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webrtc / src / f472c5b6722dfb221f929fc4d3a2b4ca54647701
  1. f472c5b Implement the NackModule as part of the new jitter buffer. by philipel · 9 years ago
  2. b624d8c Removed the inheritance from ProcessingComponent for EchoCancellerImpl. by peah · 9 years ago
  3. 6ec641b Fixing some issues with payload type mappings. by Taylor Brandstetter · 9 years ago
  4. e26e787 Roll chromium_revision ee31124..508edd3 (378158:379249) by kjellander · 9 years ago
  5. 6a4a03c Add an option to soft reset HW decoder. by Alex Glaznev · 9 years ago
  6. 92586f0 Revert of Removed the inheritance from ProcessingComponent for EchoCancellerImpl. (patchset #4 id:60001 of https://codereview.webrtc.org/1761813002/ ) by solenberg · 9 years ago
  7. 3af0a00 Removed the inheritance from ProcessingComponent for EchoCancellerImpl. by peah · 9 years ago
  8. 20028c4 Removing the use of the soon-to-be-removed echo_cancellation_impl by peah · 9 years ago
  9. 6d8e011 Change NetEq::GetAudio to use AudioFrame by henrik.lundin · 9 years ago
  10. 6459f84 Create QuicTransportChannel by mikescarlett · 9 years ago
  11. a2f7798 Tweaks for new Objective-C API. by hjon · 9 years ago
  12. 78417cf Fix VideoTrack VideoSinkWants for renderers. by perkj · 9 years ago
  13. a2a49d9 This CL provides interfaces that do not use constraints for by hta · 9 years ago
  14. 2bb7080 CQ: Remove libfuzzer trybot from default trybot set. by kjellander@webrtc.org · 9 years ago
  15. 4f43fcf Renamed new EncodeInternal to EncodeImpl to ensure proper backwards compatibility. by ossu · 9 years ago
  16. 88950cf Moved the file aec_resampler.c to be built using C++. by peah · 9 years ago
  17. 0023fdf Remove the ID from AcmReceiver by henrik.lundin · 9 years ago
  18. 4f735d1 Enable iOS AppRTCDemo send side BWE. by tkchin · 9 years ago
  19. 0197363 A bitexactness test for the highpass filter in the audio processing module. by peah · 9 years ago
  20. fc4ff2d Fixed Aec handle index in EchoCancellationImpl by peah · 9 years ago
  21. e2af9ef Keep on sending stun binding requests on zero-cost networks. by honghaiz · 9 years ago
  22. ab12c47 Modifies SDK and iOS detection for helper method that needs iOS 9+ by henrika · 9 years ago
  23. a4c7688 Move encoder thread to VideoSendStream. by Peter Boström · 9 years ago
  24. 313afba Lazily allocate input buffer for AsyncTCPSocket. by jbauch · 9 years ago
  25. 79a5cf9 Fix last commit error successfully. by Peter Boström · 9 years ago
  26. 770012c Fix potentially-uninitialized compilation error. by Peter Boström · 9 years ago
  27. a03785e Move all calls after SetEncoder into SetEncoder. by Peter Boström · 9 years ago
  28. 81e8e37 Android SurfaceTextureHelper: Add stopListening() function by magjed · 9 years ago
  29. f2880a0 Change webrtc::VideoSourceInterface to inherit rtc::VideoSourceInterface. by perkj · 9 years ago
  30. c8da45f Set prefer_fixed_point=1 for the MIPS architecture. by peah · 9 years ago
  31. 0f13ec1 Removed VideoSource dependency to ChannelManager. by Per · 9 years ago
  32. a2abdf2 Remove usage of fmaf in IntelligibilityEnhancer by aluebs · 9 years ago
  33. 36f0137 Implement Turn/Turn first logic for connection selection. by guoweis · 9 years ago
  34. 4aee2a9 Add android specific audio mute function. by Alex Glaznev · 9 years ago
  35. 0e73934 Remove webrtc/test/webrtc_test_common.gyp by kjellander · 9 years ago
  36. 905f8e7 Make ReconfigureVideoEncoder void. by Peter Boström · 9 years ago
  37. 25359e0 DtlsIdentityStoreInterface::RequestIdentity gets optional expires param. by hbos · 9 years ago
  38. 0a9fc05 Move RTP module send status outside of ViEChannel. by Peter Boström · 9 years ago
  39. 7b19b08 Reland "Calculating ERLE in AEC more properly." by minyue · 9 years ago
  40. 4fa7eca Remove add/removal of ViEReceiver RTP modules. by Peter Boström · 9 years ago
  41. f475277 Rename constants files in webrtc/{media,p2p} by kjellander · 9 years ago
  42. 91e1c15 Make sure rotation is not applied by the capturer if the CVO exenstion is set before the send stream is created. by perkj · 9 years ago
  43. 7ecc163 Report all packets to bitrate probing. by Peter Boström · 9 years ago
  44. b65f3e3 [cleanup] Remove unused fields/functions from rtcp module. by Danil Chapovalov · 9 years ago
  45. c891eb4 Replace scoped_ptr with unique_ptr in webrtc/common_video/ by kwiberg · 9 years ago
  46. 60653ba New flag is_screencast in VideoOptions. by Niels Möller · 9 years ago
  47. e065fcf Replace scoped_ptr with unique_ptr in webrtc/modules/video_*/ by kwiberg · 9 years ago
  48. 4eb1ddd Fixing a possible crash in CopyCandidatesFromSessionDescription. by Taylor Brandstetter · 9 years ago
  49. 03d6d57 Late initialize MediaController, for less resource i.e. ProcessThread, usage by PeerConnection. by solenberg · 9 years ago
  50. 27f982b Replace scoped_ptr with unique_ptr in webrtc/video/ by kwiberg · 9 years ago
  51. d802b5b Fix some signed overflow errors causing undefined behavior (in theory). by terelius · 9 years ago
  52. 5711c8d Change transport sequence number extension strings to specify what revision is implemented. by Stefan Holmer · 9 years ago
  53. b0fdfea Add stats (histograms) for vp8 screenshare layers by sprang · 9 years ago
  54. 92931b1 Replace scoped_ptr with unique_ptr in webrtc/modules/remote_bitrate_estimator/ by kwiberg · 9 years ago
  55. 0db023a Move suspend_below_min_bitrate from VideoOptions to MediaConfig. by nisse · 9 years ago
  56. e496bad autoroller: Remove sending tryjobs (done via CQ). by Henrik Kjellander · 9 years ago
  57. e3d9922 rtc::Buffer: Use RTC_DCHECK instead of assert by kwiberg · 9 years ago
  58. ffdd41e jni_helpers: Optimize IsNull() by Magnus Jedvert · 9 years ago
  59. dc29780 Re-enable DCHECKs for increasing timestamps in paced_sender by sprang · 9 years ago
  60. 10a029e Changed AudioEncoder::Encode to take an rtc::Buffer* instead of uint8_t* and a maximum size. by ossu · 9 years ago
  61. 22c2b48 Move RTP stats histograms from VieChannel to SendStatisticsProxy. by Erik Språng · 9 years ago
  62. 681e20e .gitignore: remove no longer needed entries. by Henrik Kjellander · 9 years ago
  63. ac287ee VideoCaptureInput enforce VideoFrame::render_time to be generated by webrtc clock. by danilchap · 9 years ago
  64. b9338ac Added an operator[] to Buffer, to make reading data easier. by ossu · 9 years ago
  65. 012f8c0 Remove unused encoder_config_ variable. by Peter Boström · 9 years ago
  66. c9bbbe4 Revert "Calculating ERLE in AEC more properly." by minyuel · 9 years ago
  67. 7d9112c Make it possible to exclude device management code from rtc_media target. by kjellander · 9 years ago
  68. dda8a83 Trace tracing Start/Stop events. by Peter Boström · 9 years ago
  69. 3f55dea Replace scoped_ptr with unique_ptr in webrtc/modules/video_coding/ by kwiberg · 9 years ago
  70. a4f31bd TMMBRSet become vector<rtcp::TmmbItem> by danilchap · 9 years ago
  71. dffb894 Enable CQ by Henrik Kjellander · 9 years ago
  72. 739fcb9 Cleanup of webrtc::VideoFrame. by Niels Möller · 9 years ago
  73. 944744b Calculating ERLE in AEC more properly. by minyuel · 9 years ago
  74. fb45d17 Reland Remove unused cricket::VideoCapturer methods. Originally reviewed and landed as patchset #2 id:30001 of https://codereview.webrtc.org/1733673002/) by Per · 9 years ago
  75. f3ed9d7 Remove thread checker from CongestionController. by Stefan Holmer · 9 years ago
  76. 7352804 Disable CQ since being flooded with jobs by kjellander@webrtc.org · 9 years ago
  77. f15a7e9 Removed adresp from video Added danilchap to rtcp by Danil Chapovalov · 9 years ago
  78. 7e937e9 Remove workaround for Opus DTX noise pumping issue. by minyuel · 9 years ago
  79. 2d5f091 Move direct use of VideoCapturer::VideoAdapter to VideoSinkWants. by perkj · 9 years ago
  80. 50772f1 GN: Update audio_sink.h location by kjellander@webrtc.org · 9 years ago
  81. 804e082 CQ: Add android_compile_mips_dbg to default trybots by kjellander@webrtc.org · 9 years ago
  82. 250fc65 Lazily allocate output buffer for AsyncTCPSocket. by jbauch · 9 years ago
  83. 2b31a90 Roll chromium_revision a5a1e78..ee31124 (378154:378158) by kjellander · 9 years ago
  84. 3379edc Roll chromium_revision ba93bd2..a5a1e78 (378149:378154) by kjellander · 9 years ago
  85. 73eb679 Roll chromium_revision fd4c78b..ba93bd2 (378132:378149) by kjellander · 9 years ago
  86. 215f228 Roll chromium_revision b815f03..fd4c78b (378096:378132) by kjellander · 9 years ago
  87. 7b9601e Roll chromium_revision 968c1e7..b815f03 (377935:378096) by kjellander · 9 years ago
  88. 0a00759 Fix the stereo support in IntelligibilityEnhancer by aluebs · 9 years ago
  89. 7ffeab5 Reland "Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies." by kjellander@webrtc.org · 9 years ago
  90. f0fcbf3 Roll chromium_revision 405f35f..968c1e7 (377868:377935) by kjellander · 9 years ago
  91. cedddbd Android MediaCodecVideoDecoder: Limit measured decode time to 200ms by magjed · 9 years ago
  92. 3c16576 Don't allocate buffers for listening sockets. by jbauch · 9 years ago
  93. 9e69dfd Java SurfaceTextureHelper: Remove support for external thread by magjed · 9 years ago
  94. 54ebfca Revert of Cleanup of webrtc::VideoFrame. (patchset #6 id:100001 of https://codereview.webrtc.org/1679323002/ ) by kjellander · 9 years ago
  95. f8136ba Remove add/removal of RTP modules in PacketRouter. by Peter Boström · 9 years ago
  96. 8b79b07 Move RTP module activation into PayloadRouter. by Peter Boström · 9 years ago
  97. 9c01725 Simplify registration of RTP-header extensions. by Peter Boström · 9 years ago
  98. ff474da Roll chromium_revision 38664e7..405f35f (377790:377868) by kjellander · 9 years ago
  99. d6f6743 CQ: Add linux_ubsan to default trybots. by kjellander@webrtc.org · 9 years ago
  100. 2080196 Cleanup of webrtc::VideoFrame. by nisse · 9 years ago