June 2021 - Alsa-devel - mailman.alsa-project.org

[PATCH v1 1/4] ASoC: dt-bindings: nxp,tfa989x: Add tfa9897 support
by Vincent Knecht 02 Jun '21

02 Jun '21

Document TFA9897 bindings. Signed-off-by: Vincent Knecht <vincent.knecht(a)mailoo.org> --- Documentation/devicetree/bindings/sound/nxp,tfa989x.yaml | 1 + 1 file changed, 1 insertion(+) diff --git a/Documentation/devicetree/bindings/sound/nxp,tfa989x.yaml b/Documentation/devicetree/bindings/sound/nxp,tfa989x.yaml index 45db5776550c..46ddc1f3fc0c 100644 --- a/Documentation/devicetree/bindings/sound/nxp,tfa989x.yaml +++ b/Documentation/devicetree/bindings/sound/nxp,tfa989x.yaml @@ -13,6 +13,7 @@ properties: compatible: enum: - nxp,tfa9895 + - nxp,tfa9897 reg: maxItems: 1 -- 2.31.1

3 7

[PATCH -next] ASoC: Intel: Skylake: use DEVICE_ATTR_RO macro
by YueHaibing 02 Jun '21

02 Jun '21

Use DEVICE_ATTR_RO() helper instead of plain DEVICE_ATTR(), which makes the code a bit shorter and easier to read. Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/intel/skylake/skl-nhlt.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/skylake/skl-nhlt.c b/sound/soc/intel/skylake/skl-nhlt.c index 87c891c46291..64226072f0ee 100644 --- a/sound/soc/intel/skylake/skl-nhlt.c +++ b/sound/soc/intel/skylake/skl-nhlt.c @@ -149,8 +149,8 @@ int skl_nhlt_update_topology_bin(struct skl_dev *skl) return 0; } -static ssize_t skl_nhlt_platform_id_show(struct device *dev, - struct device_attribute *attr, char *buf) +static ssize_t platform_id_show(struct device *dev, + struct device_attribute *attr, char *buf) { struct pci_dev *pci = to_pci_dev(dev); struct hdac_bus *bus = pci_get_drvdata(pci); @@ -166,7 +166,7 @@ static ssize_t skl_nhlt_platform_id_show(struct device *dev, return sprintf(buf, "%s\n", platform_id); } -static DEVICE_ATTR(platform_id, 0444, skl_nhlt_platform_id_show, NULL); +static DEVICE_ATTR_RO(platform_id); int skl_nhlt_create_sysfs(struct skl_dev *skl) { -- 2.17.1

3 2

[PATCH] ASoC: sigmadsp: Remove the repeated declaration
by Shaokun Zhang 02 Jun '21

02 Jun '21

Function 'sigmadsp_reset' is declared twice, so remove the repeated declaration. Cc: Mark Brown <broonie(a)kernel.org> Signed-off-by: Shaokun Zhang <zhangshaokun(a)hisilicon.com> --- sound/soc/codecs/sigmadsp.h | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/sigmadsp.h b/sound/soc/codecs/sigmadsp.h index d63b8c366efb..2783eff633a1 100644 --- a/sound/soc/codecs/sigmadsp.h +++ b/sound/soc/codecs/sigmadsp.h @@ -44,7 +44,6 @@ struct sigmadsp { struct sigmadsp *devm_sigmadsp_init(struct device *dev, const struct sigmadsp_ops *ops, const char *firmware_name); -void sigmadsp_reset(struct sigmadsp *sigmadsp); int sigmadsp_restrict_params(struct sigmadsp *sigmadsp, struct snd_pcm_substream *substream); -- 2.7.4

2 1

[v4] ASoC: qcom: lpass-cpu: Fix pop noise during audio capture begin
by Judy Hsiao 02 Jun '21

02 Jun '21

From: Srinivasa Rao Mandadapu <srivasam(a)codeaurora.org> This patch fixes PoP noise of around 15ms observed during audio capture begin. Enables BCLK and LRCLK in snd_soc_dai_ops prepare call for introducing some delay before capture start and clock enable. Co-developed-by: Judy Hsiao <judyhsiao(a)chromium.org> Signed-off-by: Judy Hsiao <judyhsiao(a)chromium.org> Signed-off-by: Srinivasa Rao Mandadapu <srivasam(a)codeaurora.org> (am from https://patchwork.kernel.org/patch/12276369/) (also found at https://lore.kernel.org/r/20210524142114.18676-1-srivasam@codeaurora.org) --- Chages Since V3: -- Check BCLK is off before enabling it in lpass_cpu_daiops_prepare as lpass_cpu_daiops_prepare can be called multiple times. -- Check BCLK is on before disabling it in lpass_cpu_daiops_shutdown to fix the WARN. It is because BCLK may not be enabled if lpass_cpu_daiops_prepare is not called before lpass_cpu_daiops_shutdown. -- Adds more comments. Changes Since V2: -- Updated comments as per linux style -- Removed unrelated changes. Changes Since V1: -- Enableed BCLK and LRCLK in dai ops prepare API instead of startup API -- Added comments sound/soc/qcom/lpass-cpu.c | 83 +++++++++++++++++++++++++++++++++++++- 1 file changed, 82 insertions(+), 1 deletion(-) diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index af8cb64924a0..b572d7874554 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -6,6 +6,7 @@ */ #include <linux/clk.h> +#include <linux/clk-provider.h> #include <linux/kernel.h> #include <linux/module.h> #include <linux/of.h> @@ -93,8 +94,28 @@ static void lpass_cpu_daiops_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct lpass_data *drvdata = snd_soc_dai_get_drvdata(dai); + struct lpaif_i2sctl *i2sctl = drvdata->i2sctl; + unsigned int id = dai->driver->id; clk_disable_unprepare(drvdata->mi2s_osr_clk[dai->driver->id]); + /* + * To ensure LRCLK disabled even in device node validation + * Will not impact if disabled in lpass_cpu_daiops_trigger() + * suspend. + */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + regmap_fields_write(i2sctl->spken, id, LPAIF_I2SCTL_SPKEN_DISABLE); + else + regmap_fields_write(i2sctl->micen, id, LPAIF_I2SCTL_MICEN_DISABLE); + + /* + * BCLK may not be enabled if lpass_cpu_daiops_prepare is called before + * lpass_cpu_daiops_shutdown. It's paired with the clk_enable in + * lpass_cpu_daiops_prepare. + */ + if (__clk_is_enabled(drvdata->mi2s_bit_clk[dai->driver->id])) + clk_disable(drvdata->mi2s_bit_clk[dai->driver->id]); + clk_unprepare(drvdata->mi2s_bit_clk[dai->driver->id]); } @@ -275,6 +296,18 @@ static int lpass_cpu_daiops_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + /* + * To ensure lpass BCLK/LRCLK is enabled during + * device resume as lpass_cpu_daiops_prepare() is not called + * after the device resumes. We don't check BCLK status before + * enable/disable BCLK in trigger event because: + * 1. These trigger events are paired, so the BCLK + * enable_count is balanced. + * 2. the BCLK can be shared (ex: headset and headset mic), + * we need to increase the enable_count so that we don't + * turn off the shared BCLK while other devices are using + * it. + */ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { ret = regmap_fields_write(i2sctl->spken, id, LPAIF_I2SCTL_SPKEN_ENABLE); @@ -288,7 +321,8 @@ static int lpass_cpu_daiops_trigger(struct snd_pcm_substream *substream, ret = clk_enable(drvdata->mi2s_bit_clk[id]); if (ret) { - dev_err(dai->dev, "error in enabling mi2s bit clk: %d\n", ret); + dev_err(dai->dev, + "error in enabling mi2s bit clk: %d\n", ret); clk_disable(drvdata->mi2s_osr_clk[id]); return ret; } @@ -296,6 +330,10 @@ static int lpass_cpu_daiops_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + /* + * To ensure lpass BCLK/LRCLK is disabled during + * device suspend. + */ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { ret = regmap_fields_write(i2sctl->spken, id, LPAIF_I2SCTL_SPKEN_DISABLE); @@ -315,12 +353,55 @@ static int lpass_cpu_daiops_trigger(struct snd_pcm_substream *substream, return ret; } +static int lpass_cpu_daiops_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct lpass_data *drvdata = snd_soc_dai_get_drvdata(dai); + struct lpaif_i2sctl *i2sctl = drvdata->i2sctl; + unsigned int id = dai->driver->id; + int ret; + /* + * To ensure lpass BCLK/LRCLK is enabled bit before playback/capture + * data flow starts. This allows other codec to have some delay before + * the data flow. + * (ex: to drop start up pop noise before capture starts). + */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + ret = regmap_fields_write(i2sctl->spken, id, + LPAIF_I2SCTL_SPKEN_ENABLE); + else + ret = regmap_fields_write(i2sctl->micen, id, + LPAIF_I2SCTL_MICEN_ENABLE); + + if (ret) { + dev_err(dai->dev, "error writing to i2sctl reg: %d\n", ret); + return ret; + } + + /* + * Check BCLK is off before enabling it as lpass_cpu_daiops_prepare can + * be called multiple times. It's paired with the clk_disable in + * lpass_cpu_daiops_shutdown. + */ + if (!__clk_is_enabled(drvdata->mi2s_bit_clk[dai->driver->id])) { + ret = clk_enable(drvdata->mi2s_bit_clk[id]); + if (ret) { + dev_err(dai->dev, + "error in enabling mi2s bit clk: %d\n", ret); + clk_disable(drvdata->mi2s_osr_clk[id]); + return ret; + } + } + return 0; +} + const struct snd_soc_dai_ops asoc_qcom_lpass_cpu_dai_ops = { .set_sysclk = lpass_cpu_daiops_set_sysclk, .startup = lpass_cpu_daiops_startup, .shutdown = lpass_cpu_daiops_shutdown, .hw_params = lpass_cpu_daiops_hw_params, .trigger = lpass_cpu_daiops_trigger, + .prepare = lpass_cpu_daiops_prepare, }; EXPORT_SYMBOL_GPL(asoc_qcom_lpass_cpu_dai_ops); -- 2.31.0

2 1

[PATCH] ALSA: hda: update the power_state during the direct-complete
by Hui Wang 02 Jun '21

02 Jun '21

The patch_realtek.c needs to check if the power_state.event equals PM_EVENT_SUSPEND, after using the direct-complete, the suspend() and resume() will be skipped if the codec is already rt_suspended, in this case, the patch_realtek.c will always get PM_EVENT_ON even the system is really resumed from S3. We could set power_state to PMSG_SUSPEND in the prepare(), if other PM functions are called before complete(), those functions will override power_state; if no other PM functions are called before complete(), we could know the suspend() and resume() are skipped since only S3 pm functions could be skipped by direct-complete, in this case set power_state to PMSG_RESUME in the complete(). This could guarantee the first time of calling hda_codec_runtime_resume() after complete() has the correct power_state. Fixes: 215a22ed31a1 ("ALSA: hda: Refactor codec PM to use direct-complete optimization") Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/hda_codec.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index a31009afc025..5462f771c2f9 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2917,6 +2917,7 @@ static int hda_codec_runtime_resume(struct device *dev) #ifdef CONFIG_PM_SLEEP static int hda_codec_pm_prepare(struct device *dev) { + dev->power.power_state = PMSG_SUSPEND; return pm_runtime_suspended(dev); } @@ -2924,6 +2925,10 @@ static void hda_codec_pm_complete(struct device *dev) { struct hda_codec *codec = dev_to_hda_codec(dev); + /* If no other pm-functions are called between prepare() and complete() */ + if (dev->power.power_state.event == PM_EVENT_SUSPEND) + dev->power.power_state = PMSG_RESUME; + if (pm_runtime_suspended(dev) && (codec->jackpoll_interval || hda_codec_need_resume(codec) || codec->forced_resume)) pm_request_resume(dev); -- 2.25.1

2 1

[syzbot] UBSAN: shift-out-of-bounds in snd_timer_user_ccallback
by Dongliang Mu 02 Jun '21

02 Jun '21

> Hello, > > syzbot found the following issue on: > > HEAD commit: 5ff2756a Merge tag 'nfs-for-5.13-2' of git://git.linux-nfs.. > git tree: upstream > console output: https://syzkaller.appspot.com/x/log.txt?x=17872d5bd00000 > kernel config: https://syzkaller.appspot.com/x/.config?x=770708ea7cfd4916 > dashboard link: https://syzkaller.appspot.com/bug?extid=d102fa5b35335a7e544e > > Unfortunately, I don't have any reproducer for this issue yet. > > IMPORTANT: if you fix the issue, please add the following tag to the commit: > Reported-by: syzbot+d102fa...(a)syzkaller.appspotmail.com > > ================================================================================ > UBSAN: shift-out-of-bounds in sound/core/timer.c:1376:23 > shift exponent 105 is too large for 32-bit type 'int' > CPU: 1 PID: 10368 Comm: syz-executor.1 Not tainted 5.13.0-rc3-syzkaller #0 > Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 > Call Trace: > __dump_stack lib/dump_stack.c:79 [inline] > dump_stack+0x141/0x1d7 lib/dump_stack.c:120 > ubsan_epilogue+0xb/0x5a lib/ubsan.c:148 > __ubsan_handle_shift_out_of_bounds.cold+0xb1/0x181 lib/ubsan.c:327 > snd_timer_user_ccallback.cold+0x19/0x1e sound/core/timer.c:1376 > > snd_timer_notify1+0x243/0x3b0 sound/core/timer.c:525 The root cause of this bug is in the snd_timer_notify1 [1]. At the end of this function, it calls "ts->ccallback(ts, event + 100, &tstamp, resolution)". Here the variable event is 5. It adds 100 and is passed as 2nd argument of snd_timer_user_ccallback. >From the variable naming, the 2nd argument should an event, and in the range of event enumeration. In fact, 105 (event + 100) is out of this range. I don't quite understand the meaning of adding 100. Any thought here? [1] https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/tree/sou… -- My best regards to you. No System Is Safe! Dongliang Mu

4 4

[PATCH RESEND] ALSA: timer: Fix master timer notification
by Takashi Iwai 02 Jun '21

02 Jun '21

snd_timer_notify1() calls the notification to each slave for a master event, but it passes a wrong event number. It should be +10 offset, corresponding to SNDRV_TIMER_EVENT_MXXX, but it's incorrectly with +100 offset. Casually this was spotted by UBSAN check via syzkaller. Reported-by: syzbot+d102fa5b35335a7e544e(a)syzkaller.appspotmail.com Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/000000000000e5560e05c3bd1d63@google.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/timer.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/core/timer.c b/sound/core/timer.c index 6898b1ac0d7f..92b7008fcdb8 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -520,9 +520,10 @@ static void snd_timer_notify1(struct snd_timer_instance *ti, int event) return; if (timer->hw.flags & SNDRV_TIMER_HW_SLAVE) return; + event += 10; /* convert to SNDRV_TIMER_EVENT_MXXX */ list_for_each_entry(ts, &ti->slave_active_head, active_list) if (ts->ccallback) - ts->ccallback(ts, event + 100, &tstamp, resolution); + ts->ccallback(ts, event, &tstamp, resolution); } /* start/continue a master timer */ -- 2.26.2

1 0

[PATCH][next] ASoC: rk817: remove redundant assignment to pointer node
by Colin King 02 Jun '21

02 Jun '21

From: Colin Ian King <colin.king(a)canonical.com> The pointer node is being initialized with a value that is never read and it is being updated later with a new value. The initialization is redundant and can be removed. Addresses-Coverity: ("Unused value") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/codecs/rk817_codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rk817_codec.c b/sound/soc/codecs/rk817_codec.c index 17e672b85ee5..0d7cc26ded57 100644 --- a/sound/soc/codecs/rk817_codec.c +++ b/sound/soc/codecs/rk817_codec.c @@ -457,7 +457,7 @@ static const struct snd_soc_component_driver soc_codec_dev_rk817 = { static void rk817_codec_parse_dt_property(struct device *dev, struct rk817_codec_priv *rk817) { - struct device_node *node = dev->parent->of_node; + struct device_node *node; node = of_get_child_by_name(dev->parent->of_node, "codec"); if (!node) { -- 2.31.1

3 2

no sound via HDMI
by reinerschrutz＠web.de 02 Jun '21

02 Jun '21

Hi, missing sound via monitor and HDMI cable following ubuntuusers "HDA" article I found: 0[generic ] - HD-Audio Generic HD-Audio Generic at 0xfcc 88000 irg 62 1[generic_1 ] - HD-Audio Generic HD-Audio Generic at 0xfcc 88000 irq 63 looking for a pci device I found: 05:00.1 Audio device Advanced Micro Devices, Inc. [AMD/ATI] Device 1637 05:00.6 Audio device Advanced Mirco Devices, Inc. [AMD] Family 17h (Models 10h 1fh) HD Audio Controller Codec found: ATI R6xx HDMI My system: Asus Prime A520M-K, AMD Ryzen 7 Pro 4750G, 8x 3,6 GHz, 16 GD DDR4 RAM, 500 GB Samsung 870 EVO Advanced Linux Sound Architecture - Driver Cofiguration Guide said to give notice. Thanks. Stefan Ocken

1 0

[PATCH] ASoC: rk817: fix a warning in rk817_probe()
by Chris Morgan 02 Jun '21

02 Jun '21

From: Chris Morgan <macromorgan(a)hotmail.com> The return value of snd_soc_component_write() is stored but not evaluated and this results in a warning when W=1 is set. Stop storing the return value to be consistent with all other calls of snd_soc_component_write() and to remove the warning. Fixes: 0d6a04da9b25 ("ASoC: Add Rockchip rk817 audio CODEC support") Signed-off-by: Chris Morgan <macromorgan(a)hotmail.com> --- sound/soc/codecs/rk817_codec.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/codecs/rk817_codec.c b/sound/soc/codecs/rk817_codec.c index 17e672b85ee5..fd3a5ba034a9 100644 --- a/sound/soc/codecs/rk817_codec.c +++ b/sound/soc/codecs/rk817_codec.c @@ -415,12 +415,11 @@ static int rk817_probe(struct snd_soc_component *component) { struct rk817_codec_priv *rk817 = snd_soc_component_get_drvdata(component); struct rk808 *rk808 = dev_get_drvdata(component->dev->parent); - int ret; snd_soc_component_init_regmap(component, rk808->regmap); rk817->component = component; - ret = snd_soc_component_write(component, RK817_CODEC_DTOP_LPT_SRST, 0x40); + snd_soc_component_write(component, RK817_CODEC_DTOP_LPT_SRST, 0x40); rk817_init(component); -- 2.25.1

2 1

[PATCH 0/3] ALSA: firewire-motu: media clock recovery for sph-aware devices
by Takashi Sakamoto 02 Jun '21

02 Jun '21

Hi, In a commit f9e5ecdfc2c2 ("ALSA: firewire-lib: add replay target to cache sequence of packet"), I categorize devices supported by drivers in ALSA firewire stack in terms of the way to deliver effective sampling transfer frequency. This patchset is for the devices in group 3. The devices are known to have problems when ALSA firewire-motu driver handles. Many of them generate sound with noise. In the worst case, it generates no sound. The devices interpret presentation time to decide playback timing. Unlike the syt-aware devices, the devices interpret the presentation time in source packet header (SPH) per data block, instead of the presentation time in syt field of CIP header. Current implementation of the driver processes the sequence of outgoing packet by computation according to nominal sampling transfer frequency. However, the ideal sequence is not adequate to the devices, actually. With this patchset, the drivers are going to replay the sequence of incoming packets for media clock recovery, instead of nominal sampling transfer frequency. For the detail of sequence replay, please refer to a commit 39c2649c71d8 ("ALSA: firewire-lib: replay sequence of incoming packets for outgoing packets"). The sequence replay is done by two levels; the sequence of the number of data blocks per packet, and the sequence of SPH per data blocks in the packet. Takashi Sakamoto (3): ALSA: firewire-motu: use macro for magic numbers relevant to IEC 61883-1 ALSA: firewire-motu: cache event ticks in source packet header per data block ALSA: firewire-motu: sequence replay for source packet header sound/firewire/motu/amdtp-motu.c | 136 +++++++++++++++--------------- sound/firewire/motu/motu-stream.c | 27 +++++- sound/firewire/motu/motu.h | 14 ++- 3 files changed, 104 insertions(+), 73 deletions(-) -- 2.27.0

2 4

[PATCH v2] ALSA: control led: fix memory leak in snd_ctl_led_register
by Dongliang Mu 02 Jun '21

02 Jun '21

The snd_ctl_led_sysfs_add and snd_ctl_led_sysfs_remove should contain the refcount operations in pair. However, snd_ctl_led_sysfs_remove fails to decrease the refcount to zero, which causes device_release never to be invoked. This leads to memory leak to some resources, like struct device_private. In addition, we also free some other similar memory leaks in snd_ctl_led_init/snd_ctl_led_exit. Fix this by replacing device_del to device_unregister in snd_ctl_led_sysfs_remove/snd_ctl_led_init/snd_ctl_led_exit. Note that, when CONFIG_DEBUG_KOBJECT_RELEASE is enabled, put_device will call kobject_release and delay the release of kobject, which will cause use-after-free when the memory backing the kobject is freed at once. Reported-by: syzbot+08a7d8b51ea048a74ffb(a)syzkaller.appspotmail.com Fixes: a135dfb5de1 ("ALSA: led control - add sysfs kcontrol LED marking layer") Signed-off-by: Dongliang Mu <mudongliangabcd(a)gmail.com> --- v1->v2: fix other similar memory leaks; move kfree(led_card) to a release function. sound/core/control_led.c | 33 ++++++++++++++++++++++++++------- 1 file changed, 26 insertions(+), 7 deletions(-) diff --git a/sound/core/control_led.c b/sound/core/control_led.c index 25f57c14f294..a90e31dbde61 100644 --- a/sound/core/control_led.c +++ b/sound/core/control_led.c @@ -17,6 +17,9 @@ MODULE_LICENSE("GPL"); #define MAX_LED (((SNDRV_CTL_ELEM_ACCESS_MIC_LED - SNDRV_CTL_ELEM_ACCESS_SPK_LED) \ >> SNDRV_CTL_ELEM_ACCESS_LED_SHIFT) + 1) +#define to_led_card_dev(_dev) \ + container_of(_dev, struct snd_ctl_led_card, dev) + enum snd_ctl_led_mode { MODE_FOLLOW_MUTE = 0, MODE_FOLLOW_ROUTE, @@ -371,6 +374,21 @@ static void snd_ctl_led_disconnect(struct snd_card *card) snd_ctl_led_refresh(); } +static void snd_ctl_led_card_release(struct device *dev) +{ + struct snd_ctl_led_card *led_card = to_led_card_dev(dev); + + kfree(led_card); +} + +static void snd_ctl_led_release(struct device *dev) +{ +} + +static void snd_ctl_led_dev_release(struct device *dev) +{ +} + /* * sysfs */ @@ -663,6 +681,7 @@ static void snd_ctl_led_sysfs_add(struct snd_card *card) led_card->number = card->number; led_card->led = led; device_initialize(&led_card->dev); + led_card->dev.release = snd_ctl_led_card_release; if (dev_set_name(&led_card->dev, "card%d", card->number) < 0) goto cerr; led_card->dev.parent = &led->dev; @@ -681,7 +700,6 @@ static void snd_ctl_led_sysfs_add(struct snd_card *card) put_device(&led_card->dev); cerr2: printk(KERN_ERR "snd_ctl_led: unable to add card%d", card->number); - kfree(led_card); } } @@ -700,8 +718,7 @@ static void snd_ctl_led_sysfs_remove(struct snd_card *card) snprintf(link_name, sizeof(link_name), "led-%s", led->name); sysfs_remove_link(&card->ctl_dev.kobj, link_name); sysfs_remove_link(&led_card->dev.kobj, "card"); - device_del(&led_card->dev); - kfree(led_card); + device_unregister(&led_card->dev); led->cards[card->number] = NULL; } } @@ -723,6 +740,7 @@ static int __init snd_ctl_led_init(void) device_initialize(&snd_ctl_led_dev); snd_ctl_led_dev.class = sound_class; + snd_ctl_led_dev.release = snd_ctl_led_dev_release; dev_set_name(&snd_ctl_led_dev, "ctl-led"); if (device_add(&snd_ctl_led_dev)) { put_device(&snd_ctl_led_dev); @@ -733,15 +751,16 @@ static int __init snd_ctl_led_init(void) INIT_LIST_HEAD(&led->controls); device_initialize(&led->dev); led->dev.parent = &snd_ctl_led_dev; + led->dev.release = snd_ctl_led_release; led->dev.groups = snd_ctl_led_dev_attr_groups; dev_set_name(&led->dev, led->name); if (device_add(&led->dev)) { put_device(&led->dev); for (; group > 0; group--) { led = &snd_ctl_leds[group - 1]; - device_del(&led->dev); + device_unregister(&led->dev); } - device_del(&snd_ctl_led_dev); + device_unregister(&snd_ctl_led_dev); return -ENOMEM; } } @@ -767,9 +786,9 @@ static void __exit snd_ctl_led_exit(void) } for (group = 0; group < MAX_LED; group++) { led = &snd_ctl_leds[group]; - device_del(&led->dev); + device_unregister(&led->dev); } - device_del(&snd_ctl_led_dev); + device_unregister(&snd_ctl_led_dev); snd_ctl_led_clean(NULL); } -- 2.25.1

4 3

bytcr-rt5640 initiializes but has wrong settings
by GitHub issues - opened 02 Jun '21

02 Jun '21

alsa-project/alsa-ucm-conf issue #96 was opened from AnterCreeper: My device is Elitepad 1000 G2. The headphone and the mic inside cannot work. However, The speaker works properly. The mic IN1 is the headphone mic insted of the mic inside. I think that it may be because the audio was ported to the wrong place. Sorry for my bad English :D Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/96 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[syzbot] UBSAN: shift-out-of-bounds in snd_timer_user_ccallback
by syzbot 02 Jun '21

02 Jun '21

Hello, syzbot found the following issue on: HEAD commit: 5ff2756a Merge tag 'nfs-for-5.13-2' of git://git.linux-nfs.. git tree: upstream console output: https://syzkaller.appspot.com/x/log.txt?x=17872d5bd00000 kernel config: https://syzkaller.appspot.com/x/.config?x=770708ea7cfd4916 dashboard link: https://syzkaller.appspot.com/bug?extid=d102fa5b35335a7e544e Unfortunately, I don't have any reproducer for this issue yet. IMPORTANT: if you fix the issue, please add the following tag to the commit: Reported-by: syzbot+d102fa5b35335a7e544e(a)syzkaller.appspotmail.com ================================================================================ UBSAN: shift-out-of-bounds in sound/core/timer.c:1376:23 shift exponent 105 is too large for 32-bit type 'int' CPU: 1 PID: 10368 Comm: syz-executor.1 Not tainted 5.13.0-rc3-syzkaller #0 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 Call Trace: __dump_stack lib/dump_stack.c:79 [inline] dump_stack+0x141/0x1d7 lib/dump_stack.c:120 ubsan_epilogue+0xb/0x5a lib/ubsan.c:148 __ubsan_handle_shift_out_of_bounds.cold+0xb1/0x181 lib/ubsan.c:327 snd_timer_user_ccallback.cold+0x19/0x1e sound/core/timer.c:1376 snd_timer_notify1+0x243/0x3b0 sound/core/timer.c:525 snd_timer_stop1+0x641/0x890 sound/core/timer.c:656 snd_timer_pause+0x6e/0x90 sound/core/timer.c:738 seq_timer_stop sound/core/seq/seq_timer.c:331 [inline] snd_seq_timer_stop+0x93/0xd0 sound/core/seq/seq_timer.c:341 queue_delete+0x42/0xa0 sound/core/seq/seq_queue.c:133 snd_seq_queue_client_leave+0x37/0x1a0 sound/core/seq/seq_queue.c:552 seq_free_client1.part.0+0x10a/0x260 sound/core/seq/seq_clientmgr.c:280 seq_free_client1 sound/core/seq/seq_clientmgr.c:273 [inline] seq_free_client+0x7b/0xf0 sound/core/seq/seq_clientmgr.c:301 snd_seq_release+0x4d/0xe0 sound/core/seq/seq_clientmgr.c:382 __fput+0x288/0x920 fs/file_table.c:280 task_work_run+0xdd/0x1a0 kernel/task_work.c:164 tracehook_notify_resume include/linux/tracehook.h:189 [inline] exit_to_user_mode_loop kernel/entry/common.c:174 [inline] exit_to_user_mode_prepare+0x272/0x280 kernel/entry/common.c:208 __syscall_exit_to_user_mode_work kernel/entry/common.c:290 [inline] syscall_exit_to_user_mode+0x19/0x60 kernel/entry/common.c:301 do_syscall_64+0x47/0xb0 arch/x86/entry/common.c:57 entry_SYSCALL_64_after_hwframe+0x44/0xae RIP: 0033:0x4193eb Code: 0f 05 48 3d 00 f0 ff ff 77 45 c3 0f 1f 40 00 48 83 ec 18 89 7c 24 0c e8 63 fc ff ff 8b 7c 24 0c 41 89 c0 b8 03 00 00 00 0f 05 <48> 3d 00 f0 ff ff 77 35 44 89 c7 89 44 24 0c e8 a1 fc ff ff 8b 44 RSP: 002b:0000000000a9fb80 EFLAGS: 00000293 ORIG_RAX: 0000000000000003 RAX: 0000000000000000 RBX: 0000000000000005 RCX: 00000000004193eb RDX: 0000000000571b58 RSI: ffffffff89166efe RDI: 0000000000000004 RBP: 0000000000000001 R08: 0000000000000000 R09: 0000001b2ea280a8 R10: 00000000000001d1 R11: 0000000000000293 R12: 000000000014d248 R13: 00000000000003e8 R14: 000000000056bf80 R15: 000000000014d1dd ================================================================================ --- This report is generated by a bot. It may contain errors. See https://goo.gl/tpsmEJ for more information about syzbot. syzbot engineers can be reached at syzkaller(a)googlegroups.com. syzbot will keep track of this issue. See: https://goo.gl/tpsmEJ#status for how to communicate with syzbot.

1 0

[PATCH 0/4] ASoC: rsnd: adjust disabled module for R-Car D3
by Kuninori Morimoto 01 Jun '21

01 Jun '21

Hi Mark, Geert Renesas Sound driver is assuming that all SSI/SRC are connected. But some SSI/SRC are not connected on Some Renesas SoC. ex) H2 E2 SRC0 <= SRC1 SRC1 SRC2 SRC2 ... ... We accepted it by using "status = disabled" on DT before. ex) rcar_sound,src { src-0 { => status = "disabled"; }; src1: src-1 { ... }; ... But R-Car D3 have many disabled modules (It has SSI3/SSI4, SRC5/SRC6), and Renesas SoC maintainer don't want above style on DT. ex) rcar_sound,src { => src0: src-0 { status = "disabled"; }; => src1: src-1 { status = "disabled"; }; => src2: src-2 { status = "disabled"; }; => src3: src-3 { status = "disabled"; }; => src4: src-4 { status = "disabled"; }; src5: src-5 { ... }; ... This patch-set adjust to this situation, and enables to intuitive DT settings. rcar_sound,src { src5: src-5 { ... }; src6: src-6 { ... }; }; Kuninori Morimoto (4): ASoC: rsnd: tidyup rsnd_parse_connect_common() ASoC: rsnd: tidyup rsnd_dma_request_channel() ASoC: rsnd: tidyup rsnd_parse_connect_xxx() ASoC: rsnd: adjust disabled module sound/soc/sh/rcar/core.c | 58 ++++++++++++++++++++++++++++++++++++++-- sound/soc/sh/rcar/dma.c | 8 +++--- sound/soc/sh/rcar/dvc.c | 2 +- sound/soc/sh/rcar/rsnd.h | 16 ++++++----- sound/soc/sh/rcar/src.c | 6 +++-- sound/soc/sh/rcar/ssi.c | 12 ++++++--- sound/soc/sh/rcar/ssiu.c | 10 ++++--- 7 files changed, 91 insertions(+), 21 deletions(-) -- 2.25.1

2 5

[PATCH 1/2] ASoC: SOF: Intel: hda: clean up hda_dsp_dump()
by Kai Vehmanen 01 Jun '21

01 Jun '21

From: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Clean up the hda_dsp_dump() function to avoid duplicating the ROM status and error. Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Bard Liao <bard.liao(a)intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/soc/sof/intel/hda.c | 17 +++++------------ 1 file changed, 5 insertions(+), 12 deletions(-) diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index 126232a76a10..e1e368ff2b12 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -394,28 +394,21 @@ void hda_dsp_dump(struct snd_sof_dev *sdev, u32 flags) struct sof_ipc_dsp_oops_xtensa xoops; struct sof_ipc_panic_info panic_info; u32 stack[HDA_DSP_STACK_DUMP_SIZE]; - u32 status, panic; - /* try APL specific status message types first */ + /* print ROM/FW status */ hda_dsp_get_status(sdev); - /* now try generic SOF status messages */ - status = snd_sof_dsp_read(sdev, HDA_DSP_BAR, - HDA_DSP_SRAM_REG_FW_STATUS); - panic = snd_sof_dsp_read(sdev, HDA_DSP_BAR, HDA_DSP_SRAM_REG_FW_TRACEP); - + /* print panic info if FW boot is complete. Otherwise, print the extended ROM status */ if (sdev->fw_state == SOF_FW_BOOT_COMPLETE) { + u32 status = snd_sof_dsp_read(sdev, HDA_DSP_BAR, HDA_DSP_SRAM_REG_FW_STATUS); + u32 panic = snd_sof_dsp_read(sdev, HDA_DSP_BAR, HDA_DSP_SRAM_REG_FW_TRACEP); + hda_dsp_get_registers(sdev, &xoops, &panic_info, stack, HDA_DSP_STACK_DUMP_SIZE); snd_sof_get_status(sdev, status, panic, &xoops, &panic_info, stack, HDA_DSP_STACK_DUMP_SIZE); } else { - sof_dev_dbg_or_err(sdev->dev, flags & SOF_DBG_DUMP_FORCE_ERR_LEVEL, - "status = 0x%8.8x panic = 0x%8.8x\n", - status, panic); - hda_dsp_dump_ext_rom_status(sdev, flags); - hda_dsp_get_status(sdev); } } base-commit: 058efb40641845432c52777443b3372dbc97c032 -- 2.31.1

2 2

[PATCH] ASoC: SOF: Intel: pci-tgl: add ADL-M support
by Kai Vehmanen 01 Jun '21

01 Jun '21

Add PCI DID for Intel AlderLake-M. Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/sof/intel/pci-tgl.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/sof/intel/pci-tgl.c b/sound/soc/sof/intel/pci-tgl.c index 88c3bf404dd7..a00262184efa 100644 --- a/sound/soc/sof/intel/pci-tgl.c +++ b/sound/soc/sof/intel/pci-tgl.c @@ -116,6 +116,8 @@ static const struct pci_device_id sof_pci_ids[] = { .driver_data = (unsigned long)&adls_desc}, { PCI_DEVICE(0x8086, 0x51c8), /* ADL-P */ .driver_data = (unsigned long)&adl_desc}, + { PCI_DEVICE(0x8086, 0x51cc), /* ADL-M */ + .driver_data = (unsigned long)&adl_desc}, { 0, } }; MODULE_DEVICE_TABLE(pci, sof_pci_ids); base-commit: 058efb40641845432c52777443b3372dbc97c032 -- 2.31.1

3 2

[PATCH] ASoC: fsl-asoc-card: Set .owner attribute when registering card.
by Nicolas Cavallari 01 Jun '21

01 Jun '21

Otherwise, when compiled as module, a WARN_ON is triggered: WARNING: CPU: 0 PID: 5 at sound/core/init.c:208 snd_card_new+0x310/0x39c [snd] [...] CPU: 0 PID: 5 Comm: kworker/0:0 Not tainted 5.10.39 #1 Hardware name: Freescale i.MX6 Quad/DualLite (Device Tree) Workqueue: events deferred_probe_work_func [<c0111988>] (unwind_backtrace) from [<c010c8ac>] (show_stack+0x10/0x14) [<c010c8ac>] (show_stack) from [<c092784c>] (dump_stack+0xdc/0x104) [<c092784c>] (dump_stack) from [<c0129710>] (__warn+0xd8/0x114) [<c0129710>] (__warn) from [<c0922a48>] (warn_slowpath_fmt+0x5c/0xc4) [<c0922a48>] (warn_slowpath_fmt) from [<bf0496f8>] (snd_card_new+0x310/0x39c [snd]) [<bf0496f8>] (snd_card_new [snd]) from [<bf1d7df8>] (snd_soc_bind_card+0x334/0x9c4 [snd_soc_core]) [<bf1d7df8>] (snd_soc_bind_card [snd_soc_core]) from [<bf1e9cd8>] (devm_snd_soc_register_card+0x30/0x6c [snd_soc_core]) [<bf1e9cd8>] (devm_snd_soc_register_card [snd_soc_core]) from [<bf22d964>] (fsl_asoc_card_probe+0x550/0xcc8 [snd_soc_fsl_asoc_card]) [<bf22d964>] (fsl_asoc_card_probe [snd_soc_fsl_asoc_card]) from [<c060c930>] (platform_drv_probe+0x48/0x98) [...] Signed-off-by: Nicolas Cavallari <nicolas.cavallari(a)green-communications.fr> --- sound/soc/fsl/fsl-asoc-card.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index c62bfd1c3ac7..4f55b316cf0f 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -744,6 +744,7 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) /* Initialize sound card */ priv->pdev = pdev; priv->card.dev = &pdev->dev; + priv->card.owner = THIS_MODULE; ret = snd_soc_of_parse_card_name(&priv->card, "model"); if (ret) { snprintf(priv->name, sizeof(priv->name), "%s-audio", -- 2.32.0.rc0

3 2

[PATCH] ASoC: SOF: reset enabled_cores state at suspend
by Kai Vehmanen 01 Jun '21

01 Jun '21

The recent changes to use common code to power up/down DSP cores also removed the reset of the core state at suspend. It turns out this is still needed. When the firmware state is reset to SOF_FW_BOOT_NOT_STARTED, also enabled_cores should be reset, and existing DSP drivers depend on this. BugLink: https://github.com/thesofproject/linux/issues/2824 Fixes: 42077f08b3 ("ASoC: SOF: update dsp core power status in common APIs") Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> --- sound/soc/sof/pm.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sof/pm.c b/sound/soc/sof/pm.c index fd265803f7bc..c83fb6255961 100644 --- a/sound/soc/sof/pm.c +++ b/sound/soc/sof/pm.c @@ -256,6 +256,7 @@ static int sof_suspend(struct device *dev, bool runtime_suspend) /* reset FW state */ sdev->fw_state = SOF_FW_BOOT_NOT_STARTED; + sdev->enabled_cores_mask = 0; return ret; } base-commit: 058efb40641845432c52777443b3372dbc97c032 -- 2.31.1

2 1

[PATCH] ASoC: topology: Fix spelling mistake "vesion" -> "version"
by Colin King 01 Jun '21

01 Jun '21

From: Colin Ian King <colin.king(a)canonical.com> There are spelling mistakes in comments. Fix them. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/soc-topology.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 5e65b72910e9..0a24d0d409d2 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1881,7 +1881,7 @@ static void stream_caps_new_ver(struct snd_soc_tplg_stream_caps *dest, * @src: older version of pcm as a source * @pcm: latest version of pcm created from the source * - * Support from vesion 4. User should free the returned pcm manually. + * Support from version 4. User should free the returned pcm manually. */ static int pcm_new_ver(struct soc_tplg *tplg, struct snd_soc_tplg_pcm *src, @@ -2069,7 +2069,7 @@ static void set_link_hw_format(struct snd_soc_dai_link *link, * @src: old version of phyical link config as a source * @link: latest version of physical link config created from the source * - * Support from vesion 4. User need free the returned link config manually. + * Support from version 4. User need free the returned link config manually. */ static int link_new_ver(struct soc_tplg *tplg, struct snd_soc_tplg_link_config *src, @@ -2380,7 +2380,7 @@ static int soc_tplg_dai_elems_load(struct soc_tplg *tplg, * @src: old version of manifest as a source * @manifest: latest version of manifest created from the source * - * Support from vesion 4. Users need free the returned manifest manually. + * Support from version 4. Users need free the returned manifest manually. */ static int manifest_new_ver(struct soc_tplg *tplg, struct snd_soc_tplg_manifest *src, -- 2.31.1

2 1

[PATCH -next] ASoC: ti: omap-mcbsp: use DEVICE_ATTR_RW macro
by YueHaibing 01 Jun '21

01 Jun '21

Use DEVICE_ATTR_RW() helper instead of plain DEVICE_ATTR(), which makes the code a bit shorter and easier to read. Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/ti/omap-mcbsp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/ti/omap-mcbsp.c b/sound/soc/ti/omap-mcbsp.c index db47981768c5..4479d74f0a45 100644 --- a/sound/soc/ti/omap-mcbsp.c +++ b/sound/soc/ti/omap-mcbsp.c @@ -539,7 +539,7 @@ static ssize_t prop##_store(struct device *dev, \ return size; \ } \ \ -static DEVICE_ATTR(prop, 0644, prop##_show, prop##_store) +static DEVICE_ATTR_RW(prop) THRESHOLD_PROP_BUILDER(max_tx_thres); THRESHOLD_PROP_BUILDER(max_rx_thres); -- 2.17.1

4 3

[PATCH 0/5] ASoC: Constify snd_compress_ops
by Rikard Falkeborn 01 Jun '21

01 Jun '21

The only use of the static and global snd_compress_ops structs is to assign their address to the compress_ops field in the snd_soc_component_driver struct which is a pointer to const. Make them const to allow the compiler to put them in read-only memory. Rikard Falkeborn (5): ASoC: cs47125: Constify static struct snd_compress_ops ASoC: wm5102: Constify static struct snd_compress_ops ASoC: wm5110: Constify static struct snd_compress_ops ASoC: qcom: q6asm-dai: Constify static struct snd_compress_ops ASoC: SOF: Intel: Constify sof_probe_compressed_ops sound/soc/codecs/cs47l24.c | 2 +- sound/soc/codecs/wm5102.c | 2 +- sound/soc/codecs/wm5110.c | 2 +- sound/soc/qcom/qdsp6/q6asm-dai.c | 2 +- sound/soc/sof/compress.c | 2 +- sound/soc/sof/compress.h | 2 +- 6 files changed, 6 insertions(+), 6 deletions(-) -- 2.31.1

3 7

alsa-lib 1.2.5 no devices on Lenovo Carbon (gen 7) - pipewire
by GitHub issues - edited 01 Jun '21

01 Jun '21

alsa-project/alsa-lib issue #142 was edited from sambo57u: I use Fedora 34 and updated alsa-lib and all other alsa products to version 1.2.5. They work fine on my desktop machines but on Lenovo laptop I am getting the message alsactl cannot find hw:0 and daemon deactivates with no devices showing. Downgrading to 1.2.4 solves the problem. Issue URL : https://github.com/alsa-project/alsa-lib/issues/142 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Incomplete sound device detection in PulseAudio with alsa-lib 1.2.5
by GitHub issues - opened 01 Jun '21

01 Jun '21

alsa-project/alsa-lib issue #143 was opened from foutrelis: (Copying my comments over from #142 since it appears to be a different issue to that one. Sorry for the noise there!) After upgrading to alsa-lib 1.2.5 Chromium stopped playing audio until pulseaudio was restarted. Upon closer inspection, `pactl list cards` is now missing the `HDA Intel HDMI` card and only shows the `HDA Intel PCH` one. Some error messages I've gathered: - After upgrading to alsa-lib and before restarting pulseaudio, attempting to play a video in Chromium would result in: `pulseaudio[625]: Error opening PCM device front:0: Invalid argument` - Restarting pulseaudio with alsa-lib 1.2.5 installed logs the following to the journal: ``` pulseaudio[66345]: Failed to find a working profile. pulseaudio[66345]: Failed to load module "module-alsa-card" (argument: "device_id="1" name="pci-0000_00_03.0" card_name="alsa_card.pci-0000_00_03.0" namereg_fail=false tsched=yes fixed_latency_range=no ignore_dB=no deferred_volume=yes use_ucm=yes avoid_resampling=no card_properties="module-udev-detect.discovered=1""): initialization failed. ``` Git bisect points to commit 63f7745be504e447923f2cde421177a2fca99340 as the first commit that results in the `Failed to find a working profile` error. After restarting pulseaudio with alsa-lib 1.2.5 installed, it appears that `pactl list cards` shows one fewer cards on my laptop which has this audio hardware: ``` $ lspci | grep Audio 00:03.0 Audio device: Intel Corporation Haswell-ULT HD Audio Controller (rev 0b) 00:1b.0 Audio device: Intel Corporation 8 Series HD Audio Controller (rev 04) ``` [alsa-lib-1.2.4-pactl-list-cards.txt](https://github.com/alsa-project/alsa-l… [alsa-lib-1.2.5-pactl-list-cards.txt](https://github.com/alsa-project/alsa-l… `pulseaudio -v` output: [pulseaudio-verbose.txt](https://github.com/alsa-project/alsa-lib/files/6577… Issue URL : https://github.com/alsa-project/alsa-lib/issues/143 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH 0/5] ALSA: usb-audio: Delay account cleanup and latency reduction
by Takashi Iwai 01 Jun '21

01 Jun '21

Hi, this is a patch set for USB-audio driver for the code refactoring of PCM delay account as well as the reduction of the playback latency. The last change is a bit intrusive (despite of the smallness), and hopefully it won't break things. My initial tests through various backends were OK, but if anyone finds a problem, please let me know. Takashi === Takashi Iwai (5): ALSA: usb-audio: Make snd_usb_pcm_delay() static ALSA: usb-audio: Pre-calculate buffer byte size ALSA: usb-audio: Refactoring delay account code ALSA: usb-audio: Factor out DSD bitrev copy function ALSA: usb-audio: Reduce latency at playback start sound/usb/card.h | 8 +- sound/usb/endpoint.c | 1 + sound/usb/pcm.c | 202 ++++++++++++++++++++----------------------- sound/usb/pcm.h | 3 - 4 files changed, 101 insertions(+), 113 deletions(-) -- 2.26.2

1 5