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gerrit-public.fairphone.software / platform / external / webrtc / 3c3aef44deb4cca21ece8532e8dfae1b92c7989a / webrtc / video / BUILD.gn
  1. 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
  2. 223641f Reland "Move smoothing filter to common audio". by minyue · 8 years ago
  3. f6acc2a Move VideoDecoderSoftwareFallbackWrapper from webrtc/video_decoder.h to webrtc/media/engine/ by magjed · 8 years ago
  4. 509e4fe Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ ) by magjed · 8 years ago
  5. eacbaea Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ ) by magjed · 8 years ago
  6. 42043b9 Stop using hardcoded payload types for video codecs by Magnus Jedvert · 8 years ago
  7. 614d5b7 Move VideoEncoderSoftwareFallbackWrapper from webrtc/video_encoder.h to webrtc/media/engine/ by magjed · 8 years ago
  8. 445fb8f Use correct define in H264 end-to-end tests. by brandtr · 8 years ago
  9. 13ceeea Revert of H.264 packetization mode 0 (try 2) (patchset #27 id:520001 of https://codereview.webrtc.org/2337453002/ ) by magjed · 8 years ago
  10. 3bba101 Implement H.264 packetization mode 0. by hta · 8 years ago
  11. bf6a45b Moved transport_adapter.h/.cc from call/ to video/ dir to remove circular dependency by charujain · 8 years ago
  12. e5ba44e Implement framesDecoded stat in video receive ssrc stats. by sakal · 8 years ago
  13. a3cac05 GN: move webrtc/video/ targets from webrtc/BUILD.gn into webrtc/video/BUILD.gn by kjellander · 8 years ago
  14. e40a7ee GN: Exclude suppressions of Chromium Clang warnings for Chromium builds. by kjellander · 8 years ago
  15. ae04e36 Cleanup unused dependency on video_capture_module. by perkj · 8 years ago
  16. 15d8357 Remove OnLocalSsrcChanged and rename EncoderStateFeedback. by mflodman · 8 years ago
  17. cc91d28 Moved RtcEventLog files from call/ to logging/ by skvlad · 8 years ago
  18. 89a3a1a Moved Gn target rtc_event_log to one directory above. by charujain · 8 years ago
  19. b62dbbe GN: Change rtc_source_set targets --> rtc_static_library by kjellander · 8 years ago
  20. e9cc686 GN Templates: Move common_inherited_config to the template. by ehmaldonado · 8 years ago
  21. 7a2ce0b GN Templates: Move common_config to the template. by ehmaldonado · 8 years ago
  22. 38a2132 GN: Introduce templates. by ehmaldonado · 8 years ago
  23. 26091b1 This reverts commit 8eb37a39e79fe1098d3503dcb8c8c2d196203fed. Chrome now have its own implementation of TaskQueues that is based on Chrome threads. by perkj · 8 years ago
  24. 8eb37a3 Revert of Add task queue to Call. (patchset #42 id:840001 of https://codereview.webrtc.org/2060403002/ ) by perkj · 8 years ago
  25. cc16836 - Add task queue to Call with the intent of replacing the use of one of the process threads. by perkj · 8 years ago
  26. 4cd2790 Move RTP for synchroninzation and rename classes, files and variables. by mflodman · 8 years ago
  27. 4e417b2 Reland of Switch to use SequencedTaskChecker instead of ThreadChecker where needed. by perkj · 8 years ago
  28. efd902c Revert of Switch to use SequencedTaskChecker instead of ThreadChecker where needed. (patchset #1 id:1 of https://codereview.webrtc.org/2149553002/ ) by perkj · 8 years ago
  29. ec7cef8 Switch to use SequencedTaskChecker instead of ThreadChecker where needed. by perkj · 8 years ago
  30. 37ad337 Remove EncodedFrameCallbackAdapter. by sergeyu · 8 years ago
  31. 0208322 GN: Add video_engine_tests by Peter Boström · 8 years ago
  32. 0e9d6d9 Add class which periodically computes statistics. by asapersson · 8 years ago
  33. cfc8e3b Removed all RTP dependencies from ViEChannel and renamed class. by mflodman · 8 years ago
  34. 35151f3 Add histogram stats for average send delay of sent packets for a sent video stream. The delay is measured from a packet is sent to the transport until leaving the socket. by asapersson · 8 years ago
  35. b99395a Reland of Delete video_render module. (patchset #1 id:1 of https://codereview.webrtc.org/1923613003/ ) by nisse · 8 years ago
  36. fa66659 Rename ViEReceiver and move ownership to VideoReceiveStream. by mflodman · 8 years ago
  37. 0190367 Revert of Delete video_render module. (patchset #12 id:220001 of https://codereview.webrtc.org/1912143002/ ) by kjellander · 8 years ago
  38. 97cfd1e Delete video_render module. by nisse · 8 years ago
  39. 7ffeab5 Reland "Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies." by kjellander@webrtc.org · 9 years ago
  40. 7324eb9 Revert of Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies. (patchset #2 id:40001 of https://codereview.webrtc.org/1737593002/ ) by kjellander · 9 years ago
  41. 99b345c Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies. by kjellander@webrtc.org · 9 years ago
  42. 7623ce4 Reland of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:300001 of https://codereview.webrtc.org/1507903005/ ) by Peter Boström · 9 years ago
  43. 8237abf Revert of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:20001 of https://codereview.webrtc.org/1506773002/ ) by kjellander · 9 years ago
  44. 03ef053 Merge webrtc/video_engine/ into webrtc/video/ by Peter Boström · 9 years ago
  45. c4a1c37 Removed vie_defines.h by mflodman · 9 years ago
  46. 0c478b3 Rename ChannelGroup to CongestionController and move to webrtc/call/. by mflodman · 9 years ago
  47. 5c389d3 Split webrtc/video into webrtc/{audio,call,video}. by Peter Boström · 9 years ago
  48. ebbf8a8 Make sure rtp_rtcp module doesn't directly reference anything in the pacer module, and remove build dependencies on it. by sprang · 9 years ago
  49. 3641185 Includes webrtc/build/protoc.gypi instead of build/protoc.gypi by Bjorn Terelius · 9 years ago
  50. b933667 Revert "Renamed the ACMDump to RtcEventLog and moved it to webrtc/video, since it is not specific to the audio coding module. Updated .gyp and .gn files accordingly." by Bjorn Terelius · 9 years ago
  51. c159b04 Renamed the ACMDump to RtcEventLog and moved it to webrtc/video, since it is not specific to the audio coding module. Updated .gyp and .gn files accordingly. by Bjorn Terelius · 9 years ago
  52. 4b91bd0 Move frame input (ViECapturer) to webrtc/video/. by Peter Boström · 9 years ago
  53. 2ee2439 Merge video_engine_core into webrtc target. by Peter Boström · 9 years ago
  54. 57e5fd2 PRESUBMIT: Improve PyLint check and add GN format check. by Henrik Kjellander · 9 years ago
  55. 4d71ede Add HW fallback option to software encoding. by Peter Boström · 9 years ago
  56. 7252a2b Add HW fallback option to software decoding. by Peter Boström · 9 years ago
  57. 23fba1f Add AudioReceiveStream to Call API. by Fredrik Solenberg · 9 years ago
  58. ac2d27d Fix style violations in common_types.h and config.h by kwiberg@webrtc.org · 10 years ago
  59. f21ea91 GN: Add common configs to all targets. by kjellander@webrtc.org · 10 years ago
  60. 1b9a188 GN: Fix webrtc/video/BUILD.gn for Chromium build. by kjellander@webrtc.org · 10 years ago
  61. 788f058 GN: Implement video_engine, video_capture and video_render. by kjellander@webrtc.org · 10 years ago
  62. 1227ab8 GN: Add BUILD.gn files + kjellander to OWNERS by kjellander@webrtc.org · 10 years ago