October 2023 - Alsa-devel - mailman.alsa-project.org

[PATCH 00/10] ASoC: Intel: more machine driver updates for 6.7
by Pierre-Louis Bossart 19 Oct '23

19 Oct '23

Two nice cleanups from Brent Lu and Charles Keepax, and one RaptorLake update. Brent Lu (5): ASoC: Intel: board_helpers: support dmic link initialization ASoC: Intel: sof_cs42l42: use common module for DMIC links ASoC: Intel: sof_nau8825: use common module for DMIC links ASoC: Intel: sof_rt5682: use common module for DMIC links ASoC: Intel: sof_ssp_amp: use common module for DMIC links Charles Keepax (4): ASoC: intel: sof_sdw: Stop processing CODECs when enough are found ASoC: intel: sof_sdw_cs42l43: Some trivial formatting clean ups ASoC: intel: sof_sdw_cs42l43: Create separate jacks for hp and mic ASoC: intel: sof_sdw: Move the builtin microphones to dataport 1 Terry Cheong (1): ASoC: Intel: sof_nau8825: add RPL support for MAX98360A amp sound/soc/intel/boards/sof_board_helpers.c | 92 +++++++++++++ sound/soc/intel/boards/sof_board_helpers.h | 10 ++ sound/soc/intel/boards/sof_cs42l42.c | 121 +++++------------- sound/soc/intel/boards/sof_nau8825.c | 89 ++++--------- sound/soc/intel/boards/sof_rt5682.c | 81 +++--------- sound/soc/intel/boards/sof_sdw.c | 16 +-- sound/soc/intel/boards/sof_sdw_cs42l43.c | 31 +++-- sound/soc/intel/boards/sof_ssp_amp.c | 65 +++------- .../intel/common/soc-acpi-intel-rpl-match.c | 7 + 9 files changed, 233 insertions(+), 279 deletions(-) -- 2.39.2

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[PATCH] ASoC: tlv320aic31xx: switch to gpiod_set_value_cansleep
by Marco Felsch 19 Oct '23

19 Oct '23

Switch to gpiod_set_value_cansleep() to support gpiochips which can sleep like i2c gpio expanders. Signed-off-by: Marco Felsch <m.felsch(a)pengutronix.de> --- sound/soc/codecs/tlv320aic31xx.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index 9611aa8acb0d..4d7c5a80c6ed 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -1208,7 +1208,7 @@ static int aic31xx_regulator_event(struct notifier_block *nb, * supplies was disabled. */ if (aic31xx->gpio_reset) - gpiod_set_value(aic31xx->gpio_reset, 1); + gpiod_set_value_cansleep(aic31xx->gpio_reset, 1); regcache_mark_dirty(aic31xx->regmap); dev_dbg(aic31xx->dev, "## %s: DISABLE received\n", __func__); @@ -1222,9 +1222,9 @@ static int aic31xx_reset(struct aic31xx_priv *aic31xx) int ret = 0; if (aic31xx->gpio_reset) { - gpiod_set_value(aic31xx->gpio_reset, 1); + gpiod_set_value_cansleep(aic31xx->gpio_reset, 1); ndelay(10); /* At least 10ns */ - gpiod_set_value(aic31xx->gpio_reset, 0); + gpiod_set_value_cansleep(aic31xx->gpio_reset, 0); } else { ret = regmap_write(aic31xx->regmap, AIC31XX_RESET, 1); } -- 2.39.2

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[v2 0/2] ASoC: mediatek: mt8186_mt6366_rt1019_rt5682s: add rt5650 support
by xiazhengqiao 19 Oct '23

19 Oct '23

To use RT5650 as the codec and the amp, add a new sound card named mt8186_rt5650. Changes in v2: - PATCH 2/2: Modify the code format in line 1058 - Link to v1: https://lore.kernel.org/all/54ec6435-e69b-444a-a31a-c878a4b3564a@collabora.… https://lore.kernel.org/all/8de0772b-3145-44e3-8174-702c7dc729bb@collabora.… xiazhengqiao (2): ASoC: dt-bindings: mt8186-mt6366-rt1019-rt5682s: add RT5650 support ASoC: mediatek: mt8186_mt6366_rt1019_rt5682s: add rt5650 support .../sound/mt8186-mt6366-rt1019-rt5682s.yaml | 1 + sound/soc/mediatek/Kconfig | 1 + .../mt8186/mt8186-mt6366-rt1019-rt5682s.c | 44 ++++++++++++++++++- 3 files changed, 45 insertions(+), 1 deletion(-) -- 2.17.1

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[PATCH 0/3] ASoC: qcom: check ADSP version when setting clocks
by Otto Pflüger 19 Oct '23

19 Oct '23

The apq8016_sbc driver currently works on APQ8016 and MSM8916 devices. It should also work on MSM8909 (and newer SoCs like MSM8917 and MSM8953 if the quinary MI2S line is added); however, newer devices with these SoCs ship with newer firmware that uses a different interface for controlling the digital codec and bit clocks, which causes the driver to fail because it cannot set LPAIF_BIT_CLK. In order to fix this problem, modify the LPAIF_* clock implementation in the qdsp6 driver to use the newer clock API if a newer firmware version is detected. This seems to be a better solution than exposing the firmware version to other drivers like apq8016_sbc and forcing them to figure out which clock to use. On MSM8916, a hack is currently used to control the LPAIF_DIG_CLK directly through the GCC driver, but on devices with the newer firmware, the INTERNAL_DIGITAL_CODEC_CORE clock provided by q6afe-clocks in the qdsp6 driver can be used instead. Add a fallback to make this clock work with the older firmware too, allowing one to specify it as the codec's "mclk" in the device tree: compatible = "qcom,msm8916-wcd-digital-codec"; clocks = <&xo_board>, <&q6afecc LPASS_CLK_ID_INTERNAL_DIGITAL_CODEC_CORE LPASS_CLK_ATTRIBUTE_COUPLE_NO>; clock-names = "ahbix", "mclk"; assigned-clocks = <&q6afecc LPASS_CLK_ID_INTERNAL_DIGITAL_CODEC_CORE LPASS_CLK_ATTRIBUTE_COUPLE_NO>; assigned-clock-rates = <9600000>; This works both on MSM8916 and on the newer SoCs mentioned above. Otto Pflüger (3): ASoC: qcom: q6core: expose ADSP core firmware version ASoC: qcom: q6afe: check ADSP version when setting clocks ASoC: qcom: q6afe: remove "port already open" error sound/soc/qcom/qdsp6/q6afe.c | 85 +++++++++++++++++++++++++++++++++-- sound/soc/qcom/qdsp6/q6core.c | 61 +++++++++++++++++++++++++ sound/soc/qcom/qdsp6/q6core.h | 9 ++++ 3 files changed, 152 insertions(+), 3 deletions(-) base-commit: e3b18f7200f45d66f7141136c25554ac1e82009b -- 2.39.2

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[PATCH] ASoC: amd: yc: Add DMI entry to support System76 Pangolin 13
by Tim Crawford 19 Oct '23

19 Oct '23

From: Jeremy Soller <jeremy(a)system76.com> Add pang13 quirk to enable the internal microphone. Signed-off-by: Jeremy Soller <jeremy(a)system76.com> Signed-off-by: Tim Crawford <tcrawford(a)system76.com> --- sound/soc/amd/yc/acp6x-mach.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/amd/yc/acp6x-mach.c b/sound/soc/amd/yc/acp6x-mach.c index 3ec15b46fa35..fd5b7d6c6eec 100644 --- a/sound/soc/amd/yc/acp6x-mach.c +++ b/sound/soc/amd/yc/acp6x-mach.c @@ -353,6 +353,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_VERSION, "pang12"), } }, + { + .driver_data = &acp6x_card, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "System76"), + DMI_MATCH(DMI_PRODUCT_VERSION, "pang13"), + } + }, {} }; -- 2.41.0

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[PATCH] ASoC: amd: yc: Add DMI entry to support System76 Pangolin 13
by Tim Crawford 19 Oct '23

19 Oct '23

From: Jeremy Soller <jeremy(a)system76.com> Add pang13 quirk to enable the internal microphone. Signed-off-by: Jeremy Soller <jeremy(a)system76.com> Signed-off-by: Tim Crawford <tcrawford(a)system76.com> --- sound/soc/amd/yc/acp6x-mach.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/amd/yc/acp6x-mach.c b/sound/soc/amd/yc/acp6x-mach.c index 3ec15b46fa35..fd5b7d6c6eec 100644 --- a/sound/soc/amd/yc/acp6x-mach.c +++ b/sound/soc/amd/yc/acp6x-mach.c @@ -353,6 +353,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_VERSION, "pang12"), } }, + { + .driver_data = &acp6x_card, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "System76"), + DMI_MATCH(DMI_PRODUCT_VERSION, "pang13"), + } + }, {} }; -- 2.41.0

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[PATCH v7 00/13] sound: Use -EPROBE_DEFER instead of i915 module loading.
by Maarten Lankhorst 19 Oct '23

19 Oct '23

Explicitly loading i915 becomes a problem when upstreaming the new intel driver for Tiger Lake and higher graphics (xe). By loading i915, it doesn't wait for driver load of xe, and will fail completely before it loads. -EPROBE_DEFER has to be returned before any device is created in probe(), otherwise the removal of the device will cause EPROBE_DEFER to try again in an infinite loop. The conversion is done in gradual steps. First I add an argument to snd_hdac_i915_init to allow for -EPROBE_DEFER so I can convert each driver separately. Then I convert each driver to move snd_hdac_i915_init out of the workqueue. Finally I drop the ability to choose modprobe behavior after the last user is converted. Compared to previous version, I added the patch "ASoC: SOF: Intel: Fix error handling in hda_init()" Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Cezary Rojewski <cezary.rojewski(a)intel.com> Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Cc: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> Cc: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Cc: Bard Liao <yung-chuan.liao(a)linux.intel.com> Cc: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Cc: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Daniel Baluta <daniel.baluta(a)nxp.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-kernel(a)vger.kernel.org Cc: sound-open-firmware(a)alsa-project.org Maarten Lankhorst (11): ASoC: SOF: core: Ensure sof_ops_free() is still called when probe never ran. ASoC: SOF: Intel: Fix error handling in hda_init() ALSA: hda: Intel: Fix error handling in azx_probe() ALSA: hda: i915: Allow override of gpu binding. ALSA: hda: i915: Add an allow_modprobe argument to snd_hdac_i915_init ALSA: hda: i915: Allow xe as match for i915_component_master_match ASoC: Intel: avs: Move snd_hdac_i915_init to before probe_work. ALSA: hda: Intel: Move snd_hdac_i915_init to before probe_work. ASoC: Intel: Skylake: Move snd_hdac_i915_init to before probe_work. ASoC: SOF: Intel: Move binding to display driver outside of deferred probe ALSA: hda: i915: Remove extra argument from snd_hdac_i915_init Pierre-Louis Bossart (2): ASoC: SOF: core: Add probe_early and remove_late callbacks ASoC: SOF: Intel: hda: start splitting the probe sound/hda/hdac_i915.c | 24 ++++++----- sound/pci/hda/hda_intel.c | 60 ++++++++++++++-------------- sound/soc/intel/avs/core.c | 13 ++++-- sound/soc/intel/skylake/skl.c | 31 +++++--------- sound/soc/sof/core.c | 17 +++++++- sound/soc/sof/intel/hda-common-ops.c | 2 + sound/soc/sof/intel/hda.c | 46 +++++++++++++-------- sound/soc/sof/intel/hda.h | 2 + sound/soc/sof/ops.h | 16 ++++++++ sound/soc/sof/sof-priv.h | 2 + 10 files changed, 129 insertions(+), 84 deletions(-) -- 2.39.2

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[PATCH] ASoC: da7213: Add new kcontrol for tonegen
by David Rau 19 Oct '23

19 Oct '23

Add new kcontrol for tone generator Signed-off-by: David Rau <David.Rau.opensource(a)dm.renesas.com> --- sound/soc/codecs/da7213.c | 171 +++++++++++++++++++++++++++++++++++++- sound/soc/codecs/da7213.h | 64 +++++++++++++- 2 files changed, 233 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/da7213.c b/sound/soc/codecs/da7213.c index 3a6449c44b23..31f32de0e1ed 100644 --- a/sound/soc/codecs/da7213.c +++ b/sound/soc/codecs/da7213.c @@ -55,6 +55,7 @@ static const DECLARE_TLV_DB_SCALE(hp_vol_tlv, -5700, 100, 0); static const DECLARE_TLV_DB_SCALE(lineout_vol_tlv, -4800, 100, 0); static const DECLARE_TLV_DB_SCALE(alc_threshold_tlv, -9450, 150, 0); static const DECLARE_TLV_DB_SCALE(alc_gain_tlv, 0, 600, 0); +static const DECLARE_TLV_DB_SCALE(da7213_tonegen_gain_tlv, -4500, 300, 0); /* ADC and DAC voice mode (8kHz) high pass cutoff value */ static const char * const da7213_voice_hpf_corner_txt[] = { @@ -86,6 +87,23 @@ static SOC_ENUM_SINGLE_DECL(da7213_adc_audio_hpf_corner, DA7213_AUDIO_HPF_CORNER_SHIFT, da7213_audio_hpf_corner_txt); +static const char * const da7213_tonegen_dtmf_key_txt[] = { + "0", "1", "2", "3", "4", "5", "6", "7", "8", "9", "A", "B", "C", "D", + "*", "#" +}; + +static const struct soc_enum da7213_tonegen_dtmf_key = + SOC_ENUM_SINGLE(DA7213_TONE_GEN_CFG1, DA7213_DTMF_REG_SHIFT, + DA7213_DTMF_REG_MAX, da7213_tonegen_dtmf_key_txt); + +static const char * const da7213_tonegen_swg_sel_txt[] = { + "Sum", "SWG1", "SWG2", "Sum" +}; + +static const struct soc_enum da7213_tonegen_swg_sel = + SOC_ENUM_SINGLE(DA7213_TONE_GEN_CFG2, DA7213_SWG_SEL_SHIFT, + DA7213_SWG_SEL_MAX, da7213_tonegen_swg_sel_txt); + /* Gain ramping rate value */ static const char * const da7213_gain_ramp_rate_txt[] = { "nominal rate * 8", "nominal rate * 16", "nominal rate / 16", @@ -191,6 +209,64 @@ static SOC_ENUM_SINGLE_DECL(da7213_alc_integ_release_rate, * Control Functions */ +/* Locked Kcontrol calls */ +static int da7213_volsw_locked_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct da7213_priv *da7213 = snd_soc_component_get_drvdata(component); + int ret; + + mutex_lock(&da7213->ctrl_lock); + ret = snd_soc_get_volsw(kcontrol, ucontrol); + mutex_unlock(&da7213->ctrl_lock); + + return ret; +} + +static int da7213_volsw_locked_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct da7213_priv *da7213 = snd_soc_component_get_drvdata(component); + int ret; + + mutex_lock(&da7213->ctrl_lock); + ret = snd_soc_put_volsw(kcontrol, ucontrol); + mutex_unlock(&da7213->ctrl_lock); + + return ret; +} + +static int da7213_enum_locked_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct da7213_priv *da7213 = snd_soc_component_get_drvdata(component); + int ret; + + mutex_lock(&da7213->ctrl_lock); + ret = snd_soc_get_enum_double(kcontrol, ucontrol); + mutex_unlock(&da7213->ctrl_lock); + + return ret; +} + +static int da7213_enum_locked_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct da7213_priv *da7213 = snd_soc_component_get_drvdata(component); + int ret; + + mutex_lock(&da7213->ctrl_lock); + ret = snd_soc_put_enum_double(kcontrol, ucontrol); + mutex_unlock(&da7213->ctrl_lock); + + return ret; +} + +/* ALC */ static int da7213_get_alc_data(struct snd_soc_component *component, u8 reg_val) { int mid_data, top_data; @@ -376,6 +452,64 @@ static int da7213_put_alc_sw(struct snd_kcontrol *kcontrol, return snd_soc_put_volsw(kcontrol, ucontrol); } +/* ToneGen */ +static int da7213_tonegen_freq_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct da7213_priv *da7213 = snd_soc_component_get_drvdata(component); + struct soc_mixer_control *mixer_ctrl = + (struct soc_mixer_control *) kcontrol->private_value; + unsigned int reg = mixer_ctrl->reg; + __le16 val; + int ret; + + mutex_lock(&da7213->ctrl_lock); + ret = regmap_raw_read(da7213->regmap, reg, &val, sizeof(val)); + mutex_unlock(&da7213->ctrl_lock); + + if (ret) + return ret; + + /* + * Frequency value spans two 8-bit registers, lower then upper byte. + * Therefore we need to convert to host endianness here. + */ + ucontrol->value.integer.value[0] = le16_to_cpu(val); + + return 0; +} + +static int da7213_tonegen_freq_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct da7213_priv *da7213 = snd_soc_component_get_drvdata(component); + struct soc_mixer_control *mixer_ctrl = + (struct soc_mixer_control *) kcontrol->private_value; + unsigned int reg = mixer_ctrl->reg; + __le16 val_new, val_old; + int ret; + + /* + * Frequency value spans two 8-bit registers, lower then upper byte. + * Therefore we need to convert to little endian here to align with + * HW registers. + */ + val_new = cpu_to_le16(ucontrol->value.integer.value[0]); + + mutex_lock(&da7213->ctrl_lock); + ret = regmap_raw_read(da7213->regmap, reg, &val_old, sizeof(val_old)); + if (ret == 0 && (val_old != val_new)) + ret = regmap_raw_write(da7213->regmap, reg, + &val_new, sizeof(val_new)); + mutex_unlock(&da7213->ctrl_lock); + + if (ret < 0) + return ret; + + return val_old != val_new; +} /* * KControls @@ -477,6 +611,37 @@ static const struct snd_kcontrol_new da7213_snd_controls[] = { SOC_DOUBLE_R("Headphone ZC Switch", DA7213_HP_L_CTRL, DA7213_HP_R_CTRL, DA7213_ZC_EN_SHIFT, DA7213_ZC_EN_MAX, DA7213_NO_INVERT), + /* Tone Generator */ + SOC_SINGLE_EXT_TLV("ToneGen Volume", DA7213_TONE_GEN_CFG2, + DA7213_TONE_GEN_GAIN_SHIFT, DA7213_TONE_GEN_GAIN_MAX, + DA7213_NO_INVERT, da7213_volsw_locked_get, + da7213_volsw_locked_put, da7213_tonegen_gain_tlv), + SOC_ENUM_EXT("ToneGen DTMF Key", da7213_tonegen_dtmf_key, + da7213_enum_locked_get, da7213_enum_locked_put), + SOC_SINGLE_EXT("ToneGen DTMF Switch", DA7213_TONE_GEN_CFG1, + DA7213_DTMF_EN_SHIFT, DA7213_SWITCH_EN_MAX, + DA7213_NO_INVERT, da7213_volsw_locked_get, + da7213_volsw_locked_put), + SOC_SINGLE_EXT("ToneGen Start", DA7213_TONE_GEN_CFG1, + DA7213_START_STOPN_SHIFT, DA7213_SWITCH_EN_MAX, + DA7213_NO_INVERT, da7213_volsw_locked_get, + da7213_volsw_locked_put), + SOC_ENUM_EXT("ToneGen Sinewave Gen Type", da7213_tonegen_swg_sel, + da7213_enum_locked_get, da7213_enum_locked_put), + SOC_SINGLE_EXT("ToneGen Sinewave1 Freq", DA7213_TONE_GEN_FREQ1_L, + DA7213_FREQ1_L_SHIFT, DA7213_FREQ_MAX, DA7213_NO_INVERT, + da7213_tonegen_freq_get, da7213_tonegen_freq_put), + SOC_SINGLE_EXT("ToneGen Sinewave2 Freq", DA7213_TONE_GEN_FREQ2_L, + DA7213_FREQ2_L_SHIFT, DA7213_FREQ_MAX, DA7213_NO_INVERT, + da7213_tonegen_freq_get, da7213_tonegen_freq_put), + SOC_SINGLE_EXT("ToneGen On Time", DA7213_TONE_GEN_ON_PER, + DA7213_BEEP_ON_PER_SHIFT, DA7213_BEEP_ON_OFF_MAX, + DA7213_NO_INVERT, da7213_volsw_locked_get, + da7213_volsw_locked_put), + SOC_SINGLE("ToneGen Off Time", DA7213_TONE_GEN_OFF_PER, + DA7213_BEEP_OFF_PER_SHIFT, DA7213_BEEP_ON_OFF_MAX, + DA7213_NO_INVERT), + /* Gain Ramping controls */ SOC_DOUBLE_R("Aux Gain Ramping Switch", DA7213_AUX_L_CTRL, DA7213_AUX_R_CTRL, DA7213_GAIN_RAMP_EN_SHIFT, @@ -765,7 +930,7 @@ static int da7213_dai_event(struct snd_soc_dapm_widget *w, /* Check SRM has locked */ do { pll_status = snd_soc_component_read(component, DA7213_PLL_STATUS); - if (pll_status & DA7219_PLL_SRM_LOCK) { + if (pll_status & DA7213_PLL_SRM_LOCK) { srm_lock = true; } else { ++i; @@ -1931,6 +2096,9 @@ static int da7213_probe(struct snd_soc_component *component) da7213->fixed_clk_auto_pll = true; } + /* Default infinite tone gen, start/stop by Kcontrol */ + snd_soc_component_write(component, DA7213_TONE_GEN_CYCLES, DA7213_BEEP_CYCLES_MASK); + return 0; } @@ -2078,4 +2246,5 @@ module_i2c_driver(da7213_i2c_driver); MODULE_DESCRIPTION("ASoC DA7213 Codec driver"); MODULE_AUTHOR("Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com>"); +MODULE_AUTHOR("David Rau <David.Rau.opensource(a)dm.renesas.com>"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/da7213.h b/sound/soc/codecs/da7213.h index 4ca9cfdea06d..505b731c0adb 100644 --- a/sound/soc/codecs/da7213.h +++ b/sound/soc/codecs/da7213.h @@ -5,6 +5,7 @@ * Copyright (c) 2013 Dialog Semiconductor * * Author: Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com> + * Author: David Rau <David.Rau.opensource(a)dm.renesas.com> */ #ifndef _DA7213_H @@ -135,13 +136,24 @@ #define DA7213_DAC_NG_ON_THRESHOLD 0xB1 #define DA7213_DAC_NG_CTRL 0xB2 +#define DA7213_TONE_GEN_CFG1 0xB4 +#define DA7213_TONE_GEN_CFG2 0xB5 +#define DA7213_TONE_GEN_CYCLES 0xB6 +#define DA7213_TONE_GEN_FREQ1_L 0xB7 +#define DA7213_TONE_GEN_FREQ1_U 0xB8 +#define DA7213_TONE_GEN_FREQ2_L 0xB9 +#define DA7213_TONE_GEN_FREQ2_U 0xBA +#define DA7213_TONE_GEN_ON_PER 0xBB +#define DA7213_TONE_GEN_OFF_PER 0xBC /* * Bit fields */ +#define DA7213_SWITCH_EN_MAX 0x1 + /* DA7213_PLL_STATUS = 0x03 */ -#define DA7219_PLL_SRM_LOCK (0x1 << 1) +#define DA7213_PLL_SRM_LOCK (0x1 << 1) /* DA7213_SR = 0x22 */ #define DA7213_SR_8000 (0x1 << 0) @@ -484,6 +496,55 @@ #define DA7213_DAC_NG_EN_SHIFT 7 #define DA7213_DAC_NG_EN_MAX 0x1 +/* DA7213_TONE_GEN_CFG1 = 0xB4 */ +#define DA7213_DTMF_REG_SHIFT 0 +#define DA7213_DTMF_REG_MASK (0xF << 0) +#define DA7213_DTMF_REG_MAX 16 +#define DA7213_DTMF_EN_SHIFT 4 +#define DA7213_DTMF_EN_MASK (0x1 << 4) +#define DA7213_START_STOPN_SHIFT 7 +#define DA7213_START_STOPN_MASK (0x1 << 7) + +/* DA7213_TONE_GEN_CFG2 = 0xB5 */ +#define DA7213_SWG_SEL_SHIFT 0 +#define DA7213_SWG_SEL_MASK (0x3 << 0) +#define DA7213_SWG_SEL_MAX 4 +#define DA7213_SWG_SEL_SRAMP (0x3 << 0) +#define DA7213_TONE_GEN_GAIN_SHIFT 4 +#define DA7213_TONE_GEN_GAIN_MASK (0xF << 4) +#define DA7213_TONE_GEN_GAIN_MAX 0xF +#define DA7213_TONE_GEN_GAIN_MINUS_9DB (0x3 << 4) +#define DA7213_TONE_GEN_GAIN_MINUS_15DB (0x5 << 4) + +/* DA7213_TONE_GEN_CYCLES = 0xB6 */ +#define DA7213_BEEP_CYCLES_SHIFT 0 +#define DA7213_BEEP_CYCLES_MASK (0x7 << 0) + +/* DA7213_TONE_GEN_FREQ1_L = 0xB7 */ +#define DA7213_FREQ1_L_SHIFT 0 +#define DA7213_FREQ1_L_MASK (0xFF << 0) +#define DA7213_FREQ_MAX 0xFFFF + +/* DA7213_TONE_GEN_FREQ1_U = 0xB8 */ +#define DA7213_FREQ1_U_SHIFT 0 +#define DA7213_FREQ1_U_MASK (0xFF << 0) + +/* DA7213_TONE_GEN_FREQ2_L = 0xB9 */ +#define DA7213_FREQ2_L_SHIFT 0 +#define DA7213_FREQ2_L_MASK (0xFF << 0) + +/* DA7213_TONE_GEN_FREQ2_U = 0xBA */ +#define DA7213_FREQ2_U_SHIFT 0 +#define DA7213_FREQ2_U_MASK (0xFF << 0) + +/* DA7213_TONE_GEN_ON_PER = 0xBB */ +#define DA7213_BEEP_ON_PER_SHIFT 0 +#define DA7213_BEEP_ON_PER_MASK (0x3F << 0) +#define DA7213_BEEP_ON_OFF_MAX 0x3F + +/* DA7213_TONE_GEN_OFF_PER = 0xBC */ +#define DA7213_BEEP_OFF_PER_SHIFT 0 +#define DA7213_BEEP_OFF_PER_MASK (0x3F << 0) /* * General defines @@ -534,6 +595,7 @@ enum da7213_supplies { /* Codec private data */ struct da7213_priv { struct regmap *regmap; + struct mutex ctrl_lock; struct regulator_bulk_data supplies[DA7213_NUM_SUPPLIES]; struct clk *mclk; unsigned int mclk_rate; -- 2.17.1

2 1

[RFC PATCH v6 00/11] Add audio support in v4l2 framework
by Shengjiu Wang 19 Oct '23

19 Oct '23

Audio signal processing also has the requirement for memory to memory similar as Video. This asrc memory to memory (memory ->asrc->memory) case is a non real time use case. User fills the input buffer to the asrc module, after conversion, then asrc sends back the output buffer to user. So it is not a traditional ALSA playback and capture case. It is a specific use case, there is no reference in current kernel. v4l2 memory to memory is the closed implementation, v4l2 current support video, image, radio, tuner, touch devices, so it is not complicated to add support for this specific audio case. Because we had implemented the "memory -> asrc ->i2s device-> codec" use case in ALSA. Now the "memory->asrc->memory" needs to reuse the code in asrc driver, so the first 3 patches is for refining the code to make it can be shared by the "memory->asrc->memory" driver. The main change is in the v4l2 side, A /dev/vl4-audioX will be created, user applications only use the ioctl of v4l2 framework. Other change is to add memory to memory support for two kinds of i.MX ASRC module. changes in v6: - use m2m_prepare/m2m_unprepare/m2m_start/m2m_stop to replace m2m_start_part_one/m2m_stop_part_one, m2m_start_part_two/m2m_stop_part_two. - change V4L2_CTRL_TYPE_ASRC_RATE to V4L2_CTRL_TYPE_FIXED_POINT - fix warning by kernel test rebot - remove some unused format V4L2_AUDIO_FMT_XX - Get SNDRV_PCM_FORMAT from V4L2_AUDIO_FMT in driver. - rename audm2m to viaudm2m. changes in v5: - remove V4L2_AUDIO_FMT_LPCM - define audio pixel format like V4L2_AUDIO_FMT_S8... - remove rate and format in struct v4l2_audio_format. - Add V4L2_CID_ASRC_SOURCE_RATE and V4L2_CID_ASRC_DEST_RATE controls - updata document accordingly. changes in v4: - update document style - separate V4L2_AUDIO_FMT_LPCM and V4L2_CAP_AUDIO_M2M in separate commit changes in v3: - Modify documents for adding audio m2m support - Add audio virtual m2m driver - Defined V4L2_AUDIO_FMT_LPCM format type for audio. - Defined V4L2_CAP_AUDIO_M2M capability type for audio m2m case. - with modification in v4l-utils, pass v4l2-compliance test. changes in v2: - decouple the implementation in v4l2 and ALSA - implement the memory to memory driver as a platfrom driver and move it to driver/media - move fsl_asrc_common.h to include/sound folder Shengjiu Wang (11): ASoC: fsl_asrc: define functions for memory to memory usage ASoC: fsl_easrc: define functions for memory to memory usage ASoC: fsl_asrc: move fsl_asrc_common.h to include/sound ASoC: fsl_asrc: register m2m platform device ASoC: fsl_easrc: register m2m platform device media: uapi: Add V4L2_CAP_AUDIO_M2M capability flag media: v4l2: Add audio capture and output support media: uapi: define audio sample format fourcc type media: uapi: Add audio rate controls support media: imx-asrc: Add memory to memory driver media: viaudm2m: add virtual driver for audio memory to memory .../userspace-api/media/v4l/buffer.rst | 6 + .../userspace-api/media/v4l/common.rst | 1 + .../media/v4l/dev-audio-mem2mem.rst | 71 + .../userspace-api/media/v4l/devices.rst | 1 + .../media/v4l/ext-ctrls-fixed-point.rst | 36 + .../userspace-api/media/v4l/pixfmt-audio.rst | 202 +++ .../userspace-api/media/v4l/pixfmt.rst | 1 + .../media/v4l/vidioc-enum-fmt.rst | 2 + .../media/v4l/vidioc-g-ext-ctrls.rst | 4 + .../userspace-api/media/v4l/vidioc-g-fmt.rst | 4 + .../media/v4l/vidioc-querycap.rst | 3 + .../media/v4l/vidioc-queryctrl.rst | 7 + .../media/videodev2.h.rst.exceptions | 4 + .../media/common/videobuf2/videobuf2-v4l2.c | 4 + drivers/media/platform/nxp/Kconfig | 12 + drivers/media/platform/nxp/Makefile | 1 + drivers/media/platform/nxp/imx-asrc.c | 1248 +++++++++++++++++ drivers/media/test-drivers/Kconfig | 9 + drivers/media/test-drivers/Makefile | 1 + drivers/media/test-drivers/viaudm2m.c | 707 ++++++++++ drivers/media/v4l2-core/v4l2-ctrls-core.c | 5 + drivers/media/v4l2-core/v4l2-ctrls-defs.c | 4 + drivers/media/v4l2-core/v4l2-dev.c | 17 + drivers/media/v4l2-core/v4l2-ioctl.c | 89 ++ include/media/v4l2-ctrls.h | 2 + include/media/v4l2-dev.h | 2 + include/media/v4l2-ioctl.h | 34 + .../fsl => include/sound}/fsl_asrc_common.h | 60 + include/uapi/linux/v4l2-controls.h | 13 + include/uapi/linux/videodev2.h | 69 + sound/soc/fsl/fsl_asrc.c | 144 ++ sound/soc/fsl/fsl_asrc.h | 4 +- sound/soc/fsl/fsl_asrc_dma.c | 2 +- sound/soc/fsl/fsl_easrc.c | 212 +++ sound/soc/fsl/fsl_easrc.h | 6 +- 35 files changed, 2984 insertions(+), 3 deletions(-) create mode 100644 Documentation/userspace-api/media/v4l/dev-audio-mem2mem.rst create mode 100644 Documentation/userspace-api/media/v4l/ext-ctrls-fixed-point.rst create mode 100644 Documentation/userspace-api/media/v4l/pixfmt-audio.rst create mode 100644 drivers/media/platform/nxp/imx-asrc.c create mode 100644 drivers/media/test-drivers/viaudm2m.c rename {sound/soc/fsl => include/sound}/fsl_asrc_common.h (60%) -- 2.34.1

4 36

[v1 0/2] ASoC: mediatek: mt8186_mt6366_rt1019_rt5682s: add rt5650 support
by xiazhengqiao 19 Oct '23

19 Oct '23

To use RT5650 as the codec and the amp, add a new sound card named mt8186_rt5650. xiazhengqiao (2): ASoC: dt-bindings: mt8186-mt6366-rt1019-rt5682s: add RT5650 support ASoC: mediatek: mt8186_mt6366_rt1019_rt5682s: add rt5650 support .../sound/mt8186-mt6366-rt1019-rt5682s.yaml | 1 + sound/soc/mediatek/Kconfig | 1 + .../mt8186/mt8186-mt6366-rt1019-rt5682s.c | 45 ++++++++++++++++++- 3 files changed, 46 insertions(+), 1 deletion(-) -- 2.17.1

2 3