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

[alsa-devel] [PATCH 00/10] ALSA: aloop: Support selection of snd_timer
by twischer＠de.adit-jv.com 05 Apr '19

05 Apr '19

With the following patch set any sound timer can be used as timer source for the aloop devices if selected. It can be select by the additional timer_source kernel module parameter: -2 the jiffies timer will be used as done without this patch (default) -1 the sound timer will be selected depending on the linked sound cards (snd_pcm_link()). The sound timer of the first linked card which is not a aloop device nor a dummy device will be used. 0..N Select the sound timer by the card number. This patch set is required for forwarding audio data between a HW sound card and an aloop device without the usage of an asynchronous sample rate converter. Best regards, Timo

3 25

[alsa-devel] [PATCH 1/3] ASoC: wcd9335: fix a leaked reference by adding missing of_node_put
by Wen Yang 05 Apr '19

05 Apr '19

The call to of_parse_phandle returns a node pointer with refcount incremented thus it must be explicitly decremented after the last usage. Detected by coccinelle with the following warnings: ./sound/soc/codecs/wcd9335.c:5193:2-8: ERROR: missing of_node_put; acquired a node pointer with refcount incremented on line 5183, but without a correspon ding object release within this function. Signed-off-by: Wen Yang <wen.yang99(a)zte.com.cn> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Cc: Vinod Koul <vkoul(a)kernel.org> Cc: Dan Carpenter <dan.carpenter(a)oracle.com> (commit_signer:1/11=9%,authored:1/11=9%) Cc: alsa-devel(a)alsa-project.org Cc: linux-kernel(a)vger.kernel.org --- sound/soc/codecs/wcd9335.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/wcd9335.c b/sound/soc/codecs/wcd9335.c index 981f88a..a04a7ce 100644 --- a/sound/soc/codecs/wcd9335.c +++ b/sound/soc/codecs/wcd9335.c @@ -5188,6 +5188,7 @@ static int wcd9335_slim_status(struct slim_device *sdev, wcd->slim = sdev; wcd->slim_ifc_dev = of_slim_get_device(sdev->ctrl, ifc_dev_np); + of_node_put(ifc_dev_np); if (!wcd->slim_ifc_dev) { dev_err(dev, "Unable to get SLIM Interface device\n"); return -EINVAL; -- 2.9.5

2 3

[alsa-devel] [PATCH v4 00/20] ASoC: Sound Open Firmware (SOF) - Intel support
by Pierre-Louis Bossart 04 Apr '19

04 Apr '19

This patch series introduces the support for Intel devices (Baytrail, CherryTrail, Broadwell, ApolloLake, GeminiLake, CannonLake, IceLake). Support for Haswell, Skylake and KabyLake is incomplete at the moment and is not shared for now. The series starts with the 'legacy' devices, then all the more recent ones and concludes with ACPI/PCI hooks. SKL+ platforms can be compiled without HDaudio link/codec support, in which case the hdac library is not used. When support for HDMI or HDaudio codecs is selected, parts of this library are used. A small set of machine drivers supported by SOF is now provided as part of this series, more to follow. Support for SPI-based devices, compressed streams is also not provided for now since it's not ready for upstream just yet. Thank you for reviews and comments, we appreciate the time spent commenting on this large patchset. Thanks in particular to Alan Cox and Andy Shevchenko for their comments on an earlier version. Changes since v3: Removed Haswell, Skylake and Kabylake support (code is not tested or is still under development). These devices will be re-added when there is functional firmware and for Skylake/Kabylake a signed version of the firmware usable on existing platforms. Added support for existing machine drivers when relevant (more machine drivers will be added in next batch) Fixed hardware programming sequences (ROM init timeout, HDaudio capture, simultaneous capture and playback, DMA preload) Avoid reading IPC reply on CTX_SAVE reply since memory windows might be powered off. Simplified NOCODEC support, now mutually exclusive with HDaudio link to avoid suspend/resume issues. Changes since v2: Addressed dozens of comments from Takashi Iwai, Mark Brown, Andy Shevchenko, Daniel Baluta (Thanks!) Transition to new display power management Fixed suspend issues Optimized register polling, better handling of timeouts and firmware download retries Detect number of HDaudio codec and select machine driver accordingly Added workqueue based probe for HDaudio (dependency on modules) Fixed platform name override and conflicts with DT deferred probe changes Added prefix for firmware and topology files and module arguments Split Baytrail and Edison Fixed warnings (smatch, cppcheck) Keyon Jie (2): ASoC: SOF: Intel: Add hda-bus support and initialization ASoC: SOF: Intel: add SKL+ platform DAIs Liam Girdwood (15): ASoC: SOF: Intel: Add BYT, CHT and BSW DSP HW support. ASoC: SOF: Intel: Add BDW HW DSP support ASoC: SOF: Intel: Add APL/CNL HW DSP support ASoC: SOF: Intel: Add HDA controller for Intel DSP ASoC: SOF: Intel: Add Intel specific HDA DSP HW operations ASoC: SOF: Intel: Add Intel specific HDA IPC mechanisms. ASoC: SOF: Intel: Add Intel specific HDA firmware loader ASoC: SOF: Intel: Add Intel specific HDA PCM operations ASoC: SOF: Intel: Add Intel specific HDA stream operations ASoC: SOF: Intel: Add Intel specific HDA trace operations ASoC: SOF: Intel: Add platform differentiation for APL and CNL ASoC: SOF: Add ACPI device support ASoC: SOF: Add PCI device support ASoC: SOF: Add Build support for SOF core and Intel drivers ASoC: Intel: Make sure HSW/BDW based machine drivers build for SOF Pierre-Louis Bossart (3): ASoC: SOF: Intel: Add support for HDAudio codecs ASoC: Intel: Kconfig: expose common option between SST and SOF drivers ASoC: Intel: select relevant machine drivers for SOF sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/intel/Kconfig | 7 +- sound/soc/intel/boards/Kconfig | 22 +- sound/soc/intel/boards/bdw-rt5677.c | 4 + sound/soc/intel/boards/broadwell.c | 4 + sound/soc/intel/boards/haswell.c | 4 + sound/soc/sof/Kconfig | 141 +++++ sound/soc/sof/Makefile | 18 + sound/soc/sof/intel/Kconfig | 222 ++++++++ sound/soc/sof/intel/Makefile | 15 + sound/soc/sof/intel/apl.c | 112 ++++ sound/soc/sof/intel/bdw.c | 694 ++++++++++++++++++++++ sound/soc/sof/intel/byt.c | 853 ++++++++++++++++++++++++++++ sound/soc/sof/intel/cnl.c | 257 +++++++++ sound/soc/sof/intel/hda-bus.c | 108 ++++ sound/soc/sof/intel/hda-codec.c | 171 ++++++ sound/soc/sof/intel/hda-ctrl.c | 165 ++++++ sound/soc/sof/intel/hda-dai.c | 318 +++++++++++ sound/soc/sof/intel/hda-dsp.c | 428 ++++++++++++++ sound/soc/sof/intel/hda-ipc.c | 402 +++++++++++++ sound/soc/sof/intel/hda-loader.c | 366 ++++++++++++ sound/soc/sof/intel/hda-pcm.c | 242 ++++++++ sound/soc/sof/intel/hda-stream.c | 682 ++++++++++++++++++++++ sound/soc/sof/intel/hda-trace.c | 98 ++++ sound/soc/sof/intel/hda.c | 672 ++++++++++++++++++++++ sound/soc/sof/intel/hda.h | 546 ++++++++++++++++++ sound/soc/sof/intel/shim.h | 179 ++++++ sound/soc/sof/sof-acpi-dev.c | 310 ++++++++++ sound/soc/sof/sof-pci-dev.c | 373 ++++++++++++ 30 files changed, 7407 insertions(+), 8 deletions(-) create mode 100644 sound/soc/sof/Kconfig create mode 100644 sound/soc/sof/Makefile create mode 100644 sound/soc/sof/intel/Kconfig create mode 100644 sound/soc/sof/intel/Makefile create mode 100644 sound/soc/sof/intel/apl.c create mode 100644 sound/soc/sof/intel/bdw.c create mode 100644 sound/soc/sof/intel/byt.c create mode 100644 sound/soc/sof/intel/cnl.c create mode 100644 sound/soc/sof/intel/hda-bus.c create mode 100644 sound/soc/sof/intel/hda-codec.c create mode 100644 sound/soc/sof/intel/hda-ctrl.c create mode 100644 sound/soc/sof/intel/hda-dai.c create mode 100644 sound/soc/sof/intel/hda-dsp.c create mode 100644 sound/soc/sof/intel/hda-ipc.c create mode 100644 sound/soc/sof/intel/hda-loader.c create mode 100644 sound/soc/sof/intel/hda-pcm.c create mode 100644 sound/soc/sof/intel/hda-stream.c create mode 100644 sound/soc/sof/intel/hda-trace.c create mode 100644 sound/soc/sof/intel/hda.c create mode 100644 sound/soc/sof/intel/hda.h create mode 100644 sound/soc/sof/intel/shim.h create mode 100644 sound/soc/sof/sof-acpi-dev.c create mode 100644 sound/soc/sof/sof-pci-dev.c -- 2.17.1

5 34

[alsa-devel] [PATCH] ASoC: soc-core: Fix probe deferral following prelink failure
by Jon Hunter 03 Apr '19

03 Apr '19

From: Jonathan Hunter <jonathanh(a)nvidia.com> Commit 78a24e10cd94 ("ASoC: soc-core: clear platform pointers on error") re-worked the clean-up of any platform pointers that may have been initialised by the function snd_soc_init_platform(). This commit missed one error path where if any of the prelinks for a soundcard failed to initialise, then these platform pointers would not be cleaned-up. This then prevents the soundcard from being initialised following a probe deferral when any of the soundcard prelinks cannot be found. Fix this by ensuring that soc_cleanup_platform() is called when initialising the soundcard prelinks fails. Fixes: 78a24e10cd94 ("ASoC: soc-core: clear platform pointers on error") Signed-off-by: Jonathan Hunter <jonathanh(a)nvidia.com> --- sound/soc/soc-core.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 93d316d5bf8e..5a5764dba147 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2797,6 +2797,7 @@ int snd_soc_register_card(struct snd_soc_card *card) ret = soc_init_dai_link(card, link); if (ret) { + soc_cleanup_platform(card); dev_err(card->dev, "ASoC: failed to init link %s\n", link->name); mutex_unlock(&client_mutex); -- 1.9.1

2 9

[alsa-devel] [PATCH v3 00/10] ACPI / utils: Replace leaky function
by Andy Shevchenko 02 Apr '19

02 Apr '19

The acpi_dev_get_first_match_name() is missing put_device() call and thus keeping reference counting unbalanced. In order to fix the issue introduce a new helper to convert existing users one-by-one to a better API. Since v3: - convert all existing users and drop old API (Mika) - add Hans' tag Cc: MyungJoo Ham <myungjoo.ham(a)samsung.com> Cc: Chanwoo Choi <cw00.choi(a)samsung.com> Cc: Chen-Yu Tsai <wens(a)csie.org> Cc: Linus Walleij <linus.walleij(a)linaro.org> Cc: Bartosz Golaszewski <bgolaszewski(a)baylibre.com> Andy Shevchenko (10): ACPI / utils: Introduce acpi_dev_get_first_match_dev() helper extcon: axp288: Convert to use acpi_dev_get_first_match_dev() gpio: merrifield: Convert to use acpi_dev_get_first_match_dev() ASoC: Intel: bytcht_da7213: Convert to use acpi_dev_get_first_match_dev() ASoC: Intel: bytcht_es8316: Convert to use acpi_dev_get_first_match_dev() ASoC: Intel: bytcr_rt5640: Convert to use acpi_dev_get_first_match_dev() ASoC: Intel: bytcr_rt5651: Convert to use acpi_dev_get_first_match_dev() ASoC: Intel: cht_bsw_rt5645: Convert to use acpi_dev_get_first_match_dev() ASoC: Intel: cht_bsw_rt5672: Convert to use acpi_dev_get_first_match_dev() ACPI / utils: Remove deprecated function since no user left drivers/acpi/utils.c | 16 ++++++++++------ drivers/extcon/extcon-axp288.c | 9 +++++---- drivers/gpio/gpio-merrifield.c | 18 ++++++++++++++---- include/acpi/acpi_bus.h | 4 ++-- include/linux/acpi.h | 4 ++-- sound/soc/intel/boards/bytcht_da7213.c | 9 +++++---- sound/soc/intel/boards/bytcht_es8316.c | 9 +++++---- sound/soc/intel/boards/bytcr_rt5640.c | 10 +++++----- sound/soc/intel/boards/bytcr_rt5651.c | 14 ++++++++------ sound/soc/intel/boards/cht_bsw_rt5645.c | 9 +++++---- sound/soc/intel/boards/cht_bsw_rt5672.c | 9 +++++---- 11 files changed, 66 insertions(+), 45 deletions(-) -- 2.20.1

8 26

[alsa-devel] regression: bad sound card type detection in audio graph
by Arnaud Pouliquen 02 Apr '19

02 Apr '19

Hello Kuninori, We are seeing a regression while rebasing on kernel V5.1 (from v4.19). The regression is linked to your update of the audio graph to merge the support of the normal and DPCM sound cards. In our sound card (described here: https://github.com/STMicroelectronics/linux/blob/v4.19-stm32mp/arch/arm/boo…) We declare a normal sound card using the WM8994 codec: Playback: CPU-DAI (SAI2a)-> CODEC-DAI/port0 Record: CPU-DAI (SAI2B)<- CODEC-DAI/port1 ... The WM8994 codec is declared with 2 ports as it supports 2 interfaces. When we probe the sound card we are detecting a wrong dai link type because the condition if (of_get_child_count(codec_port) > 1) return true. The dai link is detected as DPCM instead of the expected normal link. I checked your last commits sent on mailing list, seems that this criteria is still valid. Please could you elaborate a little bit on this criteria for the normal/DPCM selection? What is you feeling about this limitation. Do you see a way to refine the condition to allow normal sound card with multi-port codecs. Or do we have to migrate our sound card to DPCM. Thanks in advance for your answer Regards Arnaud

2 7

[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

2 2

[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

3 8

[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

2 1

[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

2 1