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gerrit-public.fairphone.software / platform / external / webrtc / 53e048d83aa47de7bf98d1f10c667bd0e79551e7 / audio / channel.cc
  1. a8b7c7f Move remaining traces of VoiceEngine by Fredrik Solenberg · 7 years ago[Renamed (98%) from voice_engine/channel.cc]
  2. 90ea504 Delete Channel::OnRecoveredPacket. by Niels Möller · 7 years ago
  3. 8f5787a Move ownership of voe::Channel into Audio[Receive|Send]Stream. by Fredrik Solenberg · 7 years ago
  4. 3b903d0 Reconfigure, not reconstruct, AudioReceiveStreams. by Fredrik Solenberg · 7 years ago
  5. 2a87797 Remove voe::TransmitMixer by Fredrik Solenberg · 7 years ago
  6. abbff89 Add new UMA metric for NetEq target buffer delay by Henrik Lundin · 7 years ago
  7. 606c882 Optional: Use nullopt and implicit construction in /voice_engine by Oskar Sundbom · 7 years ago
  8. 55900fd Move APM initialization into WebRtcVoiceEngine by Fredrik Solenberg · 7 years ago
  9. de93943 Revert "Revert "Encode log events periodically instead of for every event."" by Bjorn Terelius · 7 years ago
  10. 33c5c7f Revert "Encode log events periodically instead of for every event." by Zhi Huang · 7 years ago
  11. b154c27 Encode log events periodically instead of for every event. by Bjorn Terelius · 7 years ago
  12. 675513b Stop using LOG macros in favor of RTC_ prefixed macros. by Mirko Bonadei · 7 years ago
  13. 9bde6b7 Add new UMA metric for the audio receiver delay by Henrik Lundin · 7 years ago
  14. 78609d5 Reland of BWE allocation strategy by Alex Narest · 7 years ago
  15. dc9ca93 Revert "BWE allocation strategy" by Alex Narest · 7 years ago
  16. 8818237 voe::Channel: Don't use CodecManager and RentACodec by Karl Wiberg · 7 years ago
  17. a5fbc23 BWE allocation strategy by Alex Narest · 7 years ago
  18. 39260c4 Revert "BWE allocation strategy allows controlling of bitrate allocation with WEBRTC external logic." by Lu Liu · 7 years ago
  19. 54d1da1 BWE allocation strategy allows controlling of bitrate allocation with WEBRTC external logic. by Alex Narest · 7 years ago
  20. 604c14d Reland "Remove deprecated functions from RtcEventLog" by Elad Alon · 7 years ago
  21. d312a91 Revert "Remove deprecated functions from RtcEventLog" by Danil Chapovalov · 7 years ago
  22. 5fd6e5e Remove deprecated functions from RtcEventLog by Elad Alon · 7 years ago
  23. 22ec952 Delete in_order argument to RtpReceiver::IncomingRtpPacket by Niels Möller · 7 years ago
  24. c62f6c7 RTPPayloadRegistry: Use SdpAudioFormat to represent audio codecs by Karl Wiberg · 7 years ago
  25. 83ccca1 Create and use RtcEventLogOutput for output by Elad Alon · 7 years ago
  26. 4a87e1c Remove encoding code from RtcEventLogImpl and use RtcEventLogEncoder instead by Elad Alon · 7 years ago
  27. ecc51e9 Change to LOG(...) logging in most of voice_engine/channel.cc by Sam Zackrisson · 7 years ago
  28. 440216f Split LogRtpHeader and LogRtcpPacket into separate versions for incoming and outgoing packets. by Bjorn Terelius · 7 years ago
  29. 1c239d4 Remove voe::Statistics. by solenberg · 7 years ago
  30. c7b4a45 Remove various IDs: by solenberg · 7 years ago
  31. 4580217 Adds WebRTC.Audio.EncodingTaskQueueLatencyMs by henrika · 7 years ago
  32. e423a9de Revert of Remove various IDs (patchset #7 id:120001 of https://codereview.webrtc.org/3019543002/ ) by solenberg · 7 years ago
  33. 2d0f775 Remove various IDs: by solenberg · 7 years ago
  34. 6df16bf Remove unnecessary send codec initialization from voe::Channel. by solenberg · 7 years ago
  35. fc3a2e3 Remove the VoiceEngineObserver callback interface. by solenberg · 7 years ago
  36. 2397b9a Remove voe::OutputMixer and AudioConferenceMixer. by solenberg · 7 years ago
  37. 73b60b8 Remove the redundant method GetPayloadSpecifics by Karl Wiberg · 7 years ago
  38. 946d886 Remove VoENetwork by solenberg · 7 years ago
  39. dd3abbb Remove VoERTP_RTCP. by solenberg · 7 years ago
  40. 6dc2038 Remove VoECodec. by solenberg · 7 years ago
  41. b63310a Remove VoEFile and things it uses. by solenberg · 7 years ago
  42. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  43. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/voice_engine/channel.cc]
  44. 479d3d7 Drop return value from RtpRtcp::IncomingRtcpPacket. by nisse · 7 years ago
  45. a37de39 Update thread annotiation macros to use RTC_ prefix by danilchap · 7 years ago
  46. e1198e0 Add new ANA stats to the old GetStats() to count the number of actions taken by each controller. by ivoc · 7 years ago
  47. 529662a Move array_view.h to webrtc/api/ by kwiberg · 7 years ago
  48. 1acbd68 Move RtpExtension to api/ directory and config.h/.cc to call/. by Stefan Holmer · 7 years ago
  49. da194e7 Delete remnants of RTX support in voice_engine. by nisse · 7 years ago
  50. e9ef907 Revert of Add logging of host lookups made by TurnPort to the RtcEventLog. (patchset #11 id:200001 of https://codereview.webrtc.org/2996933003/ ) by maxmorin · 7 years ago
  51. c251cb1 Add logging host lookups made by TurnPort to the RtcEventLog. by jonaso · 7 years ago
  52. ee89e78 Replace CHECK(x && y) with two separate CHECK() calls by kwiberg · 7 years ago
  53. 3e69e5c Renamed fields in rtp_rtcp_defines.h/RTCPReportBlock by srte · 7 years ago
  54. 186d9c3 Renamed fields in common_types.h/RtcpStatistics. by srte · 7 years ago
  55. 822ff2b Explicitly inform PacketRouter which RTP-RTCP modules are REMB-candidates by eladalon · 7 years ago
  56. 3c45186 Move total audio energy and duration tracking to AudioLevel and protect with existing critial section. by zstein · 7 years ago
  57. 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
  58. 950c1c9 TransmitMixer: Check GetSendCodec return value. by ossu · 7 years ago
  59. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 7 years ago
  60. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 7 years ago
  61. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 7 years ago
  62. 36b1a5f Add mute state field to AudioFrame and switch some callers to use it. Also make AudioFrame::data_ private and instead provide: by yujo · 7 years ago
  63. 76d29f9 Fix Channel::GetSendCodec when used together with SetEncoder. by ossu · 7 years ago
  64. 77cd58e This cl removes RtcEventLog deps to call:call_interfaces. The purpose is to be able to use the event log from the upcoming RtpTransport. by perkj · 7 years ago
  65. f472699 Replace AudioSendStream::Config with rtclog::StreamConfig. by perkj · 7 years ago
  66. ac8f52d Replace AudioReceiveStream::Config with rtclog::StreamConfig. by perkj · 7 years ago
  67. c0876aa Replace VideoSendStream::Config with new rtclog::StreamConfig in RtcEventLog. by perkj · 7 years ago
  68. 09e71da Replace VideoReceiveStream::Config with new rtclog::StreamConfig in RtcEventLog. by perkj · 7 years ago
  69. 4515fa0 Resolves race between Channel::ProcessAndEncodeAudio() and Channel::StopSend() by henrika · 7 years ago
  70. 20a4b3f Injectable audio encoders: WebRtcVoiceEngine and company by ossu · 7 years ago
  71. cae45d0 Move RtpTransportControllerSend to a new file. by nisse · 7 years ago
  72. 92aef17 Replace Clock with timeutils in AudioEncoder. by michaelt · 7 years ago
  73. 1ffbd6c Injectable audio encoders: voice_engine/channel changes. by ossu · 7 years ago
  74. cde46b7 Resolve cyclic dependency between audio network adaptor and event log api by michaelt · 7 years ago
  75. fdbfdc9 Let PacketRouter separate send and receive modules. by nisse · 7 years ago
  76. ec6fbd2 Moves channel-dependent audio input processing to separate encoder task queue. by henrika · 7 years ago
  77. 2877048 Experiment-driven configuration of PLR/RPLR-based FecController by elad.alon · 7 years ago
  78. 1c07c70 Reland "WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType" by kwiberg · 7 years ago
  79. b8f9a32 Define RtpTransportControllerSendInterface. by nisse · 7 years ago
  80. 670a7f3 Revert of WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType (patchset #13 id:260001 of https://codereview.webrtc.org/2686043006/ ) by kwiberg · 7 years ago
  81. 1724cfb WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType by kwiberg · 7 years ago
  82. dadb4dc Allow ANA to receive RPLR (recoverable packet loss rate) indications by elad.alon · 7 years ago
  83. d12a8e1 Attach TransportFeedbackPacketLossTracker to ANA (PLR only) by elad.alon · 7 years ago
  84. 0a2391f Add thread check to ModuleProcessThread::DeRegisterModule and remove all unnecessary locking that was there due to one implementation calling from a different thread. by tommi · 7 years ago
  85. fe7dd6d Remove VoEAudioProcessing interface. by solenberg · 7 years ago
  86. 67fdb80 Reland of Enable cpplint and fix cpplint errors in webrtc/*audio (patchset #1 id:1 of https://codereview.webrtc.org/2739143002/ ) by oprypin · 7 years ago
  87. e47c1d3 Revert of Enable cpplint and fix cpplint errors in webrtc/*audio (patchset #4 id:180001 of https://codereview.webrtc.org/2683033004/ ) by oprypin · 7 years ago
  88. aebe55c Enable cpplint and fix cpplint errors in webrtc/*audio by oprypin · 7 years ago
  89. 8d73f8c Remove VoEVolumeControl interface. by solenberg · 7 years ago
  90. f949000 Rename webrtc::PacketInfo to webrtc::PacketFeedback. This resolves ambiguity with a similarly named RTCPReceiver::PacketInformation and RtpPacketizerVp9::PacketInfo. by elad.alon · 7 years ago
  91. dea489f Add support for Location (RTC_FROM_HERE) to ProcessThread::RegisterModule. by tommi · 7 years ago
  92. 796b8f9 Remove usage of VoEVolumeControl from WVoE and Audio[Send|Receive]Stream. by solenberg · 7 years ago
  93. 09f090c Remove workaround for bug 6986 by kwiberg · 7 years ago
  94. 3fd31fe Fix TSAN race in webrtc::voe::Channel. by hbos · 7 years ago
  95. 32d0010 Add probe logging to RtcEventLog. by philipel · 7 years ago
  96. 8aadd50 Propagate packet pacing information to SendTimeHistory. by philipel · 7 years ago
  97. 657bab2 Replace AudioReceiveStream::DeliverRtp with OnRtpPacket. by nisse · 7 years ago
  98. 7d59f6b Reland of Delete class SSRCDatabase, and its global ssrc registry. (patchset #1 id:1 of https://codereview.webrtc.org/2700413002/ ) by nisse · 7 years ago
  99. 424e6cf Rename some variables and methods in RTC event log. by terelius · 7 years ago
  100. b5848ec Revert of Delete class SSRCDatabase, and its global ssrc registry. (patchset #20 id:370001 of https://codereview.webrtc.org/2644303002/ ) by kjellander · 7 years ago