When HDMI PCM devices are opened in a specific order, with at least one
HDMI/DP receiver connected, ALSA PCM open fails to -EBUSY on the
connected monitor, on recent Intel platforms (ICL/JSL and newer). While
this is not a typical sequence, at least Pulseaudio does this every time
when it is started, to discover the available PCMs.
The rootcause is an invalid assumption in hdmi_add_pin(), where the
total number of converters is assumed to be known at the time the
function is called. On older Intel platforms this held true, but after
ICL/JSL, the order how pins and converters are in the subnode list as
returned by snd_hda_get_sub_nodes(), was changed. As a result,
information for some converters was not stored to per_pin->mux_nids.
And this means some pins cannot be connected to all converters, and
application instead gets -EBUSY instead at open.
The assumption that converters are always before pins in the subnode
list, is not really a valid one. Fix the problem in hdmi_parse_codec()
by introducing separate loops for discovering converters and pins.
BugLink: https://github.com/thesofproject/linux/issues/1978
BugLink: https://github.com/thesofproject/linux/issues/2216
BugLink: https://github.com/thesofproject/linux/issues/2217
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20200703153818.2808592-1-kai.vehmanen@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
