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gerrit.openfyde.cn / webrtc.googlesource.com / src / 603171647318927390e10c04d918d4ed7075717e / media
  1. e43b531 Nuke p2p/base/stun.h by Patrik Höglund · 6 years ago
  2. 108a2f0 Preventively fix missing braces warnings. by Mirko Bonadei · 6 years ago
  3. 287e464 Change VideoAdapter::OnResolutionFramerateRequest to VideoAdapter::OnSinkWants by Rasmus Brandt · 6 years ago
  4. 56d9452 Move stun.h to api/. by Patrik Höglund · 6 years ago
  5. a7a2ab4 Remove dead kDummyVideoSsrc and FPS_TO_INTERVAL from video_common.h. by Rasmus Brandt · 6 years ago
  6. cb459ca Remove double declaration of cricket::kH264CodecName. by Mirko Bonadei · 6 years ago
  7. 2b9317a Stop checking VP8BaseHeavyTl3RateAllocation field trial on every frame. by Rasmus Brandt · 6 years ago
  8. 9560d7d Make update_rect optional in VideoFrame by Ilya Nikolaevskiy · 6 years ago
  9. 6e4e688 Fixed MSAN issue with usrsctp reliability test. by Yura Yaroshevich · 6 years ago
  10. e114fb6 Added usrsctp reliablitiy stress test. by Yura Yaroshevich · 6 years ago
  11. 16cec3b Added allow_codec_switching parameter to RTCConfig. by philipel · 6 years ago
  12. fcf79cc Add estimatedPlayoutTimestamp to RTCInboundRTPStreamStats. by Åsa Persson · 6 years ago
  13. 0855e2d Delete unused members of MediaReceiverInfo and MediaSenderInfo by Niels Möller · 6 years ago
  14. 03fbace Remove apm_helpers, consolidate audio config in WebRtcVoiceEngine by Sam Zackrisson · 6 years ago
  15. 3f7e0ed Add option to make first scale factor depend on input resolution. by Åsa Persson · 6 years ago
  16. 86d053c Use source_sets in component builds and static_library in release builds. by Mirko Bonadei · 6 years ago
  17. 9429888 Delete deprecated bytes_sent/bytes_rcvd stat values by Niels Möller · 6 years ago
  18. 0bad15f Remove the noise_suppression() pointer to submodule interface by saza · 6 years ago
  19. 8038541 Update the header extensions capabilities with mid, rid and rrid by Florent Castelli · 6 years ago
  20. ac0a4cb Reland "Fix GetStats bytesSent/Received, wireup headerBytesSent/Received" by Niels Möller · 6 years ago
  21. 41478c7 Remove AudioProcessing::gain_control() getter by Sam Zackrisson · 6 years ago
  22. 35214fc Add missing RTC_EXPORT for the component build. by Mirko Bonadei · 6 years ago
  23. ef0627f Revert "Fix GetStats bytesSent/Received, wireup headerBytesSent/Received" by Mirko Bonadei · 6 years ago
  24. fbde32e Fix GetStats bytesSent/Received, wireup headerBytesSent/Received by Niels Möller · 6 years ago
  25. 80f53b7 Extend WebRTC-Video-MinVideoBitrate to experiment per-codec by Elad Alon · 6 years ago
  26. 5740f3e Clarify expectation on GlobalLock by Danil Chapovalov · 6 years ago
  27. ff27da5 Add/remove receive streams with SSRC 0 from media channels by Saurav Das · 6 years ago
  28. f4e0c29 SimulcastEncoderAdapter: support per layer fallback and single encoder proxying by Erik Språng · 6 years ago
  29. 9d7eb28 Don't limit simulcast layers number for screenshare based on resolution by Ilya Nikolaevskiy · 6 years ago
  30. 09f1195 Always pass arguments to INSTANTIATE_TEST_SUITE_P. by Mirko Bonadei · 6 years ago
  31. 27b0e0d Remove obsolete todo comment in simulcast.h by Åsa Persson · 6 years ago
  32. e942b14 New build target api:media_interface by Niels Möller · 6 years ago
  33. 1b83a9e Only handle each RTCP once. by Sebastian Jansson · 6 years ago
  34. 53227cc Remove webrtc::MinPositive from api/. by Mirko Bonadei · 6 years ago
  35. 738bfa7 Remove api/bitrate_constraints.h. by Mirko Bonadei · 6 years ago
  36. 317a1f0 Use std::make_unique instead of absl::make_unique. by Mirko Bonadei · 6 years ago
  37. d9cc8c0 Encoder switching based on network and/or resolution conditions. by philipel · 6 years ago
  38. 73ceed5 Update simulcast bitrate calculations for non-standard resolutions. by Ilya Nikolaevskiy · 6 years ago
  39. 7bf7a42 Delete flag VideoReceiveStream::Config::Rtp::remb by Niels Möller · 6 years ago
  40. eaaaf41 Introduce api/crypto/BUILD.gn. by Mirko Bonadei · 6 years ago
  41. 70dd165 Delete CoreAudio include from media_engine.h by Niels Möller · 6 years ago
  42. 65f17ca Move MediaTransportInterface out of the libjingle_peerconnection_api target by Niels Möller · 6 years ago
  43. fcfeefe Move rtc_error.{h,cc} to its own build target. by Mirko Bonadei · 6 years ago
  44. cc62b16 Add qualityLimitationResolutionChanges stat by Evan Shrubsole · 6 years ago
  45. 0bd2eff Reland "New build target p2p:stun_types" by Niels Möller · 6 years ago
  46. 91c824f Revert "New build target p2p:stun_types" by Hannes Landeholm · 6 years ago
  47. 66d6c3b Buffers non atomic message send with usrsctp lib. by Seth Hampson · 6 years ago
  48. 8c5520c Reland "Make the min video bitrate in VideoSendStream configurable." by Ying Wang · 6 years ago
  49. 1d2149c Revert "Make the min video bitrate in VideoSendStream configurable." by Alessio Bazzica · 6 years ago
  50. b2fb0b9 Make the min video bitrate in VideoSendStream configurable. by Ying Wang · 6 years ago
  51. a837030 Split out RtpSource from libjingle_peerconnection_api by Niels Möller · 6 years ago
  52. 5b4fcb5 New build target p2p:stun_types by Niels Möller · 6 years ago
  53. 25eb47c Make the RtpHeaderParserImpl available to tests and tools only. by Tommi · 6 years ago
  54. b4a6128 Delete unneeded dependencies on libjingle_peerconnection_api by Niels Möller · 6 years ago
  55. 6dcd4dc New target for api/rtp_parameters.h and api/media_types.h. by Niels Möller · 6 years ago
  56. 4271afb Fix the bug and reland "Make min video target bitrate configurable." by Ying Wang · 6 years ago
  57. 0c141c5 Fix frames dropped statistics by Johannes Kron · 6 years ago
  58. 7e896d0 Revert "Make min video target bitrate configurable." by Mirko Bonadei · 6 years ago
  59. a471e79 Make min video target bitrate configurable. by Ying Wang · 6 years ago
  60. d77cc24 New const method StreamStatistician::GetStats by Niels Möller · 6 years ago
  61. 224c69d Delete ext_seqnum member from VoiceSenderInfo and VoiceReceiverInfo by Niels Möller · 6 years ago
  62. b689af4 Changes to enable use of DatagramTransport as a data channel transport. by Bjorn A Mellem · 6 years ago
  63. 70efdde Set local ssrc at construction of Rtp module by Erik Språng · 6 years ago
  64. 587991c Remove jeroendb@webrtc.org from OWNERS by Steve Anton · 6 years ago
  65. 6b43086 Reland "[GetStats] Expose video codec implementation in standardized metrics." by Henrik Boström · 6 years ago
  66. df625f4 Revert "[GetStats] Expose video codec implementation in standardized metrics." by Henrik Andreassson · 6 years ago
  67. 2b9fa09 [GetStats] Expose video codec implementation in standardized metrics. by Henrik Boström · 6 years ago
  68. bbeb109 Reporting audio device underrun counter by Alex Narest · 6 years ago
  69. 71c6b56 Allow sending abs-send-time for audio streams. by Sebastian Jansson · 6 years ago
  70. 78a7138 Remove MediaTransport from Call. by Tommi · 6 years ago
  71. 5b5d97c Reland of "Reporting of decoding_codec_plc events"" by Alex Narest · 6 years ago
  72. 2d2bbb1 Filter out duplicate receive codecs in the media engine by Steve Anton · 6 years ago
  73. 83bbe91 Delete deprecated rtc_event_log header by Danil Chapovalov · 6 years ago
  74. f40a340 Remove deprecated code related to AEC2 by Per Åhgren · 6 years ago
  75. d2845f8 Removes unused AudioAllocationSettings from voice engine. by Sebastian Jansson · 6 years ago
  76. 9b1700c Enable field trial LegacySimulcastLayerLimit by default by Florent Castelli · 6 years ago
  77. d7ee76c Wire up field trials for some experimental screenshare settings by Erik Språng · 6 years ago
  78. 8bbdb5b Update VideoBitrateAllocator allocate to take a struct with more fields by Florent Castelli · 6 years ago
  79. da4f093 Reland "Only include payload in bytes sent/received." by Bjorn A Mellem · 6 years ago
  80. bedb7a8 Revert "Reporting of decoding_codec_plc events" by Mirko Bonadei · 6 years ago
  81. bcd068d Revert "Only include payload in bytes sent/received." by Bjorn Mellem · 6 years ago
  82. 0a88ea0 Reporting of decoding_codec_plc events by Alex Narest · 6 years ago
  83. a9fbb22 Add a field trial for older applications to reduce the simulcast layer count by Florent Castelli · 6 years ago
  84. e1795f4 Adds remote estimate RTCP packet. by Sebastian Jansson · 6 years ago
  85. 74a1b4b Only include payload in bytes sent/received. by Bjorn A Mellem · 6 years ago
  86. 0182a03 Reland "Remove the injectable bitrate allocation strategy API." by Jonas Olsson · 6 years ago
  87. e95b57c Revert "Remove the injectable bitrate allocation strategy API." by Mirko Bonadei · 6 years ago
  88. 0bb0881 Add VideoEncoderFactory::GetImplementations function. by philipel · 6 years ago
  89. 66b3860 Remove WebRTC-SimulcastScreenshare and enable it by default by Florent Castelli · 6 years ago
  90. 41300af Poison default task queue factory by Danil Chapovalov · 6 years ago
  91. 80cb3f6 Remove the injectable bitrate allocation strategy API. by Jonas Olsson · 6 years ago
  92. 495a1ae Remove cricket::WebRtcMediaEngineFactory as now unused by Danil Chapovalov · 6 years ago
  93. a4d8737 Format almost everything. by Jonas Olsson · 6 years ago
  94. 668ce0c Remove trial WebRTC-SimulcastMaxLayers and make its behavior default by Florent Castelli · 6 years ago
  95. fdf74bd Remove non implemented function from WebRtcVideoChannel. by philipel · 6 years ago
  96. d2c336f [getStats] Implement "media-source" audio levels, fixing Chrome bug. by Henrik Boström · 6 years ago
  97. 53d45ba Make TaskQueueFactory required construction parameter for Call by Danil Chapovalov · 6 years ago
  98. e8ed830 WebRtcVideoChannel encoder fallback. by philipel · 6 years ago
  99. 5ee6967 Don't reset encoder on max/min bitrate change. by Sergey Silkin · 6 years ago
  100. bfd343b Add totalDecodeTime to RTCInboundRTPStreamStats by Johannes Kron · 6 years ago