Update the call sites of cpu_is_msm8960() to include an
additional check for the MSM8960AB target where
appropriate.
Change-Id: I54b1b9dccde2f21ada27bc64df02c2cb313ff1d1
Signed-off-by: Stepan Moskovchenko <stepanm@codeaurora.org>
Get session time command don't use the command state
value used for control commands. Don't set that state
value as it leads to volume command being timed out
which is waiting for this value to be reset.
Change-Id: I734a1ed7f4fda8d1367c27b71d7bfe5070f2ffc6
CRs-fixed: 377431
Signed-off-by: Swaminathan Sathappan <Swami@codeaurora.org>
Add CVS session variable with VoLTE string. MVM is
used wrongly with VoLTE string for passive stream create
command. This is replaced with correct CVS variable with
VoLTE string.
Change-Id: I1eb764a87368807cd7faad8ef4c7f3bff2e4328c
Signed-off-by: Venkat Sudhir <vsudhir@codeaurora.org>
- AFE does not support sampling rate 44.1k
- This fix addresses the issue by setting backend proxy device
sampling rate to 48k.
Change-Id: I4cd1ac6566d3230fa16fd70d99b8e758d8c606ad
CRs-fixed: 374556
Signed-off-by: Jayasena Sangaraboina <jsanga@codeaurora.org>
Microphone Bias may or may not have an external bypass capacitor
depending on the board configurations. Add the microphone bias
capless mode setting to the platform data for codec
CRs-fixed: 363941
Change-Id: Ia949d240b3b3122bc4bd6aca02ee5b6cd785d246
Signed-off-by: Bhalchandra Gajare <gajare@codeaurora.org>
After upgrading to kernel 3.4, there is 5 second delay
at the closing of PCM playback. The delay is due to missing
EOS from QDSP6 audio session manager causing pcm close function
of PCM platform driver to wait for 5 seconds. The root cause
for missing EOS is that ALSA dynmic PCM shutdown sequence has
changed. Now, trigger stop is called on the back-end DAI-LINK.
Furthermore, back-end trigger stop is called before front-end
trigger stop. Since sink stops rendering data, data at source
will never get consumed. EOS event will not arrive. As trigger
operation has to be atomic, it is very difficult to guarantee
sequence on shutting down various modules in QDSP6. The decision
is to abandon starting and stopping QDSP6 AFE port in trigger
function. This decision is considered acceptable as playback
and capture over SLIMBUS is no longer subject to strict sequence
which Q6 AFE port must be started after CODEC configuration.
Change-Id: I0cc1d8b7d058052d7fae55c84b6be46b5b0678e9
CRs-fixed: 373966
Signed-off-by: Patrick Lai <plai@codeaurora.org>
Add IIR2 filter interface for the wcd9304 codec.
Control the two 5 band IIR filters in the audio
codec through mixer controls. Enable individual
IIR filter bands and set band coefficients.
Change the IIR filter code to use snd_soc_write
instead of snd_soc_update_bits. If update bits
is used the IIR registers may not be correctly
updated.
Change-Id: I92fc147641e9eb270d8176f20445371fe5cc2f92
Signed-off-by: Asish Bhattacharya <asishb@codeaurora.org>
- Two different locks (spin lock and mutex lock) are used
to protect the shared data, this may cause kernel panic.
- Use spin lock to protect the shared data between interrupt
function and non-interrupt functions.
CRs-fixed: 375637
Change-Id: I10c93e2ca80d821908b93c22525695d89143825a
Signed-off-by: Helen Zeng <xiaoyunz@codeaurora.org>
If Q6 does not support DTS, LA driver has to exit gracefully.
Introducing a new member cmd_response in audio_client structure
to indicate format is supported or not, and use this cmd_response
to return error from open_write.
Change-Id: Icad30c787e8a5f26ead92584e163721b94ba509d
Signed-off-by: Srikanth Uyyala <suyyala@codeaurora.org>
Update the call sites of cpu_is_msm8930() to include checks
for the MSM8930AA() variant. Relevant drivers will be
updated for more driver-specific specific MSM8930AA checks
at a later time.
Change-Id: Iff1af7a5454ec56c40390682ce2b4b6d1d325c91
Signed-off-by: Stepan Moskovchenko <stepanm@codeaurora.org>
Per revised design decisions, cpu_is_msm8930() shall only
return true on 8930, and not on the 8627 variant. Modify
the cpu_is_xxx functions to reflect this change, and update
call sites accordingly.
Change-Id: I50b943f80c731717e6cd5d7fffb13aeec0f85a40
Signed-off-by: Stepan Moskovchenko <stepanm@codeaurora.org>
For the digital gain to be applied on the codec it is required to write
the digital gain register after the digital portion of the codec is
turned ON. This applies both for RX and TX digital path setup. Fix digital
gain setting sequence for RX and TX paths by rewriting the gain register
once the digital path is turned ON
Change-Id: I7b9c59c1b29b838845d27e406ba0f8a004c868b1
Signed-off-by: Bhalchandra Gajare <gajare@codeaurora.org>
msm8930 uses external mic biasing for headset mic. Correct the
microphone bias for headset microphone by setting it to external
biasing
Change-Id: I0324f6f9922e12a3263ff803a7fa882ac08a956c
Signed-off-by: Bhalchandra Gajare <gajare@codeaurora.org>
Add device tree support to sound soc audio drivers.
These drivers get registered to the alsa framework
and thus aid detection of soundcard.
Change the device tree entries to follow the new
design approach of having individual probe functions
for each audio interface.
Change-Id: Ie8f0bddd5ba6e2cfb66c6a23efdcb434c5082d7d
Signed-off-by: Phani Kumar Uppalapati <phanik@codeaurora.org>
During slow insertion of headset, it may be possible that the
headset is wrongly detected as a headphone. This results in
the headset mic being non-functional.
Fix by polling the microphone voltage after a plug is detected
as a valid Headphone. In case the microphone voltage settles to
a valid headset voltage, correct the plug type from Headphone to
Headset.
CRs-fixed: 370332
Change-Id: I5280542e857940f8d228c5f0ded1d2fde301168f
Signed-off-by: Bhalchandra Gajare <gajare@codeaurora.org>
WCD9304 supports 4 gain values for Earpiece PA. Only 2 of them are
exposed through the mixer control. Fix to add ability to program
all of the available gain levels
Change-Id: Ie768dc3aebb476ac47dd739654703f7e3cccfd5a
Signed-off-by: Bhalchandra Gajare <gajare@codeaurora.org>
If mbhc polling is active, enable mbhc path to avoid polling noise.
CRs-fixed: 347090
Change-Id: I3d9d1d6ec64620e24244091d735ef71c605c64fd
Signed-off-by: Joonwoo Park <joonwoop@codeaurora.org>
When ANC is enabled it's needed to enable mbhc's micbias to avoid mbhc
polling noise.
Change-Id: Ib9ddf28800c7c2d993089fecb20371f3d3444a52
Signed-off-by: Joonwoo Park <joonwoop@codeaurora.org>
- Remove codec dai shutdown under cpu dai check
- Don't shutdown codec dai when it's still used by
other capture stream
Change-Id: I1b9eae17ee95d05a8feb07b2369db3936b783e3f
Signed-off-by: Helen Zeng <xiaoyunz@codeaurora.org>
The official name for copper is MSM8974.
Switch to it.
Change-Id: Ifb241232111139912477bf7b5f2e9cf5d38d0f9e
Signed-off-by: Abhimanyu Kapur <abhimany@codeaurora.org>
Add primary and secondary PCM RX and TX to the routing
table to support AUX PCM over primary and secondary
audio interface.
Change-Id: Ieca8f0af6479087d86625bec1a38e6357bb5faa3
Signed-off-by: Shiv Maliyappanahalli <smaliyap@codeaurora.org>
Adding voip and voice driver support for copper target.
Change-Id: Ib64f08b79819895bea0507ee7a89748cd4c43016
Signed-off-by: Phani Kumar Uppalapati <phanik@codeaurora.org>
Problem Description:
Open and close the same set of slimbus ports after
certain iterations will fail port open, as that port
was not disconnected successfully.
Fix Description:
Handle sequence of closing slimbus ports. Store
the channel masks associated with each codec dai
and reset them after they are closed from slimbus
Then, release the close slimbus port event, after
all the channels are closed completely
Change-Id: Ie14b9f0920b37f905151b48f18df181503acc21d
CRs-fixed: 370761
Signed-off-by: Swaminathan Sathappan <Swami@codeaurora.org>
There is use case that the HDMI input goes through MI2S TX
interface to ADSP. Add 8-channel Multi-PCM TX support for this
use case.
Change-Id: Ie26e188da8d15988452103f11277944551344cd1
Signed-off-by: Subhash Chandra Bose Naripeddy <snariped@codeaurora.org>
Compressed driver changes for the DTS support
Change-Id: I595e638da78cced02142f4ee430afb7357eb336c
Signed-off-by: Srikanth Uyyala <suyyala@codeaurora.org>
There is use case that the HDMI input goes through MI2S
TX interface to ADSP. Add compressed TX support for
this use case.
Change-Id: I510e3e63b68ea1887e4c99ebf1c6f76112abbed5
Signed-off-by: Subhash Chandra Bose Naripeddy <snariped@codeaurora.org>
When there is a failure while opening q6asm capturing stream,
it releases the q6asm data structure which is accessed by
stream closing function and lead to a crash. Resolving the
issue by freeing the data structure in closing function
instead of during opening failure.
Change-Id: Ie45335a98be21d3b6035115241f657185a918be0
CRs-Fixed: 373438
Signed-off-by: Jay Wang <jaywang@codeaurora.org>
Currently when ever hw_params is called on a codec dai the sample
rate is set of all Interpolators(RX) and decimators(TX) which are
not active. This causes issues when one TX codec dai active with one
sample rate and a side tone is enabled from one active RX path to
another TX path with different sample rate. When First TX DAI is
enabled all the non-active decimators sample rate are set to its DAI
rate. When a RX Dai is enabled and the mixer commands are given for side
tone path to complete, it will cause the other TX path to be enabled
with sample rate of first TX DAI. So when second TX DAI hw_params is
called, since the decimator is already active its sample rate will not
be set. So only set sample rates of decimators and interpolators a DAI
is going to use.
CRs-Fixed: 370230
Change-Id: Ic916fc7680b51345cfcc83011a6df30c4b3320c8
Signed-off-by: Kiran Kandi <kkandi@codeaurora.org>
- Add Mixer controls for Reference Rx device to be
used as a endpoint2 in adm open for echo cancellation.
- Add logic to support echo cancellation for audio
recording with fluence topology.
Change-Id: I7b83c3fc1a19fef7826bc8c3671e2565e393566a
Signed-off-by: Jayasena Sangaraboina <jsanga@codeaurora.org>
There is a usecase to capture PCM from HDMI input through ADSP using
MI2S TX interface and play or route the same to the desired output
device. To support this more buffering is required. Add configuring
buffer size in multi channel PCM TX driver to support the usecase.
Change-Id: Icefa803b02cd5edac0f67fe2186b44030c38c8b9
Signed-off-by: Subhash Chandra Bose Naripeddy <snariped@codeaurora.org>
Initialize spare register variable. This causes Mic path
not to work if activated after voice call or combination of
Rx and TX path.
Change-Id: Ie431e2df4a8000489cc9763785c2182a608fcd3b
Signed-off-by: Venkat Sudhir <vsudhir@codeaurora.org>
CPU, platform and codec drivers do not support bespoke trigger.
Update the front end dai links trigger option from bespoke to
dpcm trigger post.
Also update the front end dai definition with proper aif name.
Change-Id: Iab655809f9b209bbe1e2cd51a51f191ad1e408d6
Signed-off-by: Santosh Mardi <gsantosh@codeaurora.org>
This is needed to support the compressed audio capture so that the
IOCTL commands for capture can pass to the ALSA SOC audio driver
Change-Id: I78c796275946e6e02f61aeab6579f3e9362f208b
Signed-off-by: Subhash Chandra Bose Naripeddy <snariped@codeaurora.org>
For compressed playback to bypass ADM, AFE connect command
Is used when the session is closed AFE disconnect command
Should be issued.
Add the support for AFE disconnect command.
Change-Id: I4cc4e867c1be36fbc2659520fd14a356c8405f7b
Signed-off-by: Santosh Mardi <gsantosh@codeaurora.org>
Add support for I2C\I2S interface for sitar codec along
With SLIMBUS interface.
Change-Id: I68666fd10cf9fb8d871d4b2a3d9b2e454dd1efe7
Signed-off-by: Asish Bhattacharya <asishb@codeaurora.org>
SRS parameters are not updated sometimes when new adm session is
opened and DSP picks up invalid key from default values and plays
some undesired demo ring or noise. Fix this by sending the SRS
parameters everytime adm session is opened. Add a flag to check
if SRS is ever enabled and send parameters only if flag is set
Change-Id: Ib22e6ff74e4376936caa510a632a6a3c3727e034
Signed-off-by: Sriranjan Srikantam <cssrika@codeaurora.org>
When external modem is paired with the apps processor, voice call
over BT uses Slimbus to transfer voice packets from the modem to
the application processor. Enable run-time PM so that Slimbus
driver keeps the clocks enabled in the BT usecase.
CRs-fixed: 368527
Change-Id: Ic4653e304bdef7ea6303c89918ce4cfa195ba968
Signed-off-by: Neema Shetty <nshetty@codeaurora.org>
- During voice and normal recording concurrency case, both voice
and recording streams share the same tx channel. If one stream
already opens the tx channel, another stream will get error when
trying to open the same channel again. If one stream ends and closes
the channel, but another stream will lose the sound if it's still using
it.
- To prevent the above issues, only send SND_SOC_DAPM_STREAM_START event
when capture active count is one. And send SND_SOC_DAPM_STREAM_STOP event
when capture active count is zero.
Change-Id: Ic6dcd5d8d1949c2b96d46915a4399a454075fbb7
CRs-Fixed: 357022
Signed-off-by: Helen Zeng <xiaoyunz@codeaurora.org>
The default value for the OCP current setting register was wrongly updated
causing OCP to trigger when the volume on headphone is maximum. Fix by
correcting the default current setting value for OCP
Change-Id: I9aa6bfe7e4f9dbacdbd1cf1030f83660418bc37f
Signed-off-by: Bhalchandra Gajare <gajare@codeaurora.org>
If the value of channel active variable is zero, don't decrease it.
Change-Id: Ic9cf9faacc10c37b30f2e3d91700015669061c24
Signed-off-by: Helen Zeng <xiaoyunz@codeaurora.org>
Accessing buffer pointer before initialization results in device crash.
To memset the buffer using physical address also results in a device crash.
Fixed this by initializing the buffer pointer before it is accessed and
memset the buffer using the virtual address.
Change-Id: I3b03f56cf988c9471c7988665bcec3c467e60bfc
Signed-off-by: Amal Paul <amal@codeaurora.org>
- When paused and press next button to play next song,
sometimes CMD_EOS fails to get Ack from LPASS and
is wait timeout for 5sec causing delay for next
playback start.
- In the failure case, even before trigger start
of driver is done,LPA driver receives pcm_close.
In this case, though EOS is issued, it is not
getting honored from LPASS.
- If trigger start not happen in LPA driver,
avoid CMD_EOS to LPASS as it will not be handled.
CRs-Fixed: 368519, 366926
Change-Id: Ib5ff21925bb44849b27ed4709b72efcccf412b5d
Signed-off-by: Laxminath Kasam <lkasam@codeaurora.org>
The current US/EURO headset detection algorithm is overwriting detected
unsupported headset detection with invalid headset detection.
Don't make it to overwrite to report unsupported headset correctly.
CRs-fixed: 359290, 368319
Change-Id: If2d02c0d68be1c6f3f2eb1aa89c7a08ffe166446
Signed-off-by: Joonwoo Park <joonwoop@codeaurora.org>
WCD9310 requires the charge pump to be enable for lineout (speaker)
as well. Fix to add charge pump in the routing for lineout
CRs-fixed: 369639
Change-Id: Ia6f699d1e659c68062d599820768a495d1f8d05a
Signed-off-by: Bhalchandra Gajare <gajare@codeaurora.org>
Updated dai links to use dpcm trigger posts based on
the latest alsa sound soc framework.
Change-Id: I80819c87e7307ed24874e3ea237ff9d09770818f
Signed-off-by: Phani Kumar Uppalapati <phanik@codeaurora.org>
- Add all codec driver common functionality in common files.
- Add separate files for callback function definitions.
- Add header files according to the platform.
Change-Id: I906811043d4bb33571f719f79988fbdb89f5c385
Signed-off-by: Harmandeep Singh <hsingh@codeaurora.org>
-RT proxy driver is using DMA alloc. Migrate them to
use ION memory to save memory and avoid stability
issues.
Change-Id: Idc2296bc5ecf76d6b846f204478c33f58423bb72
Signed-off-by: Laxminath Kasam <lkasam@codeaurora.org>
There are use cases that 16kHz sample rate AUX PCM needed
besides 8kHz. Add the amix control so that the AUX PCM
sample rate can be configured by application.
Change-Id: Ie2eaa5e844057065af4cedd64aa2c040f392721b
Signed-off-by: Shiv Maliyappanahalli <smaliyap@codeaurora.org>
CPU, platform and codec drivers does not support bespoke trigger.
update the front end dai links trigger option from bespoke to
dpcm trigger post.
Change-Id: I15bbb3b3fb1611bc3f77f729830fc8a6ba72b40a
Signed-off-by: Santosh Mardi <gsantosh@codeaurora.org>
The WCD9320 Codec Driver is an ALSA-compliant codec driver. This driver
constructs the internal codec audio paths with DAPM widgets and
controls and provides the controls to the upper layers to enable and
configure audio paths.
Change-Id: Iee29359bebfc838cd200732b7191a2eb6a2087ee
Signed-off-by: Kiran Kandi <kkandi@codeaurora.org>
wcd9310's MUX for calibration is connected only to micbias 2.
Therefore if MBHC micbias is other than micbias 2, it's required to route
override to micbias 2 during calibration.
Set cfilt which is associated with micbias 2 as fast mode during
calibration for the same reason.
CRs-fixed: 369684
Change-Id: I910a8d6747ad013fbd7a006662a650f048ffc545
Signed-off-by: Joonwoo Park <joonwoop@codeaurora.org>
Both audio path and MBHC accessory detection logic need master clock
enabled along with bandgap and clock block of CODEC. Clock control
is managed by the machine driver. Calls to clock control function from
audio and MBHC detection paths can be nested. As a result, reference
counter of master clock is incremen/decrement out of order and cause
master clock not enabled when audio path is enabled. Without master clock
CODEC will not consume data and this leads to SLIMBUS overflow error.
CRs-fixed: 370335
Change-Id: Id8d3b98496c95bb7ada9ca102fb867f52c0e500c
Signed-off-by: Joonwoo Park <joonwoop@codeaurora.org>
Add 4 FE to support multiple playback use case as mentioned
4 parallel pcm and 4 parallel compressed playback
1) system tone
2) local file playback
3) picture in picture channel 1
4) picture in picture channel 2
Change-Id: Ibab38f18146aeb20273546005586846a59d1cd46
Signed-off-by: Santosh Mardi <gsantosh@codeaurora.org>
This patch ads core.c, the file which implements the ioctls and
registers the devices
Change-Id: I30a1d8b561ecd15b6a862d6948368394edd49665
Signed-off-by: Vinod Koul <vinod.koul@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Asish Bhattacharya <asishb@codeaurora.org>
ASoC core will scan for path, get the list of widgets
when playback and capture is started.When a mixer
command is issued it needs to scan only the path
to find if back end or front end dai needs to be shutdown.
Change ehances asoc core path finding functionality
to provide support for different usecases.
Signed-off-by: Gopikrishnaiah Anandan <agopik@codeaurora.org>
Change the wigdet power up and power down sequnce
in the ASOC as per msm platform requirements.
Signed-off-by: Gopikrishnaiah Anandan <agopik@codeaurora.org>
When hardware params are set for the session alsa
core will query for the RX/TX channels for the
current back end dai.
Signed-off-by: Gopikrishnaiah Anandan <agopik@codeaurora.org>
Change adds debugfs support which gives information
on params, state of active pcm instances.
Signed-off-by: Gopikrishnaiah Anandan <agopik@codeaurora.org>
Hostless PCM nodes will not exchange data with
the userspace clients.Control paths will
be setup by userspace clients.
Signed-off-by: Gopikrishnaiah Anandan <agopik@codeaurora.org>
The Dynamic PCM core allows digital audio data to be dynamically
routed between different ALSA PCMs and DAI links on SoC CPUs with
on chip DSP devices. e.g. audio data could be played on pcm:0,0 and
routed to any (or all) SoC DAI links.
Dynamic PCM introduces the concept of Front End (FE) PCMs and Back
End (BE) PCMs. The FE PCMs are normal ALSA PCM devices except that
they can dynamically route digital audio data to any supported BE
PCM. A BE PCM has no ALSA device, but represents a DAI link and it's
substream and audio HW parameters.
Signed-off-by: Gopikrishnaiah Anandan <agopik@codeaurora.org>
Change creates a PCM stream for ASOC backend which will only be used
internally by kernel drivers.It provides existing ASoC components
drivers with a substream and access to any private data.
- During bootup, create a file system storing the
information about usb device state and name, and
broadcast the message to userspace whenever the
device is plugged or unplugged.
- Register the device at USB sound card initialization
for creating file system and toggle the state in that
filesystem when USB is plugged/unplugged. This will
also send a message to user space informing it about
state change.
Change-Id: Icd78273bb765a26bbc70725afebe8955de8f7315
Signed-off-by: Harmandeep Singh <hsingh@codeaurora.org>
Use SPECIAL format for AMR. Add mixer control to
input mode and rate.
Change-Id: I8746d86ce323744995575a22b6128b39daaa3d13
Signed-off-by: Helen Zeng <xiaoyunz@codeaurora.org>
[sboyd: Drop soc/msm parts]
Signed-off-by: Stephen Boyd <sboyd@codeaurora.org>
A few last-minute regression fixes for 3.4 final kernel.
All trivial, and Cc'ed to stable kernel.
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Merge tag 'sound-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"A few last-minute regression fixes for 3.4 final kernel. All trivial,
and Cc'ed to stable kernel."
* tag 'sound-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ASoC: wm8994: Fix AIF2ADC power down
ALSA: hda/idt - Fix power-map for speaker-pins with some HP laptops
ASoC: cs42l73: Sync digital mixer kcontrols to allow for 0dB
Some last minute fixes for ASoC. Small, focused changes to specific
drivers.
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Merge tag 'asoc-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Last minute fixes
Some last minute fixes for ASoC. Small, focused changes to specific
drivers.
BIOS on some HP laptops don't set the speaker-pins as fixed but expose
as jacks, and this confuses the driver as if these pins are
jack-detectable. As a result, the machine doesn't get sounds from
speakers because the driver prepares the power-map update via jack
unsol events which never come up in reality. The bug was introduced
in some time in 3.2 for enabling the power-mapping feature.
This patch fixes the problem by replacing the check of the persistent
power-map bits with a proper is_jack_detectable() call.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=43240
Cc: <stable@vger.kernel.org> [v3.2+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some of the Digital mixer kcontrol max values were off by 1 not allowing a max of 0dB.
Signed-off-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
Slightly more than expected as rc7, but all are reasonablly small fixes.
A few additions of HD-audio fixup entries, a couple of other regression
fixes including a revert, and a few other trivial oneliners.
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Merge tag 'sound-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"Slightly more than expected as rc7, but all are reasonablly small
fixes. A few additions of HD-audio fixup entries, a couple of other
regression fixes including a revert, and a few other trivial
oneliners."
* tag 'sound-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ASoC: sh: fix migor.c compilation
ALSA: HDA: Lessen CPU usage when waiting for chip to respond
Revert "ALSA: hda - Set codec to D3 forcibly even if not used"
ALSA: hda/realtek - Call alc_auto_parse_customize_define() always after fixup
ALSA: hdsp - Provide ioctl_compat
ALSA: hda/realtek - Add missing CD-input pin for MSI-7350 mobo
ALSA: hda/realtek - Add a fixup for Acer Aspire 5739G
ALSA: echoaudio: Remove incorrect part of assertion
An API update which wasn't sufficiently thorough in updating the tree...
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Merge tag 'asoc-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Build fix for SH in 3.4
An API update which wasn't sufficiently thorough in updating the tree...
Fix a recent compilation breakage, caused by a change in SH clock API.
Signed-off-by: Guennadi Liakhovetski <g.liakhovetski@gmx.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
When an IRQ for some reason gets lost, we wait up to a second using
udelay, which is CPU intensive. This patch improves the situation by
waiting about 30 ms in the CPU intensive mode, then stepping down to
using msleep(2) instead. In essence, we trade some granularity in
exchange for less CPU consumption when the waiting time is a bit longer.
As a result, PulseAudio should no longer be killed by the kernel
for taking up to much RT-prio CPU time. At least not for *this* reason.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Tested-by: Arun Raghavan <arun.raghavan@collabora.co.uk>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This reverts commit 785f857d1c.
The commit causes a problem with the wrong D3 state after suspend
because the call of hda_set_power_state() involves with the power-up
sequence, which changes the power_count, and this confuses the resume
sequence that checks the power_count as well.
Originally, this go-to-D3 sequence should be a simple task without the
power-up sequence. But, it'd need some proper sanity checks in the
case of power-saved state, so it's not too easy to write now in the
3.4-rc cycle.
In short, the safest option now is to revert this affecting commit.
Of course, we need to clean up and robustify the power-saving code
better for 3.5 kernel.
Reported-by: Konstantin Khlebnikov <khlebnikov@openvz.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The call for alc_auto_parse_customize_define() must be done after the
fixup pre-probe initialization. Otherwise SKU_IGNORE fixup won't work
properly (e.g. HP RP5800 with ALC662 codec).
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_hdsp uses its own ioctls to acquire config- and status information.
Expose the corresponding ioctl handler via ioctl_compat, so that 32bit applications can use it on 64bit kernels.
Signed-off-by: Andre Schramm <andre.schramm@iosono-sound.com>
Reviewed-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Acer Aspire 5739G requires the same fix-up for 4930G to support the
surround / bass speakers.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=43180
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This assertion seems to imply that chip->dsp_code_to_load is a pointer.
It's actually an integer handle on the actual firmware, and 0 has no
special meaning.
The assertion prevents initialisation of a Darla20 card, but would also
affect other models. It seems it was introduced in commit dd7b254d.
ALSA sound/pci/echoaudio/echoaudio.c:2061 Echoaudio driver starting...
ALSA sound/pci/echoaudio/echoaudio.c:1969 chip=ebe4e000
ALSA sound/pci/echoaudio/echoaudio.c:2007 pci=ed568000 irq=19 subdev=0010 Init hardware...
ALSA sound/pci/echoaudio/darla20_dsp.c:36 init_hw() - Darla20
------------[ cut here ]------------
WARNING: at sound/pci/echoaudio/echoaudio_dsp.c:478 init_hw+0x1d1/0x86c [snd_darla20]()
Hardware name: Dell DM051
BUG? (!chip->dsp_code_to_load || !chip->comm_page)
Signed-off-by: Mark Hills <mark@pogo.org.uk>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As good as nothing exciting here; just a few trivial fixes for
various ASoC stuff.
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Merge tag 'sound-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound sound fixes from Takashi Iwai:
"As good as nothing exciting here; just a few trivial fixes for various
ASoC stuff."
* tag 'sound-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ASoC: omap-pcm: Free dma buffers in case of error.
ASoC: s3c2412-i2s: Fix dai registration
ASoC: wm8350: Don't use locally allocated codec struct
ASoC: tlv312aic23: unbreak resume
ASoC: bf5xx-ssm2602: Set DAI format
ASoC: core: check of_property_count_strings failure
ASoC: dt: sgtl5000.txt: Add description for 'reg' field
ASoC: wm_hubs: Make sure we don't disable differential line outputs
Nothing terribly exciting here, a bunch of small and simple fixes
scattered around the place.
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Merge tag 'asoc-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Updates for 3.4
Nothing terribly exciting here, a bunch of small and simple fixes
scattered around the place.
Signed-off-by: Oleg Matcovschi <oleg.matcovschi@ti.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@ti.com>
As s3c2412-i2s is using the s3c_i2sv2 it should call the more specialised
s3c_i2sv2_register_dai instead of simply calling snd_soc_register_dai.
Without this call the snd_soc_dai_ops structure isn't initialised correctly.
Signed-off-by: Heiko Stuebner <heiko@sntech.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The core allocates the live copies, we shouldn't try to duplicate it and
were buggy trying to do so as we were using uninitialised data for the
control data.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* commit f9dfbf9 "ASoC: tlv320aic23: convert to soc-cache" leads to
a bug preventing resumeof the codec as regmap expects a 9 bits data
register but 0xFFFF is passed in tlv320aic23_set_bias_level and this
values gets cached preventing any write to the TLV320AIC23_PWR
register as the final value produced by regmap is (register << 9) | value
* this patch solves the problem by only working on the 9 bits the
register contains.
Signed-off-by: Eric Bénard <eric@eukrea.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
