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gerrit-public.fairphone.software / platform / external / webrtc / 65438812ba0d84db07b39ed8000bf15be58b5d35 / modules / BUILD.gn
  1. 6543881 2nd reland of https://webrtc-review.googlesource.com/c/src/+/114883 by Alex Loiko · 6 years ago
  2. 8b3db59 Revert "Reland of https://webrtc-review.googlesource.com/c/src/+/114883" by Alex Loiko · 6 years ago
  3. 5341aac Reland of https://webrtc-review.googlesource.com/c/src/+/114883 by Alex Loiko · 6 years ago
  4. ffd1f93 Revert "Tests for multi-stream Opus." by Mirko Bonadei · 6 years ago
  5. 9c31ac2 Tests for multi-stream Opus. by Alex Loiko · 6 years ago
  6. 7a3e43a Reland of Opus multistream. by Alex Loiko · 6 years ago
  7. 1fa51d6 Revert "Opus multistream." by Amit Hilbuch · 6 years ago
  8. 83ed89a Opus multistream. by Alex Loiko · 6 years ago
  9. 3e70781 [Cleanup] Add missing #include. Remove useless ones. IWYU part 2. by Yves Gerey · 6 years ago
  10. 276827c Export symbols needed by the Chromium component build (part 3). by Mirko Bonadei · 6 years ago
  11. 40a7a35 Extract functionality of test_main into separate library. by Artem Titov · 6 years ago
  12. db12856 Cleanup modules_common_types by Danil Chapovalov · 6 years ago
  13. a12c42a Delete root header file typedef.h. by Niels Möller · 6 years ago
  14. 692409f Enabling clang::find_bad_constructs in modules/BUILD.gn. by Mirko Bonadei · 6 years ago
  15. 1a4746a Change RTPVideoTypeHeader to absl::variant and move RTPVideoHeader into its own h/cc file. by philipel · 6 years ago
  16. 196100e Replace rtc::Optional with absl::optional by Danil Chapovalov · 6 years ago
  17. c6ce9c5 New file api/video/BUILD.gn by Niels Möller · 7 years ago
  18. c6c4426 Moves network control interface to API. by Sebastian Jansson · 7 years ago
  19. ae8d8a1 Remove audio_frame.h from module_common_types.h by Fredrik Solenberg · 7 years ago
  20. 3285897 Cleaning up modules_tests resources. by Sergey Silkin · 7 years ago
  21. f35c666 Separate build targets for aec3 and aec3_unittests by Gustaf Ullberg · 7 years ago
  22. 0dd1b0a Revert "Revert "Enables PeerConnectionFactory using external fec controller"" by Ying Wang · 7 years ago
  23. 2ae140a BUILD.gn file for api/audio. by Gustaf Ullberg · 7 years ago
  24. 0073301 Revert "Enables PeerConnectionFactory using external fec controller" by Taylor Brandstetter · 7 years ago
  25. 4f07bdb Enables PeerConnectionFactory using external fec controller by Ying Wang · 7 years ago
  26. d377f04 Move AudioFrame to its own header file and target in api/. by Niels Möller · 7 years ago
  27. 10d9d59 Adding simulcast/spatial layering support to VideoProcessor. by Sergey Silkin · 7 years ago
  28. e48c61f Delete unused MediaFile module. by Niels Möller · 7 years ago
  29. 65ce311 Removing useless dependencies on //testing/gmock. by Mirko Bonadei · 7 years ago
  30. 34924c2 Fix warning 4373. by Patrik Höglund · 7 years ago
  31. 9c68613 Update gn files to support Mozilla build by Dan Minor · 7 years ago
  32. a7f2d84 Revert "Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*""" by Per Kjellander · 7 years ago
  33. c73e1f4 Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*"" by Per Kjellander · 7 years ago
  34. 588c548 GN rtc_* templates: Set default visibility to webrtc_root + "/*" by Karl Wiberg · 7 years ago
  35. 75f18fc Make building with X11 libraries optional. by Joachim Bauch · 7 years ago
  36. 3e11343 Fix circular dependencies in webrtc_common. by Patrik Höglund · 7 years ago
  37. 28fe510 Stop using public_deps in modules/BUILD.gn. by Mirko Bonadei · 7 years ago
  38. e9619f8 Add a new NetEq decoding unit test for Opus with DTX by Henrik Lundin · 7 years ago
  39. 558cabf Refactor RtpToNtpEstimator and MovingMedianFilter by Ilya Nikolaevskiy · 7 years ago
  40. b5b5bce Separate i420 and i444 implementations to separate targets. by Patrik Höglund · 7 years ago
  41. 2397b9a Remove voe::OutputMixer and AudioConferenceMixer. by solenberg · 7 years ago
  42. cb728ea Fix Gn Untracked headers in webrtc/modules/video_coding. by charujain · 7 years ago
  43. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  44. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/modules/BUILD.gn]
  45. df23299 Add VideoProcessorIntegrationTest for MediaCodec implementations. by brandtr · 7 years ago
  46. 84f6a3f Move optional.h to webrtc/api/ by kwiberg · 7 years ago
  47. f6a861a Remove remains of webrtc/base by ehmaldonado · 7 years ago
  48. c024740 Use relative paths in GN files. by jianjun.zhu · 7 years ago
  49. 370dd47 Revert of Remove remains of webrtc/base (patchset #7 id:120001 of https://codereview.webrtc.org/2973183002/ ) by ehmaldonado · 7 years ago
  50. 9483b49 Remove remains of webrtc/base by ehmaldonado · 7 years ago
  51. 6dcdf10 This is an initial cl, which contains small amount of implemented functions, and large amount of unimplemented ones. by gnish · 7 years ago
  52. 1140f97 Reland of Creating webrtc/modules:module_api (patchset #1 id:1 of https://codereview.webrtc.org/2839963005/ ) by mbonadei · 8 years ago
  53. bb08c3e Revert of Creating webrtc/modules:module_api (patchset #5 id:80001 of https://codereview.webrtc.org/2838873002/ ) by mbonadei · 8 years ago
  54. 5a1a092 Creating webrtc/modules:module_api by mbonadei · 8 years ago
  55. 656610f Move frame_generator_capture.{cc, h} and video_capturer.h to video_test_common. by ehmaldonado · 8 years ago
  56. 9cbb0a1 Reland of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #1 id:1 of https://codereview.webrtc.org/2651023005/ ) by ehmaldonado · 8 years ago
  57. cc7213e Remove "video_capture" from modules' public_deps. by jianjun.zhu · 8 years ago
  58. 3373eaa Revert of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #4 id:130001 of https://codereview.webrtc.org/2649563002/ ) by ehmaldonado · 8 years ago
  59. 36cb55d GN: Refactor modules_unittests to eliminate package boundary violations. by ehmaldonado · 8 years ago
  60. 9aa3f0a Reland of Moving webrtc.gni up one level from build/ (patchset #1 id:1 of https://codereview.webrtc.org/2657563002/ ) by mbonadei · 8 years ago
  61. 69dc7db Revert of Moving webrtc.gni up one level from build/ (patchset #1 id:1 of https://codereview.webrtc.org/2651543003/ ) by mbonadei · 8 years ago
  62. 35a3270 Moving webrtc.gni up one level from build/ by mbonadei · 8 years ago
  63. 9410b51 GN: Add audio_conference_mixer_unittests to modules_unittests. by ehmaldonado · 8 years ago
  64. 3626865 GN: Refactor modules_unittests to eliminate package boundary violations. by ehmaldonado · 8 years ago
  65. 4e477a1 Added a new echo likelihood stat that reports the maximum value from a previous time period. by ivoc · 8 years ago
  66. 363a291 Revert of Log audio network adapter decisions in event log. (patchset #14 id:320001 of https://codereview.webrtc.org/2559953002/ ) by sakal · 8 years ago
  67. 3663681 Log audio network adapter decisions in event log. by michaelt · 8 years ago
  68. 566d820 Update smoothed bitrate. by michaelt · 8 years ago
  69. 9774447 Move FilePlayer and FileRecorder to Voice Engine by kwiberg · 8 years ago
  70. de770dd Remove AudioClassifier by flim · 8 years ago
  71. d026354 Reland of Added first layer of the echo canceller 3 functionality. by peah · 8 years ago
  72. 7946b54 Revert of Added first layer of the echo canceller 3 functionality (patchset #13 id:240001 of https://codereview.webrtc.org/2584493002/ ) by terelius · 8 years ago
  73. 38fd175 Added first layer of the echo canceller 3 functionality. by peah · 8 years ago
  74. ebafdc8 Refactor webrtc/modules/rtp_rtcp for GN check by mbonadei · 8 years ago
  75. 5a38836 Implement Theil-Sen's method for fitting a line to noisy data (used in bandwidth estimation). by terelius · 8 years ago
  76. b0a1111 Decode h264 fmtp sprop-parameter-sets to binary. by johan · 8 years ago
  77. 676e08f Refactor webrtc/{api,audio} and modules/audio_coding for GN check by kjellander · 8 years ago
  78. 9aa9688 Reland of H.264 packetization mode 0 (try 3) (patchset #1 id:1 of https://codereview.webrtc.org/2558453002/ ) by hta · 8 years ago
  79. 768c648 Move /webrtc/api/android files to /webrtc/sdk/android by magjed · 8 years ago
  80. 243a0a7 Revert of H.264 packetization mode 0 (try 3) (patchset #13 id:490001 of https://codereview.webrtc.org/2528343002/ ) by hta · 8 years ago
  81. e59647b This approach passes packetization mode to the encoder as part of by hta · 8 years ago
  82. 6321b49 Move functionality out from AudioFrame and into AudioFrameOperations. by aleloi · 8 years ago
  83. 13d38fb Delete all of the video_processing module but the denoiser code. by nisse · 8 years ago
  84. cb861e0 Templatize percentile_filter.h and move it to base/analytics. by terelius · 8 years ago
  85. 26bddb9 Replace test_support_main by test_main and get rid of test_support_main_threaded_mac by ehmaldonado · 8 years ago
  86. ceecea4 Pass selected cricket::VideoCodec down to internal H264 encoder by magjed · 8 years ago
  87. a8eb756 Moved transport.h from webrtc/ to webrtc/api, created build target and updated WebRTC dependencies. by aleloi · 8 years ago
  88. f3feeff Reland of move RTPPayloadStrategy and simplify RTPPayloadRegistry (patchset #1 id:1 of https://codereview.webrtc.org/2528993002/ ) by magjed · 8 years ago
  89. 76622ce Adding a unit test for RMSLevel by henrik.lundin · 8 years ago
  90. 33c81d0 Revert of Remove RTPPayloadStrategy and simplify RTPPayloadRegistry (patchset #7 id:240001 of https://codereview.webrtc.org/2524923002/ ) by magjed · 8 years ago
  91. 69b627d Move smoothing filter to common audio and exp_filter to base/analytics. by minyue · 8 years ago
  92. b881254 Remove RTPPayloadStrategy and simplify RTPPayloadRegistry by magjed · 8 years ago
  93. b7374db Fix parsing padding byte in rtp header extension by danilchap · 8 years ago
  94. 3c3aef4 Revert of Reland "Move smoothing filter to common audio". (patchset #5 id:100001 of https://codereview.webrtc.org/2520003005/ ) by minyue · 8 years ago
  95. 223641f Reland "Move smoothing filter to common audio". by minyue · 8 years ago
  96. 90ea736 Add DesktopFrame rotation functions by zijiehe · 8 years ago
  97. d661e9c WebRTC: Replace ProjectRootPath by ResourcePath by ehmaldonado · 8 years ago
  98. 8271d04 This CL introduces the new functionality for setting by peah · 8 years ago
  99. dedaf1c Modify audio_processing_unittest to use ResourcePath instead of ProjectRootPath. by ehmaldonado · 8 years ago
  100. d7ac0a9 Revert of Move smoothing filter to common audio. (patchset #3 id:60001 of https://codereview.webrtc.org/2484153002/ ) by magjed · 8 years ago