, + bool remove_hash) +{ + unsigned int idx; + + if (snd_BUG_ON(!card || !kcontrol)) + return -EINVAL; + list_del(&kcontrol->list); + + if (remove_hash) + remove_hash_entries(card, kcontrol); + + card->controls_count -= kcontrol->count; + for (idx = 0; idx < kcontrol->count; idx++) + snd_ctl_notify_one(card, SNDRV_CTL_EVENT_MASK_REMOVE, kcontrol, idx); + snd_ctl_free_one(kcontrol); + return 0; +} + /** * snd_ctl_remove - remove the control from the card and release it * @card: the card instance @@ -492,16 +586,7 @@ EXPORT_SYMBOL(snd_ctl_replace); */ int snd_ctl_remove(struct snd_card *card, struct snd_kcontrol *kcontrol) { - unsigned int idx; - - if (snd_BUG_ON(!card || !kcontrol)) - return -EINVAL; - list_del(&kcontrol->list); - card->controls_count -= kcontrol->count; - for (idx = 0; idx < kcontrol->count; idx++) - snd_ctl_notify_one(card, SNDRV_CTL_EVENT_MASK_REMOVE, kcontrol, idx); - snd_ctl_free_one(kcontrol); - return 0; + return __snd_ctl_remove(card, kcontrol, true); } EXPORT_SYMBOL(snd_ctl_remove); @@ -642,9 +727,11 @@ int snd_ctl_rename_id(struct snd_card *card, struct snd_ctl_elem_id *src_id, up_write(&card->controls_rwsem); return -ENOENT; } + remove_hash_entries(card, kctl); kctl->id = *dst_id; kctl->id.numid = card->last_numid + 1; card->last_numid += kctl->count; + add_hash_entries(card, kctl); up_write(&card->controls_rwsem); return 0; } @@ -665,15 +752,9 @@ EXPORT_SYMBOL(snd_ctl_rename_id); */ struct snd_kcontrol *snd_ctl_find_numid(struct snd_card *card, unsigned int numid) { - struct snd_kcontrol *kctl; - if (snd_BUG_ON(!card || !numid)) return NULL; - list_for_each_entry(kctl, &card->controls, list) { - if (kctl->id.numid <= numid && kctl->id.numid + kctl->count > numid) - return kctl; - } - return NULL; + return xa_load(&card->ctl_numids, numid); } EXPORT_SYMBOL(snd_ctl_find_numid); @@ -699,21 +780,15 @@ struct snd_kcontrol *snd_ctl_find_id(struct snd_card *card, return NULL; if (id->numid != 0) return snd_ctl_find_numid(card, id->numid); - list_for_each_entry(kctl, &card->controls, list) { - if (kctl->id.iface != id->iface) - continue; - if (kctl->id.device != id->device) - continue; - if (kctl->id.subdevice != id->subdevice) - continue; - if (strncmp(kctl->id.name, id->name, sizeof(kctl->id.name))) - continue; - if (kctl->id.index > id->index) - continue; - if (kctl->id.index + kctl->count <= id->index) - continue; + kctl = xa_load(&card->ctl_hash, get_ctl_id_hash(id)); + if (kctl && elem_id_matches(kctl, id)) return kctl; - } + + /* no matching in hash table - try all as the last resort */ + list_for_each_entry(kctl, &card->controls, list) + if (elem_id_matches(kctl, id)) + return kctl; + return NULL; } EXPORT_SYMBOL(snd_ctl_find_id); @@ -2195,8 +2270,11 @@ static int snd_ctl_dev_free(struct snd_device *device) down_write(&card->controls_rwsem); while (!list_empty(&card->controls)) { control = snd_kcontrol(card->controls.next); - snd_ctl_remove(card, control); + __snd_ctl_remove(card, control, false); } + + xa_destroy(&card->ctl_numids); + xa_destroy(&card->ctl_hash); up_write(&card->controls_rwsem); put_device(&card->ctl_dev); return 0; diff --git a/sound/core/init.c b/sound/core/init.c index ac335f5906c6..a1979fe27d53 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -282,6 +282,8 @@ static int snd_card_init(struct snd_card *card, struct device *parent, rwlock_init(&card->ctl_files_rwlock); INIT_LIST_HEAD(&card->controls); INIT_LIST_HEAD(&card->ctl_files); + xa_init(&card->ctl_numids); + xa_init(&card->ctl_hash); spin_lock_init(&card->files_lock); INIT_LIST_HEAD(&card->files_list); mutex_init(&card->memory_mutex); -- 2.31.1

1 0

[PATCH] ASoC: qdsp6: audioreach: Fix clang -Wimplicit-fallthrough
by Nathan Chancellor 28 Oct '21

28 Oct '21

Clang warns: sound/soc/qcom/qdsp6/topology.c:465:3: warning: unannotated fall-through between switch labels [-Wimplicit-fallthrough] default: ^ sound/soc/qcom/qdsp6/topology.c:465:3: note: insert 'break;' to avoid fall-through default: ^ break; 1 warning generated. Clang is a little more pedantic than GCC, which permits implicit fallthroughs to cases that contain just break or return. Clang's version is more in line with the kernel's own stance in deprecated.rst, which states that all switch/case blocks must end in either break, fallthrough, continue, goto, or return. Add the missing break to fix the warning. Link: https://github.com/ClangBuiltLinux/linux/issues/1495 Signed-off-by: Nathan Chancellor <nathan(a)kernel.org> --- sound/soc/qcom/qdsp6/topology.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/qcom/qdsp6/topology.c b/sound/soc/qcom/qdsp6/topology.c index f31895379925..bd649c232a06 100644 --- a/sound/soc/qcom/qdsp6/topology.c +++ b/sound/soc/qcom/qdsp6/topology.c @@ -461,7 +461,7 @@ static struct audioreach_module *audioreach_parse_common_tokens(struct q6apm *ap break; case AR_TKN_U32_MODULE_DST_IN_PORT_ID: dst_mod_ip_port_id = le32_to_cpu(mod_elem->value); - + break; default: break; base-commit: 1dcc81d95b910dc404c40ff6101bfa2520a3528e -- 2.33.1.637.gf443b226ca

2 1

[PATCH] dt-bindings: sound: wlf,wm8903: Convert txt bindings to yaml
by David Heidelberg 28 Oct '21

28 Oct '21

Convert the Wolfson WM8903 Ultra-Low Power Stereo CODEC Device Tree binding documentation to json-schema. Signed-off-by: David Heidelberg <david(a)ixit.cz> --- .../devicetree/bindings/sound/wlf,wm8903.yaml | 116 ++++++++++++++++++ .../devicetree/bindings/sound/wm8903.txt | 82 ------------- 2 files changed, 116 insertions(+), 82 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/wlf,wm8903.yaml delete mode 100644 Documentation/devicetree/bindings/sound/wm8903.txt diff --git a/Documentation/devicetree/bindings/sound/wlf,wm8903.yaml b/Documentation/devicetree/bindings/sound/wlf,wm8903.yaml new file mode 100644 index 000000000000..afdcf9002b60 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wlf,wm8903.yaml @@ -0,0 +1,116 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: "http://devicetree.org/schemas/sound/wlf,wm8903.yaml#" +$schema: "http://devicetree.org/meta-schemas/core.yaml#" + +title: WM8903 audio codec + +description: | + This device supports I2C only. + Pins on the device (for linking into audio routes): + * IN1L + * IN1R + * IN2L + * IN2R + * IN3L + * IN3R + * DMICDAT + * HPOUTL + * HPOUTR + * LINEOUTL + * LINEOUTR + * LOP + * LON + * ROP + * RON + * MICBIAS + +maintainers: + - Mark Brown <broonie(a)opensource.wolfsonmicro.com> + +properties: + compatible: + const: wlf,wm8903 + + reg: + maxItems: 1 + + gpio-controller: true + '#gpio-cells': + const: 2 + + interrupts: + maxItems: 1 + + micdet-cfg: + $ref: /schemas/types.yaml#/definitions/uint32 + default: 0 + description: Default register value for R6 (Mic Bias). + + micdet-delay: + $ref: /schemas/types.yaml#/definitions/uint32 + default: 100 + description: The debounce delay for microphone detection in mS. + + gpio-cfg: + $ref: /schemas/types.yaml#/definitions/uint32-array + description: | + minItems: 5 + maxItems: 5 + A list of GPIO configuration register values. + If absent, no configuration of these registers is performed. + If any entry has the value 0xffffffff, that GPIO's + configuration will not be modified. + + AVDD-supply: + description: Analog power supply regulator on the AVDD pin. + + CPVDD-supply: + description: Charge pump supply regulator on the CPVDD pin. + + DBVDD-supply: + description: Digital buffer supply regulator for the DBVDD pin. + + DCVDD-supply: + description: Digital core supply regulator for the DCVDD pin. + + +required: + - compatible + - reg + - gpio-controller + - '#gpio-cells' + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + wm8903: codec@1a { + compatible = "wlf,wm8903"; + reg = <0x1a>; + interrupts = <347>; + + AVDD-supply = <&fooreg_a>; + CPVDD-supply = <&fooreg_b>; + DBVDD-supply = <&fooreg_c>; + DCVDD-supply = <&fooreg_d>; + + gpio-controller; + #gpio-cells = <2>; + + micdet-cfg = <0>; + micdet-delay = <100>; + gpio-cfg = < + 0x0600 /* DMIC_LR, output */ + 0x0680 /* DMIC_DAT, input */ + 0x0000 /* GPIO, output, low */ + 0x0200 /* Interrupt, output */ + 0x01a0 /* BCLK, input, active high */ + >; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/wm8903.txt b/Documentation/devicetree/bindings/sound/wm8903.txt deleted file mode 100644 index 6371c2434afe..000000000000 --- a/Documentation/devicetree/bindings/sound/wm8903.txt +++ /dev/null @@ -1,82 +0,0 @@ -WM8903 audio CODEC - -This device supports I2C only. - -Required properties: - - - compatible : "wlf,wm8903" - - - reg : the I2C address of the device. - - - gpio-controller : Indicates this device is a GPIO controller. - - - #gpio-cells : Should be two. The first cell is the pin number and the - second cell is used to specify optional parameters (currently unused). - -Optional properties: - - - interrupts : The interrupt line the codec is connected to. - - - micdet-cfg : Default register value for R6 (Mic Bias). If absent, the - default is 0. - - - micdet-delay : The debounce delay for microphone detection in mS. If - absent, the default is 100. - - - gpio-cfg : A list of GPIO configuration register values. The list must - be 5 entries long. If absent, no configuration of these registers is - performed. If any entry has the value 0xffffffff, that GPIO's - configuration will not be modified. - - - AVDD-supply : Analog power supply regulator on the AVDD pin. - - - CPVDD-supply : Charge pump supply regulator on the CPVDD pin. - - - DBVDD-supply : Digital buffer supply regulator for the DBVDD pin. - - - DCVDD-supply : Digital core supply regulator for the DCVDD pin. - -Pins on the device (for linking into audio routes): - - * IN1L - * IN1R - * IN2L - * IN2R - * IN3L - * IN3R - * DMICDAT - * HPOUTL - * HPOUTR - * LINEOUTL - * LINEOUTR - * LOP - * LON - * ROP - * RON - * MICBIAS - -Example: - -wm8903: codec@1a { - compatible = "wlf,wm8903"; - reg = <0x1a>; - interrupts = < 347 >; - - AVDD-supply = <&fooreg_a>; - CPVDD-supply = <&fooreg_b>; - DBVDD-supply = <&fooreg_c>; - DCVDC-supply = <&fooreg_d>; - - gpio-controller; - #gpio-cells = <2>; - - micdet-cfg = <0>; - micdet-delay = <100>; - gpio-cfg = < - 0x0600 /* DMIC_LR, output */ - 0x0680 /* DMIC_DAT, input */ - 0x0000 /* GPIO, output, low */ - 0x0200 /* Interrupt, output */ - 0x01a0 /* BCLK, input, active high */ - >; -}; -- 2.33.0

1 0

[PATCH 1/2] alsaloop: Renamed field capt_pitch to ctl_pitch
by Pavel Hofman 28 Oct '21

28 Oct '21

Renaming capt_pitch to ctl_pitch to respect the naming style of snd_ctl_elem_value_t fields in loopback_handle. Also making the variable name suitable for planned support of playback pitch. Signed-off-by: Pavel Hofman <pavel.hofman(a)ivitera.com> --- alsaloop/alsaloop.h | 2 +- alsaloop/pcmjob.c | 16 ++++++++-------- 2 files changed, 9 insertions(+), 9 deletions(-) diff --git a/alsaloop/alsaloop.h b/alsaloop/alsaloop.h index 7a98ef3..a5539ef 100644 --- a/alsaloop/alsaloop.h +++ b/alsaloop/alsaloop.h @@ -122,7 +122,7 @@ struct loopback_handle { unsigned int ctl_pollfd_count; snd_ctl_elem_value_t *ctl_notify; snd_ctl_elem_value_t *ctl_rate_shift; - snd_ctl_elem_value_t *capt_pitch; + snd_ctl_elem_value_t *ctl_pitch; snd_ctl_elem_value_t *ctl_active; snd_ctl_elem_value_t *ctl_format; snd_ctl_elem_value_t *ctl_rate; diff --git a/alsaloop/pcmjob.c b/alsaloop/pcmjob.c index 13fa595..7b47c40 100644 --- a/alsaloop/pcmjob.c +++ b/alsaloop/pcmjob.c @@ -1063,9 +1063,9 @@ static int set_rate_shift(struct loopback_handle *lhandle, double pitch) if (lhandle->ctl_rate_shift) { snd_ctl_elem_value_set_integer(lhandle->ctl_rate_shift, 0, pitch * 100000); err = snd_ctl_elem_write(lhandle->ctl, lhandle->ctl_rate_shift); - } else if (lhandle->capt_pitch) { - snd_ctl_elem_value_set_integer(lhandle->capt_pitch, 0, (1 / pitch) * 1000000); - err = snd_ctl_elem_write(lhandle->ctl, lhandle->capt_pitch); + } else if (lhandle->ctl_pitch) { + snd_ctl_elem_value_set_integer(lhandle->ctl_pitch, 0, (1 / pitch) * 1000000); + err = snd_ctl_elem_write(lhandle->ctl, lhandle->ctl_pitch); } else { return 0; } @@ -1252,7 +1252,7 @@ static int openctl(struct loopback_handle *lhandle, int device, int subdevice) openctl_elem(lhandle, device, subdevice, "PCM Rate Shift 100000", &lhandle->ctl_rate_shift); openctl_elem(lhandle, device, subdevice, "Capture Pitch 1000000", - &lhandle->capt_pitch); + &lhandle->ctl_pitch); set_rate_shift(lhandle, 1); openctl_elem(lhandle, device, subdevice, "PCM Slave Active", &lhandle->ctl_active); @@ -1338,9 +1338,9 @@ static int closeit(struct loopback_handle *lhandle) if (lhandle->ctl_rate_shift) snd_ctl_elem_value_free(lhandle->ctl_rate_shift); lhandle->ctl_rate_shift = NULL; - if (lhandle->capt_pitch) - snd_ctl_elem_value_free(lhandle->capt_pitch); - lhandle->capt_pitch = NULL; + if (lhandle->ctl_pitch) + snd_ctl_elem_value_free(lhandle->ctl_pitch); + lhandle->ctl_pitch = NULL; if (lhandle->ctl) err = snd_ctl_close(lhandle->ctl); lhandle->ctl = NULL; @@ -1386,7 +1386,7 @@ int pcmjob_init(struct loopback *loop) snprintf(id, sizeof(id), "%s/%s", loop->play->id, loop->capt->id); id[sizeof(id)-1] = '\0'; loop->id = strdup(id); - if (loop->sync == SYNC_TYPE_AUTO && (loop->capt->ctl_rate_shift || loop->capt->capt_pitch)) + if (loop->sync == SYNC_TYPE_AUTO && (loop->capt->ctl_rate_shift || loop->capt->ctl_pitch)) loop->sync = SYNC_TYPE_CAPTRATESHIFT; if (loop->sync == SYNC_TYPE_AUTO && loop->play->ctl_rate_shift) loop->sync = SYNC_TYPE_PLAYRATESHIFT; -- 2.25.1

2 2

[PATCH alsa-utils] alsactl: Add --quiet option to suppress alsa-lib error messages
by Takashi Iwai 28 Oct '21

28 Oct '21

alsactl prints error messages from alsa-lib as is, and it's rather annoying to see the error messages like alsactl[xxx]: alsa-lib parser.c:242:(error_node) UCM is not supported for this HDA model (HDA Intel PCH...) alsactl[xxx]: alsa-lib main.c:1405:(snd_use_case_mgr_open) error: failed to import hw:0 use case configuration -6 that are recorded in the syslog at each boot. This patch adds --quiet (or -q) option to suppress those error messages from alsa-lib, and applies to the systemd services as default. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- alsactl/alsa-restore.service.in | 4 ++-- alsactl/alsa-state.service.in | 4 ++-- alsactl/alsactl.1 | 4 ++++ alsactl/alsactl.c | 5 +++++ alsactl/alsactl.h | 2 ++ alsactl/utils.c | 2 ++ 6 files changed, 17 insertions(+), 4 deletions(-) diff --git a/alsactl/alsa-restore.service.in b/alsactl/alsa-restore.service.in index 80fd5fd48203..06bc21c355de 100644 --- a/alsactl/alsa-restore.service.in +++ b/alsactl/alsa-restore.service.in @@ -11,5 +11,5 @@ ConditionPathExistsGlob=/dev/snd/control* [Service] Type=oneshot RemainAfterExit=true -ExecStart=-@sbindir@/alsactl restore -ExecStop=-@sbindir@/alsactl store +ExecStart=-@sbindir@/alsactl -q restore +ExecStop=-@sbindir@/alsactl -q store diff --git a/alsactl/alsa-state.service.in b/alsactl/alsa-state.service.in index 5a8fe5eeed7e..6b6e704ad8e5 100644 --- a/alsactl/alsa-state.service.in +++ b/alsactl/alsa-state.service.in @@ -9,5 +9,5 @@ ConditionPathExists=@daemonswitch@ [Service] Type=simple -ExecStart=-@sbindir@/alsactl -s -n 19 -c rdaemon -ExecStop=-@sbindir@/alsactl -s kill save_and_quit +ExecStart=-@sbindir@/alsactl -q -s -n 19 -c rdaemon +ExecStop=-@sbindir@/alsactl -q -s kill save_and_quit diff --git a/alsactl/alsactl.1 b/alsactl/alsactl.1 index 8296663a7c2d..0a0b4597a137 100644 --- a/alsactl/alsactl.1 +++ b/alsactl/alsactl.1 @@ -94,6 +94,10 @@ Use debug mode: a bit more verbose. \fI\-v, \-\-version\fP Print alsactl version number. +.TP +\fI\-q, \-\-quiet\fP +Suppress error messages from alsa-lib. + .TP \fI\-f, \-\-file\fP Select the configuration file to use. The default is /var/lib/alsa/asound.state. diff --git a/alsactl/alsactl.c b/alsactl/alsactl.c index 05738fb6b35f..e3bef2c69558 100644 --- a/alsactl/alsactl.c +++ b/alsactl/alsactl.c @@ -52,6 +52,7 @@ int use_syslog = 0; char *command; char *statefile = NULL; char *lockfile = SYS_LOCKFILE; +int verbose_error = 1; #define TITLE 0x0100 #define HEADER 0x0200 @@ -74,6 +75,7 @@ static struct arg args[] = { { 'h', "help", "this help" }, { 'd', "debug", "debug mode" }, { 'v', "version", "print version of this program" }, +{ 'q', "quiet", "suppress errors from alsa-lib" }, { HEADER, NULL, "Available state options:" }, { FILEARG | 'f', "file", "configuration file (default " SYS_ASOUNDRC ")" }, { FILEARG | 'a', "config-dir", "boot / hotplug configuration directory (default " SYS_ASOUND_DIR ")" }, @@ -370,6 +372,9 @@ int main(int argc, char *argv[]) printf("alsactl version " SND_UTIL_VERSION_STR "\n"); res = EXIT_SUCCESS; goto out; + case 'q': + verbose_error = 0; + break; case '?': // error msg already printed help(); res = EXIT_FAILURE; diff --git a/alsactl/alsactl.h b/alsactl/alsactl.h index bbdf6c88baeb..f9c25931c924 100644 --- a/alsactl/alsactl.h +++ b/alsactl/alsactl.h @@ -83,3 +83,5 @@ static inline int hextodigit(int c) } #define ARRAY_SIZE(a) (sizeof (a) / sizeof (a)[0]) + +extern int verbose_error; diff --git a/alsactl/utils.c b/alsactl/utils.c index a50797259f21..dfb06e1a4a15 100644 --- a/alsactl/utils.c +++ b/alsactl/utils.c @@ -182,6 +182,8 @@ void error_handler(const char *file, int line, const char *function, int err, co char buf[2048]; va_list arg; + if (!verbose_error) + return; va_start(arg, fmt); vsnprintf(buf, sizeof(buf), fmt, arg); va_end(arg); -- 2.31.1

2 3

[PATCH 0/3] Make genaral and simple for new sof machine driver
by David Lin 28 Oct '21

28 Oct '21

The series of features will make general and simple for new sof machine driver. David Lin (3): ASoC: nau8825: add set_jack coponment support ASoC: nau8825: add disable jack detection support ASoC: nau8825: add clock management for power saving sound/soc/codecs/nau8825.c | 49 ++++++++++++++++++++++++++++++++++++-- 1 file changed, 47 insertions(+), 2 deletions(-)

4 8

[PATCH] ALSA: hda/realtek: Add a quirk for HP OMEN 15 mute LED
by Takashi Iwai 28 Oct '21

28 Oct '21

HP OMEN 15 laptop requires the quirk to fiddle with COEF 0x0b bit 2 for toggling the mute LED. It's already implemented for other HP laptops, and we just need to add a proper fixup entry. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=214735 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e82d756ab850..6322fac9e694 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8634,6 +8634,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { ALC285_FIXUP_HP_GPIO_AMP_INIT), SND_PCI_QUIRK(0x103c, 0x8783, "HP ZBook Fury 15 G7 Mobile Workstation", ALC285_FIXUP_HP_GPIO_AMP_INIT), + SND_PCI_QUIRK(0x103c, 0x8788, "HP OMEN 15", ALC285_FIXUP_HP_MUTE_LED), SND_PCI_QUIRK(0x103c, 0x87c8, "HP", ALC287_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x87e5, "HP ProBook 440 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x87e7, "HP ProBook 450 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED), -- 2.31.1

1 0