1. 668eb3b Add overhead to transport feedback observer. by michaelt · 8 years ago
  2. 19223ac Ignore newly added resource files. by charujain · 8 years ago
  3. 455b512 Landmine to clobber Windows builders by Henrik Kjellander · 8 years ago
  4. 1b5b22d Fixed bug in ExtractFrameFromY4mFile API which was not extracting the frames correctly. by charujain · 8 years ago
  5. db346a7 RTCStatsIntegrationTest added. by hbos · 8 years ago
  6. 876222f Move usage of QualityScaler to ViEEncoder. by kthelgason · 8 years ago
  7. 320e45a Use RateCounter for input/sent fps stats. Reports average of periodically computed stats over a call. by asapersson · 8 years ago
  8. 65e5f5a Roll chromium_revision d74a300097..0496be2799 (434704:434847) by buildbot · 8 years ago
  9. bdd6f4c Adding memcheck suppression. by deadbeef · 8 years ago
  10. 6cf94a0 Only use BoringSSL time callback in unit tests. by deadbeef · 8 years ago
  11. 352444f RTC_[D]CHECK_op: Remove superfluous casts by kwiberg · 8 years ago
  12. af476c7 RTC_[D]CHECK_op: Remove "u" suffix on integer constants by kwiberg · 8 years ago
  13. 80ed35e Implement periodic bandwidth probing in application-limited region. by sergeyu · 8 years ago
  14. bf22be9 Roll chromium_revision 2b5aa49038..d74a300097 (434640:434704) by buildbot · 8 years ago
  15. fd87f4a Opus: Move complexity variable out of conditional build flag by henrik.lundin · 8 years ago
  16. 1bc3146 Disable more VideoProcessorIntegrationTest tests on Linux 32-bit by Henrik Kjellander · 8 years ago
  17. bb58435 Fix potential synchronization issues with framelisteners in EglRenderer. by sakal · 8 years ago
  18. 266f0a4 Now run EndToEndTest with the WebRTC-NewVideoJitterBuffer experiment. by philipel · 8 years ago
  19. d1aaaaa Remove surface size mismatch logic from EglRenderer. by sakal · 8 years ago
  20. 6287e82 Revert of Pass time constant to bwe smoothing filter. (patchset #8 id:140001 of https://codereview.webrtc.org/2518923003/ ) by ossu · 8 years ago
  21. 7703b27 Disable PeerConnectionEndToEndTest.CallWithLegacySdp on Asan bots. by philipel · 8 years ago
  22. ceecea4 Pass selected cricket::VideoCodec down to internal H264 encoder by magjed · 8 years ago
  23. 20dce34 Fixed bug in PacketBuffer to correctly detect new complete frames after ClearTo has been called. by philipel · 8 years ago
  24. e1a13f8 MB: Remove a --target-devices-file flag for JUnit tests on android. by ehmaldonado · 8 years ago
  25. a8eb756 Moved transport.h from webrtc/ to webrtc/api, created build target and updated WebRTC dependencies. by aleloi · 8 years ago
  26. 9abbf5a Pass time constanct to bwe smoothing filter. by michaelt · 8 years ago
  27. ffc6118 Don't cache video codec list in VideoEngine2. by brandtr · 8 years ago
  28. ec1a670 Only create |remote_rate| when needed in RemoteBitrateEstimatorSingleStream. by Rasmus Brandt · 8 years ago
  29. fb4a37a Add memcheck suppressions. by ehmaldonado · 8 years ago
  30. 26fa6b2 Revert of Bug in ExtractFrame API (extracts frames incorrectly) (patchset #9 id:130001 of https://codereview.webrtc.org/2529923002/ ) by charujain · 8 years ago
  31. 566cba1 Roll chromium_revision 5c22c2afac..2b5aa49038 (434448:434640) by buildbot · 8 years ago
  32. b7636b4 Fixed bug in ExtractFrameFromY4mFile API which was not extracting the frames correctly. by charujain · 8 years ago
  33. 2f58ec8 Add I420Buffer::Copy method taking plane pointers as input. by nisse · 8 years ago
  34. e441bdb Cleanup RtpSender hiding RtpHeaderExtensionLength function. by danilchap · 8 years ago
  35. 2fedf9c Smooth BWE and pass it to Audio Network Adaptor. by michaelt · 8 years ago
  36. 847f689 Roll chromium_revision 5e821a778b..5c22c2afac (432715:434448) by kjellander · 8 years ago
  37. deb95f3 Change rtc::TimeNanos and rtc::TimeMicros return value from uint64_t to int64_t. by nisse · 8 years ago
  38. 71b9b58 Revert of Move ADM specific Android files into modules/audio_device/android/ (patchset #2 id:20001 of https://codereview.webrtc.org/2533573002/ ) by solenberg · 8 years ago
  39. e8d8a2b Move ADM specific Android files into modules/audio_device/android/ by solenberg · 8 years ago
  40. e69a1a9 Reland of Add H264 profile to webrtc::VideoCodecH264 and webrtc::VideoPayload (patchset #1 id:1 of https://codereview.webrtc.org/2529143002/ ) by magjed · 8 years ago
  41. d7e6ccb Revert of Add H264 profile to webrtc::VideoCodecH264 and webrtc::VideoPayload (patchset #1 id:40001 of https://codereview.webrtc.org/2525693003/ ) by magjed · 8 years ago
  42. c7805db Fix perf regression in screenshare temporal layer bitrate allocation by sprang · 8 years ago
  43. fd34d30 iOS HW encoder: Enable H264 High profile support by magjed · 8 years ago
  44. bdbc4b7 Add H264 profile to webrtc::VideoCodecH264 and webrtc::VideoPayload by magjed · 8 years ago
  45. 1da1a09 Android HW encoder: Set constrained baseline as the profile by magjed · 8 years ago
  46. 03d6b08 Get rid of webrtc/base/latebindingsymboltable* by ehmaldonado · 8 years ago
  47. f3feeff Reland of move RTPPayloadStrategy and simplify RTPPayloadRegistry (patchset #1 id:1 of https://codereview.webrtc.org/2528993002/ ) by magjed · 8 years ago
  48. 0fa164a Make Valgrind memcheck work in swarming. by ehmaldonado · 8 years ago
  49. 5732910 Revert of CQ: Disable android_more_configs trybot (patchset #1 id:1 of https://codereview.webrtc.org/2522953003/ ) by ehmaldonado · 8 years ago
  50. 76622ce Adding a unit test for RMSLevel by henrik.lundin · 8 years ago
  51. 293bc2a Add 'Update LASTCHANGE' hook to DEPS by ehmaldonado · 8 years ago
  52. 5f7226f Turn off error resilience for vp8 for no temporal layers if nack is enabled. by asapersson · 8 years ago
  53. 5dfac56 Keep all codec parameters in VideoReceiveStream::Decoder by magjed · 8 years ago
  54. a6a699a Sent bitrate stats are incorrect if FlexFEC is configured: by asapersson · 8 years ago
  55. 6b272c5 RtpReceiver: Add RegisterReceivePayload function for VideoCodec by magjed · 8 years ago
  56. 5de9b6a Move helpers_ios.cc/.h by solenberg · 8 years ago
  57. 0928a3c Reland of Split out target rtc_media_base from rtc_media (patchset #1 id:1 of https://codereview.webrtc.org/2508163002/ ) by magjed · 8 years ago
  58. 33c81d0 Revert of Remove RTPPayloadStrategy and simplify RTPPayloadRegistry (patchset #7 id:240001 of https://codereview.webrtc.org/2524923002/ ) by magjed · 8 years ago
  59. 69b627d Move smoothing filter to common audio and exp_filter to base/analytics. by minyue · 8 years ago
  60. b881254 Remove RTPPayloadStrategy and simplify RTPPayloadRegistry by magjed · 8 years ago
  61. 56124bd Send audio and video codecs to RTPPayloadRegistry by magjed · 8 years ago
  62. b7374db Fix parsing padding byte in rtp header extension by danilchap · 8 years ago
  63. bf67663 Rename "Audio playout level" to "Audio level" on the Y-axis of the event log graph. by ivoc · 8 years ago
  64. 3c3aef4 Revert of Reland "Move smoothing filter to common audio". (patchset #5 id:100001 of https://codereview.webrtc.org/2520003005/ ) by minyue · 8 years ago
  65. 223641f Reland "Move smoothing filter to common audio". by minyue · 8 years ago
  66. b365b80 Revert of Modify the paths of the resource files to point to chromium/src/tools/... (patchset #1 id:1 of https://codereview.webrtc.org/2528893002/ ) by ehmaldonado · 8 years ago
  67. d8ae20b Modify the paths of the resource files to point to chromium/src/tools/... by ehmaldonado · 8 years ago
  68. 3cfb3ef Added a perf test for the residual echo detector. by ivoc · 8 years ago
  69. 37a2111 Increase the threshold for RunPlayoutAndRecordingInFullDuplex. Again. by ehmaldonado · 8 years ago
  70. 3edc7f0 AGC: Add a histogram for new level by henrik.lundin · 8 years ago
  71. c42d376 DataChannelInterface: Remove default implementation of methods. by hbos · 8 years ago
  72. 464d50f Set rtc_use_memcheck=true for the FYI bot. by ehmaldonado · 8 years ago
  73. ed8c8ed Add rtc_use_memcheck flag, update MB and GN to handle it, and add gni files listing the runtime deps by ehmaldonado · 8 years ago
  74. d44d0ba For VPN network, use the underlying network type as its type. by honghaiz · 8 years ago
  75. 4dfb8ce Make the default value of rtcp-mux policy to required. by zhihuang · 8 years ago
  76. e02407a Add myself to WATCHLIST for api/. by solenberg · 8 years ago
  77. 42eee12 RTCPeerConnectionStats: Removed fixed TODO comments. by hbos · 8 years ago
  78. 08be780 Reland of Allow custom metrics implementations on Android. (patchset #1 id:1 of https://codereview.webrtc.org/2516403002/ ) by sakal · 8 years ago
  79. 817208b Re-enables AudioDeviceTest.StartStopPlayout on Android by henrika · 8 years ago
  80. 8b64628 Add fps reduction API to SurfaceViewRenderer. by sakal · 8 years ago
  81. 4fe3b8d Add framelistener functionality to SurfaceViewRenderer. by sakal · 8 years ago
  82. 1c82884 Remove binding framebuffer from GlTextureFrameBuffer.setSize. by sakal · 8 years ago
  83. 8e321c8 CQ: Disable android_more_configs trybot by Henrik Kjellander · 8 years ago
  84. 0c5a154 Try to deflake VideoSendStream tests with FlexFEC. by brandtr · 8 years ago
  85. 0adb828 RTCCodecStats[1] added. by hbos · 8 years ago
  86. 71caaca Split avfoundationcapturer classes in separate files. by denicija · 8 years ago
  87. 90ea736 Add DesktopFrame rotation functions by zijiehe · 8 years ago
  88. e2b1501 Start probes only after network is connected. by Sergey Ulanov · 8 years ago
  89. 1c062bf Fix module/desktop_capture compilation on iOS by Sergey Ulanov · 8 years ago
  90. c1dd1a5 Really disable Opus complexity tests on Android by henrik.lundin · 8 years ago
  91. d661e9c WebRTC: Replace ProjectRootPath by ResourcePath by ehmaldonado · 8 years ago
  92. 10165ab Unify VideoCodecType to/from string functionality by magjed · 8 years ago
  93. 2d60e53 H264 encoder: Include QP information in encoded images by magjed · 8 years ago
  94. e60f020 iOS AppRTCMobile: Fix SDP video codec reordering for multiple H264 profiles by magjed · 8 years ago
  95. 8271d04 This CL introduces the new functionality for setting by peah · 8 years ago
  96. 30a12fb AGC: Add a histogram for clipping adjustment by henrik.lundin · 8 years ago
  97. 24d812d DEPS: Specify WebRTC hooks and add a few dependencies by kjellander · 8 years ago
  98. ab6996d Enable QP parsing from CABAC bitstreams by kthelgason · 8 years ago
  99. 04c0722 Replace AudioConferenceMixer with AudioMixer. by aleloi · 8 years ago
  100. b426040 Add Full HD and 4K camera resolutions to AppRTCMobile Android. by sakal · 8 years ago