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[PATCH 0/9] ASoC: SOF: amd: add soundwire IO support for ACP7.0 & ACP7.1 platforms
by Vijendar Mukunda 02 May '25

02 May '25

This patch series contains - ACP sof stack improvements - Add support for ACP7.1 platform. - Soundwire IO support for ACP7.0 & ACP7.1 platforms Link: https://github.com/thesofproject/linux/pull/5392 Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Vijendar Mukunda (9): ASoC: SOF: amd: remove else condition in resume sequence ASoC: SOF: amd: add missing acp descriptor field ASoC: SOF: amd: add ACP7.1 platform support ASoC: SOF: amd: update soundwire specific acp descriptor fields ASoC: SOF: amd: enable ACP_PME_EN register for ACP7.0 & ACP7.1 platforms ASoC: SOF: amd: refactor acp reset sequence ASoC: SOF: amd: enable soundwire host wake irq mask ASoC: SOF: amd: add soundwire wake irq handling ASoC: SOF: amd: add build support for soundwire sound/soc/sof/amd/Kconfig | 7 +- sound/soc/sof/amd/acp-dsp-offset.h | 10 +++ sound/soc/sof/amd/acp.c | 133 +++++++++++++++++++++++++---- sound/soc/sof/amd/acp.h | 7 ++ sound/soc/sof/amd/pci-acp70.c | 10 ++- 5 files changed, 148 insertions(+), 19 deletions(-) -- 2.45.2

1 9

[PATCH v3] ASoC: tas2781: Add a debufs node for acoustic tuning
by Shenghao Ding 02 May '25

02 May '25

"Acoustic Tuning" debufs node is a bridge to the acoustic tuning tool which can tune the chips' acoustic effect. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v3: - Remove unnessary indentation. - Add a special build CONFIG_SND_SOC_TAS2781_I2C_ACOUST for acoustic tuning debufs node v2: - Implement the "Acoustic Tuning" as debugfs node instead of kcontrol. v1: - Add "Acoustic Tuning" kcontrol for acoustic tuning. --- include/sound/tas2781.h | 16 ++++ sound/soc/codecs/Kconfig | 16 ++++ sound/soc/codecs/tas2781-i2c.c | 147 +++++++++++++++++++++++++++++++++ 3 files changed, 179 insertions(+) diff --git a/include/sound/tas2781.h b/include/sound/tas2781.h index 8192a94d783a..88f45e458936 100644 --- a/include/sound/tas2781.h +++ b/include/sound/tas2781.h @@ -161,10 +161,26 @@ struct calidata { unsigned int cali_dat_sz_per_dev; }; +#ifdef CONFIG_SND_SOC_TAS2781_I2C_ACOUST +#define TASDEV_DATA_PAYLOAD_SIZE 128 +struct acoustic_data { + unsigned char len; + unsigned char id; + unsigned char addr; + unsigned char book; + unsigned char page; + unsigned char reg; + unsigned char data[TASDEV_DATA_PAYLOAD_SIZE]; +}; +#endif + struct tasdevice_priv { struct tasdevice tasdevice[TASDEVICE_MAX_CHANNELS]; struct tasdevice_rca rcabin; struct calidata cali_data; +#ifdef CONFIG_SND_SOC_TAS2781_I2C_ACOUST + struct acoustic_data acou_data; +#endif struct tasdevice_fw *fmw; struct gpio_desc *speaker_id; struct gpio_desc *reset; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index fe10aaf9856e..498c63eb6837 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -2016,6 +2016,22 @@ config SND_SOC_TAS2781_I2C algo coefficient setting, for one, two or even multiple TAS2781 chips. +if SND_SOC_TAS2781_I2C + +config SND_SOC_TAS2781_I2C_ACOUST + tristate "Acoustic Tuning tool for TAS2781" + depends on DEBUG_FS + default n + help + Enable debufs node for acoustic tuning for TAS2563/TAS2781, which + is a bridge between the acoustic tuning tool and chips. With this + node, acoustic can be tuned on chips freely from user space. + Customers also can develop his own acoustic tool with this node. + + If unsure select "N". + +endif ## SND_SOC_TAS2781_I2C + config SND_SOC_TAS5086 tristate "Texas Instruments TAS5086 speaker amplifier" depends on I2C diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index 9a329c561a53..d8c9c5e41ab6 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -14,6 +14,9 @@ // #include <linux/crc8.h> +#ifdef CONFIG_SND_SOC_TAS2781_I2C_ACOUST +#include <linux/debugfs.h> +#endif #include <linux/firmware.h> #include <linux/gpio/consumer.h> #include <linux/i2c.h> @@ -1423,10 +1426,149 @@ static int tasdevice_create_cali_ctrls(struct tasdevice_priv *priv) nctrls < i ? nctrls : i); } +#ifdef CONFIG_SND_SOC_TAS2781_I2C_ACOUST +/* + * This debugfs node is a bridge to the acoustic tuning application + * tool which can tune the chips' acoustic effect. + * + * package structure for PPC3 communications: + * Pkg len (1 byte) + * Pkg id (1 byte, 'r' or 'w') + * Dev id (1 byte, i2c address) + * Book id (1 byte) + * Page id (1 byte) + * Reg id (1 byte) + * switch (pkg id) { + * case 'w': + * 1 byte, length of data to read + * case 'r': + * data payload (1~128 bytes) + * } + */ +static ssize_t acoustic_ctl_read(struct file *file, char __user *to, + size_t count, loff_t *ppos) +{ + struct snd_soc_component *comp = file->private_data; + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + struct acoustic_data *p = &tas_priv->acou_data; + int ret = -1; + + if (p->id == 'r' && p->len == count && count <= sizeof(*p)) + ret = simple_read_from_buffer(to, count, ppos, p, p->len); + else + dev_err(tas_priv->dev, "Not ready for get.\n"); + return ret; +} + +static ssize_t acoustic_ctl_write(struct file *file, + const char __user *from, size_t count, loff_t *ppos) +{ + struct snd_soc_component *comp = file->private_data; + struct tasdevice_priv *priv = snd_soc_component_get_drvdata(comp); + struct acoustic_data *p = &priv->acou_data; + unsigned int max_pkg_len = sizeof(*p); + unsigned char *src; + int j, len, reg, val; + unsigned short chn; + int ret = -1; + + if (count > sizeof(*p)) { + dev_err(priv->dev, "count(%u) is larger than max(%u).\n", + (unsigned int)count, max_pkg_len); + return ret; + } + + src = memdup_user(from, count); + if (IS_ERR(src)) + return PTR_ERR(src); + + if (src[0] > max_pkg_len && src[0] != count) { + dev_err(priv->dev, "pkg(%u), max(%u), count(%u) dismatch.\n", + src[0], max_pkg_len, (unsigned int)count); + ret = 0; + goto exit; + } + + switch (src[1]) { + case 'r': + /* length of data to read */ + len = src[6]; + break; + case 'w': + /* Skip 6 bytes for package type and register address */ + len = src[0] - 6; + break; + default: + dev_err(priv->dev, "%s Wrong code %02x.\n", __func__, src[1]); + ret = 0; + goto exit; + } + + if (len < 1) { + dev_err(priv->dev, "pkg fmt invalid %02x.\n", len); + ret = 0; + goto exit; + } + + for (j = 0; j < priv->ndev; j++) + if (src[2] == priv->tasdevice[j].dev_addr) { + chn = j; + break; + } + if (j >= priv->ndev) { + dev_err(priv->dev, "no such device 0x%02x.\n", src[2]); + ret = 0; + goto exit; + } + + reg = TASDEVICE_REG(src[3], src[4], src[5]); + + guard(mutex)(&priv->codec_lock); + + if (src[1] == 'w') { + if (len > 1) + ret = tasdevice_dev_bulk_write(priv, chn, reg, + &src[6], len); + else + ret = tasdevice_dev_write(priv, chn, reg, src[6]); + } else { + struct acoustic_data *p = &priv->acou_data; + + memcpy(p, src, 6); + if (len > 1) { + ret = tasdevice_dev_bulk_read(priv, chn, reg, + p->data, len); + } else { + ret = tasdevice_dev_read(priv, chn, reg, &val); + p->data[0] = val; + } + p->len = len + 6; + } + + if (ret) + dev_err(priv->dev, "i2c communication error.\n"); + else + ret = count; +exit: + kfree(src); + return ret; +} + +static const struct file_operations acoustic_ctl_fops = { + .open = simple_open, + .read = acoustic_ctl_read, + .write = acoustic_ctl_write, +}; +#endif + static void tasdevice_fw_ready(const struct firmware *fmw, void *context) { struct tasdevice_priv *tas_priv = context; +#ifdef CONFIG_SND_SOC_TAS2781_I2C_ACOUST + struct snd_soc_component *comp = tas_priv->codec; + struct dentry *debugfs_root = comp->debugfs_root; +#endif int ret = 0; int i; @@ -1500,6 +1642,11 @@ static void tasdevice_fw_ready(const struct firmware *fmw, tasdevice_prmg_load(tas_priv, 0); tas_priv->cur_prog = 0; + +#ifdef CONFIG_SND_SOC_TAS2781_I2C_ACOUST + debugfs_create_file("acoustic_control", 0644, debugfs_root, + comp, &acoustic_ctl_fops); +#endif out: if (tas_priv->fw_state == TASDEVICE_RCA_FW_OK) { /* If DSP FW fail, DSP kcontrol won't be created. */ -- 2.43.0

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No `appl_ptr` update or `.ack` callback call on driver side when using dmix plugin. Works fine using `hw:`
by GitHub issues - edited 02 May '25

02 May '25

alsa-project/alsa-lib issue #454 was edited from dbrignoli: This issue affects an out of tree `snd-asihpi` driver which uses a different h/w pointer update implementation compared to the in-tree `snd-asihpi` driver. In particular, this version of the driver uses the `.ack` callback to manage h/w pointer update. Driver source is available for download here <https://www.audioscience.com/internet/download/drivers/released/v4/20/47/hp…>. OS: Ubuntu 22.04.5 LTS libasound2: 1.2.6.1-1ubuntu1 LInux: 6.8.0-58-generic Using the following `.asoundrc` (with or without `slowptr true`): ``` pcm.testPlugDmix { type plug slave.pcm "dmixer" } pcm.dmixer { type dmix ipc_key 1024 slave { pcm "hw:1,0" channels 2 format S32_LE period_time 0 period_size 1024 buffer_size 4096 rate 44100 } bindings { 0 0 1 1 } # slowptr true } ``` and running aplay with `-D testPlugDmix` the trace looks like this: ``` 1) | snd_pcm_ioctl [snd_pcm]() { 1) | snd_pcm_common_ioctl [snd_pcm]() { 1) | snd_pcm_sync_ptr [snd_pcm]() { 1) 0.388 us | pcm_lib_apply_appl_ptr [snd_pcm](); /* = 0x0 */ 1) 1.155 us | } /* snd_pcm_sync_ptr [snd_pcm] = 0x0 */ 1) 1.697 us | } /* snd_pcm_common_ioctl [snd_pcm] = 0x0 */ 1) 2.055 us | } /* snd_pcm_ioctl [snd_pcm] = 0x0 */ ``` Running aplay with `-D hw:1` instead the trace looks like this: ``` 0) | snd_pcm_ioctl [snd_pcm]() { 0) | snd_pcm_common_ioctl [snd_pcm]() { 0) | __snd_pcm_lib_xfer [snd_pcm]() { 0) 0.333 us | snd_pcm_stream_lock_irq [snd_pcm](); /* = 0x0 */ 0) 0.302 us | snd_pcm_stream_unlock_irq [snd_pcm](); /* = 0x0 */ 0) | interleaved_copy [snd_pcm]() { 0) | do_transfer [snd_pcm]() { 0) 4.608 us | default_write_copy [snd_pcm](); /* = 0x0 */ 0) 5.249 us | } /* do_transfer [snd_pcm] = 0x0 */ 0) 8.835 us | } /* interleaved_copy [snd_pcm] = 0x0 */ 0) 0.311 us | snd_dma_buffer_sync [snd_pcm](); /* = 0x0 */ 0) 0.299 us | snd_pcm_stream_lock_irq [snd_pcm](); /* = 0x0 */ 0) | pcm_lib_apply_appl_ptr [snd_pcm]() { 0) | snd_card_asihpi_playback_ack [snd_asihpi]() { 0) | /* asihpi_stream_xfer: pcmC1D0p/sub0 : 4096 */ 0) | /* asihpi_stream_state: pcmC1D0p/sub0 in .ack: ALSA: aptr 4096, hptr 0, avail 12288; HPI size 16384, host 4096 (4096), dsp 0 (0, 0), fifo 0, avail 12288, xfer 0 */ 0) 1.376 us | } /* snd_card_asihpi_playback_ack [snd_asihpi] = 0x0 */ 0) 1.999 us | } /* pcm_lib_apply_appl_ptr [snd_pcm] = 0x0 */ 0) 0.364 us | snd_pcm_update_state [snd_pcm](); /* = 0x0 */ 0) 0.299 us | snd_pcm_stream_unlock_irq [snd_pcm](); /* = 0x0 */ 0) + 15.415 us | } /* __snd_pcm_lib_xfer [snd_pcm] = 0x1000 */ 0) + 16.252 us | } /* snd_pcm_common_ioctl [snd_pcm] = 0x0 */ 0) + 16.920 us | } /* snd_pcm_ioctl [snd_pcm] = 0x0 */ ``` With dmix the driver never sees changes to `appl_ptr` and the `.ack` callback is never called. The driver sets the `SNDRV_PCM_INFO_SYNC_APPLPTR` flag and and I have tried adding `SNDRV_PCM_INFO_EXPLICIT_SYNC` but it makes no difference. Using `dsnoop` on the capture side shows the same symptoms. Is this expected with dmix? Thank you. Issue URL : https://github.com/alsa-project/alsa-lib/issues/454 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

No `appl_ptr` update or `.ack` callback call on driver side when using dmix plugin. Works fine using `hw:`
by GitHub issues - edited 02 May '25

02 May '25

alsa-project/alsa-lib issue #454 was edited from dbrignoli: This issue affects an out of tree `snd-asihpi` driver which uses a different h/w pointer update implementation compared to the in-tree `snd-asihpi` driver. In particular, this version of the driver uses the `.ack` callback to manage h/w pointer update. Driver source is available for download here <https://www.audioscience.com/internet/download/drivers/released/v4/20/47/hp…>. Using the following `.asoundrc` (with or without `slowptr true`): ``` pcm.testPlugDmix { type plug slave.pcm "dmixer" } pcm.dmixer { type dmix ipc_key 1024 slave { pcm "hw:1,0" channels 2 format S32_LE period_time 0 period_size 1024 buffer_size 4096 rate 44100 } bindings { 0 0 1 1 } # slowptr true } ``` and running aplay with `-D testPlugDmix` the trace looks like this: ``` 1) | snd_pcm_ioctl [snd_pcm]() { 1) | snd_pcm_common_ioctl [snd_pcm]() { 1) | snd_pcm_sync_ptr [snd_pcm]() { 1) 0.388 us | pcm_lib_apply_appl_ptr [snd_pcm](); /* = 0x0 */ 1) 1.155 us | } /* snd_pcm_sync_ptr [snd_pcm] = 0x0 */ 1) 1.697 us | } /* snd_pcm_common_ioctl [snd_pcm] = 0x0 */ 1) 2.055 us | } /* snd_pcm_ioctl [snd_pcm] = 0x0 */ ``` Running aplay with `-D hw:1` instead the trace looks like this: ``` 0) | snd_pcm_ioctl [snd_pcm]() { 0) | snd_pcm_common_ioctl [snd_pcm]() { 0) | __snd_pcm_lib_xfer [snd_pcm]() { 0) 0.333 us | snd_pcm_stream_lock_irq [snd_pcm](); /* = 0x0 */ 0) 0.302 us | snd_pcm_stream_unlock_irq [snd_pcm](); /* = 0x0 */ 0) | interleaved_copy [snd_pcm]() { 0) | do_transfer [snd_pcm]() { 0) 4.608 us | default_write_copy [snd_pcm](); /* = 0x0 */ 0) 5.249 us | } /* do_transfer [snd_pcm] = 0x0 */ 0) 8.835 us | } /* interleaved_copy [snd_pcm] = 0x0 */ 0) 0.311 us | snd_dma_buffer_sync [snd_pcm](); /* = 0x0 */ 0) 0.299 us | snd_pcm_stream_lock_irq [snd_pcm](); /* = 0x0 */ 0) | pcm_lib_apply_appl_ptr [snd_pcm]() { 0) | snd_card_asihpi_playback_ack [snd_asihpi]() { 0) | /* asihpi_stream_xfer: pcmC1D0p/sub0 : 4096 */ 0) | /* asihpi_stream_state: pcmC1D0p/sub0 in .ack: ALSA: aptr 4096, hptr 0, avail 12288; HPI size 16384, host 4096 (4096), dsp 0 (0, 0), fifo 0, avail 12288, xfer 0 */ 0) 1.376 us | } /* snd_card_asihpi_playback_ack [snd_asihpi] = 0x0 */ 0) 1.999 us | } /* pcm_lib_apply_appl_ptr [snd_pcm] = 0x0 */ 0) 0.364 us | snd_pcm_update_state [snd_pcm](); /* = 0x0 */ 0) 0.299 us | snd_pcm_stream_unlock_irq [snd_pcm](); /* = 0x0 */ 0) + 15.415 us | } /* __snd_pcm_lib_xfer [snd_pcm] = 0x1000 */ 0) + 16.252 us | } /* snd_pcm_common_ioctl [snd_pcm] = 0x0 */ 0) + 16.920 us | } /* snd_pcm_ioctl [snd_pcm] = 0x0 */ ``` With dmix the driver never sees changes to `appl_ptr` and the `.ack` callback is never called. The driver sets the `SNDRV_PCM_INFO_SYNC_APPLPTR` flag and and I have tried adding `SNDRV_PCM_INFO_EXPLICIT_SYNC` but it makes no difference. Using `dsnoop` on the capture side shows the same symptoms. Is this expected with dmix? Thank you. Issue URL : https://github.com/alsa-project/alsa-lib/issues/454 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

No `appl_ptr` update or `.ack` callback call on driver side when using dmix plugin. Works fine using `hw:`
by GitHub issues - opened 02 May '25

02 May '25

alsa-project/alsa-lib issue #454 was opened from dbrignoli: This issue affects an out of tree `snd-asihpi` driver which uses a different h/w pointer update implementation compared to the in-tree `snd-asihpi` driver. In particular, this version of the driver uses the `.ack` callback to manage h/w pointer update. Driver source is available for download here <https://www.audioscience.com/internet/download/drivers/released/v4/20/47/hp…>. Using the following `.asoundrc` (with or without `slowptr true`): ``` pcm.testPlugDmix { type plug slave.pcm "dmixer" } pcm.dmixer { type dmix ipc_key 1024 slave { pcm "hw:1,0" channels 2 format S32_LE period_time 0 period_size 1024 buffer_size 4096 rate 44100 } bindings { 0 0 1 1 } # slowptr true } ``` and running aplay with `-D testPlugDmix` the trace looks like this: ``` 1) | snd_pcm_ioctl [snd_pcm]() { 1) | snd_pcm_common_ioctl [snd_pcm]() { 1) | snd_pcm_sync_ptr [snd_pcm]() { 1) 0.388 us | pcm_lib_apply_appl_ptr [snd_pcm](); /* = 0x0 */ 1) 1.155 us | } /* snd_pcm_sync_ptr [snd_pcm] = 0x0 */ 1) 1.697 us | } /* snd_pcm_common_ioctl [snd_pcm] = 0x0 */ 1) 2.055 us | } /* snd_pcm_ioctl [snd_pcm] = 0x0 */ ``` Running aplay with `-D hw:1` instead the trace looks like this: ``` 0) | snd_pcm_ioctl [snd_pcm]() { 0) | snd_pcm_common_ioctl [snd_pcm]() { 0) | __snd_pcm_lib_xfer [snd_pcm]() { 0) 0.333 us | snd_pcm_stream_lock_irq [snd_pcm](); /* = 0x0 */ 0) 0.302 us | snd_pcm_stream_unlock_irq [snd_pcm](); /* = 0x0 */ 0) | interleaved_copy [snd_pcm]() { 0) | do_transfer [snd_pcm]() { 0) 4.608 us | default_write_copy [snd_pcm](); /* = 0x0 */ 0) 5.249 us | } /* do_transfer [snd_pcm] = 0x0 */ 0) 8.835 us | } /* interleaved_copy [snd_pcm] = 0x0 */ 0) 0.311 us | snd_dma_buffer_sync [snd_pcm](); /* = 0x0 */ 0) 0.299 us | snd_pcm_stream_lock_irq [snd_pcm](); /* = 0x0 */ 0) | pcm_lib_apply_appl_ptr [snd_pcm]() { 0) | snd_card_asihpi_playback_ack [snd_asihpi]() { 0) | /* asihpi_stream_xfer: pcmC1D0p/sub0 : 4096 */ 0) | /* asihpi_stream_state: pcmC1D0p/sub0 in .ack: ALSA: aptr 4096, hptr 0, avail 12288; HPI size 16384, host 4096 (4096), dsp 0 (0, 0), fifo 0, avail 12288, xfer 0 */ 0) 1.376 us | } /* snd_card_asihpi_playback_ack [snd_asihpi] = 0x0 */ 0) 1.999 us | } /* pcm_lib_apply_appl_ptr [snd_pcm] = 0x0 */ 0) 0.364 us | snd_pcm_update_state [snd_pcm](); /* = 0x0 */ 0) 0.299 us | snd_pcm_stream_unlock_irq [snd_pcm](); /* = 0x0 */ 0) + 15.415 us | } /* __snd_pcm_lib_xfer [snd_pcm] = 0x1000 */ 0) + 16.252 us | } /* snd_pcm_common_ioctl [snd_pcm] = 0x0 */ 0) + 16.920 us | } /* snd_pcm_ioctl [snd_pcm] = 0x0 */ ``` With dmix the driver never sees changes to `appl_ptr` and the `.ack` callback is never called. The driver sets the `SNDRV_PCM_INFO_SYNC_APPLPTR` flag and and I have tried adding `SNDRV_PCM_INFO_EXPLICIT_SYNC` but it makes no difference. Using `dsnoop` on the capture side shows the same symptoms. Is this expected with dmix? Thank you. Issue URL : https://github.com/alsa-project/alsa-lib/issues/454 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[GIT PULL] ASoC fixes for v6.15-rc4
by Mark Brown 01 May '25

01 May '25

The following changes since commit b4432656b36e5cc1d50a1f2dc15357543add530e: Linux 6.15-rc4 (2025-04-27 15:19:23 -0700) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.15-rc4 for you to fetch changes up to 315c84567001aec7f7b74a0dbd4fce3e1c5cbd87: ASoC: intel/sdw_utils: Add volume limits to CS35L56 (2025-05-01 14:43:52 +0900) ---------------------------------------------------------------- ASoC: Fixes for v6.15 A moderately large batch of fixes for v6.15, many driver specific including cleanups for the enabling of the Cirrus KUnit tests and a fix for a nasty crash on resume on AMD systems. We also have one core fix, for an ordering issue between DAPM and DPCM which could leave things incorrectly unpowered. ---------------------------------------------------------------- Alexander Stein (1): ASoC: simple-card-utils: Fix pointer check in graph_util_parse_link_direction Arnd Bergmann (1): ASoC: Intel: catpt: avoid type mismatch in dev_dbg() format Bard Liao (1): ASoC: soc-acpi-intel-ptl-match: add empty item to ptl_cs42l43_l3[] Charles Keepax (1): ASoC: cs42l43: Disable headphone clamps during type detection Chenyuan Yang (2): ASoC: imx-card: Adjust over allocation of memory in imx_card_parse_of() ASoC: Intel: sof_sdw: Add NULL check in asoc_sdw_rt_dmic_rtd_init() Claudiu Beznea (1): ASoC: renesas: rz-ssi: Use NOIRQ_SYSTEM_SLEEP_PM_OPS() Mark Brown (3): Fix up building KUnit tests for Cirrus Logic modules ASoC: stm32: sai: fix kernel rate configuration ASoC: intel/sdw_utils: Add volume limits to CS35L56 Nico Pache (1): firmware: cs_dsp: tests: Depend on FW_CS_DSP rather then enabling it Olivier Moysan (2): ASoC: stm32: sai: skip useless iterations on kernel rate loop ASoC: stm32: sai: add a check on minimal kernel frequency Richard Fitzgerald (3): kunit: configs: Add some Cirrus Logic modules to all_tests ASoC: cs-amp-lib-test: Don't select SND_SOC_CS_AMP_LIB ASoC: Add Cirrus and Wolfson headers to ASoC section of MAINTAINERS Sheetal (1): ASoC: soc-pcm: Fix hw_params() and DAPM widget sequence Stefan Binding (2): ASoC: intel/sdw_utils: Add volume limit to cs42l43 speakers ASoC: intel/sdw_utils: Add volume limit to cs35l56 speakers Takashi Iwai (1): ASoC: Intel: bytcr_rt5640: Add DMI quirk for Acer Aspire SW3-013 Venkata Prasad Potturu (3): ASoC: amd: acp: Fix NULL pointer deref on acp resume path ASoC: amd: acp: Fix NULL pointer deref in acp_i2s_set_tdm_slot ASoC: amd: acp: Fix devm_snd_soc_register_card(acp-pdm-mach) failure Vijendar Mukunda (1): ASoC: amd: ps: fix for irq handler return status MAINTAINERS | 6 ++++++ drivers/firmware/cirrus/Kconfig | 5 +---- include/sound/soc_sdw_utils.h | 1 + sound/soc/amd/acp/acp-i2s.c | 2 +- sound/soc/amd/acp/acp-legacy-common.c | 2 +- sound/soc/amd/acp/acp-rembrandt.c | 2 +- sound/soc/amd/acp/acp-renoir.c | 2 +- sound/soc/amd/acp/acp63.c | 2 +- sound/soc/amd/acp/acp70.c | 2 +- sound/soc/amd/ps/pci-ps.c | 5 +++-- sound/soc/codecs/Kconfig | 5 ++--- sound/soc/codecs/cs42l43-jack.c | 7 +++++++ sound/soc/fsl/imx-card.c | 2 +- sound/soc/generic/simple-card-utils.c | 4 ++-- sound/soc/intel/boards/bytcr_rt5640.c | 13 ++++++++++++ sound/soc/intel/catpt/dsp.c | 2 +- sound/soc/intel/common/soc-acpi-intel-ptl-match.c | 3 ++- sound/soc/renesas/rz-ssi.c | 2 +- sound/soc/sdw_utils/soc_sdw_bridge_cs35l56.c | 4 ++++ sound/soc/sdw_utils/soc_sdw_cs42l43.c | 10 ++++++++++ sound/soc/sdw_utils/soc_sdw_cs_amp.c | 24 +++++++++++++++++++++++ sound/soc/sdw_utils/soc_sdw_rt_dmic.c | 2 ++ sound/soc/soc-pcm.c | 5 ++++- sound/soc/stm/stm32_sai_sub.c | 16 ++++++++++----- tools/testing/kunit/configs/all_tests.config | 2 ++ 25 files changed, 103 insertions(+), 27 deletions(-)

2 1

[PATCH] ASoC: amd: ps: fix for irq handler return status
by Vijendar Mukunda 01 May '25

01 May '25

If any Soundwire manager interrupt is reported, and wake interrupt is not reported, in this scenario irq_flag will be set to zero, which results in interrupt handler return status as IRQ_NONE. Add new irq flag 'wake_irq_flag' check for SoundWire wake interrupt handling to fix incorrect irq handling return status. Fixes: 3898b189079c8 ("ASoC: amd: ps: add soundwire wake interrupt handling") Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- sound/soc/amd/ps/pci-ps.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/amd/ps/pci-ps.c b/sound/soc/amd/ps/pci-ps.c index 8e57f31ef7f7..7936b3173632 100644 --- a/sound/soc/amd/ps/pci-ps.c +++ b/sound/soc/amd/ps/pci-ps.c @@ -193,6 +193,7 @@ static irqreturn_t acp63_irq_handler(int irq, void *dev_id) struct amd_sdw_manager *amd_manager; u32 ext_intr_stat, ext_intr_stat1; u16 irq_flag = 0; + u16 wake_irq_flag = 0; u16 sdw_dma_irq_flag = 0; adata = dev_id; @@ -231,7 +232,7 @@ static irqreturn_t acp63_irq_handler(int irq, void *dev_id) } if (adata->acp_rev >= ACP70_PCI_REV) - irq_flag = check_and_handle_acp70_sdw_wake_irq(adata); + wake_irq_flag = check_and_handle_acp70_sdw_wake_irq(adata); if (ext_intr_stat & BIT(PDM_DMA_STAT)) { ps_pdm_data = dev_get_drvdata(&adata->pdm_dev->dev); @@ -245,7 +246,7 @@ static irqreturn_t acp63_irq_handler(int irq, void *dev_id) if (sdw_dma_irq_flag) return IRQ_WAKE_THREAD; - if (irq_flag) + if (irq_flag | wake_irq_flag) return IRQ_HANDLED; else return IRQ_NONE; -- 2.45.2

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[PATCH v2] ALSA: hda/tas2781: Remove tas2781_spi_fwlib.c and leverage SND_SOC_TAS2781_FMWLIB
by Shenghao Ding 30 Apr '25

30 Apr '25

Most codes in tas2781_spi_fwlib.c are same as tas2781-fmwlib.c, mainly for firmware parsing, only differece is the register reading, bit update and book switching in i2c and spi. The main purpose of this patch is for code cleaup and arrange the shared part for i2c and spi. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> Acked-by: Mark Brown <broonie(a)kernel.org> --- v2: - Split tas2781_comlib into two parts. One is the shared lib for both SPI and I2C. The other is the lib for I2C use only. - Adjust the lib dependencies in Konfig and Makefile in HDA and CODECS - | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202504260724.ryu83bm2-lkp@intel.com/ v1: - Move the definition of TASDEVICE_CRC8_POLYNOMIAL to tas2781.h for both SPI and I2C. - Add isspi flag in the struct tasdevice_priv to distinguish SPI or I2C. - Define three register-read related methods due to the difference between SPI and I2C. For SPI-based TAS2781, it will have one-byte delay to read memory-mapped registers. During init, these three method will be initialized as spi or i2c register-read methods. - Define book-switching method due to the difference between SPI and I2C. - Add select SND_SOC_TAS2781_COMLIB & SND_SOC_TAS2781_FMWLIB for tas2781 hda spi driver in Kconfig. - remove tas2781-spi.h & tas2781_spi_fwlib.c. - remove tas2781_spi_fwlib.o in Makefile - Move the definition of TASDEVICE_CALIBRATION_DATA_NAME. - Remove the useless calib_data definition for code cleanup. - Define the TASDEV_CALIB_N definition in tas2781_hda.h for hda driver calibration. - Use TASDEV_CALIB_N instead of TAS2563_CAL_N. - Include tas2781.h instead of tas2781-dsp.h in tas2781_hda_spi.c. - Remove #include "tas2781-spi.h" in tas2781_hda_spi.c. - Move the macro definitions only for hda spi tas2781 driver into tas2781_hda_spi.c. - Add read_flag_mask and reg_shift setting in tasdevice_regmap. --- include/sound/tas2781-comlib-i2c.h | 37 + include/sound/tas2781.h | 45 +- sound/pci/hda/Kconfig | 4 +- sound/pci/hda/Makefile | 2 +- sound/pci/hda/tas2781-spi.h | 157 -- sound/pci/hda/tas2781_hda.h | 3 + sound/pci/hda/tas2781_hda_i2c.c | 26 +- sound/pci/hda/tas2781_hda_spi.c | 542 +++---- sound/pci/hda/tas2781_spi_fwlib.c | 2006 ------------------------- sound/soc/codecs/Kconfig | 8 +- sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tas2781-comlib-i2c.c | 375 +++++ sound/soc/codecs/tas2781-comlib.c | 406 +---- sound/soc/codecs/tas2781-fmwlib.c | 17 +- sound/soc/codecs/tas2781-i2c.c | 1 + 15 files changed, 768 insertions(+), 2863 deletions(-) create mode 100644 include/sound/tas2781-comlib-i2c.h delete mode 100644 sound/pci/hda/tas2781-spi.h delete mode 100644 sound/pci/hda/tas2781_spi_fwlib.c create mode 100644 sound/soc/codecs/tas2781-comlib-i2c.c diff --git a/include/sound/tas2781-comlib-i2c.h b/include/sound/tas2781-comlib-i2c.h new file mode 100644 index 000000000000..a1afa5c444ba --- /dev/null +++ b/include/sound/tas2781-comlib-i2c.h @@ -0,0 +1,37 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +// +// ALSA SoC Texas Instruments TAS2563/TAS2781 Audio Smart Amplifier +// +// Copyright (C) 2025 Texas Instruments Incorporated +// https://www.ti.com +// +// The TAS2563/TAS2781 driver implements a flexible and configurable +// algo coefficient setting for one, two, or even multiple +// TAS2563/TAS2781 chips. +// +// Author: Shenghao Ding <shenghao-ding(a)ti.com> +// + +#ifndef __TAS2781_COMLIB_I2C_H__ +#define __TAS2781_COMLIB_I2C_H__ + +void tasdevice_reset(struct tasdevice_priv *tas_dev); +int tascodec_init(struct tasdevice_priv *tas_priv, void *codec, + struct module *module, + void (*cont)(const struct firmware *fw, void *context)); +struct tasdevice_priv *tasdevice_kzalloc(struct i2c_client *i2c); +int tasdevice_init(struct tasdevice_priv *tas_priv); +int tasdev_chn_switch(struct tasdevice_priv *tas_priv, + unsigned short chn); +int tasdevice_dev_update_bits( + struct tasdevice_priv *tasdevice, unsigned short chn, + unsigned int reg, unsigned int mask, unsigned int value); +int tasdevice_amp_putvol(struct tasdevice_priv *tas_priv, + struct snd_ctl_elem_value *ucontrol, struct soc_mixer_control *mc); +int tasdevice_amp_getvol(struct tasdevice_priv *tas_priv, + struct snd_ctl_elem_value *ucontrol, struct soc_mixer_control *mc); +int tasdevice_digital_getvol(struct tasdevice_priv *tas_priv, + struct snd_ctl_elem_value *ucontrol, struct soc_mixer_control *mc); +int tasdevice_digital_putvol(struct tasdevice_priv *tas_priv, + struct snd_ctl_elem_value *ucontrol, struct soc_mixer_control *mc); +#endif /* __TAS2781_COMLIB_I2C_H__ */ diff --git a/include/sound/tas2781.h b/include/sound/tas2781.h index eff011444cc8..8192a94d783a 100644 --- a/include/sound/tas2781.h +++ b/include/sound/tas2781.h @@ -32,6 +32,8 @@ SNDRV_PCM_FMTBIT_S24_LE | \ SNDRV_PCM_FMTBIT_S32_LE) +#define TASDEVICE_CRC8_POLYNOMIAL 0x4d + /* PAGE Control Register (available in page0 of each book) */ #define TASDEVICE_PAGE_SELECT 0x00 #define TASDEVICE_BOOKCTL_PAGE 0x00 @@ -193,6 +195,7 @@ struct tasdevice_priv { bool force_fwload_status; bool playback_started; bool isacpi; + bool isspi; bool is_user_space_calidata; unsigned int global_addr; @@ -210,41 +213,31 @@ struct tasdevice_priv { int (*tasdevice_load_block)(struct tasdevice_priv *tas_priv, struct tasdev_blk *block); + int (*change_chn_book)(struct tasdevice_priv *tas_priv, + unsigned short chn, int book); + int (*update_bits)(struct tasdevice_priv *tas_priv, + unsigned short chn, unsigned int reg, unsigned int mask, + unsigned int value); + int (*dev_read)(struct tasdevice_priv *tas_priv, + unsigned short chn, unsigned int reg, unsigned int *value); + int (*dev_bulk_read)(struct tasdevice_priv *tas_priv, + unsigned short chn, unsigned int reg, unsigned char *p_data, + unsigned int n_length); int (*save_calibration)(struct tasdevice_priv *tas_priv); void (*apply_calibration)(struct tasdevice_priv *tas_priv); }; -void tasdevice_reset(struct tasdevice_priv *tas_dev); -int tascodec_init(struct tasdevice_priv *tas_priv, void *codec, - struct module *module, - void (*cont)(const struct firmware *fw, void *context)); -struct tasdevice_priv *tasdevice_kzalloc(struct i2c_client *i2c); -int tasdevice_init(struct tasdevice_priv *tas_priv); -void tasdevice_remove(struct tasdevice_priv *tas_priv); -int tasdevice_save_calibration(struct tasdevice_priv *tas_priv); -void tasdevice_apply_calibration(struct tasdevice_priv *tas_priv); -int tasdev_chn_switch(struct tasdevice_priv *tas_priv, - unsigned short chn); int tasdevice_dev_read(struct tasdevice_priv *tas_priv, unsigned short chn, unsigned int reg, unsigned int *value); +int tasdevice_dev_bulk_read(struct tasdevice_priv *tas_priv, + unsigned short chn, unsigned int reg, unsigned char *p_data, + unsigned int n_length); int tasdevice_dev_write(struct tasdevice_priv *tas_priv, unsigned short chn, unsigned int reg, unsigned int value); int tasdevice_dev_bulk_write( struct tasdevice_priv *tas_priv, unsigned short chn, unsigned int reg, unsigned char *p_data, unsigned int n_length); -int tasdevice_dev_bulk_read(struct tasdevice_priv *tas_priv, - unsigned short chn, unsigned int reg, unsigned char *p_data, - unsigned int n_length); -int tasdevice_dev_update_bits( - struct tasdevice_priv *tasdevice, unsigned short chn, - unsigned int reg, unsigned int mask, unsigned int value); -int tasdevice_amp_putvol(struct tasdevice_priv *tas_priv, - struct snd_ctl_elem_value *ucontrol, struct soc_mixer_control *mc); -int tasdevice_amp_getvol(struct tasdevice_priv *tas_priv, - struct snd_ctl_elem_value *ucontrol, struct soc_mixer_control *mc); -int tasdevice_digital_putvol(struct tasdevice_priv *tas_priv, - struct snd_ctl_elem_value *ucontrol, struct soc_mixer_control *mc); -int tasdevice_digital_getvol(struct tasdevice_priv *tas_priv, - struct snd_ctl_elem_value *ucontrol, struct soc_mixer_control *mc); - +void tasdevice_remove(struct tasdevice_priv *tas_priv); +int tasdevice_save_calibration(struct tasdevice_priv *tas_priv); +void tasdevice_apply_calibration(struct tasdevice_priv *tas_priv); #endif /* __TAS2781_H__ */ diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index 15cc46032583..24e935cb0cdb 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -186,7 +186,7 @@ config SND_HDA_SCODEC_TAS2781_I2C depends on ACPI depends on EFI depends on SND_SOC - select SND_SOC_TAS2781_COMLIB + select SND_SOC_TAS2781_COMLIB_I2C select SND_SOC_TAS2781_FMWLIB select CRC32 help @@ -202,6 +202,8 @@ config SND_HDA_SCODEC_TAS2781_SPI depends on ACPI depends on EFI depends on SND_SOC + select SND_SOC_TAS2781_FMWLIB + select CRC8 select CRC32 help Say Y or M here to include TAS2781 SPI HD-audio side codec support diff --git a/sound/pci/hda/Makefile b/sound/pci/hda/Makefile index 2953def7db67..13d6c7e63203 100644 --- a/sound/pci/hda/Makefile +++ b/sound/pci/hda/Makefile @@ -39,7 +39,7 @@ snd-hda-scodec-cs35l56-i2c-y := cs35l56_hda_i2c.o snd-hda-scodec-cs35l56-spi-y := cs35l56_hda_spi.o snd-hda-scodec-component-y := hda_component.o snd-hda-scodec-tas2781-i2c-y := tas2781_hda_i2c.o -snd-hda-scodec-tas2781-spi-y := tas2781_hda_spi.o tas2781_spi_fwlib.o +snd-hda-scodec-tas2781-spi-y := tas2781_hda_spi.o # common driver obj-$(CONFIG_SND_HDA) := snd-hda-codec.o diff --git a/sound/pci/hda/tas2781-spi.h b/sound/pci/hda/tas2781-spi.h deleted file mode 100644 index 7a0faceeb675..000000000000 --- a/sound/pci/hda/tas2781-spi.h +++ /dev/null @@ -1,157 +0,0 @@ -/* SPDX-License-Identifier: GPL-2.0 */ -// -// ALSA SoC Texas Instruments TAS2781 Audio Smart Amplifier -// -// Copyright (C) 2024 Texas Instruments Incorporated -// https://www.ti.com -// -// The TAS2781 driver implements a flexible and configurable -// algo coefficient setting for TAS2781 chips. -// -// Author: Baojun Xu <baojun.xu(a)ti.com> -// - -#ifndef __TAS2781_SPI_H__ -#define __TAS2781_SPI_H__ - -#define TASDEVICE_RATES \ - (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | \ - SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_88200) - -#define TASDEVICE_FORMATS \ - (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | \ - SNDRV_PCM_FMTBIT_S32_LE) - -#define TASDEVICE_MAX_BOOK_NUM 256 -#define TASDEVICE_MAX_PAGE 256 - -#define TASDEVICE_MAX_SIZE (TASDEVICE_MAX_BOOK_NUM * TASDEVICE_MAX_PAGE) - -/* PAGE Control Register (available in page0 of each book) */ -#define TASDEVICE_PAGE_SELECT 0x00 -#define TASDEVICE_BOOKCTL_PAGE 0x00 -#define TASDEVICE_BOOKCTL_REG GENMASK(7, 1) -#define TASDEVICE_BOOK_ID(reg) (((reg) & GENMASK(24, 16)) >> 16) -#define TASDEVICE_PAGE_ID(reg) (((reg) & GENMASK(15, 8)) >> 8) -#define TASDEVICE_REG_ID(reg) (((reg) & GENMASK(7, 1)) >> 1) -#define TASDEVICE_PAGE_REG(reg) ((reg) & GENMASK(15, 1)) -#define TASDEVICE_REG(book, page, reg) \ - (((book) << 16) | ((page) << 8) | ((reg) << 1)) - -/* Software Reset */ -#define TAS2781_REG_SWRESET TASDEVICE_REG(0x0, 0x0, 0x01) -#define TAS2781_REG_SWRESET_RESET BIT(0) - -/* System Reset Check Register */ -#define TAS2781_REG_CLK_CONFIG TASDEVICE_REG(0x0, 0x0, 0x5c) -#define TAS2781_REG_CLK_CONFIG_RESET (0x19) -#define TAS2781_PRE_POST_RESET_CFG 3 - -/* Block Checksum */ -#define TASDEVICE_CHECKSUM TASDEVICE_REG(0x0, 0x0, 0x7e) - -/* Volume control */ -#define TAS2781_DVC_LVL TASDEVICE_REG(0x0, 0x0, 0x1a) -#define TAS2781_AMP_LEVEL TASDEVICE_REG(0x0, 0x0, 0x03) -#define TAS2781_AMP_LEVEL_MASK GENMASK(5, 1) - -#define TASDEVICE_CMD_SING_W 0x1 -#define TASDEVICE_CMD_BURST 0x2 -#define TASDEVICE_CMD_DELAY 0x3 -#define TASDEVICE_CMD_FIELD_W 0x4 - -#define TAS2781_SPI_MAX_FREQ (4 * HZ_PER_MHZ) - -#define TASDEVICE_CRC8_POLYNOMIAL 0x4d -#define TASDEVICE_SPEAKER_CALIBRATION_SIZE 20 - -/* Flag of calibration registers address. */ -#define TASDEVICE_CALIBRATION_REG_ADDRESS BIT(7) - -#define TASDEVICE_CALIBRATION_DATA_NAME L"CALI_DATA" -#define TASDEVICE_CALIBRATION_DATA_SIZE 256 - -enum calib_data { - R0_VAL = 0, - INV_R0, - R0LOW, - POWER, - TLIM, - CALIB_MAX -}; - -struct tasdevice_priv { - struct tasdevice_fw *cali_data_fmw; - struct tasdevice_rca rcabin; - struct tasdevice_fw *fmw; - struct gpio_desc *reset; - struct mutex codec_lock; - struct regmap *regmap; - struct device *dev; - - unsigned char crc8_lkp_tbl[CRC8_TABLE_SIZE]; - unsigned char coef_binaryname[64]; - unsigned char rca_binaryname[64]; - unsigned char dev_name[32]; - - bool force_fwload_status; - bool playback_started; - bool is_loading; - bool is_loaderr; - unsigned int cali_reg_array[CALIB_MAX]; - unsigned int cali_data[CALIB_MAX]; - unsigned int err_code; - void *codec; - int cur_book; - int cur_prog; - int cur_conf; - int fw_state; - int index; - int irq; - - int (*fw_parse_variable_header)(struct tasdevice_priv *tas_priv, - const struct firmware *fmw, - int offset); - int (*fw_parse_program_data)(struct tasdevice_priv *tas_priv, - struct tasdevice_fw *tas_fmw, - const struct firmware *fmw, int offset); - int (*fw_parse_configuration_data)(struct tasdevice_priv *tas_priv, - struct tasdevice_fw *tas_fmw, - const struct firmware *fmw, - int offset); - int (*tasdevice_load_block)(struct tasdevice_priv *tas_priv, - struct tasdev_blk *block); - - int (*save_calibration)(struct tasdevice_priv *tas_priv); - void (*apply_calibration)(struct tasdevice_priv *tas_priv); -}; - -int tasdevice_spi_dev_read(struct tasdevice_priv *tas_priv, - unsigned int reg, unsigned int *value); -int tasdevice_spi_dev_write(struct tasdevice_priv *tas_priv, - unsigned int reg, unsigned int value); -int tasdevice_spi_dev_bulk_write(struct tasdevice_priv *tas_priv, - unsigned int reg, unsigned char *p_data, - unsigned int n_length); -int tasdevice_spi_dev_bulk_read(struct tasdevice_priv *tas_priv, - unsigned int reg, unsigned char *p_data, - unsigned int n_length); -int tasdevice_spi_dev_update_bits(struct tasdevice_priv *tasdevice, - unsigned int reg, unsigned int mask, - unsigned int value); - -void tasdevice_spi_select_cfg_blk(void *context, int conf_no, - unsigned char block_type); -void tasdevice_spi_config_info_remove(void *context); -int tasdevice_spi_dsp_parser(void *context); -int tasdevice_spi_rca_parser(void *context, const struct firmware *fmw); -void tasdevice_spi_dsp_remove(void *context); -void tasdevice_spi_calbin_remove(void *context); -int tasdevice_spi_select_tuningprm_cfg(void *context, int prm, int cfg_no, - int rca_conf_no); -int tasdevice_spi_prmg_load(void *context, int prm_no); -int tasdevice_spi_prmg_calibdata_load(void *context, int prm_no); -void tasdevice_spi_tuning_switch(void *context, int state); -int tas2781_spi_load_calibration(void *context, char *file_name, - unsigned short i); -#endif /* __TAS2781_SPI_H__ */ diff --git a/sound/pci/hda/tas2781_hda.h b/sound/pci/hda/tas2781_hda.h index 32d27b202591..444231f87e53 100644 --- a/sound/pci/hda/tas2781_hda.h +++ b/sound/pci/hda/tas2781_hda.h @@ -9,6 +9,9 @@ #include <sound/asound.h> +#define TASDEVICE_CALIBRATION_DATA_NAME L"CALI_DATA" +#define TASDEV_CALIB_N 5 + /* * No standard control callbacks for SNDRV_CTL_ELEM_IFACE_CARD * Define two controls, one is Volume control callbacks, the other is diff --git a/sound/pci/hda/tas2781_hda_i2c.c b/sound/pci/hda/tas2781_hda_i2c.c index 9d94ae5fcfe0..0a686e850540 100644 --- a/sound/pci/hda/tas2781_hda_i2c.c +++ b/sound/pci/hda/tas2781_hda_i2c.c @@ -22,6 +22,7 @@ #include <sound/hda_codec.h> #include <sound/soc.h> #include <sound/tas2781.h> +#include <sound/tas2781-comlib-i2c.h> #include <sound/tlv.h> #include <sound/tas2781-tlv.h> @@ -34,15 +35,6 @@ #define TASDEVICE_SPEAKER_CALIBRATION_SIZE 20 -enum calib_data { - R0_VAL = 0, - INV_R0, - R0LOW, - POWER, - TLIM, - CALIB_MAX -}; - #define TAS2563_MAX_CHANNELS 4 #define TAS2563_CAL_POWER TASDEVICE_REG(0, 0x0d, 0x3c) @@ -50,17 +42,15 @@ enum calib_data { #define TAS2563_CAL_INVR0 TASDEVICE_REG(0, 0x0f, 0x40) #define TAS2563_CAL_R0_LOW TASDEVICE_REG(0, 0x0f, 0x48) #define TAS2563_CAL_TLIM TASDEVICE_REG(0, 0x10, 0x14) -#define TAS2563_CAL_N 5 #define TAS2563_CAL_DATA_SIZE 4 #define TAS2563_CAL_CH_SIZE 20 #define TAS2563_CAL_ARRAY_SIZE 80 -static unsigned int cal_regs[TAS2563_CAL_N] = { +static unsigned int cal_regs[TASDEV_CALIB_N] = { TAS2563_CAL_POWER, TAS2563_CAL_R0, TAS2563_CAL_INVR0, TAS2563_CAL_R0_LOW, TAS2563_CAL_TLIM, }; - struct tas2781_hda { struct device *dev; struct tasdevice_priv *priv; @@ -475,7 +465,7 @@ static void tas2563_apply_calib(struct tasdevice_priv *tas_priv) int ret; for (int i = 0; i < tas_priv->ndev; i++) { - for (int j = 0; j < TAS2563_CAL_N; ++j) { + for (int j = 0; j < TASDEV_CALIB_N; ++j) { data = cpu_to_be32( *(uint32_t *)&tas_priv->cali_data.data[offset]); ret = tasdevice_dev_bulk_write(tas_priv, i, cal_regs[j], @@ -493,7 +483,7 @@ static int tas2563_save_calibration(struct tasdevice_priv *tas_priv) static efi_guid_t efi_guid = EFI_GUID(0x1f52d2a1, 0xbb3a, 0x457d, 0xbc, 0x09, 0x43, 0xa3, 0xf4, 0x31, 0x0a, 0x92); - static efi_char16_t *efi_vars[TAS2563_MAX_CHANNELS][TAS2563_CAL_N] = { + static efi_char16_t *efi_vars[TAS2563_MAX_CHANNELS][TASDEV_CALIB_N] = { { L"Power_1", L"R0_1", L"InvR0_1", L"R0_Low_1", L"TLim_1" }, { L"Power_2", L"R0_2", L"InvR0_2", L"R0_Low_2", L"TLim_2" }, { L"Power_3", L"R0_3", L"InvR0_3", L"R0_Low_3", L"TLim_3" }, @@ -511,7 +501,7 @@ static int tas2563_save_calibration(struct tasdevice_priv *tas_priv) return -ENOMEM; for (int i = 0; i < tas_priv->ndev; ++i) { - for (int j = 0; j < TAS2563_CAL_N; ++j) { + for (int j = 0; j < TASDEV_CALIB_N; ++j) { status = efi.get_variable(efi_vars[i][j], &efi_guid, &attr, &max_size, &tas_priv->cali_data.data[offset]); @@ -535,7 +525,7 @@ static void tas2781_apply_calib(struct tasdevice_priv *tas_priv) { struct calidata *cali_data = &tas_priv->cali_data; struct cali_reg *r = &cali_data->cali_reg_array; - unsigned int cali_reg[CALIB_MAX] = { + unsigned int cali_reg[TASDEV_CALIB_N] = { TASDEVICE_REG(0, 0x17, 0x74), TASDEVICE_REG(0, 0x18, 0x0c), TASDEVICE_REG(0, 0x18, 0x14), @@ -555,7 +545,7 @@ static void tas2781_apply_calib(struct tasdevice_priv *tas_priv) } for (i = 0; i < tas_priv->ndev; i++) { - for (j = 0; j < CALIB_MAX; j++) { + for (j = 0; j < TASDEV_CALIB_N; j++) { data = cpu_to_be32( *(uint32_t *)&tas_priv->cali_data.data[oft]); rc = tasdevice_dev_bulk_write(tas_priv, i, @@ -578,7 +568,7 @@ static int tas2781_save_calibration(struct tasdevice_priv *tas_priv) { efi_guid_t efi_guid = EFI_GUID(0x02f9af02, 0x7734, 0x4233, 0xb4, 0x3d, 0x93, 0xfe, 0x5a, 0xa3, 0x5d, 0xb3); - static efi_char16_t efi_name[] = L"CALI_DATA"; + static efi_char16_t efi_name[] = TASDEVICE_CALIBRATION_DATA_NAME; unsigned int attr, crc; unsigned int *tmp_val; efi_status_t status; diff --git a/sound/pci/hda/tas2781_hda_spi.c b/sound/pci/hda/tas2781_hda_spi.c index 92be591c6bfe..426fc6044c65 100644 --- a/sound/pci/hda/tas2781_hda_spi.c +++ b/sound/pci/hda/tas2781_hda_spi.c @@ -27,12 +27,10 @@ #include <sound/hda_codec.h> #include <sound/soc.h> -#include <sound/tas2781-dsp.h> +#include <sound/tas2781.h> #include <sound/tlv.h> #include <sound/tas2781-tlv.h> -#include "tas2781-spi.h" - #include "hda_local.h" #include "hda_auto_parser.h" #include "hda_component.h" @@ -40,6 +38,17 @@ #include "hda_generic.h" #include "tas2781_hda.h" +#define TASDEVICE_RANGE_MAX_SIZE (256 * 128) +#define TASDEVICE_WIN_LEN 128 +#define TAS2781_SPI_MAX_FREQ (4 * HZ_PER_MHZ) +/* Flag of calibration registers address. */ +#define TASDEVICE_CALIBRATION_REG_ADDRESS BIT(7) +#define TASDEV_UEFI_CALI_REG_ADDR_FLG BIT(7) + +/* System Reset Check Register */ +#define TAS2781_REG_CLK_CONFIG TASDEVICE_REG(0x0, 0x0, 0x5c) +#define TAS2781_REG_CLK_CONFIG_RESET 0x19 + struct tas2781_hda { struct tasdevice_priv *priv; struct acpi_device *dacpi; @@ -52,12 +61,12 @@ struct tas2781_hda { static const struct regmap_range_cfg tasdevice_ranges[] = { { .range_min = 0, - .range_max = TASDEVICE_MAX_SIZE, + .range_max = TASDEVICE_RANGE_MAX_SIZE, .selector_reg = TASDEVICE_PAGE_SELECT, .selector_mask = GENMASK(7, 0), .selector_shift = 0, .window_start = 0, - .window_len = TASDEVICE_MAX_PAGE, + .window_len = TASDEVICE_WIN_LEN, }, }; @@ -65,39 +74,19 @@ static const struct regmap_config tasdevice_regmap = { .reg_bits = 8, .val_bits = 8, .zero_flag_mask = true, + .read_flag_mask = 0x01, + .reg_shift = -1, .cache_type = REGCACHE_NONE, .ranges = tasdevice_ranges, .num_ranges = ARRAY_SIZE(tasdevice_ranges), - .max_register = TASDEVICE_MAX_SIZE, + .max_register = TASDEVICE_RANGE_MAX_SIZE, }; -static int tasdevice_spi_switch_book(struct tasdevice_priv *tas_priv, int reg) +static int tasdevice_spi_dev_read(struct tasdevice_priv *tas_priv, + unsigned short chn, unsigned int reg, unsigned int *val) { - struct regmap *map = tas_priv->regmap; - - if (tas_priv->cur_book != TASDEVICE_BOOK_ID(reg)) { - int ret = regmap_write(map, TASDEVICE_BOOKCTL_REG, - TASDEVICE_BOOK_ID(reg)); - if (ret < 0) { - dev_err(tas_priv->dev, "Switch Book E=%d\n", ret); - return ret; - } - tas_priv->cur_book = TASDEVICE_BOOK_ID(reg); - } - return 0; -} - -int tasdevice_spi_dev_read(struct tasdevice_priv *tas_priv, - unsigned int reg, - unsigned int *val) -{ - struct regmap *map = tas_priv->regmap; int ret; - ret = tasdevice_spi_switch_book(tas_priv, reg); - if (ret < 0) - return ret; - /* * In our TAS2781 SPI mode, if read from other book (not book 0), * or read from page number larger than 1 in book 0, one more byte @@ -106,11 +95,11 @@ int tasdevice_spi_dev_read(struct tasdevice_priv *tas_priv, if ((TASDEVICE_BOOK_ID(reg) > 0) || (TASDEVICE_PAGE_ID(reg) > 1)) { unsigned char data[2]; - ret = regmap_bulk_read(map, TASDEVICE_PAGE_REG(reg) | 1, + ret = tasdevice_dev_bulk_read(tas_priv, chn, reg, data, sizeof(data)); *val = data[1]; } else { - ret = regmap_read(map, TASDEVICE_PAGE_REG(reg) | 1, val); + ret = tasdevice_dev_read(tas_priv, chn, reg, val); } if (ret < 0) dev_err(tas_priv->dev, "%s, E=%d\n", __func__, ret); @@ -118,71 +107,25 @@ int tasdevice_spi_dev_read(struct tasdevice_priv *tas_priv, return ret; } -int tasdevice_spi_dev_write(struct tasdevice_priv *tas_priv, - unsigned int reg, - unsigned int value) -{ - struct regmap *map = tas_priv->regmap; - int ret; - - ret = tasdevice_spi_switch_book(tas_priv, reg); - if (ret < 0) - return ret; - - ret = regmap_write(map, TASDEVICE_PAGE_REG(reg), value); - if (ret < 0) - dev_err(tas_priv->dev, "%s, E=%d\n", __func__, ret); - - return ret; -} - -int tasdevice_spi_dev_bulk_write(struct tasdevice_priv *tas_priv, - unsigned int reg, - unsigned char *data, - unsigned int len) -{ - struct regmap *map = tas_priv->regmap; - int ret; - - ret = tasdevice_spi_switch_book(tas_priv, reg); - if (ret < 0) - return ret; - - ret = regmap_bulk_write(map, TASDEVICE_PAGE_REG(reg), data, len); - if (ret < 0) - dev_err(tas_priv->dev, "%s, E=%d\n", __func__, ret); - - return ret; -} - -int tasdevice_spi_dev_bulk_read(struct tasdevice_priv *tas_priv, - unsigned int reg, - unsigned char *data, - unsigned int len) +static int tasdevice_spi_dev_bulk_read(struct tasdevice_priv *tas_priv, + unsigned short chn, unsigned int reg, unsigned char *data, + unsigned int len) { - struct regmap *map = tas_priv->regmap; int ret; - ret = tasdevice_spi_switch_book(tas_priv, reg); - if (ret < 0) - return ret; - - if (len > TASDEVICE_MAX_PAGE) - len = TASDEVICE_MAX_PAGE; /* * In our TAS2781 SPI mode, if read from other book (not book 0), * or read from page number larger than 1 in book 0, one more byte * read is needed, and first byte is a dummy byte, need to be ignored. */ if ((TASDEVICE_BOOK_ID(reg) > 0) || (TASDEVICE_PAGE_ID(reg) > 1)) { - unsigned char buf[TASDEVICE_MAX_PAGE+1]; + unsigned char buf[TASDEVICE_WIN_LEN + 1]; - ret = regmap_bulk_read(map, TASDEVICE_PAGE_REG(reg) | 1, buf, - len + 1); + ret = tasdevice_dev_bulk_read(tas_priv, chn, reg, + buf, len + 1); memcpy(data, buf + 1, len); } else { - ret = regmap_bulk_read(map, TASDEVICE_PAGE_REG(reg) | 1, data, - len); + ret = tasdevice_dev_bulk_read(tas_priv, chn, reg, data, len); } if (ret < 0) dev_err(tas_priv->dev, "%s, E=%d\n", __func__, ret); @@ -190,31 +133,55 @@ int tasdevice_spi_dev_bulk_read(struct tasdevice_priv *tas_priv, return ret; } -int tasdevice_spi_dev_update_bits(struct tasdevice_priv *tas_priv, - unsigned int reg, - unsigned int mask, - unsigned int value) +static int tasdevice_spi_dev_update_bits(struct tasdevice_priv *tas_priv, + unsigned short chn, unsigned int reg, unsigned int mask, + unsigned int value) { - struct regmap *map = tas_priv->regmap; int ret, val; /* * In our TAS2781 SPI mode, read/write was masked in last bit of * address, it cause regmap_update_bits() not work as expected. */ - ret = tasdevice_spi_dev_read(tas_priv, reg, &val); + ret = tasdevice_dev_read(tas_priv, chn, reg, &val); if (ret < 0) { dev_err(tas_priv->dev, "%s, E=%d\n", __func__, ret); return ret; } - ret = regmap_write(map, TASDEVICE_PAGE_REG(reg), - (val & ~mask) | (mask & value)); + + ret = tasdevice_dev_write(tas_priv, chn, TASDEVICE_PAGE_REG(reg), + (val & ~mask) | (mask & value)); if (ret < 0) dev_err(tas_priv->dev, "%s, E=%d\n", __func__, ret); return ret; } +static int tasdevice_spi_change_chn_book(struct tasdevice_priv *p, + unsigned short chn, int book) +{ + int ret = 0; + + if (chn == p->index) { + struct tasdevice *tasdev = &p->tasdevice[chn]; + struct regmap *map = p->regmap; + + if (tasdev->cur_book != book) { + ret = regmap_write(map, TASDEVICE_BOOKCTL_REG, book); + if (ret < 0) + dev_err(p->dev, "%s, E=%d\n", __func__, ret); + else + tasdev->cur_book = book; + } + } else { + ret = -EXDEV; + dev_dbg(p->dev, "Not error, %s ignore channel(%d)\n", + __func__, chn); + } + + return ret; +} + static void tas2781_spi_reset(struct tasdevice_priv *tas_dev) { int ret; @@ -223,12 +190,13 @@ static void tas2781_spi_reset(struct tasdevice_priv *tas_dev) gpiod_set_value_cansleep(tas_dev->reset, 0); fsleep(800); gpiod_set_value_cansleep(tas_dev->reset, 1); + } else { + ret = tasdevice_dev_write(tas_dev, tas_dev->index, + TASDEVICE_REG_SWRESET, TASDEVICE_REG_SWRESET_RESET); + if (ret < 0) + dev_err(tas_dev->dev, "dev sw-reset fail, %d\n", ret); + fsleep(1000); } - ret = tasdevice_spi_dev_write(tas_dev, TAS2781_REG_SWRESET, - TAS2781_REG_SWRESET_RESET); - if (ret < 0) - dev_err(tas_dev->dev, "dev sw-reset fail, %d\n", ret); - fsleep(1000); } static int tascodec_spi_init(struct tasdevice_priv *tas_priv, @@ -245,7 +213,7 @@ static int tascodec_spi_init(struct tasdevice_priv *tas_priv, scnprintf(tas_priv->rca_binaryname, sizeof(tas_priv->rca_binaryname), "%sRCA%d.bin", - tas_priv->dev_name, tas_priv->index); + tas_priv->dev_name, tas_priv->ndev); crc8_populate_msb(tas_priv->crc8_lkp_tbl, TASDEVICE_CRC8_POLYNOMIAL); tas_priv->codec = codec; ret = request_firmware_nowait(module, FW_ACTION_UEVENT, @@ -260,26 +228,22 @@ static int tascodec_spi_init(struct tasdevice_priv *tas_priv, static void tasdevice_spi_init(struct tasdevice_priv *tas_priv) { - tas_priv->cur_prog = -1; - tas_priv->cur_conf = -1; + tas_priv->tasdevice[tas_priv->index].cur_book = -1; + tas_priv->tasdevice[tas_priv->index].cur_conf = -1; + tas_priv->tasdevice[tas_priv->index].cur_prog = -1; - tas_priv->cur_book = -1; - tas_priv->cur_prog = -1; - tas_priv->cur_conf = -1; + tas_priv->isspi = true; - /* Store default registers address for calibration data. */ - tas_priv->cali_reg_array[0] = TASDEVICE_REG(0, 0x17, 0x74); - tas_priv->cali_reg_array[1] = TASDEVICE_REG(0, 0x18, 0x0c); - tas_priv->cali_reg_array[2] = TASDEVICE_REG(0, 0x18, 0x14); - tas_priv->cali_reg_array[3] = TASDEVICE_REG(0, 0x13, 0x70); - tas_priv->cali_reg_array[4] = TASDEVICE_REG(0, 0x18, 0x7c); + tas_priv->update_bits = tasdevice_spi_dev_update_bits; + tas_priv->change_chn_book = tasdevice_spi_change_chn_book; + tas_priv->dev_read = tasdevice_spi_dev_read; + tas_priv->dev_bulk_read = tasdevice_spi_dev_bulk_read; mutex_init(&tas_priv->codec_lock); } static int tasdevice_spi_amp_putvol(struct tasdevice_priv *tas_priv, - struct snd_ctl_elem_value *ucontrol, - struct soc_mixer_control *mc) + struct snd_ctl_elem_value *ucontrol, struct soc_mixer_control *mc) { unsigned int invert = mc->invert; unsigned char mask; @@ -290,7 +254,8 @@ static int tasdevice_spi_amp_putvol(struct tasdevice_priv *tas_priv, mask <<= mc->shift; val = clamp(invert ? max - ucontrol->value.integer.value[0] : ucontrol->value.integer.value[0], 0, max); - ret = tasdevice_spi_dev_update_bits(tas_priv, + + ret = tasdevice_spi_dev_update_bits(tas_priv, tas_priv->index, mc->reg, mask, (unsigned int)(val << mc->shift)); if (ret) dev_err(tas_priv->dev, "set AMP vol error in dev %d\n", @@ -300,16 +265,14 @@ static int tasdevice_spi_amp_putvol(struct tasdevice_priv *tas_priv, } static int tasdevice_spi_amp_getvol(struct tasdevice_priv *tas_priv, - struct snd_ctl_elem_value *ucontrol, - struct soc_mixer_control *mc) + struct snd_ctl_elem_value *ucontrol, struct soc_mixer_control *mc) { unsigned int invert = mc->invert; unsigned char mask = 0; int max = mc->max; int ret, val; - /* Read the primary device */ - ret = tasdevice_spi_dev_read(tas_priv, mc->reg, &val); + ret = tasdevice_spi_dev_read(tas_priv, tas_priv->index, mc->reg, &val); if (ret) { dev_err(tas_priv->dev, "%s, get AMP vol error\n", __func__); return ret; @@ -324,9 +287,8 @@ static int tasdevice_spi_amp_getvol(struct tasdevice_priv *tas_priv, return ret; } -static int tasdevice_spi_digital_putvol(struct tasdevice_priv *tas_priv, - struct snd_ctl_elem_value *ucontrol, - struct soc_mixer_control *mc) +static int tasdevice_spi_digital_putvol(struct tasdevice_priv *p, + struct snd_ctl_elem_value *ucontrol, struct soc_mixer_control *mc) { unsigned int invert = mc->invert; int max = mc->max; @@ -334,26 +296,23 @@ static int tasdevice_spi_digital_putvol(struct tasdevice_priv *tas_priv, val = clamp(invert ? max - ucontrol->value.integer.value[0] : ucontrol->value.integer.value[0], 0, max); - ret = tasdevice_spi_dev_write(tas_priv, mc->reg, (unsigned int)val); + ret = tasdevice_dev_write(p, p->index, mc->reg, (unsigned int)val); if (ret) - dev_err(tas_priv->dev, "set digital vol err in dev %d\n", - tas_priv->index); + dev_err(p->dev, "set digital vol err in dev %d\n", p->index); return ret; } -static int tasdevice_spi_digital_getvol(struct tasdevice_priv *tas_priv, - struct snd_ctl_elem_value *ucontrol, - struct soc_mixer_control *mc) +static int tasdevice_spi_digital_getvol(struct tasdevice_priv *p, + struct snd_ctl_elem_value *ucontrol, struct soc_mixer_control *mc) { unsigned int invert = mc->invert; int max = mc->max; int ret, val; - /* Read the primary device as the whole */ - ret = tasdevice_spi_dev_read(tas_priv, mc->reg, &val); + ret = tasdevice_spi_dev_read(p, p->index, mc->reg, &val); if (ret) { - dev_err(tas_priv->dev, "%s, get digital vol err\n", __func__); + dev_err(p->dev, "%s, get digital vol err\n", __func__); return ret; } @@ -392,7 +351,7 @@ static int tas2781_read_acpi(struct tas2781_hda *tas_hda, ret = -EINVAL; goto err; } - nval = ret; + p->ndev = nval = ret; ret = device_property_read_u32_array(physdev, property, values, nval); if (ret) @@ -435,14 +394,17 @@ static int tas2781_read_acpi(struct tas2781_hda *tas_hda, static void tas2781_hda_playback_hook(struct device *dev, int action) { struct tas2781_hda *tas_hda = dev_get_drvdata(dev); + struct tasdevice_priv *tas_priv = tas_hda->priv; if (action == HDA_GEN_PCM_ACT_OPEN) { pm_runtime_get_sync(dev); - guard(mutex)(&tas_hda->priv->codec_lock); - tasdevice_spi_tuning_switch(tas_hda->priv, 0); + guard(mutex)(&tas_priv->codec_lock); + if (tas_priv->fw_state == TASDEVICE_DSP_FW_ALL_OK) + tasdevice_tuning_switch(tas_hda->priv, 0); } else if (action == HDA_GEN_PCM_ACT_CLOSE) { - guard(mutex)(&tas_hda->priv->codec_lock); - tasdevice_spi_tuning_switch(tas_hda->priv, 1); + guard(mutex)(&tas_priv->codec_lock); + if (tas_priv->fw_state == TASDEVICE_DSP_FW_ALL_OK) + tasdevice_tuning_switch(tas_priv, 1); pm_runtime_mark_last_busy(dev); pm_runtime_put_autosuspend(dev); } @@ -589,6 +551,7 @@ static int tas2781_digital_getvol(struct snd_kcontrol *kcontrol, struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; + guard(mutex)(&tas_priv->codec_lock); return tasdevice_spi_digital_getvol(tas_priv, ucontrol, mc); } @@ -599,6 +562,7 @@ static int tas2781_amp_getvol(struct snd_kcontrol *kcontrol, struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; + guard(mutex)(&tas_priv->codec_lock); return tasdevice_spi_amp_getvol(tas_priv, ucontrol, mc); } @@ -609,7 +573,7 @@ static int tas2781_digital_putvol(struct snd_kcontrol *kcontrol, struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; - /* The check of the given value is in tasdevice_digital_putvol. */ + guard(mutex)(&tas_priv->codec_lock); return tasdevice_spi_digital_putvol(tas_priv, ucontrol, mc); } @@ -620,7 +584,7 @@ static int tas2781_amp_putvol(struct snd_kcontrol *kcontrol, struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; - /* The check of the given value is in tasdevice_amp_putvol. */ + guard(mutex)(&tas_priv->codec_lock); return tasdevice_spi_amp_putvol(tas_priv, ucontrol, mc); } @@ -725,137 +689,164 @@ static const struct snd_kcontrol_new tas2781_dsp_conf_ctrl[] = { static void tas2781_apply_calib(struct tasdevice_priv *tas_priv) { - int i, rc; - - /* - * If no calibration data exist in tasdevice_priv *tas_priv, - * calibration apply will be ignored, and use default values - * in firmware binary, which was loaded during firmware download. - */ - if (tas_priv->cali_data[0] == 0) - return; - /* - * Calibration data was saved in tasdevice_priv *tas_priv as: - * unsigned int cali_data[CALIB_MAX]; - * and every data (in 4 bytes) will be saved in register which in - * book 0, and page number in page_array[], offset was saved in - * rgno_array[]. - */ - for (i = 0; i < CALIB_MAX; i++) { - rc = tasdevice_spi_dev_bulk_write(tas_priv, - tas_priv->cali_reg_array[i], - (unsigned char *)&tas_priv->cali_data[i], 4); - if (rc < 0) - dev_err(tas_priv->dev, - "chn %d calib %d bulk_wr err = %d\n", - tas_priv->index, i, rc); - } -} - -/* - * Update the calibration data, including speaker impedance, f0, etc, - * into algo. Calibrate data is done by manufacturer in the factory. - * These data are used by Algo for calculating the speaker temperature, - * speaker membrane excursion and f0 in real time during playback. - * Calibration data format in EFI is V2, since 2024. - */ -static int tas2781_save_calibration(struct tasdevice_priv *tas_priv) -{ - /* - * GUID was used for data access in BIOS, it was provided by board - * manufactory, like HP: "{02f9af02-7734-4233-b43d-93fe5aa35db3}" - */ - efi_guid_t efi_guid = - EFI_GUID(0x02f9af02, 0x7734, 0x4233, - 0xb4, 0x3d, 0x93, 0xfe, 0x5a, 0xa3, 0x5d, 0xb3); - static efi_char16_t efi_name[] = TASDEVICE_CALIBRATION_DATA_NAME; - unsigned char data[TASDEVICE_CALIBRATION_DATA_SIZE], *buf; - unsigned int attr, crc, offset, *tmp_val; - unsigned long total_sz = 0; - efi_status_t status; + struct calidata *cali_data = &tas_priv->cali_data; + struct cali_reg *r = &cali_data->cali_reg_array; + unsigned char *data = cali_data->data; + unsigned int *tmp_val = (unsigned int *)data; + unsigned int cali_reg[TASDEV_CALIB_N] = { + TASDEVICE_REG(0, 0x17, 0x74), + TASDEVICE_REG(0, 0x18, 0x0c), + TASDEVICE_REG(0, 0x18, 0x14), + TASDEVICE_REG(0, 0x13, 0x70), + TASDEVICE_REG(0, 0x18, 0x7c), + }; + unsigned int crc, oft; + unsigned char *buf; + int i, j, k, l; - tas_priv->cali_data[0] = 0; - status = efi.get_variable(efi_name, &efi_guid, &attr, &total_sz, data); - if (status == EFI_BUFFER_TOO_SMALL) { - if (total_sz > TASDEVICE_CALIBRATION_DATA_SIZE) - return -ENOMEM; - /* Get variable contents into buffer */ - status = efi.get_variable(efi_name, &efi_guid, &attr, - &total_sz, data); - } - if (status != EFI_SUCCESS) - return status; - - tmp_val = (unsigned int *)data; if (tmp_val[0] == 2781) { /* * New features were added in calibrated Data V3: * 1. Added calibration registers address define in - * a node, marked as Device id == 0x80. + * a node, marked as Device id == 0x80. * New features were added in calibrated Data V2: * 1. Added some the fields to store the link_id and - * uniqie_id for multi-link solutions + * uniqie_id for multi-link solutions * 2. Support flexible number of devices instead of - * fixed one in V1. + * fixed one in V1. * Layout of calibrated data V2 in UEFI(total 256 bytes): * ChipID (2781, 4 bytes) - * Device-Sum (4 bytes) + * Data-Group-Sum (4 bytes) * TimeStamp of Calibration (4 bytes) - * for (i = 0; i < Device-Sum; i++) { - * Device #i index_info () { - * SDW link id (2bytes) - * SDW unique_id (2bytes) - * } // if Device number is 0x80, mean it's - * calibration registers address. - * Calibrated Data of Device #i (20 bytes) + * for (i = 0; i < Data-Group-Sum; i++) { + * if (Data type != 0x80) (4 bytes) + * Calibrated Data of Device #i (20 bytes) + * else + * Calibration registers address (5*4 = 20 bytes) + * # V2: No reg addr in data grp section. + * # V3: Normally the last grp is the reg addr. * } * CRC (4 bytes) * Reserved (the rest) */ crc = crc32(~0, data, (3 + tmp_val[1] * 6) * 4) ^ ~0; - if (crc != tmp_val[3 + tmp_val[1] * 6]) - return 0; - - for (int j = 0; j < tmp_val[1]; j++) { - offset = j * 6 + 3; - if (tmp_val[offset] == tas_priv->index) { - for (int i = 0; i < CALIB_MAX; i++) - tas_priv->cali_data[i] = - tmp_val[offset + i + 1]; - } else if (tmp_val[offset] == - TASDEVICE_CALIBRATION_REG_ADDRESS) { - for (int i = 0; i < CALIB_MAX; i++) { - buf = &data[(offset + i + 1) * 4]; - tas_priv->cali_reg_array[i] = - TASDEVICE_REG(buf[1], buf[2], - buf[3]); + if (crc != tmp_val[3 + tmp_val[1] * 6]) { + cali_data->total_sz = 0; + dev_err(tas_priv->dev, "%s: CRC error\n", __func__); + return; + } + + for (j = 0, k = 0; j < tmp_val[1]; j++) { + oft = j * 6 + 3; + if (tmp_val[oft] == TASDEV_UEFI_CALI_REG_ADDR_FLG) { + for (i = 0; i < TASDEV_CALIB_N; i++) { + buf = &data[(oft + i + 1) * 4]; + cali_reg[i] = TASDEVICE_REG(buf[1], + buf[2], buf[3]); + } + } else { + l = j * (cali_data->cali_dat_sz_per_dev + 1); + if (k >= tas_priv->ndev || l > oft * 4) { + dev_err(tas_priv->dev, + "%s: dev sum error\n", + __func__); + cali_data->total_sz = 0; + return; } + + data[l] = k; + for (i = 0; i < TASDEV_CALIB_N * 4; i++) + data[l + i] = data[4 * oft + i]; + k++; } - tas_priv->apply_calibration(tas_priv); } } else { /* * Calibration data is in V1 format. * struct cali_data { - * char cali_data[20]; + * char cali_data[20]; * } * * struct { - * struct cali_data cali_data[4]; - * int TimeStamp of Calibration (4 bytes) - * int CRC (4 bytes) + * struct cali_data cali_data[4]; + * int TimeStamp of Calibration (4 bytes) + * int CRC (4 bytes) * } ueft; */ crc = crc32(~0, data, 84) ^ ~0; - if (crc == tmp_val[21]) { - for (int i = 0; i < CALIB_MAX; i++) - tas_priv->cali_data[i] = - tmp_val[tas_priv->index * 5 + i]; - tas_priv->apply_calibration(tas_priv); + if (crc != tmp_val[21]) { + cali_data->total_sz = 0; + dev_err(tas_priv->dev, "%s: V1 CRC error\n", __func__); + return; + } + + for (j = tas_priv->ndev - 1; j >= 0; j--) { + l = j * (cali_data->cali_dat_sz_per_dev + 1); + for (i = TASDEV_CALIB_N * 4; i > 0 ; i--) + data[l + i] = data[tas_priv->index * 5 + i]; + data[l+i] = j; + } + } + + if (tas_priv->dspbin_typ == TASDEV_BASIC) { + r->r0_reg = cali_reg[0]; + r->invr0_reg = cali_reg[1]; + r->r0_low_reg = cali_reg[2]; + r->pow_reg = cali_reg[3]; + r->tlimit_reg = cali_reg[4]; + } + + tas_priv->is_user_space_calidata = true; + cali_data->total_sz = + tas_priv->ndev * (cali_data->cali_dat_sz_per_dev + 1); +} + +/* + * Update the calibration data, including speaker impedance, f0, etc, + * into algo. Calibrate data is done by manufacturer in the factory. + * These data are used by Algo for calculating the speaker temperature, + * speaker membrane excursion and f0 in real time during playback. + * Calibration data format in EFI is V2, since 2024. + */ +static int tas2781_save_calibration(struct tasdevice_priv *tas_priv) +{ + /* + * GUID was used for data access in BIOS, it was provided by board + * manufactory, like HP: "{02f9af02-7734-4233-b43d-93fe5aa35db3}" + */ + efi_guid_t efi_guid = + EFI_GUID(0x02f9af02, 0x7734, 0x4233, + 0xb4, 0x3d, 0x93, 0xfe, 0x5a, 0xa3, 0x5d, 0xb3); + static efi_char16_t efi_name[] = TASDEVICE_CALIBRATION_DATA_NAME; + struct calidata *cali_data = &tas_priv->cali_data; + unsigned long total_sz = 0; + unsigned int attr, size; + efi_status_t status; + unsigned char *data; + + cali_data->cali_dat_sz_per_dev = 20; + size = tas_priv->ndev * (cali_data->cali_dat_sz_per_dev + 1); + status = efi.get_variable(efi_name, &efi_guid, &attr, &total_sz, NULL); + cali_data->total_sz = total_sz > size ? total_sz : size; + if (status == EFI_BUFFER_TOO_SMALL) { + /* Allocate data buffer of data_size bytes */ + data = tas_priv->cali_data.data = devm_kzalloc(tas_priv->dev, + tas_priv->cali_data.total_sz, GFP_KERNEL); + if (!data) { + tas_priv->cali_data.total_sz = 0; + return -ENOMEM; } + /* Get variable contents into buffer */ + status = efi.get_variable(efi_name, &efi_guid, &attr, + &tas_priv->cali_data.total_sz, data); + } + if (status != EFI_SUCCESS) { + tas_priv->cali_data.total_sz = 0; + return status; } + tas_priv->apply_calibration(tas_priv); return 0; } @@ -883,7 +874,7 @@ static void tasdev_fw_ready(const struct firmware *fmw, void *context) pm_runtime_get_sync(tas_priv->dev); guard(mutex)(&tas_priv->codec_lock); - ret = tasdevice_spi_rca_parser(tas_priv, fmw); + ret = tasdevice_rca_parser(tas_priv, fmw); if (ret) goto out; @@ -910,12 +901,12 @@ static void tasdev_fw_ready(const struct firmware *fmw, void *context) } } - tasdevice_spi_dsp_remove(tas_priv); + tasdevice_dsp_remove(tas_priv); tas_priv->fw_state = TASDEVICE_DSP_FW_PENDING; scnprintf(tas_priv->coef_binaryname, 64, "TAS2XXX%08X-%01d.bin", codec->core.subsystem_id, tas_priv->index); - ret = tasdevice_spi_dsp_parser(tas_priv); + ret = tasdevice_dsp_parser(tas_priv); if (ret) { dev_err(tas_priv->dev, "dspfw load %s error\n", tas_priv->coef_binaryname); @@ -945,26 +936,27 @@ static void tasdev_fw_ready(const struct firmware *fmw, void *context) goto out; } - /* Perform AMP reset before firmware download. */ - tas_priv->rcabin.profile_cfg_id = TAS2781_PRE_POST_RESET_CFG; tas2781_spi_reset(tas_priv); tas_priv->rcabin.profile_cfg_id = 0; tas_priv->fw_state = TASDEVICE_DSP_FW_ALL_OK; - ret = tasdevice_spi_dev_read(tas_priv, TAS2781_REG_CLK_CONFIG, &val); + ret = tas_priv->dev_read(tas_priv, tas_priv->index, + TAS2781_REG_CLK_CONFIG, &val); if (ret < 0) goto out; - if (val == TAS2781_REG_CLK_CONFIG_RESET) - ret = tasdevice_spi_prmg_load(tas_priv, 0); - if (ret < 0) { - dev_err(tas_priv->dev, "FW download failed = %d\n", ret); - goto out; + if (val == TAS2781_REG_CLK_CONFIG_RESET) { + ret = tasdevice_prmg_load(tas_priv, 0); + if (ret < 0) { + dev_err(tas_priv->dev, "FW download failed = %d\n", + ret); + goto out; + } } if (tas_priv->fmw->nr_programs > 0) - tas_priv->cur_prog = 0; + tas_priv->tasdevice[tas_priv->index].cur_prog = 0; if (tas_priv->fmw->nr_configurations > 0) - tas_priv->cur_conf = 0; + tas_priv->tasdevice[tas_priv->index].cur_conf = 0; /* * If calibrated data occurs error, dsp will still works with default @@ -1017,9 +1009,10 @@ static void tas2781_hda_unbind(struct device *dev, struct device *master, { struct tas2781_hda *tas_hda = dev_get_drvdata(dev); struct hda_component_parent *parent = master_data; + struct tasdevice_priv *tas_priv = tas_hda->priv; struct hda_component *comp; - comp = hda_component_from_index(parent, tas_hda->priv->index); + comp = hda_component_from_index(parent, tas_priv->index); if (comp && (comp->dev == dev)) { comp->dev = NULL; memset(comp->name, 0, sizeof(comp->name)); @@ -1028,8 +1021,8 @@ static void tas2781_hda_unbind(struct device *dev, struct device *master, tas2781_hda_remove_controls(tas_hda); - tasdevice_spi_config_info_remove(tas_hda->priv); - tasdevice_spi_dsp_remove(tas_hda->priv); + tasdevice_config_info_remove(tas_priv); + tasdevice_dsp_remove(tas_priv); tas_hda->priv->fw_state = TASDEVICE_DSP_FW_PENDING; } @@ -1124,14 +1117,16 @@ static void tas2781_hda_spi_remove(struct spi_device *spi) static int tas2781_runtime_suspend(struct device *dev) { struct tas2781_hda *tas_hda = dev_get_drvdata(dev); + struct tasdevice_priv *tas_priv = tas_hda->priv; - guard(mutex)(&tas_hda->priv->codec_lock); + guard(mutex)(&tas_priv->codec_lock); - if (tas_hda->priv->playback_started) - tasdevice_spi_tuning_switch(tas_hda->priv, 1); + if (tas_priv->fw_state == TASDEVICE_DSP_FW_ALL_OK + && tas_priv->playback_started) + tasdevice_tuning_switch(tas_priv, 1); - tas_hda->priv->cur_book = -1; - tas_hda->priv->cur_conf = -1; + tas_priv->tasdevice[tas_priv->index].cur_book = -1; + tas_priv->tasdevice[tas_priv->index].cur_conf = -1; return 0; } @@ -1139,11 +1134,13 @@ static int tas2781_runtime_suspend(struct device *dev) static int tas2781_runtime_resume(struct device *dev) { struct tas2781_hda *tas_hda = dev_get_drvdata(dev); + struct tasdevice_priv *tas_priv = tas_hda->priv; - guard(mutex)(&tas_hda->priv->codec_lock); + guard(mutex)(&tas_priv->codec_lock); - if (tas_hda->priv->playback_started) - tasdevice_spi_tuning_switch(tas_hda->priv, 0); + if (tas_priv->fw_state == TASDEVICE_DSP_FW_ALL_OK + && tas_priv->playback_started) + tasdevice_tuning_switch(tas_priv, 0); return 0; } @@ -1151,6 +1148,7 @@ static int tas2781_runtime_resume(struct device *dev) static int tas2781_system_suspend(struct device *dev) { struct tas2781_hda *tas_hda = dev_get_drvdata(dev); + struct tasdevice_priv *tas_priv = tas_hda->priv; int ret; ret = pm_runtime_force_suspend(dev); @@ -1158,8 +1156,9 @@ static int tas2781_system_suspend(struct device *dev) return ret; /* Shutdown chip before system suspend */ - if (tas_hda->priv->playback_started) - tasdevice_spi_tuning_switch(tas_hda->priv, 1); + if (tas_priv->fw_state == TASDEVICE_DSP_FW_ALL_OK + && tas_priv->playback_started) + tasdevice_tuning_switch(tas_priv, 1); return 0; } @@ -1167,32 +1166,34 @@ static int tas2781_system_suspend(struct device *dev) static int tas2781_system_resume(struct device *dev) { struct tas2781_hda *tas_hda = dev_get_drvdata(dev); + struct tasdevice_priv *tas_priv = tas_hda->priv; int ret, val; ret = pm_runtime_force_resume(dev); if (ret) return ret; - guard(mutex)(&tas_hda->priv->codec_lock); - ret = tasdevice_spi_dev_read(tas_hda->priv, TAS2781_REG_CLK_CONFIG, - &val); + guard(mutex)(&tas_priv->codec_lock); + ret = tas_priv->dev_read(tas_priv, tas_priv->index, + TAS2781_REG_CLK_CONFIG, &val); if (ret < 0) return ret; if (val == TAS2781_REG_CLK_CONFIG_RESET) { - tas_hda->priv->cur_book = -1; - tas_hda->priv->cur_conf = -1; - tas_hda->priv->cur_prog = -1; + tas_priv->tasdevice[tas_priv->index].cur_book = -1; + tas_priv->tasdevice[tas_priv->index].cur_conf = -1; + tas_priv->tasdevice[tas_priv->index].cur_prog = -1; - ret = tasdevice_spi_prmg_load(tas_hda->priv, 0); + ret = tasdevice_prmg_load(tas_priv, 0); if (ret < 0) { - dev_err(tas_hda->priv->dev, + dev_err(tas_priv->dev, "FW download failed = %d\n", ret); return ret; } + tas_priv->fw_state = TASDEVICE_DSP_FW_ALL_OK; - if (tas_hda->priv->playback_started) - tasdevice_spi_tuning_switch(tas_hda->priv, 0); + if (tas_priv->playback_started) + tasdevice_tuning_switch(tas_priv, 0); } return ret; @@ -1229,3 +1230,4 @@ module_spi_driver(tas2781_hda_spi_driver); MODULE_DESCRIPTION("TAS2781 HDA SPI Driver"); MODULE_AUTHOR("Baojun, Xu, <baojun.xug(a)ti.com>"); MODULE_LICENSE("GPL"); +MODULE_IMPORT_NS("SND_SOC_TAS2781_FMWLIB"); diff --git a/sound/pci/hda/tas2781_spi_fwlib.c b/sound/pci/hda/tas2781_spi_fwlib.c deleted file mode 100644 index d90d022d8449..000000000000 --- a/sound/pci/hda/tas2781_spi_fwlib.c +++ /dev/null @@ -1,2006 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0 -// -// TAS2781 HDA SPI driver -// -// Copyright 2024 - 2025 Texas Instruments, Inc. -// -// Author: Baojun Xu <baojun.xu(a)ti.com> - -#include <linux/crc8.h> -#include <linux/firmware.h> -#include <linux/init.h> -#include <linux/interrupt.h> -#include <linux/module.h> -#include <linux/slab.h> -#include <linux/types.h> -#include <linux/unaligned.h> -#include <sound/pcm_params.h> -#include <sound/soc.h> -#include <sound/tas2781-dsp.h> -#include <sound/tlv.h> - -#include "tas2781-spi.h" - -#define OFFSET_ERROR_BIT BIT(31) - -#define ERROR_PRAM_CRCCHK 0x0000000 -#define ERROR_YRAM_CRCCHK 0x0000001 -#define PPC_DRIVER_CRCCHK 0x00000200 - -#define TAS2781_SA_COEFF_SWAP_REG TASDEVICE_REG(0, 0x35, 0x2c) -#define TAS2781_YRAM_BOOK1 140 -#define TAS2781_YRAM1_PAGE 42 -#define TAS2781_YRAM1_START_REG 88 - -#define TAS2781_YRAM2_START_PAGE 43 -#define TAS2781_YRAM2_END_PAGE 49 -#define TAS2781_YRAM2_START_REG 8 -#define TAS2781_YRAM2_END_REG 127 - -#define TAS2781_YRAM3_PAGE 50 -#define TAS2781_YRAM3_START_REG 8 -#define TAS2781_YRAM3_END_REG 27 - -/* should not include B0_P53_R44-R47 */ -#define TAS2781_YRAM_BOOK2 0 -#define TAS2781_YRAM4_START_PAGE 50 -#define TAS2781_YRAM4_END_PAGE 60 - -#define TAS2781_YRAM5_PAGE 61 -#define TAS2781_YRAM5_START_REG TAS2781_YRAM3_START_REG -#define TAS2781_YRAM5_END_REG TAS2781_YRAM3_END_REG - -#define TASDEVICE_MAXPROGRAM_NUM_KERNEL 5 -#define TASDEVICE_MAXCONFIG_NUM_KERNEL_MULTIPLE_AMPS 64 -#define TASDEVICE_MAXCONFIG_NUM_KERNEL 10 -#define MAIN_ALL_DEVICES_1X 0x01 -#define MAIN_DEVICE_A_1X 0x02 -#define MAIN_DEVICE_B_1X 0x03 -#define MAIN_DEVICE_C_1X 0x04 -#define MAIN_DEVICE_D_1X 0x05 -#define COEFF_DEVICE_A_1X 0x12 -#define COEFF_DEVICE_B_1X 0x13 -#define COEFF_DEVICE_C_1X 0x14 -#define COEFF_DEVICE_D_1X 0x15 -#define PRE_DEVICE_A_1X 0x22 -#define PRE_DEVICE_B_1X 0x23 -#define PRE_DEVICE_C_1X 0x24 -#define PRE_DEVICE_D_1X 0x25 -#define PRE_SOFTWARE_RESET_DEVICE_A 0x41 -#define PRE_SOFTWARE_RESET_DEVICE_B 0x42 -#define PRE_SOFTWARE_RESET_DEVICE_C 0x43 -#define PRE_SOFTWARE_RESET_DEVICE_D 0x44 -#define POST_SOFTWARE_RESET_DEVICE_A 0x45 -#define POST_SOFTWARE_RESET_DEVICE_B 0x46 -#define POST_SOFTWARE_RESET_DEVICE_C 0x47 -#define POST_SOFTWARE_RESET_DEVICE_D 0x48 - -struct tas_crc { - unsigned char offset; - unsigned char len; -}; - -struct blktyp_devidx_map { - unsigned char blktyp; - unsigned char dev_idx; -}; - -/* fixed m68k compiling issue: mapping table can save code field */ -static const struct blktyp_devidx_map ppc3_tas2781_mapping_table[] = { - { MAIN_ALL_DEVICES_1X, 0x80 }, - { MAIN_DEVICE_A_1X, 0x81 }, - { COEFF_DEVICE_A_1X, 0x81 }, - { PRE_DEVICE_A_1X, 0x81 }, - { PRE_SOFTWARE_RESET_DEVICE_A, 0xC1 }, - { POST_SOFTWARE_RESET_DEVICE_A, 0xC1 }, - { MAIN_DEVICE_B_1X, 0x82 }, - { COEFF_DEVICE_B_1X, 0x82 }, - { PRE_DEVICE_B_1X, 0x82 }, - { PRE_SOFTWARE_RESET_DEVICE_B, 0xC2 }, - { POST_SOFTWARE_RESET_DEVICE_B, 0xC2 }, - { MAIN_DEVICE_C_1X, 0x83 }, - { COEFF_DEVICE_C_1X, 0x83 }, - { PRE_DEVICE_C_1X, 0x83 }, - { PRE_SOFTWARE_RESET_DEVICE_C, 0xC3 }, - { POST_SOFTWARE_RESET_DEVICE_C, 0xC3 }, - { MAIN_DEVICE_D_1X, 0x84 }, - { COEFF_DEVICE_D_1X, 0x84 }, - { PRE_DEVICE_D_1X, 0x84 }, - { PRE_SOFTWARE_RESET_DEVICE_D, 0xC4 }, - { POST_SOFTWARE_RESET_DEVICE_D, 0xC4 }, -}; - -static const struct blktyp_devidx_map ppc3_mapping_table[] = { - { MAIN_ALL_DEVICES_1X, 0x80 }, - { MAIN_DEVICE_A_1X, 0x81 }, - { COEFF_DEVICE_A_1X, 0xC1 }, - { PRE_DEVICE_A_1X, 0xC1 }, - { MAIN_DEVICE_B_1X, 0x82 }, - { COEFF_DEVICE_B_1X, 0xC2 }, - { PRE_DEVICE_B_1X, 0xC2 }, - { MAIN_DEVICE_C_1X, 0x83 }, - { COEFF_DEVICE_C_1X, 0xC3 }, - { PRE_DEVICE_C_1X, 0xC3 }, - { MAIN_DEVICE_D_1X, 0x84 }, - { COEFF_DEVICE_D_1X, 0xC4 }, - { PRE_DEVICE_D_1X, 0xC4 }, -}; - -static const struct blktyp_devidx_map non_ppc3_mapping_table[] = { - { MAIN_ALL_DEVICES, 0x80 }, - { MAIN_DEVICE_A, 0x81 }, - { COEFF_DEVICE_A, 0xC1 }, - { PRE_DEVICE_A, 0xC1 }, - { MAIN_DEVICE_B, 0x82 }, - { COEFF_DEVICE_B, 0xC2 }, - { PRE_DEVICE_B, 0xC2 }, - { MAIN_DEVICE_C, 0x83 }, - { COEFF_DEVICE_C, 0xC3 }, - { PRE_DEVICE_C, 0xC3 }, - { MAIN_DEVICE_D, 0x84 }, - { COEFF_DEVICE_D, 0xC4 }, - { PRE_DEVICE_D, 0xC4 }, -}; - -/* - * Device support different configurations for different scene, - * like voice, music, calibration, was write in regbin file. - * Will be stored into tas_priv after regbin was loaded. - */ -static struct tasdevice_config_info *tasdevice_add_config( - struct tasdevice_priv *tas_priv, unsigned char *config_data, - unsigned int config_size, int *status) -{ - struct tasdevice_config_info *cfg_info; - struct tasdev_blk_data **bk_da; - unsigned int config_offset = 0; - unsigned int i; - - /* - * In most projects are many audio cases, such as music, handfree, - * receiver, games, audio-to-haptics, PMIC record, bypass mode, - * portrait, landscape, etc. Even in multiple audios, one or - * two of the chips will work for the special case, such as - * ultrasonic application. In order to support these variable-numbers - * of audio cases, flexible configs have been introduced in the - * DSP firmware. - */ - cfg_info = kzalloc(sizeof(*cfg_info), GFP_KERNEL); - if (!cfg_info) { - *status = -ENOMEM; - return NULL; - } - - if (tas_priv->rcabin.fw_hdr.binary_version_num >= 0x105) { - if ((config_offset + 64) > config_size) { - *status = -EINVAL; - dev_err(tas_priv->dev, "add conf: Out of boundary\n"); - goto config_err; - } - config_offset += 64; - } - - if ((config_offset + 4) > config_size) { - *status = -EINVAL; - dev_err(tas_priv->dev, "add config: Out of boundary\n"); - goto config_err; - } - - /* - * convert data[offset], data[offset + 1], data[offset + 2] and - * data[offset + 3] into host - */ - cfg_info->nblocks = get_unaligned_be32(&config_data[config_offset]); - config_offset += 4; - - /* - * Several kinds of dsp/algorithm firmwares can run on tas2781, - * the number and size of blk are not fixed and different among - * these firmwares. - */ - bk_da = cfg_info->blk_data = kcalloc(cfg_info->nblocks, - sizeof(*bk_da), GFP_KERNEL); - if (!bk_da) { - *status = -ENOMEM; - goto config_err; - } - cfg_info->real_nblocks = 0; - for (i = 0; i < cfg_info->nblocks; i++) { - if (config_offset + 12 > config_size) { - *status = -EINVAL; - dev_err(tas_priv->dev, - "%s: Out of boundary: i = %d nblocks = %u!\n", - __func__, i, cfg_info->nblocks); - goto block_err; - } - bk_da[i] = kzalloc(sizeof(*bk_da[i]), GFP_KERNEL); - if (!bk_da[i]) { - *status = -ENOMEM; - goto block_err; - } - - bk_da[i]->dev_idx = config_data[config_offset]; - config_offset++; - - bk_da[i]->block_type = config_data[config_offset]; - config_offset++; - - bk_da[i]->yram_checksum = - get_unaligned_be16(&config_data[config_offset]); - config_offset += 2; - bk_da[i]->block_size = - get_unaligned_be32(&config_data[config_offset]); - config_offset += 4; - - bk_da[i]->n_subblks = - get_unaligned_be32(&config_data[config_offset]); - - config_offset += 4; - - if (config_offset + bk_da[i]->block_size > config_size) { - *status = -EINVAL; - dev_err(tas_priv->dev, - "%s: Out of boundary: i = %d blks = %u!\n", - __func__, i, cfg_info->nblocks); - goto block_err; - } - /* instead of kzalloc+memcpy */ - bk_da[i]->regdata = kmemdup(&config_data[config_offset], - bk_da[i]->block_size, GFP_KERNEL); - if (!bk_da[i]->regdata) { - *status = -ENOMEM; - i++; - goto block_err; - } - - config_offset += bk_da[i]->block_size; - cfg_info->real_nblocks += 1; - } - - return cfg_info; -block_err: - for (int j = 0; j < i; j++) - kfree(bk_da[j]); - kfree(bk_da); -config_err: - kfree(cfg_info); - return NULL; -} - -/* Regbin file parser function. */ -int tasdevice_spi_rca_parser(void *context, const struct firmware *fmw) -{ - struct tasdevice_priv *tas_priv = context; - struct tasdevice_config_info **cfg_info; - struct tasdevice_rca_hdr *fw_hdr; - struct tasdevice_rca *rca; - unsigned int total_config_sz = 0; - int offset = 0, ret = 0, i; - unsigned char *buf; - - rca = &tas_priv->rcabin; - fw_hdr = &rca->fw_hdr; - if (!fmw || !fmw->data) { - dev_err(tas_priv->dev, "Failed to read %s\n", - tas_priv->rca_binaryname); - tas_priv->fw_state = TASDEVICE_DSP_FW_FAIL; - return -EINVAL; - } - buf = (unsigned char *)fmw->data; - fw_hdr->img_sz = get_unaligned_be32(&buf[offset]); - offset += 4; - if (fw_hdr->img_sz != fmw->size) { - dev_err(tas_priv->dev, - "File size not match, %d %u", (int)fmw->size, - fw_hdr->img_sz); - tas_priv->fw_state = TASDEVICE_DSP_FW_FAIL; - return -EINVAL; - } - - fw_hdr->checksum = get_unaligned_be32(&buf[offset]); - offset += 4; - fw_hdr->binary_version_num = get_unaligned_be32(&buf[offset]); - if (fw_hdr->binary_version_num < 0x103) { - dev_err(tas_priv->dev, "File version 0x%04x is too low", - fw_hdr->binary_version_num); - tas_priv->fw_state = TASDEVICE_DSP_FW_FAIL; - return -EINVAL; - } - offset += 4; - fw_hdr->drv_fw_version = get_unaligned_be32(&buf[offset]); - offset += 8; - fw_hdr->plat_type = buf[offset++]; - fw_hdr->dev_family = buf[offset++]; - fw_hdr->reserve = buf[offset++]; - fw_hdr->ndev = buf[offset++]; - if (offset + TASDEVICE_DEVICE_SUM > fw_hdr->img_sz) { - dev_err(tas_priv->dev, "rca_ready: Out of boundary!\n"); - tas_priv->fw_state = TASDEVICE_DSP_FW_FAIL; - return -EINVAL; - } - - for (i = 0; i < TASDEVICE_DEVICE_SUM; i++, offset++) - fw_hdr->devs[i] = buf[offset]; - - fw_hdr->nconfig = get_unaligned_be32(&buf[offset]); - offset += 4; - - for (i = 0; i < TASDEVICE_CONFIG_SUM; i++) { - fw_hdr->config_size[i] = get_unaligned_be32(&buf[offset]); - offset += 4; - total_config_sz += fw_hdr->config_size[i]; - } - - if (fw_hdr->img_sz - total_config_sz != (unsigned int)offset) { - dev_err(tas_priv->dev, "Bin file err %d - %d != %d!\n", - fw_hdr->img_sz, total_config_sz, (int)offset); - tas_priv->fw_state = TASDEVICE_DSP_FW_FAIL; - return -EINVAL; - } - - cfg_info = kcalloc(fw_hdr->nconfig, sizeof(*cfg_info), GFP_KERNEL); - if (!cfg_info) { - tas_priv->fw_state = TASDEVICE_DSP_FW_FAIL; - return -ENOMEM; - } - rca->cfg_info = cfg_info; - rca->ncfgs = 0; - for (i = 0; i < (int)fw_hdr->nconfig; i++) { - rca->ncfgs += 1; - cfg_info[i] = tasdevice_add_config(tas_priv, &buf[offset], - fw_hdr->config_size[i], &ret); - if (ret) { - tas_priv->fw_state = TASDEVICE_DSP_FW_FAIL; - return ret; - } - offset += (int)fw_hdr->config_size[i]; - } - - return ret; -} - -/* fixed m68k compiling issue: mapping table can save code field */ -static unsigned char map_dev_idx(struct tasdevice_fw *tas_fmw, - struct tasdev_blk *block) -{ - struct blktyp_devidx_map *p = - (struct blktyp_devidx_map *)non_ppc3_mapping_table; - struct tasdevice_dspfw_hdr *fw_hdr = &tas_fmw->fw_hdr; - struct tasdevice_fw_fixed_hdr *fw_fixed_hdr = &fw_hdr->fixed_hdr; - int i, n = ARRAY_SIZE(non_ppc3_mapping_table); - unsigned char dev_idx = 0; - - if (fw_fixed_hdr->ppcver >= PPC3_VERSION_TAS2781_BASIC_MIN) { - p = (struct blktyp_devidx_map *)ppc3_tas2781_mapping_table; - n = ARRAY_SIZE(ppc3_tas2781_mapping_table); - } else if (fw_fixed_hdr->ppcver >= PPC3_VERSION_BASE) { - p = (struct blktyp_devidx_map *)ppc3_mapping_table; - n = ARRAY_SIZE(ppc3_mapping_table); - } - - for (i = 0; i < n; i++) { - if (block->type == p[i].blktyp) { - dev_idx = p[i].dev_idx; - break; - } - } - - return dev_idx; -} - -/* Block parser function. */ -static int fw_parse_block_data_kernel(struct tasdevice_fw *tas_fmw, - struct tasdev_blk *block, const struct firmware *fmw, int offset) -{ - const unsigned char *data = fmw->data; - - if (offset + 16 > fmw->size) { - dev_err(tas_fmw->dev, "%s: File Size error\n", __func__); - return -EINVAL; - } - - /* - * Convert data[offset], data[offset + 1], data[offset + 2] and - * data[offset + 3] into host. - */ - block->type = get_unaligned_be32(&data[offset]); - offset += 4; - - block->is_pchksum_present = data[offset++]; - block->pchksum = data[offset++]; - block->is_ychksum_present = data[offset++]; - block->ychksum = data[offset++]; - block->blk_size = get_unaligned_be32(&data[offset]); - offset += 4; - block->nr_subblocks = get_unaligned_be32(&data[offset]); - offset += 4; - - /* - * Fixed m68k compiling issue: - * 1. mapping table can save code field. - * 2. storing the dev_idx as a member of block can reduce unnecessary - * time and system resource comsumption of dev_idx mapping every - * time the block data writing to the dsp. - */ - block->dev_idx = map_dev_idx(tas_fmw, block); - - if (offset + block->blk_size > fmw->size) { - dev_err(tas_fmw->dev, "%s: nSublocks error\n", __func__); - return -EINVAL; - } - /* instead of kzalloc+memcpy */ - block->data = kmemdup(&data[offset], block->blk_size, GFP_KERNEL); - if (!block->data) - return -ENOMEM; - - offset += block->blk_size; - - return offset; -} - -/* Data of block parser function. */ -static int fw_parse_data_kernel(struct tasdevice_fw *tas_fmw, - struct tasdevice_data *img_data, const struct firmware *fmw, - int offset) -{ - const unsigned char *data = fmw->data; - struct tasdev_blk *blk; - unsigned int i; - - if (offset + 4 > fmw->size) { - dev_err(tas_fmw->dev, "%s: File Size error\n", __func__); - return -EINVAL; - } - img_data->nr_blk = get_unaligned_be32(&data[offset]); - offset += 4; - - img_data->dev_blks = kcalloc(img_data->nr_blk, - sizeof(struct tasdev_blk), GFP_KERNEL); - if (!img_data->dev_blks) - return -ENOMEM; - - for (i = 0; i < img_data->nr_blk; i++) { - blk = &img_data->dev_blks[i]; - offset = fw_parse_block_data_kernel( - tas_fmw, blk, fmw, offset); - if (offset < 0) { - kfree(img_data->dev_blks); - return -EINVAL; - } - } - - return offset; -} - -/* Data of DSP program parser function. */ -static int fw_parse_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 72 unused byts */ - offset += 72; - - offset = fw_parse_data_kernel(tas_fmw, &program->dev_data, - fmw, offset); - if (offset < 0) - break; - } - - return offset; -} - -/* Data of DSP configurations parser function. */ -static int fw_parse_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 16 bytes */ - offset += 80; - - offset = fw_parse_data_kernel(tas_fmw, &config->dev_data, - fmw, offset); - if (offset < 0) - break; - } - - return offset; -} - -/* DSP firmware file header parser function for early PPC3 firmware binary. */ -static int fw_parse_variable_header_kernel(struct tasdevice_priv *tas_priv, - const struct firmware *fmw, int offset) -{ - struct tasdevice_fw *tas_fmw = tas_priv->fmw; - struct tasdevice_dspfw_hdr *fw_hdr = &tas_fmw->fw_hdr; - struct tasdevice_config *config; - struct tasdevice_prog *program; - const unsigned char *buf = fmw->data; - unsigned short max_confs; - unsigned int i; - - if (offset + 12 + 4 * TASDEVICE_MAXPROGRAM_NUM_KERNEL > fmw->size) { - dev_err(tas_priv->dev, "%s: File Size error\n", __func__); - return -EINVAL; - } - fw_hdr->device_family = get_unaligned_be16(&buf[offset]); - if (fw_hdr->device_family != 0) { - dev_err(tas_priv->dev, "%s:not TAS device\n", __func__); - return -EINVAL; - } - offset += 2; - fw_hdr->device = get_unaligned_be16(&buf[offset]); - if (fw_hdr->device >= TASDEVICE_DSP_TAS_MAX_DEVICE || - fw_hdr->device == 6) { - dev_err(tas_priv->dev, "Unsupported dev %d\n", fw_hdr->device); - return -EINVAL; - } - offset += 2; - - tas_fmw->nr_programs = get_unaligned_be32(&buf[offset]); - offset += 4; - - if (tas_fmw->nr_programs == 0 || - tas_fmw->nr_programs > TASDEVICE_MAXPROGRAM_NUM_KERNEL) { - dev_err(tas_priv->dev, "mnPrograms is invalid\n"); - return -EINVAL; - } - - tas_fmw->programs = kcalloc(tas_fmw->nr_programs, - sizeof(*tas_fmw->programs), GFP_KERNEL); - if (!tas_fmw->programs) - return -ENOMEM; - - for (i = 0; i < tas_fmw->nr_programs; i++) { - program = &tas_fmw->programs[i]; - program->prog_size = get_unaligned_be32(&buf[offset]); - offset += 4; - } - - /* Skip the unused prog_size */ - offset += 4 * (TASDEVICE_MAXPROGRAM_NUM_KERNEL - tas_fmw->nr_programs); - - tas_fmw->nr_configurations = get_unaligned_be32(&buf[offset]); - offset += 4; - - /* - * The max number of config in firmware greater than 4 pieces of - * tas2781s is different from the one lower than 4 pieces of - * tas2781s. - */ - max_confs = TASDEVICE_MAXCONFIG_NUM_KERNEL; - if (tas_fmw->nr_configurations == 0 || - tas_fmw->nr_configurations > max_confs) { - dev_err(tas_priv->dev, "%s: Conf is invalid\n", __func__); - kfree(tas_fmw->programs); - return -EINVAL; - } - - if (offset + 4 * max_confs > fmw->size) { - dev_err(tas_priv->dev, "%s: mpConfigurations err\n", __func__); - kfree(tas_fmw->programs); - return -EINVAL; - } - - tas_fmw->configs = kcalloc(tas_fmw->nr_configurations, - sizeof(*tas_fmw->configs), GFP_KERNEL); - if (!tas_fmw->configs) { - kfree(tas_fmw->programs); - return -ENOMEM; - } - - for (i = 0; i < tas_fmw->nr_programs; i++) { - config = &tas_fmw->configs[i]; - config->cfg_size = get_unaligned_be32(&buf[offset]); - offset += 4; - } - - /* Skip the unused configs */ - offset += 4 * (max_confs - tas_fmw->nr_programs); - - return offset; -} - -/* - * In sub-block data, have three type sub-block: - * 1. Single byte write. - * 2. Multi-byte write. - * 3. Delay. - * 4. Bits update. - * This function perform single byte write to device. - */ -static int tasdevice_single_byte_wr(void *context, int dev_idx, - unsigned char *data, int sublocksize) -{ - struct tasdevice_priv *tas_priv = context; - unsigned short len = get_unaligned_be16(&data[2]); - int i, subblk_offset, rc; - - subblk_offset = 4; - if (subblk_offset + 4 * len > sublocksize) { - dev_err(tas_priv->dev, "process_block: Out of boundary\n"); - return 0; - } - - for (i = 0; i < len; i++) { - if (dev_idx == (tas_priv->index + 1) || dev_idx == 0) { - rc = tasdevice_spi_dev_write(tas_priv, - TASDEVICE_REG(data[subblk_offset], - data[subblk_offset + 1], - data[subblk_offset + 2]), - data[subblk_offset + 3]); - if (rc < 0) { - dev_err(tas_priv->dev, - "process_block: single write error\n"); - subblk_offset |= OFFSET_ERROR_BIT; - } - } - subblk_offset += 4; - } - - return subblk_offset; -} - -/* - * In sub-block data, have three type sub-block: - * 1. Single byte write. - * 2. Multi-byte write. - * 3. Delay. - * 4. Bits update. - * This function perform multi-write to device. - */ -static int tasdevice_burst_wr(void *context, int dev_idx, unsigned char *data, - int sublocksize) -{ - struct tasdevice_priv *tas_priv = context; - unsigned short len = get_unaligned_be16(&data[2]); - int subblk_offset, rc; - - subblk_offset = 4; - if (subblk_offset + 4 + len > sublocksize) { - dev_err(tas_priv->dev, "%s: BST Out of boundary\n", __func__); - subblk_offset |= OFFSET_ERROR_BIT; - } - if (len % 4) { - dev_err(tas_priv->dev, "%s:Bst-len(%u)not div by 4\n", - __func__, len); - subblk_offset |= OFFSET_ERROR_BIT; - } - - if (dev_idx == (tas_priv->index + 1) || dev_idx == 0) { - rc = tasdevice_spi_dev_bulk_write(tas_priv, - TASDEVICE_REG(data[subblk_offset], - data[subblk_offset + 1], - data[subblk_offset + 2]), - &data[subblk_offset + 4], len); - if (rc < 0) { - dev_err(tas_priv->dev, "%s: bulk_write error = %d\n", - __func__, rc); - subblk_offset |= OFFSET_ERROR_BIT; - } - } - subblk_offset += (len + 4); - - return subblk_offset; -} - -/* Just delay for ms.*/ -static int tasdevice_delay(void *context, int dev_idx, unsigned char *data, - int sublocksize) -{ - struct tasdevice_priv *tas_priv = context; - unsigned int sleep_time, subblk_offset = 2; - - if (subblk_offset + 2 > sublocksize) { - dev_err(tas_priv->dev, "%s: delay Out of boundary\n", - __func__); - subblk_offset |= OFFSET_ERROR_BIT; - } - if (dev_idx == (tas_priv->index + 1) || dev_idx == 0) { - sleep_time = get_unaligned_be16(&data[2]) * 1000; - fsleep(sleep_time); - } - subblk_offset += 2; - - return subblk_offset; -} - -/* - * In sub-block data, have three type sub-block: - * 1. Single byte write. - * 2. Multi-byte write. - * 3. Delay. - * 4. Bits update. - * This function perform bits update. - */ -static int tasdevice_field_wr(void *context, int dev_idx, unsigned char *data, - int sublocksize) -{ - struct tasdevice_priv *tas_priv = context; - int rc, subblk_offset = 2; - - if (subblk_offset + 6 > sublocksize) { - dev_err(tas_priv->dev, "%s: bit write Out of boundary\n", - __func__); - subblk_offset |= OFFSET_ERROR_BIT; - } - if (dev_idx == (tas_priv->index + 1) || dev_idx == 0) { - rc = tasdevice_spi_dev_update_bits(tas_priv, - TASDEVICE_REG(data[subblk_offset + 2], - data[subblk_offset + 3], - data[subblk_offset + 4]), - data[subblk_offset + 1], - data[subblk_offset + 5]); - if (rc < 0) { - dev_err(tas_priv->dev, "%s: update_bits error = %d\n", - __func__, rc); - subblk_offset |= OFFSET_ERROR_BIT; - } - } - subblk_offset += 6; - - return subblk_offset; -} - -/* Data block process function. */ -static int tasdevice_process_block(void *context, unsigned char *data, - unsigned char dev_idx, int sublocksize) -{ - struct tasdevice_priv *tas_priv = context; - int blktyp = dev_idx & 0xC0, subblk_offset; - unsigned char subblk_typ = data[1]; - - switch (subblk_typ) { - case TASDEVICE_CMD_SING_W: - subblk_offset = tasdevice_single_byte_wr(tas_priv, - dev_idx & 0x3f, data, sublocksize); - break; - case TASDEVICE_CMD_BURST: - subblk_offset = tasdevice_burst_wr(tas_priv, - dev_idx & 0x3f, data, sublocksize); - break; - case TASDEVICE_CMD_DELAY: - subblk_offset = tasdevice_delay(tas_priv, - dev_idx & 0x3f, data, sublocksize); - break; - case TASDEVICE_CMD_FIELD_W: - subblk_offset = tasdevice_field_wr(tas_priv, - dev_idx & 0x3f, data, sublocksize); - break; - default: - subblk_offset = 2; - break; - } - if (((subblk_offset & OFFSET_ERROR_BIT) != 0) && blktyp != 0) { - if (blktyp == 0x80) { - tas_priv->cur_prog = -1; - tas_priv->cur_conf = -1; - } else - tas_priv->cur_conf = -1; - } - subblk_offset &= ~OFFSET_ERROR_BIT; - - return subblk_offset; -} - -/* - * Device support different configurations for different scene, - * this function was used for choose different config. - */ -void tasdevice_spi_select_cfg_blk(void *pContext, int conf_no, - unsigned char block_type) -{ - struct tasdevice_priv *tas_priv = pContext; - struct tasdevice_rca *rca = &tas_priv->rcabin; - struct tasdevice_config_info **cfg_info = rca->cfg_info; - struct tasdev_blk_data **blk_data; - unsigned int j, k; - - if (conf_no >= rca->ncfgs || conf_no < 0 || !cfg_info) { - dev_err(tas_priv->dev, "conf_no should be not more than %u\n", - rca->ncfgs); - return; - } - blk_data = cfg_info[conf_no]->blk_data; - - for (j = 0; j < cfg_info[conf_no]->real_nblocks; j++) { - unsigned int length = 0, rc = 0; - - if (block_type > 5 || block_type < 2) { - dev_err(tas_priv->dev, - "block_type should be in range from 2 to 5\n"); - break; - } - if (block_type != blk_data[j]->block_type) - continue; - - for (k = 0; k < blk_data[j]->n_subblks; k++) { - tas_priv->is_loading = true; - - rc = tasdevice_process_block(tas_priv, - blk_data[j]->regdata + length, - blk_data[j]->dev_idx, - blk_data[j]->block_size - length); - length += rc; - if (blk_data[j]->block_size < length) { - dev_err(tas_priv->dev, - "%s: %u %u out of boundary\n", - __func__, length, - blk_data[j]->block_size); - break; - } - } - if (length != blk_data[j]->block_size) - dev_err(tas_priv->dev, "%s: %u %u size is not same\n", - __func__, length, blk_data[j]->block_size); - } -} - -/* Block process function. */ -static int tasdevice_load_block_kernel( - struct tasdevice_priv *tasdevice, struct tasdev_blk *block) -{ - const unsigned int blk_size = block->blk_size; - unsigned char *data = block->data; - unsigned int i, length; - - for (i = 0, length = 0; i < block->nr_subblocks; i++) { - int rc = tasdevice_process_block(tasdevice, data + length, - block->dev_idx, blk_size - length); - if (rc < 0) { - dev_err(tasdevice->dev, - "%s: %u %u sublock write error\n", - __func__, length, blk_size); - return rc; - } - length += rc; - if (blk_size < length) { - dev_err(tasdevice->dev, "%s: %u %u out of boundary\n", - __func__, length, blk_size); - rc = -ENOMEM; - return rc; - } - } - - return 0; -} - -/* DSP firmware file header parser function. */ -static int fw_parse_variable_hdr(struct tasdevice_priv *tas_priv, - struct tasdevice_dspfw_hdr *fw_hdr, - const struct firmware *fmw, int offset) -{ - const unsigned char *buf = fmw->data; - int len = strlen((char *)&buf[offset]); - - len++; - - if (offset + len + 8 > fmw->size) { - dev_err(tas_priv->dev, "%s: File Size error\n", __func__); - return -EINVAL; - } - - offset += len; - - fw_hdr->device_family = get_unaligned_be32(&buf[offset]); - if (fw_hdr->device_family != 0) { - dev_err(tas_priv->dev, "%s: not TAS device\n", __func__); - return -EINVAL; - } - offset += 4; - - fw_hdr->device = get_unaligned_be32(&buf[offset]); - if (fw_hdr->device >= TASDEVICE_DSP_TAS_MAX_DEVICE || - fw_hdr->device == 6) { - dev_err(tas_priv->dev, "Unsupported dev %d\n", fw_hdr->device); - return -EINVAL; - } - offset += 4; - fw_hdr->ndev = 1; - - return offset; -} - -/* DSP firmware file header parser function for size variabled header. */ -static int fw_parse_variable_header_git(struct tasdevice_priv - *tas_priv, const struct firmware *fmw, int offset) -{ - struct tasdevice_fw *tas_fmw = tas_priv->fmw; - struct tasdevice_dspfw_hdr *fw_hdr = &tas_fmw->fw_hdr; - - offset = fw_parse_variable_hdr(tas_priv, fw_hdr, fmw, offset); - - return offset; -} - -/* DSP firmware file block parser function. */ -static int fw_parse_block_data(struct tasdevice_fw *tas_fmw, - struct tasdev_blk *block, const struct firmware *fmw, int offset) -{ - unsigned char *data = (unsigned char *)fmw->data; - int n; - - if (offset + 8 > fmw->size) { - dev_err(tas_fmw->dev, "%s: Type error\n", __func__); - return -EINVAL; - } - block->type = get_unaligned_be32(&data[offset]); - offset += 4; - - if (tas_fmw->fw_hdr.fixed_hdr.drv_ver >= PPC_DRIVER_CRCCHK) { - if (offset + 8 > fmw->size) { - dev_err(tas_fmw->dev, "PChkSumPresent error\n"); - return -EINVAL; - } - block->is_pchksum_present = data[offset]; - offset++; - - block->pchksum = data[offset]; - offset++; - - block->is_ychksum_present = data[offset]; - offset++; - - block->ychksum = data[offset]; - offset++; - } else { - block->is_pchksum_present = 0; - block->is_ychksum_present = 0; - } - - block->nr_cmds = get_unaligned_be32(&data[offset]); - offset += 4; - - n = block->nr_cmds * 4; - if (offset + n > fmw->size) { - dev_err(tas_fmw->dev, - "%s: File Size(%lu) error offset = %d n = %d\n", - __func__, (unsigned long)fmw->size, offset, n); - return -EINVAL; - } - /* instead of kzalloc+memcpy */ - block->data = kmemdup(&data[offset], n, GFP_KERNEL); - if (!block->data) - return -ENOMEM; - - offset += n; - - return offset; -} - -/* - * When parsing error occurs, all the memory resource will be released - * in the end of tasdevice_rca_ready. - */ -static int fw_parse_data(struct tasdevice_fw *tas_fmw, - struct tasdevice_data *img_data, const struct firmware *fmw, - int offset) -{ - const unsigned char *data = (unsigned char *)fmw->data; - struct tasdev_blk *blk; - unsigned int i, n; - - if (offset + 64 > fmw->size) { - dev_err(tas_fmw->dev, "%s: Name error\n", __func__); - return -EINVAL; - } - memcpy(img_data->name, &data[offset], 64); - offset += 64; - - n = strlen((char *)&data[offset]); - n++; - if (offset + n + 2 > fmw->size) { - dev_err(tas_fmw->dev, "%s: Description error\n", __func__); - return -EINVAL; - } - offset += n; - img_data->nr_blk = get_unaligned_be16(&data[offset]); - offset += 2; - - img_data->dev_blks = kcalloc(img_data->nr_blk, - sizeof(*img_data->dev_blks), GFP_KERNEL); - if (!img_data->dev_blks) - return -ENOMEM; - - for (i = 0; i < img_data->nr_blk; i++) { - blk = &img_data->dev_blks[i]; - offset = fw_parse_block_data(tas_fmw, blk, fmw, offset); - if (offset < 0) - return -EINVAL; - } - - return offset; -} - -/* - * When parsing error occurs, all the memory resource will be released - * in the end of tasdevice_rca_ready. - */ -static int fw_parse_program_data(struct tasdevice_priv *tas_priv, - struct tasdevice_fw *tas_fmw, const struct firmware *fmw, int offset) -{ - unsigned char *buf = (unsigned char *)fmw->data; - struct tasdevice_prog *program; - int i; - - if (offset + 2 > fmw->size) { - dev_err(tas_priv->dev, "%s: File Size error\n", __func__); - return -EINVAL; - } - tas_fmw->nr_programs = get_unaligned_be16(&buf[offset]); - offset += 2; - - if (tas_fmw->nr_programs == 0) { - /* Not error in calibration Data file, return directly */ - dev_dbg(tas_priv->dev, "%s: No Programs data, maybe calbin\n", - __func__); - return offset; - } - - tas_fmw->programs = - kcalloc(tas_fmw->nr_programs, sizeof(*tas_fmw->programs), - GFP_KERNEL); - if (!tas_fmw->programs) - return -ENOMEM; - - for (i = 0; i < tas_fmw->nr_programs; i++) { - int n = 0; - - program = &tas_fmw->programs[i]; - if (offset + 64 > fmw->size) { - dev_err(tas_priv->dev, "%s: mpName error\n", __func__); - return -EINVAL; - } - offset += 64; - - n = strlen((char *)&buf[offset]); - /* skip '\0' and 5 unused bytes */ - n += 6; - if (offset + n > fmw->size) { - dev_err(tas_priv->dev, "Description err\n"); - return -EINVAL; - } - - offset += n; - - offset = fw_parse_data(tas_fmw, &program->dev_data, fmw, - offset); - if (offset < 0) - return offset; - } - - return offset; -} - -/* - * When parsing error occurs, all the memory resource will be released - * in the end of tasdevice_rca_ready. - */ -static int fw_parse_configuration_data(struct tasdevice_priv *tas_priv, - struct tasdevice_fw *tas_fmw, const struct firmware *fmw, int offset) -{ - unsigned char *data = (unsigned char *)fmw->data; - struct tasdevice_config *config; - unsigned int i, n; - - if (offset + 2 > fmw->size) { - dev_err(tas_priv->dev, "%s: File Size error\n", __func__); - return -EINVAL; - } - tas_fmw->nr_configurations = get_unaligned_be16(&data[offset]); - offset += 2; - - if (tas_fmw->nr_configurations == 0) { - dev_err(tas_priv->dev, "%s: Conf is zero\n", __func__); - /* Not error for calibration Data file, return directly */ - return offset; - } - tas_fmw->configs = kcalloc(tas_fmw->nr_configurations, - sizeof(*tas_fmw->configs), GFP_KERNEL); - if (!tas_fmw->configs) - return -ENOMEM; - for (i = 0; i < tas_fmw->nr_configurations; i++) { - config = &tas_fmw->configs[i]; - if (offset + 64 > fmw->size) { - dev_err(tas_priv->dev, "File Size err\n"); - return -EINVAL; - } - memcpy(config->name, &data[offset], 64); - offset += 64; - - n = strlen((char *)&data[offset]); - n += 15; - if (offset + n > fmw->size) { - dev_err(tas_priv->dev, "Description err\n"); - return -EINVAL; - } - offset += n; - offset = fw_parse_data(tas_fmw, &config->dev_data, - fmw, offset); - if (offset < 0) - break; - } - - return offset; -} - -/* yram5 page check. */ -static bool check_inpage_yram_rg(struct tas_crc *cd, - unsigned char reg, unsigned char len) -{ - bool in = false; - - if (reg <= TAS2781_YRAM5_END_REG && - reg >= TAS2781_YRAM5_START_REG) { - if (reg + len > TAS2781_YRAM5_END_REG) - cd->len = TAS2781_YRAM5_END_REG - reg + 1; - else - cd->len = len; - cd->offset = reg; - in = true; - } else if (reg < TAS2781_YRAM5_START_REG) { - if (reg + len > TAS2781_YRAM5_START_REG) { - cd->offset = TAS2781_YRAM5_START_REG; - cd->len = len - TAS2781_YRAM5_START_REG + reg; - in = true; - } - } - - return in; -} - -/* DSP firmware yram block check. */ -static bool check_inpage_yram_bk1(struct tas_crc *cd, - unsigned char page, unsigned char reg, unsigned char len) -{ - bool in = false; - - if (page == TAS2781_YRAM1_PAGE) { - if (reg >= TAS2781_YRAM1_START_REG) { - cd->offset = reg; - cd->len = len; - in = true; - } else if (reg + len > TAS2781_YRAM1_START_REG) { - cd->offset = TAS2781_YRAM1_START_REG; - cd->len = len - TAS2781_YRAM1_START_REG + reg; - in = true; - } - } else if (page == TAS2781_YRAM3_PAGE) { - in = check_inpage_yram_rg(cd, reg, len); - } - - return in; -} - -/* - * Return Code: - * true -- the registers are in the inpage yram - * false -- the registers are NOT in the inpage yram - */ -static bool check_inpage_yram(struct tas_crc *cd, unsigned char book, - unsigned char page, unsigned char reg, unsigned char len) -{ - bool in = false; - - if (book == TAS2781_YRAM_BOOK1) - in = check_inpage_yram_bk1(cd, page, reg, len); - else if (book == TAS2781_YRAM_BOOK2 && page == TAS2781_YRAM5_PAGE) - in = check_inpage_yram_rg(cd, reg, len); - - return in; -} - -/* yram4 page check. */ -static bool check_inblock_yram_bk(struct tas_crc *cd, - unsigned char page, unsigned char reg, unsigned char len) -{ - bool in = false; - - if ((page >= TAS2781_YRAM4_START_PAGE && - page <= TAS2781_YRAM4_END_PAGE) || - (page >= TAS2781_YRAM2_START_PAGE && - page <= TAS2781_YRAM2_END_PAGE)) { - if (reg <= TAS2781_YRAM2_END_REG && - reg >= TAS2781_YRAM2_START_REG) { - cd->offset = reg; - cd->len = len; - in = true; - } else if (reg < TAS2781_YRAM2_START_REG) { - if (reg + len - 1 >= TAS2781_YRAM2_START_REG) { - cd->offset = TAS2781_YRAM2_START_REG; - cd->len = reg + len - TAS2781_YRAM2_START_REG; - in = true; - } - } - } - - return in; -} - -/* - * Return Code: - * true -- the registers are in the inblock yram - * false -- the registers are NOT in the inblock yram - */ -static bool check_inblock_yram(struct tas_crc *cd, unsigned char book, - unsigned char page, unsigned char reg, unsigned char len) -{ - bool in = false; - - if (book == TAS2781_YRAM_BOOK1 || book == TAS2781_YRAM_BOOK2) - in = check_inblock_yram_bk(cd, page, reg, len); - - return in; -} - -/* yram page check. */ -static bool check_yram(struct tas_crc *cd, unsigned char book, - unsigned char page, unsigned char reg, unsigned char len) -{ - bool in; - - in = check_inpage_yram(cd, book, page, reg, len); - if (!in) - in = check_inblock_yram(cd, book, page, reg, len); - - return in; -} - -/* Checksum for data block. */ -static int tasdev_multibytes_chksum(struct tasdevice_priv *tasdevice, - unsigned char book, unsigned char page, - unsigned char reg, unsigned int len) -{ - struct tas_crc crc_data; - unsigned char crc_chksum = 0; - unsigned char nBuf1[128]; - int ret = 0, i; - bool in; - - if ((reg + len - 1) > 127) { - ret = -EINVAL; - dev_err(tasdevice->dev, "firmware error\n"); - goto end; - } - - if ((book == TASDEVICE_BOOK_ID(TAS2781_SA_COEFF_SWAP_REG)) && - (page == TASDEVICE_PAGE_ID(TAS2781_SA_COEFF_SWAP_REG)) && - (reg == TASDEVICE_REG_ID(TAS2781_SA_COEFF_SWAP_REG)) && - (len == 4)) { - /* DSP swap command, pass */ - ret = 0; - goto end; - } - - in = check_yram(&crc_data, book, page, reg, len); - if (!in) - goto end; - - if (len == 1) { - dev_err(tasdevice->dev, "firmware error\n"); - ret = -EINVAL; - goto end; - } - - ret = tasdevice_spi_dev_bulk_read(tasdevice, - TASDEVICE_REG(book, page, crc_data.offset), - nBuf1, crc_data.len); - if (ret < 0) - goto end; - - for (i = 0; i < crc_data.len; i++) { - if ((book == TASDEVICE_BOOK_ID(TAS2781_SA_COEFF_SWAP_REG)) && - (page == TASDEVICE_PAGE_ID(TAS2781_SA_COEFF_SWAP_REG)) && - ((i + crc_data.offset) >= - TASDEVICE_REG_ID(TAS2781_SA_COEFF_SWAP_REG)) && - ((i + crc_data.offset) <= - (TASDEVICE_REG_ID(TAS2781_SA_COEFF_SWAP_REG) + 4))) - /* DSP swap command, bypass */ - continue; - else - crc_chksum += crc8(tasdevice->crc8_lkp_tbl, &nBuf1[i], - 1, 0); - } - - ret = crc_chksum; - -end: - return ret; -} - -/* Checksum for single register. */ -static int do_singlereg_checksum(struct tasdevice_priv *tasdevice, - unsigned char book, unsigned char page, - unsigned char reg, unsigned char val) -{ - struct tas_crc crc_data; - unsigned int nData1; - int ret = 0; - bool in; - - /* DSP swap command, pass */ - if ((book == TASDEVICE_BOOK_ID(TAS2781_SA_COEFF_SWAP_REG)) && - (page == TASDEVICE_PAGE_ID(TAS2781_SA_COEFF_SWAP_REG)) && - (reg >= TASDEVICE_REG_ID(TAS2781_SA_COEFF_SWAP_REG)) && - (reg <= (TASDEVICE_REG_ID(TAS2781_SA_COEFF_SWAP_REG) + 4))) - return 0; - - in = check_yram(&crc_data, book, page, reg, 1); - if (!in) - return 0; - ret = tasdevice_spi_dev_read(tasdevice, - TASDEVICE_REG(book, page, reg), &nData1); - if (ret < 0) - return ret; - - if (nData1 != val) { - dev_err(tasdevice->dev, - "B[0x%x]P[0x%x]R[0x%x] W[0x%x], R[0x%x]\n", - book, page, reg, val, nData1); - tasdevice->err_code |= ERROR_YRAM_CRCCHK; - return -EAGAIN; - } - - ret = crc8(tasdevice->crc8_lkp_tbl, &val, 1, 0); - - return ret; -} - -/* Block type check. */ -static void set_err_prg_cfg(unsigned int type, struct tasdevice_priv *p) -{ - if ((type == MAIN_ALL_DEVICES) || (type == MAIN_DEVICE_A) || - (type == MAIN_DEVICE_B) || (type == MAIN_DEVICE_C) || - (type == MAIN_DEVICE_D)) - p->cur_prog = -1; - else - p->cur_conf = -1; -} - -/* Checksum for data bytes. */ -static int tasdev_bytes_chksum(struct tasdevice_priv *tas_priv, - struct tasdev_blk *block, unsigned char book, - unsigned char page, unsigned char reg, unsigned int len, - unsigned char val, unsigned char *crc_chksum) -{ - int ret; - - if (len > 1) - ret = tasdev_multibytes_chksum(tas_priv, book, page, reg, - len); - else - ret = do_singlereg_checksum(tas_priv, book, page, reg, val); - - if (ret > 0) { - *crc_chksum += ret; - goto end; - } - - if (ret != -EAGAIN) - goto end; - - block->nr_retry--; - if (block->nr_retry > 0) - goto end; - - set_err_prg_cfg(block->type, tas_priv); - -end: - return ret; -} - -/* Multi-data byte write. */ -static int tasdev_multibytes_wr(struct tasdevice_priv *tas_priv, - struct tasdev_blk *block, unsigned char book, - unsigned char page, unsigned char reg, unsigned char *data, - unsigned int len, unsigned int *nr_cmds, - unsigned char *crc_chksum) -{ - int ret; - - if (len > 1) { - ret = tasdevice_spi_dev_bulk_write(tas_priv, - TASDEVICE_REG(book, page, reg), data + 3, len); - if (ret < 0) - return ret; - if (block->is_ychksum_present) - ret = tasdev_bytes_chksum(tas_priv, block, - book, page, reg, len, 0, crc_chksum); - } else { - ret = tasdevice_spi_dev_write(tas_priv, - TASDEVICE_REG(book, page, reg), data[3]); - if (ret < 0) - return ret; - if (block->is_ychksum_present) - ret = tasdev_bytes_chksum(tas_priv, block, book, - page, reg, 1, data[3], crc_chksum); - } - - if (!block->is_ychksum_present || ret >= 0) { - *nr_cmds += 1; - if (len >= 2) - *nr_cmds += ((len - 2) / 4) + 1; - } - - return ret; -} - -/* Checksum for block. */ -static int tasdev_block_chksum(struct tasdevice_priv *tas_priv, - struct tasdev_blk *block) -{ - unsigned int nr_value; - int ret; - - ret = tasdevice_spi_dev_read(tas_priv, TASDEVICE_CHECKSUM, &nr_value); - if (ret < 0) { - dev_err(tas_priv->dev, "%s: read error %d.\n", __func__, ret); - set_err_prg_cfg(block->type, tas_priv); - return ret; - } - - if ((nr_value & 0xff) != block->pchksum) { - dev_err(tas_priv->dev, "%s: PChkSum err %d ", __func__, ret); - dev_err(tas_priv->dev, "PChkSum = 0x%x, Reg = 0x%x\n", - block->pchksum, (nr_value & 0xff)); - tas_priv->err_code |= ERROR_PRAM_CRCCHK; - ret = -EAGAIN; - block->nr_retry--; - - if (block->nr_retry <= 0) - set_err_prg_cfg(block->type, tas_priv); - } else { - tas_priv->err_code &= ~ERROR_PRAM_CRCCHK; - } - - return ret; -} - -/* Firmware block load function. */ -static int tasdev_load_blk(struct tasdevice_priv *tas_priv, - struct tasdev_blk *block) -{ - unsigned int sleep_time, len, nr_cmds; - unsigned char offset, book, page, val; - unsigned char *data = block->data; - unsigned char crc_chksum = 0; - int ret = 0; - - while (block->nr_retry > 0) { - if (block->is_pchksum_present) { - ret = tasdevice_spi_dev_write(tas_priv, - TASDEVICE_CHECKSUM, 0); - if (ret < 0) - break; - } - - if (block->is_ychksum_present) - crc_chksum = 0; - - nr_cmds = 0; - - while (nr_cmds < block->nr_cmds) { - data = block->data + nr_cmds * 4; - - book = data[0]; - page = data[1]; - offset = data[2]; - val = data[3]; - - nr_cmds++; - /* Single byte write */ - if (offset <= 0x7F) { - ret = tasdevice_spi_dev_write(tas_priv, - TASDEVICE_REG(book, page, offset), - val); - if (ret < 0) - break; - if (block->is_ychksum_present) { - ret = tasdev_bytes_chksum(tas_priv, - block, book, page, offset, - 1, val, &crc_chksum); - if (ret < 0) - break; - } - continue; - } - /* sleep command */ - if (offset == 0x81) { - /* book -- data[0] page -- data[1] */ - sleep_time = ((book << 8) + page)*1000; - fsleep(sleep_time); - continue; - } - /* Multiple bytes write */ - if (offset == 0x85) { - data += 4; - len = (book << 8) + page; - book = data[0]; - page = data[1]; - offset = data[2]; - ret = tasdev_multibytes_wr(tas_priv, - block, book, page, offset, data, - len, &nr_cmds, &crc_chksum); - if (ret < 0) - break; - } - } - if (ret == -EAGAIN) { - if (block->nr_retry > 0) - continue; - } else if (ret < 0) { - /* err in current device, skip it */ - break; - } - - if (block->is_pchksum_present) { - ret = tasdev_block_chksum(tas_priv, block); - if (ret == -EAGAIN) { - if (block->nr_retry > 0) - continue; - } else if (ret < 0) { - /* err in current device, skip it */ - break; - } - } - - if (block->is_ychksum_present) { - /* TBD, open it when FW ready */ - dev_err(tas_priv->dev, - "Blk YChkSum: FW = 0x%x, YCRC = 0x%x\n", - block->ychksum, crc_chksum); - - tas_priv->err_code &= - ~ERROR_YRAM_CRCCHK; - ret = 0; - } - /* skip current blk */ - break; - } - - return ret; -} - -/* Firmware block load function. */ -static int tasdevice_load_block(struct tasdevice_priv *tas_priv, - struct tasdev_blk *block) -{ - int ret = 0; - - block->nr_retry = 6; - if (tas_priv->is_loading == false) - return 0; - ret = tasdev_load_blk(tas_priv, block); - if (ret < 0) - dev_err(tas_priv->dev, "Blk (%d) load error\n", block->type); - - return ret; -} - -/* - * Select firmware binary parser & load callback functions by ppc3 version - * and firmware binary version. - */ -static int dspfw_default_callback(struct tasdevice_priv *tas_priv, - unsigned int drv_ver, unsigned int ppcver) -{ - int rc = 0; - - if (drv_ver == 0x100) { - if (ppcver >= PPC3_VERSION_BASE) { - tas_priv->fw_parse_variable_header = - fw_parse_variable_header_kernel; - tas_priv->fw_parse_program_data = - fw_parse_program_data_kernel; - tas_priv->fw_parse_configuration_data = - fw_parse_configuration_data_kernel; - tas_priv->tasdevice_load_block = - tasdevice_load_block_kernel; - } else if (ppcver == 0x00) { - tas_priv->fw_parse_variable_header = - fw_parse_variable_header_git; - tas_priv->fw_parse_program_data = - fw_parse_program_data; - tas_priv->fw_parse_configuration_data = - fw_parse_configuration_data; - tas_priv->tasdevice_load_block = - tasdevice_load_block; - } else { - dev_err(tas_priv->dev, - "Wrong PPCVer :0x%08x\n", ppcver); - rc = -EINVAL; - } - } else { - dev_err(tas_priv->dev, "Wrong DrvVer : 0x%02x\n", drv_ver); - rc = -EINVAL; - } - - return rc; -} - -/* DSP firmware binary file header parser function. */ -static int fw_parse_header(struct tasdevice_priv *tas_priv, - struct tasdevice_fw *tas_fmw, const struct firmware *fmw, int offset) -{ - struct tasdevice_dspfw_hdr *fw_hdr = &tas_fmw->fw_hdr; - struct tasdevice_fw_fixed_hdr *fw_fixed_hdr = &fw_hdr->fixed_hdr; - static const unsigned char magic_number[] = {0x35, 0x35, 0x35, 0x32, }; - const unsigned char *buf = (unsigned char *)fmw->data; - - if (offset + 92 > fmw->size) { - dev_err(tas_priv->dev, "%s: File Size error\n", __func__); - offset = -EINVAL; - goto out; - } - if (memcmp(&buf[offset], magic_number, 4)) { - dev_err(tas_priv->dev, "%s: Magic num NOT match\n", __func__); - offset = -EINVAL; - goto out; - } - offset += 4; - - /* - * Convert data[offset], data[offset + 1], data[offset + 2] and - * data[offset + 3] into host - */ - fw_fixed_hdr->fwsize = get_unaligned_be32(&buf[offset]); - offset += 4; - if (fw_fixed_hdr->fwsize != fmw->size) { - dev_err(tas_priv->dev, "File size not match, %lu %u", - (unsigned long)fmw->size, fw_fixed_hdr->fwsize); - offset = -EINVAL; - goto out; - } - offset += 4; - fw_fixed_hdr->ppcver = get_unaligned_be32(&buf[offset]); - offset += 8; - fw_fixed_hdr->drv_ver = get_unaligned_be32(&buf[offset]); - offset += 72; - -out: - return offset; -} - -/* DSP firmware binary file parser function. */ -static int tasdevice_dspfw_ready(const struct firmware *fmw, void *context) -{ - struct tasdevice_priv *tas_priv = context; - struct tasdevice_fw_fixed_hdr *fw_fixed_hdr; - struct tasdevice_fw *tas_fmw; - int offset = 0, ret = 0; - - if (!fmw || !fmw->data) { - dev_err(tas_priv->dev, "%s: Failed to read firmware %s\n", - __func__, tas_priv->coef_binaryname); - return -EINVAL; - } - - tas_priv->fmw = kzalloc(sizeof(*tas_priv->fmw), GFP_KERNEL); - if (!tas_priv->fmw) - return -ENOMEM; - tas_fmw = tas_priv->fmw; - tas_fmw->dev = tas_priv->dev; - offset = fw_parse_header(tas_priv, tas_fmw, fmw, offset); - - if (offset == -EINVAL) - return -EINVAL; - - fw_fixed_hdr = &tas_fmw->fw_hdr.fixed_hdr; - /* Support different versions of firmware */ - switch (fw_fixed_hdr->drv_ver) { - case 0x301: - case 0x302: - case 0x502: - case 0x503: - tas_priv->fw_parse_variable_header = - fw_parse_variable_header_kernel; - tas_priv->fw_parse_program_data = - fw_parse_program_data_kernel; - tas_priv->fw_parse_configuration_data = - fw_parse_configuration_data_kernel; - tas_priv->tasdevice_load_block = - tasdevice_load_block_kernel; - break; - case 0x202: - case 0x400: - tas_priv->fw_parse_variable_header = - fw_parse_variable_header_git; - tas_priv->fw_parse_program_data = - fw_parse_program_data; - tas_priv->fw_parse_configuration_data = - fw_parse_configuration_data; - tas_priv->tasdevice_load_block = - tasdevice_load_block; - break; - default: - ret = dspfw_default_callback(tas_priv, - fw_fixed_hdr->drv_ver, fw_fixed_hdr->ppcver); - if (ret) - return ret; - break; - } - - offset = tas_priv->fw_parse_variable_header(tas_priv, fmw, offset); - if (offset < 0) - return offset; - - offset = tas_priv->fw_parse_program_data(tas_priv, tas_fmw, fmw, - offset); - if (offset < 0) - return offset; - - offset = tas_priv->fw_parse_configuration_data(tas_priv, - tas_fmw, fmw, offset); - if (offset < 0) - ret = offset; - - return ret; -} - -/* DSP firmware binary file parser function. */ -int tasdevice_spi_dsp_parser(void *context) -{ - struct tasdevice_priv *tas_priv = context; - const struct firmware *fw_entry; - int ret; - - ret = request_firmware(&fw_entry, tas_priv->coef_binaryname, - tas_priv->dev); - if (ret) { - dev_err(tas_priv->dev, "%s: load %s error\n", __func__, - tas_priv->coef_binaryname); - return ret; - } - - ret = tasdevice_dspfw_ready(fw_entry, tas_priv); - release_firmware(fw_entry); - fw_entry = NULL; - - return ret; -} - -/* DSP firmware program block data remove function. */ -static void tasdev_dsp_prog_blk_remove(struct tasdevice_prog *prog) -{ - struct tasdevice_data *tas_dt; - struct tasdev_blk *blk; - unsigned int i; - - if (!prog) - return; - - tas_dt = &prog->dev_data; - - if (!tas_dt->dev_blks) - return; - - for (i = 0; i < tas_dt->nr_blk; i++) { - blk = &tas_dt->dev_blks[i]; - kfree(blk->data); - } - kfree(tas_dt->dev_blks); -} - -/* DSP firmware program block data remove function. */ -static void tasdev_dsp_prog_remove(struct tasdevice_prog *prog, - unsigned short nr) -{ - int i; - - for (i = 0; i < nr; i++) - tasdev_dsp_prog_blk_remove(&prog[i]); - kfree(prog); -} - -/* DSP firmware config block data remove function. */ -static void tasdev_dsp_cfg_blk_remove(struct tasdevice_config *cfg) -{ - struct tasdevice_data *tas_dt; - struct tasdev_blk *blk; - unsigned int i; - - if (cfg) { - tas_dt = &cfg->dev_data; - - if (!tas_dt->dev_blks) - return; - - for (i = 0; i < tas_dt->nr_blk; i++) { - blk = &tas_dt->dev_blks[i]; - kfree(blk->data); - } - kfree(tas_dt->dev_blks); - } -} - -/* DSP firmware config remove function. */ -static void tasdev_dsp_cfg_remove(struct tasdevice_config *config, - unsigned short nr) -{ - int i; - - for (i = 0; i < nr; i++) - tasdev_dsp_cfg_blk_remove(&config[i]); - kfree(config); -} - -/* DSP firmware remove function. */ -void tasdevice_spi_dsp_remove(void *context) -{ - struct tasdevice_priv *tas_dev = context; - - if (!tas_dev->fmw) - return; - - if (tas_dev->fmw->programs) - tasdev_dsp_prog_remove(tas_dev->fmw->programs, - tas_dev->fmw->nr_programs); - if (tas_dev->fmw->configs) - tasdev_dsp_cfg_remove(tas_dev->fmw->configs, - tas_dev->fmw->nr_configurations); - kfree(tas_dev->fmw); - tas_dev->fmw = NULL; -} - -/* DSP firmware calibration data remove function. */ -static void tas2781_clear_calfirmware(struct tasdevice_fw *tas_fmw) -{ - struct tasdevice_calibration *calibration; - struct tasdev_blk *block; - unsigned int blks; - int i; - - if (!tas_fmw->calibrations) - goto out; - - for (i = 0; i < tas_fmw->nr_calibrations; i++) { - calibration = &tas_fmw->calibrations[i]; - if (!calibration) - continue; - - if (!calibration->dev_data.dev_blks) - continue; - - for (blks = 0; blks < calibration->dev_data.nr_blk; blks++) { - block = &calibration->dev_data.dev_blks[blks]; - if (!block) - continue; - kfree(block->data); - } - kfree(calibration->dev_data.dev_blks); - } - kfree(tas_fmw->calibrations); -out: - kfree(tas_fmw); -} - -/* Calibration data from firmware remove function. */ -void tasdevice_spi_calbin_remove(void *context) -{ - struct tasdevice_priv *tas_priv = context; - - if (tas_priv->cali_data_fmw) { - tas2781_clear_calfirmware(tas_priv->cali_data_fmw); - tas_priv->cali_data_fmw = NULL; - } -} - -/* Configuration remove function. */ -void tasdevice_spi_config_info_remove(void *context) -{ - struct tasdevice_priv *tas_priv = context; - struct tasdevice_rca *rca = &tas_priv->rcabin; - struct tasdevice_config_info **ci = rca->cfg_info; - unsigned int i, j; - - if (!ci) - return; - for (i = 0; i < rca->ncfgs; i++) { - if (!ci[i]) - continue; - if (ci[i]->blk_data) { - for (j = 0; j < ci[i]->real_nblocks; j++) { - if (!ci[i]->blk_data[j]) - continue; - kfree(ci[i]->blk_data[j]->regdata); - kfree(ci[i]->blk_data[j]); - } - kfree(ci[i]->blk_data); - } - kfree(ci[i]); - } - kfree(ci); -} - -/* DSP firmware program block data load function. */ -static int tasdevice_load_data(struct tasdevice_priv *tas_priv, - struct tasdevice_data *dev_data) -{ - struct tasdev_blk *block; - unsigned int i; - int ret = 0; - - for (i = 0; i < dev_data->nr_blk; i++) { - block = &dev_data->dev_blks[i]; - ret = tas_priv->tasdevice_load_block(tas_priv, block); - if (ret < 0) - break; - } - - return ret; -} - -/* DSP firmware program load interface function. */ -int tasdevice_spi_prmg_load(void *context, int prm_no) -{ - struct tasdevice_priv *tas_priv = context; - struct tasdevice_fw *tas_fmw = tas_priv->fmw; - struct tasdevice_prog *program; - struct tasdevice_config *conf; - int ret = 0; - - if (!tas_fmw) { - dev_err(tas_priv->dev, "%s: Firmware is NULL\n", __func__); - return -EINVAL; - } - if (prm_no >= 0 && prm_no <= tas_fmw->nr_programs) { - tas_priv->cur_conf = 0; - tas_priv->is_loading = true; - program = &tas_fmw->programs[prm_no]; - ret = tasdevice_load_data(tas_priv, &program->dev_data); - if (ret < 0) { - dev_err(tas_priv->dev, "Program failed %d.\n", ret); - return ret; - } - tas_priv->cur_prog = prm_no; - - conf = &tas_fmw->configs[tas_priv->cur_conf]; - ret = tasdevice_load_data(tas_priv, &conf->dev_data); - if (ret < 0) - dev_err(tas_priv->dev, "Config failed %d.\n", ret); - } else { - dev_err(tas_priv->dev, - "%s: prm(%d) is not in range of Programs %u\n", - __func__, prm_no, tas_fmw->nr_programs); - return -EINVAL; - } - - return ret; -} - -/* RCABIN configuration switch interface function. */ -void tasdevice_spi_tuning_switch(void *context, int state) -{ - struct tasdevice_priv *tas_priv = context; - int profile_cfg_id = tas_priv->rcabin.profile_cfg_id; - - if (tas_priv->fw_state == TASDEVICE_DSP_FW_FAIL) { - dev_err(tas_priv->dev, "DSP bin file not loaded\n"); - return; - } - - if (state == 0) - tasdevice_spi_select_cfg_blk(tas_priv, profile_cfg_id, - TASDEVICE_BIN_BLK_PRE_POWER_UP); - else - tasdevice_spi_select_cfg_blk(tas_priv, profile_cfg_id, - TASDEVICE_BIN_BLK_PRE_SHUTDOWN); -} diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 40bb7a1d44bc..2e5b03be2f20 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -260,6 +260,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_TAS2770 imply SND_SOC_TAS2780 imply SND_SOC_TAS2781_COMLIB + imply SND_SOC_TAS2781_COMLIB_I2C imply SND_SOC_TAS2781_FMWLIB imply SND_SOC_TAS2781_I2C imply SND_SOC_TAS5086 @@ -1991,20 +1992,23 @@ config SND_SOC_TAS2780 digital input mono Class-D audio power amplifiers. config SND_SOC_TAS2781_COMLIB + tristate + +config SND_SOC_TAS2781_COMLIB_I2C depends on I2C select CRC8 select REGMAP_I2C tristate config SND_SOC_TAS2781_FMWLIB - depends on SND_SOC_TAS2781_COMLIB + select SND_SOC_TAS2781_COMLIB tristate default n config SND_SOC_TAS2781_I2C tristate "Texas Instruments TAS2781 speaker amplifier based on I2C" depends on I2C - select SND_SOC_TAS2781_COMLIB + select SND_SOC_TAS2781_COMLIB_I2C select SND_SOC_TAS2781_FMWLIB help Enable support for Texas Instruments TAS2781 Smart Amplifier diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 10f726066b6c..337d72c435f4 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -305,6 +305,7 @@ snd-soc-tas6424-y := tas6424.o snd-soc-tda7419-y := tda7419.o snd-soc-tas2770-y := tas2770.o snd-soc-tas2781-comlib-y := tas2781-comlib.o +snd-soc-tas2781-comlib-i2c-y := tas2781-comlib-i2c.o snd-soc-tas2781-fmwlib-y := tas2781-fmwlib.o snd-soc-tas2781-i2c-y := tas2781-i2c.o snd-soc-tfa9879-y := tfa9879.o @@ -713,6 +714,7 @@ obj-$(CONFIG_SND_SOC_TAS2562) += snd-soc-tas2562.o obj-$(CONFIG_SND_SOC_TAS2764) += snd-soc-tas2764.o obj-$(CONFIG_SND_SOC_TAS2780) += snd-soc-tas2780.o obj-$(CONFIG_SND_SOC_TAS2781_COMLIB) += snd-soc-tas2781-comlib.o +obj-$(CONFIG_SND_SOC_TAS2781_COMLIB_I2C) += snd-soc-tas2781-comlib-i2c.o obj-$(CONFIG_SND_SOC_TAS2781_FMWLIB) += snd-soc-tas2781-fmwlib.o obj-$(CONFIG_SND_SOC_TAS2781_I2C) += snd-soc-tas2781-i2c.o obj-$(CONFIG_SND_SOC_TAS5086) += snd-soc-tas5086.o diff --git a/sound/soc/codecs/tas2781-comlib-i2c.c b/sound/soc/codecs/tas2781-comlib-i2c.c new file mode 100644 index 000000000000..a75b812618ee --- /dev/null +++ b/sound/soc/codecs/tas2781-comlib-i2c.c @@ -0,0 +1,375 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// TAS2563/TAS2781 Common functions for HDA and ASoC Audio drivers based on I2C +// +// Copyright 2025 Texas Instruments, Inc. +// +// Author: Shenghao Ding <shenghao-ding(a)ti.com> + +#include <linux/crc8.h> +#include <linux/firmware.h> +#include <linux/gpio/consumer.h> +#include <linux/i2c.h> +#include <linux/init.h> +#include <linux/interrupt.h> +#include <linux/module.h> +#include <linux/of.h> +#include <linux/of_irq.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tas2781.h> +#include <sound/tas2781-comlib-i2c.h> + +static const struct regmap_range_cfg tasdevice_ranges[] = { + { + .range_min = 0, + .range_max = 256 * 128, + .selector_reg = TASDEVICE_PAGE_SELECT, + .selector_mask = 0xff, + .selector_shift = 0, + .window_start = 0, + .window_len = 128, + }, +}; + +static const struct regmap_config tasdevice_regmap = { + .reg_bits = 8, + .val_bits = 8, + .cache_type = REGCACHE_NONE, + .ranges = tasdevice_ranges, + .num_ranges = ARRAY_SIZE(tasdevice_ranges), + .max_register = 256 * 128, +}; + +static int tasdevice_change_chn_book(struct tasdevice_priv *tas_priv, + unsigned short chn, int book) +{ + struct i2c_client *client = (struct i2c_client *)tas_priv->client; + int ret = 0; + + if (chn < tas_priv->ndev) { + struct tasdevice *tasdev = &tas_priv->tasdevice[chn]; + struct regmap *map = tas_priv->regmap; + + if (client->addr != tasdev->dev_addr) { + client->addr = tasdev->dev_addr; + /* All tas2781s share the same regmap, clear the page + * inside regmap once switching to another tas2781. + * Register 0 at any pages and any books inside tas2781 + * is the same one for page-switching. + */ + ret = regmap_write(map, TASDEVICE_PAGE_SELECT, 0); + if (ret < 0) { + dev_err(tas_priv->dev, "%s, E=%d channel:%d\n", + __func__, ret, chn); + goto out; + } + } + + if (tasdev->cur_book != book) { + ret = regmap_write(map, TASDEVICE_BOOKCTL_REG, book); + if (ret < 0) { + dev_err(tas_priv->dev, "%s, E=%d\n", + __func__, ret); + goto out; + } + tasdev->cur_book = book; + } + } else { + ret = -EINVAL; + dev_err(tas_priv->dev, "%s, no such channel(%d)\n", __func__, + chn); + } + +out: + return ret; +} + +int tasdev_chn_switch(struct tasdevice_priv *tas_priv, + unsigned short chn) +{ + struct i2c_client *client = (struct i2c_client *)tas_priv->client; + struct tasdevice *tasdev = &tas_priv->tasdevice[chn]; + struct regmap *map = tas_priv->regmap; + int ret; + + if (client->addr != tasdev->dev_addr) { + client->addr = tasdev->dev_addr; + /* All devices share the same regmap, clear the page + * inside regmap once switching to another device. + * Register 0 at any pages and any books inside tas2781 + * is the same one for page-switching. + */ + ret = regmap_write(map, TASDEVICE_PAGE_SELECT, 0); + if (ret < 0) { + dev_err(tas_priv->dev, "%s, E=%d\n", __func__, ret); + return ret; + } + return 1; + } + return 0; +} +EXPORT_SYMBOL_GPL(tasdev_chn_switch); + +int tasdevice_dev_update_bits( + struct tasdevice_priv *tas_priv, unsigned short chn, + unsigned int reg, unsigned int mask, unsigned int value) +{ + int ret = 0; + + if (chn < tas_priv->ndev) { + struct regmap *map = tas_priv->regmap; + + ret = tas_priv->change_chn_book(tas_priv, chn, + TASDEVICE_BOOK_ID(reg)); + if (ret < 0) + goto out; + + ret = regmap_update_bits(map, TASDEVICE_PGRG(reg), + mask, value); + if (ret < 0) + dev_err(tas_priv->dev, "%s, E=%d\n", __func__, ret); + } else { + dev_err(tas_priv->dev, "%s, no such channel(%d)\n", __func__, + chn); + ret = -EINVAL; + } + +out: + return ret; +} +EXPORT_SYMBOL_GPL(tasdevice_dev_update_bits); + +struct tasdevice_priv *tasdevice_kzalloc(struct i2c_client *i2c) +{ + struct tasdevice_priv *tas_priv; + + tas_priv = devm_kzalloc(&i2c->dev, sizeof(*tas_priv), GFP_KERNEL); + if (!tas_priv) + return NULL; + tas_priv->dev = &i2c->dev; + tas_priv->client = (void *)i2c; + + return tas_priv; +} +EXPORT_SYMBOL_GPL(tasdevice_kzalloc); + +int tasdevice_init(struct tasdevice_priv *tas_priv) +{ + int ret = 0; + int i; + + tas_priv->regmap = devm_regmap_init_i2c(tas_priv->client, + &tasdevice_regmap); + if (IS_ERR(tas_priv->regmap)) { + ret = PTR_ERR(tas_priv->regmap); + dev_err(tas_priv->dev, "Failed to allocate register map: %d\n", + ret); + goto out; + } + + tas_priv->cur_prog = -1; + tas_priv->cur_conf = -1; + tas_priv->isspi = false; + + for (i = 0; i < tas_priv->ndev; i++) { + tas_priv->tasdevice[i].cur_book = -1; + tas_priv->tasdevice[i].cur_prog = -1; + tas_priv->tasdevice[i].cur_conf = -1; + } + + tas_priv->update_bits = tasdevice_dev_update_bits; + tas_priv->change_chn_book = tasdevice_change_chn_book; + tas_priv->dev_read = tasdevice_dev_read; + tas_priv->dev_bulk_read = tasdevice_dev_bulk_read; + + mutex_init(&tas_priv->codec_lock); + +out: + return ret; +} +EXPORT_SYMBOL_GPL(tasdevice_init); + +static int tasdevice_clamp(int val, int max, unsigned int invert) +{ + if (val > max) + val = max; + if (invert) + val = max - val; + if (val < 0) + val = 0; + return val; +} + +int tasdevice_amp_putvol(struct tasdevice_priv *tas_priv, + struct snd_ctl_elem_value *ucontrol, struct soc_mixer_control *mc) +{ + unsigned int invert = mc->invert; + unsigned char mask; + int max = mc->max; + int err_cnt = 0; + int val, i, ret; + + mask = (1 << fls(max)) - 1; + mask <<= mc->shift; + val = tasdevice_clamp(ucontrol->value.integer.value[0], max, invert); + for (i = 0; i < tas_priv->ndev; i++) { + ret = tasdevice_dev_update_bits(tas_priv, i, + mc->reg, mask, (unsigned int)(val << mc->shift)); + if (!ret) + continue; + err_cnt++; + dev_err(tas_priv->dev, "set AMP vol error in dev %d\n", i); + } + + /* All the devices set error, return 0 */ + return (err_cnt == tas_priv->ndev) ? 0 : 1; +} +EXPORT_SYMBOL_GPL(tasdevice_amp_putvol); + +int tasdevice_amp_getvol(struct tasdevice_priv *tas_priv, + struct snd_ctl_elem_value *ucontrol, struct soc_mixer_control *mc) +{ + unsigned int invert = mc->invert; + unsigned char mask = 0; + int max = mc->max; + int ret = 0; + int val; + + /* Read the primary device */ + ret = tasdevice_dev_read(tas_priv, 0, mc->reg, &val); + if (ret) { + dev_err(tas_priv->dev, "%s, get AMP vol error\n", __func__); + goto out; + } + + mask = (1 << fls(max)) - 1; + mask <<= mc->shift; + val = (val & mask) >> mc->shift; + val = tasdevice_clamp(val, max, invert); + ucontrol->value.integer.value[0] = val; + +out: + return ret; + +} +EXPORT_SYMBOL_GPL(tasdevice_amp_getvol); + +int tasdevice_digital_getvol(struct tasdevice_priv *tas_priv, + struct snd_ctl_elem_value *ucontrol, struct soc_mixer_control *mc) +{ + unsigned int invert = mc->invert; + int max = mc->max; + int ret, val; + + /* Read the primary device as the whole */ + ret = tasdevice_dev_read(tas_priv, 0, mc->reg, &val); + if (ret) { + dev_err(tas_priv->dev, "%s, get digital vol error\n", + __func__); + goto out; + } + + val = tasdevice_clamp(val, max, invert); + ucontrol->value.integer.value[0] = val; + +out: + return ret; + +} +EXPORT_SYMBOL_GPL(tasdevice_digital_getvol); + +int tasdevice_digital_putvol(struct tasdevice_priv *tas_priv, + struct snd_ctl_elem_value *ucontrol, struct soc_mixer_control *mc) +{ + unsigned int invert = mc->invert; + int max = mc->max; + int err_cnt = 0; + int ret; + int val, i; + + val = tasdevice_clamp(ucontrol->value.integer.value[0], max, invert); + + for (i = 0; i < tas_priv->ndev; i++) { + ret = tasdevice_dev_write(tas_priv, i, mc->reg, + (unsigned int)val); + if (!ret) + continue; + err_cnt++; + dev_err(tas_priv->dev, + "set digital vol err in dev %d\n", i); + } + + /* All the devices set error, return 0 */ + return (err_cnt == tas_priv->ndev) ? 0 : 1; + +} +EXPORT_SYMBOL_GPL(tasdevice_digital_putvol); + +void tasdevice_reset(struct tasdevice_priv *tas_dev) +{ + int ret, i; + + if (tas_dev->reset) { + gpiod_set_value_cansleep(tas_dev->reset, 0); + usleep_range(500, 1000); + gpiod_set_value_cansleep(tas_dev->reset, 1); + } else { + for (i = 0; i < tas_dev->ndev; i++) { + ret = tasdevice_dev_write(tas_dev, i, + TASDEVICE_REG_SWRESET, + TASDEVICE_REG_SWRESET_RESET); + if (ret < 0) + dev_err(tas_dev->dev, + "dev %d swreset fail, %d\n", + i, ret); + } + } + usleep_range(1000, 1050); +} +EXPORT_SYMBOL_GPL(tasdevice_reset); + +int tascodec_init(struct tasdevice_priv *tas_priv, void *codec, + struct module *module, + void (*cont)(const struct firmware *fw, void *context)) +{ + int ret = 0; + + /* Codec Lock Hold to ensure that codec_probe and firmware parsing and + * loading do not simultaneously execute. + */ + mutex_lock(&tas_priv->codec_lock); + + if (tas_priv->name_prefix) + scnprintf(tas_priv->rca_binaryname, 64, "%s-%sRCA%d.bin", + tas_priv->name_prefix, tas_priv->dev_name, + tas_priv->ndev); + else + scnprintf(tas_priv->rca_binaryname, 64, "%sRCA%d.bin", + tas_priv->dev_name, tas_priv->ndev); + crc8_populate_msb(tas_priv->crc8_lkp_tbl, TASDEVICE_CRC8_POLYNOMIAL); + tas_priv->codec = codec; + ret = request_firmware_nowait(module, FW_ACTION_UEVENT, + tas_priv->rca_binaryname, tas_priv->dev, GFP_KERNEL, tas_priv, + cont); + if (ret) + dev_err(tas_priv->dev, "request_firmware_nowait err:0x%08x\n", + ret); + + /* Codec Lock Release*/ + mutex_unlock(&tas_priv->codec_lock); + return ret; +} +EXPORT_SYMBOL_GPL(tascodec_init); + +void tasdevice_remove(struct tasdevice_priv *tas_priv) +{ + mutex_destroy(&tas_priv->codec_lock); +} +EXPORT_SYMBOL_GPL(tasdevice_remove); + +MODULE_DESCRIPTION("TAS2781 common library for I2C"); +MODULE_AUTHOR("Shenghao Ding, TI, <shenghao-ding(a)ti.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tas2781-comlib.c b/sound/soc/codecs/tas2781-comlib.c index 1e0b3aa95749..ad2f65359b15 100644 --- a/sound/soc/codecs/tas2781-comlib.c +++ b/sound/soc/codecs/tas2781-comlib.c @@ -2,14 +2,14 @@ // // TAS2563/TAS2781 Common functions for HDA and ASoC Audio drivers // -// Copyright 2023 - 2024 Texas Instruments, Inc. +// Copyright 2023 - 2025 Texas Instruments, Inc. // // Author: Shenghao Ding <shenghao-ding(a)ti.com> #include <linux/crc8.h> +#include <linux/dev_printk.h> #include <linux/firmware.h> #include <linux/gpio/consumer.h> -#include <linux/i2c.h> #include <linux/init.h> #include <linux/interrupt.h> #include <linux/module.h> @@ -17,67 +17,24 @@ #include <linux/of_irq.h> #include <linux/regmap.h> #include <linux/slab.h> -#include <sound/pcm_params.h> -#include <sound/soc.h> #include <sound/tas2781.h> -#define TASDEVICE_CRC8_POLYNOMIAL 0x4d - -static const struct regmap_range_cfg tasdevice_ranges[] = { - { - .range_min = 0, - .range_max = 256 * 128, - .selector_reg = TASDEVICE_PAGE_SELECT, - .selector_mask = 0xff, - .selector_shift = 0, - .window_start = 0, - .window_len = 128, - }, -}; - -static const struct regmap_config tasdevice_regmap = { - .reg_bits = 8, - .val_bits = 8, - .cache_type = REGCACHE_NONE, - .ranges = tasdevice_ranges, - .num_ranges = ARRAY_SIZE(tasdevice_ranges), - .max_register = 256 * 128, -}; - -static int tasdevice_change_chn_book(struct tasdevice_priv *tas_priv, - unsigned short chn, int book) +int tasdevice_dev_read(struct tasdevice_priv *tas_priv, + unsigned short chn, unsigned int reg, unsigned int *val) { - struct i2c_client *client = (struct i2c_client *)tas_priv->client; int ret = 0; if (chn < tas_priv->ndev) { - struct tasdevice *tasdev = &tas_priv->tasdevice[chn]; struct regmap *map = tas_priv->regmap; - if (client->addr != tasdev->dev_addr) { - client->addr = tasdev->dev_addr; - /* All tas2781s share the same regmap, clear the page - * inside regmap once switching to another tas2781. - * Register 0 at any pages and any books inside tas2781 - * is the same one for page-switching. - */ - ret = regmap_write(map, TASDEVICE_PAGE_SELECT, 0); - if (ret < 0) { - dev_err(tas_priv->dev, "%s, E=%d channel:%d\n", - __func__, ret, chn); - goto out; - } - } + ret = tas_priv->change_chn_book(tas_priv, chn, + TASDEVICE_BOOK_ID(reg)); + if (ret < 0) + goto out; - if (tasdev->cur_book != book) { - ret = regmap_write(map, TASDEVICE_BOOKCTL_REG, book); - if (ret < 0) { - dev_err(tas_priv->dev, "%s, E=%d\n", - __func__, ret); - goto out; - } - tasdev->cur_book = book; - } + ret = regmap_read(map, TASDEVICE_PGRG(reg), val); + if (ret < 0) + dev_err(tas_priv->dev, "%s, E=%d\n", __func__, ret); } else { ret = -EINVAL; dev_err(tas_priv->dev, "%s, no such channel(%d)\n", __func__, @@ -87,59 +44,33 @@ static int tasdevice_change_chn_book(struct tasdevice_priv *tas_priv, out: return ret; } +EXPORT_SYMBOL_GPL(tasdevice_dev_read); -int tasdev_chn_switch(struct tasdevice_priv *tas_priv, - unsigned short chn) -{ - struct i2c_client *client = (struct i2c_client *)tas_priv->client; - struct tasdevice *tasdev = &tas_priv->tasdevice[chn]; - struct regmap *map = tas_priv->regmap; - int ret; - - if (client->addr != tasdev->dev_addr) { - client->addr = tasdev->dev_addr; - /* All devices share the same regmap, clear the page - * inside regmap once switching to another device. - * Register 0 at any pages and any books inside tas2781 - * is the same one for page-switching. - */ - ret = regmap_write(map, TASDEVICE_PAGE_SELECT, 0); - if (ret < 0) { - dev_err(tas_priv->dev, "%s, E=%d\n", __func__, ret); - return ret; - } - return 1; - } - return 0; -} -EXPORT_SYMBOL_GPL(tasdev_chn_switch); - -int tasdevice_dev_read(struct tasdevice_priv *tas_priv, - unsigned short chn, unsigned int reg, unsigned int *val) +int tasdevice_dev_bulk_read(struct tasdevice_priv *tas_priv, + unsigned short chn, unsigned int reg, unsigned char *data, + unsigned int len) { int ret = 0; if (chn < tas_priv->ndev) { struct regmap *map = tas_priv->regmap; - ret = tasdevice_change_chn_book(tas_priv, chn, + ret = tas_priv->change_chn_book(tas_priv, chn, TASDEVICE_BOOK_ID(reg)); if (ret < 0) goto out; - ret = regmap_read(map, TASDEVICE_PGRG(reg), val); + ret = regmap_bulk_read(map, TASDEVICE_PGRG(reg), data, len); if (ret < 0) dev_err(tas_priv->dev, "%s, E=%d\n", __func__, ret); - } else { - ret = -EINVAL; + } else dev_err(tas_priv->dev, "%s, no such channel(%d)\n", __func__, chn); - } out: return ret; } -EXPORT_SYMBOL_GPL(tasdevice_dev_read); +EXPORT_SYMBOL_GPL(tasdevice_dev_bulk_read); int tasdevice_dev_write(struct tasdevice_priv *tas_priv, unsigned short chn, unsigned int reg, unsigned int value) @@ -149,7 +80,7 @@ int tasdevice_dev_write(struct tasdevice_priv *tas_priv, if (chn < tas_priv->ndev) { struct regmap *map = tas_priv->regmap; - ret = tasdevice_change_chn_book(tas_priv, chn, + ret = tas_priv->change_chn_book(tas_priv, chn, TASDEVICE_BOOK_ID(reg)); if (ret < 0) goto out; @@ -179,7 +110,7 @@ int tasdevice_dev_bulk_write( if (chn < tas_priv->ndev) { struct regmap *map = tas_priv->regmap; - ret = tasdevice_change_chn_book(tas_priv, chn, + ret = tas_priv->change_chn_book(tas_priv, chn, TASDEVICE_BOOK_ID(reg)); if (ret < 0) goto out; @@ -199,160 +130,20 @@ int tasdevice_dev_bulk_write( } EXPORT_SYMBOL_GPL(tasdevice_dev_bulk_write); -int tasdevice_dev_bulk_read(struct tasdevice_priv *tas_priv, - unsigned short chn, unsigned int reg, unsigned char *data, - unsigned int len) -{ - int ret = 0; - - if (chn < tas_priv->ndev) { - struct regmap *map = tas_priv->regmap; - - ret = tasdevice_change_chn_book(tas_priv, chn, - TASDEVICE_BOOK_ID(reg)); - if (ret < 0) - goto out; - - ret = regmap_bulk_read(map, TASDEVICE_PGRG(reg), data, len); - if (ret < 0) - dev_err(tas_priv->dev, "%s, E=%d\n", __func__, ret); - } else - dev_err(tas_priv->dev, "%s, no such channel(%d)\n", __func__, - chn); - -out: - return ret; -} -EXPORT_SYMBOL_GPL(tasdevice_dev_bulk_read); - -int tasdevice_dev_update_bits( - struct tasdevice_priv *tas_priv, unsigned short chn, - unsigned int reg, unsigned int mask, unsigned int value) -{ - int ret = 0; - - if (chn < tas_priv->ndev) { - struct regmap *map = tas_priv->regmap; - - ret = tasdevice_change_chn_book(tas_priv, chn, - TASDEVICE_BOOK_ID(reg)); - if (ret < 0) - goto out; - - ret = regmap_update_bits(map, TASDEVICE_PGRG(reg), - mask, value); - if (ret < 0) - dev_err(tas_priv->dev, "%s, E=%d\n", __func__, ret); - } else { - dev_err(tas_priv->dev, "%s, no such channel(%d)\n", __func__, - chn); - ret = -EINVAL; - } - -out: - return ret; -} -EXPORT_SYMBOL_GPL(tasdevice_dev_update_bits); - -struct tasdevice_priv *tasdevice_kzalloc(struct i2c_client *i2c) -{ - struct tasdevice_priv *tas_priv; - - tas_priv = devm_kzalloc(&i2c->dev, sizeof(*tas_priv), GFP_KERNEL); - if (!tas_priv) - return NULL; - tas_priv->dev = &i2c->dev; - tas_priv->client = (void *)i2c; - - return tas_priv; -} -EXPORT_SYMBOL_GPL(tasdevice_kzalloc); - -void tasdevice_reset(struct tasdevice_priv *tas_dev) -{ - int ret, i; - - if (tas_dev->reset) { - gpiod_set_value_cansleep(tas_dev->reset, 0); - usleep_range(500, 1000); - gpiod_set_value_cansleep(tas_dev->reset, 1); - } else { - for (i = 0; i < tas_dev->ndev; i++) { - ret = tasdevice_dev_write(tas_dev, i, - TASDEVICE_REG_SWRESET, - TASDEVICE_REG_SWRESET_RESET); - if (ret < 0) - dev_err(tas_dev->dev, - "dev %d swreset fail, %d\n", - i, ret); - } - } - usleep_range(1000, 1050); -} -EXPORT_SYMBOL_GPL(tasdevice_reset); - -int tascodec_init(struct tasdevice_priv *tas_priv, void *codec, - struct module *module, - void (*cont)(const struct firmware *fw, void *context)) +int tasdevice_save_calibration(struct tasdevice_priv *tas_priv) { - int ret = 0; - - /* Codec Lock Hold to ensure that codec_probe and firmware parsing and - * loading do not simultaneously execute. - */ - mutex_lock(&tas_priv->codec_lock); - - if (tas_priv->name_prefix) - scnprintf(tas_priv->rca_binaryname, 64, "%s-%sRCA%d.bin", - tas_priv->name_prefix, tas_priv->dev_name, - tas_priv->ndev); - else - scnprintf(tas_priv->rca_binaryname, 64, "%sRCA%d.bin", - tas_priv->dev_name, tas_priv->ndev); - crc8_populate_msb(tas_priv->crc8_lkp_tbl, TASDEVICE_CRC8_POLYNOMIAL); - tas_priv->codec = codec; - ret = request_firmware_nowait(module, FW_ACTION_UEVENT, - tas_priv->rca_binaryname, tas_priv->dev, GFP_KERNEL, tas_priv, - cont); - if (ret) - dev_err(tas_priv->dev, "request_firmware_nowait err:0x%08x\n", - ret); - - /* Codec Lock Release*/ - mutex_unlock(&tas_priv->codec_lock); - return ret; + if (tas_priv->save_calibration) + return tas_priv->save_calibration(tas_priv); + return -EINVAL; } -EXPORT_SYMBOL_GPL(tascodec_init); +EXPORT_SYMBOL_GPL(tasdevice_save_calibration); -int tasdevice_init(struct tasdevice_priv *tas_priv) +void tasdevice_apply_calibration(struct tasdevice_priv *tas_priv) { - int ret = 0; - int i; - - tas_priv->regmap = devm_regmap_init_i2c(tas_priv->client, - &tasdevice_regmap); - if (IS_ERR(tas_priv->regmap)) { - ret = PTR_ERR(tas_priv->regmap); - dev_err(tas_priv->dev, "Failed to allocate register map: %d\n", - ret); - goto out; - } - - tas_priv->cur_prog = -1; - tas_priv->cur_conf = -1; - - for (i = 0; i < tas_priv->ndev; i++) { - tas_priv->tasdevice[i].cur_book = -1; - tas_priv->tasdevice[i].cur_prog = -1; - tas_priv->tasdevice[i].cur_conf = -1; - } - - mutex_init(&tas_priv->codec_lock); - -out: - return ret; + if (tas_priv->apply_calibration && tas_priv->cali_data.total_sz) + tas_priv->apply_calibration(tas_priv); } -EXPORT_SYMBOL_GPL(tasdevice_init); +EXPORT_SYMBOL_GPL(tasdevice_apply_calibration); static void tasdev_dsp_prog_blk_remove(struct tasdevice_prog *prog) { @@ -434,143 +225,6 @@ void tasdevice_dsp_remove(void *context) } EXPORT_SYMBOL_GPL(tasdevice_dsp_remove); -void tasdevice_remove(struct tasdevice_priv *tas_priv) -{ - mutex_destroy(&tas_priv->codec_lock); -} -EXPORT_SYMBOL_GPL(tasdevice_remove); - -int tasdevice_save_calibration(struct tasdevice_priv *tas_priv) -{ - if (tas_priv->save_calibration) - return tas_priv->save_calibration(tas_priv); - return -EINVAL; -} -EXPORT_SYMBOL_GPL(tasdevice_save_calibration); - -void tasdevice_apply_calibration(struct tasdevice_priv *tas_priv) -{ - if (tas_priv->apply_calibration && tas_priv->cali_data.total_sz) - tas_priv->apply_calibration(tas_priv); -} -EXPORT_SYMBOL_GPL(tasdevice_apply_calibration); - -static int tasdevice_clamp(int val, int max, unsigned int invert) -{ - if (val > max) - val = max; - if (invert) - val = max - val; - if (val < 0) - val = 0; - return val; -} - -int tasdevice_amp_putvol(struct tasdevice_priv *tas_priv, - struct snd_ctl_elem_value *ucontrol, struct soc_mixer_control *mc) -{ - unsigned int invert = mc->invert; - unsigned char mask; - int max = mc->max; - int err_cnt = 0; - int val, i, ret; - - mask = (1 << fls(max)) - 1; - mask <<= mc->shift; - val = tasdevice_clamp(ucontrol->value.integer.value[0], max, invert); - for (i = 0; i < tas_priv->ndev; i++) { - ret = tasdevice_dev_update_bits(tas_priv, i, - mc->reg, mask, (unsigned int)(val << mc->shift)); - if (!ret) - continue; - err_cnt++; - dev_err(tas_priv->dev, "set AMP vol error in dev %d\n", i); - } - - /* All the devices set error, return 0 */ - return (err_cnt == tas_priv->ndev) ? 0 : 1; -} -EXPORT_SYMBOL_GPL(tasdevice_amp_putvol); - -int tasdevice_amp_getvol(struct tasdevice_priv *tas_priv, - struct snd_ctl_elem_value *ucontrol, struct soc_mixer_control *mc) -{ - unsigned int invert = mc->invert; - unsigned char mask = 0; - int max = mc->max; - int ret = 0; - int val; - - /* Read the primary device */ - ret = tasdevice_dev_read(tas_priv, 0, mc->reg, &val); - if (ret) { - dev_err(tas_priv->dev, "%s, get AMP vol error\n", __func__); - goto out; - } - - mask = (1 << fls(max)) - 1; - mask <<= mc->shift; - val = (val & mask) >> mc->shift; - val = tasdevice_clamp(val, max, invert); - ucontrol->value.integer.value[0] = val; - -out: - return ret; - -} -EXPORT_SYMBOL_GPL(tasdevice_amp_getvol); - -int tasdevice_digital_putvol(struct tasdevice_priv *tas_priv, - struct snd_ctl_elem_value *ucontrol, struct soc_mixer_control *mc) -{ - unsigned int invert = mc->invert; - int max = mc->max; - int err_cnt = 0; - int ret; - int val, i; - - val = tasdevice_clamp(ucontrol->value.integer.value[0], max, invert); - - for (i = 0; i < tas_priv->ndev; i++) { - ret = tasdevice_dev_write(tas_priv, i, mc->reg, - (unsigned int)val); - if (!ret) - continue; - err_cnt++; - dev_err(tas_priv->dev, - "set digital vol err in dev %d\n", i); - } - - /* All the devices set error, return 0 */ - return (err_cnt == tas_priv->ndev) ? 0 : 1; - -} -EXPORT_SYMBOL_GPL(tasdevice_digital_putvol); - -int tasdevice_digital_getvol(struct tasdevice_priv *tas_priv, - struct snd_ctl_elem_value *ucontrol, struct soc_mixer_control *mc) -{ - unsigned int invert = mc->invert; - int max = mc->max; - int ret, val; - - /* Read the primary device as the whole */ - ret = tasdevice_dev_read(tas_priv, 0, mc->reg, &val); - if (ret) { - dev_err(tas_priv->dev, "%s, get digital vol error\n", - __func__); - goto out; - } - - val = tasdevice_clamp(val, max, invert); - ucontrol->value.integer.value[0] = val; - -out: - return ret; - -} -EXPORT_SYMBOL_GPL(tasdevice_digital_getvol); - MODULE_DESCRIPTION("TAS2781 common library"); MODULE_AUTHOR("Shenghao Ding, TI, <shenghao-ding(a)ti.com>"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tas2781-fmwlib.c b/sound/soc/codecs/tas2781-fmwlib.c index 13a197468193..3dd76aff1b6e 100644 --- a/sound/soc/codecs/tas2781-fmwlib.c +++ b/sound/soc/codecs/tas2781-fmwlib.c @@ -803,8 +803,13 @@ static int tasdevice_process_block(void *context, unsigned char *data, chn = idx - 1; chnend = idx; } else { - chn = 0; - chnend = tas_priv->ndev; + if (tas_priv->isspi) { + chn = tas_priv->index; + chnend = chn + 1; + } else { + chn = 0; + chnend = tas_priv->ndev; + } } for (; chn < chnend; chn++) { @@ -896,7 +901,7 @@ static int tasdevice_process_block(void *context, unsigned char *data, is_err = true; break; } - rc = tasdevice_dev_update_bits(tas_priv, chn, + rc = tas_priv->update_bits(tas_priv, chn, TASDEVICE_REG(data[subblk_offset + 2], data[subblk_offset + 3], data[subblk_offset + 4]), @@ -1461,7 +1466,7 @@ static int tasdev_multibytes_chksum(struct tasdevice_priv *tasdevice, goto end; } - ret = tasdevice_dev_bulk_read(tasdevice, chn, + ret = tasdevice->dev_bulk_read(tasdevice, chn, TASDEVICE_REG(book, page, crc_data.offset), nBuf1, crc_data.len); if (ret < 0) @@ -1511,7 +1516,7 @@ static int do_singlereg_checksum(struct tasdevice_priv *tasdevice, in = check_yram(&crc_data, book, page, reg, 1); if (!in) goto end; - ret = tasdevice_dev_read(tasdevice, chl, + ret = tasdevice->dev_read(tasdevice, chl, TASDEVICE_REG(book, page, reg), &nData1); if (ret < 0) goto end; @@ -1615,7 +1620,7 @@ static int tasdev_block_chksum(struct tasdevice_priv *tas_priv, unsigned int nr_value; int ret; - ret = tasdevice_dev_read(tas_priv, chn, TASDEVICE_CHECKSUM_REG, + ret = tas_priv->dev_read(tas_priv, chn, TASDEVICE_CHECKSUM_REG, &nr_value); if (ret < 0) { dev_err(tas_priv->dev, "%s: Chn %d\n", __func__, chn); diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index b950914b7d48..9a329c561a53 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -28,6 +28,7 @@ #include <sound/pcm_params.h> #include <sound/soc.h> #include <sound/tas2781.h> +#include <sound/tas2781-comlib-i2c.h> #include <sound/tlv.h> #include <sound/tas2563-tlv.h> #include <sound/tas2781-tlv.h> -- 2.43.0

2 1

[PATCH v2] ASoC: tas2781: Add a debufs node for acoustic tuning
by Shenghao Ding 30 Apr '25

30 Apr '25

"Acoustic Tuning" debufs node is a bridge to the acoustic tuning tool which can tune the chips' acoustic effect. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v2: - Implement the "Acoustic Tuning" as debugfs node instead of kcontrol. v1: - Add "Acoustic Tuning" kcontrol for acoustic tuning. --- include/sound/tas2781.h | 16 ++++ sound/soc/codecs/tas2781-i2c.c | 147 +++++++++++++++++++++++++++++++++ 2 files changed, 163 insertions(+) diff --git a/include/sound/tas2781.h b/include/sound/tas2781.h index eff011444cc8..907f89d24e97 100644 --- a/include/sound/tas2781.h +++ b/include/sound/tas2781.h @@ -159,10 +159,26 @@ struct calidata { unsigned int cali_dat_sz_per_dev; }; +#ifdef CONFIG_DEBUG_FS + #define TASDEV_DATA_PAYLOAD_SIZE 128 + struct acoustic_data { + unsigned char len; + unsigned char id; + unsigned char addr; + unsigned char book; + unsigned char page; + unsigned char reg; + unsigned char data[TASDEV_DATA_PAYLOAD_SIZE]; + }; +#endif + struct tasdevice_priv { struct tasdevice tasdevice[TASDEVICE_MAX_CHANNELS]; struct tasdevice_rca rcabin; struct calidata cali_data; +#ifdef CONFIG_DEBUG_FS + struct acoustic_data acou_data; +#endif struct tasdevice_fw *fmw; struct gpio_desc *speaker_id; struct gpio_desc *reset; diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index b950914b7d48..e5b377539d44 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -14,6 +14,9 @@ // #include <linux/crc8.h> +#ifdef CONFIG_DEBUG_FS + #include <linux/debugfs.h> +#endif #include <linux/firmware.h> #include <linux/gpio/consumer.h> #include <linux/i2c.h> @@ -1422,10 +1425,149 @@ static int tasdevice_create_cali_ctrls(struct tasdevice_priv *priv) nctrls < i ? nctrls : i); } +#ifdef CONFIG_DEBUG_FS +/* + * This debugfs node is a bridge to the acoustic tuning application + * tool which can tune the chips' acoustic effect. + * + * package structure for PPC3 communications: + * Pkg len (1 byte) + * Pkg id (1 byte, 'r' or 'w') + * Dev id (1 byte, i2c address) + * Book id (1 byte) + * Page id (1 byte) + * Reg id (1 byte) + * switch (pkg id) { + * case 'w': + * 1 byte, length of data to read + * case 'r': + * data payload (1~128 bytes) + * } + */ +static ssize_t acoustic_ctl_read(struct file *file, char __user *to, + size_t count, loff_t *ppos) +{ + struct snd_soc_component *comp = file->private_data; + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + struct acoustic_data *p = &tas_priv->acou_data; + int ret = -1; + + if (p->id == 'r' && p->len == count && count <= sizeof(*p)) + ret = simple_read_from_buffer(to, count, ppos, p, p->len); + else + dev_err(tas_priv->dev, "Not ready for get.\n"); + return ret; +} + +static ssize_t acoustic_ctl_write(struct file *file, + const char __user *from, size_t count, loff_t *ppos) +{ + struct snd_soc_component *comp = file->private_data; + struct tasdevice_priv *priv = snd_soc_component_get_drvdata(comp); + struct acoustic_data *p = &priv->acou_data; + unsigned int max_pkg_len = sizeof(*p); + unsigned char *src; + int j, len, reg, val; + unsigned short chn; + int ret = -1; + + if (count > sizeof(*p)) { + dev_err(priv->dev, "count(%u) is larger than max(%u).\n", + (unsigned int)count, max_pkg_len); + return ret; + } + + src = memdup_user(from, count); + if (IS_ERR(src)) + return PTR_ERR(src); + + if (src[0] > max_pkg_len && src[0] != count) { + dev_err(priv->dev, "pkg(%u), max(%u), count(%u) dismatch.\n", + src[0], max_pkg_len, (unsigned int)count); + ret = 0; + goto exit; + } + + switch (src[1]) { + case 'r': + /* length of data to read */ + len = src[6]; + break; + case 'w': + /* Skip 6 bytes for package type and register address */ + len = src[0] - 6; + break; + default: + dev_err(priv->dev, "%s Wrong code %02x.\n", __func__, src[1]); + ret = 0; + goto exit; + } + + if (len < 1) { + dev_err(priv->dev, "pkg fmt invalid %02x.\n", len); + ret = 0; + goto exit; + } + + for (j = 0; j < priv->ndev; j++) + if (src[2] == priv->tasdevice[j].dev_addr) { + chn = j; + break; + } + if (j >= priv->ndev) { + dev_err(priv->dev, "no such device 0x%02x.\n", src[2]); + ret = 0; + goto exit; + } + + reg = TASDEVICE_REG(src[3], src[4], src[5]); + + guard(mutex)(&priv->codec_lock); + + if (src[1] == 'w') { + if (len > 1) + ret = tasdevice_dev_bulk_write(priv, chn, reg, + &src[6], len); + else + ret = tasdevice_dev_write(priv, chn, reg, src[6]); + } else { + struct acoustic_data *p = &priv->acou_data; + + memcpy(p, src, 6); + if (len > 1) { + ret = tasdevice_dev_bulk_read(priv, chn, reg, + p->data, len); + } else { + ret = tasdevice_dev_read(priv, chn, reg, &val); + p->data[0] = val; + } + p->len = len + 6; + } + + if (ret) + dev_err(priv->dev, "i2c communication error.\n"); + else + ret = count; +exit: + kfree(src); + return ret; +} + +static const struct file_operations acoustic_ctl_fops = { + .open = simple_open, + .read = acoustic_ctl_read, + .write = acoustic_ctl_write, +}; +#endif + static void tasdevice_fw_ready(const struct firmware *fmw, void *context) { struct tasdevice_priv *tas_priv = context; +#ifdef CONFIG_DEBUG_FS + struct snd_soc_component *comp = tas_priv->codec; + struct dentry *debugfs_root = comp->debugfs_root; +#endif int ret = 0; int i; @@ -1499,6 +1641,11 @@ static void tasdevice_fw_ready(const struct firmware *fmw, tasdevice_prmg_load(tas_priv, 0); tas_priv->cur_prog = 0; + +#ifdef CONFIG_DEBUG_FS + debugfs_create_file("acoustic_control", 0644, debugfs_root, + comp, &acoustic_ctl_fops); +#endif out: if (tas_priv->fw_state == TASDEVICE_RCA_FW_OK) { /* If DSP FW fail, DSP kcontrol won't be created. */ -- 2.43.0

2 1

Bug: src/topology/data.c: - Uninitialized 'id', Maybe Coding Error
by GitHub issues - opened 29 Apr '25

29 Apr '25

alsa-project/alsa-lib issue #453 was opened from zkh8227: #Description During the code review, I detected an issue with an uninitialized variable in the src/topology/data.c file. In the following calls, I couldn't find an initialization operation for id. Is it possible that the intention here was to call parent->id? #Error Details Bug Type: Uninitialized Variable File Path: src/topology/data.c Line Number: 1905,1908 Code Snippet： 1883 char id[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; ........ 1900 if (tplg_verify_tuples(tplg, pos, tp->array, tp->size) < 0) { 1901 if (tuples) { 1902 err = tplg_ref_add(elem, SND_TPLG_TYPE_TOKEN, parent->id); 1903 if (err < 0) 1904 return err; 1905 err = tplg_ref_add(elem2, SND_TPLG_TYPE_TUPLE, id); 1906 if (err < 0) 1907 return err; 1908 err = tplg_ref_add(parent, SND_TPLG_TYPE_DATA, id); 1909 if (err < 0) 1910 return err; 1911 tuples = NULL; 1912 } Issue URL : https://github.com/alsa-project/alsa-lib/issues/453 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

tegra: max98089: update cset names
by GitHub pull_request - opened 27 Apr '25

27 Apr '25

alsa-project/alsa-ucm-conf pull request #556 was opened from clamor-s: Adjust configs to match Linux kernel MAX98089 codec driver change. Fixes: 725570f9 ("ASoC: max98088: Remove duplicate DACs") Link to Linux kernel commit which caused regression: https://git.kernel.org/pub/scm/linux/kernel/git/stable/linux.git/commit/sou… Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/556 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/556.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v1] ALSA: hda/tas2781: Remove tas2781_spi_fwlib.c and leverage SND_SOC_TAS2781_FMWLIB
by Shenghao Ding 26 Apr '25

26 Apr '25

Most codes in tas2781_spi_fwlib.c are same as tas2781-fmwlib.c, mainly for firmware parsing, only differece is the register reading, bit update and book switching in i2c and spi. The main purpose of this patch is for code cleaup and arrange the shared part for i2c and spi. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v1: - Move the definition of TASDEVICE_CRC8_POLYNOMIAL to tas2781.h for both SPI use. - Add isspi flag in the struct tasdevice_priv to distinguish SPI or I2C. - Define three register-read related methods due to the difference between SPI and I2C. For SPI-based TAS2781, it will have one-byte delay to read memory-mapped registers. During init, these three method will be initialized as spi or i2c register-read methods. - Define book-switching method due to the difference between SPI and I2C. - Add select SND_SOC_TAS2781_COMLIB & SND_SOC_TAS2781_FMWLIB for tas2781 hda spi driver in Kconfig. - remove tas2781-spi.h & tas2781_spi_fwlib.c. - remove tas2781_spi_fwlib.o in Makefile - Move the definition of TASDEVICE_CALIBRATION_DATA_NAME. - Remove the useless calib_data definition for code cleanup. - Define the TASDEV_CALIB_N definition in tas2781_hda.h for hda driver calibration. - Use TASDEV_CALIB_N instead of TAS2563_CAL_N. - Include tas2781.h instead of tas2781-dsp.h in tas2781_hda_spi.c. - Remove #include "tas2781-spi.h" in tas2781_hda_spi.c. - Move the macro definitions only for hda spi tas2781 driver into tas2781_hda_spi.c. - Add read_flag_mask and reg_shift setting in tasdevice_regmap. --- include/sound/tas2781.h | 13 + sound/pci/hda/Kconfig | 2 + sound/pci/hda/Makefile | 2 +- sound/pci/hda/tas2781-spi.h | 157 --- sound/pci/hda/tas2781_hda.h | 3 + sound/pci/hda/tas2781_hda_i2c.c | 25 +- sound/pci/hda/tas2781_hda_spi.c | 540 ++++---- sound/pci/hda/tas2781_spi_fwlib.c | 2006 ----------------------------- sound/soc/codecs/tas2781-comlib.c | 18 +- sound/soc/codecs/tas2781-fmwlib.c | 17 +- 10 files changed, 320 insertions(+), 2463 deletions(-) delete mode 100644 sound/pci/hda/tas2781-spi.h delete mode 100644 sound/pci/hda/tas2781_spi_fwlib.c diff --git a/include/sound/tas2781.h b/include/sound/tas2781.h index eff011444cc8..a86a82f4c544 100644 --- a/include/sound/tas2781.h +++ b/include/sound/tas2781.h @@ -32,6 +32,8 @@ SNDRV_PCM_FMTBIT_S24_LE | \ SNDRV_PCM_FMTBIT_S32_LE) +#define TASDEVICE_CRC8_POLYNOMIAL 0x4d + /* PAGE Control Register (available in page0 of each book) */ #define TASDEVICE_PAGE_SELECT 0x00 #define TASDEVICE_BOOKCTL_PAGE 0x00 @@ -193,6 +195,7 @@ struct tasdevice_priv { bool force_fwload_status; bool playback_started; bool isacpi; + bool isspi; bool is_user_space_calidata; unsigned int global_addr; @@ -210,6 +213,16 @@ struct tasdevice_priv { int (*tasdevice_load_block)(struct tasdevice_priv *tas_priv, struct tasdev_blk *block); + int (*change_chn_book)(struct tasdevice_priv *tas_priv, + unsigned short chn, int book); + int (*update_bits)(struct tasdevice_priv *tas_priv, + unsigned short chn, unsigned int reg, unsigned int mask, + unsigned int value); + int (*dev_read)(struct tasdevice_priv *tas_priv, + unsigned short chn, unsigned int reg, unsigned int *value); + int (*dev_bulk_read)(struct tasdevice_priv *tas_priv, + unsigned short chn, unsigned int reg, unsigned char *p_data, + unsigned int n_length); int (*save_calibration)(struct tasdevice_priv *tas_priv); void (*apply_calibration)(struct tasdevice_priv *tas_priv); }; diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index 9c427270ff4f..90f12e0227eb 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -210,6 +210,8 @@ config SND_HDA_SCODEC_TAS2781_SPI depends on ACPI depends on EFI depends on SND_SOC + select SND_SOC_TAS2781_COMLIB + select SND_SOC_TAS2781_FMWLIB select CRC32 help Say Y or M here to include TAS2781 SPI HD-audio side codec support diff --git a/sound/pci/hda/Makefile b/sound/pci/hda/Makefile index 210c406dfbc5..72ef3a64d9db 100644 --- a/sound/pci/hda/Makefile +++ b/sound/pci/hda/Makefile @@ -40,7 +40,7 @@ snd-hda-scodec-cs35l56-spi-y := cs35l56_hda_spi.o snd-hda-cs-dsp-ctls-y := hda_cs_dsp_ctl.o snd-hda-scodec-component-y := hda_component.o snd-hda-scodec-tas2781-i2c-y := tas2781_hda_i2c.o -snd-hda-scodec-tas2781-spi-y := tas2781_hda_spi.o tas2781_spi_fwlib.o +snd-hda-scodec-tas2781-spi-y := tas2781_hda_spi.o # common driver obj-$(CONFIG_SND_HDA) := snd-hda-codec.o diff --git a/sound/pci/hda/tas2781-spi.h b/sound/pci/hda/tas2781-spi.h deleted file mode 100644 index 7a0faceeb675..000000000000 --- a/sound/pci/hda/tas2781-spi.h +++ /dev/null @@ -1,157 +0,0 @@ -/* SPDX-License-Identifier: GPL-2.0 */ -// -// ALSA SoC Texas Instruments TAS2781 Audio Smart Amplifier -// -// Copyright (C) 2024 Texas Instruments Incorporated -// https://www.ti.com -// -// The TAS2781 driver implements a flexible and configurable -// algo coefficient setting for TAS2781 chips. -// -// Author: Baojun Xu <baojun.xu(a)ti.com> -// - -#ifndef __TAS2781_SPI_H__ -#define __TAS2781_SPI_H__ - -#define TASDEVICE_RATES \ - (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | \ - SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_88200) - -#define TASDEVICE_FORMATS \ - (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | \ - SNDRV_PCM_FMTBIT_S32_LE) - -#define TASDEVICE_MAX_BOOK_NUM 256 -#define TASDEVICE_MAX_PAGE 256 - -#define TASDEVICE_MAX_SIZE (TASDEVICE_MAX_BOOK_NUM * TASDEVICE_MAX_PAGE) - -/* PAGE Control Register (available in page0 of each book) */ -#define TASDEVICE_PAGE_SELECT 0x00 -#define TASDEVICE_BOOKCTL_PAGE 0x00 -#define TASDEVICE_BOOKCTL_REG GENMASK(7, 1) -#define TASDEVICE_BOOK_ID(reg) (((reg) & GENMASK(24, 16)) >> 16) -#define TASDEVICE_PAGE_ID(reg) (((reg) & GENMASK(15, 8)) >> 8) -#define TASDEVICE_REG_ID(reg) (((reg) & GENMASK(7, 1)) >> 1) -#define TASDEVICE_PAGE_REG(reg) ((reg) & GENMASK(15, 1)) -#define TASDEVICE_REG(book, page, reg) \ - (((book) << 16) | ((page) << 8) | ((reg) << 1)) - -/* Software Reset */ -#define TAS2781_REG_SWRESET TASDEVICE_REG(0x0, 0x0, 0x01) -#define TAS2781_REG_SWRESET_RESET BIT(0) - -/* System Reset Check Register */ -#define TAS2781_REG_CLK_CONFIG TASDEVICE_REG(0x0, 0x0, 0x5c) -#define TAS2781_REG_CLK_CONFIG_RESET (0x19) -#define TAS2781_PRE_POST_RESET_CFG 3 - -/* Block Checksum */ -#define TASDEVICE_CHECKSUM TASDEVICE_REG(0x0, 0x0, 0x7e) - -/* Volume control */ -#define TAS2781_DVC_LVL TASDEVICE_REG(0x0, 0x0, 0x1a) -#define TAS2781_AMP_LEVEL TASDEVICE_REG(0x0, 0x0, 0x03) -#define TAS2781_AMP_LEVEL_MASK GENMASK(5, 1) - -#define TASDEVICE_CMD_SING_W 0x1 -#define TASDEVICE_CMD_BURST 0x2 -#define TASDEVICE_CMD_DELAY 0x3 -#define TASDEVICE_CMD_FIELD_W 0x4 - -#define TAS2781_SPI_MAX_FREQ (4 * HZ_PER_MHZ) - -#define TASDEVICE_CRC8_POLYNOMIAL 0x4d -#define TASDEVICE_SPEAKER_CALIBRATION_SIZE 20 - -/* Flag of calibration registers address. */ -#define TASDEVICE_CALIBRATION_REG_ADDRESS BIT(7) - -#define TASDEVICE_CALIBRATION_DATA_NAME L"CALI_DATA" -#define TASDEVICE_CALIBRATION_DATA_SIZE 256 - -enum calib_data { - R0_VAL = 0, - INV_R0, - R0LOW, - POWER, - TLIM, - CALIB_MAX -}; - -struct tasdevice_priv { - struct tasdevice_fw *cali_data_fmw; - struct tasdevice_rca rcabin; - struct tasdevice_fw *fmw; - struct gpio_desc *reset; - struct mutex codec_lock; - struct regmap *regmap; - struct device *dev; - - unsigned char crc8_lkp_tbl[CRC8_TABLE_SIZE]; - unsigned char coef_binaryname[64]; - unsigned char rca_binaryname[64]; - unsigned char dev_name[32]; - - bool force_fwload_status; - bool playback_started; - bool is_loading; - bool is_loaderr; - unsigned int cali_reg_array[CALIB_MAX]; - unsigned int cali_data[CALIB_MAX]; - unsigned int err_code; - void *codec; - int cur_book; - int cur_prog; - int cur_conf; - int fw_state; - int index; - int irq; - - int (*fw_parse_variable_header)(struct tasdevice_priv *tas_priv, - const struct firmware *fmw, - int offset); - int (*fw_parse_program_data)(struct tasdevice_priv *tas_priv, - struct tasdevice_fw *tas_fmw, - const struct firmware *fmw, int offset); - int (*fw_parse_configuration_data)(struct tasdevice_priv *tas_priv, - struct tasdevice_fw *tas_fmw, - const struct firmware *fmw, - int offset); - int (*tasdevice_load_block)(struct tasdevice_priv *tas_priv, - struct tasdev_blk *block); - - int (*save_calibration)(struct tasdevice_priv *tas_priv); - void (*apply_calibration)(struct tasdevice_priv *tas_priv); -}; - -int tasdevice_spi_dev_read(struct tasdevice_priv *tas_priv, - unsigned int reg, unsigned int *value); -int tasdevice_spi_dev_write(struct tasdevice_priv *tas_priv, - unsigned int reg, unsigned int value); -int tasdevice_spi_dev_bulk_write(struct tasdevice_priv *tas_priv, - unsigned int reg, unsigned char *p_data, - unsigned int n_length); -int tasdevice_spi_dev_bulk_read(struct tasdevice_priv *tas_priv, - unsigned int reg, unsigned char *p_data, - unsigned int n_length); -int tasdevice_spi_dev_update_bits(struct tasdevice_priv *tasdevice, - unsigned int reg, unsigned int mask, - unsigned int value); - -void tasdevice_spi_select_cfg_blk(void *context, int conf_no, - unsigned char block_type); -void tasdevice_spi_config_info_remove(void *context); -int tasdevice_spi_dsp_parser(void *context); -int tasdevice_spi_rca_parser(void *context, const struct firmware *fmw); -void tasdevice_spi_dsp_remove(void *context); -void tasdevice_spi_calbin_remove(void *context); -int tasdevice_spi_select_tuningprm_cfg(void *context, int prm, int cfg_no, - int rca_conf_no); -int tasdevice_spi_prmg_load(void *context, int prm_no); -int tasdevice_spi_prmg_calibdata_load(void *context, int prm_no); -void tasdevice_spi_tuning_switch(void *context, int state); -int tas2781_spi_load_calibration(void *context, char *file_name, - unsigned short i); -#endif /* __TAS2781_SPI_H__ */ diff --git a/sound/pci/hda/tas2781_hda.h b/sound/pci/hda/tas2781_hda.h index 32d27b202591..444231f87e53 100644 --- a/sound/pci/hda/tas2781_hda.h +++ b/sound/pci/hda/tas2781_hda.h @@ -9,6 +9,9 @@ #include <sound/asound.h> +#define TASDEVICE_CALIBRATION_DATA_NAME L"CALI_DATA" +#define TASDEV_CALIB_N 5 + /* * No standard control callbacks for SNDRV_CTL_ELEM_IFACE_CARD * Define two controls, one is Volume control callbacks, the other is diff --git a/sound/pci/hda/tas2781_hda_i2c.c b/sound/pci/hda/tas2781_hda_i2c.c index 9d94ae5fcfe0..1be86d1b4a53 100644 --- a/sound/pci/hda/tas2781_hda_i2c.c +++ b/sound/pci/hda/tas2781_hda_i2c.c @@ -34,15 +34,6 @@ #define TASDEVICE_SPEAKER_CALIBRATION_SIZE 20 -enum calib_data { - R0_VAL = 0, - INV_R0, - R0LOW, - POWER, - TLIM, - CALIB_MAX -}; - #define TAS2563_MAX_CHANNELS 4 #define TAS2563_CAL_POWER TASDEVICE_REG(0, 0x0d, 0x3c) @@ -50,17 +41,15 @@ enum calib_data { #define TAS2563_CAL_INVR0 TASDEVICE_REG(0, 0x0f, 0x40) #define TAS2563_CAL_R0_LOW TASDEVICE_REG(0, 0x0f, 0x48) #define TAS2563_CAL_TLIM TASDEVICE_REG(0, 0x10, 0x14) -#define TAS2563_CAL_N 5 #define TAS2563_CAL_DATA_SIZE 4 #define TAS2563_CAL_CH_SIZE 20 #define TAS2563_CAL_ARRAY_SIZE 80 -static unsigned int cal_regs[TAS2563_CAL_N] = { +static unsigned int cal_regs[TASDEV_CALIB_N] = { TAS2563_CAL_POWER, TAS2563_CAL_R0, TAS2563_CAL_INVR0, TAS2563_CAL_R0_LOW, TAS2563_CAL_TLIM, }; - struct tas2781_hda { struct device *dev; struct tasdevice_priv *priv; @@ -475,7 +464,7 @@ static void tas2563_apply_calib(struct tasdevice_priv *tas_priv) int ret; for (int i = 0; i < tas_priv->ndev; i++) { - for (int j = 0; j < TAS2563_CAL_N; ++j) { + for (int j = 0; j < TASDEV_CALIB_N; ++j) { data = cpu_to_be32( *(uint32_t *)&tas_priv->cali_data.data[offset]); ret = tasdevice_dev_bulk_write(tas_priv, i, cal_regs[j], @@ -493,7 +482,7 @@ static int tas2563_save_calibration(struct tasdevice_priv *tas_priv) static efi_guid_t efi_guid = EFI_GUID(0x1f52d2a1, 0xbb3a, 0x457d, 0xbc, 0x09, 0x43, 0xa3, 0xf4, 0x31, 0x0a, 0x92); - static efi_char16_t *efi_vars[TAS2563_MAX_CHANNELS][TAS2563_CAL_N] = { + static efi_char16_t *efi_vars[TAS2563_MAX_CHANNELS][TASDEV_CALIB_N] = { { L"Power_1", L"R0_1", L"InvR0_1", L"R0_Low_1", L"TLim_1" }, { L"Power_2", L"R0_2", L"InvR0_2", L"R0_Low_2", L"TLim_2" }, { L"Power_3", L"R0_3", L"InvR0_3", L"R0_Low_3", L"TLim_3" }, @@ -511,7 +500,7 @@ static int tas2563_save_calibration(struct tasdevice_priv *tas_priv) return -ENOMEM; for (int i = 0; i < tas_priv->ndev; ++i) { - for (int j = 0; j < TAS2563_CAL_N; ++j) { + for (int j = 0; j < TASDEV_CALIB_N; ++j) { status = efi.get_variable(efi_vars[i][j], &efi_guid, &attr, &max_size, &tas_priv->cali_data.data[offset]); @@ -535,7 +524,7 @@ static void tas2781_apply_calib(struct tasdevice_priv *tas_priv) { struct calidata *cali_data = &tas_priv->cali_data; struct cali_reg *r = &cali_data->cali_reg_array; - unsigned int cali_reg[CALIB_MAX] = { + unsigned int cali_reg[TASDEV_CALIB_N] = { TASDEVICE_REG(0, 0x17, 0x74), TASDEVICE_REG(0, 0x18, 0x0c), TASDEVICE_REG(0, 0x18, 0x14), @@ -555,7 +544,7 @@ static void tas2781_apply_calib(struct tasdevice_priv *tas_priv) } for (i = 0; i < tas_priv->ndev; i++) { - for (j = 0; j < CALIB_MAX; j++) { + for (j = 0; j < TASDEV_CALIB_N; j++) { data = cpu_to_be32( *(uint32_t *)&tas_priv->cali_data.data[oft]); rc = tasdevice_dev_bulk_write(tas_priv, i, @@ -578,7 +567,7 @@ static int tas2781_save_calibration(struct tasdevice_priv *tas_priv) { efi_guid_t efi_guid = EFI_GUID(0x02f9af02, 0x7734, 0x4233, 0xb4, 0x3d, 0x93, 0xfe, 0x5a, 0xa3, 0x5d, 0xb3); - static efi_char16_t efi_name[] = L"CALI_DATA"; + static efi_char16_t efi_name[] = TASDEVICE_CALIBRATION_DATA_NAME; unsigned int attr, crc; unsigned int *tmp_val; efi_status_t status; diff --git a/sound/pci/hda/tas2781_hda_spi.c b/sound/pci/hda/tas2781_hda_spi.c index 92be591c6bfe..977569ee4652 100644 --- a/sound/pci/hda/tas2781_hda_spi.c +++ b/sound/pci/hda/tas2781_hda_spi.c @@ -27,12 +27,10 @@ #include <sound/hda_codec.h> #include <sound/soc.h> -#include <sound/tas2781-dsp.h> +#include <sound/tas2781.h> #include <sound/tlv.h> #include <sound/tas2781-tlv.h> -#include "tas2781-spi.h" - #include "hda_local.h" #include "hda_auto_parser.h" #include "hda_component.h" @@ -40,6 +38,17 @@ #include "hda_generic.h" #include "tas2781_hda.h" +#define TASDEVICE_RANGE_MAX_SIZE (256 * 128) +#define TASDEVICE_WIN_LEN 128 +#define TAS2781_SPI_MAX_FREQ (4 * HZ_PER_MHZ) +/* Flag of calibration registers address. */ +#define TASDEVICE_CALIBRATION_REG_ADDRESS BIT(7) +#define TASDEV_UEFI_CALI_REG_ADDR_FLG BIT(7) + +/* System Reset Check Register */ +#define TAS2781_REG_CLK_CONFIG TASDEVICE_REG(0x0, 0x0, 0x5c) +#define TAS2781_REG_CLK_CONFIG_RESET 0x19 + struct tas2781_hda { struct tasdevice_priv *priv; struct acpi_device *dacpi; @@ -52,12 +61,12 @@ struct tas2781_hda { static const struct regmap_range_cfg tasdevice_ranges[] = { { .range_min = 0, - .range_max = TASDEVICE_MAX_SIZE, + .range_max = TASDEVICE_RANGE_MAX_SIZE, .selector_reg = TASDEVICE_PAGE_SELECT, .selector_mask = GENMASK(7, 0), .selector_shift = 0, .window_start = 0, - .window_len = TASDEVICE_MAX_PAGE, + .window_len = TASDEVICE_WIN_LEN, }, }; @@ -65,39 +74,19 @@ static const struct regmap_config tasdevice_regmap = { .reg_bits = 8, .val_bits = 8, .zero_flag_mask = true, + .read_flag_mask = 0x01, + .reg_shift = -1, .cache_type = REGCACHE_NONE, .ranges = tasdevice_ranges, .num_ranges = ARRAY_SIZE(tasdevice_ranges), - .max_register = TASDEVICE_MAX_SIZE, + .max_register = TASDEVICE_RANGE_MAX_SIZE, }; -static int tasdevice_spi_switch_book(struct tasdevice_priv *tas_priv, int reg) +static int tasdevice_spi_dev_read(struct tasdevice_priv *tas_priv, + unsigned short chn, unsigned int reg, unsigned int *val) { - struct regmap *map = tas_priv->regmap; - - if (tas_priv->cur_book != TASDEVICE_BOOK_ID(reg)) { - int ret = regmap_write(map, TASDEVICE_BOOKCTL_REG, - TASDEVICE_BOOK_ID(reg)); - if (ret < 0) { - dev_err(tas_priv->dev, "Switch Book E=%d\n", ret); - return ret; - } - tas_priv->cur_book = TASDEVICE_BOOK_ID(reg); - } - return 0; -} - -int tasdevice_spi_dev_read(struct tasdevice_priv *tas_priv, - unsigned int reg, - unsigned int *val) -{ - struct regmap *map = tas_priv->regmap; int ret; - ret = tasdevice_spi_switch_book(tas_priv, reg); - if (ret < 0) - return ret; - /* * In our TAS2781 SPI mode, if read from other book (not book 0), * or read from page number larger than 1 in book 0, one more byte @@ -106,11 +95,11 @@ int tasdevice_spi_dev_read(struct tasdevice_priv *tas_priv, if ((TASDEVICE_BOOK_ID(reg) > 0) || (TASDEVICE_PAGE_ID(reg) > 1)) { unsigned char data[2]; - ret = regmap_bulk_read(map, TASDEVICE_PAGE_REG(reg) | 1, + ret = tasdevice_dev_bulk_read(tas_priv, chn, reg, data, sizeof(data)); *val = data[1]; } else { - ret = regmap_read(map, TASDEVICE_PAGE_REG(reg) | 1, val); + ret = tasdevice_dev_read(tas_priv, chn, reg, val); } if (ret < 0) dev_err(tas_priv->dev, "%s, E=%d\n", __func__, ret); @@ -118,71 +107,25 @@ int tasdevice_spi_dev_read(struct tasdevice_priv *tas_priv, return ret; } -int tasdevice_spi_dev_write(struct tasdevice_priv *tas_priv, - unsigned int reg, - unsigned int value) -{ - struct regmap *map = tas_priv->regmap; - int ret; - - ret = tasdevice_spi_switch_book(tas_priv, reg); - if (ret < 0) - return ret; - - ret = regmap_write(map, TASDEVICE_PAGE_REG(reg), value); - if (ret < 0) - dev_err(tas_priv->dev, "%s, E=%d\n", __func__, ret); - - return ret; -} - -int tasdevice_spi_dev_bulk_write(struct tasdevice_priv *tas_priv, - unsigned int reg, - unsigned char *data, - unsigned int len) -{ - struct regmap *map = tas_priv->regmap; - int ret; - - ret = tasdevice_spi_switch_book(tas_priv, reg); - if (ret < 0) - return ret; - - ret = regmap_bulk_write(map, TASDEVICE_PAGE_REG(reg), data, len); - if (ret < 0) - dev_err(tas_priv->dev, "%s, E=%d\n", __func__, ret); - - return ret; -} - -int tasdevice_spi_dev_bulk_read(struct tasdevice_priv *tas_priv, - unsigned int reg, - unsigned char *data, - unsigned int len) +static int tasdevice_spi_dev_bulk_read(struct tasdevice_priv *tas_priv, + unsigned short chn, unsigned int reg, unsigned char *data, + unsigned int len) { - struct regmap *map = tas_priv->regmap; int ret; - ret = tasdevice_spi_switch_book(tas_priv, reg); - if (ret < 0) - return ret; - - if (len > TASDEVICE_MAX_PAGE) - len = TASDEVICE_MAX_PAGE; /* * In our TAS2781 SPI mode, if read from other book (not book 0), * or read from page number larger than 1 in book 0, one more byte * read is needed, and first byte is a dummy byte, need to be ignored. */ if ((TASDEVICE_BOOK_ID(reg) > 0) || (TASDEVICE_PAGE_ID(reg) > 1)) { - unsigned char buf[TASDEVICE_MAX_PAGE+1]; + unsigned char buf[TASDEVICE_WIN_LEN + 1]; - ret = regmap_bulk_read(map, TASDEVICE_PAGE_REG(reg) | 1, buf, - len + 1); + ret = tasdevice_dev_bulk_read(tas_priv, chn, reg, + buf, len + 1); memcpy(data, buf + 1, len); } else { - ret = regmap_bulk_read(map, TASDEVICE_PAGE_REG(reg) | 1, data, - len); + ret = tasdevice_dev_bulk_read(tas_priv, chn, reg, data, len); } if (ret < 0) dev_err(tas_priv->dev, "%s, E=%d\n", __func__, ret); @@ -190,31 +133,55 @@ int tasdevice_spi_dev_bulk_read(struct tasdevice_priv *tas_priv, return ret; } -int tasdevice_spi_dev_update_bits(struct tasdevice_priv *tas_priv, - unsigned int reg, - unsigned int mask, - unsigned int value) +static int tasdevice_spi_dev_update_bits(struct tasdevice_priv *tas_priv, + unsigned short chn, unsigned int reg, unsigned int mask, + unsigned int value) { - struct regmap *map = tas_priv->regmap; int ret, val; /* * In our TAS2781 SPI mode, read/write was masked in last bit of * address, it cause regmap_update_bits() not work as expected. */ - ret = tasdevice_spi_dev_read(tas_priv, reg, &val); + ret = tasdevice_dev_read(tas_priv, chn, reg, &val); if (ret < 0) { dev_err(tas_priv->dev, "%s, E=%d\n", __func__, ret); return ret; } - ret = regmap_write(map, TASDEVICE_PAGE_REG(reg), - (val & ~mask) | (mask & value)); + + ret = tasdevice_dev_write(tas_priv, chn, TASDEVICE_PAGE_REG(reg), + (val & ~mask) | (mask & value)); if (ret < 0) dev_err(tas_priv->dev, "%s, E=%d\n", __func__, ret); return ret; } +static int tasdevice_spi_change_chn_book(struct tasdevice_priv *p, + unsigned short chn, int book) +{ + int ret = 0; + + if (chn == p->index) { + struct tasdevice *tasdev = &p->tasdevice[chn]; + struct regmap *map = p->regmap; + + if (tasdev->cur_book != book) { + ret = regmap_write(map, TASDEVICE_BOOKCTL_REG, book); + if (ret < 0) + dev_err(p->dev, "%s, E=%d\n", __func__, ret); + else + tasdev->cur_book = book; + } + } else { + ret = -EXDEV; + dev_dbg(p->dev, "Not error, %s ignore channel(%d)\n", + __func__, chn); + } + + return ret; +} + static void tas2781_spi_reset(struct tasdevice_priv *tas_dev) { int ret; @@ -223,12 +190,13 @@ static void tas2781_spi_reset(struct tasdevice_priv *tas_dev) gpiod_set_value_cansleep(tas_dev->reset, 0); fsleep(800); gpiod_set_value_cansleep(tas_dev->reset, 1); + } else { + ret = tasdevice_dev_write(tas_dev, tas_dev->index, + TASDEVICE_REG_SWRESET, TASDEVICE_REG_SWRESET_RESET); + if (ret < 0) + dev_err(tas_dev->dev, "dev sw-reset fail, %d\n", ret); + fsleep(1000); } - ret = tasdevice_spi_dev_write(tas_dev, TAS2781_REG_SWRESET, - TAS2781_REG_SWRESET_RESET); - if (ret < 0) - dev_err(tas_dev->dev, "dev sw-reset fail, %d\n", ret); - fsleep(1000); } static int tascodec_spi_init(struct tasdevice_priv *tas_priv, @@ -245,7 +213,7 @@ static int tascodec_spi_init(struct tasdevice_priv *tas_priv, scnprintf(tas_priv->rca_binaryname, sizeof(tas_priv->rca_binaryname), "%sRCA%d.bin", - tas_priv->dev_name, tas_priv->index); + tas_priv->dev_name, tas_priv->ndev); crc8_populate_msb(tas_priv->crc8_lkp_tbl, TASDEVICE_CRC8_POLYNOMIAL); tas_priv->codec = codec; ret = request_firmware_nowait(module, FW_ACTION_UEVENT, @@ -260,26 +228,22 @@ static int tascodec_spi_init(struct tasdevice_priv *tas_priv, static void tasdevice_spi_init(struct tasdevice_priv *tas_priv) { - tas_priv->cur_prog = -1; - tas_priv->cur_conf = -1; + tas_priv->tasdevice[tas_priv->index].cur_book = -1; + tas_priv->tasdevice[tas_priv->index].cur_conf = -1; + tas_priv->tasdevice[tas_priv->index].cur_prog = -1; - tas_priv->cur_book = -1; - tas_priv->cur_prog = -1; - tas_priv->cur_conf = -1; + tas_priv->isspi = true; - /* Store default registers address for calibration data. */ - tas_priv->cali_reg_array[0] = TASDEVICE_REG(0, 0x17, 0x74); - tas_priv->cali_reg_array[1] = TASDEVICE_REG(0, 0x18, 0x0c); - tas_priv->cali_reg_array[2] = TASDEVICE_REG(0, 0x18, 0x14); - tas_priv->cali_reg_array[3] = TASDEVICE_REG(0, 0x13, 0x70); - tas_priv->cali_reg_array[4] = TASDEVICE_REG(0, 0x18, 0x7c); + tas_priv->update_bits = tasdevice_spi_dev_update_bits; + tas_priv->change_chn_book = tasdevice_spi_change_chn_book; + tas_priv->dev_read = tasdevice_spi_dev_read; + tas_priv->dev_bulk_read = tasdevice_spi_dev_bulk_read; mutex_init(&tas_priv->codec_lock); } static int tasdevice_spi_amp_putvol(struct tasdevice_priv *tas_priv, - struct snd_ctl_elem_value *ucontrol, - struct soc_mixer_control *mc) + struct snd_ctl_elem_value *ucontrol, struct soc_mixer_control *mc) { unsigned int invert = mc->invert; unsigned char mask; @@ -290,7 +254,8 @@ static int tasdevice_spi_amp_putvol(struct tasdevice_priv *tas_priv, mask <<= mc->shift; val = clamp(invert ? max - ucontrol->value.integer.value[0] : ucontrol->value.integer.value[0], 0, max); - ret = tasdevice_spi_dev_update_bits(tas_priv, + + ret = tasdevice_spi_dev_update_bits(tas_priv, tas_priv->index, mc->reg, mask, (unsigned int)(val << mc->shift)); if (ret) dev_err(tas_priv->dev, "set AMP vol error in dev %d\n", @@ -300,16 +265,14 @@ static int tasdevice_spi_amp_putvol(struct tasdevice_priv *tas_priv, } static int tasdevice_spi_amp_getvol(struct tasdevice_priv *tas_priv, - struct snd_ctl_elem_value *ucontrol, - struct soc_mixer_control *mc) + struct snd_ctl_elem_value *ucontrol, struct soc_mixer_control *mc) { unsigned int invert = mc->invert; unsigned char mask = 0; int max = mc->max; int ret, val; - /* Read the primary device */ - ret = tasdevice_spi_dev_read(tas_priv, mc->reg, &val); + ret = tasdevice_spi_dev_read(tas_priv, tas_priv->index, mc->reg, &val); if (ret) { dev_err(tas_priv->dev, "%s, get AMP vol error\n", __func__); return ret; @@ -324,9 +287,8 @@ static int tasdevice_spi_amp_getvol(struct tasdevice_priv *tas_priv, return ret; } -static int tasdevice_spi_digital_putvol(struct tasdevice_priv *tas_priv, - struct snd_ctl_elem_value *ucontrol, - struct soc_mixer_control *mc) +static int tasdevice_spi_digital_putvol(struct tasdevice_priv *p, + struct snd_ctl_elem_value *ucontrol, struct soc_mixer_control *mc) { unsigned int invert = mc->invert; int max = mc->max; @@ -334,26 +296,23 @@ static int tasdevice_spi_digital_putvol(struct tasdevice_priv *tas_priv, val = clamp(invert ? max - ucontrol->value.integer.value[0] : ucontrol->value.integer.value[0], 0, max); - ret = tasdevice_spi_dev_write(tas_priv, mc->reg, (unsigned int)val); + ret = tasdevice_dev_write(p, p->index, mc->reg, (unsigned int)val); if (ret) - dev_err(tas_priv->dev, "set digital vol err in dev %d\n", - tas_priv->index); + dev_err(p->dev, "set digital vol err in dev %d\n", p->index); return ret; } -static int tasdevice_spi_digital_getvol(struct tasdevice_priv *tas_priv, - struct snd_ctl_elem_value *ucontrol, - struct soc_mixer_control *mc) +static int tasdevice_spi_digital_getvol(struct tasdevice_priv *p, + struct snd_ctl_elem_value *ucontrol, struct soc_mixer_control *mc) { unsigned int invert = mc->invert; int max = mc->max; int ret, val; - /* Read the primary device as the whole */ - ret = tasdevice_spi_dev_read(tas_priv, mc->reg, &val); + ret = tasdevice_spi_dev_read(p, p->index, mc->reg, &val); if (ret) { - dev_err(tas_priv->dev, "%s, get digital vol err\n", __func__); + dev_err(p->dev, "%s, get digital vol err\n", __func__); return ret; } @@ -392,7 +351,7 @@ static int tas2781_read_acpi(struct tas2781_hda *tas_hda, ret = -EINVAL; goto err; } - nval = ret; + p->ndev = nval = ret; ret = device_property_read_u32_array(physdev, property, values, nval); if (ret) @@ -435,14 +394,17 @@ static int tas2781_read_acpi(struct tas2781_hda *tas_hda, static void tas2781_hda_playback_hook(struct device *dev, int action) { struct tas2781_hda *tas_hda = dev_get_drvdata(dev); + struct tasdevice_priv *tas_priv = tas_hda->priv; if (action == HDA_GEN_PCM_ACT_OPEN) { pm_runtime_get_sync(dev); - guard(mutex)(&tas_hda->priv->codec_lock); - tasdevice_spi_tuning_switch(tas_hda->priv, 0); + guard(mutex)(&tas_priv->codec_lock); + if (tas_priv->fw_state == TASDEVICE_DSP_FW_ALL_OK) + tasdevice_tuning_switch(tas_hda->priv, 0); } else if (action == HDA_GEN_PCM_ACT_CLOSE) { - guard(mutex)(&tas_hda->priv->codec_lock); - tasdevice_spi_tuning_switch(tas_hda->priv, 1); + guard(mutex)(&tas_priv->codec_lock); + if (tas_priv->fw_state == TASDEVICE_DSP_FW_ALL_OK) + tasdevice_tuning_switch(tas_priv, 1); pm_runtime_mark_last_busy(dev); pm_runtime_put_autosuspend(dev); } @@ -589,6 +551,7 @@ static int tas2781_digital_getvol(struct snd_kcontrol *kcontrol, struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; + guard(mutex)(&tas_priv->codec_lock); return tasdevice_spi_digital_getvol(tas_priv, ucontrol, mc); } @@ -599,6 +562,7 @@ static int tas2781_amp_getvol(struct snd_kcontrol *kcontrol, struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; + guard(mutex)(&tas_priv->codec_lock); return tasdevice_spi_amp_getvol(tas_priv, ucontrol, mc); } @@ -610,6 +574,7 @@ static int tas2781_digital_putvol(struct snd_kcontrol *kcontrol, (struct soc_mixer_control *)kcontrol->private_value; /* The check of the given value is in tasdevice_digital_putvol. */ + guard(mutex)(&tas_priv->codec_lock); return tasdevice_spi_digital_putvol(tas_priv, ucontrol, mc); } @@ -621,6 +586,7 @@ static int tas2781_amp_putvol(struct snd_kcontrol *kcontrol, (struct soc_mixer_control *)kcontrol->private_value; /* The check of the given value is in tasdevice_amp_putvol. */ + guard(mutex)(&tas_priv->codec_lock); return tasdevice_spi_amp_putvol(tas_priv, ucontrol, mc); } @@ -725,137 +691,164 @@ static const struct snd_kcontrol_new tas2781_dsp_conf_ctrl[] = { static void tas2781_apply_calib(struct tasdevice_priv *tas_priv) { - int i, rc; - - /* - * If no calibration data exist in tasdevice_priv *tas_priv, - * calibration apply will be ignored, and use default values - * in firmware binary, which was loaded during firmware download. - */ - if (tas_priv->cali_data[0] == 0) - return; - /* - * Calibration data was saved in tasdevice_priv *tas_priv as: - * unsigned int cali_data[CALIB_MAX]; - * and every data (in 4 bytes) will be saved in register which in - * book 0, and page number in page_array[], offset was saved in - * rgno_array[]. - */ - for (i = 0; i < CALIB_MAX; i++) { - rc = tasdevice_spi_dev_bulk_write(tas_priv, - tas_priv->cali_reg_array[i], - (unsigned char *)&tas_priv->cali_data[i], 4); - if (rc < 0) - dev_err(tas_priv->dev, - "chn %d calib %d bulk_wr err = %d\n", - tas_priv->index, i, rc); - } -} - -/* - * Update the calibration data, including speaker impedance, f0, etc, - * into algo. Calibrate data is done by manufacturer in the factory. - * These data are used by Algo for calculating the speaker temperature, - * speaker membrane excursion and f0 in real time during playback. - * Calibration data format in EFI is V2, since 2024. - */ -static int tas2781_save_calibration(struct tasdevice_priv *tas_priv) -{ - /* - * GUID was used for data access in BIOS, it was provided by board - * manufactory, like HP: "{02f9af02-7734-4233-b43d-93fe5aa35db3}" - */ - efi_guid_t efi_guid = - EFI_GUID(0x02f9af02, 0x7734, 0x4233, - 0xb4, 0x3d, 0x93, 0xfe, 0x5a, 0xa3, 0x5d, 0xb3); - static efi_char16_t efi_name[] = TASDEVICE_CALIBRATION_DATA_NAME; - unsigned char data[TASDEVICE_CALIBRATION_DATA_SIZE], *buf; - unsigned int attr, crc, offset, *tmp_val; - unsigned long total_sz = 0; - efi_status_t status; + struct calidata *cali_data = &tas_priv->cali_data; + struct cali_reg *r = &cali_data->cali_reg_array; + unsigned char *data = cali_data->data; + unsigned int *tmp_val = (unsigned int *)data; + unsigned int cali_reg[TASDEV_CALIB_N] = { + TASDEVICE_REG(0, 0x17, 0x74), + TASDEVICE_REG(0, 0x18, 0x0c), + TASDEVICE_REG(0, 0x18, 0x14), + TASDEVICE_REG(0, 0x13, 0x70), + TASDEVICE_REG(0, 0x18, 0x7c), + }; + unsigned int crc, oft; + unsigned char *buf; + int i, j, k, l; - tas_priv->cali_data[0] = 0; - status = efi.get_variable(efi_name, &efi_guid, &attr, &total_sz, data); - if (status == EFI_BUFFER_TOO_SMALL) { - if (total_sz > TASDEVICE_CALIBRATION_DATA_SIZE) - return -ENOMEM; - /* Get variable contents into buffer */ - status = efi.get_variable(efi_name, &efi_guid, &attr, - &total_sz, data); - } - if (status != EFI_SUCCESS) - return status; - - tmp_val = (unsigned int *)data; if (tmp_val[0] == 2781) { /* * New features were added in calibrated Data V3: * 1. Added calibration registers address define in - * a node, marked as Device id == 0x80. + * a node, marked as Device id == 0x80. * New features were added in calibrated Data V2: * 1. Added some the fields to store the link_id and - * uniqie_id for multi-link solutions + * uniqie_id for multi-link solutions * 2. Support flexible number of devices instead of - * fixed one in V1. + * fixed one in V1. * Layout of calibrated data V2 in UEFI(total 256 bytes): * ChipID (2781, 4 bytes) - * Device-Sum (4 bytes) + * Data-Group-Sum (4 bytes) * TimeStamp of Calibration (4 bytes) - * for (i = 0; i < Device-Sum; i++) { - * Device #i index_info () { - * SDW link id (2bytes) - * SDW unique_id (2bytes) - * } // if Device number is 0x80, mean it's - * calibration registers address. - * Calibrated Data of Device #i (20 bytes) + * for (i = 0; i < Data-Group-Sum; i++) { + * if (Data type != 0x80) (4 bytes) + * Calibrated Data of Device #i (20 bytes) + * else + * Calibration registers address (5*4 = 20 bytes) + * # V2: No reg addr in data grp section. + * # V3: Normally the last grp is the reg addr. * } * CRC (4 bytes) * Reserved (the rest) */ crc = crc32(~0, data, (3 + tmp_val[1] * 6) * 4) ^ ~0; - if (crc != tmp_val[3 + tmp_val[1] * 6]) - return 0; - - for (int j = 0; j < tmp_val[1]; j++) { - offset = j * 6 + 3; - if (tmp_val[offset] == tas_priv->index) { - for (int i = 0; i < CALIB_MAX; i++) - tas_priv->cali_data[i] = - tmp_val[offset + i + 1]; - } else if (tmp_val[offset] == - TASDEVICE_CALIBRATION_REG_ADDRESS) { - for (int i = 0; i < CALIB_MAX; i++) { - buf = &data[(offset + i + 1) * 4]; - tas_priv->cali_reg_array[i] = - TASDEVICE_REG(buf[1], buf[2], - buf[3]); + if (crc != tmp_val[3 + tmp_val[1] * 6]) { + cali_data->total_sz = 0; + dev_err(tas_priv->dev, "%s: CRC error\n", __func__); + return; + } + + for (j = 0, k = 0; j < tmp_val[1]; j++) { + oft = j * 6 + 3; + if (tmp_val[oft] == TASDEV_UEFI_CALI_REG_ADDR_FLG) { + for (i = 0; i < TASDEV_CALIB_N; i++) { + buf = &data[(oft + i + 1) * 4]; + cali_reg[i] = TASDEVICE_REG(buf[1], + buf[2], buf[3]); + } + } else { + l = j * (cali_data->cali_dat_sz_per_dev + 1); + if (k >= tas_priv->ndev || l > oft * 4) { + dev_err(tas_priv->dev, + "%s: dev sum error\n", + __func__); + cali_data->total_sz = 0; + return; } + + data[l] = k; + for (i = 0; i < TASDEV_CALIB_N * 4; i++) + data[l + i] = data[4 * oft + i]; + k++; } - tas_priv->apply_calibration(tas_priv); } } else { /* * Calibration data is in V1 format. * struct cali_data { - * char cali_data[20]; + * char cali_data[20]; * } * * struct { - * struct cali_data cali_data[4]; - * int TimeStamp of Calibration (4 bytes) - * int CRC (4 bytes) + * struct cali_data cali_data[4]; + * int TimeStamp of Calibration (4 bytes) + * int CRC (4 bytes) * } ueft; */ crc = crc32(~0, data, 84) ^ ~0; - if (crc == tmp_val[21]) { - for (int i = 0; i < CALIB_MAX; i++) - tas_priv->cali_data[i] = - tmp_val[tas_priv->index * 5 + i]; - tas_priv->apply_calibration(tas_priv); + if (crc != tmp_val[21]) { + cali_data->total_sz = 0; + dev_err(tas_priv->dev, "%s: V1 CRC error\n", __func__); + return; + } + + for (j = tas_priv->ndev - 1; j >= 0; j--) { + l = j * (cali_data->cali_dat_sz_per_dev + 1); + for (i = TASDEV_CALIB_N * 4; i > 0 ; i--) + data[l + i] = data[tas_priv->index * 5 + i]; + data[l+i] = j; + } + } + + if (tas_priv->dspbin_typ == TASDEV_BASIC) { + r->r0_reg = cali_reg[0]; + r->invr0_reg = cali_reg[1]; + r->r0_low_reg = cali_reg[2]; + r->pow_reg = cali_reg[3]; + r->tlimit_reg = cali_reg[4]; + } + + tas_priv->is_user_space_calidata = true; + cali_data->total_sz = + tas_priv->ndev * (cali_data->cali_dat_sz_per_dev + 1); +} + +/* + * Update the calibration data, including speaker impedance, f0, etc, + * into algo. Calibrate data is done by manufacturer in the factory. + * These data are used by Algo for calculating the speaker temperature, + * speaker membrane excursion and f0 in real time during playback. + * Calibration data format in EFI is V2, since 2024. + */ +static int tas2781_save_calibration(struct tasdevice_priv *tas_priv) +{ + /* + * GUID was used for data access in BIOS, it was provided by board + * manufactory, like HP: "{02f9af02-7734-4233-b43d-93fe5aa35db3}" + */ + efi_guid_t efi_guid = + EFI_GUID(0x02f9af02, 0x7734, 0x4233, + 0xb4, 0x3d, 0x93, 0xfe, 0x5a, 0xa3, 0x5d, 0xb3); + static efi_char16_t efi_name[] = TASDEVICE_CALIBRATION_DATA_NAME; + struct calidata *cali_data = &tas_priv->cali_data; + unsigned long total_sz = 0; + unsigned int attr, size; + efi_status_t status; + unsigned char *data; + + cali_data->cali_dat_sz_per_dev = 20; + size = tas_priv->ndev * (cali_data->cali_dat_sz_per_dev + 1); + status = efi.get_variable(efi_name, &efi_guid, &attr, &total_sz, NULL); + cali_data->total_sz = total_sz > size ? total_sz : size; + if (status == EFI_BUFFER_TOO_SMALL) { + /* Allocate data buffer of data_size bytes */ + data = tas_priv->cali_data.data = devm_kzalloc(tas_priv->dev, + tas_priv->cali_data.total_sz, GFP_KERNEL); + if (!data) { + tas_priv->cali_data.total_sz = 0; + return -ENOMEM; } + /* Get variable contents into buffer */ + status = efi.get_variable(efi_name, &efi_guid, &attr, + &tas_priv->cali_data.total_sz, data); + } + if (status != EFI_SUCCESS) { + tas_priv->cali_data.total_sz = 0; + return status; } + tas_priv->apply_calibration(tas_priv); return 0; } @@ -883,7 +876,7 @@ static void tasdev_fw_ready(const struct firmware *fmw, void *context) pm_runtime_get_sync(tas_priv->dev); guard(mutex)(&tas_priv->codec_lock); - ret = tasdevice_spi_rca_parser(tas_priv, fmw); + ret = tasdevice_rca_parser(tas_priv, fmw); if (ret) goto out; @@ -910,12 +903,12 @@ static void tasdev_fw_ready(const struct firmware *fmw, void *context) } } - tasdevice_spi_dsp_remove(tas_priv); + tasdevice_dsp_remove(tas_priv); tas_priv->fw_state = TASDEVICE_DSP_FW_PENDING; scnprintf(tas_priv->coef_binaryname, 64, "TAS2XXX%08X-%01d.bin", codec->core.subsystem_id, tas_priv->index); - ret = tasdevice_spi_dsp_parser(tas_priv); + ret = tasdevice_dsp_parser(tas_priv); if (ret) { dev_err(tas_priv->dev, "dspfw load %s error\n", tas_priv->coef_binaryname); @@ -945,26 +938,27 @@ static void tasdev_fw_ready(const struct firmware *fmw, void *context) goto out; } - /* Perform AMP reset before firmware download. */ - tas_priv->rcabin.profile_cfg_id = TAS2781_PRE_POST_RESET_CFG; tas2781_spi_reset(tas_priv); tas_priv->rcabin.profile_cfg_id = 0; tas_priv->fw_state = TASDEVICE_DSP_FW_ALL_OK; - ret = tasdevice_spi_dev_read(tas_priv, TAS2781_REG_CLK_CONFIG, &val); + ret = tas_priv->dev_read(tas_priv, tas_priv->index, + TAS2781_REG_CLK_CONFIG, &val); if (ret < 0) goto out; - if (val == TAS2781_REG_CLK_CONFIG_RESET) - ret = tasdevice_spi_prmg_load(tas_priv, 0); - if (ret < 0) { - dev_err(tas_priv->dev, "FW download failed = %d\n", ret); - goto out; + if (val == TAS2781_REG_CLK_CONFIG_RESET) { + ret = tasdevice_prmg_load(tas_priv, 0); + if (ret < 0) { + dev_err(tas_priv->dev, "FW download failed = %d\n", + ret); + goto out; + } } if (tas_priv->fmw->nr_programs > 0) - tas_priv->cur_prog = 0; + tas_priv->tasdevice[tas_priv->index].cur_prog = 0; if (tas_priv->fmw->nr_configurations > 0) - tas_priv->cur_conf = 0; + tas_priv->tasdevice[tas_priv->index].cur_conf = 0; /* * If calibrated data occurs error, dsp will still works with default @@ -1017,9 +1011,10 @@ static void tas2781_hda_unbind(struct device *dev, struct device *master, { struct tas2781_hda *tas_hda = dev_get_drvdata(dev); struct hda_component_parent *parent = master_data; + struct tasdevice_priv *tas_priv = tas_hda->priv; struct hda_component *comp; - comp = hda_component_from_index(parent, tas_hda->priv->index); + comp = hda_component_from_index(parent, tas_priv->index); if (comp && (comp->dev == dev)) { comp->dev = NULL; memset(comp->name, 0, sizeof(comp->name)); @@ -1028,8 +1023,8 @@ static void tas2781_hda_unbind(struct device *dev, struct device *master, tas2781_hda_remove_controls(tas_hda); - tasdevice_spi_config_info_remove(tas_hda->priv); - tasdevice_spi_dsp_remove(tas_hda->priv); + tasdevice_config_info_remove(tas_priv); + tasdevice_dsp_remove(tas_priv); tas_hda->priv->fw_state = TASDEVICE_DSP_FW_PENDING; } @@ -1124,14 +1119,16 @@ static void tas2781_hda_spi_remove(struct spi_device *spi) static int tas2781_runtime_suspend(struct device *dev) { struct tas2781_hda *tas_hda = dev_get_drvdata(dev); + struct tasdevice_priv *tas_priv = tas_hda->priv; - guard(mutex)(&tas_hda->priv->codec_lock); + guard(mutex)(&tas_priv->codec_lock); - if (tas_hda->priv->playback_started) - tasdevice_spi_tuning_switch(tas_hda->priv, 1); + if (tas_priv->fw_state == TASDEVICE_DSP_FW_ALL_OK + && tas_priv->playback_started) + tasdevice_tuning_switch(tas_priv, 1); - tas_hda->priv->cur_book = -1; - tas_hda->priv->cur_conf = -1; + tas_priv->tasdevice[tas_priv->index].cur_book = -1; + tas_priv->tasdevice[tas_priv->index].cur_conf = -1; return 0; } @@ -1139,11 +1136,13 @@ static int tas2781_runtime_suspend(struct device *dev) static int tas2781_runtime_resume(struct device *dev) { struct tas2781_hda *tas_hda = dev_get_drvdata(dev); + struct tasdevice_priv *tas_priv = tas_hda->priv; - guard(mutex)(&tas_hda->priv->codec_lock); + guard(mutex)(&tas_priv->codec_lock); - if (tas_hda->priv->playback_started) - tasdevice_spi_tuning_switch(tas_hda->priv, 0); + if (tas_priv->fw_state == TASDEVICE_DSP_FW_ALL_OK + && tas_priv->playback_started) + tasdevice_tuning_switch(tas_priv, 0); return 0; } @@ -1151,6 +1150,7 @@ static int tas2781_runtime_resume(struct device *dev) static int tas2781_system_suspend(struct device *dev) { struct tas2781_hda *tas_hda = dev_get_drvdata(dev); + struct tasdevice_priv *tas_priv = tas_hda->priv; int ret; ret = pm_runtime_force_suspend(dev); @@ -1158,8 +1158,9 @@ static int tas2781_system_suspend(struct device *dev) return ret; /* Shutdown chip before system suspend */ - if (tas_hda->priv->playback_started) - tasdevice_spi_tuning_switch(tas_hda->priv, 1); + if (tas_priv->fw_state == TASDEVICE_DSP_FW_ALL_OK + && tas_priv->playback_started) + tasdevice_tuning_switch(tas_priv, 1); return 0; } @@ -1167,32 +1168,34 @@ static int tas2781_system_suspend(struct device *dev) static int tas2781_system_resume(struct device *dev) { struct tas2781_hda *tas_hda = dev_get_drvdata(dev); + struct tasdevice_priv *tas_priv = tas_hda->priv; int ret, val; ret = pm_runtime_force_resume(dev); if (ret) return ret; - guard(mutex)(&tas_hda->priv->codec_lock); - ret = tasdevice_spi_dev_read(tas_hda->priv, TAS2781_REG_CLK_CONFIG, - &val); + guard(mutex)(&tas_priv->codec_lock); + ret = tas_priv->dev_read(tas_priv, tas_priv->index, + TAS2781_REG_CLK_CONFIG, &val); if (ret < 0) return ret; if (val == TAS2781_REG_CLK_CONFIG_RESET) { - tas_hda->priv->cur_book = -1; - tas_hda->priv->cur_conf = -1; - tas_hda->priv->cur_prog = -1; + tas_priv->tasdevice[tas_priv->index].cur_book = -1; + tas_priv->tasdevice[tas_priv->index].cur_conf = -1; + tas_priv->tasdevice[tas_priv->index].cur_prog = -1; - ret = tasdevice_spi_prmg_load(tas_hda->priv, 0); + ret = tasdevice_prmg_load(tas_priv, 0); if (ret < 0) { - dev_err(tas_hda->priv->dev, + dev_err(tas_priv->dev, "FW download failed = %d\n", ret); return ret; } + tas_priv->fw_state = TASDEVICE_DSP_FW_ALL_OK; - if (tas_hda->priv->playback_started) - tasdevice_spi_tuning_switch(tas_hda->priv, 0); + if (tas_priv->playback_started) + tasdevice_tuning_switch(tas_priv, 0); } return ret; @@ -1229,3 +1232,4 @@ module_spi_driver(tas2781_hda_spi_driver); MODULE_DESCRIPTION("TAS2781 HDA SPI Driver"); MODULE_AUTHOR("Baojun, Xu, <baojun.xug(a)ti.com>"); MODULE_LICENSE("GPL"); +MODULE_IMPORT_NS("SND_SOC_TAS2781_FMWLIB"); diff --git a/sound/pci/hda/tas2781_spi_fwlib.c b/sound/pci/hda/tas2781_spi_fwlib.c deleted file mode 100644 index d90d022d8449..000000000000 --- a/sound/pci/hda/tas2781_spi_fwlib.c +++ /dev/null @@ -1,2006 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0 -// -// TAS2781 HDA SPI driver -// -// Copyright 2024 - 2025 Texas Instruments, Inc. -// -// Author: Baojun Xu <baojun.xu(a)ti.com> - -#include <linux/crc8.h> -#include <linux/firmware.h> -#include <linux/init.h> -#include <linux/interrupt.h> -#include <linux/module.h> -#include <linux/slab.h> -#include <linux/types.h> -#include <linux/unaligned.h> -#include <sound/pcm_params.h> -#include <sound/soc.h> -#include <sound/tas2781-dsp.h> -#include <sound/tlv.h> - -#include "tas2781-spi.h" - -#define OFFSET_ERROR_BIT BIT(31) - -#define ERROR_PRAM_CRCCHK 0x0000000 -#define ERROR_YRAM_CRCCHK 0x0000001 -#define PPC_DRIVER_CRCCHK 0x00000200 - -#define TAS2781_SA_COEFF_SWAP_REG TASDEVICE_REG(0, 0x35, 0x2c) -#define TAS2781_YRAM_BOOK1 140 -#define TAS2781_YRAM1_PAGE 42 -#define TAS2781_YRAM1_START_REG 88 - -#define TAS2781_YRAM2_START_PAGE 43 -#define TAS2781_YRAM2_END_PAGE 49 -#define TAS2781_YRAM2_START_REG 8 -#define TAS2781_YRAM2_END_REG 127 - -#define TAS2781_YRAM3_PAGE 50 -#define TAS2781_YRAM3_START_REG 8 -#define TAS2781_YRAM3_END_REG 27 - -/* should not include B0_P53_R44-R47 */ -#define TAS2781_YRAM_BOOK2 0 -#define TAS2781_YRAM4_START_PAGE 50 -#define TAS2781_YRAM4_END_PAGE 60 - -#define TAS2781_YRAM5_PAGE 61 -#define TAS2781_YRAM5_START_REG TAS2781_YRAM3_START_REG -#define TAS2781_YRAM5_END_REG TAS2781_YRAM3_END_REG - -#define TASDEVICE_MAXPROGRAM_NUM_KERNEL 5 -#define TASDEVICE_MAXCONFIG_NUM_KERNEL_MULTIPLE_AMPS 64 -#define TASDEVICE_MAXCONFIG_NUM_KERNEL 10 -#define MAIN_ALL_DEVICES_1X 0x01 -#define MAIN_DEVICE_A_1X 0x02 -#define MAIN_DEVICE_B_1X 0x03 -#define MAIN_DEVICE_C_1X 0x04 -#define MAIN_DEVICE_D_1X 0x05 -#define COEFF_DEVICE_A_1X 0x12 -#define COEFF_DEVICE_B_1X 0x13 -#define COEFF_DEVICE_C_1X 0x14 -#define COEFF_DEVICE_D_1X 0x15 -#define PRE_DEVICE_A_1X 0x22 -#define PRE_DEVICE_B_1X 0x23 -#define PRE_DEVICE_C_1X 0x24 -#define PRE_DEVICE_D_1X 0x25 -#define PRE_SOFTWARE_RESET_DEVICE_A 0x41 -#define PRE_SOFTWARE_RESET_DEVICE_B 0x42 -#define PRE_SOFTWARE_RESET_DEVICE_C 0x43 -#define PRE_SOFTWARE_RESET_DEVICE_D 0x44 -#define POST_SOFTWARE_RESET_DEVICE_A 0x45 -#define POST_SOFTWARE_RESET_DEVICE_B 0x46 -#define POST_SOFTWARE_RESET_DEVICE_C 0x47 -#define POST_SOFTWARE_RESET_DEVICE_D 0x48 - -struct tas_crc { - unsigned char offset; - unsigned char len; -}; - -struct blktyp_devidx_map { - unsigned char blktyp; - unsigned char dev_idx; -}; - -/* fixed m68k compiling issue: mapping table can save code field */ -static const struct blktyp_devidx_map ppc3_tas2781_mapping_table[] = { - { MAIN_ALL_DEVICES_1X, 0x80 }, - { MAIN_DEVICE_A_1X, 0x81 }, - { COEFF_DEVICE_A_1X, 0x81 }, - { PRE_DEVICE_A_1X, 0x81 }, - { PRE_SOFTWARE_RESET_DEVICE_A, 0xC1 }, - { POST_SOFTWARE_RESET_DEVICE_A, 0xC1 }, - { MAIN_DEVICE_B_1X, 0x82 }, - { COEFF_DEVICE_B_1X, 0x82 }, - { PRE_DEVICE_B_1X, 0x82 }, - { PRE_SOFTWARE_RESET_DEVICE_B, 0xC2 }, - { POST_SOFTWARE_RESET_DEVICE_B, 0xC2 }, - { MAIN_DEVICE_C_1X, 0x83 }, - { COEFF_DEVICE_C_1X, 0x83 }, - { PRE_DEVICE_C_1X, 0x83 }, - { PRE_SOFTWARE_RESET_DEVICE_C, 0xC3 }, - { POST_SOFTWARE_RESET_DEVICE_C, 0xC3 }, - { MAIN_DEVICE_D_1X, 0x84 }, - { COEFF_DEVICE_D_1X, 0x84 }, - { PRE_DEVICE_D_1X, 0x84 }, - { PRE_SOFTWARE_RESET_DEVICE_D, 0xC4 }, - { POST_SOFTWARE_RESET_DEVICE_D, 0xC4 }, -}; - -static const struct blktyp_devidx_map ppc3_mapping_table[] = { - { MAIN_ALL_DEVICES_1X, 0x80 }, - { MAIN_DEVICE_A_1X, 0x81 }, - { COEFF_DEVICE_A_1X, 0xC1 }, - { PRE_DEVICE_A_1X, 0xC1 }, - { MAIN_DEVICE_B_1X, 0x82 }, - { COEFF_DEVICE_B_1X, 0xC2 }, - { PRE_DEVICE_B_1X, 0xC2 }, - { MAIN_DEVICE_C_1X, 0x83 }, - { COEFF_DEVICE_C_1X, 0xC3 }, - { PRE_DEVICE_C_1X, 0xC3 }, - { MAIN_DEVICE_D_1X, 0x84 }, - { COEFF_DEVICE_D_1X, 0xC4 }, - { PRE_DEVICE_D_1X, 0xC4 }, -}; - -static const struct blktyp_devidx_map non_ppc3_mapping_table[] = { - { MAIN_ALL_DEVICES, 0x80 }, - { MAIN_DEVICE_A, 0x81 }, - { COEFF_DEVICE_A, 0xC1 }, - { PRE_DEVICE_A, 0xC1 }, - { MAIN_DEVICE_B, 0x82 }, - { COEFF_DEVICE_B, 0xC2 }, - { PRE_DEVICE_B, 0xC2 }, - { MAIN_DEVICE_C, 0x83 }, - { COEFF_DEVICE_C, 0xC3 }, - { PRE_DEVICE_C, 0xC3 }, - { MAIN_DEVICE_D, 0x84 }, - { COEFF_DEVICE_D, 0xC4 }, - { PRE_DEVICE_D, 0xC4 }, -}; - -/* - * Device support different configurations for different scene, - * like voice, music, calibration, was write in regbin file. - * Will be stored into tas_priv after regbin was loaded. - */ -static struct tasdevice_config_info *tasdevice_add_config( - struct tasdevice_priv *tas_priv, unsigned char *config_data, - unsigned int config_size, int *status) -{ - struct tasdevice_config_info *cfg_info; - struct tasdev_blk_data **bk_da; - unsigned int config_offset = 0; - unsigned int i; - - /* - * In most projects are many audio cases, such as music, handfree, - * receiver, games, audio-to-haptics, PMIC record, bypass mode, - * portrait, landscape, etc. Even in multiple audios, one or - * two of the chips will work for the special case, such as - * ultrasonic application. In order to support these variable-numbers - * of audio cases, flexible configs have been introduced in the - * DSP firmware. - */ - cfg_info = kzalloc(sizeof(*cfg_info), GFP_KERNEL); - if (!cfg_info) { - *status = -ENOMEM; - return NULL; - } - - if (tas_priv->rcabin.fw_hdr.binary_version_num >= 0x105) { - if ((config_offset + 64) > config_size) { - *status = -EINVAL; - dev_err(tas_priv->dev, "add conf: Out of boundary\n"); - goto config_err; - } - config_offset += 64; - } - - if ((config_offset + 4) > config_size) { - *status = -EINVAL; - dev_err(tas_priv->dev, "add config: Out of boundary\n"); - goto config_err; - } - - /* - * convert data[offset], data[offset + 1], data[offset + 2] and - * data[offset + 3] into host - */ - cfg_info->nblocks = get_unaligned_be32(&config_data[config_offset]); - config_offset += 4; - - /* - * Several kinds of dsp/algorithm firmwares can run on tas2781, - * the number and size of blk are not fixed and different among - * these firmwares. - */ - bk_da = cfg_info->blk_data = kcalloc(cfg_info->nblocks, - sizeof(*bk_da), GFP_KERNEL); - if (!bk_da) { - *status = -ENOMEM; - goto config_err; - } - cfg_info->real_nblocks = 0; - for (i = 0; i < cfg_info->nblocks; i++) { - if (config_offset + 12 > config_size) { - *status = -EINVAL; - dev_err(tas_priv->dev, - "%s: Out of boundary: i = %d nblocks = %u!\n", - __func__, i, cfg_info->nblocks); - goto block_err; - } - bk_da[i] = kzalloc(sizeof(*bk_da[i]), GFP_KERNEL); - if (!bk_da[i]) { - *status = -ENOMEM; - goto block_err; - } - - bk_da[i]->dev_idx = config_data[config_offset]; - config_offset++; - - bk_da[i]->block_type = config_data[config_offset]; - config_offset++; - - bk_da[i]->yram_checksum = - get_unaligned_be16(&config_data[config_offset]); - config_offset += 2; - bk_da[i]->block_size = - get_unaligned_be32(&config_data[config_offset]); - config_offset += 4; - - bk_da[i]->n_subblks = - get_unaligned_be32(&config_data[config_offset]); - - config_offset += 4; - - if (config_offset + bk_da[i]->block_size > config_size) { - *status = -EINVAL; - dev_err(tas_priv->dev, - "%s: Out of boundary: i = %d blks = %u!\n", - __func__, i, cfg_info->nblocks); - goto block_err; - } - /* instead of kzalloc+memcpy */ - bk_da[i]->regdata = kmemdup(&config_data[config_offset], - bk_da[i]->block_size, GFP_KERNEL); - if (!bk_da[i]->regdata) { - *status = -ENOMEM; - i++; - goto block_err; - } - - config_offset += bk_da[i]->block_size; - cfg_info->real_nblocks += 1; - } - - return cfg_info; -block_err: - for (int j = 0; j < i; j++) - kfree(bk_da[j]); - kfree(bk_da); -config_err: - kfree(cfg_info); - return NULL; -} - -/* Regbin file parser function. */ -int tasdevice_spi_rca_parser(void *context, const struct firmware *fmw) -{ - struct tasdevice_priv *tas_priv = context; - struct tasdevice_config_info **cfg_info; - struct tasdevice_rca_hdr *fw_hdr; - struct tasdevice_rca *rca; - unsigned int total_config_sz = 0; - int offset = 0, ret = 0, i; - unsigned char *buf; - - rca = &tas_priv->rcabin; - fw_hdr = &rca->fw_hdr; - if (!fmw || !fmw->data) { - dev_err(tas_priv->dev, "Failed to read %s\n", - tas_priv->rca_binaryname); - tas_priv->fw_state = TASDEVICE_DSP_FW_FAIL; - return -EINVAL; - } - buf = (unsigned char *)fmw->data; - fw_hdr->img_sz = get_unaligned_be32(&buf[offset]); - offset += 4; - if (fw_hdr->img_sz != fmw->size) { - dev_err(tas_priv->dev, - "File size not match, %d %u", (int)fmw->size, - fw_hdr->img_sz); - tas_priv->fw_state = TASDEVICE_DSP_FW_FAIL; - return -EINVAL; - } - - fw_hdr->checksum = get_unaligned_be32(&buf[offset]); - offset += 4; - fw_hdr->binary_version_num = get_unaligned_be32(&buf[offset]); - if (fw_hdr->binary_version_num < 0x103) { - dev_err(tas_priv->dev, "File version 0x%04x is too low", - fw_hdr->binary_version_num); - tas_priv->fw_state = TASDEVICE_DSP_FW_FAIL; - return -EINVAL; - } - offset += 4; - fw_hdr->drv_fw_version = get_unaligned_be32(&buf[offset]); - offset += 8; - fw_hdr->plat_type = buf[offset++]; - fw_hdr->dev_family = buf[offset++]; - fw_hdr->reserve = buf[offset++]; - fw_hdr->ndev = buf[offset++]; - if (offset + TASDEVICE_DEVICE_SUM > fw_hdr->img_sz) { - dev_err(tas_priv->dev, "rca_ready: Out of boundary!\n"); - tas_priv->fw_state = TASDEVICE_DSP_FW_FAIL; - return -EINVAL; - } - - for (i = 0; i < TASDEVICE_DEVICE_SUM; i++, offset++) - fw_hdr->devs[i] = buf[offset]; - - fw_hdr->nconfig = get_unaligned_be32(&buf[offset]); - offset += 4; - - for (i = 0; i < TASDEVICE_CONFIG_SUM; i++) { - fw_hdr->config_size[i] = get_unaligned_be32(&buf[offset]); - offset += 4; - total_config_sz += fw_hdr->config_size[i]; - } - - if (fw_hdr->img_sz - total_config_sz != (unsigned int)offset) { - dev_err(tas_priv->dev, "Bin file err %d - %d != %d!\n", - fw_hdr->img_sz, total_config_sz, (int)offset); - tas_priv->fw_state = TASDEVICE_DSP_FW_FAIL; - return -EINVAL; - } - - cfg_info = kcalloc(fw_hdr->nconfig, sizeof(*cfg_info), GFP_KERNEL); - if (!cfg_info) { - tas_priv->fw_state = TASDEVICE_DSP_FW_FAIL; - return -ENOMEM; - } - rca->cfg_info = cfg_info; - rca->ncfgs = 0; - for (i = 0; i < (int)fw_hdr->nconfig; i++) { - rca->ncfgs += 1; - cfg_info[i] = tasdevice_add_config(tas_priv, &buf[offset], - fw_hdr->config_size[i], &ret); - if (ret) { - tas_priv->fw_state = TASDEVICE_DSP_FW_FAIL; - return ret; - } - offset += (int)fw_hdr->config_size[i]; - } - - return ret; -} - -/* fixed m68k compiling issue: mapping table can save code field */ -static unsigned char map_dev_idx(struct tasdevice_fw *tas_fmw, - struct tasdev_blk *block) -{ - struct blktyp_devidx_map *p = - (struct blktyp_devidx_map *)non_ppc3_mapping_table; - struct tasdevice_dspfw_hdr *fw_hdr = &tas_fmw->fw_hdr; - struct tasdevice_fw_fixed_hdr *fw_fixed_hdr = &fw_hdr->fixed_hdr; - int i, n = ARRAY_SIZE(non_ppc3_mapping_table); - unsigned char dev_idx = 0; - - if (fw_fixed_hdr->ppcver >= PPC3_VERSION_TAS2781_BASIC_MIN) { - p = (struct blktyp_devidx_map *)ppc3_tas2781_mapping_table; - n = ARRAY_SIZE(ppc3_tas2781_mapping_table); - } else if (fw_fixed_hdr->ppcver >= PPC3_VERSION_BASE) { - p = (struct blktyp_devidx_map *)ppc3_mapping_table; - n = ARRAY_SIZE(ppc3_mapping_table); - } - - for (i = 0; i < n; i++) { - if (block->type == p[i].blktyp) { - dev_idx = p[i].dev_idx; - break; - } - } - - return dev_idx; -} - -/* Block parser function. */ -static int fw_parse_block_data_kernel(struct tasdevice_fw *tas_fmw, - struct tasdev_blk *block, const struct firmware *fmw, int offset) -{ - const unsigned char *data = fmw->data; - - if (offset + 16 > fmw->size) { - dev_err(tas_fmw->dev, "%s: File Size error\n", __func__); - return -EINVAL; - } - - /* - * Convert data[offset], data[offset + 1], data[offset + 2] and - * data[offset + 3] into host. - */ - block->type = get_unaligned_be32(&data[offset]); - offset += 4; - - block->is_pchksum_present = data[offset++]; - block->pchksum = data[offset++]; - block->is_ychksum_present = data[offset++]; - block->ychksum = data[offset++]; - block->blk_size = get_unaligned_be32(&data[offset]); - offset += 4; - block->nr_subblocks = get_unaligned_be32(&data[offset]); - offset += 4; - - /* - * Fixed m68k compiling issue: - * 1. mapping table can save code field. - * 2. storing the dev_idx as a member of block can reduce unnecessary - * time and system resource comsumption of dev_idx mapping every - * time the block data writing to the dsp. - */ - block->dev_idx = map_dev_idx(tas_fmw, block); - - if (offset + block->blk_size > fmw->size) { - dev_err(tas_fmw->dev, "%s: nSublocks error\n", __func__); - return -EINVAL; - } - /* instead of kzalloc+memcpy */ - block->data = kmemdup(&data[offset], block->blk_size, GFP_KERNEL); - if (!block->data) - return -ENOMEM; - - offset += block->blk_size; - - return offset; -} - -/* Data of block parser function. */ -static int fw_parse_data_kernel(struct tasdevice_fw *tas_fmw, - struct tasdevice_data *img_data, const struct firmware *fmw, - int offset) -{ - const unsigned char *data = fmw->data; - struct tasdev_blk *blk; - unsigned int i; - - if (offset + 4 > fmw->size) { - dev_err(tas_fmw->dev, "%s: File Size error\n", __func__); - return -EINVAL; - } - img_data->nr_blk = get_unaligned_be32(&data[offset]); - offset += 4; - - img_data->dev_blks = kcalloc(img_data->nr_blk, - sizeof(struct tasdev_blk), GFP_KERNEL); - if (!img_data->dev_blks) - return -ENOMEM; - - for (i = 0; i < img_data->nr_blk; i++) { - blk = &img_data->dev_blks[i]; - offset = fw_parse_block_data_kernel( - tas_fmw, blk, fmw, offset); - if (offset < 0) { - kfree(img_data->dev_blks); - return -EINVAL; - } - } - - return offset; -} - -/* Data of DSP program parser function. */ -static int fw_parse_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 72 unused byts */ - offset += 72; - - offset = fw_parse_data_kernel(tas_fmw, &program->dev_data, - fmw, offset); - if (offset < 0) - break; - } - - return offset; -} - -/* Data of DSP configurations parser function. */ -static int fw_parse_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 16 bytes */ - offset += 80; - - offset = fw_parse_data_kernel(tas_fmw, &config->dev_data, - fmw, offset); - if (offset < 0) - break; - } - - return offset; -} - -/* DSP firmware file header parser function for early PPC3 firmware binary. */ -static int fw_parse_variable_header_kernel(struct tasdevice_priv *tas_priv, - const struct firmware *fmw, int offset) -{ - struct tasdevice_fw *tas_fmw = tas_priv->fmw; - struct tasdevice_dspfw_hdr *fw_hdr = &tas_fmw->fw_hdr; - struct tasdevice_config *config; - struct tasdevice_prog *program; - const unsigned char *buf = fmw->data; - unsigned short max_confs; - unsigned int i; - - if (offset + 12 + 4 * TASDEVICE_MAXPROGRAM_NUM_KERNEL > fmw->size) { - dev_err(tas_priv->dev, "%s: File Size error\n", __func__); - return -EINVAL; - } - fw_hdr->device_family = get_unaligned_be16(&buf[offset]); - if (fw_hdr->device_family != 0) { - dev_err(tas_priv->dev, "%s:not TAS device\n", __func__); - return -EINVAL; - } - offset += 2; - fw_hdr->device = get_unaligned_be16(&buf[offset]); - if (fw_hdr->device >= TASDEVICE_DSP_TAS_MAX_DEVICE || - fw_hdr->device == 6) { - dev_err(tas_priv->dev, "Unsupported dev %d\n", fw_hdr->device); - return -EINVAL; - } - offset += 2; - - tas_fmw->nr_programs = get_unaligned_be32(&buf[offset]); - offset += 4; - - if (tas_fmw->nr_programs == 0 || - tas_fmw->nr_programs > TASDEVICE_MAXPROGRAM_NUM_KERNEL) { - dev_err(tas_priv->dev, "mnPrograms is invalid\n"); - return -EINVAL; - } - - tas_fmw->programs = kcalloc(tas_fmw->nr_programs, - sizeof(*tas_fmw->programs), GFP_KERNEL); - if (!tas_fmw->programs) - return -ENOMEM; - - for (i = 0; i < tas_fmw->nr_programs; i++) { - program = &tas_fmw->programs[i]; - program->prog_size = get_unaligned_be32(&buf[offset]); - offset += 4; - } - - /* Skip the unused prog_size */ - offset += 4 * (TASDEVICE_MAXPROGRAM_NUM_KERNEL - tas_fmw->nr_programs); - - tas_fmw->nr_configurations = get_unaligned_be32(&buf[offset]); - offset += 4; - - /* - * The max number of config in firmware greater than 4 pieces of - * tas2781s is different from the one lower than 4 pieces of - * tas2781s. - */ - max_confs = TASDEVICE_MAXCONFIG_NUM_KERNEL; - if (tas_fmw->nr_configurations == 0 || - tas_fmw->nr_configurations > max_confs) { - dev_err(tas_priv->dev, "%s: Conf is invalid\n", __func__); - kfree(tas_fmw->programs); - return -EINVAL; - } - - if (offset + 4 * max_confs > fmw->size) { - dev_err(tas_priv->dev, "%s: mpConfigurations err\n", __func__); - kfree(tas_fmw->programs); - return -EINVAL; - } - - tas_fmw->configs = kcalloc(tas_fmw->nr_configurations, - sizeof(*tas_fmw->configs), GFP_KERNEL); - if (!tas_fmw->configs) { - kfree(tas_fmw->programs); - return -ENOMEM; - } - - for (i = 0; i < tas_fmw->nr_programs; i++) { - config = &tas_fmw->configs[i]; - config->cfg_size = get_unaligned_be32(&buf[offset]); - offset += 4; - } - - /* Skip the unused configs */ - offset += 4 * (max_confs - tas_fmw->nr_programs); - - return offset; -} - -/* - * In sub-block data, have three type sub-block: - * 1. Single byte write. - * 2. Multi-byte write. - * 3. Delay. - * 4. Bits update. - * This function perform single byte write to device. - */ -static int tasdevice_single_byte_wr(void *context, int dev_idx, - unsigned char *data, int sublocksize) -{ - struct tasdevice_priv *tas_priv = context; - unsigned short len = get_unaligned_be16(&data[2]); - int i, subblk_offset, rc; - - subblk_offset = 4; - if (subblk_offset + 4 * len > sublocksize) { - dev_err(tas_priv->dev, "process_block: Out of boundary\n"); - return 0; - } - - for (i = 0; i < len; i++) { - if (dev_idx == (tas_priv->index + 1) || dev_idx == 0) { - rc = tasdevice_spi_dev_write(tas_priv, - TASDEVICE_REG(data[subblk_offset], - data[subblk_offset + 1], - data[subblk_offset + 2]), - data[subblk_offset + 3]); - if (rc < 0) { - dev_err(tas_priv->dev, - "process_block: single write error\n"); - subblk_offset |= OFFSET_ERROR_BIT; - } - } - subblk_offset += 4; - } - - return subblk_offset; -} - -/* - * In sub-block data, have three type sub-block: - * 1. Single byte write. - * 2. Multi-byte write. - * 3. Delay. - * 4. Bits update. - * This function perform multi-write to device. - */ -static int tasdevice_burst_wr(void *context, int dev_idx, unsigned char *data, - int sublocksize) -{ - struct tasdevice_priv *tas_priv = context; - unsigned short len = get_unaligned_be16(&data[2]); - int subblk_offset, rc; - - subblk_offset = 4; - if (subblk_offset + 4 + len > sublocksize) { - dev_err(tas_priv->dev, "%s: BST Out of boundary\n", __func__); - subblk_offset |= OFFSET_ERROR_BIT; - } - if (len % 4) { - dev_err(tas_priv->dev, "%s:Bst-len(%u)not div by 4\n", - __func__, len); - subblk_offset |= OFFSET_ERROR_BIT; - } - - if (dev_idx == (tas_priv->index + 1) || dev_idx == 0) { - rc = tasdevice_spi_dev_bulk_write(tas_priv, - TASDEVICE_REG(data[subblk_offset], - data[subblk_offset + 1], - data[subblk_offset + 2]), - &data[subblk_offset + 4], len); - if (rc < 0) { - dev_err(tas_priv->dev, "%s: bulk_write error = %d\n", - __func__, rc); - subblk_offset |= OFFSET_ERROR_BIT; - } - } - subblk_offset += (len + 4); - - return subblk_offset; -} - -/* Just delay for ms.*/ -static int tasdevice_delay(void *context, int dev_idx, unsigned char *data, - int sublocksize) -{ - struct tasdevice_priv *tas_priv = context; - unsigned int sleep_time, subblk_offset = 2; - - if (subblk_offset + 2 > sublocksize) { - dev_err(tas_priv->dev, "%s: delay Out of boundary\n", - __func__); - subblk_offset |= OFFSET_ERROR_BIT; - } - if (dev_idx == (tas_priv->index + 1) || dev_idx == 0) { - sleep_time = get_unaligned_be16(&data[2]) * 1000; - fsleep(sleep_time); - } - subblk_offset += 2; - - return subblk_offset; -} - -/* - * In sub-block data, have three type sub-block: - * 1. Single byte write. - * 2. Multi-byte write. - * 3. Delay. - * 4. Bits update. - * This function perform bits update. - */ -static int tasdevice_field_wr(void *context, int dev_idx, unsigned char *data, - int sublocksize) -{ - struct tasdevice_priv *tas_priv = context; - int rc, subblk_offset = 2; - - if (subblk_offset + 6 > sublocksize) { - dev_err(tas_priv->dev, "%s: bit write Out of boundary\n", - __func__); - subblk_offset |= OFFSET_ERROR_BIT; - } - if (dev_idx == (tas_priv->index + 1) || dev_idx == 0) { - rc = tasdevice_spi_dev_update_bits(tas_priv, - TASDEVICE_REG(data[subblk_offset + 2], - data[subblk_offset + 3], - data[subblk_offset + 4]), - data[subblk_offset + 1], - data[subblk_offset + 5]); - if (rc < 0) { - dev_err(tas_priv->dev, "%s: update_bits error = %d\n", - __func__, rc); - subblk_offset |= OFFSET_ERROR_BIT; - } - } - subblk_offset += 6; - - return subblk_offset; -} - -/* Data block process function. */ -static int tasdevice_process_block(void *context, unsigned char *data, - unsigned char dev_idx, int sublocksize) -{ - struct tasdevice_priv *tas_priv = context; - int blktyp = dev_idx & 0xC0, subblk_offset; - unsigned char subblk_typ = data[1]; - - switch (subblk_typ) { - case TASDEVICE_CMD_SING_W: - subblk_offset = tasdevice_single_byte_wr(tas_priv, - dev_idx & 0x3f, data, sublocksize); - break; - case TASDEVICE_CMD_BURST: - subblk_offset = tasdevice_burst_wr(tas_priv, - dev_idx & 0x3f, data, sublocksize); - break; - case TASDEVICE_CMD_DELAY: - subblk_offset = tasdevice_delay(tas_priv, - dev_idx & 0x3f, data, sublocksize); - break; - case TASDEVICE_CMD_FIELD_W: - subblk_offset = tasdevice_field_wr(tas_priv, - dev_idx & 0x3f, data, sublocksize); - break; - default: - subblk_offset = 2; - break; - } - if (((subblk_offset & OFFSET_ERROR_BIT) != 0) && blktyp != 0) { - if (blktyp == 0x80) { - tas_priv->cur_prog = -1; - tas_priv->cur_conf = -1; - } else - tas_priv->cur_conf = -1; - } - subblk_offset &= ~OFFSET_ERROR_BIT; - - return subblk_offset; -} - -/* - * Device support different configurations for different scene, - * this function was used for choose different config. - */ -void tasdevice_spi_select_cfg_blk(void *pContext, int conf_no, - unsigned char block_type) -{ - struct tasdevice_priv *tas_priv = pContext; - struct tasdevice_rca *rca = &tas_priv->rcabin; - struct tasdevice_config_info **cfg_info = rca->cfg_info; - struct tasdev_blk_data **blk_data; - unsigned int j, k; - - if (conf_no >= rca->ncfgs || conf_no < 0 || !cfg_info) { - dev_err(tas_priv->dev, "conf_no should be not more than %u\n", - rca->ncfgs); - return; - } - blk_data = cfg_info[conf_no]->blk_data; - - for (j = 0; j < cfg_info[conf_no]->real_nblocks; j++) { - unsigned int length = 0, rc = 0; - - if (block_type > 5 || block_type < 2) { - dev_err(tas_priv->dev, - "block_type should be in range from 2 to 5\n"); - break; - } - if (block_type != blk_data[j]->block_type) - continue; - - for (k = 0; k < blk_data[j]->n_subblks; k++) { - tas_priv->is_loading = true; - - rc = tasdevice_process_block(tas_priv, - blk_data[j]->regdata + length, - blk_data[j]->dev_idx, - blk_data[j]->block_size - length); - length += rc; - if (blk_data[j]->block_size < length) { - dev_err(tas_priv->dev, - "%s: %u %u out of boundary\n", - __func__, length, - blk_data[j]->block_size); - break; - } - } - if (length != blk_data[j]->block_size) - dev_err(tas_priv->dev, "%s: %u %u size is not same\n", - __func__, length, blk_data[j]->block_size); - } -} - -/* Block process function. */ -static int tasdevice_load_block_kernel( - struct tasdevice_priv *tasdevice, struct tasdev_blk *block) -{ - const unsigned int blk_size = block->blk_size; - unsigned char *data = block->data; - unsigned int i, length; - - for (i = 0, length = 0; i < block->nr_subblocks; i++) { - int rc = tasdevice_process_block(tasdevice, data + length, - block->dev_idx, blk_size - length); - if (rc < 0) { - dev_err(tasdevice->dev, - "%s: %u %u sublock write error\n", - __func__, length, blk_size); - return rc; - } - length += rc; - if (blk_size < length) { - dev_err(tasdevice->dev, "%s: %u %u out of boundary\n", - __func__, length, blk_size); - rc = -ENOMEM; - return rc; - } - } - - return 0; -} - -/* DSP firmware file header parser function. */ -static int fw_parse_variable_hdr(struct tasdevice_priv *tas_priv, - struct tasdevice_dspfw_hdr *fw_hdr, - const struct firmware *fmw, int offset) -{ - const unsigned char *buf = fmw->data; - int len = strlen((char *)&buf[offset]); - - len++; - - if (offset + len + 8 > fmw->size) { - dev_err(tas_priv->dev, "%s: File Size error\n", __func__); - return -EINVAL; - } - - offset += len; - - fw_hdr->device_family = get_unaligned_be32(&buf[offset]); - if (fw_hdr->device_family != 0) { - dev_err(tas_priv->dev, "%s: not TAS device\n", __func__); - return -EINVAL; - } - offset += 4; - - fw_hdr->device = get_unaligned_be32(&buf[offset]); - if (fw_hdr->device >= TASDEVICE_DSP_TAS_MAX_DEVICE || - fw_hdr->device == 6) { - dev_err(tas_priv->dev, "Unsupported dev %d\n", fw_hdr->device); - return -EINVAL; - } - offset += 4; - fw_hdr->ndev = 1; - - return offset; -} - -/* DSP firmware file header parser function for size variabled header. */ -static int fw_parse_variable_header_git(struct tasdevice_priv - *tas_priv, const struct firmware *fmw, int offset) -{ - struct tasdevice_fw *tas_fmw = tas_priv->fmw; - struct tasdevice_dspfw_hdr *fw_hdr = &tas_fmw->fw_hdr; - - offset = fw_parse_variable_hdr(tas_priv, fw_hdr, fmw, offset); - - return offset; -} - -/* DSP firmware file block parser function. */ -static int fw_parse_block_data(struct tasdevice_fw *tas_fmw, - struct tasdev_blk *block, const struct firmware *fmw, int offset) -{ - unsigned char *data = (unsigned char *)fmw->data; - int n; - - if (offset + 8 > fmw->size) { - dev_err(tas_fmw->dev, "%s: Type error\n", __func__); - return -EINVAL; - } - block->type = get_unaligned_be32(&data[offset]); - offset += 4; - - if (tas_fmw->fw_hdr.fixed_hdr.drv_ver >= PPC_DRIVER_CRCCHK) { - if (offset + 8 > fmw->size) { - dev_err(tas_fmw->dev, "PChkSumPresent error\n"); - return -EINVAL; - } - block->is_pchksum_present = data[offset]; - offset++; - - block->pchksum = data[offset]; - offset++; - - block->is_ychksum_present = data[offset]; - offset++; - - block->ychksum = data[offset]; - offset++; - } else { - block->is_pchksum_present = 0; - block->is_ychksum_present = 0; - } - - block->nr_cmds = get_unaligned_be32(&data[offset]); - offset += 4; - - n = block->nr_cmds * 4; - if (offset + n > fmw->size) { - dev_err(tas_fmw->dev, - "%s: File Size(%lu) error offset = %d n = %d\n", - __func__, (unsigned long)fmw->size, offset, n); - return -EINVAL; - } - /* instead of kzalloc+memcpy */ - block->data = kmemdup(&data[offset], n, GFP_KERNEL); - if (!block->data) - return -ENOMEM; - - offset += n; - - return offset; -} - -/* - * When parsing error occurs, all the memory resource will be released - * in the end of tasdevice_rca_ready. - */ -static int fw_parse_data(struct tasdevice_fw *tas_fmw, - struct tasdevice_data *img_data, const struct firmware *fmw, - int offset) -{ - const unsigned char *data = (unsigned char *)fmw->data; - struct tasdev_blk *blk; - unsigned int i, n; - - if (offset + 64 > fmw->size) { - dev_err(tas_fmw->dev, "%s: Name error\n", __func__); - return -EINVAL; - } - memcpy(img_data->name, &data[offset], 64); - offset += 64; - - n = strlen((char *)&data[offset]); - n++; - if (offset + n + 2 > fmw->size) { - dev_err(tas_fmw->dev, "%s: Description error\n", __func__); - return -EINVAL; - } - offset += n; - img_data->nr_blk = get_unaligned_be16(&data[offset]); - offset += 2; - - img_data->dev_blks = kcalloc(img_data->nr_blk, - sizeof(*img_data->dev_blks), GFP_KERNEL); - if (!img_data->dev_blks) - return -ENOMEM; - - for (i = 0; i < img_data->nr_blk; i++) { - blk = &img_data->dev_blks[i]; - offset = fw_parse_block_data(tas_fmw, blk, fmw, offset); - if (offset < 0) - return -EINVAL; - } - - return offset; -} - -/* - * When parsing error occurs, all the memory resource will be released - * in the end of tasdevice_rca_ready. - */ -static int fw_parse_program_data(struct tasdevice_priv *tas_priv, - struct tasdevice_fw *tas_fmw, const struct firmware *fmw, int offset) -{ - unsigned char *buf = (unsigned char *)fmw->data; - struct tasdevice_prog *program; - int i; - - if (offset + 2 > fmw->size) { - dev_err(tas_priv->dev, "%s: File Size error\n", __func__); - return -EINVAL; - } - tas_fmw->nr_programs = get_unaligned_be16(&buf[offset]); - offset += 2; - - if (tas_fmw->nr_programs == 0) { - /* Not error in calibration Data file, return directly */ - dev_dbg(tas_priv->dev, "%s: No Programs data, maybe calbin\n", - __func__); - return offset; - } - - tas_fmw->programs = - kcalloc(tas_fmw->nr_programs, sizeof(*tas_fmw->programs), - GFP_KERNEL); - if (!tas_fmw->programs) - return -ENOMEM; - - for (i = 0; i < tas_fmw->nr_programs; i++) { - int n = 0; - - program = &tas_fmw->programs[i]; - if (offset + 64 > fmw->size) { - dev_err(tas_priv->dev, "%s: mpName error\n", __func__); - return -EINVAL; - } - offset += 64; - - n = strlen((char *)&buf[offset]); - /* skip '\0' and 5 unused bytes */ - n += 6; - if (offset + n > fmw->size) { - dev_err(tas_priv->dev, "Description err\n"); - return -EINVAL; - } - - offset += n; - - offset = fw_parse_data(tas_fmw, &program->dev_data, fmw, - offset); - if (offset < 0) - return offset; - } - - return offset; -} - -/* - * When parsing error occurs, all the memory resource will be released - * in the end of tasdevice_rca_ready. - */ -static int fw_parse_configuration_data(struct tasdevice_priv *tas_priv, - struct tasdevice_fw *tas_fmw, const struct firmware *fmw, int offset) -{ - unsigned char *data = (unsigned char *)fmw->data; - struct tasdevice_config *config; - unsigned int i, n; - - if (offset + 2 > fmw->size) { - dev_err(tas_priv->dev, "%s: File Size error\n", __func__); - return -EINVAL; - } - tas_fmw->nr_configurations = get_unaligned_be16(&data[offset]); - offset += 2; - - if (tas_fmw->nr_configurations == 0) { - dev_err(tas_priv->dev, "%s: Conf is zero\n", __func__); - /* Not error for calibration Data file, return directly */ - return offset; - } - tas_fmw->configs = kcalloc(tas_fmw->nr_configurations, - sizeof(*tas_fmw->configs), GFP_KERNEL); - if (!tas_fmw->configs) - return -ENOMEM; - for (i = 0; i < tas_fmw->nr_configurations; i++) { - config = &tas_fmw->configs[i]; - if (offset + 64 > fmw->size) { - dev_err(tas_priv->dev, "File Size err\n"); - return -EINVAL; - } - memcpy(config->name, &data[offset], 64); - offset += 64; - - n = strlen((char *)&data[offset]); - n += 15; - if (offset + n > fmw->size) { - dev_err(tas_priv->dev, "Description err\n"); - return -EINVAL; - } - offset += n; - offset = fw_parse_data(tas_fmw, &config->dev_data, - fmw, offset); - if (offset < 0) - break; - } - - return offset; -} - -/* yram5 page check. */ -static bool check_inpage_yram_rg(struct tas_crc *cd, - unsigned char reg, unsigned char len) -{ - bool in = false; - - if (reg <= TAS2781_YRAM5_END_REG && - reg >= TAS2781_YRAM5_START_REG) { - if (reg + len > TAS2781_YRAM5_END_REG) - cd->len = TAS2781_YRAM5_END_REG - reg + 1; - else - cd->len = len; - cd->offset = reg; - in = true; - } else if (reg < TAS2781_YRAM5_START_REG) { - if (reg + len > TAS2781_YRAM5_START_REG) { - cd->offset = TAS2781_YRAM5_START_REG; - cd->len = len - TAS2781_YRAM5_START_REG + reg; - in = true; - } - } - - return in; -} - -/* DSP firmware yram block check. */ -static bool check_inpage_yram_bk1(struct tas_crc *cd, - unsigned char page, unsigned char reg, unsigned char len) -{ - bool in = false; - - if (page == TAS2781_YRAM1_PAGE) { - if (reg >= TAS2781_YRAM1_START_REG) { - cd->offset = reg; - cd->len = len; - in = true; - } else if (reg + len > TAS2781_YRAM1_START_REG) { - cd->offset = TAS2781_YRAM1_START_REG; - cd->len = len - TAS2781_YRAM1_START_REG + reg; - in = true; - } - } else if (page == TAS2781_YRAM3_PAGE) { - in = check_inpage_yram_rg(cd, reg, len); - } - - return in; -} - -/* - * Return Code: - * true -- the registers are in the inpage yram - * false -- the registers are NOT in the inpage yram - */ -static bool check_inpage_yram(struct tas_crc *cd, unsigned char book, - unsigned char page, unsigned char reg, unsigned char len) -{ - bool in = false; - - if (book == TAS2781_YRAM_BOOK1) - in = check_inpage_yram_bk1(cd, page, reg, len); - else if (book == TAS2781_YRAM_BOOK2 && page == TAS2781_YRAM5_PAGE) - in = check_inpage_yram_rg(cd, reg, len); - - return in; -} - -/* yram4 page check. */ -static bool check_inblock_yram_bk(struct tas_crc *cd, - unsigned char page, unsigned char reg, unsigned char len) -{ - bool in = false; - - if ((page >= TAS2781_YRAM4_START_PAGE && - page <= TAS2781_YRAM4_END_PAGE) || - (page >= TAS2781_YRAM2_START_PAGE && - page <= TAS2781_YRAM2_END_PAGE)) { - if (reg <= TAS2781_YRAM2_END_REG && - reg >= TAS2781_YRAM2_START_REG) { - cd->offset = reg; - cd->len = len; - in = true; - } else if (reg < TAS2781_YRAM2_START_REG) { - if (reg + len - 1 >= TAS2781_YRAM2_START_REG) { - cd->offset = TAS2781_YRAM2_START_REG; - cd->len = reg + len - TAS2781_YRAM2_START_REG; - in = true; - } - } - } - - return in; -} - -/* - * Return Code: - * true -- the registers are in the inblock yram - * false -- the registers are NOT in the inblock yram - */ -static bool check_inblock_yram(struct tas_crc *cd, unsigned char book, - unsigned char page, unsigned char reg, unsigned char len) -{ - bool in = false; - - if (book == TAS2781_YRAM_BOOK1 || book == TAS2781_YRAM_BOOK2) - in = check_inblock_yram_bk(cd, page, reg, len); - - return in; -} - -/* yram page check. */ -static bool check_yram(struct tas_crc *cd, unsigned char book, - unsigned char page, unsigned char reg, unsigned char len) -{ - bool in; - - in = check_inpage_yram(cd, book, page, reg, len); - if (!in) - in = check_inblock_yram(cd, book, page, reg, len); - - return in; -} - -/* Checksum for data block. */ -static int tasdev_multibytes_chksum(struct tasdevice_priv *tasdevice, - unsigned char book, unsigned char page, - unsigned char reg, unsigned int len) -{ - struct tas_crc crc_data; - unsigned char crc_chksum = 0; - unsigned char nBuf1[128]; - int ret = 0, i; - bool in; - - if ((reg + len - 1) > 127) { - ret = -EINVAL; - dev_err(tasdevice->dev, "firmware error\n"); - goto end; - } - - if ((book == TASDEVICE_BOOK_ID(TAS2781_SA_COEFF_SWAP_REG)) && - (page == TASDEVICE_PAGE_ID(TAS2781_SA_COEFF_SWAP_REG)) && - (reg == TASDEVICE_REG_ID(TAS2781_SA_COEFF_SWAP_REG)) && - (len == 4)) { - /* DSP swap command, pass */ - ret = 0; - goto end; - } - - in = check_yram(&crc_data, book, page, reg, len); - if (!in) - goto end; - - if (len == 1) { - dev_err(tasdevice->dev, "firmware error\n"); - ret = -EINVAL; - goto end; - } - - ret = tasdevice_spi_dev_bulk_read(tasdevice, - TASDEVICE_REG(book, page, crc_data.offset), - nBuf1, crc_data.len); - if (ret < 0) - goto end; - - for (i = 0; i < crc_data.len; i++) { - if ((book == TASDEVICE_BOOK_ID(TAS2781_SA_COEFF_SWAP_REG)) && - (page == TASDEVICE_PAGE_ID(TAS2781_SA_COEFF_SWAP_REG)) && - ((i + crc_data.offset) >= - TASDEVICE_REG_ID(TAS2781_SA_COEFF_SWAP_REG)) && - ((i + crc_data.offset) <= - (TASDEVICE_REG_ID(TAS2781_SA_COEFF_SWAP_REG) + 4))) - /* DSP swap command, bypass */ - continue; - else - crc_chksum += crc8(tasdevice->crc8_lkp_tbl, &nBuf1[i], - 1, 0); - } - - ret = crc_chksum; - -end: - return ret; -} - -/* Checksum for single register. */ -static int do_singlereg_checksum(struct tasdevice_priv *tasdevice, - unsigned char book, unsigned char page, - unsigned char reg, unsigned char val) -{ - struct tas_crc crc_data; - unsigned int nData1; - int ret = 0; - bool in; - - /* DSP swap command, pass */ - if ((book == TASDEVICE_BOOK_ID(TAS2781_SA_COEFF_SWAP_REG)) && - (page == TASDEVICE_PAGE_ID(TAS2781_SA_COEFF_SWAP_REG)) && - (reg >= TASDEVICE_REG_ID(TAS2781_SA_COEFF_SWAP_REG)) && - (reg <= (TASDEVICE_REG_ID(TAS2781_SA_COEFF_SWAP_REG) + 4))) - return 0; - - in = check_yram(&crc_data, book, page, reg, 1); - if (!in) - return 0; - ret = tasdevice_spi_dev_read(tasdevice, - TASDEVICE_REG(book, page, reg), &nData1); - if (ret < 0) - return ret; - - if (nData1 != val) { - dev_err(tasdevice->dev, - "B[0x%x]P[0x%x]R[0x%x] W[0x%x], R[0x%x]\n", - book, page, reg, val, nData1); - tasdevice->err_code |= ERROR_YRAM_CRCCHK; - return -EAGAIN; - } - - ret = crc8(tasdevice->crc8_lkp_tbl, &val, 1, 0); - - return ret; -} - -/* Block type check. */ -static void set_err_prg_cfg(unsigned int type, struct tasdevice_priv *p) -{ - if ((type == MAIN_ALL_DEVICES) || (type == MAIN_DEVICE_A) || - (type == MAIN_DEVICE_B) || (type == MAIN_DEVICE_C) || - (type == MAIN_DEVICE_D)) - p->cur_prog = -1; - else - p->cur_conf = -1; -} - -/* Checksum for data bytes. */ -static int tasdev_bytes_chksum(struct tasdevice_priv *tas_priv, - struct tasdev_blk *block, unsigned char book, - unsigned char page, unsigned char reg, unsigned int len, - unsigned char val, unsigned char *crc_chksum) -{ - int ret; - - if (len > 1) - ret = tasdev_multibytes_chksum(tas_priv, book, page, reg, - len); - else - ret = do_singlereg_checksum(tas_priv, book, page, reg, val); - - if (ret > 0) { - *crc_chksum += ret; - goto end; - } - - if (ret != -EAGAIN) - goto end; - - block->nr_retry--; - if (block->nr_retry > 0) - goto end; - - set_err_prg_cfg(block->type, tas_priv); - -end: - return ret; -} - -/* Multi-data byte write. */ -static int tasdev_multibytes_wr(struct tasdevice_priv *tas_priv, - struct tasdev_blk *block, unsigned char book, - unsigned char page, unsigned char reg, unsigned char *data, - unsigned int len, unsigned int *nr_cmds, - unsigned char *crc_chksum) -{ - int ret; - - if (len > 1) { - ret = tasdevice_spi_dev_bulk_write(tas_priv, - TASDEVICE_REG(book, page, reg), data + 3, len); - if (ret < 0) - return ret; - if (block->is_ychksum_present) - ret = tasdev_bytes_chksum(tas_priv, block, - book, page, reg, len, 0, crc_chksum); - } else { - ret = tasdevice_spi_dev_write(tas_priv, - TASDEVICE_REG(book, page, reg), data[3]); - if (ret < 0) - return ret; - if (block->is_ychksum_present) - ret = tasdev_bytes_chksum(tas_priv, block, book, - page, reg, 1, data[3], crc_chksum); - } - - if (!block->is_ychksum_present || ret >= 0) { - *nr_cmds += 1; - if (len >= 2) - *nr_cmds += ((len - 2) / 4) + 1; - } - - return ret; -} - -/* Checksum for block. */ -static int tasdev_block_chksum(struct tasdevice_priv *tas_priv, - struct tasdev_blk *block) -{ - unsigned int nr_value; - int ret; - - ret = tasdevice_spi_dev_read(tas_priv, TASDEVICE_CHECKSUM, &nr_value); - if (ret < 0) { - dev_err(tas_priv->dev, "%s: read error %d.\n", __func__, ret); - set_err_prg_cfg(block->type, tas_priv); - return ret; - } - - if ((nr_value & 0xff) != block->pchksum) { - dev_err(tas_priv->dev, "%s: PChkSum err %d ", __func__, ret); - dev_err(tas_priv->dev, "PChkSum = 0x%x, Reg = 0x%x\n", - block->pchksum, (nr_value & 0xff)); - tas_priv->err_code |= ERROR_PRAM_CRCCHK; - ret = -EAGAIN; - block->nr_retry--; - - if (block->nr_retry <= 0) - set_err_prg_cfg(block->type, tas_priv); - } else { - tas_priv->err_code &= ~ERROR_PRAM_CRCCHK; - } - - return ret; -} - -/* Firmware block load function. */ -static int tasdev_load_blk(struct tasdevice_priv *tas_priv, - struct tasdev_blk *block) -{ - unsigned int sleep_time, len, nr_cmds; - unsigned char offset, book, page, val; - unsigned char *data = block->data; - unsigned char crc_chksum = 0; - int ret = 0; - - while (block->nr_retry > 0) { - if (block->is_pchksum_present) { - ret = tasdevice_spi_dev_write(tas_priv, - TASDEVICE_CHECKSUM, 0); - if (ret < 0) - break; - } - - if (block->is_ychksum_present) - crc_chksum = 0; - - nr_cmds = 0; - - while (nr_cmds < block->nr_cmds) { - data = block->data + nr_cmds * 4; - - book = data[0]; - page = data[1]; - offset = data[2]; - val = data[3]; - - nr_cmds++; - /* Single byte write */ - if (offset <= 0x7F) { - ret = tasdevice_spi_dev_write(tas_priv, - TASDEVICE_REG(book, page, offset), - val); - if (ret < 0) - break; - if (block->is_ychksum_present) { - ret = tasdev_bytes_chksum(tas_priv, - block, book, page, offset, - 1, val, &crc_chksum); - if (ret < 0) - break; - } - continue; - } - /* sleep command */ - if (offset == 0x81) { - /* book -- data[0] page -- data[1] */ - sleep_time = ((book << 8) + page)*1000; - fsleep(sleep_time); - continue; - } - /* Multiple bytes write */ - if (offset == 0x85) { - data += 4; - len = (book << 8) + page; - book = data[0]; - page = data[1]; - offset = data[2]; - ret = tasdev_multibytes_wr(tas_priv, - block, book, page, offset, data, - len, &nr_cmds, &crc_chksum); - if (ret < 0) - break; - } - } - if (ret == -EAGAIN) { - if (block->nr_retry > 0) - continue; - } else if (ret < 0) { - /* err in current device, skip it */ - break; - } - - if (block->is_pchksum_present) { - ret = tasdev_block_chksum(tas_priv, block); - if (ret == -EAGAIN) { - if (block->nr_retry > 0) - continue; - } else if (ret < 0) { - /* err in current device, skip it */ - break; - } - } - - if (block->is_ychksum_present) { - /* TBD, open it when FW ready */ - dev_err(tas_priv->dev, - "Blk YChkSum: FW = 0x%x, YCRC = 0x%x\n", - block->ychksum, crc_chksum); - - tas_priv->err_code &= - ~ERROR_YRAM_CRCCHK; - ret = 0; - } - /* skip current blk */ - break; - } - - return ret; -} - -/* Firmware block load function. */ -static int tasdevice_load_block(struct tasdevice_priv *tas_priv, - struct tasdev_blk *block) -{ - int ret = 0; - - block->nr_retry = 6; - if (tas_priv->is_loading == false) - return 0; - ret = tasdev_load_blk(tas_priv, block); - if (ret < 0) - dev_err(tas_priv->dev, "Blk (%d) load error\n", block->type); - - return ret; -} - -/* - * Select firmware binary parser & load callback functions by ppc3 version - * and firmware binary version. - */ -static int dspfw_default_callback(struct tasdevice_priv *tas_priv, - unsigned int drv_ver, unsigned int ppcver) -{ - int rc = 0; - - if (drv_ver == 0x100) { - if (ppcver >= PPC3_VERSION_BASE) { - tas_priv->fw_parse_variable_header = - fw_parse_variable_header_kernel; - tas_priv->fw_parse_program_data = - fw_parse_program_data_kernel; - tas_priv->fw_parse_configuration_data = - fw_parse_configuration_data_kernel; - tas_priv->tasdevice_load_block = - tasdevice_load_block_kernel; - } else if (ppcver == 0x00) { - tas_priv->fw_parse_variable_header = - fw_parse_variable_header_git; - tas_priv->fw_parse_program_data = - fw_parse_program_data; - tas_priv->fw_parse_configuration_data = - fw_parse_configuration_data; - tas_priv->tasdevice_load_block = - tasdevice_load_block; - } else { - dev_err(tas_priv->dev, - "Wrong PPCVer :0x%08x\n", ppcver); - rc = -EINVAL; - } - } else { - dev_err(tas_priv->dev, "Wrong DrvVer : 0x%02x\n", drv_ver); - rc = -EINVAL; - } - - return rc; -} - -/* DSP firmware binary file header parser function. */ -static int fw_parse_header(struct tasdevice_priv *tas_priv, - struct tasdevice_fw *tas_fmw, const struct firmware *fmw, int offset) -{ - struct tasdevice_dspfw_hdr *fw_hdr = &tas_fmw->fw_hdr; - struct tasdevice_fw_fixed_hdr *fw_fixed_hdr = &fw_hdr->fixed_hdr; - static const unsigned char magic_number[] = {0x35, 0x35, 0x35, 0x32, }; - const unsigned char *buf = (unsigned char *)fmw->data; - - if (offset + 92 > fmw->size) { - dev_err(tas_priv->dev, "%s: File Size error\n", __func__); - offset = -EINVAL; - goto out; - } - if (memcmp(&buf[offset], magic_number, 4)) { - dev_err(tas_priv->dev, "%s: Magic num NOT match\n", __func__); - offset = -EINVAL; - goto out; - } - offset += 4; - - /* - * Convert data[offset], data[offset + 1], data[offset + 2] and - * data[offset + 3] into host - */ - fw_fixed_hdr->fwsize = get_unaligned_be32(&buf[offset]); - offset += 4; - if (fw_fixed_hdr->fwsize != fmw->size) { - dev_err(tas_priv->dev, "File size not match, %lu %u", - (unsigned long)fmw->size, fw_fixed_hdr->fwsize); - offset = -EINVAL; - goto out; - } - offset += 4; - fw_fixed_hdr->ppcver = get_unaligned_be32(&buf[offset]); - offset += 8; - fw_fixed_hdr->drv_ver = get_unaligned_be32(&buf[offset]); - offset += 72; - -out: - return offset; -} - -/* DSP firmware binary file parser function. */ -static int tasdevice_dspfw_ready(const struct firmware *fmw, void *context) -{ - struct tasdevice_priv *tas_priv = context; - struct tasdevice_fw_fixed_hdr *fw_fixed_hdr; - struct tasdevice_fw *tas_fmw; - int offset = 0, ret = 0; - - if (!fmw || !fmw->data) { - dev_err(tas_priv->dev, "%s: Failed to read firmware %s\n", - __func__, tas_priv->coef_binaryname); - return -EINVAL; - } - - tas_priv->fmw = kzalloc(sizeof(*tas_priv->fmw), GFP_KERNEL); - if (!tas_priv->fmw) - return -ENOMEM; - tas_fmw = tas_priv->fmw; - tas_fmw->dev = tas_priv->dev; - offset = fw_parse_header(tas_priv, tas_fmw, fmw, offset); - - if (offset == -EINVAL) - return -EINVAL; - - fw_fixed_hdr = &tas_fmw->fw_hdr.fixed_hdr; - /* Support different versions of firmware */ - switch (fw_fixed_hdr->drv_ver) { - case 0x301: - case 0x302: - case 0x502: - case 0x503: - tas_priv->fw_parse_variable_header = - fw_parse_variable_header_kernel; - tas_priv->fw_parse_program_data = - fw_parse_program_data_kernel; - tas_priv->fw_parse_configuration_data = - fw_parse_configuration_data_kernel; - tas_priv->tasdevice_load_block = - tasdevice_load_block_kernel; - break; - case 0x202: - case 0x400: - tas_priv->fw_parse_variable_header = - fw_parse_variable_header_git; - tas_priv->fw_parse_program_data = - fw_parse_program_data; - tas_priv->fw_parse_configuration_data = - fw_parse_configuration_data; - tas_priv->tasdevice_load_block = - tasdevice_load_block; - break; - default: - ret = dspfw_default_callback(tas_priv, - fw_fixed_hdr->drv_ver, fw_fixed_hdr->ppcver); - if (ret) - return ret; - break; - } - - offset = tas_priv->fw_parse_variable_header(tas_priv, fmw, offset); - if (offset < 0) - return offset; - - offset = tas_priv->fw_parse_program_data(tas_priv, tas_fmw, fmw, - offset); - if (offset < 0) - return offset; - - offset = tas_priv->fw_parse_configuration_data(tas_priv, - tas_fmw, fmw, offset); - if (offset < 0) - ret = offset; - - return ret; -} - -/* DSP firmware binary file parser function. */ -int tasdevice_spi_dsp_parser(void *context) -{ - struct tasdevice_priv *tas_priv = context; - const struct firmware *fw_entry; - int ret; - - ret = request_firmware(&fw_entry, tas_priv->coef_binaryname, - tas_priv->dev); - if (ret) { - dev_err(tas_priv->dev, "%s: load %s error\n", __func__, - tas_priv->coef_binaryname); - return ret; - } - - ret = tasdevice_dspfw_ready(fw_entry, tas_priv); - release_firmware(fw_entry); - fw_entry = NULL; - - return ret; -} - -/* DSP firmware program block data remove function. */ -static void tasdev_dsp_prog_blk_remove(struct tasdevice_prog *prog) -{ - struct tasdevice_data *tas_dt; - struct tasdev_blk *blk; - unsigned int i; - - if (!prog) - return; - - tas_dt = &prog->dev_data; - - if (!tas_dt->dev_blks) - return; - - for (i = 0; i < tas_dt->nr_blk; i++) { - blk = &tas_dt->dev_blks[i]; - kfree(blk->data); - } - kfree(tas_dt->dev_blks); -} - -/* DSP firmware program block data remove function. */ -static void tasdev_dsp_prog_remove(struct tasdevice_prog *prog, - unsigned short nr) -{ - int i; - - for (i = 0; i < nr; i++) - tasdev_dsp_prog_blk_remove(&prog[i]); - kfree(prog); -} - -/* DSP firmware config block data remove function. */ -static void tasdev_dsp_cfg_blk_remove(struct tasdevice_config *cfg) -{ - struct tasdevice_data *tas_dt; - struct tasdev_blk *blk; - unsigned int i; - - if (cfg) { - tas_dt = &cfg->dev_data; - - if (!tas_dt->dev_blks) - return; - - for (i = 0; i < tas_dt->nr_blk; i++) { - blk = &tas_dt->dev_blks[i]; - kfree(blk->data); - } - kfree(tas_dt->dev_blks); - } -} - -/* DSP firmware config remove function. */ -static void tasdev_dsp_cfg_remove(struct tasdevice_config *config, - unsigned short nr) -{ - int i; - - for (i = 0; i < nr; i++) - tasdev_dsp_cfg_blk_remove(&config[i]); - kfree(config); -} - -/* DSP firmware remove function. */ -void tasdevice_spi_dsp_remove(void *context) -{ - struct tasdevice_priv *tas_dev = context; - - if (!tas_dev->fmw) - return; - - if (tas_dev->fmw->programs) - tasdev_dsp_prog_remove(tas_dev->fmw->programs, - tas_dev->fmw->nr_programs); - if (tas_dev->fmw->configs) - tasdev_dsp_cfg_remove(tas_dev->fmw->configs, - tas_dev->fmw->nr_configurations); - kfree(tas_dev->fmw); - tas_dev->fmw = NULL; -} - -/* DSP firmware calibration data remove function. */ -static void tas2781_clear_calfirmware(struct tasdevice_fw *tas_fmw) -{ - struct tasdevice_calibration *calibration; - struct tasdev_blk *block; - unsigned int blks; - int i; - - if (!tas_fmw->calibrations) - goto out; - - for (i = 0; i < tas_fmw->nr_calibrations; i++) { - calibration = &tas_fmw->calibrations[i]; - if (!calibration) - continue; - - if (!calibration->dev_data.dev_blks) - continue; - - for (blks = 0; blks < calibration->dev_data.nr_blk; blks++) { - block = &calibration->dev_data.dev_blks[blks]; - if (!block) - continue; - kfree(block->data); - } - kfree(calibration->dev_data.dev_blks); - } - kfree(tas_fmw->calibrations); -out: - kfree(tas_fmw); -} - -/* Calibration data from firmware remove function. */ -void tasdevice_spi_calbin_remove(void *context) -{ - struct tasdevice_priv *tas_priv = context; - - if (tas_priv->cali_data_fmw) { - tas2781_clear_calfirmware(tas_priv->cali_data_fmw); - tas_priv->cali_data_fmw = NULL; - } -} - -/* Configuration remove function. */ -void tasdevice_spi_config_info_remove(void *context) -{ - struct tasdevice_priv *tas_priv = context; - struct tasdevice_rca *rca = &tas_priv->rcabin; - struct tasdevice_config_info **ci = rca->cfg_info; - unsigned int i, j; - - if (!ci) - return; - for (i = 0; i < rca->ncfgs; i++) { - if (!ci[i]) - continue; - if (ci[i]->blk_data) { - for (j = 0; j < ci[i]->real_nblocks; j++) { - if (!ci[i]->blk_data[j]) - continue; - kfree(ci[i]->blk_data[j]->regdata); - kfree(ci[i]->blk_data[j]); - } - kfree(ci[i]->blk_data); - } - kfree(ci[i]); - } - kfree(ci); -} - -/* DSP firmware program block data load function. */ -static int tasdevice_load_data(struct tasdevice_priv *tas_priv, - struct tasdevice_data *dev_data) -{ - struct tasdev_blk *block; - unsigned int i; - int ret = 0; - - for (i = 0; i < dev_data->nr_blk; i++) { - block = &dev_data->dev_blks[i]; - ret = tas_priv->tasdevice_load_block(tas_priv, block); - if (ret < 0) - break; - } - - return ret; -} - -/* DSP firmware program load interface function. */ -int tasdevice_spi_prmg_load(void *context, int prm_no) -{ - struct tasdevice_priv *tas_priv = context; - struct tasdevice_fw *tas_fmw = tas_priv->fmw; - struct tasdevice_prog *program; - struct tasdevice_config *conf; - int ret = 0; - - if (!tas_fmw) { - dev_err(tas_priv->dev, "%s: Firmware is NULL\n", __func__); - return -EINVAL; - } - if (prm_no >= 0 && prm_no <= tas_fmw->nr_programs) { - tas_priv->cur_conf = 0; - tas_priv->is_loading = true; - program = &tas_fmw->programs[prm_no]; - ret = tasdevice_load_data(tas_priv, &program->dev_data); - if (ret < 0) { - dev_err(tas_priv->dev, "Program failed %d.\n", ret); - return ret; - } - tas_priv->cur_prog = prm_no; - - conf = &tas_fmw->configs[tas_priv->cur_conf]; - ret = tasdevice_load_data(tas_priv, &conf->dev_data); - if (ret < 0) - dev_err(tas_priv->dev, "Config failed %d.\n", ret); - } else { - dev_err(tas_priv->dev, - "%s: prm(%d) is not in range of Programs %u\n", - __func__, prm_no, tas_fmw->nr_programs); - return -EINVAL; - } - - return ret; -} - -/* RCABIN configuration switch interface function. */ -void tasdevice_spi_tuning_switch(void *context, int state) -{ - struct tasdevice_priv *tas_priv = context; - int profile_cfg_id = tas_priv->rcabin.profile_cfg_id; - - if (tas_priv->fw_state == TASDEVICE_DSP_FW_FAIL) { - dev_err(tas_priv->dev, "DSP bin file not loaded\n"); - return; - } - - if (state == 0) - tasdevice_spi_select_cfg_blk(tas_priv, profile_cfg_id, - TASDEVICE_BIN_BLK_PRE_POWER_UP); - else - tasdevice_spi_select_cfg_blk(tas_priv, profile_cfg_id, - TASDEVICE_BIN_BLK_PRE_SHUTDOWN); -} diff --git a/sound/soc/codecs/tas2781-comlib.c b/sound/soc/codecs/tas2781-comlib.c index 1e0b3aa95749..4df2edcfac40 100644 --- a/sound/soc/codecs/tas2781-comlib.c +++ b/sound/soc/codecs/tas2781-comlib.c @@ -21,8 +21,6 @@ #include <sound/soc.h> #include <sound/tas2781.h> -#define TASDEVICE_CRC8_POLYNOMIAL 0x4d - static const struct regmap_range_cfg tasdevice_ranges[] = { { .range_min = 0, @@ -122,7 +120,7 @@ int tasdevice_dev_read(struct tasdevice_priv *tas_priv, if (chn < tas_priv->ndev) { struct regmap *map = tas_priv->regmap; - ret = tasdevice_change_chn_book(tas_priv, chn, + ret = tas_priv->change_chn_book(tas_priv, chn, TASDEVICE_BOOK_ID(reg)); if (ret < 0) goto out; @@ -149,7 +147,7 @@ int tasdevice_dev_write(struct tasdevice_priv *tas_priv, if (chn < tas_priv->ndev) { struct regmap *map = tas_priv->regmap; - ret = tasdevice_change_chn_book(tas_priv, chn, + ret = tas_priv->change_chn_book(tas_priv, chn, TASDEVICE_BOOK_ID(reg)); if (ret < 0) goto out; @@ -179,7 +177,7 @@ int tasdevice_dev_bulk_write( if (chn < tas_priv->ndev) { struct regmap *map = tas_priv->regmap; - ret = tasdevice_change_chn_book(tas_priv, chn, + ret = tas_priv->change_chn_book(tas_priv, chn, TASDEVICE_BOOK_ID(reg)); if (ret < 0) goto out; @@ -208,7 +206,7 @@ int tasdevice_dev_bulk_read(struct tasdevice_priv *tas_priv, if (chn < tas_priv->ndev) { struct regmap *map = tas_priv->regmap; - ret = tasdevice_change_chn_book(tas_priv, chn, + ret = tas_priv->change_chn_book(tas_priv, chn, TASDEVICE_BOOK_ID(reg)); if (ret < 0) goto out; @@ -234,7 +232,7 @@ int tasdevice_dev_update_bits( if (chn < tas_priv->ndev) { struct regmap *map = tas_priv->regmap; - ret = tasdevice_change_chn_book(tas_priv, chn, + ret = tas_priv->change_chn_book(tas_priv, chn, TASDEVICE_BOOK_ID(reg)); if (ret < 0) goto out; @@ -340,6 +338,7 @@ int tasdevice_init(struct tasdevice_priv *tas_priv) tas_priv->cur_prog = -1; tas_priv->cur_conf = -1; + tas_priv->isspi = false; for (i = 0; i < tas_priv->ndev; i++) { tas_priv->tasdevice[i].cur_book = -1; @@ -347,6 +346,11 @@ int tasdevice_init(struct tasdevice_priv *tas_priv) tas_priv->tasdevice[i].cur_conf = -1; } + tas_priv->update_bits = tasdevice_dev_update_bits; + tas_priv->change_chn_book = tasdevice_change_chn_book; + tas_priv->dev_read = tasdevice_dev_read; + tas_priv->dev_bulk_read = tasdevice_dev_bulk_read; + mutex_init(&tas_priv->codec_lock); out: diff --git a/sound/soc/codecs/tas2781-fmwlib.c b/sound/soc/codecs/tas2781-fmwlib.c index 13a197468193..3dd76aff1b6e 100644 --- a/sound/soc/codecs/tas2781-fmwlib.c +++ b/sound/soc/codecs/tas2781-fmwlib.c @@ -803,8 +803,13 @@ static int tasdevice_process_block(void *context, unsigned char *data, chn = idx - 1; chnend = idx; } else { - chn = 0; - chnend = tas_priv->ndev; + if (tas_priv->isspi) { + chn = tas_priv->index; + chnend = chn + 1; + } else { + chn = 0; + chnend = tas_priv->ndev; + } } for (; chn < chnend; chn++) { @@ -896,7 +901,7 @@ static int tasdevice_process_block(void *context, unsigned char *data, is_err = true; break; } - rc = tasdevice_dev_update_bits(tas_priv, chn, + rc = tas_priv->update_bits(tas_priv, chn, TASDEVICE_REG(data[subblk_offset + 2], data[subblk_offset + 3], data[subblk_offset + 4]), @@ -1461,7 +1466,7 @@ static int tasdev_multibytes_chksum(struct tasdevice_priv *tasdevice, goto end; } - ret = tasdevice_dev_bulk_read(tasdevice, chn, + ret = tasdevice->dev_bulk_read(tasdevice, chn, TASDEVICE_REG(book, page, crc_data.offset), nBuf1, crc_data.len); if (ret < 0) @@ -1511,7 +1516,7 @@ static int do_singlereg_checksum(struct tasdevice_priv *tasdevice, in = check_yram(&crc_data, book, page, reg, 1); if (!in) goto end; - ret = tasdevice_dev_read(tasdevice, chl, + ret = tasdevice->dev_read(tasdevice, chl, TASDEVICE_REG(book, page, reg), &nData1); if (ret < 0) goto end; @@ -1615,7 +1620,7 @@ static int tasdev_block_chksum(struct tasdevice_priv *tas_priv, unsigned int nr_value; int ret; - ret = tasdevice_dev_read(tas_priv, chn, TASDEVICE_CHECKSUM_REG, + ret = tas_priv->dev_read(tas_priv, chn, TASDEVICE_CHECKSUM_REG, &nr_value); if (ret < 0) { dev_err(tas_priv->dev, "%s: Chn %d\n", __func__, chn); -- 2.43.0

3 2

Re: [PATCH v3 2/5] ASoC: qcom: sm8250: set card driver name from match data
by Mark Brown 25 Apr '25

25 Apr '25

On Fri, Apr 25, 2025 at 10:47:42PM +0200, Luca Weiss wrote: > On Fri Apr 25, 2025 at 9:03 PM CEST, Mark Brown wrote: > > Or base it on my tree and let things get sorted in the merge, I don't > > know what the conflicts might be? > For this patch here it might be okay but patch 3/5 from this series very > much depends on the patch in Greg's tree, given it refactors/expands on > the USB_RX if there. Resolving this through merge wouldn't be very > pretty. Well, unfortunately Greg didn't put things on a branch so yeah waiting for next release might be easiest.

1 0

Re: [PATCH v3 2/5] ASoC: qcom: sm8250: set card driver name from match data
by Mark Brown 25 Apr '25

25 Apr '25

On Fri, Apr 25, 2025 at 08:19:39PM +0200, Luca Weiss wrote: > I've based this series on next-20250417 tag, so this is probably due to > the changes from the USB sound offloading series that Greg has picked > up. > So either Greg also picks up these changes when they're ready, or we > wait until 6.17? Or base it on my tree and let things get sorted in the merge, I don't know what the conflicts might be?

1 0

[PATCH 1/3] ASoC: amd: acp: Fix NULL pointer deref on acp resume path
by Venkata Prasad Potturu 25 Apr '25

25 Apr '25

update chip data using dev_get_drvdata(dev->parent) instead of dev_get_platdata(dev). BUG: kernel NULL pointer dereference, address: 0000000000000010 Call Trace: <TASK> ? __pfx_platform_pm_resume+0x10/0x10 platform_pm_resume+0x28/0x60 dpm_run_callback+0x51/0x1a0 device_resume+0x1a6/0x2b0 dpm_resume+0x168/0x230 Fixes: e3933683b25e ("ASoC: amd: acp: Remove redundant acp_dev_data structure") Signed-off-by: Venkata Prasad Potturu <venkataprasad.potturu(a)amd.com> --- sound/soc/amd/acp/acp-rembrandt.c | 2 +- sound/soc/amd/acp/acp-renoir.c | 2 +- sound/soc/amd/acp/acp63.c | 2 +- sound/soc/amd/acp/acp70.c | 2 +- 4 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/amd/acp/acp-rembrandt.c b/sound/soc/amd/acp/acp-rembrandt.c index 746b6ed72029..cccdd10c345e 100644 --- a/sound/soc/amd/acp/acp-rembrandt.c +++ b/sound/soc/amd/acp/acp-rembrandt.c @@ -199,7 +199,7 @@ static void rembrandt_audio_remove(struct platform_device *pdev) static int rmb_pcm_resume(struct device *dev) { - struct acp_chip_info *chip = dev_get_platdata(dev); + struct acp_chip_info *chip = dev_get_drvdata(dev->parent); struct acp_stream *stream; struct snd_pcm_substream *substream; snd_pcm_uframes_t buf_in_frames; diff --git a/sound/soc/amd/acp/acp-renoir.c b/sound/soc/amd/acp/acp-renoir.c index ebf0106fc737..04f6d70b6a92 100644 --- a/sound/soc/amd/acp/acp-renoir.c +++ b/sound/soc/amd/acp/acp-renoir.c @@ -146,7 +146,7 @@ static void renoir_audio_remove(struct platform_device *pdev) static int rn_pcm_resume(struct device *dev) { - struct acp_chip_info *chip = dev_get_platdata(dev); + struct acp_chip_info *chip = dev_get_drvdata(dev->parent); struct acp_stream *stream; struct snd_pcm_substream *substream; snd_pcm_uframes_t buf_in_frames; diff --git a/sound/soc/amd/acp/acp63.c b/sound/soc/amd/acp/acp63.c index 52d895e624c7..1f15c96a9b94 100644 --- a/sound/soc/amd/acp/acp63.c +++ b/sound/soc/amd/acp/acp63.c @@ -250,7 +250,7 @@ static void acp63_audio_remove(struct platform_device *pdev) static int acp63_pcm_resume(struct device *dev) { - struct acp_chip_info *chip = dev_get_platdata(dev); + struct acp_chip_info *chip = dev_get_drvdata(dev->parent); struct acp_stream *stream; struct snd_pcm_substream *substream; snd_pcm_uframes_t buf_in_frames; diff --git a/sound/soc/amd/acp/acp70.c b/sound/soc/amd/acp/acp70.c index 6d5f5ade075c..217b717e9beb 100644 --- a/sound/soc/amd/acp/acp70.c +++ b/sound/soc/amd/acp/acp70.c @@ -182,7 +182,7 @@ static void acp_acp70_audio_remove(struct platform_device *pdev) static int acp70_pcm_resume(struct device *dev) { - struct acp_chip_info *chip = dev_get_platdata(dev); + struct acp_chip_info *chip = dev_get_drvdata(dev->parent); struct acp_stream *stream; struct snd_pcm_substream *substream; snd_pcm_uframes_t buf_in_frames; -- 2.43.0

4 6

Re: [PATCH v3 2/5] ASoC: qcom: sm8250: set card driver name from match data
by Mark Brown 25 Apr '25

25 Apr '25

On Fri, Apr 25, 2025 at 10:07:26AM +0200, Luca Weiss wrote: > Sound machine drivers for Qualcomm SoCs can be reused across multiple > SoCs. But user space ALSA UCM files depend on the card driver name which > should be set per board/SoC. This doesn't apply against current code, please check and resend.

1 0

SSL2+ MK II "ucm is not supported for this device"
by GitHub issues - opened 25 Apr '25

25 Apr '25

alsa-project/alsa-ucm-conf issue #555 was opened from ATRAhassis: " ALSA lib parser.c:244:(error_node) UCM is not supported for this USB device (Solid State Logic SSL 2+ Mk II at usb-0000:0b:00.1-6, high speed @ USB31e9:0009) ALSA lib main.c:1554:(snd_use_case_mgr_open) error: failed to import hw:0 use case configuration -6 alsaucm: error failed to open sound card hw:0: No such device or address " i believe i just need to make a file for it but i am too new to understand exactly how. willing to help if i can get any direction. was told if a UCM can be made for this device that its helpful for all distros so i figured id make this issue ticket Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/555 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

DRAFT [for alsa-lib 1.2.15]: HDA: Handle better Bass speaker including 4 channel mode
by GitHub pull_request - edited 25 Apr '25

25 Apr '25

alsa-project/alsa-ucm-conf pull request #533 was edited from perexg: It's a follow up of #410 . Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/533 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/533.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

DRAFT [for alsa-lib 1.2.15] USB-Audio: Steinberg UR22C - fix capture channels for older firmware
by GitHub pull_request - edited 24 Apr '25

24 Apr '25

alsa-project/alsa-ucm-conf pull request #554 was edited from perexg: It seems that firmware version 3.20 added 4 capture channels while previous firmware has only 2 capture channels. Fixes: https://github.com/alsa-project/alsa-ucm-conf/issues/547 Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/554 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/554.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

USB-Audio: Steinberg UR22C - fix capture channels for older firmware
by GitHub pull_request - opened 24 Apr '25

24 Apr '25

alsa-project/alsa-ucm-conf pull request #554 was opened from perexg: It seems that firmware version 3.20 added 4 capture channels while previous firmware has only 2 capture channels. Fixes: https://github.com/alsa-project/alsa-ucm-conf/issues/547 Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/554 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/554.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

DRAFT [for alsa-lib-1.2.15]: USB-Audio: GoXLR: enable detection of beta firmware (25 channels)
by GitHub pull_request - edited 24 Apr '25

24 Apr '25

alsa-project/alsa-ucm-conf pull request #524 was edited from perexg: 'Syntax 8' is required (alsa-lib 1.2.14+) Fixes: 17f9b4f ("USB-Audio: GoXLR - fix the channel detection for mini, cleanups") Link: https://github.com/alsa-project/alsa-ucm-conf/issues/444 Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/524 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/524.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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GoXLR Mini (ALSA UCM error)
by GitHub issues - edited 24 Apr '25

24 Apr '25

alsa-project/alsa-ucm-conf issue #553 was edited from Crav07: The device seems to be functioning, but this error has came up after updating to Bazzite 42 from Bazzite 41. Here is the result after running "journalctl --user -u wireplumber --boot 0" Apr 24 10:52:12 fedora wireplumber[2409]: wp-internal-comp-loader: Loading profile 'main' Apr 24 10:52:12 fedora wireplumber[2409]: spa.alsa: Error in ALSA UCM profile for _ucm0003.hw:GoXLRMini,0 (HiFi: Line5: source> Apr 24 10:52:13 fedora wireplumber[2409]: spa.alsa: Errors in ALSA UCM profile for card GoXLRMini Apr 24 10:52:13 fedora wireplumber[2409]: spa.bluez5: BlueZ system service is not available Apr 24 11:30:31 fedora wireplumber[2409]: wp-event-dispatcher: <WpAsyncEventHook:0x557d66add4c0> failed: <WpSiStandardLink:0x5> (Edit: added results) Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/553 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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GoXLR Mini (ALSA UCM error)
by GitHub issues - opened 24 Apr '25

24 Apr '25

alsa-project/alsa-ucm-conf issue #553 was opened from Crav07: The device seems to be functioning, but this error has came up after updating to Bazzite 42 from Bazzite 41. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/553 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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SSL2+ reports [ALSA UCM error] but seems to be working fine.
by GitHub issues - opened 24 Apr '25

24 Apr '25

alsa-project/alsa-ucm-conf issue #552 was opened from Steamvoker: Got redirected here from https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/4653 Don't want to just copy paste and issue here, but everything seems to be working fine, except that the device reports errors in KDE sound settings. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/552 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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Question about conversion kcontrol to component
by Kuninori Morimoto 24 Apr '25

24 Apr '25

Hi ALSA ML I have noticed that we have similar functions snd_soc_kcontrol_component() snd_soc_dapm_kcontrol_component() Both converts snd_kcontrol to snd_soc_component. I wonder can we merge these ? or which we can use is case-by-case ? Thank you for your help !! Best regards --- Kuninori Morimoto

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Pioneer DJ DJM mixers channel names
by František Kučera 24 Apr '25

24 Apr '25

When working on DJM-V10, I noticed that other Pioneer mixers (like my older DJM-250MK2) have control names that do not comply with conventions described at <https://docs.kernel.org/sound/designs/control-names.html> and thus do not show in alsamixer under proper category (playback/capture) and they are all under playback. I am preparing a patch that will add "Playback Switch" or "Capture Switch" suffix to control names. But I am not sure about the prefix. Should it be "aux" or "PCM"? It is often the main function, so I tend rather to PCM than auxiliary... Could you give some advice regarding the mixer control names? Thanks, Franta

1 0

1.2.14: test suite build fails with `-Wincompatible-pointer-types` and `-Wold-style-definition` errors
by GitHub issues - opened 23 Apr '25

23 Apr '25

alsa-project/alsa-lib issue #452 was opened from kloczek: gcc 15.0.1 from fedora rawhide. <details> <summary>Test suite output</summary> ``` [tkloczko@pers-tkloczko alsa-lib-1.2.14]$ make check -k Making check in doc make[1]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.14/doc' Making check in pictures make[2]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.14/doc/pictures' make[2]: Nothing to be done for 'check'. make[2]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.14/doc/pictures' make[2]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.14/doc' make[2]: Nothing to be done for 'check-am'. make[2]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.14/doc' make[1]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.14/doc' Making check in include make[1]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.14/include' Making check in sound make[2]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.14/include/sound' Making check in uapi make[3]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.14/include/sound/uapi' make[3]: Nothing to be done for 'check'. make[3]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.14/include/sound/uapi' make[3]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.14/include/sound' make[3]: Nothing to be done for 'check-am'. make[3]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.14/include/sound' make[2]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.14/include/sound' make[2]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.14/include' if ! test -r alsa/local.h; then \ ln -s ../include alsa; \ fi for f in asoundlib.h version.h; do \ if ! test -r ../include/$f; then \ ln -s /home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.14/include/$f ../include/$f; \ fi; \ done make[2]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.14/include' make[1]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.14/include' Making check in src make[1]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.14/src' make check-recursive make[2]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.14/src' Making check in control make[3]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.14/src/control' make[3]: Nothing to be done for 'check'. make[3]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.14/src/control' Making check in mixer make[3]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.14/src/mixer' make[3]: Nothing to be done for 'check'. make[3]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.14/src/mixer' Making check in pcm make[3]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.14/src/pcm' make[4]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.14/src/pcm' make[4]: Nothing to be done for 'check-am'. make[4]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.14/src/pcm' make[3]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.14/src/pcm' Making check in timer make[3]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.14/src/timer' make[3]: Nothing to be done for 'check'. make[3]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.14/src/timer' Making check in rawmidi make[3]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.14/src/rawmidi' make[3]: Nothing to be done for 'check'. make[3]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.14/src/rawmidi' Making check in hwdep make[3]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.14/src/hwdep' make[3]: Nothing to be done for 'check'. make[3]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.14/src/hwdep' Making check in seq make[3]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.14/src/seq' make[3]: Nothing to be done for 'check'. make[3]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.14/src/seq' Making check in ucm make[3]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.14/src/ucm' make[3]: Nothing to be done for 'check'. make[3]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.14/src/ucm' Making check in conf make[3]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.14/src/conf' Making check in cards make[4]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.14/src/conf/cards' make[4]: Nothing to be done for 'check'. make[4]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.14/src/conf/cards' Making check in ctl make[4]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.14/src/conf/ctl' make[4]: Nothing to be done for 'check'. make[4]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.14/src/conf/ctl' Making check in pcm make[4]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.14/src/conf/pcm' make[4]: Nothing to be done for 'check'. make[4]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.14/src/conf/pcm' make[4]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.14/src/conf' make[4]: Nothing to be done for 'check-am'. make[4]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.14/src/conf' make[3]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.14/src/conf' make[3]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.14/src' make[3]: Nothing to be done for 'check-am'. make[3]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.14/src' make[2]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.14/src' make[1]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.14/src' Making check in src/topology make[1]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.14/src/topology' make[1]: Nothing to be done for 'check'. make[1]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.14/src/topology' Making check in modules make[1]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.14/modules' Making check in mixer make[2]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.14/modules/mixer' Making check in simple make[3]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.14/modules/mixer/simple' make[3]: Nothing to be done for 'check'. make[3]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.14/modules/mixer/simple' make[3]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.14/modules/mixer' make[3]: Nothing to be done for 'check-am'. make[3]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.14/modules/mixer' make[2]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.14/modules/mixer' make[2]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.14/modules' make[2]: Nothing to be done for 'check-am'. make[2]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.14/modules' make[1]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.14/modules' Making check in aserver make[1]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.14/aserver' make[1]: Nothing to be done for 'check'. make[1]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.14/aserver' Making check in test make[1]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.14/test' Making check in . make[2]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.14/test' make control pcm pcm_min latency seq seq-ump-example playmidi1 timer rawmidi midiloop umpinfo oldapi queue_timer namehint client_event_filter chmap audio_time user-ctl-element-set pcm-multi-thread make[3]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.14/test' make[3]: 'control' is up to date. make[3]: 'pcm' is up to date. make[3]: 'pcm_min' is up to date. make[3]: 'latency' is up to date. make[3]: 'seq' is up to date. make[3]: 'seq-ump-example' is up to date. CC playmidi1.o In file included from playmidi1.c:51: midifile.c: In function ‘readmt’: midifile.c:191:1: warning: old-style function definition [-Wold-style-definition] 191 | readmt (s) /* read through the "MThd" or "MTrk" header string */ | ^~~~~~ midifile.c: In function ‘readheader’: midifile.c:274:6: error: too many arguments to function ‘Mf_header’; expected 0, have 3 274 | (*Mf_header) (format, ntrks, division); | ~^~~~~~~~~~~ ~~~~~~ midifile.c: In function ‘readtrack’: midifile.c:437:11: error: too many arguments to function ‘chanmessage’; expected 0, have 3 437 | chanmessage (status, c1, (needed > 1) ? egetc () : 0); | ^~~~~~~~~~~ ~~~~~~ midifile.c:146:13: note: declared here 146 | static void chanmessage (); | ^~~~~~~~~~~ midifile.c:451:13: error: too many arguments to function ‘msgadd’; expected 0, have 1 451 | msgadd (egetc ()); | ^~~~~~ ~~~~~~~~ midifile.c:149:13: note: declared here 149 | static void msgadd (); | ^~~~~~ midifile.c:453:11: error: too many arguments to function ‘metaevent’; expected 0, have 1 453 | metaevent (type); | ^~~~~~~~~ ~~~~ midifile.c:144:13: note: declared here 144 | static void metaevent (); | ^~~~~~~~~ midifile.c:460:11: error: too many arguments to function ‘msgadd’; expected 0, have 1 460 | msgadd (0xf0); | ^~~~~~ ~~~~ midifile.c:149:13: note: declared here 149 | static void msgadd (); | ^~~~~~ midifile.c:463:13: error: too many arguments to function ‘msgadd’; expected 0, have 1 463 | msgadd (c = egetc ()); | ^~~~~~ ~~~~~~~~~~~~ midifile.c:149:13: note: declared here 149 | static void msgadd (); | ^~~~~~ midifile.c:479:13: error: too many arguments to function ‘msgadd’; expected 0, have 1 479 | msgadd (c = egetc ()); | ^~~~~~ ~~~~~~~~~~~~ midifile.c:149:13: note: declared here 149 | static void msgadd (); | ^~~~~~ midifile.c:484:18: error: too many arguments to function ‘Mf_arbitrary’; expected 0, have 2 484 | (*Mf_arbitrary) (msgleng (), msg ()); | ~^~~~~~~~~~~~~~ ~~~~~~~~~~ midifile.c:493:11: error: too many arguments to function ‘badbyte’; expected 0, have 1 493 | badbyte (c); | ^~~~~~~ ~ midifile.c:143:13: note: declared here 143 | static void badbyte (); | ^~~~~~~ midifile.c: In function ‘badbyte’: midifile.c:504:1: warning: old-style function definition [-Wold-style-definition] 504 | badbyte (c) | ^~~~~~~ midifile.c: At top level: midifile.c:515:1: error: conflicting types for ‘metaevent’; have ‘void(int)’ 515 | metaevent (int type) | ^~~~~~~~~ midifile.c:144:13: note: previous declaration of ‘metaevent’ with type ‘void(void)’ 144 | static void metaevent (); | ^~~~~~~~~ midifile.c: In function ‘metaevent’: midifile.c:524:23: error: too many arguments to function ‘to16bit’; expected 0, have 2 524 | (*Mf_seqnum) (to16bit (m[0], m[1])); | ^~~~~~~ ~~~~ midifile.c:139:12: note: declared here 139 | static int to16bit (); | ^~~~~~~ midifile.c:524:10: error: too many arguments to function ‘Mf_seqnum’; expected 0, have 1 524 | (*Mf_seqnum) (to16bit (m[0], m[1])); | ~^~~~~~~~~~~ ~~~~~~~~~~~~~~~~~~~~ midifile.c:543:10: error: too many arguments to function ‘Mf_text’; expected 0, have 3 543 | (*Mf_text) (type, leng, m); | ~^~~~~~~~~ ~~~~ midifile.c:551:37: error: too many arguments to function ‘to32bit’; expected 0, have 4 551 | (*Mf_tempo) (Mf_currtempo = to32bit (0, m[0], m[1], m[2])); | ^~~~~~~ ~ midifile.c:137:13: note: declared here 137 | static long to32bit (); | ^~~~~~~ midifile.c:551:10: error: too many arguments to function ‘Mf_tempo’; expected 0, have 1 551 | (*Mf_tempo) (Mf_currtempo = to32bit (0, m[0], m[1], m[2])); | ~^~~~~~~~~~ ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ midifile.c:560:10: error: too many arguments to function ‘Mf_smpte’; expected 0, have 5 560 | (*Mf_smpte) (m[0], m[1], m[2], m[3], m[4]); | ~^~~~~~~~~~ ~~~~ midifile.c:564:10: error: too many arguments to function ‘Mf_timesig’; expected 0, have 4 564 | (*Mf_timesig) (m[0], m[1], m[2], m[3]); | ~^~~~~~~~~~~~ ~~~~ midifile.c:568:10: error: too many arguments to function ‘Mf_keysig’; expected 0, have 2 568 | (*Mf_keysig) (m[0], m[1]); | ~^~~~~~~~~~~ ~~~~ midifile.c:572:10: error: too many arguments to function ‘Mf_seqspecific’; expected 0, have 2 572 | (*Mf_seqspecific) (leng, m); | ~^~~~~~~~~~~~~~~~ ~~~~ midifile.c:576:10: error: too many arguments to function ‘Mf_metamisc’; expected 0, have 3 576 | (*Mf_metamisc) (type, leng, m); | ~^~~~~~~~~~~~~ ~~~~ midifile.c: In function ‘sysex’: midifile.c:585:6: error: too many arguments to function ‘Mf_sysex’; expected 0, have 2 585 | (*Mf_sysex) (msgleng (), msg ()); | ~^~~~~~~~~~ ~~~~~~~~~~ midifile.c: In function ‘chanmessage’: midifile.c:590:1: warning: old-style function definition [-Wold-style-definition] 590 | chanmessage (status, c1, c2) | ^~~~~~~~~~~ midifile.c:605:10: error: too many arguments to function ‘Mf_noteoff’; expected 0, have 3 605 | (*Mf_noteoff) (chan, c1, c2); | ~^~~~~~~~~~~~ ~~~~ midifile.c:609:10: error: too many arguments to function ‘Mf_noteon’; expected 0, have 3 609 | (*Mf_noteon) (chan, c1, c2); | ~^~~~~~~~~~~ ~~~~ midifile.c:613:10: error: too many arguments to function ‘Mf_pressure’; expected 0, have 3 613 | (*Mf_pressure) (chan, c1, c2); | ~^~~~~~~~~~~~~ ~~~~ midifile.c:617:10: error: too many arguments to function ‘Mf_parameter’; expected 0, have 3 617 | (*Mf_parameter) (chan, c1, c2); | ~^~~~~~~~~~~~~~ ~~~~ midifile.c:621:10: error: too many arguments to function ‘Mf_pitchbend’; expected 0, have 3 621 | (*Mf_pitchbend) (chan, c1, c2); | ~^~~~~~~~~~~~~~ ~~~~ midifile.c:625:10: error: too many arguments to function ‘Mf_program’; expected 0, have 2 625 | (*Mf_program) (chan, c1); | ~^~~~~~~~~~~~ ~~~~ midifile.c:629:10: error: too many arguments to function ‘Mf_chanpressure’; expected 0, have 2 629 | (*Mf_chanpressure) (chan, c1); | ~^~~~~~~~~~~~~~~~~ ~~~~ midifile.c: At top level: midifile.c:659:1: error: conflicting types for ‘to32bit’; have ‘long int(int, int, int, int)’ 659 | to32bit (int c1, int c2, int c3, int c4) | ^~~~~~~ midifile.c:137:13: note: previous declaration of ‘to32bit’ with type ‘long int(void)’ 137 | static long to32bit (); | ^~~~~~~ midifile.c: In function ‘to16bit’: midifile.c:671:1: warning: old-style function definition [-Wold-style-definition] 671 | to16bit (c1, c2) | ^~~~~~~ midifile.c: In function ‘read16bit’: midifile.c:695:10: error: too many arguments to function ‘to16bit’; expected 0, have 2 695 | return to16bit (c1, c2); | ^~~~~~~ ~~ midifile.c:671:1: note: declared here 671 | to16bit (c1, c2) | ^~~~~~~ midifile.c: In function ‘mferror’: midifile.c:700:1: warning: old-style function definition [-Wold-style-definition] 700 | mferror (s) | ^~~~~~~ midifile.c:704:6: error: too many arguments to function ‘Mf_error’; expected 0, have 1 704 | (*Mf_error) (s); | ~^~~~~~~~~~ ~ midifile.c: In function ‘msgadd’: midifile.c:739:1: warning: old-style function definition [-Wold-style-definition] 739 | msgadd (c) | ^~~~~~ midifile.c: In function ‘mfwrite’: midifile.c:808:1: warning: old-style function definition [-Wold-style-definition] 808 | mfwrite (format, ntracks, division, fp) | ^~~~~~~ midifile.c:811:1: error: number of arguments doesn’t match prototype 811 | { | ^ In file included from midifile.c:53: midifile.h:42:13: error: prototype declaration 42 | extern void mfwrite(); | ^~~~~~~ midifile.c:824:3: error: too many arguments to function ‘mf_write_header_chunk’; expected 0, have 3 824 | mf_write_header_chunk (format, ntracks, division); | ^~~~~~~~~~~~~~~~~~~~~ ~~~~~~ midifile.c:813:33: note: declared here 813 | void mf_write_track_chunk (), mf_write_header_chunk (); | ^~~~~~~~~~~~~~~~~~~~~ midifile.c:836:5: error: too many arguments to function ‘mf_write_track_chunk’; expected 0, have 2 836 | mf_write_track_chunk (i, fp); | ^~~~~~~~~~~~~~~~~~~~ ~ midifile.c:813:8: note: declared here 813 | void mf_write_track_chunk (), mf_write_header_chunk (); | ^~~~~~~~~~~~~~~~~~~~ midifile.c: In function ‘mf_write_track_chunk’: midifile.c:813:8: warning: old-style function definition [-Wold-style-definition] midifile.c:843:1: error: number of arguments doesn’t match prototype 843 | { | ^ midifile.c:813:8: error: prototype declaration 813 | void mf_write_track_chunk (), mf_write_header_chunk (); | ^~~~~~~~~~~~~~~~~~~~ midifile.c:863:3: error: too many arguments to function ‘write32bit’; expected 0, have 1 863 | write32bit (trkhdr); | ^~~~~~~~~~ ~~~~~~ midifile.c:846:23: note: declared here 846 | void write16bit (), write32bit (); | ^~~~~~~~~~ midifile.c:864:3: error: too many arguments to function ‘write32bit’; expected 0, have 1 864 | write32bit (trklength); | ^~~~~~~~~~ ~~~~~~~~~ midifile.c:846:23: note: declared here 846 | void write16bit (), write32bit (); | ^~~~~~~~~~ midifile.c:870:8: error: too many arguments to function ‘Mf_writetrack’; expected 0, have 1 870 | (*Mf_writetrack) (which_track); | ~^~~~~~~~~~~~~~~ ~~~~~~~~~~~ midifile.c:905:3: error: too many arguments to function ‘write32bit’; expected 0, have 1 905 | write32bit (trkhdr); | ^~~~~~~~~~ ~~~~~~ midifile.c:846:23: note: declared here 846 | void write16bit (), write32bit (); | ^~~~~~~~~~ midifile.c:906:3: error: too many arguments to function ‘write32bit’; expected 0, have 1 906 | write32bit (trklength); | ^~~~~~~~~~ ~~~~~~~~~ midifile.c:846:23: note: declared here 846 | void write16bit (), write32bit (); | ^~~~~~~~~~ midifile.c: In function ‘mf_write_header_chunk’: midifile.c:813:33: warning: old-style function definition [-Wold-style-definition] 813 | void mf_write_track_chunk (), mf_write_header_chunk (); | ^~~~~~~~~~~~~~~~~~~~~ midifile.c:915:1: error: number of arguments doesn’t match prototype 915 | { | ^ midifile.c:813:33: error: prototype declaration 813 | void mf_write_track_chunk (), mf_write_header_chunk (); | ^~~~~~~~~~~~~~~~~~~~~ midifile.c:924:3: error: too many arguments to function ‘write32bit’; expected 0, have 1 924 | write32bit (ident); | ^~~~~~~~~~ ~~~~~ midifile.c:917:23: note: declared here 917 | void write16bit (), write32bit (); | ^~~~~~~~~~ midifile.c:925:3: error: too many arguments to function ‘write32bit’; expected 0, have 1 925 | write32bit (length); | ^~~~~~~~~~ ~~~~~~ midifile.c:917:23: note: declared here 917 | void write16bit (), write32bit (); | ^~~~~~~~~~ midifile.c:926:3: error: too many arguments to function ‘write16bit’; expected 0, have 1 926 | write16bit (format); | ^~~~~~~~~~ ~~~~~~ midifile.c:917:8: note: declared here 917 | void write16bit (), write32bit (); | ^~~~~~~~~~ midifile.c:927:3: error: too many arguments to function ‘write16bit’; expected 0, have 1 927 | write16bit (ntracks); | ^~~~~~~~~~ ~~~~~~~ midifile.c:917:8: note: declared here 917 | void write16bit (), write32bit (); | ^~~~~~~~~~ midifile.c:928:3: error: too many arguments to function ‘write16bit’; expected 0, have 1 928 | write16bit (division); | ^~~~~~~~~~ ~~~~~~~~ midifile.c:917:8: note: declared here 917 | void write16bit (), write32bit (); | ^~~~~~~~~~ midifile.c: In function ‘mf_write_midi_event’: midifile.c:951:1: warning: old-style function definition [-Wold-style-definition] 951 | mf_write_midi_event (delta_time, type, chan, data, size) | ^~~~~~~~~~~~~~~~~~~ midifile.c:960:3: error: too many arguments to function ‘WriteVarLen’; expected 0, have 1 960 | WriteVarLen (delta_time); | ^~~~~~~~~~~ ~~~~~~~~~~ midifile.c:157:6: note: declared here 157 | void WriteVarLen (); | ^~~~~~~~~~~ midifile.c:976:5: error: too many arguments to function ‘eputc’; expected 0, have 1 976 | eputc (laststatus = c); | ^~~~~ ~~~~~~~~~~~~~~ midifile.c:151:12: note: declared here 151 | static int eputc (); | ^~~~~ midifile.c:979:5: error: too many arguments to function ‘WriteVarLen’; expected 0, have 1 979 | WriteVarLen (size); | ^~~~~~~~~~~ ~~~~ midifile.c:157:6: note: declared here 157 | void WriteVarLen (); | ^~~~~~~~~~~ midifile.c:983:5: error: too many arguments to function ‘eputc’; expected 0, have 1 983 | eputc (data[i]); | ^~~~~ ~~~~~~~ midifile.c:151:12: note: declared here 151 | static int eputc (); | ^~~~~ midifile.c: In function ‘mf_write_meta_event’: midifile.c:1003:1: warning: old-style function definition [-Wold-style-definition] 1003 | mf_write_meta_event (delta_time, type, data, size) | ^~~~~~~~~~~~~~~~~~~ midifile.c:1007:1: error: number of arguments doesn’t match prototype 1007 | { | ^ midifile.c:154:5: error: prototype declaration 154 | int mf_write_meta_event (); | ^~~~~~~~~~~~~~~~~~~ midifile.c:1010:3: error: too many arguments to function ‘WriteVarLen’; expected 0, have 1 1010 | WriteVarLen (delta_time); | ^~~~~~~~~~~ ~~~~~~~~~~ midifile.c:157:6: note: declared here 157 | void WriteVarLen (); | ^~~~~~~~~~~ midifile.c:1013:3: error: too many arguments to function ‘eputc’; expected 0, have 1 1013 | eputc (laststatus = meta_event); | ^~~~~ midifile.c:151:12: note: declared here 151 | static int eputc (); | ^~~~~ midifile.c:1016:3: error: too many arguments to function ‘eputc’; expected 0, have 1 1016 | eputc (type); | ^~~~~ ~~~~ midifile.c:151:12: note: declared here 151 | static int eputc (); | ^~~~~ midifile.c:1019:3: error: too many arguments to function ‘WriteVarLen’; expected 0, have 1 1019 | WriteVarLen (size); | ^~~~~~~~~~~ ~~~~ midifile.c:157:6: note: declared here 157 | void WriteVarLen (); | ^~~~~~~~~~~ midifile.c:1023:11: error: too many arguments to function ‘eputc’; expected 0, have 1 1023 | if (eputc (data[i]) != data[i]) | ^~~~~ ~~~~~~~ midifile.c:151:12: note: declared here 151 | static int eputc (); | ^~~~~ midifile.c: In function ‘mf_write_tempo’: midifile.c:1030:1: warning: old-style function definition [-Wold-style-definition] 1030 | mf_write_tempo (delta_time, tempo) | ^~~~~~~~~~~~~~ midifile.c:1033:1: error: number of arguments doesn’t match prototype 1033 | { | ^ midifile.c:155:6: error: prototype declaration 155 | void mf_write_tempo (); | ^~~~~~~~~~~~~~ midifile.c:1038:3: error: too many arguments to function ‘WriteVarLen’; expected 0, have 1 1038 | WriteVarLen (delta_time); | ^~~~~~~~~~~ ~~~~~~~~~~ midifile.c:157:6: note: declared here 157 | void WriteVarLen (); | ^~~~~~~~~~~ midifile.c:1039:3: error: too many arguments to function ‘eputc’; expected 0, have 1 1039 | eputc (laststatus = meta_event); | ^~~~~ midifile.c:151:12: note: declared here 151 | static int eputc (); | ^~~~~ midifile.c:1040:3: error: too many arguments to function ‘eputc’; expected 0, have 1 1040 | eputc (set_tempo); | ^~~~~ midifile.c:151:12: note: declared here 151 | static int eputc (); | ^~~~~ midifile.c:1042:3: error: too many arguments to function ‘eputc’; expected 0, have 1 1042 | eputc (3); | ^~~~~ ~ midifile.c:151:12: note: declared here 151 | static int eputc (); | ^~~~~ midifile.c:1043:3: error: too many arguments to function ‘eputc’; expected 0, have 1 1043 | eputc ((unsigned) (0xff & (tempo >> 16))); | ^~~~~ ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ midifile.c:151:12: note: declared here 151 | static int eputc (); | ^~~~~ midifile.c:1044:3: error: too many arguments to function ‘eputc’; expected 0, have 1 1044 | eputc ((unsigned) (0xff & (tempo >> 8))); | ^~~~~ ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ midifile.c:151:12: note: declared here 151 | static int eputc (); | ^~~~~ midifile.c:1045:3: error: too many arguments to function ‘eputc’; expected 0, have 1 1045 | eputc ((unsigned) (0xff & tempo)); | ^~~~~ ~~~~~~~~~~~~~~~~~~~~~~~~~ midifile.c:151:12: note: declared here 151 | static int eputc (); | ^~~~~ midifile.c: In function ‘mf_write_seqnum’: midifile.c:1049:1: warning: old-style function definition [-Wold-style-definition] 1049 | mf_write_seqnum (delta_time, seqnum) | ^~~~~~~~~~~~~~~ midifile.c:1052:1: error: number of arguments doesn’t match prototype 1052 | { | ^ midifile.c:156:6: error: prototype declaration 156 | void mf_write_seqnum (); | ^~~~~~~~~~~~~~~ midifile.c:1054:3: error: too many arguments to function ‘WriteVarLen’; expected 0, have 1 1054 | WriteVarLen (delta_time); | ^~~~~~~~~~~ ~~~~~~~~~~ midifile.c:157:6: note: declared here 157 | void WriteVarLen (); | ^~~~~~~~~~~ midifile.c:1055:3: error: too many arguments to function ‘eputc’; expected 0, have 1 1055 | eputc (laststatus = meta_event); | ^~~~~ midifile.c:151:12: note: declared here 151 | static int eputc (); | ^~~~~ midifile.c:1056:3: error: too many arguments to function ‘eputc’; expected 0, have 1 1056 | eputc (0); | ^~~~~ ~ midifile.c:151:12: note: declared here 151 | static int eputc (); | ^~~~~ midifile.c:1058:3: error: too many arguments to function ‘eputc’; expected 0, have 1 1058 | eputc ((unsigned) (0xff & (seqnum >> 8))); | ^~~~~ ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ midifile.c:151:12: note: declared here 151 | static int eputc (); | ^~~~~ midifile.c:1059:3: error: too many arguments to function ‘eputc’; expected 0, have 1 1059 | eputc ((unsigned) (0xff & seqnum)); | ^~~~~ ~~~~~~~~~~~~~~~~~~~~~~~~~~ midifile.c:151:12: note: declared here 151 | static int eputc (); | ^~~~~ midifile.c: In function ‘mf_sec2ticks’: midifile.c:1063:1: warning: old-style function definition [-Wold-style-definition] 1063 | mf_sec2ticks (secs, division, tempo) | ^~~~~~~~~~~~ midifile.c:1067:1: error: number of arguments doesn’t match prototype 1067 | { | ^ midifile.h:41:22: error: prototype declaration 41 | extern unsigned long mf_sec2ticks(); | ^~~~~~~~~~~~ midifile.c: In function ‘WriteVarLen’: midifile.c:1075:1: warning: old-style function definition [-Wold-style-definition] 1075 | WriteVarLen (value) | ^~~~~~~~~~~ midifile.c:1077:1: error: number of arguments doesn’t match prototype 1077 | { | ^ midifile.c:157:6: error: prototype declaration 157 | void WriteVarLen (); | ^~~~~~~~~~~ midifile.c:1089:7: error: too many arguments to function ‘eputc’; expected 0, have 1 1089 | eputc ((unsigned) (buffer & 0xff)); | ^~~~~ ~~~~~~~~~~~~~~~~~~~~~~~~~~ midifile.c:151:12: note: declared here 151 | static int eputc (); | ^~~~~ midifile.c: In function ‘mf_ticks2sec’: midifile.c:1105:1: warning: old-style function definition [-Wold-style-definition] 1105 | mf_ticks2sec (ticks, division, tempo) | ^~~~~~~~~~~~ midifile.c: In function ‘write32bit’: midifile.c:917:23: warning: old-style function definition [-Wold-style-definition] 917 | void write16bit (), write32bit (); | ^~~~~~~~~~ midifile.c:1138:1: error: number of arguments doesn’t match prototype 1138 | { | ^ midifile.c:917:23: error: prototype declaration 917 | void write16bit (), write32bit (); | ^~~~~~~~~~ midifile.c:1139:3: error: too many arguments to function ‘eputc’; expected 0, have 1 1139 | eputc ((unsigned) ((data >> 24) & 0xff)); | ^~~~~ ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ midifile.c:151:12: note: declared here 151 | static int eputc (); | ^~~~~ midifile.c:1140:3: error: too many arguments to function ‘eputc’; expected 0, have 1 1140 | eputc ((unsigned) ((data >> 16) & 0xff)); | ^~~~~ ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ midifile.c:151:12: note: declared here 151 | static int eputc (); | ^~~~~ midifile.c:1141:3: error: too many arguments to function ‘eputc’; expected 0, have 1 1141 | eputc ((unsigned) ((data >> 8) & 0xff)); | ^~~~~ ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ midifile.c:151:12: note: declared here 151 | static int eputc (); | ^~~~~ midifile.c:1142:3: error: too many arguments to function ‘eputc’; expected 0, have 1 1142 | eputc ((unsigned) (data & 0xff)); | ^~~~~ ~~~~~~~~~~~~~~~~~~~~~~~~ midifile.c:151:12: note: declared here 151 | static int eputc (); | ^~~~~ midifile.c: In function ‘write16bit’: midifile.c:917:8: warning: old-style function definition [-Wold-style-definition] 917 | void write16bit (), write32bit (); | ^~~~~~~~~~ midifile.c:1148:1: error: number of arguments doesn’t match prototype 1148 | { | ^ midifile.c:917:8: error: prototype declaration 917 | void write16bit (), write32bit (); | ^~~~~~~~~~ midifile.c:1149:3: error: too many arguments to function ‘eputc’; expected 0, have 1 1149 | eputc ((unsigned) ((data & 0xff00) >> 8)); | ^~~~~ ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ midifile.c:151:12: note: declared here 151 | static int eputc (); | ^~~~~ midifile.c:1150:3: error: too many arguments to function ‘eputc’; expected 0, have 1 1150 | eputc ((unsigned) (data & 0xff)); | ^~~~~ ~~~~~~~~~~~~~~~~~~~~~~~~ midifile.c:151:12: note: declared here 151 | static int eputc (); | ^~~~~ midifile.c: In function ‘eputc’: midifile.c:1155:1: warning: old-style function definition [-Wold-style-definition] 1155 | eputc (c) | ^~~~~ midifile.c:1166:17: error: too many arguments to function ‘Mf_putc’; expected 0, have 1 1166 | return_val = (*Mf_putc) (c); | ~^~~~~~~~~ ~ playmidi1.c: In function ‘main’: playmidi1.c:591:19: error: assignment to ‘void (*)(void)’ from incompatible pointer type ‘void (*)(int, int, int)’ [-Wincompatible-pointer-types] 591 | Mf_header = do_header; | ^ playmidi1.c:151:13: note: ‘do_header’ declared here 151 | static void do_header(int format, int ntracks, int division) | ^~~~~~~~~ playmidi1.c:592:18: error: assignment to ‘void (*)(void)’ from incompatible pointer type ‘void (*)(int)’ [-Wincompatible-pointer-types] 592 | Mf_tempo = do_tempo; | ^ playmidi1.c:246:13: note: ‘do_tempo’ declared here 246 | static void do_tempo(int us) | ^~~~~~~~ playmidi1.c:594:17: error: assignment to ‘void (*)(void)’ from incompatible pointer type ‘void (*)(int, int, char *)’ [-Wincompatible-pointer-types] 594 | Mf_text = mytext; | ^ playmidi1.c:139:13: note: ‘mytext’ declared here 139 | static void mytext(int type ATTRIBUTE_UNUSED, int leng, char *msg) | ^~~~~~ playmidi1.c:596:19: error: assignment to ‘void (*)(void)’ from incompatible pointer type ‘void (*)(int, int, int)’ [-Wincompatible-pointer-types] 596 | Mf_noteon = do_noteon; | ^ playmidi1.c:266:13: note: ‘do_noteon’ declared here 266 | static void do_noteon(int chan, int pitch, int vol) | ^~~~~~~~~ playmidi1.c:597:20: error: assignment to ‘void (*)(void)’ from incompatible pointer type ‘void (*)(int, int, int)’ [-Wincompatible-pointer-types] 597 | Mf_noteoff = do_noteoff; | ^ playmidi1.c:278:13: note: ‘do_noteoff’ declared here 278 | static void do_noteoff(int chan, int pitch, int vol) | ^~~~~~~~~~ playmidi1.c:598:20: error: assignment to ‘void (*)(void)’ from incompatible pointer type ‘void (*)(int, int)’ [-Wincompatible-pointer-types] 598 | Mf_program = do_program; | ^ playmidi1.c:290:13: note: ‘do_program’ declared here 290 | static void do_program(int chan, int program) | ^~~~~~~~~~ playmidi1.c:599:22: error: assignment to ‘void (*)(void)’ from incompatible pointer type ‘void (*)(int, int, int)’ [-Wincompatible-pointer-types] 599 | Mf_parameter = do_parameter; | ^ playmidi1.c:302:13: note: ‘do_parameter’ declared here 302 | static void do_parameter(int chan, int control, int value) | ^~~~~~~~~~~~ playmidi1.c:600:22: error: assignment to ‘void (*)(void)’ from incompatible pointer type ‘void (*)(int, int, int)’ [-Wincompatible-pointer-types] 600 | Mf_pitchbend = do_pitchbend; | ^ playmidi1.c:314:13: note: ‘do_pitchbend’ declared here 314 | static void do_pitchbend(int chan, int lsb, int msb) | ^~~~~~~~~~~~ playmidi1.c:601:21: error: assignment to ‘void (*)(void)’ from incompatible pointer type ‘void (*)(int, int, int)’ [-Wincompatible-pointer-types] 601 | Mf_pressure = do_pressure; | ^ playmidi1.c:325:13: note: ‘do_pressure’ declared here 325 | static void do_pressure(int chan, int pitch, int pressure) | ^~~~~~~~~~~ playmidi1.c:602:25: error: assignment to ‘void (*)(void)’ from incompatible pointer type ‘void (*)(int, int)’ [-Wincompatible-pointer-types] 602 | Mf_chanpressure = do_chanpressure; | ^ playmidi1.c:336:13: note: ‘do_chanpressure’ declared here 336 | static void do_chanpressure(int chan, int pressure) | ^~~~~~~~~~~~~~~ playmidi1.c:603:18: error: assignment to ‘void (*)(void)’ from incompatible pointer type ‘void (*)(int, char *)’ [-Wincompatible-pointer-types] 603 | Mf_sysex = do_sysex; | ^ playmidi1.c:347:13: note: ‘do_sysex’ declared here 347 | static void do_sysex(int len, char *msg) | ^~~~~~~~ midifile.c: At top level: midifile.c:137:13: warning: ‘to32bit’ used but never defined 137 | static long to32bit (); | ^~~~~~~ midifile.c:144:13: warning: ‘metaevent’ used but never defined 144 | static void metaevent (); | ^~~~~~~~~ midifile.c:515:1: warning: ‘metaevent’ defined but not used [-Wunused-function] 515 | metaevent (int type) | ^~~~~~~~~ make[3]: *** [Makefile:626: playmidi1.o] Error 1 make[3]: Target 'playmidi1' not remade because of errors. make[3]: 'timer' is up to date. make[3]: 'rawmidi' is up to date. make[3]: 'midiloop' is up to date. make[3]: 'umpinfo' is up to date. make[3]: 'oldapi' is up to date. make[3]: 'queue_timer' is up to date. make[3]: 'namehint' is up to date. make[3]: 'client_event_filter' is up to date. make[3]: 'chmap' is up to date. make[3]: 'audio_time' is up to date. make[3]: 'user-ctl-element-set' is up to date. make[3]: 'pcm-multi-thread' is up to date. make[3]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.14/test' make[2]: *** [Makefile:819: check-am] Error 2 make[2]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.14/test' Making check in lsb make[2]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.14/test/lsb' make config midi_event make[3]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.14/test/lsb' make[3]: 'config' is up to date. make[3]: 'midi_event' is up to date. make[3]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.14/test/lsb' make check-TESTS make[3]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.14/test/lsb' make[4]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.14/test/lsb' PASS: config PASS: midi_event ============================================================================ Testsuite summary for alsa-lib 1.2.14 ============================================================================ # TOTAL: 2 # PASS: 2 # SKIP: 0 # XFAIL: 0 # FAIL: 0 # XPASS: 0 # ERROR: 0 ============================================================================ make[4]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.14/test/lsb' make[3]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.14/test/lsb' make[2]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.14/test/lsb' make[1]: *** [Makefile:669: check-recursive] Error 1 make[1]: Target 'check' not remade because of errors. make[1]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.14/test' Making check in utils make[1]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.14/utils' make[1]: Nothing to be done for 'check'. make[1]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.14/utils' make[1]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.14' make[1]: Nothing to be done for 'check-am'. make[1]: Leaving directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.2.14' make: *** [Makefile:412: check-recursive] Error 1 make: Target 'check' not remade because of errors. ``` </details> Issue URL : https://github.com/alsa-project/alsa-lib/issues/452 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH v2 0/5] Add DisplayPort sound support for Fairphone 5 smartphone
by Luca Weiss 23 Apr '25

23 Apr '25

Add the necessary sound card bits and some dts additions to enable sound over DisplayPort-over-USB-C, e.g. to a connected TV or monitor. The UCM files can be found here: https://gitlab.postmarketos.org/postmarketOS/pmaports/-/tree/master/device/… This series - in spirit - depends on the series enabling DisplayPort in the first place, but can land pretty independently, especially the ASoC bits: https://lore.kernel.org/linux-arm-msm/20250312-fp5-pmic-glink-dp-v2-0-a5592… Signed-off-by: Luca Weiss <luca.weiss(a)fairphone.com> --- Changes in v2: - Revamp series based on comments on v1, doesn't have much too much in common anymore - Use sm8250 instead of sc8280xp sndcard file, so port other required changes from sc8280xp.c to sm8250.c - This also changes the sound card compatible to from qcom,qcm6490-sndcard to fairphone,fp5-sndcard - Link to v1: https://lore.kernel.org/r/20240809-fp5-dp-sound-v1-0-d7ba2c24f6b9@fairphone… --- Luca Weiss (5): ASoC: dt-bindings: qcom,sm8250: Add Fairphone 5 sound card ASoC: qcom: sm8250: set card driver name from match data ASoC: qcom: sm8250: add DisplayPort Jack support ASoC: qcom: sm8250: Add Fairphone 5 soundcard compatible arm64: dts: qcom: qcm6490-fairphone-fp5: Add DisplayPort sound support .../devicetree/bindings/sound/qcom,sm8250.yaml | 1 + arch/arm64/boot/dts/qcom/qcm6490-fairphone-fp5.dts | 31 ++++++++++++++++++++++ sound/soc/qcom/sm8250.c | 28 +++++++++++-------- 3 files changed, 49 insertions(+), 11 deletions(-) --- base-commit: 7f1dd4e6634d085f3c9652d4e0f1903659fb96f9 change-id: 20240809-fp5-dp-sound-b3768f3019bd Best regards, -- Luca Weiss <luca.weiss(a)fairphone.com>

4 12

[PATCH] ALSA: usb-audio: Add Pioneer DJ DJM-V10 support
by František Kučera 22 Apr '25

22 Apr '25

From: František Kučera <franta-linux(a)frantovo.cz> Pioneer DJ DJM-V10 is a high end mixer that acts like a USB sound card. The MIDI controller part is standard but the PCM part is "vendor specific". Input and output is enabled by this patch: - 12 channels (6 stereo) in each direction - 44 100 / 48 000 / 96 000 Hz - S24_3LE Capture channels can be configured through alsamixer. Playback channels are fixed and controlled by hardware knobs on the mixer. Signed-off-by: František Kučera <franta-linux(a)frantovo.cz> --- sound/usb/mixer_quirks.c | 62 ++++++++++++++++++++++++++++++++++++++-- sound/usb/quirks-table.h | 51 +++++++++++++++++++++++++++++++++ sound/usb/quirks.c | 1 + 3 files changed, 111 insertions(+), 3 deletions(-) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index a90673d43822..01091c3a83fc 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -3525,10 +3525,13 @@ static int snd_rme_digiface_controls_create(struct usb_mixer_interface *mixer) #define SND_DJM_CAP_RECOUT 0x0a #define SND_DJM_CAP_RECOUT_NOMIC 0x0e #define SND_DJM_CAP_NONE 0x0f +#define SND_DJM_CAP_FXSEND 0x10 #define SND_DJM_CAP_CH1PFADER 0x11 #define SND_DJM_CAP_CH2PFADER 0x12 #define SND_DJM_CAP_CH3PFADER 0x13 #define SND_DJM_CAP_CH4PFADER 0x14 +#define SND_DJM_CAP_EXT1SEND 0x21 +#define SND_DJM_CAP_EXT2SEND 0x22 #define SND_DJM_CAP_CH1PREFADER 0x31 #define SND_DJM_CAP_CH2PREFADER 0x32 #define SND_DJM_CAP_CH3PREFADER 0x33 @@ -3559,6 +3562,7 @@ static int snd_rme_digiface_controls_create(struct usb_mixer_interface *mixer) #define SND_DJM_750MK2_IDX 0x4 #define SND_DJM_450_IDX 0x5 #define SND_DJM_A9_IDX 0x6 +#define SND_DJM_V10_IDX 0x7 #define SND_DJM_CTL(_name, suffix, _default_value, _windex) { \ @@ -3598,8 +3602,8 @@ static const char *snd_djm_get_label_caplevel_common(u16 wvalue) } }; -// The DJM-A9 has different capture levels than other, older models -static const char *snd_djm_get_label_caplevel_a9(u16 wvalue) +// Models like DJM-A9 or DJM-V10 have different capture levels than others +static const char *snd_djm_get_label_caplevel_high(u16 wvalue) { switch (wvalue) { case 0x0000: return "+15dB"; @@ -3627,6 +3631,7 @@ static const char *snd_djm_get_label_cap_common(u16 wvalue) case SND_DJM_CAP_RECOUT_NOMIC: return "Rec Out without Mic"; case SND_DJM_CAP_AUX: return "Aux"; case SND_DJM_CAP_NONE: return "None"; + case SND_DJM_CAP_FXSEND: return "FX SEND"; case SND_DJM_CAP_CH1PREFADER: return "Pre Fader Ch1"; case SND_DJM_CAP_CH2PREFADER: return "Pre Fader Ch2"; case SND_DJM_CAP_CH3PREFADER: return "Pre Fader Ch3"; @@ -3635,6 +3640,8 @@ static const char *snd_djm_get_label_cap_common(u16 wvalue) case SND_DJM_CAP_CH2PFADER: return "Post Fader Ch2"; case SND_DJM_CAP_CH3PFADER: return "Post Fader Ch3"; case SND_DJM_CAP_CH4PFADER: return "Post Fader Ch4"; + case SND_DJM_CAP_EXT1SEND: return "EXT1 SEND"; + case SND_DJM_CAP_EXT2SEND: return "EXT2 SEND"; default: return NULL; } }; @@ -3653,7 +3660,8 @@ static const char *snd_djm_get_label_cap_850(u16 wvalue) static const char *snd_djm_get_label_caplevel(u8 device_idx, u16 wvalue) { switch (device_idx) { - case SND_DJM_A9_IDX: return snd_djm_get_label_caplevel_a9(wvalue); + case SND_DJM_A9_IDX: return snd_djm_get_label_caplevel_high(wvalue); + case SND_DJM_V10_IDX: return snd_djm_get_label_caplevel_high(wvalue); default: return snd_djm_get_label_caplevel_common(wvalue); } }; @@ -3853,6 +3861,50 @@ static const struct snd_djm_ctl snd_djm_ctls_a9[] = { SND_DJM_CTL("Ch4 Input", a9_cap5, 2, SND_DJM_WINDEX_CAP) }; +// DJM-V10 +static const u16 snd_djm_opts_v10_cap_level[] = { + 0x0000, 0x0100, 0x0200, 0x0300, 0x0400, 0x0500 +}; +static const u16 snd_djm_opts_v10_cap1[] = { + 0x0103, + 0x0100, 0x0102, 0x0106, 0x0110, 0x0107, + 0x0108, 0x0109, 0x010a, 0x0121, 0x0122 +}; +static const u16 snd_djm_opts_v10_cap2[] = { + 0x0200, 0x0202, 0x0206, 0x0210, 0x0207, + 0x0208, 0x0209, 0x020a, 0x0221, 0x0222 +}; +static const u16 snd_djm_opts_v10_cap3[] = { + 0x0303, + 0x0300, 0x0302, 0x0306, 0x0310, 0x0307, + 0x0308, 0x0309, 0x030a, 0x0321, 0x0322 +}; +static const u16 snd_djm_opts_v10_cap4[] = { + 0x0403, + 0x0400, 0x0402, 0x0406, 0x0410, 0x0407, + 0x0408, 0x0409, 0x040a, 0x0421, 0x0422 +}; +static const u16 snd_djm_opts_v10_cap5[] = { + 0x0500, 0x0502, 0x0506, 0x0510, 0x0507, + 0x0508, 0x0509, 0x050a, 0x0521, 0x0522 +}; +static const u16 snd_djm_opts_v10_cap6[] = { + 0x0603, + 0x0600, 0x0602, 0x0606, 0x0610, 0x0607, + 0x0608, 0x0609, 0x060a, 0x0621, 0x0622 +}; + +static const struct snd_djm_ctl snd_djm_ctls_v10[] = { + SND_DJM_CTL("Master Capture Level Capture Switch", v10_cap_level, 0, SND_DJM_WINDEX_CAPLVL), + SND_DJM_CTL("Capture Ch1 Capture Switch", v10_cap1, 2, SND_DJM_WINDEX_CAP), + SND_DJM_CTL("Capture Ch2 Capture Switch", v10_cap2, 2, SND_DJM_WINDEX_CAP), + SND_DJM_CTL("Capture Ch3 Capture Switch", v10_cap3, 0, SND_DJM_WINDEX_CAP), + SND_DJM_CTL("Capture Ch4 Capture Switch", v10_cap4, 0, SND_DJM_WINDEX_CAP), + SND_DJM_CTL("Capture Ch5 Capture Switch", v10_cap5, 0, SND_DJM_WINDEX_CAP), + SND_DJM_CTL("Capture Ch6 Capture Switch", v10_cap6, 0, SND_DJM_WINDEX_CAP) + // playback channels are fixed and controlled by hardware knobs on the mixer +}; + static const struct snd_djm_device snd_djm_devices[] = { [SND_DJM_250MK2_IDX] = SND_DJM_DEVICE(250mk2), [SND_DJM_750_IDX] = SND_DJM_DEVICE(750), @@ -3861,6 +3913,7 @@ static const struct snd_djm_device snd_djm_devices[] = { [SND_DJM_750MK2_IDX] = SND_DJM_DEVICE(750mk2), [SND_DJM_450_IDX] = SND_DJM_DEVICE(450), [SND_DJM_A9_IDX] = SND_DJM_DEVICE(a9), + [SND_DJM_V10_IDX] = SND_DJM_DEVICE(v10), }; @@ -4151,6 +4204,9 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) case USB_ID(0x2b73, 0x003c): /* Pioneer DJ / AlphaTheta DJM-A9 */ err = snd_djm_controls_create(mixer, SND_DJM_A9_IDX); break; + case USB_ID(0x2b73, 0x0034): /* Pioneer DJ DJM-V10 */ + err = snd_djm_controls_create(mixer, SND_DJM_V10_IDX); + break; } return err; diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 8954be23325c..eafc0d73cca1 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3187,6 +3187,57 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, +{ + /* + * Pioneer DJ DJM-V10 + */ + USB_DEVICE_VENDOR_SPEC(0x2b73, 0x0034), + QUIRK_DRIVER_INFO { + QUIRK_DATA_COMPOSITE { + { + QUIRK_DATA_AUDIOFORMAT(0) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 12, // outputs + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .endpoint = 0x01, + .ep_attr = USB_ENDPOINT_XFER_ISOC| + USB_ENDPOINT_SYNC_ASYNC, + .rates = SNDRV_PCM_RATE_44100| + SNDRV_PCM_RATE_48000| + SNDRV_PCM_RATE_96000, + .rate_min = 44100, + .rate_max = 96000, + .nr_rates = 3, + .rate_table = (unsigned int[]) { 44100, 48000, 96000 } + } + }, + { + QUIRK_DATA_AUDIOFORMAT(0) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 12, // inputs + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .endpoint = 0x82, + .ep_idx = 1, + .ep_attr = USB_ENDPOINT_XFER_ISOC| + USB_ENDPOINT_SYNC_ASYNC| + USB_ENDPOINT_USAGE_IMPLICIT_FB, + .rates = SNDRV_PCM_RATE_44100| + SNDRV_PCM_RATE_48000| + SNDRV_PCM_RATE_96000, + .rate_min = 44100, + .rate_max = 96000, + .nr_rates = 3, + .rate_table = (unsigned int[]) { 44100, 48000, 96000 } + } + }, + QUIRK_COMPOSITE_END + } + } +}, /* * MacroSilicon MS2100/MS2106 based AV capture cards diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 9112313a9dbc..4dcddab1a0e3 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1872,6 +1872,7 @@ void snd_usb_set_format_quirk(struct snd_usb_substream *subs, break; case USB_ID(0x2b73, 0x000a): /* Pioneer DJM-900NXS2 */ case USB_ID(0x2b73, 0x0013): /* Pioneer DJM-450 */ + case USB_ID(0x2b73, 0x0034): /* Pioneer DJM-V10 */ pioneer_djm_set_format_quirk(subs, 0x0082); break; case USB_ID(0x08e4, 0x017f): /* Pioneer DJM-750 */ -- 2.43.0

2 1

[PATCH 0/9] ALSA: usb-audio: More fixes / improvements on PCM
by Takashi Iwai 22 Apr '25

22 Apr '25

Hi, this is a patch set for USB-audio to fix and improve the PCM stream handling. The primary target is the enhancement of the low-latency playback that was introduced recently for dealing with the small period size (that is used typically by JACK). There are a few other fixes that are relevant with it. The current patches are found in topic/usb-audio branch of sound.git tree, too. Takashi === Takashi Iwai (9): ALSA: usb-audio: Restrict rates for the shared clocks ALSA: usb-audio: Fix possible race at sync of urb completions ALSA: usb-audio: Rename early_playback_start flag with lowlatency_playback ALSA: usb-audio: Disable low-latency playback for free-wheel mode ALSA: usb-audio: Disable low-latency mode for implicit feedback sync ALSA: usb-audio: Check available frames for the next packet size ALSA: usb-audio: Add spinlock to stop_urbs() ALSA: usb-audio: Improved lowlatency playback support ALSA: usb-audio: Avoid killing in-flight URBs during draining sound/usb/card.h | 11 ++- sound/usb/endpoint.c | 230 +++++++++++++++++++++++++++++++------------ sound/usb/endpoint.h | 13 ++- sound/usb/pcm.c | 157 ++++++++++++++++++++++------- 4 files changed, 305 insertions(+), 106 deletions(-) -- 2.26.2

2 23

acp3x-alc5682-max98357: Fix path of HiFi.conf
by GitHub pull_request - opened 22 Apr '25

22 Apr '25

alsa-project/alsa-ucm-conf pull request #551 was opened from meithecatte: Closes: https://github.com/alsa-project/alsa-ucm-conf/issues/550 Fixes: 1048796e7fa9 ("Rename ucm2/AMD/acp3xalc5682m98 to ucm2/AMD/acp3x-alc5682-max98357") Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/551 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/551.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Audio doesn't work on acp3xalc5682m98357 since 1.2.14
by GitHub issues - opened 22 Apr '25

22 Apr '25

alsa-project/alsa-ucm-conf issue #550 was opened from meithecatte: Since 1.2.14, pipewire doesn't detect the audio device, and trying to use `aplay` results in ``` ALSA lib pcm_dmix.c:1000:(snd_pcm_dmix_open) unable to open slave aplay: main:850: audio open error: No such file or directory ``` `aplay -l` lists ``` card 1: acp3xalc5682m98 [acp3xalc5682m98357], device 0: Playback rt5682-aif1-0 [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: acp3xalc5682m98 [acp3xalc5682m98357], device 1: HiFi Playback HiFi-1 [] Subdevices: 1/1 Subdevice #0: subdevice #0 ``` `git bisect` says 0a87bd36b77e4e22397450d7ba3f9d36ce9eb5e0 is the first bad commit. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/550 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v1] ASoC: tas2781: Add "Acoustic Tuning" kcontrol for acoustic application
by Shenghao Ding 21 Apr '25

21 Apr '25

"Acoustic Tuning" kcontrol is a bridge to the acoustic tuning application tool which can tune the chips' acoustic effect. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- include/sound/tas2781.h | 13 ++++ sound/soc/codecs/tas2781-i2c.c | 138 ++++++++++++++++++++++++++++++++- 2 files changed, 149 insertions(+), 2 deletions(-) diff --git a/include/sound/tas2781.h b/include/sound/tas2781.h index eff011444cc8..a9c236ab8cba 100644 --- a/include/sound/tas2781.h +++ b/include/sound/tas2781.h @@ -32,6 +32,8 @@ SNDRV_PCM_FMTBIT_S24_LE | \ SNDRV_PCM_FMTBIT_S32_LE) +#define TASDEV_DATA_PAYLOAD_SIZE 128 + /* PAGE Control Register (available in page0 of each book) */ #define TASDEVICE_PAGE_SELECT 0x00 #define TASDEVICE_BOOKCTL_PAGE 0x00 @@ -159,10 +161,21 @@ struct calidata { unsigned int cali_dat_sz_per_dev; }; +struct acoustic_data { + unsigned char len; + unsigned char id; + unsigned char addr; + unsigned char book; + unsigned char page; + unsigned char reg; + unsigned char data[TASDEV_DATA_PAYLOAD_SIZE]; +}; + struct tasdevice_priv { struct tasdevice tasdevice[TASDEVICE_MAX_CHANNELS]; struct tasdevice_rca rcabin; struct calidata cali_data; + struct acoustic_data acou_data; struct tasdevice_fw *fmw; struct gpio_desc *speaker_id; struct gpio_desc *reset; diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index b950914b7d48..0d52cfc12743 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -328,6 +328,24 @@ static int tasdev_cali_data_get(struct snd_kcontrol *kcontrol, return 0; } +static int tasdev_acoustic_data_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *priv = snd_soc_component_get_drvdata(comp); + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *) kcontrol->private_value; + unsigned char *dst = ucontrol->value.bytes.data; + struct acoustic_data *p = &priv->acou_data; + + if (p->id == 'r' && p->len <= bytes_ext->max) + memcpy(dst, p, p->len); + else + dev_err(priv->dev, "Not ready for get.\n"); + + return 0; +} + static int calib_data_get(struct tasdevice_priv *tas_priv, int reg, unsigned char *dst) { @@ -700,6 +718,84 @@ static int tasdev_tf_data_get(struct snd_kcontrol *kcontrol, return calib_data_get(tas_priv, reg, &dst[1]); } +static int tasdev_acoustic_data_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *priv = snd_soc_component_get_drvdata(comp); + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *) kcontrol->private_value; + unsigned char *src = ucontrol->value.bytes.data; + int j, len, ret, value, reg; + unsigned short chn; + + if (src[0] > bytes_ext->max) { + dev_err(priv->dev, "pkg len(%u) is larger than max(%d).\n", + src[0], bytes_ext->max); + return 0; + } + + switch (src[1]) { + case 'r': + /* length of data to read */ + len = src[6]; + break; + case 'w': + /* Skip 6 bytes for package type and register address */ + len = src[0] - 6; + break; + default: + dev_err(priv->dev, "%s Wrong code %02x.\n", __func__, src[1]); + return 0; + } + + if (len < 1) { + dev_err(priv->dev, "pkg fmt invalid %02x.\n", len); + return 0; + } + + for (j = 0; j < priv->ndev; j++) + if (src[2] == priv->tasdevice[j].dev_addr) { + chn = j; + break; + } + if (j >= priv->ndev) { + dev_err(priv->dev, "no such device 0x%02x.\n", src[2]); + return 0; + } + + reg = TASDEVICE_REG(src[3], src[4], src[5]); + + guard(mutex)(&priv->codec_lock); + + if (src[1] == 'w') { + if (len > 1) + ret = tasdevice_dev_bulk_write(priv, chn, reg, + &src[6], len); + else + ret = tasdevice_dev_write(priv, chn, reg, src[6]); + } else { + struct acoustic_data *p = &priv->acou_data; + + memcpy(p, src, 6); + if (len > 1) { + ret = tasdevice_dev_bulk_read(priv, chn, reg, + p->data, len); + } else { + ret = tasdevice_dev_read(priv, chn, reg, &value); + p->data[0] = value; + } + p->len = len + 6; + } + + if (ret) { + dev_err(priv->dev, "i2c communication error.\n"); + return 0; + } + + return 1; +} + static int tasdev_re_data_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -1043,10 +1139,11 @@ static int tasdevice_get_chip_id(struct snd_kcontrol *kcontrol, static int tasdevice_create_control(struct tasdevice_priv *tas_priv) { struct snd_kcontrol_new *prof_ctrls; - int nr_controls = 1; + struct soc_bytes_ext *ext_acoustic; + char *acoustic_name, *name; + int nr_controls = 2; int mix_index = 0; int ret; - char *name; prof_ctrls = devm_kcalloc(tas_priv->dev, nr_controls, sizeof(prof_ctrls[0]), GFP_KERNEL); @@ -1068,6 +1165,43 @@ static int tasdevice_create_control(struct tasdevice_priv *tas_priv) prof_ctrls[mix_index].put = tasdevice_set_profile_id; mix_index++; + ext_acoustic = devm_kzalloc(tas_priv->dev, sizeof(*ext_acoustic), + GFP_KERNEL); + if (!ext_acoustic) + return -ENOMEM; + + /* + * This kcontrol is a bridge to the acoustic tuning application + * tool which can tune the chips' acoustic effect. + */ + acoustic_name = devm_kstrdup(tas_priv->dev, "Acoustic Tuning", + GFP_KERNEL); + if (!acoustic_name) + return -ENOMEM; + /* + * package structure for PPC3 communications: + * Pkg len (1 byte) + * Pkg id (1 byte, 'r' or 'w') + * Dev id (1 byte, i2c address) + * Book id (1 byte) + * Page id (1 byte) + * Reg id (1 byte) + * switch (pkg id) { + * case 'w': + * 1 byte, length of data to read + * case 'r': + * data payload (1~128 bytes) + * } + */ + ext_acoustic->max = sizeof(tas_priv->acou_data); + prof_ctrls[mix_index].name = acoustic_name; + prof_ctrls[mix_index].iface = SNDRV_CTL_ELEM_IFACE_MIXER; + prof_ctrls[mix_index].info = snd_soc_bytes_info_ext; + prof_ctrls[mix_index].put = tasdev_acoustic_data_put; + prof_ctrls[mix_index].get = tasdev_acoustic_data_get; + prof_ctrls[mix_index].private_value = (unsigned long)ext_acoustic; + mix_index++; + ret = snd_soc_add_component_controls(tas_priv->codec, prof_ctrls, nr_controls < mix_index ? nr_controls : mix_index); -- 2.34.1

3 5

Speaker can't mute when headphone was plugged
by Kailang 20 Apr '25

20 Apr '25

Hi Takashi, Speaker can't mute when headphone was plugged for HP platform. Add attach patch will solve issue. BR, Kailang

2 1

ucm2: MediaTek: mt8395-evk: Add support for SOF
by GitHub pull_request - opened 17 Apr '25

17 Apr '25

alsa-project/alsa-ucm-conf pull request #549 was opened from julien-massot: Similar to #513, add support for MT8395 with SOF enabled. We use different playback and capture PCM. Also, split the mixers that are not yet available upstream. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/549 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/549.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

No audio on Lenovo Yoga Slim 7 14ILL10
by GitHub issues - edited 16 Apr '25

16 Apr '25

alsa-project/alsa-ucm-conf issue #548 was edited from miguel1117: Only "Dummy Output" is available as an output device on the Lenovo Yoga Slim 7 14ILL10. I'm not absolutely sure if this is the right place to report this so apologies if it's not. [ALSA information](https://alsa-project.org/db/?f=71342ec47d52681c81335d75d2947d8… Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/548 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

No audio on Lenovo Yoga Slim 7 14ILL10
by GitHub issues - edited 16 Apr '25

16 Apr '25

alsa-project/alsa-ucm-conf issue #548 was edited from miguel1117: Only "Dummy Output" is available as an output device on the Lenovo Yoga Slim 7 14ILL1. I'm not absolutely sure if this is the right place to report this so apologies if it's not. [ALSA information](https://alsa-project.org/db/?f=71342ec47d52681c81335d75d2947d8… Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/548 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v3] ALSA: hda/tas2781: Create a common header for both spi and i2c tas2781 hda driver
by Shenghao Ding 16 Apr '25

16 Apr '25

Move the common macro definition of kcontrols into a common header for code cleanup, and create a common header to store the common declaration for both spi and i2c hda driver. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v3: - Add the purpose of this patch. v2: - Follow IWYU principle, add sound/asound.h into the header file. v1: - Revise the year of spi and i2c tas2781 hda drivers. - Create a common header for both spi and i2c tas2781 hda driver to define the common macros and declare the common functions. --- sound/pci/hda/tas2781_hda.h | 44 +++++++++++++++++++++++++++++++++ sound/pci/hda/tas2781_hda_i2c.c | 29 ++-------------------- sound/pci/hda/tas2781_hda_spi.c | 35 ++------------------------ 3 files changed, 48 insertions(+), 60 deletions(-) create mode 100644 sound/pci/hda/tas2781_hda.h diff --git a/sound/pci/hda/tas2781_hda.h b/sound/pci/hda/tas2781_hda.h new file mode 100644 index 000000000000..fc741fac419a --- /dev/null +++ b/sound/pci/hda/tas2781_hda.h @@ -0,0 +1,44 @@ +/* SPDX-License-Identifier: GPL-2.0-only + * + * HDA audio driver for Texas Instruments TAS2781 smart amp + * + * Copyright (C) 2025 Texas Instruments, Inc. + */ +#ifndef __TAS2781_HDA_H__ +#define __TAS2781_HDA_H__ + +#include <sound/asound.h> + +/* + * No standard control callbacks for SNDRV_CTL_ELEM_IFACE_CARD + * Define two controls, one is Volume control callbacks, the other is + * flag setting control callbacks. + */ + +/* Volume control callbacks for tas2781 */ +#define ACARD_SINGLE_RANGE_EXT_TLV(xname, xreg, xshift, xmin, xmax, xinvert, \ + xhandler_get, xhandler_put, tlv_array) { \ + .iface = SNDRV_CTL_ELEM_IFACE_CARD, .name = (xname), \ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ + SNDRV_CTL_ELEM_ACCESS_READWRITE, \ + .tlv.p = (tlv_array), \ + .info = snd_soc_info_volsw, \ + .get = xhandler_get, .put = xhandler_put, \ + .private_value = (unsigned long)&(struct soc_mixer_control) { \ + .reg = xreg, .rreg = xreg, \ + .shift = xshift, .rshift = xshift,\ + .min = xmin, .max = xmax, .invert = xinvert, \ + } \ +} + +/* Flag control callbacks for tas2781 */ +#define ACARD_SINGLE_BOOL_EXT(xname, xdata, xhandler_get, xhandler_put) { \ + .iface = SNDRV_CTL_ELEM_IFACE_CARD, \ + .name = xname, \ + .info = snd_ctl_boolean_mono_info, \ + .get = xhandler_get, \ + .put = xhandler_put, \ + .private_value = xdata, \ +} + +#endif diff --git a/sound/pci/hda/tas2781_hda_i2c.c b/sound/pci/hda/tas2781_hda_i2c.c index 29dc4f500580..9d94ae5fcfe0 100644 --- a/sound/pci/hda/tas2781_hda_i2c.c +++ b/sound/pci/hda/tas2781_hda_i2c.c @@ -2,7 +2,7 @@ // // TAS2781 HDA I2C driver // -// Copyright 2023 - 2024 Texas Instruments, Inc. +// Copyright 2023 - 2025 Texas Instruments, Inc. // // Author: Shenghao Ding <shenghao-ding(a)ti.com> // Current maintainer: Baojun Xu <baojun.xu(a)ti.com> @@ -30,35 +30,10 @@ #include "hda_component.h" #include "hda_jack.h" #include "hda_generic.h" +#include "tas2781_hda.h" #define TASDEVICE_SPEAKER_CALIBRATION_SIZE 20 -/* No standard control callbacks for SNDRV_CTL_ELEM_IFACE_CARD - * Define two controls, one is Volume control callbacks, the other is - * flag setting control callbacks. - */ - -/* Volume control callbacks for tas2781 */ -#define ACARD_SINGLE_RANGE_EXT_TLV(xname, xreg, xshift, xmin, xmax, xinvert, \ - xhandler_get, xhandler_put, tlv_array) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_CARD, .name = (xname),\ - .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\ - SNDRV_CTL_ELEM_ACCESS_READWRITE,\ - .tlv.p = (tlv_array), \ - .info = snd_soc_info_volsw, \ - .get = xhandler_get, .put = xhandler_put, \ - .private_value = (unsigned long)&(struct soc_mixer_control) \ - {.reg = xreg, .rreg = xreg, .shift = xshift, \ - .rshift = xshift, .min = xmin, .max = xmax, \ - .invert = xinvert} } - -/* Flag control callbacks for tas2781 */ -#define ACARD_SINGLE_BOOL_EXT(xname, xdata, xhandler_get, xhandler_put) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_CARD, .name = xname, \ - .info = snd_ctl_boolean_mono_info, \ - .get = xhandler_get, .put = xhandler_put, \ - .private_value = xdata } - enum calib_data { R0_VAL = 0, INV_R0, diff --git a/sound/pci/hda/tas2781_hda_spi.c b/sound/pci/hda/tas2781_hda_spi.c index 399f2e4c3b62..c6be2be1b53e 100644 --- a/sound/pci/hda/tas2781_hda_spi.c +++ b/sound/pci/hda/tas2781_hda_spi.c @@ -2,7 +2,7 @@ // // TAS2781 HDA SPI driver // -// Copyright 2024 Texas Instruments, Inc. +// Copyright 2024 - 2025 Texas Instruments, Inc. // // Author: Baojun Xu <baojun.xu(a)ti.com> @@ -38,38 +38,7 @@ #include "hda_component.h" #include "hda_jack.h" #include "hda_generic.h" - -/* - * No standard control callbacks for SNDRV_CTL_ELEM_IFACE_CARD - * Define two controls, one is Volume control callbacks, the other is - * flag setting control callbacks. - */ - -/* Volume control callbacks for tas2781 */ -#define ACARD_SINGLE_RANGE_EXT_TLV(xname, xreg, xshift, xmin, xmax, xinvert, \ - xhandler_get, xhandler_put, tlv_array) { \ - .iface = SNDRV_CTL_ELEM_IFACE_CARD, .name = (xname), \ - .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ - SNDRV_CTL_ELEM_ACCESS_READWRITE, \ - .tlv.p = (tlv_array), \ - .info = snd_soc_info_volsw, \ - .get = xhandler_get, .put = xhandler_put, \ - .private_value = (unsigned long)&(struct soc_mixer_control) { \ - .reg = xreg, .rreg = xreg, \ - .shift = xshift, .rshift = xshift,\ - .min = xmin, .max = xmax, .invert = xinvert, \ - } \ -} - -/* Flag control callbacks for tas2781 */ -#define ACARD_SINGLE_BOOL_EXT(xname, xdata, xhandler_get, xhandler_put) { \ - .iface = SNDRV_CTL_ELEM_IFACE_CARD, \ - .name = xname, \ - .info = snd_ctl_boolean_mono_info, \ - .get = xhandler_get, \ - .put = xhandler_put, \ - .private_value = xdata, \ -} +#include "tas2781_hda.h" struct tas2781_hda { struct tasdevice_priv *priv; -- 2.34.1

2 1

No audio on Lenovo Yoga Slim 7 14ILL1
by GitHub issues - edited 16 Apr '25

16 Apr '25

alsa-project/alsa-ucm-conf issue #548 was edited from miguel1117: Only "Dummy Output" is available as an output device on the Lenovo Yoga Slim 7 14ILL1. I'm not absolutely sure if this is the right place to report this so apologies if it's not. [ALSA information](https://alsa-project.org/db/?f=71342ec47d52681c81335d75d2947d8… Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/548 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

No audio on Lenovo Yoga Slim 7 14ILL1
by GitHub issues - opened 16 Apr '25

16 Apr '25

alsa-project/alsa-ucm-conf issue #548 was opened from miguel1117: Only "Dummy Output" is available as an output device on the Lenovo Yoga Slim 7 14ILL1. I'm not absolutely sure if this is the right place to report this so apologies if it's not. [ALSA information](https://alsa-project.org/db/?f=80c739775b022831be9c3aad038f8de… Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/548 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Steinberg UR22C: Version 1.2.14 introduced "[ALSA UCM error]" on older firmware.
by GitHub issues - edited 15 Apr '25

15 Apr '25

alsa-project/alsa-ucm-conf issue #547 was edited from NoXPhasma: Since version 1.2.14 the `node.description` of the Steinberg UR22C shows `[ALSA UCM error]`: ``` node.description = "Steinberg UR22C [ALSA UCM error] Direct Steinberg UR22C" ``` ``` > journalctl --user -u wireplumber --boot 0 systemd[1286]: Started Multimedia Service Session Manager. wp-internal-comp-loader: Loading profile 'main' spa.alsa: Error in ALSA UCM profile for _ucm0001.hw:UR22C,0 (HiFi: Line 3: source): CaptureChannels=4 > avail 2 spa.alsa: Errors in ALSA UCM profile for card Steinberg UR22C wireplumber[1359]: wp-device: SPA handle 'api.libcamera.enum.manager' could not be loaded; is it installed? wireplumber[1359]: s-monitors-libcamera: PipeWire's libcamera SPA plugin is missing or broken. Some camera types may not be supported. wireplumber[1359]: default: Failed to get percentage from UPower: org.freedesktop.DBus.Error.NameHasNoOwner wireplumber[1359]: spa.bluez5: BlueZ system service is not available ``` Reverting to version 1.2.13 fixes this issue. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/547 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Steinberg UR22C: Version 1.2.14 introduced "[ALSA UCM error]"
by GitHub issues - edited 15 Apr '25

15 Apr '25

alsa-project/alsa-ucm-conf issue #547 was edited from NoXPhasma: Since version 1.2.14 the `node.description` of the Steinberg UR22C shows `[ALSA UCM error]`: ``` node.description = "Steinberg UR22C [ALSA UCM error] Direct Steinberg UR22C" ``` ``` > journalctl --user -u wireplumber --boot 0 systemd[1286]: Started Multimedia Service Session Manager. wp-internal-comp-loader: Loading profile 'main' spa.alsa: Error in ALSA UCM profile for _ucm0001.hw:UR22C,0 (HiFi: Line 3: source): CaptureChannels=4 > avail 2 spa.alsa: Errors in ALSA UCM profile for card Steinberg UR22C wireplumber[1359]: wp-device: SPA handle 'api.libcamera.enum.manager' could not be loaded; is it installed? wireplumber[1359]: s-monitors-libcamera: PipeWire's libcamera SPA plugin is missing or broken. Some camera types may not be supported. wireplumber[1359]: default: Failed to get percentage from UPower: org.freedesktop.DBus.Error.NameHasNoOwner wireplumber[1359]: spa.bluez5: BlueZ system service is not available ``` Reverting to version 1.2.13 fixes this issue. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/547 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Steinberg UR22C: Version 1.2.14 introduced "[ALSA UCM error]"
by GitHub issues - opened 15 Apr '25

15 Apr '25

alsa-project/alsa-ucm-conf issue #547 was opened from NoXPhasma: Since version 1.2.14 the `node.description` of the Steinberg UR22C shows `[ALSA UCM error]`: ``` node.description = "Steinberg UR22C [ALSA UCM error] Direct Steinberg UR22C" ``` Reverting to version 1.2.13 fixes this issue. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/547 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[RESEND PATCH v2] ALSA: hda/tas2781: Create a common header for both spi and i2c tas2781 hda driver
by Shenghao Ding 15 Apr '25

15 Apr '25

Move the common macro definition of kcontrols into a common header. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v2: - Follow IWYU principle, add sound/asound.h into the header file. v1: - Revise the year of spi and i2c tas2781 hda drivers. - Create a common header for both spi and i2c tas2781 hda driver to define the common macros and declare the common functions. --- sound/pci/hda/tas2781_hda.h | 44 +++++++++++++++++++++++++++++++++ sound/pci/hda/tas2781_hda_i2c.c | 29 ++-------------------- sound/pci/hda/tas2781_hda_spi.c | 35 ++------------------------ 3 files changed, 48 insertions(+), 60 deletions(-) create mode 100644 sound/pci/hda/tas2781_hda.h diff --git a/sound/pci/hda/tas2781_hda.h b/sound/pci/hda/tas2781_hda.h new file mode 100644 index 000000000000..fc741fac419a --- /dev/null +++ b/sound/pci/hda/tas2781_hda.h @@ -0,0 +1,44 @@ +/* SPDX-License-Identifier: GPL-2.0-only + * + * HDA audio driver for Texas Instruments TAS2781 smart amp + * + * Copyright (C) 2025 Texas Instruments, Inc. + */ +#ifndef __TAS2781_HDA_H__ +#define __TAS2781_HDA_H__ + +#include <sound/asound.h> + +/* + * No standard control callbacks for SNDRV_CTL_ELEM_IFACE_CARD + * Define two controls, one is Volume control callbacks, the other is + * flag setting control callbacks. + */ + +/* Volume control callbacks for tas2781 */ +#define ACARD_SINGLE_RANGE_EXT_TLV(xname, xreg, xshift, xmin, xmax, xinvert, \ + xhandler_get, xhandler_put, tlv_array) { \ + .iface = SNDRV_CTL_ELEM_IFACE_CARD, .name = (xname), \ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ + SNDRV_CTL_ELEM_ACCESS_READWRITE, \ + .tlv.p = (tlv_array), \ + .info = snd_soc_info_volsw, \ + .get = xhandler_get, .put = xhandler_put, \ + .private_value = (unsigned long)&(struct soc_mixer_control) { \ + .reg = xreg, .rreg = xreg, \ + .shift = xshift, .rshift = xshift,\ + .min = xmin, .max = xmax, .invert = xinvert, \ + } \ +} + +/* Flag control callbacks for tas2781 */ +#define ACARD_SINGLE_BOOL_EXT(xname, xdata, xhandler_get, xhandler_put) { \ + .iface = SNDRV_CTL_ELEM_IFACE_CARD, \ + .name = xname, \ + .info = snd_ctl_boolean_mono_info, \ + .get = xhandler_get, \ + .put = xhandler_put, \ + .private_value = xdata, \ +} + +#endif diff --git a/sound/pci/hda/tas2781_hda_i2c.c b/sound/pci/hda/tas2781_hda_i2c.c index 29dc4f500580..9d94ae5fcfe0 100644 --- a/sound/pci/hda/tas2781_hda_i2c.c +++ b/sound/pci/hda/tas2781_hda_i2c.c @@ -2,7 +2,7 @@ // // TAS2781 HDA I2C driver // -// Copyright 2023 - 2024 Texas Instruments, Inc. +// Copyright 2023 - 2025 Texas Instruments, Inc. // // Author: Shenghao Ding <shenghao-ding(a)ti.com> // Current maintainer: Baojun Xu <baojun.xu(a)ti.com> @@ -30,35 +30,10 @@ #include "hda_component.h" #include "hda_jack.h" #include "hda_generic.h" +#include "tas2781_hda.h" #define TASDEVICE_SPEAKER_CALIBRATION_SIZE 20 -/* No standard control callbacks for SNDRV_CTL_ELEM_IFACE_CARD - * Define two controls, one is Volume control callbacks, the other is - * flag setting control callbacks. - */ - -/* Volume control callbacks for tas2781 */ -#define ACARD_SINGLE_RANGE_EXT_TLV(xname, xreg, xshift, xmin, xmax, xinvert, \ - xhandler_get, xhandler_put, tlv_array) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_CARD, .name = (xname),\ - .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\ - SNDRV_CTL_ELEM_ACCESS_READWRITE,\ - .tlv.p = (tlv_array), \ - .info = snd_soc_info_volsw, \ - .get = xhandler_get, .put = xhandler_put, \ - .private_value = (unsigned long)&(struct soc_mixer_control) \ - {.reg = xreg, .rreg = xreg, .shift = xshift, \ - .rshift = xshift, .min = xmin, .max = xmax, \ - .invert = xinvert} } - -/* Flag control callbacks for tas2781 */ -#define ACARD_SINGLE_BOOL_EXT(xname, xdata, xhandler_get, xhandler_put) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_CARD, .name = xname, \ - .info = snd_ctl_boolean_mono_info, \ - .get = xhandler_get, .put = xhandler_put, \ - .private_value = xdata } - enum calib_data { R0_VAL = 0, INV_R0, diff --git a/sound/pci/hda/tas2781_hda_spi.c b/sound/pci/hda/tas2781_hda_spi.c index 399f2e4c3b62..c6be2be1b53e 100644 --- a/sound/pci/hda/tas2781_hda_spi.c +++ b/sound/pci/hda/tas2781_hda_spi.c @@ -2,7 +2,7 @@ // // TAS2781 HDA SPI driver // -// Copyright 2024 Texas Instruments, Inc. +// Copyright 2024 - 2025 Texas Instruments, Inc. // // Author: Baojun Xu <baojun.xu(a)ti.com> @@ -38,38 +38,7 @@ #include "hda_component.h" #include "hda_jack.h" #include "hda_generic.h" - -/* - * No standard control callbacks for SNDRV_CTL_ELEM_IFACE_CARD - * Define two controls, one is Volume control callbacks, the other is - * flag setting control callbacks. - */ - -/* Volume control callbacks for tas2781 */ -#define ACARD_SINGLE_RANGE_EXT_TLV(xname, xreg, xshift, xmin, xmax, xinvert, \ - xhandler_get, xhandler_put, tlv_array) { \ - .iface = SNDRV_CTL_ELEM_IFACE_CARD, .name = (xname), \ - .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ - SNDRV_CTL_ELEM_ACCESS_READWRITE, \ - .tlv.p = (tlv_array), \ - .info = snd_soc_info_volsw, \ - .get = xhandler_get, .put = xhandler_put, \ - .private_value = (unsigned long)&(struct soc_mixer_control) { \ - .reg = xreg, .rreg = xreg, \ - .shift = xshift, .rshift = xshift,\ - .min = xmin, .max = xmax, .invert = xinvert, \ - } \ -} - -/* Flag control callbacks for tas2781 */ -#define ACARD_SINGLE_BOOL_EXT(xname, xdata, xhandler_get, xhandler_put) { \ - .iface = SNDRV_CTL_ELEM_IFACE_CARD, \ - .name = xname, \ - .info = snd_ctl_boolean_mono_info, \ - .get = xhandler_get, \ - .put = xhandler_put, \ - .private_value = xdata, \ -} +#include "tas2781_hda.h" struct tas2781_hda { struct tasdevice_priv *priv; -- 2.34.1

2 1

ALSA 1.2.8 release
by Jaroslav Kysela 15 Apr '25

15 Apr '25

Hello all, new ALSA userspace packages were released. You may download them from the ALSA website http://www.alsa-project.org or directly: HTTP: https://www.alsa-project.org/files/pub FTP: ftp://ftp.alsa-project.org/pub Released packages: alsa-lib alsa-utils alsa-ucm-conf tinycompress Full list of changes: https://www.alsa-project.org/wiki/Changes_v1.2.7.2_v1.2.8 The fingerprint of the public signing key is: F04D F507 37AC 1A88 4C4B 3D71 8380 596D A6E5 9C91 Have fun, Jaroslav -- Jaroslav Kysela <perex(a)perex.cz> Linux Sound Maintainer; ALSA Project; Red Hat, Inc.

2 8

1.2.14: SetuptoolsDeprecationWarning due to the use of `sys.exit(0)`
by GitHub issues - opened 15 Apr '25

15 Apr '25

alsa-project/alsa-python issue #18 was opened from dvzrv: Hi! 👋 I package this project for Arch Linux. During update to 1.2.14 I noticed that setuptools now raises the following warning: ``` /usr/lib/python3.13/site-packages/setuptools/build_meta.py:522: SetuptoolsDeprecationWarning: Running `setup.py` directly as CLI tool is deprecated. !! ******************************************************************************** Please avoid using `sys.exit(0)` or similar statements that don't fit in the paradigm of a configuration file. See https://blog.ganssle.io/articles/2021/10/setup-py-deprecated.html for details. ******************************************************************************** !! super().run_setup(setup_script=setup_script) ``` This appears to be triggered by the `sys.exit(0)` call in setup.py: https://github.com/alsa-project/alsa-python/blob/20b2b3999f24af462815b9c86e… (Setuptools now offers one of the various PEP517 build backends and calling setup.py directly as a script is deprecated). Looking at the file I noticed that there are still conditionals that accommodate Python2: https://github.com/alsa-project/alsa-python/blob/20b2b3999f24af462815b9c86e… Those should be removed, as the script itself is marked as Python3 (and well of course because Python2 is long discontinued 😅). Issue URL : https://github.com/alsa-project/alsa-python/issues/18 Repository URL: https://github.com/alsa-project/alsa-python

1 0

recent change in seqmid.c is broken?
by Takashi Iwai 15 Apr '25

15 Apr '25

Hi Jaroslav, I'm afraid that your recent fix for alsa-lib commit a4e47461eca1 doesn't work as expected: ``` @@ -664,15 +663,11 @@ static void update_group_ports(snd_seq_t *seq, snd_ump_endpoint_info_t *ep) break; } - if (!*bp->name) + if (bp->name[0] == '\0') continue; - len = strlen(blknames); - if (len) - snprintf(blknames + len, sizeof(blknames) - len, - ", %s", bp->name); - else - snd_strlcpy(blknames, (const char *)bp->name, - sizeof(blknames)); + if (blknames[0]) + snd_strlcpy(blknames, ", ", sizeof(blknames)); + snd_strlcpy(blknames, (const char *)bp->name, sizeof(blknames)); } if (!*blknames) ``` The original code appended the new bp->name string with the prefix of ", " if blknames is already present, but the new code looks as if it overwrites onto blknames with strlcpy() from scratch for each bp->name. FWIW, the code there used to be with strlcat(), but it was rewritten in the way above because strlcat() isn't always available in commit d9694398130c. thanks, Takashi

2 3

On output devices, `snd_pcm_open()` makes a bad loud tick.
by GitHub issues - opened 13 Apr '25

13 Apr '25

alsa-project/alsa-lib issue #451 was opened from ewtoombs: Minimal reproduction: ``` #include <alsa/asoundlib.h> #include <stdbool.h> int main(int argc, const char **argv) { snd_pcm_t *out; assert(snd_pcm_open(&out, "default", // the name of the device being opened SND_PCM_STREAM_PLAYBACK, // it is an output stream false // iff nonblocking stream ) == 0); } ``` (Compile with `gcc -o test test.c -lasound`.) Relevant `lspci`: ``` 00:1f.3 Audio device: Intel Corporation Sunrise Point-LP HD Audio (rev 21) ``` `uname`: ``` Linux robert 6.13.8-arch1-1 #1 SMP PREEMPT_DYNAMIC Sun, 23 Mar 2025 17:17:30 +0000 x86_64 GNU/Linux ``` alsa-lib version: 1.2.13-1 (archlinux) I've heard similar artefacts on many different sound cards, though, across many different architectures, always on linux. One must wait about five minutes in between trials, or the artefact does not occur. This isn't a minor detail. The tick is very loud. It might even damage speakers in a worst case scenario. Issue URL : https://github.com/alsa-project/alsa-lib/issues/451 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[GIT PULL] ASoC fixes for v6.15-rc1
by Mark Brown 11 Apr '25

11 Apr '25

The following changes since commit 0af2f6be1b4281385b618cb86ad946eded089ac8: Linux 6.15-rc1 (2025-04-06 13:11:33 -0700) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.15-rc1 for you to fetch changes up to e6c4618422a25cb266bf752040eb509865caeb0e: MAINTAINERS: update my email address (2025-04-11 06:17:24 +0100) ---------------------------------------------------------------- ASoC: Fixes for v6.15 A set of small fixes, quirks and device ID additions that came in since -rc1, none of them super stand out. There's also a change to Srini's email address in MAINTAINERS. ---------------------------------------------------------------- Amadeusz Sławiński (1): ASoC: Intel: avs: Constrain path based on BE capabilities Brady Norander (1): ASoC: dwc: always enable/disable i2s irqs Charles Keepax (1): ASoC: cs42l43: Reset clamp override on jack removal Evgeny Pimenov (1): ASoC: qcom: Fix sc7280 lpass potential buffer overflow Henry Martin (1): ASoC: Intel: avs: Fix null-ptr-deref in avs_component_probe() Herve Codina (1): ASoC: fsl: fsl_qmc_audio: Reset audio data pointers on TRIGGER_START event Kuninori Morimoto (1): ASoC: hdmi-codec: use RTD ID instead of DAI ID for ELD entry Mark Brown (2): ASoC: codecs: lpass-wsa: fix VI capture setup. MAINTAINERS: update my email address Peter Ujfalusi (1): ASoC: Intel: sof_sdw: Add quirk for Asus Zenbook S16 Richard Fitzgerald (1): firmware: cs_dsp: test_bin_error: Fix uninitialized data used as fw version Shengjiu Wang (1): ASoC: fsl_asrc_dma: get codec or cpu dai from backend Srinivas Kandagatla (4): ASoC: codecs:lpass-wsa-macro: Fix vi feedback rate ASoC: codecs:lpass-wsa-macro: Fix logic of enabling vi channels MAINTAINERS: use kernel.org alias mailmap: Add entry for Srinivas Kandagatla Weidong Wang (1): ASoC: codecs: Add of_match_table for aw888081 driver .mailmap | 2 + MAINTAINERS | 8 +- .../firmware/cirrus/test/cs_dsp_mock_mem_maps.c | 30 ------ drivers/firmware/cirrus/test/cs_dsp_test_bin.c | 2 +- .../firmware/cirrus/test/cs_dsp_test_bin_error.c | 2 +- include/linux/firmware/cirrus/cs_dsp_test_utils.h | 1 - sound/soc/codecs/aw88081.c | 10 ++ sound/soc/codecs/cs42l43-jack.c | 3 + sound/soc/codecs/hdmi-codec.c | 22 +++- sound/soc/codecs/lpass-wsa-macro.c | 117 +++++++++++++++------ sound/soc/dwc/dwc-i2s.c | 13 ++- sound/soc/fsl/fsl_asrc_dma.c | 15 ++- sound/soc/fsl/fsl_qmc_audio.c | 3 + sound/soc/intel/avs/path.c | 72 +++++++++++++ sound/soc/intel/avs/path.h | 5 + sound/soc/intel/avs/pcm.c | 52 ++++++++- sound/soc/intel/boards/sof_sdw.c | 1 + sound/soc/qcom/lpass.h | 3 +- 18 files changed, 277 insertions(+), 84 deletions(-)

2 1

[PATCH 0/2] MAINTAINERS: update my email address
by srinivas.kandagatla＠linaro.org 11 Apr '25

11 Apr '25

From: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Update .mailmap and MAINTAINERS to point to the @kernel.org instead of the @linaro.org. Linaro address will stop working in few days. Mark or Greg, could you please pick these two patches? Srinivas Kandagatla (2): MAINTAINERS: use kernel.org alias mailmap: Add entry for Srinivas Kandagatla .mailmap | 2 ++ MAINTAINERS | 8 ++++---- 2 files changed, 6 insertions(+), 4 deletions(-) -- 2.25.1

2 3

Add support of HyperX SoloCast
by GitHub pull_request - opened 11 Apr '25

11 Apr '25

alsa-project/alsa-ucm-conf pull request #545 was opened from serhii-nakon: Support of - volume control - un/mute control PS: About un/mute control - only from PC to Mic, vise versa does not work - like when you un/mute Mic on PC it works as expected (even LED show actual status) but when you click button on Mic side it does not reflect on PC side. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/545 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/545.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ucm2: Qualcomm: add Asus Zenbook A14
by GitHub pull_request - opened 10 Apr '25

10 Apr '25

alsa-project/alsa-ucm-conf pull request #544 was opened from alexVinarskis: Add a Regex string, seems compatible with Lenovo T14s x2 speakers, x2 DMICs, x3 DP (x2 USB-C, x1 DP->HDMI), headphone jack. Initial support for the platform is available on the [lists](https://lore.kernel.org/all/20250402084646.10098-1-alex.vinarskis@gm…, with [Audio part ](https://github.com/alexVinarskis/linux-x1e80100-zenbook-a14/blob/master/0013-arm64-dts-qcom-support-sound-on-Asus-Zenbook-A14.patch)not being upstreamed just yet. Tested that speakers, MICs and jack are working. Respective audioreach-topology changes are open [for PR.](https://github.com/linux-msm/audioreach-topology/pull/14) Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/544 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/544.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH 1/1] bat: Fix buffer time configuration
by Alexander Stein 10 Apr '25

10 Apr '25

By swapping time configuration some hardware constraints can be supported. Fixes the error: Set parameter to device error: buffer time: 500000 hw:tqmtlv320aic32: Invalid argument(-22) Signed-off-by: Alexander Stein <alexander.stein(a)ew.tq-group.com> --- Our common 'alsabat -Dhw:tqmtlv320aic32 -f S16_LE -c 2 -r 44100 -F 997:500' failed ona i.MX95 based platform, while it worked flawlessly on others, e.g. i.MX8M Plus. Upon searching I stumbled across [1], which this patch is inspired/based on. [1] https://github.com/arkq/bluez-alsa/pull/290/commits/da78b82ff77310093b4b4a9… --- bat/alsa.c | 18 +++++++++--------- 1 file changed, 9 insertions(+), 9 deletions(-) diff --git a/bat/alsa.c b/bat/alsa.c index 67489c4..457e79a 100644 --- a/bat/alsa.c +++ b/bat/alsa.c @@ -221,23 +221,23 @@ static int set_snd_pcm_params(struct bat *bat, struct pcm_container *sndpcm) period_time = buffer_time / DIV_BUFFERTIME; - /* Set buffer time and period time */ - err = snd_pcm_hw_params_set_buffer_time_near(sndpcm->handle, - params, &buffer_time, 0); + /* Set period time and buffer time */ + err = snd_pcm_hw_params_set_period_time_near(sndpcm->handle, + params, &period_time, 0); if (err < 0) { fprintf(bat->err, _("Set parameter to device error: ")); - fprintf(bat->err, _("buffer time: %d %s: %s(%d)\n"), - buffer_time, + fprintf(bat->err, _("period time: %d %s: %s(%d)\n"), + period_time, device_name, snd_strerror(err), err); return err; } - err = snd_pcm_hw_params_set_period_time_near(sndpcm->handle, - params, &period_time, 0); + err = snd_pcm_hw_params_set_buffer_time_near(sndpcm->handle, + params, &buffer_time, 0); if (err < 0) { fprintf(bat->err, _("Set parameter to device error: ")); - fprintf(bat->err, _("period time: %d %s: %s(%d)\n"), - period_time, + fprintf(bat->err, _("buffer time: %d %s: %s(%d)\n"), + buffer_time, device_name, snd_strerror(err), err); return err; } -- 2.43.0

1 0

[PATCH v2] ALSA: hda/tas2781: Create a common header for both spi and i2c tas2781 hda driver
by Shenghao Ding 10 Apr '25

10 Apr '25

Move the common macro definition of kcontrols into a common header. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v2: - Follow IWYU principle, add sound/asound.h into the header file. v1: - Revise the year of spi and i2c tas2781 hda drivers. - Create a common header for both spi and i2c tas2781 hda driver to define the common macros and declare the common functions. --- sound/pci/hda/tas2781_hda.h | 44 +++++++++++++++++++++++++++++++++ sound/pci/hda/tas2781_hda_i2c.c | 29 ++-------------------- sound/pci/hda/tas2781_hda_spi.c | 35 ++------------------------ 3 files changed, 48 insertions(+), 60 deletions(-) create mode 100644 sound/pci/hda/tas2781_hda.h diff --git a/sound/pci/hda/tas2781_hda.h b/sound/pci/hda/tas2781_hda.h new file mode 100644 index 000000000000..fc741fac419a --- /dev/null +++ b/sound/pci/hda/tas2781_hda.h @@ -0,0 +1,44 @@ +/* SPDX-License-Identifier: GPL-2.0-only + * + * HDA audio driver for Texas Instruments TAS2781 smart amp + * + * Copyright (C) 2025 Texas Instruments, Inc. + */ +#ifndef __TAS2781_HDA_H__ +#define __TAS2781_HDA_H__ + +#include <sound/asound.h> + +/* + * No standard control callbacks for SNDRV_CTL_ELEM_IFACE_CARD + * Define two controls, one is Volume control callbacks, the other is + * flag setting control callbacks. + */ + +/* Volume control callbacks for tas2781 */ +#define ACARD_SINGLE_RANGE_EXT_TLV(xname, xreg, xshift, xmin, xmax, xinvert, \ + xhandler_get, xhandler_put, tlv_array) { \ + .iface = SNDRV_CTL_ELEM_IFACE_CARD, .name = (xname), \ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ + SNDRV_CTL_ELEM_ACCESS_READWRITE, \ + .tlv.p = (tlv_array), \ + .info = snd_soc_info_volsw, \ + .get = xhandler_get, .put = xhandler_put, \ + .private_value = (unsigned long)&(struct soc_mixer_control) { \ + .reg = xreg, .rreg = xreg, \ + .shift = xshift, .rshift = xshift,\ + .min = xmin, .max = xmax, .invert = xinvert, \ + } \ +} + +/* Flag control callbacks for tas2781 */ +#define ACARD_SINGLE_BOOL_EXT(xname, xdata, xhandler_get, xhandler_put) { \ + .iface = SNDRV_CTL_ELEM_IFACE_CARD, \ + .name = xname, \ + .info = snd_ctl_boolean_mono_info, \ + .get = xhandler_get, \ + .put = xhandler_put, \ + .private_value = xdata, \ +} + +#endif diff --git a/sound/pci/hda/tas2781_hda_i2c.c b/sound/pci/hda/tas2781_hda_i2c.c index 29dc4f500580..9d94ae5fcfe0 100644 --- a/sound/pci/hda/tas2781_hda_i2c.c +++ b/sound/pci/hda/tas2781_hda_i2c.c @@ -2,7 +2,7 @@ // // TAS2781 HDA I2C driver // -// Copyright 2023 - 2024 Texas Instruments, Inc. +// Copyright 2023 - 2025 Texas Instruments, Inc. // // Author: Shenghao Ding <shenghao-ding(a)ti.com> // Current maintainer: Baojun Xu <baojun.xu(a)ti.com> @@ -30,35 +30,10 @@ #include "hda_component.h" #include "hda_jack.h" #include "hda_generic.h" +#include "tas2781_hda.h" #define TASDEVICE_SPEAKER_CALIBRATION_SIZE 20 -/* No standard control callbacks for SNDRV_CTL_ELEM_IFACE_CARD - * Define two controls, one is Volume control callbacks, the other is - * flag setting control callbacks. - */ - -/* Volume control callbacks for tas2781 */ -#define ACARD_SINGLE_RANGE_EXT_TLV(xname, xreg, xshift, xmin, xmax, xinvert, \ - xhandler_get, xhandler_put, tlv_array) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_CARD, .name = (xname),\ - .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\ - SNDRV_CTL_ELEM_ACCESS_READWRITE,\ - .tlv.p = (tlv_array), \ - .info = snd_soc_info_volsw, \ - .get = xhandler_get, .put = xhandler_put, \ - .private_value = (unsigned long)&(struct soc_mixer_control) \ - {.reg = xreg, .rreg = xreg, .shift = xshift, \ - .rshift = xshift, .min = xmin, .max = xmax, \ - .invert = xinvert} } - -/* Flag control callbacks for tas2781 */ -#define ACARD_SINGLE_BOOL_EXT(xname, xdata, xhandler_get, xhandler_put) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_CARD, .name = xname, \ - .info = snd_ctl_boolean_mono_info, \ - .get = xhandler_get, .put = xhandler_put, \ - .private_value = xdata } - enum calib_data { R0_VAL = 0, INV_R0, diff --git a/sound/pci/hda/tas2781_hda_spi.c b/sound/pci/hda/tas2781_hda_spi.c index 399f2e4c3b62..c6be2be1b53e 100644 --- a/sound/pci/hda/tas2781_hda_spi.c +++ b/sound/pci/hda/tas2781_hda_spi.c @@ -2,7 +2,7 @@ // // TAS2781 HDA SPI driver // -// Copyright 2024 Texas Instruments, Inc. +// Copyright 2024 - 2025 Texas Instruments, Inc. // // Author: Baojun Xu <baojun.xu(a)ti.com> @@ -38,38 +38,7 @@ #include "hda_component.h" #include "hda_jack.h" #include "hda_generic.h" - -/* - * No standard control callbacks for SNDRV_CTL_ELEM_IFACE_CARD - * Define two controls, one is Volume control callbacks, the other is - * flag setting control callbacks. - */ - -/* Volume control callbacks for tas2781 */ -#define ACARD_SINGLE_RANGE_EXT_TLV(xname, xreg, xshift, xmin, xmax, xinvert, \ - xhandler_get, xhandler_put, tlv_array) { \ - .iface = SNDRV_CTL_ELEM_IFACE_CARD, .name = (xname), \ - .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ - SNDRV_CTL_ELEM_ACCESS_READWRITE, \ - .tlv.p = (tlv_array), \ - .info = snd_soc_info_volsw, \ - .get = xhandler_get, .put = xhandler_put, \ - .private_value = (unsigned long)&(struct soc_mixer_control) { \ - .reg = xreg, .rreg = xreg, \ - .shift = xshift, .rshift = xshift,\ - .min = xmin, .max = xmax, .invert = xinvert, \ - } \ -} - -/* Flag control callbacks for tas2781 */ -#define ACARD_SINGLE_BOOL_EXT(xname, xdata, xhandler_get, xhandler_put) { \ - .iface = SNDRV_CTL_ELEM_IFACE_CARD, \ - .name = xname, \ - .info = snd_ctl_boolean_mono_info, \ - .get = xhandler_get, \ - .put = xhandler_put, \ - .private_value = xdata, \ -} +#include "tas2781_hda.h" struct tas2781_hda { struct tasdevice_priv *priv; -- 2.34.1

1 0

Enable Headset Mic for ASUS platform
by Kailang 09 Apr '25

09 Apr '25

Hi Takashi, Attach patch was fixed headset Mic issue for Asus platform. Many Thanks. Kailang.

2 1

Dell Precision 3571: DMIC not detected by SOF driver, silent mic under Linux, works in Windows
by Adam Larios 08 Apr '25

08 Apr '25

*System*: Dell Precision 3571 Realtek ALC3204 codec Intel Alder Lake (SOF platform) *Symptoms*: Internal mic is silent or muffled in all Linux applications Works perfectly in Windows SOF driver loads correctly with firmware sof-adl.ri and topology sof-hda-generic.tplg But dmesg shows: sof-audio-pci-intel-tgl 0000:00:1f.3: DMICs detected in NHLT tables: 0 No working audio input nodes exposed in PipeWire *Tested*: - Ubuntu 24.04 with kernel 6.8.0-57-generic - snd-intel-dspcfg.dsp_driver=3 set - PulseAudio and PipeWire backends - alsa-base.conf overrides for model options - hdajackretask pin remapping - SOF firmware and topology confirmed present *Conclusion*: Firmware/ACPI likely missing DMIC NHLT entry; requesting kernel quirk or guidance

1 0

[PATCH v1] ALSA: hda/tas2781: Create a common header for both spi and i2c tas2781 hda driver
by Shenghao Ding 08 Apr '25

08 Apr '25

Move the common macro definition of kcontrols into a common header. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v1: - revise the year of spi and i2c tas2781 hda drivers. - Create a common header for both spi and i2c tas2781 hda driver to define the common macros and declare the common functions. --- sound/pci/hda/tas2781_hda.h | 42 +++++++++++++++++++++++++++++++++ sound/pci/hda/tas2781_hda_i2c.c | 29 ++--------------------- sound/pci/hda/tas2781_hda_spi.c | 35 ++------------------------- 3 files changed, 46 insertions(+), 60 deletions(-) create mode 100644 sound/pci/hda/tas2781_hda.h diff --git a/sound/pci/hda/tas2781_hda.h b/sound/pci/hda/tas2781_hda.h new file mode 100644 index 000000000000..c57909c0ed65 --- /dev/null +++ b/sound/pci/hda/tas2781_hda.h @@ -0,0 +1,42 @@ +/* SPDX-License-Identifier: GPL-2.0-only + * + * HDA audio driver for Texas Instruments TAS2781 smart amp + * + * Copyright (C) 2025 Texas Instruments, Inc. + */ +#ifndef __TAS2781_HDA_H__ +#define __TAS2781_HDA_H__ + +/* + * No standard control callbacks for SNDRV_CTL_ELEM_IFACE_CARD + * Define two controls, one is Volume control callbacks, the other is + * flag setting control callbacks. + */ + +/* Volume control callbacks for tas2781 */ +#define ACARD_SINGLE_RANGE_EXT_TLV(xname, xreg, xshift, xmin, xmax, xinvert, \ + xhandler_get, xhandler_put, tlv_array) { \ + .iface = SNDRV_CTL_ELEM_IFACE_CARD, .name = (xname), \ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ + SNDRV_CTL_ELEM_ACCESS_READWRITE, \ + .tlv.p = (tlv_array), \ + .info = snd_soc_info_volsw, \ + .get = xhandler_get, .put = xhandler_put, \ + .private_value = (unsigned long)&(struct soc_mixer_control) { \ + .reg = xreg, .rreg = xreg, \ + .shift = xshift, .rshift = xshift,\ + .min = xmin, .max = xmax, .invert = xinvert, \ + } \ +} + +/* Flag control callbacks for tas2781 */ +#define ACARD_SINGLE_BOOL_EXT(xname, xdata, xhandler_get, xhandler_put) { \ + .iface = SNDRV_CTL_ELEM_IFACE_CARD, \ + .name = xname, \ + .info = snd_ctl_boolean_mono_info, \ + .get = xhandler_get, \ + .put = xhandler_put, \ + .private_value = xdata, \ +} + +#endif diff --git a/sound/pci/hda/tas2781_hda_i2c.c b/sound/pci/hda/tas2781_hda_i2c.c index 29dc4f500580..9d94ae5fcfe0 100644 --- a/sound/pci/hda/tas2781_hda_i2c.c +++ b/sound/pci/hda/tas2781_hda_i2c.c @@ -2,7 +2,7 @@ // // TAS2781 HDA I2C driver // -// Copyright 2023 - 2024 Texas Instruments, Inc. +// Copyright 2023 - 2025 Texas Instruments, Inc. // // Author: Shenghao Ding <shenghao-ding(a)ti.com> // Current maintainer: Baojun Xu <baojun.xu(a)ti.com> @@ -30,35 +30,10 @@ #include "hda_component.h" #include "hda_jack.h" #include "hda_generic.h" +#include "tas2781_hda.h" #define TASDEVICE_SPEAKER_CALIBRATION_SIZE 20 -/* No standard control callbacks for SNDRV_CTL_ELEM_IFACE_CARD - * Define two controls, one is Volume control callbacks, the other is - * flag setting control callbacks. - */ - -/* Volume control callbacks for tas2781 */ -#define ACARD_SINGLE_RANGE_EXT_TLV(xname, xreg, xshift, xmin, xmax, xinvert, \ - xhandler_get, xhandler_put, tlv_array) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_CARD, .name = (xname),\ - .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\ - SNDRV_CTL_ELEM_ACCESS_READWRITE,\ - .tlv.p = (tlv_array), \ - .info = snd_soc_info_volsw, \ - .get = xhandler_get, .put = xhandler_put, \ - .private_value = (unsigned long)&(struct soc_mixer_control) \ - {.reg = xreg, .rreg = xreg, .shift = xshift, \ - .rshift = xshift, .min = xmin, .max = xmax, \ - .invert = xinvert} } - -/* Flag control callbacks for tas2781 */ -#define ACARD_SINGLE_BOOL_EXT(xname, xdata, xhandler_get, xhandler_put) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_CARD, .name = xname, \ - .info = snd_ctl_boolean_mono_info, \ - .get = xhandler_get, .put = xhandler_put, \ - .private_value = xdata } - enum calib_data { R0_VAL = 0, INV_R0, diff --git a/sound/pci/hda/tas2781_hda_spi.c b/sound/pci/hda/tas2781_hda_spi.c index 399f2e4c3b62..c6be2be1b53e 100644 --- a/sound/pci/hda/tas2781_hda_spi.c +++ b/sound/pci/hda/tas2781_hda_spi.c @@ -2,7 +2,7 @@ // // TAS2781 HDA SPI driver // -// Copyright 2024 Texas Instruments, Inc. +// Copyright 2024 - 2025 Texas Instruments, Inc. // // Author: Baojun Xu <baojun.xu(a)ti.com> @@ -38,38 +38,7 @@ #include "hda_component.h" #include "hda_jack.h" #include "hda_generic.h" - -/* - * No standard control callbacks for SNDRV_CTL_ELEM_IFACE_CARD - * Define two controls, one is Volume control callbacks, the other is - * flag setting control callbacks. - */ - -/* Volume control callbacks for tas2781 */ -#define ACARD_SINGLE_RANGE_EXT_TLV(xname, xreg, xshift, xmin, xmax, xinvert, \ - xhandler_get, xhandler_put, tlv_array) { \ - .iface = SNDRV_CTL_ELEM_IFACE_CARD, .name = (xname), \ - .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ - SNDRV_CTL_ELEM_ACCESS_READWRITE, \ - .tlv.p = (tlv_array), \ - .info = snd_soc_info_volsw, \ - .get = xhandler_get, .put = xhandler_put, \ - .private_value = (unsigned long)&(struct soc_mixer_control) { \ - .reg = xreg, .rreg = xreg, \ - .shift = xshift, .rshift = xshift,\ - .min = xmin, .max = xmax, .invert = xinvert, \ - } \ -} - -/* Flag control callbacks for tas2781 */ -#define ACARD_SINGLE_BOOL_EXT(xname, xdata, xhandler_get, xhandler_put) { \ - .iface = SNDRV_CTL_ELEM_IFACE_CARD, \ - .name = xname, \ - .info = snd_ctl_boolean_mono_info, \ - .get = xhandler_get, \ - .put = xhandler_put, \ - .private_value = xdata, \ -} +#include "tas2781_hda.h" struct tas2781_hda { struct tasdevice_priv *priv; -- 2.34.1

2 1

ucm2: Qualcomm: add Lenovo ThinkBook 16 support
by GitHub pull_request - opened 08 Apr '25

08 Apr '25

alsa-project/alsa-ucm-conf pull request #543 was opened from jglathe: add a Regex string, seems compatible with Thinkpad T14s Tested-by: Jens Glathe <jens.glathe(a)oldschoolsolutions.biz> Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/543 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/543.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Fireworks Alsa Driver Buffer SIze
by jayt0808＠gmail.com 08 Apr '25

08 Apr '25

Hi I'm using an AudioFire 8a that has 16 channels over the firewire bus and the snd-fireworks alsa firewire driver. In Jack, when I set Frames/Period to anything higher than 64, alsa fails with an error saying Mon Apr 7 22:15:35 2025: ERROR: ALSA: cannot set hardware parameters for capture Mon Apr 7 22:15:35 2025: ERROR: ALSA: cannot configure capture channel I noticed that the output of "aplay -D hw:1,0 --dump-hw-params /dev/zero" indicated the following alsa settings: Playing raw data '/dev/zero' : Unsigned 8 bit, Rate 8000 Hz, Mono HW Params of device "hw:1,0": -------------------- ACCESS: MMAP_INTERLEAVED RW_INTERLEAVED FORMAT: S32_LE SUBFORMAT: STD SAMPLE_BITS: 32 FRAME_BITS: [384 512] CHANNELS: [12 16] RATE: [32000 96000] PERIOD_TIME: [250 32000] PERIOD_SIZE: [16 2720] PERIOD_BYTES: [768 131072] PERIODS: [2 341] BUFFER_TIME: (333 170500] BUFFER_SIZE: [32 5456] BUFFER_BYTES: [1536 262144] TICK_TIME: ALL -------------------- aplay: set_params:1371: Sample format non available Available formats: - S32_LE I was concerned that the BUFFER_SIZE was too small to support 16 channels at a 128 Frames/Period size. But when I modded the driver to increase this figure, it did not fix the issue. Anyone have any insight on this issue? Running Ubuntu 24.04 with kernel 6.8.0-57-lowlatency. Everything seems to work fine when I set 64 Frames/Period (except the occasional annoying stuttering that one can expect in that scenario; with a buffer that is too small to support the system load). Thanks Jay

1 0

[PATCH v2] ASoC: dwc: always enable/disable i2s irqs
by Brady Norander 08 Apr '25

08 Apr '25

Commit a42e988 ("ASoC: dwc: add DMA handshake control") changed the behavior of the driver to not enable or disable i2s irqs if using DMA. This breaks platforms such as AMD ACP. Audio playback appears to work but no audio can be heard. Revert to the old behavior by always enabling and disabling i2s irqs while keeping DMA handshake control. Fixes: a42e988 ("ASoC: dwc: add DMA handshake control") Signed-off-by: Brady Norander <bradynorander(a)gmail.com> --- v2: reword commit message to better describe the issue --- sound/soc/dwc/dwc-i2s.c | 13 ++++++------- 1 file changed, 6 insertions(+), 7 deletions(-) diff --git a/sound/soc/dwc/dwc-i2s.c b/sound/soc/dwc/dwc-i2s.c index 57b789d7fbed..5b4f20dbf7bb 100644 --- a/sound/soc/dwc/dwc-i2s.c +++ b/sound/soc/dwc/dwc-i2s.c @@ -199,12 +199,10 @@ static void i2s_start(struct dw_i2s_dev *dev, else i2s_write_reg(dev->i2s_base, IRER, 1); - /* I2S needs to enable IRQ to make a handshake with DMAC on the JH7110 SoC */ - if (dev->use_pio || dev->is_jh7110) - i2s_enable_irqs(dev, substream->stream, config->chan_nr); - else + if (!(dev->use_pio || dev->is_jh7110)) i2s_enable_dma(dev, substream->stream); + i2s_enable_irqs(dev, substream->stream, config->chan_nr); i2s_write_reg(dev->i2s_base, CER, 1); } @@ -218,11 +216,12 @@ static void i2s_stop(struct dw_i2s_dev *dev, else i2s_write_reg(dev->i2s_base, IRER, 0); - if (dev->use_pio || dev->is_jh7110) - i2s_disable_irqs(dev, substream->stream, 8); - else + if (!(dev->use_pio || dev->is_jh7110)) i2s_disable_dma(dev, substream->stream); + i2s_disable_irqs(dev, substream->stream, 8); + + if (!dev->active) { i2s_write_reg(dev->i2s_base, CER, 0); i2s_write_reg(dev->i2s_base, IER, 0); -- 2.49.0

1 0

pcm: hw: do not reset tstamp_type in SND_PCM_APPEND mode
by GitHub pull_request - opened 07 Apr '25

07 Apr '25

alsa-project/alsa-lib pull request #450 was opened from kgroeneveld: When the first client of plugins such as dshare open the hw device they set a default tstamp_type in snd_pcm_direct_initialize_slave based on tstamp_type from the config file. But when subsequent clients open the same plugin the snd_pcm_hw_open_fd function clobbers this default. Request URL : https://github.com/alsa-project/alsa-lib/pull/450 Patch URL : https://github.com/alsa-project/alsa-lib/pull/450.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

alsaseq: Fix type check in set_time()
by GitHub pull_request - opened 06 Apr '25

06 Apr '25

alsa-project/alsa-python pull request #17 was opened from ernestask: SeqEvent_set_time() has fallen victim to the inverted get_long1() return value here, erroring out when called with actual integers. Request URL : https://github.com/alsa-project/alsa-python/pull/17 Patch URL : https://github.com/alsa-project/alsa-python/pull/17.patch Repository URL: https://github.com/alsa-project/alsa-python

1 0

envy24control: port to GTK 3
by GitHub pull_request - opened 05 Apr '25

05 Apr '25

alsa-project/alsa-tools pull request #35 was opened from andreas56: This is a port envy24control from GTK 2 to GTK 3. Note: There is another very similar port in #27, but that one isn't quite finished. I did my port before noticing that there already was one, but I have compared them now to check that I didn't miss anything. My version has more changes, as I wanted to avoid using deprecated GTK functions. It also fixes the rendering of the level meters and optimizes it a bit so the CPU load isn't much higher than it was in GTK 2. Note 2: the fix from #31 is included in this pull request, as it is necessary for the spin buttons to work in GTK 3. Note 3: my other envy24control pull requests are not included here, as they are not related to GTK 3. #29 will probably cause merge conflicts with this one, but I will fix them if you decide to merge any of the pull requests. Request URL : https://github.com/alsa-project/alsa-tools/pull/35 Patch URL : https://github.com/alsa-project/alsa-tools/pull/35.patch Repository URL: https://github.com/alsa-project/alsa-tools

1 0

ALC4080 - general discussion (driver support)
by GitHub issues - edited 04 Apr '25

04 Apr '25

alsa-project/alsa-ucm-conf issue #541 was edited from perexg: UCM cannot add basic functionalities when the driver does not expose them. I already explained situation in [kernel bugzilla](https://bugzilla.kernel.org/show_bug.cgi?id=218913). **Note** This issue is for the kernel driver related problems. If the driver works and something is missing in UCM configuration, please, create a new issue in this tracker. It may be difficult to distinguish the right place, but if the developers redirected you here, follow the instructions bellow. **For issues like** 1. jack functionality assignment is unstable (jack functionality is not fixed - e.g. changes on warm reboot from windows) 2. low microphone volume (which cannot be controlled using native ALSA mixer controls) 3. sound is activated only after some magic plug/unplug sequence **Further steps** Try to analyze the audio driver using direct ALSA API. See [DEBUG.md](https://github.com/alsa-project/alsa-ucm-conf/blob/master/ucm2/DE…. Very nice report is in #528. Please, use it as a reference what to do. Developers usually do not have exact hardware, so observations describing which ALSA PCM device, which ALSA Mixer control, which ALSA Jack control can be used for given playback/capture device are essential. If the functionality cannot be handled with current ALSA audio driver, go to paragraphs bellow. Reverse engineering of the Windows driver. Capture USB communication and save it to a file which can be replayed by [wireshark](https://www.wireshark.org). Try to write an explanation including timing (what happened in given time slots). Driver reports should go to [kernel bugzilla](https://bugzilla.kernel.org). Add a comment with a link here. You may also push hardware vendors to create USB device which fully follows the USB audio specification, so the ALSA driver will work out-of-box (without "quirks"). **Documentation links** The documentation is not complete. Realtek support is generally very limited to provide full specification to their codecs / audio chips those years. They are providing details only to hardware vendors with a contract. If you know more, please, let us know. - [ALC4080 block scheme](https://bugzilla.kernel.org/attachment.cgi?id=306370) - [ALC4080 datasheet](https://bugzilla.kernel.org/attachment.cgi?id=307224) **Some details** ALC4080 is USB codec, thus it exposes functionality through the USB bus. The block scheme is nice to know internals, but look to the MCU paragraph in the datasheet: ``` 2.1.2. Micro Controller Unit - On-chip high-performance and low-power MCU - Ultra-low power consumption when MCU is at idle state - MCU controls connection to USB bus for re-enumeration without hot-plug - Internal programmable memory support for customized audio function - Watchdog control for MCU reset and interrupt - Configurable VID (Vendor ID), PID (Product ID), and serial number string ``` This means, that the internal firmware for this MCU unit can change the behaviour of audio hardware. Usually, the firmware is vendor specific. The USB audio hardware should comply with the USB Audio specification (refer https://www.usb.org/). The control units parsed by the ALSA driver can be obtained using 'cat /proc/asound/cardX/usbmixer' command (replace X with the correct ALSA card number). Those units are mapped to the standard mixer controls. My guess is that the firmware is not complaint with the USB audio specification and some functionality is controlled using specific USB handshake (data blocks). Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/541 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ALC4080 - general discussion (driver support)
by GitHub issues - edited 04 Apr '25

04 Apr '25

alsa-project/alsa-ucm-conf issue #541 was edited from perexg: UCM cannot add basic functionalities when the driver does not expose them. I already explained situation in [kernel bugzilla](https://bugzilla.kernel.org/show_bug.cgi?id=218913). **Note** This issue is for the kernel driver related problems. If the driver works and something is missing in UCM configuration, please, create a new issue in this tracker. It may be difficult to distinguish the right place, but if the developers redirected you here, follow the instructions bellow. **For issues like** 1. jack functionality assignment is unstable (jack functionality is not fixed - e.g. changes on warm reboot from windows) 2. low microphone volume (which cannot be controlled using native ALSA mixer controls) 3. sound is activated only after some magic plug/unplug sequence **Further steps** Try to analyze the audio driver using direct ALSA API. See [DEBUG.md](https://github.com/alsa-project/alsa-ucm-conf/blob/master/ucm2/DE…. Very nice report is in #528. Please, use it as a reference what to do. Developers usually do not have exact hardware, so observations describing which ALSA PCM device, which ALSA Mixer control, which ALSA Jack control can be used for given playback/capture device are essential. Reverse engineering of the Windows driver. Capture USB communication and save it to a file which can be replayed by [wireshark](https://www.wireshark.org). Try to write an explanation including timing (what happened in given time slots). Driver reports should go to [kernel bugzilla](https://bugzilla.kernel.org). Add a comment with a link here. You may also push hardware vendors to create USB device which fully follows the USB audio specification, so the ALSA driver will work out-of-box (without "quirks"). **Documentation links** The documentation is not complete. Realtek support is generally very limited to provide full specification to their codecs / audio chips those years. They are providing details only to hardware vendors with a contract. If you know more, please, let us know. - [ALC4080 block scheme](https://bugzilla.kernel.org/attachment.cgi?id=306370) - [ALC4080 datasheet](https://bugzilla.kernel.org/attachment.cgi?id=307224) **Some details** ALC4080 is USB codec, thus it exposes functionality through the USB bus. The block scheme is nice to know internals, but look to the MCU paragraph in the datasheet: ``` 2.1.2. Micro Controller Unit - On-chip high-performance and low-power MCU - Ultra-low power consumption when MCU is at idle state - MCU controls connection to USB bus for re-enumeration without hot-plug - Internal programmable memory support for customized audio function - Watchdog control for MCU reset and interrupt - Configurable VID (Vendor ID), PID (Product ID), and serial number string ``` This means, that the internal firmware for this MCU unit can change the behaviour of audio hardware. Usually, the firmware is vendor specific. The USB audio hardware should comply with the USB Audio specification (refer https://www.usb.org/). The control units parsed by the ALSA driver can be obtained using 'cat /proc/asound/cardX/usbmixer' command (replace X with the correct ALSA card number). Those units are mapped to the standard mixer controls. My guess is that the firmware is not complaint with the USB audio specification and some functionality is controlled using specific USB handshake (data blocks). Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/541 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ALC4080 - general discussion (driver support)
by GitHub issues - edited 04 Apr '25

04 Apr '25

alsa-project/alsa-ucm-conf issue #541 was edited from perexg: UCM cannot add basic functionalities when the driver does not expose them. I already explained situation in [kernel bugzilla](https://bugzilla.kernel.org/show_bug.cgi?id=218913). **Note** This issue is for the kernel driver related problems. If the driver works and something is missing in UCM configuration, please, create a new issue in this tracker. It may be difficult to distinguish the right place, but if the developers redirected you here, follow the instructions bellow. **For issues like** 1. jack functionality assignment is unstable (jack functionality is not fixed - e.g. changes on warm reboot from windows) 2. low microphone volume (which cannot be controlled using native ALSA mixer controls) 3. sound is activated only after some magic plug/unplug sequence **Further steps** Try to analyze the audio driver using direct ALSA API. See [DEBUG.md](https://github.com/alsa-project/alsa-ucm-conf/blob/master/ucm2/DE…. Very nice report is in #528. Please, use it as a reference what to do. Developers usually do not have exact hardware, so observations describing which ALSA PCM device, which ALSA Mixer control, which ALSA Jack control can be used for given playback/capture device is essential. Reverse engineering of the Windows driver. Capture USB communication and save it to a file which can be replayed by [wireshark](https://www.wireshark.org). Try to write an explanation including timing (what happened in given time slots). Driver reports should go to [kernel bugzilla](https://bugzilla.kernel.org). Add a comment with a link here. You may also push hardware vendors to create USB device which fully follows the USB audio specification, so the ALSA driver will work out-of-box (without "quirks"). **Documentation links** The documentation is not complete. Realtek support is generally very limited to provide full specification to their codecs / audio chips those years. They are providing details only to hardware vendors with a contract. If you know more, please, let us know. - [ALC4080 block scheme](https://bugzilla.kernel.org/attachment.cgi?id=306370) - [ALC4080 datasheet](https://bugzilla.kernel.org/attachment.cgi?id=307224) **Some details** ALC4080 is USB codec, thus it exposes functionality through the USB bus. The block scheme is nice to know internals, but look to the MCU paragraph in the datasheet: ``` 2.1.2. Micro Controller Unit - On-chip high-performance and low-power MCU - Ultra-low power consumption when MCU is at idle state - MCU controls connection to USB bus for re-enumeration without hot-plug - Internal programmable memory support for customized audio function - Watchdog control for MCU reset and interrupt - Configurable VID (Vendor ID), PID (Product ID), and serial number string ``` This means, that the internal firmware for this MCU unit can change the behaviour of audio hardware. Usually, the firmware is vendor specific. The USB audio hardware should comply with the USB Audio specification (refer https://www.usb.org/). The control units parsed by the ALSA driver can be obtained using 'cat /proc/asound/cardX/usbmixer' command (replace X with the correct ALSA card number). Those units are mapped to the standard mixer controls. My guess is that the firmware is not complaint with the USB audio specification and some functionality is controlled using specific USB handshake (data blocks). Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/541 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ucm2: sof-soundwire: Correct include file path for dsp.conf
by GitHub pull_request - opened 04 Apr '25

04 Apr '25

alsa-project/alsa-ucm-conf pull request #542 was opened from ujfalusi: The file path must be given as absolut and not relative, otherwise the dsp.conf file will not be found: $ alsaucm dump text ALSA lib utils.c:364:(uc_mgr_config_load_into) could not open configuration file /usr/share/alsa/ucm2/conf.d/sof-soundwire/../Intel/sof-hda-dsp/dsp.conf ALSA lib parser.c:78:(uc_mgr_config_load_file) error: failed to open file /usr/share/alsa/ucm2/conf.d/sof-soundwire/../Intel/sof-hda-dsp/dsp.conf: -2 ALSA lib main.c:1554:(snd_use_case_mgr_open) error: failed to import hw:0 use case configuration -2 alsaucm: error failed to open sound card hw:0: No such file or directory Fixes: c9d1f7eafa6c ("UCM2: sof-soundwire: Enable DRC and equalizers for") Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/542 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/542.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

USB-Audio: Add Roland BridgeCast One | adapted
by GitHub pull_request - edited 03 Apr '25

03 Apr '25

alsa-project/alsa-ucm-conf pull request #536 was edited from Simon99de: Add Roland BridgeCast One uses the same config as the bridge cast v2 as it has the same outputs and it worked in my test. 0582:030d Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/536 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/536.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Alsa lib is not able to find the UCM .conf files
by GitHub issues - opened 03 Apr '25

03 Apr '25

alsa-project/alsa-lib issue #449 was opened from 3bitbrain: Note : the sound card is a Xonar DX ( AV200 ) The alsa UCM files are installed at PATH=/usr/share/alsa When I try to start alsasound, I get this error == ` alsa-lib-1.2.11/src/umc/main.c:1554: (snd_use_case_mgr_open) error : failed to import hw:1 use case configuration -2 ` The CMD alsaucm listcards gets this response : list is empty. [alsa-info.txt](https://github.com/alsa-project/alsa-ucm-conf/files/15455180… [aplay-info.txt](https://github.com/alsa-project/alsa-ucm-conf/files/1545518… Issue URL : https://github.com/alsa-project/alsa-lib/issues/449 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

ALC4080 - general discussion (driver support)
by GitHub issues - edited 03 Apr '25

03 Apr '25

alsa-project/alsa-ucm-conf issue #541 was edited from perexg: UCM cannot add basic functionalities when the driver does not expose them. I already explained situation in [kernel bugzilla](https://bugzilla.kernel.org/show_bug.cgi?id=218913). **Note** This issue is for the kernel driver related problems. If the driver works and something is missing in UCM configuration, please, create a new issue in this tracker. It may be difficult to distinguish the right place, but if the developers redirected you here, follow the instructions bellow. **For issues like** 1. jack functionality assignment is unstable (jack functionality is not fixed - e.g. changes on warm reboot from windows) 2. low microphone volume (which cannot be controlled using native ALSA mixer controls) 3. sound is activated only after some magic plug/unplug sequence **Further steps** Try to analyze the audio driver using direct ALSA API. See [DEBUG.md](https://github.com/alsa-project/alsa-ucm-conf/blob/master/ucm2/DE…. Very nice report is in #528. Please, use it as a reference what to do. Reverse engineering of the Windows driver. Capture USB communication and save it to a file which can be replayed by [wireshark](https://www.wireshark.org). Try to write an explanation including timing (what happened in given time slots). Driver reports should go to [kernel bugzilla](https://bugzilla.kernel.org). Add a comment with a link here. You may also push hardware vendors to create USB device which fully follows the USB audio specification, so the ALSA driver will work out-of-box (without "quirks"). **Documentation links** The documentation is not complete. Realtek support is generally very limited to provide full specification to their codecs / audio chips those years. They are providing details only to hardware vendors with a contract. If you know more, please, let us know. - [ALC4080 block scheme](https://bugzilla.kernel.org/attachment.cgi?id=306370) - [ALC4080 datasheet](https://bugzilla.kernel.org/attachment.cgi?id=307224) **Some details** ALC4080 is USB codec, thus it exposes functionality through the USB bus. The block scheme is nice to know internals, but look to the MCU paragraph in the datasheet: ``` 2.1.2. Micro Controller Unit - On-chip high-performance and low-power MCU - Ultra-low power consumption when MCU is at idle state - MCU controls connection to USB bus for re-enumeration without hot-plug - Internal programmable memory support for customized audio function - Watchdog control for MCU reset and interrupt - Configurable VID (Vendor ID), PID (Product ID), and serial number string ``` This means, that the internal firmware for this MCU unit can change the behaviour of audio hardware. Usually, the firmware is vendor specific. The USB audio hardware should comply with the USB Audio specification (refer https://www.usb.org/). The control units parsed by the ALSA driver can be obtained using 'cat /proc/asound/cardX/usbmixer' command (replace X with the correct ALSA card number). Those units are mapped to the standard mixer controls. My guess is that the firmware is not complaint with the USB audio specification and some functionality is controlled using specific USB handshake (data blocks). Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/541 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ALC4080 - general discussion (driver support)
by GitHub issues - edited 03 Apr '25

03 Apr '25

alsa-project/alsa-ucm-conf issue #541 was edited from perexg: UCM cannot add basic functionalities when the driver does not expose them. I already explained situation in [kernel bugzilla](https://bugzilla.kernel.org/show_bug.cgi?id=218913). **Note** This issue is for the kernel driver related problems. If the driver works and something is missing in UCM configuration, please, create a new issue in this tracker. It may be difficult to distinguish the right place, but if the developers redirected you here, follow the instructions bellow. **For issues like** 1. jack functionality assignment is unstable (jack functionality is not fixed - e.g. changes on warm reboot from windows) 2. low microphone volume (which cannot be controlled using native ALSA mixer controls) 3. sound is activated only after some magic plug/unplug sequence **Further steps** Try to analyze the audio driver using direct ALSA API. See [DEBUG.md](https://github.com/alsa-project/alsa-ucm-conf/blob/master/ucm2/DE…. Very nice report is in #528. Please, use it as a reference what to do. Reverse engineering of the Windows driver. Capture USB communication and save it to a file which can be replayed by [wireshark] (https://www.wireshark.org). Try to write an explanation including timing (what happened in given time slots). Driver reports should go to [kernel bugzilla](https://bugzilla.kernel.org). Add a comment with a link here. You may also push hardware vendors to create USB device which fully follows the USB audio specification, so the ALSA driver will work out-of-box (without "quirks"). **Documentation links** The documentation is not complete. Realtek support is generally very limited to provide full specification to their codecs / audio chips those years. They are providing details only to hardware vendors with a contract. If you know more, please, let us know. - [ALC4080 block scheme](https://bugzilla.kernel.org/attachment.cgi?id=306370) - [ALC4080 datasheet](https://bugzilla.kernel.org/attachment.cgi?id=307224) **Some details** ALC4080 is USB codec, thus it exposes functionality through the USB bus. The block scheme is nice to know internals, but look to the MCU paragraph in the datasheet: ``` 2.1.2. Micro Controller Unit - On-chip high-performance and low-power MCU - Ultra-low power consumption when MCU is at idle state - MCU controls connection to USB bus for re-enumeration without hot-plug - Internal programmable memory support for customized audio function - Watchdog control for MCU reset and interrupt - Configurable VID (Vendor ID), PID (Product ID), and serial number string ``` This means, that the internal firmware for this MCU unit can change the behaviour of audio hardware. Usually, the firmware is vendor specific. The USB audio hardware should comply with the USB Audio specification (refer https://www.usb.org/). The control units parsed by the ALSA driver can be obtained using 'cat /proc/asound/cardX/usbmixer' command (replace X with the correct ALSA card number). Those units are mapped to the standard mixer controls. My guess is that the firmware is not complaint with the USB audio specification and some functionality is controlled using specific USB handshake (data blocks). Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/541 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ALC4080 - general discussion (driver support)
by GitHub issues - opened 03 Apr '25

03 Apr '25

alsa-project/alsa-ucm-conf issue #541 was opened from perexg: UCM cannot add basic functionalities when the driver does not expose them. I already explained situation in [kernel bugzilla](https://bugzilla.kernel.org/show_bug.cgi?id=218913). **For issues like** 1. jack functionality assignment is unstable (jack functionality is not fixed - e.g. changes on warm reboot from windows) 2. low microphone volume (which cannot be controlled using native ALSA mixer controls) 3. sound is activated only after some magic plug/unplug sequence **Further steps** Try to analyze the audio driver using direct ALSA API. See [DEBUG.md](https://github.com/alsa-project/alsa-ucm-conf/blob/master/ucm2/DE…. Very nice report is in #528. Please, use it as a reference what to do. Reverse engineering of the Windows driver. Capture USB communication and save it to a file which can be replayed by [wireshark] (https://www.wireshark.org). Try to write an explanation including timing (what happened in given time slots). Driver reports should go to [kernel bugzilla](https://bugzilla.kernel.org). Add a comment with a link here. You may also push hardware vendors to create USB device which fully follows the USB audio specification, so the ALSA driver will work out-of-box (without "quirks"). **Documentation links** The documentation is not complete. Realtek support is generally very limited to provide full specification to their codecs / audio chips those years. They are providing details only to hardware vendors with a contract. If you know more, please, let us know. - [ALC4080 block scheme](https://bugzilla.kernel.org/attachment.cgi?id=306370) - [ALC4080 datasheet](https://bugzilla.kernel.org/attachment.cgi?id=307224) **Some details** ALC4080 is USB codec, thus it exposes functionality through the USB bus. The block scheme is nice to know internals, but look to the MCU paragraph in the datasheet: ``` 2.1.2. Micro Controller Unit - On-chip high-performance and low-power MCU - Ultra-low power consumption when MCU is at idle state - MCU controls connection to USB bus for re-enumeration without hot-plug - Internal programmable memory support for customized audio function - Watchdog control for MCU reset and interrupt - Configurable VID (Vendor ID), PID (Product ID), and serial number string ``` This means, that the internal firmware for this MCU unit can change the behaviour of audio hardware. Usually, the firmware is vendor specific. The USB audio hardware should comply with the USB Audio specification (refer https://www.usb.org/). The control units parsed by the ALSA driver can be obtained using 'cat /proc/asound/cardX/usbmixer' command (replace X with the correct ALSA card number). Those units are mapped to the standard mixer controls. My guess is that the firmware is not complaint with the USB audio specification and some functionality is controlled using specific USB handshake (data blocks). Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/541 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Delete alsalisp code
by GitHub pull_request - opened 02 Apr '25

02 Apr '25

alsa-project/alsa-lib pull request #448 was opened from fragglet: Install of the alsalisp binary has been disabled since 2006 (in commit 8d382ccd), and building of it was disabled by default in 2018 (in commit 32ceab21), so it is reasonable to assume that nobody is using it. Use within the alsa-lib project is limited to an aliases file that looks like it is intended as an example, plus some very small .alisp files associated with the SiS SI7018 PCI sound card which has not been manufactured in years. These too have not been installed since 2018 when commit 32ceab21 disabled building of the alsalisp binary. In preparing this change, I searched the Github issue tracker for "lisp", "alisp" and "alsalisp", and found no complaints about the above changes. I also did a Github code search for projects that might be including the `alisp.h` header and found none. Therefore I think this code can be safely deleted and nobody is likely to object. Request URL : https://github.com/alsa-project/alsa-lib/pull/448 Patch URL : https://github.com/alsa-project/alsa-lib/pull/448.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

sof-soundwire: cs42l43: Switch mixer based on output volume
by GitHub pull_request - opened 02 Apr '25

02 Apr '25

alsa-project/alsa-ucm-conf pull request #540 was opened from mstrozek: Hardware volume control in cs42l43 can go up to +31.5dB which can cause audio degradation. For best results, limit hardware volume to 0dB. Link: https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/4622 Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/540 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/540.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[GIT PULL] ASoC fixes for v6.15-merge-window
by Mark Brown 02 Apr '25

02 Apr '25

The following changes since commit 9ef52d529bb75071e03cf85078f724d69c4abe89: ASoC: SDCA: Correct handling of selected mode DisCo property (2025-03-21 15:45:25 +0000) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.15-merge-window for you to fetch changes up to 93d34608fd162f725172e780b1c60cc93a920719: ASoC: imx-card: Add NULL check in imx_card_probe() (2025-04-01 17:01:10 +0100) ---------------------------------------------------------------- ASoC: Fixes for v6.15 A relatively large set of fixes that came in since the release, mostly for Qualcomm platforms. The biggest block of fixes is the set from Srini which fixes various quality and glitching issues on AudioReach systems. ---------------------------------------------------------------- Alexey Klimov (1): ASoC: qdsp6: q6asm-dai: fix q6asm_dai_compr_set_params error path Christophe JAILLET (1): ASoC: codecs: rt5665: Fix some error handling paths in rt5665_probe() Dan Carpenter (1): ASoC: sma1307: Fix error handling in sma1307_setting_loaded() Henry Martin (1): ASoC: imx-card: Add NULL check in imx_card_probe() Krzysztof Kozlowski (2): ASoC: codecs: wsa883x: Correct VI sense channel mask ASoC: codecs: wsa884x: Correct VI sense channel mask Mark Brown (2): ASoC: codecs: wsa88xx: Correct VI sense channel mask ASoC: q6apm: fix under runs and fragment sizes Peter Ujfalusi (1): ASoC: SOF: hda/ptl: Move mic privacy change notification sending to a work Richard Fitzgerald (1): firmware: cs_dsp: Ensure cs_dsp_load[_coeff]() returns 0 on success Srinivas Kandagatla (5): ASoC: q6apm-dai: schedule all available frames to avoid dsp under-runs ASoC: q6apm: add q6apm_get_hw_pointer helper ASoC: qdsp6: q6apm-dai: set 10 ms period and buffer alignment. ASoC: qdsp6: q6apm-dai: fix capture pipeline overruns. ASoC: q6apm-dai: make use of q6apm_get_hw_pointer drivers/firmware/cirrus/cs_dsp.c | 2 ++ sound/soc/codecs/rt5665.c | 24 +++------------- sound/soc/codecs/sma1307.c | 11 ++------ sound/soc/codecs/wsa883x.c | 2 +- sound/soc/codecs/wsa884x.c | 2 +- sound/soc/fsl/imx-card.c | 4 +++ sound/soc/qcom/qdsp6/q6apm-dai.c | 60 +++++++++++++++++++++------------------- sound/soc/qcom/qdsp6/q6apm.c | 18 +++++++++++- sound/soc/qcom/qdsp6/q6apm.h | 3 ++ sound/soc/qcom/qdsp6/q6asm-dai.c | 19 +++++++++---- sound/soc/sof/intel/hda-dsp.c | 8 ++++++ sound/soc/sof/intel/hda.c | 4 +++ sound/soc/sof/intel/hda.h | 8 ++++++ sound/soc/sof/intel/ptl.c | 33 +++++++++++++++++++--- 14 files changed, 128 insertions(+), 70 deletions(-)

2 1

Support mute led function for HP platform
by Kailang 02 Apr '25

02 Apr '25

Hi Takashi, Attach was fixed mute led no function with HP platform. Many Thanks. BR, Kailang

2 1

aplay: add support for G.711 A_LAW enconding in AU file format
by GitHub pull_request - opened 02 Apr '25

02 Apr '25

alsa-project/alsa-utils pull request #297 was opened from chleroy: At the time being, aplay only supports G.711 µ-law, 8-bits PCM and 16-bits PCM encodings for the AU format while the AU format supports many more. In Europe the most G.711 used encoding is the A-law, which is also part of the encodings supported by the AU file format. Add support for that encoding in aplay. Request URL : https://github.com/alsa-project/alsa-utils/pull/297 Patch URL : https://github.com/alsa-project/alsa-utils/pull/297.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH alsa-ucm-conf 1/2] sof-soundwire: cs42l43: Drop headset mic from mic mute LED
by Maciej Strozek 01 Apr '25

01 Apr '25

As per conversation in [1], let headsets be dropped from LED bindings. Also remove turning off the mics on device disable since they are now not needed. [1]: https://github.com/alsa-project/alsa-ucm-conf/pull/512 Signed-off-by: Maciej Strozek <mstrozek(a)opensource.cirrus.com> --- ucm2/codecs/cs42l43/init.conf | 6 ------ ucm2/sof-soundwire/cs42l43-dmic.conf | 2 -- ucm2/sof-soundwire/cs42l43.conf | 1 - 3 files changed, 9 deletions(-) diff --git a/ucm2/codecs/cs42l43/init.conf b/ucm2/codecs/cs42l43/init.conf index 0e987c21..c0fa863c 100644 --- a/ucm2/codecs/cs42l43/init.conf +++ b/ucm2/codecs/cs42l43/init.conf @@ -10,9 +10,3 @@ LibraryConfig.remap.Config { } } } - -BootSequence [ - cset "name='cs42l43 Decimator 1 Switch' 0" -] - -Macro [{ SetLED { LED="mic" Action="attach" CtlId="cs42l43 Decimator 1 Switch" } }] diff --git a/ucm2/sof-soundwire/cs42l43-dmic.conf b/ucm2/sof-soundwire/cs42l43-dmic.conf index 70ce40ff..b68a70d1 100644 --- a/ucm2/sof-soundwire/cs42l43-dmic.conf +++ b/ucm2/sof-soundwire/cs42l43-dmic.conf @@ -13,8 +13,6 @@ SectionDevice."Mic" { ] DisableSequence [ - cset "name='cs42l43 Decimator 3 Switch' 0" - cset "name='cs42l43 Decimator 4 Switch' 0" cset "name='cs42l43 DP1TX1 Input' 'None'" cset "name='cs42l43 DP1TX2 Input' 'None'" ] diff --git a/ucm2/sof-soundwire/cs42l43.conf b/ucm2/sof-soundwire/cs42l43.conf index 4b4e1187..51832126 100644 --- a/ucm2/sof-soundwire/cs42l43.conf +++ b/ucm2/sof-soundwire/cs42l43.conf @@ -33,7 +33,6 @@ SectionDevice."Headset" { ] DisableSequence [ - cset "name='cs42l43 Decimator 1 Switch' 0" cset "name='cs42l43 DP2TX1 Input' 'None'" cset "name='cs42l43 DP2TX2 Input' 'None'" ] -- 2.39.5

2 6

Difference in behaviour between physical devices and virtual endpoints
by GitHub issues - opened 31 Mar '25

31 Mar '25

alsa-project/alsa-lib issue #447 was opened from reuk: As a disclaimer, there's a good chance this is user error/misunderstanding rather than a bug, but I'd appreciate any insights to help me understand the behaviour I'm seeing. # Environment - Fedora 41 x86_64 - alsa-lib-devel-1.2.13-3 # Steps - Run `aseqdump -u 1` in a terminal - Connect ProtoZOA hardware via USB - Launch QjackCtl -> Graph, then connect the ProtoZOA's MIDI 2.0 output port (0) to `aseqdump`'s input port. <img width="535" alt="Image" src="https://github.com/user-attachments/assets/17bf6a74-f83d-41b4-afc6-05413a7a…" /> - Observe that `aseqdump` reports a new subscription from `24:0 -> 128:0` (the client IDs may vary, the important thing is that the subscription is between to ports with ID `0`). - Move one of the controllers on the ProtoZOA. Observe that `aseqdump` reports channel voice events on group 0 with source `24:1`. <img width="502" alt="Image" src="https://github.com/user-attachments/assets/2dfc5929-123f-4e33-abbb-ea0c2635…" /> - Connect `aseqdump` to port `1:Group 1 (FB1, FB2)`, and interact with the ProtoZOA. Observe that `aseqdump` still reports channel voice events on group 0 with source `24:1`. - Now, build and run the [seq-ump-example](https://github.com/alsa-project/alsa-lib/blob/master/test/…. - Connect the ProtoZOA to the ACMESynth input, and the ACMESynth output to `aseqdump` as shown. Observe that `aseqdump` does **not** report channel voice events when interacting with the ProtoZOA. <img width="790" alt="Image" src="https://github.com/user-attachments/assets/71349cd6-14b8-4df9-8ee6-ca1091c2…" /> - Finally, update the connections as shown, so that `aseqdump` is connected to ACMESynth's `1:Group 1 (Filter 1)` output. Observe that `aseqdump` now reports channel voice events with source `128:1` when interacting with the ProtoZOA. <img width="772" alt="Image" src="https://github.com/user-attachments/assets/57f4e767-b2d6-4156-8514-578851d5…" /> # TL;DR When we subscribe to a USB `MIDI 2.0` port, we receive messages from all of the endpoint's ports. Messages are dispatched to the appropriate port based on the message group. When we subscribe to a virtual UMP endpoint's `MIDI 2.0` port, this appears to *only* subscribe to that single port, and not to the rest of the ports on the endpoint, so that messages addressed to specific groups get lost. # Questions - Is this difference in behaviour between hardware and virtual endpoints intentional? - What is the "correct" way to subscribe to all UMP events from an endpoint? Should I subscribe just to the `MIDI 2.0` endpoint, or to all ports on the endpoint, or something else? Issue URL : https://github.com/alsa-project/alsa-lib/issues/447 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH alsa-ucm-conf v1] sof-soundwire: Add LED support for cs35l56 amplifiers
by Stefan Binding 31 Mar '25

31 Mar '25

Systems may have up to 8 CS35L56 amplifiers, and each amplifier should be attached to the speaker LED. A mapped Playback switch is required to ensure all amps are muted and unmute appropriately. Signed-off-by: Stefan Binding <sbinding(a)opensource.cirrus.com> --- ucm2/codecs/cs35l56-bridge/init.conf | 11 ++++++++ ucm2/codecs/cs35l56/init.conf | 36 +++++++++++++++++++++++++++ ucm2/sof-soundwire/cs35l56.conf | 32 ++---------------------- ucm2/sof-soundwire/sof-soundwire.conf | 4 +-- 4 files changed, 51 insertions(+), 32 deletions(-) create mode 100644 ucm2/codecs/cs35l56-bridge/init.conf create mode 100644 ucm2/codecs/cs35l56/init.conf diff --git a/ucm2/codecs/cs35l56-bridge/init.conf b/ucm2/codecs/cs35l56-bridge/init.conf new file mode 100644 index 000000000000..407bb0937559 --- /dev/null +++ b/ucm2/codecs/cs35l56-bridge/init.conf @@ -0,0 +1,11 @@ +# cs35l56 specific control settings + +FixedBootSequence [ + cset "name='AMPL Speaker Switch' 0" + cset "name='AMPR Speaker Switch' 0" +] + +Macro [ + { SetLED { LED="speaker" Action="attach" CtlId="AMPL Speaker Switch" } } + { SetLED { LED="speaker" Action="attach" CtlId="AMPR Speaker Switch" } } +] \ No newline at end of file diff --git a/ucm2/codecs/cs35l56/init.conf b/ucm2/codecs/cs35l56/init.conf new file mode 100644 index 000000000000..dca14c709fda --- /dev/null +++ b/ucm2/codecs/cs35l56/init.conf @@ -0,0 +1,36 @@ +# cs35l56 specific control settings + +# +# Arguments: +# ForAmps - regex filter for var:SpeakerAmps +# Amp - amplifier number 1-8 +# +DefineMacro.cs35l56spkInit.If.0 { + Condition { + Type RegexMatch + Regex "${var:__ForAmps}" + String "${var:SpeakerAmps}" + } + True { + LibraryConfig.remap.Config.ctl.default.map { + "name='cs35l56 Speaker Playback Switch'" { + "name='AMP${var:__Amp} Speaker Switch'".vindex."${eval:($__Amp-1)}" 0 + } + } + FixedBootSequence [ + cset "name='AMP${var:__Amp} Speaker Switch' 0" + ] + Macro [{ SetLED { LED="speaker" Action="attach" CtlId="AMP${var:__Amp} Speaker Switch" } }] + } +} + +Macro [ + { cs35l56spkInit { ForAmps "[1-8]" Amp 1 } } + { cs35l56spkInit { ForAmps "[2-8]" Amp 2 } } + { cs35l56spkInit { ForAmps "[3-8]" Amp 3 } } + { cs35l56spkInit { ForAmps "[4-8]" Amp 4 } } + { cs35l56spkInit { ForAmps "[5-8]" Amp 5 } } + { cs35l56spkInit { ForAmps "[6-8]" Amp 6 } } + { cs35l56spkInit { ForAmps "[78]" Amp 7 } } + { cs35l56spkInit { ForAmps "8" Amp 8 } } +] diff --git a/ucm2/sof-soundwire/cs35l56.conf b/ucm2/sof-soundwire/cs35l56.conf index f74b99afbb55..53e4d6dd8eda 100644 --- a/ucm2/sof-soundwire/cs35l56.conf +++ b/ucm2/sof-soundwire/cs35l56.conf @@ -1,40 +1,12 @@ # Use case Configuration for sof-soundwire card -# -# Arguments: -# ForAmps - regex filter for var:SpeakerAmps -# Amp - amplifier number 1-8 -# -DefineMacro.cs42l43spk.If.0 { - Condition { - Type RegexMatch - Regex "${var:__ForAmps}" - String "${var:SpeakerAmps}" - } - True { - EnableSequence [ - cset "name='AMP${var:__Amp} Speaker Switch' 1" - ] - DisableSequence [ - cset "name='AMP${var:__Amp} Speaker Switch' 0" - ] - } -} - SectionDevice."Speaker" { Comment "Speaker" - Macro.num1.cs42l43spk { ForAmps "[1-8]" Amp 1 } - Macro.num2.cs42l43spk { ForAmps "[2-8]" Amp 2 } - Macro.num3.cs42l43spk { ForAmps "[3-8]" Amp 3 } - Macro.num4.cs42l43spk { ForAmps "[4-8]" Amp 4 } - Macro.num5.cs42l43spk { ForAmps "[5-8]" Amp 5 } - Macro.num6.cs42l43spk { ForAmps "[6-8]" Amp 6 } - Macro.num7.cs42l43spk { ForAmps "[78]" Amp 7 } - Macro.num8.cs42l43spk { ForAmps "8" Amp 8 } - Value { PlaybackPriority 100 PlaybackPCM "hw:${CardId},2" + PlaybackMixer "default:${CardId}" + PlaybackMixerElem "cs35l56 Speaker" } } diff --git a/ucm2/sof-soundwire/sof-soundwire.conf b/ucm2/sof-soundwire/sof-soundwire.conf index 1cddfbf4cfbf..8f9bc82c2d3b 100644 --- a/ucm2/sof-soundwire/sof-soundwire.conf +++ b/ucm2/sof-soundwire/sof-soundwire.conf @@ -102,10 +102,10 @@ If.multi_headset { } } -If.spk_init_rt1318 { +If.spk_init { Condition { Type RegexMatch - Regex "(rt1318(-1)?)" + Regex "(rt1318(-1)?|cs35l56(-bridge)?)" String "${var:SpeakerCodec1}" } True.Include.spk_init.File "/codecs/${var:SpeakerCodec1}/init.conf" -- 2.43.0

2 1

[PATCH v2] ASoC: amd: use new ACP dev names for DAI links
by Brady Norander 30 Mar '25

30 Mar '25

On AMD SoC platforms with an ACP2x gpu ip block (such as stoneyridge), the amdgpu driver will create several platform devices for the ACP ASoC driver to communicate with the ACP hardware block on the gpu. These platform devices include dma for audio and one or multiple i2s interfaces. The amdgpu driver has always created these platform devices with automatic ids. The ASoC machine drives hardcode the platform device name. This creates an issue where if the ACP platform devices are not the first to be created, the ids can be different to what the machine drivers expect, causing them to not find the ACP platform devices and failing to load. Switch to using static ids for these ACP platform devices so that the names never change. Depends on patch: drm/amdgpu: use static ids for ACP platform devs [1] [1] https://lore.kernel.org/all/20250325210517.2097188-1-bradynorander@gmail.co… Signed-off-by: Brady Norander <bradynorander(a)gmail.com> --- v2: rewrite commit message to better describe the issue --- sound/soc/amd/acp-da7219-max98357a.c | 8 ++++---- sound/soc/amd/acp-es8336.c | 4 ++-- sound/soc/amd/acp-rt5645.c | 6 +++--- 3 files changed, 9 insertions(+), 9 deletions(-) diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index 02b04f355ca6..42aa009c4e13 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -517,11 +517,11 @@ static const struct snd_soc_ops cz_rt5682_dmic1_cap_ops = { }; SND_SOC_DAILINK_DEF(designware1, - DAILINK_COMP_ARRAY(COMP_CPU("designware-i2s.1.auto"))); + DAILINK_COMP_ARRAY(COMP_CPU("designware-i2s.1"))); SND_SOC_DAILINK_DEF(designware2, - DAILINK_COMP_ARRAY(COMP_CPU("designware-i2s.2.auto"))); + DAILINK_COMP_ARRAY(COMP_CPU("designware-i2s.2"))); SND_SOC_DAILINK_DEF(designware3, - DAILINK_COMP_ARRAY(COMP_CPU("designware-i2s.3.auto"))); + DAILINK_COMP_ARRAY(COMP_CPU("designware-i2s.3"))); SND_SOC_DAILINK_DEF(dlgs, DAILINK_COMP_ARRAY(COMP_CODEC("i2c-DLGS7219:00", "da7219-hifi"))); @@ -533,7 +533,7 @@ SND_SOC_DAILINK_DEF(adau, DAILINK_COMP_ARRAY(COMP_CODEC("ADAU7002:00", "adau7002-hifi"))); SND_SOC_DAILINK_DEF(platform, - DAILINK_COMP_ARRAY(COMP_PLATFORM("acp_audio_dma.0.auto"))); + DAILINK_COMP_ARRAY(COMP_PLATFORM("acp_audio_dma.0"))); static struct snd_soc_dai_link cz_dai_7219_98357[] = { { diff --git a/sound/soc/amd/acp-es8336.c b/sound/soc/amd/acp-es8336.c index 0193b3eae7a6..b16dde0e2987 100644 --- a/sound/soc/amd/acp-es8336.c +++ b/sound/soc/amd/acp-es8336.c @@ -137,11 +137,11 @@ static const struct snd_soc_ops st_es8336_ops = { }; SND_SOC_DAILINK_DEF(designware1, - DAILINK_COMP_ARRAY(COMP_CPU("designware-i2s.2.auto"))); + DAILINK_COMP_ARRAY(COMP_CPU("designware-i2s.1"))); SND_SOC_DAILINK_DEF(codec, DAILINK_COMP_ARRAY(COMP_CODEC("i2c-ESSX8336:00", "ES8316 HiFi"))); SND_SOC_DAILINK_DEF(platform, - DAILINK_COMP_ARRAY(COMP_PLATFORM("acp_audio_dma.1.auto"))); + DAILINK_COMP_ARRAY(COMP_PLATFORM("acp_audio_dma.0"))); static struct snd_soc_dai_link st_dai_es8336[] = { { diff --git a/sound/soc/amd/acp-rt5645.c b/sound/soc/amd/acp-rt5645.c index 72ddad24dbda..11d373169380 100644 --- a/sound/soc/amd/acp-rt5645.c +++ b/sound/soc/amd/acp-rt5645.c @@ -108,15 +108,15 @@ static const struct snd_soc_ops cz_aif1_ops = { }; SND_SOC_DAILINK_DEF(designware1, - DAILINK_COMP_ARRAY(COMP_CPU("designware-i2s.1.auto"))); + DAILINK_COMP_ARRAY(COMP_CPU("designware-i2s.1"))); SND_SOC_DAILINK_DEF(designware2, - DAILINK_COMP_ARRAY(COMP_CPU("designware-i2s.2.auto"))); + DAILINK_COMP_ARRAY(COMP_CPU("designware-i2s.2"))); SND_SOC_DAILINK_DEF(codec, DAILINK_COMP_ARRAY(COMP_CODEC("i2c-10EC5650:00", "rt5645-aif1"))); SND_SOC_DAILINK_DEF(platform, - DAILINK_COMP_ARRAY(COMP_PLATFORM("acp_audio_dma.0.auto"))); + DAILINK_COMP_ARRAY(COMP_PLATFORM("acp_audio_dma.0"))); static struct snd_soc_dai_link cz_dai_rt5650[] = { { -- 2.49.0

1 0

USB-Audio: Add focusrite scarlett 18i20 lineup
by GitHub pull_request - opened 30 Mar '25

30 Mar '25

alsa-project/alsa-ucm-conf pull request #539 was opened from ScarletRav3n: Other solutions exist to add full mixing and routing with these devices. This PR adds basic functionality in an out-of-the-box style for 1st gen, 2nd gen, 3rd gen, and 4th gen. IO includes XLR, Line, S/PDIF, ADAT, and Headphone ports. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/539 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/539.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

USB-Audio: Add Focusrite Scarlett 18i20 Lineup
by GitHub pull_request - edited 30 Mar '25

30 Mar '25

alsa-project/alsa-ucm-conf pull request #538 was edited from ScarletRav3n: Other solutions exist to add full mixing and routing with these devices. This PR adds basic functionality in an out-of-the-box style for 1st gen, 2nd gen, 3rd gen, and 4th gen. I/O includes all XLR, Line, S/PDIF, ADAT, and Headphone ports. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/538 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/538.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

USB-Audio: Add Focusrite Scarlett 18i20 Lineup
by GitHub pull_request - opened 30 Mar '25

30 Mar '25

alsa-project/alsa-ucm-conf pull request #538 was opened from ScarletRav3n: Other solutions exist to add full mixing and routing with these devices. This PR adds basic functionality in an out-of-the-box style for 2nd gen, 3rd gen, and 4th gen. 1st gen is not supported with this iteration of PR. I/O includes all XLR, Line, S/PDIF, ADAT, and Headphone ports. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/538 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/538.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

envy24control: fix memory access errors in profiles
by GitHub pull_request - opened 29 Mar '25

29 Mar '25

alsa-project/alsa-tools pull request #34 was opened from andreas56: Fix two memory errors in the profiles parser: an invalid read and a "source and destination overlap in strncpy" warning. When the profiles page is initialized it fetches the profile names from the profiles file. When a profile wasn't defined in the file, the parser made invalid reads outside the buffer. Request URL : https://github.com/alsa-project/alsa-tools/pull/34 Patch URL : https://github.com/alsa-project/alsa-tools/pull/34.patch Repository URL: https://github.com/alsa-project/alsa-tools

1 0

envy24control: fix file descriptor leaks in profiles
by GitHub pull_request - opened 29 Mar '25

29 Mar '25

alsa-project/alsa-tools pull request #33 was opened from andreas56: File descriptors were leaked when "Save active profile" was pressed. Add the missing calls to close. Request URL : https://github.com/alsa-project/alsa-tools/pull/33 Patch URL : https://github.com/alsa-project/alsa-tools/pull/33.patch Repository URL: https://github.com/alsa-project/alsa-tools

1 0

envy24control: fix crash when using system profiles file
by GitHub pull_request - opened 29 Mar '25

29 Mar '25

alsa-project/alsa-tools pull request #32 was opened from andreas56: envy24control crashed if you tried to start it with -f /etc/envy24control/profiles.conf. Request URL : https://github.com/alsa-project/alsa-tools/pull/32 Patch URL : https://github.com/alsa-project/alsa-tools/pull/32.patch Repository URL: https://github.com/alsa-project/alsa-tools

1 0

envy24control: fix GtkSpinButton runtime warnings
by GitHub pull_request - opened 29 Mar '25

29 Mar '25

alsa-project/alsa-tools pull request #31 was opened from andreas56: Change the parameters to gtk_adjustment_new to get rid of this runtime warning from GTK: GtkSpinButton: setting an adjustment with non-zero page size is deprecated The change is also important when envy24control is ported to GTK 3, as the spin buttons will stop working if it's not done. Request URL : https://github.com/alsa-project/alsa-tools/pull/31 Patch URL : https://github.com/alsa-project/alsa-tools/pull/31.patch Repository URL: https://github.com/alsa-project/alsa-tools

1 0

envy24control: fix control callback for spdif output
by GitHub pull_request - opened 29 Mar '25

29 Mar '25

alsa-project/alsa-tools pull request #30 was opened from andreas56: Update S/PDIF Output Settings GUI when settings change. Request URL : https://github.com/alsa-project/alsa-tools/pull/30 Patch URL : https://github.com/alsa-project/alsa-tools/pull/30.patch Repository URL: https://github.com/alsa-project/alsa-tools

1 0

[PATCH alsa-ucm-conf v2 1/2] sof-soundwire: cs42l43: Drop headset mic from mic mute LED
by Maciej Strozek 28 Mar '25

28 Mar '25

As per conversation in [1], let headsets be dropped from LED bindings. [1]: https://github.com/alsa-project/alsa-ucm-conf/pull/512 Signed-off-by: Maciej Strozek <mstrozek(a)opensource.cirrus.com> --- v2: Added DisableSequences back --- ucm2/codecs/cs42l43/init.conf | 6 ------ 1 file changed, 6 deletions(-) diff --git a/ucm2/codecs/cs42l43/init.conf b/ucm2/codecs/cs42l43/init.conf index 0e987c21..c0fa863c 100644 --- a/ucm2/codecs/cs42l43/init.conf +++ b/ucm2/codecs/cs42l43/init.conf @@ -10,9 +10,3 @@ LibraryConfig.remap.Config { } } } - -BootSequence [ - cset "name='cs42l43 Decimator 1 Switch' 0" -] - -Macro [{ SetLED { LED="mic" Action="attach" CtlId="cs42l43 Decimator 1 Switch" } }] -- 2.39.5

1 1

[PATCH v1] ALSA: hda/tas2781: Upgrade calibratd-data writing code to support Alpha and Beta dsp firmware
by Shenghao Ding 28 Mar '25

28 Mar '25

Since 2025, the firmware for tas2781 has been added more audio acoustic features, such as non-linear compensation, advanced battery guard, rattle-noise suppression, etc. The version was divided into two different series. Both series have a slight change on the calibrated data storage addresses, which becames flexible instead of fixed. In order to support new firwmares in time, the code have some related upgrades. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v2: - revise the subject and description. v1: - Add updating calibration addresses code into tas2781_apply_calib in case of Alpha and Beta firmware. --- sound/pci/hda/tas2781_hda_i2c.c | 30 ++++++++++++++++++++---------- 1 file changed, 20 insertions(+), 10 deletions(-) diff --git a/sound/pci/hda/tas2781_hda_i2c.c b/sound/pci/hda/tas2781_hda_i2c.c index 9ed49b0dbe6b..29dc4f500580 100644 --- a/sound/pci/hda/tas2781_hda_i2c.c +++ b/sound/pci/hda/tas2781_hda_i2c.c @@ -558,28 +558,38 @@ static int tas2563_save_calibration(struct tasdevice_priv *tas_priv) static void tas2781_apply_calib(struct tasdevice_priv *tas_priv) { - static const unsigned char page_array[CALIB_MAX] = { - 0x17, 0x18, 0x18, 0x13, 0x18, + struct calidata *cali_data = &tas_priv->cali_data; + struct cali_reg *r = &cali_data->cali_reg_array; + unsigned int cali_reg[CALIB_MAX] = { + TASDEVICE_REG(0, 0x17, 0x74), + TASDEVICE_REG(0, 0x18, 0x0c), + TASDEVICE_REG(0, 0x18, 0x14), + TASDEVICE_REG(0, 0x13, 0x70), + TASDEVICE_REG(0, 0x18, 0x7c), }; - static const unsigned char rgno_array[CALIB_MAX] = { - 0x74, 0x0c, 0x14, 0x70, 0x7c, - }; - int offset = 0; int i, j, rc; + int oft = 0; __be32 data; + if (tas_priv->dspbin_typ != TASDEV_BASIC) { + cali_reg[0] = r->r0_reg; + cali_reg[1] = r->invr0_reg; + cali_reg[2] = r->r0_low_reg; + cali_reg[3] = r->pow_reg; + cali_reg[4] = r->tlimit_reg; + } + for (i = 0; i < tas_priv->ndev; i++) { for (j = 0; j < CALIB_MAX; j++) { data = cpu_to_be32( - *(uint32_t *)&tas_priv->cali_data.data[offset]); + *(uint32_t *)&tas_priv->cali_data.data[oft]); rc = tasdevice_dev_bulk_write(tas_priv, i, - TASDEVICE_REG(0, page_array[j], rgno_array[j]), - (unsigned char *)&data, 4); + cali_reg[j], (unsigned char *)&data, 4); if (rc < 0) dev_err(tas_priv->dev, "chn %d calib %d bulk_wr err = %d\n", i, j, rc); - offset += 4; + oft += 4; } } } -- 2.34.1

2 1

[ alsa-lib ] OpenHarmony Adaptation Proposal
by GitHub issues - opened 28 Mar '25

28 Mar '25

alsa-project/alsa-lib issue #445 was opened from Asercn: # [ alsa-lib ] OpenHarmony Adaptation Proposal ## Background OpenHarmony is an open-source project incubated and operated by the OpenAtom Open Source Foundation. Its goal is to build an operating system framework and platform for intelligent terminal devices in the era of full scenarios, full connectivity, and full intelligence. ALSA, as an important audio architecture in Linux systems, allows developers to use Linux system calls in programs to control and manage the underlying operating system. Due to its efficient and easy-to-use features, it is also suitable for embedded devices, which aligns well with OpenHarmony's concept of supporting a variety of devices. Currently, ALSA is widely used in the mobile application development field of OpenHarmony. We will submit the adaptation code via a pull request (PR) as soon as possible. This issue is created to engage in in-depth discussions with the community about integrating the adaptation code. We sincerely welcome any valuable suggestions. ## Adaptation Plan 1. Details of the ALSA adaptation plan: 2. Submit back to the main community. 3. Ongoing community maintenance. ## Test Results We have completed large-scale compatibility testing, covering different versions of HarmonyOS and various types of HarmonyOS devices. The test results show that the modified version is highly compatible with the HarmonyOS system and no significant stability issues have been observed. Benefits to the Upstream Community * Can reach OpenHarmony’s large user base and developer community. * By supporting the development trends of the operating system, it ensures the long-term development potential of the project. * Enhances the project's processing capability in diverse operating environments. Issue URL : https://github.com/alsa-project/alsa-lib/issues/445 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Add UCM2 support for ASUS ROG Strix B850-I Gaming WiFi
by GitHub pull_request - edited 28 Mar '25

28 Mar '25

alsa-project/alsa-ucm-conf pull request #537 was edited from austinthiel: [Link to open issue](https://github.com/alsa-project/alsa-ucm-conf/issues/534#issue-29483… I referenced [this guide](https://askubuntu.com/a/1469066) to get SPDIF out working on my ASUS B850-I Gaming WiFi [alsa-info.txt](https://github.com/user-attachments/files/19499096/alsa-info… In case it's relevant, using a sample rate of 48kHz is required. I found that other sample rate produces very low quality, garbled audio. I read that this is because SPDIF only supports 48kHz in hardware, and any other sample rates require the Realtek drivers. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/537 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/537.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Add UCM2 support for ASUS ROG Strix B850-I Gaming WiFi
by GitHub pull_request - edited 28 Mar '25

28 Mar '25

alsa-project/alsa-ucm-conf pull request #537 was edited from austinthiel: Referenced [this guide](https://askubuntu.com/a/1469066) to get SPDIF out working on my ASUS B850-I Gaming WiFi [alsa-info.txt](https://github.com/user-attachments/files/19499096/alsa-info… In case it's relevant, using a sample rate of 48kHz is required. I found that other sample rate produces very low quality, garbled audio. I read that this is because SPDIF only supports 48kHz in hardware, and any other sample rates require the Realtek drivers. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/537 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/537.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Add UCM2 support for ASUS ROG Strix B850-I Gaming WiFi
by GitHub pull_request - edited 28 Mar '25

28 Mar '25

alsa-project/alsa-ucm-conf pull request #537 was edited from austinthiel: Referenced [this guide](https://askubuntu.com/a/1469066) to get SPDIF out working on my ASUS B850-I Gaming WiFi [alsa-info.txt](https://github.com/user-attachments/files/19499096/alsa-info… In case it's relevant, using a sample rate of 48kHz is required. I found that other sample rate produces very low quality, garbled audio. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/537 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/537.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Add UCM2 support for ASUS ROG Strix B850-I Gaming WiFi
by GitHub pull_request - opened 28 Mar '25

28 Mar '25

alsa-project/alsa-ucm-conf pull request #537 was opened from austinthiel: Referenced [this guide](https://askubuntu.com/a/1469066) to get SPDIF out working on my ASUS B850-I Gaming WiFi [alsa-info.txt](https://github.com/user-attachments/files/19498803/alsa-info… In case it's relevant, using a sample rate of 48kHz is required. I found that other sample rate produces very low quality, garbled audio. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/537 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/537.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

USB-Audio Add Roland BridgeCast One USB-Audio.conf
by GitHub pull_request - opened 27 Mar '25

27 Mar '25

alsa-project/alsa-ucm-conf pull request #536 was opened from Simon99de: Add Roland BridgeCast One uses the same config as the bridge cast v2 as it has the same outputs and it worked in my test. 0582:030d Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/536 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/536.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

USB-Audio: ALC4080 - add rear microphone support for 0414:a014
by GitHub pull_request - opened 26 Mar '25

26 Mar '25

alsa-project/alsa-ucm-conf pull request #535 was opened from perexg: 0414:a014 Gigabyte B850I Aorus Pro (rev 1.0) Link: https://github.com/alsa-project/alsa-ucm-conf/issues/528 Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/535 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/535.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH 2/3] ASoC: amd: use new ACP dev names for DAI links
by Brady Norander 26 Mar '25

26 Mar '25

The old names used automatic platform device ids, which means they could change. Use the new device names which will never change. Depends on patch: drm/amdgpu: use static ids for ACP platform devs [1] [1] https://lore.kernel.org/all/20250325210517.2097188-1-bradynorander@gmail.co… Signed-off-by: Brady Norander <bradynorander(a)gmail.com> --- sound/soc/amd/acp-da7219-max98357a.c | 8 ++++---- sound/soc/amd/acp-es8336.c | 4 ++-- sound/soc/amd/acp-rt5645.c | 6 +++--- 3 files changed, 9 insertions(+), 9 deletions(-) diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index 02b04f355ca6..42aa009c4e13 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -517,11 +517,11 @@ static const struct snd_soc_ops cz_rt5682_dmic1_cap_ops = { }; SND_SOC_DAILINK_DEF(designware1, - DAILINK_COMP_ARRAY(COMP_CPU("designware-i2s.1.auto"))); + DAILINK_COMP_ARRAY(COMP_CPU("designware-i2s.1"))); SND_SOC_DAILINK_DEF(designware2, - DAILINK_COMP_ARRAY(COMP_CPU("designware-i2s.2.auto"))); + DAILINK_COMP_ARRAY(COMP_CPU("designware-i2s.2"))); SND_SOC_DAILINK_DEF(designware3, - DAILINK_COMP_ARRAY(COMP_CPU("designware-i2s.3.auto"))); + DAILINK_COMP_ARRAY(COMP_CPU("designware-i2s.3"))); SND_SOC_DAILINK_DEF(dlgs, DAILINK_COMP_ARRAY(COMP_CODEC("i2c-DLGS7219:00", "da7219-hifi"))); @@ -533,7 +533,7 @@ SND_SOC_DAILINK_DEF(adau, DAILINK_COMP_ARRAY(COMP_CODEC("ADAU7002:00", "adau7002-hifi"))); SND_SOC_DAILINK_DEF(platform, - DAILINK_COMP_ARRAY(COMP_PLATFORM("acp_audio_dma.0.auto"))); + DAILINK_COMP_ARRAY(COMP_PLATFORM("acp_audio_dma.0"))); static struct snd_soc_dai_link cz_dai_7219_98357[] = { { diff --git a/sound/soc/amd/acp-es8336.c b/sound/soc/amd/acp-es8336.c index 0193b3eae7a6..b16dde0e2987 100644 --- a/sound/soc/amd/acp-es8336.c +++ b/sound/soc/amd/acp-es8336.c @@ -137,11 +137,11 @@ static const struct snd_soc_ops st_es8336_ops = { }; SND_SOC_DAILINK_DEF(designware1, - DAILINK_COMP_ARRAY(COMP_CPU("designware-i2s.2.auto"))); + DAILINK_COMP_ARRAY(COMP_CPU("designware-i2s.1"))); SND_SOC_DAILINK_DEF(codec, DAILINK_COMP_ARRAY(COMP_CODEC("i2c-ESSX8336:00", "ES8316 HiFi"))); SND_SOC_DAILINK_DEF(platform, - DAILINK_COMP_ARRAY(COMP_PLATFORM("acp_audio_dma.1.auto"))); + DAILINK_COMP_ARRAY(COMP_PLATFORM("acp_audio_dma.0"))); static struct snd_soc_dai_link st_dai_es8336[] = { { diff --git a/sound/soc/amd/acp-rt5645.c b/sound/soc/amd/acp-rt5645.c index 72ddad24dbda..11d373169380 100644 --- a/sound/soc/amd/acp-rt5645.c +++ b/sound/soc/amd/acp-rt5645.c @@ -108,15 +108,15 @@ static const struct snd_soc_ops cz_aif1_ops = { }; SND_SOC_DAILINK_DEF(designware1, - DAILINK_COMP_ARRAY(COMP_CPU("designware-i2s.1.auto"))); + DAILINK_COMP_ARRAY(COMP_CPU("designware-i2s.1"))); SND_SOC_DAILINK_DEF(designware2, - DAILINK_COMP_ARRAY(COMP_CPU("designware-i2s.2.auto"))); + DAILINK_COMP_ARRAY(COMP_CPU("designware-i2s.2"))); SND_SOC_DAILINK_DEF(codec, DAILINK_COMP_ARRAY(COMP_CODEC("i2c-10EC5650:00", "rt5645-aif1"))); SND_SOC_DAILINK_DEF(platform, - DAILINK_COMP_ARRAY(COMP_PLATFORM("acp_audio_dma.0.auto"))); + DAILINK_COMP_ARRAY(COMP_PLATFORM("acp_audio_dma.0"))); static struct snd_soc_dai_link cz_dai_rt5650[] = { { -- 2.49.0

2 2

[PATCH 3/3] ASoC: dwc: always enable/disable i2s irqs
by Brady Norander 26 Mar '25

26 Mar '25

Commit a42e988 ("ASoC: dwc: add DMA handshake control") changed the behavior of the driver to not enable or disable i2s irqs if using DMA. This breaks platforms such as AMD ACP. Revert to the old behavior by always enabling and disabling i2s irqs while keeping DMA handshake control. Fixes: a42e988 ("ASoC: dwc: add DMA handshake control") Signed-off-by: Brady Norander <bradynorander(a)gmail.com> --- sound/soc/dwc/dwc-i2s.c | 13 ++++++------- 1 file changed, 6 insertions(+), 7 deletions(-) diff --git a/sound/soc/dwc/dwc-i2s.c b/sound/soc/dwc/dwc-i2s.c index 57b789d7fbed..5b4f20dbf7bb 100644 --- a/sound/soc/dwc/dwc-i2s.c +++ b/sound/soc/dwc/dwc-i2s.c @@ -199,12 +199,10 @@ static void i2s_start(struct dw_i2s_dev *dev, else i2s_write_reg(dev->i2s_base, IRER, 1); - /* I2S needs to enable IRQ to make a handshake with DMAC on the JH7110 SoC */ - if (dev->use_pio || dev->is_jh7110) - i2s_enable_irqs(dev, substream->stream, config->chan_nr); - else + if (!(dev->use_pio || dev->is_jh7110)) i2s_enable_dma(dev, substream->stream); + i2s_enable_irqs(dev, substream->stream, config->chan_nr); i2s_write_reg(dev->i2s_base, CER, 1); } @@ -218,11 +216,12 @@ static void i2s_stop(struct dw_i2s_dev *dev, else i2s_write_reg(dev->i2s_base, IRER, 0); - if (dev->use_pio || dev->is_jh7110) - i2s_disable_irqs(dev, substream->stream, 8); - else + if (!(dev->use_pio || dev->is_jh7110)) i2s_disable_dma(dev, substream->stream); + i2s_disable_irqs(dev, substream->stream, 8); + + if (!dev->active) { i2s_write_reg(dev->i2s_base, CER, 0); i2s_write_reg(dev->i2s_base, IER, 0); -- 2.49.0

2 1

No output when selecting S/PDIF on Asus B850-I with Realtek ALC4080
by GitHub issues - opened 26 Mar '25

26 Mar '25

alsa-project/alsa-ucm-conf issue #534 was opened from bcrescimanno: Motherobard site: https://rog.asus.com/us/motherboards/rog-strix/rog-strix-b850-i-gaming-wifi/ I get no output from the S/PDIF port on this motherboard. I'm able to workaround it using `pavucontrol` and switching the USB Audio device to the "Pro Audio" configuration and then choosing "USB Pro Audio 2" as the output device. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/534 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

DRAFT: HDA: Handle better Bass speaker including 4 channel mode
by GitHub pull_request - edited 25 Mar '25

25 Mar '25

alsa-project/alsa-ucm-conf pull request #533 was edited from perexg: It's a follow up of #410 . Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/533 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/533.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

HDA: Handle better Bass speaker including 4 channel mode
by GitHub pull_request - opened 25 Mar '25

25 Mar '25

alsa-project/alsa-ucm-conf pull request #533 was opened from perexg: It's a follow up of #410 . Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/533 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/533.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v1] ALSA: hda/tas2781: Support dsp firmware Alpha and Beta seaies
by Shenghao Ding 25 Mar '25

25 Mar '25

For calibration data, basic version firmware does not contain any calibration addresses, it depends on calibration tool to convey the addresses to the driver. Since Alpha and Beta firmware, all the calibration addresses are saved into the firmware. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v1: - Add updating calibration addresses code into tas2781_apply_calib in case of Alpha and Beta firmware. --- sound/pci/hda/tas2781_hda_i2c.c | 30 ++++++++++++++++++++---------- 1 file changed, 20 insertions(+), 10 deletions(-) diff --git a/sound/pci/hda/tas2781_hda_i2c.c b/sound/pci/hda/tas2781_hda_i2c.c index 50c5e5f26589..cb3d683013d9 100644 --- a/sound/pci/hda/tas2781_hda_i2c.c +++ b/sound/pci/hda/tas2781_hda_i2c.c @@ -558,28 +558,38 @@ static int tas2563_save_calibration(struct tasdevice_priv *tas_priv) static void tas2781_apply_calib(struct tasdevice_priv *tas_priv) { - static const unsigned char page_array[CALIB_MAX] = { - 0x17, 0x18, 0x18, 0x13, 0x18, + struct calidata *cali_data = &tas_priv->cali_data; + struct cali_reg *r = &cali_data->cali_reg_array; + unsigned int cali_reg[CALIB_MAX] = { + TASDEVICE_REG(0, 0x17, 0x74), + TASDEVICE_REG(0, 0x18, 0x0c), + TASDEVICE_REG(0, 0x18, 0x14), + TASDEVICE_REG(0, 0x13, 0x70), + TASDEVICE_REG(0, 0x18, 0x7c), }; - static const unsigned char rgno_array[CALIB_MAX] = { - 0x74, 0x0c, 0x14, 0x70, 0x7c, - }; - int offset = 0; int i, j, rc; + int oft = 0; __be32 data; + if (tas_priv->dspbin_typ != TASDEV_BASIC) { + cali_reg[0] = r->r0_reg; + cali_reg[1] = r->invr0_reg; + cali_reg[2] = r->r0_low_reg; + cali_reg[3] = r->pow_reg; + cali_reg[4] = r->tlimit_reg; + } + for (i = 0; i < tas_priv->ndev; i++) { for (j = 0; j < CALIB_MAX; j++) { data = cpu_to_be32( - *(uint32_t *)&tas_priv->cali_data.data[offset]); + *(uint32_t *)&tas_priv->cali_data.data[oft]); rc = tasdevice_dev_bulk_write(tas_priv, i, - TASDEVICE_REG(0, page_array[j], rgno_array[j]), - (unsigned char *)&data, 4); + cali_reg[j], (unsigned char *)&data, 4); if (rc < 0) dev_err(tas_priv->dev, "chn %d calib %d bulk_wr err = %d\n", i, j, rc); - offset += 4; + oft += 4; } } } -- 2.34.1

2 1

[GIT PULL] ASoC updates for v6.15
by Mark Brown 24 Mar '25

24 Mar '25

The following changes since commit 4701f33a10702d5fc577c32434eb62adde0a1ae1: Linux 6.14-rc7 (2025-03-16 12:55:17 -1000) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v6.15 for you to fetch changes up to 9ef52d529bb75071e03cf85078f724d69c4abe89: ASoC: SDCA: Correct handling of selected mode DisCo property (2025-03-21 15:45:25 +0000) ---------------------------------------------------------------- ASoC: Updates for v6.15 This is a very big release due to a combination of some big new work, mainly new drivers and generic SoundWire support, and some wide ranging cleanup work that made small changes to a lot of drivers. - Morimoto-san has completed the conversion to use modern terminology for the clocking configuration, and several other cleanups with narrower impact. - All the power management operation configuration was updated to use current idioms by Takashi Iwai. - Clarification of the control operations from Charles Keepax. - Prepartory work for more generic SoundWire SCDA controls from Charles Keepax. - Support for AMD ACP 7.x, AWINC WM88166, Everest ES8388, Intel AVS PEAKVOL and GAIN DSP modules Mediatek MT8188 DMIC, NXP i.MX95, nVidia Tegra interconnects, Rockchip RK3588 S/PDIF, Texas Instruments SN012776 and TAS5770L, and Wolfson WM8904 DMICs, Some changes from the tip tree adding APIs needed by the AMD code are included, these were unfortunately rebased in the tip tree after being pulled in. There's also some regmap changes supporting the SCDA work and some devres refactoring that was pulled in to support other changes. ---------------------------------------------------------------- Alexey Charkov (1): dt-bindings: ASoC: rockchip: Add compatible for RK3588 SPDIF Alexey Klimov (2): ASoC: codecs: wsa883x: Implement temperature reading and hwmon ASoC: qcom: sm8250: explicitly set format in sm8250_be_hw_params_fixup() Amadeusz Sławiński (3): ASoC: Intel: avs: Configure basefw on TGL-based platforms ASoC: Intel: avs: Add volume control for GAIN module ASoC: Intel: avs: Add support for mute for PEAKVOL and GAIN Andrei Simion (1): ASoC: dt-bindings: atmel,at91-ssc: Convert to YAML format Andy Shevchenko (4): ASoC: tas2781: Switch to use %ptTsr driver core: Split devres APIs to device/devres.h iio: imu: st_lsm9ds0: Replace device.h with what is needed ASoC: SOF: ipc4-topology: Allocate ref_params on stack AngeloGioacchino Del Regno (1): ASoC: SOF: mediatek: Commonize duplicated functions Arnd Bergmann (2): ASoC: mediatek: mt8188: avoid uninitialized variable use ASoC: cs42l43: convert to SYSTEM_SLEEP_PM_OPS Balakrishnan Sambath (1): ASoC: dt-bindings: atmel-at91sam9g20ek: convert to json-schema Baojun Xu (1): ASoC: tas2781: Clean up for some define Bard Liao (8): ASoC: rt722: get lane mapping property ASoC: Intel: adl: add 2xrt1316 audio configuration ASoC: sdw_utils: add mic and amp dais to 0xaaaa codec ASoC: Intel: add multi-function SDW mockup codec match ASoC: Intel: soc-acpi-intel-lnl-match: add sdw multi function mockup codec ASoC: Intel: soc-acpi-intel-ptl-match: add sdw multi function mockup codec ASoC: Intel: soc-acpi-intel-lnl-match: add cs42l43 6x cs35l56 support ASoC: Intel: soc-acpi-intel-ptl-match: add cs42l43 6x cs35l56 support Cezary Rojewski (40): ASoC: codecs: pcm3168a: Add ACPI match table ASoC: codecs: pcm3168a: Relax probing conditions ASoC: codecs: pcm3168a: Allow for 24-bit in provider mode ASoC: Intel: avs: Add pcm3168a machine board ASoC: Intel: avs: pcm3168a board selection ASoC: Intel: avs: Move DSP-boot steps into individual functions ASoC: Intel: avs: New gateway configuration mechanism ASoC: Intel: avs: Remove unused gateway configuration code ASoC: Intel: avs: Add WHM module support ALSA: hda: Select avs-driver by default on MBL ASoC: topology: Create kcontrols based on their type ASoC: topology: Save num_channels value for mixer controls ASoC: Intel: avs: Make PEAKVOL configurable from topology ASoC: Intel: avs: Update VOLUME and add MUTE IPCs ASoC: Intel: avs: New volume control operations ASoC: Intel: avs: Move to the new control operations ASoC: Intel: avs: Honor the invert flag for mixer controls ASoC: Intel: avs: Support multi-channel PEAKVOL instantiation ASoC: codecs: rt1011: Fix definition of device_id tables ASoC: codecs: rt1015: Fix definition of device_id tables ASoC: codecs: rt1016: Fix definition of device_id tables ASoC: codecs: rt1019: Fix definition of device_id tables ASoC: codecs: rt1305: Fix definition of device_id tables ASoC: codecs: rt1308: Fix definition of device_id tables ASoC: codecs: rt1318: Fix definition of device_id tables ASoC: codecs: rt274: Fix definition of device_id tables ASoC: codecs: rt5514: Fix definition of device_id tables ASoC: codecs: rt5640: Fix definition of device_id tables ASoC: codecs: rt5645: Fix definition of device_id tables ASoC: codecs: rt5651: Fix definition of device_id tables ASoC: codecs: rt5659: Fix definition of device_id tables ASoC: codecs: rt5660: Fix definition of device_id tables ASoC: codecs: rt5663: Fix definition of device_id tables ASoC: codecs: rt5665: Fix definition of device_id tables ASoC: codecs: rt5668: Fix definition of device_id tables ASoC: codecs: rt5670: Fix definition of device_id tables ASoC: codecs: rt5682: Fix definition of device_id tables ASoC: codecs: rt286: Update definition of device_id tables ASoC: codecs: rt298: Update definition of device_id tables ASoC: codecs: rt5677: Update definition of device_id tables Charles Keepax (50): ASoC: rt722-sdca: Add some missing readable registers ASoC: rt722-sdca: Make use of new expanded MBQ regmap ASoC: SDCA: Minor formatting and naming tweaks ASoC: SDCA: Add support for Entity 0 ASoC: SDCA: Add SDCA Control parsing ASoC: SDCA: Add parsing for Control range structures ASoC: SDCA: Add Channel Cluster parsing ASoC: SDCA: Add support for IT/OT Entity properties ASoC: SDCA: Add support for clock Entity properties ASoC: SDCA: Add support for PDE Entity properties regcache: Add support for sorting defaults arrays ASoC: SDCA: Add generic regmap SDCA helpers ASoC: SDCA: Add regmap helpers for parsing for DisCo Constant values ASoC: SDCA: Add helper to write out defaults and fixed values ASoC: Remove unused helper macro ASoC: rt715: Remove duplicate SOC_DOUBLE_R_EXT() helper macro ASoC: sma1307: Use SOC_SINGLE_EXT() helper macro ASoC: tas2562: Use SOC_SINGLE_EXT_TLV() helper macro ASoC: wcd938x: Use SOC_SINGLE_EXT_TLV() helper macro ASoC: wm9712: Use SOC_SINGLE_EXT() helper macro ASoC: wm9713: Use SOC_DOUBLE_EXT() helper macro ASoC: wsa881x: Use SOC_SINGLE_EXT_TLV() helper macro ASoC: atmel: tse850-pcm5142: Use SOC_SINGLE_EXT() helper macro ASoC: dapm: Add missing SOC_DAPM_DOUBLE_R_TLV() helper ASoC: dapm: Use ASoC control macros where possible ASoC: Tidy up SOC_DOUBLE_R_* helpers ASoC: Tidy up SOC_DOUBLE_* and SOC_SINGLE_* helpers ASoC: dapm: Fix changes to DECLARE_ADAU17X1_DSP_MUX_CTRL ASoC: SDCA: Tidy up initialization write parsing ASoC: SDCA: Use __free() to manage local buffers ASoC: SDCA: Allow naming of imp def controls ASoC: SDCA: Add type flag for Controls ASoC: SDCA: Add SDCA Control Range data access helper ASoC: SDCA: Add support for GE Entity properties ASoC: ops-test: Add some basic kunit tests for soc-ops ASoC: ops: Minor formatting fixups ASoC: ops: Update comments for xr_sx control helpers ASoC: ops: Update mask generation to use GENMASK ASoC: ops: Factor out helper to check valid control values ASoC: ops: Replace snd_soc_read_signed() with new helper ASoC: ops: Add control to register value helper ASoC: ops: Remove snd_soc_info_volsw_range() ASoC: ops: Remove snd_soc_get_volsw_range() ASoC: ops: Remove snd_soc_put_volsw_range() ASoC: ops: Factor out common code from info callbacks ASoC: ops: Factor out common code from put callbacks ASoC: ops: Factor out common code from get callbacks ASoC: ops: Remove some unnecessary local variables ASoC: ops: Apply platform_max after deciding control type ASoC: SDCA: Correct handling of selected mode DisCo property Chen Ni (5): ASoC: ti: davinci-i2s: remove unnecessary NULL check before clk_disable_unprepare() ASoC: mt8365: remove unnecessary NULL check before clk_disable_unprepare() ASoC: tlv320adc3xxx: remove unnecessary NULL check before clk_disable_unprepare() ASoC: cs35l56: Remove redundant 'flush_workqueue()' calls ASoC: codecs: ntp8918: Remove duplicate clk.h header Chenyuan Yang (1): ASoC: sma1307: Add NULL check in sma1307_setting_loaded() Chris Morgan (1): ASoC: dt-bindings: sun4i-a10-codec: add hp-det-gpios Dan Carpenter (3): ASoC: SOF: imx: Fix an IS_ERR() vs NULL bug in imx_parse_ioremap_memory() ASoC: SOF: imx: Fix error code in probe() ASoC: amd: acp: Fix leak in acp_pci_probe() David Lechner (2): gpiolib: add gpiod_multi_set_value_cansleep() ASoC: adau1701: use gpiod_multi_set_value_cansleep Dr. David Alan Gilbert (3): ASoC: mediatek: Remove unused mtk_memif_set_rate ASoC: mediatek: mt6358: Remove unused functions ASoC: mediatek: mt8186: Remove unused mt8186_afe_(suspend|resume)_clock Dzmitry Sankouski (2): ASoC: dt-bindings: Add bindings for WCD934x DAIs ASoC: codecs: wcd934x: use wcd934x binding header Ernest Van Hoecke (5): of: Add of_property_read_u16_index ASoC: wm8904: Don't touch GPIO configs set to 0xFFFF ASoC: dt-bindings: wm8904: Add DMIC, GPIO, MIC and EQ support ASoC: wm8904: get platform data from DT ASoC: wm8904: add DMIC support Ethan Carter Edwards (5): ASoC: q6dsp: q6apm: replace kzalloc() with kcalloc() in q6apm_map_memory_regions() ASoC: Intel: avs: Replace devm_kzalloc() with devm_kcalloc() ASoC: Intel: avs: ssm4567: Replace devm_kzalloc() with devm_kcalloc() ASoC: Intel: avs: max98373: Replace devm_kzalloc() with devm_kcalloc() ASoC: Intel: avs: max98927: Replace devm_kzalloc() with devm_kcalloc() Geert Uytterhoeven (1): ASoC: soc-core: Stop using of_property_read_bool() for non-boolean properties Haoxiang Li (1): ASoC: codecs: wcd937x: fix a potential memory leak in wcd937x_soc_codec_probe() Hector Martin (4): ASoC: tas2764: Power up/down amp on mute ops ASoC: tas2764: Wait for ramp-down after shutdown ASoC: tas2764: Mark SW_RESET as volatile ASoC: tas2764: Add reg defaults for TAS2764_INT_CLK_CFG Ivaylo Dimitrov (3): ASoC: cpcap: Implement .set_bias_level dt-bindings: mfd: motorola-cpcap: Document audio-codec interrupts ASoC: cpcap: Implement jack detection James Calligeros (2): ASoC: dt-bindings: tas27xx: add compatible for SN012776 ASoC: dt-bindings: tas2770: add compatible for TAS5770L Jayesh Choudhary (1): ASoC: ti: j721e-evm: Reduce log level for EPROBE_DEFER Jiri Slaby (SUSE) (1): irqdomain: sound: Switch to irq_domain_create_linear() Johan Hovold (1): ASoC: codecs: wcd93xx-sdw: fix of_property_read_bool() warnings Krzysztof Kozlowski (2): ASoC: dt-bindings: everest,es8328: Mark ES8388 compatible with ES8328 ASoC: dt-bindings: everest,es8328: Require reg property Kuninori Morimoto (188): ASoC: soc-ops: remove soc-dpcm.h ASoC: add common snd_soc_ret() and use it ASoC: soc-utils: care -EOPNOTSUPP on snd_soc_ret() ASoC: soc-pcm: use snd_soc_ret() ASoC: simple-card: use snd_soc_ret() ASoC: simple-card-utils: use snd_soc_ret() ASoC: audio-graph-card: use snd_soc_ret() ASoC: audio-graph-card2: use snd_soc_ret() ASoC: soc-pcm: no need to check dpcm->fe on dpcm_be_connect() ASoC: soc-core: makes snd_soc_set_dmi_name() local ASoC: soc-ops: makes snd_soc_read_signed() void ASoC: soc-pcm: makes dpcm_dapm_stream_event() void ASoC: soc-dai: check return value at snd_soc_dai_set_tdm_slot() ASoC: soc-pcm: remove duplicate param from __soc_pcm_hw_params() ASoC: remove update from snd_soc_card ASoC: remove dpcm_process_paths() ASoC: soc-pcm: cleanup dpcm_dai_trigger_fe_be() ASoC: soc-pcm: cleanup dpcm_fe_dai_do_trigger() ASoC: Documentation: DPCM: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: kirkwood: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: ti: rx51: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: sh: migor: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: Documentation: Codec to Codec: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: ti: n810: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: ti: osk5912: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: ti: ams-delta: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: ti: j721e-evm: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: ti: davinci-evm: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: ti: omap-twl4030: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: ti: omap3pandora: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: amd: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: fsl: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: mxs: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: pxa: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: qcom: sc7180: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: qcom: sc7280: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: codec: cs4234: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: codec: cs4265: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: codec: cs4270: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: codec: cs4271: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: codec: cs4341: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: codec: cs530x: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: codec: cs35l32: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: codec: cs35l33: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: codec: cs35l34: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: codec: cs42l42: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: codec: cs42l51: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: codec: cs42l52: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: codec: cs42l56: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: codec: cs42l73: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: codec: cs42xx8: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: codec: cs43130: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: codec: cs53l30: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: codec: da7210: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: codec: da7218: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: codec: da7219: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: codec: da732x: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: codec: da9055: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: codec: nau8540: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: codec: nau8810: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: codec: nau8822: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: codec: nau8824: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: codec: nau8825: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: codec: rk817: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: codec: rt274: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: codec: rt286: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: codec: rt298: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: codec: rt1011: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: codec: rt1015: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: codec: rt1016: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: codec: rt1305: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: codec: rt1308: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: codec: rt5616: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: codec: rt5631: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: codec: rt5640: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: codec: rt5645: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: codec: rt5651: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: codec: rt5659: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: codec: rt5660: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: codec: rt5663: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: codec: rt5665: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: codec: rt5668: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: codec: rt5670: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: codec: rt5677: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: codec: rt5682: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: codec: rt5682s: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: codec: wm2200: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: codec: wm5100: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: codec: wm8350: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: codec: wm8400: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: codec: wm8510: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: codec: wm8523: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: codec: wm8524: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: codec: wm8580: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: codec: wm8711: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: codec: wm8728: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: codec: wm8737: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: codec: wm8741: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: codec: wm8750: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: codec: wm8753: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: codec: wm8770: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: codec: wm8776: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: codec: wm8804: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: codec: wm8900: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: codec: wm8903: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: codec: wm8904: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: codec: wm8940: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: codec: wm8955: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: codec: wm8960: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: codec: wm8961: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: codec: wm8962: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: codec: wm8971: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: codec: wm8974: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: codec: wm8978: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: codec: wm8983: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: codec: wm8985: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: codec: wm8988: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: codec: wm8990: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: codec: wm8991: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: codec: wm8993: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: codec: wm8994: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: codec: wm8995: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: codec: wm8996: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: codec: wm9081: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: codec: wm9713: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: codec: madera: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: codec: arizona: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: codec: twl4030: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: codec: src4xxx: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: codec: sgtl5000: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: codec: lochnagar-sc: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: ux500: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: meson: t9015: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: meson: meson-card-utils: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: tegra: tegra_asoc_machine: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: tegra: tegra_wm8903: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: intel: rt274: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: intel: rt286: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: intel: rt298: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: intel: rt5514: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: intel: da7219: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: intel: ssm4567: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: intel: nau8825: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: intel: max98927: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: intel: max98357a: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: samsung: snow: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: samsung: bells: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: samsung: odroid: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: samsung: arndale: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: samsung: lowland: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: samsung: speyside: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: samsung: tobermory: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: samsung: tm2_wm5110: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: samsung: littlemill: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: samsung: smdk_wm8994: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: samsung: aries_wm8994: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: samsung: midas_wm1811: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: samsung: smdk_wm8994pcm: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: mediatek: mt2701-wm8960: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: mediatek: mt2701-cs42448: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: mediatek: mt7986-wm8960: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: mediatek: mt7986-dai-etdm: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: mediatek: mt8173-rt5650: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: mediatek: mt8173-max98090: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: mediatek: mt8173-rt5650-rt5676: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: mediatek: mt8173-rt5650-rt5514: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: mediatek: mt8183-da7219-max98357: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: mediatek: mt8183-mt6358-ts3a227-max98357: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: mediatek: mt8186-mt6366: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: mediatek: mt8192-mt6359-rt1015-rt5682: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: mediatek: mt8195-mt6359: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: mediatek: mt8365-mt6357: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: mediatek: mt8365-dai-i2s: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: mediatek: mt8365-dai-pcm: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: rockchip: rk3288_hdmi_analog: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: rockchip: rk3399_gru_sound: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: rockchip: rockchip_max98090: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: rockchip: rockchip_rt5645: use inclusive language for SND_SOC_DAIFMT_CBx_CFx ASoC: soc-dai: remove SND_SOC_DAIFMT_CB{MS}_CF{MS} ASoC: audio-graph-card2-custom-sample1.dtsi: Separate Sample DT ASoC: audio-graph-card2-custom-sample2.dtsi: Separate Sample DT ASoC: audio-graph-card2-custom-sample.dtsi: remove original sample ASoC: doc: use SND_SOC_DAILINK_xxx() macro ASoC: soc-pcm: merge soc_pcm_hw_update_format/subformat() ASoC: soc-pcm: reuse dpcm_state_string() ASoC: soc-dai: add snd_soc_dai_mute_is_ctrled_at_trigger() ASoC: simple-card-utils: Don't use __free(device_node) at graph_util_parse_dai() ASoC: soc-pcm: tidyup function name to snd_soc_dpcm_be_can_xxx() Laurentiu Mihalcea (11): ASoC: SOF: imx: introduce more common structures and functions ASoC: SOF: imx8: use common imx chip interface ASoC: SOF: imx8: use IMX_SOF_* macros ASoC: SOF: imx8: drop unneeded/unused macros/header includes ASoC: SOF: imx: merge imx8 and imx8m drivers ASoC: SOF: imx: merge imx8 and imx8ulp drivers ASoC: SOF: imx: add driver for the imx95 chip ASoC: SOF: imx-common: set sdev->pdata->hw_pdata after common is alloc'd ASoC: dt-bindings: wlf,wm8960: add 'port' property ASoC: dt-bindings: support imx95's CM7 core ASoC: dt-bindings: audio-graph-card2: add widgets and hp-det-gpios support Linus Walleij (10): ASoC: samsung: bells: Drop unused include ASoC: samsung: littlemill: Drop unused include ASoC: samsung: lowland: Drop unused include ASoC: samsung: speyside: Convert to GPIO descriptor ASoC: samsung: tobermory: Drop unused include ASoC: samsung: tm2_wm5110: Drop unused include ASoC: samsung: speyside: Free gpiod table ASoC: pcm1681: Drop unused include ASoC: pcm3008: Convert to GPIO descriptors ASoC: pcm6240: Drop bogus code handling IRQ as GPIO Louis-Alexis Eyraud (1): ASoC: mediatek: mt6359: Fix DT parse error due to wrong child node name Mario Limonciello (12): x86/amd_node: Add SMN offsets to exclusive region access x86/amd_node: Add support for debugfs access to SMN registers x86/amd_node: Add a smn_read_register() helper ASoC: amd: acp: rembrandt: Use AMD_NODE ASoC: amd: acp: acp70: Use AMD_NODE ASoC: amd: acp: acp63: Use AMD_NODE ASoC: SOF: amd: Use AMD_NODE ASoC: amd: acp: Drop local symbols for smn read/write ASoC: SOF: amd: Drop host bridge ID from struct ASoC: SOF: amd: Add depends on CPU_SUP_AMD ASoC: SOF: amd: Move depends on AMD_NODE to consumers ASoC: dmic: Fix NULL pointer dereference Mark Brown (43): ASoC: Intel: avs: Add support for MalibouLake This is continued work on Samsung S9(SM-9600) ASoC: and adn use snd_soc_ret() ASoC: cpcap: Implement jack headset detection Add static channel mapping between soundwire master ASoC: SOF: Improve the spcm and ipc4 copier prints ASoC: Intel: soc-acpi-intel-ptl-match typo fixups ASoC: amd: Add support for ACP7.0 & ACP7.1 Refactor imx drivers and introduce support for Add SDCA DisCo parsing support arm64: dts: rockchip: Add SPDIF on RK3588 gpiolib: add gpiod_multi_set_value_cansleep ASoC: tas2764: Random patches from the Asahi Linux ASoC: random cleanup Adjust all AMD audio drivers to use AMD_NODE ASoC: imx-card: support playback or capture only ASoC: Intel: avs: Mute and multi-channel controls ASoC: dt-bindings: fsl: Reference common DAI Enable DMIC for Genio 700/510 EVK xlnx: dt-bindings: Convert to json-schema Add SDCA register map support Convert sound drivers to use devm_kmemdup_array() ASoC: audio-graph-card2-custom-sample: Separate Sample ASoC: use inclusive language for ASoC: codecs: Update device_id tables for Realtek ASoC: samsung: GPIO descriptor conversion Tidy up ASoC VALUE control macros ASoC: fsl_audmix: support audio graph card for audmix ASoC: Intel: boards: updates for 6.15 ASoC: Merge up fixes ASoC: dmic: add regulator support ASoC: SOF: Intel: Add support for ACE3+ mic privacy ASoC: Merge up fixes ASoC: sun4i-codec: add headphone dectection for Allow retrieving accessory detection reference on ASoC: Convert PCM codecs to GPIO descriptors Some minor SDCA preparation ASoC: Convert to modern PM macros ASoC: Intel: avs: Replace devm_kzalloc() with add sof support on imx95 ASoC: codecs: Add aw88166 amplifier driver Tidy up ASoC control get and put handlers ASoC: wm8904: Add DMIC and DRC support Martin Povišer (5): ASoC: ops: Enforce platform maximum on initial value ASoC: tas2764: Extend driver to SN012776 ASoC: tas2764: Add control concerning overcurrent events ASoC: tas2770: Factor out set_ivsense_slots ASoC: tas2770: Fix and redo I/V sense TDM slot setting logic Masahiro Yamada (3): ASoC: dapm: unexport snd_soc_dapm_init() ASoC: dapm: unexport snd_soc_dapm_update_dai() ASoC: dapm: unexport dapm_mark_endpoints_dirty() Mohammad Rafi Shaik (4): ASoC: dt-bindings: wcd937x-sdw: Add static channel mapping support ASoC: codecs: wcd937x: Add static channel mapping support in wcd937x-sdw soundwire: qcom: Add set_channel_map api support ASoC: qcom: sdw: Add get and set channel maps support from codec to cpu dais Naveen Manohar (1): ASoC: Intel: soc-acpi-intel-ptl-match: Add cs42l43 support Nícolas F. R. A. Prado (9): ASoC: mediatek: mt8188: Add audsys hires clocks ASoC: mediatek: mt8188: Add reference for dmic clocks ASoC: mediatek: mt8188: Treat DMIC_GAINx_CUR as non-volatile ASoC: mediatek: mt8188-mt6359: Add DMIC support ASoC: dt-bindings: mediatek,mt8188-mt6359: Add DMIC backend to dai-link ASoC: dt-bindings: mediatek,mt8188-mt6359: Add mediatek,accdet ASoC: mediatek: common: Handle mediatek,accdet property ASoC: mediatek: mt6359: Add stub for mt6359_accdet_enable_jack_detect ASoC: mediatek: mt8188-mt6359: Add accdet headset jack detect support Olivier Moysan (2): ASoC: dt-bindings: add regulator support to dmic codec ASoC: dmic: add regulator support Peter Ujfalusi (18): ASoC: SOF: Relocate and rework functionality for PCM stream freeing ASoC: SOF: pcm: Move period/buffer configuration print after platform open ASoC: SOF: pcm: Add snd_sof_pcm specific wrappers for dev_dbg() and dev_err() ASoC: SOF: ipc4-topology: Improve the information in prepare_copier prints ASoC: SOF: ipc4-pcm: Move out be_rate initialization from for loop in fixup ASoC: SOF: ipc4: Add support for split firmware releases ASoC: Intel: soc-acpi-intel-ptl-match: add rt712_vb + rt1320 support ASoC: Intel: soc-acpi-intel-ptl-match: add rt713_vb_l2_rt1320_l13 ASoC: Intel: soc-acpi-intel-ptl-match: add rt712_vb_l3_rt1320_l2 support ASoC: Intel: soc-acpi-intel-ptl-match: add rt713_vb_l3_rt1320_l12 support ASoC: SOF: Intel: mtl: Split up dsp_ops setup code ASoC: SOF: Intel: lnl/ptl: Only set dsp_ops which differs from MTL ASoC: SOF: Intel: mtl: Stop exporting dsp_ops callback functions ASoC: SOF: Intel: Create ptl.c as placeholder for Panther Lake features ASoC: SOF: ipc4: Add support for Intel HW managed mic privacy messaging ASoC: SOF: Intel: hda-mlink: Add support for mic privacy in VS SHIM registers ASoC: SOF: hda/shim: Add callbacks to handle mic privacy change for sdw ASoC: SOF: Intel: ptl: Add support for mic privacy Pierre-Louis Bossart (2): ASoC: SDCA: Add code to parse Function information ASoC: SDCA: Parse initialization write table Raag Jadav (8): err.h: move IOMEM_ERR_PTR() to err.h devres: Introduce devm_kmemdup_array() ASoC: Intel: avs: use devm_kmemdup_array() ASoC: hdac_hdmi: use devm_kmemdup_array() ASoC: tlv320dac33: use devm_kmemdup_array() ASoC: uda1380: use devm_kmemdup_array() ASoC: meson: axg-tdm-interface: use devm_kmemdup_array() ASoC: uniphier: use devm_kmemdup_array() Richard Fitzgerald (1): firmware: cs_dsp: test_bin_error: Use same test cases for adsp2 and Halo Core Ritu Chaudhary (1): ASoC: tegra: Use non-atomic timeout for ADX status register Ryan Walklin (3): ASoC: sun4i-codec: correct dapm widgets and controls for h616 ASoC: sun4i-codec: support hp-det-gpios property ASoC: sun4i-codec: add h616 card long_name Sheetal (2): ASoC: tegra: Add interconnect support ASoC: tegra: Remove the isomgr_bw APIs export Shengjiu Wang (11): ASoC: dmic: Add DSD big endian format support ASoC: fsl_micfil: Add decimation filter bypass mode support ASoC: dt-bindings: imx-card: Add playback-only and capture-only property ASoC: imx-card: Add playback_only or capture_only support ASoC: dt-bindings: fsl,easrc: Reference common DAI properties ASoC: dt-bindings: fsl,imx-asrc: Reference common DAI properties ASoC: dt-bindings: fsl,sai: Document audio graph port ASoC: dt-bindings: fsl,audmix: Document audio graph port ASoC: dt-bindings: fsl,audmix: make 'dais' property to be optional ASoC: fsl_audmix: register card device depends on 'dais' property ASoC: dt-bindings: fsl,imx-asrc: Document audio graph port Shuming Fan (1): ASoC: rt712-sdca: Add FU05 playback switch control Sudeep Holla (1): ASoC: soc-utils: Transition to the faux device interface Syed Saba kareem (1): ASoC: amd: yc: update quirk data for new Lenovo model Takashi Iwai (88): ASoC: au1x: Convert to DEFINE_SIMPLE_DEV_PM_OPS() ASoC: ak4375: Convert to RUNTIME_PM_OPS() & co ASoC: ak4458: Convert to RUNTIME_PM_OPS() & co ASoC: ak5558: Convert to RUNTIME_PM_OPS() & co ASoC: cs35l32: Convert to RUNTIME_PM_OPS() ASoC: cs35l33: Convert to RUNTIME_PM_OPS() ASoC: cs35l34: Convert to RUNTIME_PM_OPS() ASoC: cs4234: Convert to RUNTIME_PM_OPS() ASoC: cs42l42: Convert to RUNTIME_PM_OPS() & co ASoC: cs42l51: Convert to SYSTEM_SLEEP_PM_OPS() ASoC: cs42l83: Convert to SYSTEM_SLEEP_PM_OPS() ASoC: cs42xx8: Convert to EXPORT_GPL_DEV_PM_OPS() ASoC: cs43130: Convert to RUNTIME_PM_OPS() ASoC: cs4349: Convert to RUNTIME_PM_OPS() ASoC: cs53l30: Convert to RUNTIME_PM_OPS() ASoC: cx2072x: Convert to RUNTIME_PM_OPS() & co ASoC: da7213: Convert to RUNTIME_PM_OPS() & co ASoC: hdac_hdmi: Convert to RUNTIME_PM_OPS() & co ASoC: lpass: Convert to RUNTIME_PM_OPS() ASoC: max98090: Convert to RUNTIME_PM_OPS() & co ASoC: max98373: Convert to RUNTIME_PM_OPS() & co ASoC: max98390: Convert to SYSTEM_SLEEP_PM_OPS() ASoC: max98396: Convert to SYSTEM_SLEEP_PM_OPS() ASoC: max98520: Convert to SYSTEM_SLEEP_PM_OPS() ASoC: max9860: Convert to RUNTIME_PM_OPS() ASoC: max98927: Convert to SYSTEM_SLEEP_PM_OPS() ASoC: mt6660: Convert to RUNTIME_PM_OPS() ASoC: pcm512x: Convert to EXPORT_GPL_DEV_PM_OPS() ASoC: rt1017-sdca-sdw: Convert to RUNTIME_PM_OPS() & co ASoC: rt1308-sdw: Convert to RUNTIME_PM_OPS() & co ASoC: rt1316-sdw: Convert to RUNTIME_PM_OPS() & co ASoC: rt1318-sdw: Convert to RUNTIME_PM_OPS() & co ASoC: rt1320-sdw: Convert to RUNTIME_PM_OPS() & co ASoC: rt5514: Convert to SYSTEM_SLEEP_PM_OPS() ASoC: rt5645: Convert to SYSTEM_SLEEP_PM_OPS() ASoC: rt5682-sdw: Convert to RUNTIME_PM_OPS() & co ASoC: rt700-sdw: Convert to RUNTIME_PM_OPS() & co ASoC: rt711: Convert to RUNTIME_PM_OPS() & co ASoC: rt712: Convert to RUNTIME_PM_OPS() & co ASoC: rt715: Convert to RUNTIME_PM_OPS() & co ASoC: rt721: Convert to RUNTIME_PM_OPS() & co ASoC: rt722: Convert to RUNTIME_PM_OPS() & co ASoC: rt9120: Convert to RUNTIME_PM_OPS() ASoC: rtq9128: Convert to RUNTIME_PM_OPS() ASoC: tas2552: Convert to RUNTIME_PM_OPS() ASoC: ts3a227e: Convert to SYSTEM_SLEEP_PM_OPS() ASoC: wcd937x: Convert to RUNTIME_PM_OPS() ASoC: wcd938x: Convert to RUNTIME_PM_OPS() ASoC: wcd939x: Convert to RUNTIME_PM_OPS() ASoC: wm2200: Convert to RUNTIME_PM_OPS() ASoC: wm5100: Convert to RUNTIME_PM_OPS() ASoC: wm8804: Convert to EXPORT_GPL_DEV_PM_OPS() ASoC: wm8962: Convert to RUNTIME_PM_OPS() & co ASoC: wm8994: Convert to SYSTEM_SLEEP_PM_OPS() ASoC: wsa881x: Convert to RUNTIME_PM_OPS() ASoC: wsa883x: Convert to RUNTIME_PM_OPS() ASoC: wsa884x: Convert to RUNTIME_PM_OPS() ASoC: dwc: Convert to RUNTIME_PM_OPS() ASoC: fsl: Convert to RUNTIME_PM_OPS() and co ASoC: img: Convert to RUNTIME_PM_OPS() and co ASoC: intel: avs: Convert to RUNTIME_PM_OPS() ASoC: intel: catpt: Convert to RUNTIME_PM_OPS() & co ASoC: mediatek: mt2701: Convert to RUNTIME_PM_OPS() ASoC: mediatek: mt6797: Convert to RUNTIME_PM_OPS() ASoC: mediatek: mt7986: Convert to RUNTIME_PM_OPS() ASoC: mediatek: mt8173: Convert to RUNTIME_PM_OPS() ASoC: mediatek: mt8183: Convert to RUNTIME_PM_OPS() ASoC: mediatek: mt8186: Convert to RUNTIME_PM_OPS() ASoC: mediatek: mt8188: Convert to RUNTIME_PM_OPS() ASoC: mediatek: mt8192: Convert to RUNTIME_PM_OPS() ASoC: mediatek: mt8195: Convert to RUNTIME_PM_OPS() ASoC: mediatek: mt8365: Convert to RUNTIME_PM_OPS() & co ASoC: qcom: Convert to SYSTEM_SLEEP_PM_OPS() ASoC: rcar: Convert to SYSTEM_SLEEP_PM_OPS() ASoC: rockchip: Convert to RUNTIME_PM_OPS() & co ASoC: samsung: Convert to RUNTIME_PM_OPS() & co ASoC: SOF: acpi: Convert to EXPORT_NS_DEV_PM_OPS() ASoC: SOF: pci: Convert to EXPORT_NS_DEV_PM_OPS() ASoC: SOF: of: Convert to EXPORT_DEV_PM_OPS() ASoC: stm: Convert to SYSTEM_SLEEP_PM_OPS() ASoC: sunxi: Convert to RUNTIME_PM_OPS() ASoC: tegra186: Convert to RUNTIME_PM_OPS() & co ASoC: tegra210: Convert to RUNTIME_PM_OPS() & co ASoC: tegra20: Convert to RUNTIME_PM_OPS() & co ASoC: tegra30: Convert to RUNTIME_PM_OPS() & co ASoC: xtensa: Convert to RUNTIME_PM_OPS() ASoC: amd: Convert to RUNTIME_PM_OPS() & co ASoC: pcm3168a: Convert to EXPORT_GPL_DEV_PM_OPS() Thomas Weißschuh (1): firmware: cs_dsp: test_control_parse: null-terminate test strings Thorsten Blum (14): x86/mtrr: Use str_enabled_disabled() helper in print_mtrr_state() ASoC: amd: acp: Use str_low_high() helper function ASoC: SOF: mediatek: Use str_on_off() helper function ASoC: tscs454: Use str_enable_disable() in pll_power_event() ASoC: Intel: avs: Use str_on_off() in avs_dsp_core_power() ASoC: SOF: ipc3: Use str_enabled_disabled() helper function ASoC: wm_hubs: Use str_enable_disable() in wm_hubs_update_class_w() ASoC: SOF: Intel: Use str_enable_disable() helper ASoC: soc-core: Use str_yes_no() in snd_soc_close_delayed_work() ASoC: fsl: fsl_qmc_audio: Remove unnecessary bool conversions ASoC: atmel: atmel-classd: Use str_enabled_disabled() helper ASoC: cros_ec_codec: Use str_enable_disable() helper in wov_enable_put() ASoC: amd: acp: acp70: Remove unnecessary if-check ASoC: SOF: Intel: hda-dai: Remove unnecessary bool conversion Uwe Kleine-König (1): ASoC: SOF: Intel: Don't import non-existing module namespace Venkata Prasad Potturu (17): ASoC: amd: acp: Remove redundant acp70 chip->name ASoC: amd: acp: Implement acp_common_hw_ops support for acp platforms ASoC: amd: acp: Refactor dmic-codec platform device creation ASoC: amd: acp: Refactor acp platform device creation ASoC: amd: acp: Refactor acp machine select ASoC: amd: acp: Add new interrupt handle callbacks in acp_common_hw_ops ASoC: amd: acp: Remove redundant acp_dev_data structure ASoC: amd: acp: Move spin_lock and list initialization to acp-pci driver ASoC: amd: acp: Remove white line ASoC: amd: acp: Refactor acp70 platform resource structure ASoC: amd: acp: Refactor acp63 platform resource structure ASoC: amd: acp: Refactor rembrant platform resource structure ASoC: amd: acp: Refactor renoir platform resource structure ASoC: amd: acp: Fix for enabling DMIC on acp platforms via _DSD entry ASoC: amd: acp: Fix snd_soc_acpi_mach id's duplicate symbol error ASoC: amd: acp: Fix acp_resource duplicate symbol error ASoC: amd: acp: Fix acp_common_hw_ops declaration error Vijendar Mukunda (27): ASoC: amd: ps: rename structure names, variable and other macros ASoC: amd: ps: use macro for ACP6.3 pci revision id ASoC: amd: ps: add acp pci driver hw_ops for acp6.3 platform ASoC: amd: ps: add callback functions for acp pci driver pm ops ASoC: amd: ps: add callback to read acp pin configuration ASoC: amd: ps: add soundwire dma irq thread callback ASoC: amd: ps: refactor soundwire dma interrupt handling ASoC: amd: ps: store acp revision id in SoundWire dma driver private data ASoC: amd: ps: refactor soundwire dma driver code ASoC: amd: ps: refactor soundwire dma interrupts enable/disable sequence ASoC: amd: ps: rename acp_restore_sdw_dma_config() function ASoC: amd: ps: add pci driver hw_ops for ACP7.0 & ACP7.1 variants ASoC: amd: ps: add pm ops related hw_ops for ACP7.0 & ACP7.1 platforms ASoC: amd: ps: add ACP7.0 & ACP7.1 specific soundwire dma driver changes ASoC: amd: ps: implement function to restore dma config for ACP7.0 platform ASoC: amd: ps: add soundwire dma interrupts handling for ACP7.0 platform ASoC: amd: ps: add soundwire wake interrupt handling ASoC: amd: ps: update module description ASoC: amd: ps: update file description and copyright year ASoC: amd: update Pink Sardine platform Kconfig description ASoC: amd: acp: add machine driver changes for ACP7.0 and ACP7.1 platforms ASoC: amd: acp: add RT711, RT714 & RT1316 support for ACP7.0 platform ASoC: amd: acp: amd-acp70-acpi-match: Add rt722 support ASoC: amd: acp: amd-acp70-acpi-match: Add RT1320 & RT722 combination soundwire machine ASoC: amd: amd_sdw: Add quirks for Dell SKU's ASoC: amd: ps: use switch statements for acp pci revision id check ASoC: amd: ps: fix inconsistent indenting warning in check_and_handle_sdw_dma_irq() Vincenzo Frascino (4): ASoC: dt-bindings: xlnx,i2s: Convert to json-schema ASoC: dt-bindings: xlnx,audio-formatter: Convert to json-schema ASoC: dt-bindings: xlnx,spdif: Convert to json-schema MAINTAINERS: Add Vincenzo Frascino as Xilinx Sound Driver Maintainer Vitaliy Shevtsov (1): ASoC: cs35l41: check the return value from spi_setup() Vitaly Kuznetsov (1): x86/entry: Add __init to ia32_emulation_override_cmdline() Weidong Wang (2): ASoC: dt-bindings: Add schema for "awinic,aw88166" ASoC: codecs: Add aw88166 amplifier driver Yazen Ghannam (1): x86/amd_node, platform/x86/amd/hsmp: Have HSMP use SMN through AMD_NODE Zhang Heng (1): ASoC: SOF: topology: Use krealloc_array() to replace krealloc() parkeryang (1): ASoC: mediatek: mt8188: Add support for DMIC .../devicetree/bindings/mfd/motorola-cpcap.txt | 6 + .../devicetree/bindings/misc/atmel-ssc.txt | 50 - .../bindings/sound/allwinner,sun4i-a10-codec.yaml | 6 + .../devicetree/bindings/sound/atmel,at91-ssc.yaml | 104 ++ .../bindings/sound/atmel,at91sam9g20ek-wm8731.yaml | 72 + .../sound/atmel-at91sam9g20ek-wm8731-audio.txt | 26 - .../bindings/sound/audio-graph-card2.yaml | 4 + .../devicetree/bindings/sound/awinic,aw88395.yaml | 1 + .../devicetree/bindings/sound/dmic-codec.yaml | 3 + .../devicetree/bindings/sound/everest,es8328.yaml | 11 +- .../devicetree/bindings/sound/fsl,audmix.yaml | 61 +- .../devicetree/bindings/sound/fsl,easrc.yaml | 5 +- .../devicetree/bindings/sound/fsl,imx-asrc.yaml | 15 +- .../bindings/sound/fsl,imx95-cm7-sof.yaml | 64 + .../devicetree/bindings/sound/fsl,sai.yaml | 51 + .../devicetree/bindings/sound/fsl,sof-cpu.yaml | 27 + .../devicetree/bindings/sound/imx-audio-card.yaml | 14 + .../bindings/sound/mediatek,mt8188-mt6359.yaml | 9 + .../bindings/sound/qcom,wcd937x-sdw.yaml | 36 + .../devicetree/bindings/sound/rockchip-spdif.yaml | 4 + .../devicetree/bindings/sound/ti,tas2770.yaml | 1 + .../devicetree/bindings/sound/ti,tas27xx.yaml | 1 + .../devicetree/bindings/sound/wlf,wm8904.yaml | 129 ++ .../devicetree/bindings/sound/wlf,wm8960.yaml | 4 + .../bindings/sound/xlnx,audio-formatter.txt | 29 - .../bindings/sound/xlnx,audio-formatter.yaml | 72 + .../devicetree/bindings/sound/xlnx,i2s.txt | 28 - .../devicetree/bindings/sound/xlnx,i2s.yaml | 65 + .../devicetree/bindings/sound/xlnx,spdif.txt | 28 - .../devicetree/bindings/sound/xlnx,spdif.yaml | 77 + Documentation/sound/soc/codec-to-codec.rst | 4 +- Documentation/sound/soc/dpcm.rst | 21 +- MAINTAINERS | 11 +- arch/x86/entry/common.c | 2 +- arch/x86/include/asm/amd_nb.h | 1 - arch/x86/include/asm/amd_node.h | 24 + arch/x86/kernel/amd_nb.c | 1 - arch/x86/kernel/amd_node.c | 149 ++ arch/x86/kernel/cpu/mtrr/generic.c | 13 +- drivers/base/regmap/regcache.c | 31 + .../firmware/cirrus/test/cs_dsp_test_bin_error.c | 13 +- .../cirrus/test/cs_dsp_test_control_parse.c | 51 +- drivers/iio/imu/st_lsm9ds0/st_lsm9ds0_i2c.c | 2 +- drivers/iio/imu/st_lsm9ds0/st_lsm9ds0_spi.c | 2 +- drivers/of/property.c | 33 + drivers/platform/x86/amd/hsmp/Kconfig | 2 +- drivers/platform/x86/amd/hsmp/acpi.c | 7 +- drivers/platform/x86/amd/hsmp/hsmp.c | 1 - drivers/platform/x86/amd/hsmp/hsmp.h | 3 - drivers/platform/x86/amd/hsmp/plat.c | 36 +- drivers/soundwire/qcom.c | 26 + include/dt-bindings/sound/qcom,wcd934x.h | 16 + include/linux/device.h | 119 +- include/linux/device/devres.h | 129 ++ include/linux/err.h | 3 + include/linux/gpio/consumer.h | 11 + include/linux/io.h | 2 - include/linux/of.h | 9 + include/linux/regmap.h | 7 + include/sound/hda-mlink.h | 25 + include/sound/sdca.h | 22 +- include/sound/sdca_function.h | 1179 +++++++++++- include/sound/sdca_regmap.h | 31 + include/sound/simple_card_utils.h | 8 +- include/sound/soc-dai.h | 8 +- include/sound/soc-dapm.h | 61 +- include/sound/soc-dpcm.h | 8 +- include/sound/soc.h | 133 +- include/sound/sof/ipc4/header.h | 13 + include/sound/tas2781.h | 22 +- include/sound/wm8904.h | 3 + include/uapi/sound/intel/avs/tokens.h | 11 + sound/hda/intel-dsp-config.c | 4 + sound/pci/hda/tas2781-spi.h | 1 - sound/pci/hda/tas2781_hda_i2c.c | 10 +- sound/pci/hda/tas2781_hda_spi.c | 5 +- sound/soc/Kconfig | 7 + sound/soc/Makefile | 4 + sound/soc/amd/Kconfig | 10 +- sound/soc/amd/acp/Kconfig | 12 + sound/soc/amd/acp/Makefile | 4 +- sound/soc/amd/acp/acp-i2s.c | 185 +- sound/soc/amd/acp/acp-legacy-common.c | 308 +++- sound/soc/amd/acp/acp-mach-common.c | 2 +- sound/soc/amd/acp/acp-pci.c | 220 ++- sound/soc/amd/acp/acp-pdm.c | 53 +- sound/soc/amd/acp/acp-platform.c | 145 +- sound/soc/amd/acp/acp-rembrandt.c | 149 +- sound/soc/amd/acp/acp-renoir.c | 116 +- sound/soc/amd/acp/acp-sdw-legacy-mach.c | 45 + sound/soc/amd/acp/acp-sdw-mach-common.c | 34 + sound/soc/amd/acp/acp3x-es83xx/acp3x-es83xx.c | 5 +- sound/soc/amd/acp/acp63.c | 163 +- sound/soc/amd/acp/acp70.c | 132 +- sound/soc/amd/acp/amd-acp70-acpi-match.c | 160 ++ sound/soc/amd/acp/amd-acpi-mach.c | 93 + sound/soc/amd/acp/amd.h | 168 +- sound/soc/amd/acp/chip_offset_byte.h | 12 +- sound/soc/amd/acp/soc_amd_sdw_common.h | 12 + sound/soc/amd/mach-config.h | 1 + sound/soc/amd/ps/Makefile | 2 +- sound/soc/amd/ps/acp63.h | 246 ++- sound/soc/amd/ps/pci-ps.c | 434 ++--- sound/soc/amd/ps/ps-common.c | 475 +++++ sound/soc/amd/ps/ps-pdm-dma.c | 18 +- sound/soc/amd/ps/ps-sdw-dma.c | 381 +++- sound/soc/amd/rpl/rpl-pci-acp6x.c | 10 +- sound/soc/amd/vangogh/acp5x-pcm-dma.c | 11 +- sound/soc/amd/yc/acp6x-mach.c | 7 + sound/soc/amd/yc/acp6x-pdm-dma.c | 12 +- sound/soc/amd/yc/pci-acp6x.c | 10 +- sound/soc/atmel/atmel-classd.c | 3 +- sound/soc/atmel/tse850-pcm5142.c | 11 +- sound/soc/au1x/i2sc.c | 17 +- sound/soc/au1x/psc-ac97.c | 17 +- sound/soc/au1x/psc-i2s.c | 17 +- sound/soc/codecs/Kconfig | 13 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/adau1701.c | 4 +- sound/soc/codecs/adau17x1.c | 2 +- sound/soc/codecs/ak4375.c | 11 +- sound/soc/codecs/ak4458.c | 13 +- sound/soc/codecs/ak5558.c | 11 +- sound/soc/codecs/arizona.c | 12 +- sound/soc/codecs/aw88166.c | 1933 ++++++++++++++++++++ sound/soc/codecs/aw88166.h | 534 ++++++ sound/soc/codecs/cpcap.c | 200 +- sound/soc/codecs/cros_ec_codec.c | 3 +- sound/soc/codecs/cs35l32.c | 11 +- sound/soc/codecs/cs35l33.c | 14 +- sound/soc/codecs/cs35l34.c | 14 +- sound/soc/codecs/cs35l41-spi.c | 5 +- sound/soc/codecs/cs35l56.c | 1 - sound/soc/codecs/cs4234.c | 12 +- sound/soc/codecs/cs4265.c | 4 +- sound/soc/codecs/cs4270.c | 4 +- sound/soc/codecs/cs4271.c | 4 +- sound/soc/codecs/cs42l42-i2c.c | 6 +- sound/soc/codecs/cs42l42-sdw.c | 16 +- sound/soc/codecs/cs42l42.c | 4 +- sound/soc/codecs/cs42l51-i2c.c | 2 +- sound/soc/codecs/cs42l51.c | 8 +- sound/soc/codecs/cs42l52.c | 4 +- sound/soc/codecs/cs42l56.c | 4 +- sound/soc/codecs/cs42l73.c | 4 +- sound/soc/codecs/cs42l83-i2c.c | 6 +- sound/soc/codecs/cs42xx8-i2c.c | 2 +- sound/soc/codecs/cs42xx8.c | 14 +- sound/soc/codecs/cs43130.c | 33 +- sound/soc/codecs/cs4341.c | 2 +- sound/soc/codecs/cs4349.c | 7 +- sound/soc/codecs/cs530x.c | 4 +- sound/soc/codecs/cs53l30.c | 11 +- sound/soc/codecs/cx2072x.c | 12 +- sound/soc/codecs/da7210.c | 4 +- sound/soc/codecs/da7213.c | 10 +- sound/soc/codecs/da7218.c | 4 +- sound/soc/codecs/da7219.c | 4 +- sound/soc/codecs/da732x.c | 4 +- sound/soc/codecs/da9055.c | 4 +- sound/soc/codecs/dmic.c | 27 +- sound/soc/codecs/hdac_hdmi.c | 18 +- sound/soc/codecs/jz4760.c | 32 +- sound/soc/codecs/jz4770.c | 40 +- sound/soc/codecs/lochnagar-sc.c | 4 +- sound/soc/codecs/lpass-rx-macro.c | 8 +- sound/soc/codecs/lpass-tx-macro.c | 8 +- sound/soc/codecs/lpass-va-macro.c | 8 +- sound/soc/codecs/lpass-wsa-macro.c | 8 +- sound/soc/codecs/madera.c | 12 +- sound/soc/codecs/max98090.c | 11 +- sound/soc/codecs/max98373-i2c.c | 6 +- sound/soc/codecs/max98373-sdw.c | 10 +- sound/soc/codecs/max98390.c | 6 +- sound/soc/codecs/max98396.c | 6 +- sound/soc/codecs/max98520.c | 8 +- sound/soc/codecs/max9860.c | 6 +- sound/soc/codecs/max98927.c | 6 +- sound/soc/codecs/mt6358.c | 104 -- sound/soc/codecs/mt6358.h | 4 - sound/soc/codecs/mt6359-accdet.h | 9 + sound/soc/codecs/mt6359.c | 9 +- sound/soc/codecs/mt6660.c | 9 +- sound/soc/codecs/nau8540.c | 4 +- sound/soc/codecs/nau8810.c | 4 +- sound/soc/codecs/nau8822.c | 4 +- sound/soc/codecs/nau8824.c | 4 +- sound/soc/codecs/nau8825.c | 4 +- sound/soc/codecs/ntp8918.c | 1 - sound/soc/codecs/pcm1681.c | 1 - sound/soc/codecs/pcm3008.c | 61 +- sound/soc/codecs/pcm3008.h | 19 - sound/soc/codecs/pcm3168a-i2c.c | 11 +- sound/soc/codecs/pcm3168a-spi.c | 2 +- sound/soc/codecs/pcm3168a.c | 18 +- sound/soc/codecs/pcm512x-i2c.c | 2 +- sound/soc/codecs/pcm512x-spi.c | 2 +- sound/soc/codecs/pcm512x.c | 7 +- sound/soc/codecs/pcm6240.c | 28 +- sound/soc/codecs/pcm6240.h | 7 +- sound/soc/codecs/rk817_codec.c | 4 +- sound/soc/codecs/rt1011.c | 8 +- sound/soc/codecs/rt1015.c | 10 +- sound/soc/codecs/rt1016.c | 10 +- sound/soc/codecs/rt1017-sdca-sdw.c | 10 +- sound/soc/codecs/rt1019.c | 6 +- sound/soc/codecs/rt1305.c | 12 +- sound/soc/codecs/rt1308-sdw.c | 10 +- sound/soc/codecs/rt1308.c | 8 +- sound/soc/codecs/rt1316-sdw.c | 10 +- sound/soc/codecs/rt1318-sdw.c | 10 +- sound/soc/codecs/rt1318.c | 6 +- sound/soc/codecs/rt1320-sdw.c | 10 +- sound/soc/codecs/rt274.c | 12 +- sound/soc/codecs/rt286.c | 9 +- sound/soc/codecs/rt298.c | 9 +- sound/soc/codecs/rt5514-spi.c | 8 +- sound/soc/codecs/rt5514.c | 12 +- sound/soc/codecs/rt5616.c | 4 +- sound/soc/codecs/rt5631.c | 4 +- sound/soc/codecs/rt5640.c | 18 +- sound/soc/codecs/rt5645.c | 24 +- sound/soc/codecs/rt5651.c | 12 +- sound/soc/codecs/rt5659.c | 12 +- sound/soc/codecs/rt5660.c | 12 +- sound/soc/codecs/rt5663.c | 10 +- sound/soc/codecs/rt5665.c | 12 +- sound/soc/codecs/rt5668.c | 10 +- sound/soc/codecs/rt5670.c | 12 +- sound/soc/codecs/rt5677-spi.c | 3 +- sound/soc/codecs/rt5677.c | 5 +- sound/soc/codecs/rt5682-i2c.c | 6 +- sound/soc/codecs/rt5682-sdw.c | 12 +- sound/soc/codecs/rt5682.c | 4 +- sound/soc/codecs/rt5682s.c | 4 +- sound/soc/codecs/rt700-sdw.c | 12 +- sound/soc/codecs/rt711-sdca-sdw.c | 12 +- sound/soc/codecs/rt711-sdw.c | 12 +- sound/soc/codecs/rt712-sdca-dmic.c | 12 +- sound/soc/codecs/rt712-sdca-sdw.c | 12 +- sound/soc/codecs/rt712-sdca.c | 80 +- sound/soc/codecs/rt712-sdca.h | 3 + sound/soc/codecs/rt715-sdca-sdw.c | 10 +- sound/soc/codecs/rt715-sdca.c | 8 - sound/soc/codecs/rt715-sdw.c | 10 +- sound/soc/codecs/rt715.c | 8 - sound/soc/codecs/rt721-sdca-sdw.c | 12 +- sound/soc/codecs/rt722-sdca-sdw.c | 142 +- sound/soc/codecs/rt722-sdca-sdw.h | 99 +- sound/soc/codecs/rt722-sdca.c | 135 +- sound/soc/codecs/rt722-sdca.h | 4 +- sound/soc/codecs/rt9120.c | 8 +- sound/soc/codecs/rtq9128.c | 8 +- sound/soc/codecs/sgtl5000.c | 4 +- sound/soc/codecs/sma1307.c | 39 +- sound/soc/codecs/src4xxx.c | 4 +- sound/soc/codecs/tas2552.c | 9 +- sound/soc/codecs/tas2562.c | 14 +- sound/soc/codecs/tas2764.c | 113 +- sound/soc/codecs/tas2764.h | 7 + sound/soc/codecs/tas2770.c | 57 +- sound/soc/codecs/tas2781-fmwlib.c | 7 +- sound/soc/codecs/tlv320adc3xxx.c | 3 +- sound/soc/codecs/tlv320dac33.c | 6 +- sound/soc/codecs/ts3a227e.c | 6 +- sound/soc/codecs/tscs454.c | 5 +- sound/soc/codecs/twl4030.c | 2 +- sound/soc/codecs/uda1380.c | 6 +- sound/soc/codecs/wcd934x.c | 16 +- sound/soc/codecs/wcd937x-sdw.c | 49 +- sound/soc/codecs/wcd937x.c | 57 +- sound/soc/codecs/wcd937x.h | 7 +- sound/soc/codecs/wcd938x-sdw.c | 10 +- sound/soc/codecs/wcd938x.c | 11 +- sound/soc/codecs/wcd939x-sdw.c | 10 +- sound/soc/codecs/wcd939x.c | 2 +- sound/soc/codecs/wm2200.c | 15 +- sound/soc/codecs/wm5100.c | 15 +- sound/soc/codecs/wm5110.c | 4 +- sound/soc/codecs/wm8350.c | 4 +- sound/soc/codecs/wm8400.c | 4 +- sound/soc/codecs/wm8510.c | 4 +- sound/soc/codecs/wm8523.c | 4 +- sound/soc/codecs/wm8524.c | 2 +- sound/soc/codecs/wm8580.c | 4 +- sound/soc/codecs/wm8711.c | 4 +- sound/soc/codecs/wm8728.c | 2 +- sound/soc/codecs/wm8737.c | 4 +- sound/soc/codecs/wm8741.c | 2 +- sound/soc/codecs/wm8750.c | 4 +- sound/soc/codecs/wm8753.c | 12 +- sound/soc/codecs/wm8770.c | 4 +- sound/soc/codecs/wm8776.c | 4 +- sound/soc/codecs/wm8804-i2c.c | 2 +- sound/soc/codecs/wm8804-spi.c | 2 +- sound/soc/codecs/wm8804.c | 11 +- sound/soc/codecs/wm8900.c | 8 +- sound/soc/codecs/wm8903.c | 8 +- sound/soc/codecs/wm8904.c | 325 +++- sound/soc/codecs/wm8940.c | 4 +- sound/soc/codecs/wm8955.c | 4 +- sound/soc/codecs/wm8960.c | 4 +- sound/soc/codecs/wm8961.c | 4 +- sound/soc/codecs/wm8962.c | 12 +- sound/soc/codecs/wm8971.c | 4 +- sound/soc/codecs/wm8974.c | 4 +- sound/soc/codecs/wm8978.c | 4 +- sound/soc/codecs/wm8983.c | 4 +- sound/soc/codecs/wm8985.c | 4 +- sound/soc/codecs/wm8988.c | 4 +- sound/soc/codecs/wm8990.c | 4 +- sound/soc/codecs/wm8991.c | 4 +- sound/soc/codecs/wm8993.c | 8 +- sound/soc/codecs/wm8994.c | 10 +- sound/soc/codecs/wm8995.c | 4 +- sound/soc/codecs/wm8996.c | 8 +- sound/soc/codecs/wm9081.c | 8 +- sound/soc/codecs/wm9712.c | 10 +- sound/soc/codecs/wm9713.c | 18 +- sound/soc/codecs/wm_hubs.c | 3 +- sound/soc/codecs/wsa881x.c | 17 +- sound/soc/codecs/wsa883x.c | 202 +- sound/soc/codecs/wsa884x.c | 8 +- sound/soc/dwc/dwc-i2s.c | 6 +- sound/soc/fsl/fsl-asoc-card.c | 4 +- sound/soc/fsl/fsl_aud2htx.c | 3 +- sound/soc/fsl/fsl_audmix.c | 16 +- sound/soc/fsl/fsl_esai.c | 9 +- sound/soc/fsl/fsl_micfil.c | 58 +- sound/soc/fsl/fsl_micfil.h | 1 + sound/soc/fsl/fsl_qmc_audio.c | 4 +- sound/soc/fsl/fsl_sai.c | 8 +- sound/soc/fsl/fsl_xcvr.c | 3 +- sound/soc/fsl/imx-card.c | 5 + sound/soc/generic/audio-graph-card.c | 97 +- .../generic/audio-graph-card2-custom-sample.dtsi | 702 ------- .../generic/audio-graph-card2-custom-sample1.dtsi | 396 ++++ .../generic/audio-graph-card2-custom-sample2.dtsi | 382 ++++ sound/soc/generic/audio-graph-card2.c | 95 +- sound/soc/generic/simple-card-utils.c | 177 +- sound/soc/generic/simple-card.c | 132 +- sound/soc/img/img-i2s-in.c | 9 +- sound/soc/img/img-i2s-out.c | 9 +- sound/soc/img/img-parallel-out.c | 5 +- sound/soc/img/img-spdif-in.c | 9 +- sound/soc/img/img-spdif-out.c | 10 +- sound/soc/img/pistachio-internal-dac.c | 8 +- sound/soc/intel/avs/avs.h | 1 + sound/soc/intel/avs/board_selection.c | 17 +- sound/soc/intel/avs/boards/Kconfig | 10 + sound/soc/intel/avs/boards/Makefile | 2 + sound/soc/intel/avs/boards/da7219.c | 5 +- sound/soc/intel/avs/boards/es8336.c | 3 +- sound/soc/intel/avs/boards/max98357a.c | 2 +- sound/soc/intel/avs/boards/max98373.c | 2 +- sound/soc/intel/avs/boards/max98927.c | 4 +- sound/soc/intel/avs/boards/nau8825.c | 5 +- sound/soc/intel/avs/boards/pcm3168a.c | 143 ++ sound/soc/intel/avs/boards/rt274.c | 5 +- sound/soc/intel/avs/boards/rt286.c | 5 +- sound/soc/intel/avs/boards/rt298.c | 7 +- sound/soc/intel/avs/boards/rt5514.c | 2 +- sound/soc/intel/avs/boards/rt5663.c | 3 +- sound/soc/intel/avs/boards/rt5682.c | 3 +- sound/soc/intel/avs/boards/ssm4567.c | 4 +- sound/soc/intel/avs/control.c | 182 +- sound/soc/intel/avs/control.h | 12 +- sound/soc/intel/avs/core.c | 24 +- sound/soc/intel/avs/dsp.c | 3 +- sound/soc/intel/avs/loader.c | 64 +- sound/soc/intel/avs/messages.c | 149 +- sound/soc/intel/avs/messages.h | 46 +- sound/soc/intel/avs/path.c | 375 ++-- sound/soc/intel/avs/path.h | 5 + sound/soc/intel/avs/pcm.c | 2 +- sound/soc/intel/avs/tgl.c | 33 + sound/soc/intel/avs/topology.c | 89 +- sound/soc/intel/avs/topology.h | 16 +- sound/soc/intel/catpt/device.c | 14 +- sound/soc/intel/common/soc-acpi-intel-adl-match.c | 29 + sound/soc/intel/common/soc-acpi-intel-lnl-match.c | 137 ++ sound/soc/intel/common/soc-acpi-intel-ptl-match.c | 284 +++ .../intel/common/soc-acpi-intel-sdw-mockup-match.c | 42 + .../intel/common/soc-acpi-intel-sdw-mockup-match.h | 1 + sound/soc/kirkwood/armada-370-db.c | 6 +- sound/soc/mediatek/common/mtk-afe-fe-dai.c | 20 - sound/soc/mediatek/common/mtk-afe-fe-dai.h | 2 - sound/soc/mediatek/common/mtk-soc-card.h | 1 + sound/soc/mediatek/common/mtk-soundcard-driver.c | 19 +- sound/soc/mediatek/mt2701/mt2701-afe-pcm.c | 6 +- sound/soc/mediatek/mt2701/mt2701-cs42448.c | 8 +- sound/soc/mediatek/mt2701/mt2701-wm8960.c | 2 +- sound/soc/mediatek/mt6797/mt6797-afe-pcm.c | 6 +- sound/soc/mediatek/mt7986/mt7986-afe-pcm.c | 6 +- sound/soc/mediatek/mt7986/mt7986-dai-etdm.c | 4 +- sound/soc/mediatek/mt7986/mt7986-wm8960.c | 2 +- sound/soc/mediatek/mt8173/mt8173-afe-pcm.c | 6 +- sound/soc/mediatek/mt8173/mt8173-max98090.c | 2 +- sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c | 2 +- sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c | 4 +- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 2 +- sound/soc/mediatek/mt8183/mt8183-afe-pcm.c | 6 +- sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c | 2 +- .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 4 +- sound/soc/mediatek/mt8186/mt8186-afe-clk.c | 55 - sound/soc/mediatek/mt8186/mt8186-afe-clk.h | 2 - sound/soc/mediatek/mt8186/mt8186-afe-pcm.c | 6 +- sound/soc/mediatek/mt8186/mt8186-mt6366.c | 2 +- sound/soc/mediatek/mt8188/Makefile | 1 + sound/soc/mediatek/mt8188/mt8188-afe-clk.c | 8 + sound/soc/mediatek/mt8188/mt8188-afe-clk.h | 8 + sound/soc/mediatek/mt8188/mt8188-afe-common.h | 1 + sound/soc/mediatek/mt8188/mt8188-afe-pcm.c | 34 +- sound/soc/mediatek/mt8188/mt8188-audsys-clk.c | 4 + sound/soc/mediatek/mt8188/mt8188-audsys-clkid.h | 4 + sound/soc/mediatek/mt8188/mt8188-dai-dmic.c | 683 +++++++ sound/soc/mediatek/mt8188/mt8188-mt6359.c | 57 + sound/soc/mediatek/mt8188/mt8188-reg.h | 17 +- sound/soc/mediatek/mt8192/mt8192-afe-pcm.c | 6 +- .../mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c | 2 +- sound/soc/mediatek/mt8195/mt8195-afe-pcm.c | 6 +- sound/soc/mediatek/mt8195/mt8195-mt6359.c | 12 +- sound/soc/mediatek/mt8365/mt8365-afe-clk.c | 3 +- sound/soc/mediatek/mt8365/mt8365-afe-pcm.c | 19 +- sound/soc/mediatek/mt8365/mt8365-dai-i2s.c | 12 +- sound/soc/mediatek/mt8365/mt8365-dai-pcm.c | 4 +- sound/soc/mediatek/mt8365/mt8365-mt6357.c | 2 +- sound/soc/meson/axg-tdm-interface.c | 9 +- sound/soc/meson/meson-card-utils.c | 4 +- sound/soc/meson/t9015.c | 4 +- sound/soc/mxs/mxs-sgtl5000.c | 2 +- sound/soc/pxa/spitz.c | 2 +- sound/soc/qcom/lpass-sc7180.c | 8 +- sound/soc/qcom/lpass-sc7280.c | 8 +- sound/soc/qcom/qdsp6/q6apm.c | 2 +- sound/soc/qcom/sc7180.c | 2 +- sound/soc/qcom/sc7280.c | 4 +- sound/soc/qcom/sdw.c | 34 +- sound/soc/qcom/sm8250.c | 3 + sound/soc/renesas/migor.c | 2 +- sound/soc/renesas/rcar/core.c | 8 +- sound/soc/rockchip/rk3288_hdmi_analog.c | 2 +- sound/soc/rockchip/rk3399_gru_sound.c | 10 +- sound/soc/rockchip/rockchip_i2s.c | 5 +- sound/soc/rockchip/rockchip_i2s_tdm.c | 16 +- sound/soc/rockchip/rockchip_max98090.c | 8 +- sound/soc/rockchip/rockchip_pdm.c | 10 +- sound/soc/rockchip/rockchip_rt5645.c | 2 +- sound/soc/rockchip/rockchip_spdif.c | 9 +- sound/soc/samsung/aries_wm8994.c | 6 +- sound/soc/samsung/arndale.c | 4 +- sound/soc/samsung/bells.c | 21 +- sound/soc/samsung/i2s.c | 10 +- sound/soc/samsung/littlemill.c | 5 +- sound/soc/samsung/lowland.c | 7 +- sound/soc/samsung/midas_wm1811.c | 2 +- sound/soc/samsung/odroid.c | 2 +- sound/soc/samsung/smdk_wm8994.c | 4 +- sound/soc/samsung/smdk_wm8994pcm.c | 2 +- sound/soc/samsung/snow.c | 2 +- sound/soc/samsung/speyside.c | 53 +- sound/soc/samsung/tm2_wm5110.c | 9 +- sound/soc/samsung/tobermory.c | 3 +- sound/soc/sdca/Makefile | 2 +- sound/soc/sdca/sdca_device.c | 3 +- sound/soc/sdca/sdca_functions.c | 1697 ++++++++++++++++- sound/soc/sdca/sdca_regmap.c | 321 ++++ sound/soc/sdw_utils/soc_sdw_utils.c | 14 +- sound/soc/soc-card.c | 14 +- sound/soc/soc-component.c | 38 +- sound/soc/soc-compress.c | 2 +- sound/soc/soc-core.c | 50 +- sound/soc/soc-dai.c | 38 +- sound/soc/soc-dapm.c | 51 +- sound/soc/soc-link.c | 18 +- sound/soc/soc-ops-test.c | 541 ++++++ sound/soc/soc-ops.c | 731 +++----- sound/soc/soc-pcm.c | 281 ++- sound/soc/soc-topology.c | 59 +- sound/soc/soc-utils.c | 61 +- sound/soc/sof/amd/Kconfig | 5 + sound/soc/sof/amd/acp.c | 56 +- sound/soc/sof/amd/acp.h | 2 - sound/soc/sof/amd/pci-acp63.c | 3 +- sound/soc/sof/amd/pci-acp70.c | 3 +- sound/soc/sof/amd/pci-rmb.c | 1 - sound/soc/sof/amd/pci-rn.c | 3 +- sound/soc/sof/amd/pci-vangogh.c | 3 +- sound/soc/sof/imx/Kconfig | 17 +- sound/soc/sof/imx/Makefile | 6 +- sound/soc/sof/imx/imx-common.c | 430 ++++- sound/soc/sof/imx/imx-common.h | 151 ++ sound/soc/sof/imx/imx8.c | 755 +++----- sound/soc/sof/imx/imx8m.c | 567 ------ sound/soc/sof/imx/imx8ulp.c | 520 ------ sound/soc/sof/imx/imx9.c | 137 ++ sound/soc/sof/intel/Makefile | 2 +- sound/soc/sof/intel/bdw.c | 3 +- sound/soc/sof/intel/byt.c | 3 +- sound/soc/sof/intel/hda-dai.c | 2 +- sound/soc/sof/intel/hda-mlink.c | 127 ++ sound/soc/sof/intel/hda.c | 34 + sound/soc/sof/intel/hda.h | 4 - sound/soc/sof/intel/lnl.c | 117 +- sound/soc/sof/intel/lnl.h | 6 + sound/soc/sof/intel/mtl.c | 81 +- sound/soc/sof/intel/mtl.h | 15 +- sound/soc/sof/intel/pci-apl.c | 2 +- sound/soc/sof/intel/pci-cnl.c | 2 +- sound/soc/sof/intel/pci-icl.c | 2 +- sound/soc/sof/intel/pci-lnl.c | 14 +- sound/soc/sof/intel/pci-mtl.c | 10 +- sound/soc/sof/intel/pci-ptl.c | 19 +- sound/soc/sof/intel/pci-skl.c | 2 +- sound/soc/sof/intel/pci-tgl.c | 2 +- sound/soc/sof/intel/pci-tng.c | 3 +- sound/soc/sof/intel/ptl.c | 106 ++ sound/soc/sof/intel/ptl.h | 19 + sound/soc/sof/intel/shim.h | 2 + sound/soc/sof/ipc3-pcm.c | 13 +- sound/soc/sof/ipc3-topology.c | 20 +- sound/soc/sof/ipc3.c | 12 +- sound/soc/sof/ipc4-loader.c | 176 +- sound/soc/sof/ipc4-pcm.c | 21 +- sound/soc/sof/ipc4-priv.h | 6 + sound/soc/sof/ipc4-topology.c | 65 +- sound/soc/sof/ipc4.c | 26 +- sound/soc/sof/mediatek/mt8186/mt8186.c | 107 +- sound/soc/sof/mediatek/mt8195/mt8195-clk.c | 3 +- sound/soc/sof/mediatek/mt8195/mt8195.c | 107 +- sound/soc/sof/mediatek/mtk-adsp-common.c | 130 ++ sound/soc/sof/mediatek/mtk-adsp-common.h | 10 + sound/soc/sof/pcm.c | 169 +- sound/soc/sof/sof-acpi-dev.c | 9 +- sound/soc/sof/sof-audio.c | 49 - sound/soc/sof/sof-audio.h | 17 +- sound/soc/sof/sof-of-dev.c | 8 +- sound/soc/sof/sof-pci-dev.c | 9 +- sound/soc/sof/topology.c | 4 +- sound/soc/stm/stm32_i2s.c | 6 +- sound/soc/stm/stm32_sai.c | 6 +- sound/soc/stm/stm32_sai_sub.c | 6 +- sound/soc/stm/stm32_spdifrx.c | 6 +- sound/soc/sunxi/sun4i-codec.c | 57 +- sound/soc/sunxi/sun4i-spdif.c | 6 +- sound/soc/sunxi/sun50i-dmic.c | 6 +- sound/soc/sunxi/sun8i-codec.c | 6 +- sound/soc/tegra/Makefile | 2 +- sound/soc/tegra/tegra186_asrc.c | 13 +- sound/soc/tegra/tegra186_dspk.c | 13 +- sound/soc/tegra/tegra20_i2s.c | 13 +- sound/soc/tegra/tegra20_spdif.c | 13 +- sound/soc/tegra/tegra210_admaif.c | 37 +- sound/soc/tegra/tegra210_admaif.h | 9 +- sound/soc/tegra/tegra210_adx.c | 19 +- sound/soc/tegra/tegra210_ahub.c | 13 +- sound/soc/tegra/tegra210_amx.c | 13 +- sound/soc/tegra/tegra210_dmic.c | 13 +- sound/soc/tegra/tegra210_i2s.c | 13 +- sound/soc/tegra/tegra210_mixer.c | 13 +- sound/soc/tegra/tegra210_mvc.c | 13 +- sound/soc/tegra/tegra210_ope.c | 13 +- sound/soc/tegra/tegra210_sfc.c | 13 +- sound/soc/tegra/tegra30_ahub.c | 13 +- sound/soc/tegra/tegra30_i2s.c | 13 +- sound/soc/tegra/tegra_asoc_machine.c | 18 +- sound/soc/tegra/tegra_isomgr_bw.c | 129 ++ sound/soc/tegra/tegra_isomgr_bw.h | 31 + sound/soc/tegra/tegra_wm8903.c | 2 +- sound/soc/ti/ams-delta.c | 2 +- sound/soc/ti/davinci-evm.c | 2 +- sound/soc/ti/davinci-i2s.c | 6 +- sound/soc/ti/j721e-evm.c | 7 +- sound/soc/ti/n810.c | 2 +- sound/soc/ti/omap-twl4030.c | 6 +- sound/soc/ti/omap3pandora.c | 4 +- sound/soc/ti/osk5912.c | 2 +- sound/soc/ti/rx51.c | 2 +- sound/soc/uniphier/aio-cpu.c | 8 +- sound/soc/ux500/mop500_ab8500.c | 4 +- sound/soc/xtensa/xtfpga-i2s.c | 6 +- 581 files changed, 18610 insertions(+), 7971 deletions(-) delete mode 100644 Documentation/devicetree/bindings/misc/atmel-ssc.txt create mode 100644 Documentation/devicetree/bindings/sound/atmel,at91-ssc.yaml create mode 100644 Documentation/devicetree/bindings/sound/atmel,at91sam9g20ek-wm8731.yaml delete mode 100644 Documentation/devicetree/bindings/sound/atmel-at91sam9g20ek-wm8731-audio.txt create mode 100644 Documentation/devicetree/bindings/sound/fsl,imx95-cm7-sof.yaml create mode 100644 Documentation/devicetree/bindings/sound/fsl,sof-cpu.yaml delete mode 100644 Documentation/devicetree/bindings/sound/xlnx,audio-formatter.txt create mode 100644 Documentation/devicetree/bindings/sound/xlnx,audio-formatter.yaml delete mode 100644 Documentation/devicetree/bindings/sound/xlnx,i2s.txt create mode 100644 Documentation/devicetree/bindings/sound/xlnx,i2s.yaml delete mode 100644 Documentation/devicetree/bindings/sound/xlnx,spdif.txt create mode 100644 Documentation/devicetree/bindings/sound/xlnx,spdif.yaml create mode 100644 include/dt-bindings/sound/qcom,wcd934x.h create mode 100644 include/linux/device/devres.h create mode 100644 include/sound/sdca_regmap.h create mode 100644 sound/soc/amd/acp/amd-acp70-acpi-match.c create mode 100644 sound/soc/amd/acp/amd-acpi-mach.c create mode 100644 sound/soc/amd/ps/ps-common.c create mode 100644 sound/soc/codecs/aw88166.c create mode 100644 sound/soc/codecs/aw88166.h delete mode 100644 sound/soc/codecs/pcm3008.h delete mode 100644 sound/soc/generic/audio-graph-card2-custom-sample.dtsi create mode 100644 sound/soc/generic/audio-graph-card2-custom-sample1.dtsi create mode 100644 sound/soc/generic/audio-graph-card2-custom-sample2.dtsi create mode 100644 sound/soc/intel/avs/boards/pcm3168a.c create mode 100644 sound/soc/mediatek/mt8188/mt8188-dai-dmic.c create mode 100644 sound/soc/sdca/sdca_regmap.c create mode 100644 sound/soc/soc-ops-test.c delete mode 100644 sound/soc/sof/imx/imx8m.c delete mode 100644 sound/soc/sof/imx/imx8ulp.c create mode 100644 sound/soc/sof/imx/imx9.c create mode 100644 sound/soc/sof/intel/ptl.c create mode 100644 sound/soc/sof/intel/ptl.h create mode 100644 sound/soc/tegra/tegra_isomgr_bw.c create mode 100644 sound/soc/tegra/tegra_isomgr_bw.h

2 1

ucm2: Qualcomm: fix typo in Lenovo T14s matching
by GitHub pull_request - opened 24 Mar '25

24 Mar '25

alsa-project/alsa-ucm-conf pull request #532 was opened from obbardc: The original commit referenced an incorrect laptop model in the matching code. Use the correct model. This introduces no functional changes. Fixes: e17dde6fd1c3 ("ucm2: Qualcomm: add Lenovo T14s support") cc @Srinivas-Kandagatla Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/532 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/532.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ucm2: Qualcomm: add HP Omnibook X14 support
by GitHub pull_request - opened 24 Mar '25

24 Mar '25

alsa-project/alsa-ucm-conf pull request #531 was opened from jglathe: assume same as T14s (2 speakers) Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/531 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/531.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

sof-soundwire: rt1318: add playback control switch
by GitHub pull_request - edited 24 Mar '25

24 Mar '25

alsa-project/alsa-ucm-conf pull request #525 was edited from shumingfan: rt1318: add playback control switch [v2] This patch defines a remapped one 'rt1318 DAC Playback Switch' whether the cases are in one amp or two amps. [v3] This patch defines 'rt1318-1 DAC' as PlaybackMixerElem when equipped with only one amp and defines a remapped one 'rt1318 DAC Playback Switch' when using two amps. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/525 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/525.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

envy24control: simplify code by using gtk_widget_show_all
by GitHub pull_request - opened 23 Mar '25

23 Mar '25

alsa-project/alsa-tools pull request #29 was opened from andreas56: Remove all gtk_widget_show calls and replace it with just one gtk_widget_show_all. This makes the code shorter and it will also help a future port to GTK 4, where gtk_widget_show is gone. Request URL : https://github.com/alsa-project/alsa-tools/pull/29 Patch URL : https://github.com/alsa-project/alsa-tools/pull/29.patch Repository URL: https://github.com/alsa-project/alsa-tools

1 0

envy24control: fix spdif output settings
by GitHub pull_request - edited 23 Mar '25

23 Mar '25

alsa-project/alsa-tools pull request #28 was edited from andreas56: The S/PDIF Output Settings in envy24control had some bugs: wrong bits were set when user changed "Stream" under "Professional" or any of the settings under "Consumer". Request URL : https://github.com/alsa-project/alsa-tools/pull/28 Patch URL : https://github.com/alsa-project/alsa-tools/pull/28.patch Repository URL: https://github.com/alsa-project/alsa-tools

1 0

envy24control: fix spdif output settings
by GitHub pull_request - opened 23 Mar '25

23 Mar '25

alsa-project/alsa-tools pull request #28 was opened from andreas56: The S/PDIF Output Settings in envy24control had some bugs: wrong bits were set when user changed "Stream" under "Professional" or any of the the settings under "Consumer". Request URL : https://github.com/alsa-project/alsa-tools/pull/28 Patch URL : https://github.com/alsa-project/alsa-tools/pull/28.patch Repository URL: https://github.com/alsa-project/alsa-tools

1 0

[Regression] v1.2.13 broke my laptop's headphone microphone
by GitHub issues - opened 22 Mar '25

22 Mar '25

alsa-project/alsa-ucm-conf issue #530 was opened from vixalien: Hello everyone! I have a HP ProBook 440 G8 Notebook PC, and sound and microphones were working correctly on v1.2.12, but now the microphone is not working after upgrading to v1.2.13. Here's my system details: ``` OS: Alpine Linux edge x86_64 Host: HP ProBook 440 G8 Notebook PC Kernel: Linux 6.13.7-0-edge ``` ``` $ arecord -l **** List of CAPTURE Hardware Devices **** card 0: sofhdadsp [sof-hda-dsp], device 0: HDA Analog (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: sofhdadsp [sof-hda-dsp], device 6: DMIC (*) [] Subdevices: 0/1 Subdevice #0: subdevice #0 card 0: sofhdadsp [sof-hda-dsp], device 7: DMIC16kHz (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 ``` The `arecord` commands only works when I set the device to 0,6 and 0,7 ``` arecord --duration=5 --format=dat --device=hw:0,0 test-mic.wav # does not work, empty audio arecord --duration=5 --format=dat --device=hw:0,6 test-mic.wav # works arecord --duration=5 --format=dat --device=hw:0,7 test-mic.wav # works ``` On v1.2.12, pipewire is automatically configured to use `0,6`, but on v1.12.13, it seems like it's configured to use `0,0` which causes that audio does not work. I'm sorry this is my first issue report. I'm happy to provie more details if necessary. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/530 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

sof-soundwire: rt1318: add playback control switch
by GitHub pull_request - edited 22 Mar '25

22 Mar '25

alsa-project/alsa-ucm-conf pull request #525 was edited from shumingfan: rt1318: add playback control switch [v2] This patch defines a remapped one 'rt1318 DAC Playback Switch' whether the cases are in one amp or two amps. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/525 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/525.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

arecord --max-file-time behavior
by GitHub issues - opened 22 Mar '25

22 Mar '25

alsa-project/alsa-utils issue #296 was opened from thk686: I do environmental sound recordings where arecord runs for months to years with --max-file-time set to something like 15 minutes. It is very nice to have this feature, yet there is one aspect that is problematic. It is not simple to know which file is currently being written. One can time stamp the file names and ignore the most recent, which is probably the simplest. Nonetheless, I wonder if it could be the case that the current write file could be named something else, such as the file name but preceded by a . or other indicator, and have it renamed to its intended name once it is closed and arecord has moved on to another file. Issue URL : https://github.com/alsa-project/alsa-utils/issues/296 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH 6.1] ALSA: control: Avoid WARN() for symlink errors
by Alexey Nepomnyashih 21 Mar '25

21 Mar '25

From: Takashi Iwai <tiwai(a)suse.de> commit b2e538a9827dd04ab5273bf4be8eb2edb84357b0 upstream. Using WARN() for showing the error of symlink creations don't give more information than telling that something goes wrong, since the usual code path is a lregister callback from each control element creation. More badly, the use of WARN() rather confuses fuzzer as if it were serious issues. This patch downgrades the warning messages to use the normal dev_err() instead of WARN(). For making it clearer, add the function name to the prefix, too. Fixes: a135dfb5de15 ("ALSA: led control - add sysfs kcontrol LED marking layer") Reported-by: syzbot+4e7919b09c67ffd198ae(a)syzkaller.appspotmail.com Closes: https://lore.kernel.org/675664c7.050a0220.a30f1.018c.GAE@google.com Link: https://patch.msgid.link/20241209095614.4273-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> [ Alexey Nepomnyashih: changed &card->ctl_dev->kobj to &card->ctl_dev.kobj. ] Signed-off-by: Alexey Nepomnyashih <sdl(a)nppct.ru> --- sound/core/control_led.c | 14 ++++++++++---- 1 file changed, 10 insertions(+), 4 deletions(-) diff --git a/sound/core/control_led.c b/sound/core/control_led.c index 3cadd40100f3..6d81a6a59f29 100644 --- a/sound/core/control_led.c +++ b/sound/core/control_led.c @@ -688,10 +688,16 @@ static void snd_ctl_led_sysfs_add(struct snd_card *card) goto cerr; led->cards[card->number] = led_card; snprintf(link_name, sizeof(link_name), "led-%s", led->name); - WARN(sysfs_create_link(&card->ctl_dev.kobj, &led_card->dev.kobj, link_name), - "can't create symlink to controlC%i device\n", card->number); - WARN(sysfs_create_link(&led_card->dev.kobj, &card->card_dev.kobj, "card"), - "can't create symlink to card%i\n", card->number); + if (sysfs_create_link(&card->ctl_dev.kobj, &led_card->dev.kobj, + link_name)) + dev_err(card->dev, + "%s: can't create symlink to controlC%i device\n", + __func__, card->number); + if (sysfs_create_link(&led_card->dev.kobj, &card->card_dev.kobj, + "card")) + dev_err(card->dev, + "%s: can't create symlink to card%i\n", + __func__, card->number); continue; cerr: -- 2.43.0

1 0

[PATCH] ASoC: amd: yc: update quirk data for new Lenovo model
by Syed Saba Kareem 21 Mar '25

21 Mar '25

From: Syed Saba kareem <syed.sabakareem(a)amd.com> Update Quirk data for new Lenovo model 83J2 for YC platform. Signed-off-by: Syed Saba kareem <syed.sabakareem(a)amd.com> --- sound/soc/amd/yc/acp6x-mach.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/amd/yc/acp6x-mach.c b/sound/soc/amd/yc/acp6x-mach.c index bd3808f98ec9..e632f16c9102 100644 --- a/sound/soc/amd/yc/acp6x-mach.c +++ b/sound/soc/amd/yc/acp6x-mach.c @@ -339,6 +339,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "83Q3"), } }, + { + .driver_data = &acp6x_card, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_MATCH(DMI_PRODUCT_NAME, "83J2"), + } + }, { .driver_data = &acp6x_card, .matches = { -- 2.43.0

3 2

[GIT PULL] ASoC fixes for v6.14-rc7
by Mark Brown 21 Mar '25

21 Mar '25

The following changes since commit 658fb7fe8e7f4014ea17a4da0e0c1d9bc319fa35: ASoC: cs42l43: convert to SYSTEM_SLEEP_PM_OPS (2025-03-12 16:12:24 +0000) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.14-rc7 for you to fetch changes up to 45ff65e30deb919604e68faed156ad96ce7474d9: ASoC: ti: j721e-evm: Fix clock configuration for ti,j7200-cpb-audio compatible (2025-03-18 12:27:34 +0000) ---------------------------------------------------------------- ASoC: Fixes for v6.14 A couple of small fixes, plus some new device IDs - nothing super urgent here. ---------------------------------------------------------------- Frank Li (1): ASoC: dt-bindings: fsl,sai: Add i.MX94 support Jayesh Choudhary (1): ASoC: ti: j721e-evm: Fix clock configuration for ti,j7200-cpb-audio compatible Kuninori Morimoto (1): ASoC: simple-card-utils: Don't use __free(device_node) at graph_util_parse_dai() keenplify (1): ASoC: amd: Add DMI quirk for ACP6X mic support Documentation/devicetree/bindings/sound/fsl,sai.yaml | 4 ++++ sound/soc/amd/yc/acp6x-mach.c | 7 +++++++ sound/soc/generic/simple-card-utils.c | 7 +++++-- sound/soc/ti/j721e-evm.c | 2 ++ 4 files changed, 18 insertions(+), 2 deletions(-)

2 1

[tiwai-sound:for-next 8/18] ld: sound/pci/oxygen/oxygen.o:undefined reference to `oxygen_pci_pm'
by kernel test robot 21 Mar '25

21 Mar '25

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-next head: ee3d1f09e04643b1b48bd894de1de3c600a01c05 commit: 5ea0a2206b58356779cc686630b6548a3402991c [8/18] ALSA: oxygen: Convert to EXPORT_SIMPLE_DEV_PM_OPS() config: i386-randconfig-061-20250320 (https://download.01.org/0day-ci/archive/20250321/202503211242.zaTMusOT-lkp@…) compiler: gcc-12 (Debian 12.2.0-14) 12.2.0 reproduce (this is a W=1 build): (https://download.01.org/0day-ci/archive/20250321/202503211242.zaTMusOT-lkp@…) If you fix the issue in a separate patch/commit (i.e. not just a new version of the same patch/commit), kindly add following tags | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202503211242.zaTMusOT-lkp@intel.com/ Note: the tiwai-sound/for-next HEAD ee3d1f09e04643b1b48bd894de1de3c600a01c05 builds fine. It only hurts bisectability. All errors (new ones prefixed by >>): >> ld: sound/pci/oxygen/oxygen.o:(.data+0x454): undefined reference to `oxygen_pci_pm' >> ld: sound/pci/oxygen/virtuoso.o:(.data+0xf4): undefined reference to `oxygen_pci_pm' -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

1 0

[tiwai-sound:for-next 8/17] ld: vmlinux.o:undefined reference to `oxygen_pci_pm'
by kernel test robot 20 Mar '25

20 Mar '25

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-next head: 765fee28fe0e7d6a5ff8f2eeb7990b7d3ad20825 commit: 5ea0a2206b58356779cc686630b6548a3402991c [8/17] ALSA: oxygen: Convert to EXPORT_SIMPLE_DEV_PM_OPS() config: x86_64-buildonly-randconfig-002-20250320 (https://download.01.org/0day-ci/archive/20250320/202503201853.7kB0BPRw-lkp@…) compiler: gcc-12 (Debian 12.2.0-14) 12.2.0 reproduce (this is a W=1 build): (https://download.01.org/0day-ci/archive/20250320/202503201853.7kB0BPRw-lkp@…) If you fix the issue in a separate patch/commit (i.e. not just a new version of the same patch/commit), kindly add following tags | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202503201853.7kB0BPRw-lkp@intel.com/ All errors (new ones prefixed by >>): >> ld: vmlinux.o:(.data+0x57d120): undefined reference to `oxygen_pci_pm' ld: vmlinux.o:(.data+0x57d620): undefined reference to `oxygen_pci_pm' -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

1 0

Fiio K7 Pro USB DAC no Alsamixer Volume Level
by GitHub issues - opened 19 Mar '25

19 Mar '25

alsa-project/alsa-lib issue #444 was opened from Starfoxfs: Hi, i cant set any Volume Level on Fiio K7 Pro USB DAC. Alsamixer says "This Sound Device does not have any controls" I use Wine with sound-alsa Output directly because of SampleRate for Audio Apps, but Volume is to high and i cant change it. I can change the Pipewire or Pulse Volumes but this has no effect to the Alsa Volume. Heres the alsa-info [alsa-info.txt](https://github.com/user-attachments/files/19347989/alsa-info… Issue URL : https://github.com/alsa-project/alsa-lib/issues/444 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Version Bump
by GitHub issues - opened 19 Mar '25

19 Mar '25

alsa-project/alsa-ucm-conf issue #529 was opened from Stormfox2: Would it be possible to get a version bump to 1.2.13 soon? There has been quite a lot of features now that would get submitted with the new release. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/529 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[RESEND PATCH v5] ASoC: tas2781: Support dsp firmware Alpha and Beta seaies
by Shenghao Ding 19 Mar '25

19 Mar '25

For calibration, basic version does not contain any calibration addresses, it depends on calibration tool to convey the addresses to the driver. Since Alpha and Beta firmware, all the calibration addresses are saved into the firmware. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v5: - Instead of the lengthy approach, just replace if-continue with else if, has reduced this by 2/3 of LoCs - Use proper error code in fw_parse_fct_param_address() - return directly in tasdevice_dspfw_ready() - drop Unneeded assignment v4: - drop unnecceary "unsigned short size;" and "char *data;" in struct fct_param_address - replace strstr() with strncmp() in fct_param_address_parser() - replace hex_parse_u8() with memcpy() - replace hex_parse_u24() with direct assignment - drop hex_parse_u8() & hex_parse_u24() - Just return directly and use proper error codes and drop "goto out" and "out:" label in fw_parse_fct_param_address - fixed indentation issues in cali_reg_update - fixed the logical splits in tas2781_calib_stop_put() - return -ENOMEM if kmemdup fails in fw_parse_fct_param_address() - remove unnecessary "out:" label v3: - Merge the 4 patches into one patch v2: - | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: | https://urldefense.com/v3/__https://lore.kernel.org/oe-kbuild-all/2025 | 02160812.P7szOygN-lkp(a)intel.com/__;!!G3vK!WriwhtbQGlg6p00bcWutlhb-y9vS | pDrSu5_g9oOzzGOoABdirEARTntFzyX6Ss3UPRDxi_E-IA$ - | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: | https://urldefense.com/v3/__https://lore.kernel.org/oe-kbuild-all/2025 | 02160632.u0zxG7yD-lkp(a)intel.com/__;!!G3vK!VtEdfur3fZklvPVfHFFfFP60Ub2E | JYvmjSUZ65lJ6BsQPJDl1fjYRDqh3c6Q5y6sA70ssXDG5g$ v1: - update the year from 2024 to 2025 in the files. - Add dspbin_type to distinguish different dspbin types, such as TASDEV_BASIC, TASDEV_ALPHA, and TASDEV_BETA. - Add struct bulk_reg_val alp_cali_bckp to store the value of pilot tone threshold only for TASDEV_ALPHA & TASDEV_BETA firmware. - Add the declaration of (*fw_parse_fct_param_address)() to parse the addresses of calibration params. - Add the minimum version of TASDEV_ALPHA and TASDEV_BETA firmware. - Add the definition of struct fct_param_address. - update the year from 2024 to 2025. - release the memory to save calibration addresses in tasdevice_dsp_remove(). - Add TASDEV_ALPHA and TASDEV_BETA firmware checking code. - Add fct_param_address_parser for Alpha and Beta firmware. - Add an extra branch to handle calibration address for TASDEV_ALPHA and TASDEV_BETA firmware. --- include/sound/tas2781-dsp.h | 30 ++++- include/sound/tas2781.h | 11 ++ sound/pci/hda/tas2781_spi_fwlib.c | 8 +- sound/soc/codecs/tas2781-fmwlib.c | 190 ++++++++++++++++++++++---- sound/soc/codecs/tas2781-i2c.c | 215 ++++++++++++++++++++++-------- 5 files changed, 368 insertions(+), 86 deletions(-) diff --git a/include/sound/tas2781-dsp.h b/include/sound/tas2781-dsp.h index 3cda9da14f6d..c3a9efa73d5d 100644 --- a/include/sound/tas2781-dsp.h +++ b/include/sound/tas2781-dsp.h @@ -2,7 +2,7 @@ // // ALSA SoC Texas Instruments TAS2781 Audio Smart Amplifier // -// Copyright (C) 2022 - 2024 Texas Instruments Incorporated +// Copyright (C) 2022 - 2025 Texas Instruments Incorporated // https://www.ti.com // // The TAS2781 driver implements a flexible and configurable @@ -30,8 +30,10 @@ #define PRE_DEVICE_C 0x12 #define PRE_DEVICE_D 0x16 -#define PPC3_VERSION 0x4100 -#define PPC3_VERSION_TAS2781 0x14600 +#define PPC3_VERSION_BASE 0x4100 +#define PPC3_VERSION_TAS2781_BASIC_MIN 0x14600 +#define PPC3_VERSION_TAS2781_ALPHA_MIN 0x4a00 +#define PPC3_VERSION_TAS2781_BETA_MIN 0x19400 #define TASDEVICE_DEVICE_SUM 8 #define TASDEVICE_CONFIG_SUM 64 @@ -106,6 +108,27 @@ struct tasdevice_calibration { struct tasdevice_data dev_data; }; +struct fct_param_address { + /* Thermal data for PG 1.0 device */ + unsigned char thr[3]; + /* Thermal data for PG 2.0 device */ + unsigned char thr2[3]; + /* Pilot tone enable flag, usually the sine wave */ + unsigned char plt_flg[3]; + /* Pilot tone gain for calibration */ + unsigned char sin_gn[3]; + /* Pilot tone gain for calibration */ + unsigned char sin_gn2[3]; + /* high 32-bit of real-time spk impedance */ + unsigned char r0_reg[3]; + /* check spk connection */ + unsigned char tf_reg[3]; + /* check spk resonant frequency */ + unsigned char a1_reg[3]; + /* check spk resonant frequency */ + unsigned char a2_reg[3]; +}; + struct tasdevice_fw { struct tasdevice_dspfw_hdr fw_hdr; unsigned short nr_programs; @@ -114,6 +137,7 @@ struct tasdevice_fw { struct tasdevice_config *configs; unsigned short nr_calibrations; struct tasdevice_calibration *calibrations; + struct fct_param_address fct_par_addr; struct device *dev; }; diff --git a/include/sound/tas2781.h b/include/sound/tas2781.h index 4d9a37544354..eff011444cc8 100644 --- a/include/sound/tas2781.h +++ b/include/sound/tas2781.h @@ -113,6 +113,12 @@ enum audio_device { TAS2781, }; +enum dspbin_type { + TASDEV_BASIC, + TASDEV_ALPHA, + TASDEV_BETA, +}; + enum device_catlog_id { LENOVO = 0, OTHERS @@ -127,6 +133,7 @@ struct bulk_reg_val { struct tasdevice { struct bulk_reg_val *cali_data_backup; + struct bulk_reg_val alp_cali_bckp; struct tasdevice_fw *cali_data_fmw; unsigned int dev_addr; unsigned int err_code; @@ -171,6 +178,7 @@ struct tasdevice_priv { unsigned char dev_name[32]; const char *name_prefix; unsigned char ndev; + unsigned int dspbin_typ; unsigned int magic_num; unsigned int chip_id; unsigned int sysclk; @@ -196,6 +204,9 @@ struct tasdevice_priv { int (*fw_parse_configuration_data)(struct tasdevice_priv *tas_priv, struct tasdevice_fw *tas_fmw, const struct firmware *fmw, int offset); + int (*fw_parse_fct_param_address)(struct tasdevice_priv *tas_priv, + struct tasdevice_fw *tas_fmw, + const struct firmware *fmw, int offset); int (*tasdevice_load_block)(struct tasdevice_priv *tas_priv, struct tasdev_blk *block); diff --git a/sound/pci/hda/tas2781_spi_fwlib.c b/sound/pci/hda/tas2781_spi_fwlib.c index 0e2acbc3c900..19a693d58ee8 100644 --- a/sound/pci/hda/tas2781_spi_fwlib.c +++ b/sound/pci/hda/tas2781_spi_fwlib.c @@ -2,7 +2,7 @@ // // TAS2781 HDA SPI driver // -// Copyright 2024 Texas Instruments, Inc. +// Copyright 2024 - 2025 Texas Instruments, Inc. // // Author: Baojun Xu <baojun.xu(a)ti.com> @@ -370,10 +370,10 @@ static unsigned char map_dev_idx(struct tasdevice_fw *tas_fmw, int i, n = ARRAY_SIZE(non_ppc3_mapping_table); unsigned char dev_idx = 0; - if (fw_fixed_hdr->ppcver >= PPC3_VERSION_TAS2781) { + if (fw_fixed_hdr->ppcver >= PPC3_VERSION_TAS2781_BASIC_MIN) { p = (struct blktyp_devidx_map *)ppc3_tas2781_mapping_table; n = ARRAY_SIZE(ppc3_tas2781_mapping_table); - } else if (fw_fixed_hdr->ppcver >= PPC3_VERSION) { + } else if (fw_fixed_hdr->ppcver >= PPC3_VERSION_BASE) { p = (struct blktyp_devidx_map *)ppc3_mapping_table; n = ARRAY_SIZE(ppc3_mapping_table); } @@ -1605,7 +1605,7 @@ static int dspfw_default_callback(struct tasdevice_priv *tas_priv, int rc = 0; if (drv_ver == 0x100) { - if (ppcver >= PPC3_VERSION) { + if (ppcver >= PPC3_VERSION_BASE) { tas_priv->fw_parse_variable_header = fw_parse_variable_header_kernel; tas_priv->fw_parse_program_data = diff --git a/sound/soc/codecs/tas2781-fmwlib.c b/sound/soc/codecs/tas2781-fmwlib.c index bc2f68bd9846..13a197468193 100644 --- a/sound/soc/codecs/tas2781-fmwlib.c +++ b/sound/soc/codecs/tas2781-fmwlib.c @@ -390,10 +390,10 @@ static unsigned char map_dev_idx(struct tasdevice_fw *tas_fmw, int i, n = ARRAY_SIZE(non_ppc3_mapping_table); unsigned char dev_idx = 0; - if (fw_fixed_hdr->ppcver >= PPC3_VERSION_TAS2781) { + if (fw_fixed_hdr->ppcver >= PPC3_VERSION_TAS2781_BASIC_MIN) { p = (struct blktyp_devidx_map *)ppc3_tas2781_mapping_table; n = ARRAY_SIZE(ppc3_tas2781_mapping_table); - } else if (fw_fixed_hdr->ppcver >= PPC3_VERSION) { + } else if (fw_fixed_hdr->ppcver >= PPC3_VERSION_BASE) { p = (struct blktyp_devidx_map *)ppc3_mapping_table; n = ARRAY_SIZE(ppc3_mapping_table); } @@ -560,6 +560,124 @@ static int fw_parse_configuration_data_kernel( return offset; } +static void fct_param_address_parser(struct cali_reg *r, + struct tasdevice_fw *tas_fmw, const unsigned char *data) +{ + struct fct_param_address *p = &tas_fmw->fct_par_addr; + unsigned int i; + + /* + * Calibration parameters locations and data schema in dsp firmware. + * The number of items are flexible, but not more than 20. The dsp tool + * will reseve 20*24-byte space for fct params. In some cases, the + * number of fct param is less than 20, the data will be saved from the + * beginning, the rest part will be stuffed with zero. + * + * fct_param_num (not more than 20) + * for (i = 0; i < fct_param_num; i++) { + * Alias of fct param (20 bytes) + * Book (1 byte) + * Page (1 byte) + * Offset (1 byte) + * CoeffLength (1 byte) = 0x1 + * } + * if (20 - fct_param_num) + * 24*(20 - fct_param_num) pieces of '0' as stuffing + * + * As follow: + * umg_SsmKEGCye = Book, Page, Offset, CoeffLength + * iks_E0 = Book, Page, Offset, CoeffLength + * yep_LsqM0 = Book, Page, Offset, CoeffLength + * oyz_U0_ujx = Book, Page, Offset, CoeffLength + * iks_GC_GMgq = Book, Page, Offset, CoeffLength + * gou_Yao = Book, Page, Offset, CoeffLength + * kgd_Wsc_Qsbp = Book, Page, Offset, CoeffLength + * yec_CqseSsqs = Book, Page, Offset, CoeffLength + * iks_SogkGgog2 = Book, Page, Offset, CoeffLength + * yec_Sae_Y = Book, Page, Offset, CoeffLength + * Re_Int = Book, Page, Offset, CoeffLength + * SigFlag = Book, Page, Offset, CoeffLength + * a1_Int = Book, Page, Offset, CoeffLength + * a2_Int = Book, Page, Offset, CoeffLength + */ + for (i = 0; i < 20; i++) { + const unsigned char *dat = &data[24 * i]; + + /* + * check whether current fct param is empty. + */ + if (dat[23] != 1) + break; + + if (!strncmp(dat, "umg_SsmKEGCye", 20)) + r->pow_reg = TASDEVICE_REG(dat[20], dat[21], dat[22]); + /* high 32-bit of real-time spk impedance */ + else if (!strncmp(dat, "iks_E0", 20)) + r->r0_reg = TASDEVICE_REG(dat[20], dat[21], dat[22]); + /* inverse of real-time spk impedance */ + else if (!strncmp(dat, "yep_LsqM0", 20)) + r->invr0_reg = + TASDEVICE_REG(dat[20], dat[21], dat[22]); + /* low 32-bit of real-time spk impedance */ + else if (!strncmp(dat, "oyz_U0_ujx", 20)) + r->r0_low_reg = + TASDEVICE_REG(dat[20], dat[21], dat[22]); + /* Delta Thermal Limit */ + else if (!strncmp(dat, "iks_GC_GMgq", 20)) + r->tlimit_reg = + TASDEVICE_REG(dat[20], dat[21], dat[22]); + /* Thermal data for PG 1.0 device */ + else if (!strncmp(dat, "gou_Yao", 20)) + memcpy(p->thr, &dat[20], 3); + /* Pilot tone enable flag, usually the sine wave */ + else if (!strncmp(dat, "kgd_Wsc_Qsbp", 20)) + memcpy(p->plt_flg, &dat[20], 3); + /* Pilot tone gain for calibration */ + else if (!strncmp(dat, "yec_CqseSsqs", 20)) + memcpy(p->sin_gn, &dat[20], 3); + /* Pilot tone gain for calibration, useless in PG 2.0 */ + else if (!strncmp(dat, "iks_SogkGgog2", 20)) + memcpy(p->sin_gn2, &dat[20], 3); + /* Thermal data for PG 2.0 device */ + else if (!strncmp(dat, "yec_Sae_Y", 20)) + memcpy(p->thr2, &dat[20], 3); + /* Spk Equivalent Resistance in fixed-point format */ + else if (!strncmp(dat, "Re_Int", 20)) + memcpy(p->r0_reg, &dat[20], 3); + /* Check whether the spk connection is open */ + else if (!strncmp(dat, "SigFlag", 20)) + memcpy(p->tf_reg, &dat[20], 3); + /* check spk resonant frequency */ + else if (!strncmp(dat, "a1_Int", 20)) + memcpy(p->a1_reg, &dat[20], 3); + /* check spk resonant frequency */ + else if (!strncmp(dat, "a2_Int", 20)) + memcpy(p->a2_reg, &dat[20], 3); + } +} + +static int fw_parse_fct_param_address(struct tasdevice_priv *tas_priv, + struct tasdevice_fw *tas_fmw, const struct firmware *fmw, int offset) +{ + struct calidata *cali_data = &tas_priv->cali_data; + struct cali_reg *r = &cali_data->cali_reg_array; + const unsigned char *data = fmw->data; + + if (offset + 520 > fmw->size) { + dev_err(tas_priv->dev, "%s: File Size error\n", __func__); + return -EINVAL; + } + + /* skip reserved part */ + offset += 40; + + fct_param_address_parser(r, tas_fmw, &data[offset]); + + offset += 480; + + return offset; +} + static int fw_parse_variable_header_kernel( struct tasdevice_priv *tas_priv, const struct firmware *fmw, int offset) @@ -1687,13 +1805,29 @@ static int tasdevice_load_block(struct tasdevice_priv *tas_priv, return rc; } +static void dspbin_type_check(struct tasdevice_priv *tas_priv, + unsigned int ppcver) +{ + if (ppcver >= PPC3_VERSION_TAS2781_ALPHA_MIN) { + if (ppcver >= PPC3_VERSION_TAS2781_BETA_MIN) + tas_priv->dspbin_typ = TASDEV_BETA; + else if (ppcver >= PPC3_VERSION_TAS2781_BASIC_MIN) + tas_priv->dspbin_typ = TASDEV_BASIC; + else + tas_priv->dspbin_typ = TASDEV_ALPHA; + } + if (tas_priv->dspbin_typ != TASDEV_BASIC) + tas_priv->fw_parse_fct_param_address = + fw_parse_fct_param_address; +} + static int dspfw_default_callback(struct tasdevice_priv *tas_priv, unsigned int drv_ver, unsigned int ppcver) { int rc = 0; if (drv_ver == 0x100) { - if (ppcver >= PPC3_VERSION) { + if (ppcver >= PPC3_VERSION_BASE) { tas_priv->fw_parse_variable_header = fw_parse_variable_header_kernel; tas_priv->fw_parse_program_data = @@ -1702,6 +1836,7 @@ static int dspfw_default_callback(struct tasdevice_priv *tas_priv, fw_parse_configuration_data_kernel; tas_priv->tasdevice_load_block = tasdevice_load_block_kernel; + dspbin_type_check(tas_priv, ppcver); } else { switch (ppcver) { case 0x00: @@ -1717,7 +1852,7 @@ static int dspfw_default_callback(struct tasdevice_priv *tas_priv, default: dev_err(tas_priv->dev, "%s: PPCVer must be 0x0 or 0x%02x", - __func__, PPC3_VERSION); + __func__, PPC3_VERSION_BASE); dev_err(tas_priv->dev, " Current:0x%02x\n", ppcver); rc = -EINVAL; @@ -1953,28 +2088,25 @@ static int tasdevice_dspfw_ready(const struct firmware *fmw, struct tasdevice_fw_fixed_hdr *fw_fixed_hdr; struct tasdevice_fw *tas_fmw; int offset = 0; - int ret = 0; + int ret; if (!fmw || !fmw->data) { dev_err(tas_priv->dev, "%s: Failed to read firmware %s\n", __func__, tas_priv->coef_binaryname); - ret = -EINVAL; - goto out; + return -EINVAL; } tas_priv->fmw = kzalloc(sizeof(struct tasdevice_fw), GFP_KERNEL); - if (!tas_priv->fmw) { - ret = -ENOMEM; - goto out; - } + if (!tas_priv->fmw) + return -ENOMEM; + tas_fmw = tas_priv->fmw; tas_fmw->dev = tas_priv->dev; offset = fw_parse_header(tas_priv, tas_fmw, fmw, offset); - if (offset == -EINVAL) { - ret = -EINVAL; - goto out; - } + if (offset == -EINVAL) + return -EINVAL; + fw_fixed_hdr = &(tas_fmw->fw_hdr.fixed_hdr); /* Support different versions of firmware */ switch (fw_fixed_hdr->drv_ver) { @@ -2007,28 +2139,32 @@ static int tasdevice_dspfw_ready(const struct firmware *fmw, ret = dspfw_default_callback(tas_priv, fw_fixed_hdr->drv_ver, fw_fixed_hdr->ppcver); if (ret) - goto out; + return ret; break; } offset = tas_priv->fw_parse_variable_header(tas_priv, fmw, offset); - if (offset < 0) { - ret = offset; - goto out; - } + if (offset < 0) + return offset; + offset = tas_priv->fw_parse_program_data(tas_priv, tas_fmw, fmw, offset); - if (offset < 0) { - ret = offset; - goto out; - } + if (offset < 0) + return offset; + offset = tas_priv->fw_parse_configuration_data(tas_priv, tas_fmw, fmw, offset); if (offset < 0) - ret = offset; + return offset; -out: - return ret; + if (tas_priv->fw_parse_fct_param_address) { + offset = tas_priv->fw_parse_fct_param_address(tas_priv, + tas_fmw, fmw, offset); + if (offset < 0) + return offset; + } + + return 0; } int tasdevice_dsp_parser(void *context) diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index 90c5b2e74d12..b950914b7d48 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -349,13 +349,31 @@ static int calib_data_get(struct tasdevice_priv *tas_priv, int reg, return rc; } +static int partial_cali_data_update(int *reg, int j) +{ + switch (tas2781_cali_start_reg[j].reg) { + case 0: + return reg[0]; + case TAS2781_PRM_PLT_FLAG_REG: + return reg[1]; + case TAS2781_PRM_SINEGAIN_REG: + return reg[2]; + case TAS2781_PRM_SINEGAIN2_REG: + return reg[3]; + default: + return 0; + } +} + static void sngl_calib_start(struct tasdevice_priv *tas_priv, int i, int *reg, unsigned char *dat) { struct tasdevice *tasdev = tas_priv->tasdevice; struct bulk_reg_val *p = tasdev[i].cali_data_backup; + struct bulk_reg_val *t = &tasdev[i].alp_cali_bckp; const int sum = ARRAY_SIZE(tas2781_cali_start_reg); - int j; + unsigned char val[4]; + int j, r; if (p == NULL) return; @@ -370,30 +388,23 @@ static void sngl_calib_start(struct tasdevice_priv *tas_priv, int i, tasdevice_dev_read(tas_priv, i, p[j].reg, (int *)&p[j].val[0]); } else { - switch (tas2781_cali_start_reg[j].reg) { - case 0: { - if (!reg[0]) - continue; - p[j].reg = reg[0]; + if (!tas_priv->dspbin_typ) { + r = partial_cali_data_update(reg, j); + if (r) + p[j].reg = r; } - break; - case TAS2781_PRM_PLT_FLAG_REG: - p[j].reg = reg[1]; - break; - case TAS2781_PRM_SINEGAIN_REG: - p[j].reg = reg[2]; - break; - case TAS2781_PRM_SINEGAIN2_REG: - p[j].reg = reg[3]; - break; - } - tasdevice_dev_bulk_read(tas_priv, i, p[j].reg, - p[j].val, 4); + + if (p[j].reg) + tasdevice_dev_bulk_read(tas_priv, i, p[j].reg, + p[j].val, 4); } } + if (tas_priv->dspbin_typ == TASDEV_ALPHA) + tasdevice_dev_bulk_read(tas_priv, i, t->reg, t->val, 4); + /* Update the setting for calibration */ - for (j = 0; j < sum - 2; j++) { + for (j = 0; j < sum - 4; j++) { if (p[j].val_len == 1) { if (p[j].is_locked) tasdevice_dev_write(tas_priv, i, @@ -401,17 +412,33 @@ static void sngl_calib_start(struct tasdevice_priv *tas_priv, int i, TAS2781_TEST_PAGE_UNLOCK); tasdevice_dev_write(tas_priv, i, p[j].reg, tas2781_cali_start_reg[j].val[0]); - } else { - if (!p[j].reg) - continue; - tasdevice_dev_bulk_write(tas_priv, i, p[j].reg, - (unsigned char *) - tas2781_cali_start_reg[j].val, 4); } } - tasdevice_dev_bulk_write(tas_priv, i, p[j].reg, &dat[1], 4); - tasdevice_dev_bulk_write(tas_priv, i, p[j + 1].reg, &dat[5], 4); + if (tas_priv->dspbin_typ == TASDEV_ALPHA) { + val[0] = 0x00; + val[1] = 0x00; + val[2] = 0x21; + val[3] = 0x8e; + } else { + val[0] = tas2781_cali_start_reg[j].val[0]; + val[1] = tas2781_cali_start_reg[j].val[1]; + val[2] = tas2781_cali_start_reg[j].val[2]; + val[3] = tas2781_cali_start_reg[j].val[3]; + } + tasdevice_dev_bulk_write(tas_priv, i, p[j].reg, val, 4); + tasdevice_dev_bulk_write(tas_priv, i, p[j + 1].reg, + (unsigned char *)tas2781_cali_start_reg[j + 1].val, 4); + tasdevice_dev_bulk_write(tas_priv, i, p[j + 2].reg, &dat[1], 4); + tasdevice_dev_bulk_write(tas_priv, i, p[j + 3].reg, &dat[5], 4); + if (tas_priv->dspbin_typ == TASDEV_ALPHA) { + val[0] = 0x00; + val[1] = 0x00; + val[2] = 0x2a; + val[3] = 0x0b; + + tasdevice_dev_bulk_read(tas_priv, i, t->reg, val, 4); + } } static int tas2781_calib_start_put(struct snd_kcontrol *kcontrol, @@ -452,14 +479,15 @@ static int tas2781_calib_start_put(struct snd_kcontrol *kcontrol, return 1; } -static void tas2781_calib_stop_put(struct tasdevice_priv *tas_priv) +static void tas2781_calib_stop_put(struct tasdevice_priv *priv) { const int sum = ARRAY_SIZE(tas2781_cali_start_reg); int i, j; - for (i = 0; i < tas_priv->ndev; i++) { - struct tasdevice *tasdev = tas_priv->tasdevice; + for (i = 0; i < priv->ndev; i++) { + struct tasdevice *tasdev = priv->tasdevice; struct bulk_reg_val *p = tasdev[i].cali_data_backup; + struct bulk_reg_val *t = &tasdev[i].alp_cali_bckp; if (p == NULL) continue; @@ -467,18 +495,21 @@ static void tas2781_calib_stop_put(struct tasdevice_priv *tas_priv) for (j = 0; j < sum; j++) { if (p[j].val_len == 1) { if (p[j].is_locked) - tasdevice_dev_write(tas_priv, i, + tasdevice_dev_write(priv, i, TAS2781_TEST_UNLOCK_REG, TAS2781_TEST_PAGE_UNLOCK); - tasdevice_dev_write(tas_priv, i, p[j].reg, + tasdevice_dev_write(priv, i, p[j].reg, p[j].val[0]); } else { if (!p[j].reg) continue; - tasdevice_dev_bulk_write(tas_priv, i, p[j].reg, + tasdevice_dev_bulk_write(priv, i, p[j].reg, p[j].val, 4); } } + + if (priv->dspbin_typ == TASDEV_ALPHA) + tasdevice_dev_bulk_write(priv, i, t->reg, t->val, 4); } } @@ -590,16 +621,20 @@ static int tasdev_cali_data_put(struct snd_kcontrol *kcontrol, i += 2; priv->is_user_space_calidata = true; - p->r0_reg = TASDEVICE_REG(src[i], src[i + 1], src[i + 2]); - i += 3; - p->r0_low_reg = TASDEVICE_REG(src[i], src[i + 1], src[i + 2]); - i += 3; - p->invr0_reg = TASDEVICE_REG(src[i], src[i + 1], src[i + 2]); - i += 3; - p->pow_reg = TASDEVICE_REG(src[i], src[i + 1], src[i + 2]); - i += 3; - p->tlimit_reg = TASDEVICE_REG(src[i], src[i + 1], src[i + 2]); - i += 3; + if (priv->dspbin_typ == TASDEV_BASIC) { + p->r0_reg = TASDEVICE_REG(src[i], src[i + 1], src[i + 2]); + i += 3; + p->r0_low_reg = TASDEVICE_REG(src[i], src[i + 1], src[i + 2]); + i += 3; + p->invr0_reg = TASDEVICE_REG(src[i], src[i + 1], src[i + 2]); + i += 3; + p->pow_reg = TASDEVICE_REG(src[i], src[i + 1], src[i + 2]); + i += 3; + p->tlimit_reg = TASDEVICE_REG(src[i], src[i + 1], src[i + 2]); + i += 3; + } else { + i += 15; + } memcpy(dst, &src[i], cali_data->total_sz); return 1; @@ -646,12 +681,19 @@ static int tasdev_tf_data_get(struct snd_kcontrol *kcontrol, struct soc_bytes_ext *bytes_ext = (struct soc_bytes_ext *) kcontrol->private_value; unsigned char *dst = ucontrol->value.bytes.data; - unsigned int reg; + unsigned int reg = TAS2781_RUNTIME_RE_REG_TF; + + if (tas_priv->chip_id == TAS2781) { + struct tasdevice_fw *tas_fmw = tas_priv->fmw; + struct fct_param_address *p = &(tas_fmw->fct_par_addr); - if (tas_priv->chip_id == TAS2781) reg = TAS2781_RUNTIME_RE_REG_TF; - else + if (tas_priv->dspbin_typ) + reg = TASDEVICE_REG(p->tf_reg[0], p->tf_reg[1], + p->tf_reg[2]); + } else { reg = TAS2563_RUNTIME_RE_REG_TF; + } guard(mutex)(&tas_priv->codec_lock); dst[0] = bytes_ext->max; @@ -666,12 +708,19 @@ static int tasdev_re_data_get(struct snd_kcontrol *kcontrol, struct soc_bytes_ext *bytes_ext = (struct soc_bytes_ext *) kcontrol->private_value; unsigned char *dst = ucontrol->value.bytes.data; - unsigned int reg; + unsigned int reg = TAS2781_RUNTIME_RE_REG; - if (tas_priv->chip_id == TAS2781) - reg = TAS2781_RUNTIME_RE_REG; - else + if (tas_priv->chip_id == TAS2781) { + struct tasdevice_fw *tas_fmw = tas_priv->fmw; + struct fct_param_address *p = &(tas_fmw->fct_par_addr); + + if (tas_priv->dspbin_typ) + reg = TASDEVICE_REG(p->r0_reg[0], p->r0_reg[1], + p->r0_reg[2]); + } else { reg = TAS2563_RUNTIME_RE_REG; + } + guard(mutex)(&tas_priv->codec_lock); dst[0] = bytes_ext->max; return calib_data_get(tas_priv, reg, &dst[1]); @@ -705,11 +754,16 @@ static int tasdev_XMA1_data_get(struct snd_kcontrol *kcontrol, { struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + struct tasdevice_fw *tas_fmw = tas_priv->fmw; + struct fct_param_address *p = &(tas_fmw->fct_par_addr); struct soc_bytes_ext *bytes_ext = (struct soc_bytes_ext *) kcontrol->private_value; unsigned char *dst = ucontrol->value.bytes.data; unsigned int reg = TASDEVICE_XM_A1_REG; + if (tas_priv->dspbin_typ) + reg = TASDEVICE_REG(p->a1_reg[0], p->a1_reg[1], p->a1_reg[2]); + guard(mutex)(&tas_priv->codec_lock); dst[0] = bytes_ext->max; return calib_data_get(tas_priv, reg, &dst[1]); @@ -720,11 +774,16 @@ static int tasdev_XMA2_data_get(struct snd_kcontrol *kcontrol, { struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + struct tasdevice_fw *tas_fmw = tas_priv->fmw; + struct fct_param_address *p = &(tas_fmw->fct_par_addr); struct soc_bytes_ext *bytes_ext = (struct soc_bytes_ext *) kcontrol->private_value; unsigned char *dst = ucontrol->value.bytes.data; unsigned int reg = TASDEVICE_XM_A2_REG; + if (tas_priv->dspbin_typ) + reg = TASDEVICE_REG(p->a2_reg[0], p->a2_reg[1], p->a2_reg[2]); + guard(mutex)(&tas_priv->codec_lock); dst[0] = bytes_ext->max; return calib_data_get(tas_priv, reg, &dst[1]); @@ -1172,10 +1231,51 @@ static int tasdevice_dsp_create_ctrls(struct tasdevice_priv *tas_priv) nr_controls < mix_index ? nr_controls : mix_index); } +static void cali_reg_update(struct bulk_reg_val *p, + struct fct_param_address *t) +{ + const int sum = ARRAY_SIZE(tas2781_cali_start_reg); + int reg, j; + + for (j = 0; j < sum; j++) { + switch (tas2781_cali_start_reg[j].reg) { + case 0: + reg = TASDEVICE_REG(t->thr[0], t->thr[1], t->thr[2]); + break; + case TAS2781_PRM_PLT_FLAG_REG: + reg = TASDEVICE_REG(t->plt_flg[0], t->plt_flg[1], + t->plt_flg[2]); + break; + case TAS2781_PRM_SINEGAIN_REG: + reg = TASDEVICE_REG(t->sin_gn[0], t->sin_gn[1], + t->sin_gn[2]); + break; + case TAS2781_PRM_SINEGAIN2_REG: + reg = TASDEVICE_REG(t->sin_gn[0], t->sin_gn[1], + t->sin_gn[2]); + break; + default: + reg = 0; + break; + } + if (reg) + p[j].reg = reg; + } +} + +static void alpa_cali_update(struct bulk_reg_val *p, + struct fct_param_address *t) +{ + p->is_locked = false; + p->reg = TASDEVICE_REG(t->thr2[0], t->thr2[1], t->thr2[2]); + p->val_len = 4; +} + static int tasdevice_create_cali_ctrls(struct tasdevice_priv *priv) { struct calidata *cali_data = &priv->cali_data; struct tasdevice *tasdev = priv->tasdevice; + struct tasdevice_fw *fmw = priv->fmw; struct soc_bytes_ext *ext_cali_data; struct snd_kcontrol_new *cali_ctrls; unsigned int nctrls; @@ -1191,14 +1291,25 @@ static int tasdevice_create_cali_ctrls(struct tasdevice_priv *priv) } if (priv->chip_id == TAS2781) { + struct fct_param_address *t = &(fmw->fct_par_addr); + cali_ctrls = (struct snd_kcontrol_new *)tas2781_cali_controls; nctrls = ARRAY_SIZE(tas2781_cali_controls); for (i = 0; i < priv->ndev; i++) { - tasdev[i].cali_data_backup = + struct bulk_reg_val *p; + + p = tasdev[i].cali_data_backup = kmemdup(tas2781_cali_start_reg, sizeof(tas2781_cali_start_reg), GFP_KERNEL); if (!tasdev[i].cali_data_backup) return -ENOMEM; + if (priv->dspbin_typ) { + cali_reg_update(p, t); + if (priv->dspbin_typ == TASDEV_ALPHA) { + p = &tasdev[i].alp_cali_bckp; + alpa_cali_update(p, t); + } + } } } else { cali_ctrls = (struct snd_kcontrol_new *)tas2563_cali_controls; -- 2.34.1

5 4

No Micrphone detected on ALC4080 - Gigabyte B850I Aorus Pro (rev 1.0)
by GitHub issues - edited 18 Mar '25

18 Mar '25

alsa-project/alsa-ucm-conf issue #528 was edited from chanster: The Rear Microphone is not detected when I plug in my mic. I don't use the Front Audio connection as my case doesn't have I/O for it, I just used the 2 audio ports in the rear. When I try recording directly I can get the mic to work, but it does not show up in the sound settings. `arecord --duration=3 -D hw:CARD=2,DEV=1 -f dat -vvv a.wav` ``` Hardware PCM card 2 'USB Audio' device 1 subdevice 0 Its setup is: stream : CAPTURE access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 24000 period_size : 6000 period_time : 125000 tstamp_mode : ENABLE tstamp_type : MONOTONIC period_step : 1 avail_min : 6000 period_event : 0 start_threshold : 1 stop_threshold : 24000 silence_threshold: 0 silence_size : 0 boundary : 6755399441055744000 appl_ptr : 0 hw_ptr : 0 Recording WAVE 'a.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo Max peak (12000 samples): 0x00000167 # 1% Max peak (12000 samples): 0x000000da # 0% Max peak (12000 samples): 0x000006b1 ## 5% Max peak (12000 samples): 0x00000547 # 4% Max peak (12000 samples): 0x00000729 ## 5% Max peak (12000 samples): 0x00000a77 ## 8% Max peak (12000 samples): 0x000006d2 ## 5% Max peak (12000 samples): 0x000005ec # 4% Max peak (12000 samples): 0x00000627 # 4% Max peak (12000 samples): 0x00000d32 ### 10% Max peak (12000 samples): 0x00000e9f ### 11% Max peak (12000 samples): 0x00003813 ######### 43% Max peak (12000 samples): 0x00006278 ################ 76% Max peak (12000 samples): 0x00007fff #################### 99% Max peak (12000 samples): 0x00006694 ################# 80% Max peak (12000 samples): 0x000068ea ################# 81% Max peak (12000 samples): 0x00004ba0 ############ 59% Max peak (12000 samples): 0x0000207c ###### 25% Max peak (12000 samples): 0x000017e9 #### 18% Max peak (12000 samples): 0x00003938 ######### 44% Max peak (12000 samples): 0x00001f11 ##### 24% Max peak (12000 samples): 0x00000ee5 ### 11% Max peak (12000 samples): 0x00003944 ######### 44% Max peak (12000 samples): 0x00003942 ######### 44% ``` Motherboard: Gigabyte B850I Aorus Pro (rev 1.0) Audio Device: ALC4080 Distro: Fedora 41 Window Manager: Gnome-Shell Packages: ``` dnf list --installed | grep 'pipewire\|pulseaudio\|alsa' alsa-lib.i686 1.2.13-3.fc41 updates alsa-lib.x86_64 1.2.13-3.fc41 updates alsa-ucm.noarch 1.2.13-3.fc41 updates alsa-ucm-utils.x86_64 1.2.13-2.fc41 updates alsa-utils.x86_64 1.2.13-2.fc41 updates pipewire.i686 1.2.7-7.fc41 updates pipewire.x86_64 1.2.7-7.fc41 updates pipewire-alsa.i686 1.2.7-7.fc41 updates pipewire-alsa.x86_64 1.2.7-7.fc41 updates pipewire-gstreamer.x86_64 1.2.7-7.fc41 updates pipewire-libs.i686 1.2.7-7.fc41 updates pipewire-libs.x86_64 1.2.7-7.fc41 updates pipewire-pulseaudio.x86_64 1.2.7-7.fc41 updates pipewire-utils.x86_64 1.2.7-7.fc41 updates pulseaudio-libs.i686 17.0-2.fc41 fedora pulseaudio-libs.x86_64 17.0-2.fc41 fedora pulseaudio-libs-glib2.x86_64 17.0-2.fc41 fedora wine-alsa.i686 10.1-1.fc41 updates wine-alsa.x86_64 10.1-1.fc41 updates wine-pulseaudio.i686 10.1-1.fc41 updates wine-pulseaudio.x86_64 10.1-1.fc41 updates ``` `lsusb | grep -i audio` ``` Bus 001 Device 002: ID 0414:a014 Giga-Byte Technology Co., Ltd USB Audio ``` `alsactl monitor` (unplugging and plugging mic) ``` node hw:2, #9 (0,0,0,Mic - Input Jack,1) VALUE node hw:2, #9 (0,0,0,Mic - Input Jack,1) VALUE ``` `amixer -c events` (unplugging and plugging mic) ``` event value: numid=9,iface=CARD,name='Mic - Input Jack',index=1 event value: numid=9,iface=CARD,name='Mic - Input Jack',index=1 ``` `amixer -c 2 contents` ``` numid=8,iface=CARD,name='Clock Source 3 Validity' ; type=BOOLEAN,access=r-------,values=1 : values=on numid=12,iface=CARD,name='Clock Source 4 Validity' ; type=BOOLEAN,access=r-------,values=1 : values=on numid=15,iface=CARD,name='Clock Source 5 Validity' ; type=BOOLEAN,access=r-------,values=1 : values=on numid=19,iface=CARD,name='Clock Source 6 Validity' ; type=BOOLEAN,access=r-------,values=1 : values=on numid=20,iface=CARD,name='Headphone - Output Jack' ; type=BOOLEAN,access=r-------,values=1 : values=off numid=5,iface=CARD,name='Mic - Input Jack' ; type=BOOLEAN,access=r-------,values=1 : values=off numid=9,iface=CARD,name='Mic - Input Jack',index=1 ; type=BOOLEAN,access=r-------,values=1 : values=on numid=16,iface=CARD,name='Speaker - Output Jack' ; type=BOOLEAN,access=r-------,values=1 : values=off numid=13,iface=MIXER,name='PCM Playback Switch' ; type=BOOLEAN,access=rw------,values=1 : values=off numid=17,iface=MIXER,name='PCM Playback Switch',index=1 ; type=BOOLEAN,access=rw------,values=1 : values=off numid=14,iface=MIXER,name='PCM Playback Volume' ; type=INTEGER,access=rw---R--,values=2,min=0,max=87,step=0 : values=62,62 | dBminmax-min=-65.25dB,max=0.00dB numid=18,iface=MIXER,name='PCM Playback Volume',index=1 ; type=INTEGER,access=rw---R--,values=2,min=0,max=87,step=0 : values=62,62 | dBminmax-min=-65.25dB,max=0.00dB numid=6,iface=MIXER,name='Mic Capture Switch' ; type=BOOLEAN,access=rw------,values=1 : values=off numid=10,iface=MIXER,name='Mic Capture Switch',index=1 ; type=BOOLEAN,access=rw------,values=1 : values=on numid=7,iface=MIXER,name='Mic Capture Volume' ; type=INTEGER,access=rw---R--,values=2,min=0,max=39,step=0 : values=39,39 | dBminmax-min=-17.25dB,max=12.00dB numid=11,iface=MIXER,name='Mic Capture Volume',index=1 ; type=INTEGER,access=rw---R--,values=2,min=0,max=39,step=0 : values=39,39 | dBminmax-min=-17.25dB,max=12.00dB numid=1,iface=PCM,name='Capture Channel Map' ; type=INTEGER,access=r--v-R--,values=2,min=0,max=36,step=0 : values=0,0 | container | chmap-fixed=FL,FR numid=3,iface=PCM,name='Playback Channel Map' ; type=INTEGER,access=r--v-R--,values=2,min=0,max=36,step=0 : values=0,0 | container | chmap-fixed=FL,FR numid=2,iface=PCM,name='Capture Channel Map',device=1 ; type=INTEGER,access=r--v-R--,values=2,min=0,max=36,step=0 : values=0,0 | container | chmap-fixed=FL,FR numid=4,iface=PCM,name='Playback Channel Map',device=1 ; type=INTEGER,access=r--v-R--,values=2,min=0,max=36,step=0 : values=0,0 | container | chmap-fixed=FL,FR ``` `arecord -l` ``` **** List of CAPTURE Hardware Devices **** card 2: Audio [USB Audio], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 card 2: Audio [USB Audio], device 1: USB Audio [USB Audio #1] Subdevices: 1/1 Subdevice #0: subdevice #0 ``` `alsaucm -c hw:2 dump text` ``` Verb.HiFi { Comment "HiFi 2.0 channels" Device.Speaker { Comment Speakers Values { CaptureCTL "_ucm0001.hw:Audio" JackControl "Speaker - Output Jack" PlaybackCTL "_ucm0001.hw:Audio" PlaybackChannels 2 PlaybackMixerElem PCM,0 PlaybackPCM "_ucm0001.hw:Audio" PlaybackPriority 200 TQ HiFi } } Device.Headphones { Comment "Front Headphones" Values { CaptureCTL "_ucm0001.hw:Audio" JackControl "Headphone - Output Jack" PlaybackCTL "_ucm0001.hw:Audio" PlaybackMixerElem PCM,1 PlaybackPCM "_ucm0001.hw:Audio,1" PlaybackPriority 300 TQ HiFi } } Device.Mic1 { Comment "Front Microphone" Values { CaptureCTL "_ucm0001.hw:Audio" CaptureMixerElem Mic,0 CapturePCM "_ucm0001.hw:Audio,0" CapturePriority 300 JackControl "Mic - Input Jack" PlaybackCTL "_ucm0001.hw:Audio" TQ HiFi } } } Verb."HiFi 5+1" { Comment "HiFi 5.1 channels" Device.Speaker { Comment Speakers Values { CaptureCTL "_ucm0001.hw:Audio" JackControl "Speaker - Output Jack" PlaybackCTL "_ucm0001.hw:Audio" PlaybackChannels 6 PlaybackMixerElem PCM,0 PlaybackPCM "_ucm0001.hw:Audio" PlaybackPriority 200 TQ HiFi } } Device.Headphones { Comment "Front Headphones" Values { CaptureCTL "_ucm0001.hw:Audio" JackControl "Headphone - Output Jack" PlaybackCTL "_ucm0001.hw:Audio" PlaybackMixerElem PCM,1 PlaybackPCM "_ucm0001.hw:Audio,1" PlaybackPriority 300 TQ HiFi } } Device.Mic1 { Comment "Front Microphone" Values { CaptureCTL "_ucm0001.hw:Audio" CaptureMixerElem Mic,0 CapturePCM "_ucm0001.hw:Audio,0" CapturePriority 300 JackControl "Mic - Input Jack" PlaybackCTL "_ucm0001.hw:Audio" TQ HiFi } } } Verb."HiFi 7+1" { Comment "HiFi 7.1 channels" Device.Speaker { Comment Speakers Values { CaptureCTL "_ucm0001.hw:Audio" JackControl "Speaker - Output Jack" PlaybackCTL "_ucm0001.hw:Audio" PlaybackChannels 8 PlaybackMixerElem PCM,0 PlaybackPCM "_ucm0001.hw:Audio" PlaybackPriority 200 TQ HiFi } } Device.Headphones { Comment "Front Headphones" Values { CaptureCTL "_ucm0001.hw:Audio" JackControl "Headphone - Output Jack" PlaybackCTL "_ucm0001.hw:Audio" PlaybackMixerElem PCM,1 PlaybackPCM "_ucm0001.hw:Audio,1" PlaybackPriority 300 TQ HiFi } } Device.Mic1 { Comment "Front Microphone" Values { CaptureCTL "_ucm0001.hw:Audio" CaptureMixerElem Mic,0 CapturePCM "_ucm0001.hw:Audio,0" CapturePriority 300 JackControl "Mic - Input Jack" PlaybackCTL "_ucm0001.hw:Audio" TQ HiFi } } } ``` Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/528 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

No Micrphone detected on ALC4080 - Gigabyte B850I Aorus Pro (rev 1.0)
by GitHub issues - edited 18 Mar '25

18 Mar '25

alsa-project/alsa-ucm-conf issue #528 was edited from chanster: The Rear Microphone is not detected when I plug in my mic. I don't use the Front Audio connection as my case doesn't have I/O for it, I just used the 2 audio ports in the rear. When I try recording directly I can get the mic to work, but it does not show up in the sound settings. `arecord --duration=3 -D hw:CARD=2,DEV=1 -f dat -vvv a.wav` ``` Hardware PCM card 2 'USB Audio' device 1 subdevice 0 Its setup is: stream : CAPTURE access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 24000 period_size : 6000 period_time : 125000 tstamp_mode : ENABLE tstamp_type : MONOTONIC period_step : 1 avail_min : 6000 period_event : 0 start_threshold : 1 stop_threshold : 24000 silence_threshold: 0 silence_size : 0 boundary : 6755399441055744000 appl_ptr : 0 hw_ptr : 0 Recording WAVE 'a.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo Max peak (12000 samples): 0x00000167 # 1% Max peak (12000 samples): 0x000000da # 0% Max peak (12000 samples): 0x000006b1 ## 5% Max peak (12000 samples): 0x00000547 # 4% Max peak (12000 samples): 0x00000729 ## 5% Max peak (12000 samples): 0x00000a77 ## 8% Max peak (12000 samples): 0x000006d2 ## 5% Max peak (12000 samples): 0x000005ec # 4% Max peak (12000 samples): 0x00000627 # 4% Max peak (12000 samples): 0x00000d32 ### 10% Max peak (12000 samples): 0x00000e9f ### 11% Max peak (12000 samples): 0x00003813 ######### 43% Max peak (12000 samples): 0x00006278 ################ 76% Max peak (12000 samples): 0x00007fff #################### 99% Max peak (12000 samples): 0x00006694 ################# 80% Max peak (12000 samples): 0x000068ea ################# 81% Max peak (12000 samples): 0x00004ba0 ############ 59% Max peak (12000 samples): 0x0000207c ###### 25% Max peak (12000 samples): 0x000017e9 #### 18% Max peak (12000 samples): 0x00003938 ######### 44% Max peak (12000 samples): 0x00001f11 ##### 24% Max peak (12000 samples): 0x00000ee5 ### 11% Max peak (12000 samples): 0x00003944 ######### 44% Max peak (12000 samples): 0x00003942 ######### 44% ``` Motherboard: Gigabyte B850I Aorus Pro (rev 1.0) Audio Device: ALC4080 Distro: Fedora 41 Window Manager: Gnome-Shell Packages: ``` dnf list --installed | grep 'pipewire\|pulseaudio\|alsa' alsa-lib.i686 1.2.13-3.fc41 updates alsa-lib.x86_64 1.2.13-3.fc41 updates alsa-ucm.noarch 1.2.13-3.fc41 updates alsa-ucm-utils.x86_64 1.2.13-2.fc41 updates alsa-utils.x86_64 1.2.13-2.fc41 updates pipewire.i686 1.2.7-7.fc41 updates pipewire.x86_64 1.2.7-7.fc41 updates pipewire-alsa.i686 1.2.7-7.fc41 updates pipewire-alsa.x86_64 1.2.7-7.fc41 updates pipewire-gstreamer.x86_64 1.2.7-7.fc41 updates pipewire-libs.i686 1.2.7-7.fc41 updates pipewire-libs.x86_64 1.2.7-7.fc41 updates pipewire-pulseaudio.x86_64 1.2.7-7.fc41 updates pipewire-utils.x86_64 1.2.7-7.fc41 updates pulseaudio-libs.i686 17.0-2.fc41 fedora pulseaudio-libs.x86_64 17.0-2.fc41 fedora pulseaudio-libs-glib2.x86_64 17.0-2.fc41 fedora wine-alsa.i686 10.1-1.fc41 updates wine-alsa.x86_64 10.1-1.fc41 updates wine-pulseaudio.i686 10.1-1.fc41 updates wine-pulseaudio.x86_64 10.1-1.fc41 updates ``` `lsusb | grep -i audio` ``` Bus 001 Device 002: ID 0414:a014 Giga-Byte Technology Co., Ltd USB Audio ``` `alsactl monitor` (unplugging and plugging mic) ``` node hw:2, #9 (0,0,0,Mic - Input Jack,1) VALUE node hw:2, #9 (0,0,0,Mic - Input Jack,1) VALUE ``` `amixer -c events` (unplugging and plugging mic) ``` event value: numid=9,iface=CARD,name='Mic - Input Jack',index=1 event value: numid=9,iface=CARD,name='Mic - Input Jack',index=1 ``` `amixer -c 2 contents` ``` numid=8,iface=CARD,name='Clock Source 3 Validity' ; type=BOOLEAN,access=r-------,values=1 : values=on numid=12,iface=CARD,name='Clock Source 4 Validity' ; type=BOOLEAN,access=r-------,values=1 : values=on numid=15,iface=CARD,name='Clock Source 5 Validity' ; type=BOOLEAN,access=r-------,values=1 : values=on numid=19,iface=CARD,name='Clock Source 6 Validity' ; type=BOOLEAN,access=r-------,values=1 : values=on numid=20,iface=CARD,name='Headphone - Output Jack' ; type=BOOLEAN,access=r-------,values=1 : values=off numid=5,iface=CARD,name='Mic - Input Jack' ; type=BOOLEAN,access=r-------,values=1 : values=off numid=9,iface=CARD,name='Mic - Input Jack',index=1 ; type=BOOLEAN,access=r-------,values=1 : values=on numid=16,iface=CARD,name='Speaker - Output Jack' ; type=BOOLEAN,access=r-------,values=1 : values=off numid=13,iface=MIXER,name='PCM Playback Switch' ; type=BOOLEAN,access=rw------,values=1 : values=off numid=17,iface=MIXER,name='PCM Playback Switch',index=1 ; type=BOOLEAN,access=rw------,values=1 : values=off numid=14,iface=MIXER,name='PCM Playback Volume' ; type=INTEGER,access=rw---R--,values=2,min=0,max=87,step=0 : values=62,62 | dBminmax-min=-65.25dB,max=0.00dB numid=18,iface=MIXER,name='PCM Playback Volume',index=1 ; type=INTEGER,access=rw---R--,values=2,min=0,max=87,step=0 : values=62,62 | dBminmax-min=-65.25dB,max=0.00dB numid=6,iface=MIXER,name='Mic Capture Switch' ; type=BOOLEAN,access=rw------,values=1 : values=off numid=10,iface=MIXER,name='Mic Capture Switch',index=1 ; type=BOOLEAN,access=rw------,values=1 : values=on numid=7,iface=MIXER,name='Mic Capture Volume' ; type=INTEGER,access=rw---R--,values=2,min=0,max=39,step=0 : values=39,39 | dBminmax-min=-17.25dB,max=12.00dB numid=11,iface=MIXER,name='Mic Capture Volume',index=1 ; type=INTEGER,access=rw---R--,values=2,min=0,max=39,step=0 : values=39,39 | dBminmax-min=-17.25dB,max=12.00dB numid=1,iface=PCM,name='Capture Channel Map' ; type=INTEGER,access=r--v-R--,values=2,min=0,max=36,step=0 : values=0,0 | container | chmap-fixed=FL,FR numid=3,iface=PCM,name='Playback Channel Map' ; type=INTEGER,access=r--v-R--,values=2,min=0,max=36,step=0 : values=0,0 | container | chmap-fixed=FL,FR numid=2,iface=PCM,name='Capture Channel Map',device=1 ; type=INTEGER,access=r--v-R--,values=2,min=0,max=36,step=0 : values=0,0 | container | chmap-fixed=FL,FR numid=4,iface=PCM,name='Playback Channel Map',device=1 ; type=INTEGER,access=r--v-R--,values=2,min=0,max=36,step=0 : values=0,0 | container | chmap-fixed=FL,FR ``` `arecord -l` ``` **** List of CAPTURE Hardware Devices **** card 2: Audio [USB Audio], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 card 2: Audio [USB Audio], device 1: USB Audio [USB Audio #1] Subdevices: 1/1 Subdevice #0: subdevice #0 ``` `alsaucm -c hw:2 dump text` ``` Verb.HiFi { Comment "HiFi 2.0 channels" Device.Speaker { Comment Speakers Values { CaptureCTL "_ucm0001.hw:Audio" JackControl "Speaker - Output Jack" PlaybackCTL "_ucm0001.hw:Audio" PlaybackChannels 2 PlaybackMixerElem PCM,0 PlaybackPCM "_ucm0001.hw:Audio" PlaybackPriority 200 TQ HiFi } } Device.Headphones { Comment "Front Headphones" Values { CaptureCTL "_ucm0001.hw:Audio" JackControl "Headphone - Output Jack" PlaybackCTL "_ucm0001.hw:Audio" PlaybackMixerElem PCM,1 PlaybackPCM "_ucm0001.hw:Audio,1" PlaybackPriority 300 TQ HiFi } } Device.Mic1 { Comment "Front Microphone" Values { CaptureCTL "_ucm0001.hw:Audio" CaptureMixerElem Mic,0 CapturePCM "_ucm0001.hw:Audio,1" CapturePriority 300 JackControl "Mic - Input Jack" PlaybackCTL "_ucm0001.hw:Audio" TQ HiFi } } } Verb."HiFi 5+1" { Comment "HiFi 5.1 channels" Device.Speaker { Comment Speakers Values { CaptureCTL "_ucm0001.hw:Audio" JackControl "Speaker - Output Jack" PlaybackCTL "_ucm0001.hw:Audio" PlaybackChannels 6 PlaybackMixerElem PCM,0 PlaybackPCM "_ucm0001.hw:Audio" PlaybackPriority 200 TQ HiFi } } Device.Headphones { Comment "Front Headphones" Values { CaptureCTL "_ucm0001.hw:Audio" JackControl "Headphone - Output Jack" PlaybackCTL "_ucm0001.hw:Audio" PlaybackMixerElem PCM,1 PlaybackPCM "_ucm0001.hw:Audio,1" PlaybackPriority 300 TQ HiFi } } Device.Mic1 { Comment "Front Microphone" Values { CaptureCTL "_ucm0001.hw:Audio" CaptureMixerElem Mic,0 CapturePCM "_ucm0001.hw:Audio,1" CapturePriority 300 JackControl "Mic - Input Jack" PlaybackCTL "_ucm0001.hw:Audio" TQ HiFi } } } Verb."HiFi 7+1" { Comment "HiFi 7.1 channels" Device.Speaker { Comment Speakers Values { CaptureCTL "_ucm0001.hw:Audio" JackControl "Speaker - Output Jack" PlaybackCTL "_ucm0001.hw:Audio" PlaybackChannels 8 PlaybackMixerElem PCM,0 PlaybackPCM "_ucm0001.hw:Audio" PlaybackPriority 200 TQ HiFi } } Device.Headphones { Comment "Front Headphones" Values { CaptureCTL "_ucm0001.hw:Audio" JackControl "Headphone - Output Jack" PlaybackCTL "_ucm0001.hw:Audio" PlaybackMixerElem PCM,1 PlaybackPCM "_ucm0001.hw:Audio,1" PlaybackPriority 300 TQ HiFi } } Device.Mic1 { Comment "Front Microphone" Values { CaptureCTL "_ucm0001.hw:Audio" CaptureMixerElem Mic,0 CapturePCM "_ucm0001.hw:Audio,1" CapturePriority 300 JackControl "Mic - Input Jack" PlaybackCTL "_ucm0001.hw:Audio" TQ HiFi } } } ``` Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/528 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

No Micrphone detected on ALC4080 - Gigabyte B850I Aorus Pro (rev 1.0)
by GitHub issues - edited 18 Mar '25

18 Mar '25

alsa-project/alsa-ucm-conf issue #528 was edited from chanster: The Rear Microphone is not detected when I plug in my mic. I don't use the Front Audio connection as my case doesn't have I/O for it, I just used the 2 audio ports in the rear. When I try recording directly I can get the mic to work, but it does not show up in the sound settings. `arecord --duration=3 -D hw:CARD=2,DEV=1 -f dat -vvv a.wav` ``` Hardware PCM card 2 'USB Audio' device 1 subdevice 0 Its setup is: stream : CAPTURE access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 24000 period_size : 6000 period_time : 125000 tstamp_mode : ENABLE tstamp_type : MONOTONIC period_step : 1 avail_min : 6000 period_event : 0 start_threshold : 1 stop_threshold : 24000 silence_threshold: 0 silence_size : 0 boundary : 6755399441055744000 appl_ptr : 0 hw_ptr : 0 Recording WAVE 'a.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo Max peak (12000 samples): 0x00000167 # 1% Max peak (12000 samples): 0x000000da # 0% Max peak (12000 samples): 0x000006b1 ## 5% Max peak (12000 samples): 0x00000547 # 4% Max peak (12000 samples): 0x00000729 ## 5% Max peak (12000 samples): 0x00000a77 ## 8% Max peak (12000 samples): 0x000006d2 ## 5% Max peak (12000 samples): 0x000005ec # 4% Max peak (12000 samples): 0x00000627 # 4% Max peak (12000 samples): 0x00000d32 ### 10% Max peak (12000 samples): 0x00000e9f ### 11% Max peak (12000 samples): 0x00003813 ######### 43% Max peak (12000 samples): 0x00006278 ################ 76% Max peak (12000 samples): 0x00007fff #################### 99% Max peak (12000 samples): 0x00006694 ################# 80% Max peak (12000 samples): 0x000068ea ################# 81% Max peak (12000 samples): 0x00004ba0 ############ 59% Max peak (12000 samples): 0x0000207c ###### 25% Max peak (12000 samples): 0x000017e9 #### 18% Max peak (12000 samples): 0x00003938 ######### 44% Max peak (12000 samples): 0x00001f11 ##### 24% Max peak (12000 samples): 0x00000ee5 ### 11% Max peak (12000 samples): 0x00003944 ######### 44% Max peak (12000 samples): 0x00003942 ######### 44% ``` Motherboard: Gigabyte B850I Aorus Pro (rev 1.0) Audio Device: ALC4080 Distro: Fedora 41 Window Manager: Gnome-Shell Packages: ``` dnf list --installed | grep 'pipewire\|pulseaudio\|alsa' alsa-lib.i686 1.2.13-3.fc41 updates alsa-lib.x86_64 1.2.13-3.fc41 updates alsa-ucm.noarch 1.2.13-3.fc41 updates alsa-ucm-utils.x86_64 1.2.13-2.fc41 updates alsa-utils.x86_64 1.2.13-2.fc41 updates pipewire.i686 1.2.7-7.fc41 updates pipewire.x86_64 1.2.7-7.fc41 updates pipewire-alsa.i686 1.2.7-7.fc41 updates pipewire-alsa.x86_64 1.2.7-7.fc41 updates pipewire-gstreamer.x86_64 1.2.7-7.fc41 updates pipewire-libs.i686 1.2.7-7.fc41 updates pipewire-libs.x86_64 1.2.7-7.fc41 updates pipewire-pulseaudio.x86_64 1.2.7-7.fc41 updates pipewire-utils.x86_64 1.2.7-7.fc41 updates pulseaudio-libs.i686 17.0-2.fc41 fedora pulseaudio-libs.x86_64 17.0-2.fc41 fedora pulseaudio-libs-glib2.x86_64 17.0-2.fc41 fedora wine-alsa.i686 10.1-1.fc41 updates wine-alsa.x86_64 10.1-1.fc41 updates wine-pulseaudio.i686 10.1-1.fc41 updates wine-pulseaudio.x86_64 10.1-1.fc41 updates ``` `lsusb | grep -i audio` ``` Bus 001 Device 002: ID 0414:a014 Giga-Byte Technology Co., Ltd USB Audio ``` `alsactl monitor` (unplugging and plugging mic) ``` node hw:2, #9 (0,0,0,Mic - Input Jack,1) VALUE node hw:2, #9 (0,0,0,Mic - Input Jack,1) VALUE ``` `arecord -l` ``` **** List of CAPTURE Hardware Devices **** card 2: Audio [USB Audio], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 card 2: Audio [USB Audio], device 1: USB Audio [USB Audio #1] Subdevices: 1/1 Subdevice #0: subdevice #0 ``` `alsaucm -c hw:2 dump text` ``` Verb.HiFi { Comment "HiFi 2.0 channels" Device.Speaker { Comment Speakers Values { CaptureCTL "_ucm0001.hw:Audio" JackControl "Speaker - Output Jack" PlaybackCTL "_ucm0001.hw:Audio" PlaybackChannels 2 PlaybackMixerElem PCM,0 PlaybackPCM "_ucm0001.hw:Audio" PlaybackPriority 200 TQ HiFi } } Device.Headphones { Comment "Front Headphones" Values { CaptureCTL "_ucm0001.hw:Audio" JackControl "Headphone - Output Jack" PlaybackCTL "_ucm0001.hw:Audio" PlaybackMixerElem PCM,1 PlaybackPCM "_ucm0001.hw:Audio,1" PlaybackPriority 300 TQ HiFi } } Device.Mic1 { Comment "Front Microphone" Values { CaptureCTL "_ucm0001.hw:Audio" CaptureMixerElem Mic,0 CapturePCM "_ucm0001.hw:Audio,1" CapturePriority 300 JackControl "Mic - Input Jack" PlaybackCTL "_ucm0001.hw:Audio" TQ HiFi } } } Verb."HiFi 5+1" { Comment "HiFi 5.1 channels" Device.Speaker { Comment Speakers Values { CaptureCTL "_ucm0001.hw:Audio" JackControl "Speaker - Output Jack" PlaybackCTL "_ucm0001.hw:Audio" PlaybackChannels 6 PlaybackMixerElem PCM,0 PlaybackPCM "_ucm0001.hw:Audio" PlaybackPriority 200 TQ HiFi } } Device.Headphones { Comment "Front Headphones" Values { CaptureCTL "_ucm0001.hw:Audio" JackControl "Headphone - Output Jack" PlaybackCTL "_ucm0001.hw:Audio" PlaybackMixerElem PCM,1 PlaybackPCM "_ucm0001.hw:Audio,1" PlaybackPriority 300 TQ HiFi } } Device.Mic1 { Comment "Front Microphone" Values { CaptureCTL "_ucm0001.hw:Audio" CaptureMixerElem Mic,0 CapturePCM "_ucm0001.hw:Audio,1" CapturePriority 300 JackControl "Mic - Input Jack" PlaybackCTL "_ucm0001.hw:Audio" TQ HiFi } } } Verb."HiFi 7+1" { Comment "HiFi 7.1 channels" Device.Speaker { Comment Speakers Values { CaptureCTL "_ucm0001.hw:Audio" JackControl "Speaker - Output Jack" PlaybackCTL "_ucm0001.hw:Audio" PlaybackChannels 8 PlaybackMixerElem PCM,0 PlaybackPCM "_ucm0001.hw:Audio" PlaybackPriority 200 TQ HiFi } } Device.Headphones { Comment "Front Headphones" Values { CaptureCTL "_ucm0001.hw:Audio" JackControl "Headphone - Output Jack" PlaybackCTL "_ucm0001.hw:Audio" PlaybackMixerElem PCM,1 PlaybackPCM "_ucm0001.hw:Audio,1" PlaybackPriority 300 TQ HiFi } } Device.Mic1 { Comment "Front Microphone" Values { CaptureCTL "_ucm0001.hw:Audio" CaptureMixerElem Mic,0 CapturePCM "_ucm0001.hw:Audio,1" CapturePriority 300 JackControl "Mic - Input Jack" PlaybackCTL "_ucm0001.hw:Audio" TQ HiFi } } } ``` Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/528 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

No Micrphone detected on ALC4080
by GitHub issues - opened 18 Mar '25

18 Mar '25

alsa-project/alsa-ucm-conf issue #528 was opened from chanster: The Rear Microphone is not detected when I plug in my mic. I don't use the Front Audio connection as my case doesn't have I/O for it, I just used the 2 audio ports in the rear. When I try recording directly I can get the mic to work, but it does not show up in the sound settings. `arecord --duration=3 -D hw:CARD=2,DEV=1 -f dat -vvv a.wav` ``` Hardware PCM card 2 'USB Audio' device 1 subdevice 0 Its setup is: stream : CAPTURE access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 24000 period_size : 6000 period_time : 125000 tstamp_mode : ENABLE tstamp_type : MONOTONIC period_step : 1 avail_min : 6000 period_event : 0 start_threshold : 1 stop_threshold : 24000 silence_threshold: 0 silence_size : 0 boundary : 6755399441055744000 appl_ptr : 0 hw_ptr : 0 Recording WAVE 'a.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo Max peak (12000 samples): 0x00000167 # 1% Max peak (12000 samples): 0x000000da # 0% Max peak (12000 samples): 0x000006b1 ## 5% Max peak (12000 samples): 0x00000547 # 4% Max peak (12000 samples): 0x00000729 ## 5% Max peak (12000 samples): 0x00000a77 ## 8% Max peak (12000 samples): 0x000006d2 ## 5% Max peak (12000 samples): 0x000005ec # 4% Max peak (12000 samples): 0x00000627 # 4% Max peak (12000 samples): 0x00000d32 ### 10% Max peak (12000 samples): 0x00000e9f ### 11% Max peak (12000 samples): 0x00003813 ######### 43% Max peak (12000 samples): 0x00006278 ################ 76% Max peak (12000 samples): 0x00007fff #################### 99% Max peak (12000 samples): 0x00006694 ################# 80% Max peak (12000 samples): 0x000068ea ################# 81% Max peak (12000 samples): 0x00004ba0 ############ 59% Max peak (12000 samples): 0x0000207c ###### 25% Max peak (12000 samples): 0x000017e9 #### 18% Max peak (12000 samples): 0x00003938 ######### 44% Max peak (12000 samples): 0x00001f11 ##### 24% Max peak (12000 samples): 0x00000ee5 ### 11% Max peak (12000 samples): 0x00003944 ######### 44% Max peak (12000 samples): 0x00003942 ######### 44% ``` Motherboard: Gigabyte B850I Aorus Pro (rev 1.0) Audio Device: ALC4080 Distro: Fedora 41 Window Manager: Gnome-Shell Packages: ``` dnf list --installed | grep 'pipewire\|pulseaudio\|alsa' alsa-lib.i686 1.2.13-3.fc41 updates alsa-lib.x86_64 1.2.13-3.fc41 updates alsa-ucm.noarch 1.2.13-3.fc41 updates alsa-ucm-utils.x86_64 1.2.13-2.fc41 updates alsa-utils.x86_64 1.2.13-2.fc41 updates pipewire.i686 1.2.7-7.fc41 updates pipewire.x86_64 1.2.7-7.fc41 updates pipewire-alsa.i686 1.2.7-7.fc41 updates pipewire-alsa.x86_64 1.2.7-7.fc41 updates pipewire-gstreamer.x86_64 1.2.7-7.fc41 updates pipewire-libs.i686 1.2.7-7.fc41 updates pipewire-libs.x86_64 1.2.7-7.fc41 updates pipewire-pulseaudio.x86_64 1.2.7-7.fc41 updates pipewire-utils.x86_64 1.2.7-7.fc41 updates pulseaudio-libs.i686 17.0-2.fc41 fedora pulseaudio-libs.x86_64 17.0-2.fc41 fedora pulseaudio-libs-glib2.x86_64 17.0-2.fc41 fedora wine-alsa.i686 10.1-1.fc41 updates wine-alsa.x86_64 10.1-1.fc41 updates wine-pulseaudio.i686 10.1-1.fc41 updates wine-pulseaudio.x86_64 10.1-1.fc41 updates ``` `lsusb | grep -i audio` ``` Bus 001 Device 002: ID 0414:a014 Giga-Byte Technology Co., Ltd USB Audio ``` `alsactl monitor` (unplugging and plugging mic) ``` node hw:2, #9 (0,0,0,Mic - Input Jack,1) VALUE node hw:2, #9 (0,0,0,Mic - Input Jack,1) VALUE ``` `arecord -l` ``` **** List of CAPTURE Hardware Devices **** card 2: Audio [USB Audio], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 card 2: Audio [USB Audio], device 1: USB Audio [USB Audio #1] Subdevices: 1/1 Subdevice #0: subdevice #0 ``` `alsaucm -c hw:2 dump text` ``` Verb.HiFi { Comment "HiFi 2.0 channels" Device.Speaker { Comment Speakers Values { CaptureCTL "_ucm0001.hw:Audio" JackControl "Speaker - Output Jack" PlaybackCTL "_ucm0001.hw:Audio" PlaybackChannels 2 PlaybackMixerElem PCM,0 PlaybackPCM "_ucm0001.hw:Audio" PlaybackPriority 200 TQ HiFi } } Device.Headphones { Comment "Front Headphones" Values { CaptureCTL "_ucm0001.hw:Audio" JackControl "Headphone - Output Jack" PlaybackCTL "_ucm0001.hw:Audio" PlaybackMixerElem PCM,1 PlaybackPCM "_ucm0001.hw:Audio,1" PlaybackPriority 300 TQ HiFi } } Device.Mic1 { Comment "Front Microphone" Values { CaptureCTL "_ucm0001.hw:Audio" CaptureMixerElem Mic,0 CapturePCM "_ucm0001.hw:Audio,1" CapturePriority 300 JackControl "Mic - Input Jack" PlaybackCTL "_ucm0001.hw:Audio" TQ HiFi } } } Verb."HiFi 5+1" { Comment "HiFi 5.1 channels" Device.Speaker { Comment Speakers Values { CaptureCTL "_ucm0001.hw:Audio" JackControl "Speaker - Output Jack" PlaybackCTL "_ucm0001.hw:Audio" PlaybackChannels 6 PlaybackMixerElem PCM,0 PlaybackPCM "_ucm0001.hw:Audio" PlaybackPriority 200 TQ HiFi } } Device.Headphones { Comment "Front Headphones" Values { CaptureCTL "_ucm0001.hw:Audio" JackControl "Headphone - Output Jack" PlaybackCTL "_ucm0001.hw:Audio" PlaybackMixerElem PCM,1 PlaybackPCM "_ucm0001.hw:Audio,1" PlaybackPriority 300 TQ HiFi } } Device.Mic1 { Comment "Front Microphone" Values { CaptureCTL "_ucm0001.hw:Audio" CaptureMixerElem Mic,0 CapturePCM "_ucm0001.hw:Audio,1" CapturePriority 300 JackControl "Mic - Input Jack" PlaybackCTL "_ucm0001.hw:Audio" TQ HiFi } } } Verb."HiFi 7+1" { Comment "HiFi 7.1 channels" Device.Speaker { Comment Speakers Values { CaptureCTL "_ucm0001.hw:Audio" JackControl "Speaker - Output Jack" PlaybackCTL "_ucm0001.hw:Audio" PlaybackChannels 8 PlaybackMixerElem PCM,0 PlaybackPCM "_ucm0001.hw:Audio" PlaybackPriority 200 TQ HiFi } } Device.Headphones { Comment "Front Headphones" Values { CaptureCTL "_ucm0001.hw:Audio" JackControl "Headphone - Output Jack" PlaybackCTL "_ucm0001.hw:Audio" PlaybackMixerElem PCM,1 PlaybackPCM "_ucm0001.hw:Audio,1" PlaybackPriority 300 TQ HiFi } } Device.Mic1 { Comment "Front Microphone" Values { CaptureCTL "_ucm0001.hw:Audio" CaptureMixerElem Mic,0 CapturePCM "_ucm0001.hw:Audio,1" CapturePriority 300 JackControl "Mic - Input Jack" PlaybackCTL "_ucm0001.hw:Audio" TQ HiFi } } } ``` Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/528 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Unable to attach an LED to headphones without HW mute ALSA Control
by GitHub issues - opened 18 Mar '25

18 Mar '25

alsa-project/alsa-ucm-conf issue #527 was opened from stefdb: I have a laptop with a CS42L43 Codec which supplies the Headphone Jack. I'm trying to attach the mute state of the Headphones to the Speaker LED. However, the issue is that the CS42L43 does not possess a Hardware Mute ALSA Control which could be attached to the Speaker LED. I first tried to use the Volume Control to attach to the LED, however, that was a non-starter because the Mute is software controlled, and thus muting headphones did not touch the Volume Control. I then tried to add a new control using cset-new, and link it to the Headphones device ([0001-sof-soundwire-Attach-cs42l43-Headphone-Switch-to-spe.patch](https://gi…), and whilst that worked with the LED, the mute no longer actually muted the Headphone, since there was now a "real" control to set the mute which did not actually mute the Device. Other things I have tried include mapping the Volume control as a Switch, but that does not work, I assume, because the switch control needs to be Boolean. Is there a way to create a new control to track the LED status, whilst also ensuring that the Headphones still mute? Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/527 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

UCM2 Conditional cset in BootSequence
by Giedrius Trainavičius 18 Mar '25

18 Mar '25

Hello, I'm writing a UCM2 .conf file, depending on Device Tree options, some kcontrols may or may not exist - how do I handle this in the BootSequence, so alsactl does not error out if some controls don't exist? I know there's a possible If condition to check if a control exists, but how do I use that to conditionally append to the BootSequence? Thank you! Giedrius.

2 2

[PATCH] ALSA: hda/realtek: Support mute LED on HP Laptop 15s-du3xxx The mute LED on this HP laptop uses ALC236 and requires a quirk to function. This patch enables the existing quirk for the device.
by Dhruv Deshpande 18 Mar '25

18 Mar '25

Signed-off-by: Dhruv Deshpande <dhrv.d(a)proton.me> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a84857a3c2bf..c0b97c29bf89 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10519,6 +10519,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8811, "HP Spectre x360 15-eb1xxx", ALC285_FIXUP_HP_SPECTRE_X360_EB1), SND_PCI_QUIRK(0x103c, 0x8812, "HP Spectre x360 15-eb1xxx", ALC285_FIXUP_HP_SPECTRE_X360_EB1), SND_PCI_QUIRK(0x103c, 0x881d, "HP 250 G8 Notebook PC", ALC236_FIXUP_HP_MUTE_LED_COEFBIT2), + SND_PCI_QUIRK(0x103c, 0x881e, "HP Laptop 15s-du3xxx", ALC236_FIXUP_HP_MUTE_LED_COEFBIT2), SND_PCI_QUIRK(0x103c, 0x8846, "HP EliteBook 850 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8847, "HP EliteBook x360 830 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x884b, "HP EliteBook 840 Aero G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), -- 2.48.1

2 3

[PATCH] ALSA: usb-audio: separate DJM-A9 cap lvl options
by Olivia Mackintosh 18 Mar '25

18 Mar '25

Mixer quicks for the Pioneer DJM-A9 mixer was added in 5289d00 with additional capture level values added to the common DJM array of values. This breaks the existing DJM mixers however as alsa-utils relies on enumeration of the actual mixer options based on the value array which results in error when storing state. This commit just separates the A9 values into a separate array and references them in the corresponding mixer control. Signed-off-by: Olivia Mackintosh <livvy(a)base.nu> --- sound/usb/mixer_quirks.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 79c8c1540ee2..49fb849f7d40 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -3688,8 +3688,7 @@ static const char *snd_djm_get_label(u8 device_idx, u16 wvalue, u16 windex) // common DJM capture level option values static const u16 snd_djm_opts_cap_level[] = { - 0x0000, 0x0100, 0x0200, 0x0300, 0x400, 0x500 }; - + 0x0000, 0x0100, 0x0200, 0x0300 }; // DJM-250MK2 static const u16 snd_djm_opts_250mk2_cap1[] = { @@ -3831,6 +3830,8 @@ static const struct snd_djm_ctl snd_djm_ctls_750mk2[] = { // DJM-A9 +static const u16 snd_djm_opts_a9_cap_level[] = { + 0x0000, 0x0100, 0x0200, 0x0300, 0x0400, 0x0500 }; static const u16 snd_djm_opts_a9_cap1[] = { 0x0107, 0x0108, 0x0109, 0x010a, 0x010e, 0x111, 0x112, 0x113, 0x114, 0x0131, 0x132, 0x133, 0x134 }; @@ -3844,7 +3845,7 @@ static const u16 snd_djm_opts_a9_cap5[] = { 0x0501, 0x0502, 0x0503, 0x0505, 0x0506, 0x0507, 0x0508, 0x0509, 0x050a, 0x050e }; static const struct snd_djm_ctl snd_djm_ctls_a9[] = { - SND_DJM_CTL("Capture Level", cap_level, 0, SND_DJM_WINDEX_CAPLVL), + SND_DJM_CTL("Capture Level", a9_cap_level, 0, SND_DJM_WINDEX_CAPLVL), SND_DJM_CTL("Master Input", a9_cap1, 3, SND_DJM_WINDEX_CAP), SND_DJM_CTL("Ch1 Input", a9_cap2, 2, SND_DJM_WINDEX_CAP), SND_DJM_CTL("Ch2 Input", a9_cap3, 2, SND_DJM_WINDEX_CAP), -- 2.48.1

2 1

ucm2: HDA: HiFi-analog/mic: Refactor the analog mic discovery and pre…
by GitHub pull_request - opened 18 Mar '25

18 Mar '25

alsa-project/alsa-ucm-conf pull request #526 was opened from ujfalusi: …sentation The current mic device creation works on certain machines and fails on others. There are several places of conflicts and setups which can only just fail, but this is mostly not an issue if the user never uses the mic, only the speaker/headset - which, to be honest is what most of us do ;) As an example: The mic selection in most codecs are via enum and it is assumed to be named 'Input Source', which is not always the case as some device uses 'Capture Source' for the control's name. There is also different sets of mics that one can select from: Exhibit A numid=6,iface=MIXER,name='Input Source' ; type=ENUMERATED,access=rw------,values=1,items=2 ; Item #0 'Headset Mic' ; Item #1 'Headphone Mic' : values=1 Exhibit B numid=6,iface=MIXER,name='Capture Source' ; type=ENUMERATED,access=rw------,values=1,items=2 ; Item #0 'Internal Mic' ; Item #1 'Headset Mic' : values=0 Exhibit C numid=6,iface=MIXER,name='Capture Source' ; type=ENUMERATED,access=rw------,values=1,items=3 ; Item #0 'Internal Mic' ; Item #1 'Headset Mic' ; Item #2 'Headphone Mic' : values=0 Exhibit D (this pushes the limits... The patch will ignore item 1) numid=6,iface=MIXER,name='Capture Source' ; type=ENUMERATED,access=rw------,values=1,items=3 ; Item #0 'Internal Mic' ; Item #1 'Internal Mic 1' ; Item #2 'Mic' : values=2 Other issue is that we have this 'Headphone Mic', which turned out to be a 'Stereo Microphone in Headphone Jack', so if it is selected then the Headphone cannot work, they conflict, they use the same rings for different direction and purpose. This patch aims to make the mic discovery a bit more deterministic and pragmatic. But even if the UCM creates the use case profiles correctly, it is still up to UIs (KDE/GNOME/etc) to misunderstand how UCM presents the profiles, what they mean and most of all what 'Mic1', `Mic2', etc is. KDE presents the profiles as they are and user can selct between them to pick the right combination of output and input. GNOME goes further with simplification (and fails with it) and presents 'random' Configuration profiles for Output and Input, plus a device selection and they do work in an interesting way. GNOME also have popup for specifying the type of the plugged accessory, which does not worl at all with UCM profiles. But, this patch is meant for a small step to have clear rules based mic presentation for HDA. The expectation is that what have worked will work as it used to and what did not worked should be detected and presented correctly. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/526 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/526.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH 1/3] ASoC: amd: acp: Fix snd_soc_acpi_mach id's duplicate symbol error
by Venkata Prasad Potturu 17 Mar '25

17 Mar '25

Move snd_soc_acpi_mach id's of all acp platforms form header file to amd-acpi-mach.c file to avoid below errors. ld.lld: error: duplicate symbol: amp_rt1019 ld.lld: error: duplicate symbol: amp_max ld.lld: error: duplicate symbol: snd_soc_acpi_amd_acp63_acp_machines ld.lld: error: duplicate symbol: snd_soc_acpi_amd_acp70_acp_machines ld.lld: error: duplicate symbol: snd_soc_acpi_amd_rmb_acp_machines ld.lld: error: duplicate symbol: snd_soc_acpi_amd_acp_machines Fixes: 6e60db74b69c ("ASoC: amd: acp: Refactor acp machine select") Reported-by: kernel test robot <lkp(a)intel.com> Closes: https://lore.kernel.org/oe-kbuild-all/202503160801.yExt0K2E-lkp@intel.com/ Signed-off-by: Venkata Prasad Potturu <venkataprasad.potturu(a)amd.com> --- sound/soc/amd/acp/Kconfig | 8 +++ sound/soc/amd/acp/Makefile | 2 + sound/soc/amd/acp/acp-pci.c | 8 +-- sound/soc/amd/acp/amd-acpi-mach.c | 93 +++++++++++++++++++++++++++++++ sound/soc/amd/acp/amd.h | 79 ++------------------------ 5 files changed, 111 insertions(+), 79 deletions(-) create mode 100644 sound/soc/amd/acp/amd-acpi-mach.c diff --git a/sound/soc/amd/acp/Kconfig b/sound/soc/amd/acp/Kconfig index 157c124570c8..b9432052c638 100644 --- a/sound/soc/amd/acp/Kconfig +++ b/sound/soc/amd/acp/Kconfig @@ -28,6 +28,9 @@ config SND_SOC_AMD_ACP_LEGACY_COMMON config SND_SOC_AMD_ACP_I2S tristate +config SND_SOC_AMD_ACPI_MACH + tristate + config SND_SOC_AMD_ACP_PCM tristate select SND_SOC_ACPI if ACPI @@ -37,6 +40,7 @@ config SND_SOC_AMD_ACP_PCI depends on X86 && PCI depends on ACPI select SND_SOC_AMD_ACP_LEGACY_COMMON + select SND_SOC_AMD_ACPI_MACH help This options enables generic PCI driver for ACP device. @@ -47,6 +51,7 @@ config SND_AMD_ASOC_RENOIR select SND_SOC_AMD_ACP_I2S select SND_SOC_AMD_ACP_PDM select SND_SOC_AMD_ACP_LEGACY_COMMON + select SND_SOC_AMD_ACPI_MACH depends on X86 && PCI help This option enables Renoir I2S support on AMD platform. @@ -58,6 +63,7 @@ config SND_AMD_ASOC_REMBRANDT select SND_SOC_AMD_ACP_I2S select SND_SOC_AMD_ACP_PDM select SND_SOC_AMD_ACP_LEGACY_COMMON + select SND_SOC_AMD_ACPI_MACH depends on AMD_NODE depends on X86 && PCI help @@ -74,6 +80,7 @@ config SND_AMD_ASOC_ACP63 select SND_SOC_AMD_ACP_I2S select SND_SOC_AMD_ACP_PDM select SND_SOC_AMD_ACP_LEGACY_COMMON + select SND_SOC_AMD_ACPI_MACH help This option enables Acp6.3 I2S support on AMD platform. Say Y if you want to enable AUDIO on ACP6.3 @@ -88,6 +95,7 @@ config SND_AMD_ASOC_ACP70 select SND_SOC_AMD_ACP_I2S select SND_SOC_AMD_ACP_PDM select SND_SOC_AMD_ACP_LEGACY_COMMON + select SND_SOC_AMD_ACPI_MACH help This option enables Acp7.0 PDM support on AMD platform. Say Y if you want to enable AUDIO on ACP7.0 diff --git a/sound/soc/amd/acp/Makefile b/sound/soc/amd/acp/Makefile index 7c75892e678b..08220b9a3802 100644 --- a/sound/soc/amd/acp/Makefile +++ b/sound/soc/amd/acp/Makefile @@ -11,6 +11,7 @@ snd-acp-pdm-y := acp-pdm.o snd-acp-legacy-common-y := acp-legacy-common.o snd-acp-pci-y := acp-pci.o snd-amd-sdw-acpi-y := amd-sdw-acpi.o +snd-amd-acpi-mach-y := amd-acpi-mach.o #platform specific driver snd-acp-renoir-y := acp-renoir.o @@ -32,6 +33,7 @@ obj-$(CONFIG_SND_SOC_AMD_ACP_I2S) += snd-acp-i2s.o obj-$(CONFIG_SND_SOC_AMD_ACP_PDM) += snd-acp-pdm.o obj-$(CONFIG_SND_SOC_AMD_ACP_LEGACY_COMMON) += snd-acp-legacy-common.o obj-$(CONFIG_SND_SOC_AMD_ACP_PCI) += snd-acp-pci.o +obj-$(CONFIG_SND_SOC_AMD_ACPI_MACH) += snd-amd-acpi-mach.o obj-$(CONFIG_SND_AMD_ASOC_RENOIR) += snd-acp-renoir.o obj-$(CONFIG_SND_AMD_ASOC_REMBRANDT) += snd-acp-rembrandt.o diff --git a/sound/soc/amd/acp/acp-pci.c b/sound/soc/amd/acp/acp-pci.c index 9322379cb36f..e15904aabc06 100644 --- a/sound/soc/amd/acp/acp-pci.c +++ b/sound/soc/amd/acp/acp-pci.c @@ -137,26 +137,26 @@ static int acp_pci_probe(struct pci_dev *pci, const struct pci_device_id *pci_id chip->name = "acp_asoc_renoir"; chip->rsrc = &rn_rsrc; chip->acp_hw_ops_init = acp31_hw_ops_init; - chip->machines = snd_soc_acpi_amd_acp_machines; + chip->machines = &snd_soc_acpi_amd_acp_machines; break; case 0x6f: chip->name = "acp_asoc_rembrandt"; chip->rsrc = &rmb_rsrc; chip->acp_hw_ops_init = acp6x_hw_ops_init; - chip->machines = snd_soc_acpi_amd_rmb_acp_machines; + chip->machines = &snd_soc_acpi_amd_rmb_acp_machines; break; case 0x63: chip->name = "acp_asoc_acp63"; chip->rsrc = &acp63_rsrc; chip->acp_hw_ops_init = acp63_hw_ops_init; - chip->machines = snd_soc_acpi_amd_acp63_acp_machines; + chip->machines = &snd_soc_acpi_amd_acp63_acp_machines; break; case 0x70: case 0x71: chip->name = "acp_asoc_acp70"; chip->rsrc = &acp70_rsrc; chip->acp_hw_ops_init = acp70_hw_ops_init; - chip->machines = snd_soc_acpi_amd_acp70_acp_machines; + chip->machines = &snd_soc_acpi_amd_acp70_acp_machines; break; default: dev_err(dev, "Unsupported device revision:0x%x\n", pci->revision); diff --git a/sound/soc/amd/acp/amd-acpi-mach.c b/sound/soc/amd/acp/amd-acpi-mach.c new file mode 100644 index 000000000000..d95047d2ee94 --- /dev/null +++ b/sound/soc/amd/acp/amd-acpi-mach.c @@ -0,0 +1,93 @@ +// SPDX-License-Identifier: GPL-2.0-only +/* + * amd-acpi-match.c - tables and support for ACP platforms + * ACPI enumeration. + * + * Copyright 2025 Advanced Micro Devices, Inc. + */ + +#include <sound/soc-acpi.h> + +struct snd_soc_acpi_codecs amp_rt1019 = { + .num_codecs = 1, + .codecs = {"10EC1019"} +}; + +struct snd_soc_acpi_codecs amp_max = { + .num_codecs = 1, + .codecs = {"MX98360A"} +}; + +struct snd_soc_acpi_mach snd_soc_acpi_amd_acp_machines[] = { + { + .id = "10EC5682", + .drv_name = "acp3xalc56821019", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &amp_rt1019, + }, + { + .id = "RTL5682", + .drv_name = "acp3xalc5682sm98360", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &amp_max, + }, + { + .id = "RTL5682", + .drv_name = "acp3xalc5682s1019", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &amp_rt1019, + }, + { + .id = "AMDI1019", + .drv_name = "renoir-acp", + }, + { + .id = "ESSX8336", + .drv_name = "acp3x-es83xx", + }, + {}, +}; +EXPORT_SYMBOL_NS_GPL(snd_soc_acpi_amd_acp_machines, "SND_SOC_ACP_COMMON"); + +struct snd_soc_acpi_mach snd_soc_acpi_amd_rmb_acp_machines[] = { + { + .id = "10508825", + .drv_name = "rmb-nau8825-max", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &amp_max, + }, + { + .id = "AMDI0007", + .drv_name = "rembrandt-acp", + }, + { + .id = "RTL5682", + .drv_name = "rmb-rt5682s-rt1019", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &amp_rt1019, + }, + {}, +}; +EXPORT_SYMBOL_NS_GPL(snd_soc_acpi_amd_rmb_acp_machines, "SND_SOC_ACP_COMMON"); + +struct snd_soc_acpi_mach snd_soc_acpi_amd_acp63_acp_machines[] = { + { + .id = "AMDI0052", + .drv_name = "acp63-acp", + }, + {}, +}; +EXPORT_SYMBOL_NS_GPL(snd_soc_acpi_amd_acp63_acp_machines, "SND_SOC_ACP_COMMON"); + +struct snd_soc_acpi_mach snd_soc_acpi_amd_acp70_acp_machines[] = { + { + .id = "AMDI0029", + .drv_name = "acp70-acp", + }, + {}, +}; +EXPORT_SYMBOL_NS_GPL(snd_soc_acpi_amd_acp70_acp_machines, "SND_SOC_ACP_COMMON"); + +MODULE_DESCRIPTION("AMD ACP tables and support for ACPI enumeration"); +MODULE_LICENSE("GPL"); +MODULE_AUTHOR("Venkataprasad.potturu(a)amd.com"); diff --git a/sound/soc/amd/acp/amd.h b/sound/soc/amd/acp/amd.h index 796f8efd395c..928aa60ab386 100644 --- a/sound/soc/amd/acp/amd.h +++ b/sound/soc/amd/acp/amd.h @@ -277,81 +277,10 @@ struct acp_resource acp70_rsrc = { .sram_pte_offset = 0x03800000, }; -struct snd_soc_acpi_codecs amp_rt1019 = { - .num_codecs = 1, - .codecs = {"10EC1019"} -}; - -struct snd_soc_acpi_codecs amp_max = { - .num_codecs = 1, - .codecs = {"MX98360A"} -}; - -struct snd_soc_acpi_mach snd_soc_acpi_amd_acp_machines[] = { - { - .id = "10EC5682", - .drv_name = "acp3xalc56821019", - .machine_quirk = snd_soc_acpi_codec_list, - .quirk_data = &amp_rt1019, - }, - { - .id = "RTL5682", - .drv_name = "acp3xalc5682sm98360", - .machine_quirk = snd_soc_acpi_codec_list, - .quirk_data = &amp_max, - }, - { - .id = "RTL5682", - .drv_name = "acp3xalc5682s1019", - .machine_quirk = snd_soc_acpi_codec_list, - .quirk_data = &amp_rt1019, - }, - { - .id = "AMDI1019", - .drv_name = "renoir-acp", - }, - { - .id = "ESSX8336", - .drv_name = "acp3x-es83xx", - }, - {}, -}; - -struct snd_soc_acpi_mach snd_soc_acpi_amd_rmb_acp_machines[] = { - { - .id = "10508825", - .drv_name = "rmb-nau8825-max", - .machine_quirk = snd_soc_acpi_codec_list, - .quirk_data = &amp_max, - }, - { - .id = "AMDI0007", - .drv_name = "rembrandt-acp", - }, - { - .id = "RTL5682", - .drv_name = "rmb-rt5682s-rt1019", - .machine_quirk = snd_soc_acpi_codec_list, - .quirk_data = &amp_rt1019, - }, - {}, -}; - -struct snd_soc_acpi_mach snd_soc_acpi_amd_acp63_acp_machines[] = { - { - .id = "AMDI0052", - .drv_name = "acp63-acp", - }, - {}, -}; - -struct snd_soc_acpi_mach snd_soc_acpi_amd_acp70_acp_machines[] = { - { - .id = "AMDI0029", - .drv_name = "acp70-acp", - }, - {}, -}; +extern struct snd_soc_acpi_mach snd_soc_acpi_amd_acp_machines; +extern struct snd_soc_acpi_mach snd_soc_acpi_amd_rmb_acp_machines; +extern struct snd_soc_acpi_mach snd_soc_acpi_amd_acp63_acp_machines; +extern struct snd_soc_acpi_mach snd_soc_acpi_amd_acp70_acp_machines; extern const struct snd_soc_dai_ops asoc_acp_cpu_dai_ops; extern const struct snd_soc_dai_ops acp_dmic_dai_ops; -- 2.39.2

2 3

[PATCH v2 00/14] Implement acp_common_hw_ops support for all acp platforms
by Venkata Prasad Potturu 17 Mar '25

17 Mar '25

This patch series is to implement acp_common_hw_ops support and refactor interrupt handler registration and interrupt handling. Venkata Prasad Potturu (14): ASoC: amd: acp: Remove redundant acp70 chip->name ASoC: amd: acp: Implement acp_common_hw_ops support for acp platforms ASoC: amd: acp: Refactor dmic-codec platform device creation ASoC: amd: acp: Refactor acp platform device creation ASoC: amd: acp: Refactor acp machine select ASoC: amd: acp: Add new interrupt handle callbacks in acp_common_hw_ops ASoC: amd: acp: Remove redundant acp_dev_data structure ASoC: amd: acp: Move spin_lock and list initialization to acp-pci driver ASoC: amd: acp: Remove white line ASoC: amd: acp: Refactor acp70 platform resource structure ASoC: amd: acp: Refactor acp63 platform resource structure ASoC: amd: acp: Refactor rembrant platform resource structure ASoC: amd: acp: Refactor renoir platform resource structure ASoC: amd: acp: Fix for enabling DMIC on acp platforms via _DSD entry sound/soc/amd/acp/acp-i2s.c | 185 +++++++++--------- sound/soc/amd/acp/acp-legacy-common.c | 254 ++++++++++++++++++------ sound/soc/amd/acp/acp-pci.c | 210 ++++++++++++-------- sound/soc/amd/acp/acp-pdm.c | 53 +++-- sound/soc/amd/acp/acp-platform.c | 145 ++++---------- sound/soc/amd/acp/acp-rembrandt.c | 111 +++-------- sound/soc/amd/acp/acp-renoir.c | 110 ++--------- sound/soc/amd/acp/acp63.c | 94 +++------ sound/soc/amd/acp/acp70.c | 84 +++----- sound/soc/amd/acp/amd.h | 270 +++++++++++++++++++++----- sound/soc/amd/acp/chip_offset_byte.h | 12 +- 11 files changed, 792 insertions(+), 736 deletions(-) -- 2.39.2

4 17

sof-soundwire: rt1318: add playback control switch
by GitHub pull_request - opened 17 Mar '25

17 Mar '25

alsa-project/alsa-ucm-conf pull request #525 was opened from shumingfan: rt1318: add playback control switch Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/525 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/525.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

iecset.1: Some remarks and a patch with editorial changes for this man page
by Bjarni Ingi Gislason 15 Mar '25

15 Mar '25

Package: alsa-utils Version: 1.2.13-1 Severity: minor Tags: patch * What led up to the situation? Checking for defects with a new version test-[g|n]roff -mandoc -t -K utf8 -rF0 -rHY=0 -rCHECKSTYLE=10 -ww -z < "man page" [Use "groff -e ' $' -e '\\~$' <file>" to find obvious trailing spaces.] ["test-groff" is a script in the repository for "groff"; is not shipped] (local copy and "troff" slightly changed by me). [The fate of "test-nroff" was decided in groff bug #55941.] * What was the outcome of this action? an.tmac:<stdin>:1: style: .TH missing fourth argument; consider package/project name and version (e.g., "groff 1.23.0") an.tmac:<stdin>:24: misuse, warning: .BI is for at least 2 arguments, got 1 Use macro '.B' for one argument or split argument. an.tmac:<stdin>:28: misuse, warning: .BI is for at least 2 arguments, got 1 Use macro '.B' for one argument or split argument. an.tmac:<stdin>:33: misuse, warning: .BI is for at least 2 arguments, got 1 Use macro '.B' for one argument or split argument. * What outcome did you expect instead? No output (no warnings). -.- General remarks and further material, if a diff-file exist, are in the attachments. -- System Information: Debian Release: trixie/sid APT prefers testing APT policy: (500, 'testing') Architecture: amd64 (x86_64) Kernel: Linux 6.12.17-amd64 (SMP w/2 CPU threads; PREEMPT) Locale: LANG=is_IS.iso88591, LC_CTYPE=is_IS.iso88591 (charmap=ISO-8859-1), LANGUAGE not set Shell: /bin/sh linked to /usr/bin/dash Init: sysvinit (via /sbin/init) Versions of packages alsa-utils depends on: ii kmod 34-3 ii libasound2t64 1.2.13-1+b1 ii libatopology2t64 1.2.13-1+b1 ii libc6 2.41-4 ii libfftw3-single3 3.3.10-2+b1 ii libncursesw6 6.5+20250216-2 ii libsamplerate0 0.2.2-4+b2 ii libtinfo6 6.5+20250216-2 alsa-utils recommends no packages. Versions of packages alsa-utils suggests: pn dialog <none> -- no debconf information

1 0

DRAFT [for alsa-lib-1.2.14]: USB-Audio: GoXLR: enable detection of beta firmware (25 channels)
by GitHub pull_request - opened 15 Mar '25

15 Mar '25

alsa-project/alsa-ucm-conf pull request #524 was opened from perexg: 'Syntax 8' is required (alsa-lib 1.2.14+) Fixes: 17f9b4f ("USB-Audio: GoXLR - fix the channel detection for mini, cleanups") Link: https://github.com/alsa-project/alsa-ucm-conf/issues/444 Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/524 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/524.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

GoXLR: Add 'Broadcast Stream Mix 2' to Capture if channels are available
by GitHub pull_request - opened 15 Mar '25

15 Mar '25

alsa-project/alsa-ucm-conf pull request #523 was opened from FrostyCoolSlug: Fixes: #444 Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/523 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/523.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

pyalsa.alsamixer.Element.get_volume_dB returns error status instead of actual value
by GitHub issues - opened 15 Mar '25

15 Mar '25

alsa-project/alsa-python issue #16 was opened from tespent: I am writing a status collector for my PC, and I noticed that `pyalsa.alsamixer.Element.get_volume_dB` always returns 0 in my test, which lead to invalid result for the collected status. After digging into code I notice that the following code returns `res` instead of `val` as the result: https://github.com/alsa-project/alsa-python/blob/f8f9260282eb9c97f53e4689e0… I am able to workaround this issue by using `elem.ask_volume_dB(elem.get_volume())` but this looks silly. Btw, I'm willing to contribute and subscribed to alsa-devel mailing list but I didn't figure out exactly how. Can someone tell me? Issue URL : https://github.com/alsa-project/alsa-python/issues/16 Repository URL: https://github.com/alsa-project/alsa-python

1 0

ucm2: Add support for Lenovo Yoga 7x and T14s
by GitHub pull_request - opened 14 Mar '25

14 Mar '25

alsa-project/alsa-ucm-conf pull request #522 was opened from Srinivas-Kandagatla: This patch series adds support for Lenovo T14s and Yoga Slim 7x based on Qualcomm x1e80100 SoC. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/522 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/522.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

USB-Audio: GoXLR - fix the channel detection for mini, cleanups
by GitHub pull_request - opened 14 Mar '25

14 Mar '25

alsa-project/alsa-ucm-conf pull request #521 was opened from perexg: Note that detection for 25 channels requires commit [1] (alsa-lib 1.2.14+). [1] https://github.com/alsa-project/alsa-lib/commit/7fbd47ce797939862025ef42486… BugLink: https://github.com/alsa-project/alsa-ucm-conf/issues/444 Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/521 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/521.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

FFADO Save / Load for M-Audio
by Nick 14 Mar '25

14 Mar '25

Hello there, Is Save / Load implemented for FFADO - Mixer at this time? This is pertaining to M-Audio ProFire (610 and 2626). thank you! Nick

1 0

Split the GoXLR configuration into Full Device and Mini
by GitHub pull_request - opened 13 Mar '25

13 Mar '25

alsa-project/alsa-ucm-conf pull request #520 was opened from FrostyCoolSlug: This commit splits the GoXLR and GoXLR Mini into separate configurations. The GoXLR Mini will always (at this point) have 21 channels defined, where as the full sized GoXLR will have either 23 or 25 depending on Firmware (23 is FAR more common at this point, as the firmware with 25 channels is still in beta). Separating out the two devices should at least partially solve the recent issue with Pipewire. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/520 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/520.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Need to edit Steinberg configuration files
by GitHub issues - opened 13 Mar '25

13 Mar '25

alsa-project/alsa-ucm-conf issue #519 was opened from AdrianusWest: I faced a problem with disappearance of HiFi profile in Steinberg UR22C audio card settings. Attempting to fix it myself did not lead to anything. I addressed the issue to pipewire developers, who pointed out that the existing configuration file needs to be corrected: "The problem here is that previously, the channel splitting was done with ALSA dshare plugin which as it turns out does not require the exact channel numbers to match. As this introduces additional latency, in 1.4.0 the splitting is now done on Pipewire side. But of course there was a nasty surprise (and affects multiple devices), so probably there will be 1.4.1 with a workaround for wrong channel numbers in profiles." The link to the thread is attached: https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/4609 Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/519 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Dynamic PCM switching in ALSA with multi room audio sync support
by GitHub issues - opened 13 Mar '25

13 Mar '25

alsa-project/alsa-lib issue #443 was opened from Asikbct: **Challenges:** The PCM device is already opened by applications, making it difficult to switch to a new output seamlessly. Since most Music Service SDKs initialize the PCM device at startup, re-initializing it would require users to manually reopen their streaming app and re-select the device, negatively impacting the user experience. **Requirement:** User can switch between different audio output options (Line Out, Optical, USB Out, Bluetooth, etc.) at any time. The transition to a new output should be seamless, whether the user is currently playing a song from any music sources. Playback should resume on the newly selected output without requiring the user to manually reopen the content provider app and restart playback from the beginning. **Problems:** 1. **Reopening the PCM** Most music services open and configure the PCM device during initialization. Changing the PCM device (e.g., switching from Line Out to Optical Out) requires reopening it, which forces music services to reinitialize. This may require the end user to relaunch the respective app, reselect the device, and restart playback, impacting the user experience. 2. **Multiroom Sync** Many music services that support multiroom synchronization rely on Linux ALSA APIs like snd_pcm_delay() to estimate the kernel buffer fill level and synchronize audio playback across devices. Any new approach to meet the requirements, should not disrupt this synchronization. I've tried virtual loopback, PulseAudio, and a customized dswitch plugin. While all these methods enable dynamic PCM switching, I couldn't achieve stable long-term latency on the same PCM. Any guidance or suggestions on achieving stable long-term latency along with dynamic PCM switching would be greatly appreciated. Issue URL : https://github.com/alsa-project/alsa-lib/issues/443 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[GIT PULL] ASoC fixes for v6.14-rc6
by Mark Brown 13 Mar '25

13 Mar '25

The following changes since commit 7eb172143d5508b4da468ed59ee857c6e5e01da6: Linux 6.14-rc5 (2025-03-02 11:48:20 -0800) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.14-rc6 for you to fetch changes up to 658fb7fe8e7f4014ea17a4da0e0c1d9bc319fa35: ASoC: cs42l43: convert to SYSTEM_SLEEP_PM_OPS (2025-03-12 16:12:24 +0000) ---------------------------------------------------------------- ASoC: Fixes for v6.14 The bulk of this is driver specific fixes, mostly unremarkable. There's also one core fix from Charles, fixing up confusion around the limiting of maximum control values. ---------------------------------------------------------------- Alexey Klimov (1): ASoC: codecs: wsa884x: report temps to hwmon in millidegree of Celsius Arnd Bergmann (1): ASoC: cs42l43: convert to SYSTEM_SLEEP_PM_OPS Bard Liao (2): ASoC: rt1320: set wake_capable = 0 explicitly ASoC: rt722-sdca: add missing readable registers Charles Keepax (2): ASoC: ops: Consistently treat platform_max as control value ASoC: cs42l43: Fix maximum ADC Volume Christophe JAILLET (1): ASoC: codecs: wm0010: Fix error handling path in wm0010_spi_probe() Maciej Strozek (1): ASoC: cs42l43: Add jack delay debounce after suspend Peter Ujfalusi (1): ASoC: Intel: sof_sdw: Fix unlikely uninitialized variable use in create_sdw_dailinks() Thomas Mizrahi (1): ASoC: amd: yc: Support mic on another Lenovo ThinkPad E16 Gen 2 model Thorsten Blum (1): ASoC: tegra: Fix ADX S24_LE audio format include/sound/soc.h | 5 ++++- sound/soc/amd/yc/acp6x-mach.c | 7 +++++++ sound/soc/codecs/cs42l43-jack.c | 13 ++++++++++--- sound/soc/codecs/cs42l43.c | 17 +++++++++++++++-- sound/soc/codecs/cs42l43.h | 3 +++ sound/soc/codecs/rt1320-sdw.c | 3 +++ sound/soc/codecs/rt722-sdca-sdw.c | 4 ++++ sound/soc/codecs/wm0010.c | 13 +++++++++++-- sound/soc/codecs/wsa884x.c | 4 ++-- sound/soc/intel/boards/sof_sdw.c | 2 +- sound/soc/soc-ops.c | 15 +++++++-------- sound/soc/tegra/tegra210_adx.c | 4 ++-- 12 files changed, 69 insertions(+), 21 deletions(-)

2 1

Channel count in GoXLR SplitPCM configuration
by GitHub issues - edited 12 Mar '25

12 Mar '25

alsa-project/alsa-ucm-conf issue #518 was edited from pv: Similar issue as #515, and related to #444: We have a report https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/4602 that GoXLR UCM profile with SplitPCM=1 specifies always 23 channels, but the device can't be opened with exactly 23 channels. According to https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/4602#note_2819696 the device may have 21, 23, or 25 depending on firmware/etc details. In Pipewire, I think we add a workaround that just uses the available channel count even if it is different from what UCM profile specifies, but logs warnings if the counts are different so these issues in the profiles can be caught. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/518 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Channel count in GoXLR SplitPCM configuration
by GitHub issues - opened 12 Mar '25

12 Mar '25

alsa-project/alsa-ucm-conf issue #518 was opened from pv: Similar issue as #515, and related to #444: We have a report https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/4602#note_2819696 that GoXLR UCM profile with SplitPCM=1 specifies always 23 channels, but the device can't be opened with exactly 23 channels. According to https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/4602#note_2819696 the device may have 21, 23, or 25 depending on firmware/etc details. In Pipewire, I think we add a workaround that just uses the available channel count even if it is different from what UCM profile specifies, but logs warnings if the counts are different so these issues in the profiles can be caught. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/518 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[GIT PULL] ASoC fixes for v6.14-rc6
by Mark Brown 12 Mar '25

12 Mar '25

The following changes since commit 7eb172143d5508b4da468ed59ee857c6e5e01da6: Linux 6.14-rc5 (2025-03-02 11:48:20 -0800) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.14-rc6 for you to fetch changes up to ed92bc5264c4357d4fca292c769ea9967cd3d3b6: ASoC: codecs: wm0010: Fix error handling path in wm0010_spi_probe() (2025-03-11 13:51:52 +0000) ---------------------------------------------------------------- ASoC: Fixes for v6.14 The bulk of this is driver specific fixes, mostly unremarkable. There's also one core fix from Charles, fixing up confusion around the limiting of maximum control values. ---------------------------------------------------------------- Alexey Klimov (1): ASoC: codecs: wsa884x: report temps to hwmon in millidegree of Celsius Bard Liao (2): ASoC: rt1320: set wake_capable = 0 explicitly ASoC: rt722-sdca: add missing readable registers Charles Keepax (2): ASoC: ops: Consistently treat platform_max as control value ASoC: cs42l43: Fix maximum ADC Volume Christophe JAILLET (1): ASoC: codecs: wm0010: Fix error handling path in wm0010_spi_probe() Maciej Strozek (1): ASoC: cs42l43: Add jack delay debounce after suspend Peter Ujfalusi (1): ASoC: Intel: sof_sdw: Fix unlikely uninitialized variable use in create_sdw_dailinks() Thomas Mizrahi (1): ASoC: amd: yc: Support mic on another Lenovo ThinkPad E16 Gen 2 model Thorsten Blum (1): ASoC: tegra: Fix ADX S24_LE audio format include/sound/soc.h | 5 ++++- sound/soc/amd/yc/acp6x-mach.c | 7 +++++++ sound/soc/codecs/cs42l43-jack.c | 13 ++++++++++--- sound/soc/codecs/cs42l43.c | 17 +++++++++++++++-- sound/soc/codecs/cs42l43.h | 3 +++ sound/soc/codecs/rt1320-sdw.c | 3 +++ sound/soc/codecs/rt722-sdca-sdw.c | 4 ++++ sound/soc/codecs/wm0010.c | 13 +++++++++++-- sound/soc/codecs/wsa884x.c | 4 ++-- sound/soc/intel/boards/sof_sdw.c | 2 +- sound/soc/soc-ops.c | 15 +++++++-------- sound/soc/tegra/tegra210_adx.c | 4 ++-- 12 files changed, 69 insertions(+), 21 deletions(-)

2 1

[PATCH 5.10 1/1] ASoC: ops: Check for negative values before reading them
by d.privalov 12 Mar '25

12 Mar '25

From: Mark Brown <broonie(a)kernel.org> commit 1601033da2dd2052e0489137f7788a46a8fcd82f upstream. The controls allow inputs to be specified as negative but our manipulating them into register fields need to be done on unsigned variables so the checks for negative numbers weren't taking effect properly. Do the checks for negative values on the variable in the ABI struct rather than on our local unsigned copy. Signed-off-by: Mark Brown <broonie(a)kernel.org> Link: https://lore.kernel.org/r/20220128192443.3504823-1-broonie@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Dmitriy Privalov <d.privalov(a)omp.ru> --- sound/soc/soc-ops.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index a83cd8d8a9633..a1087dfee532d 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -316,26 +316,26 @@ int snd_soc_put_volsw(struct snd_kcontrol *kcontrol, if (sign_bit) mask = BIT(sign_bit + 1) - 1; + if (ucontrol->value.integer.value[0] < 0) + return -EINVAL; val = ucontrol->value.integer.value[0]; if (mc->platform_max && ((int)val + min) > mc->platform_max) return -EINVAL; if (val > max - min) return -EINVAL; - if (val < 0) - return -EINVAL; val = (val + min) & mask; if (invert) val = max - val; val_mask = mask << shift; val = val << shift; if (snd_soc_volsw_is_stereo(mc)) { + if (ucontrol->value.integer.value[1] < 0) + return -EINVAL; val2 = ucontrol->value.integer.value[1]; if (mc->platform_max && ((int)val2 + min) > mc->platform_max) return -EINVAL; if (val2 > max - min) return -EINVAL; - if (val2 < 0) - return -EINVAL; val2 = (val2 + min) & mask; if (invert) val2 = max - val2; @@ -429,13 +429,13 @@ int snd_soc_put_volsw_sx(struct snd_kcontrol *kcontrol, int err = 0; unsigned int val, val_mask, val2 = 0; + if (ucontrol->value.integer.value[0] < 0) + return -EINVAL; val = ucontrol->value.integer.value[0]; if (mc->platform_max && val > mc->platform_max) return -EINVAL; if (val > max) return -EINVAL; - if (val < 0) - return -EINVAL; val_mask = mask << shift; val = (val + min) & mask; val = val << shift; -- 2.34.1

1 0

Add SetLED macro to hide implementation details for LED handling
by GitHub pull_request - opened 11 Mar '25

11 Mar '25

alsa-project/alsa-ucm-conf pull request #517 was opened from perexg: This change also reduces modprobe calls when settings is split to multiple configuration files. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/517 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/517.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

USB-Audio: Improve support for Focusrite 4th Gen devices
by GitHub pull_request - opened 10 Mar '25

10 Mar '25

alsa-project/alsa-ucm-conf pull request #516 was opened from perexg: There are 4 capture channels. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/516 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/516.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH 00/14] Implement acp_common_hw_ops support for all platforms
by Venkata Prasad Potturu 10 Mar '25

10 Mar '25

This patch series is to implement acp_common_hw_ops support and refactor interrupt handler registration and interrupt handling. Venkata Prasad Potturu (14): ASoC: amd: acp: Remove redundant acp70 chip->name ASoC: amd: acp: Implement acp_common_hw_ops support for acp platforms ASoC: amd: acp: Refactor dmic-codec platform device creation ASoC: amd: acp: Refactor acp platform device creation ASoC: amd: acp: Refactor acp machine select ASoC: amd: acp: Add new interrupt handle callbacks in acp_common_hw_ops ASoC: amd: acp: Remove redundant acp_dev_data structure ASoC: amd: acp: Move spin_lock and list initialization to acp-pci driver ASoC: amd: acp: Remove white line ASoC: amd: acp: Refactor acp70 platform resource structure ASoC: amd: acp: Refactor acp63 platform resource structure ASoC: amd: acp: Refactor rembrant platform resource structure ASoC: amd: acp: Refactor renoir platform resource structure ASoC: amd: acp: Fix for enabling DMIC on acp platforms via _DSD entry sound/soc/amd/acp/acp-i2s.c | 185 +++++++++--------- sound/soc/amd/acp/acp-legacy-common.c | 254 ++++++++++++++++++------ sound/soc/amd/acp/acp-pci.c | 210 ++++++++++++-------- sound/soc/amd/acp/acp-pdm.c | 53 +++-- sound/soc/amd/acp/acp-platform.c | 145 ++++---------- sound/soc/amd/acp/acp-rembrandt.c | 111 +++-------- sound/soc/amd/acp/acp-renoir.c | 110 ++--------- sound/soc/amd/acp/acp63.c | 94 +++------ sound/soc/amd/acp/acp70.c | 84 +++----- sound/soc/amd/acp/amd.h | 270 +++++++++++++++++++++----- sound/soc/amd/acp/chip_offset_byte.h | 12 +- 11 files changed, 792 insertions(+), 736 deletions(-) -- 2.39.2

3 18

[PATCH alsa-ucm-conf] sof-soundwire: Split cs42l43 dmic initialisation
by Charles Keepax 10 Mar '25

10 Mar '25

Split the initialisation of cs42l43 into a headset part and a dmic part. This will prevent the dmic mute controls getting attached to the mute LED when the codec dmics are not being used. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- ucm2/codecs/cs42l43-dmic/init.conf | 23 +++++++++++++++++++++++ ucm2/codecs/cs42l43/init.conf | 11 +---------- ucm2/sof-soundwire/sof-soundwire.conf | 2 +- 3 files changed, 25 insertions(+), 11 deletions(-) create mode 100644 ucm2/codecs/cs42l43-dmic/init.conf diff --git a/ucm2/codecs/cs42l43-dmic/init.conf b/ucm2/codecs/cs42l43-dmic/init.conf new file mode 100644 index 0000000..441d082 --- /dev/null +++ b/ucm2/codecs/cs42l43-dmic/init.conf @@ -0,0 +1,23 @@ +# cs42l43 specific control settings + +LibraryConfig.remap.Config { + ctl.default.map { + "name='cs42l43 Microphone Capture Switch'" { + "name='cs42l43 Decimator 3 Switch'".vindex.0 0 + "name='cs42l43 Decimator 4 Switch'".vindex.1 0 + } + "name='cs42l43 Microphone Capture Volume'" { + "name='cs42l43 Decimator 3 Volume'".vindex.0 0 + "name='cs42l43 Decimator 4 Volume'".vindex.1 0 + } + } +} + +FixedBootSequence [ + cset "name='cs42l43 Decimator 3 Switch' 0" + cset "name='cs42l43 Decimator 4 Switch' 0" + exec "-/sbin/modprobe snd_ctl_led" + sysw "-/class/sound/ctl-led/mic/card${CardNumber}/attach:cs42l43 Decimator 3 Switch" + sysw "-/class/sound/ctl-led/mic/card${CardNumber}/attach:cs42l43 Decimator 4 Switch" +] + diff --git a/ucm2/codecs/cs42l43/init.conf b/ucm2/codecs/cs42l43/init.conf index 0fbb517..8e41a0f 100644 --- a/ucm2/codecs/cs42l43/init.conf +++ b/ucm2/codecs/cs42l43/init.conf @@ -2,14 +2,6 @@ LibraryConfig.remap.Config { ctl.default.map { - "name='cs42l43 Microphone Capture Switch'" { - "name='cs42l43 Decimator 3 Switch'".vindex.0 0 - "name='cs42l43 Decimator 4 Switch'".vindex.1 0 - } - "name='cs42l43 Microphone Capture Volume'" { - "name='cs42l43 Decimator 3 Volume'".vindex.0 0 - "name='cs42l43 Decimator 4 Volume'".vindex.1 0 - } "name='cs42l43 Headset Microphone Capture Switch'" { "name='cs42l43 Decimator 1 Switch'".vindex.0 0 } @@ -20,8 +12,7 @@ LibraryConfig.remap.Config { } FixedBootSequence [ + cset "name='cs42l43 Decimator 1 Switch' 0" exec "-/sbin/modprobe snd_ctl_led" sysw "-/class/sound/ctl-led/mic/card${CardNumber}/attach:cs42l43 Decimator 1 Switch" - sysw "-/class/sound/ctl-led/mic/card${CardNumber}/attach:cs42l43 Decimator 3 Switch" - sysw "-/class/sound/ctl-led/mic/card${CardNumber}/attach:cs42l43 Decimator 4 Switch" ] diff --git a/ucm2/sof-soundwire/sof-soundwire.conf b/ucm2/sof-soundwire/sof-soundwire.conf index 96d2966..57e5610 100644 --- a/ucm2/sof-soundwire/sof-soundwire.conf +++ b/ucm2/sof-soundwire/sof-soundwire.conf @@ -114,7 +114,7 @@ If.hs_init { If.mic_init { Condition { Type RegexMatch - Regex "(rt712-dmic|rt713-dmic|rt715(-sdca)?)" + Regex "(cs42l43-dmic|rt712-dmic|rt713-dmic|rt715(-sdca)?)" String "${var:MicCodec1}" } True.Include.mic_init.File "/codecs/${var:MicCodec1}/init.conf" -- 2.39.5

2 1

Channel count in Focusrite Scarlett Gen 4 (1235:8218) SplitPCM configuration
by GitHub issues - edited 09 Mar '25

09 Mar '25

alsa-project/alsa-ucm-conf issue #515 was edited from pv: The configuration https://github.com/alsa-project/alsa-ucm-conf/blob/master/ucm2/USB-Audio/Fo… specifies `HWChannels 2` for the Mic1 and Mic2 devices. This gets translated to `CaptureChannels 2` for `alsaucm open '<<<SplitPCM=1>>>hw:2'`. However, we've got a report https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/4597#note_2815906 that looks like snd_pcm_hw_params_set_channels(pcm_handle, hwparams, 2) for the CapturePCM "_ucm0001.hw:Gen,0" fails with error. The PCM can be opened with 4 channels, though. Without `SplitPCM=1`, it also probably works but I'm not sure what channel count the `dshare` plugin is using. The issue also affects the `Direct` profile, which also specifies `CaptureChannels 2`. Is the `HWChannels 2` for Mic1/2 input direction right? Shall `CaptureChannels 2` in UCM device be interpreted as that the device can be opened with exactly 2 channels, or that it can be opened with some number of channels >= 2? <details> ``` $ alsaucm open '<<<SplitPCM=1>>>hw:2' dump text Verb.HiFi { Comment Default Device.Line1 { Comment "Headphones / Line 1-2" Values { CaptureCTL "_ucm0001.hw:Gen" PlaybackCTL "_ucm0001.hw:Gen" PlaybackPCM "_ucm0001.hw:Gen" PlaybackPriority 200 } } Device.Mic1 { Comment "Input 1 Mic/Inst/Line" Values { CaptureCTL "_ucm0001.hw:Gen" CaptureChannel0 0 CaptureChannelPos0 MONO CaptureChannels 2 CapturePCM "_ucm0001.hw:Gen,0" CapturePriority 200 PlaybackCTL "_ucm0001.hw:Gen" } } Device.Mic2 { Comment "Input 2 Mic/Inst/Line" Values { CaptureCTL "_ucm0001.hw:Gen" CaptureChannel0 1 CaptureChannelPos0 MONO CaptureChannels 2 CapturePCM "_ucm0001.hw:Gen,0" CapturePriority 100 PlaybackCTL "_ucm0001.hw:Gen" } } } Verb.Direct { Comment "Direct Scarlett Solo 4th Gen" Device.Direct { Comment "Direct Scarlett Solo 4th Gen" Values { CaptureCTL "_ucm0001.hw:Gen" CaptureChannels 2 CapturePCM "_ucm0001.hw:Gen" CapturePriority 1000 PlaybackCTL "_ucm0001.hw:Gen" PlaybackChannels 2 PlaybackPCM "_ucm0001.hw:Gen" PlaybackPriority 1000 } } } ``` </details> Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/515 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Channel count in Focusrite Scarlett Gen 4 (1235:8218) SplitPCM configuration
by GitHub issues - opened 09 Mar '25

09 Mar '25

alsa-project/alsa-ucm-conf issue #515 was opened from pv: The configuration https://github.com/alsa-project/alsa-ucm-conf/blob/master/ucm2/USB-Audio/Fo… specifies `HWChannels 2` for the Mic1 and Mic2 devices. This gets translated to `CaptureChannels 2` for `alsaucm open '<<<SplitPCM=1>>>hw:2'`. However, we've got a report https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/4597#note_2815906 that looks like snd_pcm_hw_params_set_channels(pcm_handle, hwparams, 2) for the CapturePCM "_ucm0001.hw:Gen,0" fails with error. The PCM can be opened with 4 channels, though. Without `SplitPCM=1`, it also probably works but I'm not sure what channel count the `dshare` plugin is using. Is the `HWChannels 2` for Mic1/2 input direction right? Shall `CaptureChannels 2` in UCM device be interpreted as that the device can be opened with exactly 2 channels, or that it can be opened with some number of channels >= 2? <details> ``` $ alsaucm open '<<<SplitPCM=1>>>hw:2' dump text Verb.HiFi { Comment Default Device.Line1 { Comment "Headphones / Line 1-2" Values { CaptureCTL "_ucm0001.hw:Gen" PlaybackCTL "_ucm0001.hw:Gen" PlaybackPCM "_ucm0001.hw:Gen" PlaybackPriority 200 } } Device.Mic1 { Comment "Input 1 Mic/Inst/Line" Values { CaptureCTL "_ucm0001.hw:Gen" CaptureChannel0 0 CaptureChannelPos0 MONO CaptureChannels 2 CapturePCM "_ucm0001.hw:Gen,0" CapturePriority 200 PlaybackCTL "_ucm0001.hw:Gen" } } Device.Mic2 { Comment "Input 2 Mic/Inst/Line" Values { CaptureCTL "_ucm0001.hw:Gen" CaptureChannel0 1 CaptureChannelPos0 MONO CaptureChannels 2 CapturePCM "_ucm0001.hw:Gen,0" CapturePriority 100 PlaybackCTL "_ucm0001.hw:Gen" } } } Verb.Direct { Comment "Direct Scarlett Solo 4th Gen" Device.Direct { Comment "Direct Scarlett Solo 4th Gen" Values { CaptureCTL "_ucm0001.hw:Gen" CaptureChannels 2 CapturePCM "_ucm0001.hw:Gen" CapturePriority 1000 PlaybackCTL "_ucm0001.hw:Gen" PlaybackChannels 2 PlaybackPCM "_ucm0001.hw:Gen" PlaybackPriority 1000 } } } ``` </details> Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/515 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

RE: [PATCH v5 2/2] ASoC: dt-bindings: Add Everest ES8389 audio CODEC
by Zhang Yi 09 Mar '25

09 Mar '25

> > 1 file changed, 50 insertions(+) > > create mode 100644 Documentation/devicetree/bindings/sound/everest,es8389.yaml > > > > diff --git a/Documentation/devicetree/bindings/sound/everest,es8389.yaml b/Documentation/devicetree/bindings/sound/everest,es8389.yaml > > > You got feedback how to create CC list. Two times. > > You still create it incorrectly and Cc address which does not exist in > the kernel and is a proof you work on wrong tree, which I asked to correct. > > What's more, there is still no changelog and you never responded to > review comments. > > Can you slow down with resubmits and rethink the received feedback? I > expect that feedback to be responded to, so I am not wasting time here > repeating the same every time. I'm sorry for the trouble I've caused you. And I have responded to the comments at v3 and v4.

1 0

PORT: GTK2 to GTK3
by GitHub pull_request - opened 07 Mar '25

07 Mar '25

alsa-project/alsa-tools pull request #27 was opened from genbtc: I've undertaken a task of porting alsa-tools programs from GTK2 to GTK3. These patches start with a base of 5eaadd15bda4cf0424ee2470a6d304a0c2ce10cb (master) Date: Mon Jan 27 (latest pull) Results: Compilation completes successfully, Linked to GTK3, but I do not have the hardware cards to test run them. All work has been done by me, and I request a code review on the GDK 2 Pixmap to GDK 3 Pixbuf + Cairo surfaces. Like it needs new logic and new plan. not simply change statement A to B (whatever I did to get it to compile) I tried to comply with the existing style, and There was also a fair bit of inconsequential whitespace removed like (\n\t\n) or trailing space chars as I discovered them. Looking forward to upstreaming these changes somehow if you would like to work on this. - genBTC @ gentoo Request URL : https://github.com/alsa-project/alsa-tools/pull/27 Patch URL : https://github.com/alsa-project/alsa-tools/pull/27.patch Repository URL: https://github.com/alsa-project/alsa-tools

1 0

[PATCH v5 0/2] ASoC: codecs: add support for ES8389
by Zhang Yi 07 Mar '25

07 Mar '25

The driver is for codec ES8389 of everest-semi. Zhang Yi (2): ASoC: codecs: add support for ES8389 ASoC: dt-bindings: Add Everest ES8389 audio CODEC .../bindings/sound/everest,es8389.yaml | 50 + sound/soc/codecs/Kconfig | 6 +- sound/soc/codecs/Makefile | 2 + sound/soc/codecs/es8389.c | 966 ++++++++++++++++++ sound/soc/codecs/es8389.h | 140 +++ 5 files changed, 1163 insertions(+), 1 deletion(-) create mode 100644 Documentation/devicetree/bindings/sound/everest,es8389.yaml create mode 100644 sound/soc/codecs/es8389.c create mode 100644 sound/soc/codecs/es8389.h -- 2.17.1

3 4

RE: [PATCH v4 1/2] ASoC: codecs: add support for ES8389
by Zhang Yi 07 Mar '25

07 Mar '25

> > > No, the machine driver should be configuring different TDM slots for > > > each device - that's the whole point of the API. > > > We are using multiple codecs as a single device.So we can't use set_tdm_slot to configure > > different slots for multiple codecs under one device. > > What do you mean by using it "as a single device"? Multiple CODECs on > the same link is the main use case for set_tdm_slot(). Thanks for the advice.The method you mentioned could work. I will modify my driver.

1 0

regresion: snd_hdsp
by GitHub issues - edited 07 Mar '25

07 Mar '25

alsa-project/alsa-firmware issue #11 was edited from juanpc2018: ![Image](https://github.com/user-attachments/assets/e57192a7-ad1c-44f9-b51d-7806efed9a32) ``` $ pulseaudio W: [pulseaudio] module.c: module-combine is deprecated: Please use module-combine-sink instead of module-combine! W: [pulseaudio] module-combine.c: We will now load module-combine-sink. Please make sure to remove module-combine from your configuration. E: [alsa-sink-RME Hammerfall HDSP 9632] alsa-sink.c: ALSA woke us up to write new data to the device, but there was actually nothing to write. E: [alsa-sink-RME Hammerfall HDSP 9632] alsa-sink.c: Most likely this is a bug in the ALSA driver 'snd_hdsp'. Please report this issue to the ALSA developers. E: [alsa-sink-RME Hammerfall HDSP 9632] alsa-sink.c: We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. ``` when i play any video or sound file, most of the time PulseAudio takes between 7 to 15 seconds to fill the buffer in proper order, audio is played in wrong order, chopped, like [stutter](https://www.youtube.com/watch?v=9ZaDn5wne8Y&t=609s).[2](https://ww… i have tested /etc/pulse/deamon.conf s24le to s24be problem persists. forced sample rate to 48Khz, APIC ACPI Tables v2 & v3 in Bios, Audacity 3.6. ALSA hw:0 also Crash. but Jack Audio works ok. i remember hdsp9632 worked ok on U/Kubuntu 10.04 & 10.10 Now 20.04.4 LTS has problems. im using Liquorix 6.3 Kernel. ``` $ sudo apt show alsa-firmware* Package: alsa-firmware-loaders Version: 1.1.7-1ubuntu1 Priority: extra Section: multiverse/sound Source: alsa-tools Origin: Ubuntu Maintainer: Ubuntu Developers <ubuntu-devel-discuss(a)lists.ubuntu.com> Original-Maintainer: Debian ALSA Maintainers <pkg-alsa-devel(a)lists.alioth.debian.org> Bugs: https://bugs.launchpad.net/ubuntu/+filebug Installed-Size: 211 kB Depends: fxload, udev, libasound2 (>= 1.0.24.1), libc6 (>= 2.4) Homepage: https://www.alsa-project.org/ Download-Size: 25,0 kB APT-Manual-Installed: yes APT-Sources: http://archive.ubuntu.com/ubuntu focal/multiverse amd64 Packages Description: ALSA software loaders for specific hardware $ sudo apt show alsa-tools-gui Package: alsa-tools-gui Version: 1.1.7-1ubuntu1 Priority: extra Section: universe/sound Source: alsa-tools Origin: Ubuntu Maintainer: Ubuntu Developers <ubuntu-devel-discuss(a)lists.ubuntu.com> Original-Maintainer: Debian ALSA Maintainers <pkg-alsa-devel(a)lists.alioth.debian.org> Bugs: https://bugs.launchpad.net/ubuntu/+filebug Installed-Size: 1.002 kB Depends: libasound2 (>= 1.0.24.1), libc6 (>= 2.29), libfltk1.1 (>= 1.1.6), libglib2.0-0 (>= 2.35.9), libgtk-3-0 (>= 3.0.0), libgtk2.0-0 (>= 2.8.0), libstdc++6 (>= 5.2) Homepage: https://www.alsa-project.org/ Task: ubuntustudio-audio Download-Size: 220 kB APT-Manual-Installed: yes APT-Sources: http://archive.ubuntu.com/ubuntu focal/universe amd64 Packages Description: GUI based ALSA utilities for specific hardware ``` ---------------------------------- found another way to trigger the bug.... if i play a video on YT, the first 7 seconds are corrupted, but if in the middle of the video i pause, and wait 7 seconds, and continue to play, the corrupted 7 seconds happen again. Issue URL : https://github.com/alsa-project/alsa-firmware/issues/11 Repository URL: https://github.com/alsa-project/alsa-firmware

1 0

HDSPconf missing setting...
by GitHub issues - edited 06 Mar '25

06 Mar '25

alsa-project/alsa-tools issue #26 was edited from juanpc2018: $ alsamixer v1.2.2 ![Image](https://github.com/user-attachments/assets/6b2860d4-73a0-4526-9a97-13a7c5d6785a) ![Image](https://github.com/user-attachments/assets/ec8253d0-627a-4dda-a45e-1a2080d20563) HDSPconf 1.4 from alsa-tools-gui ![Image](https://github.com/user-attachments/assets/d2d1dba5-8da6-43ae-a594-cfd89bdd7e1f) ![Image](https://github.com/user-attachments/assets/17d070d3-067e-4d1b-adf9-131f843a634c) i dont know why is called DDS Sample Offset but it should be like HDSPmixer ![Image](https://github.com/user-attachments/assets/3ac4ca5c-df61-4cb3-8e8d-2af8fc44019c) DDS "Sample Offset" is similar to a PLL with a more aggressive rejection of the external clock signal. IF external clock coming from s/pdif is better vs. RME DDS 200pico second jitter clock, worsens the clock. like a blend control: 0% preference for internal DDS clock = 100% preference for external clock, 100% preference for internal DDS clock = 0% preference for external clock. at: [100% DDS](https://www.youtube.com/watch?v=zJLw2SF_L-g&t=27s) DDS is the Truck, still follows ext.clock, as a recommendation, Not like an [Orchestra](https://www.youtube.com/watch?v=HqCIYWZ5b7k&t=0s)_[Director](htt… 0% DDS, ext.clock becomes like a Whip, internal like a Horse / Slave / [Circus Lion](https://www.youtube.com/watch?v=zJLw2SF_L-g&t=105s) DDS is the Lion. Clock [Jitter](https://www.youtube.com/watch?v=zJLw2SF_L-g&t=152s) Issue URL : https://github.com/alsa-project/alsa-tools/issues/26 Repository URL: https://github.com/alsa-project/alsa-tools

1 0

HDSPconf missing setting...
by GitHub issues - edited 06 Mar '25

06 Mar '25

alsa-project/alsa-tools issue #26 was edited from juanpc2018: $ alsamixer v1.2.2 ![Image](https://github.com/user-attachments/assets/6b2860d4-73a0-4526-9a97-13a7c5d6785a) ![Image](https://github.com/user-attachments/assets/ec8253d0-627a-4dda-a45e-1a2080d20563) HDSPconf 1.4 from alsa-tools-gui ![Image](https://github.com/user-attachments/assets/d2d1dba5-8da6-43ae-a594-cfd89bdd7e1f) ![Image](https://github.com/user-attachments/assets/17d070d3-067e-4d1b-adf9-131f843a634c) i dont know why iis called DDS Sample Offset but it should be like HDSPmixer ![Image](https://github.com/user-attachments/assets/3ac4ca5c-df61-4cb3-8e8d-2af8fc44019c) DDS "Sample Offset" is similar to a PLL with a more aggressive rejection of the external clock signal. IF external clock coming from s/pdif is better vs. RME DDS 200pico second jitter clock, worsens the clock. like a blend control: 0% preference for internal DDS clock = 100% preference for external clock, 100% preference for internal DDS clock = 0% preference for external clock. at: [100% DDS](https://www.youtube.com/watch?v=zJLw2SF_L-g&t=27s) DDS is the Truck, still follows ext.clock, as a recommendation, Not like an [Orchestra](https://www.youtube.com/watch?v=HqCIYWZ5b7k&t=0s)_[Director](htt… 0% DDS, ext.clock becomes like a Whip, internal like a Horse / Slave / [Circus Lion](https://www.youtube.com/watch?v=zJLw2SF_L-g&t=105s) DDS is the Lion. Clock [Jitter](https://www.youtube.com/watch?v=zJLw2SF_L-g&t=152s) Issue URL : https://github.com/alsa-project/alsa-tools/issues/26 Repository URL: https://github.com/alsa-project/alsa-tools

1 0

HDSPconf missing setting...
by GitHub issues - edited 06 Mar '25

06 Mar '25

alsa-project/alsa-tools issue #26 was edited from juanpc2018: $ alsamixer v1.2.2 ![Image](https://github.com/user-attachments/assets/6b2860d4-73a0-4526-9a97-13a7c5d6785a) ![Image](https://github.com/user-attachments/assets/ec8253d0-627a-4dda-a45e-1a2080d20563) HDSPconf 1.4 from alsa-tools-gui ![Image](https://github.com/user-attachments/assets/d2d1dba5-8da6-43ae-a594-cfd89bdd7e1f) ![Image](https://github.com/user-attachments/assets/17d070d3-067e-4d1b-adf9-131f843a634c) i dont know why iis called DDS Sample Offset but it should be like HDSPmixer ![Image](https://github.com/user-attachments/assets/3ac4ca5c-df61-4cb3-8e8d-2af8fc44019c) DDS "Sample Offsett" is similar to a PLL with a more aggressive rejection of the external clock signal. IF external clock coming from s/pdif is better vs. RME DDS 200pico second jitter clock, worsens the clock. like a blend control: 0% preference for internal DDS clock = 100% preference for external clock, 100% preference for internal DDS clock = 0% preference for external clock. at: [100% DDS](https://www.youtube.com/watch?v=zJLw2SF_L-g&t=27s) DDS is the Truck, still follows ext.clock, as a recommendation, Not like an [Orchestra](https://www.youtube.com/watch?v=HqCIYWZ5b7k&t=0s)_[Director](htt… 0% DDS, ext.clock becomes like a Whip, internal like a Horse / Slave / [Circus Lion](https://www.youtube.com/watch?v=zJLw2SF_L-g&t=105s) DDS is the Lion. Clock [Jitter](https://www.youtube.com/watch?v=zJLw2SF_L-g&t=152s) Issue URL : https://github.com/alsa-project/alsa-tools/issues/26 Repository URL: https://github.com/alsa-project/alsa-tools

1 0

HDSPconf missing setting...
by GitHub issues - edited 06 Mar '25

06 Mar '25

alsa-project/alsa-tools issue #26 was edited from juanpc2018: $ alsamixer v1.2.2 ![Image](https://github.com/user-attachments/assets/6b2860d4-73a0-4526-9a97-13a7c5d6785a) ![Image](https://github.com/user-attachments/assets/ec8253d0-627a-4dda-a45e-1a2080d20563) HDSPconf 1.4 from alsa-tools-gui ![Image](https://github.com/user-attachments/assets/d2d1dba5-8da6-43ae-a594-cfd89bdd7e1f) ![Image](https://github.com/user-attachments/assets/17d070d3-067e-4d1b-adf9-131f843a634c) i dont know why iis called DDS Sample Offset but it should be a slider similar to HDSPmixer ![Image](https://github.com/user-attachments/assets/3ac4ca5c-df61-4cb3-8e8d-2af8fc44019c) DDS "Sample Offsett" is similar to a PLL with a more aggressive rejection of the external clock signal. IF external clock coming from s/pdif is better vs. RME DDS 200pico second jitter clock, worsens the clock. like a blend control: 0% preference for internal DDS clock = 100% preference for external clock, 100% preference for internal DDS clock = 0% preference for external clock. at: [100% DDS](https://www.youtube.com/watch?v=zJLw2SF_L-g&t=27s) DDS is the Truck, still follows ext.clock, as a recommendation, Not like an [Orchestra](https://www.youtube.com/watch?v=HqCIYWZ5b7k&t=0s)_[Director](htt… 0% DDS, ext.clock becomes like a Whip, internal like a Horse / Slave / [Circus Lion](https://www.youtube.com/watch?v=zJLw2SF_L-g&t=105s) DDS is the Lion. Clock [Jitter](https://www.youtube.com/watch?v=zJLw2SF_L-g&t=152s) Issue URL : https://github.com/alsa-project/alsa-tools/issues/26 Repository URL: https://github.com/alsa-project/alsa-tools

1 0

HDSPconf missing setting...
by GitHub issues - edited 06 Mar '25

06 Mar '25

alsa-project/alsa-tools issue #26 was edited from juanpc2018: $ alsamixer v1.2.2 ![Image](https://github.com/user-attachments/assets/6b2860d4-73a0-4526-9a97-13a7c5d6785a) ![Image](https://github.com/user-attachments/assets/ec8253d0-627a-4dda-a45e-1a2080d20563) HDSPconf 1.4 from alsa-tools-gui ![Image](https://github.com/user-attachments/assets/d2d1dba5-8da6-43ae-a594-cfd89bdd7e1f) ![Image](https://github.com/user-attachments/assets/17d070d3-067e-4d1b-adf9-131f843a634c) i dont know why iis called DDS Sample Offset but it should be a slider similar to HDSPmixer ![Image](https://github.com/user-attachments/assets/3ac4ca5c-df61-4cb3-8e8d-2af8fc44019c) DDS "Sample Offsett" is similar to a PLL with a more aggressive rejection of the external clock signal. IF external clock coming from s/pdif is better vs. RME DDS 200pico second jitter clock, worsens the clock. like a blend control: 0% preference for internal DDS clock = 100% preference for external clock, 100% preference for internal DDS clock = 0% preference for external clock. at: [100% DDS](https://www.youtube.com/watch?v=zJLw2SF_L-g&t=27s) DDS is the Truck, still follows ext.clock, as a recommendation, Not like an [Orchestra](https://www.youtube.com/watch?v=HqCIYWZ5b7k&t=0s) [Director](https://www.youtube.com/watch?v=VSWpVId72Jo&t=68s) 0% DDS, ext.clock becomes like a Whip, internal like a Horse / Slave / [Circus Lion](https://www.youtube.com/watch?v=zJLw2SF_L-g&t=105s) DDS is the Lion. Clock [Jitter](https://www.youtube.com/watch?v=zJLw2SF_L-g&t=152s) Issue URL : https://github.com/alsa-project/alsa-tools/issues/26 Repository URL: https://github.com/alsa-project/alsa-tools

1 0

HDSPconf missing setting...
by GitHub issues - edited 06 Mar '25

06 Mar '25

alsa-project/alsa-tools issue #26 was edited from juanpc2018: $ alsamixer v1.2.2 ![Image](https://github.com/user-attachments/assets/6b2860d4-73a0-4526-9a97-13a7c5d6785a) ![Image](https://github.com/user-attachments/assets/ec8253d0-627a-4dda-a45e-1a2080d20563) HDSPconf 1.4 from alsa-tools-gui ![Image](https://github.com/user-attachments/assets/d2d1dba5-8da6-43ae-a594-cfd89bdd7e1f) ![Image](https://github.com/user-attachments/assets/17d070d3-067e-4d1b-adf9-131f843a634c) i dont know why iis called DDS Sample Offset but it should be a slider similar to HDSPmixer ![Image](https://github.com/user-attachments/assets/3ac4ca5c-df61-4cb3-8e8d-2af8fc44019c) DDS "Sample Offsett" is similar to a PLL with a more aggressive rejection of the external clock signal. IF external clock coming from s/pdif is better vs. RME DDS 200pico second jitter clock, worsens the clock. like a blend control: 0% preference for internal DDS clock = 100% preference for external clock, 100% preference for internal DDS clock = 0% preference for external clock. at: [100% DDS](https://www.youtube.com/watch?v=zJLw2SF_L-g&t=27s) DDS is the Truck, still follows ext.clock, as a recommendation, Not like an [Orchestra](https://www.youtube.com/watch?v=HqCIYWZ5b7k&t=0s) [Director](https://www.youtube.com/watch?v=VSWpVId72Jo&t=67s) 0% DDS, ext.clock becomes like a Whip, internal like a Horse / Slave / [Circus Lion](https://www.youtube.com/watch?v=zJLw2SF_L-g&t=105s) DDS is the Lion. Clock [Jitter](https://www.youtube.com/watch?v=zJLw2SF_L-g&t=152s) Issue URL : https://github.com/alsa-project/alsa-tools/issues/26 Repository URL: https://github.com/alsa-project/alsa-tools

1 0

HDSPconf missing setting...
by GitHub issues - edited 06 Mar '25

06 Mar '25

alsa-project/alsa-tools issue #26 was edited from juanpc2018: $ alsamixer v1.2.2 ![Image](https://github.com/user-attachments/assets/6b2860d4-73a0-4526-9a97-13a7c5d6785a) ![Image](https://github.com/user-attachments/assets/ec8253d0-627a-4dda-a45e-1a2080d20563) HDSPconf 1.4 from alsa-tools-gui ![Image](https://github.com/user-attachments/assets/d2d1dba5-8da6-43ae-a594-cfd89bdd7e1f) ![Image](https://github.com/user-attachments/assets/17d070d3-067e-4d1b-adf9-131f843a634c) i dont know why iis called DDS Sample Offset but it should be a slider similar to HDSPmixer ![Image](https://github.com/user-attachments/assets/3ac4ca5c-df61-4cb3-8e8d-2af8fc44019c) DDS "Sample Offsett" is similar to a PLL with a more aggressive rejection of the external clock signal. IF external clock coming from s/pdif is better vs. RME DDS 200pico second jitter clock, worsens the clock. like a blend control: 0% preference for internal DDS clock = 100% preference for external clock, 100% preference for internal DDS clock = 0% preference for external clock. at: [100% DDS](https://www.youtube.com/watch?v=zJLw2SF_L-g&t=27s) DDS is the Truck, still follows ext.clock, as a recommendation, Not like an [Orchestra](https://www.youtube.com/watch?v=HqCIYWZ5b7k&t=0s)[Director](http… 0% DDS, ext.clock becomes like a Whip, internal like a Horse / Slave / [Circus Lion](https://www.youtube.com/watch?v=zJLw2SF_L-g&t=105s) DDS is the Lion. Clock [Jitter](https://www.youtube.com/watch?v=zJLw2SF_L-g&t=152s) Issue URL : https://github.com/alsa-project/alsa-tools/issues/26 Repository URL: https://github.com/alsa-project/alsa-tools

1 0

HDSPconf missing setting...
by GitHub issues - edited 06 Mar '25

06 Mar '25

alsa-project/alsa-tools issue #26 was edited from juanpc2018: $ alsamixer v1.2.2 ![Image](https://github.com/user-attachments/assets/6b2860d4-73a0-4526-9a97-13a7c5d6785a) ![Image](https://github.com/user-attachments/assets/ec8253d0-627a-4dda-a45e-1a2080d20563) HDSPconf 1.4 from alsa-tools-gui ![Image](https://github.com/user-attachments/assets/d2d1dba5-8da6-43ae-a594-cfd89bdd7e1f) ![Image](https://github.com/user-attachments/assets/17d070d3-067e-4d1b-adf9-131f843a634c) i dont know why iis called DDS Sample Offset but it should be a slider similar to HDSPmixer ![Image](https://github.com/user-attachments/assets/3ac4ca5c-df61-4cb3-8e8d-2af8fc44019c) DDS "Sample Offsett" is similar to a PLL with a more aggressive rejection of the external clock signal. IF external clock coming from s/pdif is better vs. RME DDS 200pico second jitter clock, worsens the clock. like a blend control: 0% preference for internal DDS clock = 100% preference for external clock, 100% preference for internal DDS clock = 0% preference for external clock. at: [100% DDS](https://www.youtube.com/watch?v=zJLw2SF_L-g&t=27s) DDS is the Truck, still follows ext.clock, as a recommendation, Not like an [Orchestra Director](https://www.youtube.com/watch?v=HqCIYWZ5b7k&t=0s) 0% DDS, ext.clock becomes like a Whip, internal like a Horse / Slave / [Circus Lion](https://www.youtube.com/watch?v=zJLw2SF_L-g&t=105s) DDS is the Lion. Clock [Jitter](https://www.youtube.com/watch?v=zJLw2SF_L-g&t=152s) Issue URL : https://github.com/alsa-project/alsa-tools/issues/26 Repository URL: https://github.com/alsa-project/alsa-tools

1 0

HDSPconf missing setting...
by GitHub issues - edited 06 Mar '25

06 Mar '25

alsa-project/alsa-tools issue #26 was edited from juanpc2018: $ alsamixer v1.2.2 ![Image](https://github.com/user-attachments/assets/6b2860d4-73a0-4526-9a97-13a7c5d6785a) ![Image](https://github.com/user-attachments/assets/ec8253d0-627a-4dda-a45e-1a2080d20563) HDSPconf 1.4 from alsa-tools-gui ![Image](https://github.com/user-attachments/assets/d2d1dba5-8da6-43ae-a594-cfd89bdd7e1f) ![Image](https://github.com/user-attachments/assets/17d070d3-067e-4d1b-adf9-131f843a634c) i dont know why iis called DDS Sample Offset but it should be a slider similar to HDSPmixer ![Image](https://github.com/user-attachments/assets/3ac4ca5c-df61-4cb3-8e8d-2af8fc44019c) DDS "Sample Offsett" is similar to a PLL with a more aggressive rejection of the external clock signal. IF external clock coming from s/pdif is better vs. RME DDS 200pico second jitter clock, worsens the clock. like a blend control: 0% preference for internal DDS clock = 100% preference for external clock, 100% preference for internal DDS clock = 0% preference for external clock. at: [100% DDS](https://www.youtube.com/watch?v=zJLw2SF_L-g&t=27s) still follows ext.clock, as a recommendation, Not like an [Orchestra Director](https://www.youtube.com/watch?v=HqCIYWZ5b7k&t=0s) 0% DDS, ext.clock becomes like a Whip, internal like a Horse / Slave / [Circus Lion](https://www.youtube.com/watch?v=zJLw2SF_L-g&t=105s) Clock [Jitter](https://www.youtube.com/watch?v=zJLw2SF_L-g&t=152s) Issue URL : https://github.com/alsa-project/alsa-tools/issues/26 Repository URL: https://github.com/alsa-project/alsa-tools

1 0

HDSPconf missing setting...
by GitHub issues - edited 06 Mar '25

06 Mar '25

alsa-project/alsa-tools issue #26 was edited from juanpc2018: $ alsamixer v1.2.2 ![Image](https://github.com/user-attachments/assets/6b2860d4-73a0-4526-9a97-13a7c5d6785a) ![Image](https://github.com/user-attachments/assets/ec8253d0-627a-4dda-a45e-1a2080d20563) HDSPconf 1.4 from alsa-tools-gui ![Image](https://github.com/user-attachments/assets/d2d1dba5-8da6-43ae-a594-cfd89bdd7e1f) ![Image](https://github.com/user-attachments/assets/17d070d3-067e-4d1b-adf9-131f843a634c) i dont know why iis called DDS Sample Offset but it should be a slider similar to HDSPmixer ![Image](https://github.com/user-attachments/assets/3ac4ca5c-df61-4cb3-8e8d-2af8fc44019c) DDS "Sample Offsett" is similar to a PLL with a more aggressive rejection of the external clock signal. IF external clock coming from s/pdif is better vs. RME DDS 200pico second jitter clock, worsens the clock. like a blend control: 0% preference for internal DDS clock = 100% preference for external clock, 100% preference for internal DDS clock = 0% preference for external clock. at: [100% DDS](https://www.youtube.com/watch?v=zJLw2SF_L-g&t=27s) still follows ext.clock, as a recommendation, Not like an Orchestra Director. 0% DDS, ext.clock becomes like a Whip, internal like a Horse / Slave / [Circus Lion](https://www.youtube.com/watch?v=zJLw2SF_L-g&t=105s) Clock [Jitter](https://www.youtube.com/watch?v=zJLw2SF_L-g&t=152s) Issue URL : https://github.com/alsa-project/alsa-tools/issues/26 Repository URL: https://github.com/alsa-project/alsa-tools

1 0

HDSPconf missing setting...
by GitHub issues - edited 06 Mar '25

06 Mar '25

alsa-project/alsa-tools issue #26 was edited from juanpc2018: $ alsamixer v1.2.2 ![Image](https://github.com/user-attachments/assets/6b2860d4-73a0-4526-9a97-13a7c5d6785a) ![Image](https://github.com/user-attachments/assets/ec8253d0-627a-4dda-a45e-1a2080d20563) HDSPconf 1.4 from alsa-tools-gui ![Image](https://github.com/user-attachments/assets/d2d1dba5-8da6-43ae-a594-cfd89bdd7e1f) ![Image](https://github.com/user-attachments/assets/17d070d3-067e-4d1b-adf9-131f843a634c) i dont know why iis called DDS Sample Offset but it should be a slider similar to HDSPmixer ![Image](https://github.com/user-attachments/assets/3ac4ca5c-df61-4cb3-8e8d-2af8fc44019c) DDS "Sample Offsett" is similar to a PLL with a more aggressive rejection of the external clock signal. IF external clock coming from s/pdif is better vs. RME DDS 200pico second jitter clock, worsens the clock. like a blend control: 0% preference for internal DDS clock = 100% preference for external clock, 100% preference for internal DDS clock = 0% preference for external clock. at: [100% DDS](https://www.youtube.com/watch?v=zJLw2SF_L-g&t=27s) still follows ext.clock, as a recommendation, Not like an Orchestra Director. 0% DDS, ext.clock becomes like a Whip, internal like a Horse / Slave / [Circus Lion](https://www.youtube.com/watch?v=zJLw2SF_L-g&t=105s) [Jitter](https://www.youtube.com/watch?v=zJLw2SF_L-g&t=152s) Issue URL : https://github.com/alsa-project/alsa-tools/issues/26 Repository URL: https://github.com/alsa-project/alsa-tools

1 0

HDSPconf missing setting...
by GitHub issues - edited 06 Mar '25

06 Mar '25

alsa-project/alsa-tools issue #26 was edited from juanpc2018: $ alsamixer v1.2.2 ![Image](https://github.com/user-attachments/assets/6b2860d4-73a0-4526-9a97-13a7c5d6785a) ![Image](https://github.com/user-attachments/assets/ec8253d0-627a-4dda-a45e-1a2080d20563) HDSPconf 1.4 from alsa-tools-gui ![Image](https://github.com/user-attachments/assets/d2d1dba5-8da6-43ae-a594-cfd89bdd7e1f) ![Image](https://github.com/user-attachments/assets/17d070d3-067e-4d1b-adf9-131f843a634c) i dont know why iis called DDS Sample Offset but it should be a slider similar to HDSPmixer ![Image](https://github.com/user-attachments/assets/3ac4ca5c-df61-4cb3-8e8d-2af8fc44019c) DDS "Sample Offsett" is similar to a PLL with a more aggressive rejection of the external clock signal. IF external clock coming from s/pdif is better vs. RME DDS 200pico second jitter clock, worsens the clock. like a blend control: 0% preference for internal DDS clock = 100% preference for external clock, 100% preference for internal DDS clock = 0% preference for external clock. at: [100% DDS](https://www.youtube.com/watch?v=zJLw2SF_L-g&t=27s) still follows ext.clock, as a recommendation, Not like an Orchestra Director. 0% DDS, ext.clock becomes like a Whip, internal like a Horse / Slave / [Circus Lion](https://www.youtube.com/watch?v=zJLw2SF_L-g&t=105s) Issue URL : https://github.com/alsa-project/alsa-tools/issues/26 Repository URL: https://github.com/alsa-project/alsa-tools

1 0

HDSPconf missing setting...
by GitHub issues - edited 06 Mar '25

06 Mar '25

alsa-project/alsa-tools issue #26 was edited from juanpc2018: $ alsamixer v1.2.2 ![Image](https://github.com/user-attachments/assets/6b2860d4-73a0-4526-9a97-13a7c5d6785a) ![Image](https://github.com/user-attachments/assets/ec8253d0-627a-4dda-a45e-1a2080d20563) HDSPconf 1.4 from alsa-tools-gui ![Image](https://github.com/user-attachments/assets/d2d1dba5-8da6-43ae-a594-cfd89bdd7e1f) ![Image](https://github.com/user-attachments/assets/17d070d3-067e-4d1b-adf9-131f843a634c) i dont know why iis called DDS Sample Offset but it should be a slider similar to HDSPmixer ![Image](https://github.com/user-attachments/assets/3ac4ca5c-df61-4cb3-8e8d-2af8fc44019c) DDS "Sample Offsett", is similar to a PLL with a more aggressive rejection of the external clock signal. IF the external clock coming from s/pdif is better, vs. RME DDS 200pico second jitter clock, worsens the clock. like a blend control: 0% preference for internal DDS clock = 100% preference for external clock, 100% preference for internal DDS clock = 0% preference for external clock. at: 100% DDS still follows external clock, but as a suggestion/recommendation, Not like an Orchestra Director. 0% DDS, ext. clock becomes like a Whip, and internal like a Horse / Slave / Circus Lion. Issue URL : https://github.com/alsa-project/alsa-tools/issues/26 Repository URL: https://github.com/alsa-project/alsa-tools

1 0

HDSPconf missing setting...
by GitHub issues - edited 06 Mar '25

06 Mar '25

alsa-project/alsa-tools issue #26 was edited from juanpc2018: $ alsamixer v1.2.2 ![Image](https://github.com/user-attachments/assets/6b2860d4-73a0-4526-9a97-13a7c5d6785a) ![Image](https://github.com/user-attachments/assets/ec8253d0-627a-4dda-a45e-1a2080d20563) HDSPconf 1.4 from alsa-tools-gui ![Image](https://github.com/user-attachments/assets/d2d1dba5-8da6-43ae-a594-cfd89bdd7e1f) ![Image](https://github.com/user-attachments/assets/17d070d3-067e-4d1b-adf9-131f843a634c) i dont know why iis called DDS Sample Offset but it should be a slider similar to HDSPmixer ![Image](https://github.com/user-attachments/assets/3ac4ca5c-df61-4cb3-8e8d-2af8fc44019c) DDS "Sample Offsett", is similar to a PLL with a more aggressive rejection of the external clock signal. IF the external clock coming from s/pdif is better, vs. RME DDS 200pico second jitter clock, worsens the clock. like a blend control: 0% preference for internal DDS clock = 100% preference for external clock, 100% preference for internal DDS clock = 0% preference for external clock. Issue URL : https://github.com/alsa-project/alsa-tools/issues/26 Repository URL: https://github.com/alsa-project/alsa-tools

1 0

HDSPconf missing setting...
by GitHub issues - edited 06 Mar '25

06 Mar '25

alsa-project/alsa-tools issue #26 was edited from juanpc2018: $ alsamixer 1.2.2. ![Image](https://github.com/user-attachments/assets/6b2860d4-73a0-4526-9a97-13a7c5d6785a) ![Image](https://github.com/user-attachments/assets/ec8253d0-627a-4dda-a45e-1a2080d20563) HDSPconf 1.4 from alsa-tools-gui ![Image](https://github.com/user-attachments/assets/d2d1dba5-8da6-43ae-a594-cfd89bdd7e1f) ![Image](https://github.com/user-attachments/assets/17d070d3-067e-4d1b-adf9-131f843a634c) i dont know why iis called DDS Sample Offset but it should be a slider similar to HDSPmixer ![Image](https://github.com/user-attachments/assets/3ac4ca5c-df61-4cb3-8e8d-2af8fc44019c) DDS "Sample Offsett", is similar to a PLL with a more aggressive rejection of the external clock signal. IF the external clock coming from s/pdif is better, vs. RME DDS 200pico second jitter clock, it worsens the clock. acts like a blend control: 0% preference for internal DDS clock = 100% preference for external clock, 100% preference for internal DDS clock = 0% preference for external clock. Issue URL : https://github.com/alsa-project/alsa-tools/issues/26 Repository URL: https://github.com/alsa-project/alsa-tools

1 0

HDSPconf missing setting...
by GitHub issues - opened 06 Mar '25

06 Mar '25

alsa-project/alsa-tools issue #26 was opened from juanpc2018: $ alsamixer 1.2.2. ![Image](https://github.com/user-attachments/assets/6b2860d4-73a0-4526-9a97-13a7c5d6785a) ![Image](https://github.com/user-attachments/assets/ec8253d0-627a-4dda-a45e-1a2080d20563) HDSPconf 1.4 from alsa-tools-gui ![Image](https://github.com/user-attachments/assets/d2d1dba5-8da6-43ae-a594-cfd89bdd7e1f) ![Image](https://github.com/user-attachments/assets/17d070d3-067e-4d1b-adf9-131f843a634c) i dont know why iis called DDS Sample Offset should be a slider similar to HDSPmixer ![Image](https://github.com/user-attachments/assets/3ac4ca5c-df61-4cb3-8e8d-2af8fc44019c) DDS "Sample Offsett", is similar to a PLL with a more aggressive rejection of the external clock signal. IF the external clock coming from s/pdif is better, vs. RME DDS 200pico second jitter clock, it worsens the clock. acts like a blend control: 0% preference for internal DDS clock = 100% preference for external clock, 100% preference for internal DDS clock = 0% preference for external clock. Issue URL : https://github.com/alsa-project/alsa-tools/issues/26 Repository URL: https://github.com/alsa-project/alsa-tools

1 0

Midi Ports are ambiguously named after some alsa update, should be distinct.
by GitHub issues - opened 06 Mar '25

06 Mar '25

alsa-project/alsa-lib issue #442 was opened from Thrilled-Oaf: Greetings~. I'm on Slackware, which is a pretty languid, laid back distro... so I have to admit that I don't know when this happened... But I recently updated it, cause I bought a new midi keyboard, a launchkey mk4, and it should have a 'DAW' port, both on its USB connection and the classic pinned midi cable thing. But instead, aconnect -i and all that just gives me ``` client 24: 'Launchkey MK4 61' [type=kernel,card=2] 0 'Launchkey MK4 61 Launchkey MK4 ' 1 'Launchkey MK4 61 Launchkey MK4 ' ``` I actually use JACK audio on top of alsa but even without jack, this still occurs. I can sort of pretty print this for JACK but it would be best to have this working properly at the very base... cause if I disable all alsa seq/raw support from JACK, then DAWs like ardour won't even see any midi ports anymore. I also tried being 'clever' and loading `snd-seq-dummy` and setting up some routing that way... didn't work. Then I actually edited the source file for the dummy device to specifically give me a properly named DAW port if the index of the dummy port was like, 3 or something. And while the naming worked, it was ultimately ...not able to help out with this issue. What is basically assumed by many clients, including the configuration utility `components` for the keyboard is that it has a distinguishable second midi port with a DAW suffix, thus...this is what `aconnect -i` should ideally show. ``` client 24: 'Launchkey MK4 61' [type=kernel,card=2] 0 'Launchkey MK4 61 Launchkey MK4 ' 1 'Launchkey MK4 61 Launchkey MK4 DAW' ``` I think I read an email somewhere that alsa might be running into a name length limit, but ...perhaps there's a way to configure this? If not through a configuration file, then maybe somewhere in the source itself...? I am not shy about that but I would need directions...100% lol. I also looked up other issues here and some talk about portaudio and mixxx and stuff but this is definitely a system wide issue for me. And so sad too, it's a text descriptor... a text label of all things... bringing the mighty ALSA to its knees. A measly, peasly, text label for a little bittle MIDI port... I tried udev too but ...that was utterly fruitless due to my lack of knowledge/skill... but this all should signal that I'm willing to put in some work, too. I just don't know what to do anymore. maybe .asoundrc but all the examples are about PCM and my PCM cards work fine, it's just this MIDI ambiguity thing... Kernel: ` 6.12.17 #1 SMP PREEMPT_DYNAMIC Thu Feb 27 15:22:43 CST 2025 x86_64 AMD Ryzen 7 2700 Eight-Core Processor AuthenticAMD GNU/Linux` All the alsa related packages on my system (many are third party but compiled by me) ``` alsa-lib-1.2.13-x86_64-2 alsa-oss-1.1.8-x86_64-3 alsa-plugin-jack-1.2.5-x86_64-1SBo-git alsa-plugins-1.2.12-x86_64-2 alsa-plugins-compat32-1.2.12-x86_64-2compat32 alsa-tools-1.0.28-x86_64-1ponce alsa-utils-1.2.13-x86_64-3 alsamixergui-0.9.0rc2-x86_64-2_SBo bluez-alsa-3.1.0-x86_64-2ponce zita-alsa-pcmi-0.3.2-x86_64-1alien ``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/442 Repository URL: https://github.com/alsa-project/alsa-lib

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RE: [PATCH v4 1/2] ASoC: codecs: add support for ES8389
by Zhang Yi 06 Mar '25

06 Mar '25

> > > set_tdm_slot() > > > We will register multiple codecs inside a single dai_link and differentiate these > > codecs by of_node. And the adc_slot and the dac_slot may be different, they can > > be decided by the user.If we use set_tdm_slot,the adc_slot and the dac_slot will > > be same. > > No, the machine driver should be configuring different TDM slots for > each device - that's the whole point of the API. We are using multiple codecs as a single device.So we can't use set_tdm_slot to configure different slots for multiple codecs under one device. > > > > + ret = clk_prepare_enable(es8389->mclk); > > > > + if (ret) { > > > > + dev_err(codec->dev, "%s, unable to enable mclk\n", __func__); > > > > + return ret; > > > > + } > > > > + } > > > > Making the use of a MCLK depend on the configuration of a TDM slot for > > > the ADC seems *very* unusual, what's going on there? > > > Because we are associating multiple codecs under a single dai_link, we will be > > executing probe and init many times during initialization.But MCLK only needs > > to be used once.So we decided making the use of a MCLK depend on the configuration > > of a TDM slot for the ADC > > No, each device should just get and enable the MCLK itself - the clock > API does reference counting so there's no problem with this, it's normal > for a clok to have multiple consumers. ok,I'll fix it

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regresion: snd_hdsp
by GitHub issues - edited 06 Mar '25

06 Mar '25

alsa-project/alsa-firmware issue #11 was edited from juanpc2018: ![Image](https://github.com/user-attachments/assets/e57192a7-ad1c-44f9-b51d-7806efed9a32) ``` $ pulseaudio W: [pulseaudio] module.c: module-combine is deprecated: Please use module-combine-sink instead of module-combine! W: [pulseaudio] module-combine.c: We will now load module-combine-sink. Please make sure to remove module-combine from your configuration. E: [alsa-sink-RME Hammerfall HDSP 9632] alsa-sink.c: ALSA woke us up to write new data to the device, but there was actually nothing to write. E: [alsa-sink-RME Hammerfall HDSP 9632] alsa-sink.c: Most likely this is a bug in the ALSA driver 'snd_hdsp'. Please report this issue to the ALSA developers. E: [alsa-sink-RME Hammerfall HDSP 9632] alsa-sink.c: We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. ``` when i play any video or sound file, most of the time PulseAudio takes between 7 to 15 seconds to fill the buffer in proper order, audio is played in wrong order, chopped, like [stutter](https://www.youtube.com/watch?v=9ZaDn5wne8Y&t=609s).[2](https://ww… i have tested /etc/pulse/deamon.conf s24le to s24be problem persists. forced sample rate to 48Khz, APIC ACPI Tables v2 & v3 in Bios, Audacity 3.6. ALSA hw:0 also Crash. but Jack Audio works ok. i remember hdsp9632 worked ok on U/Kubuntu 10.04 & 10.10 Now 20.04.4 LTS has problems. im using Liquorix 6.3 Kernel. ``` $ sudo apt show alsa-firmware* Package: alsa-firmware-loaders Version: 1.1.7-1ubuntu1 Priority: extra Section: multiverse/sound Source: alsa-tools Origin: Ubuntu Maintainer: Ubuntu Developers <ubuntu-devel-discuss(a)lists.ubuntu.com> Original-Maintainer: Debian ALSA Maintainers <pkg-alsa-devel(a)lists.alioth.debian.org> Bugs: https://bugs.launchpad.net/ubuntu/+filebug Installed-Size: 211 kB Depends: fxload, udev, libasound2 (>= 1.0.24.1), libc6 (>= 2.4) Homepage: https://www.alsa-project.org/ Download-Size: 25,0 kB APT-Manual-Installed: yes APT-Sources: http://archive.ubuntu.com/ubuntu focal/multiverse amd64 Packages Description: ALSA software loaders for specific hardware $ sudo apt show alsa-tools-gui Package: alsa-tools-gui Version: 1.1.7-1ubuntu1 Priority: extra Section: universe/sound Source: alsa-tools Origin: Ubuntu Maintainer: Ubuntu Developers <ubuntu-devel-discuss(a)lists.ubuntu.com> Original-Maintainer: Debian ALSA Maintainers <pkg-alsa-devel(a)lists.alioth.debian.org> Bugs: https://bugs.launchpad.net/ubuntu/+filebug Installed-Size: 1.002 kB Depends: libasound2 (>= 1.0.24.1), libc6 (>= 2.29), libfltk1.1 (>= 1.1.6), libglib2.0-0 (>= 2.35.9), libgtk-3-0 (>= 3.0.0), libgtk2.0-0 (>= 2.8.0), libstdc++6 (>= 5.2) Homepage: https://www.alsa-project.org/ Task: ubuntustudio-audio Download-Size: 220 kB APT-Manual-Installed: yes APT-Sources: http://archive.ubuntu.com/ubuntu focal/universe amd64 Packages Description: GUI based ALSA utilities for specific hardware ``` Issue URL : https://github.com/alsa-project/alsa-firmware/issues/11 Repository URL: https://github.com/alsa-project/alsa-firmware

1 0

regresion: snd_hdsp
by GitHub issues - edited 06 Mar '25

06 Mar '25

alsa-project/alsa-firmware issue #11 was edited from juanpc2018: ![Image](https://github.com/user-attachments/assets/e57192a7-ad1c-44f9-b51d-7806efed9a32) ``` $ pulseaudio W: [pulseaudio] module.c: module-combine is deprecated: Please use module-combine-sink instead of module-combine! W: [pulseaudio] module-combine.c: We will now load module-combine-sink. Please make sure to remove module-combine from your configuration. E: [alsa-sink-RME Hammerfall HDSP 9632] alsa-sink.c: ALSA woke us up to write new data to the device, but there was actually nothing to write. E: [alsa-sink-RME Hammerfall HDSP 9632] alsa-sink.c: Most likely this is a bug in the ALSA driver 'snd_hdsp'. Please report this issue to the ALSA developers. E: [alsa-sink-RME Hammerfall HDSP 9632] alsa-sink.c: We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. ``` when i play any video or sound file, most of the time PulseAudio takes between 7 to 15 seconds to fill the buffer in proper order, audio is played in wrong order, chopped, like [stutter](https://www.youtube.com/watch?v=9ZaDn5wne8Y&t=609s).[2](https://ww… i have tested /etc/pulse/deamon.conf s24le to s24be problem persists. forced sample rate to 48Khz, Audacity 3.6. ALSA hw:0 also Crash. but Jack Audio works ok. i remember hdsp9632 worked ok on U/Kubuntu 10.04 & 10.10 Now 20.04.4 LTS has problems. im using Liquorix 6.3 Kernel. ``` $ sudo apt show alsa-firmware* Package: alsa-firmware-loaders Version: 1.1.7-1ubuntu1 Priority: extra Section: multiverse/sound Source: alsa-tools Origin: Ubuntu Maintainer: Ubuntu Developers <ubuntu-devel-discuss(a)lists.ubuntu.com> Original-Maintainer: Debian ALSA Maintainers <pkg-alsa-devel(a)lists.alioth.debian.org> Bugs: https://bugs.launchpad.net/ubuntu/+filebug Installed-Size: 211 kB Depends: fxload, udev, libasound2 (>= 1.0.24.1), libc6 (>= 2.4) Homepage: https://www.alsa-project.org/ Download-Size: 25,0 kB APT-Manual-Installed: yes APT-Sources: http://archive.ubuntu.com/ubuntu focal/multiverse amd64 Packages Description: ALSA software loaders for specific hardware $ sudo apt show alsa-tools-gui Package: alsa-tools-gui Version: 1.1.7-1ubuntu1 Priority: extra Section: universe/sound Source: alsa-tools Origin: Ubuntu Maintainer: Ubuntu Developers <ubuntu-devel-discuss(a)lists.ubuntu.com> Original-Maintainer: Debian ALSA Maintainers <pkg-alsa-devel(a)lists.alioth.debian.org> Bugs: https://bugs.launchpad.net/ubuntu/+filebug Installed-Size: 1.002 kB Depends: libasound2 (>= 1.0.24.1), libc6 (>= 2.29), libfltk1.1 (>= 1.1.6), libglib2.0-0 (>= 2.35.9), libgtk-3-0 (>= 3.0.0), libgtk2.0-0 (>= 2.8.0), libstdc++6 (>= 5.2) Homepage: https://www.alsa-project.org/ Task: ubuntustudio-audio Download-Size: 220 kB APT-Manual-Installed: yes APT-Sources: http://archive.ubuntu.com/ubuntu focal/universe amd64 Packages Description: GUI based ALSA utilities for specific hardware ``` Issue URL : https://github.com/alsa-project/alsa-firmware/issues/11 Repository URL: https://github.com/alsa-project/alsa-firmware

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regresion: snd_hdsp
by GitHub issues - edited 06 Mar '25

06 Mar '25

alsa-project/alsa-firmware issue #11 was edited from juanpc2018: ![Image](https://github.com/user-attachments/assets/e57192a7-ad1c-44f9-b51d-7806efed9a32) ``` $ pulseaudio W: [pulseaudio] module.c: module-combine is deprecated: Please use module-combine-sink instead of module-combine! W: [pulseaudio] module-combine.c: We will now load module-combine-sink. Please make sure to remove module-combine from your configuration. E: [alsa-sink-RME Hammerfall HDSP 9632] alsa-sink.c: ALSA woke us up to write new data to the device, but there was actually nothing to write. E: [alsa-sink-RME Hammerfall HDSP 9632] alsa-sink.c: Most likely this is a bug in the ALSA driver 'snd_hdsp'. Please report this issue to the ALSA developers. E: [alsa-sink-RME Hammerfall HDSP 9632] alsa-sink.c: We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. ``` when i play any video or sound file, most of the time PulseAudio takes between 7 to 15 seconds to fill the buffer in proper order, audio is played in wrong order, chopped, like [stutter](https://www.youtube.com/watch?v=9ZaDn5wne8Y&t=609s).[2](https://ww… i have tested /etc/pulse/deamon.conf s24le to s24be problem persists. forced sample rate to 48Khz, Audacity 3.6. ALSA hw:0 also Crash. i remember hdsp9632 worked ok on U/Kubuntu 10.04 & 10.10 Now 20.04.4 LTS has problems. im using Liquorix 6.3 Kernel. ``` $ sudo apt show alsa-firmware* Package: alsa-firmware-loaders Version: 1.1.7-1ubuntu1 Priority: extra Section: multiverse/sound Source: alsa-tools Origin: Ubuntu Maintainer: Ubuntu Developers <ubuntu-devel-discuss(a)lists.ubuntu.com> Original-Maintainer: Debian ALSA Maintainers <pkg-alsa-devel(a)lists.alioth.debian.org> Bugs: https://bugs.launchpad.net/ubuntu/+filebug Installed-Size: 211 kB Depends: fxload, udev, libasound2 (>= 1.0.24.1), libc6 (>= 2.4) Homepage: https://www.alsa-project.org/ Download-Size: 25,0 kB APT-Manual-Installed: yes APT-Sources: http://archive.ubuntu.com/ubuntu focal/multiverse amd64 Packages Description: ALSA software loaders for specific hardware $ sudo apt show alsa-tools-gui Package: alsa-tools-gui Version: 1.1.7-1ubuntu1 Priority: extra Section: universe/sound Source: alsa-tools Origin: Ubuntu Maintainer: Ubuntu Developers <ubuntu-devel-discuss(a)lists.ubuntu.com> Original-Maintainer: Debian ALSA Maintainers <pkg-alsa-devel(a)lists.alioth.debian.org> Bugs: https://bugs.launchpad.net/ubuntu/+filebug Installed-Size: 1.002 kB Depends: libasound2 (>= 1.0.24.1), libc6 (>= 2.29), libfltk1.1 (>= 1.1.6), libglib2.0-0 (>= 2.35.9), libgtk-3-0 (>= 3.0.0), libgtk2.0-0 (>= 2.8.0), libstdc++6 (>= 5.2) Homepage: https://www.alsa-project.org/ Task: ubuntustudio-audio Download-Size: 220 kB APT-Manual-Installed: yes APT-Sources: http://archive.ubuntu.com/ubuntu focal/universe amd64 Packages Description: GUI based ALSA utilities for specific hardware ``` Issue URL : https://github.com/alsa-project/alsa-firmware/issues/11 Repository URL: https://github.com/alsa-project/alsa-firmware

1 0

regresion: snd_hdsp
by GitHub issues - edited 06 Mar '25

06 Mar '25

alsa-project/alsa-firmware issue #11 was edited from juanpc2018: ![Image](https://github.com/user-attachments/assets/e57192a7-ad1c-44f9-b51d-7806efed9a32) ``` $ pulseaudio W: [pulseaudio] module.c: module-combine is deprecated: Please use module-combine-sink instead of module-combine! W: [pulseaudio] module-combine.c: We will now load module-combine-sink. Please make sure to remove module-combine from your configuration. E: [alsa-sink-RME Hammerfall HDSP 9632] alsa-sink.c: ALSA woke us up to write new data to the device, but there was actually nothing to write. E: [alsa-sink-RME Hammerfall HDSP 9632] alsa-sink.c: Most likely this is a bug in the ALSA driver 'snd_hdsp'. Please report this issue to the ALSA developers. E: [alsa-sink-RME Hammerfall HDSP 9632] alsa-sink.c: We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. ``` when i play any video or sound file, most of the time PulseAudio takes between 7 to 15 seconds to fill the buffer in proper order, audio is played in wrong order, chopped, like [stutter](https://www.youtube.com/watch?v=9ZaDn5wne8Y&t=609s).[2](https://ww… i have tested /etc/pulse/deamon.conf s24le to s24be problem persists. forced sample rate to 48Khz, Audacity 3.6. ALSA also crash. i remember hdsp9632 worked ok on U/Kubuntu 10.04 & 10.10 Now 20.04.4 LTS has problems. im using Liquorix 6.3 Kernel. ``` $ sudo apt show alsa-firmware* Package: alsa-firmware-loaders Version: 1.1.7-1ubuntu1 Priority: extra Section: multiverse/sound Source: alsa-tools Origin: Ubuntu Maintainer: Ubuntu Developers <ubuntu-devel-discuss(a)lists.ubuntu.com> Original-Maintainer: Debian ALSA Maintainers <pkg-alsa-devel(a)lists.alioth.debian.org> Bugs: https://bugs.launchpad.net/ubuntu/+filebug Installed-Size: 211 kB Depends: fxload, udev, libasound2 (>= 1.0.24.1), libc6 (>= 2.4) Homepage: https://www.alsa-project.org/ Download-Size: 25,0 kB APT-Manual-Installed: yes APT-Sources: http://archive.ubuntu.com/ubuntu focal/multiverse amd64 Packages Description: ALSA software loaders for specific hardware $ sudo apt show alsa-tools-gui Package: alsa-tools-gui Version: 1.1.7-1ubuntu1 Priority: extra Section: universe/sound Source: alsa-tools Origin: Ubuntu Maintainer: Ubuntu Developers <ubuntu-devel-discuss(a)lists.ubuntu.com> Original-Maintainer: Debian ALSA Maintainers <pkg-alsa-devel(a)lists.alioth.debian.org> Bugs: https://bugs.launchpad.net/ubuntu/+filebug Installed-Size: 1.002 kB Depends: libasound2 (>= 1.0.24.1), libc6 (>= 2.29), libfltk1.1 (>= 1.1.6), libglib2.0-0 (>= 2.35.9), libgtk-3-0 (>= 3.0.0), libgtk2.0-0 (>= 2.8.0), libstdc++6 (>= 5.2) Homepage: https://www.alsa-project.org/ Task: ubuntustudio-audio Download-Size: 220 kB APT-Manual-Installed: yes APT-Sources: http://archive.ubuntu.com/ubuntu focal/universe amd64 Packages Description: GUI based ALSA utilities for specific hardware ``` Issue URL : https://github.com/alsa-project/alsa-firmware/issues/11 Repository URL: https://github.com/alsa-project/alsa-firmware

1 0

regresion: snd_hdsp
by GitHub issues - edited 06 Mar '25

06 Mar '25

alsa-project/alsa-firmware issue #11 was edited from juanpc2018: ![Image](https://github.com/user-attachments/assets/e57192a7-ad1c-44f9-b51d-7806efed9a32) ``` $ pulseaudio W: [pulseaudio] module.c: module-combine is deprecated: Please use module-combine-sink instead of module-combine! W: [pulseaudio] module-combine.c: We will now load module-combine-sink. Please make sure to remove module-combine from your configuration. E: [alsa-sink-RME Hammerfall HDSP 9632] alsa-sink.c: ALSA woke us up to write new data to the device, but there was actually nothing to write. E: [alsa-sink-RME Hammerfall HDSP 9632] alsa-sink.c: Most likely this is a bug in the ALSA driver 'snd_hdsp'. Please report this issue to the ALSA developers. E: [alsa-sink-RME Hammerfall HDSP 9632] alsa-sink.c: We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. ``` when i play any video or sound file, most of the time PulseAudio takes between 7 to 15 seconds to fill the buffer in proper order, audio is played in wrong order, chopped, like [stutter](https://www.youtube.com/watch?v=9ZaDn5wne8Y&t=609s).[2](https://ww… i have tested /etc/pulse/deamon.conf s24le to s24be problem persists. forced sample rate to 48Khz, i remember hdsp9632 worked ok on U/Kubuntu 10.04 & 10.10 Now 20.04.4 LTS has problems. im using Liquorix 6.3 Kernel. ``` $ sudo apt show alsa-firmware* Package: alsa-firmware-loaders Version: 1.1.7-1ubuntu1 Priority: extra Section: multiverse/sound Source: alsa-tools Origin: Ubuntu Maintainer: Ubuntu Developers <ubuntu-devel-discuss(a)lists.ubuntu.com> Original-Maintainer: Debian ALSA Maintainers <pkg-alsa-devel(a)lists.alioth.debian.org> Bugs: https://bugs.launchpad.net/ubuntu/+filebug Installed-Size: 211 kB Depends: fxload, udev, libasound2 (>= 1.0.24.1), libc6 (>= 2.4) Homepage: https://www.alsa-project.org/ Download-Size: 25,0 kB APT-Manual-Installed: yes APT-Sources: http://archive.ubuntu.com/ubuntu focal/multiverse amd64 Packages Description: ALSA software loaders for specific hardware $ sudo apt show alsa-tools-gui Package: alsa-tools-gui Version: 1.1.7-1ubuntu1 Priority: extra Section: universe/sound Source: alsa-tools Origin: Ubuntu Maintainer: Ubuntu Developers <ubuntu-devel-discuss(a)lists.ubuntu.com> Original-Maintainer: Debian ALSA Maintainers <pkg-alsa-devel(a)lists.alioth.debian.org> Bugs: https://bugs.launchpad.net/ubuntu/+filebug Installed-Size: 1.002 kB Depends: libasound2 (>= 1.0.24.1), libc6 (>= 2.29), libfltk1.1 (>= 1.1.6), libglib2.0-0 (>= 2.35.9), libgtk-3-0 (>= 3.0.0), libgtk2.0-0 (>= 2.8.0), libstdc++6 (>= 5.2) Homepage: https://www.alsa-project.org/ Task: ubuntustudio-audio Download-Size: 220 kB APT-Manual-Installed: yes APT-Sources: http://archive.ubuntu.com/ubuntu focal/universe amd64 Packages Description: GUI based ALSA utilities for specific hardware ``` Issue URL : https://github.com/alsa-project/alsa-firmware/issues/11 Repository URL: https://github.com/alsa-project/alsa-firmware

1 0

regresion: snd_hdsp
by GitHub issues - edited 06 Mar '25

06 Mar '25

alsa-project/alsa-firmware issue #11 was edited from juanpc2018: ![Image](https://github.com/user-attachments/assets/e57192a7-ad1c-44f9-b51d-7806efed9a32) ``` $ pulseaudio W: [pulseaudio] module.c: module-combine is deprecated: Please use module-combine-sink instead of module-combine! W: [pulseaudio] module-combine.c: We will now load module-combine-sink. Please make sure to remove module-combine from your configuration. E: [alsa-sink-RME Hammerfall HDSP 9632] alsa-sink.c: ALSA woke us up to write new data to the device, but there was actually nothing to write. E: [alsa-sink-RME Hammerfall HDSP 9632] alsa-sink.c: Most likely this is a bug in the ALSA driver 'snd_hdsp'. Please report this issue to the ALSA developers. E: [alsa-sink-RME Hammerfall HDSP 9632] alsa-sink.c: We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. ``` when i play any video or sound file, most of the time PulseAudio takes between 7 to 15 seconds to fill the buffer in proper order, audio is played in wrong order, chopped, like [stutter](https://www.youtube.com/watch?v=9ZaDn5wne8Y&t=609s) i have tested /etc/pulse/deamon.conf s24le to s24be problem persists. forced sample rate to 48Khz, i remember hdsp9632 worked ok on U/Kubuntu 10.04 & 10.10 Now 20.04.4 LTS has problems. im using Liquorix 6.3 Kernel. ``` $ sudo apt show alsa-firmware* Package: alsa-firmware-loaders Version: 1.1.7-1ubuntu1 Priority: extra Section: multiverse/sound Source: alsa-tools Origin: Ubuntu Maintainer: Ubuntu Developers <ubuntu-devel-discuss(a)lists.ubuntu.com> Original-Maintainer: Debian ALSA Maintainers <pkg-alsa-devel(a)lists.alioth.debian.org> Bugs: https://bugs.launchpad.net/ubuntu/+filebug Installed-Size: 211 kB Depends: fxload, udev, libasound2 (>= 1.0.24.1), libc6 (>= 2.4) Homepage: https://www.alsa-project.org/ Download-Size: 25,0 kB APT-Manual-Installed: yes APT-Sources: http://archive.ubuntu.com/ubuntu focal/multiverse amd64 Packages Description: ALSA software loaders for specific hardware $ sudo apt show alsa-tools-gui Package: alsa-tools-gui Version: 1.1.7-1ubuntu1 Priority: extra Section: universe/sound Source: alsa-tools Origin: Ubuntu Maintainer: Ubuntu Developers <ubuntu-devel-discuss(a)lists.ubuntu.com> Original-Maintainer: Debian ALSA Maintainers <pkg-alsa-devel(a)lists.alioth.debian.org> Bugs: https://bugs.launchpad.net/ubuntu/+filebug Installed-Size: 1.002 kB Depends: libasound2 (>= 1.0.24.1), libc6 (>= 2.29), libfltk1.1 (>= 1.1.6), libglib2.0-0 (>= 2.35.9), libgtk-3-0 (>= 3.0.0), libgtk2.0-0 (>= 2.8.0), libstdc++6 (>= 5.2) Homepage: https://www.alsa-project.org/ Task: ubuntustudio-audio Download-Size: 220 kB APT-Manual-Installed: yes APT-Sources: http://archive.ubuntu.com/ubuntu focal/universe amd64 Packages Description: GUI based ALSA utilities for specific hardware ``` Issue URL : https://github.com/alsa-project/alsa-firmware/issues/11 Repository URL: https://github.com/alsa-project/alsa-firmware

1 0

regresion: snd_hdsp
by GitHub issues - edited 06 Mar '25

06 Mar '25

alsa-project/alsa-firmware issue #11 was edited from juanpc2018: ![Image](https://github.com/user-attachments/assets/e57192a7-ad1c-44f9-b51d-7806efed9a32) ``` $ pulseaudio W: [pulseaudio] module.c: module-combine is deprecated: Please use module-combine-sink instead of module-combine! W: [pulseaudio] module-combine.c: We will now load module-combine-sink. Please make sure to remove module-combine from your configuration. E: [alsa-sink-RME Hammerfall HDSP 9632] alsa-sink.c: ALSA woke us up to write new data to the device, but there was actually nothing to write. E: [alsa-sink-RME Hammerfall HDSP 9632] alsa-sink.c: Most likely this is a bug in the ALSA driver 'snd_hdsp'. Please report this issue to the ALSA developers. E: [alsa-sink-RME Hammerfall HDSP 9632] alsa-sink.c: We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. ``` when i play any video or sound file, most of the time PulseAudio takes between 7 to 15 seconds to fill the buffer in proper order, audio is played in wrong order, chopped, like stutter. i have tested /etc/pulse/deamon.conf s24le to s24be problem persists. forced sample rate to 48Khz, i remember hdsp9632 worked ok on U/Kubuntu 10.04 & 10.10 Now 20.04.4 LTS has problems. im using Liquorix 6.3 Kernel. ``` $ sudo apt show alsa-firmware* Package: alsa-firmware-loaders Version: 1.1.7-1ubuntu1 Priority: extra Section: multiverse/sound Source: alsa-tools Origin: Ubuntu Maintainer: Ubuntu Developers <ubuntu-devel-discuss(a)lists.ubuntu.com> Original-Maintainer: Debian ALSA Maintainers <pkg-alsa-devel(a)lists.alioth.debian.org> Bugs: https://bugs.launchpad.net/ubuntu/+filebug Installed-Size: 211 kB Depends: fxload, udev, libasound2 (>= 1.0.24.1), libc6 (>= 2.4) Homepage: https://www.alsa-project.org/ Download-Size: 25,0 kB APT-Manual-Installed: yes APT-Sources: http://archive.ubuntu.com/ubuntu focal/multiverse amd64 Packages Description: ALSA software loaders for specific hardware $ sudo apt show alsa-tools-gui Package: alsa-tools-gui Version: 1.1.7-1ubuntu1 Priority: extra Section: universe/sound Source: alsa-tools Origin: Ubuntu Maintainer: Ubuntu Developers <ubuntu-devel-discuss(a)lists.ubuntu.com> Original-Maintainer: Debian ALSA Maintainers <pkg-alsa-devel(a)lists.alioth.debian.org> Bugs: https://bugs.launchpad.net/ubuntu/+filebug Installed-Size: 1.002 kB Depends: libasound2 (>= 1.0.24.1), libc6 (>= 2.29), libfltk1.1 (>= 1.1.6), libglib2.0-0 (>= 2.35.9), libgtk-3-0 (>= 3.0.0), libgtk2.0-0 (>= 2.8.0), libstdc++6 (>= 5.2) Homepage: https://www.alsa-project.org/ Task: ubuntustudio-audio Download-Size: 220 kB APT-Manual-Installed: yes APT-Sources: http://archive.ubuntu.com/ubuntu focal/universe amd64 Packages Description: GUI based ALSA utilities for specific hardware ``` Issue URL : https://github.com/alsa-project/alsa-firmware/issues/11 Repository URL: https://github.com/alsa-project/alsa-firmware

1 0

regresion: snd_hdsp
by GitHub issues - opened 06 Mar '25

06 Mar '25

alsa-project/alsa-firmware issue #11 was opened from juanpc2018: ![Image](https://github.com/user-attachments/assets/e57192a7-ad1c-44f9-b51d-7806efed9a32) ``` $ pulseaudio W: [pulseaudio] module.c: module-combine is deprecated: Please use module-combine-sink instead of module-combine! W: [pulseaudio] module-combine.c: We will now load module-combine-sink. Please make sure to remove module-combine from your configuration. E: [alsa-sink-RME Hammerfall HDSP 9632] alsa-sink.c: ALSA woke us up to write new data to the device, but there was actually nothing to write. E: [alsa-sink-RME Hammerfall HDSP 9632] alsa-sink.c: Most likely this is a bug in the ALSA driver 'snd_hdsp'. Please report this issue to the ALSA developers. E: [alsa-sink-RME Hammerfall HDSP 9632] alsa-sink.c: We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. ``` when i play any video or sound file, most of the time PulseAudio takes between 7 to 15 seconds to fill the buffer in proper order, audio is played in wrong order, chopped, like stutter. i have tested /etc/pulse/deamon.conf s24le to s24be problem persists. forced sample rate to 48Khz, i remember hdsp9632 worked ok on U/Kubuntu 10.04 & 10.10 Now 20.04.4 LTS has problems. im using Liquorix 6.3 Kernel. Issue URL : https://github.com/alsa-project/alsa-firmware/issues/11 Repository URL: https://github.com/alsa-project/alsa-firmware

1 0

XRUN handling in pcm driver
by noman pouigt 05 Mar '25

05 Mar '25

Hello Maintainers, I have a question regarding XRUN handling. After an XRUN is triggered, userspace generally calls TRIGGER_STOP followed by PREPARE and TRIGGER_START. In the PREPARE stage, we are currently not sending the ALSA ring buffer (physical address, rate, format, etc.) IPC to the DSP but instead doing that in HW_PARAMS which is causing bad audio after XRUN is triggered. I was able to resolve the issue by moving the IPC to PREPARE, which fixes the XRUN problem and results in good audio even after an XRUN. I have also observed a similar approach in Qualcomm drivers, which I assume is for the same reason. Is moving the IPC to PREPARE the correct approach for handling this issue? I would appreciate any insights or guidance on whether this is the recommended way to handle XRUN recovery. Thanks,

2 1

RE: [PATCH v4 1/2] ASoC: codecs: add support for ES8389
by Zhang Yi 05 Mar '25

05 Mar '25

> > + ret = device_property_read_u8(codec->dev, "everest,adc-slot", &es8389->adc_slot); > > + if (ret != 0) { > > + dev_dbg(codec->dev, "adc-slot return %d", ret); > > + es8389->adc_slot = 0x00; > > + } > > + > > + ret = device_property_read_u8(codec->dev, "everest,dac-slot", &es8389->dac_slot); > > + if (ret != 0) { > > + dev_dbg(codec->dev, "dac-slot return %d", ret); > > + es8389->dac_slot = 0x00; > > + } > > set_tdm_slot() > > Please don't ignore review comments, people are generally making them > for a reason and are likely to have the same concerns if issues remain > unaddressed. Having to repeat the same comments can get repetitive and > make people question the value of time spent reviewing. If you disagree > with the review comments that's fine but you need to reply and discuss > your concerns so that the reviewer can understand your decisions. We will register multiple codecs inside a single dai_link and differentiate these codecs by of_node. And the adc_slot and the dac_slot may be different, they can be decided by the user.If we use set_tdm_slot,the adc_slot and the dac_slot will be same. > > + if (!es8389->adc_slot) { > > + es8389->mclk = devm_clk_get(codec->dev, "mclk"); > > + if (IS_ERR(es8389->mclk)) > > + return dev_err_probe(codec->dev, PTR_ERR(es8389->mclk), > > + "ES8389 is unable to get mclk\n"); > > + > > + if (!es8389->mclk) > > + dev_err(codec->dev, "%s, assuming static mclk\n", __func__); > > + > > + ret = clk_prepare_enable(es8389->mclk); > > + if (ret) { > > + dev_err(codec->dev, "%s, unable to enable mclk\n", __func__); > > + return ret; > > + } > > + } > > Making the use of a MCLK depend on the configuration of a TDM slot for > the ADC seems *very* unusual, what's going on there? Because we are associating multiple codecs under a single dai_link, we will be executing probe and init many times during initialization.But MCLK only needs to be used once.So we decided making the use of a MCLK depend on the configuration of a TDM slot for the ADC

2 1

update ALC222 depop optimize
by Kailang 05 Mar '25

05 Mar '25

Hi Takashi, Update ALC222 depop optimize as attach. BR, Kailang

2 8

[PATCH] ALSA: hda: intel: Add Dell ALC3271 to power_save denylist
by Hoku Ishibe 05 Mar '25

05 Mar '25

Dell XPS 13 7390 with the Realtek ALC3271 codec experiences persistent humming noise when the power_save mode is enabled. This issue occurs when the codec enters power saving mode, leading to unwanted noise from the speakers. This patch adds the affected model (PCI ID 0x1028:0x0962) to the power_save denylist to ensure power_save is disabled by default, preventing power-off related noise issues. Steps to Reproduce 1. Boot the system with `snd_hda_intel` loaded. 2. Verify that `power_save` mode is enabled: ```sh cat /sys/module/snd_hda_intel/parameters/power_save ```` output: 10 (default power save timeout) 3. Wait for the power save timeout 4. Observe a persistent humming noise from the speakers 5. Disable `power_save` manually: ```sh echo 0 | sudo tee /sys/module/snd_hda_intel/parameters/power_save ```` 6. Confirm that the noise disappears immediately. This issue has been observed on my system, and this patch successfully eliminates the unwanted noise. If other users experience similar issues, additional reports would be helpful. Signed-off-by: Hoku Ishibe <me(a)hokuishi.be> --- sound/pci/hda/hda_intel.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 7d7f9aac50a9..ff536a0e480c 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2232,6 +2232,8 @@ static const struct snd_pci_quirk power_save_denylist[] = { SND_PCI_QUIRK(0x1631, 0xe017, "Packard Bell NEC IMEDIA 5204", 0), /* KONTRON SinglePC may cause a stall at runtime resume */ SND_PCI_QUIRK(0x1734, 0x1232, "KONTRON SinglePC", 0), + /* Dell ALC3271 */ + SND_PCI_QUIRK(0x1028, 0x0962, "Dell ALC3271", 0), {} }; -- 2.48.1

2 1

RE: [PATCH v4 1/2] ASoC: codecs: add support for ES8389
by Zhang Yi 05 Mar '25

05 Mar '25

> > @@ -0,0 +1,961 @@ > > +// SPDX-License-Identifier: GPL-2.0-only > > +/* > > + * es8389.c -- ES8389/ES8390 ALSA SoC Audio Codec > > + * > > + * Copyright (C) 2025 Everest Semiconductor Co., Ltd > > Please make the entire comment block a C++ one so things look more > consistent. I will fix it > > + if (es8389->dmic == true) { > > + regmap_update_bits(es8389->regmap, ES8389_DMIC_EN, 0xC0, 0xC0); > > + regmap_update_bits(es8389->regmap, ES8389_ADC_MODE, 0x03, 0x03); > > + } else { > > + regmap_update_bits(es8389->regmap, ES8389_DMIC_EN, 0xC0, 0x00); > > + regmap_update_bits(es8389->regmap, ES8389_ADC_MODE, 0x03, 0x00); > > + } > > We also had the DMIC mux, is that useful as a runtime control when we > have firmware data telling us if there's a DMIC? Can both a DMIC and > analog input be present in the same system? > > It does still look like a lot of these settings might be things that > should be user controllable... I'm going to remove es8389->dmic and everest,dmic-enabled and use only DMIC_MUX

1 0

RE: [PATCH v3 2/2] ASoC: dt-bindings: Add Everest ES8389 audio CODEC
by Zhang Yi 05 Mar '25

05 Mar '25

> > > >>> + > >>> + prefix_name: > >> > >> No underscores in node names, missing vendor prefix... but more > >> important: I don't understand the need for this property. Description copies property name so it is not useful. > > And these comments as well. The prefix_name is used to distinguish the dapm and kcontrol of each individual codec. And I will update name of the property > > > >>> + everest,dmic-enabled: > >>> + $ref: /schemas/types.yaml#/definitions/flag > >>> + description: > >>> + The property is a choice between PDM and AMIC > >>> + > >> > >> No supplies? > > > > I will drop the property > > I did not comment about dmic, but meant missing power supplies, which > usually are placed here. I apologize for misunderstanding your comment. But we don't have properties about the power supplies

1 0

RE: [PATCH v3 2/2] ASoC: dt-bindings: Add Everest ES8389 audio CODEC
by Zhang Yi 05 Mar '25

05 Mar '25

> > + > > +properties: > > + compatible: > > + const: everest,es8389 > > Device is really different than es8388? yes, it's different from es8388 > > + "#sound-dai-cells": > > + const: 0 > > + > > + everest,adc-slot: > > + $ref: /schemas/types.yaml#/definitions/uint8 > > + description: | > > + This property is used to set the slots of recording data when multiple > > + codecs are connected in PTDM mode. > > + please set this property to default if you are setting STDM mode. > > + minimum: 0x00 > > Drop, unsigned does not allow -1. > > > + maximum: 0x07 > > + default: 0x00 > > All of these should be rather decimal. I will try to fix the error > > + > > + prefix_name: > > No underscores in node names, missing vendor prefix... but more > important: I don't understand the need for this property. Description copies property name so it is not useful. > > Drop. You maybe wanted dai prefix, but this is already in dai-common. I will update description at v4 PATCH > > + everest,dmic-enabled: > > + $ref: /schemas/types.yaml#/definitions/flag > > + description: > > + The property is a choice between PDM and AMIC > > + > > No supplies? I will drop the property > > +required: > > + - compatible > > + - reg > > + - "#sound-dai-cells" > > + > > +additionalProperties: false > > + > > +examples: > > + - | > > + i2c{ > > Missing space. > > > + #address-cells = <1>; > > + #size-cells = <0>; > > + es8389: es8389@10 { > > Node names should be generic. See also an explanation and list of examples (not exhaustive) in DT specification: > https://devicetree-specification.readthedocs.io/en/latest/chapter2-devicetr… > > > + compatible = "everest,es8389"; > > + reg = <0x10>; > > + everest,adc-slot = [00]; > > Please open other bindings or DTS and take a look how single number is encoded. In general, please rather base your work on latest bindings and DTS accepted by reviewers and use them as learning/template to avoid common mistakes. At least two comments in this review could be avoided if you did this. > > > + everest,dac-slot = [00]; > > + prefix_name = "es8389_0"; > > Drop I will try to fix it

1 0

[PATCH v4 0/2] ASoC: codecs: add support for ES8389
by Zhang Yi 04 Mar '25

04 Mar '25

The driver is for codec ES8389 of everest-semi. Zhang Yi (2): ASoC: codecs: add support for ES8389 ASoC: dt-bindings: Add Everest ES8389 audio CODEC .../bindings/sound/everest,es8389.yaml | 83 ++ sound/soc/codecs/Kconfig | 6 +- sound/soc/codecs/Makefile | 2 + sound/soc/codecs/es8389.c | 961 ++++++++++++++++++ sound/soc/codecs/es8389.h | 139 +++ 5 files changed, 1190 insertions(+), 1 deletion(-) create mode 100644 Documentation/devicetree/bindings/sound/everest,es8389.yaml create mode 100644 sound/soc/codecs/es8389.c create mode 100644 sound/soc/codecs/es8389.h -- 2.17.1

2 3

[PATCH v3 0/2] ASoC: codecs: add support for ES8389
by Zhang Yi 04 Mar '25

04 Mar '25

The driver is for codec ES8389 of everest-semi. change in v3: - Added reg_format at everest,es8389.yaml Zhang Yi (2): ASoC: codecs: add support for ES8389 ASoC: dt-bindings: Add Everest ES8389 audio CODEC .../bindings/sound/everest,es8389.yaml | 82 ++ sound/soc/codecs/Kconfig | 6 +- sound/soc/codecs/Makefile | 2 + sound/soc/codecs/es8389.c | 971 ++++++++++++++++++ sound/soc/codecs/es8389.h | 139 +++ 5 files changed, 1199 insertions(+), 1 deletion(-) create mode 100644 Documentation/devicetree/bindings/sound/everest,es8389.yaml create mode 100644 sound/soc/codecs/es8389.c create mode 100644 sound/soc/codecs/es8389.h -- 2.17.1

2 3

[PATCH v2 0/2] ASoC: codecs: add support for ES8389
by Zhang Yi 04 Mar '25

04 Mar '25

The driver is for codec ES8389 of everest-semi. Zhang Yi (2): ASoC: codecs: add support for ES8389 ASoC: dt-bindings: Add Everest ES8389 audio CODEC .../bindings/sound/everest,es8389.yaml | 78 ++ sound/soc/codecs/Kconfig | 6 +- sound/soc/codecs/Makefile | 2 + sound/soc/codecs/es8389.c | 971 ++++++++++++++++++ sound/soc/codecs/es8389.h | 139 +++ 5 files changed, 1195 insertions(+), 1 deletion(-) create mode 100644 Documentation/devicetree/bindings/sound/everest,es8389.yaml create mode 100644 sound/soc/codecs/es8389.c create mode 100644 sound/soc/codecs/es8389.h -- 2.17.1

2 3

[PATCH v5] ASoC: tas2781: Support dsp firmware Alpha and Beta seaies
by Shenghao Ding 03 Mar '25

03 Mar '25

For calibration, basic version does not contain any calibration addresses, it depends on calibration tool to convey the addresses to the driver. Since Alpha and Beta firmware, all the calibration addresses are saved into the firmware. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v5: - Instead of the lengthy approach, just replace if-continue with else if, has reduced this by 2/3 of LoCs - Use proper error code in fw_parse_fct_param_address() - return directly in tasdevice_dspfw_ready() - drop Unneeded assignment v4: - drop unnecceary "unsigned short size;" and "char *data;" in struct fct_param_address - replace strstr() with strncmp() in fct_param_address_parser() - replace hex_parse_u8() with memcpy() - replace hex_parse_u24() with direct assignment - drop hex_parse_u8() & hex_parse_u24() - Just return directly and use proper error codes and drop "goto out" and "out:" label in fw_parse_fct_param_address - fixed indentation issues in cali_reg_update - fixed the logical splits in tas2781_calib_stop_put() - return -ENOMEM if kmemdup fails in fw_parse_fct_param_address() - remove unnecessary "out:" label v3: - Merge the 4 patches into one patch v2: - | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: | https://urldefense.com/v3/__https://lore.kernel.org/oe-kbuild-all/2025 | 02160812.P7szOygN-lkp(a)intel.com/__;!!G3vK!WriwhtbQGlg6p00bcWutlhb-y9vS | pDrSu5_g9oOzzGOoABdirEARTntFzyX6Ss3UPRDxi_E-IA$ - | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: | https://urldefense.com/v3/__https://lore.kernel.org/oe-kbuild-all/2025 | 02160632.u0zxG7yD-lkp(a)intel.com/__;!!G3vK!VtEdfur3fZklvPVfHFFfFP60Ub2E | JYvmjSUZ65lJ6BsQPJDl1fjYRDqh3c6Q5y6sA70ssXDG5g$ v1: - update the year from 2024 to 2025 in the files. - Add dspbin_type to distinguish different dspbin types, such as TASDEV_BASIC, TASDEV_ALPHA, and TASDEV_BETA. - Add struct bulk_reg_val alp_cali_bckp to store the value of pilot tone threshold only for TASDEV_ALPHA & TASDEV_BETA firmware. - Add the declaration of (*fw_parse_fct_param_address)() to parse the addresses of calibration params. - Add the minimum version of TASDEV_ALPHA and TASDEV_BETA firmware. - Add the definition of struct fct_param_address. - update the year from 2024 to 2025. - release the memory to save calibration addresses in tasdevice_dsp_remove(). - Add TASDEV_ALPHA and TASDEV_BETA firmware checking code. - Add fct_param_address_parser for Alpha and Beta firmware. - Add an extra branch to handle calibration address for TASDEV_ALPHA and TASDEV_BETA firmware. --- include/sound/tas2781-dsp.h | 30 ++++- include/sound/tas2781.h | 11 ++ sound/pci/hda/tas2781_spi_fwlib.c | 8 +- sound/soc/codecs/tas2781-fmwlib.c | 190 ++++++++++++++++++++++---- sound/soc/codecs/tas2781-i2c.c | 215 ++++++++++++++++++++++-------- 5 files changed, 368 insertions(+), 86 deletions(-) diff --git a/include/sound/tas2781-dsp.h b/include/sound/tas2781-dsp.h index 3cda9da14f6d..c3a9efa73d5d 100644 --- a/include/sound/tas2781-dsp.h +++ b/include/sound/tas2781-dsp.h @@ -2,7 +2,7 @@ // // ALSA SoC Texas Instruments TAS2781 Audio Smart Amplifier // -// Copyright (C) 2022 - 2024 Texas Instruments Incorporated +// Copyright (C) 2022 - 2025 Texas Instruments Incorporated // https://www.ti.com // // The TAS2781 driver implements a flexible and configurable @@ -30,8 +30,10 @@ #define PRE_DEVICE_C 0x12 #define PRE_DEVICE_D 0x16 -#define PPC3_VERSION 0x4100 -#define PPC3_VERSION_TAS2781 0x14600 +#define PPC3_VERSION_BASE 0x4100 +#define PPC3_VERSION_TAS2781_BASIC_MIN 0x14600 +#define PPC3_VERSION_TAS2781_ALPHA_MIN 0x4a00 +#define PPC3_VERSION_TAS2781_BETA_MIN 0x19400 #define TASDEVICE_DEVICE_SUM 8 #define TASDEVICE_CONFIG_SUM 64 @@ -106,6 +108,27 @@ struct tasdevice_calibration { struct tasdevice_data dev_data; }; +struct fct_param_address { + /* Thermal data for PG 1.0 device */ + unsigned char thr[3]; + /* Thermal data for PG 2.0 device */ + unsigned char thr2[3]; + /* Pilot tone enable flag, usually the sine wave */ + unsigned char plt_flg[3]; + /* Pilot tone gain for calibration */ + unsigned char sin_gn[3]; + /* Pilot tone gain for calibration */ + unsigned char sin_gn2[3]; + /* high 32-bit of real-time spk impedance */ + unsigned char r0_reg[3]; + /* check spk connection */ + unsigned char tf_reg[3]; + /* check spk resonant frequency */ + unsigned char a1_reg[3]; + /* check spk resonant frequency */ + unsigned char a2_reg[3]; +}; + struct tasdevice_fw { struct tasdevice_dspfw_hdr fw_hdr; unsigned short nr_programs; @@ -114,6 +137,7 @@ struct tasdevice_fw { struct tasdevice_config *configs; unsigned short nr_calibrations; struct tasdevice_calibration *calibrations; + struct fct_param_address fct_par_addr; struct device *dev; }; diff --git a/include/sound/tas2781.h b/include/sound/tas2781.h index 4d9a37544354..eff011444cc8 100644 --- a/include/sound/tas2781.h +++ b/include/sound/tas2781.h @@ -113,6 +113,12 @@ enum audio_device { TAS2781, }; +enum dspbin_type { + TASDEV_BASIC, + TASDEV_ALPHA, + TASDEV_BETA, +}; + enum device_catlog_id { LENOVO = 0, OTHERS @@ -127,6 +133,7 @@ struct bulk_reg_val { struct tasdevice { struct bulk_reg_val *cali_data_backup; + struct bulk_reg_val alp_cali_bckp; struct tasdevice_fw *cali_data_fmw; unsigned int dev_addr; unsigned int err_code; @@ -171,6 +178,7 @@ struct tasdevice_priv { unsigned char dev_name[32]; const char *name_prefix; unsigned char ndev; + unsigned int dspbin_typ; unsigned int magic_num; unsigned int chip_id; unsigned int sysclk; @@ -196,6 +204,9 @@ struct tasdevice_priv { int (*fw_parse_configuration_data)(struct tasdevice_priv *tas_priv, struct tasdevice_fw *tas_fmw, const struct firmware *fmw, int offset); + int (*fw_parse_fct_param_address)(struct tasdevice_priv *tas_priv, + struct tasdevice_fw *tas_fmw, + const struct firmware *fmw, int offset); int (*tasdevice_load_block)(struct tasdevice_priv *tas_priv, struct tasdev_blk *block); diff --git a/sound/pci/hda/tas2781_spi_fwlib.c b/sound/pci/hda/tas2781_spi_fwlib.c index 0e2acbc3c900..19a693d58ee8 100644 --- a/sound/pci/hda/tas2781_spi_fwlib.c +++ b/sound/pci/hda/tas2781_spi_fwlib.c @@ -2,7 +2,7 @@ // // TAS2781 HDA SPI driver // -// Copyright 2024 Texas Instruments, Inc. +// Copyright 2024 - 2025 Texas Instruments, Inc. // // Author: Baojun Xu <baojun.xu(a)ti.com> @@ -370,10 +370,10 @@ static unsigned char map_dev_idx(struct tasdevice_fw *tas_fmw, int i, n = ARRAY_SIZE(non_ppc3_mapping_table); unsigned char dev_idx = 0; - if (fw_fixed_hdr->ppcver >= PPC3_VERSION_TAS2781) { + if (fw_fixed_hdr->ppcver >= PPC3_VERSION_TAS2781_BASIC_MIN) { p = (struct blktyp_devidx_map *)ppc3_tas2781_mapping_table; n = ARRAY_SIZE(ppc3_tas2781_mapping_table); - } else if (fw_fixed_hdr->ppcver >= PPC3_VERSION) { + } else if (fw_fixed_hdr->ppcver >= PPC3_VERSION_BASE) { p = (struct blktyp_devidx_map *)ppc3_mapping_table; n = ARRAY_SIZE(ppc3_mapping_table); } @@ -1605,7 +1605,7 @@ static int dspfw_default_callback(struct tasdevice_priv *tas_priv, int rc = 0; if (drv_ver == 0x100) { - if (ppcver >= PPC3_VERSION) { + if (ppcver >= PPC3_VERSION_BASE) { tas_priv->fw_parse_variable_header = fw_parse_variable_header_kernel; tas_priv->fw_parse_program_data = diff --git a/sound/soc/codecs/tas2781-fmwlib.c b/sound/soc/codecs/tas2781-fmwlib.c index bc2f68bd9846..13a197468193 100644 --- a/sound/soc/codecs/tas2781-fmwlib.c +++ b/sound/soc/codecs/tas2781-fmwlib.c @@ -390,10 +390,10 @@ static unsigned char map_dev_idx(struct tasdevice_fw *tas_fmw, int i, n = ARRAY_SIZE(non_ppc3_mapping_table); unsigned char dev_idx = 0; - if (fw_fixed_hdr->ppcver >= PPC3_VERSION_TAS2781) { + if (fw_fixed_hdr->ppcver >= PPC3_VERSION_TAS2781_BASIC_MIN) { p = (struct blktyp_devidx_map *)ppc3_tas2781_mapping_table; n = ARRAY_SIZE(ppc3_tas2781_mapping_table); - } else if (fw_fixed_hdr->ppcver >= PPC3_VERSION) { + } else if (fw_fixed_hdr->ppcver >= PPC3_VERSION_BASE) { p = (struct blktyp_devidx_map *)ppc3_mapping_table; n = ARRAY_SIZE(ppc3_mapping_table); } @@ -560,6 +560,124 @@ static int fw_parse_configuration_data_kernel( return offset; } +static void fct_param_address_parser(struct cali_reg *r, + struct tasdevice_fw *tas_fmw, const unsigned char *data) +{ + struct fct_param_address *p = &tas_fmw->fct_par_addr; + unsigned int i; + + /* + * Calibration parameters locations and data schema in dsp firmware. + * The number of items are flexible, but not more than 20. The dsp tool + * will reseve 20*24-byte space for fct params. In some cases, the + * number of fct param is less than 20, the data will be saved from the + * beginning, the rest part will be stuffed with zero. + * + * fct_param_num (not more than 20) + * for (i = 0; i < fct_param_num; i++) { + * Alias of fct param (20 bytes) + * Book (1 byte) + * Page (1 byte) + * Offset (1 byte) + * CoeffLength (1 byte) = 0x1 + * } + * if (20 - fct_param_num) + * 24*(20 - fct_param_num) pieces of '0' as stuffing + * + * As follow: + * umg_SsmKEGCye = Book, Page, Offset, CoeffLength + * iks_E0 = Book, Page, Offset, CoeffLength + * yep_LsqM0 = Book, Page, Offset, CoeffLength + * oyz_U0_ujx = Book, Page, Offset, CoeffLength + * iks_GC_GMgq = Book, Page, Offset, CoeffLength + * gou_Yao = Book, Page, Offset, CoeffLength + * kgd_Wsc_Qsbp = Book, Page, Offset, CoeffLength + * yec_CqseSsqs = Book, Page, Offset, CoeffLength + * iks_SogkGgog2 = Book, Page, Offset, CoeffLength + * yec_Sae_Y = Book, Page, Offset, CoeffLength + * Re_Int = Book, Page, Offset, CoeffLength + * SigFlag = Book, Page, Offset, CoeffLength + * a1_Int = Book, Page, Offset, CoeffLength + * a2_Int = Book, Page, Offset, CoeffLength + */ + for (i = 0; i < 20; i++) { + const unsigned char *dat = &data[24 * i]; + + /* + * check whether current fct param is empty. + */ + if (dat[23] != 1) + break; + + if (!strncmp(dat, "umg_SsmKEGCye", 20)) + r->pow_reg = TASDEVICE_REG(dat[20], dat[21], dat[22]); + /* high 32-bit of real-time spk impedance */ + else if (!strncmp(dat, "iks_E0", 20)) + r->r0_reg = TASDEVICE_REG(dat[20], dat[21], dat[22]); + /* inverse of real-time spk impedance */ + else if (!strncmp(dat, "yep_LsqM0", 20)) + r->invr0_reg = + TASDEVICE_REG(dat[20], dat[21], dat[22]); + /* low 32-bit of real-time spk impedance */ + else if (!strncmp(dat, "oyz_U0_ujx", 20)) + r->r0_low_reg = + TASDEVICE_REG(dat[20], dat[21], dat[22]); + /* Delta Thermal Limit */ + else if (!strncmp(dat, "iks_GC_GMgq", 20)) + r->tlimit_reg = + TASDEVICE_REG(dat[20], dat[21], dat[22]); + /* Thermal data for PG 1.0 device */ + else if (!strncmp(dat, "gou_Yao", 20)) + memcpy(p->thr, &dat[20], 3); + /* Pilot tone enable flag, usually the sine wave */ + else if (!strncmp(dat, "kgd_Wsc_Qsbp", 20)) + memcpy(p->plt_flg, &dat[20], 3); + /* Pilot tone gain for calibration */ + else if (!strncmp(dat, "yec_CqseSsqs", 20)) + memcpy(p->sin_gn, &dat[20], 3); + /* Pilot tone gain for calibration, useless in PG 2.0 */ + else if (!strncmp(dat, "iks_SogkGgog2", 20)) + memcpy(p->sin_gn2, &dat[20], 3); + /* Thermal data for PG 2.0 device */ + else if (!strncmp(dat, "yec_Sae_Y", 20)) + memcpy(p->thr2, &dat[20], 3); + /* Spk Equivalent Resistance in fixed-point format */ + else if (!strncmp(dat, "Re_Int", 20)) + memcpy(p->r0_reg, &dat[20], 3); + /* Check whether the spk connection is open */ + else if (!strncmp(dat, "SigFlag", 20)) + memcpy(p->tf_reg, &dat[20], 3); + /* check spk resonant frequency */ + else if (!strncmp(dat, "a1_Int", 20)) + memcpy(p->a1_reg, &dat[20], 3); + /* check spk resonant frequency */ + else if (!strncmp(dat, "a2_Int", 20)) + memcpy(p->a2_reg, &dat[20], 3); + } +} + +static int fw_parse_fct_param_address(struct tasdevice_priv *tas_priv, + struct tasdevice_fw *tas_fmw, const struct firmware *fmw, int offset) +{ + struct calidata *cali_data = &tas_priv->cali_data; + struct cali_reg *r = &cali_data->cali_reg_array; + const unsigned char *data = fmw->data; + + if (offset + 520 > fmw->size) { + dev_err(tas_priv->dev, "%s: File Size error\n", __func__); + return -EINVAL; + } + + /* skip reserved part */ + offset += 40; + + fct_param_address_parser(r, tas_fmw, &data[offset]); + + offset += 480; + + return offset; +} + static int fw_parse_variable_header_kernel( struct tasdevice_priv *tas_priv, const struct firmware *fmw, int offset) @@ -1687,13 +1805,29 @@ static int tasdevice_load_block(struct tasdevice_priv *tas_priv, return rc; } +static void dspbin_type_check(struct tasdevice_priv *tas_priv, + unsigned int ppcver) +{ + if (ppcver >= PPC3_VERSION_TAS2781_ALPHA_MIN) { + if (ppcver >= PPC3_VERSION_TAS2781_BETA_MIN) + tas_priv->dspbin_typ = TASDEV_BETA; + else if (ppcver >= PPC3_VERSION_TAS2781_BASIC_MIN) + tas_priv->dspbin_typ = TASDEV_BASIC; + else + tas_priv->dspbin_typ = TASDEV_ALPHA; + } + if (tas_priv->dspbin_typ != TASDEV_BASIC) + tas_priv->fw_parse_fct_param_address = + fw_parse_fct_param_address; +} + static int dspfw_default_callback(struct tasdevice_priv *tas_priv, unsigned int drv_ver, unsigned int ppcver) { int rc = 0; if (drv_ver == 0x100) { - if (ppcver >= PPC3_VERSION) { + if (ppcver >= PPC3_VERSION_BASE) { tas_priv->fw_parse_variable_header = fw_parse_variable_header_kernel; tas_priv->fw_parse_program_data = @@ -1702,6 +1836,7 @@ static int dspfw_default_callback(struct tasdevice_priv *tas_priv, fw_parse_configuration_data_kernel; tas_priv->tasdevice_load_block = tasdevice_load_block_kernel; + dspbin_type_check(tas_priv, ppcver); } else { switch (ppcver) { case 0x00: @@ -1717,7 +1852,7 @@ static int dspfw_default_callback(struct tasdevice_priv *tas_priv, default: dev_err(tas_priv->dev, "%s: PPCVer must be 0x0 or 0x%02x", - __func__, PPC3_VERSION); + __func__, PPC3_VERSION_BASE); dev_err(tas_priv->dev, " Current:0x%02x\n", ppcver); rc = -EINVAL; @@ -1953,28 +2088,25 @@ static int tasdevice_dspfw_ready(const struct firmware *fmw, struct tasdevice_fw_fixed_hdr *fw_fixed_hdr; struct tasdevice_fw *tas_fmw; int offset = 0; - int ret = 0; + int ret; if (!fmw || !fmw->data) { dev_err(tas_priv->dev, "%s: Failed to read firmware %s\n", __func__, tas_priv->coef_binaryname); - ret = -EINVAL; - goto out; + return -EINVAL; } tas_priv->fmw = kzalloc(sizeof(struct tasdevice_fw), GFP_KERNEL); - if (!tas_priv->fmw) { - ret = -ENOMEM; - goto out; - } + if (!tas_priv->fmw) + return -ENOMEM; + tas_fmw = tas_priv->fmw; tas_fmw->dev = tas_priv->dev; offset = fw_parse_header(tas_priv, tas_fmw, fmw, offset); - if (offset == -EINVAL) { - ret = -EINVAL; - goto out; - } + if (offset == -EINVAL) + return -EINVAL; + fw_fixed_hdr = &(tas_fmw->fw_hdr.fixed_hdr); /* Support different versions of firmware */ switch (fw_fixed_hdr->drv_ver) { @@ -2007,28 +2139,32 @@ static int tasdevice_dspfw_ready(const struct firmware *fmw, ret = dspfw_default_callback(tas_priv, fw_fixed_hdr->drv_ver, fw_fixed_hdr->ppcver); if (ret) - goto out; + return ret; break; } offset = tas_priv->fw_parse_variable_header(tas_priv, fmw, offset); - if (offset < 0) { - ret = offset; - goto out; - } + if (offset < 0) + return offset; + offset = tas_priv->fw_parse_program_data(tas_priv, tas_fmw, fmw, offset); - if (offset < 0) { - ret = offset; - goto out; - } + if (offset < 0) + return offset; + offset = tas_priv->fw_parse_configuration_data(tas_priv, tas_fmw, fmw, offset); if (offset < 0) - ret = offset; + return offset; -out: - return ret; + if (tas_priv->fw_parse_fct_param_address) { + offset = tas_priv->fw_parse_fct_param_address(tas_priv, + tas_fmw, fmw, offset); + if (offset < 0) + return offset; + } + + return 0; } int tasdevice_dsp_parser(void *context) diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index 90c5b2e74d12..b950914b7d48 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -349,13 +349,31 @@ static int calib_data_get(struct tasdevice_priv *tas_priv, int reg, return rc; } +static int partial_cali_data_update(int *reg, int j) +{ + switch (tas2781_cali_start_reg[j].reg) { + case 0: + return reg[0]; + case TAS2781_PRM_PLT_FLAG_REG: + return reg[1]; + case TAS2781_PRM_SINEGAIN_REG: + return reg[2]; + case TAS2781_PRM_SINEGAIN2_REG: + return reg[3]; + default: + return 0; + } +} + static void sngl_calib_start(struct tasdevice_priv *tas_priv, int i, int *reg, unsigned char *dat) { struct tasdevice *tasdev = tas_priv->tasdevice; struct bulk_reg_val *p = tasdev[i].cali_data_backup; + struct bulk_reg_val *t = &tasdev[i].alp_cali_bckp; const int sum = ARRAY_SIZE(tas2781_cali_start_reg); - int j; + unsigned char val[4]; + int j, r; if (p == NULL) return; @@ -370,30 +388,23 @@ static void sngl_calib_start(struct tasdevice_priv *tas_priv, int i, tasdevice_dev_read(tas_priv, i, p[j].reg, (int *)&p[j].val[0]); } else { - switch (tas2781_cali_start_reg[j].reg) { - case 0: { - if (!reg[0]) - continue; - p[j].reg = reg[0]; + if (!tas_priv->dspbin_typ) { + r = partial_cali_data_update(reg, j); + if (r) + p[j].reg = r; } - break; - case TAS2781_PRM_PLT_FLAG_REG: - p[j].reg = reg[1]; - break; - case TAS2781_PRM_SINEGAIN_REG: - p[j].reg = reg[2]; - break; - case TAS2781_PRM_SINEGAIN2_REG: - p[j].reg = reg[3]; - break; - } - tasdevice_dev_bulk_read(tas_priv, i, p[j].reg, - p[j].val, 4); + + if (p[j].reg) + tasdevice_dev_bulk_read(tas_priv, i, p[j].reg, + p[j].val, 4); } } + if (tas_priv->dspbin_typ == TASDEV_ALPHA) + tasdevice_dev_bulk_read(tas_priv, i, t->reg, t->val, 4); + /* Update the setting for calibration */ - for (j = 0; j < sum - 2; j++) { + for (j = 0; j < sum - 4; j++) { if (p[j].val_len == 1) { if (p[j].is_locked) tasdevice_dev_write(tas_priv, i, @@ -401,17 +412,33 @@ static void sngl_calib_start(struct tasdevice_priv *tas_priv, int i, TAS2781_TEST_PAGE_UNLOCK); tasdevice_dev_write(tas_priv, i, p[j].reg, tas2781_cali_start_reg[j].val[0]); - } else { - if (!p[j].reg) - continue; - tasdevice_dev_bulk_write(tas_priv, i, p[j].reg, - (unsigned char *) - tas2781_cali_start_reg[j].val, 4); } } - tasdevice_dev_bulk_write(tas_priv, i, p[j].reg, &dat[1], 4); - tasdevice_dev_bulk_write(tas_priv, i, p[j + 1].reg, &dat[5], 4); + if (tas_priv->dspbin_typ == TASDEV_ALPHA) { + val[0] = 0x00; + val[1] = 0x00; + val[2] = 0x21; + val[3] = 0x8e; + } else { + val[0] = tas2781_cali_start_reg[j].val[0]; + val[1] = tas2781_cali_start_reg[j].val[1]; + val[2] = tas2781_cali_start_reg[j].val[2]; + val[3] = tas2781_cali_start_reg[j].val[3]; + } + tasdevice_dev_bulk_write(tas_priv, i, p[j].reg, val, 4); + tasdevice_dev_bulk_write(tas_priv, i, p[j + 1].reg, + (unsigned char *)tas2781_cali_start_reg[j + 1].val, 4); + tasdevice_dev_bulk_write(tas_priv, i, p[j + 2].reg, &dat[1], 4); + tasdevice_dev_bulk_write(tas_priv, i, p[j + 3].reg, &dat[5], 4); + if (tas_priv->dspbin_typ == TASDEV_ALPHA) { + val[0] = 0x00; + val[1] = 0x00; + val[2] = 0x2a; + val[3] = 0x0b; + + tasdevice_dev_bulk_read(tas_priv, i, t->reg, val, 4); + } } static int tas2781_calib_start_put(struct snd_kcontrol *kcontrol, @@ -452,14 +479,15 @@ static int tas2781_calib_start_put(struct snd_kcontrol *kcontrol, return 1; } -static void tas2781_calib_stop_put(struct tasdevice_priv *tas_priv) +static void tas2781_calib_stop_put(struct tasdevice_priv *priv) { const int sum = ARRAY_SIZE(tas2781_cali_start_reg); int i, j; - for (i = 0; i < tas_priv->ndev; i++) { - struct tasdevice *tasdev = tas_priv->tasdevice; + for (i = 0; i < priv->ndev; i++) { + struct tasdevice *tasdev = priv->tasdevice; struct bulk_reg_val *p = tasdev[i].cali_data_backup; + struct bulk_reg_val *t = &tasdev[i].alp_cali_bckp; if (p == NULL) continue; @@ -467,18 +495,21 @@ static void tas2781_calib_stop_put(struct tasdevice_priv *tas_priv) for (j = 0; j < sum; j++) { if (p[j].val_len == 1) { if (p[j].is_locked) - tasdevice_dev_write(tas_priv, i, + tasdevice_dev_write(priv, i, TAS2781_TEST_UNLOCK_REG, TAS2781_TEST_PAGE_UNLOCK); - tasdevice_dev_write(tas_priv, i, p[j].reg, + tasdevice_dev_write(priv, i, p[j].reg, p[j].val[0]); } else { if (!p[j].reg) continue; - tasdevice_dev_bulk_write(tas_priv, i, p[j].reg, + tasdevice_dev_bulk_write(priv, i, p[j].reg, p[j].val, 4); } } + + if (priv->dspbin_typ == TASDEV_ALPHA) + tasdevice_dev_bulk_write(priv, i, t->reg, t->val, 4); } } @@ -590,16 +621,20 @@ static int tasdev_cali_data_put(struct snd_kcontrol *kcontrol, i += 2; priv->is_user_space_calidata = true; - p->r0_reg = TASDEVICE_REG(src[i], src[i + 1], src[i + 2]); - i += 3; - p->r0_low_reg = TASDEVICE_REG(src[i], src[i + 1], src[i + 2]); - i += 3; - p->invr0_reg = TASDEVICE_REG(src[i], src[i + 1], src[i + 2]); - i += 3; - p->pow_reg = TASDEVICE_REG(src[i], src[i + 1], src[i + 2]); - i += 3; - p->tlimit_reg = TASDEVICE_REG(src[i], src[i + 1], src[i + 2]); - i += 3; + if (priv->dspbin_typ == TASDEV_BASIC) { + p->r0_reg = TASDEVICE_REG(src[i], src[i + 1], src[i + 2]); + i += 3; + p->r0_low_reg = TASDEVICE_REG(src[i], src[i + 1], src[i + 2]); + i += 3; + p->invr0_reg = TASDEVICE_REG(src[i], src[i + 1], src[i + 2]); + i += 3; + p->pow_reg = TASDEVICE_REG(src[i], src[i + 1], src[i + 2]); + i += 3; + p->tlimit_reg = TASDEVICE_REG(src[i], src[i + 1], src[i + 2]); + i += 3; + } else { + i += 15; + } memcpy(dst, &src[i], cali_data->total_sz); return 1; @@ -646,12 +681,19 @@ static int tasdev_tf_data_get(struct snd_kcontrol *kcontrol, struct soc_bytes_ext *bytes_ext = (struct soc_bytes_ext *) kcontrol->private_value; unsigned char *dst = ucontrol->value.bytes.data; - unsigned int reg; + unsigned int reg = TAS2781_RUNTIME_RE_REG_TF; + + if (tas_priv->chip_id == TAS2781) { + struct tasdevice_fw *tas_fmw = tas_priv->fmw; + struct fct_param_address *p = &(tas_fmw->fct_par_addr); - if (tas_priv->chip_id == TAS2781) reg = TAS2781_RUNTIME_RE_REG_TF; - else + if (tas_priv->dspbin_typ) + reg = TASDEVICE_REG(p->tf_reg[0], p->tf_reg[1], + p->tf_reg[2]); + } else { reg = TAS2563_RUNTIME_RE_REG_TF; + } guard(mutex)(&tas_priv->codec_lock); dst[0] = bytes_ext->max; @@ -666,12 +708,19 @@ static int tasdev_re_data_get(struct snd_kcontrol *kcontrol, struct soc_bytes_ext *bytes_ext = (struct soc_bytes_ext *) kcontrol->private_value; unsigned char *dst = ucontrol->value.bytes.data; - unsigned int reg; + unsigned int reg = TAS2781_RUNTIME_RE_REG; - if (tas_priv->chip_id == TAS2781) - reg = TAS2781_RUNTIME_RE_REG; - else + if (tas_priv->chip_id == TAS2781) { + struct tasdevice_fw *tas_fmw = tas_priv->fmw; + struct fct_param_address *p = &(tas_fmw->fct_par_addr); + + if (tas_priv->dspbin_typ) + reg = TASDEVICE_REG(p->r0_reg[0], p->r0_reg[1], + p->r0_reg[2]); + } else { reg = TAS2563_RUNTIME_RE_REG; + } + guard(mutex)(&tas_priv->codec_lock); dst[0] = bytes_ext->max; return calib_data_get(tas_priv, reg, &dst[1]); @@ -705,11 +754,16 @@ static int tasdev_XMA1_data_get(struct snd_kcontrol *kcontrol, { struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + struct tasdevice_fw *tas_fmw = tas_priv->fmw; + struct fct_param_address *p = &(tas_fmw->fct_par_addr); struct soc_bytes_ext *bytes_ext = (struct soc_bytes_ext *) kcontrol->private_value; unsigned char *dst = ucontrol->value.bytes.data; unsigned int reg = TASDEVICE_XM_A1_REG; + if (tas_priv->dspbin_typ) + reg = TASDEVICE_REG(p->a1_reg[0], p->a1_reg[1], p->a1_reg[2]); + guard(mutex)(&tas_priv->codec_lock); dst[0] = bytes_ext->max; return calib_data_get(tas_priv, reg, &dst[1]); @@ -720,11 +774,16 @@ static int tasdev_XMA2_data_get(struct snd_kcontrol *kcontrol, { struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + struct tasdevice_fw *tas_fmw = tas_priv->fmw; + struct fct_param_address *p = &(tas_fmw->fct_par_addr); struct soc_bytes_ext *bytes_ext = (struct soc_bytes_ext *) kcontrol->private_value; unsigned char *dst = ucontrol->value.bytes.data; unsigned int reg = TASDEVICE_XM_A2_REG; + if (tas_priv->dspbin_typ) + reg = TASDEVICE_REG(p->a2_reg[0], p->a2_reg[1], p->a2_reg[2]); + guard(mutex)(&tas_priv->codec_lock); dst[0] = bytes_ext->max; return calib_data_get(tas_priv, reg, &dst[1]); @@ -1172,10 +1231,51 @@ static int tasdevice_dsp_create_ctrls(struct tasdevice_priv *tas_priv) nr_controls < mix_index ? nr_controls : mix_index); } +static void cali_reg_update(struct bulk_reg_val *p, + struct fct_param_address *t) +{ + const int sum = ARRAY_SIZE(tas2781_cali_start_reg); + int reg, j; + + for (j = 0; j < sum; j++) { + switch (tas2781_cali_start_reg[j].reg) { + case 0: + reg = TASDEVICE_REG(t->thr[0], t->thr[1], t->thr[2]); + break; + case TAS2781_PRM_PLT_FLAG_REG: + reg = TASDEVICE_REG(t->plt_flg[0], t->plt_flg[1], + t->plt_flg[2]); + break; + case TAS2781_PRM_SINEGAIN_REG: + reg = TASDEVICE_REG(t->sin_gn[0], t->sin_gn[1], + t->sin_gn[2]); + break; + case TAS2781_PRM_SINEGAIN2_REG: + reg = TASDEVICE_REG(t->sin_gn[0], t->sin_gn[1], + t->sin_gn[2]); + break; + default: + reg = 0; + break; + } + if (reg) + p[j].reg = reg; + } +} + +static void alpa_cali_update(struct bulk_reg_val *p, + struct fct_param_address *t) +{ + p->is_locked = false; + p->reg = TASDEVICE_REG(t->thr2[0], t->thr2[1], t->thr2[2]); + p->val_len = 4; +} + static int tasdevice_create_cali_ctrls(struct tasdevice_priv *priv) { struct calidata *cali_data = &priv->cali_data; struct tasdevice *tasdev = priv->tasdevice; + struct tasdevice_fw *fmw = priv->fmw; struct soc_bytes_ext *ext_cali_data; struct snd_kcontrol_new *cali_ctrls; unsigned int nctrls; @@ -1191,14 +1291,25 @@ static int tasdevice_create_cali_ctrls(struct tasdevice_priv *priv) } if (priv->chip_id == TAS2781) { + struct fct_param_address *t = &(fmw->fct_par_addr); + cali_ctrls = (struct snd_kcontrol_new *)tas2781_cali_controls; nctrls = ARRAY_SIZE(tas2781_cali_controls); for (i = 0; i < priv->ndev; i++) { - tasdev[i].cali_data_backup = + struct bulk_reg_val *p; + + p = tasdev[i].cali_data_backup = kmemdup(tas2781_cali_start_reg, sizeof(tas2781_cali_start_reg), GFP_KERNEL); if (!tasdev[i].cali_data_backup) return -ENOMEM; + if (priv->dspbin_typ) { + cali_reg_update(p, t); + if (priv->dspbin_typ == TASDEV_ALPHA) { + p = &tasdev[i].alp_cali_bckp; + alpa_cali_update(p, t); + } + } } } else { cali_ctrls = (struct snd_kcontrol_new *)tas2563_cali_controls; -- 2.34.1

1 0

[PATCH v4] ASoC: tas2781: Support dsp firmware Alpha and Beta seaies
by Shenghao Ding 03 Mar '25

03 Mar '25

For calibration, basic version does not contain any calibration addresses, it depends on calibration tool to convey the addresses to the driver. Since Alpha and Beta firmware, all the calibration addresses are saved into the firmware. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v4: - drop unnecceary "unsigned short size;" and "char *data;" in struct fct_param_address - replace strstr() with strncmp() in fct_param_address_parser() - replace hex_parse_u8() with memcpy() - replace hex_parse_u24() with direct assignment - drop hex_parse_u8() & hex_parse_u24() - Just return directly and use proper error codes and drop "goto out" and "out:" label in fw_parse_fct_param_address - fixed indentation issues in cali_reg_update - fixed the logical splits in tas2781_calib_stop_put() - return -ENOMEM if kmemdup fails in fw_parse_fct_param_address() - remove unnecessary "out:" label v3: - Merge the 4 patches into one patch v2: - | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: | https://urldefense.com/v3/__https://lore.kernel.org/oe-kbuild-all/2025 | 02160812.P7szOygN-lkp(a)intel.com/__;!!G3vK!WriwhtbQGlg6p00bcWutlhb-y9vS | pDrSu5_g9oOzzGOoABdirEARTntFzyX6Ss3UPRDxi_E-IA$ - | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: | https://urldefense.com/v3/__https://lore.kernel.org/oe-kbuild-all/2025 | 02160632.u0zxG7yD-lkp(a)intel.com/__;!!G3vK!VtEdfur3fZklvPVfHFFfFP60Ub2E | JYvmjSUZ65lJ6BsQPJDl1fjYRDqh3c6Q5y6sA70ssXDG5g$ v1: - update the year from 2024 to 2025 in the files. - Add dspbin_type to distinguish different dspbin types, such as TASDEV_BASIC, TASDEV_ALPHA, and TASDEV_BETA. - Add struct bulk_reg_val alp_cali_bckp to store the value of pilot tone threshold only for TASDEV_ALPHA & TASDEV_BETA firmware. - Add the declaration of (*fw_parse_fct_param_address)() to parse the addresses of calibration params. - Add the minimum version of TASDEV_ALPHA and TASDEV_BETA firmware. - Add the definition of struct fct_param_address. - update the year from 2024 to 2025. - release the memory to save calibration addresses in tasdevice_dsp_remove(). - Add TASDEV_ALPHA and TASDEV_BETA firmware checking code. - Add fct_param_address_parser for Alpha and Beta firmware. - Add an extra branch to handle calibration address for TASDEV_ALPHA and TASDEV_BETA firmware. --- include/sound/tas2781-dsp.h | 30 ++++- include/sound/tas2781.h | 11 ++ sound/pci/hda/tas2781_spi_fwlib.c | 8 +- sound/soc/codecs/tas2781-fmwlib.c | 214 +++++++++++++++++++++++++---- sound/soc/codecs/tas2781-i2c.c | 216 +++++++++++++++++++++++------- 5 files changed, 395 insertions(+), 84 deletions(-) diff --git a/include/sound/tas2781-dsp.h b/include/sound/tas2781-dsp.h index 3cda9da14f6d..c3a9efa73d5d 100644 --- a/include/sound/tas2781-dsp.h +++ b/include/sound/tas2781-dsp.h @@ -2,7 +2,7 @@ // // ALSA SoC Texas Instruments TAS2781 Audio Smart Amplifier // -// Copyright (C) 2022 - 2024 Texas Instruments Incorporated +// Copyright (C) 2022 - 2025 Texas Instruments Incorporated // https://www.ti.com // // The TAS2781 driver implements a flexible and configurable @@ -30,8 +30,10 @@ #define PRE_DEVICE_C 0x12 #define PRE_DEVICE_D 0x16 -#define PPC3_VERSION 0x4100 -#define PPC3_VERSION_TAS2781 0x14600 +#define PPC3_VERSION_BASE 0x4100 +#define PPC3_VERSION_TAS2781_BASIC_MIN 0x14600 +#define PPC3_VERSION_TAS2781_ALPHA_MIN 0x4a00 +#define PPC3_VERSION_TAS2781_BETA_MIN 0x19400 #define TASDEVICE_DEVICE_SUM 8 #define TASDEVICE_CONFIG_SUM 64 @@ -106,6 +108,27 @@ struct tasdevice_calibration { struct tasdevice_data dev_data; }; +struct fct_param_address { + /* Thermal data for PG 1.0 device */ + unsigned char thr[3]; + /* Thermal data for PG 2.0 device */ + unsigned char thr2[3]; + /* Pilot tone enable flag, usually the sine wave */ + unsigned char plt_flg[3]; + /* Pilot tone gain for calibration */ + unsigned char sin_gn[3]; + /* Pilot tone gain for calibration */ + unsigned char sin_gn2[3]; + /* high 32-bit of real-time spk impedance */ + unsigned char r0_reg[3]; + /* check spk connection */ + unsigned char tf_reg[3]; + /* check spk resonant frequency */ + unsigned char a1_reg[3]; + /* check spk resonant frequency */ + unsigned char a2_reg[3]; +}; + struct tasdevice_fw { struct tasdevice_dspfw_hdr fw_hdr; unsigned short nr_programs; @@ -114,6 +137,7 @@ struct tasdevice_fw { struct tasdevice_config *configs; unsigned short nr_calibrations; struct tasdevice_calibration *calibrations; + struct fct_param_address fct_par_addr; struct device *dev; }; diff --git a/include/sound/tas2781.h b/include/sound/tas2781.h index 4d9a37544354..eff011444cc8 100644 --- a/include/sound/tas2781.h +++ b/include/sound/tas2781.h @@ -113,6 +113,12 @@ enum audio_device { TAS2781, }; +enum dspbin_type { + TASDEV_BASIC, + TASDEV_ALPHA, + TASDEV_BETA, +}; + enum device_catlog_id { LENOVO = 0, OTHERS @@ -127,6 +133,7 @@ struct bulk_reg_val { struct tasdevice { struct bulk_reg_val *cali_data_backup; + struct bulk_reg_val alp_cali_bckp; struct tasdevice_fw *cali_data_fmw; unsigned int dev_addr; unsigned int err_code; @@ -171,6 +178,7 @@ struct tasdevice_priv { unsigned char dev_name[32]; const char *name_prefix; unsigned char ndev; + unsigned int dspbin_typ; unsigned int magic_num; unsigned int chip_id; unsigned int sysclk; @@ -196,6 +204,9 @@ struct tasdevice_priv { int (*fw_parse_configuration_data)(struct tasdevice_priv *tas_priv, struct tasdevice_fw *tas_fmw, const struct firmware *fmw, int offset); + int (*fw_parse_fct_param_address)(struct tasdevice_priv *tas_priv, + struct tasdevice_fw *tas_fmw, + const struct firmware *fmw, int offset); int (*tasdevice_load_block)(struct tasdevice_priv *tas_priv, struct tasdev_blk *block); diff --git a/sound/pci/hda/tas2781_spi_fwlib.c b/sound/pci/hda/tas2781_spi_fwlib.c index 0e2acbc3c900..19a693d58ee8 100644 --- a/sound/pci/hda/tas2781_spi_fwlib.c +++ b/sound/pci/hda/tas2781_spi_fwlib.c @@ -2,7 +2,7 @@ // // TAS2781 HDA SPI driver // -// Copyright 2024 Texas Instruments, Inc. +// Copyright 2024 - 2025 Texas Instruments, Inc. // // Author: Baojun Xu <baojun.xu(a)ti.com> @@ -370,10 +370,10 @@ static unsigned char map_dev_idx(struct tasdevice_fw *tas_fmw, int i, n = ARRAY_SIZE(non_ppc3_mapping_table); unsigned char dev_idx = 0; - if (fw_fixed_hdr->ppcver >= PPC3_VERSION_TAS2781) { + if (fw_fixed_hdr->ppcver >= PPC3_VERSION_TAS2781_BASIC_MIN) { p = (struct blktyp_devidx_map *)ppc3_tas2781_mapping_table; n = ARRAY_SIZE(ppc3_tas2781_mapping_table); - } else if (fw_fixed_hdr->ppcver >= PPC3_VERSION) { + } else if (fw_fixed_hdr->ppcver >= PPC3_VERSION_BASE) { p = (struct blktyp_devidx_map *)ppc3_mapping_table; n = ARRAY_SIZE(ppc3_mapping_table); } @@ -1605,7 +1605,7 @@ static int dspfw_default_callback(struct tasdevice_priv *tas_priv, int rc = 0; if (drv_ver == 0x100) { - if (ppcver >= PPC3_VERSION) { + if (ppcver >= PPC3_VERSION_BASE) { tas_priv->fw_parse_variable_header = fw_parse_variable_header_kernel; tas_priv->fw_parse_program_data = diff --git a/sound/soc/codecs/tas2781-fmwlib.c b/sound/soc/codecs/tas2781-fmwlib.c index bc2f68bd9846..2007c25c5778 100644 --- a/sound/soc/codecs/tas2781-fmwlib.c +++ b/sound/soc/codecs/tas2781-fmwlib.c @@ -390,10 +390,10 @@ static unsigned char map_dev_idx(struct tasdevice_fw *tas_fmw, int i, n = ARRAY_SIZE(non_ppc3_mapping_table); unsigned char dev_idx = 0; - if (fw_fixed_hdr->ppcver >= PPC3_VERSION_TAS2781) { + if (fw_fixed_hdr->ppcver >= PPC3_VERSION_TAS2781_BASIC_MIN) { p = (struct blktyp_devidx_map *)ppc3_tas2781_mapping_table; n = ARRAY_SIZE(ppc3_tas2781_mapping_table); - } else if (fw_fixed_hdr->ppcver >= PPC3_VERSION) { + } else if (fw_fixed_hdr->ppcver >= PPC3_VERSION_BASE) { p = (struct blktyp_devidx_map *)ppc3_mapping_table; n = ARRAY_SIZE(ppc3_mapping_table); } @@ -560,6 +560,152 @@ static int fw_parse_configuration_data_kernel( return offset; } +static void fct_param_address_parser(struct cali_reg *r, + struct tasdevice_fw *tas_fmw, const unsigned char *data) +{ + struct fct_param_address *p = &tas_fmw->fct_par_addr; + unsigned int i; + + /* + * Calibration parameters locations and data schema in dsp firmware. + * The number of items are flexible, but not more than 20. The dsp tool + * will reseve 20*24-byte space for fct params. In some cases, the + * number of fct param is less than 20, the data will be saved from the + * beginning, the rest part will be stuffed with zero. + * + * fct_param_num (not more than 20) + * for (i = 0; i < fct_param_num; i++) { + * Alias of fct param (20 bytes) + * Book (1 byte) + * Page (1 byte) + * Offset (1 byte) + * CoeffLength (1 byte) = 0x1 + * } + * if (20 - fct_param_num) + * 24*(20 - fct_param_num) pieces of '0' as stuffing + * + * As follow: + * umg_SsmKEGCye = Book, Page, Offset, CoeffLength + * iks_E0 = Book, Page, Offset, CoeffLength + * yep_LsqM0 = Book, Page, Offset, CoeffLength + * oyz_U0_ujx = Book, Page, Offset, CoeffLength + * iks_GC_GMgq = Book, Page, Offset, CoeffLength + * gou_Yao = Book, Page, Offset, CoeffLength + * kgd_Wsc_Qsbp = Book, Page, Offset, CoeffLength + * yec_CqseSsqs = Book, Page, Offset, CoeffLength + * iks_SogkGgog2 = Book, Page, Offset, CoeffLength + * yec_Sae_Y = Book, Page, Offset, CoeffLength + * Re_Int = Book, Page, Offset, CoeffLength + * SigFlag = Book, Page, Offset, CoeffLength + * a1_Int = Book, Page, Offset, CoeffLength + * a2_Int = Book, Page, Offset, CoeffLength + */ + for (i = 0; i < 20; i++) { + const unsigned char *dat = &data[24 * i]; + + /* + * check whether current fct param is empty. + */ + if (dat[23] != 1) + break; + + if (!strncmp(dat, "umg_SsmKEGCye", 20)) { + r->pow_reg = TASDEVICE_REG(dat[20], dat[21], dat[22]); + continue; + } + /* high 32-bit of real-time spk impedance */ + if (!strncmp(dat, "iks_E0", 20)) { + r->r0_reg = TASDEVICE_REG(dat[20], dat[21], dat[22]); + continue; + } + /* inverse of real-time spk impedance */ + if (!strncmp(dat, "yep_LsqM0", 20)) { + r->invr0_reg = + TASDEVICE_REG(dat[20], dat[21], dat[22]); + continue; + } + /* low 32-bit of real-time spk impedance */ + if (!strncmp(dat, "oyz_U0_ujx", 20)) { + r->r0_low_reg = + TASDEVICE_REG(dat[20], dat[21], dat[22]); + continue; + } + /* Delta Thermal Limit */ + if (!strncmp(dat, "iks_GC_GMgq", 20)) { + r->tlimit_reg = + TASDEVICE_REG(dat[20], dat[21], dat[22]); + continue; + } + /* Thermal data for PG 1.0 device */ + if (!strncmp(dat, "gou_Yao", 20)) { + memcpy(p->thr, &dat[20], 3); + continue; + } + /* Pilot tone enable flag, usually the sine wave */ + if (!strncmp(dat, "kgd_Wsc_Qsbp", 20)) { + memcpy(p->plt_flg, &dat[20], 3); + continue; + } + /* Pilot tone gain for calibration */ + if (!strncmp(dat, "yec_CqseSsqs", 20)) { + memcpy(p->sin_gn, &dat[20], 3); + continue; + } + /* Pilot tone gain for calibration, useless in PG 2.0 */ + if (!strncmp(dat, "iks_SogkGgog2", 20)) { + memcpy(p->sin_gn2, &dat[20], 3); + continue; + } + /* Thermal data for PG 2.0 device */ + if (!strncmp(dat, "yec_Sae_Y", 20)) { + memcpy(p->thr2, &dat[20], 3); + continue; + } + /* Spk Equivalent Resistance in fixed-point format */ + if (!strncmp(dat, "Re_Int", 20)) { + memcpy(p->r0_reg, &dat[20], 3); + continue; + } + /* Check whether the spk connection is open */ + if (!strncmp(dat, "SigFlag", 20)) { + memcpy(p->tf_reg, &dat[20], 3); + continue; + } + /* check spk resonant frequency */ + if (!strncmp(dat, "a1_Int", 20)) { + memcpy(p->a1_reg, &dat[20], 3); + continue; + } + /* check spk resonant frequency */ + if (!strncmp(dat, "a2_Int", 20)) { + memcpy(p->a2_reg, &dat[20], 3); + continue; + } + } +} + +static int fw_parse_fct_param_address(struct tasdevice_priv *tas_priv, + struct tasdevice_fw *tas_fmw, const struct firmware *fmw, int offset) +{ + struct calidata *cali_data = &tas_priv->cali_data; + struct cali_reg *r = &cali_data->cali_reg_array; + const unsigned char *data = fmw->data; + + if (offset + 520 > fmw->size) { + dev_err(tas_priv->dev, "%s: File Size error\n", __func__); + return -1; + } + + /* skip reserved part */ + offset += 40; + + fct_param_address_parser(r, tas_fmw, &data[offset]); + + offset += 480; + + return offset; +} + static int fw_parse_variable_header_kernel( struct tasdevice_priv *tas_priv, const struct firmware *fmw, int offset) @@ -1687,13 +1833,29 @@ static int tasdevice_load_block(struct tasdevice_priv *tas_priv, return rc; } +static void dspbin_type_check(struct tasdevice_priv *tas_priv, + unsigned int ppcver) +{ + if (ppcver >= PPC3_VERSION_TAS2781_ALPHA_MIN) { + if (ppcver >= PPC3_VERSION_TAS2781_BETA_MIN) + tas_priv->dspbin_typ = TASDEV_BETA; + else if (ppcver >= PPC3_VERSION_TAS2781_BASIC_MIN) + tas_priv->dspbin_typ = TASDEV_BASIC; + else + tas_priv->dspbin_typ = TASDEV_ALPHA; + } + if (tas_priv->dspbin_typ != TASDEV_BASIC) + tas_priv->fw_parse_fct_param_address = + fw_parse_fct_param_address; +} + static int dspfw_default_callback(struct tasdevice_priv *tas_priv, unsigned int drv_ver, unsigned int ppcver) { int rc = 0; if (drv_ver == 0x100) { - if (ppcver >= PPC3_VERSION) { + if (ppcver >= PPC3_VERSION_BASE) { tas_priv->fw_parse_variable_header = fw_parse_variable_header_kernel; tas_priv->fw_parse_program_data = @@ -1702,6 +1864,7 @@ static int dspfw_default_callback(struct tasdevice_priv *tas_priv, fw_parse_configuration_data_kernel; tas_priv->tasdevice_load_block = tasdevice_load_block_kernel; + dspbin_type_check(tas_priv, ppcver); } else { switch (ppcver) { case 0x00: @@ -1717,7 +1880,7 @@ static int dspfw_default_callback(struct tasdevice_priv *tas_priv, default: dev_err(tas_priv->dev, "%s: PPCVer must be 0x0 or 0x%02x", - __func__, PPC3_VERSION); + __func__, PPC3_VERSION_BASE); dev_err(tas_priv->dev, " Current:0x%02x\n", ppcver); rc = -EINVAL; @@ -1958,23 +2121,20 @@ static int tasdevice_dspfw_ready(const struct firmware *fmw, if (!fmw || !fmw->data) { dev_err(tas_priv->dev, "%s: Failed to read firmware %s\n", __func__, tas_priv->coef_binaryname); - ret = -EINVAL; - goto out; + return -EINVAL; } tas_priv->fmw = kzalloc(sizeof(struct tasdevice_fw), GFP_KERNEL); - if (!tas_priv->fmw) { - ret = -ENOMEM; - goto out; - } + if (!tas_priv->fmw) + return -ENOMEM; + tas_fmw = tas_priv->fmw; tas_fmw->dev = tas_priv->dev; offset = fw_parse_header(tas_priv, tas_fmw, fmw, offset); - if (offset == -EINVAL) { - ret = -EINVAL; - goto out; - } + if (offset == -EINVAL) + return -EINVAL; + fw_fixed_hdr = &(tas_fmw->fw_hdr.fixed_hdr); /* Support different versions of firmware */ switch (fw_fixed_hdr->drv_ver) { @@ -2007,27 +2167,31 @@ static int tasdevice_dspfw_ready(const struct firmware *fmw, ret = dspfw_default_callback(tas_priv, fw_fixed_hdr->drv_ver, fw_fixed_hdr->ppcver); if (ret) - goto out; + return ret; break; } offset = tas_priv->fw_parse_variable_header(tas_priv, fmw, offset); - if (offset < 0) { - ret = offset; - goto out; - } + if (offset < 0) + return offset; + offset = tas_priv->fw_parse_program_data(tas_priv, tas_fmw, fmw, offset); - if (offset < 0) { - ret = offset; - goto out; - } + if (offset < 0) + return offset; + offset = tas_priv->fw_parse_configuration_data(tas_priv, tas_fmw, fmw, offset); if (offset < 0) - ret = offset; + return offset; + + if (tas_priv->fw_parse_fct_param_address) { + offset = tas_priv->fw_parse_fct_param_address(tas_priv, + tas_fmw, fmw, offset); + if (offset < 0) + ret = offset; + } -out: return ret; } diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index 90c5b2e74d12..7d0b119c828a 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -349,13 +349,31 @@ static int calib_data_get(struct tasdevice_priv *tas_priv, int reg, return rc; } +static int partial_cali_data_update(int *reg, int j) +{ + switch (tas2781_cali_start_reg[j].reg) { + case 0: + return reg[0]; + case TAS2781_PRM_PLT_FLAG_REG: + return reg[1]; + case TAS2781_PRM_SINEGAIN_REG: + return reg[2]; + case TAS2781_PRM_SINEGAIN2_REG: + return reg[3]; + default: + return 0; + } +} + static void sngl_calib_start(struct tasdevice_priv *tas_priv, int i, int *reg, unsigned char *dat) { struct tasdevice *tasdev = tas_priv->tasdevice; struct bulk_reg_val *p = tasdev[i].cali_data_backup; + struct bulk_reg_val *t = &tasdev[i].alp_cali_bckp; const int sum = ARRAY_SIZE(tas2781_cali_start_reg); - int j; + unsigned char val[4]; + int j, r; if (p == NULL) return; @@ -370,30 +388,23 @@ static void sngl_calib_start(struct tasdevice_priv *tas_priv, int i, tasdevice_dev_read(tas_priv, i, p[j].reg, (int *)&p[j].val[0]); } else { - switch (tas2781_cali_start_reg[j].reg) { - case 0: { - if (!reg[0]) - continue; - p[j].reg = reg[0]; + if (!tas_priv->dspbin_typ) { + r = partial_cali_data_update(reg, j); + if (r) + p[j].reg = r; } - break; - case TAS2781_PRM_PLT_FLAG_REG: - p[j].reg = reg[1]; - break; - case TAS2781_PRM_SINEGAIN_REG: - p[j].reg = reg[2]; - break; - case TAS2781_PRM_SINEGAIN2_REG: - p[j].reg = reg[3]; - break; - } - tasdevice_dev_bulk_read(tas_priv, i, p[j].reg, - p[j].val, 4); + + if (p[j].reg) + tasdevice_dev_bulk_read(tas_priv, i, p[j].reg, + p[j].val, 4); } } + if (tas_priv->dspbin_typ == TASDEV_ALPHA) + tasdevice_dev_bulk_read(tas_priv, i, t->reg, t->val, 4); + /* Update the setting for calibration */ - for (j = 0; j < sum - 2; j++) { + for (j = 0; j < sum - 4; j++) { if (p[j].val_len == 1) { if (p[j].is_locked) tasdevice_dev_write(tas_priv, i, @@ -401,17 +412,33 @@ static void sngl_calib_start(struct tasdevice_priv *tas_priv, int i, TAS2781_TEST_PAGE_UNLOCK); tasdevice_dev_write(tas_priv, i, p[j].reg, tas2781_cali_start_reg[j].val[0]); - } else { - if (!p[j].reg) - continue; - tasdevice_dev_bulk_write(tas_priv, i, p[j].reg, - (unsigned char *) - tas2781_cali_start_reg[j].val, 4); } } - tasdevice_dev_bulk_write(tas_priv, i, p[j].reg, &dat[1], 4); - tasdevice_dev_bulk_write(tas_priv, i, p[j + 1].reg, &dat[5], 4); + if (tas_priv->dspbin_typ == TASDEV_ALPHA) { + val[0] = 0x00; + val[1] = 0x00; + val[2] = 0x21; + val[3] = 0x8e; + } else { + val[0] = tas2781_cali_start_reg[j].val[0]; + val[1] = tas2781_cali_start_reg[j].val[1]; + val[2] = tas2781_cali_start_reg[j].val[2]; + val[3] = tas2781_cali_start_reg[j].val[3]; + } + tasdevice_dev_bulk_write(tas_priv, i, p[j].reg, val, 4); + tasdevice_dev_bulk_write(tas_priv, i, p[j + 1].reg, + (unsigned char *)tas2781_cali_start_reg[j + 1].val, 4); + tasdevice_dev_bulk_write(tas_priv, i, p[j + 2].reg, &dat[1], 4); + tasdevice_dev_bulk_write(tas_priv, i, p[j + 3].reg, &dat[5], 4); + if (tas_priv->dspbin_typ == TASDEV_ALPHA) { + val[0] = 0x00; + val[1] = 0x00; + val[2] = 0x2a; + val[3] = 0x0b; + + tasdevice_dev_bulk_read(tas_priv, i, t->reg, val, 4); + } } static int tas2781_calib_start_put(struct snd_kcontrol *kcontrol, @@ -452,14 +479,15 @@ static int tas2781_calib_start_put(struct snd_kcontrol *kcontrol, return 1; } -static void tas2781_calib_stop_put(struct tasdevice_priv *tas_priv) +static void tas2781_calib_stop_put(struct tasdevice_priv *priv) { const int sum = ARRAY_SIZE(tas2781_cali_start_reg); int i, j; - for (i = 0; i < tas_priv->ndev; i++) { - struct tasdevice *tasdev = tas_priv->tasdevice; + for (i = 0; i < priv->ndev; i++) { + struct tasdevice *tasdev = priv->tasdevice; struct bulk_reg_val *p = tasdev[i].cali_data_backup; + struct bulk_reg_val *t = &tasdev[i].alp_cali_bckp; if (p == NULL) continue; @@ -467,18 +495,21 @@ static void tas2781_calib_stop_put(struct tasdevice_priv *tas_priv) for (j = 0; j < sum; j++) { if (p[j].val_len == 1) { if (p[j].is_locked) - tasdevice_dev_write(tas_priv, i, + tasdevice_dev_write(priv, i, TAS2781_TEST_UNLOCK_REG, TAS2781_TEST_PAGE_UNLOCK); - tasdevice_dev_write(tas_priv, i, p[j].reg, + tasdevice_dev_write(priv, i, p[j].reg, p[j].val[0]); } else { if (!p[j].reg) continue; - tasdevice_dev_bulk_write(tas_priv, i, p[j].reg, + tasdevice_dev_bulk_write(priv, i, p[j].reg, p[j].val, 4); } } + + if (priv->dspbin_typ == TASDEV_ALPHA) + tasdevice_dev_bulk_write(priv, i, t->reg, t->val, 4); } } @@ -590,16 +621,20 @@ static int tasdev_cali_data_put(struct snd_kcontrol *kcontrol, i += 2; priv->is_user_space_calidata = true; - p->r0_reg = TASDEVICE_REG(src[i], src[i + 1], src[i + 2]); - i += 3; - p->r0_low_reg = TASDEVICE_REG(src[i], src[i + 1], src[i + 2]); - i += 3; - p->invr0_reg = TASDEVICE_REG(src[i], src[i + 1], src[i + 2]); - i += 3; - p->pow_reg = TASDEVICE_REG(src[i], src[i + 1], src[i + 2]); - i += 3; - p->tlimit_reg = TASDEVICE_REG(src[i], src[i + 1], src[i + 2]); - i += 3; + if (priv->dspbin_typ == TASDEV_BASIC) { + p->r0_reg = TASDEVICE_REG(src[i], src[i + 1], src[i + 2]); + i += 3; + p->r0_low_reg = TASDEVICE_REG(src[i], src[i + 1], src[i + 2]); + i += 3; + p->invr0_reg = TASDEVICE_REG(src[i], src[i + 1], src[i + 2]); + i += 3; + p->pow_reg = TASDEVICE_REG(src[i], src[i + 1], src[i + 2]); + i += 3; + p->tlimit_reg = TASDEVICE_REG(src[i], src[i + 1], src[i + 2]); + i += 3; + } else { + i += 15; + } memcpy(dst, &src[i], cali_data->total_sz); return 1; @@ -646,12 +681,19 @@ static int tasdev_tf_data_get(struct snd_kcontrol *kcontrol, struct soc_bytes_ext *bytes_ext = (struct soc_bytes_ext *) kcontrol->private_value; unsigned char *dst = ucontrol->value.bytes.data; - unsigned int reg; + unsigned int reg = TAS2781_RUNTIME_RE_REG_TF; + + if (tas_priv->chip_id == TAS2781) { + struct tasdevice_fw *tas_fmw = tas_priv->fmw; + struct fct_param_address *p = &(tas_fmw->fct_par_addr); - if (tas_priv->chip_id == TAS2781) reg = TAS2781_RUNTIME_RE_REG_TF; - else + if (tas_priv->dspbin_typ) + reg = TASDEVICE_REG(p->tf_reg[0], p->tf_reg[1], + p->tf_reg[2]); + } else { reg = TAS2563_RUNTIME_RE_REG_TF; + } guard(mutex)(&tas_priv->codec_lock); dst[0] = bytes_ext->max; @@ -666,12 +708,19 @@ static int tasdev_re_data_get(struct snd_kcontrol *kcontrol, struct soc_bytes_ext *bytes_ext = (struct soc_bytes_ext *) kcontrol->private_value; unsigned char *dst = ucontrol->value.bytes.data; - unsigned int reg; + unsigned int reg = TAS2781_RUNTIME_RE_REG; - if (tas_priv->chip_id == TAS2781) - reg = TAS2781_RUNTIME_RE_REG; - else + if (tas_priv->chip_id == TAS2781) { + struct tasdevice_fw *tas_fmw = tas_priv->fmw; + struct fct_param_address *p = &(tas_fmw->fct_par_addr); + + if (tas_priv->dspbin_typ) + reg = TASDEVICE_REG(p->r0_reg[0], p->r0_reg[1], + p->r0_reg[2]); + } else { reg = TAS2563_RUNTIME_RE_REG; + } + guard(mutex)(&tas_priv->codec_lock); dst[0] = bytes_ext->max; return calib_data_get(tas_priv, reg, &dst[1]); @@ -705,11 +754,16 @@ static int tasdev_XMA1_data_get(struct snd_kcontrol *kcontrol, { struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + struct tasdevice_fw *tas_fmw = tas_priv->fmw; + struct fct_param_address *p = &(tas_fmw->fct_par_addr); struct soc_bytes_ext *bytes_ext = (struct soc_bytes_ext *) kcontrol->private_value; unsigned char *dst = ucontrol->value.bytes.data; unsigned int reg = TASDEVICE_XM_A1_REG; + if (tas_priv->dspbin_typ) + reg = TASDEVICE_REG(p->a1_reg[0], p->a1_reg[1], p->a1_reg[2]); + guard(mutex)(&tas_priv->codec_lock); dst[0] = bytes_ext->max; return calib_data_get(tas_priv, reg, &dst[1]); @@ -720,11 +774,16 @@ static int tasdev_XMA2_data_get(struct snd_kcontrol *kcontrol, { struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + struct tasdevice_fw *tas_fmw = tas_priv->fmw; + struct fct_param_address *p = &(tas_fmw->fct_par_addr); struct soc_bytes_ext *bytes_ext = (struct soc_bytes_ext *) kcontrol->private_value; unsigned char *dst = ucontrol->value.bytes.data; unsigned int reg = TASDEVICE_XM_A2_REG; + if (tas_priv->dspbin_typ) + reg = TASDEVICE_REG(p->a2_reg[0], p->a2_reg[1], p->a2_reg[2]); + guard(mutex)(&tas_priv->codec_lock); dst[0] = bytes_ext->max; return calib_data_get(tas_priv, reg, &dst[1]); @@ -1172,10 +1231,52 @@ static int tasdevice_dsp_create_ctrls(struct tasdevice_priv *tas_priv) nr_controls < mix_index ? nr_controls : mix_index); } +static void cali_reg_update(struct bulk_reg_val *p, + struct fct_param_address *t) +{ + const int sum = ARRAY_SIZE(tas2781_cali_start_reg); + int reg = 0; + int j; + + for (j = 0; j < sum; j++) { + switch (tas2781_cali_start_reg[j].reg) { + case 0: + reg = TASDEVICE_REG(t->thr[0], t->thr[1], t->thr[2]); + break; + case TAS2781_PRM_PLT_FLAG_REG: + reg = TASDEVICE_REG(t->plt_flg[0], t->plt_flg[1], + t->plt_flg[2]); + break; + case TAS2781_PRM_SINEGAIN_REG: + reg = TASDEVICE_REG(t->sin_gn[0], t->sin_gn[1], + t->sin_gn[2]); + break; + case TAS2781_PRM_SINEGAIN2_REG: + reg = TASDEVICE_REG(t->sin_gn[0], t->sin_gn[1], + t->sin_gn[2]); + break; + default: + reg = 0; + break; + } + if (reg) + p[j].reg = reg; + } +} + +static void alpa_cali_update(struct bulk_reg_val *p, + struct fct_param_address *t) +{ + p->is_locked = false; + p->reg = TASDEVICE_REG(t->thr2[0], t->thr2[1], t->thr2[2]); + p->val_len = 4; +} + static int tasdevice_create_cali_ctrls(struct tasdevice_priv *priv) { struct calidata *cali_data = &priv->cali_data; struct tasdevice *tasdev = priv->tasdevice; + struct tasdevice_fw *fmw = priv->fmw; struct soc_bytes_ext *ext_cali_data; struct snd_kcontrol_new *cali_ctrls; unsigned int nctrls; @@ -1191,14 +1292,25 @@ static int tasdevice_create_cali_ctrls(struct tasdevice_priv *priv) } if (priv->chip_id == TAS2781) { + struct fct_param_address *t = &(fmw->fct_par_addr); + cali_ctrls = (struct snd_kcontrol_new *)tas2781_cali_controls; nctrls = ARRAY_SIZE(tas2781_cali_controls); for (i = 0; i < priv->ndev; i++) { - tasdev[i].cali_data_backup = + struct bulk_reg_val *p; + + p = tasdev[i].cali_data_backup = kmemdup(tas2781_cali_start_reg, sizeof(tas2781_cali_start_reg), GFP_KERNEL); if (!tasdev[i].cali_data_backup) return -ENOMEM; + if (priv->dspbin_typ) { + cali_reg_update(p, t); + if (priv->dspbin_typ == TASDEV_ALPHA) { + p = &tasdev[i].alp_cali_bckp; + alpa_cali_update(p, t); + } + } } } else { cali_ctrls = (struct snd_kcontrol_new *)tas2563_cali_controls; -- 2.34.1

2 1

RE: [PATCH v1 1/2] ASoC: codecs: add support for ES8389
by Zhang Yi 03 Mar '25

03 Mar '25

> > + /* clock inversion */ > > + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { > > + case SND_SOC_DAIFMT_NB_NF: > > + break; > > + case SND_SOC_DAIFMT_IB_IF: > > + break; > > + case SND_SOC_DAIFMT_IB_NF: > > + break; > > + case SND_SOC_DAIFMT_NB_IF: > > + break; > > + default: > > + break; > > + } > > This switch does nothing. Remove it? Ok, I will remove it

1 0

[PATCH v1 0/2] ASoC: codecs: add ES8389 codec driver
by Zhang Yi 03 Mar '25

03 Mar '25

The driver is for codec ES8389 of everest-semi. Zhang Yi (2): ASoC: codecs: add support for ES8389 ASoC: dt-bindings: Add Everest ES8389 audio CODEC .../bindings/sound/everest,es8389.yaml | 98 ++ sound/soc/codecs/Kconfig | 6 +- sound/soc/codecs/Makefile | 2 + sound/soc/codecs/es8389.c | 985 ++++++++++++++++++ sound/soc/codecs/es8389.h | 139 +++ 5 files changed, 1229 insertions(+), 1 deletion(-) create mode 100755 Documentation/devicetree/bindings/sound/everest,es8389.yaml create mode 100644 sound/soc/codecs/es8389.c create mode 100644 sound/soc/codecs/es8389.h -- 2.17.1

2 3