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

[PATCH AUTOSEL 6.4 31/54] ASoC: rt711: fix for JD event handling in ClockStop Mode0
by Sasha Levin 13 Aug '23

13 Aug '23

From: Shuming Fan <shumingf(a)realtek.com> [ Upstream commit b69de265bd0e877015a00fbba453ef72af162e0f ] When the system suspends, peripheral Imp-defined interrupt is disabled. When system level resume is invoked, the peripheral Imp-defined interrupts should be enabled to handle JD events. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> Reported-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> Link: https://lore.kernel.org/r/20230721090654.128230-1-shumingf@realtek.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt711-sdw.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt711-sdw.c b/sound/soc/codecs/rt711-sdw.c index 4fe68bcf2a7c2..9545b8a7eb192 100644 --- a/sound/soc/codecs/rt711-sdw.c +++ b/sound/soc/codecs/rt711-sdw.c @@ -541,8 +541,15 @@ static int __maybe_unused rt711_dev_resume(struct device *dev) if (!rt711->first_hw_init) return 0; - if (!slave->unattach_request) + if (!slave->unattach_request) { + if (rt711->disable_irq == true) { + mutex_lock(&rt711->disable_irq_lock); + sdw_write_no_pm(slave, SDW_SCP_INTMASK1, SDW_SCP_INT1_IMPL_DEF); + rt711->disable_irq = false; + mutex_unlock(&rt711->disable_irq_lock); + } goto regmap_sync; + } time = wait_for_completion_timeout(&slave->initialization_complete, msecs_to_jiffies(RT711_PROBE_TIMEOUT)); -- 2.40.1

1 0

[PATCH AUTOSEL 6.4 30/54] ASoC: rt712-sdca: fix for JD event handling in ClockStop Mode0
by Sasha Levin 13 Aug '23

13 Aug '23

From: Shuming Fan <shumingf(a)realtek.com> [ Upstream commit 7a8735c1551e489351172d0da96128f6f8b52b2d ] When the system suspends, peripheral SDCA interrupts are disabled. When system level resume is invoked, the peripheral SDCA interrupts should be enabled to handle JD events. Enable SDCA interrupts in resume sequence when ClockStop Mode0 is applied. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> Reported-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> Link: https://lore.kernel.org/r/20230721090721.128264-1-shumingf@realtek.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt712-sdca-sdw.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt712-sdca-sdw.c b/sound/soc/codecs/rt712-sdca-sdw.c index 3f319459dfec3..1c9e10fea3ddd 100644 --- a/sound/soc/codecs/rt712-sdca-sdw.c +++ b/sound/soc/codecs/rt712-sdca-sdw.c @@ -441,8 +441,16 @@ static int __maybe_unused rt712_sdca_dev_resume(struct device *dev) if (!rt712->first_hw_init) return 0; - if (!slave->unattach_request) + if (!slave->unattach_request) { + if (rt712->disable_irq == true) { + mutex_lock(&rt712->disable_irq_lock); + sdw_write_no_pm(slave, SDW_SCP_SDCA_INTMASK1, SDW_SCP_SDCA_INTMASK_SDCA_0); + sdw_write_no_pm(slave, SDW_SCP_SDCA_INTMASK2, SDW_SCP_SDCA_INTMASK_SDCA_8); + rt712->disable_irq = false; + mutex_unlock(&rt712->disable_irq_lock); + } goto regmap_sync; + } time = wait_for_completion_timeout(&slave->initialization_complete, msecs_to_jiffies(RT712_PROBE_TIMEOUT)); -- 2.40.1

1 0

[PATCH AUTOSEL 6.4 29/54] ASoc: codecs: ES8316: Fix DMIC config
by Sasha Levin 13 Aug '23

13 Aug '23

From: Edgar <ljijcj(a)163.com> [ Upstream commit d20d35d1ad62c6cca36368c1e8f29335a068659e ] According to the datasheet, the DMIC config should be changed to { 0, 2 ,3 } Signed-off-by: Edgar <ljijcj(a)163.com> Link: https://lore.kernel.org/r/20230719054722.401954-1-ljijcj@163.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/es8316.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c index ccecfdf700649..9f5522dee501d 100644 --- a/sound/soc/codecs/es8316.c +++ b/sound/soc/codecs/es8316.c @@ -153,7 +153,7 @@ static const char * const es8316_dmic_txt[] = { "dmic data at high level", "dmic data at low level", }; -static const unsigned int es8316_dmic_values[] = { 0, 1, 2 }; +static const unsigned int es8316_dmic_values[] = { 0, 2, 3 }; static const struct soc_enum es8316_dmic_src_enum = SOC_VALUE_ENUM_SINGLE(ES8316_ADC_DMIC, 0, 3, ARRAY_SIZE(es8316_dmic_txt), -- 2.40.1

1 0

[PATCH AUTOSEL 6.4 28/54] ASoC: rt5682-sdw: fix for JD event handling in ClockStop Mode0
by Sasha Levin 13 Aug '23

13 Aug '23

From: Shuming Fan <shumingf(a)realtek.com> [ Upstream commit 02fb23d72720df2b6be3f29fc5787ca018eb92c3 ] When the system suspends, peripheral Imp-defined interrupt is disabled. When system level resume is invoked, the peripheral Imp-defined interrupts should be enabled to handle JD events. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> Reported-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> Link: https://lore.kernel.org/r/20230721090643.128213-1-shumingf@realtek.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5682-sdw.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5682-sdw.c b/sound/soc/codecs/rt5682-sdw.c index 23f17f70d7e9b..9622aaf1b3e63 100644 --- a/sound/soc/codecs/rt5682-sdw.c +++ b/sound/soc/codecs/rt5682-sdw.c @@ -753,8 +753,15 @@ static int __maybe_unused rt5682_dev_resume(struct device *dev) if (!rt5682->first_hw_init) return 0; - if (!slave->unattach_request) + if (!slave->unattach_request) { + if (rt5682->disable_irq == true) { + mutex_lock(&rt5682->disable_irq_lock); + sdw_write_no_pm(slave, SDW_SCP_INTMASK1, SDW_SCP_INT1_IMPL_DEF); + rt5682->disable_irq = false; + mutex_unlock(&rt5682->disable_irq_lock); + } goto regmap_sync; + } time = wait_for_completion_timeout(&slave->initialization_complete, msecs_to_jiffies(RT5682_PROBE_TIMEOUT)); -- 2.40.1

1 0

[PATCH AUTOSEL 6.4 17/54] ASoC: nau8821: Add DMI quirk mechanism for active-high jack-detect
by Sasha Levin 13 Aug '23

13 Aug '23

From: Edson Juliano Drosdeck <edson.drosdeck(a)gmail.com> [ Upstream commit 1bc40efdaf4a0ccfdb10a1c8e4b458f4764e8e5f ] Add a quirk mechanism to allow specifying that active-high jack-detection should be used on platforms where this info is not available in devicetree. And add an entry for the Positivo CW14Q01P-V2 to the DMI table, so that jack-detection will work properly on this laptop. Signed-off-by: Edson Juliano Drosdeck <edson.drosdeck(a)gmail.com> Link: https://lore.kernel.org/r/20230719200241.4865-1-edson.drosdeck@gmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/nau8821.c | 41 ++++++++++++++++++++++++++++++++++++++ 1 file changed, 41 insertions(+) diff --git a/sound/soc/codecs/nau8821.c b/sound/soc/codecs/nau8821.c index fee970427a243..42de7588fdb68 100644 --- a/sound/soc/codecs/nau8821.c +++ b/sound/soc/codecs/nau8821.c @@ -10,6 +10,7 @@ #include <linux/acpi.h> #include <linux/clk.h> #include <linux/delay.h> +#include <linux/dmi.h> #include <linux/init.h> #include <linux/i2c.h> #include <linux/module.h> @@ -25,6 +26,13 @@ #include <sound/tlv.h> #include "nau8821.h" +#define NAU8821_JD_ACTIVE_HIGH BIT(0) + +static int nau8821_quirk; +static int quirk_override = -1; +module_param_named(quirk, quirk_override, uint, 0444); +MODULE_PARM_DESC(quirk, "Board-specific quirk override"); + #define NAU_FREF_MAX 13500000 #define NAU_FVCO_MAX 100000000 #define NAU_FVCO_MIN 90000000 @@ -1792,6 +1800,33 @@ static int nau8821_setup_irq(struct nau8821 *nau8821) return 0; } +/* Please keep this list alphabetically sorted */ +static const struct dmi_system_id nau8821_quirk_table[] = { + { + /* Positivo CW14Q01P-V2 */ + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Positivo Tecnologia SA"), + DMI_MATCH(DMI_BOARD_NAME, "CW14Q01P-V2"), + }, + .driver_data = (void *)(NAU8821_JD_ACTIVE_HIGH), + }, + {} +}; + +static void nau8821_check_quirks(void) +{ + const struct dmi_system_id *dmi_id; + + if (quirk_override != -1) { + nau8821_quirk = quirk_override; + return; + } + + dmi_id = dmi_first_match(nau8821_quirk_table); + if (dmi_id) + nau8821_quirk = (unsigned long)dmi_id->driver_data; +} + static int nau8821_i2c_probe(struct i2c_client *i2c) { struct device *dev = &i2c->dev; @@ -1812,6 +1847,12 @@ static int nau8821_i2c_probe(struct i2c_client *i2c) nau8821->dev = dev; nau8821->irq = i2c->irq; + + nau8821_check_quirks(); + + if (nau8821_quirk & NAU8821_JD_ACTIVE_HIGH) + nau8821->jkdet_polarity = 0; + nau8821_print_device_properties(nau8821); nau8821_reset_chip(nau8821->regmap); -- 2.40.1

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[PATCH AUTOSEL 6.4 16/54] ASoC: da7219: Check for failure reading AAD IRQ events
by Sasha Levin 13 Aug '23

13 Aug '23

From: Dmytro Maluka <dmy(a)semihalf.com> [ Upstream commit f0691dc16206f21b13c464434366e2cd632b8ed7 ] When handling an AAD interrupt, if IRQ events read failed (for example, due to i2c "Transfer while suspended" failure, i.e. when attempting to read it while DA7219 is suspended, which may happen due to a spurious AAD interrupt), the events array contains garbage uninitialized values. So instead of trying to interprete those values and doing any actions based on them (potentially resulting in misbehavior, e.g. reporting bogus events), refuse to handle the interrupt. Signed-off-by: Dmytro Maluka <dmy(a)semihalf.com> Link: https://lore.kernel.org/r/20230717193737.161784-3-dmy@semihalf.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/da7219-aad.c | 10 +++++++--- 1 file changed, 7 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/da7219-aad.c b/sound/soc/codecs/da7219-aad.c index 07c5d9d354b6d..70c175744772c 100644 --- a/sound/soc/codecs/da7219-aad.c +++ b/sound/soc/codecs/da7219-aad.c @@ -361,11 +361,15 @@ static irqreturn_t da7219_aad_irq_thread(int irq, void *data) struct da7219_priv *da7219 = snd_soc_component_get_drvdata(component); u8 events[DA7219_AAD_IRQ_REG_MAX]; u8 statusa; - int i, report = 0, mask = 0; + int i, ret, report = 0, mask = 0; /* Read current IRQ events */ - regmap_bulk_read(da7219->regmap, DA7219_ACCDET_IRQ_EVENT_A, - events, DA7219_AAD_IRQ_REG_MAX); + ret = regmap_bulk_read(da7219->regmap, DA7219_ACCDET_IRQ_EVENT_A, + events, DA7219_AAD_IRQ_REG_MAX); + if (ret) { + dev_warn_ratelimited(component->dev, "Failed to read IRQ events: %d\n", ret); + return IRQ_NONE; + } if (!events[DA7219_AAD_IRQ_REG_A] && !events[DA7219_AAD_IRQ_REG_B]) return IRQ_NONE; -- 2.40.1

1 0

[PATCH AUTOSEL 6.4 15/54] ASoC: da7219: Flush pending AAD IRQ when suspending
by Sasha Levin 13 Aug '23

13 Aug '23

From: Dmytro Maluka <dmy(a)semihalf.com> [ Upstream commit 91e292917dad64ab8d1d5ca2ab3069ad9dac6f72 ] da7219_aad_suspend() disables jack detection, which should prevent generating new interrupts by DA7219 while suspended. However, there is a theoretical possibility that there is a pending interrupt generated just before suspending DA7219 and not handled yet, so the IRQ handler may still run after DA7219 is suspended. To prevent that, wait until the pending IRQ handling is done. This patch arose as an attempt to fix the following I2C failure occurring sometimes during system suspend or resume: [ 355.876211] i2c_designware i2c_designware.3: Transfer while suspended [ 355.876245] WARNING: CPU: 2 PID: 3576 at drivers/i2c/busses/i2c-designware-master.c:570 i2c_dw_xfer+0x411/0x440 ... [ 355.876462] Call Trace: [ 355.876468] <TASK> [ 355.876475] ? update_load_avg+0x1b3/0x615 [ 355.876484] __i2c_transfer+0x101/0x1d8 [ 355.876494] i2c_transfer+0x74/0x10d [ 355.876504] regmap_i2c_read+0x6a/0x9c [ 355.876513] _regmap_raw_read+0x179/0x223 [ 355.876521] regmap_raw_read+0x1e1/0x28e [ 355.876527] regmap_bulk_read+0x17d/0x1ba [ 355.876532] ? __wake_up+0xed/0x1bb [ 355.876542] da7219_aad_irq_thread+0x54/0x2c9 [snd_soc_da7219 5fb8ebb2179cf2fea29af090f3145d68ed8e2184] [ 355.876556] irq_thread+0x13c/0x231 [ 355.876563] ? irq_forced_thread_fn+0x5f/0x5f [ 355.876570] ? irq_thread_fn+0x4d/0x4d [ 355.876576] kthread+0x13a/0x152 [ 355.876581] ? synchronize_irq+0xc3/0xc3 [ 355.876587] ? kthread_blkcg+0x31/0x31 [ 355.876592] ret_from_fork+0x1f/0x30 [ 355.876601] </TASK> which indicates that the AAD IRQ handler is unexpectedly running when DA7219 is suspended, and as a result, is trying to read data from DA7219 over I2C and is hitting the I2C driver "Transfer while suspended" failure. However, with this patch the above failure is still reproducible. So this patch does not fix any real observed issue so far, but at least is useful for confirming that the above issue is not caused by a pending IRQ but rather looks like a DA7219 hardware issue with an IRQ unexpectedly generated after jack detection is already disabled. Signed-off-by: Dmytro Maluka <dmy(a)semihalf.com> Link: https://lore.kernel.org/r/20230717193737.161784-2-dmy@semihalf.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/da7219-aad.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/da7219-aad.c b/sound/soc/codecs/da7219-aad.c index 993a0d00bc48d..07c5d9d354b6d 100644 --- a/sound/soc/codecs/da7219-aad.c +++ b/sound/soc/codecs/da7219-aad.c @@ -910,6 +910,8 @@ void da7219_aad_suspend(struct snd_soc_component *component) } } } + + synchronize_irq(da7219_aad->irq); } void da7219_aad_resume(struct snd_soc_component *component) -- 2.40.1

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"alsactl restore" does not load the driver state
by GitHub issues - edited 13 Aug '23

13 Aug '23

alsa-project/alsa-utils issue #229 was edited from YuGiOhJCJ: Hello, When I type "alsactl restore", it does not load the driver state. Steps to reproduce: 1. Open a first terminal dedicated to alsa-mixer 2. Open a second terminal where you will type the following commands 3. In the first terminal, press F4, set the audio volume for capture to 40 4. In the second terminal, type "sudo alsactl store" in order to save the driver state to /var/lib/alsa/asound.state 5. In the first terminal, set the audio volume for capture to 35 6. In the second terminal, type "sudo alsactl restore" in order to load the driver state from /var/lib/alsa/asound.state Expected results: I should see in the first terminal that the audio volume for capture is restored to 40. Current results: I see in the first terminal that the audio volume for capture is not restored to 40. It is still set to 35! How to fix that please? Thank you. Best regards. About my system: Packages: - alsa-lib-1.2.9 - alsa-plugins-1.2.7.1 - alsa-utils-1.2.9 Operating system: Slackware 14.2 (64-bit) Architecture: x86_64 Kernel: Linux 5.18.10 ![image](https://github.com/alsa-project/alsa-utils/assets/2247896/d4cc947f-b8db-493f-bf03-f18af9c821b3) Issue URL : https://github.com/alsa-project/alsa-utils/issues/229 Repository URL: https://github.com/alsa-project/alsa-utils

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"alsactl restore" does not load the driver state
by GitHub issues - opened 13 Aug '23

13 Aug '23

alsa-project/alsa-utils issue #229 was opened from YuGiOhJCJ: Hello, When I type "alsactl restore", it does not load the driver state. Steps to reproduce: 1. Open a first terminal dedicated to alsa-mixer 2. Open a second terminal where you will type the following commands 3. In the first terminal, press F4, set the audio volume for capture to 40 4. In the second terminal, type "sudo alsactl store" in order to save the driver state to /var/lib/alsa/asound.state 5. In the first terminal, set the audio volume for capture to 35 6. In the second terminal, type "sudo alsactl restore" in order to load the driver state to /var/lib/alsa/asound.state Expected results: I should see in the first terminal that the audio volume for capture is restored to 40. Current results: I see in the first terminal that the audio volume for capture is not restored to 40. It is still set to 35! How to fix that please? Thank you. Best regards. About my system: Packages: - alsa-lib-1.2.9 - alsa-plugins-1.2.7.1 - alsa-utils-1.2.9 Operating system: Slackware 14.2 (64-bit) Architecture: x86_64 Kernel: Linux 5.18.10 ![image](https://github.com/alsa-project/alsa-utils/assets/2247896/d4cc947f-b8db-493f-bf03-f18af9c821b3) Issue URL : https://github.com/alsa-project/alsa-utils/issues/229 Repository URL: https://github.com/alsa-project/alsa-utils

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[PATCH v2 0/4] Device tree and config updates for the Librem 5 devkit
by Guido Günther 12 Aug '23

12 Aug '23

The device tree updates ensure the A53 cores don't get powered off and fix a DT warning. This isn't testable with the arm64 default config unless we enable the rsi wifi modules too so do this as well. While at that include a binding file update. This is against next-20230809 that include David's option,gtm601 conversion. v2: - drop "dt-bindings: mmc: Fix reference to pwr-seq-simple" already picked up by Ulf Hansson, thanks! - add acked-by from Conor Dooley, thanks! - rework sdio changes as proposed by Sherry Sun Guido Günther (4): dt-bindings: sound: gtm601: Add description arm64: dts: imx8mq-librem5-devkit: Mark buck2 as always on arm64: dts: imx8mq-librem5-devkit: Drop power-supply arm64: defconfig: Enable Redpine 91X wlan driver .../bindings/sound/option,gtm601.yaml | 6 +++-- .../dts/freescale/imx8mq-librem5-devkit.dts | 26 ++++++++----------- arch/arm64/configs/defconfig | 1 + 3 files changed, 16 insertions(+), 17 deletions(-) -- 2.40.1

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[tiwai-sound:test/pcm-iov_iter 3/25] sound/core/memory.c:108:53: warning: passing argument 2 of 'copy_from_iter' makes integer from pointer without a cast
by kernel test robot 12 Aug '23

12 Aug '23

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git test/pcm-iov_iter head: da1be13e050f24d088d24fd65517d56f27a2eeb1 commit: 516e284c8b17a5129fcfbfb1f2f4e11093e13d3f [3/25] ALSA: core: Add memory copy helpers between iov_iter and iomem config: i386-randconfig-r033-20230812 (https://download.01.org/0day-ci/archive/20230812/202308120931.Aj1gaY7K-lkp@…) compiler: gcc-12 (Debian 12.2.0-14) 12.2.0 reproduce: (https://download.01.org/0day-ci/archive/20230812/202308120931.Aj1gaY7K-lkp@…) If you fix the issue in a separate patch/commit (i.e. not just a new version of the same patch/commit), kindly add following tags | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202308120931.Aj1gaY7K-lkp@intel.com/ All warnings (new ones prefixed by >>): sound/core/memory.c: In function 'copy_from_iter_toio': >> sound/core/memory.c:108:53: warning: passing argument 2 of 'copy_from_iter' makes integer from pointer without a cast [-Wint-conversion] 108 | return !copy_from_iter((void __force *)dst, src, count) ? -EFAULT : 0; | ^~~ | | | struct iov_iter * In file included from include/sound/pcm.h:19, from sound/core/memory.c:12: include/linux/uio.h:199:42: note: expected 'size_t' {aka 'unsigned int'} but argument is of type 'struct iov_iter *' 199 | size_t copy_from_iter(void *addr, size_t bytes, struct iov_iter *i) | ~~~~~~~^~~~~ >> sound/core/memory.c:108:58: warning: passing argument 3 of 'copy_from_iter' makes pointer from integer without a cast [-Wint-conversion] 108 | return !copy_from_iter((void __force *)dst, src, count) ? -EFAULT : 0; | ^~~~~ | | | size_t {aka unsigned int} include/linux/uio.h:199:66: note: expected 'struct iov_iter *' but argument is of type 'size_t' {aka 'unsigned int'} 199 | size_t copy_from_iter(void *addr, size_t bytes, struct iov_iter *i) | ~~~~~~~~~~~~~~~~~^ vim +/copy_from_iter +108 sound/core/memory.c 93 94 /** 95 * copy_from_iter_toio - copy data from iov_iter to mmio-space 96 * @dst: the destination pointer on mmio-space 97 * @src: the source iov_iter 98 * @count: the data size to copy in bytes 99 * 100 * Copies the data from iov_iter to mmio-space. 101 * Only the single segment iov_iter is assumed for now. 102 * 103 * Return: Zero if successful, or non-zero on failure. 104 */ 105 int copy_from_iter_toio(void __iomem *dst, struct iov_iter *src, size_t count) 106 { 107 #if defined(__i386__) || defined(CONFIG_SPARC32) > 108 return !copy_from_iter((void __force *)dst, src, count) ? -EFAULT : 0; -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

1 0

MSI x570s Carbon Max WiFi ALC 4080
by GitHub issues - edited 11 Aug '23

11 Aug '23

alsa-project/alsa-ucm-conf issue #338 was edited from zpackrat: Fedora 38 patched to current, issue has existed for months. Rear output jacks for Center/Sub and Rear Speakers are reversed. I use a 5.1 speaker setup. When I run sound tests in the gnome audio settings the issue is verifiable. Also audio that requires center channel for dialogue has issues. This is specifically the rear audio I/O ports. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/338 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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MSI x570s Carbon Max WiFi ALC 4080
by GitHub issues - opened 11 Aug '23

11 Aug '23

alsa-project/alsa-ucm-conf issue #338 was opened from zpackrat: Fedora 38 patched to current, issue has existed for months. Rear output jacks for Center/Sub and Rear Speakers are reversed. I use a 5.1 speaker setup. When I run sound tests in the gnome audio settings the issue is verifiable. Also audio that requires center channel for dialogue has issues. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/338 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[RFC] pcm: Add MSBITS subformat options
by GitHub pull_request - edited 11 Aug '23

11 Aug '23

alsa-project/alsa-lib pull request #342 was edited from crojewsk-intel: Improve granularity of format selection for S32/U32 formats by adding masks representing 20, 24 and 32 most significant bits. As a preparation step introduce `snd_pcm_subformat_value()` for easy name-to-subformat conversions. While at it, fix few warnings that pop up in subformat-related code during compilation. Request URL : https://github.com/alsa-project/alsa-lib/pull/342 Patch URL : https://github.com/alsa-project/alsa-lib/pull/342.patch Repository URL: https://github.com/alsa-project/alsa-lib

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[RFC] pcm: Add MSBITS subformat options
by GitHub pull_request - edited 11 Aug '23

11 Aug '23

alsa-project/alsa-lib pull request #342 was edited from crojewsk-intel: Improve granularity of format selection for S32/U32 formats by adding masks representing 20 and 24 most significant bits. As a preparation step introduce `snd_pcm_subformat_value()` for easy name-to-subformat conversions. While at it, fix few warnings that pop up in subformat-related code during compilation. Request URL : https://github.com/alsa-project/alsa-lib/pull/342 Patch URL : https://github.com/alsa-project/alsa-lib/pull/342.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH v1] ALSA: hda/cs8409: Support new Dell Dolphin Variants
by Stefan Binding 11 Aug '23

11 Aug '23

Add 4 new Dell Dolphin Systems, same configuration as older systems. Signed-off-by: Stefan Binding <sbinding(a)opensource.cirrus.com> --- sound/pci/hda/patch_cs8409-tables.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/pci/hda/patch_cs8409-tables.c b/sound/pci/hda/patch_cs8409-tables.c index b288874e401e..36b411d1a960 100644 --- a/sound/pci/hda/patch_cs8409-tables.c +++ b/sound/pci/hda/patch_cs8409-tables.c @@ -550,6 +550,10 @@ const struct snd_pci_quirk cs8409_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0C50, "Dolphin", CS8409_DOLPHIN), SND_PCI_QUIRK(0x1028, 0x0C51, "Dolphin", CS8409_DOLPHIN), SND_PCI_QUIRK(0x1028, 0x0C52, "Dolphin", CS8409_DOLPHIN), + SND_PCI_QUIRK(0x1028, 0x0C73, "Dolphin", CS8409_DOLPHIN), + SND_PCI_QUIRK(0x1028, 0x0C75, "Dolphin", CS8409_DOLPHIN), + SND_PCI_QUIRK(0x1028, 0x0C7D, "Dolphin", CS8409_DOLPHIN), + SND_PCI_QUIRK(0x1028, 0x0C7F, "Dolphin", CS8409_DOLPHIN), {} /* terminator */ }; -- 2.34.1

2 1

[PATCH v2 0/3] soundwire: allow for more than 8 devices, keep IDA for wake-capable devices
by Bard Liao 11 Aug '23

11 Aug '23

This series suggests a hybrid strategy for device number allocation, where only wake-capable devices use a system-unique Device Number which will be used on LunarLake to handle wake-ups using the HDaudio WAKEEN and WAKESTS. Pierre-Louis Bossart (3): soundwire: extend parameters of new_peripheral_assigned() callback soundwire: bus: add callbacks for device_number allocation soundwire: intel_auxdevice: add hybrid IDA-based device_number allocation v2: - Move the device numver allocation from bus.c to intel_auxdevice.c - List the codecs with wake capability in a quirk table, and assign device number according to that quirk table. drivers/soundwire/bus.c | 18 +++--- drivers/soundwire/intel_auxdevice.c | 85 ++++++++++++++++++++++++++--- include/linux/soundwire/sdw.h | 12 ++-- include/linux/soundwire/sdw_intel.h | 7 +++ 4 files changed, 99 insertions(+), 23 deletions(-) -- 2.25.1

2 4

[PATCH 0/2] soundwire: improve pm_runtime handling
by Bard Liao 11 Aug '23

11 Aug '23

This patchset improves the pm_runtime behavior in rare corner cases identified by the Intel CI in the last 6 months. a) in stress-tests, it's not uncommon to see the following type of warnings when the codec reports as ATTACHED "rt711 sdw:0:025d:0711:00: runtime PM trying to activate child device sdw:0:025d:0711:00 but parent (sdw-master-0) is not active" This warning was not correlated with any functional issue, but it exposed a design issue on when to enable pm_runtime. The recommended practice in the pm_runtime documentation is to keep the devices in 'suspended' mode and mark them as 'active' when they are really functional. Pierre-Louis Bossart (2): soundwire: intel_auxdevice: enable pm_runtime earlier on startup soundWire: intel_auxdevice: resume 'sdw-master' on startup and system resume drivers/soundwire/intel_auxdevice.c | 27 +++++++++++++++++++-------- 1 file changed, 19 insertions(+), 8 deletions(-) -- 2.25.1

3 5

[PATCH -next] ASoC: SOF: Intel: Remove duplicated include in lnl.c
by Yang Li 10 Aug '23

10 Aug '23

./sound/soc/sof/intel/lnl.c: hda.h is included more than once. Signed-off-by: Yang Li <yang.lee(a)linux.alibaba.com> --- sound/soc/sof/intel/lnl.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/sof/intel/lnl.c b/sound/soc/sof/intel/lnl.c index 3d919b0b6891..db94b45e53af 100644 --- a/sound/soc/sof/intel/lnl.c +++ b/sound/soc/sof/intel/lnl.c @@ -16,7 +16,6 @@ #include "hda-ipc.h" #include "../sof-audio.h" #include "mtl.h" -#include "hda.h" #include <sound/hda-mlink.h> /* LunarLake ops */ -- 2.20.1.7.g153144c

3 2

[PATCH v1 0/5] Device tree and config updates for the Librem 5 devkit
by Guido Günther 10 Aug '23

10 Aug '23

The device tree updates ensure the A53 don't get powered off and fix a DT warning. This isn't testable with the arm64 default config unless we enable the rsi wifi modules too so do this as well. While at that include two binding file updates. This is against next-20230809 that include David's option,gtm601 conversion. Guido Günther (5): dt-bindings: sound: gtm601: Add description dt-bindings: mmc: Fix reference to pwr-seq-simple arm64: dts: imx8mq-librem5-devkit: Mark buck2 as always on arm64: dts: imx8mq-librem5-devkit: Drop power-supply arm64: defconfig: Enable Redpine 91X wlan driver .../devicetree/bindings/mmc/mmc-controller.yaml | 2 +- .../devicetree/bindings/sound/option,gtm601.yaml | 6 ++++-- .../boot/dts/freescale/imx8mq-librem5-devkit.dts | 14 +++++--------- arch/arm64/configs/defconfig | 1 + 4 files changed, 11 insertions(+), 12 deletions(-) -- 2.40.1

6 10

[PATCH] ASoC: rt715: Add software reset in io init
by Jack Yu 10 Aug '23

10 Aug '23

Add software reset before setting preset registers to make sure all the registers are the default value before preset. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- sound/soc/codecs/rt715.c | 56 ++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt715.h | 7 +++++ 2 files changed, 63 insertions(+) diff --git a/sound/soc/codecs/rt715.c b/sound/soc/codecs/rt715.c index 79416bb48814..c95f481cd235 100644 --- a/sound/soc/codecs/rt715.c +++ b/sound/soc/codecs/rt715.c @@ -52,6 +52,60 @@ static int rt715_index_write(struct regmap *regmap, unsigned int reg, return ret; } +static int rt715_index_write_nid(struct regmap *regmap, + unsigned int nid, unsigned int reg, unsigned int value) +{ + int ret; + unsigned int addr = ((RT715_PRIV_INDEX_W_H_2 | nid) << 8) | reg; + + ret = regmap_write(regmap, addr, value); + if (ret < 0) + pr_err("Failed to set private value: %06x <= %04x ret=%d\n", + addr, value, ret); + + return ret; +} + +static int rt715_index_read_nid(struct regmap *regmap, + unsigned int nid, unsigned int reg, unsigned int *value) +{ + int ret; + unsigned int addr = ((RT715_PRIV_INDEX_W_H_2 | nid) << 8) | reg; + + *value = 0; + ret = regmap_read(regmap, addr, value); + if (ret < 0) + pr_err("Failed to get private value: %06x => %04x ret=%d\n", + addr, *value, ret); + + return ret; +} + +static int rt715_index_update_bits(struct regmap *regmap, unsigned int nid, + unsigned int reg, unsigned int mask, unsigned int val) +{ + unsigned int tmp, orig; + int ret; + + ret = rt715_index_read_nid(regmap, nid, reg, &orig); + if (ret < 0) + return ret; + + tmp = orig & ~mask; + tmp |= val & mask; + + return rt715_index_write_nid(regmap, nid, reg, tmp); +} + +static void rt715_reset(struct regmap *regmap) +{ + regmap_write(regmap, RT715_FUNC_RESET, 0); + rt715_index_update_bits(regmap, RT715_VENDOR_REGISTERS, + RT715_VD_CLEAR_CTRL, RT715_CLEAR_HIDDEN_REG, + RT715_CLEAR_HIDDEN_REG); +} + + static void rt715_get_gain(struct rt715_priv *rt715, unsigned int addr_h, unsigned int addr_l, unsigned int val_h, unsigned int *r_val, unsigned int *l_val) @@ -1040,6 +1094,8 @@ int rt715_io_init(struct device *dev, struct sdw_slave *slave) pm_runtime_get_noresume(&slave->dev); + rt715_reset(rt715->regmap); + /* Mute nid=08h/09h */ regmap_write(rt715->regmap, RT715_SET_GAIN_LINE_ADC_H, 0xb080); regmap_write(rt715->regmap, RT715_SET_GAIN_MIX_ADC_H, 0xb080); diff --git a/sound/soc/codecs/rt715.h b/sound/soc/codecs/rt715.h index 12a0ae656d09..6e37bf64e12f 100644 --- a/sound/soc/codecs/rt715.h +++ b/sound/soc/codecs/rt715.h @@ -48,6 +48,7 @@ struct rt715_priv { #define RT715_INLINE_CMD 0x55 /* Index (NID:20h) */ +#define RT715_VD_CLEAR_CTRL 0x01 #define RT715_SDW_INPUT_SEL 0x39 #define RT715_EXT_DMIC_CLK_CTRL2 0x54 @@ -71,6 +72,8 @@ struct rt715_priv { #define RT715_READ_HDA_0 0x2015 #define RT715_PRIV_INDEX_W_H 0x7520 #define RT715_PRIV_INDEX_W_L 0x85a0 +#define RT715_PRIV_INDEX_W_H_2 0x7500 +#define RT715_PRIV_INDEX_W_L_2 0x8580 #define RT715_PRIV_DATA_W_H 0x7420 #define RT715_PRIV_DATA_W_L 0x84a0 #define RT715_PRIV_INDEX_R_H 0x9d20 @@ -198,6 +201,10 @@ struct rt715_priv { #define RT715_SET_DMIC4_CONFIG_DEFAULT4\ (RT715_VERB_SET_CONFIG_DEFAULT4 | RT715_DMIC4) +/* vendor register clear ctrl-1 (0x01)(NID:20h) */ +#define RT715_CLEAR_HIDDEN_REG (0x1 << 15) + + #define RT715_MUTE_SFT 7 #define RT715_DIR_IN_SFT 6 #define RT715_DIR_OUT_SFT 7 -- 2.34.1

2 1

[PATCH] ASoC: meson: axg-tdm-formatter: fix channel slot allocation
by Jerome Brunet 10 Aug '23

10 Aug '23

When the tdm lane mask is computed, the driver currently fills the 1st lane before moving on to the next. If the stream has less channels than the lanes can accommodate, slots will be disabled on the last lanes. Unfortunately, the HW distribute channels in a different way. It distribute channels in pair on each lanes before moving on the next slots. This difference leads to problems if a device has an interface with more than 1 lane and with more than 2 slots per lane. For example: a playback interface with 2 lanes and 4 slots each (total 8 slots - zero based numbering) - Playing a 8ch stream: - All slots activated by the driver - channel #2 will be played on lane #1 - slot #0 following HW placement - Playing a 4ch stream: - Lane #1 disabled by the driver - channel #2 will be played on lane #0 - slot #2 This behaviour is obviously not desirable. Change the way slots are activated on the TDM lanes to follow what the HW does and make sure each channel always get mapped to the same slot/lane. Fixes: 1a11d88f499c ("ASoC: meson: add tdm formatter base driver") Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> --- sound/soc/meson/axg-tdm-formatter.c | 42 ++++++++++++++++++----------- 1 file changed, 26 insertions(+), 16 deletions(-) diff --git a/sound/soc/meson/axg-tdm-formatter.c b/sound/soc/meson/axg-tdm-formatter.c index 9883dc777f63..63333a2b0a9c 100644 --- a/sound/soc/meson/axg-tdm-formatter.c +++ b/sound/soc/meson/axg-tdm-formatter.c @@ -30,27 +30,32 @@ int axg_tdm_formatter_set_channel_masks(struct regmap *map, struct axg_tdm_stream *ts, unsigned int offset) { - unsigned int val, ch = ts->channels; - unsigned long mask; - int i, j; + unsigned int ch = ts->channels; + u32 val[AXG_TDM_NUM_LANES]; + int i, j, k; + + /* + * We need to mimick the slot distribution used by the HW to keep the + * channel placement consistent regardless of the number of channel + * in the stream. This is why the odd algorithm below is used. + */ + memset(val, 0, sizeof(*val) * AXG_TDM_NUM_LANES); /* * Distribute the channels of the stream over the available slots - * of each TDM lane + * of each TDM lane. We need to go over the 32 slots ... */ - for (i = 0; i < AXG_TDM_NUM_LANES; i++) { - val = 0; - mask = ts->mask[i]; - - for (j = find_first_bit(&mask, 32); - (j < 32) && ch; - j = find_next_bit(&mask, 32, j + 1)) { - val |= 1 << j; - ch -= 1; + for (i = 0; (i < 32) && ch; i += 2) { + /* ... of all the lanes ... */ + for (j = 0; j < AXG_TDM_NUM_LANES; j++) { + /* ... then distribute the channels in pairs */ + for (k = 0; k < 2; k++) { + if ((BIT(i + k) & ts->mask[j]) && ch) { + val[j] |= BIT(i + k); + ch -= 1; + } + } } - - regmap_write(map, offset, val); - offset += regmap_get_reg_stride(map); } /* @@ -63,6 +68,11 @@ int axg_tdm_formatter_set_channel_masks(struct regmap *map, return -EINVAL; } + for (i = 0; i < AXG_TDM_NUM_LANES; i++) { + regmap_write(map, offset, val[i]); + offset += regmap_get_reg_stride(map); + } + return 0; } EXPORT_SYMBOL_GPL(axg_tdm_formatter_set_channel_masks); -- 2.40.1

2 1

[PATCH] ASoC: SOF: Intel: Initialize chip in hda_sdw_check_wakeen_irq()
by Nathan Chancellor 10 Aug '23

10 Aug '23

Clang warns (or errors with CONFIG_WERROR): sound/soc/sof/intel/hda.c:423:6: error: variable 'chip' is uninitialized when used here [-Werror,-Wuninitialized] 423 | if (chip && chip->check_sdw_wakeen_irq) | ^~~~ sound/soc/sof/intel/hda.c:418:39: note: initialize the variable 'chip' to silence this warning 418 | const struct sof_intel_dsp_desc *chip; | ^ | = NULL 1 error generated. Add the missing initialization, following the pattern of the other irq functions. Fixes: 9362ab78f175 ("ASoC: SOF: Intel: add abstraction for SoundWire wake-ups") Signed-off-by: Nathan Chancellor <nathan(a)kernel.org> --- sound/soc/sof/intel/hda.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index 04c748a72b13..15e6779efaa3 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -420,6 +420,7 @@ static bool hda_sdw_check_wakeen_irq(struct snd_sof_dev *sdev) if (!(interface_mask & BIT(SOF_DAI_INTEL_ALH))) return false; + chip = get_chip_info(sdev->pdata); if (chip && chip->check_sdw_wakeen_irq) return chip->check_sdw_wakeen_irq(sdev); --- base-commit: 59146c3cd326a622e9041614842346aada11ca99 change-id: 20230809-intel-hda-missing-chip-init-dcccbe6365a4 Best regards, -- Nathan Chancellor <nathan(a)kernel.org>

3 7

[PATCH -next] ASoC: SOF: amd: Remove duplicated include
by GUO Zihua 10 Aug '23

10 Aug '23

Remove duplicated include of sof-audio.h. Resolves checkincludes message. Signed-off-by: GUO Zihua <guozihua(a)huawei.com> --- sound/soc/sof/amd/acp-common.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/sof/amd/acp-common.c b/sound/soc/sof/amd/acp-common.c index 3a0c7688dcfe..2d72c6d55dc8 100644 --- a/sound/soc/sof/amd/acp-common.c +++ b/sound/soc/sof/amd/acp-common.c @@ -13,7 +13,6 @@ #include "../sof-priv.h" #include "../sof-audio.h" #include "../ops.h" -#include "../sof-audio.h" #include "acp.h" #include "acp-dsp-offset.h" #include <sound/sof/xtensa.h> -- 2.17.1

1 0

[PATCH] ALSA: dice: add stream format parameters for Weiss devices
by Takashi Sakamoto 10 Aug '23

10 Aug '23

This patch is for kernel prepatch v6.5. Hard-coded stream format parameters are added for Weiss Engineering FireWire devices. When the device vendor and model match, the parameters are copied into the stream format cache. This allows for setting all supported sampling rates up to 192kHz, and consequently adjusting the number of available I/O channels. Signed-off-by: Rolf Anderegg <rolf.anderegg(a)weiss.ch> Signed-off-by: Michele Perrone <michele.perrone(a)weiss.ch> Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/dice/Makefile | 2 +- sound/firewire/dice/dice-weiss.c | 104 +++++++++++++++++++++++++++++++ sound/firewire/dice/dice.c | 63 +++++++++++++++++++ sound/firewire/dice/dice.h | 1 + 4 files changed, 169 insertions(+), 1 deletion(-) create mode 100644 sound/firewire/dice/dice-weiss.c diff --git a/sound/firewire/dice/Makefile b/sound/firewire/dice/Makefile index a5f3fbf28b8c..bac8712f9014 100644 --- a/sound/firewire/dice/Makefile +++ b/sound/firewire/dice/Makefile @@ -2,5 +2,5 @@ snd-dice-objs := dice-transaction.o dice-stream.o dice-proc.o dice-midi.o \ dice-pcm.o dice-hwdep.o dice.o dice-tcelectronic.o \ dice-alesis.o dice-extension.o dice-mytek.o dice-presonus.o \ - dice-harman.o dice-focusrite.o + dice-harman.o dice-focusrite.o dice-weiss.o obj-$(CONFIG_SND_DICE) += snd-dice.o diff --git a/sound/firewire/dice/dice-weiss.c b/sound/firewire/dice/dice-weiss.c new file mode 100644 index 000000000000..129d43408956 --- /dev/null +++ b/sound/firewire/dice/dice-weiss.c @@ -0,0 +1,104 @@ +// SPDX-License-Identifier: GPL-2.0 +// dice-weiss.c - a part of driver for DICE based devices +// +// Copyright (c) 2023 Rolf Anderegg and Michele Perrone + +#include "dice.h" + +struct dice_weiss_spec { + unsigned int tx_pcm_chs[MAX_STREAMS][SND_DICE_RATE_MODE_COUNT]; + unsigned int rx_pcm_chs[MAX_STREAMS][SND_DICE_RATE_MODE_COUNT]; +}; + +// Weiss DAC202: 192kHz 2-channel DAC +static const struct dice_weiss_spec dac202 = { + .tx_pcm_chs = {{2, 2, 2}, {0, 0, 0} }, + .rx_pcm_chs = {{2, 2, 2}, {0, 0, 0} }, +}; + +// Weiss MAN301: 192kHz 2-channel music archive network player +static const struct dice_weiss_spec man301 = { + .tx_pcm_chs = {{2, 2, 2}, {0, 0, 0} }, + .rx_pcm_chs = {{2, 2, 2}, {0, 0, 0} }, +}; + +// Weiss INT202: 192kHz unidirectional 2-channel digital Firewire nterface +static const struct dice_weiss_spec int202 = { + .tx_pcm_chs = {{2, 2, 2}, {0, 0, 0} }, + .rx_pcm_chs = {{2, 2, 2}, {0, 0, 0} }, +}; + +// Weiss INT203: 192kHz bidirectional 2-channel digital Firewire nterface +static const struct dice_weiss_spec int203 = { + .tx_pcm_chs = {{2, 2, 2}, {0, 0, 0} }, + .rx_pcm_chs = {{2, 2, 2}, {0, 0, 0} }, +}; + +// Weiss ADC2: 192kHz A/D converter with microphone preamps and line nputs +static const struct dice_weiss_spec adc2 = { + .tx_pcm_chs = {{2, 2, 2}, {0, 0, 0} }, + .rx_pcm_chs = {{2, 2, 2}, {0, 0, 0} }, +}; + +// Weiss DAC2/Minerva: 192kHz 2-channel DAC +static const struct dice_weiss_spec dac2_minerva = { + .tx_pcm_chs = {{2, 2, 2}, {0, 0, 0} }, + .rx_pcm_chs = {{2, 2, 2}, {0, 0, 0} }, +}; + +// Weiss Vesta: 192kHz 2-channel Firewire to AES/EBU interface +static const struct dice_weiss_spec vesta = { + .tx_pcm_chs = {{2, 2, 2}, {0, 0, 0} }, + .rx_pcm_chs = {{2, 2, 2}, {0, 0, 0} }, +}; + +// Weiss AFI1: 192kHz 24-channel Firewire to ADAT or AES/EBU interface +static const struct dice_weiss_spec afi1 = { + .tx_pcm_chs = {{24, 16, 8}, {0, 0, 0} }, + .rx_pcm_chs = {{24, 16, 8}, {0, 0, 0} }, +}; + +int snd_dice_detect_weiss_formats(struct snd_dice *dice) +{ + static const struct { + u32 model_id; + const struct dice_weiss_spec *spec; + } *entry, entries[] = { + {0x000007, &dac202}, + {0x000008, &dac202}, // Maya edition: same audio I/O as DAC202. + {0x000006, &int202}, + {0x00000a, &int203}, + {0x00000b, &man301}, + {0x000001, &adc2}, + {0x000003, &dac2_minerva}, + {0x000002, &vesta}, + {0x000004, &afi1}, + }; + struct fw_csr_iterator it; + int key, val, model_id; + int i; + + model_id = 0; + fw_csr_iterator_init(&it, dice->unit->directory); + while (fw_csr_iterator_next(&it, &key, &val)) { + if (key == CSR_MODEL) { + model_id = val; + break; + } + } + + for (i = 0; i < ARRAY_SIZE(entries); ++i) { + entry = entries + i; + if (entry->model_id == model_id) + break; + } + if (i == ARRAY_SIZE(entries)) + return -ENODEV; + + memcpy(dice->tx_pcm_chs, entry->spec->tx_pcm_chs, + MAX_STREAMS * SND_DICE_RATE_MODE_COUNT * sizeof(unsigned int)); + memcpy(dice->rx_pcm_chs, entry->spec->rx_pcm_chs, + MAX_STREAMS * SND_DICE_RATE_MODE_COUNT * sizeof(unsigned int)); + + return 0; +} diff --git a/sound/firewire/dice/dice.c b/sound/firewire/dice/dice.c index 6c93e6e4982c..d362e4251c68 100644 --- a/sound/firewire/dice/dice.c +++ b/sound/firewire/dice/dice.c @@ -392,6 +392,69 @@ static const struct ieee1394_device_id dice_id_table[] = { .model_id = 0x0000de, .driver_data = (kernel_ulong_t)snd_dice_detect_focusrite_pro40_tcd3070_formats, }, + // Weiss DAC202: 192kHz 2-channel DAC + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | IEEE1394_MATCH_MODEL_ID, + .vendor_id = OUI_WEISS, + .model_id = 0x000007, + .driver_data = (kernel_ulong_t)snd_dice_detect_weiss_formats, + }, + // Weiss DAC202: 192kHz 2-channel DAC (Maya edition) + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | IEEE1394_MATCH_MODEL_ID, + .vendor_id = OUI_WEISS, + .model_id = 0x000008, + .driver_data = (kernel_ulong_t)snd_dice_detect_weiss_formats, + }, + // Weiss MAN301: 192kHz 2-channel music archive network player + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | IEEE1394_MATCH_MODEL_ID, + .vendor_id = OUI_WEISS, + .model_id = 0x00000b, + .driver_data = (kernel_ulong_t)snd_dice_detect_weiss_formats, + }, + // Weiss INT202: 192kHz unidirectional 2-channel digital Firewire face + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | IEEE1394_MATCH_MODEL_ID, + .vendor_id = OUI_WEISS, + .model_id = 0x000006, + .driver_data = (kernel_ulong_t)snd_dice_detect_weiss_formats, + }, + // Weiss INT203: 192kHz bidirectional 2-channel digital Firewire face + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | IEEE1394_MATCH_MODEL_ID, + .vendor_id = OUI_WEISS, + .model_id = 0x00000a, + .driver_data = (kernel_ulong_t)snd_dice_detect_weiss_formats, + }, + // Weiss ADC2: 192kHz A/D converter with microphone preamps and inputs + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | IEEE1394_MATCH_MODEL_ID, + .vendor_id = OUI_WEISS, + .model_id = 0x000001, + .driver_data = (kernel_ulong_t)snd_dice_detect_weiss_formats, + }, + // Weiss DAC2/Minerva: 192kHz 2-channel DAC + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | IEEE1394_MATCH_MODEL_ID, + .vendor_id = OUI_WEISS, + .model_id = 0x000003, + .driver_data = (kernel_ulong_t)snd_dice_detect_weiss_formats, + }, + // Weiss Vesta: 192kHz 2-channel Firewire to AES/EBU interface + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | IEEE1394_MATCH_MODEL_ID, + .vendor_id = OUI_WEISS, + .model_id = 0x000002, + .driver_data = (kernel_ulong_t)snd_dice_detect_weiss_formats, + }, + // Weiss AFI1: 192kHz 24-channel Firewire to ADAT or AES/EBU face + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | IEEE1394_MATCH_MODEL_ID, + .vendor_id = OUI_WEISS, + .model_id = 0x000004, + .driver_data = (kernel_ulong_t)snd_dice_detect_weiss_formats, + }, { .match_flags = IEEE1394_MATCH_VERSION, .version = DICE_INTERFACE, diff --git a/sound/firewire/dice/dice.h b/sound/firewire/dice/dice.h index 674f7d552c2e..4c0ad7335998 100644 --- a/sound/firewire/dice/dice.h +++ b/sound/firewire/dice/dice.h @@ -232,5 +232,6 @@ int snd_dice_detect_mytek_formats(struct snd_dice *dice); int snd_dice_detect_presonus_formats(struct snd_dice *dice); int snd_dice_detect_harman_formats(struct snd_dice *dice); int snd_dice_detect_focusrite_pro40_tcd3070_formats(struct snd_dice *dice); +int snd_dice_detect_weiss_formats(struct snd_dice *dice); #endif -- 2.39.2

2 6

[PATCH v2] ASoC: SOF: ipc4-topology: Update the basecfg for copier earlier
by Peter Ujfalusi 10 Aug '23

10 Aug '23

The sof_ipc4_update_resource_usage() call updates the CPC value in basecfg and it must be done prior to making a copy of the copier configuration for the init message. Other module types do the resource update as last step or in case of a process module at the correct time, before the memcpy. Fixes: d8a2c9879349 ("ASoC: SOF: ipc4-loader/topology: Query the CPC value from manifest") Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Adrian Bonislawski <adrian.bonislawski(a)intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Chao Song <chao.song(a)linux.intel.com> --- Hi, changes since v1: - created on top of for-6.5 branch The conflict caused by this for 6.6 is trivial as it is just a move two lines patch. Regards, Peter sound/soc/sof/ipc4-topology.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index a4e1a70b607d..11361e1cd688 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -1731,6 +1731,9 @@ sof_ipc4_prepare_copier_module(struct snd_sof_widget *swidget, *ipc_config_size = ipc_size; + /* update pipeline memory usage */ + sof_ipc4_update_resource_usage(sdev, swidget, &copier_data->base_config); + /* copy IPC data */ memcpy(*ipc_config_data, (void *)copier_data, sizeof(*copier_data)); if (gtw_cfg_config_length) @@ -1743,9 +1746,6 @@ sof_ipc4_prepare_copier_module(struct snd_sof_widget *swidget, gtw_cfg_config_length, &ipc4_copier->dma_config_tlv, dma_config_tlv_size); - /* update pipeline memory usage */ - sof_ipc4_update_resource_usage(sdev, swidget, &copier_data->base_config); - return 0; } -- 2.41.0

2 1

[PATCH] ASoC: SOF: ipc4-topology: Update the basecfg for copier earlier
by Peter Ujfalusi 10 Aug '23

10 Aug '23

The sof_ipc4_update_resource_usage() call updates the CPC value in basecfg and it must be done prior to making a copy of the copier configuration for the init message. Other module types do the resource update as last step or in case of a process module at the correct time, before the memcpy. Fixes: d8a2c9879349 ("ASoC: SOF: ipc4-loader/topology: Query the CPC value from manifest") Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Adrian Bonislawski <adrian.bonislawski(a)intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Chao Song <chao.song(a)linux.intel.com> --- sound/soc/sof/ipc4-topology.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index 0196cbfc0998..633f6040d712 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -1731,6 +1731,9 @@ sof_ipc4_prepare_copier_module(struct snd_sof_widget *swidget, *ipc_config_size = ipc_size; + /* update pipeline memory usage */ + sof_ipc4_update_resource_usage(sdev, swidget, &copier_data->base_config); + /* copy IPC data */ memcpy(*ipc_config_data, (void *)copier_data, sizeof(*copier_data)); if (gtw_cfg_config_length) @@ -1743,9 +1746,6 @@ sof_ipc4_prepare_copier_module(struct snd_sof_widget *swidget, gtw_cfg_config_length, &ipc4_copier->dma_config_tlv, dma_config_tlv_size); - /* update pipeline memory usage */ - sof_ipc4_update_resource_usage(sdev, swidget, &copier_data->base_config); - /* * Restore gateway config length now that IPC payload is prepared. This avoids * counting the DMA CONFIG TLV multiple times -- 2.41.0

2 4

[PATCH] MAINTAINERS: update Claudiu Beznea's email address
by Claudiu Beznea 10 Aug '23

10 Aug '23

Update MAINTAINERS entries with a valid email address as the Microchip one is no longer valid. Acked-by: Conor Dooley <conor.dooley(a)microchip.com> Acked-by: Nicolas Ferre <nicolas.ferre(a)microchip.com> Signed-off-by: Claudiu Beznea <claudiu.beznea(a)tuxon.dev> --- Changes in v2: - collected tags - extended the recipients list to include individual subsystem maintainers and lists instead using only linux-kernel(a)vger.kernel.org as suggested initially by get_maintainers.pl MAINTAINERS | 22 +++++++++++----------- 1 file changed, 11 insertions(+), 11 deletions(-) diff --git a/MAINTAINERS b/MAINTAINERS index 128fd295f86f..d48d8e857f57 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -2351,7 +2351,7 @@ F: drivers/phy/mediatek/ ARM/MICROCHIP (ARM64) SoC support M: Conor Dooley <conor(a)kernel.org> M: Nicolas Ferre <nicolas.ferre(a)microchip.com> -M: Claudiu Beznea <claudiu.beznea(a)microchip.com> +M: Claudiu Beznea <claudiu.beznea(a)tuxon.dev> L: linux-arm-kernel(a)lists.infradead.org (moderated for non-subscribers) S: Supported T: git https://git.kernel.org/pub/scm/linux/kernel/git/at91/linux.git @@ -2360,7 +2360,7 @@ F: arch/arm64/boot/dts/microchip/ ARM/Microchip (AT91) SoC support M: Nicolas Ferre <nicolas.ferre(a)microchip.com> M: Alexandre Belloni <alexandre.belloni(a)bootlin.com> -M: Claudiu Beznea <claudiu.beznea(a)microchip.com> +M: Claudiu Beznea <claudiu.beznea(a)tuxon.dev> L: linux-arm-kernel(a)lists.infradead.org (moderated for non-subscribers) S: Supported W: http://www.linux4sam.org @@ -3265,7 +3265,7 @@ F: include/uapi/linux/atm* ATMEL MACB ETHERNET DRIVER M: Nicolas Ferre <nicolas.ferre(a)microchip.com> -M: Claudiu Beznea <claudiu.beznea(a)microchip.com> +M: Claudiu Beznea <claudiu.beznea(a)tuxon.dev> S: Supported F: drivers/net/ethernet/cadence/ @@ -13880,7 +13880,7 @@ F: Documentation/devicetree/bindings/serial/atmel,at91-usart.yaml F: drivers/spi/spi-at91-usart.c MICROCHIP AUDIO ASOC DRIVERS -M: Claudiu Beznea <claudiu.beznea(a)microchip.com> +M: Claudiu Beznea <claudiu.beznea(a)tuxon.dev> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) S: Supported F: Documentation/devicetree/bindings/sound/atmel* @@ -13903,7 +13903,7 @@ S: Maintained F: drivers/crypto/atmel-ecc.* MICROCHIP EIC DRIVER -M: Claudiu Beznea <claudiu.beznea(a)microchip.com> +M: Claudiu Beznea <claudiu.beznea(a)tuxon.dev> L: linux-arm-kernel(a)lists.infradead.org (moderated for non-subscribers) S: Supported F: Documentation/devicetree/bindings/interrupt-controller/microchip,sama7g5-eic.yaml @@ -13976,7 +13976,7 @@ F: drivers/video/fbdev/atmel_lcdfb.c F: include/video/atmel_lcdc.h MICROCHIP MCP16502 PMIC DRIVER -M: Claudiu Beznea <claudiu.beznea(a)microchip.com> +M: Claudiu Beznea <claudiu.beznea(a)tuxon.dev> L: linux-arm-kernel(a)lists.infradead.org (moderated for non-subscribers) S: Supported F: Documentation/devicetree/bindings/regulator/mcp16502-regulator.txt @@ -14003,7 +14003,7 @@ F: Documentation/devicetree/bindings/mtd/atmel-nand.txt F: drivers/mtd/nand/raw/atmel/* MICROCHIP OTPC DRIVER -M: Claudiu Beznea <claudiu.beznea(a)microchip.com> +M: Claudiu Beznea <claudiu.beznea(a)tuxon.dev> L: linux-arm-kernel(a)lists.infradead.org (moderated for non-subscribers) S: Supported F: Documentation/devicetree/bindings/nvmem/microchip,sama7g5-otpc.yaml @@ -14042,7 +14042,7 @@ F: Documentation/devicetree/bindings/fpga/microchip,mpf-spi-fpga-mgr.yaml F: drivers/fpga/microchip-spi.c MICROCHIP PWM DRIVER -M: Claudiu Beznea <claudiu.beznea(a)microchip.com> +M: Claudiu Beznea <claudiu.beznea(a)tuxon.dev> L: linux-arm-kernel(a)lists.infradead.org (moderated for non-subscribers) L: linux-pwm(a)vger.kernel.org S: Supported @@ -14058,7 +14058,7 @@ F: drivers/iio/adc/at91-sama5d2_adc.c F: include/dt-bindings/iio/adc/at91-sama5d2_adc.h MICROCHIP SAMA5D2-COMPATIBLE SHUTDOWN CONTROLLER -M: Claudiu Beznea <claudiu.beznea(a)microchip.com> +M: Claudiu Beznea <claudiu.beznea(a)tuxon.dev> S: Supported F: Documentation/devicetree/bindings/power/reset/atmel,sama5d2-shdwc.yaml F: drivers/power/reset/at91-sama5d2_shdwc.c @@ -14075,7 +14075,7 @@ S: Supported F: drivers/spi/spi-atmel.* MICROCHIP SSC DRIVER -M: Claudiu Beznea <claudiu.beznea(a)microchip.com> +M: Claudiu Beznea <claudiu.beznea(a)tuxon.dev> L: linux-arm-kernel(a)lists.infradead.org (moderated for non-subscribers) S: Supported F: Documentation/devicetree/bindings/misc/atmel-ssc.txt @@ -14104,7 +14104,7 @@ F: drivers/usb/gadget/udc/atmel_usba_udc.* MICROCHIP WILC1000 WIFI DRIVER M: Ajay Singh <ajay.kathat(a)microchip.com> -M: Claudiu Beznea <claudiu.beznea(a)microchip.com> +M: Claudiu Beznea <claudiu.beznea(a)tuxon.dev> L: linux-wireless(a)vger.kernel.org S: Supported F: drivers/net/wireless/microchip/wilc1000/ -- 2.39.2

6 7

[PATCH v2 0/2] Add HP G11 Quirks and update Dell Oasis Quirks
by Stefan Binding 10 Aug '23

10 Aug '23

Add quirks to support HP G11 using CS35l41 HDA, using both SPI and I2C. One older HP laptop has an SSID removed. In addition, all Dell Oasis laptops using CS35L41 have been switch to use SPI, with one SSID removed. Changes from v1: - Add fixes tags Stefan Binding (2): ALSA: hda/realtek: Add quirks for HP G11 Laptops ALSA: hda/realtek: Switch Dell Oasis models to use SPI sound/pci/hda/patch_realtek.c | 17 +++++++++++------ 1 file changed, 11 insertions(+), 6 deletions(-) -- 2.34.1

2 3

[PATCH v3 0/3] ASoC: mt8188-mt6359: add SOF support
by Trevor Wu 10 Aug '23

10 Aug '23

This series introduces dynamic pinctrl and adds support for the SOF on the mt8188-mt6359 machine driver. Changes since v2: - revise mtk_sof_check_tplg_be_dai_link_fixup(). Changes since v1: - minor changes to incorporate the suggestions from the reviewer. Trevor Wu (3): ASoC: mediatek: mt8188-mt6359: support dynamic pinctrl ASoC: mediatek: common: revise SOF common code ASoC: mediatek: mt8188-mt6359: add SOF support .../soc/mediatek/common/mtk-dsp-sof-common.c | 113 +++++++-- .../soc/mediatek/common/mtk-dsp-sof-common.h | 8 + sound/soc/mediatek/mt8188/mt8188-mt6359.c | 239 +++++++++++++++++- 3 files changed, 332 insertions(+), 28 deletions(-) -- 2.18.0

1 3

Question: license compliance for libatopology 2.0.0
by GitHub issues - opened 09 Aug '23

09 Aug '23

alsa-project/alsa-lib issue #343 was opened from lucasgonze: Given a binary linked with libatopology 2.0.0, and given that the LGPL requires source code to be bundled, it is unclear how to find the corresponding repo and tag (or release). Thanks in advance for tips. Issue URL : https://github.com/alsa-project/alsa-lib/issues/343 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH] ASoC: samsung: midas_wm1811: Fix 'Headphone Switch' control creation
by Marek Szyprowski 09 Aug '23

09 Aug '23

'Headphone Switch' control is already registered from sound/soc/codecs/wm_hubs.c:479, so duplicating it in midas_wm1811 causes following probe failure: midas-audio sound: control 2:0:0:Headphone Switch:0 is already present midas-audio sound: ASoC: Failed to add Headphone Switch: -16 midas-audio sound: Failed to register card: -16 midas-audio: probe of sound failed with error -16 Fix this by dropping duplicated control. Fixes: d27224a45e54 ("ASoC: samsung: midas_wm1811: Map missing jack kcontrols") Signed-off-by: Marek Szyprowski <m.szyprowski(a)samsung.com> --- sound/soc/samsung/midas_wm1811.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/samsung/midas_wm1811.c b/sound/soc/samsung/midas_wm1811.c index 44b32f5cddca..126098fdcf1b 100644 --- a/sound/soc/samsung/midas_wm1811.c +++ b/sound/soc/samsung/midas_wm1811.c @@ -257,7 +257,6 @@ static const struct snd_kcontrol_new midas_controls[] = { SOC_DAPM_PIN_SWITCH("Main Mic"), SOC_DAPM_PIN_SWITCH("Sub Mic"), SOC_DAPM_PIN_SWITCH("Headset Mic"), - SOC_DAPM_PIN_SWITCH("Headphone"), SOC_DAPM_PIN_SWITCH("FM In"), }; -- 2.34.1

2 1

[PATCH] ASoC: rsnd: call of_node_put() when break
by Kuninori Morimoto 09 Aug '23

09 Aug '23

We need to call of_node_put() when break from for_each_child_of_node(). This patch add missing of_node_put(). Reported-by: kernel test robot <lkp(a)intel.com> Reported-by: Julia Lawall <julia.lawall(a)inria.fr> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/core.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 069add7b1b71..e29c2fee9521 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -1282,6 +1282,7 @@ static int rsnd_dai_of_node(struct rsnd_priv *priv, int *is_graph) i++; if (i >= RSND_MAX_COMPONENT) { dev_info(dev, "reach to max component\n"); + of_node_put(node); break; } } @@ -1301,6 +1302,7 @@ static int rsnd_dai_of_node(struct rsnd_priv *priv, int *is_graph) i++; if (i >= RSND_MAX_COMPONENT) { dev_info(dev, "reach to max component\n"); + of_node_put(node); break; } } -- 2.25.1

2 1

[PATCH 0/5] ASoC: cs35l56: Bugfixes
by Richard Fitzgerald 09 Aug '23

09 Aug '23

Miscellaneous bugfixes for the cs35l56 codec driver. Richard Fitzgerald (3): ASoC: cs35l56: Avoid uninitialized variable in cs35l56_set_asp_slot_positions() ASoC: cs35l56: Don't rely on GPIOD_OUT_LOW to set RESET initially low ASoC: cs35l56: Wait for control port ready during system-resume Simon Trimmer (2): ASoC: wm_adsp: Expose the DSP power down actions as wm_adsp_power_down() ASoC: cs35l56: Call wm_adsp_power_down() before reloading firmware sound/soc/codecs/cs35l56.c | 33 +++++++++++++++++---------------- sound/soc/codecs/wm_adsp.c | 8 +++++++- sound/soc/codecs/wm_adsp.h | 1 + 3 files changed, 25 insertions(+), 17 deletions(-) -- 2.30.2

2 6

[PATCH v1 0/2] Add HP G11 Quirks and update Dell Oasis Quirks
by Stefan Binding 09 Aug '23

09 Aug '23

Add quirks to support HP G11 using CS35l41 HDA, using both SPI and I2C. One older HP laptop has an SSID removed. In addition, all Dell Oasis laptops using CS35L41 have been switch to use SPI, with one SSID removed. Stefan Binding (2): ALSA: hda/realtek: Add quirks for HP G11 Laptops ALSA: hda/realtek: Switch Dell Oasis models to use SPI sound/pci/hda/patch_realtek.c | 17 +++++++++++------ 1 file changed, 11 insertions(+), 6 deletions(-) -- 2.34.1

2 3

[PATCH] ASoC: codecs: aw88261: avoid uninitialized variable warning
by Arnd Bergmann 08 Aug '23

08 Aug '23

From: Arnd Bergmann <arnd(a)arndb.de> aw88261_reg_update() returns an unintialized error code in the success path: sound/soc/codecs/aw88261.c:651:7: error: variable 'ret' is used uninitialized whenever 'if' condition is false [-Werror,-Wsometimes-uninitialized] if (aw_dev->prof_cur != aw_dev->prof_index) { ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ sound/soc/codecs/aw88261.c:660:9: note: uninitialized use occurs here return ret; ^~~ sound/soc/codecs/aw88261.c:651:3: note: remove the 'if' if its condition is always true if (aw_dev->prof_cur != aw_dev->prof_index) { ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ Return zero instead here. Fixes: 028a2ae256916 ("ASoC: codecs: Add aw88261 amplifier driver") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/codecs/aw88261.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/aw88261.c b/sound/soc/codecs/aw88261.c index 82923b454dd47..6e2266b713862 100644 --- a/sound/soc/codecs/aw88261.c +++ b/sound/soc/codecs/aw88261.c @@ -652,6 +652,8 @@ static int aw88261_reg_update(struct aw88261 *aw88261, bool force) ret = aw88261_dev_fw_update(aw88261); if (ret) return ret; + } else { + ret = 0; } } -- 2.39.2

3 2

[PATCH 01/12] ASoC: intel: sof_sdw: Printk's should end with a newline
by Charles Keepax 08 Aug '23

08 Aug '23

Add the missing new lines. Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/intel/boards/sof_sdw.c | 22 +++++++++++----------- 1 file changed, 11 insertions(+), 11 deletions(-) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index fd27e211211bd..8f3329dcf7062 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -524,7 +524,7 @@ int sdw_prepare(struct snd_pcm_substream *substream) sdw_stream = snd_soc_dai_get_stream(dai, substream->stream); if (IS_ERR(sdw_stream)) { - dev_err(rtd->dev, "no stream found for DAI %s", dai->name); + dev_err(rtd->dev, "no stream found for DAI %s\n", dai->name); return PTR_ERR(sdw_stream); } @@ -543,7 +543,7 @@ int sdw_trigger(struct snd_pcm_substream *substream, int cmd) sdw_stream = snd_soc_dai_get_stream(dai, substream->stream); if (IS_ERR(sdw_stream)) { - dev_err(rtd->dev, "no stream found for DAI %s", dai->name); + dev_err(rtd->dev, "no stream found for DAI %s\n", dai->name); return PTR_ERR(sdw_stream); } @@ -565,7 +565,7 @@ int sdw_trigger(struct snd_pcm_substream *substream, int cmd) } if (ret) - dev_err(rtd->dev, "%s trigger %d failed: %d", __func__, cmd, ret); + dev_err(rtd->dev, "%s trigger %d failed: %d\n", __func__, cmd, ret); return ret; } @@ -630,7 +630,7 @@ int sdw_hw_free(struct snd_pcm_substream *substream) sdw_stream = snd_soc_dai_get_stream(dai, substream->stream); if (IS_ERR(sdw_stream)) { - dev_err(rtd->dev, "no stream found for DAI %s", dai->name); + dev_err(rtd->dev, "no stream found for DAI %s\n", dai->name); return PTR_ERR(sdw_stream); } @@ -1339,7 +1339,7 @@ static int get_slave_info(const struct snd_soc_acpi_link_adr *adr_link, return -EINVAL; if (index >= SDW_MAX_CPU_DAIS) { - dev_err(dev, " cpu_dai_id array overflows"); + dev_err(dev, "cpu_dai_id array overflows\n"); return -EINVAL; } @@ -1490,7 +1490,7 @@ static int create_sdw_dailink(struct snd_soc_card *card, int *link_index, return -ENOMEM; if (cpu_dai_index >= sdw_cpu_dai_num) { - dev_err(dev, "invalid cpu dai index %d", + dev_err(dev, "invalid cpu dai index %d\n", cpu_dai_index); return -EINVAL; } @@ -1503,12 +1503,12 @@ static int create_sdw_dailink(struct snd_soc_card *card, int *link_index, * not be larger than sdw_be_num */ if (*link_index >= sdw_be_num) { - dev_err(dev, "invalid dai link index %d", *link_index); + dev_err(dev, "invalid dai link index %d\n", *link_index); return -EINVAL; } if (*cpu_id >= sdw_cpu_dai_num) { - dev_err(dev, " invalid cpu dai index %d", *cpu_id); + dev_err(dev, "invalid cpu dai index %d\n", *cpu_id); return -EINVAL; } @@ -1531,7 +1531,7 @@ static int create_sdw_dailink(struct snd_soc_card *card, int *link_index, ret = set_codec_init_func(card, adr_link, dai_links + (*link_index)++, playback, group_id, adr_index, dai_index); if (ret < 0) { - dev_err(dev, "failed to init codec %d", codec_index); + dev_err(dev, "failed to init codec %d\n", codec_index); return ret; } @@ -1675,7 +1675,7 @@ static int sof_card_dai_links_create(struct snd_soc_card *card) endpoint = adr_link->adr_d[i].endpoints; if (endpoint->aggregated && !endpoint->group_id) { - dev_err(dev, "invalid group id on link %x", + dev_err(dev, "invalid group id on link %x\n", adr_link->mask); continue; } @@ -1698,7 +1698,7 @@ static int sof_card_dai_links_create(struct snd_soc_card *card) &be_id, &codec_conf_index, &ignore_pch_dmic, append_dai_type, i, j); if (ret < 0) { - dev_err(dev, "failed to create dai link %d", link_index); + dev_err(dev, "failed to create dai link %d\n", link_index); return ret; } } -- 2.30.2

2 12

[PATCH -next] ASoC: imx-audio-rpmsg: Remove redundant initialization owner in imx_audio_rpmsg_driver
by Li Zetao 08 Aug '23

08 Aug '23

The module_rpmsg_driver() will set "THIS_MODULE" to driver.owner when register a rpmsg_driver driver, so it is redundant initialization to set driver.owner in the statement. Remove it for clean code. Signed-off-by: Li Zetao <lizetao1(a)huawei.com> --- sound/soc/fsl/imx-audio-rpmsg.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/fsl/imx-audio-rpmsg.c b/sound/soc/fsl/imx-audio-rpmsg.c index d5234ac4b09b..289e47c03d40 100644 --- a/sound/soc/fsl/imx-audio-rpmsg.c +++ b/sound/soc/fsl/imx-audio-rpmsg.c @@ -116,7 +116,6 @@ static struct rpmsg_device_id imx_audio_rpmsg_id_table[] = { static struct rpmsg_driver imx_audio_rpmsg_driver = { .drv.name = "imx_audio_rpmsg", - .drv.owner = THIS_MODULE, .id_table = imx_audio_rpmsg_id_table, .probe = imx_audio_rpmsg_probe, .callback = imx_audio_rpmsg_cb, -- 2.34.1

2 1

[linux-next:master] BUILD REGRESSION 71cd4fc492ec41e4acd85e98bbf7a13753fc1e03
by kernel test robot 08 Aug '23

08 Aug '23

tree/branch: https://git.kernel.org/pub/scm/linux/kernel/git/next/linux-next.git master branch HEAD: 71cd4fc492ec41e4acd85e98bbf7a13753fc1e03 Add linux-next specific files for 20230808 Error/Warning reports: https://lore.kernel.org/oe-kbuild-all/202307251531.p8ZLFTMZ-lkp@intel.com https://lore.kernel.org/oe-kbuild-all/202308081459.US5rLYAY-lkp@intel.com Error/Warning: (recently discovered and may have been fixed) ../lib/gcc/loongarch64-linux/12.3.0/plugin/include/config/loongarch/loongarch-opts.h:31:10: fatal error: loongarch-def.h: No such file or directory drivers/gpu/drm/amd/amdgpu/../display/dc/dce/dmub_replay.c:37: warning: This comment starts with '/**', but isn't a kernel-doc comment. Refer Documentation/doc-guide/kernel-doc.rst sound/soc/codecs/aw88261.c:651:7: warning: variable 'ret' is used uninitialized whenever 'if' condition is false [-Wsometimes-uninitialized] Unverified Error/Warning (likely false positive, please contact us if interested): drivers/gpu/drm/tests/drm_exec_test.c:166 test_prepare_array() error: uninitialized symbol 'ret'. drivers/mtd/nand/raw/qcom_nandc.c:2590 qcom_op_cmd_mapping() error: uninitialized symbol 'ret'. drivers/mtd/nand/raw/qcom_nandc.c:3017 qcom_check_op() warn: was && intended here instead of ||? kernel/futex/waitwake.c:422 futex_wait_multiple_setup() warn: bitwise AND condition is false here sh4-linux-gcc: internal compiler error: Segmentation fault signal terminated program cc1 {standard input}: Warning: end of file not at end of a line; newline inserted {standard input}:927: Error: pcrel too far Error/Warning ids grouped by kconfigs: gcc_recent_errors |-- alpha-allyesconfig | `-- drivers-gpu-drm-amd-amdgpu-..-display-dc-dce-dmub_replay.c:warning:This-comment-starts-with-but-isn-t-a-kernel-doc-comment.-Refer-Documentation-doc-guide-kernel-doc.rst |-- arc-allyesconfig | `-- drivers-gpu-drm-amd-amdgpu-..-display-dc-dce-dmub_replay.c:warning:This-comment-starts-with-but-isn-t-a-kernel-doc-comment.-Refer-Documentation-doc-guide-kernel-doc.rst |-- arm-allmodconfig | `-- drivers-gpu-drm-amd-amdgpu-..-display-dc-dce-dmub_replay.c:warning:This-comment-starts-with-but-isn-t-a-kernel-doc-comment.-Refer-Documentation-doc-guide-kernel-doc.rst |-- arm-allyesconfig | `-- drivers-gpu-drm-amd-amdgpu-..-display-dc-dce-dmub_replay.c:warning:This-comment-starts-with-but-isn-t-a-kernel-doc-comment.-Refer-Documentation-doc-guide-kernel-doc.rst |-- arm64-allyesconfig | `-- drivers-gpu-drm-amd-amdgpu-..-display-dc-dce-dmub_replay.c:warning:This-comment-starts-with-but-isn-t-a-kernel-doc-comment.-Refer-Documentation-doc-guide-kernel-doc.rst |-- csky-randconfig-m041-20230808 | `-- drivers-gpu-drm-tests-drm_exec_test.c-test_prepare_array()-error:uninitialized-symbol-ret-. |-- i386-allyesconfig | `-- drivers-gpu-drm-amd-amdgpu-..-display-dc-dce-dmub_replay.c:warning:This-comment-starts-with-but-isn-t-a-kernel-doc-comment.-Refer-Documentation-doc-guide-kernel-doc.rst |-- loongarch-allmodconfig | `-- lib-gcc-loongarch64-linux-..-plugin-include-config-loongarch-loongarch-opts.h:fatal-error:loongarch-def.h:No-such-file-or-directory |-- mips-allmodconfig | `-- drivers-gpu-drm-amd-amdgpu-..-display-dc-dce-dmub_replay.c:warning:This-comment-starts-with-but-isn-t-a-kernel-doc-comment.-Refer-Documentation-doc-guide-kernel-doc.rst |-- mips-allyesconfig | `-- drivers-gpu-drm-amd-amdgpu-..-display-dc-dce-dmub_replay.c:warning:This-comment-starts-with-but-isn-t-a-kernel-doc-comment.-Refer-Documentation-doc-guide-kernel-doc.rst |-- parisc-allyesconfig | `-- drivers-gpu-drm-amd-amdgpu-..-display-dc-dce-dmub_replay.c:warning:This-comment-starts-with-but-isn-t-a-kernel-doc-comment.-Refer-Documentation-doc-guide-kernel-doc.rst |-- powerpc-allmodconfig | `-- drivers-gpu-drm-amd-amdgpu-..-display-dc-dce-dmub_replay.c:warning:This-comment-starts-with-but-isn-t-a-kernel-doc-comment.-Refer-Documentation-doc-guide-kernel-doc.rst |-- riscv-allmodconfig | `-- drivers-gpu-drm-amd-amdgpu-..-display-dc-dce-dmub_replay.c:warning:This-comment-starts-with-but-isn-t-a-kernel-doc-comment.-Refer-Documentation-doc-guide-kernel-doc.rst |-- riscv-allyesconfig | `-- drivers-gpu-drm-amd-amdgpu-..-display-dc-dce-dmub_replay.c:warning:This-comment-starts-with-but-isn-t-a-kernel-doc-comment.-Refer-Documentation-doc-guide-kernel-doc.rst |-- riscv-randconfig-m031-20230808 | |-- drivers-gpu-drm-tests-drm_exec_test.c-test_prepare_array()-error:uninitialized-symbol-ret-. | |-- drivers-mtd-nand-raw-qcom_nandc.c-qcom_check_op()-warn:was-intended-here-instead-of | |-- drivers-mtd-nand-raw-qcom_nandc.c-qcom_op_cmd_mapping()-error:uninitialized-symbol-ret-. | `-- kernel-futex-waitwake.c-futex_wait_multiple_setup()-warn:bitwise-AND-condition-is-false-here |-- s390-allyesconfig | `-- drivers-gpu-drm-amd-amdgpu-..-display-dc-dce-dmub_replay.c:warning:This-comment-starts-with-but-isn-t-a-kernel-doc-comment.-Refer-Documentation-doc-guide-kernel-doc.rst |-- s390-randconfig-r044-20230808 | `-- drivers-gpu-drm-amd-amdgpu-..-display-dc-dce-dmub_replay.c:warning:This-comment-starts-with-but-isn-t-a-kernel-doc-comment.-Refer-Documentation-doc-guide-kernel-doc.rst |-- sh-allmodconfig | |-- sh4-linux-gcc:internal-compiler-error:Segmentation-fault-signal-terminated-program-cc1 | |-- standard-input:Error:pcrel-too-far | `-- standard-input:Warning:end-of-file-not-at-end-of-a-line-newline-inserted |-- sparc-allyesconfig | `-- drivers-gpu-drm-amd-amdgpu-..-display-dc-dce-dmub_replay.c:warning:This-comment-starts-with-but-isn-t-a-kernel-doc-comment.-Refer-Documentation-doc-guide-kernel-doc.rst |-- x86_64-allyesconfig | `-- drivers-gpu-drm-amd-amdgpu-..-display-dc-dce-dmub_replay.c:warning:This-comment-starts-with-but-isn-t-a-kernel-doc-comment.-Refer-Documentation-doc-guide-kernel-doc.rst `-- xtensa-randconfig-r022-20230808 `-- drivers-gpu-drm-amd-amdgpu-..-display-dc-dce-dmub_replay.c:warning:This-comment-starts-with-but-isn-t-a-kernel-doc-comment.-Refer-Documentation-doc-guide-kernel-doc.rst clang_recent_errors `-- arm64-randconfig-r006-20230808 `-- sound-soc-codecs-aw88261.c:warning:variable-ret-is-used-uninitialized-whenever-if-condition-is-false elapsed time: 721m configs tested: 106 configs skipped: 5 tested configs: alpha allyesconfig gcc alpha defconfig gcc arc allyesconfig gcc arc defconfig gcc arc randconfig-r034-20230808 gcc arc randconfig-r043-20230808 gcc arm allmodconfig gcc arm allyesconfig gcc arm defconfig gcc arm randconfig-r012-20230808 clang arm randconfig-r046-20230808 clang arm64 allyesconfig gcc arm64 defconfig gcc arm64 randconfig-r001-20230808 clang csky defconfig gcc csky randconfig-r024-20230808 gcc hexagon randconfig-r015-20230808 clang hexagon randconfig-r031-20230808 clang hexagon randconfig-r041-20230808 clang hexagon randconfig-r045-20230808 clang i386 allyesconfig gcc i386 buildonly-randconfig-r004-20230808 clang i386 buildonly-randconfig-r005-20230808 clang i386 buildonly-randconfig-r006-20230808 clang i386 debian-10.3 gcc i386 defconfig gcc i386 randconfig-i001-20230808 clang i386 randconfig-i002-20230808 clang i386 randconfig-i003-20230808 clang i386 randconfig-i004-20230808 clang i386 randconfig-i005-20230808 clang i386 randconfig-i006-20230808 clang i386 randconfig-i011-20230808 gcc i386 randconfig-i012-20230808 gcc i386 randconfig-i013-20230808 gcc i386 randconfig-i014-20230808 gcc i386 randconfig-i015-20230808 gcc i386 randconfig-i016-20230808 gcc i386 randconfig-r014-20230808 gcc loongarch allmodconfig gcc loongarch allnoconfig gcc loongarch defconfig gcc m68k allmodconfig gcc m68k allyesconfig gcc m68k defconfig gcc m68k randconfig-r004-20230808 gcc mips allmodconfig gcc mips allyesconfig gcc nios2 defconfig gcc parisc allyesconfig gcc parisc defconfig gcc parisc randconfig-r002-20230808 gcc parisc randconfig-r016-20230808 gcc parisc64 defconfig gcc powerpc allmodconfig gcc powerpc allnoconfig gcc powerpc randconfig-r005-20230808 clang riscv allmodconfig gcc riscv allnoconfig gcc riscv allyesconfig gcc riscv defconfig gcc riscv randconfig-r042-20230808 gcc riscv rv32_defconfig gcc s390 allmodconfig gcc s390 allyesconfig gcc s390 defconfig gcc s390 randconfig-r011-20230808 gcc s390 randconfig-r023-20230808 gcc s390 randconfig-r044-20230808 gcc sh allmodconfig gcc sh randconfig-r006-20230808 gcc sh randconfig-r025-20230808 gcc sparc allyesconfig gcc sparc defconfig gcc um allmodconfig clang um allnoconfig clang um allyesconfig clang um defconfig gcc um i386_defconfig gcc um randconfig-r033-20230808 gcc um x86_64_defconfig gcc x86_64 allyesconfig gcc x86_64 buildonly-randconfig-r001-20230808 clang x86_64 buildonly-randconfig-r002-20230808 clang x86_64 buildonly-randconfig-r003-20230808 clang x86_64 defconfig gcc x86_64 kexec gcc x86_64 randconfig-r003-20230808 clang x86_64 randconfig-r021-20230808 gcc x86_64 randconfig-x001-20230808 gcc x86_64 randconfig-x002-20230808 gcc x86_64 randconfig-x003-20230808 gcc x86_64 randconfig-x004-20230808 gcc x86_64 randconfig-x005-20230808 gcc x86_64 randconfig-x006-20230808 gcc x86_64 randconfig-x011-20230808 clang x86_64 randconfig-x012-20230808 clang x86_64 randconfig-x013-20230808 clang x86_64 randconfig-x014-20230808 clang x86_64 randconfig-x015-20230808 clang x86_64 randconfig-x016-20230808 clang x86_64 rhel-8.3-rust clang x86_64 rhel-8.3 gcc xtensa randconfig-r013-20230808 gcc xtensa randconfig-r022-20230808 gcc xtensa randconfig-r035-20230808 gcc -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

1 0

[PATCH] ASoC: amd: acp3x-rt5682-max9836: Configure jack as not detecting Line Out
by Alper Nebi Yasak 08 Aug '23

08 Aug '23

The RT5682, RT1015 and RT1015p codecs used in this driver do not seem capable of distinguishing Line Out connections from Headphone, but the driver configures its jack object as if it can. Remove the wrong value from the jack creation call to avoid any confusion. Signed-off-by: Alper Nebi Yasak <alpernebiyasak(a)gmail.com> --- Depends on [1] which applies onto of linux-next. Untested, see [2]: [1] https://lore.kernel.org/alsa-devel/20230802175737.263412-5-alpernebiyasak@g… [2] https://lore.kernel.org/alsa-devel/3d6794c9-60d3-b8f3-99d2-afdc69593d9f@gma… sound/soc/amd/acp3x-rt5682-max9836.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/amd/acp3x-rt5682-max9836.c b/sound/soc/amd/acp3x-rt5682-max9836.c index 3bf7e1e83c36..28ad5f5b9a76 100644 --- a/sound/soc/amd/acp3x-rt5682-max9836.c +++ b/sound/soc/amd/acp3x-rt5682-max9836.c @@ -98,7 +98,7 @@ static int acp3x_5682_init(struct snd_soc_pcm_runtime *rtd) rt5682_dai_bclk = clk_get(component->dev, "rt5682-dai-bclk"); ret = snd_soc_card_jack_new_pins(card, "Headset Jack", - SND_JACK_HEADSET | SND_JACK_LINEOUT | + SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2 | SND_JACK_BTN_3, &pco_jack, base-commit: 626c67169f9972fffcdf3bc3864de421f162ebf5 prerequisite-patch-id: 7158530856f8f97324906b342528a5ffe7b4032b -- 2.40.1

4 3

[PATCH 00/20] ASoC: SOF: Intel: add LunarLake support
by Pierre-Louis Bossart 08 Aug '23

08 Aug '23

IMPORTANT NOTE: this patchset is dependent on Takashi's hda-intel-6.6 tag. The PCI parts will not compile without merging this tag into the ASoC tree. This patchset first fixes a number of errors made in the hda-mlink support, then adds Lunar Lake definitions. The main contribution is the hda-dai changes where the HDaudio DMA is now used for SSP, DMIC and SoundWire. In previous hardware the GPDMA (aka DesignWare) was used and controlled by the audio firmware. The volume of code is minimized with the abstraction added in previous kernel cycles. Due to cross-dependencies between ASoC and SoundWire trees, the full support for jack detection will be deferred to the next kernel cycle. There's not much point to ask for a sync of the two trees to support one patch for each tree - we are at -rc5 already. Pierre-Louis Bossart (18): ASoC: SOF: Intel: hda-mlink: fix off-by-one error ASoC: SOF: Intel: fix u16/32 confusion in LSDIID ASoC: SOF: ipc4-topology: restore gateway config length ASoC: SOF: Intel: hda: add interface definitions for ACE2.x ASoC: SOF: Intel: split MTL and LNL operations ASoC: SOF: Intel: LNL: enable DMIC/SSP offload in probe/resume ASoC: SOF: Intel: hda-dai-ops: add/select DMA ops for SSP ASoC: SOF: Intel: hda-dai: add ops for SSP ASoC: SOF: Intel: hda-dai: add DMIC support ASoC: SOF: Intel: hda-dai-ops: only allocate/release streams for first CPU DAI ASoC: SOF: Intel: hda-dai-ops: add ops for SoundWire ASoC: SOF: Intel: hda-dai: add helpers for SoundWire callbacks ASoC: SOF: Intel: hda: add hw_params/free/trigger callbacks ASoC: SOF: Intel: add abstraction for SoundWire wake-ups ASoC: SOF: Intel: hda-mlink: add helper to get sublink LSDIID register ASoC: SOF: Intel: hda-dai-ops: reset device count for SoundWire DAIs ASoC: SOF: IPC4: clarify 'pipeline_ids' usage and logs ASoC: SOF: Intel: hda-mlink: add sublink to dev_dbg() log Ranjani Sridharan (2): ASoC: SOF: ipc4: avoid uninitialized default instance 0 ASoC: SOF: Intel: LNL: Add support for Lunarlake platform include/sound/hda-mlink.h | 4 + sound/soc/sof/intel/Kconfig | 16 +++ sound/soc/sof/intel/Makefile | 4 +- sound/soc/sof/intel/cnl.c | 2 + sound/soc/sof/intel/hda-dai-ops.c | 179 ++++++++++++++++++++++++- sound/soc/sof/intel/hda-dai.c | 211 +++++++++++++++++++++++++++++- sound/soc/sof/intel/hda-mlink.c | 45 ++++++- sound/soc/sof/intel/hda.c | 58 +++++++- sound/soc/sof/intel/hda.h | 21 +++ sound/soc/sof/intel/icl.c | 1 + sound/soc/sof/intel/lnl.c | 189 ++++++++++++++++++++++++++ sound/soc/sof/intel/mtl.c | 23 ++-- sound/soc/sof/intel/mtl.h | 22 +++- sound/soc/sof/intel/pci-lnl.c | 71 ++++++++++ sound/soc/sof/intel/shim.h | 1 + sound/soc/sof/intel/tgl.c | 4 + sound/soc/sof/ipc4-pcm.c | 21 +-- sound/soc/sof/ipc4-topology.c | 7 + sound/soc/sof/ipc4-topology.h | 4 +- sound/soc/sof/topology.c | 2 + 20 files changed, 842 insertions(+), 43 deletions(-) create mode 100644 sound/soc/sof/intel/lnl.c create mode 100644 sound/soc/sof/intel/pci-lnl.c -- 2.39.2

2 23

[PATCH 0/2] ASoC: Intel: sof_sdw: fix two settings
by Pierre-Louis Bossart 08 Aug '23

08 Aug '23

Fix button mapping for CS42L42 and jack detection for LunarLake RVP. Pierre-Louis Bossart (1): ASoC: Intel: sof-sdw: update jack detection quirk for LunarLake RVP jairaj-arava (1): ASoC: Intel: sof-sdw-cs42142: fix for codec button mapping sound/soc/intel/boards/sof_sdw.c | 2 +- sound/soc/intel/boards/sof_sdw_cs42l42.c | 6 +++--- 2 files changed, 4 insertions(+), 4 deletions(-) -- 2.39.2

2 5

[PATCH -next] ASoC: SOF: ipc3-dtrace: Switch to memdup_user_nul() helper
by Yang Yingliang 08 Aug '23

08 Aug '23

Use memdup_user_nul() helper instead of open-coding to simplify the code. Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/soc/sof/ipc3-dtrace.c | 12 +++--------- 1 file changed, 3 insertions(+), 9 deletions(-) diff --git a/sound/soc/sof/ipc3-dtrace.c b/sound/soc/sof/ipc3-dtrace.c index 35da85a45a9a..bd07f0472efd 100644 --- a/sound/soc/sof/ipc3-dtrace.c +++ b/sound/soc/sof/ipc3-dtrace.c @@ -196,15 +196,9 @@ static ssize_t dfsentry_trace_filter_write(struct file *file, const char __user return -EINVAL; } - string = kmalloc(count + 1, GFP_KERNEL); - if (!string) - return -ENOMEM; - - if (copy_from_user(string, from, count)) { - ret = -EFAULT; - goto error; - } - string[count] = '\0'; + string = memdup_user_nul(from, count); + if (IS_ERR(string)) + return PTR_ERR(string); ret = trace_filter_parse(sdev, string, &num_elems, &elems); if (ret < 0) -- 2.25.1

2 1

Regression fix not progressing towards mainline/stable
by Thorsten Leemhuis 08 Aug '23

08 Aug '23

Hi everyone, I noticed a regression fix that afaics lingers in the thesofproject's git repo instead of progressing towards the stable trees and wondered what's up here. I'm talking about about the fix "ASoC: SOF: intel: hda: Clean up link DMA for IPC3 during stop" for this ticket: https://github.com/thesofproject/linux/issues/4455 Two more tickets about it: https://github.com/thesofproject/linux/issues/4482 https://bugzilla.kernel.org/show_bug.cgi?id=217673 It seems the fix is ready and reviewed for two weeks now, but not even in -next by now. That's not how it should be for regression fixes, as per https://docs.kernel.org/process/handling-regressions.html it ideally should be in -stable soon or already. Is this a mistake, am I missing something, or is there a good reason for this? Ciao, Thorsten

2 2

[PATCH] ALSA: hda/tegra: refactor deprecated strncpy
by Justin Stitt 08 Aug '23

08 Aug '23

`strncpy` is deprecated for use on NUL-terminated destination strings [1]. A suitable replacement is `strscpy` [2] due to the fact that it guarantees NUL-termination on its destination buffer argument which is _not_ the case for `strncpy`! It should be noted that the current implementation is unlikely to have a bug because `drv_name` is a string literal with a size of 9 while `card->driver` has a size of 16. However, it is probably worthwhile to switch to a more robust and less ambiguous interface. [1]: www.kernel.org/doc/html/latest/process/deprecated.html#strncpy-on-nul-termi… [2]: https://manpages.debian.org/testing/linux-manual-4.8/strscpy.9.en.html Link: https://github.com/KSPP/linux/issues/90 Cc: linux-hardening(a)vger.kernel.org Signed-off-by: Justin Stitt <justinstitt(a)google.com> --- sound/pci/hda/hda_tegra.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c index 9d0ab043880b..f16740500d46 100644 --- a/sound/pci/hda/hda_tegra.c +++ b/sound/pci/hda/hda_tegra.c @@ -378,14 +378,14 @@ static int hda_tegra_first_init(struct azx *chip, struct platform_device *pdev) } /* driver name */ - strncpy(card->driver, drv_name, sizeof(card->driver)); + strscpy(card->driver, drv_name, sizeof(card->driver)); /* shortname for card */ sname = of_get_property(np, "nvidia,model", NULL); if (!sname) sname = drv_name; if (strlen(sname) > sizeof(card->shortname)) dev_info(card->dev, "truncating shortname for card\n"); - strncpy(card->shortname, sname, sizeof(card->shortname)); + strscpy(card->shortname, sname, sizeof(card->shortname)); /* longname for card */ snprintf(card->longname, sizeof(card->longname), --- base-commit: c1a515d3c0270628df8ae5f5118ba859b85464a2 change-id: 20230803-sound-pci-hda-c3a712b72a35 Best regards, -- Justin Stitt <justinstitt(a)google.com>

2 1

[PATCH -next] ALSA: info: Remove unused function declarations
by Yue Haibing 08 Aug '23

08 Aug '23

These declarations is never used since beginning of git history. Signed-off-by: Yue Haibing <yuehaibing(a)huawei.com> --- include/sound/info.h | 2 -- 1 file changed, 2 deletions(-) diff --git a/include/sound/info.h b/include/sound/info.h index 7c13bf52cc81..adbc506860d6 100644 --- a/include/sound/info.h +++ b/include/sound/info.h @@ -118,8 +118,6 @@ struct snd_info_entry *snd_info_create_card_entry(struct snd_card *card, const char *name, struct snd_info_entry *parent); void snd_info_free_entry(struct snd_info_entry *entry); -int snd_info_store_text(struct snd_info_entry *entry); -int snd_info_restore_text(struct snd_info_entry *entry); int snd_info_card_create(struct snd_card *card); int snd_info_card_register(struct snd_card *card); -- 2.34.1

2 1

[PATCH v2 1/2] ALSA: pcmtest: Move buffer iterator initialization to prepare callback
by Ivan Orlov 08 Aug '23

08 Aug '23

Trigger callback is not the best place for buffer iterator initialization, so move it out to the prepare callback, where it have to be. Minor enhancement: remove blank line. Signed-off-by: Ivan Orlov <ivan.orlov0322(a)gmail.com> --- V1 -> V2: - Split changes in the patch into two different patches sound/drivers/pcmtest.c | 33 ++++++++++++++++----------------- 1 file changed, 16 insertions(+), 17 deletions(-) diff --git a/sound/drivers/pcmtest.c b/sound/drivers/pcmtest.c index 08e14b5eb772..dc7c9c758537 100644 --- a/sound/drivers/pcmtest.c +++ b/sound/drivers/pcmtest.c @@ -92,7 +92,6 @@ MODULE_PARM_DESC(inject_trigger_err, "Inject EINVAL error in the 'trigger' callb module_param(inject_open_err, bool, 0600); MODULE_PARM_DESC(inject_open_err, "Inject EBUSY error in the 'open' callback"); - struct pcmtst { struct snd_pcm *pcm; struct snd_card *card; @@ -405,25 +404,9 @@ static int snd_pcmtst_pcm_close(struct snd_pcm_substream *substream) static int snd_pcmtst_pcm_trigger(struct snd_pcm_substream *substream, int cmd) { - struct snd_pcm_runtime *runtime = substream->runtime; - struct pcmtst_buf_iter *v_iter = runtime->private_data; - if (inject_trigger_err) return -EINVAL; - v_iter->sample_bytes = runtime->sample_bits / 8; - v_iter->period_bytes = frames_to_bytes(runtime, runtime->period_size); - if (runtime->access == SNDRV_PCM_ACCESS_RW_NONINTERLEAVED || - runtime->access == SNDRV_PCM_ACCESS_MMAP_NONINTERLEAVED) { - v_iter->chan_block = runtime->dma_bytes / runtime->channels; - v_iter->interleaved = false; - } else { - v_iter->interleaved = true; - } - // We want to record RATE * ch_cnt samples per sec, it is rate * sample_bytes * ch_cnt bytes - v_iter->s_rw_ch = runtime->rate / TIMER_PER_SEC; - v_iter->b_rw = v_iter->s_rw_ch * v_iter->sample_bytes * runtime->channels; - return 0; } @@ -454,8 +437,24 @@ static void pcmtst_pdev_release(struct device *dev) static int snd_pcmtst_pcm_prepare(struct snd_pcm_substream *substream) { + struct snd_pcm_runtime *runtime = substream->runtime; + struct pcmtst_buf_iter *v_iter = runtime->private_data; + if (inject_prepare_err) return -EINVAL; + + v_iter->sample_bytes = samples_to_bytes(runtime, 1); + v_iter->period_bytes = snd_pcm_lib_period_bytes(substream); + v_iter->interleaved = true; + if (runtime->access == SNDRV_PCM_ACCESS_RW_NONINTERLEAVED || + runtime->access == SNDRV_PCM_ACCESS_MMAP_NONINTERLEAVED) { + v_iter->chan_block = snd_pcm_lib_buffer_bytes(substream) / runtime->channels; + v_iter->interleaved = false; + } + // We want to record RATE * ch_cnt samples per sec, it is rate * sample_bytes * ch_cnt bytes + v_iter->s_rw_ch = runtime->rate / TIMER_PER_SEC; + v_iter->b_rw = v_iter->s_rw_ch * v_iter->sample_bytes * runtime->channels; + return 0; } -- 2.34.1

2 2

[PATCH v2 00/28] Add support for QMC HDLC, framer infrastruture and PEF2256 framer
by Herve Codina 08 Aug '23

08 Aug '23

Hi, I have a system where I need to handle an HDLC interface and some audio data. The HDLC data are transferred using a TDM bus on which a PEF2256 (E1/T1 framer) is present. The PEF2256 transfers data from/to the TDM bus to/from the E1 line. This PEF2256 is connected to a PowerQUICC SoC for the control path and the TDM is connected to the SoC (QMC component) for the data path. From the QMC HDLC driver, I need to handle HDLC data using the QMC, carrier detection using the PEF2256 (E1 line carrier) and set/get some PEF2256 configuration. The QMC HDLC driver considers the PEF2256 as a generic framer. It performs operations that involve the PEF2256 through the generic framer API. The audio data are exchanged with the PEF2256 using a CPU DAI connected to the TDM bus through the QMC and the PEF2256 needs to be seen as a codec in order to be linked to the CPU DAI. The codec handles the carrier detection using the PEF2256 and reports the carrier state to the ALSA subsystem using the ASoC jack detection. The codec, even if instantiated by the PEF2256 driver, considers the PEF2256 as a generic framer. The generic framer has: - 2 consumers (QMC HDLC drv and codec) - 1 provider (PEF2256) So, the design is the following: +------------------+ +---------+ | QMC | <- TDM -> | PEF2256 | <-> E1 +---------+ | +-------------+ | | | | CPU DAI | <-data--> | QMC channel | | | | +---------+ | +-------------+ | | | +--------------+ | +-------------+ | | | | QMC HDLC drv | <-data--> | QMC channel | | | | +--------------+ | +-------------+ | | | ^ +------------------+ | | | +--------+ +-------------+ | | +-> | framer | <-> | PEF2256 drv | <- local bus ->| | | | | | +---------+ +-> | | | | | +--------+ | +-------+ | +-------------------> | codec | | | +-------+ | +-------------+ Further more, the TDM timeslots used by the QMC HDLC driver need to be configured at runtime (QMC dynamic timeslots). Several weeks ago, I sent two series related to this topic: - Add the Lantiq PEF2256 audio support [1] - RFC Add support for QMC HDLC and PHY [2] This current series is a rework of these two series taking into account feedbacks previously received. In order to implement all of this, I do the following: 1) Perform some fixes (patches 1, 2, 3, 4) 2) Introduce the QMC HDLC driver (patches 5, 6, 7) 3) Add QMC dynamic timeslot support (patches 8 - 18) 4) Add timeslots change support in QMC HDLC (patch 19) 5) Introduce framer infrastructure (patch 20) 6) Add PEF2256 framer provider (patches 11, 22, 23, 24, 25) 7) Add framer codec as a framer consumer (patch 26) 8) Add framer support as a framer consumer in QMC HDLC (patch 27, 28) The series contains the full story and detailed modifications. If needed, the series can be split and/or commmits can be squashed. Let me know. Compare to the previous iteration https://lore.kernel.org/linux-kernel/20230725092417.43706-1-herve.codina@bo… This v2 series mainly: - Adds 2 patches to fix some compilation warning present on existing files (__iomem addresses declaration issues). - Fixes compilation warning on some patches. - Reword a commit log. Best regards, Hervé [1]: https://lore.kernel.org/all/20230417171601.74656-1-herve.codina@bootlin.com/ [2]: https://lore.kernel.org/all/20230323103154.264546-1-herve.codina@bootlin.co… Changes v1 -> v2 - Patches 1, 2 (New in v2) Fix __iomem addresses declaration - Patch 19 (17 in v1) Fix a compilation warning - Patch 26 (24 in v1) Fix a typo in Kconfig file Fix issues raised by sparse (make C=1) Herve Codina (28): soc: fsl: cpm1: tsa: Fix __iomem addresses declaration soc: fsl: cpm1: qmc: Fix __iomem addresses declaration soc: fsl: cpm1: qmc: Fix rx channel reset soc: fsl: cpm1: qmc: Extend the API to provide Rx status dt-bindings: net: Add support for QMC HDLC net: wan: Add support for QMC HDLC MAINTAINERS: Add the Freescale QMC HDLC driver entry soc: fsl: cpm1: qmc: Introduce available timeslots masks soc: fsl: cpm1: qmc: Rename qmc_setup_tsa* to qmc_init_tsa* soc: fsl: cpm1: qmc: Introduce qmc_chan_setup_tsa* soc: fsl: cpm1: qmc: Remove no more needed checks from qmc_check_chans() soc: fsl: cpm1: qmc: Check available timeslots in qmc_check_chans() soc: fsl: cpm1: qmc: Add support for disabling channel TSA entries soc: fsl: cpm1: qmc: Split Tx and Rx TSA entries setup soc: fsl: cpm1: qmc: Introduce is_tsa_64rxtx flag soc: fsl: cpm1: qmc: Handle timeslot entries at channel start() and stop() soc: fsl: cpm1: qmc: Remove timeslots handling from setup_chan() soc: fsl: cpm1: qmc: Introduce functions to change timeslots at runtime wan: qmc_hdlc: Add runtime timeslots changes support net: wan: Add framer framework support dt-bindings: net: Add the Lantiq PEF2256 E1/T1/J1 framer mfd: core: Ensure disabled devices are skiped without aborting net: wan: framer: Add support for the Lantiq PEF2256 framer pinctrl: Add support for the Lantic PEF2256 pinmux MAINTAINERS: Add the Lantiq PEF2256 driver entry ASoC: codecs: Add support for the framer codec dt-bindings: net: fsl,qmc-hdlc: Add framer support net: wan: fsl_qmc_hdlc: Add framer support .../devicetree/bindings/net/fsl,qmc-hdlc.yaml | 46 + .../bindings/net/lantiq,pef2256.yaml | 226 +++++ MAINTAINERS | 17 + drivers/mfd/mfd-core.c | 18 +- drivers/net/wan/Kconfig | 14 + drivers/net/wan/Makefile | 3 + drivers/net/wan/framer/Kconfig | 35 + drivers/net/wan/framer/Makefile | 7 + drivers/net/wan/framer/framer-core.c | 935 ++++++++++++++++++ drivers/net/wan/framer/pef2256/Makefile | 8 + drivers/net/wan/framer/pef2256/pef2256-regs.h | 250 +++++ drivers/net/wan/framer/pef2256/pef2256.c | 880 +++++++++++++++++ drivers/net/wan/fsl_qmc_hdlc.c | 820 +++++++++++++++ drivers/pinctrl/Kconfig | 14 + drivers/pinctrl/Makefile | 1 + drivers/pinctrl/pinctrl-pef2256-regs.h | 65 ++ drivers/pinctrl/pinctrl-pef2256.c | 310 ++++++ drivers/soc/fsl/qe/qmc.c | 501 ++++++++-- drivers/soc/fsl/qe/tsa.c | 22 +- include/linux/framer/framer-provider.h | 194 ++++ include/linux/framer/framer.h | 215 ++++ include/linux/framer/pef2256.h | 31 + include/soc/fsl/qe/qmc.h | 25 +- sound/soc/codecs/Kconfig | 15 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/framer-codec.c | 423 ++++++++ sound/soc/fsl/fsl_qmc_audio.c | 2 +- 27 files changed, 4957 insertions(+), 122 deletions(-) create mode 100644 Documentation/devicetree/bindings/net/fsl,qmc-hdlc.yaml create mode 100644 Documentation/devicetree/bindings/net/lantiq,pef2256.yaml create mode 100644 drivers/net/wan/framer/Kconfig create mode 100644 drivers/net/wan/framer/Makefile create mode 100644 drivers/net/wan/framer/framer-core.c create mode 100644 drivers/net/wan/framer/pef2256/Makefile create mode 100644 drivers/net/wan/framer/pef2256/pef2256-regs.h create mode 100644 drivers/net/wan/framer/pef2256/pef2256.c create mode 100644 drivers/net/wan/fsl_qmc_hdlc.c create mode 100644 drivers/pinctrl/pinctrl-pef2256-regs.h create mode 100644 drivers/pinctrl/pinctrl-pef2256.c create mode 100644 include/linux/framer/framer-provider.h create mode 100644 include/linux/framer/framer.h create mode 100644 include/linux/framer/pef2256.h create mode 100644 sound/soc/codecs/framer-codec.c -- 2.41.0

8 90

[PATCH 00/27] ASoC: Map missing jack kcontrols
by Alper Nebi Yasak 08 Aug '23

08 Aug '23

This patchset adds missing jack kcontrols for each independently detectable audio peripheral and maps jack pins to those kcontrols accordingly, with the primary intent to enable/improve jack detection handling in PulseAudio and PipeWire through JackControl UCM values. Usually it's just splitting a joint anything-is-connected "Headset Jack" kcontrol (from [1]) into those like "Headphone Jack" and "Headset Mic" (similar to a previous series for Intel Chromebooks [2]). This split is important to avoid automatically switching to a nonexistent external microphone when a headphone-only device is connected. When the underlying hardware seems to support it, this also adds a "Line Out" kcontrol. This is important in case the hardware can actually support a line-level connection via a different configuration (bypassing output amplifiers?), or simply for userspace to display "Line Out" instead of "Headphones" to the user for connected line-out devices. Beyond the mappings, I had to add PIN_SWITCH card kcontrols and DAPM widgets to avoid "unknown pin" errors on my devices, so tried to do them for all. For Intel devices I saw a pattern of routing things to "Platform Clock" and added to that as well. Looking at patch 5/7 of a Mediatek-related series [3], I can only guess that routes could be further improved, but don't know exactly how for each device. And one more concern is I don't know if the names conflict with any controls from codecs, although I tried to keep to names of existing widgets. As far as I can tell, the root cause for most of why these are missing originates to things being developed for ChromeOS, whose userspace reads the jack input device and doesn't care for these kcontrols. There's non-ChromeOS cases as well, maybe things got copy-pasted around and people didn't need or couldn't figure out how to get more specific than a single jack kcontrol. The secondary intent in this patchset is to fix this *everywhere*, so future copy-pastes result in the right behaviour. For more context also see: [1] ASoC: soc-card: Create jack kcontrol without pins https://lore.kernel.org/alsa-devel/20220408041114.6024-1-akihiko.odaki@gmai… [2] ASoC: Intel: Chromebooks: remap jack pins https://lore.kernel.org/alsa-devel/20220616214055.134943-1-pierre-louis.bos… [3] ASoC: mediatek: Allow separate handling of headphone and headset mic jack https://lore.kernel.org/alsa-devel/20220922235951.252532-1-nfraprado@collab… [4] ASoC: rk3399_gru_sound: Add DAPM pins, kcontrols for jack detection https://lore.kernel.org/alsa-devel/20200721182709.6895-1-alpernebiyasak@gma… This applies onto next-20230802. Unfortunately most of it is untested except for a few Chromebooks I have (Kevin, Lick, Hana, Cozmo), because I'm intentionally generalizing to everything. Alper Nebi Yasak (27): ASoC: amd: acp-da7219-max98357a: Map missing jack kcontrols ASoC: amd: acp-rt5645: Map missing jack kcontrols ASoC: amd: acp: Map missing jack kcontrols ASoC: amd: acp3x-rt5682-max9836: Map missing jack kcontrols ASoC: Intel: avs: da7219: Map missing jack kcontrols ASoC: Intel: bxt_da7219_max98357a: Map missing Line Out jack kcontrol ASoC: Intel: bytcr_wm5102: Map missing Line Out jack kcontrol ASoC: Intel: kbl_da7219_max98357a: Map missing Line Out jack kcontrol ASoC: Intel: kbl_da7219_max98927: Map missing Line Out jack kcontrol ASoC: Intel: sof_da7219_max98373: Map missing Line Out jack kcontrol ASoC: imx-es8328: Map missing jack kcontrols ASoC: mediatek: mt8173-max98090: Configure jack as a Headset jack ASoC: mediatek: mt8173-rt5650-rt5514: Map missing jack kcontrols ASoC: mediatek: mt8173-rt5650-rt5676: Map missing jack kcontrols ASoC: mediatek: mt8173-rt5650: Map missing jack kcontrols ASoC: mediatek: mt8183-da7219-max98357: Map missing jack kcontrols ASoC: mediatek: mt8183-mt6358-ts3a227-max98357: Map missing jack kcontrols ASoC: mediatek: mt8186-mt6366-da7219-max98357: Map missing jack kcontrols ASoC: qcom: apq8016_sbc: Map missing jack kcontrols ASoC: qcom: sc7180: Map missing jack kcontrols ASoC: qcom: sc7280: Map missing jack kcontrols ASoC: qcom: sdm845: Map missing jack kcontrols ASoC: rk3399-gru-sound: Map missing Line Out jack kcontrol ASoC: rockchip: rockchip_rt5645: Map missing jack kcontrols ASoC: samsung: littlemill: Map missing jack kcontrols ASoC: samsung: lowland: Split Line Out jack kcontrol from Headphone ASoC: samsung: midas_wm1811: Map missing jack kcontrols sound/soc/amd/acp-da7219-max98357a.c | 41 +++- sound/soc/amd/acp-rt5645.c | 22 +- sound/soc/amd/acp/acp-mach-common.c | 226 ++++++++++++++---- sound/soc/amd/acp3x-rt5682-max9836.c | 23 +- sound/soc/fsl/imx-es8328.c | 25 +- sound/soc/intel/avs/boards/da7219.c | 34 ++- sound/soc/intel/boards/bxt_da7219_max98357a.c | 7 + sound/soc/intel/boards/bytcr_wm5102.c | 7 + sound/soc/intel/boards/kbl_da7219_max98357a.c | 7 + sound/soc/intel/boards/kbl_da7219_max98927.c | 7 + sound/soc/intel/boards/sof_da7219_max98373.c | 10 + sound/soc/mediatek/mt8173/mt8173-max98090.c | 2 +- .../mediatek/mt8173/mt8173-rt5650-rt5514.c | 23 +- .../mediatek/mt8173/mt8173-rt5650-rt5676.c | 23 +- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 23 +- .../mediatek/mt8183/mt8183-da7219-max98357.c | 43 +++- .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 53 +++- .../mt8186/mt8186-mt6366-da7219-max98357.c | 6 + sound/soc/qcom/apq8016_sbc.c | 37 ++- sound/soc/qcom/sc7180.c | 40 +++- sound/soc/qcom/sc7280.c | 34 ++- sound/soc/qcom/sdm845.c | 32 ++- sound/soc/rockchip/rk3399_gru_sound.c | 7 +- sound/soc/rockchip/rockchip_rt5645.c | 22 +- sound/soc/samsung/littlemill.c | 27 ++- sound/soc/samsung/lowland.c | 8 +- sound/soc/samsung/midas_wm1811.c | 25 +- 27 files changed, 668 insertions(+), 146 deletions(-) base-commit: 626c67169f9972fffcdf3bc3864de421f162ebf5 -- 2.40.1

4 34

Re: linux-next: Tree for Aug 7 (sound/soc/fsl/)
by Randy Dunlap 08 Aug '23

08 Aug '23

On 8/6/23 22:47, Stephen Rothwell wrote: > Hi all, > > Changes since 20230804: > on PPC32: WARNING: unmet direct dependencies detected for SND_SOC_MPC5200_AC97 Depends on [n]: SOUND [=y] && SND [=y] && SND_SOC [=y] && SND_POWERPC_SOC [=y] && PPC_MPC52xx [=y] && PPC_BESTCOMM [=n] Selected by [y]: - SND_MPC52xx_SOC_PCM030 [=y] && SOUND [=y] && SND [=y] && SND_SOC [=y] && SND_POWERPC_SOC [=y] && PPC_MPC5200_SIMPLE [=y] - SND_MPC52xx_SOC_EFIKA [=y] && SOUND [=y] && SND [=y] && SND_SOC [=y] && SND_POWERPC_SOC [=y] && PPC_EFIKA [=y] powerpc-linux-ld: sound/soc/fsl/mpc5200_psc_ac97.o: in function `psc_ac97_of_remove': mpc5200_psc_ac97.c:(.text.psc_ac97_of_remove+0x1c): undefined reference to `mpc5200_audio_dma_destroy' powerpc-linux-ld: sound/soc/fsl/mpc5200_psc_ac97.o: in function `psc_ac97_of_probe': mpc5200_psc_ac97.c:(.text.psc_ac97_of_probe+0x20): undefined reference to `mpc5200_audio_dma_create' Full randconfig file is attached. -- ~Randy

1 0

[PATCH] ASoC: lower "no backend DAIs enabled for ... Port" log severity
by Hans de Goede 08 Aug '23

08 Aug '23

If SNDRV_PCM_IOCTL_PREPARE is called when the mixer settings linking frontend and backend have not been setup yet this results in e.g. the following errors getting logged: [ 43.244549] Baytrail Audio Port: ASoC: no backend DAIs enabled for Baytrail Audio Port [ 43.244744] Baytrail Audio Port: ASoC: error at dpcm_fe_dai_prepare on Baytrail Audio Port: -22 pipewire triggers this leading to 96 repeats of this in the log after the user has logged into a GNOME session. IMHO userspace should not be able to get the kernel to spam dmesg like this. Lower the severity of the "no backend DAIs enabled" log message for dai-s with the dynamic flag set to avoid this. And also changes _soc_pcm_ret() to not log for -EINVAL errors, to fix the other error. Userspace passing wrong parameters should not lead to dmesg messages. Link: https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/3407 Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/soc/soc-pcm.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 8896227e4fb7..fdc1a1c654d6 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -38,6 +38,7 @@ static inline int _soc_pcm_ret(struct snd_soc_pcm_runtime *rtd, switch (ret) { case -EPROBE_DEFER: case -ENOTSUPP: + case -EINVAL: break; default: dev_err(rtd->dev, @@ -2466,7 +2467,11 @@ static int dpcm_fe_dai_prepare(struct snd_pcm_substream *substream) /* there is no point preparing this FE if there are no BEs */ if (list_empty(&fe->dpcm[stream].be_clients)) { - dev_err(fe->dev, "ASoC: no backend DAIs enabled for %s\n", + if (fe->dai_link->dynamic) + dev_dbg(fe->dev, "ASoC: no backend DAIs enabled for %s\n", + fe->dai_link->name); + else + dev_err(fe->dev, "ASoC: no backend DAIs enabled for %s\n", fe->dai_link->name); ret = -EINVAL; goto out; -- 2.41.0

4 9

[PATCH V4 0/3] ASoC: codecs: Add awinic AW88261 audio amplifier driver
by wangweidong.a＠awinic.com 07 Aug '23

07 Aug '23

From: Weidong Wang <wangweidong.a(a)awinic.com> The awinic AW88261 is an I2S/TDM input, high efficiency digital Smart K audio amplifier v3 -> v4: Modify the dev_err_probe usage Changed the use of sizeof in kzalloc Changed the function name Merge aw88261_device.c into aw88261.c Weidong Wang (3): ASoC: dt-bindings: Add schema for "awinic,aw88261" ASoC: codecs: Add code for bin parsing compatible with aw88261 ASoC: codecs: Add aw88261 amplifier driver .../bindings/sound/awinic,aw88395.yaml | 4 +- sound/soc/codecs/Kconfig | 15 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/aw88261.c | 1298 +++++++++++++++++ sound/soc/codecs/aw88261.h | 459 ++++++ sound/soc/codecs/aw88395/aw88395_lib.c | 193 ++- sound/soc/codecs/aw88395/aw88395_reg.h | 1 + 7 files changed, 1954 insertions(+), 18 deletions(-) create mode 100644 sound/soc/codecs/aw88261.c create mode 100644 sound/soc/codecs/aw88261.h base-commit: c1a515d3c0270628df8ae5f5118ba859b85464a2 -- 2.41.0

2 4

[PATCH] ASoC: jz4740: Update to use maple tree register cache
by Mark Brown 07 Aug '23

07 Aug '23

The maple tree register cache is based on a much more modern data structure than the rbtree cache and makes optimisation choices which are probably more appropriate for modern systems than those made by the rbtree cache. In v6.5 it has also acquired the ability to generate multi-register writes in sync operations, bringing performance up to parity with the rbtree cache there. Update the jz4740 driver to use the more modern data structure. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/jz4740.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/jz4740.c b/sound/soc/codecs/jz4740.c index 7c25acf6ff0d..d1cea93bdb59 100644 --- a/sound/soc/codecs/jz4740.c +++ b/sound/soc/codecs/jz4740.c @@ -301,7 +301,7 @@ static const struct regmap_config jz4740_codec_regmap_config = { .reg_defaults = jz4740_codec_reg_defaults, .num_reg_defaults = ARRAY_SIZE(jz4740_codec_reg_defaults), - .cache_type = REGCACHE_RBTREE, + .cache_type = REGCACHE_MAPLE, }; static int jz4740_codec_probe(struct platform_device *pdev) --- base-commit: 06c2afb862f9da8dc5efa4b6076a0e48c3fbaaa5 change-id: 20230630-asoc-jz-maple-aacbd3a12bb6 Best regards, -- Mark Brown <broonie(a)kernel.org>

1 1

[PATCH 00/16] ASoC: SoundWire codecs: improve pm_runtime handling
by Pierre-Louis Bossart 07 Aug '23

07 Aug '23

This patchset improves the pm_runtime behavior in rare corner cases identified by the Intel CI in the last 6 months. a) in stress-tests, it's not uncommon to see the following type of warnings when the codec reports as ATTACHED "rt711 sdw:0:025d:0711:00: runtime PM trying to activate child device sdw:0:025d:0711:00 but parent (sdw-master-0) is not active" This warning was not correlated with any functional issue, but it exposed a design issue on when to enable pm_runtime. The recommended practice in the pm_runtime documentation is to keep the devices in 'suspended' mode and mark them as 'active' when they are really functional. b) enabling pm_runtime when the codec reports as ATTACHED also creates a problematic case when the ASoC pm_runtime_get_sync() will silently fail due to the -EACCESS error handling. This can happen when playback starts before the codec is enumerated. This patchset modifies the initial stages so that codecs are pm_runtime enabled in the .probe() callback, but become pm_runtime 'active' only when they report present. This is better aligned with the design of the pm_runtime helpers and improved CI results significantly. This patchset modifies all existing SoundWire codecs (except Qualcomm ones), but the pattern of changes is exactly the same in all patches. Pierre-Louis Bossart (16): ASoC: SoundWire codecs: return error status in probe ASoC: SoundWire codecs: make regmap cache-only in probe ASoC: rt5682-sdw: make regmap cache-only in probe ASoC: rt711: enable pm_runtime in probe, keep status as 'suspended' ASoC: rt711-sdca: enable pm_runtime in probe, keep status as 'suspended' ASoC: rt700: enable pm_runtime in probe, keep status as 'suspended' ASoC: rt1712-sdca: enable pm_runtime in probe, keep status as 'suspended' ASoC: rt712-sdca-dmic: enable pm_runtime in probe, keep status as 'suspended' ASoC: rt715: enable pm_runtime in probe, keep status as 'suspended' ASoC: rt715-sdca: enable pm_runtime in probe, keep status as 'suspended' ASoC: rt1308-sdw: enable pm_runtime in probe, keep status as 'suspended' ASoC: rt1316-sdw: enable pm_runtime in probe, keep status as 'suspended' ASoC: rt1318-sdw: enable pm_runtime in probe, keep status as 'suspended' ASoC: rt5682-sdw: enable pm_runtime in probe, keep status as 'suspended' ASoC: max98363: enable pm_runtime in probe, keep status as 'suspended' ASoC: max98373-sdw: enable pm_runtime in probe, keep status as 'suspended' sound/soc/codecs/max98363.c | 41 +++++++++++++--------- sound/soc/codecs/max98373-sdw.c | 47 ++++++++++++++----------- sound/soc/codecs/rt1308-sdw.c | 54 +++++++++++++++------------- sound/soc/codecs/rt1316-sdw.c | 44 ++++++++++++++--------- sound/soc/codecs/rt1318-sdw.c | 43 ++++++++++++++--------- sound/soc/codecs/rt5682-sdw.c | 56 ++++++++++++++++++------------ sound/soc/codecs/rt5682.c | 3 ++ sound/soc/codecs/rt700-sdw.c | 7 ++-- sound/soc/codecs/rt700.c | 46 +++++++++++++++--------- sound/soc/codecs/rt711-sdca-sdw.c | 3 +- sound/soc/codecs/rt711-sdca.c | 48 +++++++++++++++++-------- sound/soc/codecs/rt711-sdw.c | 7 ++-- sound/soc/codecs/rt711.c | 47 ++++++++++++++++--------- sound/soc/codecs/rt712-sdca-dmic.c | 47 +++++++++++++++---------- sound/soc/codecs/rt712-sdca-sdw.c | 3 +- sound/soc/codecs/rt712-sdca.c | 45 ++++++++++++++++-------- sound/soc/codecs/rt715-sdca-sdw.c | 5 +-- sound/soc/codecs/rt715-sdca.c | 41 ++++++++++++++++------ sound/soc/codecs/rt715-sdw.c | 9 ++--- sound/soc/codecs/rt715.c | 41 +++++++++++++++------- 20 files changed, 392 insertions(+), 245 deletions(-) -- 2.39.2

2 17

[PATCH v2 1/3] ASoC: dt-bindings: Convert maxim,max98925 to DT schema
by Krzysztof Kozlowski 07 Aug '23

07 Aug '23

Convert the Maxim Integrated MAX98925/MAX98926/MAX98927 speaker amplifier bindings to DT schema format. Changes during conversion: 1. Add "sound-dai-cells", already used by DTS. 2. Use "maxim,interleave-mode" instead previous "interleave-mode" and undocumented but used interleave_mode. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- Changes in v2: 1. Use maxim,interleave-mode as bool. Ryan, As original author, I added you as bindings maintainer. Is that okay or maybe someone else from Maxim should take care about the bindings? --- .../devicetree/bindings/sound/max9892x.txt | 44 --------- .../bindings/sound/maxim,max98925.yaml | 98 +++++++++++++++++++ 2 files changed, 98 insertions(+), 44 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/max9892x.txt create mode 100644 Documentation/devicetree/bindings/sound/maxim,max98925.yaml diff --git a/Documentation/devicetree/bindings/sound/max9892x.txt b/Documentation/devicetree/bindings/sound/max9892x.txt deleted file mode 100644 index 98cb9ba5b328..000000000000 --- a/Documentation/devicetree/bindings/sound/max9892x.txt +++ /dev/null @@ -1,44 +0,0 @@ -Maxim Integrated MAX98925/MAX98926/MAX98927 Speaker Amplifier - -This device supports I2C. - -Required properties: - - - compatible : should be one of the following - - "maxim,max98925" - - "maxim,max98926" - - "maxim,max98927" - - - vmon-slot-no : slot number used to send voltage information - or in inteleave mode this will be used as - interleave slot. - MAX98925/MAX98926 slot range : 0 ~ 30, Default : 0 - MAX98927 slot range : 0 ~ 15, Default : 0 - - - imon-slot-no : slot number used to send current information - MAX98925/MAX98926 slot range : 0 ~ 30, Default : 0 - MAX98927 slot range : 0 ~ 15, Default : 0 - - - interleave-mode : When using two MAX9892X in a system it is - possible to create ADC data that that will - overflow the frame size. Digital Audio Interleave - mode provides a means to output VMON and IMON data - from two devices on a single DOUT line when running - smaller frames sizes such as 32 BCLKS per LRCLK or - 48 BCLKS per LRCLK. - Range : 0 (off), 1 (on), Default : 0 - - - reg : the I2C address of the device for I2C - -Optional properties: - - reset-gpios : GPIO to reset the device - -Example: - -codec: max98927@3a { - compatible = "maxim,max98927"; - vmon-slot-no = <0>; - imon-slot-no = <1>; - interleave-mode = <0>; - reg = <0x3a>; -}; diff --git a/Documentation/devicetree/bindings/sound/maxim,max98925.yaml b/Documentation/devicetree/bindings/sound/maxim,max98925.yaml new file mode 100644 index 000000000000..32fd86204a7a --- /dev/null +++ b/Documentation/devicetree/bindings/sound/maxim,max98925.yaml @@ -0,0 +1,98 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/maxim,max98925.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Maxim Integrated MAX98925/MAX98926/MAX98927 speaker amplifier + +maintainers: + - Ryan Lee <ryans.lee(a)maximintegrated.com> + +properties: + compatible: + enum: + - maxim,max98925 + - maxim,max98926 + - maxim,max98927 + + reg: + maxItems: 1 + + reset-gpios: + maxItems: 1 + + '#sound-dai-cells': + const: 0 + + vmon-slot-no: + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 0 + maximum: 30 + default: 0 + description: + Slot number used to send voltage information or in inteleave mode this + will be used as interleave slot. + + imon-slot-no: + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 0 + maximum: 30 + default: 0 + description: + Slot number used to send current information. + + maxim,interleave-mode: + type: boolean + description: + When using two MAX9892X in a system it is possible to create ADC data + that will overflow the frame size. When enabled, the Digital Audio + Interleave mode provides a means to output VMON and IMON data from two + devices on a single DOUT line when running smaller frames sizes such as + 32 BCLKS per LRCLK or 48 BCLKS per LRCLK. + +required: + - compatible + - reg + +allOf: + - $ref: dai-common.yaml# + - if: + properties: + compatible: + contains: + enum: + - maxim,max98927 + then: + properties: + vmon-slot-no: + minimum: 0 + maximum: 15 + + imon-slot-no: + minimum: 0 + maximum: 15 + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + #include <dt-bindings/gpio/gpio.h> + audio-codec@3a { + compatible = "maxim,max98927"; + reg = <0x3a>; + #sound-dai-cells = <0>; + + pinctrl-0 = <&speaker_default>; + pinctrl-names = "default"; + + reset-gpios = <&tlmm 69 GPIO_ACTIVE_LOW>; + + vmon-slot-no = <1>; + imon-slot-no = <0>; + }; + }; -- 2.34.1

2 3

[PATCH v2] ASoC: rt5665: add missed regulator_bulk_disable
by Zhang Shurong 07 Aug '23

07 Aug '23

The driver forgets to call regulator_bulk_disable() Add the missed call to fix it. Fixes: 33ada14a26c8 ("ASoC: add rt5665 codec driver") Signed-off-by: Zhang Shurong <zhang_shurong(a)foxmail.com> --- Changes in v2: - move regulator_bulk_disable() in rt5665_remove() sound/soc/codecs/rt5665.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/rt5665.c b/sound/soc/codecs/rt5665.c index 83c367af91da..525713c33d71 100644 --- a/sound/soc/codecs/rt5665.c +++ b/sound/soc/codecs/rt5665.c @@ -4472,6 +4472,8 @@ static void rt5665_remove(struct snd_soc_component *component) struct rt5665_priv *rt5665 = snd_soc_component_get_drvdata(component); regmap_write(rt5665->regmap, RT5665_RESET, 0); + + regulator_bulk_disable(ARRAY_SIZE(rt5665->supplies), rt5665->supplies); } #ifdef CONFIG_PM -- 2.30.2

2 1

[PATCH 00/24] ALSA: Generic PCM copy ops using sockptr_t
by Takashi Iwai 07 Aug '23

07 Aug '23

Hi, this is a patch set to clean up the PCM copy ops using sockptr_t as a "universal" pointer, inspired by the recent patch from Andy Shevchenko: https://lore.kernel.org/r/20230721100146.67293-1-andriy.shevchenko@linux.in… Even though it sounds a bit weird, sockptr_t is a generic type that is used already in wide ranges, and it can fit our purpose, too. With sockptr_t, the former split of copy_user and copy_kernel PCM ops can be unified again gracefully. The patch set introduces the new PCM ops, converting users, and drops the old PCM ops. Most of conversions are straightforward, simply replacing copy_*_user() with copy_*_sockptr() variants. Note that the conversion in ASoC will fix a potential problem of ASoC PCM that has been for long time. Since ASoC component takes care of only copy_user, the conversion form/to kernel space might have been missing. With this patch set, both cases are handled with sockptr_t by a single callback. The patches are lightly tested (with a faked PCM copy implementation on HD-audio), while most of patches are only compile-tested. Takashi === Cc: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> Cc: Andrey Utkin <andrey_utkin(a)fastmail.com> Cc: Anton Sviridenko <anton(a)corp.bluecherry.net> Cc: Arnaud Pouliquen <arnaud.pouliquen(a)foss.st.com> Cc: Banajit Goswami <bgoswami(a)quicinc.com> Cc: Bluecherry Maintainers <maintainers(a)bluecherrydvr.com> Cc: Claudiu Beznea <claudiu.beznea(a)microchip.com> Cc: Ismael Luceno <ismael(a)iodev.co.uk> Cc: Lars-Peter Clausen <lars(a)metafoo.de> Cc: Mark Brown <broonie(a)kernel.org> Cc: Mauro Carvalho Chehab <mchehab(a)kernel.org> Cc: Oleksandr Andrushchenko <oleksandr_andrushchenko(a)epam.com> Cc: Olivier Moysan <olivier.moysan(a)foss.st.com> Cc: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Cc: linux-media(a)vger.kernel.org Cc: xen-devel(a)lists.xenproject.org === Takashi Iwai (24): ALSA: pcm: Add copy ops with universal sockptr_t ALSA: core: Add memory copy helpers between sockptr and iomem ALSA: dummy: Convert to generic PCM copy ops ALSA: gus: Convert to generic PCM copy ops ALSA: emu8000: Convert to generic PCM copy ops ALSA: es1938: Convert to generic PCM copy ops ALSA: korg1212: Convert to generic PCM copy ops ALSA: nm256: Convert to generic PCM copy ops ALSA: rme32: Convert to generic PCM copy ops ALSA: rme96: Convert to generic PCM copy ops ALSA: hdsp: Convert to generic PCM copy ops ALSA: rme9652: Convert to generic PCM copy ops ALSA: sh: Convert to generic PCM copy ops ALSA: xen: Convert to generic PCM copy ops ALSA: pcmtest: Update comment about PCM copy ops media: solo6x10: Convert to generic PCM copy ops ASoC: component: Add generic PCM copy ops ASoC: mediatek: Convert to generic PCM copy ops ASoC: qcom: Convert to generic PCM copy ops ASoC: dmaengine: Convert to generic PCM copy ops ASoC: dmaengine: Use sockptr_t for process callback, too ALSA: doc: Update description for the new PCM copy ops ASoC: pcm: Drop obsoleted PCM copy_user ops ALSA: pcm: Drop obsoleted PCM copy_user and copy_kernel ops .../kernel-api/writing-an-alsa-driver.rst | 59 +++++--------- drivers/media/pci/solo6x10/solo6x10-g723.c | 41 ++-------- include/sound/dmaengine_pcm.h | 2 +- include/sound/pcm.h | 12 +-- include/sound/soc-component.h | 14 ++-- sound/core/memory.c | 39 +++++++++ sound/core/pcm_lib.c | 81 +++++++++---------- sound/core/pcm_native.c | 2 +- sound/drivers/dummy.c | 12 +-- sound/drivers/pcmtest.c | 2 +- sound/isa/gus/gus_pcm.c | 23 +----- sound/isa/sb/emu8000_pcm.c | 79 +++++------------- sound/pci/es1938.c | 31 ++----- sound/pci/korg1212/korg1212.c | 46 +++-------- sound/pci/nm256/nm256.c | 42 ++-------- sound/pci/rme32.c | 50 +++--------- sound/pci/rme96.c | 48 +++-------- sound/pci/rme9652/hdsp.c | 42 ++-------- sound/pci/rme9652/rme9652.c | 46 ++--------- sound/sh/sh_dac_audio.c | 25 +----- sound/soc/atmel/mchp-pdmc.c | 2 +- sound/soc/mediatek/common/mtk-btcvsd.c | 22 ++--- sound/soc/qcom/lpass-platform.c | 12 +-- sound/soc/soc-component.c | 10 +-- sound/soc/soc-generic-dmaengine-pcm.c | 18 ++--- sound/soc/soc-pcm.c | 4 +- sound/soc/stm/stm32_sai_sub.c | 2 +- sound/xen/xen_snd_front_alsa.c | 55 +++---------- 28 files changed, 251 insertions(+), 570 deletions(-) -- 2.35.3

3 33

[PATCH v2] ASoC: lower "no backend DAIs enabled for ... Port" log severity
by Hans de Goede 07 Aug '23

07 Aug '23

If SNDRV_PCM_IOCTL_PREPARE is called when the mixer settings linking frontend and backend have not been setup yet this results in e.g. the following errors getting logged: [ 43.244549] Baytrail Audio Port: ASoC: no backend DAIs enabled for Baytrail Audio Port [ 43.244744] Baytrail Audio Port: ASoC: error at dpcm_fe_dai_prepare on Baytrail Audio Port: -22 pipewire triggers this leading to 96 lines getting logged after the user has logged into a GNOME session. Change the actual "no backend DAIs enabled for ... Port" error to dev_err_once() to avoid it getting repeated 48 times. While at it also improve the error by hinting the user how to fix this. To not make developing new UCM profiles harder, also log the error at dev_dbg() level all the time (vs once). So that e.g. dyndbg can be used to (re)enable the messages. Also changes _soc_pcm_ret() to not log for -EINVAL errors, to fix the other error getting logged 48 times. Userspace passing wrong parameters should not lead to dmesg messages. Link: https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/3407 Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- Changes in v2: - Switch the logging to a combination of dev_err_once() with improved log-message + dev_dbg() with shorter log message --- sound/soc/soc-pcm.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 8896227e4fb7..3aa6b988cb4b 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -38,6 +38,7 @@ static inline int _soc_pcm_ret(struct snd_soc_pcm_runtime *rtd, switch (ret) { case -EPROBE_DEFER: case -ENOTSUPP: + case -EINVAL: break; default: dev_err(rtd->dev, @@ -2466,8 +2467,11 @@ static int dpcm_fe_dai_prepare(struct snd_pcm_substream *substream) /* there is no point preparing this FE if there are no BEs */ if (list_empty(&fe->dpcm[stream].be_clients)) { - dev_err(fe->dev, "ASoC: no backend DAIs enabled for %s\n", - fe->dai_link->name); + /* dev_err_once() for visibility, dev_dbg() for debugging UCM profiles */ + dev_err_once(fe->dev, "ASoC: no backend DAIs enabled for %s, possibly missing ALSA mixer-based routing or UCM profile\n", + fe->dai_link->name); + dev_dbg(fe->dev, "ASoC: no backend DAIs enabled for %s\n", + fe->dai_link->name); ret = -EINVAL; goto out; } -- 2.41.0

3 2

[PATCH] sound: core: fix device ownership model in card and pcm
by cujomalainey＠chromium.org 07 Aug '23

07 Aug '23

From: Curtis Malainey <cujomalainey(a)chromium.org> The current implementation of how devices are released is valid for production use cases (root control of memory is handled by card_dev, all other devices are no-ops). This model does not work though in a kernel hacking environment where KASAN and delayed release on kobj is enabled. If the card_dev device is released before any of the child objects a use-after-free bug is caught by KASAN as the delayed release still has a reference to the devices that were freed by the card_dev release. Also both snd_card and snd_pcm both own two devices internally, so even if they released independently, the shared struct would result in another use after free. Solution is to move the child devices into their own memory so they can be handled independently and released on their own schedule. Signed-off-by: Curtis Malainey <cujomalainey(a)chromium.org> Cc: Doug Anderson <dianders(a)chromium.org> --- include/sound/core.h | 2 +- include/sound/pcm.h | 2 +- sound/core/control.c | 21 +++++++++++++++------ sound/core/control_led.c | 4 ++-- sound/core/pcm.c | 30 ++++++++++++++++++++---------- sound/usb/media.c | 4 ++-- 6 files changed, 41 insertions(+), 22 deletions(-) diff --git a/include/sound/core.h b/include/sound/core.h index f6e0dd648b80c..a08ab8c8cfb6d 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -96,7 +96,7 @@ struct snd_card { private data */ struct list_head devices; /* devices */ - struct device ctl_dev; /* control device */ + struct device *ctl_dev; /* control device */ unsigned int last_numid; /* last used numeric ID */ struct rw_semaphore controls_rwsem; /* controls lock (list and values) */ rwlock_t ctl_files_rwlock; /* ctl_files list lock */ diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 19f564606ac42..0243a13e9ac47 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -510,7 +510,7 @@ struct snd_pcm_str { #endif #endif struct snd_kcontrol *chmap_kctl; /* channel-mapping controls */ - struct device dev; + struct device *dev; }; struct snd_pcm { diff --git a/sound/core/control.c b/sound/core/control.c index 8386b53acdcd4..8bbaf3dffce62 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -2315,7 +2315,7 @@ static int snd_ctl_dev_register(struct snd_device *device) int err; err = snd_register_device(SNDRV_DEVICE_TYPE_CONTROL, card, -1, - &snd_ctl_f_ops, card, &card->ctl_dev); + &snd_ctl_f_ops, card, card->ctl_dev); if (err < 0) return err; down_read(&card->controls_rwsem); @@ -2351,7 +2351,7 @@ static int snd_ctl_dev_disconnect(struct snd_device *device) up_read(&snd_ctl_layer_rwsem); up_read(&card->controls_rwsem); - return snd_unregister_device(&card->ctl_dev); + return snd_unregister_device(card->ctl_dev); } /* @@ -2373,10 +2373,15 @@ static int snd_ctl_dev_free(struct snd_device *device) xa_destroy(&card->ctl_hash); #endif up_write(&card->controls_rwsem); - put_device(&card->ctl_dev); + put_device(card->ctl_dev); return 0; } +static void snd_ctl_dev_release(struct device *dev) +{ + kfree(dev); +} + /* * create control core: * called from init.c @@ -2394,13 +2399,17 @@ int snd_ctl_create(struct snd_card *card) return -ENXIO; if (snd_BUG_ON(card->number < 0 || card->number >= SNDRV_CARDS)) return -ENXIO; + card->ctl_dev = kzalloc(sizeof(*card->ctl_dev), GFP_KERNEL); + if (!card->ctl_dev) + return -ENOMEM; - snd_device_initialize(&card->ctl_dev, card); - dev_set_name(&card->ctl_dev, "controlC%d", card->number); + snd_device_initialize(card->ctl_dev, card); + card->ctl_dev->release = snd_ctl_dev_release; + dev_set_name(card->ctl_dev, "controlC%d", card->number); err = snd_device_new(card, SNDRV_DEV_CONTROL, card, &ops); if (err < 0) - put_device(&card->ctl_dev); + put_device(card->ctl_dev); return err; } diff --git a/sound/core/control_led.c b/sound/core/control_led.c index ee77547bf8dcb..760e46cf25cc8 100644 --- a/sound/core/control_led.c +++ b/sound/core/control_led.c @@ -688,7 +688,7 @@ static void snd_ctl_led_sysfs_add(struct snd_card *card) goto cerr; led->cards[card->number] = led_card; snprintf(link_name, sizeof(link_name), "led-%s", led->name); - WARN(sysfs_create_link(&card->ctl_dev.kobj, &led_card->dev.kobj, link_name), + WARN(sysfs_create_link(&card->ctl_dev->kobj, &led_card->dev.kobj, link_name), "can't create symlink to controlC%i device\n", card->number); WARN(sysfs_create_link(&led_card->dev.kobj, &card->card_dev.kobj, "card"), "can't create symlink to card%i\n", card->number); @@ -714,7 +714,7 @@ static void snd_ctl_led_sysfs_remove(struct snd_card *card) if (!led_card) continue; snprintf(link_name, sizeof(link_name), "led-%s", led->name); - sysfs_remove_link(&card->ctl_dev.kobj, link_name); + sysfs_remove_link(&card->ctl_dev->kobj, link_name); sysfs_remove_link(&led_card->dev.kobj, "card"); device_unregister(&led_card->dev); led->cards[card->number] = NULL; diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 9d95e37311230..9026ccc56dbe7 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -604,7 +604,7 @@ static const struct attribute_group *pcm_dev_attr_groups[]; #ifdef CONFIG_PM_SLEEP static int do_pcm_suspend(struct device *dev) { - struct snd_pcm_str *pstr = container_of(dev, struct snd_pcm_str, dev); + struct snd_pcm_str *pstr = dev_get_drvdata(dev); if (!pstr->pcm->no_device_suspend) snd_pcm_suspend_all(pstr->pcm); @@ -622,6 +622,11 @@ static const struct device_type pcm_dev_type = { .pm = &pcm_dev_pm_ops, }; +static void snd_pcm_dev_release(struct device *dev) +{ + kfree(dev); +} + /** * snd_pcm_new_stream - create a new PCM stream * @pcm: the pcm instance @@ -641,6 +646,10 @@ int snd_pcm_new_stream(struct snd_pcm *pcm, int stream, int substream_count) struct snd_pcm_str *pstr = &pcm->streams[stream]; struct snd_pcm_substream *substream, *prev; + pstr->dev = kzalloc(sizeof(*pstr->dev), GFP_KERNEL); + if (!pstr->dev) + return -ENOMEM; + dev_set_drvdata(pstr->dev, pstr); #if IS_ENABLED(CONFIG_SND_PCM_OSS) mutex_init(&pstr->oss.setup_mutex); #endif @@ -650,10 +659,11 @@ int snd_pcm_new_stream(struct snd_pcm *pcm, int stream, int substream_count) if (!substream_count) return 0; - snd_device_initialize(&pstr->dev, pcm->card); - pstr->dev.groups = pcm_dev_attr_groups; - pstr->dev.type = &pcm_dev_type; - dev_set_name(&pstr->dev, "pcmC%iD%i%c", pcm->card->number, pcm->device, + snd_device_initialize(pstr->dev, pcm->card); + pstr->dev->release = snd_pcm_dev_release; + pstr->dev->groups = pcm_dev_attr_groups; + pstr->dev->type = &pcm_dev_type; + dev_set_name(pstr->dev, "pcmC%iD%i%c", pcm->card->number, pcm->device, stream == SNDRV_PCM_STREAM_PLAYBACK ? 'p' : 'c'); if (!pcm->internal) { @@ -699,7 +709,7 @@ int snd_pcm_new_stream(struct snd_pcm *pcm, int stream, int substream_count) prev = substream; } return 0; -} +} EXPORT_SYMBOL(snd_pcm_new_stream); static int _snd_pcm_new(struct snd_card *card, const char *id, int device, @@ -847,7 +857,7 @@ static void snd_pcm_free_stream(struct snd_pcm_str * pstr) #endif free_chmap(pstr); if (pstr->substream_count) - put_device(&pstr->dev); + put_device(pstr->dev); } #if IS_ENABLED(CONFIG_SND_PCM_OSS) @@ -1017,7 +1027,7 @@ void snd_pcm_detach_substream(struct snd_pcm_substream *substream) static ssize_t pcm_class_show(struct device *dev, struct device_attribute *attr, char *buf) { - struct snd_pcm_str *pstr = container_of(dev, struct snd_pcm_str, dev); + struct snd_pcm_str *pstr = dev_get_drvdata(dev); struct snd_pcm *pcm = pstr->pcm; const char *str; static const char *strs[SNDRV_PCM_CLASS_LAST + 1] = { @@ -1078,7 +1088,7 @@ static int snd_pcm_dev_register(struct snd_device *device) /* register pcm */ err = snd_register_device(devtype, pcm->card, pcm->device, &snd_pcm_f_ops[cidx], pcm, - &pcm->streams[cidx].dev); + pcm->streams[cidx].dev); if (err < 0) { list_del_init(&pcm->list); goto unlock; @@ -1125,7 +1135,7 @@ static int snd_pcm_dev_disconnect(struct snd_device *device) pcm_call_notify(pcm, n_disconnect); for (cidx = 0; cidx < 2; cidx++) { - snd_unregister_device(&pcm->streams[cidx].dev); + snd_unregister_device(pcm->streams[cidx].dev); free_chmap(&pcm->streams[cidx]); } mutex_unlock(&pcm->open_mutex); diff --git a/sound/usb/media.c b/sound/usb/media.c index 840f42cb9272c..d48db6f3ae659 100644 --- a/sound/usb/media.c +++ b/sound/usb/media.c @@ -35,7 +35,7 @@ int snd_media_stream_init(struct snd_usb_substream *subs, struct snd_pcm *pcm, { struct media_device *mdev; struct media_ctl *mctl; - struct device *pcm_dev = &pcm->streams[stream].dev; + struct device *pcm_dev = pcm->streams[stream].dev; u32 intf_type; int ret = 0; u16 mixer_pad; @@ -163,7 +163,7 @@ void snd_media_stop_pipeline(struct snd_usb_substream *subs) static int snd_media_mixer_init(struct snd_usb_audio *chip) { - struct device *ctl_dev = &chip->card->ctl_dev; + struct device *ctl_dev = chip->card->ctl_dev; struct media_intf_devnode *ctl_intf; struct usb_mixer_interface *mixer; struct media_device *mdev = chip->media_dev; -- 2.41.0.585.gd2178a4bd4-goog

6 16

[PATCH v1 0/5] Add I2S support for the StarFive JH7110 SoC
by Xingyu Wu 07 Aug '23

07 Aug '23

This patch series adds I2S support for the StarFive JH7110 RISC-V SoC based on Designware I2S controller. There has three I2S channels (RX/TX0/TX1) on the JH7110 SoC, one of which is for record(RX) and two for playback(TX). The first patch adds the ops to get data from platform bus in the I2S driver. The second patch adds support for the StarFive JH7110 SoC in the Designware I2S bindings. The third patch adds support for the StarFive JH7110 SoC in the Designware I2S driver. The fourth patch fixes the name of I2STX1 pinmux. The last patch adds device node of I2S RX/TX0/TX1 in JH7110 dts. This patch series is based on Linux-next which is merge clock, syscon and dma nodes for the StarFive JH7110 SoC. And these should be applied after the following patchset: https://lore.kernel.org/all/20230724055440.100947-1-xingyu.wu@starfivetech.… The series has been tested and works normally on the VisionFive 2 board by plugging an audio expansion board. Xingyu Wu (5): ASoC: dwc: Use ops to get platform data ASoC: dt-bindings: snps,designware-i2s: Add StarFive JH7110 SoC support ASoC: dwc: i2s: Add StarFive JH7110 SoC support riscv: dts: starfive: pinfunc: Fix the pins name of I2STX1 riscv: dts: starfive: Add the nodes and pins of I2Srx/I2Stx0/I2Stx1 .../bindings/sound/snps,designware-i2s.yaml | 101 +++++- arch/riscv/boot/dts/starfive/jh7110-pinfunc.h | 4 +- .../jh7110-starfive-visionfive-2.dtsi | 58 ++++ arch/riscv/boot/dts/starfive/jh7110.dtsi | 65 ++++ include/sound/designware_i2s.h | 3 + sound/soc/dwc/dwc-i2s.c | 318 ++++++++++++++++-- sound/soc/dwc/local.h | 1 + 7 files changed, 523 insertions(+), 27 deletions(-) base-commit: 0fee53365d051781156b75c6f6e6015757e71fd8 prerequisite-patch-id: 0159f09bb0a1ff711a00ae17ef5b12662c9c7d3d -- 2.25.1

3 13

[PATCH v2 0/3] ASoC: mt8188-mt6359: add SOF support
by Trevor Wu 07 Aug '23

07 Aug '23

This series introduces dynamic pinctrl and adds support for the SOF on the mt8188-mt6359 machine driver. Changes since v1: - minor changes to incorporate the suggestions from the reviewer. Trevor Wu (3): ASoC: mediatek: mt8188-mt6359: support dynamic pinctrl ASoC: mediatek: common: revise SOF common code ASoC: mediatek: mt8188-mt6359: add SOF support .../soc/mediatek/common/mtk-dsp-sof-common.c | 106 ++++++-- .../soc/mediatek/common/mtk-dsp-sof-common.h | 8 + sound/soc/mediatek/mt8188/mt8188-mt6359.c | 239 +++++++++++++++++- 3 files changed, 325 insertions(+), 28 deletions(-) -- 2.18.0

3 6

[PATCH v5] um/drivers: fix hostaudio build errors
by Randy Dunlap 06 Aug '23

06 Aug '23

Use "select" to ensure that the required kconfig symbols are set as expected. Drop HOSTAUDIO since it is now equivalent to UML_SOUND. Set CONFIG_SOUND=m in ARCH=um defconfig files to maintain the status quo of the default configs. Allow SOUND with UML regardless of HAS_IOMEM. Otherwise there is a kconfig warning for unmet dependencies. (This was not an issue when SOUND was defined in arch/um/drivers/Kconfig. I have done 50 randconfig builds and didn't find any issues.) This fixes build errors when CONFIG_SOUND is not set: ld: arch/um/drivers/hostaudio_kern.o: in function `hostaudio_cleanup_module': hostaudio_kern.c:(.exit.text+0xa): undefined reference to `unregister_sound_mixer' ld: hostaudio_kern.c:(.exit.text+0x15): undefined reference to `unregister_sound_dsp' ld: arch/um/drivers/hostaudio_kern.o: in function `hostaudio_init_module': hostaudio_kern.c:(.init.text+0x19): undefined reference to `register_sound_dsp' ld: hostaudio_kern.c:(.init.text+0x31): undefined reference to `register_sound_mixer' ld: hostaudio_kern.c:(.init.text+0x49): undefined reference to `unregister_sound_dsp' and this kconfig warning: WARNING: unmet direct dependencies detected for SOUND Fixes: 1da177e4c3f4 ("Linux-2.6.12-rc2") Fixes: d886e87cb82b ("sound: make OSS sound core optional") Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Reported-by: kernel test robot <lkp(a)intel.com> Closes: lore.kernel.org/r/202307141416.vxuRVpFv-lkp@intel.com Cc: Richard Weinberger <richard(a)nod.at> Cc: Anton Ivanov <anton.ivanov(a)cambridgegreys.com> Cc: Johannes Berg <johannes(a)sipsolutions.net> Cc: linux-um(a)lists.infradead.org Cc: Tejun Heo <tj(a)kernel.org> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Masahiro Yamada <masahiroy(a)kernel.org> Cc: Nathan Chancellor <nathan(a)kernel.org> Cc: Nick Desaulniers <ndesaulniers(a)google.com> Cc: Nicolas Schier <nicolas(a)fjasle.eu> Cc: linux-kbuild(a)vger.kernel.org Cc: alsa-devel(a)alsa-project.org --- v2: don't delete the HOSTAUDIO Kconfig entry (Masahiro) v3: drop HOSTAUDIO and use CONFIG_UML_SOUND for it in Makefile (Takashi); add SOUND depends on "|| UML" to HAS_IOMEM v4: use depends on instead of select for SOUND (Masahiro); use Closes: instead of Link: v5: update ARCH=um defconfig files (Masahiro) arch/um/configs/i386_defconfig | 1 + arch/um/configs/x86_64_defconfig | 1 + arch/um/drivers/Kconfig | 16 +++------------- arch/um/drivers/Makefile | 2 +- sound/Kconfig | 2 +- 5 files changed, 7 insertions(+), 15 deletions(-) diff -- a/arch/um/drivers/Kconfig b/arch/um/drivers/Kconfig --- a/arch/um/drivers/Kconfig +++ b/arch/um/drivers/Kconfig @@ -111,24 +111,14 @@ config SSL_CHAN config UML_SOUND tristate "Sound support" + depends on SOUND + select SOUND_OSS_CORE help This option enables UML sound support. If enabled, it will pull in - soundcore and the UML hostaudio relay, which acts as a intermediary + the UML hostaudio relay, which acts as a intermediary between the host's dsp and mixer devices and the UML sound system. It is safe to say 'Y' here. -config SOUND - tristate - default UML_SOUND - -config SOUND_OSS_CORE - bool - default UML_SOUND - -config HOSTAUDIO - tristate - default UML_SOUND - endmenu menu "UML Network Devices" diff -- a/sound/Kconfig b/sound/Kconfig --- a/sound/Kconfig +++ b/sound/Kconfig @@ -1,7 +1,7 @@ # SPDX-License-Identifier: GPL-2.0-only menuconfig SOUND tristate "Sound card support" - depends on HAS_IOMEM + depends on HAS_IOMEM || UML help If you have a sound card in your computer, i.e. if it can say more than an occasional beep, say Y. diff -- a/arch/um/drivers/Makefile b/arch/um/drivers/Makefile --- a/arch/um/drivers/Makefile +++ b/arch/um/drivers/Makefile @@ -54,7 +54,7 @@ obj-$(CONFIG_UML_NET) += net.o obj-$(CONFIG_MCONSOLE) += mconsole.o obj-$(CONFIG_MMAPPER) += mmapper_kern.o obj-$(CONFIG_BLK_DEV_UBD) += ubd.o -obj-$(CONFIG_HOSTAUDIO) += hostaudio.o +obj-$(CONFIG_UML_SOUND) += hostaudio.o obj-$(CONFIG_NULL_CHAN) += null.o obj-$(CONFIG_PORT_CHAN) += port.o obj-$(CONFIG_PTY_CHAN) += pty.o diff -- a/arch/um/configs/i386_defconfig b/arch/um/configs/i386_defconfig --- a/arch/um/configs/i386_defconfig +++ b/arch/um/configs/i386_defconfig @@ -34,6 +34,7 @@ CONFIG_TTY_CHAN=y CONFIG_XTERM_CHAN=y CONFIG_CON_CHAN="pts" CONFIG_SSL_CHAN="pts" +CONFIG_SOUND=m CONFIG_UML_SOUND=m CONFIG_DEVTMPFS=y CONFIG_DEVTMPFS_MOUNT=y diff -- a/arch/um/configs/x86_64_defconfig b/arch/um/configs/x86_64_defconfig --- a/arch/um/configs/x86_64_defconfig +++ b/arch/um/configs/x86_64_defconfig @@ -32,6 +32,7 @@ CONFIG_TTY_CHAN=y CONFIG_XTERM_CHAN=y CONFIG_CON_CHAN="pts" CONFIG_SSL_CHAN="pts" +CONFIG_SOUND=m CONFIG_UML_SOUND=m CONFIG_DEVTMPFS=y CONFIG_DEVTMPFS_MOUNT=y

2 1

change order in which kernel initializes devices
by Fourhundred Thecat 06 Aug '23

06 Aug '23

Hello, I am using monolithic kernel without loadable module support. Everything is compiled in statically. Now I have snd-aloop (CONFIG_SND_ALOOP=y) and when i boot my kernel, it is recognized as primary device. My real hw card is as second device. Is there any way I can change the order, in which kernel initializes the sound devices, so that my real (hw) sound card is recognized first ? If I was using modules, I could simply load snd-aloop later. I am compiling my own kernel, so I would be happy to patch the code if necessary, if somebody could point me into the right section. Here is output of aplay -l, in case it helps: **** List of PLAYBACK Hardware Devices **** card 0: Loopback [Loopback], device 0: Loopback PCM [Loopback PCM] Subdevices: 8/8 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 Subdevice #2: subdevice #2 Subdevice #3: subdevice #3 Subdevice #4: subdevice #4 Subdevice #5: subdevice #5 Subdevice #6: subdevice #6 Subdevice #7: subdevice #7 card 0: Loopback [Loopback], device 1: Loopback PCM [Loopback PCM] Subdevices: 8/8 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 Subdevice #2: subdevice #2 Subdevice #3: subdevice #3 Subdevice #4: subdevice #4 Subdevice #5: subdevice #5 Subdevice #6: subdevice #6 Subdevice #7: subdevice #7 card 1: PCH [HDA Intel PCH], device 0: Generic Analog [Generic Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 PS: why are there 2 loopback devices, with 8 subdevices? When I am alredy at it, can I change tat in the kernel, so that I only have 1 loopback device for simplicity ? thank you,

1 0

[PATCH 1/2] ASoC: dt-bindings: wlf,wm8960: Describe the power supplies
by Fabio Estevam 06 Aug '23

06 Aug '23

From: Fabio Estevam <festevam(a)denx.de> WM8960 has the following power supplies: - AVDD - DBVDD - DCVDD - SPKVDD1 - SPKVDD1 Add them to bindings to improve the hardware description. Signed-off-by: Fabio Estevam <festevam(a)denx.de> --- .../devicetree/bindings/sound/wlf,wm8960.yaml | 20 +++++++++++++++++++ 1 file changed, 20 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/wlf,wm8960.yaml b/Documentation/devicetree/bindings/sound/wlf,wm8960.yaml index ee8eba7f0104..82519f6720ef 100644 --- a/Documentation/devicetree/bindings/sound/wlf,wm8960.yaml +++ b/Documentation/devicetree/bindings/sound/wlf,wm8960.yaml @@ -26,6 +26,21 @@ properties: '#sound-dai-cells': const: 0 + AVDD-supply: + description: Analogue supply. + + DBVDD-supply: + description: Digital Buffer Supply. + + DCVDD-supply: + description: Digital Core Supply. + + SPKVDD1-supply: + description: Supply for speaker drivers 1. + + SPKVDD2-supply: + description: Supply for speaker drivers 2. + wlf,capless: type: boolean description: @@ -84,5 +99,10 @@ examples: wlf,hp-cfg = <3 2 3>; wlf,gpio-cfg = <1 3>; wlf,shared-lrclk; + DCVDD-supply = <&reg_audio>; + DBVDD-supply = <&reg_audio>; + AVDD-supply = <&reg_audio>; + SPKVDD1-supply = <&reg_audio>; + SPKVDD2-supply = <&reg_audio>; }; }; -- 2.34.1

5 6

[PATCH] ASoC: max98363: don't return on success reading revision ID
by Bard Liao 04 Aug '23

04 Aug '23

max98363_io_init needs to keep going when we read revision ID successfully. Fixes: 18c0af945fa3 ("ASoC: max98363: add soundwire amplifier driver") Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> --- sound/soc/codecs/max98363.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/max98363.c b/sound/soc/codecs/max98363.c index b5c69bba0e48..2dfaf4fcfbd3 100644 --- a/sound/soc/codecs/max98363.c +++ b/sound/soc/codecs/max98363.c @@ -185,10 +185,10 @@ static int max98363_io_init(struct sdw_slave *slave) pm_runtime_get_noresume(dev); ret = regmap_read(max98363->regmap, MAX98363_R21FF_REV_ID, &reg); - if (!ret) { + if (!ret) dev_info(dev, "Revision ID: %X\n", reg); - return ret; - } + else + goto out; if (max98363->first_hw_init) { regcache_cache_bypass(max98363->regmap, false); @@ -198,10 +198,11 @@ static int max98363_io_init(struct sdw_slave *slave) max98363->first_hw_init = true; max98363->hw_init = true; +out: pm_runtime_mark_last_busy(dev); pm_runtime_put_autosuspend(dev); - return 0; + return ret; } #define MAX98363_RATES SNDRV_PCM_RATE_8000_192000 -- 2.25.1

2 1

[PATCH v2 00/15] ASoC: add multi Component support
by Kuninori Morimoto 04 Aug '23

04 Aug '23

Hi Mark These are v2 of multi Component support. On below HW case, we would like to use it as "2 Cards", but unfortunately it is impossible in intuitive way, or possible but not intuitive way. In reality, it is handled as "1 big Card" today. +-- basic board --------+ |+--------+ | || CPU ch0| <--> CodecA | || ch1| <-+ | |+--------+ | | +-------------|---------+ +-- expansion board ----+ | | | | +-> CodecB| +-----------------------+ To handling it as intuitive "2 Cards", this patch-set adds multi Component support. To enable this patch-set, I included [01/15] patch into this patch-set which is posted but not yet accepted. v1 -> v2 - include [01/15] patch into this patch-set - add related member to To - add DT member to [14/15][15/15] Link: https://lore.kernel.org/all/87a6b6cofh.wl-kuninori.morimoto.gx@renesas.com/ Link: https://lore.kernel.org/r/87r0q5blta.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/20230623-asoc-fix-meson-probe-v1-1-82b2c2ec5ca4@k… Kuninori Morimoto (15): ASoC: soc-core: protect dlc->of_node under mutex ASoC: soc-core.c: initialize dlc on snd_soc_get_dai_id() ASoC: soc-core.c: cleanup soc_dai_link_sanity_check() ASoC: soc-dai.c: add DAI get/match functions ASoC: soc-core.c: enable multi Component ASoC: soc-core.c: add snd_soc_get_dai_via_args() ASoC: soc-core.c: add snd_soc_dlc_use_cpu_as_platform() ASoC: soc-core.c: add snd_soc_copy_dai_args() ASoC: simple-card-utils.c: enable multi Component support ASoC: simple-card.c: enable multi Component support ASoC: rsnd: use DAI driver ID instead of DAI ID ASoC: rsnd: cleanup rsnd_dai_of_node() ASoC: rsnd: enable multi Component support for Audio Graph Card/Card2 ASoC: dt-bindings: renesas,rsnd.yaml: add common port-def ASoC: dt-bindings: renesas,rsnd.yaml: enable multi ports for multi Component support .../bindings/sound/renesas,rsnd.yaml | 60 ++-- include/sound/simple_card_utils.h | 5 +- include/sound/soc-dai.h | 3 + include/sound/soc.h | 6 + sound/soc/generic/audio-graph-card.c | 2 +- sound/soc/generic/audio-graph-card2.c | 2 +- sound/soc/generic/simple-card-utils.c | 23 +- sound/soc/generic/simple-card.c | 20 +- sound/soc/sh/rcar/core.c | 146 +++++++--- sound/soc/sh/rcar/rsnd.h | 4 + sound/soc/soc-core.c | 273 +++++++++++++----- 11 files changed, 378 insertions(+), 166 deletions(-) -- 2.25.1

3 24

[PATCH] ASoC: SOF: mediatek: mt8186 modify dram type as non-cache
by Trevor Wu 04 Aug '23

04 Aug '23

To prevent incorrect access between the host and DSP sides, we need to modify DRAM as a non-cache memory type. Additionally, we can retrieve the size of shared DMA from the device tree. Signed-off-by: Trevor Wu <trevor.wu(a)mediatek.com> Reviewed-by: Yaochun Hung <yc.hung(a)mediatek.com> Reviewed-by: Kuan-Hsun Cheng <Allen-KH.Cheng(a)mediatek.com> --- sound/soc/sof/mediatek/mt8186/mt8186.c | 40 +++++++++++++++----------- 1 file changed, 23 insertions(+), 17 deletions(-) diff --git a/sound/soc/sof/mediatek/mt8186/mt8186.c b/sound/soc/sof/mediatek/mt8186/mt8186.c index 3e0ea0e109e2..f587edf9e0a7 100644 --- a/sound/soc/sof/mediatek/mt8186/mt8186.c +++ b/sound/soc/sof/mediatek/mt8186/mt8186.c @@ -111,6 +111,14 @@ static int platform_parse_resource(struct platform_device *pdev, void *data) dev_dbg(dev, "DMA %pR\n", &res); + adsp->pa_shared_dram = (phys_addr_t)res.start; + adsp->shared_size = resource_size(&res); + if (adsp->pa_shared_dram & DRAM_REMAP_MASK) { + dev_err(dev, "adsp shared dma memory(%#x) is not 4K-aligned\n", + (u32)adsp->pa_shared_dram); + return -EINVAL; + } + ret = of_reserved_mem_device_init(dev); if (ret) { dev_err(dev, "of_reserved_mem_device_init failed\n"); @@ -244,23 +252,18 @@ static int adsp_shared_base_ioremap(struct platform_device *pdev, void *data) { struct device *dev = &pdev->dev; struct mtk_adsp_chip_info *adsp = data; - u32 shared_size; /* remap shared-dram base to be non-cachable */ - shared_size = TOTAL_SIZE_SHARED_DRAM_FROM_TAIL; - adsp->pa_shared_dram = adsp->pa_dram + adsp->dramsize - shared_size; - if (adsp->va_dram) { - adsp->shared_dram = adsp->va_dram + DSP_DRAM_SIZE - shared_size; - } else { - adsp->shared_dram = devm_ioremap(dev, adsp->pa_shared_dram, - shared_size); - if (!adsp->shared_dram) { - dev_err(dev, "ioremap failed for shared DRAM\n"); - return -ENOMEM; - } + adsp->shared_dram = devm_ioremap(dev, adsp->pa_shared_dram, + adsp->shared_size); + if (!adsp->shared_dram) { + dev_err(dev, "failed to ioremap base %pa size %#x\n", + adsp->shared_dram, adsp->shared_size); + return -ENOMEM; } - dev_dbg(dev, "shared-dram vbase=%p, phy addr :%pa, size=%#x\n", - adsp->shared_dram, &adsp->pa_shared_dram, shared_size); + + dev_dbg(dev, "shared-dram vbase=%p, phy addr :%pa, size=%#x\n", + adsp->shared_dram, &adsp->pa_shared_dram, adsp->shared_size); return 0; } @@ -307,9 +310,12 @@ static int mt8186_dsp_probe(struct snd_sof_dev *sdev) return -ENOMEM; } - sdev->bar[SOF_FW_BLK_TYPE_SRAM] = devm_ioremap_wc(sdev->dev, - priv->adsp->pa_dram, - priv->adsp->dramsize); + priv->adsp->va_sram = sdev->bar[SOF_FW_BLK_TYPE_IRAM]; + + sdev->bar[SOF_FW_BLK_TYPE_SRAM] = devm_ioremap(sdev->dev, + priv->adsp->pa_dram, + priv->adsp->dramsize); + if (!sdev->bar[SOF_FW_BLK_TYPE_SRAM]) { dev_err(sdev->dev, "failed to ioremap base %pa size %#x\n", &priv->adsp->pa_dram, priv->adsp->dramsize); -- 2.18.0

4 3

[PATCH -next] ASoC: ti: davinci-evm: simplify the code with module_platform_driver
by Yang Yingliang 04 Aug '23

04 Aug '23

The init/exit() of driver only calls platform_driver_register/unregister, it can be simpilfied with module_platform_driver. Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/soc/ti/davinci-evm.c | 13 +------------ 1 file changed, 1 insertion(+), 12 deletions(-) diff --git a/sound/soc/ti/davinci-evm.c b/sound/soc/ti/davinci-evm.c index 983d69b951b0..544cb3da50eb 100644 --- a/sound/soc/ti/davinci-evm.c +++ b/sound/soc/ti/davinci-evm.c @@ -257,18 +257,7 @@ static struct platform_driver davinci_evm_driver = { }, }; -static int __init evm_init(void) -{ - return platform_driver_register(&davinci_evm_driver); -} - -static void __exit evm_exit(void) -{ - platform_driver_unregister(&davinci_evm_driver); -} - -module_init(evm_init); -module_exit(evm_exit); +module_platform_driver(davinci_evm_driver); MODULE_AUTHOR("Vladimir Barinov"); MODULE_DESCRIPTION("TI DAVINCI EVM ASoC driver"); -- 2.25.1

2 1

[PATCH v2 0/9] sound: Use -EPROBE_DEFER instead of i915 module loading.
by Maarten Lankhorst 04 Aug '23

04 Aug '23

Explicitly loading i915 becomes a problem when upstreaming the new intel driver for Tiger Lake and higher graphics (xe). By loading i915, it doesn't wait for driver load of xe, and will fail completely before it loads. -EPROBE_DEFER has to be returned before any device is created in probe(), otherwise the removal of the device will cause EPROBE_DEFER to try again in an infinite loop. The conversion is done in gradual steps. First I add an argument to snd_hdac_i915_init to allow for -EPROBE_DEFER so I can convert each driver separately. Then I convert each driver to move snd_hdac_i915_init out of the workqueue. Finally I drop the ability to choose modprobe behavior after the last user is converted. I suspect the avs and skylake drivers used snd_hdac_i915_init purely for the modprobe, but I don't have the hardware to test if it can be safely removed. It can still be done easily in a followup patch to simplify probing. --- New since first version: - snd_hda_core.gpu_bind is added as a mechanism to force gpu binding, for testing. snd_hda_core.gpu_bind=0 forces waiting for GPU bind to off, snd_hda_core.gpu_bind=1 forces waiting for gpu bind. Default setting depends on whether kernel booted with nomodeset. - Incorporated all feedback review. Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Cezary Rojewski <cezary.rojewski(a)intel.com> Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Cc: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> Cc: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Cc: Bard Liao <yung-chuan.liao(a)linux.intel.com> Cc: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Cc: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Daniel Baluta <daniel.baluta(a)nxp.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-kernel(a)vger.kernel.org Cc: sound-open-firmware(a)alsa-project.org Maarten Lankhorst (9): ALSA: hda/intel: Fix error handling in azx_probe() ALSA: hda/i915: Allow override of gpu binding. ALSA: hda/i915: Add an allow_modprobe argument to snd_hdac_i915_init ALSA: hda/i915: Allow xe as match for i915_component_master_match ASoC: Intel: avs: Move snd_hdac_i915_init to before probe_work. ASoC: Intel: Skylake: Move snd_hdac_i915_init to before probe_work. ALSA: hda/intel: Move snd_hdac_i915_init to before probe_work. ASoC: SOF: Intel: Remove deferred probe for SOF ALSA: hda/i915: Remove extra argument from snd_hdac_i915_init sound/hda/hdac_i915.c | 25 ++++++++------- sound/pci/hda/hda_intel.c | 60 ++++++++++++++++++----------------- sound/soc/intel/avs/core.c | 13 +++++--- sound/soc/intel/skylake/skl.c | 31 ++++++------------ sound/soc/sof/Kconfig | 19 ----------- sound/soc/sof/core.c | 38 ++-------------------- sound/soc/sof/intel/Kconfig | 1 - sound/soc/sof/intel/hda.c | 32 +++++++++++-------- sound/soc/sof/sof-pci-dev.c | 3 +- sound/soc/sof/sof-priv.h | 5 --- 10 files changed, 85 insertions(+), 142 deletions(-) -- 2.39.2

6 36

[PATCH] ALSA: usb-audio: Add support for Mythware XA001AU capture and playback interfaces.
by dengxiang 04 Aug '23

04 Aug '23

This patch adds a USB quirk for Mythware XA001AU USB interface. Signed-off-by: dengxiang <dengxiang(a)nfschina.com> --- sound/usb/quirks-table.h | 29 +++++++++++++++++++++++++++++ 1 file changed, 29 insertions(+) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index efb4a3311cc5..5d72dc8441cb 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -4507,6 +4507,35 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, +{ + /* Advanced modes of the Mythware XA001AU. + * For the standard mode, Mythware XA001AU has ID ffad:a001 + */ + USB_DEVICE_VENDOR_SPEC(0xffad, 0xa001), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .vendor_name = "Mythware", + .product_name = "XA001AU", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_IGNORE_INTERFACE, + }, + { + .ifnum = 1, + .type = QUIRK_AUDIO_STANDARD_INTERFACE, + }, + { + .ifnum = 2, + .type = QUIRK_AUDIO_STANDARD_INTERFACE, + }, + { + .ifnum = -1 + } + } + } +}, #undef USB_DEVICE_VENDOR_SPEC #undef USB_AUDIO_DEVICE -- 2.30.2

2 2

[GIT PULL] ASoC fixes for v6.5-rc4
by Mark Brown 04 Aug '23

04 Aug '23

The following changes since commit 5d0c230f1de8c7515b6567d9afba1f196fb4e2f4: Linux 6.5-rc4 (2023-07-30 13:23:47 -0700) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.5-rc4 for you to fetch changes up to f803ec63686dec863a33cad87218d7d99c4b5e92: ASoC: fsl: micfil: Use dual license micfil code (2023-08-03 14:54:06 +0100) ---------------------------------------------------------------- ASoC: Fix for v6.5 Not really a fix, but rather a licensing update for the fsl_micfil driver. ---------------------------------------------------------------- Daniel Baluta (1): ASoC: fsl: micfil: Use dual license micfil code sound/soc/fsl/fsl_micfil.c | 4 ++-- sound/soc/fsl/fsl_micfil.h | 2 +- 2 files changed, 3 insertions(+), 3 deletions(-)

2 1

[PATCH 0/5] ALSA/PCI: hda: add ARL-S support, config for MTL/LNL
by Pierre-Louis Bossart 04 Aug '23

04 Aug '23

Add a new PCI ID and i915 connectivity check for ArrowLake-S, and configuration updates for MTL/LNL. Kai Vehmanen (2): ALSA: hda: add HD Audio PCI ID for Intel Arrow Lake-S ALSA: hda/i915: extend connectivity check to cover Intel ARL Pierre-Louis Bossart (3): PCI: add ArrowLake-S PCI ID for Intel HDAudio subsystem. ALSA: hda: intel-dsp-cfg: use common include for MeteorLake ALSA: hda: intel-dsp-cfg: add LunarLake support include/linux/pci_ids.h | 1 + sound/hda/hdac_i915.c | 14 ++++++++++---- sound/hda/intel-dsp-config.c | 10 +++++++++- sound/pci/hda/hda_intel.c | 2 ++ 4 files changed, 22 insertions(+), 5 deletions(-) -- 2.39.2

4 12

[PATCH] ALSA: pcmtest: Move buffer iterator initialization to prepare callback
by Ivan Orlov 04 Aug '23

04 Aug '23

Trigger callback is not the best place for buffer iterator initialization, so move it out to the prepare callback, where it have to be. Minor enhancement: remove redundant definitions and blank line. Signed-off-by: Ivan Orlov <ivan.orlov0322(a)gmail.com> --- sound/drivers/pcmtest.c | 41 +++++++++++++++++++---------------------- 1 file changed, 19 insertions(+), 22 deletions(-) diff --git a/sound/drivers/pcmtest.c b/sound/drivers/pcmtest.c index 08e14b5eb772..7f170429eb8f 100644 --- a/sound/drivers/pcmtest.c +++ b/sound/drivers/pcmtest.c @@ -41,8 +41,6 @@ #include <linux/debugfs.h> #include <linux/delay.h> -#define DEVNAME "pcmtestd" -#define CARD_NAME "pcm-test-card" #define TIMER_PER_SEC 5 #define TIMER_INTERVAL (HZ / TIMER_PER_SEC) #define DELAY_JIFFIES HZ @@ -74,11 +72,11 @@ static u8 ioctl_reset_test; static struct dentry *driver_debug_dir; module_param(index, int, 0444); -MODULE_PARM_DESC(index, "Index value for " CARD_NAME " soundcard"); +MODULE_PARM_DESC(index, "Index value for pcmtest soundcard"); module_param(id, charp, 0444); -MODULE_PARM_DESC(id, "ID string for " CARD_NAME " soundcard"); +MODULE_PARM_DESC(id, "ID string for pcmtest soundcard"); module_param(enable, bool, 0444); -MODULE_PARM_DESC(enable, "Enable " CARD_NAME " soundcard."); +MODULE_PARM_DESC(enable, "Enable pcmtest soundcard."); module_param(fill_mode, short, 0600); MODULE_PARM_DESC(fill_mode, "Buffer fill mode: rand(0) or pattern(1)"); module_param(inject_delay, int, 0600); @@ -92,7 +90,6 @@ MODULE_PARM_DESC(inject_trigger_err, "Inject EINVAL error in the 'trigger' callb module_param(inject_open_err, bool, 0600); MODULE_PARM_DESC(inject_open_err, "Inject EBUSY error in the 'open' callback"); - struct pcmtst { struct snd_pcm *pcm; struct snd_card *card; @@ -405,25 +402,9 @@ static int snd_pcmtst_pcm_close(struct snd_pcm_substream *substream) static int snd_pcmtst_pcm_trigger(struct snd_pcm_substream *substream, int cmd) { - struct snd_pcm_runtime *runtime = substream->runtime; - struct pcmtst_buf_iter *v_iter = runtime->private_data; - if (inject_trigger_err) return -EINVAL; - v_iter->sample_bytes = runtime->sample_bits / 8; - v_iter->period_bytes = frames_to_bytes(runtime, runtime->period_size); - if (runtime->access == SNDRV_PCM_ACCESS_RW_NONINTERLEAVED || - runtime->access == SNDRV_PCM_ACCESS_MMAP_NONINTERLEAVED) { - v_iter->chan_block = runtime->dma_bytes / runtime->channels; - v_iter->interleaved = false; - } else { - v_iter->interleaved = true; - } - // We want to record RATE * ch_cnt samples per sec, it is rate * sample_bytes * ch_cnt bytes - v_iter->s_rw_ch = runtime->rate / TIMER_PER_SEC; - v_iter->b_rw = v_iter->s_rw_ch * v_iter->sample_bytes * runtime->channels; - return 0; } @@ -454,8 +435,24 @@ static void pcmtst_pdev_release(struct device *dev) static int snd_pcmtst_pcm_prepare(struct snd_pcm_substream *substream) { + struct snd_pcm_runtime *runtime = substream->runtime; + struct pcmtst_buf_iter *v_iter = runtime->private_data; + if (inject_prepare_err) return -EINVAL; + + v_iter->sample_bytes = samples_to_bytes(runtime, 1); + v_iter->period_bytes = snd_pcm_lib_period_bytes(substream); + v_iter->interleaved = true; + if (runtime->access == SNDRV_PCM_ACCESS_RW_NONINTERLEAVED || + runtime->access == SNDRV_PCM_ACCESS_MMAP_NONINTERLEAVED) { + v_iter->chan_block = snd_pcm_lib_buffer_bytes(substream) / runtime->channels; + v_iter->interleaved = false; + } + // We want to record RATE * ch_cnt samples per sec, it is rate * sample_bytes * ch_cnt bytes + v_iter->s_rw_ch = runtime->rate / TIMER_PER_SEC; + v_iter->b_rw = v_iter->s_rw_ch * v_iter->sample_bytes * runtime->channels; + return 0; } -- 2.34.1

2 2

[PATCH v1 1/2] ASoC: tas2783: Add source files for tas2783 soundwire driver
by Baojun.Xu 04 Aug '23

04 Aug '23

From: Baojun Xu <baojun.xu(a)ti.com> Add source file and header file for tas2783 driver. Update Kconfig and Makefile for tas2783 driver. Signed-off-by: Baojun Xu <baojun.xu(a)ti.com> --- Change in v1: - Add tas2783-sdw.c - Add tas2783.h - Update Makefile for tas2783 driver. - Update Kconfig for tas2783 driver. --- sound/soc/codecs/Kconfig | 12 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tas2783-sdw.c | 1166 ++++++++++++++++++++++++++++++++ sound/soc/codecs/tas2783.h | 178 +++++ 4 files changed, 1358 insertions(+) create mode 100644 sound/soc/codecs/tas2783-sdw.c create mode 100644 sound/soc/codecs/tas2783.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index c2de4ee72183..318ad867030c 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -240,6 +240,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_TAS2781_COMLIB imply SND_SOC_TAS2781_FMWLIB imply SND_SOC_TAS2781_I2C + imply SND_SOC_TAS2783 imply SND_SOC_TAS5086 imply SND_SOC_TAS571X imply SND_SOC_TAS5720 @@ -1762,6 +1763,17 @@ config SND_SOC_TAS5086 tristate "Texas Instruments TAS5086 speaker amplifier" depends on I2C +config SND_SOC_TAS2783 + tristate "Texas Instruments TAS2781 speaker amplifier (sdw)" + depends on SOUNDWIRE + select REGMAP + select CRC8 + help + Enable support for Texas Instruments TAS2783 Smart Amplifier + Digital input mono Class-D and DSP-inside audio power amplifiers. + Note the TAS2783 driver implements a flexible and configurable + algo coff setting, for one, two, even multiple TAS2783 chips. + config SND_SOC_TAS571X tristate "Texas Instruments TAS571x power amplifiers" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index b48a9a323b84..3298dfa3b9d5 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -276,6 +276,7 @@ snd-soc-tas2770-objs := tas2770.o snd-soc-tas2781-comlib-objs := tas2781-comlib.o snd-soc-tas2781-fmwlib-objs := tas2781-fmwlib.o snd-soc-tas2781-i2c-objs := tas2781-i2c.o +snd-soc-tas2783-objs := tas2783-sdw.o snd-soc-tfa9879-objs := tfa9879.o snd-soc-tfa989x-objs := tfa989x.o snd-soc-tlv320adc3xxx-objs := tlv320adc3xxx.o @@ -648,6 +649,7 @@ obj-$(CONFIG_SND_SOC_TAS2780) += snd-soc-tas2780.o obj-$(CONFIG_SND_SOC_TAS2781_COMLIB) += snd-soc-tas2781-comlib.o obj-$(CONFIG_SND_SOC_TAS2781_FMWLIB) += snd-soc-tas2781-fmwlib.o obj-$(CONFIG_SND_SOC_TAS2781_I2C) += snd-soc-tas2781-i2c.o +obj-$(CONFIG_SND_SOC_TAS2783) += snd-soc-tas2783.o obj-$(CONFIG_SND_SOC_TAS5086) += snd-soc-tas5086.o obj-$(CONFIG_SND_SOC_TAS571X) += snd-soc-tas571x.o obj-$(CONFIG_SND_SOC_TAS5720) += snd-soc-tas5720.o diff --git a/sound/soc/codecs/tas2783-sdw.c b/sound/soc/codecs/tas2783-sdw.c new file mode 100644 index 000000000000..93b1e7a1f311 --- /dev/null +++ b/sound/soc/codecs/tas2783-sdw.c @@ -0,0 +1,1165 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// ALSA SoC Texas Instruments TAS2783 Audio Smart Amplifier +// +// Copyright (C) 2023 Texas Instruments Incorporated +// https://www.ti.com +// +// The TAS2783 driver implements a flexible and configurable +// algo coefficient setting for single TAS2783 chips. +// +// Author: Baojun Xu <baojun.xu(a)ti.com> +// Author: Kevin Lu <kevin-lu(a)ti.com> +// + +#include <linux/delay.h> +#include <linux/device.h> +#include <linux/pm_runtime.h> +#include <linux/err.h> +#include <linux/module.h> +#include <linux/regmap.h> +#include <linux/soundwire/sdw.h> +#include <linux/soundwire/sdw_registers.h> +#include <linux/soundwire/sdw_type.h> +#include <linux/init.h> +#include <linux/pm.h> +#include <linux/of.h> +#include <linux/slab.h> +#include <linux/crc8.h> +#include <linux/crc32.h> +#include <linux/efi.h> +#include <linux/firmware.h> +#include <linux/of_gpio.h> +#include <linux/interrupt.h> +#include <sound/soc.h> +#include <sound/sdw.h> +#include <sound/pcm_params.h> +#include <sound/tlv.h> +#include <sound/tas2781-tlv.h> + +#include "tas2783.h" + +static const unsigned int tas2783_calibration_reg[] = { + TAS2783_CALIBRATION_RE, + TAS2783_CALIBRATION_RE_LOW, + TAS2783_CALIBRATION_INV_RE, + TAS2783_CALIBRATION_POW, + TAS2783_CALIBRATION_TLIMIT, + 0, +}; + +static const struct reg_default tas2783_reg_defaults[] = { + // Default values for ROM mode. Activated. + { 0x8002, 0x1a}, + { 0x800e, 0x44}, + { 0x800f, 0x40}, + { 0x8097, 0xc8}, + { 0x80b5, 0x74}, + { 0x8099, 0x20}, + { 0xfe8d, 0x0d}, + { 0xfebe, 0x4a}, + { 0x8230, 0x00}, + { 0x8231, 0x00}, + { 0x8232, 0x00}, + { 0x8233, 0x01}, + { 0x8418, 0x00}, + { 0x8419, 0x00}, + { 0x841a, 0x00}, + { 0x841b, 0x00}, + { 0x8428, 0x40}, + { 0x8429, 0x00}, + { 0x842a, 0x00}, + { 0x842b, 0x00}, + { 0x8548, 0x00}, + { 0x8549, 0x00}, + { 0x854a, 0x00}, + { 0x854b, 0x00}, + { 0x8558, 0x40}, + { 0x8559, 0x00}, + { 0x855a, 0x00}, + { 0x855b, 0x00}, + { 0x800a, 0x3a}, + { 0x800e, 0x44}, + { 0x800f, 0x40}, + { 0x805c, 0xd9}, + { 0x8002, 0x00}, +}; + +static bool tas2783_readable_register(struct device *dev, + unsigned int reg) +{ + switch (reg) { + case 0x8000 ... 0xc000: // Page 0 ~ 127. + case 0xfe80 ... 0xfeff: // Page 253. + case SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, TAS2783_SDCA_ENT_UDMPU21, + TAS2783_SDCA_CTL_UDMPU_CLUSTER, 0): + case SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, TAS2783_SDCA_ENT_FU21, + TAS2783_SDCA_CTL_FU_MUTE, CH_L): + case SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, TAS2783_SDCA_ENT_FU21, + TAS2783_SDCA_CTL_FU_MUTE, CH_R): + case SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, TAS2783_SDCA_ENT_PDE23, + TAS2783_SDCA_CTL_REQ_POWER_STATE, 0): + case SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, TAS2783_SDCA_ENT_PDE22, + TAS2783_SDCA_CTL_REQ_POWER_STATE, 0): + return true; + default: + return false; + } +} + +static bool tas2783_volatile_register(struct device *dev, + unsigned int reg) +{ + switch (reg) { + case 0x8001: + // Only reset register was volatiled. + return true; + default: + return false; + } +} + +static const struct regmap_config tasdevice_regmap = { + .reg_bits = 32, + .val_bits = 8, + .readable_reg = tas2783_readable_register, + .volatile_reg = tas2783_volatile_register, + .max_register = 0x41008000 + TASDEVICE_REG(0xa1, 0x60, 0x7f), + .reg_defaults = tas2783_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(tas2783_reg_defaults), + .cache_type = REGCACHE_RBTREE, + .use_single_read = true, + .use_single_write = true, +}; + +static int tas2783_sdw_read(void *context, const void *reg_buf, + const size_t reg_size, void *val_buf, size_t val_size) +{ + struct sdw_slave *peripheral = context; + u8 *buf8 = val_buf; + unsigned int reg, bytes; + int ret; + + reg = le32_to_cpu(*(const __le32 *)reg_buf); + reg += TAS2783_SDW_ADDR_OFFSET; + + while (val_size) { + /* to end of page */ + bytes = SDW_REG_NO_PAGE - (reg & SDW_REGADDR); + if (bytes > val_size) + bytes = val_size; + + ret = sdw_nread_no_pm(peripheral, reg, bytes, buf8); + if (ret != 0) { + dev_err(&peripheral->dev, "Read @%#x..%#x:%d\n", + reg, reg + bytes - 1, ret); + return ret; + } + + swab32_array((u32 *)buf8, bytes / 4); + val_size -= bytes; + reg += bytes; + buf8 += bytes; + } + + return 0; +} + +/* Convert for LSB or MSB. */ +static inline void tas2783_swab_copy( + void *dest, const void *src, size_t nbytes) +{ + u32 *dest32 = dest; + const u32 *src32 = src; + + for (; nbytes > 0; nbytes -= 4) + *dest32++ = swab32(*src32++); +} + +static int tas2783_sdw_gather_write(void *context, const void *reg_buf, + size_t reg_size, const void *val_buf, size_t val_size) +{ + struct sdw_slave *peripheral = context; + const u8 *src_be = val_buf; + u32 val_le_buf[64]; /* Define u32 so it is 32-bit aligned */ + unsigned int reg, bytes; + int ret; + + reg = le32_to_cpu(*(const __le32 *)reg_buf); + + while (val_size) { + /* to end of page */ + bytes = SDW_REG_NO_PAGE - (reg & SDW_REGADDR); + if (bytes > val_size) + bytes = val_size; + if (bytes > sizeof(val_le_buf)) + bytes = sizeof(val_le_buf); + + tas2783_swab_copy(val_le_buf, src_be, bytes); + + ret = sdw_nwrite_no_pm(peripheral, reg, bytes, + (u8 *)val_le_buf); + if (ret != 0) { + dev_err(&peripheral->dev, "Write @%#x..%#x:%d\n", + reg, reg + bytes - 1, ret); + return ret; + } + + val_size -= bytes; + reg += bytes; + src_be += bytes; + } + + return 0; +} + +static int tas2783_sdw_write( + void *context, const void *val_buf, size_t val_size) +{ + const u8 *src_buf = val_buf; + + /* First word of val_buf contains the destination address */ + return tas2783_sdw_gather_write(context, &src_buf[0], 4, + &src_buf[4], val_size - 4); +} + +static int tasdevice_clamp(int val, int max, unsigned int invert) +{ + if (val > max) + val = max; + if (invert) + val = max - val; + if (val < 0) + val = 0; + return val; +} + +/* + * Registers are big-endian on I2C and SPI but little-endian on SoundWire. + * Exported firmware controls are big-endian on I2C/SPI but little-endian + * on SoundWire. + * Firmware files are always big-endian and are opaque blobs. + * Present a big-endian regmap and hide the endianness swap, + * so that the ALSA byte controls always have the same byte order, + * and firmware file blobs can be written verbatim. + */ +static const struct regmap_bus tas2783_regmap_bus_sdw = { + .read = tas2783_sdw_read, + .write = tas2783_sdw_write, + .gather_write = tas2783_sdw_gather_write, + .reg_format_endian_default = REGMAP_ENDIAN_LITTLE, + .val_format_endian_default = REGMAP_ENDIAN_BIG, +}; + +static int tas2783_digital_getvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec + = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_dev = + snd_soc_component_get_drvdata(codec); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + int val; + unsigned int invert = mc->invert; + int max = mc->max; + int ret = 0; + + /* Read the primary device as the whole */ + ret = tasdevice_dev_read(tas_dev, &mc->reg, (unsigned int *)&val); + dev_dbg(tas_dev->dev, "%s, get digital vol %d from %x with %d\n", + __func__, val, mc->reg, ret); + if (ret) { + dev_err(tas_dev->dev, "%s, get digital vol error %x.\n", + __func__, ret); + goto out; + } + + if (val > max) + val = max; + if (invert) + val = max - val; + if (val < 0) + val = 0; + ucontrol->value.integer.value[0] = val; + +out: + return ret; +} + +static int tas2783_digital_putvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec + = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_dev = + snd_soc_component_get_drvdata(codec); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + int val, ret = 0; + unsigned int invert = mc->invert; + int max = mc->max; + unsigned char value = 0; + + val = ucontrol->value.integer.value[0]; + if (val > max) + val = max; + if (invert) + val = max - val; + if (val < 0) + val = 0; + + value = (u8)val; + ret = sdw_nwrite_no_pm(tas_dev->sdw_slave, mc->reg, 1, &value); + if (ret != 0) { + dev_err(tas_dev->dev, "Write @%#x..%#x:%d\n", + mc->reg, val, ret); + return ret; + } + dev_dbg(tas_dev->dev, "%s, Put digital vol %d into %x.\n", + __func__, val, mc->reg); + + return 1; +} + +static int tas2783_amp_getvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec + = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_dev = + snd_soc_component_get_drvdata(codec); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + int val; + unsigned char mask = 0; + int max = mc->max; + int ret = 0; + unsigned int invert = mc->invert; + unsigned char value = 0; + + /* Read the primary device */ + ret = sdw_nread_no_pm(tas_dev->sdw_slave, mc->reg, 1, &value); + dev_dbg(tas_dev->dev, "%s, get AMP vol %d from %x with %d\n", + __func__, value, mc->reg, ret); + + mask = (1 << fls(mc->max)) - 1; + mask <<= mc->shift; + val = value; + val = (val & mask) >> mc->shift; + if (val > max) + val = max; + if (invert) + val = max - val; + if (val < 0) + val = 0; + ucontrol->value.integer.value[0] = val; + + return ret; +} + +static int tas2783_amp_putvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec + = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_dev = + snd_soc_component_get_drvdata(codec); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + int val = 0, ret = 0, max = mc->max; + unsigned char mask = 0, value = 0; + unsigned int invert = mc->invert; + + mask = (1 << fls(mc->max)) - 1; + mask <<= mc->shift; + val = tasdevice_clamp(ucontrol->value.integer.value[0], max, + invert); + + ret = sdw_nread_no_pm(tas_dev->sdw_slave, mc->reg, 1, &value); + if (ret != 0) { + dev_err(tas_dev->dev, "Read @%x.%x:%d\n", + mc->reg, value, ret); + return ret; + } + value &= (~TAS2783_AMP_LEVEL_MASK); + value |= (val << mc->shift); + ret = sdw_nwrite_no_pm(tas_dev->sdw_slave, mc->reg, 1, &value); + if (ret != 0) { + dev_err(tas_dev->dev, "Write @%#x..%#x:%d\n", + mc->reg, val, ret); + return ret; + } + dev_dbg(tas_dev->dev, "amp %x into 0x%08x by %lx shift %d.\n", + val, mc->reg, TAS2783_AMP_LEVEL_MASK, mc->shift); + + return ret; +} + +static const struct snd_kcontrol_new tas2783_snd_controls[] = { + SOC_SINGLE_RANGE_EXT_TLV("Amp Gain Volume", TAS2783_AMP_LEVEL, + 1, 0, 20, 0, tas2783_amp_getvol, + tas2783_amp_putvol, amp_vol_tlv), + SOC_SINGLE_RANGE_EXT_TLV("Digital Volume", TAS2783_DVC_LVL, + 0, 0, 200, 1, tas2783_digital_getvol, + tas2783_digital_putvol, dvc_tlv), +}; + +static void tas2783_apply_calib( + struct tasdevice_priv *tas_priv, unsigned int *cali_data) +{ + unsigned int reg_address = 0, i = 0; + unsigned int *reg_start; + + if (!tas_priv->sdw_slave) + return; + if ((tas_priv->sdw_slave->id.unique_id < TAS2783_DEVICE_ID_START) + || (tas_priv->sdw_slave->id.unique_id > TAS2783_DEVICE_ID_END)) + return; + reg_start = cali_data+(tas_priv->sdw_slave->id.unique_id - + TAS2783_DEVICE_ID_START)*TAS2783_CALIBRATION_SIZE; + while (tas2783_calibration_reg[i] != 0) { + reg_address = tas2783_calibration_reg[i]; + tas2783_sdw_gather_write(tas_priv->sdw_slave, + &reg_address, 4, reg_start+i, 4); + } +} + +static int tas2783_calibration(struct tasdevice_priv *tas_priv) +{ + efi_guid_t efi_guid = EFI_GUID(0x1f52d2a1, 0xbb3a, 0x457d, 0xbc, + 0x09, 0x43, 0xa3, 0xf4, 0x31, 0x0a, 0x92); + static efi_char16_t efi_name[] = L"CALI_DATA"; + struct tm *tm = &tas_priv->tm; + unsigned int attr, crc; + unsigned int *tmp_val; + efi_status_t status; + + tas_priv->cali_data.total_sz = 128; + /* Get real size of UEFI variable */ + status = efi.get_variable(efi_name, &efi_guid, &attr, + &tas_priv->cali_data.total_sz, tas_priv->cali_data.data); + dev_dbg(tas_priv->dev, "cali get %lx bytes with result : %ld\n", + tas_priv->cali_data.total_sz, status); + if (status == EFI_BUFFER_TOO_SMALL) { + status = efi.get_variable(efi_name, &efi_guid, &attr, + &tas_priv->cali_data.total_sz, + tas_priv->cali_data.data); + dev_dbg(tas_priv->dev, "cali get %lx bytes result:%ld\n", + tas_priv->cali_data.total_sz, status); + } + for (int i = 0; i < tas_priv->cali_data.total_sz; i += 4) { + dev_dbg(tas_priv->dev, "cali get %02x %02x %02x %02x", + tas_priv->cali_data.data[i], + tas_priv->cali_data.data[i+1], + tas_priv->cali_data.data[i+2], + tas_priv->cali_data.data[i+3]); + } + + tmp_val = (unsigned int *)tas_priv->cali_data.data; + + crc = crc32(~0, tas_priv->cali_data.data, 84) ^ ~0; + dev_dbg(tas_priv->dev, "cali crc 0x%08x PK tmp_val 0x%08x\n", + crc, tmp_val[21]); + + if (crc == tmp_val[21]) { + time64_to_tm(tmp_val[20], 0, tm); + dev_dbg(tas_priv->dev, "%4ld-%2d-%2d, %2d:%2d:%2d\n", + tm->tm_year, tm->tm_mon, tm->tm_mday, + tm->tm_hour, tm->tm_min, tm->tm_sec); + tas2783_apply_calib(tas_priv, tmp_val); + } else { + dev_dbg(tas_priv->dev, "CRC error!\n"); + tas_priv->cali_data.total_sz = 0; + } + + return 0; +} + +static void tasdevice_rca_ready(const struct firmware *fmw, void *context) +{ + struct tasdevice_priv *tas_dev = + (struct tasdevice_priv *) context; + struct MYSWFTFile *p; + unsigned char *buf = NULL, value_sdw; + int offset = 0, i = 0, num_files = 0, img_sz = 0; + int ret = 0; + + mutex_lock(&tas_dev->codec_lock); + + if (!fmw || !fmw->data) { + dev_err(tas_dev->dev, + "Failed to read %s, no side - effect on driver running\n", + tas_dev->rca_binaryname); + ret = -1; + goto out; + } + buf = (unsigned char *)fmw->data; + + img_sz = SMS_HTONL(buf[offset+3], buf[offset + 2], + buf[offset + 1], buf[offset]); + dev_dbg(tas_dev->dev, "Got %x:%lx.\n", img_sz, fmw->size); + offset += sizeof(img_sz); + if (img_sz != fmw->size) { + dev_err(tas_dev->dev, + "Size not match, %d %u", (int)fmw->size, img_sz); + ret = -1; + goto out; + } + + while ((offset < img_sz) && (num_files < MAX_NO_FILES)) { + p = (struct MYSWFTFile *)(buf + offset); + tas_dev->m_swftFile[num_files].m_vendor_id = + p->m_vendor_id; + tas_dev->m_swftFile[num_files].m_file_id = p->m_file_id; + tas_dev->m_swftFile[num_files].m_version = p->m_version; + tas_dev->m_swftFile[num_files].m_Length = p->m_Length; + tas_dev->m_swftFile[num_files].m_downloadAddr = + p->m_downloadAddr; + tas_dev->m_swftFile[num_files].m_startAdress = + ((char *)p) + sizeof(unsigned int)*5; + + ret = sdw_nwrite_no_pm(tas_dev->sdw_peripheral, + tas_dev->m_swftFile[num_files].m_downloadAddr, + tas_dev->m_swftFile[num_files].m_Length, + tas_dev->m_swftFile[num_files].m_startAdress); + dev_dbg(tas_dev->dev, "Write %x into %x result : %d\n", + tas_dev->m_swftFile[num_files].m_Length, + tas_dev->m_swftFile[num_files].m_downloadAddr, + ret); + + dev_dbg(tas_dev->dev, "Write %d :%x:%x:%x:%x:%x:%x.\n", + num_files, + tas_dev->m_swftFile[num_files].m_vendor_id, + tas_dev->m_swftFile[num_files].m_file_id, + tas_dev->m_swftFile[num_files].m_version, + tas_dev->m_swftFile[num_files].m_Length, + tas_dev->m_swftFile[num_files].m_downloadAddr, + tas_dev->m_swftFile[num_files].m_startAdress[0]); + + ret = sdw_nread_no_pm(tas_dev->sdw_peripheral, + tas_dev->m_swftFile[num_files].m_downloadAddr, + 1, &value_sdw); + dev_dbg(tas_dev->dev, "Got block %d :%x.\n", + num_files, value_sdw); + + offset += sizeof(unsigned int)*5 + p->m_Length; + num_files++; + } + + i = 0x50; + while (i < 0x56) { + ret = sdw_nread_no_pm(tas_dev->sdw_peripheral, i, 1, + &value_sdw); + dev_dbg(tas_dev->dev, "Got device id:%x\n", value_sdw); + i++; + } + tas2783_calibration(tas_dev); + +out: + mutex_unlock(&tas_dev->codec_lock); + if (fmw) + release_firmware(fmw); +} + +static void tasdevice_config_info_remove(void *context) +{ + struct tasdevice_priv *tas_dev = + (struct tasdevice_priv *) context; + dev_dbg(tas_dev->dev, "%s was performed\n", __func__); +} + +static int tasdevice_dapm_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *codec = + snd_soc_dapm_to_component(w->dapm); + struct tasdevice_priv *tas_dev = + snd_soc_component_get_drvdata(codec); + int state = 1; + + /* Codec Lock Hold */ + mutex_lock(&tas_dev->codec_lock); + dev_dbg(tas_dev->dev, "dapm event : %x\n", event); + if (event == SND_SOC_DAPM_PRE_PMD) + state = 0; + /* Codec Lock Release*/ + mutex_unlock(&tas_dev->codec_lock); + + return 0; +} + +static const struct snd_soc_dapm_widget tasdevice_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN("ASI", "ASI Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT_E("ASI OUT", "ASI Capture", 0, SND_SOC_NOPM, + 0, 0, tasdevice_dapm_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_SPK("SPK", tasdevice_dapm_event), + SND_SOC_DAPM_OUTPUT("OUT"), + SND_SOC_DAPM_INPUT("DMIC") +}; + +static const struct snd_soc_dapm_route tasdevice_audio_map[] = { + {"SPK", NULL, "ASI"}, + {"OUT", NULL, "SPK"}, + {"ASI OUT", NULL, "DMIC"} +}; + +static int tasdevice_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + int ret = 0; + + return ret; +} + +static int tasdevice_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct tasdevice_priv *tas_dev = + snd_soc_dai_get_drvdata(codec_dai); + + tas_dev->sysclk = freq; + + return 0; +} + +static int tasdevice_set_sdw_stream( + struct snd_soc_dai *dai, void *sdw_stream, int direction) +{ + struct sdw_stream_data *stream; + + if (!sdw_stream) + return 0; + + stream = kzalloc(sizeof(*stream), GFP_KERNEL); + if (!stream) + return -ENOMEM; + + stream->sdw_stream = sdw_stream; + + /* Use tx_mask or rx_mask to set dma_data */ + snd_soc_dai_dma_data_set(dai, direction, stream); + + return 0; +} + +static void tasdevice_sdw_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct sdw_stream_data *stream; + + stream = snd_soc_dai_get_dma_data(dai, substream); + snd_soc_dai_set_dma_data(dai, substream, NULL); + kfree(stream); +} + +static int tasdevice_sdw_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct tasdevice_priv *tasdevice = + snd_soc_component_get_drvdata(component); + struct sdw_stream_config stream_config = {0}; + struct sdw_port_config port_config = {0}; + struct sdw_stream_data *stream; + int retval; + + dev_dbg(dai->dev, "%s %s", __func__, dai->name); + stream = snd_soc_dai_get_dma_data(dai, substream); + + if (!stream) + return -EINVAL; + + if (!tasdevice->sdw_slave) + return -EINVAL; + + /* SoundWire specific configuration */ + snd_sdw_params_to_config(substream, params, + &stream_config, &port_config); + + /* port 1 for playback */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + port_config.num = 1; + else + port_config.num = 2; + + retval = sdw_stream_add_slave(tasdevice->sdw_slave, + &stream_config, &port_config, 1, stream->sdw_stream); + if (retval) { + dev_err(dai->dev, "Unable to configure port\n"); + return retval; + } + + return 0; +} + +static int tasdevice_sdw_pcm_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct tasdevice_priv *tasdevice = + snd_soc_component_get_drvdata(component); + struct sdw_stream_data *stream = + snd_soc_dai_get_dma_data(dai, substream); + + if (!tasdevice->sdw_slave) + return -EINVAL; + + sdw_stream_remove_slave(tasdevice->sdw_slave, stream->sdw_stream); + return 0; +} + +static int tasdevice_mute(struct snd_soc_dai *dai, int mute, + int direction) +{ + struct snd_soc_component *component = dai->component; + struct tasdevice_priv *tasdevice = + snd_soc_component_get_drvdata(component); + int ret = 0; + unsigned char value_sdw = 0; + + dev_dbg(tasdevice->dev, "%s: mute %d, %d\n", + __func__, mute, direction); + if (mute == 0) {// Unmute. + // FU21 Unmute + value_sdw = 0; + ret = sdw_nwrite_no_pm(tasdevice->sdw_peripheral, + SDW_SDCA_CTL(1, 1, 1, 0), 1, &value_sdw); + dev_dbg(tasdevice->dev, "Unmuted %lx %x\n", + SDW_SDCA_CTL(1, 1, 1, 0), ret); + value_sdw = 0; + ret = sdw_nwrite_no_pm(tasdevice->sdw_peripheral, + 0x40400090, 1, &value_sdw); + dev_dbg(tasdevice->dev, "Unmuted %x\n", ret); + ret = sdw_nwrite_no_pm(tasdevice->sdw_peripheral, + 0x40402090, 1, &value_sdw); + dev_dbg(tasdevice->dev, "Unmuted %x\n", ret); + + // FU23 Unmute + value_sdw = 0; + ret = sdw_nwrite_no_pm(tasdevice->sdw_peripheral, + SDW_SDCA_CTL(1, 2, 1, 0), 1, &value_sdw); + dev_dbg(tasdevice->dev, "Unmuted %lx %x\n", + SDW_SDCA_CTL(1, 2, 1, 0), ret); + value_sdw = 0; + ret = sdw_nwrite_no_pm(tasdevice->sdw_peripheral, + 0x40400108, 1, &value_sdw); + dev_dbg(tasdevice->dev, "Unmuted %x\n", ret); + value_sdw = 0; + ret = sdw_nwrite_no_pm(tasdevice->sdw_peripheral, + 0x8002, 1, &value_sdw); + dev_dbg(tasdevice->dev, "Unmuted %x\n", ret); + } else { + // FU21 mute + value_sdw = 1; + ret = sdw_nwrite_no_pm(tasdevice->sdw_peripheral, + SDW_SDCA_CTL(1, 1, 1, 0), 1, &value_sdw); + dev_dbg(tasdevice->dev, "muted %lx %x\n", + SDW_SDCA_CTL(1, 1, 1, 0), ret); + value_sdw = 1; + ret = sdw_nwrite_no_pm(tasdevice->sdw_peripheral, + 0x40400090, 1, &value_sdw); + dev_dbg(tasdevice->dev, "muted %x\n", ret); + ret = sdw_nwrite_no_pm(tasdevice->sdw_peripheral, + 0x40402090, 1, &value_sdw); + dev_dbg(tasdevice->dev, "muted %x\n", ret); + // FU23 mute + value_sdw = 1; + ret = sdw_nwrite_no_pm(tasdevice->sdw_peripheral, + SDW_SDCA_CTL(1, 2, 1, 0), 1, &value_sdw); + dev_dbg(tasdevice->dev, "muted %lx %x\n", + SDW_SDCA_CTL(1, 2, 1, 0), ret); + ret = sdw_nwrite_no_pm(tasdevice->sdw_peripheral, + 0x40400108, 1, &value_sdw); + dev_dbg(tasdevice->dev, "muted %x\n", ret); + value_sdw = 0x1a; + ret = sdw_nwrite_no_pm(tasdevice->sdw_peripheral, + 0x8002, 1, &value_sdw); + dev_dbg(tasdevice->dev, "muted %x\n", ret); + } + + if (!tasdevice->sdw_slave) + return -EINVAL; + + return 0; +} + +static const struct snd_soc_dai_ops tasdevice_dai_ops = { + .startup = tasdevice_startup, + .mute_stream = tasdevice_mute, + .hw_params = tasdevice_sdw_hw_params, + .set_sysclk = tasdevice_set_dai_sysclk, + .hw_free = tasdevice_sdw_pcm_hw_free, + .set_stream = tasdevice_set_sdw_stream, + .shutdown = tasdevice_sdw_shutdown, +}; + +static struct snd_soc_dai_driver tasdevice_dai_driver[] = { + { + .name = "tas2783_codec", + .id = 0, + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 4, + .rates = TASDEVICE_RATES, + .formats = TASDEVICE_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 1, + .rates = TASDEVICE_RATES, + .formats = TASDEVICE_FORMATS, + }, + .ops = &tasdevice_dai_ops, + .symmetric_rate = 1, + }, +}; + +static void tas2783_reset(struct tasdevice_priv *tas_dev) +{ + int ret = 0; + unsigned char value_sdw; + + value_sdw = TAS2873_REG_SWRESET_RESET; + ret = sdw_nwrite_no_pm(tas_dev->sdw_peripheral, + TAS2873_REG_SWRESET, 1, &value_sdw); + dev_dbg(tas_dev->dev, "%s TAS2783 was reseted.\n", __func__); + usleep_range(1000, 1050); +} + +static int tasdevice_codec_probe(struct snd_soc_component *codec) +{ + struct tasdevice_priv *tas_dev = + snd_soc_component_get_drvdata(codec); + int ret = 0; + + dev_dbg(tas_dev->dev, "%s called for TAS2783 start.\n", + __func__); + /* Codec Lock Hold */ + mutex_lock(&tas_dev->codec_lock); + + crc8_populate_msb(tas_dev->crc8_lkp_tbl, + TASDEVICE_CRC8_POLYNOMIAL); + tas_dev->codec = codec; + + scnprintf(tas_dev->rca_binaryname, 64, "MY_SWFT_x%01x.bin", + tas_dev->sdw_peripheral->id.unique_id); + + ret = request_firmware_nowait(THIS_MODULE, FW_ACTION_UEVENT, + tas_dev->rca_binaryname, tas_dev->dev, GFP_KERNEL, + tas_dev, tasdevice_rca_ready); + dev_dbg(tas_dev->dev, + "%s: request_firmware %x open status: 0x%08x\n", + __func__, tas_dev->sdw_peripheral->id.unique_id, ret); + + /* Codec Lock Release*/ + mutex_unlock(&tas_dev->codec_lock); + + tas2783_reset(tas_dev); + + dev_dbg(tas_dev->dev, "%s was called end.\n", __func__); + return ret; +} + +static void tasdevice_deinit(void *context) +{ + struct tasdevice_priv *tas_dev = + (struct tasdevice_priv *) context; + + tasdevice_config_info_remove(tas_dev); +} + +static void tasdevice_codec_remove( + struct snd_soc_component *codec) +{ + struct tasdevice_priv *tas_dev = + snd_soc_component_get_drvdata(codec); + /* Codec Lock Hold */ + mutex_lock(&tas_dev->codec_lock); + tasdevice_deinit(tas_dev); + /* Codec Lock Release*/ + mutex_unlock(&tas_dev->codec_lock); +} + +static const struct snd_soc_component_driver + soc_codec_driver_tasdevice = { + .probe = tasdevice_codec_probe, + .remove = tasdevice_codec_remove, + .controls = tas2783_snd_controls, + .num_controls = ARRAY_SIZE(tas2783_snd_controls), + .dapm_widgets = tasdevice_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tasdevice_dapm_widgets), + .dapm_routes = tasdevice_audio_map, + .num_dapm_routes = ARRAY_SIZE(tasdevice_audio_map), + .idle_bias_on = 1, + .endianness = 1, +}; + +int tasdevice_dev_read(struct tasdevice_priv *tas_dev, unsigned int *reg, + unsigned int *val) +{ + int ret = 0; + + mutex_lock(&tas_dev->dev_lock); + ret = tas2783_sdw_read(tas_dev->sdw_peripheral, + reg, 4, (void *)val, 4); + dev_dbg(tas_dev->dev, "%s, read %x from reg %x. Result is (%d)\n", + __func__, *val, *reg, ret); + mutex_unlock(&tas_dev->dev_lock); + return ret; +} + +static int tasdevice_init(struct tasdevice_priv *tas_dev) +{ + int ret = 0; + + tas_dev->cur_prog = -1; + tas_dev->cur_conf = -1; + + dev_dbg(tas_dev->dev, "%s was called for TAS2783.\n", __func__); + mutex_init(&tas_dev->dev_lock); + + dev_set_drvdata(tas_dev->dev, tas_dev); + + mutex_init(&tas_dev->codec_lock); + ret = devm_snd_soc_register_component(tas_dev->dev, + &soc_codec_driver_tasdevice, + tasdevice_dai_driver, ARRAY_SIZE(tasdevice_dai_driver)); + if (ret) { + dev_err(tas_dev->dev, "%s: codec register error:0x%08x\n", + __func__, ret); + } + dev_dbg(tas_dev->dev, "%s was called for TAS2783.\n", __func__); + + return ret; +} + +static int tasdevice_read_prop(struct sdw_slave *slave) +{ + struct sdw_slave_prop *prop = &slave->prop; + int nval; + int i, j; + u32 bit; + unsigned long addr; + struct sdw_dpn_prop *dpn; + + prop->scp_int1_mask = + SDW_SCP_INT1_BUS_CLASH | SDW_SCP_INT1_PARITY; + prop->quirks = SDW_SLAVE_QUIRKS_INVALID_INITIAL_PARITY; + + prop->paging_support = true; + + /* first we need to allocate memory for set bits in port lists */ + prop->source_ports = 0x04; /* BITMAP: 00000100 */ + prop->sink_ports = 0x2; /* BITMAP: 00000010 */ + + nval = hweight32(prop->source_ports); + prop->src_dpn_prop = devm_kcalloc(&slave->dev, nval, + sizeof(*prop->src_dpn_prop), GFP_KERNEL); + if (!prop->src_dpn_prop) + return -ENOMEM; + + i = 0; + dpn = prop->src_dpn_prop; + addr = prop->source_ports; + for_each_set_bit(bit, &addr, 32) { + dpn[i].num = bit; + dpn[i].type = SDW_DPN_FULL; + dpn[i].simple_ch_prep_sm = true; + dpn[i].ch_prep_timeout = 10; + i++; + } + + /* do this again for sink now */ + nval = hweight32(prop->sink_ports); + prop->sink_dpn_prop = devm_kcalloc(&slave->dev, nval, + sizeof(*prop->sink_dpn_prop), GFP_KERNEL); + if (!prop->sink_dpn_prop) + return -ENOMEM; + + j = 0; + dpn = prop->sink_dpn_prop; + addr = prop->sink_ports; + for_each_set_bit(bit, &addr, 32) { + dpn[j].num = bit; + dpn[j].type = SDW_DPN_FULL; + dpn[j].simple_ch_prep_sm = true; + dpn[j].ch_prep_timeout = 10; + j++; + } + + /* set the timeout values */ + prop->clk_stop_timeout = 20; + + dev_dbg(&slave->dev, "%s was performed.\n", __func__); + + return 0; +} + +static int tasdevice_io_init(struct device *dev, struct sdw_slave *slave) +{ + struct tasdevice_priv *tasdevice = dev_get_drvdata(dev); + + if (tasdevice->hw_init) + return 0; + + if (tasdevice->first_hw_init) { + regcache_cache_only(tasdevice->regmap, false); + regcache_cache_bypass(tasdevice->regmap, true); + } else { + /* PM runtime is only enabled when + * a Slave reports as Attached + * set autosuspend parameters + */ + pm_runtime_set_autosuspend_delay(&slave->dev, 3000); + pm_runtime_use_autosuspend(&slave->dev); + + /* update count of parent 'active' children */ + pm_runtime_set_active(&slave->dev); + + /* make sure the device does not suspend immediately */ + pm_runtime_mark_last_busy(&slave->dev); + + pm_runtime_enable(&slave->dev); + } + + pm_runtime_get_noresume(&slave->dev); + + if (tasdevice->first_hw_init) { + regcache_cache_bypass(tasdevice->regmap, false); + regcache_mark_dirty(tasdevice->regmap); + } else + tasdevice->first_hw_init = true; + + /* Mark Slave initialization complete */ + tasdevice->hw_init = true; + + pm_runtime_mark_last_busy(&slave->dev); + pm_runtime_put_autosuspend(&slave->dev); + + dev_dbg(&slave->dev, "%s hw_init complete\n", __func__); + return 0; +} + +static int tasdevice_update_status(struct sdw_slave *slave, + enum sdw_slave_status status) +{ + struct tasdevice_priv *tasdevice = dev_get_drvdata(&slave->dev); + + /* Update the status */ + tasdevice->status = status; + + if (status == SDW_SLAVE_UNATTACHED) + tasdevice->hw_init = false; + + /* Perform initialization only if slave status + * is present and hw_init flag is false + */ + if (tasdevice->hw_init || tasdevice->status != SDW_SLAVE_ATTACHED) + return 0; + + /* perform I/O transfers required for Slave initialization */ + return tasdevice_io_init(&slave->dev, slave); +} + +/* + * slave_ops: callbacks for get_clock_stop_mode, clock_stop and + * port_prep are not defined for now + */ +static const struct sdw_slave_ops tasdevice_sdw_ops = { + .read_prop = tasdevice_read_prop, + .update_status = tasdevice_update_status, +}; + +static void tasdevice_remove(struct tasdevice_priv *tas_dev) +{ + snd_soc_unregister_component(tas_dev->dev); + if (gpio_is_valid(tas_dev->rst_gpio)) + gpio_free(tas_dev->rst_gpio); + + if (gpio_is_valid(tas_dev->irq_info.irq_gpio)) + gpio_free(tas_dev->irq_info.irq_gpio); + + mutex_destroy(&tas_dev->dev_lock); + mutex_destroy(&tas_dev->codec_lock); +} + +static int tasdevice_sdw_probe(struct sdw_slave *peripheral, + const struct sdw_device_id *id) +{ + struct device *dev = &peripheral->dev; + struct tasdevice_priv *tas_dev; + int ret = 0; + + dev_dbg(dev, "%s was called for TAS2783.\n", __func__); + + tas_dev = devm_kzalloc(dev, sizeof(*tas_dev), GFP_KERNEL); + if (!tas_dev) { + ret = -ENOMEM; + goto out; + } + tas_dev->dev = &peripheral->dev; + tas_dev->sdw_peripheral = peripheral; + tas_dev->chip_id = id->driver_data; + tas_dev->sdw_slave = peripheral; + tas_dev->hw_init = false; + tas_dev->first_hw_init = false; + + dev_dbg(dev, "%d chip id %x for TAS2783.\n", + peripheral->id.unique_id, tas_dev->chip_id); + + dev_set_drvdata(dev, tas_dev); + + tas_dev->regmap = devm_regmap_init(dev, &tas2783_regmap_bus_sdw, + peripheral, &tasdevice_regmap); + if (IS_ERR(tas_dev->regmap)) { + ret = PTR_ERR(tas_dev->regmap); + dev_err(dev, "Failed devm_regmap_init: %d\n", ret); + goto out; + } + ret = tasdevice_init(tas_dev); + +out: + if (ret < 0 && tas_dev != NULL) + tasdevice_remove(tas_dev); + + dev_dbg(dev, "%s was called for TAS2783.\n", __func__); + return ret; + +} + +static int tasdevice_sdw_remove(struct sdw_slave *peripheral) +{ + struct tasdevice_priv *tas_dev = + dev_get_drvdata(&peripheral->dev); + + if (tas_dev) + tasdevice_remove(tas_dev); + + return 0; +} + +static const struct sdw_device_id tasdevice_sdw_id[] = { + SDW_SLAVE_ENTRY(0x0102, 0x0, 0), + {}, +}; +MODULE_DEVICE_TABLE(sdw, tasdevice_sdw_id); + +static struct sdw_driver tasdevice_sdw_driver = { + .driver = { + .name = "slave-tas2783", + }, + .probe = tasdevice_sdw_probe, + .remove = tasdevice_sdw_remove, + .ops = &tasdevice_sdw_ops, + .id_table = tasdevice_sdw_id, +}; + +module_sdw_driver(tasdevice_sdw_driver); + +MODULE_AUTHOR("Baojun Xu <baojun.xu(a)ti.com>"); +MODULE_DESCRIPTION("ASoC TAS2783 SoundWire Driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tas2783.h b/sound/soc/codecs/tas2783.h new file mode 100644 index 000000000000..0fa5b3855b51 --- /dev/null +++ b/sound/soc/codecs/tas2783.h @@ -0,0 +1,178 @@ +/* SPDX-License-Identifier: GPL-2.0 + * + * ALSA SoC Texas Instruments TAS2781 Audio Smart Amplifier + * + * Copyright (C) 2023 Texas Instruments Incorporated + * https://www.ti.com + * + * The TAS2783 driver implements a flexible and configurable + * algo coff setting for single TAS2783 chips. + * + * Author: Baojun Xu <baojun.xu(a)ti.com> + */ + +#ifndef __TAS2783_H__ +#define __TAS2783_H__ + +#define TASDEVICE_RATES (SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | \ + SNDRV_PCM_RATE_96000 | \ + SNDRV_PCM_RATE_88200) +#define TAS2783_SDW_ADDR_OFFSET 0x8000 + +#define TASDEVICE_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +/* BOOK, PAGE Control Register */ +#define TASDEVICE_REG(book, page, reg) ((book * 256 * 256) + 0x8000 +\ + (page * 128) + reg) + +/*Software Reset */ +#define TAS2873_REG_SWRESET TASDEVICE_REG(0x0, 0X0, 0x01) +#define TAS2873_REG_SWRESET_RESET BIT(0) + +/* Volume control */ +#define TAS2783_DVC_LVL TASDEVICE_REG(0x0, 0x00, 0x1A) +#define TAS2783_AMP_LEVEL TASDEVICE_REG(0x0, 0x00, 0x03) +#define TAS2783_AMP_LEVEL_MASK GENMASK(5, 1) + +/* Calibration data */ +#define TAS2783_CALIBRATION_RE TASDEVICE_REG(0x0, 0x17, 0x74) +#define TAS2783_CALIBRATION_RE_LOW TASDEVICE_REG(0x0, 0x18, 0x14) +#define TAS2783_CALIBRATION_INV_RE TASDEVICE_REG(0x0, 0x18, 0x0c) +#define TAS2783_CALIBRATION_POW TASDEVICE_REG(0x0, 0x0d, 0x3c) +#define TAS2783_CALIBRATION_TLIMIT TASDEVICE_REG(0x0, 0x18, 0x7c) + +#define TAS2783_CALIBRATION_SIZE 5 // int data number. + +#define TAS2783_DEVICE_ID_START 0x08 // Unique id start +#define TAS2783_DEVICE_ID_END 0x0F // Unique id end + +/* TAS2783 SDCA Control - function number */ +#define FUNC_NUM_SMART_AMP 0x01 + +/* TAS2783 SDCA entity */ +#define TAS2783_SDCA_ENT_PDE23 0x0C +#define TAS2783_SDCA_ENT_PDE22 0x0B +#define TAS2783_SDCA_ENT_FU21 0x01 +#define TAS2783_SDCA_ENT_UDMPU21 0x10 + +/* TAS2783 SDCA control */ +#define TAS2783_SDCA_CTL_REQ_POWER_STATE 0x01 +#define TAS2783_SDCA_CTL_FU_MUTE 0x01 +#define TAS2783_SDCA_CTL_UDMPU_CLUSTER 0x10 + +#define CH_L 1 +#define CH_R 2 + +#define TASDEVICE_CRC8_POLYNOMIAL 0x4d + +#define MAX_NO_FILES 100 +#define MAX_CALIBRATION_DATA_SIZE 252 + +#define SMS_HTONL(a, b, c, d) ((((a)&0x000000FF)<<24) | \ + (((b)&0x000000FF)<<16) | \ + (((c)&0x000000FF)<<8) | \ + ((d)&0x000000FF)) + +enum channel { + top_left_Chn, + top_right_chn, + bottom_left_Chn, + bottom_right_chn, + max_chn, +}; + +struct tasdevice { + unsigned int dev_addr; + unsigned int err_code; + short cur_prog; + short cur_conf; + bool is_loading; + bool is_loaderr; + bool is_calibrated_data_loaded; + unsigned char cur_book; + struct tasdevice_fw *cali_data_fmw; +}; + +struct MYSWFTFile { + unsigned int m_vendor_id; + unsigned int m_file_id; + unsigned int m_version; + unsigned int m_Length; + unsigned int m_downloadAddr; + unsigned char *m_startAdress; +}; + +/* + * This item is used to store the generic i2c address of + * all the tas2781 devices for I2C broadcast during the multi-device + * writes, useless in mono case. + */ +struct global_addr { + unsigned char cur_book; + unsigned int dev_addr; + int ref_cnt; +}; + +struct calibration_data { + unsigned long total_sz; + unsigned char data[MAX_CALIBRATION_DATA_SIZE]; +}; + +struct tas_control { + struct snd_kcontrol_new *tasdevice_profile_controls; + int nr_controls; +}; + +struct tasdevice_irqinfo { + int irq_gpio; + int irq; +}; + +struct tasdevice_priv { + struct device *dev; + struct snd_soc_component *component; + struct sdw_slave *sdw_slave; + enum sdw_slave_status status; + struct sdw_bus_params params; + struct regmap *regmap; + struct mutex codec_lock; + struct mutex dev_lock; + int rst_gpio; + struct tasdevice tasdevice[max_chn]; + struct tasdevice_fw *fmw; + struct tasdevice_irqinfo irq_info; + struct tas_control tas_ctrl; + struct sdw_slave *sdw_peripheral; + struct global_addr glb_addr; + struct calibration_data cali_data; + struct tm tm; + int cur_prog; + int cur_conf; + unsigned int chip_id; + bool hw_init; + bool first_hw_init; + unsigned char *bq_params; + unsigned int magic_num; + unsigned char ndev; + unsigned char dev_name[32]; + unsigned char rca_binaryname[64]; + unsigned char coef_binaryname[64]; + unsigned char crc8_lkp_tbl[CRC8_TABLE_SIZE]; + + struct MYSWFTFile m_swftFile[MAX_NO_FILES]; + + void *codec; + unsigned int sysclk; + bool is_glb_calibrated_data_loaded; +}; + +struct sdw_stream_data { + struct sdw_stream_runtime *sdw_stream; +}; + +int tasdevice_dev_read(struct tasdevice_priv *tasdevice, + unsigned int *reg, unsigned int *value); + +#endif /*__TAS2783_H__ */ -- 2.34.1

4 4

[PATCH v3 0/6] ASoC: mediatek: Add support for MT7986 SoC
by Maso Huang 04 Aug '23

04 Aug '23

Changes in v3: - merge clk api to mt7986-afe-pcm.c, remove [2/7] in v2 - refine based on reviewer's suggestions [1/6] [2/6] - fix the comment format, move in clk api, and simplify based on reviewer's suggestions [3/6] - remove redundant prefix in dt-binding [6/6] Changes in v2: - v1 title: [PATCH 0/7] ASoC: mediatek: Add support for MT79xx SoC - add missing maintainers - rename mt79xx to mt7986 in all files - use clk bulk api in mt7986-afe-clk.c [2/7] - refine mt79xx-afe-pcm.c based on reviewer's suggestions [4/7] - refine dt-binding files based on reviewer's suggestions [6/7] [7/7] - transpose [3/7] and [4/7] in v1 to fix test build errors Maso Huang (6): ASoC: mediatek: mt7986: add common header ASoC: mediatek: mt7986: support etdm in platform driver ASoC: mediatek: mt7986: add platform driver ASoC: mediatek: mt7986: add machine driver with wm8960 ASoC: dt-bindings: mediatek,mt7986-wm8960: add mt7986-wm8960 document ASoC: dt-bindings: mediatek,mt7986-afe: add audio afe document .../bindings/sound/mediatek,mt7986-afe.yaml | 89 +++ .../sound/mediatek,mt7986-wm8960.yaml | 53 ++ sound/soc/mediatek/Kconfig | 20 + sound/soc/mediatek/Makefile | 1 + sound/soc/mediatek/mt7986/Makefile | 9 + sound/soc/mediatek/mt7986/mt7986-afe-common.h | 49 ++ sound/soc/mediatek/mt7986/mt7986-afe-pcm.c | 622 ++++++++++++++++++ sound/soc/mediatek/mt7986/mt7986-dai-etdm.c | 420 ++++++++++++ sound/soc/mediatek/mt7986/mt7986-reg.h | 206 ++++++ sound/soc/mediatek/mt7986/mt7986-wm8960.c | 184 ++++++ 10 files changed, 1653 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/mediatek,mt7986-afe.yaml create mode 100644 Documentation/devicetree/bindings/sound/mediatek,mt7986-wm8960.yaml create mode 100644 sound/soc/mediatek/mt7986/Makefile create mode 100644 sound/soc/mediatek/mt7986/mt7986-afe-common.h create mode 100644 sound/soc/mediatek/mt7986/mt7986-afe-pcm.c create mode 100644 sound/soc/mediatek/mt7986/mt7986-dai-etdm.c create mode 100644 sound/soc/mediatek/mt7986/mt7986-reg.h create mode 100644 sound/soc/mediatek/mt7986/mt7986-wm8960.c -- 2.18.0

5 16

Re: [PATCH v3 6/6] ASoC: dt-bindings: mediatek,mt7986-afe: add audio afe document
by Krzysztof Kozlowski 04 Aug '23

04 Aug '23

On 01/08/2023 10:25, Maso Huang (黃加竹) wrote: > On Fri, 2023-07-28 at 14:51 +0200, Krzysztof Kozlowski wrote: >> >> External email : Please do not click links or open attachments until >> you have verified the sender or the content. >> On 28/07/2023 11:08, Maso Huang wrote: >>> Add mt7986 audio afe document. >>> >>> Signed-off-by: Maso Huang <maso.huang(a)mediatek.com> >> >> Thank you for your patch. There is something to discuss/improve. >> >>> + >>> +required: >>> + - compatible >>> + - reg >>> + - interrupts >>> + - clocks >>> + - clock-names >>> + - assigned-clocks >>> + - assigned-clock-parents >> >> You should constrain your clocks per variants. I doubt that they are >> really so flexible/optional on each SoC... or maybe missing clocks >> are >> result of unimplemented parts in the driver? But then this should not >> really affect bindings. Bindings still should require such clocks. >> Your >> DTS can always provide a <0>, if needed. >> >> > > Hi Krzysztof, > > After internal check, assigned-clocks and assigned-clock-parents are > not required on this SoC. > Maybe we can just drop these two options? It's separate issue, but yes - why requiring them? I wrote about missing constraints for your clocks in the bindings. Best regards, Krzysztof

1 0

[PATCH -next] ASoC: q6dsp: Make g_apm staic
by GUO Zihua 04 Aug '23

04 Aug '23

g_apm is not referred elsewhere, so make it static. This resolves sparse warning: warning: symbol 'g_apm' was not declared. Should it be static? Signed-off-by: GUO Zihua <guozihua(a)huawei.com> --- sound/soc/qcom/qdsp6/q6apm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/qcom/qdsp6/q6apm.c b/sound/soc/qcom/qdsp6/q6apm.c index 5d44d07acd69..2a2a5bd98110 100644 --- a/sound/soc/qcom/qdsp6/q6apm.c +++ b/sound/soc/qcom/qdsp6/q6apm.c @@ -27,7 +27,7 @@ struct apm_graph_mgmt_cmd { #define APM_GRAPH_MGMT_PSIZE(p, n) ALIGN(struct_size(p, sub_graph_id_list, n), 8) -struct q6apm *g_apm; +static struct q6apm *g_apm; int q6apm_send_cmd_sync(struct q6apm *apm, struct gpr_pkt *pkt, uint32_t rsp_opcode) { -- 2.17.1

1 0

[RFC] aplay: Add option for specifying subformat
by GitHub pull_request - opened 04 Aug '23

04 Aug '23

alsa-project/alsa-utils pull request #228 was opened from crojewsk-intel: Make subformat first-class citizen by allowing users to specify it just like it is the case for the format. Default to SND_PCM_SUBFORMAT_STD so they are no surprises. Request URL : https://github.com/alsa-project/alsa-utils/pull/228 Patch URL : https://github.com/alsa-project/alsa-utils/pull/228.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[RFC] pcm: Add MSBITS_20 and MSBITS_24 subformat options
by GitHub pull_request - opened 04 Aug '23

04 Aug '23

alsa-project/alsa-lib pull request #342 was opened from crojewsk-intel: Improve granularity of format selection for S32/U32 formats by adding masks representing 20 and 24 most significant bits. As a preparation step introduce `snd_pcm_subformat_value()` for easy name-to-subformat conversions. While at it, fix few warnings that pop up in subformat-related code during compilation. Request URL : https://github.com/alsa-project/alsa-lib/pull/342 Patch URL : https://github.com/alsa-project/alsa-lib/pull/342.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH] ASoC: soc-acpi: Add missing kernel doc
by Charles Keepax 03 Aug '23

03 Aug '23

The UID field in snd_soc_acpi_link_adr is not documented, add kernel doc for it. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- include/sound/soc-acpi.h | 1 + 1 file changed, 1 insertion(+) diff --git a/include/sound/soc-acpi.h b/include/sound/soc-acpi.h index 630bf7367fe69..6d31d535e8f6d 100644 --- a/include/sound/soc-acpi.h +++ b/include/sound/soc-acpi.h @@ -151,6 +151,7 @@ struct snd_soc_acpi_link_adr { * all firmware/topology related fields. * * @id: ACPI ID (usually the codec's) used to find a matching machine driver. + * @uid: ACPI Unique ID, can be used to disambiguate matches. * @comp_ids: list of compatible audio codecs using the same machine driver, * firmware and topology * @link_mask: describes required board layout, e.g. for SoundWire. -- 2.30.2

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[PATCH v2 RESEND*3] ASoC: fsl MPC52xx drivers require PPC_BESTCOMM
by Randy Dunlap 03 Aug '23

03 Aug '23

Both SND_MPC52xx_SOC_PCM030 and SND_MPC52xx_SOC_EFIKA select SND_SOC_MPC5200_AC97. The latter symbol depends on PPC_BESTCOMM, so the 2 former symbols should also depend on PPC_BESTCOMM since "select" does not follow any dependency chains. This prevents a kconfig warning and build errors: WARNING: unmet direct dependencies detected for SND_SOC_MPC5200_AC97 Depends on [n]: SOUND [=y] && !UML && SND [=m] && SND_SOC [=m] && SND_POWERPC_SOC [=m] && PPC_MPC52xx [=y] && PPC_BESTCOMM [=n] Selected by [m]: - SND_MPC52xx_SOC_PCM030 [=m] && SOUND [=y] && !UML && SND [=m] && SND_SOC [=m] && SND_POWERPC_SOC [=m] && PPC_MPC5200_SIMPLE [=y] - SND_MPC52xx_SOC_EFIKA [=m] && SOUND [=y] && !UML && SND [=m] && SND_SOC [=m] && SND_POWERPC_SOC [=m] && PPC_EFIKA [=y] ERROR: modpost: "mpc5200_audio_dma_destroy" [sound/soc/fsl/mpc5200_psc_ac97.ko] undefined! ERROR: modpost: "mpc5200_audio_dma_create" [sound/soc/fsl/mpc5200_psc_ac97.ko] undefined! Fixes: 40d9ec14e7e1 ("ASoC: remove BROKEN from Efika and pcm030 fabric drivers") Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Cc: Grant Likely <grant.likely(a)secretlab.ca> Cc: Mark Brown <broonie(a)kernel.org> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Shengjiu Wang <shengjiu.wang(a)gmail.com> Cc: Xiubo Li <Xiubo.Lee(a)gmail.com> Cc: alsa-devel(a)alsa-project.org Cc: linuxppc-dev(a)lists.ozlabs.org Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Acked-by: Shengjiu Wang <shengjiu.wang(a)gmail.com> --- v2: use correct email address for Mark Brown. sound/soc/fsl/Kconfig | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff -- a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -243,7 +243,7 @@ config SND_SOC_MPC5200_AC97 config SND_MPC52xx_SOC_PCM030 tristate "SoC AC97 Audio support for Phytec pcm030 and WM9712" - depends on PPC_MPC5200_SIMPLE + depends on PPC_MPC5200_SIMPLE && PPC_BESTCOMM select SND_SOC_MPC5200_AC97 select SND_SOC_WM9712 help @@ -252,7 +252,7 @@ config SND_MPC52xx_SOC_PCM030 config SND_MPC52xx_SOC_EFIKA tristate "SoC AC97 Audio support for bbplan Efika and STAC9766" - depends on PPC_EFIKA + depends on PPC_EFIKA && PPC_BESTCOMM select SND_SOC_MPC5200_AC97 select SND_SOC_STAC9766 help

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[PATCH] ASoC: fsl: micfil: Use dual license micfil code
by Daniel Baluta 03 Aug '23

03 Aug '23

From: Daniel Baluta <daniel.baluta(a)nxp.com> We need this in order to easily reuse register definitions and some functions with Sound Open Firmware driver. According to Documentation/process/license-rules.rst: "Dual BSD/GPL" The module is dual licensed under a GPL v2 variant or BSD license choice. The exact variant of the BSD license can only be determined via the license information in the corresponding source files. so use "Dual BSD/GPL" for license string. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> Signed-off-by: Daniel Baluta <daniel.baluta(a)nxp.com> --- sound/soc/fsl/fsl_micfil.c | 4 ++-- sound/soc/fsl/fsl_micfil.h | 2 +- 2 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/fsl/fsl_micfil.c b/sound/soc/fsl/fsl_micfil.c index fe28b27e50d0..97cf315781ab 100644 --- a/sound/soc/fsl/fsl_micfil.c +++ b/sound/soc/fsl/fsl_micfil.c @@ -1,4 +1,4 @@ -// SPDX-License-Identifier: GPL-2.0 +// SPDX-License-Identifier: GPL-2.0 OR BSD-3-Clause // Copyright 2018 NXP #include <linux/bitfield.h> @@ -1254,4 +1254,4 @@ module_platform_driver(fsl_micfil_driver); MODULE_AUTHOR("Cosmin-Gabriel Samoila <cosmin.samoila(a)nxp.com>"); MODULE_DESCRIPTION("NXP PDM Microphone Interface (MICFIL) driver"); -MODULE_LICENSE("GPL v2"); +MODULE_LICENSE("Dual BSD/GPL"); diff --git a/sound/soc/fsl/fsl_micfil.h b/sound/soc/fsl/fsl_micfil.h index 9237a1c4cb8f..fee9fe3d9119 100644 --- a/sound/soc/fsl/fsl_micfil.h +++ b/sound/soc/fsl/fsl_micfil.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: GPL-2.0 */ +/* SPDX-License-Identifier: GPL-2.0 OR BSD-3-Clause */ /* * PDM Microphone Interface for the NXP i.MX SoC * Copyright 2018 NXP -- 2.25.1

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[PATCH] ASoC: mediatek: mt8186: Remove unused mutex.
by Sebastian Andrzej Siewior 03 Aug '23

03 Aug '23

The mutex mutex_request_dram has no user. Remove mutex_request_dram. Signed-off-by: Sebastian Andrzej Siewior <bigeasy(a)linutronix.de> --- sound/soc/mediatek/mt8186/mt8186-afe-clk.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/mediatek/mt8186/mt8186-afe-clk.c b/sound/soc/mediatek/mt8186/mt8186-afe-clk.c index 539e3a023bc4e..70ec101890d36 100644 --- a/sound/soc/mediatek/mt8186/mt8186-afe-clk.c +++ b/sound/soc/mediatek/mt8186/mt8186-afe-clk.c @@ -13,8 +13,6 @@ #include "mt8186-afe-clk.h" #include "mt8186-audsys-clk.h" -static DEFINE_MUTEX(mutex_request_dram); - static const char *aud_clks[CLK_NUM] = { [CLK_AFE] = "aud_afe_clk", [CLK_DAC] = "aud_dac_clk", -- 2.40.1

3 2

[PATCH v2 0/3] Add PWM-DAC audio support for StarFive JH7110 RISC-V SoC
by Hal Feng 03 Aug '23

03 Aug '23

This patchset adds PWM-DAC audio support for the StarFive JH7110 SoC. The PWM-DAC module does not require a hardware codec, but a dummy codec is needed for the driver. The dummy spdif codec driver, which is already upstream, is compatible with the one which JH7110 PWM-DAC needed. So we use it as the dummy codec driver for the JH7110 PWM-DAC module. The third patch depends on tag next-20230728 in linux-next branch. Changes since v1: - Rebase on tag v6.5-rc3. - Drop patch 1 and 2. - Drop the unneeded space and line in patch 3. - Use the dummy spdif codec driver instead of adding a new one. - Change "dai_link->stop_dma_first = 1" to "dai_link->trigger_stop = SND_SOC_TRIGGER_ORDER_LDC" in patch 4. - Drop the unneeded "status = "okay;" in patch 5. - Change some node names in patch 5. --- v1: https://lore.kernel.org/all/20230626110909.38718-1-hal.feng@starfivetech.co… Hal Feng (3): ASoC: dt-bindings: Add StarFive JH7110 PWM-DAC controller ASoC: starfive: Add JH7110 PWM-DAC driver riscv: dts: starfive: Add JH7110 PWM-DAC support .../sound/starfive,jh7110-pwmdac.yaml | 76 ++ MAINTAINERS | 7 + .../jh7110-starfive-visionfive-2.dtsi | 99 +++ arch/riscv/boot/dts/starfive/jh7110.dtsi | 156 ++++ sound/soc/starfive/Kconfig | 9 + sound/soc/starfive/Makefile | 1 + sound/soc/starfive/jh7110_pwmdac.c | 787 ++++++++++++++++++ 7 files changed, 1135 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/starfive,jh7110-pwmdac.yaml create mode 100644 sound/soc/starfive/jh7110_pwmdac.c base-commit: 6eaae198076080886b9e7d57f4ae06fa782f90ef -- 2.38.1

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[PATCH 0/6] ASoC: SOF: misc updates for 6.6
by Pierre-Louis Bossart 03 Aug '23

03 Aug '23

Cleanups, move of code to promote reuse and Mediatek minor updates. Curtis Malainey (1): ASoC: SOF: Deprecate invalid enums in IPC3 Pierre-Louis Bossart (3): ASoC: SOF: Intel: start simplify the signature of link_slaves_found() ASoC: soc-acpi: move link_slaves_found() ASoC: soc-acpi: improve log messagesin link_slaves_found() Trevor Wu (2): ASoC: SOF: ipc3: update dai_link_fixup for SOF_DAI_MEDIATEK_AFE ASoC: SOF: mediatek: mt8186 modify dram type as non-cache include/linux/soundwire/sdw.h | 5 ++ include/linux/soundwire/sdw_intel.h | 7 +-- include/sound/soc-acpi.h | 6 ++ include/sound/sof/topology.h | 4 +- sound/soc/soc-acpi.c | 73 +++++++++++++++++++++++ sound/soc/sof/intel/hda.c | 81 +------------------------- sound/soc/sof/ipc3-pcm.c | 17 ++++++ sound/soc/sof/mediatek/mt8186/mt8186.c | 40 +++++++------ 8 files changed, 130 insertions(+), 103 deletions(-) -- 2.39.2

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[PATCH 00/38] ASoC: merge DAI call back functions into ops
by Kuninori Morimoto 03 Aug '23

03 Aug '23

Hi Mark snd_soc_dai_driver has .ops for call back functions (A), but it also has other call back functions (B). It is duplicated and confusable. struct snd_soc_dai_driver { ... ^ int (*probe)(...); | int (*remove)(...); (B) int (*compress_new)(...); | int (*pcm_new)(...); v ... (A) const struct snd_soc_dai_ops *ops; ... } This patch merges (B) into (A). Kuninori Morimoto (38): ASoC: soc-dai.h: merge DAI call back functions into ops ASoC: ti: merge DAI call back functions into ops ASoC: adi: merge DAI call back functions into ops ASoC: adi: merge DAI call back functions into ops ASoC: amd: merge DAI call back functions into ops ASoC: dwc: merge DAI call back functions into ops ASoC: pxa: merge DAI call back functions into ops ASoC: bcm: merge DAI call back functions into ops ASoC: fsl: merge DAI call back functions into ops ASoC: img: merge DAI call back functions into ops ASoC: sof: merge DAI call back functions into ops ASoC: sti: merge DAI call back functions into ops ASoC: stm: merge DAI call back functions into ops ASoC: pxa: merge DAI call back functions into ops ASoC: rsnd: merge DAI call back functions into ops ASoC: qcom: merge DAI call back functions into ops ASoC: ux500: merge DAI call back functions into ops ASoC: sunxi: merge DAI call back functions into ops ASoC: tegra: merge DAI call back functions into ops ASoC: atmel: merge DAI call back functions into ops ASoC: intel: merge DAI call back functions into ops ASoC: meson: merge DAI call back functions into ops ASoC: jz4740: merge DAI call back functions into ops ASoC: cirrus: merge DAI call back functions into ops ASoC: drm/vc4: merge DAI call back functions into ops ASoC: samsung: merge DAI call back functions into ops ASoC: mediatek: merge DAI call back functions into ops ASoC: rockchip: merge DAI call back functions into ops ASoC: uniphier: merge DAI call back functions into ops ASoC: loongson: merge DAI call back functions into ops ASoC: starfive: merge DAI call back functions into ops ASoC: hisilicon: merge DAI call back functions into ops ASoC: codecs/wm*: merge DAI call back functions into ops ASoC: soc-topology: merge DAI call back functions into ops ASoC: codecs/cs47lxx: merge DAI call back functions into ops ASoC: codecs/cx2072x: merge DAI call back functions into ops ASoC: codecs/hdmi-codec: merge DAI call back functions into ops ASoC: soc-dai.h: remove unused call back functions drivers/gpu/drm/vc4/vc4_hdmi.c | 6 +- include/sound/soc-dai.h | 26 +-- sound/soc/adi/axi-i2s.c | 2 +- sound/soc/adi/axi-spdif.c | 2 +- sound/soc/amd/acp/acp-i2s.c | 24 +-- sound/soc/amd/acp/acp-rembrandt.c | 3 - sound/soc/amd/acp/acp-renoir.c | 2 - sound/soc/amd/acp/amd.h | 1 - sound/soc/atmel/atmel-i2s.c | 16 +- sound/soc/atmel/mchp-i2s-mcc.c | 24 +-- sound/soc/atmel/mchp-pdmc.c | 18 +-- sound/soc/atmel/mchp-spdifrx.c | 14 +- sound/soc/atmel/mchp-spdiftx.c | 18 +-- sound/soc/bcm/bcm2835-i2s.c | 24 +-- sound/soc/cirrus/ep93xx-i2s.c | 2 +- sound/soc/codecs/cs47l15.c | 6 +- sound/soc/codecs/cs47l24.c | 8 +- sound/soc/codecs/cs47l35.c | 8 +- sound/soc/codecs/cs47l85.c | 8 +- sound/soc/codecs/cs47l90.c | 8 +- sound/soc/codecs/cs47l92.c | 6 +- sound/soc/codecs/cx2072x.c | 11 +- sound/soc/codecs/hdmi-codec.c | 44 ++--- sound/soc/codecs/wm2200.c | 12 +- sound/soc/codecs/wm5102.c | 6 +- sound/soc/codecs/wm5110.c | 8 +- sound/soc/codecs/wm8994.c | 2 +- sound/soc/dwc/dwc-i2s.c | 18 +-- sound/soc/fsl/fsl_asrc.c | 16 +- sound/soc/fsl/fsl_aud2htx.c | 10 +- sound/soc/fsl/fsl_easrc.c | 16 +- sound/soc/fsl/fsl_esai.c | 20 +-- sound/soc/fsl/fsl_micfil.c | 14 +- sound/soc/fsl/fsl_sai.c | 24 +-- sound/soc/fsl/fsl_spdif.c | 17 +- sound/soc/fsl/fsl_ssi.c | 3 +- sound/soc/fsl/fsl_xcvr.c | 16 +- sound/soc/generic/audio-graph-card.c | 2 +- sound/soc/hisilicon/hi6210-i2s.c | 2 +- sound/soc/img/img-i2s-in.c | 14 +- sound/soc/img/img-i2s-out.c | 14 +- sound/soc/img/img-parallel-out.c | 14 +- sound/soc/img/img-spdif-in.c | 12 +- sound/soc/img/img-spdif-out.c | 12 +- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 2 +- sound/soc/intel/avs/probes.c | 10 +- sound/soc/intel/catpt/pcm.c | 8 +- sound/soc/intel/keembay/kmb_platform.c | 4 +- sound/soc/jz4740/jz4740-i2s.c | 3 +- sound/soc/loongson/loongson_i2s.c | 16 +- sound/soc/mediatek/mt8195/mt8195-dai-etdm.c | 56 ++++--- sound/soc/meson/aiu-acodec-ctrl.c | 4 +- sound/soc/meson/aiu-codec-ctrl.c | 4 +- sound/soc/meson/aiu-fifo-i2s.c | 3 + sound/soc/meson/aiu-fifo-spdif.c | 3 + sound/soc/meson/aiu.c | 6 - sound/soc/meson/axg-frddr.c | 4 +- sound/soc/meson/axg-pdm.c | 18 +-- sound/soc/meson/axg-spdifin.c | 4 +- sound/soc/meson/axg-tdm-interface.c | 6 +- sound/soc/meson/axg-toddr.c | 4 +- sound/soc/meson/g12a-toacodec.c | 4 +- sound/soc/meson/g12a-tohdmitx.c | 4 +- sound/soc/pxa/mmp-sspa.c | 2 +- sound/soc/pxa/pxa-ssp.c | 4 +- sound/soc/qcom/lpass-apq8016.c | 4 - sound/soc/qcom/lpass-cpu.c | 40 +++-- sound/soc/qcom/lpass-ipq806x.c | 1 - sound/soc/qcom/lpass-sc7180.c | 5 +- sound/soc/qcom/lpass-sc7280.c | 2 - sound/soc/qcom/lpass.h | 4 +- sound/soc/qcom/qdsp6/q6afe-dai.c | 88 +++++----- sound/soc/qcom/qdsp6/q6asm-dai.c | 6 +- sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c | 3 - sound/soc/rockchip/rockchip_i2s.c | 2 +- sound/soc/rockchip/rockchip_i2s_tdm.c | 2 +- sound/soc/rockchip/rockchip_pdm.c | 2 +- sound/soc/rockchip/rockchip_spdif.c | 2 +- sound/soc/samsung/i2s.c | 5 +- sound/soc/samsung/pcm.c | 18 +-- sound/soc/sh/rcar/core.c | 27 ++-- sound/soc/soc-dai.c | 44 ++--- sound/soc/soc-topology.c | 6 +- sound/soc/sof/sof-client-probes.c | 6 +- sound/soc/starfive/jh7110_tdm.c | 16 +- sound/soc/sti/sti_uniperif.c | 6 +- sound/soc/stm/stm32_i2s.c | 2 +- sound/soc/stm/stm32_sai_sub.c | 18 ++- sound/soc/stm/stm32_spdifrx.c | 2 +- sound/soc/sunxi/sun4i-codec.c | 6 +- sound/soc/sunxi/sun4i-i2s.c | 18 +-- sound/soc/sunxi/sun4i-spdif.c | 2 +- sound/soc/sunxi/sun50i-dmic.c | 2 +- sound/soc/tegra/tegra20_ac97.c | 10 +- sound/soc/tegra/tegra20_i2s.c | 2 +- sound/soc/tegra/tegra20_spdif.c | 2 +- sound/soc/tegra/tegra210_admaif.c | 12 +- sound/soc/tegra/tegra30_i2s.c | 2 +- sound/soc/ti/davinci-i2s.c | 22 +-- sound/soc/ti/davinci-mcasp.c | 27 ++-- sound/soc/ti/omap-dmic.c | 22 +-- sound/soc/ti/omap-mcbsp.c | 28 ++-- sound/soc/ti/omap-mcpdm.c | 22 +-- sound/soc/uniphier/aio-cpu.c | 161 +++++++++++++++---- sound/soc/uniphier/aio-ld11.c | 62 ++----- sound/soc/uniphier/aio-pxs2.c | 55 +------ sound/soc/uniphier/aio.h | 10 +- sound/soc/ux500/ux500_msp_dai.c | 2 +- 108 files changed, 792 insertions(+), 688 deletions(-) -- 2.25.1

1 0

[PATCH v2 0/3] Update the register list of MICFIL
by Chancel Liu 02 Aug '23

02 Aug '23

MICFIL IP is upgraded on i.MX93 platform. Add new registers and new bit definition. changes in v2: - rename check_version to use_verid to make it more explicit - rename fsl_micfil_check_version to fsl_micfil_use_verid Chancel Liu (3): ASoC: fsl_micfil: Add new registers and new bit definition ASoC: fsl_micfil: Add fsl_micfil_use_verid function ASoC: fsl_micfil: Use SET_SYSTEM_SLEEP_PM_OPS to simplify PM sound/soc/fsl/fsl_micfil.c | 100 ++++++++++++++++++++++++++++++------- sound/soc/fsl/fsl_micfil.h | 64 ++++++++++++++++++++++++ 2 files changed, 146 insertions(+), 18 deletions(-) -- 2.25.1

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[PATCH 0/2] ALSA: hda/realtek: Add headphone quirks for Aya devices
by ruinairas1992＠gmail.com 02 Aug '23

02 Aug '23

The content of this message was lost. It was probably cross-posted to multiple lists and previously handled on another list.

3 3

[PATCH v1] ALSA: hda: cs35l41: Print amp configuration after bind
by Stefan Binding 02 Aug '23

02 Aug '23

Print amp configuration information to be able to confirm ACPI _DSD information (and other useful info) for each amp on each system using CS35L41, without having to get the acpidump. Signed-off-by: Stefan Binding <sbinding(a)opensource.cirrus.com> --- sound/pci/hda/cs35l41_hda.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/pci/hda/cs35l41_hda.c b/sound/pci/hda/cs35l41_hda.c index 98feb5ccd5866..825e551be9bb8 100644 --- a/sound/pci/hda/cs35l41_hda.c +++ b/sound/pci/hda/cs35l41_hda.c @@ -1124,6 +1124,13 @@ static int cs35l41_hda_bind(struct device *dev, struct device *master, void *mas pm_runtime_mark_last_busy(dev); pm_runtime_put_autosuspend(dev); + dev_info(cs35l41->dev, + "CS35L41 Bound - SSID: %s, BST: %d, VSPK: %d, CH: %c, FW EN: %d, SPKID: %d\n", + cs35l41->acpi_subsystem_id, cs35l41->hw_cfg.bst_type, + cs35l41->hw_cfg.gpio1.func == CS35l41_VSPK_SWITCH, + cs35l41->hw_cfg.spk_pos ? 'R' : 'L', + cs35l41->firmware_running, cs35l41->speaker_id); + return ret; } -- 2.34.1

2 1

[PATCH -next] ALSA: ac97: set variables dev_attr_vendor_id to static
by Yu Liao 02 Aug '23

02 Aug '23

sparse reports sound/ac97/bus.c:465:1: sparse: sparse: symbol 'dev_attr_vendor_id' was not declared. Should it be static? This variable is only used in its defining file, so it should be static. Reported-by: kernel test robot <lkp(a)intel.com> Closes: https://lore.kernel.org/oe-kbuild-all/202307242300.Oy0Dp2QI-lkp@intel.com/ Signed-off-by: Yu Liao <liaoyu15(a)huawei.com> --- sound/ac97/bus.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/ac97/bus.c b/sound/ac97/bus.c index 6067c04ce4c0..3173e9d98927 100644 --- a/sound/ac97/bus.c +++ b/sound/ac97/bus.c @@ -462,7 +462,7 @@ static ssize_t vendor_id_show(struct device *dev, return sysfs_emit(buf, "%08x", codec->vendor_id); } -DEVICE_ATTR_RO(vendor_id); +static DEVICE_ATTR_RO(vendor_id); static struct attribute *ac97_dev_attrs[] = { &dev_attr_vendor_id.attr, -- 2.25.1

1 0

[PATCH] ASoC: intel: use devm_snd_soc_register_dai()
by Kuninori Morimoto 02 Aug '23

02 Aug '23

It is still using snd_soc_{un}register_dai() manually. Let's use devm_snd_soc_register_dai(). Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/intel/avs/pcm.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/intel/avs/pcm.c b/sound/soc/intel/avs/pcm.c index 1fbb2c2fadb5..dd03db31c2d8 100644 --- a/sound/soc/intel/avs/pcm.c +++ b/sound/soc/intel/avs/pcm.c @@ -1309,8 +1309,6 @@ static void avs_component_hda_unregister_dais(struct snd_soc_component *componen for_each_pcm_streams(stream) snd_soc_dapm_free_widget(snd_soc_dai_get_widget(dai, stream)); - - snd_soc_unregister_dai(dai); } } @@ -1375,7 +1373,7 @@ static int avs_component_hda_probe(struct snd_soc_component *component) } } - dai = snd_soc_register_dai(component, &dais[i], false); + dai = devm_snd_soc_register_dai(component->dev, component, &dais[i], false); if (!dai) { dev_err(component->dev, "register dai for %s failed\n", pcm->name); -- 2.25.1

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How do I set up multiple codecs on one I2S - without TDM?
by John Watts 01 Aug '23

01 Aug '23

Hello there! I have an interesting problem that I'm trying to solve. It will require writing adding driver support but right now I'm struggling to understand how I should go about it all. As a background, this is with the Allwinner D1 or T113. It has a 16 channel I2S controller and each channel can be configured to use a specific pin and TDM time slice. Even channels are low LRCLK, odd are high LRCLK. So for my situation I want to have six channels: - Channel 0: Timeslot 0, Pin 0, LRCLK 0 -> ADC 1 - Channel 1: Timeslot 0, Pin 0, LRCLK 1 -> ADC 1 - Channel 2: Timeslot 0, Pin 1, LRCLK 0 -> ADC 2 - Channel 3: Timeslot 0, Pin 1, LRCLK 1 -> ADC 2 - Channel 4: Timeslot 0, Pin 2, LRCLK 0 -> ADC 3 - Channel 5: Timeslot 0, Pin 2, LRCLK 1 -> ADC 3 These ADCs are WM8782s, completely TDM unaware. If they were TDM aware then I could run even more ADCs by having multiple separate TDMs going on each pin. I have tried to look online for I2S controllers that do the same thing but haven't had much luck. The current driver for this controller only supports TDM. So I will have to add support for this. So my questions are: - How do I express this in terms of ALSA and sound card concepts? - How do I express this in the device tree? My immediate thought is that this is a situation where I have a card with 3 dai-links between the I2S controller and each separate codec. I have written this tree as a test: wm8782_1: stereo-adc { compatible = "wlf,wm8782"; Vdda-supply = <&reg_vcc>; Vdd-supply = <&reg_vcc>; #sound-dai-cells = <0>; }; wm8782_2: stereo-adc { compatible = "wlf,wm8782"; Vdda-supply = <&reg_vcc>; Vdd-supply = <&reg_vcc>; #sound-dai-cells = <0>; }; wm8782_3: stereo-adc { compatible = "wlf,wm8782"; Vdda-supply = <&reg_vcc>; Vdd-supply = <&reg_vcc>; #sound-dai-cells = <0>; }; sound { compatible = "simple-audio-card"; simple-audio-card,name = "Testcard In"; simple-audio-card,format = "i2s"; simple-audio-card,bitclock-master = <&sound_cpu>; simple-audio-card,frame-master = <&sound_cpu>; simple-audio-card,mclk-fs = <128>; #address-cells = <1>; #size-cells = <0>; simple-audio-card,dai-link@0 { reg = <0>; sound_cpu: cpu { sound-dai = <&i2s2>; }; codec { sound-dai = <&wm8782_1>; }; }; simple-audio-card,dai-link@1 { reg = <1>; cpu { sound-dai = <&i2s2>; }; codec { sound-dai = <&wm8782_2>; }; }; simple-audio-card,dai-link@2 { reg = <2>; cpu { sound-dai = <&i2s2>; }; codec { sound-dai = <&wm8782_3>; }; }; }; But I get this kernel error: kobject_add_internal failed for 2034000.i2s-wm8782 with -EEXIST, don't try to register things with the same name asoc-simple-card: probe of sound failed with error -12 The WM8782 doesn't support TDM or multiplexing so I'm not exactly sure if this is correct behaviour or my mistake configuring. I tried adding TDM nodes which didn't help. I also tried having a single link with multiple codecs on sound-dai which did create a sound card but it was unclear to me if this was ignoring the other codecs I had listed or not. I also can't test this at this point as the sunxi codec driver assumes that the number of channels is equal to TDM slots, which isn't true any more. Any pointers in the right direction would be a huge help. Surely someone else has had this problem. Thanks, John.

2 3

[PATCH] ASoC: rt5665: add missed regulator_bulk_disable
by Zhang Shurong 01 Aug '23

01 Aug '23

The driver forgets to call regulator_bulk_disable() Add the missed call to fix it. Fixes: 33ada14a26c8 ("ASoC: add rt5665 codec driver") Signed-off-by: Zhang Shurong <zhang_shurong(a)foxmail.com> --- sound/soc/codecs/rt5665.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/codecs/rt5665.c b/sound/soc/codecs/rt5665.c index 83c367af91da..de36cad8a559 100644 --- a/sound/soc/codecs/rt5665.c +++ b/sound/soc/codecs/rt5665.c @@ -4756,6 +4756,13 @@ static void rt5665_calibrate_handler(struct work_struct *work) rt5665_calibrate(rt5665); } +static void rt5665_i2c_disable_regulators(void *data) +{ + struct rt5665_priv *rt5665 = data; + + regulator_bulk_disable(ARRAY_SIZE(rt5665->supplies), rt5665->supplies); +} + static int rt5665_i2c_probe(struct i2c_client *i2c) { struct rt5665_platform_data *pdata = dev_get_platdata(&i2c->dev); @@ -4786,6 +4793,11 @@ static int rt5665_i2c_probe(struct i2c_client *i2c) return ret; } + ret = devm_add_action_or_reset(&i2c->dev, rt5665_i2c_disable_regulators, + rt5665); + if (ret) + return ret; + ret = regulator_bulk_enable(ARRAY_SIZE(rt5665->supplies), rt5665->supplies); if (ret != 0) { -- 2.41.0

2 2

[PATCH 00/23] ASoC: Intel: machine driver updates for 6.6
by Pierre-Louis Bossart 01 Aug '23

01 Aug '23

Lots of small cleanups coming from Bard Liao and Charles Keepax for SoundWire platforms, and minor additions for RVPs and Chromebooks. Balamurugan C (1): ASoC: Intel: soc-acpi: Add entry for rt711-sdca-sdw at link 0 in RPL match table Bard Liao (2): ASoC: Intel: sof_sdw: rename link_id to be_id ASoC: Intel: sof_sdw: break earlier when a adr link contains different codecs Chao Song (1): ASoC: Intel: sof_sdw: add cs35l56 codec info Charles Keepax (13): ASoC: intel: sof_sdw: Use consistent variable naming for links ASoC: intel: sof_sdw: Rename codec_idx to codec_dlc_index ASoC: intel: sof_sdw: Remove some extra line breaks ASoC: intel: sof_sdw: Use a module device table ASoC: intel: sof_sdw: Simplify find_codec_info_acpi ASoC: intel: sof_sdw: Constify parameter to find_codec_part_info ASoC: intel: sof_sdw: Minor tidy up of mc_probe ASoC: intel: sof_sdw: Remove redundant parameters in dai creation ASoC: intel: sof_sdw: Move amp_num initialisation to mc_probe ASoC: intel: sof_sdw: Allow direct specification of CODEC name ASoC: intel: sof_sdw: Clean up DAI link counting ASoC: intel: sof_sdw: Merge codec_conf_alloc into dailink_info ASoC: intel: sof_sdw: Move group_generated logic Pierre-Louis Bossart (3): ASoC: Intel: sof_sdw: reorder SoundWire codecs in Kconfig ASoC: Intel: sof_sdw: allow mockup amplifier to provide feedback ASoC: Intel: sof_sdw: add support for SKU 0AFE Terry Cheong (1): ASoC: Intel: sof_rt5682: add RPL support for MAX98357A speaker Uday M Bhat (1): ASoC: Intel: sof_sdw: Update BT offload config for soundwire config apoorv (1): ASoC: Intel: sof_rt5682 add support for HDMI_In capture sound/soc/intel/boards/Kconfig | 5 +- sound/soc/intel/boards/Makefile | 2 +- sound/soc/intel/boards/sof_rt5682.c | 55 +++ sound/soc/intel/boards/sof_sdw.c | 404 +++++++++--------- sound/soc/intel/boards/sof_sdw_common.h | 7 + sound/soc/intel/boards/sof_sdw_cs_amp.c | 47 ++ .../intel/common/soc-acpi-intel-adl-match.c | 7 + .../intel/common/soc-acpi-intel-rpl-match.c | 27 ++ 8 files changed, 342 insertions(+), 212 deletions(-) create mode 100644 sound/soc/intel/boards/sof_sdw_cs_amp.c -- 2.39.2

2 24

[PATCH -next] ALSA: usb-audio: Remove unused function declaration
by Yue Haibing 01 Aug '23

01 Aug '23

Commit 68e67f40b734 ("ALSA: snd-usb: move calls to usb_set_interface") leave this unused declaration. Signed-off-by: Yue Haibing <yuehaibing(a)huawei.com> --- sound/usb/endpoint.h | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/usb/endpoint.h b/sound/usb/endpoint.h index c09f68ce08b1..ba70f52f6860 100644 --- a/sound/usb/endpoint.h +++ b/sound/usb/endpoint.h @@ -44,7 +44,6 @@ int snd_usb_endpoint_start(struct snd_usb_endpoint *ep); void snd_usb_endpoint_stop(struct snd_usb_endpoint *ep, bool keep_pending); void snd_usb_endpoint_sync_pending_stop(struct snd_usb_endpoint *ep); void snd_usb_endpoint_suspend(struct snd_usb_endpoint *ep); -int snd_usb_endpoint_activate(struct snd_usb_endpoint *ep); void snd_usb_endpoint_release(struct snd_usb_endpoint *ep); void snd_usb_endpoint_free_all(struct snd_usb_audio *chip); -- 2.34.1

2 1

[PATCH v1 2/2] MAINTAINERS: Add the MAINTAINERS entries for TEXAS INSTRUMENTS AUDIO DRIVERS
by Baojun.Xu 01 Aug '23

01 Aug '23

From: Baojun Xu <baojun.xu(a)ti.com> Add the MAINTAINERS entries for TEXAS INSTRUMENTS AUDIO DRIVERS. Signed-off-by: Baojun Xu <baojun.xu(a)ti.com> --- Change in v1: - Add the MAINTAINERS entries for TI. --- MAINTAINERS | 32 ++++++++++++++++++++++++++++++++ 1 file changed, 32 insertions(+) diff --git a/MAINTAINERS b/MAINTAINERS index 53b7ca804465..2c16168167e2 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -21063,6 +21063,38 @@ S: Maintained F: Documentation/devicetree/bindings/sound/davinci-mcasp-audio.yaml F: sound/soc/ti/ +TEXAS INSTRUMENTS AUDIO (ASoC/HDA) DRIVERS +M: Shenghao Ding <shenghao-ding(a)ti.com> +M: Kevin Lu <kevin-lu(a)ti.com> +M: Baojun Xu <baojun.xu(a)ti.com> +L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) +S: Maintained +F: Documentation/devicetree/bindings/sound/tas2552.txt +F: Documentation/devicetree/bindings/sound/tas2562.yaml +F: Documentation/devicetree/bindings/sound/tas2770.yaml +F: Documentation/devicetree/bindings/sound/tas27xx.yaml +F: Documentation/devicetree/bindings/sound/ti,pcm1681.txt +F: Documentation/devicetree/bindings/sound/ti,pcm3168a.yaml +F: Documentation/devicetree/bindings/sound/ti,tlv320*.yaml +F: Documentation/devicetree/bindings/sound/tlv320adcx140.yaml +F: Documentation/devicetree/bindings/sound/tlv320aic31xx.txt +F: Documentation/devicetree/bindings/sound/tpa6130a2.txt +F: include/sound/tas2*.h +F: include/sound/tlv320*.h +F: include/sound/tpa6130a2-plat.h +F: sound/soc/codecs/pcm1681.c +F: sound/soc/codecs/pcm1789*.* +F: sound/soc/codecs/pcm179x*.* +F: sound/soc/codecs/pcm186x*.* +F: sound/soc/codecs/pcm3008.* +F: sound/soc/codecs/pcm3060*.* +F: sound/soc/codecs/pcm3168a*.* +F: sound/soc/codecs/pcm5102a.c +F: sound/soc/codecs/pcm512x*.* +F: sound/soc/codecs/tas2*.* +F: sound/soc/codecs/tlv320*.* +F: sound/soc/codecs/tpa6130a2.* + TEXAS INSTRUMENTS DMA DRIVERS M: Peter Ujfalusi <peter.ujfalusi(a)gmail.com> L: dmaengine(a)vger.kernel.org -- 2.34.1

1 0

[PATCH v1 1/2] ASoC: tas2783: Add source files for tas2783 soundwire driver
by Baojun.Xu 01 Aug '23

01 Aug '23

From: Baojun Xu <baojun.xu(a)ti.com> Add source file and header file for tas2783 driver. Update Kconfig and Makefile for tas2783 driver. Signed-off-by: Baojun Xu <baojun.xu(a)ti.com> --- Change in v1: - Add tas2783-sdw.c - Add tas2783.h - Update Makefile for tas2783 driver. - Update Kconfig for tas2783 driver. --- sound/soc/codecs/Kconfig | 12 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tas2783-sdw.c | 1166 ++++++++++++++++++++++++++++++++ sound/soc/codecs/tas2783.h | 178 +++++ 4 files changed, 1358 insertions(+) create mode 100644 sound/soc/codecs/tas2783-sdw.c create mode 100644 sound/soc/codecs/tas2783.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index c2de4ee72183..318ad867030c 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -240,6 +240,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_TAS2781_COMLIB imply SND_SOC_TAS2781_FMWLIB imply SND_SOC_TAS2781_I2C + imply SND_SOC_TAS2783 imply SND_SOC_TAS5086 imply SND_SOC_TAS571X imply SND_SOC_TAS5720 @@ -1762,6 +1763,17 @@ config SND_SOC_TAS5086 tristate "Texas Instruments TAS5086 speaker amplifier" depends on I2C +config SND_SOC_TAS2783 + tristate "Texas Instruments TAS2781 speaker amplifier (sdw)" + depends on SOUNDWIRE + select REGMAP + select CRC8 + help + Enable support for Texas Instruments TAS2783 Smart Amplifier + Digital input mono Class-D and DSP-inside audio power amplifiers. + Note the TAS2783 driver implements a flexible and configurable + algo coff setting, for one, two, even multiple TAS2783 chips. + config SND_SOC_TAS571X tristate "Texas Instruments TAS571x power amplifiers" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index b48a9a323b84..3298dfa3b9d5 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -276,6 +276,7 @@ snd-soc-tas2770-objs := tas2770.o snd-soc-tas2781-comlib-objs := tas2781-comlib.o snd-soc-tas2781-fmwlib-objs := tas2781-fmwlib.o snd-soc-tas2781-i2c-objs := tas2781-i2c.o +snd-soc-tas2783-objs := tas2783-sdw.o snd-soc-tfa9879-objs := tfa9879.o snd-soc-tfa989x-objs := tfa989x.o snd-soc-tlv320adc3xxx-objs := tlv320adc3xxx.o @@ -648,6 +649,7 @@ obj-$(CONFIG_SND_SOC_TAS2780) += snd-soc-tas2780.o obj-$(CONFIG_SND_SOC_TAS2781_COMLIB) += snd-soc-tas2781-comlib.o obj-$(CONFIG_SND_SOC_TAS2781_FMWLIB) += snd-soc-tas2781-fmwlib.o obj-$(CONFIG_SND_SOC_TAS2781_I2C) += snd-soc-tas2781-i2c.o +obj-$(CONFIG_SND_SOC_TAS2783) += snd-soc-tas2783.o obj-$(CONFIG_SND_SOC_TAS5086) += snd-soc-tas5086.o obj-$(CONFIG_SND_SOC_TAS571X) += snd-soc-tas571x.o obj-$(CONFIG_SND_SOC_TAS5720) += snd-soc-tas5720.o diff --git a/sound/soc/codecs/tas2783-sdw.c b/sound/soc/codecs/tas2783-sdw.c new file mode 100644 index 000000000000..93b1e7a1f311 --- /dev/null +++ b/sound/soc/codecs/tas2783-sdw.c @@ -0,0 +1,1165 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// ALSA SoC Texas Instruments TAS2783 Audio Smart Amplifier +// +// Copyright (C) 2023 Texas Instruments Incorporated +// https://www.ti.com +// +// The TAS2783 driver implements a flexible and configurable +// algo coefficient setting for single TAS2783 chips. +// +// Author: Baojun Xu <baojun.xu(a)ti.com> +// Author: Kevin Lu <kevin-lu(a)ti.com> +// + +#include <linux/delay.h> +#include <linux/device.h> +#include <linux/pm_runtime.h> +#include <linux/err.h> +#include <linux/module.h> +#include <linux/regmap.h> +#include <linux/soundwire/sdw.h> +#include <linux/soundwire/sdw_registers.h> +#include <linux/soundwire/sdw_type.h> +#include <linux/init.h> +#include <linux/pm.h> +#include <linux/of.h> +#include <linux/slab.h> +#include <linux/crc8.h> +#include <linux/crc32.h> +#include <linux/efi.h> +#include <linux/firmware.h> +#include <linux/of_gpio.h> +#include <linux/interrupt.h> +#include <sound/soc.h> +#include <sound/sdw.h> +#include <sound/pcm_params.h> +#include <sound/tlv.h> +#include <sound/tas2781-tlv.h> + +#include "tas2783.h" + +static const unsigned int tas2783_calibration_reg[] = { + TAS2783_CALIBRATION_RE, + TAS2783_CALIBRATION_RE_LOW, + TAS2783_CALIBRATION_INV_RE, + TAS2783_CALIBRATION_POW, + TAS2783_CALIBRATION_TLIMIT, + 0, +}; + +static const struct reg_default tas2783_reg_defaults[] = { + // Default values for ROM mode. Activated. + { 0x8002, 0x1a}, + { 0x800e, 0x44}, + { 0x800f, 0x40}, + { 0x8097, 0xc8}, + { 0x80b5, 0x74}, + { 0x8099, 0x20}, + { 0xfe8d, 0x0d}, + { 0xfebe, 0x4a}, + { 0x8230, 0x00}, + { 0x8231, 0x00}, + { 0x8232, 0x00}, + { 0x8233, 0x01}, + { 0x8418, 0x00}, + { 0x8419, 0x00}, + { 0x841a, 0x00}, + { 0x841b, 0x00}, + { 0x8428, 0x40}, + { 0x8429, 0x00}, + { 0x842a, 0x00}, + { 0x842b, 0x00}, + { 0x8548, 0x00}, + { 0x8549, 0x00}, + { 0x854a, 0x00}, + { 0x854b, 0x00}, + { 0x8558, 0x40}, + { 0x8559, 0x00}, + { 0x855a, 0x00}, + { 0x855b, 0x00}, + { 0x800a, 0x3a}, + { 0x800e, 0x44}, + { 0x800f, 0x40}, + { 0x805c, 0xd9}, + { 0x8002, 0x00}, +}; + +static bool tas2783_readable_register(struct device *dev, + unsigned int reg) +{ + switch (reg) { + case 0x8000 ... 0xc000: // Page 0 ~ 127. + case 0xfe80 ... 0xfeff: // Page 253. + case SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, TAS2783_SDCA_ENT_UDMPU21, + TAS2783_SDCA_CTL_UDMPU_CLUSTER, 0): + case SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, TAS2783_SDCA_ENT_FU21, + TAS2783_SDCA_CTL_FU_MUTE, CH_L): + case SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, TAS2783_SDCA_ENT_FU21, + TAS2783_SDCA_CTL_FU_MUTE, CH_R): + case SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, TAS2783_SDCA_ENT_PDE23, + TAS2783_SDCA_CTL_REQ_POWER_STATE, 0): + case SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, TAS2783_SDCA_ENT_PDE22, + TAS2783_SDCA_CTL_REQ_POWER_STATE, 0): + return true; + default: + return false; + } +} + +static bool tas2783_volatile_register(struct device *dev, + unsigned int reg) +{ + switch (reg) { + case 0x8001: + // Only reset register was volatiled. + return true; + default: + return false; + } +} + +static const struct regmap_config tasdevice_regmap = { + .reg_bits = 32, + .val_bits = 8, + .readable_reg = tas2783_readable_register, + .volatile_reg = tas2783_volatile_register, + .max_register = 0x41008000 + TASDEVICE_REG(0xa1, 0x60, 0x7f), + .reg_defaults = tas2783_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(tas2783_reg_defaults), + .cache_type = REGCACHE_RBTREE, + .use_single_read = true, + .use_single_write = true, +}; + +static int tas2783_sdw_read(void *context, const void *reg_buf, + const size_t reg_size, void *val_buf, size_t val_size) +{ + struct sdw_slave *peripheral = context; + u8 *buf8 = val_buf; + unsigned int reg, bytes; + int ret; + + reg = le32_to_cpu(*(const __le32 *)reg_buf); + reg += TAS2783_SDW_ADDR_OFFSET; + + while (val_size) { + /* to end of page */ + bytes = SDW_REG_NO_PAGE - (reg & SDW_REGADDR); + if (bytes > val_size) + bytes = val_size; + + ret = sdw_nread_no_pm(peripheral, reg, bytes, buf8); + if (ret != 0) { + dev_err(&peripheral->dev, "Read @%#x..%#x:%d\n", + reg, reg + bytes - 1, ret); + return ret; + } + + swab32_array((u32 *)buf8, bytes / 4); + val_size -= bytes; + reg += bytes; + buf8 += bytes; + } + + return 0; +} + +/* Convert for LSB or MSB. */ +static inline void tas2783_swab_copy( + void *dest, const void *src, size_t nbytes) +{ + u32 *dest32 = dest; + const u32 *src32 = src; + + for (; nbytes > 0; nbytes -= 4) + *dest32++ = swab32(*src32++); +} + +static int tas2783_sdw_gather_write(void *context, const void *reg_buf, + size_t reg_size, const void *val_buf, size_t val_size) +{ + struct sdw_slave *peripheral = context; + const u8 *src_be = val_buf; + u32 val_le_buf[64]; /* Define u32 so it is 32-bit aligned */ + unsigned int reg, bytes; + int ret; + + reg = le32_to_cpu(*(const __le32 *)reg_buf); + + while (val_size) { + /* to end of page */ + bytes = SDW_REG_NO_PAGE - (reg & SDW_REGADDR); + if (bytes > val_size) + bytes = val_size; + if (bytes > sizeof(val_le_buf)) + bytes = sizeof(val_le_buf); + + tas2783_swab_copy(val_le_buf, src_be, bytes); + + ret = sdw_nwrite_no_pm(peripheral, reg, bytes, + (u8 *)val_le_buf); + if (ret != 0) { + dev_err(&peripheral->dev, "Write @%#x..%#x:%d\n", + reg, reg + bytes - 1, ret); + return ret; + } + + val_size -= bytes; + reg += bytes; + src_be += bytes; + } + + return 0; +} + +static int tas2783_sdw_write( + void *context, const void *val_buf, size_t val_size) +{ + const u8 *src_buf = val_buf; + + /* First word of val_buf contains the destination address */ + return tas2783_sdw_gather_write(context, &src_buf[0], 4, + &src_buf[4], val_size - 4); +} + +static int tasdevice_clamp(int val, int max, unsigned int invert) +{ + if (val > max) + val = max; + if (invert) + val = max - val; + if (val < 0) + val = 0; + return val; +} + +/* + * Registers are big-endian on I2C and SPI but little-endian on SoundWire. + * Exported firmware controls are big-endian on I2C/SPI but little-endian + * on SoundWire. + * Firmware files are always big-endian and are opaque blobs. + * Present a big-endian regmap and hide the endianness swap, + * so that the ALSA byte controls always have the same byte order, + * and firmware file blobs can be written verbatim. + */ +static const struct regmap_bus tas2783_regmap_bus_sdw = { + .read = tas2783_sdw_read, + .write = tas2783_sdw_write, + .gather_write = tas2783_sdw_gather_write, + .reg_format_endian_default = REGMAP_ENDIAN_LITTLE, + .val_format_endian_default = REGMAP_ENDIAN_BIG, +}; + +static int tas2783_digital_getvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec + = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_dev = + snd_soc_component_get_drvdata(codec); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + int val; + unsigned int invert = mc->invert; + int max = mc->max; + int ret = 0; + + /* Read the primary device as the whole */ + ret = tasdevice_dev_read(tas_dev, &mc->reg, (unsigned int *)&val); + dev_dbg(tas_dev->dev, "%s, get digital vol %d from %x with %d\n", + __func__, val, mc->reg, ret); + if (ret) { + dev_err(tas_dev->dev, "%s, get digital vol error %x.\n", + __func__, ret); + goto out; + } + + if (val > max) + val = max; + if (invert) + val = max - val; + if (val < 0) + val = 0; + ucontrol->value.integer.value[0] = val; + +out: + return ret; +} + +static int tas2783_digital_putvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec + = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_dev = + snd_soc_component_get_drvdata(codec); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + int val, ret = 0; + unsigned int invert = mc->invert; + int max = mc->max; + unsigned char value = 0; + + val = ucontrol->value.integer.value[0]; + if (val > max) + val = max; + if (invert) + val = max - val; + if (val < 0) + val = 0; + + value = (u8)val; + ret = sdw_nwrite_no_pm(tas_dev->sdw_slave, mc->reg, 1, &value); + if (ret != 0) { + dev_err(tas_dev->dev, "Write @%#x..%#x:%d\n", + mc->reg, val, ret); + return ret; + } + dev_dbg(tas_dev->dev, "%s, Put digital vol %d into %x.\n", + __func__, val, mc->reg); + + return 1; +} + +static int tas2783_amp_getvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec + = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_dev = + snd_soc_component_get_drvdata(codec); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + int val; + unsigned char mask = 0; + int max = mc->max; + int ret = 0; + unsigned int invert = mc->invert; + unsigned char value = 0; + + /* Read the primary device */ + ret = sdw_nread_no_pm(tas_dev->sdw_slave, mc->reg, 1, &value); + dev_dbg(tas_dev->dev, "%s, get AMP vol %d from %x with %d\n", + __func__, value, mc->reg, ret); + + mask = (1 << fls(mc->max)) - 1; + mask <<= mc->shift; + val = value; + val = (val & mask) >> mc->shift; + if (val > max) + val = max; + if (invert) + val = max - val; + if (val < 0) + val = 0; + ucontrol->value.integer.value[0] = val; + + return ret; +} + +static int tas2783_amp_putvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec + = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_dev = + snd_soc_component_get_drvdata(codec); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + int val = 0, ret = 0, max = mc->max; + unsigned char mask = 0, value = 0; + unsigned int invert = mc->invert; + + mask = (1 << fls(mc->max)) - 1; + mask <<= mc->shift; + val = tasdevice_clamp(ucontrol->value.integer.value[0], max, + invert); + + ret = sdw_nread_no_pm(tas_dev->sdw_slave, mc->reg, 1, &value); + if (ret != 0) { + dev_err(tas_dev->dev, "Read @%x.%x:%d\n", + mc->reg, value, ret); + return ret; + } + value &= (~TAS2783_AMP_LEVEL_MASK); + value |= (val << mc->shift); + ret = sdw_nwrite_no_pm(tas_dev->sdw_slave, mc->reg, 1, &value); + if (ret != 0) { + dev_err(tas_dev->dev, "Write @%#x..%#x:%d\n", + mc->reg, val, ret); + return ret; + } + dev_dbg(tas_dev->dev, "amp %x into 0x%08x by %lx shift %d.\n", + val, mc->reg, TAS2783_AMP_LEVEL_MASK, mc->shift); + + return ret; +} + +static const struct snd_kcontrol_new tas2783_snd_controls[] = { + SOC_SINGLE_RANGE_EXT_TLV("Amp Gain Volume", TAS2783_AMP_LEVEL, + 1, 0, 20, 0, tas2783_amp_getvol, + tas2783_amp_putvol, amp_vol_tlv), + SOC_SINGLE_RANGE_EXT_TLV("Digital Volume", TAS2783_DVC_LVL, + 0, 0, 200, 1, tas2783_digital_getvol, + tas2783_digital_putvol, dvc_tlv), +}; + +static void tas2783_apply_calib( + struct tasdevice_priv *tas_priv, unsigned int *cali_data) +{ + unsigned int reg_address = 0, i = 0; + unsigned int *reg_start; + + if (!tas_priv->sdw_slave) + return; + if ((tas_priv->sdw_slave->id.unique_id < TAS2783_DEVICE_ID_START) + || (tas_priv->sdw_slave->id.unique_id > TAS2783_DEVICE_ID_END)) + return; + reg_start = cali_data+(tas_priv->sdw_slave->id.unique_id - + TAS2783_DEVICE_ID_START)*TAS2783_CALIBRATION_SIZE; + while (tas2783_calibration_reg[i] != 0) { + reg_address = tas2783_calibration_reg[i]; + tas2783_sdw_gather_write(tas_priv->sdw_slave, + &reg_address, 4, reg_start+i, 4); + } +} + +static int tas2783_calibration(struct tasdevice_priv *tas_priv) +{ + efi_guid_t efi_guid = EFI_GUID(0x1f52d2a1, 0xbb3a, 0x457d, 0xbc, + 0x09, 0x43, 0xa3, 0xf4, 0x31, 0x0a, 0x92); + static efi_char16_t efi_name[] = L"CALI_DATA"; + struct tm *tm = &tas_priv->tm; + unsigned int attr, crc; + unsigned int *tmp_val; + efi_status_t status; + + tas_priv->cali_data.total_sz = 128; + /* Get real size of UEFI variable */ + status = efi.get_variable(efi_name, &efi_guid, &attr, + &tas_priv->cali_data.total_sz, tas_priv->cali_data.data); + dev_dbg(tas_priv->dev, "cali get %lx bytes with result : %ld\n", + tas_priv->cali_data.total_sz, status); + if (status == EFI_BUFFER_TOO_SMALL) { + status = efi.get_variable(efi_name, &efi_guid, &attr, + &tas_priv->cali_data.total_sz, + tas_priv->cali_data.data); + dev_dbg(tas_priv->dev, "cali get %lx bytes result:%ld\n", + tas_priv->cali_data.total_sz, status); + } + for (int i = 0; i < tas_priv->cali_data.total_sz; i += 4) { + dev_dbg(tas_priv->dev, "cali get %02x %02x %02x %02x", + tas_priv->cali_data.data[i], + tas_priv->cali_data.data[i+1], + tas_priv->cali_data.data[i+2], + tas_priv->cali_data.data[i+3]); + } + + tmp_val = (unsigned int *)tas_priv->cali_data.data; + + crc = crc32(~0, tas_priv->cali_data.data, 84) ^ ~0; + dev_dbg(tas_priv->dev, "cali crc 0x%08x PK tmp_val 0x%08x\n", + crc, tmp_val[21]); + + if (crc == tmp_val[21]) { + time64_to_tm(tmp_val[20], 0, tm); + dev_dbg(tas_priv->dev, "%4ld-%2d-%2d, %2d:%2d:%2d\n", + tm->tm_year, tm->tm_mon, tm->tm_mday, + tm->tm_hour, tm->tm_min, tm->tm_sec); + tas2783_apply_calib(tas_priv, tmp_val); + } else { + dev_dbg(tas_priv->dev, "CRC error!\n"); + tas_priv->cali_data.total_sz = 0; + } + + return 0; +} + +static void tasdevice_rca_ready(const struct firmware *fmw, void *context) +{ + struct tasdevice_priv *tas_dev = + (struct tasdevice_priv *) context; + struct MYSWFTFile *p; + unsigned char *buf = NULL, value_sdw; + int offset = 0, i = 0, num_files = 0, img_sz = 0; + int ret = 0; + + mutex_lock(&tas_dev->codec_lock); + + if (!fmw || !fmw->data) { + dev_err(tas_dev->dev, + "Failed to read %s, no side - effect on driver running\n", + tas_dev->rca_binaryname); + ret = -1; + goto out; + } + buf = (unsigned char *)fmw->data; + + img_sz = SMS_HTONL(buf[offset+3], buf[offset + 2], + buf[offset + 1], buf[offset]); + dev_dbg(tas_dev->dev, "Got %x:%lx.\n", img_sz, fmw->size); + offset += sizeof(img_sz); + if (img_sz != fmw->size) { + dev_err(tas_dev->dev, + "Size not match, %d %u", (int)fmw->size, img_sz); + ret = -1; + goto out; + } + + while ((offset < img_sz) && (num_files < MAX_NO_FILES)) { + p = (struct MYSWFTFile *)(buf + offset); + tas_dev->m_swftFile[num_files].m_vendor_id = + p->m_vendor_id; + tas_dev->m_swftFile[num_files].m_file_id = p->m_file_id; + tas_dev->m_swftFile[num_files].m_version = p->m_version; + tas_dev->m_swftFile[num_files].m_Length = p->m_Length; + tas_dev->m_swftFile[num_files].m_downloadAddr = + p->m_downloadAddr; + tas_dev->m_swftFile[num_files].m_startAdress = + ((char *)p) + sizeof(unsigned int)*5; + + ret = sdw_nwrite_no_pm(tas_dev->sdw_peripheral, + tas_dev->m_swftFile[num_files].m_downloadAddr, + tas_dev->m_swftFile[num_files].m_Length, + tas_dev->m_swftFile[num_files].m_startAdress); + dev_dbg(tas_dev->dev, "Write %x into %x result : %d\n", + tas_dev->m_swftFile[num_files].m_Length, + tas_dev->m_swftFile[num_files].m_downloadAddr, + ret); + + dev_dbg(tas_dev->dev, "Write %d :%x:%x:%x:%x:%x:%x.\n", + num_files, + tas_dev->m_swftFile[num_files].m_vendor_id, + tas_dev->m_swftFile[num_files].m_file_id, + tas_dev->m_swftFile[num_files].m_version, + tas_dev->m_swftFile[num_files].m_Length, + tas_dev->m_swftFile[num_files].m_downloadAddr, + tas_dev->m_swftFile[num_files].m_startAdress[0]); + + ret = sdw_nread_no_pm(tas_dev->sdw_peripheral, + tas_dev->m_swftFile[num_files].m_downloadAddr, + 1, &value_sdw); + dev_dbg(tas_dev->dev, "Got block %d :%x.\n", + num_files, value_sdw); + + offset += sizeof(unsigned int)*5 + p->m_Length; + num_files++; + } + + i = 0x50; + while (i < 0x56) { + ret = sdw_nread_no_pm(tas_dev->sdw_peripheral, i, 1, + &value_sdw); + dev_dbg(tas_dev->dev, "Got device id:%x\n", value_sdw); + i++; + } + tas2783_calibration(tas_dev); + +out: + mutex_unlock(&tas_dev->codec_lock); + if (fmw) + release_firmware(fmw); +} + +static void tasdevice_config_info_remove(void *context) +{ + struct tasdevice_priv *tas_dev = + (struct tasdevice_priv *) context; + dev_dbg(tas_dev->dev, "%s was performed\n", __func__); +} + +static int tasdevice_dapm_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *codec = + snd_soc_dapm_to_component(w->dapm); + struct tasdevice_priv *tas_dev = + snd_soc_component_get_drvdata(codec); + int state = 1; + + /* Codec Lock Hold */ + mutex_lock(&tas_dev->codec_lock); + dev_dbg(tas_dev->dev, "dapm event : %x\n", event); + if (event == SND_SOC_DAPM_PRE_PMD) + state = 0; + /* Codec Lock Release*/ + mutex_unlock(&tas_dev->codec_lock); + + return 0; +} + +static const struct snd_soc_dapm_widget tasdevice_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN("ASI", "ASI Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT_E("ASI OUT", "ASI Capture", 0, SND_SOC_NOPM, + 0, 0, tasdevice_dapm_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_SPK("SPK", tasdevice_dapm_event), + SND_SOC_DAPM_OUTPUT("OUT"), + SND_SOC_DAPM_INPUT("DMIC") +}; + +static const struct snd_soc_dapm_route tasdevice_audio_map[] = { + {"SPK", NULL, "ASI"}, + {"OUT", NULL, "SPK"}, + {"ASI OUT", NULL, "DMIC"} +}; + +static int tasdevice_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + int ret = 0; + + return ret; +} + +static int tasdevice_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct tasdevice_priv *tas_dev = + snd_soc_dai_get_drvdata(codec_dai); + + tas_dev->sysclk = freq; + + return 0; +} + +static int tasdevice_set_sdw_stream( + struct snd_soc_dai *dai, void *sdw_stream, int direction) +{ + struct sdw_stream_data *stream; + + if (!sdw_stream) + return 0; + + stream = kzalloc(sizeof(*stream), GFP_KERNEL); + if (!stream) + return -ENOMEM; + + stream->sdw_stream = sdw_stream; + + /* Use tx_mask or rx_mask to set dma_data */ + snd_soc_dai_dma_data_set(dai, direction, stream); + + return 0; +} + +static void tasdevice_sdw_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct sdw_stream_data *stream; + + stream = snd_soc_dai_get_dma_data(dai, substream); + snd_soc_dai_set_dma_data(dai, substream, NULL); + kfree(stream); +} + +static int tasdevice_sdw_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct tasdevice_priv *tasdevice = + snd_soc_component_get_drvdata(component); + struct sdw_stream_config stream_config = {0}; + struct sdw_port_config port_config = {0}; + struct sdw_stream_data *stream; + int retval; + + dev_dbg(dai->dev, "%s %s", __func__, dai->name); + stream = snd_soc_dai_get_dma_data(dai, substream); + + if (!stream) + return -EINVAL; + + if (!tasdevice->sdw_slave) + return -EINVAL; + + /* SoundWire specific configuration */ + snd_sdw_params_to_config(substream, params, + &stream_config, &port_config); + + /* port 1 for playback */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + port_config.num = 1; + else + port_config.num = 2; + + retval = sdw_stream_add_slave(tasdevice->sdw_slave, + &stream_config, &port_config, 1, stream->sdw_stream); + if (retval) { + dev_err(dai->dev, "Unable to configure port\n"); + return retval; + } + + return 0; +} + +static int tasdevice_sdw_pcm_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct tasdevice_priv *tasdevice = + snd_soc_component_get_drvdata(component); + struct sdw_stream_data *stream = + snd_soc_dai_get_dma_data(dai, substream); + + if (!tasdevice->sdw_slave) + return -EINVAL; + + sdw_stream_remove_slave(tasdevice->sdw_slave, stream->sdw_stream); + return 0; +} + +static int tasdevice_mute(struct snd_soc_dai *dai, int mute, + int direction) +{ + struct snd_soc_component *component = dai->component; + struct tasdevice_priv *tasdevice = + snd_soc_component_get_drvdata(component); + int ret = 0; + unsigned char value_sdw = 0; + + dev_dbg(tasdevice->dev, "%s: mute %d, %d\n", + __func__, mute, direction); + if (mute == 0) {// Unmute. + // FU21 Unmute + value_sdw = 0; + ret = sdw_nwrite_no_pm(tasdevice->sdw_peripheral, + SDW_SDCA_CTL(1, 1, 1, 0), 1, &value_sdw); + dev_dbg(tasdevice->dev, "Unmuted %lx %x\n", + SDW_SDCA_CTL(1, 1, 1, 0), ret); + value_sdw = 0; + ret = sdw_nwrite_no_pm(tasdevice->sdw_peripheral, + 0x40400090, 1, &value_sdw); + dev_dbg(tasdevice->dev, "Unmuted %x\n", ret); + ret = sdw_nwrite_no_pm(tasdevice->sdw_peripheral, + 0x40402090, 1, &value_sdw); + dev_dbg(tasdevice->dev, "Unmuted %x\n", ret); + + // FU23 Unmute + value_sdw = 0; + ret = sdw_nwrite_no_pm(tasdevice->sdw_peripheral, + SDW_SDCA_CTL(1, 2, 1, 0), 1, &value_sdw); + dev_dbg(tasdevice->dev, "Unmuted %lx %x\n", + SDW_SDCA_CTL(1, 2, 1, 0), ret); + value_sdw = 0; + ret = sdw_nwrite_no_pm(tasdevice->sdw_peripheral, + 0x40400108, 1, &value_sdw); + dev_dbg(tasdevice->dev, "Unmuted %x\n", ret); + value_sdw = 0; + ret = sdw_nwrite_no_pm(tasdevice->sdw_peripheral, + 0x8002, 1, &value_sdw); + dev_dbg(tasdevice->dev, "Unmuted %x\n", ret); + } else { + // FU21 mute + value_sdw = 1; + ret = sdw_nwrite_no_pm(tasdevice->sdw_peripheral, + SDW_SDCA_CTL(1, 1, 1, 0), 1, &value_sdw); + dev_dbg(tasdevice->dev, "muted %lx %x\n", + SDW_SDCA_CTL(1, 1, 1, 0), ret); + value_sdw = 1; + ret = sdw_nwrite_no_pm(tasdevice->sdw_peripheral, + 0x40400090, 1, &value_sdw); + dev_dbg(tasdevice->dev, "muted %x\n", ret); + ret = sdw_nwrite_no_pm(tasdevice->sdw_peripheral, + 0x40402090, 1, &value_sdw); + dev_dbg(tasdevice->dev, "muted %x\n", ret); + // FU23 mute + value_sdw = 1; + ret = sdw_nwrite_no_pm(tasdevice->sdw_peripheral, + SDW_SDCA_CTL(1, 2, 1, 0), 1, &value_sdw); + dev_dbg(tasdevice->dev, "muted %lx %x\n", + SDW_SDCA_CTL(1, 2, 1, 0), ret); + ret = sdw_nwrite_no_pm(tasdevice->sdw_peripheral, + 0x40400108, 1, &value_sdw); + dev_dbg(tasdevice->dev, "muted %x\n", ret); + value_sdw = 0x1a; + ret = sdw_nwrite_no_pm(tasdevice->sdw_peripheral, + 0x8002, 1, &value_sdw); + dev_dbg(tasdevice->dev, "muted %x\n", ret); + } + + if (!tasdevice->sdw_slave) + return -EINVAL; + + return 0; +} + +static const struct snd_soc_dai_ops tasdevice_dai_ops = { + .startup = tasdevice_startup, + .mute_stream = tasdevice_mute, + .hw_params = tasdevice_sdw_hw_params, + .set_sysclk = tasdevice_set_dai_sysclk, + .hw_free = tasdevice_sdw_pcm_hw_free, + .set_stream = tasdevice_set_sdw_stream, + .shutdown = tasdevice_sdw_shutdown, +}; + +static struct snd_soc_dai_driver tasdevice_dai_driver[] = { + { + .name = "tas2783_codec", + .id = 0, + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 4, + .rates = TASDEVICE_RATES, + .formats = TASDEVICE_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 1, + .rates = TASDEVICE_RATES, + .formats = TASDEVICE_FORMATS, + }, + .ops = &tasdevice_dai_ops, + .symmetric_rate = 1, + }, +}; + +static void tas2783_reset(struct tasdevice_priv *tas_dev) +{ + int ret = 0; + unsigned char value_sdw; + + value_sdw = TAS2873_REG_SWRESET_RESET; + ret = sdw_nwrite_no_pm(tas_dev->sdw_peripheral, + TAS2873_REG_SWRESET, 1, &value_sdw); + dev_dbg(tas_dev->dev, "%s TAS2783 was reseted.\n", __func__); + usleep_range(1000, 1050); +} + +static int tasdevice_codec_probe(struct snd_soc_component *codec) +{ + struct tasdevice_priv *tas_dev = + snd_soc_component_get_drvdata(codec); + int ret = 0; + + dev_dbg(tas_dev->dev, "%s called for TAS2783 start.\n", + __func__); + /* Codec Lock Hold */ + mutex_lock(&tas_dev->codec_lock); + + crc8_populate_msb(tas_dev->crc8_lkp_tbl, + TASDEVICE_CRC8_POLYNOMIAL); + tas_dev->codec = codec; + + scnprintf(tas_dev->rca_binaryname, 64, "MY_SWFT_x%01x.bin", + tas_dev->sdw_peripheral->id.unique_id); + + ret = request_firmware_nowait(THIS_MODULE, FW_ACTION_UEVENT, + tas_dev->rca_binaryname, tas_dev->dev, GFP_KERNEL, + tas_dev, tasdevice_rca_ready); + dev_dbg(tas_dev->dev, + "%s: request_firmware %x open status: 0x%08x\n", + __func__, tas_dev->sdw_peripheral->id.unique_id, ret); + + /* Codec Lock Release*/ + mutex_unlock(&tas_dev->codec_lock); + + tas2783_reset(tas_dev); + + dev_dbg(tas_dev->dev, "%s was called end.\n", __func__); + return ret; +} + +static void tasdevice_deinit(void *context) +{ + struct tasdevice_priv *tas_dev = + (struct tasdevice_priv *) context; + + tasdevice_config_info_remove(tas_dev); +} + +static void tasdevice_codec_remove( + struct snd_soc_component *codec) +{ + struct tasdevice_priv *tas_dev = + snd_soc_component_get_drvdata(codec); + /* Codec Lock Hold */ + mutex_lock(&tas_dev->codec_lock); + tasdevice_deinit(tas_dev); + /* Codec Lock Release*/ + mutex_unlock(&tas_dev->codec_lock); +} + +static const struct snd_soc_component_driver + soc_codec_driver_tasdevice = { + .probe = tasdevice_codec_probe, + .remove = tasdevice_codec_remove, + .controls = tas2783_snd_controls, + .num_controls = ARRAY_SIZE(tas2783_snd_controls), + .dapm_widgets = tasdevice_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tasdevice_dapm_widgets), + .dapm_routes = tasdevice_audio_map, + .num_dapm_routes = ARRAY_SIZE(tasdevice_audio_map), + .idle_bias_on = 1, + .endianness = 1, +}; + +int tasdevice_dev_read(struct tasdevice_priv *tas_dev, unsigned int *reg, + unsigned int *val) +{ + int ret = 0; + + mutex_lock(&tas_dev->dev_lock); + ret = tas2783_sdw_read(tas_dev->sdw_peripheral, + reg, 4, (void *)val, 4); + dev_dbg(tas_dev->dev, "%s, read %x from reg %x. Result is (%d)\n", + __func__, *val, *reg, ret); + mutex_unlock(&tas_dev->dev_lock); + return ret; +} + +static int tasdevice_init(struct tasdevice_priv *tas_dev) +{ + int ret = 0; + + tas_dev->cur_prog = -1; + tas_dev->cur_conf = -1; + + dev_dbg(tas_dev->dev, "%s was called for TAS2783.\n", __func__); + mutex_init(&tas_dev->dev_lock); + + dev_set_drvdata(tas_dev->dev, tas_dev); + + mutex_init(&tas_dev->codec_lock); + ret = devm_snd_soc_register_component(tas_dev->dev, + &soc_codec_driver_tasdevice, + tasdevice_dai_driver, ARRAY_SIZE(tasdevice_dai_driver)); + if (ret) { + dev_err(tas_dev->dev, "%s: codec register error:0x%08x\n", + __func__, ret); + } + dev_dbg(tas_dev->dev, "%s was called for TAS2783.\n", __func__); + + return ret; +} + +static int tasdevice_read_prop(struct sdw_slave *slave) +{ + struct sdw_slave_prop *prop = &slave->prop; + int nval; + int i, j; + u32 bit; + unsigned long addr; + struct sdw_dpn_prop *dpn; + + prop->scp_int1_mask = + SDW_SCP_INT1_BUS_CLASH | SDW_SCP_INT1_PARITY; + prop->quirks = SDW_SLAVE_QUIRKS_INVALID_INITIAL_PARITY; + + prop->paging_support = true; + + /* first we need to allocate memory for set bits in port lists */ + prop->source_ports = 0x04; /* BITMAP: 00000100 */ + prop->sink_ports = 0x2; /* BITMAP: 00000010 */ + + nval = hweight32(prop->source_ports); + prop->src_dpn_prop = devm_kcalloc(&slave->dev, nval, + sizeof(*prop->src_dpn_prop), GFP_KERNEL); + if (!prop->src_dpn_prop) + return -ENOMEM; + + i = 0; + dpn = prop->src_dpn_prop; + addr = prop->source_ports; + for_each_set_bit(bit, &addr, 32) { + dpn[i].num = bit; + dpn[i].type = SDW_DPN_FULL; + dpn[i].simple_ch_prep_sm = true; + dpn[i].ch_prep_timeout = 10; + i++; + } + + /* do this again for sink now */ + nval = hweight32(prop->sink_ports); + prop->sink_dpn_prop = devm_kcalloc(&slave->dev, nval, + sizeof(*prop->sink_dpn_prop), GFP_KERNEL); + if (!prop->sink_dpn_prop) + return -ENOMEM; + + j = 0; + dpn = prop->sink_dpn_prop; + addr = prop->sink_ports; + for_each_set_bit(bit, &addr, 32) { + dpn[j].num = bit; + dpn[j].type = SDW_DPN_FULL; + dpn[j].simple_ch_prep_sm = true; + dpn[j].ch_prep_timeout = 10; + j++; + } + + /* set the timeout values */ + prop->clk_stop_timeout = 20; + + dev_dbg(&slave->dev, "%s was performed.\n", __func__); + + return 0; +} + +static int tasdevice_io_init(struct device *dev, struct sdw_slave *slave) +{ + struct tasdevice_priv *tasdevice = dev_get_drvdata(dev); + + if (tasdevice->hw_init) + return 0; + + if (tasdevice->first_hw_init) { + regcache_cache_only(tasdevice->regmap, false); + regcache_cache_bypass(tasdevice->regmap, true); + } else { + /* PM runtime is only enabled when + * a Slave reports as Attached + * set autosuspend parameters + */ + pm_runtime_set_autosuspend_delay(&slave->dev, 3000); + pm_runtime_use_autosuspend(&slave->dev); + + /* update count of parent 'active' children */ + pm_runtime_set_active(&slave->dev); + + /* make sure the device does not suspend immediately */ + pm_runtime_mark_last_busy(&slave->dev); + + pm_runtime_enable(&slave->dev); + } + + pm_runtime_get_noresume(&slave->dev); + + if (tasdevice->first_hw_init) { + regcache_cache_bypass(tasdevice->regmap, false); + regcache_mark_dirty(tasdevice->regmap); + } else + tasdevice->first_hw_init = true; + + /* Mark Slave initialization complete */ + tasdevice->hw_init = true; + + pm_runtime_mark_last_busy(&slave->dev); + pm_runtime_put_autosuspend(&slave->dev); + + dev_dbg(&slave->dev, "%s hw_init complete\n", __func__); + return 0; +} + +static int tasdevice_update_status(struct sdw_slave *slave, + enum sdw_slave_status status) +{ + struct tasdevice_priv *tasdevice = dev_get_drvdata(&slave->dev); + + /* Update the status */ + tasdevice->status = status; + + if (status == SDW_SLAVE_UNATTACHED) + tasdevice->hw_init = false; + + /* Perform initialization only if slave status + * is present and hw_init flag is false + */ + if (tasdevice->hw_init || tasdevice->status != SDW_SLAVE_ATTACHED) + return 0; + + /* perform I/O transfers required for Slave initialization */ + return tasdevice_io_init(&slave->dev, slave); +} + +/* + * slave_ops: callbacks for get_clock_stop_mode, clock_stop and + * port_prep are not defined for now + */ +static const struct sdw_slave_ops tasdevice_sdw_ops = { + .read_prop = tasdevice_read_prop, + .update_status = tasdevice_update_status, +}; + +static void tasdevice_remove(struct tasdevice_priv *tas_dev) +{ + snd_soc_unregister_component(tas_dev->dev); + if (gpio_is_valid(tas_dev->rst_gpio)) + gpio_free(tas_dev->rst_gpio); + + if (gpio_is_valid(tas_dev->irq_info.irq_gpio)) + gpio_free(tas_dev->irq_info.irq_gpio); + + mutex_destroy(&tas_dev->dev_lock); + mutex_destroy(&tas_dev->codec_lock); +} + +static int tasdevice_sdw_probe(struct sdw_slave *peripheral, + const struct sdw_device_id *id) +{ + struct device *dev = &peripheral->dev; + struct tasdevice_priv *tas_dev; + int ret = 0; + + dev_dbg(dev, "%s was called for TAS2783.\n", __func__); + + tas_dev = devm_kzalloc(dev, sizeof(*tas_dev), GFP_KERNEL); + if (!tas_dev) { + ret = -ENOMEM; + goto out; + } + tas_dev->dev = &peripheral->dev; + tas_dev->sdw_peripheral = peripheral; + tas_dev->chip_id = id->driver_data; + tas_dev->sdw_slave = peripheral; + tas_dev->hw_init = false; + tas_dev->first_hw_init = false; + + dev_dbg(dev, "%d chip id %x for TAS2783.\n", + peripheral->id.unique_id, tas_dev->chip_id); + + dev_set_drvdata(dev, tas_dev); + + tas_dev->regmap = devm_regmap_init(dev, &tas2783_regmap_bus_sdw, + peripheral, &tasdevice_regmap); + if (IS_ERR(tas_dev->regmap)) { + ret = PTR_ERR(tas_dev->regmap); + dev_err(dev, "Failed devm_regmap_init: %d\n", ret); + goto out; + } + ret = tasdevice_init(tas_dev); + +out: + if (ret < 0 && tas_dev != NULL) + tasdevice_remove(tas_dev); + + dev_dbg(dev, "%s was called for TAS2783.\n", __func__); + return ret; + +} + +static int tasdevice_sdw_remove(struct sdw_slave *peripheral) +{ + struct tasdevice_priv *tas_dev = + dev_get_drvdata(&peripheral->dev); + + if (tas_dev) + tasdevice_remove(tas_dev); + + return 0; +} + +static const struct sdw_device_id tasdevice_sdw_id[] = { + SDW_SLAVE_ENTRY(0x0102, 0x0, 0), + {}, +}; +MODULE_DEVICE_TABLE(sdw, tasdevice_sdw_id); + +static struct sdw_driver tasdevice_sdw_driver = { + .driver = { + .name = "slave-tas2783", + }, + .probe = tasdevice_sdw_probe, + .remove = tasdevice_sdw_remove, + .ops = &tasdevice_sdw_ops, + .id_table = tasdevice_sdw_id, +}; + +module_sdw_driver(tasdevice_sdw_driver); + +MODULE_AUTHOR("Baojun Xu <baojun.xu(a)ti.com>"); +MODULE_DESCRIPTION("ASoC TAS2783 SoundWire Driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tas2783.h b/sound/soc/codecs/tas2783.h new file mode 100644 index 000000000000..0fa5b3855b51 --- /dev/null +++ b/sound/soc/codecs/tas2783.h @@ -0,0 +1,178 @@ +/* SPDX-License-Identifier: GPL-2.0 + * + * ALSA SoC Texas Instruments TAS2781 Audio Smart Amplifier + * + * Copyright (C) 2023 Texas Instruments Incorporated + * https://www.ti.com + * + * The TAS2783 driver implements a flexible and configurable + * algo coff setting for single TAS2783 chips. + * + * Author: Baojun Xu <baojun.xu(a)ti.com> + */ + +#ifndef __TAS2783_H__ +#define __TAS2783_H__ + +#define TASDEVICE_RATES (SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | \ + SNDRV_PCM_RATE_96000 | \ + SNDRV_PCM_RATE_88200) +#define TAS2783_SDW_ADDR_OFFSET 0x8000 + +#define TASDEVICE_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +/* BOOK, PAGE Control Register */ +#define TASDEVICE_REG(book, page, reg) ((book * 256 * 256) + 0x8000 +\ + (page * 128) + reg) + +/*Software Reset */ +#define TAS2873_REG_SWRESET TASDEVICE_REG(0x0, 0X0, 0x01) +#define TAS2873_REG_SWRESET_RESET BIT(0) + +/* Volume control */ +#define TAS2783_DVC_LVL TASDEVICE_REG(0x0, 0x00, 0x1A) +#define TAS2783_AMP_LEVEL TASDEVICE_REG(0x0, 0x00, 0x03) +#define TAS2783_AMP_LEVEL_MASK GENMASK(5, 1) + +/* Calibration data */ +#define TAS2783_CALIBRATION_RE TASDEVICE_REG(0x0, 0x17, 0x74) +#define TAS2783_CALIBRATION_RE_LOW TASDEVICE_REG(0x0, 0x18, 0x14) +#define TAS2783_CALIBRATION_INV_RE TASDEVICE_REG(0x0, 0x18, 0x0c) +#define TAS2783_CALIBRATION_POW TASDEVICE_REG(0x0, 0x0d, 0x3c) +#define TAS2783_CALIBRATION_TLIMIT TASDEVICE_REG(0x0, 0x18, 0x7c) + +#define TAS2783_CALIBRATION_SIZE 5 // int data number. + +#define TAS2783_DEVICE_ID_START 0x08 // Unique id start +#define TAS2783_DEVICE_ID_END 0x0F // Unique id end + +/* TAS2783 SDCA Control - function number */ +#define FUNC_NUM_SMART_AMP 0x01 + +/* TAS2783 SDCA entity */ +#define TAS2783_SDCA_ENT_PDE23 0x0C +#define TAS2783_SDCA_ENT_PDE22 0x0B +#define TAS2783_SDCA_ENT_FU21 0x01 +#define TAS2783_SDCA_ENT_UDMPU21 0x10 + +/* TAS2783 SDCA control */ +#define TAS2783_SDCA_CTL_REQ_POWER_STATE 0x01 +#define TAS2783_SDCA_CTL_FU_MUTE 0x01 +#define TAS2783_SDCA_CTL_UDMPU_CLUSTER 0x10 + +#define CH_L 1 +#define CH_R 2 + +#define TASDEVICE_CRC8_POLYNOMIAL 0x4d + +#define MAX_NO_FILES 100 +#define MAX_CALIBRATION_DATA_SIZE 252 + +#define SMS_HTONL(a, b, c, d) ((((a)&0x000000FF)<<24) | \ + (((b)&0x000000FF)<<16) | \ + (((c)&0x000000FF)<<8) | \ + ((d)&0x000000FF)) + +enum channel { + top_left_Chn, + top_right_chn, + bottom_left_Chn, + bottom_right_chn, + max_chn, +}; + +struct tasdevice { + unsigned int dev_addr; + unsigned int err_code; + short cur_prog; + short cur_conf; + bool is_loading; + bool is_loaderr; + bool is_calibrated_data_loaded; + unsigned char cur_book; + struct tasdevice_fw *cali_data_fmw; +}; + +struct MYSWFTFile { + unsigned int m_vendor_id; + unsigned int m_file_id; + unsigned int m_version; + unsigned int m_Length; + unsigned int m_downloadAddr; + unsigned char *m_startAdress; +}; + +/* + * This item is used to store the generic i2c address of + * all the tas2781 devices for I2C broadcast during the multi-device + * writes, useless in mono case. + */ +struct global_addr { + unsigned char cur_book; + unsigned int dev_addr; + int ref_cnt; +}; + +struct calibration_data { + unsigned long total_sz; + unsigned char data[MAX_CALIBRATION_DATA_SIZE]; +}; + +struct tas_control { + struct snd_kcontrol_new *tasdevice_profile_controls; + int nr_controls; +}; + +struct tasdevice_irqinfo { + int irq_gpio; + int irq; +}; + +struct tasdevice_priv { + struct device *dev; + struct snd_soc_component *component; + struct sdw_slave *sdw_slave; + enum sdw_slave_status status; + struct sdw_bus_params params; + struct regmap *regmap; + struct mutex codec_lock; + struct mutex dev_lock; + int rst_gpio; + struct tasdevice tasdevice[max_chn]; + struct tasdevice_fw *fmw; + struct tasdevice_irqinfo irq_info; + struct tas_control tas_ctrl; + struct sdw_slave *sdw_peripheral; + struct global_addr glb_addr; + struct calibration_data cali_data; + struct tm tm; + int cur_prog; + int cur_conf; + unsigned int chip_id; + bool hw_init; + bool first_hw_init; + unsigned char *bq_params; + unsigned int magic_num; + unsigned char ndev; + unsigned char dev_name[32]; + unsigned char rca_binaryname[64]; + unsigned char coef_binaryname[64]; + unsigned char crc8_lkp_tbl[CRC8_TABLE_SIZE]; + + struct MYSWFTFile m_swftFile[MAX_NO_FILES]; + + void *codec; + unsigned int sysclk; + bool is_glb_calibrated_data_loaded; +}; + +struct sdw_stream_data { + struct sdw_stream_runtime *sdw_stream; +}; + +int tasdevice_dev_read(struct tasdevice_priv *tasdevice, + unsigned int *reg, unsigned int *value); + +#endif /*__TAS2783_H__ */ -- 2.34.1

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[PATCH v4 0/1] Intel: sof_rt5682: remove quirk flag
by Brent Lu 01 Aug '23

01 Aug '23

We add a helper function to detect amplifier number according to device instance in ACPI table so the SOF_MAX98390_TWEETER_SPEAKER_PRESENT flag and a dmi quirk for 4-amplifier configuration could be safely removed. Also refactor the max_98390_hw_params() function to use an array to handle the TDM parameter. Amplifier number detection and TDM parameter are tested on two Chromebooks. One with 2 MAX98390 and one with 4 MAX98390 amplifier. V2 Changes: - fix some typos in cover letter - use for_each_acpi_dev_match() macro to simplify the codec - use 'unsign int' for codec number variable V3 Changes: - replace pr_err() with dev_err() V4 Changes: - merge two patches to avoid build break - remove redundant assignment in max_98390_hw_params() - refactor MAX_98390_DEVX_NAME macros Brent Lu (2): ASoC: Intel: maxim-common: get codec number from ACPI ASoC: Intel: sof_rt5682: remove SOF_MAX98390_TWEETER_SPEAKER_PRESENT flag Brent Lu (1): ASoC: Intel: maxim-common: get codec number from ACPI table sound/soc/intel/boards/sof_maxim_common.c | 170 +++++++++++++--------- sound/soc/intel/boards/sof_maxim_common.h | 21 ++- sound/soc/intel/boards/sof_rt5682.c | 37 +---- 3 files changed, 111 insertions(+), 117 deletions(-) -- 2.34.1

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[PATCH 0/8] ASoC/SOF/Intel/AMD: cleanups for GCC11 -fanalyzer checks
by Pierre-Louis Bossart 01 Aug '23

01 Aug '23

GCC11 provides an '-fanalyzer' static analysis option which does not provide too many false-positives. This patch cleans-up known problematic code paths to help enable this capability in CI. We've used this for about a month already. Pierre-Louis Bossart (8): ASoC: SOF: sof-client-probes-ipc4: add checks to prevent static analysis warnings ASoC: SOF: ipc3: add checks to prevent static analysis warnings ASoC: SOF: topology: simplify code to prevent static analysis warnings ASoC: SOF: imx: remove error checks on NULL ipc ASoC: SOF: mediatek: remove error checks on NULL ipc ASoC: Intel: bdw_rt286: add checks to avoid static analysis warnings ASoC: Intel: atom: remove static analysis false positive ASoC: amd: acp5x-mach:add checks to avoid static analysis warnings include/linux/firmware/imx/dsp.h | 6 ------ include/linux/firmware/mediatek/mtk-adsp-ipc.h | 6 ------ sound/soc/amd/vangogh/acp5x-mach.c | 3 +++ sound/soc/intel/atom/sst/sst_stream.c | 13 +++++++------ sound/soc/intel/boards/bdw_rt286.c | 6 ++++++ sound/soc/sof/ipc3.c | 2 +- sound/soc/sof/sof-client-probes-ipc4.c | 9 +++++++++ sound/soc/sof/topology.c | 5 +++-- 8 files changed, 29 insertions(+), 21 deletions(-) -- 2.39.2

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[PATCH v4] um/drivers: fix hostaudio build errors
by Randy Dunlap 01 Aug '23

01 Aug '23

Use "select"s to ensure that the required kconfig symbols are set as expected. Drop HOSTAUDIO since it is now equivalent to UML_SOUND. Allow SOUND with UML regardless of HAS_IOMEM. Otherwise there is a kconfig warning for unmet dependencies. (This was not an issue when SOUND was defined in arch/um/drivers/Kconfig. I have done 50 randconfig builds and didn't find any issues.) This fixes build errors when CONFIG_SOUND is not set: ld: arch/um/drivers/hostaudio_kern.o: in function `hostaudio_cleanup_module': hostaudio_kern.c:(.exit.text+0xa): undefined reference to `unregister_sound_mixer' ld: hostaudio_kern.c:(.exit.text+0x15): undefined reference to `unregister_sound_dsp' ld: arch/um/drivers/hostaudio_kern.o: in function `hostaudio_init_module': hostaudio_kern.c:(.init.text+0x19): undefined reference to `register_sound_dsp' ld: hostaudio_kern.c:(.init.text+0x31): undefined reference to `register_sound_mixer' ld: hostaudio_kern.c:(.init.text+0x49): undefined reference to `unregister_sound_dsp' and this kconfig warning: WARNING: unmet direct dependencies detected for SOUND Fixes: 1da177e4c3f4 ("Linux-2.6.12-rc2") Fixes: d886e87cb82b ("sound: make OSS sound core optional") Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Reported-by: kernel test robot <lkp(a)intel.com> Closes: lore.kernel.org/r/202307141416.vxuRVpFv-lkp@intel.com Cc: Richard Weinberger <richard(a)nod.at> Cc: Anton Ivanov <anton.ivanov(a)cambridgegreys.com> Cc: Johannes Berg <johannes(a)sipsolutions.net> Cc: linux-um(a)lists.infradead.org Cc: Tejun Heo <tj(a)kernel.org> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Masahiro Yamada <masahiroy(a)kernel.org> Cc: Nathan Chancellor <nathan(a)kernel.org> Cc: Nick Desaulniers <ndesaulniers(a)google.com> Cc: Nicolas Schier <nicolas(a)fjasle.eu> Cc: linux-kbuild(a)vger.kernel.org Cc: alsa-devel(a)alsa-project.org --- v2: don't delete the HOSTAUDIO Kconfig entry (Masahiro) v3: drop HOSTAUDIO and use CONFIG_UML_SOUND for it in Makefile (Takashi); add SOUND depends on "|| UML" to HAS_IOMEM v4: use depends on instead of select for SOUND (Masahiro); use Closes: instead of Link: arch/um/drivers/Kconfig | 16 +++------------- arch/um/drivers/Makefile | 2 +- sound/Kconfig | 2 +- 3 files changed, 5 insertions(+), 15 deletions(-) diff -- a/arch/um/drivers/Kconfig b/arch/um/drivers/Kconfig --- a/arch/um/drivers/Kconfig +++ b/arch/um/drivers/Kconfig @@ -111,24 +111,14 @@ config SSL_CHAN config UML_SOUND tristate "Sound support" + depends on SOUND + select SOUND_OSS_CORE help This option enables UML sound support. If enabled, it will pull in - soundcore and the UML hostaudio relay, which acts as a intermediary + the UML hostaudio relay, which acts as a intermediary between the host's dsp and mixer devices and the UML sound system. It is safe to say 'Y' here. -config SOUND - tristate - default UML_SOUND - -config SOUND_OSS_CORE - bool - default UML_SOUND - -config HOSTAUDIO - tristate - default UML_SOUND - endmenu menu "UML Network Devices" diff -- a/sound/Kconfig b/sound/Kconfig --- a/sound/Kconfig +++ b/sound/Kconfig @@ -1,7 +1,7 @@ # SPDX-License-Identifier: GPL-2.0-only menuconfig SOUND tristate "Sound card support" - depends on HAS_IOMEM + depends on HAS_IOMEM || UML help If you have a sound card in your computer, i.e. if it can say more than an occasional beep, say Y. diff -- a/arch/um/drivers/Makefile b/arch/um/drivers/Makefile --- a/arch/um/drivers/Makefile +++ b/arch/um/drivers/Makefile @@ -54,7 +54,7 @@ obj-$(CONFIG_UML_NET) += net.o obj-$(CONFIG_MCONSOLE) += mconsole.o obj-$(CONFIG_MMAPPER) += mmapper_kern.o obj-$(CONFIG_BLK_DEV_UBD) += ubd.o -obj-$(CONFIG_HOSTAUDIO) += hostaudio.o +obj-$(CONFIG_UML_SOUND) += hostaudio.o obj-$(CONFIG_NULL_CHAN) += null.o obj-$(CONFIG_PORT_CHAN) += port.o obj-$(CONFIG_PTY_CHAN) += pty.o

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