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Added Beacn Mic and Studio Support
by GitHub pull_request - opened 06 May '25

06 May '25

alsa-project/alsa-ucm-conf pull request #558 was opened from FrostyCoolSlug: This PR adds support for the Beacn Mic and Beacn Studio Split PCM config Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/558 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/558.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[PATCH v4] ASoC: tas2781: Add a debufs node for acoustic tuning
by Shenghao Ding 06 May '25

06 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> --- v4: - Add name_prefix as part of the name of debugfs node in case of multiple devices. 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 | 154 +++++++++++++++++++++++++++++++++ 3 files changed, 186 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..059a46bde2d1 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 +1426,150 @@ 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; + char *acoustic_debugfs_node; +#endif int ret = 0; int i; @@ -1500,6 +1643,17 @@ 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 + if (tas_priv->name_prefix) + acoustic_debugfs_node = devm_kasprintf(tas_priv->dev, + GFP_KERNEL, "%s_acoustic_ctl", tas_priv->name_prefix); + else + acoustic_debugfs_node = devm_kstrdup(tas_priv->dev, + "acoustic_ctl", GFP_KERNEL); + debugfs_create_file(acoustic_debugfs_node, 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

3 3

Memory Leak in bag_del_all function of src/mixer/bag.c
by GitHub issues - reopened 06 May '25

06 May '25

alsa-project/alsa-lib issue #453 was reopened from zkh8227: #Issue Summary The bag_del_all function in src/mixer/bag.c has a memory leak. It removes elements from a linked list but doesn't free the memory allocated for them in bag_add function, which uses malloc. Repeated calls to bag_add and bag_del_all will waste memory. #Fix Suggestion Change bag_del_all to free memory when removing elements. Issue URL : https://github.com/alsa-project/alsa-lib/issues/453 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Memory Leak in bag_del_all function of src/mixer/bag.c
by GitHub issues - edited 06 May '25

06 May '25

alsa-project/alsa-lib issue #453 was edited from zkh8227: #Issue Summary The bag_del_all function in src/mixer/bag.c has a memory leak. It removes elements from a linked list but doesn't free the memory allocated for them in bag_add function, which uses malloc. Repeated calls to bag_add and bag_del_all will waste memory. #Fix Suggestion Change bag_del_all to free memory when removing elements. Issue URL : https://github.com/alsa-project/alsa-lib/issues/453 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH 0/9] ASoC: SOF: amd: add soundwire IO support for ACP7.0 & ACP7.1 platforms
by Vijendar Mukunda 06 May '25

06 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

2 10

[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

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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 - 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

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[PATCH 1/3] ASoC: amd: acp: Fix NULL pointer deref on acp resume path
by Venkata Prasad Potturu 02 May '25

02 May '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

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