August 2025 - Alsa-devel - mailman.alsa-project.org

aseqsend silently fails (ENODEV) with hardware port
by GitHub issues - opened 30 Oct '25

30 Oct '25

alsa-project/alsa-utils issue #300 was opened from michaelforney: When I use `aseqsend` to send a sysex message to a device, it doesn't seem to work, though there is no error message. Running with strace, it seems it is failing with ENODEV. ```sh $ strace aseqsend -p 32:0 'F0 43 7D 10 41 30 01 00 00 01 F7' ... open("/dev/snd/seq", O_WRONLY|O_LARGEFILE|O_CLOEXEC) = 3 fcntl(3, F_SETFD, FD_CLOEXEC) = 0 ioctl(3, SNDRV_SEQ_IOCTL_PVERSION, 0x7fffef1ae678) = 0 ioctl(3, SNDRV_SEQ_IOCTL_USER_PVERSION, 0x7fffef1ae67c) = 0 mmap(NULL, 20480, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_ANONYMOUS, -1, 0) = 0x7f6ae25f7000 ioctl(3, SNDRV_SEQ_IOCTL_CLIENT_ID, 0x7fffef1ae67c) = 0 ioctl(3, SNDRV_SEQ_IOCTL_RUNNING_MODE, 0x7fffef1ae680) = 0 ioctl(3, SNDRV_SEQ_IOCTL_GET_CLIENT_INFO, 0x7fffef1ae940) = 0 ioctl(3, SNDRV_SEQ_IOCTL_SET_CLIENT_INFO, 0x7fffef1ae940) = 0 ioctl(3, SNDRV_SEQ_IOCTL_GET_CLIENT_INFO, 0x7fffef1ae940) = 0 ioctl(3, SNDRV_SEQ_IOCTL_SET_CLIENT_INFO, 0x7fffef1ae940) = 0 ioctl(3, SNDRV_SEQ_IOCTL_CREATE_PORT, 0x7fffef1ae960) = 0 write(3, "\202\4\0\375\0\0\0\0\0\0\0\0\0\0 \0\v\0\0\0\320\212WN\361U\0\0\360C}\20"..., 39) = -1 ENODEV (No such device) nanosleep({tv_sec=0, tv_nsec=1000000}, 0x7fffef1aea20) = 0 close(3) = 0 munmap(0x7f6ae25f7000, 20480) = 0 exit_group(0) = ? +++ exited with 0 +++ $ ``` `amidi -S ...` works as expected. If I modify `aseqsend` to first subscribe to the port, it seems to work: ```diff diff --git a/seq/aseqsend/aseqsend.c b/seq/aseqsend/aseqsend.c index 92354eb..ad0a636 100644 --- a/seq/aseqsend/aseqsend.c +++ b/seq/aseqsend/aseqsend.c @@ -364,6 +364,7 @@ int main(int argc, char *argv[]) char do_port_list = 0; char verbose = 0; int k; + int err; while ((c = getopt_long(argc, argv, "hi:Vvlp:s:u:", long_options, NULL)) != -1) { switch (c) { @@ -439,6 +440,8 @@ int main(int argc, char *argv[]) error("Unable to parse port name!"); exit(EXIT_FAILURE); } + err = snd_seq_connect_to(seq, 0, addr.client, addr.port); + check_snd("connect to port", err); sent_data_c = 0; //counter of actually sent bytes ``` From what I've read, I don't think it should be necessary to subscribe to a port to send it messages. However, I did find one sentence in the alsa-lib docs that seem to indicate that it is needed for hardware ports: > There is another subscription type for opposite direction: Suppose a MIDI sequencer program which sends events to a MIDI output device. In ALSA system, MIDI device is not opened until the associated MIDI port is accessed. Thus, in order to activate MIDI device, we have to subscribe to MIDI port for write. After this connection is established, events will be properly sent to MIDI output device. This seems to be a pretty basic use of `aseqsend`, so I am a bit surprised that it wasn't working. Is the subscription the right way to fix the problem, or am I doing something else wrong? Issue URL : https://github.com/alsa-project/alsa-utils/issues/300 Repository URL: https://github.com/alsa-project/alsa-utils

2 1

[PATCH AUTOSEL 6.1 02/13] ASoC: amd: yc: Add quirk for Lenovo Yoga Pro 7 14ASP9
by Sasha Levin 30 Oct '25

30 Oct '25

From: Talhah Peerbhai <talhah.peerbhai(a)gmail.com> [ Upstream commit a28206060dc5848a1a2a15b7f6ac6223d869084d ] Similar to many other Lenovo models with AMD chips, the Lenovo Yoga Pro 7 14ASP9 (product name 83HN) requires a specific quirk to ensure internal mic detection. This patch adds a quirk fixing this. Signed-off-by: Talhah Peerbhai <talhah.peerbhai(a)gmail.com> Link: https://patch.msgid.link/20250515222741.144616-1-talhah.peerbhai@gmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- Based on my analysis of this commit, I can provide a comprehensive assessment for backporting. **YES** This commit should be backported to stable kernel trees for the following reasons: **1. Identical Pattern to Previously Backported Commits:** All five similar commits provided in the historical context (commits for Lenovo models 82UU, 82, 82YM, 83AS, and 21M1) have "Backport Status: YES" and follow the exact same pattern as this commit. This establishes a clear precedent for backporting such changes. **2. Hardware Enablement Bug Fix:** The commit fixes a clear user-affecting bug where the internal microphone on the Lenovo Yoga Pro 7 14ASP9 (product name 83HN) is non-functional. This is a hardware compatibility issue that directly impacts users' ability to use their device's built-in microphone. **3. Minimal and Contained Change:** The code change is extremely small and safe: - Adds only a single 7-line quirk table entry to `sound/soc/amd/yc/acp6x-mach.c` - The change is purely additive - no existing code is modified - The quirk entry follows the exact same pattern as dozens of other entries in the same table - Zero risk of regression to existing functionality **4. Critical Subsystem with Safe Pattern:** While this touches audio (a critical subsystem), the change follows a well-established, safe pattern used extensively in this driver. The quirk table approach is designed specifically for adding device-specific compatibility without affecting other hardware. **5. Clear User Benefit:** Users with this specific Lenovo model will have their internal microphone functionality restored, which is essential for video calls, voice recording, and other audio input tasks. **6. Code Quality Improvement:** The commit also includes a minor whitespace fix (changing spaces to tab at line 350), improving code formatting consistency. **7. Follows Stable Tree Rules:** - Important bugfix: ✓ (enables hardware functionality) - Minimal risk: ✓ (purely additive quirk entry) - Small and contained: ✓ (7 lines added) - No architectural changes: ✓ - Confined to subsystem: ✓ (AMD YC audio driver) The commit message clearly explains the issue and solution, and the change is identical in nature to numerous other successfully backported commits for similar Lenovo audio quirks. This represents a textbook example of a stable-appropriate hardware enablement fix. sound/soc/amd/yc/acp6x-mach.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) diff --git a/sound/soc/amd/yc/acp6x-mach.c b/sound/soc/amd/yc/acp6x-mach.c index 1f94269e121af..d5dc1d48fca94 100644 --- a/sound/soc/amd/yc/acp6x-mach.c +++ b/sound/soc/amd/yc/acp6x-mach.c @@ -304,6 +304,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "83AS"), } }, + { + .driver_data = &acp6x_card, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_MATCH(DMI_PRODUCT_NAME, "83HN"), + } + }, { .driver_data = &acp6x_card, .matches = { @@ -353,7 +360,7 @@ static const struct dmi_system_id yc_acp_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "M5402RA"), } }, - { + { .driver_data = &acp6x_card, .matches = { DMI_MATCH(DMI_BOARD_VENDOR, "ASUSTeK COMPUTER INC."), -- 2.39.5

2 1

[alsa-devel] Applied "ASoC: tlv320aic32x4: Model CODEC_CLKIN in CCF" to the asoc tree
by Mark Brown 29 Oct '25

29 Oct '25

The patch ASoC: tlv320aic32x4: Model CODEC_CLKIN in CCF has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From fd2df3aeafa4b4cc468d58e147e0822967034b71 Mon Sep 17 00:00:00 2001 From: Annaliese McDermond <nh6z(a)nh6z.net> Date: Thu, 21 Mar 2019 17:58:46 -0700 Subject: [PATCH] ASoC: tlv320aic32x4: Model CODEC_CLKIN in CCF Model and manage codec clock input as a component in the Core Clock Framework. This should allow us to do some more complex clock management and power control. Also, some of the on-board chip clocks can be exposed to the outside, and this change will make those clocks easier to consume by other parts of the kernel. Signed-off-by: Annaliese McDermond <nh6z(a)nh6z.net> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/tlv320aic32x4-clk.c | 34 ++++++++++++++++++++++++++++ sound/soc/codecs/tlv320aic32x4.c | 18 +++++++++++---- 2 files changed, 47 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/tlv320aic32x4-clk.c b/sound/soc/codecs/tlv320aic32x4-clk.c index 5e495fc8d931..cded85009c8c 100644 --- a/sound/soc/codecs/tlv320aic32x4-clk.c +++ b/sound/soc/codecs/tlv320aic32x4-clk.c @@ -265,6 +265,30 @@ static const struct clk_ops aic32x4_pll_ops = { .get_parent = clk_aic32x4_pll_get_parent, }; +static int clk_aic32x4_codec_clkin_set_parent(struct clk_hw *hw, u8 index) +{ + struct clk_aic32x4 *mux = to_clk_aic32x4(hw); + + return regmap_update_bits(mux->regmap, + AIC32X4_CLKMUX, + AIC32X4_CODEC_CLKIN_MASK, index << AIC32X4_CODEC_CLKIN_SHIFT); +} + +static u8 clk_aic32x4_codec_clkin_get_parent(struct clk_hw *hw) +{ + struct clk_aic32x4 *mux = to_clk_aic32x4(hw); + unsigned int val; + + regmap_read(mux->regmap, AIC32X4_CLKMUX, &val); + + return (val & AIC32X4_CODEC_CLKIN_MASK) >> AIC32X4_CODEC_CLKIN_SHIFT; +} + +static const struct clk_ops aic32x4_codec_clkin_ops = { + .set_parent = clk_aic32x4_codec_clkin_set_parent, + .get_parent = clk_aic32x4_codec_clkin_get_parent, +}; + static struct aic32x4_clkdesc aic32x4_clkdesc_array[] = { { .name = "pll", @@ -274,6 +298,14 @@ static struct aic32x4_clkdesc aic32x4_clkdesc_array[] = { .ops = &aic32x4_pll_ops, .reg = 0, }, + { + .name = "codec_clkin", + .parent_names = + (const char *[]) { "mclk", "bclk", "gpio", "pll" }, + .num_parents = 4, + .ops = &aic32x4_codec_clkin_ops, + .reg = 0, + }, }; static struct clk *aic32x4_register_clk(struct device *dev, @@ -314,6 +346,8 @@ int aic32x4_register_clocks(struct device *dev, const char *mclk_name) */ aic32x4_clkdesc_array[0].parent_names = (const char* []) { mclk_name, "bclk", "gpio", "din" }; + aic32x4_clkdesc_array[1].parent_names = + (const char *[]) { mclk_name, "bclk", "gpio", "pll" }; for (i = 0; i < ARRAY_SIZE(aic32x4_clkdesc_array); ++i) aic32x4_register_clk(dev, &aic32x4_clkdesc_array[i]); diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index 7cf8c7cedfe1..5496e4e080f4 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -735,12 +735,9 @@ static int aic32x4_setup_clocks(struct snd_soc_component *component, aic32x4_set_processing_blocks(component, aic32x4_divs[i].r_block, aic32x4_divs[i].p_block); - /* PLL as CODEC_CLKIN */ - snd_soc_component_update_bits(component, AIC32X4_CLKMUX, - AIC32X4_CODEC_CLKIN_MASK, - AIC32X4_CODEC_CLKIN_PLL << AIC32X4_CODEC_CLKIN_SHIFT); /* DAC_MOD_CLK as BDIV_CLKIN */ - snd_soc_component_update_bits(component, AIC32X4_IFACE3, AIC32X4_BDIVCLK_MASK, + snd_soc_component_update_bits(component, AIC32X4_IFACE3, + AIC32X4_BDIVCLK_MASK, AIC32X4_DACMOD2BCLK << AIC32X4_BDIVCLK_SHIFT); /* NDAC divider value */ @@ -987,6 +984,15 @@ static int aic32x4_component_probe(struct snd_soc_component *component) { struct aic32x4_priv *aic32x4 = snd_soc_component_get_drvdata(component); u32 tmp_reg; + int ret; + + struct clk_bulk_data clocks[] = { + { .id = "codec_clkin" }, + }; + + ret = devm_clk_bulk_get(component->dev, ARRAY_SIZE(clocks), clocks); + if (ret) + return ret; if (gpio_is_valid(aic32x4->rstn_gpio)) { ndelay(10); @@ -999,6 +1005,8 @@ static int aic32x4_component_probe(struct snd_soc_component *component) if (aic32x4->setup) aic32x4_setup_gpios(component); + clk_set_parent(clocks[0].clk, clocks[1].clk); + /* Power platform configuration */ if (aic32x4->power_cfg & AIC32X4_PWR_MICBIAS_2075_LDOIN) { snd_soc_component_write(component, AIC32X4_MICBIAS, -- 2.20.1

6 7

[PATCH v4 1/2] ASoC: tas2781: Add tas2118, tas2x20, tas5825 support
by Baojun Xu 09 Sep '25

09 Sep '25

Add tas2020, tas2118, tas2120, tas2320, tas2570, tas2572, tas5825 tas5827 support in tas2781 driver. Tas2118, tas2x20, tas257x have no on-chip DSP, tas582x have on-chip DSP but have no calibration required stereo smart amplifier. Signed-off-by: Baojun Xu <baojun.xu(a)ti.com> --- v4: - Change description for adding tas2570, tas2572 and tas5827 - Add tas2570, tas2572, tas5827 in device enumerate - Change the check for reset value, from "== TAS5825" to ">= TAS5825" as TAS5827 was added. - Add tas2570, tas2572, tas5827 in i2c_device_id list - Add tas2570, tas2572, tas5827 in of_device_id list - Change max of device ID from TAS2781 to TAS_OTHERS - Change the check for non-DSP chip, from "< TAS2563" to switch case, as TAS2570 and TAS2572 also have no on-chip DSP - Add TAS5827 in comments for calibration check - Change the check for on-chip DSP, from ">= TAS2563" to switch case, as TAS2570 and TAS2572 also have no on-chip DSP - Add tas2570, tas2572, tas5827 initialyze in codec probe - Add tas2570, tas2572, tas5827 in ACPI device id list v3: - No update. v2: - Update the mail list for maintainers of yaml file --- include/sound/tas2781.h | 14 +- include/sound/tas2x20-tlv.h | 259 ++++++++++++++++++++++++++ sound/soc/codecs/tas2781-comlib-i2c.c | 2 + sound/soc/codecs/tas2781-i2c.c | 184 +++++++++++++----- 4 files changed, 410 insertions(+), 49 deletions(-) create mode 100644 include/sound/tas2x20-tlv.h diff --git a/include/sound/tas2781.h b/include/sound/tas2781.h index f0aefc04a957..ddd997ac3216 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 +112,17 @@ #define TAS2781_RUNTIME_RE_REG TASDEVICE_REG(0x64, 0x63, 0x44) enum audio_device { + TAS2020, + TAS2118, + TAS2120, + TAS2320, TAS2563, + TAS2570, + TAS2572, TAS2781, + TAS5825, + TAS5827, + TAS_OTHERS, }; enum dspbin_type { @@ -194,6 +205,7 @@ 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..b3fd7350143b 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..1539b70881d1 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 +100,32 @@ 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 }, + { "tas2570", TAS2570 }, + { "tas2572", TAS2572 }, { "tas2781", TAS2781 }, + { "tas5825", TAS5825 }, + { "tas5827", TAS5827 }, {} }; 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,tas2570" }, + { .compatible = "ti,tas2572" }, { .compatible = "ti,tas2781" }, + { .compatible = "ti,tas5825" }, + { .compatible = "ti,tas5827" }, {}, }; MODULE_DEVICE_TABLE(of, tasdevice_of_match); @@ -797,7 +815,7 @@ 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 +841,15 @@ 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 +859,7 @@ 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 +885,7 @@ 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 +894,7 @@ 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 +910,6 @@ 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 +920,16 @@ 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 +939,15 @@ 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 +955,7 @@ 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), }; @@ -968,8 +1000,8 @@ 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.max = TAS2781; + uinfo->value.integer.min = TAS2020; + uinfo->value.integer.max = TAS_OTHERS; return 0; } @@ -1168,9 +1200,9 @@ 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 +1260,19 @@ 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 +1632,16 @@ 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 TAS2118/2X20/257X. */ + switch (tas_priv->chip_id) { + case TAS2020: + case TAS2118: + case TAS2120: + case TAS2320: + case TAS2570: + case TAS2572: + 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 +1663,37 @@ 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/TAS5827. */ + 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); @@ -1659,8 +1717,14 @@ static void tasdevice_fw_ready(const struct firmware *fmw, #endif out: if (tas_priv->fw_state == TASDEVICE_RCA_FW_OK) { - /* If DSP FW fail, DSP kcontrol won't be created. */ - tasdevice_dsp_remove(tas_priv); + switch (tas_priv->chip_id) { + case TAS2563: + case TAS2781: + case TAS5825: + case TAS5827: + /* If DSP FW fail, DSP kcontrol won't be created. */ + tasdevice_dsp_remove(tas_priv); + } } mutex_unlock(&tas_priv->codec_lock); release_firmware(fmw); @@ -1804,13 +1868,30 @@ 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: + case TAS2570: + case TAS2572: + 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: + case TAS5827: + 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 +1931,6 @@ 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 +2046,16 @@ 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 }, + { "TXNW2570", TAS2570 }, + { "TXNW2572", TAS2572 }, + { "TXNW2781", TAS2781 }, + { "TXNW5825", TAS5825 }, + { "TXNW5827", TAS5827 }, {}, }; -- 2.43.0

4 6

ALSAfirewire broken
by M. Armsby 04 Sep '25

04 Sep '25

Hello everyone. 2 Firewire problems with ALSAfirewire. 1. Firewire Midi was working fine with kernel 5.10 with Reaper. Then obviously a lot changed with the Firewire Stack. The midi output from Reaper has not changed according to it's developer - Justin. The problem is direct ALSA-Midi. We tested the midi app from www.alsa-project.org/alsa-doc/alsa-lib/_2test_2rawmidi_8c-example.html#a0 yesterday and it freezes Debian and just breaks Arch with kernel 6.15 Thread at Reaper Bugs forum: https://forum.cockos.com/showthread.php?t=300278 Has low level ALSA-Midi API changed? 2. If using Pipewire-Jack with ALSAfirewire the Reaper midi output works fine but unfortunately Pipewire adds 90ms delay when playing back through any Firewire interface compared with an onboard soundcard. The pipewire dev Wim Tayman, says it's ALSAfirewire fault but I proved it is not - see thread: https://gitlab.freedesktop.org/pipewire/pipewire/-/issues?show=eyJpaWQiOiI0… The ALSA recording and playback is confirmed by my tests to be working under 10ms RTT with Echo, Motu and Bebop interfaces. Is this problem related to first? Low level communication error? Cheers - and let me know if I can help in any way, although I'm not a programmer. P. S. I hope you can read my post as I loaded up K-9 eMail client so as to deliver text only. -- Martin Armsby

3 11

[PATCH v1] ALSA: hda/tas2781: Fix the order of TAS2563 calibrated-data
by Shenghao Ding 01 Sep '25

01 Sep '25

A bug reported by one of my customers that the order of TAS2563 calibrated-data is incorrect, the correct way is to move R0_Low_%d and insert it between R0_%d and InvR0_%d. 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> --- sound/hda/codecs/side-codecs/tas2781_hda_i2c.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c b/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c index dbd71e173119..853ff5d87966 100644 --- a/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c +++ b/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c @@ -313,7 +313,7 @@ static int tas2563_save_calibration(struct tas2781_hda *h) { efi_guid_t efi_guid = tasdev_fct_efi_guid[LENOVO]; char *vars[TASDEV_CALIB_N] = { - "R0_%d", "InvR0_%d", "R0_Low_%d", "Power_%d", "TLim_%d" + "R0_%d", "R0_Low_%d", "InvR0_%d", "Power_%d", "TLim_%d" }; efi_char16_t efi_name[TAS2563_CAL_VAR_NAME_MAX]; unsigned long max_size = TAS2563_CAL_DATA_SIZE; -- 2.43.0

3 2

[PATCH 1/2] ALSA: hda: tas2781: fix tas2563 EFI data endianness
by Gergo Koteles 30 Aug '25

30 Aug '25

Before conversion to unify the calibration data management, the tas2563_apply_calib() function performed the big endian conversion and wrote the calibration data to the device. The writing is now done by the common tasdev_load_calibrated_data() function, but without conversion. Put the values into the calibration data buffer with the expected endianness. Fixes: 4fe238513407 ("ALSA: hda/tas2781: Move and unified the calibrated-data getting function for SPI and I2C into the tas2781_hda lib") Cc: <stable(a)vger.kernel.org> Signed-off-by: Gergo Koteles <soyer(a)irl.hu> --- sound/hda/codecs/side-codecs/tas2781_hda_i2c.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c b/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c index e34b17f0c9b9..1eac751ab2a6 100644 --- a/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c +++ b/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c @@ -310,6 +310,7 @@ static int tas2563_save_calibration(struct tas2781_hda *h) struct cali_reg *r = &cd->cali_reg_array; unsigned int offset = 0; unsigned char *data; + __be32 bedata; efi_status_t status; unsigned int attr; int ret, i, j, k; @@ -351,6 +352,8 @@ static int tas2563_save_calibration(struct tas2781_hda *h) i, j, status); return -EINVAL; } + bedata = cpu_to_be32(*(uint32_t *)&data[offset]); + memcpy(&data[offset], &bedata, sizeof(bedata)); offset += TAS2563_CAL_DATA_SIZE; } } -- 2.51.0

2 2

[BUG] ALSA FireWire: poll timeout causes JACK to stop (6.10 – 6.17-rc2)
by Илья Кузнецов 30 Aug '25

30 Aug '25

------------------------------ Hello ALSA developers, I would like to report a reproducible issue with *snd-fireface* on Linux kernels 6.10 through 6.17-rc2. ------------------------------ System Information - *Kernel versions tested:* 6.10 → 6.17-rc2 - *Audio interface:* RME Fireface 800 - *Driver in use:* snd-fireface - *Audio server:* JACK (tested with jackd and QJackCtl) - *Distribution:* Debian 12 - *Kernel config:* ALSA FireWire drivers enabled, no FFADO in use - *Chip 1394: *Texas Instrumets ------------------------------ Steps to Reproduce 1. Boot kernel 6.10 … 6.17-rc2 with snd-fireface loaded. 2. Start JACK with ALSA backend, for example: jackd -d alsa -d hw:Fireface -p 64 -n 3 3. Use the system normally — during playback, recording, or even when idle. The issue may occur at any time without apparent external cause. Actual Behavior After some idle time, ALSA stops responding and JACK prints: ERROR: ALSA: poll time out, polled for ... usecs ERROR: JackAudioDriver::ProcessAsync: read error, stopping... JACK immediately terminates. The Fireface 800 does not recover until JACK is restarted. ------------------------------ Expected Behavior The ALSA FireWire driver should maintain the stream and allow JACK to continue running without manual restart, even when idle. ------------------------------ Notes 4. With *FFADO* backend (jackd -d firewire), the same hardware works reliably and does not exhibit this issue. 5. This suggests that the problem lies in the ALSA FireWire streaming engine rather than the hardware itself. 6. The problem has been reproducible across all tested kernels (vanila, rt, liquorix) from *6.10 through 6.17-rc2*. ------------------------------ Request Could you please confirm if this issue is known? Is there any ongoing work on the ALSA FireWire streaming engine ( snd-fireface) to address these poll timeout conditions? I can provide additional logs (dmesg, journalctl) or test patches if needed. Thank you, Ilya Kuznetsov.

1 0

URGENT: ALSAfirewire Drivers Introduce 90ms delay - Linux is Unusable for Pro Audio
by M. Armsby 29 Aug '25

29 Aug '25

Dear maintainers, Linux is being ridiculed in the professional audio community — not because it can't do low-latency audio, but because the ALSA FireWire stack actively breaks it. Professional gear costing thousands of dollars like Motu RME Focusrite etc. etc. Since kernel 5.16+, FireWire audio devices are correctly detected and functional — but they are burdened with an artificial +90ms latency offset due to missing timing metadata (e.g., cycle-match delay, transmission latency) in the driver's interface to userspace. PipeWire and WirePlumber rely on accurate latency reporting. When it's missing, they fall back to worst-case assumptions — adding ~90ms of delay. This is NOT a PipeWire bug. It is a FAILURE of the ALSA FireWire drivers to expose timing data that the hardware and firmware already provide. This bug: - Renders high-end interfaces (RME, Focusrite, Tascam, Motu, Terratec) unusable for real-time monitoring - Forces users to hack WirePlumber with hardcoded quantum rules - Damages Linux's credibility in music production - Makes Linux a JOKE in studios worldwide There is NO excuse for this in 2025. We have workarounds (see: https://forum.cockos.com/showthread.php?t=274978), but they are fragile, device-specific, and do not scale. We demand: 1. Immediate exposure of accurate device latency via ALSA control or hwdep interface 2. Proper reporting of transmission/cycle delay to userspace 3. A timeline for fixing this in mainline This is not a "nice-to-have". It is a **critical defect** in a core multimedia subsystem now using pipewire as standard. Linux can and must do better. Many are waiting to move to Linux, finally leaving windows. — Martin Armsby (Martin A - maa) Professional Audio Producer / Linux Advocate Please see: Imagine if Linux Audio just worked - Firewire <https://forum.cockos.com/showthread.php?t=302305> and <https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/4785> Thank you for reading. -- Martin Armsby -- Martin Armsby

1 1

[GIT PULL] ASoC fixes for v6.17-rc3
by Mark Brown 29 Aug '25

29 Aug '25

The following changes since commit 1b237f190eb3d36f52dffe07a40b5eb210280e00: Linux 6.17-rc3 (2025-08-24 12:04:12 -0400) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.17-rc3 for you to fetch changes up to 3e7fd1febc3156d3d98fba229399a13b12d69707: ASoC: SOF: Intel: WCL: Add the sdw_process_wakeen op (2025-08-28 21:15:49 +0200) ---------------------------------------------------------------- ASoC: Fixes for v6.17 The main fixes here are for some of the cleanups done in the core in this release, we had broken component lookup in the case with a single bus and DMA controller. Otherwise it's driver specific changes, the shortlogs for the Intel WCL and rsnd drivers look like minor cleanups but are actually bugfixes (adding an op needed for correct functionality and reverting an inappropriate helper usage). ---------------------------------------------------------------- Ajye Huang (1): ASoC: SOF: Intel: WCL: Add the sdw_process_wakeen op Cryolitia PukNgae (1): ASoC: codecs: idt821034: fix wrong log in idt821034_chip_direction_output() Kuninori Morimoto (3): ASoC: soc-core: care NULL dirver name on snd_soc_lookup_component_nolocked() ASoC: soc-core: tidyup snd_soc_lookup_component_nolocked() ASoC: rsnd: tidyup direction name on rsnd_dai_connect() Mark Brown (1): ASoC: fixup snd_soc_lookup_component_nolocked() sound/soc/codecs/idt821034.c | 2 +- sound/soc/renesas/rcar/core.c | 2 +- sound/soc/soc-core.c | 25 +++++++++++++++---------- sound/soc/sof/intel/ptl.c | 1 + 4 files changed, 18 insertions(+), 12 deletions(-)

2 1

[PATCH v3 1/2] ASoC: tas2781: Add tas2118, tas2x20, tas5825 support
by Baojun Xu 29 Aug '25

29 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> --- v3: - No update. 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

1 1

when i click apply HDAjackretask is not respoding
by GitHub issues - opened 29 Aug '25

29 Aug '25

alsa-project/alsa-tools issue #40 was opened from m33rxj: I'm using a hp envy x360 and when i try change and override the pins to allow the front speakers and click apply, it says on the gui 'its not responding' it happened to me on ubuntu and now on cachyos. I'm not sure if my codec is unsupported? its RealTek ALC245 I'm trying to work on but even on the intel codec I get the same error. I forgot to mention I have pipewire but I'm unsure what I used for ubuntu at the time Any chance I can get some help incase I'm doing something wrong. <img width="656" height="869" alt="Image" src="https://github.com/user-attachments/assets/dfdd4d42-3979-4e89-9234-b49726d2…" /> <img width="808" height="295" alt="Image" src="https://github.com/user-attachments/assets/42f65770-e19d-40d3-8ec7-0521e345…" /> Issue URL : https://github.com/alsa-project/alsa-tools/issues/40 Repository URL: https://github.com/alsa-project/alsa-tools

1 0

[PATCH v3] ASoC: SOF: Intel: WCL: Add the sdw_process_wakeen op
by Ajye Huang 28 Aug '25

28 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") Acked-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> 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") v3: Add Acked-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.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 1

ucm2: MediaTek: mt8391-evk: Add alsa-ucm support
by GitHub pull_request - opened 28 Aug '25

28 Aug '25

alsa-project/alsa-ucm-conf pull request #607 was opened from zoran055: Add alsa-ucm support for the MediaTek mt8391-evk platform. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/607 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/607.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v2] ALSA: hda/tas2781: Fix EFI name for calibration beginning with 1 instead of 0
by Shenghao Ding 27 Aug '25

27 Aug '25

A bug reported by one of my customers that EFI name beginning with 0 instead of 1. 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> --- v2: - remove unrelated change v1: - Fix EFI name beginning with 1 instead of 0 - Add extra comments on EFI name for calibration - Remove an extra space --- sound/hda/codecs/side-codecs/tas2781_hda_i2c.c | 5 +++-- 1 file changed, 3 insertions(+), 2 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..635cbd8820ac 100644 --- a/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c +++ b/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c @@ -340,7 +340,8 @@ 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) { dev_err(p->dev, "%s: Read %s failed\n", @@ -349,7 +350,7 @@ static int tas2563_save_calibration(struct tas2781_hda *h) } /* * 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

4 5

Add UCM for RME Fireface UCX
by GitHub pull_request - opened 27 Aug '25

27 Aug '25

alsa-project/alsa-ucm-conf pull request #606 was opened from vascosantos: Adds UCM for RME Fireface UCX (heavily based on RME Fireface UCX II config). Device is assumed in CC mode. #605 Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/606 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/606.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Feature request: Add support for RME Fireface UCX
by GitHub issues - opened 27 Aug '25

27 Aug '25

alsa-project/alsa-ucm-conf issue #605 was opened from vascosantos: I see there is already support for the RME Fireface UCX II. Could you please add support for the previous version, RME Fireface UCX? Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/605 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v3] ALSA: hda/tas2781: Fix EFI name for calibration beginning with 1 instead of 0
by Shenghao Ding 27 Aug '25

27 Aug '25

A bug reported by one of my customers that EFI name beginning with 0 instead of 1. 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> --- v3: - remove unrelated change v1: - Fix EFI name beginning with 1 instead of 0 - Add extra comments on EFI name for calibration --- sound/hda/codecs/side-codecs/tas2781_hda_i2c.c | 4 ++-- 1 file changed, 2 insertions(+), 2 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..dbd71e173119 100644 --- a/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c +++ b/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c @@ -340,8 +340,8 @@ 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) { dev_err(p->dev, "%s: Read %s failed\n", __func__, var8); -- 2.43.0

2 1

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

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[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

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[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

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[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

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