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webrtc / src / webrtc / cc7862d112bfbb4ba112a0380c406c99955971b4
  1. cc7862d Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ ) by magjed · 8 years ago
  2. 3fde4a9 Revert of Move smoothing filter to common audio. (patchset #3 id:60001 of https://codereview.webrtc.org/2484153002/ ) by magjed · 8 years ago
  3. cac1a8f Move smoothing filter to common audio. by michaelt · 8 years ago
  4. e9dbdd1 Add Datachannel support to Android AppRTCMobile by hekra01 · 8 years ago
  5. 9a1d49f Expose RtpCodecParameters to VoiceMediaInfo stats. by hbos · 8 years ago
  6. 62d59ce Fix PayloadRouter::OnEncodedImage() to handle errors properly. by sergeyu · 8 years ago
  7. 3f3a686 Added a callback function OnAddTrack to PeerConnectionObserver by zhihuang · 8 years ago
  8. 5f01bcc iOS: Add AudioSendSideBwe field trial. by tkchin · 8 years ago
  9. bc59f39 Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ ) by magjed · 8 years ago
  10. 11a53a5 Revert of Split out target rtc_media_base from rtc_media (patchset #3 id:40001 of https://codereview.webrtc.org/2471573003/ ) by magjed · 8 years ago
  11. a2033f4 MB: Add new perf desktop bots and remove DCHECK from Android perf by kjellander · 8 years ago
  12. 614a1fb Split out target rtc_media_base from rtc_media by magjed · 8 years ago
  13. 2942121 Update the alpha value in the echo detector. by ivoc · 8 years ago
  14. fac364f Stop using hardcoded payload types for video codecs by Magnus Jedvert · 8 years ago
  15. d85b51d Passed AudioMixer to AudioState::Config. by aleloi · 8 years ago
  16. 7713f33 Added an empty AudioTransportProxy to AudioState. by aleloi · 8 years ago
  17. ac14ca2 Remove Absolute Send Time from list of supported header extensions for audio streams. by solenberg · 8 years ago
  18. ea541cf Add a reliability term to the echo detector. by ivoc · 8 years ago
  19. f24b273 Delete unused files httprequest.h and httprequest.cc. by nisse · 8 years ago
  20. ff9d77c Support multiple timestamp rates for sending DTMF. by solenberg · 8 years ago
  21. cb8ce03 Support receiving DTMF for multiple RTP clock rates. by solenberg · 8 years ago
  22. 5e5f5cb Explicitly enable RED over RTX in rampup tests. by brandtr · 8 years ago
  23. faa6c4d Add a new overuse estimator for the delay based BWE behind experiment. by terelius · 8 years ago
  24. 08ae82e Use NtpTime in RtcpMeasurement instead of uint sec/uint frac. by asapersson · 8 years ago
  25. 57beae3 Add overhead per packet observer to the rtp_sender. by michaelt · 8 years ago
  26. 57cbf49 Add interval estimator to remote bitrate estimator. by michaelt · 8 years ago
  27. a0ae62b Only enable residual echo detector when needed in level controller perf tests. by ivoc · 8 years ago
  28. 251d1da Renamed -red to -ed and -red_graph to -ed_graph in audioproc_f. by ivoc · 8 years ago
  29. 826436a Propagate bitrate setting to RTCRtpSender. by denicija · 8 years ago
  30. ff63b34 Integrate FlexFEC in video_loopback. by brandtr · 8 years ago
  31. c87bfde Reduce full stack test time to 45 secs and add H264 and FlexFEC. by brandtr · 8 years ago
  32. 8ee7814 Reland of Declare VideoCodec.codec_specific_info private (patchset #1 id:1 of https://codereview.webrtc.org/2491613005/ ) by hta · 8 years ago
  33. be17d21 Replace c-style cast and constrain value in VCMFecMethod::ProtectionFactor. by brandtr · 8 years ago
  34. 9b90f60 Add VideoSendStreamTest for FlexFEC. by brandtr · 8 years ago
  35. 9b3747e Configure FlexFEC in VideoQualityTest. by brandtr · 8 years ago
  36. 87777cb Add FlexFEC end-to-end test. by brandtr · 8 years ago
  37. 4d0d012 Adding GetConfiguration to PeerConnection. by deadbeef · 8 years ago
  38. 7c36a75 More reliable ALR detection by Sergey Ulanov · 8 years ago
  39. 9daf602 MB: Remove configuration for unexisting bots. by ehmaldonado · 8 years ago
  40. abef9e9 Remove all references to GYP by Henrik Kjellander · 8 years ago
  41. a8bd6b6 Relax the PostDelayed expectations a little more to address flakiness. by tommi · 8 years ago
  42. fafcfc0 Reland #2 of Issue 2434073003: Extract bitrate allocation ... by Erik Språng · 8 years ago
  43. 709aab7 Adds stereo support for Java-based input and output audio on Android by henrika · 8 years ago
  44. 9a67d0b Better delete of file in loopback script by mandermo · 8 years ago
  45. fa4cf85 Add a JNI boot test to catch ARM dynamic linker regressions. by phoglund · 8 years ago
  46. d24ad38 Fix unit for logged bitrates at the end of a call. by Åsa Persson · 8 years ago
  47. f815e5d Use different RTX payload types for different H264 profiles by magjed · 8 years ago
  48. 3b1361e Revert of Start probes only after network is connected. (patchset #9 id:240001 of https://codereview.webrtc.org/2458863002/ ) by honghaiz · 8 years ago
  49. 9e178ab Fix Android lint error. by sakal · 8 years ago
  50. b07851d Start probes only after network is connected. by sergeyu · 8 years ago
  51. 94b1c9b MB: Run test with gtest-parallel on swarming. by ehmaldonado · 8 years ago
  52. feec627 mac: Fix screen capture on secondary displays. by erikchen · 8 years ago
  53. 2e3a8c6 Prepare iOS H264 HW encoder for High Profile by magjed · 8 years ago
  54. e550306 Simplify creating RtpHeaderExtensionMap in EventLogAnalyzer by danilchap · 8 years ago
  55. f2bbb29 Make SendStatisticsProxy let through FlexFEC packets. by brandtr · 8 years ago
  56. 7bdb258 Add ToString method to AggregatedStats and log stats at the end of a call. by asapersson · 8 years ago
  57. 256593c Add FlexFEC to CallTest. by brandtr · 8 years ago
  58. cdd2ac3 Add support for field trials to event log visualizer. by stefan · 8 years ago
  59. a98271c Move VideoEncoderSoftwareFallbackWrapper from webrtc/video_encoder.h to webrtc/media/engine/ by magjed · 8 years ago
  60. d756112 Removes usage of system_wrappers/include/clock.h in audio_device/ by henrika · 8 years ago
  61. 6c39963 Make configuration logic harsher in FlexfecReceiveStream. by brandtr · 8 years ago
  62. 6b820ad Wire up FlexfecSender in RTP module and VideoSendStream. by brandtr · 8 years ago
  63. 453cc42 Make sure that multiband processing is active when the residual echo detector is active. by ivoc · 8 years ago
  64. ab81d98 Rename the adapt audio bitrate experiment. by stefan · 8 years ago
  65. ca1d233 Add AudioOption for residual echo detector, and enable the echo detector by default on non-mobile platforms. by ivoc · 8 years ago
  66. fa09b17 Avoid left-shifting negative values in a number of places by henrik.lundin · 8 years ago
  67. ba3a19d New jitter buffer experiment. by philipel · 8 years ago
  68. 8de1842 Add ARDSettingsModelTests to apprtcmobile_test target. by denicija · 8 years ago
  69. 067f3ae DirectX capturer flickers on the second monitor by zijiehe · 8 years ago
  70. d3ada78 Use a default mouse cursor if XFixes is not supported. by jamiewalch · 8 years ago
  71. dbe2c77 Remove usage of VoEBase::AssociateSendChannel() from WVoMC. by solenberg · 8 years ago
  72. 3359df0 Fix BitrateControllerImpl not to ignore BW probe results mid-call. by sergeyu · 8 years ago
  73. b7c5e52 NetEq: Don't interpolate longer than the output size by henrik.lundin · 8 years ago
  74. a6cd561 Add support to audioproc_f for running the residual echo detector and producing an echo likelihood graph. by ivoc · 8 years ago
  75. 36f224d Update RTPSender::IsFecPacket for FlexFEC. by brandtr · 8 years ago
  76. dc4b6e0 Make FlexFEC packets paceable through RTPSender. by brandtr · 8 years ago
  77. 2c7aecc Make use of new APM statistics interface. by ivoc · 8 years ago
  78. eae6019 Update header formatters to FlexFEC draft 03. by brandtr · 8 years ago
  79. c364057 Use correct define in H264 end-to-end tests. by brandtr · 8 years ago
  80. 4968913 Explicitly use RTX for RED in VideoQualityTest and video_loopback. by brandtr · 8 years ago
  81. ae9cdd5 Expose RtpCodecParameters to VideoMediaInfo stats. by hbos · 8 years ago
  82. 0177daa Correct stats for RTCPeerConnectionStats.dataChannels[Opened/Closed]. by hbos · 8 years ago
  83. a7f3617 Allowing resetting of AudioNetworkAdaptor in AudioSendStream. by minyue · 8 years ago
  84. 155a5b4 Negotiate H264 profiles in SDP by magjed · 8 years ago
  85. 5403adc Revert of H.264 packetization mode 0 (try 2) (patchset #27 id:520001 of https://codereview.webrtc.org/2337453002/ ) by magjed · 8 years ago
  86. ea2de0e Remove screen_capturer_mock_objects.h by zijiehe · 8 years ago
  87. 6510335 Crash in DirectX capturer by zijiehe · 8 years ago
  88. 5c35113 CroppingWindowCapturer::CreateCapturer() function to replace raw pointer version by zijiehe · 8 years ago
  89. 0a94a8c Remove unused warning suppression by kthelgason · 8 years ago
  90. 8ca3481 Revert of Declare VideoCodec.codec_specific_info private (patchset #1 id:1 of https://codereview.webrtc.org/2494683006/ ) by hta · 8 years ago
  91. 4f4e859 Reland of Declare VideoCodec.codec_specific_info private (patchset #1 id:1 of https://codereview.webrtc.org/2491933002/ ) by hta · 8 years ago
  92. 8261e17 Optimize FindCodecById and ReferencedCodecsMatch by magjed · 8 years ago
  93. e843185 Reland of Delete all use of cricket::VideoFrame and cricket::WebRtcVideoFrame. (patchset #1 id:1 of https://codereview.webrtc.org/2471783002/ ) by nisse · 8 years ago
  94. c1d9346 Remove RED/RTX workaround from sender/receiver and VideoEngine2. by brandtr · 8 years ago
  95. 48bb72d Ensures that AudioDeviceBuffer::StopPeriodicLogging works as intended. by henrika · 8 years ago
  96. df15b3d Reduce taking locks in RTPSenderVideo::SendVideo by danilchap · 8 years ago
  97. d311569 MB: Add new Win8 and Win10 bots. by Henrik Kjellander · 8 years ago
  98. 99d036c Remove ScreenCapturer and WindowCapturer by zijiehe · 8 years ago
  99. 9b86501 Implement H.264 packetization mode 0. by hta · 8 years ago
  100. 15f2f09 Re-enable the P2PTransportChannelMultihomedTest.TestBasic by zhihuang · 8 years ago