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webrtc / src.git / 9fea80f50daab46f20d4a6fc67b0144fbbbf56cd / talk
  1. 13f61df Move fake-handle frame creation into test target. by Peter Boström · 9 years ago
  2. 60ca31b Roll chromium_revision d66326c..4df108a (367167:367307) by kjellander · 9 years ago
  3. 0c7e9f5 Removing webrtc::PortAllocatorFactoryInterface. by Taylor Brandstetter · 9 years ago
  4. 3f7219b Fixing issue where description contains empty ICE ufrag/pwd. by deadbeef · 9 years ago
  5. e6bf587 Deleted VideoCapturer::screencast_max_pixels, together with by nisse · 9 years ago
  6. 2f042f2 Roll chromium_revision 1b6c421..db567a8 (365999:366304) by kjellander · 9 years ago
  7. 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 · 9 years ago
  8. f4f5cb0 Reland "Added option to specify a maximum file size when recording an AEC dump.", commit ae2c5ad12afc8cc29fe9c59dea432b697b871a87. by ivoc · 9 years ago
  9. bd7d8f7 Adding a MediaStream parameter to createSender. by deadbeef · 9 years ago
  10. 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 · 9 years ago
  11. b7d9a97 Expose codec implementation names in stats. by Peter Boström · 9 years ago
  12. ae2c5ad Added option to specify a maximum file size when recording an AEC dump. by ivoc · 9 years ago
  13. 88518a2 Use NV21 instead of YUV12 and clean up. by perkj · 9 years ago
  14. 48477c1 MediaCodecVideoEncoder, set timestamp on the encoder surface when drawing a texture. by perkj · 9 years ago
  15. 77fa59d Fix build break in google3 import caused by https://codereview.webrtc.org/1532543003 by guoweis · 9 years ago
  16. 4638331 DTLS-SRTP set up is bypassed when the channel has been writable. by guoweis · 9 years ago
  17. 0eb15ed Don't call the Pass methods of rtc::Buffer, rtc::scoped_ptr, and rtc::ScopedVector by kwiberg · 9 years ago
  18. a54a080 Add ufrag to the ICE candidate signaling. by honghaiz · 9 years ago
  19. 7cae30c Disable warnings failing when using Clang on Windows. by kjellander · 9 years ago
  20. 672aba3 Fix error prone code in VideoCapturerAndroid by perkj · 9 years ago
  21. 66085be Bugfix that fixes the error where the audio processing module is called by peah · 9 years ago
  22. eb45981 Restoring behavior where PeerConnection tracks changes to MediaStreams. by deadbeef · 9 years ago
  23. 44f0819 Fixing bug where "mid" wasn't preserved across re-offers. by deadbeef · 9 years ago
  24. 5125433 Android: Refactor renderers to allow apps to inject custom shaders by Magnus Jedvert · 9 years ago
  25. 32d989b Disable transport sequence numbers for audio. by Stefan Holmer · 9 years ago
  26. 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
  27. 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
  28. 1588793 Fixing flaky LocalP2PTestSctpDataChannel test. by deadbeef · 9 years ago
  29. c9be007 Fixing and re-enabling some flaky PeerConnection tests. by deadbeef · 9 years ago
  30. 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
  31. e22e1cb3 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
  32. 3207916 Made EglBase an abstract class and cleaned up. by perkj · 9 years ago
  33. 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
  34. a78c021 Add APK targets to build libjingle_peerconnection_unittests for Android. by perkj · 9 years ago
  35. 17821db Wire up bandwidth limitation info to GetStats and adapt_reason. by asapersson · 9 years ago
  36. 1d5c19d Address comments from code review 1505253004 by tommi · 9 years ago
  37. 4759bfb Roll chromium_revision 7de03ed..4bc4277 (364770:364953) by kjellander · 9 years ago
  38. cb95f54e Remove pointless move() to fix build on clang/win. by Tommi · 9 years ago
  39. f888bb5 Support for unmixed remote audio into tracks. by Tommi · 9 years ago
  40. 04e9146 Discard old-generation candidates when ICE restarts by Honghai Zhang · 9 years ago
  41. 822bdf9 Remove cricket::VideoEncoderConfig. by Peter Boström · 9 years ago
  42. 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
  43. cf846ad Adding stub files needed for https://codereview.webrtc.org/1507973003/ by Taylor Brandstetter · 9 years ago
  44. 7c73bdb Renaming JsepPeerConnectionP2PTestClient back to P2PTestConductor. by deadbeef · 9 years ago
  45. 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
  46. 796cfaf Add VideoCodec::PreferDecodeLate by perkj · 9 years ago
  47. c490e01 Implement NativeToI420Buffer in C++, calling java SurfaceTextureHelper, new method .textureToYUV, to by nisse · 9 years ago
  48. 1387149 Adding reduced size RTCP configuration down to the video stream level. by deadbeef · 9 years ago
  49. 434aca8 Add empty placeholder files for remote audio tracks. by tommi · 9 years ago
  50. 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
  51. bda7e0b Fixing issue with default stream upon setting 2nd remote description. by deadbeef · 9 years ago
  52. d02b0fa Add oldest rotation type option to RTCFileLogger by haysc · 9 years ago
  53. 1a9d615 Add tracing to public PeerConnection methods. by Peter Boström · 9 years ago
  54. 7b2f762 Don't call SetPreviewFormat if capturing to textures. by perkj · 9 years ago
  55. edd8fef Add new view that renders local video using AVCaptureLayerPreview. by haysc · 9 years ago
  56. 246b817 Refactor handling of AudioOptions. by solenberg · 9 years ago
  57. 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
  58. 9f45a45 Add tracing to upper-level WebRTC calls. by Peter Boström · 9 years ago
  59. 03ef053 Merge webrtc/video_engine/ into webrtc/video/ by Peter Boström · 9 years ago
  60. 3868692 Free SCTP data channels asynchronously in PeerConnection. by deadbeef · 9 years ago
  61. 46ad542 Revert of "Create rtc::AtomicInt POD struct." (patchset #3 id:40001 of https://codereview.webrtc.org/1498953002/ ) by pbos · 9 years ago
  62. 6f28cf0 Implement standalone event tracing in AppRTCDemo. by Peter Boström · 9 years ago
  63. 84f0970 Reland of "Create rtc::AtomicInt POD struct." by Peter Boström · 9 years ago
  64. cd4003f Use @webrtc.org addresses for OWNERS. by Peter Boström · 9 years ago
  65. cf890bc Roll gtest-parallel. by Peter Boström · 9 years ago
  66. 9d69c3f Return a copy of the supported RTP header extensions instead of a reference. by Stefan Holmer · 9 years ago
  67. b86d4e4 Prepare the AudioSendStream to be hooked up to send-side BWE. by Stefan Holmer · 9 years ago
  68. 03f80eb Refactor EglBase configuration. by nisse · 9 years ago
  69. 1218d7a Allow remote fingerprint update during a call by Guo-wei Shieh · 9 years ago
  70. 86aaa4b Revert "Allow remote fingerprint update during a call" by Guo-wei Shieh · 9 years ago
  71. 9c38c2d Allow remote fingerprint update during a call by Guo-wei Shieh · 9 years ago
  72. 381b421 Ping backup connection at a slower rate by Honghai Zhang · 9 years ago
  73. 9e1b992 Clear old decoders after recreating the receiver. by Peter Boström · 9 years ago
  74. b572768 - Remove calls to VoEDtmf from WVoE/MC. by Fredrik Solenberg · 9 years ago
  75. 1a5cf6e Remove the unused NullMediaEngine (and NullVoiceEngine+NullVideoEngine). by Fredrik Solenberg · 9 years ago
  76. 9cf0c3d Removes MAYBE_ from several test case names in JsepPeerConnectionP2PTestClient. by Ivo Creusen · 9 years ago
  77. 7635684 Fix Mac ObjC PeerConnection API compilation. by tkchin · 9 years ago
  78. 9462052 In some rare Android systems ConnectivityManager may be null. by honghaiz · 9 years ago
  79. 3c28d0d Disable PeerConnectionEndToEndTest.Call on Mac. by kjellander@webrtc.org · 9 years ago
  80. 1d63dd0 - Remove cricket::VoiceChannel::PressDtmf(); AFAICT unused. by solenberg · 9 years ago
  81. ee524f7 Adding Java binding for CreateSender. by deadbeef · 9 years ago
  82. 7e4e01a Add header extension filtering for WebRtcVoiceEngine/MediaChannel. by solenberg · 9 years ago
  83. 2515af2 Removing some unnecessary string manipulation code from VoEBase::GetVersion(). by solenberg · 9 years ago
  84. d20d247 Fix VideoCaptureAndroid, drop frame when switching camera using textures. by perkj · 9 years ago
  85. 226a602 Fix problem when drawing to the Android Media encoder surface. by perkj · 9 years ago
  86. 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
  87. 41b0798 Adding CreatePeerConnection method that uses new PC Initialize method. by deadbeef · 9 years ago
  88. 0de97f1 WebRtcVideoCapturer: SetCaptureState(CS_STOPPED) on Stop and ensure state changes in unittest. by hbos · 9 years ago
  89. cb9792e Fix VideoCapturerAndroidTest.testStartWhileCameraIsAlreadyOpen on Android M. by perkj · 9 years ago
  90. 14f4144 Add helper KeepRefUntilDone. by perkj · 9 years ago
  91. ee69ed5 Add separate event for camera freeze. by glaznev · 9 years ago
  92. 70c0e29 Disable PeerConnectionEndToEndTest.Call for TSan. by kjellander@webrtc.org · 9 years ago
  93. ae54b83 Android SurfaceViewRenderer: Add resetStatistics() method by magjed · 9 years ago
  94. 2fe1cb0 Don't overwrite audio stats when they're not available. by andrew · 9 years ago
  95. 26c8c91 Using Rent-A-Codec for static Codec access in WVoE/MC. by solenberg · 9 years ago
  96. 727dbc2 VideoCapturerAndroid - allow lower frame rate in bad lightning by Per · 9 years ago
  97. 598242a Support texture scaling in Androids MediaEncoder. by Per · 9 years ago
  98. a3c20bb Add support for scaling textures in AndroidVideoCapturer. by Per · 9 years ago
  99. fac0655 Reland of Adding the ability to create an RtpSender without a track. by deadbeef · 9 years ago
  100. 444682a Remove frame time scheduing in IncomingVideoStream by qiangchen · 9 years ago