- eb90e6f Merge SendTask implementation for SingleThreadedTaskQueueForTesting and TaskQueueForTest by Danil Chapovalov · 4 years, 6 months ago
- 35214fc Add missing RTC_EXPORT for the component build. by Mirko Bonadei · 4 years, 6 months ago
- ef0627f Revert "Fix GetStats bytesSent/Received, wireup headerBytesSent/Received" by Mirko Bonadei · 4 years, 6 months ago
- 55c7694 Roll chromium_revision 18d4117247..88a7a88286 (705754:705863) by chromium-webrtc-autoroll · 4 years, 6 months ago
- 36d171b Add Ramprakash Jelari to AUTHORS. by Sami Kalliomäki · 4 years, 6 months ago
- 17608dc RtpRtcp modules and below: Make media, RTX and FEC SSRCs const by Erik Språng · 4 years, 6 months ago
- 2f28370 Move --resources_dir to its right place. by Patrik Höglund · 4 years, 6 months ago
- 3f0d8e4 Revert "Delete methods EncodedImage::Allocate and EncodedImageBufferInterface::Realloc" by Mirko Bonadei · 4 years, 6 months ago
- c122c29 Roll chromium_revision 2e1ac8de05..18d4117247 (705654:705754) by chromium-webrtc-autoroll · 4 years, 6 months ago
- d942282 Roll chromium_revision 02833e653c..2e1ac8de05 (705539:705654) by chromium-webrtc-autoroll · 4 years, 6 months ago
- f8998cf Add a turn port prune policy to keep the first ready turn port. by Honghai Zhang · 4 years, 6 months ago
- ef98ae6 Use GlobalLock to protect logging by Danil Chapovalov · 4 years, 6 months ago
- 65c57ff Adds logging of NetworkStateEstimator estimates. by Sebastian Jansson · 4 years, 6 months ago
- c6404a1 Add field trial to reduce STUN pings. by Jonas Oreland · 4 years, 6 months ago
- b259b0a Roll chromium_revision c1f96a7b93..02833e653c (705365:705539) by chromium-webrtc-autoroll · 4 years, 6 months ago
- 24c678f Adds test for loss based controller under cross traffic induced loss. by Sebastian Jansson · 4 years, 6 months ago
- 4af7882 Add feature to skip RELAY to non-RELAY connections by Jonas Oreland · 4 years, 6 months ago
- 0deef72 Remove deprecated functions in RTPSenderVideo by Danil Chapovalov · 4 years, 6 months ago
- fbde32e Fix GetStats bytesSent/Received, wireup headerBytesSent/Received by Niels Möller · 4 years, 6 months ago
- 41c650b Use bitrate limits provided by encoder. by Sergey Silkin · 4 years, 6 months ago
- 5ab79e6 Reland "Implement rollback for setRemoteDescription" by Eldar Rello · 4 years, 6 months ago
- 75acef3 Reject invalid spatial index by Kuang-che Wu · 4 years, 6 months ago
- d6bb184 Delete methods EncodedImage::Allocate and EncodedImageBufferInterface::Realloc by Niels Möller · 4 years, 6 months ago
- 8bbf9e2 Roll chromium_revision 002d8b5c6a..c1f96a7b93 (705236:705365) by chromium-webrtc-autoroll · 4 years, 6 months ago
- 8be669f AEC3: Add support for multiple channels to the reverb modelling by Per Åhgren · 4 years, 6 months ago
- 373b149 Roll chromium_revision da0e48ef9f..002d8b5c6a (705127:705236) by chromium-webrtc-autoroll · 4 years, 6 months ago
- 6787f23 Remove AudioProcessing::level_estimator() getter by saza · 4 years, 6 months ago
- c67a4d6 Fix WebRTC-Video-MinVideoBitrate for VP9 by Elad Alon · 4 years, 6 months ago
- db3d81f Roll chromium_revision 3d7980bda8..da0e48ef9f (705004:705127) by chromium-webrtc-autoroll · 4 years, 6 months ago
- d8aff21 Adds support for stopping fake TCP cross traffic. by Sebastian Jansson · 4 years, 6 months ago
- 80f53b7 Extend WebRTC-Video-MinVideoBitrate to experiment per-codec by Elad Alon · 4 years, 6 months ago
- e62a588 Merging TransportFeedbackAdapter and SendTimeHistory. by Sebastian Jansson · 4 years, 6 months ago
- c69c1bb Plot delay feedback in RTCP arrival order. by Björn Terelius · 4 years, 6 months ago
- 5740f3e Clarify expectation on GlobalLock by Danil Chapovalov · 4 years, 6 months ago
- 3c918b1 Fix bypass of unnecessary resampling by Gustaf Ullberg · 4 years, 6 months ago
- 51bf200 Reduce number of RTPVideoSender::SendVideo parameters by Danil Chapovalov · 4 years, 6 months ago
- 4b64411 NetEqImpl::GetDecoderFormat: Return RTP clockrate, not codec sample rate by Karl Wiberg · 4 years, 6 months ago
- 3b819f3 Move video_sources_.clear() call to CallTest::DestroyStreams by Niels Möller · 4 years, 6 months ago
- 7c3b100 Roll chromium_revision 3fcb948181..3d7980bda8 (704895:705004) by chromium-webrtc-autoroll · 4 years, 6 months ago
- e6f9bd0 Roll chromium_revision d66030f8c3..3fcb948181 (704779:704895) by chromium-webrtc-autoroll · 4 years, 6 months ago
- 3273b5e Roll chromium_revision a1c9c88904..d66030f8c3 (704650:704779) by chromium-webrtc-autoroll · 4 years, 6 months ago
- d62ac3f Use fake clock for replay fuzzing by Kuang-che Wu · 4 years, 6 months ago
- d0704ce Remove RTCP tests from channel_unittest. by Bjorn A Mellem · 4 years, 6 months ago
- ee153c9 Send rtcp::RemoteEstimate and rtcp::TransportFeedback in one packet by Per Kjellander · 4 years, 6 months ago
- 9e70f36 Roll chromium_revision 651f5a2987..a1c9c88904 (704530:704650) by chromium-webrtc-autoroll · 4 years, 6 months ago
- f17976d Use single thread vp9 decoder for fuzzing by Kuang-che Wu · 4 years, 6 months ago
- 45eb135 Remove the unused `receive_timestamp` arg to NetEq::InsertPacket by Karl Wiberg · 4 years, 6 months ago
- c466f08 Cap vp9 fuzzer frame size to prevent OOM by Kuang-che Wu · 4 years, 6 months ago
- cd0eedb Don't allocate audio if we have no transport sequence number. by Sebastian Jansson · 4 years, 6 months ago
- 9afdddf Enable capturing from camera in PC framework by Artem Titov · 4 years, 6 months ago
- 1699981 Add void::RtcpFeedbackSenderInterface::SendCombinedRtcpPacket by Per Kjellander · 4 years, 6 months ago
- 03f4b36 Roll chromium_revision d9b4f45e42..651f5a2987 (704251:704530) by chromium-webrtc-autoroll · 4 years, 6 months ago
- cbbfd08 Replace virtual RtcpPacket::SetSenderSsrc with base member by Danil Chapovalov · 4 years, 6 months ago
- 907f154 Revert "Implement rollback for setRemoteDescription" by Alex Loiko · 4 years, 6 months ago
- 28214cd Fix handling of large packets in RtxReceiveStream by Niels Möller · 4 years, 6 months ago
- 8675eee Bypass unnecessary resampling. by Gustaf Ullberg · 4 years, 6 months ago
- ba700de Add missing dependencies to the static library. by Mirko Bonadei · 4 years, 6 months ago
- 066c2ab Roll chromium_revision 8e1616e4fc..d9b4f45e42 (704145:704251) by chromium-webrtc-autoroll · 4 years, 6 months ago
- 16d4c4d Implement rollback for setRemoteDescription by Eldar Rello · 4 years, 6 months ago
- 5963c7c Count disabled due to low bw streams or layers as bw limited quality in GetStats by Ilya Nikolaevskiy · 4 years, 6 months ago
- 955f8fd Add virtual method rtcp::RtcpPacket::SetSenderSsrc by Per Kjellander · 4 years, 6 months ago
- 6f41f8e Roll chromium_revision b2d00427a6..8e1616e4fc (703937:704145) by chromium-webrtc-autoroll · 4 years, 6 months ago
- f3f03e2 Removing outdated tests. by Alex Loiko · 4 years, 6 months ago
- f980725 AEC3: Send the spectral power estimates for all channels to AecState by Per Åhgren · 4 years, 6 months ago
- d9755ee Delete large up-front allocation in LibvpxVp8Encoder::InitEncode by Niels Möller · 4 years, 6 months ago
- 422b9e0 Run fullband processing at output rate on ARM by Gustaf Ullberg · 4 years, 6 months ago
- 1d3008b AEC3: Remove redundant class by Per Åhgren · 4 years, 6 months ago
- 9ddd729 Add Duration field to EventRateCounter by Evan Shrubsole · 4 years, 6 months ago
- 0169a3e Delete AecState::EchoPathGain() by Sam Zackrisson · 4 years, 6 months ago
- e1092c0 Roll chromium_revision a78cc9b4cc..b2d00427a6 (703818:703937) by chromium-webrtc-autoroll · 4 years, 6 months ago
- 6e9395c Roll chromium_revision baa7b58596..a78cc9b4cc (703669:703818) by chromium-webrtc-autoroll · 4 years, 6 months ago
- f77b939 Makes render time > decode time in VideoFrameMatcher. by Sebastian Jansson · 4 years, 6 months ago
- 46b0140 Update filter analyzer for multi channel by Sam Zackrisson · 4 years, 6 months ago
- 43bd760 Fix build errors of RTCAudioDeviceTests by Byoungchan Lee · 4 years, 6 months ago
- cfe5e2a Stop using goma for MSVC bots. by Mirko Bonadei · 4 years, 6 months ago
- fa77ba6 SetStreams API of RtpSender wrapped for iOS and Android by Cyril Lashkevich · 4 years, 6 months ago
- 999afa9 Fix cropping in H264 decoder wrapper. by Sergey Silkin · 4 years, 6 months ago
- 7f9a0f3 Roll chromium_revision 977e732442..baa7b58596 (703537:703669) by chromium-webrtc-autoroll · 4 years, 6 months ago
- d46d1e9 Add #COMPONENT to WebRTC. by Patrik Höglund · 4 years, 6 months ago
- e93b1fe Improve bitstream dumping logic to handle multiple SLs correctly by Ilya Nikolaevskiy · 4 years, 6 months ago
- b4161d3 AEC3: Add multichannel support to the residual echo estimator by Per Åhgren · 4 years, 6 months ago
- 7e6abf0 Roll chromium_revision 5ac2340a23..977e732442 (703358:703537) by chromium-webrtc-autoroll · 4 years, 6 months ago
- ff27da5 Add/remove receive streams with SSRC 0 from media channels by Saurav Das · 4 years, 6 months ago
- a639f7a Roll chromium_revision 10156469d6..5ac2340a23 (703248:703358) by chromium-webrtc-autoroll · 4 years, 6 months ago
- 7c06777 Cleanup includes in modules/include/module_common_types.h by Danil Chapovalov · 4 years, 6 months ago
- 0824c6f Delete voice_detection() pointer to submodule by Sam Zackrisson · 4 years, 6 months ago
- 24d251f Add 100 ms network delay to the SupportsFlexFEC* tests. by Björn Terelius · 4 years, 6 months ago
- 0a6510d Removes rtp_transport checks in AudioSendStream by Sebastian Jansson · 4 years, 6 months ago
- 99a2096 Added support for skipping get_audio events, adding dummy packets and setting a field trial string. by Ivo Creusen · 4 years, 6 months ago
- 35cf9e7 Replaces static modifier functions in AudioSendStream. by Sebastian Jansson · 4 years, 6 months ago
- db0b3bc Roll chromium_revision 35431c5114..10156469d6 (703133:703248) by chromium-webrtc-autoroll · 4 years, 6 months ago
- b441acf AEC3: Add support in the echo subtractor for handling multiple channels by Per Åhgren · 4 years, 6 months ago
- d21db5d Roll chromium_revision e2b55cc552..35431c5114 (703005:703133) by chromium-webrtc-autoroll · 4 years, 6 months ago
- 0e0a04c Roll chromium_revision b5ead1daa2..e2b55cc552 (702047:703005) by chromium-webrtc-autoroll · 4 years, 6 months ago
- 2b84dad Fixed issue with H264 packet buffer where it was not detecting presence of sps/pps for idr frames by Shyam Sadhwani · 4 years, 6 months ago
- 4f2e940 ACM: Adding support for more than 2 channels in the send pipeline by Per Åhgren · 4 years, 6 months ago
- dc34a25 Adds RTPSenderVideo::Config struct with red/ulpfec config by Erik Språng · 4 years, 6 months ago
- b9bfe65 Delete VCMEncodedFrame::VerifyAndAllocate by Niels Möller · 4 years, 6 months ago
- 7536bc5 Account for IP and UDP headers in emulated network by Niels Möller · 4 years, 6 months ago
- ed8eadc Update RTC_LOGs in DtlsTransport to be able to distinguish errors. by Henrik Boström · 4 years, 6 months ago