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[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>

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[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

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[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

[PATCH 0/9] ASoC: dt-bindings: meson: covert axg audio to schema
by Jerome Brunet 03 Feb '23

03 Feb '23

Convert AXG audio dt-binding documentation to schema Jerome Brunet (9): ASoC: dt-bindings: create component common schema ASoC: dt-bindings: meson: fix gx-card codec node regex ASoC: dt-bindings: meson: convert axg tdm interface to schema ASoC: dt-bindings: meson: convert axg tdm formatters to schema ASoC: dt-bindings: meson: convert axg pdm to schema ASoC: dt-bindings: meson: convert axg fifo to schema ASoC: dt-bindings: meson: convert axg spdif input to schema ASoC: dt-bindings: meson: convert axg spdif output to schema ASoC: dt-bindings: meson: convert axg sound card control to schema .../bindings/sound/amlogic,axg-fifo.txt | 34 ---- .../bindings/sound/amlogic,axg-fifo.yaml | 116 +++++++++++ .../bindings/sound/amlogic,axg-pdm.txt | 29 --- .../bindings/sound/amlogic,axg-pdm.yaml | 86 +++++++++ .../bindings/sound/amlogic,axg-sound-card.txt | 124 ------------ .../sound/amlogic,axg-sound-card.yaml | 182 ++++++++++++++++++ .../bindings/sound/amlogic,axg-spdifin.txt | 27 --- .../bindings/sound/amlogic,axg-spdifin.yaml | 90 +++++++++ .../bindings/sound/amlogic,axg-spdifout.txt | 25 --- .../bindings/sound/amlogic,axg-spdifout.yaml | 83 ++++++++ .../sound/amlogic,axg-tdm-formatters.txt | 36 ---- .../sound/amlogic,axg-tdm-formatters.yaml | 90 +++++++++ .../bindings/sound/amlogic,axg-tdm-iface.txt | 22 --- .../bindings/sound/amlogic,axg-tdm-iface.yaml | 59 ++++++ .../bindings/sound/amlogic,gx-sound-card.yaml | 2 +- .../bindings/sound/component-common.yaml | 21 ++ .../devicetree/bindings/sound/dai-common.yaml | 11 +- 17 files changed, 731 insertions(+), 306 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-fifo.txt create mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-fifo.yaml delete mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-pdm.txt create mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-pdm.yaml delete mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-sound-card.txt create mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-sound-card.yaml delete mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-spdifin.txt create mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-spdifin.yaml delete mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-spdifout.txt create mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-spdifout.yaml delete mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-tdm-formatters.txt create mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-tdm-formatters.yaml delete mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-tdm-iface.txt create mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-tdm-iface.yaml create mode 100644 Documentation/devicetree/bindings/sound/component-common.yaml -- 2.39.0

3 28

[PATCH][next] ASoC: uapi: Replace zero-length arrays with __DECLARE_FLEX_ARRAY() helper
by Gustavo A. R. Silva 03 Feb '23

03 Feb '23

Zero-length arrays are deprecated and we are moving towards adopting C99 flexible-array members, instead. So, replace zero-length arrays declarations in anonymous union with the new __DECLARE_FLEX_ARRAY() helper macro. This helper allows for flexible-array members in unions. Link: https://github.com/KSPP/linux/issues/193 Link: https://github.com/KSPP/linux/issues/227 Link: https://gcc.gnu.org/onlinedocs/gcc/Zero-Length.html Signed-off-by: Gustavo A. R. Silva <gustavoars(a)kernel.org> --- include/uapi/sound/asoc.h | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/include/uapi/sound/asoc.h b/include/uapi/sound/asoc.h index 053949287ce8..dd8ad790fc15 100644 --- a/include/uapi/sound/asoc.h +++ b/include/uapi/sound/asoc.h @@ -226,9 +226,9 @@ struct snd_soc_tplg_vendor_array { __le32 type; /* SND_SOC_TPLG_TUPLE_TYPE_ */ __le32 num_elems; /* number of elements in array */ union { - struct snd_soc_tplg_vendor_uuid_elem uuid[0]; - struct snd_soc_tplg_vendor_value_elem value[0]; - struct snd_soc_tplg_vendor_string_elem string[0]; + __DECLARE_FLEX_ARRAY(struct snd_soc_tplg_vendor_uuid_elem, uuid); + __DECLARE_FLEX_ARRAY(struct snd_soc_tplg_vendor_value_elem, value); + __DECLARE_FLEX_ARRAY(struct snd_soc_tplg_vendor_string_elem, string); }; } __attribute__((packed)); -- 2.34.1

5 11

[PATCH v2] ALSA: hda: Fix the control element identification for multiple codecs
by Jaroslav Kysela 03 Feb '23

03 Feb '23

Some motherboards have multiple HDA codecs connected to the serial bus. The current code may create multiple mixer controls with the almost identical identification. The current code use id.device field from the control element structure to store the codec address to avoid such clashes for multiple codecs. Unfortunately, the user space do not handle this correctly. For mixer controls, only name and index are used for the identifiers. This patch fixes this problem to compose the index using the codec address as an offset in case, when the control already exists. It is really unlikely that one codec will create 10 similar controls. This patch adds new kernel module parameter 'ctl_dev_id' to allow select the old behaviour, too. The CONFIG_SND_HDA_CTL_DEV_ID Kconfig option sets the default value. BugLink: https://github.com/alsa-project/alsa-lib/issues/294 BugLink: https://github.com/alsa-project/alsa-lib/issues/205 Fixes: 54d174031576 ("[ALSA] hda-codec - Fix connection list parsing") Fixes: 1afe206ab699 ("ALSA: hda - Try to find an empty control index when it's occupied") Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> -- rfc..v1: - added CONFIG_SND_HDA_CTL_DEV_ID Kconfig option v1..v2: - ctl_dev_id is not an array - use IS_ENABLED() macro to check the CONFIG_SND_HDA_CTL_DEV_ID setup --- include/sound/hda_codec.h | 1 + sound/pci/hda/Kconfig | 14 ++++++++++++++ sound/pci/hda/hda_codec.c | 13 ++++++++++--- sound/pci/hda/hda_controller.c | 1 + sound/pci/hda/hda_controller.h | 1 + sound/pci/hda/hda_intel.c | 5 +++++ 6 files changed, 32 insertions(+), 3 deletions(-) diff --git a/include/sound/hda_codec.h b/include/sound/hda_codec.h index eba23daf2c29..bbb7805e85d8 100644 --- a/include/sound/hda_codec.h +++ b/include/sound/hda_codec.h @@ -259,6 +259,7 @@ struct hda_codec { unsigned int relaxed_resume:1; /* don't resume forcibly for jack */ unsigned int forced_resume:1; /* forced resume for jack */ unsigned int no_stream_clean_at_suspend:1; /* do not clean streams at suspend */ + unsigned int ctl_dev_id:1; /* old control element id build behaviour */ #ifdef CONFIG_PM unsigned long power_on_acct; diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index 06d304db4183..886255a03e8b 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -302,6 +302,20 @@ config SND_HDA_INTEL_HDMI_SILENT_STREAM This feature can impact power consumption as resources are kept reserved both at transmitter and receiver. +config SND_HDA_CTL_DEV_ID + bool "Use the device identifier field for controls" + depends on SND_HDA_INTEL + help + Say Y to use the device identifier field for (mixer) + controls (old behaviour until this option is available). + + When enabled, the multiple HDA codecs may set the device + field in control (mixer) element identifiers. The use + of this field is not recommended and defined for mixer controls. + + The old behaviour (Y) is obsolete and will be removed. Consider + to not enable this option. + endif endmenu diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index edd653ece70d..8f20abc036bf 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3390,7 +3390,12 @@ int snd_hda_add_new_ctls(struct hda_codec *codec, kctl = snd_ctl_new1(knew, codec); if (!kctl) return -ENOMEM; - if (addr > 0) + /* Do not use the id.device field for MIXER elements. + * This field is for real device numbers (like PCM) but codecs + * are hidden components from the user space view (unrelated + * to the mixer element identification). + */ + if (addr > 0 && codec->ctl_dev_id) kctl->id.device = addr; if (idx > 0) kctl->id.index = idx; @@ -3401,9 +3406,11 @@ int snd_hda_add_new_ctls(struct hda_codec *codec, * the codec addr; if it still fails (or it's the * primary codec), then try another control index */ - if (!addr && codec->core.addr) + if (!addr && codec->core.addr) { addr = codec->core.addr; - else if (!idx && !knew->index) { + if (!codec->ctl_dev_id) + idx += 10 * addr; + } else if (!idx && !knew->index) { idx = find_empty_mixer_ctl_idx(codec, knew->name, 0); if (idx <= 0) diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index 0ff286b7b66b..083df287c1a4 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -1231,6 +1231,7 @@ int azx_probe_codecs(struct azx *chip, unsigned int max_slots) continue; codec->jackpoll_interval = chip->jackpoll_interval; codec->beep_mode = chip->beep_mode; + codec->ctl_dev_id = chip->ctl_dev_id; codecs++; } } diff --git a/sound/pci/hda/hda_controller.h b/sound/pci/hda/hda_controller.h index f5bf295eb830..8556031bcd68 100644 --- a/sound/pci/hda/hda_controller.h +++ b/sound/pci/hda/hda_controller.h @@ -124,6 +124,7 @@ struct azx { /* HD codec */ int codec_probe_mask; /* copied from probe_mask option */ unsigned int beep_mode; + bool ctl_dev_id; #ifdef CONFIG_SND_HDA_PATCH_LOADER const struct firmware *fw; diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 87002670c0c9..2dbc082076f6 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -119,6 +119,7 @@ static bool beep_mode[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = CONFIG_SND_HDA_INPUT_BEEP_MODE}; #endif static bool dmic_detect = 1; +static bool ctl_dev_id = IS_ENABLED(CONFIG_SND_HDA_CTL_DEV_ID) ? 1 : 0; module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for Intel HD audio interface."); @@ -157,6 +158,8 @@ module_param(dmic_detect, bool, 0444); MODULE_PARM_DESC(dmic_detect, "Allow DSP driver selection (bypass this driver) " "(0=off, 1=on) (default=1); " "deprecated, use snd-intel-dspcfg.dsp_driver option instead"); +module_param(ctl_dev_id, bool, 0444); +MODULE_PARM_DESC(ctl_dev_id, "Use control device identifier (based on codec address)."); #ifdef CONFIG_PM static int param_set_xint(const char *val, const struct kernel_param *kp); @@ -2278,6 +2281,8 @@ static int azx_probe_continue(struct azx *chip) chip->beep_mode = beep_mode[dev]; #endif + chip->ctl_dev_id = ctl_dev_id; + /* create codec instances */ if (bus->codec_mask) { err = azx_probe_codecs(chip, azx_max_codecs[chip->driver_type]); -- 2.39.0

2 1

[PATCH v3 0/9] ASoC: SOF: core/ipc4/mtl: Add support for PCM delay reporting
by Peter Ujfalusi 03 Feb '23

03 Feb '23

Hi, Changes since v2: - Use div64_u64_rem() to make the code compile and work on non 64bit architectures Changes since v1: - The delay calculation updated to take into account the counter wrapping on both ends (host and link side). The following series adds support for the PCM delay reporting in SOF core level and implements the needed infrastructure with IPC4 to finally enable it for MTL. Currently this is only supported on MTL (and via IPC4), but with the infrastructure in place it will be possible to support other platforms with DeepBuffer. Regards, Peter --- Rander Wang (9): ASoC: SOF: add ipc4_fw_reg header file ASoC: SOF: add fw_info_box support ASoC: SOF: add time info structure for ipc4 path ASoC: SOF: ipc4-pcm: allocate time info for pcm delay feature ASoC: SOF: ipc4-pcm: add hw_params ASoC: SOF: add delay function support in sof framework ASoC: SOF: add get_stream_position ops for pcm delay ASoC: SOF: Intel: mtl: add get_stream_position support ASoC: SOF: ipc4-pcm: add delay function support sound/soc/sof/intel/mtl.c | 14 +++ sound/soc/sof/intel/mtl.h | 6 + sound/soc/sof/ipc4-fw-reg.h | 155 ++++++++++++++++++++++++ sound/soc/sof/ipc4-pcm.c | 232 ++++++++++++++++++++++++++++++++++++ sound/soc/sof/ipc4-priv.h | 15 +++ sound/soc/sof/ipc4.c | 3 + sound/soc/sof/ops.h | 10 ++ sound/soc/sof/pcm.c | 13 ++ sound/soc/sof/sof-audio.h | 6 + sound/soc/sof/sof-priv.h | 10 ++ 10 files changed, 464 insertions(+) create mode 100644 sound/soc/sof/ipc4-fw-reg.h -- 2.39.1

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[PATCH 0/9] ASoC: SOF: core/ipc4/mtl: Add support for PCM delay reporting
by Peter Ujfalusi 03 Feb '23

03 Feb '23

Hi, The following series adds support for the PCM delay reporting in SOF core level and implements the needed infrastructure with IPC4 to finally enable it for MTL. Currently this is only supported on MTL (and via IPC4), but with the infrastructure in place it will be possible to support other platforms with DeepBuffer. Regards, Peter --- Rander Wang (9): ASoC: SOF: add ipc4_fw_reg header file ASoC: SOF: add fw_info_box support ASoC: SOF: add time info structure for ipc4 path ASoC: SOF: ipc4-pcm: allocate time info for pcm delay feature ASoC: SOF: ipc4-pcm: add hw_params ASoC: SOF: add delay function support in sof framework ASoC: SOF: add get_stream_position ops for pcm delay ASoC: SOF: Intel: mtl: add get_stream_position support ASoC: SOF: ipc4-pcm: add delay function support sound/soc/sof/intel/mtl.c | 14 +++ sound/soc/sof/intel/mtl.h | 6 + sound/soc/sof/ipc4-fw-reg.h | 155 ++++++++++++++++++++++++ sound/soc/sof/ipc4-pcm.c | 230 ++++++++++++++++++++++++++++++++++++ sound/soc/sof/ipc4-priv.h | 15 +++ sound/soc/sof/ipc4.c | 3 + sound/soc/sof/ops.h | 10 ++ sound/soc/sof/pcm.c | 13 ++ sound/soc/sof/sof-audio.h | 6 + sound/soc/sof/sof-priv.h | 10 ++ 10 files changed, 462 insertions(+) create mode 100644 sound/soc/sof/ipc4-fw-reg.h -- 2.39.1

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[PATCH] ASoC: mediatek: mt8188: remove etdm dead code
by Trevor Wu 03 Feb '23

03 Feb '23

Some Smatch static checker warning like below was found. sound/soc/mediatek/mt8188/mt8188-dai-etdm.c:2487 mt8188_dai_etdm_parse_of() warn: 'ret' returned from snprintf() might be larger than 48 2479 for (i = 0; i < MT8188_AFE_IO_ETDM_NUM; i++) { 2480 dai_id = ETDM_TO_DAI_ID(i); 2481 etdm_data = afe_priv->dai_priv[dai_id]; 2482 2483 ret = snprintf(prop, sizeof(prop), 2484 "mediatek,%s-multi-pin-mode", 2485 of_afe_etdms[i].name); 2486 if (ret < 0) { --> 2487 dev_err(afe->dev, "%s snprintf err=%d\n", 2488 In linux kernel, snprintf() never returns negatives. On the other hand, the format string like "mediatek,%s-multi-pin-mode" must be smaller than sizeof(prop)=48. After discussing in the mail thread[1], I remove the dead code to fix the Smatch warnings. [1]: https://lore.kernel.org/all/Y9EdBg641tJDDrt%2F@kili/ Signed-off-by: Trevor Wu <trevor.wu(a)mediatek.com> --- sound/soc/mediatek/mt8188/mt8188-dai-etdm.c | 33 ++++++--------------- 1 file changed, 9 insertions(+), 24 deletions(-) diff --git a/sound/soc/mediatek/mt8188/mt8188-dai-etdm.c b/sound/soc/mediatek/mt8188/mt8188-dai-etdm.c index 0b79c1cc293b..071841903c62 100644 --- a/sound/soc/mediatek/mt8188/mt8188-dai-etdm.c +++ b/sound/soc/mediatek/mt8188/mt8188-dai-etdm.c @@ -2480,24 +2480,14 @@ static void mt8188_dai_etdm_parse_of(struct mtk_base_afe *afe) dai_id = ETDM_TO_DAI_ID(i); etdm_data = afe_priv->dai_priv[dai_id]; - ret = snprintf(prop, sizeof(prop), - "mediatek,%s-multi-pin-mode", - of_afe_etdms[i].name); - if (ret < 0) { - dev_err(afe->dev, "%s snprintf err=%d\n", - __func__, ret); - return; - } + snprintf(prop, sizeof(prop), "mediatek,%s-multi-pin-mode", + of_afe_etdms[i].name); + etdm_data->data_mode = of_property_read_bool(of_node, prop); - ret = snprintf(prop, sizeof(prop), - "mediatek,%s-cowork-source", - of_afe_etdms[i].name); - if (ret < 0) { - dev_err(afe->dev, "%s snprintf err=%d\n", - __func__, ret); - return; - } + snprintf(prop, sizeof(prop), "mediatek,%s-cowork-source", + of_afe_etdms[i].name); + ret = of_property_read_u32(of_node, prop, &sel); if (ret == 0) { if (sel >= MT8188_AFE_IO_ETDM_NUM) { @@ -2516,14 +2506,9 @@ static void mt8188_dai_etdm_parse_of(struct mtk_base_afe *afe) /* etdm in only */ for (i = 0; i < 2; i++) { - ret = snprintf(prop, sizeof(prop), - "mediatek,%s-chn-disabled", - of_afe_etdms[i].name); - if (ret < 0) { - dev_err(afe->dev, "%s snprintf err=%d\n", - __func__, ret); - return; - } + snprintf(prop, sizeof(prop), "mediatek,%s-chn-disabled", + of_afe_etdms[i].name); + ret = of_property_read_variable_u8_array(of_node, prop, disable_chn, 1, max_chn); -- 2.18.0

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