1. cde4b67 [SourceTracker] Move state to the worker thread, remove mutex. by Tommi · 1 year, 9 months ago
  2. 6a7bf10 Replace "rcvd" with "received" for readability by Philipp Hancke · 1 year, 9 months ago
  3. 217b384 Remove rtp header extension from config of Call audio and video receivers by Per K · 2 years ago
  4. 9253240 Reland "Remove use of ReceiveStreamRtpConfig:transport_cc" by Per K · 2 years ago
  5. be5c713 Revert "Remove use of ReceiveStreamRtpConfig:transport_cc" by Olga Sharonova · 2 years ago
  6. 97ba853 Remove use of ReceiveStreamRtpConfig:transport_cc by Per K · 2 years ago
  7. 1a84b56 Implement RTCInboundRTPStreamStats.JitterBufferMinimumDelay by Ivo Creusen · 2 years, 6 months ago
  8. a136ed4 Add SetTransportCc to ReceiveStreamInterface. by Tommi · 2 years, 8 months ago
  9. 3176ef7 Rename AudioReceiveStream to AudioReceiveStreamInterface by Tommi · 2 years, 8 months ago
  10. dddbbeb Rename internal::AudioReceiveStream to AudioReceiveStreamImpl by Tommi · 2 years, 8 months ago
  11. cc50b04 Remove config() getter from AudioReceiveStream(). by Tommi · 2 years, 8 months ago
  12. 28a2c63 Adding packetsDiscarded to RTCReceivedRtpStreamStats. by Minyue Li · 3 years, 6 months ago
  13. 6eda26c Reland "Remove AudioReceiveStream::Reconfigure() method." by Tommi · 3 years, 7 months ago
  14. 8a18e5b Revert "Remove AudioReceiveStream::Reconfigure() method." by Andrey Logvin · 3 years, 7 months ago
  15. e2561e1 Remove AudioReceiveStream::Reconfigure() method. by Tommi · 3 years, 7 months ago
  16. 02df2eb Split AudioStream initialization into worker / network steps. by Tommi · 3 years, 8 months ago
  17. c1d5891 Replace `new rtc::RefCountedObject` with `rtc::make_ref_counted` in a few files by Tomas Gunnarsson · 3 years, 9 months ago
  18. dea374a Deliver packet info to source listeners when audio playout is disabled. by Ranveer Aggarwal · 4 years ago
  19. c20baf6 Remove nesting of Naggy/Strict/NiceMock by Alex Konradi · 4 years, 1 month ago
  20. 42cafa5 Delete legacy stats minWaitingTimeMs and medianWaitingTimeMs from ACM. by Niels Möller · 4 years, 2 months ago
  21. 4461f05 Delete unused NetEq stats currentPacketLossRate, currentDiscardRate and addedSamples by Niels Möller · 4 years, 4 months ago
  22. 6b4d962 Fix standard GetStats to not modify NetEq state. by Niels Möller · 4 years, 4 months ago
  23. 9e9c8b7 Delete obsolete method AudioReceiveStream::OnRtpPacket by Niels Möller · 4 years, 5 months ago
  24. cc73ed3 APM: Add build flag to allow building WebRTC without APM by Per Åhgren · 4 years, 9 months ago
  25. 3e9af7f Insert audio frame transformer between depacketizer and decoder. by Marina Ciocea · 4 years, 10 months ago
  26. e618cc9 Add jitterBufferTargetDelay as RTCNonStandardStatsMember to new GetStats API by Artem Titov · 4 years, 10 months ago
  27. 7a9a092 Delete media transport integration. by Bjorn A Mellem · 5 years ago
  28. fcf79cc Add estimatedPlayoutTimestamp to RTCInboundRTPStreamStats. by Åsa Persson · 5 years ago
  29. ac0a4cb Reland "Fix GetStats bytesSent/Received, wireup headerBytesSent/Received" by Niels Möller · 5 years ago
  30. ef0627f Revert "Fix GetStats bytesSent/Received, wireup headerBytesSent/Received" by Mirko Bonadei · 5 years ago
  31. fbde32e Fix GetStats bytesSent/Received, wireup headerBytesSent/Received by Niels Möller · 5 years ago
  32. 65024d9 Remove clock drift metric from NetEq. by Jakob Ivarsson · 5 years ago
  33. 224c69d Delete ext_seqnum member from VoiceSenderInfo and VoiceReceiverInfo by Niels Möller · 5 years ago
  34. 70efdde Set local ssrc at construction of Rtp module by Erik Språng · 5 years ago
  35. 5b5d97c Reland of "Reporting of decoding_codec_plc events"" by Alex Narest · 5 years ago
  36. bedb7a8 Revert "Reporting of decoding_codec_plc events" by Mirko Bonadei · 5 years ago
  37. 0a88ea0 Reporting of decoding_codec_plc events by Alex Narest · 5 years ago
  38. a4d8737 Format almost everything. by Jonas Olsson · 6 years ago
  39. 3472b9a Delete RTCInboundRTPStreamStats::fraction_lost by Niels Möller · 6 years ago
  40. 4f08faa Introduce MediaTransportConfig by Anton Sukhanov · 6 years ago
  41. 8d8ffdb Expose new audio stats on the API by Ivo Creusen · 6 years ago
  42. 6a489f2 Fully qualify googletest symbols. by Mirko Bonadei · 6 years ago
  43. 17b050f Fixes ClangTidy errors in audio/ by Benjamin Wright · 6 years ago
  44. 9ffb5df Removes unused mock_bitrate_controller. by Sebastian Jansson · 6 years ago
  45. 8fb1a6a Delete a few return values from audio streams and video send streams. by Niels Möller · 6 years ago
  46. 977b335 Injecting Clock into audio streams. by Sebastian Jansson · 6 years ago
  47. 5c2f1f0 Add some missing includes and dependencies. by Bjorn Terelius · 6 years ago
  48. 0acffb5 Expose `jitterBufferEmittedCount` in addition to the existing `jitterBufferDelay` for `getStats()`. by Chen Xing · 6 years ago
  49. f693bfa Remove CodecInst pt.2 by Fredrik Solenberg · 6 years ago
  50. 349ade3 Delete class ChannelReceiveProxy. by Niels Möller · 6 years ago
  51. 7d76a31 Use MediaTransportInterface, for audio streams. by Niels Möller · 6 years ago
  52. 78410ad Fixes use after free error when setting a new FrameEncryptor on ChannelSend. by Benjamin Wright · 6 years ago
  53. ae4237e Set ChannelReceive transport at construction time. by Niels Möller · 6 years ago
  54. 530ead4 Split voe::Channel into ChannelSend and ChannelReceive by Niels Möller · 6 years ago
  55. b222f49 Split ChannelProxy into send and receive classes. by Niels Möller · 6 years ago
  56. 30b4839 Refactor voe::Channel to not use RtpReceiver. by Niels Möller · 6 years ago
  57. fa4e185 Delete class voe::RtcEventLogProxy by Niels Möller · 6 years ago
  58. 7008287 Delete struct webrtc::PacketTime. by Niels Möller · 6 years ago
  59. 665174f Reformat the WebRTC code base by Yves Gerey · 7 years ago
  60. f782492 Delete RtpFeedback. The ssrc for a receive stream should be known at by Niels Möller · 7 years ago
  61. a8b7c7f Move remaining traces of VoiceEngine by Fredrik Solenberg · 7 years ago
  62. 90ea504 Delete Channel::OnRecoveredPacket. by Niels Möller · 7 years ago
  63. 8f5787a Move ownership of voe::Channel into Audio[Receive|Send]Stream. by Fredrik Solenberg · 7 years ago
  64. 3b903d0 Reconfigure, not reconstruct, AudioReceiveStreams. by Fredrik Solenberg · 7 years ago
  65. d524751 Replace VoEBase::[Start/Stop]Playout(). by Fredrik Solenberg · 7 years ago
  66. 2a87797 Remove voe::TransmitMixer by Fredrik Solenberg · 7 years ago
  67. d319534 Move ADM initialization into WebRtcVoiceEngine by Fredrik Solenberg · 7 years ago
  68. b0a0207 Added RTCMediaStreamTrackStats.jitterBufferDelay for audio by Gustaf Ullberg · 7 years ago
  69. 1c239d4 Remove voe::Statistics. by solenberg · 7 years ago
  70. fc3a2e3 Remove the VoiceEngineObserver callback interface. by solenberg · 7 years ago
  71. 9a2e906 Added RTCMediaStreamTrackStats.concealmentEvents by Gustaf Ullberg · 7 years ago
  72. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  73. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/audio/audio_receive_stream_unittest.cc]
  74. 0e320ec Wiring discard rate of audio FEC/RED packets up to StatsReport. by minyue-webrtc · 7 years ago
  75. 2dbc69f Add stats totalSamplesReceived and concealedSamples by Steve Anton · 7 years ago
  76. 0c3ca75 Replacing NetEq discard rate with secondary discarded rate. by minyue-webrtc · 7 years ago
  77. e76bd3a 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 · 7 years ago
  78. a9cc40b Allow an external audio processing module to be used in WebRTC by peah · 8 years ago
  79. 0f15f92 Introduce RtpStreamReceiverInterface and RtpStreamReceiverControllerInterface. by nisse · 8 years ago
  80. 37e99fd Move AudioDecoder and AudioDecoderFactory mocks to webrtc/test/ by kwiberg · 8 years ago
  81. fdbfdc9 Let PacketRouter separate send and receive modules. by nisse · 8 years ago
  82. 1c07c70 Reland "WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType" by kwiberg · 8 years ago
  83. 670a7f3 Revert of WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType (patchset #13 id:260001 of https://codereview.webrtc.org/2686043006/ ) by kwiberg · 8 years ago
  84. 1724cfb WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType by kwiberg · 8 years ago
  85. 922246a Replace NULL with nullptr or null in webrtc/audio/ and common_audio/. by deadbeef · 8 years ago
  86. 657bab2 Replace AudioReceiveStream::DeliverRtp with OnRtpPacket. by nisse · 8 years ago
  87. 4709e89 Move RemoteBitrateEstimator::RemoveStream calls from receive streams to Call. by nisse · 8 years ago
  88. bd9a77f Remove most of the remaining calls to VoE APIs from Audio[Send|Receive]Stream. by solenberg · 8 years ago
  89. d44ce05 Reland of Always call RemoteBitrateEstimator::IncomingPacket from Call. (patchset #1 id:1 of https://codereview.webrtc.org/2668973003/ ) by nisse · 8 years ago
  90. 14245cc Revert of Always call RemoteBitrateEstimator::IncomingPacket from Call. (patchset #9 id:160001 of https://codereview.webrtc.org/2659563002/ ) by nisse · 8 years ago
  91. 6d4dd59 Always call RemoteBitrateEstimator::IncomingPacket from Call. by nisse · 8 years ago
  92. 0245da0 Move ownership of PacketRouter from CongestionController to Call. by nisse · 8 years ago
  93. 04c0722 Replace AudioConferenceMixer with AudioMixer. by aleloi · 8 years ago
  94. 10111bc Passed AudioMixer to AudioState::Config. by aleloi · 8 years ago
  95. dd31071 Added an empty AudioTransportProxy to AudioState. by aleloi · 8 years ago
  96. 7602aab Remove usage of VoEBase::AssociateSendChannel() from WVoMC. by solenberg · 8 years ago
  97. b521aa7 Clean up abs-send-time for audio. by stefan · 8 years ago
  98. 2d81eb3 Fix BWE simulations so that it uses the delay based BWE. by terelius · 8 years ago
  99. 18e0b67 Restarting channel when swapping AudioReceiveStreams in WebrtcVoE. by aleloi · 8 years ago
  100. ac9f876 Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/ by kwiberg · 8 years ago