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>
