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webrtc_m130/webrtc/call
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eladalon d9827b554e Removed an unused variable from CallPerfTest::TestAudioVideoSync() 
fake_decoder (of type test::FakeDecoder) was not used.

BUG=None

Review-Url: https://codereview.webrtc.org/3001543002
Cr-Commit-Position: refs/heads/master@{#19322}
2017-08-11 09:08:33 +00:00
..
test
Only one implementation of MockRtpPacketSink once
2017-07-28 17:05:45 +00:00
audio_receive_stream.h
Adding stats that can be used to compute output audio levels as described here https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudioenergy.
2017-07-14 19:17:49 +00:00
audio_send_stream.cc
Injectable audio encoders: WebRtcVoiceEngine and company
2017-04-27 09:08:52 +00:00
audio_send_stream.h
Make ~webrtc::AudioSendStream public, and s/config()/GetConfig(), as well as make public.
2017-07-26 09:09:44 +00:00
audio_state.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +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
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
bitrate_allocator.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
bitrate_estimator_tests.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
BUILD.gn
Wire up RTP keep-alive in ortc api.
2017-08-09 13:42:32 +00:00
call_perf_tests.cc
Removed an unused variable from CallPerfTest::TestAudioVideoSync()
2017-08-11 09:08:33 +00:00
call_unittest.cc
Wire up RTP keep-alive in ortc api.
2017-08-09 13:42:32 +00:00
call.cc
Add functionality which limits the number of bytes on the network.
2017-08-09 14:17:33 +00:00
call.h
Wire up RTP keep-alive in ortc api.
2017-08-09 13:42:32 +00:00
callfactory.cc
Support building WebRTC without audio and video.
2017-06-15 19:52:32 +00:00
callfactory.h
Support building WebRTC without audio and video.
2017-06-15 19:52:32 +00:00
callfactoryinterface.h
Support building WebRTC without audio and video.
2017-06-15 19:52:32 +00:00
DEPS
Passed AudioMixer to AudioState::Config.
2016-11-17 14:48:56 +00:00
fake_rtp_transport_controller_send.h
Wire up RTP keep-alive in ortc api.
2017-08-09 13:42:32 +00:00
flexfec_receive_stream_impl.cc
Protected streams report RTP messages directly to the FlexFec streams
2017-08-02 14:39:07 +00:00
flexfec_receive_stream_impl.h
Protected streams report RTP messages directly to the FlexFec streams
2017-08-02 14:39:07 +00:00
flexfec_receive_stream_unittest.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
flexfec_receive_stream.h
Protected streams report RTP messages directly to the FlexFec streams
2017-08-02 14:39:07 +00:00
OWNERS
Remove all references to GYP
2016-11-16 19:11:38 +00:00
rampup_tests.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
rampup_tests.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
rsid_resolution_observer.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
rtcp_demuxer_unittest.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
rtcp_demuxer.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
rtcp_demuxer.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
rtcp_packet_sink_interface.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
rtp_demuxer_unittest.cc
Reland of SSRC and RSID may only refer to one sink each in RtpDemuxer (patchset #1 id:1 of https://codereview.webrtc.org/2993633002/ )
2017-08-04 13:34:54 +00:00
rtp_demuxer.cc
Reland of SSRC and RSID may only refer to one sink each in RtpDemuxer (patchset #1 id:1 of https://codereview.webrtc.org/2993633002/ )
2017-08-04 13:34:54 +00:00
rtp_demuxer.h
Reland of SSRC and RSID may only refer to one sink each in RtpDemuxer (patchset #1 id:1 of https://codereview.webrtc.org/2993633002/ )
2017-08-04 13:34:54 +00:00
rtp_packet_sink_interface.h
Reland of Create RtcpDemuxer (patchset #1 id:1 of https://codereview.webrtc.org/2957763002/ )
2017-06-26 18:23:54 +00:00
rtp_rtcp_demuxer_helper_unittest.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
rtp_rtcp_demuxer_helper.cc
Reland of Create RtcpDemuxer (patchset #1 id:1 of https://codereview.webrtc.org/2957763002/ )
2017-06-26 18:23:54 +00:00
rtp_rtcp_demuxer_helper.h
Reland of SSRC and RSID may only refer to one sink each in RtpDemuxer (patchset #1 id:1 of https://codereview.webrtc.org/2993633002/ )
2017-08-04 13:34:54 +00:00
rtp_stream_receiver_controller_interface.h
Reland of SSRC and RSID may only refer to one sink each in RtpDemuxer (patchset #1 id:1 of https://codereview.webrtc.org/2993633002/ )
2017-08-04 13:34:54 +00:00
rtp_stream_receiver_controller.cc
Reland of SSRC and RSID may only refer to one sink each in RtpDemuxer (patchset #1 id:1 of https://codereview.webrtc.org/2993633002/ )
2017-08-04 13:34:54 +00:00
rtp_stream_receiver_controller.h
Reland of SSRC and RSID may only refer to one sink each in RtpDemuxer (patchset #1 id:1 of https://codereview.webrtc.org/2993633002/ )
2017-08-04 13:34:54 +00:00
rtp_transport_controller_send_interface.h
Wire up RTP keep-alive in ortc api.
2017-08-09 13:42:32 +00:00
rtp_transport_controller_send.cc
Wire up RTP keep-alive in ortc api.
2017-08-09 13:42:32 +00:00
rtp_transport_controller_send.h
Wire up RTP keep-alive in ortc api.
2017-08-09 13:42:32 +00:00
rtx_receive_stream_unittest.cc
Only one implementation of MockRtpPacketSink once
2017-07-28 17:05:45 +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
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00