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[PATCH v2] ASoC: rsnd: Add check for rsnd_mod_power_on
by Jiasheng Jiang 02 Sep '22

02 Sep '22

As rsnd_mod_power_on() can return negative numbers, it should be better to check the return value and deal with the exception. Fixes: e7d850dd10f4 ("ASoC: rsnd: use mod base common method on SSI-parent") Signed-off-by: Jiasheng Jiang <jiasheng(a)iscas.ac.cn> --- Changelog: v1 -> v2: 1. Add check for rsnd_mod_power_on in other files. --- sound/soc/sh/rcar/ctu.c | 6 +++++- sound/soc/sh/rcar/dvc.c | 6 +++++- sound/soc/sh/rcar/mix.c | 6 +++++- sound/soc/sh/rcar/src.c | 5 ++++- sound/soc/sh/rcar/ssi.c | 4 +++- 5 files changed, 22 insertions(+), 5 deletions(-) diff --git a/sound/soc/sh/rcar/ctu.c b/sound/soc/sh/rcar/ctu.c index 6156445bcb69..e39eb2ac7e95 100644 --- a/sound/soc/sh/rcar/ctu.c +++ b/sound/soc/sh/rcar/ctu.c @@ -171,7 +171,11 @@ static int rsnd_ctu_init(struct rsnd_mod *mod, struct rsnd_dai_stream *io, struct rsnd_priv *priv) { - rsnd_mod_power_on(mod); + int ret; + + ret = rsnd_mod_power_on(mod); + if (ret < 0) + return ret; rsnd_ctu_activation(mod); diff --git a/sound/soc/sh/rcar/dvc.c b/sound/soc/sh/rcar/dvc.c index 5137e03a9d7c..16befcbc312c 100644 --- a/sound/soc/sh/rcar/dvc.c +++ b/sound/soc/sh/rcar/dvc.c @@ -186,7 +186,11 @@ static int rsnd_dvc_init(struct rsnd_mod *mod, struct rsnd_dai_stream *io, struct rsnd_priv *priv) { - rsnd_mod_power_on(mod); + int ret; + + ret = rsnd_mod_power_on(mod); + if (ret < 0) + return ret; rsnd_dvc_activation(mod); diff --git a/sound/soc/sh/rcar/mix.c b/sound/soc/sh/rcar/mix.c index 3572c2c5686c..1de0e085804c 100644 --- a/sound/soc/sh/rcar/mix.c +++ b/sound/soc/sh/rcar/mix.c @@ -146,7 +146,11 @@ static int rsnd_mix_init(struct rsnd_mod *mod, struct rsnd_dai_stream *io, struct rsnd_priv *priv) { - rsnd_mod_power_on(mod); + int ret; + + ret = rsnd_mod_power_on(mod); + if (ret < 0) + return ret; rsnd_mix_activation(mod); diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c index 0ea84ae57c6a..f832165e46bc 100644 --- a/sound/soc/sh/rcar/src.c +++ b/sound/soc/sh/rcar/src.c @@ -463,11 +463,14 @@ static int rsnd_src_init(struct rsnd_mod *mod, struct rsnd_priv *priv) { struct rsnd_src *src = rsnd_mod_to_src(mod); + int ret; /* reset sync convert_rate */ src->sync.val = 0; - rsnd_mod_power_on(mod); + ret = rsnd_mod_power_on(mod); + if (ret < 0) + return ret; rsnd_src_activation(mod); diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 43c5e27dc5c8..7ade6c5ed96f 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -480,7 +480,9 @@ static int rsnd_ssi_init(struct rsnd_mod *mod, ssi->usrcnt++; - rsnd_mod_power_on(mod); + ret = rsnd_mod_power_on(mod); + if (ret < 0) + return ret; rsnd_ssi_config_init(mod, io); -- 2.25.1

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[PATCH v4 0/2] lib/string_helpers: Introduce parse_int_array_user()
by Cezary Rojewski 02 Sep '22

02 Sep '22

Continuation of recent upstream discussion [1] regarding user string tokenization. First, parse_int_array_user() is introduced to allow for splitting specified user string into a sequence of integers. Makes use of get_options() internally so the parsing logic is not duplicated. With that done, redundant parts of the sound driver are removed. Originally similar functionality was added for the SOF sound driver. As more users are on the horizon, it is desirable to update existing string_helpers code and provide a unified solution. Changes in v4: - renamed the function to parse_int_array_user() - at the name several local variable names have been reworded to match the above Changes in v3: - relocated tokenize_user_input() implementation to string_helpers as requested by Matthew Changes in v2: - reused get_options() so no parsing logic is duplicated - simplified __user variant with help of memdup_user_nul() Both suggested by Andy, thanks for thourough review [1]: https://lore.kernel.org/alsa-devel/20220707091301.1282291-1-cezary.rojewski… Cezary Rojewski (2): lib/string_helpers: Introduce parse_int_array_user() ASoC: SOF: Remove strsplit_u32() and tokenize_input() include/linux/string_helpers.h | 2 + lib/string_helpers.c | 45 +++++++++++++ sound/soc/sof/sof-client-probes.c | 104 ++++++------------------------ 3 files changed, 65 insertions(+), 86 deletions(-) -- 2.25.1

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[PATCH v3 0/2] lib/string_helpers: Introduce tokenize_user_input()
by Cezary Rojewski 02 Sep '22

02 Sep '22

Continuation of recent upstream discussion [1] regarding user string tokenization. First, tokenize_user_input() is introduced to allow for splitting specified user string into a sequence of integers. Makes use of get_options() internally so the parsing logic is not duplicated. With that done, redundant parts of the sound driver are removed. Originally similar functionality was added for the SOF sound driver. As more users are on the horizon, it is desirable to update existing string_helpers code and provide a unified solution. Changes in v3: - relocated tokenize_user_input() implementation to string_helpers as requested by Matthew Changes in v2: - reused get_options() so no parsing logic is duplicated - simplified __user variant with help of memdup_user_nul() Both suggested by Andy, thanks for thourough review [1]: https://lore.kernel.org/alsa-devel/20220707091301.1282291-1-cezary.rojewski… Cezary Rojewski (2): lib/string_helpers: Introduce tokenize_user_input() ASoC: SOF: Remove strsplit_u32() and tokenize_input() include/linux/string_helpers.h | 2 + lib/string_helpers.c | 45 +++++++++++++++ sound/soc/sof/sof-client-probes.c | 93 ++++--------------------------- 3 files changed, 59 insertions(+), 81 deletions(-) -- 2.25.1

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[PATCH v2 0/2] Introduce sof_of_machine_select
by Chunxu Li 02 Sep '22

02 Sep '22

From: "chunxu.li" <chunxu.li(a)mediatek.com> Change since v1: - remove the callback of_machine_select defined in sof-priv.h - move sof_of_machine_select to common code, and called in sof_machine_check - rename .board field to .compatible in structure snd_sof_of_mach In these patches, we introduce function sof_of_machine_select for SOF Chunxu Li (2): ASoC: SOF: Introduce function sof_of_machine_select ASoC: SOF: mediatek: Add sof_mt8186_machs for mt8186 include/sound/sof.h | 2 ++ sound/soc/sof/mediatek/mt8186/mt8186.c | 9 ++++++++ sound/soc/sof/pcm.c | 8 ++++++- sound/soc/sof/sof-audio.c | 30 ++++++++++++++++++++++++++ sound/soc/sof/sof-of-dev.h | 7 ++++++ 5 files changed, 55 insertions(+), 1 deletion(-) -- 2.25.1

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[PATCH] ALSA: aloop: Fix random zeros in capture data when using jiffies timer
by Pattara Teerapong 02 Sep '22

02 Sep '22

In loopback_jiffies_timer_pos_update(), we are getting jiffies twice. First time for playback, second time for capture. Jiffies can be updated between these two calls and if the capture jiffies is larger, extra zeros will be filled in the capture buffer. Change to get jiffies once and use it for both playback and capture. Signed-off-by: Pattara Teerapong <pteerapong(a)chromium.org> --- sound/drivers/aloop.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c index 9b4a7cdb103a..12f12a294df5 100644 --- a/sound/drivers/aloop.c +++ b/sound/drivers/aloop.c @@ -605,17 +605,18 @@ static unsigned int loopback_jiffies_timer_pos_update cable->streams[SNDRV_PCM_STREAM_PLAYBACK]; struct loopback_pcm *dpcm_capt = cable->streams[SNDRV_PCM_STREAM_CAPTURE]; - unsigned long delta_play = 0, delta_capt = 0; + unsigned long delta_play = 0, delta_capt = 0, cur_jiffies; unsigned int running, count1, count2; + cur_jiffies = jiffies; running = cable->running ^ cable->pause; if (running & (1 << SNDRV_PCM_STREAM_PLAYBACK)) { - delta_play = jiffies - dpcm_play->last_jiffies; + delta_play = cur_jiffies - dpcm_play->last_jiffies; dpcm_play->last_jiffies += delta_play; } if (running & (1 << SNDRV_PCM_STREAM_CAPTURE)) { - delta_capt = jiffies - dpcm_capt->last_jiffies; + delta_capt = cur_jiffies - dpcm_capt->last_jiffies; dpcm_capt->last_jiffies += delta_capt; } -- 2.37.2.789.g6183377224-goog

2 1

retire_capture_urb: Corrected urb data len
by Sean Anderson 02 Sep '22

02 Sep '22

Hi all, I have a "FiiO DigiHug USB Audio" sound card (1852:7022) [3]. I have had no problems with the audio, but I did notice a large number of message like retire_capture_urb: 4992 callbacks suppressed in my dmesg [1]. This is caused by the "Corrected urb data len." warning. The patch adding this warning [2] makes it seem like this warning should be an uncommon occurance. However, based on the number of suppressed callbacks, this seems to be happening at a rate of around 500 Hz. Is this buggy hardware? Or is this a bug in the driver? Does there need to be a quirk? Or perhaps the warning above should be a debug instead? I'm using 5.15.0-46-generic. I know, I know; this is a vendor kernel. However, I went through the commits for sound/usb/pcm.c and didn't see any major differences. I also previously saw this issue on 5.4, so this probably isn't a regression. --Sean [1] I would have liked to post the actual message, but I was unable to figure out how to disable ratelimiting. I tried setting /proc/sys/kernel/printk_ratelimit to 0, /proc/sys/kernel/printk_ratelimit_burst to 5000, and /proc/sys/kernel/printk to 8. However, nothing seemed to have any effect (except that the callbacks suppressed messages were disabled). I find this very strange, given that the callbacks suppressed message uses KERN_WARN, and so does dev_warn_ratelimited. [2] https://lore.kernel.org/all/4B28A659.6070303@ladisch.de/T/ [3] Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 1.10 bDeviceClass 0 bDeviceSubClass 0 bDeviceProtocol 0 bMaxPacketSize0 8 idVendor 0x1852 GYROCOM C&C Co., LTD idProduct 0x7022 bcdDevice 0.01 iManufacturer 1 FiiO iProduct 2 DigiHug USB Audio iSerial 0 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 0x0182 bNumInterfaces 4 bConfigurationValue 1 iConfiguration 0 bmAttributes 0x80 (Bus Powered) MaxPower 500mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 1 bInterfaceClass 3 Human Interface Device bInterfaceSubClass 0 bInterfaceProtocol 0 iInterface 0 HID Device Descriptor: bLength 9 bDescriptorType 33 bcdHID 1.00 bCountryCode 0 Not supported bNumDescriptors 1 bDescriptorType 34 Report wDescriptorLength 58 Report Descriptors: ** UNAVAILABLE ** Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0012 1x 18 bytes bInterval 32 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 1 Control Device bInterfaceProtocol 0 iInterface 3 FiiO USB DAC-E10 AudioControl Interface Descriptor: bLength 10 bDescriptorType 36 bDescriptorSubtype 1 (HEADER) bcdADC 1.00 wTotalLength 0x003e bInCollection 2 baInterfaceNr(0) 2 baInterfaceNr(1) 3 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 5 wTerminalType 0x0605 SPDIF interface bAssocTerminal 0 bNrChannels 2 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 9 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bNrChannels 2 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 3 wTerminalType 0x0605 SPDIF interface bAssocTerminal 0 bSourceID 16 iTerminal 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 7 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bSourceID 5 iTerminal 0 AudioControl Interface Descriptor: bLength 10 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 16 bSourceID 9 bControlSize 1 bmaControls(0) 0x01 Mute Control bmaControls(1) 0x02 Volume Control bmaControls(2) 0x02 Volume Control iFeature 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 7 bDelay 0 frames wFormatTag 0x0001 PCM AudioStreaming Interface Descriptor: bLength 26 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 2 bBitResolution 16 bSamFreqType 6 Discrete tSamFreq[ 0] 8000 tSamFreq[ 1] 16000 tSamFreq[ 2] 32000 tSamFreq[ 3] 44100 tSamFreq[ 4] 48000 tSamFreq[ 5] 96000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x82 EP 2 IN bmAttributes 9 Transfer Type Isochronous Synch Type Adaptive Usage Type Data wMaxPacketSize 0x0184 1x 388 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioStreaming Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 2 Decoded PCM samples wLockDelay 0x0002 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 2 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 7 bDelay 0 frames wFormatTag 0x0001 PCM AudioStreaming Interface Descriptor: bLength 26 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 3 bBitResolution 24 bSamFreqType 6 Discrete tSamFreq[ 0] 8000 tSamFreq[ 1] 16000 tSamFreq[ 2] 32000 tSamFreq[ 3] 44100 tSamFreq[ 4] 48000 tSamFreq[ 5] 96000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x82 EP 2 IN bmAttributes 9 Transfer Type Isochronous Synch Type Adaptive Usage Type Data wMaxPacketSize 0x0246 1x 582 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioStreaming Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 2 Decoded PCM samples wLockDelay 0x0002 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 9 bDelay 0 frames wFormatTag 0x0001 PCM AudioStreaming Interface Descriptor: bLength 20 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 2 bBitResolution 16 bSamFreqType 4 Discrete tSamFreq[ 0] 32000 tSamFreq[ 1] 44100 tSamFreq[ 2] 48000 tSamFreq[ 3] 96000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x03 EP 3 OUT bmAttributes 9 Transfer Type Isochronous Synch Type Adaptive Usage Type Data wMaxPacketSize 0x0184 1x 388 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioStreaming Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 2 Decoded PCM samples wLockDelay 0x0002 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 2 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 9 bDelay 0 frames wFormatTag 0x0001 PCM AudioStreaming Interface Descriptor: bLength 20 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 3 bBitResolution 24 bSamFreqType 4 Discrete tSamFreq[ 0] 32000 tSamFreq[ 1] 44100 tSamFreq[ 2] 48000 tSamFreq[ 3] 96000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x03 EP 3 OUT bmAttributes 9 Transfer Type Isochronous Synch Type Adaptive Usage Type Data wMaxPacketSize 0x0246 1x 582 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioStreaming Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 2 Decoded PCM samples wLockDelay 0x0002 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 3 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 9 bDelay 0 frames wFormatTag 0x2001 IEC1937_AC-3 AudioStreaming Interface Descriptor: bLength 11 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 3 (FORMAT_TYPE_III) bNrChannels 2 bSubframeSize 2 bBitResolution 16 bSamFreqType 1 Discrete tSamFreq[ 0] 48000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x03 EP 3 OUT bmAttributes 9 Transfer Type Isochronous Synch Type Adaptive Usage Type Data wMaxPacketSize 0x00c0 1x 192 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioStreaming Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 2 Decoded PCM samples wLockDelay 0x0002 can't get debug descriptor: Resource temporarily unavailable Device Status: 0x0000 (Bus Powered)

2 5

[PATCH] ASoC: rsnd: Add check for rsnd_mod_power_on
by Jiasheng Jiang 02 Sep '22

02 Sep '22

As rsnd_mod_power_on() can return negative numbers, it should be better to check the return value and deal with the exception. Fixes: e7d850dd10f4 ("ASoC: rsnd: use mod base common method on SSI-parent") Signed-off-by: Jiasheng Jiang <jiasheng(a)iscas.ac.cn> --- sound/soc/sh/rcar/ssi.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 43c5e27dc5c8..7ade6c5ed96f 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -480,7 +480,9 @@ static int rsnd_ssi_init(struct rsnd_mod *mod, ssi->usrcnt++; - rsnd_mod_power_on(mod); + ret = rsnd_mod_power_on(mod); + if (ret < 0) + return ret; rsnd_ssi_config_init(mod, io); -- 2.25.1

2 1

[PATCH] ALSA: usb-audio: Clean up endpoint setups at PCM prepare
by Takashi Iwai 01 Sep '22

01 Sep '22

This patch cleans up the superfluous checks and calls for setting up the endpoints at PCM prepare callback: - Drop stop_endpoints() and sync_pending_stops() calls; the stream is guaranteed to have been already stopped and synced at each PCM prepare call by ALSA PCM core - Call snd_usb_endpoint_prepare() unconditionally; the check for endpoint->need_setup is done in snd_pcm_hw_endpoint_prepare() itself - Apply snd_usb_set_format_quirk() only when the endpoint is actually set up (i.e. the return code from snd_usb_endpoint_prepare() > 0) - Move a few lines back into snd_usb_pcm_prepare(); it's even easier to follow than a small useless function Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/pcm.c | 40 ++++++++++------------------------------ 1 file changed, 10 insertions(+), 30 deletions(-) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index b604f7e95e82..4ed53a3dc922 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -433,35 +433,6 @@ static void close_endpoints(struct snd_usb_audio *chip, } } -static int configure_endpoints(struct snd_usb_audio *chip, - struct snd_usb_substream *subs) -{ - int err; - - if (subs->data_endpoint->need_setup) { - /* stop any running stream beforehand */ - if (stop_endpoints(subs, false)) - sync_pending_stops(subs); - if (subs->sync_endpoint) { - err = snd_usb_endpoint_prepare(chip, subs->sync_endpoint); - if (err < 0) - return err; - } - err = snd_usb_endpoint_prepare(chip, subs->data_endpoint); - if (err < 0) - return err; - snd_usb_set_format_quirk(subs, subs->cur_audiofmt); - } else { - if (subs->sync_endpoint) { - err = snd_usb_endpoint_prepare(chip, subs->sync_endpoint); - if (err < 0) - return err; - } - } - - return 0; -} - /* * hw_params callback * @@ -640,9 +611,18 @@ static int snd_usb_pcm_prepare(struct snd_pcm_substream *substream) goto unlock; } - ret = configure_endpoints(chip, subs); + if (subs->sync_endpoint) { + ret = snd_usb_endpoint_prepare(chip, subs->sync_endpoint); + if (ret < 0) + goto unlock; + } + + ret = snd_usb_endpoint_prepare(chip, subs->data_endpoint); if (ret < 0) goto unlock; + else if (ret > 0) + snd_usb_set_format_quirk(subs, subs->cur_audiofmt); + ret = 0; /* reset the pointer */ subs->buffer_bytes = frames_to_bytes(runtime, runtime->buffer_size); -- 2.35.3

1 0

[PATCH 1/2] ASoC: apple: mca: Unselect COMMON_CLK in Kconfig
by Martin Povišer 01 Sep '22

01 Sep '22

The MCA driver selects COMMON_CLK, which leads to misconfiguration on platforms with HAVE_LEGACY_CLK (under compile test). Kconfig catches the conflict with the following warning: WARNING: unmet direct dependencies detected for COMMON_CLK Depends on [n]: !HAVE_LEGACY_CLK [=y] Selected by [m]: - SND_SOC_APPLE_MCA [=m] && SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && (ARCH_APPLE || COMPILE_TEST [=y]) Eventually the build fails with errors like: >> drivers/clk/clk.c:867:6: error: redefinition of 'clk_unprepare' 867 | void clk_unprepare(struct clk *clk) | ^~~~~~~~~~~~~ In file included from drivers/clk/clk.c:9: include/linux/clk.h:303:20: note: previous definition of 'clk_unprepare' with type 'void(struct clk *)' 303 | static inline void clk_unprepare(struct clk *clk) | ^~~~~~~~~~~~~ which appears to be because COMMON_CLK is selected but HAVE_CLK_PREPARE is not. In the end it seems we had no business selecting COMMON_CLK from an unrelated driver like that, so remove the selection. The linux/clk.h API is there anyway. Fixes: 3df5d0d97289 ("ASoC: apple: mca: Start new platform driver") Reported-by: kernel test robot <lkp(a)intel.com> Signed-off-by: Martin Povišer <povik+lin(a)cutebit.org> --- sound/soc/apple/Kconfig | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/apple/Kconfig b/sound/soc/apple/Kconfig index 0ba955657e98..793f7782e0d7 100644 --- a/sound/soc/apple/Kconfig +++ b/sound/soc/apple/Kconfig @@ -2,7 +2,6 @@ config SND_SOC_APPLE_MCA tristate "Apple Silicon MCA driver" depends on ARCH_APPLE || COMPILE_TEST select SND_DMAENGINE_PCM - select COMMON_CLK default ARCH_APPLE help This option enables an ASoC platform driver for MCA peripherals found -- 2.33.0

2 2

[PATCH] ALSA: usb-audio: Split endpoint setups for hw_params and prepare
by Takashi Iwai 01 Sep '22

01 Sep '22

One of the former changes for the endpoint management was the more consistent setup of endpoints at hw_params. snd_usb_endpoint_configure() is a single function that does the full setup, and it's called from both PCM hw_params and prepare callbacks. Although the EP setup at the prepare phase is usually skipped (by checking need_setup flag), it may be still effective in some cases like suspend/resume that requires the interface setup again. As it's a full and single setup, the invocation of snd_usb_endpoint_configure() includes not only the USB interface setup but also the buffer release and allocation. OTOH, doing the buffer release and re-allocation at PCM prepare phase is rather superfluous, and better to be done only in the hw_params phase. For those optimizations, this patch splits the endpoint setup to two phases: snd_usb_endpoint_set_params() and snd_usb_endpoint_prepare(), to be called from hw_params and from prepare, respectively. Note that this patch changes the driver operation slightly, effectively moving the USB interface setup again to PCM prepare stage instead of hw_params stage, while the buffer allocation and such initializations are still done at hw_params stage. And, the change of the USB interface setup timing (moving to prepare) gave an interesting "fix", too: it was reported that the recent kernels caused silent output at the beginning on playbacks on some devices on Android, and this change casually fixed the regression. It seems that those devices are picky about the sample rate change (or the interface change?), and don't follow the too immediate rate changes. Meanwhile, Android operates the PCM in the following order: - open, then hw_params with the possibly highest sample rate - close without prepare - re-open, hw_params with the normal sample rate - prepare, and start streaming This procedure ended up the hw_params twice with different rates, and because the recent kernel did set up the sample rate twice one and after, it screwed up the device. OTOH, the earlier kernels didn't set up the USB interface at hw_params, hence this problem didn't appear. Now, with this patch, the USB interface setup is again back to the prepare phase, and it works around the problem automagically. Although we should address the sample rate problem in a more solid way in future, let's keep things working as before for now. Fixes: bf6313a0ff76 ("ALSA: usb-audio: Refactor endpoint management") Cc: <stable(a)vger.kernel.org> Reported-by: chihhao chen <chihhao.chen(a)mediatek.com> Link: https://lore.kernel.org/r/87e6d6ae69d68dc588ac9acc8c0f24d6188375c3.camel@me… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/endpoint.c | 23 +++++++++-------------- sound/usb/endpoint.h | 6 ++++-- sound/usb/pcm.c | 14 ++++++++++---- 3 files changed, 23 insertions(+), 20 deletions(-) diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 0d7b73bf7945..a42f2ce19455 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -758,7 +758,8 @@ bool snd_usb_endpoint_compatible(struct snd_usb_audio *chip, * The endpoint needs to be closed via snd_usb_endpoint_close() later. * * Note that this function doesn't configure the endpoint. The substream - * needs to set it up later via snd_usb_endpoint_configure(). + * needs to set it up later via snd_usb_endpoint_set_params() and + * snd_usb_endpoint_prepare(). */ struct snd_usb_endpoint * snd_usb_endpoint_open(struct snd_usb_audio *chip, @@ -1290,12 +1291,13 @@ static int sync_ep_set_params(struct snd_usb_endpoint *ep) /* * snd_usb_endpoint_set_params: configure an snd_usb_endpoint * + * It's called either from hw_params callback. * Determine the number of URBs to be used on this endpoint. * An endpoint must be configured before it can be started. * An endpoint that is already running can not be reconfigured. */ -static int snd_usb_endpoint_set_params(struct snd_usb_audio *chip, - struct snd_usb_endpoint *ep) +int snd_usb_endpoint_set_params(struct snd_usb_audio *chip, + struct snd_usb_endpoint *ep) { const struct audioformat *fmt = ep->cur_audiofmt; int err; @@ -1378,18 +1380,18 @@ static int init_sample_rate(struct snd_usb_audio *chip, } /* - * snd_usb_endpoint_configure: Configure the endpoint + * snd_usb_endpoint_prepare: Prepare the endpoint * * This function sets up the EP to be fully usable state. - * It's called either from hw_params or prepare callback. + * It's called either from prepare callback. * The function checks need_setup flag, and performs nothing unless needed, * so it's safe to call this multiple times. * * This returns zero if unchanged, 1 if the configuration has changed, * or a negative error code. */ -int snd_usb_endpoint_configure(struct snd_usb_audio *chip, - struct snd_usb_endpoint *ep) +int snd_usb_endpoint_prepare(struct snd_usb_audio *chip, + struct snd_usb_endpoint *ep) { bool iface_first; int err = 0; @@ -1410,9 +1412,6 @@ int snd_usb_endpoint_configure(struct snd_usb_audio *chip, if (err < 0) goto unlock; } - err = snd_usb_endpoint_set_params(chip, ep); - if (err < 0) - goto unlock; goto done; } @@ -1440,10 +1439,6 @@ int snd_usb_endpoint_configure(struct snd_usb_audio *chip, if (err < 0) goto unlock; - err = snd_usb_endpoint_set_params(chip, ep); - if (err < 0) - goto unlock; - err = snd_usb_select_mode_quirk(chip, ep->cur_audiofmt); if (err < 0) goto unlock; diff --git a/sound/usb/endpoint.h b/sound/usb/endpoint.h index 6a9af04cf175..e67ea28faa54 100644 --- a/sound/usb/endpoint.h +++ b/sound/usb/endpoint.h @@ -17,8 +17,10 @@ snd_usb_endpoint_open(struct snd_usb_audio *chip, bool is_sync_ep); void snd_usb_endpoint_close(struct snd_usb_audio *chip, struct snd_usb_endpoint *ep); -int snd_usb_endpoint_configure(struct snd_usb_audio *chip, - struct snd_usb_endpoint *ep); +int snd_usb_endpoint_set_params(struct snd_usb_audio *chip, + struct snd_usb_endpoint *ep); +int snd_usb_endpoint_prepare(struct snd_usb_audio *chip, + struct snd_usb_endpoint *ep); int snd_usb_endpoint_get_clock_rate(struct snd_usb_audio *chip, int clock); bool snd_usb_endpoint_compatible(struct snd_usb_audio *chip, diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index d45d1d7e6664..b604f7e95e82 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -443,17 +443,17 @@ static int configure_endpoints(struct snd_usb_audio *chip, if (stop_endpoints(subs, false)) sync_pending_stops(subs); if (subs->sync_endpoint) { - err = snd_usb_endpoint_configure(chip, subs->sync_endpoint); + err = snd_usb_endpoint_prepare(chip, subs->sync_endpoint); if (err < 0) return err; } - err = snd_usb_endpoint_configure(chip, subs->data_endpoint); + err = snd_usb_endpoint_prepare(chip, subs->data_endpoint); if (err < 0) return err; snd_usb_set_format_quirk(subs, subs->cur_audiofmt); } else { if (subs->sync_endpoint) { - err = snd_usb_endpoint_configure(chip, subs->sync_endpoint); + err = snd_usb_endpoint_prepare(chip, subs->sync_endpoint); if (err < 0) return err; } @@ -551,7 +551,13 @@ static int snd_usb_hw_params(struct snd_pcm_substream *substream, subs->cur_audiofmt = fmt; mutex_unlock(&chip->mutex); - ret = configure_endpoints(chip, subs); + if (subs->sync_endpoint) { + ret = snd_usb_endpoint_set_params(chip, subs->sync_endpoint); + if (ret < 0) + goto unlock; + } + + ret = snd_usb_endpoint_set_params(chip, subs->data_endpoint); unlock: if (ret < 0) -- 2.35.3

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