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webrtc / src / 57c1ad3b16b222d5e63bf94d96a32c652e2e59d8 / webrtc
  1. 57c1ad3 Don't declare function arguments of array type by kwiberg · 8 years ago
  2. 6280960 Correctly pass drawn frame size when layout aspect ratio is used in EglRenderer. by sakal · 8 years ago
  3. 96c1587 RtpPacket::payload() return rtc::ArrayView instead of raw pointer by danilchap · 8 years ago
  4. f880285 iOS: Cleanup buildbot JSON files + bump iOS version to 10.0 by kjellander · 8 years ago
  5. 3898944 Remove unused files linux.cc/.h and linuxfdwalk.c/.h. by solenberg · 8 years ago
  6. 2184155 Add more logging in ScreenCapturerIntegrationTest by zijiehe · 8 years ago
  7. ed9dccf Revert of Remove unused HttpClient class. (patchset #1 id:1 of https://codereview.webrtc.org/2511883005/ ) by honghaiz · 8 years ago
  8. 4a698f6 Remove unused HttpClient class. by solenberg · 8 years ago
  9. 01af3a3 Remove unused dbus.cc/.h and related things. by solenberg · 8 years ago
  10. 90c024f Move FirewallSocketServer to test code. by nisse · 8 years ago
  11. 00f2ee0 Changed the way we find the ProjectRootPath. by ehmaldonado · 8 years ago
  12. dedaf1c Modify audio_processing_unittest to use ResourcePath instead of ProjectRootPath. by ehmaldonado · 8 years ago
  13. bbc747c Delete WindowPicker class and subclasses. by nisse · 8 years ago
  14. 76b3049 Changed the interface AudioMixer::RemoveSource to have a void return type. by aleloi · 8 years ago
  15. a28780e Introduce ArrayView::subview function to return portion of the original view by danilchap · 8 years ago
  16. 509e4fe 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
  17. d7ac0a9 Revert of Move smoothing filter to common audio. (patchset #3 id:60001 of https://codereview.webrtc.org/2484153002/ ) by magjed · 8 years ago
  18. a82395b Move smoothing filter to common audio. by michaelt · 8 years ago
  19. 610c454 Add Datachannel support to Android AppRTCMobile by hekra01 · 8 years ago
  20. 1acfbd2 Expose RtpCodecParameters to VoiceMediaInfo stats. by hbos · 8 years ago
  21. 7b9feee Fix PayloadRouter::OnEncodedImage() to handle errors properly. by sergeyu · 8 years ago
  22. 81c3a03 Added a callback function OnAddTrack to PeerConnectionObserver by zhihuang · 8 years ago
  23. 5b93db2 iOS: Add AudioSendSideBwe field trial. by tkchin · 8 years ago
  24. eacbaea 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
  25. 0d0d753 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
  26. de49803 MB: Add new perf desktop bots and remove DCHECK from Android perf by kjellander · 8 years ago
  27. aae7e7c Split out target rtc_media_base from rtc_media by magjed · 8 years ago
  28. 765edc3 Update the alpha value in the echo detector. by ivoc · 8 years ago
  29. 42043b9 Stop using hardcoded payload types for video codecs by Magnus Jedvert · 8 years ago
  30. 10111bc Passed AudioMixer to AudioState::Config. by aleloi · 8 years ago
  31. dd31071 Added an empty AudioTransportProxy to AudioState. by aleloi · 8 years ago
  32. d4adce4 Remove Absolute Send Time from list of supported header extensions for audio streams. by solenberg · 8 years ago
  33. fbb374d Add a reliability term to the echo detector. by ivoc · 8 years ago
  34. d51c4dc Delete unused files httprequest.h and httprequest.cc. by nisse · 8 years ago
  35. ffbbcac Support multiple timestamp rates for sending DTMF. by solenberg · 8 years ago
  36. 2779bab Support receiving DTMF for multiple RTP clock rates. by solenberg · 8 years ago
  37. fbfb536 Explicitly enable RED over RTX in rampup tests. by brandtr · 8 years ago
  38. afaef8b Add a new overuse estimator for the delay based BWE behind experiment. by terelius · 8 years ago
  39. b7e7b49 Use NtpTime in RtcpMeasurement instead of uint sec/uint frac. by asapersson · 8 years ago
  40. 4da3044 Add overhead per packet observer to the rtp_sender. by michaelt · 8 years ago
  41. 4a4b3cf Add interval estimator to remote bitrate estimator. by michaelt · 8 years ago
  42. 377b60c Only enable residual echo detector when needed in level controller perf tests. by ivoc · 8 years ago
  43. 0bff12a Renamed -red to -ed and -red_graph to -ed_graph in audioproc_f. by ivoc · 8 years ago
  44. 9af2b60 Propagate bitrate setting to RTCRtpSender. by denicija · 8 years ago
  45. a62f582 Integrate FlexFEC in video_loopback. by brandtr · 8 years ago
  46. dd369c6 Reduce full stack test time to 45 secs and add H264 and FlexFEC. by brandtr · 8 years ago
  47. 527d347 Reland of Declare VideoCodec.codec_specific_info private (patchset #1 id:1 of https://codereview.webrtc.org/2491613005/ ) by hta · 8 years ago
  48. 05f845d Replace c-style cast and constrain value in VCMFecMethod::ProtectionFactor. by brandtr · 8 years ago
  49. 39f9729 Add VideoSendStreamTest for FlexFEC. by brandtr · 8 years ago
  50. 1293aca Configure FlexFEC in VideoQualityTest. by brandtr · 8 years ago
  51. 1e3dfbf Add FlexFEC end-to-end test. by brandtr · 8 years ago
  52. 46c7389 Adding GetConfiguration to PeerConnection. by deadbeef · 8 years ago
  53. 0182f85 More reliable ALR detection by Sergey Ulanov · 8 years ago
  54. 3a1c40a MB: Remove configuration for unexisting bots. by ehmaldonado · 8 years ago
  55. b4af3d6 Remove all references to GYP by Henrik Kjellander · 8 years ago
  56. 67fcad8 Relax the PostDelayed expectations a little more to address flakiness. by tommi · 8 years ago
  57. 08127a9 Reland #2 of Issue 2434073003: Extract bitrate allocation ... by Erik Språng · 8 years ago
  58. 779017d Adds stereo support for Java-based input and output audio on Android by henrika · 8 years ago
  59. 007cdb5 Better delete of file in loopback script by mandermo · 8 years ago
  60. 613152a Add a JNI boot test to catch ARM dynamic linker regressions. by phoglund · 8 years ago
  61. a814941 Fix unit for logged bitrates at the end of a call. by Åsa Persson · 8 years ago
  62. 725e484 Use different RTX payload types for different H264 profiles by magjed · 8 years ago
  63. 906c5dc Revert of Start probes only after network is connected. (patchset #9 id:240001 of https://codereview.webrtc.org/2458863002/ ) by honghaiz · 8 years ago
  64. e19649b Fix Android lint error. by sakal · 8 years ago
  65. 5c99c76 Start probes only after network is connected. by sergeyu · 8 years ago
  66. b2fcf6d MB: Run test with gtest-parallel on swarming. by ehmaldonado · 8 years ago
  67. 2a3eb9f mac: Fix screen capture on secondary displays. by erikchen · 8 years ago
  68. 5d54e18 Prepare iOS H264 HW encoder for High Profile by magjed · 8 years ago
  69. 4aecc58 Simplify creating RtpHeaderExtensionMap in EventLogAnalyzer by danilchap · 8 years ago
  70. cd188f6 Make SendStatisticsProxy let through FlexFEC packets. by brandtr · 8 years ago
  71. 43cb716 Add ToString method to AggregatedStats and log stats at the end of a call. by asapersson · 8 years ago
  72. 841de6a Add FlexFEC to CallTest. by brandtr · 8 years ago
  73. 985d280 Add support for field trials to event log visualizer. by stefan · 8 years ago
  74. 614d5b7 Move VideoEncoderSoftwareFallbackWrapper from webrtc/video_encoder.h to webrtc/media/engine/ by magjed · 8 years ago
  75. 92fd8e6 Removes usage of system_wrappers/include/clock.h in audio_device/ by henrika · 8 years ago
  76. 43c31e7 Make configuration logic harsher in FlexfecReceiveStream. by brandtr · 8 years ago
  77. e950cad Wire up FlexfecSender in RTP module and VideoSendStream. by brandtr · 8 years ago
  78. 20270be Make sure that multiband processing is active when the residual echo detector is active. by ivoc · 8 years ago
  79. b2b61b3 Rename the adapt audio bitrate experiment. by stefan · 8 years ago
  80. b829d9f Add AudioOption for residual echo detector, and enable the echo detector by default on non-mobile platforms. by ivoc · 8 years ago
  81. 79dfdad Avoid left-shifting negative values in a number of places by henrik.lundin · 8 years ago
  82. fd5a20f New jitter buffer experiment. by philipel · 8 years ago
  83. 77bfd7c Add ARDSettingsModelTests to apprtcmobile_test target. by denicija · 8 years ago
  84. 8bc9326 DirectX capturer flickers on the second monitor by zijiehe · 8 years ago
  85. 69a0e3e Use a default mouse cursor if XFixes is not supported. by jamiewalch · 8 years ago
  86. 7602aab Remove usage of VoEBase::AssociateSendChannel() from WVoMC. by solenberg · 8 years ago
  87. 26b6756 Fix BitrateControllerImpl not to ignore BW probe results mid-call. by sergeyu · 8 years ago
  88. 80c06fa NetEq: Don't interpolate longer than the output size by henrik.lundin · 8 years ago
  89. 87d1a78 Add support to audioproc_f for running the residual echo detector and producing an echo likelihood graph. by ivoc · 8 years ago
  90. 9e795c6 Update RTPSender::IsFecPacket for FlexFEC. by brandtr · 8 years ago
  91. 9dfff29 Make FlexFEC packets paceable through RTPSender. by brandtr · 8 years ago
  92. 7aba029 Make use of new APM statistics interface. by ivoc · 8 years ago
  93. 25b57ce Update header formatters to FlexFEC draft 03. by brandtr · 8 years ago
  94. 445fb8f Use correct define in H264 end-to-end tests. by brandtr · 8 years ago
  95. 8e75a52 Explicitly use RTX for RED in VideoQualityTest and video_loopback. by brandtr · 8 years ago
  96. a65704b Expose RtpCodecParameters to VideoMediaInfo stats. by hbos · 8 years ago
  97. 82ebe02 Correct stats for RTCPeerConnectionStats.dataChannels[Opened/Closed]. by hbos · 8 years ago
  98. 6f0b9fd Allowing resetting of AudioNetworkAdaptor in AudioSendStream. by minyue · 8 years ago
  99. f823ede Negotiate H264 profiles in SDP by magjed · 8 years ago
  100. 13ceeea Revert of H.264 packetization mode 0 (try 2) (patchset #27 id:520001 of https://codereview.webrtc.org/2337453002/ ) by magjed · 8 years ago