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

[alsa-devel] [PATCH] Documentation: soundwire: Ensure that code is inside the code blocks
by Jonathan Neuschäfer 02 Apr '19

02 Apr '19

The way the document is written now, Sphinx renders empty code blocks followed by the lines that should be inside them. Unindent the code-block directives to fix this. Fixes: 502c00d9c315 ("Documentation: soundwire: fix stream.rst markup warnings") Signed-off-by: Jonathan Neuschäfer <j.neuschaefer(a)gmx.net> --- Documentation/driver-api/soundwire/stream.rst | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/Documentation/driver-api/soundwire/stream.rst b/Documentation/driver-api/soundwire/stream.rst index 26a6064503fd..5351bd2f34a8 100644 --- a/Documentation/driver-api/soundwire/stream.rst +++ b/Documentation/driver-api/soundwire/stream.rst @@ -201,7 +201,7 @@ Bus implements below API for allocate a stream which needs to be called once per stream. From ASoC DPCM framework, this stream state maybe linked to .startup() operation. - .. code-block:: c +.. code-block:: c int sdw_alloc_stream(char * stream_name); @@ -228,7 +228,7 @@ the respective Master(s) and Slave(s) associated with stream. These APIs can only be invoked once by respective Master(s) and Slave(s). From ASoC DPCM framework, this stream state is linked to .hw_params() operation. - .. code-block:: c +.. code-block:: c int sdw_stream_add_master(struct sdw_bus * bus, struct sdw_stream_config * stream_config, @@ -274,7 +274,7 @@ Bus implements below API for PREPARE state which needs to be called once per stream. From ASoC DPCM framework, this stream state is linked to .prepare() operation. - .. code-block:: c +.. code-block:: c int sdw_prepare_stream(struct sdw_stream_runtime * stream); @@ -304,7 +304,7 @@ Bus implements below API for ENABLE state which needs to be called once per stream. From ASoC DPCM framework, this stream state is linked to .trigger() start operation. - .. code-block:: c +.. code-block:: c int sdw_enable_stream(struct sdw_stream_runtime * stream); @@ -332,7 +332,7 @@ Bus implements below API for DISABLED state which needs to be called once per stream. From ASoC DPCM framework, this stream state is linked to .trigger() stop operation. - .. code-block:: c +.. code-block:: c int sdw_disable_stream(struct sdw_stream_runtime * stream); @@ -357,7 +357,7 @@ Bus implements below API for DEPREPARED state which needs to be called once per stream. From ASoC DPCM framework, this stream state is linked to .trigger() stop operation. - .. code-block:: c +.. code-block:: c int sdw_deprepare_stream(struct sdw_stream_runtime * stream); @@ -382,7 +382,7 @@ Bus implements below APIs for RELEASE state which needs to be called by all the Master(s) and Slave(s) associated with stream. From ASoC DPCM framework, this stream state is linked to .hw_free() operation. - .. code-block:: c +.. code-block:: c int sdw_stream_remove_master(struct sdw_bus * bus, struct sdw_stream_runtime * stream); @@ -395,7 +395,7 @@ stream assigned as part of ALLOCATED state. In .shutdown() the data structure maintaining stream state are freed up. - .. code-block:: c +.. code-block:: c void sdw_release_stream(struct sdw_stream_runtime * stream); -- 2.20.1

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[alsa-devel] [PATCH v14 0/6] Media Device Allocator API
by Shuah Khan 01 Apr '19

01 Apr '19

Media Device Allocator API to allows multiple drivers share a media device. This API solves a very common use-case for media devices where one physical device (an USB stick) provides both audio and video. When such media device exposes a standard USB Audio class, a proprietary Video class, two or more independent drivers will share a single physical USB bridge. In such cases, it is necessary to coordinate access to the shared resource. Using this API, drivers can allocate a media device with the shared struct device as the key. Once the media device is allocated by a driver, other drivers can get a reference to it. The media device is released when all the references are released. The primary focus for testing The patch series is making sure media device is released when both drivers release the media device with a series of unbind/binds on both drivers. - both au0828 and snd-usb-aduio as built-in - both au0828 and snd-usb-aduio as modules - au0828 as module and snd-usb-aduio as built-in - au0828 as built-in and snd-usb-aduio as module Test results can be found at: https://docs.google.com/document/d/1RMF8Rwj7xHJEoOx6_K2f-REgZJ63BMeAVEf-CV-… Changes since v13: - Minor changes to variable names and other minor changes to copyright and typos suggested by Hans Verkuil. Changes since v12: - Patch 1: Fixed prototype warns from media_dev_allocator.c. Removed dependency on find_module() by adding struct module to input args. Still need module name to pass into media_device API. - Patch 2 & 4: Update media dev allocator api calls to pass in struct module pointer. - No changes to Patches 3 & 5. - Added patch 6 with a test. It can go in separately. Changes since v11: - Patch 1: Add CONFIG_MODULES dependency in media_dev_allocator files. to fix compile errors when CONFIG_MODULES is disabled. - Patch 2, 3: No changes. - Patch 4: Fix sparse error reported by Dan Carpenter. - Patch 5: Fix warns found by Hans Verkuil. - v11 was tested on 5.0-rc7 and addresses comments on v10 series from Hans Verkuil. Fixed problems found in resource sharing logic in au0828 adding a patch 5 to this series. The test plan used for testing resource sharing could serve as a regression test plan and the test results can be found at: - v10 was tested on 5.0-rc3 and addresses comments on v9 series from Hans Verkuil. Changes since v10: - Patch 1: Fixed SPDX tag and removed redundant IS_ENABLED(CONFIG_USB) around media_device_usb_allocate() - Sakari Ailus's review comment. - Patch 2 and 3: No changes - Patch 4: Fixed SPDX tag - Sakari Ailus's review comment. - Carried Reviewed-by tag from Takashi Iwai for the sound from v9. - Patch 5: This is a new patch added to fix resource sharing inconsistencies and problem found during testing using Han's tests. Changes since v9: - Patch 1: Fix mutex assert warning from find_module() calls. This code was written before the change to find_module() that requires callers to hold module_mutex. I missed this during my testing on 4.20-rc6. Hans Verkuil reported the problem. - Patch 4: sound/usb: Initializes all the entities it can before registering the device based on comments from Hans Verkuil - Carried Reviewed-by tag from Takashi Iwai for the sound from v9. - No changes to Patches 2 and 3. References: https://lkml.org/lkml/2018/11/2/169 https://www.mail-archive.com/linux-media@vger.kernel.org/msg105854.html Shuah Khan (6): media: Media Device Allocator API media: change au0828 to use Media Device Allocator API media: media.h: Enable ALSA MEDIA_INTF_T* interface types sound/usb: Use Media Controller API to share media resources au0828: fix enable and disable source audio and video inconsistencies selftests: media_dev_allocator api test Documentation/media/kapi/mc-core.rst | 41 +++ drivers/media/Makefile | 6 + drivers/media/media-dev-allocator.c | 135 ++++++++ drivers/media/usb/au0828/Kconfig | 2 + drivers/media/usb/au0828/au0828-core.c | 195 ++++++++--- drivers/media/usb/au0828/au0828.h | 6 +- include/media/media-dev-allocator.h | 63 ++++ include/uapi/linux/media.h | 25 +- sound/usb/Kconfig | 4 + sound/usb/Makefile | 2 + sound/usb/card.c | 14 + sound/usb/card.h | 3 + sound/usb/media.c | 327 ++++++++++++++++++ sound/usb/media.h | 74 ++++ sound/usb/mixer.h | 3 + sound/usb/pcm.c | 29 +- sound/usb/quirks-table.h | 1 + sound/usb/stream.c | 2 + sound/usb/usbaudio.h | 6 + .../media_tests/media_dev_allocator.sh | 85 +++++ 20 files changed, 962 insertions(+), 61 deletions(-) create mode 100644 drivers/media/media-dev-allocator.c create mode 100644 include/media/media-dev-allocator.h create mode 100644 sound/usb/media.c create mode 100644 sound/usb/media.h create mode 100755 tools/testing/selftests/media_tests/media_dev_allocator.sh -- 2.17.1

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[alsa-devel] [PATCH] ALSA: uapi: #include <time.h> in asound.h
by Daniel Mentz 01 Apr '19

01 Apr '19

The uapi header asound.h defines types based on struct timespec. We need to #include <time.h> to get access to the definition of this struct. Previously, we encountered the following error message when building applications with a clang/bionic toolchain: kernel-headers/sound/asound.h:350:19: error: field has incomplete type 'struct timespec' struct timespec trigger_tstamp; ^ The absence of the time.h #include statement does not cause build errors with glibc, because its version of stdlib.h indirectly includes time.h. Signed-off-by: Daniel Mentz <danielmentz(a)google.com> --- include/uapi/sound/asound.h | 1 + 1 file changed, 1 insertion(+) diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index 404d4b9ffe76..df1153cea0b7 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -32,6 +32,7 @@ #ifndef __KERNEL__ #include <stdlib.h> +#include <time.h> #endif /* -- 2.21.0.392.gf8f6787159e-goog

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[alsa-devel] [PATCH] ASoC: cs42l51: add multi endpoint support
by Olivier Moysan 01 Apr '19

01 Apr '19

Support multiple endpoints on cs42L51 codec port when used in of_graph context. This patch allows to share the codec port between two CPU DAIs. Example: STM32MP157C-DK2 board uses CS42L51 audio codec. This codec is connected to two serial audio interfaces, which are configured either as rx or tx. >From AsoC point of view the topolgy is the following: // 2 CPU DAIs (SAI2A/B), 1 Codec (CS42L51) Playback: CPU-A-DAI(slave) -> (master)CODEC-DAI/port0 Record: CPU-B-DAI(slave) <- (master)CODEC-DAI/port0 In the DT two endpoints have to be associated to the codec port: cs42l51_port: port { cs42l51_tx_endpoint: endpoint@0 { remote-endpoint = <&sai2a_endpoint>; }; cs42l51_rx_endpoint: endpoint@1 { remote-endpoint = <&sai2b_endpoint>; }; }; However, when the audio graph card parses the codec nodes, it expects to find DAI interface indexes matching the endpoints indexes. The current patch forces the use of DAI id 0 for both endpoints, which allows to share the codec DAI between the two CPU DAIs for playback and capture streams respectively. Signed-off-by: Olivier Moysan <olivier.moysan(a)st.com> --- sound/soc/codecs/cs42l51.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index fd2bd74024c1..80da3cd73e04 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -464,6 +464,13 @@ static int cs42l51_dai_mute(struct snd_soc_dai *dai, int mute) return snd_soc_component_write(component, CS42L51_DAC_OUT_CTL, reg); } +static int cs42l51_of_xlate_dai_id(struct snd_soc_component *component, + struct device_node *endpoint) +{ + /* return dai id 0, whatever the endpoint index */ + return 0; +} + static const struct snd_soc_dai_ops cs42l51_dai_ops = { .hw_params = cs42l51_hw_params, .set_sysclk = cs42l51_set_dai_sysclk, @@ -526,6 +533,7 @@ static const struct snd_soc_component_driver soc_component_device_cs42l51 = { .num_dapm_widgets = ARRAY_SIZE(cs42l51_dapm_widgets), .dapm_routes = cs42l51_routes, .num_dapm_routes = ARRAY_SIZE(cs42l51_routes), + .of_xlate_dai_id = cs42l51_of_xlate_dai_id, .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, -- 2.7.4

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[alsa-devel] [PATCH] ASoC: rt5677-spi: Add ACPI ID
by Oder Chiou 01 Apr '19

01 Apr '19

Add the ACPI ID for the product "chromebook pixel 2015" to match the coreboot settings. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5677-spi.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/codecs/rt5677-spi.c b/sound/soc/codecs/rt5677-spi.c index 84501c2020c7..167a02773a0b 100644 --- a/sound/soc/codecs/rt5677-spi.c +++ b/sound/soc/codecs/rt5677-spi.c @@ -25,6 +25,7 @@ #include <linux/sysfs.h> #include <linux/clk.h> #include <linux/firmware.h> +#include <linux/acpi.h> #include "rt5677-spi.h" @@ -226,9 +227,16 @@ static int rt5677_spi_probe(struct spi_device *spi) return 0; } +static const struct acpi_device_id rt5677_spi_acpi_id[] = { + { "RT5677AA", 0 }, + { } +}; +MODULE_DEVICE_TABLE(acpi, rt5677_spi_acpi_id); + static struct spi_driver rt5677_spi_driver = { .driver = { .name = "rt5677", + .acpi_match_table = ACPI_PTR(rt5677_spi_acpi_id), }, .probe = rt5677_spi_probe, }; -- 2.21.0

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[alsa-devel] [PATCH 0/4] ASoC: Mediatek: MT8183: Add support for machine driver with DA7219 or TS3A227
by Shunli Wang 01 Apr '19

01 Apr '19

ASoC: Mediatek: MT8183: Add support for machine driver with DA7219 or TS3A227 Shunli Wang (4): ASoC: Mediatek: MT8183: Add machine driver with TS3A227 ASoC: Mediatek: Add document for mt8183-mt6358-ts3a227-max98357 ASoC: Mediatek: MT8183: Add machine driver with DA7219 ASoC: Mediatek: Add document for mt8183-da7219-max98357 .../bindings/sound/mt8183-da7219-max98357.txt | 15 + .../sound/mt8183-mt6358-ts3a227-max98357.txt | 15 + sound/soc/mediatek/Kconfig | 26 ++ sound/soc/mediatek/mt8183/Makefile | 2 + sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c | 472 +++++++++++++++++++++ .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 382 +++++++++++++++++ 6 files changed, 912 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/mt8183-da7219-max98357.txt create mode 100644 Documentation/devicetree/bindings/sound/mt8183-mt6358-ts3a227-max98357.txt create mode 100644 sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c create mode 100644 sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c -- 1.9.1

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[alsa-devel] [PATCH][next] ASoC: fsl: fix spelling mistake: "missign" -> "missing"
by Colin King 01 Apr '19

01 Apr '19

From: Colin Ian King <colin.king(a)canonical.com> There is a spelling mistake in a dev_err message. Fix this. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/fsl/imx-audmix.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/imx-audmix.c b/sound/soc/fsl/imx-audmix.c index 72e37ca6cfbb..7983bd339c01 100644 --- a/sound/soc/fsl/imx-audmix.c +++ b/sound/soc/fsl/imx-audmix.c @@ -161,7 +161,7 @@ static int imx_audmix_probe(struct platform_device *pdev) } if (!audmix_np) { - dev_err(&pdev->dev, "Missign DT node for parent device.\n"); + dev_err(&pdev->dev, "Missing DT node for parent device.\n"); return -EINVAL; } -- 2.20.1

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[alsa-devel] [PATCH] ASoC: wm_adsp: Fix typo in comment
by Charles Keepax 01 Apr '19

01 Apr '19

Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/codecs/wm_adsp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 5e615fbddf9a7..0aa62b26f61cc 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -3263,7 +3263,7 @@ static void wm_halo_stop_core(struct wm_adsp *dsp) regmap_update_bits(dsp->regmap, dsp->base + HALO_CCM_CORE_CONTROL, HALO_CORE_EN, 0); - /* reset halo core with CORE_SOFT_REEST */ + /* reset halo core with CORE_SOFT_RESET */ regmap_update_bits(dsp->regmap, dsp->base + HALO_CORE_SOFT_RESET, HALO_CORE_SOFT_RESET_MASK, 1); } -- 2.11.0

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[alsa-devel] [PATCH v3 1/2] ASoC: tlv320aic31xx: Add headphone/headset detection
by Andrew F. Davis 01 Apr '19

01 Apr '19

This device can detect the insertion/removal of headphones and headsets. Enable reporting this status by enabling this interrupt and forwarding this to upper-layers if a jack has been defined. This jack definition and the resulting operation from a jack detection event must currently be defined by sound card platform code until CODEC outputs to jack mappings can be defined generically. Signed-off-by: Andrew F. Davis <afd(a)ti.com> --- Changes from v1: - Rebase Changes from v2: - Rebase sound/soc/codecs/tlv320aic31xx.c | 48 +++++++++++++++++++++++++++++++- sound/soc/codecs/tlv320aic31xx.h | 11 ++++++++ 2 files changed, 58 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index 6744f736501f..3e11b3d41f3d 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -25,6 +25,7 @@ #include <linux/of_gpio.h> #include <linux/slab.h> #include <sound/core.h> +#include <sound/jack.h> #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> @@ -89,6 +90,7 @@ static bool aic31xx_volatile(struct device *dev, unsigned int reg) case AIC31XX_INTRADCFLAG: /* Sticky interrupt flags */ case AIC31XX_INTRDACFLAG2: case AIC31XX_INTRADCFLAG2: + case AIC31XX_HSDETECT: return true; } return false; @@ -157,6 +159,7 @@ struct aic31xx_priv { int micbias_vg; struct aic31xx_pdata pdata; struct regulator_bulk_data supplies[AIC31XX_NUM_SUPPLIES]; + struct snd_soc_jack *jack; unsigned int sysclk; u8 p_div; int rate_div_line; @@ -1233,6 +1236,20 @@ static int aic31xx_set_bias_level(struct snd_soc_component *component, return 0; } +int aic31xx_set_jack(struct snd_soc_component *component, + struct snd_soc_jack *jack, void *data) +{ + struct aic31xx_priv *aic31xx = snd_soc_component_get_drvdata(component); + + aic31xx->jack = jack; + + /* Enable/Disable jack detection */ + regmap_write(aic31xx->regmap, AIC31XX_HSDETECT, + jack ? AIC31XX_HSD_ENABLE : 0); + + return 0; +} + static int aic31xx_codec_probe(struct snd_soc_component *component) { struct aic31xx_priv *aic31xx = snd_soc_component_get_drvdata(component); @@ -1258,6 +1275,7 @@ static int aic31xx_codec_probe(struct snd_soc_component *component) static const struct snd_soc_component_driver soc_codec_driver_aic31xx = { .probe = aic31xx_codec_probe, + .set_jack = aic31xx_set_jack, .set_bias_level = aic31xx_set_bias_level, .controls = common31xx_snd_controls, .num_controls = ARRAY_SIZE(common31xx_snd_controls), @@ -1362,8 +1380,35 @@ static irqreturn_t aic31xx_irq(int irq, void *data) dev_err(dev, "Short circuit on Left output is detected\n"); if (value & AIC31XX_HPRSCDETECT) dev_err(dev, "Short circuit on Right output is detected\n"); + if (value & AIC31XX_HSPLUG) { + unsigned int val; + int status = 0; + + ret = regmap_read(aic31xx->regmap, AIC31XX_HSDETECT, &val); + if (ret) { + dev_err(dev, "Failed to read headset type: %d\n", ret); + goto exit; + } + + switch ((val & AIC31XX_HSD_TYPE_MASK) >> + AIC31XX_HSD_TYPE_SHIFT) { + case AIC31XX_HSD_HP: + status |= SND_JACK_HEADPHONE; + break; + case AIC31XX_HSD_HS: + status |= SND_JACK_HEADSET; + break; + default: + break; + } + + if (aic31xx->jack) + snd_soc_jack_report(aic31xx->jack, status, + AIC31XX_JACK_MASK); + } if (value & ~(AIC31XX_HPLSCDETECT | - AIC31XX_HPRSCDETECT)) + AIC31XX_HPRSCDETECT | + AIC31XX_HSPLUG)) dev_err(dev, "Unknown DAC interrupt flags: 0x%08x\n", value); read_overflow: @@ -1476,6 +1521,7 @@ static int aic31xx_i2c_probe(struct i2c_client *i2c, AIC31XX_GPIO1_FUNC_SHIFT); regmap_write(aic31xx->regmap, AIC31XX_INT1CTRL, + AIC31XX_HSPLUGDET | AIC31XX_SC | AIC31XX_ENGINE); diff --git a/sound/soc/codecs/tlv320aic31xx.h b/sound/soc/codecs/tlv320aic31xx.h index 2636f2c6bc79..9985ceed97ae 100644 --- a/sound/soc/codecs/tlv320aic31xx.h +++ b/sound/soc/codecs/tlv320aic31xx.h @@ -20,6 +20,9 @@ #define AIC31XX_MINIDSP_BIT BIT(2) #define DAC31XX_BIT BIT(3) +#define AIC31XX_JACK_MASK (SND_JACK_HEADPHONE | \ + SND_JACK_HEADSET) + enum aic31xx_type { AIC3100 = 0, AIC3110 = AIC31XX_STEREO_CLASS_D_BIT, @@ -220,6 +223,14 @@ struct aic31xx_pdata { /* AIC31XX_DACMUTE */ #define AIC31XX_DACMUTE_MASK GENMASK(3, 2) +/* AIC31XX_HSDETECT */ +#define AIC31XX_HSD_ENABLE BIT(7) +#define AIC31XX_HSD_TYPE_MASK GENMASK(6, 5) +#define AIC31XX_HSD_TYPE_SHIFT 5 +#define AIC31XX_HSD_NONE 0x00 +#define AIC31XX_HSD_HP 0x01 +#define AIC31XX_HSD_HS 0x03 + /* AIC31XX_MICBIAS */ #define AIC31XX_MICBIAS_MASK GENMASK(1, 0) #define AIC31XX_MICBIAS_SHIFT 0 -- 2.21.0

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Re: [alsa-devel] INFO: task hung in evdev_release
by syzbot 01 Apr '19

01 Apr '19

syzbot has bisected this bug to: commit e32d99af6830c9a8f37b4f2637ef0cdc60fa79fb Author: Jerome Brunet <jbrunet(a)baylibre.com> Date: Tue Jul 17 15:42:50 2018 +0000 ASoC: meson: add axg fifos DT binding documentation bisection log: https://syzkaller.appspot.com/x/bisect.txt?x=13fba69b200000 start commit: 9dcd936c Merge tag 'for-4.19/dm-fixes-4' of git://git.kern.. git tree: upstream final crash: https://syzkaller.appspot.com/x/report.txt?x=1007a69b200000 console output: https://syzkaller.appspot.com/x/log.txt?x=17fba69b200000 kernel config: https://syzkaller.appspot.com/x/.config?x=88e9a8a39dc0be2d dashboard link: https://syzkaller.appspot.com/bug?extid=a979743610b4755d4d57 syz repro: https://syzkaller.appspot.com/x/repro.syz?x=10fcd826400000 Reported-by: syzbot+a979743610b4755d4d57(a)syzkaller.appspotmail.com Fixes: e32d99af6830 ("ASoC: meson: add axg fifos DT binding documentation") For information about bisection process see: https://goo.gl/tpsmEJ#bisection

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