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gerrit-public.fairphone.software / platform / external / webrtc / refs/heads/int/11/fp3 / talk
  1. 2622ea7 Leave only an empty top level OWNERS file. by Chih-Hung Hsieh · 7 years ago
  2. fcfc804 Eliminate defines in talk/ by kjellander · 8 years ago
  3. 3542013 Revert of Update with new default boringssl no-aes cipher suites. Re-enable tests. (patchset #3 id:40001 of https://codereview.webrtc.org/1550773002/ ) by sprang · 8 years ago
  4. 31c8d2e Update with new default boringssl no-aes cipher suites. Re-enable tests. by Torbjorn Granlund · 8 years ago
  5. 688e308 Re-land: "Use an explicit identifier in Config" by aluebs · 8 years ago
  6. 268493a Revert of Delete remnants of non-square pixel support from cricket::VideoFrame. (patchset #1 id:1 of https://codereview.webrtc.org/1586613002/ ) by nisse · 8 years ago
  7. 709513d Delete remnants of non-square pixel support from cricket::VideoFrame. by nisse · 8 years ago
  8. 2d110be Revert of Storing raw audio sink for default audio track. (patchset #7 id:120001 of https://codereview.chromium.org/1551813002/ ) by deadbeef · 8 years ago
  9. fca54f4 Revert of Use an explicit identifier in Config (patchset #4 id:60001 of https://codereview.webrtc.org/1538643004/ ) by tommi · 8 years ago
  10. 306efad Disable WebRtcVideoChannel2BaseTest.SendManyResizeOnce for TSan by kjellander · 8 years ago
  11. 25249d9 Use an explicit identifier in Config by aluebs · 8 years ago
  12. e591f93 Storing raw audio sink for default audio track. by deadbeef · 8 years ago
  13. 6955870 Convert channel counts to size_t. by Peter Kasting · 8 years ago
  14. 3e1cfa7 Delete unused method webrtc::VideoRendererInterface::SetSize. by nisse · 8 years ago
  15. 127782b Add default dummy implementation of cricket::VideoRenderer::SetSize, to easy later removal. by nisse · 8 years ago
  16. b2328d1 Remove additional channel constraints when Beamforming is enabled in AudioProcessing by aluebs · 8 years ago
  17. a7446d2 Change DTLS default from 1.0 to 1.2 for webrtc. by Guo-wei Shieh · 8 years ago
  18. 27ed3cc SCTP: Stopped accepting SSRCs higher than max. Seems to fix asan-related crash. by lally · 8 years ago
  19. f475d36 Properly handle different transports having different SSL roles. by Taylor Brandstetter · 8 years ago
  20. 25702cb Misc. small cleanups. by pkasting · 8 years ago
  21. 37ebcf0 Reland "Add APK targets to build libjingle tests for Android." by phoglund · 8 years ago
  22. fbeb97e Fix clang warning in peerconnection_jni.cc by perkj · 8 years ago
  23. 893505d Adding unit test to ensure TURN server priorities are unique. by Taylor Brandstetter · 8 years ago
  24. e5ba13b Adding a way for a Java RtpSender to set a track without taking ownership. by Taylor Brandstetter · 8 years ago
  25. 13f61df Move fake-handle frame creation into test target. by Peter Boström · 8 years ago
  26. 60ca31b Roll chromium_revision d66326c..4df108a (367167:367307) by kjellander · 8 years ago
  27. 0c7e9f5 Removing webrtc::PortAllocatorFactoryInterface. by Taylor Brandstetter · 8 years ago
  28. 3f7219b Fixing issue where description contains empty ICE ufrag/pwd. by deadbeef · 8 years ago
  29. e6bf587 Deleted VideoCapturer::screencast_max_pixels, together with by nisse · 8 years ago
  30. 2f042f2 Roll chromium_revision 1b6c421..db567a8 (365999:366304) by kjellander · 8 years ago
  31. a4df27b Revert of Reland "Added option to specify a maximum file size when recording an AEC dump." (patchset #2 id:20001 of https://codereview.webrtc.org/1541633002/ ) by ivoc · 8 years ago
  32. f4f5cb0 Reland "Added option to specify a maximum file size when recording an AEC dump.", commit ae2c5ad12afc8cc29fe9c59dea432b697b871a87. by ivoc · 8 years ago
  33. bd7d8f7 Adding a MediaStream parameter to createSender. by deadbeef · 8 years ago
  34. 36d4c54 Revert of Added option to specify a maximum file size when recording an AEC dump. (patchset #5 id:120001 of https://codereview.webrtc.org/1413483003/ ) by ivoc · 8 years ago
  35. b7d9a97 Expose codec implementation names in stats. by Peter Boström · 8 years ago
  36. ae2c5ad Added option to specify a maximum file size when recording an AEC dump. by ivoc · 8 years ago
  37. 88518a2 Use NV21 instead of YUV12 and clean up. by perkj · 8 years ago
  38. 48477c1 MediaCodecVideoEncoder, set timestamp on the encoder surface when drawing a texture. by perkj · 8 years ago
  39. 77fa59d Fix build break in google3 import caused by https://codereview.webrtc.org/1532543003 by guoweis · 9 years ago
  40. 4638331 DTLS-SRTP set up is bypassed when the channel has been writable. by guoweis · 9 years ago
  41. 0eb15ed Don't call the Pass methods of rtc::Buffer, rtc::scoped_ptr, and rtc::ScopedVector by kwiberg · 9 years ago
  42. a54a080 Add ufrag to the ICE candidate signaling. by honghaiz · 9 years ago
  43. 7cae30c Disable warnings failing when using Clang on Windows. by kjellander · 9 years ago
  44. 672aba3 Fix error prone code in VideoCapturerAndroid by perkj · 9 years ago
  45. 66085be Bugfix that fixes the error where the audio processing module is called by peah · 9 years ago
  46. eb45981 Restoring behavior where PeerConnection tracks changes to MediaStreams. by deadbeef · 9 years ago
  47. 44f0819 Fixing bug where "mid" wasn't preserved across re-offers. by deadbeef · 9 years ago
  48. 5125433 Android: Refactor renderers to allow apps to inject custom shaders by Magnus Jedvert · 9 years ago
  49. 32d989b Disable transport sequence numbers for audio. by Stefan Holmer · 9 years ago
  50. 6eca7e3 Add a 'remote' property to MediaSourceInterface. Also adding an implementation to the relevant sources we have (audio/video) and an extra check where we're casting a source into a local audio source :( by tommi · 9 years ago
  51. 9638143 Reland of Made EglBase an abstract class and cleaned up. (patchset #1 id:1 of https://codereview.webrtc.org/1522073002/ ) by perkj · 9 years ago
  52. 1588793 Fixing flaky LocalP2PTestSctpDataChannel test. by deadbeef · 9 years ago
  53. c9be007 Fixing and re-enabling some flaky PeerConnection tests. by deadbeef · 9 years ago
  54. bd29246 Reland of Free SCTP data channels asynchronously in PeerConnection. (patchset #1 id:1 of https://codereview.webrtc.org/1513143003/ ) by deadbeef · 9 years ago
  55. e22e1cb Revert of Made EglBase an abstract class and cleaned up. (patchset #4 id:60001 of https://codereview.webrtc.org/1526463002/ ) by perkj · 9 years ago
  56. 3207916 Made EglBase an abstract class and cleaned up. by perkj · 9 years ago
  57. bc14164 Revert of Add APK targets to build libjingle tests for Android. (patchset #10 id:180001 of https://codereview.webrtc.org/1511633002/ ) by stefan · 9 years ago
  58. a78c021 Add APK targets to build libjingle_peerconnection_unittests for Android. by perkj · 9 years ago
  59. 17821db Wire up bandwidth limitation info to GetStats and adapt_reason. by asapersson · 9 years ago
  60. 1d5c19d Address comments from code review 1505253004 by tommi · 9 years ago
  61. 4759bfb Roll chromium_revision 7de03ed..4bc4277 (364770:364953) by kjellander · 9 years ago
  62. cb95f54e Remove pointless move() to fix build on clang/win. by Tommi · 9 years ago
  63. f888bb5 Support for unmixed remote audio into tracks. by Tommi · 9 years ago
  64. 04e9146 Discard old-generation candidates when ICE restarts by Honghai Zhang · 9 years ago
  65. 822bdf9 Remove cricket::VideoEncoderConfig. by Peter Boström · 9 years ago
  66. 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
  67. cf846ad Adding stub files needed for https://codereview.webrtc.org/1507973003/ by Taylor Brandstetter · 9 years ago
  68. 7c73bdb Renaming JsepPeerConnectionP2PTestClient back to P2PTestConductor. by deadbeef · 9 years ago
  69. 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
  70. 796cfaf Add VideoCodec::PreferDecodeLate by perkj · 9 years ago
  71. c490e01 Implement NativeToI420Buffer in C++, calling java SurfaceTextureHelper, new method .textureToYUV, to by nisse · 9 years ago
  72. 1387149 Adding reduced size RTCP configuration down to the video stream level. by deadbeef · 9 years ago
  73. 434aca8 Add empty placeholder files for remote audio tracks. by tommi · 9 years ago
  74. 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
  75. bda7e0b Fixing issue with default stream upon setting 2nd remote description. by deadbeef · 9 years ago
  76. d02b0fa Add oldest rotation type option to RTCFileLogger by haysc · 9 years ago
  77. 1a9d615 Add tracing to public PeerConnection methods. by Peter Boström · 9 years ago
  78. 7b2f762 Don't call SetPreviewFormat if capturing to textures. by perkj · 9 years ago
  79. edd8fef Add new view that renders local video using AVCaptureLayerPreview. by haysc · 9 years ago
  80. 246b817 Refactor handling of AudioOptions. by solenberg · 9 years ago
  81. 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
  82. 9f45a45 Add tracing to upper-level WebRTC calls. by Peter Boström · 9 years ago
  83. 03ef053 Merge webrtc/video_engine/ into webrtc/video/ by Peter Boström · 9 years ago
  84. 3868692 Free SCTP data channels asynchronously in PeerConnection. by deadbeef · 9 years ago
  85. 46ad542 Revert of "Create rtc::AtomicInt POD struct." (patchset #3 id:40001 of https://codereview.webrtc.org/1498953002/ ) by pbos · 9 years ago
  86. 6f28cf0 Implement standalone event tracing in AppRTCDemo. by Peter Boström · 9 years ago
  87. 84f0970 Reland of "Create rtc::AtomicInt POD struct." by Peter Boström · 9 years ago
  88. cd4003f Use @webrtc.org addresses for OWNERS. by Peter Boström · 9 years ago
  89. cf890bc Roll gtest-parallel. by Peter Boström · 9 years ago
  90. 9d69c3f Return a copy of the supported RTP header extensions instead of a reference. by Stefan Holmer · 9 years ago
  91. b86d4e4 Prepare the AudioSendStream to be hooked up to send-side BWE. by Stefan Holmer · 9 years ago
  92. 03f80eb Refactor EglBase configuration. by nisse · 9 years ago
  93. 1218d7a Allow remote fingerprint update during a call by Guo-wei Shieh · 9 years ago
  94. 86aaa4b Revert "Allow remote fingerprint update during a call" by Guo-wei Shieh · 9 years ago
  95. 9c38c2d Allow remote fingerprint update during a call by Guo-wei Shieh · 9 years ago
  96. 381b421 Ping backup connection at a slower rate by Honghai Zhang · 9 years ago
  97. 9e1b992 Clear old decoders after recreating the receiver. by Peter Boström · 9 years ago
  98. b572768 - Remove calls to VoEDtmf from WVoE/MC. by Fredrik Solenberg · 9 years ago
  99. 1a5cf6e Remove the unused NullMediaEngine (and NullVoiceEngine+NullVideoEngine). by Fredrik Solenberg · 9 years ago
  100. 9cf0c3d Removes MAYBE_ from several test case names in JsepPeerConnectionP2PTestClient. by Ivo Creusen · 9 years ago