September 2024 - Alsa-devel - mailman.alsa-project.org

[PATCH] ASoC: tlv320aic31xx: Add support for loading filter coefficients
by Romain Gantois 09 Sep '24

09 Sep '24

The TLV320DAC3100 Audio DAC has 25 built-in digital audio processing blocks (PRBs). Each of these PRBs has a static filter structure with programmable coefficients. Once a PRB is selected for use by the DAC, its filter coefficients can be configured via a dedicated set of registers. Define a new optional firmware which can be loaded by the TLV320DAC driver. This firmware describes a full set of filter coefficients for all blocks used by the various PRBs. The firmware's binary format is heavily inspired by the one used in the peb2466 driver. It includes a version marker to allow for potential evolutions of the format. Note that adaptive filtering is not supported i.e. filter coefficients are loaded once before power-on and then cannot be changed while the DAC is powered. This is why only page A coefficients are modified. Page B coefficients are only used for adaptive filtering. Signed-off-by: Romain Gantois <romain.gantois(a)bootlin.com> --- sound/soc/codecs/tlv320aic31xx.c | 100 +++++++++++++++++++++++++++++++++++++++ 1 file changed, 100 insertions(+) diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index 2f94cfda0e33..7e624c4b77b6 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -12,6 +12,7 @@ * and mono/stereo Class-D speaker driver. */ +#include <asm/unaligned.h> #include <linux/module.h> #include <linux/moduleparam.h> #include <linux/init.h> @@ -22,6 +23,7 @@ #include <linux/gpio/consumer.h> #include <linux/regulator/consumer.h> #include <linux/acpi.h> +#include <linux/firmware.h> #include <linux/of.h> #include <linux/slab.h> #include <sound/core.h> @@ -1638,6 +1640,98 @@ static const struct i2c_device_id aic31xx_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, aic31xx_i2c_id); +static int tlv320dac3100_fw_load(struct aic31xx_priv *aic31xx, + const u8 *data, size_t size) +{ + int ret, reg; + u16 val16; + + /* + * Coefficients firmware binary structure. Multi-byte values are big-endian. + * + * @0, 16bits: Magic (0xB30C) + * @2, 16bits: Version (0x0100 for version 1.0) + * @4, 8bits: DAC Processing Block Selection + * @5, 62 16-bit values: Page 8 buffer A DAC programmable filter coefficients + * @129, 12 16-bit values: Page 9 Buffer A DAC programmable filter coefficients + * + * Filter coefficients are interpreted as two's complement values + * ranging from -32 768 to 32 767. For more details on filter coefficients, + * please refer to the TLV320DAC3100 datasheet, tables 6-120 and 6-123. + */ + + if (size != 153) { + dev_err(aic31xx->dev, "firmware size is %zu, expected 153 bytes\n", size); + return -EINVAL; + } + + /* Check magic */ + val16 = get_unaligned_be16(data); + if (val16 != 0xb30c) { + dev_err(aic31xx->dev, "fw magic is 0x%04x expected 0xb30c\n", val16); + return -EINVAL; + } + data += 2; + + /* Check version */ + val16 = get_unaligned_be16(data); + if (val16 != 0x0100) { + dev_err(aic31xx->dev, "invalid firmware version 0x%04x! expected 1", val16); + return -EINVAL; + } + data += 2; + + ret = regmap_write(aic31xx->regmap, AIC31XX_DACPRB, *data); + if (ret) { + dev_err(aic31xx->dev, "failed to write PRB index: err %d\n", ret); + return ret; + } + data += 1; + + /* Page 8 Buffer A coefficients */ + for (reg = 2; reg < 126; reg++) { + ret = regmap_write(aic31xx->regmap, AIC31XX_REG(8, reg), *data); + if (ret) { + dev_err(aic31xx->dev, + "failed to write page 8 filter coefficient %d: err %d\n", reg, ret); + return ret; + } + data++; + } + + /* Page 9 Buffer A coefficients */ + for (reg = 2; reg < 26; reg++) { + ret = regmap_write(aic31xx->regmap, AIC31XX_REG(9, reg), *data); + if (ret) { + dev_err(aic31xx->dev, + "failed to write page 9 filter coefficient %d: err %d\n", reg, ret); + return ret; + } + data++; + } + + dev_info(aic31xx->dev, "done loading DAC filter coefficients\n"); + + return ret; +} + +static int tlv320dac3100_load_coeffs(struct aic31xx_priv *aic31xx, + const char *fw_name) +{ + const struct firmware *fw; + int ret; + + ret = request_firmware(&fw, fw_name, aic31xx->dev); + if (ret) + return ret; + + ret = tlv320dac3100_fw_load(aic31xx, fw->data, fw->size); + + release_firmware(fw); + + return ret; +} + static int aic31xx_i2c_probe(struct i2c_client *i2c) { struct aic31xx_priv *aic31xx; @@ -1727,6 +1821,12 @@ static int aic31xx_i2c_probe(struct i2c_client *i2c) } } + if (aic31xx->codec_type == DAC3100) { + ret = tlv320dac3100_load_coeffs(aic31xx, "tlv320dac3100-coeffs.bin"); + if (ret) + dev_warn(aic31xx->dev, "Did not load any filter coefficients\n"); + } + if (aic31xx->codec_type & DAC31XX_BIT) return devm_snd_soc_register_component(&i2c->dev, &soc_codec_driver_aic31xx, --- base-commit: 8400291e289ee6b2bf9779ff1c83a291501f017b change-id: 20240806-tlv320-filter-189edd66496c Best regards, -- Romain Gantois <romain.gantois(a)bootlin.com>

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[PATCH] soundwire: stream: Revert "soundwire: stream: fix programming slave ports for non-continous port maps"
by Krzysztof Kozlowski 09 Sep '24

09 Sep '24

This reverts commit ab8d66d132bc8f1992d3eb6cab8d32dda6733c84 because it breaks codecs using non-continuous masks in source and sink ports. The commit missed the point that port numbers are not used as indices for iterating over prop.sink_ports or prop.source_ports. Soundwire core and existing codecs expect that the array passed as prop.sink_ports and prop.source_ports is continuous. The port mask still might be non-continuous, but that's unrelated. Reported-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Closes: https://lore.kernel.org/all/b6c75eee-761d-44c8-8413-2a5b34ee2f98@linux.inte… Fixes: ab8d66d132bc ("soundwire: stream: fix programming slave ports for non-continous port maps") Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- I will need to fix my codec drivers, but that's independent. --- drivers/soundwire/stream.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/drivers/soundwire/stream.c b/drivers/soundwire/stream.c index f275143d7b18..7aa4900dcf31 100644 --- a/drivers/soundwire/stream.c +++ b/drivers/soundwire/stream.c @@ -1291,18 +1291,18 @@ struct sdw_dpn_prop *sdw_get_slave_dpn_prop(struct sdw_slave *slave, unsigned int port_num) { struct sdw_dpn_prop *dpn_prop; - unsigned long mask; + u8 num_ports; int i; if (direction == SDW_DATA_DIR_TX) { - mask = slave->prop.source_ports; + num_ports = hweight32(slave->prop.source_ports); dpn_prop = slave->prop.src_dpn_prop; } else { - mask = slave->prop.sink_ports; + num_ports = hweight32(slave->prop.sink_ports); dpn_prop = slave->prop.sink_dpn_prop; } - for_each_set_bit(i, &mask, 32) { + for (i = 0; i < num_ports; i++) { if (dpn_prop[i].num == port_num) return &dpn_prop[i]; } -- 2.43.0

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[PATCH] ASoC: tas2781: fix to save the dsp bin file name into the correct array in case name_prefix is not NULL
by Shenghao Ding 09 Sep '24

09 Sep '24

fix to save the dsp bin file name into the correct array, coef_binaryname, instead of rca_binaryname, in case name_prefix is not NULL. Fixes: 55f0a1fa6ea4 ("ASoC: tas2781: Add name_prefix as the prefix name of DSP firmwares and calibrated data files") Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- sound/soc/codecs/tas2781-i2c.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index 59fcce7fef7b..3bc10d615fd3 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -498,7 +498,7 @@ static void tasdevice_fw_ready(const struct firmware *fmw, */ tas_priv->fw_state = TASDEVICE_RCA_FW_OK; if (tas_priv->name_prefix) - scnprintf(tas_priv->rca_binaryname, 64, "%s-%s_coef.bin", + scnprintf(tas_priv->coef_binaryname, 64, "%s-%s_coef.bin", tas_priv->name_prefix, tas_priv->dev_name); else scnprintf(tas_priv->coef_binaryname, 64, "%s_coef.bin", -- 2.34.1

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[PATCH] soundwire: stream: fix programming slave ports for non-continous port maps
by Krzysztof Kozlowski 09 Sep '24

09 Sep '24

Two bitmasks in 'struct sdw_slave_prop' - 'source_ports' and 'sink_ports' - define which ports to program in sdw_program_slave_port_params(). The masks are used to get the appropriate data port properties ('struct sdw_get_slave_dpn_prop') from an array. Bitmasks can be non-continuous or can start from index different than 0, thus when looking for matching port property for given port, we must iterate over mask bits, not from 0 up to number of ports. This fixes allocation and programming slave ports, when a source or sink masks start from further index. Fixes: f8101c74aa54 ("soundwire: Add Master and Slave port programming") Cc: <stable(a)vger.kernel.org> Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- drivers/soundwire/stream.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/drivers/soundwire/stream.c b/drivers/soundwire/stream.c index 7aa4900dcf31..f275143d7b18 100644 --- a/drivers/soundwire/stream.c +++ b/drivers/soundwire/stream.c @@ -1291,18 +1291,18 @@ struct sdw_dpn_prop *sdw_get_slave_dpn_prop(struct sdw_slave *slave, unsigned int port_num) { struct sdw_dpn_prop *dpn_prop; - u8 num_ports; + unsigned long mask; int i; if (direction == SDW_DATA_DIR_TX) { - num_ports = hweight32(slave->prop.source_ports); + mask = slave->prop.source_ports; dpn_prop = slave->prop.src_dpn_prop; } else { - num_ports = hweight32(slave->prop.sink_ports); + mask = slave->prop.sink_ports; dpn_prop = slave->prop.sink_dpn_prop; } - for (i = 0; i < num_ports; i++) { + for_each_set_bit(i, &mask, 32) { if (dpn_prop[i].num == port_num) return &dpn_prop[i]; } -- 2.43.0

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[PATCH v2] ASoC: dt-bindings: renesas,rsnd: add post-init-providers property
by Kuninori Morimoto 09 Sep '24

09 Sep '24

At least if rsnd is using DPCM connection with Audio-Graph-Card2, fw_devlink might doesn't have enough information to break the cycle (Same problem might occur with Multi-CPU/Codec or Codec2Codec). In such case, rsnd driver will not be probed. Add post-init-providers support to break the link cycle. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- v1 -> v2 - remove basic description/type - add detail description - update git-log Documentation/devicetree/bindings/sound/renesas,rsnd.yaml | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml b/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml index 07ec6247d9def..458877ce4aa42 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml @@ -112,6 +112,13 @@ properties: description: List of necessary clock names. # details are defined below + post-init-providers: + description: | + At least if rsnd is using DPCM connection with Audio-Graph-Card2, fw_devlink might doesn't + have enough information to break the cycle. rsnd driver will not be probed in such case. + Add post-init-providers property to indicate which link in the cycle to break. + Same problem might occur with Multi-CPU/Codec or Codec2Codec. + # ports is below port: $ref: audio-graph-port.yaml#/definitions/port-base -- 2.43.0

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How to handle complex Codec2Codec
by Kuninori Morimoto 09 Sep '24

09 Sep '24

Hi ALSA ML ALSA SoC is supporting Codec2Codec connection, and Audio Graph Card2 is supporting it. I think its image is... +------+ +--- |Codec1| <-- | +------+ | +------+ +--> |Codec2| --> +------+ This is OK, but how to handle below ? +----------------------+ |CPU | +------+ | +--[rate convert]<--| <-- |Codec1| <-- | | | +------+ | | | +------+ | +------------------>| --> |Codec2| --> +----------------------+ +------+ I think there are 2 issues. (1) How to handle [CPU] part on Codec2Codec ? (2) How to handle [rate convert] on Codec2Codec ? I guess (1) can be handled if [CPU] was worked as platform, but do you have any other idea/solution for it ? I think (2) [rate convert] needs DPCM connection, but how to use it with Codec2Codec ? Or, can we handle Codec2Codec via DPCM somehow ?? FE BE +------+ +------+ --> |Codec1| -> |Codec2| -> +------+ +------+ If we can do this, both (1) and (2) can be solved, like below ? In such case, [Codec1] need to work as [capture] when playback. But I'm not sure how to implement it... FE BE +------+ +-----------------+ +-----+ +------+ --> |Codec1|-|CPU[rate convert]| -> |dummy|-|Codec2| -> +------+ +-----------------+ +-----+ +------+ Thank you for your help !! Best regards --- Kuninori Morimoto

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Re: [PATCH] ALSA: Add logitech Audio profile quirk
by Joshua Pius 06 Sep '24

06 Sep '24

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[PATCH] ASoC: codecs: Use devm_clk_get_enabled() helpers
by ying zuxin 06 Sep '24

06 Sep '24

From: ying zuxin <11154159(a)vivo.com> The devm_clk_get_enabled() helpers: - call devm_clk_get() - call clk_prepare_enable() and register what is needed in order to call clk_disable_unprepare() when needed, as a managed resource. This simplifies the code and avoids the calls to clk_disable_unprepare(). Signed-off-by: ying zuxin <11154159(a)vivo.com> --- sound/soc/codecs/peb2466.c | 14 +------------- 1 file changed, 1 insertion(+), 13 deletions(-) diff --git a/sound/soc/codecs/peb2466.c b/sound/soc/codecs/peb2466.c index 76ee7e3f4d9b..b67cfad4fc32 100644 --- a/sound/soc/codecs/peb2466.c +++ b/sound/soc/codecs/peb2466.c @@ -1975,12 +1975,9 @@ static int peb2466_spi_probe(struct spi_device *spi) if (IS_ERR(peb2466->reset_gpio)) return PTR_ERR(peb2466->reset_gpio); - peb2466->mclk = devm_clk_get(&peb2466->spi->dev, "mclk"); + peb2466->mclk = devm_clk_get_enabled(&peb2466->spi->dev, "mclk"); if (IS_ERR(peb2466->mclk)) return PTR_ERR(peb2466->mclk); - ret = clk_prepare_enable(peb2466->mclk); - if (ret) - return ret; if (peb2466->reset_gpio) { gpiod_set_value_cansleep(peb2466->reset_gpio, 1); @@ -2031,17 +2028,9 @@ static int peb2466_spi_probe(struct spi_device *spi) return 0; failed: - clk_disable_unprepare(peb2466->mclk); return ret; } -static void peb2466_spi_remove(struct spi_device *spi) -{ - struct peb2466 *peb2466 = spi_get_drvdata(spi); - - clk_disable_unprepare(peb2466->mclk); -} - static const struct of_device_id peb2466_of_match[] = { { .compatible = "infineon,peb2466", }, { } @@ -2061,7 +2050,6 @@ static struct spi_driver peb2466_spi_driver = { }, .id_table = peb2466_id_table, .probe = peb2466_spi_probe, - .remove = peb2466_spi_remove, }; module_spi_driver(peb2466_spi_driver); -- 2.39.0

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[PATCH] ASoC: makes rtd->initialized bit field
by Kuninori Morimoto 06 Sep '24

06 Sep '24

rtd->initialized is used to know whether soc_init_pcm_runtime() was correctly fined, and used to call snd_soc_link_exit(). We don't need to have it as bool, let's make it bit-field same as other flags. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Cc: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> Cc: Cezary Rojewski <cezary.rojewski(a)intel.com> --- include/sound/soc.h | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 81ef380b2e06..e6e359c1a2ac 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1207,8 +1207,7 @@ struct snd_soc_pcm_runtime { /* bit field */ unsigned int pop_wait:1; unsigned int fe_compr:1; /* for Dynamic PCM */ - - bool initialized; + unsigned int initialized:1; /* CPU/Codec/Platform */ int num_components; -- 2.43.0

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[PATCH 1/2] pcm: dmix: Fix resume with multiple instances
by Takashi Iwai 06 Sep '24

06 Sep '24

The fix for PCM dmix suspend/resume checks spcm->info bit of SND_PCM_INFO_RESUME for applying a workaround for drivers with the full resume support. This assumed that scpm->info is exposed from the underlying slave PCM device. The above is true for the first opened instance, but for the second opened instance, it's a copy from the saved data in shmem. And, we dropped SND_PCM_INFO_RESUME bit there to assure not to expose the full resume capability to applications. This resulted in the inconsistencies, and when the second instance is resumed at first, it misses the snd_pcm_resume() call, hence the driver doesn't react properly any longer. For addressing it, we keep SND_PCM_INFO_RESUME bit in shmptr->s.info bits as is, while dropping the bit exposed to apps in snd_pcm_direct_hw_refine() and *_hw_params() callbacks. Fixes: 6d1d620eadf3 ("pcm: dmix: resume workaround for buggy driver") Reported-and-tested-by: Chancel Liu <chancel.liu(a)nxp.com> Closes: https://lore.kernel.org/DB9PR04MB94988752ED7C43B399E0BC00E3942@DB9PR04MB949… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- src/pcm/pcm_direct.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/src/pcm/pcm_direct.c b/src/pcm/pcm_direct.c index e53e59238119..17e677f60b19 100644 --- a/src/pcm/pcm_direct.c +++ b/src/pcm/pcm_direct.c @@ -1018,6 +1018,7 @@ int snd_pcm_direct_hw_refine(snd_pcm_t *pcm, snd_pcm_hw_params_t *params) } dshare->timer_ticks = hw_param_interval(params, SND_PCM_HW_PARAM_PERIOD_SIZE)->max / dshare->slave_period_size; params->info = dshare->shmptr->s.info; + params->info &= ~SND_PCM_INFO_RESUME; #ifdef REFINE_DEBUG snd_output_puts(log, "DMIX REFINE (end):\n"); snd_pcm_hw_params_dump(params, log); @@ -1031,6 +1032,7 @@ int snd_pcm_direct_hw_params(snd_pcm_t *pcm, snd_pcm_hw_params_t * params) snd_pcm_direct_t *dmix = pcm->private_data; params->info = dmix->shmptr->s.info; + params->info &= ~SND_PCM_INFO_RESUME; params->rate_num = dmix->shmptr->s.rate; params->rate_den = 1; params->fifo_size = 0; @@ -1183,8 +1185,6 @@ static void save_slave_setting(snd_pcm_direct_t *dmix, snd_pcm_t *spcm) COPY_SLAVE(buffer_time); COPY_SLAVE(sample_bits); COPY_SLAVE(frame_bits); - - dmix->shmptr->s.info &= ~SND_PCM_INFO_RESUME; } #undef COPY_SLAVE -- 2.43.0

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Suspend/resume Issue on pcm_dmix.c in alsa-lib
by Chancel Liu 06 Sep '24

06 Sep '24

Hi Takashi Iwai, Jaroslav Kysela We found an issue on dmix in alsa-lib when do suspend and resume. It can be easily reproduced by following steps: 1. Run two dmix clients in parallel. (Only one client doesn't has such issue) ~# aplay xxx1.wav & ~# aplay xxx2.wav & Here I attach the asound.conf we're using. ~# cat /etc/asound.conf defaults.pcm.rate_converter "linear" pcm.dmix_44100{ type dmix ipc_key 5678293 ipc_key_add_uid yes slave{ pcm "hw:0,0" period_time 40000 format S16_LE rate 44100 } } pcm.asymed{ type asym playback.pcm "dmix_44100" capture.pcm "dsnoop_44100" } pcm.!default{ type plug route_policy "average" slave.pcm "asymed" } 2. Let linux enter into suspend and then resume(Repeat this step if not reproduced) 3. After resume, aplay will get stuck in snd_pcm_wait(). The GDB shows: (gdb) bt #0 0x0000fffff7da9264 in __GI___poll (fds=fds@entry=0xfffffffff480, nfds=nfds@entry=1, timeout=timeout@entry=240) at /usr/src/debug/glibc/2.39+git/sysdeps/unix/sysv/linux/poll.c:41 #1 0x0000fffff7edf468 in poll (__timeout=240, __nfds=1, __fds=0xfffffffff480) #2 snd1_pcm_wait_nocheck (pcm=pcm@entry=0xaaaaaaad2cb0, timeout=240, timeout@entry=-10001) at pcm.c:2993 #3 0x0000fffff7ee54a0 in snd1_pcm_write_areas (pcm=pcm@entry=0xaaaaaaad2cb0, areas=areas@entry=0xfffffffff560, offset=<optimized out>, offset@entry=0, size=<optimized out>, size@entry=1768, func=func@entry=0xfffff7ef5190 <snd_pcm_plugin_write_areas>) at pcm.c:7699 #4 0x0000fffff7ef5020 in snd_pcm_plugin_writei (pcm=0xaaaaaaad2cb0, buffer=<optimized out>, size=1768) at pcm_plugin.c:354 It seems that sometimes after suspend and resume there's no available space for data written into buffer. Then aplay keeps stuck in snd_pcm_wait(). I checked the hw_ptr of dmix and found that hw_ptr is always 0 after resume. I don't have a solution now so I turn to you for help. The version of alsa-lib is v1.2.11. Could you please help check it? Regards, Chancel Liu

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[GIT PULL] ASoC fixes for v6.11-rc6
by Mark Brown 06 Sep '24

06 Sep '24

The following changes since commit bb4485562f5907708f1c218b5d70dce04165d1e1: ASoC: cs-amp-lib: Ignore empty UEFI calibration entries (2024-08-22 17:52:26 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.11-rc6 for you to fetch changes up to 77212f300bfd6fb3edaabd1daf863cabb521854a: ASoC: codecs: lpass-va-macro: set the default codec version for sm8250 (2024-08-30 13:54:24 +0100) ---------------------------------------------------------------- ASoC: Fixes for v6.11 A larger set of fixes than I'd like at this point, but mainly due to people working on fixing module autoloading by adding missing exports of ID tables rather than anything particularly concerning. There are some other runtime fixes and quirks, and a tweak to the ABI definition for SOF which ensures that a struct layout doesn't vary depending on the architecture of the host. ---------------------------------------------------------------- Chen-Yu Tsai (1): ASoc: SOF: topology: Clear SOF link platform name upon unload Hans de Goede (2): ASoC: Intel: Boards: Fix NULL pointer deref in BYT/CHT boards harder ASoC: Intel: soc-acpi-cht: Make Lenovo Yoga Tab 3 X90F DMI match less strict Laurentiu Mihalcea (1): ASoC: SOF: ipc: replace "enum sof_comp_type" field with "uint32_t" Liao Chen (4): ASoC: intel: fix module autoloading ASoC: google: fix module autoloading ASoC: tda7419: fix module autoloading ASoC: fix module autoloading Mark Brown (1): ASoC: fix module autoloading Markuss Broks (1): ASoC: amd: yc: Add a quirk for MSI Bravo 17 (D7VEK) Matteo Martelli (1): ASoC: sunxi: sun4i-i2s: fix LRCLK polarity in i2s mode Mohan Kumar (1): ASoC: tegra: Fix CBB error during probe() Srinivas Kandagatla (1): ASoC: codecs: lpass-va-macro: set the default codec version for sm8250 Zhang Yi (1): ASoC: mediatek: mt8188-mt6359: Modify key robelin (1): ASoC: dapm: Fix UAF for snd_soc_pcm_runtime object include/sound/sof/topology.h | 2 +- include/uapi/sound/sof/abi.h | 2 +- sound/soc/amd/yc/acp6x-mach.c | 7 ++ sound/soc/codecs/chv3-codec.c | 1 + sound/soc/codecs/lpass-va-macro.c | 11 +- sound/soc/codecs/tda7419.c | 1 + sound/soc/google/chv3-i2s.c | 1 + sound/soc/intel/boards/bxt_rt298.c | 2 +- sound/soc/intel/boards/bytcht_cx2072x.c | 2 +- sound/soc/intel/boards/bytcht_da7213.c | 2 +- sound/soc/intel/boards/bytcht_es8316.c | 2 +- sound/soc/intel/boards/bytcr_rt5640.c | 2 +- sound/soc/intel/boards/bytcr_rt5651.c | 2 +- sound/soc/intel/boards/bytcr_wm5102.c | 2 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 2 +- sound/soc/intel/boards/cht_bsw_rt5672.c | 2 +- sound/soc/intel/common/soc-acpi-intel-cht-match.c | 1 - sound/soc/intel/keembay/kmb_platform.c | 1 + sound/soc/mediatek/mt8188/mt8188-mt6359.c | 17 ++- sound/soc/soc-dapm.c | 1 + sound/soc/sof/topology.c | 2 + sound/soc/sunxi/sun4i-i2s.c | 143 +++++++++++----------- sound/soc/tegra/tegra210_ahub.c | 10 +- 23 files changed, 127 insertions(+), 91 deletions(-)

2 1

[PATCH] ASoC: Intel: skl_hda_dsp_generic: convert comma to semicolon
by Chen Ni 05 Sep '24

05 Sep '24

Replace comma between expressions with semicolons. Using a ',' in place of a ';' can have unintended side effects. Although that is not the case here, it is seems best to use ';' unless ',' is intended. Found by inspection. No functional change intended. Compile tested only. Signed-off-by: Chen Ni <nichen(a)iscas.ac.cn> --- sound/soc/intel/boards/skl_hda_dsp_generic.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/intel/boards/skl_hda_dsp_generic.c b/sound/soc/intel/boards/skl_hda_dsp_generic.c index 225867bb3310..8e104874d58c 100644 --- a/sound/soc/intel/boards/skl_hda_dsp_generic.c +++ b/sound/soc/intel/boards/skl_hda_dsp_generic.c @@ -217,14 +217,14 @@ static int skl_hda_audio_probe(struct platform_device *pdev) return -ENOMEM; card = &ctx->card; - card->name = "hda-dsp", - card->owner = THIS_MODULE, - card->dai_link = skl_hda_be_dai_links, - card->dapm_widgets = skl_hda_widgets, - card->dapm_routes = skl_hda_map, - card->add_dai_link = skl_hda_add_dai_link, - card->fully_routed = true, - card->late_probe = skl_hda_card_late_probe, + card->name = "hda-dsp"; + card->owner = THIS_MODULE; + card->dai_link = skl_hda_be_dai_links; + card->dapm_widgets = skl_hda_widgets; + card->dapm_routes = skl_hda_map; + card->add_dai_link = skl_hda_add_dai_link; + card->fully_routed = true; + card->late_probe = skl_hda_card_late_probe; snd_soc_card_set_drvdata(card, ctx); -- 2.25.1

2 1

[PATCH 1/2 4.19.y] ALSA: usb-audio: Sanity checks for each pipe and EP types
by Takashi Iwai 05 Sep '24

05 Sep '24

[ Upstream commit 801ebf1043ae7b182588554cc9b9ad3c14bc2ab5 ] The recent USB core code performs sanity checks for the given pipe and EP types, and it can be hit by manipulated USB descriptors by syzbot. For making syzbot happier, this patch introduces a local helper for a sanity check in the driver side and calls it at each place before the message handling, so that we can avoid the WARNING splats. Reported-by: syzbot+d952e5e28f5fb7718d23(a)syzkaller.appspotmail.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Dan Carpenter <dan.carpenter(a)linaro.org> --- sound/usb/helper.c | 17 +++++++++++++++++ sound/usb/helper.h | 1 + sound/usb/quirks.c | 14 +++++++++++--- 3 files changed, 29 insertions(+), 3 deletions(-) diff --git a/sound/usb/helper.c b/sound/usb/helper.c index 7712e2b84183..b1cc9499c57e 100644 --- a/sound/usb/helper.c +++ b/sound/usb/helper.c @@ -76,6 +76,20 @@ void *snd_usb_find_csint_desc(void *buffer, int buflen, void *after, u8 dsubtype return NULL; } +/* check the validity of pipe and EP types */ +int snd_usb_pipe_sanity_check(struct usb_device *dev, unsigned int pipe) +{ + static const int pipetypes[4] = { + PIPE_CONTROL, PIPE_ISOCHRONOUS, PIPE_BULK, PIPE_INTERRUPT + }; + struct usb_host_endpoint *ep; + + ep = usb_pipe_endpoint(dev, pipe); + if (usb_pipetype(pipe) != pipetypes[usb_endpoint_type(&ep->desc)]) + return -EINVAL; + return 0; +} + /* * Wrapper for usb_control_msg(). * Allocates a temp buffer to prevent dmaing from/to the stack. @@ -88,6 +102,9 @@ int snd_usb_ctl_msg(struct usb_device *dev, unsigned int pipe, __u8 request, void *buf = NULL; int timeout; + if (snd_usb_pipe_sanity_check(dev, pipe)) + return -EINVAL; + if (size > 0) { buf = kmemdup(data, size, GFP_KERNEL); if (!buf) diff --git a/sound/usb/helper.h b/sound/usb/helper.h index f5b4c6647e4d..5e8a18b4e7b9 100644 --- a/sound/usb/helper.h +++ b/sound/usb/helper.h @@ -7,6 +7,7 @@ unsigned int snd_usb_combine_bytes(unsigned char *bytes, int size); void *snd_usb_find_desc(void *descstart, int desclen, void *after, u8 dtype); void *snd_usb_find_csint_desc(void *descstart, int desclen, void *after, u8 dsubtype); +int snd_usb_pipe_sanity_check(struct usb_device *dev, unsigned int pipe); int snd_usb_ctl_msg(struct usb_device *dev, unsigned int pipe, __u8 request, __u8 requesttype, __u16 value, __u16 index, void *data, __u16 size); diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 43cbaaff163f..087eef5e249d 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -743,11 +743,13 @@ static int snd_usb_novation_boot_quirk(struct usb_device *dev) static int snd_usb_accessmusic_boot_quirk(struct usb_device *dev) { int err, actual_length; - /* "midi send" enable */ static const u8 seq[] = { 0x4e, 0x73, 0x52, 0x01 }; + void *buf; - void *buf = kmemdup(seq, ARRAY_SIZE(seq), GFP_KERNEL); + if (snd_usb_pipe_sanity_check(dev, usb_sndintpipe(dev, 0x05))) + return -EINVAL; + buf = kmemdup(seq, ARRAY_SIZE(seq), GFP_KERNEL); if (!buf) return -ENOMEM; err = usb_interrupt_msg(dev, usb_sndintpipe(dev, 0x05), buf, @@ -772,7 +774,11 @@ static int snd_usb_accessmusic_boot_quirk(struct usb_device *dev) static int snd_usb_nativeinstruments_boot_quirk(struct usb_device *dev) { - int ret = usb_control_msg(dev, usb_sndctrlpipe(dev, 0), + int ret; + + if (snd_usb_pipe_sanity_check(dev, usb_sndctrlpipe(dev, 0))) + return -EINVAL; + ret = usb_control_msg(dev, usb_sndctrlpipe(dev, 0), 0xaf, USB_TYPE_VENDOR | USB_RECIP_DEVICE, 1, 0, NULL, 0, 1000); @@ -879,6 +885,8 @@ static int snd_usb_axefx3_boot_quirk(struct usb_device *dev) dev_dbg(&dev->dev, "Waiting for Axe-Fx III to boot up...\n"); + if (snd_usb_pipe_sanity_check(dev, usb_sndctrlpipe(dev, 0))) + return -EINVAL; /* If the Axe-Fx III has not fully booted, it will timeout when trying * to enable the audio streaming interface. A more generous timeout is * used here to detect when the Axe-Fx III has finished booting as the -- 2.45.2

4 5

[PATCH] ASoC: atmel: mchp-i2s-mcc: Improve maxburst calculation for better performance
by Andrei Simion 05 Sep '24

05 Sep '24

From: Codrin Ciubotariu <codrin.ciubotariu(a)microchip.com> The period size represents the size of the DMA descriptor. To ensure all DMA descriptors start from a well-aligned address, the period size must be divided by (sample size * maxburst), not just by maxburst. This adjustment allows for computing a higher maxburst value, thereby increasing the performance of the DMA transfer. Previously, snd_pcm_lib_period_bytes() returned 0 because the runtime HW parameters are computed after the hw_params() callbacks are used. To address this, we now use params_*() functions to compute the period size accurately. This change optimizes the DMA transfer performance by ensuring proper alignment and efficient use of maxburst values. Signed-off-by: Codrin Ciubotariu <codrin.ciubotariu(a)microchip.com> [andrei.simion(a)microchip.com: Reword commit message and commit title. Add macros with values for maximum DMA chunk size allowed. Add DMA_BURST_ALIGNED preprocessor function to check the alignment of the DMA burst] Signed-off-by: Andrei Simion <andrei.simion(a)microchip.com> --- sound/soc/atmel/mchp-i2s-mcc.c | 38 +++++++++++++++++++++++++++++----- 1 file changed, 33 insertions(+), 5 deletions(-) diff --git a/sound/soc/atmel/mchp-i2s-mcc.c b/sound/soc/atmel/mchp-i2s-mcc.c index 193dd7acceb0..35a56b266b8d 100644 --- a/sound/soc/atmel/mchp-i2s-mcc.c +++ b/sound/soc/atmel/mchp-i2s-mcc.c @@ -221,6 +221,15 @@ #define MCHP_I2SMCC_MAX_CHANNELS 8 #define MCHP_I2MCC_TDM_SLOT_WIDTH 32 +/* + * ---- DMA chunk size allowed ---- + */ +#define MCHP_I2SMCC_DMA_8_WORD_CHUNK 8 +#define MCHP_I2SMCC_DMA_4_WORD_CHUNK 4 +#define MCHP_I2SMCC_DMA_2_WORD_CHUNK 2 +#define MCHP_I2SMCC_DMA_1_WORD_CHUNK 1 +#define DMA_BURST_ALIGNED(_p, _s, _w) !(_p % (_s * _w)) + static const struct regmap_config mchp_i2s_mcc_regmap_config = { .reg_bits = 32, .reg_stride = 4, @@ -504,12 +513,30 @@ static int mchp_i2s_mcc_is_running(struct mchp_i2s_mcc_dev *dev) return !!(sr & (MCHP_I2SMCC_SR_TXEN | MCHP_I2SMCC_SR_RXEN)); } +static inline int mchp_i2s_mcc_period_to_maxburst(int period_size, int sample_size) +{ + int p_size = period_size; + int s_size = sample_size; + + if (DMA_BURST_ALIGNED(p_size, s_size, MCHP_I2SMCC_DMA_8_WORD_CHUNK)) + return MCHP_I2SMCC_DMA_8_WORD_CHUNK; + if (DMA_BURST_ALIGNED(p_size, s_size, MCHP_I2SMCC_DMA_4_WORD_CHUNK)) + return MCHP_I2SMCC_DMA_4_WORD_CHUNK; + if (DMA_BURST_ALIGNED(p_size, s_size, MCHP_I2SMCC_DMA_2_WORD_CHUNK)) + return MCHP_I2SMCC_DMA_2_WORD_CHUNK; + return MCHP_I2SMCC_DMA_1_WORD_CHUNK; +} + static int mchp_i2s_mcc_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { unsigned long rate = 0; struct mchp_i2s_mcc_dev *dev = snd_soc_dai_get_drvdata(dai); + int sample_bytes = params_physical_width(params) / 8; + int period_bytes = params_period_size(params) * + params_channels(params) * sample_bytes; + int maxburst; u32 mra = 0; u32 mrb = 0; unsigned int channels = params_channels(params); @@ -519,9 +546,9 @@ static int mchp_i2s_mcc_hw_params(struct snd_pcm_substream *substream, int ret; bool is_playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); - dev_dbg(dev->dev, "%s() rate=%u format=%#x width=%u channels=%u\n", + dev_dbg(dev->dev, "%s() rate=%u format=%#x width=%u channels=%u period_bytes=%d\n", __func__, params_rate(params), params_format(params), - params_width(params), params_channels(params)); + params_width(params), params_channels(params), period_bytes); switch (dev->fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: @@ -630,11 +657,12 @@ static int mchp_i2s_mcc_hw_params(struct snd_pcm_substream *substream, * We must have the same burst size configured * in the DMA transfer and in out IP */ - mrb |= MCHP_I2SMCC_MRB_DMACHUNK(channels); + maxburst = mchp_i2s_mcc_period_to_maxburst(period_bytes, sample_bytes); + mrb |= MCHP_I2SMCC_MRB_DMACHUNK(maxburst); if (is_playback) - dev->playback.maxburst = 1 << (fls(channels) - 1); + dev->playback.maxburst = maxburst; else - dev->capture.maxburst = 1 << (fls(channels) - 1); + dev->capture.maxburst = maxburst; switch (params_format(params)) { case SNDRV_PCM_FORMAT_S8: base-commit: fdadd93817f124fd0ea6ef251d4a1068b7feceba -- 2.34.1

2 1

[PATCH 2/2 4.19.y] ALSA: usb-audio: Fix gpf in snd_usb_pipe_sanity_check
by Hillf Danton 05 Sep '24

05 Sep '24

[ Upstream commit 5d78e1c2b7f4be00bbe62141603a631dc7812f35 ] syzbot found the following crash on: general protection fault: 0000 [#1] SMP KASAN RIP: 0010:snd_usb_pipe_sanity_check+0x80/0x130 sound/usb/helper.c:75 Call Trace: snd_usb_motu_microbookii_communicate.constprop.0+0xa0/0x2fb sound/usb/quirks.c:1007 snd_usb_motu_microbookii_boot_quirk sound/usb/quirks.c:1051 [inline] snd_usb_apply_boot_quirk.cold+0x163/0x370 sound/usb/quirks.c:1280 usb_audio_probe+0x2ec/0x2010 sound/usb/card.c:576 usb_probe_interface+0x305/0x7a0 drivers/usb/core/driver.c:361 really_probe+0x281/0x650 drivers/base/dd.c:548 .... It was introduced in commit 801ebf1043ae for checking pipe and endpoint types. It is fixed by adding a check of the ep pointer in question. BugLink: https://syzkaller.appspot.com/bug?extid=d59c4387bfb6eced94e2 Reported-by: syzbot <syzbot+d59c4387bfb6eced94e2(a)syzkaller.appspotmail.com> Fixes: 801ebf1043ae ("ALSA: usb-audio: Sanity checks for each pipe and EP types") Cc: Andrey Konovalov <andreyknvl(a)google.com> Signed-off-by: Hillf Danton <hdanton(a)sina.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Dan Carpenter <dan.carpenter(a)linaro.org> --- sound/usb/helper.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/helper.c b/sound/usb/helper.c index b1cc9499c57e..029489b490ca 100644 --- a/sound/usb/helper.c +++ b/sound/usb/helper.c @@ -85,7 +85,7 @@ int snd_usb_pipe_sanity_check(struct usb_device *dev, unsigned int pipe) struct usb_host_endpoint *ep; ep = usb_pipe_endpoint(dev, pipe); - if (usb_pipetype(pipe) != pipetypes[usb_endpoint_type(&ep->desc)]) + if (!ep || usb_pipetype(pipe) != pipetypes[usb_endpoint_type(&ep->desc)]) return -EINVAL; return 0; } -- 2.45.2

2 1

[PATCH 0/2 4.19.y] ALSA: usb-audio: sanity checks for pipes
by Dan Carpenter 05 Sep '24

05 Sep '24

Backporting the sanity checks on pipes seems like a good idea. These are basically the same as Takashi Iwai's patches upstream. The difference is that upstream added two sanity checks to code that doesn't exist in 4.19. I had talked about backporting fcc2cc1f3561 ("USB: move snd_usb_pipe_sanity_check into the USB core") but that's just a refactor and not a bug fix. Hillf Danton (1): ALSA: usb-audio: Fix gpf in snd_usb_pipe_sanity_check Takashi Iwai (1): ALSA: usb-audio: Sanity checks for each pipe and EP types sound/usb/helper.c | 17 +++++++++++++++++ sound/usb/helper.h | 1 + sound/usb/quirks.c | 14 +++++++++++--- 3 files changed, 29 insertions(+), 3 deletions(-) -- 2.45.2

1 0

[PATCH] ASoC/HDA cs35l56: Remove redundant IRQ handling
by Simon Trimmer 05 Sep '24

05 Sep '24

The IRQ handling in the cs35l56 driver was purely informational. It was not necessary to support the HDA or ASoC driver functionality and added unnecessary complexity to the drivers. As the IRQ signal GPIO line could be connected and shared with other components the handling is replaced with a regmap patch to ensure the cs35l56 IRQ sources are masked and will not generate interrupts that go unhandled. Signed-off-by: Simon Trimmer <simont(a)opensource.cirrus.com> --- v1 - As this changes both the ASoC and HDA cs35l56 device drivers we considered a series of patches that stepped through the removal of the common code but thought that the co-ordination would be unnecessarily complicated so we initially offer a combined patch that that applies to both Mark and Takashi's for-next branches so that either route can be taken. include/sound/cs35l56.h | 4 -- sound/pci/hda/cs35l56_hda.c | 53 +-------------- sound/pci/hda/cs35l56_hda_i2c.c | 9 +-- sound/pci/hda/cs35l56_hda_spi.c | 9 +-- sound/soc/codecs/cs35l56-i2c.c | 2 - sound/soc/codecs/cs35l56-sdw.c | 104 +----------------------------- sound/soc/codecs/cs35l56-shared.c | 101 +---------------------------- sound/soc/codecs/cs35l56-spi.c | 2 - sound/soc/codecs/cs35l56.c | 92 ++------------------------ sound/soc/codecs/cs35l56.h | 10 --- 10 files changed, 14 insertions(+), 372 deletions(-) diff --git a/include/sound/cs35l56.h b/include/sound/cs35l56.h index 94e8185c4795..acc43d85f5df 100644 --- a/include/sound/cs35l56.h +++ b/include/sound/cs35l56.h @@ -257,8 +257,6 @@ struct cs35l56_base { struct device *dev; struct regmap *regmap; - int irq; - struct mutex irq_lock; u8 type; u8 rev; bool init_done; @@ -298,8 +296,6 @@ int cs35l56_wait_for_firmware_boot(struct cs35l56_base *cs35l56_base); void cs35l56_wait_control_port_ready(void); void cs35l56_wait_min_reset_pulse(void); void cs35l56_system_reset(struct cs35l56_base *cs35l56_base, bool is_soundwire); -int cs35l56_irq_request(struct cs35l56_base *cs35l56_base, int irq); -irqreturn_t cs35l56_irq(int irq, void *data); int cs35l56_is_fw_reload_needed(struct cs35l56_base *cs35l56_base); int cs35l56_runtime_suspend_common(struct cs35l56_base *cs35l56_base); int cs35l56_runtime_resume_common(struct cs35l56_base *cs35l56_base, bool is_soundwire); diff --git a/sound/pci/hda/cs35l56_hda.c b/sound/pci/hda/cs35l56_hda.c index 31cc92bac89a..8ec06a3f4765 100644 --- a/sound/pci/hda/cs35l56_hda.c +++ b/sound/pci/hda/cs35l56_hda.c @@ -626,8 +626,6 @@ static void cs35l56_hda_fw_load(struct cs35l56_hda *cs35l56) goto err_fw_release; } - mutex_lock(&cs35l56->base.irq_lock); - /* * If the firmware hasn't been patched it must be shutdown before * doing a full patch and reset afterwards. If it is already @@ -637,14 +635,14 @@ static void cs35l56_hda_fw_load(struct cs35l56_hda *cs35l56) if (firmware_missing && (wmfw_firmware || coeff_firmware)) { ret = cs35l56_firmware_shutdown(&cs35l56->base); if (ret) - goto err; + goto err_fw_release; } ret = cs_dsp_power_up(&cs35l56->cs_dsp, wmfw_firmware, wmfw_filename, coeff_firmware, coeff_filename, "misc"); if (ret) { dev_dbg(cs35l56->base.dev, "%s: cs_dsp_power_up ret %d\n", __func__, ret); - goto err; + goto err_fw_release; } if (wmfw_filename) @@ -687,8 +685,6 @@ static void cs35l56_hda_fw_load(struct cs35l56_hda *cs35l56) err_powered_up: if (!cs35l56->base.fw_patched) cs_dsp_power_down(&cs35l56->cs_dsp); -err: - mutex_unlock(&cs35l56->base.irq_lock); err_fw_release: cs35l56_hda_release_firmware_files(wmfw_firmware, wmfw_filename, coeff_firmware, coeff_filename); @@ -778,15 +774,6 @@ static int cs35l56_hda_system_suspend(struct device *dev) cs35l56->suspended = true; - /* - * The interrupt line is normally shared, but after we start suspending - * we can't check if our device is the source of an interrupt, and can't - * clear it. Prevent this race by temporarily disabling the parent irq - * until we reach _no_irq. - */ - if (cs35l56->base.irq) - disable_irq(cs35l56->base.irq); - return pm_runtime_force_suspend(dev); } @@ -806,34 +793,6 @@ static int cs35l56_hda_system_suspend_late(struct device *dev) return 0; } -static int cs35l56_hda_system_suspend_no_irq(struct device *dev) -{ - struct cs35l56_hda *cs35l56 = dev_get_drvdata(dev); - - /* Handlers are now disabled so the parent IRQ can safely be re-enabled. */ - if (cs35l56->base.irq) - enable_irq(cs35l56->base.irq); - - return 0; -} - -static int cs35l56_hda_system_resume_no_irq(struct device *dev) -{ - struct cs35l56_hda *cs35l56 = dev_get_drvdata(dev); - - /* - * WAKE interrupts unmask if the CS35L56 hibernates, which can cause - * spurious interrupts, and the interrupt line is normally shared. - * We can't check if our device is the source of an interrupt, and can't - * clear it, until it has fully resumed. Prevent this race by temporarily - * disabling the parent irq until we complete resume(). - */ - if (cs35l56->base.irq) - disable_irq(cs35l56->base.irq); - - return 0; -} - static int cs35l56_hda_system_resume_early(struct device *dev) { struct cs35l56_hda *cs35l56 = dev_get_drvdata(dev); @@ -856,11 +815,8 @@ static int cs35l56_hda_system_resume(struct device *dev) struct cs35l56_hda *cs35l56 = dev_get_drvdata(dev); int ret; - /* Undo pm_runtime_force_suspend() before re-enabling the irq */ + /* Undo pm_runtime_force_suspend() */ ret = pm_runtime_force_resume(dev); - if (cs35l56->base.irq) - enable_irq(cs35l56->base.irq); - if (ret) return ret; @@ -987,7 +943,6 @@ int cs35l56_hda_common_probe(struct cs35l56_hda *cs35l56, int hid, int id) { int ret; - mutex_init(&cs35l56->base.irq_lock); dev_set_drvdata(cs35l56->base.dev, cs35l56); INIT_WORK(&cs35l56->dsp_work, cs35l56_hda_dsp_work); @@ -1120,8 +1075,6 @@ const struct dev_pm_ops cs35l56_hda_pm_ops = { SYSTEM_SLEEP_PM_OPS(cs35l56_hda_system_suspend, cs35l56_hda_system_resume) LATE_SYSTEM_SLEEP_PM_OPS(cs35l56_hda_system_suspend_late, cs35l56_hda_system_resume_early) - NOIRQ_SYSTEM_SLEEP_PM_OPS(cs35l56_hda_system_suspend_no_irq, - cs35l56_hda_system_resume_no_irq) }; EXPORT_SYMBOL_NS_GPL(cs35l56_hda_pm_ops, SND_HDA_SCODEC_CS35L56); diff --git a/sound/pci/hda/cs35l56_hda_i2c.c b/sound/pci/hda/cs35l56_hda_i2c.c index 40f2f97944d5..822c1251f703 100644 --- a/sound/pci/hda/cs35l56_hda_i2c.c +++ b/sound/pci/hda/cs35l56_hda_i2c.c @@ -34,14 +34,7 @@ static int cs35l56_hda_i2c_probe(struct i2c_client *clt) return ret; } - ret = cs35l56_hda_common_probe(cs35l56, id->driver_data, clt->addr); - if (ret) - return ret; - ret = cs35l56_irq_request(&cs35l56->base, clt->irq); - if (ret < 0) - cs35l56_hda_remove(cs35l56->base.dev); - - return ret; + return cs35l56_hda_common_probe(cs35l56, id->driver_data, clt->addr); } static void cs35l56_hda_i2c_remove(struct i2c_client *clt) diff --git a/sound/pci/hda/cs35l56_hda_spi.c b/sound/pci/hda/cs35l56_hda_spi.c index 7f02155fe61e..37a159546e59 100644 --- a/sound/pci/hda/cs35l56_hda_spi.c +++ b/sound/pci/hda/cs35l56_hda_spi.c @@ -34,14 +34,7 @@ static int cs35l56_hda_spi_probe(struct spi_device *spi) return ret; } - ret = cs35l56_hda_common_probe(cs35l56, id->driver_data, spi_get_chipselect(spi, 0)); - if (ret) - return ret; - ret = cs35l56_irq_request(&cs35l56->base, spi->irq); - if (ret < 0) - cs35l56_hda_remove(cs35l56->base.dev); - - return ret; + return cs35l56_hda_common_probe(cs35l56, id->driver_data, spi_get_chipselect(spi, 0)); } static void cs35l56_hda_spi_remove(struct spi_device *spi) diff --git a/sound/soc/codecs/cs35l56-i2c.c b/sound/soc/codecs/cs35l56-i2c.c index 2bd2ff75cd50..4419a94c5e73 100644 --- a/sound/soc/codecs/cs35l56-i2c.c +++ b/sound/soc/codecs/cs35l56-i2c.c @@ -41,8 +41,6 @@ static int cs35l56_i2c_probe(struct i2c_client *client) return ret; ret = cs35l56_init(cs35l56); - if (ret == 0) - ret = cs35l56_irq_request(&cs35l56->base, client->irq); if (ret < 0) cs35l56_remove(cs35l56); diff --git a/sound/soc/codecs/cs35l56-sdw.c b/sound/soc/codecs/cs35l56-sdw.c index 7c9a17fe2195..134631eff861 100644 --- a/sound/soc/codecs/cs35l56-sdw.c +++ b/sound/soc/codecs/cs35l56-sdw.c @@ -266,75 +266,14 @@ static void cs35l56_sdw_init(struct sdw_slave *peripheral) } ret = cs35l56_init(cs35l56); - if (ret < 0) { + if (ret < 0) regcache_cache_only(cs35l56->base.regmap, true); - goto out; - } - - /* - * cs35l56_init can return with !init_done if it triggered - * a soft reset. - */ - if (cs35l56->base.init_done) { - /* Enable SoundWire interrupts */ - sdw_write_no_pm(peripheral, CS35L56_SDW_GEN_INT_MASK_1, - CS35L56_SDW_INT_MASK_CODEC_IRQ); - } out: pm_runtime_mark_last_busy(cs35l56->base.dev); pm_runtime_put_autosuspend(cs35l56->base.dev); } -static int cs35l56_sdw_interrupt(struct sdw_slave *peripheral, - struct sdw_slave_intr_status *status) -{ - struct cs35l56_private *cs35l56 = dev_get_drvdata(&peripheral->dev); - - /* SoundWire core holds our pm_runtime when calling this function. */ - - dev_dbg(cs35l56->base.dev, "int control_port=%#x\n", status->control_port); - - if ((status->control_port & SDW_SCP_INT1_IMPL_DEF) == 0) - return 0; - - /* - * Prevent bus manager suspending and possibly issuing a - * bus-reset before the queued work has run. - */ - pm_runtime_get_noresume(cs35l56->base.dev); - - /* - * Mask and clear until it has been handled. The read of GEN_INT_STAT_1 - * is required as per the SoundWire spec for interrupt status bits - * to clear. GEN_INT_MASK_1 masks the _inputs_ to GEN_INT_STAT1. - * None of the interrupts are time-critical so use the - * power-efficient queue. - */ - sdw_write_no_pm(peripheral, CS35L56_SDW_GEN_INT_MASK_1, 0); - sdw_read_no_pm(peripheral, CS35L56_SDW_GEN_INT_STAT_1); - sdw_write_no_pm(peripheral, CS35L56_SDW_GEN_INT_STAT_1, 0xFF); - queue_work(system_power_efficient_wq, &cs35l56->sdw_irq_work); - - return 0; -} - -static void cs35l56_sdw_irq_work(struct work_struct *work) -{ - struct cs35l56_private *cs35l56 = container_of(work, - struct cs35l56_private, - sdw_irq_work); - - cs35l56_irq(-1, &cs35l56->base); - - /* unmask interrupts */ - if (!cs35l56->sdw_irq_no_unmask) - sdw_write_no_pm(cs35l56->sdw_peripheral, CS35L56_SDW_GEN_INT_MASK_1, - CS35L56_SDW_INT_MASK_CODEC_IRQ); - - pm_runtime_put_autosuspend(cs35l56->base.dev); -} - static int cs35l56_sdw_read_prop(struct sdw_slave *peripheral) { struct cs35l56_private *cs35l56 = dev_get_drvdata(&peripheral->dev); @@ -349,7 +288,7 @@ static int cs35l56_sdw_read_prop(struct sdw_slave *peripheral) prop->sink_ports = BIT(CS35L56_SDW1_PLAYBACK_PORT); prop->paging_support = true; prop->quirks = SDW_SLAVE_QUIRKS_INVALID_INITIAL_PARITY; - prop->scp_int1_mask = SDW_SCP_INT1_BUS_CLASH | SDW_SCP_INT1_PARITY | SDW_SCP_INT1_IMPL_DEF; + prop->scp_int1_mask = SDW_SCP_INT1_BUS_CLASH | SDW_SCP_INT1_PARITY; /* DP1 - playback */ ports[0].num = CS35L56_SDW1_PLAYBACK_PORT; @@ -406,7 +345,6 @@ static int __maybe_unused cs35l56_sdw_clk_stop(struct sdw_slave *peripheral, static const struct sdw_slave_ops cs35l56_sdw_ops = { .read_prop = cs35l56_sdw_read_prop, - .interrupt_callback = cs35l56_sdw_interrupt, .update_status = cs35l56_sdw_update_status, #ifdef DEBUG .clk_stop = cs35l56_sdw_clk_stop, @@ -465,10 +403,6 @@ static int __maybe_unused cs35l56_sdw_runtime_resume(struct device *dev) if (ret) return ret; - /* Re-enable SoundWire interrupts */ - sdw_write_no_pm(cs35l56->sdw_peripheral, CS35L56_SDW_GEN_INT_MASK_1, - CS35L56_SDW_INT_MASK_CODEC_IRQ); - return 0; } @@ -479,32 +413,9 @@ static int __maybe_unused cs35l56_sdw_system_suspend(struct device *dev) if (!cs35l56->base.init_done) return 0; - /* - * Disable SoundWire interrupts. - * Flush - don't cancel because that could leave an unbalanced pm_runtime_get. - */ - cs35l56->sdw_irq_no_unmask = true; - flush_work(&cs35l56->sdw_irq_work); - - /* Mask interrupts and flush in case sdw_irq_work was queued again */ - sdw_write_no_pm(cs35l56->sdw_peripheral, CS35L56_SDW_GEN_INT_MASK_1, 0); - sdw_read_no_pm(cs35l56->sdw_peripheral, CS35L56_SDW_GEN_INT_STAT_1); - sdw_write_no_pm(cs35l56->sdw_peripheral, CS35L56_SDW_GEN_INT_STAT_1, 0xFF); - flush_work(&cs35l56->sdw_irq_work); - return cs35l56_system_suspend(dev); } -static int __maybe_unused cs35l56_sdw_system_resume(struct device *dev) -{ - struct cs35l56_private *cs35l56 = dev_get_drvdata(dev); - - cs35l56->sdw_irq_no_unmask = false; - /* runtime_resume re-enables the interrupt */ - - return cs35l56_system_resume(dev); -} - static int cs35l56_sdw_probe(struct sdw_slave *peripheral, const struct sdw_device_id *id) { struct device *dev = &peripheral->dev; @@ -517,7 +428,6 @@ static int cs35l56_sdw_probe(struct sdw_slave *peripheral, const struct sdw_devi cs35l56->base.dev = dev; cs35l56->sdw_peripheral = peripheral; - INIT_WORK(&cs35l56->sdw_irq_work, cs35l56_sdw_irq_work); dev_set_drvdata(dev, cs35l56); @@ -542,13 +452,6 @@ static int cs35l56_sdw_remove(struct sdw_slave *peripheral) { struct cs35l56_private *cs35l56 = dev_get_drvdata(&peripheral->dev); - /* Disable SoundWire interrupts */ - cs35l56->sdw_irq_no_unmask = true; - cancel_work_sync(&cs35l56->sdw_irq_work); - sdw_write_no_pm(peripheral, CS35L56_SDW_GEN_INT_MASK_1, 0); - sdw_read_no_pm(peripheral, CS35L56_SDW_GEN_INT_STAT_1); - sdw_write_no_pm(peripheral, CS35L56_SDW_GEN_INT_STAT_1, 0xFF); - cs35l56_remove(cs35l56); return 0; @@ -556,9 +459,8 @@ static int cs35l56_sdw_remove(struct sdw_slave *peripheral) static const struct dev_pm_ops cs35l56_sdw_pm = { SET_RUNTIME_PM_OPS(cs35l56_sdw_runtime_suspend, cs35l56_sdw_runtime_resume, NULL) - SYSTEM_SLEEP_PM_OPS(cs35l56_sdw_system_suspend, cs35l56_sdw_system_resume) + SYSTEM_SLEEP_PM_OPS(cs35l56_sdw_system_suspend, cs35l56_system_resume) LATE_SYSTEM_SLEEP_PM_OPS(cs35l56_system_suspend_late, cs35l56_system_resume_early) - /* NOIRQ stage not needed, SoundWire doesn't use a hard IRQ */ }; static const struct sdw_device_id cs35l56_sdw_id[] = { diff --git a/sound/soc/codecs/cs35l56-shared.c b/sound/soc/codecs/cs35l56-shared.c index e45e9ae01bc6..55adbc3d26ae 100644 --- a/sound/soc/codecs/cs35l56-shared.c +++ b/sound/soc/codecs/cs35l56-shared.c @@ -37,6 +37,7 @@ static const struct reg_sequence cs35l56_patch[] = { { CS35L56_SWIRE_DP3_CH3_INPUT, 0x00000029 }, { CS35L56_SWIRE_DP3_CH4_INPUT, 0x00000028 }, { CS35L56_IRQ1_MASK_18, 0x1f7df0ff }, + { CS35L56_IRQ1_MASK_20, 0x15fff19c }, /* These are not reset by a soft-reset, so patch to defaults. */ { CS35L56_MAIN_RENDER_USER_MUTE, 0x00000000 }, @@ -70,12 +71,6 @@ static const struct reg_default cs35l56_reg_defaults[] = { { CS35L56_SWIRE_DP3_CH2_INPUT, 0x00000019 }, { CS35L56_SWIRE_DP3_CH3_INPUT, 0x00000029 }, { CS35L56_SWIRE_DP3_CH4_INPUT, 0x00000028 }, - { CS35L56_IRQ1_MASK_1, 0x83ffffff }, - { CS35L56_IRQ1_MASK_2, 0xffff7fff }, - { CS35L56_IRQ1_MASK_4, 0xe0ffffff }, - { CS35L56_IRQ1_MASK_8, 0xfc000fff }, - { CS35L56_IRQ1_MASK_18, 0x1f7df0ff }, - { CS35L56_IRQ1_MASK_20, 0x15c00000 }, { CS35L56_MAIN_RENDER_USER_MUTE, 0x00000000 }, { CS35L56_MAIN_RENDER_USER_VOLUME, 0x00000000 }, { CS35L56_MAIN_POSTURE_NUMBER, 0x00000000 }, @@ -329,91 +324,6 @@ void cs35l56_system_reset(struct cs35l56_base *cs35l56_base, bool is_soundwire) } EXPORT_SYMBOL_NS_GPL(cs35l56_system_reset, SND_SOC_CS35L56_SHARED); -int cs35l56_irq_request(struct cs35l56_base *cs35l56_base, int irq) -{ - int ret; - - if (irq < 1) - return 0; - - ret = devm_request_threaded_irq(cs35l56_base->dev, irq, NULL, cs35l56_irq, - IRQF_ONESHOT | IRQF_SHARED | IRQF_TRIGGER_LOW, - "cs35l56", cs35l56_base); - if (!ret) - cs35l56_base->irq = irq; - else - dev_err(cs35l56_base->dev, "Failed to get IRQ: %d\n", ret); - - return ret; -} -EXPORT_SYMBOL_NS_GPL(cs35l56_irq_request, SND_SOC_CS35L56_SHARED); - -irqreturn_t cs35l56_irq(int irq, void *data) -{ - struct cs35l56_base *cs35l56_base = data; - unsigned int status1 = 0, status8 = 0, status20 = 0; - unsigned int mask1, mask8, mask20; - unsigned int val; - int rv; - - irqreturn_t ret = IRQ_NONE; - - if (!cs35l56_base->init_done) - return IRQ_NONE; - - mutex_lock(&cs35l56_base->irq_lock); - - rv = pm_runtime_resume_and_get(cs35l56_base->dev); - if (rv < 0) { - dev_err(cs35l56_base->dev, "irq: failed to get pm_runtime: %d\n", rv); - goto err_unlock; - } - - regmap_read(cs35l56_base->regmap, CS35L56_IRQ1_STATUS, &val); - if ((val & CS35L56_IRQ1_STS_MASK) == 0) { - dev_dbg(cs35l56_base->dev, "Spurious IRQ: no pending interrupt\n"); - goto err; - } - - /* Ack interrupts */ - regmap_read(cs35l56_base->regmap, CS35L56_IRQ1_EINT_1, &status1); - regmap_read(cs35l56_base->regmap, CS35L56_IRQ1_MASK_1, &mask1); - status1 &= ~mask1; - regmap_write(cs35l56_base->regmap, CS35L56_IRQ1_EINT_1, status1); - - regmap_read(cs35l56_base->regmap, CS35L56_IRQ1_EINT_8, &status8); - regmap_read(cs35l56_base->regmap, CS35L56_IRQ1_MASK_8, &mask8); - status8 &= ~mask8; - regmap_write(cs35l56_base->regmap, CS35L56_IRQ1_EINT_8, status8); - - regmap_read(cs35l56_base->regmap, CS35L56_IRQ1_EINT_20, &status20); - regmap_read(cs35l56_base->regmap, CS35L56_IRQ1_MASK_20, &mask20); - status20 &= ~mask20; - /* We don't want EINT20 but they default to unmasked: force mask */ - regmap_write(cs35l56_base->regmap, CS35L56_IRQ1_MASK_20, 0xffffffff); - - dev_dbg(cs35l56_base->dev, "%s: %#x %#x\n", __func__, status1, status8); - - /* Check to see if unmasked bits are active */ - if (!status1 && !status8 && !status20) - goto err; - - if (status1 & CS35L56_AMP_SHORT_ERR_EINT1_MASK) - dev_crit(cs35l56_base->dev, "Amp short error\n"); - - if (status8 & CS35L56_TEMP_ERR_EINT1_MASK) - dev_crit(cs35l56_base->dev, "Overtemp error\n"); - - ret = IRQ_HANDLED; - -err: - pm_runtime_put(cs35l56_base->dev); -err_unlock: - mutex_unlock(&cs35l56_base->irq_lock); - - return ret; -} -EXPORT_SYMBOL_NS_GPL(cs35l56_irq, SND_SOC_CS35L56_SHARED); int cs35l56_is_fw_reload_needed(struct cs35l56_base *cs35l56_base) { @@ -763,15 +673,6 @@ int cs35l56_hw_init(struct cs35l56_base *cs35l56_base) cs35l56_base->type, cs35l56_base->secured ? "s" : "", cs35l56_base->rev, otpid, fw_ver >> 16, (fw_ver >> 8) & 0xff, fw_ver & 0xff, !fw_missing); - /* Wake source and *_BLOCKED interrupts default to unmasked, so mask them */ - regmap_write(cs35l56_base->regmap, CS35L56_IRQ1_MASK_20, 0xffffffff); - regmap_update_bits(cs35l56_base->regmap, CS35L56_IRQ1_MASK_1, - CS35L56_AMP_SHORT_ERR_EINT1_MASK, - 0); - regmap_update_bits(cs35l56_base->regmap, CS35L56_IRQ1_MASK_8, - CS35L56_TEMP_ERR_EINT1_MASK, - 0); - return 0; } EXPORT_SYMBOL_NS_GPL(cs35l56_hw_init, SND_SOC_CS35L56_SHARED); diff --git a/sound/soc/codecs/cs35l56-spi.c b/sound/soc/codecs/cs35l56-spi.c index b07b798b0b45..1382b8903ea7 100644 --- a/sound/soc/codecs/cs35l56-spi.c +++ b/sound/soc/codecs/cs35l56-spi.c @@ -39,8 +39,6 @@ static int cs35l56_spi_probe(struct spi_device *spi) return ret; ret = cs35l56_init(cs35l56); - if (ret == 0) - ret = cs35l56_irq_request(&cs35l56->base, spi->irq); if (ret < 0) cs35l56_remove(cs35l56); diff --git a/sound/soc/codecs/cs35l56.c b/sound/soc/codecs/cs35l56.c index 757ade6373ed..cf2aee6cda6c 100644 --- a/sound/soc/codecs/cs35l56.c +++ b/sound/soc/codecs/cs35l56.c @@ -12,7 +12,6 @@ #include <linux/delay.h> #include <linux/err.h> #include <linux/gpio/consumer.h> -#include <linux/interrupt.h> #include <linux/math.h> #include <linux/module.h> #include <linux/pm.h> @@ -702,23 +701,9 @@ static void cs35l56_patch(struct cs35l56_private *cs35l56, bool firmware_missing { int ret; - /* - * Disable SoundWire interrupts to prevent race with IRQ work. - * Setting sdw_irq_no_unmask prevents the handler re-enabling - * the SoundWire interrupt. - */ - if (cs35l56->sdw_peripheral) { - cs35l56->sdw_irq_no_unmask = true; - flush_work(&cs35l56->sdw_irq_work); - sdw_write_no_pm(cs35l56->sdw_peripheral, CS35L56_SDW_GEN_INT_MASK_1, 0); - sdw_read_no_pm(cs35l56->sdw_peripheral, CS35L56_SDW_GEN_INT_STAT_1); - sdw_write_no_pm(cs35l56->sdw_peripheral, CS35L56_SDW_GEN_INT_STAT_1, 0xFF); - flush_work(&cs35l56->sdw_irq_work); - } - ret = cs35l56_firmware_shutdown(&cs35l56->base); if (ret) - goto err; + return; /* * Use wm_adsp to load and apply the firmware patch and coefficient files, @@ -728,11 +713,9 @@ static void cs35l56_patch(struct cs35l56_private *cs35l56, bool firmware_missing ret = wm_adsp_power_up(&cs35l56->dsp, !!firmware_missing); if (ret) { dev_dbg(cs35l56->base.dev, "%s: wm_adsp_power_up ret %d\n", __func__, ret); - goto err; + return; } - mutex_lock(&cs35l56->base.irq_lock); - reinit_completion(&cs35l56->init_completion); cs35l56->soft_resetting = true; @@ -748,10 +731,10 @@ static void cs35l56_patch(struct cs35l56_private *cs35l56, bool firmware_missing msecs_to_jiffies(5000))) { dev_err(cs35l56->base.dev, "%s: init_completion timed out (SDW)\n", __func__); - goto err_unlock; + return; } } else if (cs35l56_init(cs35l56)) { - goto err_unlock; + return; } regmap_clear_bits(cs35l56->base.regmap, CS35L56_PROTECTION_STATUS, @@ -760,16 +743,6 @@ static void cs35l56_patch(struct cs35l56_private *cs35l56, bool firmware_missing if (cs35l56_write_cal(cs35l56) == 0) cs35l56_mbox_send(&cs35l56->base, CS35L56_MBOX_CMD_AUDIO_REINIT); - -err_unlock: - mutex_unlock(&cs35l56->base.irq_lock); -err: - /* Re-enable SoundWire interrupts */ - if (cs35l56->sdw_peripheral) { - cs35l56->sdw_irq_no_unmask = false; - sdw_write_no_pm(cs35l56->sdw_peripheral, CS35L56_SDW_GEN_INT_MASK_1, - CS35L56_SDW_INT_MASK_CODEC_IRQ); - } } static void cs35l56_dsp_work(struct work_struct *work) @@ -956,15 +929,6 @@ int cs35l56_system_suspend(struct device *dev) if (cs35l56->component) flush_work(&cs35l56->dsp_work); - /* - * The interrupt line is normally shared, but after we start suspending - * we can't check if our device is the source of an interrupt, and can't - * clear it. Prevent this race by temporarily disabling the parent irq - * until we reach _no_irq. - */ - if (cs35l56->base.irq) - disable_irq(cs35l56->base.irq); - return pm_runtime_force_suspend(dev); } EXPORT_SYMBOL_GPL(cs35l56_system_suspend); @@ -991,40 +955,6 @@ int cs35l56_system_suspend_late(struct device *dev) } EXPORT_SYMBOL_GPL(cs35l56_system_suspend_late); -int cs35l56_system_suspend_no_irq(struct device *dev) -{ - struct cs35l56_private *cs35l56 = dev_get_drvdata(dev); - - dev_dbg(dev, "system_suspend_no_irq\n"); - - /* Handlers are now disabled so the parent IRQ can safely be re-enabled. */ - if (cs35l56->base.irq) - enable_irq(cs35l56->base.irq); - - return 0; -} -EXPORT_SYMBOL_GPL(cs35l56_system_suspend_no_irq); - -int cs35l56_system_resume_no_irq(struct device *dev) -{ - struct cs35l56_private *cs35l56 = dev_get_drvdata(dev); - - dev_dbg(dev, "system_resume_no_irq\n"); - - /* - * WAKE interrupts unmask if the CS35L56 hibernates, which can cause - * spurious interrupts, and the interrupt line is normally shared. - * We can't check if our device is the source of an interrupt, and can't - * clear it, until it has fully resumed. Prevent this race by temporarily - * disabling the parent irq until we complete resume(). - */ - if (cs35l56->base.irq) - disable_irq(cs35l56->base.irq); - - return 0; -} -EXPORT_SYMBOL_GPL(cs35l56_system_resume_no_irq); - int cs35l56_system_resume_early(struct device *dev) { struct cs35l56_private *cs35l56 = dev_get_drvdata(dev); @@ -1065,11 +995,8 @@ int cs35l56_system_resume(struct device *dev) */ cs35l56_wait_control_port_ready(); - /* Undo pm_runtime_force_suspend() before re-enabling the irq */ + /* Undo pm_runtime_force_suspend() */ ret = pm_runtime_force_resume(dev); - if (cs35l56->base.irq) - enable_irq(cs35l56->base.irq); - if (ret) return ret; @@ -1267,7 +1194,6 @@ int cs35l56_common_probe(struct cs35l56_private *cs35l56) int ret; init_completion(&cs35l56->init_completion); - mutex_init(&cs35l56->base.irq_lock); cs35l56->base.cal_index = -1; cs35l56->speaker_id = -ENOENT; @@ -1428,13 +1354,6 @@ void cs35l56_remove(struct cs35l56_private *cs35l56) { cs35l56->base.init_done = false; - /* - * WAKE IRQs unmask if CS35L56 hibernates so free the handler to - * prevent it racing with remove(). - */ - if (cs35l56->base.irq) - devm_free_irq(cs35l56->base.dev, cs35l56->base.irq, &cs35l56->base); - flush_workqueue(cs35l56->dsp_wq); destroy_workqueue(cs35l56->dsp_wq); @@ -1454,7 +1373,6 @@ EXPORT_NS_GPL_DEV_PM_OPS(cs35l56_pm_ops_i2c_spi, SND_SOC_CS35L56_CORE) = { SET_RUNTIME_PM_OPS(cs35l56_runtime_suspend_i2c_spi, cs35l56_runtime_resume_i2c_spi, NULL) SYSTEM_SLEEP_PM_OPS(cs35l56_system_suspend, cs35l56_system_resume) LATE_SYSTEM_SLEEP_PM_OPS(cs35l56_system_suspend_late, cs35l56_system_resume_early) - NOIRQ_SYSTEM_SLEEP_PM_OPS(cs35l56_system_suspend_no_irq, cs35l56_system_resume_no_irq) }; #endif diff --git a/sound/soc/codecs/cs35l56.h b/sound/soc/codecs/cs35l56.h index 8a987ec01507..0ebae11c1cd0 100644 --- a/sound/soc/codecs/cs35l56.h +++ b/sound/soc/codecs/cs35l56.h @@ -16,10 +16,6 @@ #include <sound/cs35l56.h> #include "wm_adsp.h" -#define CS35L56_SDW_GEN_INT_STAT_1 0xc0 -#define CS35L56_SDW_GEN_INT_MASK_1 0xc1 -#define CS35L56_SDW_INT_MASK_CODEC_IRQ BIT(0) - #define CS35L56_SDW_INVALID_BUS_SCALE 0xf #define CS35L56_RX_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE) @@ -38,8 +34,6 @@ struct cs35l56_private { struct snd_soc_component *component; struct regulator_bulk_data supplies[CS35L56_NUM_BULK_SUPPLIES]; struct sdw_slave *sdw_peripheral; - struct work_struct sdw_irq_work; - bool sdw_irq_no_unmask; bool soft_resetting; bool sdw_attached; struct completion init_completion; @@ -57,12 +51,8 @@ extern const struct dev_pm_ops cs35l56_pm_ops_i2c_spi; int cs35l56_system_suspend(struct device *dev); int cs35l56_system_suspend_late(struct device *dev); -int cs35l56_system_suspend_no_irq(struct device *dev); -int cs35l56_system_resume_no_irq(struct device *dev); int cs35l56_system_resume_early(struct device *dev); int cs35l56_system_resume(struct device *dev); -irqreturn_t cs35l56_irq(int irq, void *data); -int cs35l56_irq_request(struct cs35l56_base *cs35l56_base, int irq); int cs35l56_common_probe(struct cs35l56_private *cs35l56); int cs35l56_init(struct cs35l56_private *cs35l56); void cs35l56_remove(struct cs35l56_private *cs35l56); -- 2.43.0

2 5

[PATCH v26 00/33] Introduce QC USB SND audio offloading support
by Wesley Cheng 04 Sep '24

04 Sep '24

Requesting to see if we can get some Acked-By tags, and merge on usb-next. Several Qualcomm based chipsets can support USB audio offloading to a dedicated audio DSP, which can take over issuing transfers to the USB host controller. The intention is to reduce the load on the main processors in the SoC, and allow them to be placed into lower power modes. There are several parts to this design: 1. Adding ASoC binding layer 2. Create a USB backend for Q6DSP 3. Introduce XHCI interrupter support 4. Create vendor ops for the USB SND driver USB | ASoC -------------------------------------------------------------------- | _________________________ | |sm8250 platform card | | |_________________________| | | | | ___V____ ____V____ | |Q6USB | |Q6AFE | | |"codec" | |"cpu" | | |________| |_________| | ^ ^ ^ | | |________| | ___V____ | | |SOC-USB | | ________ ________ | | | |USB SND |<--->|QC offld|<------------>|________| | |(card.c)| | |<---------- | |________| |________|___ | | | ^ ^ | | | ____________V_________ | | | | | |APR/GLINK | __ V_______________V_____ | | | |______________________| |USB SND (endpoint.c) | | | | ^ |_________________________| | | | | ^ | | | ___________V___________ | | | |->|audio DSP | ___________V_____________ | | |_______________________| |XHCI HCD |<- | |_________________________| | Adding ASoC binding layer ========================= soc-usb: Intention is to treat a USB port similar to a headphone jack. The port is always present on the device, but cable/pin status can be enabled/disabled. Expose mechanisms for USB backend ASoC drivers to communicate with USB SND. Create a USB backend for Q6DSP ============================== q6usb: Basic backend driver that will be responsible for maintaining the resources needed to initiate a playback stream using the Q6DSP. Will be the entity that checks to make sure the connected USB audio device supports the requested PCM format. If it does not, the PCM open call will fail, and userspace ALSA can take action accordingly. Introduce XHCI interrupter support ================================== XHCI HCD supports multiple interrupters, which allows for events to be routed to different event rings. This is determined by "Interrupter Target" field specified in Section "6.4.1.1 Normal TRB" of the XHCI specification. Events in the offloading case will be routed to an event ring that is assigned to the audio DSP. Create vendor ops for the USB SND driver ======================================== qc_audio_offload: This particular driver has several components associated with it: - QMI stream request handler - XHCI interrupter and resource management - audio DSP memory management When the audio DSP wants to enable a playback stream, the request is first received by the ASoC platform sound card. Depending on the selected route, ASoC will bring up the individual DAIs in the path. The Q6USB backend DAI will send an AFE port start command (with enabling the USB playback path), and the audio DSP will handle the request accordingly. Part of the AFE USB port start handling will have an exchange of control messages using the QMI protocol. The qc_audio_offload driver will populate the buffer information: - Event ring base address - EP transfer ring base address and pass it along to the audio DSP. All endpoint management will now be handed over to the DSP, and the main processor is not involved in transfers. Overall, implementing this feature will still expose separate sound card and PCM devices for both the platform card and USB audio device: 0 [SM8250MTPWCD938]: sm8250 - SM8250-MTP-WCD9380-WSA8810-VA-D SM8250-MTP-WCD9380-WSA8810-VA-DMIC 1 [Audio ]: USB-Audio - USB Audio Generic USB Audio at usb-xhci-hcd.1.auto-1.4, high speed This is to ensure that userspace ALSA entities can decide which route to take when executing the audio playback. In the above, if card#1 is selected, then USB audio data will take the legacy path over the USB PCM drivers, etc... The current limitation is that the latest USB audio device that is identified will be automatically selected by the Q6USB BE DAI for offloading. Future patches can be added to possibly add for more flexibility, but until the userpace applications can be better defined, having these mechanisms will complicate the overall implementation. USB offload Kcontrols ===================== Part of the vendor offload package will have a mixer driver associated with it (mixer_usb_offload.c). This entity will be responsible for coordinating with SOC USB and the Q6USB backend DAI to fetch information about the sound card and PCM device indices associated with the offload path. The logic is done based on the current implementation of how paths are controlled within the QC ASoC implementation. QC ASoC Q6Routing ----------------- Within the Q6 ASOC design, the registered ASoC platform card will expose a set of kcontrols for enabling the BE DAI links to the FE DAI link. For example: tinymix -D 0 contents Number of controls: 1033 ctl type num name value ... 1025 BOOL 1 USB Mixer MultiMedia1 Off 1026 BOOL 1 USB Mixer MultiMedia2 Off 1027 BOOL 1 USB Mixer MultiMedia3 Off 1028 BOOL 1 USB Mixer MultiMedia4 Off 1029 BOOL 1 USB Mixer MultiMedia5 Off 1030 BOOL 1 USB Mixer MultiMedia6 Off 1031 BOOL 1 USB Mixer MultiMedia7 Off 1032 BOOL 1 USB Mixer MultiMedia8 Off Each of these kcontrols will enable the USB BE DAI link (q6usb) to be connected to a FE DAI link (q6asm). Since each of these controls are DAPM widgets, when it is enabled, the DAPM widget's "connect" flag is updated accordingly. USB Offload Mapping ------------------- Based on the Q6routing, the USB BE DAI link can determine which sound card and PCM device is enabled for offloading. Fetching the ASoC platform sound card's information is fairly straightforward, and the bulk of the work goes to finding the corresponding PCM device index. As mentioned above, the USB BE DAI can traverse the DAPM widgets to find the DAPM path that is related to the control for the "USB Mixer." Based on which "USB Mixer" is enabled, it can find the corresponding DAPM widget associated w/ the FE DAI link (Multimedia*). From there it can find the PCM device created for the Multimedia* stream. Only one BE DAI link can be enabled per FE DAI. For example, if the HDMI path is enabled for Multimedia1, the USB Mixer will be disabled and switched over. Examples of kcontrol -------------------- tinymix -D 0 contents Number of controls: 1033 ctl type num name ... 1025 BOOL 1 USB Mixer MultiMedia1 Off 1026 BOOL 1 USB Mixer MultiMedia2 On 1027 BOOL 1 USB Mixer MultiMedia3 Off 1028 BOOL 1 USB Mixer MultiMedia4 Off 1029 BOOL 1 USB Mixer MultiMedia5 Off 1030 BOOL 1 USB Mixer MultiMedia6 Off 1031 BOOL 1 USB Mixer MultiMedia7 Off 1032 BOOL 1 USB Mixer MultiMedia8 Off tinymix -D 2 contents Number of controls: 7 ctl type num name value 0 INT 2 Playback Channel Map 0, 0 (range 0->36) 1 BOOL 2 MDR-1ADAC Playback Switch On, On 2 BOOL 1 MDR-1ADAC Playback Switch On 3 INT 2 MDR-1ADAC Playback Volume 127, 127 (range 0->127) 4 INT 1 MDR-1ADAC Playback Volume 127 (range 0->127) 5 BOOL 1 Sony Internal Clock Validity On 6 INT 2 USB Offload Playback Route PCM#0 0, 1 (range -1->255) The example highlights that the userspace/application can utilize the offload path for the USB device on card#0 PCM device#1. When dealing with multiple USB audio devices, only the latest USB device identified is going to be selected for offload capable. tinymix -D 1 contents Number of controls: 9 ctl type num name value 0 INT 2 Capture Channel Map 0, 0 (range 0->36) 1 INT 2 Playback Channel Map 0, 0 (range 0->36) 2 BOOL 1 Headset Capture Switch On 3 INT 1 Headset Capture Volume 1 (range 0->4) 4 BOOL 1 Sidetone Playback Switch On 5 INT 1 Sidetone Playback Volume 4096 (range 0->8192) 6 BOOL 1 Headset Playback Switch On 7 INT 2 Headset Playback Volume 20, 20 (range 0->24) 8 INT 2 USB Offload Playback Route PCM#0 -1, -1 (range -1->255) "-1, -1" shows that this device has no route to the offload path. This feature was validated using: - tinymix: set/enable the multimedia path to route to USB backend - tinyplay: issue playback on platform card Changelog -------------------------------------------- Changes in v26: - Cleaned up drivers based on errors from checkpatch - Fixed several typos using codespell - Removed any vendor specific notation from USB SND offload mixer patch Changes in v25: - Cleanups on typos mentioned within the xHCI layers - Modified the xHCI interrupter search if clients specify interrupter index - Moved mixer_usb_offload into its own module, so that other vendor offload USB modules can utilize it also. - Added support for USB audio devices that may have multiple PCM streams, as previous implementation only assumed a single PCM device. SOC USB will be able to handle an array of PCM indexes supported by the USB audio device. - Added some additional checks in the QC USB offload driver to check that device has at least one playback stream before allowing to bind - Reordered DT bindings to fix the error found by Rob's bot. The patch that added USB_RX was after the example was updated. - Updated comments within SOC USB to clarify terminology and to keep it consistent - Added SND_USB_JACK type for notifying of USB device audio connections Changes in v24: - Simplified the kcontrols involved in determining how to utilize the offload path. - There is one kcontrol registered to each USB audio device that will output which card/pcm device it is mapped to for the offload route. - Removed kcontrols to track offload status and device selection. - Default to last USB audio device plugged in as offload capable. - kcontrol will reside on USB SND device. - Reworked the tracking of connected USB devices from the Q6USB BE DAI link. Previously, it was convoluted by doing it over an array, but moved to using a list made it much simpler. Logic is still unchanged in that the last USB headset plugged in will be selected for offloading. - Updated the USB SOC RST documentation accordingly with new kcontrol updates. - Added logic to fetch mapped ASoC card and pcm device index that the offload path is mapped to for the USB SND kcontrol (for offload route). - Re-ordered series to hopefully make reviews more readable by combining patches based on the layer modified (ie QC ASoC, ASoC, USB sound, and USB XHCI). Changes in v23: - Added MODULE_DESCRIPTION() fields to drivers that needed it. Changes in v22: - Removed components tag for the ASoC platform card, as the USB SND kcontrol for notifying userspace of offload capable card achieves similar results. - Due to the above, had to remove the review-by tag for the RST documentation, as changes were made to remove the components tag section. - Took in feedback to make the SOC USB add/remove ports void. - Fixed an issue w/ the USB SND kcontrol management for devices that have multi UAC interfaces. (would attempt to create the kcontrol more than once) - Modified SOC USB card and PCM index select to be based off the num_supported streams that is specified by the USB BE DAI. - Modified comments on selecting the latest USB headset for offloading. Changes in v21: - Added an offload jack disable path from the ASoC platform driver and SOC USB. - Refactored some of the existing SOC USB context look up APIs and created some new helpers to search for the USB context. - Renamed snd_soc_usb_find_format to snd_soc_usb_find_supported_format - Removed some XHCI sideband calls that would allow clients to actually enable the IRQ line associated w/ the secondary interrupter. This is removed because there are other dependencies that are required for that to happen, which are not covered as part of this series, and to avoid confusion. - Due to the above, removed the need to export IMOD setting, and enable/disable interrupter APIs. Changes in v20: - Fixed up some formatting changes pointed out in the usb.rst - Added SB null check during XHCI sideband unregister in case caller passes improper argument (xhci_sideband_unregister()) Changes in v19: - Rebased to usb-next to account for some new changes in dependent drivers. Changes in v18: - Rebased to usb-next, which merged in part of the series. Removed these patches. - Reworked Kconfigs for the ASoC USB related components from QCOM Q6DSP drivers to keep dependencies in place for SoC USB and USB SND. - Removed the repurposing of the stop ep sync API into existing XHCI operations. This will be solely used by the XHCI sideband for now. Changes in v17: - Fixed an issue where one patch was squashed into another. - Re-added some kconfig checks for helpers exposed in USB SND for the soc usb driver, after running different kconfigs. Changes in v16: - Modified some code layer dependencies so that soc usb can be split as a separate module. - Split the kcontrols from ASoC QCOM common layer into a separate driver - Reworked SOC USB kcontrols for controlling card + pcm offload routing and status so that there are individual controls for card and pcm devices. - Added a kcontrol remove API in SOC USB to remove the controls on the fly. This required to add some kcontrol management to SOC USB. - Removed the disconnect work and workqueue for the QC USB offload as it is not required, since QMI interface driver ensures events are handled in its own WQ. Changes in v15: - Removed some already merged XHCI changes - Separated SOC USB driver from being always compiled into SOC core. Now configurable from kconfig. - Fixed up ASoC kcontrol naming to fit guidelines. - Removed some unnecessary dummy ifdefs. - Moved usb snd offload capable kcontrol to be initialized by the platform offloading driver. Changes in v14: - Cleaned up some USB SND related feedback: - Renamed SNDUSB OFFLD playback available --> USB offload capable card - Fixed locking while checking if stream is in use - Replaced some mutex pairs with guard(mutex) Changes in v13: - Pulled in secondary/primary interrupter rework from Mathias from: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/drivers… - Did some cleanup and commit message updates, and tested on current code base. - Added mutex locking to xhci sideband to help prevent any race conditions, esp. for when accessing shared references. - Addressed concerns from Hillf about gfp_flags and locking used in qc_usb_audio_offload. - Rebased onto usb-next Changes in v12: - Updated copyright year to 2024. Happy new years! - Fixed newline format on mixer offload driver. Changes in v11: - Modified QMI format structures to be const Changes in v10: - Added new mixer for exposing kcontrol for sound card created by USB SND. This allows for applications to know which platform sound card has offload support. Will return the card number. - Broke down and cleaned up some functions/APIs within qc_audio_offload driver. - Exported xhci_initialize_ring_info(), and modified XHCI makefile to allow for the XHCI sideband to exist as a module. - Reworked the jack registration and moved it to the QCOM platform card driver, ie sm8250. - Added an SOC USB API to fetch a standard component tag that can be appended to the platform sound card. Added this tag to sm8250 if any USB path exists within the DT node. - Moved kcontrols that existed in the Q6USB driver, and made it a bit more generic, so that naming can be standardized across solutions. SOC USB is now responsible for creation of these kcontrols. - Added a SOC USB RST document explaining some code flows and implementation details so that other vendors can utilize the framework. - Addressed a case where USB device connection events are lost if usb offload driver (qc_audio_offload) is not probed when everything else has been initialized, ie USB SND, SOC USB and ASoC sound card. Add a rediscover device call during module init, to ensure that connection events will be propagated. - Rebased to usb-next. Changes in v9: - Fixed the dt binding check issue with regards to num-hc-interrupters. Changes in v8: - Cleaned up snd_soc_usb_find_priv_data() based on Mark's feedback. Removed some of the duplicate looping code that was present on previous patches. Also renamed the API. - Integrated Mathias' suggestions on his new sideband changes: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/?h=feat… - Addressed some of Mathias' fixme tags, such as: - Resetting transfer ring dequeue/enqueue pointers - Issuing stop endpoint command during ep removal - Reset ERDP properly to first segment ring during interrupter removal. (this is currently just being cleared to 0, but should be pointing to a valid segment if controller is still running. Changes in v7: - Fixed dt check error for q6usb bindings - Updated q6usb property from qcom,usb-audio-intr-num --> qcom,usb-audio-intr-idx - Removed separate DWC3 HC interrupters num property, and place limits to XHCI one. - Modified xhci_ring_to_sgtable() to use assigned IOVA/DMA address to fetch pages, as it is not ensured event ring allocated is always done in the vmalloc range. Changes in v6: - Fixed limits and description on several DT bindings (XHCI and Q6USB) - Fixed patch subjects to follow other ALSA/ASoC notations. USB SND - Addressed devices which expose multiple audio (UAC) interfaces. These devices will create a single USB sound card with multiple audio streams, and receive multiple interface probe routines. QC offload was not properly considering cases with multiple probe calls. - Renamed offload module name and kconfig to fit within the SND domain. - Renamed attach/detach endpoint API to keep the hw_params notation. Changes in v5: - Removed some unnecessary files that were included - Fixed some typos mentioned - Addressed dt-binding issues and added hc-interrupters definition to usb-xhci.yaml XHCI: - Moved secondary skip events API to xhci-ring and updated implementation - Utilized existing XHCI APIs, such as inc_deq and xhci_update_erst_dequeue() USB SND - Renamed and reworked the APIs in "sound: usb: Export USB SND APIs for modules" patch to include suggestions to utilize snd_usb_hw_params/free and to avoid generic naming. - Added a resume_cb() op for completion sake. - Addressed some locking concerns with regards to when registering for platform hooks. - Added routine to disconnect all offloaded devices during module unbind. ASoC - Replaced individual PCM parameter arguments in snd_soc_usb_connect() with new snd_soc_usb_device structure to pass along PCM info. - Modified snd_jack set report to notify HEADPHONE event, as we do not support record path. Changes in v4: - Rebased to xhci/for-usb-next - Addressed some dt-bindings comments XHCI: - Pulled in latest changes from Mathias' feature_interrupters branch: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/?h=feat… - Fixed commit text and signage for the XHCI sideband/interrupter related changes - Added some logic to address the FIXME tags mentioned throughout the commits, such as handling multi segment rings and building the SGT, locking concerns, and ep cleanup operations. - Removed some fixme tags for conditions that may not be needed/addressed. - Repurposed the new endpoint stop sync API to be utilized in other places. - Fixed potential compile issue if XHCI sideband config is not defined. ASoC: - Added sound jack control into the Q6USB driver. Allows for userspsace to know when an offload capable device is connected. USB SND: - Avoided exporting _snd_pcm_hw_param_set based on Takashi's recommendation. - Split USB QMI packet header definitions into a separate commit. This is used to properly allow the QMI interface driver to parse and route QMI packets accordingly - Added a "depends on" entry when enabling QC audio offload to avoid compile time issues. Changes in v3: - Changed prefix from RFC to PATCH - Rebased entire series to usb-next - Updated copyright years XHCI: - Rebased changes on top of XHCI changes merged into usb-next, and only added changes that were still under discussion. - Added change to read in the "num-hc-interrupters" device property. ASoC: - qusb6 USB backend - Incorporated suggestions to fetch iommu information with existing APIs - Added two new sound kcontrols to fetch offload status and offload device selection. - offload status - will return the card and pcm device in use tinymix -D 0 get 1 --> 1, 0 (offload in progress on card#1 pcm#0) - device selection - set the card and pcm device to enable offload on. Ex.: tinymix -D 0 set 1 2 0 --> sets offload on card#2 pcm#0 (this should be the USB card) USB SND: - Fixed up some locking related concerns for registering platform ops. - Moved callbacks under the register_mutex, so that - Modified APIs to properly pass more information about the USB SND device, so that the Q6USB backend can build a device list/map, in order to monitor offload status and device selection. Changes in v2: XHCI: - Replaced XHCI and HCD changes with Mathias' XHCI interrupter changes in his tree: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/?h=feat… Adjustments made to Mathias' changes: - Created xhci-intr.h to export/expose interrupter APIs versus exposing xhci.h. Moved dependent structures to this file as well. (so clients can parse out information from "struct xhci_interrupter") - Added some basic locking when requesting interrupters. - Fixed up some sanity checks. - Removed clearing of the ERSTBA during freeing of the interrupter. (pending issue where SMMU fault occurs if DMA addr returned is 64b - TODO) - Clean up pending events in the XHCI secondary interrupter. While testing USB bus suspend, it was seen that on bus resume, the xHCI HC would run into a command timeout. - Added offloading APIs to xHCI to fetch transfer and event ring information. ASoC: - Modified soc-usb to allow for multiple USB port additions. For this to work, the USB offload driver has to have a reference to the USB backend by adding a "usb-soc-be" DT entry to the device saved into XHCI sysdev. - Created separate dt-bindings for defining USB_RX port. - Increased APR timeout to accommodate the situation where the AFE port start command could be delayed due to having to issue a USB bus resume while handling the QMI stream start command. Mathias Nyman (2): xhci: add helper to stop endpoint and wait for completion xhci: sideband: add initial api to register a sideband entity Wesley Cheng (31): usb: host: xhci: Repurpose event handler for skipping interrupter events usb: xhci: Allow for secondary interrupter to set IMOD usb: host: xhci-mem: Cleanup pending secondary event ring events usb: host: xhci-mem: Allow for interrupter clients to choose specific index usb: host: xhci-plat: Set XHCI max interrupters if property is present usb: dwc3: Specify maximum number of XHCI interrupters ALSA: Add USB audio device jack type ALSA: usb-audio: Export USB SND APIs for modules ALSA: usb-audio: Check for support for requested audio format ASoC: Add SOC USB APIs for adding an USB backend ASoC: usb: Add PCM format check API for USB backend ASoC: usb: Create SOC USB SND jack kcontrol ASoC: usb: Fetch ASoC card and pcm device information ASoC: doc: Add documentation for SOC USB ASoC: dt-bindings: qcom,q6dsp-lpass-ports: Add USB_RX port ASoC: dt-bindings: Update example for enabling USB offload on SM8250 ASoC: qcom: qdsp6: Introduce USB AFE port to q6dsp ASoC: qcom: qdsp6: q6afe: Increase APR timeout ASoC: qcom: qdsp6: Add USB backend ASoC driver for Q6 ASoC: qcom: qdsp6: Add headphone jack for offload connection status ASoC: qcom: qdsp6: Fetch USB offload mapped card and PCM device ALSA: usb-audio: Introduce USB SND platform op callbacks ALSA: usb-audio: Save UAC sample size information ALSA: usb-audio: Prevent starting of audio stream if in use ALSA: usb-audio: qcom: Add USB QMI definitions ALSA: usb-audio: qcom: Introduce QC USB SND offloading support ALSA: usb-audio: qcom: Don't allow USB offload path if PCM device is in use ALSA: usb-audio: qcom: Use card and PCM index from QMI request ALSA: usb-audio: Add USB offload route kcontrol ALSA: usb-audio: Allow for rediscovery of connected USB SND devices ASoC: usb: Rediscover USB SND devices on USB port add .../bindings/sound/qcom,sm8250.yaml | 15 + Documentation/sound/soc/index.rst | 1 + Documentation/sound/soc/usb.rst | 429 ++++ drivers/usb/dwc3/core.c | 12 + drivers/usb/dwc3/core.h | 2 + drivers/usb/dwc3/host.c | 3 + drivers/usb/host/Kconfig | 9 + drivers/usb/host/Makefile | 2 + drivers/usb/host/xhci-mem.c | 34 +- drivers/usb/host/xhci-plat.c | 2 + drivers/usb/host/xhci-ring.c | 54 +- drivers/usb/host/xhci-sideband.c | 425 ++++ drivers/usb/host/xhci.c | 45 +- drivers/usb/host/xhci.h | 19 +- .../sound/qcom,q6dsp-lpass-ports.h | 1 + include/linux/mod_devicetable.h | 2 +- include/linux/usb/xhci-sideband.h | 71 + include/sound/jack.h | 4 +- include/sound/q6usboffload.h | 20 + include/sound/soc-usb.h | 135 ++ include/uapi/linux/input-event-codes.h | 3 +- sound/core/jack.c | 6 +- sound/soc/Kconfig | 10 + sound/soc/Makefile | 2 + sound/soc/qcom/Kconfig | 15 + sound/soc/qcom/Makefile | 2 + sound/soc/qcom/qdsp6/Makefile | 1 + sound/soc/qcom/qdsp6/q6afe-dai.c | 60 + sound/soc/qcom/qdsp6/q6afe.c | 194 +- sound/soc/qcom/qdsp6/q6afe.h | 36 +- sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c | 23 + sound/soc/qcom/qdsp6/q6dsp-lpass-ports.h | 1 + sound/soc/qcom/qdsp6/q6routing.c | 9 + sound/soc/qcom/qdsp6/q6usb.c | 391 ++++ sound/soc/qcom/sm8250.c | 24 +- sound/soc/qcom/usb_offload_utils.c | 56 + sound/soc/qcom/usb_offload_utils.h | 30 + sound/soc/soc-usb.c | 338 +++ sound/usb/Kconfig | 25 + sound/usb/Makefile | 4 +- sound/usb/card.c | 110 + sound/usb/card.h | 17 + sound/usb/endpoint.c | 1 + sound/usb/format.c | 1 + sound/usb/helper.c | 1 + sound/usb/mixer_usb_offload.c | 102 + sound/usb/mixer_usb_offload.h | 17 + sound/usb/pcm.c | 104 +- sound/usb/pcm.h | 11 + sound/usb/qcom/Makefile | 2 + sound/usb/qcom/qc_audio_offload.c | 1968 +++++++++++++++++ sound/usb/qcom/usb_audio_qmi_v01.c | 863 ++++++++ sound/usb/qcom/usb_audio_qmi_v01.h | 164 ++ 53 files changed, 5820 insertions(+), 56 deletions(-) create mode 100644 Documentation/sound/soc/usb.rst create mode 100644 drivers/usb/host/xhci-sideband.c create mode 100644 include/linux/usb/xhci-sideband.h create mode 100644 include/sound/q6usboffload.h create mode 100644 include/sound/soc-usb.h create mode 100644 sound/soc/qcom/qdsp6/q6usb.c create mode 100644 sound/soc/qcom/usb_offload_utils.c create mode 100644 sound/soc/qcom/usb_offload_utils.h create mode 100644 sound/soc/soc-usb.c create mode 100644 sound/usb/mixer_usb_offload.c create mode 100644 sound/usb/mixer_usb_offload.h create mode 100644 sound/usb/qcom/Makefile create mode 100644 sound/usb/qcom/qc_audio_offload.c create mode 100644 sound/usb/qcom/usb_audio_qmi_v01.c create mode 100644 sound/usb/qcom/usb_audio_qmi_v01.h

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[PATCH] ASoC: dt-bindings: renesas,rsnd: add post-init-providers property
by Kuninori Morimoto 04 Sep '24

04 Sep '24

sometimes, fwnode dependency cycles can be broken, and in such case, rsnd driver will never be probed. To avoid such case, we would like to use post-init-providers to avoid it. Enable post-init-providers property on rsnd. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- Documentation/devicetree/bindings/sound/renesas,rsnd.yaml | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml b/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml index 07ec6247d9def..cb44478d9e420 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml @@ -112,6 +112,10 @@ properties: description: List of necessary clock names. # details are defined below + post-init-providers: + description: ignore fw_devlink dependency cycles and avoid driver probe dead-lock + $ref: /schemas/types.yaml#/definitions/phandle-array + # ports is below port: $ref: audio-graph-port.yaml#/definitions/port-base -- 2.43.0

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[PATCH 00/12] Add i2s/tdm support for acp7.0 and acp7.1 platforms
by Venkata Prasad Potturu 04 Sep '24

04 Sep '24

1. Refactor acp generic driver to support all platforms. 2. Add i2s/tdm and support for acp7.0 and acp7.1 platforms. Venkata Prasad Potturu (12): ASoC: amd: acp: Refactor TDM slots selction based on acp revision id ASoC: amd: acp: Refactor I2S dai driver ASoC: amd: acp: Update pcm hardware capabilities for acp6.3 platform ASoC: amd: acp: Add I2S TDM support for acp6.3 platform ASoC: amd: acp: Update pcm hardware capabilities for acp7.0 platform ASoC: amd: acp: Add I2S master clock generation support for acp7.0 platform ASoC: amd: acp: Set i2s clock for acp7.0 platform ASoC: amd: acp: Modify max channels and sample rate support for acp70 dai driver ASoC: amd: acp: Add I2S TDM support for acp7.0 platform ASoC: amd: acp: Add pte configuration for ACP7.0 platform ASoC: amd: acp: Add i2s master clock generation support for acp7.1 platform ASoC: amd: acp: Add I2S TDM support for acp7.1 platform sound/soc/amd/acp/acp-i2s.c | 188 ++++++++++++++++++++------ sound/soc/amd/acp/acp-legacy-common.c | 32 ++--- sound/soc/amd/acp/acp-pdm.c | 7 +- sound/soc/amd/acp/acp-platform.c | 117 ++++++++++++++-- sound/soc/amd/acp/acp70.c | 73 +++++++--- sound/soc/amd/acp/amd.h | 24 ++-- sound/soc/amd/acp/chip_offset_byte.h | 84 +++++++----- 7 files changed, 387 insertions(+), 138 deletions(-) -- 2.39.2

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[PATCH] ALSA: hda/realtek: Support mute LED on HP Laptop 14-dq2xxx
by Maximilien Perreault 04 Sep '24

04 Sep '24

The mute LED on this HP laptop uses ALC236 and requires a quirk to function. This patch enables the existing quirk for the device. Signed-off-by: Maximilien Perreault <maximilienperreault(a)gmail.com> --- 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 f67190a8c..6f96c0c67 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10183,6 +10183,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x87f5, "HP", ALC287_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x87f6, "HP Spectre x360 14", ALC245_FIXUP_HP_X360_AMP), SND_PCI_QUIRK(0x103c, 0x87f7, "HP Spectre x360 14", ALC245_FIXUP_HP_X360_AMP), + SND_PCI_QUIRK(0x103c, 0x87fd, "HP Laptop 14-dq2xxx", ALC236_FIXUP_HP_MUTE_LED_COEFBIT2), SND_PCI_QUIRK(0x103c, 0x87fe, "HP Laptop 15s-fq2xxx", ALC236_FIXUP_HP_MUTE_LED_COEFBIT2), SND_PCI_QUIRK(0x103c, 0x8805, "HP ProBook 650 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x880d, "HP EliteBook 830 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), -- 2.46.0

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[PATCH] soc: qcom: apr: Fix some parenthesis alignment coding style issues
by Riyan Dhiman 04 Sep '24

04 Sep '24

Adhere to Linux kernel coding style. Issue reported by checkpatch: - CHECK: Alignment should match open parenthesis It has no functional changes. Signed-off-by: Riyan Dhiman <riyandhiman14(a)gmail.com> --- drivers/soc/qcom/apr.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/drivers/soc/qcom/apr.c b/drivers/soc/qcom/apr.c index 4fbff3a890e2..db1326bdb170 100644 --- a/drivers/soc/qcom/apr.c +++ b/drivers/soc/qcom/apr.c @@ -88,7 +88,7 @@ void gpr_free_port(gpr_port_t *port) EXPORT_SYMBOL_GPL(gpr_free_port); gpr_port_t *gpr_alloc_port(struct apr_device *gdev, struct device *dev, - gpr_port_cb cb, void *priv) + gpr_port_cb cb, void *priv) { struct packet_router *pr = dev_get_drvdata(gdev->dev.parent); gpr_port_t *port; @@ -159,7 +159,7 @@ static void apr_dev_release(struct device *dev) } static int apr_callback(struct rpmsg_device *rpdev, void *buf, - int len, void *priv, u32 addr) + int len, void *priv, u32 addr) { struct packet_router *apr = dev_get_drvdata(&rpdev->dev); struct apr_rx_buf *abuf; @@ -222,9 +222,9 @@ static int apr_do_rx_callback(struct packet_router *apr, struct apr_rx_buf *abuf } if (hdr->src_domain >= APR_DOMAIN_MAX || - hdr->dest_domain >= APR_DOMAIN_MAX || - hdr->src_svc >= APR_SVC_MAX || - hdr->dest_svc >= APR_SVC_MAX) { + hdr->dest_domain >= APR_DOMAIN_MAX || + hdr->src_svc >= APR_SVC_MAX || + hdr->dest_svc >= APR_SVC_MAX) { dev_err(apr->dev, "APR: Wrong APR header\n"); return -EINVAL; } -- 2.46.0

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Qestion: of property has dead-lock to call .probe()
by Kuninori Morimoto 04 Sep '24

04 Sep '24

Hi Saravana, Rob Cc Mark/ALSA-ML, Geert, Laurent I got dead-lock issue on drivers/of/property.c to call .probe() I'm now using v6.11-rc5. I'm using 3 devices, and these are connected via OF-Graph. These are connected like this so far. Let's call it as [Style-A] [Style-A] +---+ +---+ |(A)+--+ | +-+-+ | | | |(B)| +-+-+ | | |(C)+--+ | +---+ +---+ These are using port/endpoint to connect. It works well for now, no issue I have so far. (B) is connector for (A)-(C) data, thus, (A) and (C) should be probed first, and (B) needs probe later (For ALSA member, this (B) is Audio Graph Card2). Here, (A)-(C) can connect directly for some data. Now it is using both data connection (= (A)-(B)-(C) and (A)-(C)) Now, I want to disconnect (A)-(C) connection, like below Let's call it as [Style-B] [Style-B] +---+ +---+ |(A)+--+ | +---+ | | |(B)| +---+ | | |(C)+--+ | +---+ +---+ Then, it seems dead-lock happen. In my debug, it seems... - (B) is handled as supplier for (A). - (B) probe() is called, but it needs (A) info which is not yet probed. So it returns -EPROBE_DEFER. - Because (B) is not probed, (A) probe() never called In [Style-A], it seems __fwnode_link_cycle() is called for (A)-(B) connection, so the dead-lock will be solved. But it is not called for [Style-B]. Because of that (B) is always handled as supplier for (A). If I used below patch, and use "non-supplier" property on (B), this dead-lock issue was solved. But I know this is not a good solution. I think it is very normal connection, not super special. How can I solve this issue on correct way ? Or how can I indicate you my issue more detail ? I can add debug patch and test it if you can indicate it to me. ----------------------------------------------- diff --git a/drivers/of/property.c b/drivers/of/property.c index def152c61049..2f08210c2ea4 100644 --- a/drivers/of/property.c +++ b/drivers/of/property.c @@ -1502,11 +1502,22 @@ static struct device_node *parse_remote_endpoint(struct device_node *np, const char *prop_name, int index) { + struct device_node *node; + /* Return NULL for index > 0 to signify end of remote-endpoints. */ if (index > 0 || strcmp(prop_name, "remote-endpoint")) return NULL; - return of_graph_get_remote_port_parent(np); + node = of_graph_get_remote_port_parent(np); + + /* + * There is clearly non-supplier node which is connected via "remote-endpoint". + * Ignore it, otherwise dead-lock might occur + */ + if (of_property_present(node, "non-supplier")) + return NULL; + + return node; } ----------------------------------------------- Thank you for your help !! Best regards --- Kuninori Morimoto

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[PATCH] ASoC: tlv320aic32x4: Add multi endpoint support
by Marek Vasut 04 Sep '24

04 Sep '24

Support multiple endpoints on TLV320AIC32x4 codec port when used in of_graph context. This patch allows to share the codec port between two CPU DAIs. Example: Custom STM32MP157C board uses TLV320AIC32x4 audio codec. This codec is connected to two serial audio interfaces, which are configured either as rx or tx. From AsoC point of view the topolgy is the following: // 2 CPU DAIs (SAI2A/B), 1 Codec (TLV320AIC32x4) Playback: CPU-A-DAI(slave) -> (master)CODEC-DAI/port0 Record: CPU-B-DAI(slave) <- (master)CODEC-DAI/port0 In the DT two endpoints have to be associated to the codec port: tlv320aic32x4_port: port { tlv320aic32x4_tx_endpoint: endpoint@0 { remote-endpoint = <&sai2a_endpoint>; }; tlv320aic32x4_rx_endpoint: endpoint@1 { remote-endpoint = <&sai2b_endpoint>; }; }; However, when the audio graph card parses the codec nodes, it expects to find DAI interface indexes matching the endpoints indexes. The current patch forces the use of DAI id 0 for both endpoints, which allows to share the codec DAI between the two CPU DAIs for playback and capture streams respectively. Signed-off-by: Marek Vasut <marex(a)denx.de> --- Cc: Baojun Xu <baojun.xu(a)ti.com> Cc: Charles Keepax <ckeepax(a)opensource.cirrus.com> Cc: Christoph Niedermaier <cniedermaier(a)dh-electronics.com> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Kevin Lu <kevin-lu(a)ti.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Marek Vasut <marex(a)denx.de> Cc: Mark Brown <broonie(a)kernel.org> Cc: Philipp Zabel <p.zabel(a)pengutronix.de> Cc: Shenghao Ding <shenghao-ding(a)ti.com> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Cc: kernel(a)dh-electronics.com Cc: linux-sound(a)vger.kernel.org --- sound/soc/codecs/tlv320aic32x4.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index 5c0c81da06dba..54ea4bc58c276 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -1073,6 +1073,13 @@ static int aic32x4_component_probe(struct snd_soc_component *component) return 0; } +static int aic32x4_of_xlate_dai_id(struct snd_soc_component *component, + struct device_node *endpoint) +{ + /* return dai id 0, whatever the endpoint index */ + return 0; +} + static const struct snd_soc_component_driver soc_component_dev_aic32x4 = { .probe = aic32x4_component_probe, .set_bias_level = aic32x4_set_bias_level, @@ -1082,6 +1089,7 @@ static const struct snd_soc_component_driver soc_component_dev_aic32x4 = { .num_dapm_widgets = ARRAY_SIZE(aic32x4_dapm_widgets), .dapm_routes = aic32x4_dapm_routes, .num_dapm_routes = ARRAY_SIZE(aic32x4_dapm_routes), + .of_xlate_dai_id = aic32x4_of_xlate_dai_id, .suspend_bias_off = 1, .idle_bias_on = 1, .use_pmdown_time = 1, @@ -1203,6 +1211,7 @@ static const struct snd_soc_component_driver soc_component_dev_aic32x4_tas2505 = .num_dapm_widgets = ARRAY_SIZE(aic32x4_tas2505_dapm_widgets), .dapm_routes = aic32x4_tas2505_dapm_routes, .num_dapm_routes = ARRAY_SIZE(aic32x4_tas2505_dapm_routes), + .of_xlate_dai_id = aic32x4_of_xlate_dai_id, .suspend_bias_off = 1, .idle_bias_on = 1, .use_pmdown_time = 1, -- 2.45.2

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