Google Git
Sign in
webrtc / src / / e55686e9d36a585630035e274e5a842519c37d19 / audio / audio_receive_stream.cc
  1. 2397b9a Remove voe::OutputMixer and AudioConferenceMixer. by solenberg · 8 years ago
  2. 9a2e906 Added RTCMediaStreamTrackStats.concealmentEvents by Gustaf Ullberg · 8 years ago
  3. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 8 years ago
  4. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 8 years ago[Renamed from webrtc/audio/audio_receive_stream.cc]
  5. 0e320ec Wiring discard rate of audio FEC/RED packets up to StatsReport. by minyue-webrtc · 8 years ago
  6. 2dbc69f Add stats totalSamplesReceived and concealedSamples by Steve Anton · 8 years ago
  7. abbc430 Make ~webrtc::AudioSendStream public, and s/config()/GetConfig(), as well as make public. by eladalon · 8 years ago
  8. 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
  9. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 8 years ago
  10. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 8 years ago
  11. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 8 years ago
  12. 0f15f92 Introduce RtpStreamReceiverInterface and RtpStreamReceiverControllerInterface. by nisse · 8 years ago
  13. 8d609f6 Reland of Implemented the GetSources() in native code. (patchset #1 id:1 of https://codereview.webrtc.org/2809613002/ ) by hbos · 8 years ago
  14. fbcc5cb Revert of Implemented the GetSources() in native code. (patchset #11 id:510001 of https://codereview.webrtc.org/2770233003/ ) by olka · 8 years ago
  15. 292084c Added the GetSources() to the RtpReceiverInterface and implemented by zhihuang · 8 years ago
  16. fdbfdc9 Let PacketRouter separate send and receive modules. by nisse · 8 years ago
  17. 1c07c70 Reland "WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType" by kwiberg · 8 years ago
  18. 670a7f3 Revert of WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType (patchset #13 id:260001 of https://codereview.webrtc.org/2686043006/ ) by kwiberg · 8 years ago
  19. 1724cfb WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType by kwiberg · 8 years ago
  20. 796b8f9 Remove usage of VoEVolumeControl from WVoE and Audio[Send|Receive]Stream. by solenberg · 8 years ago
  21. 922246a Replace NULL with nullptr or null in webrtc/audio/ and common_audio/. by deadbeef · 8 years ago
  22. 657bab2 Replace AudioReceiveStream::DeliverRtp with OnRtpPacket. by nisse · 8 years ago
  23. 08b19df Remove VoEVideoSync interface. by solenberg · 8 years ago
  24. 4709e89 Move RemoteBitrateEstimator::RemoveStream calls from receive streams to Call. by nisse · 8 years ago
  25. 6b34124 Remove unnecessary RTPHeaderParser, following https://codereview.webrtc.org/2659563002/ by solenberg · 8 years ago
  26. bd9a77f Remove most of the remaining calls to VoE APIs from Audio[Send|Receive]Stream. by solenberg · 8 years ago
  27. d44ce05 Reland of Always call RemoteBitrateEstimator::IncomingPacket from Call. (patchset #1 id:1 of https://codereview.webrtc.org/2668973003/ ) by nisse · 8 years ago
  28. 14245cc Revert of Always call RemoteBitrateEstimator::IncomingPacket from Call. (patchset #9 id:160001 of https://codereview.webrtc.org/2659563002/ ) by nisse · 8 years ago
  29. 6d4dd59 Always call RemoteBitrateEstimator::IncomingPacket from Call. by nisse · 8 years ago
  30. 3ebbcb5 Stop using VoEVideoSync in Call/VideoReceiveStream. by solenberg · 8 years ago
  31. d32bf75 Pass SdpAudioFormat through Channel, without converting to CodecInst by kwiberg · 8 years ago
  32. 95aa964 Support external audio mixer in webrtc 2. by gyzhou · 8 years ago
  33. 39ce11f Revert of Support external audio mixer. (patchset #5 id:140001 of https://codereview.webrtc.org/2539213003/ ) by gyzhou · 8 years ago
  34. f6bcac5 Support external audio mixer in webrtc. by gyzhou · 8 years ago
  35. 0245da0 Move ownership of PacketRouter from CongestionController to Call. by nisse · 8 years ago
  36. 04c0722 Replace AudioConferenceMixer with AudioMixer. by aleloi · 8 years ago
  37. 1acfbd2 Expose RtpCodecParameters to VoiceMediaInfo stats. by hbos · 8 years ago
  38. d4adce4 Remove Absolute Send Time from list of supported header extensions for audio streams. by solenberg · 8 years ago
  39. 7602aab Remove usage of VoEBase::AssociateSendChannel() from WVoMC. by solenberg · 8 years ago
  40. 572ae12 Fix crash when registering abs-send-time to AudioSend/ReceiveStream. by stefan · 8 years ago
  41. b521aa7 Clean up abs-send-time for audio. by stefan · 8 years ago
  42. 051f678 Add a NeededFrequency() method to the AudioMixer::Source interface. by aleloi · 8 years ago
  43. 6c27849 Move audio frame memory handling inside AudioMixer. by aleloi · 8 years ago
  44. aed581a Made AudioReceiveStream a mixer participant. by aleloi · 8 years ago
  45. 18e0b67 Restarting channel when swapping AudioReceiveStreams in WebrtcVoE. by aleloi · 8 years ago
  46. 6348978 Add new decoding statistics for muted output by henrik.lundin · 9 years ago
  47. a69d973 Move webrtc/audio_*.h to webrtc/api/call by kjellander · 9 years ago
  48. 84ef615 Removed calls to VoE::SetPlayout() from WebRTCVoiceEngine. by aleloi · 9 years ago
  49. 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
  50. 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
  51. 1895526 Move RtcEventLog object from inside VoiceEngine to Call. by Ivo Creusen · 9 years ago
  52. 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
  53. 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
  54. a7d88d3 Remove audio/video distinction for probe packets. by Peter Boström · 9 years ago
  55. 217fb66 Add AudioReceiveStream::SetGain() method and use that in WVoMC::SetOutputVolume(). by solenberg · 9 years ago
  56. 8189b02 Configure VoE NACK through AudioReceiveStream::Config, for receive streams. Also minor refactoring of WVoE unit test. by solenberg · 9 years ago
  57. 29b1a8d Moved creation of AudioDecoderFactory to inside PeerConnectionFactory. by ossu · 9 years ago
  58. 6f8d686 Remove use of RtpHeaderExtension and clean up by isheriff · 9 years ago
  59. d28db7f Delete all use of tick_util.h. by Niels Möller · 9 years ago
  60. 1ba8d39 Remove webrtc/stream.h and unutilized inheritance. by pbos · 9 years ago
  61. 3d7db26 Switch voice transport to use Call and Stream instead of VoENetwork. by mflodman · 9 years ago
  62. 7ffeab5 Reland "Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies." by kjellander@webrtc.org · 9 years ago
  63. 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
  64. 99b345c Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies. by kjellander@webrtc.org · 9 years ago
  65. fffa42b Replace scoped_ptr with unique_ptr in webrtc/audio/ by kwiberg · 9 years ago
  66. 80e1207 Move congestion controller to a separate module. by Stefan Holmer · 9 years ago
  67. b7f89d6 Replace scoped_ptr with unique_ptr in webrtc/voice_engine/ by kwiberg · 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. 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
  72. e591f93 Storing raw audio sink for default audio track. by deadbeef · 9 years ago
  73. 3842c5c Wire-up BWE feedback for audio receive streams. by Stefan Holmer · 9 years ago
  74. f888bb5 Support for unmixed remote audio into tracks. by Tommi · 9 years ago
  75. 358057b Use ChannelProxy for most calls on voe::Channel in Audio[Receive|Send]Stream. by solenberg · 9 years ago
  76. 1372508 Open backdoor in VoiceEngineImpl to get at the actual voe::Channel objects from an ID. by solenberg · 9 years ago
  77. 7add058 Move some receive stream configuration into webrtc::AudioReceiveStream. by solenberg · 9 years ago
  78. 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
  79. 566ef24 Move VoiceEngineObserver into AudioSendStream so that we detect typing noises and return properly in GetStats(). by solenberg · 9 years ago
  80. 98f5351 system_wrappers: rename interface -> include by Henrik Kjellander · 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. a2f30de Log Call {audio, video} stream deletions. by pbos · 9 years ago
  86. 5c389d3 Split webrtc/video into webrtc/{audio,call,video}. by Peter Boström · 10 years ago[Renamed (98%) from webrtc/video/audio_receive_stream.cc]
  87. 8bffba7 Fix BWE bug where audio has timestamps in us. by Stefan Holmer · 10 years ago
  88. 91d6ede Add RTC_ prefix to (D)CHECKs and related macros. by henrikg · 10 years ago
  89. 68786d2 Wire up PacketTime to ReceiveStreams. by stefan · 10 years ago
  90. 867fb52 Add support for transport wide sequence numbers by sprang · 10 years ago
  91. 6bb1b6e Control combined_audio_video_bwe with config bool. by pbos · 10 years ago
  92. cd67022 Define Stream base classes by Jelena Marusic · 10 years ago
  93. 8fc7fa7 Base A/V synchronization on sync_labels. by pbos · 10 years ago
  94. ff4ea93 Only use paced packets for estimating bitrate probes. by Stefan Holmer · 10 years ago
  95. 04f4931 VoE2 API draft by Fredrik Solenberg · 10 years ago
  96. 23fba1f Add AudioReceiveStream to Call API. by Fredrik Solenberg · 10 years ago