1. 32ad4a4 Made ViEToFileRenderer use a separate thread for rendering frames to file. by stefan@webrtc.org · 12 years ago
  2. 1b23416 logical 'and' of mutually exclusive tests is always false in ViECodecImpl::CodecValid() by braveyao@webrtc.org · 12 years ago
  3. 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
  4. 1d4568f Disable PSNR/SSIM thresholds for the Gilber-Elliot test. by stefan@webrtc.org · 12 years ago
  5. b011c6a Generalized mechanism for excluding gtests on platforms, disabled broken tests on mac. by phoglund@webrtc.org · 12 years ago
  6. c702d28 Disabled GQoS since it breaks ViE auto test. by henrika@webrtc.org · 12 years ago
  7. 8be968f Enable external encoders with internal picture source. by stefan@webrtc.org · 12 years ago
  8. 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
  9. 8be556d Updated version number to 3.20 by elham@webrtc.org · 12 years ago
  10. 5e22650 Removed spaces from full stack test labels, consolidated graphs by phoglund@webrtc.org · 12 years ago
  11. c54e675 Changed assert to log. by mflodman@webrtc.org · 12 years ago
  12. 42264f2 Make protection method, filename and resolution configurable for FullStackTest. by stefan@webrtc.org · 12 years ago
  13. c302ff2 vie auto test: Adding a constructor for NetworkParameters by mikhal@webrtc.org · 12 years ago
  14. b8029db ViE autotest: Adding loss models to the external transport by mikhal@webrtc.org · 12 years ago
  15. b4e5d10 Updated version number to 3.19 by elham@webrtc.org · 12 years ago
  16. b36efe3 Added API to get receive side video delay. by mflodman@webrtc.org · 12 years ago
  17. da80bad Remove latency excl network and add render time diff stats. by stefan@webrtc.org · 12 years ago
  18. 2188300 Fix for buffer overflow, WebRTC issue 1196 by elham@webrtc.org · 12 years ago
  19. 4c8b31e Added jitter to fake network pipe. by mflodman@webrtc.org · 12 years ago
  20. 2b6c051 Track the actual render time rather than the decode time. by stefan@webrtc.org · 12 years ago
  21. f0bf6f6 Will now only require near-perfect PSNR and SSIM. by phoglund@webrtc.org · 12 years ago
  22. 7940bbb Revert 3269 by andrew@webrtc.org · 12 years ago
  23. 6db19bd Will now only require near-perfect PSNR and SSIM. by phoglund@webrtc.org · 12 years ago
  24. c1b0f7d Use TRACE_EVENT to track time spent in VP8 encoding by hclam@chromium.org · 12 years ago
  25. fb537e2 Add a third full stack test and support for random jitter in ext transport. by stefan@webrtc.org · 12 years ago
  26. 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
  27. 3e5b30b Add more audio codec information into codec list by leozwang@webrtc.org · 12 years ago
  28. 9470f64 Added auto-call feature to WebRTCDemo. by fischman@webrtc.org · 12 years ago
  29. e5a2710 Adds two full stack performance metrics for end-to-end delay. by stefan@webrtc.org · 12 years ago
  30. 0cf911a First pass of MediaCodecDecoder which uses Android MediaCodec API. by dwkang@webrtc.org · 12 years ago
  31. 3f4b16d Delete {start,stop}CPULoad() since they're broken. by fischman@webrtc.org · 12 years ago
  32. bc23d31 Enable building WebRTCDemo apk using Release webrtc libs, take 2. by fischman@webrtc.org · 12 years ago
  33. 8adceb0 Corrected .h path. by phoglund@webrtc.org · 12 years ago
  34. a7920db Fixed standard PSNR/SSIM test. by phoglund@webrtc.org · 12 years ago
  35. 2255427 Properly remove the bitrate observer when ViEEncoder is destructed. by stefan@webrtc.org · 12 years ago
  36. 0c9d201 Disabled some more flaky tests. Memcheck vie_auto_test should be very stable after this. by phoglund@webrtc.org · 12 years ago
  37. 4bbb260 Revert 3190 - Enable building WebRTCDemo apk using Release webrtc libs. by fischman@webrtc.org · 12 years ago
  38. 4a2fab0 Enable building WebRTCDemo apk using Release webrtc libs. by fischman@webrtc.org · 12 years ago
  39. 5dea525 Remove ringtone from test app by leozwang@webrtc.org · 12 years ago
  40. 4591d9b Fixing vie and voe auto test project paths for test execution. by kjellander@webrtc.org · 12 years ago
  41. 73d3490 Updated version number to 3.18 by elham@webrtc.org · 12 years ago
  42. 6318790 Wire up CallStats to provide modules with correct RTT. by mflodman@webrtc.org · 12 years ago
  43. 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
  44. 1ec1bc9 Removed codec comparison test: it didn't work and probably never will. by phoglund@webrtc.org · 12 years ago
  45. b743278 Remove ViE lint warnings that should have been caught at upload time. by mflodman@webrtc.org · 12 years ago
  46. a39ac68 Reorganize gyp for Android by leozwang@webrtc.org · 12 years ago
  47. 020b350 Fix possible race condition and access into an empty list. by stefan@webrtc.org · 12 years ago
  48. d51d166 Move SSRC list to RemoteBitrateEstimator. by stefan@webrtc.org · 12 years ago
  49. 8a8517a Adding ViE CallStats to keep track of call statistics. As a start, only rtt is handled. by mflodman@webrtc.org · 12 years ago
  50. 36fdd24 Replaced remb unittest sleep with fake clock. by mflodman@webrtc.org · 12 years ago
  51. 8dd4b98 Revert 3111 (revert of a revert). by tommi@webrtc.org · 12 years ago
  52. b159bd8 Revert 3105 - Don't crash the unit test host when tests fail. by mikhal@webrtc.org · 12 years ago
  53. 6be5b2f Don't crash the unit test host when tests fail. by tommi@webrtc.org · 12 years ago
  54. c06c66d Fixed test memory leak + disabled base test. by phoglund@webrtc.org · 12 years ago
  55. 68783ae Add libpaced_sender to Android makefile by leozwang@webrtc.org · 12 years ago
  56. 32f05a7 Enable paced sender. Review URL: https://webrtc-codereview.appspot.com/965016 by pwestin@webrtc.org · 12 years ago
  57. c4f9c04 Fixes an incorrect if statement in vie_sync_module.cc. by stefan@webrtc.org · 12 years ago
  58. 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
  59. 3f78c6c Add Android OWNER files by leozwang@webrtc.org · 12 years ago
  60. 7121008 Can now fully control custom calls from the command line. by phoglund@webrtc.org · 12 years ago
  61. 215428c Adding codecType to OnIncomingCapturedEncodedFrame partially reverting r3013. by mikhal@webrtc.org · 12 years ago
  62. bf4bba9 Made TickTime immutable, rewrote tick utils to be fakeable. by phoglund@webrtc.org · 12 years ago
  63. 77085af Removed ViEBaseObserver. by mflodman@webrtc.org · 12 years ago
  64. 0558a74 Fix for webrtc issue 1052 on windows with vie_auto_test. by vikasmarwaha@webrtc.org · 12 years ago
  65. 42c2116 Landing http://review.webrtc.org/914006/ by niklas.enbom@webrtc.org · 12 years ago
  66. f6cc3b7 Fixes a bitrate mismatch between sender and receiver. by stefan@webrtc.org · 12 years ago
  67. b43b611 Reorganize modules/video_render. by andrew@webrtc.org · 12 years ago
  68. fea2238 Fix Android build after video_capture reorg. by andrew@webrtc.org · 12 years ago
  69. 5f6856f Reorganize modules/video_capture. by andrew@webrtc.org · 12 years ago
  70. 904b81f Init capturePicture with GetCaptureDeviceSnapshot so that the SetRenderStartImage test won't depend on the previous test which may be disabled by the include_timing_dependent_tests flag. This is a fix for LinuxLargeTests. by wu@webrtc.org · 12 years ago
  71. 6baede5 Check if opus exists when build test app on Android by leozwang@webrtc.org · 12 years ago
  72. 25d9ea0 Fix uninitialzed memory and cleanup. by pwestin@webrtc.org · 12 years ago
  73. eb4840f Removing codecType from capture API by mikhal@webrtc.org · 12 years ago
  74. 1e25690 Enable Opus by leozwang@webrtc.org · 12 years ago
  75. f6547d3 Only remove encoder state feedback for send channels. by mflodman@webrtc.org · 12 years ago
  76. be86bb6 Revert the revert in r2988 since that wasn't the issue. by mflodman@webrtc.org · 12 years ago
  77. f5197ca Reverse Merged r2884 & r2888 from trunk. by vikasmarwaha@webrtc.org · 12 years ago
  78. dc7e6cf Switching to I420VideoFrame by mikhal@webrtc.org · 12 years ago
  79. 20dfb6e Update version to 3.15. by vikasmarwaha@webrtc.org · 12 years ago
  80. 4b3e611 Change android NDK library path by leozwang@webrtc.org · 12 years ago
  81. fae3665 Fix file path in vie_auto_test. by andrew@webrtc.org · 12 years ago
  82. a7b57da Move src/ -> webrtc/ by andrew@webrtc.org · 12 years ago