- 19cf6be Unwrap picture ids in the RtpFrameReferencerFinder. by philipel · 7 years ago
- dfa8f5e Revert of Use RtxReceiveStream. (patchset #5 id:80001 of https://codereview.webrtc.org/3008773002/ ) by nisse · 7 years ago
- daf49be Use RtxReceiveStream. by nisse · 7 years ago
- 36189cd Move RtpExtension to api/ directory and config.h/.cc to call/. by Stefan Holmer · 7 years ago
- 3ef80a2 Reverse |rtx_payload_types| map, and rename. by nisse · 7 years ago
- 7f62201 Eliminate RtpVideoStreamReceiver::receive_cs_ in favor of using a SequencedTaskChecker by eladalon · 7 years ago
- 0df2c66 Reland of Add a flags field to video timing extension. (patchset #1 id:1 of https://codereview.webrtc.org/2995953002/ ) by sprang · 7 years ago
- 786989c Revert of Add a flags field to video timing extension. (patchset #15 id:280001 of https://codereview.webrtc.org/3000753002/ ) by emircan · 7 years ago
- a58c48e Add a flags field to video timing extension. by sprang · 7 years ago
- d8a48fb Workaround for PacketBuffer bug. by philipel · 7 years ago
- 2c1cbc1 Protected streams report RTP messages directly to the FlexFec streams by eladalon · 7 years ago
- a207a3a Explicitly inform PacketRouter which RTP-RTCP modules are REMB-candidates by eladalon · 7 years ago
- 76de83e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 8 years ago
- bc32410 Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 8 years ago
- 60154fd Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 8 years ago
- 0e9e5a2 Reland of Only compare sequence numbers from the same SSRC in ForwardErrorCorrection. (patchset #1 id:1 of https://codereview.webrtc.org/2919313005/ ) by brandtr · 8 years ago
- 39e693a Implement timing frames. by ilnik · 8 years ago
- e65f137 Only create H264 frames if there are no gaps in the packet sequence number. by philipel · 8 years ago
- 9fe4191 Use the configured remote ssrc instead of relying on the first received packet RtpStreamReceiver. by stefan · 8 years ago
- 14ec7e0 Rename class RtpStreamReceiver --> RtpVideoStreamReceiver. by nisse · 8 years ago[Renamed (88%) from video/rtp_stream_receiver.cc]
- 032d66e Delete RtpData::OnRecoveredPacket, use RecoveredPacketReceiver instead. by nisse · 8 years ago
- 47be53a Dont request keyframes if the stream is inactive or if we are currently receiving a keyframe. by philipel · 8 years ago
- 22e0182 Request keyframe if the first received frame is not a keyframe. by philipel · 8 years ago
- ba6f478 Make Call::OnRecoveredPacket parse RTP header and call OnRtpPacket. by nisse · 8 years ago
- 7e3c920 Delete VieRemb class, move functionality to PacketRouter. by nisse · 8 years ago
- cb436f6 Reland of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2809653004/ ) by ilnik · 8 years ago
- 8f162b3 Revert of Add content type information to encoded images and corresponding rtp extension header (patchset #4 id:400001 of https://codereview.webrtc.org/2812913002/ ) by ilnik · 8 years ago
- 1087c1e Reland of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2816463002/ ) by ilnik · 8 years ago
- f888bbf Revert of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2811963002/ ) by ilnik · 8 years ago
- 6b65c51 Reland of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2816463002/ ) by ilnik · 8 years ago
- fa3ebb0 Revert of Add content type information to encoded images and corresponding rtp extension header (patchset #31 id:600001 of https://codereview.webrtc.org/2772033002/ ) by ilnik · 8 years ago
- e849cea Add content type information to Encoded Images and add corresponding RTP extension header. by ilnik · 8 years ago
- 5e9233b Let PacketRouter separate send and receive modules. by nisse · 8 years ago
- 82efb01 Delete unneeded includes of deprecated system_wrappers include files. by nisse · 8 years ago
- 59601fc Allow padding packet in video streams. by philipel · 8 years ago
- 5081145 Don't allocate any RTPSender object for a receive only RtpRtcp module. by nisse · 8 years ago
- 8dc857c Delete support for sending RTCP RPSI and SLI messages. by nisse · 8 years ago
- c72b1c8 Add support for Location (RTC_FROM_HERE) to ProcessThread::RegisterModule. by tommi · 8 years ago
- 451d9c2 Remove |running_| state from NackModule to avoid running a tight loop. by tommi · 8 years ago
- a7cca31 Revert of Revert Make the new jitter buffer the default jitter buffer. (patchset #1 id:1 of https://codereview.chromium.org/2682073003/ ) by philipel · 8 years ago
- 6d5d814 Replace RtpStreamReceiver::DeliverRtp with OnRtpPacket. by nisse · 8 years ago
- 5bb6b3b Revert Make the new jitter buffer the default jitter buffer. by stefan · 8 years ago
- 49cdd35 Move RemoteBitrateEstimator::RemoveStream calls from receive streams to Call. by nisse · 8 years ago
- a0add70 Reland of Always call RemoteBitrateEstimator::IncomingPacket from Call. (patchset #1 id:1 of https://codereview.webrtc.org/2668973003/ ) by nisse · 8 years ago
- 07dbf94 Reland of Make the new jitter buffer the default jitter buffer. (patchset #2 id:260001 of https://codereview.chromium.org/2656983002/ ) by philipel · 8 years ago
- 5cacd3d Revert of Always call RemoteBitrateEstimator::IncomingPacket from Call. (patchset #9 id:160001 of https://codereview.webrtc.org/2659563002/ ) by nisse · 8 years ago
- 40ab430 Always call RemoteBitrateEstimator::IncomingPacket from Call. by nisse · 8 years ago
- 685e50e Only update VCMTiming on every received frame instead of every received packet. by philipel · 8 years ago
- bcf3fd3 Reland of Make RTX pt/apt reconfigurable by calling WebRtcVideoChannel2::SetRecvParameters. (patchset #1 id:1 of https://codereview.webrtc.org/2649323010/ ) by brandtr · 8 years ago
- db89bb8 Revert of Make the new jitter buffer the default jitter buffer. (patchset #2 id:290001 of https://codereview.chromium.org/2652043005/ ) by philipel · 8 years ago
- 3bdef22 Revert of Make RTX pt/apt reconfigurable by calling WebRtcVideoChannel2::SetRecvParameters. (patchset #7 id:160001 of https://codereview.webrtc.org/2646073004/ ) by kjellander · 8 years ago
- 914a215 Make RTX pt/apt reconfigurable by calling WebRtcVideoChannel2::SetRecvParameters. by brandtr · 8 years ago
- 911c399 Reland of Make the new jitter buffer the default jitter buffer. (patchset #1 id:1 of https://codereview.webrtc.org/2638423003/ ) by philipel · 8 years ago
- 900d66d Unit test out of band H264 SPS,PPS within RtpStreamReceiver. by johan · 8 years ago
- 684009a Reland of H264SpsPpsTracker.InsertSpsPpsNalus() should accept Nalus with header. by johan · 8 years ago
- dbce297 Drop pacer and retransmission_rate_limiter from RtpStreamReceiver constructor. by nisse · 8 years ago
- 20776fa Revert of H264SpsPpsTracker.InsertSpsPpsNalus() should accept Nalus with header. (patchset #3 id:40001 of https://codereview.webrtc.org/2638933002/ ) by kjellander · 8 years ago
- 27afadb H264SpsPpsTracker.InsertSpsPpsNalus() should accept Nalus with header. by johan · 8 years ago
- dfc7880 Revert of Make the new jitter buffer the default jitter buffer. (patchset #2 id:230001 of https://codereview.webrtc.org/2642753002/ ) by kjellander · 8 years ago
- 87b55bf Reland of Make the new jitter buffer the default jitter buffer. (patchset #1 id:1 of https://codereview.chromium.org/2632123005/ ) by philipel · 8 years ago
- 24d7b7e Revert of Make the new jitter buffer the default jitter buffer. (patchset #7 id:120001 of https://codereview.chromium.org/2627463004/ ) by philipel · 8 years ago
- 4612a95 Make the new jitter buffer the default jitter buffer. by philipel · 8 years ago
- 09c99cf Wire up H264 fmtp sprop-parameter-sets with H264SpsPpsTracker. by philipel · 8 years ago
- f3b4b21 Rename RtpStreamReceiver::SetCodec() to ::AddCodec(). by johan · 8 years ago
- cc24846 Rename RtpStreamReceiver::IsFecEnabled to RtpStreamReceiver::IsUlpfecEnabled. by brandtr · 8 years ago
- fc3c29b Update video histograms that do not have a minimum lifetime limit before being recorded. by asapersson · 8 years ago
- c138a32 Fix memory leak in video_coding::PacketBuffer::InsertPacket. by philipel · 8 years ago
- 1cbcced Reland of move RTPPayloadStrategy and simplify RTPPayloadRegistry (patchset #1 id:1 of https://codereview.webrtc.org/2528993002/ ) by magjed · 8 years ago
- e82ac9a Revert of Remove RTPPayloadStrategy and simplify RTPPayloadRegistry (patchset #7 id:240001 of https://codereview.webrtc.org/2524923002/ ) by magjed · 8 years ago
- 402fe33 Remove RTPPayloadStrategy and simplify RTPPayloadRegistry by magjed · 8 years ago
- 4c1eb05 Send audio and video codecs to RTPPayloadRegistry by magjed · 8 years ago
- ba3a19d New jitter buffer experiment. by philipel · 8 years ago
- c1d9346 Remove RED/RTX workaround from sender/receiver and VideoEngine2. by brandtr · 8 years ago
- e4bac0a Simplify {,Set}UlpfecConfig interface. by brandtr · 8 years ago
- 0125e3f Rename {,Set}GenericFECStatus to {,Set}UlpfecConfig. by brandtr · 8 years ago
- 001769c Rename FecReceiver to UlpfecReceiver. by brandtr · 8 years ago
- d984c57 Rename FecConfig to UlpfecConfig in config.h. by brandtr · 8 years ago
- 56fc637 Remove RTC_LOGGED_* macro. by asapersson · 8 years ago
- 78706f4 Reland of Ignore Camera and Flip bits in CVO when parsing video rotation (patchset #1 id:1 of https://codereview.webrtc.org/2300323002/ ) by magjed · 8 years ago
- b437f62 Revert of Ignore Camera and Flip bits in CVO when parsing video rotation (patchset #3 id:80001 of https://codereview.webrtc.org/2280703002/ ) by magjed · 8 years ago
- c123161 Ignore Camera and Flip bits in CVO when parsing video rotation by Magnus Jedvert · 8 years ago
- 46e4b5a Move RTP for synchroninzation and rename classes, files and variables. by mflodman · 8 years ago
- 0077997 Add NACK rate throttling for audio channels. by Erik Språng · 8 years ago
- 4c991f1 Reland of actor NACK bitrate allocation (patchset #1 id:1 of https://codereview.webrtc.org/2131913003/ ) by sprang · 9 years ago
- b78878b Revert of Reland Issue 2061423003: Refactor NACK bitrate allocation (patchset #1 id:1 of https://codereview.webrtc.org/2131313002/ ) by aluebs · 9 years ago
- 209767c Reland Issue 2061423003: Refactor NACK bitrate allocation by Erik Språng · 9 years ago
- 5158958 Revert of Refactor NACK bitrate allocation (patchset #16 id:300001 of https://codereview.webrtc.org/2061423003/ ) by sprang · 9 years ago
- 9eb26bd Refactor NACK bitrate allocation by Erik Språng · 9 years ago
- 6ba7dfc Reland of move audio/video distinction for probe packets. (patchset #1 id:1 of https://codereview.webrtc.org/2086633002/ ) by pbos · 9 years ago
- f4d4351 Revert of Remove audio/video distinction for probe packets. (patchset #2 id:20001 of https://codereview.webrtc.org/2061193002/ ) by honghaiz · 9 years ago
- b0d0745 Remove audio/video distinction for probe packets. by Peter Boström · 9 years ago
- c9da01c GN: Add video_engine_tests by Peter Boström · 9 years ago
- 87875ef Movable support for VideoReceiveStream::Config and avoid copies. by Tommi · 9 years ago
- 00cc045 Add sender controlled playout delay limits by isheriff · 9 years ago
- c4921f4 Remove use of RtpHeaderExtension and clean up by isheriff · 9 years ago
- 6418d5f Removing some old code which looked like it had to do with NACK handling but in reality did nothing. by Fredrik Solenberg · 9 years ago
- 7738985 Delete all use of tick_util.h. by Niels Möller · 9 years ago
- 5c3920a Jitter delay now depend on protection mode (FEC/NACK). by philipel · 9 years ago
- 33927c7 Move, almost, all receive side references to RTP to RtpStreamReceiver. by mflodman · 9 years ago
- 639986e Removed all RTP dependencies from ViEChannel and renamed class. by mflodman · 9 years ago