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webrtc / src / / e687f7816c9e6b7e374d62a4d44ce4ee14a49168
  1. e687f78 Moved the functionality in aec_core_internal.h into other files. by peah · 9 years ago
  2. ae28408 Jitter delay now depend on protection mode (FEC/NACK). by philipel · 9 years ago
  3. a105987 Convert Vp9 Rtp headers to frame references. by philipel · 9 years ago
  4. ba6371e Delete unused video capture support for cropping, non-square pixels, and ARGB screencast scaling. by nisse · 9 years ago
  5. 85d5108 Add test annotation to PeerConnectionClientTest.testLoopbackVp9 test. by kjellander@webrtc.org · 9 years ago
  6. d939d48 Remove Android x86 compilation trybot from CQ. by kjellander@webrtc.org · 9 years ago
  7. e69c37b Separated the functionalities in the OverdriveAndSuppress by peah · 9 years ago
  8. 23868b6 Broke apart the functionalities in the SubbandCoherence method in the AEC. by peah · 9 years ago
  9. 6c9b65a Made the method PartitionDelay independent of the AEC state. by peah · 9 years ago
  10. 779e97e Removed the MIPS optimized code for the comfort noise generation in by peah · 9 years ago
  11. 8d13c4f Changed the AEC SubbandCoherence function to not use the full AEC state by peah · 9 years ago
  12. d251196 Provide isAudioEnabled flag to control audio unit. by tkchin · 9 years ago
  13. 8f65cdf Only generate one CNAME per PeerConnection. by zhihuang · 9 years ago
  14. 630d9ba Fixed a crash in Objective-C clients when data channel becomes closed. by skvlad · 9 years ago
  15. 4f543d0 Remove Scanner usage from CPU Monitor. by Alex Glaznev · 9 years ago
  16. c7a6569 Revert of Disable failing modules_unittests for UBSan. (patchset #1 id:40001 of https://codereview.webrtc.org/1915813002/ ) by pbos · 9 years ago
  17. 82d7862 Change default timestamp to 64 bits in all webrtc directories. by Honghai Zhang · 9 years ago
  18. e76db89 Fix BoringSSL license path. by tkchin · 9 years ago
  19. dd32486 Bitrate prober now keep track of probing cluster id. by philipel · 9 years ago
  20. f2eae33 Corrected bug in checking the third number and added extra checks by dkirovbroadsoft · 9 years ago
  21. dc7d0d2 Move, almost, all receive side references to RTP to RtpStreamReceiver. by mflodman · 9 years ago
  22. b56069e Enable NACK for audio even if there are no send streams. by deadbeef · 9 years ago
  23. 31fec40 Set rtcp_send_transport for AudioReceiveStreams. This was forgotten in https://codereview.webrtc.org/1909333002/. by solenberg · 9 years ago
  24. 3a33465 Fix the flaky WebRtcSessionTest.TestRtxRemovedByCreateAnswer. by zhihuang · 9 years ago
  25. 44c8a37 Removed the file echo_cancellation_internal.h and moved by peah · 9 years ago
  26. cf5b37c Accept all the media profiles required by JSEP. by zhihuang · 9 years ago
  27. 39a3670 Rename OpenH264 frame-type conversion function. by pbos · 9 years ago
  28. 3f08dc6 Introduced the new APM data logging functionality into the AEC echo_cancellation.* API layer. by peah · 9 years ago
  29. e84cd2e Cache a ClientHello received before the DTLS handshake has started. by deadbeef · 9 years ago
  30. fac23f0 Tune QP threshold for HW codecs. by Alex Glaznev · 9 years ago
  31. 600246e Removed SSRC knowledge from ViEEncoder. by perkj · 9 years ago
  32. ef00ec1 Update CPU monitor to use moving averages. by Alex Glaznev · 9 years ago
  33. 28a4456 Revert "Revert of Remove SendPacer from ViEEncoder (patchset #13 id:240001 of https://codereview.webrtc.org/1917793002/ )" by Per · 9 years ago
  34. b49ac78 Revert of Use RC_TIMESTAMP_MODE for OpenH264. (patchset #1 id:1 of https://codereview.webrtc.org/1945763002/ ) by pbos · 9 years ago
  35. 1aa435c Reland of Android GlDrawer: Add frame size as argument to draw functions (patchset #1 id:1 of https://codereview.webrtc.org/1950953002/ ) by ivoc · 9 years ago
  36. 1726831 Revert of Android GlDrawer: Add frame size as argument to draw functions (patchset #2 id:20001 of https://codereview.webrtc.org/1948473002/ ) by ivoc · 9 years ago
  37. 274c1dc Added flag for FEC for video_loopback. by philipel · 9 years ago
  38. 73987c9 Run "git cl format --full" on a pair of files with ancient formatting by kwiberg · 9 years ago
  39. 053f917 Partial revert of Enable -Winconsistent-missing-override flag. (patchset #5 id:80001 of https://codereview.webrtc.org/1921653002/ ) by ivoc · 9 years ago
  40. 71af75d Android GlDrawer: Add frame size as argument to draw functions by magjed · 9 years ago
  41. c6c00b3 Revert of Remove the rtc_relative_path GYP variable and similar defines (patchset #1 id:1 of https://codereview.webrtc.org/1925733002/ ) by phoglund · 9 years ago
  42. 825eb58 Revert of Remove SendPacer from ViEEncoder (patchset #13 id:240001 of https://codereview.webrtc.org/1917793002/ ) by perkj · 9 years ago
  43. 857c5cc Remove SendPacer from ViEEncoder by perkj · 9 years ago
  44. cfc8e3b Removed all RTP dependencies from ViEChannel and renamed class. by mflodman · 9 years ago
  45. fe4b216 Roll chromium_revision 0b4adfd25e..58963e5878 (390907:391406) by buildbot · 9 years ago
  46. 3815655 Change aggregation window of aecDivergentFilterFraction to 1 second. by minyue · 9 years ago
  47. 7dd7ab5 Changed the name of the variable overdriveSm and removed the by peah · 9 years ago
  48. 55dd708 Support RtpEncodingParameters::active in voice engine. by Taylor Brandstetter · 9 years ago
  49. 1746179 Reducing neteq sync buffer size. by minyue · 9 years ago
  50. 4adbbcf Move ADM Create() method to public interface. by Peter Boström · 9 years ago
  51. 9bfa106 Change the threshold for external VNR. by jackychen · 9 years ago
  52. c4deee4 Use RC_TIMESTAMP_MODE for OpenH264. by Peter Boström · 9 years ago
  53. c8fe991 Removing SpatialAudio test code by henrik.lundin · 9 years ago
  54. 87f8c0d Adding in objc vars for WebRTC GN config. by Patrik Höglund · 9 years ago
  55. b1fb72b NetEq: Move counting of generated CNG samples from DecisionLogic by henrik.lundin · 9 years ago
  56. b46083e This CL introduces a new data logging functionality by peah · 9 years ago
  57. 696a802 Re-enable Vp9FlexModeRefCount by philipel · 9 years ago
  58. 35fdb2a Log WebRTC.Video.AVSyncOffsetInMs. by pbos · 9 years ago
  59. 5178ee8 NetEq: Use a BuiltinAudioDecoderFactory to create decoders by kwiberg · 9 years ago
  60. ddf1653 Android EGL: Synchronize calls to eglCreateContext by magjed · 9 years ago
  61. 30f118e This cl deletes the class webrtc::VideoRendererCallback. by nisse · 9 years ago
  62. fc88ffe Fix allocation size in CricketToJavaI420Frame, taking stride into account. by nisse · 9 years ago
  63. 35151f3 Add histogram stats for average send delay of sent packets for a sent video stream. The delay is measured from a packet is sent to the transport until leaving the socket. by asapersson · 9 years ago
  64. 5a24637 Do not stop a session unless the candidate of a writable connection belongs to the by Honghai Zhang · 9 years ago
  65. 5dd42fd Fixing a segfault that can occur when changing the track of an RtpSender. by deadbeef · 9 years ago
  66. acf1431 Removing unused resources from building files. by minyue · 9 years ago
  67. 376b192 Remove VideoCodingModule::VCMPacketizationCallback by perkj · 9 years ago
  68. 16ac328 Remove VCMRenderBufferSizeCallback. by Peter Boström · 9 years ago
  69. 1bffc1d Rename rtc::Time64 --> rtc::TimeMillis. by nisse · 9 years ago
  70. bc75d97 Remove PayloadRouter dependency from ViEEncoder. by perkj · 9 years ago
  71. 5bd3397 Adding 120 ms frame length support in NetEq. by minyue · 9 years ago
  72. e4246b6 Roll chromium_revision 8e44a16c27..0b4adfd25e (390884:390907) by buildbot · 9 years ago
  73. 7d4a6c3 Adds timeout for audio record thread in Java layer by henrika · 9 years ago
  74. 53ff70f Reland "Avoiding overflow in cross correlation in NetEq." by minyue · 9 years ago
  75. a017b8e Remove asapersson from webrtc/modules/utility/OWNERS. by asapersson · 9 years ago
  76. d1d96b2 VideoCapturerAndroid: Remove deprecated create function with egl context argument by Magnus Jedvert · 9 years ago
  77. 1ba8d39 Remove webrtc/stream.h and unutilized inheritance. by pbos · 9 years ago
  78. 60d7e7c Roll chromium_revision 927a9e3510..8e44a16c27 (390871:390884) by buildbot · 9 years ago
  79. 7b94c69 Remove DropOldEvents from list of excluded tests. As of last week, the test no longer exists. (https://codereview.webrtc.org/1687703002/) by terelius · 9 years ago
  80. c043052 Reland of move VCMQmRobustness. (patchset #1 id:1 of https://codereview.webrtc.org/1935753002/ ) by pbos · 9 years ago
  81. 0a1f14e Roll chromium_revision 9cd13b2cd1..927a9e3510 (390867:390871) by buildbot · 9 years ago
  82. bfefb03 Replace scoped_ptr with unique_ptr everywhere by kwiberg · 9 years ago
  83. e319059 Roll chromium_revision c6dd4461f5..9cd13b2cd1 (390857:390867) by buildbot · 9 years ago
  84. a1aad8f Roll chromium_revision 551bb334c0..c6dd4461f5 (390850:390857) by buildbot · 9 years ago
  85. d130ecc Roll chromium_revision b7ebd042d1..551bb334c0 (390693:390850) by buildbot · 9 years ago
  86. 322c4a0 Replace scoped_ptr with unique_ptr in webrtc/libjingle/ by kwiberg · 9 years ago
  87. a97611a Stop QuicDataChannel and QuicDataTransport unit tests from segfaulting by mikescarlett · 9 years ago
  88. e774867 Allow TransportController to create a QuicTransportChannel by mikescarlett · 9 years ago
  89. 9bc517f Add QuicDataChannel and QuicDataTransport classes by mikescarlett · 9 years ago
  90. 70035ca Fix QuicSession to unbuffer data when the QuicTransportChannel reconnects by mikescarlett · 9 years ago
  91. 602316c Revert of Remove VCMQmRobustness. (patchset #1 id:1 of https://codereview.webrtc.org/1917083003/ ) by pbos · 9 years ago
  92. 8d37d29 Update QuicTransportChannel to latest version of libquic (Chromium: f03d2c62) by mikescarlett · 9 years ago
  93. f3569c8 Added the API to create an RTCRtpSender to the Objective C wrapper. by skvlad · 9 years ago
  94. b3bedca Roll chromium_revision d908daba0f..b7ebd042d1 (390420:390693) by buildbot · 9 years ago
  95. 0bdebd4 Re-add a (dummy) webrtc/base/buffer.cc to hopefully unbreak the Chromium build by Karl Wiberg · 9 years ago
  96. 4f90677 Making NetEq bitexactness test independent on reference files. by minyue · 9 years ago
  97. 05e61ed Remove usage of VoENetwork from VoEWrapper and FakeWebRtcVoiceEngine. by solenberg · 9 years ago
  98. 79e2842 Add tracing to MessageQueue::Dispatch. by pbos · 9 years ago
  99. a4ac478 Define rtc::BufferT, like rtc::Buffer but for any trivial type by kwiberg · 9 years ago
  100. ef8b61e Enable -Winconsistent-missing-override flag. by nisse · 9 years ago