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webrtc_m130/webrtc/call
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asapersson fc5e81c979 Replace first_packet_sent_ms_ in Call. 
Instead of using the time on the first callback to Call::OnSentPacket, use the time when the first packet is sent from the pacer (to make sure this packet corresponds to an audio/video RTP packet).

BUG=webrtc:6244

Review-Url: https://codereview.webrtc.org/2825333002
Cr-Commit-Position: refs/heads/master@{#17777}
2017-04-20 06:28:53 +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
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
Making FakeNetworkPipe demux audio and video packets.
2017-04-10 23:57:57 +00:00
BUILD.gn
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
call_perf_tests.cc
Making FakeNetworkPipe demux audio and video packets.
2017-04-10 23:57:57 +00:00
call_unittest.cc
Move AudioDecoder and AudioDecoderFactory mocks to webrtc/test/
2017-04-10 12:15:48 +00:00
call.cc
Replace first_packet_sent_ms_ in Call.
2017-04-20 06:28:53 +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
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_transport_controller_send.h
Delete VieRemb class, move functionality to PacketRouter.
2017-04-19 06:38:35 +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