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webrtc / src / webrtc / 63ab26e1fc88ba3219e40784c61675e203e319e9 / media / base
  1. acbc4e8 Simplify passing video coded factories in media engine by magjed · 8 years ago
  2. 642a074 Update thread annotiation macros to use RTC_ prefix by danilchap · 8 years ago
  3. 3aa28f0 Add new ANA stats to the old GetStats() to count the number of actions taken by each controller. by ivoc · 8 years ago
  4. 6ae8262 Add reporting of googContentType via GetStats on send side by ilnik · 8 years ago
  5. c4f78c0 Enable UBSan float-cast-overflow warnings and fix existing ones by oprypin · 8 years ago
  6. 5b18967 Move optional.h to webrtc/api/ by kwiberg · 8 years ago
  7. 1f11d1a Implement googContentType GetStats metric reported on receive side. by ilnik · 8 years ago
  8. 36189cd Move RtpExtension to api/ directory and config.h/.cc to call/. by Stefan Holmer · 8 years ago
  9. 443f9a9 Wiring discard rate of audio FEC/RED packets up to StatsReport. by minyue-webrtc · 8 years ago
  10. 94ac82f Add stats totalSamplesReceived and concealedSamples by Steve Anton · 8 years ago
  11. 6a5ac8f Report max interframe delay over window insdead of interframe delay sum by ilnik · 8 years ago
  12. f0c86c0 Move video send/receive stream headers to webrtc/call. by aleloi · 8 years ago
  13. 6ab33cf Wire up RTP keep-alive in ortc api. by sprang · 8 years ago
  14. d1701d0 Adding stats that can be used to compute output audio levels as described here https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudioenergy. by zstein · 8 years ago
  15. 4b941f0 Report interframe delay sum in old GetStats by ilnik · 8 years ago
  16. 76de83e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 8 years ago
  17. 37342b9 Report timing frames info in GetStats. by ilnik · 8 years ago
  18. bc32410 Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 8 years ago
  19. 60154fd Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 8 years ago
  20. 0d58090 Support encrypted RTP extensions (RFC 6904) by jbauch · 8 years ago
  21. 588f761 Allow an external audio processing module to be used in WebRTC by peah · 8 years ago
  22. b49c6a7 Remove unused #include "libyuv/compare.h" by eladalon · 8 years ago
  23. 5e343ec Delete SignalSrtpError. by nisse · 8 years ago
  24. f91805c Support building WebRTC without audio and video. by zhihuang · 8 years ago
  25. f94a820 Allow WebRtcMediaEngine to be created from any thread. by deadbeef · 8 years ago
  26. f7f8eb4 Move MinPositive to call.h as discussed here: https://codereview.chromium.org/2888303005/#msg19 by zstein · 8 years ago
  27. 5dc4393 Update webrtc/media and webrtc/modules to new VideoFrameBuffer interface by Magnus Jedvert · 8 years ago
  28. 3d545a2 s/WebRtcVideoChannel2/WebRtcVideoChannel and s/WebRtcVideoEngine2/WebRtcVideoEngine by eladalon · 8 years ago
  29. 4e13fcb Revert "Revert of Wire up BWE stats through WebrtcSession so that they are filled in both for audio and video calls. (patchset #8 id:140001 of https://codereview.webrtc.org/2863123002/ )" by stefan · 8 years ago
  30. df24c99 Move RTP/RTCP demuxing logic from BaseChannel to RtpTransport. by zstein · 8 years ago
  31. 5acd6fa Update I420Buffer to new VideoFrameBuffer interface by magjed · 8 years ago
  32. b6a2c8f Revert of Wire up BWE stats through WebrtcSession so that they are filled in both for audio and video calls. (patchset #8 id:140001 of https://codereview.webrtc.org/2863123002/ ) by charujain · 8 years ago
  33. ddb82e2 Wire up BWE stats through WebrtcSession so that they are filled in both for audio and video calls. by Stefan Holmer · 8 years ago
  34. d1df7af Reland of Delete deprecated and transitional stuff related to video frame refactoring. (patchset #1 id:1 of https://codereview.webrtc.org/2853383005/ ) by nisse · 8 years ago
  35. e6bd325 Update comments for removal of MediaController. by nisse · 8 years ago
  36. 5af64de Revert of Delete deprecated and transitional stuff related to video frame refactoring. (patchset #1 id:1 of https://codereview.webrtc.org/2854873003/ ) by nisse · 8 years ago
  37. d9704a0 Reland of Delete deprecated and transitional stuff related to video frame refactoring. (patchset #1 id:1 of https://codereview.webrtc.org/2854883002/ ) by nisse · 8 years ago
  38. d477b8c Revert of Delete deprecated and transitional stuff related to video frame refactoring. (patchset #1 id:1 of https://codereview.webrtc.org/2852303002/ ) by nisse · 8 years ago
  39. f26202b Injectable audio encoders: Moved audio encoder, factory and builtin factory to api/. by ossu · 8 years ago
  40. 47f48ce Reland of Delete deprecated and transitional stuff related to video frame refactoring. (patchset #1 id:1 of https://codereview.webrtc.org/2845333002/ ) by nisse · 8 years ago
  41. 0173390 Revert of Delete deprecated and transitional stuff related to video frame refactoring. (patchset #17 id:320001 of https://codereview.webrtc.org/2622263002/ ) by nisse · 8 years ago
  42. 3244692 Delete deprecated and transitional stuff related to video frame refactoring. by nisse · 8 years ago
  43. 5c7d0c4 Normalize codec names to those used by AcmCodecDatabase. by ossu · 8 years ago
  44. 37a786a Fix RtpReceiver.GetParameters when SSRCs aren't signaled. by deadbeef · 8 years ago
  45. ff493f2 Fix SDP stream ID mismatch issue when a track's stream changes. by deadbeef · 8 years ago
  46. 04c660e Don't add stuff to namespace std by kwiberg · 8 years ago
  47. 7950bab Reland of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #1 id:1 of https://codereview.webrtc.org/2783183003/ ) by sprang · 8 years ago
  48. 14e0920 Revert of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #8 id:410001 of https://codereview.webrtc.org/2781433002/ ) by lliuu · 8 years ago
  49. acef169 Reland of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #1 id:1 of https://codereview.webrtc.org/2764133002/ ) by sprang · 8 years ago
  50. cd05a9e Rewrite PeerConnection integration tests using better testing practices. by deadbeef · 8 years ago
  51. 63d1630 Revert of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #14 id:250001 of https://codereview.webrtc.org/2716643002/ ) by skvlad · 8 years ago
  52. 3647ae7 Add framerate to VideoSinkWants and ability to signal on overuse by sprang · 8 years ago
  53. d22d96d Fix non-functional nit in videoadapter by magjed · 8 years ago
  54. ad5d1e6 Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc. by deadbeef · 8 years ago
  55. eb8fabc Fix issue where video scaling gets stuck at low resolution by Magnus Jedvert · 8 years ago
  56. fa96717 Implement operator<< for AudioCodec by kwiberg · 8 years ago
  57. f2c0aed Switching some interfaces to use std::unique_ptr<>. by deadbeef · 8 years ago
  58. f7c7480 Move AudioDecoder and related stuff to the api/ directory by kwiberg · 8 years ago
  59. 6d680fa Change rtc::VideoSinkWants to have target and a max pixel count by sprang · 8 years ago
  60. 68f7c03 Remove unused methods in WebRtcVoiceEngine and VoiceMediaChannel. by solenberg · 8 years ago
  61. bc6405b Reland of Add QP sum stats for received streams. (patchset #2 id:300001 of https://codereview.webrtc.org/2680893002/ ) by sakal · 8 years ago
  62. a8773b4 Revert of Add QP sum stats for received streams. (patchset #10 id:180001 of https://codereview.webrtc.org/2649133005/ ) by skvlad · 8 years ago
  63. 647a225 Replace all use of the VERIFY macro. by nisse · 8 years ago
  64. f2d1765 Add QP sum stats for received streams. by sakal · 8 years ago
  65. 8124838 Adding C++ versions of currently spec'd "RtpParameters" structs. by deadbeef · 8 years ago
  66. 7d23ddd Delete unneeded includes of base/common.h. by nisse · 8 years ago
  67. 3390d21 Delete unused class CompositeMediaEngineWithFakeVoiceEngine. by nisse · 8 years ago
  68. 5e90008 Delete unused classes DesktopId and ScreencastEventCatcher. by nisse · 8 years ago
  69. e43c0e3 RTCMediaStreamTrackStats.framesDropped collected by RTCStatsCollector. by hbos · 8 years ago
  70. d481738 RTCMediaStreamTrackStats.framesReceived collected by RTCStatsCollector. by hbos · 8 years ago
  71. 7367842 Revert of Fix flaky WebRtcVideoChannel2BaseTest.GetStats T (patchset #1 id:1 of https://codereview.webrtc.org/2634273002/ ) by perkj · 8 years ago
  72. 66d1941 Disable automatic scaling in tests. by nisse · 8 years ago
  73. e5e0de5 Fix flaky WebRtcVideoChannel2BaseTest.GetStats T by perkj · 8 years ago
  74. ee5a316 Added a new echo likelihood stat that reports the maximum value from a previous time period. by ivoc · 8 years ago
  75. 174ae8e Remove WebRtcVideoSendStream2::VideoSink inheritance. Remove sending black frame on source removal. by perkj · 8 years ago
  76. 0a3331c Delete unused rtpdump code in media/base. by nisse · 8 years ago
  77. a875ff8 Replace ASSERT by RTC_DCHECK in all non-test code. by nisse · 8 years ago
  78. a496afa Reland of: Adding error output param to SetConfiguration, using new RTCError type. by deadbeef · 8 years ago
  79. f0daa06 Move VideoFrame and related declarations to webrtc/api/video. by nisse · 8 years ago
  80. 9a71660 Reland of: Separating SCTP code from BaseChannel/MediaChannel. by deadbeef · 8 years ago
  81. b479ec0 Reland of Delete unused code from systeminfo. (patchset #1 id:1 of https://codereview.webrtc.org/2584563004/ ) by kthelgason · 8 years ago
  82. e57f3ae Revert of Separating SCTP code from BaseChannel/MediaChannel. (patchset #14 id:240001 of https://codereview.webrtc.org/2564333002/ ) by deadbeef · 8 years ago
  83. 7163e1b Separating SCTP code from BaseChannel/MediaChannel. by deadbeef · 8 years ago
  84. 67d12ed Reland of place basictypes.h with stdint.h for int_t types. (patchset #1 id:1 of https://codereview.webrtc.org/2603203003/ ) by pbos · 8 years ago
  85. f86f083 Revert of Replace basictypes.h with stdint.h for int_t types. (patchset #1 id:1 of https://codereview.webrtc.org/2604043002/ ) by pbos · 8 years ago
  86. 693be15 Replace basictypes.h with stdint.h for int_t types. by pbos · 8 years ago
  87. 6c8666b Revert of Adding error output param to SetConfiguration, using new RTCError type. (patchset #4 id:60001 of https://codereview.webrtc.org/2587133004/ ) by deadbeef · 8 years ago
  88. ec77eb3 Adding error output param to SetConfiguration, using new RTCError type. by deadbeef · 8 years ago
  89. 8495a81 Revert of Delete unused code from systeminfo. (patchset #3 id:40001 of https://codereview.webrtc.org/2578323005/ ) by skvlad · 8 years ago
  90. 767c6f5 Delete unused code from systeminfo. by kthelgason · 8 years ago
  91. 0e41e0d Reland of Delete VideoFrame default constructor, and IsZeroSize method. (patchset #1 id:1 of https://codereview.webrtc.org/2574123002/ ) by nisse · 8 years ago
  92. d8cd9cc Revert of Delete VideoFrame default constructor, and IsZeroSize method. (patchset #5 id:80001 of https://codereview.webrtc.org/2541863002/ ) by nisse · 8 years ago
  93. 800a8f1 Support external audio mixer in webrtc 2. by gyzhou · 8 years ago
  94. 6dab241 Delete VideoFrame default constructor, and IsZeroSize method. by nisse · 8 years ago
  95. 818a3ee Revert of Support external audio mixer. (patchset #5 id:140001 of https://codereview.webrtc.org/2539213003/ ) by gyzhou · 8 years ago
  96. 0808545 Support external audio mixer in webrtc. by gyzhou · 8 years ago
  97. 469292f Reland of Add ability to scale to arbitrary factors (patchset #1 id:1 of https://codereview.webrtc.org/2557323002/ ) by kthelgason · 8 years ago
  98. 77e631c Revert of Add ability to scale to arbitrary factors (patchset #7 id:120001 of https://codereview.webrtc.org/2555483005/ ) by kthelgason · 8 years ago
  99. 921fffc Add ability to scale to arbitrary factors by kthelgason · 8 years ago
  100. cc206ac Refactoring: Declare cricket::Codec constructors protected. by hta · 8 years ago