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gerrit-public.fairphone.software / platform / external / webrtc / 5932149c9aeaa7679ad0bc3183047766832ca907
  1. 5932149 Use different restrictions of acked bitrate lag depending on operating point. by stefan · 8 years ago
  2. a790d83 Wire up rtcp xr target bitrate on receive side. by sprang · 8 years ago
  3. 9afbfc0 Roll chromium_revision f50152dfc4..dd10720676 (435897:435924) by buildbot · 8 years ago
  4. 0f01c7f Added tool for reference less video analysis (go/refless-video-analysis) by charujain · 8 years ago
  5. b2250e5 New gn target video_frame_api. by nisse · 8 years ago
  6. 969b12f Remove xdisplaycheck by kjellander · 8 years ago
  7. df28e47 fix coding and documentary ambiguity in AimdRateControl::TimeToReduceFurther. by howtofly · 8 years ago
  8. a28a1b9 Disable P2PTestConductor.LocalP2PTestDtlsBundleInIceRestart under msan by henrik.lundin · 8 years ago
  9. 5c71166 VP8DecoderImpl: Fix uninitialized memory crash by magjed · 8 years ago
  10. 00bceb1 Deprecated SetAudioPacketSize from RTPSender and removed calls to it. by ossu · 8 years ago
  11. e066b30 Remove API-related #defines from voice_engine_configurations.h by henrik.lundin · 8 years ago
  12. 15cf9f9 Roll chromium_revision 601e4f48a3..f50152dfc4 (435870:435897) by buildbot · 8 years ago
  13. b336392 Sanity check parsed QP values from H264 bitstream by kthelgason · 8 years ago
  14. 864f58b Roll chromium_revision 353f713f3d..601e4f48a3 (435846:435870) by buildbot · 8 years ago
  15. ea3d923 Roll chromium_revision c6d437f401..353f713f3d (435797:435846) by buildbot · 8 years ago
  16. 51cb31c Roll chromium_revision 47f73d2d11..c6d437f401 (435727:435797) by buildbot · 8 years ago
  17. b465980 In end-to-end PeerConnection tests, allow video to be downscaled. by deadbeef · 8 years ago
  18. 897530e Roll chromium_revision bffc40ae72..47f73d2d11 (435660:435727) by buildbot · 8 years ago
  19. 8f89bff Revert of Disabled flaky P2PTestConductor tests on ASAN and MSAN. (patchset #1 id:1 of https://codereview.webrtc.org/2539103002/ ) by deadbeef · 8 years ago
  20. c6b6e09 Relaxing timeouts for TestMediaMonitor. by deadbeef · 8 years ago
  21. 8f425f9 Relaxing DCHECK for packets sent before SRTP is enabled. by deadbeef · 8 years ago
  22. 183d51a Roll chromium_revision 3045382d44..bffc40ae72 (435618:435660) by buildbot · 8 years ago
  23. f29e05d Add linearly spaced counting histograms by henrik.lundin · 8 years ago
  24. 1454669 Cleanup RtpHeaderExtensionMap removing use of two legacy functions by danilchap · 8 years ago
  25. 73f2ee2 Roll chromium_revision a6a331af3e..3045382d44 (435597:435618) by buildbot · 8 years ago
  26. 182e4a4 Remove bitrate cap for AdaptiveVideoSource and increase other caps to 25 Mbps. by terelius · 8 years ago
  27. 2d9c877 Revert of Whitespace. by tandrii · 8 years ago
  28. 106edae Revert of Whitespace. by Andrii Shyshkalov · 8 years ago
  29. aa7a9a2 Whitespace. by Andrii Shyshkalov · 8 years ago
  30. 3b56e1f Whitespace. by tandrii · 8 years ago
  31. 1a646ee Wire up BitrateAllocation to be sent as RTCP TargetBitrate by sprang · 8 years ago
  32. 5e38c96 Wire up RTCP XR target bitrate in rtp/rtcp module by sprang · 8 years ago
  33. b57e0f6 Roll chromium_revision 698c4714e8..a6a331af3e (435589:435597) by buildbot · 8 years ago
  34. 5e13d41 Remove limit on how often quality scaling downscales by kthelgason · 8 years ago
  35. 86cf9a2 Increase test timeout to combat flakiness. by kthelgason · 8 years ago
  36. e90adce Remove OnLocalSsrcChanged by mflodman · 8 years ago
  37. 847f294 Roll chromium_revision b5ce7e0a17..698c4714e8 (435129:435589) by buildbot · 8 years ago
  38. 665bc3c Move webrtc/api/androidtests to webrtc/sdk/android/instrumentationtests by magjed · 8 years ago
  39. 2bc324c Add method on AVFoundation capturer to adapt output format. by kthelgason · 8 years ago
  40. dd40702 Move VideoDecoder::Create() logic to separate internal video decoder factory by magjed · 8 years ago
  41. aa354c9 Rename full_stack.cc to full_stack_tests.cc. by brandtr · 8 years ago
  42. a974d76 Enable VideoToolbox encoder on mac by kthelgason · 8 years ago
  43. 024c90e Whitespace CL to trigger bots. by Henrik Kjellander · 8 years ago
  44. 3a70cc3 Whitespace change to trigger bots. by Henrik Kjellander · 8 years ago
  45. 5c13c33 Roll chromium_revision b66d8ae9dc..b5ce7e0a17 (435081:435129) by buildbot · 8 years ago
  46. ed6e077 Make SurfaceTextureHelper and I420Frame public in Java. by skvlad · 8 years ago
  47. a15948c Change assert to RTC_DCHECK in bwe_test_logging.cc by terelius · 8 years ago
  48. 3a9bc17 Allow custom drawers to be added to framelisteners. by sakal · 8 years ago
  49. 06c1d64 Prep to remove API-related #defines from voice_engine_configurations.h by henrik.lundin · 8 years ago
  50. 9332b7d Reland "Update rtt on audio only calls". by michaelt · 8 years ago
  51. 93c5d03 Start gathering perf data for VP8 + FlexFEC. by brandtr · 8 years ago
  52. 13d38fb Delete all of the video_processing module but the denoiser code. by nisse · 8 years ago
  53. 13f1a0a Wire up x-google-{min,start,max}-bitrate to WebRtcVoiceMediaChannel. by stefan · 8 years ago
  54. 127387e Delete nalu parser in mediaencoder by kthelgason · 8 years ago
  55. 2305b70 Remove unused build override by kthelgason · 8 years ago
  56. cb861e0 Templatize percentile_filter.h and move it to base/analytics. by terelius · 8 years ago
  57. 4b9a2cb Reland "Renaming AudioEncoder::SetTargetBitrate and SetProjectedPacketLossRate." by minyue · 8 years ago
  58. 26bddb9 Replace test_support_main by test_main and get rid of test_support_main_threaded_mac by ehmaldonado · 8 years ago
  59. 18f7c8d Remove warning suppression from VideoToolboxEncoder by kthelgason · 8 years ago
  60. 076c011 Change unit of logged bitrate stats in bytes/s to bits/s. by asapersson · 8 years ago
  61. aff9636 Greatly reduce number of level controller tests. by phoglund · 8 years ago
  62. 6a2e20a Make sure GetLastError on a PlatformThread return an error that is relevant to the thread. by perkj · 8 years ago
  63. b49fc14 RtpDataEngine, FindCodecByName: Don't reassign codecs by magjed · 8 years ago
  64. 998df1d Create webrtc/sdk/android folder by magjed · 8 years ago
  65. 78b4d56 Relanding "Pass time constant to bwe smoothing filter." by minyue · 8 years ago
  66. 8d66a5a Disabled flaky P2PTestConductor tests on ASAN and MSAN. by ossu · 8 years ago
  67. 0245da0 Move ownership of PacketRouter from CongestionController to Call. by nisse · 8 years ago
  68. 06251f1 Reduce ProbeController::kDefaultMaxProbingBitrateBps to 10 mbps. by philipel · 8 years ago
  69. f4a5942 Disabled all ScreenCapturerIntegrationTests on Windows by ossu · 8 years ago
  70. 3a864d2 MB: Add swarming bots in the FYI waterfall and remove memcheck swarming. by ehmaldonado · 8 years ago
  71. b7e1dd7 Revert of Adding memcheck suppression. (patchset #1 id:1 of https://codereview.webrtc.org/2537563003/ ) by philipel · 8 years ago
  72. 706a45e Added missing include to fix waterfall compile error. by hbos · 8 years ago
  73. f15a2c5 Delete deprecated versions of Copy, ScaleFrom and CropAndScaleFrom. by nisse · 8 years ago
  74. 0583b28 Collecting RTCIceCandidatePairStats.transport_id and improved unittests. by hbos · 8 years ago
  75. 0c43f77 Update video histograms that do not have a minimum lifetime limit before being recorded. by asapersson · 8 years ago
  76. 759e0b7 Fix memory leak in video_coding::PacketBuffer::InsertPacket. by philipel · 8 years ago
  77. be74270 Calculate JitterBufferDelayInMs in the new jitter buffer. by philipel · 8 years ago
  78. e69b468 Revert of Renaming AudioEncoder::SetTargetBitrate and SetProjectedPacketLossRate. (patchset #5 id:240001 of https://codereview.webrtc.org/2411613002/ ) by minyue · 8 years ago
  79. 1731c9c Use swap instead of copy in RtcHistogram::GetAndReset. by asapersson · 8 years ago
  80. 84e56d5 Renaming AudioEncoder::SetTargetBitrate and SetProjectedPacketLossRate. by minyue · 8 years ago
  81. 097529f Remove 3 defines in voice_engine_configurations.h by henrik.lundin · 8 years ago
  82. e61fbff Use RotateDesktopFrame in DirectX capturer by zijiehe · 8 years ago
  83. 166e59a Enable ScreenCapturerIntegrationTests by zijiehe · 8 years ago
  84. 6a46cf7 Roll chromium_revision db14e1adbd..b66d8ae9dc (435041:435081) by buildbot · 8 years ago
  85. c9e80ee Adding packet overhead to audio network adaptor. by minyue · 8 years ago
  86. 821dc7a Roll chromium_revision 683745f53c..db14e1adbd (434997:435041) by buildbot · 8 years ago
  87. a332877 Remove overhead from video bitrate. by michaelt · 8 years ago
  88. c4dc4a5 Adding RTCStatsIntegrationTest to memcheck supressions. by deadbeef · 8 years ago
  89. 75f9d8c Roll chromium_revision ffe8e7b51d..683745f53c (434954:434997) by buildbot · 8 years ago
  90. 290d43a Add a new UMA metric in APM to track incoming capture-side audio level by henrik.lundin · 8 years ago
  91. 939e08f Added webrtc/audio/utility directory and empty GN target. by aleloi · 8 years ago
  92. ee414d9 Added sanity check to VCMDecodingState::UsingFlexibleMode to prevent OOB error. by philipel · 8 years ago
  93. ad6f646 Use //build/dotfile_settings.gni to reduce blocked auto-rolls by kjellander · 8 years ago
  94. 768d625 Fix spelling mistake in RTP module declaration. by brandtr · 8 years ago
  95. b890c95c Replace some asserts with DCHECKs by kwiberg · 8 years ago
  96. 5049942 Refactor RMSLevel and give it new functionality by henrik.lundin · 8 years ago
  97. 1308c69 Roll chromium_revision 0496be2799..ffe8e7b51d (434847:434954) by buildbot · 8 years ago
  98. f17cae2 Cleanup unused rules in webrtc/DEPS + add kjellander to OWNERS for it by kjellander · 8 years ago
  99. 668eb3b Add overhead to transport feedback observer. by michaelt · 8 years ago
  100. 19223ac Ignore newly added resource files. by charujain · 8 years ago