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
Update acdb extn and platform to pass Instance ID support info
in order to enable ACDB loader use this info and call the respective
ACDB SW APIs. Update GEF APIs to support instance ID feature.
Add a property to query the instance ID support and use that
to call ACDB loader for instance ID support.
CRs-Fixed: 2199639
Change-Id: I64b41faf23cb42ec2af1695adb8a7021cad996dc
Vendor created android system properties are required to have
vendor/qti prefix. Update vendor specific audio system
properties used in HAL.
CRs-Fixed: 2034869
Change-Id: Ibdb1d12fedee1ce12073e8b90fdec814318f3acf
Add support for multi-copps for multiple BEs with independent
calibration data. This allows for more accurate calibration of device
copps.
Change-Id: I2628cf16564a6cd596e7f2663063a7cd64749bdc
CRs-Fixed: 1110415
Add changes to read acdb metainfo key and module name from platform
info xml. Changes to add acdb key values to a list and query key value
based on module name.
CRs-Fixed: 1081424
Change-Id: Ia287a27c86f63fea16cdb35d553de6e2e853b4e9
- if customer mapping is added in platform info file, use that
- in absence of customer table use default mapping
CRs-Fixed: 1089771
Change-Id: I4bf8bcf1913f16ad6298ff1f3fa5dd649c889b2a
When a new playback usecase has to be started, existing playback
usecases might have to be re-routed (or not) to the new device
based on usecase requirements or h/w limitations
(e.g headphones and speaker sharing the same backend).
This change addresses this requirement by deriving the
new device based on pre-defined cases.
CRs-Fixed: 1077122
Bug: 31671778
Change-Id: Ic0fd4e8d2c1119e7198dc5bb5e5a51817f0110c1
When aanc is enabled during voice call, TX and RX path mixer
controls are issued back to back, this causes the codec path to be
enabled from TX to RX as the internal loopback for aanc exists. Due
to enabling this path, mute is observed on handset. This issue
is fixed by treating aanc as special usecase and changing the enable
and disable sequence of aanc mixer controls.
Change-Id: I305156cb80e6faf35bd6ab4fc089740ffa847f43
- Enable ASRC mode for incoming mix path use case(Headphone backend)
if Headphone 44.1 or Native DSD backends are enabled for any of
the current use case.
e.g. 48-> + (Naitve DSD or Headphone 44.1)
- Disable current mix path use case(Headphone backend) and re-enable it with
ASRC mode for incoming Headphone 44.1 or Native DSD use case.
e.g. Naitve DSD or Headphone 44.1 -> + 48
Change-Id: I94df832ba5e2a26e9179d66d4639f168b5c30eeb
Add devices to support right speaker as mono device along with
speaker protection support. It also provides option to switch
between left and right speaker using setparameter.
CRs-Fixed: 1051075
Change-Id: Iac140448bd6f49444d22a9e496b71e3502dfaefe
Signed-off-by: Rohit kumar <rohitkr@codeaurora.org>
- Added support for DD/DDP/AAC/MP4 decode through QAF.
- Added support for Rencoding to DD/DDP through QAF..
- Added support for DAP through QAF.
- Added suport for mixing DD/DDP with PCM through QAF.
- Added suport for mixing AAC with PCM through QAF.
- Added support for Dual Decode through QAF.
CRs-Fixed: 1063555
Change-Id: Icc3ddf62f9bc1c45e1027fb93bf9fa0e154a783e
-Enable ASRC mode for headphone backend if headphone 44.1 or
native DSD backend is active.
-Disable headphone backend and reopen it with ASRC mode for upcoming
use case over headphone 44.1 or native DSD backend
Change-Id: Ic8fb0ab9f254d3472fda49dbb824f622d518a451
-set passthrough flag and diable gapless for native dsd
-Select new backend for native dsd.
-Add support for e2e playback of clips with sampling rate
multiple of 44.1.
Change-Id: I6aa0ef5ea176a0923b0b88924ab046f9a11b7b12
- Add new audio extension module for a2dp
- Add support to configure AFE encoder based
on BT device capabilities
- Add support for soft handoff to re-configure
AFE based on new BT device capability
Change-Id: I5d710fda1a8d8fa3b5d85aaa2b3096cff8fd7ce5
-check EDID info for supported audio configuration,
and config audio backend for HDMI audio playback
with proper bitwidth and sample rate.
Change-Id: Iae65e00b0f4d2f74f303ef03610f8b3a098dc7aa
Add device sidetone support for USB ADSP solution if the USB headset
has the sidetone capability.
CRs-Fixed: 1019158
Change-Id: Ie40c286eb9988aeb5093f32c7f81cdb142e88eb6
-Add support for 24 bit packed audio in audio hal.
-Disable gapless for AV playback and direct pcm usecase,
this ensures that the buffering in DSP is not more.
-Simulate rendered time stamp for direct pcm usecase
based on the number of frames written to the compress
driver, bufferring in the driver and DSP latency.
-Pass mixer instance to offload effects library to avoid
an unnecessary mixer_open call, this optimizes audio
start delay in compress playback.
Change-Id: I422a53af5632eaf6cc362a6c44f62ff8412965f7
-Changes to support 24 bit record if input format request is
AUDIO_FORMAT_PCM_8_24_BIT or AUDIO_FORMAT_PCM_24_BIT_PACKED
Change-Id: I68076524ccccbf9f0be3c88bb01180ae7e4fd8b1
- update read_hdmi_sink_caps to support dynamic sample
rate, channel, and format based on HDMI EDID information
Change-Id: I69b3f19cb1d81c3e51695adae4f2379cbb439dc8
If the wired headset/headphone/line devices are handled by
a different backend than speaker/earpiece devices, the combo
devices such as speaker+headphones can be split into individual
devices and enabled/disabled independently.
Conflicts:
hal/msm8916/platform.c
hal/msm8974/platform.c
Change-Id: I30e90f398ef204d106b5d8dac8fba166583bdbc8
If the wired headset/headphone/line devices are handled by
a different backend than speaker/earpiece devices, the combo
devices such as speaker+headphones can be split into individual
devices and enabled/disabled independently.
CRs-fixed: 989410
Change-Id: I379b56940eaf384d3cd1d9fe392b7fec65df023b
Feedback speaker protection(FBSP) is supported in WSA
analog mode. Add the required changes to support this
feature on 8937/8953 internal codec platforms.
CRs-Fixed: 970156
Change-Id: I1e5f6c1408ba6eb1c216cf8b254ac29551837dbb
-Issue: Two sessions with fluence topology are created
in case of voice call + incall voice recording.
-Rootcause: To avoid fluence topology for incall
recording usecase when voice call is active, default
input device's audio calibration was used earlier.
But while re-organizing audio_extn_utils_send_audio_calibration()
incall recording usecase is not considered.
-Fix: Send default input device's audio calibration
for incall recording usecases
Change-Id: I0d3734be3c793d350b1c73d8f8fc4380bf06fe30
Some of the parameters passed to functions in older platforms
are not used anymore, though we maintain the platform code for
backward compatibility.
Mark the unused parameter explicitely as __unused.
Change-Id: I814f5f233590ecfd3e79bf58b97ad6d37be74601
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 enable vbat feature. This includes required changes
to select vbat speaker devices for audio, voice usecases and
sending the vbat calibration data to codec driver.
Change-Id: I2234bee3cc4d6f16934a4b27ef9da21cde74bb48
During device switch, all active capture usecases need to be
switched to new sound device if they all share same backend.
This is not happening because of an improper check of capture
usecase's device against output backend devices. Correct the
check to compare against input backend devices.
Change-Id: Iaf43c30b091f78c2ffb8b2740f241f2edfbe5a08
CRs-fixed: 873528
Update required acdb ids and channel config for supporting
speaker protection in 8952 WSA Analog mode.
Change-Id: I2642ba8cf0fc5854cd26ceaf6f773786b87bff74
During device switch, all active capture usecases need to be
switched to new sound device if they all share same backend.
This is not happening because of an improper check of capture
usecase's device against output backend devices. Correct the
check to compare against input backend devices.
Change-Id: Iaf43c30b091f78c2ffb8b2740f241f2edfbe5a08
CRs-fixed: 873528
- expose new api from audio HAL that volume listner module will call
to set volume calibration level
Change-Id: If74bf66d32def85022d79ccb9f84c3b85c8a2dc9
Voice call setup latency value is exceeding 500 msec. It is due to
primary output selecting handset instead of voice handset causing
the extra delay. Fix this by selecting voice handset when the mode
is set to MODE_IN_CALL.
CRs-Fixed: 855783
Change-Id: Ibfaee75d658bffbcf2e9cdeaa589e0487c214b41
Voice call setup latency value is exceeding 500 msec. It is due to
primary output selecting handset instead of voice handset causing
the extra delay. Fix this by selecting voice handset when the mode
is set to MODE_IN_CALL.
CRs-Fixed: 855783
Change-Id: Ibfaee75d658bffbcf2e9cdeaa589e0487c214b41
Naresh Tanniru
Weiyin Jiang
Aditya Bavanari
Haynes Mathew George
Aniket Kumar Lata
Siena Richard
Dhanalakshmi Siddani
Vidyakumar Athota
Preetam Singh Ranawat
Rohit kumar
Lakshman Chaluvaraju
Shiv Maliyappanahalli
Mingming Yin
Kuirong Wang
Ashish Jain
Manish Dewangan
Linux Build Service Account
Sidipotu Ashok
Venkata Narendra Kumar Gutta
Xiaojun Sang
Karthik Reddy Katta
Banajit Goswami
Narsinga Rao Chella
Laxminath Kasam
Panwar Vivek
vivek mehta