For the use case of Playback over HDMI device, the number of
channels supported varies with the sink capabilities.
Add support to fix the channels based on the sink capabilites
configured by userspace for msm8960
Change-Id: Iba4f1fead17832d7832fabdcba02da671e5fd005
Signed-off-by: Subhash Chandra Bose Naripeddy <snariped@codeaurora.org>
There is a missing step to clear OTHR_RESET_CTR before power down.
It could cause a glitch and power hit without it. It has to be reset.
original gerrit is https://review-android.quicinc.com/#/c/191312/
This is cherry-pick gerrit by developer to jb_2.5
Change-Id: Id96070352480be6ea412844d4f71765e2df76afe
CRs-fixed: 420142
Signed-off-by: Fred Oh <fred@codeaurora.org>
Add support for configuring ISO or ARIB stereo mixing
coefficients. This change introduces a function with whichi,
AAC multi-channel driver can configure ISO or ARIB coefficients
according to configuration done by the client.
Change-Id: I72b74033532a276fa3bc1d305a04720ff6767409
Signed-off-by: Amal Paul <amal@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>
Signed-off-by: Iliyan Malchev <malchev@google.com>
Conflicts:
sound/soc/msm/apq8064.c
-In afe_open, trying to unlock mutex on variable
that is already freed.
-correct the sequence by unlock mutex first then
free the variable.
CRs-Fixed: 423196
Change-Id: If7df0e0f46c7d16843c6d52ae821974cc74539ff
Signed-off-by: Laxminath Kasam <lkasam@codeaurora.org>
(cherry picked from commit 4c120ae85c6c5ac2adfdf61956b45d02461a93e3)
WCD9310 codec supports loopback through IIR filters. Either IIR1
or IIR2 can be used to setup loopback path. Add support and routing
to enable loopback through IIR2 filter path
CRs-fixed: 364236
Change-Id: I52cb455ad31aa129f81bac3852ac50d4a5a99bc2
Signed-off-by: Bhalchandra Gajare <gajare@codeaurora.org>
Signed-off-by: Kiran Kandi <kkandi@codeaurora.org>
(cherry picked from commit 8ba4446a571ed5f7033dc6cb651e19f5d9485abb)
Time stamp is unsigned. There is no need to check for
negative value.
Change-Id: I15e380f81e46908f6a9fe8d29c174de1b7c33173
CRs-fixed: 423372
Signed-off-by: Patrick Lai <plai@codeaurora.org>
The clock is not usable when either the clock source changes between
external MCLK and internal RCO or the bandgap mode changes to fast mode.
In such cases, turn off master clock and turn on again later to avoid
codec from going into bad state
CRs-fixed: 442464
Change-Id: Id8d07bbbd4bf6c7be16016800d19ab332b960d1a
Signed-off-by: Joonwoo Park <joonwoop@codeaurora.org>
For the playback use of compressed pass through on HDMI device,
post processing features are not supported and should not be sent to
DSP. Hence the change is now to send the volume only for the playback
of decode and render in DSP and skip for compressed pass through
CRs-Fixed: 442140
Change-Id: I2125f08baf1676f6bb1d13463a1f01dfe280963f
Signed-off-by: Subhash Chandra Bose Naripeddy <snariped@codeaurora.org>
There is maximum number of COPPs ASM session can be routed
to even though there are far more number of AFE ports.
Add the check to make sure caller is not routing to more
than ADM can actually support.
Change-Id: Iee97dd7e5d21820c87244a422fa252c2228185f9
Signed-off-by: Patrick Lai <plai@codeaurora.org>
Remove unnecessary check as it should be reasonable
to assume valid parameter would be issued through
debugfs interface
Change-Id: I1a23d8c5324b813bf9501c5462684234926b5dcc
Signed-off-by: Patrick Lai <plai@codeaurora.org>
When playing a full scale tone with highest gain in the codec, OCP
(Over Current Protection) event is generated. This is caused due to
lower switching frequency for the class G amplifier on headphone
path. Fix the false OCP event trigger by appropriately setting the
class G switching frequency
CRs-fixed: 415990
Change-Id: I9563ab2d06c970b47693293175fe7ae4a711348c
Signed-off-by: Bhalchandra Gajare <gajare@codeaurora.org>
For compressed and LPA drivers, set pending_buffers flag to 1 only if the client
issues a trigger_start command, not a pause_release command. This flag
is used to initiate writes to the dsp.
Change-Id: I503156863646210b460613a0b9a26a6bfb8854ed
CRs-Fixed: 442269, 449126
Signed-off-by: Haynes Mathew George <hgeorge@codeaurora.org>
Function which converts port ID to port index can return
error value if port ID is invalid. Caller should not
blindly use return value without checking
Change-Id: I02b8112b6714e3364758907e7078bcbb88d45b6b
Signed-off-by: Patrick Lai <plai@codeaurora.org>
For compressed drivers, allow the drain ioctl only if
there are pending writes. Return a proper error (-EWOULDBLOCK) otherwise.
Change-Id: I4799351a8930c44d88d6968802df98896fc1531a
Signed-off-by: Haynes Mathew George <hgeorge@codeaurora.org>
CRs-Fixed: 445370
If a drain ioctl on a compressed driver has been interrupted
by a reset ioctl, we need to return a proper error code for the
drain ioctl
Change-Id: I15f42a22b1cc13b3a0e2ae873adfcd1db2ac1e53
Signed-off-by: Haynes Mathew George <hgeorge@codeaurora.org>
CRs-Fixed: 445370
When playback is paused and proxy device disabled, afe is
not closed. This results in failure of proxy port open in the second
iteration. Fix the same by closing and opening the proxy.
Change-Id: Idfa94e5756d87e59dc0101fd688d6accfe6287e7
CRs-Fixed: 447001
CRs-Fixed: 446504
Signed-off-by: Satya Krishna Pindiproli <satyak@codeaurora.org>
Update the Compander register settings to remain the default
settings after the system bootup and optimize compander
configuration sequence.
Change-Id: Ib475f3bb84414546b4ead1d3f013dc173317e1b3
Signed-off-by: Kuirong Wang <kuirongw@codeaurora.org>
- When a running WFD session is paused/disconnected, there is
a delay of about 10-15 seconds in switching audio from WFD
sink to target.
- pcm_write is blocked because pausing/disconnecting the WFD
session closes the backend DAI before a new link is established.
- Avoid closing afe if front end PCM session is still active.
CRs-Fixed: 444163
CRs-Fixed: 425512
Change-Id: I1ad6de1e201aa6130c397007219c0d370c7be2a1
Signed-off-by: Satya Krishna Pindiproli <satyak@codeaurora.org>
- Compander1 never gets enabled though the user space sets the
corresponding mixer control.
- Compander switch kcontrols are configured wrongly as integer
type with max value as 0.
- Correct kcontrol creation to fix the issue.
CRs-Fixed: 441091
Change-Id: Iad99b44c9226da3bcc1132e61ea989a0543ae56e
Signed-off-by: Ravi Kumar Alamanda <ralama@codeaurora.org>
-Non tunnel playback is use case where compressed
data is sent to driver(for decoding) and
caller expects PCM data back from the driver.
-Add non-tunnel driver for amr-wb+.
-Add support for amr-wb+ in asm driver
Change-Id: I266e70dff0466a20b0435da840a31a8a1774cf35
Signed-off-by: Ajit Khare <ajitk@codeaurora.org>
Add virtual ANC widgets which are used to enable ANC
before PA is turned on, and to disable ANC before TX is
turned off.
CRs-fixed: 387904
Change-Id: I8c2fa670062f1486499c79653a2c223f7daba84f
Signed-off-by: Damir Didjusto <damird@codeaurora.org>
During error cases, there might be situation, when slimbus channels
are being closed even when the number of active channels for
that DAI is 0. Avoid decrementing ch_act field if it is already 0.
Change-Id: Ie988eb049a20fc7cf5725ddc183ee1ae5fcb758d
Signed-off-by: Banajit Goswami <bgoswa@codeaurora.org>
-While music playback is done and PCM close sequence
is in progress, if headset is inserted this will report
headset insert to snd_soc_jack_report. But it will get
stuck here waiting for lock that is already acquired by
DAPM at PCM stream close snd_soc_dapm_stream_event.
-Change the function call used to report insert/removal
from snd_soc_jack_report to snd_soc_jack_report_no_dapm
that fixes this deadlock.
CRs-Fixed: 429766
Change-Id: I167e45bdce9790ff7e878528629c4d652029bf26
Signed-off-by: Laxminath Kasam <lkasam@codeaurora.org>
As the ASM session is opened in hw_param the soft volume does
not take effect. Hence move it to hw_param so its applied after
asm open.
CRs-Fixed: 434323
Change-Id: I1ce24d74382bd2c1926df2d63ff2eb40b0652f95
Signed-off-by: Asish Bhattacharya <asishb@codeaurora.org>
To support the deep buffer output, the playback buffer size is
increased to 4k from 2k. Reverting the change as the support for
deep buffer output is no longer valid.
Change-Id: I5034245bde4b55a5816506fbdfeac970a8e7804b
CRs-Fixed: 438177
Signed-off-by: Subhash Chandra Bose Naripeddy <snariped@codeaurora.org>
Modify min and max period bytes and number of periods in
compressed driver so as to be consistent with lpa and to
meet ION memory requirements. This avoids using huge
buffer sizes for tunnel decode.
Change-Id: Ib71da4cebfc097e7aeaf0ca0e65ee847ba7cceb7
Signed-off-by: Phani Kumar Uppalapati <phanik@codeaurora.org>
Buffer flags are going to be specified at allocation time rather than
map time. This removes the flags argument from the ion kernel map
function.
Change-Id: Ib983ecd0dcd7befb36287ae7037c71d4ca475f90
Signed-off-by: Mitchel Humpherys <mitchelh@codeaurora.org>
Refactor common combinations of cpu_is macros into logical
SoC class macros to reduce the maintenance burden of
implementing support for additional product variants.
Conflicts:
arch/arm/mach-msm/include/mach/socinfo.h
Change-Id: I6916dcb30378cd3ba33ea8d863403f93f62a13f2
Signed-off-by: Stepan Moskovchenko <stepanm@codeaurora.org>
Signed-off-by: Neha Pandey <nehap@codeaurora.org>
Heap mask field passes as argument at allocation time to specify ion
heap.
Change-Id: I6951da81b11bfcb0e006d2539c1707aabbb12db6
Signed-off-by: Hanumant Singh <hanumant@codeaurora.org>
Signed-off-by: Mitchel Humpherys <mitchelh@codeaurora.org>
Update LPA platform driver to register restart callback.
The implementation of the callback is to issue a pending buffer
written by the application to restart the session when the dequeue
process is stopped due to underrun issue.
The callback is triggered from ALSA core when the framework
detects that the renderer stopped pulling data but the application
is still writing data. It runs in atomic context.
Fixes:
-A disturbance was heard when switching from one song
to the other, or when the song is paused for greater
than 3 seconds and then resumed
-This was due to the previous value of the track
volume being used in the lpa driver
-This was fixed by setting 0 as the initial volume
Change-Id: I035760937d032038fb3f3383dc87ecbdc5f4700d
Signed-off-by: Subhash Chandra Bose Naripeddy <snariped@codeaurora.org>
Populate bus vectors for the audio ocmem driver with use of new
bus device tree APIs.
Change-Id: If965c269706763a3a67b22243f265429f23427d6
Signed-off-by: Phani Kumar Uppalapati <phanik@codeaurora.org>
Signed-off-by: Rajeev Kulkarni <krajeev@codeaurora.org>
Fix issues when alloc grow, shrink requests received while display
off/on. Add return statement to the audio/voice workqueue for the safe
exit.
Change-Id: Ifda7823f68ef0f26b9fe781c4c509c7633291169
CRs-fixed: 404445
Signed-off-by: Phani Kumar Uppalapati <phanik@codeaurora.org>
Signed-off-by: Rajeev Kulkarni <krajeev@codeaurora.org>
Updating the sample rate for the compressed stream playback of the HDMI
interface through the mixer control. For APQ8064, MSM8930 and MSM8960
Change-Id: I61d598e2eff546bfc053c5301f992fb7d9ba3268
CRs-fixed: 405772
Signed-off-by: Vicky Sehrawat <vickys@codeaurora.org>
Update compressed platform driver to register an restart callback.
Implementation of the restart callback is to issue an pending buffer
Written by application to restart the session when the de queue
Process is stopped due to underrun issue.
Restart callback is triggered from the ALSA core framework when
Framework detects that the render stopped pulling the data
And application is writtig the data.
This callback will be running in atomic context.
Change-Id: Iada382f39bd052beeb0b1d165959d510fa51ff7c
Signed-off-by: Santosh Mardi <gsantosh@codeaurora.org>
Update SoC framework to update the restart callback registerd
By platform driver, which inturn be used by ALSA core framework
To trigger the restart callback during the under run issue to
To recover from hang state.
Change-Id: I70f47c0d295526e7f35d4a4f9024e0e87dbc2455
Signed-off-by: Santosh Mardi <gsantosh@codeaurora.org>
Enhance the multi channel pcm platform driver to support SYNC and
ASYNC mode
Change-Id: Ieca18864ce68055f7806be487b487d305bee2af8
Signed-off-by: Santosh Mardi <gsantosh@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: Ief351168b08d5cca0a76405834cf3d6aa14a3941
Signed-off-by: Phani Kumar Uppalapati <phanik@codeaurora.org>
Currently, at the end of compress audio playback, session is
terminated without ensuring compressed audio stream is completely
rendered. Use ioctl drain to drain out all the data before EOS is
signalled and session is closed. This is to make sure all the
data is played out by DSP.
CRs-Fixed: 390897
Change-Id: I3e39b210a8924180de9eb4453c6f6f6435ded0ee
Signed-off-by: Krishnankutty Kolathappilly <kkolat@codeaurora.org>
There is use case that the HDMI input goes through MI2S TX
interface to ADSP. Add 8-channel L-PCM TX support for this
use case.
Change-Id: Iaf10ec357a25fe4cf80e36a2ceba3d113e9b0416
Signed-off-by: Subhash Chandra Bose Naripeddy <snariped@codeaurora.org>
If MBHC GPIO IRQ for mechanical headset detection asserted while power
management's suspend call chain is being called, wcd9xxx_lock_sleep will
fail to wait for system's resume since power management interface has
to finish suspend in order to resume system.
In this case, interrupt cannot be handled as codec driver cannot
guarantee if underlying bus is awake.
Wake up system and resend the IRQ for this situation.
(cherry picked from commit ecf379cd17b7fe94e6f0df0598a472b6f1828ad5)
(cherry picked from commit bdda99b93915d217150f5484f9c718c905703f1e)
CRs-fixed: 404167
Signed-off-by: Joonwoo Park <joonwoop@codeaurora.org>
Change-Id: Id5970b359b80e0cd1f701b30c61ade23e0f67b9e
Signed-off-by: Neha Pandey <nehap@codeaurora.org>
When recording is started, it is observed that the pop level is
much higher than expected. Fix this issue by enabling the High
Pass Filter on microphone path with a cut off frequency of 150Hz
when recording starts and reprogram the cut off frequency back
after a timeout of 300 millisecond.
CRs-fixed: 380966
(cherry picked from commit f9ed163835d1634a86cabc233d85bfe857f0eacf)
Signed-off-by: Bhalchandra Gajare <gajare@codeaurora.org>
Change-Id: If98dfaf0f9b8f14c306f2d6a59154a615752823b
Signed-off-by: Neha Pandey <nehap@codeaurora.org>
When front-end PCM session is in paused state, back-end PCM
session will be put in paused state as well if given front-end
PCM session is the only client of the given back-end. Then,
even though the application closes front-end PCM session, DPCM
framework will not allow back-end to enter into HW_FREE state,
and hence back-end will never get shutdown completely.
(cherry picked from commit 6479fa2d82c5fa8b756a4755bc84e2ce48dad850)
CRs-fixed: 411089
CRs-fixed: 409894
Signed-off-by: Banajit Goswami <bgoswa@codeaurora.org>
Change-Id: I4d4f856a0401ff96cef98d374c1d12b10e124528
Signed-off-by: Neha Pandey <nehap@codeaurora.org>
- AFE does not support sampling rate 44.1k
- This fix addresses the issue by setting backend proxy device
sampling rate to 48k
(cherry picked from commit 69ab7356a72981032d5d403ec1508dcfaeb7075f)
CRs-fixed: 413871
Signed-off-by: Jayasena Sangaraboina <jsanga@codeaurora.org>
Change-Id: Ic067178f9d4cdb51b8fee5292b960922088c8539
Signed-off-by: Neha Pandey <nehap@codeaurora.org>
During playback of audio on earpiece path, the PA on the path
is the last component to be turned on. The turn on of PA can
cause a glitch in the output signal because the level of the PA
is not yet settled. Waiting for 20 msec after turning on the PA
avoids this glitch.
CRs-fixed: 417031
Signed-off-by: Bhalchandra Gajare <gajare@codeaurora.org>
(cherry picked from commit 1fa994d4aa256a536523f116232c4822b3495400)
Change-Id: Iaa7a47526c71d8b037df3d09cf30ede296c21356
Signed-off-by: Neha Pandey <nehap@codeaurora.org>
PM8917 does not have speaker amplifier, hence the codec lineout
is directed to the speaker through a onboard amplifier. The
amplifier needs to be enabled in order for speaker playback
to work on a target with PM8917
Change-Id: Ic9f16d335bc1a0cef6fb1eb93c1617fd086cc5b7
Signed-off-by: Bhalchandra Gajare <gajare@codeaurora.org>
(cherry picked from commit f6d1f7261de810bb07429c25b99187328359b8c4)
