March 2018 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 1/5] ASoC: samsung: odroid: Fix 32000 sample rate handling
by Sylwester Nawrocki 19 Mar '18

19 Mar '18

In case of sample rates lower than 44100 currently there is too low MCLK frequency set for the CODEC. Playback fails with following errors: $ speaker-test -c2 -t sine -f 1500 -l2 -r 32000 Sine wave rate is 1500.0000Hz Rate set to 32000Hz (requested 32000Hz) Buffer size range from 128 to 131072 Period size range from 64 to 65536 Using max buffer size 131072 Periods = 4 Unable to set hw params for playback: Invalid argument Setting of hwparams failed: Invalid argument [ 497.883700] max98090 1-0010: Invalid master clock frequency To fix this the I2S root clock's frequency is increased, depending on sampling rate. Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- sound/soc/samsung/odroid.c | 11 +++++++---- 1 file changed, 7 insertions(+), 4 deletions(-) diff --git a/sound/soc/samsung/odroid.c b/sound/soc/samsung/odroid.c index 92d750806d1d..e698e67f5f84 100644 --- a/sound/soc/samsung/odroid.c +++ b/sound/soc/samsung/odroid.c @@ -36,23 +36,26 @@ static int odroid_card_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct odroid_priv *priv = snd_soc_card_get_drvdata(rtd->card); - unsigned int pll_freq, rclk_freq; + unsigned int pll_freq, rclk_freq, rfs; int ret; switch (params_rate(params)) { - case 32000: case 64000: - pll_freq = 131072006U; + pll_freq = 196608001U; + rfs = 384; break; case 44100: case 88200: case 176400: pll_freq = 180633609U; + rfs = 512; break; + case 32000: case 48000: case 96000: case 192000: pll_freq = 196608001U; + rfs = 512; break; default: return -EINVAL; @@ -67,7 +70,7 @@ static int odroid_card_hw_params(struct snd_pcm_substream *substream, * frequency values due to the EPLL output frequency not being exact * multiple of the audio sampling rate. */ - rclk_freq = params_rate(params) * 256 + 1; + rclk_freq = params_rate(params) * rfs + 1; ret = clk_set_rate(priv->sclk_i2s, rclk_freq); if (ret < 0) -- 2.14.2

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[alsa-devel] [PATCH v2 0/2] TDA7419 audio processor driver
by Matt Porter 19 Mar '18

19 Mar '18

This series adds an ASoC component driver for the ST TDA7419 audio processor which is commonly used in automotive audio applications. The datasheet can be found at http://www.st.com/resource/en/datasheet/tda7419.pdf Changes since v1: * add supply and mute gpios to binding * remove writeable register, already the default * add "Switch" to all switch controls * move register default reset to probe and comment * rework part-specific volume handling to handle register power-on default values * remove custom register init values * add audio route pins to binding * add DAPM support Matt Porter (2): ASoC: add tda7419 audio processor binding ASoC: add tda7419 audio processor driver .../devicetree/bindings/sound/tda7419.txt | 38 ++ sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tda7419.c | 654 +++++++++++++++++++++ 4 files changed, 700 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/tda7419.txt create mode 100644 sound/soc/codecs/tda7419.c -- 2.11.0

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[alsa-devel] [PATCH v3 0/2] ASoC: Add support for DAC PCM1789
by Mylène Josserand 19 Mar '18

19 Mar '18

Hello everyone, The current series is the 3rd version to add the support of Texas Instrument's DAC PCM1789. This DAC is very minimalist and does not have many registers. It is important to notice that this DAC needs to always have clocks enabled (even without any data) otherwise it will be in a "desynchronized" state and can not send data correctly. This issue has been solved by performing a reset each time a sound is played. This reset can produce a "pop" noise. Depending on your DAI, you will need to provide and enable the MCLK to be able to communicate with this codec throught i2c. Changes since v2: - Update SPDX headers according to Mark Brown's review. - Update the dt bindings according to Rob Herring's review. Changes since v1: - Create a new file to support pcm1789 instead of converting the existing pcm179x driver. All the patches are merged into one patch. - Update the code to use gpiod for the reset. - Add some fixes according to Thomas Petazzoni's reviews - Create a new patch to add device-tree bindings for this new DAC. Thank you in advance for any review. Best regards, Mylène Mylène Josserand (2): ASoC: codecs: Add support for PCM1789 ASoC: Add bindings for PCM1789 .../devicetree/bindings/sound/pcm1789.txt | 22 ++ sound/soc/codecs/Kconfig | 12 + sound/soc/codecs/Makefile | 4 + sound/soc/codecs/pcm1789-i2c.c | 62 +++++ sound/soc/codecs/pcm1789.c | 274 +++++++++++++++++++++ sound/soc/codecs/pcm1789.h | 17 ++ 6 files changed, 391 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/pcm1789.txt create mode 100644 sound/soc/codecs/pcm1789-i2c.c create mode 100644 sound/soc/codecs/pcm1789.c create mode 100644 sound/soc/codecs/pcm1789.h -- 2.11.0

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[alsa-devel] [PATCH 0/2] add syscon property to binding document
by Katsuhiro Suzuki 19 Mar '18

19 Mar '18

This patch adds syscon property for specifying soc-glue core. The soc-glue core is used for changing the state of S/PDIF signal output pin to signal output state or Hi-Z state. After resetting of SoC Hi-Z state is selected. Currently, we set this mode in local changed U-Boot or some bootloaders fixedly. This patch removes dependencies of bootloader from UniPhier audio system driver. Katsuhiro Suzuki (2): ASoC: uniphier: add syscon property to binding document ASoC: uniphier: add syscon property for UniPhier sound system .../devicetree/bindings/sound/uniphier,aio.txt | 8 ++++++++ sound/soc/uniphier/aio-core.c | 21 +++++++++++++++++++++ sound/soc/uniphier/aio-cpu.c | 11 +++++++++++ sound/soc/uniphier/aio-reg.h | 3 +++ sound/soc/uniphier/aio.h | 2 ++ 5 files changed, 45 insertions(+) -- 2.16.2

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[alsa-devel] [PATCH -next] ASoC: da7210: Use devm_snd_soc_register_component()
by Wei Yongjun 19 Mar '18

19 Mar '18

Since the remove callback is removed, the snd_soc_unregister_component() is missing when remove device. Using devm_snd_soc_register_component() instead of snd_soc_register_component(). Fixes: d06f33aed85c ("ASoC: da7210: replace codec to component") Signed-off-by: Wei Yongjun <weiyongjun1(a)huawei.com> --- sound/soc/codecs/da7210.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 07153be..a664111 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -1233,7 +1233,7 @@ static int da7210_i2c_probe(struct i2c_client *i2c, if (ret != 0) dev_warn(&i2c->dev, "Failed to apply regmap patch: %d\n", ret); - ret = snd_soc_register_component(&i2c->dev, + ret = devm_snd_soc_register_component(&i2c->dev, &soc_component_dev_da7210, &da7210_dai, 1); if (ret < 0) dev_err(&i2c->dev, "Failed to register component: %d\n", ret);

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[alsa-devel] [PATCH] ASoC: rsnd: Document R-Car M3-W support
by Geert Uytterhoeven 19 Mar '18

19 Mar '18

Document support for the sound modules in the Renesas M3-W (r8a7796) SoC. No driver update is needed. Signed-off-by: Geert Uytterhoeven <geert+renesas(a)glider.be> --- "renesas,rcar_sound-r8a7796" has been in use since quite a while, but the binding update was still missing. --- Documentation/devicetree/bindings/sound/renesas,rsnd.txt | 1 + 1 file changed, 1 insertion(+) diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt index 5bed9a5957729453..b86d790f630ff79a 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt @@ -351,6 +351,7 @@ Required properties: - "renesas,rcar_sound-r8a7793" (R-Car M2-N) - "renesas,rcar_sound-r8a7794" (R-Car E2) - "renesas,rcar_sound-r8a7795" (R-Car H3) + - "renesas,rcar_sound-r8a7796" (R-Car M3-W) - reg : Should contain the register physical address. required register is SRU/ADG/SSI if generation1 -- 2.7.4

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[alsa-devel] [PATCH 0/2] TDA7419 audio processor driver
by Matt Porter 18 Mar '18

18 Mar '18

This series adds an ASoC component driver for the ST TDA7419 audio processor which is commonly used in automotive audio applications. The datasheet can be found at http://www.st.com/resource/en/datasheet/tda7419.pdf Matt Porter (2): ASoC: add tda7419 audio processor binding ASoC: add tda7419 audio processor driver .../devicetree/bindings/sound/tda7419.txt | 15 + sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tda7419.c | 571 +++++++++++++++++++++ 4 files changed, 594 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/tda7419.txt create mode 100644 sound/soc/codecs/tda7419.c -- 2.11.0

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[alsa-devel] [PATCH 1/3] sound: usb: line6: Replace mdelay with msleep in line6_read_data
by Jia-Ju Bai 18 Mar '18

18 Mar '18

line6_read_data() is never called in atomic context. The call chains ending up at line6_read_data() are: [1] line6_read_data() <- line6_read_serial_number() -> pod_startup4() [2] line6_read_data() <- line6_read_serial_number() <- podhd_startup_workqueue() [3] line6_read_data() <- podhd_dev_start() <- podhd_startup_workqueue() [4] line6_read_data() <- line6_read_serial_number() <- toneport_init() <- line6_probe() [5] line6_read_data() <- toneport_init() <- line6_probe() pod_startup4() and podhd_startup_workqueue() are set in INIT_WORK(). line6_probe() is called in toneport_probe(), pod_probe(), podhd_probe() and variax_probe(), which are set as ".probe" in usb_driver structure. These functions are not called in atomic context. Despite never getting called from atomic context, line6_read_data() calls mdelay for busy wait. That is not necessary and can be replaced with msleep to avoid busy waiting. This is found by a static analysis tool named DCNS written by myself. Signed-off-by: Jia-Ju Bai <baijiaju1990(a)gmail.com> --- sound/usb/line6/driver.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/line6/driver.c b/sound/usb/line6/driver.c index 0ff5a7d..af99e73 100644 --- a/sound/usb/line6/driver.c +++ b/sound/usb/line6/driver.c @@ -356,7 +356,7 @@ int line6_read_data(struct usb_line6 *line6, unsigned address, void *data, /* Wait for data length. We'll get 0xff until length arrives. */ for (count = 0; count < LINE6_READ_WRITE_MAX_RETRIES; count++) { - mdelay(LINE6_READ_WRITE_STATUS_DELAY); + msleep(LINE6_READ_WRITE_STATUS_DELAY); ret = usb_control_msg(usbdev, usb_rcvctrlpipe(usbdev, 0), 0x67, USB_TYPE_VENDOR | USB_RECIP_DEVICE | -- 1.9.1

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[alsa-devel] [PATCH] ASoC: samsung: Mark unused Odroid compatibles as deprecated
by Krzysztof Kozlowski 18 Mar '18

18 Mar '18

Compatible for XU4 audio is not being used. Instead the board uses the same compatible as XU3. The devices are now just compatible so they should use the same value. Mark hardkernel,odroid-xu4-audio as being deprecated so in future could be removed to limit useless compatibles. Additionally add older compatibles appearing in the bindings. Signed-off-by: Krzysztof Kozlowski <krzk(a)kernel.org> --- Documentation/devicetree/bindings/sound/samsung,odroid.txt | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/samsung,odroid.txt b/Documentation/devicetree/bindings/sound/samsung,odroid.txt index f35a2694eb04..4a18d83456c3 100644 --- a/Documentation/devicetree/bindings/sound/samsung,odroid.txt +++ b/Documentation/devicetree/bindings/sound/samsung,odroid.txt @@ -3,7 +3,9 @@ Samsung Exynos Odroid XU3/XU4 audio complex with MAX98090 codec Required properties: - compatible - "hardkernel,odroid-xu3-audio" - for Odroid XU3 board, - "hardkernel,odroid-xu4-audio" - for Odroid XU4 board + "hardkernel,odroid-xu4-audio" - for Odroid XU4 board (deprecated) + "samsung,odroid-xu3-audio" - for Odroid XU4 board (deprecated) + "samsung,odroid-xu4-audio" - for Odroid XU4 board (deprecated) - model - the user-visible name of this sound complex - clocks - should contain entries matching clock names in the clock-names property -- 2.14.1

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[alsa-devel] [PATCH 3/3] sound: usb: line6: Replace GFP_ATOMIC with GFP_KERNEL in line6_alloc_sysex_buffer
by Jia-Ju Bai 18 Mar '18

18 Mar '18

line6_alloc_sysex_buffer() is never called in atomic context. The call chain ending up at line6_alloc_sysex_buffer() are: [1] line6_alloc_sysex_buffer() <- pod_alloc_sysex_buffer() <- pod_set_system_param_int() <- snd_pod_control_monitor_put() snd_pod_control_monitor_put() is only set as ".put" in snd_kcontrol_new structure "pod_control_monitor". And ->put() is called in snd_ctl_elem_write() (sound/core/control.c) snd_ctl_elem_write() are called in ctl_elem_write_user() and snd_ctl_elem_write_user(). These functions are not called in atomic context. Despite never getting called from atomic context, line6_alloc_sysex_buffer() calls kmalloc with GFP_ATOMIC, which waits busily for allocation. GFP_ATOMIC is not necessary and can be replaced with GFP_KERNEL to avoid busy waiting. This is found by a static analysis tool named DCNS written by myself. Signed-off-by: Jia-Ju Bai <baijiaju1990(a)gmail.com> --- sound/usb/line6/driver.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/line6/driver.c b/sound/usb/line6/driver.c index 0ff5a7d..70a509e 100644 --- a/sound/usb/line6/driver.c +++ b/sound/usb/line6/driver.c @@ -265,7 +265,7 @@ int line6_send_sysex_message(struct usb_line6 *line6, const char *buffer, char *line6_alloc_sysex_buffer(struct usb_line6 *line6, int code1, int code2, int size) { - char *buffer = kmalloc(size + SYSEX_EXTRA_SIZE, GFP_ATOMIC); + char *buffer = kmalloc(size + SYSEX_EXTRA_SIZE, GFP_KERNEL); if (!buffer) return NULL; -- 1.9.1

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