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[PATCH v2] ALSA: ASoc/tas2781: Fix wrong loading calibrated data sequence
by Shenghao Ding 01 May '24

01 May '24

Calibrated data will be set to default after loading DSP config params, which will cause speaker protection work abnormally. Reload calibrated data after loading DSP config params. 'Fixes: 0a0877812628 ("ASoc: tas2781: Fix spelling mistake "calibraiton" -> "calibration"")' Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v2: - In the Subject, fixed --> Fix - In tas2781-fmwlib.c, tasdevice-fmw.c ---> tas2781-fmwlib.c - dsp --> DSP - Remove unneeded parentheses for & (dereference) operator - Add Fixes tag - Changed the copyright year to 2024 in the related files - In tas2781-dsp.h, __TASDEVICE_DSP_H__ --> __TAS2781_DSP_H__ v1: - Download calibrated data after loading the new DSP config params - call tasdevice_prmg_load instead of tasdevice_prmg_calibdata_load, it is unnecessary to load calibrated data after loading DSP program. Load it after loading DSP config params each time. - Remove tasdevice_prmg_calibdata_load, because it is unnecessary to load calibrated data after loading DSP program. --- include/sound/tas2781-dsp.h | 7 +-- sound/soc/codecs/tas2781-fmwlib.c | 99 +++++++------------------------ sound/soc/codecs/tas2781-i2c.c | 4 +- 3 files changed, 28 insertions(+), 82 deletions(-) diff --git a/include/sound/tas2781-dsp.h b/include/sound/tas2781-dsp.h index ea9af2726a53..7fba7ea26a4b 100644 --- a/include/sound/tas2781-dsp.h +++ b/include/sound/tas2781-dsp.h @@ -2,7 +2,7 @@ // // ALSA SoC Texas Instruments TAS2781 Audio Smart Amplifier // -// Copyright (C) 2022 - 2023 Texas Instruments Incorporated +// Copyright (C) 2022 - 2024 Texas Instruments Incorporated // https://www.ti.com // // The TAS2781 driver implements a flexible and configurable @@ -13,8 +13,8 @@ // Author: Kevin Lu <kevin-lu(a)ti.com> // -#ifndef __TASDEVICE_DSP_H__ -#define __TASDEVICE_DSP_H__ +#ifndef __TAS2781_DSP_H__ +#define __TAS2781_DSP_H__ #define MAIN_ALL_DEVICES 0x0d #define MAIN_DEVICE_A 0x01 @@ -180,7 +180,6 @@ void tasdevice_calbin_remove(void *context); int tasdevice_select_tuningprm_cfg(void *context, int prm, int cfg_no, int rca_conf_no); int tasdevice_prmg_load(void *context, int prm_no); -int tasdevice_prmg_calibdata_load(void *context, int prm_no); void tasdevice_tuning_switch(void *context, int state); int tas2781_load_calibration(void *context, char *file_name, unsigned short i); diff --git a/sound/soc/codecs/tas2781-fmwlib.c b/sound/soc/codecs/tas2781-fmwlib.c index 45760fe19523..5a4095c6a954 100644 --- a/sound/soc/codecs/tas2781-fmwlib.c +++ b/sound/soc/codecs/tas2781-fmwlib.c @@ -2,7 +2,7 @@ // // tasdevice-fmw.c -- TASDEVICE firmware support // -// Copyright 2023 Texas Instruments, Inc. +// Copyright 2023 - 2024 Texas Instruments, Inc. // // Author: Shenghao Ding <shenghao-ding(a)ti.com> @@ -2151,6 +2151,20 @@ static int tasdevice_load_data(struct tasdevice_priv *tas_priv, return ret; } +static void tasdev_load_calibrated_data(struct tasdevice_priv *priv, + int i) +{ + struct tasdevice_fw *cal_fmw = priv->tasdevice[i].cali_data_fmw; + + if (cal_fmw) { + struct tasdevice_calibration *cal = cal_fmw->calibrations; + + if (cal) + load_calib_data(priv, &cal->dev_data); + return; + } +} + int tasdevice_select_tuningprm_cfg(void *context, int prm_no, int cfg_no, int rca_conf_no) { @@ -2210,21 +2224,9 @@ int tasdevice_select_tuningprm_cfg(void *context, int prm_no, for (i = 0; i < tas_priv->ndev; i++) { if (tas_priv->tasdevice[i].is_loaderr == true) continue; - else if (tas_priv->tasdevice[i].is_loaderr == false - && tas_priv->tasdevice[i].is_loading == true) { - struct tasdevice_fw *cal_fmw = - tas_priv->tasdevice[i].cali_data_fmw; - - if (cal_fmw) { - struct tasdevice_calibration - *cal = cal_fmw->calibrations; - - if (cal) - load_calib_data(tas_priv, - &(cal->dev_data)); - } + if (tas_priv->tasdevice[i].is_loaderr == false + && tas_priv->tasdevice[i].is_loading == true) tas_priv->tasdevice[i].cur_prog = prm_no; - } } } @@ -2247,9 +2249,13 @@ int tasdevice_select_tuningprm_cfg(void *context, int prm_no, if (tas_priv->tasdevice[i].is_loaderr == true) { status |= 1 << (i + 4); continue; - } else if (tas_priv->tasdevice[i].is_loaderr == false - && tas_priv->tasdevice[i].is_loading == true) + } + + if (tas_priv->tasdevice[i].is_loaderr == false + && tas_priv->tasdevice[i].is_loading == true) { + tasdev_load_calibrated_data(tas_priv, i); tas_priv->tasdevice[i].cur_conf = cfg_no; + } } } else dev_dbg(tas_priv->dev, "%s: Unneeded loading dsp conf %d\n", @@ -2308,65 +2314,6 @@ int tasdevice_prmg_load(void *context, int prm_no) } EXPORT_SYMBOL_NS_GPL(tasdevice_prmg_load, SND_SOC_TAS2781_FMWLIB); -int tasdevice_prmg_calibdata_load(void *context, int prm_no) -{ - struct tasdevice_priv *tas_priv = (struct tasdevice_priv *) context; - struct tasdevice_fw *tas_fmw = tas_priv->fmw; - struct tasdevice_prog *program; - int prog_status = 0; - int i; - - if (!tas_fmw) { - dev_err(tas_priv->dev, "%s: Firmware is NULL\n", __func__); - goto out; - } - - if (prm_no >= tas_fmw->nr_programs) { - dev_err(tas_priv->dev, - "%s: prm(%d) is not in range of Programs %u\n", - __func__, prm_no, tas_fmw->nr_programs); - goto out; - } - - for (i = 0, prog_status = 0; i < tas_priv->ndev; i++) { - if (prm_no >= 0 && tas_priv->tasdevice[i].cur_prog != prm_no) { - tas_priv->tasdevice[i].cur_conf = -1; - tas_priv->tasdevice[i].is_loading = true; - prog_status++; - } - tas_priv->tasdevice[i].is_loaderr = false; - } - - if (prog_status) { - program = &(tas_fmw->programs[prm_no]); - tasdevice_load_data(tas_priv, &(program->dev_data)); - for (i = 0; i < tas_priv->ndev; i++) { - if (tas_priv->tasdevice[i].is_loaderr == true) - continue; - else if (tas_priv->tasdevice[i].is_loaderr == false - && tas_priv->tasdevice[i].is_loading == true) { - struct tasdevice_fw *cal_fmw = - tas_priv->tasdevice[i].cali_data_fmw; - - if (cal_fmw) { - struct tasdevice_calibration *cal = - cal_fmw->calibrations; - - if (cal) - load_calib_data(tas_priv, - &(cal->dev_data)); - } - tas_priv->tasdevice[i].cur_prog = prm_no; - } - } - } - -out: - return prog_status; -} -EXPORT_SYMBOL_NS_GPL(tasdevice_prmg_calibdata_load, - SND_SOC_TAS2781_FMWLIB); - void tasdevice_tuning_switch(void *context, int state) { struct tasdevice_priv *tas_priv = (struct tasdevice_priv *) context; diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index b5abff230e43..9350972dfefe 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -2,7 +2,7 @@ // // ALSA SoC Texas Instruments TAS2563/TAS2781 Audio Smart Amplifier // -// Copyright (C) 2022 - 2023 Texas Instruments Incorporated +// Copyright (C) 2022 - 2024 Texas Instruments Incorporated // https://www.ti.com // // The TAS2563/TAS2781 driver implements a flexible and configurable @@ -414,7 +414,7 @@ static void tasdevice_fw_ready(const struct firmware *fmw, __func__, tas_priv->cal_binaryname[i]); } - tasdevice_prmg_calibdata_load(tas_priv, 0); + tasdevice_prmg_load(tas_priv, 0); tas_priv->cur_prog = 0; out: if (tas_priv->fw_state == TASDEVICE_DSP_FW_FAIL) { -- 2.34.1

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[PATCH 0/5] ASoc: SOF: misc updates for 6.10
by Pierre-Louis Bossart 01 May '24

01 May '24

Fixes when fw_lib_prefix is not set, updated error messages, improved dmesg logs to SoundWire configurations not supported by ACPI tables/topology and support for IEC61937 passthrough. Peter Ujfalusi (2): ASoC: SOF: debug: Handle cases when fw_lib_prefix is not set, NULL ASoC: SOF: ipc4-topology: Advertise passthrough capable PCMs (using ChainDMA) Pierre-Louis Bossart (2): ASoC: SOF: Intel: hda: simplify and reduce indentation for hda_sdw_machine_select() ASoC: SOF: Intel: hda: list SoundWire peripherals on mismatch Yong Zhi (1): ASoC: SOF: topology: remove incorrect widget id in error message sound/soc/sof/debug.c | 23 ++++- sound/soc/sof/intel/hda.c | 154 +++++++++++++++++++--------------- sound/soc/sof/ipc4-topology.c | 50 +++++++++++ sound/soc/sof/topology.c | 7 +- 4 files changed, 160 insertions(+), 74 deletions(-) -- 2.40.1

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[PATCH 0/4] ASoC: meson: tdm fixes
by Jerome Brunet 01 May '24

01 May '24

This patchset fixes 2 problems on TDM which both find a solution by properly implementing the .trigger() callback for the TDM backend. ATM, enabling the TDM formatters is done by the .prepare() callback because handling the formatter is slow due to necessary calls to CCF. The first problem affects the TDMIN. Because .prepare() is called on DPCM backend first, the formatter are started before the FIFOs and this may cause a random channel shifts if the TDMIN use multiple lanes with more than 2 slots per lanes. Using trigger() allows to set the FE/BE order, solving the problem. There has already been an attempt to fix this 3y ago [1] and reverted [2] It triggered a 'sleep in irq' error on the period IRQ. The solution is to just use the bottom half of threaded IRQ. This is patch #1. Patch #2 and #3 remain mostly the same as 3y ago. For TDMOUT, the problem is on pause. ATM pause only stops the FIFO and the TDMOUT just starves. When it does, it will actually repeat the last sample continuously. Depending on the platform, if there is no high-pass filter on the analog path, this may translate to a constant position of the speaker membrane. There is no audible glitch but it may damage the speaker coil. Properly stopping the TDMOUT in pause solves the problem. There is behaviour change associated with that fix. Clocks used to be continuous on pause because of the problem above. They will now be gated on pause by default, as they should. The last change introduce the proper support for continuous clocks, if needed. [1]: https://lore.kernel.org/linux-amlogic/20211020114217.133153-1-jbrunet@bayli… [2]: https://lore.kernel.org/linux-amlogic/20220421155725.2589089-1-narmstrong@b… Jerome Brunet (4): ASoC: meson: axg-fifo: use threaded irq to check periods ASoC: meson: axg-card: make links nonatomic ASoC: meson: axg-tdm-interface: manage formatters in trigger ASoC: meson: axg-tdm: add continuous clock support sound/soc/meson/axg-card.c | 1 + sound/soc/meson/axg-fifo.c | 29 +++++++++++++-------- sound/soc/meson/axg-tdm-formatter.c | 40 +++++++++++++++++++++++++++++ sound/soc/meson/axg-tdm-interface.c | 38 +++++++++++++++++++-------- sound/soc/meson/axg-tdm.h | 5 ++++ 5 files changed, 93 insertions(+), 20 deletions(-) -- 2.43.0

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[PATCH 00/23] ASoC: Intel: updates for 6.10 - part6
by Pierre-Louis Bossart 01 May '24

01 May '24

Last batch of cleanups from Brent Lu, with Chromebooks now supported with fewer modular machine drivers. Bard Liao (1): ASoC: Intel: sof_rt5682: add missing MAX98357A config Brent Lu (22): ASoC: Intel: nau8825/rt5682: move speaker widget to common modules ASoC: Intel: sof_maxim_common: support MAX98390 on cml boards ASoC: Intel: sof_da7219: support MAX98390 ASoC: Intel: sof_da7219: add cml_da7219_def for cml boards ASoC: Intel: sof_da7219: support MAX98390 on cml boards ASoC: Intel: bxt_da7219_max98357a: remove MAX98390 support ASoC: Intel: sof_da7219: support MAX98357A ASoC: Intel: sof_da7219: support MAX98357A on cml boards ASoC: Intel: bxt_da7219_max98357a: remove cml support ASoC: Intel: sof_da7219: add glk_da7219_def for glk boards ASoC: Intel: sof_da7219: support MAX98357A on glk boards ASoC: Intel: bxt_da7219_max98357a: remove glk support ASoC: Intel: sof_rt5682: add glk_rt5682_def for glk boards ASoC: Intel: sof_rt5682: support MAX98357A on glk boards ASoC: Intel: glk_rt5682_max98357a: delete driver ASoC: Intel: sof-rt5682: add mclk_en to sof_rt5682_private ASoC: Intel: sof-rt5682: remove SOF_RT5682_MCLK_BYTCHT_EN ASoC: Intel: sof_rt5682: add icl_rt5682_def for icl boards ASoC: Intel: sof-rt5682: add driver_data to sof_rt5682 board ASoC: Intel: sof-rt5682: setup pll_id only when needed ASoC: Intel: sof-rt5682: get bclk frequency from topology ASoC: Intel: sof-rt5682: support bclk as PLL source on rt5682s sound/soc/intel/boards/Kconfig | 41 +- sound/soc/intel/boards/Makefile | 4 +- sound/soc/intel/boards/bxt_da7219_max98357a.c | 179 +---- sound/soc/intel/boards/glk_rt5682_max98357a.c | 693 ------------------ sound/soc/intel/boards/sof_board_helpers.h | 2 + sound/soc/intel/boards/sof_da7219.c | 110 ++- sound/soc/intel/boards/sof_maxim_common.c | 108 ++- sound/soc/intel/boards/sof_nau8825.c | 4 - sound/soc/intel/boards/sof_realtek_common.c | 93 ++- sound/soc/intel/boards/sof_rt5682.c | 233 +++--- .../intel/common/soc-acpi-intel-cml-match.c | 4 +- .../intel/common/soc-acpi-intel-glk-match.c | 4 +- .../intel/common/soc-acpi-intel-icl-match.c | 2 +- 13 files changed, 461 insertions(+), 1016 deletions(-) delete mode 100644 sound/soc/intel/boards/glk_rt5682_max98357a.c -- 2.40.1

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[PATCH 00/12] ASoC: Intel: updates for 6.10 - part5
by Pierre-Louis Bossart 01 May '24

01 May '24

This patchset corrects a couple of mistakes corrected, improves snd_soc_card allocation. The new functionality is mostly for SoundWire platforms, with new SKUs for Dell and Acer, and support for the Cirrus Logic bridge/sidecar amplifier topology. Bard Liao (1): ASoC: Intel: sof_sdw: add a space before cfg-amp in components Charles Keepax (2): ASoC: Intel: sof_sdw: Delay update of the codec_conf array ASoC: Intel: sof_sdw: Add callbacks to register sidecar devices Mac Chiang (1): ASoC: Intel: soc-acpi-intel-lnl-match: adds RT714 and RT1318 support Maciej Strozek (1): ASoC: intel: sof_sdw: Add support for cs42l43-cs35l56 sidecar amps Peter Ujfalusi (3): ASoC: Intel: skl_hda_dsp_generic: Allocate snd_soc_card dynamically ASoC: Intel: skl_hda_dsp_generic: Use devm_kasprintf for the components string ASoC: Intel: sof_sdw: Allocate snd_soc_card dynamically Pierre-Louis Bossart (4): ASoC: Intel: soc-acpi: mtl: add Dell SKU 0C64 and 0CC6 ASoC: Intel: soc-acpi: mtl: add support for Acer Swift Go 14 ASoC: Intel: sof-sdw: don't set card long_name ASoC: Intel: sof-sdw: really remove FOUR_SPEAKER quirk sound/soc/intel/boards/Kconfig | 1 + sound/soc/intel/boards/Makefile | 1 + sound/soc/intel/boards/bridge_cs35l56.c | 137 ++++++++++++++++++ sound/soc/intel/boards/skl_hda_dsp_common.h | 1 + sound/soc/intel/boards/skl_hda_dsp_generic.c | 50 +++---- sound/soc/intel/boards/sof_sdw.c | 106 ++++++++------ sound/soc/intel/boards/sof_sdw_common.h | 27 ++++ sound/soc/intel/boards/sof_sdw_cs42l43.c | 14 +- .../intel/common/soc-acpi-intel-lnl-match.c | 52 +++++++ .../intel/common/soc-acpi-intel-mtl-match.c | 83 ++++++++++- 10 files changed, 397 insertions(+), 75 deletions(-) create mode 100644 sound/soc/intel/boards/bridge_cs35l56.c -- 2.40.1

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[PATCH 0/8] ASoC: Intel: avs: PCM code cleanup
by Cezary Rojewski 01 May '24

01 May '24

A set of changes that aims to improve readability of cohesiveness of the pcm code for the avs-driver. Start off with a change that synchronizes DAI open/close - DAIs are started up in ascending order yet their shutdown does not follow the scheme - it is done in the ascending order too, rather than desceding one. This patch is a dependency for the next one in line. To align the HDAudio DAI startup/shutdown with the non-HDAudio equivalents, relocate the code from component to DAI. The reason above is a dependency stems from codec driver requirements - HDAudio code found in sound/pci/hda/ expects substream->runtime->private_data to point to a valid stream (HOST) pointer. With the hard part done, the follow up changes update the existing code to reduce it is complexity - removal of duplicates, renaming of ambiguous functions and adding new fields to DAI-data object so that the number of local variables and casts is reduced. Cezary Rojewski (8): ASoC: pcm: Reverse iterate DAIs when shutting them down ASoC: Intel: avs: Relocate HDA BE DAI specific operations ASoC: Intel: avs: Remove redundancy around DAI shutdown ASoC: Intel: avs: Store pointer to adev in DAI dma_data ASoC: Intel: avs: Remove redundancy around DAI startup ASoC: Intel: avs: Remove redundancy around DAI prepare ASoC: Intel: avs: Store pointer to link_stream in dma_data ASoC: Intel: avs: Clean up hw constraints initialization include/sound/soc.h | 4 + sound/soc/intel/avs/pcm.c | 250 +++++++++++++++++--------------------- sound/soc/soc-pcm.c | 2 +- 3 files changed, 117 insertions(+), 139 deletions(-) -- 2.25.1

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[PATCH 1/1] ALSA: aoa: soundbus: i2sbus: pcm: use 'time_left' variable with wait_for_completion_timeout()
by Wolfram Sang 01 May '24

01 May '24

There is a confusing pattern in the kernel to use a variable named 'timeout' to store the result of wait_for_completion_timeout() causing patterns like: timeout = wait_for_completion_timeout(...) if (!timeout) return -ETIMEDOUT; with all kinds of permutations. Use 'time_left' as a variable to make the code self explaining. Fix to the proper variable type 'unsigned long' while here. Signed-off-by: Wolfram Sang <wsa+renesas(a)sang-engineering.com> --- sound/aoa/soundbus/i2sbus/pcm.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/aoa/soundbus/i2sbus/pcm.c b/sound/aoa/soundbus/i2sbus/pcm.c index 07df5cc0f2d7..98b812ffbde6 100644 --- a/sound/aoa/soundbus/i2sbus/pcm.c +++ b/sound/aoa/soundbus/i2sbus/pcm.c @@ -255,24 +255,24 @@ static void i2sbus_wait_for_stop(struct i2sbus_dev *i2sdev, { unsigned long flags; DECLARE_COMPLETION_ONSTACK(done); - long timeout; + unsigned long time_left; spin_lock_irqsave(&i2sdev->low_lock, flags); if (pi->dbdma_ring.stopping) { pi->stop_completion = &done; spin_unlock_irqrestore(&i2sdev->low_lock, flags); - timeout = wait_for_completion_timeout(&done, HZ); + time_left = wait_for_completion_timeout(&done, HZ); spin_lock_irqsave(&i2sdev->low_lock, flags); pi->stop_completion = NULL; - if (timeout == 0) { + if (time_left == 0) { /* timeout expired, stop dbdma forcefully */ printk(KERN_ERR "i2sbus_wait_for_stop: timed out\n"); /* make sure RUN, PAUSE and S0 bits are cleared */ out_le32(&pi->dbdma->control, (RUN | PAUSE | 1) << 16); pi->dbdma_ring.stopping = 0; - timeout = 10; + time_left = 10; while (in_le32(&pi->dbdma->status) & ACTIVE) { - if (--timeout <= 0) + if (--time_left <= 0) break; udelay(1); } -- 2.43.0

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Hacker/stalkerALERT
by GitHub issues - opened 01 May '24

01 May '24

alsa-project/alsa-tools issue #24 was opened from interstellar08: same problems no matter what i do. new mail ... everything new phone .. com ALL i cannot get rid of this!!! Issue URL : https://github.com/alsa-project/alsa-tools/issues/24 Repository URL: https://github.com/alsa-project/alsa-tools

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topology: Add SND_PCM_RATE_24000 for 24 kHz rate support
by GitHub pull_request - opened 30 Apr '24

30 Apr '24

alsa-project/alsa-lib pull request #396 was opened from yongzhi1: The Low Complexity Communication Codec (LC3) supports 24 kHz sample rate, so add SND_PCM_RATE_24000 to the supported rates. Request URL : https://github.com/alsa-project/alsa-lib/pull/396 Patch URL : https://github.com/alsa-project/alsa-lib/pull/396.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH v2] ASoC: sunxi: DMIC: Add controls for adjusting the mic gains
by Joao Schim 30 Apr '24

30 Apr '24

The AllWinner H6 and later SoCs that sport a DMIC block contain a set of registers to control the gain (left + right) of each of the four supported channels. Add ASoC controls for changing each of the stereo channel gains using alsamixer and alike Signed-off-by: Joao Schim <joao(a)schimsalabim.eu> --- sound/soc/sunxi/sun50i-dmic.c | 34 ++++++++++++++++++++++++++++++++++ 1 file changed, 34 insertions(+) diff --git a/sound/soc/sunxi/sun50i-dmic.c b/sound/soc/sunxi/sun50i-dmic.c index c76628bc86c6..dd32780fb6a4 100644 --- a/sound/soc/sunxi/sun50i-dmic.c +++ b/sound/soc/sunxi/sun50i-dmic.c @@ -14,6 +14,7 @@ #include <sound/dmaengine_pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> +#include <sound/tlv.h> #define SUN50I_DMIC_EN_CTL (0x00) #define SUN50I_DMIC_EN_CTL_GLOBE BIT(8) @@ -43,6 +44,17 @@ #define SUN50I_DMIC_CH_NUM_N_MASK GENMASK(2, 0) #define SUN50I_DMIC_CNT (0x2c) #define SUN50I_DMIC_CNT_N (1 << 0) +#define SUN50I_DMIC_D0D1_VOL_CTR (0x30) + #define SUN50I_DMIC_D0D1_VOL_CTR_0R (0) + #define SUN50I_DMIC_D0D1_VOL_CTR_0L (8) + #define SUN50I_DMIC_D0D1_VOL_CTR_1R (16) + #define SUN50I_DMIC_D0D1_VOL_CTR_1L (24) +#define SUN50I_DMIC_D2D3_VOL_CTR (0x34) + #define SUN50I_DMIC_D2D3_VOL_CTR_2R (0) + #define SUN50I_DMIC_D2D3_VOL_CTR_2L (8) + #define SUN50I_DMIC_D2D3_VOL_CTR_3R (16) + #define SUN50I_DMIC_D2D3_VOL_CTR_3L (24) + #define SUN50I_DMIC_HPF_CTRL (0x38) #define SUN50I_DMIC_VERSION (0x50) @@ -273,8 +285,30 @@ static const struct of_device_id sun50i_dmic_of_match[] = { }; MODULE_DEVICE_TABLE(of, sun50i_dmic_of_match); +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(sun50i_dmic_vol_scale, -12000, 75, 1); + +static const struct snd_kcontrol_new sun50i_dmic_controls[] = { + + SOC_DOUBLE_TLV("DMIC Channel 0 Capture Volume", SUN50I_DMIC_D0D1_VOL_CTR, + SUN50I_DMIC_D0D1_VOL_CTR_0L, SUN50I_DMIC_D0D1_VOL_CTR_0R, + 0xFF, 0, sun50i_dmic_vol_scale), + SOC_DOUBLE_TLV("DMIC Channel 1 Capture Volume", SUN50I_DMIC_D0D1_VOL_CTR, + SUN50I_DMIC_D0D1_VOL_CTR_1L, SUN50I_DMIC_D0D1_VOL_CTR_1R, + 0xFF, 0, sun50i_dmic_vol_scale), + SOC_DOUBLE_TLV("DMIC Channel 2 Capture Volume", SUN50I_DMIC_D2D3_VOL_CTR, + SUN50I_DMIC_D2D3_VOL_CTR_2L, SUN50I_DMIC_D2D3_VOL_CTR_2R, + 0xFF, 0, sun50i_dmic_vol_scale), + SOC_DOUBLE_TLV("DMIC Channel 3 Capture Volume", SUN50I_DMIC_D2D3_VOL_CTR, + SUN50I_DMIC_D2D3_VOL_CTR_3L, SUN50I_DMIC_D2D3_VOL_CTR_3R, + 0xFF, 0, sun50i_dmic_vol_scale), + + +}; + static const struct snd_soc_component_driver sun50i_dmic_component = { .name = "sun50i-dmic", + .controls = sun50i_dmic_controls, + .num_controls = ARRAY_SIZE(sun50i_dmic_controls), }; static int sun50i_dmic_runtime_suspend(struct device *dev) -- 2.25.1

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