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[PATCH v1] ALSA: hda: Add TAS5825 support
by Baojun Xu 11 Aug '25

11 Aug '25

Add TAS5825 support in TI's HDA driver. TAS5825 is an on-chip DSP, but no calibration is required, and no global address support smart amplifier devices. Signed-off-by: Baojun Xu <baojun.xu(a)ti.com> --- include/sound/tas2781-dsp.h | 3 + include/sound/tas2781.h | 4 +- include/sound/tas5825-tlv.h | 24 +++++++ .../hda/codecs/side-codecs/tas2781_hda_i2c.c | 28 ++++++++ sound/soc/codecs/tas2781-fmwlib.c | 67 ++++++++++++++++++- 5 files changed, 121 insertions(+), 5 deletions(-) create mode 100644 include/sound/tas5825-tlv.h diff --git a/include/sound/tas2781-dsp.h b/include/sound/tas2781-dsp.h index c3a9efa73d5d..49bbf24d6559 100644 --- a/include/sound/tas2781-dsp.h +++ b/include/sound/tas2781-dsp.h @@ -34,6 +34,7 @@ #define PPC3_VERSION_TAS2781_BASIC_MIN 0x14600 #define PPC3_VERSION_TAS2781_ALPHA_MIN 0x4a00 #define PPC3_VERSION_TAS2781_BETA_MIN 0x19400 +#define PPC3_VERSION_TAS5825_BASE 0x114200 #define TASDEVICE_DEVICE_SUM 8 #define TASDEVICE_CONFIG_SUM 64 @@ -53,6 +54,8 @@ enum tasdevice_dsp_dev_idx { TASDEVICE_DSP_TAS_2781_DUAL_MONO, TASDEVICE_DSP_TAS_2781_21, TASDEVICE_DSP_TAS_2781_QUAD, + TASDEVICE_DSP_TAS_5825_MONO, + TASDEVICE_DSP_TAS_5825_DUAL, TASDEVICE_DSP_TAS_MAX_DEVICE }; diff --git a/include/sound/tas2781.h b/include/sound/tas2781.h index 3875e92f1ec5..f0aefc04a957 100644 --- a/include/sound/tas2781.h +++ b/include/sound/tas2781.h @@ -49,9 +49,9 @@ #define TASDEVICE_REG(book, page, reg) (((book * 256 * 128) + \ (page * 128)) + reg) -/* Software Reset */ +/* Software Reset, compatble with new device (TAS5825). */ #define TASDEVICE_REG_SWRESET TASDEVICE_REG(0x0, 0x0, 0x01) -#define TASDEVICE_REG_SWRESET_RESET BIT(0) +#define TASDEVICE_REG_SWRESET_RESET (BIT(0) | BIT(4)) /* Checksum */ #define TASDEVICE_CHECKSUM_REG TASDEVICE_REG(0x0, 0x0, 0x7e) diff --git a/include/sound/tas5825-tlv.h b/include/sound/tas5825-tlv.h new file mode 100644 index 000000000000..95f2d3fad120 --- /dev/null +++ b/include/sound/tas5825-tlv.h @@ -0,0 +1,24 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +// +// ALSA SoC Texas Instruments TAS5825 Audio Smart Amplifier +// +// Copyright (C) 2025 Texas Instruments Incorporated +// https://www.ti.com +// +// The TAS5825 hda driver implements for one or two TAS5825 chips. +// +// Author: Baojun Xu <baojun.xu(a)ti.com> +// + +#ifndef __TAS5825_TLV_H__ +#define __TAS5825_TLV_H__ + +#define TAS5825_DVC_LEVEL TASDEVICE_REG(0x0, 0x0, 0x4c) +#define TAS5825_AMP_LEVEL TASDEVICE_REG(0x0, 0x0, 0x54) + +static const __maybe_unused DECLARE_TLV_DB_SCALE( + tas5825_dvc_tlv, -10300, 50, 0); +static const __maybe_unused DECLARE_TLV_DB_SCALE( + tas5825_amp_tlv, -1550, 50, 0); + +#endif diff --git a/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c b/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c index 45ac5e41bd4f..71ee946eb748 100644 --- a/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c +++ b/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c @@ -26,6 +26,7 @@ #include <sound/tlv.h> #include <sound/tas2770-tlv.h> #include <sound/tas2781-tlv.h> +#include <sound/tas5825-tlv.h> #include "hda_local.h" #include "hda_auto_parser.h" @@ -50,6 +51,7 @@ enum device_chip_id { HDA_TAS2563, HDA_TAS2770, HDA_TAS2781, + HDA_TAS5825, HDA_OTHERS }; @@ -272,6 +274,17 @@ static const struct snd_kcontrol_new tas2781_snd_controls[] = { tas2781_force_fwload_get, tas2781_force_fwload_put), }; +static const struct snd_kcontrol_new tas5825_snd_controls[] = { + ACARD_SINGLE_RANGE_EXT_TLV("Speaker Analog Volume", TAS5825_AMP_LEVEL, + 0, 0, 31, 1, tas2781_amp_getvol, + tas2781_amp_putvol, tas5825_amp_tlv), + ACARD_SINGLE_RANGE_EXT_TLV("Speaker Digital Volume", TAS5825_DVC_LEVEL, + 0, 0, 254, 1, tas2781_amp_getvol, + tas2781_amp_putvol, tas5825_dvc_tlv), + ACARD_SINGLE_BOOL_EXT("Speaker Force Firmware Load", 0, + tas2781_force_fwload_get, tas2781_force_fwload_put), +}; + static const struct snd_kcontrol_new tasdevice_prof_ctrl = { .name = "Speaker Profile Id", .iface = SNDRV_CTL_ELEM_IFACE_CARD, @@ -501,6 +514,12 @@ static void tasdev_fw_ready(const struct firmware *fmw, void *context) ARRAY_SIZE(tas2781_snd_controls)); tasdevice_dspfw_init(context); break; + case HDA_TAS5825: + tasdev_add_kcontrols(tas_priv, hda_priv->snd_ctls, codec, + &tas5825_snd_controls[0], + ARRAY_SIZE(tas5825_snd_controls)); + tasdevice_dspfw_init(context); + break; case HDA_TAS2563: tasdevice_dspfw_init(context); break; @@ -628,6 +647,7 @@ static int tas2781_hda_i2c_probe(struct i2c_client *clt) } else if (strstarts(dev_name(&clt->dev), "i2c-TXNW2781:00-tas2781-hda.0")) { device_name = "TXNW2781"; + hda_priv->hda_chip_id = HDA_TAS2781; hda_priv->save_calibration = tas2781_save_calibration; tas_hda->priv->global_addr = TAS2781_GLOBAL_ADDR; } else if (strstr(dev_name(&clt->dev), "INT8866")) { @@ -639,6 +659,13 @@ static int tas2781_hda_i2c_probe(struct i2c_client *clt) hda_priv->hda_chip_id = HDA_TAS2563; hda_priv->save_calibration = tas2563_save_calibration; tas_hda->priv->global_addr = TAS2563_GLOBAL_ADDR; + } else if (strstarts(dev_name(&clt->dev), "i2c-TXNW5825")) { + /* + * TAS5825, integrated on-chip DSP without + * global I2C address and calibration supported. + */ + device_name = "TXNW5825"; + hda_priv->hda_chip_id = HDA_TAS5825; } else { return -ENODEV; } @@ -775,6 +802,7 @@ static const struct acpi_device_id tas2781_acpi_hda_match[] = { {"TIAS2781", 0 }, {"TXNW2770", 0 }, {"TXNW2781", 0 }, + {"TXNW5825", 0 }, {} }; MODULE_DEVICE_TABLE(acpi, tas2781_acpi_hda_match); diff --git a/sound/soc/codecs/tas2781-fmwlib.c b/sound/soc/codecs/tas2781-fmwlib.c index c9c1e608ddb7..d69faef8a4d7 100644 --- a/sound/soc/codecs/tas2781-fmwlib.c +++ b/sound/soc/codecs/tas2781-fmwlib.c @@ -91,7 +91,7 @@ struct blktyp_devidx_map { }; static const char deviceNumber[TASDEVICE_DSP_TAS_MAX_DEVICE] = { - 1, 2, 1, 2, 1, 1, 0, 2, 4, 3, 1, 2, 3, 4 + 1, 2, 1, 2, 1, 1, 0, 2, 4, 3, 1, 2, 3, 4, 1, 2 }; /* fixed m68k compiling issue: mapping table can save code field */ @@ -509,6 +509,56 @@ static int fw_parse_data_kernel(struct tasdevice_fw *tas_fmw, return offset; } +static int fw_parse_tas5825_program_data_kernel( + struct tasdevice_priv *tas_priv, struct tasdevice_fw *tas_fmw, + const struct firmware *fmw, int offset) +{ + struct tasdevice_prog *program; + unsigned int i; + + for (i = 0; i < tas_fmw->nr_programs; i++) { + program = &(tas_fmw->programs[i]); + if (offset + 72 > fmw->size) { + dev_err(tas_priv->dev, "%s: mpName error\n", __func__); + return -EINVAL; + } + /* Skip 65 unused byts*/ + offset += 65; + offset = fw_parse_data_kernel(tas_fmw, &(program->dev_data), + fmw, offset); + if (offset < 0) + return offset; + } + + return offset; +} + +static int fw_parse_tas5825_configuration_data_kernel( + struct tasdevice_priv *tas_priv, + struct tasdevice_fw *tas_fmw, const struct firmware *fmw, int offset) +{ + const unsigned char *data = fmw->data; + struct tasdevice_config *config; + unsigned int i; + + for (i = 0; i < tas_fmw->nr_configurations; i++) { + config = &(tas_fmw->configs[i]); + if (offset + 80 > fmw->size) { + dev_err(tas_priv->dev, "%s: mpName error\n", __func__); + return -EINVAL; + } + memcpy(config->name, &data[offset], 64); + /* Skip extra 8 bytes*/ + offset += 72; + offset = fw_parse_data_kernel(tas_fmw, &(config->dev_data), + fmw, offset); + if (offset < 0) + return offset; + } + + return offset; +} + static int fw_parse_program_data_kernel( struct tasdevice_priv *tas_priv, struct tasdevice_fw *tas_fmw, const struct firmware *fmw, int offset) @@ -1826,7 +1876,8 @@ static void dspbin_type_check(struct tasdevice_priv *tas_priv, else tas_priv->dspbin_typ = TASDEV_ALPHA; } - if (tas_priv->dspbin_typ != TASDEV_BASIC) + if ((tas_priv->dspbin_typ != TASDEV_BASIC) && + (ppcver < PPC3_VERSION_TAS5825_BASE)) tas_priv->fw_parse_fct_param_address = fw_parse_fct_param_address; } @@ -1837,7 +1888,17 @@ static int dspfw_default_callback(struct tasdevice_priv *tas_priv, int rc = 0; if (drv_ver == 0x100) { - if (ppcver >= PPC3_VERSION_BASE) { + if (ppcver >= PPC3_VERSION_TAS5825_BASE) { + tas_priv->fw_parse_variable_header = + fw_parse_variable_header_kernel; + tas_priv->fw_parse_program_data = + fw_parse_tas5825_program_data_kernel; + tas_priv->fw_parse_configuration_data = + fw_parse_tas5825_configuration_data_kernel; + tas_priv->tasdevice_load_block = + tasdevice_load_block_kernel; + dspbin_type_check(tas_priv, ppcver); + } else if (ppcver >= PPC3_VERSION_BASE) { tas_priv->fw_parse_variable_header = fw_parse_variable_header_kernel; tas_priv->fw_parse_program_data = -- 2.43.0

1 0

[alsa-devel] Applied "ASoC: tlv320aic32x4: Model CODEC_CLKIN in CCF" to the asoc tree
by Mark Brown 11 Aug '25

11 Aug '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

4 5

ALSAfirewire broken
by M. Armsby 10 Aug '25

10 Aug '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

2 3

ucm2: Qualcomm: add Radxa Dragon Q6A
by GitHub pull_request - opened 10 Aug '25

10 Aug '25

alsa-project/alsa-ucm-conf pull request #601 was opened from strongtz: This is an upcoming SBC (single board computer) based on QCS6490. Full specs: - Headphone jack using WCD9380 codec - Onboard DisplayPort to HDMI bridge using Radxa RA620 - MI2S0 available via the 40-Pin GPIO header (unused by default) - AMICs using WCD9380 codec via a dedicated MIC header (unused by default) The kernel device tree part is not yet but going to be upstreamed soon. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/601 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/601.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[HDA][HP EliteDesk 800 G5][8086:a348] – HDMI audio missing after BIOS update
by Hermanns Cyrille 09 Aug '25

09 Aug '25

Hello ALSA team, On an HP EliteDesk 800 G5 Desktop Mini with BIOS R21 Ver. 02.22.00, audio controller Intel Cannon Lake PCH cAVS [8086:a348], HDMI audio outputs are no longer detected after this BIOS update. System: - Linux Mint 22.1 Cinnamon 6.4.8 - Kernel: 6.8.0-71-generic - Codecs: Conexant CX20632 (analog), Intel Kabylake HDMI (digital) Issue: After boot, HDMI outputs appear as “Not connected”. HDMI audio works only if I use hdajackretask (Intel Kabylake HDMI → Show unconnected pins → set to HDMI/DisplayPort) and select “Install boot override”. ALSA report: http://alsa-project.org/db/?f=a1ee220062b23ad5c13c6d38cde3f1d046ce6260 Request: Would it be possible to add a quirk for this model so HDMI pins are enabled automatically at boot? Thank you for your help.

1 0

Problems with "snd_pcm_hw_params_set_rate_near" and "snd_pcm_hw_params_set_period_size_near" in x86_64 assembly.
by GitHub issues - opened 08 Aug '25

08 Aug '25

alsa-project/alsa-lib issue #471 was opened from sc1ntilla: ### **PREAMBLE:** I have never worked with any sound system before, so I'm almost sure that it's just me doing something wrong, but I can't see what, so I'm asking for help. I'm doing this project for fun, and to learn a lot of stuff, so feel free to give your opinions / advice if you want to. ### **WHAT HAPPENS:** When calling `snd_pcm_hw_params_set_rate_near` and `snd_pcm_hw_params_set_period_size_near` memory protection violation is issued. I find that strange, because `snd_pcm_hw_params_set_buffer_size_near` works perfectly fine, and it's similiar to these two. ### **WHAT I TRIED:** - Making sure that stack after call is on address divisible by 96 (by adding 8 to stack pointer), - Making sure that `*val` argument is on address divisible by 96, - Misaligning stack by 8, 16, 24 and 32 before calling these functions (by subtracting from stack pointer), - Providing 0 to `*val` argument, to rule out that writing to output crashes, - Providing valid output address to `*dir` argument to rule out that 0 is causing crashes, - Calling these functions in different order, - Trying to call `snd_pcm_set_params` instead, but it says _"Channels count (0) not available for PLAYBACK: Invalid argument"_ however, I'm sure that I provided a valid number (2). I even hardcoded it. But to no avail. Is it my misunderstanding of how alsa-lib works? Very likely. Is my assembly code faulty? Quite possible. I just can't see where the problem lies. If you see error somewhere, please let me know. ### **ADDITIONAL INFORMATION:** I have multiple audio outputs: monitors with built-in speakers, standalone speakers, and analog jack headphones. I hardcoded `hw:2,0` as device name to open. Sometimes it results in `snd_pcm_open` returning _"File descriptor in bad state"_. When I tried using `sysdefault` it often returns _"ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.14/work/alsa-lib-1.2.14/src/pcm/pcm_dmix.c:1000:(snd_pcm_dmix_open) unable to open slave"_ Below is .data section in which globals are stored, and source of function that will be used to play audio buffers. .data section: ``` .section .data .align 32 test2: .8byte 0 test: .8byte 0 SND_PCM_NONBLOCK: .8byte 0x1 SND_PCM_ASYNC: .8byte 0x2 SND_PCM_STREAM_PLAYBACK: .8byte 0 SND_PCM_FORMAT_U8: .8byte 1 SND_PCM_FORMAT_U16_LE: .8byte 4 SND_PCM_FORMAT_U24_LE: .8byte 8 SND_PCM_FORMAT_U32_LE: .8byte 12 SND_PCM_FORMAT_S8: .8byte 0 SND_PCM_FORMAT_S16_LE: .8byte 2 SND_PCM_FORMAT_S24_LE: .8byte 6 SND_PCM_FORMAT_S32_LE: .8byte 10 SND_PCM_ACCESS_MMAP_INTERLEAVED: .8byte 0 SND_PCM_STATE_PREPARED: .8byte 2 AUDIO_BUFFER_STATE_IDLE: .8byte 0 AUDIO_BUFFER_STATE_WRITING: .8byte 1 AUDIO_BUFFER_STATE_PLAYING: .8byte 2 t1: .8byte 0 t2: .8byte 0 alsa: alsa.pcm_handle: .8byte 0 alsa.hw_params: .8byte 0 alsa.last_error: .8byte 0 alsa.initialized: .8byte 0 alsa.device_name: .ascii "hw:2,0\0" audio_buffers.count_max: .8byte 16 amd64_audio_last_error: .8byte 0 ``` function: ``` amd64_audio_play: /* rdi: (input) address to audio data: * (+0/2) audio format (1: PCM, 2: IEEE754) * (+2/2) amount of channels * (+4/4) rate in hertz * (+8/4) bytes per sec (frequency * bytes per block) * (+12/2) bytes per block (number of channels * bits per sample / 8) * (+14/2) bits per sample * (+16/8) total size of samples data * (+24/8) address to samples data */ /* Align stack on 96byte boundary: */ push %r15 mov %rsp, %r15 and $-32, %rsp mov %rsp, %rax mov $96, %r14 xor %rdx, %rdx div %r14 sub %rdx, %rsp sub $96, %rsp mov %r15, (%rsp) /* Store registers: */ sub $192, %rsp call amd64_util_store_gpr_regs /* Create stack space: */ sub $192, %rsp mov %rdi, (%rsp) /* stack layout: * (+0/8) rdi (address to audio data) * (+8/8) buffer size (altered by *buffer_size_near) * (+16/8) period size (set when setting period size) * (+24/8) rate in hertz * (+32/8) temp */ xor %r15, %r15 mov 16(%rdi), %r15 mov %r15, 8(%rsp) xor %r15, %r15 mov 4(%rdi), %r15d mov %r15d, 24(%rsp) movq $0, 32(%rsp) /* Open pcm if needed: */ cmpq $0, (alsa.pcm_handle) jne .amd64_audio_play.pcm_is_open lea alsa.pcm_handle, %rdi mov %rdi, (%rdi) lea alsa.device_name, %rsi mov (SND_PCM_ACCESS_MMAP_INTERLEAVED), %rdx mov (SND_PCM_NONBLOCK), %r10 call snd_pcm_open cmp $0, %rax jne .amd64_audio_play.error.snd_pcm_open .amd64_audio_play.pcm_is_open: /* Allocate hw params: */ cmpq $0, (alsa.hw_params) jne .amd64_audio_play.hw_params_are_allocated lea alsa.hw_params, %rdi mov %rdi, (%rdi) call snd_pcm_hw_params_malloc cmp $0, %rax jne .amd64_audio_play.error.snd_pcm_hw_params_malloc .amd64_audio_play.hw_params_are_allocated: /* Get all possible combinations into hw params: */ mov (alsa.pcm_handle), %rdi mov (alsa.hw_params), %rsi call snd_pcm_hw_params_any cmp $0, %rax jne .amd64_audio_play.error.snd_pcm_hw_params_any /* Set access: */ mov (alsa.pcm_handle), %rdi mov (alsa.hw_params), %rsi mov (SND_PCM_ACCESS_MMAP_INTERLEAVED), %rdx call snd_pcm_hw_params_set_access cmp $0, %rax jne .amd64_audio_play.error.snd_pcm_hw_params_set_access /* Set format: */ xor %rdi, %rdi mov (%rsp), %rdi mov 14(%rdi), %r15w mov (%rdi), %r14w cmp $1, %r14w jne .amd64_audio_play.error.format_unsupported cmpw $32, %r15w je .amd64_audio_play.format32 cmpw $24, %r15w je .amd64_audio_play.format24 cmpw $16, %r15w je .amd64_audio_play.format16 cmpw $8, %r15w je .amd64_audio_play.format8 jmp .amd64_audio_play.error.format_unsupported .amd64_audio_play.format32: mov (SND_PCM_FORMAT_S32_LE), %rdx jmp .amd64_audio_play.format_end .amd64_audio_play.format24: mov (SND_PCM_FORMAT_S24_LE), %rdx jmp .amd64_audio_play.format_end .amd64_audio_play.format16: mov (SND_PCM_FORMAT_S16_LE), %rdx jmp .amd64_audio_play.format_end .amd64_audio_play.format8: mov (SND_PCM_FORMAT_S8), %rdx jmp .amd64_audio_play.format_end .amd64_audio_play.format_end: mov (alsa.pcm_handle), %rdi mov (alsa.hw_params), %rsi call snd_pcm_hw_params_set_format cmp $0, %rax jne .amd64_audio_play.error.snd_pcm_hw_params_set_format /* Set channels: */ xor %rdx, %rdx mov (%rsp), %r15 mov (alsa.pcm_handle), %rdi mov (alsa.hw_params), %rsi mov 2(%r15), %dx call snd_pcm_hw_params_set_channels cmp $0, %rax jne .amd64_audio_play.error.snd_pcm_hw_params_set_channels /* Set buffer size near: */ /* Returns 0, meaning it succeeded. No crashes generated. */ mov (alsa.pcm_handle), %rdi mov (alsa.hw_params), %rsi mov %rsp, %rdx add $8, %rdx mov $0, %r10 call snd_pcm_hw_params_set_buffer_size_near cmp $0, %rax jne .amd64_audio_play.error.snd_pcm_hw_params_set_buffer_size_near /* Set rate near: */ /* CRASHES */ mov (alsa.pcm_handle), %rdi mov (alsa.hw_params), %rsi mov %rsp, %rdx add $24, %rdx mov $0, %r10 call snd_pcm_hw_params_set_rate_near cmp $0, %rax jne .amd64_audio_play.error.snd_pcm_hw_params_set_rate_near /* Set period size near: */ /* CRASHES */ mov 8(%rsp), %rax mov $4, %r15 xor %rdx, %rdx div %r15 mov %rax, 16(%rsp) mov (alsa.pcm_handle), %rdi mov (alsa.hw_params), %rsi mov %rsp, %rdx add $16, %rdx mov $0, %r10 call snd_pcm_hw_params_set_period_size_near cmp $0, %rax jne .amd64_audio_play.error.snd_pcm_hw_params_set_period_size_near /* Free hw params: */ cmpq $0, (alsa.hw_params) je .amd64_audio_play.hw_params_not_allocated mov (alsa.hw_params), %rdi call snd_pcm_hw_params_free movq $0, (alsa.hw_params) .amd64_audio_play.hw_params_not_allocated: /* Register errors: */ xor %rax, %rax jmp .amd64_audio_play.exit .amd64_audio_play.error.snd_pcm_open: mov %rax, (amd64_audio_last_error) mov $-1, %rax jmp .amd64_audio_play.exit .amd64_audio_play.error.snd_pcm_hw_params_malloc: mov %rax, (amd64_audio_last_error) mov $-2, %rax jmp .amd64_audio_play.exit .amd64_audio_play.error.snd_pcm_hw_params_any: mov %rax, (amd64_audio_last_error) mov $-3, %rax jmp .amd64_audio_play.exit .amd64_audio_play.error.snd_pcm_hw_params_set_access: mov %rax, (amd64_audio_last_error) mov $-4, %rax jmp .amd64_audio_play.exit .amd64_audio_play.error.format_unsupported: mov $-5, %rax jmp .amd64_audio_play.exit .amd64_audio_play.error.snd_pcm_hw_params_set_format: mov %rax, (amd64_audio_last_error) mov $-6, %rax jmp .amd64_audio_play.exit .amd64_audio_play.error.snd_pcm_hw_params_set_channels: mov %rax, (amd64_audio_last_error) mov $-7, %rax jmp .amd64_audio_play.exit .amd64_audio_play.error.snd_pcm_hw_params_set_buffer_size_near: mov %rax, (amd64_audio_last_error) mov $-8, %rax jmp .amd64_audio_play.exit .amd64_audio_play.error.snd_pcm_hw_params_set_period_size_near: mov %rax, (amd64_audio_last_error) mov $-9, %rax jmp .amd64_audio_play.exit .amd64_audio_play.error.snd_pcm_hw_params_set_rate_near: mov %rax, (amd64_audio_last_error) mov $-10, %rax jmp .amd64_audio_play.exit /* Exit: */ .amd64_audio_play.exit: /* Print error to stdout: */ mov %rax, %rdi call amd64_util_print_dec call amd64_util_print_newline cmpq $0, (amd64_audio_last_error) je .amd64_audio_play.no_error mov (amd64_audio_last_error), %rdi call snd_strerror mov %rax, %rdi call amd64_util_print2 call amd64_util_print_newline .amd64_audio_play.no_error: /* Restore stack: */ add $192, %rsp /* Restore registers without rax: */ mov %rax, 128(%rsp) call amd64_util_restore_gpr_regs mov 128(%rsp), %rax add $192, %rsp /* Restore stack alignment: */ pop %rsp pop %r15 ret ``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/471 Repository URL: https://github.com/alsa-project/alsa-lib

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Update hdajackretask.desktop
by GitHub pull_request - edited 08 Aug '25

08 Aug '25

alsa-project/alsa-tools pull request #39 was edited from trebmuh: - adds a FR comment - change the "GenericName" to "Comment" - adds a Icon= field (closes #38 ) Request URL : https://github.com/alsa-project/alsa-tools/pull/39 Patch URL : https://github.com/alsa-project/alsa-tools/pull/39.patch Repository URL: https://github.com/alsa-project/alsa-tools

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Update hdajackretask.desktop
by GitHub pull_request - edited 08 Aug '25

08 Aug '25

alsa-project/alsa-tools pull request #39 was edited from trebmuh: - adds a FR comment - change the "genericname" to "Comment" - adds a Icon= field (closes #38 ) Request URL : https://github.com/alsa-project/alsa-tools/pull/39 Patch URL : https://github.com/alsa-project/alsa-tools/pull/39.patch Repository URL: https://github.com/alsa-project/alsa-tools

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Update hdajackretask.desktop
by GitHub pull_request - opened 08 Aug '25

08 Aug '25

alsa-project/alsa-tools pull request #39 was opened from trebmuh: - adds a FR comment - change the "genericname" to "Comment" - adds a Icon= field (close #38 ) Request URL : https://github.com/alsa-project/alsa-tools/pull/39 Patch URL : https://github.com/alsa-project/alsa-tools/pull/39.patch Repository URL: https://github.com/alsa-project/alsa-tools

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ucm2: Qualcomm: Add MONACO-EVK HiFi config
by GitHub pull_request - edited 08 Aug '25

08 Aug '25

alsa-project/alsa-ucm-conf pull request #597 was edited from mohsRafi: Add UCM2 configs for the Qualcomm MONACO-EVK Board to handle: - I2S Speaker Amplifier - I2S Mic Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/597 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/597.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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