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webrtc / src.git / e5c4a810e21e74acec09c987f20fb0e6788062fd / webrtc / call
  1. e5c4a81 Move RTP keep-alive config from VideoSendStream::Config to Call::Config by sprang · 8 years ago
  2. 370dd47 Revert of Remove remains of webrtc/base (patchset #7 id:120001 of https://codereview.webrtc.org/2973183002/ ) by ehmaldonado · 8 years ago
  3. 9483b49 Remove remains of webrtc/base by ehmaldonado · 8 years ago
  4. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 8 years ago
  5. 0d7f04d Reland of Add received audio/video call duration metrics based on packets. by saza · 8 years ago
  6. 382f21c Revert of Add received audio and video call duration metrics based on packets. (patchset #4 id:140001 of https://codereview.webrtc.org/2957073002/ ) by saza · 8 years ago
  7. 7467492 Add received audio/video call duration metrics based on packets. by saza · 8 years ago
  8. 2a2b297 Add underscore at end of Call members' names by eladalon · 8 years ago
  9. 7ab7fd6 Fix gmock warnings emanating from FlexfecReceiveStreamTest by eladalon · 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. 9addbeb Remove RtpDemuxer tweak for preventing multiple RSID inspections by eladalon · 8 years ago
  13. a9cc40b Allow an external audio processing module to be used in WebRTC by peah · 8 years ago
  14. c3e3e60 nit: Rename RtpDemuxer::sink_ to RtpDemuxer::ssrc_sinks_ by eladalon · 8 years ago
  15. 4847ae6 Reland of Periodically update codec bit/frame rate settings. by sprang · 8 years ago
  16. a52722f Reland of Create RtcpDemuxer (patchset #1 id:1 of https://codereview.webrtc.org/2957763002/ ) by eladalon · 8 years ago
  17. 0e7e786 Revert of Create RtcpDemuxer (patchset #13 id:240001 of https://codereview.webrtc.org/2943693003/ ) by guidou · 8 years ago
  18. cb83bdf Create RtcpDemuxer. Capabilities: by eladalon · 8 years ago
  19. 0f15f92 Introduce RtpStreamReceiverInterface and RtpStreamReceiverControllerInterface. by nisse · 8 years ago
  20. 38ede13 Support building WebRTC without audio and video. by zhihuang · 8 years ago
  21. a5e0df6 Move MinPositive to call.h as discussed here: https://codereview.chromium.org/2888303005/#msg19 by zstein · 8 years ago
  22. dea075c Log an error in RtpDemuxer::FindSsrcAssociations() if kMaxProcessedSsrcs exceeded by eladalon · 8 years ago
  23. 84b4d2c Use rtp_header_extension_map.h instead of rtp_header_extension.h by Danil Chapovalov · 8 years ago
  24. f184138 s/WebRtcVideoChannel2/WebRtcVideoChannel and s/WebRtcVideoEngine2/WebRtcVideoEngine by eladalon · 8 years ago
  25. d0244c2 Add RSID-based demuxing to RtpDemuxer by eladalon · 8 years ago
  26. 4b97980 Relanding: Adds PeerConnectionInterface::UpdateCallBitrate to give clients more control of the bandwidth estimator. PeerConnection implements this method by passing a BitrateConfigMask to its associated Call, which is combined with the existing BitrateConfig and passed on to the SendSideCongestionController as necessary. The existing BitrateConfig generally comes from the x-google-{min,start,max}-bitrate params in the SDP. by zstein · 8 years ago
  27. 441718e Revert of Add PeerConnectionInterface::UpdateCallBitrate. (patchset #7 id:120001 of https://codereview.webrtc.org/2888303005/ ) by charujain · 8 years ago
  28. 9641c13 Adds PeerConnectionInterface::UpdateCallBitrate to give clients more control of the bandwidth estimator. PeerConnection implements this method by passing a BitrateConfigMask to its associated Call, which is combined with the existing BitrateConfig and passed on to the SendSideCongestionController as necessary. The existing BitrateConfig generally comes from the x-google-{min,start,max}-bitrate params in the SDP. by zstein · 8 years ago
  29. d76b7b2 New targets call:rtp_interfaces, call:rtp_receiver, call:rtp_sender. by nisse · 8 years ago
  30. 760a076 Create unit tests for RtpDemuxer by eladalon · 8 years ago
  31. 76e62b0 Address some violations of chromium-style. by nisse · 8 years ago
  32. 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
  33. c3d4b48 Store/restore RTP state for audio streams with same SSRC within a call by ossu · 8 years ago
  34. f472699 Replace AudioSendStream::Config with rtclog::StreamConfig. by perkj · 8 years ago
  35. ac8f52d Replace AudioReceiveStream::Config with rtclog::StreamConfig. by perkj · 8 years ago
  36. c0876aa Replace VideoSendStream::Config with new rtclog::StreamConfig in RtcEventLog. by perkj · 8 years ago
  37. 09e71da Replace VideoReceiveStream::Config with new rtclog::StreamConfig in RtcEventLog. by perkj · 8 years ago
  38. eed52bf New class RtxReceiveStream. by nisse · 8 years ago
  39. 93e4522 Renaming probing_interval to bwe_period globally. by minyue · 8 years ago
  40. 8c96a14 Simple tests for Call::SetBitrateConfig. by zstein · 8 years ago
  41. e4bcd6d New class RtpDemuxer and RtpPacketSinkInterface, use in Call. by nisse · 8 years ago
  42. d2ef314 Make Call::OnRecoveredPacket parse RTP header and call OnRtpPacket. by nisse · 8 years ago
  43. 2d9d21f Add untracked headers in modules/rtp_rtcp by danilchap · 8 years ago
  44. c467520 Delete helper class MediaTypePacketReceiver. by nisse · 8 years ago
  45. 7cb69d5 This will allow me to test that Call invokes SendSideCongestionController::SetBweBitrates as expected (for https://codereview.chromium.org/2793913008). by zstein · 8 years ago
  46. eb1fde4 Injectable audio encoders: Moved audio encoder, factory and builtin factory to api/. by ossu · 8 years ago
  47. 20a4b3f Injectable audio encoders: WebRtcVoiceEngine and company by ossu · 8 years ago
  48. 81c79f5 Creating webrtc:video_stream_api by mbonadei · 8 years ago
  49. e0629c0 GN: Tighten up test target visibility + refactorings by kjellander · 8 years ago
  50. b8a654c Delete declaration of non-existing function webrtc::Version(). by nisse · 8 years ago
  51. cae45d0 Move RtpTransportControllerSend to a new file. by nisse · 8 years ago
  52. fc5e81c Replace first_packet_sent_ms_ in Call. by asapersson · 8 years ago
  53. 0584331 Delete VieRemb class, move functionality to PacketRouter. by nisse · 8 years ago
  54. 20c84cc Making FakeNetworkPipe demux audio and video packets. by minyue · 8 years ago
  55. 8d609f6 Reland of Implemented the GetSources() in native code. (patchset #1 id:1 of https://codereview.webrtc.org/2809613002/ ) by hbos · 8 years ago
  56. 37e99fd Move AudioDecoder and AudioDecoderFactory mocks to webrtc/test/ by kwiberg · 8 years ago
  57. fbcc5cb Revert of Implemented the GetSources() in native code. (patchset #11 id:510001 of https://codereview.webrtc.org/2770233003/ ) by olka · 8 years ago
  58. 4fb651d Event log cleanup in tests. by philipel · 8 years ago
  59. fca900a Fix two invalid DCHECKs related to audio BWE. by stefan · 8 years ago
  60. 292084c Added the GetSources() to the RtpReceiverInterface and implemented by zhihuang · 8 years ago
  61. 6167b26 Make RtpTransportControllerSend::send_side_cc_ a direct member. by nisse · 8 years ago
  62. d8ce1e1 Move SelectMediaType from RampUpTester to BaseTest. by nisse · 8 years ago
  63. c5d62e2 Reland of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #1 id:1 of https://codereview.webrtc.org/2783183003/ ) by sprang · 8 years ago
  64. 76d9c9c Reland of Enable trendline experiment and bayesian bitrate estimator experiment by default. by stefan · 8 years ago
  65. 029f7cc Revert of Enable trendline experiment and bayesian bitrate estimator experiment by default. (patchset #6 id:100001 of https://codereview.webrtc.org/2777333003/ ) by lliuu · 8 years ago
  66. 27925de Enable trendline experiment and bayesian bitrate estimator experiment by default. by stefan · 8 years ago
  67. f9ed235 Revert of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #8 id:410001 of https://codereview.webrtc.org/2781433002/ ) by lliuu · 8 years ago
  68. 7a3615b Revert of Enable the bayesian bitrate estimator by default. (patchset #5 id:80001 of https://codereview.webrtc.org/2749803002/ ) by lliuu · 8 years ago
  69. c53a17f Enable the bayesian bitrate estimator by default. by stefan · 8 years ago
  70. 3ea3c77 Reland of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #1 id:1 of https://codereview.webrtc.org/2764133002/ ) by sprang · 8 years ago
  71. 0ffdcc5 Delete unneeded includes of deprecated system_wrappers include files. by nisse · 8 years ago
  72. e5ad5ca Reland of Don't hardcode MediaType::ANY in FakeNetworkPipe. (patchset #1 id:1 of https://codereview.webrtc.org/2784543002/ ) by nisse · 8 years ago
  73. 3a3bd50 Revert of Don't hardcode MediaType::ANY in FakeNetworkPipe. (patchset #4 id:60001 of https://codereview.webrtc.org/2774463003/ ) by lliuu · 8 years ago
  74. 9c47b00 Don't hardcode MediaType::ANY in FakeNetworkPipe. by nisse · 8 years ago
  75. bcbaf74 Let Call register ReceiveSideCongestionController as CallStatsObserver. by nisse · 8 years ago
  76. 1c07c70 Reland "WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType" by kwiberg · 8 years ago
  77. b8f9a32 Define RtpTransportControllerSendInterface. by nisse · 8 years ago
  78. 670a7f3 Revert of WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType (patchset #13 id:260001 of https://codereview.webrtc.org/2686043006/ ) by kwiberg · 8 years ago
  79. 1724cfb WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType by kwiberg · 8 years ago
  80. 8b45b11 Revert of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #14 id:250001 of https://codereview.webrtc.org/2716643002/ ) by skvlad · 8 years ago
  81. 72acf25 Add framerate to VideoSinkWants and ability to signal on overuse by sprang · 8 years ago
  82. 559af38 Split CongestionController into send- and receive-side classes. by nisse · 8 years ago
  83. 14adba7 Don't allocate any RTPSender object for a receive only RtpRtcp module. by nisse · 8 years ago
  84. baded15 Reland of write frame generator capturer to use TaskQueue instead of EventTimeWrapper (patchset #1 id:1 of https://codereview.webrtc.org/2751063005/ ) by ilnik · 8 years ago
  85. 2a420ce Revert of write frame generator capturer to use TaskQueue instead of EventTimeWrapper (patchset #11 id:300001 of https://codereview.webrtc.org/2750473002/ ) by ilnik · 8 years ago
  86. 2549ad4 Reland of write frame generator capturer to use TaskQueue instead of EventTimeWrapper (patchset #1 id:1 of https://codereview.webrtc.org/2748643002/ ) by ilnik · 8 years ago
  87. 9ea46b5 Ignore packets sent on old network route when receiving feedback. by Stefan Holmer · 8 years ago
  88. a514584 Add the ability to read/write to WAV files in FakeAudioDevice by oprypin · 8 years ago
  89. 382a72a Revert of rewrite frame generator capturer to use TaskQueue instead of EventTimeWrapper (patchset #2 id:90001 of https://codereview.webrtc.org/2744003002/ ) by ilnik · 8 years ago
  90. ff2ebf5 Clean up perf metrics and report ramp-up stats for fewer tests. by stefan · 8 years ago
  91. 8c0a589 Reland of rewrite frame generator capturer to use TaskQueue instead of EventTimeWrapper (patchset #1 id:1 of https://codereview.webrtc.org/2743993002/ ) by ilnik · 8 years ago
  92. 1cb27c2 Revert of rewrite frame generator capturer to use TaskQueue instead of EventTimeWrapper (patchset #2 id:70001 of https://codereview.webrtc.org/2745583006/ ) by ilnik · 8 years ago
  93. b007425 Reland of rewrite frame generator capturer to use TaskQueue instead of EventTimeWrapper. by ilnik · 8 years ago
  94. 45b5fe5 Don't report perf metrics for packet loss ramp-up tests. by stefan · 8 years ago
  95. c69385d Add |protected_by_flexfec| flag to VideoReceiveStream::Config. by nisse · 8 years ago
  96. dea489f Add support for Location (RTC_FROM_HERE) to ProcessThread::RegisterModule. by tommi · 8 years ago
  97. 796b8f9 Remove usage of VoEVolumeControl from WVoE and Audio[Send|Receive]Stream. by solenberg · 8 years ago
  98. e5d3a3e Fix quick perf test setting that was accidentally inverted. by sprang · 8 years ago
  99. 16ccfdf Reland Move fake_audio_device to its own target. by perkj · 8 years ago
  100. c1b57a1 Test field trial group with startswith rather than equals. by sprang · 8 years ago