- 4de6622 Fix a bug in computing audio delay on ios device. Converts seconds to by Jiawei Ou · 10 years ago
- 3449faa Remove GICE (gone forever!) and PORTALLOCATOR_ENABLE_SHARED_UFRAG (enabled forever). by Peter Thatcher · 10 years ago
- 4cee419 Separating voice activity flag from audio level in RtpHeaderExtension. by Minyue · 10 years ago
- c2ee2c8 Refactor the relationship between BaseChannel and MediaChannel so that we send over all the parameters in one method call rather then having them broken up into multiple method calls. This should allow future refactorings of the WebRtcVideoEngine2 to not recreate configurations so many times, and have more simple code as well. by Peter Thatcher · 10 years ago
- eb04d68 Moved project configs to infra/config branch by nodir · 10 years ago
- 25c96d0 Add thread checker to StatsCollection. by jbauch · 10 years ago
- 2328a94 Add average rtt to CallStatsObserver and an average rtt histogram. by stefan · 10 years ago
- 0482dcc Enable HW H.264 decoding on Intel platforms. by Alex Glaznev · 10 years ago
- 8381b37 Removed bjornv from OWNERS and added two new owners by peah · 10 years ago
- 2e1d8bb Suppress a race in libjingle_peerconnection_unittest by henrik.lundin · 10 years ago
- fcf8ece AndroidVideoCapturer: Return frames that have been dropped by magjed · 10 years ago
- c937139 Regenerate bind.h using pump.py BUG=webrtc:4690 R=pthatcher@webrtc.org by Fredrik Solenberg · 10 years ago
- a873644 Move all the examples from the talk directory into the webrtc examples directory. by Donald E Curtis · 10 years ago
- 5b4ce33 DtlsIdentityStoreInterface added. by Henrik Boström · 10 years ago
- 0c02264 Get rid of media_engine_ from BaseChannel; only VoiceChannel needs it. by Fredrik Solenberg · 10 years ago
- bd10ee8 Tiny cleanups. by Fredrik Solenberg · 10 years ago
- 62dae19 Use RtcpPacket to send FIR in RtcpSender by sprang · 10 years ago
- ef7228c Selectable number of TL screenshare loopback test. Also contains some tweaks to make a single TL perform better. by sprang · 10 years ago
- 907dcfd Increase packet limit in jitter buffer. by sprang · 10 years ago
- 37ec733 VideoCapturerAndroid: Check if data is null in onPreviewFrame() by magjed · 10 years ago
- 0c85020 Add list of devices with HW H.264 encoder non suitable for WebRTC. by Alex Glaznev · 10 years ago
- 8d62971 Fix race condition in EndToEndTest.AssignsTransportSequenceNumbers by Erik Språng · 10 years ago
- b19eba3 Fix Turn TCP port issue. by honghaiz · 10 years ago
- 867fb52 Add support for transport wide sequence numbers by sprang · 10 years ago
- d67a219 Switch to base/logging.h in neteq_impl.cc by Henrik Lundin · 10 years ago
- 62cde2c Disabling VP9 perf test by ivica · 10 years ago
- 503726c Fix the generation mismatch assertion error. by honghaiz · 10 years ago
- 72aa9a6 Use RtcpPacket to send PLI in RtcpSender by Erik Språng · 10 years ago
- a9455ab Integration of VP9 packetization. by asapersson · 10 years ago
- 2386a45 Supporting Pause/Resume, Sending Estimate logging. Corrected plot colors by Cesar Magalhaes · 10 years ago
- a12ba55 Added protection for GetCapabilities() failure. by dkirovbroadsoft · 10 years ago
- 5f5f11c FEC protect H264 delta frames as well. by pbos · 10 years ago
- 3641185 Includes webrtc/build/protoc.gypi instead of build/protoc.gypi by Bjorn Terelius · 10 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 · 10 years ago
- 9a6e741 Move audio_coding_module.gypi from main/acm2 to main/. by Peter Boström · 10 years ago
- e2cb1f1 Efficient Metric Recorder by Cesar Magalhaes · 10 years ago
- 028cf48 Added FullStack performance test for screensharing with VP9 by ivica · 10 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 · 10 years ago
- ee66016 Added IsInBeam to mock_nonlinear_beamformer.h by bloch · 10 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 · 10 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 · 10 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 · 10 years ago
- a3b8769 Add packetization and coding/decoding of feedback message format. by Erik Språng · 10 years ago
- f1828e8 Prevent OOB reads for truncated H264 STAP-A packets. by pbos · 10 years ago
- f38ea3c Add support for VP9 packetization/depacketization. by asapersson · 10 years ago
- 95b8718 Fix to "Removing AudioMixerStatusReceiver and ParticipantStatistics" by Minyue Li · 10 years ago
- 4540ffa Removing AudioMixerStatusReceiver and ParticipantStatistics. by Minyue · 10 years ago
- d40af69 Split MoveReadPosition into Forward and Backward versions. by andrew · 10 years ago
- b3cc77f Re-enable WebRtcIsacfix_AllpassFilter2FixDec16Neon by Zhongwei Yao · 10 years ago
- a446609 When we trace to file, add eol of each trace message. by Brave Yao · 10 years ago
- b3b79b6 Clean up the Config to enable 48kHz support in AudioProcessing by aluebs · 10 years ago
- ef35f06 Remove webrtc::Config from ViEChannelGroup. by pbos · 10 years ago
- 081af25 Remove kProtectionKey* and VCMKeyRequestMode. by pbos · 10 years ago
- fa37e33 Add pbos@webrtc.org to webrtc/video_engine/OWNERS. by pbos · 10 years ago
- fe0c905 Improve probing by ignoring small packets which otherwise break the mechanism. by stefan · 10 years ago
- b28678c Add unittest to GlRectDrawer by magjed · 10 years ago
- 013a580 VideoCapturerAndroid: Revert elapsedRealtimeNanos to elapsedRealtime by magjed · 10 years ago
- d55ce2d BWE Simulation Framework: Standard plot logging by Cesar Magalhaes · 10 years ago
- 7a1c24f Remove "multichannel" from parameter to match interface name. by andrew · 10 years ago
- e2b34b7 Bug fix: camera frames are dropped before wideo encoder. by jackychen · 10 years ago
- 6bb1b6e Control combined_audio_video_bwe with config bool. by pbos · 10 years ago
- cfd5f96 Ignore packets with reordered timestamps when doing BWE. by stefan · 10 years ago
- a38233a Removed extended jitter report from RtcpSender. by Erik Språng · 10 years ago
- 6718e97 Add encode and decode time to histograms stats: by asapersson · 10 years ago
- c3f46a9 iOS: Move AppRTC logging methods to public headers. by tkchin · 10 years ago
- 28bae02 Remove CircularFileStream / replace it with CallSessionFileRotatingStream. by tkchin · 10 years ago
- 3ab2f14 Remove C++11 calls from intelligibility_utils by ekmeyerson · 10 years ago
- 86c6d33 Allow more than 2 input channels in AudioProcessing. by Michael Graczyk · 10 years ago
- fcfdb08 Update AUTHORS file. by tkchin · 10 years ago
- d6fc47e Remove base channel for video receivers. by pbos · 10 years ago
- 59adf34 Evaluation test cases. by Cesar Magalhaes · 10 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 · 10 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 · 10 years ago
- b21fd94 Temporarily disable ScreenshareSlides on Android. by Peter Boström · 10 years ago
- c204754 Allow more than 2 input channels in AudioProcessing. by Michael Graczyk · 10 years ago
- 0b6a204 Configure AudioProcessing directly in agc_harness. by andrew · 10 years ago
- b297c5a Miscellaneous changes split from https://codereview.webrtc.org/1230503003 . by pkasting · 10 years ago
- 7c5304c Allow webrtc compilation with stlport by Jared Duke · 10 years ago
- 9341191 Provides log sinks for rotating logs. Intended for use on mobile devices to record call logs. by tkchin · 10 years ago
- f24b2bc Modified histogram shell plot script, added python dynamics plot script by Cesar Magalhaes · 10 years ago
- 235c35f Implement store as an explicit atomic operation. by pbos · 10 years ago
- 085856c Extend full stack tests with more stats by Erik Språng · 10 years ago
- d89920b Add resolution and fps stats to histograms: by asapersson · 10 years ago
- 65eb1c3 Disable testcase NatTcpTest.TestConnectOut by magjed · 10 years ago
- d60a799 Mark WebRTC project as public in luci-config by sergiyb · 10 years ago
- b69ab79 VideoCapturerAndroid: Add function to change capture format while camera is running by magjed · 10 years ago
- 496019c If the array size is even, the median should be average of its two middlemost elements. by Cesar Magalhaes · 10 years ago
- 83d6b0c2 Ignore genperf lib in merge_libs.py. by noahric · 10 years ago
- 343714e Fix the problom that on Linux no external audio device can be selected since #9243. by Brave Yao · 10 years ago
- 2981945 Moved arrray_util include to beamformer.h by bloch · 10 years ago
- 8ff04d6 Remove UpdateSsrcs from EncoderStateFeedback. by pbos · 10 years ago
- 324d9c9 Avoids error message about unknown selected data source for Port iPhone Microphone by henrika · 10 years ago
- f421bdc Fix an NPE when creating TurnPort with a NULL socket. by honghaiz · 10 years ago
- be37888 Fixing scenario where track is rejected and later un-rejected. by deadbeef · 10 years ago
- b947f28 Add pcap support to bwe tools. Allow filtering on SSRCs. by stefan · 10 years ago
- fabe2c9 Remove deprecated functions. by jbauch · 10 years ago
- c27d89f Let WebRtcVideoChannel2::WebRtcVideoSendStream::InputFrame carry the input frame's timestamp to output frame. by qiangchen · 10 years ago
- c5d0d95 Ensuring that UDP TURN servers are always used as STUN servers. by deadbeef · 10 years ago
- d848d5e Enable cropping window capturing for Win7 when Aero is disabled. by Jiayang Liu · 10 years ago
- bd38428 Don't use result of "field_trial::FindFullName" as string reference. by jbauch · 10 years ago
