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gerrit.openfyde.cn / webrtc.googlesource.com / src / 603171647318927390e10c04d918d4ed7075717e / audio / audio_receive_stream_unittest.cc
  1. fcf79cc Add estimatedPlayoutTimestamp to RTCInboundRTPStreamStats. by Åsa Persson · 6 years ago
  2. ac0a4cb Reland "Fix GetStats bytesSent/Received, wireup headerBytesSent/Received" by Niels Möller · 6 years ago
  3. ef0627f Revert "Fix GetStats bytesSent/Received, wireup headerBytesSent/Received" by Mirko Bonadei · 6 years ago
  4. fbde32e Fix GetStats bytesSent/Received, wireup headerBytesSent/Received by Niels Möller · 6 years ago
  5. 65024d9 Remove clock drift metric from NetEq. by Jakob Ivarsson · 6 years ago
  6. 224c69d Delete ext_seqnum member from VoiceSenderInfo and VoiceReceiverInfo by Niels Möller · 6 years ago
  7. 70efdde Set local ssrc at construction of Rtp module by Erik Språng · 6 years ago
  8. 5b5d97c Reland of "Reporting of decoding_codec_plc events"" by Alex Narest · 6 years ago
  9. bedb7a8 Revert "Reporting of decoding_codec_plc events" by Mirko Bonadei · 6 years ago
  10. 0a88ea0 Reporting of decoding_codec_plc events by Alex Narest · 6 years ago
  11. a4d8737 Format almost everything. by Jonas Olsson · 6 years ago
  12. 3472b9a Delete RTCInboundRTPStreamStats::fraction_lost by Niels Möller · 6 years ago
  13. 4f08faa Introduce MediaTransportConfig by Anton Sukhanov · 6 years ago
  14. 8d8ffdb Expose new audio stats on the API by Ivo Creusen · 6 years ago
  15. 6a489f2 Fully qualify googletest symbols. by Mirko Bonadei · 6 years ago
  16. 17b050f Fixes ClangTidy errors in audio/ by Benjamin Wright · 6 years ago
  17. 9ffb5df Removes unused mock_bitrate_controller. by Sebastian Jansson · 6 years ago
  18. 8fb1a6a Delete a few return values from audio streams and video send streams. by Niels Möller · 6 years ago
  19. 977b335 Injecting Clock into audio streams. by Sebastian Jansson · 6 years ago
  20. 5c2f1f0 Add some missing includes and dependencies. by Bjorn Terelius · 7 years ago
  21. 0acffb5 Expose `jitterBufferEmittedCount` in addition to the existing `jitterBufferDelay` for `getStats()`. by Chen Xing · 7 years ago
  22. f693bfa Remove CodecInst pt.2 by Fredrik Solenberg · 7 years ago
  23. 349ade3 Delete class ChannelReceiveProxy. by Niels Möller · 7 years ago
  24. 7d76a31 Use MediaTransportInterface, for audio streams. by Niels Möller · 7 years ago
  25. 78410ad Fixes use after free error when setting a new FrameEncryptor on ChannelSend. by Benjamin Wright · 7 years ago
  26. ae4237e Set ChannelReceive transport at construction time. by Niels Möller · 7 years ago
  27. 530ead4 Split voe::Channel into ChannelSend and ChannelReceive by Niels Möller · 7 years ago
  28. b222f49 Split ChannelProxy into send and receive classes. by Niels Möller · 7 years ago
  29. 30b4839 Refactor voe::Channel to not use RtpReceiver. by Niels Möller · 7 years ago
  30. fa4e185 Delete class voe::RtcEventLogProxy by Niels Möller · 7 years ago
  31. 7008287 Delete struct webrtc::PacketTime. by Niels Möller · 7 years ago
  32. 665174f Reformat the WebRTC code base by Yves Gerey · 7 years ago
  33. f782492 Delete RtpFeedback. The ssrc for a receive stream should be known at by Niels Möller · 7 years ago
  34. a8b7c7f Move remaining traces of VoiceEngine by Fredrik Solenberg · 8 years ago
  35. 90ea504 Delete Channel::OnRecoveredPacket. by Niels Möller · 8 years ago
  36. 8f5787a Move ownership of voe::Channel into Audio[Receive|Send]Stream. by Fredrik Solenberg · 8 years ago
  37. 3b903d0 Reconfigure, not reconstruct, AudioReceiveStreams. by Fredrik Solenberg · 8 years ago
  38. d524751 Replace VoEBase::[Start/Stop]Playout(). by Fredrik Solenberg · 8 years ago
  39. 2a87797 Remove voe::TransmitMixer by Fredrik Solenberg · 8 years ago
  40. d319534 Move ADM initialization into WebRtcVoiceEngine by Fredrik Solenberg · 8 years ago
  41. b0a0207 Added RTCMediaStreamTrackStats.jitterBufferDelay for audio by Gustaf Ullberg · 8 years ago
  42. 1c239d4 Remove voe::Statistics. by solenberg · 8 years ago
  43. fc3a2e3 Remove the VoiceEngineObserver callback interface. by solenberg · 8 years ago
  44. 9a2e906 Added RTCMediaStreamTrackStats.concealmentEvents by Gustaf Ullberg · 8 years ago
  45. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 8 years ago
  46. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 8 years ago[Renamed from webrtc/audio/audio_receive_stream_unittest.cc]
  47. 0e320ec Wiring discard rate of audio FEC/RED packets up to StatsReport. by minyue-webrtc · 8 years ago
  48. 2dbc69f Add stats totalSamplesReceived and concealedSamples by Steve Anton · 8 years ago
  49. 0c3ca75 Replacing NetEq discard rate with secondary discarded rate. by minyue-webrtc · 8 years ago
  50. 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 · 8 years ago
  51. a9cc40b Allow an external audio processing module to be used in WebRTC by peah · 8 years ago
  52. 0f15f92 Introduce RtpStreamReceiverInterface and RtpStreamReceiverControllerInterface. by nisse · 8 years ago
  53. 37e99fd Move AudioDecoder and AudioDecoderFactory mocks to webrtc/test/ by kwiberg · 8 years ago
  54. fdbfdc9 Let PacketRouter separate send and receive modules. by nisse · 8 years ago
  55. 1c07c70 Reland "WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType" by kwiberg · 8 years ago
  56. 670a7f3 Revert of WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType (patchset #13 id:260001 of https://codereview.webrtc.org/2686043006/ ) by kwiberg · 8 years ago
  57. 1724cfb WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType by kwiberg · 8 years ago
  58. 922246a Replace NULL with nullptr or null in webrtc/audio/ and common_audio/. by deadbeef · 8 years ago
  59. 657bab2 Replace AudioReceiveStream::DeliverRtp with OnRtpPacket. by nisse · 8 years ago
  60. 4709e89 Move RemoteBitrateEstimator::RemoveStream calls from receive streams to Call. by nisse · 8 years ago
  61. bd9a77f Remove most of the remaining calls to VoE APIs from Audio[Send|Receive]Stream. by solenberg · 8 years ago
  62. d44ce05 Reland of Always call RemoteBitrateEstimator::IncomingPacket from Call. (patchset #1 id:1 of https://codereview.webrtc.org/2668973003/ ) by nisse · 8 years ago
  63. 14245cc Revert of Always call RemoteBitrateEstimator::IncomingPacket from Call. (patchset #9 id:160001 of https://codereview.webrtc.org/2659563002/ ) by nisse · 8 years ago
  64. 6d4dd59 Always call RemoteBitrateEstimator::IncomingPacket from Call. by nisse · 8 years ago
  65. 0245da0 Move ownership of PacketRouter from CongestionController to Call. by nisse · 9 years ago
  66. 04c0722 Replace AudioConferenceMixer with AudioMixer. by aleloi · 9 years ago
  67. 10111bc Passed AudioMixer to AudioState::Config. by aleloi · 9 years ago
  68. dd31071 Added an empty AudioTransportProxy to AudioState. by aleloi · 9 years ago
  69. 7602aab Remove usage of VoEBase::AssociateSendChannel() from WVoMC. by solenberg · 9 years ago
  70. b521aa7 Clean up abs-send-time for audio. by stefan · 9 years ago
  71. 2d81eb3 Fix BWE simulations so that it uses the delay based BWE. by terelius · 9 years ago
  72. 18e0b67 Restarting channel when swapping AudioReceiveStreams in WebrtcVoE. by aleloi · 9 years ago
  73. ac9f876 Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/ by kwiberg · 9 years ago
  74. 77eab70 Enable the -Wundef warning for clang by kwiberg · 9 years ago
  75. 6348978 Add new decoding statistics for muted output by henrik.lundin · 9 years ago
  76. 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 · 9 years ago
  77. 9e03c3b Revert of Move RtcEventLog object from inside VoiceEngine to Call. (patchset #16 id:420001 of https://codereview.webrtc.org/1748403002/ ) by ivoc · 9 years ago
  78. 1895526 Move RtcEventLog object from inside VoiceEngine to Call. by Ivo Creusen · 9 years ago
  79. 2169d8b Reland of move audio/video distinction for probe packets. (patchset #1 id:1 of https://codereview.webrtc.org/2086633002/ ) by pbos · 9 years ago
  80. 17bde8c Revert of Remove audio/video distinction for probe packets. (patchset #2 id:20001 of https://codereview.webrtc.org/2061193002/ ) by honghaiz · 9 years ago
  81. a7d88d3 Remove audio/video distinction for probe packets. by Peter Boström · 9 years ago
  82. 217fb66 Add AudioReceiveStream::SetGain() method and use that in WVoMC::SetOutputVolume(). by solenberg · 9 years ago
  83. 8189b02 Configure VoE NACK through AudioReceiveStream::Config, for receive streams. Also minor refactoring of WVoE unit test. by solenberg · 9 years ago
  84. 0208322 GN: Add video_engine_tests by Peter Boström · 9 years ago
  85. 29b1a8d Moved creation of AudioDecoderFactory to inside PeerConnectionFactory. by ossu · 9 years ago
  86. 6f8d686 Remove use of RtpHeaderExtension and clean up by isheriff · 9 years ago
  87. ec81bcd Remove SendPacer from ViEEncoder and make sure SendPacer starts at a valid bitrate by perkj · 9 years ago
  88. e30c272 Revert "Reland of Remove SendPacer from ViEEncoder by perkj · 9 years ago
  89. 28a4456 Revert "Revert of Remove SendPacer from ViEEncoder (patchset #13 id:240001 of https://codereview.webrtc.org/1917793002/ )" by Per · 9 years ago
  90. 825eb58 Revert of Remove SendPacer from ViEEncoder (patchset #13 id:240001 of https://codereview.webrtc.org/1917793002/ ) by perkj · 9 years ago
  91. 857c5cc Remove SendPacer from ViEEncoder by perkj · 9 years ago
  92. 3d7db26 Switch voice transport to use Call and Stream instead of VoENetwork. by mflodman · 9 years ago
  93. 80e1207 Move congestion controller to a separate module. by Stefan Holmer · 9 years ago
  94. 789ba92 Simplify CongestionController. by Stefan Holmer · 9 years ago
  95. 58c664c Clean up of CongestionController. by Stefan Holmer · 9 years ago
  96. ba4c0e4 Add send-side BWE to WebRtcVoiceEngine under a finch experiment. by stefan · 9 years ago
  97. bba9dec Use separate rtp module lists for send and receive in PacketRouter. by stefan · 9 years ago
  98. 3313ec9 Enable transport seq num extension on receive channel to suppress log warning. by stefan · 9 years ago
  99. 3842c5c Wire-up BWE feedback for audio receive streams. by Stefan Holmer · 10 years ago
  100. 25702cb Misc. small cleanups. by pkasting · 10 years ago