henrika@webrtc.org a954c07ee1 AppRTCDemo (Android): built-in AEC should be enabled if device supports it and in combination with Java-based audio layer ...

BUG=4034
R=andrew@webrtc.org, perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32179004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7849 4adac7df-926f-26a2-2b94-8c16560cd09d

2014-12-09 16:22:09 +00:00

..

mock

Mark all virtual overrides in the hierarchy of Transport as virtual + OVERRIDE.

2014-10-10 09:42:53 +00:00

voe_audio_processing.h

…

voe_base.h

Remove Get/SetNetEQPlayoutMode APIs

2014-09-22 14:30:10 +00:00

voe_codec.h

Renaming SetOpusMaxBandwidth to SetOpusMaxPlaybackRate

2014-09-03 12:28:06 +00:00

voe_dtmf.h

Reland "Remove DTMF status methods from Voice Engine" r7276

2014-10-01 08:23:21 +00:00

voe_errors.h

This CL is to adding feedback of packet loss rate to encoder in voice engine. A direct reason for doing it is to make use of Opus FEC, which can adapt itself to changes in the packet loss rate.

2014-05-28 09:52:06 +00:00

voe_external_media.h

Removes parts of the webrtc::VoEExternalMedia sub API as part of a clean-up operation where the goal is to remove unused APIs.

2014-05-12 12:19:19 +00:00

voe_file.h

Removes parts of the webrtc::VoEFile sub API as part of a clean-up operation where the goal is to remove unused APIs.

2014-05-12 12:24:10 +00:00

voe_hardware.h

AppRTCDemo (Android): built-in AEC should be enabled if device supports it and in combination with Java-based audio layer

2014-12-09 16:22:09 +00:00

voe_neteq_stats.h

…

voe_network.h

Use size_t more consistently for packet/payload lengths.

2014-11-20 22:28:14 +00:00

voe_rtp_rtcp.h

Use size_t more consistently for packet/payload lengths.

2014-11-20 22:28:14 +00:00

voe_video_sync.h

…

voe_volume_control.h

Removes parts of the webrtc::VoEVolumeControl sub API as part of a clean-up operation where the goal is to remove unused APIs.

2014-05-14 11:51:45 +00:00