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gerrit-public.fairphone.software / platform / external / webrtc / d0136b8afba4b4ed68c39f9d50c5a787d8bc3ba8 / audio / channel.cc
  1. 918f50c Use absl::make_unique and absl::WrapUnique directly by Karl Wiberg · 7 years ago
  2. 64b17c2 Remove StreamStatistician::IsPacketInOrder by Danil Chapovalov · 7 years ago
  3. 8491693 Update packetsLost and jitter stats any time a packet is received. by Taylor Brandstetter · 7 years ago
  4. 665174f Reformat the WebRTC code base by Yves Gerey · 7 years ago
  5. b9b146c Replace rtc::Optional with absl::optional in audio, call and video by Danil Chapovalov · 7 years ago
  6. f782492 Delete RtpFeedback. The ssrc for a receive stream should be known at by Niels Möller · 7 years ago
  7. eda0087 Drop the RTT as input to IsRetransmitOfOldPacket. by Niels Möller · 7 years ago
  8. bbf21a3 Remove dependencies on modules:module_api from AudioProcessing. by Fredrik Solenberg · 7 years ago
  9. ef99888 Delete OnIncomingCSRCChanged and related code. by Niels Möller · 7 years ago
  10. bb50ce5 Wire up MID send value to the PeerConnection API by Steve Anton · 7 years ago
  11. 5f22365 Remove unnecessary proxy+lock code around RtcpRttStats pointer by Tommi · 7 years ago
  12. 9cfb18c Delete obsolete method RtpFeedback::OnInitializeDecoder. by Niels Möller · 7 years ago
  13. 0812634 Pass a real audio codec pair ID to decoders that we create by Karl Wiberg · 7 years ago
  14. 6fed924 Delete RTPPayloadRegistry::SetIncomingPayloadType. by Niels Möller · 7 years ago
  15. 8493594 Cleanup of TransportFeedbackObserver interface by Erik Språng · 7 years ago
  16. f120cba Delete AudioMonitor and related code. by Niels Möller · 7 years ago
  17. 24ea822 Remove logging in audio/* from release builds. by Jonas Olsson · 7 years ago
  18. a8b7c7f Move remaining traces of VoiceEngine by Fredrik Solenberg · 7 years ago[Renamed (98%) from voice_engine/channel.cc]
  19. 90ea504 Delete Channel::OnRecoveredPacket. by Niels Möller · 7 years ago
  20. 8f5787a Move ownership of voe::Channel into Audio[Receive|Send]Stream. by Fredrik Solenberg · 7 years ago
  21. 3b903d0 Reconfigure, not reconstruct, AudioReceiveStreams. by Fredrik Solenberg · 7 years ago
  22. 2a87797 Remove voe::TransmitMixer by Fredrik Solenberg · 7 years ago
  23. abbff89 Add new UMA metric for NetEq target buffer delay by Henrik Lundin · 7 years ago
  24. 606c882 Optional: Use nullopt and implicit construction in /voice_engine by Oskar Sundbom · 7 years ago
  25. 55900fd Move APM initialization into WebRtcVoiceEngine by Fredrik Solenberg · 7 years ago
  26. de93943 Revert "Revert "Encode log events periodically instead of for every event."" by Bjorn Terelius · 7 years ago
  27. 33c5c7f Revert "Encode log events periodically instead of for every event." by Zhi Huang · 7 years ago
  28. b154c27 Encode log events periodically instead of for every event. by Bjorn Terelius · 7 years ago
  29. 675513b Stop using LOG macros in favor of RTC_ prefixed macros. by Mirko Bonadei · 7 years ago
  30. 9bde6b7 Add new UMA metric for the audio receiver delay by Henrik Lundin · 7 years ago
  31. 78609d5 Reland of BWE allocation strategy by Alex Narest · 7 years ago
  32. dc9ca93 Revert "BWE allocation strategy" by Alex Narest · 7 years ago
  33. 8818237 voe::Channel: Don't use CodecManager and RentACodec by Karl Wiberg · 7 years ago
  34. a5fbc23 BWE allocation strategy by Alex Narest · 7 years ago
  35. 39260c4 Revert "BWE allocation strategy allows controlling of bitrate allocation with WEBRTC external logic." by Lu Liu · 7 years ago
  36. 54d1da1 BWE allocation strategy allows controlling of bitrate allocation with WEBRTC external logic. by Alex Narest · 7 years ago
  37. 604c14d Reland "Remove deprecated functions from RtcEventLog" by Elad Alon · 7 years ago
  38. d312a91 Revert "Remove deprecated functions from RtcEventLog" by Danil Chapovalov · 7 years ago
  39. 5fd6e5e Remove deprecated functions from RtcEventLog by Elad Alon · 7 years ago
  40. 22ec952 Delete in_order argument to RtpReceiver::IncomingRtpPacket by Niels Möller · 7 years ago
  41. c62f6c7 RTPPayloadRegistry: Use SdpAudioFormat to represent audio codecs by Karl Wiberg · 7 years ago
  42. 83ccca1 Create and use RtcEventLogOutput for output by Elad Alon · 7 years ago
  43. 4a87e1c Remove encoding code from RtcEventLogImpl and use RtcEventLogEncoder instead by Elad Alon · 7 years ago
  44. ecc51e9 Change to LOG(...) logging in most of voice_engine/channel.cc by Sam Zackrisson · 7 years ago
  45. 440216f Split LogRtpHeader and LogRtcpPacket into separate versions for incoming and outgoing packets. by Bjorn Terelius · 7 years ago
  46. 1c239d4 Remove voe::Statistics. by solenberg · 7 years ago
  47. c7b4a45 Remove various IDs: by solenberg · 7 years ago
  48. 4580217 Adds WebRTC.Audio.EncodingTaskQueueLatencyMs by henrika · 7 years ago
  49. e423a9de Revert of Remove various IDs (patchset #7 id:120001 of https://codereview.webrtc.org/3019543002/ ) by solenberg · 7 years ago
  50. 2d0f775 Remove various IDs: by solenberg · 7 years ago
  51. 6df16bf Remove unnecessary send codec initialization from voe::Channel. by solenberg · 7 years ago
  52. fc3a2e3 Remove the VoiceEngineObserver callback interface. by solenberg · 7 years ago
  53. 2397b9a Remove voe::OutputMixer and AudioConferenceMixer. by solenberg · 7 years ago
  54. 73b60b8 Remove the redundant method GetPayloadSpecifics by Karl Wiberg · 7 years ago
  55. 946d886 Remove VoENetwork by solenberg · 7 years ago
  56. dd3abbb Remove VoERTP_RTCP. by solenberg · 7 years ago
  57. 6dc2038 Remove VoECodec. by solenberg · 7 years ago
  58. b63310a Remove VoEFile and things it uses. by solenberg · 7 years ago
  59. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  60. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/voice_engine/channel.cc]
  61. 479d3d7 Drop return value from RtpRtcp::IncomingRtcpPacket. by nisse · 7 years ago
  62. a37de39 Update thread annotiation macros to use RTC_ prefix by danilchap · 7 years ago
  63. e1198e0 Add new ANA stats to the old GetStats() to count the number of actions taken by each controller. by ivoc · 7 years ago
  64. 529662a Move array_view.h to webrtc/api/ by kwiberg · 7 years ago
  65. 1acbd68 Move RtpExtension to api/ directory and config.h/.cc to call/. by Stefan Holmer · 7 years ago
  66. da194e7 Delete remnants of RTX support in voice_engine. by nisse · 7 years ago
  67. 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
  68. c251cb1 Add logging host lookups made by TurnPort to the RtcEventLog. by jonaso · 7 years ago
  69. ee89e78 Replace CHECK(x && y) with two separate CHECK() calls by kwiberg · 7 years ago
  70. 3e69e5c Renamed fields in rtp_rtcp_defines.h/RTCPReportBlock by srte · 7 years ago
  71. 186d9c3 Renamed fields in common_types.h/RtcpStatistics. by srte · 7 years ago
  72. 822ff2b Explicitly inform PacketRouter which RTP-RTCP modules are REMB-candidates by eladalon · 7 years ago
  73. 3c45186 Move total audio energy and duration tracking to AudioLevel and protect with existing critial section. by zstein · 7 years ago
  74. 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
  75. 950c1c9 TransmitMixer: Check GetSendCodec return value. by ossu · 7 years ago
  76. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 8 years ago
  77. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 8 years ago
  78. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 8 years ago
  79. 36b1a5f Add mute state field to AudioFrame and switch some callers to use it. Also make AudioFrame::data_ private and instead provide: by yujo · 8 years ago
  80. 76d29f9 Fix Channel::GetSendCodec when used together with SetEncoder. by ossu · 8 years ago
  81. 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 · 8 years ago
  82. f472699 Replace AudioSendStream::Config with rtclog::StreamConfig. by perkj · 8 years ago
  83. ac8f52d Replace AudioReceiveStream::Config with rtclog::StreamConfig. by perkj · 8 years ago
  84. c0876aa Replace VideoSendStream::Config with new rtclog::StreamConfig in RtcEventLog. by perkj · 8 years ago
  85. 09e71da Replace VideoReceiveStream::Config with new rtclog::StreamConfig in RtcEventLog. by perkj · 8 years ago
  86. 4515fa0 Resolves race between Channel::ProcessAndEncodeAudio() and Channel::StopSend() by henrika · 8 years ago
  87. 20a4b3f Injectable audio encoders: WebRtcVoiceEngine and company by ossu · 8 years ago
  88. cae45d0 Move RtpTransportControllerSend to a new file. by nisse · 8 years ago
  89. 92aef17 Replace Clock with timeutils in AudioEncoder. by michaelt · 8 years ago
  90. 1ffbd6c Injectable audio encoders: voice_engine/channel changes. by ossu · 8 years ago
  91. cde46b7 Resolve cyclic dependency between audio network adaptor and event log api by michaelt · 8 years ago
  92. fdbfdc9 Let PacketRouter separate send and receive modules. by nisse · 8 years ago
  93. ec6fbd2 Moves channel-dependent audio input processing to separate encoder task queue. by henrika · 8 years ago
  94. 2877048 Experiment-driven configuration of PLR/RPLR-based FecController by elad.alon · 8 years ago
  95. 1c07c70 Reland "WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType" by kwiberg · 8 years ago
  96. b8f9a32 Define RtpTransportControllerSendInterface. by nisse · 8 years ago
  97. 670a7f3 Revert of WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType (patchset #13 id:260001 of https://codereview.webrtc.org/2686043006/ ) by kwiberg · 8 years ago
  98. 1724cfb WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType by kwiberg · 8 years ago
  99. dadb4dc Allow ANA to receive RPLR (recoverable packet loss rate) indications by elad.alon · 8 years ago
  100. d12a8e1 Attach TransportFeedbackPacketLossTracker to ANA (PLR only) by elad.alon · 8 years ago