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webrtc / src.git / 50d79babcd37e0de20d9333f6dee9c789dc841e7 / media / BUILD.gn
  1. e904161 Replace RTC_DEPRECATED with ABSL_DEPRECATED by Danil Chapovalov · 4 years, 1 month ago
  2. e71b55f build: merge media_constants and engine_constants by Philipp Hancke · 4 years, 1 month ago
  3. d15a575 Use SequenceChecker from public API by Artem Titov · 4 years, 2 months ago
  4. 0e3cb9f Create and initialize encoders only for active streams by Sergey Silkin · 4 years, 2 months ago
  5. e7c79fd Remove from chromium build targets that are not compatible with it. by Andrey Logvin · 4 years, 2 months ago
  6. 7864600 Add absl_deps field for rtc_test and rtc_executable by Andrey Logvin · 4 years, 2 months ago
  7. a7e34d3 Add resolution_bitrate_limits to EncoderInfo field trial. by Åsa Persson · 4 years, 2 months ago
  8. 33c0ab4 Call MediaChannel::OnPacketReceived on the network thread. by Tomas Gunnarsson · 4 years, 2 months ago
  9. 2397b6e SimulcastEncoderAdapter: Add field trial for EncoderInfo settings. by Åsa Persson · 4 years, 2 months ago
  10. e5f4c6b Reland "Refactor rtc_base build targets." by Mirko Bonadei · 4 years, 2 months ago
  11. 5ab6a8c Refactors SimulcastEncoder Adapter. by Erik Språng · 4 years, 2 months ago
  12. 7acc2d9 Revert "Refactor rtc_base build targets." by Mirko Bonadei · 4 years, 2 months ago
  13. 69241a9 Refactor rtc_base build targets. by Mirko Bonadei · 4 years, 3 months ago
  14. 08d2c2b Delete unneeded dependencies on the Module abstraction by Niels Möller · 4 years, 3 months ago
  15. 20e4c80 Reland "Introduce RTC_NO_UNIQUE_ADDRESS." by Mirko Bonadei · 4 years, 4 months ago
  16. bff717e Remove dependency on AsyncInvoker in SctpTransport by Tomas Gunnarsson · 4 years, 4 months ago
  17. 38e9b06 Reland "Add scaling interface to VideoFrameBuffer" by Ilya Nikolaevskiy · 4 years, 6 months ago
  18. 441dbf9 Revert "Add scaling interface to VideoFrameBuffer" by Ilya Nikolaevskiy · 4 years, 6 months ago
  19. c79f1d8 Add scaling interface to VideoFrameBuffer by Ilya Nikolaevskiy · 4 years, 6 months ago
  20. 0abd518 Revert "Introduce RTC_NO_UNIQUE_ADDRESS." by Mirko Bonadei · 4 years, 6 months ago
  21. 09ceed2 Async audio processing API by Olga Sharonova · 4 years, 6 months ago
  22. f5e261a Introduce RTC_NO_UNIQUE_ADDRESS. by Mirko Bonadei · 4 years, 6 months ago
  23. ceb4495 Reland: Wires up WebrtcKeyValueBasedConfig in media engines. by Erik Språng · 4 years, 6 months ago
  24. 5956a17 Revert "Wires up WebrtcKeyValueBasedConfig in media engines." by Artem Titov · 4 years, 6 months ago
  25. 591b2ab Wires up WebrtcKeyValueBasedConfig in media engines. by Erik Språng · 4 years, 6 months ago
  26. 2bca008 Reland "Make cricket::SctpTransportInternalFactory injectable through PeerConnectionFactory Deps" by Per Kjellander · 4 years, 7 months ago
  27. 1f580a9 Revert "Make cricket::SctpTransportInternalFactory injectable through PeerConnectionFactory Deps" by Björn Terelius · 4 years, 7 months ago
  28. 4c0a381 Make cricket::SctpTransportInternalFactory injectable through PeerConnectionFactory Deps by Per Kjellander · 4 years, 7 months ago
  29. 6761546 Move H264::Profile to h264_profile_level_id.h by Niels Möller · 4 years, 8 months ago
  30. 1e257ca Migrate media/ to webrtc::Mutex. by Markus Handell · 4 years, 9 months ago
  31. c212873 Relanding: Fix data channel message integrity violation by Taylor Brandstetter · 4 years, 9 months ago
  32. 51e08b8 Revert "Fix data channel message integrity violation" by Alessio Bazzica · 4 years, 9 months ago
  33. 6cbff75 Fix data channel message integrity violation by Lennart Grahl · 4 years, 9 months ago
  34. 39adce1 Add RtpEncodingParameters.adaptive_ptime. by Jakob Ivarsson · 4 years, 9 months ago
  35. 2a70703 Delete MediaTransportInterface and DatagramTransportInterface by Niels Möller · 4 years, 9 months ago
  36. 2dcf348 Use absl_deps in order to preapre to the Abseil component build release. by Mirko Bonadei · 4 years, 10 months ago
  37. 6415dca Remove WebRTC-ExperimentalScreenshareSettings. by Mirko Bonadei · 5 years ago
  38. 4553f45 Add AV1 to default video encoder factory by Danil Chapovalov · 5 years ago
  39. 3e98368 Reland "Distinguish between send and receive codecs" by Johannes Kron · 5 years ago
  40. 0357b3e RtpTransceiverInterface: add header_extensions_to_offer() by Markus Handell · 5 years ago
  41. c46385c Add Av1 Decoder wrapper behind a build flag by Danil Chapovalov · 5 years ago
  42. b05ca4b Implement new specification for degradation preference by Florent Castelli · 5 years ago
  43. e77912b Insert frame transformer between Encoded and Packetizer. by Marina Ciocea · 5 years ago
  44. ccd49d9 Use I420Buffer::ScaleFrom to clean scaling in SimulcastEncoderAdapter by Evan Shrubsole · 5 years ago
  45. ccbe95f Reformat GN files. by Mirko Bonadei · 5 years ago
  46. c8f3134 Parse max-fr and max-fs from SDP FMTP line by Johannes Kron · 5 years ago
  47. 5cad55b Signal requested resolution alignment requirements from sinks to sources. by Rasmus Brandt · 5 years ago
  48. 32565f6 WebRtcVideoEngine: Enable encoded frame sink. by Markus Handell · 5 years ago
  49. 831ce5f Export more symbols to fix Chromecast component build by Ken MacKay · 5 years ago
  50. 014dd3c Trials should always be populated in call config. by Erik Språng · 5 years ago
  51. 7a9a092 Delete media transport integration. by Bjorn A Mellem · 5 years ago
  52. e43b531 Nuke p2p/base/stun.h by Patrik Höglund · 5 years ago
  53. 56d9452 Move stun.h to api/. by Patrik Höglund · 5 years ago
  54. 2b9317a Stop checking VP8BaseHeavyTl3RateAllocation field trial on every frame. by Rasmus Brandt · 5 years ago
  55. e114fb6 Added usrsctp reliablitiy stress test. by Yura Yaroshevich · 5 years ago
  56. 03fbace Remove apm_helpers, consolidate audio config in WebRtcVoiceEngine by Sam Zackrisson · 5 years ago
  57. 3f7e0ed Add option to make first scale factor depend on input resolution. by Åsa Persson · 5 years ago
  58. 86d053c Use source_sets in component builds and static_library in release builds. by Mirko Bonadei · 5 years ago
  59. 41478c7 Remove AudioProcessing::gain_control() getter by Sam Zackrisson · 5 years ago
  60. 80f53b7 Extend WebRTC-Video-MinVideoBitrate to experiment per-codec by Elad Alon · 5 years ago
  61. 5740f3e Clarify expectation on GlobalLock by Danil Chapovalov · 5 years ago
  62. f4e0c29 SimulcastEncoderAdapter: support per layer fallback and single encoder proxying by Erik Språng · 5 years ago
  63. e942b14 New build target api:media_interface by Niels Möller · 6 years ago
  64. 738bfa7 Remove api/bitrate_constraints.h. by Mirko Bonadei · 6 years ago
  65. 317a1f0 Use std::make_unique instead of absl::make_unique. by Mirko Bonadei · 6 years ago
  66. eaaaf41 Introduce api/crypto/BUILD.gn. by Mirko Bonadei · 6 years ago
  67. 65f17ca Move MediaTransportInterface out of the libjingle_peerconnection_api target by Niels Möller · 6 years ago
  68. fcfeefe Move rtc_error.{h,cc} to its own build target. by Mirko Bonadei · 6 years ago
  69. 0bd2eff Reland "New build target p2p:stun_types" by Niels Möller · 6 years ago
  70. 91c824f Revert "New build target p2p:stun_types" by Hannes Landeholm · 6 years ago
  71. 66d6c3b Buffers non atomic message send with usrsctp lib. by Seth Hampson · 6 years ago
  72. a837030 Split out RtpSource from libjingle_peerconnection_api by Niels Möller · 6 years ago
  73. 5b4fcb5 New build target p2p:stun_types by Niels Möller · 6 years ago
  74. 25eb47c Make the RtpHeaderParserImpl available to tests and tools only. by Tommi · 6 years ago
  75. b4a6128 Delete unneeded dependencies on libjingle_peerconnection_api by Niels Möller · 6 years ago
  76. 6dcd4dc New target for api/rtp_parameters.h and api/media_types.h. by Niels Möller · 6 years ago
  77. 83bbe91 Delete deprecated rtc_event_log header by Danil Chapovalov · 6 years ago
  78. d2845f8 Removes unused AudioAllocationSettings from voice engine. by Sebastian Jansson · 6 years ago
  79. d7ee76c Wire up field trials for some experimental screenshare settings by Erik Språng · 6 years ago
  80. 41300af Poison default task queue factory by Danil Chapovalov · 6 years ago
  81. 495a1ae Remove cricket::WebRtcMediaEngineFactory as now unused by Danil Chapovalov · 6 years ago
  82. 8f01c4e Define FecControllerOverride and plumb it down to VideoEncoder by Elad Alon · 6 years ago
  83. 4ba04b7 Delete RtcEventLogFactory factory as now unused by Danil Chapovalov · 6 years ago
  84. e8e4dc4 Change StartAecDump methods to work with FILE* and FileWrapper by Niels Möller · 6 years ago
  85. 5d24b16 Prepare for splitting the api/video:video_frames build rule. by Chen Xing · 6 years ago
  86. 85b8ce2 In media engine replace forward declaration with proper includes by Danil Chapovalov · 6 years ago
  87. 220f4be Remove some media/ --> pc/ test dependencies by Steve Anton · 6 years ago
  88. 87e3f9d [video] Plumbing of ReportBlockData from RTCPReceiver to MediaSenderInfo by Henrik Boström · 6 years ago
  89. 04a3cc1 Delete rtc_base/unittest_main.cc by Niels Möller · 6 years ago
  90. 9e79e6b Consolidate mock video encoders by Erik Språng · 6 years ago
  91. a3aa9bd Make VideoBitrateAllocatorFactory injectable. by Jonas Oreland · 6 years ago
  92. 4844c5f Introduce media engine factory where TaskQueueFactory dependency can be set. by Danil Chapovalov · 6 years ago
  93. 6fe6b5a merge deps= and unconditional deps+= blocks in media/BUILD.gn by Danil Chapovalov · 6 years ago
  94. b55015e Replacing SequencedTaskChecker with SequenceChecker. by Sebastian Jansson · 6 years ago
  95. 8581877 Delete interface class VideoCaptureExternal by Niels Möller · 6 years ago
  96. 66e7679 Export symbols needed by the Chromium component build (part 8). by Mirko Bonadei · 6 years ago
  97. 9d8eaac Delete unneeded direct includes of common_types.h by Niels Möller · 6 years ago
  98. a7d8f19 Use 'public_deps' to depend on nacl_io [1]. by Mirko Bonadei · 6 years ago
  99. 4c7112a Reland "in WebrtcVoiceEngine allow to set TaskQueueFactory" by Danil Chapovalov · 6 years ago
  100. e27ccf9 Revert "in WebrtcVoiceEngine allow to set TaskQueueFactory" by Amit Hilbuch · 6 years ago