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[PATCH alsa-lib] seq: Add snd_seq_{get|set}_ump_is_midi1() API functions
by Takashi Iwai 07 Aug '24

07 Aug '24

Implement the API function calls corresponding to the new sequencer port flag bit that has been added recently to the kernel. A UMP MIDI 2.0 device allow to have an optional MIDI 1.0 port while speaking in MIDI 2.0 protocol for other UMP Groups. The new seq port flag indicates that. This is rather a minor difference, and since ALSA sequencer core covers the all conversions, usually you don't have to worry about it at all. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/seq.h | 2 ++ include/sound/uapi/asequencer.h | 2 ++ src/Versions.in.in | 2 ++ src/seq/seq.c | 29 +++++++++++++++++++++++++++++ 4 files changed, 35 insertions(+) diff --git a/include/seq.h b/include/seq.h index 4f03df438288..2eee95a643b4 100644 --- a/include/seq.h +++ b/include/seq.h @@ -318,6 +318,7 @@ int snd_seq_port_info_get_timestamp_real(const snd_seq_port_info_t *info); int snd_seq_port_info_get_timestamp_queue(const snd_seq_port_info_t *info); int snd_seq_port_info_get_direction(const snd_seq_port_info_t *info); int snd_seq_port_info_get_ump_group(const snd_seq_port_info_t *info); +int snd_seq_port_info_get_ump_is_midi1(const snd_seq_port_info_t *info); void snd_seq_port_info_set_client(snd_seq_port_info_t *info, int client); void snd_seq_port_info_set_port(snd_seq_port_info_t *info, int port); @@ -334,6 +335,7 @@ void snd_seq_port_info_set_timestamp_real(snd_seq_port_info_t *info, int realtim void snd_seq_port_info_set_timestamp_queue(snd_seq_port_info_t *info, int queue); void snd_seq_port_info_set_direction(snd_seq_port_info_t *info, int direction); void snd_seq_port_info_set_ump_group(snd_seq_port_info_t *info, int ump_group); +void snd_seq_port_info_set_ump_is_midi1(snd_seq_port_info_t *info, int is_midi1); int snd_seq_create_port(snd_seq_t *handle, snd_seq_port_info_t *info); int snd_seq_delete_port(snd_seq_t *handle, int port); diff --git a/include/sound/uapi/asequencer.h b/include/sound/uapi/asequencer.h index 923dfdddfc76..b3e9df5894af 100644 --- a/include/sound/uapi/asequencer.h +++ b/include/sound/uapi/asequencer.h @@ -477,6 +477,8 @@ struct snd_seq_remove_events { #define SNDRV_SEQ_PORT_FLG_TIMESTAMP (1<<1) #define SNDRV_SEQ_PORT_FLG_TIME_REAL (1<<2) +#define SNDRV_SEQ_PORT_FLG_IS_MIDI1 (1<<3) /* Keep MIDI 1.0 protocol */ + /* port direction */ #define SNDRV_SEQ_PORT_DIR_UNKNOWN 0 #define SNDRV_SEQ_PORT_DIR_INPUT 1 diff --git a/src/Versions.in.in b/src/Versions.in.in index 90849277c983..298e610c2099 100644 --- a/src/Versions.in.in +++ b/src/Versions.in.in @@ -209,6 +209,8 @@ ALSA_1.2.13 { @SYMBOL_PREFIX@snd_seq_queue_tempo_get_tempo_base; @SYMBOL_PREFIX@snd_seq_queue_tempo_set_tempo_base; @SYMBOL_PREFIX@snd_seq_has_tempo_base; + @SYMBOL_PREFIX@snd_seq_port_info_get_ump_is_midi1; + @SYMBOL_PREFIX@snd_seq_port_info_set_ump_is_midi1; #endif #ifdef HAVE_RAWMIDI_SYMS @SYMBOL_PREFIX@snd_ump_endpoint_info_clear; diff --git a/src/seq/seq.c b/src/seq/seq.c index d0ecedf0838f..347ff45544fa 100644 --- a/src/seq/seq.c +++ b/src/seq/seq.c @@ -2431,6 +2431,19 @@ int snd_seq_port_info_get_ump_group(const snd_seq_port_info_t *info) return info->ump_group; } +/** + * \brief Get the status of the optional MIDI 1.0 port in MIDI 2.0 UMP Endpoint + * \param info port_info container + * \return 1 if it's an optional MIDI 1.0 port in MIDI 2.0 UMP Endpoint + * + * \sa snd_seq_get_port_info(), snd_seq_port_info_set_ump_is_midi1() + */ +int snd_seq_port_info_get_ump_is_midi1(const snd_seq_port_info_t *info) +{ + assert(info); + return !!(info->flags & SNDRV_SEQ_PORT_FLG_IS_MIDI1); +} + /** * \brief Set the client id of a port_info container * \param info port_info container @@ -2635,6 +2648,22 @@ void snd_seq_port_info_set_ump_group(snd_seq_port_info_t *info, int ump_group) info->ump_group = ump_group; } +/** + * \brief Set the optional MIDI 1.0 port in MIDI 2.0 UMP Endpoint + * \param info port_info container + * \param is_midi1 non-zero for MIDI 1.0 port in MIDI 2.0 EP + * + * \sa snd_seq_get_port_info(), snd_seq_port_info_get_ump_is_midi1() + */ +void snd_seq_port_info_set_ump_is_midi1(snd_seq_port_info_t *info, int is_midi1) +{ + assert(info); + if (is_midi1) + info->flags |= SNDRV_SEQ_PORT_FLG_IS_MIDI1; + else + info->flags &= ~SNDRV_SEQ_PORT_FLG_IS_MIDI1; +} + /** * \brief create a sequencer port on the current client * \param seq sequencer handle -- 2.43.0

1 0

Re: [PATCH v11 07/10] mtd: spi-nor: Add stacked memories support in spi-nor
by Michael Walle 07 Aug '24

07 Aug '24

Hi, > Based on the inputs/suggestions from Tudor, i am planning to add a new > layer between the SPI-NOR and MTD layers to support stacked and parallel > configurations. This new layer will be part of the spi-nor and located in > mtd/spi-nor/ Will AMD submit to maintain this layer? What happens if the maintainer will leave AMD? TBH, personally, I don't like to maintain such a niche feature. I'd really like to see some use cases and performance reports for this, like actual boards (and no evaluation boards don't count). Why wouldn't someone just use an octal flash? And as already mentioned there is also mtdcat, which seems to duplicate some features? -michael

5 13

[PATCH v1] ALSA: ASoC/tas2781: fix wrong calibrated data order
by Shenghao Ding 07 Aug '24

07 Aug '24

From: Baojun Xu <baojun.xu(a)ti.com> Wrong calibration data order cause sound too low in some device. Fix wrong calibrated data order, add calibration data converssion by get_unaligned_be32() after reading from UEFI. Fixes: 5be27f1e3ec9 ("ALSA: hda/tas2781: Add tas2781 HDA driver") Signed-off-by: Baojun Xu <baojun.xu(a)ti.com> --- v1: - Change copyright date, and add new maintainer. - Add unaligned.h included for get_unaligned_be32(). - In tas2781_apply_calib(), change data address transfer directly to get data by get_unaligned_be32(), and send address to device. --- sound/pci/hda/tas2781_hda_i2c.c | 15 +++++++++------ 1 file changed, 9 insertions(+), 6 deletions(-) diff --git a/sound/pci/hda/tas2781_hda_i2c.c b/sound/pci/hda/tas2781_hda_i2c.c index 49bd7097d..4dc3350d5 100644 --- a/sound/pci/hda/tas2781_hda_i2c.c +++ b/sound/pci/hda/tas2781_hda_i2c.c @@ -2,10 +2,12 @@ // // TAS2781 HDA I2C driver // -// Copyright 2023 Texas Instruments, Inc. +// Copyright 2023 - 2024 Texas Instruments, Inc. // // Author: Shenghao Ding <shenghao-ding(a)ti.com> +// Current maintainer: Baojun Xu <baojun.xu(a)ti.com> +#include <asm/unaligned.h> #include <linux/acpi.h> #include <linux/crc8.h> #include <linux/crc32.h> @@ -519,20 +521,21 @@ static void tas2781_apply_calib(struct tasdevice_priv *tas_priv) static const unsigned char rgno_array[CALIB_MAX] = { 0x74, 0x0c, 0x14, 0x70, 0x7c, }; - unsigned char *data; - int i, j, rc; + int i, j, rc, data; + int offset = 0; for (i = 0; i < tas_priv->ndev; i++) { - data = tas_priv->cali_data.data + - i * TASDEVICE_SPEAKER_CALIBRATION_SIZE; for (j = 0; j < CALIB_MAX; j++) { + data = get_unaligned_be32( + &tas_priv->cali_data.data[offset]); rc = tasdevice_dev_bulk_write(tas_priv, i, TASDEVICE_REG(0, page_array[j], rgno_array[j]), - &(data[4 * j]), 4); + (unsigned char *)&data, 4); if (rc < 0) dev_err(tas_priv->dev, "chn %d calib %d bulk_wr err = %d\n", i, j, rc); + offset += 4; } } } -- 2.40.1

2 1

[PATCH] ALSA: hda/realtek: Add Framework Laptop 13 (Intel Core Ultra) to quirks
by Dustin L. Howett 07 Aug '24

07 Aug '24

The Framework Laptop 13 (Intel Core Ultra) has an ALC285 that ships in a similar configuration to the ALC295 in previous models. It requires the same quirk for headset detection. --- Signed-off-by: Dustin L. Howett <dustin(a)howett.net> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1645d21d422f..480e82df7a4c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10678,6 +10678,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x8086, 0x3038, "Intel NUC 13", ALC295_FIXUP_CHROME_BOOK), SND_PCI_QUIRK(0xf111, 0x0001, "Framework Laptop", ALC295_FIXUP_FRAMEWORK_LAPTOP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0xf111, 0x0006, "Framework Laptop", ALC295_FIXUP_FRAMEWORK_LAPTOP_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0xf111, 0x0009, "Framework Laptop", ALC295_FIXUP_FRAMEWORK_LAPTOP_MIC_NO_PRESENCE), #if 0 /* Below is a quirk table taken from the old code. --- base-commit: de9c2c66ad8e787abec7c9d7eff4f8c3cdd28aed change-id: 20240606-alsa-hda-realtek-add-framework-laptop-13-intel-core-ultra-to-quirks-2af5081e2f2c Best regards, -- Dustin L. Howett <dustin(a)howett.net>

2 1

[PATCH v2] ASoC: remove bespoke trigger support
by Kuninori Morimoto 06 Aug '24

06 Aug '24

Bespoke trigger support was added when Linux v3.5 by this patch. commit 07bf84aaf736781a283b1bd36eaa911453b14574 ("ASoC: dpcm: Add bespoke trigger()") test-component driver is using it, but this is because it indicates used function for debug/trace purpose. Except test-component driver, bespoke trigger has never been used over 10 years in upstream. We can re-support it if needed in the future, but let's remove it for now, because it just noise in upstream. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- v1 -> v2 - rebased to latest asoc/for-6.12 branch include/sound/soc-dai.h | 5 +--- include/sound/soc-dpcm.h | 1 - sound/soc/generic/test-component.c | 9 ------ sound/soc/soc-dai.c | 20 ------------- sound/soc/soc-pcm.c | 47 +++--------------------------- 5 files changed, 5 insertions(+), 77 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index bbb72ad4c9518..ab4e109fe71d2 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -240,8 +240,6 @@ int snd_soc_pcm_dai_new(struct snd_soc_pcm_runtime *rtd); int snd_soc_pcm_dai_prepare(struct snd_pcm_substream *substream); int snd_soc_pcm_dai_trigger(struct snd_pcm_substream *substream, int cmd, int rollback); -int snd_soc_pcm_dai_bespoke_trigger(struct snd_pcm_substream *substream, - int cmd); void snd_soc_pcm_dai_delay(struct snd_pcm_substream *substream, snd_pcm_sframes_t *cpu_delay, snd_pcm_sframes_t *codec_delay); @@ -345,8 +343,7 @@ struct snd_soc_dai_ops { */ int (*trigger)(struct snd_pcm_substream *, int, struct snd_soc_dai *); - int (*bespoke_trigger)(struct snd_pcm_substream *, int, - struct snd_soc_dai *); + /* * For hardware based FIFO caused delay reporting. * Optional. diff --git a/include/sound/soc-dpcm.h b/include/sound/soc-dpcm.h index ebd24753dd000..773f2db8c31c8 100644 --- a/include/sound/soc-dpcm.h +++ b/include/sound/soc-dpcm.h @@ -58,7 +58,6 @@ enum snd_soc_dpcm_state { enum snd_soc_dpcm_trigger { SND_SOC_DPCM_TRIGGER_PRE = 0, SND_SOC_DPCM_TRIGGER_POST, - SND_SOC_DPCM_TRIGGER_BESPOKE, }; /* diff --git a/sound/soc/generic/test-component.c b/sound/soc/generic/test-component.c index e9e5e235a8a65..df2487b700cca 100644 --- a/sound/soc/generic/test-component.c +++ b/sound/soc/generic/test-component.c @@ -181,14 +181,6 @@ static int test_dai_trigger(struct snd_pcm_substream *substream, int cmd, struct return 0; } -static int test_dai_bespoke_trigger(struct snd_pcm_substream *substream, - int cmd, struct snd_soc_dai *dai) -{ - mile_stone(dai); - - return 0; -} - static const u64 test_dai_formats = /* * Select below from Sound Card, not auto @@ -228,7 +220,6 @@ static const struct snd_soc_dai_ops test_verbose_ops = { .hw_params = test_dai_hw_params, .hw_free = test_dai_hw_free, .trigger = test_dai_trigger, - .bespoke_trigger = test_dai_bespoke_trigger, .auto_selectable_formats = &test_dai_formats, .num_auto_selectable_formats = 1, }; diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c index 9e47053419c16..302ca753a1f35 100644 --- a/sound/soc/soc-dai.c +++ b/sound/soc/soc-dai.c @@ -685,26 +685,6 @@ int snd_soc_pcm_dai_trigger(struct snd_pcm_substream *substream, return ret; } -int snd_soc_pcm_dai_bespoke_trigger(struct snd_pcm_substream *substream, - int cmd) -{ - struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); - struct snd_soc_dai *dai; - int i, ret; - - for_each_rtd_dais(rtd, i, dai) { - if (dai->driver->ops && - dai->driver->ops->bespoke_trigger) { - ret = dai->driver->ops->bespoke_trigger(substream, - cmd, dai); - if (ret < 0) - return soc_dai_ret(dai, ret); - } - } - - return 0; -} - void snd_soc_pcm_dai_delay(struct snd_pcm_substream *substream, snd_pcm_sframes_t *cpu_delay, snd_pcm_sframes_t *codec_delay) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 5520944ac9ddc..1edcb8d6f6eea 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2388,14 +2388,6 @@ static int dpcm_fe_dai_do_trigger(struct snd_pcm_substream *substream, int cmd) break; } break; - case SND_SOC_DPCM_TRIGGER_BESPOKE: - /* bespoke trigger() - handles both FE and BEs */ - - dev_dbg(fe->dev, "ASoC: bespoke trigger FE %s cmd %d\n", - fe->dai_link->name, cmd); - - ret = snd_soc_pcm_dai_bespoke_trigger(substream, cmd); - break; default: dev_err(fe->dev, "ASoC: invalid trigger cmd %d for %s\n", cmd, fe->dai_link->name); @@ -2525,26 +2517,12 @@ static int dpcm_fe_dai_prepare(struct snd_pcm_substream *substream) static int dpcm_run_update_shutdown(struct snd_soc_pcm_runtime *fe, int stream) { - struct snd_pcm_substream *substream = - snd_soc_dpcm_get_substream(fe, stream); - enum snd_soc_dpcm_trigger trigger = fe->dai_link->trigger[stream]; int err; dev_dbg(fe->dev, "ASoC: runtime %s close on FE %s\n", snd_pcm_direction_name(stream), fe->dai_link->name); - if (trigger == SND_SOC_DPCM_TRIGGER_BESPOKE) { - /* call bespoke trigger - FE takes care of all BE triggers */ - dev_dbg(fe->dev, "ASoC: bespoke trigger FE %s cmd stop\n", - fe->dai_link->name); - - err = snd_soc_pcm_dai_bespoke_trigger(substream, SNDRV_PCM_TRIGGER_STOP); - } else { - dev_dbg(fe->dev, "ASoC: trigger FE %s cmd stop\n", - fe->dai_link->name); - - err = dpcm_be_dai_trigger(fe, stream, SNDRV_PCM_TRIGGER_STOP); - } + err = dpcm_be_dai_trigger(fe, stream, SNDRV_PCM_TRIGGER_STOP); dpcm_be_dai_hw_free(fe, stream); @@ -2558,10 +2536,7 @@ static int dpcm_run_update_shutdown(struct snd_soc_pcm_runtime *fe, int stream) static int dpcm_run_update_startup(struct snd_soc_pcm_runtime *fe, int stream) { - struct snd_pcm_substream *substream = - snd_soc_dpcm_get_substream(fe, stream); struct snd_soc_dpcm *dpcm; - enum snd_soc_dpcm_trigger trigger = fe->dai_link->trigger[stream]; int ret = 0; dev_dbg(fe->dev, "ASoC: runtime %s open on FE %s\n", @@ -2605,23 +2580,9 @@ static int dpcm_run_update_startup(struct snd_soc_pcm_runtime *fe, int stream) fe->dpcm[stream].state == SND_SOC_DPCM_STATE_STOP) return 0; - if (trigger == SND_SOC_DPCM_TRIGGER_BESPOKE) { - /* call trigger on the frontend - FE takes care of all BE triggers */ - dev_dbg(fe->dev, "ASoC: bespoke trigger FE %s cmd start\n", - fe->dai_link->name); - - ret = snd_soc_pcm_dai_bespoke_trigger(substream, SNDRV_PCM_TRIGGER_START); - if (ret < 0) - goto hw_free; - } else { - dev_dbg(fe->dev, "ASoC: trigger FE %s cmd start\n", - fe->dai_link->name); - - ret = dpcm_be_dai_trigger(fe, stream, - SNDRV_PCM_TRIGGER_START); - if (ret < 0) - goto hw_free; - } + ret = dpcm_be_dai_trigger(fe, stream, SNDRV_PCM_TRIGGER_START); + if (ret < 0) + goto hw_free; return 0; -- 2.43.0

2 1

Re: [PATCH 0/2] ASoC: Extend wm_adsp so cs35l56 can suppress controls
by Mark Brown 06 Aug '24

06 Aug '24

On Mon, 05 Aug 2024 10:27:19 +0000, Simon Trimmer wrote: > This pair of patches extend wm_adsp to add a callback that can be used > to control whether ALSA controls are added and then tweak cs35l56 to use > it to suppress controls made from firmware coefficients. > > Simon Trimmer (2): > ASoC: wm_adsp: Add control_add callback and export > wm_adsp_control_add() > ASoC: cs35l56: Stop creating ALSA controls for firmware coefficients > > [...] Applied to https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next Thanks! [1/2] ASoC: wm_adsp: Add control_add callback and export wm_adsp_control_add() commit: 45b4acab4cac79503663f0a4be9eb3752db04d4b [2/2] ASoC: cs35l56: Stop creating ALSA controls for firmware coefficients commit: 2c3640b82213cf2beb7c1cc3cfce2ecf5349b0de All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark

1 0

[PATCH] ASoC: remove bespoke trigger support
by Kuninori Morimoto 06 Aug '24

06 Aug '24

Bespoke trigger support was added when Linux v3.5 by this patch. commit 07bf84aaf736781a283b1bd36eaa911453b14574 ("ASoC: dpcm: Add bespoke trigger()") test-component driver is using it, but this is because it indicates used function for debug/trace purpose. Except test-component driver, bespoke trigger has never been used over 10 years in upstream. We can re-support it if needed in the future, but let's remove it for now, because it just noise in upstream. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- include/sound/soc-dai.h | 5 +--- include/sound/soc-dpcm.h | 1 - sound/soc/generic/test-component.c | 9 ------ sound/soc/soc-dai.c | 20 ------------- sound/soc/soc-pcm.c | 47 +++--------------------------- 5 files changed, 5 insertions(+), 77 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index bbb72ad4c9518..ab4e109fe71d2 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -240,8 +240,6 @@ int snd_soc_pcm_dai_new(struct snd_soc_pcm_runtime *rtd); int snd_soc_pcm_dai_prepare(struct snd_pcm_substream *substream); int snd_soc_pcm_dai_trigger(struct snd_pcm_substream *substream, int cmd, int rollback); -int snd_soc_pcm_dai_bespoke_trigger(struct snd_pcm_substream *substream, - int cmd); void snd_soc_pcm_dai_delay(struct snd_pcm_substream *substream, snd_pcm_sframes_t *cpu_delay, snd_pcm_sframes_t *codec_delay); @@ -345,8 +343,7 @@ struct snd_soc_dai_ops { */ int (*trigger)(struct snd_pcm_substream *, int, struct snd_soc_dai *); - int (*bespoke_trigger)(struct snd_pcm_substream *, int, - struct snd_soc_dai *); + /* * For hardware based FIFO caused delay reporting. * Optional. diff --git a/include/sound/soc-dpcm.h b/include/sound/soc-dpcm.h index ebd24753dd000..773f2db8c31c8 100644 --- a/include/sound/soc-dpcm.h +++ b/include/sound/soc-dpcm.h @@ -58,7 +58,6 @@ enum snd_soc_dpcm_state { enum snd_soc_dpcm_trigger { SND_SOC_DPCM_TRIGGER_PRE = 0, SND_SOC_DPCM_TRIGGER_POST, - SND_SOC_DPCM_TRIGGER_BESPOKE, }; /* diff --git a/sound/soc/generic/test-component.c b/sound/soc/generic/test-component.c index e9e5e235a8a65..df2487b700cca 100644 --- a/sound/soc/generic/test-component.c +++ b/sound/soc/generic/test-component.c @@ -181,14 +181,6 @@ static int test_dai_trigger(struct snd_pcm_substream *substream, int cmd, struct return 0; } -static int test_dai_bespoke_trigger(struct snd_pcm_substream *substream, - int cmd, struct snd_soc_dai *dai) -{ - mile_stone(dai); - - return 0; -} - static const u64 test_dai_formats = /* * Select below from Sound Card, not auto @@ -228,7 +220,6 @@ static const struct snd_soc_dai_ops test_verbose_ops = { .hw_params = test_dai_hw_params, .hw_free = test_dai_hw_free, .trigger = test_dai_trigger, - .bespoke_trigger = test_dai_bespoke_trigger, .auto_selectable_formats = &test_dai_formats, .num_auto_selectable_formats = 1, }; diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c index 9e47053419c16..302ca753a1f35 100644 --- a/sound/soc/soc-dai.c +++ b/sound/soc/soc-dai.c @@ -685,26 +685,6 @@ int snd_soc_pcm_dai_trigger(struct snd_pcm_substream *substream, return ret; } -int snd_soc_pcm_dai_bespoke_trigger(struct snd_pcm_substream *substream, - int cmd) -{ - struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); - struct snd_soc_dai *dai; - int i, ret; - - for_each_rtd_dais(rtd, i, dai) { - if (dai->driver->ops && - dai->driver->ops->bespoke_trigger) { - ret = dai->driver->ops->bespoke_trigger(substream, - cmd, dai); - if (ret < 0) - return soc_dai_ret(dai, ret); - } - } - - return 0; -} - void snd_soc_pcm_dai_delay(struct snd_pcm_substream *substream, snd_pcm_sframes_t *cpu_delay, snd_pcm_sframes_t *codec_delay) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 4571785736f76..5646525ad7924 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2371,14 +2371,6 @@ static int dpcm_fe_dai_do_trigger(struct snd_pcm_substream *substream, int cmd) break; } break; - case SND_SOC_DPCM_TRIGGER_BESPOKE: - /* bespoke trigger() - handles both FE and BEs */ - - dev_dbg(fe->dev, "ASoC: bespoke trigger FE %s cmd %d\n", - fe->dai_link->name, cmd); - - ret = snd_soc_pcm_dai_bespoke_trigger(substream, cmd); - break; default: dev_err(fe->dev, "ASoC: invalid trigger cmd %d for %s\n", cmd, fe->dai_link->name); @@ -2508,26 +2500,12 @@ static int dpcm_fe_dai_prepare(struct snd_pcm_substream *substream) static int dpcm_run_update_shutdown(struct snd_soc_pcm_runtime *fe, int stream) { - struct snd_pcm_substream *substream = - snd_soc_dpcm_get_substream(fe, stream); - enum snd_soc_dpcm_trigger trigger = fe->dai_link->trigger[stream]; int err; dev_dbg(fe->dev, "ASoC: runtime %s close on FE %s\n", stream ? "capture" : "playback", fe->dai_link->name); - if (trigger == SND_SOC_DPCM_TRIGGER_BESPOKE) { - /* call bespoke trigger - FE takes care of all BE triggers */ - dev_dbg(fe->dev, "ASoC: bespoke trigger FE %s cmd stop\n", - fe->dai_link->name); - - err = snd_soc_pcm_dai_bespoke_trigger(substream, SNDRV_PCM_TRIGGER_STOP); - } else { - dev_dbg(fe->dev, "ASoC: trigger FE %s cmd stop\n", - fe->dai_link->name); - - err = dpcm_be_dai_trigger(fe, stream, SNDRV_PCM_TRIGGER_STOP); - } + err = dpcm_be_dai_trigger(fe, stream, SNDRV_PCM_TRIGGER_STOP); dpcm_be_dai_hw_free(fe, stream); @@ -2541,10 +2519,7 @@ static int dpcm_run_update_shutdown(struct snd_soc_pcm_runtime *fe, int stream) static int dpcm_run_update_startup(struct snd_soc_pcm_runtime *fe, int stream) { - struct snd_pcm_substream *substream = - snd_soc_dpcm_get_substream(fe, stream); struct snd_soc_dpcm *dpcm; - enum snd_soc_dpcm_trigger trigger = fe->dai_link->trigger[stream]; int ret = 0; dev_dbg(fe->dev, "ASoC: runtime %s open on FE %s\n", @@ -2588,23 +2563,9 @@ static int dpcm_run_update_startup(struct snd_soc_pcm_runtime *fe, int stream) fe->dpcm[stream].state == SND_SOC_DPCM_STATE_STOP) return 0; - if (trigger == SND_SOC_DPCM_TRIGGER_BESPOKE) { - /* call trigger on the frontend - FE takes care of all BE triggers */ - dev_dbg(fe->dev, "ASoC: bespoke trigger FE %s cmd start\n", - fe->dai_link->name); - - ret = snd_soc_pcm_dai_bespoke_trigger(substream, SNDRV_PCM_TRIGGER_START); - if (ret < 0) - goto hw_free; - } else { - dev_dbg(fe->dev, "ASoC: trigger FE %s cmd start\n", - fe->dai_link->name); - - ret = dpcm_be_dai_trigger(fe, stream, - SNDRV_PCM_TRIGGER_START); - if (ret < 0) - goto hw_free; - } + ret = dpcm_be_dai_trigger(fe, stream, SNDRV_PCM_TRIGGER_START); + if (ret < 0) + goto hw_free; return 0; -- 2.43.0

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[PATCH v6 0/9] iio: adc: dfsdm: add scaling support
by Olivier Moysan 06 Aug '24

06 Aug '24

The aim of this serie is to add scaling support to STM32 DFSDM peripheral in the analog context. The DFSDM currently operates as a consumer of IIO channels provided by a generic SD modulator. As previously discussed in RFC [1], this topology is not suitable for implementing scaling. This series brings the integration of the DFSDM driver with the new IIO backend framework [2], enabling the DFSDM IIO device to offer scaling feature based on reference voltage data obtained from the IIO SD modulator backend. This generic SD modulator backend takes the place of the former SD modulator, used with legacy implementation. The DFSDM driver has been updated to adopt the generic ADC channel binding [3]. The reasons for this include: - Reducing the use of proprietary properties - Simplifying the coexistence of legacy and new backend bindings - Prepare the support of the MDF peripheral on STM32MP25 SoC Backward compatibility is maintained through legacy support. This series extends the backend framework with the following APIs: - iio_backend_read_raw: This API is intented to retrieve channel attributes from the backend. It is based on IIO framework read_raw API. - iio_backend_read_scale / iio_backend_read_offset Dedicated APIs to read scale and offset channel values. - iio_backend_disable / iio_backend_enable: backend enable/disable to be used for PM management - devm_iio_backend_fwnode_get Intended for parsing DT subnodes to allow generic channel binding support, as generic channel DT nodes are not populated as devices. [1]: https://lore.kernel.org/lkml/20200204101008.11411-1-olivier.moysan@st.com/ [2]: https://lore.kernel.org/all/20240206-iio-backend-v9-0-df66d159c000@analog.c… [3]: devicetree/bindings/iio/adc/adc.yaml Changes in v2: - Update enable/disable backend API - Rename devm_iio_backend_subnode_get(), as devm_iio_backend_fwnode_get() - Update iio_backend_read_raw() prototype to fully match IIO framework read_raw callback prototype. - Change st,adc-channel-type property name and type in DFSDM binding - Remove sd-backend and rename ads1201 compatibles in SD binding Changes in v3: - Add iio_backend_read_scale and iio_backend_read_offset - Remove sd-backend compatible. Adapt existing sd modulator driver and binding to support backend topology. - Misc corrections in DFSDM binding & minor changes in DFSDM driver Changes in v4: - Remove allOf and rework examples in SD modulator binding - Wrap lines to 80 characters in IIO code Changes in v5: - Move iio_backend_read_scale and iio_backend_read_offset to inline and update prototype - Fix channel number & remove legacy binding example in DFSDM binding - Rearrange DFSDM patches to fix build issue in serie - Split SD modulator binding patch. Move reference voltage change in a separate patch Changes in v6: - Minor cleanup in DFSDM binding - Fix vref-supply dependency condition in SD modulator binding Olivier Moysan (9): iio: add read scale and offset services to iio backend framework iio: add enable and disable services to iio backend framework iio: add child nodes support in iio backend framework dt-bindings: iio: dfsdm: move to backend framework dt-bindings: iio: add backend support to sd modulator dt-bindings: iio: add vref support to sd modulator iio: adc: stm32-dfsdm: adopt generic channels bindings iio: add iio backend support to sd modulator iio: adc: stm32-dfsdm: add scaling support to dfsdm .../iio/adc/sigma-delta-modulator.yaml | 25 +- .../bindings/iio/adc/st,stm32-dfsdm-adc.yaml | 122 +++++++- drivers/iio/adc/Kconfig | 2 + drivers/iio/adc/sd_adc_modulator.c | 92 +++++- drivers/iio/adc/stm32-dfsdm-adc.c | 293 +++++++++++++++--- drivers/iio/industrialio-backend.c | 101 +++++- include/linux/iio/backend.h | 29 +- 7 files changed, 592 insertions(+), 72 deletions(-) base-commit: bf829a98bc4bb9076a2fd70f4a69ea7e281863fa -- 2.25.1

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[PATCH] ALSA: usb-audio: Add input gain and master output mixer controls for RME Babyface Pro
by Stefan Stistrup 06 Aug '24

06 Aug '24

Add missing input gain and master output mixer controls for RME Babyface Pro This patch implements: 1. Input gain controls for 2 mic and 2 line inputs 2. Master output volume controls for all 12 output channels These additions allow for more complete control of the Babyface Pro under Linux. Signed-off-by: Stefan Stistrup <sstistrup(a)gmail.com> --- sound/usb/mixer_quirks.c | 162 ++++++++++++++++++++++++++++++++++++++- 1 file changed, 161 insertions(+), 1 deletion(-) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 065a4be0d771..26298e06b838 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -2541,14 +2541,23 @@ enum { #define SND_BBFPRO_CTL_REG2_PAD_AN1 4 #define SND_BBFPRO_CTL_REG2_PAD_AN2 5 -#define SND_BBFPRO_MIXER_IDX_MASK 0x1ff +#define SND_BBFPRO_MIXER_MAIN_OUT_CH_OFFSET 992 +#define SND_BBFPRO_MIXER_IDX_MASK 0x3ff #define SND_BBFPRO_MIXER_VAL_MASK 0x3ffff #define SND_BBFPRO_MIXER_VAL_SHIFT 9 #define SND_BBFPRO_MIXER_VAL_MIN 0 // -inf #define SND_BBFPRO_MIXER_VAL_MAX 65536 // +6dB +#define SND_BBFPRO_GAIN_CHANNEL_MASK 0x03 +#define SND_BBFPRO_GAIN_CHANNEL_SHIFT 7 +#define SND_BBFPRO_GAIN_VAL_MASK 0x7f +#define SND_BBFPRO_GAIN_VAL_MIN 0 +#define SND_BBFPRO_GAIN_VAL_MIC_MAX 65 +#define SND_BBFPRO_GAIN_VAL_LINE_MAX 18 // 9db in 0.5db incraments + #define SND_BBFPRO_USBREQ_CTL_REG1 0x10 #define SND_BBFPRO_USBREQ_CTL_REG2 0x17 +#define SND_BBFPRO_USBREQ_GAIN 0x1a #define SND_BBFPRO_USBREQ_MIXER 0x12 static int snd_bbfpro_ctl_update(struct usb_mixer_interface *mixer, u8 reg, @@ -2695,6 +2704,113 @@ static int snd_bbfpro_ctl_resume(struct usb_mixer_elem_list *list) return snd_bbfpro_ctl_update(list->mixer, reg, idx, value); } +static int snd_bbfpro_gain_update(struct usb_mixer_interface *mixer, + u8 channel, u8 gain) +{ + int err; + + struct snd_usb_audio *chip = mixer->chip; + + if (channel < 2) { + // XLR preamp: 3-bit fine, 5-bit coarse; special case >60 + if (gain < 60) + gain = ((gain % 3) << 5) | (gain / 3); + else + gain = ((gain % 6) << 5) | (60 / 3); + } + + err = snd_usb_lock_shutdown(chip); + if (err < 0) + return err; + + err = snd_usb_ctl_msg(chip->dev, + usb_sndctrlpipe(chip->dev, 0), + SND_BBFPRO_USBREQ_GAIN, + USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_DEVICE, + gain, channel, NULL, 0); + + snd_usb_unlock_shutdown(chip); + return err; +} + +static int snd_bbfpro_gain_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + int value = kcontrol->private_value & SND_BBFPRO_GAIN_VAL_MASK; + ucontrol->value.integer.value[0] = value; + return 0; +} + +static int snd_bbfpro_gain_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + int pv, channel; + + pv = kcontrol->private_value; + channel = (pv >> SND_BBFPRO_GAIN_CHANNEL_SHIFT) & + SND_BBFPRO_GAIN_CHANNEL_MASK; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = SND_BBFPRO_GAIN_VAL_MIN; + + if (channel < 2) { + uinfo->value.integer.max = SND_BBFPRO_GAIN_VAL_MIC_MAX; + } else { + uinfo->value.integer.max = SND_BBFPRO_GAIN_VAL_LINE_MAX; + } + + return 0; +} + +static int snd_bbfpro_gain_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + int pv, channel, old_value, value, err; + + struct usb_mixer_elem_list *list = snd_kcontrol_chip(kcontrol); + struct usb_mixer_interface *mixer = list->mixer; + + pv = kcontrol->private_value; + channel = (pv >> SND_BBFPRO_GAIN_CHANNEL_SHIFT) & + SND_BBFPRO_GAIN_CHANNEL_MASK; + old_value = pv & SND_BBFPRO_GAIN_VAL_MASK; + value = ucontrol->value.integer.value[0]; + + if (channel < 2) { + if (value > SND_BBFPRO_GAIN_VAL_MIC_MAX) + return -EINVAL; + } else { + if (value > SND_BBFPRO_GAIN_VAL_LINE_MAX) + return -EINVAL; + } + + if (value == old_value) + return 0; + + err = snd_bbfpro_gain_update(mixer, channel, value); + if (err < 0) + return err; + + kcontrol->private_value = + (channel << SND_BBFPRO_GAIN_CHANNEL_SHIFT) | value; + return 1; +} + +static int snd_bbfpro_gain_resume(struct usb_mixer_elem_list *list) +{ + int pv, channel, value; + + struct snd_kcontrol *kctl = list->kctl; + + pv = kctl->private_value; + channel = (pv >> SND_BBFPRO_GAIN_CHANNEL_SHIFT) & + SND_BBFPRO_GAIN_CHANNEL_MASK; + value = pv & SND_BBFPRO_GAIN_VAL_MASK; + + return snd_bbfpro_gain_update(list->mixer, channel, value); +} + static int snd_bbfpro_vol_update(struct usb_mixer_interface *mixer, u16 index, u32 value) { @@ -2790,6 +2906,15 @@ static const struct snd_kcontrol_new snd_bbfpro_ctl_control = { .put = snd_bbfpro_ctl_put }; +static const struct snd_kcontrol_new snd_bbfpro_gain_control = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .index = 0, + .info = snd_bbfpro_gain_info, + .get = snd_bbfpro_gain_get, + .put = snd_bbfpro_gain_put +}; + static const struct snd_kcontrol_new snd_bbfpro_vol_control = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, @@ -2813,6 +2938,18 @@ static int snd_bbfpro_ctl_add(struct usb_mixer_interface *mixer, u8 reg, &knew, NULL); } +static int snd_bbfpro_gain_add(struct usb_mixer_interface *mixer, u8 channel, + char *name) +{ + struct snd_kcontrol_new knew = snd_bbfpro_gain_control; + + knew.name = name; + knew.private_value = channel << SND_BBFPRO_GAIN_CHANNEL_SHIFT; + + return add_single_ctl_with_resume(mixer, 0, snd_bbfpro_gain_resume, + &knew, NULL); +} + static int snd_bbfpro_vol_add(struct usb_mixer_interface *mixer, u16 index, char *name) { @@ -2860,6 +2997,29 @@ static int snd_bbfpro_controls_create(struct usb_mixer_interface *mixer) } } + // Main out volume + for (i = 0 ; i < 12 ; ++i) { + snprintf(name, sizeof(name), "Main-Out %s", output[i]); + // Main outs are offset to 992 + err = snd_bbfpro_vol_add(mixer, + i + SND_BBFPRO_MIXER_MAIN_OUT_CH_OFFSET, + name); + if (err < 0) + return err; + } + + // Input gain + for (i = 0 ; i < 4 ; ++i) { + if (i < 2) { + snprintf(name, sizeof(name), "Mic-%s Gain", input[i]); + } else { + snprintf(name, sizeof(name), "Line-%s Gain", input[i]); + } + err = snd_bbfpro_gain_add(mixer, i, name); + if (err < 0) + return err; + } + // Control Reg 1 err = snd_bbfpro_ctl_add(mixer, SND_BBFPRO_CTL_REG1, SND_BBFPRO_CTL_REG1_CLK_OPTICAL, -- 2.45.2

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[RFC PATCH 0/6] ASoC: fsl: add memory to memory function for ASRC
by Shengjiu Wang 06 Aug '24

06 Aug '24

This function is base on the accelerator implementation for compress API: https://patchwork.kernel.org/project/alsa-devel/patch/20240731083843.59911-… Audio signal processing also has the requirement for memory to memory similar as Video. This asrc memory to memory (memory ->asrc->memory) case is a non real time use case. User fills the input buffer to the asrc module, after conversion, then asrc sends back the output buffer to user. So it is not a traditional ALSA playback and capture case. Because we had implemented the "memory -> asrc ->i2s device-> codec" use case in ALSA. Now the "memory->asrc->memory" needs to reuse the code in asrc driver, so the patch 1 and patch 2 is for refining the code to make it can be shared by the "memory->asrc->memory" driver. Other change is to add memory to memory support for two kinds of i.MX ASRC modules. Shengjiu Wang (6): ALSA: compress: add Sample Rate Converter codec support ASoC: fsl_asrc: define functions for memory to memory usage ASoC: fsl_easrc: define functions for memory to memory usage ASoC: fsl_asrc_m2m: Add memory to memory function ASoC: fsl_asrc: register m2m platform device ASoC: fsl_easrc: register m2m platform device include/uapi/sound/compress_offload.h | 2 + include/uapi/sound/compress_params.h | 9 +- sound/soc/fsl/Kconfig | 1 + sound/soc/fsl/Makefile | 2 +- sound/soc/fsl/fsl_asrc.c | 176 +++++- sound/soc/fsl/fsl_asrc.h | 2 + sound/soc/fsl/fsl_asrc_common.h | 68 +++ sound/soc/fsl/fsl_asrc_m2m.c | 769 ++++++++++++++++++++++++++ sound/soc/fsl/fsl_easrc.c | 259 ++++++++- sound/soc/fsl/fsl_easrc.h | 4 + 10 files changed, 1284 insertions(+), 8 deletions(-) create mode 100644 sound/soc/fsl/fsl_asrc_m2m.c -- 2.34.1

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