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[PATCH] ASoC: ak5558: Fix s/show/slow/ typo
by Shengjiu Wang 06 Apr '21

06 Apr '21

s/show/slow/ Fixes: 920884777480 ("ASoC: ak5558: Add support for AK5558 ADC driver") Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/codecs/ak5558.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/ak5558.c b/sound/soc/codecs/ak5558.c index 3ac62a281807..8e4dca753f0b 100644 --- a/sound/soc/codecs/ak5558.c +++ b/sound/soc/codecs/ak5558.c @@ -75,8 +75,8 @@ static const struct soc_enum ak5552_mono_enum[] = { }; static const char * const digfil_texts[] = { - "Sharp Roll-Off", "Show Roll-Off", - "Short Delay Sharp Roll-Off", "Short Delay Show Roll-Off", + "Sharp Roll-Off", "Slow Roll-Off", + "Short Delay Sharp Roll-Off", "Short Delay Slow Roll-Off", }; static const struct soc_enum ak5558_adcset_enum[] = { -- 2.27.0

2 1

[PATCH v3 0/2] kbl_da7219_max9357a machine changes for wov and MST
by vamshi.krishna.gopal＠intel.com 06 Apr '21

06 Apr '21

From: Vamshi Krishna Gopal <vamshi.krishna.gopal(a)intel.corp-partner.google.com> Hi all, This patch series about creating dailink for Wake on voice functionality and also adding MST route changes. changes in v3: -Remove DP from widgets -Add MST support in other routing table for kbl_rt5663_max98927.c v2 is here: https://patchwork.kernel.org/project/alsa-devel/cover/20210325174325.31802-… v1 is here: https://patchwork.kernel.org/project/alsa-devel/cover/20210324175200.44922-… Mac Chiang (1): ASoc: Intel: board: add BE DAI link for WoV Vamshi Krishna Gopal (1): ASoC: Intel: kbl: Add MST route change to kbl machine drivers sound/soc/intel/boards/kbl_da7219_max98357a.c | 58 +++++++++++++++++-- sound/soc/intel/boards/kbl_da7219_max98927.c | 10 ++-- sound/soc/intel/boards/kbl_rt5663_max98927.c | 14 +++-- 3 files changed, 70 insertions(+), 12 deletions(-) -- 2.17.1

3 4

[PATCH] ASoC: max98390: Add support for tx slot configuration.
by Steve Lee 06 Apr '21

06 Apr '21

Update voltage/current tx slot configuration support. Signed-off-by: Steve Lee <steves.lee(a)maximintegrated.com> --- sound/soc/codecs/max98390.c | 62 +++++++++++++++++++++++++++++++++++++ sound/soc/codecs/max98390.h | 2 ++ 2 files changed, 64 insertions(+) diff --git a/sound/soc/codecs/max98390.c b/sound/soc/codecs/max98390.c index bb736c44e68a..94773ccee9d5 100644 --- a/sound/soc/codecs/max98390.c +++ b/sound/soc/codecs/max98390.c @@ -856,6 +856,48 @@ static void max98390_init_regs(struct snd_soc_component *component) regmap_write(max98390->regmap, MAX98390_ENV_TRACK_VOUT_HEADROOM, 0x0e); regmap_write(max98390->regmap, MAX98390_BOOST_BYPASS1, 0x46); regmap_write(max98390->regmap, MAX98390_FET_SCALING3, 0x03); + + /* voltage, current slot configuration */ + regmap_write(max98390->regmap, + MAX98390_PCM_CH_SRC_2, + (max98390->i_l_slot << 4 | + max98390->v_l_slot)&0xFF); + + if (max98390->v_l_slot < 8) { + regmap_update_bits(max98390->regmap, + MAX98390_PCM_TX_HIZ_CTRL_A, + 1 << max98390->v_l_slot, 0); + regmap_update_bits(max98390->regmap, + MAX98390_PCM_TX_EN_A, + 1 << max98390->v_l_slot, + 1 << max98390->v_l_slot); + } else { + regmap_update_bits(max98390->regmap, + MAX98390_PCM_TX_HIZ_CTRL_B, + 1 << (max98390->v_l_slot - 8), 0); + regmap_update_bits(max98390->regmap, + MAX98390_PCM_TX_EN_B, + 1 << (max98390->v_l_slot - 8), + 1 << (max98390->v_l_slot - 8)); + } + + if (max98390->i_l_slot < 8) { + regmap_update_bits(max98390->regmap, + MAX98390_PCM_TX_HIZ_CTRL_A, + 1 << max98390->i_l_slot, 0); + regmap_update_bits(max98390->regmap, + MAX98390_PCM_TX_EN_A, + 1 << max98390->i_l_slot, + 1 << max98390->i_l_slot); + } else { + regmap_update_bits(max98390->regmap, + MAX98390_PCM_TX_HIZ_CTRL_B, + 1 << (max98390->i_l_slot - 8), 0); + regmap_update_bits(max98390->regmap, + MAX98390_PCM_TX_EN_B, + 1 << (max98390->i_l_slot - 8), + 1 << (max98390->i_l_slot - 8)); + } } static int max98390_probe(struct snd_soc_component *component) @@ -946,6 +988,23 @@ static const struct regmap_config max98390_regmap = { .cache_type = REGCACHE_RBTREE, }; +static void max98390_slot_config(struct i2c_client *i2c, + struct max98390_priv *max98390) +{ + int value; + struct device *dev = &i2c->dev; + + if (!device_property_read_u32(dev, "maxim,vmon-slot-no", &value)) + max98390->v_l_slot = value & 0xF; + else + max98390->v_l_slot = 0; + + if (!device_property_read_u32(dev, "maxim,imon-slot-no", &value)) + max98390->i_l_slot = value & 0xF; + else + max98390->i_l_slot = 1; +} + static int max98390_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -988,6 +1047,9 @@ static int max98390_i2c_probe(struct i2c_client *i2c, __func__, max98390->ref_rdc_value, max98390->ambient_temp_value); + /* voltage/current slot configuration */ + max98390_slot_config(i2c, max98390); + /* regmap initialization */ max98390->regmap = devm_regmap_init_i2c(i2c, &max98390_regmap); if (IS_ERR(max98390->regmap)) { diff --git a/sound/soc/codecs/max98390.h b/sound/soc/codecs/max98390.h index dff884f68e3e..e31516717d3b 100644 --- a/sound/soc/codecs/max98390.h +++ b/sound/soc/codecs/max98390.h @@ -658,6 +658,8 @@ struct max98390_priv { unsigned int sysclk; unsigned int master; unsigned int tdm_mode; + unsigned int v_l_slot; + unsigned int i_l_slot; unsigned int ref_rdc_value; unsigned int ambient_temp_value; }; -- 2.17.1

2 1

dsnoop/dshare/dmix: snd_pcm_htimestamp returns time past since last boot not clock time
by GitHub issues - edited 06 Apr '21

06 Apr '21

alsa-project/alsa-lib issue #131 was edited from RonaldAJ: While on a PC snd_pcm_htimestamp returns the clock time since the epoch, under Raspbian on an RPI it returns the time since the last boot. (A similar issue was described here: https://stackoverflow.com/questions/45841230/alsa-retrieving-audio-buffer-t…). The problem shows if we run a slightly modified version of recordtest.py using the latest pyalsaaudio versions. ``` #!/usr/bin/env python3 # -*- mode: python; indent-tabs-mode: t; c-basic-offset: 4; tab-width: 4 -*- ## recordtest.py ## ## This is an example of a simple sound capture script. ## ## The script opens an ALSA pcm device for sound capture, sets ## various attributes of the capture, and reads in a loop, ## writing the data to standard out. ## ## To test it out do the following: ## python recordtest.py out.raw # talk to the microphone ## aplay -r 8000 -f S16_LE -c 1 out.raw #!/usr/bin/env python from __future__ import print_function import sys import time import getopt import alsaaudio def usage(): print('usage: recordtest.py [-d <device>] <file>', file=sys.stderr) sys.exit(2) if __name__ == '__main__': device = 'default' opts, args = getopt.getopt(sys.argv[1:], 'd:') for o, a in opts: if o == '-d': device = a if not args: usage() f = open(args[0], 'wb') # Open the device in nonblocking capture mode in mono, with a sampling rate of 44100 Hz # and 16 bit little endian samples # The period size controls the internal number of frames per period. # The significance of this parameter is documented in the ALSA api. # For our purposes, it is suficcient to know that reads from the device # will return this many frames. Each frame being 2 bytes long. # This means that the reads below will return either 320 bytes of data # or 0 bytes of data. The latter is possible because we are in nonblocking # mode. inp = alsaaudio.PCM(alsaaudio.PCM_CAPTURE, alsaaudio.PCM_NONBLOCK, channels=1, rate=44100, format=alsaaudio.PCM_FORMAT_S16_LE, periodsize=160, device=device) print(inp.info()) # help(inp.htimestamp) loops = 100000 while loops > 0: loops -= 1 # Read data from device l, data = inp.read() if l: print(l, inp.htimestamp(), time.time()) f.write(data) time.sleep(.001) ``` When executed on an RPi it yields: ``` {'name': 'plugsnoop:1,0', 'state': 'PREPARED', 'access_type': 'RW_INTERLEAVED', ' (call value) type': 1, ' (call value) type_name': 'CAPTURE', ' (call value) mode': 1, ' (call value) mode_name': 'PCM_NONBLOCK', 'format': 2, 'format_name': 'S16_LE', 'format_description': 'Signed 16 bit Little Endian', 'subformat_name': 'STD', 'subformat_description': 'Standard', 'channels': 1, 'rate': 44100, 'period_time': 21333, 'period_size': 940, 'buffer_time': 0, 'buffer_size': 4704, 'get_periods': 0, 'rate_numden': (44100, 1), 'significant_bits': 16, 'is_batch': True, 'is_block_transfer': True, 'is_double': False, 'is_half_duplex': False, 'is_joint_duplex': False, 'can_overrange': False, 'can_mmap_sample_resolution': False, 'can_pause': False, 'can_resume': False, 'can_sync_start': False} 940 (5596, 851796145, 0) 1617702646.0637105 940 (5596, 872796030, 0) 1617702646.084714 940 (5596, 893795916, 0) 1617702646.105702 940 (5596, 915796158, 0) 1617702646.1277127 940 (5596, 936795314, 0) 1617702646.1486921 940 (5596, 957795199, 0) 1617702646.1697078 940 (5596, 979795858, 0) 1617702646.191694 940 (5597, 795171, 0) 1617702646.212712 ... ``` I read that for the timestamp function to work there should be hardware support. But here we have it half working. This behavior is shown if I use the following in my .asoundrc ``` pcm.plugsnoop { type plug @args [ CARD DEV ] @args.CARD { type string } @args.DEV { type integer } slave { pcm { @func concat strings [ "dsnoop:" $CARD "," $DEV ] } } } ``` and call it with: `python recordtest.py -d plugsnoop:1,0 test.raw` Using plughw all timestamps become zero. The attached microphone is an USB microphone: GoMic by SAMSON. Is there a fix or workaround for this behavior which gives me clocktime instead of seconds since last boot? Issue URL : https://github.com/alsa-project/alsa-lib/issues/131 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

dsnoop: snd_pcm_htimestamp returns time past since last boot not clock time
by GitHub issues - edited 06 Apr '21

06 Apr '21

alsa-project/alsa-lib issue #131 was edited from RonaldAJ: While on a PC snd_pcm_htimestamp returns the clock time since the epoch, under Raspbian on an RPI it returns the time since the last boot. (A similar issue was described here: https://stackoverflow.com/questions/45841230/alsa-retrieving-audio-buffer-t…). The problem shows if we run a slightly modified version of recordtest.py using the latest pyalsaaudio versions. ``` #!/usr/bin/env python3 # -*- mode: python; indent-tabs-mode: t; c-basic-offset: 4; tab-width: 4 -*- ## recordtest.py ## ## This is an example of a simple sound capture script. ## ## The script opens an ALSA pcm device for sound capture, sets ## various attributes of the capture, and reads in a loop, ## writing the data to standard out. ## ## To test it out do the following: ## python recordtest.py out.raw # talk to the microphone ## aplay -r 8000 -f S16_LE -c 1 out.raw #!/usr/bin/env python from __future__ import print_function import sys import time import getopt import alsaaudio def usage(): print('usage: recordtest.py [-d <device>] <file>', file=sys.stderr) sys.exit(2) if __name__ == '__main__': device = 'default' opts, args = getopt.getopt(sys.argv[1:], 'd:') for o, a in opts: if o == '-d': device = a if not args: usage() f = open(args[0], 'wb') # Open the device in nonblocking capture mode in mono, with a sampling rate of 44100 Hz # and 16 bit little endian samples # The period size controls the internal number of frames per period. # The significance of this parameter is documented in the ALSA api. # For our purposes, it is suficcient to know that reads from the device # will return this many frames. Each frame being 2 bytes long. # This means that the reads below will return either 320 bytes of data # or 0 bytes of data. The latter is possible because we are in nonblocking # mode. inp = alsaaudio.PCM(alsaaudio.PCM_CAPTURE, alsaaudio.PCM_NONBLOCK, channels=1, rate=44100, format=alsaaudio.PCM_FORMAT_S16_LE, periodsize=160, device=device) print(inp.info()) # help(inp.htimestamp) loops = 100000 while loops > 0: loops -= 1 # Read data from device l, data = inp.read() if l: print(l, inp.htimestamp(), time.time()) f.write(data) time.sleep(.001) ``` When executed on an RPi it yields: ``` {'name': 'plugsnoop:1,0', 'state': 'PREPARED', 'access_type': 'RW_INTERLEAVED', ' (call value) type': 1, ' (call value) type_name': 'CAPTURE', ' (call value) mode': 1, ' (call value) mode_name': 'PCM_NONBLOCK', 'format': 2, 'format_name': 'S16_LE', 'format_description': 'Signed 16 bit Little Endian', 'subformat_name': 'STD', 'subformat_description': 'Standard', 'channels': 1, 'rate': 44100, 'period_time': 21333, 'period_size': 940, 'buffer_time': 0, 'buffer_size': 4704, 'get_periods': 0, 'rate_numden': (44100, 1), 'significant_bits': 16, 'is_batch': True, 'is_block_transfer': True, 'is_double': False, 'is_half_duplex': False, 'is_joint_duplex': False, 'can_overrange': False, 'can_mmap_sample_resolution': False, 'can_pause': False, 'can_resume': False, 'can_sync_start': False} 940 (5596, 851796145, 0) 1617702646.0637105 940 (5596, 872796030, 0) 1617702646.084714 940 (5596, 893795916, 0) 1617702646.105702 940 (5596, 915796158, 0) 1617702646.1277127 940 (5596, 936795314, 0) 1617702646.1486921 940 (5596, 957795199, 0) 1617702646.1697078 940 (5596, 979795858, 0) 1617702646.191694 940 (5597, 795171, 0) 1617702646.212712 ... ``` I read that for the timestamp function to work there should be hardware support. But here we have it half working. This behavior is shown if I use the following in my .asoundrc ``` pcm.plugsnoop { type plug @args [ CARD DEV ] @args.CARD { type string } @args.DEV { type integer } slave { pcm { @func concat strings [ "dsnoop:" $CARD "," $DEV ] } } } ``` and call it with: `python recordtest.py -d plugsnoop:1,0 test.raw` Using plughw all timestamps become zero. The attached microphone is an USB microphone: GoMic by SAMSON. Is there a fix or workaround for this behavior which gives me clocktime instead of seconds since last boot? Issue URL : https://github.com/alsa-project/alsa-lib/issues/131 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

cplay: Add suport for AAC audio codec
by GitHub pull_request - opened 06 Apr '21

06 Apr '21

alsa-project/tinycompress pull request #10 was opened from TangleZ: Add parse_aac_header API to parse AAC streams, support ADTS and ADIF format. Signed-off-by: Zhang Peng <peng.zhang_8(a)nxp.com> Request URL : https://github.com/alsa-project/tinycompress/pull/10 Patch URL : https://github.com/alsa-project/tinycompress/pull/10.patch Repository URL: https://github.com/alsa-project/tinycompress

1 0

[PATCH] ALSA: usb-audio: Drop implicit fb quirk entries dubbed for capture
by Takashi Iwai 06 Apr '21

06 Apr '21

The implicit feedback quirk table contains the entries that also appear in the capture quirks, and those are all handled to be skipped. For the code simplicity, drop the duped entries in the playback quirk table, and check the match with the capture quirk table instead. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=212519 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/implicit.c | 15 +++++---------- 1 file changed, 5 insertions(+), 10 deletions(-) diff --git a/sound/usb/implicit.c b/sound/usb/implicit.c index 11a85e66aa96..19622c58b72b 100644 --- a/sound/usb/implicit.c +++ b/sound/usb/implicit.c @@ -70,16 +70,6 @@ static const struct snd_usb_implicit_fb_match playback_implicit_fb_quirks[] = { .type = IMPLICIT_FB_FIXED, .ep_num = 0x84, .iface = 0 }, /* MOTU MicroBook II */ - /* No quirk for playback but with capture quirk (see below) */ - IMPLICIT_FB_SKIP_DEV(0x0582, 0x0130), /* BOSS BR-80 */ - IMPLICIT_FB_SKIP_DEV(0x0582, 0x0171), /* BOSS RC-505 */ - IMPLICIT_FB_SKIP_DEV(0x0582, 0x0185), /* BOSS GP-10 */ - IMPLICIT_FB_SKIP_DEV(0x0582, 0x0189), /* BOSS GT-100v2 */ - IMPLICIT_FB_SKIP_DEV(0x0582, 0x01d6), /* BOSS GT-1 */ - IMPLICIT_FB_SKIP_DEV(0x0582, 0x01d8), /* BOSS Katana */ - IMPLICIT_FB_SKIP_DEV(0x0582, 0x01e5), /* BOSS GT-001 */ - IMPLICIT_FB_SKIP_DEV(0x0582, 0x0203), /* BOSS AD-10 */ - {} /* terminator */ }; @@ -278,6 +268,11 @@ static int audioformat_implicit_fb_quirk(struct snd_usb_audio *chip, } } + /* Don't apply playback quirks for the devices with capture quirk */ + p = find_implicit_fb_entry(chip, capture_implicit_fb_quirks, alts); + if (p && p->type == IMPLICIT_FB_FIXED) + return 0; /* no quirk */ + /* Generic UAC2 implicit feedback */ if (attr == USB_ENDPOINT_SYNC_ASYNC && alts->desc.bInterfaceClass == USB_CLASS_AUDIO && -- 2.26.2

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[PATCH] ALSA: usb-audio: Add error checks for usb_driver_claim_interface() calls
by Takashi Iwai 06 Apr '21

06 Apr '21

There are a few calls of usb_driver_claim_interface() but all of those miss the proper error checks, as reported by Coverity. This patch adds those missing checks. Along with it, replace the magic pointer with -1 with a constant USB_AUDIO_IFACE_UNUSED for better readability. Reported-by: coverity-bot <keescook+coverity-bot(a)chromium.org> Addresses-Coverity-ID: 1475943 ("Error handling issues") Addresses-Coverity-ID: 1475944 ("Error handling issues") Addresses-Coverity-ID: 1475945 ("Error handling issues") Fixes: b1ce7ba619d9 ("ALSA: usb-audio: claim autodetected PCM interfaces all at once") Fixes: e5779998bf8b ("ALSA: usb-audio: refactor code") Link: https://lore.kernel.org/r/202104051059.FB7F3016@keescook Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/card.c | 14 +++++++------- sound/usb/quirks.c | 16 ++++++++++++---- sound/usb/usbaudio.h | 2 ++ 3 files changed, 21 insertions(+), 11 deletions(-) diff --git a/sound/usb/card.c b/sound/usb/card.c index 0826a437f8fc..7b7526d3a56e 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -181,9 +181,8 @@ static int snd_usb_create_stream(struct snd_usb_audio *chip, int ctrlif, int int ctrlif, interface); return -EINVAL; } - usb_driver_claim_interface(&usb_audio_driver, iface, (void *)-1L); - - return 0; + return usb_driver_claim_interface(&usb_audio_driver, iface, + USB_AUDIO_IFACE_UNUSED); } if ((altsd->bInterfaceClass != USB_CLASS_AUDIO && @@ -203,7 +202,8 @@ static int snd_usb_create_stream(struct snd_usb_audio *chip, int ctrlif, int int if (! snd_usb_parse_audio_interface(chip, interface)) { usb_set_interface(dev, interface, 0); /* reset the current interface */ - usb_driver_claim_interface(&usb_audio_driver, iface, (void *)-1L); + return usb_driver_claim_interface(&usb_audio_driver, iface, + USB_AUDIO_IFACE_UNUSED); } return 0; @@ -862,7 +862,7 @@ static void usb_audio_disconnect(struct usb_interface *intf) struct snd_card *card; struct list_head *p; - if (chip == (void *)-1L) + if (chip == USB_AUDIO_IFACE_UNUSED) return; card = chip->card; @@ -992,7 +992,7 @@ static int usb_audio_suspend(struct usb_interface *intf, pm_message_t message) struct usb_mixer_interface *mixer; struct list_head *p; - if (chip == (void *)-1L) + if (chip == USB_AUDIO_IFACE_UNUSED) return 0; if (!chip->num_suspended_intf++) { @@ -1022,7 +1022,7 @@ static int __usb_audio_resume(struct usb_interface *intf, bool reset_resume) struct list_head *p; int err = 0; - if (chip == (void *)-1L) + if (chip == USB_AUDIO_IFACE_UNUSED) return 0; atomic_inc(&chip->active); /* avoid autopm */ diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index ae667cf77e61..63f27ce64bcb 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -55,8 +55,12 @@ static int create_composite_quirk(struct snd_usb_audio *chip, if (!iface) continue; if (quirk->ifnum != probed_ifnum && - !usb_interface_claimed(iface)) - usb_driver_claim_interface(driver, iface, (void *)-1L); + !usb_interface_claimed(iface)) { + err = usb_driver_claim_interface(driver, iface, + USB_AUDIO_IFACE_UNUSED); + if (err < 0) + return err; + } } return 0; @@ -426,8 +430,12 @@ static int create_autodetect_quirks(struct snd_usb_audio *chip, continue; err = create_autodetect_quirk(chip, iface, driver); - if (err >= 0) - usb_driver_claim_interface(driver, iface, (void *)-1L); + if (err >= 0) { + err = usb_driver_claim_interface(driver, iface, + USB_AUDIO_IFACE_UNUSED); + if (err < 0) + return err; + } } return 0; diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index 60b9dd7df6bb..8794c8658ab9 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -61,6 +61,8 @@ struct snd_usb_audio { struct media_intf_devnode *ctl_intf_media_devnode; }; +#define USB_AUDIO_IFACE_UNUSED ((void *)-1L) + #define usb_audio_err(chip, fmt, args...) \ dev_err(&(chip)->dev->dev, fmt, ##args) #define usb_audio_warn(chip, fmt, args...) \ -- 2.26.2

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snd_pcm_htimestamp returns time past since last boot not clock time
by GitHub issues - opened 06 Apr '21

06 Apr '21

alsa-project/alsa-lib issue #131 was opened from RonaldAJ: While on a PC snd_pcm_htimestamp returns the clock time since the epoch, under Raspbian on an RPI it returns the time since the last boot. (A similar issue was described here: https://stackoverflow.com/questions/45841230/alsa-retrieving-audio-buffer-t…). The problem shows if we run a slightly modified version of recordtest.py using the latest pyalsaaudio versions. ``` #!/usr/bin/env python3 # -*- mode: python; indent-tabs-mode: t; c-basic-offset: 4; tab-width: 4 -*- ## recordtest.py ## ## This is an example of a simple sound capture script. ## ## The script opens an ALSA pcm device for sound capture, sets ## various attributes of the capture, and reads in a loop, ## writing the data to standard out. ## ## To test it out do the following: ## python recordtest.py out.raw # talk to the microphone ## aplay -r 8000 -f S16_LE -c 1 out.raw #!/usr/bin/env python from __future__ import print_function import sys import time import getopt import alsaaudio def usage(): print('usage: recordtest.py [-d <device>] <file>', file=sys.stderr) sys.exit(2) if __name__ == '__main__': device = 'default' opts, args = getopt.getopt(sys.argv[1:], 'd:') for o, a in opts: if o == '-d': device = a if not args: usage() f = open(args[0], 'wb') # Open the device in nonblocking capture mode in mono, with a sampling rate of 44100 Hz # and 16 bit little endian samples # The period size controls the internal number of frames per period. # The significance of this parameter is documented in the ALSA api. # For our purposes, it is suficcient to know that reads from the device # will return this many frames. Each frame being 2 bytes long. # This means that the reads below will return either 320 bytes of data # or 0 bytes of data. The latter is possible because we are in nonblocking # mode. inp = alsaaudio.PCM(alsaaudio.PCM_CAPTURE, alsaaudio.PCM_NONBLOCK, channels=1, rate=44100, format=alsaaudio.PCM_FORMAT_S16_LE, periodsize=160, device=device) print(inp.info()) # help(inp.htimestamp) loops = 100000 while loops > 0: loops -= 1 # Read data from device l, data = inp.read() if l: print(l, inp.htimestamp(), time.time()) f.write(data) time.sleep(.001) ``` When executed on an RPi it yields: ``` {'name': 'plugsnoop:1,0', 'state': 'PREPARED', 'access_type': 'RW_INTERLEAVED', ' (call value) type': 1, ' (call value) type_name': 'CAPTURE', ' (call value) mode': 1, ' (call value) mode_name': 'PCM_NONBLOCK', 'format': 2, 'format_name': 'S16_LE', 'format_description': 'Signed 16 bit Little Endian', 'subformat_name': 'STD', 'subformat_description': 'Standard', 'channels': 1, 'rate': 44100, 'period_time': 21333, 'period_size': 940, 'buffer_time': 0, 'buffer_size': 4704, 'get_periods': 0, 'rate_numden': (44100, 1), 'significant_bits': 16, 'is_batch': True, 'is_block_transfer': True, 'is_double': False, 'is_half_duplex': False, 'is_joint_duplex': False, 'can_overrange': False, 'can_mmap_sample_resolution': False, 'can_pause': False, 'can_resume': False, 'can_sync_start': False} 940 (5596, 851796145, 0) 1617702646.0637105 940 (5596, 872796030, 0) 1617702646.084714 940 (5596, 893795916, 0) 1617702646.105702 940 (5596, 915796158, 0) 1617702646.1277127 940 (5596, 936795314, 0) 1617702646.1486921 940 (5596, 957795199, 0) 1617702646.1697078 940 (5596, 979795858, 0) 1617702646.191694 940 (5597, 795171, 0) 1617702646.212712 ... ``` I read that for the timestamp function to work there should be hardware support. But here we have it half working. This behavior is shown if I use the following in my .asoundrc ``` pcm.plugsnoop { type plug @args [ CARD DEV ] @args.CARD { type string } @args.DEV { type integer } slave { pcm { @func concat strings [ "dsnoop:" $CARD "," $DEV ] } } } ``` and call it with: `python recordtest.py -d plugsnoop:1,0 test.raw` Using plughw all timestamps become zero. The attached microphone is an USB microphone: GoMic by SAMSON. Is there a fix or workaround for this behavior which gives me clocktime instead of seconds since last boot? Issue URL : https://github.com/alsa-project/alsa-lib/issues/131 Repository URL: https://github.com/alsa-project/alsa-lib

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RE: [PATCH v6 2/2] ASoC: rt715:add micmute led state control supportspobrn@protonmail.com
by Yuan, Perry 06 Apr '21

06 Apr '21

Hi Jaroslav: > -----Original Message----- > From: Yuan, Perry <Perry_Yuan(a)Dell.com> > Sent: Sunday, April 4, 2021 4:32 PM > To: pobrn(a)protonmail.com; pierre-louis.bossart(a)linux.intel.com; > oder_chiou(a)realtek.com; perex(a)perex.cz; tiwai(a)suse.com; > hdegoede(a)redhat.com; mgross(a)linux.intel.com; Limonciello, Mario > Cc: lgirdwood(a)gmail.com; broonie(a)kernel.org; alsa-devel(a)alsa-project.org; > linux-kernel(a)vger.kernel.org; platform-driver-x86(a)vger.kernel.org; Yuan, > Perry; mario.limonciello(a)outlook.com; Dell Client Kernel > Subject: [PATCH v6 2/2] ASoC: rt715:add micmute led state control supports > > From: Perry Yuan <perry_yuan(a)dell.com> > > Some new Dell system is going to support audio internal micphone privacy > setting from hardware level with micmute led state changing When micmute > hotkey pressed by user, soft mute will need to be enabled firstly in case of > pop noise, and codec driver need to react to mic mute event to > EC(embedded controller) notifying that SW mute is completed Then EC will > do the hardware mute physically within the timeout reached > > This patch allow codec rt715 and rt715 sdca driver to change the local > micmute led state. Dell privacy led trigger driver will ack EC when micmute > key pressed through this micphone led control interface like hda_generic > provided ACPI method defined in dell-privacy micmute led trigger will be > called for notifying the EC that software mute has been completed, then > hardware audio circuit solution controlled by EC will switch the audio input > source off/on > > Signed-off-by: Perry Yuan <perry_yuan(a)dell.com> > > -------- > v5 -> v6: > * addresed review comments from Jaroslav > * add quirks for micmute led control as short term solution to control > micmute led state change > v4 -> v5: > * rebase to latest 5.12 rc4 upstream kernel > v3 -> v4: > * remove unused debug log > * remove compile flag of DELL privacy > * move the micmute_led to local from rt715_priv > * when Jaroslav upstream his gerneric LED trigger driver,I will rebase > this patch,please consider merge this at first > https://lore.kernel.org/alsa-devel/20210211111400.1131020-1- > perex(a)perex.cz/ > v2 -> v3: > * simplify the patch to reuse some val value > * add more detail to the commit info > v1 -> v2: > * fix some format issue > -------- > --- > sound/soc/codecs/rt715-sdca.c | 41 > ++++++++++++++++++++++++++++++++++- > sound/soc/codecs/rt715.c | 41 > +++++++++++++++++++++++++++++++++++ > 2 files changed, 81 insertions(+), 1 deletion(-) > > diff --git a/sound/soc/codecs/rt715-sdca.c b/sound/soc/codecs/rt715-sdca.c > index 20528afbdc57..7bd7ad0ba7d7 100644 > --- a/sound/soc/codecs/rt715-sdca.c > +++ b/sound/soc/codecs/rt715-sdca.c > @@ -11,12 +11,14 @@ > #include <linux/moduleparam.h> > #include <linux/kernel.h> > #include <linux/init.h> > +#include <linux/leds.h> > #include <linux/pm_runtime.h> > #include <linux/pm.h> > #include <linux/soundwire/sdw.h> > #include <linux/regmap.h> > #include <linux/slab.h> > #include <linux/platform_device.h> > +#include <linux/dmi.h> > #include <sound/core.h> > #include <sound/pcm.h> > #include <sound/pcm_params.h> > @@ -344,6 +346,34 @@ static int rt715_sdca_get_volsw(struct snd_kcontrol > *kcontrol, > return 0; > } > > +static bool micmute_led_set; > +static int dmi_matched(const struct dmi_system_id *dmi) { > + micmute_led_set = 1; > + return 1; > +} > + > +/* Some systems will need to use this to trigger mic mute LED state > +changed */ static const struct dmi_system_id micmute_led_dmi_table[] = { > + { > + .callback = dmi_matched, > + .ident = "Dell Latitude 9420", > + .matches = { > + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc."), > + DMI_MATCH(DMI_PRODUCT_NAME, "Latitude > 9420"), > + }, > + }, > + { > + .callback = dmi_matched, > + .ident = "Dell Latitude 9520", > + .matches = { > + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc."), > + DMI_MATCH(DMI_PRODUCT_NAME, "Latitude > 9520"), > + }, > + }, > + {}, > +}; > + > static int rt715_sdca_put_volsw(struct snd_kcontrol *kcontrol, > struct snd_ctl_elem_value *ucontrol) > { > @@ -358,6 +388,7 @@ static int rt715_sdca_put_volsw(struct snd_kcontrol > *kcontrol, > unsigned int mask = (1 << fls(max)) - 1; > unsigned int invert = p->invert; > int err; > + bool micmute_led; > > for (i = 0; i < 4; i++) { > if (ucontrol->value.integer.value[i] != rt715- > >kctl_switch_orig[i]) { @@ -393,7 +424,15 @@ static int > rt715_sdca_put_volsw(struct snd_kcontrol *kcontrol, > if (err < 0) > return err; > } > - > + /* Micmute LED state changed by muted/unmute switch */ > + dmi_check_system(micmute_led_dmi_table); > + if (invert && micmute_led_set) { > + if (ucontrol->value.integer.value[0] || ucontrol- > >value.integer.value[1]) > + micmute_led = LED_OFF; > + else > + micmute_led = LED_ON; > + ledtrig_audio_set(LED_AUDIO_MICMUTE, micmute_led); > + } > return k_changed; > } > > diff --git a/sound/soc/codecs/rt715.c b/sound/soc/codecs/rt715.c index > 34c3357e943b..490cf9e63c2d 100644 > --- a/sound/soc/codecs/rt715.c > +++ b/sound/soc/codecs/rt715.c > @@ -13,6 +13,7 @@ > #include <linux/init.h> > #include <linux/delay.h> > #include <linux/i2c.h> > +#include <linux/leds.h> > #include <linux/pm_runtime.h> > #include <linux/pm.h> > #include <linux/soundwire/sdw.h> > @@ -25,6 +26,7 @@ > #include <linux/of.h> > #include <linux/of_gpio.h> > #include <linux/of_device.h> > +#include <linux/dmi.h> > #include <sound/core.h> > #include <sound/pcm.h> > #include <sound/pcm_params.h> > @@ -70,6 +72,34 @@ static void rt715_get_gain(struct rt715_priv *rt715, > unsigned int addr_h, > pr_err("Failed to get L channel gain.\n"); } > > +static bool micmute_led_set; > +static int dmi_matched(const struct dmi_system_id *dmi) { > + micmute_led_set = 1; > + return 1; > +} > + > +/* Some systems will need to use this to trigger mic mute LED state > +changed */ static const struct dmi_system_id micmute_led_dmi_table[] = { > + { > + .callback = dmi_matched, > + .ident = "Dell Latitude 9420", > + .matches = { > + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc."), > + DMI_MATCH(DMI_PRODUCT_NAME, "Latitude > 9420"), > + }, > + }, > + { > + .callback = dmi_matched, > + .ident = "Dell Latitude 9520", > + .matches = { > + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc."), > + DMI_MATCH(DMI_PRODUCT_NAME, "Latitude > 9520"), > + }, > + }, > + {}, > +}; > + > /* For Verb-Set Amplifier Gain (Verb ID = 3h) */ static int > rt715_set_amp_gain_put(struct snd_kcontrol *kcontrol, > struct snd_ctl_elem_value *ucontrol) > @@ -88,6 +118,7 @@ static int rt715_set_amp_gain_put(struct snd_kcontrol > *kcontrol, > RT715_SET_GAIN_MIX_ADC2_L}; > unsigned int addr_h, addr_l, val_h, val_ll, val_lr; > unsigned int read_ll, read_rl, i, j, loop_cnt; > + bool micmute_led; > > if (strstr(ucontrol->id.name, "Main Capture Switch") || > strstr(ucontrol->id.name, "Main Capture Volume")) @@ - > 185,6 +216,16 @@ static int rt715_set_amp_gain_put(struct snd_kcontrol > *kcontrol, > if (dapm->bias_level <= SND_SOC_BIAS_STANDBY) > regmap_write(rt715->regmap, > RT715_SET_AUDIO_POWER_STATE, > AC_PWRST_D3); > + > + /* Micmute LED state changed by muted/unmute switch */ > + dmi_check_system(micmute_led_dmi_table); > + if (mc->invert && micmute_led_set) { > + if (ucontrol->value.integer.value[0] || ucontrol- > >value.integer.value[1]) > + micmute_led = LED_OFF; > + else > + micmute_led = LED_ON; > + ledtrig_audio_set(LED_AUDIO_MICMUTE, micmute_led); > + } > return 0; > } > > -- > 2.25.1 Could you help to review if the patch is acceptable as short term solution? I added dmi checking when changing the led state for rt715 and rt715-sdca codec. Thank you! Perry

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