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gerrit-public.fairphone.software / platform / external / webrtc / 8fdcac3f06ae09fe17bbed02037ef136f086cc00 / audio / audio_receive_stream_unittest.cc
  1. 30b4839 Refactor voe::Channel to not use RtpReceiver. by Niels Möller · 6 years ago
  2. fa4e185 Delete class voe::RtcEventLogProxy by Niels Möller · 6 years ago
  3. 7008287 Delete struct webrtc::PacketTime. by Niels Möller · 6 years ago
  4. 665174f Reformat the WebRTC code base by Yves Gerey · 6 years ago
  5. f782492 Delete RtpFeedback. The ssrc for a receive stream should be known at by Niels Möller · 6 years ago
  6. a8b7c7f Move remaining traces of VoiceEngine by Fredrik Solenberg · 7 years ago
  7. 90ea504 Delete Channel::OnRecoveredPacket. by Niels Möller · 7 years ago
  8. 8f5787a Move ownership of voe::Channel into Audio[Receive|Send]Stream. by Fredrik Solenberg · 7 years ago
  9. 3b903d0 Reconfigure, not reconstruct, AudioReceiveStreams. by Fredrik Solenberg · 7 years ago
  10. d524751 Replace VoEBase::[Start/Stop]Playout(). by Fredrik Solenberg · 7 years ago
  11. 2a87797 Remove voe::TransmitMixer by Fredrik Solenberg · 7 years ago
  12. d319534 Move ADM initialization into WebRtcVoiceEngine by Fredrik Solenberg · 7 years ago
  13. b0a0207 Added RTCMediaStreamTrackStats.jitterBufferDelay for audio by Gustaf Ullberg · 7 years ago
  14. 1c239d4 Remove voe::Statistics. by solenberg · 7 years ago
  15. fc3a2e3 Remove the VoiceEngineObserver callback interface. by solenberg · 7 years ago
  16. 9a2e906 Added RTCMediaStreamTrackStats.concealmentEvents by Gustaf Ullberg · 7 years ago
  17. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  18. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/audio/audio_receive_stream_unittest.cc]
  19. 0e320ec Wiring discard rate of audio FEC/RED packets up to StatsReport. by minyue-webrtc · 7 years ago
  20. 2dbc69f Add stats totalSamplesReceived and concealedSamples by Steve Anton · 7 years ago
  21. 0c3ca75 Replacing NetEq discard rate with secondary discarded rate. by minyue-webrtc · 7 years ago
  22. 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
  23. a9cc40b Allow an external audio processing module to be used in WebRTC by peah · 7 years ago
  24. 0f15f92 Introduce RtpStreamReceiverInterface and RtpStreamReceiverControllerInterface. by nisse · 7 years ago
  25. 37e99fd Move AudioDecoder and AudioDecoderFactory mocks to webrtc/test/ by kwiberg · 8 years ago
  26. fdbfdc9 Let PacketRouter separate send and receive modules. by nisse · 8 years ago
  27. 1c07c70 Reland "WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType" by kwiberg · 8 years ago
  28. 670a7f3 Revert of WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType (patchset #13 id:260001 of https://codereview.webrtc.org/2686043006/ ) by kwiberg · 8 years ago
  29. 1724cfb WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType by kwiberg · 8 years ago
  30. 922246a Replace NULL with nullptr or null in webrtc/audio/ and common_audio/. by deadbeef · 8 years ago
  31. 657bab2 Replace AudioReceiveStream::DeliverRtp with OnRtpPacket. by nisse · 8 years ago
  32. 4709e89 Move RemoteBitrateEstimator::RemoveStream calls from receive streams to Call. by nisse · 8 years ago
  33. bd9a77f Remove most of the remaining calls to VoE APIs from Audio[Send|Receive]Stream. by solenberg · 8 years ago
  34. d44ce05 Reland of Always call RemoteBitrateEstimator::IncomingPacket from Call. (patchset #1 id:1 of https://codereview.webrtc.org/2668973003/ ) by nisse · 8 years ago
  35. 14245cc Revert of Always call RemoteBitrateEstimator::IncomingPacket from Call. (patchset #9 id:160001 of https://codereview.webrtc.org/2659563002/ ) by nisse · 8 years ago
  36. 6d4dd59 Always call RemoteBitrateEstimator::IncomingPacket from Call. by nisse · 8 years ago
  37. 0245da0 Move ownership of PacketRouter from CongestionController to Call. by nisse · 8 years ago
  38. 04c0722 Replace AudioConferenceMixer with AudioMixer. by aleloi · 8 years ago
  39. 10111bc Passed AudioMixer to AudioState::Config. by aleloi · 8 years ago
  40. dd31071 Added an empty AudioTransportProxy to AudioState. by aleloi · 8 years ago
  41. 7602aab Remove usage of VoEBase::AssociateSendChannel() from WVoMC. by solenberg · 8 years ago
  42. b521aa7 Clean up abs-send-time for audio. by stefan · 8 years ago
  43. 2d81eb3 Fix BWE simulations so that it uses the delay based BWE. by terelius · 8 years ago
  44. 18e0b67 Restarting channel when swapping AudioReceiveStreams in WebrtcVoE. by aleloi · 8 years ago
  45. ac9f876 Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/ by kwiberg · 8 years ago
  46. 77eab70 Enable the -Wundef warning for clang by kwiberg · 8 years ago
  47. 6348978 Add new decoding statistics for muted output by henrik.lundin · 8 years ago
  48. 14d5dbe Reland of "Move RtcEventLog object from inside VoiceEngine to Call.", "Fix to make the start/stop functions for the Rtc Eventlog non-virtual." and "Fix for RtcEventLog ObjC interface" by ivoc · 8 years ago
  49. 9e03c3b Revert of Move RtcEventLog object from inside VoiceEngine to Call. (patchset #16 id:420001 of https://codereview.webrtc.org/1748403002/ ) by ivoc · 8 years ago
  50. 1895526 Move RtcEventLog object from inside VoiceEngine to Call. by Ivo Creusen · 8 years ago
  51. 2169d8b Reland of move audio/video distinction for probe packets. (patchset #1 id:1 of https://codereview.webrtc.org/2086633002/ ) by pbos · 8 years ago
  52. 17bde8c Revert of Remove audio/video distinction for probe packets. (patchset #2 id:20001 of https://codereview.webrtc.org/2061193002/ ) by honghaiz · 8 years ago
  53. a7d88d3 Remove audio/video distinction for probe packets. by Peter Boström · 8 years ago
  54. 217fb66 Add AudioReceiveStream::SetGain() method and use that in WVoMC::SetOutputVolume(). by solenberg · 8 years ago
  55. 8189b02 Configure VoE NACK through AudioReceiveStream::Config, for receive streams. Also minor refactoring of WVoE unit test. by solenberg · 8 years ago
  56. 0208322 GN: Add video_engine_tests by Peter Boström · 8 years ago
  57. 29b1a8d Moved creation of AudioDecoderFactory to inside PeerConnectionFactory. by ossu · 8 years ago
  58. 6f8d686 Remove use of RtpHeaderExtension and clean up by isheriff · 8 years ago
  59. ec81bcd Remove SendPacer from ViEEncoder and make sure SendPacer starts at a valid bitrate by perkj · 8 years ago
  60. e30c272 Revert "Reland of Remove SendPacer from ViEEncoder by perkj · 8 years ago
  61. 28a4456 Revert "Revert of Remove SendPacer from ViEEncoder (patchset #13 id:240001 of https://codereview.webrtc.org/1917793002/ )" by Per · 8 years ago
  62. 825eb58 Revert of Remove SendPacer from ViEEncoder (patchset #13 id:240001 of https://codereview.webrtc.org/1917793002/ ) by perkj · 8 years ago
  63. 857c5cc Remove SendPacer from ViEEncoder by perkj · 8 years ago
  64. 3d7db26 Switch voice transport to use Call and Stream instead of VoENetwork. by mflodman · 8 years ago
  65. 80e1207 Move congestion controller to a separate module. by Stefan Holmer · 9 years ago
  66. 789ba92 Simplify CongestionController. by Stefan Holmer · 9 years ago
  67. 58c664c Clean up of CongestionController. by Stefan Holmer · 9 years ago
  68. ba4c0e4 Add send-side BWE to WebRtcVoiceEngine under a finch experiment. by stefan · 9 years ago
  69. bba9dec Use separate rtp module lists for send and receive in PacketRouter. by stefan · 9 years ago
  70. 3313ec9 Enable transport seq num extension on receive channel to suppress log warning. by stefan · 9 years ago
  71. 3842c5c Wire-up BWE feedback for audio receive streams. by Stefan Holmer · 9 years ago
  72. 25702cb Misc. small cleanups. by pkasting · 9 years ago
  73. ea07373 Enable cpplint for webrtc/audio and webrtc/call, and fix all uncovered cpplint errors. by Fredrik Solenberg · 9 years ago
  74. 358057b Use ChannelProxy for most calls on voe::Channel in Audio[Receive|Send]Stream. by solenberg · 9 years ago
  75. 1372508 Open backdoor in VoiceEngineImpl to get at the actual voe::Channel objects from an ID. by solenberg · 9 years ago
  76. 7add058 Move some receive stream configuration into webrtc::AudioReceiveStream. by solenberg · 9 years ago
  77. 8b85de2 Converted a bunch of error checking in Audio[Receive|Send]Stream to RTC_CHECKs instead. They should never fail. by solenberg · 9 years ago
  78. 3a94154 Move some send stream configuration into webrtc::AudioSendStream. by solenberg · 9 years ago
  79. 566ef24 Move VoiceEngineObserver into AudioSendStream so that we detect typing noises and return properly in GetStats(). by solenberg · 9 years ago
  80. 0ccae13 Changed FakeVoiceEngine into a MockVoiceEngine. by Fredrik Solenberg · 9 years ago
  81. 85a0496 Implement AudioSendStream::GetStats(). by solenberg · 9 years ago
  82. 4f4ec0a Re-Land: Implement AudioReceiveStream::GetStats(). by Fredrik Solenberg · 9 years ago
  83. 43e83d4 Revert of Implement AudioReceiveStream::GetStats(). (patchset #19 id:360001 of https://codereview.webrtc.org/1390753002/ ) by solenberg · 9 years ago
  84. a457752 Implement AudioReceiveStream::GetStats(). by Fredrik Solenberg · 9 years ago
  85. 5c389d3 Split webrtc/video into webrtc/{audio,call,video}. by Peter Boström · 9 years ago[Renamed (98%) from webrtc/video/audio_receive_stream_unittest.cc]
  86. 8bffba7 Fix BWE bug where audio has timestamps in us. by Stefan Holmer · 9 years ago