1. 823f908 Switch voice transport to use Call and Stream instead of VoENetwork. by mflodman · 9 years ago
  2. 29dbd8a Remove calls to ScopedToUnique and UniqueToScoped by kwiberg · 9 years ago
  3. 5fb5bd2 #include "webrtc/base/constructormagic.h" where appropriate by kwiberg · 9 years ago
  4. 413dd56 Fix test.gyp dependency. by nisse · 9 years ago
  5. 450784a Refactored CL for moving the output to a separate thread. by terelius · 9 years ago
  6. cad4296 Disable SwitchesToASTThenBackToTOFForVideo test completely. by deadbeef · 9 years ago
  7. 512897c Update the call when the network route changes by Honghai Zhang · 9 years ago
  8. 46b2024 Disabling SwitchesToASTThenBackToTOFForVideo test for MSan bot. by deadbeef · 9 years ago
  9. b6a1179 Suppress a flaky test: SwitchesToASTThenBackToTOFForVideo. by minyuel · 9 years ago
  10. 63b8d3f Add macros for ability to log samples that are added to histograms (RTC_LOGGED_*). by asapersson · 9 years ago
  11. a2761a5 Delete class webrtc::VideoRenderer and its header file. by nisse · 9 years ago
  12. ee1e4ed Make the audio channel communicate network state changes to the call. by skvlad · 9 years ago
  13. f8e52ce This is an initial cleanup step, aiming to delete the by nisse · 9 years ago
  14. 0b54e5a Reland "Add check_deps rules in DEPS files." by kjellander@webrtc.org · 9 years ago
  15. de82d23 Revert of Add check_deps rules in DEPS files. (patchset #2 id:60001 of https://codereview.webrtc.org/1796413002/ ) by kjellander · 9 years ago
  16. 181e867 Add check_deps rules in DEPS files. by kjellander@webrtc.org · 9 years ago
  17. cf2fc20 Clean away use of RtpAudioFeedback interface from RTP/RTCP sender code. by solenberg · 9 years ago
  18. a637d2c Add histogram stats for AV sync stream offset: "WebRTC.Video.AVSyncOffsetInMs" by asapersson · 9 years ago
  19. 4185d51 Replace scoped_ptr with unique_ptr in webrtc/call/ by kwiberg · 9 years ago
  20. 3e089d3 Move BitrateAllocator reference from ViEEncoder to VideoSendStream. by mflodman · 9 years ago
  21. 06e85e9 Make ReconfigureVideoEncoder void. by Peter Boström · 9 years ago
  22. 9c0599f VideoCaptureInput enforce VideoFrame::render_time to be generated by webrtc clock. by danilchap · 9 years ago
  23. 579eea4 Break out pacer thread from CongestionController to increase testability. by Stefan Holmer · 9 years ago
  24. 85150f6 Disabled flaky tests by philipel · 9 years ago
  25. ead3cf2 Move congestion controller to a separate module. by Stefan Holmer · 9 years ago
  26. 5ba7bdf Use CallStats for RTT in Call, rather than VideoSendStream::GetRtt() by sprang · 9 years ago
  27. 1262b9d Simplify CongestionController. by Stefan Holmer · 9 years ago
  28. 54fc3d3 [rtp_rtcp] rtc::scoped_ptr<rtcp::RawPacket> replaced with rtc::Buffer by danilchap · 9 years ago
  29. 8d1de81 Update bitrate only when we have incoming packet. by Stefan Holmer · 9 years ago
  30. a96ea33 removed five redundant avsync tests to make webrtc_perf_test faster by Danil Chapovalov · 9 years ago
  31. 25b4a62 Initialize VideoSendStream members in constructor. by Peter Boström · 9 years ago
  32. a4777ed Added A/V sync tests with drifting clocks. by danilchap · 9 years ago
  33. ed50be1 Clean up of CongestionController. by Stefan Holmer · 9 years ago
  34. 74c29e2 Add send-side BWE to WebRtcVoiceEngine under a finch experiment. by stefan · 9 years ago
  35. 3dad57b Use separate rtp module lists for send and receive in PacketRouter. by stefan · 9 years ago
  36. b295e77 Switch to use new implementation in metrics.h. by asapersson · 9 years ago
  37. 1e5b805 Remove mutable from rtc::CriticalSection members. by pbos · 9 years ago
  38. f0c9607 Fixes a bug which incorrectly logs incoming RTCP as outgoing. by terelius · 9 years ago
  39. 10a9538 Support REMB in combination with send-side BWE. by stefan · 9 years ago
  40. e2095e4 Switch use of CriticalSectionWrapper -> rtc::CriticalSection in call/ by tommi · 9 years ago
  41. d146002 Remove extra_options from VideoCodec. by Peter Boström · 9 years ago
  42. 8d5f298 [rtp_rtcp] rtcp::SenderReport moved into own file and got Parse function by danilchap · 9 years ago
  43. 5bfff64 Add ramp-up tests for transport sequence number with and w/o audio. by Stefan Holmer · 9 years ago
  44. cbbb3b1 Fix capture ntp time issue introduced with r11187. by Stefan Holmer · 9 years ago
  45. 5af14d7 Fix test bug introduced in r11101. by Stefan Holmer · 9 years ago
  46. 87f3db7 Wire-up BWE feedback for audio receive streams. by Stefan Holmer · 9 years ago
  47. a21e4b9 Disable RampUpTest.UpDownUp* in webrtc_perf_tests on Mac by kjellander · 9 years ago
  48. 26f9c18 Add CreateSend/ReceiveTransport() methods to CallTest. by stefan · 9 years ago
  49. 8c0d4cb Add audio streams to CallTest and a first A/V call test. by Stefan Holmer · 9 years ago
  50. 98836c0 Remove DISABLED_ON_ macros. by Peter Boström · 9 years ago
  51. 19cc283 Step 1 to prepare call_test.* for combined audio/video tests. by stefan · 9 years ago
  52. 4de01f8 Rename RTC_HISTOGRAM_* macros to RTC_HISTOGRAM_*_SPARSE_* to indicate that these are for infrequent updates. by asapersson · 9 years ago
  53. 1cddafe Replace EventWrapper in video/, test/ and call/. by Peter Boström · 9 years ago
  54. 0739714 Rewrote the PRNG using an xorshift* algorithm and moved the files from test/ to base/. by terelius · 9 years ago
  55. e8f0735 Reland of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:300001 of https://codereview.webrtc.org/1507903005/ ) by Peter Boström · 9 years ago
  56. a24951b Nuke TickTime::UseFakeClock. by Peter Boström · 9 years ago
  57. 32949e5 Revert of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:20001 of https://codereview.webrtc.org/1506773002/ ) by kjellander · 9 years ago
  58. 5be013d Merge webrtc/video_engine/ into webrtc/video/ by Peter Boström · 9 years ago
  59. 155895c Implement standalone event tracing in AppRTCDemo. by Peter Boström · 9 years ago
  60. f95302f Prepare the AudioSendStream to be hooked up to send-side BWE. by Stefan Holmer · 9 years ago
  61. e0dd44e Use webrtc/base/logging.h in stefan@'s ownership. by Peter Boström · 9 years ago
  62. edbb7ba Enable cpplint for webrtc/audio and webrtc/call, and fix all uncovered cpplint errors. by Fredrik Solenberg · 9 years ago
  63. 1834d13 Remove duplicate decoders in BitrateEstimatorTest. by Peter Boström · 9 years ago
  64. 9780212 Rewrote pacer and bandwidth UMA stats. by Stefan Holmer · 9 years ago
  65. 0c1546c audio_coding: remove "main" directory by kjellander · 9 years ago
  66. 9ab9f46 Add UMA for send bwe and pacer bitrate. by stefan · 9 years ago
  67. ffe1ce0 Move some send stream configuration into webrtc::AudioSendStream. by solenberg · 9 years ago
  68. 273352e Remove include dirs from modules/{media_file,pacing} by Henrik Kjellander · 9 years ago
  69. 1b78cc3 Move BitrateAllocator from BitrateController logic to Call. by mflodman · 9 years ago
  70. 73807f4 Re-add a thread check in Call::Call that was removed by mistake in a rebase. by solenberg · 9 years ago
  71. 0e2506e Add receive bitrate UMA stats. by stefan · 9 years ago
  72. b0f22c5 Move VoiceEngineObserver into AudioSendStream so that we detect typing noises and return properly in GetStats(). by solenberg · 9 years ago
  73. e37a013 Removed cname and receiver_reference_time_report from proto and logging code. Changed logging of RTCP to omit messages of type SDES and APP. by terelius · 9 years ago
  74. ad8bc01 Change to use local Random object instead of global rand() in the RtcEventLog unit test. by terelius · 9 years ago
  75. cfdad3c Added protobuf message for loss-based BWE events, and wired it up to the send side bandwidth estimator. by terelius · 9 years ago
  76. 2cded7a Set pacer target bitrate to max of bwe and bitrate allocation. by sprang · 9 years ago
  77. 36a14b5 modules: more interface -> include renames by Henrik Kjellander · 9 years ago
  78. 4f247a6 Changed FakeVoiceEngine into a MockVoiceEngine. by Fredrik Solenberg · 9 years ago
  79. 602ff1f audio_coding: rename interface -> include by Henrik Kjellander · 9 years ago
  80. 78f65d0 system_wrappers: rename interface -> include by Henrik Kjellander · 9 years ago
  81. f5a44e8 Call OnSentPacket for all packets sent in the test framework. by stefan · 9 years ago
  82. 1c28f5c Implement AudioSendStream::GetStats(). by solenberg · 9 years ago
  83. 18a79dc Remove network_enabled_crit_ in call.cc. by mflodman · 9 years ago
  84. 379f274 Set send times in send time history via OnSentPacket. by stefan · 9 years ago
  85. 10762d3 Re-Land: Implement AudioReceiveStream::GetStats(). by Fredrik Solenberg · 9 years ago
  86. b62f41a Rename ChannelGroup to CongestionController and move to webrtc/call/. by mflodman · 9 years ago
  87. d5b6d22 ChannelGroup cleanup. by mflodman · 9 years ago
  88. 512a784 Remove system_wrappers/interface/trace_event.h by tommi · 9 years ago
  89. 0e9f679 Revert of Implement AudioReceiveStream::GetStats(). (patchset #19 id:360001 of https://codereview.webrtc.org/1390753002/ ) by solenberg · 9 years ago
  90. bbb922f Implement AudioReceiveStream::GetStats(). by Fredrik Solenberg · 9 years ago
  91. fd3f2cd Remove the video channel id completely. by mflodman · 9 years ago
  92. 6db0478 Added thread checker to webrtc::Call. by solenberg · 9 years ago
  93. 7d39383 Move ownership of receive ViEChannel to VideoReceiveStream. by mflodman · 9 years ago
  94. f707c68 Add webrtc::AudioSendStream and methods on webrtc::Call to create and delete AudioSendStreams. by solenberg · 9 years ago
  95. 2676348 Old config events are no longer removed from RtcEventLog. by terelius · 9 years ago
  96. 922e8ed Added functions on libjingle API to start and stop the recording of an RtcEventLog. by ivoc · 9 years ago
  97. 15b2099 Wire up packet_id / send time callbacks to webrtc via libjingle. by stefan · 9 years ago
  98. f863304 Log Call {audio, video} stream deletions. by pbos · 9 years ago
  99. df65b2e Pause/resume pacer from Call instead of via SendStreams. by stefan · 9 years ago
  100. 3939297 Update to the RtcEventLog protobuf to remove the DebugEvent message. by Ivo Creusen · 9 years ago