1. f15a2c5 Delete deprecated versions of Copy, ScaleFrom and CropAndScaleFrom. by nisse · 8 years ago
  2. 0583b28 Collecting RTCIceCandidatePairStats.transport_id and improved unittests. by hbos · 8 years ago
  3. 0c43f77 Update video histograms that do not have a minimum lifetime limit before being recorded. by asapersson · 8 years ago
  4. 759e0b7 Fix memory leak in video_coding::PacketBuffer::InsertPacket. by philipel · 8 years ago
  5. be74270 Calculate JitterBufferDelayInMs in the new jitter buffer. by philipel · 8 years ago
  6. e69b468 Revert of Renaming AudioEncoder::SetTargetBitrate and SetProjectedPacketLossRate. (patchset #5 id:240001 of https://codereview.webrtc.org/2411613002/ ) by minyue · 8 years ago
  7. 1731c9c Use swap instead of copy in RtcHistogram::GetAndReset. by asapersson · 8 years ago
  8. 84e56d5 Renaming AudioEncoder::SetTargetBitrate and SetProjectedPacketLossRate. by minyue · 8 years ago
  9. 097529f Remove 3 defines in voice_engine_configurations.h by henrik.lundin · 8 years ago
  10. e61fbff Use RotateDesktopFrame in DirectX capturer by zijiehe · 8 years ago
  11. 166e59a Enable ScreenCapturerIntegrationTests by zijiehe · 8 years ago
  12. 6a46cf7 Roll chromium_revision db14e1adbd..b66d8ae9dc (435041:435081) by buildbot · 8 years ago
  13. c9e80ee Adding packet overhead to audio network adaptor. by minyue · 8 years ago
  14. 821dc7a Roll chromium_revision 683745f53c..db14e1adbd (434997:435041) by buildbot · 8 years ago
  15. a332877 Remove overhead from video bitrate. by michaelt · 8 years ago
  16. c4dc4a5 Adding RTCStatsIntegrationTest to memcheck supressions. by deadbeef · 8 years ago
  17. 75f9d8c Roll chromium_revision ffe8e7b51d..683745f53c (434954:434997) by buildbot · 8 years ago
  18. 290d43a Add a new UMA metric in APM to track incoming capture-side audio level by henrik.lundin · 8 years ago
  19. 939e08f Added webrtc/audio/utility directory and empty GN target. by aleloi · 8 years ago
  20. ee414d9 Added sanity check to VCMDecodingState::UsingFlexibleMode to prevent OOB error. by philipel · 8 years ago
  21. ad6f646 Use //build/dotfile_settings.gni to reduce blocked auto-rolls by kjellander · 8 years ago
  22. 768d625 Fix spelling mistake in RTP module declaration. by brandtr · 8 years ago
  23. b890c95c Replace some asserts with DCHECKs by kwiberg · 8 years ago
  24. 5049942 Refactor RMSLevel and give it new functionality by henrik.lundin · 8 years ago
  25. 1308c69 Roll chromium_revision 0496be2799..ffe8e7b51d (434847:434954) by buildbot · 8 years ago
  26. f17cae2 Cleanup unused rules in webrtc/DEPS + add kjellander to OWNERS for it by kjellander · 8 years ago
  27. 668eb3b Add overhead to transport feedback observer. by michaelt · 8 years ago
  28. 19223ac Ignore newly added resource files. by charujain · 8 years ago
  29. 455b512 Landmine to clobber Windows builders by Henrik Kjellander · 8 years ago
  30. 1b5b22d Fixed bug in ExtractFrameFromY4mFile API which was not extracting the frames correctly. by charujain · 8 years ago
  31. db346a7 RTCStatsIntegrationTest added. by hbos · 8 years ago
  32. 876222f Move usage of QualityScaler to ViEEncoder. by kthelgason · 8 years ago
  33. 320e45a Use RateCounter for input/sent fps stats. Reports average of periodically computed stats over a call. by asapersson · 8 years ago
  34. 65e5f5a Roll chromium_revision d74a300097..0496be2799 (434704:434847) by buildbot · 8 years ago
  35. bdd6f4c Adding memcheck suppression. by deadbeef · 8 years ago
  36. 6cf94a0 Only use BoringSSL time callback in unit tests. by deadbeef · 8 years ago
  37. 352444f RTC_[D]CHECK_op: Remove superfluous casts by kwiberg · 8 years ago
  38. af476c7 RTC_[D]CHECK_op: Remove "u" suffix on integer constants by kwiberg · 8 years ago
  39. 80ed35e Implement periodic bandwidth probing in application-limited region. by sergeyu · 8 years ago
  40. bf22be9 Roll chromium_revision 2b5aa49038..d74a300097 (434640:434704) by buildbot · 8 years ago
  41. fd87f4a Opus: Move complexity variable out of conditional build flag by henrik.lundin · 8 years ago
  42. 1bc3146 Disable more VideoProcessorIntegrationTest tests on Linux 32-bit by Henrik Kjellander · 8 years ago
  43. bb58435 Fix potential synchronization issues with framelisteners in EglRenderer. by sakal · 8 years ago
  44. 266f0a4 Now run EndToEndTest with the WebRTC-NewVideoJitterBuffer experiment. by philipel · 8 years ago
  45. d1aaaaa Remove surface size mismatch logic from EglRenderer. by sakal · 8 years ago
  46. 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
  47. 7703b27 Disable PeerConnectionEndToEndTest.CallWithLegacySdp on Asan bots. by philipel · 8 years ago
  48. ceecea4 Pass selected cricket::VideoCodec down to internal H264 encoder by magjed · 8 years ago
  49. 20dce34 Fixed bug in PacketBuffer to correctly detect new complete frames after ClearTo has been called. by philipel · 8 years ago
  50. e1a13f8 MB: Remove a --target-devices-file flag for JUnit tests on android. by ehmaldonado · 8 years ago
  51. a8eb756 Moved transport.h from webrtc/ to webrtc/api, created build target and updated WebRTC dependencies. by aleloi · 8 years ago
  52. 9abbf5a Pass time constanct to bwe smoothing filter. by michaelt · 8 years ago
  53. ffc6118 Don't cache video codec list in VideoEngine2. by brandtr · 8 years ago
  54. ec1a670 Only create |remote_rate| when needed in RemoteBitrateEstimatorSingleStream. by Rasmus Brandt · 8 years ago
  55. fb4a37a Add memcheck suppressions. by ehmaldonado · 8 years ago
  56. 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
  57. 566cba1 Roll chromium_revision 5c22c2afac..2b5aa49038 (434448:434640) by buildbot · 8 years ago
  58. b7636b4 Fixed bug in ExtractFrameFromY4mFile API which was not extracting the frames correctly. by charujain · 8 years ago
  59. 2f58ec8 Add I420Buffer::Copy method taking plane pointers as input. by nisse · 8 years ago
  60. e441bdb Cleanup RtpSender hiding RtpHeaderExtensionLength function. by danilchap · 8 years ago
  61. 2fedf9c Smooth BWE and pass it to Audio Network Adaptor. by michaelt · 8 years ago
  62. 847f689 Roll chromium_revision 5e821a778b..5c22c2afac (432715:434448) by kjellander · 8 years ago
  63. deb95f3 Change rtc::TimeNanos and rtc::TimeMicros return value from uint64_t to int64_t. by nisse · 8 years ago
  64. 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
  65. e8d8a2b Move ADM specific Android files into modules/audio_device/android/ by solenberg · 8 years ago
  66. 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
  67. 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
  68. c7805db Fix perf regression in screenshare temporal layer bitrate allocation by sprang · 8 years ago
  69. fd34d30 iOS HW encoder: Enable H264 High profile support by magjed · 8 years ago
  70. bdbc4b7 Add H264 profile to webrtc::VideoCodecH264 and webrtc::VideoPayload by magjed · 8 years ago
  71. 1da1a09 Android HW encoder: Set constrained baseline as the profile by magjed · 8 years ago
  72. 03d6b08 Get rid of webrtc/base/latebindingsymboltable* by ehmaldonado · 8 years ago
  73. f3feeff Reland of move RTPPayloadStrategy and simplify RTPPayloadRegistry (patchset #1 id:1 of https://codereview.webrtc.org/2528993002/ ) by magjed · 8 years ago
  74. 0fa164a Make Valgrind memcheck work in swarming. by ehmaldonado · 8 years ago
  75. 5732910 Revert of CQ: Disable android_more_configs trybot (patchset #1 id:1 of https://codereview.webrtc.org/2522953003/ ) by ehmaldonado · 8 years ago
  76. 76622ce Adding a unit test for RMSLevel by henrik.lundin · 8 years ago
  77. 293bc2a Add 'Update LASTCHANGE' hook to DEPS by ehmaldonado · 8 years ago
  78. 5f7226f Turn off error resilience for vp8 for no temporal layers if nack is enabled. by asapersson · 8 years ago
  79. 5dfac56 Keep all codec parameters in VideoReceiveStream::Decoder by magjed · 8 years ago
  80. a6a699a Sent bitrate stats are incorrect if FlexFEC is configured: by asapersson · 8 years ago
  81. 6b272c5 RtpReceiver: Add RegisterReceivePayload function for VideoCodec by magjed · 8 years ago
  82. 5de9b6a Move helpers_ios.cc/.h by solenberg · 8 years ago
  83. 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
  84. 33c81d0 Revert of Remove RTPPayloadStrategy and simplify RTPPayloadRegistry (patchset #7 id:240001 of https://codereview.webrtc.org/2524923002/ ) by magjed · 8 years ago
  85. 69b627d Move smoothing filter to common audio and exp_filter to base/analytics. by minyue · 8 years ago
  86. b881254 Remove RTPPayloadStrategy and simplify RTPPayloadRegistry by magjed · 8 years ago
  87. 56124bd Send audio and video codecs to RTPPayloadRegistry by magjed · 8 years ago
  88. b7374db Fix parsing padding byte in rtp header extension by danilchap · 8 years ago
  89. bf67663 Rename "Audio playout level" to "Audio level" on the Y-axis of the event log graph. by ivoc · 8 years ago
  90. 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
  91. 223641f Reland "Move smoothing filter to common audio". by minyue · 8 years ago
  92. 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
  93. d8ae20b Modify the paths of the resource files to point to chromium/src/tools/... by ehmaldonado · 8 years ago
  94. 3cfb3ef Added a perf test for the residual echo detector. by ivoc · 8 years ago
  95. 37a2111 Increase the threshold for RunPlayoutAndRecordingInFullDuplex. Again. by ehmaldonado · 8 years ago
  96. 3edc7f0 AGC: Add a histogram for new level by henrik.lundin · 8 years ago
  97. c42d376 DataChannelInterface: Remove default implementation of methods. by hbos · 8 years ago
  98. 464d50f Set rtc_use_memcheck=true for the FYI bot. by ehmaldonado · 8 years ago
  99. 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
  100. d44d0ba For VPN network, use the underlying network type as its type. by honghaiz · 8 years ago