The token field is for identification to determine response packet's
source. Fill this field correctly to address afe_loopback timeout.
Signed-off-by: Joonwoo Park <joonwoop@codeaurora.org>
If the vote for pm runtime is done while codec shutdown, it is possible
that the runtime pm vote occurs even before the slimbus port for tx/rx
audio channel is disconnected. This can cause problem in audio playback/
record. Fix by moving the vote for runtime pm after slimbus port has
been disconnected
Change-Id: I711bc5cfee5b832575ea0b91cf68e826f1a3c0f5
Signed-off-by: Bhalchandra Gajare <gajare@codeaurora.org>
Headphone path can either be in CLASS G mode or CLASS AB mode.
The class G mode in the codec hardware is used to adjust the
supply voltage of the PA's according to the signal level. This
makes the system as efficient as possible. Fix the register
sequence to enable headphone in class G mode
CRs-fixed: 380598
Change-Id: I110c4e0ea958ef55c0b407c566deb7da58f4d99a
Signed-off-by: Bhalchandra Gajare <gajare@codeaurora.org>
Upgrade the board file to change the microphone bias used by MBHC to
2.7 Volts. The button voltage ranges need to be updated accordingly
Change-Id: Ia2f91271864e0f8fe25b866bff8006af0dd6f20b
Signed-off-by: Bhalchandra Gajare <gajare@codeaurora.org>
The debugfs TRRS entry is for overriding headset button polling.
Fix regression not to start button polling when this flag is set.
CRs-fixed: 386038
Change-Id: I469c366bc111f37ecbb46708d2800200dd3d7584
Signed-off-by: Joonwoo Park <joonwoop@codeaurora.org>
Some of accessories on the headset jack takes longer to settle down
voltage. Make adjust time to be configurable so those accessories can
be detected correctly.
Change-Id: I3c2f68c8a4bb1a8f94669bd910728f014ee39874
Signed-off-by: Joonwoo Park <joonwoop@codeaurora.org>
Read the codec specific data from device tree instead of board file.
Change-Id: Iad382b89692903d2434b63d34c7121fe0b4b9dda
Signed-off-by: Kiran Kandi <kkandi@codeaurora.org>
Resetting this channel actual flag without
checking whether the set of channels are already in
use, could cause failures in disconnecting those
set of ports associated with that channels.
Change-Id: If0b917023c8f6d11d6b5cd92708715e10a1408ab
CRs-fixed: 384055
Signed-off-by: Swaminathan Sathappan <Swami@codeaurora.org>
Update the number of supported voice networks
in the ACDB driver and change the voice driver
to support concurrent calibration for VOLTE,
VOIP & voice.
Change-Id: I945fa40cbb4ac079a79fa0cb829971f1aa9d07f6
Signed-off-by: Ben Romberger <bromberg@codeaurora.org>
When SRS Tru media topology is enabled the mixer controls set from
userspace are not sent to driver as the validation fails with
the missing field initialization in the structure. Update the same
to receive the SRS parameters from userspace
Change-Id: Ia2bf800a540adf3bcc2a99fc6e0d9b043c826272
Signed-off-by: Subhash Chandra Bose Naripeddy <snariped@codeaurora.org>
This change sets rate on pcm0 root clock (pcm0_src_clk)
and pcm out enable (PCM_OE) clock for AUX PCM CPU DAI.
Once the rate is set on the source clocks, corresponding
branch clocks get enabled.
Change-Id: I59bc339d69fe836f613c1bebf7561184dced2dcf
Signed-off-by: Phani Kumar Uppalapati <phanik@codeaurora.org>
After recent movement of afe_start to prepare the interrupts start
coming earlier from proxy port.
This results in proxy port driver not ready for interrupts. Start the
timer later to ensure processing of packet once trigger start happens.
Change-Id: Ie82d982b6c147afdef46a8c7bfe7234dd14486f1
Signed-off-by: Asish Bhattacharya <asishb@codeaurora.org>
If interrupt handler is not quicker than voltage ramp up by button
release, driver requested measurement won't see stable voltage.
Enhance button press detection performance by comparing voltage
measurements from codec hardware only.
CRs-fixed: 376825
Change-Id: I294239df02fb5afeb3527dda85924c06ab15e54c
Signed-off-by: Joonwoo Park <joonwoop@codeaurora.org>
Enhance button release detection performance by adopting dynamic release
threshold adjusting logic.
Button release is now detected more quickly.
Change-Id: I3a2379e10663cf91df671e8a3894b8805d1ccf9c
Signed-off-by: Joonwoo Park <joonwoop@codeaurora.org>
copy_to_user and copy_from_user can block(go to sleep). These two functions
should be removed from spinlock protection.
Change-Id: I0c10796ece4ac218600ef3b214c88a06004a0a0d
CRs-fixed: 381839
Signed-off-by: Helen Zeng <xiaoyunz@codeaurora.org>
If Mic Bias is high Z by default and when the LDOH is enabled,
the Mic Bias output can pull-up to a non-zero voltage if there
is no loading or if the load leakage is very small. Pulling
down the Mic Bias output so it is not in high Z to eliminate
the floating voltage.
Change-Id: I38e76a8c03107879727564f177b2713c9dfa4631
CRs-Fixed: 368898
Signed-off-by: Kuirong Wang <kuirongw@codeaurora.org>
The value for the amix command with the numeric strings is
the actual value not the order of the string in the array
of text so the put function has to be updated to handle it.
For example, the following amix comand:
amix 'Internal BTSCO SampleRatee' 16000'
The value passed from the alsa-lib is not 2, i.e., the 2nd
paramater in the command string:
static const char *btsco_rate_text[] = {"8000", "16000"};
Instead, it is 16000.
CRs-fixed: 364832
Change-Id: Ie7c83a460900b54e2b317e1c77a064efc22e6bcd
Signed-off-by: Kuirong Wang <kuirongw@codeaurora.org>
-Voice call recording is not working because of missing
backends configuration for incall record Tx/Rx.
Change-Id: Ie69112e2929ed98a5bc19164cf9a9c66d73cc8dc
Signed-off-by: Vidyakumar Athota <vathota@codeaurora.org>
Update DAI link to connect slimbus RX/TX port 0 to CPU.
Change-Id: Id8aa2ca92ea9b1e0d51b6d3ccf113860a9147c44
Signed-off-by: Joonwoo Park <joonwoop@codeaurora.org>
Add audio OCMEM driver support to exercise On-Chip
Memory (OCMEM) for low power audio and voice. The
driver is implemented as standalone and it gets
exercised based on the usecase. Also, this design
reduces the latency associated with OCMEM handshaking
protocol. The audio OCMEM driver is enabled by default
with a Kconfig option using select. Add device tree node
for audio OCMEM to retrieve platform data.
Change-Id: Iba46ce675fc03843d88cd7cf2aa9bc92fe70a955
Signed-off-by: Phani Kumar Uppalapati <phanik@codeaurora.org>
Register slimbus CPU dai link to support slimbus data path.
Change-Id: I3584306ac1e0ad6561a19cecfe71f2a63aadafa9
Signed-off-by: Joonwoo Park <joonwoop@codeaurora.org>
During rapid open and close of slimbus ports, it may happen that when
port open is performed, the port is not disconnected from the previous
open. This will cause inconsistent state and may result in failure to
playback audio. Fix by waiting for the slimbus port disconnect to happen
before opening the port.
CRs-fixed: 375689
Change-Id: Id1303deae296eb6842074837183ab231aa2b4dad
Signed-off-by: Bhalchandra Gajare <gajare@codeaurora.org>
In AFE-PCM RX and TX dai links of MPQ8064 machine driver there
was a conflict in codec dai name and codec name, with this ALSA
Framework is not creating the node in /dev/snd/
CRs-Fixed: 377509
Change-Id: I5337b216a3d0a2cdc36292ccdafe3e144e7f1d41
Signed-off-by: Santosh Mardi <gsantosh@codeaurora.org>
The wcd9310 codec driver already can detect and report unsupported headset
plugging. Create headset jack with unsupported headset mask to be able to
report via sound core.
Change-Id: I0119d01c039362cc7b185f9f3407d78c958bc49a
Signed-off-by: Joonwoo Park <joonwoop@codeaurora.org>
HDMI 1.3 supports Multi channel PCM up to 8 channels with sample
size of 16/20/24 bits and sample rate of 32, 44.1, 48, 96, 176.4,
192K. This patch add supports for 6 channel PCM at 48K sample rate
with sample size of 16 bits.
CRs-Fixed: 380370
Change-Id: I01cb7eb509a0d21072e2e8dcc63624384a1edb0d
Signed-off-by: Asish Bhattacharya <asishb@codeaurora.org>
Found that the register sequence for enabling recording on analog
microphone was updating a wrong register. Fix by correcting the
register sequence
CRs-fixed: 378189
Change-Id: I3598cbc77b279684b28677943ac13f446bc2f78e
Signed-off-by: Bhalchandra Gajare <gajare@codeaurora.org>
When closing the slimbus ports, clear only the slimbus
ports which were active at the time of closing on that
slimbus interrupt, which will avoid clearing further
interrupts for other slimbus ports
Change-Id: I43ce9963c0ecb4b8fb79527b8a9c6b13d7780aad
CRs-fixed: 380535
Signed-off-by: Swaminathan Sathappan <Swami@codeaurora.org>
- Add a front end DAI link for compressed audio driver.
- Add device tree specific changes to sound soc compressed
drivers which will help the detection of sound card.
Change-Id: I4a8076df8c82cd4e444fc0d68e8f7a228bd1dc02
Signed-off-by: Harmandeep Singh <hsingh@codeaurora.org>
Some physical pools ids may become unsupported in future targets.
Remove target specific information from API to make it robust against
such architectural changes.
Also different platforms such as MDM, MSM might support some physical
pools but not the other, so choosing a generic name which will could be
mapped internally to different memory pools
Change-Id: I4f003662d9a2a28c17eefa5230530b8608b26c09
Signed-off-by: Harmandeep Singh <hsingh@codeaurora.org>
Volume and mute settings before voice call is started are cached
in the driver, apply the cached settings at call start.
Change-Id: Iabc1f47c46a8e986c79106545ac3ee977fbca99c
Signed-off-by: Neema Shetty <nshetty@codeaurora.org>
There is a usecase where compressed data is sent over HDMI IN to
ADSP. The format of compressed is detected in ADSP and sent through
the meta data to compressed driver. Add support for meta data in
compressed TX for this use case.
Change-Id: Idbb18fe4a0ad828e9c2e9d7beec048b3cedf002d
Signed-off-by: Subhash Chandra Bose Naripeddy <snariped@codeaurora.org>
QDSP6 AFE module produces error message whenever afe
loopback gain control command is issued. The reason is that
loopback gain control function sets wrong payload size.
Make change to set appropriate payload size for a given
SET_PARAM command
Change-Id: Ida2bf76baf56c35e89fe29f887f5b43af8bceabe
Signed-off-by: Patrick Lai <plai@codeaurora.org>
AV switch and US Euro headset switches are not supported
on apq8064 target. Hence removing unnecessary gpio pins
configuration.
Change-Id: Ia4747b59b63b0bf7c37054fb1bcebfc54079b481
Signed-off-by: Ravi Kumar Alamanda <ralama@codeaurora.org>
Add support for controlling pcm audio volume in DSP through
Multimedia5. Add TLV mixer control to set the pcm stream volume
Change-Id: Ie5f50c4f47ea57fe4be0aef1320c79a9d3fe7600
Signed-off-by: Subhash Chandra Bose Naripeddy <snariped@codeaurora.org>
If the vote for pm runtime is done while codec shutdown, it is possible
that the runtime pm vote occurs even before the slimbus port for tx/rx
audio channel is disconnected. This can cause problem in audio playback/
record. Fix by moving the vote for runtime pm after slimbus port has
been disconnected
Change-Id: I959a83be7bc381e80dfc0176c50cb60e59ce227b
Signed-off-by: Swaminathan Sathappan <Swami@codeaurora.org>
Signed-off-by: Patrick Lai <plai@codeaurora.org>
When session CLOSE command is sent right before session RUN command
is acknowledged, callback function can mistakenly think that
the next received acknowledgement is for CLOSE command instead of
RUN command. This triggers driver to send memory unmap command to
the Q6 while it is still processing the CLOSE command. Eventually,
this leads to an invalid memory access and causes Q6 crash.
Change-Id: Ib5d560fbcb7e8ced79cc1075a9f6bea3b55a86b6
CRs-Fixed: 377281
Signed-off-by: Jay Wang <jaywang@codeaurora.org>
ALSA framework in kernel 3.4 requires all CPU DAIs to be routed to
the repective back-end input or output. Add the routing for STUB_1,
SLIMBUS_1, SLIMBUS_3, and SLIMBUS_4 CPU DAIs.
CRs-Fixed: 376720
Change-Id: Ie7799777d500194c53520320302e667f2ed07480
Signed-off-by: Neema Shetty <nshetty@codeaurora.org>
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>
