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gerrit-public.fairphone.software / fp2-dev / platform / external / chromium_org / third_party / webrtc / ace0823ddb33a8970e285c1be1722b51d385802f / video_engine
  1. ace0823 Enabling bufffering mode with no sync module or VoE by mikhal@webrtc.org · 11 years ago
  2. 87d8f2d Updated version number to 3.25 by elham@webrtc.org · 11 years ago
  3. 3da576e Update integration tests for idempotent RTP header settings. by bemasc@google.com · 11 years ago
  4. 1dcba31 Destroy VCM and VPM instead of delete. by mflodman@webrtc.org · 11 years ago
  5. ca65c51 Handle multiple calls to set initial delay by mikhal@webrtc.org · 11 years ago
  6. 213217c Stop and restart fix. by mflodman@webrtc.org · 11 years ago
  7. 2325284 Fixed typo in vie_autotest_loopback.cc. by pbos@webrtc.org · 11 years ago
  8. cb139b1 Rename webrtc::StatsObserver to webrtc::CallStatsObserver by fischman@webrtc.org · 11 years ago
  9. 432bc1a fixing nack list size calculation by mikhal@webrtc.org · 11 years ago
  10. 39eb955 Updated version number to 3.24 by elham@webrtc.org · 11 years ago
  11. 85e2e0e Refactoring temporal layers implementation and adding VideoCodecMode for easier control of codec settings. by stefan@webrtc.org · 11 years ago
  12. ce3f2ca Add VoE interface to VieRTP test by mikhal@webrtc.org · 11 years ago
  13. 4db69af Adding a receive side API for buffering mode. by mikhal@webrtc.org · 11 years ago
  14. 64506e2 Roll Chromium revision 176094:182149 by kjellander@webrtc.org · 11 years ago
  15. e740a7b Remove MultiStreamMode from test. by stefan@webrtc.org · 11 years ago
  16. 4c6689a Reset ssrc when calling SetSendCodec. by mflodman@webrtc.org · 11 years ago
  17. 33c6e92 Sync libvpx and its gyp wrapper from Chromium. by andrew@webrtc.org · 11 years ago
  18. 1fb8372 Increase maximum resolution to 4k x 3k. by fbarchard@google.com · 11 years ago
  19. 9c4707e Android NDK build tools by kjellander@webrtc.org · 11 years ago
  20. 4da62e0 Set SingleStream BWE in unittests. by stefan@webrtc.org · 11 years ago
  21. 6cd34e5 Updates to send side streaming mode: by mikhal@webrtc.org · 11 years ago
  22. 6bcf2ab Update version number to 3.23 by tnakamura@webrtc.org · 11 years ago
  23. 75e6669 Made it possible to render custom call output to file. by phoglund@webrtc.org · 11 years ago
  24. 89c3de3 Don't report an error for GetEstimatedReceiveBandwidth if there is no valid by mflodman@webrtc.org · 11 years ago
  25. 34d1110 Enable indefinitely running vie_auto_test option by kjellander@webrtc.org · 11 years ago
  26. db325e2 Updated version number to 3.22 by elham@webrtc.org · 11 years ago
  27. cc895d1 Fixing/disabling Windows x64 warnings by kjellander@webrtc.org · 11 years ago
  28. d6739c8 Adding a send side API for streaming by mikhal@webrtc.org · 11 years ago
  29. a7761c7 Fix mismatch between different NACK list lengths and packet buffers. by stefan@webrtc.org · 11 years ago
  30. 3442158 Break out RemoteBitrateEstimator from RtpRtcp module and make RemoteBitrateEstimator::Process trigger new REMB messages. by stefan@webrtc.org · 11 years ago
  31. fd2dd1a Set frame length for frame converting in external renderer by braveyao@webrtc.org · 11 years ago
  32. 8d759af VP8: Making key frame interval a tunnable parameter by mikhal@webrtc.org · 11 years ago
  33. b4575c1 Fix webrtc compilation errors for Chrome Win64 by andrew@webrtc.org · 11 years ago
  34. ceca869 Moving ViE test files and deleting files no longer used. by mflodman@webrtc.org · 12 years ago
  35. 3d7848b Updated version number to 3.21 by elham@webrtc.org · 12 years ago
  36. 81cfcb5 Remove '<(library)' in gyp files. by wjia@webrtc.org · 12 years ago
  37. fc37398 Convert psnr and ssim to strings before printing them. by stefan@webrtc.org · 12 years ago
  38. 087c593 Removing outdated comment by mikhal@webrtc.org · 12 years ago
  39. 32ad4a4 Made ViEToFileRenderer use a separate thread for rendering frames to file. by stefan@webrtc.org · 12 years ago
  40. 1b23416 logical 'and' of mutually exclusive tests is always false in ViECodecImpl::CodecValid() by braveyao@webrtc.org · 12 years ago
  41. ee92f9d Disable full stack PSNR/SSIM triggers on Mac and Win for now due to flakiness. Adding plots of PSNR and SSIM. by stefan@webrtc.org · 12 years ago
  42. 1d4568f Disable PSNR/SSIM thresholds for the Gilber-Elliot test. by stefan@webrtc.org · 12 years ago
  43. b011c6a Generalized mechanism for excluding gtests on platforms, disabled broken tests on mac. by phoglund@webrtc.org · 12 years ago
  44. c702d28 Disabled GQoS since it breaks ViE auto test. by henrika@webrtc.org · 12 years ago
  45. 8be968f Enable external encoders with internal picture source. by stefan@webrtc.org · 12 years ago
  46. e91de87 Using Convert in lieu of ExtractBuffer: Less error prone (as we don't need to compute buffer sizes etc.). This cl is first in a series (doing all of WebRtc would make it quite a big cl). While at it, fixing a few headers. by mikhal@webrtc.org · 12 years ago
  47. 8be556d Updated version number to 3.20 by elham@webrtc.org · 12 years ago
  48. 5e22650 Removed spaces from full stack test labels, consolidated graphs by phoglund@webrtc.org · 12 years ago
  49. c54e675 Changed assert to log. by mflodman@webrtc.org · 12 years ago
  50. 42264f2 Make protection method, filename and resolution configurable for FullStackTest. by stefan@webrtc.org · 12 years ago
  51. c302ff2 vie auto test: Adding a constructor for NetworkParameters by mikhal@webrtc.org · 12 years ago
  52. b8029db ViE autotest: Adding loss models to the external transport by mikhal@webrtc.org · 12 years ago
  53. b4e5d10 Updated version number to 3.19 by elham@webrtc.org · 12 years ago
  54. b36efe3 Added API to get receive side video delay. by mflodman@webrtc.org · 12 years ago
  55. da80bad Remove latency excl network and add render time diff stats. by stefan@webrtc.org · 12 years ago
  56. 2188300 Fix for buffer overflow, WebRTC issue 1196 by elham@webrtc.org · 12 years ago
  57. 4c8b31e Added jitter to fake network pipe. by mflodman@webrtc.org · 12 years ago
  58. 2b6c051 Track the actual render time rather than the decode time. by stefan@webrtc.org · 12 years ago
  59. f0bf6f6 Will now only require near-perfect PSNR and SSIM. by phoglund@webrtc.org · 12 years ago
  60. 7940bbb Revert 3269 by andrew@webrtc.org · 12 years ago
  61. 6db19bd Will now only require near-perfect PSNR and SSIM. by phoglund@webrtc.org · 12 years ago
  62. c1b0f7d Use TRACE_EVENT to track time spent in VP8 encoding by hclam@chromium.org · 12 years ago
  63. fb537e2 Add a third full stack test and support for random jitter in ext transport. by stefan@webrtc.org · 12 years ago
  64. 2082b3a Adding a simple fake network pipe to use for testing. Next CL will contain an external transport implementation using this link and I'll follow up later making this more advanced. by mflodman@webrtc.org · 12 years ago
  65. 3e5b30b Add more audio codec information into codec list by leozwang@webrtc.org · 12 years ago
  66. 9470f64 Added auto-call feature to WebRTCDemo. by fischman@webrtc.org · 12 years ago
  67. e5a2710 Adds two full stack performance metrics for end-to-end delay. by stefan@webrtc.org · 12 years ago
  68. 0cf911a First pass of MediaCodecDecoder which uses Android MediaCodec API. by dwkang@webrtc.org · 12 years ago
  69. 3f4b16d Delete {start,stop}CPULoad() since they're broken. by fischman@webrtc.org · 12 years ago
  70. bc23d31 Enable building WebRTCDemo apk using Release webrtc libs, take 2. by fischman@webrtc.org · 12 years ago
  71. 8adceb0 Corrected .h path. by phoglund@webrtc.org · 12 years ago
  72. a7920db Fixed standard PSNR/SSIM test. by phoglund@webrtc.org · 12 years ago
  73. 2255427 Properly remove the bitrate observer when ViEEncoder is destructed. by stefan@webrtc.org · 12 years ago
  74. 0c9d201 Disabled some more flaky tests. Memcheck vie_auto_test should be very stable after this. by phoglund@webrtc.org · 12 years ago
  75. 4bbb260 Revert 3190 - Enable building WebRTCDemo apk using Release webrtc libs. by fischman@webrtc.org · 12 years ago
  76. 4a2fab0 Enable building WebRTCDemo apk using Release webrtc libs. by fischman@webrtc.org · 12 years ago
  77. 5dea525 Remove ringtone from test app by leozwang@webrtc.org · 12 years ago
  78. 4591d9b Fixing vie and voe auto test project paths for test execution. by kjellander@webrtc.org · 12 years ago
  79. 73d3490 Updated version number to 3.18 by elham@webrtc.org · 12 years ago
  80. 6318790 Wire up CallStats to provide modules with correct RTT. by mflodman@webrtc.org · 12 years ago
  81. 0993f8b Fixes (or at least reduces) the flakiness in the full stack test by making sure the different frame monitors are registered and deregistered in the right order. Also makes sure only local preview frames which are actually transmitted are rendered by moving the local preview rendering to an effect filter. by stefan@webrtc.org · 12 years ago
  82. 1ec1bc9 Removed codec comparison test: it didn't work and probably never will. by phoglund@webrtc.org · 12 years ago
  83. b743278 Remove ViE lint warnings that should have been caught at upload time. by mflodman@webrtc.org · 12 years ago
  84. a39ac68 Reorganize gyp for Android by leozwang@webrtc.org · 12 years ago
  85. 020b350 Fix possible race condition and access into an empty list. by stefan@webrtc.org · 12 years ago
  86. d51d166 Move SSRC list to RemoteBitrateEstimator. by stefan@webrtc.org · 12 years ago
  87. 8a8517a Adding ViE CallStats to keep track of call statistics. As a start, only rtt is handled. by mflodman@webrtc.org · 12 years ago
  88. 36fdd24 Replaced remb unittest sleep with fake clock. by mflodman@webrtc.org · 12 years ago
  89. 8dd4b98 Revert 3111 (revert of a revert). by tommi@webrtc.org · 12 years ago
  90. b159bd8 Revert 3105 - Don't crash the unit test host when tests fail. by mikhal@webrtc.org · 12 years ago
  91. 6be5b2f Don't crash the unit test host when tests fail. by tommi@webrtc.org · 12 years ago
  92. c06c66d Fixed test memory leak + disabled base test. by phoglund@webrtc.org · 12 years ago
  93. 68783ae Add libpaced_sender to Android makefile by leozwang@webrtc.org · 12 years ago
  94. 32f05a7 Enable paced sender. Review URL: https://webrtc-codereview.appspot.com/965016 by pwestin@webrtc.org · 12 years ago
  95. c4f9c04 Fixes an incorrect if statement in vie_sync_module.cc. by stefan@webrtc.org · 12 years ago
  96. b95093a Re-initialize enough state on "Stop Call" to be able to stop/start multiple calls in succession. by fischman@webrtc.org · 12 years ago
  97. 3f78c6c Add Android OWNER files by leozwang@webrtc.org · 12 years ago
  98. 7121008 Can now fully control custom calls from the command line. by phoglund@webrtc.org · 12 years ago
  99. 215428c Adding codecType to OnIncomingCapturedEncodedFrame partially reverting r3013. by mikhal@webrtc.org · 12 years ago
  100. bf4bba9 Made TickTime immutable, rewrote tick utils to be fakeable. by phoglund@webrtc.org · 12 years ago