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

[PATCH 1/3] ASoC: amd: amd_sdw: Fix unlikely uninitialized variable use in create_sdw_dailinks()
by Vijendar Mukunda 07 May '25

07 May '25

Initialize current_be_id to 0 in AMD legacy stack(NO DSP enabled) SoundWire generic machine driver code to handle the unlikely case when there are no devices connected to a DAI. In this case create_sdw_dailink() would return without touching the passed pointer to current_be_id. Found by gcc -fanalyzer Cc: stable(a)vger.kernel.org Fixes: 2981d9b0789c4 ("ASoC: amd: acp: add soundwire machine driver for legacy stack") Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- sound/soc/amd/acp/acp-sdw-legacy-mach.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/amd/acp/acp-sdw-legacy-mach.c b/sound/soc/amd/acp/acp-sdw-legacy-mach.c index 2020c5cfb3d5..582c68aee6e5 100644 --- a/sound/soc/amd/acp/acp-sdw-legacy-mach.c +++ b/sound/soc/amd/acp/acp-sdw-legacy-mach.c @@ -272,7 +272,7 @@ static int create_sdw_dailinks(struct snd_soc_card *card, /* generate DAI links by each sdw link */ while (soc_dais->initialised) { - int current_be_id; + int current_be_id = 0; ret = create_sdw_dailink(card, soc_dais, dai_links, &current_be_id, codec_conf, sdw_platform_component); -- 2.45.2

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Re: sound/pci/hda: add quirk for HP Spectre x360 15-eb0xxx
by Ezra Khuzadi 06 May '25

06 May '25

Hi Takashi, Jaroslav, all maintainers, Could you please review it or let me know if any changes are needed? This is my first kernel patch as a student, and I’d appreciate any feedback. Thanks, Ezra Khuzadi On Wed, Apr 30, 2025 at 1:43 AM Ezra Khuzadi <ekhuzadi(a)uci.edu> wrote: > sound/pci/hda/patch_realtek.c: add quirk for HP Spectre x360 15-eb0xxx > > Add subsystem ID 0x86e5 for HP Spectre x360 15-eb0xxx so that > ALC285_FIXUP_HP_SPECTRE_X360_EB1 (GPIO amp-enable, mic-mute LED and > pinconfigs) is applied. > > Tested on HP Spectre x360 15-eb0043dx (Vendor 0x10ec0285, Subsys > 0x103c86e5) > with legacy HDA driver and hda-verb toggles: > > $ cat /proc/asound/card0/codec#0 \ > | sed -n -e '1,5p;/Vendor Id:/p;/Subsystem Id:/p' > Codec: Realtek ALC285 > Vendor Id: 0x10ec0285 > Subsystem Id: 0x103c86e5 > > $ dmesg | grep -i realtek > [ 5.828728] snd_hda_codec_realtek ehdaudio0D0: ALC285: picked fixup > for PCI SSID 103c:86e5 > > Signed-off-by: Ezra Khuzadi <ekhuzadi(a)uci.edu> > Cc: stable(a)vger.kernel.org > > --- > 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 877137cb09ac..82ad105e7fa9 100644 > --- a/sound/pci/hda/patch_realtek.c > +++ b/sound/pci/hda/patch_realtek.c > @@ -10563,6 +10563,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = > { > SND_PCI_QUIRK(0x103c, 0x86c7, "HP Envy AiO 32", > ALC274_FIXUP_HP_ENVY_GPIO), > + SND_PCI_QUIRK(0x103c, 0x86e5, "HP Spectre x360 15-eb0xxx", > ALC285_FIXUP_HP_SPECTRE_X360_EB1), > SND_PCI_QUIRK(0x103c, 0x86e7, "HP Spectre x360 15-eb0xxx", > ALC285_FIXUP_HP_SPECTRE_X360_EB1), > SND_PCI_QUIRK(0x103c, 0x86e8, "HP Spectre x360 15-eb0xxx", > ALC285_FIXUP_HP_SPECTRE_X360_EB1), > SND_PCI_QUIRK(0x103c, 0x86f9, "HP Spectre x360 13-aw0xxx", > ALC285_FIXUP_HP_SPECTRE_X360_MUTE_LED), > > On Wed, Apr 30, 2025 at 1:33 AM kernel test robot <lkp(a)intel.com> wrote: > > > > Hi, > > > > Thanks for your patch. > > > > FYI: kernel test robot notices the stable kernel rule is not satisfied. > > > > The check is based on > https://urldefense.com/v3/__https://www.kernel.org/doc/html/latest/process/… > > > > Rule: add the tag "Cc: stable(a)vger.kernel.org" in the sign-off area to > have the patch automatically included in the stable tree. > > Subject: sound/pci/hda: add quirk for HP Spectre x360 15-eb0xxx > > Link: > https://urldefense.com/v3/__https://lore.kernel.org/stable/CAPXr0uxh0c_2b2-… > > > > -- > > 0-DAY CI Kernel Test Service > > > https://urldefense.com/v3/__https://github.com/intel/lkp-tests/wiki__;!!CzA… > > > > > > >

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ucm2: Added Beacn Mic and Studio Support
by GitHub pull_request - edited 06 May '25

06 May '25

alsa-project/alsa-ucm-conf pull request #558 was edited 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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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

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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

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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

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[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

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[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

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