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webrtc_m130/webrtc/call
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eladalon 9addbebf42 Remove RtpDemuxer tweak for preventing multiple RSID inspections 
We have a tweak preventing multiple deep-examinations of packets; packets with a given SSRC are only inspected deeply (RSID) once (only the first received packet). Once we move to many-to-one stream-to-sink associations, this becomes less useful, and is better removed.

BUG=webrtc:7135

Review-Url: https://codereview.webrtc.org/2955373002
Cr-Commit-Position: refs/heads/master@{#18859}
2017-06-30 13:26:54 +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
Allow an external audio processing module to be used in WebRTC
2017-06-29 15:32:09 +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
Reland of Create RtcpDemuxer (patchset #1 id:1 of https://codereview.webrtc.org/2957763002/ )
2017-06-26 18:23:54 +00:00
call_perf_tests.cc
Allow an external audio processing module to be used in WebRTC
2017-06-29 15:32:09 +00:00
call_unittest.cc
Allow an external audio processing module to be used in WebRTC
2017-06-29 15:32:09 +00:00
call.cc
Introduce RtpStreamReceiverInterface and RtpStreamReceiverControllerInterface.
2017-06-21 08:05:22 +00:00
call.h
Support building WebRTC without audio and video.
2017-06-15 19:52:32 +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
Simple tests for Call::SetBitrateConfig.
2017-05-17 18:49:12 +00:00
flexfec_receive_stream_impl.cc
Introduce RtpStreamReceiverInterface and RtpStreamReceiverControllerInterface.
2017-06-21 08:05:22 +00:00
flexfec_receive_stream_impl.h
Introduce RtpStreamReceiverInterface and RtpStreamReceiverControllerInterface.
2017-06-21 08:05:22 +00:00
flexfec_receive_stream_unittest.cc
Introduce RtpStreamReceiverInterface and RtpStreamReceiverControllerInterface.
2017-06-21 08:05:22 +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
Reland of Periodically update codec bit/frame rate settings.
2017-06-27 14:06:52 +00:00
rampup_tests.h
Move SelectMediaType from RampUpTester to BaseTest.
2017-04-05 07:33:40 +00:00
rsid_resolution_observer.h
Reland of Create RtcpDemuxer (patchset #1 id:1 of https://codereview.webrtc.org/2957763002/ )
2017-06-26 18:23:54 +00:00
rtcp_demuxer_unittest.cc
Reland of Create RtcpDemuxer (patchset #1 id:1 of https://codereview.webrtc.org/2957763002/ )
2017-06-26 18:23:54 +00:00
rtcp_demuxer.cc
Reland of Create RtcpDemuxer (patchset #1 id:1 of https://codereview.webrtc.org/2957763002/ )
2017-06-26 18:23:54 +00:00
rtcp_demuxer.h
Reland of Create RtcpDemuxer (patchset #1 id:1 of https://codereview.webrtc.org/2957763002/ )
2017-06-26 18:23:54 +00:00
rtcp_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_demuxer_unittest.cc
Remove RtpDemuxer tweak for preventing multiple RSID inspections
2017-06-30 13:26:54 +00:00
rtp_demuxer.cc
Remove RtpDemuxer tweak for preventing multiple RSID inspections
2017-06-30 13:26:54 +00:00
rtp_demuxer.h
Remove RtpDemuxer tweak for preventing multiple RSID inspections
2017-06-30 13:26:54 +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
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.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 Create RtcpDemuxer (patchset #1 id:1 of https://codereview.webrtc.org/2957763002/ )
2017-06-26 18:23:54 +00:00
rtp_stream_receiver_controller_interface.h
Introduce RtpStreamReceiverInterface and RtpStreamReceiverControllerInterface.
2017-06-21 08:05:22 +00:00
rtp_stream_receiver_controller.cc
Introduce RtpStreamReceiverInterface and RtpStreamReceiverControllerInterface.
2017-06-21 08:05:22 +00:00
rtp_stream_receiver_controller.h
Introduce RtpStreamReceiverInterface and RtpStreamReceiverControllerInterface.
2017-06-21 08:05:22 +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
Use rtp_header_extension_map.h instead of rtp_header_extension.h
2017-06-12 14:01:20 +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