Update the source files for ma_listener, cirrus_playback and
maxxaudio to latest versions.
CRs-Fixed: 2388992
Change-Id: I62eecf0356ee942d085ee94b848225a91704e688
- use effect module to monitor stream's volume
- the control(maxxaudio) monitors active device and
changed volume from HAL, and send audio calibration
to acdb via acdb_loader.
- support specific USB headset
Enabled function:
- LR channels swap
- load external tuning table for volume preset feature
Bug: 74360112
Test: manual effect test, usb headset test
Change-Id: I9b35436d6abebcfa250beb4857dfa0be9248f2b5
(cherry picked from commit 270b7762c1c6c7b69400ba28a2ea2bed7af1906a)
Add auto platform specific pcm device id and
playback volume control to support hfp with
dual asm loopback.
Change-Id: I6bd2ae63c8733dd770f48c1b2557c4d3d5a6b101
- Add changes to support BT IPC sink library.
- Add mixer control support to send decoder configuration for
split A2DP sink usecase.
- Add hw loopback support for split A2DP sink usecase.
Change-Id: I3b015385ce1fdd5c0cbf512fe06b73143a32ae2d
Until now speaker playback only supports stereo
and is limited to default sample rate.
Update code to support QCS405 configurations with
CSRA soundcards that can have up to 16 or 32 speaker
output channels and allow sample rates up to 384kHz.
Change-Id: Ib4ed5edafca6f8b15134ca66db4bf1ef719ec15d
During in call recording, input device is same as that of voice
use case but the backend used in routing driver is different.
Due to mismatch of backend ids, Wrong acdb id and app type are
being set in routing driver when adm port is opened.
Defined new devices for in call recording and used the new devices
to retrive the backend ids.
CRs-Fixed: 2183315
Change-Id: I278996185badad640e16f60730176916d81e3626
Define module id for tmic devices.
Add tri mic handset and speaker devices to effect config
table.
CRs-Fixed: 2280317
Change-Id: Ia3d22f92a7b4fd483a54192f3119814c8c112229
- Single mic fluence devices are never used when fluence type is
not enabled. When fluence is not enabled, single mic devices
should use ECNS.
- Remove checks for fluence type to enable set echo reference for
non fluence usecases.
CRs-Fixed: 2275470
Change-Id: I6d76e6332960663583a082c0de933854403255f5
Vipertooth has hardware support for LINE IN device.
Adding support in audio hal to select line-in device.
Change-Id: Ia783ef986724a51a25a33f91df05b58984ecc87d
Add support to get active microphones on input stream.
Get microphone device id and channel map based on active
usecase and sound device. Active microphone characteristics
are retrieved from platform info using this device id.
Change-Id: I63ec99e70980869097fa896bc4b309fc0583fb3a
Add device entries and additional support to enable
voice speaker stereo and stereo-protected features.
Change-Id: Iad40d6a297824181ed380a5a3f37254f58144de5
During in call recording, input device is same as that of voice
use case but the backend used in routing driver is different.
Due to mismatch of backend ids, Wrong acdb id and app type are
being set in routing driver when adm port is opened.
Defined new devices for in call recording and used the new devices
to retrive the backend ids.
CRs-Fixed: 2183315
Change-Id: I278996185badad640e16f60730176916d81e3626
Add new sound devices and acdb device ids to
support Fluence V8 feature. Enable the sound
devices based on the tri mic fluence support
property.
CRs-Fixed: 2155314
Change-Id: Ibfcd022d69bdd11a280caee6d0f03e5909ed2961
-Add generic handset mic to support different use-cases
on this single device using tx app_type config.
-The change is guarded with a system property which should
be enabled only if tx app_type support is already available
and generic device definition is added in acdb files.
Change-Id: I3fea2e12bd99a0d229fb5753d47dba2501ea528b
Loopback gain setting failed in DSP due to invalid FM
port configuration. Assign correct FM backend port
to fix this issue.
CRs-Fixed: 2127308
Change-Id: Iaf8106f1dda40e84e8128331f1e3c0cc4556ea62
Add support for 6mic Fluence pre-processing of
captured stream using Far Field Voice (FFV) algorithm.
Change-Id: I7a643014f3e090482db7772dce2fc812fd469b2d
- Increase direct PCM buffer duration (80ms) to receive
linear time acks on write for fractional sample rates
- Fixes 44.1 Khz timestamp updates.
Change-Id: I0f657bae0d6e4645f2fcf2b37b97014b89544008
Matrix limiter is used to prevent saturation
when mixing multiple audio streams. It is
enabled when both the BD output flag is set
and the property audio.matrix.limiter.enable
is set to true.
Change-Id: Ibf06bd2444874ae555e98a26ed645f2455a3e1a3
Add support in HAL to send mix matrix params using downmix,
Pan-Scale mixer controls. Add new interactive usecases and
also add support to open and teardown for interactive streams.
Add support in test app for supporting up to 9 concurrent
streams. Add support for mixer matrix params handling.
Change-Id: I0dc5b908ee779b2b2c526a67609c057f591f26e7
- Support compressed input format in transcode loopback extension
- Support compressed input format in target platform
- Test application changes to support pcm and compressed input formats
CRs-Fixed: 2061945
Change-Id: I693594164e05c09ffece8fa705149c7b45c2e5d3
Signed-off-by: Siddartha Shaik <sshaik@codeaurora.org>
Add dedicated streams for VoIP usecase. This includes adding
a new direct output and input types that use audio playback
and record paths respectively. It falls back to compress VoIP
wherever this feature is not applicable.
Change-Id: I9b7654819b9d8deb00f8888fbfc31f0fbbae5266
Hari Veerubhotla
Linux Build Service Account
Samyak Jain
Meng Wang
Aalique Grahame
jasmine cha
Surendar Karka
Derek Chen
Florian Pfister
Naresh Tanniru
Dieter Luecking
Divya Narayanan Poojary
Karthikeyan Mani
Sharad Sangle
Vignesh Kulothungan
Ralf Herz
Ramu Gottipati
Vikram Panduranga
Dhananjay Kumar
Tanya Dixit
Xiaoyu Ye
Linux Build Service Account
Aditya Bavanari
Josh Kirsch
Ashok Sidipotu
Haynes Mathew George
Garmond Leung
Garmond Leung
Aniket Kumar Lata
Ben Romberger
Varun Balaraj
kunleiz
Siddartha Shaik