We have a Dell AIO, there is neither internal speaker nor internal
mic, only a multi-function audio jack on it.
Users reported that after freshly installing the OS and plug
a headset to the audio jack, the headset can't output sound. I
reproduced this bug, at that moment, the Input Source is as below:
Simple mixer control 'Input Source',0
Capabilities: cenum
Items: 'Headphone Mic' 'Headset Mic'
Item0: 'Headphone Mic'
That is because the patch_realtek will set this audio jack as mic_in
mode if Input Source's value is hp_mic.
If it is not fresh installing, this issue will not happen since the
systemd will run alsactl restore -f /var/lib/alsa/asound.state, this
will set the 'Input Source' according to history value.
If there is internal speaker or internal mic, this issue will not
happen since there is valid sink/source in the pulseaudio, the PA will
set the 'Input Source' according to active_port.
To fix this issue, change the parser function to let the hs_mic be
stored ahead of hp_mic.
Cc: stable@vger.kernel.org
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Link: https://lore.kernel.org/r/20200625083833.11264-1-hui.wang@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The USB-audio mixer code holds a linked list of usb_mixer_elem_list,
and several operations are performed for each mixer element. A few of
them (snd_usb_mixer_notify_id() and snd_usb_mixer_interrupt_v2())
assume each mixer element being a usb_mixer_elem_info object that is a
subclass of usb_mixer_elem_list, cast via container_of() and access it
members. This may result in an out-of-bound access when a
non-standard list element has been added, as spotted by syzkaller
recently.
This patch adds a new field, is_std_info, in usb_mixer_elem_list to
indicate that the element is the usb_mixer_elem_info type or not, and
skip the access to such an element if needed.
Reported-by: syzbot+fb14314433463ad51625@syzkaller.appspotmail.com
Reported-by: syzbot+2405ca3401e943c538b5@syzkaller.appspotmail.com
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200624122340.9615-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ASoC: Fixes for v5.8
This is a collection of mostly small fixes, mostly fixing fallout from
some of the DPCM changes that went in last time around which shook out
some issues on i.MX and Qualcomm platforms. The addition of a managed
version of snd_soc_register_dai() is to fix resource leaks.
There's also a few new device IDs for x86 systems.
With the recent full-duplex support of implicit feedback streams, an
endpoint can be still running after closing the capture stream as long
as the playback stream with the sync-endpoint is running. In such a
state, the URBs are still be handled and they may call retire_data_urb
callback, which tries to transfer the data from the PCM buffer. Since
the PCM stream gets closed, this may lead to use-after-free.
This patch adds the proper clearance of the callback at stopping the
capture stream for addressing the possible UAF above.
Fixes: 10ce77e481 ("ALSA: usb-audio: Add duplex sound support for USB devices using implicit feedback")
Link: https://lore.kernel.org/r/20200616120921.12249-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For mono channel, SSI will switch to Normal mode.
In Normal mode and Network mode, the Word Length Control bits
control the word length divider in clock generator, which is
different with I2S Master mode (the word length is fixed to
32bit), it should be the value of params_width(hw_params).
The condition "slots == 2" is not good for I2S Master mode,
because for Network mode and Normal mode, the slots can also
be 2. Then we need to use (ssi->i2s_net & SSI_SCR_I2S_MODE_MASK)
to check if it is I2S Master mode.
So we refine the formula for mono channel, otherwise there
will be sound issue for S24_LE.
Fixes: b0a7043d5c ("ASoC: fsl_ssi: Caculate bit clock rate using slot number and width")
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Reviewed-by: Nicolin Chen <nicoleotsuka@gmail.com>
Link: https://lore.kernel.org/r/034eff1435ff6ce300b6c781130cefd9db22ab9a.1592276147.git.shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This patchset fixes a memory allocation issue and removes a 100%
reproducible use-after-free report thrown by KASAN in automated module
removal tests across multiple platforms.
All the credit goes to Bard Liao for root-causing the issue. DAIs may
be registered at the same time as a component, or when the topology is
loaded. This two-step registration causes the memory for
topology-based DAIs to allocated last, and conversely to be released
first by devres, before the component is released and the DAIs removed
from the component DAI list with snd_soc_unregister_dais().
When we remove a component, by the time we walk through its dai list
to unregister all dais, the dais allocated by the topology have been
freed already by devres and the list is corrupted with pointers that
are no longer valid.
The suggestion is to add an explicit devm_ based registration for
topology-based dais, so that each dai is cleanly removed from the
component dai list in the release operation before devres releases the
allocated memory.
Pierre-Louis Bossart (2):
ASoC: soc-devres: add devm_snd_soc_register_dai()
ASoC: soc-topology: use devm_snd_soc_register_dai()
include/sound/soc.h | 4 ++++
sound/soc/soc-devres.c | 37 +++++++++++++++++++++++++++++++++++++
sound/soc/soc-topology.c | 3 +--
3 files changed, 42 insertions(+), 2 deletions(-)
--
2.20.1
fix error "clock source 41 is not valid, cannot use"
[] New USB device found, idVendor=154e, idProduct=1002, bcdDevice= 1.00
[] New USB device strings: Mfr=1, Product=2, SerialNumber=0
[] Product: DCD-1500RE
[] Manufacturer: D & M Holdings Inc.
[]
[] clock source 41 is not valid, cannot use
[] usbcore: registered new interface driver snd-usb-audio
Signed-off-by: Yick W. Tse <y_w_tse@yahoo.com.hk>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/1373857985.210365.1592048406997@mail.yahoo.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
With EDMA, there is two dma channels can be used for dev_to_dev,
one is from ASRC, one is from another peripheral (ESAI or SAI).
If we select the dma channel of ASRC, there is an issue for ideal
ratio case, the speed of copy data is faster than sample
frequency, because ASRC output data is very fast in ideal ratio
mode.
So it is reasonable to use the dma channel of Back-End peripheral.
then copying speed of DMA is controlled by data consumption
speed in the peripheral FIFO,
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Reviewed-by: Nicolin Chen <nicoleotsuka@gmail.com>
Link: https://lore.kernel.org/r/424ed6c249bafcbe30791c9de0352821c5ea67e2.1591947428.git.shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The dma channel has been requested by Back-End cpu dai driver already.
If fsl_asrc_dma requests dma chan with same dma:tx symlink, then
there will be below warning with SDMA.
[ 48.174236] fsl-esai-dai 2024000.esai: Cannot create DMA dma:tx symlink
So if we can reuse the dma channel of Back-End, then the issue can be
fixed.
In order to get the dma channel which is already requested in Back-End.
we use the exported two functions (snd_soc_lookup_component_nolocked
and soc_component_to_pcm). If we can get the dma channel, then reuse it,
if can't, then request a new one.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Reviewed-by: Nicolin Chen <nicoleotsuka@gmail.com>
Link: https://lore.kernel.org/r/3a79f0442cb4930c633cf72145cfe95a45b9c78e.1591947428.git.shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Successful send of EOS command does not indicate that EOS is actually
finished, correct event to wait EOS is finished is EOS_RENDERED event.
EOS_RENDERED means that the DSP has finished processing all the buffers
for that particular session and stream.
This patch fixes EOS handling!
Fixes: 68fd8480bb ("ASoC: qdsp6: q6asm: Add support to audio stream apis")
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20200611124159.20742-3-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
We've had a couple of changes that introduce regressions with the
multi-cpu DAI solutions, and while trying to fix them we found
additional inconsistencies that should also go to stable branches.
Bard Liao (1):
ASoC: core: only convert non DPCM link to DPCM link
Pierre-Louis Bossart (3):
ASoC: soc-pcm: dpcm: fix playback/capture checks
ASoC: Intel: boards: replace capture_only by dpcm_capture
ASoC: SOF: nocodec: conditionally set dpcm_capture/dpcm_playback flags
sound/soc/intel/boards/glk_rt5682_max98357a.c | 2 +-
sound/soc/intel/boards/kbl_da7219_max98927.c | 4 +-
sound/soc/intel/boards/kbl_rt5663_max98927.c | 2 +-
.../intel/boards/kbl_rt5663_rt5514_max98927.c | 2 +-
sound/soc/soc-core.c | 22 ++++++++--
sound/soc/soc-pcm.c | 44 ++++++++++++++-----
sound/soc/sof/nocodec.c | 6 ++-
7 files changed, 62 insertions(+), 20 deletions(-)
base-commit: 8a9144c1cf
--
2.20.1
Additional checks for valid DAIs expose a corner case, where existing
BE dailinks get modified, e.g. HDMI links are tagged with
dpcm_capture=1 even if the DAIs are for playback.
This patch makes those changes conditional and flags configuration
issues when a BE dailink is has no_pcm=0 but dpcm_playback or
dpcm_capture=1 (which makes no sense).
As discussed on the alsa-devel mailing list, there are redundant flags
for dpcm_playback, dpcm_capture, playback_only, capture_only. This
will have to be cleaned-up in a future update. For now only correct
and flag problematic configurations.
Fixes: 218fe9b7ec ("ASoC: soc-core: Set dpcm_playback / dpcm_capture")
Suggested-by: Daniel Baluta <daniel.baluta@nxp.com>
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Reviewed-by: Daniel Baluta <daniel.baluta@gmail.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Link: https://lore.kernel.org/r/20200608194415.4663-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently USB-audio driver manages the auto-pm of the primary
interface although a card may consist of multiple interfaces.
This may leave the secondary and other interfaces left running
unnecessarily after the auto-suspend.
This patch allows the driver managing the auto-pm of all bundled
interfaces per card. The chip->pm_intf field is extended as
chip->intf[] to contain the array of assigned interfaces, and the
runtime-PM is performed to all those interfaces.
Tested-by: Macpaul Lin <macpaul.lin@mediatek.com>
Link: https://lore.kernel.org/r/20200605064117.28504-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
