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topology: pre-processor: Add support for object arrays
by GitHub pull_request - edited 24 May '23

24 May '23

alsa-project/alsa-utils pull request #216 was edited from ranj063: Introduce a new keyword, "ObjectArray" to instantiate multiple objects of the same class with different attirbutes. For example ``` ObjectArray.Base.input_audio_format [ { in_rate [ 8000 16000 48000 ] in_bit_depth [ 32 ] in_valid_bit_depth [ 24 32 ] } ] ``` This would be expanded into 6 objects with the rate, bit_depth and valid_bit_depth combinations of the arrays of values above. ``` Object.Base.input_audio_format [ { in_rate 8000 in_bit_depth 32 in_valid_bit_depth 32 } { in_rate 16000 in_bit_depth 32 in_valid_bit_depth 32 } { in_rate 48000 in_bit_depth 32 in_valid_bit_depth 32 } { in_rate 8000 in_bit_depth 32 in_valid_bit_depth 24 } { in_rate 16000 in_bit_depth 32 in_valid_bit_depth 24 } { in_rate 48000 in_bit_depth 32 in_valid_bit_depth 24 } ] ``` The Objects definition is an array so that multiple combinations can be specified so that we deal with only valid combinations. For example, 16-bit bit_depth is only valid with 16-bit valid_bit_depth. So if we also want to add those combinations with the multiple rates, the definition would look like: ``` ObjectArray.Base.input_audio_format [ { in_rate [ 8000 16000 48000 ] in_bit_depth [ 32 ] in_valid_bit_depth [ 24 32 ] } { in_rate [ 8000 16000 48000 ] in_bit_depth [ 16 ] in_valid_bit_depth [ 16 ] } ] ``` Request URL : https://github.com/alsa-project/alsa-utils/pull/216 Patch URL : https://github.com/alsa-project/alsa-utils/pull/216.patch Repository URL: https://github.com/alsa-project/alsa-utils

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unsuscribe
by Juan DCG 24 May '23

24 May '23

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[PATCH] ALSA: hda/realtek: Enable headset onLenovo M70/M90
by Bin Li 24 May '23

24 May '23

Lenovo M70/M90 Gen4 are equipped with ALC897, and they need ALC897_FIXUP_HEADSET_MIC_PIN quirk to make its headset mic work. The previous quirk for M70/M90 is for Gen3. Signed-off-by: Bin Li <bin.li(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a7e4765eff80..7b5f194513c7 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -11719,6 +11719,8 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x32cb, "Lenovo ThinkCentre M70", ALC897_FIXUP_HEADSET_MIC_PIN), SND_PCI_QUIRK(0x17aa, 0x32cf, "Lenovo ThinkCentre M950", ALC897_FIXUP_HEADSET_MIC_PIN), SND_PCI_QUIRK(0x17aa, 0x32f7, "Lenovo ThinkCentre M90", ALC897_FIXUP_HEADSET_MIC_PIN), + SND_PCI_QUIRK(0x17aa, 0x3321, "Lenovo ThinkCentre M70 Gen4", ALC897_FIXUP_HEADSET_MIC_PIN), + SND_PCI_QUIRK(0x17aa, 0x331b, "Lenovo ThinkCentre M90 Gen4", ALC897_FIXUP_HEADSET_MIC_PIN), SND_PCI_QUIRK(0x17aa, 0x3742, "Lenovo TianYi510Pro-14IOB", ALC897_FIXUP_HEADSET_MIC_PIN2), SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo Ideapad Y550P", ALC662_FIXUP_IDEAPAD), SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo Ideapad Y550", ALC662_FIXUP_IDEAPAD), -- 2.34.1

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[PATCH] regmap: sdw: check for invalid multi-register writes config
by Srinivas Kandagatla 24 May '23

24 May '23

SoundWire code as it is only supports Bulk register writes and it does not support multi-register writes. Any drivers that set can_multi_write and use regmap_multi_reg_write() will easily endup with programming the hardware incorrectly without any errors. So, add this check in bus code to be able to validate the drivers config. Fixes: 522272047dc6 ("regmap: sdw: Remove 8-bit value size restriction") Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- drivers/base/regmap/regmap-sdw.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/drivers/base/regmap/regmap-sdw.c b/drivers/base/regmap/regmap-sdw.c index 09899ae99fc1..159c0b740b00 100644 --- a/drivers/base/regmap/regmap-sdw.c +++ b/drivers/base/regmap/regmap-sdw.c @@ -59,6 +59,10 @@ static int regmap_sdw_config_check(const struct regmap_config *config) if (config->pad_bits != 0) return -ENOTSUPP; + /* Only bulk writes are supported not multi-register writes */ + if (config->can_multi_write) + return -ENOTSUPP; + return 0; } -- 2.21.0

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[PATCH] regmap: maple: Drop the RCU read lock while syncing registers
by Mark Brown 24 May '23

24 May '23

Unfortunately the maple tree requires us to explicitly lock it so we need to take the RCU read lock while iterating. When syncing this means that we end up trying to write out register values while holding the RCU read lock which triggers lockdep issues since that is an atomic context but most buses can't be used in atomic context. Pause the iteration and drop the lock for each register we check to avoid this. Reported-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- drivers/base/regmap/regcache-maple.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/drivers/base/regmap/regcache-maple.c b/drivers/base/regmap/regcache-maple.c index 9b1b559107ef..c2e3a0f6c218 100644 --- a/drivers/base/regmap/regcache-maple.c +++ b/drivers/base/regmap/regcache-maple.c @@ -203,15 +203,18 @@ static int regcache_maple_sync(struct regmap *map, unsigned int min, mas_for_each(&mas, entry, max) { for (r = max(mas.index, lmin); r <= min(mas.last, lmax); r++) { + mas_pause(&mas); + rcu_read_unlock(); ret = regcache_sync_val(map, r, entry[r - mas.index]); if (ret != 0) goto out; + rcu_read_lock(); } } -out: rcu_read_unlock(); +out: map->cache_bypass = false; return ret; --- base-commit: 44c026a73be8038f03dbdeef028b642880cf1511 change-id: 20230523-regcache-maple-sync-lock-57ea356dc60b Best regards, -- Mark Brown <broonie(a)kernel.org>

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[PATCH] ASoC: codecs: wcd938x-sdw: do not set can_multi_write flag
by Srinivas Kandagatla 24 May '23

24 May '23

regmap-sdw does not support multi register writes, so there is no point in setting this flag. This also leads to incorrect programming of WSA codecs with regmap_multi_reg_write() call. This invalid configuration should have been rejected by regmap-sdw. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/codecs/wcd938x-sdw.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/wcd938x-sdw.c b/sound/soc/codecs/wcd938x-sdw.c index 402286dfaea4..9c10200ff34b 100644 --- a/sound/soc/codecs/wcd938x-sdw.c +++ b/sound/soc/codecs/wcd938x-sdw.c @@ -1190,7 +1190,6 @@ static const struct regmap_config wcd938x_regmap_config = { .readable_reg = wcd938x_readable_register, .writeable_reg = wcd938x_writeable_register, .volatile_reg = wcd938x_volatile_register, - .can_multi_write = true, }; static const struct sdw_slave_ops wcd9380_slave_ops = { -- 2.21.0

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[PATCH 1/1] ASoC: dwc: move DMA init to snd_soc_dai_driver probe()
by Maxim Kochetkov 24 May '23

24 May '23

When using DMA mode we are facing with Oops: [ 396.458157] Unable to handle kernel access to user memory without uaccess routines at virtual address 000000000000000c [ 396.469374] Oops [#1] [ 396.471839] Modules linked in: [ 396.475144] CPU: 0 PID: 114 Comm: arecord Not tainted 6.0.0-00164-g9a8eccdaf2be-dirty #68 [ 396.483619] Hardware name: YMP ELCT FPGA (DT) [ 396.488156] epc : dmaengine_pcm_open+0x1d2/0x342 [ 396.493227] ra : dmaengine_pcm_open+0x1d2/0x342 [ 396.498140] epc : ffffffff807fe346 ra : ffffffff807fe346 sp : ffffffc804e138f0 [ 396.505602] gp : ffffffff817bf730 tp : ffffffd8042c8ac0 t0 : 6500000000000000 [ 396.513045] t1 : 0000000000000064 t2 : 656e69676e65616d s0 : ffffffc804e13990 [ 396.520477] s1 : ffffffd801b86a18 a0 : 0000000000000026 a1 : ffffffff816920f8 [ 396.527897] a2 : 0000000000000010 a3 : fffffffffffffffe a4 : 0000000000000000 [ 396.535319] a5 : 0000000000000000 a6 : ffffffd801b87040 a7 : 0000000000000038 [ 396.542740] s2 : ffffffd801b94a00 s3 : 0000000000000000 s4 : ffffffd80427f5e8 [ 396.550153] s5 : ffffffd80427f5e8 s6 : ffffffd801b44410 s7 : fffffffffffffff5 [ 396.557569] s8 : 0000000000000800 s9 : 0000000000000001 s10: ffffffff8066d254 [ 396.564978] s11: ffffffd8059cf768 t3 : ffffffff817d5577 t4 : ffffffff817d5577 [ 396.572391] t5 : ffffffff817d5578 t6 : ffffffc804e136e8 [ 396.577876] status: 0000000200000120 badaddr: 000000000000000c cause: 000000000000000d [ 396.586007] [<ffffffff806839f4>] snd_soc_component_open+0x1a/0x68 [ 396.592439] [<ffffffff807fdd62>] __soc_pcm_open+0xf0/0x502 [ 396.598217] [<ffffffff80685d86>] soc_pcm_open+0x2e/0x4e [ 396.603741] [<ffffffff8066cea4>] snd_pcm_open_substream+0x442/0x68e [ 396.610313] [<ffffffff8066d1ea>] snd_pcm_open+0xfa/0x212 [ 396.615868] [<ffffffff8066d39c>] snd_pcm_capture_open+0x3a/0x60 [ 396.622048] [<ffffffff8065b35a>] snd_open+0xa8/0x17a [ 396.627421] [<ffffffff801ae036>] chrdev_open+0xa0/0x218 [ 396.632893] [<ffffffff801a5a28>] do_dentry_open+0x17c/0x2a6 [ 396.638713] [<ffffffff801a6d9a>] vfs_open+0x1e/0x26 [ 396.643850] [<ffffffff801b8544>] path_openat+0x96e/0xc96 [ 396.649518] [<ffffffff801b9390>] do_filp_open+0x7c/0xf6 [ 396.655034] [<ffffffff801a6ff2>] do_sys_openat2+0x8a/0x11e [ 396.660765] [<ffffffff801a735a>] sys_openat+0x50/0x7c [ 396.666068] [<ffffffff80003aca>] ret_from_syscall+0x0/0x2 [ 396.674964] ---[ end trace 0000000000000000 ]--- It happens because of play_dma_data/capture_dma_data pointers are NULL. Current implementation assigns these pointers at snd_soc_dai_driver startup() callback and reset them back to NULL at shutdown(). But soc_pcm_open() sequence uses DMA pointers in dmaengine_pcm_open() before snd_soc_dai_driver startup(). Most generic DMA capable I2S drivers use snd_soc_dai_driver probe() callback to init DMA pointers only once at probe. So move DMA init to dw_i2s_dai_probe and drop shutdown() and startup() callbacks. Signed-off-by: Maxim Kochetkov <fido_max(a)inbox.ru> --- sound/soc/dwc/dwc-i2s.c | 41 +++++++++-------------------------------- 1 file changed, 9 insertions(+), 32 deletions(-) diff --git a/sound/soc/dwc/dwc-i2s.c b/sound/soc/dwc/dwc-i2s.c index dd5d8d77bdc9..9e7065dd854c 100644 --- a/sound/soc/dwc/dwc-i2s.c +++ b/sound/soc/dwc/dwc-i2s.c @@ -184,30 +184,6 @@ static void i2s_stop(struct dw_i2s_dev *dev, } } -static int dw_i2s_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *cpu_dai) -{ - struct dw_i2s_dev *dev = snd_soc_dai_get_drvdata(cpu_dai); - union dw_i2s_snd_dma_data *dma_data = NULL; - - if (!(dev->capability & DWC_I2S_RECORD) && - (substream->stream == SNDRV_PCM_STREAM_CAPTURE)) - return -EINVAL; - - if (!(dev->capability & DWC_I2S_PLAY) && - (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)) - return -EINVAL; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - dma_data = &dev->play_dma_data; - else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) - dma_data = &dev->capture_dma_data; - - snd_soc_dai_set_dma_data(cpu_dai, substream, (void *)dma_data); - - return 0; -} - static void dw_i2s_config(struct dw_i2s_dev *dev, int stream) { u32 ch_reg; @@ -306,12 +282,6 @@ static int dw_i2s_hw_params(struct snd_pcm_substream *substream, return 0; } -static void dw_i2s_shutdown(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - snd_soc_dai_set_dma_data(dai, substream, NULL); -} - static int dw_i2s_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { @@ -383,8 +353,6 @@ static int dw_i2s_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) } static const struct snd_soc_dai_ops dw_i2s_dai_ops = { - .startup = dw_i2s_startup, - .shutdown = dw_i2s_shutdown, .hw_params = dw_i2s_hw_params, .prepare = dw_i2s_prepare, .trigger = dw_i2s_trigger, @@ -626,6 +594,14 @@ static int dw_configure_dai_by_dt(struct dw_i2s_dev *dev, } +static int dw_i2s_dai_probe(struct snd_soc_dai *dai) +{ + struct dw_i2s_dev *dev = snd_soc_dai_get_drvdata(dai); + + snd_soc_dai_init_dma_data(dai, &dev->play_dma_data, &dev->capture_dma_data); + return 0; +} + static int dw_i2s_probe(struct platform_device *pdev) { const struct i2s_platform_data *pdata = pdev->dev.platform_data; @@ -644,6 +620,7 @@ static int dw_i2s_probe(struct platform_device *pdev) return -ENOMEM; dw_i2s_dai->ops = &dw_i2s_dai_ops; + dw_i2s_dai->probe = dw_i2s_dai_probe; dev->i2s_base = devm_platform_get_and_ioremap_resource(pdev, 0, &res); if (IS_ERR(dev->i2s_base)) -- 2.39.2

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[PATCH 0/3] ASoC: da7219: Add jack insertion detection polarity
by David Rau 24 May '23

24 May '23

This patchset adds the support of DA7219 Jack insertion detection polarity selection. The first patch replaces the old txt binding with a new schema binding. The second patch adds `dlg,jack-ins-det-pty` property for Jack insertion detection polarity selection. The last patch adds the driver support for this topic. The series has been verified on the DA7219 development kit. David Rau (3): ASoC: dt-bindings: dialog,da7219: convert to dtschema ASoC: dt-bindings: da7219: Add jack-ins-det-pty property ASoC: da7219: Add Jack insertion detection polarity .../devicetree/bindings/sound/da7219.txt | 112 --------- .../bindings/sound/dialog,da7219.yaml | 237 ++++++++++++++++++ include/sound/da7219-aad.h | 6 + sound/soc/codecs/da7219-aad.c | 42 +++- 4 files changed, 281 insertions(+), 116 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/da7219.txt create mode 100644 Documentation/devicetree/bindings/sound/dialog,da7219.yaml -- 2.17.1

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[GIT PULL] ASoC fixes for v6.4-rc3
by Mark Brown 24 May '23

24 May '23

The following changes since commit 9be0b3a0074a61df1c94c37faea35ec8b9ea130b: ASoC: SOF: Intel: hda-mlink: fixes and extensions (2023-05-16 00:13:09 +0900) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.4-rc3 for you to fetch changes up to 011a8719d6105dcb48077ea7a6a88ac019d4aa50: ASoC: dwc: move DMA init to snd_soc_dai_driver probe() (2023-05-23 21:39:30 +0100) ---------------------------------------------------------------- ASoC: Fixes for v6.4 A collection of fixes for v6.4, mostly driver specific but there's also one fix for DPCM to avoid incorrectly repeated calls to prepare() which can trigger issues on some systems. ---------------------------------------------------------------- Amadeusz Sławiński (4): ASoC: Intel: avs: Fix module lookup ASoC: Intel: avs: Access path components under lock ASoC: Intel: avs: Fix avs_path_module::instance_id size ASoC: Intel: avs: Add missing checks on FE startup Cezary Rojewski (3): ASoC: Intel: Skylake: Fix declaration of enum skl_ch_cfg ASoC: Intel: avs: Fix declaration of enum avs_channel_config ASoC: Intel: avs: Account for UID of ACPI device David Epping (1): ASoC: dt-bindings: tlv320aic32x4: Fix supply names Mark Brown (1): ASoC: Intel: Fixes Matthias Kaehlcke (1): ASoC: rt5682: Disable jack detection interrupt during suspend Maxim Kochetkov (1): ASoC: dwc: move DMA init to snd_soc_dai_driver probe() Ranjani Sridharan (1): ASoC: soc-pcm: test if a BE can be prepared Ravulapati Vishnu Vardhan Rao (1): ASoC: lpass: Fix for KASAN use_after_free out of bounds Stefan Binding (1): ASoC: cs35l41: Fix default regmap values for some registers .../devicetree/bindings/sound/tlv320aic32x4.txt | 2 +- include/sound/soc-acpi.h | 1 + include/sound/soc-dpcm.h | 4 +++ include/uapi/sound/skl-tplg-interface.h | 3 +- sound/soc/codecs/cs35l41-lib.c | 6 ++-- sound/soc/codecs/lpass-tx-macro.c | 5 +++ sound/soc/codecs/rt5682-i2c.c | 4 ++- sound/soc/codecs/rt5682.c | 6 ++++ sound/soc/codecs/rt5682.h | 1 + sound/soc/dwc/dwc-i2s.c | 41 +++++----------------- sound/soc/intel/avs/apl.c | 6 +++- sound/soc/intel/avs/avs.h | 4 +-- sound/soc/intel/avs/board_selection.c | 2 +- sound/soc/intel/avs/control.c | 22 ++++++++---- sound/soc/intel/avs/dsp.c | 4 +-- sound/soc/intel/avs/messages.h | 2 +- sound/soc/intel/avs/path.h | 2 +- sound/soc/intel/avs/pcm.c | 23 +++++++++--- sound/soc/intel/avs/probes.c | 2 +- sound/soc/soc-pcm.c | 20 +++++++++++ 20 files changed, 101 insertions(+), 59 deletions(-)

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[PATCH 0/4] ALSA: emu10k1: capture-related improvements
by Oswald Buddenhagen 24 May '23

24 May '23

Oswald Buddenhagen (4): ALSA: emu10k1: fix capture buffer size confusion ALSA: emu10k1: fix support for 24 kHz capture ALSA: emu10k1: don't restrict capture channel count to powers of two ALSA: emu10k1: fix multi-channel capture config for E-MU cards sound/pci/emu10k1/emupcm.c | 122 +++++++++++++++++++------------------ 1 file changed, 62 insertions(+), 60 deletions(-) -- 2.40.0.152.g15d061e6df

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[PATCH 0/3] ALSA: emu10k1: multi-channel playback related improvements
by Oswald Buddenhagen 24 May '23

24 May '23

Oswald Buddenhagen (3): ALSA: emu10k1: don't limit multi-channel playback to two periods ALSA: emu10k1: add synchronized start of multi-channel playback ALSA: emu10k1: make channel count of multi-channel playback flexible include/sound/emu10k1.h | 10 +++- sound/pci/emu10k1/emupcm.c | 115 +++++++++++++++++++++++++++++-------- sound/pci/emu10k1/io.c | 82 ++++++++++++++++++++++++++ 3 files changed, 181 insertions(+), 26 deletions(-) -- 2.40.0.152.g15d061e6df

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[PATCH v4 0/3] Add TDM audio on StarFive JH7110
by Walker Chen 24 May '23

24 May '23

This patchset adds TDM audio driver for the StarFive JH7110 SoC. The first patch adds device tree binding for TDM module. The second patch adds tdm driver support for JH7110 SoC. The last patch adds device tree node and pins configuration of tdm to JH7110 dts. The series has been tested on the VisionFive 2 board by plugging an audio expansion board. For more information of audio expansion board, you can take a look at the following webpage: https://wiki.seeedstudio.com/ReSpeaker_2_Mics_Pi_HAT/ Changes since v3: - Modified the subject of patch [1/3] to reflect which subsystem it is. - Use the set of functions with 'clk_bulk_' to get/enable/disable clocks. - Dropped the overlay from patch [3/3]. - Dropped the redundant macro 'CONFIG_PM' and 'CONFIG_PM_SLEEP' around suspend() and resume(). Changes since v2: - Use dt-overlay to describe sound card because need to plug the audio expansion board into the VisionFive2 board. - Modified the coding style for driver. - Moved assignment of stop_dma_first to startup function of dai_driver. - Dropped some useless macro definition. - Use loops to get/enable/disable clocks. Changes since v1: - Rebased on Linux 6.3-rc4. - Added the dts file dedicated to describe audio card. - Added the item for JH7110 audio board to the dt-binding of StarFive SoC-based boards. --- v3: https://lore.kernel.org/all/20230506090116.9206-1-walker.chen@starfivetech.… v2: https://lore.kernel.org/all/20230420024118.22677-1-walker.chen@starfivetech… v1: https://lore.kernel.org/all/20230329153320.31390-1-walker.chen@starfivetech… Walker Chen (3): ASoC: dt-bindings: Add TDM controller bindings for StarFive JH7110 ASoC: starfive: Add JH7110 TDM driver riscv: dts: starfive: add the node and pins configuration for tdm .../bindings/sound/starfive,jh7110-tdm.yaml | 98 ++++ MAINTAINERS | 6 + .../jh7110-starfive-visionfive-2.dtsi | 40 ++ arch/riscv/boot/dts/starfive/jh7110.dtsi | 21 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/starfive/Kconfig | 15 + sound/soc/starfive/Makefile | 2 + sound/soc/starfive/jh7110_tdm.c | 554 ++++++++++++++++++ sound/soc/starfive/jh7110_tdm.h | 138 +++++ 10 files changed, 876 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/starfive,jh7110-tdm.yaml create mode 100644 sound/soc/starfive/Kconfig create mode 100644 sound/soc/starfive/Makefile create mode 100644 sound/soc/starfive/jh7110_tdm.c create mode 100644 sound/soc/starfive/jh7110_tdm.h base-commit: 197b6b60ae7bc51dd0814953c562833143b292aa prerequisite-patch-id: 30bec4dba6f250a6edd0c2cbab2ce09442e50e8a prerequisite-patch-id: bb939c0c7c26b08addfccd890f9d3974b6eaec53 prerequisite-patch-id: 8a6f135bcabdad4a4bfb21f0c6a0ffd2bb57efe7 prerequisite-patch-id: c2366f993a9d85e28c06d8d09f064dd5e8b29a61 prerequisite-patch-id: 50d53a21f91f4087fc80b6f1f72864adfb0002b9 prerequisite-patch-id: 0df3703af91c30f1ca2c47f5609012f2d7200028 prerequisite-patch-id: 89f049f951e5acf75aab92541992f816fd0acc0d prerequisite-patch-id: 551fae54377090044c3612fca9740a9b359abdd2 prerequisite-patch-id: c7fdf904f398d478f0ed6d57eb878982bc73329d prerequisite-patch-id: 1b2d0982b18da060c82134f05bf3ce16425bac8d prerequisite-patch-id: 090ba4b78d47bc19204916e76fdbc70021785388 prerequisite-patch-id: a5d9e0f7d4f8163f566678894cf693015119f2d9 prerequisite-patch-id: 4637a8fa2334a45fa6b64351f4e9e28d3e2d60d3 prerequisite-patch-id: 32647ec60a3b614e1c59ec8e54cb511ae832c22f prerequisite-patch-id: aa06658ecf89c92d0dfdd6a4ba6d9e6e67532971 prerequisite-patch-id: 1387a7e87b446329dfc21f3e575ceae7ebcf954c prerequisite-patch-id: 258ea5f9b8bf41b6981345dcc81795f25865d38f prerequisite-patch-id: 8b6f2c9660c0ac0ee4e73e4c21aca8e6b75e81b9 prerequisite-patch-id: dbb0c0151b8bdf093e6ce79fd2fe3f60791a6e0b prerequisite-patch-id: 9007c8610fdcd387592475949864edde874c20a2 prerequisite-patch-id: d57e95d31686772abc4c4d5aa1cadc344dc293cd prerequisite-patch-id: 9f911969d0a550648493952c99096d26e05d4d83 prerequisite-patch-id: 2ddada18ab6ea5cd1da14212aaf59632f5203d40 prerequisite-patch-id: 80042661ff6156ce577a72e9eb8c0b218b624829 prerequisite-patch-id: 398744c61913c76a35754de867c4f820ca7a8d99 prerequisite-patch-id: f59269382164b5d642a5e10443ca447f5caa595c prerequisite-patch-id: 1babe83d6bf999bad17584dc595480f9070a5369 prerequisite-patch-id: 77be3d122d66df813f13088141ce27b21107a341 prerequisite-patch-id: 9fbb7ad1dd258bb8ff5946c4a0e59de4bfd82a04 prerequisite-patch-id: 6f6984916dffd0cc66aa733c9b6bd3a55495a50c prerequisite-patch-id: 39e1be2a3d1593577ab997f55f59367cba665aa7 prerequisite-patch-id: 584c256c9acb52ee2773d0c81c3f4977fc18155a prerequisite-patch-id: b37ac15032973e1fcd918f157c82a0606775c9e9 prerequisite-patch-id: 32deea16304859842af5c2151bc41d91cf6dfc9b prerequisite-patch-id: 20ac2450fb93b3f69f83fc720fd4800a95e618a6 prerequisite-patch-id: 6abf359fa445f4104432ddee27044dfbfb128417 -- 2.17.1

4 12

ucm2: Rockchip: Add UCM support for ES8316 on Rock 5B
by GitHub pull_request - opened 24 May '23

24 May '23

alsa-project/alsa-ucm-conf pull request #316 was opened from cristicc: Add UCM configuration for the RK3588 SoC based Rock 5B board to enable the analog audio support provided by the Everest Semi ES8316 codec. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/316 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/316.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH RFC] ALSA: control: Avoid nested locks at notification
by Takashi Iwai 24 May '23

24 May '23

The new control layer stuff introduced the nested rwsem for managing the list of registered control layer ops. Since then, a global snd_ctl_layer_rwsem is always read at each time a control notification is sent via snd_ctl_notify*() in a nested matter inside the card's controls_rwsem lock. This also required a bit complicated way of the lock at snd_ctl_activate_id() and snd_ctl_elem_write() with the downgrade of rwsem. This patch is an attempt to simplify the handling of ctl layer ops. Now, instead of traversing the global linked list, we keep a local list of lops in each card instance. This reduces the need of the global snd_ctl_layer_rwsem lock at snd_ctl_notify*() invocation. And, since the nested lock is avoided in most places, we can also avoid the rwsem downgrade hack in the above, too. Since the local list entry is created dynamically, snd_ctl_register_layer() may return an error, and the caller needs to check the return value. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- I noticed the nested lock while looking at the pending bug report ("USB sound card freezes USB after resume from suspend" -- which hasn't been resolved yet). Only very lightly tested. include/sound/control.h | 2 +- include/sound/core.h | 9 +++ sound/core/control.c | 138 +++++++++++++++++++++++++++------------ sound/core/control_led.c | 3 +- sound/core/init.c | 1 + 5 files changed, 108 insertions(+), 45 deletions(-) diff --git a/include/sound/control.h b/include/sound/control.h index cc3dcc6cfb0f..c40c693853e6 100644 --- a/include/sound/control.h +++ b/include/sound/control.h @@ -156,7 +156,7 @@ int snd_ctl_unregister_ioctl_compat(snd_kctl_ioctl_func_t fcn); #endif int snd_ctl_request_layer(const char *module_name); -void snd_ctl_register_layer(struct snd_ctl_layer_ops *lops); +int snd_ctl_register_layer(struct snd_ctl_layer_ops *lops); void snd_ctl_disconnect_layer(struct snd_ctl_layer_ops *lops); int snd_ctl_get_preferred_subdevice(struct snd_card *card, int type); diff --git a/include/sound/core.h b/include/sound/core.h index 4ea5f66b59d7..3e49fb483340 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -75,6 +75,14 @@ struct snd_device { #define snd_device(n) list_entry(n, struct snd_device, list) +/* control layer ops */ +struct snd_ctl_layer_ops; + +struct snd_ctl_lops_list { + struct snd_ctl_layer_ops *lops; + struct list_head list; +}; + /* main structure for soundcard */ struct snd_card { @@ -104,6 +112,7 @@ struct snd_card { size_t user_ctl_alloc_size; // current memory allocation by user controls. struct list_head controls; /* all controls for this card */ struct list_head ctl_files; /* active control files */ + struct list_head lops_list; /* list of associated control layer ops */ #ifdef CONFIG_SND_CTL_FAST_LOOKUP struct xarray ctl_numids; /* hash table for numids */ struct xarray ctl_hash; /* hash table for ctl id matching */ diff --git a/sound/core/control.c b/sound/core/control.c index 82aa1af1d1d8..70515521fc49 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -202,15 +202,13 @@ void snd_ctl_notify_one(struct snd_card *card, unsigned int mask, struct snd_kcontrol *kctl, unsigned int ioff) { struct snd_ctl_elem_id id = kctl->id; - struct snd_ctl_layer_ops *lops; + struct snd_ctl_lops_list *llist; id.index += ioff; id.numid += ioff; snd_ctl_notify(card, mask, &id); - down_read(&snd_ctl_layer_rwsem); - for (lops = snd_ctl_layer; lops; lops = lops->next) - lops->lnotify(card, mask, kctl, ioff); - up_read(&snd_ctl_layer_rwsem); + list_for_each_entry(llist, &card->lops_list, list) + llist->lops->lnotify(card, mask, kctl, ioff); } EXPORT_SYMBOL(snd_ctl_notify_one); @@ -710,11 +708,8 @@ int snd_ctl_activate_id(struct snd_card *card, struct snd_ctl_elem_id *id, vd->access |= SNDRV_CTL_ELEM_ACCESS_INACTIVE; } snd_ctl_build_ioff(id, kctl, index_offset); - downgrade_write(&card->controls_rwsem); snd_ctl_notify_one(card, SNDRV_CTL_EVENT_MASK_INFO, kctl, index_offset); - up_read(&card->controls_rwsem); - return 1; - + ret = 1; unlock: up_write(&card->controls_rwsem); return ret; @@ -1283,16 +1278,16 @@ static int snd_ctl_elem_write(struct snd_card *card, struct snd_ctl_file *file, down_write(&card->controls_rwsem); kctl = snd_ctl_find_id(card, &control->id); if (kctl == NULL) { - up_write(&card->controls_rwsem); - return -ENOENT; + result = -ENOENT; + goto unlock; } index_offset = snd_ctl_get_ioff(kctl, &control->id); vd = &kctl->vd[index_offset]; if (!(vd->access & SNDRV_CTL_ELEM_ACCESS_WRITE) || kctl->put == NULL || (file && vd->owner && vd->owner != file)) { - up_write(&card->controls_rwsem); - return -EPERM; + result = -EPERM; + goto unlock; } snd_ctl_build_ioff(&control->id, kctl, index_offset); @@ -1311,20 +1306,15 @@ static int snd_ctl_elem_write(struct snd_card *card, struct snd_ctl_file *file, if (!result) result = kctl->put(kctl, control); snd_power_unref(card); - if (result < 0) { - up_write(&card->controls_rwsem); - return result; - } + if (result < 0) + goto unlock; - if (result > 0) { - downgrade_write(&card->controls_rwsem); + if (result > 0) snd_ctl_notify_one(card, SNDRV_CTL_EVENT_MASK_VALUE, kctl, index_offset); - up_read(&card->controls_rwsem); - } else { - up_write(&card->controls_rwsem); - } - - return 0; + result = 0; + unlock: + up_write(&card->controls_rwsem); + return result; } static int snd_ctl_elem_write_user(struct snd_ctl_file *file, @@ -2225,6 +2215,58 @@ int snd_ctl_request_layer(const char *module_name) } EXPORT_SYMBOL_GPL(snd_ctl_request_layer); +/* register a control layer to the given card */ +static int snd_ctl_card_register_lops(struct snd_card *card, + struct snd_ctl_layer_ops *lops) +{ + struct snd_ctl_lops_list *llist; + + llist = kmalloc(sizeof(*llist), GFP_KERNEL); + if (!llist) + return -ENOMEM; + llist->lops = lops; + down_read(&card->controls_rwsem); + list_add_tail(&llist->list, &card->lops_list); + lops->lregister(card); + up_read(&card->controls_rwsem); + return 0; +} + +static void delete_lops_list(struct snd_card *card, + struct snd_ctl_lops_list *llist) +{ + list_del(&llist->list); + llist->lops->ldisconnect(card); + kfree(llist); +} + +/* disconnect a control layer from the given card */ +static void snd_ctl_card_disconnect_lops(struct snd_card *card, + struct snd_ctl_layer_ops *lops) +{ + struct snd_ctl_lops_list *llist; + + down_read(&card->controls_rwsem); + list_for_each_entry(llist, &card->lops_list, list) { + if (llist->lops == lops) { + delete_lops_list(card, llist); + break; + } + } + up_read(&card->controls_rwsem); +} + +/* disconnect all control layers from the given card */ +static void snd_ctl_card_disconnect_all_lops(struct snd_card *card) +{ + struct snd_ctl_lops_list *llist, *next; + + down_read(&card->controls_rwsem); + list_for_each_entry_safe(llist, next, &card->lops_list, list) + delete_lops_list(card, llist); + up_read(&card->controls_rwsem); +} + /** * snd_ctl_register_layer - register new control layer * @lops: operation structure @@ -2232,10 +2274,10 @@ EXPORT_SYMBOL_GPL(snd_ctl_request_layer); * The new layer can track all control elements and do additional * operations on top (like audio LED handling). */ -void snd_ctl_register_layer(struct snd_ctl_layer_ops *lops) +int snd_ctl_register_layer(struct snd_ctl_layer_ops *lops) { struct snd_card *card; - int card_number; + int ret, card_number; down_write(&snd_ctl_layer_rwsem); lops->next = snd_ctl_layer; @@ -2244,12 +2286,17 @@ void snd_ctl_register_layer(struct snd_ctl_layer_ops *lops) for (card_number = 0; card_number < SNDRV_CARDS; card_number++) { card = snd_card_ref(card_number); if (card) { - down_read(&card->controls_rwsem); - lops->lregister(card); - up_read(&card->controls_rwsem); + ret = snd_ctl_card_register_lops(card, lops); + if (ret < 0) + goto error; snd_card_unref(card); } } + return 0; + + error: + snd_ctl_disconnect_layer(lops); + return ret; } EXPORT_SYMBOL_GPL(snd_ctl_register_layer); @@ -2264,6 +2311,16 @@ EXPORT_SYMBOL_GPL(snd_ctl_register_layer); void snd_ctl_disconnect_layer(struct snd_ctl_layer_ops *lops) { struct snd_ctl_layer_ops *lops2, *prev_lops2; + struct snd_card *card; + int card_number; + + for (card_number = 0; card_number < SNDRV_CARDS; card_number++) { + card = snd_card_ref(card_number); + if (card) { + snd_ctl_card_disconnect_lops(card, lops); + snd_card_unref(card); + } + } down_write(&snd_ctl_layer_rwsem); for (lops2 = snd_ctl_layer, prev_lops2 = NULL; lops2; lops2 = lops2->next) { @@ -2310,13 +2367,16 @@ static int snd_ctl_dev_register(struct snd_device *device) &snd_ctl_f_ops, card, &card->ctl_dev); if (err < 0) return err; - down_read(&card->controls_rwsem); down_read(&snd_ctl_layer_rwsem); - for (lops = snd_ctl_layer; lops; lops = lops->next) - lops->lregister(card); + for (lops = snd_ctl_layer; lops; lops = lops->next) { + err = snd_ctl_card_register_lops(card, lops); + if (err < 0) { + snd_ctl_card_disconnect_all_lops(card); + break; + } + } up_read(&snd_ctl_layer_rwsem); - up_read(&card->controls_rwsem); - return 0; + return err; } /* @@ -2326,7 +2386,6 @@ static int snd_ctl_dev_disconnect(struct snd_device *device) { struct snd_card *card = device->device_data; struct snd_ctl_file *ctl; - struct snd_ctl_layer_ops *lops; unsigned long flags; read_lock_irqsave(&card->ctl_files_rwlock, flags); @@ -2336,12 +2395,7 @@ static int snd_ctl_dev_disconnect(struct snd_device *device) } read_unlock_irqrestore(&card->ctl_files_rwlock, flags); - down_read(&card->controls_rwsem); - down_read(&snd_ctl_layer_rwsem); - for (lops = snd_ctl_layer; lops; lops = lops->next) - lops->ldisconnect(card); - up_read(&snd_ctl_layer_rwsem); - up_read(&card->controls_rwsem); + snd_ctl_card_disconnect_all_lops(card); return snd_unregister_device(&card->ctl_dev); } diff --git a/sound/core/control_led.c b/sound/core/control_led.c index 3cadd40100f3..18b366207a9c 100644 --- a/sound/core/control_led.c +++ b/sound/core/control_led.c @@ -762,8 +762,7 @@ static int __init snd_ctl_led_init(void) return -ENOMEM; } } - snd_ctl_register_layer(&snd_ctl_led_lops); - return 0; + return snd_ctl_register_layer(&snd_ctl_led_lops); } static void __exit snd_ctl_led_exit(void) diff --git a/sound/core/init.c b/sound/core/init.c index df0c22480375..34bda15ad9ef 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -314,6 +314,7 @@ static int snd_card_init(struct snd_card *card, struct device *parent, rwlock_init(&card->ctl_files_rwlock); INIT_LIST_HEAD(&card->controls); INIT_LIST_HEAD(&card->ctl_files); + INIT_LIST_HEAD(&card->lops_list); #ifdef CONFIG_SND_CTL_FAST_LOOKUP xa_init(&card->ctl_numids); xa_init(&card->ctl_hash); -- 2.35.3

2 2

topology: pre-processor: Add support for object arrays
by GitHub pull_request - edited 24 May '23

24 May '23

alsa-project/alsa-utils pull request #216 was edited from ranj063: Introduce a new keyword, "Objects" to instantiate multiple objects of the same class with different attirbutes. For example ``` Objects.Base.input_audio_format [ { in_rate [ 8000 16000 48000 ] in_bit_depth [ 32 ] in_valid_bit_depth [ 24 32 ] } ] ``` This would be expanded into 6 objects with the rate, bit_depth and valid_bit_depth combinations of the arrays of values above. ``` Object.Base.input_audio_format [ { in_rate 8000 in_bit_depth 32 in_valid_bit_depth 32 } { in_rate 16000 in_bit_depth 32 in_valid_bit_depth 32 } { in_rate 48000 in_bit_depth 32 in_valid_bit_depth 32 } { in_rate 8000 in_bit_depth 32 in_valid_bit_depth 24 } { in_rate 16000 in_bit_depth 32 in_valid_bit_depth 24 } { in_rate 48000 in_bit_depth 32 in_valid_bit_depth 24 } ] ``` The Objects definition is an array so that multiple combinations can be specified so that we deal with only valid combinations. For example, 16-bit bit_depth is only valid with 16-bit valid_bit_depth. So if we also want to add those combinations with the multiple rates, the definition would look like: ``` Objects.Base.input_audio_format [ { in_rate [ 8000 16000 48000 ] in_bit_depth [ 32 ] in_valid_bit_depth [ 24 32 ] } { in_rate [ 8000 16000 48000 ] in_bit_depth [ 16 ] in_valid_bit_depth [ 16 ] } ] ``` Request URL : https://github.com/alsa-project/alsa-utils/pull/216 Patch URL : https://github.com/alsa-project/alsa-utils/pull/216.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH v2 1/3] docs: sound: add 'pcmtest' driver documentation
by Ivan Orlov 24 May '23

24 May '23

Add documentation for the new Virtual PCM Test Driver. It covers all possible usage cases: errors and delay injections, random and pattern-based data generation, playback and ioctl redefinition functionalities testing. We have a lot of different virtual media drivers, which can be used for testing of the userspace applications and media subsystem middle layer. However, all of them are aimed at testing the video functionality and simulating the video devices. For audio devices we have only snd-dummy module, which is good in simulating the correct behavior of an ALSA device. I decided to write a tool, which would help to test the userspace ALSA programs (and the PCM middle layer as well) under unusual circumstances to figure out how they would behave. So I came up with this Virtual PCM Test Driver. This new Virtual PCM Test Driver has several features which can be useful during the userspace ALSA applications testing/fuzzing, or testing/fuzzing of the PCM middle layer. Not all of them can be implemented using the existing virtual drivers (like dummy or loopback). Here is what can this driver do: - Simulate both capture and playback processes - Check the playback stream for containing the looped pattern - Generate random or pattern-based capture data - Inject delays into the playback and capturing processes - Inject errors during the PCM callbacks Also, this driver can check the playback stream for containing the predefined pattern, which is used in the corresponding selftest to check the PCM middle layer data transferring functionality. Additionally, this driver redefines the default RESET ioctl, and the selftest covers this PCM API functionality as well. Signed-off-by: Ivan Orlov <ivan.orlov0322(a)gmail.com> --- V1 -> V2: - Rename the driver from from 'valsa' to 'pcmtest'. - Implement support for interleaved and non-interleaved access modes - Add support for 8 substreams and 4 channels - Extend supported formats - Extend and rewrite in C the selftest for the driver Documentation/sound/cards/index.rst | 1 + Documentation/sound/cards/pcmtest.rst | 119 ++++++++++++++++++++++++++ 2 files changed, 120 insertions(+) create mode 100644 Documentation/sound/cards/pcmtest.rst diff --git a/Documentation/sound/cards/index.rst b/Documentation/sound/cards/index.rst index c016f8c3b88b..49c1f2f688f8 100644 --- a/Documentation/sound/cards/index.rst +++ b/Documentation/sound/cards/index.rst @@ -17,3 +17,4 @@ Card-Specific Information hdspm serial-u16550 img-spdif-in + pcmtest diff --git a/Documentation/sound/cards/pcmtest.rst b/Documentation/sound/cards/pcmtest.rst new file mode 100644 index 000000000000..ea8070eaa44e --- /dev/null +++ b/Documentation/sound/cards/pcmtest.rst @@ -0,0 +1,119 @@ +.. SPDX-License-Identifier: GPL-2.0 + +The Virtual PCM Test Driver +=========================== + +The Virtual PCM Test Driver emulates a generic PCM device, and can be used for +testing/fuzzing of the userspace ALSA applications, as well as for testing/fuzzing of +the PCM middle layer. Additionally, it can be used for simulating hard to reproduce +problems with PCM devices. + +What can this driver do? +~~~~~~~~~~~~~~~~~~~~~~~~ + +At this moment the driver can do the following things: + * Simulate both capture and playback processes + * Generate random or pattern-based capturing data + * Inject delays into the playback and capturing processes + * Inject errors during the PCM callbacks + +It supports up to 8 substreams and 4 channels. Also it supports both interleaved and +non-interleaved access modes. + +Also, this driver can check the playback stream for containing the predefined pattern, +which is used in the corresponding selftest (alsa/test-pcmtest-driver.c). To check the +PCM middle layer data transferring functionality. Additionally, this driver redefines +the default RESET ioctl, and the selftest covers this PCM API functionality as well. + +Configuration +------------- + +The driver has several parameters besides the common ALSA module parameters: + + * fill_mode (bool) - Buffer fill mode (see below) + * inject_delay (int) + * inject_hwpars_err (bool) + * inject_prepare_err (bool) + * inject_trigger_err (bool) + + +Capture Data Generation +----------------------- + +The driver has two modes of data generation: the first (0 in the fill_mode parameter) +means random data generation, the second (1 in the fill_mode) - pattern-based +data generation. Let's look at the second mode. + +First of all, you may want to specify the pattern for data generation. You can do it +by writing the pattern to the debugfs file (/sys/kernel/debug/pcmtest/fill_pattern). +Like that: + +.. code-block:: bash + + echo -n mycoolpattern > /sys/kernel/debug/pcmtest/fill_pattern + +After this, every capture action performed on the 'pcmtest' device will return +'mycoolpatternmycoolpatternmycoolpatternmy...' for every channel buffer. + +In case of interleaved access, the capture buffer will contain the repeated pattern +for every channel. Otherwise, every channel buffer will contain the repeated pattern. + +The pattern itself can be up to 4096 bytes long. + +Delay injection +--------------- + +The driver has 'inject_delay' parameter, which has very self-descriptive name and +can be used for time delay/speedup simulations. The parameter has integer type, and +it means the delay added between module's internal timer ticks. + +If the 'inject_delay' value is positive, the buffer will be filled slower, if it is +negative - faster. You can try it yourself by starting a recording in any +audio recording application (like Audacity) and selecting the 'pcmtest' device as a +source. + +This parameter can be also used for generating a huge amount of sound data in a very +short period of time (with the negative 'inject_delay' value). + +Errors injection +---------------- + +This module can be used for injecting errors into the PCM communication process. This +action can help you to figure out how the userspace ALSA program behaves under unusual +circumstances. + +For example, you can make all 'hw_params' PCM callback calls return EBUSY error by +writing '1' to the 'inject_hwpars_err' module parameter: + +.. code-block:: bash + + echo 1 > /sys/module/snd_pcmtest/parameters/inject_hwpars_err + +Errors can be injected into the following PCM callbacks: + + * hw_params (EBUSY) + * prepare (EINVAL) + * trigger (EINVAL) + + +Playback test +------------- + +This driver can be also used for the playback functionality testing - every time you +write the playback data to the 'pcmtest' PCM device and close it, the driver checks the +buffer of each channel for containing the looped pattern (which is specified in the +fill_pattern debugfs file). If the playback buffer content represents the looped pattern, +'pc_test' debugfs entry is set into '1'. Otherwise, the driver sets it to '0'. + +ioctl redefinition test +----------------------- + +The driver redefines the 'reset' ioctl, which is default for all PCM devices. To test +this functionality, we can trigger the reset ioctl and check the 'ioctl_test' debugfs +entry: + +.. code-block:: bash + + cat /sys/kernel/debug/pcmtest/ioctl_test + +If the ioctl is triggered successfully, this file will contain '1', and '0' otherwise. -- 2.34.1

2 4

topology: pre-processor: Add support for object arrays
by GitHub pull_request - edited 24 May '23

24 May '23

alsa-project/alsa-utils pull request #216 was edited from ranj063: Introduce a new keyword, "Objects" to instantiate multiple objects of the same class with different attirbutes. For example Objects.Base.input_audio_format [ { in_rate [ 8000 16000 48000 ] in_bit_depth [ 32 ] in_valid_bit_depth [ 24 32 ] } ] This would be expanded into 6 objects with the rate, bit_depth and valid_bit_depth combinations of the arrays of values above. Object.Base.input_audio_format [ { in_rate 8000 in_bit_depth 32 in_valid_bit_depth 32 } { in_rate 16000 in_bit_depth 32 in_valid_bit_depth 32 } { in_rate 48000 in_bit_depth 32 in_valid_bit_depth 32 } { in_rate 8000 in_bit_depth 32 in_valid_bit_depth 24 } { in_rate 16000 in_bit_depth 32 in_valid_bit_depth 24 } { in_rate 48000 in_bit_depth 32 in_valid_bit_depth 24 } ] The Objects definition is an array so that multiple combinations can be specified so that we deal with only valid combinations. For example, 16-bit bit_depth is only valid with 16-bit valid_bit_depth. So if we also want to add those combinations with the multiple rates, the definition would look like: Objects.Base.input_audio_format [ { in_rate [ 8000 16000 48000 ] in_bit_depth [ 32 ] in_valid_bit_depth [ 24 32 ] } { in_rate [ 8000 16000 48000 ] in_bit_depth [ 16 ] in_valid_bit_depth [ 16 ] } ] Request URL : https://github.com/alsa-project/alsa-utils/pull/216 Patch URL : https://github.com/alsa-project/alsa-utils/pull/216.patch Repository URL: https://github.com/alsa-project/alsa-utils

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tooplogy: pre-processor: Add support for object arrays
by GitHub pull_request - opened 24 May '23

24 May '23

alsa-project/alsa-utils pull request #216 was opened from ranj063: Introduce a new keyword, "Objects" to instantiate multiple objects of the same class with different attirbutes. For example Objects.Base.input_audio_format [ { in_rate [ 8000 16000 48000 ] in_bit_depth [ 32 ] in_valid_bit_depth [ 24 32 ] } ] This would be expanded into 6 objects with the rate, bit_depth and valid_bit_depth combinations of the arrays of values above. Object.Base.input_audio_format [ { in_rate 8000 in_bit_depth 32 in_valid_bit_depth 32 } { in_rate 16000 in_bit_depth 32 in_valid_bit_depth 32 } { in_rate 48000 in_bit_depth 32 in_valid_bit_depth 32 } { in_rate 8000 in_bit_depth 32 in_valid_bit_depth 24 } { in_rate 16000 in_bit_depth 32 in_valid_bit_depth 24 } { in_rate 48000 in_bit_depth 32 in_valid_bit_depth 24 } ] The Objects definition is an array so that multiple combinations can be specified so that we deal with only valid combinations. For example, 16-bit bit_depth is only valid with 16-bit valid_bit_depth. So if we also want to add those combinations with the multiple rates, the definition would look like: Objects.Base.input_audio_format [ { in_rate [ 8000 16000 48000 ] in_bit_depth [ 32 ] in_valid_bit_depth [ 24 32 ] } { in_rate [ 8000 16000 48000 ] in_bit_depth [ 16 ] in_valid_bit_depth [ 16 ] } ] Request URL : https://github.com/alsa-project/alsa-utils/pull/216 Patch URL : https://github.com/alsa-project/alsa-utils/pull/216.patch Repository URL: https://github.com/alsa-project/alsa-utils

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[PATCH 0/2] Improve support for sof_ipc{3|4}_bytes_ext_put
by Daniel Baluta 24 May '23

24 May '23

From: Daniel Baluta <daniel.baluta(a)nxp.com> This patch series provides better handling of cases where sending a data blob to FW results in a validation error. In this case we restore to the last good known value instead of keeping the data that firwmare rejected. Paul Olaru (2): ASoC: sof: Improve sof_ipc3_bytes_ext_put function ASoC: sof: Improve sof_ipc4_bytes_ext_put function sound/soc/sof/ipc3-control.c | 54 ++++++++++++++++++++++++++++++++---- sound/soc/sof/ipc4-control.c | 39 +++++++++++++++++++++++--- sound/soc/sof/sof-audio.h | 1 + 3 files changed, 84 insertions(+), 10 deletions(-) -- 2.25.1

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[PATCH 0/4] ASoC: mt6359: kselftest fix and driver extension
by Trevor Wu 24 May '23

24 May '23

The patch series includes a kselftest fix and changes for extending driver capability to support more use cases. Trevor Wu (4): ASoC: mediatek: mt6359: add supply for MTKAIF ASoC: mediatek: mt6359: fix kselftest error of playback gain ASoC: mediatek: mt6359: add mtkaif gpio setting ASoC: mediatek: mt6359: update route for lineout mux sound/soc/codecs/mt6359.c | 152 ++++++++++++++++++++++++++++++++++---- 1 file changed, 137 insertions(+), 15 deletions(-) -- 2.18.0

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[PATCH v1 0/4] Fixes and cleanup for CS35L41 HDA
by Stefan Binding 23 May '23

23 May '23

Several minor issues were found during additional testing and static analysis. These patches fix these minor issues. Stefan Binding (4): ALSA: hda: cs35l41: Clean up Firmware Load Controls ALSA: hda: cs35l41: Fix endian conversions ALSA: cs35l41: Fix default regmap values for some registers ALSA: hda/realtek: Delete cs35l41 component master during free sound/pci/hda/cs35l41_hda.c | 40 +++++++++++++++------------------- sound/pci/hda/patch_realtek.c | 2 ++ sound/soc/codecs/cs35l41-lib.c | 6 ++--- 3 files changed, 23 insertions(+), 25 deletions(-) -- 2.34.1

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[PATCH v2 0/2] ASoC: do not include runtime_pm.h if not needed
by Claudiu Beznea 23 May '23

23 May '23

Hi, Series removes the pm_runtime.h inclusion in files where APIs exported though pm_runtime.h are not used. In case of files that make use of pm.h which comes form pm_runtime.h added patch 2/2. Changes were built with allmodconfig on ARM and x86_64 and checked all the changed files were built at least once. Thank you, Claudiu Beznea Changes in v2: - removed cs35l45 handling - changed a bit commit description - added patch 2/2 - collected Jarkko Nikula's tag Claudiu Beznea (2): ASoC: do not include pm_runtime.h if not used ASoC: use pm.h instead of runtime_pm.h sound/hda/hdac_regmap.c | 1 - sound/pci/hda/hda_bind.c | 1 - sound/soc/amd/acp/acp-pci.c | 1 - sound/soc/amd/acp/acp-platform.c | 1 - sound/soc/codecs/max98090.c | 1 - sound/soc/codecs/max98373-i2c.c | 2 +- sound/soc/codecs/pcm186x.c | 1 - sound/soc/codecs/rk3328_codec.c | 1 - sound/soc/codecs/rt5682-i2c.c | 1 - sound/soc/codecs/rt5682s.c | 1 - sound/soc/codecs/tas2562.c | 1 - sound/soc/codecs/tas5720.c | 1 - sound/soc/codecs/tas6424.c | 1 - sound/soc/codecs/wm_adsp.c | 1 - sound/soc/fsl/imx-audmix.c | 1 - sound/soc/intel/atom/sst/sst_acpi.c | 1 - sound/soc/intel/atom/sst/sst_ipc.c | 1 - sound/soc/intel/atom/sst/sst_loader.c | 1 - sound/soc/intel/atom/sst/sst_pci.c | 1 - sound/soc/intel/atom/sst/sst_stream.c | 1 - sound/soc/mediatek/mt8186/mt8186-afe-control.c | 1 - sound/soc/mediatek/mt8186/mt8186-mt6366-da7219-max98357.c | 1 - sound/soc/mediatek/mt8186/mt8186-mt6366-rt1019-rt5682s.c | 1 - sound/soc/mediatek/mt8192/mt8192-afe-control.c | 2 -- sound/soc/qcom/lpass-sc7180.c | 2 +- sound/soc/qcom/lpass-sc7280.c | 2 +- sound/soc/soc-compress.c | 1 - sound/soc/soc-pcm.c | 1 - sound/soc/sof/intel/hda-loader-skl.c | 1 - sound/soc/sof/intel/hda-stream.c | 1 - sound/soc/sof/intel/skl.c | 1 - sound/soc/sof/mediatek/mt8186/mt8186-clk.c | 1 - sound/soc/sof/mediatek/mt8195/mt8195-clk.c | 1 - sound/soc/tegra/tegra20_ac97.c | 1 - sound/soc/ti/omap-mcbsp-st.c | 1 - 35 files changed, 3 insertions(+), 36 deletions(-) -- 2.34.1

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[PATCH next] ASoC: SOF: Intel: hda-dai: Fix locking in hda_ipc4_pre_trigger()
by Harshit Mogalapalli 23 May '23

23 May '23

hda_ipc4_pre_trigger() has two issues: 1. In the default case, we are returning without unlocking the mutex. 2. In case SNDRV_PCM_TRIGGER_STOP: when ret is less than zero it goes to out, unlocks but returns zero instead of a negative value. Fix this by changing the final return value to 'ret' instead of zero, and initialize 'ret' to zero in the start of the function. Fixes: 225f37b578a9 ("ASoC: SOF: ipc4-pcm: reset all pipelines during FE DAI hw_free") Signed-off-by: Harshit Mogalapalli <harshit.m.mogalapalli(a)oracle.com> --- Only compile tested. This is found using static anlysis with Smatch. --- sound/soc/sof/intel/hda-dai-ops.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/sof/intel/hda-dai-ops.c b/sound/soc/sof/intel/hda-dai-ops.c index 5a508e118e3d..1e58256c8003 100644 --- a/sound/soc/sof/intel/hda-dai-ops.c +++ b/sound/soc/sof/intel/hda-dai-ops.c @@ -183,7 +183,7 @@ static int hda_ipc4_pre_trigger(struct snd_sof_dev *sdev, struct snd_soc_dai *cp struct sof_ipc4_pipeline *pipeline; struct snd_sof_widget *swidget; struct snd_soc_dapm_widget *w; - int ret; + int ret = 0; w = snd_soc_dai_get_widget(cpu_dai, substream->stream); swidget = w->dobj.private; @@ -208,11 +208,11 @@ static int hda_ipc4_pre_trigger(struct snd_sof_dev *sdev, struct snd_soc_dai *cp break; default: dev_err(sdev->dev, "unknown trigger command %d\n", cmd); - return -EINVAL; + ret = -EINVAL; } out: mutex_unlock(&ipc4_data->pipeline_state_mutex); - return 0; + return ret; } static int hda_trigger(struct snd_sof_dev *sdev, struct snd_soc_dai *cpu_dai, -- 2.31.1

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[PATCH -next] ASoC: SOF: ipc4-topology: Fix an unsigned comparison which can never be negative
by Yang Li 23 May '23

23 May '23

The return value from the call to sof_ipc4_get_valid_bits() is int. However, the return value is being assigned to an unsigned int variable 'out_ref_valid_bits', so making it an int. Eliminate the following warning: ./sound/soc/sof/ipc4-topology.c:1537:6-24: WARNING: Unsigned expression compared with zero: out_ref_valid_bits < 0 Reported-by: Abaci Robot <abaci(a)linux.alibaba.com> Link: https://bugzilla.openanolis.cn/show_bug.cgi?id=4985 Signed-off-by: Yang Li <yang.lee(a)linux.alibaba.com> --- sound/soc/sof/ipc4-topology.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index dce2f8f7f518..d82ac5701eb2 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -1381,8 +1381,8 @@ sof_ipc4_prepare_copier_module(struct snd_sof_widget *swidget, void **ipc_config_data; int *ipc_config_size; u32 **data; - int ipc_size, ret; - u32 out_ref_rate, out_ref_channels, out_ref_valid_bits; + int ipc_size, ret, out_ref_valid_bits; + u32 out_ref_rate, out_ref_channels; u32 deep_buffer_dma_ms = 0; int output_fmt_index; -- 2.20.1.7.g153144c

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[PATCH] ASoC: rt5682: Use a maple tree based register cache
by Mark Brown 23 May '23

23 May '23

regmap has introduced a maple tree based register cache which makes use of this more advanced data structure which has been added to the kernel recently. Maple trees are much flatter than rbtrees, meaning that they do not grow to such depths when the register map is sparse which makes access a bit more efficient. The maple tree cache type is still a bit of a work in progress but should be effective for some devices already. RT5682 seems like a good candidate for maple tree. It only supports single register read/write operations so will gain minimal benefit from storing the register data in device native format like rbtree does (none for SoundWire) and has some sparsity in the register map which is a good fit for maple tree. Convert to use maple tree. There should be little if any visible difference at runtime. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/rt5682-sdw.c | 2 +- sound/soc/codecs/rt5682s.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5682-sdw.c b/sound/soc/codecs/rt5682-sdw.c index 5f80a5d59b65..fb7951f11c92 100644 --- a/sound/soc/codecs/rt5682-sdw.c +++ b/sound/soc/codecs/rt5682-sdw.c @@ -79,7 +79,7 @@ static const struct regmap_config rt5682_sdw_indirect_regmap = { .max_register = RT5682_I2C_MODE, .volatile_reg = rt5682_volatile_register, .readable_reg = rt5682_readable_register, - .cache_type = REGCACHE_RBTREE, + .cache_type = REGCACHE_MAPLE, .reg_defaults = rt5682_reg, .num_reg_defaults = RT5682_REG_NUM, .use_single_read = true, diff --git a/sound/soc/codecs/rt5682s.c b/sound/soc/codecs/rt5682s.c index 9c34dca58f54..36102fa2b806 100644 --- a/sound/soc/codecs/rt5682s.c +++ b/sound/soc/codecs/rt5682s.c @@ -3046,7 +3046,7 @@ static const struct regmap_config rt5682s_regmap = { .max_register = RT5682S_MAX_REG, .volatile_reg = rt5682s_volatile_register, .readable_reg = rt5682s_readable_register, - .cache_type = REGCACHE_RBTREE, + .cache_type = REGCACHE_MAPLE, .reg_defaults = rt5682s_reg, .num_reg_defaults = ARRAY_SIZE(rt5682s_reg), .use_single_read = true, --- base-commit: 4a670ac3e75e517c96cbd01ef870dbd598c3ce71 change-id: 20230419-asoc-rt5682-maple-7da060991ca4 Best regards, -- Mark Brown <broonie(a)kernel.org>

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[ASoC: da7219: Add Jack insertion detection polarity selection 1/2] ASoC: dt-bindings: dialog,da7219: convert to dtschema
by David Rau 23 May '23

23 May '23

- Convert Dialog DA7219 bindings to DT schema format. - Remove unused `dlg,ldo-lvl` property. Signed-off-by: David Rau <David.Rau.opensource(a)dm.renesas.com> --- .../devicetree/bindings/sound/da7219.txt | 112 --------- .../bindings/sound/dialog,da7219.yaml | 230 ++++++++++++++++++ 2 files changed, 230 insertions(+), 112 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/da7219.txt create mode 100644 Documentation/devicetree/bindings/sound/dialog,da7219.yaml diff --git a/Documentation/devicetree/bindings/sound/da7219.txt b/Documentation/devicetree/bindings/sound/da7219.txt deleted file mode 100644 index add1caf26ac2..000000000000 --- a/Documentation/devicetree/bindings/sound/da7219.txt +++ /dev/null @@ -1,112 +0,0 @@ -Dialog Semiconductor DA7219 Audio Codec bindings - -DA7219 is an audio codec with advanced accessory detect features. - -====== - -Required properties: -- compatible : Should be "dlg,da7219" -- reg: Specifies the I2C slave address - -- interrupts : IRQ line info for DA7219. - (See Documentation/devicetree/bindings/interrupt-controller/interrupts.txt for - further information relating to interrupt properties) - -- VDD-supply: VDD power supply for the device -- VDDMIC-supply: VDDMIC power supply for the device -- VDDIO-supply: VDDIO power supply for the device - (See Documentation/devicetree/bindings/regulator/regulator.txt for further - information relating to regulators) - -Optional properties: -- interrupt-names : Name associated with interrupt line. Should be "wakeup" if - interrupt is to be used to wake system, otherwise "irq" should be used. -- wakeup-source: Flag to indicate this device can wake system (suspend/resume). - -- #clock-cells : Should be set to '<1>', two clock sources provided; -- clock-output-names : Names given for DAI clock outputs (WCLK & BCLK); - -- clocks : phandle and clock specifier for codec MCLK. -- clock-names : Clock name string for 'clocks' attribute, should be "mclk". - -- dlg,micbias-lvl : Voltage (mV) for Mic Bias - [<1600>, <1800>, <2000>, <2200>, <2400>, <2600>] -- dlg,mic-amp-in-sel : Mic input source type - ["diff", "se_p", "se_n"] - -Deprecated properties: -- dlg,ldo-lvl : Required internal LDO voltage (mV) level for digital engine - (LDO unavailable in production HW so property no longer required). - -====== - -Child node - 'da7219_aad': - -Optional properties: -- dlg,micbias-pulse-lvl : Mic bias higher voltage pulse level (mV). - [<2800>, <2900>] -- dlg,micbias-pulse-time : Mic bias higher voltage pulse duration (ms) -- dlg,btn-cfg : Periodic button press measurements for 4-pole jack (ms) - [<2>, <5>, <10>, <50>, <100>, <200>, <500>] -- dlg,mic-det-thr : Impedance threshold for mic detection measurement (Ohms) - [<200>, <500>, <750>, <1000>] -- dlg,jack-ins-deb : Debounce time for jack insertion (ms) - [<5>, <10>, <20>, <50>, <100>, <200>, <500>, <1000>] -- dlg,jack-det-rate: Jack type detection latency (3/4 pole) - ["32ms_64ms", "64ms_128ms", "128ms_256ms", "256ms_512ms"] -- dlg,jack-rem-deb : Debounce time for jack removal (ms) - [<1>, <5>, <10>, <20>] -- dlg,a-d-btn-thr : Impedance threshold between buttons A and D - [0x0 - 0xFF] -- dlg,d-b-btn-thr : Impedance threshold between buttons D and B - [0x0 - 0xFF] -- dlg,b-c-btn-thr : Impedance threshold between buttons B and C - [0x0 - 0xFF] -- dlg,c-mic-btn-thr : Impedance threshold between button C and Mic - [0x0 - 0xFF] -- dlg,btn-avg : Number of 8-bit readings for averaged button measurement - [<1>, <2>, <4>, <8>] -- dlg,adc-1bit-rpt : Repeat count for 1-bit button measurement - [<1>, <2>, <4>, <8>] - -====== - -Example: - - codec: da7219@1a { - compatible = "dlg,da7219"; - reg = <0x1a>; - - interrupt-parent = <&gpio6>; - interrupts = <11 IRQ_TYPE_LEVEL_LOW>; - - VDD-supply = <&reg_audio>; - VDDMIC-supply = <&reg_audio>; - VDDIO-supply = <&reg_audio>; - - #clock-cells = <1>; - clock-output-names = "dai-wclk", "dai-bclk"; - - clocks = <&clks 201>; - clock-names = "mclk"; - - dlg,ldo-lvl = <1200>; - dlg,micbias-lvl = <2600>; - dlg,mic-amp-in-sel = "diff"; - - da7219_aad { - dlg,btn-cfg = <50>; - dlg,mic-det-thr = <500>; - dlg,jack-ins-deb = <20>; - dlg,jack-det-rate = "32ms_64ms"; - dlg,jack-rem-deb = <1>; - - dlg,a-d-btn-thr = <0xa>; - dlg,d-b-btn-thr = <0x16>; - dlg,b-c-btn-thr = <0x21>; - dlg,c-mic-btn-thr = <0x3E>; - - dlg,btn-avg = <4>; - dlg,adc-1bit-rpt = <1>; - }; - }; diff --git a/Documentation/devicetree/bindings/sound/dialog,da7219.yaml b/Documentation/devicetree/bindings/sound/dialog,da7219.yaml new file mode 100644 index 000000000000..9160f68696dd --- /dev/null +++ b/Documentation/devicetree/bindings/sound/dialog,da7219.yaml @@ -0,0 +1,230 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/dialog,da7219.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Dialog Semiconductor DA7219 Audio Codec + +maintainers: + - David Rau <David.Rau.opensource(a)dm.renesas.com> + +description: + The DA7219 is an ultra low-power audio codec with + in-built advanced accessory detection (AAD) for mobile + computing and accessory applications, which supports + sample rates up to 96 kHz at 24-bit resolution. + +properties: + compatible: + const: dlg,da7219 + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + VDD-supply: + description: + VDD power supply for the device. + + VDDMIC-supply: + description: + VDDMIC power supply for the device. + + VDDIO-supply: + description: + VDDIO power supply for the device. + + interrupt-names: + description: + Should be "wakeup" if interrupt is to be used to wake system, + otherwise "irq" should be used. + enum: + - wakeup + - irq + + wakeup-source: + type: boolean + description: + Flag to indicate this device can wake system (suspend/resume). + + "#clock-cells": + const: 1 + + clock-output-names: + minItems: 2 + maxItems: 2 + description: + Name given for DAI WCLK and BCLK outputs. + + clocks: + maxItems: 1 + description: + phandle and clock specifier for codec MCLK. + + clock-names: + const: mclk + + dlg,micbias-lvl: + enum: [1600, 1800, 2000, 2200, 2400, 2600] + description: + Voltage (mV) for Mic Bias. + $ref: /schemas/types.yaml#/definitions/uint32 + + dlg,mic-amp-in-sel: + enum: ["diff", "se_p", "se_n"] + description: + Mic input source type. + + diff - Differential. + + se_p - MIC_P. + Positive differential analog microphone input. + + se_n - MIC_N. + Negative differential analog microphone input. + $ref: /schemas/types.yaml#/definitions/string + + da7219_aad: + type: object + description: + Configuration of advanced accessory detection. + properties: + dlg,micbias-pulse-lvl: + enum: [2800, 2900] + description: + Mic bias higher voltage pulse level (mV). + $ref: /schemas/types.yaml#/definitions/uint32 + + dlg,micbias-pulse-time: + description: + Mic bias higher voltage pulse duration (ms). + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 0 + + dlg,btn-cfg: + enum: [2, 5, 10, 50, 100, 200, 500] + description: + Periodic button press measurements for 4-pole jack (ms). + $ref: /schemas/types.yaml#/definitions/uint32 + + dlg,mic-det-thr: + enum: [200, 500, 750, 1000] + description: + Impedance threshold for mic detection measurement (Ohms). + $ref: /schemas/types.yaml#/definitions/uint32 + + dlg,jack-ins-deb: + enum: [5, 10, 20, 50, 100, 200, 500, 1000] + description: + Debounce time for jack insertion (ms). + $ref: /schemas/types.yaml#/definitions/uint32 + + dlg,jack-det-rate: + enum: ["32_64", "64_128", "128_256", "256_512"] + description: + Jack type (3/4 pole) detection latency (ms). + $ref: /schemas/types.yaml#/definitions/string + + dlg,jack-rem-deb: + enum: [1, 5, 10, 20] + description: + Debounce time for jack removal (ms). + $ref: /schemas/types.yaml#/definitions/uint32 + + dlg,a-d-btn-thr: + description: + Impedance threshold between buttons A and D. + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 0 + maximum: 255 + + dlg,d-b-btn-thr: + description: + Impedance threshold between buttons D and B. + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 0 + maximum: 255 + + dlg,b-c-btn-thr: + description: + Impedance threshold between buttons B and C. + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 0 + maximum: 255 + + dlg,c-mic-btn-thr: + description: + Impedance threshold between button C and Mic. + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 0 + maximum: 255 + + dlg,btn-avg: + enum: [1, 2, 4, 8] + description: + Number of 8-bit readings for averaged button measurement. + $ref: /schemas/types.yaml#/definitions/uint32 + + dlg,adc-1bit-rpt: + enum: [1, 2, 4, 8] + description: + Repeat count for 1-bit button measurement. + $ref: /schemas/types.yaml#/definitions/uint32 + +required: + - compatible + - reg + - interrupts + - VDD-supply + - VDDMIC-supply + - VDDIO-supply + +additionalProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/irq.h> + i2c { + #address-cells = <1>; + #size-cells = <0>; + + codec: da7219@1a { + compatible = "dlg,da7219"; + reg = <0x1a>; + + interrupt-parent = <&gpio6>; + interrupts = <11 IRQ_TYPE_LEVEL_LOW>; + + VDD-supply = <&vdd_reg>; + VDDMIC-supply = <&vddmic_reg>; + VDDIO-supply = <&vddio_reg>; + + #clock-cells = <1>; + clock-output-names = "da7219-dai-wclk", "da7219-dai-bclk"; + + clocks = <&clks 201>; + clock-names = "mclk"; + + dlg,micbias-lvl = <2600>; + dlg,mic-amp-in-sel = "diff"; + + da7219_aad { + dlg,btn-cfg = <50>; + dlg,mic-det-thr = <500>; + dlg,jack-ins-deb = <20>; + dlg,jack-det-rate = "32_64"; + dlg,jack-rem-deb = <1>; + + dlg,a-d-btn-thr = <0xa>; + dlg,d-b-btn-thr = <0x16>; + dlg,b-c-btn-thr = <0x21>; + dlg,c-mic-btn-thr = <0x3E>; + + dlg,btn-avg = <4>; + dlg,adc-1bit-rpt = <1>; + }; + }; + }; -- 2.17.1

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[PATCH] ASoC: SOF: amd: Add pci revision id check
by Venkata Prasad Potturu 23 May '23

23 May '23

Add pci revision id check for renoir and rembrandt platforms. Signed-off-by: Venkata Prasad Potturu <venkataprasad.potturu(a)amd.com> --- sound/soc/sof/amd/acp.h | 3 +++ sound/soc/sof/amd/pci-rmb.c | 3 +++ sound/soc/sof/amd/pci-rn.c | 3 +++ 3 files changed, 9 insertions(+) diff --git a/sound/soc/sof/amd/acp.h b/sound/soc/sof/amd/acp.h index 1c535cc6c3a9..dc624f727aa3 100644 --- a/sound/soc/sof/amd/acp.h +++ b/sound/soc/sof/amd/acp.h @@ -55,6 +55,9 @@ #define ACP_DSP_TO_HOST_IRQ 0x04 +#define ACP_RN_PCI_ID 0x01 +#define ACP_RMB_PCI_ID 0x6F + #define HOST_BRIDGE_CZN 0x1630 #define HOST_BRIDGE_RMB 0x14B5 #define ACP_SHA_STAT 0x8000 diff --git a/sound/soc/sof/amd/pci-rmb.c b/sound/soc/sof/amd/pci-rmb.c index eaf70ea6e556..58b3092425f1 100644 --- a/sound/soc/sof/amd/pci-rmb.c +++ b/sound/soc/sof/amd/pci-rmb.c @@ -65,6 +65,9 @@ static int acp_pci_rmb_probe(struct pci_dev *pci, const struct pci_device_id *pc { unsigned int flag; + if (pci->revision != ACP_RMB_PCI_ID) + return -ENODEV; + flag = snd_amd_acp_find_config(pci); if (flag != FLAG_AMD_SOF && flag != FLAG_AMD_SOF_ONLY_DMIC) return -ENODEV; diff --git a/sound/soc/sof/amd/pci-rn.c b/sound/soc/sof/amd/pci-rn.c index 4809cb644152..7409e21ce5aa 100644 --- a/sound/soc/sof/amd/pci-rn.c +++ b/sound/soc/sof/amd/pci-rn.c @@ -65,6 +65,9 @@ static int acp_pci_rn_probe(struct pci_dev *pci, const struct pci_device_id *pci { unsigned int flag; + if (pci->revision != ACP_RN_PCI_ID) + return -ENODEV; + flag = snd_amd_acp_find_config(pci); if (flag != FLAG_AMD_SOF && flag != FLAG_AMD_SOF_ONLY_DMIC) return -ENODEV; -- 2.25.1

2 1

[PATCH] ASoC: nau8825: Add pre-charge actions for input
by David Lin 23 May '23

23 May '23

Adding pre-charge actions to make FEPGA power stable faster. It improve the recording quality at the beginning. Thus, it is also meaningfully to decrease the final adc delay time. Signed-off-by: David Lin <CTLIN0(a)nuvoton.com> --- sound/soc/codecs/nau8825.c | 30 ++++++++++++++++++++++++++++-- sound/soc/codecs/nau8825.h | 7 +++++++ 2 files changed, 35 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index 775c8e0cb09e..cc3e18207c42 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -911,6 +911,32 @@ static bool nau8825_volatile_reg(struct device *dev, unsigned int reg) } } +static int nau8825_fepga_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + struct nau8825 *nau8825 = snd_soc_component_get_drvdata(component); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + regmap_update_bits(nau8825->regmap, NAU8825_REG_FEPGA, + NAU8825_ACDC_CTRL_MASK, + NAU8825_ACDC_VREF_MICP | NAU8825_ACDC_VREF_MICN); + regmap_update_bits(nau8825->regmap, NAU8825_REG_BOOST, + NAU8825_DISCHRG_EN, NAU8825_DISCHRG_EN); + msleep(40); + regmap_update_bits(nau8825->regmap, NAU8825_REG_BOOST, + NAU8825_DISCHRG_EN, 0); + regmap_update_bits(nau8825->regmap, NAU8825_REG_FEPGA, + NAU8825_ACDC_CTRL_MASK, 0); + break; + default: + break; + } + + return 0; +} + static int nau8825_adc_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -1127,8 +1153,8 @@ static const struct snd_soc_dapm_widget nau8825_dapm_widgets[] = { SND_SOC_DAPM_INPUT("MIC"), SND_SOC_DAPM_MICBIAS("MICBIAS", NAU8825_REG_MIC_BIAS, 8, 0), - SND_SOC_DAPM_PGA("Frontend PGA", NAU8825_REG_POWER_UP_CONTROL, 14, 0, - NULL, 0), + SND_SOC_DAPM_PGA_E("Frontend PGA", NAU8825_REG_POWER_UP_CONTROL, 14, 0, + NULL, 0, nau8825_fepga_event, SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_ADC_E("ADC", NULL, SND_SOC_NOPM, 0, 0, nau8825_adc_event, SND_SOC_DAPM_POST_PMU | diff --git a/sound/soc/codecs/nau8825.h b/sound/soc/codecs/nau8825.h index 44b62bc3880f..38ce052aed50 100644 --- a/sound/soc/codecs/nau8825.h +++ b/sound/soc/codecs/nau8825.h @@ -442,10 +442,17 @@ /* BOOST (0x76) */ #define NAU8825_PRECHARGE_DIS (1 << 13) #define NAU8825_GLOBAL_BIAS_EN (1 << 12) +#define NAU8825_DISCHRG_EN (1 << 11) #define NAU8825_HP_BOOST_DIS (1 << 9) #define NAU8825_HP_BOOST_G_DIS (1 << 8) #define NAU8825_SHORT_SHUTDOWN_EN (1 << 6) +/* FEPGA (0x77) */ +#define NAU8825_ACDC_CTRL_SFT 14 +#define NAU8825_ACDC_CTRL_MASK (0x3 << NAU8825_ACDC_CTRL_SFT) +#define NAU8825_ACDC_VREF_MICP (0x1 << NAU8825_ACDC_CTRL_SFT) +#define NAU8825_ACDC_VREF_MICN (0x2 << NAU8825_ACDC_CTRL_SFT) + /* POWER_UP_CONTROL (0x7f) */ #define NAU8825_POWERUP_INTEGR_R (1 << 5) #define NAU8825_POWERUP_INTEGR_L (1 << 4) -- 2.25.1

2 1

[PATCH] ASoC: dt-bindings: rt1016: Convert to dtschema
by Anup Sharma 23 May '23

23 May '23

Convert the RT1016 Stereo Audio Amplifier bindings to DT schema Signed-off-by: Anup Sharma <anupnewsmail(a)gmail.com> --- .../bindings/sound/realtek,rt1016.yaml | 40 +++++++++++++++++++ .../devicetree/bindings/sound/rt1016.txt | 17 -------- 2 files changed, 40 insertions(+), 17 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/realtek,rt1016.yaml delete mode 100644 Documentation/devicetree/bindings/sound/rt1016.txt diff --git a/Documentation/devicetree/bindings/sound/realtek,rt1016.yaml b/Documentation/devicetree/bindings/sound/realtek,rt1016.yaml new file mode 100644 index 000000000000..5287e9c9197e --- /dev/null +++ b/Documentation/devicetree/bindings/sound/realtek,rt1016.yaml @@ -0,0 +1,40 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/realtek,rt1016.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Reaktek RT1016 Stereo Class D Audio Amplifier + +maintainers: + - oder_chiou(a)realtek.com + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: realtek,rt1016 + + reg: + maxItems: 1 + + "#sound-dai-cells": + const: 0 + +required: + - compatible + - reg + +unevaluatedProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + codec@1a { + compatible = "realtek,rt1016"; + reg = <0x1a>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/rt1016.txt b/Documentation/devicetree/bindings/sound/rt1016.txt deleted file mode 100644 index 2310f8ff259b..000000000000 --- a/Documentation/devicetree/bindings/sound/rt1016.txt +++ /dev/null @@ -1,17 +0,0 @@ -RT1016 Stereo Class D Audio Amplifier - -This device supports I2C only. - -Required properties: - -- compatible : "realtek,rt1016". - -- reg : The I2C address of the device. - - -Example: - -rt1016: codec@1a { - compatible = "realtek,rt1016"; - reg = <0x1a>; -}; -- 2.34.1

2 4

[PATCH alsa-lib v2 00/10] Add MIDI 2.0 support
by Takashi Iwai 23 May '23

23 May '23

This is a revised series of patches for alsa-lib to add the support of MIDI 2.0 features. Since v1 [*], two more control API helpers to obtain UMP EP / block info are added and the CTL protocol version is bumped. Takashi [*] http://lore.kernel.org/r/20230520070021.1301-1-tiwai@suse.de === Takashi Iwai (10): uapi: Update rawmidi API to 2.0.3 uapi: Update control API to 2.0.9 rawmidi: Add UMP ioctl support ump: Add initial support control: Add UMP device query support control: Add UMP Endpoint and Block info query support ump: Add helpers to parse / set UMP packet data ump: Add helpers for handling SysEx data uapi: Update asequencer.h definitions for 1.0.3 seq: Add UMP support configure.ac | 1 + include/Makefile.am | 2 +- include/control.h | 3 + include/local.h | 5 + include/rawmidi.h | 3 + include/seq.h | 44 ++ include/seq_event.h | 42 +- include/seqmid.h | 24 ++ include/sound/uapi/asequencer.h | 91 ++++- include/sound/uapi/asound.h | 62 ++- include/ump.h | 119 ++++++ include/ump_msg.h | 598 +++++++++++++++++++++++++++ src/Versions.in | 26 ++ src/control/control.c | 38 ++ src/control/control_hw.c | 29 ++ src/control/control_local.h | 3 + src/rawmidi/Makefile.am | 5 +- src/rawmidi/rawmidi.c | 19 + src/rawmidi/rawmidi_hw.c | 36 +- src/rawmidi/rawmidi_local.h | 7 + src/rawmidi/ump.c | 702 ++++++++++++++++++++++++++++++++ src/rawmidi/ump_local.h | 10 + src/seq/seq.c | 410 +++++++++++++++++-- src/seq/seq_hw.c | 72 +++- src/seq/seq_local.h | 6 +- src/seq/seqmid.c | 38 ++ 26 files changed, 2316 insertions(+), 79 deletions(-) create mode 100644 include/ump.h create mode 100644 include/ump_msg.h create mode 100644 src/rawmidi/ump.c create mode 100644 src/rawmidi/ump_local.h -- 2.35.3

1 10

[PATCH 1/2] ALSA: emu10k1: introduce higher-level voice manipulation functions
by Oswald Buddenhagen 23 May '23

23 May '23

This adds snd_emu10k1_pcm_init_{voices,extra_voice}() and snd_emu10k1_playback_{un,}mute_voices() to slightly abstract by voice function and potential stereo property. Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen(a)gmx.de> --- sound/pci/emu10k1/emupcm.c | 70 +++++++++++++++++++++++++++++--------- 1 file changed, 53 insertions(+), 17 deletions(-) diff --git a/sound/pci/emu10k1/emupcm.c b/sound/pci/emu10k1/emupcm.c index 0036593cca7c..65af94d08b47 100644 --- a/sound/pci/emu10k1/emupcm.c +++ b/sound/pci/emu10k1/emupcm.c @@ -312,6 +312,30 @@ static void snd_emu10k1_pcm_init_voice(struct snd_emu10k1 *emu, spin_unlock_irqrestore(&emu->reg_lock, flags); } +static void snd_emu10k1_pcm_init_voices(struct snd_emu10k1 *emu, + struct snd_emu10k1_voice *evoice, + bool w_16, bool stereo, + unsigned int start_addr, + unsigned int end_addr, + struct snd_emu10k1_pcm_mixer *mix) +{ + snd_emu10k1_pcm_init_voice(emu, 1, 0, evoice, w_16, stereo, + start_addr, end_addr, mix); + if (stereo) + snd_emu10k1_pcm_init_voice(emu, 0, 0, evoice + 1, w_16, true, + start_addr, end_addr, mix); +} + +static void snd_emu10k1_pcm_init_extra_voice(struct snd_emu10k1 *emu, + struct snd_emu10k1_voice *evoice, + bool w_16, + unsigned int start_addr, + unsigned int end_addr) +{ + snd_emu10k1_pcm_init_voice(emu, 1, 1, evoice, w_16, false, + start_addr, end_addr, NULL); +} + static int snd_emu10k1_playback_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *hw_params) { @@ -393,18 +417,15 @@ static int snd_emu10k1_playback_prepare(struct snd_pcm_substream *substream) start_addr = epcm->start_addr >> w_16; end_addr = start_addr + runtime->period_size; - snd_emu10k1_pcm_init_voice(emu, 1, 1, epcm->extra, w_16, false, - start_addr, end_addr, NULL); + snd_emu10k1_pcm_init_extra_voice(emu, epcm->extra, w_16, + start_addr, end_addr); start_addr >>= stereo; epcm->ccca_start_addr = start_addr; end_addr = start_addr + runtime->buffer_size; - snd_emu10k1_pcm_init_voice(emu, 1, 0, epcm->voices[0], w_16, stereo, - start_addr, end_addr, - &emu->pcm_mixer[substream->number]); - if (stereo) - snd_emu10k1_pcm_init_voice(emu, 0, 0, epcm->voices[0] + 1, w_16, true, - start_addr, end_addr, - &emu->pcm_mixer[substream->number]); + snd_emu10k1_pcm_init_voices(emu, epcm->voices[0], w_16, stereo, + start_addr, end_addr, + &emu->pcm_mixer[substream->number]); + return 0; } @@ -421,8 +442,8 @@ static int snd_emu10k1_efx_playback_prepare(struct snd_pcm_substream *substream) channel_size = runtime->buffer_size; - snd_emu10k1_pcm_init_voice(emu, 1, 1, epcm->extra, true, false, - start_addr, start_addr + (channel_size / 2), NULL); + snd_emu10k1_pcm_init_extra_voice(emu, epcm->extra, true, + start_addr, start_addr + (channel_size / 2)); epcm->ccca_start_addr = start_addr; for (i = 0; i < NUM_EFX_PLAYBACK; i++) { @@ -598,12 +619,31 @@ static void snd_emu10k1_playback_unmute_voice(struct snd_emu10k1 *emu, snd_emu10k1_playback_commit_volume(emu, evoice, vattn); } +static void snd_emu10k1_playback_unmute_voices(struct snd_emu10k1 *emu, + struct snd_emu10k1_voice *evoice, + bool stereo, + struct snd_emu10k1_pcm_mixer *mix) +{ + snd_emu10k1_playback_unmute_voice(emu, evoice, stereo, true, mix); + if (stereo) + snd_emu10k1_playback_unmute_voice(emu, evoice + 1, true, false, mix); +} + static void snd_emu10k1_playback_mute_voice(struct snd_emu10k1 *emu, struct snd_emu10k1_voice *evoice) { snd_emu10k1_playback_commit_volume(emu, evoice, 0); } +static void snd_emu10k1_playback_mute_voices(struct snd_emu10k1 *emu, + struct snd_emu10k1_voice *evoice, + bool stereo) +{ + snd_emu10k1_playback_mute_voice(emu, evoice); + if (stereo) + snd_emu10k1_playback_mute_voice(emu, evoice + 1); +} + static void snd_emu10k1_playback_commit_pitch(struct snd_emu10k1 *emu, u32 voice, u32 pitch_target) { @@ -680,22 +720,18 @@ static int snd_emu10k1_playback_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: case SNDRV_PCM_TRIGGER_RESUME: mix = &emu->pcm_mixer[substream->number]; - snd_emu10k1_playback_unmute_voice(emu, epcm->voices[0], stereo, true, mix); - if (stereo) - snd_emu10k1_playback_unmute_voice(emu, epcm->voices[0] + 1, true, false, mix); + snd_emu10k1_playback_unmute_voices(emu, epcm->voices[0], stereo, mix); snd_emu10k1_playback_set_running(emu, epcm); snd_emu10k1_playback_trigger_voice(emu, epcm->voices[0]); snd_emu10k1_playback_trigger_voice(emu, epcm->extra); break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: case SNDRV_PCM_TRIGGER_SUSPEND: snd_emu10k1_playback_stop_voice(emu, epcm->voices[0]); snd_emu10k1_playback_stop_voice(emu, epcm->extra); snd_emu10k1_playback_set_stopped(emu, epcm); - snd_emu10k1_playback_mute_voice(emu, epcm->voices[0]); - if (stereo) - snd_emu10k1_playback_mute_voice(emu, epcm->voices[0] + 1); + snd_emu10k1_playback_mute_voices(emu, epcm->voices[0], stereo); break; default: result = -EINVAL; -- 2.40.0.152.g15d061e6df

2 2

[PATCH v1 1/3] ASoC: jz4740-i2s: Add support for X1000 SoC
by Aidan MacDonald 23 May '23

23 May '23

The X1000's AIC is similar to the AIC found on other Ingenic SoCs. It has symmetric playback/capture rates like the JZ4740, but more flexible clocking when outputting the system or bit clocks. Signed-off-by: Aidan MacDonald <aidanmacdonald.0x0(a)gmail.com> --- sound/soc/jz4740/jz4740-i2s.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/jz4740/jz4740-i2s.c b/sound/soc/jz4740/jz4740-i2s.c index 6d9cfe0a5041..7cb563bb8b09 100644 --- a/sound/soc/jz4740/jz4740-i2s.c +++ b/sound/soc/jz4740/jz4740-i2s.c @@ -329,6 +329,14 @@ static const struct i2s_soc_info jz4760_i2s_soc_info = { .field_i2sdiv_playback = REG_FIELD(JZ_REG_AIC_CLK_DIV, 0, 3), }; +static const struct i2s_soc_info x1000_i2s_soc_info = { + .dai = &jz4740_i2s_dai, + .field_rx_fifo_thresh = REG_FIELD(JZ_REG_AIC_CONF, 24, 27), + .field_tx_fifo_thresh = REG_FIELD(JZ_REG_AIC_CONF, 16, 20), + .field_i2sdiv_capture = REG_FIELD(JZ_REG_AIC_CLK_DIV, 0, 8), + .field_i2sdiv_playback = REG_FIELD(JZ_REG_AIC_CLK_DIV, 0, 8), +}; + static struct snd_soc_dai_driver jz4770_i2s_dai = { .probe = jz4740_i2s_dai_probe, .playback = { @@ -440,6 +448,7 @@ static const struct of_device_id jz4740_of_matches[] = { { .compatible = "ingenic,jz4760-i2s", .data = &jz4760_i2s_soc_info }, { .compatible = "ingenic,jz4770-i2s", .data = &jz4770_i2s_soc_info }, { .compatible = "ingenic,jz4780-i2s", .data = &jz4780_i2s_soc_info }, + { .compatible = "ingenic,x1000-i2s", .data = &x1000_i2s_soc_info }, { /* sentinel */ } }; MODULE_DEVICE_TABLE(of, jz4740_of_matches); -- 2.39.2

4 5

[PATCH 0/5] ASoC: topology: Clean up error messages handling
by Amadeusz Sławiński 22 May '23

22 May '23

Series of patches cleaning up error messages when loading topology. In few places instead of logging in place of failure message is logged in caller. Additionally there are places where both caller and failing function log error, leading to unnecessary logs. Clean all of the above up. Amadeusz Sławiński (5): ASoC: topology: Log control load errors in soc_tplg_control_load() ASoC: topology: Remove redundant logs ASoC: topology: Do not split message string on multiple lines ASoC: topology: Remove redundant log ASoC: topology: Remove redundant log sound/soc/soc-topology.c | 63 ++++++++++++---------------------------- 1 file changed, 19 insertions(+), 44 deletions(-) -- 2.34.1

4 8

[PATCH 0/5] ASoC: SOF: ipc4: Querry CPC value from firmware's manifest
by Peter Ujfalusi 22 May '23

22 May '23

Hi, Hi, The MOD_INIT_INSTANCE message contains a CPC (Cycles Per Chunk/processing unit) parameter. This CPC value is used by the firmware to calculate the total cycles needed by the enabled module instances and based on this it can decide to set the frequency of the DSP core(s). The manifest section of the firmware image contains a module configuration section, where a per module table of configurations are listed with measured CPC values as triplet of IBS/IBS/CPC (Input/Output buffer size - corresponding to the selected audio format). In case the CPC value is 0 (missing from the manifest or the configuration cannot be matched) the firmware will force the DSP cores to maximum speed to avoid audio glitches due to starvation. In these cases the kernel will print a warning message to let the SOF developers know about the gap and provide information to correct it with a firmware update. Regards, Peter --- Peter Ujfalusi (5): ASoC: SOF: ipc4-loader: Drop unused bss_size from struct sof_ipc4_fw_module ASoC: SOF: ipc4-loader: Save a pointer to fm_config in sof_ipc4_fw_module ASoC: SOF: ipc4-topology: Rename sof_ipc4_update_pipeline_mem_usage() to be generic ASoC: SOF: ipc4-topology: Do not use the CPC value from topology ASoC: SOF: ipc4-loader/topology: Query the CPC value from manifest sound/soc/sof/ipc4-loader.c | 72 ++++++++++++++++++++++++++++++++--- sound/soc/sof/ipc4-priv.h | 10 ++++- sound/soc/sof/ipc4-topology.c | 33 +++++++++------- 3 files changed, 94 insertions(+), 21 deletions(-) -- 2.40.1

2 6

[PATCH] ASoC: dt-bindings: tlv320aic32x4: Fix supply names
by David Epping 22 May '23

22 May '23

The term "-supply" is a suffix to regulator names. Signed-off-by: David Epping <david.epping(a)missinglinkelectronics.com> --- Documentation/devicetree/bindings/sound/tlv320aic32x4.txt | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/tlv320aic32x4.txt b/Documentation/devicetree/bindings/sound/tlv320aic32x4.txt index f59125bc79d1..0b4e21bde5bc 100644 --- a/Documentation/devicetree/bindings/sound/tlv320aic32x4.txt +++ b/Documentation/devicetree/bindings/sound/tlv320aic32x4.txt @@ -8,7 +8,7 @@ Required properties: "ti,tlv320aic32x6" TLV320AIC3206, TLV320AIC3256 "ti,tas2505" TAS2505, TAS2521 - reg: I2C slave address - - supply-*: Required supply regulators are: + - *-supply: Required supply regulators are: "iov" - digital IO power supply "ldoin" - LDO power supply "dv" - Digital core power supply -- 2.17.1

2 1

[PATCH] ASoC: stm32: sai: Use the devm_clk_get_optional() helper
by Christophe JAILLET 22 May '23

22 May '23

Use devm_clk_get_optional() instead of hand writing it. This saves some LoC and improves the semantic. Signed-off-by: Christophe JAILLET <christophe.jaillet(a)wanadoo.fr> --- sound/soc/stm/stm32_sai_sub.c | 9 +++------ 1 file changed, 3 insertions(+), 6 deletions(-) diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index f6695dee353b..271ec5b3378d 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -1485,12 +1485,9 @@ static int stm32_sai_sub_parse_of(struct platform_device *pdev, if (ret < 0) return ret; } else { - sai->sai_mclk = devm_clk_get(&pdev->dev, "MCLK"); - if (IS_ERR(sai->sai_mclk)) { - if (PTR_ERR(sai->sai_mclk) != -ENOENT) - return PTR_ERR(sai->sai_mclk); - sai->sai_mclk = NULL; - } + sai->sai_mclk = devm_clk_get_optional(&pdev->dev, "MCLK"); + if (IS_ERR(sai->sai_mclk)) + return PTR_ERR(sai->sai_mclk); } return 0; -- 2.34.1

2 1

[PATCH] ASoC: rt5682s: Use the devm_clk_get_optional() helper
by Christophe JAILLET 22 May '23

22 May '23

Use devm_clk_get_optional() instead of hand writing it. This saves some LoC and improves the semantic. Signed-off-by: Christophe JAILLET <christophe.jaillet(a)wanadoo.fr> --- sound/soc/codecs/rt5682s.c | 11 +++-------- 1 file changed, 3 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/rt5682s.c b/sound/soc/codecs/rt5682s.c index 81163b026b9e..a01de405c22c 100644 --- a/sound/soc/codecs/rt5682s.c +++ b/sound/soc/codecs/rt5682s.c @@ -2848,14 +2848,9 @@ static int rt5682s_dai_probe_clks(struct snd_soc_component *component) int ret; /* Check if MCLK provided */ - rt5682s->mclk = devm_clk_get(component->dev, "mclk"); - if (IS_ERR(rt5682s->mclk)) { - if (PTR_ERR(rt5682s->mclk) != -ENOENT) { - ret = PTR_ERR(rt5682s->mclk); - return ret; - } - rt5682s->mclk = NULL; - } + rt5682s->mclk = devm_clk_get_optional(component->dev, "mclk"); + if (IS_ERR(rt5682s->mclk)) + return PTR_ERR(rt5682s->mclk); /* Register CCF DAI clock control */ ret = rt5682s_register_dai_clks(component); -- 2.34.1

2 1

[PATCH] ASoC: rt5659: Use the devm_clk_get_optional() helper
by Christophe JAILLET 22 May '23

22 May '23

Use devm_clk_get_optional() instead of hand writing it. This saves some LoC and improves the semantic. Signed-off-by: Christophe JAILLET <christophe.jaillet(a)wanadoo.fr> --- sound/soc/codecs/rt5659.c | 10 +++------- 1 file changed, 3 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/rt5659.c b/sound/soc/codecs/rt5659.c index 22bb57029bc0..df6f0d769bbd 100644 --- a/sound/soc/codecs/rt5659.c +++ b/sound/soc/codecs/rt5659.c @@ -4141,13 +4141,9 @@ static int rt5659_i2c_probe(struct i2c_client *i2c) regmap_write(rt5659->regmap, RT5659_RESET, 0); /* Check if MCLK provided */ - rt5659->mclk = devm_clk_get(&i2c->dev, "mclk"); - if (IS_ERR(rt5659->mclk)) { - if (PTR_ERR(rt5659->mclk) != -ENOENT) - return PTR_ERR(rt5659->mclk); - /* Otherwise mark the mclk pointer to NULL */ - rt5659->mclk = NULL; - } + rt5659->mclk = devm_clk_get_optional(&i2c->dev, "mclk"); + if (IS_ERR(rt5659->mclk)) + return PTR_ERR(rt5659->mclk); rt5659_calibrate(rt5659); -- 2.34.1

2 1

[PATCH] ASoC: cs53l30: Use the devm_clk_get_optional() helper
by Christophe JAILLET 22 May '23

22 May '23

Use devm_clk_get_optional() instead of hand writing it. This saves some LoC and improves the semantic. Signed-off-by: Christophe JAILLET <christophe.jaillet(a)wanadoo.fr> --- sound/soc/codecs/cs53l30.c | 10 +++------- 1 file changed, 3 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/cs53l30.c b/sound/soc/codecs/cs53l30.c index 51ca66e7b3ea..21962b828ab1 100644 --- a/sound/soc/codecs/cs53l30.c +++ b/sound/soc/codecs/cs53l30.c @@ -990,14 +990,10 @@ static int cs53l30_i2c_probe(struct i2c_client *client) } /* Check if MCLK provided */ - cs53l30->mclk = devm_clk_get(dev, "mclk"); + cs53l30->mclk = devm_clk_get_optional(dev, "mclk"); if (IS_ERR(cs53l30->mclk)) { - if (PTR_ERR(cs53l30->mclk) != -ENOENT) { - ret = PTR_ERR(cs53l30->mclk); - goto error; - } - /* Otherwise mark the mclk pointer to NULL */ - cs53l30->mclk = NULL; + ret = PTR_ERR(cs53l30->mclk); + goto error; } /* Fetch the MUTE control */ -- 2.34.1

2 1

[PATCH] ASoC: cs42l51: Use the devm_clk_get_optional() helper
by Christophe JAILLET 22 May '23

22 May '23

Use devm_clk_get_optional() instead of hand writing it. This saves some LoC and improves the semantic. Signed-off-by: Christophe JAILLET <christophe.jaillet(a)wanadoo.fr> --- sound/soc/codecs/cs42l51.c | 9 +++------ 1 file changed, 3 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index e88d9ff95cdf..a67cd3ee84e0 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -724,12 +724,9 @@ int cs42l51_probe(struct device *dev, struct regmap *regmap) dev_set_drvdata(dev, cs42l51); cs42l51->regmap = regmap; - cs42l51->mclk_handle = devm_clk_get(dev, "MCLK"); - if (IS_ERR(cs42l51->mclk_handle)) { - if (PTR_ERR(cs42l51->mclk_handle) != -ENOENT) - return PTR_ERR(cs42l51->mclk_handle); - cs42l51->mclk_handle = NULL; - } + cs42l51->mclk_handle = devm_clk_get_optional(dev, "MCLK"); + if (IS_ERR(cs42l51->mclk_handle)) + return PTR_ERR(cs42l51->mclk_handle); for (i = 0; i < ARRAY_SIZE(cs42l51->supplies); i++) cs42l51->supplies[i].supply = cs42l51_supply_names[i]; -- 2.34.1

2 1

[PATCH] ASoC: adau1761: Use the devm_clk_get_optional() helper
by Christophe JAILLET 22 May '23

22 May '23

Use devm_clk_get_optional() instead of hand writing it. This saves some LoC and improves the semantic. Signed-off-by: Christophe JAILLET <christophe.jaillet(a)wanadoo.fr> --- sound/soc/codecs/adau17x1.c | 13 ++++++------- 1 file changed, 6 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/adau17x1.c b/sound/soc/codecs/adau17x1.c index 634d4dbca5ec..f2932713b4de 100644 --- a/sound/soc/codecs/adau17x1.c +++ b/sound/soc/codecs/adau17x1.c @@ -1059,13 +1059,12 @@ int adau17x1_probe(struct device *dev, struct regmap *regmap, if (!adau) return -ENOMEM; - adau->mclk = devm_clk_get(dev, "mclk"); - if (IS_ERR(adau->mclk)) { - if (PTR_ERR(adau->mclk) != -ENOENT) - return PTR_ERR(adau->mclk); - /* Clock is optional (for the driver) */ - adau->mclk = NULL; - } else if (adau->mclk) { + /* Clock is optional (for the driver) */ + adau->mclk = devm_clk_get_optional(dev, "mclk"); + if (IS_ERR(adau->mclk)) + return PTR_ERR(adau->mclk); + + if (adau->mclk) { adau->clk_src = ADAU17X1_CLK_SRC_PLL_AUTO; /* -- 2.34.1

2 1

[PATCH] ASoC: atmel: sam9g20_wm8731: Remove the unneeded include <linux/i2c.h>
by Christophe JAILLET 22 May '23

22 May '23

This driver does not use i2c, so there is no point in including <linux/i2c.h> Remove it. Signed-off-by: Christophe JAILLET <christophe.jaillet(a)wanadoo.fr> --- Based on one of my scripts, this reduce the number of included files from 589 to 498 --- sound/soc/atmel/sam9g20_wm8731.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c index baf38964b491..0405e9e49140 100644 --- a/sound/soc/atmel/sam9g20_wm8731.c +++ b/sound/soc/atmel/sam9g20_wm8731.c @@ -23,7 +23,6 @@ #include <linux/timer.h> #include <linux/interrupt.h> #include <linux/platform_device.h> -#include <linux/i2c.h> #include <linux/of.h> #include <linux/atmel-ssc.h> -- 2.34.1

2 1

[PATCH 0/7] ASoC: Intel: Fixes
by Amadeusz Sławiński 22 May '23

22 May '23

Series of fixes for issues found during development and testing, primarly for avs driver. Amadeusz Sławiński (4): ASoC: Intel: avs: Fix module lookup ASoC: Intel: avs: Access path components under lock ASoC: Intel: avs: Fix avs_path_module::instance_id size ASoC: Intel: avs: Add missing checks on FE startup Cezary Rojewski (3): ASoC: Intel: Skylake: Fix declaration of enum skl_ch_cfg ASoC: Intel: avs: Fix declaration of enum avs_channel_config ASoC: Intel: avs: Account for UID of ACPI device include/sound/soc-acpi.h | 1 + include/uapi/sound/skl-tplg-interface.h | 3 ++- sound/soc/intel/avs/apl.c | 6 +++++- sound/soc/intel/avs/avs.h | 4 ++-- sound/soc/intel/avs/board_selection.c | 2 +- sound/soc/intel/avs/control.c | 22 +++++++++++++++------- sound/soc/intel/avs/dsp.c | 4 ++-- sound/soc/intel/avs/messages.h | 2 +- sound/soc/intel/avs/path.h | 2 +- sound/soc/intel/avs/pcm.c | 23 ++++++++++++++++++----- sound/soc/intel/avs/probes.c | 2 +- 11 files changed, 49 insertions(+), 22 deletions(-) -- 2.34.1

2 8

[PATCH v5 32/44] sound: add HAS_IOPORT dependencies
by Niklas Schnelle 22 May '23

22 May '23

In a future patch HAS_IOPORT=n will result in inb()/outb() and friends not being declared. We thus need to add HAS_IOPORT as dependency for those drivers using them. Co-developed-by: Arnd Bergmann <arnd(a)kernel.org> Signed-off-by: Arnd Bergmann <arnd(a)kernel.org> Signed-off-by: Niklas Schnelle <schnelle(a)linux.ibm.com> --- sound/drivers/Kconfig | 3 +++ sound/isa/Kconfig | 1 + sound/pci/Kconfig | 45 ++++++++++++++++++++++++++++++++----------- sound/pcmcia/Kconfig | 1 + 4 files changed, 39 insertions(+), 11 deletions(-) diff --git a/sound/drivers/Kconfig b/sound/drivers/Kconfig index be3009746f3a..864991d8776d 100644 --- a/sound/drivers/Kconfig +++ b/sound/drivers/Kconfig @@ -128,6 +128,7 @@ config SND_VIRMIDI config SND_MTPAV tristate "MOTU MidiTimePiece AV multiport MIDI" + depends on HAS_IOPORT select SND_RAWMIDI help To use a MOTU MidiTimePiece AV multiport MIDI adapter @@ -152,6 +153,7 @@ config SND_MTS64 config SND_SERIAL_U16550 tristate "UART16550 serial MIDI driver" + depends on HAS_IOPORT select SND_RAWMIDI help To include support for MIDI serial port interfaces, say Y here @@ -185,6 +187,7 @@ config SND_SERIAL_GENERIC config SND_MPU401 tristate "Generic MPU-401 UART driver" + depends on HAS_IOPORT select SND_MPU401_UART help Say Y here to include support for MIDI ports compatible with diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig index 6ffa48dd5983..f8159179e38d 100644 --- a/sound/isa/Kconfig +++ b/sound/isa/Kconfig @@ -23,6 +23,7 @@ menuconfig SND_ISA bool "ISA sound devices" depends on ISA || COMPILE_TEST depends on ISA_DMA_API + depends on HAS_IOPORT default y help Support for sound devices connected via the ISA bus. diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index 861958451ef5..787868c9e91b 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -26,7 +26,7 @@ config SND_ALS300 select SND_PCM select SND_AC97_CODEC select SND_OPL3_LIB - depends on ZONE_DMA + depends on ZONE_DMA && HAS_IOPORT help Say 'Y' or 'M' to include support for Avance Logic ALS300/ALS300+ @@ -36,6 +36,7 @@ config SND_ALS300 config SND_ALS4000 tristate "Avance Logic ALS4000" depends on ISA_DMA_API + depends on HAS_IOPORT select SND_OPL3_LIB select SND_MPU401_UART select SND_PCM @@ -51,7 +52,7 @@ config SND_ALI5451 tristate "ALi M5451 PCI Audio Controller" select SND_MPU401_UART select SND_AC97_CODEC - depends on ZONE_DMA + depends on ZONE_DMA && HAS_IOPORT help Say Y here to include support for the integrated AC97 sound device on motherboards using the ALi M5451 Audio Controller @@ -96,6 +97,7 @@ config SND_ATIIXP_MODEM config SND_AU8810 tristate "Aureal Advantage" + depends on HAS_IOPORT select SND_MPU401_UART select SND_AC97_CODEC help @@ -110,6 +112,7 @@ config SND_AU8810 config SND_AU8820 tristate "Aureal Vortex" + depends on HAS_IOPORT select SND_MPU401_UART select SND_AC97_CODEC help @@ -123,6 +126,7 @@ config SND_AU8820 config SND_AU8830 tristate "Aureal Vortex 2" + depends on HAS_IOPORT select SND_MPU401_UART select SND_AC97_CODEC help @@ -157,7 +161,7 @@ config SND_AZT3328 select SND_RAWMIDI select SND_AC97_CODEC select SND_TIMER - depends on ZONE_DMA + depends on ZONE_DMA && HAS_IOPORT help Say Y here to include support for Aztech AZF3328 (PCI168) soundcards. @@ -193,6 +197,7 @@ config SND_BT87X_OVERCLOCK config SND_CA0106 tristate "SB Audigy LS / Live 24bit" + depends on HAS_IOPORT select SND_AC97_CODEC select SND_RAWMIDI select SND_VMASTER @@ -205,6 +210,7 @@ config SND_CA0106 config SND_CMIPCI tristate "C-Media 8338, 8738, 8768, 8770" + depends on HAS_IOPORT select SND_OPL3_LIB select SND_MPU401_UART select SND_PCM @@ -221,6 +227,7 @@ config SND_OXYGEN_LIB config SND_OXYGEN tristate "C-Media 8786, 8787, 8788 (Oxygen)" + depends on HAS_IOPORT select SND_OXYGEN_LIB select SND_PCM select SND_MPU401_UART @@ -246,6 +253,7 @@ config SND_OXYGEN config SND_CS4281 tristate "Cirrus Logic (Sound Fusion) CS4281" + depends on HAS_IOPORT select SND_OPL3_LIB select SND_RAWMIDI select SND_AC97_CODEC @@ -257,6 +265,7 @@ config SND_CS4281 config SND_CS46XX tristate "Cirrus Logic (Sound Fusion) CS4280/CS461x/CS462x/CS463x" + depends on HAS_IOPORT select SND_RAWMIDI select SND_AC97_CODEC select FW_LOADER @@ -290,6 +299,7 @@ config SND_CS5530 config SND_CS5535AUDIO tristate "CS5535/CS5536 Audio" depends on X86_32 || MIPS || COMPILE_TEST + depends on HAS_IOPORT select SND_PCM select SND_AC97_CODEC help @@ -307,6 +317,7 @@ config SND_CS5535AUDIO config SND_CTXFI tristate "Creative Sound Blaster X-Fi" + depends on HAS_IOPORT select SND_PCM help If you want to use soundcards based on Creative Sound Blastr X-Fi @@ -468,7 +479,7 @@ config SND_EMU10K1 select SND_AC97_CODEC select SND_TIMER select SND_SEQ_DEVICE if SND_SEQUENCER != n - depends on ZONE_DMA + depends on ZONE_DMA && HAS_IOPORT help Say Y to include support for Sound Blaster PCI 512, Live!, Audigy and E-MU APS/0404/1010/1212/1616/1820 soundcards. @@ -491,7 +502,7 @@ config SND_EMU10K1X tristate "Emu10k1X (Dell OEM Version)" select SND_AC97_CODEC select SND_RAWMIDI - depends on ZONE_DMA + depends on ZONE_DMA && HAS_IOPORT help Say Y here to include support for the Dell OEM version of the Sound Blaster Live!. @@ -501,6 +512,7 @@ config SND_EMU10K1X config SND_ENS1370 tristate "(Creative) Ensoniq AudioPCI 1370" + depends on HAS_IOPORT select SND_RAWMIDI select SND_PCM help @@ -511,6 +523,7 @@ config SND_ENS1370 config SND_ENS1371 tristate "(Creative) Ensoniq AudioPCI 1371/1373" + depends on HAS_IOPORT select SND_RAWMIDI select SND_AC97_CODEC help @@ -525,7 +538,7 @@ config SND_ES1938 select SND_OPL3_LIB select SND_MPU401_UART select SND_AC97_CODEC - depends on ZONE_DMA + depends on ZONE_DMA && HAS_IOPORT help Say Y here to include support for soundcards based on ESS Solo-1 (ES1938, ES1946, ES1969) chips. @@ -537,7 +550,7 @@ config SND_ES1968 tristate "ESS ES1968/1978 (Maestro-1/2/2E)" select SND_MPU401_UART select SND_AC97_CODEC - depends on ZONE_DMA + depends on ZONE_DMA && HAS_IOPORT help Say Y here to include support for soundcards based on ESS Maestro 1/2/2E chips. @@ -569,6 +582,7 @@ config SND_ES1968_RADIO config SND_FM801 tristate "ForteMedia FM801" + depends on HAS_IOPORT select SND_OPL3_LIB select SND_MPU401_UART select SND_AC97_CODEC @@ -624,7 +638,7 @@ config SND_ICE1712 select SND_MPU401_UART select SND_AC97_CODEC select BITREVERSE - depends on ZONE_DMA + depends on ZONE_DMA && HAS_IOPORT help Say Y here to include support for soundcards based on the ICE1712 (Envy24) chip. @@ -640,6 +654,7 @@ config SND_ICE1712 config SND_ICE1724 tristate "ICE/VT1724/1720 (Envy24HT/PT)" + depends on HAS_IOPORT select SND_RAWMIDI select SND_AC97_CODEC select SND_VMASTER @@ -712,7 +727,7 @@ config SND_LX6464ES config SND_MAESTRO3 tristate "ESS Allegro/Maestro3" select SND_AC97_CODEC - depends on ZONE_DMA + depends on ZONE_DMA && HAS_IOPORT help Say Y here to include support for soundcards based on ESS Maestro 3 (Allegro) chips. @@ -753,6 +768,7 @@ config SND_NM256 config SND_PCXHR tristate "Digigram PCXHR" + depends on HAS_IOPORT select FW_LOADER select SND_PCM select SND_HWDEP @@ -764,6 +780,7 @@ config SND_PCXHR config SND_RIPTIDE tristate "Conexant Riptide" + depends on HAS_IOPORT select FW_LOADER select SND_OPL3_LIB select SND_MPU401_UART @@ -808,6 +825,7 @@ config SND_RME9652 config SND_SE6X tristate "Studio Evolution SE6X" depends on SND_OXYGEN=n && SND_VIRTUOSO=n # PCI ID conflict + depends on HAS_IOPORT select SND_OXYGEN_LIB select SND_PCM select SND_MPU401_UART @@ -830,7 +848,7 @@ config SND_SONICVIBES select SND_OPL3_LIB select SND_MPU401_UART select SND_AC97_CODEC - depends on ZONE_DMA + depends on ZONE_DMA && HAS_IOPORT help Say Y here to include support for soundcards based on the S3 SonicVibes chip. @@ -842,7 +860,7 @@ config SND_TRIDENT tristate "Trident 4D-Wave DX/NX; SiS 7018" select SND_MPU401_UART select SND_AC97_CODEC - depends on ZONE_DMA + depends on ZONE_DMA && HAS_IOPORT help Say Y here to include support for soundcards based on Trident 4D-Wave DX/NX or SiS 7018 chips. @@ -852,6 +870,7 @@ config SND_TRIDENT config SND_VIA82XX tristate "VIA 82C686A/B, 8233/8235 AC97 Controller" + depends on HAS_IOPORT select SND_MPU401_UART select SND_AC97_CODEC help @@ -863,6 +882,7 @@ config SND_VIA82XX config SND_VIA82XX_MODEM tristate "VIA 82C686A/B, 8233 based Modems" + depends on HAS_IOPORT select SND_AC97_CODEC help Say Y here to include support for the integrated MC97 modem on @@ -873,6 +893,7 @@ config SND_VIA82XX_MODEM config SND_VIRTUOSO tristate "Asus Virtuoso 66/100/200 (Xonar)" + depends on HAS_IOPORT select SND_OXYGEN_LIB select SND_PCM select SND_MPU401_UART @@ -889,6 +910,7 @@ config SND_VIRTUOSO config SND_VX222 tristate "Digigram VX222" + depends on HAS_IOPORT select SND_VX_LIB help Say Y here to include support for Digigram VX222 soundcards. @@ -898,6 +920,7 @@ config SND_VX222 config SND_YMFPCI tristate "Yamaha YMF724/740/744/754" + depends on HAS_IOPORT select SND_OPL3_LIB select SND_MPU401_UART select SND_AC97_CODEC diff --git a/sound/pcmcia/Kconfig b/sound/pcmcia/Kconfig index 10291c43cb18..2e3dfc1ff540 100644 --- a/sound/pcmcia/Kconfig +++ b/sound/pcmcia/Kconfig @@ -4,6 +4,7 @@ menuconfig SND_PCMCIA bool "PCMCIA sound devices" depends on PCMCIA + depends on HAS_IOPORT default y help Support for sound devices connected via the PCMCIA bus. -- 2.39.2

3 4

[PATCH] ASoC: dt-bindings: dialog,da7219: convert to dtschema
by David Rau 22 May '23

22 May '23

Convert Dialog DA7219 bindings to DT schema format. Signed-off-by: David Rau <David.Rau.opensource(a)dm.renesas.com> --- .../devicetree/bindings/sound/da7219.txt | 112 --------- .../bindings/sound/dialog,da7219.yaml | 232 ++++++++++++++++++ 2 files changed, 232 insertions(+), 112 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/da7219.txt create mode 100644 Documentation/devicetree/bindings/sound/dialog,da7219.yaml diff --git a/Documentation/devicetree/bindings/sound/da7219.txt b/Documentation/devicetree/bindings/sound/da7219.txt deleted file mode 100644 index add1caf26ac2..000000000000 --- a/Documentation/devicetree/bindings/sound/da7219.txt +++ /dev/null @@ -1,112 +0,0 @@ -Dialog Semiconductor DA7219 Audio Codec bindings - -DA7219 is an audio codec with advanced accessory detect features. - -====== - -Required properties: -- compatible : Should be "dlg,da7219" -- reg: Specifies the I2C slave address - -- interrupts : IRQ line info for DA7219. - (See Documentation/devicetree/bindings/interrupt-controller/interrupts.txt for - further information relating to interrupt properties) - -- VDD-supply: VDD power supply for the device -- VDDMIC-supply: VDDMIC power supply for the device -- VDDIO-supply: VDDIO power supply for the device - (See Documentation/devicetree/bindings/regulator/regulator.txt for further - information relating to regulators) - -Optional properties: -- interrupt-names : Name associated with interrupt line. Should be "wakeup" if - interrupt is to be used to wake system, otherwise "irq" should be used. -- wakeup-source: Flag to indicate this device can wake system (suspend/resume). - -- #clock-cells : Should be set to '<1>', two clock sources provided; -- clock-output-names : Names given for DAI clock outputs (WCLK & BCLK); - -- clocks : phandle and clock specifier for codec MCLK. -- clock-names : Clock name string for 'clocks' attribute, should be "mclk". - -- dlg,micbias-lvl : Voltage (mV) for Mic Bias - [<1600>, <1800>, <2000>, <2200>, <2400>, <2600>] -- dlg,mic-amp-in-sel : Mic input source type - ["diff", "se_p", "se_n"] - -Deprecated properties: -- dlg,ldo-lvl : Required internal LDO voltage (mV) level for digital engine - (LDO unavailable in production HW so property no longer required). - -====== - -Child node - 'da7219_aad': - -Optional properties: -- dlg,micbias-pulse-lvl : Mic bias higher voltage pulse level (mV). - [<2800>, <2900>] -- dlg,micbias-pulse-time : Mic bias higher voltage pulse duration (ms) -- dlg,btn-cfg : Periodic button press measurements for 4-pole jack (ms) - [<2>, <5>, <10>, <50>, <100>, <200>, <500>] -- dlg,mic-det-thr : Impedance threshold for mic detection measurement (Ohms) - [<200>, <500>, <750>, <1000>] -- dlg,jack-ins-deb : Debounce time for jack insertion (ms) - [<5>, <10>, <20>, <50>, <100>, <200>, <500>, <1000>] -- dlg,jack-det-rate: Jack type detection latency (3/4 pole) - ["32ms_64ms", "64ms_128ms", "128ms_256ms", "256ms_512ms"] -- dlg,jack-rem-deb : Debounce time for jack removal (ms) - [<1>, <5>, <10>, <20>] -- dlg,a-d-btn-thr : Impedance threshold between buttons A and D - [0x0 - 0xFF] -- dlg,d-b-btn-thr : Impedance threshold between buttons D and B - [0x0 - 0xFF] -- dlg,b-c-btn-thr : Impedance threshold between buttons B and C - [0x0 - 0xFF] -- dlg,c-mic-btn-thr : Impedance threshold between button C and Mic - [0x0 - 0xFF] -- dlg,btn-avg : Number of 8-bit readings for averaged button measurement - [<1>, <2>, <4>, <8>] -- dlg,adc-1bit-rpt : Repeat count for 1-bit button measurement - [<1>, <2>, <4>, <8>] - -====== - -Example: - - codec: da7219@1a { - compatible = "dlg,da7219"; - reg = <0x1a>; - - interrupt-parent = <&gpio6>; - interrupts = <11 IRQ_TYPE_LEVEL_LOW>; - - VDD-supply = <&reg_audio>; - VDDMIC-supply = <&reg_audio>; - VDDIO-supply = <&reg_audio>; - - #clock-cells = <1>; - clock-output-names = "dai-wclk", "dai-bclk"; - - clocks = <&clks 201>; - clock-names = "mclk"; - - dlg,ldo-lvl = <1200>; - dlg,micbias-lvl = <2600>; - dlg,mic-amp-in-sel = "diff"; - - da7219_aad { - dlg,btn-cfg = <50>; - dlg,mic-det-thr = <500>; - dlg,jack-ins-deb = <20>; - dlg,jack-det-rate = "32ms_64ms"; - dlg,jack-rem-deb = <1>; - - dlg,a-d-btn-thr = <0xa>; - dlg,d-b-btn-thr = <0x16>; - dlg,b-c-btn-thr = <0x21>; - dlg,c-mic-btn-thr = <0x3E>; - - dlg,btn-avg = <4>; - dlg,adc-1bit-rpt = <1>; - }; - }; diff --git a/Documentation/devicetree/bindings/sound/dialog,da7219.yaml b/Documentation/devicetree/bindings/sound/dialog,da7219.yaml new file mode 100644 index 000000000000..5015f5797901 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/dialog,da7219.yaml @@ -0,0 +1,232 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/dialog,da7219.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Dialog Semiconductor DA7219 Audio Codec + +maintainers: + - David Rau <David.Rau.opensource(a)dm.renesas.com> + +description: + The DA7219 is an ultra low-power audio codec with + in-built advanced accessory detection (AAD) for mobile + computing and accessory applications, which supports + sample rates up to 96 kHz at 24-bit resolution. + +properties: + compatible: + const: dlg,da7219 + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + VDD-supply: + description: + VDD power supply for the device. + + VDDMIC-supply: + description: + VDDMIC power supply for the device. + + VDDIO-supply: + description: + VDDIO power supply for the device. + + interrupt-names: + oneOf: + - items: + - const: wakeup + - items: + - const: irq + description: + Name associated with interrupt line. + Should be "wakeup" if interrupt is to be used to wake system, + otherwise "irq" should be used. + + wakeup-source: + type: boolean + description: + Flag to indicate this device can wake system (suspend/resume). + + "#clock-cells": + const: 1 + + clock-output-names: + minItems: 2 + maxItems: 2 + description: + Name given for DAI word clock and bit clock outputs. + + clocks: + description: + phandle and clock specifier for codec MCLK. + + clock-names: + const: mclk + + dlg,micbias-lvl: + enum: [1600, 1800, 2000, 2200, 2400, 2600] + description: + Voltage (mV) for Mic Bias. + $ref: /schemas/types.yaml#/definitions/uint32 + + dlg,mic-amp-in-sel: + enum: ["diff", "se_p", "se_n"] + description: + Mic input source type. + + diff - Differential. + + se_p - MIC_P. + Positive differential analog microphone input. + + se_n - MIC_N. + Negative differential analog microphone input. + $ref: /schemas/types.yaml#/definitions/string + + da7219_aad: + type: object + description: + Configuration of advanced accessory detection. + properties: + dlg,micbias-pulse-lvl: + enum: [2800, 2900] + description: + Mic bias higher voltage pulse level (mV). + $ref: /schemas/types.yaml#/definitions/uint32 + + dlg,micbias-pulse-time: + description: + Mic bias higher voltage pulse duration (ms). + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 0 + + dlg,btn-cfg: + enum: [2, 5, 10, 50, 100, 200, 500] + description: + Periodic button press measurements for 4-pole jack (ms). + $ref: /schemas/types.yaml#/definitions/uint32 + + dlg,mic-det-thr: + enum: [200, 500, 750, 1000] + description: + Impedance threshold for mic detection measurement (Ohms). + $ref: /schemas/types.yaml#/definitions/uint32 + + dlg,jack-ins-deb: + enum: [5, 10, 20, 50, 100, 200, 500, 1000] + description: + Debounce time for jack insertion (ms). + $ref: /schemas/types.yaml#/definitions/uint32 + + dlg,jack-det-rate: + enum: ["32ms_64ms", "64ms_128ms", "128ms_256ms", "256ms_512ms"] + description: + Jack type detection latency (3/4 pole). + $ref: /schemas/types.yaml#/definitions/string + + dlg,jack-rem-deb: + enum: [1, 5, 10, 20] + description: + Debounce time for jack removal (ms). + $ref: /schemas/types.yaml#/definitions/uint32 + + dlg,a-d-btn-thr: + description: + Impedance threshold between buttons A and D. + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 0 + maximum: 255 + + dlg,d-b-btn-thr: + description: + Impedance threshold between buttons D and B. + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 0 + maximum: 255 + + dlg,b-c-btn-thr: + description: + Impedance threshold between buttons B and C. + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 0 + maximum: 255 + + dlg,c-mic-btn-thr: + description: + Impedance threshold between button C and Mic. + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 0 + maximum: 255 + + dlg,btn-avg: + enum: [1, 2, 4, 8] + description: + Number of 8-bit readings for averaged button measurement. + $ref: /schemas/types.yaml#/definitions/uint32 + + dlg,adc-1bit-rpt: + enum: [1, 2, 4, 8] + description: + Repeat count for 1-bit button measurement. + $ref: /schemas/types.yaml#/definitions/uint32 + +required: + - compatible + - reg + - interrupts + - VDD-supply + - VDDMIC-supply + - VDDIO-supply + +additionalProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/irq.h> + i2c { + #address-cells = <1>; + #size-cells = <0>; + + codec: da7219@1a { + compatible = "dlg,da7219"; + reg = <0x1a>; + + interrupt-parent = <&gpio6>; + interrupts = <11 IRQ_TYPE_LEVEL_LOW>; + + VDD-supply = <&vdd_reg>; + VDDMIC-supply = <&vddmic_reg>; + VDDIO-supply = <&vddio_reg>; + + #clock-cells = <1>; + clock-output-names = "dai-wclk", "dai-bclk"; + + clocks = <&clks 201>; + clock-names = "mclk"; + + dlg,micbias-lvl = <2600>; + dlg,mic-amp-in-sel = "diff"; + + da7219_aad { + dlg,btn-cfg = <50>; + dlg,mic-det-thr = <500>; + dlg,jack-ins-deb = <20>; + dlg,jack-det-rate = "32ms_64ms"; + dlg,jack-rem-deb = <1>; + + dlg,a-d-btn-thr = <0xa>; + dlg,d-b-btn-thr = <0x16>; + dlg,b-c-btn-thr = <0x21>; + dlg,c-mic-btn-thr = <0x3E>; + + dlg,btn-avg = <4>; + dlg,adc-1bit-rpt = <1>; + }; + }; + }; -- 2.17.1

3 2

[PATCH] ALSA: hda/ca0132: add quirk for EVGA X299 DARK
by Adam Stylinski 22 May '23

22 May '23

This quirk is necessary for surround and other DSP effects to work with the onboard ca0132 based audio chipset for the EVGA X299 dark mainboard. Signed-off-by: Adam Stylinski <kungfujesus06(a)gmail.com> --- sound/pci/hda/patch_ca0132.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 099722ebaed8..748a3c40966e 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -1306,6 +1306,7 @@ static const struct snd_pci_quirk ca0132_quirks[] = { SND_PCI_QUIRK(0x1458, 0xA026, "Gigabyte G1.Sniper Z97", QUIRK_R3DI), SND_PCI_QUIRK(0x1458, 0xA036, "Gigabyte GA-Z170X-Gaming 7", QUIRK_R3DI), SND_PCI_QUIRK(0x3842, 0x1038, "EVGA X99 Classified", QUIRK_R3DI), + SND_PCI_QUIRK(0x3842, 0x104b, "EVGA X299 Dark", QUIRK_R3DI), SND_PCI_QUIRK(0x3842, 0x1055, "EVGA Z390 DARK", QUIRK_R3DI), SND_PCI_QUIRK(0x1102, 0x0013, "Recon3D", QUIRK_R3D), SND_PCI_QUIRK(0x1102, 0x0018, "Recon3D", QUIRK_R3D), -- 2.40.1

2 1

[PATCH v3 3/5] ASoC: tas2781: Add tas2781 driver
by Shenghao Ding 22 May '23

22 May '23

Create tas2781 driver. Signed-off-by: Shenghao Ding <13916275206(a)139.com> --- Changes in v3: - fixed issue | Reported-by: kernel test robot <lkp(a)intel.com> | Link: https://lore.kernel.org/oe-kbuild-all/202305022033.LiI7Ojm4-lkp@intel.com/ - fixed issue | Reported-by: kernel test robot <lkp(a)intel.com> | Link: https://lore.kernel.org/oe-kbuild-all/202305022338.ZA1okoZW-lkp@intel.com/ Changes to be committed: modified: sound/soc/codecs/Kconfig modified: sound/soc/codecs/Makefile new file: sound/soc/codecs/tas2781-comlib.c new file: sound/soc/codecs/tas2781-i2c.c --- sound/soc/codecs/Kconfig | 24 + sound/soc/codecs/Makefile | 6 + sound/soc/codecs/tas2781-comlib.c | 581 ++++++++++++++++++++++++ sound/soc/codecs/tas2781-i2c.c | 722 ++++++++++++++++++++++++++++++ 4 files changed, 1333 insertions(+) create mode 100644 sound/soc/codecs/tas2781-comlib.c create mode 100644 sound/soc/codecs/tas2781-i2c.c diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 8020097d4e4c..4ee7d8fee526 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -234,6 +234,9 @@ config SND_SOC_ALL_CODECS imply SND_SOC_TAS2764 imply SND_SOC_TAS2770 imply SND_SOC_TAS2780 + imply SND_SOC_TAS2781_COMLIB + imply SND_SOC_TAS2781_FMWLIB + imply SND_SOC_TAS2781 imply SND_SOC_TAS5086 imply SND_SOC_TAS571X imply SND_SOC_TAS5720 @@ -1699,6 +1702,27 @@ config SND_SOC_TAS2780 Enable support for Texas Instruments TAS2780 high-efficiency digital input mono Class-D audio power amplifiers. +config SND_SOC_TAS2781_COMLIB + select CRC8 + select REGMAP_I2C + tristate + +config SND_SOC_TAS2781_FMWLIB + tristate + default n + +config SND_SOC_TAS2781 + tristate "Texas Instruments TAS2781 speaker amplifier" + depends on I2C + select SND_SOC_TAS2781_COMLIB + select SND_SOC_TAS2781_FMWLIB + help + Enable support for Texas Instruments TAS2781 Smart Amplifier + Digital input mono Class-D and DSP-inside audio power amplifiers. + Note the TAS2781 driver implements a flexible and configurable + algo coefficient setting, for one, two or even multiple TAS2781 + chips. + config SND_SOC_TAS5086 tristate "Texas Instruments TAS5086 speaker amplifier" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 5cdbae88e6e3..97c3881ed587 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -269,6 +269,9 @@ snd-soc-tas5805m-objs := tas5805m.o snd-soc-tas6424-objs := tas6424.o snd-soc-tda7419-objs := tda7419.o 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-objs := tas2781-i2c.o snd-soc-tfa9879-objs := tfa9879.o snd-soc-tfa989x-objs := tfa989x.o snd-soc-tlv320adc3xxx-objs := tlv320adc3xxx.o @@ -633,6 +636,9 @@ obj-$(CONFIG_SND_SOC_TAS2552) += snd-soc-tas2552.o obj-$(CONFIG_SND_SOC_TAS2562) += snd-soc-tas2562.o obj-$(CONFIG_SND_SOC_TAS2764) += snd-soc-tas2764.o obj-$(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-$(CONFIG_SND_SOC_TAS2781) += snd-soc-tas2781.o obj-$(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/tas2781-comlib.c b/sound/soc/codecs/tas2781-comlib.c new file mode 100644 index 000000000000..5994698326b1 --- /dev/null +++ b/sound/soc/codecs/tas2781-comlib.c @@ -0,0 +1,581 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// tas2781-lib.c -- TAS2781 Common functions for HDA and ASoC Audio drivers +// +// Copyright 2023 Texas Instruments, Inc. +// +// Author: Shenghao Ding <shenghao-ding(a)ti.com> + +#include <linux/crc8.h> +#include <linux/firmware.h> +#include <linux/gpio/consumer.h> +#include <linux/i2c.h> +#include <linux/init.h> +#include <linux/interrupt.h> +#include <linux/module.h> +#include <linux/of.h> +#include <linux/of_gpio.h> +#include <linux/of_irq.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tas2781.h> + +#define TASDEVICE_CRC8_POLYNOMIAL 0x4d + +static const struct regmap_range_cfg tasdevice_ranges[] = { + { + .range_min = 0, + .range_max = 256 * 128, + .selector_reg = TASDEVICE_PAGE_SELECT, + .selector_mask = 0xff, + .selector_shift = 0, + .window_start = 0, + .window_len = 128, + }, +}; + +static const struct regmap_config tasdevice_regmap = { + .reg_bits = 8, + .val_bits = 8, + .cache_type = REGCACHE_RBTREE, + .ranges = tasdevice_ranges, + .num_ranges = ARRAY_SIZE(tasdevice_ranges), + .max_register = 256 * 128, +}; + +static int tasdevice_change_chn_book(struct tasdevice_priv *tas_priv, + enum channel chn, int book) +{ + struct i2c_client *client = (struct i2c_client *)tas_priv->client; + int ret = 0; + int i; + + if (chn < tas_priv->ndev) { + struct tasdevice *tasdev = &tas_priv->tasdevice[chn]; + + if (tas_priv->glb_addr.ref_cnt != 0) { + tas_priv->glb_addr.ref_cnt = 0; + tas_priv->glb_addr.cur_book = -1; + } + client->addr = tasdev->dev_addr; + if (tasdev->cur_book != book) { + ret = regmap_write(tas_priv->regmap, + TASDEVICE_BOOKCTL_REG, book); + if (ret < 0) { + dev_err(tas_priv->dev, "%s, E=%d\n", + __func__, ret); + goto out; + } + tasdev->cur_book = book; + } + } else if (chn == tas_priv->ndev) { + struct global_addr *glb_addr = &tas_priv->glb_addr; + + /* Chn == device number will enable global broadcast mode*/ + if (tas_priv->glb_addr.ref_cnt == 0) + for (i = 0; i < tas_priv->ndev; i++) + tas_priv->tasdevice[i].cur_book = -1; + client->addr = glb_addr->dev_addr; + if (glb_addr->cur_book != book) { + ret = regmap_write(tas_priv->regmap, + TASDEVICE_BOOKCTL_REG, book); + if (ret < 0) { + dev_err(tas_priv->dev, "%s, book%x, E=%d\n", + __func__, book, ret); + goto out; + } + glb_addr->cur_book = book; + } + + glb_addr->ref_cnt++; + } else { + ret = -EINVAL; + dev_err(tas_priv->dev, "%s, no such channel(%d)\n", __func__, + chn); + } + +out: + return ret; +} + +int tasdevice_dev_read(struct tasdevice_priv *tas_priv, + enum channel chn, unsigned int reg, unsigned int *val) +{ + int ret = 0; + + if (chn < tas_priv->ndev) { + ret = tasdevice_change_chn_book(tas_priv, chn, + TASDEVICE_BOOK_ID(reg)); + if (ret < 0) + goto out; + + ret = regmap_read(tas_priv->regmap, TASDEVICE_PGRG(reg), val); + if (ret < 0) + dev_err(tas_priv->dev, "%s, E=%d\n", __func__, ret); + } else { + ret = -EINVAL; + dev_err(tas_priv->dev, "%s, no such channel(%d)\n", __func__, + chn); + } + +out: + return ret; +} +EXPORT_SYMBOL_GPL(tasdevice_dev_read); + +int tasdevice_dev_write(struct tasdevice_priv *tas_priv, + enum channel chn, unsigned int reg, unsigned int value) +{ + int ret = 0; + + /* chn <= tas_dev->ndev will support global broadcast mode*/ + if (chn <= tas_priv->ndev) { + ret = tasdevice_change_chn_book(tas_priv, chn, + TASDEVICE_BOOK_ID(reg)); + if (ret < 0) + goto out; + + ret = regmap_write(tas_priv->regmap, TASDEVICE_PGRG(reg), + value); + if (ret < 0) + dev_err(tas_priv->dev, "%s, E=%d\n", __func__, ret); + } else { + ret = -EINVAL; + dev_err(tas_priv->dev, "%s, no such channel(%d)\n", __func__, + chn); + } + +out: + return ret; +} +EXPORT_SYMBOL_GPL(tasdevice_dev_write); + +int tasdevice_dev_bulk_write( + struct tasdevice_priv *tas_priv, enum channel chn, + unsigned int reg, unsigned char *data, + unsigned int len) +{ + int ret = 0; + + /* chn <= tas_dev->ndev will support global broadcast mode*/ + if (chn <= tas_priv->ndev) { + ret = tasdevice_change_chn_book(tas_priv, chn, + TASDEVICE_BOOK_ID(reg)); + if (ret < 0) + goto out; + + ret = regmap_bulk_write(tas_priv->regmap, TASDEVICE_PGRG(reg), + data, len); + if (ret < 0) + dev_err(tas_priv->dev, "%s, E=%d\n", __func__, ret); + } else { + ret = -EINVAL; + dev_err(tas_priv->dev, "%s, no such channel(%d)\n", __func__, + chn); + } + +out: + return ret; +} +EXPORT_SYMBOL_GPL(tasdevice_dev_bulk_write); + +int tasdevice_dev_bulk_read(struct tasdevice_priv *tas_priv, + enum channel chn, unsigned int reg, unsigned char *data, + unsigned int len) +{ + int ret = 0; + + if (chn < tas_priv->ndev) { + ret = tasdevice_change_chn_book(tas_priv, chn, + TASDEVICE_BOOK_ID(reg)); + if (ret < 0) + goto out; + + ret = regmap_bulk_read(tas_priv->regmap, TASDEVICE_PGRG(reg), + data, len); + if (ret < 0) + dev_err(tas_priv->dev, "%s, E=%d\n", __func__, ret); + } else + dev_err(tas_priv->dev, "%s, no such channel(%d)\n", __func__, + chn); + +out: + return ret; +} +EXPORT_SYMBOL_GPL(tasdevice_dev_bulk_read); + +int tasdevice_dev_update_bits( + struct tasdevice_priv *tas_priv, enum channel chn, + unsigned int reg, unsigned int mask, unsigned int value) +{ + int ret = 0; + + /* update-bits can't support global broadcast mode, + * because this operation contains a reading operation + * for register value. In global mode, reading register + * from all of the tas2781 will cause confusion. + */ + if (chn < tas_priv->ndev) { + ret = tasdevice_change_chn_book(tas_priv, chn, + TASDEVICE_BOOK_ID(reg)); + if (ret < 0) + goto out; + + ret = regmap_update_bits(tas_priv->regmap, + TASDEVICE_PGRG(reg), mask, value); + if (ret < 0) + dev_err(tas_priv->dev, "%s, E=%d\n", __func__, ret); + } else { + dev_err(tas_priv->dev, "%s, no such channel(%d)\n", __func__, + chn); + ret = -EINVAL; + } + +out: + return ret; +} +EXPORT_SYMBOL_GPL(tasdevice_dev_update_bits); + +/* Only register write and bulk write support global mode + * Other cases, such as bits update, register read and bulk + * only support device-by-device operation + */ +static void tas2781_set_global_mode(struct tasdevice_priv *tas_priv) +{ + int i, ret; + + for (i = 0; i < tas_priv->ndev; i++) { + ret = tasdevice_dev_update_bits(tas_priv, i, + TAS2781_MISC_CFG2, TAS2781_GLOBAL_ADDR_MASK, + TAS2781_GLOBAL_ADDR_ENABLE); + if (ret < 0) { + dev_err(tas_priv->dev, "%s: chn %d set glb fail, %d\n", + __func__, i, ret); + continue; + } + } +} + +struct tasdevice_priv *tasdevice_kzalloc(struct i2c_client *i2c) +{ + struct tasdevice_priv *tas_priv; + + tas_priv = devm_kzalloc(&i2c->dev, sizeof(*tas_priv), GFP_KERNEL); + if (!tas_priv) + return NULL; + tas_priv->dev = &i2c->dev; + tas_priv->client = (void *)i2c; + + return tas_priv; +} +EXPORT_SYMBOL_GPL(tasdevice_kzalloc); + +void tas2781_reset(struct tasdevice_priv *tas_dev) +{ + int ret, i; + + if (tas_dev->reset) { + gpiod_set_value_cansleep(tas_dev->reset, 0); + usleep_range(500, 1000); + gpiod_set_value_cansleep(tas_dev->reset, 1); + } else { + for (i = 0; i < tas_dev->ndev; i++) { + ret = tasdevice_dev_write(tas_dev, i, + TAS2781_REG_SWRESET, + TAS2781_REG_SWRESET_RESET); + if (ret < 0) + dev_err(tas_dev->dev, + "dev %d swreset fail, %d\n", + i, ret); + } + } + usleep_range(1000, 1050); +} +EXPORT_SYMBOL_GPL(tas2781_reset); + +int tascodec_init(struct tasdevice_priv *tas_priv, void *codec, + void (*cont)(const struct firmware *fw, void *context)) +{ + int ret = 0; + + /* Codec Lock Hold to ensure that codec_probe and firmware parsing and + * loading do not simultaneously execute. + */ + mutex_lock(&tas_priv->codec_lock); + + scnprintf(tas_priv->rca_binaryname, 64, "%srca%d.bin", + tas_priv->dev_name, tas_priv->ndev); + crc8_populate_msb(tas_priv->crc8_lkp_tbl, TASDEVICE_CRC8_POLYNOMIAL); + tas_priv->codec = codec; + ret = request_firmware_nowait(THIS_MODULE, FW_ACTION_UEVENT, + tas_priv->rca_binaryname, tas_priv->dev, GFP_KERNEL, tas_priv, + cont); + if (ret) + dev_err(tas_priv->dev, "request_firmware_nowait err:0x%08x\n", + ret); + + tas2781_reset(tas_priv); + if (tas_priv->set_global_mode) + tas_priv->set_global_mode(tas_priv); + + /* Codec Lock Release*/ + mutex_unlock(&tas_priv->codec_lock); + return ret; +} +EXPORT_SYMBOL_GPL(tascodec_init); + +int tasdevice_init(struct tasdevice_priv *tas_priv) +{ + int ret = 0; + int i; + + tas_priv->regmap = devm_regmap_init_i2c(tas_priv->client, + &tasdevice_regmap); + if (IS_ERR(tas_priv->regmap)) { + ret = PTR_ERR(tas_priv->regmap); + dev_err(tas_priv->dev, "Failed to allocate register map: %d\n", + ret); + goto out; + } + + tas_priv->cur_prog = -1; + tas_priv->cur_conf = -1; + + for (i = 0; i < tas_priv->ndev; i++) { + tas_priv->tasdevice[i].cur_book = -1; + tas_priv->tasdevice[i].cur_prog = -1; + tas_priv->tasdevice[i].cur_conf = -1; + } + + if (tas_priv->glb_addr.dev_addr != 0 + && tas_priv->glb_addr.dev_addr < 0x7F) + tas_priv->set_global_mode = tas2781_set_global_mode; + dev_set_drvdata(tas_priv->dev, tas_priv); + + mutex_init(&tas_priv->codec_lock); + +out: + return ret; +} +EXPORT_SYMBOL_GPL(tasdevice_init); + +static void tasdev_dsp_prog_blk_remove(struct tasdevice_prog *prog) +{ + struct tasdevice_data *im; + struct tasdev_blk *blk; + unsigned int nr_blk; + + if (!prog) + return; + + im = &(prog->dev_data); + + if (!im->dev_blks) + return; + + for (nr_blk = 0; nr_blk < im->nr_blk; nr_blk++) { + blk = &(im->dev_blks[nr_blk]); + kfree(blk->data); + } + kfree(im->dev_blks); +} + +static void tasdev_dsp_prog_remove(struct tasdevice_prog *prog, + unsigned short nr) +{ + int i; + + for (i = 0; i < nr; i++) + tasdev_dsp_prog_blk_remove(&prog[i]); + kfree(prog); +} + +static void tasdev_dsp_cfg_blk_remove(struct tasdevice_config *cfg) +{ + struct tasdevice_data *im; + struct tasdev_blk *blk; + unsigned int nr_blk; + + if (cfg) { + im = &(cfg->dev_data); + + if (!im->dev_blks) + return; + + for (nr_blk = 0; nr_blk < im->nr_blk; nr_blk++) { + blk = &(im->dev_blks[nr_blk]); + kfree(blk->data); + } + kfree(im->dev_blks); + } +} + +static void tasdev_dsp_cfg_remove(struct tasdevice_config *config, + unsigned short nr) +{ + int i; + + for (i = 0; i < nr; i++) + tasdev_dsp_cfg_blk_remove(&config[i]); + kfree(config); +} + +void tasdevice_dsp_remove(void *context) +{ + struct tasdevice_priv *tas_dev = (struct tasdevice_priv *) context; + struct tasdevice_fw *tas_fmw = tas_dev->fmw; + + if (!tas_dev->fmw) + return; + + if (tas_fmw->programs) + tasdev_dsp_prog_remove(tas_fmw->programs, + tas_fmw->nr_programs); + if (tas_fmw->configs) + tasdev_dsp_cfg_remove(tas_fmw->configs, + tas_fmw->nr_configurations); + kfree(tas_fmw); + tas_dev->fmw = NULL; +} +EXPORT_SYMBOL_GPL(tasdevice_dsp_remove); + +void tasdevice_remove(struct tasdevice_priv *tas_priv) +{ + if (gpio_is_valid(tas_priv->irq_info.irq_gpio)) + gpio_free(tas_priv->irq_info.irq_gpio); + kfree(tas_priv->acpi_subsystem_id); + mutex_destroy(&tas_priv->codec_lock); +} +EXPORT_SYMBOL_GPL(tasdevice_remove); + +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; +} + +int tasdevice_amp_putvol(struct tasdevice_priv *tas_priv, + struct snd_ctl_elem_value *ucontrol, struct soc_mixer_control *mc) +{ + unsigned int invert = mc->invert; + unsigned char mask; + int max = mc->max; + int err_cnt = 0; + int val, i, ret; + + mask = (1 << fls(max)) - 1; + mask <<= mc->shift; + val = tasdevice_clamp(ucontrol->value.integer.value[0], max, invert); + for (i = 0; i < tas_priv->ndev; i++) { + ret = tasdevice_dev_update_bits(tas_priv, i, + mc->reg, mask, (unsigned int)(val << mc->shift)); + if (!ret) + continue; + err_cnt++; + dev_err(tas_priv->dev, "set AMP vol error in dev %d\n", i); + } + + /* All the devices set error, return 0 */ + return (err_cnt == tas_priv->ndev) ? 0 : 1; +} +EXPORT_SYMBOL_GPL(tasdevice_amp_putvol); + +int tasdevice_amp_getvol(struct tasdevice_priv *tas_priv, + struct snd_ctl_elem_value *ucontrol, struct soc_mixer_control *mc) +{ + unsigned int invert = mc->invert; + unsigned char mask = 0; + int max = mc->max; + int ret = 0; + int val; + + /* Read the primary device */ + ret = tasdevice_dev_read(tas_priv, 0, mc->reg, &val); + if (ret) { + dev_err(tas_priv->dev, "%s, get AMP vol error\n", __func__); + goto out; + } + + mask = (1 << fls(max)) - 1; + mask <<= mc->shift; + val = (val & mask) >> mc->shift; + val = tasdevice_clamp(val, max, invert); + ucontrol->value.integer.value[0] = val; + +out: + return ret; + +} +EXPORT_SYMBOL_GPL(tasdevice_amp_getvol); + +int tasdevice_digital_putvol(struct tasdevice_priv *tas_priv, + struct snd_ctl_elem_value *ucontrol, struct soc_mixer_control *mc) +{ + unsigned int invert = mc->invert; + int max = mc->max; + int err_cnt = 0; + int ret = 1; + int val, i; + + val = tasdevice_clamp(ucontrol->value.integer.value[0], max, invert); + + if (tas_priv->set_global_mode) { + ret = tasdevice_dev_write(tas_priv, tas_priv->ndev, + mc->reg, (unsigned int)val); + if (ret) + dev_err(tas_priv->dev, "%s, error in global mode\n", + __func__); + } + + if (!tas_priv->set_global_mode || ret != 0) { + for (i = 0; i < tas_priv->ndev; i++) { + ret = tasdevice_dev_write(tas_priv, i, mc->reg, + (unsigned int)val); + if (!ret) + continue; + err_cnt++; + dev_err(tas_priv->dev, + "set digital vol err in dev %d\n", i); + } + } + + /* All the devices set error, return 0 */ + return (err_cnt == tas_priv->ndev) ? 0 : 1; + +} +EXPORT_SYMBOL_GPL(tasdevice_digital_putvol); + +int tasdevice_digital_getvol(struct tasdevice_priv *tas_priv, + struct snd_ctl_elem_value *ucontrol, struct soc_mixer_control *mc) +{ + unsigned int invert = mc->invert; + int max = mc->max; + int ret, val; + + /* Read the primary device as the whole */ + ret = tasdevice_dev_read(tas_priv, 0, mc->reg, &val); + if (ret) { + dev_err(tas_priv->dev, "%s, get digital vol error\n", + __func__); + goto out; + } + + val = tasdevice_clamp(val, max, invert); + ucontrol->value.integer.value[0] = val; + +out: + return ret; + +} +EXPORT_SYMBOL_GPL(tasdevice_digital_getvol); + +MODULE_DESCRIPTION("TAS2781 common library"); +MODULE_AUTHOR("Shenghao Ding, TI, <shenghao-ding(a)ti.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c new file mode 100644 index 000000000000..343cb04d1ecf --- /dev/null +++ b/sound/soc/codecs/tas2781-i2c.c @@ -0,0 +1,722 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// ALSA SoC Texas Instruments TAS2781 Audio Smart Amplifier +// +// Copyright (C) 2022 - 2023 Texas Instruments Incorporated +// https://www.ti.com +// +// The TAS2781 driver implements a flexible and configurable +// algo coefficient setting for one, two, or even multiple +// TAS2781 chips. +// +// Author: Shenghao Ding <shenghao-ding(a)ti.com> +// Author: Kevin Lu <kevin-lu(a)ti.com> +// + +#include <linux/crc8.h> +#include <linux/firmware.h> +#include <linux/gpio/consumer.h> +#include <linux/i2c.h> +#include <linux/init.h> +#include <linux/interrupt.h> +#include <linux/module.h> +#include <linux/of.h> +#include <linux/of_gpio.h> +#include <linux/of_irq.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tas2781.h> +#include <sound/tlv.h> +#include <sound/tas2781-tlv.h> + +static const struct i2c_device_id tasdevice_id[] = { + { "tas2781", TAS2781 }, + {} +}; +MODULE_DEVICE_TABLE(i2c, tasdevice_id); + +#ifdef CONFIG_OF +static const struct of_device_id tasdevice_of_match[] = { + { .compatible = "ti,tas2781" }, + {}, +}; +MODULE_DEVICE_TABLE(of, tasdevice_of_match); +#endif + +/** + * tas2781_digital_getvol - get the volum control + * @kcontrol: control pointer + * @ucontrol: User data + * Customer Kcontrol for tas2781 is primarily for regmap booking, paging + * depends on internal regmap mechanism. + * tas2781 contains book and page two-level register map, especially + * book switching will set the register BXXP00R7F, after switching to the + * correct book, then leverage the mechanism for paging to access the + * register. + */ +static int tas2781_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_priv = snd_soc_component_get_drvdata(codec); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + + return tasdevice_digital_getvol(tas_priv, ucontrol, mc); +} + +static int tas2781_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_priv = snd_soc_component_get_drvdata(codec); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + + return tasdevice_digital_putvol(tas_priv, ucontrol, mc); +} + +static int tas2781_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_priv = snd_soc_component_get_drvdata(codec); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + + return tasdevice_amp_getvol(tas_priv, ucontrol, mc); +} + +static int tas2781_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_priv = + snd_soc_component_get_drvdata(codec); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + + return tasdevice_amp_putvol(tas_priv, ucontrol, mc); +} + +static const struct snd_kcontrol_new tas2781_snd_controls[] = { + SOC_SINGLE_RANGE_EXT_TLV("Amp Gain Volume", TAS2781_AMP_LEVEL, + 1, 0, 20, 0, tas2781_amp_getvol, + tas2781_amp_putvol, amp_vol_tlv), + SOC_SINGLE_RANGE_EXT_TLV("Digital Volume", TAS2781_DVC_LVL, + 0, 0, 200, 1, tas2781_digital_getvol, + tas2781_digital_putvol, dvc_tlv), +}; + +static int tasdevice_set_profile_id(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); + + tas_priv->rcabin.profile_cfg_id = ucontrol->value.integer.value[0]; + + return 1; +} + +static int tasdevice_info_programs(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); + struct tasdevice_fw *tas_fw = tas_priv->fmw; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = (int)tas_fw->nr_programs; + + return 0; +} + +static int tasdevice_info_configurations( + struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) +{ + struct snd_soc_component *codec = + snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); + struct tasdevice_fw *tas_fw = tas_priv->fmw; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = (int)tas_fw->nr_configurations - 1; + + return 0; +} + +static int tasdevice_info_profile(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 1; + uinfo->value.integer.max = max(0, tas_priv->rcabin.ncfgs); + + return 0; +} + +static int tasdevice_get_profile_id(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); + + ucontrol->value.integer.value[0] = tas_priv->rcabin.profile_cfg_id; + + return 0; +} + +static int tasdevice_create_control(struct tasdevice_priv *tas_priv) +{ + struct snd_kcontrol_new *prof_ctrls; + int nr_controls = 1; + int mix_index = 0; + int ret; + char *name; + + prof_ctrls = devm_kcalloc(tas_priv->dev, nr_controls, + sizeof(prof_ctrls[0]), GFP_KERNEL); + if (!prof_ctrls) { + ret = -ENOMEM; + goto out; + } + + /* Create a mixer item for selecting the active profile */ + name = devm_kzalloc(tas_priv->dev, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, + GFP_KERNEL); + if (!name) { + ret = -ENOMEM; + goto out; + } + scnprintf(name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, "tasdev-profile-id"); + prof_ctrls[mix_index].name = name; + prof_ctrls[mix_index].iface = SNDRV_CTL_ELEM_IFACE_MIXER; + prof_ctrls[mix_index].info = tasdevice_info_profile; + prof_ctrls[mix_index].get = tasdevice_get_profile_id; + prof_ctrls[mix_index].put = tasdevice_set_profile_id; + mix_index++; + + ret = snd_soc_add_component_controls(tas_priv->codec, + prof_ctrls, nr_controls < mix_index ? nr_controls : mix_index); + +out: + return ret; +} + +static int tasdevice_program_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); + + ucontrol->value.integer.value[0] = tas_priv->cur_prog; + + return 0; +} + +static int tasdevice_program_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); + unsigned int nr_program = ucontrol->value.integer.value[0]; + + tas_priv->cur_prog = nr_program; + + return 1; +} + +static int tasdevice_configuration_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); + + ucontrol->value.integer.value[0] = tas_priv->cur_conf; + + return 0; +} + +static int tasdevice_configuration_put( + struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); + unsigned int nr_configuration = ucontrol->value.integer.value[0]; + + tas_priv->cur_conf = nr_configuration; + + return 1; +} + +static int tasdevice_dsp_create_ctrls( + struct tasdevice_priv *tas_priv) +{ + struct snd_kcontrol_new *dsp_ctrls; + char *prog_name, *conf_name; + int nr_controls = 2; + int mix_index = 0; + int ret; + + /* Alloc kcontrol via devm_kzalloc, which don't manually + * free the kcontrol + */ + dsp_ctrls = devm_kcalloc(tas_priv->dev, nr_controls, + sizeof(dsp_ctrls[0]), GFP_KERNEL); + if (!dsp_ctrls) { + ret = -ENOMEM; + goto out; + } + + /* Create a mixer item for selecting the active profile */ + prog_name = devm_kzalloc(tas_priv->dev, + SNDRV_CTL_ELEM_ID_NAME_MAXLEN, GFP_KERNEL); + conf_name = devm_kzalloc(tas_priv->dev, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, + GFP_KERNEL); + if (!prog_name || !conf_name) { + ret = -ENOMEM; + goto out; + } + + scnprintf(prog_name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, + "tasdev-prog-id"); + dsp_ctrls[mix_index].name = prog_name; + dsp_ctrls[mix_index].iface = SNDRV_CTL_ELEM_IFACE_MIXER; + dsp_ctrls[mix_index].info = tasdevice_info_programs; + dsp_ctrls[mix_index].get = tasdevice_program_get; + dsp_ctrls[mix_index].put = tasdevice_program_put; + mix_index++; + + scnprintf(conf_name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, + "tasdev-conf-id"); + dsp_ctrls[mix_index].name = conf_name; + dsp_ctrls[mix_index].iface = SNDRV_CTL_ELEM_IFACE_MIXER; + dsp_ctrls[mix_index].info = tasdevice_info_configurations; + dsp_ctrls[mix_index].get = tasdevice_configuration_get; + dsp_ctrls[mix_index].put = tasdevice_configuration_put; + mix_index++; + + ret = snd_soc_add_component_controls(tas_priv->codec, dsp_ctrls, + nr_controls < mix_index ? nr_controls : mix_index); + +out: + return ret; +} + +static void tasdevice_fw_ready(const struct firmware *fmw, + void *context) +{ + struct tasdevice_priv *tas_priv = (struct tasdevice_priv *)context; + int ret = 0; + int i; + + mutex_lock(&tas_priv->codec_lock); + + ret = tasdevice_rca_parser(tas_priv, fmw); + if (ret) + goto out; + tasdevice_create_control(tas_priv); + + tasdevice_dsp_remove(tas_priv); + tasdevice_calbin_remove(tas_priv); + tas_priv->fw_state = TASDEVICE_DSP_FW_PENDING; + scnprintf(tas_priv->coef_binaryname, 64, "%s_coef.bin", + tas_priv->dev_name); + ret = tasdevice_dsp_parser(tas_priv); + if (ret) { + dev_err(tas_priv->dev, "dspfw load %s error\n", + tas_priv->coef_binaryname); + tas_priv->fw_state = TASDEVICE_DSP_FW_FAIL; + goto out; + } + tasdevice_dsp_create_ctrls(tas_priv); + + tas_priv->fw_state = TASDEVICE_DSP_FW_ALL_OK; + + /* If calibrated data occurs error, dsp will still works with default + * calibrated data inside algo. + */ + for (i = 0; i < tas_priv->ndev; i++) { + scnprintf(tas_priv->cal_binaryname[i], 64, "%s_cal_0x%02x.bin", + tas_priv->dev_name, tas_priv->tasdevice[i].dev_addr); + ret = tas2781_load_calibration(tas_priv, + tas_priv->cal_binaryname[i], i); + if (ret != 0) + dev_err(tas_priv->dev, + "%s: load %s error, default will effect\n", + __func__, tas_priv->cal_binaryname[i]); + } + + tasdevice_prmg_calibdata_load(tas_priv, 0); + tas_priv->cur_prog = 0; +out: + if (tas_priv->fw_state == TASDEVICE_DSP_FW_FAIL) { + /*If DSP FW fail, kcontrol won't be created */ + tasdevice_config_info_remove(tas_priv); + tasdevice_dsp_remove(tas_priv); + } + mutex_unlock(&tas_priv->codec_lock); + if (fmw) + release_firmware(fmw); +} + +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_priv = snd_soc_component_get_drvdata(codec); + int state = 0; + + /* Codec Lock Hold */ + mutex_lock(&tas_priv->codec_lock); + if (event == SND_SOC_DAPM_PRE_PMD) + state = 1; + tasdevice_tuning_switch(tas_priv, state); + /* Codec Lock Release*/ + mutex_unlock(&tas_priv->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) +{ + struct snd_soc_component *codec = dai->component; + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); + int ret = 0; + + if (tas_priv->fw_state != TASDEVICE_DSP_FW_ALL_OK) { + dev_err(tas_priv->dev, "DSP bin file not loaded\n"); + ret = -EINVAL; + } + + return ret; +} + +static int tasdevice_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct tasdevice_priv *tas_priv = snd_soc_dai_get_drvdata(dai); + unsigned int slot_width; + unsigned int fsrate; + int bclk_rate; + int rc = 0; + + fsrate = params_rate(params); + switch (fsrate) { + case 48000: + case 44100: + break; + default: + dev_err(tas_priv->dev, "%s: incorrect sample rate = %u\n", + __func__, fsrate); + rc = -EINVAL; + goto out; + } + + slot_width = params_width(params); + switch (slot_width) { + case 16: + case 20: + case 24: + case 32: + break; + default: + dev_err(tas_priv->dev, "%s: incorrect slot width = %u\n", + __func__, slot_width); + rc = -EINVAL; + goto out; + } + + bclk_rate = snd_soc_params_to_bclk(params); + if (bclk_rate < 0) { + dev_err(tas_priv->dev, "%s: incorrect bclk rate = %d\n", + __func__, bclk_rate); + rc = bclk_rate; + goto out; + } + +out: + return rc; +} + +static int tasdevice_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct tasdevice_priv *tas_priv = snd_soc_dai_get_drvdata(codec_dai); + + tas_priv->sysclk = freq; + + return 0; +} + +static const struct snd_soc_dai_ops tasdevice_dai_ops = { + .startup = tasdevice_startup, + .hw_params = tasdevice_hw_params, + .set_sysclk = tasdevice_set_dai_sysclk, +}; + +static struct snd_soc_dai_driver tasdevice_dai_driver[] = { + { + .name = "tas2781_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 = 4, + .rates = TASDEVICE_RATES, + .formats = TASDEVICE_FORMATS, + }, + .ops = &tasdevice_dai_ops, + .symmetric_rate = 1, + }, +}; + +static int tasdevice_codec_probe(struct snd_soc_component *codec) +{ + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); + + return tascodec_init(tas_priv, codec, tasdevice_fw_ready); +} + +static void tasdevice_deinit(void *context) +{ + struct tasdevice_priv *tas_priv = (struct tasdevice_priv *) context; + + tasdevice_config_info_remove(tas_priv); + tasdevice_dsp_remove(tas_priv); + tasdevice_calbin_remove(tas_priv); + tas_priv->fw_state = TASDEVICE_DSP_FW_PENDING; +} + +static void tasdevice_codec_remove( + struct snd_soc_component *codec) +{ + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); + + tasdevice_deinit(tas_priv); +} + +static const struct snd_soc_component_driver + soc_codec_driver_tasdevice = { + .probe = tasdevice_codec_probe, + .remove = tasdevice_codec_remove, + .controls = tas2781_snd_controls, + .num_controls = ARRAY_SIZE(tas2781_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, +}; + +static void tasdevice_parse_dt(struct tasdevice_priv *tas_priv) +{ + struct i2c_client *client = (struct i2c_client *)tas_priv->client; + unsigned int dev_addrs[max_chn]; + int rc, i, ndev = 0; + + if (tas_priv->isacpi) { + ndev = device_property_read_u32_array(&client->dev, + "ti,audio-slots", NULL, 0); + if (ndev <= 0) { + ndev = 1; + dev_addrs[0] = client->addr; + } else { + ndev = (ndev < ARRAY_SIZE(dev_addrs)) + ? ndev : ARRAY_SIZE(dev_addrs); + ndev = device_property_read_u32_array(&client->dev, + "ti,audio-slots", dev_addrs, ndev); + } + + tas_priv->irq_info.irq_gpio = + acpi_dev_gpio_irq_get(ACPI_COMPANION(&client->dev), 0); + } else { + struct device_node *np = tas_priv->dev->of_node; +#ifdef CONFIG_OF + const unsigned int *reg, *reg_end; + int len, sw, aw; + + aw = of_n_addr_cells(np); + sw = of_n_size_cells(np); + if (sw == 0) { + reg = (const unsigned int *)of_get_property(np, + "reg", &len); + reg_end = reg + len/sizeof(*reg); + ndev = 0; + do { + dev_addrs[ndev] = of_read_number(reg, aw); + reg += aw; + ndev++; + } while (reg < reg_end); + } else { + ndev = 1; + dev_addrs[0] = client->addr; + } +#else + ndev = 1; + dev_addrs[0] = client->addr; +#endif + tas_priv->irq_info.irq_gpio = of_irq_get(np, 0); + } + tas_priv->ndev = ndev; + for (i = 0; i < ndev; i++) + tas_priv->tasdevice[i].dev_addr = dev_addrs[i]; + + if (ndev > 1) { + rc = device_property_read_u32(&client->dev, + "ti,broadcast-addr", + &(tas_priv->glb_addr.dev_addr)); + if (rc) { + dev_err(tas_priv->dev, + "Looking up broadcast-addr failed %d\n", rc); + tas_priv->glb_addr.dev_addr = 0; + } + } + + tas_priv->reset = devm_gpiod_get_optional(&client->dev, + "reset-gpios", GPIOD_OUT_HIGH); + if (IS_ERR(tas_priv->reset)) + dev_err(tas_priv->dev, "%s Can't get reset GPIO\n", + __func__); + + strcpy(tas_priv->dev_name, tasdevice_id[tas_priv->chip_id].name); + + if (gpio_is_valid(tas_priv->irq_info.irq_gpio)) { + rc = gpio_request(tas_priv->irq_info.irq_gpio, + "AUDEV-IRQ"); + if (!rc) { + gpio_direction_input( + tas_priv->irq_info.irq_gpio); + + tas_priv->irq_info.irq = + gpio_to_irq(tas_priv->irq_info.irq_gpio); + } else + dev_err(tas_priv->dev, "%s: GPIO %d request error\n", + __func__, tas_priv->irq_info.irq_gpio); + } else + dev_err(tas_priv->dev, + "Looking up irq-gpio property failed %d\n", + tas_priv->irq_info.irq_gpio); +} + +static int tasdevice_i2c_probe(struct i2c_client *i2c) +{ + const struct i2c_device_id *id = i2c_match_id(tasdevice_id, i2c); + const struct acpi_device_id *acpi_id; + struct tasdevice_priv *tas_priv; + int ret; + + tas_priv = tasdevice_kzalloc(i2c); + if (!tas_priv) + return -ENOMEM; + + if (ACPI_HANDLE(&i2c->dev)) { + acpi_id = acpi_match_device(i2c->dev.driver->acpi_match_table, + &i2c->dev); + if (!acpi_id) { + dev_err(&i2c->dev, "No driver data\n"); + ret = -EINVAL; + goto err; + } + tas_priv->chip_id = acpi_id->driver_data; + tas_priv->isacpi = true; + } else { + tas_priv->chip_id = id ? id->driver_data : 0; + tas_priv->isacpi = false; + } + + tasdevice_parse_dt(tas_priv); + + ret = tasdevice_init(tas_priv); + if (ret) + goto err; + + ret = devm_snd_soc_register_component(tas_priv->dev, + &soc_codec_driver_tasdevice, + tasdevice_dai_driver, ARRAY_SIZE(tasdevice_dai_driver)); + if (ret) { + dev_err(tas_priv->dev, "%s: codec register error:0x%08x\n", + __func__, ret); + goto err; + } +err: + if (ret < 0) + tasdevice_remove(tas_priv); + return ret; +} + +static void tasdevice_i2c_remove(struct i2c_client *client) +{ + struct tasdevice_priv *tas_priv = i2c_get_clientdata(client); + + tasdevice_remove(tas_priv); +} + +#ifdef CONFIG_ACPI +static const struct acpi_device_id tasdevice_acpi_match[] = { + { "TAS2781", TAS2781 }, + {}, +}; + +MODULE_DEVICE_TABLE(acpi, tasdevice_acpi_match); +#endif + +static struct i2c_driver tasdevice_i2c_driver = { + .driver = { + .name = "tas2781-codec", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(tasdevice_of_match), +#ifdef CONFIG_ACPI + .acpi_match_table = ACPI_PTR(tasdevice_acpi_match), +#endif + }, + .probe = tasdevice_i2c_probe, + .remove = tasdevice_i2c_remove, + .id_table = tasdevice_id, +}; + +module_i2c_driver(tasdevice_i2c_driver); + +MODULE_AUTHOR("Shenghao Ding <shenghao-ding(a)ti.com>"); +MODULE_AUTHOR("Kevin Lu <kevin-lu(a)ti.com>"); +MODULE_DESCRIPTION("ASoC TAS2781 Driver"); +MODULE_LICENSE("GPL"); +MODULE_IMPORT_NS(SND_SOC_TAS2781_FMWLIB); -- 2.34.1

2 2

Updated single and double quotes throughout
by GitHub pull_request - opened 21 May '23

21 May '23

alsa-project/alsa-utils pull request #215 was opened from wileyhy: This commit: - Added double quotes to common expansions; removed quotes from strings; - Changed REQUIRES to an indexed array and changed the corresponding for loop; - Added curly braces where feasibly necessary; - Removed a few unnecessary trailing semi-colons; - Put URL's in single quotes to prevent any possible expansions, except for within the sed command on line 988. - Some logic structures may depend on word splitting, on lines 641, 643, 654, and 667, so I left those as-is for this commit. - Double exclamation marks in double quotes allow history expansion which is usually disabled in non-interactive shells, so I left those as-is also. - Single quoted instances of 'yes' as a parameter value, since it's also a command name, which in certain contexts could be executed. - Removed quotes from instances of '$?' since it can only expand to an integer and cannot be unset. Some additional issues for consideration of further efforts (credit: shellcheck): - variables 'inp' on line 87 and 'KERNEL_RELEASE' on line 418 appear unused; - https://pastebin.ca is offline Request URL : https://github.com/alsa-project/alsa-utils/pull/215 Patch URL : https://github.com/alsa-project/alsa-utils/pull/215.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

Add Korean translations
by GitHub pull_request - opened 21 May '23

21 May '23

alsa-project/alsa-utils pull request #214 was opened from yi-yunseok: As a sub. Request URL : https://github.com/alsa-project/alsa-utils/pull/214 Patch URL : https://github.com/alsa-project/alsa-utils/pull/214.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[tiwai-sound:topic/midi20 13/36] sound/core/ump.c:608:15: warning: variable 'info_flags' set but not used
by kernel test robot 21 May '23

21 May '23

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git topic/midi20 head: 9e08987ec08f99983e56b33fa1762a57b6216b2a commit: 3a425eaae00098e9fe5d6a5eaaac1760f15f0fa8 [13/36] ALSA: ump: Add legacy raw MIDI support config: hexagon-allmodconfig compiler: clang version 17.0.0 (https://github.com/llvm/llvm-project b0fb98227c90adf2536c9ad644a74d5e92961111) reproduce (this is a W=1 build): wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross # https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git/commit/?id=… git remote add tiwai-sound https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git git fetch --no-tags tiwai-sound topic/midi20 git checkout 3a425eaae00098e9fe5d6a5eaaac1760f15f0fa8 # save the config file mkdir build_dir && cp config build_dir/.config COMPILER_INSTALL_PATH=$HOME/0day COMPILER=clang make.cross W=1 O=build_dir ARCH=hexagon olddefconfig COMPILER_INSTALL_PATH=$HOME/0day COMPILER=clang make.cross W=1 O=build_dir ARCH=hexagon SHELL=/bin/bash drivers/tty/serial/ sound/core/ If you fix the issue, kindly add following tag where applicable | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202305211419.wjYDzRY7-lkp@intel.com/ All warnings (new ones prefixed by >>): In file included from sound/core/ump.c:12: In file included from include/sound/rawmidi.h:11: In file included from include/linux/interrupt.h:11: In file included from include/linux/hardirq.h:11: In file included from ./arch/hexagon/include/generated/asm/hardirq.h:1: In file included from include/asm-generic/hardirq.h:17: In file included from include/linux/irq.h:20: In file included from include/linux/io.h:13: In file included from arch/hexagon/include/asm/io.h:334: include/asm-generic/io.h:547:31: warning: performing pointer arithmetic on a null pointer has undefined behavior [-Wnull-pointer-arithmetic] val = __raw_readb(PCI_IOBASE + addr); ~~~~~~~~~~ ^ include/asm-generic/io.h:560:61: warning: performing pointer arithmetic on a null pointer has undefined behavior [-Wnull-pointer-arithmetic] val = __le16_to_cpu((__le16 __force)__raw_readw(PCI_IOBASE + addr)); ~~~~~~~~~~ ^ include/uapi/linux/byteorder/little_endian.h:37:51: note: expanded from macro '__le16_to_cpu' #define __le16_to_cpu(x) ((__force __u16)(__le16)(x)) ^ In file included from sound/core/ump.c:12: In file included from include/sound/rawmidi.h:11: In file included from include/linux/interrupt.h:11: In file included from include/linux/hardirq.h:11: In file included from ./arch/hexagon/include/generated/asm/hardirq.h:1: In file included from include/asm-generic/hardirq.h:17: In file included from include/linux/irq.h:20: In file included from include/linux/io.h:13: In file included from arch/hexagon/include/asm/io.h:334: include/asm-generic/io.h:573:61: warning: performing pointer arithmetic on a null pointer has undefined behavior [-Wnull-pointer-arithmetic] val = __le32_to_cpu((__le32 __force)__raw_readl(PCI_IOBASE + addr)); ~~~~~~~~~~ ^ include/uapi/linux/byteorder/little_endian.h:35:51: note: expanded from macro '__le32_to_cpu' #define __le32_to_cpu(x) ((__force __u32)(__le32)(x)) ^ In file included from sound/core/ump.c:12: In file included from include/sound/rawmidi.h:11: In file included from include/linux/interrupt.h:11: In file included from include/linux/hardirq.h:11: In file included from ./arch/hexagon/include/generated/asm/hardirq.h:1: In file included from include/asm-generic/hardirq.h:17: In file included from include/linux/irq.h:20: In file included from include/linux/io.h:13: In file included from arch/hexagon/include/asm/io.h:334: include/asm-generic/io.h:584:33: warning: performing pointer arithmetic on a null pointer has undefined behavior [-Wnull-pointer-arithmetic] __raw_writeb(value, PCI_IOBASE + addr); ~~~~~~~~~~ ^ include/asm-generic/io.h:594:59: warning: performing pointer arithmetic on a null pointer has undefined behavior [-Wnull-pointer-arithmetic] __raw_writew((u16 __force)cpu_to_le16(value), PCI_IOBASE + addr); ~~~~~~~~~~ ^ include/asm-generic/io.h:604:59: warning: performing pointer arithmetic on a null pointer has undefined behavior [-Wnull-pointer-arithmetic] __raw_writel((u32 __force)cpu_to_le32(value), PCI_IOBASE + addr); ~~~~~~~~~~ ^ >> sound/core/ump.c:608:15: warning: variable 'info_flags' set but not used [-Wunused-but-set-variable] unsigned int info_flags = SNDRV_RAWMIDI_INFO_UMP; ^ 7 warnings generated. vim +/info_flags +608 sound/core/ump.c 604 605 int snd_ump_attach_legacy_rawmidi(struct snd_ump_endpoint *ump, 606 char *id, int device) 607 { > 608 unsigned int info_flags = SNDRV_RAWMIDI_INFO_UMP; 609 struct snd_rawmidi *rmidi; 610 bool input, output; 611 int err; 612 613 err = snd_ump_convert_init(ump); 614 if (err < 0) 615 return err; 616 617 input = ump->core.info_flags & SNDRV_RAWMIDI_INFO_INPUT; 618 output = ump->core.info_flags & SNDRV_RAWMIDI_INFO_OUTPUT; 619 if (input) 620 info_flags |= SNDRV_RAWMIDI_INFO_INPUT; 621 if (output) 622 info_flags |= SNDRV_RAWMIDI_INFO_OUTPUT; 623 if (input && output) 624 info_flags |= SNDRV_RAWMIDI_INFO_DUPLEX; 625 626 err = snd_rawmidi_new(ump->core.card, id, device, 627 output ? 16 : 0, input ? 16 : 0, 628 &rmidi); 629 if (err < 0) { 630 snd_ump_convert_free(ump); 631 return err; 632 } 633 634 if (input) 635 snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT, 636 &snd_ump_legacy_input_ops); 637 if (output) 638 snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT, 639 &snd_ump_legacy_output_ops); 640 rmidi->info_flags = ump->core.info_flags & ~SNDRV_RAWMIDI_INFO_UMP; 641 rmidi->ops = &snd_ump_legacy_ops; 642 rmidi->private_data = ump; 643 ump->legacy_rmidi = rmidi; 644 ump_dbg(ump, "Created a legacy rawmidi #%d (%s)\n", device, id); 645 return 0; 646 } 647 EXPORT_SYMBOL_GPL(snd_ump_attach_legacy_rawmidi); 648 #endif /* CONFIG_SND_UMP_LEGACY_RAWMIDI */ 649 -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

1 0

[tiwai-sound:topic/midi20 11/36] sound/usb/midi2.c:826:59: warning: variable 'type' is uninitialized when used here
by kernel test robot 21 May '23

21 May '23

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git topic/midi20 head: 9e08987ec08f99983e56b33fa1762a57b6216b2a commit: a9822209474cc230d43b7df9833b25cb3c544b6f [11/36] ALSA: usb-audio: Create UMP blocks from USB MIDI GTBs config: hexagon-allmodconfig compiler: clang version 17.0.0 (https://github.com/llvm/llvm-project b0fb98227c90adf2536c9ad644a74d5e92961111) reproduce (this is a W=1 build): wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross # https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git/commit/?id=… git remote add tiwai-sound https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git git fetch --no-tags tiwai-sound topic/midi20 git checkout a9822209474cc230d43b7df9833b25cb3c544b6f # save the config file mkdir build_dir && cp config build_dir/.config COMPILER_INSTALL_PATH=$HOME/0day COMPILER=clang make.cross W=1 O=build_dir ARCH=hexagon olddefconfig COMPILER_INSTALL_PATH=$HOME/0day COMPILER=clang make.cross W=1 O=build_dir ARCH=hexagon SHELL=/bin/bash drivers/tty/serial/ sound/core/ sound/usb/ If you fix the issue, kindly add following tag where applicable | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202305211230.xsXqNmFC-lkp@intel.com/ All warnings (new ones prefixed by >>): In file included from sound/usb/midi2.c:10: In file included from include/linux/usb.h:16: In file included from include/linux/interrupt.h:11: In file included from include/linux/hardirq.h:11: In file included from ./arch/hexagon/include/generated/asm/hardirq.h:1: In file included from include/asm-generic/hardirq.h:17: In file included from include/linux/irq.h:20: In file included from include/linux/io.h:13: In file included from arch/hexagon/include/asm/io.h:334: include/asm-generic/io.h:547:31: warning: performing pointer arithmetic on a null pointer has undefined behavior [-Wnull-pointer-arithmetic] val = __raw_readb(PCI_IOBASE + addr); ~~~~~~~~~~ ^ include/asm-generic/io.h:560:61: warning: performing pointer arithmetic on a null pointer has undefined behavior [-Wnull-pointer-arithmetic] val = __le16_to_cpu((__le16 __force)__raw_readw(PCI_IOBASE + addr)); ~~~~~~~~~~ ^ include/uapi/linux/byteorder/little_endian.h:37:51: note: expanded from macro '__le16_to_cpu' #define __le16_to_cpu(x) ((__force __u16)(__le16)(x)) ^ In file included from sound/usb/midi2.c:10: In file included from include/linux/usb.h:16: In file included from include/linux/interrupt.h:11: In file included from include/linux/hardirq.h:11: In file included from ./arch/hexagon/include/generated/asm/hardirq.h:1: In file included from include/asm-generic/hardirq.h:17: In file included from include/linux/irq.h:20: In file included from include/linux/io.h:13: In file included from arch/hexagon/include/asm/io.h:334: include/asm-generic/io.h:573:61: warning: performing pointer arithmetic on a null pointer has undefined behavior [-Wnull-pointer-arithmetic] val = __le32_to_cpu((__le32 __force)__raw_readl(PCI_IOBASE + addr)); ~~~~~~~~~~ ^ include/uapi/linux/byteorder/little_endian.h:35:51: note: expanded from macro '__le32_to_cpu' #define __le32_to_cpu(x) ((__force __u32)(__le32)(x)) ^ In file included from sound/usb/midi2.c:10: In file included from include/linux/usb.h:16: In file included from include/linux/interrupt.h:11: In file included from include/linux/hardirq.h:11: In file included from ./arch/hexagon/include/generated/asm/hardirq.h:1: In file included from include/asm-generic/hardirq.h:17: In file included from include/linux/irq.h:20: In file included from include/linux/io.h:13: In file included from arch/hexagon/include/asm/io.h:334: include/asm-generic/io.h:584:33: warning: performing pointer arithmetic on a null pointer has undefined behavior [-Wnull-pointer-arithmetic] __raw_writeb(value, PCI_IOBASE + addr); ~~~~~~~~~~ ^ include/asm-generic/io.h:594:59: warning: performing pointer arithmetic on a null pointer has undefined behavior [-Wnull-pointer-arithmetic] __raw_writew((u16 __force)cpu_to_le16(value), PCI_IOBASE + addr); ~~~~~~~~~~ ^ include/asm-generic/io.h:604:59: warning: performing pointer arithmetic on a null pointer has undefined behavior [-Wnull-pointer-arithmetic] __raw_writel((u32 __force)cpu_to_le32(value), PCI_IOBASE + addr); ~~~~~~~~~~ ^ >> sound/usb/midi2.c:826:59: warning: variable 'type' is uninitialized when used here [-Wuninitialized] usb_audio_dbg(umidi->chip, "Unsupported GTB type %d\n", type); ^~~~ sound/usb/usbaudio.h:79:36: note: expanded from macro 'usb_audio_dbg' dev_dbg(&(chip)->dev->dev, fmt, ##args) ^~~~ include/linux/dev_printk.h:155:39: note: expanded from macro 'dev_dbg' dynamic_dev_dbg(dev, dev_fmt(fmt), ##__VA_ARGS__) ^~~~~~~~~~~ include/linux/dynamic_debug.h:272:19: note: expanded from macro 'dynamic_dev_dbg' dev, fmt, ##__VA_ARGS__) ^~~~~~~~~~~ include/linux/dynamic_debug.h:248:59: note: expanded from macro '_dynamic_func_call' _dynamic_func_call_cls(_DPRINTK_CLASS_DFLT, fmt, func, ##__VA_ARGS__) ^~~~~~~~~~~ include/linux/dynamic_debug.h:246:65: note: expanded from macro '_dynamic_func_call_cls' __dynamic_func_call_cls(__UNIQUE_ID(ddebug), cls, fmt, func, ##__VA_ARGS__) ^~~~~~~~~~~ include/linux/dynamic_debug.h:222:15: note: expanded from macro '__dynamic_func_call_cls' func(&id, ##__VA_ARGS__); \ ^~~~~~~~~~~ sound/usb/midi2.c:801:10: note: initialize the variable 'type' to silence this warning int type, err; ^ = 0 7 warnings generated. vim +/type +826 sound/usb/midi2.c 794 795 /* create a UMP block from a GTB entry */ 796 static int create_gtb_block(struct snd_usb_midi2_ump *rmidi, int dir, int blk) 797 { 798 struct snd_usb_midi2_interface *umidi = rmidi->umidi; 799 const struct usb_ms20_gr_trm_block_descriptor *desc; 800 struct snd_ump_block *fb; 801 int type, err; 802 803 desc = find_group_terminal_block(umidi, blk); 804 if (!desc) 805 return 0; 806 807 usb_audio_dbg(umidi->chip, 808 "GTB %d: type=%d, group=%d/%d, protocol=%d, in bw=%d, out bw=%d\n", 809 blk, desc->bGrpTrmBlkType, desc->nGroupTrm, 810 desc->nNumGroupTrm, desc->bMIDIProtocol, 811 __le16_to_cpu(desc->wMaxInputBandwidth), 812 __le16_to_cpu(desc->wMaxOutputBandwidth)); 813 814 /* assign the direction */ 815 switch (desc->bGrpTrmBlkType) { 816 case USB_MS_GR_TRM_BLOCK_TYPE_BIDIRECTIONAL: 817 type = SNDRV_UMP_DIR_BIDIRECTION; 818 break; 819 case USB_MS_GR_TRM_BLOCK_TYPE_INPUT_ONLY: 820 type = SNDRV_UMP_DIR_INPUT; 821 break; 822 case USB_MS_GR_TRM_BLOCK_TYPE_OUTPUT_ONLY: 823 type = SNDRV_UMP_DIR_OUTPUT; 824 break; 825 default: > 826 usb_audio_dbg(umidi->chip, "Unsupported GTB type %d\n", type); 827 return 0; /* unsupported */ 828 } 829 830 /* guess work: set blk-1 as the (0-based) block ID */ 831 err = snd_ump_block_new(rmidi->ump, blk - 1, type, 832 desc->nGroupTrm, desc->nNumGroupTrm, 833 &fb); 834 if (err == -EBUSY) 835 return 0; /* already present */ 836 else if (err) 837 return err; 838 839 if (desc->iBlockItem) 840 usb_string(rmidi->dev, desc->iBlockItem, 841 fb->info.name, sizeof(fb->info.name)); 842 843 if (__le16_to_cpu(desc->wMaxInputBandwidth) == 1 || 844 __le16_to_cpu(desc->wMaxOutputBandwidth) == 1) 845 fb->info.flags |= SNDRV_UMP_BLOCK_IS_MIDI1 | 846 SNDRV_UMP_BLOCK_IS_LOWSPEED; 847 848 usb_audio_dbg(umidi->chip, 849 "Created a UMP block %d from GTB, name=%s\n", 850 blk, fb->info.name); 851 return 0; 852 } 853 -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

1 0

[PATCH v2][next] ALSA: mixart: Replace one-element arrays with simple object declarations
by Gustavo A. R. Silva 20 May '23

20 May '23

One-element arrays are deprecated, and we are replacing them with flexible array members instead. However, in this case it seems those one-element arrays have never actually been used as fake flexible arrays. See this code that dates from Linux-2.6.12-rc2 initial git repository build (commit 1da177e4c3f4 ("Linux-2.6.12-rc2")): sound/pci/mixart/mixart_core.h: 215 struct mixart_stream_state_req 216 { 217 u32 delayed; 218 u64 scheduler; 219 u32 reserved4np[3]; 220 u32 stream_count; /* set to 1 for instance */ 221 struct mixart_flow_info stream_info; /* could be an array[stream_count] */ 222 } __attribute__((packed)); sound/pci/mixart/mixart.c: 388 389 memset(&stream_state_req, 0, sizeof(stream_state_req)); 390 stream_state_req.stream_count = 1; 391 stream_state_req.stream_info.stream_desc.uid_pipe = stream->pipe->group_uid; 392 stream_state_req.stream_info.stream_desc.stream_idx = stream->substream->number; 393 So, taking the code above as example, replace multiple one-element arrays with simple object declarations, and refactor the rest of the code, accordingly. This helps with the ongoing efforts to tighten the FORTIFY_SOURCE routines on memcpy() and help us make progress towards globally enabling -fstrict-flex-arrays=3 [1]. This results in no differences in binary output. Link: https://github.com/KSPP/linux/issues/79 Link: https://github.com/KSPP/linux/issues/296 Link: https://gcc.gnu.org/pipermail/gcc-patches/2022-October/602902.html [1] Signed-off-by: Gustavo A. R. Silva <gustavoars(a)kernel.org> --- Changes in v2: - Keep the code comments (by Takashi Iwai) and explicitly mention that the objects could be arrays, in theory. v1: - Link: https://lore.kernel.org/linux-hardening/ZGVlcpuvx1rSOMP8@work/ sound/pci/mixart/mixart.c | 8 ++++---- sound/pci/mixart/mixart_core.h | 7 +++---- 2 files changed, 7 insertions(+), 8 deletions(-) diff --git a/sound/pci/mixart/mixart.c b/sound/pci/mixart/mixart.c index 1b078b789604..7ceaf6a7a77e 100644 --- a/sound/pci/mixart/mixart.c +++ b/sound/pci/mixart/mixart.c @@ -98,7 +98,7 @@ static int mixart_set_pipe_state(struct mixart_mgr *mgr, memset(&group_state, 0, sizeof(group_state)); group_state.pipe_count = 1; - group_state.pipe_uid[0] = pipe->group_uid; + group_state.pipe_uid = pipe->group_uid; if(start) request.message_id = MSG_STREAM_START_STREAM_GRP_PACKET; @@ -185,7 +185,7 @@ static int mixart_set_clock(struct mixart_mgr *mgr, clock_properties.clock_mode = CM_STANDALONE; clock_properties.frequency = rate; clock_properties.nb_callers = 1; /* only one entry in uid_caller ! */ - clock_properties.uid_caller[0] = pipe->group_uid; + clock_properties.uid_caller = pipe->group_uid; dev_dbg(&mgr->pci->dev, "mixart_set_clock to %d kHz\n", rate); @@ -565,8 +565,8 @@ static int mixart_set_format(struct mixart_stream *stream, snd_pcm_format_t form stream_param.pipe_count = 1; /* set to 1 */ stream_param.stream_count = 1; /* set to 1 */ - stream_param.stream_desc[0].uid_pipe = stream->pipe->group_uid; - stream_param.stream_desc[0].stream_idx = stream->substream->number; + stream_param.stream_desc.uid_pipe = stream->pipe->group_uid; + stream_param.stream_desc.stream_idx = stream->substream->number; request.message_id = MSG_STREAM_SET_INPUT_STAGE_PARAM; request.uid = (struct mixart_uid){0,0}; diff --git a/sound/pci/mixart/mixart_core.h b/sound/pci/mixart/mixart_core.h index 2f0e29ed5d63..d39233e0e070 100644 --- a/sound/pci/mixart/mixart_core.h +++ b/sound/pci/mixart/mixart_core.h @@ -231,7 +231,7 @@ struct mixart_group_state_req u64 scheduler; u32 reserved4np[2]; u32 pipe_count; /* set to 1 for instance */ - struct mixart_uid pipe_uid[1]; /* could be an array[pipe_count] */ + struct mixart_uid pipe_uid; /* could be an array[pipe_count], in theory */ } __attribute__((packed)); struct mixart_group_state_resp @@ -314,7 +314,7 @@ struct mixart_clock_properties u32 format; u32 board_mask; u32 nb_callers; /* set to 1 (see below) */ - struct mixart_uid uid_caller[1]; + struct mixart_uid uid_caller; } __attribute__((packed)); struct mixart_clock_properties_resp @@ -401,8 +401,7 @@ struct mixart_stream_param_desc u32 reserved4np[3]; u32 pipe_count; /* set to 1 (array size !) */ u32 stream_count; /* set to 1 (array size !) */ - struct mixart_txx_stream_desc stream_desc[1]; /* only one stream per command, but this could be an array */ - + struct mixart_txx_stream_desc stream_desc; /* only one stream per command, but this could be an array, in theory */ } __attribute__((packed)); -- 2.34.1

2 1

[PATCH v2] ALSA: emu10k1: centralize freeing PCM voices
by Oswald Buddenhagen 20 May '23

20 May '23

This saves some code duplication; no functional change. Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen(a)gmx.de> --- v2: - add minimal description (the two subsequent patches are unmodified, so not re-posting.) --- sound/pci/emu10k1/emupcm.c | 28 +++++++++++++--------------- 1 file changed, 13 insertions(+), 15 deletions(-) diff --git a/sound/pci/emu10k1/emupcm.c b/sound/pci/emu10k1/emupcm.c index 324db1141479..9d6eb58e773f 100644 --- a/sound/pci/emu10k1/emupcm.c +++ b/sound/pci/emu10k1/emupcm.c @@ -76,16 +76,22 @@ static void snd_emu10k1_pcm_efx_interrupt(struct snd_emu10k1 *emu, snd_pcm_period_elapsed(emu->pcm_capture_efx_substream); } -static int snd_emu10k1_pcm_channel_alloc(struct snd_emu10k1_pcm * epcm, int voices) +static void snd_emu10k1_pcm_free_voices(struct snd_emu10k1_pcm *epcm) { - int err, i; - - for (i = 0; i < ARRAY_SIZE(epcm->voices); i++) { + for (unsigned i = 0; i < ARRAY_SIZE(epcm->voices); i++) { if (epcm->voices[i]) { snd_emu10k1_voice_free(epcm->emu, epcm->voices[i]); epcm->voices[i] = NULL; } } +} + +static int snd_emu10k1_pcm_channel_alloc(struct snd_emu10k1_pcm * epcm, int voices) +{ + int err, i; + + snd_emu10k1_pcm_free_voices(epcm); + err = snd_emu10k1_voice_alloc(epcm->emu, epcm->type == PLAYBACK_EMUVOICE ? EMU10K1_PCM : EMU10K1_EFX, voices, @@ -115,15 +121,13 @@ static int snd_emu10k1_pcm_channel_alloc(struct snd_emu10k1_pcm * epcm, int voic "failed extra: voices=%d, frame=%d\n", voices, frame); */ - for (i = 0; i < voices; i++) { - snd_emu10k1_voice_free(epcm->emu, epcm->voices[i]); - epcm->voices[i] = NULL; - } + snd_emu10k1_pcm_free_voices(epcm); return err; } epcm->extra->epcm = epcm; epcm->extra->interrupt = snd_emu10k1_pcm_interrupt; } + return 0; } @@ -359,21 +363,15 @@ static int snd_emu10k1_playback_hw_free(struct snd_pcm_substream *substream) struct snd_emu10k1 *emu = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; struct snd_emu10k1_pcm *epcm; - int i; if (runtime->private_data == NULL) return 0; epcm = runtime->private_data; if (epcm->extra) { snd_emu10k1_voice_free(epcm->emu, epcm->extra); epcm->extra = NULL; } - for (i = 0; i < NUM_EFX_PLAYBACK; i++) { - if (epcm->voices[i]) { - snd_emu10k1_voice_free(epcm->emu, epcm->voices[i]); - epcm->voices[i] = NULL; - } - } + snd_emu10k1_pcm_free_voices(epcm); if (epcm->memblk) { snd_emu10k1_free_pages(emu, epcm->memblk); epcm->memblk = NULL; -- 2.40.0.152.g15d061e6df

2 1

snd_hda_codec_cirrus kernel oops
by Olliver Schinagl 20 May '23

20 May '23

Hey list, I've noticed the following crash since the last few days, causing audio to no longer to work. It could be related to me updating the firmware (bios) not too long ago (I load Mac OS every few months via an external drive) which caused the firmware to be updated. Or, it was a kernel update, that came along every few weeks. I'll try to find my archlinux bugzilla stuff, to report it there as well, but I don't think they are actively patching the sound stuff. The last commit I'm seeing on `patch_cirrus.c` seems to be from aug 2022, so that can't be it, though incidentally, the change is with regards to support for iMac 12,1 model, which is interesting as the number is the same. I understand you guys probably get tons of bug reports, but best to leave it here, just in case. [ 90.497004] CPU: 3 PID: 343 Comm: modprobe Not tainted 6.3.1-arch2-1 #1 4c16b0b90f71a940c7f1bb2eb00cdd9db2a83452 [ 90.497008] Hardware name: Apple Inc. MacBookPro12,1/Mac-E43C1C25D4880AD6, BIOS 481.0.0.0.0 01/12/2023 [ 90.497010] RIP: 0010:get_line_out_pfx+0x2dd/0x3e0 [snd_hda_codec_generic] [ 90.497019] Code: f9 01 0f 84 20 fe ff ff 48 c7 c2 4d 45 f9 c0 41 83 f9 02 0f 85 fe fd ff ff e9 0a fe ff ff 48 c7 c7 6e 45 f9 c0 e8 13 26 32 e6 <0f> 0b 48 c7 c2 dd 45 f9 c0 e9 f0 fd ff ff 8b 91 10 0b 00 00 48 8b [ 90.497021] RSP: 0018:ffffa50740cebad8 EFLAGS: 00010282 [ 90.497023] RAX: 0000000000000000 RBX: ffff90ea048d1cf0 RCX: 0000000000000027 [ 90.497025] RDX: ffff90ed5eda1688 RSI: 0000000000000001 RDI: ffff90ed5eda1680 [ 90.497027] RBP: 0000000000000004 R08: 0000000000000000 R09: ffffa50740ceb968 [ 90.497028] R10: 0000000000000003 R11: ffffffffa90ca208 R12: ffffffffc0f948c0 [ 90.497030] R13: ffff90ea13961000 R14: ffff90ea13940000 R15: 0000000010000013 [ 90.497032] FS: 00007f1db7091740(0000) GS:ffff90ed5ed80000(0000) knlGS:0000000000000000 [ 90.497034] CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 [ 90.497036] CR2: 000056301fb6f000 CR3: 00000001018ec002 CR4: 00000000003706e0 [ 90.497038] Call Trace: [ 90.497040] <TASK> [ 90.497043] snd_hda_gen_parse_auto_config+0xa54/0x2d50 [snd_hda_codec_generic 81639eb673de159e5e532c3e233834b733e4a85d] [ 90.497066] ? snd_hda_parse_pin_defcfg+0xaf7/0xda0 [snd_hda_codec 5b5014cb067d97e046acd84ac2db87d5fa068fb6] [ 90.497086] cs_parse_auto_config+0x3b/0xa0 [snd_hda_codec_cirrus 9ac52229b60afb26837192841054ac3dbe8aecc3] [ 90.497094] patch_cs4208+0x155/0x1a0 [snd_hda_codec_cirrus 9ac52229b60afb26837192841054ac3dbe8aecc3] [ 90.497101] ? snd_hdac_regmap_init+0x24/0x50 [snd_hda_core b180f4281d829fe0e1d0ef6f041e0fc5adf64817] [ 90.497128] hda_codec_driver_probe+0x98/0x150 [snd_hda_codec 5b5014cb067d97e046acd84ac2db87d5fa068fb6] [ 90.497143] really_probe+0x19b/0x3e0 [ 90.497149] ? __pfx___driver_attach+0x10/0x10 [ 90.497152] __driver_probe_device+0x78/0x160 [ 90.497156] driver_probe_device+0x1f/0x90 [ 90.497159] __driver_attach+0xd2/0x1c0 [ 90.497163] bus_for_each_dev+0x85/0xd0 [ 90.497166] bus_add_driver+0x116/0x220 [ 90.497170] driver_register+0x59/0x100 [ 90.497173] ? __pfx_init_module+0x10/0x10 [snd_hda_codec_cirrus 9ac52229b60afb26837192841054ac3dbe8aecc3] [ 90.497179] do_one_initcall+0x5a/0x240 [ 90.497184] do_init_module+0x4a/0x200 [ 90.497190] __do_sys_init_module+0x17f/0x1b0 [ 90.497195] do_syscall_64+0x5d/0x90 [ 90.497213] ? do_user_addr_fault+0x1e0/0x720 [ 90.497217] ? exc_page_fault+0x7c/0x180 [ 90.497222] entry_SYSCALL_64_after_hwframe+0x72/0xdc [ 90.497225] RIP: 0033:0x7f1db6b21f9e [ 90.497242] Code: 48 8b 0d bd ed 0c 00 f7 d8 64 89 01 48 83 c8 ff c3 66 2e 0f 1f 84 00 00 00 00 00 90 f3 0f 1e fa 49 89 ca b8 af 00 00 00 0f 05 <48> 3d 01 f0 ff ff 73 01 c3 48 8b 0d 8a ed 0c 00 f7 d8 64 89 01 48 [ 90.497244] RSP: 002b:00007fff2ac9ec58 EFLAGS: 00000246 ORIG_RAX: 00000000000000af [ 90.497246] RAX: ffffffffffffffda RBX: 00005614a9a07e00 RCX: 00007f1db6b21f9e [ 90.497248] RDX: 00005614a8e1dcb2 RSI: 0000000000010660 RDI: 00005614a9a6cdb0 [ 90.497250] RBP: 00005614a8e1dcb2 R08: 0000000000010660 R09: 0000000000000000 [ 90.497251] R10: 000000000005d8a1 R11: 0000000000000246 R12: 0000000000040000 [ 90.497253] R13: 00005614a9a07c50 R14: 0000000000000000 R15: 00005614a9a0bbb0 [ 90.497270] </TASK> [ 90.497271] ---[ end trace 0000000000000000 ]--- [ 90.497273] ------------[ cut here ]------------ [ 90.497274] BUG? [ 90.497284] WARNING: CPU: 3 PID: 343 at sound/pci/hda/hda_generic.c:1238 get_line_out_pfx+0x2dd/0x3e0 [snd_hda_codec_generic] [ 90.497293] Modules linked in: snd_hda_codec_cirrus(+) brcmfmac(+) snd_hda_codec_generic ledtrig_audio brcmutil iTCO_wdt snd_hda_intel mei_pxp mei_hdcp snd_intel_dspcfg btusb irqbypass snd_intel_sdw_acpi btrtl intel_pmc_bxt snd_hda_codec btbcm drm_buddy spi_intel_platform btintel snd_hda_core spi_intel iTCO_vendor_support rapl btmtk cfg80211 applesmc i2c_i801 mei_me joydev i2c_algo_bit intel_cstate snd_hwdep ttm i2c_smbus pcspkr mei intel_uncore intel_pch_thermal bluetooth lpc_ich snd_pcm thunderbolt mmc_core drm_display_helper snd_timer snd acpi_als ecdh_generic cec mousedev bcm5974 rfkill intel_gtt industrialio_triggered_buffer soundcore sbs kfifo_buf video apple_mfi_fastcharge sbshc industrialio wmi apple_bl mac_hid dm_multipath sg loop crypto_user fuse bpf_preload ip_tables x_tables usbhid crct10dif_pclmul crc32_pclmul polyval_clmulni polyval_generic gf128mul ghash_clmulni_intel sha512_ssse3 aesni_intel crypto_simd cryptd applespi crc16 xhci_pci spi_pxa2xx_pci spi_pxa2xx_platform xhci_pci_renesas dw_dmac hid_apple [ 90.497374] tg3 libphy vfat fat btrfs blake2b_generic xor raid6_pq libcrc32c crc32c_generic crc32c_intel dm_crypt cbc encrypted_keys trusted asn1_encoder tee dm_mod [ 90.497388] CPU: 3 PID: 343 Comm: modprobe Tainted: G W 6.3.1-arch2-1 #1 4c16b0b90f71a940c7f1bb2eb00cdd9db2a83452 [ 90.497391] Hardware name: Apple Inc. MacBookPro12,1/Mac-E43C1C25D4880AD6, BIOS 481.0.0.0.0 01/12/2023 [ 90.497392] RIP: 0010:get_line_out_pfx+0x2dd/0x3e0 [snd_hda_codec_generic] [ 90.497401] Code: f9 01 0f 84 20 fe ff ff 48 c7 c2 4d 45 f9 c0 41 83 f9 02 0f 85 fe fd ff ff e9 0a fe ff ff 48 c7 c7 6e 45 f9 c0 e8 13 26 32 e6 <0f> 0b 48 c7 c2 dd 45 f9 c0 e9 f0 fd ff ff 8b 91 10 0b 00 00 48 8b [ 90.497403] RSP: 0018:ffffa50740cebad8 EFLAGS: 00010282 [ 90.497418] RAX: 0000000000000000 RBX: ffff90ea048d1cf0 RCX: 0000000000000027 [ 90.497420] RDX: ffff90ed5eda1688 RSI: 0000000000000001 RDI: ffff90ed5eda1680 [ 90.497422] RBP: 0000000000000004 R08: 0000000000000000 R09: ffffa50740ceb968 [ 90.497423] R10: 0000000000000003 R11: ffffffffa90ca208 R12: 0000000000000004 [ 90.497425] R13: ffff90ea13961000 R14: ffff90ea13940000 R15: ffffffffc0f945dd [ 90.497427] FS: 00007f1db7091740(0000) GS:ffff90ed5ed80000(0000) knlGS:0000000000000000 [ 90.497429] CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 [ 90.497430] CR2: 000056301fb6f000 CR3: 00000001018ec002 CR4: 00000000003706e0 [ 90.497432] Call Trace: [ 90.497433] <TASK> [ 90.497435] snd_hda_gen_parse_auto_config+0x9a5/0x2d50 [snd_hda_codec_generic 81639eb673de159e5e532c3e233834b733e4a85d] [ 90.497444] ? snd_hda_parse_pin_defcfg+0xaf7/0xda0 [snd_hda_codec 5b5014cb067d97e046acd84ac2db87d5fa068fb6] [ 90.497463] cs_parse_auto_config+0x3b/0xa0 [snd_hda_codec_cirrus 9ac52229b60afb26837192841054ac3dbe8aecc3] [ 90.497469] patch_cs4208+0x155/0x1a0 [snd_hda_codec_cirrus 9ac52229b60afb26837192841054ac3dbe8aecc3] [ 90.497476] ? snd_hdac_regmap_init+0x24/0x50 [snd_hda_core b180f4281d829fe0e1d0ef6f041e0fc5adf64817] [ 90.497489] hda_codec_driver_probe+0x98/0x150 [snd_hda_codec 5b5014cb067d97e046acd84ac2db87d5fa068fb6] [ 90.497516] really_probe+0x19b/0x3e0 [ 90.497520] ? __pfx___driver_attach+0x10/0x10 [ 90.497523] __driver_probe_device+0x78/0x160 [ 90.497527] driver_probe_device+0x1f/0x90 [ 90.497530] __driver_attach+0xd2/0x1c0 [ 90.497533] bus_for_each_dev+0x85/0xd0 [ 90.497537] bus_add_driver+0x116/0x220 [ 90.497540] driver_register+0x59/0x100 [ 90.497543] ? __pfx_init_module+0x10/0x10 [snd_hda_codec_cirrus 9ac52229b60afb26837192841054ac3dbe8aecc3] [ 90.497549] do_one_initcall+0x5a/0x240 [ 90.497553] do_init_module+0x4a/0x200 [ 90.497570] __do_sys_init_module+0x17f/0x1b0 [ 90.497576] do_syscall_64+0x5d/0x90 [ 90.497579] ? do_user_addr_fault+0x1e0/0x720 [ 90.497583] ? exc_page_fault+0x7c/0x180 [ 90.497587] entry_SYSCALL_64_after_hwframe+0x72/0xdc [ 90.497589] RIP: 0033:0x7f1db6b21f9e [ 90.497595] Code: 48 8b 0d bd ed 0c 00 f7 d8 64 89 01 48 83 c8 ff c3 66 2e 0f 1f 84 00 00 00 00 00 90 f3 0f 1e fa 49 89 ca b8 af 00 00 00 0f 05 <48> 3d 01 f0 ff ff 73 01 c3 48 8b 0d 8a ed 0c 00 f7 d8 64 89 01 48 [ 90.497596] RSP: 002b:00007fff2ac9ec58 EFLAGS: 00000246 ORIG_RAX: 00000000000000af [ 90.497599] RAX: ffffffffffffffda RBX: 00005614a9a07e00 RCX: 00007f1db6b21f9e [ 90.497601] RDX: 00005614a8e1dcb2 RSI: 0000000000010660 RDI: 00005614a9a6cdb0 [ 90.497602] RBP: 00005614a8e1dcb2 R08: 0000000000010660 R09: 0000000000000000 [ 90.497604] R10: 000000000005d8a1 R11: 0000000000000246 R12: 0000000000040000 [ 90.497605] R13: 00005614a9a07c50 R14: 0000000000000000 R15: 00005614a9a0bbb0 [ 90.497609] </TASK> [ 90.497610] ---[ end trace 0000000000000000 ]--- [ 90.512670] bcm5974: bad trackpad package, length: 8 [ 90.569769] input: HDA Intel PCH Mic as /devices/pci0000:00/0000:00:1b.0/sound/card1/input9 [ 90.569831] input: HDA Intel PCH Line as /devices/pci0000:00/0000:00:1b.0/sound/card1/input10 [ 90.569886] input: HDA Intel PCH Headphone as /devices/pci0000:00/0000:00:1b.0/sound/card1/input11 [ 90.569926] input: HDA Intel PCH SPDIF as /devices/pci0000:00/0000:00:1b.0/sound/card1/input12 [ 90.569977] input: HDA Intel PCH SPDIF In as /devices/pci0000:00/0000:00:1b.0/sound/card1/input13

2 10

[PATCH alsa-utils 0/3] Add MIDI 2.0 support
by Takashi Iwai 20 May '23

20 May '23

This is a small patch set for alsa-utils to add the MIDI 2.0 support. Very minimalistic changes, covering only the sequencer programs: aseqdump, aconnect and aplaymidi. Takashi --- Takashi Iwai (3): aseqdump: Add UMP support aconnect: Add UMP support aplaymidi: Add UMP support configure.ac | 4 + seq/aconnect/aconnect.1 | 3 + seq/aconnect/aconnect.c | 91 +++++++++++++--- seq/aplaymidi/aplaymidi.1 | 6 ++ seq/aplaymidi/aplaymidi.c | 216 ++++++++++++++++++++++++++++++-------- seq/aseqdump/aseqdump.1 | 9 ++ seq/aseqdump/aseqdump.c | 211 ++++++++++++++++++++++++++++++++++++- 7 files changed, 478 insertions(+), 62 deletions(-) -- 2.35.3

1 3

[PATCH alsa-lib 0/8] Add MIDI 2.0 support
by Takashi Iwai 20 May '23

20 May '23

This is a series of patches for alsa-lib to add the support of MIDI 2.0 features, corresponding to the kernel patches that have been submitted recently: https://lore.kernel.org/r/20230519093114.28813-1-tiwai@suse.de The API extensions are minimalistic, and only a few basic accessors have been defined. Takashi --- Takashi Iwai (8): uapi: Update rawmidi API to 2.0.3 rawmidi: Add UMP ioctl support ump: Add initial support control: Add UMP device query support ump: Add helpers to parse / set UMP packet data ump: Add helpers for handling SysEx data uapi: Update asequencer.h definitions for 1.0.3 seq: Add UMP support configure.ac | 1 + include/Makefile.am | 2 +- include/control.h | 1 + include/local.h | 4 + include/rawmidi.h | 3 + include/seq.h | 44 ++ include/seq_event.h | 42 +- include/seqmid.h | 24 ++ include/sound/uapi/asequencer.h | 91 ++++- include/sound/uapi/asound.h | 58 ++- include/ump.h | 119 ++++++ include/ump_msg.h | 598 +++++++++++++++++++++++++++ src/Versions.in | 23 ++ src/control/control.c | 14 + src/control/control_hw.c | 9 + src/control/control_local.h | 1 + src/rawmidi/Makefile.am | 5 +- src/rawmidi/rawmidi.c | 19 + src/rawmidi/rawmidi_hw.c | 36 +- src/rawmidi/rawmidi_local.h | 7 + src/rawmidi/ump.c | 702 ++++++++++++++++++++++++++++++++ src/rawmidi/ump_local.h | 10 + src/seq/seq.c | 410 +++++++++++++++++-- src/seq/seq_hw.c | 72 +++- src/seq/seq_local.h | 6 +- src/seq/seqmid.c | 38 ++ 26 files changed, 2261 insertions(+), 78 deletions(-) create mode 100644 include/ump.h create mode 100644 include/ump_msg.h create mode 100644 src/rawmidi/ump.c create mode 100644 src/rawmidi/ump_local.h -- 2.35.3

1 8

[PATCH] ALSA: ti: Use pcm_for_each_format() macro for PCM format iteration
by Min-Hua Chen 20 May '23

20 May '23

Use pcm_for_each_format for the PCM format iteration and fix the following sparse warnings. sound/soc/ti/davinci-mcasp.c:1336:26: sparse: warning: restricted snd_pcm_format_t degrades to integer sound/soc/ti/davinci-mcasp.c:1358:26: sparse: warning: restricted snd_pcm_format_t degrades to integer sound/soc/ti/davinci-mcasp.c:1438:26: sparse: warning: restricted snd_pcm_format_t degrades to integer No functional changes. Signed-off-by: Min-Hua Chen <minhuadotchen(a)gmail.com> --- sound/soc/ti/davinci-mcasp.c | 27 +++++++++++++++------------ 1 file changed, 15 insertions(+), 12 deletions(-) diff --git a/sound/soc/ti/davinci-mcasp.c b/sound/soc/ti/davinci-mcasp.c index c0892be2992b..172fea764a31 100644 --- a/sound/soc/ti/davinci-mcasp.c +++ b/sound/soc/ti/davinci-mcasp.c @@ -1328,15 +1328,16 @@ static int davinci_mcasp_hw_rule_slot_width(struct snd_pcm_hw_params *params, struct davinci_mcasp_ruledata *rd = rule->private; struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); struct snd_mask nfmt; - int i, slot_width; + int slot_width; + snd_pcm_format_t i; snd_mask_none(&nfmt); slot_width = rd->mcasp->slot_width; - for (i = 0; i <= SNDRV_PCM_FORMAT_LAST; i++) { - if (snd_mask_test(fmt, i)) { + pcm_for_each_format(i) { + if (snd_mask_test_format(fmt, i)) { if (snd_pcm_format_width(i) <= slot_width) { - snd_mask_set(&nfmt, i); + snd_mask_set_format(&nfmt, i); } } } @@ -1350,15 +1351,16 @@ static int davinci_mcasp_hw_rule_format_width(struct snd_pcm_hw_params *params, struct davinci_mcasp_ruledata *rd = rule->private; struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); struct snd_mask nfmt; - int i, format_width; + int format_width; + snd_pcm_format_t i; snd_mask_none(&nfmt); format_width = rd->mcasp->max_format_width; - for (i = 0; i <= SNDRV_PCM_FORMAT_LAST; i++) { - if (snd_mask_test(fmt, i)) { + pcm_for_each_format(i) { + if (snd_mask_test_format(fmt, i)) { if (snd_pcm_format_width(i) == format_width) { - snd_mask_set(&nfmt, i); + snd_mask_set_format(&nfmt, i); } } } @@ -1431,12 +1433,13 @@ static int davinci_mcasp_hw_rule_format(struct snd_pcm_hw_params *params, struct snd_mask nfmt; int rate = params_rate(params); int slots = rd->mcasp->tdm_slots; - int i, count = 0; + int count = 0; + snd_pcm_format_t i; snd_mask_none(&nfmt); - for (i = 0; i <= SNDRV_PCM_FORMAT_LAST; i++) { - if (snd_mask_test(fmt, i)) { + pcm_for_each_format(i) { + if (snd_mask_test_format(fmt, i)) { uint sbits = snd_pcm_format_width(i); unsigned int sysclk_freq; int ppm; @@ -1454,7 +1457,7 @@ static int davinci_mcasp_hw_rule_format(struct snd_pcm_hw_params *params, sbits * slots * rate, false); if (abs(ppm) < DAVINCI_MAX_RATE_ERROR_PPM) { - snd_mask_set(&nfmt, i); + snd_mask_set_format(&nfmt, i); count++; } } -- 2.34.1

2 4

[tiwai-sound:topic/midi20 20/36] sound/core/seq/seq_ports.c:138:37: warning: variable 'num' is uninitialized when used here
by kernel test robot 19 May '23

19 May '23

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git topic/midi20 head: cc8e2c3b2d2b9c2af16151d30335492e2d0e3240 commit: e046e543faaff41051415c97bcce0b8b31d3b3f9 [20/36] ALSA: seq: Check the conflicting port at port creation config: hexagon-randconfig-r016-20230517 compiler: clang version 17.0.0 (https://github.com/llvm/llvm-project b0fb98227c90adf2536c9ad644a74d5e92961111) reproduce (this is a W=1 build): wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross # https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git/commit/?id=… git remote add tiwai-sound https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git git fetch --no-tags tiwai-sound topic/midi20 git checkout e046e543faaff41051415c97bcce0b8b31d3b3f9 # save the config file mkdir build_dir && cp config build_dir/.config COMPILER_INSTALL_PATH=$HOME/0day COMPILER=clang make.cross W=1 O=build_dir ARCH=hexagon olddefconfig COMPILER_INSTALL_PATH=$HOME/0day COMPILER=clang make.cross W=1 O=build_dir ARCH=hexagon SHELL=/bin/bash sound/core/seq/ If you fix the issue, kindly add following tag where applicable | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202305192228.T2dDkOaw-lkp@intel.com/ All warnings (new ones prefixed by >>): >> sound/core/seq/seq_ports.c:138:37: warning: variable 'num' is uninitialized when used here [-Wuninitialized] sprintf(new_port->name, "port-%d", num); ^~~ sound/core/seq/seq_ports.c:117:9: note: initialize the variable 'num' to silence this warning int num; ^ = 0 1 warning generated. vim +/num +138 sound/core/seq/seq_ports.c ^1da177e4c3f41 Linus Torvalds 2005-04-16 108 ^1da177e4c3f41 Linus Torvalds 2005-04-16 109 e046e543faaff4 Takashi Iwai 2023-01-05 110 /* create a port, port number or a negative error code is returned 71105998845fb0 Takashi Iwai 2017-10-09 111 * the caller needs to unref the port via snd_seq_port_unlock() appropriately 71105998845fb0 Takashi Iwai 2017-10-09 112 */ e046e543faaff4 Takashi Iwai 2023-01-05 113 int snd_seq_create_port(struct snd_seq_client *client, int port, e046e543faaff4 Takashi Iwai 2023-01-05 114 struct snd_seq_client_port **port_ret) ^1da177e4c3f41 Linus Torvalds 2005-04-16 115 { 9244b2c3079faa Johannes Berg 2006-10-05 116 struct snd_seq_client_port *new_port, *p; e046e543faaff4 Takashi Iwai 2023-01-05 117 int num; e046e543faaff4 Takashi Iwai 2023-01-05 118 e046e543faaff4 Takashi Iwai 2023-01-05 119 *port_ret = NULL; ^1da177e4c3f41 Linus Torvalds 2005-04-16 120 ^1da177e4c3f41 Linus Torvalds 2005-04-16 121 /* sanity check */ 7eaa943c8ed8e9 Takashi Iwai 2008-08-08 122 if (snd_BUG_ON(!client)) e046e543faaff4 Takashi Iwai 2023-01-05 123 return -EINVAL; ^1da177e4c3f41 Linus Torvalds 2005-04-16 124 de20b572a30ac3 Clemens Ladisch 2015-01-25 125 if (client->num_ports >= SNDRV_SEQ_MAX_PORTS) { 04cc79a048ee21 Takashi Iwai 2014-02-04 126 pr_warn("ALSA: seq: too many ports for client %d\n", client->number); e046e543faaff4 Takashi Iwai 2023-01-05 127 return -EINVAL; ^1da177e4c3f41 Linus Torvalds 2005-04-16 128 } ^1da177e4c3f41 Linus Torvalds 2005-04-16 129 ^1da177e4c3f41 Linus Torvalds 2005-04-16 130 /* create a new port */ ecca82b4b447f8 Takashi Iwai 2005-09-09 131 new_port = kzalloc(sizeof(*new_port), GFP_KERNEL); 24db8bbaa3fcfa Takashi Iwai 2015-03-10 132 if (!new_port) e046e543faaff4 Takashi Iwai 2023-01-05 133 return -ENOMEM; /* failure, out of memory */ ^1da177e4c3f41 Linus Torvalds 2005-04-16 134 /* init port data */ ^1da177e4c3f41 Linus Torvalds 2005-04-16 135 new_port->addr.client = client->number; ^1da177e4c3f41 Linus Torvalds 2005-04-16 136 new_port->addr.port = -1; ^1da177e4c3f41 Linus Torvalds 2005-04-16 137 new_port->owner = THIS_MODULE; ^1da177e4c3f41 Linus Torvalds 2005-04-16 @138 sprintf(new_port->name, "port-%d", num); ^1da177e4c3f41 Linus Torvalds 2005-04-16 139 snd_use_lock_init(&new_port->use_lock); ^1da177e4c3f41 Linus Torvalds 2005-04-16 140 port_subs_info_init(&new_port->c_src); ^1da177e4c3f41 Linus Torvalds 2005-04-16 141 port_subs_info_init(&new_port->c_dest); 71105998845fb0 Takashi Iwai 2017-10-09 142 snd_use_lock_use(&new_port->use_lock); ^1da177e4c3f41 Linus Torvalds 2005-04-16 143 44d30762cde76f Guo Zhengkui 2022-05-17 144 num = max(port, 0); 1a60d4c5a0c402 Ingo Molnar 2006-01-16 145 mutex_lock(&client->ports_mutex); f823b8a75527dc Takashi Iwai 2019-03-28 146 write_lock_irq(&client->ports_lock); 9244b2c3079faa Johannes Berg 2006-10-05 147 list_for_each_entry(p, &client->ports_list_head, list) { e046e543faaff4 Takashi Iwai 2023-01-05 148 if (p->addr.port == port) { e046e543faaff4 Takashi Iwai 2023-01-05 149 num = -EBUSY; e046e543faaff4 Takashi Iwai 2023-01-05 150 goto unlock; e046e543faaff4 Takashi Iwai 2023-01-05 151 } ^1da177e4c3f41 Linus Torvalds 2005-04-16 152 if (p->addr.port > num) ^1da177e4c3f41 Linus Torvalds 2005-04-16 153 break; ^1da177e4c3f41 Linus Torvalds 2005-04-16 154 if (port < 0) /* auto-probe mode */ ^1da177e4c3f41 Linus Torvalds 2005-04-16 155 num = p->addr.port + 1; ^1da177e4c3f41 Linus Torvalds 2005-04-16 156 } ^1da177e4c3f41 Linus Torvalds 2005-04-16 157 /* insert the new port */ 9244b2c3079faa Johannes Berg 2006-10-05 158 list_add_tail(&new_port->list, &p->list); ^1da177e4c3f41 Linus Torvalds 2005-04-16 159 client->num_ports++; ^1da177e4c3f41 Linus Torvalds 2005-04-16 160 new_port->addr.port = num; /* store the port number in the port */ 71105998845fb0 Takashi Iwai 2017-10-09 161 sprintf(new_port->name, "port-%d", num); e046e543faaff4 Takashi Iwai 2023-01-05 162 *port_ret = new_port; e046e543faaff4 Takashi Iwai 2023-01-05 163 unlock: f823b8a75527dc Takashi Iwai 2019-03-28 164 write_unlock_irq(&client->ports_lock); 1a60d4c5a0c402 Ingo Molnar 2006-01-16 165 mutex_unlock(&client->ports_mutex); ^1da177e4c3f41 Linus Torvalds 2005-04-16 166 e046e543faaff4 Takashi Iwai 2023-01-05 167 return num; ^1da177e4c3f41 Linus Torvalds 2005-04-16 168 } ^1da177e4c3f41 Linus Torvalds 2005-04-16 169 :::::: The code at line 138 was first introduced by commit :::::: 1da177e4c3f41524e886b7f1b8a0c1fc7321cac2 Linux-2.6.12-rc2 :::::: TO: Linus Torvalds <torvalds(a)ppc970.osdl.org> :::::: CC: Linus Torvalds <torvalds(a)ppc970.osdl.org> -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

1 0

[RESEND PATCH 00/12] ASoC: codecs: do not store status in state containe
by Krzysztof Kozlowski 19 May '23

19 May '23

Hi, Resend due to missing cover letter, so adding per-series tags was not possible. Added Rb tag. Best regards, Krzysztof Krzysztof Kozlowski (12): ASoC: codecs: rt1308: do not store status in state container ASoC: codecs: rt1316: do not store status in state container ASoC: codecs: rt1318: do not store status in state container ASoC: codecs: rt5682: do not store status in state container ASoC: codecs: rt700: do not store status in state container ASoC: codecs: rt711-sdca: do not store status in state container ASoC: codecs: rt711: do not store status in state container ASoC: codecs: rt712-sdca-dmic: do not store status in state container ASoC: codecs: rt712-sdca: do not store status in state container ASoC: codecs: rt715-sdca: do not store status in state container ASoC: codecs: rt715: do not store status in state container ASoC: codecs: rt722-sdca: do not store status in state container sound/soc/codecs/rt1308-sdw.c | 5 +---- sound/soc/codecs/rt1308-sdw.h | 1 - sound/soc/codecs/rt1316-sdw.c | 5 +---- sound/soc/codecs/rt1316-sdw.h | 1 - sound/soc/codecs/rt1318-sdw.c | 5 +---- sound/soc/codecs/rt1318-sdw.h | 1 - sound/soc/codecs/rt5682-sdw.c | 5 +---- sound/soc/codecs/rt5682.h | 1 - sound/soc/codecs/rt700-sdw.c | 5 +---- sound/soc/codecs/rt700.h | 1 - sound/soc/codecs/rt711-sdca-sdw.c | 5 +---- sound/soc/codecs/rt711-sdca.h | 1 - sound/soc/codecs/rt711-sdw.c | 5 +---- sound/soc/codecs/rt711.h | 1 - sound/soc/codecs/rt712-sdca-dmic.c | 5 +---- sound/soc/codecs/rt712-sdca-dmic.h | 1 - sound/soc/codecs/rt712-sdca-sdw.c | 5 +---- sound/soc/codecs/rt712-sdca.h | 1 - sound/soc/codecs/rt715-sdca-sdw.c | 5 +---- sound/soc/codecs/rt715-sdca.h | 1 - sound/soc/codecs/rt715-sdw.c | 4 +--- sound/soc/codecs/rt715.h | 1 - sound/soc/codecs/rt722-sdca-sdw.c | 5 +---- sound/soc/codecs/rt722-sdca.h | 1 - 24 files changed, 12 insertions(+), 59 deletions(-) -- 2.34.1

2 16

[PATCH 0/7] ALSA: emu10k1: refactoring of the playback voice management
by Oswald Buddenhagen 19 May '23

19 May '23

Oswald Buddenhagen (7): ALSA: emu10k1: simplify freeing synth voices ALSA: emu10k1: don't forget to reset reclaimed synth voices ALSA: emu10k1: improve voice status display in /proc ALSA: emu10k1: make freeing untouched playback voices cheap ALSA: emu10k1: centralize freeing PCM voices ALSA: emu10k1: make snd_emu10k1_voice_alloc() assign voices' epcm ALSA: emu10k1: revamp playback voice allocator include/sound/emu10k1.h | 18 ++-- sound/pci/emu10k1/emu10k1_callback.c | 8 +- sound/pci/emu10k1/emumixer.c | 24 ++--- sound/pci/emu10k1/emupcm.c | 88 ++++++++--------- sound/pci/emu10k1/emuproc.c | 18 ++-- sound/pci/emu10k1/voice.c | 136 +++++++++++++-------------- 6 files changed, 136 insertions(+), 156 deletions(-) -- 2.40.0.152.g15d061e6df

2 13

[PATCH] ASoC: dt-bindings: dialog,da7219: convert to dtschema
by David Rau 19 May '23

19 May '23

- Convert Dialog DA7219 bindings to DT schema format. - Add Jack insertion detection polarity selection. - Remove unused `dlg,ldo-lvl` parameter. Signed-off-by: David Rau <David.Rau.opensource(a)dm.renesas.com> --- .../devicetree/bindings/sound/da7219.txt | 112 -------- .../bindings/sound/dialog,da7219.yaml | 240 ++++++++++++++++++ 2 files changed, 240 insertions(+), 112 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/da7219.txt create mode 100644 Documentation/devicetree/bindings/sound/dialog,da7219.yaml diff --git a/Documentation/devicetree/bindings/sound/da7219.txt b/Documentation/devicetree/bindings/sound/da7219.txt deleted file mode 100644 index add1caf26ac2..000000000000 --- a/Documentation/devicetree/bindings/sound/da7219.txt +++ /dev/null @@ -1,112 +0,0 @@ -Dialog Semiconductor DA7219 Audio Codec bindings - -DA7219 is an audio codec with advanced accessory detect features. - -====== - -Required properties: -- compatible : Should be "dlg,da7219" -- reg: Specifies the I2C slave address - -- interrupts : IRQ line info for DA7219. - (See Documentation/devicetree/bindings/interrupt-controller/interrupts.txt for - further information relating to interrupt properties) - -- VDD-supply: VDD power supply for the device -- VDDMIC-supply: VDDMIC power supply for the device -- VDDIO-supply: VDDIO power supply for the device - (See Documentation/devicetree/bindings/regulator/regulator.txt for further - information relating to regulators) - -Optional properties: -- interrupt-names : Name associated with interrupt line. Should be "wakeup" if - interrupt is to be used to wake system, otherwise "irq" should be used. -- wakeup-source: Flag to indicate this device can wake system (suspend/resume). - -- #clock-cells : Should be set to '<1>', two clock sources provided; -- clock-output-names : Names given for DAI clock outputs (WCLK & BCLK); - -- clocks : phandle and clock specifier for codec MCLK. -- clock-names : Clock name string for 'clocks' attribute, should be "mclk". - -- dlg,micbias-lvl : Voltage (mV) for Mic Bias - [<1600>, <1800>, <2000>, <2200>, <2400>, <2600>] -- dlg,mic-amp-in-sel : Mic input source type - ["diff", "se_p", "se_n"] - -Deprecated properties: -- dlg,ldo-lvl : Required internal LDO voltage (mV) level for digital engine - (LDO unavailable in production HW so property no longer required). - -====== - -Child node - 'da7219_aad': - -Optional properties: -- dlg,micbias-pulse-lvl : Mic bias higher voltage pulse level (mV). - [<2800>, <2900>] -- dlg,micbias-pulse-time : Mic bias higher voltage pulse duration (ms) -- dlg,btn-cfg : Periodic button press measurements for 4-pole jack (ms) - [<2>, <5>, <10>, <50>, <100>, <200>, <500>] -- dlg,mic-det-thr : Impedance threshold for mic detection measurement (Ohms) - [<200>, <500>, <750>, <1000>] -- dlg,jack-ins-deb : Debounce time for jack insertion (ms) - [<5>, <10>, <20>, <50>, <100>, <200>, <500>, <1000>] -- dlg,jack-det-rate: Jack type detection latency (3/4 pole) - ["32ms_64ms", "64ms_128ms", "128ms_256ms", "256ms_512ms"] -- dlg,jack-rem-deb : Debounce time for jack removal (ms) - [<1>, <5>, <10>, <20>] -- dlg,a-d-btn-thr : Impedance threshold between buttons A and D - [0x0 - 0xFF] -- dlg,d-b-btn-thr : Impedance threshold between buttons D and B - [0x0 - 0xFF] -- dlg,b-c-btn-thr : Impedance threshold between buttons B and C - [0x0 - 0xFF] -- dlg,c-mic-btn-thr : Impedance threshold between button C and Mic - [0x0 - 0xFF] -- dlg,btn-avg : Number of 8-bit readings for averaged button measurement - [<1>, <2>, <4>, <8>] -- dlg,adc-1bit-rpt : Repeat count for 1-bit button measurement - [<1>, <2>, <4>, <8>] - -====== - -Example: - - codec: da7219@1a { - compatible = "dlg,da7219"; - reg = <0x1a>; - - interrupt-parent = <&gpio6>; - interrupts = <11 IRQ_TYPE_LEVEL_LOW>; - - VDD-supply = <&reg_audio>; - VDDMIC-supply = <&reg_audio>; - VDDIO-supply = <&reg_audio>; - - #clock-cells = <1>; - clock-output-names = "dai-wclk", "dai-bclk"; - - clocks = <&clks 201>; - clock-names = "mclk"; - - dlg,ldo-lvl = <1200>; - dlg,micbias-lvl = <2600>; - dlg,mic-amp-in-sel = "diff"; - - da7219_aad { - dlg,btn-cfg = <50>; - dlg,mic-det-thr = <500>; - dlg,jack-ins-deb = <20>; - dlg,jack-det-rate = "32ms_64ms"; - dlg,jack-rem-deb = <1>; - - dlg,a-d-btn-thr = <0xa>; - dlg,d-b-btn-thr = <0x16>; - dlg,b-c-btn-thr = <0x21>; - dlg,c-mic-btn-thr = <0x3E>; - - dlg,btn-avg = <4>; - dlg,adc-1bit-rpt = <1>; - }; - }; diff --git a/Documentation/devicetree/bindings/sound/dialog,da7219.yaml b/Documentation/devicetree/bindings/sound/dialog,da7219.yaml new file mode 100644 index 000000000000..32dfc4559052 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/dialog,da7219.yaml @@ -0,0 +1,240 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/dialog,da7219.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Dialog Semiconductor DA7219 Audio Codec + +maintainers: + - David Rau <David.Rau.opensource(a)dm.renesas.com> + +description: + The DA7219 is an ultra low-power audio codec with + in-built advanced accessory detection (AAD) for mobile + computing and accessory applications, which supports + sample rates up to 96 kHz at 24-bit resolution. + +properties: + compatible: + enum: + - dlg,da7219 + + reg: + description: + The I2C slave address of the DA7219. + maxItems: 1 + + interrupts: + description: + Interrupt for DA7219 IRQ line. + maxItems: 1 + + VDD-supply: + description: + VDD power supply for the device. + + VDDMIC-supply: + description: + VDDMIC power supply for the device. + + VDDIO-supply: + description: + VDDIO power supply for the device. + + interrupt-names: + description: + Name associated with interrupt line. + Should be "wakeup" if interrupt is to be used to wake system, + otherwise "irq" should be used. + + wakeup-source: + description: + Flag to indicate this device can wake system (suspend/resume). + + "#clock-cells": + description: + Two clock sources provided. + const: 1 + + clock-output-names: + description: + Names given for DAI clock outputs (WCLK & BCLK). + + clocks: + description: + phandle and clock specifier for codec MCLK. + + clock-names: + description: + Clock name string for "clocks" attribute. + const: mclk + + dlg,micbias-lvl: + enum: [1600, 1800, 2000, 2200, 2400, 2600] + description: + Voltage (mV) for Mic Bias. + $ref: /schemas/types.yaml#/definitions/uint32 + + dlg,mic-amp-in-sel: + enum: ["diff", "se_p", "se_n"] + description: + Mic input source type. + + diff - Differential. + + se_p - MIC_P. + Positive differential analog microphone input. + + se_n - MIC_N. + Negative differential analog microphone input. + $ref: /schemas/types.yaml#/definitions/string + + "da7219_aad": + type: object + description: + Configuration of advanced accessory detection. + properties: + dlg,micbias-pulse-lvl: + enum: [2800, 2900] + description: + Mic bias higher voltage pulse level (mV). + $ref: /schemas/types.yaml#/definitions/uint32 + + dlg,micbias-pulse-time: + description: + Mic bias higher voltage pulse duration (ms). + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 0 + + dlg,btn-cfg: + enum: [2, 5, 10, 50, 100, 200, 500] + description: + Periodic button press measurements for 4-pole jack (ms). + $ref: /schemas/types.yaml#/definitions/uint32 + + dlg,mic-det-thr: + enum: [200, 500, 750, 1000] + description: + Impedance threshold for mic detection measurement (Ohms). + $ref: /schemas/types.yaml#/definitions/uint32 + + dlg,jack-ins-deb: + enum: [5, 10, 20, 50, 100, 200, 500, 1000] + description: + Debounce time for jack insertion (ms). + $ref: /schemas/types.yaml#/definitions/uint32 + + dlg,jack-ins-det-pty: + enum: ["low", "high"] + description: + Polarity for jack insertion detection. + $ref: /schemas/types.yaml#/definitions/string + + dlg,jack-det-rate: + enum: ["32ms_64ms", "64ms_128ms", "128ms_256ms", "256ms_512ms"] + description: + Jack type detection latency (3/4 pole). + $ref: /schemas/types.yaml#/definitions/string + + dlg,jack-rem-deb: + enum: [1, 5, 10, 20] + description: + Debounce time for jack removal (ms). + $ref: /schemas/types.yaml#/definitions/uint32 + + dlg,a-d-btn-thr: + description: + Impedance threshold between buttons A and D. + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 0 + maximum: 255 + + dlg,d-b-btn-thr: + description: + Impedance threshold between buttons D and B. + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 0 + maximum: 255 + + dlg,b-c-btn-thr: + description: + Impedance threshold between buttons B and C. + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 0 + maximum: 255 + + dlg,c-mic-btn-thr: + description: + Impedance threshold between button C and Mic. + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 0 + maximum: 255 + + dlg,btn-avg: + enum: [1, 2, 4, 8] + description: + Number of 8-bit readings for averaged button measurement. + $ref: /schemas/types.yaml#/definitions/uint32 + + dlg,adc-1bit-rpt: + enum: [1, 2, 4, 8] + description: + Repeat count for 1-bit button measurement. + $ref: /schemas/types.yaml#/definitions/uint32 + +required: + - compatible + - reg + - interrupts + - VDD-supply + - VDDMIC-supply + - VDDIO-supply + +additionalProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/irq.h> + i2c { + #address-cells = <1>; + #size-cells = <0>; + + codec: da7219@1a { + compatible = "dlg,da7219"; + reg = <0x1a>; + + interrupt-parent = <&gpio6>; + interrupts = <11 IRQ_TYPE_LEVEL_LOW>; + + VDD-supply = <&vdd_reg>; + VDDMIC-supply = <&vddmic_reg>; + VDDIO-supply = <&vddio_reg>; + + #clock-cells = <1>; + clock-output-names = "dai-wclk", "dai-bclk"; + + clocks = <&clks 201>; + clock-names = "mclk"; + + dlg,micbias-lvl = <2600>; + dlg,mic-amp-in-sel = "diff"; + + da7219_aad { + dlg,btn-cfg = <50>; + dlg,mic-det-thr = <500>; + dlg,jack-ins-deb = <20>; + dlg,jack-ins-det-pty = "low"; + dlg,jack-det-rate = "32ms_64ms"; + dlg,jack-rem-deb = <1>; + + dlg,a-d-btn-thr = <0xa>; + dlg,d-b-btn-thr = <0x16>; + dlg,b-c-btn-thr = <0x21>; + dlg,c-mic-btn-thr = <0x3E>; + + dlg,btn-avg = <4>; + dlg,adc-1bit-rpt = <1>; + }; + }; + }; -- 2.17.1

3 4

[PATCH 00/10] Add cs42l43 PC focused SoundWire CODEC
by Charles Keepax 19 May '23

19 May '23

This patch chain adds support for the Cirrus Logic cs42l43 PC focused SoundWire CODEC. Some supporting work is included in the chain, including adding an ASoC control notification helper function and adding support for IRQs generated by the in-band SoundWire alert mechanism. The chain is currently based of v6.4-rc1 because I am not 100% sure which tree we want to send everything through. The CODEC support has a build dependency on both the SoundWire change and the ASoC soc-component change. Thanks, Charles Charles Keepax (8): ASoC: soc-component: Add notify control helper function ASoC: ak4118: Update to use new component control notify helper ASoC: wm_adsp: Update to use new component control notify helepr dt-bindings: mfd: cirrus,cs42l43: Add initial DT binding mfd: cs42l43: Add support for cs42l43 core driver irqchip/cs42l43: Add support for the cs42l43 IRQs pinctrl: cs42l43: Add support for the cs42l43 ASoC: cs42l43: Add support for the cs42l43 Lucas Tanure (2): soundwire: bus: Allow SoundWire peripherals to register IRQ handlers spi: cs42l43: Add SPI controller support .../bindings/mfd/cirrus,cs42l43.yaml | 212 ++ MAINTAINERS | 7 + drivers/irqchip/Kconfig | 9 + drivers/irqchip/Makefile | 1 + drivers/irqchip/irq-cs42l43.c | 170 ++ drivers/mfd/Kconfig | 23 + drivers/mfd/Makefile | 3 + drivers/mfd/cs42l43-i2c.c | 86 + drivers/mfd/cs42l43-sdw.c | 210 ++ drivers/mfd/cs42l43.c | 1044 ++++++++ drivers/mfd/cs42l43.h | 23 + drivers/pinctrl/cirrus/Kconfig | 11 + drivers/pinctrl/cirrus/Makefile | 2 + drivers/pinctrl/cirrus/pinctrl-cs42l43.c | 614 +++++ drivers/soundwire/bus.c | 31 + drivers/soundwire/bus_type.c | 12 + drivers/spi/Kconfig | 7 + drivers/spi/Makefile | 1 + drivers/spi/spi-cs42l43.c | 287 +++ include/linux/irqchip/cs42l43.h | 61 + include/linux/mfd/cs42l43-regs.h | 1172 +++++++++ include/linux/mfd/cs42l43.h | 50 + include/linux/soundwire/sdw.h | 9 + include/sound/cs42l43.h | 84 + include/sound/soc-component.h | 4 + sound/soc/codecs/Kconfig | 16 + sound/soc/codecs/Makefile | 4 + sound/soc/codecs/ak4118.c | 11 +- sound/soc/codecs/cs42l43-jack.c | 946 +++++++ sound/soc/codecs/cs42l43-sdw.c | 75 + sound/soc/codecs/cs42l43.c | 2270 +++++++++++++++++ sound/soc/codecs/cs42l43.h | 117 + sound/soc/codecs/wm_adsp.c | 20 +- sound/soc/soc-component.c | 22 + 34 files changed, 7586 insertions(+), 28 deletions(-) create mode 100644 Documentation/devicetree/bindings/mfd/cirrus,cs42l43.yaml create mode 100644 drivers/irqchip/irq-cs42l43.c create mode 100644 drivers/mfd/cs42l43-i2c.c create mode 100644 drivers/mfd/cs42l43-sdw.c create mode 100644 drivers/mfd/cs42l43.c create mode 100644 drivers/mfd/cs42l43.h create mode 100644 drivers/pinctrl/cirrus/pinctrl-cs42l43.c create mode 100644 drivers/spi/spi-cs42l43.c create mode 100644 include/linux/irqchip/cs42l43.h create mode 100644 include/linux/mfd/cs42l43-regs.h create mode 100644 include/linux/mfd/cs42l43.h create mode 100644 include/sound/cs42l43.h create mode 100644 sound/soc/codecs/cs42l43-jack.c create mode 100644 sound/soc/codecs/cs42l43-sdw.c create mode 100644 sound/soc/codecs/cs42l43.c create mode 100644 sound/soc/codecs/cs42l43.h -- 2.30.2

9 56

[PATCH v3 5/5] ASoC: dt-bindings: Add tas2781 amplifier
by Shenghao Ding 19 May '23

19 May '23

Create tas2781.yaml for tas2781 driver. Signed-off-by: Shenghao Ding <13916275206(a)139.com> --- Changes in v3: - Add allOf with ref to saound-dai-common - remove audio-slots, put all the i2c address into <reg> - Add more description on broadcast-addr item Changes to be committed: new file: Documentation/devicetree/bindings/sound/ti,tas2781.yaml --- .../devicetree/bindings/sound/ti,tas2781.yaml | 83 +++++++++++++++++++ 1 file changed, 83 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ti,tas2781.yaml diff --git a/Documentation/devicetree/bindings/sound/ti,tas2781.yaml b/Documentation/devicetree/bindings/sound/ti,tas2781.yaml new file mode 100644 index 000000000000..c4928e88f1c5 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,tas2781.yaml @@ -0,0 +1,83 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +# Copyright (C) 2022 - 2023 Texas Instruments Incorporated +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ti,tas2781.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Texas Instruments TAS2781 SmartAMP + +maintainers: + - Shenghao Ding <shenghao-ding(a)ti.com> + +description: + The TAS2781 is a mono, digital input Class-D audio amplifier + optimized for efficiently driving high peak power into small + loudspeakers. Integrated an on-chip DSP supports Texas Instruments + Smart Amp speaker protection algorithm. The integrated speaker + voltage and current sense provides for real time + monitoring of loudspeaker behavior. + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - ti,tas2781 + + reg: + description: + I2C address, in multiple tas2781s case, all the i2c address + aggreate as one Audio Device to support multiple audio slots. + maxItems: 4 + items: + minimum: 0x38 + maximum: 0x3f + + reset-gpios: + maxItems: 1 + + interrupts: + maxItems: 1 + + ti,broadcast-addr: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + Generic I2C address for all the tas2781 devices in + purpose of I2C broadcast during the multi-device + writes, useless in mono case or remove this item to + disable broadcast mode. + + '#sound-dai-cells': + const: 1 + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + i2c { + /* example with quad support, such as tablet or pad device */ + #address-cells = <1>; + #size-cells = <0>; + quad: codec@38 { + compatible = "ti,tas2781"; + reg = < 0x38 /* Audio slot 0 */ + 0x3a /* Audio slot 1 */ + 0x39 /* Audio slot 2 */ + 0x3b /* Audio slot 3 */ + >; + #sound-dai-cells = <1>; + reset-gpios = < &gpio1 10 GPIO_ACTIVE_HIGH >; + interrupt-parent = <&gpio1>; + interrupts = <15>; + /* Generic I2C addr among all the tas2781s for i2c data broadcast */ + ti,broadcast-addr = <0x40>; + }; + }; +... -- 2.34.1

2 1

[PATCH v3 2/5] ASoC: tas2781: firmware lib
by Shenghao Ding 19 May '23

19 May '23

Create tas2781 firmware lib. Signed-off-by: Shenghao Ding <13916275206(a)139.com> --- Changes in v3: - fixed issue | Reported-by: kernel test robot <lkp(a)intel.com> | Link: https://lore.kernel.org/oe-kbuild-all/202305022033.LiI7Ojm4-lkp@intel.com/ - Move back to codecs folder, this firmware lib is special for the audio cases Changes to be committed: new file: sound/soc/codecs/tas2781-fmwlib.c --- sound/soc/codecs/tas2781-fmwlib.c | 2439 +++++++++++++++++++++++++++++ 1 file changed, 2439 insertions(+) create mode 100644 sound/soc/codecs/tas2781-fmwlib.c diff --git a/sound/soc/codecs/tas2781-fmwlib.c b/sound/soc/codecs/tas2781-fmwlib.c new file mode 100644 index 000000000000..a2ecc32a870e --- /dev/null +++ b/sound/soc/codecs/tas2781-fmwlib.c @@ -0,0 +1,2439 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// tasdevice-fmw.c -- TASDEVICE firmware support +// +// Copyright 2023 Texas Instruments, Inc. +// +// Author: Shenghao Ding <shenghao-ding(a)ti.com> + +#include <linux/crc8.h> +#include <linux/firmware.h> +#include <linux/i2c.h> +#include <linux/init.h> +#include <linux/interrupt.h> +#include <linux/module.h> +#include <linux/of.h> +#include <linux/of_gpio.h> +#include <linux/of_irq.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include <sound/tas2781.h> + + +#define ERROR_PRAM_CRCCHK 0x0000000 +#define ERROR_YRAM_CRCCHK 0x0000001 +#define PPC_DRIVER_CRCCHK 0x00000200 + +#define TAS2781_SA_COEFF_SWAP_REG TASDEVICE_REG(0, 0x35, 0x2c) +#define TAS2781_YRAM_BOOK1 140 +#define TAS2781_YRAM1_PAGE 42 +#define TAS2781_YRAM1_START_REG 88 + +#define TAS2781_YRAM2_START_PAGE 43 +#define TAS2781_YRAM2_END_PAGE 49 +#define TAS2781_YRAM2_START_REG 8 +#define TAS2781_YRAM2_END_REG 127 + +#define TAS2781_YRAM3_PAGE 50 +#define TAS2781_YRAM3_START_REG 8 +#define TAS2781_YRAM3_END_REG 27 + +/*should not include B0_P53_R44-R47 */ +#define TAS2781_YRAM_BOOK2 0 +#define TAS2781_YRAM4_START_PAGE 50 +#define TAS2781_YRAM4_END_PAGE 60 + +#define TAS2781_YRAM5_PAGE 61 +#define TAS2781_YRAM5_START_REG TAS2781_YRAM3_START_REG +#define TAS2781_YRAM5_END_REG TAS2781_YRAM3_END_REG + +#define TASDEVICE_MAXPROGRAM_NUM_KERNEL 5 +#define TASDEVICE_MAXCONFIG_NUM_KERNEL_MULTIPLE_AMPS 64 +#define TASDEVICE_MAXCONFIG_NUM_KERNEL 10 +#define MAIN_ALL_DEVICES_1X 0x01 +#define MAIN_DEVICE_A_1X 0x02 +#define MAIN_DEVICE_B_1X 0x03 +#define MAIN_DEVICE_C_1X 0x04 +#define MAIN_DEVICE_D_1X 0x05 +#define COEFF_DEVICE_A_1X 0x12 +#define COEFF_DEVICE_B_1X 0x13 +#define COEFF_DEVICE_C_1X 0x14 +#define COEFF_DEVICE_D_1X 0x15 +#define PRE_DEVICE_A_1X 0x22 +#define PRE_DEVICE_B_1X 0x23 +#define PRE_DEVICE_C_1X 0x24 +#define PRE_DEVICE_D_1X 0x25 +#define PRE_SOFTWARE_RESET_DEVICE_A 0x41 +#define PRE_SOFTWARE_RESET_DEVICE_B 0x42 +#define PRE_SOFTWARE_RESET_DEVICE_C 0x43 +#define PRE_SOFTWARE_RESET_DEVICE_D 0x44 +#define POST_SOFTWARE_RESET_DEVICE_A 0x45 +#define POST_SOFTWARE_RESET_DEVICE_B 0x46 +#define POST_SOFTWARE_RESET_DEVICE_C 0x47 +#define POST_SOFTWARE_RESET_DEVICE_D 0x48 + +struct tas_crc { + unsigned char offset; + unsigned char len; +}; + +const char deviceNumber[TASDEVICE_DSP_TAS_MAX_DEVICE] = { + 1, 2, 1, 2, 1, 1, 0, 2, 4, 3, 1, 2, 3, 4 +}; + +static struct tasdevice_config_info *tasdevice_add_config( + struct tasdevice_priv *tas_priv, unsigned char *config_data, + unsigned int config_size, int *status) +{ + struct tasdevice_config_info *cfg_info; + struct tasdev_blk_data **bk_da; + unsigned int config_offset = 0; + unsigned int i; + + /* In most projects are many audio cases, such as music, handfree, + * receiver, games, audio-to-haptics, PMIC record, bypass mode, + * portrait, landscape, etc. Even in multiple audios, one or + * two of the chips will work for the special case, such as + * ultrasonic application. In order to support these variable-numbers + * of audio cases, flexible configs have been introduced in the + * dsp firmware. + */ + cfg_info = kzalloc(sizeof(struct tasdevice_config_info), GFP_KERNEL); + if (!cfg_info) { + *status = -ENOMEM; + goto out; + } + + if (tas_priv->rcabin.fw_hdr.binary_version_num >= 0x105) { + if (config_offset + 64 > (int)config_size) { + *status = -EINVAL; + dev_err(tas_priv->dev, "add conf: Out of boundary\n"); + goto out; + } + config_offset += 64; + } + + if (config_offset + 4 > (int)config_size) { + *status = -EINVAL; + dev_err(tas_priv->dev, "add config: Out of boundary\n"); + goto out; + } + + /* convert data[offset], data[offset + 1], data[offset + 2] and + * data[offset + 3] into host + */ + cfg_info->nblocks = + be32_to_cpup((__be32 *)&config_data[config_offset]); + config_offset += 4; + + /* Several kinds of dsp/algorithm firmwares can run on tas2781, + * the number and size of blk are not fixed and different among + * these firmwares. + */ + bk_da = cfg_info->blk_data = kcalloc(cfg_info->nblocks, + sizeof(struct tasdev_blk_data *), GFP_KERNEL); + if (!bk_da) { + *status = -ENOMEM; + goto out; + } + cfg_info->real_nblocks = 0; + for (i = 0; i < cfg_info->nblocks; i++) { + if (config_offset + 12 > config_size) { + *status = -EINVAL; + dev_err(tas_priv->dev, + "%s: Out of boundary: i = %d nblocks = %u!\n", + __func__, i, cfg_info->nblocks); + break; + } + bk_da[i] = kzalloc(sizeof(struct tasdev_blk_data), GFP_KERNEL); + if (!bk_da[i]) { + *status = -ENOMEM; + break; + } + + bk_da[i]->dev_idx = config_data[config_offset]; + config_offset++; + + bk_da[i]->block_type = config_data[config_offset]; + config_offset++; + + if (bk_da[i]->block_type == TASDEVICE_BIN_BLK_PRE_POWER_UP) { + if (bk_da[i]->dev_idx == 0) + cfg_info->active_dev = 1; + else + cfg_info->active_dev = 1 << + (bk_da[i]->dev_idx - 1); + + } + bk_da[i]->yram_checksum = + be16_to_cpup((__be16 *)&config_data[config_offset]); + config_offset += 2; + bk_da[i]->block_size = + be32_to_cpup((__be32 *)&config_data[config_offset]); + config_offset += 4; + + bk_da[i]->n_subblks = + be32_to_cpup((__be32 *)&config_data[config_offset]); + + config_offset += 4; + + if (config_offset + bk_da[i]->block_size > config_size) { + *status = -EINVAL; + dev_err(tas_priv->dev, + "%s: Out of boundary: i = %d blks = %u!\n", + __func__, i, cfg_info->nblocks); + break; + } + /* instead of kzalloc+memcpy */ + bk_da[i]->regdata = kmemdup(&config_data[config_offset], + bk_da[i]->block_size, GFP_KERNEL); + if (!bk_da[i]->regdata) { + *status = -ENOMEM; + goto out; + } + + config_offset += bk_da[i]->block_size; + cfg_info->real_nblocks += 1; + } + +out: + return cfg_info; +} + +int tasdevice_rca_parser(void *context, const struct firmware *fmw) +{ + struct tasdevice_priv *tas_priv = context; + struct tasdevice_config_info **cfg_info; + struct tasdevice_rca_hdr *fw_hdr; + struct tasdevice_rca *rca; + unsigned int total_config_sz = 0; + unsigned char *buf; + int offset = 0; + int ret = 0; + int i; + + rca = &(tas_priv->rcabin); + fw_hdr = &(rca->fw_hdr); + if (!fmw || !fmw->data) { + dev_err(tas_priv->dev, "Failed to read %s\n", + tas_priv->rca_binaryname); + tas_priv->fw_state = TASDEVICE_DSP_FW_FAIL; + ret = -EINVAL; + goto out; + } + buf = (unsigned char *)fmw->data; + + fw_hdr->img_sz = be32_to_cpup((__be32 *)&buf[offset]); + offset += 4; + if (fw_hdr->img_sz != fmw->size) { + dev_err(tas_priv->dev, + "File size not match, %d %u", (int)fmw->size, + fw_hdr->img_sz); + tas_priv->fw_state = TASDEVICE_DSP_FW_FAIL; + ret = -EINVAL; + goto out; + } + + fw_hdr->checksum = be32_to_cpup((__be32 *)&buf[offset]); + offset += 4; + fw_hdr->binary_version_num = be32_to_cpup((__be32 *)&buf[offset]); + if (fw_hdr->binary_version_num < 0x103) { + dev_err(tas_priv->dev, "File version 0x%04x is too low", + fw_hdr->binary_version_num); + tas_priv->fw_state = TASDEVICE_DSP_FW_FAIL; + ret = -EINVAL; + goto out; + } + offset += 4; + fw_hdr->drv_fw_version = be32_to_cpup((__be32 *)&buf[offset]); + offset += 8; + fw_hdr->plat_type = buf[offset]; + offset += 1; + fw_hdr->dev_family = buf[offset]; + offset += 1; + fw_hdr->reserve = buf[offset]; + offset += 1; + fw_hdr->ndev = buf[offset]; + offset += 1; + if (fw_hdr->ndev != tas_priv->ndev) { + dev_err(tas_priv->dev, + "ndev(%u) in rcabin mismatch ndev(%u) in DTS\n", + fw_hdr->ndev, tas_priv->ndev); + tas_priv->fw_state = TASDEVICE_DSP_FW_FAIL; + ret = -EINVAL; + goto out; + } + if (offset + TASDEVICE_DEVICE_SUM > fw_hdr->img_sz) { + dev_err(tas_priv->dev, "rca_ready: Out of boundary!\n"); + ret = -EINVAL; + tas_priv->fw_state = TASDEVICE_DSP_FW_FAIL; + goto out; + } + + for (i = 0; i < TASDEVICE_DEVICE_SUM; i++, offset++) + fw_hdr->devs[i] = buf[offset]; + + fw_hdr->nconfig = be32_to_cpup((__be32 *)&buf[offset]); + offset += 4; + + for (i = 0; i < TASDEVICE_CONFIG_SUM; i++) { + fw_hdr->config_size[i] = be32_to_cpup((__be32 *)&buf[offset]); + offset += 4; + total_config_sz += fw_hdr->config_size[i]; + } + + if (fw_hdr->img_sz - total_config_sz != (unsigned int)offset) { + dev_err(tas_priv->dev, "Bin file error!\n"); + ret = -EINVAL; + tas_priv->fw_state = TASDEVICE_DSP_FW_FAIL; + goto out; + } + + cfg_info = kcalloc(fw_hdr->nconfig, sizeof(*cfg_info), GFP_KERNEL); + if (!cfg_info) { + ret = -ENOMEM; + tas_priv->fw_state = TASDEVICE_DSP_FW_FAIL; + goto out; + } + rca->cfg_info = cfg_info; + rca->ncfgs = 0; + for (i = 0; i < (int)fw_hdr->nconfig; i++) { + rca->ncfgs += 1; + cfg_info[i] = tasdevice_add_config(tas_priv, &buf[offset], + fw_hdr->config_size[i], &ret); + if (ret) { + tas_priv->fw_state = TASDEVICE_DSP_FW_FAIL; + goto out; + } + offset += (int)fw_hdr->config_size[i]; + } +out: + return ret; +} +EXPORT_SYMBOL_NS_GPL(tasdevice_rca_parser, SND_SOC_TAS2781_FMWLIB); + +static int fw_parse_block_data_kernel(struct tasdevice_fw *tas_fmw, + struct tasdev_blk *block, const struct firmware *fmw, int offset) +{ + const unsigned char *data = fmw->data; + + if (offset + 16 > fmw->size) { + dev_err(tas_fmw->dev, "%s: File Size error\n", __func__); + offset = -EINVAL; + goto out; + } + + /* convert data[offset], data[offset + 1], data[offset + 2] and + * data[offset + 3] into host + */ + block->type = be32_to_cpup((__be32 *)&data[offset]); + offset += 4; + + block->is_pchksum_present = data[offset]; + offset++; + + block->pchksum = data[offset]; + offset++; + + block->is_ychksum_present = data[offset]; + offset++; + + block->ychksum = data[offset]; + offset++; + + block->blk_size = be32_to_cpup((__be32 *)&data[offset]); + offset += 4; + + block->nr_subblocks = be32_to_cpup((__be32 *)&data[offset]); + offset += 4; + + if (offset + block->blk_size > fmw->size) { + dev_err(tas_fmw->dev, "%s: nSublocks error\n", __func__); + offset = -EINVAL; + goto out; + } + /* instead of kzalloc+memcpy */ + block->data = kmemdup(&data[offset], block->blk_size, GFP_KERNEL); + if (!block->data) { + offset = -ENOMEM; + goto out; + } + offset += block->blk_size; + +out: + return offset; +} + +static int fw_parse_data_kernel(struct tasdevice_fw *tas_fmw, + struct tasdevice_data *img_data, const struct firmware *fmw, + int offset) +{ + const unsigned char *data = fmw->data; + struct tasdev_blk *blk; + unsigned int i; + + if (offset + 4 > fmw->size) { + dev_err(tas_fmw->dev, "%s: File Size error\n", __func__); + offset = -EINVAL; + goto out; + } + img_data->nr_blk = be32_to_cpup((__be32 *)&data[offset]); + offset += 4; + + img_data->dev_blks = kcalloc(img_data->nr_blk, + sizeof(struct tasdev_blk), GFP_KERNEL); + if (!img_data->dev_blks) { + offset = -ENOMEM; + goto out; + } + + for (i = 0; i < img_data->nr_blk; i++) { + blk = &(img_data->dev_blks[i]); + offset = fw_parse_block_data_kernel(tas_fmw, blk, fmw, offset); + if (offset < 0) { + offset = -EINVAL; + break; + } + } + +out: + return offset; +} + +static int fw_parse_program_data_kernel( + struct tasdevice_priv *tas_priv, struct tasdevice_fw *tas_fmw, + const struct firmware *fmw, int offset) +{ + struct tasdevice_prog *program; + unsigned int i; + + for (i = 0; i < tas_fmw->nr_programs; i++) { + program = &(tas_fmw->programs[i]); + if (offset + 72 > fmw->size) { + dev_err(tas_priv->dev, "%s: mpName error\n", __func__); + offset = -EINVAL; + goto out; + } + /*skip 72 unused byts*/ + offset += 72; + + offset = fw_parse_data_kernel(tas_fmw, &(program->dev_data), + fmw, offset); + if (offset < 0) + goto out; + } + +out: + return offset; +} + +static int fw_parse_configuration_data_kernel( + struct tasdevice_priv *tas_priv, + struct tasdevice_fw *tas_fmw, const struct firmware *fmw, int offset) +{ + const unsigned char *data = fmw->data; + struct tasdevice_config *config; + unsigned int i; + + for (i = 0; i < tas_fmw->nr_configurations; i++) { + config = &(tas_fmw->configs[i]); + if (offset + 80 > fmw->size) { + dev_err(tas_priv->dev, "%s: mpName error\n", __func__); + offset = -EINVAL; + goto out; + } + memcpy(config->name, &data[offset], 64); + /*skip extra 16 bytes*/ + offset += 80; + + offset = fw_parse_data_kernel(tas_fmw, &(config->dev_data), + fmw, offset); + if (offset < 0) + goto out; + } + +out: + return offset; +} + +static int fw_parse_variable_header_kernel( + struct tasdevice_priv *tas_priv, const struct firmware *fmw, + int offset) +{ + struct tasdevice_fw *tas_fmw = tas_priv->fmw; + struct tasdevice_dspfw_hdr *fw_hdr = &(tas_fmw->fw_hdr); + struct tasdevice_prog *program; + struct tasdevice_config *config; + const unsigned char *buf = fmw->data; + unsigned short max_confs; + unsigned int i; + + if (offset + 12 + 4 * TASDEVICE_MAXPROGRAM_NUM_KERNEL > fmw->size) { + dev_err(tas_priv->dev, "%s: File Size error\n", __func__); + offset = -EINVAL; + goto out; + } + fw_hdr->device_family = be16_to_cpup((__be16 *)&buf[offset]); + if (fw_hdr->device_family != 0) { + dev_err(tas_priv->dev, "%s:not TAS device\n", __func__); + offset = -EINVAL; + goto out; + } + offset += 2; + fw_hdr->device = be16_to_cpup((__be16 *)&buf[offset]); + if (fw_hdr->device >= TASDEVICE_DSP_TAS_MAX_DEVICE || + fw_hdr->device == 6) { + dev_err(tas_priv->dev, "Unsupported dev %d\n", fw_hdr->device); + offset = -EINVAL; + goto out; + } + offset += 2; + fw_hdr->ndev = deviceNumber[fw_hdr->device]; + + if (fw_hdr->ndev != tas_priv->ndev) { + dev_err(tas_priv->dev, + "%s: ndev(%u) in dspbin mismatch ndev(%u) in DTS\n", + __func__, fw_hdr->ndev, tas_priv->ndev); + offset = -EINVAL; + goto out; + } + + tas_fmw->nr_programs = be32_to_cpup((__be32 *)&buf[offset]); + offset += 4; + + if (tas_fmw->nr_programs == 0 || tas_fmw->nr_programs > + TASDEVICE_MAXPROGRAM_NUM_KERNEL) { + dev_err(tas_priv->dev, "mnPrograms is invalid\n"); + offset = -EINVAL; + goto out; + } + + tas_fmw->programs = kcalloc(tas_fmw->nr_programs, + sizeof(struct tasdevice_prog), GFP_KERNEL); + if (!tas_fmw->programs) { + offset = -ENOMEM; + goto out; + } + + for (i = 0; i < tas_fmw->nr_programs; i++) { + program = &(tas_fmw->programs[i]); + program->prog_size = be32_to_cpup((__be32 *)&buf[offset]); + offset += 4; + } + + /* Skip the unused prog_size */ + offset += 4 * (TASDEVICE_MAXPROGRAM_NUM_KERNEL - tas_fmw->nr_programs); + + tas_fmw->nr_configurations = be32_to_cpup((__be32 *)&buf[offset]); + offset += 4; + + /* The max number of config in firmware greater than 4 pieces of + * tas2781s is different from the one lower than 4 pieces of + * tas2781s. + */ + max_confs = (fw_hdr->ndev >= 4) ? + TASDEVICE_MAXCONFIG_NUM_KERNEL_MULTIPLE_AMPS : + TASDEVICE_MAXCONFIG_NUM_KERNEL; + if (tas_fmw->nr_configurations == 0 || + tas_fmw->nr_configurations > max_confs) { + dev_err(tas_priv->dev, "%s: Conf is invalid\n", __func__); + offset = -EINVAL; + goto out; + } + + if (offset + 4 * max_confs > fmw->size) { + dev_err(tas_priv->dev, "%s: mpConfigurations err\n", __func__); + offset = -EINVAL; + goto out; + } + + tas_fmw->configs = kcalloc(tas_fmw->nr_configurations, + sizeof(struct tasdevice_config), GFP_KERNEL); + if (!tas_fmw->configs) { + offset = -ENOMEM; + goto out; + } + + for (i = 0; i < tas_fmw->nr_programs; i++) { + config = &(tas_fmw->configs[i]); + config->cfg_size = be32_to_cpup((__be32 *)&buf[offset]); + offset += 4; + } + + /* Skip the unused configs */ + offset += 4 * (max_confs - tas_fmw->nr_programs); + +out: + return offset; +} + +static int tasdevice_process_block(void *context, unsigned char *data, + unsigned char dev_idx, int sublocksize) +{ + struct tasdevice_priv *tas_priv = (struct tasdevice_priv *)context; + int subblk_offset, chn, chnend, rc; + unsigned char subblk_typ = data[1]; + int blktyp = dev_idx & 0xC0; + int idx = dev_idx & 0x3F; + bool is_err = false; + + if (idx) { + chn = idx - 1; + chnend = idx; + } else { + if (tas_priv->set_global_mode) { + chn = tas_priv->ndev; + chnend = tas_priv->ndev + 1; + } else { + chn = 0; + chnend = tas_priv->ndev; + } + } + + for (; chn < chnend; chn++) { + if (!tas_priv->set_global_mode && + tas_priv->tasdevice[chn].is_loading == false) + continue; + + is_err = false; + subblk_offset = 2; + switch (subblk_typ) { + case TASDEVICE_CMD_SING_W: { + int i; + unsigned short len = be16_to_cpup((__be16 *)&data[2]); + + subblk_offset += 2; + if (subblk_offset + 4 * len > sublocksize) { + dev_err(tas_priv->dev, + "process_block: Out of boundary\n"); + is_err = true; + break; + } + + for (i = 0; i < len; i++) { + rc = tasdevice_dev_write(tas_priv, chn, + TASDEVICE_REG(data[subblk_offset], + data[subblk_offset + 1], + data[subblk_offset + 2]), + data[subblk_offset + 3]); + if (rc < 0) { + is_err = true; + dev_err(tas_priv->dev, + "process_block: single write error\n"); + } + subblk_offset += 4; + } + } + break; + case TASDEVICE_CMD_BURST: { + unsigned short len = be16_to_cpup((__be16 *)&data[2]); + + subblk_offset += 2; + if (subblk_offset + 4 + len > sublocksize) { + dev_err(tas_priv->dev, + "%s: BST Out of boundary\n", + __func__); + is_err = true; + break; + } + if (len % 4) { + dev_err(tas_priv->dev, + "%s:Bst-len(%u)not div by 4\n", + __func__, len); + break; + } + + rc = tasdevice_dev_bulk_write(tas_priv, chn, + TASDEVICE_REG(data[subblk_offset], + data[subblk_offset + 1], + data[subblk_offset + 2]), + &(data[subblk_offset + 4]), len); + if (rc < 0) { + is_err = true; + dev_err(tas_priv->dev, + "%s: bulk_write error = %d\n", + __func__, rc); + } + subblk_offset += (len + 4); + } + break; + case TASDEVICE_CMD_DELAY: { + unsigned int sleep_time = 0; + + if (subblk_offset + 2 > sublocksize) { + dev_err(tas_priv->dev, + "%s: delay Out of boundary\n", + __func__); + is_err = true; + break; + } + sleep_time = be16_to_cpup((__be16 *)&data[2]) * 1000; + usleep_range(sleep_time, sleep_time + 50); + subblk_offset += 2; + } + break; + case TASDEVICE_CMD_FIELD_W: + if (subblk_offset + 6 > sublocksize) { + dev_err(tas_priv->dev, + "%s: bit write Out of boundary\n", + __func__); + is_err = true; + break; + } + rc = tasdevice_dev_update_bits(tas_priv, chn, + TASDEVICE_REG(data[subblk_offset + 2], + data[subblk_offset + 3], + data[subblk_offset + 4]), + data[subblk_offset + 1], + data[subblk_offset + 5]); + if (rc < 0) { + is_err = true; + dev_err(tas_priv->dev, + "%s: update_bits error = %d\n", + __func__, rc); + } + subblk_offset += 6; + break; + default: + break; + } + if (is_err == true && blktyp != 0) { + if (blktyp == 0x80) { + tas_priv->tasdevice[chn].cur_prog = -1; + tas_priv->tasdevice[chn].cur_conf = -1; + } else + tas_priv->tasdevice[chn].cur_conf = -1; + } + } + + return subblk_offset; +} + +void tasdevice_select_cfg_blk(void *pContext, int conf_no, + unsigned char block_type) +{ + struct tasdevice_priv *tas_priv = (struct tasdevice_priv *) pContext; + struct tasdevice_rca *rca = &(tas_priv->rcabin); + struct tasdevice_config_info **cfg_info = rca->cfg_info; + struct tasdev_blk_data **blk_data; + int j, k, chn, chnend; + + if (conf_no >= rca->ncfgs || conf_no < 0 || !cfg_info) { + dev_err(tas_priv->dev, "conf_no should be not more than %u\n", + rca->ncfgs); + return; + } + blk_data = cfg_info[conf_no]->blk_data; + + for (j = 0; j < (int)cfg_info[conf_no]->real_nblocks; j++) { + unsigned int length = 0, rc = 0; + + if (block_type > 5 || block_type < 2) { + dev_err(tas_priv->dev, + "block_type should be in range from 2 to 5\n"); + break; + } + if (block_type != blk_data[j]->block_type) + continue; + + for (k = 0; k < (int)blk_data[j]->n_subblks; k++) { + if (blk_data[j]->dev_idx) { + chn = blk_data[j]->dev_idx - 1; + chnend = blk_data[j]->dev_idx; + } else { + chn = 0; + chnend = tas_priv->ndev; + } + for (; chn < chnend; chn++) + tas_priv->tasdevice[chn].is_loading = true; + + rc = tasdevice_process_block(tas_priv, + blk_data[j]->regdata + length, + blk_data[j]->dev_idx, + blk_data[j]->block_size - length); + length += rc; + if (blk_data[j]->block_size < length) { + dev_err(tas_priv->dev, + "%s: %u %u out of boundary\n", + __func__, length, + blk_data[j]->block_size); + break; + } + } + if (length != blk_data[j]->block_size) + dev_err(tas_priv->dev, "%s: %u %u size is not same\n", + __func__, length, blk_data[j]->block_size); + } +} +EXPORT_SYMBOL_NS_GPL(tasdevice_select_cfg_blk, SND_SOC_TAS2781_FMWLIB); + +static int tasdevice_load_block_kernel( + struct tasdevice_priv *tasdevice, struct tasdev_blk *block) +{ + struct tasdevice_dspfw_hdr *fw_hdr = &(tasdevice->fmw->fw_hdr); + struct tasdevice_fw_fixed_hdr *fw_fixed_hdr = &(fw_hdr->fixed_hdr); + const unsigned int blk_size = block->blk_size; + unsigned int i, length; + unsigned char *data = block->data; + unsigned char dev_idx = 0; + + if (fw_fixed_hdr->ppcver >= PPC3_VERSION_TAS2781) { + switch (block->type) { + case MAIN_ALL_DEVICES_1X: + dev_idx = 0x80; + break; + case MAIN_DEVICE_A_1X: + dev_idx = 0x81; + break; + case COEFF_DEVICE_A_1X: + case PRE_DEVICE_A_1X: + case PRE_SOFTWARE_RESET_DEVICE_A: + case POST_SOFTWARE_RESET_DEVICE_A: + dev_idx = 0xC1; + break; + case MAIN_DEVICE_B_1X: + dev_idx = 0x82; + break; + case COEFF_DEVICE_B_1X: + case PRE_DEVICE_B_1X: + case PRE_SOFTWARE_RESET_DEVICE_B: + case POST_SOFTWARE_RESET_DEVICE_B: + dev_idx = 0xC2; + break; + case MAIN_DEVICE_C_1X: + dev_idx = 0x83; + break; + case COEFF_DEVICE_C_1X: + case PRE_DEVICE_C_1X: + case PRE_SOFTWARE_RESET_DEVICE_C: + case POST_SOFTWARE_RESET_DEVICE_C: + dev_idx = 0xC3; + break; + case MAIN_DEVICE_D_1X: + dev_idx = 0x84; + break; + case COEFF_DEVICE_D_1X: + case PRE_DEVICE_D_1X: + case PRE_SOFTWARE_RESET_DEVICE_D: + case POST_SOFTWARE_RESET_DEVICE_D: + dev_idx = 0xC4; + break; + default: + dev_info(tasdevice->dev, + "%s: load block: Other Type = 0x%02x\n", + __func__, block->type); + break; + } + } else if (fw_fixed_hdr->ppcver >= + PPC3_VERSION) { + switch (block->type) { + case MAIN_ALL_DEVICES_1X: + dev_idx = 0x80; + break; + case MAIN_DEVICE_A_1X: + dev_idx = 0x81; + break; + case COEFF_DEVICE_A_1X: + case PRE_DEVICE_A_1X: + dev_idx = 0xC1; + break; + case MAIN_DEVICE_B_1X: + dev_idx = 0x82; + break; + case COEFF_DEVICE_B_1X: + case PRE_DEVICE_B_1X: + dev_idx = 0xC2; + break; + case MAIN_DEVICE_C_1X: + dev_idx = 0x83; + break; + case COEFF_DEVICE_C_1X: + case PRE_DEVICE_C_1X: + dev_idx = 0xC3; + break; + case MAIN_DEVICE_D_1X: + dev_idx = 0x84; + break; + case COEFF_DEVICE_D_1X: + case PRE_DEVICE_D_1X: + dev_idx = 0xC4; + break; + default: + dev_info(tasdevice->dev, + "%s: load block: Other Type = 0x%02x\n", + __func__, block->type); + break; + } + } else { + switch (block->type) { + case MAIN_ALL_DEVICES: + dev_idx = 0|0x80; + break; + case MAIN_DEVICE_A: + dev_idx = 0x81; + break; + case COEFF_DEVICE_A: + case PRE_DEVICE_A: + dev_idx = 0xC1; + break; + case MAIN_DEVICE_B: + dev_idx = 0x82; + break; + case COEFF_DEVICE_B: + case PRE_DEVICE_B: + dev_idx = 0xC2; + break; + case MAIN_DEVICE_C: + dev_idx = 0x83; + break; + case COEFF_DEVICE_C: + case PRE_DEVICE_C: + dev_idx = 0xC3; + break; + case MAIN_DEVICE_D: + dev_idx = 0x84; + break; + case COEFF_DEVICE_D: + case PRE_DEVICE_D: + dev_idx = 0xC4; + break; + default: + dev_info(tasdevice->dev, + "%s: load block: Other Type = 0x%02x\n", + __func__, block->type); + break; + } + } + + for (i = 0, length = 0; i < block->nr_subblocks; i++) { + int rc = tasdevice_process_block(tasdevice, data + length, + dev_idx, blk_size - length); + if (rc < 0) { + dev_err(tasdevice->dev, + "%s: %u %u sublock write error\n", + __func__, length, blk_size); + break; + } + length += (unsigned int)rc; + if (blk_size < length) { + dev_err(tasdevice->dev, "%s: %u %u out of boundary\n", + __func__, length, blk_size); + break; + } + } + + return 0; +} + +static int fw_parse_variable_hdr(struct tasdevice_priv + *tas_priv, struct tasdevice_dspfw_hdr *fw_hdr, + const struct firmware *fmw, int offset) +{ + const unsigned char *buf = fmw->data; + int len = strlen((char *)&buf[offset]); + + len++; + + if (offset + len + 8 > fmw->size) { + dev_err(tas_priv->dev, "%s: File Size error\n", __func__); + offset = -EINVAL; + goto out; + } + + offset += len; + + fw_hdr->device_family = be32_to_cpup((__be32 *)&buf[offset]); + if (fw_hdr->device_family != 0) { + dev_err(tas_priv->dev, "%s: not TAS device\n", __func__); + offset = -EINVAL; + goto out; + } + offset += 4; + + fw_hdr->device = be32_to_cpup((__be32 *)&buf[offset]); + if (fw_hdr->device >= TASDEVICE_DSP_TAS_MAX_DEVICE || + fw_hdr->device == 6) { + dev_err(tas_priv->dev, "Unsupported dev %d\n", fw_hdr->device); + offset = -EINVAL; + goto out; + } + offset += 4; + fw_hdr->ndev = deviceNumber[fw_hdr->device]; + +out: + return offset; +} + +static int fw_parse_variable_header_git(struct tasdevice_priv + *tas_priv, const struct firmware *fmw, int offset) +{ + struct tasdevice_fw *tas_fmw = tas_priv->fmw; + struct tasdevice_dspfw_hdr *fw_hdr = &(tas_fmw->fw_hdr); + + offset = fw_parse_variable_hdr(tas_priv, fw_hdr, fmw, offset); + if (offset < 0) + goto out; + if (fw_hdr->ndev != tas_priv->ndev) { + dev_err(tas_priv->dev, + "%s: ndev(%u) in dspbin mismatch ndev(%u) in DTS\n", + __func__, fw_hdr->ndev, tas_priv->ndev); + offset = -EINVAL; + } + +out: + return offset; +} + +static int fw_parse_block_data(struct tasdevice_fw *tas_fmw, + struct tasdev_blk *block, const struct firmware *fmw, int offset) +{ + unsigned char *data = (unsigned char *)fmw->data; + int n; + + if (offset + 8 > fmw->size) { + dev_err(tas_fmw->dev, "%s: Type error\n", __func__); + offset = -EINVAL; + goto out; + } + block->type = be32_to_cpup((__be32 *)&data[offset]); + offset += 4; + + if (tas_fmw->fw_hdr.fixed_hdr.drv_ver >= PPC_DRIVER_CRCCHK) { + if (offset + 8 > fmw->size) { + dev_err(tas_fmw->dev, "PChkSumPresent error\n"); + offset = -EINVAL; + goto out; + } + block->is_pchksum_present = data[offset]; + offset++; + + block->pchksum = data[offset]; + offset++; + + block->is_ychksum_present = data[offset]; + offset++; + + block->ychksum = data[offset]; + offset++; + } else { + block->is_pchksum_present = 0; + block->is_ychksum_present = 0; + } + + block->nr_cmds = be32_to_cpup((__be32 *)&data[offset]); + offset += 4; + + n = block->nr_cmds * 4; + if (offset + n > fmw->size) { + dev_err(tas_fmw->dev, + "%s: File Size(%lu) error offset = %d n = %d\n", + __func__, (unsigned long)fmw->size, offset, n); + offset = -EINVAL; + goto out; + } + /* instead of kzalloc+memcpy */ + block->data = kmemdup(&data[offset], n, GFP_KERNEL); + if (!block->data) { + offset = -ENOMEM; + goto out; + } + offset += n; + +out: + return offset; +} + +/* When parsing error occurs, all the memory resource will be released + * in the end of tasdevice_rca_ready. + */ +static int fw_parse_data(struct tasdevice_fw *tas_fmw, + struct tasdevice_data *img_data, const struct firmware *fmw, + int offset) +{ + const unsigned char *data = (unsigned char *)fmw->data; + struct tasdev_blk *blk; + unsigned int i; + int n; + + if (offset + 64 > fmw->size) { + dev_err(tas_fmw->dev, "%s: Name error\n", __func__); + offset = -EINVAL; + goto out; + } + memcpy(img_data->name, &data[offset], 64); + offset += 64; + + n = strlen((char *)&data[offset]); + n++; + if (offset + n + 2 > fmw->size) { + dev_err(tas_fmw->dev, "%s: Description error\n", __func__); + offset = -EINVAL; + goto out; + } + offset += n; + img_data->nr_blk = be16_to_cpup((__be16 *)&data[offset]); + offset += 2; + + img_data->dev_blks = kcalloc(img_data->nr_blk, + sizeof(struct tasdev_blk), GFP_KERNEL); + if (!img_data->dev_blks) { + offset = -ENOMEM; + goto out; + } + for (i = 0; i < img_data->nr_blk; i++) { + blk = &(img_data->dev_blks[i]); + offset = fw_parse_block_data(tas_fmw, blk, fmw, offset); + if (offset < 0) { + offset = -EINVAL; + goto out; + } + } + +out: + return offset; +} + +/* When parsing error occurs, all the memory resource will be released + * in the end of tasdevice_rca_ready. + */ +static int fw_parse_program_data(struct tasdevice_priv *tas_priv, + struct tasdevice_fw *tas_fmw, const struct firmware *fmw, int offset) +{ + unsigned char *buf = (unsigned char *)fmw->data; + struct tasdevice_prog *program; + int i; + + if (offset + 2 > fmw->size) { + dev_err(tas_priv->dev, "%s: File Size error\n", __func__); + offset = -EINVAL; + goto out; + } + tas_fmw->nr_programs = be16_to_cpup((__be16 *)&buf[offset]); + offset += 2; + + if (tas_fmw->nr_programs == 0) { + /*Not error in calibration Data file, return directly*/ + dev_info(tas_priv->dev, "%s: No Programs data, maybe calbin\n", + __func__); + goto out; + } + + tas_fmw->programs = + kcalloc(tas_fmw->nr_programs, sizeof(struct tasdevice_prog), + GFP_KERNEL); + if (!tas_fmw->programs) { + offset = -ENOMEM; + goto out; + } + for (i = 0; i < tas_fmw->nr_programs; i++) { + int n = 0; + + program = &(tas_fmw->programs[i]); + if (offset + 64 > fmw->size) { + dev_err(tas_priv->dev, "%s: mpName error\n", __func__); + offset = -EINVAL; + goto out; + } + offset += 64; + + n = strlen((char *)&buf[offset]); + /* skip '\0' and 5 unused bytes */ + n += 6; + if (offset + n > fmw->size) { + dev_err(tas_priv->dev, "Description err\n"); + offset = -EINVAL; + goto out; + } + + offset += n; + + offset = fw_parse_data(tas_fmw, &(program->dev_data), fmw, + offset); + if (offset < 0) + goto out; + } + +out: + return offset; +} + +/* When parsing error occurs, all the memory resource will be released + * in the end of tasdevice_rca_ready. + */ +static int fw_parse_configuration_data( + struct tasdevice_priv *tas_priv, + struct tasdevice_fw *tas_fmw, + const struct firmware *fmw, int offset) +{ + unsigned char *data = (unsigned char *)fmw->data; + struct tasdevice_config *config; + unsigned int i; + int n; + + if (offset + 2 > fmw->size) { + dev_err(tas_priv->dev, "%s: File Size error\n", __func__); + offset = -EINVAL; + goto out; + } + tas_fmw->nr_configurations = be16_to_cpup((__be16 *)&data[offset]); + offset += 2; + + if (tas_fmw->nr_configurations == 0) { + dev_err(tas_priv->dev, "%s: Conf is zero\n", __func__); + /*Not error for calibration Data file, return directly*/ + goto out; + } + tas_fmw->configs = kcalloc(tas_fmw->nr_configurations, + sizeof(struct tasdevice_config), GFP_KERNEL); + if (!tas_fmw->configs) { + offset = -ENOMEM; + goto out; + } + for (i = 0; i < tas_fmw->nr_configurations; i++) { + config = &(tas_fmw->configs[i]); + if (offset + 64 > fmw->size) { + dev_err(tas_priv->dev, "File Size err\n"); + offset = -EINVAL; + goto out; + } + memcpy(config->name, &data[offset], 64); + offset += 64; + + n = strlen((char *)&data[offset]); + n += 15; + if (offset + n > fmw->size) { + dev_err(tas_priv->dev, "Description err\n"); + offset = -EINVAL; + goto out; + } + + offset += n; + + offset = fw_parse_data(tas_fmw, &(config->dev_data), + fmw, offset); + if (offset < 0) + goto out; + } + +out: + return offset; +} + +static bool check_inpage_yram_rg(struct tas_crc *cd, + unsigned char reg, unsigned char len) +{ + bool in = false; + + + if (reg <= TAS2781_YRAM5_END_REG && + reg >= TAS2781_YRAM5_START_REG) { + if (reg + len > TAS2781_YRAM5_END_REG) + cd->len = TAS2781_YRAM5_END_REG - reg + 1; + else + cd->len = len; + cd->offset = reg; + in = true; + } else if (reg < TAS2781_YRAM5_START_REG) { + if (reg + len > TAS2781_YRAM5_START_REG) { + cd->offset = TAS2781_YRAM5_START_REG; + cd->len = len - TAS2781_YRAM5_START_REG + reg; + in = true; + } + } + + return in; +} + +static bool check_inpage_yram_bk1(struct tas_crc *cd, + unsigned char page, unsigned char reg, unsigned char len) +{ + bool in = false; + + if (page == TAS2781_YRAM1_PAGE) { + if (reg >= TAS2781_YRAM1_START_REG) { + cd->offset = reg; + cd->len = len; + in = true; + } else if (reg + len > TAS2781_YRAM1_START_REG) { + cd->offset = TAS2781_YRAM1_START_REG; + cd->len = len - TAS2781_YRAM1_START_REG + reg; + in = true; + } + } else if (page == TAS2781_YRAM3_PAGE) + in = check_inpage_yram_rg(cd, reg, len); + + return in; +} + +/* Return Code: + * true -- the registers are in the inpage yram + * false -- the registers are NOT in the inpage yram + */ +static bool check_inpage_yram(struct tas_crc *cd, unsigned char book, + unsigned char page, unsigned char reg, unsigned char len) +{ + bool in = false; + + if (book == TAS2781_YRAM_BOOK1) { + in = check_inpage_yram_bk1(cd, page, reg, len); + goto end; + } + if (book == TAS2781_YRAM_BOOK2 && page == TAS2781_YRAM5_PAGE) + in = check_inpage_yram_rg(cd, reg, len); + +end: + return in; +} + +static bool check_inblock_yram_bk(struct tas_crc *cd, + unsigned char page, unsigned char reg, unsigned char len) +{ + bool in = false; + + if ((page >= TAS2781_YRAM4_START_PAGE && + page <= TAS2781_YRAM4_END_PAGE) || + (page >= TAS2781_YRAM2_START_PAGE && + page <= TAS2781_YRAM2_END_PAGE)) { + if (reg <= TAS2781_YRAM2_END_REG && + reg >= TAS2781_YRAM2_START_REG) { + cd->offset = reg; + cd->len = len; + in = true; + } else if (reg < TAS2781_YRAM2_START_REG) { + if (reg + len - 1 >= TAS2781_YRAM2_START_REG) { + cd->offset = TAS2781_YRAM2_START_REG; + cd->len = reg + len - TAS2781_YRAM2_START_REG; + in = true; + } + } + } + + return in; +} + +/* Return Code: + * true -- the registers are in the inblock yram + * false -- the registers are NOT in the inblock yram + */ +static bool check_inblock_yram(struct tas_crc *cd, unsigned char book, + unsigned char page, unsigned char reg, unsigned char len) +{ + bool in = false; + + if (book == TAS2781_YRAM_BOOK1 || book == TAS2781_YRAM_BOOK2) + in = check_inblock_yram_bk(cd, page, reg, len); + + return in; +} + +static bool check_yram(struct tas_crc *cd, unsigned char book, + unsigned char page, unsigned char reg, unsigned char len) +{ + bool in; + + in = check_inpage_yram(cd, book, page, reg, len); + if (in) + goto end; + in = check_inblock_yram(cd, book, page, reg, len); + +end: + return in; +} + +static int tasdev_multibytes_chksum(struct tasdevice_priv *tasdevice, + enum channel chn, unsigned char book, unsigned char page, + unsigned char reg, unsigned int len) +{ + struct tas_crc crc_data; + unsigned char crc_chksum = 0; + unsigned char nBuf1[128]; + int ret = 0; + int i; + bool in; + + if ((reg + len - 1) > 127) { + ret = -EINVAL; + dev_err(tasdevice->dev, "firmware error\n"); + goto end; + } + + if ((book == TASDEVICE_BOOK_ID(TAS2781_SA_COEFF_SWAP_REG)) + && (page == TASDEVICE_PAGE_ID(TAS2781_SA_COEFF_SWAP_REG)) + && (reg == TASDEVICE_PAGE_REG(TAS2781_SA_COEFF_SWAP_REG)) + && (len == 4)) { + /*DSP swap command, pass */ + ret = 0; + goto end; + } + + in = check_yram(&crc_data, book, page, reg, len); + if (!in) + goto end; + + if (len == 1) { + dev_err(tasdevice->dev, "firmware error\n"); + ret = -EINVAL; + goto end; + } + + ret = tasdevice_dev_bulk_read(tasdevice, chn, + TASDEVICE_REG(book, page, crc_data.offset), + nBuf1, crc_data.len); + if (ret < 0) + goto end; + + for (i = 0; i < crc_data.len; i++) { + if ((book == TASDEVICE_BOOK_ID(TAS2781_SA_COEFF_SWAP_REG)) + && (page == TASDEVICE_PAGE_ID( + TAS2781_SA_COEFF_SWAP_REG)) + && ((i + crc_data.offset) + >= TASDEVICE_PAGE_REG(TAS2781_SA_COEFF_SWAP_REG)) + && ((i + crc_data.offset) + <= (TASDEVICE_PAGE_REG(TAS2781_SA_COEFF_SWAP_REG) + + 4))) + /*DSP swap command, bypass */ + continue; + else + crc_chksum += crc8(tasdevice->crc8_lkp_tbl, &nBuf1[i], + 1, 0); + } + + ret = crc_chksum; + +end: + return ret; +} + +static int do_singlereg_checksum(struct tasdevice_priv *tasdevice, + enum channel chl, unsigned char book, unsigned char page, + unsigned char reg, unsigned char val) +{ + struct tas_crc crc_data; + unsigned int nData1; + int ret = 0; + bool in; + + if ((book == TASDEVICE_BOOK_ID(TAS2781_SA_COEFF_SWAP_REG)) + && (page == TASDEVICE_PAGE_ID(TAS2781_SA_COEFF_SWAP_REG)) + && (reg >= TASDEVICE_PAGE_REG(TAS2781_SA_COEFF_SWAP_REG)) + && (reg <= (TASDEVICE_PAGE_REG( + TAS2781_SA_COEFF_SWAP_REG) + 4))) { + /*DSP swap command, pass */ + ret = 0; + goto end; + } + + in = check_yram(&crc_data, book, page, reg, 1); + if (!in) + goto end; + ret = tasdevice_dev_read(tasdevice, chl, + TASDEVICE_REG(book, page, reg), &nData1); + if (ret < 0) + goto end; + + if (nData1 != val) { + dev_err(tasdevice->dev, + "B[0x%x]P[0x%x]R[0x%x] W[0x%x], R[0x%x]\n", + book, page, reg, val, nData1); + tasdevice->tasdevice[chl].err_code |= ERROR_YRAM_CRCCHK; + ret = -EAGAIN; + goto end; + } + + ret = crc8(tasdevice->crc8_lkp_tbl, &val, 1, 0); + +end: + return ret; +} + +static void set_err_prg_cfg(unsigned int type, struct tasdevice *dev) +{ + if ((type == MAIN_ALL_DEVICES) || (type == MAIN_DEVICE_A) + || (type == MAIN_DEVICE_B) || (type == MAIN_DEVICE_C) + || (type == MAIN_DEVICE_D)) + dev->cur_prog = -1; + else + dev->cur_conf = -1; +} + +static int tasdev_bytes_chksum(struct tasdevice_priv *tas_priv, + struct tasdev_blk *block, int chn, unsigned char book, + unsigned char page, unsigned char reg, unsigned int len, + unsigned char val, unsigned char *crc_chksum) +{ + int ret; + + if (len > 1) + ret = tasdev_multibytes_chksum(tas_priv, chn, book, page, reg, + len); + else + ret = do_singlereg_checksum(tas_priv, chn, book, page, reg, + val); + + if (ret > 0) { + *crc_chksum += (unsigned char)ret; + goto end; + } + + if (ret != -EAGAIN) + goto end; + + block->nr_retry--; + if (block->nr_retry > 0) + goto end; + + set_err_prg_cfg(block->type, &tas_priv->tasdevice[chn]); + +end: + return ret; +} + +static int tasdev_multibytes_wr(struct tasdevice_priv *tas_priv, + struct tasdev_blk *block, int chn, unsigned char book, + unsigned char page, unsigned char reg, unsigned char *data, + unsigned int len, unsigned int *nr_cmds, + unsigned char *crc_chksum) +{ + int ret; + + if (len > 1) { + ret = tasdevice_dev_bulk_write(tas_priv, chn, + TASDEVICE_REG(book, page, reg), data + 3, len); + if (ret < 0) + goto end; + if (block->is_ychksum_present) + ret = tasdev_bytes_chksum(tas_priv, block, chn, + book, page, reg, len, 0, crc_chksum); + } else { + ret = tasdevice_dev_write(tas_priv, chn, + TASDEVICE_REG(book, page, reg), data[3]); + if (ret < 0) + goto end; + if (block->is_ychksum_present) + ret = tasdev_bytes_chksum(tas_priv, block, chn, book, + page, reg, 1, data[3], crc_chksum); + } + + if (!block->is_ychksum_present || ret >= 0) { + *nr_cmds += 1; + if (len >= 2) + *nr_cmds += ((len - 2) / 4) + 1; + } + +end: + return ret; +} + +static int tasdev_block_chksum(struct tasdevice_priv *tas_priv, + struct tasdev_blk *block, int chn) +{ + unsigned int nr_value; + int ret; + + ret = tasdevice_dev_read(tas_priv, chn, TASDEVICE_I2CChecksum, + &nr_value); + if (ret < 0) { + dev_err(tas_priv->dev, "%s: Chn %d\n", __func__, chn); + set_err_prg_cfg(block->type, &tas_priv->tasdevice[chn]); + goto end; + } + + if ((nr_value & 0xff) != block->pchksum) { + dev_err(tas_priv->dev, "%s: Blk PChkSum Chn %d ", __func__, + chn); + dev_err(tas_priv->dev, "PChkSum = 0x%x, Reg = 0x%x\n", + block->pchksum, (nr_value & 0xff)); + tas_priv->tasdevice[chn].err_code |= ERROR_PRAM_CRCCHK; + ret = -EAGAIN; + block->nr_retry--; + + if (block->nr_retry <= 0) + set_err_prg_cfg(block->type, + &tas_priv->tasdevice[chn]); + } else + tas_priv->tasdevice[chn].err_code &= ~ERROR_PRAM_CRCCHK; + +end: + return ret; +} + +static int tasdev_load_blk(struct tasdevice_priv *tas_priv, + struct tasdev_blk *block, int chn) +{ + unsigned int sleep_time; + unsigned int len; + unsigned int nr_cmds; + unsigned char *data = block->data; + unsigned char crc_chksum = 0; + unsigned char offset; + unsigned char book; + unsigned char page; + unsigned char val; + int ret = 0; + + while (block->nr_retry > 0) { + if (block->is_pchksum_present) { + ret = tasdevice_dev_write(tas_priv, chn, + TASDEVICE_I2CChecksum, 0); + if (ret < 0) + break; + } + + if (block->is_ychksum_present) + crc_chksum = 0; + + nr_cmds = 0; + + while (nr_cmds < block->nr_cmds) { + data = block->data + nr_cmds * 4; + + book = data[0]; + page = data[1]; + offset = data[2]; + val = data[3]; + + nr_cmds++; + /*Single byte write*/ + if (offset <= 0x7F) { + ret = tasdevice_dev_write(tas_priv, chn, + TASDEVICE_REG(book, page, offset), + val); + if (ret < 0) + goto end; + if (block->is_ychksum_present) { + ret = tasdev_bytes_chksum(tas_priv, + block, chn, book, page, offset, + 1, val, &crc_chksum); + if (ret < 0) + break; + } + continue; + } + /*sleep command*/ + if (offset == 0x81) { + /*book -- data[0] page -- data[1]*/ + sleep_time = ((book << 8) + page)*1000; + usleep_range(sleep_time, sleep_time + 50); + continue; + } + /*Multiple bytes write*/ + if (offset == 0x85) { + data += 4; + len = (book << 8) + page; + book = data[0]; + page = data[1]; + offset = data[2]; + ret = tasdev_multibytes_wr(tas_priv, + block, chn, book, page, offset, data, + len, &nr_cmds, &crc_chksum); + if (ret < 0) + break; + } + } + if (ret == -EAGAIN) { + if (block->nr_retry > 0) + continue; + } else if (ret < 0) /*err in current device, skip it*/ + break; + + if (block->is_pchksum_present) { + ret = tasdev_block_chksum(tas_priv, block, chn); + if (ret == -EAGAIN) { + if (block->nr_retry > 0) + continue; + } else if (ret < 0) /*err in current device, skip it*/ + break; + } + + if (block->is_ychksum_present) { + /* TBD, open it when FW ready */ + dev_err(tas_priv->dev, + "Blk YChkSum: FW = 0x%x, YCRC = 0x%x\n", + block->ychksum, crc_chksum); + + tas_priv->tasdevice[chn].err_code &= + ~ERROR_YRAM_CRCCHK; + ret = 0; + } + /*skip current blk*/ + break; + } + +end: + return ret; +} + +static int tasdevice_load_block(struct tasdevice_priv *tas_priv, + struct tasdev_blk *block) +{ + int chnend = 0; + int ret = 0; + int chn = 0; + int rc = 0; + + switch (block->type) { + case MAIN_ALL_DEVICES: + chn = 0; + chnend = tas_priv->ndev; + break; + case MAIN_DEVICE_A: + case COEFF_DEVICE_A: + case PRE_DEVICE_A: + chn = 0; + chnend = 1; + break; + case MAIN_DEVICE_B: + case COEFF_DEVICE_B: + case PRE_DEVICE_B: + chn = 1; + chnend = 2; + break; + case MAIN_DEVICE_C: + case COEFF_DEVICE_C: + case PRE_DEVICE_C: + chn = 2; + chnend = 3; + break; + case MAIN_DEVICE_D: + case COEFF_DEVICE_D: + case PRE_DEVICE_D: + chn = 3; + chnend = 4; + break; + default: + dev_dbg(tas_priv->dev, "load blk: Other Type = 0x%02x\n", + block->type); + break; + } + + for (; chn < chnend; chn++) { + block->nr_retry = 6; + if (tas_priv->tasdevice[chn].is_loading == false) + continue; + ret = tasdev_load_blk(tas_priv, block, chn); + if (ret < 0) + dev_err(tas_priv->dev, "dev %d, Blk (%d) load error\n", + chn, block->type); + rc |= ret; + } + + return rc; +} + +static int dspfw_default_callback(struct tasdevice_priv *tas_priv, + unsigned int drv_ver, unsigned int ppcver) +{ + int rc = 0; + + if (drv_ver == 0x100) { + if (ppcver >= PPC3_VERSION) { + tas_priv->fw_parse_variable_header = + fw_parse_variable_header_kernel; + tas_priv->fw_parse_program_data = + fw_parse_program_data_kernel; + tas_priv->fw_parse_configuration_data = + fw_parse_configuration_data_kernel; + tas_priv->tasdevice_load_block = + tasdevice_load_block_kernel; + } else { + switch (ppcver) { + case 0x00: + tas_priv->fw_parse_variable_header = + fw_parse_variable_header_git; + tas_priv->fw_parse_program_data = + fw_parse_program_data; + tas_priv->fw_parse_configuration_data = + fw_parse_configuration_data; + tas_priv->tasdevice_load_block = + tasdevice_load_block; + break; + default: + dev_err(tas_priv->dev, + "%s: PPCVer must be 0x0 or 0x%02x", + __func__, PPC3_VERSION); + dev_err(tas_priv->dev, " Current:0x%02x\n", + ppcver); + rc = -EINVAL; + break; + } + } + } else { + dev_err(tas_priv->dev, + "DrvVer must be 0x0, 0x230 or above 0x230 "); + dev_err(tas_priv->dev, "current is 0x%02x\n", drv_ver); + rc = -EINVAL; + } + + return rc; +} + +static int load_calib_data(struct tasdevice_priv *tas_priv, + struct tasdevice_data *dev_data) +{ + struct tasdev_blk *block; + unsigned int i; + int ret = 0; + + for (i = 0; i < dev_data->nr_blk; i++) { + block = &(dev_data->dev_blks[i]); + ret = tasdevice_load_block(tas_priv, block); + if (ret < 0) + break; + } + + return ret; +} + +static int fw_parse_header(struct tasdevice_priv *tas_priv, + struct tasdevice_fw *tas_fmw, const struct firmware *fmw, int offset) +{ + struct tasdevice_dspfw_hdr *fw_hdr = &(tas_fmw->fw_hdr); + struct tasdevice_fw_fixed_hdr *fw_fixed_hdr = &(fw_hdr->fixed_hdr); + const unsigned char magic_number[] = { 0x35, 0x35, 0x35, 0x32 }; + const unsigned char *buf = (unsigned char *)fmw->data; + + if (offset + 92 > fmw->size) { + dev_err(tas_priv->dev, "%s: File Size error\n", __func__); + offset = -EINVAL; + goto out; + } + if (memcmp(&buf[offset], magic_number, 4)) { + dev_err(tas_priv->dev, "%s: Magic num NOT match\n", __func__); + offset = -EINVAL; + goto out; + } + offset += 4; + + /* Convert data[offset], data[offset + 1], data[offset + 2] and + * data[offset + 3] into host + */ + fw_fixed_hdr->fwsize = be32_to_cpup((__be32 *)&buf[offset]); + offset += 4; + if (fw_fixed_hdr->fwsize != fmw->size) { + dev_err(tas_priv->dev, "File size not match, %lu %u", + (unsigned long)fmw->size, fw_fixed_hdr->fwsize); + offset = -EINVAL; + goto out; + } + offset += 4; + fw_fixed_hdr->ppcver = be32_to_cpup((__be32 *)&buf[offset]); + offset += 8; + fw_fixed_hdr->drv_ver = be32_to_cpup((__be32 *)&buf[offset]); + offset += 72; + + out: + return offset; +} + +static int fw_parse_variable_hdr_cal(struct tasdevice_priv *tas_priv, + struct tasdevice_fw *tas_fmw, const struct firmware *fmw, int offset) +{ + struct tasdevice_dspfw_hdr *fw_hdr = &(tas_fmw->fw_hdr); + + offset = fw_parse_variable_hdr(tas_priv, fw_hdr, fmw, offset); + if (offset < 0) + goto out; + if (fw_hdr->ndev != 1) { + dev_err(tas_priv->dev, + "%s: calbin must be 1, but currently ndev(%u)\n", + __func__, fw_hdr->ndev); + offset = -EINVAL; + } + +out: + return offset; +} + +/* When calibrated data parsing error occurs, DSP can still work with default + * calibrated data, memory resource related to calibrated data will be + * released in the tasdevice_codec_remove. + */ +static int fw_parse_calibration_data(struct tasdevice_priv *tas_priv, + struct tasdevice_fw *tas_fmw, const struct firmware *fmw, int offset) +{ + struct tasdevice_calibration *calibration; + unsigned char *data = (unsigned char *)fmw->data; + unsigned int i, n; + + if (offset + 2 > fmw->size) { + dev_err(tas_priv->dev, "%s: Calibrations error\n", __func__); + offset = -EINVAL; + goto out; + } + tas_fmw->nr_calibrations = be16_to_cpup((__be16 *)&data[offset]); + offset += 2; + + if (tas_fmw->nr_calibrations != 1) { + dev_err(tas_priv->dev, + "%s: only support one calibraiton(%d)!\n", + __func__, tas_fmw->nr_calibrations); + goto out; + } + + tas_fmw->calibrations = kcalloc(tas_fmw->nr_calibrations, + sizeof(struct tasdevice_calibration), GFP_KERNEL); + if (!tas_fmw->calibrations) { + offset = -ENOMEM; + goto out; + } + for (i = 0; i < tas_fmw->nr_calibrations; i++) { + if (offset + 64 > fmw->size) { + dev_err(tas_priv->dev, "Calibrations error\n"); + offset = -EINVAL; + goto out; + } + calibration = &(tas_fmw->calibrations[i]); + offset += 64; + + n = strlen((char *)&data[offset]); + /* skip '\0' and 2 unused bytes */ + n += 3; + if (offset + n > fmw->size) { + dev_err(tas_priv->dev, "Description err\n"); + offset = -EINVAL; + goto out; + } + offset += n; + + offset = fw_parse_data(tas_fmw, &(calibration->dev_data), fmw, + offset); + if (offset < 0) + goto out; + } + +out: + return offset; +} + +int tas2781_load_calibration(void *context, char *file_name, + enum channel i) +{ + struct tasdevice_priv *tas_priv = (struct tasdevice_priv *)context; + struct tasdevice *tasdev = &(tas_priv->tasdevice[i]); + const struct firmware *fw_entry; + struct tasdevice_fw *tas_fmw; + struct firmware fmw; + int offset = 0; + int ret; + + ret = request_firmware(&fw_entry, file_name, tas_priv->dev); + if (ret) { + dev_info(tas_priv->dev, "%s: Request firmware %s failed\n", + __func__, file_name); + goto out; + } + + if (!fw_entry->size) { + dev_err(tas_priv->dev, "%s: file read error: size = %lu\n", + __func__, (unsigned long)fw_entry->size); + goto out; + } + fmw.size = fw_entry->size; + fmw.data = fw_entry->data; + + tas_fmw = tasdev->cali_data_fmw = kzalloc(sizeof(struct tasdevice_fw), + GFP_KERNEL); + if (!tasdev->cali_data_fmw) { + ret = -ENOMEM; + goto out; + } + tas_fmw->dev = tas_priv->dev; + offset = fw_parse_header(tas_priv, tas_fmw, &fmw, offset); + if (offset == -EINVAL) { + dev_err(tas_priv->dev, "fw_parse_header EXIT!\n"); + ret = offset; + goto out; + } + offset = fw_parse_variable_hdr_cal(tas_priv, tas_fmw, &fmw, offset); + if (offset == -EINVAL) { + dev_err(tas_priv->dev, + "%s: fw_parse_variable_header_cal EXIT!\n", __func__); + ret = offset; + goto out; + } + offset = fw_parse_program_data(tas_priv, tas_fmw, &fmw, offset); + if (offset < 0) { + dev_err(tas_priv->dev, "fw_parse_program_data EXIT!\n"); + ret = offset; + goto out; + } + offset = fw_parse_configuration_data(tas_priv, tas_fmw, &fmw, offset); + if (offset < 0) { + dev_err(tas_priv->dev, "fw_parse_configuration_data EXIT!\n"); + ret = offset; + goto out; + } + offset = fw_parse_calibration_data(tas_priv, tas_fmw, &fmw, offset); + if (offset < 0) { + dev_err(tas_priv->dev, "fw_parse_calibration_data EXIT!\n"); + ret = offset; + goto out; + } + +out: + if (fw_entry) + release_firmware(fw_entry); + + return ret; +} +EXPORT_SYMBOL_NS_GPL(tas2781_load_calibration, SND_SOC_TAS2781_FMWLIB); + +static int tasdevice_dspfw_ready(const struct firmware *fmw, + void *context) +{ + struct tasdevice_priv *tas_priv = (struct tasdevice_priv *) context; + struct tasdevice_fw_fixed_hdr *fw_fixed_hdr; + struct tasdevice_fw *tas_fmw; + int offset = 0; + int ret = 0; + + if (!fmw || !fmw->data) { + dev_err(tas_priv->dev, "%s: Failed to read firmware %s\n", + __func__, tas_priv->coef_binaryname); + ret = -EINVAL; + goto out; + } + + tas_priv->fmw = kzalloc(sizeof(struct tasdevice_fw), GFP_KERNEL); + if (!tas_priv->fmw) { + ret = -ENOMEM; + goto out; + } + tas_fmw = tas_priv->fmw; + tas_fmw->dev = tas_priv->dev; + offset = fw_parse_header(tas_priv, tas_fmw, fmw, offset); + + if (offset == -EINVAL) { + ret = -EINVAL; + goto out; + } + fw_fixed_hdr = &(tas_fmw->fw_hdr.fixed_hdr); + /* Support different versions of firmware */ + switch (fw_fixed_hdr->drv_ver) { + case 0x301: + case 0x302: + case 0x502: + tas_priv->fw_parse_variable_header = + fw_parse_variable_header_kernel; + tas_priv->fw_parse_program_data = + fw_parse_program_data_kernel; + tas_priv->fw_parse_configuration_data = + fw_parse_configuration_data_kernel; + tas_priv->tasdevice_load_block = + tasdevice_load_block_kernel; + break; + case 0x202: + case 0x400: + tas_priv->fw_parse_variable_header = + fw_parse_variable_header_git; + tas_priv->fw_parse_program_data = + fw_parse_program_data; + tas_priv->fw_parse_configuration_data = + fw_parse_configuration_data; + tas_priv->tasdevice_load_block = + tasdevice_load_block; + break; + default: + ret = dspfw_default_callback(tas_priv, + fw_fixed_hdr->drv_ver, fw_fixed_hdr->ppcver); + if (ret) + goto out; + break; + } + + offset = tas_priv->fw_parse_variable_header(tas_priv, fmw, offset); + if (offset < 0) { + ret = offset; + goto out; + } + offset = tas_priv->fw_parse_program_data(tas_priv, tas_fmw, fmw, + offset); + if (offset < 0) { + ret = offset; + goto out; + } + offset = tas_priv->fw_parse_configuration_data(tas_priv, + tas_fmw, fmw, offset); + if (offset < 0) + ret = offset; + +out: + return ret; +} + +int tasdevice_dsp_parser(void *context) +{ + struct tasdevice_priv *tas_priv = (struct tasdevice_priv *)context; + const struct firmware *fw_entry; + int ret; + + ret = request_firmware(&fw_entry, tas_priv->coef_binaryname, + tas_priv->dev); + if (ret) { + dev_err(tas_priv->dev, "%s: load %s error\n", __func__, + tas_priv->coef_binaryname); + goto out; + } + + ret = tasdevice_dspfw_ready(fw_entry, tas_priv); + release_firmware(fw_entry); + fw_entry = NULL; + +out: + return ret; +} +EXPORT_SYMBOL_NS_GPL(tasdevice_dsp_parser, SND_SOC_TAS2781_FMWLIB); + +static void tas2781_clear_calfirmware(struct tasdevice_fw *tas_fmw) +{ + struct tasdevice_calibration *calibration; + struct tasdev_blk *block; + struct tasdevice_data *im; + unsigned int blks; + int i; + + if (!tas_fmw->calibrations) + goto out; + + for (i = 0; i < tas_fmw->nr_calibrations; i++) { + calibration = &(tas_fmw->calibrations[i]); + if (!calibration) + continue; + + im = &(calibration->dev_data); + + if (!im->dev_blks) + continue; + + for (blks = 0; blks < im->nr_blk; blks++) { + block = &(im->dev_blks[blks]); + if (!block) + continue; + kfree(block->data); + } + kfree(im->dev_blks); + } + kfree(tas_fmw->calibrations); +out: + kfree(tas_fmw); +} + +void tasdevice_calbin_remove(void *context) +{ + struct tasdevice_priv *tas_priv = (struct tasdevice_priv *) context; + struct tasdevice *tasdev; + int i; + + if (!tas_priv) + return; + + for (i = 0; i < tas_priv->ndev; i++) { + tasdev = &(tas_priv->tasdevice[i]); + if (!tasdev->cali_data_fmw) + continue; + tas2781_clear_calfirmware(tasdev->cali_data_fmw); + tasdev->cali_data_fmw = NULL; + } +} +EXPORT_SYMBOL_NS_GPL(tasdevice_calbin_remove, SND_SOC_TAS2781_FMWLIB); + +void tasdevice_config_info_remove(void *context) +{ + struct tasdevice_priv *tas_priv = (struct tasdevice_priv *) context; + struct tasdevice_rca *rca = &(tas_priv->rcabin); + struct tasdevice_config_info **ci = rca->cfg_info; + int i, j; + + if (!ci) + return; + for (i = 0; i < rca->ncfgs; i++) { + if (!ci[i]) + continue; + if (ci[i]->blk_data) { + for (j = 0; j < (int)ci[i]->real_nblocks; j++) { + if (!ci[i]->blk_data[j]) + continue; + kfree(ci[i]->blk_data[j]->regdata); + kfree(ci[i]->blk_data[j]); + } + kfree(ci[i]->blk_data); + } + kfree(ci[i]); + } + kfree(ci); +} +EXPORT_SYMBOL_NS_GPL(tasdevice_config_info_remove, SND_SOC_TAS2781_FMWLIB); + +static int tasdevice_load_data(struct tasdevice_priv *tas_priv, + struct tasdevice_data *dev_data) +{ + struct tasdev_blk *block; + unsigned int i; + int ret = 0; + + for (i = 0; i < dev_data->nr_blk; i++) { + block = &(dev_data->dev_blks[i]); + ret = tas_priv->tasdevice_load_block(tas_priv, block); + if (ret < 0) + break; + } + + return ret; +} + +int tasdevice_select_tuningprm_cfg(void *context, int prm_no, + int cfg_no, int rca_conf_no) +{ + struct tasdevice_priv *tas_priv = (struct tasdevice_priv *) context; + struct tasdevice_rca *rca = &(tas_priv->rcabin); + struct tasdevice_config_info **cfg_info = rca->cfg_info; + struct tasdevice_fw *tas_fmw = tas_priv->fmw; + struct tasdevice_prog *program; + struct tasdevice_config *conf; + int prog_status = 0; + int status, i; + + if (!tas_fmw) { + dev_err(tas_priv->dev, "%s: Firmware is NULL\n", __func__); + goto out; + } + + if (cfg_no >= tas_fmw->nr_configurations) { + dev_err(tas_priv->dev, + "%s: cfg(%d) is not in range of conf %u\n", + __func__, cfg_no, tas_fmw->nr_configurations); + goto out; + } + + if (prm_no >= tas_fmw->nr_programs) { + dev_err(tas_priv->dev, + "%s: prm(%d) is not in range of Programs %u\n", + __func__, prm_no, tas_fmw->nr_programs); + goto out; + } + + if (rca_conf_no > rca->ncfgs || rca_conf_no <= 0 || + !cfg_info) { + dev_err(tas_priv->dev, + "conf_no:%d should be in range from 0 to %u\n", + rca_conf_no, rca->ncfgs-1); + goto out; + } + + conf = &(tas_fmw->configs[cfg_no]); + for (i = 0, prog_status = 0; i < tas_priv->ndev; i++) { + if (cfg_info[rca_conf_no]->active_dev & (1 << i)) { + if (tas_priv->tasdevice[i].cur_prog != prm_no) { + tas_priv->tasdevice[i].cur_conf = -1; + tas_priv->tasdevice[i].is_loading = true; + prog_status++; + } + } else + tas_priv->tasdevice[i].is_loading = false; + tas_priv->tasdevice[i].is_loaderr = false; + } + + if (prog_status) { + program = &(tas_fmw->programs[prm_no]); + tasdevice_load_data(tas_priv, &(program->dev_data)); + for (i = 0; i < tas_priv->ndev; i++) { + if (tas_priv->tasdevice[i].is_loaderr == true) + continue; + else if (tas_priv->tasdevice[i].is_loaderr == false + && tas_priv->tasdevice[i].is_loading == true) { + struct tasdevice_fw *cal_fmw = + tas_priv->tasdevice[i].cali_data_fmw; + + if (cal_fmw) { + struct tasdevice_calibration + *cal = cal_fmw->calibrations; + + if (cal) + load_calib_data(tas_priv, + &(cal->dev_data)); + } + tas_priv->tasdevice[i].cur_prog = prm_no; + } + } + } + + for (i = 0, status = 0; i < tas_priv->ndev; i++) { + if (tas_priv->tasdevice[i].cur_conf != cfg_no + && (cfg_info[rca_conf_no]->active_dev & (1 << i)) + && (tas_priv->tasdevice[i].is_loaderr == false)) { + status++; + tas_priv->tasdevice[i].is_loading = true; + } else + tas_priv->tasdevice[i].is_loading = false; + } + + if (status) { + status = 0; + tasdevice_load_data(tas_priv, &(conf->dev_data)); + for (i = 0; i < tas_priv->ndev; i++) { + if (tas_priv->tasdevice[i].is_loaderr == true) { + status |= 1 << (i + 4); + continue; + } else if (tas_priv->tasdevice[i].is_loaderr == false + && tas_priv->tasdevice[i].is_loading == true) + tas_priv->tasdevice[i].cur_conf = cfg_no; + } + } else + dev_err(tas_priv->dev, + "%s: No device is in active in conf %d\n", + __func__, rca_conf_no); + + status |= cfg_info[rca_conf_no]->active_dev; + +out: + return prog_status; +} +EXPORT_SYMBOL_NS_GPL(tasdevice_select_tuningprm_cfg, + SND_SOC_TAS2781_FMWLIB); + +int tasdevice_prmg_load(void *context, int prm_no) +{ + struct tasdevice_priv *tas_priv = (struct tasdevice_priv *) context; + struct tasdevice_fw *tas_fmw = tas_priv->fmw; + struct tasdevice_prog *program; + int prog_status = 0; + int i; + + if (!tas_fmw) { + dev_err(tas_priv->dev, "%s: Firmware is NULL\n", __func__); + goto out; + } + + if (prm_no >= tas_fmw->nr_programs) { + dev_err(tas_priv->dev, + "%s: prm(%d) is not in range of Programs %u\n", + __func__, prm_no, tas_fmw->nr_programs); + goto out; + } + + for (i = 0, prog_status = 0; i < tas_priv->ndev; i++) { + if (tas_priv->tasdevice[i].cur_prog != prm_no) { + tas_priv->tasdevice[i].cur_conf = -1; + tas_priv->tasdevice[i].is_loading = true; + prog_status++; + } + } + + if (prog_status) { + program = &(tas_fmw->programs[prm_no]); + tasdevice_load_data(tas_priv, &(program->dev_data)); + for (i = 0; i < tas_priv->ndev; i++) { + if (tas_priv->tasdevice[i].is_loaderr == true) + continue; + else if (tas_priv->tasdevice[i].is_loaderr == false + && tas_priv->tasdevice[i].is_loading == true) + tas_priv->tasdevice[i].cur_prog = prm_no; + } + } + if (tas_priv->set_global_mode) + tas_priv->set_global_mode(tas_priv); +out: + return prog_status; +} +EXPORT_SYMBOL_NS_GPL(tasdevice_prmg_load, SND_SOC_TAS2781_FMWLIB); + +int tasdevice_prmg_calibdata_load(void *context, int prm_no) +{ + struct tasdevice_priv *tas_priv = (struct tasdevice_priv *) context; + struct tasdevice_fw *tas_fmw = tas_priv->fmw; + struct tasdevice_prog *program; + int prog_status = 0; + int i; + + if (!tas_fmw) { + dev_err(tas_priv->dev, "%s: Firmware is NULL\n", __func__); + goto out; + } + + if (prm_no >= tas_fmw->nr_programs) { + dev_err(tas_priv->dev, + "%s: prm(%d) is not in range of Programs %u\n", + __func__, prm_no, tas_fmw->nr_programs); + goto out; + } + + for (i = 0, prog_status = 0; i < tas_priv->ndev; i++) { + if (tas_priv->tasdevice[i].cur_prog != prm_no) { + tas_priv->tasdevice[i].cur_conf = -1; + tas_priv->tasdevice[i].is_loading = true; + prog_status++; + } + tas_priv->tasdevice[i].is_loaderr = false; + } + + if (prog_status) { + program = &(tas_fmw->programs[prm_no]); + tasdevice_load_data(tas_priv, &(program->dev_data)); + for (i = 0; i < tas_priv->ndev; i++) { + if (tas_priv->tasdevice[i].is_loaderr == true) + continue; + else if (tas_priv->tasdevice[i].is_loaderr == false + && tas_priv->tasdevice[i].is_loading == true) { + struct tasdevice_fw *cal_fmw = + tas_priv->tasdevice[i].cali_data_fmw; + + if (cal_fmw) { + struct tasdevice_calibration *cal = + cal_fmw->calibrations; + + if (cal) + load_calib_data(tas_priv, + &(cal->dev_data)); + } + tas_priv->tasdevice[i].cur_prog = prm_no; + } + } + } + +out: + return prog_status; +} +EXPORT_SYMBOL_NS_GPL(tasdevice_prmg_calibdata_load, + SND_SOC_TAS2781_FMWLIB); + +void tasdevice_tuning_switch(void *context, int state) +{ + struct tasdevice_priv *tas_priv = (struct tasdevice_priv *) context; + struct tasdevice_fw *tas_fmw = tas_priv->fmw; + int profile_cfg_id = 0; + + if (tas_priv->fw_state == TASDEVICE_DSP_FW_FAIL) { + dev_err(tas_priv->dev, "DSP bin file not loaded\n"); + return; + } + + if (state == 0) { + if (tas_priv->cur_prog >= tas_fmw->nr_programs) + /*bypass all in rca is profile id 0*/ + profile_cfg_id = RCA_CONFIGID_BYPASS_ALL; + else { + /*dsp mode or tuning mode*/ + profile_cfg_id = tas_priv->rcabin.profile_cfg_id; + tasdevice_select_tuningprm_cfg(tas_priv, + tas_priv->cur_prog, tas_priv->cur_conf, + profile_cfg_id); + if (tas_priv->set_global_mode) + tas_priv->set_global_mode(tas_priv); + } + + tasdevice_select_cfg_blk(tas_priv, profile_cfg_id, + TASDEVICE_BIN_BLK_PRE_POWER_UP); + } else + tasdevice_select_cfg_blk(tas_priv, + tas_priv->rcabin.profile_cfg_id, + TASDEVICE_BIN_BLK_PRE_SHUTDOWN); +} +EXPORT_SYMBOL_NS_GPL(tasdevice_tuning_switch, + SND_SOC_TAS2781_FMWLIB); + +MODULE_DESCRIPTION("Texas Firmware Support"); +MODULE_AUTHOR("Shenghao Ding, TI, <shenghao-ding(a)ti.com>"); +MODULE_LICENSE("GPL"); -- 2.34.1

1 0

[PATCH v3 1/5] ASoC: tas2781: Add Header file for tas2781 driver
by Shenghao Ding 19 May '23

19 May '23

Create Header file for tas2781 driver. Signed-off-by: Shenghao Ding <13916275206(a)139.com> --- Changes in v3: - correct some enums have capitalized Chn, chn to all Changes to be committed: new file: include/sound/tas2781-dsp.h new file: include/sound/tas2781-tlv.h new file: include/sound/tas2781.h --- include/sound/tas2781-dsp.h | 190 ++++++++++++++++++++++++++++++++++++ include/sound/tas2781-tlv.h | 22 +++++ include/sound/tas2781.h | 176 +++++++++++++++++++++++++++++++++ 3 files changed, 388 insertions(+) create mode 100644 include/sound/tas2781-dsp.h create mode 100644 include/sound/tas2781-tlv.h create mode 100644 include/sound/tas2781.h diff --git a/include/sound/tas2781-dsp.h b/include/sound/tas2781-dsp.h new file mode 100644 index 000000000000..b05f20bc4a1f --- /dev/null +++ b/include/sound/tas2781-dsp.h @@ -0,0 +1,190 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +// +// ALSA SoC Texas Instruments TAS2781 Audio Smart Amplifier +// +// Copyright (C) 2022 - 2023 Texas Instruments Incorporated +// https://www.ti.com +// +// The TAS2781 driver implements a flexible and configurable +// algo coefficient setting for one, two, or even multiple +// TAS2781 chips. +// +// Author: Shenghao Ding <shenghao-ding(a)ti.com> +// Author: Kevin Lu <kevin-lu(a)ti.com> +// + +#ifndef __TASDEVICE_DSP_H__ +#define __TASDEVICE_DSP_H__ + +#define MAIN_ALL_DEVICES 0x0d +#define MAIN_DEVICE_A 0x01 +#define MAIN_DEVICE_B 0x08 +#define MAIN_DEVICE_C 0x10 +#define MAIN_DEVICE_D 0x14 +#define COEFF_DEVICE_A 0x03 +#define COEFF_DEVICE_B 0x0a +#define COEFF_DEVICE_C 0x11 +#define COEFF_DEVICE_D 0x15 +#define PRE_DEVICE_A 0x04 +#define PRE_DEVICE_B 0x0b +#define PRE_DEVICE_C 0x12 +#define PRE_DEVICE_D 0x16 + +#define PPC3_VERSION 0x4100 +#define PPC3_VERSION_TAS2781 0x14c00 +#define RCA_CONFIGID_BYPASS_ALL 0 +#define TASDEVICE_DEVICE_SUM 8 +#define TASDEVICE_CONFIG_SUM 64 + +enum channel { + top_left_Chn, + top_right_chn, + bottom_left_Chn, + bottom_right_chn, + max_chn, +}; + +enum tasdevice_dsp_dev_idx { + TASDEVICE_DSP_TAS_2555 = 0, + TASDEVICE_DSP_TAS_2555_STEREO, + TASDEVICE_DSP_TAS_2557_MONO, + TASDEVICE_DSP_TAS_2557_DUAL_MONO, + TASDEVICE_DSP_TAS_2559, + TASDEVICE_DSP_TAS_2563, + TASDEVICE_DSP_TAS_2563_DUAL_MONO = 7, + TASDEVICE_DSP_TAS_2563_QUAD, + TASDEVICE_DSP_TAS_2563_21, + TASDEVICE_DSP_TAS_2781, + TASDEVICE_DSP_TAS_2781_DUAL_MONO, + TASDEVICE_DSP_TAS_2781_21, + TASDEVICE_DSP_TAS_2781_QUAD, + TASDEVICE_DSP_TAS_MAX_DEVICE +}; + +struct tasdevice_fw_fixed_hdr { + unsigned int fwsize; + unsigned int ppcver; + unsigned int drv_ver; +}; + +struct tasdevice_dspfw_hdr { + struct tasdevice_fw_fixed_hdr fixed_hdr; + unsigned short device_family; + unsigned short device; + unsigned char ndev; +}; + +struct tasdev_blk { + int nr_retry; + unsigned int type; + unsigned char is_pchksum_present; + unsigned char pchksum; + unsigned char is_ychksum_present; + unsigned char ychksum; + unsigned int nr_cmds; + unsigned int blk_size; + unsigned int nr_subblocks; + unsigned char *data; +}; + +struct tasdevice_data { + char name[64]; + unsigned int nr_blk; + struct tasdev_blk *dev_blks; +}; + +struct tasdevice_prog { + unsigned int prog_size; + struct tasdevice_data dev_data; +}; + +struct tasdevice_config { + unsigned int cfg_size; + char name[64]; + struct tasdevice_data dev_data; +}; + +struct tasdevice_calibration { + struct tasdevice_data dev_data; +}; + +struct tasdevice_fw { + struct tasdevice_dspfw_hdr fw_hdr; + unsigned short nr_programs; + struct tasdevice_prog *programs; + unsigned short nr_configurations; + struct tasdevice_config *configs; + unsigned short nr_calibrations; + struct tasdevice_calibration *calibrations; + struct device *dev; +}; + +enum tasdevice_dsp_fw_state { + TASDEVICE_DSP_FW_NONE = 0, + TASDEVICE_DSP_FW_PENDING, + TASDEVICE_DSP_FW_FAIL, + TASDEVICE_DSP_FW_ALL_OK, +}; + +enum tasdevice_bin_blk_type { + TASDEVICE_BIN_BLK_COEFF = 1, + TASDEVICE_BIN_BLK_POST_POWER_UP, + TASDEVICE_BIN_BLK_PRE_SHUTDOWN, + TASDEVICE_BIN_BLK_PRE_POWER_UP, + TASDEVICE_BIN_BLK_POST_SHUTDOWN +}; + +struct tasdevice_rca_hdr { + unsigned int img_sz; + unsigned int checksum; + unsigned int binary_version_num; + unsigned int drv_fw_version; + unsigned char plat_type; + unsigned char dev_family; + unsigned char reserve; + unsigned char ndev; + unsigned char devs[TASDEVICE_DEVICE_SUM]; + unsigned int nconfig; + unsigned int config_size[TASDEVICE_CONFIG_SUM]; +}; + +struct tasdev_blk_data { + unsigned char dev_idx; + unsigned char block_type; + unsigned short yram_checksum; + unsigned int block_size; + unsigned int n_subblks; + unsigned char *regdata; +}; + +struct tasdevice_config_info { + unsigned int nblocks; + unsigned int real_nblocks; + unsigned char active_dev; + struct tasdev_blk_data **blk_data; +}; + +struct tasdevice_rca { + struct tasdevice_rca_hdr fw_hdr; + int ncfgs; + struct tasdevice_config_info **cfg_info; + int profile_cfg_id; +}; + +void tasdevice_select_cfg_blk(void *context, int conf_no, + unsigned char block_type); +void tasdevice_config_info_remove(void *context); +void tasdevice_dsp_remove(void *context); +int tasdevice_dsp_parser(void *context); +int tasdevice_rca_parser(void *context, const struct firmware *fmw); +void tasdevice_dsp_remove(void *context); +void tasdevice_calbin_remove(void *context); +int tasdevice_select_tuningprm_cfg(void *context, int prm, + int cfg_no, int rca_conf_no); +int tasdevice_prmg_load(void *context, int prm_no); +int tasdevice_prmg_calibdata_load(void *context, int prm_no); +void tasdevice_tuning_switch(void *context, int state); +int tas2781_load_calibration(void *context, char *file_name, + enum channel i); + +#endif diff --git a/include/sound/tas2781-tlv.h b/include/sound/tas2781-tlv.h new file mode 100644 index 000000000000..f4310dce655a --- /dev/null +++ b/include/sound/tas2781-tlv.h @@ -0,0 +1,22 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +// +// ALSA SoC Texas Instruments TAS2781 Audio Smart Amplifier +// +// Copyright (C) 2022 - 2023 Texas Instruments Incorporated +// https://www.ti.com +// +// The TAS2781 driver implements a flexible and configurable +// algo coefficient setting for one, two, or even multiple +// TAS2781 chips. +// +// Author: Shenghao Ding <shenghao-ding(a)ti.com> +// Author: Kevin Lu <kevin-lu(a)ti.com> +// + +#ifndef __TAS2781_TLV_H__ +#define __TAS2781_TLV_H__ + +static const DECLARE_TLV_DB_SCALE(dvc_tlv, -10000, 100, 0); +static const DECLARE_TLV_DB_SCALE(amp_vol_tlv, 1100, 50, 0); + +#endif /* __TAS2781_LIB_H__ */ diff --git a/include/sound/tas2781.h b/include/sound/tas2781.h new file mode 100644 index 000000000000..c98955bda07e --- /dev/null +++ b/include/sound/tas2781.h @@ -0,0 +1,176 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +// +// ALSA SoC Texas Instruments TAS2781 Audio Smart Amplifier +// +// Copyright (C) 2022 - 2023 Texas Instruments Incorporated +// https://www.ti.com +// +// The TAS2781 driver implements a flexible and configurable +// algo coefficient setting for one, two, or even multiple +// TAS2781 chips. +// +// Author: Shenghao Ding <shenghao-ding(a)ti.com> +// Author: Kevin Lu <kevin-lu(a)ti.com> +// + +#ifndef __TAS2781_H__ +#define __TAS2781_H__ + +#include <linux/kernel.h> +#include "tas2781-dsp.h" + +/* version number */ +#define TAS2781_DRV_VER 1 +#define SMARTAMP_MODULE_NAME "tas2781" +#define TAS2781_GLOBAL_ADDR 0x40 +#define TASDEVICE_RATES (SNDRV_PCM_RATE_44100 |\ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000 |\ + SNDRV_PCM_RATE_88200) +#define TASDEVICE_MAX_CHANNELS 8 + +#define TASDEVICE_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +/*PAGE Control Register (available in page0 of each book) */ +#define TASDEVICE_PAGE_SELECT 0x00 +#define TASDEVICE_BOOKCTL_PAGE 0x00 +#define TASDEVICE_BOOKCTL_REG 127 +#define TASDEVICE_BOOK_ID(reg) (reg / (256 * 128)) +#define TASDEVICE_PAGE_ID(reg) ((reg % (256 * 128)) / 128) +#define TASDEVICE_PAGE_REG(reg) ((reg % (256 * 128)) % 128) +#define TASDEVICE_PGRG(reg) (reg % (256 * 128)) +#define TASDEVICE_REG(book, page, reg) (((book * 256 * 128) + \ + (page * 128)) + reg) + +/*Software Reset */ +#define TAS2781_REG_SWRESET TASDEVICE_REG(0x0, 0X0, 0x02) +#define TAS2781_REG_SWRESET_RESET BIT(0) + +/* Enable Global addresses */ +#define TAS2781_MISC_CFG2 TASDEVICE_REG(0x0, 0X0, 0x07) +#define TAS2781_GLOBAL_ADDR_MASK BIT(1) +#define TAS2781_GLOBAL_ADDR_ENABLE BIT(1) + +/*I2C Checksum */ +#define TASDEVICE_I2CChecksum TASDEVICE_REG(0x0, 0x0, 0x7E) + +/* Volume control */ +#define TAS2781_DVC_LVL TASDEVICE_REG(0x0, 0x0, 0x1A) +#define TAS2781_AMP_LEVEL TASDEVICE_REG(0x0, 0x0, 0x03) +#define TAS2781_AMP_LEVEL_MASK GENMASK(5, 1) + +#define TASDEVICE_CMD_SING_W 0x1 +#define TASDEVICE_CMD_BURST 0x2 +#define TASDEVICE_CMD_DELAY 0x3 +#define TASDEVICE_CMD_FIELD_W 0x4 + +enum audio_device { + TAS2781 = 0, +}; + +struct tasdevice { + struct tasdevice_fw *cali_data_fmw; + unsigned int dev_addr; + unsigned int err_code; + unsigned char cur_book; + short cur_prog; + short cur_conf; + bool is_loading; + bool is_loaderr; +}; + +/* + * 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 { + int ref_cnt; + unsigned int dev_addr; + unsigned char cur_book; +}; + +struct tasdevice_irqinfo { + int irq_gpio; + int irq; +}; + +struct calidata { + unsigned char *data; + unsigned long total_sz; +}; + +struct tasdevice_priv { + struct tasdevice tasdevice[max_chn]; + struct tasdevice_irqinfo irq_info; + struct global_addr glb_addr; + struct tasdevice_rca rcabin; + struct tasdevice_fw *fmw; + struct regmap *regmap; + struct mutex codec_lock; + struct calidata cali_data; + struct gpio_desc *reset; + struct device *dev; + struct tm tm; + + unsigned char crc8_lkp_tbl[CRC8_TABLE_SIZE]; + unsigned char cal_binaryname[max_chn][64]; + unsigned char coef_binaryname[64]; + unsigned char rca_binaryname[64]; + unsigned char dev_name[32]; + unsigned char ndev; + unsigned int magic_num; + unsigned int chip_id; + unsigned int sysclk; + const char *acpi_subsystem_id; + int index; + int cur_prog; + int cur_conf; + int fw_state; + void *client; + void *codec; + bool isacpi; + bool playback_started; + void (*set_global_mode)(struct tasdevice_priv *tas_priv); + int (*fw_parse_variable_header)(struct tasdevice_priv *tas_priv, + const struct firmware *fmw, int offset); + int (*fw_parse_program_data)(struct tasdevice_priv *tas_priv, + struct tasdevice_fw *tas_fmw, + const struct firmware *fmw, int offset); + int (*fw_parse_configuration_data)(struct tasdevice_priv *tas_priv, + struct tasdevice_fw *tas_fmw, + const struct firmware *fmw, int offset); + int (*tasdevice_load_block)(struct tasdevice_priv *tas_priv, + struct tasdev_blk *block); +}; + +void tas2781_reset(struct tasdevice_priv *tas_dev); +int tascodec_init(struct tasdevice_priv *tas_priv, void *codec, + void (*cont)(const struct firmware *fw, void *context)); +struct tasdevice_priv *tasdevice_kzalloc(struct i2c_client *i2c); +int tasdevice_init(struct tasdevice_priv *tas_priv); +void tasdevice_remove(struct tasdevice_priv *tas_priv); +int tasdevice_dev_read(struct tasdevice_priv *tas_priv, + enum channel chn, unsigned int reg, unsigned int *value); +int tasdevice_dev_write(struct tasdevice_priv *tas_priv, + enum channel chn, unsigned int reg, unsigned int value); +int tasdevice_dev_bulk_write( + struct tasdevice_priv *tas_priv, enum channel chn, + unsigned int reg, unsigned char *p_data, unsigned int n_length); +int tasdevice_dev_bulk_read(struct tasdevice_priv *tas_priv, + enum channel chn, unsigned int reg, unsigned char *p_data, + unsigned int n_length); +int tasdevice_dev_update_bits( + struct tasdevice_priv *tasdevice, enum channel chn, + unsigned int reg, unsigned int mask, unsigned int value); +int tasdevice_amp_putvol(struct tasdevice_priv *tas_priv, + struct snd_ctl_elem_value *ucontrol, struct soc_mixer_control *mc); +int tasdevice_amp_getvol(struct tasdevice_priv *tas_priv, + struct snd_ctl_elem_value *ucontrol, struct soc_mixer_control *mc); +int tasdevice_digital_putvol(struct tasdevice_priv *tas_priv, + struct snd_ctl_elem_value *ucontrol, struct soc_mixer_control *mc); +int tasdevice_digital_getvol(struct tasdevice_priv *tas_priv, + struct snd_ctl_elem_value *ucontrol, struct soc_mixer_control *mc); + +#endif /* __TAS2781_H__ */ -- 2.34.1

1 0

[PATCH 0/3] ASoC: cs35l56: Bugfixes and efficiency improvement
by Richard Fitzgerald 19 May '23

19 May '23

First two patches are bugfixes. Third patch skips the overhead of rebooting the amp after applying firmware files when we know that it isn't necessary. Simon Trimmer (3): ASoC: cs35l56: Move DSP part string generation so that it is done only once ASoC: cs35l56: sdw_write_no_pm() should be performed under a pm_runtime request ASoC: cs35l56: In secure mode skip SHUTDOWN and RESET around fw download include/sound/cs35l56.h | 1 + sound/soc/codecs/cs35l56.c | 65 +++++++++++++++++++++++++++----------- 2 files changed, 47 insertions(+), 19 deletions(-) -- 2.30.2

2 4

[PATCH AUTOSEL 6.3 01/24] ASoC: jack: allow multiple interrupt per gpio
by Sasha Levin 19 May '23

19 May '23

From: Svyatoslav Ryhel <clamor95(a)gmail.com> [ Upstream commit a2d4051b0bd6dffcd736888ae89a550d6f60b060 ] This feature is required for coupled hp-mic quirk used by some Nvidia Tegra 3 based devices work properly. Signed-off-by: Svyatoslav Ryhel <clamor95(a)gmail.com> Link: https://lore.kernel.org/r/20230221183211.21964-3-clamor95@gmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/soc-jack.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index fcece5ca38c69..f951acb2ce362 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -367,6 +367,7 @@ int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count, ret = request_any_context_irq(gpiod_to_irq(gpios[i].desc), gpio_handler, + IRQF_SHARED | IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING, gpios[i].name, -- 2.39.2

2 10

[PATCH] ASoC: soc-pcm: test if a BE can be prepared
by Pierre-Louis Bossart 19 May '23

19 May '23

From: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> In the BE hw_params configuration, the existing code checks if any of the existing FEs are prepared, running, paused or suspended - and skips the configuration in those cases. This allows multiple calls of hw_params which the ALSA state machine supports. This check is not handled for the prepare stage, which can lead to the same BE being prepared multiple times. This patch adds a check similar to that of the hw_params, with the main difference being that the suspended state is allowed: the ALSA state machine allows a transition from suspended to prepared with hw_params skipped. This problem was detected on Intel IPC4/SoundWire devices, where the BE dailink .prepare stage is used to configure the SoundWire stream with a bank switch. Multiple .prepare calls lead to conflicts with the .trigger operation with IPC4 configurations. This problem was not detected earlier on Intel devices, HDaudio BE dailinks detect that the link is already prepared and skip the configuration, and for IPC3 devices there is no BE trigger. Link: https://github.com/thesofproject/sof/issues/7596 Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- include/sound/soc-dpcm.h | 4 ++++ sound/soc/soc-pcm.c | 20 ++++++++++++++++++++ 2 files changed, 24 insertions(+) diff --git a/include/sound/soc-dpcm.h b/include/sound/soc-dpcm.h index 4d6ac7699833..ebd24753dd00 100644 --- a/include/sound/soc-dpcm.h +++ b/include/sound/soc-dpcm.h @@ -122,6 +122,10 @@ int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe, struct snd_soc_pcm_runtime *be, int stream); +/* can this BE perform prepare */ +int snd_soc_dpcm_can_be_prepared(struct snd_soc_pcm_runtime *fe, + struct snd_soc_pcm_runtime *be, int stream); + /* is the current PCM operation for this FE ? */ int snd_soc_dpcm_fe_can_update(struct snd_soc_pcm_runtime *fe, int stream); diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index adb69d7820a8..4fb1ac8e1c4a 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2405,6 +2405,9 @@ int dpcm_be_dai_prepare(struct snd_soc_pcm_runtime *fe, int stream) if (!snd_soc_dpcm_be_can_update(fe, be, stream)) continue; + if (!snd_soc_dpcm_can_be_prepared(fe, be, stream)) + continue; + if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_PARAMS) && (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP) && (be->dpcm[stream].state != SND_SOC_DPCM_STATE_SUSPEND) && @@ -3042,3 +3045,20 @@ int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe, return snd_soc_dpcm_check_state(fe, be, stream, state, ARRAY_SIZE(state)); } EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_params); + +/* + * We can only prepare a BE DAI if any of it's FE are not prepared, + * running or paused for the specified stream direction. + */ +int snd_soc_dpcm_can_be_prepared(struct snd_soc_pcm_runtime *fe, + struct snd_soc_pcm_runtime *be, int stream) +{ + const enum snd_soc_dpcm_state state[] = { + SND_SOC_DPCM_STATE_START, + SND_SOC_DPCM_STATE_PAUSED, + SND_SOC_DPCM_STATE_PREPARE, + }; + + return snd_soc_dpcm_check_state(fe, be, stream, state, ARRAY_SIZE(state)); +} +EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_prepared); -- 2.37.2

3 2

[PATCH v2 0/7] Add Chameleon v3 ASoC audio
by Paweł Anikiel 19 May '23

19 May '23

The Google Chameleon v3 is a device made for testing audio and video paths of other devices. This patchset adds support for ASoC audio on this device. It has two audio sources: HDMI audio from the it68051 chip (RX only), and analog audio from the ssm2603 chip (RX and TX). The patchset adds the ASoC platform and codec drivers, as well as some changes to the existing ssm2602 codec driver. v2 changes: * Replace chv3-audio machine driver with simple-audio-card * Use regmap_register_patch() in ssm2602 workaround * Remove ssm2602 mute gpio patch, add a gpio hog instead * Simplify hw_pointer logic in chv3-i2s * Commit splits & other misc changes Paweł Anikiel (7): ASoC: Add Google Chameleon v3 i2s driver ASoC: Add Google Chameleon v3 codec driver ASoC: dt-bindings: Add Google Chameleon v3 i2s device ASoC: dt-bindings: Add Google Chameleon v3 audio codec ARM: dts: chameleonv3: Add mute gpio hog ARM: dts: chameleonv3: Add Chameleon v3 audio ASoC: ssm2602: Add workaround for playback distortions .../bindings/sound/google,chv3-codec.yaml | 31 ++ .../bindings/sound/google,chv3-i2s.yaml | 44 +++ .../boot/dts/socfpga_arria10_chameleonv3.dts | 62 ++++ sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/codecs/Kconfig | 8 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/chv3-codec.c | 41 +++ sound/soc/codecs/ssm2602.c | 15 + sound/soc/google/Kconfig | 6 + sound/soc/google/Makefile | 2 + sound/soc/google/chv3-i2s.c | 338 ++++++++++++++++++ 12 files changed, 551 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/google,chv3-codec.yaml create mode 100644 Documentation/devicetree/bindings/sound/google,chv3-i2s.yaml create mode 100644 sound/soc/codecs/chv3-codec.c create mode 100644 sound/soc/google/Kconfig create mode 100644 sound/soc/google/Makefile create mode 100644 sound/soc/google/chv3-i2s.c -- 2.40.1.521.gf1e218fcd8-goog

4 16

[PATCH] SoC: rt5682: Disable jack detection interrupt during suspend
by Matthias Kaehlcke 18 May '23

18 May '23

The rt5682 driver switches its regmap to cache-only when the device suspends and back to regular mode on resume. When the jack detect interrupt fires rt5682_irq() schedules the jack detect work. This can result in invalid reads from the regmap in cache-only mode if the work runs before the device has resumed: [ 56.245502] rt5682 9-001a: ASoC: error at soc_component_read_no_lock on rt5682.9-001a for register: [0x000000f0] -16 Disable the jack detection interrupt during suspend and re-enable it on resume. The driver already schedules the jack detection work on resume, so any state change during suspend is still handled. This is essentially the same as commit f7d00a9be147 ("SoC: rt5682s: Disable jack detection interrupt during suspend") for the rt5682s. Cc: stable(a)kernel.org Signed-off-by: Matthias Kaehlcke <mka(a)chromium.org> --- sound/soc/codecs/rt5682-i2c.c | 4 +++- sound/soc/codecs/rt5682.c | 6 ++++++ sound/soc/codecs/rt5682.h | 1 + 3 files changed, 10 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5682-i2c.c b/sound/soc/codecs/rt5682-i2c.c index 2935c1bb81f3..5bc46b041786 100644 --- a/sound/soc/codecs/rt5682-i2c.c +++ b/sound/soc/codecs/rt5682-i2c.c @@ -267,7 +267,9 @@ static int rt5682_i2c_probe(struct i2c_client *i2c) ret = devm_request_threaded_irq(&i2c->dev, i2c->irq, NULL, rt5682_irq, IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING | IRQF_ONESHOT, "rt5682", rt5682); - if (ret) + if (!ret) + rt5682->irq = i2c->irq; + else dev_err(&i2c->dev, "Failed to reguest IRQ: %d\n", ret); } diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index f6c798b65c08..5d992543b791 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -2959,6 +2959,9 @@ static int rt5682_suspend(struct snd_soc_component *component) if (rt5682->is_sdw) return 0; + if (rt5682->irq) + disable_irq(rt5682->irq); + cancel_delayed_work_sync(&rt5682->jack_detect_work); cancel_delayed_work_sync(&rt5682->jd_check_work); if (rt5682->hs_jack && (rt5682->jack_type & SND_JACK_HEADSET) == SND_JACK_HEADSET) { @@ -3027,6 +3030,9 @@ static int rt5682_resume(struct snd_soc_component *component) mod_delayed_work(system_power_efficient_wq, &rt5682->jack_detect_work, msecs_to_jiffies(0)); + if (rt5682->irq) + enable_irq(rt5682->irq); + return 0; } #else diff --git a/sound/soc/codecs/rt5682.h b/sound/soc/codecs/rt5682.h index d568c6993c33..e8efd8a84a6c 100644 --- a/sound/soc/codecs/rt5682.h +++ b/sound/soc/codecs/rt5682.h @@ -1462,6 +1462,7 @@ struct rt5682_priv { int pll_out[RT5682_PLLS]; int jack_type; + int irq; int irq_work_delay_time; }; -- 2.40.1.606.ga4b1b128d6-goog

4 4

[PATCH v2] ASoC:codecs: lpass: Fix for KASAN use_after_free out of bounds
by Ravulapati Vishnu Vardhan Rao 18 May '23

18 May '23

When we run syzkaller we get below Out of Bounds error. "KASAN: slab-out-of-bounds Read in regcache_flat_read" Below is the backtrace of the issue: BUG: KASAN: slab-out-of-bounds in regcache_flat_read+0x10c/0x110 Read of size 4 at addr ffffff8088fbf714 by task syz-executor.4/14144 CPU: 6 PID: 14144 Comm: syz-executor.4 Tainted: G W Hardware name: Qualcomm Technologies, Inc. sc7280 CRD platform (rev5+) (DT) Call trace: dump_backtrace+0x0/0x4ec show_stack+0x34/0x50 dump_stack_lvl+0xdc/0x11c print_address_description+0x30/0x2d8 kasan_report+0x178/0x1e4 __asan_report_load4_noabort+0x44/0x50 regcache_flat_read+0x10c/0x110 regcache_read+0xf8/0x5a0 _regmap_read+0x45c/0x86c _regmap_update_bits+0x128/0x290 regmap_update_bits_base+0xc0/0x15c snd_soc_component_update_bits+0xa8/0x22c snd_soc_component_write_field+0x68/0xd4 tx_macro_put_dec_enum+0x1d0/0x268 snd_ctl_elem_write+0x288/0x474 By Error checking and checking valid values issue gets rectifies. Signed-off-by: Ravulapati Vishnu Vardhan Rao <quic_visr(a)quicinc.com> --- sound/soc/codecs/lpass-tx-macro.c | 5 +++++ 1 file changed, 5 insertions(+) Changes Since v1: - Check value with enum item value getting from control. - Removing of previously added code which is unwanted as already we get out with -EINVAL if value is out of bound. diff --git a/sound/soc/codecs/lpass-tx-macro.c b/sound/soc/codecs/lpass-tx-macro.c index da6fcf7f0991..de978c3d70b7 100644 --- a/sound/soc/codecs/lpass-tx-macro.c +++ b/sound/soc/codecs/lpass-tx-macro.c @@ -746,6 +746,8 @@ static int tx_macro_put_dec_enum(struct snd_kcontrol *kcontrol, struct tx_macro *tx = snd_soc_component_get_drvdata(component); val = ucontrol->value.enumerated.item[0]; + if (val >= e->items) + return -EINVAL; switch (e->reg) { case CDC_TX_INP_MUX_ADC_MUX0_CFG0: @@ -772,6 +774,9 @@ static int tx_macro_put_dec_enum(struct snd_kcontrol *kcontrol, case CDC_TX_INP_MUX_ADC_MUX7_CFG0: mic_sel_reg = CDC_TX7_TX_PATH_CFG0; break; + default: + dev_err(component->dev, "Error in configuration!!\n"); + return -EINVAL; } if (val != 0) { -- 2.17.1

2 1

[PATCH] ASoC: mediatek: mt8192-mt6359: Remove " Jack" from Headphone pin name
by AngeloGioacchino Del Regno 18 May '23

18 May '23

Function jack_kctl_name_gen() will remove the redundant " Jack" from the name, if present, and then it will add it back, so that all of the controls are named "(pin-name) Jack". Remove " Jack" from the Headphone pin name to spare some CPU cycles. This commit brings no functional changes. Signed-off-by: AngeloGioacchino Del Regno <angelogioacchino.delregno(a)collabora.com> --- sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c b/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c index 5e163e23a207..4e0d5bf12b47 100644 --- a/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c +++ b/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c @@ -46,7 +46,7 @@ struct mt8192_mt6359_priv { /* Headset jack detection DAPM pins */ static struct snd_soc_jack_pin mt8192_jack_pins[] = { { - .pin = "Headphone Jack", + .pin = "Headphone", .mask = SND_JACK_HEADPHONE, }, { -- 2.40.1

3 2

[PATCH 01/12] ASoC: codecs: rt1308: do not store status in state container
by Krzysztof Kozlowski 18 May '23

18 May '23

Driver in its update status callback stores Soundwire device status in state container but it never uses it later. Simplify the code a bit. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- sound/soc/codecs/rt1308-sdw.c | 5 +---- sound/soc/codecs/rt1308-sdw.h | 1 - 2 files changed, 1 insertion(+), 5 deletions(-) diff --git a/sound/soc/codecs/rt1308-sdw.c b/sound/soc/codecs/rt1308-sdw.c index 1797af824f60..313e97c94532 100644 --- a/sound/soc/codecs/rt1308-sdw.c +++ b/sound/soc/codecs/rt1308-sdw.c @@ -304,9 +304,6 @@ static int rt1308_update_status(struct sdw_slave *slave, { struct rt1308_sdw_priv *rt1308 = dev_get_drvdata(&slave->dev); - /* Update the status */ - rt1308->status = status; - if (status == SDW_SLAVE_UNATTACHED) rt1308->hw_init = false; @@ -314,7 +311,7 @@ static int rt1308_update_status(struct sdw_slave *slave, * Perform initialization only if slave status is present and * hw_init flag is false */ - if (rt1308->hw_init || rt1308->status != SDW_SLAVE_ATTACHED) + if (rt1308->hw_init || status != SDW_SLAVE_ATTACHED) return 0; /* perform I/O transfers required for Slave initialization */ diff --git a/sound/soc/codecs/rt1308-sdw.h b/sound/soc/codecs/rt1308-sdw.h index 04ff18fa18e2..f816c73e247e 100644 --- a/sound/soc/codecs/rt1308-sdw.h +++ b/sound/soc/codecs/rt1308-sdw.h @@ -159,7 +159,6 @@ struct rt1308_sdw_priv { struct snd_soc_component *component; struct regmap *regmap; struct sdw_slave *sdw_slave; - enum sdw_slave_status status; struct sdw_bus_params params; bool hw_init; bool first_hw_init; -- 2.34.1

3 14

[PATCH 0/2] ALSA: emu10k1: various improvements to the DSP-based mixer code
by Oswald Buddenhagen 18 May '23

18 May '23

Oswald Buddenhagen (2): ALSA: emu10k1: enable bit-exact playback, part 3: pitch ALSA: emu10k1: enable bit-exact playback, part 4: send amounts include/sound/emu10k1.h | 2 ++ sound/pci/emu10k1/emupcm.c | 34 ++++++++++++++++++++++++++++++---- 2 files changed, 32 insertions(+), 4 deletions(-) -- 2.40.0.152.g15d061e6df

2 3

[PATCH 1/2] ALSA: emu10k1: fix synthesizer sample playback position and caching
by Oswald Buddenhagen 18 May '23

18 May '23

Compensate for the cache delay, and actually populate the cache. Without these, the playback would start with garbage (which would be (mostly?) masqueraded by the attack phase). Unlike for the PCM voices, this doesn't try to compensate for the interpolator read-ahead, because it's pointless to be super-exact here. Note that this code is probably still broken for particularly short samples, because we ignore the loop-related parts of CCR. But I'm not going to reverse-engineer that now ... Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen(a)gmx.de> --- sound/pci/emu10k1/emu10k1_callback.c | 36 ++++------------------------ 1 file changed, 5 insertions(+), 31 deletions(-) diff --git a/sound/pci/emu10k1/emu10k1_callback.c b/sound/pci/emu10k1/emu10k1_callback.c index 06440b97b5d7..aab8d64fd708 100644 --- a/sound/pci/emu10k1/emu10k1_callback.c +++ b/sound/pci/emu10k1/emu10k1_callback.c @@ -253,7 +253,7 @@ lookup_voices(struct snd_emux *emu, struct snd_emu10k1 *hw, /* check if sample is finished playing (non-looping only) */ if (bp != best + V_OFF && bp != best + V_FREE && (vp->reg.sample_mode & SNDRV_SFNT_SAMPLE_SINGLESHOT)) { - val = snd_emu10k1_ptr_read(hw, CCCA_CURRADDR, vp->ch); + val = snd_emu10k1_ptr_read(hw, CCCA_CURRADDR, vp->ch) - 64; if (val >= vp->reg.loopstart) bp = best + V_OFF; } @@ -360,7 +360,7 @@ start_voice(struct snd_emux_voice *vp) map = (hw->silent_page.addr << hw->address_mode) | (hw->address_mode ? MAP_PTI_MASK1 : MAP_PTI_MASK0); - addr = vp->reg.start; + addr = vp->reg.start + 64; temp = vp->reg.parm.filterQ; ccca = (temp << 28) | addr; if (vp->apitch < 0xe400) @@ -428,40 +428,14 @@ start_voice(struct snd_emux_voice *vp) /* Q & current address (Q 4bit value, MSB) */ CCCA, ccca, + /* cache */ + CCR, REG_VAL_PUT(CCR_CACHEINVALIDSIZE, 64), + /* reset volume */ VTFT, vtarget | vp->ftarget, CVCF, vtarget | CVCF_CURRENTFILTER_MASK, REGLIST_END); -#if 0 - /* cache */ - { - unsigned int val, sample; - val = 32; - if (vp->reg.sample_mode & SNDRV_SFNT_SAMPLE_8BITS) - sample = 0x80808080; - else { - sample = 0; - val *= 2; - } - - /* cache */ - snd_emu10k1_ptr_write(hw, CCR, ch, 0x1c << 16); - snd_emu10k1_ptr_write(hw, CDE, ch, sample); - snd_emu10k1_ptr_write(hw, CDF, ch, sample); - - /* invalidate maps */ - temp = ((unsigned int)hw->silent_page.addr << hw_address_mode) | (hw->address_mode ? MAP_PTI_MASK1 : MAP_PTI_MASK0); - snd_emu10k1_ptr_write(hw, MAPA, ch, temp); - snd_emu10k1_ptr_write(hw, MAPB, ch, temp); - - /* fill cache */ - val -= 4; - val <<= 25; - val |= 0x1c << 16; - snd_emu10k1_ptr_write(hw, CCR, ch, val); - } -#endif return 0; } -- 2.40.0.152.g15d061e6df

2 2

[PATCH] ALSA: emu10k1: set variables emu1010_routing_info and emu1010_pads_info storage-class-specifier to static
by Tom Rix 18 May '23

18 May '23

smatch reports sound/pci/emu10k1/emumixer.c:519:39: warning: symbol 'emu1010_routing_info' was not declared. Should it be static? sound/pci/emu10k1/emumixer.c:859:36: warning: symbol 'emu1010_pads_info' was not declared. Should it be static? These variables are only used in their defining file, so it should be static Signed-off-by: Tom Rix <trix(a)redhat.com> --- sound/pci/emu10k1/emumixer.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/emu10k1/emumixer.c b/sound/pci/emu10k1/emumixer.c index 3a7f25f81504..26ed7e2fe053 100644 --- a/sound/pci/emu10k1/emumixer.c +++ b/sound/pci/emu10k1/emumixer.c @@ -516,7 +516,7 @@ struct snd_emu1010_routing_info { unsigned n_ins; }; -const struct snd_emu1010_routing_info emu1010_routing_info[] = { +static const struct snd_emu1010_routing_info emu1010_routing_info[] = { { /* rev1 1010 */ .src_regs = emu1010_src_regs, @@ -856,7 +856,7 @@ struct snd_emu1010_pads_info { unsigned n_adc_ctls, n_dac_ctls; }; -const struct snd_emu1010_pads_info emu1010_pads_info[] = { +static const struct snd_emu1010_pads_info emu1010_pads_info[] = { { /* rev1 1010 */ .adc_ctls = snd_emu1010_adc_pads, -- 2.27.0

3 3

[PATCH] ALSA: hda: Fix unhandled register update during auto-suspend period
by Takashi Iwai 18 May '23

18 May '23

It's reported that the recording started right after the driver probe doesn't work properly, and it turned out that this is related with the codec auto-suspend. Namely, after the probe phase, the usage count goes zero, and the auto-suspend is programmed, but the codec is kept still active until the auto-suspend expiration. When an application (e.g. alsactl) updates the mixer values at this moment, the values are cached but not actually written. Then, starting arecord thereafter also results in the silence because of the missing unmute. The root cause is the handling of "lazy update" mode; when a mixer value is updated *after* the suspend, it should update only the cache and exits. At the resume, the cached value is written to the device, in turn. The problem is that the current code misinterprets the state of auto-suspend as if it were already suspended. Although we can add the check of the actual device state after pm_runtime_get_if_in_use() for catching the missing state, this won't suffice; the second call of regmap_update_bits_check() will skip writing the register because the cache has been already updated by the first call. So we'd need fixes in two different places. OTOH, a simpler fix is to replace pm_runtime_get_if_in_use() with pm_runtime_get_if_active() (with ign_usage_count=true). This change implies that the driver takes the pm refcount if the device is still in ACTIVE state and continues the processing. A small caveat is that this will leave the auto-suspend timer. But, since the timer callback itself checks the device state and aborts gracefully when it's active, this won't be any substantial problem. Long story short: we address the missing register-write problem just by replacing the pm_runtime_*() call in snd_hda_keep_power_up(). Fixes: fc4f000bf8c0 ("ALSA: hda - Fix unexpected resume through regmap code path") Reported-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> Closes: https://lore.kernel.org/r/a7478636-af11-92ab-731c-9b13c582a70d@linux.intel.… Suggested-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/hda/hdac_device.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index accc9d279ce5..6c043fbd606f 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -611,7 +611,7 @@ EXPORT_SYMBOL_GPL(snd_hdac_power_up_pm); int snd_hdac_keep_power_up(struct hdac_device *codec) { if (!atomic_inc_not_zero(&codec->in_pm)) { - int ret = pm_runtime_get_if_in_use(&codec->dev); + int ret = pm_runtime_get_if_active(&codec->dev, true); if (!ret) return -1; if (ret < 0) -- 2.35.3

1 0

[PATCH] ALSA: emu10k1: introduce and use snd_emu10k1_ptr_write_multiple()
by Oswald Buddenhagen 18 May '23

18 May '23

While this nicely denoises the code, the real intent is being able to write many registers pseudo-atomically, which will come in handy later. Idea stolen from kX-project. Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen(a)gmx.de> --- This patch needs to be applied on top of the patch titled "ALSA: emu10k1: fix PCM playback buffer size constraints". --- include/sound/emu10k1.h | 4 + sound/pci/emu10k1/emu10k1_callback.c | 209 +++++++++++++-------------- sound/pci/emu10k1/emu10k1_main.c | 164 +++++++++++---------- sound/pci/emu10k1/emumixer.c | 8 +- sound/pci/emu10k1/emupcm.c | 110 +++++++------- sound/pci/emu10k1/io.c | 31 ++++ 6 files changed, 292 insertions(+), 234 deletions(-) diff --git a/include/sound/emu10k1.h b/include/sound/emu10k1.h index ee662a1b0dc7..9c5de1f45566 100644 --- a/include/sound/emu10k1.h +++ b/include/sound/emu10k1.h @@ -64,6 +64,9 @@ #define REG_VAL_GET(r, v) ((v & REG_MASK(r)) >> REG_SHIFT(r)) #define REG_VAL_PUT(r, v) ((v) << REG_SHIFT(r)) +// List terminator for snd_emu10k1_ptr_write_multiple() +#define REGLIST_END ~0 + // Audigy specify registers are prefixed with 'A_' /************************************************************************************************/ @@ -1793,6 +1796,7 @@ int snd_emu10k1_done(struct snd_emu10k1 * emu); /* I/O functions */ unsigned int snd_emu10k1_ptr_read(struct snd_emu10k1 * emu, unsigned int reg, unsigned int chn); void snd_emu10k1_ptr_write(struct snd_emu10k1 *emu, unsigned int reg, unsigned int chn, unsigned int data); +void snd_emu10k1_ptr_write_multiple(struct snd_emu10k1 *emu, unsigned int chn, ...); unsigned int snd_emu10k1_ptr20_read(struct snd_emu10k1 * emu, unsigned int reg, unsigned int chn); void snd_emu10k1_ptr20_write(struct snd_emu10k1 *emu, unsigned int reg, unsigned int chn, unsigned int data); int snd_emu10k1_spi_write(struct snd_emu10k1 * emu, unsigned int data); diff --git a/sound/pci/emu10k1/emu10k1_callback.c b/sound/pci/emu10k1/emu10k1_callback.c index 9455df18f7b2..06440b97b5d7 100644 --- a/sound/pci/emu10k1/emu10k1_callback.c +++ b/sound/pci/emu10k1/emu10k1_callback.c @@ -33,9 +33,8 @@ static void release_voice(struct snd_emux_voice *vp); static void update_voice(struct snd_emux_voice *vp, int update); static void terminate_voice(struct snd_emux_voice *vp); static void free_voice(struct snd_emux_voice *vp); -static void set_fmmod(struct snd_emu10k1 *hw, struct snd_emux_voice *vp); -static void set_fm2frq2(struct snd_emu10k1 *hw, struct snd_emux_voice *vp); -static void set_filterQ(struct snd_emu10k1 *hw, struct snd_emux_voice *vp); +static u32 make_fmmod(struct snd_emux_voice *vp); +static u32 make_fm2frq2(struct snd_emux_voice *vp); /* * Ensure a value is between two points @@ -116,14 +115,13 @@ snd_emu10k1_synth_get_voice(struct snd_emu10k1 *hw) static void release_voice(struct snd_emux_voice *vp) { - int dcysusv; struct snd_emu10k1 *hw; hw = vp->hw; - dcysusv = (unsigned char)vp->reg.parm.modrelease | DCYSUSM_PHASE1_MASK; - snd_emu10k1_ptr_write(hw, DCYSUSM, vp->ch, dcysusv); - dcysusv = (unsigned char)vp->reg.parm.volrelease | DCYSUSV_PHASE1_MASK | DCYSUSV_CHANNELENABLE_MASK; - snd_emu10k1_ptr_write(hw, DCYSUSV, vp->ch, dcysusv); + snd_emu10k1_ptr_write_multiple(hw, vp->ch, + DCYSUSM, (unsigned char)vp->reg.parm.modrelease | DCYSUSM_PHASE1_MASK, + DCYSUSV, (unsigned char)vp->reg.parm.volrelease | DCYSUSV_PHASE1_MASK | DCYSUSV_CHANNELENABLE_MASK, + REGLIST_END); } @@ -192,13 +190,13 @@ update_voice(struct snd_emux_voice *vp, int update) snd_emu10k1_ptr_write(hw, PTRX_FXSENDAMOUNT_B, vp->ch, vp->aaux); } if (update & SNDRV_EMUX_UPDATE_FMMOD) - set_fmmod(hw, vp); + snd_emu10k1_ptr_write(hw, FMMOD, vp->ch, make_fmmod(vp)); if (update & SNDRV_EMUX_UPDATE_TREMFREQ) snd_emu10k1_ptr_write(hw, TREMFRQ, vp->ch, vp->reg.parm.tremfrq); if (update & SNDRV_EMUX_UPDATE_FM2FRQ2) - set_fm2frq2(hw, vp); + snd_emu10k1_ptr_write(hw, FM2FRQ2, vp->ch, make_fm2frq2(vp)); if (update & SNDRV_EMUX_UPDATE_Q) - set_filterQ(hw, vp); + snd_emu10k1_ptr_write(hw, CCCA_RESONANCE, vp->ch, vp->reg.parm.filterQ); } @@ -310,6 +308,7 @@ start_voice(struct snd_emux_voice *vp) { unsigned int temp; int ch; + u32 psst, dsl, map, ccca, vtarget; unsigned int addr, mapped_offset; struct snd_midi_channel *chan; struct snd_emu10k1 *hw; @@ -347,66 +346,93 @@ start_voice(struct snd_emux_voice *vp) snd_emu10k1_ptr_write(hw, FXRT, ch, temp); } - /* channel to be silent and idle */ - snd_emu10k1_ptr_write(hw, DCYSUSV, ch, 0); - snd_emu10k1_ptr_write(hw, VTFT, ch, VTFT_FILTERTARGET_MASK); - snd_emu10k1_ptr_write(hw, CVCF, ch, CVCF_CURRENTFILTER_MASK); - snd_emu10k1_ptr_write(hw, PTRX, ch, 0); - snd_emu10k1_ptr_write(hw, CPF, ch, 0); - - /* set pitch offset */ - snd_emu10k1_ptr_write(hw, IP, vp->ch, vp->apitch); - - /* set envelope parameters */ - snd_emu10k1_ptr_write(hw, ENVVAL, ch, vp->reg.parm.moddelay); - snd_emu10k1_ptr_write(hw, ATKHLDM, ch, vp->reg.parm.modatkhld); - snd_emu10k1_ptr_write(hw, DCYSUSM, ch, vp->reg.parm.moddcysus); - snd_emu10k1_ptr_write(hw, ENVVOL, ch, vp->reg.parm.voldelay); - snd_emu10k1_ptr_write(hw, ATKHLDV, ch, vp->reg.parm.volatkhld); - /* decay/sustain parameter for volume envelope is used - for triggerg the voice */ - - /* cutoff and volume */ - temp = (unsigned int)vp->acutoff << 8 | (unsigned char)vp->avol; - snd_emu10k1_ptr_write(hw, IFATN, vp->ch, temp); - - /* modulation envelope heights */ - snd_emu10k1_ptr_write(hw, PEFE, ch, vp->reg.parm.pefe); - - /* lfo1/2 delay */ - snd_emu10k1_ptr_write(hw, LFOVAL1, ch, vp->reg.parm.lfo1delay); - snd_emu10k1_ptr_write(hw, LFOVAL2, ch, vp->reg.parm.lfo2delay); - - /* lfo1 pitch & cutoff shift */ - set_fmmod(hw, vp); - /* lfo1 volume & freq */ - snd_emu10k1_ptr_write(hw, TREMFRQ, vp->ch, vp->reg.parm.tremfrq); - /* lfo2 pitch & freq */ - set_fm2frq2(hw, vp); - - /* reverb and loop start (reverb 8bit, MSB) */ temp = vp->reg.parm.reverb; temp += (int)vp->chan->control[MIDI_CTL_E1_REVERB_DEPTH] * 9 / 10; LIMITMAX(temp, 255); addr = vp->reg.loopstart; - snd_emu10k1_ptr_write(hw, PSST, vp->ch, (temp << 24) | addr); + psst = (temp << 24) | addr; - /* chorus & loop end (chorus 8bit, MSB) */ addr = vp->reg.loopend; temp = vp->reg.parm.chorus; temp += (int)chan->control[MIDI_CTL_E3_CHORUS_DEPTH] * 9 / 10; LIMITMAX(temp, 255); - temp = (temp <<24) | addr; - snd_emu10k1_ptr_write(hw, DSL, ch, temp); + dsl = (temp << 24) | addr; - /* clear filter delay memory */ - snd_emu10k1_ptr_write(hw, Z1, ch, 0); - snd_emu10k1_ptr_write(hw, Z2, ch, 0); + map = (hw->silent_page.addr << hw->address_mode) | (hw->address_mode ? MAP_PTI_MASK1 : MAP_PTI_MASK0); - /* invalidate maps */ - temp = (hw->silent_page.addr << hw->address_mode) | (hw->address_mode ? MAP_PTI_MASK1 : MAP_PTI_MASK0); - snd_emu10k1_ptr_write(hw, MAPA, ch, temp); - snd_emu10k1_ptr_write(hw, MAPB, ch, temp); + addr = vp->reg.start; + temp = vp->reg.parm.filterQ; + ccca = (temp << 28) | addr; + if (vp->apitch < 0xe400) + ccca |= CCCA_INTERPROM_0; + else { + unsigned int shift = (vp->apitch - 0xe000) >> 10; + ccca |= shift << 25; + } + if (vp->reg.sample_mode & SNDRV_SFNT_SAMPLE_8BITS) + ccca |= CCCA_8BITSELECT; + + vtarget = (unsigned int)vp->vtarget << 16; + + snd_emu10k1_ptr_write_multiple(hw, ch, + /* channel to be silent and idle */ + DCYSUSV, 0, + VTFT, VTFT_FILTERTARGET_MASK, + CVCF, CVCF_CURRENTFILTER_MASK, + PTRX, 0, + CPF, 0, + + /* set pitch offset */ + IP, vp->apitch, + + /* set envelope parameters */ + ENVVAL, vp->reg.parm.moddelay, + ATKHLDM, vp->reg.parm.modatkhld, + DCYSUSM, vp->reg.parm.moddcysus, + ENVVOL, vp->reg.parm.voldelay, + ATKHLDV, vp->reg.parm.volatkhld, + /* decay/sustain parameter for volume envelope is used + for triggerg the voice */ + + /* cutoff and volume */ + IFATN, (unsigned int)vp->acutoff << 8 | (unsigned char)vp->avol, + + /* modulation envelope heights */ + PEFE, vp->reg.parm.pefe, + + /* lfo1/2 delay */ + LFOVAL1, vp->reg.parm.lfo1delay, + LFOVAL2, vp->reg.parm.lfo2delay, + + /* lfo1 pitch & cutoff shift */ + FMMOD, make_fmmod(vp), + /* lfo1 volume & freq */ + TREMFRQ, vp->reg.parm.tremfrq, + /* lfo2 pitch & freq */ + FM2FRQ2, make_fm2frq2(vp), + + /* reverb and loop start (reverb 8bit, MSB) */ + PSST, psst, + + /* chorus & loop end (chorus 8bit, MSB) */ + DSL, dsl, + + /* clear filter delay memory */ + Z1, 0, + Z2, 0, + + /* invalidate maps */ + MAPA, map, + MAPB, map, + + /* Q & current address (Q 4bit value, MSB) */ + CCCA, ccca, + + /* reset volume */ + VTFT, vtarget | vp->ftarget, + CVCF, vtarget | CVCF_CURRENTFILTER_MASK, + + REGLIST_END); #if 0 /* cache */ { @@ -437,34 +463,16 @@ start_voice(struct snd_emux_voice *vp) } #endif - /* Q & current address (Q 4bit value, MSB) */ - addr = vp->reg.start; - temp = vp->reg.parm.filterQ; - temp = (temp<<28) | addr; - if (vp->apitch < 0xe400) - temp |= CCCA_INTERPROM_0; - else { - unsigned int shift = (vp->apitch - 0xe000) >> 10; - temp |= shift << 25; - } - if (vp->reg.sample_mode & SNDRV_SFNT_SAMPLE_8BITS) - temp |= CCCA_8BITSELECT; - snd_emu10k1_ptr_write(hw, CCCA, ch, temp); - - /* reset volume */ - temp = (unsigned int)vp->vtarget << 16; - snd_emu10k1_ptr_write(hw, VTFT, ch, temp | vp->ftarget); - snd_emu10k1_ptr_write(hw, CVCF, ch, temp | CVCF_CURRENTFILTER_MASK); return 0; } /* * Start envelope */ static void trigger_voice(struct snd_emux_voice *vp) { - unsigned int temp, ptarget; + unsigned int ptarget; struct snd_emu10k1 *hw; struct snd_emu10k1_memblk *emem; @@ -479,58 +487,49 @@ trigger_voice(struct snd_emux_voice *vp) #else ptarget = IP_TO_CP(vp->apitch); #endif - /* set pitch target and pan (volume) */ - temp = ptarget | (vp->apan << 8) | vp->aaux; - snd_emu10k1_ptr_write(hw, PTRX, vp->ch, temp); + snd_emu10k1_ptr_write_multiple(hw, vp->ch, + /* set pitch target and pan (volume) */ + PTRX, ptarget | (vp->apan << 8) | vp->aaux, - /* pitch target */ - snd_emu10k1_ptr_write(hw, CPF, vp->ch, ptarget); + /* current pitch and fractional address */ + CPF, ptarget, - /* trigger voice */ - snd_emu10k1_ptr_write(hw, DCYSUSV, vp->ch, vp->reg.parm.voldcysus|DCYSUSV_CHANNELENABLE_MASK); + /* enable envelope engine */ + DCYSUSV, vp->reg.parm.voldcysus | DCYSUSV_CHANNELENABLE_MASK, + + REGLIST_END); } #define MOD_SENSE 18 -/* set lfo1 modulation height and cutoff */ -static void -set_fmmod(struct snd_emu10k1 *hw, struct snd_emux_voice *vp) +/* calculate lfo1 modulation height and cutoff register */ +static u32 +make_fmmod(struct snd_emux_voice *vp) { - unsigned short fmmod; short pitch; unsigned char cutoff; int modulation; pitch = (char)(vp->reg.parm.fmmod>>8); cutoff = (vp->reg.parm.fmmod & 0xff); modulation = vp->chan->gm_modulation + vp->chan->midi_pressure; pitch += (MOD_SENSE * modulation) / 1200; LIMITVALUE(pitch, -128, 127); - fmmod = ((unsigned char)pitch<<8) | cutoff; - snd_emu10k1_ptr_write(hw, FMMOD, vp->ch, fmmod); + return ((unsigned char)pitch << 8) | cutoff; } -/* set lfo2 pitch & frequency */ -static void -set_fm2frq2(struct snd_emu10k1 *hw, struct snd_emux_voice *vp) +/* calculate set lfo2 pitch & frequency register */ +static u32 +make_fm2frq2(struct snd_emux_voice *vp) { - unsigned short fm2frq2; short pitch; unsigned char freq; int modulation; pitch = (char)(vp->reg.parm.fm2frq2>>8); freq = vp->reg.parm.fm2frq2 & 0xff; modulation = vp->chan->gm_modulation + vp->chan->midi_pressure; pitch += (MOD_SENSE * modulation) / 1200; LIMITVALUE(pitch, -128, 127); - fm2frq2 = ((unsigned char)pitch<<8) | freq; - snd_emu10k1_ptr_write(hw, FM2FRQ2, vp->ch, fm2frq2); -} - -/* set filterQ */ -static void -set_filterQ(struct snd_emu10k1 *hw, struct snd_emux_voice *vp) -{ - snd_emu10k1_ptr_write(hw, CCCA_RESONANCE, vp->ch, vp->reg.parm.filterQ); + return ((unsigned char)pitch << 8) | freq; } diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index da7c988b5c97..65207ef689cb 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -57,44 +57,49 @@ MODULE_FIRMWARE(EMU1010_NOTEBOOK_FILENAME); void snd_emu10k1_voice_init(struct snd_emu10k1 *emu, int ch) { - snd_emu10k1_ptr_write(emu, DCYSUSV, ch, 0); - snd_emu10k1_ptr_write(emu, VTFT, ch, VTFT_FILTERTARGET_MASK); - snd_emu10k1_ptr_write(emu, CVCF, ch, CVCF_CURRENTFILTER_MASK); - snd_emu10k1_ptr_write(emu, PTRX, ch, 0); - snd_emu10k1_ptr_write(emu, CPF, ch, 0); - snd_emu10k1_ptr_write(emu, CCR, ch, 0); + snd_emu10k1_ptr_write_multiple(emu, ch, + DCYSUSV, 0, + VTFT, VTFT_FILTERTARGET_MASK, + CVCF, CVCF_CURRENTFILTER_MASK, + PTRX, 0, + CPF, 0, + CCR, 0, - snd_emu10k1_ptr_write(emu, PSST, ch, 0); - snd_emu10k1_ptr_write(emu, DSL, ch, 0x10); - snd_emu10k1_ptr_write(emu, CCCA, ch, 0); - snd_emu10k1_ptr_write(emu, Z1, ch, 0); - snd_emu10k1_ptr_write(emu, Z2, ch, 0); - snd_emu10k1_ptr_write(emu, FXRT, ch, 0x32100000); + PSST, 0, + DSL, 0x10, + CCCA, 0, + Z1, 0, + Z2, 0, + FXRT, 0x32100000, - // The rest is meaningless as long as DCYSUSV_CHANNELENABLE_MASK is zero - snd_emu10k1_ptr_write(emu, DCYSUSM, ch, 0); - snd_emu10k1_ptr_write(emu, ATKHLDV, ch, 0); - snd_emu10k1_ptr_write(emu, ATKHLDM, ch, 0); - snd_emu10k1_ptr_write(emu, IP, ch, 0); - snd_emu10k1_ptr_write(emu, IFATN, ch, IFATN_FILTERCUTOFF_MASK | IFATN_ATTENUATION_MASK); - snd_emu10k1_ptr_write(emu, PEFE, ch, 0); - snd_emu10k1_ptr_write(emu, FMMOD, ch, 0); - snd_emu10k1_ptr_write(emu, TREMFRQ, ch, 24); /* 1 Hz */ - snd_emu10k1_ptr_write(emu, FM2FRQ2, ch, 24); /* 1 Hz */ - snd_emu10k1_ptr_write(emu, LFOVAL2, ch, 0); - snd_emu10k1_ptr_write(emu, LFOVAL1, ch, 0); - snd_emu10k1_ptr_write(emu, ENVVOL, ch, 0); - snd_emu10k1_ptr_write(emu, ENVVAL, ch, 0); + // The rest is meaningless as long as DCYSUSV_CHANNELENABLE_MASK is zero + DCYSUSM, 0, + ATKHLDV, 0, + ATKHLDM, 0, + IP, 0, + IFATN, IFATN_FILTERCUTOFF_MASK | IFATN_ATTENUATION_MASK, + PEFE, 0, + FMMOD, 0, + TREMFRQ, 24, /* 1 Hz */ + FM2FRQ2, 24, /* 1 Hz */ + LFOVAL2, 0, + LFOVAL1, 0, + ENVVOL, 0, + ENVVAL, 0, + + REGLIST_END); /* Audigy extra stuffs */ if (emu->audigy) { - snd_emu10k1_ptr_write(emu, A_CSBA, ch, 0); - snd_emu10k1_ptr_write(emu, A_CSDC, ch, 0); - snd_emu10k1_ptr_write(emu, A_CSFE, ch, 0); - snd_emu10k1_ptr_write(emu, A_CSHG, ch, 0); - snd_emu10k1_ptr_write(emu, A_FXRT1, ch, 0x03020100); - snd_emu10k1_ptr_write(emu, A_FXRT2, ch, 0x07060504); - snd_emu10k1_ptr_write(emu, A_SENDAMOUNTS, ch, 0); + snd_emu10k1_ptr_write_multiple(emu, ch, + A_CSBA, 0, + A_CSDC, 0, + A_CSFE, 0, + A_CSHG, 0, + A_FXRT1, 0x03020100, + A_FXRT2, 0x07060504, + A_SENDAMOUNTS, 0, + REGLIST_END); } } @@ -148,22 +153,26 @@ static int snd_emu10k1_init(struct snd_emu10k1 *emu, int enable_ir) outl(HCFG_LOCKSOUNDCACHE | HCFG_LOCKTANKCACHE_MASK | HCFG_MUTEBUTTONENABLE, emu->port + HCFG); - /* reset recording buffers */ - snd_emu10k1_ptr_write(emu, MICBS, 0, ADCBS_BUFSIZE_NONE); - snd_emu10k1_ptr_write(emu, MICBA, 0, 0); - snd_emu10k1_ptr_write(emu, FXBS, 0, ADCBS_BUFSIZE_NONE); - snd_emu10k1_ptr_write(emu, FXBA, 0, 0); - snd_emu10k1_ptr_write(emu, ADCBS, 0, ADCBS_BUFSIZE_NONE); - snd_emu10k1_ptr_write(emu, ADCBA, 0, 0); - - /* disable channel interrupt */ outl(0, emu->port + INTE); - snd_emu10k1_ptr_write(emu, CLIEL, 0, 0); - snd_emu10k1_ptr_write(emu, CLIEH, 0, 0); - /* disable stop on loop end */ - snd_emu10k1_ptr_write(emu, SOLEL, 0, 0); - snd_emu10k1_ptr_write(emu, SOLEH, 0, 0); + snd_emu10k1_ptr_write_multiple(emu, 0, + /* reset recording buffers */ + MICBS, ADCBS_BUFSIZE_NONE, + MICBA, 0, + FXBS, ADCBS_BUFSIZE_NONE, + FXBA, 0, + ADCBS, ADCBS_BUFSIZE_NONE, + ADCBA, 0, + + /* disable channel interrupt */ + CLIEL, 0, + CLIEH, 0, + + /* disable stop on loop end */ + SOLEL, 0, + SOLEH, 0, + + REGLIST_END); if (emu->audigy) { /* set SPDIF bypass mode */ @@ -177,9 +186,11 @@ static int snd_emu10k1_init(struct snd_emu10k1 *emu, int enable_ir) for (ch = 0; ch < NUM_G; ch++) snd_emu10k1_voice_init(emu, ch); - snd_emu10k1_ptr_write(emu, SPCS0, 0, emu->spdif_bits[0]); - snd_emu10k1_ptr_write(emu, SPCS1, 0, emu->spdif_bits[1]); - snd_emu10k1_ptr_write(emu, SPCS2, 0, emu->spdif_bits[2]); + snd_emu10k1_ptr_write_multiple(emu, 0, + SPCS0, emu->spdif_bits[0], + SPCS1, emu->spdif_bits[1], + SPCS2, emu->spdif_bits[2], + REGLIST_END); if (emu->card_capabilities->emu_model) { } else if (emu->card_capabilities->ca0151_chip) { /* audigy2 */ @@ -390,41 +401,48 @@ int snd_emu10k1_done(struct snd_emu10k1 *emu) outl(0, emu->port + INTE); /* - * Shutdown the chip + * Shutdown the voices */ - for (ch = 0; ch < NUM_G; ch++) - snd_emu10k1_ptr_write(emu, DCYSUSV, ch, 0); for (ch = 0; ch < NUM_G; ch++) { - snd_emu10k1_ptr_write(emu, VTFT, ch, 0); - snd_emu10k1_ptr_write(emu, CVCF, ch, 0); - snd_emu10k1_ptr_write(emu, PTRX, ch, 0); - snd_emu10k1_ptr_write(emu, CPF, ch, 0); + snd_emu10k1_ptr_write_multiple(emu, ch, + DCYSUSV, 0, + VTFT, 0, + CVCF, 0, + PTRX, 0, + CPF, 0, + REGLIST_END); } - /* reset recording buffers */ - snd_emu10k1_ptr_write(emu, MICBS, 0, 0); - snd_emu10k1_ptr_write(emu, MICBA, 0, 0); - snd_emu10k1_ptr_write(emu, FXBS, 0, 0); - snd_emu10k1_ptr_write(emu, FXBA, 0, 0); - snd_emu10k1_ptr_write(emu, FXWC, 0, 0); - snd_emu10k1_ptr_write(emu, ADCBS, 0, ADCBS_BUFSIZE_NONE); - snd_emu10k1_ptr_write(emu, ADCBA, 0, 0); - snd_emu10k1_ptr_write(emu, TCBS, 0, TCBS_BUFFSIZE_16K); - snd_emu10k1_ptr_write(emu, TCB, 0, 0); + // stop the DSP if (emu->audigy) snd_emu10k1_ptr_write(emu, A_DBG, 0, A_DBG_SINGLE_STEP); else snd_emu10k1_ptr_write(emu, DBG, 0, EMU10K1_DBG_SINGLE_STEP); - /* disable channel interrupt */ - snd_emu10k1_ptr_write(emu, CLIEL, 0, 0); - snd_emu10k1_ptr_write(emu, CLIEH, 0, 0); - snd_emu10k1_ptr_write(emu, SOLEL, 0, 0); - snd_emu10k1_ptr_write(emu, SOLEH, 0, 0); + snd_emu10k1_ptr_write_multiple(emu, 0, + /* reset recording buffers */ + MICBS, 0, + MICBA, 0, + FXBS, 0, + FXBA, 0, + FXWC, 0, + ADCBS, ADCBS_BUFSIZE_NONE, + ADCBA, 0, + TCBS, TCBS_BUFFSIZE_16K, + TCB, 0, + + /* disable channel interrupt */ + CLIEL, 0, + CLIEH, 0, + SOLEL, 0, + SOLEH, 0, + + PTB, 0, + + REGLIST_END); /* disable audio and lock cache */ outl(HCFG_LOCKSOUNDCACHE | HCFG_LOCKTANKCACHE_MASK | HCFG_MUTEBUTTONENABLE, emu->port + HCFG); - snd_emu10k1_ptr_write(emu, PTB, 0, 0); return 0; } diff --git a/sound/pci/emu10k1/emumixer.c b/sound/pci/emu10k1/emumixer.c index 3a7f25f81504..183051e846f2 100644 --- a/sound/pci/emu10k1/emumixer.c +++ b/sound/pci/emu10k1/emumixer.c @@ -1396,10 +1396,10 @@ static const struct snd_kcontrol_new snd_emu10k1_spdif_control = static void update_emu10k1_fxrt(struct snd_emu10k1 *emu, int voice, unsigned char *route) { if (emu->audigy) { - snd_emu10k1_ptr_write(emu, A_FXRT1, voice, - snd_emu10k1_compose_audigy_fxrt1(route)); - snd_emu10k1_ptr_write(emu, A_FXRT2, voice, - snd_emu10k1_compose_audigy_fxrt2(route)); + snd_emu10k1_ptr_write_multiple(emu, voice, + A_FXRT1, snd_emu10k1_compose_audigy_fxrt1(route), + A_FXRT2, snd_emu10k1_compose_audigy_fxrt2(route), + REGLIST_END); } else { snd_emu10k1_ptr_write(emu, FXRT, voice, snd_emu10k1_compose_send_routing(route)); diff --git a/sound/pci/emu10k1/emupcm.c b/sound/pci/emu10k1/emupcm.c index 9045359bb461..1ca16f0ddbed 100644 --- a/sound/pci/emu10k1/emupcm.c +++ b/sound/pci/emu10k1/emupcm.c @@ -268,47 +268,43 @@ static void snd_emu10k1_pcm_init_voice(struct snd_emu10k1 *emu, memcpy(send_routing, &mix->send_routing[tmp][0], 8); memcpy(send_amount, &mix->send_volume[tmp][0], 8); } - - if (stereo) { - // Not really necessary for the slave, but it doesn't hurt - snd_emu10k1_ptr_write(emu, CPF, voice, CPF_STEREO_MASK); - } else { - snd_emu10k1_ptr_write(emu, CPF, voice, 0); - } - - /* setup routing */ - if (emu->audigy) { - snd_emu10k1_ptr_write(emu, A_FXRT1, voice, - snd_emu10k1_compose_audigy_fxrt1(send_routing)); - snd_emu10k1_ptr_write(emu, A_FXRT2, voice, - snd_emu10k1_compose_audigy_fxrt2(send_routing)); - snd_emu10k1_ptr_write(emu, A_SENDAMOUNTS, voice, - snd_emu10k1_compose_audigy_sendamounts(send_amount)); - } else - snd_emu10k1_ptr_write(emu, FXRT, voice, - snd_emu10k1_compose_send_routing(send_routing)); - /* Assumption that PT is already 0 so no harm overwriting */ - snd_emu10k1_ptr_write(emu, PTRX, voice, (send_amount[0] << 8) | send_amount[1]); - // Stereo slaves don't need to have the addresses set, but it doesn't hurt - snd_emu10k1_ptr_write(emu, DSL, voice, end_addr | (send_amount[3] << 24)); - snd_emu10k1_ptr_write(emu, PSST, voice, start_addr | (send_amount[2] << 24)); if (emu->card_capabilities->emu_model) pitch_target = PITCH_48000; /* Disable interpolators on emu1010 card */ - else + else pitch_target = emu10k1_calc_pitch_target(runtime->rate); - snd_emu10k1_ptr_write(emu, CCCA, voice, - emu10k1_select_interprom(pitch_target) | - (w_16 ? 0 : CCCA_8BITSELECT)); - /* Clear filter delay memory */ - snd_emu10k1_ptr_write(emu, Z1, voice, 0); - snd_emu10k1_ptr_write(emu, Z2, voice, 0); - /* invalidate maps */ - silent_page = ((unsigned int)emu->silent_page.addr << emu->address_mode) | (emu->address_mode ? MAP_PTI_MASK1 : MAP_PTI_MASK0); - snd_emu10k1_ptr_write(emu, MAPA, voice, silent_page); - snd_emu10k1_ptr_write(emu, MAPB, voice, silent_page); - // Disable filter (in conjunction with CCCA_RESONANCE == 0) - snd_emu10k1_ptr_write(emu, VTFT, voice, VTFT_FILTERTARGET_MASK); - snd_emu10k1_ptr_write(emu, CVCF, voice, CVCF_CURRENTFILTER_MASK); + silent_page = ((unsigned int)emu->silent_page.addr << emu->address_mode) | + (emu->address_mode ? MAP_PTI_MASK1 : MAP_PTI_MASK0); + snd_emu10k1_ptr_write_multiple(emu, voice, + // Not really necessary for the slave, but it doesn't hurt + CPF, stereo ? CPF_STEREO_MASK : 0, + // Assumption that PT is already 0 so no harm overwriting + PTRX, (send_amount[0] << 8) | send_amount[1], + // Stereo slaves don't need to have the addresses set, but it doesn't hurt + DSL, end_addr | (send_amount[3] << 24), + PSST, start_addr | (send_amount[2] << 24), + CCCA, emu10k1_select_interprom(pitch_target) | + (w_16 ? 0 : CCCA_8BITSELECT), + // Clear filter delay memory + Z1, 0, + Z2, 0, + // Invalidate maps + MAPA, silent_page, + MAPB, silent_page, + // Disable filter (in conjunction with CCCA_RESONANCE == 0) + VTFT, VTFT_FILTERTARGET_MASK, + CVCF, CVCF_CURRENTFILTER_MASK, + REGLIST_END); + // Setup routing + if (emu->audigy) { + snd_emu10k1_ptr_write_multiple(emu, voice, + A_FXRT1, snd_emu10k1_compose_audigy_fxrt1(send_routing), + A_FXRT2, snd_emu10k1_compose_audigy_fxrt2(send_routing), + A_SENDAMOUNTS, snd_emu10k1_compose_audigy_sendamounts(send_amount), + REGLIST_END); + } else { + snd_emu10k1_ptr_write(emu, FXRT, voice, + snd_emu10k1_compose_send_routing(send_routing)); + } spin_unlock_irqrestore(&emu->reg_lock, flags); } @@ -466,8 +462,10 @@ static int snd_emu10k1_capture_prepare(struct snd_pcm_substream *substream) break; case CAPTURE_EFX: if (emu->audigy) { - snd_emu10k1_ptr_write(emu, A_FXWC1, 0, 0); - snd_emu10k1_ptr_write(emu, A_FXWC2, 0, 0); + snd_emu10k1_ptr_write_multiple(emu, 0, + A_FXWC1, 0, + A_FXWC2, 0, + REGLIST_END); } else snd_emu10k1_ptr_write(emu, FXWC, 0, 0); break; @@ -574,8 +572,10 @@ static void snd_emu10k1_playback_commit_volume(struct snd_emu10k1 *emu, struct snd_emu10k1_voice *evoice, unsigned int vattn) { - snd_emu10k1_ptr_write(emu, VTFT, evoice->number, vattn | VTFT_FILTERTARGET_MASK); - snd_emu10k1_ptr_write(emu, CVCF, evoice->number, vattn | CVCF_CURRENTFILTER_MASK); + snd_emu10k1_ptr_write_multiple(emu, evoice->number, + VTFT, vattn | VTFT_FILTERTARGET_MASK, + CVCF, vattn | CVCF_CURRENTFILTER_MASK, + REGLIST_END); } static void snd_emu10k1_playback_unmute_voice(struct snd_emu10k1 *emu, @@ -716,8 +716,10 @@ static int snd_emu10k1_capture_trigger(struct snd_pcm_substream *substream, break; case CAPTURE_EFX: if (emu->audigy) { - snd_emu10k1_ptr_write(emu, A_FXWC1, 0, epcm->capture_cr_val); - snd_emu10k1_ptr_write(emu, A_FXWC2, 0, epcm->capture_cr_val2); + snd_emu10k1_ptr_write_multiple(emu, 0, + A_FXWC1, epcm->capture_cr_val, + A_FXWC2, epcm->capture_cr_val2, + REGLIST_END); dev_dbg(emu->card->dev, "cr_val=0x%x, cr_val2=0x%x\n", epcm->capture_cr_val, @@ -744,8 +746,10 @@ static int snd_emu10k1_capture_trigger(struct snd_pcm_substream *substream, break; case CAPTURE_EFX: if (emu->audigy) { - snd_emu10k1_ptr_write(emu, A_FXWC1, 0, 0); - snd_emu10k1_ptr_write(emu, A_FXWC2, 0, 0); + snd_emu10k1_ptr_write_multiple(emu, 0, + A_FXWC1, 0, + A_FXWC2, 0, + REGLIST_END); } else snd_emu10k1_ptr_write(emu, FXWC, 0, 0); break; @@ -1562,12 +1566,14 @@ static int snd_emu10k1_fx8010_playback_prepare(struct snd_pcm_substream *substre pcm->pcm_rec.sw_buffer_size = snd_pcm_lib_buffer_bytes(substream); pcm->tram_pos = INITIAL_TRAM_POS(pcm->buffer_size); pcm->tram_shift = 0; - snd_emu10k1_ptr_write(emu, emu->gpr_base + pcm->gpr_running, 0, 0); /* reset */ - snd_emu10k1_ptr_write(emu, emu->gpr_base + pcm->gpr_trigger, 0, 0); /* reset */ - snd_emu10k1_ptr_write(emu, emu->gpr_base + pcm->gpr_size, 0, runtime->buffer_size); - snd_emu10k1_ptr_write(emu, emu->gpr_base + pcm->gpr_ptr, 0, 0); /* reset ptr number */ - snd_emu10k1_ptr_write(emu, emu->gpr_base + pcm->gpr_count, 0, runtime->period_size); - snd_emu10k1_ptr_write(emu, emu->gpr_base + pcm->gpr_tmpcount, 0, runtime->period_size); + snd_emu10k1_ptr_write_multiple(emu, 0, + emu->gpr_base + pcm->gpr_running, 0, /* reset */ + emu->gpr_base + pcm->gpr_trigger, 0, /* reset */ + emu->gpr_base + pcm->gpr_size, runtime->buffer_size, + emu->gpr_base + pcm->gpr_ptr, 0, /* reset ptr number */ + emu->gpr_base + pcm->gpr_count, runtime->period_size, + emu->gpr_base + pcm->gpr_tmpcount, runtime->period_size, + REGLIST_END); for (i = 0; i < pcm->channels; i++) snd_emu10k1_ptr_write(emu, TANKMEMADDRREGBASE + 0x80 + pcm->etram[i], 0, (TANKMEMADDRREG_READ|TANKMEMADDRREG_ALIGN) + i * (runtime->buffer_size / pcm->channels)); return 0; diff --git a/sound/pci/emu10k1/io.c b/sound/pci/emu10k1/io.c index 36fd6f7a0a2c..6419719c739c 100644 --- a/sound/pci/emu10k1/io.c +++ b/sound/pci/emu10k1/io.c @@ -94,6 +94,37 @@ void snd_emu10k1_ptr_write(struct snd_emu10k1 *emu, unsigned int reg, unsigned i EXPORT_SYMBOL(snd_emu10k1_ptr_write); +void snd_emu10k1_ptr_write_multiple(struct snd_emu10k1 *emu, unsigned int chn, ...) +{ + va_list va; + u32 addr_mask; + unsigned long flags; + + if (snd_BUG_ON(!emu)) + return; + if (snd_BUG_ON(chn & ~PTR_CHANNELNUM_MASK)) + return; + addr_mask = ~((emu->audigy ? A_PTR_ADDRESS_MASK : PTR_ADDRESS_MASK) >> 16); + + va_start(va, chn); + spin_lock_irqsave(&emu->emu_lock, flags); + for (;;) { + u32 data; + u32 reg = va_arg(va, u32); + if (reg == REGLIST_END) + break; + data = va_arg(va, u32); + if (snd_BUG_ON(reg & addr_mask)) // Only raw registers supported here + continue; + outl((reg << 16) | chn, emu->port + PTR); + outl(data, emu->port + DATA); + } + spin_unlock_irqrestore(&emu->emu_lock, flags); + va_end(va); +} + +EXPORT_SYMBOL(snd_emu10k1_ptr_write_multiple); + unsigned int snd_emu10k1_ptr20_read(struct snd_emu10k1 * emu, unsigned int reg, unsigned int chn) -- 2.40.0.152.g15d061e6df

2 1

[PATCH 1/4] ALSA: emu10k1: simplify interrupt handler, part 1
by Oswald Buddenhagen 18 May '23

18 May '23

IPR_CHANNELNUMBERMASK cannot be non-zero when IPR_CHANNELLOOP is unset, so join marking them as handled. This logically reverts part of commit f453e20d8a0 ("ALSA update 0.9.3a"), which made the inverse change with no explanation. Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen(a)gmx.de> --- sound/pci/emu10k1/irq.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/pci/emu10k1/irq.c b/sound/pci/emu10k1/irq.c index dfb44e5e69a7..0cb89bd8c16b 100644 --- a/sound/pci/emu10k1/irq.c +++ b/sound/pci/emu10k1/irq.c @@ -79,9 +79,8 @@ irqreturn_t snd_emu10k1_interrupt(int irq, void *dev_id) val >>= 1; pvoice++; } - status &= ~IPR_CHANNELLOOP; + status &= ~(IPR_CHANNELLOOP | IPR_CHANNELNUMBERMASK); } - status &= ~IPR_CHANNELNUMBERMASK; if (status & (IPR_ADCBUFFULL|IPR_ADCBUFHALFFULL)) { if (emu->capture_interrupt) emu->capture_interrupt(emu, status); -- 2.40.0.152.g15d061e6df

2 4

[PATCH v2 1/9] ALSA: emu10k1: pass frame instead of byte addresses
by Oswald Buddenhagen 18 May '23

18 May '23

... to snd_emu10k1_pcm_init_voice(). This makes the code arguably less convoluted. Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen(a)gmx.de> --- sound/pci/emu10k1/emupcm.c | 31 +++++++++---------------------- 1 file changed, 9 insertions(+), 22 deletions(-) diff --git a/sound/pci/emu10k1/emupcm.c b/sound/pci/emu10k1/emupcm.c index d669f93d8930..9f151a0a7756 100644 --- a/sound/pci/emu10k1/emupcm.c +++ b/sound/pci/emu10k1/emupcm.c @@ -270,15 +270,6 @@ static void snd_emu10k1_pcm_init_voice(struct snd_emu10k1 *emu, stereo = runtime->channels == 2; w_16 = snd_pcm_format_width(runtime->format) == 16; - if (!extra && stereo) { - start_addr >>= 1; - end_addr >>= 1; - } - if (w_16) { - start_addr >>= 1; - end_addr >>= 1; - } - spin_lock_irqsave(&emu->reg_lock, flags); /* volume parameters */ @@ -424,19 +415,16 @@ static int snd_emu10k1_playback_prepare(struct snd_pcm_substream *substream) struct snd_emu10k1 *emu = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; struct snd_emu10k1_pcm *epcm = runtime->private_data; + bool w_16 = snd_pcm_format_width(runtime->format) == 16; + bool stereo = runtime->channels == 2; unsigned int start_addr, end_addr; - start_addr = epcm->start_addr; - end_addr = snd_pcm_lib_period_bytes(substream); - if (runtime->channels == 2) { - start_addr >>= 1; - end_addr >>= 1; - } - end_addr += start_addr; + start_addr = epcm->start_addr >> w_16; + end_addr = start_addr + runtime->period_size; snd_emu10k1_pcm_init_voice(emu, 1, 1, epcm->extra, start_addr, end_addr, NULL); - start_addr = epcm->start_addr; - end_addr = epcm->start_addr + snd_pcm_lib_buffer_bytes(substream); + start_addr >>= stereo; + end_addr = start_addr + runtime->buffer_size; snd_emu10k1_pcm_init_voice(emu, 1, 0, epcm->voices[0], start_addr, end_addr, &emu->pcm_mixer[substream->number]); @@ -452,14 +440,13 @@ static int snd_emu10k1_efx_playback_prepare(struct snd_pcm_substream *substream) struct snd_emu10k1 *emu = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; struct snd_emu10k1_pcm *epcm = runtime->private_data; - unsigned int start_addr, end_addr; + unsigned int start_addr; unsigned int channel_size; int i; - start_addr = epcm->start_addr; - end_addr = epcm->start_addr + snd_pcm_lib_buffer_bytes(substream); + start_addr = epcm->start_addr >> 1; // 16-bit voices - channel_size = ( end_addr - start_addr ) / NUM_EFX_PLAYBACK; + channel_size = runtime->buffer_size; snd_emu10k1_pcm_init_voice(emu, 1, 1, epcm->extra, start_addr, start_addr + (channel_size / 2), NULL); -- 2.40.0.152.g15d061e6df

2 10

HDA, power saving and recording
by Amadeusz Sławiński 18 May '23

18 May '23

Hi Takashi, I have a question related to HDA codecs, power saving and sending verb configuration. In our tests we test avs driver with HDA codec configuration and we see problem when arecord is started before codec goes to first sleep. In ASoC code we set power save timeout to 2 seconds [1], if we start recording before codec goes to sleep, we record only silence. Now I've spend some time investigating the issue and as far as I can tell, when codec is being initialized channels are being set to mute and when one starts stream before first "sleep" it never sends proper configuration to codec. However if one allows codec to go to sleep proper configuration is being send on resume. I've also managed to reproduce it with non DSP configuration (using snd_hda_intel driver). One needs to set power_save to non zero value and start recording, before it first goes into first sleep. Here are the steps: 1. Add following to /etc/modprobe.d/test.conf options snd_intel_dspcfg dsp_driver=1 options snd_hda_intel power_save=5 blacklist snd_hda_intel 2. Reboot the machine 3. Run something like (it is time sensitive, so I just run it all in one line): echo "PRE MODPROBE" > /dev/kmsg; modprobe snd_hda_intel; echo "POST MODPROBE" > /dev/kmsg; sleep 1; echo "PRE ARECORD" > /dev/kmsg; arecord -Dhw:PCH,0 -r48 -c2 -fS16_LE -d 20 -vvv /tmp/cap.wav; echo "POST ARECORD" > /dev/kmsg; 4. Observe silence in above file 5. Run after waiting more than 5 seconds: echo "PRE SECOND ARECORD" > /dev/kmsg; arecord -Dhw:PCH,0 -r48 -c2 -fS16_LE -d 20 -vvv /tmp/cap.wav; echo "POST SECOND ARECORD" > /dev/kmsg; I'm also attaching logs from my reproduction with debug.diff (also attached) applied. Main purpose of debug.diff is to log when snd_hdac_exec_verb() is called and to log backtraces for when record input is muted on load and set on startup. When looking at attached dmesg, it is easy to see that when first arecord is being run almost no configuration happens. Now my question is what would be the proper way to fix the issue? Something like forcing the codec to suspend as soon as it is setup seems to work, but I'm pretty sure it is not the best way...: diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 2e728aad6771..251466aaef88 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1049,6 +1049,8 @@ int snd_hda_codec_device_new(struct hda_bus *bus, struct snd_card *card, pm_runtime_get_noresume(&codec->core.dev); #endif + pm_runtime_force_suspend(&codec->core.dev); + return 0; } EXPORT_SYMBOL_GPL(snd_hda_codec_device_new); Do you have any better ideas? [1] https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/tree/sou…

3 16

[PATCH 1/9] ALSA: emu10k1: pass frame instead of byte addresses
by Oswald Buddenhagen 18 May '23

18 May '23

... to snd_emu10k1_pcm_init_voice(). This makes the code arguably less convoluted. Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen(a)gmx.de> --- sound/pci/emu10k1/emupcm.c | 31 +++++++++---------------------- 1 file changed, 9 insertions(+), 22 deletions(-) diff --git a/sound/pci/emu10k1/emupcm.c b/sound/pci/emu10k1/emupcm.c index d669f93d8930..9f151a0a7756 100644 --- a/sound/pci/emu10k1/emupcm.c +++ b/sound/pci/emu10k1/emupcm.c @@ -270,15 +270,6 @@ static void snd_emu10k1_pcm_init_voice(struct snd_emu10k1 *emu, stereo = runtime->channels == 2; w_16 = snd_pcm_format_width(runtime->format) == 16; - if (!extra && stereo) { - start_addr >>= 1; - end_addr >>= 1; - } - if (w_16) { - start_addr >>= 1; - end_addr >>= 1; - } - spin_lock_irqsave(&emu->reg_lock, flags); /* volume parameters */ @@ -424,19 +415,16 @@ static int snd_emu10k1_playback_prepare(struct snd_pcm_substream *substream) struct snd_emu10k1 *emu = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; struct snd_emu10k1_pcm *epcm = runtime->private_data; + bool w_16 = snd_pcm_format_width(runtime->format) == 16; + bool stereo = runtime->channels == 2; unsigned int start_addr, end_addr; - start_addr = epcm->start_addr; - end_addr = snd_pcm_lib_period_bytes(substream); - if (runtime->channels == 2) { - start_addr >>= 1; - end_addr >>= 1; - } - end_addr += start_addr; + start_addr = epcm->start_addr >> w_16; + end_addr = start_addr + runtime->period_size; snd_emu10k1_pcm_init_voice(emu, 1, 1, epcm->extra, start_addr, end_addr, NULL); - start_addr = epcm->start_addr; - end_addr = epcm->start_addr + snd_pcm_lib_buffer_bytes(substream); + start_addr >>= stereo; + end_addr = start_addr + runtime->buffer_size; snd_emu10k1_pcm_init_voice(emu, 1, 0, epcm->voices[0], start_addr, end_addr, &emu->pcm_mixer[substream->number]); @@ -452,14 +440,13 @@ static int snd_emu10k1_efx_playback_prepare(struct snd_pcm_substream *substream) struct snd_emu10k1 *emu = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; struct snd_emu10k1_pcm *epcm = runtime->private_data; - unsigned int start_addr, end_addr; + unsigned int start_addr; unsigned int channel_size; int i; - start_addr = epcm->start_addr; - end_addr = epcm->start_addr + snd_pcm_lib_buffer_bytes(substream); + start_addr = epcm->start_addr >> 1; // 16-bit voices - channel_size = ( end_addr - start_addr ) / NUM_EFX_PLAYBACK; + channel_size = runtime->buffer_size; snd_emu10k1_pcm_init_voice(emu, 1, 1, epcm->extra, start_addr, start_addr + (channel_size / 2), NULL); -- 2.40.0.152.g15d061e6df

2 12

[PATCH v3 00/65] clk: Make determine_rate mandatory for muxes
by Maxime Ripard 18 May '23

18 May '23

Hi, This is a follow-up to a previous series that was printing a warning when a mux has a set_parent implementation but is missing determine_rate(). The rationale is that set_parent() is very likely to be useful when changing the rate, but it's determine_rate() that takes the parenting decision. If we're missing it, then the current parent is always going to be used, and thus set_parent() will not be used. The only exception being a direct call to clk_set_parent(), but those are fairly rare compared to clk_set_rate(). Stephen then asked to promote the warning to an error, and to fix up all the muxes that are in that situation first. So here it is :) It was build-tested on x86, arm and arm64. Affected drivers have been tracked down by the following coccinelle script: virtual report @ clk_ops @ identifier ops; position p; @@ struct clk_ops ops@p = { ... }; @ has_set_parent @ identifier clk_ops.ops; identifier set_parent_f; @@ struct clk_ops ops = { .set_parent = set_parent_f, }; @ has_determine_rate @ identifier clk_ops.ops; identifier determine_rate_f; @@ struct clk_ops ops = { .determine_rate = determine_rate_f, }; @ script:python depends on report && has_set_parent && !has_determine_rate @ ops << clk_ops.ops; set_parent_f << has_set_parent.set_parent_f; p << clk_ops.p; @@ coccilib.report.print_report(p[0], "ERROR: %s has set_parent (%s)" % (ops, set_parent_f)) Berlin is the only user still matching after this series has been applied, but it's because it uses a composite clock which throws the script off. The driver has been converted and shouldn't be a problem. Let me know what you think, Maxime Signed-off-by: Maxime Ripard <maxime(a)cerno.tech> --- Changes in v3: - Rebased on top of next-20230404 - Link to v2: https://lore.kernel.org/r/20221018-clk-range-checks-fixes-v2-0-f6736dec138e… Changes in v2: - Drop all the patches already applied - Promote the clk registration warning to an error - Make all muxes use determine_rate - Link to v1: https://lore.kernel.org/r/20221018-clk-range-checks-fixes-v1-0-f3ef80518140… --- Maxime Ripard (65): clk: Export clk_hw_forward_rate_request() clk: lan966x: Remove unused round_rate hook clk: nodrv: Add a determine_rate hook clk: test: Add a determine_rate hook clk: actions: composite: Add a determine_rate hook for pass clk clk: at91: main: Add a determine_rate hook clk: at91: sckc: Add a determine_rate hook clk: berlin: div: Add a determine_rate hook clk: cdce706: Add a determine_rate hook clk: k210: pll: Add a determine_rate hook clk: k210: aclk: Add a determine_rate hook clk: k210: mux: Add a determine_rate hook clk: lmk04832: clkout: Add a determine_rate hook clk: lochnagar: Add a determine_rate hook clk: qoriq: Add a determine_rate hook clk: si5341: Add a determine_rate hook clk: stm32f4: mux: Add a determine_rate hook clk: vc5: mux: Add a determine_rate hook clk: vc5: clkout: Add a determine_rate hook clk: wm831x: clkout: Add a determine_rate hook clk: davinci: da8xx-cfgchip: Add a determine_rate hook clk: davinci: da8xx-cfgchip: Add a determine_rate hook clk: imx: busy: Add a determine_rate hook clk: imx: fixup-mux: Add a determine_rate hook clk: imx: scu: Add a determine_rate hook clk: mediatek: cpumux: Add a determine_rate hook clk: pxa: Add a determine_rate hook clk: renesas: r9a06g032: Add a determine_rate hook clk: socfpga: gate: Add a determine_rate hook clk: stm32: core: Add a determine_rate hook clk: tegra: bpmp: Add a determine_rate hook clk: tegra: super: Add a determine_rate hook clk: tegra: periph: Add a determine_rate hook clk: ux500: prcmu: Add a determine_rate hook clk: ux500: sysctrl: Add a determine_rate hook clk: versatile: sp810: Add a determine_rate hook drm/tegra: sor: Add a determine_rate hook phy: cadence: sierra: Add a determine_rate hook phy: cadence: torrent: Add a determine_rate hook phy: ti: am654-serdes: Add a determine_rate hook phy: ti: j721e-wiz: Add a determine_rate hook rtc: sun6i: Add a determine_rate hook ASoC: tlv320aic32x4: Add a determine_rate hook clk: actions: composite: div: Switch to determine_rate clk: actions: composite: fact: Switch to determine_rate clk: at91: smd: Switch to determine_rate clk: axi-clkgen: Switch to determine_rate clk: cdce706: divider: Switch to determine_rate clk: cdce706: clkout: Switch to determine_rate clk: si5341: Switch to determine_rate clk: si5351: pll: Switch to determine_rate clk: si5351: msynth: Switch to determine_rate clk: si5351: clkout: Switch to determine_rate clk: da8xx: clk48: Switch to determine_rate clk: imx: scu: Switch to determine_rate clk: ingenic: cgu: Switch to determine_rate clk: ingenic: tcu: Switch to determine_rate clk: sprd: composite: Switch to determine_rate clk: st: flexgen: Switch to determine_rate clk: stm32: composite: Switch to determine_rate clk: tegra: periph: Switch to determine_rate clk: tegra: super: Switch to determine_rate ASoC: tlv320aic32x4: pll: Switch to determine_rate ASoC: tlv320aic32x4: div: Switch to determine_rate clk: Forbid to register a mux without determine_rate drivers/clk/actions/owl-composite.c | 35 +++++++++++----- drivers/clk/actions/owl-composite.h | 2 +- drivers/clk/at91/clk-main.c | 3 +- drivers/clk/at91/clk-smd.c | 29 +++++++------ drivers/clk/at91/sckc.c | 3 +- drivers/clk/berlin/berlin2-div.c | 3 +- drivers/clk/clk-axi-clkgen.c | 14 ++++--- drivers/clk/clk-cdce706.c | 31 ++++++++------ drivers/clk/clk-k210.c | 17 +++++--- drivers/clk/clk-lan966x.c | 17 -------- drivers/clk/clk-lmk04832.c | 1 + drivers/clk/clk-lochnagar.c | 2 + drivers/clk/clk-qoriq.c | 10 +++-- drivers/clk/clk-si5341.c | 21 +++++----- drivers/clk/clk-si5351.c | 67 +++++++++++++++++-------------- drivers/clk/clk-stm32f4.c | 3 +- drivers/clk/clk-versaclock5.c | 8 ++-- drivers/clk/clk-wm831x.c | 3 +- drivers/clk/clk.c | 15 +++++++ drivers/clk/clk_test.c | 1 + drivers/clk/davinci/da8xx-cfgchip.c | 15 ++++--- drivers/clk/imx/clk-busy.c | 3 +- drivers/clk/imx/clk-fixup-mux.c | 3 +- drivers/clk/imx/clk-scu.c | 27 +++++++++++-- drivers/clk/ingenic/cgu.c | 15 +++---- drivers/clk/ingenic/tcu.c | 19 +++++---- drivers/clk/mediatek/clk-cpumux.c | 3 +- drivers/clk/pxa/clk-pxa.c | 3 +- drivers/clk/renesas/r9a06g032-clocks.c | 3 +- drivers/clk/socfpga/clk-gate.c | 3 +- drivers/clk/sprd/composite.c | 16 +++++--- drivers/clk/st/clk-flexgen.c | 15 +++---- drivers/clk/stm32/clk-stm32-core.c | 33 ++++++++++----- drivers/clk/tegra/clk-bpmp.c | 7 +++- drivers/clk/tegra/clk-periph.c | 19 ++++++--- drivers/clk/tegra/clk-super.c | 18 ++++++--- drivers/clk/ux500/clk-prcmu.c | 3 +- drivers/clk/ux500/clk-sysctrl.c | 4 +- drivers/clk/versatile/clk-sp810.c | 3 +- drivers/gpu/drm/tegra/sor.c | 3 +- drivers/phy/cadence/phy-cadence-sierra.c | 1 + drivers/phy/cadence/phy-cadence-torrent.c | 1 + drivers/phy/ti/phy-am654-serdes.c | 1 + drivers/phy/ti/phy-j721e-wiz.c | 1 + drivers/rtc/rtc-sun6i.c | 2 + sound/soc/codecs/tlv320aic32x4-clk.c | 37 ++++++++++------- 46 files changed, 343 insertions(+), 200 deletions(-) --- base-commit: 6a53bda3aaf3de5edeea27d0b1d8781d067640b6 change-id: 20221018-clk-range-checks-fixes-2039f3523240 Best regards, -- Maxime Ripard <maxime(a)cerno.tech>

11 92

[PATCH 0/5] ASoC: mt8188: add new board support
by Trevor Wu 18 May '23

18 May '23

In the series, we extend the capability of mt8188-mt6359 driver. The following changes are included. 1. Divide ADDA BE dai into two dais for SOF. 2. Register hdmi/dp jack pins. 3. dai_fmt can be configured from device tree. 4. Add some I2S codecs support. In addition, new compatible string "mediatek,mt8188-nau8825" is included for a new board support. Trevor Wu (5): ASoC: mediatek: mt8188: separate ADDA playback dai from capture dai ASoC: mediatek: mt8188-mt6359: register hdmi/dp jack pins ASoC: mediatek: common: soundcard driver add dai_fmt support ASoC: mediatek: mt8188-mt6359: support new board with nau88255 ASoC: dt-bindings: mediatek,mt8188-mt6359: update properties .../sound/mediatek,mt8188-mt6359.yaml | 27 +- sound/soc/mediatek/Kconfig | 4 + .../mediatek/common/mtk-soundcard-driver.c | 49 ++- sound/soc/mediatek/mt8188/mt8188-afe-common.h | 3 +- sound/soc/mediatek/mt8188/mt8188-dai-adda.c | 76 ++-- sound/soc/mediatek/mt8188/mt8188-mt6359.c | 326 +++++++++++++++++- 6 files changed, 431 insertions(+), 54 deletions(-) -- 2.18.0

4 18

[PATCH][next] ALSA: mixart: Replace one-element arrays with simple object declarations
by Gustavo A. R. Silva 18 May '23

18 May '23

One-element arrays are deprecated, and we are replacing them with flexible array members, instead. However, in this case it seems those one-element arrays have never actually been used as fake flexible arrays. See this code that dates from Linux-2.6.12-rc2 initial git repository build (commit 1da177e4c3f4 ("Linux-2.6.12-rc2")): sound/pci/mixart/mixart_core.h: 215 struct mixart_stream_state_req 216 { 217 u32 delayed; 218 u64 scheduler; 219 u32 reserved4np[3]; 220 u32 stream_count; /* set to 1 for instance */ 221 struct mixart_flow_info stream_info; /* could be an array[stream_cou nt] */ 222 } __attribute__((packed)); sound/pci/mixart/mixart.c: 388 389 memset(&stream_state_req, 0, sizeof(stream_state_req)); 390 stream_state_req.stream_count = 1; 391 stream_state_req.stream_info.stream_desc.uid_pipe = stream->pipe->group_uid; 392 stream_state_req.stream_info.stream_desc.stream_idx = stream->substream->number; 393 So, taking the code above as example, replace multiple one-element arrays with simple object declarations, and refactor the rest of the code, accordingly. This helps with the ongoing efforts to tighten the FORTIFY_SOURCE routines on memcpy() and help us make progress towards globally enabling -fstrict-flex-arrays=3 [1]. This results in no differences in binary output. Link: https://github.com/KSPP/linux/issues/79 Link: https://github.com/KSPP/linux/issues/296 Link: https://gcc.gnu.org/pipermail/gcc-patches/2022-October/602902.html [1] Signed-off-by: Gustavo A. R. Silva <gustavoars(a)kernel.org> --- sound/pci/mixart/mixart.c | 8 ++++---- sound/pci/mixart/mixart_core.h | 7 +++---- 2 files changed, 7 insertions(+), 8 deletions(-) diff --git a/sound/pci/mixart/mixart.c b/sound/pci/mixart/mixart.c index 1b078b789604..7ceaf6a7a77e 100644 --- a/sound/pci/mixart/mixart.c +++ b/sound/pci/mixart/mixart.c @@ -98,7 +98,7 @@ static int mixart_set_pipe_state(struct mixart_mgr *mgr, memset(&group_state, 0, sizeof(group_state)); group_state.pipe_count = 1; - group_state.pipe_uid[0] = pipe->group_uid; + group_state.pipe_uid = pipe->group_uid; if(start) request.message_id = MSG_STREAM_START_STREAM_GRP_PACKET; @@ -185,7 +185,7 @@ static int mixart_set_clock(struct mixart_mgr *mgr, clock_properties.clock_mode = CM_STANDALONE; clock_properties.frequency = rate; clock_properties.nb_callers = 1; /* only one entry in uid_caller ! */ - clock_properties.uid_caller[0] = pipe->group_uid; + clock_properties.uid_caller = pipe->group_uid; dev_dbg(&mgr->pci->dev, "mixart_set_clock to %d kHz\n", rate); @@ -565,8 +565,8 @@ static int mixart_set_format(struct mixart_stream *stream, snd_pcm_format_t form stream_param.pipe_count = 1; /* set to 1 */ stream_param.stream_count = 1; /* set to 1 */ - stream_param.stream_desc[0].uid_pipe = stream->pipe->group_uid; - stream_param.stream_desc[0].stream_idx = stream->substream->number; + stream_param.stream_desc.uid_pipe = stream->pipe->group_uid; + stream_param.stream_desc.stream_idx = stream->substream->number; request.message_id = MSG_STREAM_SET_INPUT_STAGE_PARAM; request.uid = (struct mixart_uid){0,0}; diff --git a/sound/pci/mixart/mixart_core.h b/sound/pci/mixart/mixart_core.h index 2f0e29ed5d63..ac74b2133c97 100644 --- a/sound/pci/mixart/mixart_core.h +++ b/sound/pci/mixart/mixart_core.h @@ -231,7 +231,7 @@ struct mixart_group_state_req u64 scheduler; u32 reserved4np[2]; u32 pipe_count; /* set to 1 for instance */ - struct mixart_uid pipe_uid[1]; /* could be an array[pipe_count] */ + struct mixart_uid pipe_uid; } __attribute__((packed)); struct mixart_group_state_resp @@ -314,7 +314,7 @@ struct mixart_clock_properties u32 format; u32 board_mask; u32 nb_callers; /* set to 1 (see below) */ - struct mixart_uid uid_caller[1]; + struct mixart_uid uid_caller; } __attribute__((packed)); struct mixart_clock_properties_resp @@ -401,8 +401,7 @@ struct mixart_stream_param_desc u32 reserved4np[3]; u32 pipe_count; /* set to 1 (array size !) */ u32 stream_count; /* set to 1 (array size !) */ - struct mixart_txx_stream_desc stream_desc[1]; /* only one stream per command, but this could be an array */ - + struct mixart_txx_stream_desc stream_desc; } __attribute__((packed)); -- 2.34.1

2 1

[PATCH 0/9] ASoC: amd: ps: add soundwire support
by Vijendar Mukunda 18 May '23

18 May '23

This patch series add support for - Platform device creation for Soundwire Manager instances and PDM controller. - Soundwire DMA driver. - Interrupt handling for Soundwire manager related interrupts, Soundwire DMA interrupts and ACP error interrupts. - ACP PCI driver PM ops modification with respect to Soundwire Power modes. Vijendar Mukunda (9): ASoC: amd: ps: create platform devices based on acp config ASoC: amd: ps: handle soundwire interrupts in acp pci driver ASoC: amd: ps: add soundwire dma driver for ps platform ASoC: amd: ps: add soundwire dma driver dma ops ASoC: amd: ps: add support for Soundwire DMA interrupts ASoC: amd: ps: add pm ops support for soundwire dma driver ASoC: amd: ps: enable Soundwire DMA driver build ASoC: amd: update comments in Kconfig file ASoC: amd: ps: Add soundwire specific checks in pci driver in pm ops. sound/soc/amd/Kconfig | 3 +- sound/soc/amd/ps/Makefile | 2 + sound/soc/amd/ps/acp63.h | 145 ++++++++- sound/soc/amd/ps/pci-ps.c | 393 +++++++++++++++++++++-- sound/soc/amd/ps/ps-sdw-dma.c | 574 ++++++++++++++++++++++++++++++++++ 5 files changed, 1095 insertions(+), 22 deletions(-) create mode 100644 sound/soc/amd/ps/ps-sdw-dma.c -- 2.34.1

5 20

Bug Report ALC294
by Luca Adrian Lindhorst 18 May '23

18 May '23

Hello, I recently got a ASUS ROG Strix G17 13PV with a Ryzen 7945HX. I am running arch with kernel 6.3.2. Unfortunately the speakers aren't working. Headphones, Microphone (internal and on headset) are working fine. I already fiddled with some workarounds: specifying model=asus-zenbook and model=asus-zenbook-ux31a and fiddling with hda-analyzer, but nothing with success. Any help would be appreciated. See the attached alsa info dump. Kind regards Luca Lindhorst

1 0

No sound with Realtek ALC298 on a Samsung Galaxy book
by Aurélien 17 May '23

17 May '23

Hi, Several years ago I sent several mails about problem with Realtek ALC 298 sound card not working on Samsung Galaxy book. Due to a computer problem i had to stop working on it. However I succeed to get it working and I'm submitting the patch attached to this email. Internal speakers works very well. In order to get headset working I had to hack a little bit since it shares the same pin widget (0x17) with the internal speaker. It seems that the ALSA code have solutions to declare a pin widget as both internal and jack but if I use that it doesn't recognize the presence of the jack.. Maybe I'm wrong somewhere. Last but not least i saw that under windows the driver uses another mixer (0xd through the out widget 0x3 rather than mixer 0xc throught the out widget 0x2) when it uses the headset. It offers the opportunity to have a dedicated volume control for the headset independant from the internal speaker. Maybe it could be interesting to do the same under the Linux driver. Unfortunately I don't know how to do that. Do you think is it possible to integrate this patch to the ALSA code? Thanks in advance. Aurélien

2 2

[PATCH] ALSA: emu10k1: fix 64-bit integer division
by Arnd Bergmann 17 May '23

17 May '23

From: Arnd Bergmann <arnd(a)arndb.de> Division of 64-bit values causes a link failure on 32-bit targets, depending on compiler version and target architecture: ERROR: modpost: "__divdi3" [sound/pci/emu10k1/snd-emu10k1.ko] undefined! ERROR: modpost: "__udivdi3" [sound/pci/emu10k1/snd-emu10k1.ko] undefined! Replace these with the safe div_u64() helpers. Fixes: bb5ceb43b7bf ("ALSA: emu10k1: fix non-zero mixer control defaults in highres mode") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/pci/emu10k1/emufx.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/emu10k1/emufx.c b/sound/pci/emu10k1/emufx.c index f64b2b4eb348..7c28789720d1 100644 --- a/sound/pci/emu10k1/emufx.c +++ b/sound/pci/emu10k1/emufx.c @@ -1156,7 +1156,7 @@ snd_emu10k1_init_mono_control(struct snd_emu10k1_fx8010_control_gpr *ctl, ctl->max = 0x7fffffff; ctl->tlv = snd_emu10k1_db_linear; ctl->translation = EMU10K1_GPR_TRANSLATION_NEGATE; - defval = defval * 0x80000000LL / 100 - 1; + defval = div_u64(defval * 0x80000000LL, 100) - 1; } else { ctl->min = 0; ctl->max = 100; @@ -1178,7 +1178,7 @@ snd_emu10k1_init_stereo_control(struct snd_emu10k1_fx8010_control_gpr *ctl, ctl->max = 0x7fffffff; ctl->tlv = snd_emu10k1_db_linear; ctl->translation = EMU10K1_GPR_TRANSLATION_NEGATE; - defval = defval * 0x80000000LL / 100 - 1; + defval = div_u64(defval * 0x80000000LL, 100) - 1; } else { ctl->min = 0; ctl->max = 100; -- 2.39.2

2 1

[PATCH] ALSA: emu10k1: remove runtime 64-bit divisions
by Oswald Buddenhagen 17 May '23

17 May '23

32-bit platforms don't like these. As we're actually dealing with constants, factor out the calculations and pass them in as additional arguments. To keep the call sites clean, wrap the actual functions in macros which generate the arguments. Fixes: bb5ceb43b7bf ("ALSA: emu10k1: fix non-zero mixer control defaults in highres mode") Fixes: 1298bc978afb ("ALSA: emu10k1: enable bit-exact playback, part 1: DSP attenuation") Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen(a)gmx.de> --- Cc: Christophe Leroy <christophe.leroy(a)csgroup.eu> Cc: Naresh Kamboju <naresh.kamboju(a)linaro.org> --- sound/pci/emu10k1/emufx.c | 18 ++++++++++++------ 1 file changed, 12 insertions(+), 6 deletions(-) diff --git a/sound/pci/emu10k1/emufx.c b/sound/pci/emu10k1/emufx.c index f64b2b4eb348..e9855d37fa5c 100644 --- a/sound/pci/emu10k1/emufx.c +++ b/sound/pci/emu10k1/emufx.c @@ -1144,50 +1144,56 @@ static int snd_emu10k1_ipcm_peek(struct snd_emu10k1 *emu, #define SND_EMU10K1_PLAYBACK_CHANNELS 8 #define SND_EMU10K1_CAPTURE_CHANNELS 4 +#define HR_VAL(v) ((v) * 0x80000000LL / 100 - 1) + static void -snd_emu10k1_init_mono_control(struct snd_emu10k1_fx8010_control_gpr *ctl, - const char *name, int gpr, int defval) +snd_emu10k1_init_mono_control2(struct snd_emu10k1_fx8010_control_gpr *ctl, + const char *name, int gpr, int defval, int defval_hr) { ctl->id.iface = (__force int)SNDRV_CTL_ELEM_IFACE_MIXER; strcpy(ctl->id.name, name); ctl->vcount = ctl->count = 1; if (high_res_gpr_volume) { ctl->min = -1; ctl->max = 0x7fffffff; ctl->tlv = snd_emu10k1_db_linear; ctl->translation = EMU10K1_GPR_TRANSLATION_NEGATE; - defval = defval * 0x80000000LL / 100 - 1; + defval = defval_hr; } else { ctl->min = 0; ctl->max = 100; ctl->tlv = snd_emu10k1_db_scale1; ctl->translation = EMU10K1_GPR_TRANSLATION_NEG_TABLE100; } ctl->gpr[0] = gpr + 0; ctl->value[0] = defval; } +#define snd_emu10k1_init_mono_control(ctl, name, gpr, defval) \ + snd_emu10k1_init_mono_control2(ctl, name, gpr, defval, HR_VAL(defval)) static void -snd_emu10k1_init_stereo_control(struct snd_emu10k1_fx8010_control_gpr *ctl, - const char *name, int gpr, int defval) +snd_emu10k1_init_stereo_control2(struct snd_emu10k1_fx8010_control_gpr *ctl, + const char *name, int gpr, int defval, int defval_hr) { ctl->id.iface = (__force int)SNDRV_CTL_ELEM_IFACE_MIXER; strcpy(ctl->id.name, name); ctl->vcount = ctl->count = 2; if (high_res_gpr_volume) { ctl->min = -1; ctl->max = 0x7fffffff; ctl->tlv = snd_emu10k1_db_linear; ctl->translation = EMU10K1_GPR_TRANSLATION_NEGATE; - defval = defval * 0x80000000LL / 100 - 1; + defval = defval_hr; } else { ctl->min = 0; ctl->max = 100; ctl->tlv = snd_emu10k1_db_scale1; ctl->translation = EMU10K1_GPR_TRANSLATION_NEG_TABLE100; } ctl->gpr[0] = gpr + 0; ctl->value[0] = defval; ctl->gpr[1] = gpr + 1; ctl->value[1] = defval; } +#define snd_emu10k1_init_stereo_control(ctl, name, gpr, defval) \ + snd_emu10k1_init_stereo_control2(ctl, name, gpr, defval, HR_VAL(defval)) static void snd_emu10k1_init_mono_onoff_control(struct snd_emu10k1_fx8010_control_gpr *ctl, -- 2.40.0.152.g15d061e6df

3 2

next: gcc-8-ppc6xx_defconfig: ERROR: modpost: "__divdi3" [sound/pci/emu10k1/snd-emu10k1.ko] undefined!
by Naresh Kamboju 17 May '23

17 May '23

Linux next powerpc gcc-8 build failed on Linux next 20230516 and 20230517. - build/gcc-8-ppc6xx_defconfig Reported-by: Linux Kernel Functional Testing <lkft(a)linaro.org> Build log: ==== make --silent --keep-going --jobs=8 O=/home/tuxbuild/.cache/tuxmake/builds/1/build \ ARCH=powerpc CROSS_COMPILE=powerpc64le-linux-gnu- \ 'CC=sccache powerpc64le-linux-gnu-gcc' \ 'HOSTCC=sccache gcc' ERROR: modpost: "__divdi3" [sound/pci/emu10k1/snd-emu10k1.ko] undefined! ERROR: modpost: "__udivdi3" [sound/pci/emu10k1/snd-emu10k1.ko] undefined! make[2]: *** [/builds/linux/scripts/Makefile.modpost:136: Module.symvers] Error 1 make[2]: Target '__modpost' not remade because of errors. make[1]: *** [/builds/linux/Makefile:1978: modpost] Error 2 links, - https://qa-reports.linaro.org/lkft/linux-next-master/build/next-20230517/te… - https://qa-reports.linaro.org/lkft/linux-next-master/build/next-20230517/te… Steps to reproduce: ======= # To install tuxmake on your system globally: # sudo pip3 install -U tuxmake # # See https://docs.tuxmake.org/ for complete documentation. # Original tuxmake command with fragments listed below. tuxmake --runtime podman --target-arch powerpc --toolchain gcc-8 --kconfig ppc6xx_defconfig -- Linaro LKFT https://lkft.linaro.org

2 1

[PATCH 00/10] ALSA: emu10k1: improvements mostly to the I/O routing mixers for E-MU cards
by Oswald Buddenhagen 17 May '23

17 May '23

Oswald Buddenhagen (10): ALSA: emu10k1: make some initializer arrays less wasteful ALSA: emu10k1: compactize E-MU routing source arrays ALSA: emu10k1: make mixer control mass creation less wasteful ALSA: emu10k1: un-hardcode E-MU mixer control callbacks somewhat ALSA: emu10k1: fix return value of snd_emu1010_dac_pads_put() ALSA: emu10k1: make E-MU I/O routing init data-driven ALSA: emu10k1: make E-MU mixer control creation more data-driven ALSA: emu10k1: improve mixer controls for E-MU 1010 rev2 card ALSA: emu10k1: add explicit support for E-MU 0404 ALSA: emu10k1: make struct snd_emu1010 less wasteful include/sound/emu10k1.h | 47 +- sound/pci/emu10k1/emu10k1_main.c | 268 +------ sound/pci/emu10k1/emumixer.c | 1242 +++++++++++++++++------------- sound/pci/emu10k1/emuproc.c | 18 +- 4 files changed, 759 insertions(+), 816 deletions(-) -- 2.40.0.152.g15d061e6df

2 11

[PATCH 0/6] ALSA: emu10k1: assorted playback-related cleanups
by Oswald Buddenhagen 17 May '23

17 May '23

Oswald Buddenhagen (6): ALSA: emu10k1: straighten out FX send init ALSA: emu10k1: cleanup envelope register init ALSA: emu10k1: remove useless resets of stop-on-loop-end bits ALSA: emu10k1: rewire {en,dis}abling interrupts for PCM playback ALSA: emu10k1: skip needless setting of some voice registers ALSA: emu10k1: factor out snd_emu10k1_compose_audigy_sendamounts() Documentation/sound/cards/audigy-mixer.rst | 36 ++--- include/sound/emu10k1.h | 14 +- sound/pci/emu10k1/emu10k1_main.c | 12 +- sound/pci/emu10k1/emumixer.c | 51 +------ sound/pci/emu10k1/emupcm.c | 168 ++++++++++----------- sound/pci/emu10k1/io.c | 63 +------- 6 files changed, 121 insertions(+), 223 deletions(-) -- 2.40.0.152.g15d061e6df

2 7

[PATCH v2 2/5] ASoC: dt-bindings: Add tas2781 amplifier
by Shenghao Ding 17 May '23

17 May '23

Create tas2781.yaml for tas2781 driver. Signed-off-by: Shenghao Ding <13916275206(a)139.com> --- Changes in v7: - Submit together with tas2781 codec driver code - Add more detail description for ti,audio-slots - Keep consistent for "I2C" - remove reset-gpios description - For reg, express as constraints instead - remove unnecessary '|' Changes to be committed: new file: Documentation/devicetree/bindings/sound/ti,tas2781.yaml --- .../devicetree/bindings/sound/ti,tas2781.yaml | 90 +++++++++++++++++++ 1 file changed, 90 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ti,tas2781.yaml diff --git a/Documentation/devicetree/bindings/sound/ti,tas2781.yaml b/Documentation/devicetree/bindings/sound/ti,tas2781.yaml new file mode 100644 index 000000000000..96c2584855d4 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,tas2781.yaml @@ -0,0 +1,90 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +# Copyright (C) 2022 - 2023 Texas Instruments Incorporated +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ti,tas2781.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Texas Instruments TAS2781 SmartAMP + +maintainers: + - Shenghao Ding <shenghao-ding(a)ti.com> + +description: + The TAS2781 is a mono, digital input Class-D audio amplifier + optimized for efficiently driving high peak power into small + loudspeakers. Integrated an on-chip DSP supports Texas Instruments + Smart Amp speaker protection algorithm. The integrated speaker + voltage and current sense provides for real time + monitoring of loudspeaker behavior. + +properties: + compatible: + enum: + - ti,tas2781 + + reg: + description: I2C address of the primary device. + items: + minimum: 0x38 + maximum: 0x3f + + reset-gpios: + maxItems: 1 + + interrupts: + maxItems: 1 + + ti,audio-slots: + description: + Multiple tas2781s aggregate as one Audio Amp to support + multiple audio slots + $ref: /schemas/types.yaml#/definitions/uint32-array + maxItems: 4 + items: + minimum: 0x38 + maximum: 0x3f + description: + I2C address of the device for different audio slots, + useless in mono case. + + ti,broadcast-addr: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + Generic I2C address for all the tas2781 devices in + purpose of I2C broadcast during the multi-device + writes, useless in mono case. + + '#sound-dai-cells': + const: 1 + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + i2c { + /* example with quad support, such as tablet or pad device */ + #address-cells = <1>; + #size-cells = <0>; + quad: codec@38 { + compatible = "ti,tas2781"; + reg = <0x38>; + #sound-dai-cells = <1>; + reset-gpios = < &gpio1 10 GPIO_ACTIVE_HIGH >; + interrupt-parent = <&gpio1>; + interrupts = <15>; + ti,audio-slots = < 0x38 /* Audio slot 0 */ + 0x3a /* Audio slot 1 */ + 0x39 /* Audio slot 2 */ + 0x3b /* Audio slot 3 */ + >; + /* Generic I2C addr among all the tas2781 for i2c data broadcast */ + ti,broadcast-addr = <0x40>; + }; + }; +... -- 2.34.1

5 11

[PATCH] ALSA: compat_ioctl: use correct snd_ctl_elem_type_t type
by Min-Hua Chen 17 May '23

17 May '23

SNDRV_CTL_ELEM_TYPE_* are type of snd_ctl_elem_type_t, we have to __force cast them to int when comparing them with int to fix the following sparse warnings. sound/core/control_compat.c:203:14: sparse: warning: restricted snd_ctl_elem_type_t degrades to integer sound/core/control_compat.c:205:14: sparse: warning: restricted snd_ctl_elem_type_t degrades to integer sound/core/control_compat.c:207:14: sparse: warning: restricted snd_ctl_elem_type_t degrades to integer sound/core/control_compat.c:209:14: sparse: warning: restricted snd_ctl_elem_type_t degrades to integer sound/core/control_compat.c:237:21: sparse: warning: restricted snd_ctl_elem_type_t degrades to integer sound/core/control_compat.c:238:21: sparse: warning: restricted snd_ctl_elem_type_t degrades to integer sound/core/control_compat.c:270:21: sparse: warning: restricted snd_ctl_elem_type_t degrades to integer sound/core/control_compat.c:271:21: sparse: warning: restricted snd_ctl_elem_type_t degrades to integer Signed-off-by: Min-Hua Chen <minhuadotchen(a)gmail.com> --- sound/core/control_compat.c | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/core/control_compat.c b/sound/core/control_compat.c index d8a86d1a99d6..9cae5d74335c 100644 --- a/sound/core/control_compat.c +++ b/sound/core/control_compat.c @@ -197,7 +197,7 @@ static int get_ctl_type(struct snd_card *card, struct snd_ctl_elem_id *id, return err; } -static int get_elem_size(int type, int count) +static int get_elem_size(snd_ctl_elem_type_t type, int count) { switch (type) { case SNDRV_CTL_ELEM_TYPE_INTEGER64: @@ -234,8 +234,8 @@ static int copy_ctl_value_from_user(struct snd_card *card, if (type < 0) return type; - if (type == SNDRV_CTL_ELEM_TYPE_BOOLEAN || - type == SNDRV_CTL_ELEM_TYPE_INTEGER) { + if (type == (__force int)SNDRV_CTL_ELEM_TYPE_BOOLEAN || + type == (__force int)SNDRV_CTL_ELEM_TYPE_INTEGER) { for (i = 0; i < count; i++) { s32 __user *intp = valuep; int val; @@ -244,7 +244,7 @@ static int copy_ctl_value_from_user(struct snd_card *card, data->value.integer.value[i] = val; } } else { - size = get_elem_size(type, count); + size = get_elem_size((__force snd_ctl_elem_type_t)type, count); if (size < 0) { dev_err(card->dev, "snd_ioctl32_ctl_elem_value: unknown type %d\n", type); return -EINVAL; @@ -267,8 +267,8 @@ static int copy_ctl_value_to_user(void __user *userdata, struct snd_ctl_elem_value32 __user *data32 = userdata; int i, size; - if (type == SNDRV_CTL_ELEM_TYPE_BOOLEAN || - type == SNDRV_CTL_ELEM_TYPE_INTEGER) { + if (type == (__force int)SNDRV_CTL_ELEM_TYPE_BOOLEAN || + type == (__force int)SNDRV_CTL_ELEM_TYPE_INTEGER) { for (i = 0; i < count; i++) { s32 __user *intp = valuep; int val; @@ -277,7 +277,7 @@ static int copy_ctl_value_to_user(void __user *userdata, return -EFAULT; } } else { - size = get_elem_size(type, count); + size = get_elem_size((__force snd_ctl_elem_type_t)type, count); if (copy_to_user(valuep, data->value.bytes.data, size)) return -EFAULT; } -- 2.34.1

2 1

[PATCH] ALSA: hda: Add NVIDIA codec IDs a3 through a7 to patch table
by Nikhil Mahale 17 May '23

17 May '23

These IDs are for AD102, AD103, AD104, AD106, and AD107 gpus with audio functions that are largely similar to the existing ones. Tested audio using gnome-settings, over HDMI, DP-SST and DP-MST connections on AD106 gpu. Signed-off-by: Nikhil Mahale <nmahale(a)nvidia.com> --- sound/pci/hda/patch_hdmi.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 64a944016c78..5c0b1a09fd57 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -4589,6 +4589,11 @@ HDA_CODEC_ENTRY(0x10de009d, "GPU 9d HDMI/DP", patch_nvhdmi), HDA_CODEC_ENTRY(0x10de009e, "GPU 9e HDMI/DP", patch_nvhdmi), HDA_CODEC_ENTRY(0x10de009f, "GPU 9f HDMI/DP", patch_nvhdmi), HDA_CODEC_ENTRY(0x10de00a0, "GPU a0 HDMI/DP", patch_nvhdmi), +HDA_CODEC_ENTRY(0x10de00a3, "GPU a3 HDMI/DP", patch_nvhdmi), +HDA_CODEC_ENTRY(0x10de00a4, "GPU a4 HDMI/DP", patch_nvhdmi), +HDA_CODEC_ENTRY(0x10de00a5, "GPU a5 HDMI/DP", patch_nvhdmi), +HDA_CODEC_ENTRY(0x10de00a6, "GPU a6 HDMI/DP", patch_nvhdmi), +HDA_CODEC_ENTRY(0x10de00a7, "GPU a7 HDMI/DP", patch_nvhdmi), HDA_CODEC_ENTRY(0x10de8001, "MCP73 HDMI", patch_nvhdmi_2ch), HDA_CODEC_ENTRY(0x10de8067, "MCP67/68 HDMI", patch_nvhdmi_2ch), HDA_CODEC_ENTRY(0x67663d82, "Arise 82 HDMI/DP", patch_gf_hdmi), -- 2.35.3

2 1

[PATCH 1/2] ALSA: cs46xx: mark snd_cs46xx_download_image as static
by Arnd Bergmann 17 May '23

17 May '23

From: Arnd Bergmann <arnd(a)arndb.de> snd_cs46xx_download_image() was originally called from dsp_spos.c, but is now local to cs46xx_lib.c. Mark it as 'static' to avoid a warning about it lacking a declaration, and '__maybe_unused' to avoid a warning about it being unused when CONFIG_SND_CS46XX_NEW_DSP is disabled: sound/pci/cs46xx/cs46xx_lib.c:534:5: error: no previous prototype for 'snd_cs46xx_download_image' Fixes: 89f157d9e6bf ("[ALSA] cs46xx - Fix PM resume") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/pci/cs46xx/cs46xx_lib.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c index 62f45847b351..7d882b33d45e 100644 --- a/sound/pci/cs46xx/cs46xx_lib.c +++ b/sound/pci/cs46xx/cs46xx_lib.c @@ -531,7 +531,7 @@ static int load_firmware(struct snd_cs46xx *chip) return err; } -int snd_cs46xx_download_image(struct snd_cs46xx *chip) +static __maybe_unused int snd_cs46xx_download_image(struct snd_cs46xx *chip) { int idx, err; unsigned int offset = 0; -- 2.39.2

2 3

[PATCH] ASoC: tegra: tegra210_adx: fix snd_pcm_format_t type
by Min-Hua Chen 17 May '23

17 May '23

use snd_pcm_format_t instead of unsigned int to fix the following sparse warnings: sound/soc/tegra/tegra210_adx.c:125:14: sparse: warning: restricted snd_pcm_format_t degrades to integer sound/soc/tegra/tegra210_adx.c:128:14: sparse: warning: restricted snd_pcm_format_t degrades to integer sound/soc/tegra/tegra210_adx.c:131:14: sparse: warning: restricted snd_pcm_format_t degrades to integer Signed-off-by: Min-Hua Chen <minhuadotchen(a)gmail.com> --- sound/soc/tegra/tegra210_adx.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/tegra/tegra210_adx.c b/sound/soc/tegra/tegra210_adx.c index 41117c1d61fb..bd0b10c70c4c 100644 --- a/sound/soc/tegra/tegra210_adx.c +++ b/sound/soc/tegra/tegra210_adx.c @@ -109,7 +109,7 @@ static int __maybe_unused tegra210_adx_runtime_resume(struct device *dev) static int tegra210_adx_set_audio_cif(struct snd_soc_dai *dai, unsigned int channels, - unsigned int format, + snd_pcm_format_t format, unsigned int reg) { struct tegra210_adx *adx = snd_soc_dai_get_drvdata(dai); -- 2.34.1

2 1

[PATCH] ASoC: dt-bindings: nau8315: Convert to dtschema
by David Lin 17 May '23

17 May '23

Convert the NAU8315 audio CODEC bindings to DT schema. Signed-off-by: David Lin <CTLIN0(a)nuvoton.com> --- .../devicetree/bindings/sound/nau8315.txt | 24 ---------- .../bindings/sound/nuvoton,nau8315.yaml | 44 +++++++++++++++++++ 2 files changed, 44 insertions(+), 24 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/nau8315.txt create mode 100644 Documentation/devicetree/bindings/sound/nuvoton,nau8315.yaml diff --git a/Documentation/devicetree/bindings/sound/nau8315.txt b/Documentation/devicetree/bindings/sound/nau8315.txt deleted file mode 100644 index 1cd94517d45e..000000000000 --- a/Documentation/devicetree/bindings/sound/nau8315.txt +++ /dev/null @@ -1,24 +0,0 @@ -Nuvoton NAU8315 Mono Class-D Amplifier - -Required properties: -- compatible : "nuvoton,nau8315" - "nuvoton,nau8318" - -Optional properties: -- enable-gpios : GPIO specifier for the chip's device enable input(EN) pin. - If this option is not specified then driver does not manage - the pin state (e.g. chip is always on). - -Example: - -#include <dt-bindings/gpio/gpio.h> - -nau8315 { - compatible = "nuvoton,nau8315"; - enable-gpios = <&gpio1 5 GPIO_ACTIVE_HIGH>; -}; - -nau8318 { - compatible = "nuvoton,nau8318"; - enable-gpios = <&gpio1 5 GPIO_ACTIVE_HIGH>; -}; diff --git a/Documentation/devicetree/bindings/sound/nuvoton,nau8315.yaml b/Documentation/devicetree/bindings/sound/nuvoton,nau8315.yaml new file mode 100644 index 000000000000..24006e9dc501 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nuvoton,nau8315.yaml @@ -0,0 +1,44 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nuvoton,nau8315.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NAU8315/NAU8318 Mono Class-D Amplifier + +maintainers: + - David Lin <CTLIN0(a)nuvoton.com> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - nuvoton,nau8315 + - nuvoton,nau8318 + + '#sound-dai-cells': + const: 0 + + enable-gpios: + maxItems: 1 + description: + GPIO specifier for the chip's device enable input(EN) pin. + If this option is not specified then driver does not manage + the pin state (e.g. chip is always on). + +required: + - compatible + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + + codec { + compatible = "nuvoton,nau8315"; + #sound-dai-cells = <0>; + enable-gpios = <&gpio1 5 GPIO_ACTIVE_HIGH>; + }; -- 2.25.1

4 4

[PATCH] ASoC: dt-bindings: nau8810: Convert to dtschema
by David Lin 17 May '23

17 May '23

Convert the NAU8810 audio CODEC bindings to DT schema. Signed-off-by: David Lin <CTLIN0(a)nuvoton.com> --- .../devicetree/bindings/sound/nau8810.txt | 17 ------- .../bindings/sound/nuvoton,nau8810.yaml | 45 +++++++++++++++++++ 2 files changed, 45 insertions(+), 17 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/nau8810.txt create mode 100644 Documentation/devicetree/bindings/sound/nuvoton,nau8810.yaml diff --git a/Documentation/devicetree/bindings/sound/nau8810.txt b/Documentation/devicetree/bindings/sound/nau8810.txt deleted file mode 100644 index 7deaa452b200..000000000000 --- a/Documentation/devicetree/bindings/sound/nau8810.txt +++ /dev/null @@ -1,17 +0,0 @@ -NAU8810/NAU8812/NAU8814 audio CODEC - -This device supports I2C only. - -Required properties: - - - compatible : One of "nuvoton,nau8810" or "nuvoton,nau8812" or - "nuvoton,nau8814" - - - reg : the I2C address of the device. - -Example: - -codec: nau8810@1a { - compatible = "nuvoton,nau8810"; - reg = <0x1a>; -}; diff --git a/Documentation/devicetree/bindings/sound/nuvoton,nau8810.yaml b/Documentation/devicetree/bindings/sound/nuvoton,nau8810.yaml new file mode 100644 index 000000000000..d9696f6c75ed --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nuvoton,nau8810.yaml @@ -0,0 +1,45 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nuvoton,nau8810.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NAU8810/NAU8812/NAU8814 audio CODEC + +maintainers: + - David Lin <CTLIN0(a)nuvoton.com> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - nuvoton,nau8810 + - nuvoton,nau8812 + - nuvoton,nau8814 + + reg: + maxItems: 1 + + '#sound-dai-cells': + const: 0 + +required: + - compatible + - reg + +unevaluatedProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + codec@1a { + #sound-dai-cells = <0>; + compatible = "nuvoton,nau8810"; + reg = <0x1a>; + }; + }; -- 2.25.1

4 3

[PATCH] ASoC: dt-bindings: ssm2518: Convert to dtschema
by Marek Vasut 17 May '23

17 May '23

Convert the ADI SSM2518 audio CODEC bindings to DT schema. Signed-off-by: Marek Vasut <marex(a)denx.de> --- Cc: Conor Dooley <conor+dt(a)kernel.org> Cc: Krzysztof Kozlowski <krzysztof.kozlowski+dt(a)linaro.org> Cc: Lars-Peter Clausen <lars(a)metafoo.de> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Rob Herring <robh+dt(a)kernel.org> Cc: alsa-devel(a)alsa-project.org Cc: devicetree(a)vger.kernel.org --- .../bindings/sound/adi,ssm2518.yaml | 47 +++++++++++++++++++ .../devicetree/bindings/sound/ssm2518.txt | 20 -------- 2 files changed, 47 insertions(+), 20 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/adi,ssm2518.yaml delete mode 100644 Documentation/devicetree/bindings/sound/ssm2518.txt diff --git a/Documentation/devicetree/bindings/sound/adi,ssm2518.yaml b/Documentation/devicetree/bindings/sound/adi,ssm2518.yaml new file mode 100644 index 0000000000000..f3f32540779c6 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/adi,ssm2518.yaml @@ -0,0 +1,47 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/adi,ssm2518.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Analog Devices SSM2518 audio amplifier + +maintainers: + - Lars-Peter Clausen <lars(a)metafoo.de> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: adi,ssm2518 + + reg: + maxItems: 1 + description: | + I2C address of the device. This will either be 0x34 (ADDR pin low) + or 0x35 (ADDR pin high) + + gpios: + maxItems: 1 + description: | + GPIO connected to the nSD pin. If the property is not present + it is assumed that the nSD pin is hardwired to always on. + +required: + - compatible + - reg + +unevaluatedProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + codec@34 { + compatible = "adi,ssm2518"; + reg = <0x34>; + gpios = <&gpio 5 0>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/ssm2518.txt b/Documentation/devicetree/bindings/sound/ssm2518.txt deleted file mode 100644 index 59381a778c797..0000000000000 --- a/Documentation/devicetree/bindings/sound/ssm2518.txt +++ /dev/null @@ -1,20 +0,0 @@ -SSM2518 audio amplifier - -This device supports I2C only. - -Required properties: - - compatible : Must be "adi,ssm2518" - - reg : the I2C address of the device. This will either be 0x34 (ADDR pin low) - or 0x35 (ADDR pin high) - -Optional properties: - - gpios : GPIO connected to the nSD pin. If the property is not present it is - assumed that the nSD pin is hardwired to always on. - -Example: - - ssm2518: ssm2518@34 { - compatible = "adi,ssm2518"; - reg = <0x34>; - gpios = <&gpio 5 0>; - }; -- 2.39.2

3 3

[RESEND 0/4] ASoC: mt6359: kselftest fix and driver extension
by Trevor Wu 17 May '23

17 May '23

The patch series includes a kselftest fix and changes for extending driver capability to support more use cases. Trevor Wu (4): ASoC: mediatek: mt6359: add supply for MTKAIF ASoC: mediatek: mt6359: fix kselftest error of playback gain ASoC: mediatek: mt6359: add mtkaif gpio setting ASoC: mediatek: mt6359: update route for lineout mux sound/soc/codecs/mt6359.c | 152 ++++++++++++++++++++++++++++++++++---- 1 file changed, 137 insertions(+), 15 deletions(-) -- 2.18.0

1 4

[bug report] ASoC: SOF: ipc4-pcm: reset all pipelines during FE DAI hw_free
by Dan Carpenter 17 May '23

17 May '23

Hello Ranjani Sridharan, The patch 225f37b578a9: "ASoC: SOF: ipc4-pcm: reset all pipelines during FE DAI hw_free" from May 15, 2023, leads to the following Smatch static checker warning: sound/soc/sof/intel/hda-dai-ops.c:215 hda_ipc4_pre_trigger() warn: inconsistent returns '&ipc4_data->pipeline_state_mutex'. sound/soc/sof/intel/hda-dai-ops.c 178 static int hda_ipc4_pre_trigger(struct snd_sof_dev *sdev, struct snd_soc_dai *cpu_dai, 179 struct snd_pcm_substream *substream, int cmd) 180 { 181 struct sof_ipc4_fw_data *ipc4_data = sdev->private; 182 struct snd_sof_widget *pipe_widget; 183 struct sof_ipc4_pipeline *pipeline; 184 struct snd_sof_widget *swidget; 185 struct snd_soc_dapm_widget *w; 186 int ret; 187 188 w = snd_soc_dai_get_widget(cpu_dai, substream->stream); 189 swidget = w->dobj.private; 190 pipe_widget = swidget->spipe->pipe_widget; 191 pipeline = pipe_widget->private; 192 193 mutex_lock(&ipc4_data->pipeline_state_mutex); 194 195 switch (cmd) { 196 case SNDRV_PCM_TRIGGER_START: 197 case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: 198 break; 199 case SNDRV_PCM_TRIGGER_PAUSE_PUSH: 200 case SNDRV_PCM_TRIGGER_SUSPEND: 201 case SNDRV_PCM_TRIGGER_STOP: 202 ret = sof_ipc4_set_pipeline_state(sdev, pipe_widget->instance_id, 203 SOF_IPC4_PIPE_PAUSED); 204 if (ret < 0) 205 goto out; Originally we propagated the error code back on this path, but now we are returning 0. Was this intentional? 206 207 pipeline->state = SOF_IPC4_PIPE_PAUSED; 208 break; 209 default: 210 dev_err(sdev->dev, "unknown trigger command %d\n", cmd); 211 return -EINVAL; Needs an unlock before returning. 212 } 213 out: 214 mutex_unlock(&ipc4_data->pipeline_state_mutex); --> 215 return 0; 216 } regards, dan carpenter

1 0

[PATCH] ALSA: hda: Fix Oops by 9.1 surround channel names
by Takashi Iwai 16 May '23

16 May '23

get_line_out_pfx() may trigger an Oops by overflowing the static array with more than 8 channels. This was reported for MacBookPro 12,1 with Cirrus codec. As a workaround, extend for the 9.1 channels and also fix the potential Oops by unifying the code paths accessing the same array with the proper size check. Reported-by: Olliver Schinagl <oliver(a)schinagl.nl> Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/64d95eb0-dbdb-cff8-a8b1-988dc22b24cd@schinagl.nl Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_generic.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index fc114e522480..dbf7aa88e0e3 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -1155,8 +1155,8 @@ static bool path_has_mixer(struct hda_codec *codec, int path_idx, int ctl_type) return path && path->ctls[ctl_type]; } -static const char * const channel_name[4] = { - "Front", "Surround", "CLFE", "Side" +static const char * const channel_name[] = { + "Front", "Surround", "CLFE", "Side", "Back", }; /* give some appropriate ctl name prefix for the given line out channel */ @@ -1182,7 +1182,7 @@ static const char *get_line_out_pfx(struct hda_codec *codec, int ch, /* multi-io channels */ if (ch >= cfg->line_outs) - return channel_name[ch]; + goto fixed_name; switch (cfg->line_out_type) { case AUTO_PIN_SPEAKER_OUT: @@ -1234,6 +1234,7 @@ static const char *get_line_out_pfx(struct hda_codec *codec, int ch, if (cfg->line_outs == 1 && !spec->multi_ios) return "Line Out"; + fixed_name: if (ch >= ARRAY_SIZE(channel_name)) { snd_BUG(); return "PCM"; -- 2.35.3

1 0

[GIT PULL] ASoC fixes for v6.4-rc2
by Mark Brown 16 May '23

16 May '23

The following changes since commit ac9a78681b921877518763ba0e89202254349d1b: Linux 6.4-rc1 (2023-05-07 13:34:35 -0700) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.4-rc2 for you to fetch changes up to 9be0b3a0074a61df1c94c37faea35ec8b9ea130b: ASoC: SOF: Intel: hda-mlink: fixes and extensions (2023-05-16 00:13:09 +0900) ---------------------------------------------------------------- ASoC: Fixes for v6.4 More fixes that came in since the merge window, the bulk of which are for the SOF code, I suspect as a result of the wide usage, active development and large code size rather than huge quality problems. There's also a couple of MAINTAINERS updates and some new device quirks. ---------------------------------------------------------------- Aidan MacDonald (1): ASoC: jz4740-i2s: Make I2S divider calculations more robust Douglas Anderson (1): ASoC: mediatek: mt8186: Fix use-after-free in driver remove path Jeremy Soller (1): ASoC: amd: yc: Add DMI entry to support System76 Pangolin 12 Kai Vehmanen (1): ASoC: SOF: pm: save io region state in case of errors in resume Krzysztof Kozlowski (1): ASoC: MAINTAINERS: drop Krzysztof Kozlowski from Samsung audio Lucas Tanure (1): MAINTAINERS: Remove self from Cirrus Codec drivers Mark Brown (2): ASoC: SOF: Various runtime pm fixes, improvements ASoC: SOF: Intel: hda-mlink: fixes and extensions Martin Povišer (1): ASoC: dt-bindings: Adjust #sound-dai-cells on TI's single-DAI codecs Maxim Kochetkov (1): ASoC: dwc: limit the number of overrun messages Paweł Anikiel (1): ASoC: ssm2602: Add workaround for playback distortions Peter Ujfalusi (1): ASoC: SOF: ipc3-topology: Make sure that only one cmd is sent in dai_config Pierre-Louis Bossart (9): ASoC: SOF: Intel: hda-mlink: fix sublink refcounting ASoC: SOF: Intel: hda-mlink: add helper to get SoundWire hlink ASoC: SOF: Intel: hda-mlink: fix base_ptr computation ASoC: SOF: Intel: hda-mlink: use 'ml_addr' parameter consistently ASoC: SOF: Intel: hda-mlink: initialize instance_offset member ASoC: SOF: Intel: hda-mlink: add helper to program SoundWire PCMSyCM registers ASoC: SOF: debug: conditionally bump runtime_pm counter on exceptions ASoC: SOF: pcm: fix pm_runtime imbalance in error handling ASoC: SOF: sof-client-probes: fix pm_runtime imbalance in error handling Ranjani Sridharan (3): ASoC: SOF: topology: Fix logic for copying tuples ASoC: SOF: Separate the tokens for input and output pin index ASoC: SOF: topology: Fix tuples array allocation Shengjiu Wang (1): ASoC: fsl_micfil: Fix error handler with pm_runtime_enable Simon Trimmer (1): ASoC: cs35l56: Prevent unbalanced pm_runtime in dsp_work() on SoundWire V sujith kumar Reddy (1): ASoC: SOF: amd: Fix NULL pointer crash in acp_sof_ipc_msg_data function .../devicetree/bindings/sound/tas2562.yaml | 6 +- .../devicetree/bindings/sound/tas2770.yaml | 6 +- .../devicetree/bindings/sound/tas27xx.yaml | 6 +- MAINTAINERS | 4 +- include/sound/hda-mlink.h | 14 ++++ include/uapi/sound/sof/tokens.h | 3 +- sound/soc/amd/yc/acp6x-mach.c | 7 ++ sound/soc/codecs/cs35l56.c | 3 +- sound/soc/codecs/ssm2602.c | 15 ++++ sound/soc/dwc/dwc-i2s.c | 4 +- sound/soc/fsl/fsl_micfil.c | 14 +++- sound/soc/jz4740/jz4740-i2s.c | 54 +++++++++++- sound/soc/mediatek/mt8186/mt8186-afe-clk.c | 6 -- sound/soc/mediatek/mt8186/mt8186-afe-clk.h | 1 - sound/soc/mediatek/mt8186/mt8186-afe-pcm.c | 4 - sound/soc/mediatek/mt8186/mt8186-audsys-clk.c | 46 ++++++----- sound/soc/mediatek/mt8186/mt8186-audsys-clk.h | 1 - sound/soc/sof/amd/acp-ipc.c | 7 +- sound/soc/sof/debug.c | 4 +- sound/soc/sof/intel/hda-mlink.c | 96 ++++++++++++++++++++-- sound/soc/sof/ipc3-topology.c | 7 +- sound/soc/sof/ipc4-topology.c | 4 +- sound/soc/sof/pcm.c | 11 +-- sound/soc/sof/pm.c | 14 +++- sound/soc/sof/sof-client-probes.c | 14 ++-- sound/soc/sof/topology.c | 6 +- 26 files changed, 276 insertions(+), 81 deletions(-)

2 1

[RESEND PATCH 0/2] Improve support for sof_ipc{3|4}_bytes_ext_put
by Daniel Baluta 16 May '23

16 May '23

From: Daniel Baluta <daniel.baluta(a)nxp.com> This patch series provides better handling of cases where sending a data blob to FW results in a validation error. In this case we restore to the last good known value instead of keeping the data that firwmare rejected. Paul Olaru (2): ASoC: sof: Improve sof_ipc3_bytes_ext_put function ASoC: sof: Improve sof_ipc4_bytes_ext_put function sound/soc/sof/ipc3-control.c | 54 ++++++++++++++++++++++++++++++++---- sound/soc/sof/ipc4-control.c | 39 +++++++++++++++++++++++--- sound/soc/sof/sof-audio.h | 1 + 3 files changed, 84 insertions(+), 10 deletions(-) -- 2.25.1

2 3

[PATCH v2] ASoC: dt-bindings: nau8824: Convert to dtschema
by David Lin 16 May '23

16 May '23

Convert the NAU8824 audio CODEC bindings to DT schema. Signed-off-by: David Lin <CTLIN0(a)nuvoton.com> --- Change: V1 -> V2: - Correct the maximun and maxItems from 4 to 8 for sar-threshold-num and sar-threshold, respectively. .../devicetree/bindings/sound/nau8824.txt | 88 --------- .../bindings/sound/nuvoton,nau8824.yaml | 182 ++++++++++++++++++ 2 files changed, 182 insertions(+), 88 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/nau8824.txt create mode 100644 Documentation/devicetree/bindings/sound/nuvoton,nau8824.yaml diff --git a/Documentation/devicetree/bindings/sound/nau8824.txt b/Documentation/devicetree/bindings/sound/nau8824.txt deleted file mode 100644 index e0058b97e49a..000000000000 --- a/Documentation/devicetree/bindings/sound/nau8824.txt +++ /dev/null @@ -1,88 +0,0 @@ -Nuvoton NAU8824 audio codec - -This device supports I2C only. - -Required properties: - - compatible : Must be "nuvoton,nau8824" - - - reg : the I2C address of the device. This is either 0x1a (CSB=0) or 0x1b (CSB=1). - -Optional properties: - - nuvoton,jkdet-polarity: JKDET pin polarity. 0 - active high, 1 - active low. - - - nuvoton,vref-impedance: VREF Impedance selection - 0 - Open - 1 - 25 kOhm - 2 - 125 kOhm - 3 - 2.5 kOhm - - - nuvoton,micbias-voltage: Micbias voltage level. - 0 - VDDA - 1 - VDDA - 2 - VDDA * 1.1 - 3 - VDDA * 1.2 - 4 - VDDA * 1.3 - 5 - VDDA * 1.4 - 6 - VDDA * 1.53 - 7 - VDDA * 1.53 - - - nuvoton,sar-threshold-num: Number of buttons supported - - nuvoton,sar-threshold: Impedance threshold for each button. Array that contains up to 8 buttons configuration. SAR value is calculated as - SAR = 255 * MICBIAS / SAR_VOLTAGE * R / (2000 + R) - where MICBIAS is configured by 'nuvoton,micbias-voltage', SAR_VOLTAGE is configured by 'nuvoton,sar-voltage', R - button impedance. - Refer datasheet section 10.2 for more information about threshold calculation. - - - nuvoton,sar-hysteresis: Button impedance measurement hysteresis. - - - nuvoton,sar-voltage: Reference voltage for button impedance measurement. - 0 - VDDA - 1 - VDDA - 2 - VDDA * 1.1 - 3 - VDDA * 1.2 - 4 - VDDA * 1.3 - 5 - VDDA * 1.4 - 6 - VDDA * 1.53 - 7 - VDDA * 1.53 - - - nuvoton,sar-compare-time: SAR compare time - 0 - 500 ns - 1 - 1 us - 2 - 2 us - 3 - 4 us - - - nuvoton,sar-sampling-time: SAR sampling time - 0 - 2 us - 1 - 4 us - 2 - 8 us - 3 - 16 us - - - nuvoton,short-key-debounce: Button short key press debounce time. - 0 - 30 ms - 1 - 50 ms - 2 - 100 ms - - - nuvoton,jack-eject-debounce: Jack ejection debounce time. - 0 - 0 ms - 1 - 1 ms - 2 - 10 ms - - -Example: - - headset: nau8824@1a { - compatible = "nuvoton,nau8824"; - reg = <0x1a>; - interrupt-parent = <&gpio>; - interrupts = <TEGRA_GPIO(E, 6) IRQ_TYPE_LEVEL_LOW>; - nuvoton,vref-impedance = <2>; - nuvoton,micbias-voltage = <6>; - // Setup 4 buttons impedance according to Android specification - nuvoton,sar-threshold-num = <4>; - nuvoton,sar-threshold = <0xc 0x1e 0x38 0x60>; - nuvoton,sar-hysteresis = <0>; - nuvoton,sar-voltage = <6>; - nuvoton,sar-compare-time = <1>; - nuvoton,sar-sampling-time = <1>; - nuvoton,short-key-debounce = <0>; - nuvoton,jack-eject-debounce = <1>; - }; diff --git a/Documentation/devicetree/bindings/sound/nuvoton,nau8824.yaml b/Documentation/devicetree/bindings/sound/nuvoton,nau8824.yaml new file mode 100644 index 000000000000..3dbf438c3841 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nuvoton,nau8824.yaml @@ -0,0 +1,182 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nuvoton,nau8824.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NAU8824 audio CODEC + +maintainers: + - John Hsu <KCHSU0(a)nuvoton.com> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - nuvoton,nau8824 + + reg: + maxItems: 1 + + '#sound-dai-cells': + const: 0 + + interrupts: + maxItems: 1 + + nuvoton,jkdet-polarity: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + JKDET pin polarity. + enum: + - 0 # active high + - 1 # active low + default: 1 + + nuvoton,vref-impedance: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + VREF Impedance selection. + enum: + - 0 # Open + - 1 # 25 kOhm + - 2 # 125 kOhm + - 3 # 2.5 kOhm + default: 2 + + nuvoton,micbias-voltage: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + Micbias voltage level. + enum: + - 0 # VDDA + - 1 # VDDA + - 2 # VDDA * 1.1 + - 3 # VDDA * 1.2 + - 4 # VDDA * 1.3 + - 5 # VDDA * 1.4 + - 6 # VDDA * 1.53 + - 7 # VDDA * 1.53 + default: 6 + + nuvoton,sar-threshold-num: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + Number of buttons supported. + minimum: 1 + maximum: 8 + default: 4 + + nuvoton,sar-threshold: + $ref: /schemas/types.yaml#/definitions/uint32-array + description: + Impedance threshold for each button. Array that contains up to 8 buttons + configuration. SAR value is calculated as + SAR = 255 * MICBIAS / SAR_VOLTAGE * R / (2000 + R) where MICBIAS is + configured by 'nuvoton,micbias-voltage', SAR_VOLTAGE is configured by + 'nuvoton,sar-voltage', R - button impedance. + Refer datasheet section 10.2 for more information about threshold + calculation. + minItems: 1 + maxItems: 8 + items: + minimum: 0 + maximum: 255 + + nuvoton,sar-hysteresis: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + Button impedance measurement hysteresis. + default: 0 + + nuvoton,sar-voltage: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + Reference voltage for button impedance measurement. + enum: + - 0 # VDDA + - 1 # VDDA + - 2 # VDDA * 1.1 + - 3 # VDDA * 1.2 + - 4 # VDDA * 1.3 + - 5 # VDDA * 1.4 + - 6 # VDDA * 1.53 + - 7 # VDDA * 1.53 + default: 6 + + nuvoton,sar-compare-time: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + SAR compare time. + enum: + - 0 # 500ns + - 1 # 1us + - 2 # 2us + - 3 # 4us + default: 1 + + nuvoton,sar-sampling-time: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + SAR sampling time. + enum: + - 0 # 2us + - 1 # 4us + - 2 # 8us + - 3 # 16us + default: 1 + + nuvoton,short-key-debounce: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + Button short key press debounce time. + enum: + - 0 # 30 ms + - 1 # 50 ms + - 2 # 100 ms + default: 0 + + nuvoton,jack-eject-debounce: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + Jack ejection debounce time. + enum: + - 0 # 0 ms + - 1 # 1 ms + - 2 # 10 ms + default: 1 + +required: + - compatible + - reg + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + #include <dt-bindings/interrupt-controller/irq.h> + i2c { + #address-cells = <1>; + #size-cells = <0>; + codec@1a { + #sound-dai-cells = <0>; + compatible = "nuvoton,nau8824"; + reg = <0x1a>; + interrupt-parent = <&gpio>; + interrupts = <38 IRQ_TYPE_LEVEL_LOW>; + nuvoton,vref-impedance = <2>; + nuvoton,micbias-voltage = <6>; + nuvoton,sar-threshold-num = <4>; + // Setup 4 buttons impedance according to Android specification + nuvoton,sar-threshold = <0xc 0x1e 0x38 0x60>; + nuvoton,sar-hysteresis = <0>; + nuvoton,sar-voltage = <6>; + nuvoton,sar-compare-time = <1>; + nuvoton,sar-sampling-time = <1>; + nuvoton,short-key-debounce = <0>; + nuvoton,jack-eject-debounce = <1>; + }; + }; -- 2.25.1

3 2

[PATCH v4 30/41] sound: add HAS_IOPORT dependencies
by Niklas Schnelle 16 May '23

16 May '23

In a future patch HAS_IOPORT=n will result in inb()/outb() and friends not being declared. We thus need to add HAS_IOPORT as dependency for those drivers using them. Co-developed-by: Arnd Bergmann <arnd(a)kernel.org> Signed-off-by: Arnd Bergmann <arnd(a)kernel.org> Signed-off-by: Niklas Schnelle <schnelle(a)linux.ibm.com> --- Note: The HAS_IOPORT Kconfig option was added in v6.4-rc1 so per-subsystem patches may be applied independently sound/drivers/Kconfig | 3 +++ sound/isa/Kconfig | 1 + sound/pci/Kconfig | 45 ++++++++++++++++++++++++++++++++----------- sound/pcmcia/Kconfig | 1 + 4 files changed, 39 insertions(+), 11 deletions(-) diff --git a/sound/drivers/Kconfig b/sound/drivers/Kconfig index be3009746f3a..864991d8776d 100644 --- a/sound/drivers/Kconfig +++ b/sound/drivers/Kconfig @@ -128,6 +128,7 @@ config SND_VIRMIDI config SND_MTPAV tristate "MOTU MidiTimePiece AV multiport MIDI" + depends on HAS_IOPORT select SND_RAWMIDI help To use a MOTU MidiTimePiece AV multiport MIDI adapter @@ -152,6 +153,7 @@ config SND_MTS64 config SND_SERIAL_U16550 tristate "UART16550 serial MIDI driver" + depends on HAS_IOPORT select SND_RAWMIDI help To include support for MIDI serial port interfaces, say Y here @@ -185,6 +187,7 @@ config SND_SERIAL_GENERIC config SND_MPU401 tristate "Generic MPU-401 UART driver" + depends on HAS_IOPORT select SND_MPU401_UART help Say Y here to include support for MIDI ports compatible with diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig index 6ffa48dd5983..f8159179e38d 100644 --- a/sound/isa/Kconfig +++ b/sound/isa/Kconfig @@ -23,6 +23,7 @@ menuconfig SND_ISA bool "ISA sound devices" depends on ISA || COMPILE_TEST depends on ISA_DMA_API + depends on HAS_IOPORT default y help Support for sound devices connected via the ISA bus. diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index 861958451ef5..787868c9e91b 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -26,7 +26,7 @@ config SND_ALS300 select SND_PCM select SND_AC97_CODEC select SND_OPL3_LIB - depends on ZONE_DMA + depends on ZONE_DMA && HAS_IOPORT help Say 'Y' or 'M' to include support for Avance Logic ALS300/ALS300+ @@ -36,6 +36,7 @@ config SND_ALS300 config SND_ALS4000 tristate "Avance Logic ALS4000" depends on ISA_DMA_API + depends on HAS_IOPORT select SND_OPL3_LIB select SND_MPU401_UART select SND_PCM @@ -51,7 +52,7 @@ config SND_ALI5451 tristate "ALi M5451 PCI Audio Controller" select SND_MPU401_UART select SND_AC97_CODEC - depends on ZONE_DMA + depends on ZONE_DMA && HAS_IOPORT help Say Y here to include support for the integrated AC97 sound device on motherboards using the ALi M5451 Audio Controller @@ -96,6 +97,7 @@ config SND_ATIIXP_MODEM config SND_AU8810 tristate "Aureal Advantage" + depends on HAS_IOPORT select SND_MPU401_UART select SND_AC97_CODEC help @@ -110,6 +112,7 @@ config SND_AU8810 config SND_AU8820 tristate "Aureal Vortex" + depends on HAS_IOPORT select SND_MPU401_UART select SND_AC97_CODEC help @@ -123,6 +126,7 @@ config SND_AU8820 config SND_AU8830 tristate "Aureal Vortex 2" + depends on HAS_IOPORT select SND_MPU401_UART select SND_AC97_CODEC help @@ -157,7 +161,7 @@ config SND_AZT3328 select SND_RAWMIDI select SND_AC97_CODEC select SND_TIMER - depends on ZONE_DMA + depends on ZONE_DMA && HAS_IOPORT help Say Y here to include support for Aztech AZF3328 (PCI168) soundcards. @@ -193,6 +197,7 @@ config SND_BT87X_OVERCLOCK config SND_CA0106 tristate "SB Audigy LS / Live 24bit" + depends on HAS_IOPORT select SND_AC97_CODEC select SND_RAWMIDI select SND_VMASTER @@ -205,6 +210,7 @@ config SND_CA0106 config SND_CMIPCI tristate "C-Media 8338, 8738, 8768, 8770" + depends on HAS_IOPORT select SND_OPL3_LIB select SND_MPU401_UART select SND_PCM @@ -221,6 +227,7 @@ config SND_OXYGEN_LIB config SND_OXYGEN tristate "C-Media 8786, 8787, 8788 (Oxygen)" + depends on HAS_IOPORT select SND_OXYGEN_LIB select SND_PCM select SND_MPU401_UART @@ -246,6 +253,7 @@ config SND_OXYGEN config SND_CS4281 tristate "Cirrus Logic (Sound Fusion) CS4281" + depends on HAS_IOPORT select SND_OPL3_LIB select SND_RAWMIDI select SND_AC97_CODEC @@ -257,6 +265,7 @@ config SND_CS4281 config SND_CS46XX tristate "Cirrus Logic (Sound Fusion) CS4280/CS461x/CS462x/CS463x" + depends on HAS_IOPORT select SND_RAWMIDI select SND_AC97_CODEC select FW_LOADER @@ -290,6 +299,7 @@ config SND_CS5530 config SND_CS5535AUDIO tristate "CS5535/CS5536 Audio" depends on X86_32 || MIPS || COMPILE_TEST + depends on HAS_IOPORT select SND_PCM select SND_AC97_CODEC help @@ -307,6 +317,7 @@ config SND_CS5535AUDIO config SND_CTXFI tristate "Creative Sound Blaster X-Fi" + depends on HAS_IOPORT select SND_PCM help If you want to use soundcards based on Creative Sound Blastr X-Fi @@ -468,7 +479,7 @@ config SND_EMU10K1 select SND_AC97_CODEC select SND_TIMER select SND_SEQ_DEVICE if SND_SEQUENCER != n - depends on ZONE_DMA + depends on ZONE_DMA && HAS_IOPORT help Say Y to include support for Sound Blaster PCI 512, Live!, Audigy and E-MU APS/0404/1010/1212/1616/1820 soundcards. @@ -491,7 +502,7 @@ config SND_EMU10K1X tristate "Emu10k1X (Dell OEM Version)" select SND_AC97_CODEC select SND_RAWMIDI - depends on ZONE_DMA + depends on ZONE_DMA && HAS_IOPORT help Say Y here to include support for the Dell OEM version of the Sound Blaster Live!. @@ -501,6 +512,7 @@ config SND_EMU10K1X config SND_ENS1370 tristate "(Creative) Ensoniq AudioPCI 1370" + depends on HAS_IOPORT select SND_RAWMIDI select SND_PCM help @@ -511,6 +523,7 @@ config SND_ENS1370 config SND_ENS1371 tristate "(Creative) Ensoniq AudioPCI 1371/1373" + depends on HAS_IOPORT select SND_RAWMIDI select SND_AC97_CODEC help @@ -525,7 +538,7 @@ config SND_ES1938 select SND_OPL3_LIB select SND_MPU401_UART select SND_AC97_CODEC - depends on ZONE_DMA + depends on ZONE_DMA && HAS_IOPORT help Say Y here to include support for soundcards based on ESS Solo-1 (ES1938, ES1946, ES1969) chips. @@ -537,7 +550,7 @@ config SND_ES1968 tristate "ESS ES1968/1978 (Maestro-1/2/2E)" select SND_MPU401_UART select SND_AC97_CODEC - depends on ZONE_DMA + depends on ZONE_DMA && HAS_IOPORT help Say Y here to include support for soundcards based on ESS Maestro 1/2/2E chips. @@ -569,6 +582,7 @@ config SND_ES1968_RADIO config SND_FM801 tristate "ForteMedia FM801" + depends on HAS_IOPORT select SND_OPL3_LIB select SND_MPU401_UART select SND_AC97_CODEC @@ -624,7 +638,7 @@ config SND_ICE1712 select SND_MPU401_UART select SND_AC97_CODEC select BITREVERSE - depends on ZONE_DMA + depends on ZONE_DMA && HAS_IOPORT help Say Y here to include support for soundcards based on the ICE1712 (Envy24) chip. @@ -640,6 +654,7 @@ config SND_ICE1712 config SND_ICE1724 tristate "ICE/VT1724/1720 (Envy24HT/PT)" + depends on HAS_IOPORT select SND_RAWMIDI select SND_AC97_CODEC select SND_VMASTER @@ -712,7 +727,7 @@ config SND_LX6464ES config SND_MAESTRO3 tristate "ESS Allegro/Maestro3" select SND_AC97_CODEC - depends on ZONE_DMA + depends on ZONE_DMA && HAS_IOPORT help Say Y here to include support for soundcards based on ESS Maestro 3 (Allegro) chips. @@ -753,6 +768,7 @@ config SND_NM256 config SND_PCXHR tristate "Digigram PCXHR" + depends on HAS_IOPORT select FW_LOADER select SND_PCM select SND_HWDEP @@ -764,6 +780,7 @@ config SND_PCXHR config SND_RIPTIDE tristate "Conexant Riptide" + depends on HAS_IOPORT select FW_LOADER select SND_OPL3_LIB select SND_MPU401_UART @@ -808,6 +825,7 @@ config SND_RME9652 config SND_SE6X tristate "Studio Evolution SE6X" depends on SND_OXYGEN=n && SND_VIRTUOSO=n # PCI ID conflict + depends on HAS_IOPORT select SND_OXYGEN_LIB select SND_PCM select SND_MPU401_UART @@ -830,7 +848,7 @@ config SND_SONICVIBES select SND_OPL3_LIB select SND_MPU401_UART select SND_AC97_CODEC - depends on ZONE_DMA + depends on ZONE_DMA && HAS_IOPORT help Say Y here to include support for soundcards based on the S3 SonicVibes chip. @@ -842,7 +860,7 @@ config SND_TRIDENT tristate "Trident 4D-Wave DX/NX; SiS 7018" select SND_MPU401_UART select SND_AC97_CODEC - depends on ZONE_DMA + depends on ZONE_DMA && HAS_IOPORT help Say Y here to include support for soundcards based on Trident 4D-Wave DX/NX or SiS 7018 chips. @@ -852,6 +870,7 @@ config SND_TRIDENT config SND_VIA82XX tristate "VIA 82C686A/B, 8233/8235 AC97 Controller" + depends on HAS_IOPORT select SND_MPU401_UART select SND_AC97_CODEC help @@ -863,6 +882,7 @@ config SND_VIA82XX config SND_VIA82XX_MODEM tristate "VIA 82C686A/B, 8233 based Modems" + depends on HAS_IOPORT select SND_AC97_CODEC help Say Y here to include support for the integrated MC97 modem on @@ -873,6 +893,7 @@ config SND_VIA82XX_MODEM config SND_VIRTUOSO tristate "Asus Virtuoso 66/100/200 (Xonar)" + depends on HAS_IOPORT select SND_OXYGEN_LIB select SND_PCM select SND_MPU401_UART @@ -889,6 +910,7 @@ config SND_VIRTUOSO config SND_VX222 tristate "Digigram VX222" + depends on HAS_IOPORT select SND_VX_LIB help Say Y here to include support for Digigram VX222 soundcards. @@ -898,6 +920,7 @@ config SND_VX222 config SND_YMFPCI tristate "Yamaha YMF724/740/744/754" + depends on HAS_IOPORT select SND_OPL3_LIB select SND_MPU401_UART select SND_AC97_CODEC diff --git a/sound/pcmcia/Kconfig b/sound/pcmcia/Kconfig index 10291c43cb18..2e3dfc1ff540 100644 --- a/sound/pcmcia/Kconfig +++ b/sound/pcmcia/Kconfig @@ -4,6 +4,7 @@ menuconfig SND_PCMCIA bool "PCMCIA sound devices" depends on PCMCIA + depends on HAS_IOPORT default y help Support for sound devices connected via the PCMCIA bus. -- 2.39.2

1 0

Apple USB-C dongle cannot output 44100hz
by GitHub issues - edited 16 May '23

16 May '23

alsa-project/alsa-lib issue #322 was edited from tw-hx: I have the same issue as outlined here, with an A2049 Apple USB-C to 3.5mm dongle: https://github.com/mikebrady/shairport-sync/issues/1504 Running Arch Linux with kernel 6.3.1-arch2-1, alsa-lib 1.2.9-1. Pipewire reports that the output is locked to 48000hz when it should be switching to 44100hz, and ALSA reports that this is the only detected supported rate: ``` $ cat /proc/asound/card0/stream0 Playback: Status: Running Interface = 1 Altset = 1 Packet Size = 288 Momentary freq = 48000 Hz (0x30.0000) Interface 1 Altset 1 Format: S24_3LE Channels: 2 Endpoint: 0x02 (2 OUT) (SYNC) Rates: 48000 - 48000 (continuous) Bits: 0 Channel map: FL FR Interface 1 Altset 2 Format: S16_LE Channels: 2 Endpoint: 0x02 (2 OUT) (SYNC) Rates: 48000 - 48000 (continuous) Bits: 0 Channel map: FL FR ``` Thanks if this can be fixed. It seems like a regression between Linux 4.x and 5.x, and halfway through the above thread the user [quantonian](https://github.com/quantonian) added the device to the ALSA quirks table and has posted a patch. Issue URL : https://github.com/alsa-project/alsa-lib/issues/322 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Apple USB-C dongle cannot output 44100hz
by GitHub issues - opened 16 May '23

16 May '23

alsa-project/alsa-lib issue #322 was opened from tw-hx: I have the same issue as outlined here, with an A2049 Apple USB-C to 3.5mm dongle: https://github.com/mikebrady/shairport-sync/issues/1504 Running Arch Linux with kernel 6.3.1-arch2-1, alsa-lib 1.2.9-1. Pipewire reports that the output is locked to 48000hz when it should be switching to 41000hz, and ALSA reports that this is the only detected supported rate: ``` $ cat /proc/asound/card0/stream0 Playback: Status: Running Interface = 1 Altset = 1 Packet Size = 288 Momentary freq = 48000 Hz (0x30.0000) Interface 1 Altset 1 Format: S24_3LE Channels: 2 Endpoint: 0x02 (2 OUT) (SYNC) Rates: 48000 - 48000 (continuous) Bits: 0 Channel map: FL FR Interface 1 Altset 2 Format: S16_LE Channels: 2 Endpoint: 0x02 (2 OUT) (SYNC) Rates: 48000 - 48000 (continuous) Bits: 0 Channel map: FL FR ``` Thanks if this can be fixed. It seems like a regression between Linux 4.x and 5.x, and halfway through the above thread the user [quantonian](https://github.com/quantonian) added the device to the ALSA quirks table and has posted a patch. Issue URL : https://github.com/alsa-project/alsa-lib/issues/322 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH v2 0/7] ALSA: emu10k1: various improvements to the DSP-based mixer code
by Oswald Buddenhagen 16 May '23

16 May '23

Oswald Buddenhagen (7): ALSA: emu10k1: polish audigy GPR allocation ALSA: emu10k1: fix non-zero mixer control defaults in highres mode ALSA: emu10k1: validate min/max values of translated controls ALSA: emu10k1: omit non-applicable mixer controls for E-MU cards ALSA: emu10k1: skip mic capture PCM for cards without AC97 codec ALSA: emu10k1: enable bit-exact playback, part 1: DSP attenuation ALSA: emu10k1: enable bit-exact playback, part 2: voice attenuation Documentation/sound/cards/audigy-mixer.rst | 2 +- Documentation/sound/cards/sb-live-mixer.rst | 2 +- include/sound/emu10k1.h | 9 +- include/uapi/sound/emu10k1.h | 8 +- sound/pci/emu10k1/emu10k1.c | 8 +- sound/pci/emu10k1/emufx.c | 491 +++++++++++--------- sound/pci/emu10k1/emumixer.c | 15 +- sound/pci/emu10k1/emupcm.c | 4 +- 8 files changed, 291 insertions(+), 248 deletions(-) -- 2.40.0.152.g15d061e6df

2 10

Update USB-Audio.conf for ASUS ROG Strix B650E-I Gaming WiFi and ALC4080
by GitHub pull_request - edited 16 May '23

16 May '23

alsa-project/alsa-ucm-conf pull request #315 was edited from Sunsheep: Added configuration for ASUS ROG Strix B650E-I Gaming WiFi and ALC4080 Edit: Added Information from ALSA Information Script [https://alsa-project.org/db/?f=fe7e76a335dc8f1009ecaac9aa341fb68231aee5](ht… ``` Bus 001 Device 002: ID 0b05:1a5c ASUSTek Computer, Inc. USB Audio Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 2.00 bDeviceClass 239 Miscellaneous Device bDeviceSubClass 2 bDeviceProtocol 1 Interface Association bMaxPacketSize0 64 idVendor 0x0b05 ASUSTek Computer, Inc. idProduct 0x1a5c bcdDevice 0.02 iManufacturer 3 Generic iProduct 1 USB Audio iSerial 0 ``` Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/315 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/315.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Update USB-Audio.conf for ASUS ROG Strix B650E-I Gaming WiFi and ALC4080
by GitHub pull_request - edited 16 May '23

16 May '23

alsa-project/alsa-ucm-conf pull request #315 was edited from Sunsheep: Added configuration for ASUS ROG Strix B650E-I Gaming WiFi and ALC4080 Edit: Added Information from ALSA Information Script [https://alsa-project.org/db/?f=fe7e76a335dc8f1009ecaac9aa341fb68231aee5](ht… Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/315 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/315.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Update USB-Audio.conf for ASUS ROG Strix B650E-I Gaming WiFi and ALC4080
by GitHub pull_request - edited 16 May '23

16 May '23

alsa-project/alsa-ucm-conf pull request #315 was edited from Sunsheep: Added configuration for ASUS ROG Strix B650E-I Gaming WiFi and ALC4080 Edit: Added Information from ALSA Information Script [https://alsa-project.org/db/?f=fe7e76a335dc8f1009ecaac9aa341fb68231aee5](ur… Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/315 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/315.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

external rate plugin: broken 16-bit stereo stream conversion (interleaved -> non-interleaved)
by GitHub issues - edited 16 May '23

16 May '23

alsa-project/alsa-lib issue #320 was edited from ossilator: this setup: ``` Plug PCM: Rate conversion PCM (44100, sformat=S16_LE) Converter: libspeex (external) Protocol version: 10003 Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 17832 period_size : 4458 period_time : 92879 tstamp_mode : NONE tstamp_type : MONOTONIC period_step : 1 avail_min : 4458 period_event : 0 start_threshold : 17832 stop_threshold : 17832 silence_threshold: 0 silence_size : 0 boundary : 5019261784704417792 Slave: Hardware PCM card 0 'E-MU 0404b PCI [MAEM8852]' device 3 subdevice 0 Its setup is: stream : PLAYBACK access : MMAP_NONINTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 44100 exact rate : 44100 (44100/1) msbits : 16 buffer_size : 16384 period_size : 4096 period_time : 92879 tstamp_mode : NONE tstamp_type : MONOTONIC period_step : 1 avail_min : 4096 period_event : 0 start_threshold : 16384 stop_threshold : 16384 silence_threshold: 0 silence_size : 0 boundary : 4611686018427387904 appl_ptr : 0 hw_ptr : 0 ``` garbles the output rather impressively, though it's still recognizable as the input, more or less. it happens both when up-sampling and down-sampling. it doesn't happen when not re-sampling. it doesn't happen when the hardware takes 32-bit samples, so i guess it's somewhere in the optimized s16 paths. it doesn't happen with mono. this is not reproducible with the upstream emu10k1 driver, as the multichannel device demands 16 channels, which causes the route conversion plugin to be inserted, which sidesteps the issue. and the regular pcm device accepts any rate. Issue URL : https://github.com/alsa-project/alsa-lib/issues/320 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

external rate plugin: broken 16-bit stream conversion (interleaved -> non-interleaved)
by GitHub issues - edited 16 May '23

16 May '23

alsa-project/alsa-lib issue #320 was edited from ossilator: this setup: ``` Plug PCM: Rate conversion PCM (44100, sformat=S16_LE) Converter: libspeex (external) Protocol version: 10003 Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 17832 period_size : 4458 period_time : 92879 tstamp_mode : NONE tstamp_type : MONOTONIC period_step : 1 avail_min : 4458 period_event : 0 start_threshold : 17832 stop_threshold : 17832 silence_threshold: 0 silence_size : 0 boundary : 5019261784704417792 Slave: Hardware PCM card 0 'E-MU 0404b PCI [MAEM8852]' device 3 subdevice 0 Its setup is: stream : PLAYBACK access : MMAP_NONINTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 44100 exact rate : 44100 (44100/1) msbits : 16 buffer_size : 16384 period_size : 4096 period_time : 92879 tstamp_mode : NONE tstamp_type : MONOTONIC period_step : 1 avail_min : 4096 period_event : 0 start_threshold : 16384 stop_threshold : 16384 silence_threshold: 0 silence_size : 0 boundary : 4611686018427387904 appl_ptr : 0 hw_ptr : 0 ``` garbles the output rather impressively, though it's still recognizable as the input, more or less. it happens both when up-sampling and down-sampling. it doesn't happen when not re-sampling. it doesn't happen when the hardware takes 32-bit samples, so i guess it's somewhere in the optimized s16 paths. it doesn't happen with mono. this is not reproducible with the upstream emu10k1 driver, as the multichannel device demands 16 channels, which causes the route conversion plugin to be inserted, which sidesteps the issue. and the regular pcm device accepts any rate. Issue URL : https://github.com/alsa-project/alsa-lib/issues/320 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH] ASoC: da7219: Add Jack insertion detection polarity selection
by David Rau 16 May '23

16 May '23

DA7219 can support 2 kinds of insertion detection polarity - Default polarity (Low) - Inverted polarity (High) This patch adds support for selecting insertion detection polarity to the DT binding. Signed-off-by: David Rau <David.Rau.opensource(a)dm.renesas.com> --- .../devicetree/bindings/sound/da7219.txt | 3 ++ include/sound/da7219-aad.h | 6 ++++ sound/soc/codecs/da7219-aad.c | 34 +++++++++++++++++++ 3 files changed, 43 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/da7219.txt b/Documentation/devicetree/bindings/sound/da7219.txt index add1caf26ac2..357d1f61de1d 100644 --- a/Documentation/devicetree/bindings/sound/da7219.txt +++ b/Documentation/devicetree/bindings/sound/da7219.txt @@ -52,6 +52,8 @@ Optional properties: [<200>, <500>, <750>, <1000>] - dlg,jack-ins-deb : Debounce time for jack insertion (ms) [<5>, <10>, <20>, <50>, <100>, <200>, <500>, <1000>] +- dlg,jack-ins-det-pty : Polarity for jack insertion detection + ["low", "high"] - dlg,jack-det-rate: Jack type detection latency (3/4 pole) ["32ms_64ms", "64ms_128ms", "128ms_256ms", "256ms_512ms"] - dlg,jack-rem-deb : Debounce time for jack removal (ms) @@ -98,6 +100,7 @@ Example: dlg,btn-cfg = <50>; dlg,mic-det-thr = <500>; dlg,jack-ins-deb = <20>; + dlg,jack-ins-det-pty = "low"; dlg,jack-det-rate = "32ms_64ms"; dlg,jack-rem-deb = <1>; diff --git a/include/sound/da7219-aad.h b/include/sound/da7219-aad.h index 24ee7baa2589..41320522daa2 100644 --- a/include/sound/da7219-aad.h +++ b/include/sound/da7219-aad.h @@ -44,6 +44,11 @@ enum da7219_aad_jack_ins_deb { DA7219_AAD_JACK_INS_DEB_1S, }; +enum da7219_aad_jack_ins_det_pty { + DA7219_AAD_JACK_INS_DET_PTY_LOW = 0, + DA7219_AAD_JACK_INS_DET_PTY_HIGH, +}; + enum da7219_aad_jack_det_rate { DA7219_AAD_JACK_DET_RATE_32_64MS = 0, DA7219_AAD_JACK_DET_RATE_64_128MS, @@ -80,6 +85,7 @@ struct da7219_aad_pdata { enum da7219_aad_btn_cfg btn_cfg; enum da7219_aad_mic_det_thr mic_det_thr; enum da7219_aad_jack_ins_deb jack_ins_deb; + enum da7219_aad_jack_ins_det_pty jack_ins_det_pty; enum da7219_aad_jack_det_rate jack_det_rate; enum da7219_aad_jack_rem_deb jack_rem_deb; diff --git a/sound/soc/codecs/da7219-aad.c b/sound/soc/codecs/da7219-aad.c index 993a0d00bc48..a61dc965f4fc 100644 --- a/sound/soc/codecs/da7219-aad.c +++ b/sound/soc/codecs/da7219-aad.c @@ -571,6 +571,19 @@ static enum da7219_aad_jack_ins_deb } } +static enum da7219_aad_jack_ins_det_pty + da7219_aad_fw_jack_ins_det_pty(struct device *dev, const char *str) +{ + if (!strcmp(str, "low")) { + return DA7219_AAD_JACK_INS_DET_PTY_LOW; + } else if (!strcmp(str, "high")) { + return DA7219_AAD_JACK_INS_DET_PTY_HIGH; + } else { + dev_warn(dev, "Invalid jack insertion detection polarity"); + return DA7219_AAD_JACK_INS_DET_PTY_LOW; + } +} + static enum da7219_aad_jack_det_rate da7219_aad_fw_jack_det_rate(struct device *dev, const char *str) { @@ -688,6 +701,12 @@ static struct da7219_aad_pdata *da7219_aad_fw_to_pdata(struct device *dev) else aad_pdata->jack_ins_deb = DA7219_AAD_JACK_INS_DEB_20MS; + if (!fwnode_property_read_string(aad_np, "dlg,jack-ins-det-pty", &fw_str)) + aad_pdata->jack_ins_det_pty = + da7219_aad_fw_jack_ins_det_pty(dev, fw_str); + else + aad_pdata->jack_ins_det_pty = DA7219_AAD_JACK_INS_DET_PTY_LOW; + if (!fwnode_property_read_string(aad_np, "dlg,jack-det-rate", &fw_str)) aad_pdata->jack_det_rate = da7219_aad_fw_jack_det_rate(dev, fw_str); @@ -849,6 +868,21 @@ static void da7219_aad_handle_pdata(struct snd_soc_component *component) mask |= DA7219_ADC_1_BIT_REPEAT_MASK; } snd_soc_component_update_bits(component, DA7219_ACCDET_CONFIG_7, mask, cfg); + + switch (aad_pdata->jack_ins_det_pty) { + case DA7219_AAD_JACK_INS_DET_PTY_LOW: + snd_soc_component_write(component, 0xF0, 0x8B); + snd_soc_component_write(component, 0x75, 0x80); + snd_soc_component_write(component, 0xF0, 0x00); + break; + case DA7219_AAD_JACK_INS_DET_PTY_HIGH: + snd_soc_component_write(component, 0xF0, 0x8B); + snd_soc_component_write(component, 0x75, 0x00); + snd_soc_component_write(component, 0xF0, 0x00); + break; + default: + break; + } } } -- 2.17.1

3 2

alsa-lib hangs when mounting a partition while playing a sound file using aplay
by GitHub issues - opened 16 May '23

16 May '23

alsa-project/alsa-lib issue #321 was opened from nachopitt: Hello, I'm using alsa-lib to add sound playback functionality to a Embedded Linux system. Recently I noticed that whenever I mount a UBI partition while there is a sound file being played back with the help of the aplay command, the aplay process hangs forever and the sound playback stops immediately. ``` # aplay -D output1 1.wav & Playing WAVE '1.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Mono # # # mount -t ubifs /dev/ubi0_5 /opt/conf/Logging UBIFS (ubi0:5): background thread "ubifs_bgt0_5" started, PID 190 UBIFS (ubi0:5): UBIFS: mounted UBI device 0, volume 5, name "Logging" UBIFS (ubi0:5): LEB size: 126976 bytes (124 KiB), min./max. I/O unit sizes: 2048 bytes/2048 bytes UBIFS (ubi0:5): FS size: 1650688 bytes (1 MiB, 13 LEBs), journal size 1142785 bytes (1 MiB, 8 LEBs) underrun!!! (at least 0.044 ms loUBIFS (ubi0:5): reserved for root: 0 bytes (0 KiB) ng) UBIFS (ubi0:5): media format: w4/r0 (latest is w5/r0), UUID 26A3D522-A34F-4FE4-BF6C-931A9FE59854, small LPT model # ``` I also prepared a small video that captures the exact moment where the issue is reproduced in case you want it to see it in action: https://www.youtube.com/watch?v=M9bW2k2CTVs The only way to recover from this scenario is to kill the stuck aplay process manually and execute it again. Last time I tried to debug this, I attached the gdbserver to the still stuck aplay process and backtrace showed the following information: ``` (gdb) bt arm-linux-gnueabihf-gdb bin/aplay (wd: ~/home_volume/git_repos/alsa/alsa-install/usr) (gdb) bt #0 0x76d96ba4 in ?? () **#1 0x76f0e222 in snd1_pcm_wait_nocheck (pcm=0x879280, timeout=-1) at pcm.c:2898** Backtrace stopped: previous frame inner to this frame (corrupt stack?) (gdb) frame 1 #1 0x76f0e222 in snd1_pcm_wait_nocheck (pcm=0x879280, timeout=-1) at pcm.c:2898 2898 err_poll = poll(pfd, npfds, timeout); (gdb) ``` It is clear to me that the `poll` function is waiting for something to change in the file descriptors and it waits infinitely due to the `timeout = -1` argument, but none of the file descriptors in `pfd` ever change or become ready after the mount command is executed during the sound playback. I'm currently using both alsa-lib and alsa-utils v1.2.6. The SoC is an NXP i.MX6 Solo which includes an Armv7-A 32bits Cortex-A9 Core CPU Platform. Linux Kernel BSP is based on 4.14.98. Most likely this is an issue with my system configuration, but since you guys are more familiar with alsa than anyone on the planet, I thought you might have some ideas to troubleshoot and debug this because I currently do not have any clue. Do you know what could cause this behavior? Is there any possibility that there is an issue in alsa-lib? Do you have any suggestions to debug and troubleshoot the error to get more information? Any help will be really appreciated. Thanks a lot. Best regards. Issue URL : https://github.com/alsa-project/alsa-lib/issues/321 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH v2 1/4] ALSA: emu10k1: automate encoding of sub-register definitions
by Oswald Buddenhagen 15 May '23

15 May '23

The idea to encode the bitfield manipulation in the register address is quite clever, but doing that by hand is ugly and error-prone. So derive it automatically from the mask instead. Macros cannot #define other macros, so we now declare enums instead. This also adds macros for decoding the register definitions. These will be used by later commits. Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen(a)gmx.de> --- v2: added comments --- include/sound/emu10k1.h | 123 ++++++++++++++++++++-------------------- 1 file changed, 60 insertions(+), 63 deletions(-) diff --git a/include/sound/emu10k1.h b/include/sound/emu10k1.h index 7129b9249eb3..e9b1729ade60 100644 --- a/include/sound/emu10k1.h +++ b/include/sound/emu10k1.h @@ -38,6 +38,32 @@ #define IP_TO_CP(ip) ((ip == 0) ? 0 : (((0x00001000uL | (ip & 0x00000FFFL)) << (((ip >> 12) & 0x000FL) + 4)) & 0xFFFF0000uL)) +// This is used to define hardware bit-fields (sub-registers) by combining +// the bit shift and count with the actual register address. The passed +// mask must represent a single run of adjacent bits. +// The non-concatenating (_NC) variant should be used directly only for +// sub-registers that do not follow the <register>_<field> naming pattern. +#define SUB_REG_NC(reg, field, mask) \ + enum { \ + field ## _MASK = mask, \ + field = reg | \ + (__builtin_ctz(mask) << 16) | \ + (__builtin_popcount(mask) << 24), \ + }; +#define SUB_REG(reg, field, mask) SUB_REG_NC(reg, reg ## _ ## field, mask) + +// Macros for manipulating values of bit-fields declared using the above macros. +// Best used with constant register addresses, as otherwise quite some code is +// generated. The actual register read/write functions handle combined addresses +// automatically, so use of these macros conveys no advantage when accessing a +// single sub-register at a time. +#define REG_SHIFT(r) (((r) >> 16) & 0x1f) +#define REG_SIZE(r) (((r) >> 24) & 0x1f) +#define REG_MASK0(r) ((1U << REG_SIZE(r)) - 1U) +#define REG_MASK(r) (REG_MASK0(r) << REG_SHIFT(r)) +#define REG_VAL_GET(r, v) ((v & REG_MASK(r)) >> REG_SHIFT(r)) +#define REG_VAL_PUT(r, v) ((v) << REG_SHIFT(r)) + // Audigy specify registers are prefixed with 'A_' /************************************************************************************************/ @@ -148,12 +174,10 @@ #define INTE_MIDIRXENABLE 0x00000001 /* Enable MIDI receive-buffer-empty interrupts */ #define WC 0x10 /* Wall Clock register */ -#define WC_SAMPLECOUNTER_MASK 0x03FFFFC0 /* Sample periods elapsed since reset */ -#define WC_SAMPLECOUNTER 0x14060010 -#define WC_CURRENTCHANNEL_MASK 0x0000003F /* Channel [0..63] currently being serviced */ +SUB_REG(WC, SAMPLECOUNTER, 0x03FFFFC0) /* Sample periods elapsed since reset */ +SUB_REG(WC, CURRENTCHANNEL, 0x0000003F) /* Channel [0..63] currently being serviced */ /* NOTE: Each channel takes 1/64th of a sample */ /* period to be serviced. */ -#define WC_CURRENTCHANNEL 0x06000010 #define HCFG 0x14 /* Hardware config register */ /* NOTE: There is no reason to use the legacy */ @@ -225,9 +249,8 @@ /* async audio source */ #define HCFG_LOCKSOUNDCACHE 0x00000008 /* 1 = Cancel bustmaster accesses to soundcache */ /* NOTE: This should generally never be used. */ -#define HCFG_LOCKTANKCACHE_MASK 0x00000004 /* 1 = Cancel bustmaster accesses to tankcache */ +SUB_REG(HCFG, LOCKTANKCACHE, 0x00000004) /* 1 = Cancel bustmaster accesses to tankcache */ /* NOTE: This should generally never be used. */ -#define HCFG_LOCKTANKCACHE 0x01020014 #define HCFG_MUTEBUTTONENABLE 0x00000002 /* 1 = Master mute button sets AUDIOENABLE = 0. */ /* NOTE: This is a 'cheap' way to implement a */ /* master mute function on the mute button, and */ @@ -382,55 +405,38 @@ // which the current registers "swerve" gradually. #define CPF 0x00 /* Current pitch and fraction register */ -#define CPF_CURRENTPITCH_MASK 0xffff0000 /* Current pitch (linear, 0x4000 == unity pitch shift) */ -#define CPF_CURRENTPITCH 0x10100000 +SUB_REG(CPF, CURRENTPITCH, 0xffff0000) /* Current pitch (linear, 0x4000 == unity pitch shift) */ #define CPF_STEREO_MASK 0x00008000 /* 1 = Even channel interleave, odd channel locked */ #define CPF_STOP_MASK 0x00004000 /* 1 = Current pitch forced to 0 */ #define CPF_FRACADDRESS_MASK 0x00003fff /* Linear fractional address of the current channel */ #define PTRX 0x01 /* Pitch target and send A/B amounts register */ -#define PTRX_PITCHTARGET_MASK 0xffff0000 /* Pitch target of specified channel */ -#define PTRX_PITCHTARGET 0x10100001 -#define PTRX_FXSENDAMOUNT_A_MASK 0x0000ff00 /* Linear level of channel output sent to FX send bus A */ -#define PTRX_FXSENDAMOUNT_A 0x08080001 -#define PTRX_FXSENDAMOUNT_B_MASK 0x000000ff /* Linear level of channel output sent to FX send bus B */ -#define PTRX_FXSENDAMOUNT_B 0x08000001 +SUB_REG(PTRX, PITCHTARGET, 0xffff0000) /* Pitch target of specified channel */ +SUB_REG(PTRX, FXSENDAMOUNT_A, 0x0000ff00) /* Linear level of channel output sent to FX send bus A */ +SUB_REG(PTRX, FXSENDAMOUNT_B, 0x000000ff) /* Linear level of channel output sent to FX send bus B */ #define CVCF 0x02 /* Current volume and filter cutoff register */ -#define CVCF_CURRENTVOL_MASK 0xffff0000 /* Current linear volume of specified channel */ -#define CVCF_CURRENTVOL 0x10100002 -#define CVCF_CURRENTFILTER_MASK 0x0000ffff /* Current filter cutoff frequency of specified channel */ -#define CVCF_CURRENTFILTER 0x10000002 +SUB_REG(CVCF, CURRENTVOL, 0xffff0000) /* Current linear volume of specified channel */ +SUB_REG(CVCF, CURRENTFILTER, 0x0000ffff) /* Current filter cutoff frequency of specified channel */ #define VTFT 0x03 /* Volume target and filter cutoff target register */ -#define VTFT_VOLUMETARGET_MASK 0xffff0000 /* Volume target of specified channel */ -#define VTFT_VOLUMETARGET 0x10100003 -#define VTFT_FILTERTARGET_MASK 0x0000ffff /* Filter cutoff target of specified channel */ -#define VTFT_FILTERTARGET 0x10000003 +SUB_REG(VTFT, VOLUMETARGET, 0xffff0000) /* Volume target of specified channel */ +SUB_REG(VTFT, FILTERTARGET, 0x0000ffff) /* Filter cutoff target of specified channel */ #define Z1 0x05 /* Filter delay memory 1 register */ #define Z2 0x04 /* Filter delay memory 2 register */ #define PSST 0x06 /* Send C amount and loop start address register */ -#define PSST_FXSENDAMOUNT_C_MASK 0xff000000 /* Linear level of channel output sent to FX send bus C */ - -#define PSST_FXSENDAMOUNT_C 0x08180006 - -#define PSST_LOOPSTARTADDR_MASK 0x00ffffff /* Loop start address of the specified channel */ -#define PSST_LOOPSTARTADDR 0x18000006 +SUB_REG(PSST, FXSENDAMOUNT_C, 0xff000000) /* Linear level of channel output sent to FX send bus C */ +SUB_REG(PSST, LOOPSTARTADDR, 0x00ffffff) /* Loop start address of the specified channel */ #define DSL 0x07 /* Send D amount and loop end address register */ -#define DSL_FXSENDAMOUNT_D_MASK 0xff000000 /* Linear level of channel output sent to FX send bus D */ - -#define DSL_FXSENDAMOUNT_D 0x08180007 - -#define DSL_LOOPENDADDR_MASK 0x00ffffff /* Loop end address of the specified channel */ -#define DSL_LOOPENDADDR 0x18000007 +SUB_REG(DSL, FXSENDAMOUNT_D, 0xff000000) /* Linear level of channel output sent to FX send bus D */ +SUB_REG(DSL, LOOPENDADDR, 0x00ffffff) /* Loop end address of the specified channel */ #define CCCA 0x08 /* Filter Q, interp. ROM, byte size, cur. addr register */ -#define CCCA_RESONANCE_MASK 0xf0000000 /* Lowpass filter resonance (Q) height */ -#define CCCA_RESONANCE 0x041c0008 +SUB_REG(CCCA, RESONANCE, 0xf0000000) /* Lowpass filter resonance (Q) height */ #define CCCA_INTERPROM_MASK 0x0e000000 /* Selects passband of interpolation ROM */ /* 1 == full band, 7 == lowpass */ /* ROM 0 is used when pitch shifting downward or less */ @@ -447,27 +453,24 @@ #define CCCA_INTERPROM_7 0x0e000000 /* Select interpolation ROM 7 */ #define CCCA_8BITSELECT 0x01000000 /* 1 = Sound memory for this channel uses 8-bit samples */ /* 8-bit samples are unsigned, 16-bit ones signed */ -#define CCCA_CURRADDR_MASK 0x00ffffff /* Current address of the selected channel */ -#define CCCA_CURRADDR 0x18000008 +SUB_REG(CCCA, CURRADDR, 0x00ffffff) /* Current address of the selected channel */ #define CCR 0x09 /* Cache control register */ -#define CCR_CACHEINVALIDSIZE 0x07190009 -#define CCR_CACHEINVALIDSIZE_MASK 0xfe000000 /* Number of invalid samples before the read address */ +SUB_REG(CCR, CACHEINVALIDSIZE, 0xfe000000) /* Number of invalid samples before the read address */ #define CCR_CACHELOOPFLAG 0x01000000 /* 1 = Cache has a loop service pending */ #define CCR_INTERLEAVEDSAMPLES 0x00800000 /* 1 = A cache service will fetch interleaved samples */ /* Auto-set from CPF_STEREO_MASK */ #define CCR_WORDSIZEDSAMPLES 0x00400000 /* 1 = A cache service will fetch word sized samples */ /* Auto-set from CCCA_8BITSELECT */ -#define CCR_READADDRESS 0x06100009 -#define CCR_READADDRESS_MASK 0x003f0000 /* Next cached sample to play */ -#define CCR_LOOPINVALSIZE 0x0000fe00 /* Number of invalid samples in cache prior to loop */ +SUB_REG(CCR, READADDRESS, 0x003f0000) /* Next cached sample to play */ +SUB_REG(CCR, LOOPINVALSIZE, 0x0000fe00) /* Number of invalid samples in cache prior to loop */ /* NOTE: This is valid only if CACHELOOPFLAG is set */ #define CCR_LOOPFLAG 0x00000100 /* Set for a single sample period when a loop occurs */ -#define CCR_CACHELOOPADDRHI 0x000000ff /* CLP_LOOPSTARTADDR's hi byte if CACHELOOPFLAG is set */ +SUB_REG(CCR, CACHELOOPADDRHI, 0x000000ff) /* CLP_LOOPSTARTADDR's hi byte if CACHELOOPFLAG is set */ #define CLP 0x0a /* Cache loop register (valid if CCR_CACHELOOPFLAG = 1) */ /* NOTE: This register is normally not used */ -#define CLP_CACHELOOPADDR 0x0000ffff /* Cache loop address low word */ +SUB_REG(CLP, CACHELOOPADDR, 0x0000ffff) /* Cache loop address low word */ #define FXRT 0x0b /* Effects send routing register */ /* NOTE: It is illegal to assign the same routing to */ @@ -537,20 +540,17 @@ #define IP_UNITY 0x0000e000 /* Unity pitch shift */ #define IFATN 0x19 /* Initial filter cutoff and attenuation register */ -#define IFATN_FILTERCUTOFF_MASK 0x0000ff00 /* Initial filter cutoff frequency in exponential units */ +SUB_REG(IFATN, FILTERCUTOFF, 0x0000ff00) /* Initial filter cutoff frequency in exponential units */ /* 6 most significant bits are semitones */ /* 2 least significant bits are fractions */ -#define IFATN_FILTERCUTOFF 0x08080019 -#define IFATN_ATTENUATION_MASK 0x000000ff /* Initial attenuation in 0.375dB steps */ -#define IFATN_ATTENUATION 0x08000019 +SUB_REG(IFATN, ATTENUATION, 0x000000ff) /* Initial attenuation in 0.375dB steps */ #define PEFE 0x1a /* Pitch envelope and filter envelope amount register */ -#define PEFE_PITCHAMOUNT_MASK 0x0000ff00 /* Pitch envlope amount */ +SUB_REG(PEFE, PITCHAMOUNT, 0x0000ff00) /* Pitch envlope amount */ /* Signed 2's complement, +/- one octave peak extremes */ -#define PEFE_PITCHAMOUNT 0x0808001a -#define PEFE_FILTERAMOUNT_MASK 0x000000ff /* Filter envlope amount */ +SUB_REG(PEFE, FILTERAMOUNT, 0x000000ff) /* Filter envlope amount */ /* Signed 2's complement, +/- six octaves peak extremes */ -#define PEFE_FILTERAMOUNT 0x0800001a + #define FMMOD 0x1b /* Vibrato/filter modulation from LFO register */ #define FMMOD_MODVIBRATO 0x0000ff00 /* Vibrato LFO modulation depth */ @@ -793,22 +793,19 @@ #define SRCS_SPDIFRATE_96 0x00080000 #define MICIDX 0x63 /* Microphone recording buffer index register */ -#define MICIDX_MASK 0x0000ffff /* 16-bit value */ -#define MICIDX_IDX 0x10000063 +SUB_REG(MICIDX, IDX, 0x0000ffff) #define ADCIDX 0x64 /* ADC recording buffer index register */ -#define ADCIDX_MASK 0x0000ffff /* 16 bit index field */ -#define ADCIDX_IDX 0x10000064 +SUB_REG(ADCIDX, IDX, 0x0000ffff) #define A_ADCIDX 0x63 -#define A_ADCIDX_IDX 0x10000063 +SUB_REG(A_ADCIDX, IDX, 0x0000ffff) #define A_MICIDX 0x64 -#define A_MICIDX_IDX 0x10000064 +SUB_REG(A_MICIDX, IDX, 0x0000ffff) #define FXIDX 0x65 /* FX recording buffer index register */ -#define FXIDX_MASK 0x0000ffff /* 16-bit value */ -#define FXIDX_IDX 0x10000065 +SUB_REG(FXIDX, IDX, 0x0000ffff) /* The 32-bit HLIEx and HLIPx registers all have one bit per channel control/status */ #define HLIEL 0x66 /* Channel half loop interrupt enable low register */ @@ -852,8 +849,8 @@ #define A_SPDIF_44100 0x00000080 #define A_SPDIF_MUTED 0x000000c0 -#define A_I2S_CAPTURE_RATE_MASK 0x00000e00 /* This sets the capture PCM rate, but it is */ -#define A_I2S_CAPTURE_RATE 0x03090076 /* unclear if this sets the ADC rate as well. */ +SUB_REG_NC(A_EHC, A_I2S_CAPTURE_RATE, 0x00000e00) /* This sets the capture PCM rate, but it is */ + /* unclear if this sets the ADC rate as well. */ #define A_I2S_CAPTURE_48000 0x0 #define A_I2S_CAPTURE_192000 0x1 #define A_I2S_CAPTURE_96000 0x2 -- 2.40.0.152.g15d061e6df

2 4

alsactl: fix compilation when building in a subdir
by GitHub pull_request - opened 15 May '23

15 May '23

alsa-project/alsa-utils pull request #213 was opened from heitbaum: Fixes: 613372d Fixes: cff2d1c Fixes: #212 Compile errors when building in a subdir: ``` alsactl/alsactl.c:33:10: fatal error: os_compat.h: No such file or directory 33 | #include "os_compat.h" | ^~~~~~~~~~~~~ alsactl/lock.c:34:10: fatal error: os_compat.h: No such file or directory 34 | #include "os_compat.h" | ^~~~~~~~~~~~~ ``` Request URL : https://github.com/alsa-project/alsa-utils/pull/213 Patch URL : https://github.com/alsa-project/alsa-utils/pull/213.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

Build of alsa-utils 1.2.9 fails when building is a subdir
by GitHub issues - opened 15 May '23

15 May '23

alsa-project/alsa-utils issue #212 was opened from heitbaum: Compile errors when building in a subdir: ``` alsactl/alsactl.c:33:10: fatal error: os_compat.h: No such file or directory 33 | #include "os_compat.h" | ^~~~~~~~~~~~~ alsactl/lock.c:34:10: fatal error: os_compat.h: No such file or directory 34 | #include "os_compat.h" | ^~~~~~~~~~~~~ ``` Issue URL : https://github.com/alsa-project/alsa-utils/issues/212 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH 5.10 051/381] sound/oss/dmasound: fix build when drivers are mixed =y/=m
by Greg Kroah-Hartman 15 May '23

15 May '23

From: Randy Dunlap <rdunlap(a)infradead.org> commit 9dd7c46346ca4390f84a7ea9933005eb1b175c15 upstream. When CONFIG_DMASOUND_ATARI=m and CONFIG_DMASOUND_Q40=y (or vice versa), dmasound_core.o can be built without dmasound_deinit() being defined, causing a build error: ERROR: modpost: "dmasound_deinit" [sound/oss/dmasound/dmasound_atari.ko] undefined! Modify dmasound_core.c and dmasound.h so that dmasound_deinit() is always available. The mixed modes (=y/=m) also mean that several variables and structs have to be declared in all cases. Suggested-by: Arnd Bergmann <arnd(a)arndb.de> Suggested-by: Geert Uytterhoeven <geert(a)linux-m68k.org> Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Reported-by: kernel test robot <lkp(a)intel.com> Link: lore.kernel.org/r/202204032138.EFT9qGEd-lkp@intel.com Cc: Geert Uytterhoeven <geert(a)linux-m68k.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Link: https://lore.kernel.org/r/20220405234118.24830-1-rdunlap@infradead.org Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> --- sound/oss/dmasound/dmasound.h | 6 ------ sound/oss/dmasound/dmasound_core.c | 24 +----------------------- 2 files changed, 1 insertion(+), 29 deletions(-) --- a/sound/oss/dmasound/dmasound.h +++ b/sound/oss/dmasound/dmasound.h @@ -88,11 +88,7 @@ static inline int ioctl_return(int __use */ extern int dmasound_init(void); -#ifdef MODULE extern void dmasound_deinit(void); -#else -#define dmasound_deinit() do { } while (0) -#endif /* description of the set-up applies to either hard or soft settings */ @@ -114,9 +110,7 @@ typedef struct { void *(*dma_alloc)(unsigned int, gfp_t); void (*dma_free)(void *, unsigned int); int (*irqinit)(void); -#ifdef MODULE void (*irqcleanup)(void); -#endif void (*init)(void); void (*silence)(void); int (*setFormat)(int); --- a/sound/oss/dmasound/dmasound_core.c +++ b/sound/oss/dmasound/dmasound_core.c @@ -206,12 +206,10 @@ module_param(writeBufSize, int, 0); MODULE_LICENSE("GPL"); -#ifdef MODULE static int sq_unit = -1; static int mixer_unit = -1; static int state_unit = -1; static int irq_installed; -#endif /* MODULE */ /* control over who can modify resources shared between play/record */ static fmode_t shared_resource_owner; @@ -391,9 +389,6 @@ static const struct file_operations mixe static void mixer_init(void) { -#ifndef MODULE - int mixer_unit; -#endif mixer_unit = register_sound_mixer(&mixer_fops, -1); if (mixer_unit < 0) return; @@ -1176,9 +1171,6 @@ static const struct file_operations sq_f static int sq_init(void) { const struct file_operations *fops = &sq_fops; -#ifndef MODULE - int sq_unit; -#endif sq_unit = register_sound_dsp(fops, -1); if (sq_unit < 0) { @@ -1380,9 +1372,6 @@ static const struct file_operations stat static int state_init(void) { -#ifndef MODULE - int state_unit; -#endif state_unit = register_sound_special(&state_fops, SND_DEV_STATUS); if (state_unit < 0) return state_unit ; @@ -1400,10 +1389,9 @@ static int state_init(void) int dmasound_init(void) { int res ; -#ifdef MODULE + if (irq_installed) return -EBUSY; -#endif /* Set up sound queue, /dev/audio and /dev/dsp. */ @@ -1422,9 +1410,7 @@ int dmasound_init(void) printk(KERN_ERR "DMA sound driver: Interrupt initialization failed\n"); return -ENODEV; } -#ifdef MODULE irq_installed = 1; -#endif printk(KERN_INFO "%s DMA sound driver rev %03d installed\n", dmasound.mach.name, (DMASOUND_CORE_REVISION<<4) + @@ -1438,8 +1424,6 @@ int dmasound_init(void) return 0; } -#ifdef MODULE - void dmasound_deinit(void) { if (irq_installed) { @@ -1458,8 +1442,6 @@ void dmasound_deinit(void) unregister_sound_dsp(sq_unit); } -#else /* !MODULE */ - static int dmasound_setup(char *str) { int ints[6], size; @@ -1503,8 +1485,6 @@ static int dmasound_setup(char *str) __setup("dmasound=", dmasound_setup); -#endif /* !MODULE */ - /* * Conversion tables */ @@ -1591,9 +1571,7 @@ char dmasound_alaw2dma8[] = { EXPORT_SYMBOL(dmasound); EXPORT_SYMBOL(dmasound_init); -#ifdef MODULE EXPORT_SYMBOL(dmasound_deinit); -#endif EXPORT_SYMBOL(dmasound_write_sq); EXPORT_SYMBOL(dmasound_catchRadius); #ifdef HAS_8BIT_TABLES

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Re: [PATCH] ASoC: cs35l45: Relicense to GPL only
by James Schulman 15 May '23

15 May '23

On 10/05/2023 10:25, Charles Keepax wrote: > Cirrus never intended to upstream dual licensed code, convert to > GPL only. > > Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> > --- > > Apologies for this but this was never our intention and I believe that > it can be a little legally involved maintaining dual licensed kernel code. > Fortunately not that many people outside of Cirrus have contributed to > the files in question. I have CCed everyone who has contributed both > internally and externally. > > Thanks, > Charles > > sound/soc/codecs/cs35l45-i2c.c | 4 ++-- > sound/soc/codecs/cs35l45-spi.c | 4 ++-- > sound/soc/codecs/cs35l45-tables.c | 2 +- > sound/soc/codecs/cs35l45.c | 4 ++-- > sound/soc/codecs/cs35l45.h | 2 +- > 5 files changed, 8 insertions(+), 8 deletions(-) > Reviewed-by: James Schulman <james.schulman(a)cirrus.com>

1 0

[PATCH 0/9] ASoC: SOF: ipc4-topology: Improve the audio format selection logic
by Peter Ujfalusi 15 May '23

15 May '23

Hi, Improve the logic to account for the current restrictions in topology while making it extensible for future topology modifications. The current topology definitions assume that input/output formats come in pairs. For example even if there's only 1 output format for a module, we add 3 output formats to match that of the input format count with the same parameters. This is unnecessary but we have to deal with it until the topologies are modified. Additionally, choosing the input/output audio format should depend only on the pipeline params or the runtime FE hw_params depending on where the module is in the pipeline. This PR modifies the logic for selection based on this and removes unnecessary dependencies between the input and output formats. Regards, Peter --- Ranjani Sridharan (9): ASoC: SOF: ipc4-topology: Handle input/output audio format special case ASoC: SOF: ipc4-topology: Add a helper function for output format selection ASoC: SOF: ipc4-topology: Move the call to init output format ASoC: SOF: ipc4-topology: Rename sof_ipc4_init_audio_fmt() ASoC: SOF: ipc4-topology: Handle output format special case ASoC: SOF: ipc4-topology: Add a new helper function to get the valid bits ASoC: SOF: ipc4-topology: Modify the output format selection logic ASoC: SOF: ipc4-topology: New helper to check if all output formats are the same ASoC: SOF: ipc4-topology: Modify input audio format selection logic sound/soc/sof/ipc4-topology.c | 358 ++++++++++++++++++++++------------ 1 file changed, 237 insertions(+), 121 deletions(-) -- 2.40.1

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[PATCH v4 0/9] ASoC: mediatek: mt8188: revise AFE driver
by Trevor Wu 15 May '23

15 May '23

The series of patches consists of four major changes. First, remove redundant supply for ADDA DAI dirver. Second, revise ETDM control including APLL dynamic switch via DAPM, so APLL can be enabled when it is really required. Third, update AFE probe function. Bus protection change was dropped at the previous patch because the dependent change was not accepted at that time. Finally, correct some binding errors and add required clocks. Changes since v3: - correct errors in binding file - add required clocks which are possibly used in the future Changes since v2: - remove required properties in dt-binding file Changes since v1: - replace unnecessary dev_info with dev_dbg - fix dt-binging ABI breaking behavior Trevor Wu (9): ASoC: mediatek: mt8188: remove supply AUDIO_HIRES ASoC: mediatek: mt8188: complete set_tdm_slot function ASoC: mediatek: mt8188: revise ETDM control flow ASoC: mediatek: mt8188: refine APLL control ASoC: mediatek: mt8188: combine afe component registration ASoC: mediatek: mt8188: add bus protection ASoC: mediatek: mt8188: add required clocks ASoC: dt-bindings: mediatek,mt8188-afe: correct clock name ASoC: dt-bindings: mediatek,mt8188-afe: add audio properties .../bindings/sound/mediatek,mt8188-afe.yaml | 66 +- sound/soc/mediatek/mt8188/mt8188-afe-clk.c | 142 ++- sound/soc/mediatek/mt8188/mt8188-afe-clk.h | 15 + sound/soc/mediatek/mt8188/mt8188-afe-pcm.c | 113 ++- sound/soc/mediatek/mt8188/mt8188-dai-adda.c | 37 - sound/soc/mediatek/mt8188/mt8188-dai-etdm.c | 922 ++++++++++-------- sound/soc/mediatek/mt8188/mt8188-reg.h | 2 + 7 files changed, 795 insertions(+), 502 deletions(-) -- 2.18.0

3 12

[PATCH 0/2] ASoC: SOF: Intel/ipc4: Do not reset BE DAI pipeline during stop/suspend
by Peter Ujfalusi 15 May '23

15 May '23

Hi, Do not reset pipelines during the stop/suspend triggers in the BE DAI ops as the BE DAI pipeline needs to be left in the PAUSED state. It should only be reset during hw_free. This simplification is already done for the FE pipelines and the DAI trigger only toggles the states between PAUSED and RUNNING. Regards Peter --- Ranjani Sridharan (2): ASoC: SOF: Intel: hda-dai-ops: Split the get_hext_stream() op for IPC4 ASoC: SOF: ipc4-pcm: reset all pipelines during FE DAI hw_free sound/soc/sof/intel/hda-dai-ops.c | 69 +++++++++++++++++++++++-------- sound/soc/sof/ipc4-pcm.c | 4 +- sound/soc/sof/ipc4-topology.c | 1 - 3 files changed, 53 insertions(+), 21 deletions(-) -- 2.40.1

2 3

[PATCH] ASoC: SOF: ipc4-topology: Use set_get_data() to send LARGE_CONFIG message
by Peter Ujfalusi 15 May '23

15 May '23

Instead of open coding the sending of sink format of the copier with LARGE_CONFIG_SET message, use the proper function to do so. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-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> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- sound/soc/sof/ipc4-topology.c | 21 +++++++-------------- 1 file changed, 7 insertions(+), 14 deletions(-) diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index 059eebf0a687..73a45437a3a1 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -2271,11 +2271,11 @@ static int sof_ipc4_set_copier_sink_format(struct snd_sof_dev *sdev, int sink_id) { struct sof_ipc4_copier_config_set_sink_format format; + const struct sof_ipc_ops *iops = sdev->ipc->ops; struct sof_ipc4_base_module_cfg *src_config; const struct sof_ipc4_audio_format *pin_fmt; struct sof_ipc4_fw_module *fw_module; struct sof_ipc4_msg msg = {{ 0 }}; - u32 header, extension; dev_dbg(sdev->dev, "%s set copier sink %d format\n", src_widget->widget->name, sink_id); @@ -2305,22 +2305,15 @@ static int sof_ipc4_set_copier_sink_format(struct snd_sof_dev *sdev, msg.data_size = sizeof(format); msg.data_ptr = &format; - header = fw_module->man4_module_entry.id; - header |= SOF_IPC4_MOD_INSTANCE(src_widget->instance_id); - header |= SOF_IPC4_MSG_TYPE_SET(SOF_IPC4_MOD_LARGE_CONFIG_SET); - header |= SOF_IPC4_MSG_DIR(SOF_IPC4_MSG_REQUEST); - header |= SOF_IPC4_MSG_TARGET(SOF_IPC4_MODULE_MSG); + msg.primary = fw_module->man4_module_entry.id; + msg.primary |= SOF_IPC4_MOD_INSTANCE(src_widget->instance_id); + msg.primary |= SOF_IPC4_MSG_DIR(SOF_IPC4_MSG_REQUEST); + msg.primary |= SOF_IPC4_MSG_TARGET(SOF_IPC4_MODULE_MSG); - extension = SOF_IPC4_MOD_EXT_MSG_SIZE(msg.data_size); - extension |= + msg.extension = SOF_IPC4_MOD_EXT_MSG_PARAM_ID(SOF_IPC4_COPIER_MODULE_CFG_PARAM_SET_SINK_FORMAT); - extension |= SOF_IPC4_MOD_EXT_MSG_LAST_BLOCK(1); - extension |= SOF_IPC4_MOD_EXT_MSG_FIRST_BLOCK(1); - - msg.primary = header; - msg.extension = extension; - return sof_ipc_tx_message_no_reply(sdev->ipc, &msg, msg.data_size); + return iops->set_get_data(sdev, &msg, msg.data_size, true); } static int sof_ipc4_route_setup(struct snd_sof_dev *sdev, struct snd_sof_route *sroute) -- 2.40.1

2 1

[PATCH 0/9] ASoC: SOF: Intel: hda-dai: prepare LNL support
by Pierre-Louis Bossart 15 May '23

15 May '23

The HDAudio support has not changed much since CometLake/2019: the code was reshuffled for IPC4 support and we used hdac_hda for external and Display Audio codec, but the hda-dai code was only used for HDAudio codecs. In the LunarLake architecture, all endpoints (HDaudio, SSP, DMIC, SoundWire) are handled with the HDaudio DMA, which requires us to revisit the definitions of HDA_LINK, and remove the mutual exclusion between NOCODEC and HDA_LINK: we do want the ability to test SSP/DMIC in NOCODEC mode even with an HDA DMA. This code change exposed a number of issues, with a useless .prepare callback, a DAI number mismatch and the need to support SoundWire which is handled by a different component in drivers/soundwire/intel_ace2.c. Pierre-Louis Bossart (9): ASoC: SOF: Intel: hda-dai: simplify .prepare callback ASoC: SOF: Intel: hda-dai: remove use of cpu_dai->component drvdata ASoC: SOF: Intel: fix DAI number mismatch ASoC: SOF: Intel: clarify initialization when HDA_AUDIO_CODEC is not used ASoC: SOF: Intel: Kconfig: move selection of PROBE_WORK_QUEUE ASoC: SOF: Intel: hda-dai: move hda_dai_prepare() code ASoC: SOF: Intel: hda-dai: mark functions as __maybe_unused ASoC: SOF: Intel: hda-dai: use HDA_LINK instead of HDA_AUDIO_CODEC ASoC: SOF: Intel: remove mutual exclusion between NOCODEC and HDA_LINK sound/soc/sof/intel/Kconfig | 3 +- sound/soc/sof/intel/hda-bus.c | 11 +++- sound/soc/sof/intel/hda-dai-ops.c | 4 +- sound/soc/sof/intel/hda-dai.c | 104 +++++++++++++++--------------- sound/soc/sof/intel/hda.h | 2 +- 5 files changed, 63 insertions(+), 61 deletions(-) -- 2.37.2

2 10

[PATCH 0/8] ASoC: Intel: machine driver updates
by Pierre-Louis Bossart 15 May '23

15 May '23

Several updates for RPL/MTL/LNL, and one fix for SoundWire TGL device. Bard Liao (3): ASoC: Intel: sof_sdw: remove SOF_SDW_TGL_HDMI for MeteorLake devices ASoC: Intel: sof_sdw: add quirk for MTL RVP ASoC: Intel: sof_sdw: start set codec init function with an adr index Chao Song (1): ASoC: Intel: soc-acpi: add support for MTL SDCA boards Peter Ujfalusi (1): ASoC: Intel: sof_sdw: add quirk for LNL RVP Pierre-Louis Bossart (2): ASoC: Intel: sof_sdw: add new mapping for HP Spectre x360 ASoC: Intel: soc-acpi: add tables for LunarLake Terry Cheong (1): ASoC: Intel: Add rpl_max98373_8825 driver include/sound/soc-acpi-intel-match.h | 2 + sound/soc/intel/boards/sof_nau8825.c | 10 +++ sound/soc/intel/boards/sof_sdw.c | 49 +++++++++++-- sound/soc/intel/common/Makefile | 1 + .../intel/common/soc-acpi-intel-lnl-match.c | 72 +++++++++++++++++++ .../intel/common/soc-acpi-intel-mtl-match.c | 57 +++++++++++++++ .../intel/common/soc-acpi-intel-rpl-match.c | 12 ++++ 7 files changed, 197 insertions(+), 6 deletions(-) create mode 100644 sound/soc/intel/common/soc-acpi-intel-lnl-match.c -- 2.37.2

2 9

[PATCH] ASoC: SOF: Simplify the calculation of variables
by Jiapeng Chong 15 May '23

15 May '23

./sound/soc/sof/pcm.c:372:27-29: WARNING !A || A && B is equivalent to !A || B. Reported-by: Abaci Robot <abaci(a)linux.alibaba.com> Link: https://bugzilla.openanolis.cn/show_bug.cgi?id=4938 Signed-off-by: Jiapeng Chong <jiapeng.chong(a)linux.alibaba.com> --- sound/soc/sof/pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/pcm.c b/sound/soc/sof/pcm.c index 567db32173a8..d14e7fd2c2ee 100644 --- a/sound/soc/sof/pcm.c +++ b/sound/soc/sof/pcm.c @@ -369,7 +369,7 @@ static int sof_pcm_trigger(struct snd_soc_component *component, case SNDRV_PCM_TRIGGER_PAUSE_PUSH: case SNDRV_PCM_TRIGGER_STOP: /* invoke platform trigger to stop DMA even if pcm_ops isn't set or if it failed */ - if (!pcm_ops || (pcm_ops && !pcm_ops->platform_stop_during_hw_free)) + if (!pcm_ops || !pcm_ops->platform_stop_during_hw_free) snd_sof_pcm_platform_trigger(sdev, substream, cmd); break; default: -- 2.20.1.7.g153144c

3 2

[PATCH] ASoC: SOF: Separate the tokens for input and output pin index
by Peter Ujfalusi 15 May '23

15 May '23

From: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Using the same token ID for both input and output format pin index results in collisions and incorrect pin index getting parsed from topology. Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.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: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Paul Olaru <paul.olaru(a)oss.nxp.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- include/uapi/sound/sof/tokens.h | 3 ++- sound/soc/sof/ipc4-topology.c | 4 ++-- 2 files changed, 4 insertions(+), 3 deletions(-) diff --git a/include/uapi/sound/sof/tokens.h b/include/uapi/sound/sof/tokens.h index bbc37877aaff..e9ec7e4eb982 100644 --- a/include/uapi/sound/sof/tokens.h +++ b/include/uapi/sound/sof/tokens.h @@ -183,7 +183,7 @@ #define SOF_TKN_CAVS_AUDIO_FORMAT_IN_INTERLEAVING_STYLE 1906 #define SOF_TKN_CAVS_AUDIO_FORMAT_IN_FMT_CFG 1907 #define SOF_TKN_CAVS_AUDIO_FORMAT_IN_SAMPLE_TYPE 1908 -#define SOF_TKN_CAVS_AUDIO_FORMAT_PIN_INDEX 1909 +#define SOF_TKN_CAVS_AUDIO_FORMAT_INPUT_PIN_INDEX 1909 /* intentional token numbering discontinuity, reserved for future use */ #define SOF_TKN_CAVS_AUDIO_FORMAT_OUT_RATE 1930 #define SOF_TKN_CAVS_AUDIO_FORMAT_OUT_BIT_DEPTH 1931 @@ -194,6 +194,7 @@ #define SOF_TKN_CAVS_AUDIO_FORMAT_OUT_INTERLEAVING_STYLE 1936 #define SOF_TKN_CAVS_AUDIO_FORMAT_OUT_FMT_CFG 1937 #define SOF_TKN_CAVS_AUDIO_FORMAT_OUT_SAMPLE_TYPE 1938 +#define SOF_TKN_CAVS_AUDIO_FORMAT_OUTPUT_PIN_INDEX 1939 /* intentional token numbering discontinuity, reserved for future use */ #define SOF_TKN_CAVS_AUDIO_FORMAT_IBS 1970 #define SOF_TKN_CAVS_AUDIO_FORMAT_OBS 1971 diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index 059eebf0a687..5abe616a2054 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -59,7 +59,7 @@ static const struct sof_topology_token ipc4_in_audio_format_tokens[] = { audio_fmt.interleaving_style)}, {SOF_TKN_CAVS_AUDIO_FORMAT_IN_FMT_CFG, SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32, offsetof(struct sof_ipc4_pin_format, audio_fmt.fmt_cfg)}, - {SOF_TKN_CAVS_AUDIO_FORMAT_PIN_INDEX, SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32, + {SOF_TKN_CAVS_AUDIO_FORMAT_INPUT_PIN_INDEX, SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32, offsetof(struct sof_ipc4_pin_format, pin_index)}, {SOF_TKN_CAVS_AUDIO_FORMAT_IBS, SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32, offsetof(struct sof_ipc4_pin_format, buffer_size)}, @@ -79,7 +79,7 @@ static const struct sof_topology_token ipc4_out_audio_format_tokens[] = { audio_fmt.interleaving_style)}, {SOF_TKN_CAVS_AUDIO_FORMAT_OUT_FMT_CFG, SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32, offsetof(struct sof_ipc4_pin_format, audio_fmt.fmt_cfg)}, - {SOF_TKN_CAVS_AUDIO_FORMAT_PIN_INDEX, SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32, + {SOF_TKN_CAVS_AUDIO_FORMAT_OUTPUT_PIN_INDEX, SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32, offsetof(struct sof_ipc4_pin_format, pin_index)}, {SOF_TKN_CAVS_AUDIO_FORMAT_OBS, SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32, offsetof(struct sof_ipc4_pin_format, buffer_size)}, -- 2.40.1

2 1

[PATCH] ASoC: SOF: topology: Fix tuples array allocation
by Peter Ujfalusi 15 May '23

15 May '23

From: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> The memory allocated for the tuples array assumes that there's 1 instance of all tokens already. So for those tokens that have multiple instances in topology, we need to exclude the initial instance that has already been accounted for. Fixes: 4fdef47a44d6 ("ASoC: SOF: ipc4-topology: Add new tokens for input/output pin format count") Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.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> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- sound/soc/sof/topology.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c index b80566570512..f160dc454b44 100644 --- a/sound/soc/sof/topology.c +++ b/sound/soc/sof/topology.c @@ -1265,7 +1265,7 @@ static int sof_widget_parse_tokens(struct snd_soc_component *scomp, struct snd_s if (num_sets > 1) { struct snd_sof_tuple *new_tuples; - num_tuples += token_list[object_token_list[i]].count * num_sets; + num_tuples += token_list[object_token_list[i]].count * (num_sets - 1); new_tuples = krealloc(swidget->tuples, sizeof(*new_tuples) * num_tuples, GFP_KERNEL); if (!new_tuples) { -- 2.40.1

2 1

[PATCH 0/6] ASoC: SOF: Intel: hda-mlink: fixes and extensions
by Pierre-Louis Bossart 15 May '23

15 May '23

With additional testing with multiple links and multiple DAI types, we found a couple of mistakes with refcounts, base address, missing initialization. A new helper was also added due to a change in the SoundWire programming sequences, with the host driver in charge of setting up the DMA channel mapping instead of the firmware. Pierre-Louis Bossart (6): ASoC: SOF: Intel: hda-mlink: fix sublink refcounting ASoC: SOF: Intel: hda-mlink: add helper to get SoundWire hlink ASoC: SOF: Intel: hda-mlink: fix base_ptr computation ASoC: SOF: Intel: hda-mlink: use 'ml_addr' parameter consistently ASoC: SOF: Intel: hda-mlink: initialize instance_offset member ASoC: SOF: Intel: hda-mlink: add helper to program SoundWire PCMSyCM registers include/sound/hda-mlink.h | 14 +++++ sound/soc/sof/intel/hda-mlink.c | 96 +++++++++++++++++++++++++++++---- 2 files changed, 101 insertions(+), 9 deletions(-) -- 2.37.2

2 7

[PATCH v2 1/3] ASoC: Add ESS ES9218P codec driver
by Aidan MacDonald 15 May '23

15 May '23

The ES9218P is a 32-bit stereo codec featuring an integrated headphone amp with analog volume control and up to 2.0 V rms output, and a low-power bypass path for passing through an analog signal from an alternate source. The digital audio interface supports 16/24/32-bit I2S inputs ranging from 8 KHz to 384 KHz, and DSD inputs. This driver only supports I2S operation. Signed-off-by: Aidan MacDonald <aidanmacdonald.0x0(a)gmail.com> --- Notes: v1->v2 - Drop clock names, add __maybe_unused to OF match table - Add comment explaining why ess,max-clock-div exists sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ess-es9218p.c | 805 +++++++++++++++++++++++++++++++++ 3 files changed, 813 insertions(+) create mode 100644 sound/soc/codecs/ess-es9218p.c diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 8020097d4e4c..973ce164e1e5 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -108,6 +108,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_ES8328_I2C imply SND_SOC_ES7134 imply SND_SOC_ES7241 + imply SND_SOC_ESS_ES9218P imply SND_SOC_GTM601 imply SND_SOC_HDAC_HDMI imply SND_SOC_HDAC_HDA @@ -1017,6 +1018,11 @@ config SND_SOC_ES8328_SPI depends on SPI_MASTER select SND_SOC_ES8328 +config SND_SOC_ESS_ES9218P + tristate "ESS Technology ES9218P CODEC" + depends on I2C + select REGMAP_I2C + config SND_SOC_GTM601 tristate 'GTM601 UMTS modem audio codec' diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 5cdbae88e6e3..1c119f1d9cb1 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -114,6 +114,7 @@ snd-soc-es8326-objs := es8326.o snd-soc-es8328-objs := es8328.o snd-soc-es8328-i2c-objs := es8328-i2c.o snd-soc-es8328-spi-objs := es8328-spi.o +snd-soc-ess-es9218p-objs := ess-es9218p.o snd-soc-gtm601-objs := gtm601.o snd-soc-hdac-hdmi-objs := hdac_hdmi.o snd-soc-hdac-hda-objs := hdac_hda.o @@ -488,6 +489,7 @@ obj-$(CONFIG_SND_SOC_ES8326) += snd-soc-es8326.o obj-$(CONFIG_SND_SOC_ES8328) += snd-soc-es8328.o obj-$(CONFIG_SND_SOC_ES8328_I2C)+= snd-soc-es8328-i2c.o obj-$(CONFIG_SND_SOC_ES8328_SPI)+= snd-soc-es8328-spi.o +obj-$(CONFIG_SND_SOC_ESS_ES9218P) += snd-soc-ess-es9218p.o obj-$(CONFIG_SND_SOC_GTM601) += snd-soc-gtm601.o obj-$(CONFIG_SND_SOC_HDAC_HDMI) += snd-soc-hdac-hdmi.o obj-$(CONFIG_SND_SOC_HDAC_HDA) += snd-soc-hdac-hda.o diff --git a/sound/soc/codecs/ess-es9218p.c b/sound/soc/codecs/ess-es9218p.c new file mode 100644 index 000000000000..bcaacd9cdc4b --- /dev/null +++ b/sound/soc/codecs/ess-es9218p.c @@ -0,0 +1,805 @@ +// SPDX-License-Identifier: GPL-2.0-or-later +/* + * Copyright (c) 2022-2023 Aidan MacDonald + */ + +#include <linux/bitfield.h> +#include <linux/clk.h> +#include <linux/delay.h> +#include <linux/gpio/consumer.h> +#include <linux/i2c.h> +#include <linux/module.h> +#include <linux/mod_devicetable.h> +#include <linux/regmap.h> +#include <linux/regulator/consumer.h> + +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> + +/* Read/write registers */ +#define ES9218P_SYSTEM 0x00 +#define ES9218P_INPUT_SEL 0x01 +#define ES9218P_MIX_AMUTE 0x02 +#define ES9218P_ANALOGVOL 0x03 +#define ES9218P_AMUTE_TIME 0x04 +#define ES9218P_AMUTE_LEVEL 0x05 +#define ES9218P_DOP_VOL_RAMP 0x06 +#define ES9218P_FBW_MUTE 0x07 +#define ES9218P_GPIO_CONFIG 0x08 +#define ES9218P_MODE_SYNC 0x0a +#define ES9218P_OC_PROTECT 0x0b +#define ES9218P_DPLL_BANDWITH 0x0c +#define ES9218P_THD_MONO_MODE 0x0d +#define ES9218P_SS_CONFIG 0x0e +#define ES9218P_VOL_CH1 0x0f +#define ES9218P_VOL_CH2 0x10 +#define ES9218P_MASTER_TRIM 0x11 /* 32-bit */ +#define ES9218P_GPIO_INPUT_SEL 0x15 +#define ES9218P_THD_C2 0x16 /* 16-bit */ +#define ES9218P_THD_C3 0x18 /* 16-bit */ +#define ES9218P_CP_SS_DELAY 0x1a +#define ES9218P_GEN_CONFIG 0x1b +#define ES9218P_GPIO_INV_ACG 0x1c +#define ES9218P_CP_CLOCK 0x1e /* 16-bit */ +#define ES9218P_AMP_CONFIG 0x20 +#define ES9218P_IMASK 0x21 +#define ES9218P_NCO 0x22 /* 32-bit */ +#define ES9218P_FIR_ADDR 0x28 +#define ES9218P_FIR_DATA 0x29 /* 24-bit */ +#define ES9218P_FIR_CONFIG 0x2c +#define ES9218P_ANALOG_OV_1 0x2d +#define ES9218P_ANALOG_OV_2 0x2e +#define ES9218P_ANALOG_OV_3 0x2f +#define ES9218P_ANALOG_CTRL 0x30 +#define ES9218P_ACG_THRESH 0x31 /* 24-bit */ +#define ES9218P_THD_C2_CH2 0x35 /* 16-bit */ +#define ES9218P_THD_C3_CH2 0x37 /* 16-bit */ + +/* Read only registers */ +#define ES9218P_CHIP_STATUS 0x40 +#define ES9218P_GPIO_READBACK 0x41 +#define ES9218P_DPLL_NUMBER 0x42 /* 32-bit */ +#define ES9218P_INPUT_STATUS 0x48 +#define ES9218P_RAM_READBACK 0x49 /* 24-bit */ + +#define ES9218P_MAX_REGISTER (ES9218P_RAM_READBACK + 2) + +#define ES9218P_SYSTEM_OSC_DRV GENMASK(7, 4) +#define ES9218P_SYSTEM_CLK_GEAR GENMASK(3, 2) + +#define ES9218P_INPUT_SEL_SER_LEN GENMASK(7, 6) +#define ES9218P_INPUT_SEL_SER_MODE GENMASK(5, 4) +#define ES9218P_INPUT_SEL_AUTOSELECT GENMASK(3, 2) +#define ES9218P_INPUT_SEL_DECODER GENMASK(1, 0) + +#define ES9218P_MODE_SYNC_MASTER BIT(7) + +#define ES9218P_SS_CONFIG_QUAD_RAMP BIT(5) +#define ES9218P_SS_CONFIG_TIME GENMASK(4, 0) + +#define ES9218P_GEN_CONFIG_ASRC_EN BIT(7) + +#define ES9218P_AMP_CONFIG_MODE GENMASK(1, 0) +#define ES9218P_AMP_MODE_CORE_ON 0 +#define ES9218P_AMP_MODE_LOWFI 1 +#define ES9218P_AMP_MODE_HIFI_1V 2 +#define ES9218P_AMP_MODE_HIFI_2V 3 + +enum es9218p_supply_id { + ES9218P_SUPPLY_AVDD, + ES9218P_SUPPLY_VCCA, + ES9218P_SUPPLY_AVCC3V3, + ES9218P_SUPPLY_AVCC1V8, + ES9218P_NUM_SUPPLIES +}; + +static const char * const es9218p_supply_names[ES9218P_NUM_SUPPLIES] = { + "avdd", + "vcca", + "avcc3v3", + "avcc1v8", +}; + +/** + * struct es9218p_priv - private data for ES9218P codec + * @supplies: number of supplies + * @mclk: clock for the MCLK/XI input + * @reset_gpio: GPIO for the RESETB pin + * @max_clk_div: maximum safe clock divider for MCLK + * @osc_drv_bias: oscillator drive pad bias + * @soft_start_time: requested soft start time (0-20) + * @output_2v: requested output voltage (true -> 2Vrms, false -> 1Vrms) + * @analog_bypass: requested audio path (true -> AUX input, false -> I2S input) + * @fsr: current sample rate + * @master_mode: true if the codec is supplying the bit and frame clocks + * @powered: true if the codec is powered up + * @out_mux_powered: true if the output mux is powered + * @dac_powered: true if the DAC is powered + */ +struct es9218p_priv { + struct regmap *regmap; + struct regulator_bulk_data supplies[ES9218P_NUM_SUPPLIES]; + struct clk *mclk; + struct gpio_desc *reset_gpio; + unsigned int max_clk_div; + unsigned int osc_drv_bias; + + unsigned int soft_start_time; + bool output_2v; + bool analog_bypass; + + unsigned int fsr; + bool master_mode; + bool powered; + bool out_mux_powered; + bool dac_powered; +}; + +static int es9218p_wide_write(struct regmap *regmap, unsigned int reg, + int count, unsigned int value) +{ + u8 data[4]; + int i; + + for (i = 0; i < count; i++) { + data[i] = value & 0xff; + value >>= 8; + } + + return regmap_raw_write(regmap, reg, data, count); +} + +static void es9218p_apply_clk_params(struct snd_soc_component *component) +{ + struct es9218p_priv *priv = snd_soc_component_get_drvdata(component); + unsigned int clk_div, clk_gear, nco, ss_time, max_ss_time; + unsigned long mclk; + + if (!priv->powered) + return; + + /* + * Calculate the maximum safe clock gear setting based on the + * constraint CLK >= 128*FSR and the platform-specific maximum. + */ + mclk = clk_get_rate(priv->mclk); + + clk_div = mclk / (128 * priv->fsr); + if (clk_div > priv->max_clk_div) + clk_div = priv->max_clk_div; + + if (clk_div >= 8) + clk_gear = 3; + else if (clk_div >= 4) + clk_gear = 2; + else if (clk_div >= 2) + clk_gear = 1; + else + clk_gear = 0; + + snd_soc_component_write_field(component, ES9218P_SYSTEM, + ES9218P_SYSTEM_CLK_GEAR, clk_gear); + + /* + * Calculate soft start time, adjusting for clock gear and clamping + * to ensure the time does not exceed 250ms. That limit is arbitrary, + * but we need to impose _some_ limit because the maximum timeout can + * be absurdly long (minutes to hours). + */ + if (priv->soft_start_time > clk_gear) + ss_time = priv->soft_start_time - clk_gear; + else + ss_time = 0; + + max_ss_time = ilog2((mclk >> clk_gear) / 4096 / 2 / 4); + if (ss_time > max_ss_time) + ss_time = max_ss_time; + + snd_soc_component_write_field(component, ES9218P_SS_CONFIG, + ES9218P_SS_CONFIG_TIME, ss_time); + + /* + * NCO sets the frame clock when in master mode, otherwise it's unused. + */ + if (priv->master_mode) + nco = div_u64((u64)priv->fsr << 32, mclk >> clk_gear); + else + nco = 0; + + snd_soc_component_write_field(component, ES9218P_MODE_SYNC, + ES9218P_MODE_SYNC_MASTER, priv->master_mode); + snd_soc_component_write_field(component, ES9218P_GEN_CONFIG, + ES9218P_GEN_CONFIG_ASRC_EN, !priv->master_mode); + es9218p_wide_write(priv->regmap, ES9218P_NCO, 4, nco); +} + +static int es9218p_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_component *component = dai->component; + struct es9218p_priv *priv = snd_soc_component_get_drvdata(component); + + /* Select I2S decoder, disable decoder autoselection */ + snd_soc_component_write_field(component, ES9218P_INPUT_SEL, + ES9218P_INPUT_SEL_AUTOSELECT, 0); + snd_soc_component_write_field(component, ES9218P_INPUT_SEL, + ES9218P_INPUT_SEL_DECODER, 0); + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + snd_soc_component_write_field(component, ES9218P_INPUT_SEL, + ES9218P_INPUT_SEL_SER_MODE, 0); + break; + case SND_SOC_DAIFMT_LEFT_J: + snd_soc_component_write_field(component, ES9218P_INPUT_SEL, + ES9218P_INPUT_SEL_SER_MODE, 1); + break; + case SND_SOC_DAIFMT_RIGHT_J: + snd_soc_component_write_field(component, ES9218P_INPUT_SEL, + ES9218P_INPUT_SEL_SER_MODE, 2); + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_BP_FP: + priv->master_mode = true; + break; + case SND_SOC_DAIFMT_BC_FC: + priv->master_mode = false; + break; + default: + return -EINVAL; + } + + es9218p_apply_clk_params(component); + return 0; +} + +static int es9218p_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 es9218p_priv *priv = snd_soc_component_get_drvdata(component); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + snd_soc_component_write_field(component, ES9218P_INPUT_SEL, + ES9218P_INPUT_SEL_SER_LEN, 0); + break; + case SNDRV_PCM_FORMAT_S24_LE: + snd_soc_component_write_field(component, ES9218P_INPUT_SEL, + ES9218P_INPUT_SEL_SER_LEN, 1); + break; + case SNDRV_PCM_FORMAT_S32_LE: + snd_soc_component_write_field(component, ES9218P_INPUT_SEL, + ES9218P_INPUT_SEL_SER_LEN, 2); + break; + default: + return -EINVAL; + } + + priv->fsr = params_rate(params); + + es9218p_apply_clk_params(component); + return 0; +} + +static int es9218p_mute_stream(struct snd_soc_dai *dai, int mute, int direction) +{ + struct snd_soc_component *component = dai->component; + + if (mute) + snd_soc_component_update_bits(component, ES9218P_FBW_MUTE, 0x01, 0x01); + else + snd_soc_component_update_bits(component, ES9218P_FBW_MUTE, 0x01, 0); + + return 0; +} + +static const struct snd_soc_dai_ops es9218p_dai_ops = { + .set_fmt = es9218p_set_fmt, + .hw_params = es9218p_hw_params, + .mute_stream = es9218p_mute_stream, +}; + +static struct snd_soc_dai_driver es9218p_dai = { + .name = "es9218p", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 8000, + .rate_max = 384000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE, + }, + .ops = &es9218p_dai_ops, +}; + +static int es9218p_power_up(struct snd_soc_component *component) +{ + struct es9218p_priv *priv = snd_soc_component_get_drvdata(component); + int ret; + + ret = regulator_bulk_enable(ARRAY_SIZE(priv->supplies), priv->supplies); + if (ret) { + dev_err(component->dev, "Cannot enable supplies: %d\n", ret); + return ret; + } + + ret = clk_prepare_enable(priv->mclk); + if (ret) { + dev_err(component->dev, "Cannot enable mclk: %d\n", ret); + goto err_regulator; + } + + gpiod_set_value_cansleep(priv->reset_gpio, 1); + usleep_range(1500, 2000); + + priv->powered = true; + + snd_soc_component_write_field(component, ES9218P_AMP_CONFIG, + ES9218P_AMP_CONFIG_MODE, ES9218P_AMP_MODE_CORE_ON); + snd_soc_component_write_field(component, ES9218P_SYSTEM, + ES9218P_SYSTEM_OSC_DRV, priv->osc_drv_bias); + es9218p_apply_clk_params(component); + + regcache_cache_only(priv->regmap, false); + ret = regcache_sync(priv->regmap); + if (ret) { + regcache_cache_only(priv->regmap, true); + goto err_clk_reset; + } + + return 0; + +err_clk_reset: + priv->powered = false; + gpiod_set_value_cansleep(priv->reset_gpio, 0); + clk_disable_unprepare(priv->mclk); +err_regulator: + regulator_bulk_disable(ARRAY_SIZE(priv->supplies), priv->supplies); + return ret; +} + +static void es9218p_power_down(struct snd_soc_component *component) +{ + struct es9218p_priv *priv = snd_soc_component_get_drvdata(component); + + regcache_cache_only(priv->regmap, true); + regcache_mark_dirty(priv->regmap); + + priv->powered = false; + + gpiod_set_value_cansleep(priv->reset_gpio, 0); + + clk_disable_unprepare(priv->mclk); + regulator_bulk_disable(ARRAY_SIZE(priv->supplies), priv->supplies); +} + +static int es9218p_set_bias_level(struct snd_soc_component *component, + enum snd_soc_bias_level level) +{ + int ret = 0; + + switch (level) { + case SND_SOC_BIAS_STANDBY: + if (snd_soc_component_get_bias_level(component) == SND_SOC_BIAS_OFF) + ret = es9218p_power_up(component); + break; + + case SND_SOC_BIAS_OFF: + es9218p_power_down(component); + break; + + default: + break; + } + + return ret; +} + +/* Returns soft start delay in microseconds */ +static unsigned long es9218p_get_soft_start_delay(struct snd_soc_component *component) +{ + struct es9218p_priv *priv = snd_soc_component_get_drvdata(component); + unsigned long mclk, clk_gear, ss_time; + + if (!priv->powered) + return 0; + + mclk = clk_get_rate(priv->mclk); + clk_gear = snd_soc_component_read_field(component, ES9218P_SYSTEM, + ES9218P_SYSTEM_CLK_GEAR); + ss_time = snd_soc_component_read_field(component, ES9218P_SS_CONFIG, + ES9218P_SS_CONFIG_TIME); + + return div_u64((1000000ull * 4096ull) << (1 + ss_time), mclk >> clk_gear); +} + +static void es9218p_set_amp_mode(struct snd_soc_component *component, int new_mode) +{ + unsigned long ss_delay; + int cur_mode, ret; + + cur_mode = snd_soc_component_read_field(component, ES9218P_AMP_CONFIG, + ES9218P_AMP_CONFIG_MODE); + if (cur_mode == new_mode) + return; + + ss_delay = es9218p_get_soft_start_delay(component); + ret = snd_soc_component_write_field(component, ES9218P_AMP_CONFIG, + ES9218P_AMP_CONFIG_MODE, + ES9218P_AMP_MODE_CORE_ON); + if (ret) { + /* + * We need an extra ~25ms delay here, otherwise the transition + * to the new mode doesn't work. + */ + usleep_range(ss_delay + 25000, ss_delay + 26000); + } + + ret = snd_soc_component_write_field(component, ES9218P_AMP_CONFIG, + ES9218P_AMP_CONFIG_MODE, new_mode); + if (ret) + usleep_range(ss_delay, ss_delay + 1000); +} + +static void es9218p_update_amp_mode(struct snd_soc_component *component) +{ + struct es9218p_priv *priv = snd_soc_component_get_drvdata(component); + int amp_mode; + + if (!priv->out_mux_powered) + amp_mode = ES9218P_AMP_MODE_CORE_ON; + else if (priv->analog_bypass) + amp_mode = ES9218P_AMP_MODE_LOWFI; + else if (!priv->dac_powered) + amp_mode = ES9218P_AMP_MODE_CORE_ON; + else if (priv->output_2v) + amp_mode = ES9218P_AMP_MODE_HIFI_2V; + else + amp_mode = ES9218P_AMP_MODE_HIFI_1V; + + es9218p_set_amp_mode(component, amp_mode); +} + +static int es9218p_output_dapm_event(struct snd_soc_dapm_widget *widget, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = snd_soc_dapm_to_component(widget->dapm); + struct es9218p_priv *priv = snd_soc_component_get_drvdata(component); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + if (widget->id == snd_soc_dapm_dac) + priv->dac_powered = true; + else + priv->out_mux_powered = true; + break; + + case SND_SOC_DAPM_POST_PMD: + if (widget->id == snd_soc_dapm_dac) + priv->dac_powered = false; + else + priv->out_mux_powered = false; + break; + + default: + return 0; + } + + es9218p_update_amp_mode(component); + return 0; +} + +static int outlevel_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct es9218p_priv *priv = snd_soc_component_get_drvdata(component); + + ucontrol->value.enumerated.item[0] = priv->output_2v; + return 0; +} + +static int outlevel_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct es9218p_priv *priv = snd_soc_component_get_drvdata(component); + + priv->output_2v = ucontrol->value.enumerated.item[0]; + es9218p_update_amp_mode(component); + return 1; +} + +static int softstart_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct es9218p_priv *priv = snd_soc_component_get_drvdata(component); + + ucontrol->value.integer.value[0] = priv->soft_start_time; + return 0; +} + +static int softstart_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct es9218p_priv *priv = snd_soc_component_get_drvdata(component); + + priv->soft_start_time = ucontrol->value.integer.value[0]; + es9218p_apply_clk_params(component); + return 1; +} + +static int output_mux_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_dapm_kcontrol_component(kcontrol); + struct es9218p_priv *priv = snd_soc_component_get_drvdata(component); + + ucontrol->value.enumerated.item[0] = priv->analog_bypass; + return 0; +} + +static int output_mux_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_dapm_kcontrol_component(kcontrol); + struct snd_soc_dapm_context *dapm = snd_soc_dapm_kcontrol_dapm(kcontrol); + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + struct es9218p_priv *priv = snd_soc_component_get_drvdata(component); + unsigned int value = ucontrol->value.enumerated.item[0]; + + priv->analog_bypass = value; + es9218p_update_amp_mode(component); + + snd_soc_dapm_mux_update_power(dapm, kcontrol, value, e, NULL); + return 1; +} + +static DECLARE_TLV_DB_SCALE(analog_vol_scale, -2400, 100, 0); +static DECLARE_TLV_DB_SCALE(digital_vol_scale, -12750, 50, 0); + +static const char * const digfilter_texts[] = { + "Linear Phase Fast", + "Linear Phase Slow", + "Minimum Phase Fast", + "Minimum Phase Slow", + "Apodizing Fast Type 1", + "Apodizing Fast Type 2", + "Hybrid Fast", + "Brick Wall" +}; +static SOC_ENUM_SINGLE_DECL(digfilter_enum, ES9218P_FBW_MUTE, 5, digfilter_texts); + +static const char * const outlevel_texts[] = { "1V", "2V" }; +static SOC_ENUM_SINGLE_EXT_DECL(outlevel_enum, outlevel_texts); + +static const char * const channel_mux_texts[] = { "CH1", "CH2" }; +static SOC_ENUM_DOUBLE_DECL(channel_mux_enum, ES9218P_MIX_AMUTE, 0, 2, channel_mux_texts); + +static const struct snd_kcontrol_new es9218p_controls[] = { + SOC_SINGLE_TLV("Headphone Playback Volume", + ES9218P_ANALOGVOL, 0, 24, 1, analog_vol_scale), + SOC_DOUBLE_R_TLV("Master Playback Volume", + ES9218P_VOL_CH1, ES9218P_VOL_CH2, 0, 0xff, 1, digital_vol_scale), + SOC_ENUM("Digital Filter Roll-Off", digfilter_enum), + SOC_ENUM_EXT("Output Level", outlevel_enum, outlevel_get, outlevel_put), + SOC_ENUM("Channel Mux", channel_mux_enum), + SOC_SINGLE_EXT("Soft Start Time", SND_SOC_NOPM, 0, 20, 0, softstart_get, softstart_put), +}; + +static const char * const output_mux_texts[] = { "Digital Input", "AUX Input" }; +static SOC_ENUM_SINGLE_EXT_DECL(output_mux_enum, output_mux_texts); + +static const struct snd_kcontrol_new output_mux_ctl = + SOC_DAPM_ENUM_EXT("Output Source", output_mux_enum, output_mux_get, output_mux_put); + +static const struct snd_soc_dapm_widget es9218p_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("AUX_L"), + SND_SOC_DAPM_INPUT("AUX_R"), + + SND_SOC_DAPM_DAC_E("DAC", "Playback", SND_SOC_NOPM, 0, 0, + es9218p_output_dapm_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + + SND_SOC_DAPM_MUX_E("Output Mux", SND_SOC_NOPM, 0, 0, + &output_mux_ctl, es9218p_output_dapm_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + + SND_SOC_DAPM_OUTPUT("OUTL"), + SND_SOC_DAPM_OUTPUT("OUTR"), + +}; + +static const struct snd_soc_dapm_route es9218p_dapm_routes[] = { + {"Output Mux", "Digital Input", "DAC"}, + {"Output Mux", "AUX Input", "AUX_L"}, + {"Output Mux", "AUX Input", "AUX_R"}, + {"OUTL", NULL, "Output Mux"}, + {"OUTR", NULL, "Output Mux"}, +}; + +static const struct snd_soc_component_driver es9218p_component = { + .set_bias_level = es9218p_set_bias_level, + .controls = es9218p_controls, + .num_controls = ARRAY_SIZE(es9218p_controls), + .dapm_widgets = es9218p_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(es9218p_dapm_widgets), + .dapm_routes = es9218p_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(es9218p_dapm_routes), +}; + +static const struct regmap_range es9218p_readable_ranges[] = { + regmap_reg_range(ES9218P_SYSTEM, ES9218P_THD_C3_CH2+1), + regmap_reg_range(ES9218P_CHIP_STATUS, ES9218P_MAX_REGISTER), +}; + +static const struct regmap_access_table es9218p_readable_table = { + .yes_ranges = es9218p_readable_ranges, + .n_yes_ranges = ARRAY_SIZE(es9218p_readable_ranges), +}; + +static const struct regmap_range es9218p_writeable_ranges[] = { + regmap_reg_range(ES9218P_SYSTEM, ES9218P_THD_C3_CH2+1), +}; + +static const struct regmap_access_table es9218p_writeable_table = { + .yes_ranges = es9218p_writeable_ranges, + .n_yes_ranges = ARRAY_SIZE(es9218p_writeable_ranges), +}; + +static const struct regmap_range es9218p_volatile_ranges[] = { + regmap_reg_range(ES9218P_FIR_ADDR, ES9218P_FIR_CONFIG), + regmap_reg_range(ES9218P_CHIP_STATUS, ES9218P_DPLL_NUMBER+3), + regmap_reg_range(ES9218P_INPUT_STATUS, ES9218P_RAM_READBACK+2), +}; + +static const struct regmap_access_table es9218p_volatile_table = { + .yes_ranges = es9218p_volatile_ranges, + .n_yes_ranges = ARRAY_SIZE(es9218p_volatile_ranges), +}; + +static const u8 es9218p_reg_defaults[] = { + 0x00, 0x8c, 0x34, 0x58, 0x00, 0x68, 0x72, 0x80, + 0xdd, 0x18, 0x02, 0x00, 0x5a, 0x40, 0x0a, 0x50, + 0x50, 0xff, 0xff, 0xff, 0x7f, 0x0d, 0x00, 0x00, + 0x00, 0x00, 0x62, 0xc4, 0xf0, 0x00, 0x00, 0x00, + 0x00, 0x3c, 0x00, 0x00, 0x00, 0x00, 0x00, 0xcc, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x01, 0x62, 0xc0, 0x0d, 0x62, 0x00, 0x00, 0x00, + 0x00, +}; + +static const struct regmap_config es9218p_regmap = { + .reg_bits = 8, + .val_bits = 8, + .rd_table = &es9218p_readable_table, + .wr_table = &es9218p_writeable_table, + .volatile_table = &es9218p_volatile_table, + .max_register = ES9218P_MAX_REGISTER, + .reg_defaults_raw = es9218p_reg_defaults, + .num_reg_defaults_raw = ARRAY_SIZE(es9218p_reg_defaults), + .cache_type = REGCACHE_RBTREE, +}; + +static int es9218p_read_properties(struct device *dev, + struct es9218p_priv *priv) +{ + u32 val; + + /* + * NOTE: This property determines the maximum value of the MCLK/XI + * divider that supplies the codec internal CLK. A lower CLK saves + * power so we want to use the highest possible divider at all times. + * + * We _should_ be able to calculate the clock divider automatically + * because CLK is constrained only by the sample rate and I2C bus + * frequency: + * + * MCLK / DIV = CLK >= max { 128 * FSR, 20 * I2C_BUS_FREQ } + * + * The I2C subsystem doesn't yet provide a mechanism to query the + * bus frequency, so we can't implement the automatic calculation. + * Instead, ess,max-clock-div sets the maximum value of DIV that + * ensures CLK >= 20 * I2C_BUS_FREQ, and we use FSR to determine + * the value of DIV within the range [1, ess,max-clock-div]. + * + * If/when there's a way to query I2C bus frequency this property + * can be removed. + */ + if (!device_property_read_u32(dev, "ess,max-clock-div", &val)) + priv->max_clk_div = val; + + if (!device_property_read_u32(dev, "ess,oscillator-pad-bias", &val)) + priv->osc_drv_bias = val; + + return 0; +} + +static int es9218p_i2c_probe(struct i2c_client *i2c) +{ + struct device *dev = &i2c->dev; + struct es9218p_priv *priv; + int ret, i; + + priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + priv->regmap = devm_regmap_init_i2c(i2c, &es9218p_regmap); + if (IS_ERR(priv->regmap)) + return dev_err_probe(dev, PTR_ERR(priv->regmap), "Failed to create regmap\n"); + + /* Codec is only powered when needed */ + regcache_cache_only(priv->regmap, true); + + for (i = 0; i < ES9218P_NUM_SUPPLIES; i++) + priv->supplies[i].supply = es9218p_supply_names[i]; + + ret = devm_regulator_bulk_get(dev, ES9218P_NUM_SUPPLIES, priv->supplies); + if (ret) + return dev_err_probe(dev, ret, "Failed to request supplies\n"); + + priv->mclk = devm_clk_get(dev, NULL); + if (IS_ERR(priv->mclk)) + return dev_err_probe(dev, PTR_ERR(priv->mclk), "Failed to request MCLK\n"); + + priv->reset_gpio = devm_gpiod_get(dev, "reset", GPIOD_OUT_LOW); + if (IS_ERR(priv->reset_gpio)) + return dev_err_probe(dev, PTR_ERR(priv->reset_gpio), + "Failed to request reset GPIO\n"); + + ret = es9218p_read_properties(dev, priv); + if (ret) + return ret; + + /* Arbitrary; must be nonzero */ + priv->fsr = 48000; + + i2c_set_clientdata(i2c, priv); + + return devm_snd_soc_register_component(dev, &es9218p_component, &es9218p_dai, 1); +} + +static const struct of_device_id __maybe_unused es9218p_of_match[] = { + { .compatible = "ess,es9218p" }, + { } +}; +MODULE_DEVICE_TABLE(of, es9218p_of_match); + +static const struct i2c_device_id es9218p_i2c_id[] = { + { "ess-es9218p", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, es9218p_i2c_id); + +static struct i2c_driver es9218p_i2c_driver = { + .driver = { + .name = "ess-es9218p", + .of_match_table = of_match_ptr(es9218p_of_match), + }, + .probe = es9218p_i2c_probe, + .id_table = es9218p_i2c_id, +}; +module_i2c_driver(es9218p_i2c_driver); + +MODULE_DESCRIPTION("ESS ES9218P ALSA SoC Codec Driver"); +MODULE_AUTHOR("Aidan MacDonald <aidanmacdonald.0x0(a)gmail.com>"); +MODULE_LICENSE("GPL"); -- 2.39.2

3 4

[PATCH] ASoC: cs35l56: Prevent unbalanced pm_runtime in dsp_work() on SoundWire
by Richard Fitzgerald 15 May '23

15 May '23

From: Simon Trimmer <simont(a)opensource.cirrus.com> Flush the SoundWire interrupt handler work instead of cancelling it. When a SoundWire interrupt is triggered the pm_runtime is held until the work has completed. It's therefore unsafe to cancel the work, it must be flushed. Signed-off-by: Simon Trimmer <simont(a)opensource.cirrus.com> Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> --- sound/soc/codecs/cs35l56.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs35l56.c b/sound/soc/codecs/cs35l56.c index 46762f7f1449..d1677d76d018 100644 --- a/sound/soc/codecs/cs35l56.c +++ b/sound/soc/codecs/cs35l56.c @@ -852,10 +852,11 @@ static void cs35l56_dsp_work(struct work_struct *work) */ if (cs35l56->sdw_peripheral) { cs35l56->sdw_irq_no_unmask = true; - cancel_work_sync(&cs35l56->sdw_irq_work); + flush_work(&cs35l56->sdw_irq_work); sdw_write_no_pm(cs35l56->sdw_peripheral, CS35L56_SDW_GEN_INT_MASK_1, 0); sdw_read_no_pm(cs35l56->sdw_peripheral, CS35L56_SDW_GEN_INT_STAT_1); sdw_write_no_pm(cs35l56->sdw_peripheral, CS35L56_SDW_GEN_INT_STAT_1, 0xFF); + flush_work(&cs35l56->sdw_irq_work); } ret = cs35l56_mbox_send(cs35l56, CS35L56_MBOX_CMD_SHUTDOWN); -- 2.30.2

2 1

[PATCH RESEND] MAINTAINERS: Remove self from Cirrus Codec drivers
by Richard Fitzgerald 15 May '23

15 May '23

From: Lucas Tanure <tanureal(a)opensource.cirrus.com> I'm leaving Cirrus Logic, and will no longer have access to hardware and documentation necessary to be effective in a maintainership role. Signed-off-by: Lucas Tanure <tanureal(a)opensource.cirrus.com> Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> --- MAINTAINERS | 1 - 1 file changed, 1 deletion(-) diff --git a/MAINTAINERS b/MAINTAINERS index c919c282f599..5c37070138f2 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -4923,7 +4923,6 @@ F: drivers/media/cec/i2c/ch7322.c CIRRUS LOGIC AUDIO CODEC DRIVERS M: James Schulman <james.schulman(a)cirrus.com> M: David Rhodes <david.rhodes(a)cirrus.com> -M: Lucas Tanure <tanureal(a)opensource.cirrus.com> M: Richard Fitzgerald <rf(a)opensource.cirrus.com> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) L: patches(a)opensource.cirrus.com -- 2.30.2

2 1

[PATCH] ASoC: mediatek: mt8186: Fix use-after-free in driver remove path
by Douglas Anderson 15 May '23

15 May '23

When devm runs function in the "remove" path for a device it runs them in the reverse order. That means that if you have parts of your driver that aren't using devm or are using "roll your own" devm w/ devm_add_action_or_reset() you need to keep that in mind. The mt8186 audio driver didn't quite get this right. Specifically, in mt8186_init_clock() it called mt8186_audsys_clk_register() and then went on to call a bunch of other devm function. The caller of mt8186_init_clock() used devm_add_action_or_reset() to call mt8186_deinit_clock() but, because of the intervening devm functions, the order was wrong. Specifically at probe time, the order was: 1. mt8186_audsys_clk_register() 2. afe_priv->clk = devm_kcalloc(...) 3. afe_priv->clk[i] = devm_clk_get(...) At remove time, the order (which should have been 3, 2, 1) was: 1. mt8186_audsys_clk_unregister() 3. Free all of afe_priv->clk[i] 2. Free afe_priv->clk The above seemed to be causing a use-after-free. Luckily, it's easy to fix this by simply using devm more correctly. Let's move the devm_add_action_or_reset() to the right place. In addition to fixing the use-after-free, code inspection shows that this fixes a leak (missing call to mt8186_audsys_clk_unregister()) that would have happened if any of the syscon_regmap_lookup_by_phandle() calls in mt8186_init_clock() had failed. Fixes: 55b423d5623c ("ASoC: mediatek: mt8186: support audio clock control in platform driver") Signed-off-by: Douglas Anderson <dianders(a)chromium.org> --- sound/soc/mediatek/mt8186/mt8186-afe-clk.c | 6 --- sound/soc/mediatek/mt8186/mt8186-afe-clk.h | 1 - sound/soc/mediatek/mt8186/mt8186-afe-pcm.c | 4 -- sound/soc/mediatek/mt8186/mt8186-audsys-clk.c | 46 ++++++++++--------- sound/soc/mediatek/mt8186/mt8186-audsys-clk.h | 1 - 5 files changed, 24 insertions(+), 34 deletions(-) diff --git a/sound/soc/mediatek/mt8186/mt8186-afe-clk.c b/sound/soc/mediatek/mt8186/mt8186-afe-clk.c index a6b4f29049bb..539e3a023bc4 100644 --- a/sound/soc/mediatek/mt8186/mt8186-afe-clk.c +++ b/sound/soc/mediatek/mt8186/mt8186-afe-clk.c @@ -644,9 +644,3 @@ int mt8186_init_clock(struct mtk_base_afe *afe) return 0; } - -void mt8186_deinit_clock(void *priv) -{ - struct mtk_base_afe *afe = priv; - mt8186_audsys_clk_unregister(afe); -} diff --git a/sound/soc/mediatek/mt8186/mt8186-afe-clk.h b/sound/soc/mediatek/mt8186/mt8186-afe-clk.h index d5988717d8f2..a9d59e506d9a 100644 --- a/sound/soc/mediatek/mt8186/mt8186-afe-clk.h +++ b/sound/soc/mediatek/mt8186/mt8186-afe-clk.h @@ -81,7 +81,6 @@ enum { struct mtk_base_afe; int mt8186_set_audio_int_bus_parent(struct mtk_base_afe *afe, int clk_id); int mt8186_init_clock(struct mtk_base_afe *afe); -void mt8186_deinit_clock(void *priv); int mt8186_afe_enable_cgs(struct mtk_base_afe *afe); void mt8186_afe_disable_cgs(struct mtk_base_afe *afe); int mt8186_afe_enable_clock(struct mtk_base_afe *afe); diff --git a/sound/soc/mediatek/mt8186/mt8186-afe-pcm.c b/sound/soc/mediatek/mt8186/mt8186-afe-pcm.c index 41172a82103e..a868a04ed4e7 100644 --- a/sound/soc/mediatek/mt8186/mt8186-afe-pcm.c +++ b/sound/soc/mediatek/mt8186/mt8186-afe-pcm.c @@ -2848,10 +2848,6 @@ static int mt8186_afe_pcm_dev_probe(struct platform_device *pdev) return ret; } - ret = devm_add_action_or_reset(dev, mt8186_deinit_clock, (void *)afe); - if (ret) - return ret; - /* init memif */ afe->memif_32bit_supported = 0; afe->memif_size = MT8186_MEMIF_NUM; diff --git a/sound/soc/mediatek/mt8186/mt8186-audsys-clk.c b/sound/soc/mediatek/mt8186/mt8186-audsys-clk.c index 578969ca91c8..5666be6b1bd2 100644 --- a/sound/soc/mediatek/mt8186/mt8186-audsys-clk.c +++ b/sound/soc/mediatek/mt8186/mt8186-audsys-clk.c @@ -84,6 +84,29 @@ static const struct afe_gate aud_clks[CLK_AUD_NR_CLK] = { GATE_AUD2(CLK_AUD_ETDM_OUT1_BCLK, "aud_etdm_out1_bclk", "top_audio", 24), }; +static void mt8186_audsys_clk_unregister(void *data) +{ + struct mtk_base_afe *afe = data; + struct mt8186_afe_private *afe_priv = afe->platform_priv; + struct clk *clk; + struct clk_lookup *cl; + int i; + + if (!afe_priv) + return; + + for (i = 0; i < CLK_AUD_NR_CLK; i++) { + cl = afe_priv->lookup[i]; + if (!cl) + continue; + + clk = cl->clk; + clk_unregister_gate(clk); + + clkdev_drop(cl); + } +} + int mt8186_audsys_clk_register(struct mtk_base_afe *afe) { struct mt8186_afe_private *afe_priv = afe->platform_priv; @@ -124,27 +147,6 @@ int mt8186_audsys_clk_register(struct mtk_base_afe *afe) afe_priv->lookup[i] = cl; } - return 0; + return devm_add_action_or_reset(afe->dev, mt8186_audsys_clk_unregister, afe); } -void mt8186_audsys_clk_unregister(struct mtk_base_afe *afe) -{ - struct mt8186_afe_private *afe_priv = afe->platform_priv; - struct clk *clk; - struct clk_lookup *cl; - int i; - - if (!afe_priv) - return; - - for (i = 0; i < CLK_AUD_NR_CLK; i++) { - cl = afe_priv->lookup[i]; - if (!cl) - continue; - - clk = cl->clk; - clk_unregister_gate(clk); - - clkdev_drop(cl); - } -} diff --git a/sound/soc/mediatek/mt8186/mt8186-audsys-clk.h b/sound/soc/mediatek/mt8186/mt8186-audsys-clk.h index b8d6a06e11e8..897a2914dc19 100644 --- a/sound/soc/mediatek/mt8186/mt8186-audsys-clk.h +++ b/sound/soc/mediatek/mt8186/mt8186-audsys-clk.h @@ -10,6 +10,5 @@ #define _MT8186_AUDSYS_CLK_H_ int mt8186_audsys_clk_register(struct mtk_base_afe *afe); -void mt8186_audsys_clk_unregister(struct mtk_base_afe *afe); #endif -- 2.40.1.606.ga4b1b128d6-goog

2 1

[PATCH 0/3] ASoC: SOF: Various runtime pm fixes, improvements
by Peter Ujfalusi 15 May '23

15 May '23

Three patch to correct error path PM runtime handling in few places. Regards, Peter --- Pierre-Louis Bossart (3): ASoC: SOF: debug: conditionally bump runtime_pm counter on exceptions ASoC: SOF: pcm: fix pm_runtime imbalance in error handling ASoC: SOF: sof-client-probes: fix pm_runtime imbalance in error handling sound/soc/sof/debug.c | 4 ++-- sound/soc/sof/pcm.c | 11 ++++++----- sound/soc/sof/sof-client-probes.c | 14 ++++++++------ 3 files changed, 16 insertions(+), 13 deletions(-) -- 2.40.1

2 4

[PATCH] ASoC: SOF: pm: save io region state in case of errors in resume
by Peter Ujfalusi 15 May '23

15 May '23

From: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> If there are failures in DSP runtime resume, the device state will not reach active and this makes it impossible e.g. to retrieve a possible DSP panic dump via "exception" debugfs node. If CONFIG_SND_SOC_SOF_DEBUG_ENABLE_DEBUGFS_CACHE=y is set, the data in cache is stale. If debugfs cache is not used, the region simply cannot be read. To allow debugging these scenarios, update the debugfs cache contents in resume error handler. User-space can then later retrieve DSP panic and other state via debugfs (requires SOF debugfs cache to be enabled in build). Reported-by: Curtis Malainey <cujomalainey(a)chromium.org> Link: https://github.com/thesofproject/linux/issues/4274 Signed-off-by: Kai Vehmanen <kai.vehmanen(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> Reviewed-by: Curtis Malainey <cujomalainey(a)chromium.org> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- sound/soc/sof/pm.c | 14 +++++++++++++- 1 file changed, 13 insertions(+), 1 deletion(-) diff --git a/sound/soc/sof/pm.c b/sound/soc/sof/pm.c index 2fdbc53ca715..2b232442e84b 100644 --- a/sound/soc/sof/pm.c +++ b/sound/soc/sof/pm.c @@ -164,7 +164,7 @@ static int sof_resume(struct device *dev, bool runtime_resume) ret = tplg_ops->set_up_all_pipelines(sdev, false); if (ret < 0) { dev_err(sdev->dev, "Failed to restore pipeline after resume %d\n", ret); - return ret; + goto setup_fail; } } @@ -178,6 +178,18 @@ static int sof_resume(struct device *dev, bool runtime_resume) dev_err(sdev->dev, "ctx_restore IPC error during resume: %d\n", ret); } +setup_fail: +#if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_ENABLE_DEBUGFS_CACHE) + if (ret < 0) { + /* + * Debugfs cannot be read in runtime suspend, so cache + * the contents upon failure. This allows to capture + * possible DSP coredump information. + */ + sof_cache_debugfs(sdev); + } +#endif + return ret; } -- 2.40.1

2 1

[PATCH] ASoC: SOF: topology: Fix logic for copying tuples
by Peter Ujfalusi 15 May '23

15 May '23

From: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Topology could have more instances of the tokens being searched for than the number of sets that need to be copied. Stop copying token after the limit of number of token instances has been reached. This worked before only by chance as we had allocated more size for the tuples array than the number of actual tokens being parsed. Fixes: 7006d20e5e9d ("ASoC: SOF: Introduce IPC3 ops") Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- sound/soc/sof/topology.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c index d3d536b0a8f5..b80566570512 100644 --- a/sound/soc/sof/topology.c +++ b/sound/soc/sof/topology.c @@ -586,6 +586,10 @@ static int sof_copy_tuples(struct snd_sof_dev *sdev, struct snd_soc_tplg_vendor_ if (*num_copied_tuples == tuples_size) return 0; } + + /* stop when we've found the required token instances */ + if (found == num_tokens * token_instance_num) + return 0; } /* next array */ -- 2.40.1

2 1

[PATCH] ASoC: SOF: ipc3-topology: Make sure that only one cmd is sent in dai_config
by Peter Ujfalusi 15 May '23

15 May '23

The commands in sof_ipc_dai_config.flags are encoded as bits: 1 (bit0) - hw_params 2 (bit1) - hw_free 4 (bit2) - pause These are commands, they cannot be combined as one would assume, for example 3 (bit0 | bit1) is invalid. This can happen right at the second start of a stream as at the end of the first stream we set the hw_free command (bit1) and on the second start we would OR on top of it the hw_params (bit0). Fixes: b66bfc3a9810 ("ASoC: SOF: sof-audio: Fix broken early bclk feature for SSP") Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Daniel Baluta <daniel.baluta(a)nxp.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/sof/ipc3-topology.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/ipc3-topology.c b/sound/soc/sof/ipc3-topology.c index fc1eb8e2de2c..ba4ef290b634 100644 --- a/sound/soc/sof/ipc3-topology.c +++ b/sound/soc/sof/ipc3-topology.c @@ -2103,10 +2103,13 @@ static int sof_ipc3_dai_config(struct snd_sof_dev *sdev, struct snd_sof_widget * * For the case of PAUSE/HW_FREE, since there are no quirks, flags can be used as is. */ - if (flags & SOF_DAI_CONFIG_FLAGS_HW_PARAMS) + if (flags & SOF_DAI_CONFIG_FLAGS_HW_PARAMS) { + /* Clear stale command */ + config->flags &= ~SOF_DAI_CONFIG_FLAGS_CMD_MASK; config->flags |= flags; - else + } else { config->flags = flags; + } /* only send the IPC if the widget is set up in the DSP */ if (swidget->use_count > 0) { -- 2.40.1

2 1

[PATCH] ASoC: MAINTAINERS: drop Krzysztof Kozlowski from Samsung audio
by Krzysztof Kozlowski 15 May '23

15 May '23

Remove Krzysztof Kozlowski from maintainer of Samsung SoC Audio drivers and change the status to maintenance (no one is reality being paid for looking at this). Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- MAINTAINERS | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/MAINTAINERS b/MAINTAINERS index c7082004ac55..746ec43ff244 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -18581,10 +18581,9 @@ F: Documentation/admin-guide/LSM/SafeSetID.rst F: security/safesetid/ SAMSUNG AUDIO (ASoC) DRIVERS -M: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> M: Sylwester Nawrocki <s.nawrocki(a)samsung.com> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) -S: Supported +S: Maintained B: mailto:linux-samsung-soc@vger.kernel.org F: Documentation/devicetree/bindings/sound/samsung* F: sound/soc/samsung/ -- 2.34.1

3 3

[PATCH 1/2] pcm: hw: reinterpret the drain_silence setting
by Oswald Buddenhagen 15 May '23

15 May '23

It makes no sense to have a config option which determines the entire silencing time, because useful values strongly depend on the period size runtime setting. So instead we interpret the setting as an override for the "overhang" which is used to compensate FIFOs and IRQ delays. The reasonable worst case of this is determined by the hardware, so it makes sense to have this as a config option. In a next step, we may want to derive the default value from the declared FIFO size (whiche few drivers do) and the minimal period size (which is generally linked to the FIFO size). Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen(a)gmx.de> --- src/pcm/pcm_hw.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/src/pcm/pcm_hw.c b/src/pcm/pcm_hw.c index 8ffebed9..ecc47a76 100644 --- a/src/pcm/pcm_hw.c +++ b/src/pcm/pcm_hw.c @@ -737,18 +737,19 @@ static int snd_pcm_hw_drain(snd_pcm_t *pcm) { snd_pcm_hw_t *hw = pcm->private_data; snd_pcm_sw_params_t sw_params; + snd_pcm_uframes_t silence_slack; snd_pcm_uframes_t silence_size; int err; if (pcm->stream != SND_PCM_STREAM_PLAYBACK) goto __skip_silence; if (hw->drain_silence == 0 || hw->perfect_drain) goto __skip_silence; snd_pcm_sw_params_current_no_lock(pcm, &sw_params); - if (hw->drain_silence > 0) { - silence_size = (pcm->rate * hw->drain_silence) / 1000; - goto __manual_silence; - } + if (hw->drain_silence > 0) + silence_slack = (pcm->rate * hw->drain_silence) / 1000; + else + silence_slack = pcm->rate / 10; /* 1/10th of second */ /* compute end silence size, align to period size + extra time */ if ((pcm->boundary % pcm->period_size) == 0) { silence_size = pcm->period_size - (*pcm->appl.ptr % pcm->period_size); @@ -761,8 +762,7 @@ static int snd_pcm_hw_drain(snd_pcm_t *pcm) */ silence_size = pcm->period_size; } - silence_size += pcm->rate / 10; /* 1/10th of second */ -__manual_silence: + silence_size += silence_slack; if (sw_params.silence_size < silence_size) { /* fill the silence soon as possible (in the bellow ioctl * or the next period wake up) -- 2.40.0.152.g15d061e6df

2 3

[PATCH] ASoC: dt-bindings: nau8824: Convert to dtschema
by David Lin 15 May '23

15 May '23

Convert the NAU8824 audio CODEC bindings to DT schema. Signed-off-by: David Lin <CTLIN0(a)nuvoton.com> --- .../devicetree/bindings/sound/nau8824.txt | 88 --------- .../bindings/sound/nuvoton,nau8824.yaml | 182 ++++++++++++++++++ 2 files changed, 182 insertions(+), 88 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/nau8824.txt create mode 100644 Documentation/devicetree/bindings/sound/nuvoton,nau8824.yaml diff --git a/Documentation/devicetree/bindings/sound/nau8824.txt b/Documentation/devicetree/bindings/sound/nau8824.txt deleted file mode 100644 index e0058b97e49a..000000000000 --- a/Documentation/devicetree/bindings/sound/nau8824.txt +++ /dev/null @@ -1,88 +0,0 @@ -Nuvoton NAU8824 audio codec - -This device supports I2C only. - -Required properties: - - compatible : Must be "nuvoton,nau8824" - - - reg : the I2C address of the device. This is either 0x1a (CSB=0) or 0x1b (CSB=1). - -Optional properties: - - nuvoton,jkdet-polarity: JKDET pin polarity. 0 - active high, 1 - active low. - - - nuvoton,vref-impedance: VREF Impedance selection - 0 - Open - 1 - 25 kOhm - 2 - 125 kOhm - 3 - 2.5 kOhm - - - nuvoton,micbias-voltage: Micbias voltage level. - 0 - VDDA - 1 - VDDA - 2 - VDDA * 1.1 - 3 - VDDA * 1.2 - 4 - VDDA * 1.3 - 5 - VDDA * 1.4 - 6 - VDDA * 1.53 - 7 - VDDA * 1.53 - - - nuvoton,sar-threshold-num: Number of buttons supported - - nuvoton,sar-threshold: Impedance threshold for each button. Array that contains up to 8 buttons configuration. SAR value is calculated as - SAR = 255 * MICBIAS / SAR_VOLTAGE * R / (2000 + R) - where MICBIAS is configured by 'nuvoton,micbias-voltage', SAR_VOLTAGE is configured by 'nuvoton,sar-voltage', R - button impedance. - Refer datasheet section 10.2 for more information about threshold calculation. - - - nuvoton,sar-hysteresis: Button impedance measurement hysteresis. - - - nuvoton,sar-voltage: Reference voltage for button impedance measurement. - 0 - VDDA - 1 - VDDA - 2 - VDDA * 1.1 - 3 - VDDA * 1.2 - 4 - VDDA * 1.3 - 5 - VDDA * 1.4 - 6 - VDDA * 1.53 - 7 - VDDA * 1.53 - - - nuvoton,sar-compare-time: SAR compare time - 0 - 500 ns - 1 - 1 us - 2 - 2 us - 3 - 4 us - - - nuvoton,sar-sampling-time: SAR sampling time - 0 - 2 us - 1 - 4 us - 2 - 8 us - 3 - 16 us - - - nuvoton,short-key-debounce: Button short key press debounce time. - 0 - 30 ms - 1 - 50 ms - 2 - 100 ms - - - nuvoton,jack-eject-debounce: Jack ejection debounce time. - 0 - 0 ms - 1 - 1 ms - 2 - 10 ms - - -Example: - - headset: nau8824@1a { - compatible = "nuvoton,nau8824"; - reg = <0x1a>; - interrupt-parent = <&gpio>; - interrupts = <TEGRA_GPIO(E, 6) IRQ_TYPE_LEVEL_LOW>; - nuvoton,vref-impedance = <2>; - nuvoton,micbias-voltage = <6>; - // Setup 4 buttons impedance according to Android specification - nuvoton,sar-threshold-num = <4>; - nuvoton,sar-threshold = <0xc 0x1e 0x38 0x60>; - nuvoton,sar-hysteresis = <0>; - nuvoton,sar-voltage = <6>; - nuvoton,sar-compare-time = <1>; - nuvoton,sar-sampling-time = <1>; - nuvoton,short-key-debounce = <0>; - nuvoton,jack-eject-debounce = <1>; - }; diff --git a/Documentation/devicetree/bindings/sound/nuvoton,nau8824.yaml b/Documentation/devicetree/bindings/sound/nuvoton,nau8824.yaml new file mode 100644 index 000000000000..e03d7e2704b4 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nuvoton,nau8824.yaml @@ -0,0 +1,182 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nuvoton,nau8824.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NAU8824 audio CODEC + +maintainers: + - John Hsu <KCHSU0(a)nuvoton.com> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - nuvoton,nau8824 + + reg: + maxItems: 1 + + '#sound-dai-cells': + const: 0 + + interrupts: + maxItems: 1 + + nuvoton,jkdet-polarity: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + JKDET pin polarity. + enum: + - 0 # active high + - 1 # active low + default: 1 + + nuvoton,vref-impedance: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + VREF Impedance selection. + enum: + - 0 # Open + - 1 # 25 kOhm + - 2 # 125 kOhm + - 3 # 2.5 kOhm + default: 2 + + nuvoton,micbias-voltage: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + Micbias voltage level. + enum: + - 0 # VDDA + - 1 # VDDA + - 2 # VDDA * 1.1 + - 3 # VDDA * 1.2 + - 4 # VDDA * 1.3 + - 5 # VDDA * 1.4 + - 6 # VDDA * 1.53 + - 7 # VDDA * 1.53 + default: 6 + + nuvoton,sar-threshold-num: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + Number of buttons supported. + minimum: 1 + maximum: 4 + default: 4 + + nuvoton,sar-threshold: + $ref: /schemas/types.yaml#/definitions/uint32-array + description: + Impedance threshold for each button. Array that contains up to 8 buttons + configuration. SAR value is calculated as + SAR = 255 * MICBIAS / SAR_VOLTAGE * R / (2000 + R) where MICBIAS is + configured by 'nuvoton,micbias-voltage', SAR_VOLTAGE is configured by + 'nuvoton,sar-voltage', R - button impedance. + Refer datasheet section 10.2 for more information about threshold + calculation. + minItems: 1 + maxItems: 4 + items: + minimum: 0 + maximum: 255 + + nuvoton,sar-hysteresis: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + Button impedance measurement hysteresis. + default: 0 + + nuvoton,sar-voltage: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + Reference voltage for button impedance measurement. + enum: + - 0 # VDDA + - 1 # VDDA + - 2 # VDDA * 1.1 + - 3 # VDDA * 1.2 + - 4 # VDDA * 1.3 + - 5 # VDDA * 1.4 + - 6 # VDDA * 1.53 + - 7 # VDDA * 1.53 + default: 6 + + nuvoton,sar-compare-time: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + SAR compare time. + enum: + - 0 # 500ns + - 1 # 1us + - 2 # 2us + - 3 # 4us + default: 1 + + nuvoton,sar-sampling-time: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + SAR sampling time. + enum: + - 0 # 2us + - 1 # 4us + - 2 # 8us + - 3 # 16us + default: 1 + + nuvoton,short-key-debounce: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + Button short key press debounce time. + enum: + - 0 # 30 ms + - 1 # 50 ms + - 2 # 100 ms + default: 0 + + nuvoton,jack-eject-debounce: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + Jack ejection debounce time. + enum: + - 0 # 0 ms + - 1 # 1 ms + - 2 # 10 ms + default: 1 + +required: + - compatible + - reg + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + #include <dt-bindings/interrupt-controller/irq.h> + i2c { + #address-cells = <1>; + #size-cells = <0>; + codec@1a { + #sound-dai-cells = <0>; + compatible = "nuvoton,nau8824"; + reg = <0x1a>; + interrupt-parent = <&gpio>; + interrupts = <38 IRQ_TYPE_LEVEL_LOW>; + nuvoton,vref-impedance = <2>; + nuvoton,micbias-voltage = <6>; + nuvoton,sar-threshold-num = <4>; + // Setup 4 buttons impedance according to Android specification + nuvoton,sar-threshold = <0xc 0x1e 0x38 0x60>; + nuvoton,sar-hysteresis = <0>; + nuvoton,sar-voltage = <6>; + nuvoton,sar-compare-time = <1>; + nuvoton,sar-sampling-time = <1>; + nuvoton,short-key-debounce = <0>; + nuvoton,jack-eject-debounce = <1>; + }; + }; -- 2.25.1

3 4

[PATCH 1/3] docs: admin-guide: add valsa driver documentation
by Ivan Orlov 15 May '23

15 May '23

Add documentation for the new Virtual ALSA driver. It covers all possible usage cases: errors and delay injections, random and pattern-based data generation, playback and ioctl redefinition functionalities testing. We have a lot of different virtual media drivers, which can be used for testing of the userspace applications and media subsystem middle layer. However, all of them are aimed at testing the video functionality and simulating the video devices. For audio devices we have only snd-dummy module, which is good in simulating the correct behavior of an ALSA device. I decided to write a tool, which would help to test the userspace ALSA programs (and the PCM middle layer as well) under unusual circumstances to figure out how they would behave. So I came up with this Virtual ALSA Driver. This new Virtual ALSA Driver has several features which can be useful during the userspace ALSA applications testing/fuzzing, or testing/fuzzing of the PCM middle layer. Not all of them can be implemented using the existing virtual drivers (like dummy or loopback). Here is what can this driver do: - Simulate both capture and playback processes - Check the playback stream for containing the looped pattern - Generate random or pattern-based capture data - Inject delays into the playback and capturing processes - Inject errors during the PCM callbacks Also, this driver can check the playback stream for containing the predefined pattern, which is used in the corresponding selftest to check the PCM middle layer data transferring functionality. Additionally, this driver redefines the default RESET ioctl, and the selftest covers this PCM API functionality as well. Signed-off-by: Ivan Orlov <ivan.orlov0322(a)gmail.com> --- Documentation/admin-guide/index.rst | 1 + Documentation/admin-guide/valsa.rst | 114 ++++++++++++++++++++++++++++ 2 files changed, 115 insertions(+) create mode 100644 Documentation/admin-guide/valsa.rst diff --git a/Documentation/admin-guide/index.rst b/Documentation/admin-guide/index.rst index 43ea35613dfc..328cc59275a1 100644 --- a/Documentation/admin-guide/index.rst +++ b/Documentation/admin-guide/index.rst @@ -131,6 +131,7 @@ configure specific aspects of kernel behavior to your liking. thunderbolt ufs unicode + valsa vga-softcursor video-output xfs diff --git a/Documentation/admin-guide/valsa.rst b/Documentation/admin-guide/valsa.rst new file mode 100644 index 000000000000..64ffc130fb4c --- /dev/null +++ b/Documentation/admin-guide/valsa.rst @@ -0,0 +1,114 @@ +.. SPDX-License-Identifier: GPL-2.0 + +The Virtual ALSA Driver +======================= + +The Virtual ALSA Driver emulates a generic ALSA device, and can be used for +testing/fuzzing of the userspace ALSA applications, as well as for testing/fuzzing of +the ALSA middle layer. Additionally, it can be used for simulating hard to reproduce +problems with PCM devices. + +What can this driver do? +~~~~~~~~~~~~~~~~~~~~~~~~ + +At this moment the driver can do the following things: + * Simulate both capture and playback processes + * Generate random or pattern-based capturing data + * Inject delays into the playback and capturing processes + * Inject errors during the PCM callbacks + +Also, this driver can check the playback stream for containing the +predefined pattern, which is used in the corresponding selftest (alsa/valsa-test.sh) +to check the PCM middle layer data transferring functionality. Additionally, this +driver redefines the default RESET ioctl, and the selftest covers this PCM +API functionality as well. + +Configuration +------------- + +The driver has several parameters besides the common ALSA module parameters: + + * fill_mode (bool) - Buffer fill mode (see below) + * inject_delay (int) + * inject_hwpars_err (bool) + * inject_prepare_err (bool) + * inject_trigger_err (bool) + + +Capture Data Generation +----------------------- + +The driver has two modes of data generation: the first (0 in the fill_mode parameter) +means random data generation, the second (1 in the fill_mode) - pattern-based +data generation. Let's look at the second mode. + +First of all, you may want to specify the pattern for data generation. You can do it +by writing the pattern to the debugfs file (/sys/kernel/debug/valsa/fill_pattern). +Like that: + +.. code-block:: bash + + echo -n mycoolpattern > /sys/kernel/debug/valsa/fill_pattern + +After this, every capture action performed on the 'valsa' device will return +'mycoolpatternmycoolpatternmycoolpatternmy...' in the capturing buffer. + +The pattern itself can be up to 4096 bytes long. + +Delay injection +--------------- + +The driver has 'inject_delay' parameter, which has very self-descriptive name and +can be used for time delay/speedup simulations. The parameter has integer type, and +it means the delay added between module's internal timer ticks. + +If the 'inject_delay' value is positive, the buffer will be filled slower, if it is +negative - faster. You can try it yourself by starting a recording in any +audiorecording application (like Audacity) and selecting the 'valsa' device as a +source. + +This parameter can be also used for generating a huge amount of sound data in a very +short period of time (with the negative 'inject_delay' value). + +Errors injection +---------------- + +This module can be used for injecting errors into the PCM communication process. This +action can help you to figure out how the userspace ALSA program behaves under unusual +circumstances. + +For example, you can make all 'hw_params' PCM callback calls return EBUSY error by +writing '1' to the 'inject_hwpars_err' module parameter: + +.. code-block:: bash + + echo 1 > /sys/module/snd_valsa/parameters/inject_hwpars_err + +Errors can be injected into the following PCM callbacks: + + * hw_params (EBUSY) + * prepare (EINVAL) + * trigger (EINVAL) + + +Playback test +------------- + +This driver can be also used for the playback functionality testing - every time you +write the playback data to the 'valsa' PCM device and close it, the driver checks the +buffer for containing the looped pattern (which is specified in the fill_pattern +debugfs file). If the playback buffer content represents the looped pattern, 'pc_test' +debugfs entry is set into '1'. Otherwise, the driver sets it to '0'. + +ioctl redefinition test +----------------------- + +The driver redefines the 'reset' ioctl, which is default for all PCM devices. To test +this functionality, we can trigger the reset ioctl and check the 'ioctl_test' debugfs +entry: + +.. code-block:: bash + + cat /sys/kernel/debug/valsa/ioctl_test + +If the ioctl is triggered successfully, this file will contain '1', and '0' otherwise. -- 2.34.1

3 8

16-bit rate conversion broken with non-interleaved output
by GitHub issues - opened 14 May '23

14 May '23

alsa-project/alsa-lib issue #320 was opened from ossilator: this setup: ``` Plug PCM: Rate conversion PCM (44100, sformat=S16_LE) Converter: libspeex (external) Protocol version: 10003 Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 17832 period_size : 4458 period_time : 92879 tstamp_mode : NONE tstamp_type : MONOTONIC period_step : 1 avail_min : 4458 period_event : 0 start_threshold : 17832 stop_threshold : 17832 silence_threshold: 0 silence_size : 0 boundary : 5019261784704417792 Slave: Hardware PCM card 0 'E-MU 0404b PCI [MAEM8852]' device 3 subdevice 0 Its setup is: stream : PLAYBACK access : MMAP_NONINTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 44100 exact rate : 44100 (44100/1) msbits : 16 buffer_size : 16384 period_size : 4096 period_time : 92879 tstamp_mode : NONE tstamp_type : MONOTONIC period_step : 1 avail_min : 4096 period_event : 0 start_threshold : 16384 stop_threshold : 16384 silence_threshold: 0 silence_size : 0 boundary : 4611686018427387904 appl_ptr : 0 hw_ptr : 0 ``` garbles the output rather impressively, though it's still recognizable as the input, more or less. it happens both when up-sampling and down-sampling. it doesn't happen when not re-sampling. it doesn't happen when the hardware takes 32-bit samples, so i guess it's somewhere in the optimized s16 paths. it doesn't happen with mono. this is not reproducible with the upstream emu10k1 driver, as the multichannel device demands 16 channels, which causes the route conversion plugin to be inserted, which sidesteps the issue. and the regular pcm device accepts any rate. Issue URL : https://github.com/alsa-project/alsa-lib/issues/320 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Set C-Media USB 7.1 sound card to six_channel for surround40
by GitHub pull_request - opened 14 May '23

14 May '23

alsa-project/alsa-lib pull request #319 was opened from runderwo: This 7.1 analog sound card identifies as the following: 0d8c:0102 C-Media Electronics, Inc. CM106 Like Sound Device (lsusb) 1 [ICUSBAUDIO7D ]: USB-Audio - ICUSBAUDIO7D (/proc/asound/cards) It has the following four-channel analog stream layout: Interface 1 Altset 3 Format: S16_LE Channels: 4 Endpoint: 6 OUT (ADAPTIVE) Rates: 44100, 48000 Bits: 16 Channel map: FL FR FC LFE (/proc/asound/card1/stream0) This layout makes it impossible to play audio to the surround (SL/SR) speakers plugged into the "Surround" jack in 4 channel mode. Instead, set six_channel so that the six-channel layout will be used to access those speakers: Interface 1 Altset 4 Format: S16_LE Channels: 6 Endpoint: 6 OUT (ADAPTIVE) Rates: 44100, 48000 Bits: 16 Channel map: FL FR FC LFE SL SR Tested with speaker-test -Dsurround40:CARD=ICUSBAUDIO7D,DEV=0 -c4 The speaker system that required surround40 in order to downmix LFE into the front/rear mix is a Klipsch ProMedia v.2-400, which has 4 speakers and a subwoofer with an internal crossover, connected via front and rear stereo analog cables. Request URL : https://github.com/alsa-project/alsa-lib/pull/319 Patch URL : https://github.com/alsa-project/alsa-lib/pull/319.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

1.2.9: arithmetic exception in pcm/pcm_hw.c:753
by GitHub issues - opened 13 May '23

13 May '23

alsa-project/alsa-lib issue #318 was opened from engineeror: [`period_size`](https://github.com/alsa-project/alsa-lib/blob/904f0f7e7a29244c05e5874bc36486638d1f7e7f/src/pcm/pcm_hw.c#LL753C33-L753C33 ) is zero on USB DACs: https://bugs.archlinux.org/index.php?do=details&task_id=78476 @perexg Issue URL : https://github.com/alsa-project/alsa-lib/issues/318 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Update USB-Audio.conf for ASUS ROG Strix B650E-I Gaming WiFi and ALC4080
by GitHub pull_request - edited 13 May '23

13 May '23

alsa-project/alsa-ucm-conf pull request #315 was edited from Sunsheep: Added configuration for ASUS ROG Strix B650E-I Gaming WiFi and ALC4080 Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/315 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/315.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Update USB-Audio.conf
by GitHub pull_request - opened 13 May '23

13 May '23

alsa-project/alsa-ucm-conf pull request #315 was opened from Sunsheep: Added configuration for ASUS ROG Strix B650E-I Gaming WiFi and ALC4080 Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/315 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/315.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Newest Update for Fedora 37 (alsa-ucm-1.2.9-1.fc37) causes crackling with device
by GitHub issues - edited 12 May '23

12 May '23

alsa-project/alsa-ucm-conf issue #314 was edited from ClaraCrazy: First bug report here + not skilled with debugging, so please go easy on me :smile: OS: Fedora 37 Alsa-lib: 1.2.9-1 Issue: Audio crackling Audio Device: `Logic3 TX101 5.1 Soundbar` connected via `Scarlett 2i2 3rd Gen audio interface (USB)` Reverted lib back to `alsa-lib.1.2.8-2.fc37` (which in turn also downgraded alsa-ucm and alsa-utils again) and issues are fixed. I have no idea how to debug this, but would love to help in any way possible, sadly there was no issue template that would give me infos on what logs are useful / where to get those so uh.. please someone just point me in the right direction :) Thanks for your time Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/314 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] snd-hda-intel: Apply HP B&O top speaker profile to Pavilion 15
by Ryan C. Underwood 12 May '23

12 May '23

The Pavilion 15 line has B&O top speakers similar to the x360 and applying the same profile produces good sound. Without this, the sound would be tinny and underpowered without either applying model=alc295-hp-x360 or booting another OS first. Signed-off-by: Ryan Underwood <nemesis(a)icequake.net> --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e103bb3693c0..7cb232a23a08 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9335,7 +9335,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x802f, "HP Z240", ALC221_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x8077, "HP", ALC256_FIXUP_HP_HEADSET_MIC), SND_PCI_QUIRK(0x103c, 0x8158, "HP", ALC256_FIXUP_HP_HEADSET_MIC), - SND_PCI_QUIRK(0x103c, 0x820d, "HP Pavilion 15", ALC269_FIXUP_HP_MUTE_LED_MIC3), + SND_PCI_QUIRK(0x103c, 0x820d, "HP Pavilion 15", ALC295_FIXUP_HP_X360), SND_PCI_QUIRK(0x103c, 0x8256, "HP", ALC221_FIXUP_HP_FRONT_MIC), SND_PCI_QUIRK(0x103c, 0x827e, "HP x360", ALC295_FIXUP_HP_X360), SND_PCI_QUIRK(0x103c, 0x827f, "HP x360", ALC269_FIXUP_HP_MUTE_LED_MIC3), -- 2.39.2 -- Ryan C. Underwood, <nemesis(a)icequake.net>

2 2

[PATCH] ALSA: firewire-digi00x: prevent potential use after free
by Dan Carpenter 12 May '23

12 May '23

This code was supposed to return an error code if init_stream() failed, but it instead freed dg00x->rx_stream and returned success. This potentially leads to a use after free. Fixes: 9a08067ec318 ("ALSA: firewire-digi00x: support AMDTP domain") Signed-off-by: Dan Carpenter <dan.carpenter(a)linaro.org> --- sound/firewire/digi00x/digi00x-stream.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/firewire/digi00x/digi00x-stream.c b/sound/firewire/digi00x/digi00x-stream.c index a15f55b0dce3..295163bb8abb 100644 --- a/sound/firewire/digi00x/digi00x-stream.c +++ b/sound/firewire/digi00x/digi00x-stream.c @@ -259,8 +259,10 @@ int snd_dg00x_stream_init_duplex(struct snd_dg00x *dg00x) return err; err = init_stream(dg00x, &dg00x->tx_stream); - if (err < 0) + if (err < 0) { destroy_stream(dg00x, &dg00x->rx_stream); + return err; + } err = amdtp_domain_init(&dg00x->domain); if (err < 0) { -- 2.39.2

3 2

[PATCH] ASoC: SOF: topology: Fix missing error code in sof_route_load()
by Jiapeng Chong 12 May '23

12 May '23

The error code is missing in this code scenario, add the error code '-EINVAL' to the return value 'err'. sound/soc/sof/topology.c:2060 sof_route_load() warn: missing error code 'ret'. Reported-by: Abaci Robot <abaci(a)linux.alibaba.com> Link: https://bugzilla.openanolis.cn/show_bug.cgi?id=4935 Signed-off-by: Jiapeng Chong <jiapeng.chong(a)linux.alibaba.com> --- sound/soc/sof/topology.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c index d3d536b0a8f5..3ca3a72f1805 100644 --- a/sound/soc/sof/topology.c +++ b/sound/soc/sof/topology.c @@ -2045,7 +2045,6 @@ static int sof_route_load(struct snd_soc_component *scomp, int index, if (!source_swidget) { dev_err(scomp->dev, "error: source %s not found\n", route->source); - ret = -EINVAL; goto err; } @@ -2064,7 +2063,6 @@ static int sof_route_load(struct snd_soc_component *scomp, int index, if (!sink_swidget) { dev_err(scomp->dev, "error: sink %s not found\n", route->sink); - ret = -EINVAL; goto err; } @@ -2087,6 +2085,8 @@ static int sof_route_load(struct snd_soc_component *scomp, int index, return 0; err: kfree(sroute); + ret = -EINVAL; + return ret; } -- 2.20.1.7.g153144c

2 1

[PATCH 00/14] ALSA: emu10k1: various improvements to the DSP-based mixer code
by Oswald Buddenhagen 12 May '23

12 May '23

Oswald Buddenhagen (14): ALSA: emu10k1: make tone control switch mono ALSA: emu10k1: roll up loops in DSP setup code for Audigy ALSA: emu10k1: fix+optimize E-MU stereo capture DSP code ALSA: emu10k1: simplify snd_emu10k1_audigy_dsp_convert_32_to_2x16() ALSA: emu10k1: apply channel delay hack to all E-MU cards ALSA: emu10k1: simplify tone control switch DSP code ALSA: emu10k1: properly assert DSP init constraints ALSA: emu10k1: polish audigy GPR allocation ALSA: emu10k1: fix non-zero mixer control defaults in highres mode ALSA: emu10k1: validate min/max values of translated controls ALSA: emu10k1: omit non-applicable mixer controls for E-MU cards ALSA: emu10k1: skip mic capture PCM for cards without AC97 codec ALSA: emu10k1: enable bit-exact playback, part 1: DSP attenuation ALSA: emu10k1: enable bit-exact playback, part 2: voice attenuation Documentation/sound/cards/audigy-mixer.rst | 2 +- Documentation/sound/cards/sb-live-mixer.rst | 2 +- include/sound/emu10k1.h | 10 +- include/uapi/sound/emu10k1.h | 8 +- sound/pci/emu10k1/emu10k1.c | 8 +- sound/pci/emu10k1/emufx.c | 763 +++++++++----------- sound/pci/emu10k1/emumixer.c | 15 +- sound/pci/emu10k1/emupcm.c | 4 +- 8 files changed, 369 insertions(+), 443 deletions(-) -- 2.40.0.152.g15d061e6df

2 18

[PATCH] ALSA: hda/realtek: Fix mute and micmute LEDs for yet another HP laptop
by Kai-Heng Feng 12 May '23

12 May '23

There's yet another laptop that needs the fixup to enable mute and micmute LEDs. So do it accordingly. Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 172ffc2c332b..2e3f87e9b4ee 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9480,6 +9480,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8b8f, "HP", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8b92, "HP", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8b96, "HP", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), + SND_PCI_QUIRK(0x103c, 0x8b97, "HP", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), SND_PCI_QUIRK(0x103c, 0x8bf0, "HP", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300), -- 2.34.1

2 1

[PATCH] ALSA: usb-audio: Add a sample rate workaround for Line6 Pod Go
by Takashi Iwai 12 May '23

12 May '23

Line6 Pod Go (0e41:424b) requires the similar workaround for the fixed 48k sample rate like other Line6 models. This patch adds the corresponding entry to line6_parse_audio_format_rate_quirk(). Reported-by: John Humlick <john(a)humlick.org> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/format.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/format.c b/sound/usb/format.c index 4b1c5ba121f3..ab5fed9f55b6 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -423,6 +423,7 @@ static int line6_parse_audio_format_rates_quirk(struct snd_usb_audio *chip, case USB_ID(0x0e41, 0x4248): /* Line6 Helix >= fw 2.82 */ case USB_ID(0x0e41, 0x4249): /* Line6 Helix Rack >= fw 2.82 */ case USB_ID(0x0e41, 0x424a): /* Line6 Helix LT >= fw 2.82 */ + case USB_ID(0x0e41, 0x424b): /* Line6 Pod Go */ case USB_ID(0x19f7, 0x0011): /* Rode Rodecaster Pro */ return set_fixed_rate(fp, 48000, SNDRV_PCM_RATE_48000); } -- 2.35.3

1 0

[PATCH] ALSA: emu10k1: don't create regular S/PDIF controls for E-MU cards
by Oswald Buddenhagen 12 May '23

12 May '23

These ports are unused on these cards. Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen(a)gmx.de> --- sound/pci/emu10k1/emumixer.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/emu10k1/emumixer.c b/sound/pci/emu10k1/emumixer.c index 610700be1e37..48f0d3f8b8e7 100644 --- a/sound/pci/emu10k1/emumixer.c +++ b/sound/pci/emu10k1/emumixer.c @@ -2055,7 +2055,7 @@ int snd_emu10k1_mixer(struct snd_emu10k1 *emu, mix->attn[0] = 0xffff; } - if (! emu->card_capabilities->ecard) { /* FIXME: APS has these controls? */ + if (!emu->card_capabilities->ecard && !emu->card_capabilities->emu_model) { /* sb live! and audigy */ kctl = snd_ctl_new1(&snd_emu10k1_spdif_mask_control, emu); if (!kctl) -- 2.40.0.152.g15d061e6df

2 1

[PATCH 1/4] ALSA: emu10k1: automate encoding of sub-register definitions
by Oswald Buddenhagen 12 May '23

12 May '23

The idea to encode the bitfield manipulation in the register address is quite clever, but doing that by hand is ugly and error-prone. So derive it automatically from the mask instead. Macros cannot #define other macros, so we now declare enums instead. This also adds macros for decoding the register definitions. These will be used by later commits. Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen(a)gmx.de> --- include/sound/emu10k1.h | 113 ++++++++++++++++++---------------------- 1 file changed, 50 insertions(+), 63 deletions(-) diff --git a/include/sound/emu10k1.h b/include/sound/emu10k1.h index 8fe80dcee71b..f4e05a2897e8 100644 --- a/include/sound/emu10k1.h +++ b/include/sound/emu10k1.h @@ -38,6 +38,22 @@ #define IP_TO_CP(ip) ((ip == 0) ? 0 : (((0x00001000uL | (ip & 0x00000FFFL)) << (((ip >> 12) & 0x000FL) + 4)) & 0xFFFF0000uL)) +#define SUB_REG_NC(reg, field, mask) \ + enum { \ + field ## _MASK = mask, \ + field = reg | \ + (__builtin_ctz(mask) << 16) | \ + (__builtin_popcount(mask) << 24), \ + }; +#define SUB_REG(reg, field, mask) SUB_REG_NC(reg, reg ## _ ## field, mask) + +#define REG_SHIFT(r) (((r) >> 16) & 0x1f) +#define REG_SIZE(r) (((r) >> 24) & 0x1f) +#define REG_MASK0(r) ((1U << REG_SIZE(r)) - 1U) +#define REG_MASK(r) (REG_MASK0(r) << REG_SHIFT(r)) +#define REG_VAL_GET(r, v) ((v & REG_MASK(r)) >> REG_SHIFT(r)) +#define REG_VAL_PUT(r, v) ((v) << REG_SHIFT(r)) + // Audigy specify registers are prefixed with 'A_' /************************************************************************************************/ @@ -148,12 +164,10 @@ #define INTE_MIDIRXENABLE 0x00000001 /* Enable MIDI receive-buffer-empty interrupts */ #define WC 0x10 /* Wall Clock register */ -#define WC_SAMPLECOUNTER_MASK 0x03FFFFC0 /* Sample periods elapsed since reset */ -#define WC_SAMPLECOUNTER 0x14060010 -#define WC_CURRENTCHANNEL_MASK 0x0000003F /* Channel [0..63] currently being serviced */ +SUB_REG(WC, SAMPLECOUNTER, 0x03FFFFC0) /* Sample periods elapsed since reset */ +SUB_REG(WC, CURRENTCHANNEL, 0x0000003F) /* Channel [0..63] currently being serviced */ /* NOTE: Each channel takes 1/64th of a sample */ /* period to be serviced. */ -#define WC_CURRENTCHANNEL 0x06000010 #define HCFG 0x14 /* Hardware config register */ /* NOTE: There is no reason to use the legacy */ @@ -225,9 +239,8 @@ /* async audio source */ #define HCFG_LOCKSOUNDCACHE 0x00000008 /* 1 = Cancel bustmaster accesses to soundcache */ /* NOTE: This should generally never be used. */ -#define HCFG_LOCKTANKCACHE_MASK 0x00000004 /* 1 = Cancel bustmaster accesses to tankcache */ +SUB_REG(HCFG, LOCKTANKCACHE, 0x00000004) /* 1 = Cancel bustmaster accesses to tankcache */ /* NOTE: This should generally never be used. */ -#define HCFG_LOCKTANKCACHE 0x01020014 #define HCFG_MUTEBUTTONENABLE 0x00000002 /* 1 = Master mute button sets AUDIOENABLE = 0. */ /* NOTE: This is a 'cheap' way to implement a */ /* master mute function on the mute button, and */ @@ -382,55 +395,38 @@ // which the current registers "swerve" gradually. #define CPF 0x00 /* Current pitch and fraction register */ -#define CPF_CURRENTPITCH_MASK 0xffff0000 /* Current pitch (linear, 0x4000 == unity pitch shift) */ -#define CPF_CURRENTPITCH 0x10100000 +SUB_REG(CPF, CURRENTPITCH, 0xffff0000) /* Current pitch (linear, 0x4000 == unity pitch shift) */ #define CPF_STEREO_MASK 0x00008000 /* 1 = Even channel interleave, odd channel locked */ #define CPF_STOP_MASK 0x00004000 /* 1 = Current pitch forced to 0 */ #define CPF_FRACADDRESS_MASK 0x00003fff /* Linear fractional address of the current channel */ #define PTRX 0x01 /* Pitch target and send A/B amounts register */ -#define PTRX_PITCHTARGET_MASK 0xffff0000 /* Pitch target of specified channel */ -#define PTRX_PITCHTARGET 0x10100001 -#define PTRX_FXSENDAMOUNT_A_MASK 0x0000ff00 /* Linear level of channel output sent to FX send bus A */ -#define PTRX_FXSENDAMOUNT_A 0x08080001 -#define PTRX_FXSENDAMOUNT_B_MASK 0x000000ff /* Linear level of channel output sent to FX send bus B */ -#define PTRX_FXSENDAMOUNT_B 0x08000001 +SUB_REG(PTRX, PITCHTARGET, 0xffff0000) /* Pitch target of specified channel */ +SUB_REG(PTRX, FXSENDAMOUNT_A, 0x0000ff00) /* Linear level of channel output sent to FX send bus A */ +SUB_REG(PTRX, FXSENDAMOUNT_B, 0x000000ff) /* Linear level of channel output sent to FX send bus B */ #define CVCF 0x02 /* Current volume and filter cutoff register */ -#define CVCF_CURRENTVOL_MASK 0xffff0000 /* Current linear volume of specified channel */ -#define CVCF_CURRENTVOL 0x10100002 -#define CVCF_CURRENTFILTER_MASK 0x0000ffff /* Current filter cutoff frequency of specified channel */ -#define CVCF_CURRENTFILTER 0x10000002 +SUB_REG(CVCF, CURRENTVOL, 0xffff0000) /* Current linear volume of specified channel */ +SUB_REG(CVCF, CURRENTFILTER, 0x0000ffff) /* Current filter cutoff frequency of specified channel */ #define VTFT 0x03 /* Volume target and filter cutoff target register */ -#define VTFT_VOLUMETARGET_MASK 0xffff0000 /* Volume target of specified channel */ -#define VTFT_VOLUMETARGET 0x10100003 -#define VTFT_FILTERTARGET_MASK 0x0000ffff /* Filter cutoff target of specified channel */ -#define VTFT_FILTERTARGET 0x10000003 +SUB_REG(VTFT, VOLUMETARGET, 0xffff0000) /* Volume target of specified channel */ +SUB_REG(VTFT, FILTERTARGET, 0x0000ffff) /* Filter cutoff target of specified channel */ #define Z1 0x05 /* Filter delay memory 1 register */ #define Z2 0x04 /* Filter delay memory 2 register */ #define PSST 0x06 /* Send C amount and loop start address register */ -#define PSST_FXSENDAMOUNT_C_MASK 0xff000000 /* Linear level of channel output sent to FX send bus C */ - -#define PSST_FXSENDAMOUNT_C 0x08180006 - -#define PSST_LOOPSTARTADDR_MASK 0x00ffffff /* Loop start address of the specified channel */ -#define PSST_LOOPSTARTADDR 0x18000006 +SUB_REG(PSST, FXSENDAMOUNT_C, 0xff000000) /* Linear level of channel output sent to FX send bus C */ +SUB_REG(PSST, LOOPSTARTADDR, 0x00ffffff) /* Loop start address of the specified channel */ #define DSL 0x07 /* Send D amount and loop end address register */ -#define DSL_FXSENDAMOUNT_D_MASK 0xff000000 /* Linear level of channel output sent to FX send bus D */ - -#define DSL_FXSENDAMOUNT_D 0x08180007 - -#define DSL_LOOPENDADDR_MASK 0x00ffffff /* Loop end address of the specified channel */ -#define DSL_LOOPENDADDR 0x18000007 +SUB_REG(DSL, FXSENDAMOUNT_D, 0xff000000) /* Linear level of channel output sent to FX send bus D */ +SUB_REG(DSL, LOOPENDADDR, 0x00ffffff) /* Loop end address of the specified channel */ #define CCCA 0x08 /* Filter Q, interp. ROM, byte size, cur. addr register */ -#define CCCA_RESONANCE_MASK 0xf0000000 /* Lowpass filter resonance (Q) height */ -#define CCCA_RESONANCE 0x041c0008 +SUB_REG(CCCA, RESONANCE, 0xf0000000) /* Lowpass filter resonance (Q) height */ #define CCCA_INTERPROM_MASK 0x0e000000 /* Selects passband of interpolation ROM */ /* 1 == full band, 7 == lowpass */ /* ROM 0 is used when pitch shifting downward or less */ @@ -447,27 +443,24 @@ #define CCCA_INTERPROM_7 0x0e000000 /* Select interpolation ROM 7 */ #define CCCA_8BITSELECT 0x01000000 /* 1 = Sound memory for this channel uses 8-bit samples */ /* 8-bit samples are unsigned, 16-bit ones signed */ -#define CCCA_CURRADDR_MASK 0x00ffffff /* Current address of the selected channel */ -#define CCCA_CURRADDR 0x18000008 +SUB_REG(CCCA, CURRADDR, 0x00ffffff) /* Current address of the selected channel */ #define CCR 0x09 /* Cache control register */ -#define CCR_CACHEINVALIDSIZE 0x07190009 -#define CCR_CACHEINVALIDSIZE_MASK 0xfe000000 /* Number of invalid samples before the read address */ +SUB_REG(CCR, CACHEINVALIDSIZE, 0xfe000000) /* Number of invalid samples before the read address */ #define CCR_CACHELOOPFLAG 0x01000000 /* 1 = Cache has a loop service pending */ #define CCR_INTERLEAVEDSAMPLES 0x00800000 /* 1 = A cache service will fetch interleaved samples */ /* Auto-set from CPF_STEREO_MASK */ #define CCR_WORDSIZEDSAMPLES 0x00400000 /* 1 = A cache service will fetch word sized samples */ /* Auto-set from CCCA_8BITSELECT */ -#define CCR_READADDRESS 0x06100009 -#define CCR_READADDRESS_MASK 0x003f0000 /* Next cached sample to play */ -#define CCR_LOOPINVALSIZE 0x0000fe00 /* Number of invalid samples in cache prior to loop */ +SUB_REG(CCR, READADDRESS, 0x003f0000) /* Next cached sample to play */ +SUB_REG(CCR, LOOPINVALSIZE, 0x0000fe00) /* Number of invalid samples in cache prior to loop */ /* NOTE: This is valid only if CACHELOOPFLAG is set */ #define CCR_LOOPFLAG 0x00000100 /* Set for a single sample period when a loop occurs */ -#define CCR_CACHELOOPADDRHI 0x000000ff /* CLP_LOOPSTARTADDR's hi byte if CACHELOOPFLAG is set */ +SUB_REG(CCR, CACHELOOPADDRHI, 0x000000ff) /* CLP_LOOPSTARTADDR's hi byte if CACHELOOPFLAG is set */ #define CLP 0x0a /* Cache loop register (valid if CCR_CACHELOOPFLAG = 1) */ /* NOTE: This register is normally not used */ -#define CLP_CACHELOOPADDR 0x0000ffff /* Cache loop address low word */ +SUB_REG(CLP, CACHELOOPADDR, 0x0000ffff) /* Cache loop address low word */ #define FXRT 0x0b /* Effects send routing register */ /* NOTE: It is illegal to assign the same routing to */ @@ -537,20 +530,17 @@ #define IP_UNITY 0x0000e000 /* Unity pitch shift */ #define IFATN 0x19 /* Initial filter cutoff and attenuation register */ -#define IFATN_FILTERCUTOFF_MASK 0x0000ff00 /* Initial filter cutoff frequency in exponential units */ +SUB_REG(IFATN, FILTERCUTOFF, 0x0000ff00) /* Initial filter cutoff frequency in exponential units */ /* 6 most significant bits are semitones */ /* 2 least significant bits are fractions */ -#define IFATN_FILTERCUTOFF 0x08080019 -#define IFATN_ATTENUATION_MASK 0x000000ff /* Initial attenuation in 0.375dB steps */ -#define IFATN_ATTENUATION 0x08000019 +SUB_REG(IFATN, ATTENUATION, 0x000000ff) /* Initial attenuation in 0.375dB steps */ #define PEFE 0x1a /* Pitch envelope and filter envelope amount register */ -#define PEFE_PITCHAMOUNT_MASK 0x0000ff00 /* Pitch envlope amount */ +SUB_REG(PEFE, PITCHAMOUNT, 0x0000ff00) /* Pitch envlope amount */ /* Signed 2's complement, +/- one octave peak extremes */ -#define PEFE_PITCHAMOUNT 0x0808001a -#define PEFE_FILTERAMOUNT_MASK 0x000000ff /* Filter envlope amount */ +SUB_REG(PEFE, FILTERAMOUNT, 0x000000ff) /* Filter envlope amount */ /* Signed 2's complement, +/- six octaves peak extremes */ -#define PEFE_FILTERAMOUNT 0x0800001a + #define FMMOD 0x1b /* Vibrato/filter modulation from LFO register */ #define FMMOD_MODVIBRATO 0x0000ff00 /* Vibrato LFO modulation depth */ @@ -793,22 +783,19 @@ #define SRCS_SPDIFRATE_96 0x00080000 #define MICIDX 0x63 /* Microphone recording buffer index register */ -#define MICIDX_MASK 0x0000ffff /* 16-bit value */ -#define MICIDX_IDX 0x10000063 +SUB_REG(MICIDX, IDX, 0x0000ffff) #define ADCIDX 0x64 /* ADC recording buffer index register */ -#define ADCIDX_MASK 0x0000ffff /* 16 bit index field */ -#define ADCIDX_IDX 0x10000064 +SUB_REG(ADCIDX, IDX, 0x0000ffff) #define A_ADCIDX 0x63 -#define A_ADCIDX_IDX 0x10000063 +SUB_REG(A_ADCIDX, IDX, 0x0000ffff) #define A_MICIDX 0x64 -#define A_MICIDX_IDX 0x10000064 +SUB_REG(A_MICIDX, IDX, 0x0000ffff) #define FXIDX 0x65 /* FX recording buffer index register */ -#define FXIDX_MASK 0x0000ffff /* 16-bit value */ -#define FXIDX_IDX 0x10000065 +SUB_REG(FXIDX, IDX, 0x0000ffff) /* The 32-bit HLIEx and HLIPx registers all have one bit per channel control/status */ #define HLIEL 0x66 /* Channel half loop interrupt enable low register */ @@ -852,8 +839,8 @@ #define A_SPDIF_44100 0x00000080 #define A_SPDIF_MUTED 0x000000c0 -#define A_I2S_CAPTURE_RATE_MASK 0x00000e00 /* This sets the capture PCM rate, but it is */ -#define A_I2S_CAPTURE_RATE 0x03090076 /* unclear if this sets the ADC rate as well. */ +SUB_REG_NC(A_EHC, A_I2S_CAPTURE_RATE, 0x00000e00) /* This sets the capture PCM rate, but it is */ + /* unclear if this sets the ADC rate as well. */ #define A_I2S_CAPTURE_48000 0x0 #define A_I2S_CAPTURE_192000 0x1 #define A_I2S_CAPTURE_96000 0x2 -- 2.40.0.152.g15d061e6df

2 4

[PATCH v2 0/2] ASoC SSM3515 codec driver
by Martin Povišer 12 May '23

12 May '23

Hi, Analog Devices SSM3515 is a simple speaker amp that Apple is using in their 2021 iMacs, possibly elsewhere. This is a v2 of an ASoC driver for those, to be used with upcoming ASoC support (in upstream kernels anyway) of those platforms. Martin Changes since v1: https://lore.kernel.org/asahi/0fcc4617-e706-8743-e65d-1ff99f38ddac@linaro.o… - drop 'adi,ana-gain' from the binding - move device reset to bus probe - fix schema copyright, other schema tweaks Martin Povišer (2): ASoC: dt-bindings: Add adi,ssm3515 amp schema ASoC: ssm3515: Add new amp driver .../bindings/sound/adi,ssm3515.yaml | 49 ++ MAINTAINERS | 2 + sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ssm3515.c | 448 ++++++++++++++++++ 5 files changed, 507 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/adi,ssm3515.yaml create mode 100644 sound/soc/codecs/ssm3515.c -- 2.38.3

3 4

[PATCH v4] ASoC: dt-bindings: nau8825: Convert to dtschema
by David Lin 12 May '23

12 May '23

Convert the NAU8825 audio CODEC bindings to DT schema. Signed-off-by: David Lin <CTLIN0(a)nuvoton.com> Changes: V3 -> V4: - remove the description from interrupts, clocks and clock-names properites - add sound-dai-cells property and update the dts example V2 -> V3: - refine node name from "nau8825" to "codec" for generic purpose V1 -> V2: - add interrupts properties - add maximum to nuvoton,jack-insert-debounce and nuvoton,jack-eject-debounce properites - add a enum item for nuvoton,short-key-debounce properites - add default value for most properites - add maxItems to clocks properites and mclk entries to clock-names properites - refine wrong indentation from clocks and clock-names - refine dts example for interrupts and clocks - remove headset label from dts example --- .../devicetree/bindings/sound/nau8825.txt | 111 -------- .../bindings/sound/nuvoton,nau8825.yaml | 239 ++++++++++++++++++ 2 files changed, 239 insertions(+), 111 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/nau8825.txt create mode 100644 Documentation/devicetree/bindings/sound/nuvoton,nau8825.yaml diff --git a/Documentation/devicetree/bindings/sound/nau8825.txt b/Documentation/devicetree/bindings/sound/nau8825.txt deleted file mode 100644 index a9c34526f4cb..000000000000 --- a/Documentation/devicetree/bindings/sound/nau8825.txt +++ /dev/null @@ -1,111 +0,0 @@ -Nuvoton NAU8825 audio codec - -This device supports I2C only. - -Required properties: - - compatible : Must be "nuvoton,nau8825" - - - reg : the I2C address of the device. This is either 0x1a (CSB=0) or 0x1b (CSB=1). - -Optional properties: - - nuvoton,jkdet-enable: Enable jack detection via JKDET pin. - - nuvoton,jkdet-pull-enable: Enable JKDET pin pull. If set - pin pull enabled, - otherwise pin in high impedance state. - - nuvoton,jkdet-pull-up: Pull-up JKDET pin. If set then JKDET pin is pull up, otherwise pull down. - - nuvoton,jkdet-polarity: JKDET pin polarity. 0 - active high, 1 - active low. - - - nuvoton,vref-impedance: VREF Impedance selection - 0 - Open - 1 - 25 kOhm - 2 - 125 kOhm - 3 - 2.5 kOhm - - - nuvoton,micbias-voltage: Micbias voltage level. - 0 - VDDA - 1 - VDDA - 2 - VDDA * 1.1 - 3 - VDDA * 1.2 - 4 - VDDA * 1.3 - 5 - VDDA * 1.4 - 6 - VDDA * 1.53 - 7 - VDDA * 1.53 - - - nuvoton,sar-threshold-num: Number of buttons supported - - nuvoton,sar-threshold: Impedance threshold for each button. Array that contains up to 8 buttons configuration. SAR value is calculated as - SAR = 255 * MICBIAS / SAR_VOLTAGE * R / (2000 + R) - where MICBIAS is configured by 'nuvoton,micbias-voltage', SAR_VOLTAGE is configured by 'nuvoton,sar-voltage', R - button impedance. - Refer datasheet section 10.2 for more information about threshold calculation. - - - nuvoton,sar-hysteresis: Button impedance measurement hysteresis. - - - nuvoton,sar-voltage: Reference voltage for button impedance measurement. - 0 - VDDA - 1 - VDDA - 2 - VDDA * 1.1 - 3 - VDDA * 1.2 - 4 - VDDA * 1.3 - 5 - VDDA * 1.4 - 6 - VDDA * 1.53 - 7 - VDDA * 1.53 - - - nuvoton,sar-compare-time: SAR compare time - 0 - 500 ns - 1 - 1 us - 2 - 2 us - 3 - 4 us - - - nuvoton,sar-sampling-time: SAR sampling time - 0 - 2 us - 1 - 4 us - 2 - 8 us - 3 - 16 us - - - nuvoton,short-key-debounce: Button short key press debounce time. - 0 - 30 ms - 1 - 50 ms - 2 - 100 ms - 3 - 30 ms - - - nuvoton,jack-insert-debounce: number from 0 to 7 that sets debounce time to 2^(n+2) ms - - nuvoton,jack-eject-debounce: number from 0 to 7 that sets debounce time to 2^(n+2) ms - - - nuvoton,crosstalk-enable: make crosstalk function enable if set. - - - nuvoton,adcout-drive-strong: make the drive strength of ADCOUT IO PIN strong if set. - Otherwise, the drive keeps normal strength. - - - nuvoton,adc-delay-ms: Delay (in ms) to make input path stable and avoid pop noise. The - default value is 125 and range between 125 to 500 ms. - - - clocks: list of phandle and clock specifier pairs according to common clock bindings for the - clocks described in clock-names - - clock-names: should include "mclk" for the MCLK master clock - -Example: - - headset: nau8825@1a { - compatible = "nuvoton,nau8825"; - reg = <0x1a>; - interrupt-parent = <&gpio>; - interrupts = <TEGRA_GPIO(E, 6) IRQ_TYPE_LEVEL_LOW>; - nuvoton,jkdet-enable; - nuvoton,jkdet-pull-enable; - nuvoton,jkdet-pull-up; - nuvoton,jkdet-polarity = <GPIO_ACTIVE_LOW>; - nuvoton,vref-impedance = <2>; - nuvoton,micbias-voltage = <6>; - // Setup 4 buttons impedance according to Android specification - nuvoton,sar-threshold-num = <4>; - nuvoton,sar-threshold = <0xc 0x1e 0x38 0x60>; - nuvoton,sar-hysteresis = <1>; - nuvoton,sar-voltage = <0>; - nuvoton,sar-compare-time = <0>; - nuvoton,sar-sampling-time = <0>; - nuvoton,short-key-debounce = <2>; - nuvoton,jack-insert-debounce = <7>; - nuvoton,jack-eject-debounce = <7>; - nuvoton,crosstalk-enable; - - clock-names = "mclk"; - clocks = <&tegra_pmc TEGRA_PMC_CLK_OUT_2>; - }; diff --git a/Documentation/devicetree/bindings/sound/nuvoton,nau8825.yaml b/Documentation/devicetree/bindings/sound/nuvoton,nau8825.yaml new file mode 100644 index 000000000000..a54f194a0b49 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nuvoton,nau8825.yaml @@ -0,0 +1,239 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nuvoton,nau8825.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NAU8825 audio CODEC + +maintainers: + - John Hsu <KCHSU0(a)nuvoton.com> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - nuvoton,nau8825 + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + nuvoton,jkdet-enable: + description: + Enable jack detection via JKDET pin. + type: boolean + + nuvoton,jkdet-pull-enable: + description: + Enable JKDET pin pull. + If set - pin pull enabled, otherwise pin in high impedance state. + type: boolean + + nuvoton,jkdet-pull-up: + description: + Pull-up JKDET pin. + If set then JKDET pin is pull up, otherwise pull down. + type: boolean + + nuvoton,jkdet-polarity: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + JKDET pin polarity. + enum: + - 0 # active high + - 1 # active low + default: 1 + + nuvoton,vref-impedance: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + VREF Impedance selection. + enum: + - 0 # Open + - 1 # 25 kOhm + - 2 # 125 kOhm + - 3 # 2.5 kOhm + default: 2 + + nuvoton,micbias-voltage: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + Micbias voltage level. + enum: + - 0 # VDDA + - 1 # VDDA + - 2 # VDDA * 1.1 + - 3 # VDDA * 1.2 + - 4 # VDDA * 1.3 + - 5 # VDDA * 1.4 + - 6 # VDDA * 1.53 + - 7 # VDDA * 1.53 + default: 6 + + nuvoton,sar-threshold-num: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + Number of buttons supported. + minimum: 1 + maximum: 4 + default: 4 + + nuvoton,sar-threshold: + $ref: /schemas/types.yaml#/definitions/uint32-array + description: + Impedance threshold for each button. Array that contains up to 8 buttons + configuration. SAR value is calculated as + SAR = 255 * MICBIAS / SAR_VOLTAGE * R / (2000 + R) where MICBIAS is + configured by 'nuvoton,micbias-voltage', SAR_VOLTAGE is configured by + 'nuvoton,sar-voltage', R - button impedance. + Refer datasheet section 10.2 for more information about threshold + calculation. + minItems: 1 + maxItems: 4 + items: + minimum: 0 + maximum: 255 + + nuvoton,sar-hysteresis: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + Button impedance measurement hysteresis. + default: 0 + + nuvoton,sar-voltage: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + Reference voltage for button impedance measurement. + enum: + - 0 # VDDA + - 1 # VDDA + - 2 # VDDA * 1.1 + - 3 # VDDA * 1.2 + - 4 # VDDA * 1.3 + - 5 # VDDA * 1.4 + - 6 # VDDA * 1.53 + - 7 # VDDA * 1.53 + default: 6 + + nuvoton,sar-compare-time: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + SAR compare time. + enum: + - 0 # 500 ns + - 1 # 1 us + - 2 # 2 us + - 3 # 4 us + default: 1 + + nuvoton,sar-sampling-time: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + SAR sampling time. + enum: + - 0 # 2 us + - 1 # 4 us + - 2 # 8 us + - 3 # 16 us + default: 1 + + nuvoton,short-key-debounce: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + Button short key press debounce time. + enum: + - 0 # 30 ms + - 1 # 50 ms + - 2 # 100 ms + - 3 # 30 ms + default: 3 + + nuvoton,jack-insert-debounce: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + number from 0 to 7 that sets debounce time to 2^(n+2) ms. + maximum: 7 + default: 7 + + nuvoton,jack-eject-debounce: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + number from 0 to 7 that sets debounce time to 2^(n+2) ms + maximum: 7 + default: 0 + + nuvoton,crosstalk-enable: + description: + make crosstalk function enable if set. + type: boolean + + nuvoton,adcout-drive-strong: + description: + make the drive strength of ADCOUT IO PIN strong if set. + Otherwise, the drive keeps normal strength. + type: boolean + + nuvoton,adc-delay-ms: + description: + Delay (in ms) to make input path stable and avoid pop noise. + The default value is 125 and range between 125 to 500 ms. + minimum: 125 + maximum: 500 + default: 125 + + clocks: + maxItems: 1 + + clock-names: + items: + - const: mclk + + '#sound-dai-cells': + const: 0 + +required: + - compatible + - reg + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + #include <dt-bindings/interrupt-controller/irq.h> + i2c { + #address-cells = <1>; + #size-cells = <0>; + codec@1a { + #sound-dai-cells = <0>; + compatible = "nuvoton,nau8825"; + reg = <0x1a>; + interrupt-parent = <&gpio>; + interrupts = <38 IRQ_TYPE_LEVEL_LOW>; + nuvoton,jkdet-enable; + nuvoton,jkdet-pull-enable; + nuvoton,jkdet-pull-up; + nuvoton,jkdet-polarity = <GPIO_ACTIVE_LOW>; + nuvoton,vref-impedance = <2>; + nuvoton,micbias-voltage = <6>; + // Setup 4 buttons impedance according to Android specification + nuvoton,sar-threshold-num = <4>; + nuvoton,sar-threshold = <0xc 0x1e 0x38 0x60>; + nuvoton,sar-hysteresis = <1>; + nuvoton,sar-voltage = <0>; + nuvoton,sar-compare-time = <0>; + nuvoton,sar-sampling-time = <0>; + nuvoton,short-key-debounce = <2>; + nuvoton,jack-insert-debounce = <7>; + nuvoton,jack-eject-debounce = <7>; + nuvoton,crosstalk-enable; + + clock-names = "mclk"; + clocks = <&tegra_pmc 1>; + }; + }; -- 2.25.1

3 2

[PATCH 0/6] Fix soc compress playback deadlock in 5.15
by yixuanjiang 12 May '23

12 May '23

The patch in 5.15 aa9ff6a4955f ("ASoC: soc-compress: Reposition and add pcm_mutex") will cause the deadlock when close compress playback. It needs backport patch b7898396f4 ("ASoC: soc-pcm: Fix and cleanup DPCM locking commit") to prevent the deadlock issue. Rootcause is patch b7898396f4 will remove pcm_mutex in soc_pcm_hw_clean then patch aa9ff6a4955f add pcm_mutex acqurie back in snd_compr_free to fix regression cause from b7898396f4. If only apply aa9ff6a4955f, it will acqurie pcm_mutex twice cause the deadlock. There are total 6 patches on these related backport [ 212.063445] sysrq: Show Blocked State [ 212.064964] Call trace: [ 212.064974] __switch_to+0x244/0x454 [ 212.065075] __schedule+0x5fc/0xbdc [ 212.065092] schedule+0x154/0x258 [ 212.065103] schedule_preempt_disabled+0x30/0x50 [ 212.065115] __mutex_lock+0x36c/0x794 [ 212.065135] __mutex_lock_slowpath+0x1c/0x2c [ 212.065145] soc_pcm_hw_clean+0x6c/0x364 [ 212.065177] dpcm_be_dai_hw_free+0x20c/0x304 [ 212.065188] soc_compr_free_fe+0x130/0x448 [ 212.065230] snd_compr_free+0xac/0x148 [ 212.065316] __fput+0x104/0x438 [ 212.065347] ____fput+0x18/0x28 [ 212.065409] task_work_run+0x144/0x1f8 [ 212.065474] do_notify_resume+0x1d24/0x27b0 [ 212.065518] el0_svc+0x68/0xc4 [ 212.065601] el0t_64_sync_handler+0x8c/0xfc [ 212.065610] el0t_64_sync+0x1b0/0x1b4 *** BLURB HERE *** Pierre-Louis Bossart (4): ASoC: soc-pcm: use GFP_ATOMIC for dpcm structure ASoC: soc-pcm: align BE 'atomicity' with that of the FE ASoC: soc-pcm: test refcount before triggering ASoC: soc-pcm: fix BE handling of PAUSE_RELEASE Takashi Iwai (2): ASoC: soc-pcm: Fix and cleanup DPCM locking ASoC: soc-pcm: serialize BE triggers include/sound/soc-dpcm.h | 2 + include/sound/soc.h | 2 - sound/soc/soc-core.c | 1 - sound/soc/soc-pcm.c | 351 +++++++++++++++++++++++++++------------ 4 files changed, 246 insertions(+), 110 deletions(-) -- 2.40.1.521.gf1e218fcd8-goog

2 7

Looking for alsaunmute script
by GitHub issues - opened 12 May '23

12 May '23

alsa-project/alsa-utils issue #211 was opened from DenverEllis: I have a friend that is having trouble with their legion 7 laptop. Perusing a very long kernel thread I saw mention of an alsaunmute script and I am having a hard time tracking it down. I believ perex authored it and I saw it mentioned that it was a part of alsa-utils, but alas I have had no luck in finding this script. Issue URL : https://github.com/alsa-project/alsa-utils/issues/211 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH AUTOSEL 5.10 2/3] ALSA: hda: Glenfly: add HD Audio PCI IDs and HDMI Codec Vendor IDs.
by Sasha Levin 11 May '23

11 May '23

From: jasontao <jasontao(a)glenfly.com> [ Upstream commit c51e431052e2eacfb23fbf6b39bc6c8770d9827a ] Add a set of HD Audio PCI IDS, and the HDMI codec vendor IDs for Glenfly Gpus. - In default_bdl_pos_adj, set bdl to 128 as Glenfly Gpus have hardware limitation, need to increase hdac interrupt interval. - In azx_first_init, enable polling mode for Glenfly Gpu. When the codec complete the command, it sends interrupt and writes response entries to memory, howerver, the write requests sometimes are not actually synchronized to memory when driver handle hdac interrupt on Glenfly Gpus. If the RIRB status is not updated in the interrupt handler, azx_rirb_get_response keeps trying to recevie a response from rirb until 1s timeout. Enabling polling mode for Glenfly Gpu can fix the issue. - In patch_gf_hdmi, set Glenlfy Gpu Codec's no_sticky_stream as it need driver to do actual clean-ups for the linked codec when switch from one codec to another. Signed-off-by: jasontao <jasontao(a)glenfly.com> Signed-off-by: Reaper Li <reaperlioc(a)glenfly.com> Link: https://lore.kernel.org/r/20230426013059.4329-1-reaperlioc@glenfly.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/hda_intel.c | 21 +++++++++++++++++++++ sound/pci/hda/patch_hdmi.c | 22 ++++++++++++++++++++++ 2 files changed, 43 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index de1fe604905f3..1f641712233ef 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -264,6 +264,7 @@ enum { AZX_DRIVER_ATI, AZX_DRIVER_ATIHDMI, AZX_DRIVER_ATIHDMI_NS, + AZX_DRIVER_GFHDMI, AZX_DRIVER_VIA, AZX_DRIVER_SIS, AZX_DRIVER_ULI, @@ -386,6 +387,7 @@ static const char * const driver_short_names[] = { [AZX_DRIVER_ATI] = "HDA ATI SB", [AZX_DRIVER_ATIHDMI] = "HDA ATI HDMI", [AZX_DRIVER_ATIHDMI_NS] = "HDA ATI HDMI", + [AZX_DRIVER_GFHDMI] = "HDA GF HDMI", [AZX_DRIVER_VIA] = "HDA VIA VT82xx", [AZX_DRIVER_SIS] = "HDA SIS966", [AZX_DRIVER_ULI] = "HDA ULI M5461", @@ -1783,6 +1785,12 @@ static int default_bdl_pos_adj(struct azx *chip) } switch (chip->driver_type) { + /* + * increase the bdl size for Glenfly Gpus for hardware + * limitation on hdac interrupt interval + */ + case AZX_DRIVER_GFHDMI: + return 128; case AZX_DRIVER_ICH: case AZX_DRIVER_PCH: return 1; @@ -1902,6 +1910,12 @@ static int azx_first_init(struct azx *chip) pci_write_config_dword(pci, PCI_BASE_ADDRESS_1, 0); } #endif + /* + * Fix response write request not synced to memory when handle + * hdac interrupt on Glenfly Gpus + */ + if (chip->driver_type == AZX_DRIVER_GFHDMI) + bus->polling_mode = 1; err = pci_request_regions(pci, "ICH HD audio"); if (err < 0) @@ -2011,6 +2025,7 @@ static int azx_first_init(struct azx *chip) chip->playback_streams = ATIHDMI_NUM_PLAYBACK; chip->capture_streams = ATIHDMI_NUM_CAPTURE; break; + case AZX_DRIVER_GFHDMI: case AZX_DRIVER_GENERIC: default: chip->playback_streams = ICH6_NUM_PLAYBACK; @@ -2756,6 +2771,12 @@ static const struct pci_device_id azx_ids[] = { { PCI_DEVICE(0x1002, 0xab38), .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS | AZX_DCAPS_PM_RUNTIME }, + /* GLENFLY */ + { PCI_DEVICE(0x6766, PCI_ANY_ID), + .class = PCI_CLASS_MULTIMEDIA_HD_AUDIO << 8, + .class_mask = 0xffffff, + .driver_data = AZX_DRIVER_GFHDMI | AZX_DCAPS_POSFIX_LPIB | + AZX_DCAPS_NO_MSI | AZX_DCAPS_NO_64BIT }, /* VIA VT8251/VT8237A */ { PCI_DEVICE(0x1106, 0x3288), .driver_data = AZX_DRIVER_VIA }, /* VIA GFX VT7122/VX900 */ diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 1afe9cddb69eb..bd151465f2089 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -4287,6 +4287,22 @@ static int patch_via_hdmi(struct hda_codec *codec) return patch_simple_hdmi(codec, VIAHDMI_CVT_NID, VIAHDMI_PIN_NID); } +static int patch_gf_hdmi(struct hda_codec *codec) +{ + int err; + + err = patch_generic_hdmi(codec); + if (err) + return err; + + /* + * Glenfly GPUs have two codecs, stream switches from one codec to + * another, need to do actual clean-ups in codec_cleanup_stream + */ + codec->no_sticky_stream = 1; + return 0; +} + /* * patch entries */ @@ -4376,6 +4392,12 @@ HDA_CODEC_ENTRY(0x10de009f, "GPU 9f HDMI/DP", patch_nvhdmi), HDA_CODEC_ENTRY(0x10de00a0, "GPU a0 HDMI/DP", patch_nvhdmi), HDA_CODEC_ENTRY(0x10de8001, "MCP73 HDMI", patch_nvhdmi_2ch), HDA_CODEC_ENTRY(0x10de8067, "MCP67/68 HDMI", patch_nvhdmi_2ch), +HDA_CODEC_ENTRY(0x67663d82, "Arise 82 HDMI/DP", patch_gf_hdmi), +HDA_CODEC_ENTRY(0x67663d83, "Arise 83 HDMI/DP", patch_gf_hdmi), +HDA_CODEC_ENTRY(0x67663d84, "Arise 84 HDMI/DP", patch_gf_hdmi), +HDA_CODEC_ENTRY(0x67663d85, "Arise 85 HDMI/DP", patch_gf_hdmi), +HDA_CODEC_ENTRY(0x67663d86, "Arise 86 HDMI/DP", patch_gf_hdmi), +HDA_CODEC_ENTRY(0x67663d87, "Arise 87 HDMI/DP", patch_gf_hdmi), HDA_CODEC_ENTRY(0x11069f80, "VX900 HDMI/DP", patch_via_hdmi), HDA_CODEC_ENTRY(0x11069f81, "VX900 HDMI/DP", patch_via_hdmi), HDA_CODEC_ENTRY(0x11069f84, "VX11 HDMI/DP", patch_generic_hdmi), -- 2.39.2

1 0

[PATCH AUTOSEL 5.15 3/5] ALSA: hda: Glenfly: add HD Audio PCI IDs and HDMI Codec Vendor IDs.
by Sasha Levin 11 May '23

11 May '23

From: jasontao <jasontao(a)glenfly.com> [ Upstream commit c51e431052e2eacfb23fbf6b39bc6c8770d9827a ] Add a set of HD Audio PCI IDS, and the HDMI codec vendor IDs for Glenfly Gpus. - In default_bdl_pos_adj, set bdl to 128 as Glenfly Gpus have hardware limitation, need to increase hdac interrupt interval. - In azx_first_init, enable polling mode for Glenfly Gpu. When the codec complete the command, it sends interrupt and writes response entries to memory, howerver, the write requests sometimes are not actually synchronized to memory when driver handle hdac interrupt on Glenfly Gpus. If the RIRB status is not updated in the interrupt handler, azx_rirb_get_response keeps trying to recevie a response from rirb until 1s timeout. Enabling polling mode for Glenfly Gpu can fix the issue. - In patch_gf_hdmi, set Glenlfy Gpu Codec's no_sticky_stream as it need driver to do actual clean-ups for the linked codec when switch from one codec to another. Signed-off-by: jasontao <jasontao(a)glenfly.com> Signed-off-by: Reaper Li <reaperlioc(a)glenfly.com> Link: https://lore.kernel.org/r/20230426013059.4329-1-reaperlioc@glenfly.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/hda_intel.c | 21 +++++++++++++++++++++ sound/pci/hda/patch_hdmi.c | 22 ++++++++++++++++++++++ 2 files changed, 43 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 5fce1ca8a393a..1379ac07df350 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -230,6 +230,7 @@ enum { AZX_DRIVER_ATI, AZX_DRIVER_ATIHDMI, AZX_DRIVER_ATIHDMI_NS, + AZX_DRIVER_GFHDMI, AZX_DRIVER_VIA, AZX_DRIVER_SIS, AZX_DRIVER_ULI, @@ -352,6 +353,7 @@ static const char * const driver_short_names[] = { [AZX_DRIVER_ATI] = "HDA ATI SB", [AZX_DRIVER_ATIHDMI] = "HDA ATI HDMI", [AZX_DRIVER_ATIHDMI_NS] = "HDA ATI HDMI", + [AZX_DRIVER_GFHDMI] = "HDA GF HDMI", [AZX_DRIVER_VIA] = "HDA VIA VT82xx", [AZX_DRIVER_SIS] = "HDA SIS966", [AZX_DRIVER_ULI] = "HDA ULI M5461", @@ -1742,6 +1744,12 @@ static int default_bdl_pos_adj(struct azx *chip) } switch (chip->driver_type) { + /* + * increase the bdl size for Glenfly Gpus for hardware + * limitation on hdac interrupt interval + */ + case AZX_DRIVER_GFHDMI: + return 128; case AZX_DRIVER_ICH: case AZX_DRIVER_PCH: return 1; @@ -1857,6 +1865,12 @@ static int azx_first_init(struct azx *chip) pci_write_config_dword(pci, PCI_BASE_ADDRESS_1, 0); } #endif + /* + * Fix response write request not synced to memory when handle + * hdac interrupt on Glenfly Gpus + */ + if (chip->driver_type == AZX_DRIVER_GFHDMI) + bus->polling_mode = 1; err = pcim_iomap_regions(pci, 1 << 0, "ICH HD audio"); if (err < 0) @@ -1957,6 +1971,7 @@ static int azx_first_init(struct azx *chip) chip->playback_streams = ATIHDMI_NUM_PLAYBACK; chip->capture_streams = ATIHDMI_NUM_CAPTURE; break; + case AZX_DRIVER_GFHDMI: case AZX_DRIVER_GENERIC: default: chip->playback_streams = ICH6_NUM_PLAYBACK; @@ -2694,6 +2709,12 @@ static const struct pci_device_id azx_ids[] = { { PCI_DEVICE(0x1002, 0xab38), .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS | AZX_DCAPS_PM_RUNTIME }, + /* GLENFLY */ + { PCI_DEVICE(0x6766, PCI_ANY_ID), + .class = PCI_CLASS_MULTIMEDIA_HD_AUDIO << 8, + .class_mask = 0xffffff, + .driver_data = AZX_DRIVER_GFHDMI | AZX_DCAPS_POSFIX_LPIB | + AZX_DCAPS_NO_MSI | AZX_DCAPS_NO_64BIT }, /* VIA VT8251/VT8237A */ { PCI_DEVICE(0x1106, 0x3288), .driver_data = AZX_DRIVER_VIA }, /* VIA GFX VT7122/VX900 */ diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 8ed66c416c0e7..c8982c788835a 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -4298,6 +4298,22 @@ static int patch_via_hdmi(struct hda_codec *codec) return patch_simple_hdmi(codec, VIAHDMI_CVT_NID, VIAHDMI_PIN_NID); } +static int patch_gf_hdmi(struct hda_codec *codec) +{ + int err; + + err = patch_generic_hdmi(codec); + if (err) + return err; + + /* + * Glenfly GPUs have two codecs, stream switches from one codec to + * another, need to do actual clean-ups in codec_cleanup_stream + */ + codec->no_sticky_stream = 1; + return 0; +} + /* * patch entries */ @@ -4387,6 +4403,12 @@ HDA_CODEC_ENTRY(0x10de009f, "GPU 9f HDMI/DP", patch_nvhdmi), HDA_CODEC_ENTRY(0x10de00a0, "GPU a0 HDMI/DP", patch_nvhdmi), HDA_CODEC_ENTRY(0x10de8001, "MCP73 HDMI", patch_nvhdmi_2ch), HDA_CODEC_ENTRY(0x10de8067, "MCP67/68 HDMI", patch_nvhdmi_2ch), +HDA_CODEC_ENTRY(0x67663d82, "Arise 82 HDMI/DP", patch_gf_hdmi), +HDA_CODEC_ENTRY(0x67663d83, "Arise 83 HDMI/DP", patch_gf_hdmi), +HDA_CODEC_ENTRY(0x67663d84, "Arise 84 HDMI/DP", patch_gf_hdmi), +HDA_CODEC_ENTRY(0x67663d85, "Arise 85 HDMI/DP", patch_gf_hdmi), +HDA_CODEC_ENTRY(0x67663d86, "Arise 86 HDMI/DP", patch_gf_hdmi), +HDA_CODEC_ENTRY(0x67663d87, "Arise 87 HDMI/DP", patch_gf_hdmi), HDA_CODEC_ENTRY(0x11069f80, "VX900 HDMI/DP", patch_via_hdmi), HDA_CODEC_ENTRY(0x11069f81, "VX900 HDMI/DP", patch_via_hdmi), HDA_CODEC_ENTRY(0x11069f84, "VX11 HDMI/DP", patch_generic_hdmi), -- 2.39.2

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[PATCH AUTOSEL 6.1 8/9] ASoC: Intel: soc-acpi-cht: Add quirk for Nextbook Ares 8A tablet
by Sasha Levin 11 May '23

11 May '23

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit ec6f82b4c63cc68f8dc03316e725106d242706be ] The Nextbook Ares 8A tablet which has Android as factory OS, has a buggy DSDT with both ESSX8316 and 10EC5651 ACPI devices. This tablet actually uses an rt5651 codec, but the matching code ends up picking the ESSX8316 device, add a quirk to ignote the ESSX8316 device on this tablet. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Message-Id: <20230429104721.7176-1-hdegoede(a)redhat.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- .../intel/common/soc-acpi-intel-cht-match.c | 26 +++++++++++++++++++ 1 file changed, 26 insertions(+) diff --git a/sound/soc/intel/common/soc-acpi-intel-cht-match.c b/sound/soc/intel/common/soc-acpi-intel-cht-match.c index 6beb00858c33f..cdcbf04b8832f 100644 --- a/sound/soc/intel/common/soc-acpi-intel-cht-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-cht-match.c @@ -50,6 +50,31 @@ static struct snd_soc_acpi_mach *cht_quirk(void *arg) return mach; } +/* + * Some tablets with Android factory OS have buggy DSDTs with an ESSX8316 device + * in the ACPI tables. While they are not using an ESS8316 codec. These DSDTs + * also have an ACPI device for the correct codec, ignore the ESSX8316. + */ +static const struct dmi_system_id cht_ess8316_not_present_table[] = { + { + /* Nextbook Ares 8A */ + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Insyde"), + DMI_MATCH(DMI_PRODUCT_NAME, "CherryTrail"), + DMI_MATCH(DMI_BIOS_VERSION, "M882"), + }, + }, + { } +}; + +static struct snd_soc_acpi_mach *cht_ess8316_quirk(void *arg) +{ + if (dmi_check_system(cht_ess8316_not_present_table)) + return NULL; + + return arg; +} + static const struct snd_soc_acpi_codecs rt5640_comp_ids = { .num_codecs = 2, .codecs = { "10EC5640", "10EC3276" }, @@ -113,6 +138,7 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cherrytrail_machines[] = { .drv_name = "bytcht_es8316", .fw_filename = "intel/fw_sst_22a8.bin", .board = "bytcht_es8316", + .machine_quirk = cht_ess8316_quirk, .sof_tplg_filename = "sof-cht-es8316.tplg", }, /* some CHT-T platforms rely on RT5640, use Baytrail machine driver */ -- 2.39.2

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[PATCH AUTOSEL 6.1 3/9] ALSA: hda: Glenfly: add HD Audio PCI IDs and HDMI Codec Vendor IDs.
by Sasha Levin 11 May '23

11 May '23

From: jasontao <jasontao(a)glenfly.com> [ Upstream commit c51e431052e2eacfb23fbf6b39bc6c8770d9827a ] Add a set of HD Audio PCI IDS, and the HDMI codec vendor IDs for Glenfly Gpus. - In default_bdl_pos_adj, set bdl to 128 as Glenfly Gpus have hardware limitation, need to increase hdac interrupt interval. - In azx_first_init, enable polling mode for Glenfly Gpu. When the codec complete the command, it sends interrupt and writes response entries to memory, howerver, the write requests sometimes are not actually synchronized to memory when driver handle hdac interrupt on Glenfly Gpus. If the RIRB status is not updated in the interrupt handler, azx_rirb_get_response keeps trying to recevie a response from rirb until 1s timeout. Enabling polling mode for Glenfly Gpu can fix the issue. - In patch_gf_hdmi, set Glenlfy Gpu Codec's no_sticky_stream as it need driver to do actual clean-ups for the linked codec when switch from one codec to another. Signed-off-by: jasontao <jasontao(a)glenfly.com> Signed-off-by: Reaper Li <reaperlioc(a)glenfly.com> Link: https://lore.kernel.org/r/20230426013059.4329-1-reaperlioc@glenfly.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/hda_intel.c | 21 +++++++++++++++++++++ sound/pci/hda/patch_hdmi.c | 22 ++++++++++++++++++++++ 2 files changed, 43 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 77a592f219472..31e16feaa2b4f 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -227,6 +227,7 @@ enum { AZX_DRIVER_ATI, AZX_DRIVER_ATIHDMI, AZX_DRIVER_ATIHDMI_NS, + AZX_DRIVER_GFHDMI, AZX_DRIVER_VIA, AZX_DRIVER_SIS, AZX_DRIVER_ULI, @@ -349,6 +350,7 @@ static const char * const driver_short_names[] = { [AZX_DRIVER_ATI] = "HDA ATI SB", [AZX_DRIVER_ATIHDMI] = "HDA ATI HDMI", [AZX_DRIVER_ATIHDMI_NS] = "HDA ATI HDMI", + [AZX_DRIVER_GFHDMI] = "HDA GF HDMI", [AZX_DRIVER_VIA] = "HDA VIA VT82xx", [AZX_DRIVER_SIS] = "HDA SIS966", [AZX_DRIVER_ULI] = "HDA ULI M5461", @@ -1743,6 +1745,12 @@ static int default_bdl_pos_adj(struct azx *chip) } switch (chip->driver_type) { + /* + * increase the bdl size for Glenfly Gpus for hardware + * limitation on hdac interrupt interval + */ + case AZX_DRIVER_GFHDMI: + return 128; case AZX_DRIVER_ICH: case AZX_DRIVER_PCH: return 1; @@ -1858,6 +1866,12 @@ static int azx_first_init(struct azx *chip) pci_write_config_dword(pci, PCI_BASE_ADDRESS_1, 0); } #endif + /* + * Fix response write request not synced to memory when handle + * hdac interrupt on Glenfly Gpus + */ + if (chip->driver_type == AZX_DRIVER_GFHDMI) + bus->polling_mode = 1; err = pcim_iomap_regions(pci, 1 << 0, "ICH HD audio"); if (err < 0) @@ -1959,6 +1973,7 @@ static int azx_first_init(struct azx *chip) chip->playback_streams = ATIHDMI_NUM_PLAYBACK; chip->capture_streams = ATIHDMI_NUM_CAPTURE; break; + case AZX_DRIVER_GFHDMI: case AZX_DRIVER_GENERIC: default: chip->playback_streams = ICH6_NUM_PLAYBACK; @@ -2724,6 +2739,12 @@ static const struct pci_device_id azx_ids[] = { { PCI_DEVICE(0x1002, 0xab38), .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS | AZX_DCAPS_PM_RUNTIME }, + /* GLENFLY */ + { PCI_DEVICE(0x6766, PCI_ANY_ID), + .class = PCI_CLASS_MULTIMEDIA_HD_AUDIO << 8, + .class_mask = 0xffffff, + .driver_data = AZX_DRIVER_GFHDMI | AZX_DCAPS_POSFIX_LPIB | + AZX_DCAPS_NO_MSI | AZX_DCAPS_NO_64BIT }, /* VIA VT8251/VT8237A */ { PCI_DEVICE(0x1106, 0x3288), .driver_data = AZX_DRIVER_VIA }, /* VIA GFX VT7122/VX900 */ diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 5c6980394dcec..b784ef95390db 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -4489,6 +4489,22 @@ static int patch_via_hdmi(struct hda_codec *codec) return patch_simple_hdmi(codec, VIAHDMI_CVT_NID, VIAHDMI_PIN_NID); } +static int patch_gf_hdmi(struct hda_codec *codec) +{ + int err; + + err = patch_generic_hdmi(codec); + if (err) + return err; + + /* + * Glenfly GPUs have two codecs, stream switches from one codec to + * another, need to do actual clean-ups in codec_cleanup_stream + */ + codec->no_sticky_stream = 1; + return 0; +} + /* * patch entries */ @@ -4579,6 +4595,12 @@ HDA_CODEC_ENTRY(0x10de009f, "GPU 9f HDMI/DP", patch_nvhdmi), HDA_CODEC_ENTRY(0x10de00a0, "GPU a0 HDMI/DP", patch_nvhdmi), HDA_CODEC_ENTRY(0x10de8001, "MCP73 HDMI", patch_nvhdmi_2ch), HDA_CODEC_ENTRY(0x10de8067, "MCP67/68 HDMI", patch_nvhdmi_2ch), +HDA_CODEC_ENTRY(0x67663d82, "Arise 82 HDMI/DP", patch_gf_hdmi), +HDA_CODEC_ENTRY(0x67663d83, "Arise 83 HDMI/DP", patch_gf_hdmi), +HDA_CODEC_ENTRY(0x67663d84, "Arise 84 HDMI/DP", patch_gf_hdmi), +HDA_CODEC_ENTRY(0x67663d85, "Arise 85 HDMI/DP", patch_gf_hdmi), +HDA_CODEC_ENTRY(0x67663d86, "Arise 86 HDMI/DP", patch_gf_hdmi), +HDA_CODEC_ENTRY(0x67663d87, "Arise 87 HDMI/DP", patch_gf_hdmi), HDA_CODEC_ENTRY(0x11069f80, "VX900 HDMI/DP", patch_via_hdmi), HDA_CODEC_ENTRY(0x11069f81, "VX900 HDMI/DP", patch_via_hdmi), HDA_CODEC_ENTRY(0x11069f84, "VX11 HDMI/DP", patch_generic_hdmi), -- 2.39.2

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[PATCH AUTOSEL 6.2 09/10] ASoC: Intel: soc-acpi-cht: Add quirk for Nextbook Ares 8A tablet
by Sasha Levin 11 May '23

11 May '23

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit ec6f82b4c63cc68f8dc03316e725106d242706be ] The Nextbook Ares 8A tablet which has Android as factory OS, has a buggy DSDT with both ESSX8316 and 10EC5651 ACPI devices. This tablet actually uses an rt5651 codec, but the matching code ends up picking the ESSX8316 device, add a quirk to ignote the ESSX8316 device on this tablet. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Message-Id: <20230429104721.7176-1-hdegoede(a)redhat.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- .../intel/common/soc-acpi-intel-cht-match.c | 26 +++++++++++++++++++ 1 file changed, 26 insertions(+) diff --git a/sound/soc/intel/common/soc-acpi-intel-cht-match.c b/sound/soc/intel/common/soc-acpi-intel-cht-match.c index 6beb00858c33f..cdcbf04b8832f 100644 --- a/sound/soc/intel/common/soc-acpi-intel-cht-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-cht-match.c @@ -50,6 +50,31 @@ static struct snd_soc_acpi_mach *cht_quirk(void *arg) return mach; } +/* + * Some tablets with Android factory OS have buggy DSDTs with an ESSX8316 device + * in the ACPI tables. While they are not using an ESS8316 codec. These DSDTs + * also have an ACPI device for the correct codec, ignore the ESSX8316. + */ +static const struct dmi_system_id cht_ess8316_not_present_table[] = { + { + /* Nextbook Ares 8A */ + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Insyde"), + DMI_MATCH(DMI_PRODUCT_NAME, "CherryTrail"), + DMI_MATCH(DMI_BIOS_VERSION, "M882"), + }, + }, + { } +}; + +static struct snd_soc_acpi_mach *cht_ess8316_quirk(void *arg) +{ + if (dmi_check_system(cht_ess8316_not_present_table)) + return NULL; + + return arg; +} + static const struct snd_soc_acpi_codecs rt5640_comp_ids = { .num_codecs = 2, .codecs = { "10EC5640", "10EC3276" }, @@ -113,6 +138,7 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cherrytrail_machines[] = { .drv_name = "bytcht_es8316", .fw_filename = "intel/fw_sst_22a8.bin", .board = "bytcht_es8316", + .machine_quirk = cht_ess8316_quirk, .sof_tplg_filename = "sof-cht-es8316.tplg", }, /* some CHT-T platforms rely on RT5640, use Baytrail machine driver */ -- 2.39.2

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[PATCH AUTOSEL 6.2 03/10] ALSA: hda: Glenfly: add HD Audio PCI IDs and HDMI Codec Vendor IDs.
by Sasha Levin 11 May '23

11 May '23

From: jasontao <jasontao(a)glenfly.com> [ Upstream commit c51e431052e2eacfb23fbf6b39bc6c8770d9827a ] Add a set of HD Audio PCI IDS, and the HDMI codec vendor IDs for Glenfly Gpus. - In default_bdl_pos_adj, set bdl to 128 as Glenfly Gpus have hardware limitation, need to increase hdac interrupt interval. - In azx_first_init, enable polling mode for Glenfly Gpu. When the codec complete the command, it sends interrupt and writes response entries to memory, howerver, the write requests sometimes are not actually synchronized to memory when driver handle hdac interrupt on Glenfly Gpus. If the RIRB status is not updated in the interrupt handler, azx_rirb_get_response keeps trying to recevie a response from rirb until 1s timeout. Enabling polling mode for Glenfly Gpu can fix the issue. - In patch_gf_hdmi, set Glenlfy Gpu Codec's no_sticky_stream as it need driver to do actual clean-ups for the linked codec when switch from one codec to another. Signed-off-by: jasontao <jasontao(a)glenfly.com> Signed-off-by: Reaper Li <reaperlioc(a)glenfly.com> Link: https://lore.kernel.org/r/20230426013059.4329-1-reaperlioc@glenfly.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/hda_intel.c | 21 +++++++++++++++++++++ sound/pci/hda/patch_hdmi.c | 22 ++++++++++++++++++++++ 2 files changed, 43 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 77a592f219472..31e16feaa2b4f 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -227,6 +227,7 @@ enum { AZX_DRIVER_ATI, AZX_DRIVER_ATIHDMI, AZX_DRIVER_ATIHDMI_NS, + AZX_DRIVER_GFHDMI, AZX_DRIVER_VIA, AZX_DRIVER_SIS, AZX_DRIVER_ULI, @@ -349,6 +350,7 @@ static const char * const driver_short_names[] = { [AZX_DRIVER_ATI] = "HDA ATI SB", [AZX_DRIVER_ATIHDMI] = "HDA ATI HDMI", [AZX_DRIVER_ATIHDMI_NS] = "HDA ATI HDMI", + [AZX_DRIVER_GFHDMI] = "HDA GF HDMI", [AZX_DRIVER_VIA] = "HDA VIA VT82xx", [AZX_DRIVER_SIS] = "HDA SIS966", [AZX_DRIVER_ULI] = "HDA ULI M5461", @@ -1743,6 +1745,12 @@ static int default_bdl_pos_adj(struct azx *chip) } switch (chip->driver_type) { + /* + * increase the bdl size for Glenfly Gpus for hardware + * limitation on hdac interrupt interval + */ + case AZX_DRIVER_GFHDMI: + return 128; case AZX_DRIVER_ICH: case AZX_DRIVER_PCH: return 1; @@ -1858,6 +1866,12 @@ static int azx_first_init(struct azx *chip) pci_write_config_dword(pci, PCI_BASE_ADDRESS_1, 0); } #endif + /* + * Fix response write request not synced to memory when handle + * hdac interrupt on Glenfly Gpus + */ + if (chip->driver_type == AZX_DRIVER_GFHDMI) + bus->polling_mode = 1; err = pcim_iomap_regions(pci, 1 << 0, "ICH HD audio"); if (err < 0) @@ -1959,6 +1973,7 @@ static int azx_first_init(struct azx *chip) chip->playback_streams = ATIHDMI_NUM_PLAYBACK; chip->capture_streams = ATIHDMI_NUM_CAPTURE; break; + case AZX_DRIVER_GFHDMI: case AZX_DRIVER_GENERIC: default: chip->playback_streams = ICH6_NUM_PLAYBACK; @@ -2724,6 +2739,12 @@ static const struct pci_device_id azx_ids[] = { { PCI_DEVICE(0x1002, 0xab38), .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS | AZX_DCAPS_PM_RUNTIME }, + /* GLENFLY */ + { PCI_DEVICE(0x6766, PCI_ANY_ID), + .class = PCI_CLASS_MULTIMEDIA_HD_AUDIO << 8, + .class_mask = 0xffffff, + .driver_data = AZX_DRIVER_GFHDMI | AZX_DCAPS_POSFIX_LPIB | + AZX_DCAPS_NO_MSI | AZX_DCAPS_NO_64BIT }, /* VIA VT8251/VT8237A */ { PCI_DEVICE(0x1106, 0x3288), .driver_data = AZX_DRIVER_VIA }, /* VIA GFX VT7122/VX900 */ diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 5c6980394dcec..b784ef95390db 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -4489,6 +4489,22 @@ static int patch_via_hdmi(struct hda_codec *codec) return patch_simple_hdmi(codec, VIAHDMI_CVT_NID, VIAHDMI_PIN_NID); } +static int patch_gf_hdmi(struct hda_codec *codec) +{ + int err; + + err = patch_generic_hdmi(codec); + if (err) + return err; + + /* + * Glenfly GPUs have two codecs, stream switches from one codec to + * another, need to do actual clean-ups in codec_cleanup_stream + */ + codec->no_sticky_stream = 1; + return 0; +} + /* * patch entries */ @@ -4579,6 +4595,12 @@ HDA_CODEC_ENTRY(0x10de009f, "GPU 9f HDMI/DP", patch_nvhdmi), HDA_CODEC_ENTRY(0x10de00a0, "GPU a0 HDMI/DP", patch_nvhdmi), HDA_CODEC_ENTRY(0x10de8001, "MCP73 HDMI", patch_nvhdmi_2ch), HDA_CODEC_ENTRY(0x10de8067, "MCP67/68 HDMI", patch_nvhdmi_2ch), +HDA_CODEC_ENTRY(0x67663d82, "Arise 82 HDMI/DP", patch_gf_hdmi), +HDA_CODEC_ENTRY(0x67663d83, "Arise 83 HDMI/DP", patch_gf_hdmi), +HDA_CODEC_ENTRY(0x67663d84, "Arise 84 HDMI/DP", patch_gf_hdmi), +HDA_CODEC_ENTRY(0x67663d85, "Arise 85 HDMI/DP", patch_gf_hdmi), +HDA_CODEC_ENTRY(0x67663d86, "Arise 86 HDMI/DP", patch_gf_hdmi), +HDA_CODEC_ENTRY(0x67663d87, "Arise 87 HDMI/DP", patch_gf_hdmi), HDA_CODEC_ENTRY(0x11069f80, "VX900 HDMI/DP", patch_via_hdmi), HDA_CODEC_ENTRY(0x11069f81, "VX900 HDMI/DP", patch_via_hdmi), HDA_CODEC_ENTRY(0x11069f84, "VX11 HDMI/DP", patch_generic_hdmi), -- 2.39.2

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[PATCH AUTOSEL 6.3 10/11] ASoC: Intel: soc-acpi-cht: Add quirk for Nextbook Ares 8A tablet
by Sasha Levin 11 May '23

11 May '23

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit ec6f82b4c63cc68f8dc03316e725106d242706be ] The Nextbook Ares 8A tablet which has Android as factory OS, has a buggy DSDT with both ESSX8316 and 10EC5651 ACPI devices. This tablet actually uses an rt5651 codec, but the matching code ends up picking the ESSX8316 device, add a quirk to ignote the ESSX8316 device on this tablet. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Message-Id: <20230429104721.7176-1-hdegoede(a)redhat.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- .../intel/common/soc-acpi-intel-cht-match.c | 26 +++++++++++++++++++ 1 file changed, 26 insertions(+) diff --git a/sound/soc/intel/common/soc-acpi-intel-cht-match.c b/sound/soc/intel/common/soc-acpi-intel-cht-match.c index 6beb00858c33f..cdcbf04b8832f 100644 --- a/sound/soc/intel/common/soc-acpi-intel-cht-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-cht-match.c @@ -50,6 +50,31 @@ static struct snd_soc_acpi_mach *cht_quirk(void *arg) return mach; } +/* + * Some tablets with Android factory OS have buggy DSDTs with an ESSX8316 device + * in the ACPI tables. While they are not using an ESS8316 codec. These DSDTs + * also have an ACPI device for the correct codec, ignore the ESSX8316. + */ +static const struct dmi_system_id cht_ess8316_not_present_table[] = { + { + /* Nextbook Ares 8A */ + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Insyde"), + DMI_MATCH(DMI_PRODUCT_NAME, "CherryTrail"), + DMI_MATCH(DMI_BIOS_VERSION, "M882"), + }, + }, + { } +}; + +static struct snd_soc_acpi_mach *cht_ess8316_quirk(void *arg) +{ + if (dmi_check_system(cht_ess8316_not_present_table)) + return NULL; + + return arg; +} + static const struct snd_soc_acpi_codecs rt5640_comp_ids = { .num_codecs = 2, .codecs = { "10EC5640", "10EC3276" }, @@ -113,6 +138,7 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cherrytrail_machines[] = { .drv_name = "bytcht_es8316", .fw_filename = "intel/fw_sst_22a8.bin", .board = "bytcht_es8316", + .machine_quirk = cht_ess8316_quirk, .sof_tplg_filename = "sof-cht-es8316.tplg", }, /* some CHT-T platforms rely on RT5640, use Baytrail machine driver */ -- 2.39.2

1 0

[PATCH AUTOSEL 6.3 03/11] ALSA: hda: Glenfly: add HD Audio PCI IDs and HDMI Codec Vendor IDs.
by Sasha Levin 11 May '23

11 May '23

From: jasontao <jasontao(a)glenfly.com> [ Upstream commit c51e431052e2eacfb23fbf6b39bc6c8770d9827a ] Add a set of HD Audio PCI IDS, and the HDMI codec vendor IDs for Glenfly Gpus. - In default_bdl_pos_adj, set bdl to 128 as Glenfly Gpus have hardware limitation, need to increase hdac interrupt interval. - In azx_first_init, enable polling mode for Glenfly Gpu. When the codec complete the command, it sends interrupt and writes response entries to memory, howerver, the write requests sometimes are not actually synchronized to memory when driver handle hdac interrupt on Glenfly Gpus. If the RIRB status is not updated in the interrupt handler, azx_rirb_get_response keeps trying to recevie a response from rirb until 1s timeout. Enabling polling mode for Glenfly Gpu can fix the issue. - In patch_gf_hdmi, set Glenlfy Gpu Codec's no_sticky_stream as it need driver to do actual clean-ups for the linked codec when switch from one codec to another. Signed-off-by: jasontao <jasontao(a)glenfly.com> Signed-off-by: Reaper Li <reaperlioc(a)glenfly.com> Link: https://lore.kernel.org/r/20230426013059.4329-1-reaperlioc@glenfly.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/hda_intel.c | 21 +++++++++++++++++++++ sound/pci/hda/patch_hdmi.c | 22 ++++++++++++++++++++++ 2 files changed, 43 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 77a592f219472..31e16feaa2b4f 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -227,6 +227,7 @@ enum { AZX_DRIVER_ATI, AZX_DRIVER_ATIHDMI, AZX_DRIVER_ATIHDMI_NS, + AZX_DRIVER_GFHDMI, AZX_DRIVER_VIA, AZX_DRIVER_SIS, AZX_DRIVER_ULI, @@ -349,6 +350,7 @@ static const char * const driver_short_names[] = { [AZX_DRIVER_ATI] = "HDA ATI SB", [AZX_DRIVER_ATIHDMI] = "HDA ATI HDMI", [AZX_DRIVER_ATIHDMI_NS] = "HDA ATI HDMI", + [AZX_DRIVER_GFHDMI] = "HDA GF HDMI", [AZX_DRIVER_VIA] = "HDA VIA VT82xx", [AZX_DRIVER_SIS] = "HDA SIS966", [AZX_DRIVER_ULI] = "HDA ULI M5461", @@ -1743,6 +1745,12 @@ static int default_bdl_pos_adj(struct azx *chip) } switch (chip->driver_type) { + /* + * increase the bdl size for Glenfly Gpus for hardware + * limitation on hdac interrupt interval + */ + case AZX_DRIVER_GFHDMI: + return 128; case AZX_DRIVER_ICH: case AZX_DRIVER_PCH: return 1; @@ -1858,6 +1866,12 @@ static int azx_first_init(struct azx *chip) pci_write_config_dword(pci, PCI_BASE_ADDRESS_1, 0); } #endif + /* + * Fix response write request not synced to memory when handle + * hdac interrupt on Glenfly Gpus + */ + if (chip->driver_type == AZX_DRIVER_GFHDMI) + bus->polling_mode = 1; err = pcim_iomap_regions(pci, 1 << 0, "ICH HD audio"); if (err < 0) @@ -1959,6 +1973,7 @@ static int azx_first_init(struct azx *chip) chip->playback_streams = ATIHDMI_NUM_PLAYBACK; chip->capture_streams = ATIHDMI_NUM_CAPTURE; break; + case AZX_DRIVER_GFHDMI: case AZX_DRIVER_GENERIC: default: chip->playback_streams = ICH6_NUM_PLAYBACK; @@ -2724,6 +2739,12 @@ static const struct pci_device_id azx_ids[] = { { PCI_DEVICE(0x1002, 0xab38), .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS | AZX_DCAPS_PM_RUNTIME }, + /* GLENFLY */ + { PCI_DEVICE(0x6766, PCI_ANY_ID), + .class = PCI_CLASS_MULTIMEDIA_HD_AUDIO << 8, + .class_mask = 0xffffff, + .driver_data = AZX_DRIVER_GFHDMI | AZX_DCAPS_POSFIX_LPIB | + AZX_DCAPS_NO_MSI | AZX_DCAPS_NO_64BIT }, /* VIA VT8251/VT8237A */ { PCI_DEVICE(0x1106, 0x3288), .driver_data = AZX_DRIVER_VIA }, /* VIA GFX VT7122/VX900 */ diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 5c6980394dcec..b784ef95390db 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -4489,6 +4489,22 @@ static int patch_via_hdmi(struct hda_codec *codec) return patch_simple_hdmi(codec, VIAHDMI_CVT_NID, VIAHDMI_PIN_NID); } +static int patch_gf_hdmi(struct hda_codec *codec) +{ + int err; + + err = patch_generic_hdmi(codec); + if (err) + return err; + + /* + * Glenfly GPUs have two codecs, stream switches from one codec to + * another, need to do actual clean-ups in codec_cleanup_stream + */ + codec->no_sticky_stream = 1; + return 0; +} + /* * patch entries */ @@ -4579,6 +4595,12 @@ HDA_CODEC_ENTRY(0x10de009f, "GPU 9f HDMI/DP", patch_nvhdmi), HDA_CODEC_ENTRY(0x10de00a0, "GPU a0 HDMI/DP", patch_nvhdmi), HDA_CODEC_ENTRY(0x10de8001, "MCP73 HDMI", patch_nvhdmi_2ch), HDA_CODEC_ENTRY(0x10de8067, "MCP67/68 HDMI", patch_nvhdmi_2ch), +HDA_CODEC_ENTRY(0x67663d82, "Arise 82 HDMI/DP", patch_gf_hdmi), +HDA_CODEC_ENTRY(0x67663d83, "Arise 83 HDMI/DP", patch_gf_hdmi), +HDA_CODEC_ENTRY(0x67663d84, "Arise 84 HDMI/DP", patch_gf_hdmi), +HDA_CODEC_ENTRY(0x67663d85, "Arise 85 HDMI/DP", patch_gf_hdmi), +HDA_CODEC_ENTRY(0x67663d86, "Arise 86 HDMI/DP", patch_gf_hdmi), +HDA_CODEC_ENTRY(0x67663d87, "Arise 87 HDMI/DP", patch_gf_hdmi), HDA_CODEC_ENTRY(0x11069f80, "VX900 HDMI/DP", patch_via_hdmi), HDA_CODEC_ENTRY(0x11069f81, "VX900 HDMI/DP", patch_via_hdmi), HDA_CODEC_ENTRY(0x11069f84, "VX11 HDMI/DP", patch_generic_hdmi), -- 2.39.2

1 0

[PATCH v1 1/3] ASoC: Add ESS ES9218P codec driver
by Aidan MacDonald 11 May '23

11 May '23

The ES9218P is a 32-bit stereo codec featuring an integrated headphone amp with analog volume control and up to 2.0 V rms output, and a low-power bypass path for passing through an analog signal from an alternate source. The digital audio interface supports 16/24/32-bit I2S inputs ranging from 8 KHz to 384 KHz, and DSD inputs. This driver only supports I2S operation. Signed-off-by: Aidan MacDonald <aidanmacdonald.0x0(a)gmail.com> --- sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ess-es9218p.c | 785 +++++++++++++++++++++++++++++++++ 3 files changed, 793 insertions(+) create mode 100644 sound/soc/codecs/ess-es9218p.c diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 8020097d4e4c..973ce164e1e5 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -108,6 +108,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_ES8328_I2C imply SND_SOC_ES7134 imply SND_SOC_ES7241 + imply SND_SOC_ESS_ES9218P imply SND_SOC_GTM601 imply SND_SOC_HDAC_HDMI imply SND_SOC_HDAC_HDA @@ -1017,6 +1018,11 @@ config SND_SOC_ES8328_SPI depends on SPI_MASTER select SND_SOC_ES8328 +config SND_SOC_ESS_ES9218P + tristate "ESS Technology ES9218P CODEC" + depends on I2C + select REGMAP_I2C + config SND_SOC_GTM601 tristate 'GTM601 UMTS modem audio codec' diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 5cdbae88e6e3..1c119f1d9cb1 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -114,6 +114,7 @@ snd-soc-es8326-objs := es8326.o snd-soc-es8328-objs := es8328.o snd-soc-es8328-i2c-objs := es8328-i2c.o snd-soc-es8328-spi-objs := es8328-spi.o +snd-soc-ess-es9218p-objs := ess-es9218p.o snd-soc-gtm601-objs := gtm601.o snd-soc-hdac-hdmi-objs := hdac_hdmi.o snd-soc-hdac-hda-objs := hdac_hda.o @@ -488,6 +489,7 @@ obj-$(CONFIG_SND_SOC_ES8326) += snd-soc-es8326.o obj-$(CONFIG_SND_SOC_ES8328) += snd-soc-es8328.o obj-$(CONFIG_SND_SOC_ES8328_I2C)+= snd-soc-es8328-i2c.o obj-$(CONFIG_SND_SOC_ES8328_SPI)+= snd-soc-es8328-spi.o +obj-$(CONFIG_SND_SOC_ESS_ES9218P) += snd-soc-ess-es9218p.o obj-$(CONFIG_SND_SOC_GTM601) += snd-soc-gtm601.o obj-$(CONFIG_SND_SOC_HDAC_HDMI) += snd-soc-hdac-hdmi.o obj-$(CONFIG_SND_SOC_HDAC_HDA) += snd-soc-hdac-hda.o diff --git a/sound/soc/codecs/ess-es9218p.c b/sound/soc/codecs/ess-es9218p.c new file mode 100644 index 000000000000..70ccf28b2585 --- /dev/null +++ b/sound/soc/codecs/ess-es9218p.c @@ -0,0 +1,785 @@ +// SPDX-License-Identifier: GPL-2.0-or-later +/* + * Copyright (c) 2022-2023 Aidan MacDonald + */ + +#include <linux/bitfield.h> +#include <linux/clk.h> +#include <linux/delay.h> +#include <linux/gpio/consumer.h> +#include <linux/i2c.h> +#include <linux/module.h> +#include <linux/mod_devicetable.h> +#include <linux/regmap.h> +#include <linux/regulator/consumer.h> + +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> + +/* Read/write registers */ +#define ES9218P_SYSTEM 0x00 +#define ES9218P_INPUT_SEL 0x01 +#define ES9218P_MIX_AMUTE 0x02 +#define ES9218P_ANALOGVOL 0x03 +#define ES9218P_AMUTE_TIME 0x04 +#define ES9218P_AMUTE_LEVEL 0x05 +#define ES9218P_DOP_VOL_RAMP 0x06 +#define ES9218P_FBW_MUTE 0x07 +#define ES9218P_GPIO_CONFIG 0x08 +#define ES9218P_MODE_SYNC 0x0a +#define ES9218P_OC_PROTECT 0x0b +#define ES9218P_DPLL_BANDWITH 0x0c +#define ES9218P_THD_MONO_MODE 0x0d +#define ES9218P_SS_CONFIG 0x0e +#define ES9218P_VOL_CH1 0x0f +#define ES9218P_VOL_CH2 0x10 +#define ES9218P_MASTER_TRIM 0x11 /* 32-bit */ +#define ES9218P_GPIO_INPUT_SEL 0x15 +#define ES9218P_THD_C2 0x16 /* 16-bit */ +#define ES9218P_THD_C3 0x18 /* 16-bit */ +#define ES9218P_CP_SS_DELAY 0x1a +#define ES9218P_GEN_CONFIG 0x1b +#define ES9218P_GPIO_INV_ACG 0x1c +#define ES9218P_CP_CLOCK 0x1e /* 16-bit */ +#define ES9218P_AMP_CONFIG 0x20 +#define ES9218P_IMASK 0x21 +#define ES9218P_NCO 0x22 /* 32-bit */ +#define ES9218P_FIR_ADDR 0x28 +#define ES9218P_FIR_DATA 0x29 /* 24-bit */ +#define ES9218P_FIR_CONFIG 0x2c +#define ES9218P_ANALOG_OV_1 0x2d +#define ES9218P_ANALOG_OV_2 0x2e +#define ES9218P_ANALOG_OV_3 0x2f +#define ES9218P_ANALOG_CTRL 0x30 +#define ES9218P_ACG_THRESH 0x31 /* 24-bit */ +#define ES9218P_THD_C2_CH2 0x35 /* 16-bit */ +#define ES9218P_THD_C3_CH2 0x37 /* 16-bit */ + +/* Read only registers */ +#define ES9218P_CHIP_STATUS 0x40 +#define ES9218P_GPIO_READBACK 0x41 +#define ES9218P_DPLL_NUMBER 0x42 /* 32-bit */ +#define ES9218P_INPUT_STATUS 0x48 +#define ES9218P_RAM_READBACK 0x49 /* 24-bit */ + +#define ES9218P_MAX_REGISTER (ES9218P_RAM_READBACK + 2) + +#define ES9218P_SYSTEM_OSC_DRV GENMASK(7, 4) +#define ES9218P_SYSTEM_CLK_GEAR GENMASK(3, 2) + +#define ES9218P_INPUT_SEL_SER_LEN GENMASK(7, 6) +#define ES9218P_INPUT_SEL_SER_MODE GENMASK(5, 4) +#define ES9218P_INPUT_SEL_AUTOSELECT GENMASK(3, 2) +#define ES9218P_INPUT_SEL_DECODER GENMASK(1, 0) + +#define ES9218P_MODE_SYNC_MASTER BIT(7) + +#define ES9218P_SS_CONFIG_QUAD_RAMP BIT(5) +#define ES9218P_SS_CONFIG_TIME GENMASK(4, 0) + +#define ES9218P_GEN_CONFIG_ASRC_EN BIT(7) + +#define ES9218P_AMP_CONFIG_MODE GENMASK(1, 0) +#define ES9218P_AMP_MODE_CORE_ON 0 +#define ES9218P_AMP_MODE_LOWFI 1 +#define ES9218P_AMP_MODE_HIFI_1V 2 +#define ES9218P_AMP_MODE_HIFI_2V 3 + +enum es9218p_supply_id { + ES9218P_SUPPLY_AVDD, + ES9218P_SUPPLY_VCCA, + ES9218P_SUPPLY_AVCC3V3, + ES9218P_SUPPLY_AVCC1V8, + ES9218P_NUM_SUPPLIES +}; + +static const char * const es9218p_supply_names[ES9218P_NUM_SUPPLIES] = { + "avdd", + "vcca", + "avcc3v3", + "avcc1v8", +}; + +/** + * struct es9218p_priv - private data for ES9218P codec + * @supplies: number of supplies + * @mclk: clock for the MCLK/XI input + * @reset_gpio: GPIO for the RESETB pin + * @max_clk_div: maximum safe clock divider for MCLK + * @osc_drv_bias: oscillator drive pad bias + * @soft_start_time: requested soft start time (0-20) + * @output_2v: requested output voltage (true -> 2Vrms, false -> 1Vrms) + * @analog_bypass: requested audio path (true -> AUX input, false -> I2S input) + * @fsr: current sample rate + * @master_mode: true if the codec is supplying the bit and frame clocks + * @powered: true if the codec is powered up + * @out_mux_powered: true if the output mux is powered + * @dac_powered: true if the DAC is powered + */ +struct es9218p_priv { + struct regmap *regmap; + struct regulator_bulk_data supplies[ES9218P_NUM_SUPPLIES]; + struct clk *mclk; + struct gpio_desc *reset_gpio; + unsigned int max_clk_div; + unsigned int osc_drv_bias; + + unsigned int soft_start_time; + bool output_2v; + bool analog_bypass; + + unsigned int fsr; + bool master_mode; + bool powered; + bool out_mux_powered; + bool dac_powered; +}; + +static int es9218p_wide_write(struct regmap *regmap, unsigned int reg, + int count, unsigned int value) +{ + u8 data[4]; + int i; + + for (i = 0; i < count; i++) { + data[i] = value & 0xff; + value >>= 8; + } + + return regmap_raw_write(regmap, reg, data, count); +} + +static void es9218p_apply_clk_params(struct snd_soc_component *component) +{ + struct es9218p_priv *priv = snd_soc_component_get_drvdata(component); + unsigned int clk_div, clk_gear, nco, ss_time, max_ss_time; + unsigned long mclk; + + if (!priv->powered) + return; + + /* + * Calculate the maximum safe clock gear setting based on the + * constraint CLK >= 128*FSR and the platform-specific maximum. + */ + mclk = clk_get_rate(priv->mclk); + + clk_div = mclk / (128 * priv->fsr); + if (clk_div > priv->max_clk_div) + clk_div = priv->max_clk_div; + + if (clk_div >= 8) + clk_gear = 3; + else if (clk_div >= 4) + clk_gear = 2; + else if (clk_div >= 2) + clk_gear = 1; + else + clk_gear = 0; + + snd_soc_component_write_field(component, ES9218P_SYSTEM, + ES9218P_SYSTEM_CLK_GEAR, clk_gear); + + /* + * Calculate soft start time, adjusting for clock gear and clamping + * to ensure the time does not exceed 250ms. That limit is arbitrary, + * but we need to impose _some_ limit because the maximum timeout can + * be absurdly long (minutes to hours). + */ + if (priv->soft_start_time > clk_gear) + ss_time = priv->soft_start_time - clk_gear; + else + ss_time = 0; + + max_ss_time = ilog2((mclk >> clk_gear) / 4096 / 2 / 4); + if (ss_time > max_ss_time) + ss_time = max_ss_time; + + snd_soc_component_write_field(component, ES9218P_SS_CONFIG, + ES9218P_SS_CONFIG_TIME, ss_time); + + /* + * NCO sets the frame clock when in master mode, otherwise it's unused. + */ + if (priv->master_mode) + nco = div_u64((u64)priv->fsr << 32, mclk >> clk_gear); + else + nco = 0; + + snd_soc_component_write_field(component, ES9218P_MODE_SYNC, + ES9218P_MODE_SYNC_MASTER, priv->master_mode); + snd_soc_component_write_field(component, ES9218P_GEN_CONFIG, + ES9218P_GEN_CONFIG_ASRC_EN, !priv->master_mode); + es9218p_wide_write(priv->regmap, ES9218P_NCO, 4, nco); +} + +static int es9218p_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_component *component = dai->component; + struct es9218p_priv *priv = snd_soc_component_get_drvdata(component); + + /* Select I2S decoder, disable decoder autoselection */ + snd_soc_component_write_field(component, ES9218P_INPUT_SEL, + ES9218P_INPUT_SEL_AUTOSELECT, 0); + snd_soc_component_write_field(component, ES9218P_INPUT_SEL, + ES9218P_INPUT_SEL_DECODER, 0); + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + snd_soc_component_write_field(component, ES9218P_INPUT_SEL, + ES9218P_INPUT_SEL_SER_MODE, 0); + break; + case SND_SOC_DAIFMT_LEFT_J: + snd_soc_component_write_field(component, ES9218P_INPUT_SEL, + ES9218P_INPUT_SEL_SER_MODE, 1); + break; + case SND_SOC_DAIFMT_RIGHT_J: + snd_soc_component_write_field(component, ES9218P_INPUT_SEL, + ES9218P_INPUT_SEL_SER_MODE, 2); + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_BP_FP: + priv->master_mode = true; + break; + case SND_SOC_DAIFMT_BC_FC: + priv->master_mode = false; + break; + default: + return -EINVAL; + } + + es9218p_apply_clk_params(component); + return 0; +} + +static int es9218p_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 es9218p_priv *priv = snd_soc_component_get_drvdata(component); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + snd_soc_component_write_field(component, ES9218P_INPUT_SEL, + ES9218P_INPUT_SEL_SER_LEN, 0); + break; + case SNDRV_PCM_FORMAT_S24_LE: + snd_soc_component_write_field(component, ES9218P_INPUT_SEL, + ES9218P_INPUT_SEL_SER_LEN, 1); + break; + case SNDRV_PCM_FORMAT_S32_LE: + snd_soc_component_write_field(component, ES9218P_INPUT_SEL, + ES9218P_INPUT_SEL_SER_LEN, 2); + break; + default: + return -EINVAL; + } + + priv->fsr = params_rate(params); + + es9218p_apply_clk_params(component); + return 0; +} + +static int es9218p_mute_stream(struct snd_soc_dai *dai, int mute, int direction) +{ + struct snd_soc_component *component = dai->component; + + if (mute) + snd_soc_component_update_bits(component, ES9218P_FBW_MUTE, 0x01, 0x01); + else + snd_soc_component_update_bits(component, ES9218P_FBW_MUTE, 0x01, 0); + + return 0; +} + +static const struct snd_soc_dai_ops es9218p_dai_ops = { + .set_fmt = es9218p_set_fmt, + .hw_params = es9218p_hw_params, + .mute_stream = es9218p_mute_stream, +}; + +static struct snd_soc_dai_driver es9218p_dai = { + .name = "es9218p", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 8000, + .rate_max = 384000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE, + }, + .ops = &es9218p_dai_ops, +}; + +static int es9218p_power_up(struct snd_soc_component *component) +{ + struct es9218p_priv *priv = snd_soc_component_get_drvdata(component); + int ret; + + ret = regulator_bulk_enable(ARRAY_SIZE(priv->supplies), priv->supplies); + if (ret) { + dev_err(component->dev, "Cannot enable supplies: %d\n", ret); + return ret; + } + + ret = clk_prepare_enable(priv->mclk); + if (ret) { + dev_err(component->dev, "Cannot enable mclk: %d\n", ret); + goto err_regulator; + } + + gpiod_set_value_cansleep(priv->reset_gpio, 1); + usleep_range(1500, 2000); + + priv->powered = true; + + snd_soc_component_write_field(component, ES9218P_AMP_CONFIG, + ES9218P_AMP_CONFIG_MODE, ES9218P_AMP_MODE_CORE_ON); + snd_soc_component_write_field(component, ES9218P_SYSTEM, + ES9218P_SYSTEM_OSC_DRV, priv->osc_drv_bias); + es9218p_apply_clk_params(component); + + regcache_cache_only(priv->regmap, false); + ret = regcache_sync(priv->regmap); + if (ret) { + regcache_cache_only(priv->regmap, true); + goto err_clk_reset; + } + + return 0; + +err_clk_reset: + priv->powered = false; + gpiod_set_value_cansleep(priv->reset_gpio, 0); + clk_disable_unprepare(priv->mclk); +err_regulator: + regulator_bulk_disable(ARRAY_SIZE(priv->supplies), priv->supplies); + return ret; +} + +static void es9218p_power_down(struct snd_soc_component *component) +{ + struct es9218p_priv *priv = snd_soc_component_get_drvdata(component); + + regcache_cache_only(priv->regmap, true); + regcache_mark_dirty(priv->regmap); + + priv->powered = false; + + gpiod_set_value_cansleep(priv->reset_gpio, 0); + + clk_disable_unprepare(priv->mclk); + regulator_bulk_disable(ARRAY_SIZE(priv->supplies), priv->supplies); +} + +static int es9218p_set_bias_level(struct snd_soc_component *component, + enum snd_soc_bias_level level) +{ + int ret = 0; + + switch (level) { + case SND_SOC_BIAS_STANDBY: + if (snd_soc_component_get_bias_level(component) == SND_SOC_BIAS_OFF) + ret = es9218p_power_up(component); + break; + + case SND_SOC_BIAS_OFF: + es9218p_power_down(component); + break; + + default: + break; + } + + return ret; +} + +/* Returns soft start delay in microseconds */ +static unsigned long es9218p_get_soft_start_delay(struct snd_soc_component *component) +{ + struct es9218p_priv *priv = snd_soc_component_get_drvdata(component); + unsigned long mclk, clk_gear, ss_time; + + if (!priv->powered) + return 0; + + mclk = clk_get_rate(priv->mclk); + clk_gear = snd_soc_component_read_field(component, ES9218P_SYSTEM, + ES9218P_SYSTEM_CLK_GEAR); + ss_time = snd_soc_component_read_field(component, ES9218P_SS_CONFIG, + ES9218P_SS_CONFIG_TIME); + + return div_u64((1000000ull * 4096ull) << (1 + ss_time), mclk >> clk_gear); +} + +static void es9218p_set_amp_mode(struct snd_soc_component *component, int new_mode) +{ + unsigned long ss_delay; + int cur_mode, ret; + + cur_mode = snd_soc_component_read_field(component, ES9218P_AMP_CONFIG, + ES9218P_AMP_CONFIG_MODE); + if (cur_mode == new_mode) + return; + + ss_delay = es9218p_get_soft_start_delay(component); + ret = snd_soc_component_write_field(component, ES9218P_AMP_CONFIG, + ES9218P_AMP_CONFIG_MODE, + ES9218P_AMP_MODE_CORE_ON); + if (ret) { + /* + * We need an extra ~25ms delay here, otherwise the transition + * to the new mode doesn't work. + */ + usleep_range(ss_delay + 25000, ss_delay + 26000); + } + + ret = snd_soc_component_write_field(component, ES9218P_AMP_CONFIG, + ES9218P_AMP_CONFIG_MODE, new_mode); + if (ret) + usleep_range(ss_delay, ss_delay + 1000); +} + +static void es9218p_update_amp_mode(struct snd_soc_component *component) +{ + struct es9218p_priv *priv = snd_soc_component_get_drvdata(component); + int amp_mode; + + if (!priv->out_mux_powered) + amp_mode = ES9218P_AMP_MODE_CORE_ON; + else if (priv->analog_bypass) + amp_mode = ES9218P_AMP_MODE_LOWFI; + else if (!priv->dac_powered) + amp_mode = ES9218P_AMP_MODE_CORE_ON; + else if (priv->output_2v) + amp_mode = ES9218P_AMP_MODE_HIFI_2V; + else + amp_mode = ES9218P_AMP_MODE_HIFI_1V; + + es9218p_set_amp_mode(component, amp_mode); +} + +static int es9218p_output_dapm_event(struct snd_soc_dapm_widget *widget, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = snd_soc_dapm_to_component(widget->dapm); + struct es9218p_priv *priv = snd_soc_component_get_drvdata(component); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + if (widget->id == snd_soc_dapm_dac) + priv->dac_powered = true; + else + priv->out_mux_powered = true; + break; + + case SND_SOC_DAPM_POST_PMD: + if (widget->id == snd_soc_dapm_dac) + priv->dac_powered = false; + else + priv->out_mux_powered = false; + break; + + default: + return 0; + } + + es9218p_update_amp_mode(component); + return 0; +} + +static int outlevel_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct es9218p_priv *priv = snd_soc_component_get_drvdata(component); + + ucontrol->value.enumerated.item[0] = priv->output_2v; + return 0; +} + +static int outlevel_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct es9218p_priv *priv = snd_soc_component_get_drvdata(component); + + priv->output_2v = ucontrol->value.enumerated.item[0]; + es9218p_update_amp_mode(component); + return 1; +} + +static int softstart_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct es9218p_priv *priv = snd_soc_component_get_drvdata(component); + + ucontrol->value.integer.value[0] = priv->soft_start_time; + return 0; +} + +static int softstart_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct es9218p_priv *priv = snd_soc_component_get_drvdata(component); + + priv->soft_start_time = ucontrol->value.integer.value[0]; + es9218p_apply_clk_params(component); + return 1; +} + +static int output_mux_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_dapm_kcontrol_component(kcontrol); + struct es9218p_priv *priv = snd_soc_component_get_drvdata(component); + + ucontrol->value.enumerated.item[0] = priv->analog_bypass; + return 0; +} + +static int output_mux_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_dapm_kcontrol_component(kcontrol); + struct snd_soc_dapm_context *dapm = snd_soc_dapm_kcontrol_dapm(kcontrol); + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + struct es9218p_priv *priv = snd_soc_component_get_drvdata(component); + unsigned int value = ucontrol->value.enumerated.item[0]; + + priv->analog_bypass = value; + es9218p_update_amp_mode(component); + + snd_soc_dapm_mux_update_power(dapm, kcontrol, value, e, NULL); + return 1; +} + +static DECLARE_TLV_DB_SCALE(analog_vol_scale, -2400, 100, 0); +static DECLARE_TLV_DB_SCALE(digital_vol_scale, -12750, 50, 0); + +static const char * const digfilter_texts[] = { + "Linear Phase Fast", + "Linear Phase Slow", + "Minimum Phase Fast", + "Minimum Phase Slow", + "Apodizing Fast Type 1", + "Apodizing Fast Type 2", + "Hybrid Fast", + "Brick Wall" +}; +static SOC_ENUM_SINGLE_DECL(digfilter_enum, ES9218P_FBW_MUTE, 5, digfilter_texts); + +static const char * const outlevel_texts[] = { "1V", "2V" }; +static SOC_ENUM_SINGLE_EXT_DECL(outlevel_enum, outlevel_texts); + +static const char * const channel_mux_texts[] = { "CH1", "CH2" }; +static SOC_ENUM_DOUBLE_DECL(channel_mux_enum, ES9218P_MIX_AMUTE, 0, 2, channel_mux_texts); + +static const struct snd_kcontrol_new es9218p_controls[] = { + SOC_SINGLE_TLV("Headphone Playback Volume", + ES9218P_ANALOGVOL, 0, 24, 1, analog_vol_scale), + SOC_DOUBLE_R_TLV("Master Playback Volume", + ES9218P_VOL_CH1, ES9218P_VOL_CH2, 0, 0xff, 1, digital_vol_scale), + SOC_ENUM("Digital Filter Roll-Off", digfilter_enum), + SOC_ENUM_EXT("Output Level", outlevel_enum, outlevel_get, outlevel_put), + SOC_ENUM("Channel Mux", channel_mux_enum), + SOC_SINGLE_EXT("Soft Start Time", SND_SOC_NOPM, 0, 20, 0, softstart_get, softstart_put), +}; + +static const char * const output_mux_texts[] = { "Digital Input", "AUX Input" }; +static SOC_ENUM_SINGLE_EXT_DECL(output_mux_enum, output_mux_texts); + +static const struct snd_kcontrol_new output_mux_ctl = + SOC_DAPM_ENUM_EXT("Output Source", output_mux_enum, output_mux_get, output_mux_put); + +static const struct snd_soc_dapm_widget es9218p_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("AUX_L"), + SND_SOC_DAPM_INPUT("AUX_R"), + + SND_SOC_DAPM_DAC_E("DAC", "Playback", SND_SOC_NOPM, 0, 0, + es9218p_output_dapm_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + + SND_SOC_DAPM_MUX_E("Output Mux", SND_SOC_NOPM, 0, 0, + &output_mux_ctl, es9218p_output_dapm_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + + SND_SOC_DAPM_OUTPUT("OUTL"), + SND_SOC_DAPM_OUTPUT("OUTR"), + +}; + +static const struct snd_soc_dapm_route es9218p_dapm_routes[] = { + {"Output Mux", "Digital Input", "DAC"}, + {"Output Mux", "AUX Input", "AUX_L"}, + {"Output Mux", "AUX Input", "AUX_R"}, + {"OUTL", NULL, "Output Mux"}, + {"OUTR", NULL, "Output Mux"}, +}; + +static const struct snd_soc_component_driver es9218p_component = { + .set_bias_level = es9218p_set_bias_level, + .controls = es9218p_controls, + .num_controls = ARRAY_SIZE(es9218p_controls), + .dapm_widgets = es9218p_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(es9218p_dapm_widgets), + .dapm_routes = es9218p_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(es9218p_dapm_routes), +}; + +static const struct regmap_range es9218p_readable_ranges[] = { + regmap_reg_range(ES9218P_SYSTEM, ES9218P_THD_C3_CH2+1), + regmap_reg_range(ES9218P_CHIP_STATUS, ES9218P_MAX_REGISTER), +}; + +static const struct regmap_access_table es9218p_readable_table = { + .yes_ranges = es9218p_readable_ranges, + .n_yes_ranges = ARRAY_SIZE(es9218p_readable_ranges), +}; + +static const struct regmap_range es9218p_writeable_ranges[] = { + regmap_reg_range(ES9218P_SYSTEM, ES9218P_THD_C3_CH2+1), +}; + +static const struct regmap_access_table es9218p_writeable_table = { + .yes_ranges = es9218p_writeable_ranges, + .n_yes_ranges = ARRAY_SIZE(es9218p_writeable_ranges), +}; + +static const struct regmap_range es9218p_volatile_ranges[] = { + regmap_reg_range(ES9218P_FIR_ADDR, ES9218P_FIR_CONFIG), + regmap_reg_range(ES9218P_CHIP_STATUS, ES9218P_DPLL_NUMBER+3), + regmap_reg_range(ES9218P_INPUT_STATUS, ES9218P_RAM_READBACK+2), +}; + +static const struct regmap_access_table es9218p_volatile_table = { + .yes_ranges = es9218p_volatile_ranges, + .n_yes_ranges = ARRAY_SIZE(es9218p_volatile_ranges), +}; + +static const u8 es9218p_reg_defaults[] = { + 0x00, 0x8c, 0x34, 0x58, 0x00, 0x68, 0x72, 0x80, + 0xdd, 0x18, 0x02, 0x00, 0x5a, 0x40, 0x0a, 0x50, + 0x50, 0xff, 0xff, 0xff, 0x7f, 0x0d, 0x00, 0x00, + 0x00, 0x00, 0x62, 0xc4, 0xf0, 0x00, 0x00, 0x00, + 0x00, 0x3c, 0x00, 0x00, 0x00, 0x00, 0x00, 0xcc, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x01, 0x62, 0xc0, 0x0d, 0x62, 0x00, 0x00, 0x00, + 0x00, +}; + +static const struct regmap_config es9218p_regmap = { + .reg_bits = 8, + .val_bits = 8, + .rd_table = &es9218p_readable_table, + .wr_table = &es9218p_writeable_table, + .volatile_table = &es9218p_volatile_table, + .max_register = ES9218P_MAX_REGISTER, + .reg_defaults_raw = es9218p_reg_defaults, + .num_reg_defaults_raw = ARRAY_SIZE(es9218p_reg_defaults), + .cache_type = REGCACHE_RBTREE, +}; + +static int es9218p_read_properties(struct device *dev, + struct es9218p_priv *priv) +{ + u32 val; + + if (!device_property_read_u32(dev, "ess,max-clock-div", &val)) + priv->max_clk_div = val; + + if (!device_property_read_u32(dev, "ess,oscillator-pad-bias", &val)) + priv->osc_drv_bias = val; + + return 0; +} + +static int es9218p_i2c_probe(struct i2c_client *i2c) +{ + struct device *dev = &i2c->dev; + struct es9218p_priv *priv; + int ret, i; + + priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + priv->regmap = devm_regmap_init_i2c(i2c, &es9218p_regmap); + if (IS_ERR(priv->regmap)) + return dev_err_probe(dev, PTR_ERR(priv->regmap), "Failed to create regmap\n"); + + /* Codec is only powered when needed */ + regcache_cache_only(priv->regmap, true); + + for (i = 0; i < ES9218P_NUM_SUPPLIES; i++) + priv->supplies[i].supply = es9218p_supply_names[i]; + + ret = devm_regulator_bulk_get(dev, ES9218P_NUM_SUPPLIES, priv->supplies); + if (ret) + return dev_err_probe(dev, ret, "Failed to request supplies\n"); + + priv->mclk = devm_clk_get(dev, "mclk"); + if (IS_ERR(priv->mclk)) + return dev_err_probe(dev, PTR_ERR(priv->mclk), "Failed to request MCLK\n"); + + priv->reset_gpio = devm_gpiod_get(dev, "reset", GPIOD_OUT_LOW); + if (IS_ERR(priv->reset_gpio)) + return dev_err_probe(dev, PTR_ERR(priv->reset_gpio), + "Failed to request reset GPIO\n"); + + ret = es9218p_read_properties(dev, priv); + if (ret) + return ret; + + /* Arbitrary; must be nonzero */ + priv->fsr = 48000; + + i2c_set_clientdata(i2c, priv); + + return devm_snd_soc_register_component(dev, &es9218p_component, &es9218p_dai, 1); +} + +static const struct of_device_id es9218p_of_match[] = { + { .compatible = "ess,es9218p" }, + { } +}; +MODULE_DEVICE_TABLE(of, es9218p_of_match); + +static const struct i2c_device_id es9218p_i2c_id[] = { + { "ess-es9218p", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, es9218p_i2c_id); + +static struct i2c_driver es9218p_i2c_driver = { + .driver = { + .name = "ess-es9218p", + .of_match_table = of_match_ptr(es9218p_of_match), + }, + .probe = es9218p_i2c_probe, + .id_table = es9218p_i2c_id, +}; +module_i2c_driver(es9218p_i2c_driver); + +MODULE_DESCRIPTION("ESS ES9218P ALSA SoC Codec Driver"); +MODULE_AUTHOR("Aidan MacDonald <aidanmacdonald.0x0(a)gmail.com>"); +MODULE_LICENSE("GPL"); -- 2.39.2

2 9

[PATCH] snd-hda-intel: Apply HP B&O top speaker profile to Pavilion 15
by Ryan C. Underwood 11 May '23

11 May '23

The Pavilion 15 line has B&O top speakers similar to the x360 and applying the same profile produces good sound. Without this, the sound would be tinny and underpowered without either applying model=alc295-hp-x360 or booting another OS first. --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e103bb3693c0..7cb232a23a08 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9335,7 +9335,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x802f, "HP Z240", ALC221_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x8077, "HP", ALC256_FIXUP_HP_HEADSET_MIC), SND_PCI_QUIRK(0x103c, 0x8158, "HP", ALC256_FIXUP_HP_HEADSET_MIC), - SND_PCI_QUIRK(0x103c, 0x820d, "HP Pavilion 15", ALC269_FIXUP_HP_MUTE_LED_MIC3), + SND_PCI_QUIRK(0x103c, 0x820d, "HP Pavilion 15", ALC295_FIXUP_HP_X360), SND_PCI_QUIRK(0x103c, 0x8256, "HP", ALC221_FIXUP_HP_FRONT_MIC), SND_PCI_QUIRK(0x103c, 0x827e, "HP x360", ALC295_FIXUP_HP_X360), SND_PCI_QUIRK(0x103c, 0x827f, "HP x360", ALC269_FIXUP_HP_MUTE_LED_MIC3), -- 2.39.2 -- Ryan C. Underwood, <nemesis(a)icequake.net>

3 2

[PATCH v2] ALSA: hda/realtek: Add quirks for ASUS GU604V and GU603V
by Alexandru Sorodoc 11 May '23

11 May '23

These models use 2 CS35L41 amplifiers using SPI for down-facing speakers. alc285_fixup_speaker2_to_dac1 is needed to fix volume control of the down-facing speakers. Pin configs are needed to enable headset mic detection. Note that these models lack the ACPI _DSD properties needed to initialize the amplifiers. They can be added during boot to get working sound out of the speakers: https://gist.github.com/lamperez/862763881c0e1c812392b5574727f6ff Signed-off-by: Alexandru Sorodoc <ealex95(a)gmail.com> --- v2: Fixed position of quirk entries in table sound/pci/hda/patch_realtek.c | 20 ++++++++++++++++++++ 1 file changed, 20 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index bc062c0a1916..4c87a95c5d3d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7063,6 +7063,8 @@ enum { ALC225_FIXUP_DELL1_MIC_NO_PRESENCE, ALC295_FIXUP_DISABLE_DAC3, ALC285_FIXUP_SPEAKER2_TO_DAC1, + ALC285_FIXUP_ASUS_SPEAKER2_TO_DAC1, + ALC285_FIXUP_ASUS_HEADSET_MIC, ALC280_FIXUP_HP_HEADSET_MIC, ALC221_FIXUP_HP_FRONT_MIC, ALC292_FIXUP_TPT460, @@ -8033,6 +8035,22 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_THINKPAD_ACPI }, + [ALC285_FIXUP_ASUS_SPEAKER2_TO_DAC1] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc285_fixup_speaker2_to_dac1, + .chained = true, + .chain_id = ALC245_FIXUP_CS35L41_SPI_2 + }, + [ALC285_FIXUP_ASUS_HEADSET_MIC] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x03a11050 }, + { 0x1b, 0x03a11c30 }, + { } + }, + .chained = true, + .chain_id = ALC285_FIXUP_ASUS_SPEAKER2_TO_DAC1 + }, [ALC256_FIXUP_DELL_INSPIRON_7559_SUBWOOFER] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -9500,6 +9518,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1313, "Asus K42JZ", ALC269VB_FIXUP_ASUS_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x13b0, "ASUS Z550SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_ASUS_ZENBOOK), + SND_PCI_QUIRK(0x1043, 0x1473, "ASUS GU604V", ALC285_FIXUP_ASUS_HEADSET_MIC), + SND_PCI_QUIRK(0x1043, 0x1483, "ASUS GU603V", ALC285_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x1517, "Asus Zenbook UX31A", ALC269VB_FIXUP_ASUS_ZENBOOK_UX31A), SND_PCI_QUIRK(0x1043, 0x1662, "ASUS GV301QH", ALC294_FIXUP_ASUS_DUAL_SPK), SND_PCI_QUIRK(0x1043, 0x1683, "ASUS UM3402YAR", ALC287_FIXUP_CS35L41_I2C_2), -- 2.40.1

2 1

[PATCH 1/1] ALSA: hda/realtek: Add quirks for ASUS GU604V and GU603V
by Alexandru Sorodoc 11 May '23

11 May '23

These models use 2 CS35L41 amplifiers using SPI for down-facing speakers. alc285_fixup_speaker2_to_dac1 is needed to fix volume control of the down-facing speakers. Pin configs are needed to enable headset mic detection. Note that these models lack the ACPI _DSD properties needed to initialize the amplifiers. They can be added during boot to get working sound out of the speakers: https://gist.github.com/lamperez/862763881c0e1c812392b5574727f6ff Signed-off-by: Alexandru Sorodoc <ealex95(a)gmail.com> --- sound/pci/hda/patch_realtek.c | 20 ++++++++++++++++++++ 1 file changed, 20 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index bc062c0a1916..3689c0fca328 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7063,6 +7063,8 @@ enum { ALC225_FIXUP_DELL1_MIC_NO_PRESENCE, ALC295_FIXUP_DISABLE_DAC3, ALC285_FIXUP_SPEAKER2_TO_DAC1, + ALC285_FIXUP_ASUS_SPEAKER2_TO_DAC1, + ALC285_FIXUP_ASUS_HEADSET_MIC, ALC280_FIXUP_HP_HEADSET_MIC, ALC221_FIXUP_HP_FRONT_MIC, ALC292_FIXUP_TPT460, @@ -8033,6 +8035,22 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_THINKPAD_ACPI }, + [ALC285_FIXUP_ASUS_SPEAKER2_TO_DAC1] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc285_fixup_speaker2_to_dac1, + .chained = true, + .chain_id = ALC245_FIXUP_CS35L41_SPI_2 + }, + [ALC285_FIXUP_ASUS_HEADSET_MIC] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x03a11050 }, + { 0x1b, 0x03a11c30 }, + { } + }, + .chained = true, + .chain_id = ALC285_FIXUP_ASUS_SPEAKER2_TO_DAC1 + }, [ALC256_FIXUP_DELL_INSPIRON_7559_SUBWOOFER] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -9543,6 +9561,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x8398, "ASUS P1005", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x83ce, "ASUS P1005", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x8516, "ASUS X101CH", ALC269_FIXUP_ASUS_X101), + SND_PCI_QUIRK(0x1043, 0x1473, "ASUS GU604V", ALC285_FIXUP_ASUS_HEADSET_MIC), + SND_PCI_QUIRK(0x1043, 0x1483, "ASUS GU603V", ALC285_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x104d, 0x9073, "Sony VAIO", ALC275_FIXUP_SONY_VAIO_GPIO2), SND_PCI_QUIRK(0x104d, 0x907b, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ), SND_PCI_QUIRK(0x104d, 0x9084, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ), -- 2.40.1

2 2

[PATCH] ALSA: hda/realtek: Add quirk for HP EliteBook G10 laptops
by Vitaly Rodionov 11 May '23

11 May '23

Add support for HP EliteBook 835/845/845W/865 G10 laptops with CS35L41 amplifiers on I2C/SPI bus connected to Realtek codec. Signed-off-by: Vitaly Rodionov <vitalyr(a)opensource.cirrus.com> --- sound/pci/hda/patch_realtek.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 172ffc2c332b..9e509356790f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9458,7 +9458,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8aa3, "HP ProBook 450 G9 (MB 8AA1)", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8aa8, "HP EliteBook 640 G9 (MB 8AA6)", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8aab, "HP EliteBook 650 G9 (MB 8AA9)", ALC236_FIXUP_HP_GPIO_LED), - SND_PCI_QUIRK(0x103c, 0x8abb, "HP ZBook Firefly 14 G9", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x8abb, "HP ZBook Firefly 14 G9", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8ad1, "HP EliteBook 840 14 inch G9 Notebook PC", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8ad2, "HP EliteBook 860 16 inch G9 Notebook PC", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8b42, "HP", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), @@ -9469,8 +9469,13 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8b47, "HP", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8b5d, "HP", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), SND_PCI_QUIRK(0x103c, 0x8b5e, "HP", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), + SND_PCI_QUIRK(0x103c, 0x8b63, "HP Elite Dragonfly 13.5 inch G4", ALC245_FIXUP_CS35L41_SPI_4_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8b65, "HP ProBook 455 15.6 inch G10 Notebook PC", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), SND_PCI_QUIRK(0x103c, 0x8b66, "HP", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), + SND_PCI_QUIRK(0x103c, 0x8b70, "HP EliteBook 835 G10", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x103c, 0x8b72, "HP EliteBook 845 G10", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x103c, 0x8b74, "HP EliteBook 845W G10", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x103c, 0x8b77, "HP ElieBook 865 G10", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x103c, 0x8b7a, "HP", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8b7d, "HP", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8b87, "HP", ALC236_FIXUP_HP_GPIO_LED), @@ -9481,6 +9486,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8b92, "HP", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8b96, "HP", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), SND_PCI_QUIRK(0x103c, 0x8bf0, "HP", ALC236_FIXUP_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x8c26, "HP HP EliteBook 800G11", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300), SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), -- 2.34.1

2 1

Newest Update for Fedora 37 (alsa-lib-1.2.9-1.fc37) causes crackling with device
by GitHub issues - opened 11 May '23

11 May '23

alsa-project/alsa-ucm-conf issue #314 was opened from ClaraCrazy: First bug report here + not skilled with debugging, so please go easy on me :smile: OS: Fedora 37 Alsa-lib: 1.2.9-1 Issue: Audio crackling Audio Device: `Logic3 TX101 5.1 Soundbar` connected via `Scarlett 2i2 3rd Gen audio interface (USB)` Reverted lib back to `alsa-lib.1.2.8-2.fc37` (which in turn also downgraded alsa-ucm and alsa-utils again) and issues are fixed. I have no idea how to debug this, but would love to help in any way possible, sadly there was no issue template that would give me infos on what logs are useful / where to get those so uh.. please someone just point me in the right direction :) Thanks for your time Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/314 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Newest Update for Fedora 37 (alsa-lib-1.2.9-1.fc37) causes crackling with device
by GitHub issues - edited 11 May '23

11 May '23

alsa-project/alsa-ucm-conf issue #314 was edited from ClaraCrazy: First bug report here + not skilled with debugging, so please go easy on me :smile: OS: Fedora 37 Alsa-lib: 1.2.9-1 Issue: Audio crackling Audio Device: `Logic3 TX101 5.1 Soundbar` connected via `Scarlett 2i2 3rd Gen audio interface (USB)` Reverted lib back to `alsa-lib.1.2.8-2.fc37` (which in turn also downgraded alsa-ucm and alsa-utils again) and issues are fixed. I have no idea how to debug this, but would love to help in any way possible, sadly there was no issue template that would give me infos on what logs are useful / where to get those so uh.. please someone just point me in the right direction :) Thanks for your time Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/314 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] ASoC: soc-pcm: Fix and cleanup DPCM locking
by yixuanjiang 11 May '23

11 May '23

From: Takashi Iwai <tiwai(a)suse.de> The existing locking for DPCM has several issues a) a confusing mix of card->mutex and card->pcm_mutex. b) a dpcm_lock spinlock added inconsistently and on paths that could be recursively taken. The use of irqsave/irqrestore was also overkill. The suggested model is: 1) The pcm_mutex is the top-most protection of BE links in the FE. The pcm_mutex is applied always on either the top PCM callbacks or the external call from DAPM, not taken in the internal functions. 2) the FE stream lock is taken in higher levels before invoking dpcm_be_dai_trigger() 3) when adding and deleting a BE, both the pcm_mutex and FE stream lock are taken. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> [clarification of commit message by plbossart] Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Link: https://lore.kernel.org/r/20211207173745.15850-4-pierre-louis.bossart@linux… Cc: stable(a)vger.kernel.org # 5.15.x Signed-off-by: Mark Brown <broonie(a)kernel.org> --- include/sound/soc.h | 2 - sound/soc/soc-core.c | 1 - sound/soc/soc-pcm.c | 229 ++++++++++++++++++++++++++++--------------- 3 files changed, 152 insertions(+), 80 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 8e6dd8a257c5..5872a8864f3b 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -893,8 +893,6 @@ struct snd_soc_card { struct mutex pcm_mutex; enum snd_soc_pcm_subclass pcm_subclass; - spinlock_t dpcm_lock; - int (*probe)(struct snd_soc_card *card); int (*late_probe)(struct snd_soc_card *card); int (*remove)(struct snd_soc_card *card); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 1b1749b920f4..a5b3ee69fb88 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2339,7 +2339,6 @@ int snd_soc_register_card(struct snd_soc_card *card) mutex_init(&card->mutex); mutex_init(&card->dapm_mutex); mutex_init(&card->pcm_mutex); - spin_lock_init(&card->dpcm_lock); return snd_soc_bind_card(card); } diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 3b673477f621..07ad98934c7a 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -27,6 +27,31 @@ #include <sound/soc-link.h> #include <sound/initval.h> +static inline void snd_soc_dpcm_mutex_lock(struct snd_soc_pcm_runtime *rtd) +{ + mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); +} + +static inline void snd_soc_dpcm_mutex_unlock(struct snd_soc_pcm_runtime *rtd) +{ + mutex_unlock(&rtd->card->pcm_mutex); +} + +#define snd_soc_dpcm_mutex_assert_held(rtd) \ + lockdep_assert_held(&(rtd)->card->pcm_mutex) + +static inline void snd_soc_dpcm_stream_lock_irq(struct snd_soc_pcm_runtime *rtd, + int stream) +{ + snd_pcm_stream_lock_irq(snd_soc_dpcm_get_substream(rtd, stream)); +} + +static inline void snd_soc_dpcm_stream_unlock_irq(struct snd_soc_pcm_runtime *rtd, + int stream) +{ + snd_pcm_stream_unlock_irq(snd_soc_dpcm_get_substream(rtd, stream)); +} + #define DPCM_MAX_BE_USERS 8 static inline const char *soc_cpu_dai_name(struct snd_soc_pcm_runtime *rtd) @@ -73,7 +98,6 @@ static ssize_t dpcm_show_state(struct snd_soc_pcm_runtime *fe, struct snd_pcm_hw_params *params = &fe->dpcm[stream].hw_params; struct snd_soc_dpcm *dpcm; ssize_t offset = 0; - unsigned long flags; /* FE state */ offset += scnprintf(buf + offset, size - offset, @@ -101,7 +125,6 @@ static ssize_t dpcm_show_state(struct snd_soc_pcm_runtime *fe, goto out; } - spin_lock_irqsave(&fe->card->dpcm_lock, flags); for_each_dpcm_be(fe, stream, dpcm) { struct snd_soc_pcm_runtime *be = dpcm->be; params = &dpcm->hw_params; @@ -122,7 +145,6 @@ static ssize_t dpcm_show_state(struct snd_soc_pcm_runtime *fe, params_channels(params), params_rate(params)); } - spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); out: return offset; } @@ -145,11 +167,13 @@ static ssize_t dpcm_state_read_file(struct file *file, char __user *user_buf, if (!buf) return -ENOMEM; + snd_soc_dpcm_mutex_lock(fe); for_each_pcm_streams(stream) if (snd_soc_dai_stream_valid(asoc_rtd_to_cpu(fe, 0), stream)) offset += dpcm_show_state(fe, stream, buf + offset, out_count - offset); + snd_soc_dpcm_mutex_unlock(fe); ret = simple_read_from_buffer(user_buf, count, ppos, buf, offset); @@ -221,14 +245,14 @@ static void dpcm_set_fe_update_state(struct snd_soc_pcm_runtime *fe, struct snd_pcm_substream *substream = snd_soc_dpcm_get_substream(fe, stream); - snd_pcm_stream_lock_irq(substream); + snd_soc_dpcm_stream_lock_irq(fe, stream); if (state == SND_SOC_DPCM_UPDATE_NO && fe->dpcm[stream].trigger_pending) { dpcm_fe_dai_do_trigger(substream, fe->dpcm[stream].trigger_pending - 1); fe->dpcm[stream].trigger_pending = 0; } fe->dpcm[stream].runtime_update = state; - snd_pcm_stream_unlock_irq(substream); + snd_soc_dpcm_stream_unlock_irq(fe, stream); } static void dpcm_set_be_update_state(struct snd_soc_pcm_runtime *be, @@ -256,7 +280,7 @@ void snd_soc_runtime_action(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai; int i; - lockdep_assert_held(&rtd->card->pcm_mutex); + snd_soc_dpcm_mutex_assert_held(rtd); for_each_rtd_dais(rtd, i, dai) snd_soc_dai_action(dai, stream, action); @@ -309,6 +333,8 @@ int dpcm_dapm_stream_event(struct snd_soc_pcm_runtime *fe, int dir, { struct snd_soc_dpcm *dpcm; + snd_soc_dpcm_mutex_assert_held(fe); + for_each_dpcm_be(fe, dir, dpcm) { struct snd_soc_pcm_runtime *be = dpcm->be; @@ -646,14 +672,14 @@ static int soc_pcm_components_close(struct snd_pcm_substream *substream, return ret; } -static int soc_pcm_clean(struct snd_pcm_substream *substream, int rollback) +static int soc_pcm_clean(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_substream *substream, int rollback) { - struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); struct snd_soc_component *component; struct snd_soc_dai *dai; int i; - mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); + snd_soc_dpcm_mutex_assert_held(rtd); if (!rollback) snd_soc_runtime_deactivate(rtd, substream->stream); @@ -665,9 +691,6 @@ static int soc_pcm_clean(struct snd_pcm_substream *substream, int rollback) soc_pcm_components_close(substream, rollback); - - mutex_unlock(&rtd->card->pcm_mutex); - snd_soc_pcm_component_pm_runtime_put(rtd, substream, rollback); for_each_rtd_components(rtd, i, component) @@ -682,9 +705,21 @@ static int soc_pcm_clean(struct snd_pcm_substream *substream, int rollback) * freed here. The cpu DAI, codec DAI, machine and components are also * shutdown. */ +static int __soc_pcm_close(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_substream *substream) +{ + return soc_pcm_clean(rtd, substream, 0); +} + +/* PCM close ops for non-DPCM streams */ static int soc_pcm_close(struct snd_pcm_substream *substream) { - return soc_pcm_clean(substream, 0); + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + + snd_soc_dpcm_mutex_lock(rtd); + soc_pcm_clean(rtd, substream, 0); + snd_soc_dpcm_mutex_unlock(rtd); + return 0; } static int soc_hw_sanity_check(struct snd_pcm_substream *substream) @@ -730,21 +765,21 @@ static int soc_hw_sanity_check(struct snd_pcm_substream *substream) * then initialized and any private data can be allocated. This also calls * startup for the cpu DAI, component, machine and codec DAI. */ -static int soc_pcm_open(struct snd_pcm_substream *substream) +static int __soc_pcm_open(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_substream *substream) { - struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); struct snd_soc_component *component; struct snd_soc_dai *dai; int i, ret = 0; + snd_soc_dpcm_mutex_assert_held(rtd); + for_each_rtd_components(rtd, i, component) pinctrl_pm_select_default_state(component->dev); ret = snd_soc_pcm_component_pm_runtime_get(rtd, substream); if (ret < 0) - goto pm_err; - - mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); + goto err; ret = soc_pcm_components_open(substream); if (ret < 0) @@ -786,16 +821,26 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) snd_soc_runtime_activate(rtd, substream->stream); ret = 0; err: - mutex_unlock(&rtd->card->pcm_mutex); -pm_err: if (ret < 0) { - soc_pcm_clean(substream, 1); + soc_pcm_clean(rtd, substream, 1); dev_err(rtd->dev, "%s() failed (%d)", __func__, ret); } return ret; } +/* PCM open ops for non-DPCM streams */ +static int soc_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + int ret; + + snd_soc_dpcm_mutex_lock(rtd); + ret = __soc_pcm_open(rtd, substream); + snd_soc_dpcm_mutex_unlock(rtd); + return ret; +} + static void codec2codec_close_delayed_work(struct snd_soc_pcm_runtime *rtd) { /* @@ -811,13 +856,13 @@ static void codec2codec_close_delayed_work(struct snd_soc_pcm_runtime *rtd) * rate, etc. This function is non atomic and can be called multiple times, * it can refer to the runtime info. */ -static int soc_pcm_prepare(struct snd_pcm_substream *substream) +static int __soc_pcm_prepare(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_substream *substream) { - struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); struct snd_soc_dai *dai; int i, ret = 0; - mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); + snd_soc_dpcm_mutex_assert_held(rtd); ret = snd_soc_link_prepare(substream); if (ret < 0) @@ -845,14 +890,24 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) snd_soc_dai_digital_mute(dai, 0, substream->stream); out: - mutex_unlock(&rtd->card->pcm_mutex); - if (ret < 0) dev_err(rtd->dev, "ASoC: %s() failed (%d)\n", __func__, ret); return ret; } +/* PCM prepare ops for non-DPCM streams */ +static int soc_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + int ret; + + snd_soc_dpcm_mutex_lock(rtd); + ret = __soc_pcm_prepare(rtd, substream); + snd_soc_dpcm_mutex_unlock(rtd); + return ret; +} + static void soc_pcm_codec_params_fixup(struct snd_pcm_hw_params *params, unsigned int mask) { @@ -864,13 +919,13 @@ static void soc_pcm_codec_params_fixup(struct snd_pcm_hw_params *params, interval->max = channels; } -static int soc_pcm_hw_clean(struct snd_pcm_substream *substream, int rollback) +static int soc_pcm_hw_clean(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_substream *substream, int rollback) { - struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); struct snd_soc_dai *dai; int i; - mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); + snd_soc_dpcm_mutex_assert_held(rtd); /* clear the corresponding DAIs parameters when going to be inactive */ for_each_rtd_dais(rtd, i, dai) { @@ -900,16 +955,28 @@ static int soc_pcm_hw_clean(struct snd_pcm_substream *substream, int rollback) snd_soc_dai_hw_free(dai, substream, rollback); } - mutex_unlock(&rtd->card->pcm_mutex); return 0; } /* * Frees resources allocated by hw_params, can be called multiple times */ +static int __soc_pcm_hw_free(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_substream *substream) +{ + return soc_pcm_hw_clean(rtd, substream, 0); +} + +/* hw_free PCM ops for non-DPCM streams */ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) { - return soc_pcm_hw_clean(substream, 0); + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + int ret; + + snd_soc_dpcm_mutex_lock(rtd); + ret = __soc_pcm_hw_free(rtd, substream); + snd_soc_dpcm_mutex_unlock(rtd); + return ret; } /* @@ -917,15 +984,15 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) * function can also be called multiple times and can allocate buffers * (using snd_pcm_lib_* ). It's non-atomic. */ -static int soc_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) +static int __soc_pcm_hw_params(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) { - struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); struct snd_soc_dai *cpu_dai; struct snd_soc_dai *codec_dai; int i, ret = 0; - mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); + snd_soc_dpcm_mutex_assert_held(rtd); ret = soc_pcm_params_symmetry(substream, params); if (ret) @@ -997,16 +1064,27 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, ret = snd_soc_pcm_component_hw_params(substream, params); out: - mutex_unlock(&rtd->card->pcm_mutex); - if (ret < 0) { - soc_pcm_hw_clean(substream, 1); + soc_pcm_hw_clean(rtd, substream, 1); dev_err(rtd->dev, "ASoC: %s() failed (%d)\n", __func__, ret); } return ret; } +/* hw_params PCM ops for non-DPCM streams */ +static int soc_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + int ret; + + snd_soc_dpcm_mutex_lock(rtd); + ret = __soc_pcm_hw_params(rtd, substream, params); + snd_soc_dpcm_mutex_unlock(rtd); + return ret; +} + static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) { struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); @@ -1124,7 +1202,8 @@ static int dpcm_be_connect(struct snd_soc_pcm_runtime *fe, struct snd_soc_pcm_runtime *be, int stream) { struct snd_soc_dpcm *dpcm; - unsigned long flags; + + snd_soc_dpcm_mutex_assert_held(fe); /* only add new dpcms */ for_each_dpcm_be(fe, stream, dpcm) { @@ -1140,10 +1219,10 @@ static int dpcm_be_connect(struct snd_soc_pcm_runtime *fe, dpcm->fe = fe; be->dpcm[stream].runtime = fe->dpcm[stream].runtime; dpcm->state = SND_SOC_DPCM_LINK_STATE_NEW; - spin_lock_irqsave(&fe->card->dpcm_lock, flags); + snd_soc_dpcm_stream_lock_irq(fe, stream); list_add(&dpcm->list_be, &fe->dpcm[stream].be_clients); list_add(&dpcm->list_fe, &be->dpcm[stream].fe_clients); - spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); + snd_soc_dpcm_stream_unlock_irq(fe, stream); dev_dbg(fe->dev, "connected new DPCM %s path %s %s %s\n", stream ? "capture" : "playback", fe->dai_link->name, @@ -1188,8 +1267,10 @@ static void dpcm_be_reparent(struct snd_soc_pcm_runtime *fe, void dpcm_be_disconnect(struct snd_soc_pcm_runtime *fe, int stream) { struct snd_soc_dpcm *dpcm, *d; - unsigned long flags; + snd_soc_dpcm_mutex_assert_held(fe); + + snd_soc_dpcm_stream_lock_irq(fe, stream); for_each_dpcm_be_safe(fe, stream, dpcm, d) { dev_dbg(fe->dev, "ASoC: BE %s disconnect check for %s\n", stream ? "capture" : "playback", @@ -1207,12 +1288,11 @@ void dpcm_be_disconnect(struct snd_soc_pcm_runtime *fe, int stream) dpcm_remove_debugfs_state(dpcm); - spin_lock_irqsave(&fe->card->dpcm_lock, flags); list_del(&dpcm->list_be); list_del(&dpcm->list_fe); - spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); kfree(dpcm); } + snd_soc_dpcm_stream_unlock_irq(fe, stream); } /* get BE for DAI widget and stream */ @@ -1426,12 +1506,9 @@ int dpcm_process_paths(struct snd_soc_pcm_runtime *fe, void dpcm_clear_pending_state(struct snd_soc_pcm_runtime *fe, int stream) { struct snd_soc_dpcm *dpcm; - unsigned long flags; - spin_lock_irqsave(&fe->card->dpcm_lock, flags); for_each_dpcm_be(fe, stream, dpcm) dpcm_set_be_update_state(dpcm->be, stream, SND_SOC_DPCM_UPDATE_NO); - spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); } void dpcm_be_dai_stop(struct snd_soc_pcm_runtime *fe, int stream, @@ -1467,12 +1544,12 @@ void dpcm_be_dai_stop(struct snd_soc_pcm_runtime *fe, int stream, continue; if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_FREE) { - soc_pcm_hw_free(be_substream); + __soc_pcm_hw_free(be, be_substream); be->dpcm[stream].state = SND_SOC_DPCM_STATE_HW_FREE; } } - soc_pcm_close(be_substream); + __soc_pcm_close(be, be_substream); be_substream->runtime = NULL; be->dpcm[stream].state = SND_SOC_DPCM_STATE_CLOSE; } @@ -1520,7 +1597,7 @@ int dpcm_be_dai_startup(struct snd_soc_pcm_runtime *fe, int stream) stream ? "capture" : "playback", be->dai_link->name); be_substream->runtime = be->dpcm[stream].runtime; - err = soc_pcm_open(be_substream); + err = __soc_pcm_open(be, be_substream); if (err < 0) { be->dpcm[stream].users--; if (be->dpcm[stream].users < 0) @@ -1764,7 +1841,7 @@ static int dpcm_fe_dai_startup(struct snd_pcm_substream *fe_substream) dev_dbg(fe->dev, "ASoC: open FE %s\n", fe->dai_link->name); /* start the DAI frontend */ - ret = soc_pcm_open(fe_substream); + ret = __soc_pcm_open(fe, fe_substream); if (ret < 0) goto unwind; @@ -1795,6 +1872,8 @@ static int dpcm_fe_dai_shutdown(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *fe = asoc_substream_to_rtd(substream); int stream = substream->stream; + snd_soc_dpcm_mutex_assert_held(fe); + dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_FE); /* shutdown the BEs */ @@ -1803,7 +1882,7 @@ static int dpcm_fe_dai_shutdown(struct snd_pcm_substream *substream) dev_dbg(fe->dev, "ASoC: close FE %s\n", fe->dai_link->name); /* now shutdown the frontend */ - soc_pcm_close(substream); + __soc_pcm_close(fe, substream); /* run the stream stop event */ dpcm_dapm_stream_event(fe, stream, SND_SOC_DAPM_STREAM_STOP); @@ -1848,7 +1927,7 @@ void dpcm_be_dai_hw_free(struct snd_soc_pcm_runtime *fe, int stream) dev_dbg(be->dev, "ASoC: hw_free BE %s\n", be->dai_link->name); - soc_pcm_hw_free(be_substream); + __soc_pcm_hw_free(be, be_substream); be->dpcm[stream].state = SND_SOC_DPCM_STATE_HW_FREE; } @@ -1859,13 +1938,13 @@ static int dpcm_fe_dai_hw_free(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *fe = asoc_substream_to_rtd(substream); int stream = substream->stream; - mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME); + snd_soc_dpcm_mutex_lock(fe); dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_FE); dev_dbg(fe->dev, "ASoC: hw_free FE %s\n", fe->dai_link->name); /* call hw_free on the frontend */ - soc_pcm_hw_free(substream); + soc_pcm_hw_clean(fe, substream, 0); /* only hw_params backends that are either sinks or sources * to this frontend DAI */ @@ -1874,7 +1953,7 @@ static int dpcm_fe_dai_hw_free(struct snd_pcm_substream *substream) fe->dpcm[stream].state = SND_SOC_DPCM_STATE_HW_FREE; dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_NO); - mutex_unlock(&fe->card->mutex); + snd_soc_dpcm_mutex_unlock(fe); return 0; } @@ -1918,7 +1997,7 @@ int dpcm_be_dai_hw_params(struct snd_soc_pcm_runtime *fe, int stream) dev_dbg(be->dev, "ASoC: hw_params BE %s\n", be->dai_link->name); - ret = soc_pcm_hw_params(be_substream, &dpcm->hw_params); + ret = __soc_pcm_hw_params(be, be_substream, &dpcm->hw_params); if (ret < 0) goto unwind; @@ -1948,7 +2027,7 @@ int dpcm_be_dai_hw_params(struct snd_soc_pcm_runtime *fe, int stream) (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP)) continue; - soc_pcm_hw_free(be_substream); + __soc_pcm_hw_free(be, be_substream); } return ret; @@ -1960,7 +2039,7 @@ static int dpcm_fe_dai_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *fe = asoc_substream_to_rtd(substream); int ret, stream = substream->stream; - mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME); + snd_soc_dpcm_mutex_lock(fe); dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_FE); memcpy(&fe->dpcm[stream].hw_params, params, @@ -1974,7 +2053,7 @@ static int dpcm_fe_dai_hw_params(struct snd_pcm_substream *substream, params_channels(params), params_format(params)); /* call hw_params on the frontend */ - ret = soc_pcm_hw_params(substream, params); + ret = __soc_pcm_hw_params(fe, substream, params); if (ret < 0) dpcm_be_dai_hw_free(fe, stream); else @@ -1982,7 +2061,7 @@ static int dpcm_fe_dai_hw_params(struct snd_pcm_substream *substream, out: dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_NO); - mutex_unlock(&fe->card->mutex); + snd_soc_dpcm_mutex_unlock(fe); if (ret < 0) dev_err(fe->dev, "ASoC: %s failed (%d)\n", __func__, ret); @@ -2253,7 +2332,7 @@ int dpcm_be_dai_prepare(struct snd_soc_pcm_runtime *fe, int stream) dev_dbg(be->dev, "ASoC: prepare BE %s\n", be->dai_link->name); - ret = soc_pcm_prepare(be_substream); + ret = __soc_pcm_prepare(be, be_substream); if (ret < 0) break; @@ -2271,7 +2350,7 @@ static int dpcm_fe_dai_prepare(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *fe = asoc_substream_to_rtd(substream); int stream = substream->stream, ret = 0; - mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME); + snd_soc_dpcm_mutex_lock(fe); dev_dbg(fe->dev, "ASoC: prepare FE %s\n", fe->dai_link->name); @@ -2290,7 +2369,7 @@ static int dpcm_fe_dai_prepare(struct snd_pcm_substream *substream) goto out; /* call prepare on the frontend */ - ret = soc_pcm_prepare(substream); + ret = __soc_pcm_prepare(fe, substream); if (ret < 0) goto out; @@ -2298,7 +2377,7 @@ static int dpcm_fe_dai_prepare(struct snd_pcm_substream *substream) out: dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_NO); - mutex_unlock(&fe->card->mutex); + snd_soc_dpcm_mutex_unlock(fe); if (ret < 0) dev_err(fe->dev, "ASoC: %s() failed (%d)\n", __func__, ret); @@ -2349,7 +2428,6 @@ static int dpcm_run_update_startup(struct snd_soc_pcm_runtime *fe, int stream) struct snd_soc_dpcm *dpcm; enum snd_soc_dpcm_trigger trigger = fe->dai_link->trigger[stream]; int ret = 0; - unsigned long flags; dev_dbg(fe->dev, "ASoC: runtime %s open on FE %s\n", stream ? "capture" : "playback", fe->dai_link->name); @@ -2418,7 +2496,6 @@ static int dpcm_run_update_startup(struct snd_soc_pcm_runtime *fe, int stream) dpcm_be_dai_shutdown(fe, stream); disconnect: /* disconnect any pending BEs */ - spin_lock_irqsave(&fe->card->dpcm_lock, flags); for_each_dpcm_be(fe, stream, dpcm) { struct snd_soc_pcm_runtime *be = dpcm->be; @@ -2430,7 +2507,6 @@ static int dpcm_run_update_startup(struct snd_soc_pcm_runtime *fe, int stream) be->dpcm[stream].state == SND_SOC_DPCM_STATE_NEW) dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE; } - spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); if (ret < 0) dev_err(fe->dev, "ASoC: %s() failed (%d)\n", __func__, ret); @@ -2505,7 +2581,7 @@ int snd_soc_dpcm_runtime_update(struct snd_soc_card *card) struct snd_soc_pcm_runtime *fe; int ret = 0; - mutex_lock_nested(&card->mutex, SND_SOC_CARD_CLASS_RUNTIME); + mutex_lock_nested(&card->pcm_mutex, card->pcm_subclass); /* shutdown all old paths first */ for_each_card_rtds(card, fe) { ret = soc_dpcm_fe_runtime_update(fe, 0); @@ -2521,7 +2597,7 @@ int snd_soc_dpcm_runtime_update(struct snd_soc_card *card) } out: - mutex_unlock(&card->mutex); + mutex_unlock(&card->pcm_mutex); return ret; } EXPORT_SYMBOL_GPL(snd_soc_dpcm_runtime_update); @@ -2532,6 +2608,8 @@ static void dpcm_fe_dai_cleanup(struct snd_pcm_substream *fe_substream) struct snd_soc_dpcm *dpcm; int stream = fe_substream->stream; + snd_soc_dpcm_mutex_assert_held(fe); + /* mark FE's links ready to prune */ for_each_dpcm_be(fe, stream, dpcm) dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE; @@ -2546,12 +2624,12 @@ static int dpcm_fe_dai_close(struct snd_pcm_substream *fe_substream) struct snd_soc_pcm_runtime *fe = asoc_substream_to_rtd(fe_substream); int ret; - mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME); + snd_soc_dpcm_mutex_lock(fe); ret = dpcm_fe_dai_shutdown(fe_substream); dpcm_fe_dai_cleanup(fe_substream); - mutex_unlock(&fe->card->mutex); + snd_soc_dpcm_mutex_unlock(fe); return ret; } @@ -2562,7 +2640,7 @@ static int dpcm_fe_dai_open(struct snd_pcm_substream *fe_substream) int ret; int stream = fe_substream->stream; - mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME); + snd_soc_dpcm_mutex_lock(fe); fe->dpcm[stream].runtime = fe_substream->runtime; ret = dpcm_path_get(fe, stream, &list); @@ -2579,7 +2657,7 @@ static int dpcm_fe_dai_open(struct snd_pcm_substream *fe_substream) dpcm_clear_pending_state(fe, stream); dpcm_path_put(&list); open_end: - mutex_unlock(&fe->card->mutex); + snd_soc_dpcm_mutex_unlock(fe); return ret; } @@ -2840,10 +2918,8 @@ static int snd_soc_dpcm_check_state(struct snd_soc_pcm_runtime *fe, struct snd_soc_dpcm *dpcm; int state; int ret = 1; - unsigned long flags; int i; - spin_lock_irqsave(&fe->card->dpcm_lock, flags); for_each_dpcm_fe(be, stream, dpcm) { if (dpcm->fe == fe) @@ -2857,7 +2933,6 @@ static int snd_soc_dpcm_check_state(struct snd_soc_pcm_runtime *fe, } } } - spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); /* it's safe to do this BE DAI */ return ret; -- 2.40.1.495.gc816e09b53d-goog

4 7

[PATCH v3] ASoC: dt-bindings: nau8825: Convert to dtschema
by David Lin 11 May '23

11 May '23

Convert the NAU8825 audio CODEC bindings to DT schema. Signed-off-by: David Lin <CTLIN0(a)nuvoton.com> Changes: V2 -> V3: - refine node name from "nau8825" to "codec" for generic purpose V1 -> V2: - add interrupts properties - add maximum to nuvoton,jack-insert-debounce and nuvoton,jack-eject-debounce properites - add a enum item for nuvoton,short-key-debounce properites - add default value for most properites - add maxItems to clocks properites and mclk entries to clock-names properites - refine wrong indentation from clocks and clock-names - refine dts example for interrupts and clocks - remove headset label from dts example --- .../devicetree/bindings/sound/nau8825.txt | 111 -------- .../bindings/sound/nuvoton,nau8825.yaml | 242 ++++++++++++++++++ 2 files changed, 242 insertions(+), 111 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/nau8825.txt create mode 100644 Documentation/devicetree/bindings/sound/nuvoton,nau8825.yaml diff --git a/Documentation/devicetree/bindings/sound/nau8825.txt b/Documentation/devicetree/bindings/sound/nau8825.txt deleted file mode 100644 index a9c34526f4cb..000000000000 --- a/Documentation/devicetree/bindings/sound/nau8825.txt +++ /dev/null @@ -1,111 +0,0 @@ -Nuvoton NAU8825 audio codec - -This device supports I2C only. - -Required properties: - - compatible : Must be "nuvoton,nau8825" - - - reg : the I2C address of the device. This is either 0x1a (CSB=0) or 0x1b (CSB=1). - -Optional properties: - - nuvoton,jkdet-enable: Enable jack detection via JKDET pin. - - nuvoton,jkdet-pull-enable: Enable JKDET pin pull. If set - pin pull enabled, - otherwise pin in high impedance state. - - nuvoton,jkdet-pull-up: Pull-up JKDET pin. If set then JKDET pin is pull up, otherwise pull down. - - nuvoton,jkdet-polarity: JKDET pin polarity. 0 - active high, 1 - active low. - - - nuvoton,vref-impedance: VREF Impedance selection - 0 - Open - 1 - 25 kOhm - 2 - 125 kOhm - 3 - 2.5 kOhm - - - nuvoton,micbias-voltage: Micbias voltage level. - 0 - VDDA - 1 - VDDA - 2 - VDDA * 1.1 - 3 - VDDA * 1.2 - 4 - VDDA * 1.3 - 5 - VDDA * 1.4 - 6 - VDDA * 1.53 - 7 - VDDA * 1.53 - - - nuvoton,sar-threshold-num: Number of buttons supported - - nuvoton,sar-threshold: Impedance threshold for each button. Array that contains up to 8 buttons configuration. SAR value is calculated as - SAR = 255 * MICBIAS / SAR_VOLTAGE * R / (2000 + R) - where MICBIAS is configured by 'nuvoton,micbias-voltage', SAR_VOLTAGE is configured by 'nuvoton,sar-voltage', R - button impedance. - Refer datasheet section 10.2 for more information about threshold calculation. - - - nuvoton,sar-hysteresis: Button impedance measurement hysteresis. - - - nuvoton,sar-voltage: Reference voltage for button impedance measurement. - 0 - VDDA - 1 - VDDA - 2 - VDDA * 1.1 - 3 - VDDA * 1.2 - 4 - VDDA * 1.3 - 5 - VDDA * 1.4 - 6 - VDDA * 1.53 - 7 - VDDA * 1.53 - - - nuvoton,sar-compare-time: SAR compare time - 0 - 500 ns - 1 - 1 us - 2 - 2 us - 3 - 4 us - - - nuvoton,sar-sampling-time: SAR sampling time - 0 - 2 us - 1 - 4 us - 2 - 8 us - 3 - 16 us - - - nuvoton,short-key-debounce: Button short key press debounce time. - 0 - 30 ms - 1 - 50 ms - 2 - 100 ms - 3 - 30 ms - - - nuvoton,jack-insert-debounce: number from 0 to 7 that sets debounce time to 2^(n+2) ms - - nuvoton,jack-eject-debounce: number from 0 to 7 that sets debounce time to 2^(n+2) ms - - - nuvoton,crosstalk-enable: make crosstalk function enable if set. - - - nuvoton,adcout-drive-strong: make the drive strength of ADCOUT IO PIN strong if set. - Otherwise, the drive keeps normal strength. - - - nuvoton,adc-delay-ms: Delay (in ms) to make input path stable and avoid pop noise. The - default value is 125 and range between 125 to 500 ms. - - - clocks: list of phandle and clock specifier pairs according to common clock bindings for the - clocks described in clock-names - - clock-names: should include "mclk" for the MCLK master clock - -Example: - - headset: nau8825@1a { - compatible = "nuvoton,nau8825"; - reg = <0x1a>; - interrupt-parent = <&gpio>; - interrupts = <TEGRA_GPIO(E, 6) IRQ_TYPE_LEVEL_LOW>; - nuvoton,jkdet-enable; - nuvoton,jkdet-pull-enable; - nuvoton,jkdet-pull-up; - nuvoton,jkdet-polarity = <GPIO_ACTIVE_LOW>; - nuvoton,vref-impedance = <2>; - nuvoton,micbias-voltage = <6>; - // Setup 4 buttons impedance according to Android specification - nuvoton,sar-threshold-num = <4>; - nuvoton,sar-threshold = <0xc 0x1e 0x38 0x60>; - nuvoton,sar-hysteresis = <1>; - nuvoton,sar-voltage = <0>; - nuvoton,sar-compare-time = <0>; - nuvoton,sar-sampling-time = <0>; - nuvoton,short-key-debounce = <2>; - nuvoton,jack-insert-debounce = <7>; - nuvoton,jack-eject-debounce = <7>; - nuvoton,crosstalk-enable; - - clock-names = "mclk"; - clocks = <&tegra_pmc TEGRA_PMC_CLK_OUT_2>; - }; diff --git a/Documentation/devicetree/bindings/sound/nuvoton,nau8825.yaml b/Documentation/devicetree/bindings/sound/nuvoton,nau8825.yaml new file mode 100644 index 000000000000..a68c553d7464 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nuvoton,nau8825.yaml @@ -0,0 +1,242 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nuvoton,nau8825.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NAU8825 audio CODEC + +maintainers: + - John Hsu <KCHSU0(a)nuvoton.com> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - nuvoton,nau8825 + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + description: + The CODEC's interrupt output. + + nuvoton,jkdet-enable: + description: + Enable jack detection via JKDET pin. + type: boolean + + nuvoton,jkdet-pull-enable: + description: + Enable JKDET pin pull. + If set - pin pull enabled, otherwise pin in high impedance state. + type: boolean + + nuvoton,jkdet-pull-up: + description: + Pull-up JKDET pin. + If set then JKDET pin is pull up, otherwise pull down. + type: boolean + + nuvoton,jkdet-polarity: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + JKDET pin polarity. + enum: + - 0 # active high + - 1 # active low + default: 1 + + nuvoton,vref-impedance: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + VREF Impedance selection. + enum: + - 0 # Open + - 1 # 25 kOhm + - 2 # 125 kOhm + - 3 # 2.5 kOhm + default: 2 + + nuvoton,micbias-voltage: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + Micbias voltage level. + enum: + - 0 # VDDA + - 1 # VDDA + - 2 # VDDA * 1.1 + - 3 # VDDA * 1.2 + - 4 # VDDA * 1.3 + - 5 # VDDA * 1.4 + - 6 # VDDA * 1.53 + - 7 # VDDA * 1.53 + default: 6 + + nuvoton,sar-threshold-num: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + Number of buttons supported. + minimum: 1 + maximum: 4 + default: 4 + + nuvoton,sar-threshold: + $ref: /schemas/types.yaml#/definitions/uint32-array + description: + Impedance threshold for each button. Array that contains up to 8 buttons + configuration. SAR value is calculated as + SAR = 255 * MICBIAS / SAR_VOLTAGE * R / (2000 + R) where MICBIAS is + configured by 'nuvoton,micbias-voltage', SAR_VOLTAGE is configured by + 'nuvoton,sar-voltage', R - button impedance. + Refer datasheet section 10.2 for more information about threshold + calculation. + minItems: 1 + maxItems: 4 + items: + minimum: 0 + maximum: 255 + + nuvoton,sar-hysteresis: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + Button impedance measurement hysteresis. + default: 0 + + nuvoton,sar-voltage: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + Reference voltage for button impedance measurement. + enum: + - 0 # VDDA + - 1 # VDDA + - 2 # VDDA * 1.1 + - 3 # VDDA * 1.2 + - 4 # VDDA * 1.3 + - 5 # VDDA * 1.4 + - 6 # VDDA * 1.53 + - 7 # VDDA * 1.53 + default: 6 + + nuvoton,sar-compare-time: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + SAR compare time. + enum: + - 0 # 500 ns + - 1 # 1 us + - 2 # 2 us + - 3 # 4 us + default: 1 + + nuvoton,sar-sampling-time: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + SAR sampling time. + enum: + - 0 # 2 us + - 1 # 4 us + - 2 # 8 us + - 3 # 16 us + default: 1 + + nuvoton,short-key-debounce: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + Button short key press debounce time. + enum: + - 0 # 30 ms + - 1 # 50 ms + - 2 # 100 ms + - 3 # 30 ms + default: 3 + + nuvoton,jack-insert-debounce: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + number from 0 to 7 that sets debounce time to 2^(n+2) ms. + maximum: 7 + default: 7 + + nuvoton,jack-eject-debounce: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + number from 0 to 7 that sets debounce time to 2^(n+2) ms + maximum: 7 + default: 0 + + nuvoton,crosstalk-enable: + description: + make crosstalk function enable if set. + type: boolean + + nuvoton,adcout-drive-strong: + description: + make the drive strength of ADCOUT IO PIN strong if set. + Otherwise, the drive keeps normal strength. + type: boolean + + nuvoton,adc-delay-ms: + description: + Delay (in ms) to make input path stable and avoid pop noise. + The default value is 125 and range between 125 to 500 ms. + minimum: 125 + maximum: 500 + default: 125 + + clocks: + description: + list of phandle and clock specifier pairs according to common clock + bindings for the clocks described in clock-names. + maxItems: 1 + + clock-names: + description: + should include "mclk" for the MCLK master clock. + items: + - const: mclk + +required: + - compatible + - reg + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + #include <dt-bindings/interrupt-controller/irq.h> + i2c { + #address-cells = <1>; + #size-cells = <0>; + codec@1a { + compatible = "nuvoton,nau8825"; + reg = <0x1a>; + interrupt-parent = <&gpio>; + interrupts = <38 IRQ_TYPE_LEVEL_LOW>; + nuvoton,jkdet-enable; + nuvoton,jkdet-pull-enable; + nuvoton,jkdet-pull-up; + nuvoton,jkdet-polarity = <GPIO_ACTIVE_LOW>; + nuvoton,vref-impedance = <2>; + nuvoton,micbias-voltage = <6>; + // Setup 4 buttons impedance according to Android specification + nuvoton,sar-threshold-num = <4>; + nuvoton,sar-threshold = <0xc 0x1e 0x38 0x60>; + nuvoton,sar-hysteresis = <1>; + nuvoton,sar-voltage = <0>; + nuvoton,sar-compare-time = <0>; + nuvoton,sar-sampling-time = <0>; + nuvoton,short-key-debounce = <2>; + nuvoton,jack-insert-debounce = <7>; + nuvoton,jack-eject-debounce = <7>; + nuvoton,crosstalk-enable; + + clock-names = "mclk"; + clocks = <&tegra_pmc 1>; + }; + }; -- 2.25.1

3 2

[PATCH 0/4] Add support for IIO devices in ASoC
by Herve Codina 11 May '23

11 May '23

Several weeks ago, I sent a series [1] for adding a potentiometer as an auxiliary device in ASoC. The feedback was that the potentiometer should be directly handled in IIO (as other potentiometers) and something more generic should be present in ASoC in order to have a binding to import some IIO devices into sound cards. The series related to the IIO potentiometer device is already under review [2]. This series introduces simple-iio-aux. Its goal is to offer the binding between IIO and ASoC. It exposes attached IIO devices as ASoC auxiliary devices and allows to control them through mixer controls. On my system, the IIO device is a potentiometer and it is present in an amplifier design present in the audio path. Best regards, Hervé [1] https://lore.kernel.org/linux-kernel/20230203111422.142479-1-herve.codina@b… [2] https://lore.kernel.org/linux-kernel/20230421085245.302169-1-herve.codina@b… Herve Codina (4): dt-bindings: sound: Add simple-iio-aux iio: inkern: Add a helper to query an available minimum raw value ASoC: soc-dapm.h: Add a helper to build a DAPM widget dynamically ASoC: codecs: Add support for the generic IIO auxiliary devices .../bindings/sound/simple-iio-aux.yaml | 65 ++++ drivers/iio/inkern.c | 67 ++++ include/linux/iio/consumer.h | 11 + include/sound/soc-dapm.h | 12 +- sound/soc/codecs/Kconfig | 12 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/simple-iio-aux.c | 307 ++++++++++++++++++ 7 files changed, 475 insertions(+), 1 deletion(-) create mode 100644 Documentation/devicetree/bindings/sound/simple-iio-aux.yaml create mode 100644 sound/soc/codecs/simple-iio-aux.c -- 2.39.2

5 16

Can't play 44100hz, 88200hz files Natively.
by GitHub issues - opened 11 May '23

11 May '23

alsa-project/alsa-ucm-conf issue #313 was opened from RenEH01: Can't play audio files at native sample rate. Everything using the Hifi 2.0 output configuration forces a sample rate of 48000hz. When I switch to the Pro Audio driver under pavucontrol, 44100 hz can play natively. Is there a forced resampling occuring in the UCM2 driver? Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/313 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH 0/4] ASoC: qcom: add display port module support.
by Srinivas Kandagatla 11 May '23

11 May '23

This patchset adds support to displayport on AudioReach. Patches are tested on X13s with two display ports. Srinivas Kandagatla (4): ASoC: qcom: q6dsp-common: move channel allocation to common ASoC: qcom: audioreach: add support for DISPLAY PORT SINK module ASoC: qcom: q6dsp: add support to more display ports ASoC: qcom: q6apm: add support to display ports in lpass dais .../sound/qcom,q6dsp-lpass-ports.h | 8 ++ sound/soc/qcom/qdsp6/audioreach.c | 75 +++++++++++++++++++ sound/soc/qcom/qdsp6/audioreach.h | 12 +++ sound/soc/qcom/qdsp6/q6afe-dai.c | 34 ++------- sound/soc/qcom/qdsp6/q6apm-lpass-dais.c | 39 ++++++++++ sound/soc/qcom/qdsp6/q6dsp-common.c | 35 +++++++++ sound/soc/qcom/qdsp6/q6dsp-common.h | 1 + sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c | 43 +++++++---- 8 files changed, 204 insertions(+), 43 deletions(-) -- 2.21.0

2 5

[alsa-devel@alsa-project.org: [PATCH 3/5] ASoC: mediatek: mt8195-afe-pcm: Simplify runtime PM during probe]
by Mark Brown 11 May '23

11 May '23

Hi, Here's another mail (one of several in this series) that got completely mangled by the alsa-project.org mailman to the point of unusability. I didn't see any response to my last mail about this, is there any news on fixing mailman to not do this? It's extremely disruptive to working with lore. Thanks, Mark

6 30

[PATCH v1] ASoC: jz4740-i2s: Make I2S divider calculations more robust
by Aidan MacDonald 11 May '23

11 May '23

When the CPU supplies bit/frame clocks, the system clock (clk_i2s) is divided to produce the bit clock. This is a simple 1/N divider with a fairly limited range, so for a given system clock frequency only a few sample rates can be produced. Usually a wider range of sample rates is supported by varying the system clock frequency. The old calculation method was not very robust and could easily produce the wrong clock rate, especially with non-standard rates. For example, if the system clock is 1.99x the target bit clock rate, the divider would be calculated as 1 instead of the more accurate 2. Instead, use a more accurate method that considers two adjacent divider settings and selects the one that produces the least error versus the requested rate. If the error is 5% or higher then the rate setting is rejected to prevent garbled audio. Skip divider calculation when the codec is supplying both the bit and frame clock; in that case, the divider outputs are unused and we don't want to constrain the sample rate. Signed-off-by: Aidan MacDonald <aidanmacdonald.0x0(a)gmail.com> --- sound/soc/jz4740/jz4740-i2s.c | 54 ++++++++++++++++++++++++++++++++--- 1 file changed, 50 insertions(+), 4 deletions(-) diff --git a/sound/soc/jz4740/jz4740-i2s.c b/sound/soc/jz4740/jz4740-i2s.c index 6d9cfe0a5041..d0f6c945d9ae 100644 --- a/sound/soc/jz4740/jz4740-i2s.c +++ b/sound/soc/jz4740/jz4740-i2s.c @@ -218,18 +218,48 @@ static int jz4740_i2s_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) return 0; } +static int jz4740_i2s_get_i2sdiv(unsigned long mclk, unsigned long rate, + unsigned long i2sdiv_max) +{ + unsigned long div, rate1, rate2, err1, err2; + + div = mclk / (64 * rate); + if (div == 0) + div = 1; + + rate1 = mclk / (64 * div); + rate2 = mclk / (64 * (div + 1)); + + err1 = abs(rate1 - rate); + err2 = abs(rate2 - rate); + + /* + * Choose the divider that produces the smallest error in the + * output rate and reject dividers with a 5% or higher error. + * In the event that both dividers are outside the acceptable + * error margin, reject the rate to prevent distorted audio. + * (The number 5% is arbitrary.) + */ + if (div <= i2sdiv_max && err1 <= err2 && err1 < rate/20) + return div; + if (div < i2sdiv_max && err2 < rate/20) + return div + 1; + + return -EINVAL; +} + static int jz4740_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { struct jz4740_i2s *i2s = snd_soc_dai_get_drvdata(dai); struct regmap_field *div_field; + unsigned long i2sdiv_max; unsigned int sample_size; - uint32_t ctrl; - int div; + uint32_t ctrl, conf; + int div = 1; regmap_read(i2s->regmap, JZ_REG_AIC_CTRL, &ctrl); - - div = clk_get_rate(i2s->clk_i2s) / (64 * params_rate(params)); + regmap_read(i2s->regmap, JZ_REG_AIC_CONF, &conf); switch (params_format(params)) { case SNDRV_PCM_FORMAT_S8: @@ -258,11 +288,27 @@ static int jz4740_i2s_hw_params(struct snd_pcm_substream *substream, ctrl &= ~JZ_AIC_CTRL_MONO_TO_STEREO; div_field = i2s->field_i2sdiv_playback; + i2sdiv_max = GENMASK(i2s->soc_info->field_i2sdiv_playback.msb, + i2s->soc_info->field_i2sdiv_playback.lsb); } else { ctrl &= ~JZ_AIC_CTRL_INPUT_SAMPLE_SIZE; ctrl |= FIELD_PREP(JZ_AIC_CTRL_INPUT_SAMPLE_SIZE, sample_size); div_field = i2s->field_i2sdiv_capture; + i2sdiv_max = GENMASK(i2s->soc_info->field_i2sdiv_capture.msb, + i2s->soc_info->field_i2sdiv_capture.lsb); + } + + /* + * Only calculate I2SDIV if we're supplying the bit or frame clock. + * If the codec is supplying both clocks then the divider output is + * unused, and we don't want it to limit the allowed sample rates. + */ + if (conf & (JZ_AIC_CONF_BIT_CLK_MASTER | JZ_AIC_CONF_SYNC_CLK_MASTER)) { + div = jz4740_i2s_get_i2sdiv(clk_get_rate(i2s->clk_i2s), + params_rate(params), i2sdiv_max); + if (div < 0) + return div; } regmap_write(i2s->regmap, JZ_REG_AIC_CTRL, ctrl); -- 2.39.2

2 1

[PATCH] ASoC: dt-bindings: Adjust #sound-dai-cells on TI's single-DAI codecs
by Martin Povišer 11 May '23

11 May '23

A bunch of TI's codecs have binding schemas which force #sound-dai-cells to one despite those codecs only having a single DAI. Allow for bindings with zero DAI cells and deprecate the former non-zero value. Signed-off-by: Martin Povišer <povik+lin(a)cutebit.org> --- Documentation/devicetree/bindings/sound/tas2562.yaml | 6 ++++-- Documentation/devicetree/bindings/sound/tas2770.yaml | 6 ++++-- Documentation/devicetree/bindings/sound/tas27xx.yaml | 6 ++++-- 3 files changed, 12 insertions(+), 6 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/tas2562.yaml b/Documentation/devicetree/bindings/sound/tas2562.yaml index a5bb561bfcfb..31a3024ea789 100644 --- a/Documentation/devicetree/bindings/sound/tas2562.yaml +++ b/Documentation/devicetree/bindings/sound/tas2562.yaml @@ -55,7 +55,9 @@ properties: description: TDM TX current sense time slot. '#sound-dai-cells': - const: 1 + # The codec has a single DAI, the #sound-dai-cells=<1>; case is left in for backward + # compatibility but is deprecated. + enum: [0, 1] required: - compatible @@ -72,7 +74,7 @@ examples: codec: codec@4c { compatible = "ti,tas2562"; reg = <0x4c>; - #sound-dai-cells = <1>; + #sound-dai-cells = <0>; interrupt-parent = <&gpio1>; interrupts = <14>; shutdown-gpios = <&gpio1 15 0>; diff --git a/Documentation/devicetree/bindings/sound/tas2770.yaml b/Documentation/devicetree/bindings/sound/tas2770.yaml index 26088adb9dc2..8908bf1122e9 100644 --- a/Documentation/devicetree/bindings/sound/tas2770.yaml +++ b/Documentation/devicetree/bindings/sound/tas2770.yaml @@ -57,7 +57,9 @@ properties: - 1 # Falling edge '#sound-dai-cells': - const: 1 + # The codec has a single DAI, the #sound-dai-cells=<1>; case is left in for backward + # compatibility but is deprecated. + enum: [0, 1] required: - compatible @@ -74,7 +76,7 @@ examples: codec: codec@41 { compatible = "ti,tas2770"; reg = <0x41>; - #sound-dai-cells = <1>; + #sound-dai-cells = <0>; interrupt-parent = <&gpio1>; interrupts = <14>; reset-gpio = <&gpio1 15 0>; diff --git a/Documentation/devicetree/bindings/sound/tas27xx.yaml b/Documentation/devicetree/bindings/sound/tas27xx.yaml index 8cba01316855..a876545ec87d 100644 --- a/Documentation/devicetree/bindings/sound/tas27xx.yaml +++ b/Documentation/devicetree/bindings/sound/tas27xx.yaml @@ -50,7 +50,9 @@ properties: description: TDM TX voltage sense time slot. '#sound-dai-cells': - const: 1 + # The codec has a single DAI, the #sound-dai-cells=<1>; case is left in for backward + # compatibility but is deprecated. + enum: [0, 1] required: - compatible @@ -67,7 +69,7 @@ examples: codec: codec@38 { compatible = "ti,tas2764"; reg = <0x38>; - #sound-dai-cells = <1>; + #sound-dai-cells = <0>; interrupt-parent = <&gpio1>; interrupts = <14>; reset-gpios = <&gpio1 15 0>; -- 2.38.3

3 5

[PATCH v3 0/3] Add TDM audio on StarFive JH7110
by Walker Chen 11 May '23

11 May '23

This patchset adds TDM audio driver for the StarFive JH7110 SoC. The first patch adds device tree binding for TDM module. The second patch adds tdm driver support for JH7110 SoC. The last patch adds device node of tdm and sound card to JH7110 dts. The series has been tested on the VisionFive 2 board by plugging an audio expansion board. For more information of audio expansion board, you can take a look at the following webpage: https://wiki.seeedstudio.com/ReSpeaker_2_Mics_Pi_HAT/ Changes since v2: - Use dt-overlay to describe sound card because need to plug the audio expansion board into the VisionFive2 board. - Modified the coding style for driver. - Moved assignment of stop_dma_first to startup function of dai_driver. - Dropped some useless macro definition. - Use loops to get/enable/disable clocks. Changes since v1: - Rebased on Linux 6.3-rc4. - Added the dts file dedicated to describe audio card. - Added the item for JH7110 audio board to the dt-binding of StarFive SoC-based boards. --- v2: https://lore.kernel.org/all/20230420024118.22677-1-walker.chen@starfivetech… v1: https://lore.kernel.org/all/20230329153320.31390-1-walker.chen@starfivetech… Walker Chen (3): dt-bindings: sound: Add TDM for StarFive JH7110 ASoC: starfive: Add JH7110 TDM driver riscv: dts: starfive: add tdm node and sound card .../bindings/sound/starfive,jh7110-tdm.yaml | 98 +++ MAINTAINERS | 6 + arch/riscv/boot/dts/starfive/Makefile | 3 + .../jh7110-starfive-visionfive-2-wm8960.dtso | 70 +++ .../jh7110-starfive-visionfive-2.dtsi | 40 ++ arch/riscv/boot/dts/starfive/jh7110.dtsi | 21 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/starfive/Kconfig | 15 + sound/soc/starfive/Makefile | 2 + sound/soc/starfive/jh7110_tdm.c | 573 ++++++++++++++++++ sound/soc/starfive/jh7110_tdm.h | 147 +++++ 12 files changed, 977 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/starfive,jh7110-tdm.yaml create mode 100644 arch/riscv/boot/dts/starfive/jh7110-starfive-visionfive-2-wm8960.dtso create mode 100644 sound/soc/starfive/Kconfig create mode 100644 sound/soc/starfive/Makefile create mode 100644 sound/soc/starfive/jh7110_tdm.c create mode 100644 sound/soc/starfive/jh7110_tdm.h base-commit: 197b6b60ae7bc51dd0814953c562833143b292aa prerequisite-patch-id: 30bec4dba6f250a6edd0c2cbab2ce09442e50e8a prerequisite-patch-id: bb939c0c7c26b08addfccd890f9d3974b6eaec53 prerequisite-patch-id: 8a6f135bcabdad4a4bfb21f0c6a0ffd2bb57efe7 prerequisite-patch-id: c2366f993a9d85e28c06d8d09f064dd5e8b29a61 prerequisite-patch-id: 50d53a21f91f4087fc80b6f1f72864adfb0002b9 prerequisite-patch-id: 0df3703af91c30f1ca2c47f5609012f2d7200028 prerequisite-patch-id: 89f049f951e5acf75aab92541992f816fd0acc0d prerequisite-patch-id: 551fae54377090044c3612fca9740a9b359abdd2 prerequisite-patch-id: c7fdf904f398d478f0ed6d57eb878982bc73329d prerequisite-patch-id: 1b2d0982b18da060c82134f05bf3ce16425bac8d prerequisite-patch-id: 090ba4b78d47bc19204916e76fdbc70021785388 prerequisite-patch-id: a5d9e0f7d4f8163f566678894cf693015119f2d9 prerequisite-patch-id: 4637a8fa2334a45fa6b64351f4e9e28d3e2d60d3 prerequisite-patch-id: 32647ec60a3b614e1c59ec8e54cb511ae832c22f prerequisite-patch-id: aa06658ecf89c92d0dfdd6a4ba6d9e6e67532971 prerequisite-patch-id: 1387a7e87b446329dfc21f3e575ceae7ebcf954c prerequisite-patch-id: 258ea5f9b8bf41b6981345dcc81795f25865d38f prerequisite-patch-id: 8b6f2c9660c0ac0ee4e73e4c21aca8e6b75e81b9 prerequisite-patch-id: dbb0c0151b8bdf093e6ce79fd2fe3f60791a6e0b prerequisite-patch-id: 9007c8610fdcd387592475949864edde874c20a2 prerequisite-patch-id: d57e95d31686772abc4c4d5aa1cadc344dc293cd prerequisite-patch-id: 9f911969d0a550648493952c99096d26e05d4d83 prerequisite-patch-id: 2ddada18ab6ea5cd1da14212aaf59632f5203d40 prerequisite-patch-id: 80042661ff6156ce577a72e9eb8c0b218b624829 prerequisite-patch-id: 398744c61913c76a35754de867c4f820ca7a8d99 prerequisite-patch-id: f59269382164b5d642a5e10443ca447f5caa595c prerequisite-patch-id: 1babe83d6bf999bad17584dc595480f9070a5369 prerequisite-patch-id: 77be3d122d66df813f13088141ce27b21107a341 prerequisite-patch-id: 9fbb7ad1dd258bb8ff5946c4a0e59de4bfd82a04 prerequisite-patch-id: 6f6984916dffd0cc66aa733c9b6bd3a55495a50c prerequisite-patch-id: 39e1be2a3d1593577ab997f55f59367cba665aa7 prerequisite-patch-id: 584c256c9acb52ee2773d0c81c3f4977fc18155a prerequisite-patch-id: b37ac15032973e1fcd918f157c82a0606775c9e9 prerequisite-patch-id: 32deea16304859842af5c2151bc41d91cf6dfc9b prerequisite-patch-id: 20ac2450fb93b3f69f83fc720fd4800a95e618a6 prerequisite-patch-id: 6abf359fa445f4104432ddee27044dfbfb128417 -- 2.17.1

6 17

[PATCH 1/2] ASoC: dt-bindings: Add adi,ssm3515 amp schema
by Martin Povišer 11 May '23

11 May '23

Add a DT schema for the SSM3515 amp by Analog Devices. It's a simple mono amplifier with digital input. Signed-off-by: Martin Povišer <povik+lin(a)cutebit.org> --- .../bindings/sound/adi,ssm3515.yaml | 66 +++++++++++++++++++ 1 file changed, 66 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/adi,ssm3515.yaml diff --git a/Documentation/devicetree/bindings/sound/adi,ssm3515.yaml b/Documentation/devicetree/bindings/sound/adi,ssm3515.yaml new file mode 100644 index 000000000000..19b7185ae8e2 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/adi,ssm3515.yaml @@ -0,0 +1,66 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +# Copyright (C) 2019-20 Texas Instruments Incorporated +%YAML 1.2 +--- +$id: "http://devicetree.org/schemas/sound/adi,ssm3515.yaml#" +$schema: "http://devicetree.org/meta-schemas/core.yaml#" + +title: Analog Devices SSM3515 Audio Amplifier + +maintainers: + - Martin Povišer <povik+lin(a)cutebit.org> + +description: | + SSM3515 is a mono Class-D audio amplifier with digital input. + + https://www.analog.com/media/en/technical-documentation/data-sheets/SSM3515… + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - adi,ssm3515 + + reg: + maxItems: 1 + + adi,ana-gain: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: [0, 1, 2, 3] + description: | + The value to be set in the ANA_GAIN register field on the codec. This determines + the full voltage span of the codec's analog output. + + To quote the datasheet on the available options: + + 00: 8.4 V Full-Scale Gain Mapping + 01: 12.6 V Full-Scale Gain Mapping + 10: 14 V Full-Scale Gain Mapping + 11: 15 V Full-Scale Gain Mapping + + '#sound-dai-cells': + const: 0 + +required: + - compatible + - reg + - adi,ana-gain + +additionalProperties: true + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + codec@14 { + compatible = "adi,ssm3515"; + reg = <0x14>; + #sound-dai-cells = <0>; + adi,ana-gain = <0>; + sound-name-prefix = "Left Tweeter"; + }; + }; -- 2.38.3

4 13

[PATCH AUTOSEL 5.15 2/4] soundwire: qcom: gracefully handle too many ports in DT
by Sasha Levin 10 May '23

10 May '23

From: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> [ Upstream commit 2367e0ecb498764e95cfda691ff0828f7d25f9a4 ] There are two issues related to the number of ports coming from Devicetree when exceeding in total QCOM_SDW_MAX_PORTS. Both lead to incorrect memory accesses: 1. With DTS having too big value of input or output ports, the driver, when copying port parameters from local/stack arrays into 'pconfig' array in 'struct qcom_swrm_ctrl', will iterate over their sizes. 2. If DTS also has too many parameters for these ports (e.g. qcom,ports-sinterval-low), the driver will overflow buffers on the stack when reading these properties from DTS. Add a sanity check so incorrect DTS will not cause kernel memory corruption. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Reviewed-by: Konrad Dybcio <konrad.dybcio(a)linaro.org> Link: https://lore.kernel.org/r/20230222144412.237832-2-krzysztof.kozlowski@linar… Signed-off-by: Vinod Koul <vkoul(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- drivers/soundwire/qcom.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c index 500035a1fd460..e76d4cee5e7f4 100644 --- a/drivers/soundwire/qcom.c +++ b/drivers/soundwire/qcom.c @@ -1142,6 +1142,9 @@ static int qcom_swrm_get_port_config(struct qcom_swrm_ctrl *ctrl) ctrl->num_dout_ports = val; nports = ctrl->num_dout_ports + ctrl->num_din_ports; + if (nports > QCOM_SDW_MAX_PORTS) + return -EINVAL; + /* Valid port numbers are from 1-14, so mask out port 0 explicitly */ set_bit(0, &ctrl->dout_port_mask); set_bit(0, &ctrl->din_port_mask); -- 2.39.2

1 0

[PATCH AUTOSEL 6.1 1/8] soundwire: dmi-quirks: add remapping for Intel 'Rooks County' NUC M15
by Sasha Levin 10 May '23

10 May '23

From: Eugene Huang <eugene.huang99(a)gmail.com> [ Upstream commit 01b33e284ca28cc977bdcfb23be2c719f2139175 ] Same DSDT problem as the HP Omen 16-k0005TX, except rt1316 amp is on link2. Link: https://github.com/thesofproject/linux/issues/4088 Signed-off-by: Eugene Huang <eugene.huang99(a)gmail.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Link: https://lore.kernel.org/r/20230314090618.498716-1-yung-chuan.liao@linux.int… Signed-off-by: Vinod Koul <vkoul(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- drivers/soundwire/dmi-quirks.c | 25 +++++++++++++++++++++++++ 1 file changed, 25 insertions(+) diff --git a/drivers/soundwire/dmi-quirks.c b/drivers/soundwire/dmi-quirks.c index 7969881f126dc..58ea013fa918a 100644 --- a/drivers/soundwire/dmi-quirks.c +++ b/drivers/soundwire/dmi-quirks.c @@ -73,6 +73,23 @@ static const struct adr_remap hp_omen_16[] = { {} }; +/* + * Intel NUC M15 LAPRC510 and LAPRC710 + */ +static const struct adr_remap intel_rooks_county[] = { + /* rt711-sdca on link0 */ + { + 0x000020025d071100ull, + 0x000030025d071101ull + }, + /* rt1316-sdca on link2 */ + { + 0x000120025d071100ull, + 0x000230025d131601ull + }, + {} +}; + static const struct dmi_system_id adr_remap_quirk_table[] = { /* TGL devices */ { @@ -98,6 +115,14 @@ static const struct dmi_system_id adr_remap_quirk_table[] = { }, .driver_data = (void *)intel_tgl_bios, }, + { + /* quirk used for NUC15 'Rooks County' LAPRC510 and LAPRC710 skews */ + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Intel(R) Client Systems"), + DMI_MATCH(DMI_PRODUCT_NAME, "LAPRC"), + }, + .driver_data = (void *)intel_rooks_county, + }, { .matches = { DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), -- 2.39.2

1 2

[PATCH AUTOSEL 6.2 5/9] soundwire: bus: Fix unbalanced pm_runtime_put() causing usage count underflow
by Sasha Levin 10 May '23

10 May '23

From: Richard Fitzgerald <rf(a)opensource.cirrus.com> [ Upstream commit e9537962519e88969f5f69cd0571eb4f6984403c ] This reverts commit 443a98e649b4 ("soundwire: bus: use pm_runtime_resume_and_get()") Change calls to pm_runtime_resume_and_get() back to pm_runtime_get_sync(). This fixes a usage count underrun caused by doing a pm_runtime_put() even though pm_runtime_resume_and_get() returned an error. The three affected functions ignore -EACCES error from trying to get pm_runtime, and carry on, including a put at the end of the function. But pm_runtime_resume_and_get() does not increment the usage count if it returns an error. So in the -EACCES case you must not call pm_runtime_put(). The documentation for pm_runtime_get_sync() says: "Consider using pm_runtime_resume_and_get() ... as this is likely to result in cleaner code." In this case I don't think it results in cleaner code because the pm_runtime_put() at the end of the function would have to be conditional on the return value from pm_runtime_resume_and_get() at the top of the function. pm_runtime_get_sync() doesn't have this problem because it always increments the count, so always needs a put. The code can just flow through and do the pm_runtime_put() unconditionally. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20230406134640.8582-1-rf@opensource.cirrus.com Signed-off-by: Vinod Koul <vkoul(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- drivers/soundwire/bus.c | 15 ++++++++++----- 1 file changed, 10 insertions(+), 5 deletions(-) diff --git a/drivers/soundwire/bus.c b/drivers/soundwire/bus.c index 76515c33e639e..4fd221d0cc818 100644 --- a/drivers/soundwire/bus.c +++ b/drivers/soundwire/bus.c @@ -571,9 +571,11 @@ int sdw_nread(struct sdw_slave *slave, u32 addr, size_t count, u8 *val) { int ret; - ret = pm_runtime_resume_and_get(&slave->dev); - if (ret < 0 && ret != -EACCES) + ret = pm_runtime_get_sync(&slave->dev); + if (ret < 0 && ret != -EACCES) { + pm_runtime_put_noidle(&slave->dev); return ret; + } ret = sdw_nread_no_pm(slave, addr, count, val); @@ -595,9 +597,11 @@ int sdw_nwrite(struct sdw_slave *slave, u32 addr, size_t count, const u8 *val) { int ret; - ret = pm_runtime_resume_and_get(&slave->dev); - if (ret < 0 && ret != -EACCES) + ret = pm_runtime_get_sync(&slave->dev); + if (ret < 0 && ret != -EACCES) { + pm_runtime_put_noidle(&slave->dev); return ret; + } ret = sdw_nwrite_no_pm(slave, addr, count, val); @@ -1565,9 +1569,10 @@ static int sdw_handle_slave_alerts(struct sdw_slave *slave) sdw_modify_slave_status(slave, SDW_SLAVE_ALERT); - ret = pm_runtime_resume_and_get(&slave->dev); + ret = pm_runtime_get_sync(&slave->dev); if (ret < 0 && ret != -EACCES) { dev_err(&slave->dev, "Failed to resume device: %d\n", ret); + pm_runtime_put_noidle(&slave->dev); return ret; } -- 2.39.2

1 0

[PATCH AUTOSEL 6.2 4/9] soundwire: qcom: gracefully handle too many ports in DT
by Sasha Levin 10 May '23

10 May '23

From: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> [ Upstream commit 2367e0ecb498764e95cfda691ff0828f7d25f9a4 ] There are two issues related to the number of ports coming from Devicetree when exceeding in total QCOM_SDW_MAX_PORTS. Both lead to incorrect memory accesses: 1. With DTS having too big value of input or output ports, the driver, when copying port parameters from local/stack arrays into 'pconfig' array in 'struct qcom_swrm_ctrl', will iterate over their sizes. 2. If DTS also has too many parameters for these ports (e.g. qcom,ports-sinterval-low), the driver will overflow buffers on the stack when reading these properties from DTS. Add a sanity check so incorrect DTS will not cause kernel memory corruption. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Reviewed-by: Konrad Dybcio <konrad.dybcio(a)linaro.org> Link: https://lore.kernel.org/r/20230222144412.237832-2-krzysztof.kozlowski@linar… Signed-off-by: Vinod Koul <vkoul(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- drivers/soundwire/qcom.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c index 3354248702900..3e80a4d388cbb 100644 --- a/drivers/soundwire/qcom.c +++ b/drivers/soundwire/qcom.c @@ -1217,6 +1217,9 @@ static int qcom_swrm_get_port_config(struct qcom_swrm_ctrl *ctrl) ctrl->num_dout_ports = val; nports = ctrl->num_dout_ports + ctrl->num_din_ports; + if (nports > QCOM_SDW_MAX_PORTS) + return -EINVAL; + /* Valid port numbers are from 1-14, so mask out port 0 explicitly */ set_bit(0, &ctrl->dout_port_mask); set_bit(0, &ctrl->din_port_mask); -- 2.39.2

1 0

[PATCH AUTOSEL 6.2 2/9] soundwire: dmi-quirks: add remapping for Intel 'Rooks County' NUC M15
by Sasha Levin 10 May '23

10 May '23

From: Eugene Huang <eugene.huang99(a)gmail.com> [ Upstream commit 01b33e284ca28cc977bdcfb23be2c719f2139175 ] Same DSDT problem as the HP Omen 16-k0005TX, except rt1316 amp is on link2. Link: https://github.com/thesofproject/linux/issues/4088 Signed-off-by: Eugene Huang <eugene.huang99(a)gmail.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Link: https://lore.kernel.org/r/20230314090618.498716-1-yung-chuan.liao@linux.int… Signed-off-by: Vinod Koul <vkoul(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- drivers/soundwire/dmi-quirks.c | 25 +++++++++++++++++++++++++ 1 file changed, 25 insertions(+) diff --git a/drivers/soundwire/dmi-quirks.c b/drivers/soundwire/dmi-quirks.c index 7969881f126dc..58ea013fa918a 100644 --- a/drivers/soundwire/dmi-quirks.c +++ b/drivers/soundwire/dmi-quirks.c @@ -73,6 +73,23 @@ static const struct adr_remap hp_omen_16[] = { {} }; +/* + * Intel NUC M15 LAPRC510 and LAPRC710 + */ +static const struct adr_remap intel_rooks_county[] = { + /* rt711-sdca on link0 */ + { + 0x000020025d071100ull, + 0x000030025d071101ull + }, + /* rt1316-sdca on link2 */ + { + 0x000120025d071100ull, + 0x000230025d131601ull + }, + {} +}; + static const struct dmi_system_id adr_remap_quirk_table[] = { /* TGL devices */ { @@ -98,6 +115,14 @@ static const struct dmi_system_id adr_remap_quirk_table[] = { }, .driver_data = (void *)intel_tgl_bios, }, + { + /* quirk used for NUC15 'Rooks County' LAPRC510 and LAPRC710 skews */ + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Intel(R) Client Systems"), + DMI_MATCH(DMI_PRODUCT_NAME, "LAPRC"), + }, + .driver_data = (void *)intel_rooks_county, + }, { .matches = { DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), -- 2.39.2

1 0

[PATCH AUTOSEL 6.3 5/9] soundwire: bus: Fix unbalanced pm_runtime_put() causing usage count underflow
by Sasha Levin 10 May '23

10 May '23

From: Richard Fitzgerald <rf(a)opensource.cirrus.com> [ Upstream commit e9537962519e88969f5f69cd0571eb4f6984403c ] This reverts commit 443a98e649b4 ("soundwire: bus: use pm_runtime_resume_and_get()") Change calls to pm_runtime_resume_and_get() back to pm_runtime_get_sync(). This fixes a usage count underrun caused by doing a pm_runtime_put() even though pm_runtime_resume_and_get() returned an error. The three affected functions ignore -EACCES error from trying to get pm_runtime, and carry on, including a put at the end of the function. But pm_runtime_resume_and_get() does not increment the usage count if it returns an error. So in the -EACCES case you must not call pm_runtime_put(). The documentation for pm_runtime_get_sync() says: "Consider using pm_runtime_resume_and_get() ... as this is likely to result in cleaner code." In this case I don't think it results in cleaner code because the pm_runtime_put() at the end of the function would have to be conditional on the return value from pm_runtime_resume_and_get() at the top of the function. pm_runtime_get_sync() doesn't have this problem because it always increments the count, so always needs a put. The code can just flow through and do the pm_runtime_put() unconditionally. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20230406134640.8582-1-rf@opensource.cirrus.com Signed-off-by: Vinod Koul <vkoul(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- drivers/soundwire/bus.c | 15 ++++++++++----- 1 file changed, 10 insertions(+), 5 deletions(-) diff --git a/drivers/soundwire/bus.c b/drivers/soundwire/bus.c index b6aca59c31300..7fd99e581a574 100644 --- a/drivers/soundwire/bus.c +++ b/drivers/soundwire/bus.c @@ -546,9 +546,11 @@ int sdw_nread(struct sdw_slave *slave, u32 addr, size_t count, u8 *val) { int ret; - ret = pm_runtime_resume_and_get(&slave->dev); - if (ret < 0 && ret != -EACCES) + ret = pm_runtime_get_sync(&slave->dev); + if (ret < 0 && ret != -EACCES) { + pm_runtime_put_noidle(&slave->dev); return ret; + } ret = sdw_nread_no_pm(slave, addr, count, val); @@ -570,9 +572,11 @@ int sdw_nwrite(struct sdw_slave *slave, u32 addr, size_t count, const u8 *val) { int ret; - ret = pm_runtime_resume_and_get(&slave->dev); - if (ret < 0 && ret != -EACCES) + ret = pm_runtime_get_sync(&slave->dev); + if (ret < 0 && ret != -EACCES) { + pm_runtime_put_noidle(&slave->dev); return ret; + } ret = sdw_nwrite_no_pm(slave, addr, count, val); @@ -1541,9 +1545,10 @@ static int sdw_handle_slave_alerts(struct sdw_slave *slave) sdw_modify_slave_status(slave, SDW_SLAVE_ALERT); - ret = pm_runtime_resume_and_get(&slave->dev); + ret = pm_runtime_get_sync(&slave->dev); if (ret < 0 && ret != -EACCES) { dev_err(&slave->dev, "Failed to resume device: %d\n", ret); + pm_runtime_put_noidle(&slave->dev); return ret; } -- 2.39.2

1 0

[PATCH AUTOSEL 6.3 4/9] soundwire: qcom: gracefully handle too many ports in DT
by Sasha Levin 10 May '23

10 May '23

From: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> [ Upstream commit 2367e0ecb498764e95cfda691ff0828f7d25f9a4 ] There are two issues related to the number of ports coming from Devicetree when exceeding in total QCOM_SDW_MAX_PORTS. Both lead to incorrect memory accesses: 1. With DTS having too big value of input or output ports, the driver, when copying port parameters from local/stack arrays into 'pconfig' array in 'struct qcom_swrm_ctrl', will iterate over their sizes. 2. If DTS also has too many parameters for these ports (e.g. qcom,ports-sinterval-low), the driver will overflow buffers on the stack when reading these properties from DTS. Add a sanity check so incorrect DTS will not cause kernel memory corruption. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Reviewed-by: Konrad Dybcio <konrad.dybcio(a)linaro.org> Link: https://lore.kernel.org/r/20230222144412.237832-2-krzysztof.kozlowski@linar… Signed-off-by: Vinod Koul <vkoul(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- drivers/soundwire/qcom.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c index 3354248702900..3e80a4d388cbb 100644 --- a/drivers/soundwire/qcom.c +++ b/drivers/soundwire/qcom.c @@ -1217,6 +1217,9 @@ static int qcom_swrm_get_port_config(struct qcom_swrm_ctrl *ctrl) ctrl->num_dout_ports = val; nports = ctrl->num_dout_ports + ctrl->num_din_ports; + if (nports > QCOM_SDW_MAX_PORTS) + return -EINVAL; + /* Valid port numbers are from 1-14, so mask out port 0 explicitly */ set_bit(0, &ctrl->dout_port_mask); set_bit(0, &ctrl->din_port_mask); -- 2.39.2

1 0

[PATCH AUTOSEL 6.3 2/9] soundwire: dmi-quirks: add remapping for Intel 'Rooks County' NUC M15
by Sasha Levin 10 May '23

10 May '23

From: Eugene Huang <eugene.huang99(a)gmail.com> [ Upstream commit 01b33e284ca28cc977bdcfb23be2c719f2139175 ] Same DSDT problem as the HP Omen 16-k0005TX, except rt1316 amp is on link2. Link: https://github.com/thesofproject/linux/issues/4088 Signed-off-by: Eugene Huang <eugene.huang99(a)gmail.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Link: https://lore.kernel.org/r/20230314090618.498716-1-yung-chuan.liao@linux.int… Signed-off-by: Vinod Koul <vkoul(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- drivers/soundwire/dmi-quirks.c | 25 +++++++++++++++++++++++++ 1 file changed, 25 insertions(+) diff --git a/drivers/soundwire/dmi-quirks.c b/drivers/soundwire/dmi-quirks.c index 7969881f126dc..58ea013fa918a 100644 --- a/drivers/soundwire/dmi-quirks.c +++ b/drivers/soundwire/dmi-quirks.c @@ -73,6 +73,23 @@ static const struct adr_remap hp_omen_16[] = { {} }; +/* + * Intel NUC M15 LAPRC510 and LAPRC710 + */ +static const struct adr_remap intel_rooks_county[] = { + /* rt711-sdca on link0 */ + { + 0x000020025d071100ull, + 0x000030025d071101ull + }, + /* rt1316-sdca on link2 */ + { + 0x000120025d071100ull, + 0x000230025d131601ull + }, + {} +}; + static const struct dmi_system_id adr_remap_quirk_table[] = { /* TGL devices */ { @@ -98,6 +115,14 @@ static const struct dmi_system_id adr_remap_quirk_table[] = { }, .driver_data = (void *)intel_tgl_bios, }, + { + /* quirk used for NUC15 'Rooks County' LAPRC510 and LAPRC710 skews */ + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Intel(R) Client Systems"), + DMI_MATCH(DMI_PRODUCT_NAME, "LAPRC"), + }, + .driver_data = (void *)intel_rooks_county, + }, { .matches = { DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), -- 2.39.2

1 0

Newest Update for Fedora 37 (alsa-lib-1.2.9-1.fc37) causes crackling with device
by GitHub issues - opened 10 May '23

10 May '23

alsa-project/alsa-lib issue #317 was opened from ClaraCrazy: First bug report here + not skilled with debugging, so please go easy on me :smile: OS: Fedora 37 Alsa-lib: 1.2.9-1 Issue: Audio crackling Audio Device: `Logic3 TX101 5.1 Soundbar` connected via `Scarlett 2i2 3rd Gen audio interface (USB)` Reverted lib back to `alsa-lib.1.2.8-2.fc37` (which in turn also downgraded alsa-ucm and alsa-utils again) and issues are fixed. I have no idea how to debug this, but would love to help in any way possible, sadly there was no issue template that would give me infos on what logs are useful / where to get those so uh.. please someone just point me in the right direction :) Thanks for your time Issue URL : https://github.com/alsa-project/alsa-lib/issues/317 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH v3] ALSA: pcm: auto-fill buffer with silence when draining playback
by Oswald Buddenhagen 10 May '23

10 May '23

Draining will always playback somewhat beyond the end of the filled buffer. This would produce artifacts if the user did not set up the auto-silencing machinery, which is an extremely easy mistake to make, as the API strongly suggests convenient fire-and-forget semantics. This patch makes it work out of the box. Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen(a)gmx.de> --- you are NOT expected to apply this. i just needed it for my testing (it's easier to deploy as i'm hacking on the kernel anyway) and wanted to post it for posterity. v3: - rebased to updated silencing code - intro period alignment to reduce redundant fill - cut down commit message again, as it's disabled by default now v2: - fill only up to two periods, to avoid undue load with big buffers - added discussion to commit message --- sound/core/pcm_lib.c | 35 ++++++++++++++++++++++++++++++++++- sound/core/pcm_native.c | 3 ++- 2 files changed, 36 insertions(+), 2 deletions(-) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 9c121a921b04..46e63e653789 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -67,6 +67,11 @@ void snd_pcm_playback_silence(struct snd_pcm_substream *substream, snd_pcm_ufram snd_pcm_uframes_t frames, ofs, transfer; int err; + if (runtime->silence_size == 0 && + (runtime->state != SNDRV_PCM_STATE_DRAINING || + (runtime->info & SNDRV_PCM_HW_PARAMS_NO_DRAIN_SILENCE) || + (runtime->hw.info & SNDRV_PCM_INFO_PERFECT_DRAIN))) + return; if (runtime->silence_size < runtime->boundary) { snd_pcm_sframes_t noise_dist; snd_pcm_uframes_t appl_ptr = READ_ONCE(runtime->control->appl_ptr); @@ -80,6 +85,33 @@ void snd_pcm_playback_silence(struct snd_pcm_substream *substream, snd_pcm_ufram noise_dist += runtime->boundary; /* total noise distance */ noise_dist += runtime->silence_filled; + if (runtime->state == SNDRV_PCM_STATE_DRAINING) { + snd_pcm_uframes_t slack = runtime->rate / 10; + snd_pcm_sframes_t threshold; + snd_pcm_uframes_t ps = runtime->period_size; + snd_pcm_uframes_t silence_size = ps; + // Round down to start of next period. This is disabled + // if the period count is not integer. + if (runtime->periods * ps == runtime->buffer_size) + silence_size = ps - (appl_ptr + ps - 1) % ps - 1; + // Add overshoot to accomodate FIFOs and IRQ delays. + // The default 1/10th secs is very generous. But more than one + // period doesn't make sense; the driver would set the minimum + // period size accordingly. + slack = min(slack, ps); + silence_size += slack; + // This catches the periods == 1 case. + silence_size = min(silence_size, runtime->buffer_size); + + threshold = ps + slack; + if (noise_dist >= threshold) + return; + frames = threshold - noise_dist; + if (frames > silence_size) + frames = silence_size; + + goto avoid_reindent; + } if (noise_dist >= (snd_pcm_sframes_t) runtime->silence_threshold) return; frames = runtime->silence_threshold - noise_dist; @@ -118,6 +150,7 @@ void snd_pcm_playback_silence(struct snd_pcm_substream *substream, snd_pcm_ufram */ frames = runtime->buffer_size - runtime->silence_filled; } +avoid_reindent: if (snd_BUG_ON(frames > runtime->buffer_size)) return; if (frames == 0) @@ -465,7 +498,7 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, } if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && - runtime->silence_size > 0) + (runtime->silence_size > 0 || runtime->state == SNDRV_PCM_STATE_DRAINING)) snd_pcm_playback_silence(substream, new_hw_ptr); if (in_interrupt) { diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 39a65d1415ab..913dae449ba0 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -1454,7 +1454,7 @@ static void snd_pcm_post_start(struct snd_pcm_substream *substream, runtime->rate; __snd_pcm_set_state(runtime, state); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && - runtime->silence_size > 0) + (runtime->silence_size > 0 || state == SNDRV_PCM_STATE_DRAINING)) snd_pcm_playback_silence(substream, ULONG_MAX); snd_pcm_timer_notify(substream, SNDRV_TIMER_EVENT_MSTART); } @@ -2045,6 +2045,7 @@ static int snd_pcm_do_drain_init(struct snd_pcm_substream *substream, break; case SNDRV_PCM_STATE_RUNNING: __snd_pcm_set_state(runtime, SNDRV_PCM_STATE_DRAINING); + snd_pcm_playback_silence(substream, ULONG_MAX); break; case SNDRV_PCM_STATE_XRUN: __snd_pcm_set_state(runtime, SNDRV_PCM_STATE_SETUP); -- 2.40.0.152.g15d061e6df

2 2

[PATCH AUTOSEL 5.10 1/3] ASOC: Intel: sof_sdw: add quirk for Intel 'Rooks County' NUC M15
by Sasha Levin 10 May '23

10 May '23

From: Eugene Huang <eugene.huang99(a)gmail.com> [ Upstream commit 3c728b1bc5b99c5275ac5c7788ef814c0e51ef54 ] Same quirks as the 'Bishop County' NUC M15, except the rt711 is in the 'JD2 100K' jack detection mode. Link: https://github.com/thesofproject/linux/issues/4088 Signed-off-by: Eugene Huang <eugene.huang99(a)gmail.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Link: https://lore.kernel.org/r/20230314090553.498664-2-yung-chuan.liao@linux.int… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/sof_sdw.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index f5d8f7951cfc3..eb713e9c2bd22 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -175,6 +175,17 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { SOF_SDW_PCH_DMIC | SOF_RT711_JD_SRC_JD2), }, + { + /* NUC15 'Rooks County' LAPRC510 and LAPRC710 skews */ + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Intel(R) Client Systems"), + DMI_MATCH(DMI_PRODUCT_NAME, "LAPRC"), + }, + .driver_data = (void *)(SOF_SDW_TGL_HDMI | + SOF_SDW_PCH_DMIC | + RT711_JD2_100K), + }, /* TigerLake-SDCA devices */ { .callback = sof_sdw_quirk_cb, -- 2.39.2

2 2

[PATCH] ALSA: PCM: Fix codestyle issues in pcm_native.c
by Ivan Orlov 10 May '23

10 May '23

Fix some obvious codestyle issues in the pcm_native.c file. Some of them were found by checkpatch with --strict option, and the others by just looking at the code. These issues include incorrect placement of brackets, trailing and starting spaces, wrong alignment of function arguments, incorrect multiline comments, using __attribute__((packed)) instead of just "__packed", comparison with null instead of using "!". Also, today we have 100 columns per line available, and I tried to reformat the file considering this. Signed-off-by: Ivan Orlov <ivan.orlov0322(a)gmail.com> --- sound/core/pcm_native.c | 696 +++++++++++++++++++--------------------- 1 file changed, 323 insertions(+), 373 deletions(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 39a65d1415ab..4a8d2e825b1d 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -43,10 +43,9 @@ struct snd_pcm_hw_params_old { unsigned int flags; - unsigned int masks[SNDRV_PCM_HW_PARAM_SUBFORMAT - - SNDRV_PCM_HW_PARAM_ACCESS + 1]; + unsigned int masks[SNDRV_PCM_HW_PARAM_SUBFORMAT - SNDRV_PCM_HW_PARAM_ACCESS + 1]; struct snd_interval intervals[SNDRV_PCM_HW_PARAM_TICK_TIME - - SNDRV_PCM_HW_PARAM_SAMPLE_BITS + 1]; + SNDRV_PCM_HW_PARAM_SAMPLE_BITS + 1]; unsigned int rmask; unsigned int cmask; unsigned int info; @@ -62,9 +61,9 @@ struct snd_pcm_hw_params_old { #define SNDRV_PCM_IOCTL_HW_PARAMS_OLD _IOWR('A', 0x11, struct snd_pcm_hw_params_old) static int snd_pcm_hw_refine_old_user(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params_old __user * _oparams); + struct snd_pcm_hw_params_old __user *_oparams); static int snd_pcm_hw_params_old_user(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params_old __user * _oparams); + struct snd_pcm_hw_params_old __user *_oparams); #endif static int snd_pcm_open(struct file *file, struct snd_pcm *pcm, int stream); @@ -133,8 +132,7 @@ EXPORT_SYMBOL_GPL(snd_pcm_stream_unlock); */ void snd_pcm_stream_lock_irq(struct snd_pcm_substream *substream) { - snd_pcm_group_lock_irq(&substream->self_group, - substream->pcm->nonatomic); + snd_pcm_group_lock_irq(&substream->self_group, substream->pcm->nonatomic); } EXPORT_SYMBOL_GPL(snd_pcm_stream_lock_irq); @@ -164,6 +162,7 @@ EXPORT_SYMBOL_GPL(snd_pcm_stream_unlock_irq); unsigned long _snd_pcm_stream_lock_irqsave(struct snd_pcm_substream *substream) { unsigned long flags = 0; + if (substream->pcm->nonatomic) mutex_lock(&substream->self_group.mutex); else @@ -175,12 +174,11 @@ EXPORT_SYMBOL_GPL(_snd_pcm_stream_lock_irqsave); unsigned long _snd_pcm_stream_lock_irqsave_nested(struct snd_pcm_substream *substream) { unsigned long flags = 0; + if (substream->pcm->nonatomic) - mutex_lock_nested(&substream->self_group.mutex, - SINGLE_DEPTH_NESTING); + mutex_lock_nested(&substream->self_group.mutex, SINGLE_DEPTH_NESTING); else - spin_lock_irqsave_nested(&substream->self_group.lock, flags, - SINGLE_DEPTH_NESTING); + spin_lock_irqsave_nested(&substream->self_group.lock, flags, SINGLE_DEPTH_NESTING); return flags; } EXPORT_SYMBOL_GPL(_snd_pcm_stream_lock_irqsave_nested); @@ -192,8 +190,7 @@ EXPORT_SYMBOL_GPL(_snd_pcm_stream_lock_irqsave_nested); * * This is a counter-part of snd_pcm_stream_lock_irqsave(). */ -void snd_pcm_stream_unlock_irqrestore(struct snd_pcm_substream *substream, - unsigned long flags) +void snd_pcm_stream_unlock_irqrestore(struct snd_pcm_substream *substream, unsigned long flags) { if (substream->pcm->nonatomic) mutex_unlock(&substream->self_group.mutex); @@ -203,8 +200,7 @@ void snd_pcm_stream_unlock_irqrestore(struct snd_pcm_substream *substream, EXPORT_SYMBOL_GPL(snd_pcm_stream_unlock_irqrestore); /* Run PCM ioctl ops */ -static int snd_pcm_ops_ioctl(struct snd_pcm_substream *substream, - unsigned cmd, void *arg) +static int snd_pcm_ops_ioctl(struct snd_pcm_substream *substream, unsigned int cmd, void *arg) { if (substream->ops->ioctl) return substream->ops->ioctl(substream, cmd, arg); @@ -233,14 +229,13 @@ int snd_pcm_info(struct snd_pcm_substream *substream, struct snd_pcm_info *info) return 0; } -int snd_pcm_info_user(struct snd_pcm_substream *substream, - struct snd_pcm_info __user * _info) +int snd_pcm_info_user(struct snd_pcm_substream *substream, struct snd_pcm_info __user *_info) { struct snd_pcm_info *info; int err; info = kmalloc(sizeof(*info), GFP_KERNEL); - if (! info) + if (!info) return -ENOMEM; err = snd_pcm_info(substream, info); if (err >= 0) { @@ -284,8 +279,7 @@ static bool hw_support_mmap(struct snd_pcm_substream *substream) static int constrain_mask_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { - struct snd_pcm_hw_constraints *constrs = - &substream->runtime->hw_constraints; + struct snd_pcm_hw_constraints *constrs = &substream->runtime->hw_constraints; struct snd_mask *m; unsigned int k; struct snd_mask old_mask __maybe_unused; @@ -320,8 +314,7 @@ static int constrain_mask_params(struct snd_pcm_substream *substream, static int constrain_interval_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { - struct snd_pcm_hw_constraints *constrs = - &substream->runtime->hw_constraints; + struct snd_pcm_hw_constraints *constrs = &substream->runtime->hw_constraints; struct snd_interval *i; unsigned int k; struct snd_interval old_interval __maybe_unused; @@ -356,8 +349,7 @@ static int constrain_interval_params(struct snd_pcm_substream *substream, static int constrain_params_by_rules(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { - struct snd_pcm_hw_constraints *constrs = - &substream->runtime->hw_constraints; + struct snd_pcm_hw_constraints *constrs = &substream->runtime->hw_constraints; unsigned int k; unsigned int *rstamps; unsigned int vstamps[SNDRV_PCM_HW_PARAM_LAST_INTERVAL + 1]; @@ -447,14 +439,12 @@ static int constrain_params_by_rules(struct snd_pcm_substream *substream, */ if (changed && r->var >= 0) { if (hw_is_mask(r->var)) { - trace_hw_mask_param(substream, r->var, - k + 1, &old_mask, - hw_param_mask(params, r->var)); + trace_hw_mask_param(substream, r->var, k + 1, &old_mask, + hw_param_mask(params, r->var)); } if (hw_is_interval(r->var)) { - trace_hw_interval_param(substream, r->var, - k + 1, &old_interval, - hw_param_interval(params, r->var)); + trace_hw_interval_param(substream, r->var, k + 1, &old_interval, + hw_param_interval(params, r->var)); } params->cmask |= PARAM_MASK_BIT(r->var); @@ -509,18 +499,15 @@ static int fixup_unreferenced_params(struct snd_pcm_substream *substream, if (!params->info) { params->info = substream->runtime->hw.info; - params->info &= ~(SNDRV_PCM_INFO_FIFO_IN_FRAMES | - SNDRV_PCM_INFO_DRAIN_TRIGGER); + params->info &= ~(SNDRV_PCM_INFO_FIFO_IN_FRAMES | SNDRV_PCM_INFO_DRAIN_TRIGGER); if (!hw_support_mmap(substream)) - params->info &= ~(SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID); + params->info &= ~(SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID); } return 0; } -int snd_pcm_hw_refine(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) +int snd_pcm_hw_refine(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { int err; @@ -552,7 +539,7 @@ int snd_pcm_hw_refine(struct snd_pcm_substream *substream, EXPORT_SYMBOL(snd_pcm_hw_refine); static int snd_pcm_hw_refine_user(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params __user * _params) + struct snd_pcm_hw_params __user *_params) { struct snd_pcm_hw_params *params; int err; @@ -580,7 +567,7 @@ static int period_to_usecs(struct snd_pcm_runtime *runtime) { int usecs; - if (! runtime->rate) + if (!runtime->rate) return -1; /* invalid */ /* take 75% of period time as the deadline */ @@ -591,8 +578,7 @@ static int period_to_usecs(struct snd_pcm_runtime *runtime) return usecs; } -static void snd_pcm_set_state(struct snd_pcm_substream *substream, - snd_pcm_state_t state) +static void snd_pcm_set_state(struct snd_pcm_substream *substream, snd_pcm_state_t state) { snd_pcm_stream_lock_irq(substream); if (substream->runtime->state != SNDRV_PCM_STATE_DISCONNECTED) @@ -600,13 +586,11 @@ static void snd_pcm_set_state(struct snd_pcm_substream *substream, snd_pcm_stream_unlock_irq(substream); } -static inline void snd_pcm_timer_notify(struct snd_pcm_substream *substream, - int event) +static inline void snd_pcm_timer_notify(struct snd_pcm_substream *substream, int event) { #ifdef CONFIG_SND_PCM_TIMER if (substream->timer) - snd_timer_notify(substream->timer, event, - &substream->runtime->trigger_tstamp); + snd_timer_notify(substream->timer, event, &substream->runtime->trigger_tstamp); #endif } @@ -672,10 +656,8 @@ static int snd_pcm_hw_params_choose(struct snd_pcm_substream *pcm, continue; /* Trace the changed parameter. */ - if (hw_is_mask(*v)) { - trace_hw_mask_param(pcm, *v, 0, &old_mask, - hw_param_mask(params, *v)); - } + if (hw_is_mask(*v)) + trace_hw_mask_param(pcm, *v, 0, &old_mask, hw_param_mask(params, *v)); if (hw_is_interval(*v)) { trace_hw_interval_param(pcm, *v, 0, &old_interval, hw_param_interval(params, *v)); @@ -756,14 +738,13 @@ static int snd_pcm_hw_params(struct snd_pcm_substream *substream, goto _error; if (substream->managed_buffer_alloc) { - err = snd_pcm_lib_malloc_pages(substream, - params_buffer_bytes(params)); + err = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params)); if (err < 0) goto _error; runtime->buffer_changed = err > 0; } - if (substream->ops->hw_params != NULL) { + if (substream->ops->hw_params) { err = substream->ops->hw_params(substream, params); if (err < 0) goto _error; @@ -780,9 +761,8 @@ static int snd_pcm_hw_params(struct snd_pcm_substream *substream, runtime->info = params->info; runtime->rate_num = params->rate_num; runtime->rate_den = params->rate_den; - runtime->no_period_wakeup = - (params->info & SNDRV_PCM_INFO_NO_PERIOD_WAKEUP) && - (params->flags & SNDRV_PCM_HW_PARAMS_NO_PERIOD_WAKEUP); + runtime->no_period_wakeup = (params->info & SNDRV_PCM_INFO_NO_PERIOD_WAKEUP) && + (params->flags & SNDRV_PCM_HW_PARAMS_NO_PERIOD_WAKEUP); bits = snd_pcm_format_physical_width(runtime->format); runtime->sample_bits = bits; @@ -824,8 +804,7 @@ static int snd_pcm_hw_params(struct snd_pcm_substream *substream, cpu_latency_qos_remove_request(&substream->latency_pm_qos_req); usecs = period_to_usecs(runtime); if (usecs >= 0) - cpu_latency_qos_add_request(&substream->latency_pm_qos_req, - usecs); + cpu_latency_qos_add_request(&substream->latency_pm_qos_req, usecs); err = 0; _error: if (err) { @@ -834,7 +813,7 @@ static int snd_pcm_hw_params(struct snd_pcm_substream *substream, * the correct hardware parameter settings */ snd_pcm_set_state(substream, SNDRV_PCM_STATE_OPEN); - if (substream->ops->hw_free != NULL) + if (substream->ops->hw_free) substream->ops->hw_free(substream); if (substream->managed_buffer_alloc) snd_pcm_lib_free_pages(substream); @@ -845,7 +824,7 @@ static int snd_pcm_hw_params(struct snd_pcm_substream *substream, } static int snd_pcm_hw_params_user(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params __user * _params) + struct snd_pcm_hw_params __user *_params) { struct snd_pcm_hw_params *params; int err; @@ -954,10 +933,9 @@ static int snd_pcm_sw_params(struct snd_pcm_substream *substream, runtime->stop_threshold = params->stop_threshold; runtime->silence_threshold = params->silence_threshold; runtime->silence_size = params->silence_size; - params->boundary = runtime->boundary; + params->boundary = runtime->boundary; if (snd_pcm_running(substream)) { - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && - runtime->silence_size > 0) + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && runtime->silence_size > 0) snd_pcm_playback_silence(substream, ULONG_MAX); err = snd_pcm_update_state(substream, runtime); } @@ -966,10 +944,11 @@ static int snd_pcm_sw_params(struct snd_pcm_substream *substream, } static int snd_pcm_sw_params_user(struct snd_pcm_substream *substream, - struct snd_pcm_sw_params __user * _params) + struct snd_pcm_sw_params __user *_params) { struct snd_pcm_sw_params params; int err; + if (copy_from_user(&params, _params, sizeof(params))) return -EFAULT; err = snd_pcm_sw_params(substream, &params); @@ -978,8 +957,7 @@ static int snd_pcm_sw_params_user(struct snd_pcm_substream *substream, return err; } -static inline snd_pcm_uframes_t -snd_pcm_calc_delay(struct snd_pcm_substream *substream) +static inline snd_pcm_uframes_t snd_pcm_calc_delay(struct snd_pcm_substream *substream) { snd_pcm_uframes_t delay; @@ -990,15 +968,14 @@ snd_pcm_calc_delay(struct snd_pcm_substream *substream) return delay + substream->runtime->delay; } -int snd_pcm_status64(struct snd_pcm_substream *substream, - struct snd_pcm_status64 *status) +int snd_pcm_status64(struct snd_pcm_substream *substream, struct snd_pcm_status64 *status) { struct snd_pcm_runtime *runtime = substream->runtime; snd_pcm_stream_lock_irq(substream); snd_pcm_unpack_audio_tstamp_config(status->audio_tstamp_data, - &runtime->audio_tstamp_config); + &runtime->audio_tstamp_config); /* backwards compatible behavior */ if (runtime->audio_tstamp_config.type_requested == @@ -1010,8 +987,9 @@ int snd_pcm_status64(struct snd_pcm_substream *substream, runtime->audio_tstamp_config.type_requested = SNDRV_PCM_AUDIO_TSTAMP_TYPE_DEFAULT; runtime->audio_tstamp_report.valid = 0; - } else + } else { runtime->audio_tstamp_report.valid = 1; + } status->state = runtime->state; status->suspended_state = runtime->suspended_state; @@ -1036,8 +1014,8 @@ int snd_pcm_status64(struct snd_pcm_substream *substream, if (runtime->audio_tstamp_report.valid == 1) /* backwards compatibility, no report provided in COMPAT mode */ snd_pcm_pack_audio_tstamp_report(&status->audio_tstamp_data, - &status->audio_tstamp_accuracy, - &runtime->audio_tstamp_report); + &status->audio_tstamp_accuracy, + &runtime->audio_tstamp_report); goto _tstamp_end; } @@ -1055,19 +1033,18 @@ int snd_pcm_status64(struct snd_pcm_substream *substream, status->appl_ptr = runtime->control->appl_ptr; status->hw_ptr = runtime->status->hw_ptr; status->avail = snd_pcm_avail(substream); - status->delay = snd_pcm_running(substream) ? - snd_pcm_calc_delay(substream) : 0; + status->delay = snd_pcm_running(substream) ? snd_pcm_calc_delay(substream) : 0; status->avail_max = runtime->avail_max; status->overrange = runtime->overrange; runtime->avail_max = 0; runtime->overrange = 0; - _end: - snd_pcm_stream_unlock_irq(substream); +_end: + snd_pcm_stream_unlock_irq(substream); return 0; } static int snd_pcm_status_user64(struct snd_pcm_substream *substream, - struct snd_pcm_status64 __user * _status, + struct snd_pcm_status64 __user *_status, bool ext) { struct snd_pcm_status64 status; @@ -1079,8 +1056,7 @@ static int snd_pcm_status_user64(struct snd_pcm_substream *substream, * get audio_tstamp_data from user, * ignore rest of status structure */ - if (ext && get_user(status.audio_tstamp_data, - (u32 __user *)(&_status->audio_tstamp_data))) + if (ext && get_user(status.audio_tstamp_data, (u32 __user *)&_status->audio_tstamp_data)) return -EFAULT; res = snd_pcm_status64(substream, &status); if (res < 0) @@ -1091,7 +1067,7 @@ static int snd_pcm_status_user64(struct snd_pcm_substream *substream, } static int snd_pcm_status_user32(struct snd_pcm_substream *substream, - struct snd_pcm_status32 __user * _status, + struct snd_pcm_status32 __user *_status, bool ext) { struct snd_pcm_status64 status64; @@ -1105,8 +1081,7 @@ static int snd_pcm_status_user32(struct snd_pcm_substream *substream, * get audio_tstamp_data from user, * ignore rest of status structure */ - if (ext && get_user(status64.audio_tstamp_data, - (u32 __user *)(&_status->audio_tstamp_data))) + if (ext && get_user(status64.audio_tstamp_data, (u32 __user *)&_status->audio_tstamp_data)) return -EFAULT; res = snd_pcm_status64(substream, &status64); if (res < 0) @@ -1140,11 +1115,11 @@ static int snd_pcm_status_user32(struct snd_pcm_substream *substream, } static int snd_pcm_channel_info(struct snd_pcm_substream *substream, - struct snd_pcm_channel_info * info) + struct snd_pcm_channel_info *info) { struct snd_pcm_runtime *runtime; unsigned int channel; - + channel = info->channel; runtime = substream->runtime; snd_pcm_stream_lock_irq(substream); @@ -1161,11 +1136,11 @@ static int snd_pcm_channel_info(struct snd_pcm_substream *substream, } static int snd_pcm_channel_info_user(struct snd_pcm_substream *substream, - struct snd_pcm_channel_info __user * _info) + struct snd_pcm_channel_info __user *_info) { struct snd_pcm_channel_info info; int res; - + if (copy_from_user(&info, _info, sizeof(info))) return -EFAULT; res = snd_pcm_channel_info(substream, &info); @@ -1179,7 +1154,8 @@ static int snd_pcm_channel_info_user(struct snd_pcm_substream *substream, static void snd_pcm_trigger_tstamp(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; - if (runtime->trigger_master == NULL) + + if (!runtime->trigger_master) return; if (runtime->trigger_master == substream) { if (!runtime->trigger_tstamp_latched) @@ -1191,17 +1167,13 @@ static void snd_pcm_trigger_tstamp(struct snd_pcm_substream *substream) runtime->trigger_master = NULL; } -#define ACTION_ARG_IGNORE (__force snd_pcm_state_t)0 +#define ACTION_ARG_IGNORE ((__force snd_pcm_state_t)0) struct action_ops { - int (*pre_action)(struct snd_pcm_substream *substream, - snd_pcm_state_t state); - int (*do_action)(struct snd_pcm_substream *substream, - snd_pcm_state_t state); - void (*undo_action)(struct snd_pcm_substream *substream, - snd_pcm_state_t state); - void (*post_action)(struct snd_pcm_substream *substream, - snd_pcm_state_t state); + int (*pre_action)(struct snd_pcm_substream *substream, snd_pcm_state_t state); + int (*do_action)(struct snd_pcm_substream *substream, snd_pcm_state_t state); + void (*undo_action)(struct snd_pcm_substream *substream, snd_pcm_state_t state); + void (*post_action)(struct snd_pcm_substream *substream, snd_pcm_state_t state); }; /* @@ -1209,10 +1181,8 @@ struct action_ops { * Note: the stream state might be changed also on failure * Note2: call with calling stream lock + link lock */ -static int snd_pcm_action_group(const struct action_ops *ops, - struct snd_pcm_substream *substream, - snd_pcm_state_t state, - bool stream_lock) +static int snd_pcm_action_group(const struct action_ops *ops, struct snd_pcm_substream *substream, + snd_pcm_state_t state, bool stream_lock) { struct snd_pcm_substream *s = NULL; struct snd_pcm_substream *s1; @@ -1274,7 +1244,7 @@ static int snd_pcm_action_single(const struct action_ops *ops, snd_pcm_state_t state) { int res; - + res = ops->pre_action(substream, state); if (res < 0) return res; @@ -1297,8 +1267,7 @@ static void snd_pcm_group_assign(struct snd_pcm_substream *substream, * Unref and unlock the group, but keep the stream lock; * when the group becomes empty and no longer referred, destroy itself */ -static void snd_pcm_group_unref(struct snd_pcm_group *group, - struct snd_pcm_substream *substream) +static void snd_pcm_group_unref(struct snd_pcm_group *group, struct snd_pcm_substream *substream) { bool do_free; @@ -1314,8 +1283,7 @@ static void snd_pcm_group_unref(struct snd_pcm_group *group, * Lock the group inside a stream lock and reference it; * return the locked group object, or NULL if not linked */ -static struct snd_pcm_group * -snd_pcm_stream_group_ref(struct snd_pcm_substream *substream) +static struct snd_pcm_group *snd_pcm_stream_group_ref(struct snd_pcm_substream *substream) { bool nonatomic = substream->pcm->nonatomic; struct snd_pcm_group *group; @@ -1328,8 +1296,7 @@ snd_pcm_stream_group_ref(struct snd_pcm_substream *substream) /* block freeing the group object */ refcount_inc(&group->refs); - trylock = nonatomic ? mutex_trylock(&group->mutex) : - spin_trylock(&group->lock); + trylock = nonatomic ? mutex_trylock(&group->mutex) : spin_trylock(&group->lock); if (trylock) break; /* OK */ @@ -1411,10 +1378,10 @@ static int snd_pcm_pre_start(struct snd_pcm_substream *substream, snd_pcm_state_t state) { struct snd_pcm_runtime *runtime = substream->runtime; + if (runtime->state != SNDRV_PCM_STATE_PREPARED) return -EBADFD; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && - !snd_pcm_playback_data(substream)) + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && !snd_pcm_playback_data(substream)) return -EPIPE; runtime->trigger_tstamp_latched = false; runtime->trigger_master = substream; @@ -1448,13 +1415,12 @@ static void snd_pcm_post_start(struct snd_pcm_substream *substream, snd_pcm_state_t state) { struct snd_pcm_runtime *runtime = substream->runtime; + snd_pcm_trigger_tstamp(substream); runtime->hw_ptr_jiffies = jiffies; - runtime->hw_ptr_buffer_jiffies = (runtime->buffer_size * HZ) / - runtime->rate; + runtime->hw_ptr_buffer_jiffies = (runtime->buffer_size * HZ) / runtime->rate; __snd_pcm_set_state(runtime, state); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && - runtime->silence_size > 0) + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && runtime->silence_size > 0) snd_pcm_playback_silence(substream, ULONG_MAX); snd_pcm_timer_notify(substream, SNDRV_TIMER_EVENT_MSTART); } @@ -1475,15 +1441,13 @@ static const struct action_ops snd_pcm_action_start = { */ int snd_pcm_start(struct snd_pcm_substream *substream) { - return snd_pcm_action(&snd_pcm_action_start, substream, - SNDRV_PCM_STATE_RUNNING); + return snd_pcm_action(&snd_pcm_action_start, substream, SNDRV_PCM_STATE_RUNNING); } /* take the stream lock and start the streams */ static int snd_pcm_start_lock_irq(struct snd_pcm_substream *substream) { - return snd_pcm_action_lock_irq(&snd_pcm_action_start, substream, - SNDRV_PCM_STATE_RUNNING); + return snd_pcm_action_lock_irq(&snd_pcm_action_start, substream, SNDRV_PCM_STATE_RUNNING); } /* @@ -1493,6 +1457,7 @@ static int snd_pcm_pre_stop(struct snd_pcm_substream *substream, snd_pcm_state_t state) { struct snd_pcm_runtime *runtime = substream->runtime; + if (runtime->state == SNDRV_PCM_STATE_OPEN) return -EBADFD; runtime->trigger_master = substream; @@ -1502,8 +1467,7 @@ static int snd_pcm_pre_stop(struct snd_pcm_substream *substream, static int snd_pcm_do_stop(struct snd_pcm_substream *substream, snd_pcm_state_t state) { - if (substream->runtime->trigger_master == substream && - snd_pcm_running(substream)) { + if (substream->runtime->trigger_master == substream && snd_pcm_running(substream)) { substream->ops->trigger(substream, SNDRV_PCM_TRIGGER_STOP); substream->runtime->stop_operating = true; } @@ -1514,6 +1478,7 @@ static void snd_pcm_post_stop(struct snd_pcm_substream *substream, snd_pcm_state_t state) { struct snd_pcm_runtime *runtime = substream->runtime; + if (runtime->state != state) { snd_pcm_trigger_tstamp(substream); __snd_pcm_set_state(runtime, state); @@ -1555,8 +1520,7 @@ EXPORT_SYMBOL(snd_pcm_stop); */ int snd_pcm_drain_done(struct snd_pcm_substream *substream) { - return snd_pcm_action_single(&snd_pcm_action_stop, substream, - SNDRV_PCM_STATE_SETUP); + return snd_pcm_action_single(&snd_pcm_action_stop, substream, SNDRV_PCM_STATE_SETUP); } /** @@ -1583,19 +1547,20 @@ EXPORT_SYMBOL_GPL(snd_pcm_stop_xrun); /* * pause callbacks: pass boolean (to start pause or resume) as state argument */ -#define pause_pushed(state) (__force bool)(state) +#define pause_pushed(state) ((__force bool)(state)) -static int snd_pcm_pre_pause(struct snd_pcm_substream *substream, - snd_pcm_state_t state) +static int snd_pcm_pre_pause(struct snd_pcm_substream *substream, snd_pcm_state_t state) { struct snd_pcm_runtime *runtime = substream->runtime; + if (!(runtime->info & SNDRV_PCM_INFO_PAUSE)) return -ENOSYS; if (pause_pushed(state)) { if (runtime->state != SNDRV_PCM_STATE_RUNNING) return -EBADFD; - } else if (runtime->state != SNDRV_PCM_STATE_PAUSED) + } else if (runtime->state != SNDRV_PCM_STATE_PAUSED) { return -EBADFD; + } runtime->trigger_master = substream; return 0; } @@ -1605,8 +1570,7 @@ static int snd_pcm_do_pause(struct snd_pcm_substream *substream, { if (substream->runtime->trigger_master != substream) return 0; - /* some drivers might use hw_ptr to recover from the pause - - update the hw_ptr now */ + // some drivers might use hw_ptr to recover from the pause - update the hw_ptr now if (pause_pushed(state)) snd_pcm_update_hw_ptr(substream); /* The jiffies check in snd_pcm_update_hw_ptr*() is done by @@ -1614,26 +1578,23 @@ static int snd_pcm_do_pause(struct snd_pcm_substream *substream, * delta, effectively to skip the check once. */ substream->runtime->hw_ptr_jiffies = jiffies - HZ * 1000; - return substream->ops->trigger(substream, - pause_pushed(state) ? + return substream->ops->trigger(substream, pause_pushed(state) ? SNDRV_PCM_TRIGGER_PAUSE_PUSH : SNDRV_PCM_TRIGGER_PAUSE_RELEASE); } -static void snd_pcm_undo_pause(struct snd_pcm_substream *substream, - snd_pcm_state_t state) +static void snd_pcm_undo_pause(struct snd_pcm_substream *substream, snd_pcm_state_t state) { if (substream->runtime->trigger_master == substream) - substream->ops->trigger(substream, - pause_pushed(state) ? + substream->ops->trigger(substream, pause_pushed(state) ? SNDRV_PCM_TRIGGER_PAUSE_RELEASE : SNDRV_PCM_TRIGGER_PAUSE_PUSH); } -static void snd_pcm_post_pause(struct snd_pcm_substream *substream, - snd_pcm_state_t state) +static void snd_pcm_post_pause(struct snd_pcm_substream *substream, snd_pcm_state_t state) { struct snd_pcm_runtime *runtime = substream->runtime; + snd_pcm_trigger_tstamp(substream); if (pause_pushed(state)) { __snd_pcm_set_state(runtime, SNDRV_PCM_STATE_PAUSED); @@ -1658,24 +1619,23 @@ static const struct action_ops snd_pcm_action_pause = { */ static int snd_pcm_pause(struct snd_pcm_substream *substream, bool push) { - return snd_pcm_action(&snd_pcm_action_pause, substream, - (__force snd_pcm_state_t)push); + return snd_pcm_action(&snd_pcm_action_pause, substream, (__force snd_pcm_state_t)push); } -static int snd_pcm_pause_lock_irq(struct snd_pcm_substream *substream, - bool push) +static int snd_pcm_pause_lock_irq(struct snd_pcm_substream *substream, bool push) { return snd_pcm_action_lock_irq(&snd_pcm_action_pause, substream, (__force snd_pcm_state_t)push); } #ifdef CONFIG_PM + /* suspend callback: state argument ignored */ -static int snd_pcm_pre_suspend(struct snd_pcm_substream *substream, - snd_pcm_state_t state) +static int snd_pcm_pre_suspend(struct snd_pcm_substream *substream, snd_pcm_state_t state) { struct snd_pcm_runtime *runtime = substream->runtime; + switch (runtime->state) { case SNDRV_PCM_STATE_SUSPENDED: return -EBUSY; @@ -1689,23 +1649,23 @@ static int snd_pcm_pre_suspend(struct snd_pcm_substream *substream, return 0; } -static int snd_pcm_do_suspend(struct snd_pcm_substream *substream, - snd_pcm_state_t state) +static int snd_pcm_do_suspend(struct snd_pcm_substream *substream, snd_pcm_state_t state) { struct snd_pcm_runtime *runtime = substream->runtime; + if (runtime->trigger_master != substream) return 0; - if (! snd_pcm_running(substream)) + if (!snd_pcm_running(substream)) return 0; substream->ops->trigger(substream, SNDRV_PCM_TRIGGER_SUSPEND); runtime->stop_operating = true; return 0; /* suspend unconditionally */ } -static void snd_pcm_post_suspend(struct snd_pcm_substream *substream, - snd_pcm_state_t state) +static void snd_pcm_post_suspend(struct snd_pcm_substream *substream, snd_pcm_state_t state) { struct snd_pcm_runtime *runtime = substream->runtime; + snd_pcm_trigger_tstamp(substream); runtime->suspended_state = runtime->state; runtime->status->suspended_state = runtime->suspended_state; @@ -1735,8 +1695,7 @@ static int snd_pcm_suspend(struct snd_pcm_substream *substream) unsigned long flags; snd_pcm_stream_lock_irqsave(substream, flags); - err = snd_pcm_action(&snd_pcm_action_suspend, substream, - ACTION_ARG_IGNORE); + err = snd_pcm_action(&snd_pcm_action_suspend, substream, ACTION_ARG_IGNORE); snd_pcm_stream_unlock_irqrestore(substream, flags); return err; } @@ -1754,7 +1713,7 @@ int snd_pcm_suspend_all(struct snd_pcm *pcm) struct snd_pcm_substream *substream; int stream, err = 0; - if (! pcm) + if (!pcm) return 0; for_each_pcm_substream(pcm, stream, substream) { @@ -1783,20 +1742,20 @@ EXPORT_SYMBOL(snd_pcm_suspend_all); /* resume callbacks: state argument ignored */ -static int snd_pcm_pre_resume(struct snd_pcm_substream *substream, - snd_pcm_state_t state) +static int snd_pcm_pre_resume(struct snd_pcm_substream *substream, snd_pcm_state_t state) { struct snd_pcm_runtime *runtime = substream->runtime; + if (!(runtime->info & SNDRV_PCM_INFO_RESUME)) return -ENOSYS; runtime->trigger_master = substream; return 0; } -static int snd_pcm_do_resume(struct snd_pcm_substream *substream, - snd_pcm_state_t state) +static int snd_pcm_do_resume(struct snd_pcm_substream *substream, snd_pcm_state_t state) { struct snd_pcm_runtime *runtime = substream->runtime; + if (runtime->trigger_master != substream) return 0; /* DMA not running previously? */ @@ -1807,18 +1766,16 @@ static int snd_pcm_do_resume(struct snd_pcm_substream *substream, return substream->ops->trigger(substream, SNDRV_PCM_TRIGGER_RESUME); } -static void snd_pcm_undo_resume(struct snd_pcm_substream *substream, - snd_pcm_state_t state) +static void snd_pcm_undo_resume(struct snd_pcm_substream *substream, snd_pcm_state_t state) { - if (substream->runtime->trigger_master == substream && - snd_pcm_running(substream)) + if (substream->runtime->trigger_master == substream && snd_pcm_running(substream)) substream->ops->trigger(substream, SNDRV_PCM_TRIGGER_SUSPEND); } -static void snd_pcm_post_resume(struct snd_pcm_substream *substream, - snd_pcm_state_t state) +static void snd_pcm_post_resume(struct snd_pcm_substream *substream, snd_pcm_state_t state) { struct snd_pcm_runtime *runtime = substream->runtime; + snd_pcm_trigger_tstamp(substream); __snd_pcm_set_state(runtime, runtime->suspended_state); snd_pcm_timer_notify(substream, SNDRV_TIMER_EVENT_MRESUME); @@ -1833,8 +1790,7 @@ static const struct action_ops snd_pcm_action_resume = { static int snd_pcm_resume(struct snd_pcm_substream *substream) { - return snd_pcm_action_lock_irq(&snd_pcm_action_resume, substream, - ACTION_ARG_IGNORE); + return snd_pcm_action_lock_irq(&snd_pcm_action_resume, substream, ACTION_ARG_IGNORE); } #else @@ -1876,10 +1832,10 @@ static int snd_pcm_xrun(struct snd_pcm_substream *substream) * reset ioctl */ /* reset callbacks: state argument ignored */ -static int snd_pcm_pre_reset(struct snd_pcm_substream *substream, - snd_pcm_state_t state) +static int snd_pcm_pre_reset(struct snd_pcm_substream *substream, snd_pcm_state_t state) { struct snd_pcm_runtime *runtime = substream->runtime; + switch (runtime->state) { case SNDRV_PCM_STATE_RUNNING: case SNDRV_PCM_STATE_PREPARED: @@ -1891,11 +1847,12 @@ static int snd_pcm_pre_reset(struct snd_pcm_substream *substream, } } -static int snd_pcm_do_reset(struct snd_pcm_substream *substream, - snd_pcm_state_t state) +static int snd_pcm_do_reset(struct snd_pcm_substream *substream, snd_pcm_state_t state) { struct snd_pcm_runtime *runtime = substream->runtime; - int err = snd_pcm_ops_ioctl(substream, SNDRV_PCM_IOCTL1_RESET, NULL); + int err; + + err = snd_pcm_ops_ioctl(substream, SNDRV_PCM_IOCTL1_RESET, NULL); if (err < 0) return err; snd_pcm_stream_lock_irq(substream); @@ -1908,14 +1865,13 @@ static int snd_pcm_do_reset(struct snd_pcm_substream *substream, return 0; } -static void snd_pcm_post_reset(struct snd_pcm_substream *substream, - snd_pcm_state_t state) +static void snd_pcm_post_reset(struct snd_pcm_substream *substream, snd_pcm_state_t state) { struct snd_pcm_runtime *runtime = substream->runtime; + snd_pcm_stream_lock_irq(substream); runtime->control->appl_ptr = runtime->status->hw_ptr; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && - runtime->silence_size > 0) + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && runtime->silence_size > 0) snd_pcm_playback_silence(substream, ULONG_MAX); snd_pcm_stream_unlock_irq(substream); } @@ -1928,16 +1884,14 @@ static const struct action_ops snd_pcm_action_reset = { static int snd_pcm_reset(struct snd_pcm_substream *substream) { - return snd_pcm_action_nonatomic(&snd_pcm_action_reset, substream, - ACTION_ARG_IGNORE); + return snd_pcm_action_nonatomic(&snd_pcm_action_reset, substream, ACTION_ARG_IGNORE); } /* * prepare ioctl */ /* pass f_flags as state argument */ -static int snd_pcm_pre_prepare(struct snd_pcm_substream *substream, - snd_pcm_state_t state) +static int snd_pcm_pre_prepare(struct snd_pcm_substream *substream, snd_pcm_state_t state) { struct snd_pcm_runtime *runtime = substream->runtime; int f_flags = (__force int)state; @@ -1951,10 +1905,10 @@ static int snd_pcm_pre_prepare(struct snd_pcm_substream *substream, return 0; } -static int snd_pcm_do_prepare(struct snd_pcm_substream *substream, - snd_pcm_state_t state) +static int snd_pcm_do_prepare(struct snd_pcm_substream *substream, snd_pcm_state_t state) { int err; + snd_pcm_sync_stop(substream, true); err = substream->ops->prepare(substream); if (err < 0) @@ -1962,10 +1916,10 @@ static int snd_pcm_do_prepare(struct snd_pcm_substream *substream, return snd_pcm_do_reset(substream, state); } -static void snd_pcm_post_prepare(struct snd_pcm_substream *substream, - snd_pcm_state_t state) +static void snd_pcm_post_prepare(struct snd_pcm_substream *substream, snd_pcm_state_t state) { struct snd_pcm_runtime *runtime = substream->runtime; + runtime->control->appl_ptr = runtime->status->hw_ptr; snd_pcm_set_state(substream, SNDRV_PCM_STATE_PREPARED); } @@ -1983,8 +1937,7 @@ static const struct action_ops snd_pcm_action_prepare = { * * Return: Zero if successful, or a negative error code. */ -static int snd_pcm_prepare(struct snd_pcm_substream *substream, - struct file *file) +static int snd_pcm_prepare(struct snd_pcm_substream *substream, struct file *file) { int f_flags; @@ -2004,8 +1957,7 @@ static int snd_pcm_prepare(struct snd_pcm_substream *substream, } snd_pcm_stream_unlock_irq(substream); - return snd_pcm_action_nonatomic(&snd_pcm_action_prepare, - substream, + return snd_pcm_action_nonatomic(&snd_pcm_action_prepare, substream, (__force snd_pcm_state_t)f_flags); } @@ -2014,10 +1966,10 @@ static int snd_pcm_prepare(struct snd_pcm_substream *substream, */ /* drain init callbacks: state argument ignored */ -static int snd_pcm_pre_drain_init(struct snd_pcm_substream *substream, - snd_pcm_state_t state) +static int snd_pcm_pre_drain_init(struct snd_pcm_substream *substream, snd_pcm_state_t state) { struct snd_pcm_runtime *runtime = substream->runtime; + switch (runtime->state) { case SNDRV_PCM_STATE_OPEN: case SNDRV_PCM_STATE_DISCONNECTED: @@ -2028,15 +1980,15 @@ static int snd_pcm_pre_drain_init(struct snd_pcm_substream *substream, return 0; } -static int snd_pcm_do_drain_init(struct snd_pcm_substream *substream, - snd_pcm_state_t state) +static int snd_pcm_do_drain_init(struct snd_pcm_substream *substream, snd_pcm_state_t state) { struct snd_pcm_runtime *runtime = substream->runtime; + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { switch (runtime->state) { case SNDRV_PCM_STATE_PREPARED: /* start playback stream if possible */ - if (! snd_pcm_playback_empty(substream)) { + if (!snd_pcm_playback_empty(substream)) { snd_pcm_do_start(substream, SNDRV_PCM_STATE_DRAINING); snd_pcm_post_start(substream, SNDRV_PCM_STATE_DRAINING); } else { @@ -2064,17 +2016,15 @@ static int snd_pcm_do_drain_init(struct snd_pcm_substream *substream, } } - if (runtime->state == SNDRV_PCM_STATE_DRAINING && - runtime->trigger_master == substream && - (runtime->hw.info & SNDRV_PCM_INFO_DRAIN_TRIGGER)) + if (runtime->state == SNDRV_PCM_STATE_DRAINING && runtime->trigger_master == substream && + runtime->hw.info & SNDRV_PCM_INFO_DRAIN_TRIGGER) return substream->ops->trigger(substream, SNDRV_PCM_TRIGGER_DRAIN); return 0; } -static void snd_pcm_post_drain_init(struct snd_pcm_substream *substream, - snd_pcm_state_t state) +static void snd_pcm_post_drain_init(struct snd_pcm_substream *substream, snd_pcm_state_t state) { } @@ -2091,8 +2041,7 @@ static const struct action_ops snd_pcm_action_drain_init = { * After this call, all streams are supposed to be either SETUP or DRAINING * (capture only) state. */ -static int snd_pcm_drain(struct snd_pcm_substream *substream, - struct file *file) +static int snd_pcm_drain(struct snd_pcm_substream *substream, struct file *file) { struct snd_card *card; struct snd_pcm_runtime *runtime; @@ -2111,8 +2060,9 @@ static int snd_pcm_drain(struct snd_pcm_substream *substream, if (file) { if (file->f_flags & O_NONBLOCK) nonblock = 1; - } else if (substream->f_flags & O_NONBLOCK) + } else if (substream->f_flags & O_NONBLOCK) { nonblock = 1; + } snd_pcm_stream_lock_irq(substream); /* resume pause */ @@ -2120,8 +2070,7 @@ static int snd_pcm_drain(struct snd_pcm_substream *substream, snd_pcm_pause(substream, false); /* pre-start/stop - all running streams are changed to DRAINING state */ - result = snd_pcm_action(&snd_pcm_action_drain_init, substream, - ACTION_ARG_IGNORE); + result = snd_pcm_action(&snd_pcm_action_drain_init, substream, ACTION_ARG_IGNORE); if (result < 0) goto unlock; /* in non-blocking, we don't wait in ioctl but let caller poll */ @@ -2131,8 +2080,9 @@ static int snd_pcm_drain(struct snd_pcm_substream *substream, } for (;;) { - long tout; + long tout, t; struct snd_pcm_runtime *to_check; + if (signal_pending(current)) { result = -ERESTARTSYS; break; @@ -2156,12 +2106,12 @@ static int snd_pcm_drain(struct snd_pcm_substream *substream, set_current_state(TASK_INTERRUPTIBLE); add_wait_queue(&to_check->sleep, &wait); snd_pcm_stream_unlock_irq(substream); - if (runtime->no_period_wakeup) + if (runtime->no_period_wakeup) { tout = MAX_SCHEDULE_TIMEOUT; - else { + } else { tout = 100; if (runtime->rate) { - long t = runtime->buffer_size * 1100 / runtime->rate; + t = runtime->buffer_size * 1100 / runtime->rate; tout = max(t, tout); } tout = msecs_to_jiffies(tout); @@ -2183,9 +2133,9 @@ static int snd_pcm_drain(struct snd_pcm_substream *substream, break; } if (tout == 0) { - if (substream->runtime->state == SNDRV_PCM_STATE_SUSPENDED) + if (substream->runtime->state == SNDRV_PCM_STATE_SUSPENDED) { result = -ESTRPIPE; - else { + } else { dev_dbg(substream->pcm->card->dev, "playback drain timeout (DMA or IRQ trouble?)\n"); snd_pcm_stop(substream, SNDRV_PCM_STATE_SETUP); @@ -2195,7 +2145,7 @@ static int snd_pcm_drain(struct snd_pcm_substream *substream, } } - unlock: +unlock: snd_pcm_stream_unlock_irq(substream); return result; @@ -2210,7 +2160,7 @@ static int snd_pcm_drop(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime; int result = 0; - + if (PCM_RUNTIME_CHECK(substream)) return -ENXIO; runtime = substream->runtime; @@ -2231,7 +2181,6 @@ static int snd_pcm_drop(struct snd_pcm_substream *substream) return result; } - static bool is_pcm_file(struct file *file) { struct inode *inode = file_inode(file); @@ -2347,8 +2296,7 @@ static int snd_pcm_unlink(struct snd_pcm_substream *substream) /* detach the last stream, too */ if (list_is_singular(&group->substreams)) { - relink_to_local(list_first_entry(&group->substreams, - struct snd_pcm_substream, + relink_to_local(list_first_entry(&group->substreams, struct snd_pcm_substream, link_list)); do_free = refcount_dec_and_test(&group->refs); } @@ -2357,7 +2305,7 @@ static int snd_pcm_unlink(struct snd_pcm_substream *substream) if (do_free) kfree(group); - _end: +_end: up_write(&snd_pcm_link_rwsem); return res; } @@ -2365,62 +2313,62 @@ static int snd_pcm_unlink(struct snd_pcm_substream *substream) /* * hw configurator */ -static int snd_pcm_hw_rule_mul(struct snd_pcm_hw_params *params, - struct snd_pcm_hw_rule *rule) +static int snd_pcm_hw_rule_mul(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) { struct snd_interval t; + snd_interval_mul(hw_param_interval_c(params, rule->deps[0]), - hw_param_interval_c(params, rule->deps[1]), &t); + hw_param_interval_c(params, rule->deps[1]), &t); return snd_interval_refine(hw_param_interval(params, rule->var), &t); } -static int snd_pcm_hw_rule_div(struct snd_pcm_hw_params *params, - struct snd_pcm_hw_rule *rule) +static int snd_pcm_hw_rule_div(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) { struct snd_interval t; + snd_interval_div(hw_param_interval_c(params, rule->deps[0]), - hw_param_interval_c(params, rule->deps[1]), &t); + hw_param_interval_c(params, rule->deps[1]), &t); return snd_interval_refine(hw_param_interval(params, rule->var), &t); } -static int snd_pcm_hw_rule_muldivk(struct snd_pcm_hw_params *params, - struct snd_pcm_hw_rule *rule) +static int snd_pcm_hw_rule_muldivk(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) { struct snd_interval t; + snd_interval_muldivk(hw_param_interval_c(params, rule->deps[0]), - hw_param_interval_c(params, rule->deps[1]), - (unsigned long) rule->private, &t); + hw_param_interval_c(params, rule->deps[1]), + (unsigned long)rule->private, &t); return snd_interval_refine(hw_param_interval(params, rule->var), &t); } -static int snd_pcm_hw_rule_mulkdiv(struct snd_pcm_hw_params *params, - struct snd_pcm_hw_rule *rule) +static int snd_pcm_hw_rule_mulkdiv(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) { struct snd_interval t; + snd_interval_mulkdiv(hw_param_interval_c(params, rule->deps[0]), - (unsigned long) rule->private, - hw_param_interval_c(params, rule->deps[1]), &t); + (unsigned long)rule->private, + hw_param_interval_c(params, rule->deps[1]), &t); return snd_interval_refine(hw_param_interval(params, rule->var), &t); } -static int snd_pcm_hw_rule_format(struct snd_pcm_hw_params *params, - struct snd_pcm_hw_rule *rule) +static int snd_pcm_hw_rule_format(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) { snd_pcm_format_t k; - const struct snd_interval *i = - hw_param_interval_c(params, rule->deps[0]); + const struct snd_interval *i = hw_param_interval_c(params, rule->deps[0]); struct snd_mask m; struct snd_mask *mask = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + snd_mask_any(&m); pcm_for_each_format(k) { int bits; + if (!snd_mask_test_format(mask, k)) continue; bits = snd_pcm_format_physical_width(k); if (bits <= 0) continue; /* ignore invalid formats */ - if ((unsigned)bits < i->min || (unsigned)bits > i->max) - snd_mask_reset(&m, (__force unsigned)k); + if ((unsigned int)bits < i->min || (unsigned int)bits > i->max) + snd_mask_reset(&m, (__force unsigned int)k); } return snd_mask_refine(mask, &m); } @@ -2437,14 +2385,15 @@ static int snd_pcm_hw_rule_sample_bits(struct snd_pcm_hw_params *params, t.openmax = 0; pcm_for_each_format(k) { int bits; + if (!snd_mask_test_format(hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT), k)) continue; bits = snd_pcm_format_physical_width(k); if (bits <= 0) continue; /* ignore invalid formats */ - if (t.min > (unsigned)bits) + if (t.min > (unsigned int)bits) t.min = bits; - if (t.max < (unsigned)bits) + if (t.max < (unsigned int)bits) t.max = bits; } t.integer = 1; @@ -2465,27 +2414,28 @@ const struct snd_pcm_hw_constraint_list snd_pcm_known_rates = { .list = rates, }; -static int snd_pcm_hw_rule_rate(struct snd_pcm_hw_params *params, - struct snd_pcm_hw_rule *rule) +static int snd_pcm_hw_rule_rate(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) { struct snd_pcm_hardware *hw = rule->private; + return snd_interval_list(hw_param_interval(params, rule->var), snd_pcm_known_rates.count, snd_pcm_known_rates.list, hw->rates); -} +} static int snd_pcm_hw_rule_buffer_bytes_max(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) { struct snd_interval t; struct snd_pcm_substream *substream = rule->private; + t.min = 0; t.max = substream->buffer_bytes_max; t.openmin = 0; t.openmax = 0; t.integer = 1; return snd_interval_refine(hw_param_interval(params, rule->var), &t); -} +} static int snd_pcm_hw_constraints_init(struct snd_pcm_substream *substream) { @@ -2493,13 +2443,11 @@ static int snd_pcm_hw_constraints_init(struct snd_pcm_substream *substream) struct snd_pcm_hw_constraints *constrs = &runtime->hw_constraints; int k, err; - for (k = SNDRV_PCM_HW_PARAM_FIRST_MASK; k <= SNDRV_PCM_HW_PARAM_LAST_MASK; k++) { + for (k = SNDRV_PCM_HW_PARAM_FIRST_MASK; k <= SNDRV_PCM_HW_PARAM_LAST_MASK; k++) snd_mask_any(constrs_mask(constrs, k)); - } - for (k = SNDRV_PCM_HW_PARAM_FIRST_INTERVAL; k <= SNDRV_PCM_HW_PARAM_LAST_INTERVAL; k++) { + for (k = SNDRV_PCM_HW_PARAM_FIRST_INTERVAL; k <= SNDRV_PCM_HW_PARAM_LAST_INTERVAL; k++) snd_interval_any(constrs_interval(constrs, k)); - } snd_interval_setinteger(constrs_interval(constrs, SNDRV_PCM_HW_PARAM_CHANNELS)); snd_interval_setinteger(constrs_interval(constrs, SNDRV_PCM_HW_PARAM_BUFFER_SIZE)); @@ -2508,103 +2456,115 @@ static int snd_pcm_hw_constraints_init(struct snd_pcm_substream *substream) snd_interval_setinteger(constrs_interval(constrs, SNDRV_PCM_HW_PARAM_FRAME_BITS)); err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_FORMAT, - snd_pcm_hw_rule_format, NULL, + snd_pcm_hw_rule_format, NULL, SNDRV_PCM_HW_PARAM_SAMPLE_BITS, -1); if (err < 0) return err; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_SAMPLE_BITS, + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_SAMPLE_BITS, snd_pcm_hw_rule_sample_bits, NULL, - SNDRV_PCM_HW_PARAM_FORMAT, + SNDRV_PCM_HW_PARAM_FORMAT, SNDRV_PCM_HW_PARAM_SAMPLE_BITS, -1); if (err < 0) return err; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_SAMPLE_BITS, + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_SAMPLE_BITS, snd_pcm_hw_rule_div, NULL, - SNDRV_PCM_HW_PARAM_FRAME_BITS, SNDRV_PCM_HW_PARAM_CHANNELS, -1); + SNDRV_PCM_HW_PARAM_FRAME_BITS, + SNDRV_PCM_HW_PARAM_CHANNELS, -1); if (err < 0) return err; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_FRAME_BITS, + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_FRAME_BITS, snd_pcm_hw_rule_mul, NULL, - SNDRV_PCM_HW_PARAM_SAMPLE_BITS, SNDRV_PCM_HW_PARAM_CHANNELS, -1); + SNDRV_PCM_HW_PARAM_SAMPLE_BITS, + SNDRV_PCM_HW_PARAM_CHANNELS, -1); if (err < 0) return err; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_FRAME_BITS, - snd_pcm_hw_rule_mulkdiv, (void*) 8, - SNDRV_PCM_HW_PARAM_PERIOD_BYTES, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, -1); + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_FRAME_BITS, + snd_pcm_hw_rule_mulkdiv, (void *)8, + SNDRV_PCM_HW_PARAM_PERIOD_BYTES, + SNDRV_PCM_HW_PARAM_PERIOD_SIZE, -1); if (err < 0) return err; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_FRAME_BITS, - snd_pcm_hw_rule_mulkdiv, (void*) 8, - SNDRV_PCM_HW_PARAM_BUFFER_BYTES, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, -1); + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_FRAME_BITS, + snd_pcm_hw_rule_mulkdiv, (void *)8, + SNDRV_PCM_HW_PARAM_BUFFER_BYTES, + SNDRV_PCM_HW_PARAM_BUFFER_SIZE, -1); if (err < 0) return err; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, snd_pcm_hw_rule_div, NULL, - SNDRV_PCM_HW_PARAM_FRAME_BITS, SNDRV_PCM_HW_PARAM_SAMPLE_BITS, -1); + SNDRV_PCM_HW_PARAM_FRAME_BITS, + SNDRV_PCM_HW_PARAM_SAMPLE_BITS, -1); if (err < 0) return err; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, - snd_pcm_hw_rule_mulkdiv, (void*) 1000000, - SNDRV_PCM_HW_PARAM_PERIOD_SIZE, SNDRV_PCM_HW_PARAM_PERIOD_TIME, -1); + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + snd_pcm_hw_rule_mulkdiv, (void *)1000000, + SNDRV_PCM_HW_PARAM_PERIOD_SIZE, + SNDRV_PCM_HW_PARAM_PERIOD_TIME, -1); if (err < 0) return err; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, - snd_pcm_hw_rule_mulkdiv, (void*) 1000000, - SNDRV_PCM_HW_PARAM_BUFFER_SIZE, SNDRV_PCM_HW_PARAM_BUFFER_TIME, -1); + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + snd_pcm_hw_rule_mulkdiv, (void *)1000000, + SNDRV_PCM_HW_PARAM_BUFFER_SIZE, + SNDRV_PCM_HW_PARAM_BUFFER_TIME, -1); if (err < 0) return err; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_PERIODS, + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_PERIODS, snd_pcm_hw_rule_div, NULL, - SNDRV_PCM_HW_PARAM_BUFFER_SIZE, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, -1); + SNDRV_PCM_HW_PARAM_BUFFER_SIZE, + SNDRV_PCM_HW_PARAM_PERIOD_SIZE, -1); if (err < 0) return err; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, snd_pcm_hw_rule_div, NULL, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, SNDRV_PCM_HW_PARAM_PERIODS, -1); if (err < 0) return err; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, - snd_pcm_hw_rule_mulkdiv, (void*) 8, - SNDRV_PCM_HW_PARAM_PERIOD_BYTES, SNDRV_PCM_HW_PARAM_FRAME_BITS, -1); + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, + snd_pcm_hw_rule_mulkdiv, (void *)8, + SNDRV_PCM_HW_PARAM_PERIOD_BYTES, + SNDRV_PCM_HW_PARAM_FRAME_BITS, -1); if (err < 0) return err; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, - snd_pcm_hw_rule_muldivk, (void*) 1000000, + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, + snd_pcm_hw_rule_muldivk, (void *)1000000, SNDRV_PCM_HW_PARAM_PERIOD_TIME, SNDRV_PCM_HW_PARAM_RATE, -1); if (err < 0) return err; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, snd_pcm_hw_rule_mul, NULL, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, SNDRV_PCM_HW_PARAM_PERIODS, -1); if (err < 0) return err; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, - snd_pcm_hw_rule_mulkdiv, (void*) 8, - SNDRV_PCM_HW_PARAM_BUFFER_BYTES, SNDRV_PCM_HW_PARAM_FRAME_BITS, -1); + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, + snd_pcm_hw_rule_mulkdiv, (void *)8, + SNDRV_PCM_HW_PARAM_BUFFER_BYTES, + SNDRV_PCM_HW_PARAM_FRAME_BITS, -1); if (err < 0) return err; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, - snd_pcm_hw_rule_muldivk, (void*) 1000000, + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, + snd_pcm_hw_rule_muldivk, (void *)1000000, SNDRV_PCM_HW_PARAM_BUFFER_TIME, SNDRV_PCM_HW_PARAM_RATE, -1); if (err < 0) return err; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, - snd_pcm_hw_rule_muldivk, (void*) 8, - SNDRV_PCM_HW_PARAM_PERIOD_SIZE, SNDRV_PCM_HW_PARAM_FRAME_BITS, -1); + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, + snd_pcm_hw_rule_muldivk, (void *)8, + SNDRV_PCM_HW_PARAM_PERIOD_SIZE, + SNDRV_PCM_HW_PARAM_FRAME_BITS, -1); if (err < 0) return err; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, - snd_pcm_hw_rule_muldivk, (void*) 8, - SNDRV_PCM_HW_PARAM_BUFFER_SIZE, SNDRV_PCM_HW_PARAM_FRAME_BITS, -1); + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, + snd_pcm_hw_rule_muldivk, (void *)8, + SNDRV_PCM_HW_PARAM_BUFFER_SIZE, + SNDRV_PCM_HW_PARAM_FRAME_BITS, -1); if (err < 0) return err; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_TIME, - snd_pcm_hw_rule_mulkdiv, (void*) 1000000, + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_TIME, + snd_pcm_hw_rule_mulkdiv, (void *)1000000, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, SNDRV_PCM_HW_PARAM_RATE, -1); if (err < 0) return err; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_TIME, - snd_pcm_hw_rule_mulkdiv, (void*) 1000000, + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_TIME, + snd_pcm_hw_rule_mulkdiv, (void *)1000000, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, SNDRV_PCM_HW_PARAM_RATE, -1); if (err < 0) return err; @@ -2618,9 +2578,9 @@ static int snd_pcm_hw_constraints_complete(struct snd_pcm_substream *substream) int err; unsigned int mask = 0; - if (hw->info & SNDRV_PCM_INFO_INTERLEAVED) + if (hw->info & SNDRV_PCM_INFO_INTERLEAVED) mask |= PARAM_MASK_BIT(SNDRV_PCM_ACCESS_RW_INTERLEAVED); - if (hw->info & SNDRV_PCM_INFO_NONINTERLEAVED) + if (hw->info & SNDRV_PCM_INFO_NONINTERLEAVED) mask |= PARAM_MASK_BIT(SNDRV_PCM_ACCESS_RW_NONINTERLEAVED); if (hw_support_mmap(substream)) { if (hw->info & SNDRV_PCM_INFO_INTERLEAVED) @@ -2668,7 +2628,7 @@ static int snd_pcm_hw_constraints_complete(struct snd_pcm_substream *substream) if (err < 0) return err; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, snd_pcm_hw_rule_buffer_bytes_max, substream, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, -1); if (err < 0) @@ -2676,13 +2636,14 @@ static int snd_pcm_hw_constraints_complete(struct snd_pcm_substream *substream) /* FIXME: remove */ if (runtime->dma_bytes) { - err = snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 0, runtime->dma_bytes); + err = snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, + 0, runtime->dma_bytes); if (err < 0) return err; } if (!(hw->rates & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS))) { - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, snd_pcm_hw_rule_rate, hw, SNDRV_PCM_HW_PARAM_RATE, -1); if (err < 0) @@ -2724,8 +2685,7 @@ void snd_pcm_release_substream(struct snd_pcm_substream *substream) } EXPORT_SYMBOL(snd_pcm_release_substream); -int snd_pcm_open_substream(struct snd_pcm *pcm, int stream, - struct file *file, +int snd_pcm_open_substream(struct snd_pcm *pcm, int stream, struct file *file, struct snd_pcm_substream **rsubstream) { struct snd_pcm_substream *substream; @@ -2760,8 +2720,7 @@ int snd_pcm_open_substream(struct snd_pcm *pcm, int stream, /* automatically set EXPLICIT_SYNC flag in the managed mode whenever * the DMA buffer requires it */ - if (substream->managed_buffer_alloc && - substream->dma_buffer.dev.need_sync) + if (substream->managed_buffer_alloc && substream->dma_buffer.dev.need_sync) substream->runtime->hw.info |= SNDRV_PCM_INFO_EXPLICIT_SYNC; *rsubstream = substream; @@ -2773,9 +2732,7 @@ int snd_pcm_open_substream(struct snd_pcm *pcm, int stream, } EXPORT_SYMBOL(snd_pcm_open_substream); -static int snd_pcm_open_file(struct file *file, - struct snd_pcm *pcm, - int stream) +static int snd_pcm_open_file(struct file *file, struct snd_pcm *pcm, int stream) { struct snd_pcm_file *pcm_file; struct snd_pcm_substream *substream; @@ -2786,7 +2743,7 @@ static int snd_pcm_open_file(struct file *file, return err; pcm_file = kzalloc(sizeof(*pcm_file), GFP_KERNEL); - if (pcm_file == NULL) { + if (!pcm_file) { snd_pcm_release_substream(substream); return -ENOMEM; } @@ -2802,10 +2759,10 @@ static int snd_pcm_playback_open(struct inode *inode, struct file *file) { struct snd_pcm *pcm; int err = nonseekable_open(inode, file); + if (err < 0) return err; - pcm = snd_lookup_minor_data(iminor(inode), - SNDRV_DEVICE_TYPE_PCM_PLAYBACK); + pcm = snd_lookup_minor_data(iminor(inode), SNDRV_DEVICE_TYPE_PCM_PLAYBACK); err = snd_pcm_open(file, pcm, SNDRV_PCM_STREAM_PLAYBACK); if (pcm) snd_card_unref(pcm->card); @@ -2816,10 +2773,10 @@ static int snd_pcm_capture_open(struct inode *inode, struct file *file) { struct snd_pcm *pcm; int err = nonseekable_open(inode, file); + if (err < 0) return err; - pcm = snd_lookup_minor_data(iminor(inode), - SNDRV_DEVICE_TYPE_PCM_CAPTURE); + pcm = snd_lookup_minor_data(iminor(inode), SNDRV_DEVICE_TYPE_PCM_CAPTURE); err = snd_pcm_open(file, pcm, SNDRV_PCM_STREAM_CAPTURE); if (pcm) snd_card_unref(pcm->card); @@ -2831,7 +2788,7 @@ static int snd_pcm_open(struct file *file, struct snd_pcm *pcm, int stream) int err; wait_queue_entry_t wait; - if (pcm == NULL) { + if (!pcm) { err = -ENODEV; goto __error1; } @@ -2854,8 +2811,9 @@ static int snd_pcm_open(struct file *file, struct snd_pcm *pcm, int stream) err = -EBUSY; break; } - } else + } else { break; + } set_current_state(TASK_INTERRUPTIBLE); mutex_unlock(&pcm->open_mutex); schedule(); @@ -2875,12 +2833,12 @@ static int snd_pcm_open(struct file *file, struct snd_pcm *pcm, int stream) goto __error; return err; - __error: +__error: module_put(pcm->card->module); - __error2: - snd_card_file_remove(pcm->card, file); - __error1: - return err; +__error2: + snd_card_file_remove(pcm->card, file); +__error1: + return err; } static int snd_pcm_release(struct inode *inode, struct file *file) @@ -2935,7 +2893,7 @@ static int do_pcm_hwsync(struct snd_pcm_substream *substream) /* increase the appl_ptr; returns the processed frames or a negative error */ static snd_pcm_sframes_t forward_appl_ptr(struct snd_pcm_substream *substream, snd_pcm_uframes_t frames, - snd_pcm_sframes_t avail) + snd_pcm_sframes_t avail) { struct snd_pcm_runtime *runtime = substream->runtime; snd_pcm_sframes_t appl_ptr; @@ -2988,8 +2946,7 @@ static snd_pcm_sframes_t snd_pcm_rewind(struct snd_pcm_substream *substream, snd_pcm_stream_lock_irq(substream); ret = do_pcm_hwsync(substream); if (!ret) - ret = rewind_appl_ptr(substream, frames, - snd_pcm_hw_avail(substream)); + ret = rewind_appl_ptr(substream, frames, snd_pcm_hw_avail(substream)); snd_pcm_stream_unlock_irq(substream); if (ret >= 0) snd_pcm_dma_buffer_sync(substream, SNDRV_DMA_SYNC_DEVICE); @@ -3007,16 +2964,14 @@ static snd_pcm_sframes_t snd_pcm_forward(struct snd_pcm_substream *substream, snd_pcm_stream_lock_irq(substream); ret = do_pcm_hwsync(substream); if (!ret) - ret = forward_appl_ptr(substream, frames, - snd_pcm_avail(substream)); + ret = forward_appl_ptr(substream, frames, snd_pcm_avail(substream)); snd_pcm_stream_unlock_irq(substream); if (ret >= 0) snd_pcm_dma_buffer_sync(substream, SNDRV_DMA_SYNC_DEVICE); return ret; } -static int snd_pcm_delay(struct snd_pcm_substream *substream, - snd_pcm_sframes_t *delay) +static int snd_pcm_delay(struct snd_pcm_substream *substream, snd_pcm_sframes_t *delay) { int err; @@ -3029,7 +2984,7 @@ static int snd_pcm_delay(struct snd_pcm_substream *substream, return err; } - + static inline int snd_pcm_hwsync(struct snd_pcm_substream *substream) { return snd_pcm_delay(substream, NULL); @@ -3045,10 +3000,11 @@ static int snd_pcm_sync_ptr(struct snd_pcm_substream *substream, int err; memset(&sync_ptr, 0, sizeof(sync_ptr)); - if (get_user(sync_ptr.flags, (unsigned __user *)&(_sync_ptr->flags))) + if (get_user(sync_ptr.flags, (unsigned __user *)&_sync_ptr->flags)) + return -EFAULT; + if (copy_from_user(&sync_ptr.c.control, &_sync_ptr->c.control, + sizeof(struct snd_pcm_mmap_control))) return -EFAULT; - if (copy_from_user(&sync_ptr.c.control, &(_sync_ptr->c.control), sizeof(struct snd_pcm_mmap_control))) - return -EFAULT; status = runtime->status; control = runtime->control; if (sync_ptr.flags & SNDRV_PCM_SYNC_PTR_HWSYNC) { @@ -3058,8 +3014,7 @@ static int snd_pcm_sync_ptr(struct snd_pcm_substream *substream, } snd_pcm_stream_lock_irq(substream); if (!(sync_ptr.flags & SNDRV_PCM_SYNC_PTR_APPL)) { - err = pcm_lib_apply_appl_ptr(substream, - sync_ptr.c.control.appl_ptr); + err = pcm_lib_apply_appl_ptr(substream, sync_ptr.c.control.appl_ptr); if (err < 0) { snd_pcm_stream_unlock_irq(substream); return err; @@ -3093,7 +3048,7 @@ struct snd_pcm_mmap_status32 { snd_pcm_state_t suspended_state; s32 audio_tstamp_sec; s32 audio_tstamp_nsec; -} __attribute__((packed)); +} __packed; struct snd_pcm_mmap_control32 { u32 appl_ptr; @@ -3110,14 +3065,14 @@ struct snd_pcm_sync_ptr32 { struct snd_pcm_mmap_control32 control; unsigned char reserved[64]; } c; -} __attribute__((packed)); +} __packed; /* recalcuate the boundary within 32bit */ static snd_pcm_uframes_t recalculate_boundary(struct snd_pcm_runtime *runtime) { snd_pcm_uframes_t boundary; - if (! runtime->buffer_size) + if (!runtime->buffer_size) return 0; boundary = runtime->buffer_size; while (boundary * 2 <= 0x7fffffffUL - runtime->buffer_size) @@ -3152,19 +3107,19 @@ static int snd_pcm_ioctl_sync_ptr_compat(struct snd_pcm_substream *substream, status = runtime->status; control = runtime->control; boundary = recalculate_boundary(runtime); - if (! boundary) + if (!boundary) boundary = 0x7fffffff; snd_pcm_stream_lock_irq(substream); /* FIXME: we should consider the boundary for the sync from app */ if (!(sflags & SNDRV_PCM_SYNC_PTR_APPL)) { - err = pcm_lib_apply_appl_ptr(substream, - scontrol.appl_ptr); + err = pcm_lib_apply_appl_ptr(substream, scontrol.appl_ptr); if (err < 0) { snd_pcm_stream_unlock_irq(substream); return err; } - } else + } else { scontrol.appl_ptr = control->appl_ptr % boundary; + } if (!(sflags & SNDRV_PCM_SYNC_PTR_AVAIL_MIN)) control->avail_min = scontrol.avail_min; else @@ -3190,13 +3145,14 @@ static int snd_pcm_ioctl_sync_ptr_compat(struct snd_pcm_substream *substream, return 0; } + #define __SNDRV_PCM_IOCTL_SYNC_PTR32 _IOWR('A', 0x23, struct snd_pcm_sync_ptr32) static int snd_pcm_tstamp(struct snd_pcm_substream *substream, int __user *_arg) { struct snd_pcm_runtime *runtime = substream->runtime; int arg; - + if (get_user(arg, _arg)) return -EFAULT; if (arg < 0 || arg > SNDRV_PCM_TSTAMP_TYPE_LAST) @@ -3290,8 +3246,8 @@ static int snd_pcm_forward_ioctl(struct snd_pcm_substream *substream, } static int snd_pcm_common_ioctl(struct file *file, - struct snd_pcm_substream *substream, - unsigned int cmd, void __user *arg) + struct snd_pcm_substream *substream, + unsigned int cmd, void __user *arg) { struct snd_pcm_file *pcm_file = file->private_data; int res; @@ -3316,8 +3272,7 @@ static int snd_pcm_common_ioctl(struct file *file, case SNDRV_PCM_IOCTL_TTSTAMP: return snd_pcm_tstamp(substream, arg); case SNDRV_PCM_IOCTL_USER_PVERSION: - if (get_user(pcm_file->user_pversion, - (unsigned int __user *)arg)) + if (get_user(pcm_file->user_pversion, (unsigned int __user *)arg)) return -EFAULT; return 0; case SNDRV_PCM_IOCTL_HW_REFINE: @@ -3345,7 +3300,7 @@ static int snd_pcm_common_ioctl(struct file *file, case SNDRV_PCM_IOCTL_START: return snd_pcm_start_lock_irq(substream); case SNDRV_PCM_IOCTL_LINK: - return snd_pcm_link(substream, (int)(unsigned long) arg); + return snd_pcm_link(substream, (int)(unsigned long)arg); case SNDRV_PCM_IOCTL_UNLINK: return snd_pcm_unlink(substream); case SNDRV_PCM_IOCTL_RESUME: @@ -3398,8 +3353,7 @@ static int snd_pcm_common_ioctl(struct file *file, return -ENOTTY; } -static long snd_pcm_ioctl(struct file *file, unsigned int cmd, - unsigned long arg) +static long snd_pcm_ioctl(struct file *file, unsigned int cmd, unsigned long arg) { struct snd_pcm_file *pcm_file; @@ -3408,8 +3362,7 @@ static long snd_pcm_ioctl(struct file *file, unsigned int cmd, if (((cmd >> 8) & 0xff) != 'A') return -ENOTTY; - return snd_pcm_common_ioctl(file, pcm_file->substream, cmd, - (void __user *)arg); + return snd_pcm_common_ioctl(file, pcm_file->substream, cmd, (void __user *)arg); } /** @@ -3424,12 +3377,11 @@ static long snd_pcm_ioctl(struct file *file, unsigned int cmd, * * Return: zero if successful, or a negative error code */ -int snd_pcm_kernel_ioctl(struct snd_pcm_substream *substream, - unsigned int cmd, void *arg) +int snd_pcm_kernel_ioctl(struct snd_pcm_substream *substream, unsigned int cmd, void *arg) { snd_pcm_uframes_t *frames = arg; snd_pcm_sframes_t result; - + if (substream->runtime->state == SNDRV_PCM_STATE_DISCONNECTED) return -EBADFD; @@ -3462,8 +3414,7 @@ int snd_pcm_kernel_ioctl(struct snd_pcm_substream *substream, } EXPORT_SYMBOL(snd_pcm_kernel_ioctl); -static ssize_t snd_pcm_read(struct file *file, char __user *buf, size_t count, - loff_t * offset) +static ssize_t snd_pcm_read(struct file *file, char __user *buf, size_t count, loff_t *offset) { struct snd_pcm_file *pcm_file; struct snd_pcm_substream *substream; @@ -3488,7 +3439,7 @@ static ssize_t snd_pcm_read(struct file *file, char __user *buf, size_t count, } static ssize_t snd_pcm_write(struct file *file, const char __user *buf, - size_t count, loff_t * offset) + size_t count, loff_t *offset) { struct snd_pcm_file *pcm_file; struct snd_pcm_substream *substream; @@ -3539,7 +3490,7 @@ static ssize_t snd_pcm_readv(struct kiocb *iocb, struct iov_iter *to) return -EINVAL; frames = bytes_to_samples(runtime, iov->iov_len); bufs = kmalloc_array(to->nr_segs, sizeof(void *), GFP_KERNEL); - if (bufs == NULL) + if (!bufs) return -ENOMEM; for (i = 0; i < to->nr_segs; ++i) { bufs[i] = iov->iov_base; @@ -3578,7 +3529,7 @@ static ssize_t snd_pcm_writev(struct kiocb *iocb, struct iov_iter *from) return -EINVAL; frames = bytes_to_samples(runtime, iov->iov_len); bufs = kmalloc_array(from->nr_segs, sizeof(void *), GFP_KERNEL); - if (bufs == NULL) + if (!bufs) return -ENOMEM; for (i = 0; i < from->nr_segs; ++i) { bufs[i] = iov->iov_base; @@ -3656,8 +3607,8 @@ static vm_fault_t snd_pcm_mmap_status_fault(struct vm_fault *vmf) { struct snd_pcm_substream *substream = vmf->vma->vm_private_data; struct snd_pcm_runtime *runtime; - - if (substream == NULL) + + if (!substream) return VM_FAULT_SIGBUS; runtime = substream->runtime; vmf->page = virt_to_page(runtime->status); @@ -3665,8 +3616,7 @@ static vm_fault_t snd_pcm_mmap_status_fault(struct vm_fault *vmf) return 0; } -static const struct vm_operations_struct snd_pcm_vm_ops_status = -{ +static const struct vm_operations_struct snd_pcm_vm_ops_status = { .fault = snd_pcm_mmap_status_fault, }; @@ -3674,6 +3624,7 @@ static int snd_pcm_mmap_status(struct snd_pcm_substream *substream, struct file struct vm_area_struct *area) { long size; + if (!(area->vm_flags & VM_READ)) return -EINVAL; size = area->vm_end - area->vm_start; @@ -3681,8 +3632,7 @@ static int snd_pcm_mmap_status(struct snd_pcm_substream *substream, struct file return -EINVAL; area->vm_ops = &snd_pcm_vm_ops_status; area->vm_private_data = substream; - vm_flags_mod(area, VM_DONTEXPAND | VM_DONTDUMP, - VM_WRITE | VM_MAYWRITE); + vm_flags_mod(area, VM_DONTEXPAND | VM_DONTDUMP, VM_WRITE | VM_MAYWRITE); return 0; } @@ -3694,8 +3644,8 @@ static vm_fault_t snd_pcm_mmap_control_fault(struct vm_fault *vmf) { struct snd_pcm_substream *substream = vmf->vma->vm_private_data; struct snd_pcm_runtime *runtime; - - if (substream == NULL) + + if (!substream) return VM_FAULT_SIGBUS; runtime = substream->runtime; vmf->page = virt_to_page(runtime->control); @@ -3703,8 +3653,7 @@ static vm_fault_t snd_pcm_mmap_control_fault(struct vm_fault *vmf) return 0; } -static const struct vm_operations_struct snd_pcm_vm_ops_control = -{ +static const struct vm_operations_struct snd_pcm_vm_ops_control = { .fault = snd_pcm_mmap_control_fault, }; @@ -3712,6 +3661,7 @@ static int snd_pcm_mmap_control(struct snd_pcm_substream *substream, struct file struct vm_area_struct *area) { long size; + if (!(area->vm_flags & VM_READ)) return -EINVAL; size = area->vm_end - area->vm_start; @@ -3769,11 +3719,13 @@ static int snd_pcm_mmap_status(struct snd_pcm_substream *substream, struct file { return -ENXIO; } + static int snd_pcm_mmap_control(struct snd_pcm_substream *substream, struct file *file, struct vm_area_struct *area) { return -ENXIO; } + #endif /* coherent mmap */ /* @@ -3784,10 +3736,10 @@ static vm_fault_t snd_pcm_mmap_data_fault(struct vm_fault *vmf) struct snd_pcm_substream *substream = vmf->vma->vm_private_data; struct snd_pcm_runtime *runtime; unsigned long offset; - struct page * page; + struct page *page; size_t dma_bytes; - - if (substream == NULL) + + if (!substream) return VM_FAULT_SIGBUS; runtime = substream->runtime; offset = vmf->pgoff << PAGE_SHIFT; @@ -3884,7 +3836,7 @@ int snd_pcm_mmap_data(struct snd_pcm_substream *substream, struct file *file, int err; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - if (!(area->vm_flags & (VM_WRITE|VM_READ))) + if (!(area->vm_flags & (VM_WRITE | VM_READ))) return -EINVAL; } else { if (!(area->vm_flags & VM_READ)) @@ -3920,10 +3872,10 @@ EXPORT_SYMBOL(snd_pcm_mmap_data); static int snd_pcm_mmap(struct file *file, struct vm_area_struct *area) { - struct snd_pcm_file * pcm_file; - struct snd_pcm_substream *substream; + struct snd_pcm_file *pcm_file; + struct snd_pcm_substream *substream; unsigned long offset; - + pcm_file = file->private_data; substream = pcm_file->substream; if (PCM_RUNTIME_CHECK(substream)) @@ -3955,9 +3907,9 @@ static int snd_pcm_mmap(struct file *file, struct vm_area_struct *area) return 0; } -static int snd_pcm_fasync(int fd, struct file * file, int on) +static int snd_pcm_fasync(int fd, struct file *file, int on) { - struct snd_pcm_file * pcm_file; + struct snd_pcm_file *pcm_file; struct snd_pcm_substream *substream; struct snd_pcm_runtime *runtime; @@ -3985,8 +3937,8 @@ static int snd_pcm_fasync(int fd, struct file * file, int on) */ #ifdef CONFIG_SND_SUPPORT_OLD_API -#define __OLD_TO_NEW_MASK(x) ((x&7)|((x&0x07fffff8)<<5)) -#define __NEW_TO_OLD_MASK(x) ((x&7)|((x&0xffffff00)>>5)) +#define __OLD_TO_NEW_MASK(x) ((x & 7) | ((x & 0x07fffff8) << 5)) +#define __NEW_TO_OLD_MASK(x) ((x & 7) | ((x & 0xffffff00) >> 5)) static void snd_pcm_hw_convert_from_old_params(struct snd_pcm_hw_params *params, struct snd_pcm_hw_params_old *oparams) @@ -4027,7 +3979,7 @@ static void snd_pcm_hw_convert_to_old_params(struct snd_pcm_hw_params_old *opara } static int snd_pcm_hw_refine_old_user(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params_old __user * _oparams) + struct snd_pcm_hw_params_old __user *_oparams) { struct snd_pcm_hw_params *params; struct snd_pcm_hw_params_old *oparams = NULL; @@ -4062,7 +4014,7 @@ static int snd_pcm_hw_refine_old_user(struct snd_pcm_substream *substream, } static int snd_pcm_hw_params_old_user(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params_old __user * _oparams) + struct snd_pcm_hw_params_old __user *_oparams) { struct snd_pcm_hw_params *params; struct snd_pcm_hw_params_old *oparams = NULL; @@ -4095,10 +4047,8 @@ static int snd_pcm_hw_params_old_user(struct snd_pcm_substream *substream, #endif /* CONFIG_SND_SUPPORT_OLD_API */ #ifndef CONFIG_MMU -static unsigned long snd_pcm_get_unmapped_area(struct file *file, - unsigned long addr, - unsigned long len, - unsigned long pgoff, +static unsigned long snd_pcm_get_unmapped_area(struct file *file, unsigned long addr, + unsigned long len, unsigned long pgoff, unsigned long flags) { struct snd_pcm_file *pcm_file = file->private_data; @@ -4133,7 +4083,7 @@ const struct file_operations snd_pcm_f_ops[2] = { .llseek = no_llseek, .poll = snd_pcm_poll, .unlocked_ioctl = snd_pcm_ioctl, - .compat_ioctl = snd_pcm_ioctl_compat, + .compat_ioctl = snd_pcm_ioctl_compat, .mmap = snd_pcm_mmap, .fasync = snd_pcm_fasync, .get_unmapped_area = snd_pcm_get_unmapped_area, @@ -4147,7 +4097,7 @@ const struct file_operations snd_pcm_f_ops[2] = { .llseek = no_llseek, .poll = snd_pcm_poll, .unlocked_ioctl = snd_pcm_ioctl, - .compat_ioctl = snd_pcm_ioctl_compat, + .compat_ioctl = snd_pcm_ioctl_compat, .mmap = snd_pcm_mmap, .fasync = snd_pcm_fasync, .get_unmapped_area = snd_pcm_get_unmapped_area, -- 2.34.1

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