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gerrit-public.fairphone.software / platform / external / webrtc / 34029e209d6d116c338e7d65d6980cc395184658 / test
  1. 675513b Stop using LOG macros in favor of RTC_ prefixed macros. by Mirko Bonadei · 7 years ago
  2. 84a87c4 Don't include x11-specific code when use_x11 is set to false. by Edward Lemur · 7 years ago
  3. fbb3b7d Reland: "Make javac warnings errors for WebRTC targets." by Mirko Bonadei · 7 years ago
  4. 2bad72a Reland "Revert "Make javac warnings errors for WebRTC targets."" by Lu Liu · 7 years ago
  5. a1a475a Revert "Revert "Make javac warnings errors for WebRTC targets."" by Mirko Bonadei · 7 years ago
  6. 098d24c Revert "Make javac warnings errors for WebRTC targets." by Lu Liu · 7 years ago
  7. 19b7614 Make javac warnings errors for WebRTC targets. by Sami Kalliomäki · 7 years ago
  8. de69145 Remove pbos@webrtc.org from all OWNERS. by Peter Boström · 7 years ago
  9. d2817d8 Allow injection of custom network models in place of FakeNetworkPipe. by Christoffer Rodbro · 7 years ago
  10. ddfd9c5 Fix AudioProcessing fuzzer crash. by Alex Loiko · 7 years ago
  11. 0c72050 Adding libFuzzer target for UlpFEC receiver. by Mark Brand · 7 years ago
  12. b82de30 Add new neteq_signal_fuzzer by Henrik Lundin · 7 years ago
  13. 90d46c4 Add phoglund@ to various OWNERS and remove kjellander@ by Henrik Kjellander · 7 years ago
  14. bd83b91 Separate AEC3 config from AudioProcessing::Config. by Gustaf Ullberg · 7 years ago
  15. 9ce0c7a Improving APM fuzzer coverage. by Alex Loiko · 7 years ago
  16. c774d5d Make the complexity of the square generator configurable. by erikvarga@webrtc.org · 7 years ago
  17. 6e87850 Unconditionally link fuzzers with rtc_task_queue_impl. by Niels Möller · 7 years ago
  18. 84255bb Add explicit includes of refcountedobject.h where it is used. by Niels Möller · 7 years ago
  19. fb26f85 Revert "Reland "Make rtc_base/refcount.h self contained, not including refcountedobject.h."" by Niels Moller · 7 years ago
  20. bf6937a Reland "Make rtc_base/refcount.h self contained, not including refcountedobject.h." by Niels Möller · 7 years ago
  21. 4332d09 Reland "Reland "Remove WEBRTC_TRACE."" by Fredrik Solenberg · 7 years ago
  22. a32dd01 Reland "Remove AudioDeviceObserver and make ADM not inherit from the Module interface." by Fredrik Solenberg · 7 years ago
  23. 39cefdb Revert "Reland "Remove WEBRTC_TRACE."" by Fredrik Solenberg · 7 years ago
  24. 68007e9 Reland "Remove WEBRTC_TRACE." by Fredrik Solenberg · 7 years ago
  25. 729b910 Revert "Remove WEBRTC_TRACE." by Fredrik Solenberg · 7 years ago
  26. 2209b90 Remove WEBRTC_TRACE. by Fredrik Solenberg · 7 years ago
  27. d25fa78 Revert "Make rtc_base/refcount.h self contained, not including refcountedobject.h." by Niels Moller · 7 years ago
  28. b7239a9 Make rtc_base/refcount.h self contained, not including refcountedobject.h. by Niels Möller · 7 years ago
  29. d4404c2 Revert "Remove AudioDeviceObserver and make ADM not inherit from the Module interface." by Fredrik Solenberg · 7 years ago
  30. 34cdd2d Remove AudioDeviceObserver and make ADM not inherit from the Module interface. by Fredrik Solenberg · 7 years ago
  31. 3102734 Revert "Reland of Add full stack tests for MediaCodec encoder (moved from Rietveld)." by Rasmus Brandt · 7 years ago
  32. 1c239d4 Remove voe::Statistics. by solenberg · 7 years ago
  33. 2666cf7 Reland of Add full stack tests for MediaCodec encoder (moved from Rietveld). by Rasmus Brandt · 7 years ago
  34. af8659a Rename test output to test artifacts. by Edward Lemur · 7 years ago
  35. fc3a2e3 Remove the VoiceEngineObserver callback interface. by solenberg · 7 years ago
  36. fbbba3f Remove remaining mentions of gflags by oprypin · 7 years ago
  37. 5ab6854 Revert "Remove remaining mentions of gflags" by Oleh Prypin · 7 years ago
  38. 90ce84e Remove remaining mentions of gflags by Oleh Prypin · 7 years ago
  39. 2397b9a Remove voe::OutputMixer and AudioConferenceMixer. by solenberg · 7 years ago
  40. 2c30120 Revert of Add full stack tests for MediaCodec. (patchset #10 id:180001 of https://codereview.webrtc.org/3005253002/ ) by brandtr · 7 years ago
  41. 2cefac6 Add full stack tests for MediaCodec encoder. by brandtr · 7 years ago
  42. 7cd28b9 Set protected_by_flexfec flag properly in tests. by brandtr · 7 years ago
  43. 946d886 Remove VoENetwork by solenberg · 7 years ago
  44. dd3abbb Remove VoERTP_RTCP. by solenberg · 7 years ago
  45. 6dc2038 Remove VoECodec. by solenberg · 7 years ago
  46. cb728ea Fix Gn Untracked headers in webrtc/modules/video_coding. by charujain · 7 years ago
  47. b63310a Remove VoEFile and things it uses. by solenberg · 7 years ago
  48. 35dee81 Clean out unused methods from VoiceEngine and VoEBase. by solenberg · 7 years ago
  49. 18f5427 Remove voe_auto_test and add new tests to cover the missing cases. by solenberg · 7 years ago
  50. 7120742 Adding NOLINT for typedefs.h and common_types.h by Mirko Bonadei · 7 years ago
  51. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  52. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago
  53. 81cf5e4 Move test to src/test. by andrew@webrtc.org · 12 years ago
  54. f1d6e0a Removed the obsolete sanity check and added new test HTML files. by phoglund@webrtc.org · 12 years ago
  55. 9dc45da Move trunk/test/data -> trunk/data by andrew@webrtc.org · 12 years ago
  56. 22bde08 Made sanity check more flexible. by phoglund@webrtc.org · 13 years ago
  57. 4aa57b4 Extracted a helper library from vie_auto_test. by phoglund@webrtc.org · 13 years ago
  58. 4530aa3 Updates html test file to webkitDeprecatedPeerConnection. by henrikg@webrtc.org · 13 years ago
  59. 61bf8e3 Flush far-end buffers when larger than system delay. by andrew@webrtc.org · 13 years ago
  60. 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
  61. 50d9e26 Adds autoconnect and autocall functionality to web test page. by henrikg@webrtc.org · 13 years ago
  62. 29fafef Fix building errors Review URL: https://webrtc-codereview.appspot.com/399012 by leozwang@webrtc.org · 13 years ago
  63. 51198f1 More PRESUBMIT checks. by kjellander@webrtc.org · 13 years ago
  64. 0a57aae Converted old jpeg_test tool to gtest unit test. by kjellander@webrtc.org · 13 years ago
  65. cf6a295 Making video codecs test framework integration test execute in a reproducable fashion. by kjellander@webrtc.org · 13 years ago
  66. 9d9ad88 Fixed remaining warnings. by phoglund@webrtc.org · 13 years ago
  67. daacee8 Use better reference files with audioproc_unittest. by andrew@webrtc.org · 13 years ago
  68. c80d9d9 Removed default cases causing clang errors, -Wcovered-switch-default. by mflodman@webrtc.org · 13 years ago
  69. fede80c Updated test web page info for PeerConnection v2. by henrikg@webrtc.org · 13 years ago
  70. 6a81475 Removing year range in copyright statement in test web page. by henrikg@webrtc.org · 13 years ago
  71. 16a0427 Updates for web test page. by henrikg@webrtc.org · 13 years ago
  72. 12cccdd NS-SWB: Actived SWB processing at once, i.e., no startup phase. by bjornv@webrtc.org · 13 years ago
  73. 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
  74. cc33737 Changing all PSNR/SSIM calculations to use libyuv. by kjellander@webrtc.org · 13 years ago
  75. 70adcd4 Delay estimator improvements. by bjornv@webrtc.org · 13 years ago
  76. 7270a6b Merged apm-buffer branch [r1293] back to trunk. by bjornv@webrtc.org · 13 years ago
  77. 173b7bb Integration test that tracks dropped frames and compares video output. by kjellander@webrtc.org · 13 years ago
  78. 5b97b12 Splitted FileHandler into FrameReader and FrameWriter classes and moved them to testsupport in test.gyp. by kjellander@webrtc.org · 13 years ago
  79. 80b2661 Fixing invalid check for existing file. by kjellander@webrtc.org · 13 years ago
  80. 4ed4f24 New fileutils.h method for managing resources on different platforms by kjellander@webrtc.org · 13 years ago
  81. 82d91ae Fixing crash when calculating SSIM and PSNR with empty video files in video_metrics.cc by kjellander@webrtc.org · 13 years ago
  82. a919d3a Don't return a zero delay with insufficient data. by andrew@webrtc.org · 13 years ago
  83. 5483210 Fixed open file handle in fileutils.cc Thanks Henrik L for pointing this out. by kjellander@webrtc.org · 13 years ago
  84. 91617ff by henrikg@webrtc.org · 13 years ago
  85. d0e5b96 Fix Amy's email address. by andrew@webrtc.org · 13 years ago
  86. 755b04a Add RMS computation for the RTP level indicator. by andrew@webrtc.org · 13 years ago
  87. 0db7dc6 Add file-playing channels to voe_cmd_test. by andrew@webrtc.org · 13 years ago
  88. 9b18ed6 Removed incorrect dependency. by phoglund@webrtc.org · 13 years ago
  89. 1144ba2 Base and codec tests now run verify output and render to file instead of to screen. by phoglund@webrtc.org · 13 years ago
  90. 62e48eb adding owners for test by niklas.enbom@webrtc.org · 13 years ago
  91. 4d8cd9d Adding GetOutputDir method to test_support library. by kjellander@webrtc.org · 13 years ago
  92. 20a370e Changing the namespace of TestSuite to webrtc::test. by kjellander@webrtc.org · 13 years ago
  93. 1a8d08a Changing usage of gtest_main target, to use test_support_main instead. by kjellander@webrtc.org · 13 years ago
  94. 1e10bb3 Remove global std::strings from fileutils. by andrew@webrtc.org · 13 years ago
  95. 5b5c31d Update fixed point audio processing output. by andrew@webrtc.org · 13 years ago
  96. 4c63676 Updated the AEC delay logging to output values in ms. PB output updated. by bjornv@webrtc.org · 13 years ago
  97. e698eb7 Make the sanity check test a little more robust, and add a README file. by hta@webrtc.org · 13 years ago
  98. 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
  99. 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
  100. 1ba3dbe Adds possibility to log delay estimates in AEC. by bjornv@google.com · 13 years ago