- 72aa9a6 Use RtcpPacket to send PLI in RtcpSender by Erik Språng · 9 years ago
- a9455ab Integration of VP9 packetization. by asapersson · 9 years ago
- 2386a45 Supporting Pause/Resume, Sending Estimate logging. Corrected plot colors by Cesar Magalhaes · 9 years ago
- a12ba55 Added protection for GetCapabilities() failure. by dkirovbroadsoft · 9 years ago
- 5f5f11c FEC protect H264 delta frames as well. by pbos · 9 years ago
- 3641185 Includes webrtc/build/protoc.gypi instead of build/protoc.gypi by Bjorn Terelius · 9 years ago
- b933667 Revert "Renamed the ACMDump to RtcEventLog and moved it to webrtc/video, since it is not specific to the audio coding module. Updated .gyp and .gn files accordingly." by Bjorn Terelius · 9 years ago
- 9a6e741 Move audio_coding_module.gypi from main/acm2 to main/. by Peter Boström · 9 years ago
- e2cb1f1 Efficient Metric Recorder by Cesar Magalhaes · 9 years ago
- 028cf48 Added FullStack performance test for screensharing with VP9 by ivica · 9 years ago
- c159b04 Renamed the ACMDump to RtcEventLog and moved it to webrtc/video, since it is not specific to the audio coding module. Updated .gyp and .gn files accordingly. by Bjorn Terelius · 9 years ago
- ee66016 Added IsInBeam to mock_nonlinear_beamformer.h by bloch · 9 years ago
- d635895 Add a frame generator that allows scrolling over a larger still image, for use with new screen sharing quality tests. by sprang · 9 years ago
- 49c0ce3 Revert "Add a frame generator that allows scrolling over a larger still image, for use with new screen sharing quality tests." by Erik Språng · 9 years ago
- 8993413 Add a frame generator that allows scrolling over a larger still image, for use with new screen sharing quality tests. by Erik Språng · 9 years ago
- a3b8769 Add packetization and coding/decoding of feedback message format. by Erik Språng · 9 years ago
- f1828e8 Prevent OOB reads for truncated H264 STAP-A packets. by pbos · 9 years ago
- f38ea3c Add support for VP9 packetization/depacketization. by asapersson · 9 years ago
- 95b8718 Fix to "Removing AudioMixerStatusReceiver and ParticipantStatistics" by Minyue Li · 9 years ago
- 4540ffa Removing AudioMixerStatusReceiver and ParticipantStatistics. by Minyue · 9 years ago
- d40af69 Split MoveReadPosition into Forward and Backward versions. by andrew · 9 years ago
- b3cc77f Re-enable WebRtcIsacfix_AllpassFilter2FixDec16Neon by Zhongwei Yao · 9 years ago
- a446609 When we trace to file, add eol of each trace message. by Brave Yao · 9 years ago
- b3b79b6 Clean up the Config to enable 48kHz support in AudioProcessing by aluebs · 9 years ago
- ef35f06 Remove webrtc::Config from ViEChannelGroup. by pbos · 9 years ago
- 081af25 Remove kProtectionKey* and VCMKeyRequestMode. by pbos · 9 years ago
- fa37e33 Add pbos@webrtc.org to webrtc/video_engine/OWNERS. by pbos · 9 years ago
- fe0c905 Improve probing by ignoring small packets which otherwise break the mechanism. by stefan · 9 years ago
- b28678c Add unittest to GlRectDrawer by magjed · 9 years ago
- 013a580 VideoCapturerAndroid: Revert elapsedRealtimeNanos to elapsedRealtime by magjed · 9 years ago
- d55ce2d BWE Simulation Framework: Standard plot logging by Cesar Magalhaes · 9 years ago
- 7a1c24f Remove "multichannel" from parameter to match interface name. by andrew · 9 years ago
- e2b34b7 Bug fix: camera frames are dropped before wideo encoder. by jackychen · 9 years ago
- 6bb1b6e Control combined_audio_video_bwe with config bool. by pbos · 9 years ago
- cfd5f96 Ignore packets with reordered timestamps when doing BWE. by stefan · 9 years ago
- a38233a Removed extended jitter report from RtcpSender. by Erik Språng · 9 years ago
- 6718e97 Add encode and decode time to histograms stats: by asapersson · 9 years ago
- c3f46a9 iOS: Move AppRTC logging methods to public headers. by tkchin · 9 years ago
- 28bae02 Remove CircularFileStream / replace it with CallSessionFileRotatingStream. by tkchin · 9 years ago
- 3ab2f14 Remove C++11 calls from intelligibility_utils by ekmeyerson · 9 years ago
- 86c6d33 Allow more than 2 input channels in AudioProcessing. by Michael Graczyk · 9 years ago
- fcfdb08 Update AUTHORS file. by tkchin · 9 years ago
- d6fc47e Remove base channel for video receivers. by pbos · 9 years ago
- 59adf34 Evaluation test cases. by Cesar Magalhaes · 9 years ago
- 66f438f Revert of Fixing scenario where track is rejected and later un-rejected. (patchset #5 id:80001 of https://codereview.webrtc.org/1231613002/) by magjed · 9 years ago
- 64e753c Revert of Allow more than 2 input channels in AudioProcessing. (patchset #13 id:240001 of https://codereview.webrtc.org/1226093007/) by magjed · 9 years ago
- b21fd94 Temporarily disable ScreenshareSlides on Android. by Peter Boström · 9 years ago
- c204754 Allow more than 2 input channels in AudioProcessing. by Michael Graczyk · 9 years ago
- 0b6a204 Configure AudioProcessing directly in agc_harness. by andrew · 9 years ago
- b297c5a Miscellaneous changes split from https://codereview.webrtc.org/1230503003 . by pkasting · 9 years ago
- 7c5304c Allow webrtc compilation with stlport by Jared Duke · 9 years ago
- 9341191 Provides log sinks for rotating logs. Intended for use on mobile devices to record call logs. by tkchin · 9 years ago
- f24b2bc Modified histogram shell plot script, added python dynamics plot script by Cesar Magalhaes · 9 years ago
- 235c35f Implement store as an explicit atomic operation. by pbos · 9 years ago
- 085856c Extend full stack tests with more stats by Erik Språng · 9 years ago
- d89920b Add resolution and fps stats to histograms: by asapersson · 9 years ago
- 65eb1c3 Disable testcase NatTcpTest.TestConnectOut by magjed · 9 years ago
- d60a799 Mark WebRTC project as public in luci-config by sergiyb · 9 years ago
- b69ab79 VideoCapturerAndroid: Add function to change capture format while camera is running by magjed · 9 years ago
- 496019c If the array size is even, the median should be average of its two middlemost elements. by Cesar Magalhaes · 9 years ago
- 83d6b0c2 Ignore genperf lib in merge_libs.py. by noahric · 9 years ago
- 343714e Fix the problom that on Linux no external audio device can be selected since #9243. by Brave Yao · 9 years ago
- 2981945 Moved arrray_util include to beamformer.h by bloch · 9 years ago
- 8ff04d6 Remove UpdateSsrcs from EncoderStateFeedback. by pbos · 9 years ago
- 324d9c9 Avoids error message about unknown selected data source for Port iPhone Microphone by henrika · 9 years ago
- f421bdc Fix an NPE when creating TurnPort with a NULL socket. by honghaiz · 9 years ago
- be37888 Fixing scenario where track is rejected and later un-rejected. by deadbeef · 9 years ago
- b947f28 Add pcap support to bwe tools. Allow filtering on SSRCs. by stefan · 9 years ago
- fabe2c9 Remove deprecated functions. by jbauch · 9 years ago
- c27d89f Let WebRtcVideoChannel2::WebRtcVideoSendStream::InputFrame carry the input frame's timestamp to output frame. by qiangchen · 9 years ago
- c5d0d95 Ensuring that UDP TURN servers are always used as STUN servers. by deadbeef · 9 years ago
- d848d5e Enable cropping window capturing for Win7 when Aero is disabled. by Jiayang Liu · 9 years ago
- bd38428 Don't use result of "field_trial::FindFullName" as string reference. by jbauch · 9 years ago
- a9b4c32 Nuke buffered latency mode. It's not actually working, and it's not used. It's just dead code complexity. by Peter Thatcher · 9 years ago
- 083b73f Use std::string references instead of copying contents. by jbauch · 9 years ago
- cd67022 Define Stream base classes by Jelena Marusic · 9 years ago
- cddb367 Remove unused metric in overuse detector. by Asa Persson · 9 years ago
- f393829 Use "UDP/TLS/RTP/SAVPF" profile in offer when DTLS-SRTP is used. by deadbeef · 9 years ago
- fb19f49 Replaced uint32_t with standard uint16_t for sequence_number variables. by Cesar Magalhaes · 9 years ago
- bf40b42 Modified Simulation Framework Jitter Model. by Cesar Magalhaes · 9 years ago
- 8fc7fa7 Base A/V synchronization on sync_labels. by pbos · 9 years ago
- 9c261f2 Supports logging for dynamic and histogram plots on Simulation Framework. by Cesar Magalhaes · 9 years ago
- a4a8d4a Base padding bitrate for an encoder on the bitrate allocated for that encoder, rather than the total bitrate of the channel group. by stefan · 9 years ago
- 3258db2 Split iSAC encoder/decoder: Test more cases (and make sure they work) by kwiberg · 9 years ago
- 2d3b7e2 AppRTCDemo file logging. by Zeke Chin · 9 years ago
- 43e7d3b Avoid overflow in checking for emulation bytes in rbsp. by noahric · 9 years ago
- ba8c15b Merge methods for configuring NACK/FEC/hybrid. by pbos · 9 years ago
- caa498a Make sure RTCP is sent in tests when receiving packets even if REMB is delayed. by stefan · 9 years ago
- ba35d05 Cleanup of iOS AudioDevice implementation by henrika · 9 years ago
- d6f1a38 Remove ViEChannel simulcast lock. by Peter Boström · 9 years ago
- 4988ca5 Removed unused variables and the need to include the d3dx9.h file. by dkirovbroadsoft · 9 years ago
- 870eee4 Fix simulator issue where chokes didn't apply to non-congested packets. by stefan · 9 years ago
- a03cd3f 1. Override and virtual has to be consistent. by honghaiz · 9 years ago
- 6e2ce6e Allow for framerate reduction for HW encoder. by jackychen · 9 years ago
- 9009962 Add methods to set the ICE connection receiving_timeout values. by honghaiz · 9 years ago
- 45d1fde Revert of Fix simulator issue where chokes didn't apply to non-congested packets. (patchset #2 id:20001 of https://codereview.webrtc.org/1233853002/) by stefan · 9 years ago
- 662ae00 Fix simulator issue where chokes didn't apply to non-congested packets. by Stefan Holmer · 9 years ago
- 5d6e58e Improvements to rtc::Bind by Jelena Marusic · 9 years ago
- 30409b4 Add statistics gathering for packet loss. by bcornell · 9 years ago
- 35b72fb Add new variance update option and unittests for intelligibility by ekm · 9 years ago