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[tiwai-sound:for-next 32/79] sound/pcmcia/vx/vxpocket.c:208:13: error: 'struct vx_core' has no member named 'dev'
by kernel test robot 08 Aug '24

08 Aug '24

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-next head: 9b88d0890ed9adab413ea991fac90842688e1017 commit: b426b3ba9f6fe17990893c5324727dd217d420b6 [32/79] ALSA: vx_core: Drop unused dev field config: i386-randconfig-141-20240808 (https://download.01.org/0day-ci/archive/20240809/202408090016.kW0TA6fc-lkp@…) compiler: gcc-12 (Debian 12.2.0-14) 12.2.0 reproduce (this is a W=1 build): (https://download.01.org/0day-ci/archive/20240809/202408090016.kW0TA6fc-lkp@…) If you fix the issue in a separate patch/commit (i.e. not just a new version of the same patch/commit), kindly add following tags | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202408090016.kW0TA6fc-lkp@intel.com/ All errors (new ones prefixed by >>): sound/pcmcia/vx/vxpocket.c: In function 'vxpocket_config': >> sound/pcmcia/vx/vxpocket.c:208:13: error: 'struct vx_core' has no member named 'dev' 208 | chip->dev = &link->dev; | ^~ vim +208 sound/pcmcia/vx/vxpocket.c 6d00a3127972e7 Takashi Iwai 2005-06-30 170 6d00a3127972e7 Takashi Iwai 2005-06-30 171 6d00a3127972e7 Takashi Iwai 2005-06-30 172 /* 6d00a3127972e7 Takashi Iwai 2005-06-30 173 * configuration callback 6d00a3127972e7 Takashi Iwai 2005-06-30 174 */ 6d00a3127972e7 Takashi Iwai 2005-06-30 175 15b99ac1729503 Dominik Brodowski 2006-03-31 176 static int vxpocket_config(struct pcmcia_device *link) 6d00a3127972e7 Takashi Iwai 2005-06-30 177 { af26367f69a474 Takashi Iwai 2005-11-17 178 struct vx_core *chip = link->priv; 7c5af6ffd69bb2 Dominik Brodowski 2009-10-24 179 int ret; 6d00a3127972e7 Takashi Iwai 2005-06-30 180 6d00a3127972e7 Takashi Iwai 2005-06-30 181 snd_printdd(KERN_DEBUG "vxpocket_config called\n"); 6d00a3127972e7 Takashi Iwai 2005-06-30 182 6d00a3127972e7 Takashi Iwai 2005-06-30 183 /* redefine hardware record according to the VERSION1 string */ af2b3b503ad1b0 Dominik Brodowski 2006-10-25 184 if (!strcmp(link->prod_id[1], "VX-POCKET")) { 6d00a3127972e7 Takashi Iwai 2005-06-30 185 snd_printdd("VX-pocket is detected\n"); 6d00a3127972e7 Takashi Iwai 2005-06-30 186 } else { 6d00a3127972e7 Takashi Iwai 2005-06-30 187 snd_printdd("VX-pocket 440 is detected\n"); 6d00a3127972e7 Takashi Iwai 2005-06-30 188 /* overwrite the hardware information */ 6d00a3127972e7 Takashi Iwai 2005-06-30 189 chip->hw = &vxp440_hw; 6d00a3127972e7 Takashi Iwai 2005-06-30 190 chip->type = vxp440_hw.type; 6d00a3127972e7 Takashi Iwai 2005-06-30 191 strcpy(chip->card->driver, vxp440_hw.name); 6d00a3127972e7 Takashi Iwai 2005-06-30 192 } 6d00a3127972e7 Takashi Iwai 2005-06-30 193 90abdc3b973229 Dominik Brodowski 2010-07-24 194 ret = pcmcia_request_io(link); 7c5af6ffd69bb2 Dominik Brodowski 2009-10-24 195 if (ret) db0a5214b8d6cc Takashi Iwai 2014-09-09 196 goto failed_preirq; 7c5af6ffd69bb2 Dominik Brodowski 2009-10-24 197 db0a5214b8d6cc Takashi Iwai 2014-09-09 198 ret = request_threaded_irq(link->irq, snd_vx_irq_handler, db0a5214b8d6cc Takashi Iwai 2014-09-09 199 snd_vx_threaded_irq_handler, db0a5214b8d6cc Takashi Iwai 2014-09-09 200 IRQF_SHARED, link->devname, link->priv); 7c5af6ffd69bb2 Dominik Brodowski 2009-10-24 201 if (ret) db0a5214b8d6cc Takashi Iwai 2014-09-09 202 goto failed_preirq; 7c5af6ffd69bb2 Dominik Brodowski 2009-10-24 203 1ac71e5a35eebe Dominik Brodowski 2010-07-29 204 ret = pcmcia_enable_device(link); 7c5af6ffd69bb2 Dominik Brodowski 2009-10-24 205 if (ret) 7c5af6ffd69bb2 Dominik Brodowski 2009-10-24 206 goto failed; 6d00a3127972e7 Takashi Iwai 2005-06-30 207 dd2e5a156525f1 Dominik Brodowski 2009-11-03 @208 chip->dev = &link->dev; 6d00a3127972e7 Takashi Iwai 2005-06-30 209 9a017a910346af Dominik Brodowski 2010-07-24 210 if (snd_vxpocket_assign_resources(chip, link->resource[0]->start, 9a017a910346af Dominik Brodowski 2010-07-24 211 link->irq) < 0) 6d00a3127972e7 Takashi Iwai 2005-06-30 212 goto failed; 6d00a3127972e7 Takashi Iwai 2005-06-30 213 79ca4f3f625e14 Takashi Iwai 2006-04-12 214 return 0; 6d00a3127972e7 Takashi Iwai 2005-06-30 215 6d00a3127972e7 Takashi Iwai 2005-06-30 216 failed: db0a5214b8d6cc Takashi Iwai 2014-09-09 217 free_irq(link->irq, link->priv); db0a5214b8d6cc Takashi Iwai 2014-09-09 218 failed_preirq: fba395eee7d3f3 Dominik Brodowski 2006-03-31 219 pcmcia_disable_device(link); 15b99ac1729503 Dominik Brodowski 2006-03-31 220 return -ENODEV; 6d00a3127972e7 Takashi Iwai 2005-06-30 221 } 6d00a3127972e7 Takashi Iwai 2005-06-30 222 :::::: The code at line 208 was first introduced by commit :::::: dd2e5a156525f11754d9b1e0583f6bb49c253d62 pcmcia: remove deprecated handle_to_dev() macro :::::: TO: Dominik Brodowski <linux(a)dominikbrodowski.net> :::::: CC: Dominik Brodowski <linux(a)dominikbrodowski.net> -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

1 0

[PATCH 0/5] ASoC/SOF/PCI/Intel: add PantherLake support
by Pierre-Louis Bossart 08 Aug '24

08 Aug '24

Add initial support for the PantherLake platform, and initial ACPI configurations. This patchset depends on the first patch of "[PATCH 0/3] ALSA/PCI: add PantherLake audio support" Bard Liao (2): ASoC: Intel: soc-acpi-intel-ptl-match: add rt711-sdca table ASoC: Intel: soc-acpi-intel-ptl-match: Add rt722 support Fred Oh (1): ASoC: SOF: Intel: add PTL specific power control register Pierre-Louis Bossart (2): ASoC: SOF: Intel: add initial support for PTL ASoC: Intel: soc-acpi: add PTL match tables include/sound/soc-acpi-intel-match.h | 2 + sound/soc/intel/common/Makefile | 1 + .../intel/common/soc-acpi-intel-ptl-match.c | 121 ++++++++++++++++++ sound/soc/sof/intel/Kconfig | 17 +++ sound/soc/sof/intel/Makefile | 2 + sound/soc/sof/intel/hda-dsp.c | 1 + sound/soc/sof/intel/hda.h | 1 + sound/soc/sof/intel/lnl.c | 27 ++++ sound/soc/sof/intel/mtl.c | 16 ++- sound/soc/sof/intel/mtl.h | 2 + sound/soc/sof/intel/pci-ptl.c | 77 +++++++++++ sound/soc/sof/intel/shim.h | 1 + 12 files changed, 266 insertions(+), 2 deletions(-) create mode 100644 sound/soc/intel/common/soc-acpi-intel-ptl-match.c create mode 100644 sound/soc/sof/intel/pci-ptl.c -- 2.43.0

2 9

[PATCH 2/2] ASoC: cs35l56: Stop creating ALSA controls for firmware coefficients
by Simon Trimmer 07 Aug '24

07 Aug '24

A number of laptops have gone to market with old firmware versions that export controls that have since been hidden, but we can't just install a newer firmware because the firmware for each product is customized and qualified by the OEM. The issue is that alsactl save and restore has no idea what controls are good to persist which can lead to misconfiguration. There is no reason that the UCM or user should need to interact with any of the ALSA controls for the firmware coefficients so they can be removed entirely. Fixes: e49611252900 ("ASoC: cs35l56: Add driver for Cirrus Logic CS35L56") Signed-off-by: Simon Trimmer <simont(a)opensource.cirrus.com> --- sound/soc/codecs/cs35l56.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/codecs/cs35l56.c b/sound/soc/codecs/cs35l56.c index 84c34f5b1a51..757ade6373ed 100644 --- a/sound/soc/codecs/cs35l56.c +++ b/sound/soc/codecs/cs35l56.c @@ -1095,6 +1095,11 @@ int cs35l56_system_resume(struct device *dev) } EXPORT_SYMBOL_GPL(cs35l56_system_resume); +static int cs35l56_control_add_nop(struct wm_adsp *dsp, struct cs_dsp_coeff_ctl *cs_ctl) +{ + return 0; +} + static int cs35l56_dsp_init(struct cs35l56_private *cs35l56) { struct wm_adsp *dsp; @@ -1117,6 +1122,12 @@ static int cs35l56_dsp_init(struct cs35l56_private *cs35l56) dsp->fw = 12; dsp->wmfw_optional = true; + /* + * None of the firmware controls need to be exported so add a no-op + * callback that suppresses creating an ALSA control. + */ + dsp->control_add = &cs35l56_control_add_nop; + dev_dbg(cs35l56->base.dev, "DSP system name: '%s'\n", dsp->system_name); ret = wm_halo_init(dsp); -- 2.43.0

1 0

[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

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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

2 3