February 2020 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 0/2 v4] ALSA: pcm: anonymous dup implementation
by Jaroslav Kysela 12 Jun '20

12 Jun '20

This patchset contains the anonymous dup implementation with permissions checking for the ALSA's PCM interface in kernel to enable the restricted DMA sound buffer sharing for the restricted tasks. The code was tested through qemu and it seems to be pretty stable. The initial tinyalsa implementation can be found here: https://github.com/perexg/tinyalsa/commits/anondup v2 of the patches: - change clone parameter to subdevice number for the pcm attach - change SNDRV_PCM_PERM_MAX to SNDRV_PCM_PERM_MASK - the tinyalsa implementation was a little updated (restructured) v3 of the patches: - group integer declarations in snd_pcm_anonymous_dup() - replaced substream->pcm with pcm in snd_pcm_anonymous_dup() - added SNDRV_PCM_PERM_RW check for read/write/readv/writev syscalls v4 of the patches: - more simple restriction control (only two modes - full/buffer) - the tinyalsa implementation follows this change Cc: Phil Burk <philburk(a)google.com> Cc: Zach Riggle <riggle(a)google.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Leo Yan <leo.yan(a)linaro.org> Cc: Baolin Wang <baolin.wang(a)linaro.org> Jaroslav Kysela (2): ALSA: pcm: implement the anonymous dup (inode file descriptor) ALSA: pcm: implement the mmap buffer mode for the anonymous dup include/sound/pcm.h | 10 +++-- include/uapi/sound/asound.h | 6 ++- sound/core/oss/pcm_oss.c | 2 +- sound/core/pcm.c | 13 +++--- sound/core/pcm_compat.c | 1 + sound/core/pcm_native.c | 97 ++++++++++++++++++++++++++++++++++++++++++--- 6 files changed, 110 insertions(+), 19 deletions(-) -- 2.13.6

4 17

[alsa-devel] [PATCH] sound/soc: only first codec is master in multicodec setup
by Johannes Krude 11 May '20

11 May '20

When using multiple codecs, at most one codec should generate the master clock. All codecs except the first are therefore configured for slave mode. Before this patch all codecs in a multicodec setup had to be slaves. This is needed when e.g., connecting multiple sound hats for simultaneous playback to the raspberry pi I2S output and one of the sound hats generates the I2S clocks (https://github.com/raspberrypi/linux/pull/3337). I checked the raspberry pi kernel tree for multicodec usage with `fgrep num_codecs -R sound/soc/` and verified that all existing multicodec drivers with hardcoded format have indeed configured all codecs for slave mode. Doing a similar check on the current for-5.5 tree is more difficult since .num_codecs is now hidden behind a preprocessor macro. Signed-off-by: Johannes Krude <johannes(a)krude.de> --- sound/soc/soc-core.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 058e038df..cb8952527 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1688,6 +1688,14 @@ int snd_soc_runtime_set_dai_fmt(struct snd_soc_pcm_runtime *rtd, for_each_rtd_codec_dai(rtd, i, codec_dai) { + unsigned int codec_dai_fmt = dai_fmt; + - ret = snd_soc_dai_set_fmt(codec_dai, dai_fmt); + // there can only be one master when using multiple codecs + if (i && (codec_dai_fmt & SND_SOC_DAIFMT_MASTER_MASK)) { + codec_dai_fmt &= ~SND_SOC_DAIFMT_MASTER_MASK; + codec_dai_fmt |= SND_SOC_DAIFMT_CBS_CFS; + } + + ret = snd_soc_dai_set_fmt(codec_dai, codec_dai_fmt); if (ret != 0 && ret != -ENOTSUPP) { dev_warn(codec_dai->dev, "ASoC: Failed to set DAI format: %d\n", ret); -- 2.17.1

2 2

snd_dice: Clicking artifacts with TC StudioKonnekt 48
by Mathias Buhr 10 May '20

10 May '20

Hi, thanks to this group, both of my firewire interfaces are (almost) working! Big thank you! While the TC Konnekt 24D works fine (playback and recording), the StudioKonnekt 48 produces clicking artifacts during playback when using snd_dice. This interface is working flawless on Windows and on a Jack/FFADO combination. This artifact occurs in all use cases (alsa via aplay, pulseaudio and jack) and does not seem to be in recorded streams. After recording the playback with another device, it looks like the level of the artifact is scaling with the signal and its interval is interestingly 256ms. Is there anything I can do to further debug this issue? Capture firewire packets? I would love to get this device fully working. I'm using kernel version 5.5.4 but this issue has been there for a long time. Thanks, Mathias

3 9

[alsa-devel] [PATCH v4 00/12] ASoC: Add support to WCD9340/WCD9341 codec
by Srinivas Kandagatla 08 May '20

08 May '20

This patchset adds support to Qualcomm WCD9340/WCD9341 Codec which is a standalone Hi-Fi audio codec IC. This codec supports both I2S/I2C and SLIMbus audio interfaces. On slimbus interface it supports two data lanes; 16 Tx ports and 8 Rx ports. It has Five DACs and seven dedicated interpolators, Multibutton headset control (MBHC), Active noise cancellation, Sidetone paths, MAD (mic activity detection) and codec processing engine. It supports Class-H differential earpiece out and stereo single ended headphones out. This codec also has integrated SoundWire controller. Patchset for this is already sent for review at https://patchwork.kernel.org/cover/11185769/ This patchset has been tested on SDM845 based DragonBoard DB845c and Lenovo Yoga C630 laptop with WSA881x smart speaker amplifiers via soundwire and 4 DMICs. gpio controller patch does not have any link dependency, it can go by its own. Most of the code in this driver is rework of Qualcomm downstream drivers used in Andriod. Credits to Banajit Goswami and Patrick Lai's Team. Thanks, srini Changes since v3: - Added new SLIMBus common bindings suggested by Rob. - renamed microvolts to microvolt in bindings as suggested by Rob - dumped pinctrl driver and added new simple gpio driver - few cosmetic cleanups in mfd driver. Srinivas Kandagatla (12): dt-bindings: SLIMBus: add slim devices optional properties ASoC: dt-bindings: add dt bindings for WCD9340/WCD9341 audio codec mfd: wcd934x: add support to wcd9340/wcd9341 codec ASoC: wcd934x: add support to wcd9340/wcd9341 codec ASoC: wcd934x: add basic controls ASoC: wcd934x: add playback dapm widgets ASoC: wcd934x: add capture dapm widgets ASoC: wcd934x: add audio routings dt-bindings: gpio: wcd934x: Add bindings for gpio gpio: wcd934x: Add support to wcd934x gpio controller ASoC: qcom: dt-bindings: Add compatible for DB845c and Lenovo Yoga ASoC: qcom: sdm845: add support to DB845c and Lenovo Yoga .../bindings/gpio/qcom,wcd934x-gpio.yaml | 47 + .../devicetree/bindings/slimbus/bus.txt | 10 + .../devicetree/bindings/sound/qcom,sdm845.txt | 5 +- .../bindings/sound/qcom,wcd934x.yaml | 163 + drivers/gpio/Kconfig | 8 + drivers/gpio/Makefile | 1 + drivers/gpio/gpio-wcd934x.c | 104 + drivers/mfd/Kconfig | 12 + drivers/mfd/Makefile | 1 + drivers/mfd/wcd934x.c | 306 + include/linux/mfd/wcd934x/registers.h | 531 ++ include/linux/mfd/wcd934x/wcd934x.h | 31 + sound/soc/codecs/Kconfig | 10 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wcd934x.c | 5084 +++++++++++++++++ sound/soc/qcom/sdm845.c | 86 +- 16 files changed, 6399 insertions(+), 2 deletions(-) create mode 100644 Documentation/devicetree/bindings/gpio/qcom,wcd934x-gpio.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,wcd934x.yaml create mode 100644 drivers/gpio/gpio-wcd934x.c create mode 100644 drivers/mfd/wcd934x.c create mode 100644 include/linux/mfd/wcd934x/registers.h create mode 100644 include/linux/mfd/wcd934x/wcd934x.h create mode 100644 sound/soc/codecs/wcd934x.c -- 2.21.0

4 17

[alsa-devel] [RESEND PATCH] ASoC: wm8524: Add support S32_LE
by Shengjiu Wang 05 May '20

05 May '20

Allow 32bit sample with this codec. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/codecs/wm8524.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8524.c b/sound/soc/codecs/wm8524.c index 91e3d1570c45..4e9ab542f648 100644 --- a/sound/soc/codecs/wm8524.c +++ b/sound/soc/codecs/wm8524.c @@ -159,7 +159,9 @@ static int wm8524_mute_stream(struct snd_soc_dai *dai, int mute, int stream) #define WM8524_RATES SNDRV_PCM_RATE_8000_192000 -#define WM8524_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE) +#define WM8524_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S24_LE |\ + SNDRV_PCM_FMTBIT_S32_LE) static const struct snd_soc_dai_ops wm8524_dai_ops = { .startup = wm8524_startup, -- 2.21.0

2 1

[alsa-devel] [PATCH] ASoC: stm32: sai: fix master clock management
by Olivier Moysan 16 Apr '20

16 Apr '20

When master clock is used, master clock rate is set exclusively. Parent clocks of master clock cannot be changed after a call to clk_set_rate_exclusive(). So the parent clock of SAI kernel clock must be set before. Ensure also that exclusive rate operations are balanced in STM32 SAI driver. Signed-off-by: Olivier Moysan <olivier.moysan(a)st.com> --- sound/soc/stm/stm32_sai_sub.c | 64 +++++++++++++++++++++++++++++++------------ 1 file changed, 47 insertions(+), 17 deletions(-) diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index 3dd54bc54fa1..77aa38183955 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -70,6 +70,7 @@ #define SAI_IEC60958_STATUS_BYTES 24 #define SAI_MCLK_NAME_LEN 32 +#define SAI_RATE_11K 11025 /** * struct stm32_sai_sub_data - private data of SAI sub block (block A or B) @@ -311,6 +312,25 @@ static int stm32_sai_set_clk_div(struct stm32_sai_sub_data *sai, return ret; } +static int stm32_sai_set_parent_clock(struct stm32_sai_sub_data *sai, + unsigned int rate) +{ + struct platform_device *pdev = sai->pdev; + struct clk *parent_clk = sai->pdata->clk_x8k; + int ret; + + if (!(rate % SAI_RATE_11K)) + parent_clk = sai->pdata->clk_x11k; + + ret = clk_set_parent(sai->sai_ck, parent_clk); + if (ret) + dev_err(&pdev->dev, " Error %d setting sai_ck parent clock. %s", + ret, ret == -EBUSY ? + "Active stream rates conflict\n" : "\n"); + + return ret; +} + static long stm32_sai_mclk_round_rate(struct clk_hw *hw, unsigned long rate, unsigned long *prate) { @@ -492,25 +512,29 @@ static int stm32_sai_set_sysclk(struct snd_soc_dai *cpu_dai, struct stm32_sai_sub_data *sai = snd_soc_dai_get_drvdata(cpu_dai); int ret; - if (dir == SND_SOC_CLOCK_OUT) { + if (dir == SND_SOC_CLOCK_OUT && sai->sai_mclk) { ret = regmap_update_bits(sai->regmap, STM_SAI_CR1_REGX, SAI_XCR1_NODIV, (unsigned int)~SAI_XCR1_NODIV); if (ret < 0) return ret; - dev_dbg(cpu_dai->dev, "SAI MCLK frequency is %uHz\n", freq); - sai->mclk_rate = freq; + /* If master clock is used, set parent clock now */ + ret = stm32_sai_set_parent_clock(sai, freq); + if (ret) + return ret; - if (sai->sai_mclk) { - ret = clk_set_rate_exclusive(sai->sai_mclk, - sai->mclk_rate); - if (ret) { - dev_err(cpu_dai->dev, - "Could not set mclk rate\n"); - return ret; - } + ret = clk_set_rate_exclusive(sai->sai_mclk, freq); + if (ret) { + dev_err(cpu_dai->dev, + ret == -EBUSY ? + "Active streams have incompatible rates" : + "Could not set mclk rate\n"); + return ret; } + + dev_dbg(cpu_dai->dev, "SAI MCLK frequency is %uHz\n", freq); + sai->mclk_rate = freq; } return 0; @@ -917,11 +941,13 @@ static int stm32_sai_configure_clock(struct snd_soc_dai *cpu_dai, int div = 0, cr1 = 0; int sai_clk_rate, mclk_ratio, den; unsigned int rate = params_rate(params); + int ret; - if (!(rate % 11025)) - clk_set_parent(sai->sai_ck, sai->pdata->clk_x11k); - else - clk_set_parent(sai->sai_ck, sai->pdata->clk_x8k); + if (!sai->sai_mclk) { + ret = stm32_sai_set_parent_clock(sai, rate); + if (ret) + return ret; + } sai_clk_rate = clk_get_rate(sai->sai_ck); if (STM_SAI_IS_F4(sai->pdata)) { @@ -1080,9 +1106,13 @@ static void stm32_sai_shutdown(struct snd_pcm_substream *substream, regmap_update_bits(sai->regmap, STM_SAI_CR1_REGX, SAI_XCR1_NODIV, SAI_XCR1_NODIV); - clk_disable_unprepare(sai->sai_ck); + /* Release mclk rate only if rate was actually set */ + if (sai->mclk_rate) { + clk_rate_exclusive_put(sai->sai_mclk); + sai->mclk_rate = 0; + } - clk_rate_exclusive_put(sai->sai_mclk); + clk_disable_unprepare(sai->sai_ck); spin_lock_irqsave(&sai->irq_lock, flags); sai->substream = NULL; -- 2.7.4

2 3

[alsa-devel] [PATCH] ASoC: pcm: fix error handling when try_module_get() fails.
by Ranjani Sridharan 16 Apr '20

16 Apr '20

Handle error before returning when try_module_get() fails to prevent inconsistent mutex lock/unlock. Fixes: 52034add7 (ASoC: pcm: update module refcount if module_get_upon_open is set) Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> --- sound/soc/soc-pcm.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index d21247546f7f..be80a12fba27 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -518,8 +518,10 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) continue; if (component->driver->module_get_upon_open && - !try_module_get(component->dev->driver->owner)) - return -ENODEV; + !try_module_get(component->dev->driver->owner)) { + ret = -ENODEV; + goto module_err; + } ret = component->driver->ops->open(substream); if (ret < 0) { @@ -636,7 +638,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) component_err: soc_pcm_components_close(substream, component); - +module_err: if (cpu_dai->driver->ops->shutdown) cpu_dai->driver->ops->shutdown(substream, cpu_dai); out: -- 2.17.1

2 5

[alsa-devel] [PATCH] ASoC: wcd9335: Fix missing regmap requirement
by Marc Gonzalez 16 Apr '20

16 Apr '20

wcd9335.c: undefined reference to 'devm_regmap_add_irq_chip' Signed-off-by: Marc Gonzalez <marc.w.gonzalez(a)free.fr> --- sound/soc/codecs/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 419142111b6d..33778dc99108 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1162,6 +1162,7 @@ config SND_SOC_WCD9335 tristate "WCD9335 Codec" depends on SLIMBUS select REGMAP_SLIMBUS + select REGMAP_IRQ help The WCD9335 is a standalone Hi-Fi audio CODEC IC, supports Qualcomm Technologies, Inc. (QTI) multimedia solutions, -- 2.17.1

3 4

[alsa-devel] [PATCH] alsactl: avoid needless wakeups in monitor loop.
by Zev Weiss 14 Apr '20

14 Apr '20

The timeout wasn't really being used for anything; disabling it should reduce idle energy consumption slightly. Signed-off-by: Zev Weiss <zev(a)bewilderbeest.net> --- alsactl/monitor.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/alsactl/monitor.c b/alsactl/monitor.c index 6b090e4f5c92..6d9328d899a4 100644 --- a/alsactl/monitor.c +++ b/alsactl/monitor.c @@ -340,7 +340,7 @@ static int run_dispatcher(int epfd, int sigfd, int infd, struct list_head *srcs, int count; int i; - count = epoll_wait(epfd, epev, max_ev_count, 200); + count = epoll_wait(epfd, epev, max_ev_count, -1); if (count < 0) { err = count; break; -- 2.23.0

3 3

[alsa-devel] [PATCH] alsactl: don't exit on EINTR from epoll_wait().
by Zev Weiss 14 Apr '20

14 Apr '20

Previously, things like attaching strace to a running 'alsactl monitor' process would cause it to exit. Signed-off-by: Zev Weiss <zev(a)bewilderbeest.net> --- alsactl/monitor.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/alsactl/monitor.c b/alsactl/monitor.c index 6b090e4f5c92..cf4167bee165 100644 --- a/alsactl/monitor.c +++ b/alsactl/monitor.c @@ -342,6 +342,8 @@ static int run_dispatcher(int epfd, int sigfd, int infd, struct list_head *srcs, count = epoll_wait(epfd, epev, max_ev_count, 200); if (count < 0) { + if (errno == EINTR) + continue; err = count; break; } -- 2.23.0

3 2