1. 443f9a9 Wiring discard rate of audio FEC/RED packets up to StatsReport. by minyue-webrtc · 7 years ago
  2. 94ac82f Add stats totalSamplesReceived and concealedSamples by Steve Anton · 7 years ago
  3. ea04cc0 Make ~webrtc::AudioSendStream public, and s/config()/GetConfig(), as well as make public. by eladalon · 7 years ago
  4. 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
  5. 76de83e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 8 years ago
  6. bc32410 Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 8 years ago
  7. 60154fd Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 8 years ago
  8. 04ef6b0 Introduce RtpStreamReceiverInterface and RtpStreamReceiverControllerInterface. by nisse · 8 years ago
  9. 69579d7 Reland of Implemented the GetSources() in native code. (patchset #1 id:1 of https://codereview.webrtc.org/2809613002/ ) by hbos · 8 years ago
  10. fee642e Revert of Implemented the GetSources() in native code. (patchset #11 id:510001 of https://codereview.webrtc.org/2770233003/ ) by olka · 8 years ago
  11. b5eeeee Added the GetSources() to the RtpReceiverInterface and implemented by zhihuang · 8 years ago
  12. 5e9233b Let PacketRouter separate send and receive modules. by nisse · 8 years ago
  13. afd1255 Reland "WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType" by kwiberg · 8 years ago
  14. 52b9df6 Revert of WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType (patchset #13 id:260001 of https://codereview.webrtc.org/2686043006/ ) by kwiberg · 8 years ago
  15. 135be28 WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType by kwiberg · 8 years ago
  16. bf77153 Remove usage of VoEVolumeControl from WVoE and Audio[Send|Receive]Stream. by solenberg · 8 years ago
  17. 9617a87 Replace NULL with nullptr or null in webrtc/audio/ and common_audio/. by deadbeef · 8 years ago
  18. 95a13b4 Replace AudioReceiveStream::DeliverRtp with OnRtpPacket. by nisse · 8 years ago
  19. ac16f1b Remove VoEVideoSync interface. by solenberg · 8 years ago
  20. 49cdd35 Move RemoteBitrateEstimator::RemoveStream calls from receive streams to Call. by nisse · 8 years ago
  21. f65fb4c Remove unnecessary RTPHeaderParser, following https://codereview.webrtc.org/2659563002/ by solenberg · 8 years ago
  22. 6045993 Remove most of the remaining calls to VoE APIs from Audio[Send|Receive]Stream. by solenberg · 8 years ago
  23. a0add70 Reland of Always call RemoteBitrateEstimator::IncomingPacket from Call. (patchset #1 id:1 of https://codereview.webrtc.org/2668973003/ ) by nisse · 8 years ago
  24. 5cacd3d Revert of Always call RemoteBitrateEstimator::IncomingPacket from Call. (patchset #9 id:160001 of https://codereview.webrtc.org/2659563002/ ) by nisse · 8 years ago
  25. 40ab430 Always call RemoteBitrateEstimator::IncomingPacket from Call. by nisse · 8 years ago
  26. 9551ae0 Stop using VoEVideoSync in Call/VideoReceiveStream. by solenberg · 8 years ago
  27. bc2b639 Pass SdpAudioFormat through Channel, without converting to CodecInst by kwiberg · 8 years ago
  28. 800a8f1 Support external audio mixer in webrtc 2. by gyzhou · 8 years ago
  29. 818a3ee Revert of Support external audio mixer. (patchset #5 id:140001 of https://codereview.webrtc.org/2539213003/ ) by gyzhou · 8 years ago
  30. 0808545 Support external audio mixer in webrtc. by gyzhou · 8 years ago
  31. 61e42da Move ownership of PacketRouter from CongestionController to Call. by nisse · 8 years ago
  32. 441ecb3 Replace AudioConferenceMixer with AudioMixer. by aleloi · 8 years ago
  33. 9a1d49f Expose RtpCodecParameters to VoiceMediaInfo stats. by hbos · 8 years ago
  34. ac14ca2 Remove Absolute Send Time from list of supported header extensions for audio streams. by solenberg · 8 years ago
  35. dbe2c77 Remove usage of VoEBase::AssociateSendChannel() from WVoMC. by solenberg · 8 years ago
  36. def0180 Fix crash when registering abs-send-time to AudioSend/ReceiveStream. by stefan · 8 years ago
  37. e9523f0 Clean up abs-send-time for audio. by stefan · 8 years ago
  38. b87bc5d Add a NeededFrequency() method to the AudioMixer::Source interface. by aleloi · 8 years ago
  39. fbd0246 Move audio frame memory handling inside AudioMixer. by aleloi · 8 years ago
  40. b4ac6b0 Made AudioReceiveStream a mixer participant. by aleloi · 8 years ago
  41. d0dcb3b Restarting channel when swapping AudioReceiveStreams in WebrtcVoE. by aleloi · 8 years ago
  42. 4450c27 Add new decoding statistics for muted output by henrik.lundin · 8 years ago
  43. cf82062 Move webrtc/audio_*.h to webrtc/api/call by kjellander · 8 years ago
  44. 762f0f9 Removed calls to VoE::SetPlayout() from WebRTCVoiceEngine. by aleloi · 8 years ago
  45. 23ea12e 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
  46. 822f09e 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
  47. 1e2f1e5 Move RtcEventLog object from inside VoiceEngine to Call. by Ivo Creusen · 9 years ago
  48. 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
  49. 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
  50. b0d0745 Remove audio/video distinction for probe packets. by Peter Boström · 9 years ago
  51. 784336a Add AudioReceiveStream::SetGain() method and use that in WVoMC::SetOutputVolume(). by solenberg · 9 years ago
  52. ff1d51a Configure VoE NACK through AudioReceiveStream::Config, for receive streams. Also minor refactoring of WVoE unit test. by solenberg · 9 years ago
  53. e840777 Moved creation of AudioDecoderFactory to inside PeerConnectionFactory. by ossu · 9 years ago
  54. c4921f4 Remove use of RtpHeaderExtension and clean up by isheriff · 9 years ago
  55. 7738985 Delete all use of tick_util.h. by Niels Möller · 9 years ago
  56. 47a40a3 Remove webrtc/stream.h and unutilized inheritance. by pbos · 9 years ago
  57. 823f908 Switch voice transport to use Call and Stream instead of VoENetwork. by mflodman · 9 years ago
  58. caa8176 Reland "Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies." by kjellander@webrtc.org · 9 years ago
  59. 52cf08c Revert of Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies. (patchset #2 id:40001 of https://codereview.webrtc.org/1737593002/ ) by kjellander · 9 years ago
  60. 0bb951e Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies. by kjellander@webrtc.org · 9 years ago
  61. abb2e3e Replace scoped_ptr with unique_ptr in webrtc/audio/ by kwiberg · 9 years ago
  62. ead3cf2 Move congestion controller to a separate module. by Stefan Holmer · 9 years ago
  63. 484e0cd Replace scoped_ptr with unique_ptr in webrtc/voice_engine/ by kwiberg · 9 years ago
  64. 74c29e2 Add send-side BWE to WebRtcVoiceEngine under a finch experiment. by stefan · 9 years ago
  65. 3dad57b Use separate rtp module lists for send and receive in PacketRouter. by stefan · 9 years ago
  66. 9b91023 Enable transport seq num extension on receive channel to suppress log warning. by stefan · 9 years ago
  67. e3f40fb Revert of Storing raw audio sink for default audio track. (patchset #7 id:120001 of https://codereview.chromium.org/1551813002/ ) by deadbeef · 9 years ago
  68. cf354ef Storing raw audio sink for default audio track. by deadbeef · 9 years ago
  69. 87f3db7 Wire-up BWE feedback for audio receive streams. by Stefan Holmer · 9 years ago
  70. 387e90b Support for unmixed remote audio into tracks. by Tommi · 9 years ago
  71. 21ca0a4 Use ChannelProxy for most calls on voe::Channel in Audio[Receive|Send]Stream. by solenberg · 9 years ago
  72. 5bbf7f9 Open backdoor in VoiceEngineImpl to get at the actual voe::Channel objects from an ID. by solenberg · 9 years ago
  73. 775e132 Move some receive stream configuration into webrtc::AudioReceiveStream. by solenberg · 9 years ago
  74. 03d4810 Converted a bunch of error checking in Audio[Receive|Send]Stream to RTC_CHECKs instead. They should never fail. by solenberg · 9 years ago
  75. b0f22c5 Move VoiceEngineObserver into AudioSendStream so that we detect typing noises and return properly in GetStats(). by solenberg · 9 years ago
  76. 78f65d0 system_wrappers: rename interface -> include by Henrik Kjellander · 9 years ago
  77. 1c28f5c Implement AudioSendStream::GetStats(). by solenberg · 9 years ago
  78. 10762d3 Re-Land: Implement AudioReceiveStream::GetStats(). by Fredrik Solenberg · 9 years ago
  79. 0e9f679 Revert of Implement AudioReceiveStream::GetStats(). (patchset #19 id:360001 of https://codereview.webrtc.org/1390753002/ ) by solenberg · 9 years ago
  80. bbb922f Implement AudioReceiveStream::GetStats(). by Fredrik Solenberg · 9 years ago
  81. f863304 Log Call {audio, video} stream deletions. by pbos · 9 years ago
  82. bf9f73c Split webrtc/video into webrtc/{audio,call,video}. by Peter Boström · 9 years ago[Renamed (98%) from video/audio_receive_stream.cc]
  83. 4f5e451 Fix BWE bug where audio has timestamps in us. by Stefan Holmer · 9 years ago
  84. 5c075c8 Add RTC_ prefix to (D)CHECKs and related macros. by henrikg · 9 years ago
  85. 30bf778 Wire up PacketTime to ReceiveStreams. by stefan · 9 years ago
  86. 374a570 Add support for transport wide sequence numbers by sprang · 9 years ago
  87. 95b9fbb Control combined_audio_video_bwe with config bool. by pbos · 10 years ago
  88. 2fb88e4 Define Stream base classes by Jelena Marusic · 10 years ago
  89. 495b350 Base A/V synchronization on sync_labels. by pbos · 10 years ago
  90. 9dca5d7 Only use paced packets for estimating bitrate probes. by Stefan Holmer · 10 years ago
  91. ee9b72e VoE2 API draft by Fredrik Solenberg · 10 years ago
  92. ad86786 Add AudioReceiveStream to Call API. by Fredrik Solenberg · 10 years ago