November 2019 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: bdw-rt5677: enable runtime channel merge
by Brent Lu 05 Nov '19

05 Nov '19

In the DAI link "Capture PCM", the FE DAI "Capture Pin" supports 4-channel capture but the BE DAI supports only 2-channel capture. To fix the channel mismatch, we need to enable the runtime channel merge for this DAI link. Signed-off-by: Brent Lu <brent.lu(a)intel.com> --- sound/soc/intel/boards/bdw-rt5677.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/boards/bdw-rt5677.c b/sound/soc/intel/boards/bdw-rt5677.c index 4a4d335..8d07038 100644 --- a/sound/soc/intel/boards/bdw-rt5677.c +++ b/sound/soc/intel/boards/bdw-rt5677.c @@ -273,6 +273,7 @@ static struct snd_soc_dai_link bdw_rt5677_dais[] = { }, .dpcm_capture = 1, .dpcm_playback = 1, + .dpcm_merged_chan = 1, SND_SOC_DAILINK_REG(fe, dummy, platform), }, -- 2.7.4

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[alsa-devel] amixer: "5%-" or "5%+" doesn't work with cset
by GitHub issues - opened 05 Nov '19

05 Nov '19

alsa-project/alsa-utils issue #32 was opened from MarioMey: I want to raise and low volume with keys. This works, this rise the volume 20%: `amixer set Master 20%+` And every time I run this command, it rise 20% more. But, when I want to use the card (because I need to do this): `amixer -c 0 cset name='Master Playback Volume' 20%+` It doesn't work. It seems that amixer takes only "20%" value and doesn't care about "+" or "-". I have Debian Buster, Mate desktop. ``` mario@debian:~/$ apt-cache showpkg alsa-tools Package: alsa-tools Versions: 1.1.7-1 (/var/lib/apt/lists/ftp.us.debian.org_debian_dists_buster_main_binary-amd64_Packages) Description Language: File: /var/lib/apt/lists/ftp.us.debian.org_debian_dists_buster_main_binary-amd64_Packages MD5: 4d64eace7e51fb1fc1a18ed780ee5e62 Description Language: en File: /var/lib/apt/lists/ftp.us.debian.org_debian_dists_buster_main_i18n_Translation-en MD5: 4d64eace7e51fb1fc1a18ed780ee5e62 Reverse Depends: ld10k1,alsa-tools multimedia-jack,alsa-tools Dependencies: 1.1.7-1 - libasound2 (2 1.0.16) libc6 (2 2.7) libgcc1 (2 1:3.0) libglib2.0-0 (2 2.12.0) libstdc++6 (2 5) as10k1 (0 (null)) as10k1 (0 (null)) Provides: 1.1.7-1 - as10k1 (= ) Reverse Provides: ``` Issue URL : https://github.com/alsa-project/alsa-utils/issues/32 Repository URL: https://github.com/alsa-project/alsa-utils

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[alsa-devel] tegra30 tdm support
by Ben Dooks 05 Nov '19

05 Nov '19

New series for TDM audio support on tegra30 v2: - moved edge-control and data-offset to the set_fmt callbacks - fixed dev_dbg in set_tdm callback - fixed dev_dbg message contents in set_tdm callback - changed fsync width to be permanently 1 clock v3: - cleanup fsync patch - fix rebase issue with tdm patch v4: - fix comment style issues - change tdm-a to data-offset 1

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[alsa-devel] [PATCH 20/62] gpio: gpio-madera: Use new GPIO_LINE_DIRECTION
by Matti Vaittinen 05 Nov '19

05 Nov '19

It's hard for occasional GPIO code reader/writer to know if values 0/1 equal to IN or OUT. Use defined GPIO_LINE_DIRECTION_IN and GPIO_LINE_DIRECTION_OUT to help them out. Signed-off-by: Matti Vaittinen <matti.vaittinen(a)fi.rohmeurope.com> --- drivers/gpio/gpio-madera.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/drivers/gpio/gpio-madera.c b/drivers/gpio/gpio-madera.c index 7086f8b5388f..8f38303fcbc4 100644 --- a/drivers/gpio/gpio-madera.c +++ b/drivers/gpio/gpio-madera.c @@ -34,7 +34,10 @@ static int madera_gpio_get_direction(struct gpio_chip *chip, if (ret < 0) return ret; - return !!(val & MADERA_GP1_DIR_MASK); + if (val & MADERA_GP1_DIR_MASK) + return GPIO_LINE_DIRECTION_IN; + + return GPIO_LINE_DIRECTION_OUT; } static int madera_gpio_direction_in(struct gpio_chip *chip, unsigned int offset) -- 2.21.0 -- Matti Vaittinen, Linux device drivers ROHM Semiconductors, Finland SWDC Kiviharjunlenkki 1E 90220 OULU FINLAND ~~~ "I don't think so," said Rene Descartes. Just then he vanished ~~~ Simon says - in Latin please. ~~~ "non cogito me" dixit Rene Descarte, deinde evanescavit ~~~ Thanks to Simon Glass for the translation =]

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[alsa-devel] [PATCH] ALSA: hda: Disable regmap internal locking
by Takashi Iwai 05 Nov '19

05 Nov '19

Since we apply the own mutex (bus->cmd_mutex) in HDA core side, the internal locking in regmap is superfluous. This patch adds the flag to indicate that. Also, an infamous side-effect by this change is that it disables the regmap debugfs, too, and this is seen rather good; the regmap debugfs isn't quite useful for HD-audio as it provides the very sparse registers and its debugfs access tends to lead to the way too high resource usages or sometimes hang up. So it'd be rather safe to disable it altogether. Link: https://lore.kernel.org/r/2029139028.10333037.1572874551626.JavaMail.zimbra… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/hda/hdac_regmap.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/hda/hdac_regmap.c b/sound/hda/hdac_regmap.c index 286361ecd640..906b1e20bae0 100644 --- a/sound/hda/hdac_regmap.c +++ b/sound/hda/hdac_regmap.c @@ -363,6 +363,7 @@ static const struct regmap_config hda_regmap_cfg = { .reg_write = hda_reg_write, .use_single_read = true, .use_single_write = true, + .disable_locking = true, }; /** -- 2.16.4

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[alsa-devel] [PATCH v2 resend 00/18] ASoC: soc-core cleanup - step 4
by Kuninori Morimoto 05 Nov '19

05 Nov '19

Hi Mark These are resend of soc-core cleanup step4. These are based on mark/for-5.5 branch. I already got report from Intel that these had no issue on Intel CI machine. Kuninori Morimoto (18): ASoC: soc-core: remove unneeded snd_soc_tplg_component_remove() ASoC: soc-core: move soc_init_dai_link() ASoC: soc-core: tidyup soc_init_dai_link() ASoC: soc-core: remove duplicated soc_is_dai_link_bound() ASoC: soc-core: call soc_bind_dai_link() under snd_soc_add_dai_link() ASoC: soc-core: add soc_unbind_dai_link() ASoC: soc-core: move snd_soc_lookup_component() ASoC: soc-core: add snd_soc_del_component_unlocked() ASoC: soc-core: remove snd_soc_component_add/del() ASoC: soc-core: use snd_soc_lookup_component() at snd_soc_unregister_component() ASoC: soc-core: move snd_soc_register_dai() ASoC: soc-core: move snd_soc_unregister_dais() ASoC: soc-core: add snd_soc_unregister_dai() ASoC: soc-core: have legacy_dai_naming at snd_soc_register_dai() ASoC: soc-core: don't call snd_soc_dapm_new_dai_widgets() at snd_soc_register_dai() ASoC: soc-core: call snd_soc_register_dai() from snd_soc_register_dais() ASoC: soc-core: remove topology specific operation ASoC: soc.h: dobj is used only when SND_SOC_TOPOLOGY include/sound/soc.h | 15 +- sound/soc/soc-core.c | 536 +++++++++++++++++++++-------------------------- sound/soc/soc-topology.c | 17 +- 3 files changed, 263 insertions(+), 305 deletions(-) -- 2.7.4

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Re: [alsa-devel] [PATCH] ASoC: wm8524: Add support S32_LE
by S.j. Wang 05 Nov '19

05 Nov '19

Hi > On Fri, Nov 01, 2019 at 06:34:54AM +0000, S.j. Wang wrote: > > 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) > > > > The device doesn't actually support 32bit though, I guess it will ignore the > extra LSBs so it should work. But is that really supporting 32 bit? > I also think it may ignore the extra LSBs just as you said, but don't really Support 32bit. This change is to make driver accept S32_LE format. Best regards Wang shengjiu

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[alsa-devel] [PATCH v4 0/2] soundwire: Add support to Qualcomm SoundWire master
by Srinivas Kandagatla 05 Nov '19

05 Nov '19

Thanks for reviewing the v3 patchset. Here is new patchset addressing all the comments from v3 This patchset adds support for Qualcomm SoundWire Master Controller found in most of Qualcomm SoCs and WCD audio codecs. This driver along with WCD934x codec and WSA881x Class-D Smart Speaker Amplifier drivers is tested on on DragonBoard DB845c based of SDM845 SoC and Lenovo YOGA C630 Laptop based on SDM850. SoundWire controller on SDM845 is integrated in WCD934x audio codec via SlimBus interface. Currently this driver is very minimal and only supports PDM. 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. TODO: Test and add PCM support. Thanks, srini Changes since v3: - Updated bindings as suggested by Rob. - Fixed sdw_disable/unprepare order. - removed setting stream_name as suggested by Pierre - removed v1.5.0 controller support for now, will be added after testing. Srinivas Kandagatla (2): dt-bindings: soundwire: add bindings for Qcom controller soundwire: qcom: add support for SoundWire controller .../bindings/soundwire/qcom,sdw.txt | 179 ++++ drivers/soundwire/Kconfig | 9 + drivers/soundwire/Makefile | 4 + drivers/soundwire/qcom.c | 904 ++++++++++++++++++ 4 files changed, 1096 insertions(+) create mode 100644 Documentation/devicetree/bindings/soundwire/qcom,sdw.txt create mode 100644 drivers/soundwire/qcom.c -- 2.21.0

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[alsa-devel] [PATCH] slimbus: qcom: add missed clk_disable_unprepare in remove
by Chuhong Yuan 04 Nov '19

04 Nov '19

The remove misses to disable and unprepare rclk and hclk. Add calls to clk_disable_unprepare to fix it. Signed-off-by: Chuhong Yuan <hslester96(a)gmail.com> --- drivers/slimbus/qcom-ctrl.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/drivers/slimbus/qcom-ctrl.c b/drivers/slimbus/qcom-ctrl.c index a444badd8df5..4aad2566f52d 100644 --- a/drivers/slimbus/qcom-ctrl.c +++ b/drivers/slimbus/qcom-ctrl.c @@ -641,6 +641,8 @@ static int qcom_slim_remove(struct platform_device *pdev) pm_runtime_disable(&pdev->dev); slim_unregister_controller(&ctrl->ctrl); + clk_disable_unprepare(ctrl->rclk); + clk_disable_unprepare(ctrl->hclk); destroy_workqueue(ctrl->rxwq); return 0; } -- 2.23.0

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[alsa-devel] [PATCH] ASoC: max98373: Fix calls to free_gpio()
by Yong Zhi 04 Nov '19

04 Nov '19

Don't Call free_gpio() when request_gpio() fails, call it at error paths to avoid resource leak. Signed-off-by: Yong Zhi <yong.zhi(a)intel.com> --- sound/soc/codecs/max98373.c | 12 +++++++++--- 1 file changed, 9 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/max98373.c b/sound/soc/codecs/max98373.c index eb709d528259..83984b5244c5 100644 --- a/sound/soc/codecs/max98373.c +++ b/sound/soc/codecs/max98373.c @@ -964,7 +964,6 @@ static int max98373_i2c_probe(struct i2c_client *i2c, if (ret) { dev_err(&i2c->dev, "%s: Failed to request gpio %d\n", __func__, max98373->reset_gpio); - gpio_free(max98373->reset_gpio); return -EINVAL; } gpio_direction_output(max98373->reset_gpio, 0); @@ -979,16 +978,23 @@ static int max98373_i2c_probe(struct i2c_client *i2c, if (ret < 0) { dev_err(&i2c->dev, "Failed to read: 0x%02X\n", MAX98373_R21FF_REV_ID); - return ret; + goto err_gpio_free; } dev_info(&i2c->dev, "MAX98373 revisionID: 0x%02X\n", reg); /* codec registeration */ ret = devm_snd_soc_register_component(&i2c->dev, &soc_codec_dev_max98373, max98373_dai, ARRAY_SIZE(max98373_dai)); - if (ret < 0) + if (ret < 0) { dev_err(&i2c->dev, "Failed to register codec: %d\n", ret); + goto err_gpio_free; + } + + return 0; +err_gpio_free: + if (gpio_is_valid(max98373->reset_gpio)) + gpio_free(max98373->reset_gpio); return ret; } -- 2.7.4

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