Add support for routing voice calls to devices in other audio
HALs by allowing playback and capture to/from AFE proxy
Change-Id: I2c6a1ddec072e1d5f1a8b7ded874e9c082a7b810
During skype call if headset is inserted next voip
input request is failed.
VoIP data sample rate is not being reset properly is
causing the aforementioned issue.
CRs-Fixed: 725583
Change-Id: I9f85936aec0caeaf53b9aeff93a481fd38afe3ff
Fix unused param warnings.
Take care of coding style issues.
Remove unwanted code to set acdb_settings.
Fix compilation issues on 8960 platform.
Change-Id: Ifc72472b487abee83ff85e21fd324cee938973e9
When voip input started, active input stream is not updated.
Therefore, as active input stream is NULL no tx device switch
occurs for a voip call with primary output.
Fixed this by updating active input stream for voip when
voip input started.
Change-Id: I058bfaa0091eea4a0259224de17be32f271dbdc9
CRs-Fixed: 710923
If there is any difference in the signature of a function declared
as extern, it will not be reported by the compiler and may result
in unexpected results when executed.
All the API functions should be declared in a header file.
Change-Id: I89662e23da8118c3a9eac728b389498ed52e19c2
adev->voice_device_set flag is to indicate voice call
device routing update from policymanager to HAL. It is
set to true in voice_start_call and reset in update_calls()
which causes mismatch in flag update during back to back
voice calls scenario. Update adev->voice_device_set flag
in voice_stop_call instead of update_calls(). Rename
voice_device_set flag to in_call for readability.
Change-Id: Ie07105671f254899890bdb4c0635c7dc1f55dbff
-This change adds DSDA support for APQ based targets
where external modem is used for voice calls.
-Add new CSD API to support local call hold.
Change-Id: I7743a1df43dc1abac4e325ff104ec1bb64c9e12b
- All VoIP, compress, deep buffer, record usecases need to close
before voice call setup and don't allow these usecases during
voice call on 8916.
- setMode(IN_CALL) itself is not sufficient to know if we are
in actual voice call.
- Add keyvalue parameters support to know if we are in voice call
and policy hal can query the same and make decissions based on it.
CRs-Fixed: 662804
Change-Id: I20d660fadd07c4d227a7e3dfe356fc39806ca167
-When voice call interrups ongoing voip call, voip call goes
to hold state and voice call becomes active. During voice call end
stop_call is not being called because mode is not set to MODE_NORMAL.
Voip call becomes active and mode is set to MODE_IN_COMMUNICATION
after voice call end.
-Fix this issue by removing the dependency on MODE_NORMAL to end
voice call.
Change-Id: I0103f5c6833fb929cb7975f838bc2c5ec4d94313
CRs-Fixed: 633437
- Unable to apply the mute in MT SIP call during Voice call
in CALL_HOLD.
- Mute is only applied if the voice call state is INACTIVE.
Thus when SIP call is made during Voice call in CALL_HOLD,
the mute is not being reflected.
- Fix this issue by checking voice stream type to allow
the mute to be applied.
Change-Id: I3af5225edd8e9a4123867b647de9405d5c4b9efc
CRs-Fixed: 642893
Remove update_mixer (as its unused) and change function signatures
wherever used.
Remove code chunk to set BT sample rate before enabling device as
thats part of the mixer path file now.
Change-Id: I5eb3bce5183acf505596be2b0ec93cdbd728bdaa
CRs-Fixed: 581453
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
- Voip driver requires voip output stream to be opened
to start itself. But it is possible that some applications
may use voip input and normal audio output. Code changes are
done to handle this scenario.
- It is possible that some applications may use different
sample rates for voip input and voip output streams. It is
handled now in such a way that, if the sample rate of the
secondly opening stream does not match with that of the
first one, the second stream will not be allowed to open
voice path for VoIP.
Change-Id: Id9d86181eeee2c4ea0aa49a723ca92c1961a559b
CRs-fixed: 601785
String returned from str_parms_to_str for set/get parameter in
debug logs is not de-allocated. Fix is to make sure the memory
allocated for set parameter key value pair is de-allocated.
CRs-Fixed: 610079
Change-Id: I5027d8f1741c9e7223e7b4721f8d5473404998d3
This change is needed to fix a compilation warning in Compress VoIP
code when the feature flag "AUDIO_FEATURE_DISABLED_MULTI_VOICE_SESSIONS"
is defined.
Change-Id: Ie19f542dc71705bc27a7278c83236904782f3a5e
CRs-Fixed: 590219
Add support for voip_out_stream_count in get_parameters which
will be queried by Audio Track to decide whether to open direct
output or not in case of VoIP.
Change-Id: Ic0d058dff32d9002cd452689636f342e748a7fe5
HAL doesnt let voice calls start if the device is set to speaker
as voice calls generally start on devices other than speaker.
However QCHAT calls start by calling setForceUse and setting the
device to speaker.
Fix by updating a flag whenever device routing happens when
AUDIO_MODE_IN_CALL. Reset the flag when mode is AUDIO_MODE_NORMAL.
CRs-fixed: 596074
Change-Id: I546959d2b0123828562dba1f6439aa494a365c3f
When setparameters() is called and if the known keys are not found,
the implementation returns -2 error.Fix this by setting error only
when keys are found and there is an error while executing corresponding
implemention.
CRs-fixed: 589280
Change-Id: If5544d6fdeac47ddfc1f7e8e18bfa81e2ef5cbc2
- HAL always assumed SOUND_CARD/MIXER_CARD
number to be zero. This could cause HAL
not to load when some other card is
detected earlier to HAL.
- Fix by removing these Macro(s) and reading
correct sound card number runtime while loading
- Corrected its use cases across HAL replacing it
with appropriate platform API
CRs-fixed: 580226
Change-Id: I4512e4a2f7f81d8415c3ba8ec393c96867e6d69a
- Correct the framesize for 16KHz VoIP use cases
- Add proper checks to enable compress VoIP use case
- Add get_parameters() support to query audio mode
Change-Id: I470d538f1e136780264e81b509fb439baead603d
If setParameters(vsid, callstate) call is processed before
setparameter(routing), voice calls will be started on speaker.
Fix by not starting voice calls by checking if the primary output
device is speaker.
Change-Id: If3ecb83bbcb02d5da40d66340276263413adabc9
Telephony will call set_mode(AUDIO_MODE_NORMAL) when a
call is ended which will trigger audio policy manager setting
routing with audio usecase compatible device. Voice calls can
still be active if RIL has not yet called
setparameters(vsid,callstate). This would result in routing
voice call usecases with incompatible device for voice calls.
Fix this by ending all voice calls when set_mode(MODE_NORMAL)
is called.
Change-Id: Id2c9f2ff9ed46969e5cbd27b525b81735c1d49d8
Compile voice extensions based on the flag
AUDIO_FEATURE_DISABLED_MULTI_VOICE_SESSIONS defined
in BoardConfigVendor.mk.
Change-Id: I06c42bf4b79294fed2077c1d2771ed155f68de00
In multisim scenarios, local call hold tone has to be played
to the uplink on the sub on local call hold. Add incall music
delivery feature so that call hold tones can be played to the
uplink.
Change-Id: I8cecf0139926a4bfee603c02542bbedc67720bf4
The uplink, downlink and the combination of both can be
recorded using sound recorder app. Add support for incall
recording feature in audio HAL.
Change-Id: Ia828ab5b3af8044d82b6ef8eb989ae0ba9bbba16
Update HAL to add multi SIM voice support. Seperate
out voice features to voice files.
Add getprop/setprop for BT sample rate, Fluence and Slow talk.
Change-Id: Ief61466f78c7b915549981a28fc7ff32731607a5
Ravi Kumar Alamanda
Venkata Narendra Kumar Gutta
Linux Build Service Account
kunleiz
Mingming Yin
Vidyakumar Athota
Haynes Mathew George
kunleiz
Shiv Maliyappanahalli
Vicky Sehrawat
Narsinga Rao Chella
Krishnankutty Kolathappilly
Apoorv Raghuvanshi