Google Git
Sign in
webrtc / src.git / 77f3e0da5a2ad50dab6c1ebad490dad8a88a779a / webrtc
  1. 77f3e0d Screen was flickering when the picker for desktop medias showed up in Windows platform. Keeping track of window size for each window so that BitBlt() instead of PrintWindow() will be called for windows with unchanged sizes. by gyzhou · 9 years ago
  2. b1eaa8d Only average positive quality stats. by Peter Boström · 9 years ago
  3. 80e1207 Move congestion controller to a separate module. by Stefan Holmer · 9 years ago
  4. ba3e25e Simple RTCP receiver fuzzer. by Peter Boström · 9 years ago
  5. 79d7a49 Replace scoped_ptr with unique_ptr in webrtc/common_audio/ by kwiberg · 9 years ago
  6. dc0e381 Add more camera resolutions to camera scaling slider. by Alex Glaznev · 9 years ago
  7. 18fcbcf Use VAD to get a better speech power estimation in the IntelligibilityEnhancer by Alejandro Luebs · 9 years ago
  8. 67b81f9 Tune QP thresholds for HW H.264 encoder. by Alex Glaznev · 9 years ago
  9. a094fd1 RTT intermediate calculation use ntp time instead of milliseconds. by Danil Chapovalov · 9 years ago
  10. 723ead8 Move simple RtpRtcp calls to VideoSendStream. by Peter Boström · 9 years ago
  11. eee7d9e iOS: Promote iOS simulator testing to main waterfall. by kjellander@webrtc.org · 9 years ago
  12. 7ddc9de Reduce the scope of rtc::Event::Wait() locking. by Peter Boström · 9 years ago
  13. d1f718b Changes in the wav_file implementation in order to by peah · 9 years ago
  14. 253d8fa Simplified the function for detecting whether capture data is modified. by peah · 9 years ago
  15. ada8fe5 iOS: Don't run modules_unittests on iOS simulator by kjellander@webrtc.org · 9 years ago
  16. a18f638 Include "sharedexclusivelock.cc" in Chromium GN build. by jbauch · 9 years ago
  17. b9dd7c5 Remove GetTransport() from TransportChannelImpl by mikescarlett · 9 years ago
  18. 91fe304 vp9: Adjust parameter for a test in videoprocessor_integrationtest.cc by Marco · 9 years ago
  19. a9d08929 Add initial bitrate and frame resolution parameters to quality scaler. by Alex Glaznev · 9 years ago
  20. 0013dcc Simplify SSRC usage inside ViEEncoder. by Peter Boström · 9 years ago
  21. 7254890 Nuke SetSenderBufferingMode. by Peter Boström · 9 years ago
  22. e2d83d6 Use CallStats for RTT in Call, rather than VideoSendStream::GetRtt() by sprang · 9 years ago
  23. 45c44f0 Simplify EncoderStateFeedback. by Peter Boström · 9 years ago
  24. 9674d7c Revert of Prevent data race in MessageQueue. (patchset #3 id:40001 of https://codereview.webrtc.org/1675923002/ ) by jbauch · 9 years ago
  25. fc968a2 Fix sequence-number replay race for padding. by Peter Boström · 9 years ago
  26. 88788ad Replace scoped_ptr with unique_ptr in webrtc/modules/audio_processing/ by kwiberg · 9 years ago
  27. df88460 Prevent data race in MessageQueue. by jbauch · 9 years ago
  28. 1e80ce4 webrtc::RtpPacket name freed for better RtpPacket by Danil Chapovalov · 9 years ago
  29. 728012e Changed the semantics of Buffer::Clear to not alter the capacity by ossu · 9 years ago
  30. 4458d09 Drop support for playing output through aplay in intelligibility_proc by Alejandro Luebs · 9 years ago
  31. b3fb71c Add RTCAudioSession proxy class. by Zeke Chin · 9 years ago
  32. 9ac4df1 iOS: Enable modules_unittests and common_audio_unittests by kjellander · 9 years ago
  33. 235aaa7 Fix Linux 32-bit compilation after sysroot switch. by Henrik Kjellander · 9 years ago
  34. 66a9928 Roll chromium_revision 1d144ca..fa5d546 (375480:376142) by kjellander@webrtc.org · 9 years ago
  35. 0e2e50c Always append the BYE packet type at the end by aleungbroadsoft · 9 years ago
  36. f45381e VideoCapturerAndroid: Report onFirstFrameAvailable() for textures as well by Magnus Jedvert · 9 years ago
  37. 5199c74 AndroidVideoCapturer getSupportedFormats(): Change interface from JSON string to List/vector by Magnus Jedvert · 9 years ago
  38. 347c0bb Android GLShader: Check return value of glCreateShader() by magjed · 9 years ago
  39. 3ee73a5 Make RemoteBitrateEstimator::GetStats() virtual. by Stefan Holmer · 9 years ago
  40. fd22e6c Change PeerConnectionFactory.setVideoHwAccelerationOptions to be able to replace Egl context. by Per · 9 years ago
  41. 74db777 Revert of Remove GetTransport() from TransportChannelImpl (patchset #3 id:40001 of https://codereview.webrtc.org/1691673002/ ) by guidou · 9 years ago
  42. 59c634b Re-add RemoteBitrateEstimator::GetStats. by Stefan Holmer · 9 years ago
  43. 3234819 Fix and simplify the power estimation in the IntelligibilityEnhancer by Alejandro Luebs · 9 years ago
  44. ee18220 Remove GetTransport() from TransportChannelImpl by mikescarlett · 9 years ago
  45. ee75c7a Compile rtc_base_objc for Mac. by tkchin · 9 years ago
  46. e3c6c82 When doing continual gathering, remove the local ports when a corresponding network is dropped. by honghaiz · 9 years ago
  47. a08bb0d Disabled the test EndToEndTest RestartingSendStreamPreservesRtpState due to the test being flaky. by peah · 9 years ago
  48. b7f89d6 Replace scoped_ptr with unique_ptr in webrtc/voice_engine/ by kwiberg · 9 years ago
  49. dabf07f Replace scoped_ptr with unique_ptr in webrtc/modules/audio_processing/vad/ by kwiberg · 9 years ago
  50. a293ef0 Apply VideoOptions per stream. by nisse · 9 years ago
  51. 789ba92 Simplify CongestionController. by Stefan Holmer · 9 years ago
  52. bad7804 Remove unused VideoSendStream TransportAdapter. by Peter Boström · 9 years ago
  53. 62eaacf Replace scoped_ptr with unique_ptr in webrtc/modules/audio_processing/test/ by kwiberg · 9 years ago
  54. 28c99bc iOS: Include legacy objc API in all.gyp + fix H264 libyuv dependency by kjellander · 9 years ago
  55. 4b4dc86 Remove conference_mode flag from AudioOptions and VideoOptions. by nisse · 9 years ago
  56. 69e59e6 [rtp_rtcp] rtc::scoped_ptr<rtcp::RawPacket> replaced with rtc::Buffer by danilchap · 9 years ago
  57. 67680c1 Ignore padding-only RTX packets in test. by Peter Boström · 9 years ago
  58. a332e2d Added boilerplate code for being able to test the upcoming AEC functionality. by peah · 9 years ago
  59. 0206000 iOS: Add resource files for tests and implement OutputPath by kjellander · 9 years ago
  60. 85d8bb0 Replace scoped_ptr with unique_ptr in webrtc/modules/audio_processing/transient/ by kwiberg · 9 years ago
  61. 9d3584c Implementing unified plan encoding of msid. by deadbeef · 9 years ago
  62. 25d6a0f Adding TSan suppressions temporarily to fix some flaky unit tests. by deadbeef · 9 years ago
  63. e1a0c94 Add network cost as part of the connection ranking. by honghaiz · 9 years ago
  64. 2c38c20 Fix out-of-buffer write in iLBC by henrik.lundin · 9 years ago
  65. 44c65e9 Enable adaptive threshold experiment by default. by Stefan Holmer · 9 years ago
  66. 9d0c432 Remove video-codec max bitrate from TMMBN. by Peter Boström · 9 years ago
  67. d20327c Increase the allowed number of probe packets in test to please msan. by Stefan Holmer · 9 years ago
  68. ee31f0a Fix out-of-buffer read in iLBC by henrik.lundin · 9 years ago
  69. 62a5ccd Update bitrate only when we have incoming packet. by Stefan Holmer · 9 years ago
  70. 58cf5f1 Changed order of events when synthesizing a call. by peah · 9 years ago
  71. 0453ef8 Prevent busy-looping PacedSender on small packets. by Peter Boström · 9 years ago
  72. 1794b26 Extract ViESyncModule outside ViEChannel. by Peter Boström · 9 years ago
  73. a3dc79e Move SSLIdentity Generate() implementations from .h to .cc file. by Torbjorn Granlund · 9 years ago
  74. 71e92dc Avoid overflow in WebRtcSpl_Sqrt by henrik.lundin · 9 years ago
  75. e8dc081 Implement certificate lifetime parameter as required by WebRTC RFC. by torbjorng · 9 years ago
  76. b1ae3a4 Stop decoders in VideoReceiveStream destructor. by Peter Boström · 9 years ago
  77. 461121c Replaced eglbase_jni with with holding a EglBase in PeerConnectionFactory. by perkj · 9 years ago
  78. 16c5a96 Replace scoped_ptr with unique_ptr in webrtc/modules/audio_coding/ by kwiberg · 9 years ago
  79. 3959397 Reland of Don't send FEC for H.264 with NACK enabled. (patchset #1 id:1 of https://codereview.webrtc.org/1692783005/ ) by Peter Boström · 9 years ago
  80. cde5d6b removed five redundant avsync tests to make webrtc_perf_test faster by Danil Chapovalov · 9 years ago
  81. e829f58 Rename libjingle_p2p_unittest -> rtc_pc_unittests by kjellander@webrtc.org · 9 years ago
  82. 3747838 Replace scoped_ptr with unique_ptr in webrtc/modules/audio_coding/test/ by kwiberg · 9 years ago
  83. 2d0c332 Replace scoped_ptr with unique_ptr in webrtc/modules/audio_coding/neteq/ by kwiberg · 9 years ago
  84. 04af839 Move refcount.h and scoped_ref_ptr.h to rtc_base_approved. BUG= by tommi · 9 years ago
  85. 91d9756 Replace scoped_ptr with unique_ptr in webrtc/modules/audio_coding/codecs/ by kwiberg · 9 years ago
  86. be61562 Moved the GainControlForNewAGC class to be a separate file. by peah · 9 years ago
  87. 88b0a22 Add VP9 to full stack tests. by asapersson · 9 years ago
  88. 29ffdc1 Revert of Don't send FEC for H.264 with NACK enabled. (patchset #5 id:80001 of https://codereview.webrtc.org/1687303002/ ) by deadbeef · 9 years ago
  89. 5e7834e Android: Make VideoCapturer an interface for all VideoCapturers to implement by Magnus Jedvert · 9 years ago
  90. e78765b Removes Nexus 5 from AEC and NS blacklists by henrika · 9 years ago
  91. 579e832 Fix race on VCM protection callback. by Peter Boström · 9 years ago
  92. b72dada Remove Reset from conditionally-compiled decoders. by Peter Boström · 9 years ago
  93. c90e9b6 Remove no-op VideoDecoder::Reset implementation. by Peter Boström · 9 years ago
  94. 25558ad Don't send FEC for H.264 with NACK enabled. by Peter Boström · 9 years ago
  95. efce73e Rename libjingle_media_unittest -> rtc_media_unittests by kjellander@webrtc.org · 9 years ago
  96. e9de296 Add OWNERS file in webrtc/pc by kjellander@webrtc.org · 9 years ago
  97. 040f68e Fix VideoCapturer::OnMessage override by Per · 9 years ago
  98. a509241 This reland https://codereview.webrtc.org/1655793003/ with the change that cricket::VideoCapturer::SignalVideoFrame is added back and used for frame forwarding. It is used in Chrome remoting. by Per · 9 years ago
  99. 59013bc Remove spammy GetRTPStatistics() log. by Peter Boström · 9 years ago
  100. 51542be Introduce struct MediaConfig, with construction-time settings. by nisse · 9 years ago