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gerrit-public.fairphone.software / platform / external / webrtc / a3177057c7d8010d350de6e9c615331deb9e0504 / call / BUILD.gn
  1. 11b34f4 Remove chromium clang style errors affecting sdk/android/media_jni by Paulina Hensman · 7 years ago
  2. abbe841 This CL removes all usages of our custom ostream << overloads. by Jonas Olsson · 7 years ago
  3. 88614b0 Pass VideoEncoderFactory from WebrtcVideoEngine to VideoStreamEncoder. by Niels Möller · 7 years ago
  4. 2b85792 Move rw_lock_wrapper.h to rtc_base/synchronization/ by Karl Wiberg · 7 years ago
  5. b34556e Added receive time calculator under field trial. by Sebastian Jansson · 7 years ago
  6. 7696bef Remove the public_deps to fileutils from test_support. by Patrik Höglund · 7 years ago
  7. 7bd79a0 Split up audio_device build target by Paulina Hensman · 7 years ago
  8. 0970851 Reland: Add ability to emulate degraded network in Call via field trial by Erik Språng · 7 years ago
  9. 16cba5c Revert "Add ability to emulate degraded network in Call via field trial" by Ilya Nikolaevskiy · 7 years ago
  10. 31a12c5 Add ability to emulate degraded network in Call via field trial by Erik Språng · 7 years ago
  11. 19bea51 Adding task queue congestion control experiment. by Sebastian Jansson · 7 years ago
  12. 3faa832 Separate test/fake_audio_device on API and implementation. Step 2. by Artem Titov · 7 years ago
  13. 19704ec Removing AvailableBandwidth method on transport controller. by Sebastian Jansson · 7 years ago
  14. 6723cdc Revert "Separate test/fake_audio_device on API and implementation." by Artem Titov · 7 years ago
  15. 8ea5f9a Separate test/fake_audio_device on API and implementation. by Artem Titov · 7 years ago
  16. a646d30 Enables configuration of transmission max bitrate multiplier and fec protection level. by Ying Wang · 7 years ago
  17. 45087cd Moved retransmission rate limiter to Call class. by Sebastian Jansson · 7 years ago
  18. c33c0fc Moved pacer and congestion thread from call. by Sebastian Jansson · 7 years ago
  19. 35dd6cd Added dependencies to mock transport controller send. by Sebastian Jansson · 7 years ago
  20. 8f83b42 Moved bitrate config interface from Call class. by Sebastian Jansson · 7 years ago
  21. 91bb667 Moved routes tracking to rtp transport controller. by Sebastian Jansson · 7 years ago
  22. 97f61ea Moved bitrate configuration to rtp controller by Sebastian Jansson · 7 years ago
  23. e5447fb Removed fake rtp transport controller send. by Sebastian Jansson · 7 years ago
  24. df023aa Extracted bitrate configuration from call class. by Sebastian Jansson · 7 years ago
  25. fc8d26b Reland "Moved BitrateConfig out of Call::Config." by Sebastian Jansson · 7 years ago
  26. e4bf600 Revert "Moved BitrateConfig out of Call::Config." by Lu Liu · 7 years ago
  27. 5897fe2 Moved BitrateConfig out of Call::Config. by Sebastian Jansson · 7 years ago
  28. 2ae140a BUILD.gn file for api/audio. by Gustaf Ullberg · 7 years ago
  29. 8366e17 Rename Call::Config to CallConfig, keep old name as alias. by Niels Möller · 7 years ago
  30. 970b088 Reland "Break up rtc_event_log_api to solve circular dependencies." by Qingsi Wang · 7 years ago
  31. 75df728 Revert "Break up rtc_event_log_api to solve circular dependencies." by Mirko Bonadei · 7 years ago
  32. 001546d Break up rtc_event_log_api to solve circular dependencies. by Qingsi Wang · 7 years ago
  33. 65ce311 Removing useless dependencies on //testing/gmock. by Mirko Bonadei · 7 years ago
  34. 3b790f3 Make fec controller plug-able. by Ying Wang · 7 years ago
  35. 2ffe3e8 Reland Use runtime enabled features API to enable dual stream mode by Ilya Nikolaevskiy · 7 years ago
  36. c1094eb Revert "Use runtime enabled features API to enable dual stream mode" by Lu Liu · 7 years ago
  37. 6f011df Use runtime enabled features API to enable dual stream mode by Ilya Nikolaevskiy · 7 years ago
  38. a8b7c7f Move remaining traces of VoiceEngine by Fredrik Solenberg · 7 years ago
  39. 9c68613 Update gn files to support Mozilla build by Dan Minor · 7 years ago
  40. 8f5787a Move ownership of voe::Channel into Audio[Receive|Send]Stream. by Fredrik Solenberg · 7 years ago
  41. a7f2d84 Revert "Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*""" by Per Kjellander · 7 years ago
  42. c73e1f4 Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*"" by Per Kjellander · 7 years ago
  43. 588c548 GN rtc_* templates: Set default visibility to webrtc_root + "/*" by Karl Wiberg · 7 years ago
  44. e66572b Reland "iOS: Save perf results under Documents/perf_result.json" by Edward Lemur · 7 years ago
  45. 9e19403 Move videosourceinterface to api. by Patrik Höglund · 7 years ago
  46. 6213929 Add missing files to audio_processing. by Patrik Höglund · 7 years ago
  47. 2a87797 Remove voe::TransmitMixer by Fredrik Solenberg · 7 years ago
  48. 3e11343 Fix circular dependencies in webrtc_common. by Patrik Höglund · 7 years ago
  49. 712989d Revert "Reland "iOS: Save perf results under Documents/perf_result.json"" by Mirko Bonadei · 7 years ago
  50. a8005cf Fix circular dependencies between optional, array_view, and rtc_base. by Patrik Höglund · 7 years ago
  51. 8b886bb Reland "iOS: Save perf results under Documents/perf_result.json" by Edward Lemur · 7 years ago
  52. d37709b Revert "Fix circular dependencies between optional, array_view, and rtc_base." by Patrik Höglund · 7 years ago
  53. 081c651 Revert "iOS: Save perf results under Documents/perf_result.json" by Rasmus Brandt · 7 years ago
  54. a9e0924 Fix circular dependencies between optional, array_view, and rtc_base. by Patrik Höglund · 7 years ago
  55. 10a8e7a iOS: Save perf results under Documents/perf_result.json by Edward Lemur · 7 years ago
  56. a498ae8 Stop using public_deps in system_wrappers. by Mirko Bonadei · 7 years ago
  57. b5728d9 Stop using public_deps in modules/rtp_rtcp. by Mirko Bonadei · 7 years ago
  58. 03d6f2f Stop using public_deps in modules/audio_mixer. by Mirko Bonadei · 7 years ago
  59. a0e1a55 Stop using public_deps in the call module. by Mirko Bonadei · 7 years ago
  60. ad62792 Fixing hidden dependencies. by Mirko Bonadei · 7 years ago
  61. 56d4609 Use the new AudioProcessing statistics everywhere. by Ivo Creusen · 7 years ago
  62. c0e6804 Fix deps of audio:audio_tests. by Patrik Höglund · 7 years ago
  63. 61a7b14 Removing conditional visibility. by Mirko Bonadei · 7 years ago
  64. fd6c091 Delete deprecated constructor of SendSideCongestionController. by Niels Möller · 7 years ago
  65. f3850f6 Voice Engine: Require caller to supply an AudioDecoderFactory by Karl Wiberg · 7 years ago
  66. 245660a Fix Gn untracked headers in webrtc/call. by Mirko Bonadei · 7 years ago
  67. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/call/BUILD.gn]
  68. 84f6a3f Move optional.h to webrtc/api/ by kwiberg · 7 years ago
  69. 529662a Move array_view.h to webrtc/api/ by kwiberg · 7 years ago
  70. 334f9e6 Tracking mock_paced_sender.h with a GN target by mbonadei · 7 years ago
  71. 1acbd68 Move RtpExtension to api/ directory and config.h/.cc to call/. by Stefan Holmer · 7 years ago
  72. 95c8f65 Now that https://codereview.webrtc.org/3003643002 is landed we can by mbonadei · 7 years ago
  73. 440b6d9 Move video send/receive stream headers to webrtc/call. by aleloi · 7 years ago
  74. f3f5c0e Change ThreadChecker to SequencedTaskChecker in internal::Call by eladalon · 7 years ago
  75. b332917 Rename RsidResolutionObserver to SsrcBindingObserver. by Steve Anton · 7 years ago
  76. db2a9fc Wire up RTP keep-alive in ortc api. by sprang · 7 years ago
  77. 5166e54 Tracking mock_process_thread with a GN target by mbonadei · 7 years ago
  78. e2173d9 Only one implementation of MockRtpPacketSink once by eladalon · 7 years ago
  79. f6a861a Remove remains of webrtc/base by ehmaldonado · 7 years ago
  80. c024740 Use relative paths in GN files. by jianjun.zhu · 7 years ago
  81. 370dd47 Revert of Remove remains of webrtc/base (patchset #7 id:120001 of https://codereview.webrtc.org/2973183002/ ) by ehmaldonado · 7 years ago
  82. 9483b49 Remove remains of webrtc/base by ehmaldonado · 7 years ago
  83. a52722f Reland of Create RtcpDemuxer (patchset #1 id:1 of https://codereview.webrtc.org/2957763002/ ) by eladalon · 8 years ago
  84. 0e7e786 Revert of Create RtcpDemuxer (patchset #13 id:240001 of https://codereview.webrtc.org/2943693003/ ) by guidou · 8 years ago
  85. cb83bdf Create RtcpDemuxer. Capabilities: by eladalon · 8 years ago
  86. 0f15f92 Introduce RtpStreamReceiverInterface and RtpStreamReceiverControllerInterface. by nisse · 8 years ago
  87. 38ede13 Support building WebRTC without audio and video. by zhihuang · 8 years ago
  88. d76b7b2 New targets call:rtp_interfaces, call:rtp_receiver, call:rtp_sender. by nisse · 8 years ago
  89. 760a076 Create unit tests for RtpDemuxer by eladalon · 8 years ago
  90. c3d4b48 Store/restore RTP state for audio streams with same SSRC within a call by ossu · 8 years ago
  91. eed52bf New class RtxReceiveStream. by nisse · 8 years ago
  92. e4bcd6d New class RtpDemuxer and RtpPacketSinkInterface, use in Call. by nisse · 8 years ago
  93. 2d9d21f Add untracked headers in modules/rtp_rtcp by danilchap · 8 years ago
  94. 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
  95. eb1fde4 Injectable audio encoders: Moved audio encoder, factory and builtin factory to api/. by ossu · 8 years ago
  96. 20a4b3f Injectable audio encoders: WebRtcVoiceEngine and company by ossu · 8 years ago
  97. 81c79f5 Creating webrtc:video_stream_api by mbonadei · 8 years ago
  98. e0629c0 GN: Tighten up test target visibility + refactorings by kjellander · 8 years ago
  99. cae45d0 Move RtpTransportControllerSend to a new file. by nisse · 8 years ago
  100. 8d609f6 Reland of Implemented the GetSources() in native code. (patchset #1 id:1 of https://codereview.webrtc.org/2809613002/ ) by hbos · 8 years ago