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[PATCH] ASoC: wm8524: remove rate constraint for FE-BE case
by Shengjiu Wang 12 Apr '22

12 Apr '22

The constraint is propagate to Front End Bitstream for Front End and Back End share same snd_soc_pcm_runtime. The constraint is not needed for Back End Bitstream when there is be_hw_params_fixup() defined. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> Tested-by: Fabio Estevam <festevam(a)denx.de> --- sound/soc/codecs/wm8524.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm8524.c b/sound/soc/codecs/wm8524.c index 81f858f6bd67..442a59857875 100644 --- a/sound/soc/codecs/wm8524.c +++ b/sound/soc/codecs/wm8524.c @@ -61,6 +61,7 @@ static int wm8524_startup(struct snd_pcm_substream *substream, { struct snd_soc_component *component = dai->component; struct wm8524_priv *wm8524 = snd_soc_component_get_drvdata(component); + struct snd_soc_pcm_runtime *rtd = substream->private_data; /* The set of sample rates that can be supported depends on the * MCLK supplied to the CODEC - enforce this. @@ -71,9 +72,10 @@ static int wm8524_startup(struct snd_pcm_substream *substream, return -EINVAL; } - snd_pcm_hw_constraint_list(substream->runtime, 0, - SNDRV_PCM_HW_PARAM_RATE, - &wm8524->rate_constraint); + if (!rtd->dai_link->be_hw_params_fixup) + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &wm8524->rate_constraint); gpiod_set_value_cansleep(wm8524->mute, 1); -- 2.17.1

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[PATCH] ALSA: usb-audio: Limit max buffer and period sizes per time
by Takashi Iwai 12 Apr '22

12 Apr '22

In the previous fix, we increased the max buffer bytes from 1MB to 4MB so that we can use bigger buffers for the modern HiFi devices with higher rates, more channels and wider formats. OTOH, extending this has a concern that too big buffer is allowed for the lower rates, less channels and narrower formats; when an application tries to allocate as big buffer as possible, it'll lead to unexpectedly too huge size. Also, we had a problem about the inconsistent max buffer and period bytes for the implicit feedback mode when both streams have different channels. This was fixed by the (relatively complex) patch to reduce the max buffer and period bytes accordingly. This is an alternative fix for those, a patch to kill two birds with one stone (*): instead of increasing the max buffer bytes blindly and applying the reduction per channels, we simply use the hw constraints for the buffer and period "time". Meanwhile the max buffer and period bytes are set unlimited instead. Since the inconsistency of buffer (and period) bytes comes from the difference of the channels in the tied streams, as long as we care only about the buffer (and period) time, it doesn't matter; the buffer time is same for different channels, although we still allow higher buffer size. Similarly, this will allow more buffer bytes for HiFi devices while it also keeps the reasonable size for the legacy devices, too. As of this patch, the max period and buffer time are set to 1 and 2 seconds, which should be large enough for all possible use cases. (*) No animals were harmed in the making of this patch. Fixes: 98c27add5d96 ("ALSA: usb-audio: Cap upper limits of buffer/period bytes for implicit fb") Fixes: fee2ec8cceb3 ("ALSA: usb-audio: Increase max buffer size") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/pcm.c | 101 +++++++----------------------------------------- 1 file changed, 14 insertions(+), 87 deletions(-) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 37ee6df8b15a..6d699065e81a 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -659,9 +659,6 @@ static int snd_usb_pcm_prepare(struct snd_pcm_substream *substream) #define hwc_debug(fmt, args...) do { } while(0) #endif -#define MAX_BUFFER_BYTES (4 * 1024 * 1024) -#define MAX_PERIOD_BYTES (512 * 1024) - static const struct snd_pcm_hardware snd_usb_hardware = { .info = SNDRV_PCM_INFO_MMAP | @@ -672,9 +669,9 @@ static const struct snd_pcm_hardware snd_usb_hardware = SNDRV_PCM_INFO_PAUSE, .channels_min = 1, .channels_max = 256, - .buffer_bytes_max = MAX_BUFFER_BYTES, + .buffer_bytes_max = INT_MAX, /* limited by BUFFER_TIME later */ .period_bytes_min = 64, - .period_bytes_max = MAX_PERIOD_BYTES, + .period_bytes_max = INT_MAX, /* limited by PERIOD_TIME later */ .periods_min = 2, .periods_max = 1024, }; @@ -974,78 +971,6 @@ static int hw_rule_periods_implicit_fb(struct snd_pcm_hw_params *params, ep->cur_buffer_periods); } -/* get the adjusted max buffer (or period) bytes that can fit with the - * paired format for implicit fb - */ -static unsigned int -get_adjusted_max_bytes(struct snd_usb_substream *subs, - struct snd_usb_substream *pair, - struct snd_pcm_hw_params *params, - unsigned int max_bytes, - bool reverse_map) -{ - const struct audioformat *fp, *pp; - unsigned int rmax = 0, r; - - list_for_each_entry(fp, &subs->fmt_list, list) { - if (!fp->implicit_fb) - continue; - if (!reverse_map && - !hw_check_valid_format(subs, params, fp)) - continue; - list_for_each_entry(pp, &pair->fmt_list, list) { - if (pp->iface != fp->sync_iface || - pp->altsetting != fp->sync_altsetting || - pp->ep_idx != fp->sync_ep_idx) - continue; - if (reverse_map && - !hw_check_valid_format(pair, params, pp)) - break; - if (!reverse_map && pp->channels > fp->channels) - r = max_bytes * fp->channels / pp->channels; - else if (reverse_map && pp->channels < fp->channels) - r = max_bytes * pp->channels / fp->channels; - else - r = max_bytes; - rmax = max(rmax, r); - break; - } - } - return rmax; -} - -/* Reduce the period or buffer bytes depending on the paired substream; - * when a paired configuration for implicit fb has a higher number of channels, - * we need to reduce the max size accordingly, otherwise it may become unusable - */ -static int hw_rule_bytes_implicit_fb(struct snd_pcm_hw_params *params, - struct snd_pcm_hw_rule *rule) -{ - struct snd_usb_substream *subs = rule->private; - struct snd_usb_substream *pair; - struct snd_interval *it; - unsigned int max_bytes; - unsigned int rmax; - - pair = &subs->stream->substream[!subs->direction]; - if (!pair->ep_num) - return 0; - - if (rule->var == SNDRV_PCM_HW_PARAM_PERIOD_BYTES) - max_bytes = MAX_PERIOD_BYTES; - else - max_bytes = MAX_BUFFER_BYTES; - - rmax = get_adjusted_max_bytes(subs, pair, params, max_bytes, false); - if (!rmax) - rmax = get_adjusted_max_bytes(pair, subs, params, max_bytes, true); - if (!rmax) - return 0; - - it = hw_param_interval(params, rule->var); - return apply_hw_params_minmax(it, 0, rmax); -} - /* * set up the runtime hardware information. */ @@ -1139,6 +1064,18 @@ static int setup_hw_info(struct snd_pcm_runtime *runtime, struct snd_usb_substre return err; } + /* set max period and buffer sizes for 1 and 2 seconds, respectively */ + err = snd_pcm_hw_constraint_minmax(runtime, + SNDRV_PCM_HW_PARAM_PERIOD_TIME, + 0, 1000000); + if (err < 0) + return err; + err = snd_pcm_hw_constraint_minmax(runtime, + SNDRV_PCM_HW_PARAM_BUFFER_TIME, + 0, 2000000); + if (err < 0) + return err; + /* additional hw constraints for implicit fb */ err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_FORMAT, hw_rule_format_implicit_fb, subs, @@ -1160,16 +1097,6 @@ static int setup_hw_info(struct snd_pcm_runtime *runtime, struct snd_usb_substre SNDRV_PCM_HW_PARAM_PERIODS, -1); if (err < 0) return err; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, - hw_rule_bytes_implicit_fb, subs, - SNDRV_PCM_HW_PARAM_BUFFER_BYTES, -1); - if (err < 0) - return err; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, - hw_rule_bytes_implicit_fb, subs, - SNDRV_PCM_HW_PARAM_PERIOD_BYTES, -1); - if (err < 0) - return err; list_for_each_entry(fp, &subs->fmt_list, list) { if (fp->implicit_fb) { -- 2.34.1

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[PATCH 0/2] ALSA: echoaudio: Fix the probe error handling
by Takashi Iwai 12 Apr '22

12 Apr '22

Hi, this is a patch set to address the regression of the error handling in the echoaudio driver probe phase. Similar error patterns are seen in other drivers, and the newly introduced helper will be used later for those, too. Takashi === Takashi Iwai (2): ALSA: core: Add snd_card_free_on_error() helper ALSA: echoaudio: Fix the missing snd_card_free() call at probe error include/sound/core.h | 1 + sound/core/init.c | 28 ++++++++++++++++++++++++++++ sound/pci/echoaudio/echoaudio.c | 9 +++++++-- 3 files changed, 36 insertions(+), 2 deletions(-) -- 2.31.1

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[PATCH v2 1/2] rename tas2764 to tas27xx
by Raphael-Xu 12 Apr '22

12 Apr '22

rename tas2764.yaml to tas27xx.yaml Signed-off-by: Raphael-Xu <13691752556(a)139.com> --- .../devicetree/bindings/sound/{tas2764.yaml => tas27xx.yaml} | 0 1 file changed, 0 insertions(+), 0 deletions(-) rename Documentation/devicetree/bindings/sound/{tas2764.yaml => tas27xx.yaml} (100%) diff --git a/Documentation/devicetree/bindings/sound/tas2764.yaml b/Documentation/devicetree/bindings/sound/tas27xx.yaml similarity index 100% rename from Documentation/devicetree/bindings/sound/tas2764.yaml rename to Documentation/devicetree/bindings/sound/tas27xx.yaml -- 2.35.1

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Re: [bug report] firewire: add kernel API to access CYCLE_TIME register
by Takashi Sakamoto 12 Apr '22

12 Apr '22

Hi, On Mon, Apr 11, 2022 at 10:01:25AM +0300, Dan Carpenter wrote: > Hello Takashi Sakamoto, > > The patch baa914cd81f5: "firewire: add kernel API to access > CYCLE_TIME register" from Apr 5, 2022, leads to the following Smatch > static checker warning: > > drivers/firewire/core-cdev.c:1235 ioctl_get_cycle_timer2() > error: uninitialized symbol 'cycle_time'. > > drivers/firewire/core-cdev.c > 1209 static int ioctl_get_cycle_timer2(struct client *client, union ioctl_arg *arg) > 1210 { > 1211 struct fw_cdev_get_cycle_timer2 *a = &arg->get_cycle_timer2; > 1212 struct fw_card *card = client->device->card; > 1213 struct timespec64 ts = {0, 0}; > 1214 u32 cycle_time; > 1215 int ret = 0; > 1216 > 1217 local_irq_disable(); > 1218 > 1219 ret = fw_card_read_cycle_time(card, &cycle_time); > 1220 if (ret < 0) > 1221 goto end; > ^^^^^^^^ > "cycle_time" not initialized on error path. > > 1222 > 1223 switch (a->clk_id) { > 1224 case CLOCK_REALTIME: ktime_get_real_ts64(&ts); break; > 1225 case CLOCK_MONOTONIC: ktime_get_ts64(&ts); break; > 1226 case CLOCK_MONOTONIC_RAW: ktime_get_raw_ts64(&ts); break; > 1227 default: > 1228 ret = -EINVAL; > 1229 } > 1230 end: > 1231 local_irq_enable(); > 1232 > 1233 a->tv_sec = ts.tv_sec; > 1234 a->tv_nsec = ts.tv_nsec; > --> 1235 a->cycle_timer = cycle_time; > 1236 > 1237 return ret; > 1238 } Thanks for the report. Indeed, it leaks the unidentified value on kernel stack to userspace. I'll post fix later. Regards Takashi Sakamoto

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[PATCH v2] ALSA: jack: Access input_dev under mutex
by Amadeusz Sławiński 12 Apr '22

12 Apr '22

It is possible when using ASoC that input_dev is unregistered while calling snd_jack_report, which causes NULL pointer dereference. In order to prevent this serialize access to input_dev using mutex lock. Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> --- Changes: v2: - drop mutex in snd_jack_new - add comment that snd_jack_report() should be called in context where it can sleep --- include/sound/jack.h | 1 + sound/core/jack.c | 34 +++++++++++++++++++++++++++------- 2 files changed, 28 insertions(+), 7 deletions(-) diff --git a/include/sound/jack.h b/include/sound/jack.h index 1181f536557e..1ed90e2109e9 100644 --- a/include/sound/jack.h +++ b/include/sound/jack.h @@ -62,6 +62,7 @@ struct snd_jack { const char *id; #ifdef CONFIG_SND_JACK_INPUT_DEV struct input_dev *input_dev; + struct mutex input_dev_lock; int registered; int type; char name[100]; diff --git a/sound/core/jack.c b/sound/core/jack.c index d1e3055f2b6a..88493cc31914 100644 --- a/sound/core/jack.c +++ b/sound/core/jack.c @@ -42,8 +42,11 @@ static int snd_jack_dev_disconnect(struct snd_device *device) #ifdef CONFIG_SND_JACK_INPUT_DEV struct snd_jack *jack = device->device_data; - if (!jack->input_dev) + mutex_lock(&jack->input_dev_lock); + if (!jack->input_dev) { + mutex_unlock(&jack->input_dev_lock); return 0; + } /* If the input device is registered with the input subsystem * then we need to use a different deallocator. */ @@ -52,6 +55,7 @@ static int snd_jack_dev_disconnect(struct snd_device *device) else input_free_device(jack->input_dev); jack->input_dev = NULL; + mutex_unlock(&jack->input_dev_lock); #endif /* CONFIG_SND_JACK_INPUT_DEV */ return 0; } @@ -90,8 +94,11 @@ static int snd_jack_dev_register(struct snd_device *device) snprintf(jack->name, sizeof(jack->name), "%s %s", card->shortname, jack->id); - if (!jack->input_dev) + mutex_lock(&jack->input_dev_lock); + if (!jack->input_dev) { + mutex_unlock(&jack->input_dev_lock); return 0; + } jack->input_dev->name = jack->name; @@ -116,6 +123,7 @@ static int snd_jack_dev_register(struct snd_device *device) if (err == 0) jack->registered = 1; + mutex_unlock(&jack->input_dev_lock); return err; } #endif /* CONFIG_SND_JACK_INPUT_DEV */ @@ -517,9 +525,11 @@ int snd_jack_new(struct snd_card *card, const char *id, int type, return -ENOMEM; } - /* don't creat input device for phantom jack */ - if (!phantom_jack) { #ifdef CONFIG_SND_JACK_INPUT_DEV + mutex_init(&jack->input_dev_lock); + + /* don't create input device for phantom jack */ + if (!phantom_jack) { int i; jack->input_dev = input_allocate_device(); @@ -537,8 +547,8 @@ int snd_jack_new(struct snd_card *card, const char *id, int type, input_set_capability(jack->input_dev, EV_SW, jack_switch_types[i]); -#endif /* CONFIG_SND_JACK_INPUT_DEV */ } +#endif /* CONFIG_SND_JACK_INPUT_DEV */ err = snd_device_new(card, SNDRV_DEV_JACK, jack, &ops); if (err < 0) @@ -578,10 +588,14 @@ EXPORT_SYMBOL(snd_jack_new); void snd_jack_set_parent(struct snd_jack *jack, struct device *parent) { WARN_ON(jack->registered); - if (!jack->input_dev) + mutex_lock(&jack->input_dev_lock); + if (!jack->input_dev) { + mutex_unlock(&jack->input_dev_lock); return; + } jack->input_dev->dev.parent = parent; + mutex_unlock(&jack->input_dev_lock); } EXPORT_SYMBOL(snd_jack_set_parent); @@ -629,6 +643,8 @@ EXPORT_SYMBOL(snd_jack_set_key); /** * snd_jack_report - Report the current status of a jack + * Note: This function uses mutexes and should be called from a + * context which can sleep (such as a workqueue). * * @jack: The jack to report status for * @status: The current status of the jack @@ -654,8 +670,11 @@ void snd_jack_report(struct snd_jack *jack, int status) status & jack_kctl->mask_bits); #ifdef CONFIG_SND_JACK_INPUT_DEV - if (!jack->input_dev) + mutex_lock(&jack->input_dev_lock); + if (!jack->input_dev) { + mutex_unlock(&jack->input_dev_lock); return; + } for (i = 0; i < ARRAY_SIZE(jack->key); i++) { int testbit = ((SND_JACK_BTN_0 >> i) & ~mask_bits); @@ -675,6 +694,7 @@ void snd_jack_report(struct snd_jack *jack, int status) } input_sync(jack->input_dev); + mutex_unlock(&jack->input_dev_lock); #endif /* CONFIG_SND_JACK_INPUT_DEV */ } EXPORT_SYMBOL(snd_jack_report); -- 2.25.1

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[PATCH 0/2] Add support for HDA Jack poll in suspend state
by Mohan Kumar 12 Apr '22

12 Apr '22

This series is to add HDA jack polling support to update the Jack mixer control status properly even during runtime suspended state, as unsol event won't be triggered during D3 state. Mohan Kumar (2): ALSA: hda: Jack detection poll in suspend state ALSA: hda/tegra: Enable Jack poll for tegra include/sound/hda_codec.h | 3 +++ sound/pci/hda/hda_codec.c | 11 ++++++++++- sound/pci/hda/hda_tegra.c | 2 ++ 3 files changed, 15 insertions(+), 1 deletion(-) -- 2.17.1

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[BUG] ALSA: echoaudio: warning when the driver fails to probe
by Zheyu Ma 12 Apr '22

12 Apr '22

Hello, I found a bug in echoaudio.c. When the driver fails at the function snd_echo_create(), it should release resources requested before, otherwise we will get the following warning: [ 3.262866] remove_proc_entry: removing non-empty directory 'irq/21', leaking at least 'snd_indigodj' [ 3.263577] WARNING: CPU: 3 PID: 261 at fs/proc/generic.c:717 remove_proc_entry+0x389/0x3f0 [ 3.267098] RIP: 0010:remove_proc_entry+0x389/0x3f0 [ 3.269976] Call Trace: [ 3.269979] <TASK> [ 3.269988] unregister_irq_proc+0x14c/0x170 [ 3.269997] irq_free_descs+0x94/0xe0 [ 3.270004] mp_unmap_irq+0xb6/0x100 [ 3.270011] acpi_unregister_gsi_ioapic+0x27/0x40 [ 3.270017] acpi_pci_irq_disable+0x1d3/0x320 [ 3.270025] pci_disable_device+0x1ad/0x380 [ 3.270034] pcim_release+0x566/0x6d0 [ 3.270046] devres_release_all+0x1f1/0x2c0 [ 3.270057] really_probe+0xe0/0x920 Regards, Zheyu Ma

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[PATCH] slimbus: qcom: Fix IRQ check in qcom_slim_probe
by Miaoqian Lin 12 Apr '22

12 Apr '22

platform_get_irq() returns negative error number instead 0 on failure. And the doc of platform_get_irq() provides a usage example: int irq = platform_get_irq(pdev, 0); if (irq < 0) return irq; Fix the check of return value to catch errors correctly. Fixes: ad7fcbc308b0 ("slimbus: qcom: Add Qualcomm Slimbus controller driver") Signed-off-by: Miaoqian Lin <linmq006(a)gmail.com> --- drivers/slimbus/qcom-ctrl.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/drivers/slimbus/qcom-ctrl.c b/drivers/slimbus/qcom-ctrl.c index f04b961b96cd..ec58091fc948 100644 --- a/drivers/slimbus/qcom-ctrl.c +++ b/drivers/slimbus/qcom-ctrl.c @@ -510,9 +510,9 @@ static int qcom_slim_probe(struct platform_device *pdev) } ctrl->irq = platform_get_irq(pdev, 0); - if (!ctrl->irq) { + if (ctrl->irq < 0) { dev_err(&pdev->dev, "no slimbus IRQ\n"); - return -ENODEV; + return ctrl->irq; } sctrl = &ctrl->ctrl; -- 2.17.1

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[PATCH v1 1/2] rename tas2764 to tas27xx
by Raphael-Xu 11 Apr '22

11 Apr '22

rename tas2764.yaml to tas27xx.yaml Signed-off-by: Raphael-Xu <13691752556(a)139.com> --- .../devicetree/bindings/sound/{tas2764.yaml => tas27xx.yaml} | 0 1 file changed, 0 insertions(+), 0 deletions(-) rename Documentation/devicetree/bindings/sound/{tas2764.yaml => tas27xx.yaml} (100%) diff --git a/Documentation/devicetree/bindings/sound/tas2764.yaml b/Documentation/devicetree/bindings/sound/tas27xx.yaml similarity index 100% rename from Documentation/devicetree/bindings/sound/tas2764.yaml rename to Documentation/devicetree/bindings/sound/tas27xx.yaml -- 2.35.1

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