Stefan Holmer 58c664c13d Clean up of CongestionController. ...

Removes unused methods and moves out ViERemb to Call.

R=pbos@webrtc.org, solenberg@webrtc.org

Review URL: https://codereview.webrtc.org/1663413003 .

Cr-Commit-Position: refs/heads/master@{#11527}

2016-02-08 13:31:53 +00:00

..

audio_receive_stream_unittest.cc

Clean up of CongestionController.

2016-02-08 13:31:53 +00:00

audio_receive_stream.cc

Add send-side BWE to WebRtcVoiceEngine under a finch experiment.

2016-02-04 12:12:31 +00:00

audio_receive_stream.h

Revert of Storing raw audio sink for default audio track. (patchset #7 id:120001 of https://codereview.chromium.org/1551813002/ )

2016-01-13 20:00:29 +00:00

audio_send_stream_unittest.cc

Clean up of CongestionController.

2016-02-08 13:31:53 +00:00

audio_send_stream.cc

Use separate rtp module lists for send and receive in PacketRouter.

2016-02-01 12:40:04 +00:00

audio_send_stream.h

Prepare the AudioSendStream to be hooked up to send-side BWE.

2015-12-07 09:26:32 +00:00

audio_sink.h

Revert of Storing raw audio sink for default audio track. (patchset #7 id:120001 of https://codereview.chromium.org/1551813002/ )

2016-01-13 20:00:29 +00:00

audio_state_unittest.cc

Move some send stream configuration into webrtc::AudioSendStream.

2015-11-16 15:34:59 +00:00

audio_state.cc

Move VoiceEngineObserver into AudioSendStream so that we detect typing noises and return properly in GetStats().

2015-11-06 23:34:58 +00:00

audio_state.h

Remove mutable from rtc::CriticalSection members.

2016-01-25 11:52:53 +00:00

BUILD.gn

Support for unmixed remote audio into tracks.

2015-12-12 00:37:14 +00:00

conversion.h

Implement AudioSendStream::GetStats().

2015-10-27 10:35:30 +00:00

OWNERS

Split webrtc/video into webrtc/{audio,call,video}.

2015-09-25 11:58:39 +00:00

scoped_voe_interface.h

Re-Land: Implement AudioReceiveStream::GetStats().

2015-10-22 08:49:39 +00:00

webrtc_audio.gypi

Support for unmixed remote audio into tracks.

2015-12-12 00:37:14 +00:00