Gitiles
Code Review Sign In
gerrit-public.fairphone.software / platform / external / webrtc / fc29b0ad4658ccf68a51f2f170ed5641e354bf0a / audio / mock_voe_channel_proxy.h
  1. 6298b56 Cleanup: Using RtpRtcp directly from AudioSendStream by Sebastian Jansson · 4 years, 7 months ago
  2. fcf79cc Add estimatedPlayoutTimestamp to RTCInboundRTPStreamStats. by Åsa Persson · 4 years, 10 months ago
  3. 70efdde Set local ssrc at construction of Rtp module by Erik Språng · 5 years ago
  4. ee5ccbc Move ownership of RTPSenderAudio to ChannelSend. by Niels Möller · 5 years ago
  5. 8fb1a6a Delete a few return values from audio streams and video send streams. by Niels Möller · 5 years ago
  6. 14a7cf9 Adds CallEncoder to ChannelSend. by Sebastian Jansson · 6 years ago
  7. 3b50f9f Propagate base minimum delay to audio_receiver_stream by Ruslan Burakov · 6 years ago
  8. 77938e6 Simulcast work to enable RID mux. by Amit Hilbuch · 6 years ago
  9. f693bfa Remove CodecInst pt.2 by Fredrik Solenberg · 6 years ago
  10. e977199 Delete ChannelSend::RegisterTransport, replacing by construction argument by Niels Möller · 6 years ago
  11. 254d869 Routing BitrateAllocationUpdate to audio codec. by Sebastian Jansson · 6 years ago
  12. c69a56e Remove more unneeded things from ChannelSend by Fredrik Solenberg · 6 years ago
  13. dced9f6 Delete class ChannelSendProxy by Niels Möller · 6 years ago
  14. 645a3af Remove unused/unnecessary things from ChannelSend. by Fredrik Solenberg · 6 years ago
  15. 2681523 Tweak ChannelSend interface, to make it closer to ChannelSendProxy by Niels Möller · 6 years ago
  16. 349ade3 Delete class ChannelReceiveProxy. by Niels Möller · 6 years ago
  17. 9190b82 Propagate SDP negotiation of extmap-allow-mixed to RtpHeaderExtensionMap by Johannes Kron · 6 years ago
  18. 78410ad Fixes use after free error when setting a new FrameEncryptor on ChannelSend. by Benjamin Wright · 6 years ago
  19. 359d60a Adds target rate to audio send stream stats. by Sebastian Jansson · 6 years ago
  20. ae4237e Set ChannelReceive transport at construction time. by Niels Möller · 6 years ago
  21. 84583f6 Enable End-to-End Encrypted Audio Payloads. by Benjamin Wright · 6 years ago
  22. 530ead4 Split voe::Channel into ChannelSend and ChannelReceive by Niels Möller · 6 years ago
  23. b222f49 Split ChannelProxy into send and receive classes. by Niels Möller · 6 years ago
  24. 30b4839 Refactor voe::Channel to not use RtpReceiver. by Niels Möller · 6 years ago
  25. fa4e185 Delete class voe::RtcEventLogProxy by Niels Möller · 6 years ago
  26. 848d6d3 Change Channel::GetRtpRtcp to return only RtpRtcp, not RtpReceiver. by Niels Möller · 6 years ago
  27. 665174f Reformat the WebRTC code base by Yves Gerey · 6 years ago
  28. f782492 Delete RtpFeedback. The ssrc for a receive stream should be known at by Niels Möller · 6 years ago
  29. 5f22365 Remove unnecessary proxy+lock code around RtcpRttStats pointer by Tommi · 6 years ago
  30. f120cba Delete AudioMonitor and related code. by Niels Möller · 7 years ago
  31. a8b7c7f Move remaining traces of VoiceEngine by Fredrik Solenberg · 7 years ago[Renamed (95%) from test/mock_voe_channel_proxy.h]
  32. 90ea504 Delete Channel::OnRecoveredPacket. by Niels Möller · 7 years ago
  33. 8f5787a Move ownership of voe::Channel into Audio[Receive|Send]Stream. by Fredrik Solenberg · 7 years ago
  34. d524751 Replace VoEBase::[Start/Stop]Playout(). by Fredrik Solenberg · 7 years ago
  35. aaedf75 Replace VoEBase::[Start/Stop]Send(). by Fredrik Solenberg · 7 years ago
  36. 2a87797 Remove voe::TransmitMixer by Fredrik Solenberg · 7 years ago
  37. 1c239d4 Remove voe::Statistics. by solenberg · 7 years ago
  38. 2397b9a Remove voe::OutputMixer and AudioConferenceMixer. by solenberg · 7 years ago
  39. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  40. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/test/mock_voe_channel_proxy.h]
  41. e1198e0 Add new ANA stats to the old GetStats() to count the number of actions taken by each controller. by ivoc · 7 years ago
  42. 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
  43. 20a4b3f Injectable audio encoders: WebRtcVoiceEngine and company by ossu · 7 years ago
  44. 8d609f6 Reland of Implemented the GetSources() in native code. (patchset #1 id:1 of https://codereview.webrtc.org/2809613002/ ) by hbos · 7 years ago
  45. fbcc5cb Revert of Implemented the GetSources() in native code. (patchset #11 id:510001 of https://codereview.webrtc.org/2770233003/ ) by olka · 7 years ago
  46. 292084c Added the GetSources() to the RtpReceiverInterface and implemented by zhihuang · 7 years ago
  47. 1ffbd6c Injectable audio encoders: voice_engine/channel changes. by ossu · 7 years ago
  48. fdbfdc9 Let PacketRouter separate send and receive modules. by nisse · 7 years ago
  49. 1c07c70 Reland "WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType" by kwiberg · 7 years ago
  50. b8f9a32 Define RtpTransportControllerSendInterface. by nisse · 7 years ago
  51. 670a7f3 Revert of WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType (patchset #13 id:260001 of https://codereview.webrtc.org/2686043006/ ) by kwiberg · 7 years ago
  52. 1724cfb WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType by kwiberg · 7 years ago
  53. dadb4dc Allow ANA to receive RPLR (recoverable packet loss rate) indications by elad.alon · 7 years ago
  54. d12a8e1 Attach TransportFeedbackPacketLossTracker to ANA (PLR only) by elad.alon · 7 years ago
  55. 8d73f8c Remove VoEVolumeControl interface. by solenberg · 7 years ago
  56. 796b8f9 Remove usage of VoEVolumeControl from WVoE and Audio[Send|Receive]Stream. by solenberg · 7 years ago
  57. 657bab2 Replace AudioReceiveStream::DeliverRtp with OnRtpPacket. by nisse · 8 years ago
  58. 08b19df Remove VoEVideoSync interface. by solenberg · 8 years ago
  59. 7de8d64 Wire up audio packet loss to BWE. by stefan · 8 years ago
  60. bd9a77f Remove most of the remaining calls to VoE APIs from Audio[Send|Receive]Stream. by solenberg · 8 years ago
  61. 3ebbcb5 Stop using VoEVideoSync in Call/VideoReceiveStream. by solenberg · 8 years ago
  62. 9332b7d Reland "Update rtt on audio only calls". by michaelt · 8 years ago
  63. 78b4d56 Relanding "Pass time constant to bwe smoothing filter." by minyue · 8 years ago
  64. 6287e82 Revert of Pass time constant to bwe smoothing filter. (patchset #8 id:140001 of https://codereview.webrtc.org/2518923003/ ) by ossu · 8 years ago
  65. 9abbf5a Pass time constanct to bwe smoothing filter. by michaelt · 8 years ago
  66. ffbbcac Support multiple timestamp rates for sending DTMF. by solenberg · 8 years ago
  67. 7602aab Remove usage of VoEBase::AssociateSendChannel() from WVoMC. by solenberg · 8 years ago
  68. 79e0588 Set actual transport overhead in rtp_rtcp by michaelt · 8 years ago
  69. b521aa7 Clean up abs-send-time for audio. by stefan · 8 years ago
  70. 6b825df Using AudioOption to enable audio network adaptor. by minyue · 8 years ago
  71. 982bf89 Revert of Add RtcpRttStats to AudioStream (patchset #1 id:1 of https://codereview.webrtc.org/2402333002/ ) by sprang · 8 years ago
  72. e0729c5 Add RtcpRttStats to AudioStream by michaelt · 8 years ago
  73. ac9f876 Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/ by kwiberg · 8 years ago
  74. 77eab70 Enable the -Wundef warning for clang by kwiberg · 8 years ago
  75. 86cc6ff Variable audio bitrate. by mflodman · 8 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 · 8 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 · 8 years ago
  78. 1895526 Move RtcEventLog object from inside VoiceEngine to Call. by Ivo Creusen · 8 years ago
  79. 217fb66 Add AudioReceiveStream::SetGain() method and use that in WVoMC::SetOutputVolume(). by solenberg · 8 years ago
  80. 9421853 Add AudioSendStream::SetMuted() method and use it in WVoMC::MuteStream(). by solenberg · 8 years ago
  81. 971cab0 Configure VoE NACK through AudioSendStream::Config, for send streams. by solenberg · 8 years ago
  82. 29b1a8d Moved creation of AudioDecoderFactory to inside PeerConnectionFactory. by ossu · 8 years ago
  83. 3d7db26 Switch voice transport to use Call and Stream instead of VoENetwork. by mflodman · 8 years ago
  84. 8842c3e Relanding https://codereview.webrtc.org/1715883002/ in pieces. by solenberg · 8 years ago
  85. 3ecb5c8 Revert of - Clean up unused voice engine DTMF code. (patchset #4 id:60001 of https://codereview.webrtc.org/1722253002/ ) by solenberg · 8 years ago
  86. 8886c81 - Clean up unused voice engine DTMF code following removal of VoEDtmf APIs. by solenberg · 8 years ago
  87. bba9dec Use separate rtp module lists for send and receive in PacketRouter. by stefan · 9 years ago
  88. 3313ec9 Enable transport seq num extension on receive channel to suppress log warning. by stefan · 9 years ago
  89. b86d4e4 Prepare the AudioSendStream to be hooked up to send-side BWE. by Stefan Holmer · 9 years ago
  90. b572768 - Remove calls to VoEDtmf from WVoE/MC. by Fredrik Solenberg · 9 years ago
  91. 358057b Use ChannelProxy for most calls on voe::Channel in Audio[Receive|Send]Stream. by solenberg · 9 years ago
  92. 1372508 Open backdoor in VoiceEngineImpl to get at the actual voe::Channel objects from an ID. by solenberg · 9 years ago