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gerrit-public.fairphone.software / platform / external / webrtc / 29accefbb2282281dfb51c82d322e05b66cfe858 / logging / BUILD.gn
  1. a96fd7f Make rtc_event_log2text handle all events [2/2] by Elad Alon · 7 years ago
  2. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  3. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/logging/BUILD.gn]
  4. 4bb3b9c Move StreamConfig into its own file by eladalon · 7 years ago
  5. 248fd4f Reland of Make RtcEventLogImpl to use a TaskQueue instead of a helper-thread ( https://codereview.webrtc.org/3007473002/ ) by eladalon · 7 years ago
  6. 23814b7 Revert of Make RtcEventLogImpl to use a TaskQueue instead of a helper-thread (patchset #4 id:200001 of https://codereview.webrtc.org/3005153002/ ) by eladalon · 7 years ago
  7. d67cefb Reland of Make RtcEventLogImpl to use a TaskQueue instead of a helper-thread (patchset #1 id:1 of https://codereview.webrtc.org/3010143002/ ) by eladalon · 7 years ago
  8. 3eac800 Revert of Make RtcEventLogImpl to use a TaskQueue instead of a helper-thread (patchset #18 id:340001 of https://codereview.webrtc.org/3007473002/ ) by charujain · 7 years ago
  9. f33cee7 Make RtcEventLogImpl to use a TaskQueue instead of a helper-thread by eladalon · 7 years ago
  10. 1acbd68 Move RtpExtension to api/ directory and config.h/.cc to call/. by Stefan Holmer · 7 years ago
  11. 6e09d87 Replace remaining gflags usages with rtc_base/flags by oprypin · 7 years ago
  12. 440b6d9 Move video send/receive stream headers to webrtc/call. by aleloi · 7 years ago
  13. b5c319a Add rtc_event_log_unittest_helper.h to relevant BUILD.gn by eladalon · 7 years ago
  14. f6a861a Remove remains of webrtc/base by ehmaldonado · 7 years ago
  15. 370dd47 Revert of Remove remains of webrtc/base (patchset #7 id:120001 of https://codereview.webrtc.org/2973183002/ ) by ehmaldonado · 7 years ago
  16. 9483b49 Remove remains of webrtc/base by ehmaldonado · 7 years ago
  17. 38ede13 Support building WebRTC without audio and video. by zhihuang · 8 years ago
  18. f53c4cd Delete rtc_event_log/ringbuffer.h by terelius · 8 years ago
  19. 77cd58e This cl removes RtcEventLog deps to call:call_interfaces. The purpose is to be able to use the event log from the upcoming RtpTransport. by perkj · 8 years ago
  20. ee37e86 Create tool to print statistics about the file size usage of an RTC event log. by terelius · 8 years ago
  21. 81c79f5 Creating webrtc:video_stream_api by mbonadei · 8 years ago
  22. 9765370 Resolve dependency between rtc_event_log_api and remote_bitrate_estimator by michaelt · 8 years ago
  23. 7c2c843 Reland of Loosening the coupling between WebRTC and //third_party/protobuf (patchset #1 id:1 of https://codereview.webrtc.org/2786363002/ ) by mbonadei · 8 years ago
  24. cde46b7 Resolve cyclic dependency between audio network adaptor and event log api by michaelt · 8 years ago
  25. d00aad5 Revert of Loosening the coupling between WebRTC and //third_party/protobuf (patchset #16 id:300001 of https://codereview.webrtc.org/2747863003/ ) by mbonadei · 8 years ago
  26. 16ab93b To accommodate some downstream WebRTC users we need to loosen by mbonadei · 8 years ago
  27. 727ac1d Enable GN check for webrtc/logging by kjellander · 8 years ago
  28. 087bd34 Move AudioDecoder and related stuff to the api/ directory by kwiberg · 8 years ago
  29. bb46b95 Add option to print information about configured SSRCs from RTC event logs. by terelius · 8 years ago
  30. d4ed7f5 New tool for printing basic packet information from an RTC event log to stdout. by terelius · 8 years ago
  31. 54b6e98 Added gn target for rtc_event_log2rtp_dump. by ivoc · 8 years ago
  32. 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
  33. 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
  34. 4b7c952 Reland of "Log audio network adapter decisions in event log." by minyue · 8 years ago
  35. 35a3270 Moving webrtc.gni up one level from build/ by mbonadei · 8 years ago
  36. 1fd08c1 GN: Refactor so that WebRTC compiles with rtc_enable_protobuf=false. by ehmaldonado · 8 years ago
  37. 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
  38. 3663681 Log audio network adapter decisions in event log. by michaelt · 8 years ago
  39. f515ab8 Moved call.h and most of api/call/* into call/ by ossu · 8 years ago
  40. 6ceab08 GN: New conventions, default target and refactorings by kjellander · 8 years ago
  41. 940b6d6 Clean up logging in AudioSendStream::SetupSendCodec(). by solenberg · 8 years ago
  42. 189f9b1 Revert of Clean up logging in AudioSendStream::SetupSendCodec(). (patchset #3 id:40001 of https://codereview.webrtc.org/2446963003/ ) by terelius · 8 years ago
  43. 1836fd6 Clean up logging in AudioSendStream::SetupSendCodec(). by solenberg · 8 years ago
  44. e40a7ee GN: Exclude suppressions of Chromium Clang warnings for Chromium builds. by kjellander · 8 years ago
  45. cc91d28 Moved RtcEventLog files from call/ to logging/ by skvlad · 8 years ago