- add changes to support single BE mode on top of the
Double BE mode which is supported.
- support reading of version or mode from XML.
- add logic to switch between single and double BE
based on the mode in XML.
- define a new device for true native.
- add logic to read tasha codec version and allow
true native only for 2.0 version of codec
- enchance log msgs for better debugging.
Change-Id: I970487f16af32144ec26f967e8bfcb32d44d67a4
Add support to enable VBat feature. This includes required changes
to select VBat speaker devices for audio and voice usecases and
sending the VBat calibration data to codec driver.
Change-Id: I2234bee3cc4d6f16934a4b27ef9da21cde74bb48
tz_names support is added using key value pairs under
config params in audio_platform_info.xml.
This reverts commit 8225243c24
Change-Id: I5bee7225f945b31e3df9372d00aeb003c2092d8b
Add support to parse platform specific configuration as
key value pairs from audio_platform_info.xml file.
Change-Id: Id1199f6f5cb3a060476f713a69b5de05f48815ce
- We need to set codec EC reference instead of MSM/AFE
EC ref for Native audio usecase along with other
usecases running on 48K port. In this scenario we
will run two AFE ports, one regular port and the
other port running at 44.1KHz and mixing done in codec.
So set appropriate codec EC reference.
- One example of such usecase is VoIP over audio path
and a 44.1KHz stream playback. But this change is generic
enough to accommodate other possible usecases if any.
Change-Id: Ib980e36d111c1671303abe2b19341568d9fd2331
Add API to notify platform when sound card
status is changed. Use function to reinitialize
ACDB-loader or resend common custom topology.
Change-Id: I59198f0ad8412ddc0d379b0c39a7807d6e22f4f7
- expose new api from audio HAL that volume listner module will call
to set volume calibration level
Change-Id: If74bf66d32def85022d79ccb9f84c3b85c8a2dc9
- If Modem SSR happens during boot up, Audio HAL
fails to initialize ACDB. It does not check the
error from acdb_init and does not re-initialize
when Sound Card is ONLINE after SSR
- Fix is to re-initialize ACDB after Sound Card is
ONLINE
Change-Id: Icb7a7ca7bb682d55a3cc8f7b26dc0bf511c30a69
- all changes for native audio backward
compatibility and related
- add set parameters from UI
- enable dynamic device switching
- featurize the code
- various backend concurrency changes
Change-Id: Id0f824c4b4c033f42008a4e8868652c8f6fe5c42
- Changes for device/Usecase selection
- Add new and combo devices for headphones
- Combo device handling
- Add new backend and logic to handle multiple
backends
- Remove hardcoded mixers and select them
dynamically based on the device
Change-Id: I7207291f2c27da13ba0cc77c66de1144f4b9888a
SLIMBUS_0 is the backend on 8952 external codec PLs.
Whereas TERT_MI2S is the backend on 8952 internal codec PLs.
Since both these PLs are supported on same code base,
provision is required to select backend based on codec type.
Change-Id: I9c0dd7d952ab9d6b9bf738074d97597622e09590
Fluence module on ADSP is upgraded to support source
tracking, sound focus and audio zoom features. This
change adds support for the same in audio HAL and is
needed to support the features end-to-end.
Source Tracking is the capability to identify the
source of speech. Sound Focus is the capability to
configure which sectors in the 360 degrees plane to
listen to. Audio Zoom is the capability to combine
the sound focus with other technologies so that a user
can listen in on a particular source in a scene.
Change-Id: I019b22d6541e6d1a10552d808c3320a998b86e48
* quic/LA.BR.1: (37 commits)
hal: Performance mode during recording usecase
audio: Support for compress offload recovery
audio: Fix for clip skip issue during SSR
hal: add checks for calibration buffer allocation failures
mm-audio: aenc-aac: fix integer overflow for encoded buffer timestamp calculation
hal: Add support for 8909 QRD skue
hal: Fix for FM mute issue on start of FM record
hal: Fix array length computation for backend id array.
hal : set Non SA+ app type for PCM Capture
hal : Fix for SA+ app type selection for PCM Capture usecase
hal : Fix for SA+ app type selection for PCM Capture usecase
hal: use 0 as default return value for out_get_render_position
audio: hal: Add support for pm8916 on msm8909
hal: Define a new combo device for AANC and Fluence
hal: Define a new combo device for AANC and Fluence
hal: use 0 as default return value for out_get_render_position
hal: Update DS2 implementation to support ACDB based license mechanism
hal : Overwrite the APP type for PCM RX and TX path
audio: HAL to support for peripheral manager
hal : Set the input device for VOIP calls using audio path
...
Conflicts:
hal/Android.mk
hal/audio_extn/audio_extn.c
hal/audio_extn/audio_extn.h
hal/audio_extn/dolby.c
hal/audio_extn/utils.c
hal/audio_hw.c
hal/msm8916/hw_info.c
hal/msm8916/platform.c
hal/msm8974/platform.c
hal/platform_api.h
Change-Id: Ibfa171e8f3af713dbb2cffbaf2ca2b0df3e8ae73
-Add support for HDMI passthrough for DD/DDP contents to HAL and
APM.
-Add mixer control to set volume on volume module on passthrough
COPP. The mixer control takes zero volume for mute and unity
volume for max volume.
-DDP content requires sample rate to be four times native sample
rate. HAL sample rate is changed to four times native rate from
start output stream. During rapid pause/resume offload thread
calls standby when there is no active track and no activity for
over a second. On resume start output steam is called and
sample rate is converted to four times its current value.
This results in session to be started with invalid sampling
rate and playback failures.
-Add HDMI passthrough support for JOC format.
Expose DDP/JOC as pass-through supported format if sink
supports either DD or DDP. This is to allow support for
pass-through convert based on edid data.
-Deep buffer music stream does not switch back to HDMI after
ringtone playback ends. Ringtone, alarm, notification etc are
played on speaker if HDMI pass-through is enabled. The
decision is taken based on stream or strategy. The decision
to change the device from HDMI to speaker is taken based on
the stream type. When a patricular stream ends use the
reference count instead of stream type.
-DAP is not turned on when switching from passthrough stream to
deep buffer stream.DAP bypass call from HAL to DAP HAL expects
integer pointer. Address of a bool variable is passed into the
function. Corrupt value is set on driver instead of turn on/off
causing random failures in DAP on/DAP bypass. Use same data type
and use enumeration to make the function call readable.
CRs-Fixed: 761339
Change-Id: Ided9439ec5e87233a1fec3ff9f50a1e9ba4cb788
Add support for parsing hdmi edid data and setting channel map
to asm
Parse edid data on hdmi connection and invalidat on disconnection.
Fix audio playback errors on HDMI when source channel count is
greater than sink channel count for offload playback.
Change-Id: I1e0a8335611a87974cb273431659938a0b345e68
-Add device to back end mapping for APQ8084 and MSM8994.
-Set initialization info like sound card number, dmid
to ds2 dap hal.
-Add mixer control DS2 OnOff to enable or disable DS2
-A property "audio.dolby.ds2.enabled" is added to trigger
the mixer control.
-Reuse ds1 functions for checking dolby format and getting
sound codec id.
CRs-Fixed: 661095 661018
Conflicts:
hal/audio_hw.c
hal/msm8916/platform.c
Change-Id: Iff5fdc49249eb035c2ccde18e190ccc6c3ff632a
Check for 24 bit in HAL is not needed as the check for 24 bit
platform support is achieved through flags in frameworks.
Change-Id: Icc590dfc1c4b831399435c19dd505ddebf6503cd
Echo reference is not being reset currently at the end of
recording/VoIP session causing the echo reference not to
work in the further instances of recording/VoIP sessions.
Fix this by resetting echo reference while closing the input
stream used for recording/VoIP usecases.
CRs-Fixed: 717973
Change-Id: I761ba112952951777525b1bf102c9d6ccaad3269
Sound trigger lookahead buffering data is read through audio hal.
Also the audio hal has to inform sound trigger hal about concurrencies
and subsytsem restart events. Audio Hal uses proprietary interface
defined by sound trigger for communication with sound trigger hal.
Change-Id: Id344077b40b481a953769518f8a6ac0adc13b709
Extend platform parser with support for additional sections.
Supported sections now include acdb ids, pcm device ids and backend names.
Change-Id: Idfbc8a8bb490606686436c107db5b0c7d636ccbe
- Add flac in supported codecs list
- Set FLAC codec specific parameters
- Set fragment size based on bit width and sampling rate
for flac
- Configure backend to 24/16 bit based on the current
usecases running
Change-Id: If013078e277fd3053fba6489345803f8f58bd86d
-This change adds DSDA support for APQ based targets
where external modem is used for voice calls.
-Add new CSD API to support local call hold.
Change-Id: I7743a1df43dc1abac4e325ff104ec1bb64c9e12b
Add support for concurrency handling between number of
listen capture session and playback activity.
Change-Id: I4a0656f240c62f2cc6266a714a96fdcdc13ae9d8
The stream's output rendering policy requires calibration to be
sent when stream routing configuration is set instead of the
device enable as each stream can have it's own routing with
different calibration needs.
Change-Id: If8bd365c4a6c616ed86b9383ba540ee8e28c4079
For the usecases where individual streams have to be rendered
through different post processing chain on DSP an output policy
is defined. The output configuration is read from the
output policy file based on the stream atrributes to render the
playback appropriately.
Change-Id: I095592013ffa5b04bea69254026ebb78e2934822
- To turn on/off fluence we need to change the build prop
file and changes will take effect after reboot. There is no
option to turn it on/off without rebooting the target.
- Add set_params support to turn it on and off.
An option is given to user to select the dualmic fluence
based on which set_parms is invoked.
Change-Id: I51d580af820d8e0b1bd1384c941ffba3f96813ca
CRs-Fixed: 605087
Add support for muting the RX and TX devices without affecting
the voice stream. This feature is added to enable voice call
quality testing by carriers.
Change-Id: I4d0f599e44ae24ce02d29333f496ef18f889ac55
There are specific xml files used for I2S based external modem.
Add check to load the correct xml file.
For I2S based external modem, voice sample rate can be 8k or
16k based on vocoder type. Call csd client api to get the
correct sample rate to configure voice call.
Change-Id: Ie9799336620f49a890355be74fe1830eb348cabe
- Translate PCM Offload format to pcm codec id
- Set 16/24 sub format id in compress params format
- Calculate offload fragment size for pcm
- Move offload calculation functions to platform file
- Update audio policy manager isOffloadSupported
for pcm offload profile. Use audio.offload.pcm.enable
Change-Id: I2064d646b66e8283fce2cc736e96fb93cba0a552
New CSD api is added to reduce device switch delay during voice call.
This api is called before enabling the devices so that csd voice call
sequence on MDM can run in parallel with device enable configuration
on APQ.
Change-Id: I9239ff21e1c07370516d0ed668e1e94af12f9eb2
Move platform_parser to root hal directory. Rename
platform_parser to platform_info. Change name of
XML file read from platform_info.xml to
audio_platform_info.xml. The xml now only needs
information for ACDB ID's that you want overwritten.
Names in the XML now match sound device enums in
platform.c.
Change-Id: I6f9bfcf4149f08d88cb2a95ba021bdf7296aaf4d
Add XML parser which parses the platform_info.xml
on the device. That xml contains ACDB ID information
and is populated from the device project folder to
the /etc folder on the device. It is used to overwrite
hardcoded ACDB ID's in platform.c.
Change-Id: I86419bf0f48bcf7f0125da58626adab1d23fa50a
-In-call recording feature support is added in APQ.
-In-call music delivery support is added in APQ.
Change-Id: Id473eac87da33f1f566b6910ec1feadcd3f08317
Listen needs be stopped when audio capture is active. Also
Listen needs to be started again after audio capture becomes
inactive. Stop and start listen based on audio capture device.
Change-Id: I05ca7cccd59b43a163604881bd9c6ee10cc8fba3
Depends-on: 510494
Send device disable command to external modem before voice call
mixer controls are disabled on APQ side. Enable device command
also should be sent only after enabling mixer controls on APQ side.
Change-Id: I2e4dc321b16f695564460771b3656cd2c4a089a9
APQ8084 uses external modem to make voice calls.
APQ uses CSD QMI interface to communicate with external
modem. Add csd client apis to support voice calls
Change-Id: I11fe0cc614ee38c42e95b40d5cd5e2d8b3afd43a
Sidipotu Ashok
Xiaojun Sang
Srikanth Uyyala
Banajit Goswami
Linux Build Service Account
Vidyakumar Athota
Ravi Kumar Alamanda
Apoorv Raghuvanshi
Ben Romberger
vivek mehta
Ramjee Singh
Mingming Yin
Laxminath Kasam
Karthik Reddy Katta
Narsinga Rao Chella
Preetam Singh Ranawat
Pradnya Chaphekar
Amit Shekhar
Venkata Narendra Kumar Gutta
Haynes Mathew George
Dhananjay Kumar
Subhash Chandra Bose Naripeddy
Shiv Maliyappanahalli
Helen Zeng
ApurupaPattapu
Kiran Kandi