reason: hwservicemanager: getTransport: Cannot find entry android.hardware.audio@7.0::IDevicesFactory/default in either framework or device VINTF manifest.
For earpiece to be properly enumerated as a output device,
set valid and supported out_ channel mask.
Change-Id: Ic83b009956ff43567b3acbb186d0069b16e91e83
Adding stereo config channel support to usb surround sound proflle
to address CTS-V failure issue seen due to mismatch in channel mask
support for record usecase
Change-Id: Ib50511d2fa6df24b3aa4764553cd48726129b2c2
Bring raw mixport back and drop "USB Device In" and "USB Headset In" from
"primary input" again.
This partially reverts commit 0b7df4e151.
Change-Id: I547518fbd6318dcb5fcb7c4ab85e8ad9da3d48ad
It will select "fast input" or "quad mic" as a compatible profile if
the parameter from recording app can't match the profile of
usb_surround_sound completely although connecting usb device.
Remove the usb device from the supported devices list of "fast input"
and "quad mic" for exact match.
Change-Id: Ie99886552082e99ac19193907b271131a2502d33
Remove the following mixports because of the usage of unsupported
flags:
* virtual_deep_buffer
* uplink downlink input
* voip record
* Multiroute
Rewire USB Device In and USB Headset In to go through primary input,
since uplink downlink input does not exist anymore.
Also, remove raw mixport and its uses, since it can cause sound
popping.
Xiaomi overrides the default audio policy configuration from
vendor/etc/audio/sku_lahaina inside of AudioPolicyManager,
and uses vendor/etc/audio_policy_configuration_mi.xml.
Use the proper configuration, but keep copying it into the old
location, since our AudioPolicyManager only knows how to load
it from there.
Routing primary output before voip_rx output will cause unexpected
RX/TX path suddenly and unmuted voice stream during when system
plays touch sound and switches path between handset and handsfree.
Bug: 152292858
Test: manual test in VoIP
Test: Regression test for VoIP(b/153276370) - pass
Signed-off-by: Jasmine Cha <chajasmine@google.com>
Change-Id: Ia60802ec887f0dfeaa0f93d151ba2f2c50e0d41d
Add mixer controls for lito and atoll display ports on QRD
and speaker-safe display path in IDP and CDP.
Change-Id: Idfa9860248db51d0c147b4fee2df479e200ca81e
Signed-off-by: Prasad Kumpatla <nkumpat@codeaurora.org>
Add VoIP Tx app type config for pcm and low-latency captures.
Bug: 133297310
Test: make a voip call
Change-Id: I5dd67167b7568fd08a7ae979b50417f2627dc7d9
Signed-off-by: justinweng <justinweng@google.com>
Remove Telephony Rx to make in call record use case through
different front end pcm device when sva enabled to solve concurrency
capture issue
Change-Id: I8a036423da689a4de9abe767a0b6a595b4e068f1
If two recording apps both select the primary input with
fast flag, second app will fail to start because the hal
doesn't support two inputs with the same use case if input
can't be reused since of the limit of primary input's max
open count.
Add the dedicated profile for “low-latency-record” to make
the first input can be reused if possible.
CRs-Fixed: 2550228
Change-Id: I233bd3eaf614b195df0a41402747f496d3a56b44
Usecase audio-record and audio-record-compress2 can both be
mapped to record_24 profile, if both the usecases are active,
when BT SCO mic is connected, due to no more available usecases
in audio hal, check inputs for device can fail for other input
profiles which support BT SCO mic.
This makes BT SCO mic connection to be rejected and phone call
can't route to BT.
Add the support for record_24 profile, so existing input handle
can be returned when checking inputs for BT SCO mic.
Change-Id: I63046ec87e35774474e08762fb6aa6275318b2c8
Backend config is missing for voice handset when TMU operator
is used due to which hw params is not set properly.
Add backend config for handset tmu operator sound device
in kona, lito and atoll variants.
[wight554: update interface to match SND_DEVICE_OUT_VOICE_HANDSET]
Change-Id: I199fa0bd33d682a129a4e77f25d32f615b175310
Signed-off-by: Volodymyr Zhdanov <wight554@gmail.com>
This DEFAULT adm_config profile will be used whenever a session is
in non-LPI with FFECNS disabled.
AFE will connect to ADM but will have none_topology.
Change-Id: I636667c376aa4309b67897e5dd162c4d318863f8
Built-in mic and Built-In Back Mic aren't in supported devices
list of primary input, so record_24 is selected if without
connecting any headset, if there is more than 2 capture
concurrency, it will fail to open input for primary input when
connecting headset in AHAL.
Add built-in mic to primary input sources to reuse input to avoid
to open extra input stream in AHAL.
CRs-fixed: 2643298
Change-Id: I1e062520f7fb03ae97c88341dd58f77177dbe030
Add sound trigger implementer version. Set the version value to 0x0101.
This is to add SVA4.0 support from STHAL so that the client app can
make use of this version info to get to know of platform SVA4.0 support
Change-Id: I85945ba49f31e2a7b9295e77a629737ddcb8d7d5
Allocate dedicated pcm node for in call recording, this can
avoid conflicting with generic audio recording for concurrent
recording scenarios.
Change-Id: I77f78a495623a55ad9c475d2554e7af326c68d7b
