Commit graph

565033 commits

Author SHA1 Message Date
Alexy Joseph
3e18aa8eca ASoC: msm: qdsp6v2: Handle additional codec specific metadata
Codec specific  metadata is sent only for first stream in gapless
playback. This causes incorrect configuration to be set for second
stream and distortions are observed due to framedrops in adsp.
Add support to send codec specific format during start of
next stream in gapless.
Add bit rate to wma codec data structure as it can vary between
streams in gapless.

Change-Id: I39f34ea1addff720612fe3e06257e7d75889e574
Signed-off-by: Chaithanya Krishna Bacharaju <chaithan@codeaurora.org>
Signed-off-by: Alexy Joseph <alexyj@codeaurora.org>

Conflicts:
	sound/soc/msm/qdsp6v2/msm-compress-q6-v2.c
2016-03-23 20:11:07 -07:00
Kenneth Westfield
579b5a9417 ASoC: pcm: Add delay_blk feature
Add delay_blk() to pcm, platform and DAI ops.
The pcm delay_blk() op collects the audio DSP path
delays from the low-level drivers and sets the
runtime->delay field to their aggregate.

Change-Id: Ib7e10f44ab8ccb46dc2f5825081d0afef662d827
Signed-off-by: Kenneth Westfield <kwestfie@codeaurora.org>
2016-03-23 20:11:06 -07:00
Sudheer Papothi
27c5547625 ALSA: audio_codec: add hwdep interface
This commit adds hwdep interface for codec calibration.

currently codec driver uses request_firmware to get calibration data
but as this firmware file is also written by userspace process after
bootup, it is not recommended to use request_firmware. ALSA core
provides mechanism to get hardware dependent data using hwdep nodes.
Codec will use aforementioned nodes to get calibration data from
userspace.

Signed-off-by: Sudheer Papothi <spapothi@codeaurora.org>
2016-03-23 20:11:05 -07:00
Sudheer Papothi
b21e23e099 ASoC: pcm: Check for capture path in all codec_dais
During soc_pcm_prepare, check for capture path for all
the codec_dais instead of the last codec_dai. This is
will help to set pcm start event for all the codec dais
that has capture_active set to high. Change checks for
all the codec_dais capture_active bit to set start stream
event.

Signed-off-by: Sudheer Papothi <spapothi@codeaurora.org>
2016-03-23 20:11:04 -07:00
Phani Kumar Uppalapati
3dea4c44db ALSA: core: Add an API to create and register module
Add an API to create a module under the given parent
directory and then register the info entry. This API
can be used to expose ID and version related
information.

Change-Id: I06f8bc1d8f649e6db37eb65303e948bc58eab7da
Signed-off-by: Phani Kumar Uppalapati <phaniu@codeaurora.org>
2016-03-23 20:11:03 -07:00
Vidyakumar Athota
2e4fca555e ALSA: pcm: add support for 384KHz sample rate
Currenlty HW params fails to set 384KHz sample rate
due to unsupported sample rate. Change to add 384KHz
sample rate support to ALSA.

Change-Id: I381f7cdcc69e6cf9339cec53aab3fa295760c17c
Signed-off-by: Vidyakumar Athota <vathota@codeaurora.org>
2016-03-23 20:11:02 -07:00
Sudheer Papothi
fef4c4cb85 regmap: Add soundwire bus support
Add soundwire bus support to regmap. This change enables
codec drivers using soundwire hardware interface to use
regmap interface for register read/write functionality.

Signed-off-by: Sudheer Papothi <spapothi@codeaurora.org>
2016-03-23 20:11:01 -07:00
Sudheer Papothi
b2f4078131 ASoC: msm: qdsp6v2: Handles additional flac metadata
Currently, metadata such as min/max block size is sent only for first
stream in FLAC gapless playback. This causes incorrect configuration
and, subsequently, framedrops in decoding of second stream and onwards
By sending these additional flac metadata, dsp receives stream-wise
metadata and decodes without dropping

Signed-off-by: Amit Shekhar <ashekhar@codeaurora.org>
Signed-off-by: Banajit Goswami <bgoswami@codeaurora.org>
Signed-off-by: Sudheer Papothi <spapothi@codeaurora.org>
2016-03-23 20:11:00 -07:00
Banajit Goswami
a45d64c35f ASoC: msm: qdsp6v2: Add support for HDMI passthrough
Add support for compressed bitstream passthrough over HDMI
for DD/DDP contents. Use compressed driver to support
passthrough.

Change-Id: I01f9e4fa984a1f45d1f4de5250bed8f95d2a2dd0
Signed-off-by: Krishnankutty Kolathappilly <kkolat@codeaurora.org>
Signed-off-by: Banajit Goswami <bgoswami@codeaurora.org>
2016-03-23 20:11:00 -07:00
Sudheer Papothi
852b36a9c0 soundwire: add support for device table match
With this patch soundwire drivers can use id_table and
MODULE_DEVICE_TABLE() method to bind against the devices just
like I2C or SPI drivers.

Change-Id: I4e8eee3cb9626a5dc4fbfa238b5d2a578355f2a3
Signed-off-by: Sudheer Papothi <spapothi@codeaurora.org>
2016-03-23 20:10:59 -07:00
Deepa Madiregama
8152f43f0a ALSA: usb-audio: Fix the mixer control range limiting issue
- mixer_ctl_set() function is limiting the volume level
  to particular range. This results in incorrect initial
  volume setting for that device.
- In USB mixer while calculating the dBmin/dBmax values
  resolution factor is hardcoded to 256 which results in
  populating the wrong values for dBmin/dBmax.
- Fix is to use appropriate resolution factor while
  calculating the dBmin/dBmax values.

CRs-Fixed: 515012
Change-Id: I502355af66f850bb65380c27333c3341fa43a947
Signed-off-by: Deepa Madiregama <dmadireg@codeaurora.org>
2016-03-23 20:10:58 -07:00
Mayank Rana
f18c9be9d4 msm: usbaudio: Add check for NULL before dereferencing
kzalloc() and usb_ifnum_to_if() both APIs can return NULL. Current
code is not checking return value and derefencing which may crash
device if it is set to NULL. Fix this by checking return value
against NULL and handling the same.

CRs-Fixed: 562273
Change-Id: I0d2c910f43321e94fc447b19ae3e3207727e24f3
Signed-off-by: Mayank Rana <mrana@codeaurora.org>
Signed-off-by: Banajit Goswami <bgoswami@codeaurora.org>
2016-03-23 20:10:57 -07:00
Banajit Goswami
c336ae90fa ASoC: core: Remove sound card state check from read/write
Having sound card offline condition checks inside
register read/write functions will also avoid Asoc
cache writes. However, it is needed to update the
Asoc cache whenever user-space sends disable sequence
while sound card is in offline state. Remove the offline
check condition from ASoC read/writes and let the codec
driver take care of not sending bus read/writes if sound
card is in offline state.

Change-Id: I960b85a277a28dd05eb919c1c6abf7aa32a45265
Signed-off-by: Phani Kumar Uppalapati <phaniu@codeaurora.org>
Signed-off-by: Banajit Goswami <bgoswami@codeaurora.org>
2016-03-23 20:10:56 -07:00
Sudheer Papothi
a73f468d5e ASoC: pcm: update the start-up sequence for playback
Codec should be started before the CPU to ensure that there is no data
loss during playback.Current sequence enables the CPU first followed by
codec.This change updates the sequence prevent any playback data loss.

Signed-off-by: Gopikrishnaiah Anandan <agopik@codeaurora.org>
Signed-off-by: Banajit Goswami <bgoswami@codeaurora.org>
Signed-off-by: Sudheer Papothi <spapothi@codeaurora.org>
2016-03-23 20:10:55 -07:00
Anish Kumar
f1031ca0f7 ASoC: pcm: increase the hostless buffer size
PAGE_SIZE buffer is not enough in all use cases.This
limits the amount of buffer that can be passed from
userspace. Not increasing this causes failure in pcm_native
when rules are checked against all hw_params passed
from userspace.

Change-Id: Iddc280f78d54a1959be8718055bdb5ef0270653c
Signed-off-by: Anish Kumar <kanish@codeaurora.org>
2016-03-23 20:10:54 -07:00
Anish Kumar
80e8ef2c04 ASoC: pcm: Add support for fixup callback
Fixup callback is added for dai's which
do not follow the FE and BE convention
and is directly controlled by userspace
such as hostless dai's. This will restrict
the hw_params based on what is supported by
hardware rather than blindly setting what
is given by userspace.

Change-Id: I401c70ab5de1df10363ec808cb68f72d8d74af96
Signed-off-by: Anish Kumar <kanish@codeaurora.org>
2016-03-23 20:10:54 -07:00
Sudheer Papothi
d57ab70296 ASoC: dapm: mark dapm_kcontrol_get_wlist as global function
dapm_kcontrol_get_wlist() function needs to be called from
driver other than the soc-dapm.c file itself. Mark this
function as global function, so it can be accessed from
other drivers.

Signed-off-by: Banajit Goswami <bgoswami@codeaurora.org>
Signed-off-by: Sudheer Papothi <spapothi@codeaurora.org>
2016-03-23 20:10:53 -07:00
Abhimanyu Kapur
6aa90dc7dc sound: pcm: fix compilation warnings in snd_pcm_add_volume_ctls
Use explicit char* casting to fix compiler warning.

kernel/sound/core/pcm_lib.c: In function 'snd_pcm_add_volume_ctls':
kernel/sound/core/pcm_lib.c:2674:3: warning: passing argument 1 of 'snprintf' discards 'const' qualifier from pointer target type
   snprintf(knew.name, size, "%s %d %s",
   ^
Signed-off-by: Abhimanyu Kapur <abhimany@codeaurora.org>
2016-03-23 20:10:52 -07:00
Aviral Gupta
6f88ba1a69 ASoC: compress: propagate the error code from the compress framework
Propagate the error code from the compress framework for the timestamp
query. This error code will be used by the client to handle the
error case scenarios gracefully.

CRs-Fixed: 683288
Change-Id: I68ad14d52327dd0156531fe8d17ac54ba110fdf6
Signed-off-by: Aviral Gupta <aviralg@codeaurora.org>
2016-03-23 20:10:51 -07:00
Damir Didjusto
583170e962 ASoC: msm: qdsp6v2: Cleanup of compress offload drivers
Clean up the format specifiers for compress offload drivers
and add compat ioctl support

Change-Id: I0829830336ec62c66658d6a5199d73b61249f791
Signed-off-by: Damir Didjusto <damird@codeaurora.org>
Signed-off-by: Gopikrishnaiah Anandan <agopik@codeaurora.org>
Signed-off-by: Banajit Goswami <bgoswami@codeaurora.org>
2016-03-23 20:10:50 -07:00
Banajit Goswami
c58f710d7d ALSA: compress: use mutex in drain
Since we dont have lock over the function, we need to acquire mutex
when checking and modifying states in drain and partial_drain handlers

Change-Id: I3f8af90a226b772492fb6e09c625ebedc8ebfeb5
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Eric Laurent <elaurent@google.com>
Git-commit: bab68b886fb87ee82a3e7d86d54f7eaa025950ef
Git-repo: https://android.googlesource.com/kernel/msm
Signed-off-by: Dhananjay Kumar <dhakumar@codeaurora.org>
Signed-off-by: Banajit Goswami <bgoswami@codeaurora.org>
2016-03-23 20:10:49 -07:00
Banajit Goswami
5200617101 ASoC: compress: revert some code to avoid race condition
Revert some changes for compress offload path to avoid race
condition in drain and paartial-drain cases.

Signed-off-by: Banajit Goswami <bgoswami@codeaurora.org>
2016-03-23 20:10:48 -07:00
Eric Laurent
575eeabce9 ALSA: Compress - dont use lock for all ioctls
Some simple ioctls like timsetamp query, capabities query can be done anytime
and should not be under the stream lock. Move these to
snd_compress_simple_iotcls() which is invoked without lock held.

While at it, improve readblity a bit by sprinkling some empty lines.

Change-Id: Icc8ffdadd565c635f6a95e7e5bdda76257f24ea3
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Eric Laurent <elaurent@google.com>
Git-commit: 6a44374b8b92e9946dc1e5c15c2a11003aa859b1
Git-repo: https://android.googlesource.com/kernel/msm
[dhakumar@codeaurora.org: resolved merge conflicts]
Signed-off-by: Dhananjay Kumar <dhakumar@codeaurora.org>
2016-03-23 20:10:47 -07:00
Liam Girdwood
8236e4d497 ALSA: compress: Add snd_compress_free()
We are not currently freeing the compressed devices at module
unload time. This could be used to help, but more investigation
needs to be done to resolve this issue.

This patch is WIP

Change-Id: I6cf8bcc56e4d446d83e9a0e63b79db3da378a901
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Git-commit: b2ebfd20ecb255b8e575ea8b6fad8d1affc2ea77
Git-repo: https://android.googlesource.com/kernel/msm
Signed-off-by: Dhananjay Kumar <dhakumar@codeaurora.org>
2016-03-23 20:10:47 -07:00
Krishnankutty Kolathappilly
280e00c01d ALSA: compress: Memset timestamp structure to zero.
snd_compr_tstamp is initialized using aggregate initialization
that does not zero out the padded bytes. Initialize timestamp
structure to zero using memset to avoid this.

CRs-Fixed: 568717
Change-Id: I7a7d188705161f06201f1a1f2945bb6acd633d5d
Signed-off-by: Krishnankutty Kolathappilly <kkolat@codeaurora.org>
2016-03-23 20:10:46 -07:00
Eric Laurent
fea50d9677 ASoC: msm: handle write done events in pause state
If a write done event is received when the compress driver
is in paused state, treat it as an underrun.

Change-Id: I56ca867983b9139c04d135276da7344ac912065e
Signed-off-by: Haynes Mathew George <hgeorge@codeaurora.org>
Signed-off-by: Eric Laurent <elaurent@google.com>
Git-commit: 6b9c3272087127fda41ed65ee7f8536ad9748383
Git-repo: https://android.googlesource.com/kernel/msm
Signed-off-by: Dhananjay Kumar <dhakumar@codeaurora.org>
2016-03-23 20:10:45 -07:00
Eric Laurent
b6125c4081 ASoC: msm: handle partial writes in compress driver
Compress driver might have to deal with arbitrary sized writes
in the middle of playback. To deal with it, accumulate fragment_size worth
data before sending to the DSP. Use the drain trigger to push any
pending buffer to the DSP.

Change-Id: I8d95a9945076142f16adb99a295f6a9e84039cbb
Signed-off-by: Haynes Mathew George <hgeorge@codeaurora.org>
Signed-off-by: Eric Laurent <elaurent@google.com>
Git-commit: 6ac9bb553c1285266b2ac28800abec240cd76e17
Git-repo: https://android.googlesource.com/kernel/msm
[dhakumar@codeaurora.org: resolved merge conflicts]
Signed-off-by: Dhananjay Kumar <dhakumar@codeaurora.org>
2016-03-23 20:10:44 -07:00
Eric Laurent
106989c13d ASoC: msm: compr_free(), check if EOS is going on before waiting
Change-Id: I760979aa6ab1e1e10f203d8da5c9720d145869d3
Signed-off-by: Haynes Mathew George <hgeorge@codeaurora.org>
Signed-off-by: Eric Laurent <elaurent@google.com>
Git-commit: dddb29670e35ff0366f1f3dd7128b78872375f72
Git-repo: https://android.googlesource.com/kernel/msm
Signed-off-by: Dhananjay Kumar <dhakumar@codeaurora.org>
2016-03-23 20:10:43 -07:00
Eric Laurent
5871bd83cf ASoC: msm: treat partial drain like full drain
Change-Id: I3a9d7e7a62f0eefb941b68d0ee40657998549f76
Signed-off-by: Haynes Mathew George <hgeorge@codeaurora.org>
Signed-off-by: Eric Laurent <elaurent@google.com>
Git-commit: 1f1aac6eee5e080e9bd70bd6d56da3b4435273e8
Git-repo: https://android.googlesource.com/kernel/msm
Signed-off-by: Dhananjay Kumar <dhakumar@codeaurora.org>
2016-03-23 20:10:42 -07:00
Eric Laurent
21f7a9a5eb ASoc: msm: stub set_metadata ops function
Implement stub set_metadata ops function in compressed platform driver.
Just return no error and print debug log.

Change-Id: I45d69b5902b666abc12549aebf1bfc58824ab8ec
Signed-off-by: Haynes Mathew George <hgeorge@codeaurora.org>
Signed-off-by: Eric Laurent <elaurent@google.com>
Git-commit: 902e5b4af87f4649f3b093dea0c5f59c87be0dc1
Git-repo: https://android.googlesource.com/kernel/msm
Signed-off-by: Dhananjay Kumar <dhakumar@codeaurora.org>
2016-03-23 20:10:41 -07:00
Eric Laurent
db4af0b8e4 ASoc: msm: Send media format block for compress playback
Compress offload playback for AAC requires media format block to be
sent to the DSP before the bitstream can be decoded. Send this
as part of configuring the DSP

Change-Id: Ia22bc52aa0c136a1ee3f70a1e150458f8e4b6866
Signed-off-by: Haynes Mathew George <hgeorge@codeaurora.org>
Signed-off-by: Eric Laurent <elaurent@google.com>
Git-commit: ed16e26a99b2b7904d0badae4bd0d30953873b9e
Git-repo: https://android.googlesource.com/kernel/msm
Signed-off-by: Dhananjay Kumar <dhakumar@codeaurora.org>
2016-03-23 20:10:41 -07:00
Eric Laurent
0f65a07b71 ASoC: msm: Return proper error codes
Return proper error codes back to ASoc framework if
trigger operations like flush, drain fails or gets interrupted.

Change-Id: I3e2a259b55d5619eac6122083a8936a85700d657
Signed-off-by: Eric Laurent <elaurent@google.com>
Git-commit: 3a14d1af02c52d19d698732f3772f07843c90f6c
Git-repo: https://android.googlesource.com/kernel/msm
[dhakumar@codeaurora.org: resolved merge conflicts]
Signed-off-by: Dhananjay Kumar <dhakumar@codeaurora.org>
2016-03-23 20:10:40 -07:00
Haynes Mathew George
5cd46b0657 Asoc: msm: compr drain/flush fixes
Add missing synchronization between handling of flush and
drain in the compressed audio driver.

Change-Id: I01b6d2c4fb372be483022d1c3430b9369dea34f7
Signed-off-by: Haynes Mathew George <hgeorge@codeaurora.org>
Signed-off-by: Dhananjay Kumar <dhakumar@codeaurora.org>
2016-03-23 20:10:39 -07:00
Eric Laurent
75e02a8b5f ASoc: msm: Change condition to get session time
Change condition to skip request for session time from DSP
during compress offload.

Change-Id: I0192b1f4059f09a32b8848c40e91eecf3dadf966
Signed-off-by: Haynes Mathew George <hgeorge@codeaurora.org>
Signed-off-by: Eric Laurent <elaurent@google.com>
Git-commit: b31efc7e185f453da00cedbbad5b59c23560dcde
Git-repo: https://android.googlesource.com/kernel/msm
Signed-off-by: Dhananjay Kumar <dhakumar@codeaurora.org>
2016-03-23 20:10:38 -07:00
Matt Wagantall
418255d9d9 ASoC: msm: Add compress audio playback code base
Support bits and pieces to make msm-compress-q6-v2.c usable
will be added in subsequent commits.

Signed-off-by: Matt Wagantall <mattw@codeaurora.org>
2016-03-23 20:10:37 -07:00
Banajit Goswami
547025290f ASoC: msm: qdsp6v2: Fix bit alignment in snd_codec params
Pointer member variables in snd_codec params break bit alignment and
causes data corruption. By changing these pointers to fixed size
array variables, the bit alignment is rectified. The size has been
set to max possible size. Also, remove params which are not required
anymore.

Change-Id: Ib87bbeb07b0df1ce8a81166b319976fe54c0f013
Signed-off-by: Amit Shekhar <ashekhar@codeaurora.org>
Signed-off-by: Banajit Goswami <bgoswami@codeaurora.org>
2016-03-23 20:10:36 -07:00
Banajit Goswami
ab70671974 ASoC: dapm: Do not process cpu_dai widgets in bias changes
Kernel 3.10 has cpu dai widgets in the list of card
widgets in addition to platform and codec widgets.
This is causing a delay in device switch because
cpu dai widgets increase the number of widgets to be
processed for bias changes by a factor of five, however
they are not required to be processed.
Skip processing of cpu dai widgets to keep
the device switch latency at same level as in kernel 3.4.

CRs-fixed: 699168
Change-Id: I2d7d9e34dabf2cd25ec5fdd3e58be0fc657c0f6c
Signed-off-by: Damir Didjusto <damird@codeaurora.org>
Signed-off-by: Banajit Goswami <bgoswami@codeaurora.org>
2016-03-23 20:10:35 -07:00
Banajit Goswami
f48c0e7f53 revert: ALSA: compress: update struct snd_codec_desc for sample rate
This gerrit reverts the gerrit with the commit number-
      b8bab04829
    Now that we don't use SNDRV_PCM_RATE_xxx bit fields for sample rate, we need to
    change the description to an array for describing the sample rates supported by
    the sink/source

Signed-off-by: Banajit Goswami <bgoswami@codeaurora.org>
2016-03-23 20:10:34 -07:00
Banajit Goswami
dba6519f3f include: increase allowed SW INPUT device ID from 15 to 32
Increase the Input device SW ID from 15 to 32. This is needed
to accomodate more input devices.

Signed-off-by: Banajit Goswami <bgoswami@codeaurora.org>
2016-03-23 20:10:34 -07:00
Sudheer Papothi
18e3977b99 ASoC: msm: Add compressed TX support
There is use case that the HDMI input goes through MI2S
TX interface to ADSP. Add compressed TX support for
this use case.

Signed-off-by: Subhash Chandra Bose Naripeddy <snariped@codeaurora.org>
Signed-off-by: Banajit Goswami <bgoswami@codeaurora.org>
Signed-off-by: Sudheer Papothi <spapothi@codeaurora.org>
2016-03-23 20:10:33 -07:00
Banajit Goswami
0d879df1d3 ASoC: msm: qdsp6v2: Add FLAC in compress offload path
Add FLAC format in compress offload driver, and asm
module

Change-Id: I818ace8397e761b1acff7f9b2eab6e0103ed78c8
Signed-off-by: Apurupa Pattapu <apurupa@codeaurora.org>
Signed-off-by: Banajit Goswami <bgoswami@codeaurora.org>
2016-03-23 20:10:32 -07:00
Banajit Goswami
d3d2941f84 ASoC: msm: Add support for WMA DSP decode
WMA V9 and WMA Pro tunnel-mode supports are added to
compressed driver. It allows user-space application to
decode WMA V9 and WMA Pro audio stream through QDSP6.

Change-Id: I99407d00b618a627e6d762be9abea4abd2410b8b
Signed-off-by: Manish Dewangan <manish@codeaurora.org>
Signed-off-by: Dhananjay Kumar <dhakumar@codeaurora.org>
Signed-off-by: Banajit Goswami <bgoswami@codeaurora.org>
2016-03-23 20:10:31 -07:00
Banajit Goswami
64d6133b7f ALSA: compress: Update compress audio params
Add timestamp field to compress structure which indicates when
audio sample has been captured or needs to be rendered.

Change-Id: Ie61170c6645c71207e7df1c7176e0750e47590f8
Signed-off-by: Gopikrishnaiah Anandan <agopik@codeaurora.org>
Signed-off-by: Banajit Goswami <bgoswami@codeaurora.org>
2016-03-23 20:10:30 -07:00
Banajit Goswami
35b7cb1228 ASoC: msm: Add support for AC3 and EC3 playback in tunnel mode
Dolby surround1-DS1 module supports both Dolby Audio processing - post
processing and Dolby digital plus decoder in DSP. Add support for
AC3 and EC3 playback in tunnel mode so that DS1 is integrated and
functional end to end.

Change-Id: Iacb46cdfded16c9a5a9227a6ff4e072c61df2be8
Signed-off-by: Subhash Chandra Bose Naripeddy <snariped@codeaurora.org>
Signed-off-by: Banajit Goswami <bgoswami@codeaurora.org>
2016-03-23 20:10:29 -07:00
Banajit Goswami
a3447d8ba1 ASoC: msm: Implementation of transcode from MP3, WMA to DTS compressed
Set the output path of the decoder output of the compressed format
to input of the pseudo port.
Output of the pseudo port is connected to the input of
DTS encoder.
Output of the DTS encoder is connected to the selected
output ports SPDIF, HDMI.

Change-Id: I3945e53fdfd57de47fb2209ddc81ba4623999028
Signed-off-by: Aviral Gupta <aviralg@codeaurora.org>
Signed-off-by: Banajit Goswami <bgoswami@codeaurora.org>
2016-03-23 20:10:28 -07:00
Banajit Goswami
826ba45578 ASoC: msm: DTS_LBR Passthrough Support
- Added DTS_LBR for Passthrough case in Compress Driver
- Added Graceful Error Handling for Unsupported Codec
  IDs in case of open_write_compressed.

Change-Id: Ifbecb02832a2599be0e3c73cc69381f87969d78a
Signed-off-by: Dhananjay Kumar <dhakumar@codeaurora.org>
Signed-off-by: Banajit Goswami <bgoswami@codeaurora.org>
2016-03-23 20:10:28 -07:00
Banajit Goswami
d81e3e2ffd ASoC: msm: Add the support for the MP2 decoder
Support the MP2 as a format for the tunnel mode.
MP2 format to be decoded in the DSP.

Change-Id: I0d268a6ddb57b1470ee2c43449ac31520176232f
Signed-off-by: Aviral Gupta <aviralg@codeaurora.org>
Signed-off-by: Banajit Goswami <bgoswami@codeaurora.org>
2016-03-23 20:10:27 -07:00
Banajit Goswami
93e5937bda ASoC: msm : DTS security implementation
Receive DTS security modelId from userspace,
and supply to Q6 core service.

Change-Id: Ib50f3a81da60c92ceb5b521134cd3d1b6fb8e5cb
Signed-off-by: Srikanth Uyyala <suyyala@codeaurora.org>
Signed-off-by: Banajit Goswami <bgoswami@codeaurora.org>
2016-03-23 20:10:26 -07:00
Banajit Goswami
b07e54a825 ASoC: msm: Enable PCM capture in compressed driver
In the use case of HDMI input PCM capture, timestamp received from
DSP is required in userspace to propogate it on the playback path.
Timestamp mode propagation is availble through the meta data mode
in compressed driver. Add support for PCM capture in compressed
driver to address the usecase

Change-Id: I1221b8e99628dadc136df681619ed960ff7c5c1a
Signed-off-by: Subhash Chandra Bose Naripeddy <snariped@codeaurora.org>
Signed-off-by: Banajit Goswami <bgoswami@codeaurora.org>
2016-03-23 20:10:25 -07:00
Banajit Goswami
ae3cbed9e0 SoC: msm: Add support for meta data in compressed TX
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>
Signed-off-by: Banajit Goswami <bgoswami@codeaurora.org>
2016-03-23 20:10:24 -07:00