Gitiles
Code Review Sign In
gerrit-public.fairphone.software / platform / external / webrtc / 8fdcac3f06ae09fe17bbed02037ef136f086cc00 / audio / audio_receive_stream.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. 848d6d3 Change Channel::GetRtpRtcp to return only RtpRtcp, not RtpReceiver. by Niels Möller · 6 years ago
  4. 665174f Reformat the WebRTC code base by Yves Gerey · 6 years ago
  5. b9b146c Replace rtc::Optional with absl::optional in audio, call and video by Danil Chapovalov · 6 years ago
  6. f782492 Delete RtpFeedback. The ssrc for a receive stream should be known at by Niels Möller · 6 years ago
  7. 5f22365 Remove unnecessary proxy+lock code around RtcpRttStats pointer by Tommi · 6 years ago
  8. 0812634 Pass a real audio codec pair ID to decoders that we create by Karl Wiberg · 6 years ago
  9. 881f168 Make SimpleStringBuilder into a non-template by Karl Wiberg · 7 years ago
  10. fef0500 Adding a new string utility class: SimpleStringBuilder. by Tommi · 7 years ago
  11. f120cba Delete AudioMonitor and related code. by Niels Möller · 7 years ago
  12. 24ea822 Remove logging in audio/* from release builds. by Jonas Olsson · 7 years ago
  13. a8b7c7f Move remaining traces of VoiceEngine by Fredrik Solenberg · 7 years ago
  14. 90ea504 Delete Channel::OnRecoveredPacket. by Niels Möller · 7 years ago
  15. 8f5787a Move ownership of voe::Channel into Audio[Receive|Send]Stream. by Fredrik Solenberg · 7 years ago
  16. 3b903d0 Reconfigure, not reconstruct, AudioReceiveStreams. by Fredrik Solenberg · 7 years ago
  17. d524751 Replace VoEBase::[Start/Stop]Playout(). by Fredrik Solenberg · 7 years ago
  18. 2707fb2 Optional: Use nullopt and implicit construction in /audio by Oskar Sundbom · 7 years ago
  19. 675513b Stop using LOG macros in favor of RTC_ prefixed macros. by Mirko Bonadei · 7 years ago
  20. c3fa8e1 New method RtpReceiver::GetLatestTimestamps. by Niels Möller · 7 years ago
  21. b0a0207 Added RTCMediaStreamTrackStats.jitterBufferDelay for audio by Gustaf Ullberg · 7 years ago
  22. 1c239d4 Remove voe::Statistics. by solenberg · 7 years ago
  23. 2397b9a Remove voe::OutputMixer and AudioConferenceMixer. by solenberg · 7 years ago
  24. 9a2e906 Added RTCMediaStreamTrackStats.concealmentEvents by Gustaf Ullberg · 7 years ago
  25. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  26. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/audio/audio_receive_stream.cc]
  27. 0e320ec Wiring discard rate of audio FEC/RED packets up to StatsReport. by minyue-webrtc · 7 years ago
  28. 2dbc69f Add stats totalSamplesReceived and concealedSamples by Steve Anton · 7 years ago
  29. abbc430 Make ~webrtc::AudioSendStream public, and s/config()/GetConfig(), as well as make public. by eladalon · 7 years ago
  30. 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
  31. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 7 years ago
  32. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 7 years ago
  33. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 7 years ago
  34. 0f15f92 Introduce RtpStreamReceiverInterface and RtpStreamReceiverControllerInterface. by nisse · 7 years ago
  35. 8d609f6 Reland of Implemented the GetSources() in native code. (patchset #1 id:1 of https://codereview.webrtc.org/2809613002/ ) by hbos · 7 years ago
  36. fbcc5cb Revert of Implemented the GetSources() in native code. (patchset #11 id:510001 of https://codereview.webrtc.org/2770233003/ ) by olka · 7 years ago
  37. 292084c Added the GetSources() to the RtpReceiverInterface and implemented by zhihuang · 7 years ago
  38. fdbfdc9 Let PacketRouter separate send and receive modules. by nisse · 7 years ago
  39. 1c07c70 Reland "WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType" by kwiberg · 7 years ago
  40. 670a7f3 Revert of WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType (patchset #13 id:260001 of https://codereview.webrtc.org/2686043006/ ) by kwiberg · 7 years ago
  41. 1724cfb WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType by kwiberg · 7 years ago
  42. 796b8f9 Remove usage of VoEVolumeControl from WVoE and Audio[Send|Receive]Stream. by solenberg · 8 years ago
  43. 922246a Replace NULL with nullptr or null in webrtc/audio/ and common_audio/. by deadbeef · 8 years ago
  44. 657bab2 Replace AudioReceiveStream::DeliverRtp with OnRtpPacket. by nisse · 8 years ago
  45. 08b19df Remove VoEVideoSync interface. by solenberg · 8 years ago
  46. 4709e89 Move RemoteBitrateEstimator::RemoveStream calls from receive streams to Call. by nisse · 8 years ago
  47. 6b34124 Remove unnecessary RTPHeaderParser, following https://codereview.webrtc.org/2659563002/ by solenberg · 8 years ago
  48. bd9a77f Remove most of the remaining calls to VoE APIs from Audio[Send|Receive]Stream. by solenberg · 8 years ago
  49. d44ce05 Reland of Always call RemoteBitrateEstimator::IncomingPacket from Call. (patchset #1 id:1 of https://codereview.webrtc.org/2668973003/ ) by nisse · 8 years ago
  50. 14245cc Revert of Always call RemoteBitrateEstimator::IncomingPacket from Call. (patchset #9 id:160001 of https://codereview.webrtc.org/2659563002/ ) by nisse · 8 years ago
  51. 6d4dd59 Always call RemoteBitrateEstimator::IncomingPacket from Call. by nisse · 8 years ago
  52. 3ebbcb5 Stop using VoEVideoSync in Call/VideoReceiveStream. by solenberg · 8 years ago
  53. d32bf75 Pass SdpAudioFormat through Channel, without converting to CodecInst by kwiberg · 8 years ago
  54. 95aa964 Support external audio mixer in webrtc 2. by gyzhou · 8 years ago
  55. 39ce11f Revert of Support external audio mixer. (patchset #5 id:140001 of https://codereview.webrtc.org/2539213003/ ) by gyzhou · 8 years ago
  56. f6bcac5 Support external audio mixer in webrtc. by gyzhou · 8 years ago
  57. 0245da0 Move ownership of PacketRouter from CongestionController to Call. by nisse · 8 years ago
  58. 04c0722 Replace AudioConferenceMixer with AudioMixer. by aleloi · 8 years ago
  59. 1acfbd2 Expose RtpCodecParameters to VoiceMediaInfo stats. by hbos · 8 years ago
  60. d4adce4 Remove Absolute Send Time from list of supported header extensions for audio streams. by solenberg · 8 years ago
  61. 7602aab Remove usage of VoEBase::AssociateSendChannel() from WVoMC. by solenberg · 8 years ago
  62. 572ae12 Fix crash when registering abs-send-time to AudioSend/ReceiveStream. by stefan · 8 years ago
  63. b521aa7 Clean up abs-send-time for audio. by stefan · 8 years ago
  64. 051f678 Add a NeededFrequency() method to the AudioMixer::Source interface. by aleloi · 8 years ago
  65. 6c27849 Move audio frame memory handling inside AudioMixer. by aleloi · 8 years ago
  66. aed581a Made AudioReceiveStream a mixer participant. by aleloi · 8 years ago
  67. 18e0b67 Restarting channel when swapping AudioReceiveStreams in WebrtcVoE. by aleloi · 8 years ago
  68. 6348978 Add new decoding statistics for muted output by henrik.lundin · 8 years ago
  69. a69d973 Move webrtc/audio_*.h to webrtc/api/call by kjellander · 8 years ago
  70. 84ef615 Removed calls to VoE::SetPlayout() from WebRTCVoiceEngine. by aleloi · 8 years ago
  71. 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
  72. 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
  73. 1895526 Move RtcEventLog object from inside VoiceEngine to Call. by Ivo Creusen · 8 years ago
  74. 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
  75. 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
  76. a7d88d3 Remove audio/video distinction for probe packets. by Peter Boström · 8 years ago
  77. 217fb66 Add AudioReceiveStream::SetGain() method and use that in WVoMC::SetOutputVolume(). by solenberg · 8 years ago
  78. 8189b02 Configure VoE NACK through AudioReceiveStream::Config, for receive streams. Also minor refactoring of WVoE unit test. by solenberg · 8 years ago
  79. 29b1a8d Moved creation of AudioDecoderFactory to inside PeerConnectionFactory. by ossu · 8 years ago
  80. 6f8d686 Remove use of RtpHeaderExtension and clean up by isheriff · 8 years ago
  81. d28db7f Delete all use of tick_util.h. by Niels Möller · 8 years ago
  82. 1ba8d39 Remove webrtc/stream.h and unutilized inheritance. by pbos · 8 years ago
  83. 3d7db26 Switch voice transport to use Call and Stream instead of VoENetwork. by mflodman · 8 years ago
  84. 7ffeab5 Reland "Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies." by kjellander@webrtc.org · 9 years ago
  85. 7324eb9 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
  86. 99b345c Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies. by kjellander@webrtc.org · 9 years ago
  87. fffa42b Replace scoped_ptr with unique_ptr in webrtc/audio/ by kwiberg · 9 years ago
  88. 80e1207 Move congestion controller to a separate module. by Stefan Holmer · 9 years ago
  89. b7f89d6 Replace scoped_ptr with unique_ptr in webrtc/voice_engine/ by kwiberg · 9 years ago
  90. ba4c0e4 Add send-side BWE to WebRtcVoiceEngine under a finch experiment. by stefan · 9 years ago
  91. bba9dec Use separate rtp module lists for send and receive in PacketRouter. by stefan · 9 years ago
  92. 3313ec9 Enable transport seq num extension on receive channel to suppress log warning. by stefan · 9 years ago
  93. 2d110be 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
  94. e591f93 Storing raw audio sink for default audio track. by deadbeef · 9 years ago
  95. 3842c5c Wire-up BWE feedback for audio receive streams. by Stefan Holmer · 9 years ago
  96. f888bb5 Support for unmixed remote audio into tracks. by Tommi · 9 years ago
  97. 358057b Use ChannelProxy for most calls on voe::Channel in Audio[Receive|Send]Stream. by solenberg · 9 years ago
  98. 1372508 Open backdoor in VoiceEngineImpl to get at the actual voe::Channel objects from an ID. by solenberg · 9 years ago
  99. 7add058 Move some receive stream configuration into webrtc::AudioReceiveStream. by solenberg · 9 years ago
  100. 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