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gerrit-public.fairphone.software / platform / external / webrtc / 9bac68c0cc4444b852416396f0e0f31ea66a9cfe / media
  1. 9bac68c Reland "Reland "Distinguish between send and receive codecs"" by Johannes Kron · 4 years, 7 months ago
  2. 760fd52 Replace MockAudioDeviceModule mock refcounting with real refcounting by Steve Anton · 4 years, 7 months ago
  3. 00a3087 Revert "Reland "Distinguish between send and receive codecs"" by Johannes Kron · 4 years, 7 months ago
  4. 897776e Pass SDP video parameters to encoder. by Sergey Silkin · 4 years, 7 months ago
  5. 133bf2b Reland "Distinguish between send and receive codecs" by Johannes Kron · 4 years, 7 months ago
  6. 1acdc74 Split up EncoderStreamFactory::CreateEncoderStreams in two. by Rasmus Brandt · 4 years, 7 months ago
  7. 43bfe0b Enforce VideoEncoderConfig.num_temporal_layers >= 1. by Rasmus Brandt · 4 years, 7 months ago
  8. ccbe95f Reformat GN files. by Mirko Bonadei · 4 years, 7 months ago
  9. e57b266 Revert "Distinguish between send and receive codecs" by Steve Anton · 4 years, 7 months ago
  10. c0f25cf Distinguish between send and receive codecs by Johannes Kron · 4 years, 7 months ago
  11. b2b2031 Concatenate string literals at compile time. by Jonas Olsson · 4 years, 7 months ago
  12. 1546f99 Fixed timeout overflow in sctp reliability test. by Yura Yaroshevich · 4 years, 7 months ago
  13. fae6400 Add saza@ and peah@ to OWNERS of some audio files by Sam Zackrisson · 4 years, 7 months ago
  14. f5ecb5f Revert "Reland "Reland "Reland "Distinguish between send and receive video codecs"""" by Mirko Bonadei · 4 years, 7 months ago
  15. 9cad4dc Reland "Reland "Reland "Distinguish between send and receive video codecs""" by Johannes Kron · 4 years, 8 months ago
  16. 0e3a3f6 Adding deadbeef to sctp/OWNERS and removing myself. by Seth Hampson · 4 years, 8 months ago
  17. 27064ad SimulcastEncoderAdapter: In passthrough mode set correct lenght for frame_types parameter by Ilya Nikolaevskiy · 4 years, 8 months ago
  18. 4db28b5 Cleanup: Removes redundant includes on message_queue.h by Sebastian Jansson · 4 years, 8 months ago
  19. 873610c Fix updating degradation preference in SetRtpParameters. by Mirta Dvornicic · 4 years, 8 months ago
  20. b5159fe Revert "Reland "Reland "Distinguish between send and receive video codecs""" by Olga Sharonova · 4 years, 8 months ago
  21. 4e64e60 Reland "Reland "Distinguish between send and receive video codecs"" by Johannes Kron · 4 years, 8 months ago
  22. c8f3134 Parse max-fr and max-fs from SDP FMTP line by Johannes Kron · 4 years, 8 months ago
  23. 5cad55b Signal requested resolution alignment requirements from sinks to sources. by Rasmus Brandt · 4 years, 8 months ago
  24. 6fd58b3 Add maxFramerate support to SimulcastEncoderAdapter by Florent Castelli · 4 years, 8 months ago
  25. f9d92ed Revert "Reland "Distinguish between send and receive video codecs"" by Ilya Nikolaevskiy · 4 years, 8 months ago
  26. 2697ac1 Stop an SCTP connection when the DTLS transport closes. by Harald Alvestrand · 4 years, 8 months ago
  27. 77eb338 Reland "Distinguish between send and receive video codecs" by Johannes Kron · 4 years, 8 months ago
  28. f2d6fe6 Revert "Reland "Distinguish between send and receive video codecs"" by Johannes Kron · 4 years, 8 months ago
  29. 26e6afe Reland "Distinguish between send and receive video codecs" by Johannes Kron · 4 years, 9 months ago
  30. 977b265 Reduce some logging at INFO level by moving log statements by Harald Alvestrand · 4 years, 8 months ago
  31. dcb4fcc Execute cached video encoder switching request if encoder switching is allowed after the switch request was made. by philipel · 4 years, 9 months ago
  32. ded86c1 Remove remaining settings for using legacy AEC by Per Åhgren · 4 years, 9 months ago
  33. f22af3c Revert "Distinguish between send and receive video codecs" by Johannes Kron · 4 years, 9 months ago
  34. 18314bd Distinguish between send and receive video codecs by Johannes Kron · 4 years, 9 months ago
  35. 62ea0aa Remove deprecated legacy AEC code by Per Åhgren · 4 years, 9 months ago
  36. 33f9d2b Migrate WebRTC on FrameGeneratorInterface and remove FrameGenerator class by Artem Titov · 4 years, 9 months ago
  37. 907dc80 Reland "Add support for RtpEncodingParameters::max_framerate" by Florent Castelli · 4 years, 9 months ago
  38. 749f660 Enable SSRC 0 in MediaChannel methods by Saurav Das · 4 years, 9 months ago
  39. 32565f6 WebRtcVideoEngine: Enable encoded frame sink. by Markus Handell · 4 years, 9 months ago
  40. 41462d5 Always keep abs send time extension. by Sebastian Jansson · 4 years, 9 months ago
  41. 934afc6 Deprecate RtpReceiver's SetParameters method by Saurav Das · 4 years, 9 months ago
  42. 831ce5f Export more symbols to fix Chromecast component build by Ken MacKay · 4 years, 9 months ago
  43. 269ac81 VideoReceiveStream: Enable encoded frame sink. by Markus Handell · 4 years, 9 months ago
  44. 014dd3c Trials should always be populated in call config. by Erik Språng · 4 years, 9 months ago
  45. 7968530 Removes caching SimulcastEncoderAdapter::GetEncoderInfo() by Erik Språng · 4 years, 9 months ago
  46. 5cef9c3 Revert "Add support for RtpEncodingParameters::max_framerate" by Florent Castelli · 4 years, 9 months ago
  47. 7a9a092 Delete media transport integration. by Bjorn A Mellem · 4 years, 9 months ago
  48. 15be528 Add support for RtpEncodingParameters::max_framerate by Florent Castelli · 4 years, 9 months ago
  49. 00376e1 Add totalInterFrameDelay to RTCInboundRTPStreamStats by Johannes Kron · 4 years, 9 months ago
  50. e43b531 Nuke p2p/base/stun.h by Patrik Höglund · 4 years, 9 months ago
  51. 108a2f0 Preventively fix missing braces warnings. by Mirko Bonadei · 4 years, 9 months ago
  52. 287e464 Change VideoAdapter::OnResolutionFramerateRequest to VideoAdapter::OnSinkWants by Rasmus Brandt · 4 years, 9 months ago
  53. 56d9452 Move stun.h to api/. by Patrik Höglund · 4 years, 9 months ago
  54. a7a2ab4 Remove dead kDummyVideoSsrc and FPS_TO_INTERVAL from video_common.h. by Rasmus Brandt · 4 years, 10 months ago
  55. cb459ca Remove double declaration of cricket::kH264CodecName. by Mirko Bonadei · 4 years, 10 months ago
  56. 2b9317a Stop checking VP8BaseHeavyTl3RateAllocation field trial on every frame. by Rasmus Brandt · 4 years, 10 months ago
  57. 9560d7d Make update_rect optional in VideoFrame by Ilya Nikolaevskiy · 4 years, 10 months ago
  58. 6e4e688 Fixed MSAN issue with usrsctp reliability test. by Yura Yaroshevich · 4 years, 10 months ago
  59. e114fb6 Added usrsctp reliablitiy stress test. by Yura Yaroshevich · 4 years, 10 months ago
  60. 16cec3b Added allow_codec_switching parameter to RTCConfig. by philipel · 4 years, 10 months ago
  61. fcf79cc Add estimatedPlayoutTimestamp to RTCInboundRTPStreamStats. by Åsa Persson · 4 years, 10 months ago
  62. 0855e2d Delete unused members of MediaReceiverInfo and MediaSenderInfo by Niels Möller · 4 years, 10 months ago
  63. 03fbace Remove apm_helpers, consolidate audio config in WebRtcVoiceEngine by Sam Zackrisson · 4 years, 10 months ago
  64. 3f7e0ed Add option to make first scale factor depend on input resolution. by Åsa Persson · 4 years, 10 months ago
  65. 86d053c Use source_sets in component builds and static_library in release builds. by Mirko Bonadei · 4 years, 10 months ago
  66. 9429888 Delete deprecated bytes_sent/bytes_rcvd stat values by Niels Möller · 4 years, 10 months ago
  67. 0bad15f Remove the noise_suppression() pointer to submodule interface by saza · 4 years, 10 months ago
  68. 8038541 Update the header extensions capabilities with mid, rid and rrid by Florent Castelli · 4 years, 10 months ago
  69. ac0a4cb Reland "Fix GetStats bytesSent/Received, wireup headerBytesSent/Received" by Niels Möller · 4 years, 11 months ago
  70. 41478c7 Remove AudioProcessing::gain_control() getter by Sam Zackrisson · 4 years, 10 months ago
  71. 35214fc Add missing RTC_EXPORT for the component build. by Mirko Bonadei · 5 years ago
  72. ef0627f Revert "Fix GetStats bytesSent/Received, wireup headerBytesSent/Received" by Mirko Bonadei · 4 years, 10 months ago
  73. fbde32e Fix GetStats bytesSent/Received, wireup headerBytesSent/Received by Niels Möller · 4 years, 11 months ago
  74. 80f53b7 Extend WebRTC-Video-MinVideoBitrate to experiment per-codec by Elad Alon · 4 years, 11 months ago
  75. 5740f3e Clarify expectation on GlobalLock by Danil Chapovalov · 4 years, 11 months ago
  76. ff27da5 Add/remove receive streams with SSRC 0 from media channels by Saurav Das · 5 years ago
  77. f4e0c29 SimulcastEncoderAdapter: support per layer fallback and single encoder proxying by Erik Språng · 5 years ago
  78. 9d7eb28 Don't limit simulcast layers number for screenshare based on resolution by Ilya Nikolaevskiy · 5 years ago
  79. 09f1195 Always pass arguments to INSTANTIATE_TEST_SUITE_P. by Mirko Bonadei · 5 years ago
  80. 27b0e0d Remove obsolete todo comment in simulcast.h by Åsa Persson · 5 years ago
  81. e942b14 New build target api:media_interface by Niels Möller · 5 years ago
  82. 1b83a9e Only handle each RTCP once. by Sebastian Jansson · 5 years ago
  83. 53227cc Remove webrtc::MinPositive from api/. by Mirko Bonadei · 5 years ago
  84. 738bfa7 Remove api/bitrate_constraints.h. by Mirko Bonadei · 5 years ago
  85. 317a1f0 Use std::make_unique instead of absl::make_unique. by Mirko Bonadei · 5 years ago
  86. d9cc8c0 Encoder switching based on network and/or resolution conditions. by philipel · 5 years ago
  87. 73ceed5 Update simulcast bitrate calculations for non-standard resolutions. by Ilya Nikolaevskiy · 5 years ago
  88. 7bf7a42 Delete flag VideoReceiveStream::Config::Rtp::remb by Niels Möller · 5 years ago
  89. eaaaf41 Introduce api/crypto/BUILD.gn. by Mirko Bonadei · 5 years ago
  90. 70dd165 Delete CoreAudio include from media_engine.h by Niels Möller · 5 years ago
  91. 65f17ca Move MediaTransportInterface out of the libjingle_peerconnection_api target by Niels Möller · 5 years ago
  92. fcfeefe Move rtc_error.{h,cc} to its own build target. by Mirko Bonadei · 5 years ago
  93. cc62b16 Add qualityLimitationResolutionChanges stat by Evan Shrubsole · 5 years ago
  94. 0bd2eff Reland "New build target p2p:stun_types" by Niels Möller · 5 years ago
  95. 91c824f Revert "New build target p2p:stun_types" by Hannes Landeholm · 5 years ago
  96. 66d6c3b Buffers non atomic message send with usrsctp lib. by Seth Hampson · 5 years ago
  97. 8c5520c Reland "Make the min video bitrate in VideoSendStream configurable." by Ying Wang · 5 years ago
  98. 1d2149c Revert "Make the min video bitrate in VideoSendStream configurable." by Alessio Bazzica · 5 years ago
  99. b2fb0b9 Make the min video bitrate in VideoSendStream configurable. by Ying Wang · 5 years ago
  100. a837030 Split out RtpSource from libjingle_peerconnection_api by Niels Möller · 5 years ago