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gerrit-public.fairphone.software / platform / external / webrtc / 66679dcc04658e5e0dc334f53e647eb9eb3107ed / talk
  1. f888bb5 Support for unmixed remote audio into tracks. by Tommi · 9 years ago
  2. 04e9146 Discard old-generation candidates when ICE restarts by Honghai Zhang · 9 years ago
  3. 822bdf9 Remove cricket::VideoEncoderConfig. by Peter Boström · 9 years ago
  4. 71f5a9a This cl change VideoCaptureAndroid to handle CVO the same way when capturing to texture as when using ordinary byte buffers. by Per · 9 years ago
  5. cf846ad Adding stub files needed for https://codereview.webrtc.org/1507973003/ by Taylor Brandstetter · 9 years ago
  6. 7c73bdb Renaming JsepPeerConnectionP2PTestClient back to P2PTestConductor. by deadbeef · 9 years ago
  7. a1f567a Revert of Free SCTP data channels asynchronously in PeerConnection. (patchset #3 id:40001 of https://codereview.webrtc.org/1492383002/ ) by deadbeef · 9 years ago
  8. 796cfaf Add VideoCodec::PreferDecodeLate by perkj · 9 years ago
  9. c490e01 Implement NativeToI420Buffer in C++, calling java SurfaceTextureHelper, new method .textureToYUV, to by nisse · 9 years ago
  10. 1387149 Adding reduced size RTCP configuration down to the video stream level. by deadbeef · 9 years ago
  11. 434aca8 Add empty placeholder files for remote audio tracks. by tommi · 9 years ago
  12. 7623ce4 Reland of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:300001 of https://codereview.webrtc.org/1507903005/ ) by Peter Boström · 9 years ago
  13. bda7e0b Fixing issue with default stream upon setting 2nd remote description. by deadbeef · 9 years ago
  14. d02b0fa Add oldest rotation type option to RTCFileLogger by haysc · 9 years ago
  15. 1a9d615 Add tracing to public PeerConnection methods. by Peter Boström · 9 years ago
  16. 7b2f762 Don't call SetPreviewFormat if capturing to textures. by perkj · 9 years ago
  17. edd8fef Add new view that renders local video using AVCaptureLayerPreview. by haysc · 9 years ago
  18. 246b817 Refactor handling of AudioOptions. by solenberg · 9 years ago
  19. 8237abf Revert of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:20001 of https://codereview.webrtc.org/1506773002/ ) by kjellander · 9 years ago
  20. 9f45a45 Add tracing to upper-level WebRTC calls. by Peter Boström · 9 years ago
  21. 03ef053 Merge webrtc/video_engine/ into webrtc/video/ by Peter Boström · 9 years ago
  22. 3868692 Free SCTP data channels asynchronously in PeerConnection. by deadbeef · 9 years ago
  23. 46ad542 Revert of "Create rtc::AtomicInt POD struct." (patchset #3 id:40001 of https://codereview.webrtc.org/1498953002/ ) by pbos · 9 years ago
  24. 6f28cf0 Implement standalone event tracing in AppRTCDemo. by Peter Boström · 9 years ago
  25. 84f0970 Reland of "Create rtc::AtomicInt POD struct." by Peter Boström · 9 years ago
  26. cd4003f Use @webrtc.org addresses for OWNERS. by Peter Boström · 9 years ago
  27. cf890bc Roll gtest-parallel. by Peter Boström · 9 years ago
  28. 9d69c3f Return a copy of the supported RTP header extensions instead of a reference. by Stefan Holmer · 9 years ago
  29. b86d4e4 Prepare the AudioSendStream to be hooked up to send-side BWE. by Stefan Holmer · 9 years ago
  30. 03f80eb Refactor EglBase configuration. by nisse · 9 years ago
  31. 1218d7a Allow remote fingerprint update during a call by Guo-wei Shieh · 9 years ago
  32. 86aaa4b Revert "Allow remote fingerprint update during a call" by Guo-wei Shieh · 9 years ago
  33. 9c38c2d Allow remote fingerprint update during a call by Guo-wei Shieh · 9 years ago
  34. 381b421 Ping backup connection at a slower rate by Honghai Zhang · 9 years ago
  35. 9e1b992 Clear old decoders after recreating the receiver. by Peter Boström · 9 years ago
  36. b572768 - Remove calls to VoEDtmf from WVoE/MC. by Fredrik Solenberg · 9 years ago
  37. 1a5cf6e Remove the unused NullMediaEngine (and NullVoiceEngine+NullVideoEngine). by Fredrik Solenberg · 9 years ago
  38. 9cf0c3d Removes MAYBE_ from several test case names in JsepPeerConnectionP2PTestClient. by Ivo Creusen · 9 years ago
  39. 7635684 Fix Mac ObjC PeerConnection API compilation. by tkchin · 9 years ago
  40. 9462052 In some rare Android systems ConnectivityManager may be null. by honghaiz · 9 years ago
  41. 3c28d0d Disable PeerConnectionEndToEndTest.Call on Mac. by kjellander@webrtc.org · 9 years ago
  42. 1d63dd0 - Remove cricket::VoiceChannel::PressDtmf(); AFAICT unused. by solenberg · 9 years ago
  43. ee524f7 Adding Java binding for CreateSender. by deadbeef · 9 years ago
  44. 7e4e01a Add header extension filtering for WebRtcVoiceEngine/MediaChannel. by solenberg · 9 years ago
  45. 2515af2 Removing some unnecessary string manipulation code from VoEBase::GetVersion(). by solenberg · 9 years ago
  46. d20d247 Fix VideoCaptureAndroid, drop frame when switching camera using textures. by perkj · 9 years ago
  47. 226a602 Fix problem when drawing to the Android Media encoder surface. by perkj · 9 years ago
  48. 40455d6 This cl change so that we use EGL14 where it is supported and EGL10 otherwise. The idea is to make this agnostic to an application and for WebRTC except in EGLBase. by perkj · 9 years ago
  49. 41b0798 Adding CreatePeerConnection method that uses new PC Initialize method. by deadbeef · 9 years ago
  50. 0de97f1 WebRtcVideoCapturer: SetCaptureState(CS_STOPPED) on Stop and ensure state changes in unittest. by hbos · 9 years ago
  51. cb9792e Fix VideoCapturerAndroidTest.testStartWhileCameraIsAlreadyOpen on Android M. by perkj · 9 years ago
  52. 14f4144 Add helper KeepRefUntilDone. by perkj · 9 years ago
  53. ee69ed5 Add separate event for camera freeze. by glaznev · 9 years ago
  54. 70c0e29 Disable PeerConnectionEndToEndTest.Call for TSan. by kjellander@webrtc.org · 9 years ago
  55. ae54b83 Android SurfaceViewRenderer: Add resetStatistics() method by magjed · 9 years ago
  56. 2fe1cb0 Don't overwrite audio stats when they're not available. by andrew · 9 years ago
  57. 26c8c91 Using Rent-A-Codec for static Codec access in WVoE/MC. by solenberg · 9 years ago
  58. 727dbc2 VideoCapturerAndroid - allow lower frame rate in bad lightning by Per · 9 years ago
  59. 598242a Support texture scaling in Androids MediaEncoder. by Per · 9 years ago
  60. a3c20bb Add support for scaling textures in AndroidVideoCapturer. by Per · 9 years ago
  61. fac0655 Reland of Adding the ability to create an RtpSender without a track. by deadbeef · 9 years ago
  62. 444682a Remove frame time scheduing in IncomingVideoStream by qiangchen · 9 years ago
  63. b251472 Add JNI interface for functions to start and stop recording AEC dumps and RTC event logs. by ivoc · 9 years ago
  64. 4c5eea3 Android SurfaceViewRenderer: Don't rely on widthSpec/heightSpec after onMeasure() returns by Magnus Jedvert · 9 years ago
  65. 7baf79f Temporary remove spamming MediaDecoder log by perkj · 9 years ago
  66. 4f2152e Android SurfaceViewRenderer: Make sure not to call eglCreateSurface() twice by Magnus Jedvert · 9 years ago
  67. 9237559 Add SurfaceTextureHelper.disconnect(Handler handler) method by perkj · 9 years ago
  68. b5cb19b Fixing direction attribute in answer for non-RTP protocols. by deadbeef · 9 years ago
  69. 05816eb Fix target_arch for ios devices by wr.wllm · 9 years ago
  70. 1aa6efe Android ThreadUtils: Make the class public for access outside org.webrtc by Magnus Jedvert · 9 years ago
  71. 8becec3 talk: remove deprecated *processor.h files by tfarina · 9 years ago
  72. 87d5845 Fix androidmediadecoder_jni TS logging. by perkj · 9 years ago
  73. 43edf0f Require negotiation to send transport cc feedback over RTCP. by stefan · 9 years ago
  74. bd13838 Remove SetVideoLogging/SetAudioLogging from ChannelManager and down the stack. by solenberg · 9 years ago
  75. 5def7b9 Revert of Adding the ability to create an RtpSender without a track. (patchset #3 id:300001 of https://codereview.webrtc.org/1413983004/ ) by deadbeef · 9 years ago
  76. 7add058 Move some receive stream configuration into webrtc::AudioReceiveStream. by solenberg · 9 years ago
  77. 6834fa1 Reland of Adding the ability to create an RtpSender without a track. (patchset #1 id:1 of https://codereview.webrtc.org/1426443007/ ) by deadbeef · 9 years ago
  78. 30e9182 This cl add support to encode from textures to MediaCodecVideoEncoder. by perkj · 9 years ago
  79. 7e63ef0 Allow default audio receive channel to receive on any unsignalled SSRC. by solenberg · 9 years ago
  80. 17c0aff Enable VP9 HW decoder on Exynos chips. by Alex Glaznev · 9 years ago
  81. 7755e20 Chrome has now been updated. by perkj · 9 years ago
  82. 726b1f7 Removed dummy "mediastreamsignaling.h" by perkj · 9 years ago
  83. 191c1f9 Disable all JsepPeerConnectionP2PTestClient tests on Mac due to flakiness on Mac Debug bots. by ivoc · 9 years ago
  84. ef45323 Android: Make classes non-final by Magnus Jedvert · 9 years ago
  85. 1503867 Disabled several JsepPeerConnectionP2PTestClient tests on Mac, due to flakiness on Debug Mac trybots. by ivoc · 9 years ago
  86. b6755ab Revert of Adding thread timeout for audio recorer thread in Java (patchset #2 id:20001 of https://codereview.webrtc.org/1444313002/ ) by henrika · 9 years ago
  87. 488e75f Patchset 1 yet again relands without modification https://codereview.webrtc.org/1422963003/ by Per · 9 years ago
  88. 521ed7b Reland Convert internal representation of Srtp cryptos from string to int by Guo-wei Shieh · 9 years ago
  89. 318166b Revert of Convert internal representation of Srtp cryptos from string to int. (patchset #10 id:180001 of https://codereview.webrtc.org/1416673006/ ) by guoweis · 9 years ago
  90. 2764e10 Convert internal representation of Srtp cryptos from string to int. by guoweis · 9 years ago
  91. b7ce964 modules/video_coding/utility: Remove include by kjellander@webrtc.org · 9 years ago
  92. ad948c4 Preliminary support of VP9 HW encoder on Android. by Alex Glaznev · 9 years ago
  93. 2557b86 modules/video_coding refactorings by Henrik Kjellander · 9 years ago
  94. 4dd7a65 Temporarily disable VERIFY while bug is investigated. by phoglund · 9 years ago
  95. 2aff615 Remove spammy logging of RTCP delivery failures. by Peter Boström · 9 years ago
  96. fd614c2 Adding thread timeout for audio recorer thread in Java by henrika · 9 years ago
  97. 6f8ce06 common_video: rename interface -> include by kjellander · 9 years ago
  98. 482b12e Remove BundleFilter filtering of RTCP. by pbos · 9 years ago
  99. 3a94154 Move some send stream configuration into webrtc::AudioSendStream. by solenberg · 9 years ago
  100. 633a3aa ThreadUtils: Add joinUninterruptibly() with timeout by magjed · 9 years ago