February 2023 - Alsa-devel - mailman.alsa-project.org

sample_size field in snd_dec_flac structure
by Carlos Rafael Giani 06 Feb '23

06 Feb '23

In snd_dec_flac (see: https://github.com/torvalds/linux/blob/master/include/uapi/sound/compress_p… ), there is the sample_size field. It seems to me that this is meant to contain the same value as the sample size as the one in the frame header: https://xiph.org/flac/format.html#frame_header Does this mean that if I always send a full FLAC stream into the device, with its complete STREAMINFO header, I can just always set sample_size to 0? The spec does mention that 0 mean "get [the sample size] from STREAMINFO metadata block".

1 0

[PATCH] ASoC: codecs: Fix unsigned comparison with less than zero
by Jiapeng Chong 06 Feb '23

06 Feb '23

The val is defined as unsigned int type, if(val<0) is redundant, so delete it. sound/soc/codecs/idt821034.c:449 idt821034_kctrl_gain_put() warn: unsigned 'val' is never less than zero. Reported-by: Abaci Robot <abaci(a)linux.alibaba.com> Link: https://bugzilla.openanolis.cn/show_bug.cgi?id=3947 Signed-off-by: Jiapeng Chong <jiapeng.chong(a)linux.alibaba.com> --- sound/soc/codecs/idt821034.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/codecs/idt821034.c b/sound/soc/codecs/idt821034.c index 5d01787b1c1f..2cc7b9166e69 100644 --- a/sound/soc/codecs/idt821034.c +++ b/sound/soc/codecs/idt821034.c @@ -446,8 +446,6 @@ static int idt821034_kctrl_gain_put(struct snd_kcontrol *kcontrol, u8 ch; val = ucontrol->value.integer.value[0]; - if (val < 0) - return -EINVAL; if (val > max - min) return -EINVAL; -- 2.20.1.7.g153144c

3 2

[PATCH] ASoC: codecs: aw88395: initialize cur_scene_id to 0
by Tom Rix 06 Feb '23

06 Feb '23

cppcheck reports sound/soc/codecs/aw88395/aw88395_lib.c:789:6: error: Uninitialized variable: cur_scene_id [uninitvar] if (cur_scene_id == 0) { ^ Passing a garbage value to aw_dev_parse_data_by_sec_type_v1() will cause a crash when the value is used as an array index. This check assumes cur_scene_id is initialized to 0, so initialize it to 0. Fixes: 4345865b003b ("ASoC: codecs: ACF bin parsing and check library file for aw88395") Signed-off-by: Tom Rix <trix(a)redhat.com> --- sound/soc/codecs/aw88395/aw88395_lib.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/aw88395/aw88395_lib.c b/sound/soc/codecs/aw88395/aw88395_lib.c index 64dde972f3f0..d7c31a202adc 100644 --- a/sound/soc/codecs/aw88395/aw88395_lib.c +++ b/sound/soc/codecs/aw88395/aw88395_lib.c @@ -769,7 +769,7 @@ static int aw_dev_parse_dev_type_v1(struct aw_device *aw_dev, { struct aw_cfg_dde_v1 *cfg_dde = (struct aw_cfg_dde_v1 *)((char *)prof_hdr + prof_hdr->hdr_offset); - int cur_scene_id; + int cur_scene_id = 0; unsigned int i; int ret; -- 2.26.3

2 1

[PATCH] ASoC: codecs: aw88395: Uninitialized variable bug in aw_dev_parse_dev_type_v1()
by Dan Carpenter 06 Feb '23

06 Feb '23

The "cur_scene_id" variable is never initialized. It needs to be set to zero for the code to work. Fixes: 4345865b003b ("ASoC: codecs: ACF bin parsing and check library file for aw88395") Signed-off-by: Dan Carpenter <error27(a)gmail.com> --- Presumably this code was tested with CONFIG_CC_HAS_AUTO_VAR_INIT_ZERO=y or whatever to automatically initialize it to zero or it would have died specatularly. ;) sound/soc/codecs/aw88395/aw88395_lib.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/aw88395/aw88395_lib.c b/sound/soc/codecs/aw88395/aw88395_lib.c index 64dde972f3f0..d7c31a202adc 100644 --- a/sound/soc/codecs/aw88395/aw88395_lib.c +++ b/sound/soc/codecs/aw88395/aw88395_lib.c @@ -769,7 +769,7 @@ static int aw_dev_parse_dev_type_v1(struct aw_device *aw_dev, { struct aw_cfg_dde_v1 *cfg_dde = (struct aw_cfg_dde_v1 *)((char *)prof_hdr + prof_hdr->hdr_offset); - int cur_scene_id; + int cur_scene_id = 0; unsigned int i; int ret; -- 2.35.1

2 1

[PATCH v5] ASoC: dt-bindings: Add tas2781 amplifier
by Shenghao Ding 05 Feb '23

05 Feb '23

Create tas2781.yaml for tas2781 driver. Signed-off-by: Shenghao Ding <13916275206(a)139.com> --- Changes in v5: - Drop the full stop in the Subject. - Drop the reset-gpios reference to gpio.txt. - Add Changelog. Changes to be committed: new file: Documentation/devicetree/bindings/sound/ti,tas2781.yaml --- .../devicetree/bindings/sound/ti,tas2781.yaml | 87 +++++++++++++++++++ 1 file changed, 87 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ti,tas2781.yaml diff --git a/Documentation/devicetree/bindings/sound/ti,tas2781.yaml b/Documentation/devicetree/bindings/sound/ti,tas2781.yaml new file mode 100644 index 000000000000..8af44792a904 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,tas2781.yaml @@ -0,0 +1,87 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +# Copyright (C) 2022 - 2023 Texas Instruments Incorporated +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ti,tas2781.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Texas Instruments TAS2781 SmartAMP + +maintainers: + - Shenghao Ding <shenghao-ding(a)ti.com> + - Kevin Lu <kevin-lu(a)ti.com> + +description: | + The TAS2781 is a mono, digital input Class-D audio amplifier + optimized for efficiently driving high peak power into small + loudspeakers. Integrated an on-chip DSP supports Texas Instruments + Smart Amp speaker protection algorithm. The integrated speaker + voltage and current sense provides for real time + monitoring of loudspeaker behavior. + +properties: + compatible: + enum: + - ti,tas2781 + + reg: + maxItems: 1 + description: | + I2C address of the device can be in range from 0x38 to 0x40. + + reset-gpios: + maxItems: 1 + description: | + A GPIO line handling reset of the chip. As the line is active high, + it should be marked GPIO_ACTIVE_HIGH. + + interrupts: + maxItems: 1 + + ti,audio-slots: + $ref: /schemas/types.yaml#/definitions/uint32-array + minItems: 1 + maxItems: 4 + description: | + I2c address of the device for different audio slots, + useless in mono case. + + ti,broadcast-addr: + $ref: /schemas/types.yaml#/definitions/uint32 + description: | + Generic i2c address for all the tas2781 devices in + purpose of I2C broadcast during the multi-device + writes, useless in mono case. + + '#sound-dai-cells': + const: 1 + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + i2c { + /* example with quad support, such as tablet or pad device */ + #address-cells = <1>; + #size-cells = <0>; + quad: codec@38 { + compatible = "ti,tas2781"; + reg = <0x38>; + #sound-dai-cells = <1>; + reset-gpios = < &gpio1 10 GPIO_ACTIVE_HIGH >; + interrupt-parent = <&gpio1>; + interrupts = <15>; + ti,audio-slots = < 0x38 /* topleft-channel */ + 0x39 /* topright-channel */ + 0x3a /* bottomleft-channel */ + 0x3b /* bottomright-channel */ + >; + ti,broadcast-addr = <0x40>; + }; + }; +... -- 2.34.1

4 3

Make ALC4080 front panel audio work on ASUS ROG STRIX X670E-E GAMING WIFI
by Jesper Juhl 04 Feb '23

04 Feb '23

Hi In order to have working front panel audio for my computer based on a ASUS ROG STRIX X670E-E GAMING WIFI motherboard with ALC4080 audio, I need to apply the following patch. It would be great if it could be applied up-stream so I (and other users with the same hardware) don't have to do this manually in the future. Kind regards, Jesper Juhl --- /tmp/USB-Audio.conf.original 2023-02-04 12:50:27.691689435 +0100 +++ /usr/share/alsa/ucm2/USB-Audio/USB-Audio.conf 2023-02-04 12:53:43.307961781 +0100 @@ -40,6 +40,7 @@ # 0b05:1996 ASUS on multiple boards (including ASUS ROG Maximus XIII) # 0b05:1a16 ASUS ROG Strix B660-F Gaming WiFi # 0b05:1a20 ASUS ROG STRIX Z690-I Gaming Wifi + # 0b05:1a52 ASUS ROG STRIX X670E-E GAMING WIFI # 0b05:1a27 ALC4082 on ASUS ROG Maximus Z690 Hero # 0db0:005a MSI MPG Z690 CARBON WIFI # 0db0:151f MSI X570S EDGE MAX WIFI @@ -50,7 +51,7 @@ # 0db0:a47c MSI MEG X570S Ace Max # 0db0:b202 MSI MAG Z690 Tomahawk Wifi # 0db0:d6e7 MSI MPG X670E Carbon Wifi - Regex "USB((0414:a00e)|(0b05:(1996|1a(16|2[07])))|(0db0:(005a|151f|1feb|419c|82c7|a073|a47c|b202|d6e7)))" + Regex "USB((0414:a00e)|(0b05:(1996|1a(52|16|2[07])))|(0db0:(005a|151f|1feb|419c|82c7|a073|a47c|b202|d6e7)))" } True.Define.ProfileName "Realtek/ALC4080" }

1 0

[PATCH] kselftest/alsa: Run PCM tests for multiple cards in parallel
by Mark Brown 04 Feb '23

04 Feb '23

With each test taking 4 seconds the runtime of pcm-test can add up. Since generally each card in the system is physically independent and will be unaffected by what's going on with other cards we can mitigate this by testing each card in parallel. Make a list of cards as we enumerate the system and then start a thread for each, then join the threads to ensure they have all finished. The threads each run the same tests we currently run for each PCM on the card before exiting. The list of PCMs is kept global since it helps with global operations like working out our planned number of tests and identifying missing PCMs and it seemed neater to check for PCMs on the right card in the card thread than make every PCM loop iterate over cards as well. We don't run per-PCM tests in parallel since in embedded systems it can be the case that resources are shared between the PCMs and operations on one PCM on a card may constrain what can be done on another PCM on the same card leading to potentially unstable results. We use a mutex to ensure that the reporting of results is serialised and we don't have issues with anything like the current test number, we could do this in the kselftest framework but it seems like this might cause problems for other tests that are doing lower level testing and building in constrained environments such as nolibc so this seems more sensible. Note that the ordering of the tests can't be guaranteed as things stand, this does not seem like a major problem since the numbering of tests often changes as test programs are changed so results parsers are expected to rely on the test name rather than the test numbers. We also now prefix the machine generated test name when printing the description of the test since this is logged before streaming starts. On my two card desktop system this reduces the overall runtime by a third. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- tools/testing/selftests/alsa/Makefile | 2 + tools/testing/selftests/alsa/pcm-test.c | 80 +++++++++++++++++++++++++++++---- 2 files changed, 74 insertions(+), 8 deletions(-) diff --git a/tools/testing/selftests/alsa/Makefile b/tools/testing/selftests/alsa/Makefile index 77fba3e498cc..901949db80ad 100644 --- a/tools/testing/selftests/alsa/Makefile +++ b/tools/testing/selftests/alsa/Makefile @@ -8,6 +8,8 @@ LDLIBS += -lasound endif CFLAGS += -L$(OUTPUT) -Wl,-rpath=./ +LDLIBS+=-lpthread + OVERRIDE_TARGETS = 1 TEST_GEN_PROGS := mixer-test pcm-test diff --git a/tools/testing/selftests/alsa/pcm-test.c b/tools/testing/selftests/alsa/pcm-test.c index 57d3f6dcb46b..58b525a4a32c 100644 --- a/tools/testing/selftests/alsa/pcm-test.c +++ b/tools/testing/selftests/alsa/pcm-test.c @@ -15,12 +15,21 @@ #include <stdbool.h> #include <errno.h> #include <assert.h> +#include <pthread.h> #include "../kselftest.h" #include "alsa-local.h" typedef struct timespec timestamp_t; +struct card_data { + int card; + pthread_t thread; + struct card_data *next; +}; + +struct card_data *card_list = NULL; + struct pcm_data { snd_pcm_t *handle; int card; @@ -36,6 +45,11 @@ struct pcm_data *pcm_list = NULL; int num_missing = 0; struct pcm_data *pcm_missing = NULL; +snd_config_t *default_pcm_config; + +/* Lock while reporting results since kselftest doesn't */ +pthread_mutex_t results_lock = PTHREAD_MUTEX_INITIALIZER; + enum test_class { TEST_CLASS_DEFAULT, TEST_CLASS_SYSTEM, @@ -141,6 +155,7 @@ static void find_pcms(void) snd_ctl_t *handle; snd_pcm_info_t *pcm_info; snd_config_t *config, *card_config, *pcm_config; + struct card_data *card_data; snd_pcm_info_alloca(&pcm_info); @@ -162,6 +177,13 @@ static void find_pcms(void) card_config = conf_by_card(card); + card_data = calloc(1, sizeof(*card_data)); + if (!card_data) + ksft_exit_fail_msg("Out of memory\n"); + card_data->card = card; + card_data->next = card_list; + card_list = card_data; + dev = -1; while (1) { if (snd_ctl_pcm_next_device(handle, &dev) < 0) @@ -246,10 +268,6 @@ static void test_pcm_time(struct pcm_data *data, enum test_class class, bool skip = true; const char *desc; - desc = conf_get_string(pcm_cfg, "description", NULL, NULL); - if (desc) - ksft_print_msg("%s\n", desc); - switch (class) { case TEST_CLASS_DEFAULT: test_class_name = "default"; @@ -262,6 +280,15 @@ static void test_pcm_time(struct pcm_data *data, enum test_class class, break; } + desc = conf_get_string(pcm_cfg, "description", NULL, NULL); + if (desc) + ksft_print_msg("%s.%s.%d.%d.%d.%s - %s\n", + test_class_name, test_name, + data->card, data->device, data->subdevice, + snd_pcm_stream_name(data->stream), + desc); + + snd_pcm_hw_params_alloca(&hw_params); snd_pcm_sw_params_alloca(&sw_params); @@ -443,6 +470,8 @@ static void test_pcm_time(struct pcm_data *data, enum test_class class, msg[0] = '\0'; pass = true; __close: + pthread_mutex_lock(&results_lock); + switch (class) { case TEST_CLASS_SYSTEM: test_class_name = "system"; @@ -471,6 +500,9 @@ static void test_pcm_time(struct pcm_data *data, enum test_class class, data->card, data->device, data->subdevice, snd_pcm_stream_name(data->stream), msg[0] ? " " : "", msg); + + pthread_mutex_unlock(&results_lock); + free(samples); if (handle) snd_pcm_close(handle); @@ -502,11 +534,30 @@ void run_time_tests(struct pcm_data *pcm, enum test_class class, } } +void *card_thread(void *data) +{ + struct card_data *card = data; + struct pcm_data *pcm; + + for (pcm = pcm_list; pcm != NULL; pcm = pcm->next) { + if (pcm->card != card->card) + continue; + + run_time_tests(pcm, TEST_CLASS_DEFAULT, default_pcm_config); + run_time_tests(pcm, TEST_CLASS_SYSTEM, pcm->pcm_config); + } + + return 0; +} + int main(void) { + struct card_data *card; struct pcm_data *pcm; - snd_config_t *global_config, *default_pcm_config, *cfg, *pcm_cfg; + snd_config_t *global_config, *cfg, *pcm_cfg; int num_pcm_tests = 0, num_tests, num_std_pcm_tests; + int ret; + void *thread_ret; ksft_print_header(); @@ -540,9 +591,22 @@ int main(void) snd_pcm_stream_name(pcm->stream)); } - for (pcm = pcm_list; pcm != NULL; pcm = pcm->next) { - run_time_tests(pcm, TEST_CLASS_DEFAULT, default_pcm_config); - run_time_tests(pcm, TEST_CLASS_SYSTEM, pcm->pcm_config); + for (card = card_list; card != NULL; card = card->next) { + ret = pthread_create(&card->thread, NULL, card_thread, card); + if (ret != 0) { + ksft_exit_fail_msg("Failed to create card %d thread: %d (%s)\n", + card->card, ret, + strerror(errno)); + } + } + + for (card = card_list; card != NULL; card = card->next) { + ret = pthread_join(card->thread, &thread_ret); + if (ret != 0) { + ksft_exit_fail_msg("Failed to join card %d thread: %d (%s)\n", + card->card, ret, + strerror(errno)); + } } snd_config_delete(global_config); --- base-commit: 372a0d7856be29671fc03e2f28ac27114e8c6805 change-id: 20230203-alsa-pcm-test-card-thread-17b30cc309c1 Best regards, -- Mark Brown <broonie(a)kernel.org>

2 1

[PATCH] ALSA: fireface: add field for the number of messages copied to user space
by Takashi Sakamoto 04 Feb '23

04 Feb '23

Current structure includes no field to express the number of messages copied to user space, thus user space application needs to information out of the structure to parse the content of structure. This commit adds a field to express the number of messages copied to user space since It is more preferable to use self-contained structure. Kees Cook proposed an idea of annotation for bound of flexible arrays in his future improvement for flexible-length array in kernel. The additional field for message count is suitable to the idea as well. Reference: https://people.kernel.org/kees/bounded-flexible-arrays-in-c Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- include/uapi/sound/firewire.h | 2 ++ sound/firewire/fireface/ff-protocol-former.c | 28 +++++++++++--------- 2 files changed, 18 insertions(+), 12 deletions(-) diff --git a/include/uapi/sound/firewire.h b/include/uapi/sound/firewire.h index 50917581dd2b..1e86872c151f 100644 --- a/include/uapi/sound/firewire.h +++ b/include/uapi/sound/firewire.h @@ -78,6 +78,7 @@ struct snd_firewire_event_motu_register_dsp_change { * operating hardware knob. * * @type: Fixed to SNDRV_FIREWIRE_EVENT_FF400_MESSAGE. + * @message_count: The number of messages. * @messages.message: The messages expressing hardware knob operation. * @messages.tstamp: The isochronous cycle at which the request subaction of asynchronous * transaction was sent to deliver the message. It has 16 bit unsigned integer @@ -89,6 +90,7 @@ struct snd_firewire_event_motu_register_dsp_change { */ struct snd_firewire_event_ff400_message { unsigned int type; + unsigned int message_count; struct { __u32 message; __u32 tstamp; diff --git a/sound/firewire/fireface/ff-protocol-former.c b/sound/firewire/fireface/ff-protocol-former.c index fa41de978756..efd59e9d9935 100644 --- a/sound/firewire/fireface/ff-protocol-former.c +++ b/sound/firewire/fireface/ff-protocol-former.c @@ -677,23 +677,19 @@ static void ff400_handle_msg(struct snd_ff *ff, unsigned int offset, const __le3 static long ff400_copy_msg_to_user(struct snd_ff *ff, char __user *buf, long count) { + struct snd_firewire_event_ff400_message ev = { + .type = SNDRV_FIREWIRE_EVENT_FF400_MESSAGE, + .message_count = 0, + }; struct ff400_msg_parser *parser = ff->msg_parser; - u32 type = SNDRV_FIREWIRE_EVENT_FF400_MESSAGE; long consumed = 0; - int ret = 0; + long ret = 0; - if (count < 8) + if (count < sizeof(ev) || parser->pull_pos == parser->push_pos) return 0; - spin_unlock_irq(&ff->lock); - if (copy_to_user(buf, &type, sizeof(type))) - ret = -EFAULT; - spin_lock_irq(&ff->lock); - if (ret) - return ret; - - count -= sizeof(type); - consumed += sizeof(type); + count -= sizeof(ev); + consumed += sizeof(ev); while (count >= sizeof(*parser->msgs) && parser->pull_pos != parser->push_pos) { spin_unlock_irq(&ff->lock); @@ -707,10 +703,18 @@ static long ff400_copy_msg_to_user(struct snd_ff *ff, char __user *buf, long cou ++parser->pull_pos; if (parser->pull_pos >= FF400_QUEUE_SIZE) parser->pull_pos = 0; + ++ev.message_count; count -= sizeof(*parser->msgs); consumed += sizeof(*parser->msgs); } + spin_unlock_irq(&ff->lock); + if (copy_to_user(buf, &ev, sizeof(ev))) + ret = -EFAULT; + spin_lock_irq(&ff->lock); + if (ret) + return ret; + return consumed; } -- 2.37.2

2 3

[PATCH 1/3] ASoC: dt-bindings: irondevice, sma1303: Rework binding and add missing properties
by Kiseok Jo 04 Feb '23

04 Feb '23

Fix according to the writing-schema.rst file and tested. Add the missing property as compatible, reg, sys-clk-id. Signed-off-by: Kiseok Jo <kiseok.jo(a)irondevice.com> --- .../bindings/sound/irondevice,sma1303.yaml | 35 +++++++++++++++++-- 1 file changed, 32 insertions(+), 3 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/irondevice,sma1303.yaml b/Documentation/devicetree/bindings/sound/irondevice,sma1303.yaml index 162c52606635..2c6158277aee 100644 --- a/Documentation/devicetree/bindings/sound/irondevice,sma1303.yaml +++ b/Documentation/devicetree/bindings/sound/irondevice,sma1303.yaml @@ -10,22 +10,51 @@ maintainers: - Kiseok Jo <kiseok.jo(a)irondevice.com> description: - SMA1303 digital class-D audio amplifier with an integrated boost converter. + SMA1303 digital class-D audio amplifier + with an integrated boost converter. allOf: - - $ref: name-prefix.yaml# + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - irondevice,sma1303 + + reg: + maxItems: 1 + + '#sound-dai-cells': + const: 1 + + sys-clk-id: + description: + select the using system clock. + 0 = External Clock 19.2MHz + 1 = External Clock 24.576MHz + 2 = Using PLL in MCLK + 3 = Using PLL in BCLK + $ref: "/schemas/types.yaml#/definitions/uint32" + maximum: 3 + default: 3 required: - compatible - reg + - '#sound-dai-cells' additionalProperties: false examples: - | - i2c_bus { + i2c { + #address-cells = <1>; + #size-cells = <0>; + amplifier@1e { compatible = "irondevice,sma1303"; reg = <0x1e>; + #sound-dai-cells = <1>; + sys-clk-id = <3>; }; }; base-commit: afdf57285d1e16a1bbdbf6cc33fb22fb6c8e7860 -- 2.20.1

4 6

[PATCH 0/3] Add the Renesas X9250 potentiometers ASoC support
by Herve Codina 04 Feb '23

04 Feb '23

Hi, The Renesas X9250 is a quad digitally controlled potentiometers. In the audio path, it can be present on amplifiers designs and it can be used in ALSA as an auxiliary audio device to control these amplifiers. Best regards, Herve Codina Herve Codina (3): dt-bindings: sound: Add the Renesas X9250 potentiometers ASoC: codecs: Add support for the Renesas X9250 potentiometers MAINTAINERS: add the Renesas X9250 ASoC entry .../bindings/sound/renesas,x9250.yaml | 60 ++++ MAINTAINERS | 7 + sound/soc/codecs/Kconfig | 11 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/x9250.c | 276 ++++++++++++++++++ 5 files changed, 356 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/renesas,x9250.yaml create mode 100644 sound/soc/codecs/x9250.c -- 2.39.0

3 5