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gerrit-public.fairphone.software / platform / external / webrtc / e423a9de93082049029634081b5d7012521db573 / test
  1. af8659a Rename test output to test artifacts. by Edward Lemur · 7 years ago
  2. fc3a2e3 Remove the VoiceEngineObserver callback interface. by solenberg · 7 years ago
  3. fbbba3f Remove remaining mentions of gflags by oprypin · 7 years ago
  4. 5ab6854 Revert "Remove remaining mentions of gflags" by Oleh Prypin · 7 years ago
  5. 90ce84e Remove remaining mentions of gflags by Oleh Prypin · 7 years ago
  6. 2397b9a Remove voe::OutputMixer and AudioConferenceMixer. by solenberg · 7 years ago
  7. 2c30120 Revert of Add full stack tests for MediaCodec. (patchset #10 id:180001 of https://codereview.webrtc.org/3005253002/ ) by brandtr · 7 years ago
  8. 2cefac6 Add full stack tests for MediaCodec encoder. by brandtr · 7 years ago
  9. 7cd28b9 Set protected_by_flexfec flag properly in tests. by brandtr · 7 years ago
  10. 946d886 Remove VoENetwork by solenberg · 7 years ago
  11. dd3abbb Remove VoERTP_RTCP. by solenberg · 7 years ago
  12. 6dc2038 Remove VoECodec. by solenberg · 7 years ago
  13. cb728ea Fix Gn Untracked headers in webrtc/modules/video_coding. by charujain · 7 years ago
  14. b63310a Remove VoEFile and things it uses. by solenberg · 7 years ago
  15. 35dee81 Clean out unused methods from VoiceEngine and VoEBase. by solenberg · 7 years ago
  16. 18f5427 Remove voe_auto_test and add new tests to cover the missing cases. by solenberg · 7 years ago
  17. 7120742 Adding NOLINT for typedefs.h and common_types.h by Mirko Bonadei · 7 years ago
  18. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  19. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago
  20. 81cf5e4 Move test to src/test. by andrew@webrtc.org · 13 years ago
  21. f1d6e0a Removed the obsolete sanity check and added new test HTML files. by phoglund@webrtc.org · 13 years ago
  22. 9dc45da Move trunk/test/data -> trunk/data by andrew@webrtc.org · 13 years ago
  23. 22bde08 Made sanity check more flexible. by phoglund@webrtc.org · 13 years ago
  24. 4aa57b4 Extracted a helper library from vie_auto_test. by phoglund@webrtc.org · 13 years ago
  25. 4530aa3 Updates html test file to webkitDeprecatedPeerConnection. by henrikg@webrtc.org · 13 years ago
  26. 61bf8e3 Flush far-end buffers when larger than system delay. by andrew@webrtc.org · 13 years ago
  27. 754626b Fixed the sanity_check and started using the new webrtc_test.html file. Added capability for xvfb testing. by phoglund@webrtc.org · 13 years ago
  28. 50d9e26 Adds autoconnect and autocall functionality to web test page. by henrikg@webrtc.org · 13 years ago
  29. 29fafef Fix building errors Review URL: https://webrtc-codereview.appspot.com/399012 by leozwang@webrtc.org · 13 years ago
  30. 51198f1 More PRESUBMIT checks. by kjellander@webrtc.org · 13 years ago
  31. 0a57aae Converted old jpeg_test tool to gtest unit test. by kjellander@webrtc.org · 13 years ago
  32. cf6a295 Making video codecs test framework integration test execute in a reproducable fashion. by kjellander@webrtc.org · 13 years ago
  33. 9d9ad88 Fixed remaining warnings. by phoglund@webrtc.org · 13 years ago
  34. daacee8 Use better reference files with audioproc_unittest. by andrew@webrtc.org · 13 years ago
  35. c80d9d9 Removed default cases causing clang errors, -Wcovered-switch-default. by mflodman@webrtc.org · 13 years ago
  36. fede80c Updated test web page info for PeerConnection v2. by henrikg@webrtc.org · 13 years ago
  37. 6a81475 Removing year range in copyright statement in test web page. by henrikg@webrtc.org · 13 years ago
  38. 16a0427 Updates for web test page. by henrikg@webrtc.org · 13 years ago
  39. 12cccdd NS-SWB: Actived SWB processing at once, i.e., no startup phase. by bjornv@webrtc.org · 13 years ago
  40. 267b877 Add possibility to set HTML element values (e.g. server and name) in the URL for the test web page. by henrikg@webrtc.org · 13 years ago
  41. cc33737 Changing all PSNR/SSIM calculations to use libyuv. by kjellander@webrtc.org · 13 years ago
  42. 70adcd4 Delay estimator improvements. by bjornv@webrtc.org · 13 years ago
  43. 7270a6b Merged apm-buffer branch [r1293] back to trunk. by bjornv@webrtc.org · 13 years ago
  44. 173b7bb Integration test that tracks dropped frames and compares video output. by kjellander@webrtc.org · 13 years ago
  45. 5b97b12 Splitted FileHandler into FrameReader and FrameWriter classes and moved them to testsupport in test.gyp. by kjellander@webrtc.org · 13 years ago
  46. 80b2661 Fixing invalid check for existing file. by kjellander@webrtc.org · 13 years ago
  47. 4ed4f24 New fileutils.h method for managing resources on different platforms by kjellander@webrtc.org · 13 years ago
  48. 82d91ae Fixing crash when calculating SSIM and PSNR with empty video files in video_metrics.cc by kjellander@webrtc.org · 13 years ago
  49. a919d3a Don't return a zero delay with insufficient data. by andrew@webrtc.org · 13 years ago
  50. 5483210 Fixed open file handle in fileutils.cc Thanks Henrik L for pointing this out. by kjellander@webrtc.org · 13 years ago
  51. 91617ff by henrikg@webrtc.org · 13 years ago
  52. d0e5b96 Fix Amy's email address. by andrew@webrtc.org · 13 years ago
  53. 755b04a Add RMS computation for the RTP level indicator. by andrew@webrtc.org · 13 years ago
  54. 0db7dc6 Add file-playing channels to voe_cmd_test. by andrew@webrtc.org · 13 years ago
  55. 9b18ed6 Removed incorrect dependency. by phoglund@webrtc.org · 13 years ago
  56. 1144ba2 Base and codec tests now run verify output and render to file instead of to screen. by phoglund@webrtc.org · 13 years ago
  57. 62e48eb adding owners for test by niklas.enbom@webrtc.org · 13 years ago
  58. 4d8cd9d Adding GetOutputDir method to test_support library. by kjellander@webrtc.org · 13 years ago
  59. 20a370e Changing the namespace of TestSuite to webrtc::test. by kjellander@webrtc.org · 13 years ago
  60. 1a8d08a Changing usage of gtest_main target, to use test_support_main instead. by kjellander@webrtc.org · 13 years ago
  61. 1e10bb3 Remove global std::strings from fileutils. by andrew@webrtc.org · 13 years ago
  62. 5b5c31d Update fixed point audio processing output. by andrew@webrtc.org · 13 years ago
  63. 4c63676 Updated the AEC delay logging to output values in ms. PB output updated. by bjornv@webrtc.org · 13 years ago
  64. e698eb7 Make the sanity check test a little more robust, and add a README file. by hta@webrtc.org · 13 years ago
  65. a59d80d Updated fixed point output data file after changes in nsx. Verified bitexactness before that CL and the CLs afterwards towards the new file. by bjornv@webrtc.org · 13 years ago
  66. 7951e81 Simple utility method for finding the project root dir (to be used by tests loading resource files) by kjellander@webrtc.org · 13 years ago
  67. 1ba3dbe Adds possibility to log delay estimates in AEC. by bjornv@google.com · 13 years ago
  68. e90265b Commit http://webrtc-codereview.appspot.com/191001/ by tommi@webrtc.org · 13 years ago
  69. 19eefdc Add a unit testing framework. by andrew@webrtc.org · 13 years ago
  70. 9f710d0 Switch to new sqrt in NetEQ by henrik.lundin@webrtc.org · 13 years ago
  71. 35dcc23 Adding regression test to NetEQ by henrik.lundin@webrtc.org · 13 years ago
  72. af931bd Update of iLBC reference files for version 1.1.1, new SQRT. by tina.legrand@webrtc.org · 13 years ago
  73. 5daeae2 Update fixed profile data due to AECM sqrt change (no presubmit). by andrew@webrtc.org · 13 years ago
  74. 325bca7 Add unit test output. webrtc r319, ran on Xoom, synced source code on 8/8. by leozwang@google.com · 13 years ago
  75. 14acdbc Update fixed-point profile output due to r313. by andrew@webrtc.org · 13 years ago
  76. 59e4140 Add a fixed-point profile to the APM unit test. by ajm@google.com · 13 years ago
  77. a769fa5 Adding more output data checks to APM unittest. Blowing out the protobuf definition (changing the tags) since we're still in the formative stages. Later, this would be very bad. Leaving a Frame message in case we want frame-by-frame data, but we prefer to keep the output storage small in general so avoiding it thus far. by ajm@google.com · 13 years ago
  78. 1b627c7 Tests using the rtp_rtcp test data should now be run from inside trunk/test/data/rtp_rtcp. I.e. all test files were moved to the test folder. by hellner@google.com · 13 years ago
  79. 3675f9b by tlegrand@google.com · 13 years ago
  80. 7c4469b Revamp of audio_processing unit test to use protocol buffers. Chromium's protobuf version is synced to third_party. This isn't really needed for the unit test, but I'd like to use it soon for echo recordings, so I used this as a warm up. by ajm@google.com · 13 years ago
  81. c5758f8 Uploaded test files for ADM functional tests. by henrika@google.com · 13 years ago
  82. 0adca82 Move iLBC test and reference files to new location. by tlegrand@google.com · 13 years ago
  83. 2e8a1a2 Creates new test folder for VoiceEngine test files and adds the required files. by henrika@google.com · 13 years ago
  84. 95fa29e Creating a new directory for test data files, and moving audio_processing files there. by ajm@google.com · 14 years ago