- Add an audio extension for ACDB
- ACDB Init from AHAL and STHAL are routed via this extension
CRs-Fixed: 2028493
Change-Id: Ib63c8fa4bf1f89d4dbfce8b6584ac1d8271e5f38
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
-In case of playback over HDMI/USB, the device might not support
the same sample rate as of the stream, though it is preferred that
resampling happens after post processing, but if device sample rate
is not even a multiple of stream sample rate, ensure that resampling
happens in the COPP.
Change-Id: Ia8cdff5294433ec099b703001d11da905bf82c9b
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
During display port disconnect, adev_set_parameter() is calling
platform_invalidate_hdmi_config() before the audio extension.
Attempts to read the current display type in order to determine
which sysfs node to communicate with. The solution is to
relocate invalidate function in audio extension to ensure that
it does not get called prematurely.
Change-Id: Ifc7bc8cc753855edacb8fa90a9f784086fef7594
CRs-Fixed: 1082254
Surround Sound Recording(SSR) supports only channel position
representation while default channel mask for multi-channel
recording is always index representation.
Return updated channel mask in case index representation is sent
for channel count 6. This ensures audio flinger reopens the session
with updated position representation channel mask.
And also handled the case if any client requests recording with
multichannel when SSR is not enabled, will check the max_mic_count
supported by the device and update the channel mask accordingly.
Change-Id: Ib925a5f31d50182cd595e547f3412ed809224a5a
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>
Update to add BE channel configuration for USB capture and more
BE configuration options for sample rate and channel number.
Change-Id: I1e0af7a19a67aecba5afae95ecc62eec4a2d650b
CRs-Fixed: 1071868
- 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
-Update platform_check_codec api to enable support for multichannel
tunnel USB
-Update condition to check if voice call is active on a specific backend
then only force configure it
-Update the check to ensure that offload callback is triggered only for
non blocking writes.
Change-Id: I0dbd61c28b24cdba22f63b0e014eb7c362e2df34
Monitor sound card status and report online/offline
events back to the streams registered with it and to audio_dev.
Bug: 30075678
CRs-Fixed: 1053073
Change-Id: Iea74d5c83f7fe92ba4a24f4f4650ce255bdb48e8
-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 changes to support PCM_FLOAT format using direct track.
-For targets supporting true 32 bit, convert FLOAT pcm data
to 32 bit.
-Targets which dont support true 32 bit, convert FLOAT or
32 bit input to 24 bit Packed in HAL.
Change-Id: I8052c8b878bbd4d70c0453df054953d494773b0a
-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
- 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
Vignesh Kulothungan
Siena Richard
Dhanalakshmi Siddani
Aniket Kumar Lata
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Haynes Mathew George
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Kuirong Wang
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Weiyin Jiang
Shiv Maliyappanahalli
Preetam Singh Ranawat
Mingming Yin
Manish Dewangan
Sidipotu Ashok
Venkata Narendra Kumar Gutta
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