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gerrit-public.fairphone.software / platform / external / webrtc / 02ee47c1ae2a7beecae36a886994d82390eb3d57 / call
  1. 3b903d0 Reconfigure, not reconstruct, AudioReceiveStreams. by Fredrik Solenberg · 7 years ago
  2. a7f2d84 Revert "Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*""" by Per Kjellander · 7 years ago
  3. c73e1f4 Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*"" by Per Kjellander · 7 years ago
  4. 588c548 GN rtc_* templates: Set default visibility to webrtc_root + "/*" by Karl Wiberg · 7 years ago
  5. dfe9ffc Added active field to constructor and ToString() of VideoStream. by Seth Hampson · 7 years ago
  6. 62337e5 Use AudioProcessingBuilder everywhere AudioProcessing is created. by Ivo Creusen · 7 years ago
  7. 7f331fa Add metric name for MinVideoAndAudioBitRate. by Edward Lemur · 7 years ago
  8. 24722b3 Reland "Wiring the RTCRtpEncodingParameters.priority down to the bitrate allocator." by Seth Hampson · 7 years ago
  9. e66572b Reland "iOS: Save perf results under Documents/perf_result.json" by Edward Lemur · 7 years ago
  10. 947f3fe Fix reporting of perf results on PlaysOutAudioAndVideoInSync* tests by Edward Lemur · 7 years ago
  11. dd3987f Add _[no]red suffix to RampUpTests. by Edward Lemur · 7 years ago
  12. 9e19403 Move videosourceinterface to api. by Patrik Höglund · 7 years ago
  13. be214a2 Move videosinkinterface.h to common_video to solve a circular dep. by Patrik Höglund · 7 years ago
  14. 6213929 Add missing files to audio_processing. by Patrik Höglund · 7 years ago
  15. 8b77aea Revert "Wiring the RTCRtpEncodingParameters.priority down to the bitrate allocator." by Lu Liu · 7 years ago
  16. d2b912a Wiring the RTCRtpEncodingParameters.priority down to the bitrate allocator. by Seth Hampson · 7 years ago
  17. 36193c3 Adds active field to VideoStream struct. by Seth Hampson · 7 years ago
  18. f32795e Updates to video config to allow changes in google3 tests, in order to not break anything. by Seth Hampson · 7 years ago
  19. 2a87797 Remove voe::TransmitMixer by Fredrik Solenberg · 7 years ago
  20. 3e11343 Fix circular dependencies in webrtc_common. by Patrik Höglund · 7 years ago
  21. 712989d Revert "Reland "iOS: Save perf results under Documents/perf_result.json"" by Mirko Bonadei · 7 years ago
  22. a8005cf Fix circular dependencies between optional, array_view, and rtc_base. by Patrik Höglund · 7 years ago
  23. 8b886bb Reland "iOS: Save perf results under Documents/perf_result.json" by Edward Lemur · 7 years ago
  24. d37709b Revert "Fix circular dependencies between optional, array_view, and rtc_base." by Patrik Höglund · 7 years ago
  25. 081c651 Revert "iOS: Save perf results under Documents/perf_result.json" by Rasmus Brandt · 7 years ago
  26. a9e0924 Fix circular dependencies between optional, array_view, and rtc_base. by Patrik Höglund · 7 years ago
  27. 10a8e7a iOS: Save perf results under Documents/perf_result.json by Edward Lemur · 7 years ago
  28. cbf5b73 Explicitly convert size_t to int in Call::DeliverPacket by Danil Chapovalov · 7 years ago
  29. 292a73e Deliver packet to Call as rtc::CopyOnWriteBuffer by Danil Chapovalov · 7 years ago
  30. a498ae8 Stop using public_deps in system_wrappers. by Mirko Bonadei · 7 years ago
  31. b5728d9 Stop using public_deps in modules/rtp_rtcp. by Mirko Bonadei · 7 years ago
  32. 03d6f2f Stop using public_deps in modules/audio_mixer. by Mirko Bonadei · 7 years ago
  33. a0e1a55 Stop using public_deps in the call module. by Mirko Bonadei · 7 years ago
  34. cf73c96 Add AudioDeviceModule to AudioState::Config. by Fredrik Solenberg · 7 years ago
  35. ad62792 Fixing hidden dependencies. by Mirko Bonadei · 7 years ago
  36. 1eb051c Made functions on BitrateAllocator::ObserverConfig member functions by srte · 7 years ago
  37. 2f06168 Make PrintResultList receive a vector of doubles instead of a string. by Edward Lemur · 7 years ago
  38. d0e196b Adding two tests: by Alex Narest · 7 years ago
  39. 56d4609 Use the new AudioProcessing statistics everywhere. by Ivo Creusen · 7 years ago
  40. a6092a9 Deprecated the Get BitrateController method by srte · 7 years ago
  41. e40468b Move some numeric utility code from rtc_base/ to rtc_base/numerics/ by Karl Wiberg · 7 years ago
  42. d319534 Move ADM initialization into WebRtcVoiceEngine by Fredrik Solenberg · 7 years ago
  43. 63e6072 Add AudioState::audio_transport() to prepare clients for moving ADM initialization out of VoiceEngine. by Fredrik Solenberg · 7 years ago
  44. c3ed630 Add stats googHasEnteredLowResolution. by Åsa Persson · 7 years ago
  45. e0b2ff5 Add kTransmissionMaxBitrateMultiplier logic to audio priority bitrate allocation strategy similarly to default bitrate allocation logic. by Alex Narest · 7 years ago
  46. fe73d6a Extended the bitrate allocator to allow allocation to tracks based upon priorities which are planned to be defined as a relative bitrate in the RTCRtpEncodingParameters. by Seth Hampson · 7 years ago
  47. c0e6804 Fix deps of audio:audio_tests. by Patrik Höglund · 7 years ago
  48. 61a7b14 Removing conditional visibility. by Mirko Bonadei · 7 years ago
  49. 675513b Stop using LOG macros in favor of RTC_ prefixed macros. by Mirko Bonadei · 7 years ago
  50. fd6c091 Delete deprecated constructor of SendSideCongestionController. by Niels Möller · 7 years ago
  51. f3850f6 Voice Engine: Require caller to supply an AudioDecoderFactory by Karl Wiberg · 7 years ago
  52. 5f6bf24 Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API (II) by henrika · 7 years ago
  53. de69145 Remove pbos@webrtc.org from all OWNERS. by Peter Boström · 7 years ago
  54. 990d6b8 Revert "Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API" by Mirko Bonadei · 7 years ago
  55. 90bace0 Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API by henrika · 7 years ago
  56. d79314f Reland "Add fine grained dropped video frames counters on sending side" by Ilya Nikolaevskiy · 7 years ago
  57. 1c1a681 Revert "Add fine grained dropped video frames counters on sending side" by Ilya Nikolaevskiy · 7 years ago
  58. 78609d5 Reland of BWE allocation strategy by Alex Narest · 7 years ago
  59. dc9ca93 Revert "BWE allocation strategy" by Alex Narest · 7 years ago
  60. 4b1a363 Add fine grained dropped video frames counters on sending side by Ilya Nikolaevskiy · 7 years ago
  61. a5fbc23 BWE allocation strategy by Alex Narest · 7 years ago
  62. 39260c4 Revert "BWE allocation strategy allows controlling of bitrate allocation with WEBRTC external logic." by Lu Liu · 7 years ago
  63. 54d1da1 BWE allocation strategy allows controlling of bitrate allocation with WEBRTC external logic. by Alex Narest · 7 years ago
  64. b3944f0 Media track ID visibility at BWE level by Alex Narest · 7 years ago
  65. 05d9822 Disable RampUpTest.UpDownUpTransportSequenceNumberPacketLoss. by Taylor Brandstetter · 7 years ago
  66. b709cf8 Remove Call::ParseRtpPacket by Danil Chapovalov · 7 years ago
  67. 245660a Fix Gn untracked headers in webrtc/call. by Mirko Bonadei · 7 years ago
  68. 4bece9a Set RTPVideoHeader picture id in PayloadRouter if forced fallback for VP8 is enabled. by Åsa Persson · 7 years ago
  69. 4332d09 Reland "Reland "Remove WEBRTC_TRACE."" by Fredrik Solenberg · 7 years ago
  70. a32dd01 Reland "Remove AudioDeviceObserver and make ADM not inherit from the Module interface." by Fredrik Solenberg · 7 years ago
  71. 39cefdb Revert "Reland "Remove WEBRTC_TRACE."" by Fredrik Solenberg · 7 years ago
  72. 68007e9 Reland "Remove WEBRTC_TRACE." by Fredrik Solenberg · 7 years ago
  73. 4a87e1c Remove encoding code from RtcEventLogImpl and use RtcEventLogEncoder instead by Elad Alon · 7 years ago
  74. 729b910 Revert "Remove WEBRTC_TRACE." by Fredrik Solenberg · 7 years ago
  75. 2209b90 Remove WEBRTC_TRACE. by Fredrik Solenberg · 7 years ago
  76. d4404c2 Revert "Remove AudioDeviceObserver and make ADM not inherit from the Module interface." by Fredrik Solenberg · 7 years ago
  77. 34cdd2d Remove AudioDeviceObserver and make ADM not inherit from the Module interface. by Fredrik Solenberg · 7 years ago
  78. b0a0207 Added RTCMediaStreamTrackStats.jitterBufferDelay for audio by Gustaf Ullberg · 7 years ago
  79. 440216f Split LogRtpHeader and LogRtcpPacket into separate versions for incoming and outgoing packets. by Bjorn Terelius · 7 years ago
  80. 3102734 Revert "Reland of Add full stack tests for MediaCodec encoder (moved from Rietveld)." by Rasmus Brandt · 7 years ago
  81. 2666cf7 Reland of Add full stack tests for MediaCodec encoder (moved from Rietveld). by Rasmus Brandt · 7 years ago
  82. 5aea38c Disabling CallPerfTest.{CaptureNtpTimeWithNetworkDelay,CaptureNtpTimeWithNetworkJitter}. by Alex Loiko · 7 years ago
  83. 06319b7 Disable RampUpTest.UpDownUpTransportSequenceNumberPacketLoss on Mac. by Alex Loiko · 7 years ago
  84. 1405afe Disable RampUpTest.UpDownUpTransportSequenceNumberPacketLoss on Linux due to flakiness. by lliuu · 7 years ago
  85. 3b3622f Delete member VideoReceiveStream::Config::Rtp::ulpfec. by nisse · 7 years ago
  86. 2c30120 Revert of Add full stack tests for MediaCodec. (patchset #10 id:180001 of https://codereview.webrtc.org/3005253002/ ) by brandtr · 7 years ago
  87. 2cefac6 Add full stack tests for MediaCodec encoder. by brandtr · 7 years ago
  88. 99a81b6 Remove #include of rtc_stream_config.h from rtc_event_log.h by Elad Alon · 7 years ago
  89. 9a2e906 Added RTCMediaStreamTrackStats.concealmentEvents by Gustaf Ullberg · 7 years ago
  90. 7120742 Adding NOLINT for typedefs.h and common_types.h by Mirko Bonadei · 7 years ago
  91. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  92. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago