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

[alsa-devel] [PATCH] ALSA: oxfw: fix memory leak for model-dependent data at error path
by Takashi Sakamoto 17 Sep '18

17 Sep '18

After allocating model-dependent data, ALSA OXFW driver has memory leak of the data at error path. This commit releases the data at the error path. Fixes: 6c29230e2a5f ('ALSA: oxfw: delayed registration of sound card') Cc: <stable(a)vger.kernel.org> # v4.7+ Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/oxfw/oxfw.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/firewire/oxfw/oxfw.c b/sound/firewire/oxfw/oxfw.c index 63b3e1ce986e..9c8c527ebff9 100644 --- a/sound/firewire/oxfw/oxfw.c +++ b/sound/firewire/oxfw/oxfw.c @@ -277,6 +277,8 @@ static void do_registration(struct work_struct *work) oxfw->rx_stream_formats[i] = NULL; } snd_card_free(oxfw->card); + kfree(oxfw->spec); + oxfw->spec = NULL; dev_info(&oxfw->unit->device, "Sound card registration failed: %d\n", err); } -- 2.17.1

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[alsa-devel] [PATCH] ALSA: bebob: fix memory leak for M-Audio FW1814 and ProjectMix I/O at error path
by Takashi Sakamoto 17 Sep '18

17 Sep '18

After allocating model-dependent data for M-Audio FW1814 and ProjectMix I/O, ALSA bebob driver has memory leak at error path. This commit releases the allocated data at the error path. Fixes: 04a2c73c97eb('ALSA: bebob: delayed registration of sound card') Cc: <stable(a)vger.kernel.org> # v4.7+ Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/bebob/bebob.c | 2 ++ sound/firewire/bebob/bebob_maudio.c | 4 ---- 2 files changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/firewire/bebob/bebob.c b/sound/firewire/bebob/bebob.c index 730ea91d9be8..93676354f87f 100644 --- a/sound/firewire/bebob/bebob.c +++ b/sound/firewire/bebob/bebob.c @@ -263,6 +263,8 @@ do_registration(struct work_struct *work) error: mutex_unlock(&devices_mutex); snd_bebob_stream_destroy_duplex(bebob); + kfree(bebob->maudio_special_quirk); + bebob->maudio_special_quirk = NULL; snd_card_free(bebob->card); dev_info(&bebob->unit->device, "Sound card registration failed: %d\n", err); diff --git a/sound/firewire/bebob/bebob_maudio.c b/sound/firewire/bebob/bebob_maudio.c index bd55620c6a47..164f2126a676 100644 --- a/sound/firewire/bebob/bebob_maudio.c +++ b/sound/firewire/bebob/bebob_maudio.c @@ -290,10 +290,6 @@ snd_bebob_maudio_special_discover(struct snd_bebob *bebob, bool is1814) bebob->midi_output_ports = 2; } end: - if (err < 0) { - kfree(params); - bebob->maudio_special_quirk = NULL; - } mutex_unlock(&bebob->mutex); return err; } -- 2.17.1

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[alsa-devel] [GIT PULL] ASoC fixes for v4.19
by Mark Brown 17 Sep '18

17 Sep '18

The following changes since commit f2cf0ef7c0ce141bb38f315c34c56e6ef5667a27: ASoC: adav80x: mark expected switch fall-through (2018-08-09 14:41:50 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v4.19-rc4 for you to fetch changes up to 75383f8d39d4c0fb96083dd460b7b139fbdac492: sound: don't call skl_init_chip() to reset intel skl soc (2018-09-12 11:22:09 +0100) ---------------------------------------------------------------- ASoC: Fixes for v4.19 This is the usual set of small fixes scatterd around various drivers, plus one fix for DAPM and a UAPI build fix. There's not a huge amount that stands out here relative to anything else. ---------------------------------------------------------------- Akshu Agrawal (1): ASoC: AMD: Ensure reset bit is cleared before configuring Andrew F. Davis (1): ASoC: tas6424: Save last fault register even when clear Charles Keepax (1): ASoC: dapm: Fix NULL pointer deference on CODEC to CODEC DAIs Danny Smith (1): ASoC: sigmadsp: safeload should not have lower byte limit Dmitry V. Levin (1): ASoC: uapi: fix sound/skl-tplg-interface.h userspace compilation errors Hans de Goede (1): ASoC: Intel: bytcr_rt5640: Add quirks for 2 more devices Jiada Wang (1): ASoC: rsnd: fixup not to call clk_get/set under non-atomic Katsuhiro Suzuki (1): ASoC: uniphier: change status to orphan Kuninori Morimoto (2): ASoC: rsnd: adg: care clock-frequency size ASoC: rsnd: don't fallback to PIO mode when -EPROBE_DEFER Marcel Ziswiler (1): ASoC: wm9712: fix replace codec to component Oder Chiou (1): ASoC: rt5514: Fix the issue of the delay volume applied again Pierre-Louis Bossart (1): ASoC: wm8804: Add ACPI support Ryan Lee (2): ASoC: max98373: Added speaker FS gain cotnrol register to volatile. ASoC: max98373: Added 10ms sleep after amp software reset Shuming Fan (1): ASoC: rt5682: Change DAC/ADC volume scale Srinivas Kandagatla (1): ASoC: q6routing: initialize data correctly Sébastien Szymanski (1): ASoC: cs4265: fix MMTLR Data switch control Yong Zhi (1): ASoC: Intel: Skylake: Acquire irq after RIRB allocation Yu Zhao (3): Revert "ASoC: Intel: Skylake: Acquire irq after RIRB allocation" sound: enable interrupt after dma buffer initialization sound: don't call skl_init_chip() to reset intel skl soc MAINTAINERS | 3 +- include/sound/hdaudio.h | 1 + include/sound/soc-dapm.h | 1 + include/uapi/sound/skl-tplg-interface.h | 106 ++++++++++++++++---------------- sound/hda/hdac_controller.c | 15 +++-- sound/soc/amd/acp-pcm-dma.c | 21 +++++++ sound/soc/codecs/cs4265.c | 4 +- sound/soc/codecs/max98373.c | 3 + sound/soc/codecs/rt5514.c | 8 +-- sound/soc/codecs/rt5682.c | 8 +-- sound/soc/codecs/sigmadsp.c | 3 +- sound/soc/codecs/tas6424.c | 12 +++- sound/soc/codecs/wm8804-i2c.c | 15 ++++- sound/soc/codecs/wm9712.c | 2 +- sound/soc/intel/boards/bytcr_rt5640.c | 26 ++++++++ sound/soc/intel/skylake/skl.c | 2 +- sound/soc/qcom/qdsp6/q6routing.c | 4 +- sound/soc/sh/rcar/adg.c | 5 ++ sound/soc/sh/rcar/core.c | 21 ++++++- sound/soc/sh/rcar/dma.c | 4 ++ sound/soc/sh/rcar/rsnd.h | 7 +++ sound/soc/sh/rcar/ssi.c | 16 +++-- sound/soc/soc-core.c | 4 +- sound/soc/soc-dapm.c | 4 ++ 24 files changed, 208 insertions(+), 87 deletions(-)

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[alsa-devel] Bug#908938: alsa-utils: amixer does not unmute pulseaudio
by Elimar Riesebieter 17 Sep '18

17 Sep '18

Control: forwarded -1 alsa-devel(a)alsa-project.org Hi alsa devs, please notice Debian Bug https://bugs.debian.org/cgi-bin/bugreport.cgi?bug=908938 and let us know whether a fix is possible. Thanks in advance -- Elimar * Łukasz Stelmach <steelman(a)post.pl> [2018-09-16 11:23 +0200]: > Package: alsa-utils > Version: 1.1.3-1 > Severity: normal > > Dear Maintainer, > > I've got PulseAudio running, so amixer (and alsamixer) controls the > default PA output. I use the following command to control muting of > audio. > > amixer -q -c 0 sset Master toggle > > It mutes the audio but can't unmute. Neither can > > amixer -q -c 0 sset Master unmute > > Only alsamixer and pavucontrol can unmute. -- Do you smell something burning or is it me?

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[alsa-devel] [PATCH] Documentation: soundwire: fix stream.rst markup warnings
by Randy Dunlap 17 Sep '18

17 Sep '18

From: Randy Dunlap <rdunlap(a)infradead.org> Fix kernel-doc markup warnings in soundwire/stream.rst: rc4/Documentation/driver-api/soundwire/stream.rst:177: WARNING: Explicit markup ends without a blank line; unexpected unindent. rc4/Documentation/driver-api/soundwire/stream.rst:203: WARNING: Explicit markup ends without a blank line; unexpected unindent. rc4/Documentation/driver-api/soundwire/stream.rst:248: WARNING: Explicit markup ends without a blank line; unexpected unindent. rc4/Documentation/driver-api/soundwire/stream.rst:277: WARNING: Explicit markup ends without a blank line; unexpected unindent. rc4/Documentation/driver-api/soundwire/stream.rst:304: WARNING: Explicit markup ends without a blank line; unexpected unindent. rc4/Documentation/driver-api/soundwire/stream.rst:328: WARNING: Explicit markup ends without a blank line; unexpected unindent. rc4/Documentation/driver-api/soundwire/stream.rst:352: WARNING: Explicit markup ends without a blank line; unexpected unindent. rc4/Documentation/driver-api/soundwire/stream.rst:364: WARNING: Explicit markup ends without a blank line; unexpected unindent. Fixes: 89634f99a83e ("Documentation: soundwire: Add more documentation") Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Cc: Sanyog Kale <sanyog.r.kale(a)intel.com> Cc: Shreyas NC <shreyas.nc(a)intel.com> Cc: Vinod Koul <vkoul(a)kernel.org> Cc: alsa-devel(a)alsa-project.org Cc: linux-doc(a)vger.kernel.org --- Documentation/driver-api/soundwire/stream.rst | 8 ++++++++ 1 file changed, 8 insertions(+) --- lnx-419-rc4.orig/Documentation/driver-api/soundwire/stream.rst +++ lnx-419-rc4/Documentation/driver-api/soundwire/stream.rst @@ -174,6 +174,7 @@ per stream. From ASoC DPCM framework, th .startup() operation. .. code-block:: c + int sdw_alloc_stream(char * stream_name); @@ -200,6 +201,7 @@ only be invoked once by respective Maste framework, this stream state is linked to .hw_params() operation. .. code-block:: c + int sdw_stream_add_master(struct sdw_bus * bus, struct sdw_stream_config * stream_config, struct sdw_ports_config * ports_config, @@ -245,6 +247,7 @@ stream. From ASoC DPCM framework, this s .prepare() operation. .. code-block:: c + int sdw_prepare_stream(struct sdw_stream_runtime * stream); @@ -274,6 +277,7 @@ stream. From ASoC DPCM framework, this s .trigger() start operation. .. code-block:: c + int sdw_enable_stream(struct sdw_stream_runtime * stream); SDW_STREAM_DISABLED @@ -301,6 +305,7 @@ per stream. From ASoC DPCM framework, th .trigger() stop operation. .. code-block:: c + int sdw_disable_stream(struct sdw_stream_runtime * stream); @@ -325,6 +330,7 @@ per stream. From ASoC DPCM framework, th .trigger() stop operation. .. code-block:: c + int sdw_deprepare_stream(struct sdw_stream_runtime * stream); @@ -349,6 +355,7 @@ all the Master(s) and Slave(s) associate framework, this stream state is linked to .hw_free() operation. .. code-block:: c + int sdw_stream_remove_master(struct sdw_bus * bus, struct sdw_stream_runtime * stream); int sdw_stream_remove_slave(struct sdw_slave * slave, @@ -361,6 +368,7 @@ stream assigned as part of ALLOCATED sta In .shutdown() the data structure maintaining stream state are freed up. .. code-block:: c + void sdw_release_stream(struct sdw_stream_runtime * stream); Not Supported

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[alsa-devel] [PATCH] ASoC: AMD: Fix capture unstable in beginning for some runs
by Akshu Agrawal 17 Sep '18

17 Sep '18

alsa_conformance_test -C hw:0,4 -p 1024 --debug would sometime show: TIME_DIFF(s) HW_LEVEL READ RATE 0.000095970 1024 1024 10670001.041992 0.042609555 1024 2048 24032.168372 0.021330364 1024 3072 48006.681930 0.021339559 1024 4096 47985.996337 The issue is that in dma pointer function we can have stale value of the register for current descriptor of channel. The register retains the number of the last descriptor that was transferred. Fix ensures that we report position, 0, till the one period worth of data is transferred. After one period of data, in handler of period completion interrupt we update the config and correct value of descriptor starts reflecting. Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- sound/soc/amd/acp-pcm-dma.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index c7e972b..f27e20c 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -1050,6 +1050,8 @@ static snd_pcm_uframes_t acp_dma_pointer(struct snd_pcm_substream *substream) bytescount -= rtd->bytescount; delay = do_div(bytescount, period_bytes); runtime->delay = bytes_to_frames(runtime, delay); + if (bytescount < period_bytes) + pos = 0; } else { buffersize = frames_to_bytes(runtime, runtime->buffer_size); bytescount = acp_get_byte_count(rtd); -- 1.9.1

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[alsa-devel] [PATCH v3 00/13] ASoC: Add support to WCD9335 Audio Codec
by Srinivas Kandagatla 17 Sep '18

17 Sep '18

Thankyou for reviewing v2 patchset, here is v3 addressing comments from v2. Qualcomm WCD9335 Codec is a standalone Hi-Fi audio codec IC. It is integrated in multiple Qualcomm SoCs like: MSM8996, MSM8976, and MSM8956 chipsets. WCD9335 had multiple functional blocks, like: Soundwire controller, interrupt mux, pin controller, Audio codec, MBHC, MAD(Mic activity Detection), Ultrasonic proximity and pen detection, Battery-voltage monitoring and Codec processing engine. Currently this patchset has been only tested with SLIMbus interface due to hardware avaiablity, but it can be easily made to work with both SLIMbus and I2C/I2S. This patchset adds very basic playback and capture support witout much fancy features. New features will be added once the basic support is in. This patchset is tested on top of linux-next on DB820c for both playback , capture paths and MBHC. Some parts of the code has been inherited from Qualcomm andriod kernel, so credits to various authors. WCD9335 can be interfaced via I2S/I2C or SLIMbus. Here is my test branch incase someone want to try this out: https://git.linaro.org/people/srinivas.kandagatla/linux.git/log/?h=wcd9335 Thanks, Srini Changes since v2 (https://lwn.net/Articles/761107/) - Updated dt bindings according to Robs Comments. - Added MBHC support with new bindings - Moved mfd irq in to single file as suggested by Lee Jones. - Addressed various minor comments on mfd driver by Lee Jones. - Rebased on top of 4.19-rc1 - renamed clsh as clsh-v2 as its v2 block Srinivas Kandagatla (13): ASoC: dt-bindings: update wcd9335 bindings. mfd: wcd9335: add support to wcd9335 core mfd: wcd9335: add wcd irq support ASoC: wcd9335: add support to wcd9335 codec ASoC: wcd9335: add CLASS-H Controller support ASoC: wcd9335: add basic controls ASoC: wcd9335: add playback dapm widgets ASoC: wcd9335: add capture dapm widgets ASoC: wcd9335: add audio routings ASoC: dt-bindings: Add WCD9335 MBHC specific properties ASoC: wcd9335: add mbhc support ASoC: apq8096: add slim support ASoC: apq8096: add headset JACK support .../bindings/sound/qcom,wcd9335.txt | 30 +- drivers/mfd/Kconfig | 14 + drivers/mfd/Makefile | 4 + drivers/mfd/wcd9335-core.c | 369 ++ include/linux/mfd/wcd9335/registers.h | 627 ++ include/linux/mfd/wcd9335/wcd9335.h | 65 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wcd-clsh-v2.c | 577 ++ sound/soc/codecs/wcd-clsh-v2.h | 49 + sound/soc/codecs/wcd9335.c | 5234 +++++++++++++++++ sound/soc/qcom/apq8096.c | 122 +- 12 files changed, 7092 insertions(+), 6 deletions(-) create mode 100644 drivers/mfd/wcd9335-core.c create mode 100644 include/linux/mfd/wcd9335/registers.h create mode 100644 include/linux/mfd/wcd9335/wcd9335.h create mode 100644 sound/soc/codecs/wcd-clsh-v2.c create mode 100644 sound/soc/codecs/wcd-clsh-v2.h create mode 100644 sound/soc/codecs/wcd9335.c -- 2.18.0

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[alsa-devel] [PATCH 0/7] slimbus: fix module loading
by srinivas.kandagatla＠linaro.org 17 Sep '18

17 Sep '18

From: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Hi Greg, Here are some of the fixes for issues found while testing module loading on DB820c debian setup. First 3 core fixes are to do with generating uevents, calling status callback in probe sequence if the device is up and one fix is able to match device id from device tree compatible strings. Second 4 ngd patches fixes few issues firstly with registration, secondly checking the validity of logical address assigned by remote. Lastly there are few typo fixes. Thanks, Srini Srinivas Kandagatla (7): slimbus: core: add support to uevent slimbus: core: update device status in probe slimbus: core: match device tree based devices correctly slimbus: ngd: validate logical address assigned by remote silmbus: ngd: register controller after power up. slimbus: ngd: return proper error code instead of zero slimbus: ngd: register ngd driver only once. drivers/slimbus/core.c | 37 +++++++++++++++++++++++++++++++++++-- drivers/slimbus/qcom-ngd-ctrl.c | 28 +++++++++++++++++----------- 2 files changed, 52 insertions(+), 13 deletions(-) -- 2.9.3

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[alsa-devel] [PATCH 0/3] ASoC: qdsp6: add compress offload support
by Srinivas Kandagatla 17 Sep '18

17 Sep '18

This patchset adds support to very basic MP3 playback compress offload support via QDSP ASM module. Tested this patchset on DB410c with APQ8016 and DB820c with APQ8096 using tinycompress library. Adding other codec support should be trivial w.r.t qdsp6 side, however there are other dependencies like compress UAPI header changes and FastRPC which are being worked in parallel. Once ready will post them! thanks, srini Srinivas Kandagatla (3): ASoC: q6asm-dai: dt-bindings: Add support to compress dais ASoC: qdsp6: q6asm: add support to MP3 format ASoC: qdsp6: q6asm-dai: Add support to compress offload .../devicetree/bindings/sound/qcom,q6asm.txt | 27 ++ sound/soc/qcom/Kconfig | 1 + sound/soc/qcom/qdsp6/q6asm-dai.c | 418 +++++++++++++++++- sound/soc/qcom/qdsp6/q6asm.c | 5 + 4 files changed, 450 insertions(+), 1 deletion(-) -- 2.18.0

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[alsa-devel] [RFC v2 - AAF PCM plugin 0/5] Introduce AVTP Audio Format (AAF) plugin
by Andre Guedes 15 Sep '18

15 Sep '18

Hi all, This is the version 2 of AAF plugin RFC series. For general context information about the AAF plugin, please refer to the cover letter from version 1 [1] and the documentation in doc/aaf.txt. Code-wise, the main changes in this new version are: 1) The media clock was generating timestamps in UTC coordinate. However, AVTP timestamps are in TAI coordinate system so the media clock was fixed to use TAI instead. 2) All plugin resources (e.g. buffers, fds) are now allocated/freed in hw_params/hw_free ioplug callbacks. 3) The plugin constraints are applied at entry point instead of hw_params callback. Also, in this new version, the commit message from patch "aaf: Implement Playback mode support" was improved so it discusses the AVTP protocol restrictions in terms of formats and rates as well as why the plugin uses the system clock to implement a media clock. For further information about what has been discussed in the previous version, please refer to the RFC v1 patch series which starts in [1]. All versions of this series can be also found in my alsa-plugins tree in github [2]. Best regards, Andre [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2018-August/139494.Even [2] https://github.com/aguedes/alsa-plugins Andre Guedes (5): aaf: Introduce plugin skeleton aaf: Load configuration parameters aaf: Implement Playback mode support aaf: Prepare for Capture mode support aaf: Implement Capture mode support Makefile.am | 3 + aaf/Makefile.am | 9 + aaf/pcm_aaf.c | 1214 +++++++++++++++++++++++++++++++++++++++++++++++++++++++ configure.ac | 9 + doc/aaf.txt | 142 +++++++ 5 files changed, 1377 insertions(+) create mode 100644 aaf/Makefile.am create mode 100644 aaf/pcm_aaf.c create mode 100644 doc/aaf.txt -- 2.14.4

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