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webrtc_m130/webrtc/call
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nisse 76e62b0d38 Address some violations of chromium-style. 
BUG=webrtc:163

Review-Url: https://codereview.webrtc.org/2913793002
Cr-Commit-Position: refs/heads/master@{#18345}
2017-05-31 09:24:52 +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
Store/restore RTP state for audio streams with same SSRC within a call
2017-05-23 13:07:11 +00:00
call_perf_tests.cc
Delete helper class MediaTypePacketReceiver.
2017-05-09 12:12:00 +00:00
call_unittest.cc
Store/restore RTP state for audio streams with same SSRC within a call
2017-05-23 13:07:11 +00:00
call.cc
This cl removes RtcEventLog deps to call:call_interfaces. The purpose is to be able to use the event log from the upcoming RtpTransport.
2017-05-30 10:52:10 +00:00
call.h
This will allow me to test that Call invokes SendSideCongestionController::SetBweBitrates as expected (for https://codereview.chromium.org/2793913008).
2017-05-08 18:52:38 +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 class RtpDemuxer and RtpPacketSinkInterface, use in Call.
2017-05-16 11:47:04 +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.cc
New class RtpDemuxer and RtpPacketSinkInterface, use in Call.
2017-05-16 11:47:04 +00:00
rtp_demuxer.h
New class RtpDemuxer and RtpPacketSinkInterface, use in Call.
2017-05-16 11:47:04 +00:00
rtp_transport_controller_send_interface.h
Move RtpTransportControllerSend to a new file.
2017-04-24 12:53:20 +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
Address some violations of chromium-style.
2017-05-31 09:24:52 +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