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gerrit-public.fairphone.software / platform / external / webrtc / 642a49d1eb20b8c5744e745de79ddb585e0f7472
  1. 642a49d Add wrapper to normalize flags. by Mirko Bonadei · 5 years ago
  2. 52e240e Use 16000Hz audio in PC test when specified by Artem Titov · 5 years ago
  3. ff25b87 Implements method on RtpPacket to extract extension. by Anton Sukhanov · 5 years ago
  4. ca5f21e Make force_fieldtrials persistent string during entire program live. by Ruslan Burakov · 5 years ago
  5. a1471b7 Roll chromium_revision 34d38f69ba..fff68b4152 (675873:675999) by chromium-webrtc-autoroll · 5 years ago
  6. b1f2d60 Reland "Fix collection of audio metrics from PC test framework for audio test" by Artem Titov · 5 years ago
  7. ca16021 Update rtc_tools/rtp_generator to compile by Ilya Nikolaevskiy · 5 years ago
  8. 1685059 Add support of quick test mode into PC framework by Artem Titov · 5 years ago
  9. 41300af Poison default task queue factory by Danil Chapovalov · 5 years ago
  10. 80cb3f6 Remove the injectable bitrate allocation strategy API. by Jonas Olsson · 5 years ago
  11. 0d90a1d Do not use hungarian notation for DwmGetWindowAttribute's params by Julien Isorce · 5 years ago
  12. d6f4f74 Roll chromium_revision 58be81bf4b..34d38f69ba (675720:675873) by chromium-webrtc-autoroll · 5 years ago
  13. 6f420e4 Reland "Add ability to set RTCP sender ssrc at construction time" by Erik Språng · 5 years ago
  14. 4876cb2 Revert "Fix collection of audio metrics from PC test framework for audio test" by Mirko Bonadei · 5 years ago
  15. 49fa4ea Roll chromium_revision 2bc3837c3d..58be81bf4b (675350:675720) by chromium-webrtc-autoroll · 5 years ago
  16. d0679bd Enables usage of ChannelMixer in WebRTC's output mixer. by henrika · 5 years ago
  17. b249c54 Delete GlobalTaskQueueFactory as now unused by Danil Chapovalov · 5 years ago
  18. 6456e35 Use max bitrate limit recommended by encoder. by Sergey Silkin · 5 years ago
  19. 2d0880b Fix collection of audio metrics from PC test framework for audio test by Artem Titov · 5 years ago
  20. bc558ce Add support of specifying audio sample rate for PC test framework by Artem Titov · 5 years ago
  21. 495a1ae Remove cricket::WebRtcMediaEngineFactory as now unused by Danil Chapovalov · 5 years ago
  22. cb60a8b Roll chromium_revision b624ecb939..2bc3837c3d (675206:675350) by chromium-webrtc-autoroll · 5 years ago
  23. 5e25fac CroppingWindowCapturerWin: filter out cloaked window. by Julien Isorce · 5 years ago
  24. a0eefc1 Rename USE_NATIVE_MUTEX_ON_MAC to RTC_USE_NATIVE_MUTEX_ON_MAC. by Mirko Bonadei · 5 years ago
  25. 34462f5 Revert "Add ability to set RTCP sender ssrc at construction time" by Ilya Nikolaevskiy · 5 years ago
  26. 4a126e4 Rename tests to prevent clashing with old audio test by Artem Titov · 5 years ago
  27. 94c58fd Add ability to set RTCP sender ssrc at construction time by Erik Språng · 5 years ago
  28. a4d8737 Format almost everything. by Jonas Olsson · 5 years ago
  29. c93bfcf Roll chromium_revision 619b1dc518..b624ecb939 (675093:675206) by chromium-webrtc-autoroll · 5 years ago
  30. 48284b8 BalancedDegradationSettings: Add option to configure fps based on codec type. by Åsa Persson · 5 years ago
  31. 79e4c92 Remove bwe_rtp_play and add rtp_to_text to the build. by Mirko Bonadei · 5 years ago
  32. ab0b9d4 Switch rtc_event_log2rtp_dump to ABSL_FLAG. by Mirko Bonadei · 5 years ago
  33. eec86cd Fix platform-specific header dependencies to be more precise by Oleh Prypin · 5 years ago
  34. 86f8b3b Roll chromium_revision 48f88caf2e..619b1dc518 (674992:675093) by chromium-webrtc-autoroll · 5 years ago
  35. 9455c30 Remove deprecation notice for VideoFrame::ntp_time_ms by Ilya Nikolaevskiy · 5 years ago
  36. f6468d2 Wire up new PacedSender code path. by Erik Språng · 5 years ago
  37. 668ce0c Remove trial WebRTC-SimulcastMaxLayers and make its behavior default by Florent Castelli · 5 years ago
  38. ecae9cd Android: Add error callback for GL_OUT_OF_MEMORY in EglRenderer by Magnus Jedvert · 5 years ago
  39. 48b1b18 Add ability to create EmulatedNetworkNode from BuiltInNetworkBehaviorConfig by Artem Titov · 5 years ago
  40. c8263e0 Introduce PC level audio quality test. by Artem Titov · 5 years ago
  41. 386802e Move network emulation framework under test/network by Artem Titov · 5 years ago
  42. 13eb602 Roll chromium_revision 50acc956cd..48f88caf2e (674882:674992) by chromium-webrtc-autoroll · 5 years ago
  43. fdf74bd Remove non implemented function from WebRtcVideoChannel. by philipel · 5 years ago
  44. 656590d Roll chromium_revision 778353d874..50acc956cd (674780:674882) by chromium-webrtc-autoroll · 5 years ago
  45. 1efb4a2 Add field trial for forcing partition resilience mode in libvpx. by Rasmus Brandt · 5 years ago
  46. 3fbf1e2 Reduce kMaxSimulcastStreams to 3 by Florent Castelli · 5 years ago
  47. 9d96209 Switch unpack_aecdump to ABSL_FLAG. by Mirko Bonadei · 5 years ago
  48. 4580ca2 Reland: Add ability to set ssrcs of RtpSender at construction time by Erik Språng · 5 years ago
  49. 8e3cb53 Remove activity_metric tool. by Mirko Bonadei · 5 years ago
  50. 543179e Roll chromium_revision 79b588ee95..778353d874 (674667:674780) by chromium-webrtc-autoroll · 5 years ago
  51. 2250b05 Adding support for channel mixing between different channel layouts. by henrika · 5 years ago
  52. 3f2eeb8 Adding test on GetSpanSamples() for NetEq PacketBuffer. by Minyue Li · 5 years ago
  53. d2c336f [getStats] Implement "media-source" audio levels, fixing Chrome bug. by Henrik Boström · 5 years ago
  54. e8fbc5d Refactor WebRtcOpus_PacketHasFec. by Minyue Li · 5 years ago
  55. 67008df Revert "Replace the implementation of `GetContributingSources()` on the audio side." by Artem Titov · 5 years ago
  56. c6c730b Roll chromium_revision 54ec0ffb89..79b588ee95 (674538:674667) by chromium-webrtc-autoroll · 5 years ago
  57. 697f861 Roll chromium_revision 13b7238371..54ec0ffb89 (674397:674538) by chromium-webrtc-autoroll · 5 years ago
  58. 2e60217 Add speculative checks to RtpPacketHistory by Erik Språng · 5 years ago
  59. 46dda83 Improve buffer level estimation with DTX and add CNG time stretching. by Jakob Ivarsson · 5 years ago
  60. 3d642f8 Rename ..BitrateThresholds to ..BitrateLimits. by Sergey Silkin · 5 years ago
  61. cecee99 Disable VP9 2nd profile test for ios arm64 by Artem Titov · 5 years ago
  62. 51f599b Make rtc_base/base_java public. by Sami Kalliomäki · 5 years ago
  63. 81e1bf0 Remove using DegradationPreference from scenario_config.h by Artem Titov · 5 years ago
  64. 6542826 Add new tests with lossy networks on PC test framework by Artem Titov · 5 years ago
  65. cd8a6e2 Add writing and parsing of the `abs-capture-time` RTP header extension. by Chen Xing · 5 years ago
  66. 53d45ba Make TaskQueueFactory required construction parameter for Call by Danil Chapovalov · 5 years ago
  67. 84ce3c0 Macro rename s/CS_DEBUG_CHECKS/RTC_CS_DEBUG_CHECKS. by Mirko Bonadei · 5 years ago
  68. a2b30d8 Add functions to read from/write to bitstream values with known max value by Danil Chapovalov · 5 years ago
  69. 9eee121 Switch py_quality_assessment to ABSL_FLAG. by Mirko Bonadei · 5 years ago
  70. b60141b Save and serialize the receive RIDs in MediaContentDescription by Florent Castelli · 5 years ago
  71. e8ed830 WebRtcVideoChannel encoder fallback. by philipel · 5 years ago
  72. e420c6a Add missing include for memcpy/memcmp by Artem Titov · 5 years ago
  73. 6a2c1ba Roll chromium_revision ba17fd6b36..13b7238371 (674288:674397) by chromium-webrtc-autoroll · 5 years ago
  74. 8fa7151 Replace the implementation of `GetContributingSources()` on the audio side. by Chen Xing · 5 years ago
  75. 16661eb Fix: report video_bwe_stats as bytes per second, as specified in the unit by Artem Titov · 5 years ago
  76. 443b7ee Destroy existing encoder instance before creating a new one. by Sergey Silkin · 5 years ago
  77. 2c5af4f Add * and / operator into SamplesStatsCounter. by Artem Titov · 5 years ago
  78. 1d46f9c Add RtpPacket::IsExtensionReserved(). by Erik Språng · 5 years ago
  79. 6038926 Roll chromium_revision 35be8751a4..ba17fd6b36 (674036:674288) by chromium-webrtc-autoroll · 5 years ago
  80. 02d7d35 Revert "Add ability to set ssrcs of RtpSender at construction time" by Amit Hilbuch · 5 years ago
  81. c442197 Check the rid direction matches the direction in simulcast description by Florent Castelli · 5 years ago
  82. 238aab9 Fix bug in use_datagram_transport configuration. by Bjorn A Mellem · 5 years ago
  83. b073f1c Only set the RtcEventLog for media transport when it's used for media. by Bjorn A Mellem · 5 years ago
  84. 73bfc0e Roll chromium_revision 6f0434662d..35be8751a4 (673926:674036) by chromium-webrtc-autoroll · 5 years ago
  85. ed56cf4 Remove deprecated version of Vp8FrameBufferControllerFactory::Create by Elad Alon · 5 years ago
  86. e9d6e65 Add ability to set ssrcs of RtpSender at construction time by Erik Språng · 5 years ago
  87. 5ee6967 Don't reset encoder on max/min bitrate change. by Sergey Silkin · 5 years ago
  88. bc70b61 Switch rnn_vad_tool to ABSL_FLAG. by Mirko Bonadei · 5 years ago
  89. f1a7bb1 Stop using unnecessary gclient vars by Oleh Prypin · 5 years ago
  90. 45befc5 Pass FecControllerOverride to Vp8FrameBufferControllerFactory::Create by Elad Alon · 5 years ago
  91. 14be799 Switch neteq tools to ABSL_FLAG. by Mirko Bonadei · 5 years ago
  92. e731a2e Remove check on supported profile in favor of expilict disabling by Artem Titov · 5 years ago
  93. bfd343b Add totalDecodeTime to RTCInboundRTPStreamStats by Johannes Kron · 5 years ago
  94. 419aae2 Remove android_tools deps by Yun Liu · 5 years ago
  95. ebdf9f8 Roll chromium_revision 097ffaa18d..6f0434662d (673789:673926) by chromium-webrtc-autoroll · 5 years ago
  96. 6fdfec1 Add overload to CreateIceTransport that takes additional dependencies by Steve Anton · 5 years ago
  97. 1cf9470 Roll chromium_revision 05067e74f0..097ffaa18d (673689:673789) by chromium-webrtc-autoroll · 5 years ago
  98. 5985a04 Add a field trial to control datagram transport use. by Bjorn A Mellem · 5 years ago
  99. 3e8ef94 Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker. by Chen Xing · 5 years ago
  100. 62eb89d Fixing possible overflow in NetEq buffle level filter. by Minyue Li · 5 years ago