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webrtc / src / webrtc / 1b9b5778ad350cccb83b1f7b9bdcfec86f55ac36 / voice_engine / channel.cc
  1. e5d488c Drop return value from RtpRtcp::IncomingRtcpPacket. by nisse · 8 years ago
  2. 642a074 Update thread annotiation macros to use RTC_ prefix by danilchap · 8 years ago
  3. 3aa28f0 Add new ANA stats to the old GetStats() to count the number of actions taken by each controller. by ivoc · 8 years ago
  4. 23aa43d Move array_view.h to webrtc/api/ by kwiberg · 8 years ago
  5. 36189cd Move RtpExtension to api/ directory and config.h/.cc to call/. by Stefan Holmer · 8 years ago
  6. 9fdf592 Delete remnants of RTX support in voice_engine. by nisse · 8 years ago
  7. 94c183c 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 · 8 years ago
  8. 4282b1f Add logging host lookups made by TurnPort to the RtcEventLog. by jonaso · 8 years ago
  9. ea15d68 Replace CHECK(x && y) with two separate CHECK() calls by kwiberg · 8 years ago
  10. 504c822 Renamed fields in rtp_rtcp_defines.h/RTCPReportBlock by srte · 8 years ago
  11. 63b0b01 Renamed fields in common_types.h/RtcpStatistics. by srte · 8 years ago
  12. a207a3a Explicitly inform PacketRouter which RTP-RTCP modules are REMB-candidates by eladalon · 8 years ago
  13. 31d74e4 Move total audio energy and duration tracking to AudioLevel and protect with existing critial section. by zstein · 8 years ago
  14. d1701d0 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
  15. 5f13072 TransmitMixer: Check GetSendCodec return value. by ossu · 8 years ago
  16. 76de83e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 8 years ago
  17. bc32410 Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 8 years ago
  18. 60154fd Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 8 years ago
  19. 7d97307 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
  20. 34b78a2 Fix Channel::GetSendCodec when used together with SetEncoder. by ossu · 8 years ago
  21. 33f05e7 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
  22. eaf4d68 Replace AudioSendStream::Config with rtclog::StreamConfig. by perkj · 8 years ago
  23. a6d8766 Replace AudioReceiveStream::Config with rtclog::StreamConfig. by perkj · 8 years ago
  24. 433b35c Replace VideoSendStream::Config with new rtclog::StreamConfig in RtcEventLog. by perkj · 8 years ago
  25. acddd51 Replace VideoReceiveStream::Config with new rtclog::StreamConfig in RtcEventLog. by perkj · 8 years ago
  26. d100c35 Resolves race between Channel::ProcessAndEncodeAudio() and Channel::StopSend() by henrika · 8 years ago
  27. fe5f71c Injectable audio encoders: WebRtcVoiceEngine and company by ossu · 8 years ago
  28. cddf701 Move RtpTransportControllerSend to a new file. by nisse · 8 years ago
  29. 41f1066 Replace Clock with timeutils in AudioEncoder. by michaelt · 8 years ago
  30. 1aa04eb Injectable audio encoders: voice_engine/channel changes. by ossu · 8 years ago
  31. 8c39fad Resolve cyclic dependency between audio network adaptor and event log api by michaelt · 8 years ago
  32. 5e9233b Let PacketRouter separate send and receive modules. by nisse · 8 years ago
  33. ab2deb1 Moves channel-dependent audio input processing to separate encoder task queue. by henrika · 8 years ago
  34. 187f726 Experiment-driven configuration of PLR/RPLR-based FecController by elad.alon · 8 years ago
  35. afd1255 Reland "WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType" by kwiberg · 8 years ago
  36. 22de8f3 Define RtpTransportControllerSendInterface. by nisse · 8 years ago
  37. 52b9df6 Revert of WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType (patchset #13 id:260001 of https://codereview.webrtc.org/2686043006/ ) by kwiberg · 8 years ago
  38. 135be28 WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType by kwiberg · 8 years ago
  39. 1993d91 Allow ANA to receive RPLR (recoverable packet loss rate) indications by elad.alon · 8 years ago
  40. 49d1987 Attach TransportFeedbackPacketLossTracker to ANA (PLR only) by elad.alon · 8 years ago
  41. 3d8ed00 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 · 8 years ago
  42. 788c1f7 Remove VoEAudioProcessing interface. by solenberg · 8 years ago
  43. 0f20d58 Reland of Enable cpplint and fix cpplint errors in webrtc/*audio (patchset #1 id:1 of https://codereview.webrtc.org/2739143002/ ) by oprypin · 8 years ago
  44. 899c143 Revert of Enable cpplint and fix cpplint errors in webrtc/*audio (patchset #4 id:180001 of https://codereview.webrtc.org/2683033004/ ) by oprypin · 8 years ago
  45. ba76094 Enable cpplint and fix cpplint errors in webrtc/*audio by oprypin · 8 years ago
  46. 07f189f Remove VoEVolumeControl interface. by solenberg · 8 years ago
  47. 17d9d86 Rename webrtc::PacketInfo to webrtc::PacketFeedback. This resolves ambiguity with a similarly named RTCPReceiver::PacketInformation and RtpPacketizerVp9::PacketInfo. by elad.alon · 8 years ago
  48. c72b1c8 Add support for Location (RTC_FROM_HERE) to ProcessThread::RegisterModule. by tommi · 8 years ago
  49. bf77153 Remove usage of VoEVolumeControl from WVoE and Audio[Send|Receive]Stream. by solenberg · 8 years ago
  50. aca8476 Remove workaround for bug 6986 by kwiberg · 8 years ago
  51. 3528419 Fix TSAN race in webrtc::voe::Channel. by hbos · 8 years ago
  52. 1c1bd4d Add probe logging to RtcEventLog. by philipel · 8 years ago
  53. 1f8727f Propagate packet pacing information to SendTimeHistory. by philipel · 8 years ago
  54. 95a13b4 Replace AudioReceiveStream::DeliverRtp with OnRtpPacket. by nisse · 8 years ago
  55. 6bbab56 Reland of Delete class SSRCDatabase, and its global ssrc registry. (patchset #1 id:1 of https://codereview.webrtc.org/2700413002/ ) by nisse · 8 years ago
  56. 223ac94 Rename some variables and methods in RTC event log. by terelius · 8 years ago
  57. 3264fc7 Revert of Delete class SSRCDatabase, and its global ssrc registry. (patchset #20 id:370001 of https://codereview.webrtc.org/2644303002/ ) by kjellander · 8 years ago
  58. 46dc8df Delete class SSRCDatabase, and its global ssrc registry, by nisse · 8 years ago
  59. 60d2fc1 Add logging of delay-based bandwidth estimate. by terelius · 8 years ago
  60. ac16f1b Remove VoEVideoSync interface. by solenberg · 8 years ago
  61. bda935c Remove VoEExternalMedia interface. by solenberg · 8 years ago
  62. 7eb7de8 Remove the unused and untested functions from VoERTP_RTCP. by solenberg · 8 years ago
  63. c10c31d Wire up audio packet loss to BWE. by stefan · 8 years ago
  64. 4fdc30d Reland of "Log audio network adapter decisions in event log." by minyue · 8 years ago
  65. bc2b639 Pass SdpAudioFormat through Channel, without converting to CodecInst by kwiberg · 8 years ago
  66. 5b7322d Revert of Log audio network adapter decisions in event log. (patchset #14 id:320001 of https://codereview.webrtc.org/2559953002/ ) by sakal · 8 years ago
  67. dc916a1 Log audio network adapter decisions in event log. by michaelt · 8 years ago
  68. 22174cb Pass event log to ANA. by michaelt · 8 years ago
  69. 756b219 Update smoothed bitrate. by michaelt · 8 years ago
  70. 0e0db71 Make OverheadObserver::OnOverheadChanged count RTP headers only by nisse · 8 years ago
  71. c9999ff Wire-up audio BWE with overhead. by michaelt · 8 years ago
  72. d7ef04b Move functionality out from AudioFrame and into AudioFrameOperations. by aleloi · 8 years ago
  73. 0d9fb71 Deprecated SetAudioPacketSize from RTPSender and removed calls to it. by ossu · 8 years ago
  74. 2bc91bf Reland "Update rtt on audio only calls". by michaelt · 8 years ago
  75. c01dfe6 Relanding "Pass time constant to bwe smoothing filter." by minyue · 8 years ago
  76. cf7d22e Refactor RMSLevel and give it new functionality by henrik.lundin · 8 years ago
  77. acd8db6 Revert of Pass time constant to bwe smoothing filter. (patchset #8 id:140001 of https://codereview.webrtc.org/2518923003/ ) by ossu · 8 years ago
  78. 4d9c52e Pass time constanct to bwe smoothing filter. by michaelt · 8 years ago
  79. 6b56973 Smooth BWE and pass it to Audio Network Adaptor. by michaelt · 8 years ago
  80. 1cbcced Reland of move RTPPayloadStrategy and simplify RTPPayloadRegistry (patchset #1 id:1 of https://codereview.webrtc.org/2528993002/ ) by magjed · 8 years ago
  81. e82ac9a Revert of Remove RTPPayloadStrategy and simplify RTPPayloadRegistry (patchset #7 id:240001 of https://codereview.webrtc.org/2524923002/ ) by magjed · 8 years ago
  82. 402fe33 Remove RTPPayloadStrategy and simplify RTPPayloadRegistry by magjed · 8 years ago
  83. 4c1eb05 Send audio and video codecs to RTPPayloadRegistry by magjed · 8 years ago
  84. ff9d77c Support multiple timestamp rates for sending DTMF. by solenberg · 8 years ago
  85. dbe2c77 Remove usage of VoEBase::AssociateSendChannel() from WVoMC. by solenberg · 8 years ago
  86. 2454551 Set actual transport overhead in rtp_rtcp by michaelt · 8 years ago
  87. e9523f0 Clean up abs-send-time for audio. by stefan · 8 years ago
  88. 72aebf4 Remove voe::Channel::StopReceive() and associated logic. by solenberg · 8 years ago
  89. 77c17ed Stop using old AudioCodingModule::RegisterReceiveCodec overloads by kwiberg · 8 years ago
  90. fbd0246 Move audio frame memory handling inside AudioMixer. by aleloi · 8 years ago
  91. b4ac6b0 Made AudioReceiveStream a mixer participant. by aleloi · 8 years ago
  92. d8e3fa0 Call OnTransportFeedback just when feedback_observer exist. by michaelt · 8 years ago
  93. 5e0ea59 Revert of Add RtcpRttStats to AudioStream (patchset #1 id:1 of https://codereview.webrtc.org/2402333002/ ) by sprang · 8 years ago
  94. 0d87bce Voe::Channel: Turned GetPlayoutFrequency into GetRtpTimestampRateHz. by ossu · 8 years ago
  95. 2a6e87c Hooking up audio network adaptor to VoE. by minyue · 8 years ago
  96. 6dcc486 Add RtcpRttStats to AudioStream by michaelt · 8 years ago
  97. 19689f4 Added logging for audio send/receive stream configs. by ivoc · 8 years ago
  98. e59b6ff Moved RtcEventLog files from call/ to logging/ by skvlad · 8 years ago
  99. 4e6d4da Revert of Remove unnecessary interface TelephoneEventHandler (patchset #3 id:40001 of https://codereview.webrtc.org/2357583002/ ) by danilchap · 8 years ago
  100. 1bed982 Remove unnecessary interface TelephoneEventHandler. by solenberg · 8 years ago