August 2022 - Alsa-devel - mailman.alsa-project.org

[PATCH -next] ASoC: codecs: es8326: change es8326_regmap_config to static
by Yang Yingliang 17 Aug '22

17 Aug '22

es8326_regmap_config is only used in es8326.c now, change it to static. Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/soc/codecs/es8326.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/es8326.c b/sound/soc/codecs/es8326.c index 975302b2a61d..87c1cc16592b 100755 --- a/sound/soc/codecs/es8326.c +++ b/sound/soc/codecs/es8326.c @@ -207,7 +207,7 @@ static const struct regmap_access_table es8326_volatile_table = { .n_yes_ranges = ARRAY_SIZE(es8326_volatile_ranges), }; -const struct regmap_config es8326_regmap_config = { +static const struct regmap_config es8326_regmap_config = { .reg_bits = 8, .val_bits = 8, .max_register = 0xff, -- 2.25.1

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[PATCH 0/4] Enable timestamp support for SOF compress driver
by Daniel Baluta 17 Aug '22

17 Aug '22

From: Daniel Baluta <daniel.baluta(a)nxp.com> This patchseries computes pcm_io_frames from the DAI position reported by SOF firmware. Using pcm_io_frames userspace applications can later compute timestamps for compressed stream. Daniel Baluta (2): ASoC: SOF: compress: Remove byte offset computation ASoC: SOF: compress: Introduce sof_compr_stream Laurentiu Mihalcea (2): ASoC: SOF: compress: Save channel count and sample bytes ASoC: SOF: compress: Add support for computing timestamps sound/soc/sof/compress.c | 54 ++++++++++++++++++++++++---------------- sound/soc/sof/sof-priv.h | 7 ++++++ 2 files changed, 40 insertions(+), 21 deletions(-) -- 2.27.0

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[PATCH] ASoC: qcom: qdsp6: q6prm: add new clocks
by Srinivas Kandagatla 17 Aug '22

17 Aug '22

Add support to new clocks that are added in Q6DSP as part of newer version of LPASS support on SM8450 and SC8280XP. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- .../sound/qcom,q6dsp-lpass-ports.h | 18 ++++++++++++++++++ sound/soc/qcom/qdsp6/q6prm-clocks.c | 9 +++++++++ sound/soc/qcom/qdsp6/q6prm.h | 19 +++++++++++++++++++ 3 files changed, 46 insertions(+) diff --git a/include/dt-bindings/sound/qcom,q6dsp-lpass-ports.h b/include/dt-bindings/sound/qcom,q6dsp-lpass-ports.h index 0d3276c8fc11..9f7c5103bc82 100644 --- a/include/dt-bindings/sound/qcom,q6dsp-lpass-ports.h +++ b/include/dt-bindings/sound/qcom,q6dsp-lpass-ports.h @@ -193,6 +193,24 @@ #define LPASS_CLK_ID_RX_CORE_MCLK 59 #define LPASS_CLK_ID_RX_CORE_NPL_MCLK 60 #define LPASS_CLK_ID_VA_CORE_2X_MCLK 61 +/* Clock ID for MCLK for WSA2 core */ +#define LPASS_CLK_ID_WSA2_CORE_MCLK 62 +/* Clock ID for NPL MCLK for WSA2 core */ +#define LPASS_CLK_ID_WSA2_CORE_2X_MCLK 63 +/* Clock ID for RX Core TX MCLK */ +#define LPASS_CLK_ID_RX_CORE_TX_MCLK 64 +/* Clock ID for RX CORE TX 2X MCLK */ +#define LPASS_CLK_ID_RX_CORE_TX_2X_MCLK 65 +/* Clock ID for WSA core TX MCLK */ +#define LPASS_CLK_ID_WSA_CORE_TX_MCLK 66 +/* Clock ID for WSA core TX 2X MCLK */ +#define LPASS_CLK_ID_WSA_CORE_TX_2X_MCLK 67 +/* Clock ID for WSA2 core TX MCLK */ +#define LPASS_CLK_ID_WSA2_CORE_TX_MCLK 68 +/* Clock ID for WSA2 core TX 2X MCLK */ +#define LPASS_CLK_ID_WSA2_CORE_TX_2X_MCLK 69 +/* Clock ID for RX CORE MCLK2 2X MCLK */ +#define LPASS_CLK_ID_RX_CORE_MCLK2_2X_MCLK 70 #define LPASS_HW_AVTIMER_VOTE 101 #define LPASS_HW_MACRO_VOTE 102 diff --git a/sound/soc/qcom/qdsp6/q6prm-clocks.c b/sound/soc/qcom/qdsp6/q6prm-clocks.c index a26cda5140c1..73b0cbac73d4 100644 --- a/sound/soc/qcom/qdsp6/q6prm-clocks.c +++ b/sound/soc/qcom/qdsp6/q6prm-clocks.c @@ -50,6 +50,15 @@ static const struct q6dsp_clk_init q6prm_clks[] = { Q6PRM_CLK(LPASS_CLK_ID_RX_CORE_MCLK), Q6PRM_CLK(LPASS_CLK_ID_RX_CORE_NPL_MCLK), Q6PRM_CLK(LPASS_CLK_ID_VA_CORE_2X_MCLK), + Q6PRM_CLK(LPASS_CLK_ID_WSA2_CORE_MCLK), + Q6PRM_CLK(LPASS_CLK_ID_WSA2_CORE_2X_MCLK), + Q6PRM_CLK(LPASS_CLK_ID_RX_CORE_TX_MCLK), + Q6PRM_CLK(LPASS_CLK_ID_RX_CORE_TX_2X_MCLK), + Q6PRM_CLK(LPASS_CLK_ID_WSA_CORE_TX_MCLK), + Q6PRM_CLK(LPASS_CLK_ID_WSA_CORE_TX_2X_MCLK), + Q6PRM_CLK(LPASS_CLK_ID_WSA2_CORE_TX_MCLK), + Q6PRM_CLK(LPASS_CLK_ID_WSA2_CORE_TX_2X_MCLK), + Q6PRM_CLK(LPASS_CLK_ID_RX_CORE_MCLK2_2X_MCLK), Q6DSP_VOTE_CLK(LPASS_HW_MACRO_VOTE, Q6PRM_HW_CORE_ID_LPASS, "LPASS_HW_MACRO"), Q6DSP_VOTE_CLK(LPASS_HW_DCODEC_VOTE, Q6PRM_HW_CORE_ID_DCODEC, diff --git a/sound/soc/qcom/qdsp6/q6prm.h b/sound/soc/qcom/qdsp6/q6prm.h index fea4d1954bc1..a988a32086fe 100644 --- a/sound/soc/qcom/qdsp6/q6prm.h +++ b/sound/soc/qcom/qdsp6/q6prm.h @@ -64,6 +64,25 @@ #define Q6PRM_LPASS_CLK_ID_RX_CORE_MCLK 0x30e #define Q6PRM_LPASS_CLK_ID_RX_CORE_NPL_MCLK 0x30f +/* Clock ID for MCLK for WSA2 core */ +#define Q6PRM_LPASS_CLK_ID_WSA2_CORE_MCLK 0x310 +/* Clock ID for NPL MCLK for WSA2 core */ +#define Q6PRM_LPASS_CLK_ID_WSA2_CORE_2X_MCLK 0x311 +/* Clock ID for RX Core TX MCLK */ +#define Q6PRM_LPASS_CLK_ID_RX_CORE_TX_MCLK 0x312 +/* Clock ID for RX CORE TX 2X MCLK */ +#define Q6PRM_LPASS_CLK_ID_RX_CORE_TX_2X_MCLK 0x313 +/* Clock ID for WSA core TX MCLK */ +#define Q6PRM_LPASS_CLK_ID_WSA_CORE_TX_MCLK 0x314 +/* Clock ID for WSA core TX 2X MCLK */ +#define Q6PRM_LPASS_CLK_ID_WSA_CORE_TX_2X_MCLK 0x315 +/* Clock ID for WSA2 core TX MCLK */ +#define Q6PRM_LPASS_CLK_ID_WSA2_CORE_TX_MCLK 0x316 +/* Clock ID for WSA2 core TX 2X MCLK */ +#define Q6PRM_LPASS_CLK_ID_WSA2_CORE_TX_2X_MCLK 0x317 +/* Clock ID for RX CORE MCLK2 2X MCLK */ +#define Q6PRM_LPASS_CLK_ID_RX_CORE_MCLK2_2X_MCLK 0x318 + #define Q6PRM_LPASS_CLK_SRC_INTERNAL 1 #define Q6PRM_LPASS_CLK_ROOT_DEFAULT 0 #define Q6PRM_HW_CORE_ID_LPASS 1 -- 2.21.0

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Re: [PATCH] ASoC: core: remove setting platform_max in kcontrol macros
by Mark Brown 17 Aug '22

17 Aug '22

On Tue, 16 Aug 2022 18:21:29 +0100, Srinivas Kandagatla wrote: > platform_max should not be set by the driver, its intended for machine drivers > or DT to override the max value for platform specific reasons. > > So remove setting this from Kcontrol macros. > > Setting this to max in these macros would limit the range when min > value is less then zero. > > [...] Applied to https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next Thanks! [1/1] ASoC: core: remove setting platform_max in kcontrol macros commit: 26bdcc4ba12351642cd94339aa6996f96434dd47 All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark

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Re: [PATCH 0/4] ALSA: hda: Minor cleanups
by Cezary Rojewski 17 Aug '22

17 Aug '22

On 2022-08-17 3:23 PM, Cezary Rojewski wrote: > On 2022-08-17 3:11 PM, Amadeusz Sławiński wrote: >> First two patches are small refactoring of code to use poll macros >> instead of open coding register checks. >> Following two patches remove unused defines from code. >> >> Amadeusz Sławiński (4): >> ALSA: hda: Move stream-register polling macros >> ALSA: hda: Rework snd_hdac_stream_reset() to use macros >> ALSA: hda: Remove unused MAX_PIN_CONFIGS constant >> ALSA: hda: Remove unused defines > > > Reviewed by me before sending, plus from the test results patchset > causes no regression in the avs-driver CI. > > Reviewed-by: Cezary Rojewski <cezary.rojewski(a)intel.com> RESEND. Patch was blocked so my tag did not make it to the list.

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[RESEND][PATCH] ALSA: info: Fix llseek return value when using callback
by Amadeusz Sławiński 17 Aug '22

17 Aug '22

When using callback there was a flow of ret = -EINVAL if (callback) { offset = callback(); goto out; } ... offset = some other value in case of no callback; ret = offset; out: return ret; which causes the snd_info_entry_llseek() to return -EINVAL when there is callback handler. Fix this by setting "ret" directly to callback return value before jumping to "out". 73029e0ff18d ("ALSA: info - Implement common llseek for binary mode") Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> --- sound/core/info.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/core/info.c b/sound/core/info.c index b8058b341178..0b2f04dcb589 100644 --- a/sound/core/info.c +++ b/sound/core/info.c @@ -111,9 +111,9 @@ static loff_t snd_info_entry_llseek(struct file *file, loff_t offset, int orig) entry = data->entry; mutex_lock(&entry->access); if (entry->c.ops->llseek) { - offset = entry->c.ops->llseek(entry, - data->file_private_data, - file, offset, orig); + ret = entry->c.ops->llseek(entry, + data->file_private_data, + file, offset, orig); goto out; } -- 2.25.1

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[PATCH 0/4] ALSA: hda: Minor cleanups
by Amadeusz Sławiński 17 Aug '22

17 Aug '22

First two patches are small refactoring of code to use poll macros instead of open coding register checks. Following two patches remove unused defines from code. Amadeusz Sławiński (4): ALSA: hda: Move stream-register polling macros ALSA: hda: Rework snd_hdac_stream_reset() to use macros ALSA: hda: Remove unused MAX_PIN_CONFIGS constant ALSA: hda: Remove unused defines include/sound/hdaudio.h | 6 ++++++ include/sound/hdaudio_ext.h | 6 ------ sound/hda/hdac_stream.c | 27 +++++++-------------------- sound/pci/hda/hda_intel.c | 7 ------- sound/pci/hda/hda_sysfs.c | 2 -- 5 files changed, 13 insertions(+), 35 deletions(-) -- 2.25.1

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[PATCH] ASoC: Variable type completion
by Xin Gao 17 Aug '22

17 Aug '22

'unsigned int' is better than 'unsigned'. Signed-off-by: Xin Gao <gaoxin(a)cdjrlc.com> --- sound/soc/soc-ac97.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/soc-ac97.c b/sound/soc/soc-ac97.c index 5f49e3dec3fc..32c5be61e2ec 100644 --- a/sound/soc/soc-ac97.c +++ b/sound/soc/soc-ac97.c @@ -57,7 +57,7 @@ static inline struct snd_soc_component *gpio_to_component(struct gpio_chip *chip return gpio_priv->component; } -static int snd_soc_ac97_gpio_request(struct gpio_chip *chip, unsigned offset) +static int snd_soc_ac97_gpio_request(struct gpio_chip *chip, unsigned int offset) { if (offset >= AC97_NUM_GPIOS) return -EINVAL; @@ -66,7 +66,7 @@ static int snd_soc_ac97_gpio_request(struct gpio_chip *chip, unsigned offset) } static int snd_soc_ac97_gpio_direction_in(struct gpio_chip *chip, - unsigned offset) + unsigned int offset) { struct snd_soc_component *component = gpio_to_component(chip); @@ -75,7 +75,7 @@ static int snd_soc_ac97_gpio_direction_in(struct gpio_chip *chip, 1 << offset, 1 << offset); } -static int snd_soc_ac97_gpio_get(struct gpio_chip *chip, unsigned offset) +static int snd_soc_ac97_gpio_get(struct gpio_chip *chip, unsigned int offset) { struct snd_soc_component *component = gpio_to_component(chip); int ret; @@ -88,7 +88,7 @@ static int snd_soc_ac97_gpio_get(struct gpio_chip *chip, unsigned offset) return !!(ret & (1 << offset)); } -static void snd_soc_ac97_gpio_set(struct gpio_chip *chip, unsigned offset, +static void snd_soc_ac97_gpio_set(struct gpio_chip *chip, unsigned int offset, int value) { struct snd_ac97_gpio_priv *gpio_priv = gpiochip_get_data(chip); -- 2.30.2

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[PATCH] ASoC: qcom: sm8250: add missing module owner
by Srinivas Kandagatla 17 Aug '22

17 Aug '22

Add missing module owner to able to build and load this driver as module. Fixes: aa2e2785545a ("ASoC: qcom: sm8250: add sound card qrb5165-rb5 support") Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/qcom/sm8250.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/qcom/sm8250.c b/sound/soc/qcom/sm8250.c index ce4a5713386a..98a2fde9e004 100644 --- a/sound/soc/qcom/sm8250.c +++ b/sound/soc/qcom/sm8250.c @@ -270,6 +270,7 @@ static int sm8250_platform_probe(struct platform_device *pdev) if (!card) return -ENOMEM; + card->owner = THIS_MODULE; /* Allocate the private data */ data = devm_kzalloc(dev, sizeof(*data), GFP_KERNEL); if (!data) -- 2.21.0

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Channel mapping randomly change on usb soundcard and sucpicious dmesg log
by GitHub issues - opened 17 Aug '22

17 Aug '22

alsa-project/alsa-lib issue #261 was opened from sprudent: Hello erveryone, I am getting random channel swap when using speaker test to output sequencially a pink noise on every channel of the soundcard. I tested two distinct usb soundcard getting the same issue. I could identify a suspicious log showing at the same time the channels swapped. Tested and reproduced on a recent archlinux image (see system information). ``` [Aug16 18:26] xhci_hcd 0000:00:14.0: ERROR Transfer event TRB DMA ptr not part of current TD ep_index 11 comp_code 1 [ +0.000134] xhci_hcd 0000:00:14.0: Looking for event-dma 0000000014ed98e0 trb-start 0000000014ed98f0 trb-end 0000000014ed98f0 seg-start 0000000014ed9000 seg-end 0000000014ed9ff0 ``` ### system information ``` uname -r 5.18.7-arch1-1 ``` ``` aplay -l **** List of PLAYBACK Hardware Devices **** card 0: Audio [Intel HDMI/DP LPE Audio], device 0: HdmiLpeAudio [Intel HDMI/DP LPE Audi] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: Audio [Intel HDMI/DP LPE Audio], device 1: HdmiLpeAudio [Intel HDMI/DP LPE Audi] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: Audio [Intel HDMI/DP LPE Audio], device 2: HdmiLpeAudio [Intel HDMI/DP LPE Audi] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: ICUSBAUDIO7D [ICUSBAUDIO7D], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 card 2: bytchtes8316 [bytcht-es8316], device 0: Audio (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 2: bytchtes8316 [bytcht-es8316], device 1: Deep-Buffer Audio (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 ``` [alsa-info.txt](https://github.com/alsa-project/alsa-lib/files/9358483/alsa-… ### Test with dmix #### Description First test was made with alsa dmix plugin. Multiple processes running in parrallel and outputing to dmix pcm device. #### Alsa configuration ``` pcm.play_8 { @args [ CARD ] @args.CARD { type string } type dmix ipc_key 5678 slave { pcm $CARD channels 8 period_time 0 period_size 1024 buffer_size 4096 rate 44100 } bindings { 0 0 1 1 2 2 3 3 4 4 5 5 6 6 7 7 } } ``` #### Step to reproduce Run this script to simulate multiple processes writing to dmix device at the same time ``` for i in $(seq 20) do speaker-test -r44100 -D"play_8:\"hw:1\"" -c8 -t pink & done ``` Then on another tty, start a speaker-test command outputing a sine wave in order to identify channel mapping by correlating speaker test output with audio output. `speaker-test -r44100 -D"play_8:\"hw:1\"" -c8 -t sine` ``` 0 - Front Left 4 - Front Center 1 - Front Right 3 - Side Right 7 - Side Right 2 - Side Left 6 - Side Left 5 - LFE Time per period = 21.998519 ... ``` #### Expected result Audio output to the same channels during all the test #### Actual result Random audio channel swap correlated with dmesg log mentioned above. ### Second test with direct output #### Alsa configuration No specific configuration required #### Step to reproduce Run speaker-following test command and wait for channel swap `speaker-test -r44100 -Dsurround71:ICUSBAUDIO7D -c8` #### Expected result Audio output to the same channels during all the test #### Actual result Random audio channel swap correlated with dmesg log mentioned above. The chanel swap happen far less often than with dmix. ### Interpretation The DMA transfer ring buffer get messed up after a while (see dmesg log) and interleaved audio frames get shifted in soundcard buffer. If I unplug/replug usb soundcard and try again, channel mapping goes back to the correct order. My alsa, kernel and usb driver knowledge is limited, so I come to you for a little help. Tell me if you need more information to reproduce. Thank's, Sylvain Issue URL : https://github.com/alsa-project/alsa-lib/issues/261 Repository URL: https://github.com/alsa-project/alsa-lib

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