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gerrit-public.fairphone.software / platform / external / webrtc / 0740a2040a167b62e70f8f24f695263bde4a971a
  1. 0740a20 Track SSRCs configured for each media type in event log visualization tool. by terelius · 8 years ago
  2. 4496809 Changed mixing api and moved resampler. by aleloi · 8 years ago
  3. 0727f15 Create gn target for peerconnection examples by kthelgason · 8 years ago
  4. 03bc3cf GN target for test_packet_masks_metrics. by brandtr · 8 years ago
  5. d046d98 Cleaning up mb_config.pyl style. by ehmaldonado · 8 years ago
  6. f4433ef Add argument support for stand-alone WebRTC tracing. by sakal · 8 years ago
  7. d4664d4 Roll chromium_revision 039ff70eec..c020edf5c5 (410288:410323) by buildbot · 8 years ago
  8. 04dbb34 iOS: Add support for rendering native CVPixelBuffers directly by tkchin · 8 years ago
  9. c3ec1fd Adding GN targets for audioproc, audioproc_f and unpack_aecdump by peah · 8 years ago
  10. 2ddfdba GN: Add targets for AppRTCDemo on Mac+iOS by tkchin · 8 years ago
  11. 968c9cc Roll chromium_revision c792d1ca22..039ff70eec (410283:410288) by buildbot · 8 years ago
  12. 7064d64 Roll chromium_revision 7ac4976476..c792d1ca22 (410281:410283) by buildbot · 8 years ago
  13. 08bbcc4 Roll chromium_revision 8577969f1d..7ac4976476 (410275:410281) by buildbot · 8 years ago
  14. 4d14d24 Roll chromium_revision 8d0c21dddb..8577969f1d (410266:410275) by buildbot · 8 years ago
  15. 14ac655 Roll chromium_revision 066f6569a1..8d0c21dddb (410263:410266) by buildbot · 8 years ago
  16. ef38529 Roll chromium_revision d90908aa34..066f6569a1 (410251:410263) by buildbot · 8 years ago
  17. 92f4d6e Roll chromium_revision b7b51c4bcf..d90908aa34 (410104:410251) by buildbot · 8 years ago
  18. f31e87f Roll chromium_revision 9ace14b3b2..b7b51c4bcf (410022:410104) by buildbot · 8 years ago
  19. 9763d56 Modified PeerConnection and WebRtcSession for end-to-end QuicDataChannel usage. by zhihuang · 8 years ago
  20. 3859c89 Add decoder-specific settings with proper lifetime. by johan · 8 years ago
  21. 4cd2790 Move RTP for synchroninzation and rename classes, files and variables. by mflodman · 8 years ago
  22. b7b9dca Reland Enable socketservers unittests on windows. by danilchap · 8 years ago
  23. ac86c68 Roll chromium_revision 9d5d9721db..9ace14b3b2 (409966:410022) by buildbot · 8 years ago
  24. 262e486 TMMBRHelp::FindBoundingSet function cleaned by danilchap · 8 years ago
  25. 1f8ed10 Add win trybots to the default trybot set. by ehmaldonado · 8 years ago
  26. 9028216 Roll chromium_revision 8b35b05724..9d5d9721db (409830:409966) by buildbot · 8 years ago
  27. 33f00e1 Fix case-sensitive build system errors. by tkchin · 8 years ago
  28. a4e4e7a Revert of Temporarily disable android_arm64_rel trybot. (patchset #1 id:1 of https://codereview.webrtc.org/2212913002/ ) by kjellander · 8 years ago
  29. 907abe4 Revert of Modified PeerConnection and WebRtcSession for end-to-end QuicDataChannel usage. (patchset #8 id:280001 of https://codereview.webrtc.org/2166873002/ ) by deadbeef · 8 years ago
  30. e7db69b Roll chromium_revision f65883a9c0..8b35b05724 (409751:409830) by buildbot · 8 years ago
  31. 34b54c3 Modified PeerConnection and WebRtcSession for end-to-end QuicDataChannel usage. by zhihuang · 8 years ago
  32. f736d23 Add fraction loss in visualization tool. by terelius · 8 years ago
  33. ddba75c Remove rtt limit on retransmission rate limiter by sprang · 8 years ago
  34. 9f1771b Revert of Enable socketservers unittests on windows (patchset #7 id:260001 of https://codereview.webrtc.org/2199193004/ ) by ehmaldonado · 8 years ago
  35. 06bfe1f Reland of GN: Create target for video_capture_tests on Linux. (patchset #1 id:1 of https://codereview.webrtc.org/2217463002/ ) by sakal · 8 years ago
  36. 41bce13 Revert of GN: Create target for video_capture_tests on Linux. (patchset #2 id:20001 of https://codereview.webrtc.org/2207463004/ ) by guidou · 8 years ago
  37. db75438 Tune BWE to be more sensitive on low capacity networks. by stefan · 8 years ago
  38. 171c7fe GN: Create target for video_capture_tests on Linux. by sakal · 8 years ago
  39. 90860af Roll chromium_revision 33f87688e4..f65883a9c0 (409700:409751) by buildbot · 8 years ago
  40. 84ef615 Removed calls to VoE::SetPlayout() from WebRTCVoiceEngine. by aleloi · 8 years ago
  41. cb56065 Add support for GCM cipher suites from RFC 7714. by jbauch · 8 years ago
  42. bfc7a30c Temporarily disable android_arm64_rel trybot. by ehmaldonado · 8 years ago
  43. 604fc11 Creating GN version of xmllite_xmpp_unittests by charujain · 8 years ago
  44. b419d6e Roll chromium_revision 48e079d573..33f87688e4 (407510:409700) by kjellander · 8 years ago
  45. 8cd8f81 Prepare for ICE renomination. by Honghai Zhang · 8 years ago
  46. 6ce738d Disable encoder scaling on iPhone4S. by tkchin · 8 years ago
  47. 9dfa249 Use 352x288 for iPhone 4S. by tkchin · 8 years ago
  48. f98dc10 GN: Add target for modules_tests. by ehmaldonado · 8 years ago
  49. df47312 Roll gtest-parallel. by Peter Boström · 8 years ago
  50. 31e4e80 RtpPacketHistory rewritten to use RtpPacket class. by Danil Chapovalov · 8 years ago
  51. 6b42cef Add linux_arm to tryserver.webrtc. by ehmaldonado · 8 years ago
  52. 9976840 Remove a few meaningless TODOs by mflodman · 8 years ago
  53. e51e119 Enable socketservers unittests on windows. by Danil Chapovalov · 8 years ago
  54. 837a170 Remove faulty DCHECK in rtcp::TransportFeedback::ParseFrom by Danil Chapovalov · 8 years ago
  55. 4f6cd6a Wire up VCMTiming in FrameBuffer2. by philipel · 8 years ago
  56. 17dfa74 Gyp->GN migration of RTPjitter. by aleloi · 8 years ago
  57. 64636dd Add UMA for tracking which BWE versions are in use. by stefan · 8 years ago
  58. 65ae2d8 Ignore iOS SDK during presubmit. by hjon · 8 years ago
  59. 525df3f Add EncodedImageCallback::OnEncodedImage(). by Sergey Ulanov · 8 years ago
  60. 51db4dd Revert of Add EncodedImageCallback::OnEncodedImage(). (patchset #14 id:300001 of https://codereview.chromium.org/2089773002/ ) by sergeyu · 8 years ago
  61. 4c7f4cd Add EncodedImageCallback::OnEncodedImage(). by Sergey Ulanov · 8 years ago
  62. ac4dc2c Revert of Add EncodedImageCallback::OnEncodedImage(). (patchset #13 id:280001 of https://codereview.webrtc.org/2089773002/ ) by sergeyu · 8 years ago
  63. ad34dbe Add EncodedImageCallback::OnEncodedImage(). by Sergey Ulanov · 8 years ago
  64. 6e11efa Bitrate prober and paced sender improvements by Irfan Sheriff · 8 years ago
  65. 469df93 Don't build VP9 for iOS framework. by tkchin · 8 years ago
  66. c3de033 Add plot of network delay change computed based on transport feedback. by stefan · 8 years ago
  67. 3ab32dc gn: Do not include "webrtc" in Chromium/iOS builds. by Nico Weber · 8 years ago
  68. d3235f0 Re-order and remove unused Java imports. by Sami Kalliomaki · 8 years ago
  69. 5a74650 Migrated GN target RTPtimeshift. by aleloi · 8 years ago
  70. b0dc30d Improvements to bandwidth plot. See Monorail issue for comparison. by aleloi · 8 years ago
  71. 5556dcb Migrated GN target rtpcat. by aleloi · 8 years ago
  72. 8626eb5 Fix testReturnBufferLate incorrectly calling end to end test. by sakal · 8 years ago
  73. b4ad456 Remove negative include rules from webrtc_nonparallel_tests target by danilchap · 8 years ago
  74. 606daa8 Adding Danil as an owner for the RTP module. by mflodman · 8 years ago
  75. 76cbe19 Migrated RTPchange Gyp target to GN. by aleloi · 8 years ago
  76. 9eb3d5d Add sanity check for arrival timestamps. by sprang · 8 years ago
  77. 8266773 Migrated target RTPencode to GN by aleloi · 8 years ago
  78. 734262c Fixing invalid operator< implementation. by Taylor Brandstetter · 8 years ago
  79. 8853289 Un-flaking TestSrtpError by using a fake clock. by Taylor Brandstetter · 8 years ago
  80. 19ee1e6eb Add cricket::VideoFrame::transport_frame_id() and set it to RTP timestamp. by Sergey Ulanov · 8 years ago
  81. 7fbe2ee Make EGL10 class public. by Alex Glaznev · 8 years ago
  82. b578d6c congestion controller: Fix network route change handling by Irfan Sheriff · 8 years ago
  83. dc35dcd Convenience functions to set axis properties in visualization tool. by terelius · 8 years ago
  84. 8b70faf Send audio packets immediately instead of pacing them. by terelius · 8 years ago
  85. c980062 Use a new method (android_setsocknetwork) to bind socket to network in Android M and later. by Honghai Zhang · 8 years ago
  86. f734c13 Cleanup BWE SendTimeHistory class by danilchap · 8 years ago
  87. 233c4ba New ProbingCalculator class. by philipel · 8 years ago
  88. ddf3e4a GN conversion of audio_decoder_unittests by charujain · 8 years ago
  89. 32cd2c4 Fix issues with RestartingSendStreamPreservesRtpStatesWithRtx by danilchap · 8 years ago
  90. 95e7560 [rtcp] Nack::Parse updated not to use RTCPUtility by danilchap · 8 years ago
  91. 88c1d2b Don't use LOG_END timestamp to compute RtcEventLog duration. by terelius · 8 years ago
  92. 1a518fc Request ownership for visualization tool by terelius · 8 years ago
  93. 8a0b128 Fix CopyOnWriteBuffer::cdata to return const pointer by danilchap · 8 years ago
  94. d8a9c53 Implement CopyOnWriteBuffer move using scoped_refptr move by danilchap · 8 years ago
  95. 76a44d5 Log current time if we stop logging before stop_time_. by terelius · 8 years ago
  96. 7a11ae3 Print entire error object for capture session error handler by haysc · 8 years ago
  97. cd25539 Increase audio buffer duration for iPhone 4s. by tkchin · 8 years ago
  98. 6a850c3 Register header extensions for rtx streams in the event log analyzer. by stefan · 8 years ago
  99. 70cfc14 Adds move assignment operator to scoped_refptr by Danil Chapovalov · 8 years ago
  100. 3d7346f Adds delta-time logging for audio playout by henrika · 8 years ago