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27925de9512c67c9934ba535dc83ffe04ed999fe
platform-external-webrtc/webrtc/call
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stefan 27925de951 Enable trendline experiment and bayesian bitrate estimator experiment by default. 
BUG=webrtc:6566, webrtc:7415

Review-Url: https://codereview.webrtc.org/2777333003
Cr-Commit-Position: refs/heads/master@{#17491}
2017-03-31 14:38:06 +00:00
..
audio_receive_stream.h
Remove usage of VoEVolumeControl from WVoE and Audio[Send|Receive]Stream.
2017-03-02 01:02:23 +00:00
audio_send_stream.cc
Replace NULL with nullptr or null in webrtc/audio/ and common_audio/.
2017-02-26 12:18:12 +00:00
audio_send_stream.h
Added a new echo likelihood stat that reports the maximum value from a previous time period.
2017-01-15 16:29:46 +00:00
audio_state.h
Moved call.h and most of api/call/* into call/
2016-12-07 12:53:04 +00:00
bitrate_allocator_unittest.cc
Reland of Fixes a bug where a video stream can get stuck in the suspended state. (patchset #1 id:1 of https://codereview.chromium.org/2703393002/ )
2017-02-21 15:28:31 +00:00
bitrate_allocator.cc
Reland of Fixes a bug where a video stream can get stuck in the suspended state. (patchset #1 id:1 of https://codereview.chromium.org/2703393002/ )
2017-02-21 15:28:31 +00:00
bitrate_allocator.h
Reland of Fixes a bug where a video stream can get stuck in the suspended state. (patchset #1 id:1 of https://codereview.chromium.org/2703393002/ )
2017-02-21 15:28:31 +00:00
bitrate_estimator_tests.cc
Revert of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #8 id:410001 of https://codereview.webrtc.org/2781433002/ )
2017-03-30 17:44:38 +00:00
BUILD.gn
Define RtpTransportControllerSendInterface.
2017-03-27 12:36:15 +00:00
call_perf_tests.cc
Revert of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #8 id:410001 of https://codereview.webrtc.org/2781433002/ )
2017-03-30 17:44:38 +00:00
call_unittest.cc
Reland "WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType"
2017-03-27 14:15:49 +00:00
call.cc
Delete unneeded includes of deprecated system_wrappers include files.
2017-03-30 07:31:15 +00:00
call.h
Inform jitter buffer about FlexFEC protection.
2017-01-27 14:47:55 +00:00
DEPS
Passed AudioMixer to AudioState::Config.
2016-11-17 14:48:56 +00:00
flexfec_receive_stream_impl.cc
Don't allocate any RTPSender object for a receive only RtpRtcp module.
2017-03-20 10:52:39 +00:00
flexfec_receive_stream_impl.h
Rename flexfec AddAndProcessReceivedPacket --> OnRtpPacket.
2017-02-16 14:52:32 +00:00
flexfec_receive_stream_unittest.cc
Rename flexfec AddAndProcessReceivedPacket --> OnRtpPacket.
2017-02-16 14:52:32 +00:00
flexfec_receive_stream.h
Make |rtcp_send_transport| mandatory in FlexfecReceiveStream::Config.
2017-01-13 15:41:19 +00:00
OWNERS
Remove all references to GYP
2016-11-16 19:11:38 +00:00
rampup_tests.cc
Enable trendline experiment and bayesian bitrate estimator experiment by default.
2017-03-31 14:38:06 +00:00
rampup_tests.h
Reland of Don't hardcode MediaType::ANY in FakeNetworkPipe. (patchset #1 id:1 of https://codereview.webrtc.org/2784543002/ )
2017-03-30 06:57:43 +00:00
rtp_transport_controller_send.h
Define RtpTransportControllerSendInterface.
2017-03-27 12:36:15 +00:00
syncable.cc
Stop using VoEVideoSync in Call/VideoReceiveStream.
2017-01-31 11:58:40 +00:00
syncable.h
Stop using VoEVideoSync in Call/VideoReceiveStream.
2017-01-31 11:58:40 +00:00