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ADC Soundcard hack : Increase by 2 the number of channels for any adc soundcard
by Sylvain Grenier 27 Aug '25

27 Aug '25

Hello alsa team, To begin, thank you all the team for what you do, the alsa-project is a wonderfull open-source project! My goal is to increase the number of channels of my adc soundcard by separate once again a left-right for one channel. Using a small electronic scheme like this (it's a first simulation try, I have to test and improve), we can multiply the number of channels by 2. [image.png] Of course, the sampling frequency will be decreased by 2 using this method. Where can I begin with alsa to increase the number of channels and adjust the sampling frequency ? Something like a virtual soundcard maybe ? I have skills in dev but don't know where I have to start with alsa fot this hack... Best regards, Sylvain Grenier

1 0

Audio garbled when using Multi plugin to record from 2+ USB soundcards on Raspberry Pi
by GitHub issues - edited 26 Aug '25

26 Aug '25

alsa-project/alsa-lib issue #472 was edited from TommyKaneko: I have been trying to use the multi plugin to record multiple channels from multiple soundcards on the Raspberry Pi. The results are a garbled mess. ==== EDIT 26/08 ==== Having probed the issue some more, I would like to add more detail: The devices and drivers I try to use are: ```**** List of CAPTURE Hardware Devices **** card 0: Device [USB PnP Sound Device], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: Device_1 [USB PnP Sound Device], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 ``` ...Curiously, I have actually had success using the multi plugin to record using a pro audio interface AND ONE `USB PnP Sound Device`, in this set up ```**** List of CAPTURE Hardware Devices **** card 0: Device [USB PnP Sound Device], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: USB [Scarlett 2i2 USB], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 ``` HOWEVER, if I try to add another `USB PnP Sound Device` to the multi setup, then the result is a garbled mess. My conclusion is that the multi plugin works generally, but it does NOT work for two or more `USB PnP Sound Device`. Something tells me that the plugin cannot differentiate the addresses of the audio frames held in two or more of these cheap USB microphones. I have a more verbose post of my issue here: https://www.reddit.com/r/linuxaudio/comments/1m24po8/recording_on_multiple_… ===EDIT 26/08 The following diagnosis was wrong: I have diagnosed the reasons as thus: - current raspberry Pi OS uses alsa v1.2.8+rpt1 - the "fix" for the multi plugin was implemented in v1.2.9 : https://github.com/alsa-project/alsa-lib/commit/369df322577034ea872978d6470… I am aware that the multi plugin was not meant to be used on multiple soundcards that are not in sync, however if the clock drifts over time, this is ok for my application. I am here to seek advice on how I should go about creating a workaround. The limitations are that I need to use cheap USB microphones (it is a weight issue). Without going along the microcontroller route, I thought one way was to make an [external plugin](https://www.alsa-project.org/alsa-doc/alsa-lib/pcm_external_plugins…, based on the newly patched pcm_multi plugin. I am working on this, while climbing the steep learning curve. Perhaps I can contribute an external plugin for the project... In the meantime, does a developer have a better suggestion that I might have success with? Or any tips if you are familiar with the pcm_multi plugin? @perexg ? Issue URL : https://github.com/alsa-project/alsa-lib/issues/472 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Audio garbled when using Multi plugin to record from 2+ USB soundcards on Raspberry Pi
by GitHub issues - edited 26 Aug '25

26 Aug '25

alsa-project/alsa-lib issue #472 was edited from TommyKaneko: I have been trying to use the multi plugin to record multiple channels from multiple soundcards on the Raspberry Pi. The results are a garbled mess. ==== EDIT 26/08 ==== Having probed the issue some more, I would like to add more detail: The devices and drivers I try to use are: ```**** List of CAPTURE Hardware Devices **** card 0: Device [USB PnP Sound Device], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: Device_1 [USB PnP Sound Device], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 ``` ...Curiously, I have actually had success using the multi plugin to record using a pro audio interface AND ONE `USB PnP Sound Device`, in this set up ```**** List of CAPTURE Hardware Devices **** card 0: Device [USB PnP Sound Device], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: USB [Scarlett 2i2 USB], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0``` HOWEVER, if I try to add another `USB PnP Sound Device` to the multi setup, then the result is a garbled mess. My conclusion is that the multi plugin works generally, but it does NOT work for two or more `USB PnP Sound Device`. Something tells me that the plugin cannot differentiate the addresses of the audio frames held in two or more of these cheap USB microphones. I have a more verbose post of my issue here: https://www.reddit.com/r/linuxaudio/comments/1m24po8/recording_on_multiple_… ===EDIT 26/08 The following diagnosis was wrong: I have diagnosed the reasons as thus: - current raspberry Pi OS uses alsa v1.2.8+rpt1 - the "fix" for the multi plugin was implemented in v1.2.9 : https://github.com/alsa-project/alsa-lib/commit/369df322577034ea872978d6470… I am aware that the multi plugin was not meant to be used on multiple soundcards that are not in sync, however if the clock drifts over time, this is ok for my application. I am here to seek advice on how I should go about creating a workaround. The limitations are that I need to use cheap USB microphones (it is a weight issue). Without going along the microcontroller route, I thought one way was to make an [external plugin](https://www.alsa-project.org/alsa-doc/alsa-lib/pcm_external_plugins…, based on the newly patched pcm_multi plugin. I am working on this, while climbing the steep learning curve. Perhaps I can contribute an external plugin for the project... In the meantime, does a developer have a better suggestion that I might have success with? Or any tips if you are familiar with the pcm_multi plugin? @perexg ? Issue URL : https://github.com/alsa-project/alsa-lib/issues/472 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Audio garbled when using Multi plugin to record from 2+ USB soundcards on Raspberry Pi
by GitHub issues - edited 26 Aug '25

26 Aug '25

alsa-project/alsa-lib issue #472 was edited from TommyKaneko: I have been trying to use the multi plugin to record multiple channels from multiple soundcards on the Raspberry Pi. The results are a garbled mess. ==== EDIT 26/08 ==== Having probed the issue some more, I would like to add more detail: The devices and drivers I try to use are: `**** List of CAPTURE Hardware Devices **** card 0: Device [USB PnP Sound Device], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: Device_1 [USB PnP Sound Device], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 ` ...Curiously, I have actually had success using the multi plugin to record using a pro audio interface AND ONE `USB PnP Sound Device`, in this set up `**** List of CAPTURE Hardware Devices **** card 0: Device [USB PnP Sound Device], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: USB [Scarlett 2i2 USB], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0` HOWEVER, if I try to add another `USB PnP Sound Device` to the multi setup, then the result is a garbled mess. My conclusion is that the multi plugin works generally, but it does NOT work for two or more `USB PnP Sound Device`. Something tells me that the plugin cannot differentiate the addresses of the audio frames held in two or more of these cheap USB microphones. I have a more verbose post of my issue here: https://www.reddit.com/r/linuxaudio/comments/1m24po8/recording_on_multiple_… ===EDIT 26/08 The following diagnosis was wrong: I have diagnosed the reasons as thus: - current raspberry Pi OS uses alsa v1.2.8+rpt1 - the "fix" for the multi plugin was implemented in v1.2.9 : https://github.com/alsa-project/alsa-lib/commit/369df322577034ea872978d6470… I am aware that the multi plugin was not meant to be used on multiple soundcards that are not in sync, however if the clock drifts over time, this is ok for my application. I am here to seek advice on how I should go about creating a workaround. The limitations are that I need to use cheap USB microphones (it is a weight issue). Without going along the microcontroller route, I thought one way was to make an [external plugin](https://www.alsa-project.org/alsa-doc/alsa-lib/pcm_external_plugins…, based on the newly patched pcm_multi plugin. I am working on this, while climbing the steep learning curve. Perhaps I can contribute an external plugin for the project... In the meantime, does a developer have a better suggestion that I might have success with? Or any tips if you are familiar with the pcm_multi plugin? @perexg ? Issue URL : https://github.com/alsa-project/alsa-lib/issues/472 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH v2] ASoC: SOF: Intel: WCL: Add the sdw_process_wakeen op
by Ajye Huang 26 Aug '25

26 Aug '25

Add the missing op in the device description to avoid issues with jack detection. Fixes: 6b04629ae97a ("ASoC: SOF: Intel: add initial support for WCL") Signed-off-by: Ajye Huang <ajye_huang(a)compal.corp-partner.google.com> --- v2: Add Fixes: 6b04629ae97a ("ASoC: SOF: Intel: add initial support for WCL") sound/soc/sof/intel/ptl.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sof/intel/ptl.c b/sound/soc/sof/intel/ptl.c index 1bc1f54c470d..4633cd01e7dd 100644 --- a/sound/soc/sof/intel/ptl.c +++ b/sound/soc/sof/intel/ptl.c @@ -143,6 +143,7 @@ const struct sof_intel_dsp_desc wcl_chip_info = { .read_sdw_lcount = hda_sdw_check_lcount_ext, .check_sdw_irq = lnl_dsp_check_sdw_irq, .check_sdw_wakeen_irq = lnl_sdw_check_wakeen_irq, + .sdw_process_wakeen = hda_sdw_process_wakeen_common, .check_ipc_irq = mtl_dsp_check_ipc_irq, .cl_init = mtl_dsp_cl_init, .power_down_dsp = mtl_power_down_dsp, -- 2.25.1

1 1

[PATCH v1] ASoC: SOF: Intel: WCL: Add the sdw_process_wakeen op
by Ajye Huang 26 Aug '25

26 Aug '25

Add the missing op in the device description to avoid issues with jack detection. Signed-off-by: Ajye Huang <ajye_huang(a)compal.corp-partner.google.com> --- sound/soc/sof/intel/ptl.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sof/intel/ptl.c b/sound/soc/sof/intel/ptl.c index 1bc1f54c470d..4633cd01e7dd 100644 --- a/sound/soc/sof/intel/ptl.c +++ b/sound/soc/sof/intel/ptl.c @@ -143,6 +143,7 @@ const struct sof_intel_dsp_desc wcl_chip_info = { .read_sdw_lcount = hda_sdw_check_lcount_ext, .check_sdw_irq = lnl_dsp_check_sdw_irq, .check_sdw_wakeen_irq = lnl_sdw_check_wakeen_irq, + .sdw_process_wakeen = hda_sdw_process_wakeen_common, .check_ipc_irq = mtl_dsp_check_ipc_irq, .cl_init = mtl_dsp_cl_init, .power_down_dsp = mtl_power_down_dsp, -- 2.25.1

2 2

[PATCH v1] ALSA: hda/tas2781: Fix EFI name for calibration beginning with 1 instead of 0
by Shenghao Ding 26 Aug '25

26 Aug '25

A bug reported by one of my customers that EFI name beginning with 0 instead of 1, and code clean for the string checking. Fixes: 4fe238513407 ("ALSA: hda/tas2781: Move and unified the calibrated-data getting function for SPI and I2C into the tas2781_hda lib") Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v1: - Fix EFI name beginning with 1 instead of 0 - Code clean for the string checking - Add extra comments on EFI name for calibration - Remove an extra space --- sound/hda/codecs/side-codecs/tas2781_hda_i2c.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c b/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c index ed7771ab9475..fecd5eac739b 100644 --- a/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c +++ b/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c @@ -340,16 +340,17 @@ static int tas2563_save_calibration(struct tas2781_hda *h) data[offset] = i; offset++; for (j = 0; j < TASDEV_CALIB_N; ++j) { - ret = snprintf(var8, sizeof(var8), vars[j], i); + /* EFI name for calibration started with 1, not 0 */ + ret = snprintf(var8, sizeof(var8), vars[j], i + 1); - if (ret < 0 || ret >= sizeof(var8) - 1) { + if (ret != strlen(var8)) { dev_err(p->dev, "%s: Read %s failed\n", __func__, var8); return -EINVAL; } /* * Our variable names are ASCII by construction, but - * EFI names are wide chars. Convert and zero-pad. + * EFI names are wide chars. Convert and zero-pad. */ memset(efi_name, 0, sizeof(efi_name)); for (k = 0; k < sizeof(var8) && var8[k]; k++) -- 2.43.0

2 1

my duo cast volume is too low
by GitHub issues - opened 25 Aug '25

25 Aug '25

alsa-project/alsa-ucm-conf issue #604 was opened from reflexionn: hello i bought the hyperx duocast mic for recording reasons i have a small youtube channel that goes by the name vironio i was dissapointed with the mic performance because it has very low volume and if i wanna record i have to be uncomfortably close to the microphone i thought thats just how the mic is and i lived with it i set the gain to around 70% because anything more would lead to weird noise that comes from the mic itself and i just record and then put the recording in adobe audition add a lot of effects and noise cancelation and audio gate and then i amplify the audio by around 9db so its hearable but not so long ago i saw a tiktoker with the same mic but he has a much higher volume and can use the microphone from around 2 to 3 inches away with the quality being really good so i thought its not actually the mics fault and i just didnt set it up right can anyone help me? Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/604 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ucm2: MediaTek: mt8395-evk: Add support for SOF
by GitHub pull_request - edited 25 Aug '25

25 Aug '25

alsa-project/alsa-ucm-conf pull request #549 was edited from julien-massot: Add support for sof-mt8395-evk. Currently, enable only the minimum: earphone output and HDMI output. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/549 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/549.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v2 1/2] ASoC: tas2781: Add tas2118, tas2x20, tas5825 support
by Baojun Xu 22 Aug '25

22 Aug '25

Add tas2020, tas2118, tas2120, tas2320, tas5825 support in tas2781 driver. Tas2118, tas2x20 have no on-chip DSP, tas5825 have on-chip DSP but have no calibration required stereo smart amplifier. Signed-off-by: Baojun Xu <baojun.xu(a)ti.com> --- v2: - Update the mail list for maintainers of yaml file --- include/sound/tas2781.h | 11 +- include/sound/tas2x20-tlv.h | 259 ++++++++++++++++++++++++++ sound/soc/codecs/tas2781-comlib-i2c.c | 2 + sound/soc/codecs/tas2781-i2c.c | 156 +++++++++++----- 4 files changed, 381 insertions(+), 47 deletions(-) create mode 100644 include/sound/tas2x20-tlv.h diff --git a/include/sound/tas2781.h b/include/sound/tas2781.h index f0aefc04a957..73ac447b9327 100644 --- a/include/sound/tas2781.h +++ b/include/sound/tas2781.h @@ -51,7 +51,9 @@ /* Software Reset, compatble with new device (TAS5825). */ #define TASDEVICE_REG_SWRESET TASDEVICE_REG(0x0, 0x0, 0x01) -#define TASDEVICE_REG_SWRESET_RESET (BIT(0) | BIT(4)) +#define TASDEVICE_REG_SWRESET_RESET BIT(0) + +#define TAS5825_REG_SWRESET_RESET (BIT(0) | BIT(4)) /* Checksum */ #define TASDEVICE_CHECKSUM_REG TASDEVICE_REG(0x0, 0x0, 0x7e) @@ -110,8 +112,14 @@ #define TAS2781_RUNTIME_RE_REG TASDEVICE_REG(0x64, 0x63, 0x44) enum audio_device { + TAS2020, + TAS2118, + TAS2120, + TAS2320, TAS2563, TAS2781, + TAS5825, + TAS_OTHERS, }; enum dspbin_type { @@ -194,6 +202,7 @@ struct tasdevice_priv { unsigned char coef_binaryname[64]; unsigned char rca_binaryname[64]; unsigned char dev_name[32]; + const unsigned char (*dvc_tlv_table)[4]; const char *name_prefix; unsigned char ndev; unsigned int dspbin_typ; diff --git a/include/sound/tas2x20-tlv.h b/include/sound/tas2x20-tlv.h new file mode 100644 index 000000000000..6e6bcec4a0a1 --- /dev/null +++ b/include/sound/tas2x20-tlv.h @@ -0,0 +1,259 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +// +// ALSA SoC Texas Instruments TAS2x20/TAS2118 Audio Smart Amplifier +// +// Copyright (C) 2025 Texas Instruments Incorporated +// https://www.ti.com +// +// The TAS2x20/TAS2118 hda driver implements for one, two, or even multiple +// TAS2x20/TAS2118 chips. +// +// Author: Baojun Xu <baojun.xu(a)ti.com> +// + +#ifndef __TAS2X20_TLV_H__ +#define __TAS2X20_TLV_H__ + +#define TAS2X20_DVC_LEVEL TASDEVICE_REG(0x0, 0x2, 0x0c) +#define TAS2X20_AMP_LEVEL TASDEVICE_REG(0x0, 0x0, 0x07) + +static const __maybe_unused DECLARE_TLV_DB_SCALE(tas2x20_dvc_tlv, 1650, 50, 0); +static const __maybe_unused DECLARE_TLV_DB_SCALE(tas2x20_amp_tlv, 2100, 50, 0); + +/* pow(10, db/20) * pow(2,22) */ +static const __maybe_unused unsigned char tas2x20_dvc_table[][4] = { + { 0X00, 0X00, 0X0D, 0X00 }, /* -110.0db */ + { 0X00, 0X00, 0X0E, 0X00 }, /* -109.5db */ + { 0X00, 0X00, 0X0E, 0X00 }, /* -109.0db */ + { 0X00, 0X00, 0X0F, 0X00 }, /* -108.5db */ + { 0X00, 0X00, 0X10, 0X00 }, /* -108.0db */ + { 0X00, 0X00, 0X11, 0X00 }, /* -107.5db */ + { 0X00, 0X00, 0X12, 0X00 }, /* -107.0db */ + { 0X00, 0X00, 0X13, 0X00 }, /* -106.5db */ + { 0X00, 0X00, 0X15, 0X00 }, /* -106.0db */ + { 0X00, 0X00, 0X16, 0X00 }, /* -105.5db */ + { 0X00, 0X00, 0X17, 0X00 }, /* -105.0db */ + { 0X00, 0X00, 0X18, 0X00 }, /* -104.5db */ + { 0X00, 0X00, 0X1A, 0X00 }, /* -104.0db */ + { 0X00, 0X00, 0X1C, 0X00 }, /* -103.5db */ + { 0X00, 0X00, 0X1D, 0X00 }, /* -103.0db */ + { 0X00, 0X00, 0X1F, 0X00 }, /* -102.5db */ + { 0X00, 0X00, 0X21, 0X00 }, /* -102.0db */ + { 0X00, 0X00, 0X23, 0X00 }, /* -101.5db */ + { 0X00, 0X00, 0X25, 0X00 }, /* -101.0db */ + { 0X00, 0X00, 0X27, 0X00 }, /* -100.5db */ + { 0X00, 0X00, 0X29, 0X00 }, /* -100.0db */ + { 0X00, 0X00, 0X2C, 0X00 }, /* -99.5db */ + { 0X00, 0X00, 0X2F, 0X00 }, /* -99.0db */ + { 0X00, 0X00, 0X31, 0X00 }, /* -98.5db */ + { 0X00, 0X00, 0X34, 0X00 }, /* -98.0db */ + { 0X00, 0X00, 0X37, 0X00 }, /* -97.5db */ + { 0X00, 0X00, 0X3B, 0X00 }, /* -97.0db */ + { 0X00, 0X00, 0X3E, 0X00 }, /* -96.5db */ + { 0X00, 0X00, 0X42, 0X00 }, /* -96.0db */ + { 0X00, 0X00, 0X46, 0X00 }, /* -95.5db */ + { 0X00, 0X00, 0X4A, 0X00 }, /* -95.0db */ + { 0X00, 0X00, 0X4F, 0X00 }, /* -94.5db */ + { 0X00, 0X00, 0X53, 0X00 }, /* -94.0db */ + { 0X00, 0X00, 0X58, 0X00 }, /* -93.5db */ + { 0X00, 0X00, 0X5D, 0X00 }, /* -93.0db */ + { 0X00, 0X00, 0X63, 0X00 }, /* -92.5db */ + { 0X00, 0X00, 0X69, 0X00 }, /* -92.0db */ + { 0X00, 0X00, 0X6F, 0X00 }, /* -91.5db */ + { 0X00, 0X00, 0X76, 0X00 }, /* -91.0db */ + { 0X00, 0X00, 0X7D, 0X00 }, /* -90.5db */ + { 0X00, 0X00, 0X84, 0X00 }, /* -90.0db */ + { 0X00, 0X00, 0X8C, 0X00 }, /* -89.5db */ + { 0X00, 0X00, 0X94, 0X00 }, /* -89.0db */ + { 0X00, 0X00, 0X9D, 0X00 }, /* -88.5db */ + { 0X00, 0X00, 0XA6, 0X00 }, /* -88.0db */ + { 0X00, 0X00, 0XB0, 0X00 }, /* -87.5db */ + { 0X00, 0X00, 0XBB, 0X00 }, /* -87.0db */ + { 0X00, 0X00, 0XC6, 0X00 }, /* -86.5db */ + { 0X00, 0X00, 0XD2, 0X00 }, /* -86.0db */ + { 0X00, 0X00, 0XDE, 0X00 }, /* -85.5db */ + { 0X00, 0X00, 0XEB, 0X00 }, /* -85.0db */ + { 0X00, 0X00, 0XF9, 0X00 }, /* -84.5db */ + { 0X00, 0X01, 0X08, 0X00 }, /* -84.0db */ + { 0X00, 0X01, 0X18, 0X00 }, /* -83.5db */ + { 0X00, 0X01, 0X28, 0X00 }, /* -83.0db */ + { 0X00, 0X01, 0X3A, 0X00 }, /* -82.5db */ + { 0X00, 0X01, 0X4D, 0X00 }, /* -82.0db */ + { 0X00, 0X01, 0X60, 0X00 }, /* -81.5db */ + { 0X00, 0X01, 0X75, 0X00 }, /* -81.0db */ + { 0X00, 0X01, 0X8B, 0X00 }, /* -80.5db */ + { 0X00, 0X01, 0XA3, 0X00 }, /* -80.0db */ + { 0X00, 0X01, 0XBC, 0X00 }, /* -79.5db */ + { 0X00, 0X01, 0XD6, 0X00 }, /* -79.0db */ + { 0X00, 0X01, 0XF2, 0X00 }, /* -78.5db */ + { 0X00, 0X02, 0X10, 0X00 }, /* -78.0db */ + { 0X00, 0X02, 0X2F, 0X00 }, /* -77.5db */ + { 0X00, 0X02, 0X50, 0X00 }, /* -77.0db */ + { 0X00, 0X02, 0X73, 0X00 }, /* -76.5db */ + { 0X00, 0X02, 0X98, 0X00 }, /* -76.0db */ + { 0X00, 0X02, 0XC0, 0X00 }, /* -75.5db */ + { 0X00, 0X02, 0XE9, 0X00 }, /* -75.0db */ + { 0X00, 0X03, 0X16, 0X00 }, /* -74.5db */ + { 0X00, 0X03, 0X44, 0X00 }, /* -74.0db */ + { 0X00, 0X03, 0X76, 0X00 }, /* -73.5db */ + { 0X00, 0X03, 0XAA, 0X00 }, /* -73.0db */ + { 0X00, 0X03, 0XE2, 0X00 }, /* -72.5db */ + { 0X00, 0X04, 0X1D, 0X00 }, /* -72.0db */ + { 0X00, 0X04, 0X5B, 0X00 }, /* -71.5db */ + { 0X00, 0X04, 0X9E, 0X00 }, /* -71.0db */ + { 0X00, 0X04, 0XE4, 0X00 }, /* -70.5db */ + { 0X00, 0X05, 0X2E, 0X00 }, /* -70.0db */ + { 0X00, 0X05, 0X7C, 0X00 }, /* -69.5db */ + { 0X00, 0X05, 0XD0, 0X00 }, /* -69.0db */ + { 0X00, 0X06, 0X28, 0X00 }, /* -68.5db */ + { 0X00, 0X06, 0X85, 0X00 }, /* -68.0db */ + { 0X00, 0X06, 0XE8, 0X00 }, /* -67.5db */ + { 0X00, 0X07, 0X51, 0X00 }, /* -67.0db */ + { 0X00, 0X07, 0XC0, 0X00 }, /* -66.5db */ + { 0X00, 0X08, 0X36, 0X00 }, /* -66.0db */ + { 0X00, 0X08, 0XB2, 0X00 }, /* -65.5db */ + { 0X00, 0X09, 0X36, 0X00 }, /* -65.0db */ + { 0X00, 0X09, 0XC2, 0X00 }, /* -64.5db */ + { 0X00, 0X0A, 0X56, 0X00 }, /* -64.0db */ + { 0X00, 0X0A, 0XF3, 0X00 }, /* -63.5db */ + { 0X00, 0X0B, 0X99, 0X00 }, /* -63.0db */ + { 0X00, 0X0C, 0X49, 0X00 }, /* -62.5db */ + { 0X00, 0X0D, 0X03, 0X00 }, /* -62.0db */ + { 0X00, 0X0D, 0XC9, 0X00 }, /* -61.5db */ + { 0X00, 0X0E, 0X9A, 0X00 }, /* -61.0db */ + { 0X00, 0X0F, 0X77, 0X00 }, /* -60.5db */ + { 0X00, 0X10, 0X62, 0X00 }, /* -60.0db */ + { 0X00, 0X11, 0X5A, 0X00 }, /* -59.5db */ + { 0X00, 0X12, 0X62, 0X00 }, /* -59.0db */ + { 0X00, 0X13, 0X78, 0X00 }, /* -58.5db */ + { 0X00, 0X14, 0XA0, 0X00 }, /* -58.0db */ + { 0X00, 0X15, 0XD9, 0X00 }, /* -57.5db */ + { 0X00, 0X17, 0X24, 0X00 }, /* -57.0db */ + { 0X00, 0X18, 0X83, 0X00 }, /* -56.5db */ + { 0X00, 0X19, 0XF7, 0X00 }, /* -56.0db */ + { 0X00, 0X1B, 0X81, 0X00 }, /* -55.5db */ + { 0X00, 0X1D, 0X22, 0X00 }, /* -55.0db */ + { 0X00, 0X1E, 0XDC, 0X00 }, /* -54.5db */ + { 0X00, 0X20, 0XB0, 0X00 }, /* -54.0db */ + { 0X00, 0X22, 0XA0, 0X00 }, /* -53.5db */ + { 0X00, 0X24, 0XAD, 0X00 }, /* -53.0db */ + { 0X00, 0X26, 0XDA, 0X00 }, /* -52.5db */ + { 0X00, 0X29, 0X27, 0X00 }, /* -52.0db */ + { 0X00, 0X2B, 0X97, 0X00 }, /* -51.5db */ + { 0X00, 0X2E, 0X2D, 0X00 }, /* -51.0db */ + { 0X00, 0X30, 0XE9, 0X00 }, /* -50.5db */ + { 0X00, 0X33, 0XCF, 0X00 }, /* -50.0db */ + { 0X00, 0X36, 0XE1, 0X00 }, /* -49.5db */ + { 0X00, 0X3A, 0X21, 0X00 }, /* -49.0db */ + { 0X00, 0X3D, 0X93, 0X00 }, /* -48.5db */ + { 0X00, 0X41, 0X39, 0X00 }, /* -48.0db */ + { 0X00, 0X45, 0X17, 0X00 }, /* -47.5db */ + { 0X00, 0X49, 0X2F, 0X00 }, /* -47.0db */ + { 0X00, 0X4D, 0X85, 0X00 }, /* -46.5db */ + { 0X00, 0X52, 0X1D, 0X00 }, /* -46.0db */ + { 0X00, 0X56, 0XFA, 0X00 }, /* -45.5db */ + { 0X00, 0X5C, 0X22, 0X00 }, /* -45.0db */ + { 0X00, 0X61, 0X97, 0X00 }, /* -44.5db */ + { 0X00, 0X67, 0X60, 0X00 }, /* -44.0db */ + { 0X00, 0X6D, 0X80, 0X00 }, /* -43.5db */ + { 0X00, 0X73, 0XFD, 0X00 }, /* -43.0db */ + { 0X00, 0X7A, 0XDC, 0X00 }, /* -42.5db */ + { 0X00, 0X82, 0X24, 0X00 }, /* -42.0db */ + { 0X00, 0X89, 0XDA, 0X00 }, /* -41.5db */ + { 0X00, 0X92, 0X05, 0X00 }, /* -41.0db */ + { 0X00, 0X9A, 0XAC, 0X00 }, /* -40.5db */ + { 0X00, 0XA3, 0XD7, 0X00 }, /* -40.0db */ + { 0X00, 0XAD, 0X8C, 0X00 }, /* -39.5db */ + { 0X00, 0XB7, 0XD4, 0X00 }, /* -39.0db */ + { 0X00, 0XC2, 0XB9, 0X00 }, /* -38.5db */ + { 0X00, 0XCE, 0X43, 0X00 }, /* -38.0db */ + { 0X00, 0XDA, 0X7B, 0X00 }, /* -37.5db */ + { 0X00, 0XE7, 0X6E, 0X00 }, /* -37.0db */ + { 0X00, 0XF5, 0X24, 0X00 }, /* -36.5db */ + { 0X01, 0X03, 0XAB, 0X00 }, /* -36.0db */ + { 0X01, 0X13, 0X0E, 0X00 }, /* -35.5db */ + { 0X01, 0X23, 0X5A, 0X00 }, /* -35.0db */ + { 0X01, 0X34, 0X9D, 0X00 }, /* -34.5db */ + { 0X01, 0X46, 0XE7, 0X00 }, /* -34.0db */ + { 0X01, 0X5A, 0X46, 0X00 }, /* -33.5db */ + { 0X01, 0X6E, 0XCA, 0X00 }, /* -33.0db */ + { 0X01, 0X84, 0X86, 0X00 }, /* -32.5db */ + { 0X01, 0X9B, 0X8C, 0X00 }, /* -32.0db */ + { 0X01, 0XB3, 0XEE, 0X00 }, /* -31.5db */ + { 0X01, 0XCD, 0XC3, 0X00 }, /* -31.0db */ + { 0X01, 0XE9, 0X20, 0X00 }, /* -30.5db */ + { 0X02, 0X06, 0X1B, 0X00 }, /* -30.0db */ + { 0X02, 0X24, 0XCE, 0X00 }, /* -29.5db */ + { 0X02, 0X45, 0X53, 0X00 }, /* -29.0db */ + { 0X02, 0X67, 0XC5, 0X00 }, /* -28.5db */ + { 0X02, 0X8C, 0X42, 0X00 }, /* -28.0db */ + { 0X02, 0XB2, 0XE8, 0X00 }, /* -27.5db */ + { 0X02, 0XDB, 0XD8, 0X00 }, /* -27.0db */ + { 0X03, 0X07, 0X36, 0X00 }, /* -26.5db */ + { 0X03, 0X35, 0X25, 0X00 }, /* -26.0db */ + { 0X03, 0X65, 0XCD, 0X00 }, /* -25.5db */ + { 0X03, 0X99, 0X57, 0X00 }, /* -25.0db */ + { 0X03, 0XCF, 0XEE, 0X00 }, /* -24.5db */ + { 0X04, 0X09, 0XC2, 0X00 }, /* -24.0db */ + { 0X04, 0X47, 0X03, 0X00 }, /* -23.5db */ + { 0X04, 0X87, 0XE5, 0X00 }, /* -23.0db */ + { 0X04, 0XCC, 0XA0, 0X00 }, /* -22.5db */ + { 0X05, 0X15, 0X6D, 0X00 }, /* -22.0db */ + { 0X05, 0X62, 0X8A, 0X00 }, /* -21.5db */ + { 0X05, 0XB4, 0X39, 0X00 }, /* -21.0db */ + { 0X06, 0X0A, 0XBF, 0X00 }, /* -20.5db */ + { 0X06, 0X66, 0X66, 0X00 }, /* -20.0db */ + { 0X06, 0XC7, 0X7B, 0X00 }, /* -19.5db */ + { 0X07, 0X2E, 0X50, 0X00 }, /* -19.0db */ + { 0X07, 0X9B, 0X3D, 0X00 }, /* -18.5db */ + { 0X08, 0X0E, 0X9F, 0X00 }, /* -18.0db */ + { 0X08, 0X88, 0XD7, 0X00 }, /* -17.5db */ + { 0X09, 0X0A, 0X4D, 0X00 }, /* -17.0db */ + { 0X09, 0X93, 0X6E, 0X00 }, /* -16.5db */ + { 0X0A, 0X24, 0XB0, 0X00 }, /* -16.0db */ + { 0X0A, 0XBE, 0X8D, 0X00 }, /* -15.5db */ + { 0X0B, 0X61, 0X88, 0X00 }, /* -15.0db */ + { 0X0C, 0X0E, 0X2B, 0X00 }, /* -14.5db */ + { 0X0C, 0XC5, 0X09, 0X00 }, /* -14.0db */ + { 0X0D, 0X86, 0XBD, 0X00 }, /* -13.5db */ + { 0X0E, 0X53, 0XEB, 0X00 }, /* -13.0db */ + { 0X0F, 0X2D, 0X42, 0X00 }, /* -12.5db */ + { 0X10, 0X13, 0X79, 0X00 }, /* -12.0db */ + { 0X11, 0X07, 0X54, 0X00 }, /* -11.5db */ + { 0X12, 0X09, 0XA3, 0X00 }, /* -11.0db */ + { 0X13, 0X1B, 0X40, 0X00 }, /* -10.5db */ + { 0X14, 0X3D, 0X13, 0X00 }, /* -10.0db */ + { 0X15, 0X70, 0X12, 0X00 }, /* -9.5db */ + { 0X16, 0XB5, 0X43, 0X00 }, /* -9.0db */ + { 0X18, 0X0D, 0XB8, 0X00 }, /* -8.5db */ + { 0X19, 0X7A, 0X96, 0X00 }, /* -8.0db */ + { 0X1A, 0XFD, 0X13, 0X00 }, /* -7.5db */ + { 0X1C, 0X96, 0X76, 0X00 }, /* -7.0db */ + { 0X1E, 0X48, 0X1C, 0X00 }, /* -6.5db */ + { 0X20, 0X13, 0X73, 0X00 }, /* -6.0db */ + { 0X21, 0XFA, 0X02, 0X00 }, /* -5.5db */ + { 0X23, 0XFD, 0X66, 0X00 }, /* -5.0db */ + { 0X26, 0X1F, 0X54, 0X00 }, /* -4.5db */ + { 0X28, 0X61, 0X9A, 0X00 }, /* -4.0db */ + { 0X2A, 0XC6, 0X25, 0X00 }, /* -3.5db */ + { 0X2D, 0X4E, 0XFB, 0X00 }, /* -3.0db */ + { 0X2F, 0XFE, 0X44, 0X00 }, /* -2.5db */ + { 0X32, 0XD6, 0X46, 0X00 }, /* -2.0db */ + { 0X35, 0XD9, 0X6B, 0X00 }, /* -1.5db */ + { 0X39, 0X0A, 0X41, 0X00 }, /* -1.0db */ + { 0X3C, 0X6B, 0X7E, 0X00 }, /* -0.5db */ + { 0X40, 0X00, 0X00, 0X00 }, /* 0.0db */ + { 0X43, 0XCA, 0XD0, 0X00 }, /* 0.5db */ + { 0X47, 0XCF, 0X26, 0X00 }, /* 1.0db */ + { 0X4C, 0X10, 0X6B, 0X00 }, /* 1.5db */ + { 0X50, 0X92, 0X3B, 0X00 }, /* 2.0db */ + { 0X55, 0X58, 0X6A, 0X00 }, /* 2.5db */ + { 0X5A, 0X67, 0X03, 0X00 }, /* 3.0db */ + { 0X5F, 0XC2, 0X53, 0X00 }, /* 3.5db */ + { 0X65, 0X6E, 0XE3, 0X00 }, /* 4.0db */ + { 0X6B, 0X71, 0X86, 0X00 }, /* 4.5db */ + { 0X71, 0XCF, 0X54, 0X00 }, /* 5.0db */ + { 0X78, 0X8D, 0XB4, 0X00 }, /* 5.5db */ + { 0X7F, 0XB2, 0X61, 0X00 }, /* 6.0db */ +}; +#endif diff --git a/sound/soc/codecs/tas2781-comlib-i2c.c b/sound/soc/codecs/tas2781-comlib-i2c.c index c078bb0a8437..764304604e2c 100644 --- a/sound/soc/codecs/tas2781-comlib-i2c.c +++ b/sound/soc/codecs/tas2781-comlib-i2c.c @@ -320,6 +320,8 @@ void tasdevice_reset(struct tasdevice_priv *tas_dev) for (i = 0; i < tas_dev->ndev; i++) { ret = tasdevice_dev_write(tas_dev, i, TASDEVICE_REG_SWRESET, + tas_dev->chip_id == TAS5825 ? + TAS5825_REG_SWRESET_RESET : TASDEVICE_REG_SWRESET_RESET); if (ret < 0) dev_err(tas_dev->dev, diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index ea3cdb8553de..cf6bd4d63f31 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -30,8 +30,10 @@ #include <sound/tas2781.h> #include <sound/tas2781-comlib-i2c.h> #include <sound/tlv.h> +#include <sound/tas2x20-tlv.h> #include <sound/tas2563-tlv.h> #include <sound/tas2781-tlv.h> +#include <sound/tas5825-tlv.h> #include <linux/unaligned.h> #define X2563_CL_STT_VAL(xreg, xval) \ @@ -98,16 +100,26 @@ static const struct bulk_reg_val tas2781_cali_start_reg[] = { }; static const struct i2c_device_id tasdevice_id[] = { + { "tas2020", TAS2020 }, + { "tas2118", TAS2118 }, + { "tas2120", TAS2120 }, + { "tas2320", TAS2320 }, { "tas2563", TAS2563 }, { "tas2781", TAS2781 }, + { "tas5825", TAS5825 }, {} }; MODULE_DEVICE_TABLE(i2c, tasdevice_id); #ifdef CONFIG_OF static const struct of_device_id tasdevice_of_match[] = { + { .compatible = "ti,tas2020" }, + { .compatible = "ti,tas2118" }, + { .compatible = "ti,tas2120" }, + { .compatible = "ti,tas2320" }, { .compatible = "ti,tas2563" }, { .compatible = "ti,tas2781" }, + { .compatible = "ti,tas5825" }, {}, }; MODULE_DEVICE_TABLE(of, tasdevice_of_match); @@ -797,7 +809,7 @@ static int tasdev_nop_get( return 0; } -static int tas2563_digital_gain_get( +static int tasdevice_digital_gain_get( struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -823,15 +835,15 @@ static int tas2563_digital_gain_get( while (r > 1 + l) { mid = (l + r) / 2; - ar_mid = get_unaligned_be32(tas2563_dvc_table[mid]); + ar_mid = get_unaligned_be32(tas_dev->dvc_tlv_table[mid]); if (target < ar_mid) r = mid; else l = mid; } - ar_l = get_unaligned_be32(tas2563_dvc_table[l]); - ar_r = get_unaligned_be32(tas2563_dvc_table[r]); + ar_l = get_unaligned_be32(tas_dev->dvc_tlv_table[l]); + ar_r = get_unaligned_be32(tas_dev->dvc_tlv_table[r]); /* find out the member same as or closer to the current volume */ ucontrol->value.integer.value[0] = @@ -841,7 +853,7 @@ static int tas2563_digital_gain_get( return 0; } -static int tas2563_digital_gain_put( +static int tasdevice_digital_gain_put( struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -867,7 +879,7 @@ static int tas2563_digital_gain_put( } volrd = get_unaligned_be32(&data[0]); - volwr = get_unaligned_be32(tas2563_dvc_table[vol]); + volwr = get_unaligned_be32(tas_dev->dvc_tlv_table[vol]); if (volrd == volwr) { rc = 0; @@ -876,7 +888,7 @@ static int tas2563_digital_gain_put( for (i = 0; i < tas_dev->ndev; i++) { ret = tasdevice_dev_bulk_write(tas_dev, i, reg, - (unsigned char *)tas2563_dvc_table[vol], 4); + (unsigned char *)tas_dev->dvc_tlv_table[vol], 4); if (ret) { dev_err(tas_dev->dev, "%s, set digital vol error in dev %d\n", @@ -892,11 +904,6 @@ static int tas2563_digital_gain_put( return rc; } -static const struct snd_kcontrol_new tasdevice_snd_controls[] = { - SOC_SINGLE_BOOL_EXT("Speaker Force Firmware Load", 0, - tasdev_force_fwload_get, tasdev_force_fwload_put), -}; - static const struct snd_kcontrol_new tasdevice_cali_controls[] = { SOC_SINGLE_EXT("Calibration Stop", SND_SOC_NOPM, 0, 1, 0, tasdev_nop_get, tasdev_calib_stop_put), @@ -907,6 +914,16 @@ static const struct snd_kcontrol_new tasdevice_cali_controls[] = { SND_SOC_BYTES_EXT("Amp XMA2 Data", 6, tasdev_XMA2_data_get, NULL), }; +static const struct snd_kcontrol_new tas2x20_snd_controls[] = { + SOC_SINGLE_RANGE_EXT_TLV("Speaker Analog Volume", TAS2X20_AMP_LEVEL, + 0, 0, 42, 1, tas2781_amp_getvol, + tas2781_amp_putvol, tas2x20_amp_tlv), + SOC_SINGLE_RANGE_EXT_TLV("Speaker Digital Volume", TAS2X20_DVC_LEVEL, + 0, 0, ARRAY_SIZE(tas2x20_dvc_table) - 1, 0, + tasdevice_digital_gain_get, tasdevice_digital_gain_put, + tas2x20_dvc_tlv), +}; + static const struct snd_kcontrol_new tas2781_snd_controls[] = { SOC_SINGLE_RANGE_EXT_TLV("Speaker Analog Volume", TAS2781_AMP_LEVEL, 1, 0, 20, 0, tas2781_amp_getvol, @@ -916,6 +933,15 @@ static const struct snd_kcontrol_new tas2781_snd_controls[] = { tas2781_digital_putvol, tas2781_dvc_tlv), }; +static const struct snd_kcontrol_new tas5825_snd_controls[] = { + SOC_SINGLE_RANGE_EXT_TLV("Speaker Analog Volume", TAS5825_AMP_LEVEL, + 0, 0, 31, 1, tas2781_amp_getvol, + tas2781_amp_putvol, tas5825_amp_tlv), + SOC_SINGLE_RANGE_EXT_TLV("Speaker Digital Volume", TAS5825_DVC_LEVEL, + 0, 0, 254, 1, tas2781_amp_getvol, + tas2781_amp_putvol, tas5825_dvc_tlv), +}; + static const struct snd_kcontrol_new tas2781_cali_controls[] = { SND_SOC_BYTES_EXT("Amp Latch Data", 3, tas2781_latch_reg_get, NULL), }; @@ -923,7 +949,7 @@ static const struct snd_kcontrol_new tas2781_cali_controls[] = { static const struct snd_kcontrol_new tas2563_snd_controls[] = { SOC_SINGLE_RANGE_EXT_TLV("Speaker Digital Volume", TAS2563_DVC_LVL, 0, 0, ARRAY_SIZE(tas2563_dvc_table) - 1, 0, - tas2563_digital_gain_get, tas2563_digital_gain_put, + tasdevice_digital_gain_get, tasdevice_digital_gain_put, tas2563_dvc_tlv), }; @@ -968,7 +994,7 @@ static int tasdevice_info_chip_id(struct snd_kcontrol *kcontrol, { uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->count = 1; - uinfo->value.integer.min = TAS2563; + uinfo->value.integer.min = TAS2020; uinfo->value.integer.max = TAS2781; return 0; @@ -1168,9 +1194,9 @@ static int tasdevice_active_num_put(struct snd_kcontrol *kcontrol, static int tasdevice_dsp_create_ctrls(struct tasdevice_priv *tas_priv) { struct snd_kcontrol_new *dsp_ctrls; - char *active_dev_num, *chip_id; + char *active_dev_num, *chip_id, *fw_load; char *conf_name, *prog_name; - int nr_controls = 4; + int nr_controls = 5; int mix_index = 0; /* Alloc kcontrol via devm_kzalloc, which don't manually @@ -1228,6 +1254,19 @@ static int tasdevice_dsp_create_ctrls(struct tasdevice_priv *tas_priv) dsp_ctrls[mix_index].get = tasdevice_get_chip_id; mix_index++; + fw_load = devm_kstrdup(tas_priv->dev, "Speaker Force Firmware Load", + GFP_KERNEL); + if (!fw_load) + return -ENOMEM; + + dsp_ctrls[mix_index].name = fw_load; + dsp_ctrls[mix_index].iface = SNDRV_CTL_ELEM_IFACE_MIXER; + dsp_ctrls[mix_index].info = snd_soc_info_bool_ext; + dsp_ctrls[mix_index].put = tasdev_force_fwload_put; + dsp_ctrls[mix_index].get = tasdev_force_fwload_get; + dsp_ctrls[mix_index].private_value = 0UL; + mix_index++; + return snd_soc_add_component_controls(tas_priv->codec, dsp_ctrls, nr_controls < mix_index ? nr_controls : mix_index); } @@ -1587,6 +1626,9 @@ static void tasdevice_fw_ready(const struct firmware *fmw, * failing to load DSP firmware is NOT an error. */ tas_priv->fw_state = TASDEVICE_RCA_FW_OK; + /* There is no DSP firmware required for TAS2118/2X20. */ + if (tas_priv->chip_id < TAS2563) + goto out; if (tas_priv->name_prefix) scnprintf(tas_priv->coef_binaryname, 64, "%s-%s_coef.bin", tas_priv->name_prefix, tas_priv->dev_name); @@ -1608,34 +1650,37 @@ static void tasdevice_fw_ready(const struct firmware *fmw, dev_err(tas_priv->dev, "dsp controls error\n"); goto out; } - - ret = tasdevice_create_cali_ctrls(tas_priv); - if (ret) { - dev_err(tas_priv->dev, "cali controls error\n"); - goto out; - } - tas_priv->fw_state = TASDEVICE_DSP_FW_ALL_OK; - /* If calibrated data occurs error, dsp will still works with default - * calibrated data inside algo. - */ - for (i = 0; i < tas_priv->ndev; i++) { - if (tas_priv->name_prefix) - scnprintf(tas_priv->cal_binaryname[i], 64, - "%s-%s_cal_0x%02x.bin", tas_priv->name_prefix, - tas_priv->dev_name, - tas_priv->tasdevice[i].dev_addr); - else - scnprintf(tas_priv->cal_binaryname[i], 64, - "%s_cal_0x%02x.bin", tas_priv->dev_name, - tas_priv->tasdevice[i].dev_addr); - ret = tas2781_load_calibration(tas_priv, - tas_priv->cal_binaryname[i], i); - if (ret != 0) - dev_err(tas_priv->dev, - "%s: load %s error, default will effect\n", - __func__, tas_priv->cal_binaryname[i]); + /* There is no calibration required for TAS5825. */ + if (tas_priv->chip_id < TAS5825) { + ret = tasdevice_create_cali_ctrls(tas_priv); + if (ret) { + dev_err(tas_priv->dev, "cali controls error\n"); + goto out; + } + /* If calibrated data occurs error, dsp will still works + * with default calibrated data inside algo. + */ + for (i = 0; i < tas_priv->ndev; i++) { + if (tas_priv->name_prefix) + scnprintf(tas_priv->cal_binaryname[i], 64, + "%s-%s_cal_0x%02x.bin", + tas_priv->name_prefix, + tas_priv->dev_name, + tas_priv->tasdevice[i].dev_addr); + else + scnprintf(tas_priv->cal_binaryname[i], 64, + "%s_cal_0x%02x.bin", + tas_priv->dev_name, + tas_priv->tasdevice[i].dev_addr); + ret = tas2781_load_calibration(tas_priv, + tas_priv->cal_binaryname[i], i); + if (ret != 0) + dev_err(tas_priv->dev, + "%s: load %s error, keep default.\n", + __func__, tas_priv->cal_binaryname[i]); + } } tasdevice_prmg_load(tas_priv, 0); @@ -1658,7 +1703,8 @@ static void tasdevice_fw_ready(const struct firmware *fmw, comp, &acoustic_ctl_fops); #endif out: - if (tas_priv->fw_state == TASDEVICE_RCA_FW_OK) { + if ((tas_priv->fw_state == TASDEVICE_RCA_FW_OK) && + (tas_priv->chip_id >= TAS2563)) { /* If DSP FW fail, DSP kcontrol won't be created. */ tasdevice_dsp_remove(tas_priv); } @@ -1804,13 +1850,27 @@ static int tasdevice_codec_probe(struct snd_soc_component *codec) int rc; switch (tas_priv->chip_id) { + case TAS2020: + case TAS2118: + case TAS2120: + case TAS2320: + p = (struct snd_kcontrol_new *)tas2x20_snd_controls; + size = ARRAY_SIZE(tas2x20_snd_controls); + tas_priv->dvc_tlv_table = tas2x20_dvc_table; + break; case TAS2781: p = (struct snd_kcontrol_new *)tas2781_snd_controls; size = ARRAY_SIZE(tas2781_snd_controls); break; + case TAS5825: + p = (struct snd_kcontrol_new *)tas5825_snd_controls; + size = ARRAY_SIZE(tas5825_snd_controls); + break; default: p = (struct snd_kcontrol_new *)tas2563_snd_controls; size = ARRAY_SIZE(tas2563_snd_controls); + tas_priv->dvc_tlv_table = tas2563_dvc_table; + break; } rc = snd_soc_add_component_controls(codec, p, size); @@ -1850,8 +1910,6 @@ static const struct snd_soc_component_driver soc_codec_driver_tasdevice = { .probe = tasdevice_codec_probe, .remove = tasdevice_codec_remove, - .controls = tasdevice_snd_controls, - .num_controls = ARRAY_SIZE(tasdevice_snd_controls), .dapm_widgets = tasdevice_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(tasdevice_dapm_widgets), .dapm_routes = tasdevice_audio_map, @@ -1967,7 +2025,13 @@ static void tasdevice_i2c_remove(struct i2c_client *client) #ifdef CONFIG_ACPI static const struct acpi_device_id tasdevice_acpi_match[] = { - { "TAS2781", TAS2781 }, + { "TXNW2020", TAS2020 }, + { "TXNW2118", TAS2118 }, + { "TXNW2120", TAS2120 }, + { "TXNW2320", TAS2320 }, + { "TXNW2563", TAS2563 }, + { "TXNW2781", TAS2781 }, + { "TXNW5825", TAS5825 }, {}, }; -- 2.43.0

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