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

[PATCH] ASoC: amd: Add support for ALC1015P codec in acp3x machine driver
by Vijendar Mukunda 06 Apr '21

06 Apr '21

Add ALC1015p codec support for acp3x machine driver. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- sound/soc/amd/Kconfig | 1 + sound/soc/amd/acp3x-rt5682-max9836.c | 46 +++++++++++++++++++++++++++++++++++- 2 files changed, 46 insertions(+), 1 deletion(-) diff --git a/sound/soc/amd/Kconfig b/sound/soc/amd/Kconfig index 256c192..ba5a85b 100644 --- a/sound/soc/amd/Kconfig +++ b/sound/soc/amd/Kconfig @@ -35,6 +35,7 @@ config SND_SOC_AMD_RV_RT5682_MACH select SND_SOC_CROS_EC_CODEC select I2C_CROS_EC_TUNNEL select SND_SOC_RT1015 + select SND_SOC_RT1015P depends on SND_SOC_AMD_ACP3x && I2C && CROS_EC help This option enables machine driver for RT5682 and MAX9835. diff --git a/sound/soc/amd/acp3x-rt5682-max9836.c b/sound/soc/amd/acp3x-rt5682-max9836.c index cea320a..8e11bb8 100644 --- a/sound/soc/amd/acp3x-rt5682-max9836.c +++ b/sound/soc/amd/acp3x-rt5682-max9836.c @@ -275,6 +275,8 @@ SND_SOC_DAILINK_DEF(rt5682, DAILINK_COMP_ARRAY(COMP_CODEC("i2c-10EC5682:00", "rt5682-aif1"))); SND_SOC_DAILINK_DEF(max, DAILINK_COMP_ARRAY(COMP_CODEC("MX98357A:00", "HiFi"))); +SND_SOC_DAILINK_DEF(rt1015p, + DAILINK_COMP_ARRAY(COMP_CODEC("RTL1015:00", "HiFi"))); SND_SOC_DAILINK_DEF(rt1015, DAILINK_COMP_ARRAY(COMP_CODEC("i2c-10EC1015:00", "rt1015-aif"), COMP_CODEC("i2c-10EC1015:01", "rt1015-aif"))); @@ -419,6 +421,43 @@ static struct snd_soc_card acp3x_1015 = { .num_controls = ARRAY_SIZE(acp3x_mc_1015_controls), }; +static const struct snd_soc_dapm_widget acp3x_1015p_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MUX("Dmic Mux", SND_SOC_NOPM, 0, 0, + &acp3x_dmic_mux_control), + SND_SOC_DAPM_SPK("Speakers", NULL), +}; + +static const struct snd_soc_dapm_route acp3x_1015p_route[] = { + {"Headphone Jack", NULL, "HPOL"}, + {"Headphone Jack", NULL, "HPOR"}, + {"IN1P", NULL, "Headset Mic"}, + {"Dmic Mux", "Front Mic", "DMIC"}, + {"Dmic Mux", "Rear Mic", "DMIC"}, + /* speaker */ + { "Speakers", NULL, "Speaker" }, +}; + +static const struct snd_kcontrol_new acp3x_mc_1015p_controls[] = { + SOC_DAPM_PIN_SWITCH("Speakers"), + SOC_DAPM_PIN_SWITCH("Headphone Jack"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), +}; + +static struct snd_soc_card acp3x_1015p = { + .name = "acp3xalc56821015p", + .owner = THIS_MODULE, + .dai_link = acp3x_dai, + .num_links = ARRAY_SIZE(acp3x_dai), + .dapm_widgets = acp3x_1015p_widgets, + .num_dapm_widgets = ARRAY_SIZE(acp3x_1015p_widgets), + .dapm_routes = acp3x_1015p_route, + .num_dapm_routes = ARRAY_SIZE(acp3x_1015p_route), + .controls = acp3x_mc_1015p_controls, + .num_controls = ARRAY_SIZE(acp3x_mc_1015p_controls), +}; + void *soc_is_rltk_max(struct device *dev) { const struct acpi_device_id *match; @@ -435,6 +474,9 @@ static void card_spk_dai_link_present(struct snd_soc_dai_link *links, if (!strcmp(card_name, "acp3xalc56821015")) { links[1].codecs = rt1015; links[1].num_codecs = ARRAY_SIZE(rt1015); + } else if (!strcmp(card_name, "acp3xalc56821015p")) { + links[1].codecs = rt1015p; + links[1].num_codecs = ARRAY_SIZE(rt1015p); } else { links[1].codecs = max; links[1].num_codecs = ARRAY_SIZE(max); @@ -486,6 +528,7 @@ static int acp3x_probe(struct platform_device *pdev) static const struct acpi_device_id acp3x_audio_acpi_match[] = { { "AMDI5682", (unsigned long)&acp3x_5682}, { "AMDI1015", (unsigned long)&acp3x_1015}, + { "AMDP1015", (unsigned long)&acp3x_1015p}, {}, }; MODULE_DEVICE_TABLE(acpi, acp3x_audio_acpi_match); @@ -503,5 +546,6 @@ module_platform_driver(acp3x_audio); MODULE_AUTHOR("akshu.agrawal(a)amd.com"); MODULE_AUTHOR("Vishnuvardhanrao.Ravulapati(a)amd.com"); -MODULE_DESCRIPTION("ALC5682 ALC1015 & MAX98357 audio support"); +MODULE_AUTHOR("Vijendar.Mukunda(a)amd.com"); +MODULE_DESCRIPTION("ALC5682 ALC1015, ALC1015P & MAX98357 audio support"); MODULE_LICENSE("GPL v2"); -- 2.7.4

4 7

[PATCH v2 0/2] ASoC: tlv320aic32x4: Bugfixes for tlv320aic32x4 driver
by Annaliese McDermond 06 Apr '21

06 Apr '21

These are a couple of bugfixes for the tlv320aic32x4 driver that we discovered after the Raspberry Pi Foundation's move to the 5.10 kernel. The first increases the maximum value in the regmap configuration so that the powerup command actually works. The second changes the ordering of the clock registration in the module probe function so that the clocks are properly registered before the component probe function runs. Changes in v2: - Corrected commit hashes in Fixes lines to be proper length - Including inadvertently omitted maintainers on To list. Annaliese McDermond (2): ASoC: tlv320aic32x4: Increase maximum register in regmap ASoC: tlv320aic32x4: Register clocks before registering component sound/soc/codecs/tlv320aic32x4.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) -- 2.27.0

2 1

[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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