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
- When an incoming MT HFP call is accepted during an ongoing
MO HFP call, the MO call goes to held state. Later, if this held
call is rejected, Tx is not heard at the far end of the incoming
call from that moment.
- When the held call is rejected, the input device gets disabled due
to which Tx samples are not captured.
- Fix the issue by ensuring that the input device is not set to none when
HFP is active.
CRs-Fixed: 656928
Change-Id: Ie3f0886c0412bfdb3e450c54e9f4b651e992f84e
New machine driver has been added for msm8994. Update PCM device IDs
for voice use cases accordingly. This change is needed to support
voice use cases on msm8994.
Change-Id: Iba5f9423fe99668ccfe3a692c2a3148977d4dfc2
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
Add dolby audio_extn API's to set dolby license.
License needs to be set with dsp everytime it is loaded.
Change-Id: I21d58a21d65c823469d066f4ba5edc5b2660ea1e
APQ8084 uses an external chip for FM. Add support for
mixer control for external FM volume control.
Change-Id: I296d0b32fce632a296057e6d7b48698d9b0648fe
Echo cancellation is not working in VoIP calls.
Echo reference port is set as I2S_RX which inturn sets
PRIMARY_I2S_RX as echo reference port while configuring
ADM is causing the issue.
8x10 is using SECONDARY_I2S_RX port in Rx path. Update
the EC reference port accordingly.
CRs-Fixed: 646310
Change-Id: If949c6c7487c36a02e5bcedb3c779f502592ce64
- Correct the fragment size to use bytes instead of bits
per sample
- Use 80ms as pcm offload buffer duration for video streaming
usecases and 1 sec for video + audio usecases
Change-Id: Iec8e24c2b158368742a55710a46da94ce5d3c170
For APQ8084 voice call use-case, Adaptive ANC (AANC) is used
by sending an appropriate flag to CSD for enabled AANC.
Change-Id: Ifc4609c16ef3c5d905628ff22853fc44fd81a73a
- Ensure reference device is set if audio is routed
to speaker/headset during HFP usecase.
Change-Id: Ia404aaa8ec8b80e9f4be14840dfef3bf9afef435
CRs-Fixed: 641513
Due to changes in machine driver, the pcm ids for VI
feedback and Voice2 are changed, so update the PCM
ids to support these use cases.
CRs-Fixed: 642001
Change-Id: I1cb23b57448bdecc5229e39b866d1c056029f712
- Incorrect PCM device ID is being selected for
incall music delivery usecase.
- Select proper PCM device for incall music uplink-2.
Change-Id: Ic53495790fbd4bf7eac3f45811f2a8973ea3c6d9
CRs-Fixed: 642160
-Enable speaker protection compilation flag to support
speaker protection feature for audio playback and voice
calls on APQ8084
-Append speaker-protected backend name to usecase only
for APQ8084 to differentiate with regular voice call as
routing controls are different.
Change-Id: I92ed26c9df0dc2f152e2bbf8b29362e32fff9844
- Some BT solutions set sample rate only at the time of BT connection
but not for every device switch. Due to this sample rate is not
set properly after device switch.
- Update bt-sco sample rate before device enable instead of sample rate
set parameters call.
Change-Id: I07cb0c186296b00a99713b67628fb87b2f134d32
With the introduction of Quaternary MI2S Hostless Front-end
DAI for APQ8084, QCHAT FE DAI has been moved to possible at
the bottom of the list. Update the position of the QCHAT FE
as per the new position.
Change-Id: I3f72db194cfee9a7d6a6ba04ff37a293639d6293
- 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
ADSP supports Fluence Broadside algorithm in Speaker Phone mode
for voice call and audio recording usecases. Add support for
the same in audio HAL.
Change-Id: I6682f72f8f47ce1f261511cb150e506de6f3dddb
CRs-fixed: 590495
- There is no LCH tone playback on sub2 since the
pcm ID and the DAI links are missing for 8974.
- Add pcm id for 8974.
Change-Id: I518d5eb3bfeaddd3b00281db064429087b10a37b
CRs-Fixed: 626562
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
- Set proper USB headset channel mapping before
starting playback.
- Update audio_extn_set_afe_proxy_channel_mixer
to handle channel mapping based on channel count.
CRs-Fixed: 620828
Change-Id: I9660e3088bc747c30891b0aa7ec8d22096dd0fa2
- Ensure EC reference is set when the source is MIC and fluence is
enabled in the regular audio recording path.
Change-Id: I6648f41fcc578ded13e4fecdb7ef35bc6b0579d4
CRs-Fixed: 628085
Update csd volume and mute controls with new apis which
takes ramp duration as a parameter. Set default mute
ramp duration to 20 ms.
Change-Id: Id8dcbc9bc12ac4e9c56962abc58b6bd1de0530a7