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gerrit-public.fairphone.software / platform / external / webrtc / 04f4d126f84dbbb915f52d91807cabdabf08d483
  1. 04f4d12 Implement timing frames. by ilnik · 7 years ago
  2. 3b921f0 Roll chromium_revision 4b74fa1307..df32089dae (480384:480415) by buildbot · 7 years ago
  3. 1f7476f Remove explicit draw call on MTKView. by Daniela · 7 years ago
  4. 91047e5 Remove redundant std::min from ProbeBitrateEstimator. by terelius · 7 years ago
  5. 76b20b7 Roll chromium_revision 2390071bb3..4b74fa1307 (480364:480384) by buildbot · 7 years ago
  6. bed7a6b Use information about blacklisted devices in video_quality_loopback_test by oprypin · 7 years ago
  7. 429d614 Roll chromium_revision add3c68a6c..2390071bb3 (480340:480364) by buildbot · 7 years ago
  8. edf2859 Roll chromium_revision ed82d45fc0..add3c68a6c (480324:480340) by buildbot · 7 years ago
  9. fb8cf3c Roll chromium_revision e438353b8b..ed82d45fc0 (480311:480324) by buildbot · 7 years ago
  10. 0393de4 Roll chromium_revision b032878ebd..e438353b8b (480186:480311) by kjellander · 7 years ago
  11. 1a610f1 Revert of Opus implementation of the AudioEncoderFactoryTemplate API (patchset #4 id:80001 of https://codereview.webrtc.org/2930243003/ ) by charujain · 7 years ago
  12. eb2d2d3 Revert of Opus implementation of the AudioDecoderFactoryTemplate API (patchset #1 id:1 of https://codereview.webrtc.org/2942733003/ ) by charujain · 7 years ago
  13. af62935 Support building WebRTC without audio and video for Android by zhihuang · 7 years ago
  14. d053fe4 Opus implementation of the AudioDecoderFactoryTemplate API by kwiberg · 7 years ago
  15. fe1aa82 Opus implementation of the AudioEncoderFactoryTemplate API by kwiberg · 7 years ago
  16. b8727ae G722 implementation of the AudioEncoderFactoryTemplate API by kwiberg · 7 years ago
  17. b1ed7f0 G722 implementation of the AudioDecoderFactoryTemplate API by kwiberg · 7 years ago
  18. 0eacd88 Templated AudioDecoderFactory by kwiberg · 7 years ago
  19. 19b3a55 Fixing incorrect use of erase/remove idiom. by deadbeef · 7 years ago
  20. dab1d2d Enable SNI in ssl adapter. by Emad Omara · 7 years ago
  21. 43b39de Roll chromium_revision 175bf817db..b032878ebd (480112:480186) by buildbot · 7 years ago
  22. 5369a98 Roll chromium_revision 1d3617187c..175bf817db (480056:480112) by buildbot · 7 years ago
  23. 6531583 Templated AudioEncoderFactory by kwiberg · 7 years ago
  24. 9fbbdc2 Create the VideoEncoderFactory and implement it. by Bjorn Mellem · 7 years ago
  25. c1db79b Roll chromium_revision 804cd4b03e..1d3617187c (480025:480056) by buildbot · 7 years ago
  26. 5cb1982 Tune loss-based BWE to be more compatible with the low frequency loss reports of audio streams. by stefan · 7 years ago
  27. 8fa21c4 Style fixes in rtcp_packet/ by eladalon · 7 years ago
  28. 6b826ef Add cropping to VIEEncoder to match simulcast streams resolution by ilnik · 7 years ago
  29. f79dbad Add has_value() and value() methods to rtc::Optional. by terelius · 7 years ago
  30. 0ef8fb9 Roll chromium_revision ff467ab402..804cd4b03e (480004:480025) by buildbot · 7 years ago
  31. af35f83 Reduces sensitivity in audio-glitch detector for iOS by henrika · 7 years ago
  32. bf5a2fc Use RaceChecker instead of ThreadChecker in a few places. by erikvarga · 7 years ago
  33. bd09ebc Remove unused #include "libyuv/compare.h" by eladalon · 7 years ago
  34. e150058 Move setting switches in AppRTCMobile to Settings screen by Anders Carlsson · 7 years ago
  35. 7c30390 Roll chromium_revision 2696d5a95c..ff467ab402 (479974:480004) by buildbot · 7 years ago
  36. bc061b4 Create AndroidVideoBuffer and allow renderers to consume it. by Sami Kalliomäki · 7 years ago
  37. af99b6d Delete SignalSrtpError. by nisse · 7 years ago
  38. 92b607b Roll chromium_revision 0292e51a15..2696d5a95c (479940:479974) by buildbot · 7 years ago
  39. 2b6e658 Roll chromium_revision 3a8e71c423..0292e51a15 (479885:479940) by buildbot · 7 years ago
  40. 7ca9e36 Roll chromium_revision 14085ad54d..3a8e71c423 (479854:479885) by buildbot · 7 years ago
  41. 3fc2350 Support H.264 high profile encoding on Exynos devices. by glaznev · 7 years ago
  42. 24d00dd Roll chromium_revision 3a1b0a576e..14085ad54d (479771:479854) by buildbot · 7 years ago
  43. 38ede13 Support building WebRTC without audio and video. by zhihuang · 7 years ago
  44. beae92c Roll chromium_revision 408bddbca7..3a1b0a576e (479728:479771) by buildbot · 7 years ago
  45. 65f3150 Roll chromium_revision 5cda18fb97..408bddbca7 (479679:479728) by buildbot · 7 years ago
  46. 112adf9 Validate references of frames inserted into FrameBuffer2. by philipel · 7 years ago
  47. eb02c03 Allow WebRtcMediaEngine to be created from any thread. by deadbeef · 7 years ago
  48. d03a578 Roll chromium_revision 47f5ca5642..5cda18fb97 (479659:479679) by buildbot · 7 years ago
  49. 67561a6 Use the same QP max for tests as in production by sprang · 7 years ago
  50. fda496a Set overuse detector max frame interval based on target frame rate. by sprang · 7 years ago
  51. 19e087f This CL finalizes the Conversational Speech tool. by alessiob · 7 years ago
  52. b2152b7 Roll chromium_revision 5423b19357..47f5ca5642 (479628:479659) by buildbot · 7 years ago
  53. 6af9399 ACM: Make AcmReceiver's ownership of NetEq more obvious by Henrik Lundin · 7 years ago
  54. f9784f2 Reland of Conversational speech tool, simualtor + unit tests (patchset #1 id:1 of https://codereview.webrtc.org/2925123003/ ) by alessiob · 7 years ago
  55. f4dd191 Change existing aec dump tests to use webrtc::AecDump. by aleloi · 7 years ago
  56. af66f2c Roll chromium_revision c4920d627d..5423b19357 (479593:479628) by buildbot · 7 years ago
  57. 08cfc84 Roll chromium_revision e7ba6263e5..c4920d627d (479563:479593) by buildbot · 7 years ago
  58. 24b60ef Roll chromium_revision 6dec5dff0b..e7ba6263e5 (479516:479563) by buildbot · 7 years ago
  59. f7f6572 Roll chromium_revision 4062088042..6dec5dff0b (479467:479516) by buildbot · 7 years ago
  60. 4d2e0a8 Roll chromium_revision 78764cfda4..4062088042 (479277:479467) by buildbot · 7 years ago
  61. a5e0df6 Move MinPositive to call.h as discussed here: https://codereview.chromium.org/2888303005/#msg19 by zstein · 7 years ago
  62. 23bc51b Reland of ll chromium_revision 05ba7bc226..78764cfda4 (479231:479277) (patchset #1 id:1 of https://codereview.webrtc.org/2939903002/ ) by kjellander · 7 years ago
  63. 62faaab Android: Add functionality for wrapping C++ I420 buffers to Java by Magnus Jedvert · 7 years ago
  64. cca0f6c Support H.264 high profile decoding on Exynos devices. by glaznev · 7 years ago
  65. 4f1f458 Also scan stderr for audio files to test, due to change in Android test_runner by oprypin · 7 years ago
  66. 386e496 Revert "Revert "Update webrtc/sdk/objc to new VideoFrameBuffer interface"" by Magnus Jedvert · 7 years ago
  67. 26ecfcc Remove timeStampMs from EncodedImage. by Sami Kalliomäki · 7 years ago
  68. 4eccdaa Fix a numerical issue in NetEq delay plotting by henrik.lundin · 7 years ago
  69. 7a721e8 Update webrtc/media and webrtc/modules to new VideoFrameBuffer interface by Magnus Jedvert · 7 years ago
  70. 3c938fc Add NetEq delay plotting to event_log_visualizer by henrik.lundin · 7 years ago
  71. 3c81a1a Add field trial for balanced degradation preference. by asapersson · 7 years ago
  72. c417d9e NetEq: Removing LastError and LastDecoderError by Henrik Lundin · 7 years ago
  73. 2b3aa14 Fix Chromium style checker warnings for MockAudioDecoder by kwiberg · 7 years ago
  74. 96444ae Implement operator<< for AudioCodecInfo and AudioCodecSpec by kwiberg · 7 years ago
  75. 6c4ba9f Plot acknowledged bitrate when compiled with rtc_enable_bwe_test_logging. by terelius · 7 years ago
  76. 58c5a7d Revert of Roll chromium_revision 05ba7bc226..78764cfda4 (479231:479277) (patchset #1 id:1 of https://codereview.webrtc.org/2936153002/ ) by mbonadei · 7 years ago
  77. f7e294d Implement kBalanced degradation preference. by asapersson · 7 years ago
  78. b749e5e Fix for broken test BweFeedbackTest. by tschumim · 7 years ago
  79. b7d6015 Roll chromium_revision 05ba7bc226..78764cfda4 (479231:479277) by buildbot · 7 years ago
  80. 7dce727 Roll chromium_revision 53ecf9341f..05ba7bc226 (479165:479231) by buildbot · 7 years ago
  81. 6eb03b8 Remove dependency on gunit headers in virtualsocketserver. by Bjorn Mellem · 7 years ago
  82. 1ee2125 Adding PortAllocator option to support cases where sockets can't be bound. by deadbeef · 7 years ago
  83. 1d560e1 Roll chromium_revision 4ddaa6f836..53ecf9341f (479034:479165) by buildbot · 7 years ago
  84. 179f997 Remove DCHECK from PeerConnectionFactory::worker_thread. by zstein · 7 years ago
  85. da4eba1 Tune vp9 quality scaler parameters by glaznev · 7 years ago
  86. 0c61a36 Roll chromium_revision 4f7c2dc196..4ddaa6f836 (478995:479034) by buildbot · 7 years ago
  87. 5c4eebb Implement org.webrtc.VideoEncoder using the android MediaCodec. by Bjorn Mellem · 7 years ago
  88. 7be7883 Adds detection of audio glitches for playout on iOS (reland) by henrika · 7 years ago
  89. 6e286cb Revert "Adds detection of audio glitches for playout on iOS. " by Henrik Andreasson · 7 years ago
  90. 33e4e65 Adds detection of audio glitches for playout on iOS. by henrika · 7 years ago
  91. dea075c Log an error in RtpDemuxer::FindSsrcAssociations() if kMaxProcessedSsrcs exceeded by eladalon · 7 years ago
  92. 7ed35f4 Replacing WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP with WEBRTC_ENABLE_PROTOBUF. by minyue-webrtc · 7 years ago
  93. 10e1f75 Roll chromium_revision 9061a92f5c..4f7c2dc196 (478958:478995) by buildbot · 7 years ago
  94. 2986033 Remove webrtcvideoengine2.h by eladalon · 7 years ago
  95. 659a010 Delete old include file webrtc/video_frame.h. by nisse · 7 years ago
  96. a65ad22 Delete unused method FilesystemInterface::GetFileTime. by nisse · 7 years ago
  97. 8c6afef Make sure UI methods get called on the main thread by adam.fedor · 7 years ago
  98. fdfeb83 Declaring rtc_base_approved dep on webrtc_common by mbonadei · 7 years ago
  99. 7339712 Removing backward compatible header by mbonadei · 7 years ago
  100. a735d4e Roll chromium_revision 0ca6ede735..9061a92f5c (478917:478958) by buildbot · 7 years ago