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webrtc / src / 9ed29d2e4707bea65284e2fdf3a48502fccfb37a / audio / BUILD.gn
  1. 2011075 MB: Add support for isolating scripts + isolate low_bandwidth_audio_test.py. by Edward Lemur · 7 years ago
  2. 18f5427 Remove voe_auto_test and add new tests to cover the missing cases. by solenberg · 8 years ago
  3. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 8 years ago
  4. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 8 years ago[Renamed from webrtc/audio/BUILD.gn]
  5. 73276ad - Removes voe_conference_test. by Fredrik Solenberg · 8 years ago
  6. 84f6a3f Move optional.h to webrtc/api/ by kwiberg · 8 years ago
  7. 9b2f20c Replace gflags usages with rtc_base/flags in all targets based on test_main by oprypin · 8 years ago
  8. 413ee9a Use SingleThreadedTaskQueue in DirectTransport by eladalon · 8 years ago
  9. 037f3e4 Replace absolute path with relative path for GN files. by Jianjun Zhu · 8 years ago
  10. f6a861a Remove remains of webrtc/base by ehmaldonado · 8 years ago
  11. c58f8c0 Adds a histogram metric tracking for how long audio RTP packets are sent by saza · 8 years ago
  12. 9d11764 Reimplemeted "Test and fix for huge bwe drop after alr state" by tschumim · 8 years ago
  13. c024740 Use relative paths in GN files. by jianjun.zhu · 8 years ago
  14. 370dd47 Revert of Remove remains of webrtc/base (patchset #7 id:120001 of https://codereview.webrtc.org/2973183002/ ) by ehmaldonado · 8 years ago
  15. 9483b49 Remove remains of webrtc/base by ehmaldonado · 8 years ago
  16. e75d96b Revert of Test and fix for huge bwe drop after alr state. (patchset #13 id:320001 of https://codereview.webrtc.org/2931873002/ ) by terelius · 8 years ago
  17. 0f15f92 Introduce RtpStreamReceiverInterface and RtpStreamReceiverControllerInterface. by nisse · 8 years ago
  18. 37aa8ba Test and fix for huge bwe drop after alr state. by tschumim · 8 years ago
  19. d76b7b2 New targets call:rtp_interfaces, call:rtp_receiver, call:rtp_sender. by nisse · 8 years ago
  20. 7cb69d5 This will allow me to test that Call invokes SendSideCongestionController::SetBweBitrates as expected (for https://codereview.chromium.org/2793913008). by zstein · 8 years ago
  21. eb1fde4 Injectable audio encoders: Moved audio encoder, factory and builtin factory to api/. by ossu · 8 years ago
  22. 20a4b3f Injectable audio encoders: WebRtcVoiceEngine and company by ossu · 8 years ago
  23. e0629c0 GN: Tighten up test target visibility + refactorings by kjellander · 8 years ago
  24. f250100 Add POLQA to low bandwidth audio test by oprypin · 8 years ago
  25. 6d305ba Add Windows, Mac, Android support to low bandwidth audio test by oprypin · 8 years ago
  26. 92220ff Low-bandwidth audio testing by oprypin · 8 years ago
  27. 5e1ca78 Add low_bandwidth_audio_test to default build by oprypin · 8 years ago
  28. 8f8d1a0 Adding placeholder for low bandwidth audio test by kjellander · 8 years ago
  29. 7de8d64 Wire up audio packet loss to BWE. by stefan · 8 years ago
  30. 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
  31. 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
  32. 35a3270 Moving webrtc.gni up one level from build/ by mbonadei · 8 years ago
  33. 894c2bb GN: Refactor webrtc_nonparallel_tests and audio_tests to avoid crossing package boundaries. by ehmaldonado · 8 years ago
  34. 676e08f Refactor webrtc/{api,audio} and modules/audio_coding for GN check by kjellander · 8 years ago
  35. f515ab8 Moved call.h and most of api/call/* into call/ by ossu · 8 years ago
  36. 6321b49 Move functionality out from AudioFrame and into AudioFrameOperations. by aleloi · 8 years ago
  37. 939e08f Added webrtc/audio/utility directory and empty GN target. by aleloi · 8 years ago
  38. 04c0722 Replace AudioConferenceMixer with AudioMixer. by aleloi · 8 years ago
  39. 10111bc Passed AudioMixer to AudioState::Config. by aleloi · 8 years ago
  40. dd31071 Added an empty AudioTransportProxy to AudioState. by aleloi · 8 years ago
  41. aed581a Made AudioReceiveStream a mixer participant. by aleloi · 8 years ago
  42. e40a7ee GN: Exclude suppressions of Chromium Clang warnings for Chromium builds. by kjellander · 8 years ago
  43. b62dbbe GN: Change rtc_source_set targets --> rtc_static_library by kjellander · 9 years ago
  44. e9cc686 GN Templates: Move common_inherited_config to the template. by ehmaldonado · 9 years ago
  45. 7a2ce0b GN Templates: Move common_config to the template. by ehmaldonado · 9 years ago
  46. 38a2132 GN: Introduce templates. by ehmaldonado · 9 years ago
  47. a69d973 Move webrtc/audio_*.h to webrtc/api/call by kjellander · 9 years ago
  48. 0208322 GN: Add video_engine_tests by Peter Boström · 9 years ago
  49. 50772f1 GN: Update audio_sink.h location by kjellander@webrtc.org · 9 years ago
  50. f888bb5 Support for unmixed remote audio into tracks. by Tommi · 9 years ago
  51. 566ef24 Move VoiceEngineObserver into AudioSendStream so that we detect typing noises and return properly in GetStats(). by solenberg · 9 years ago
  52. 4f4ec0a Re-Land: Implement AudioReceiveStream::GetStats(). by Fredrik Solenberg · 9 years ago
  53. 43e83d4 Revert of Implement AudioReceiveStream::GetStats(). (patchset #19 id:360001 of https://codereview.webrtc.org/1390753002/ ) by solenberg · 9 years ago
  54. a457752 Implement AudioReceiveStream::GetStats(). by Fredrik Solenberg · 9 years ago
  55. c7a8b08 Add webrtc::AudioSendStream and methods on webrtc::Call to create and delete AudioSendStreams. by solenberg · 9 years ago
  56. 5c389d3 Split webrtc/video into webrtc/{audio,call,video}. by Peter Boström · 10 years ago