- 00ed864 Recently we moved webrtc/base to webrtc/rtc_base, so these by mbonadei · 7 years ago
- 950c7f2 Fix setting of recovered flag in RtxReceiveStream. by nisse · 7 years ago
- 443f9a9 Wiring discard rate of audio FEC/RED packets up to StatsReport. by minyue-webrtc · 7 years ago
- c205a94 Now that https://codereview.webrtc.org/3003643002 is landed we can by mbonadei · 7 years ago
- 3ef80a2 Reverse |rtx_payload_types| map, and rename. by nisse · 7 years ago
- 13871b5 Uncomment commented-out sequence-checks in call.cc by eladalon · 7 years ago
- 94ac82f Add stats totalSamplesReceived and concealedSamples by Steve Anton · 7 years ago
- a875cf0 New accessor function VideoReceiveStream::Config::Rtp::AddRtxBinding by Niels Möller · 7 years ago
- 6a5ac8f Report max interframe delay over window insdead of interframe delay sum by ilnik · 7 years ago
- 3a071f6 Let Call::OnRecoveredPacket parse RTP header extensions. by brandtr · 7 years ago
- 7865664 Fix an implicit narrowing conversion found by MSVC by oprypin · 7 years ago
- 53431d2 Move PacedSender ownership to RtpTransportControllerSend. by Stefan Holmer · 7 years ago
- f0c86c0 Move video send/receive stream headers to webrtc/call. by aleloi · 7 years ago
- 0f007ea Use SingleThreadedTaskQueue in DirectTransport by eladalon · 7 years ago
- 35c6258 Fix size_t to int in RtpDemuxer. by Steve Anton · 7 years ago
- d9732a5 Reduce code repetition in RtcpDemuxerTest. by Steve Anton · 7 years ago
- a70873d Add BUNDLE processing to RtpDemuxer. by Steve Anton · 7 years ago
- 8c7e2bd Change ThreadChecker to SequencedTaskChecker in internal::Call by eladalon · 7 years ago
- c59e7d2 Reduce code repetition in RtpDemuxerTest. by Steve Anton · 7 years ago
- 6f7363f Rename RsidResolutionObserver to SsrcBindingObserver. by Steve Anton · 7 years ago
- 9db127b Reland of Add functionality which limits the number of bytes on the network. (patchset #1 id:1 of https://codereview.webrtc.org/3001653002/ ) by stefan · 7 years ago
- 0a27bbc Delete unneeded Start and Stop methods on FlexfecReceiveStream. by Niels Möller · 7 years ago
- a562a59 Revert of Add functionality which limits the number of bytes on the network. (patchset #26 id:500001 of https://codereview.webrtc.org/2918323002/ ) by stefan · 7 years ago
- b96990e Removed an unused variable from CallPerfTest::TestAudioVideoSync() by eladalon · 7 years ago
- b0be841 Add functionality which limits the number of bytes on the network. by stefan · 7 years ago
- 6ab33cf Wire up RTP keep-alive in ortc api. by sprang · 7 years ago
- ec0b697 Reland of SSRC and RSID may only refer to one sink each in RtpDemuxer (patchset #1 id:1 of https://codereview.webrtc.org/2993633002/ ) by eladalon · 7 years ago
- 3a68f5d Revert of SSRC and RSID may only refer to one sink each in RtpDemuxer (patchset #15 id:280001 of https://codereview.webrtc.org/2968693002/ ) by zhihuang · 7 years ago
- 783ce68 Rename ViEEncoder to VideoStreamEncoder by mflodman · 7 years ago
- 92bdb64 Tracking mock_process_thread with a GN target by mbonadei · 7 years ago
- 9eed04a SSRC and RSID may only refer to one sink each in RtpDemuxer by eladalon · 7 years ago
- 2c1cbc1 Protected streams report RTP messages directly to the FlexFec streams by eladalon · 7 years ago
- 0360a1c Make FlexfecReceiveStreamImpl::started_ into std::atomic<bool> by eladalon · 7 years ago
- 5fb3e75 Only one implementation of MockRtpPacketSink once by eladalon · 7 years ago
- cc7d766 Remove DCHECK from Call's ctor that could never fail by eladalon · 7 years ago
- ea04cc0 Make ~webrtc::AudioSendStream public, and s/config()/GetConfig(), as well as make public. by eladalon · 8 years ago
- 0dedc27 Get rid of unnecessary cast of FlexfecReceiveStreamImpl to FlexfecReceiveStream by eladalon · 8 years ago
- 2bc62f3 Remove remains of webrtc/base by ehmaldonado · 8 years ago
- a42f080 Adds a histogram metric tracking for how long audio RTP packets are sent by saza · 8 years ago
- d1701d0 Adding stats that can be used to compute output audio levels as described here https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudioenergy. by zstein · 8 years ago
- 59d5575 Use relative paths in GN files. by jianjun.zhu · 8 years ago
- 3237bd2 Call should allow pass through of keep-alive packets. by sprang · 8 years ago
- bc05a6a Move RTP keep-alive config from VideoSendStream::Config to Call::Config by sprang · 8 years ago
- 4f870fc Revert of Remove remains of webrtc/base (patchset #7 id:120001 of https://codereview.webrtc.org/2973183002/ ) by ehmaldonado · 8 years ago
- c6c814d Remove remains of webrtc/base by ehmaldonado · 8 years ago
- 76de83e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 8 years ago
- e3d2394 Reland of Add received audio/video call duration metrics based on packets. by saza · 8 years ago
- 1480b28 Revert of Add received audio and video call duration metrics based on packets. (patchset #4 id:140001 of https://codereview.webrtc.org/2957073002/ ) by saza · 8 years ago
- 88bc2f6 Add received audio/video call duration metrics based on packets. by saza · 8 years ago
- 101c359 Add underscore at end of Call members' names by eladalon · 8 years ago
- 273eb06 Fix gmock warnings emanating from FlexfecReceiveStreamTest by eladalon · 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
- 2104994 Remove RtpDemuxer tweak for preventing multiple RSID inspections by eladalon · 8 years ago
- 588f761 Allow an external audio processing module to be used in WebRTC by peah · 8 years ago
- 242fd78 nit: Rename RtpDemuxer::sink_ to RtpDemuxer::ssrc_sinks_ by eladalon · 8 years ago
- 3a84a8f Reland of Periodically update codec bit/frame rate settings. by sprang · 8 years ago
- 9ae9fd9 Reland of Create RtcpDemuxer (patchset #1 id:1 of https://codereview.webrtc.org/2957763002/ ) by eladalon · 8 years ago
- ca3693c Revert of Create RtcpDemuxer (patchset #13 id:240001 of https://codereview.webrtc.org/2943693003/ ) by guidou · 8 years ago
- 5af9557 Create RtcpDemuxer. Capabilities: by eladalon · 8 years ago
- 04ef6b0 Introduce RtpStreamReceiverInterface and RtpStreamReceiverControllerInterface. by nisse · 8 years ago
- f91805c Support building WebRTC without audio and video. by zhihuang · 8 years ago
- f7f8eb4 Move MinPositive to call.h as discussed here: https://codereview.chromium.org/2888303005/#msg19 by zstein · 8 years ago
- 8eacbbd Log an error in RtpDemuxer::FindSsrcAssociations() if kMaxProcessedSsrcs exceeded by eladalon · 8 years ago
- f75b6fa Use rtp_header_extension_map.h instead of rtp_header_extension.h by Danil Chapovalov · 8 years ago
- 3d545a2 s/WebRtcVideoChannel2/WebRtcVideoChannel and s/WebRtcVideoEngine2/WebRtcVideoEngine by eladalon · 8 years ago
- 41449dc Add RSID-based demuxing to RtpDemuxer by eladalon · 8 years ago
- 33944ff Relanding: Adds PeerConnectionInterface::UpdateCallBitrate to give clients more control of the bandwidth estimator. PeerConnection implements this method by passing a BitrateConfigMask to its associated Call, which is combined with the existing BitrateConfig and passed on to the SendSideCongestionController as necessary. The existing BitrateConfig generally comes from the x-google-{min,start,max}-bitrate params in the SDP. by zstein · 8 years ago
- fb8fb2d Revert of Add PeerConnectionInterface::UpdateCallBitrate. (patchset #7 id:120001 of https://codereview.webrtc.org/2888303005/ ) by charujain · 8 years ago
- 9f8b6f3 Adds PeerConnectionInterface::UpdateCallBitrate to give clients more control of the bandwidth estimator. PeerConnection implements this method by passing a BitrateConfigMask to its associated Call, which is combined with the existing BitrateConfig and passed on to the SendSideCongestionController as necessary. The existing BitrateConfig generally comes from the x-google-{min,start,max}-bitrate params in the SDP. by zstein · 8 years ago
- 2d1de82 New targets call:rtp_interfaces, call:rtp_receiver, call:rtp_sender. by nisse · 8 years ago
- baa253b Create unit tests for RtpDemuxer by eladalon · 8 years ago
- ad35e29 Address some violations of chromium-style. by nisse · 8 years ago
- 33f05e7 This cl removes RtcEventLog deps to call:call_interfaces. The purpose is to be able to use the event log from the upcoming RtpTransport. by perkj · 8 years ago
- 30fa632 Store/restore RTP state for audio streams with same SSRC within a call by ossu · 8 years ago
- eaf4d68 Replace AudioSendStream::Config with rtclog::StreamConfig. by perkj · 8 years ago
- a6d8766 Replace AudioReceiveStream::Config with rtclog::StreamConfig. by perkj · 8 years ago
- 433b35c Replace VideoSendStream::Config with new rtclog::StreamConfig in RtcEventLog. by perkj · 8 years ago
- acddd51 Replace VideoReceiveStream::Config with new rtclog::StreamConfig in RtcEventLog. by perkj · 8 years ago
- 002b533 New class RtxReceiveStream. by nisse · 8 years ago
- a4c5957 Renaming probing_interval to bwe_period globally. by minyue · 8 years ago
- 82158aa Simple tests for Call::SetBitrateConfig. by zstein · 8 years ago
- 78bbf3a New class RtpDemuxer and RtpPacketSinkInterface, use in Call. by nisse · 8 years ago
- ba6f478 Make Call::OnRecoveredPacket parse RTP header and call OnRtpPacket. by nisse · 8 years ago
- cd2f080 Add untracked headers in modules/rtp_rtcp by danilchap · 8 years ago
- 10a054d Delete helper class MediaTypePacketReceiver. by nisse · 8 years ago
- 29a1f8c This will allow me to test that Call invokes SendSideCongestionController::SetBweBitrates as expected (for https://codereview.chromium.org/2793913008). by zstein · 8 years ago
- f26202b Injectable audio encoders: Moved audio encoder, factory and builtin factory to api/. by ossu · 8 years ago
- fe5f71c Injectable audio encoders: WebRtcVoiceEngine and company by ossu · 8 years ago
- 279bdc9 Creating webrtc:video_stream_api by mbonadei · 8 years ago
- 9cdb538 GN: Tighten up test target visibility + refactorings by kjellander · 8 years ago
- d1dc63e Delete declaration of non-existing function webrtc::Version(). by nisse · 8 years ago
- cddf701 Move RtpTransportControllerSend to a new file. by nisse · 8 years ago
- b660bfe Replace first_packet_sent_ms_ in Call. by asapersson · 8 years ago
- 7e3c920 Delete VieRemb class, move functionality to PacketRouter. by nisse · 8 years ago
- b9f62aa Making FakeNetworkPipe demux audio and video packets. by minyue · 8 years ago
- 69579d7 Reland of Implemented the GetSources() in native code. (patchset #1 id:1 of https://codereview.webrtc.org/2809613002/ ) by hbos · 8 years ago
- af8f6e5 Move AudioDecoder and AudioDecoderFactory mocks to webrtc/test/ by kwiberg · 8 years ago
- fee642e Revert of Implemented the GetSources() in native code. (patchset #11 id:510001 of https://codereview.webrtc.org/2770233003/ ) by olka · 8 years ago
- cdb9439 Event log cleanup in tests. by philipel · 8 years ago