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

[PATCH 1/2] ASoC: cs35l41: Change monitor widgets to siggens
by David Rhodes 16 Nov '21

16 Nov '21

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> Currently the internal monitor sources are input widgets, which means if the card is set to fully routed these will not enable unless connected to something in the machine driver. However, all these are internal monitor signals so it makes no sense to connect them to something in the machine driver. As such switch them to siggen widgets which will have the same behaviour except not require external linkage on a fully routed card. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Signed-off-by: David Rhodes <drhodes(a)opensource.cirrus.com> --- sound/soc/codecs/cs35l41.c | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/cs35l41.c b/sound/soc/codecs/cs35l41.c index 94ed21d7676f..9d0530dde996 100644 --- a/sound/soc/codecs/cs35l41.c +++ b/sound/soc/codecs/cs35l41.c @@ -612,6 +612,12 @@ static const struct snd_soc_dapm_widget cs35l41_dapm_widgets[] = { SND_SOC_DAPM_AIF_OUT("ASPTX3", NULL, 0, CS35L41_SP_ENABLES, 2, 0), SND_SOC_DAPM_AIF_OUT("ASPTX4", NULL, 0, CS35L41_SP_ENABLES, 3, 0), + SND_SOC_DAPM_SIGGEN("VSENSE"), + SND_SOC_DAPM_SIGGEN("ISENSE"), + SND_SOC_DAPM_SIGGEN("VP"), + SND_SOC_DAPM_SIGGEN("VBST"), + SND_SOC_DAPM_SIGGEN("TEMP"), + SND_SOC_DAPM_ADC("VMON ADC", NULL, CS35L41_PWR_CTRL2, 12, 0), SND_SOC_DAPM_ADC("IMON ADC", NULL, CS35L41_PWR_CTRL2, 13, 0), SND_SOC_DAPM_ADC("VPMON ADC", NULL, CS35L41_PWR_CTRL2, 8, 0), @@ -623,12 +629,6 @@ static const struct snd_soc_dapm_widget cs35l41_dapm_widgets[] = { cs35l41_main_amp_event, SND_SOC_DAPM_POST_PMD | SND_SOC_DAPM_POST_PMU), - SND_SOC_DAPM_INPUT("VP"), - SND_SOC_DAPM_INPUT("VBST"), - SND_SOC_DAPM_INPUT("ISENSE"), - SND_SOC_DAPM_INPUT("VSENSE"), - SND_SOC_DAPM_INPUT("TEMP"), - SND_SOC_DAPM_MUX("ASP TX1 Source", SND_SOC_NOPM, 0, 0, &asp_tx1_mux), SND_SOC_DAPM_MUX("ASP TX2 Source", SND_SOC_NOPM, 0, 0, &asp_tx2_mux), SND_SOC_DAPM_MUX("ASP TX3 Source", SND_SOC_NOPM, 0, 0, &asp_tx3_mux), @@ -674,8 +674,8 @@ static const struct snd_soc_dapm_route cs35l41_audio_map[] = { {"VMON ADC", NULL, "VSENSE"}, {"IMON ADC", NULL, "ISENSE"}, {"VPMON ADC", NULL, "VP"}, - {"TEMPMON ADC", NULL, "TEMP"}, {"VBSTMON ADC", NULL, "VBST"}, + {"TEMPMON ADC", NULL, "TEMP"}, {"ASPRX1", NULL, "AMP Playback"}, {"ASPRX2", NULL, "AMP Playback"}, -- 2.25.1

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[PATCH] ASoC: SOF: core: Unregister machine driver before IPC and debugfs
by Peter Ujfalusi 16 Nov '21

16 Nov '21

To ensure clean unload of the machine driver, components and topology, do the unregister before we free IPC and debugfs. It is a possibility that part of the unregister we would have IPC communication with the firmware. Suggested-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> --- sound/soc/sof/core.c | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/soc/sof/core.c b/sound/soc/sof/core.c index 2c3de295f11f..af5bbc861297 100644 --- a/sound/soc/sof/core.c +++ b/sound/soc/sof/core.c @@ -362,6 +362,13 @@ int snd_sof_device_remove(struct device *dev) if (IS_ENABLED(CONFIG_SND_SOC_SOF_PROBE_WORK_QUEUE)) cancel_work_sync(&sdev->probe_work); + /* + * Unregister machine driver. This will unbind the snd_card which + * will remove the component driver and unload the topology + * before freeing the snd_card. + */ + snd_sof_machine_unregister(sdev, pdata); + if (sdev->fw_state > SOF_FW_BOOT_NOT_STARTED) { ret = snd_sof_dsp_power_down_notify(sdev); if (ret < 0) @@ -373,13 +380,6 @@ int snd_sof_device_remove(struct device *dev) snd_sof_free_trace(sdev); } - /* - * Unregister machine driver. This will unbind the snd_card which - * will remove the component driver and unload the topology - * before freeing the snd_card. - */ - snd_sof_machine_unregister(sdev, pdata); - /* * Unregistering the machine driver results in unloading the topology. * Some widgets, ex: scheduler, attempt to power down the core they are -- 2.33.1

2 1

[PATCH v2] ASoC: cs42l42: Add control for audio slow-start switch
by Richard Fitzgerald 16 Nov '21

16 Nov '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 | 30 +++++++++++++++++++++++++++++- sound/soc/codecs/cs42l42.h | 3 +++ 2 files changed, 32 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index 27a1c4c73074..56804a3f285e 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 +389,28 @@ 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; + } + + return snd_soc_component_update_bits(component, CS42L42_SLOW_START_ENABLE, + CS42L42_SLOW_START_EN_MASK, val); +} + static const char * const cs42l42_hpf_freq_text[] = { "1.86Hz", "120Hz", "235Hz", "466Hz" }; @@ -431,7 +455,11 @@ 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

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[PATCH] ASoC: wm_adsp: wm_adsp_control_add() error: uninitialized symbol 'ret'
by Simon Trimmer 16 Nov '21

16 Nov '21

This patch fixes the static analysis warning as it is correctly indicating a possible code path, it cannot know that for the affected firmware versions subname would always be NULL. Reported-by: kernel test robot <lkp(a)intel.com> Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Signed-off-by: Simon Trimmer <simont(a)opensource.cirrus.com> --- sound/soc/codecs/wm_adsp.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index d4f0d72cbcc8..6cb01a8e08fb 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -617,8 +617,9 @@ static int wm_adsp_control_add(struct cs_dsp_coeff_ctl *cs_ctl) switch (cs_dsp->fw_ver) { case 0: case 1: - snprintf(name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, "%s %s %x", - cs_dsp->name, region_name, cs_ctl->alg_region.alg); + ret = scnprintf(name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, + "%s %s %x", cs_dsp->name, region_name, + cs_ctl->alg_region.alg); break; case 2: ret = scnprintf(name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, -- 2.33.0

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[PATCH 0/2] Add tfa9897 rcv-gpios support
by Vincent Knecht 15 Nov '21

15 Nov '21

This is the continuation of a previous series [1] where - patch 1/4 is removed in favor of using pin switch This will be posted independently of tfa989x support, since it mainly require changes to sound/soc/qcom/common.c and device DTS. - patch 2/4 is already merged so here are reworked patch 3/4 (bindings fixed and example added) and patch 4/4 unchanged. [1] https://patchwork.kernel.org/project/alsa-devel/cover/20211024085840.153643… Vincent Knecht (2): ASoC: dt-bindings: nxp, tfa989x: Add rcv-gpios property for tfa9897 ASoC: codecs: tfa989x: Add support for tfa9897 optional rcv-gpios .../bindings/sound/nxp,tfa989x.yaml | 41 +++++++++++++++++++ sound/soc/codecs/tfa989x.c | 20 ++++++++- 2 files changed, 60 insertions(+), 1 deletion(-) -- 2.31.1

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[PATCH 00/10] ASoC: Intel: sof_sdw: Use fixed DAI link id
by Bard Liao 15 Nov '21

15 Nov '21

This series provides a way to use constant dailink numbers for different devices. So that we don't need to renumber them in topologies. Some patches with different purpose are sent together in this series since they are dependent. Bard Liao (8): ASoC: intel: sof_sdw: return the original error number ASoC: intel: sof_sdw: rename be_index/link_id to link_index ASoC: intel: sof_sdw: Use a fixed DAI link id for AMP ASoC: intel: sof_sdw: move DMIC link id overwrite to create_sdw_dailink ASoC: intel: sof_sdw: remove SOF_RT715_DAI_ID_FIX quirk ASoC: intel: sof_sdw: remove sof_sdw_mic_codec_mockup_init ASoC: intel: sof_sdw: remove get_next_be_id ASoC: intel: sof_sdw: add link adr order check Pierre-Louis Bossart (2): ASoC: Intel: sof_sdw: fix jack detection on HP Spectre x360 convertible ASoC: Intel: sof_sdw: add SKU for Dell Latitude 9520 sound/soc/intel/boards/sof_sdw.c | 152 ++++++++++---------- sound/soc/intel/boards/sof_sdw_common.h | 7 +- sound/soc/intel/boards/sof_sdw_rt715.c | 7 - sound/soc/intel/boards/sof_sdw_rt715_sdca.c | 7 - 4 files changed, 85 insertions(+), 88 deletions(-) -- 2.17.1

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[PATCH 0/3] ASoC: SOF: Implement support for DMA trace free IPC command
by Peter Ujfalusi 15 Nov '21

15 Nov '21

Hi, This series adds support for the new DMA_TRACE_FREE IPC message to gracefully stop and free the DMA trace related resources on the firmware side. Regards, Peter --- Ranjani Sridharan (3): ASoC: SOF: IPC: Add new IPC command to free trace DMA ASoC: SOF: IPC: update ipc_log_header() ASoC: SOF: trace: send DMA_TRACE_FREE IPC during release include/sound/sof/header.h | 1 + sound/soc/sof/core.c | 2 +- sound/soc/sof/ipc.c | 17 ++++++++++++++++- sound/soc/sof/trace.c | 18 ++++++++++++++++++ 4 files changed, 36 insertions(+), 2 deletions(-) -- 2.33.1

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[PATCH] ASoC: sunxi: sun4i-spdif: Implement IEC958 control
by Jernej Skrabec 15 Nov '21

15 Nov '21

SPDIF core is capable of sending custom status. Implement IEC958 control handling. Signed-off-by: Jernej Skrabec <jernej.skrabec(a)gmail.com> --- sound/soc/sunxi/sun4i-spdif.c | 99 +++++++++++++++++++++++++++++++++++ 1 file changed, 99 insertions(+) diff --git a/sound/soc/sunxi/sun4i-spdif.c b/sound/soc/sunxi/sun4i-spdif.c index a10949bf0ca1..f7438e3ece65 100644 --- a/sound/soc/sunxi/sun4i-spdif.c +++ b/sound/soc/sunxi/sun4i-spdif.c @@ -21,6 +21,7 @@ #include <linux/platform_device.h> #include <linux/pm_runtime.h> #include <linux/reset.h> +#include <sound/asoundef.h> #include <sound/dmaengine_pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> @@ -385,11 +386,109 @@ static int sun4i_spdif_trigger(struct snd_pcm_substream *substream, int cmd, return ret; } +static int sun4i_spdif_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958; + uinfo->count = 1; + + return 0; +} + +static int sun4i_spdif_get_status_mask(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + u8 *status = ucontrol->value.iec958.status; + + status[0] = 0xff; + status[1] = 0xff; + status[2] = 0xff; + status[3] = 0xff; + status[4] = 0xff; + status[5] = 0x03; + + return 0; +} + +static int sun4i_spdif_get_status(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol); + struct sun4i_spdif_dev *host = snd_soc_dai_get_drvdata(cpu_dai); + u8 *status = ucontrol->value.iec958.status; + unsigned int reg; + + regmap_read(host->regmap, SUN4I_SPDIF_TXCHSTA0, &reg); + + status[0] = reg & 0xff; + status[1] = (reg >> 8) & 0xff; + status[2] = (reg >> 16) & 0xff; + status[3] = (reg >> 24) & 0xff; + + regmap_read(host->regmap, SUN4I_SPDIF_TXCHSTA1, &reg); + + status[4] = reg & 0xff; + status[5] = (reg >> 8) & 0x3; + + return 0; +} + +static int sun4i_spdif_set_status(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol); + struct sun4i_spdif_dev *host = snd_soc_dai_get_drvdata(cpu_dai); + u8 *status = ucontrol->value.iec958.status; + unsigned int reg; + + reg = (u32)status[3] << 24; + reg |= (u32)status[2] << 16; + reg |= (u32)status[1] << 8; + reg |= (u32)status[0]; + + regmap_write(host->regmap, SUN4I_SPDIF_TXCHSTA0, reg); + + reg = (u32)(status[5] & 3) << 8; + reg |= (u32)status[4]; + + regmap_write(host->regmap, SUN4I_SPDIF_TXCHSTA1, reg); + + reg = SUN4I_SPDIF_TXCFG_CHSTMODE; + if (status[0] & IEC958_AES0_NONAUDIO) + reg |= SUN4I_SPDIF_TXCFG_NONAUDIO; + + regmap_update_bits(host->regmap, SUN4I_SPDIF_TXCFG, + SUN4I_SPDIF_TXCFG_CHSTMODE | + SUN4I_SPDIF_TXCFG_NONAUDIO, reg); + + return 0; +} + +static struct snd_kcontrol_new sun4i_spdif_controls[] = { + { + .access = SNDRV_CTL_ELEM_ACCESS_READ, + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, MASK), + .info = sun4i_spdif_info, + .get = sun4i_spdif_get_status_mask + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, DEFAULT), + .info = sun4i_spdif_info, + .get = sun4i_spdif_get_status, + .put = sun4i_spdif_set_status + } +}; + static int sun4i_spdif_soc_dai_probe(struct snd_soc_dai *dai) { struct sun4i_spdif_dev *host = snd_soc_dai_get_drvdata(dai); snd_soc_dai_init_dma_data(dai, &host->dma_params_tx, NULL); + snd_soc_add_dai_controls(dai, sun4i_spdif_controls, + ARRAY_SIZE(sun4i_spdif_controls)); + return 0; } -- 2.33.1

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[PATCH v5 1/2] dt-bindings: sound: tlv320adc3xxx: New codec driver
by Ricard Wanderlof 15 Nov '21

15 Nov '21

DT bindings for tlv320adc3xxx driver, currently supporting Texas Instruments TLV320ADC3001 and TLV320ADC3101 audio ADCs. Signed-off-by: Ricard Wanderlof <ricardw(a)axis.com> --- .../bindings/sound/ti,tlv320adc3xxx.yaml | 137 ++++++++++++++++++ include/dt-bindings/sound/tlv320adc3xxx.h | 28 ++++ 2 files changed, 165 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ti,tlv320adc3xxx.yaml create mode 100644 include/dt-bindings/sound/tlv320adc3xxx.h diff --git a/Documentation/devicetree/bindings/sound/ti,tlv320adc3xxx.yaml b/Documentation/devicetree/bindings/sound/ti,tlv320adc3xxx.yaml new file mode 100644 index 000000000000..c4fed6335230 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,tlv320adc3xxx.yaml @@ -0,0 +1,137 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ti,tlv320adc3xxx.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Texas Instruments TLV320ADC3001/TLV320ADC3101 Stereo ADC + +maintainers: + - Ricard Wanderlof <ricardw(a)axis.com> + +description: | + Texas Instruments TLV320ADC3001 and TLV320ADC3101 Stereo ADC + https://www.ti.com/product/TLV320ADC3001 + https://www.ti.com/product/TLV320ADC3101 + +properties: + compatible: + enum: + - ti,tlv320adc3001 + - ti,tlv320adc3101 + + reg: + maxItems: 1 + description: I2C address + + '#sound-dai-cells': + const: 0 + + '#gpio-cells': + const: 2 + + gpio-controller: true + + reset-gpios: + maxItems: 1 + description: GPIO pin used for codec reset (RESET pin) + + clocks: + maxItems: 1 + description: Master clock (MCLK) + + ti,dmdin-gpio1: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # ADC3XXX_GPIO_DISABLED - I/O buffers powered down and not used + - 1 # ADC3XXX_GPIO_INPUT - Various non-GPIO input functions + - 2 # ADC3XXX_GPIO_GPI - General purpose input + - 3 # ADC3XXX_GPIO_GPO - General purpose output + - 4 # ADC3XXX_GPIO_CLKOUT - Clock source set in CLKOUT_MUX reg + - 5 # ADC3XXX_GPIO_INT1 - INT1 output + - 6 # ADC3XXX_GPIO_SECONDARY_BCLK - Codec interface secondary BCLK + - 7 # ADC3XXX_GPIO_SECONDARY_WCLK - Codec interface secondary WCLK + default: 0 + description: | + Configuration for DMDIN/GPIO1 pin. + + When ADC3XXX_GPIO_GPO is configured, this causes corresponding the + ALSA control "GPIOx Output" to appear, as a switch control. + + ti,dmclk-gpio2: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # ADC3XXX_GPIO_DISABLED - I/O buffers powered down and not used + - 1 # ADC3XXX_GPIO_INPUT - Various non-GPIO input functions + - 2 # ADC3XXX_GPIO_GPI - General purpose input + - 3 # ADC3XXX_GPIO_GPO - General purpose output + - 4 # ADC3XXX_GPIO_CLKOUT - Clock source set in CLKOUT_MUX reg + - 5 # ADC3XXX_GPIO_INT1 - INT1 output + - 6 # ADC3XXX_GPIO_SECONDARY_BCLK - Codec interface secondary BCLK + - 7 # ADC3XXX_GPIO_SECONDARY_WCLK - Codec interface secondary WCLK + default: 0 + description: | + Configuration for DMCLK/GPIO2 pin. + + When ADC3XXX_GPIO_GPO is configured, this causes corresponding the + ALSA control "GPIOx Output" to appear, as a switch control. + + Note that there is currently no support for reading the GPIO pins as + inputs. + + ti,micbias1-vg: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # ADC3XXX_MICBIAS_OFF - Mic bias is powered down + - 1 # ADC3XXX_MICBIAS_2_0V - Mic bias is set to 2.0V + - 2 # ADC3XXX_MICBIAS_2_5V - Mic bias is set to 2.5V + - 3 # ADC3XXX_MICBIAS_AVDD - Mic bias is same as AVDD supply + default: 0 + description: | + Mic bias voltage output on MICBIAS1 pin + + ti,micbias2-vg: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # ADC3XXX_MICBIAS_OFF - Mic bias is powered down + - 1 # ADC3XXX_MICBIAS_2_0V - Mic bias is set to 2.0V + - 2 # ADC3XXX_MICBIAS_2_5V - Mic bias is set to 2.5V + - 3 # ADC3XXX_MICBIAS_AVDD - Mic bias is same as AVDD supply + default: 0 + description: | + Mic bias voltage output on MICBIAS2 pin + +required: + - compatible + - reg + - clocks + +additionalProperties: false + +examples: + - | + + #include <dt-bindings/gpio/gpio.h> + #include <dt-bindings/sound/tlv320adc3xxx.h> + + i2c { + #address-cells = <1>; + #size-cells = <0>; + tlv320adc3101: tlv320adc3101@18 { + compatible = "ti,tlv320adc3101"; + reg = <0x18>; + reset-gpios = <&gpio_pc 3 GPIO_ACTIVE_LOW>; + clocks = <&audio_mclk>; + gpio-controller; + #gpio-cells = <2>; + ti,dmdin-gpio1 = <ADC3XXX_GPIO_GPO>; + ti,micbias1-vg = <ADC3XXX_MICBIAS_AVDD>; + }; + }; + + audio_mclk: clock { + compatible = "fixed-clock"; + #clock-cells = <0>; + clock-frequency = <24576000>; + }; +... diff --git a/include/dt-bindings/sound/tlv320adc3xxx.h b/include/dt-bindings/sound/tlv320adc3xxx.h new file mode 100644 index 000000000000..59ed510e6cf1 --- /dev/null +++ b/include/dt-bindings/sound/tlv320adc3xxx.h @@ -0,0 +1,28 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * Devicetree bindings definitions for tlv320adc3xxx driver. + * + * Copyright (C) 2021 Axis Communications AB + */ +#ifndef __DT_TLV320ADC3XXX_H +#define __DT_TLV320ADC3XXX_H + +#define ADC3XXX_GPIO_DISABLED 0 /* I/O buffers powered down */ +#define ADC3XXX_GPIO_INPUT 1 /* Various non-GPIO inputs */ +#define ADC3XXX_GPIO_GPI 2 /* General purpose input */ +#define ADC3XXX_GPIO_GPO 3 /* General purpose output */ +#define ADC3XXX_GPIO_CLKOUT 4 /* Source set in reg. CLKOUT_MUX */ +#define ADC3XXX_GPIO_INT1 5 /* INT1 output */ +#define ADC3XXX_GPIO_INT2 6 /* INT2 output */ +/* value 7 is reserved */ +#define ADC3XXX_GPIO_SECONDARY_BCLK 8 /* Codec interface secondary BCLK */ +#define ADC3XXX_GPIO_SECONDARY_WCLK 9 /* Codec interface secondary WCLK */ +#define ADC3XXX_GPIO_ADC_MOD_CLK 10 /* Clock output for digital mics */ +/* values 11-15 reserved */ + +#define ADC3XXX_MICBIAS_OFF 0 /* Micbias pin powered off */ +#define ADC3XXX_MICBIAS_2_0V 1 /* Micbias pin set to 2.0V */ +#define ADC3XXX_MICBIAS_2_5V 2 /* Micbias pin set to 2.5V */ +#define ADC3XXX_MICBIAS_AVDD 3 /* Use AVDD voltage for micbias pin */ + +#endif /* __DT_TLV320ADC3XXX_H */ -- 2.20.1 -- Ricard Wolf Wanderlof ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

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[PATCH v5 0/2] Codec driver for TLV320ADC3001 and 3101
by Ricard Wanderlof 15 Nov '21

15 Nov '21

This is a codec driver for the TLV320ADC3001 and TLV320ADC3101 2 channel audio ADC chips from Texas Instruments. Based on an old driver for these chips that I was referred to by Texas Instruments, but which apparently was never upstreamed, I've taken patches from various incarnations, ported them and augmented them with various new features in order to make a working driver. There is currently no support for the on-chip "miniDSP". Although the TLV320ADC3001 and TLV320ADC3101 have similarities with other TI codec chips for which there are existing drivers, upon closer inspection I decided the differences were too great to warrant attempting to modify an existing driver, especially without access to all supported chips and resources for testing them all. There are certainly things that could be improved, among others in the area of PLL register value calculations, but it has been tested on the Axis ARTPEC-8 platform and has been found to work satisfactorily at least with a TLV320ADC3101 operating as an I2S slave at 48 kHz with 32 bit bit depth, 1 or 2 channels, both with the on-chip PLL enabled and disabled. I have not tested this on a TLV320ADC3001 chip; in the original driver the only differences were related to the now-removed initialization code, so I've just retained the compatible strings for both chips. Limitations: - GPIO pins on chip can only be set up as outputs. - Using BCLK as an input to clock chain is supported by the hw but not by driver. Changes v4->v5: Yet another fix based on kernel test robot (lkp(a)intel.com), this time a legitimate issue: the return value from of_get_named_gpio_flags() was crammed into an unsigned int so any potential negative error codes were lost. No changes to the devicetree bindings. Changes v3->v4: More fixes reported by kernel test robot (lkp(a)intel.com). Don't know why these weren't reported the first time. Changes v2->v3: Fix a couple of issues reported by the kernel test robot (lkp(a)intel.com). Changes v1->v2: Large portions of the driver have been reworked from v1 based on remarks from Mark. More specifically: - Remove an unneeded #ifdef and EXPORT_SYMBOL_GPL. - Add ADC3XXX_ prefix to all macro names to avoid namespace contention. - Clean up certain macro names with mixed lower case - upper case names. - Do away with the D0..D7 macros representing bit values. - Fix general formatting: spaces before and after {, }, replace multi spaces with tabs, maximum 2 blank lines anywhere, and other minor issues. - Page register does not need to be volatile. - Rather than use 0 / -6dB enum for input attenuator, use standard TLV. - The dither offset control contains reserved values so use VALUE_ENUM instead of straight ENUM. - Replace vector of SOC_ENUMs with individual variables. - Fix ALSA control names: - Use Volume at the end of all gain controls, and remove "Gain" where applicable. - Use "Switch" at the end of all switch controls, either adding it or changing "switch" into "Switch" as applicable. - Switches and Volumes related to capture should be set up as such, so add "Capture" to widget names where appropriate. This is mostly important for the PGA Capture and Switch widgets, not just for consistency, but also because amixer simple Switch controls expect different names (cap/nocap instead of on/off) for control values for capture. - Make widget names have capital first letters for all consitutent words. - When resuming, the registers will not have changed, so no need to call snd_soc_component_cache_sync(). - Move reset gpio init to I2C probe - Remove needless _priv at end of private struct name. - Make hex constants lower case - _CBM_CFM/_CBS_CFS -> _CBP_CFP/_CBC_CFC - Instead of using device specific ALSA controls to control the two GPIO pins, use GPIOLIB. GPIO pin functions still can be set up in the devicetree as the pins can be used for other functions which are not GPIO related. - Limitation: The GPIO pins can only be used as outputs, even though the hardware alternatively supports using them as inputs. - Handle GPIO configs in a vector rather than separately. - Use more appropriate module_i2c_driver instead of generic module_init/module_exit. - Remove useless initialization data. - As the ADC mute function had no control, add it. - Clean up GPIO initialization which was done both in the I2C probe and in the component probe. - Remove probe, remove, suspend and resume callbacks, which either do not do anything at all anymore, or uselessly just do set_bias_level. - The PLL should not be configured in the devicetree, so move the corresponding definitions to tlv320adc3xxx.h where they can be picked up by a machine driver if needed. - Default to PLL AUTO mode. - Use "reset-gpios" instead of too-generic "gpios" for dt property for configuring codec reset pin. - Use function for printing PLL mode rather than macro. - dev_info is too verbose for most of the printouts, change them to dev_dbg. - Add support for configuring micbias voltage in DT. - Remove ALSA control for mic bias. - Move PLL and NADC, MADC divisor enable to DAPM, including 10 ms post PLL start up delay via POST_PMU event. - ADC is powered on using DAPM, so needless to do it in set_bias_level, thus remove it. - Let DAPM handle BCLK divisor enable. With this, we can remove set_bias_level as it doesn't do anything anymore. - Fix incorrect argument order to a snd_soc_component_update_bits call. - Although the chip itself can operate using BCLK as the clock source, either directly to the divider chain or via the PLL, this is not yet supported in the driver, so require the clocks property for the MCLK for now. - Move PLL rate structure out of .h file where it does not belong. /Ricard Ricard Wanderlof (2): dt-bindings: sound: tlv320adc3xxx: New codec driver ASoC: codec: tlv320adc3xxx: New codec driver .../bindings/sound/ti,tlv320adc3xxx.yaml | 137 +++ include/dt-bindings/sound/tlv320adc3xxx.h | 28 + sound/soc/codecs/Kconfig | 7 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tlv320adc3xxx.c | 1072 +++++++++++++++++ sound/soc/codecs/tlv320adc3xxx.h | 379 ++++++ 6 files changed, 1625 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ti,tlv320adc3xxx.yaml create mode 100644 include/dt-bindings/sound/tlv320adc3xxx.h create mode 100644 sound/soc/codecs/tlv320adc3xxx.c create mode 100644 sound/soc/codecs/tlv320adc3xxx.h -- 2.20.1 -- Ricard Wolf Wanderlof ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

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