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webrtc_m130/webrtc/call
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eladalon f184138a5f s/WebRtcVideoChannel2/WebRtcVideoChannel and s/WebRtcVideoEngine2/WebRtcVideoEngine 
WebRtcVideoChannel and and WebRtcVideoEngine seem to have been removed, and only WebRtcVideoChannel2 and WebRtcVideoEngine2 remain, which removes the need for the "2" postfix.

BUG=None

Review-Url: https://codereview.webrtc.org/2932073002
Cr-Commit-Position: refs/heads/master@{#18531}
2017-06-12 08:16:46 +00:00
..
audio_receive_stream.h
Reland of Implemented the GetSources() in native code. (patchset #1 id:1 of https://codereview.webrtc.org/2809613002/ )
2017-04-10 14:39:05 +00:00
audio_send_stream.cc
Injectable audio encoders: WebRtcVoiceEngine and company
2017-04-27 09:08:52 +00:00
audio_send_stream.h
Injectable audio encoders: Moved audio encoder, factory and builtin factory to api/.
2017-05-02 13:46:30 +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
Renaming probing_interval to bwe_period globally.
2017-05-18 21:32:41 +00:00
bitrate_allocator.h
Renaming probing_interval to bwe_period globally.
2017-05-18 21:32:41 +00:00
bitrate_estimator_tests.cc
Making FakeNetworkPipe demux audio and video packets.
2017-04-10 23:57:57 +00:00
BUILD.gn
New targets call:rtp_interfaces, call:rtp_receiver, call:rtp_sender.
2017-06-01 11:02:35 +00:00
call_perf_tests.cc
Delete helper class MediaTypePacketReceiver.
2017-05-09 12:12:00 +00:00
call_unittest.cc
Relanding: Adds PeerConnectionInterface::UpdateCallBitrate to give clients more control of the bandwidth estimator. PeerConnection implements this method by passing a BitrateConfigMask to its associated Call, which is combined with the existing BitrateConfig and passed on to the SendSideCongestionController as necessary. The existing BitrateConfig generally comes from the x-google-{min,start,max}-bitrate params in the SDP.
2017-06-02 21:37:37 +00:00
call.cc
Relanding: Adds PeerConnectionInterface::UpdateCallBitrate to give clients more control of the bandwidth estimator. PeerConnection implements this method by passing a BitrateConfigMask to its associated Call, which is combined with the existing BitrateConfig and passed on to the SendSideCongestionController as necessary. The existing BitrateConfig generally comes from the x-google-{min,start,max}-bitrate params in the SDP.
2017-06-02 21:37:37 +00:00
call.h
Relanding: Adds PeerConnectionInterface::UpdateCallBitrate to give clients more control of the bandwidth estimator. PeerConnection implements this method by passing a BitrateConfigMask to its associated Call, which is combined with the existing BitrateConfig and passed on to the SendSideCongestionController as necessary. The existing BitrateConfig generally comes from the x-google-{min,start,max}-bitrate params in the SDP.
2017-06-02 21:37:37 +00:00
DEPS
Passed AudioMixer to AudioState::Config.
2016-11-17 14:48:56 +00:00
fake_rtp_transport_controller_send.h
Simple tests for Call::SetBitrateConfig.
2017-05-17 18:49:12 +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
New targets call:rtp_interfaces, call:rtp_receiver, call:rtp_sender.
2017-06-01 11:02:35 +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
Making FakeNetworkPipe demux audio and video packets.
2017-04-10 23:57:57 +00:00
rampup_tests.h
Move SelectMediaType from RampUpTester to BaseTest.
2017-04-05 07:33:40 +00:00
rtp_demuxer_unittest.cc
Add RSID-based demuxing to RtpDemuxer
2017-06-08 11:19:13 +00:00
rtp_demuxer.cc
Add RSID-based demuxing to RtpDemuxer
2017-06-08 11:19:13 +00:00
rtp_demuxer.h
Add RSID-based demuxing to RtpDemuxer
2017-06-08 11:19:13 +00:00
rtp_packet_sink_interface.h
New targets call:rtp_interfaces, call:rtp_receiver, call:rtp_sender.
2017-06-01 11:02:35 +00:00
rtp_transport_controller_send_interface.h
s/WebRtcVideoChannel2/WebRtcVideoChannel and s/WebRtcVideoEngine2/WebRtcVideoEngine
2017-06-12 08:16:46 +00:00
rtp_transport_controller_send.cc
Address some violations of chromium-style.
2017-05-31 09:24:52 +00:00
rtp_transport_controller_send.h
Address some violations of chromium-style.
2017-05-31 09:24:52 +00:00
rtx_receive_stream_unittest.cc
New class RtxReceiveStream.
2017-05-19 13:15:19 +00:00
rtx_receive_stream.cc
Address some violations of chromium-style.
2017-05-31 09:24:52 +00:00
rtx_receive_stream.h
New targets call:rtp_interfaces, call:rtp_receiver, call:rtp_sender.
2017-06-01 11:02:35 +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