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webrtc / src / / bc900cb1d1810fcf678fe41cf1e3966daa39c88c / test
  1. bc900cb Move rtp-specific config out of EncoderSettings. by Niels Möller · 7 years ago
  2. e62f600 Extend WavReader and WavWriter API. by Artem Titov · 7 years ago
  3. 465a5d9 Refactor payload types constants in CallTest by Ilya Nikolaevskiy · 7 years ago
  4. 7696bef Remove the public_deps to fileutils from test_support. by Patrik Höglund · 7 years ago
  5. 7bd79a0 Split up audio_device build target by Paulina Hensman · 7 years ago
  6. 9f64b9c Reland "Remove unnecessary dependency on base." by Patrik Höglund · 7 years ago
  7. b3bac5e Revert "Remove unnecessary dependency on base." by Patrik Höglund · 7 years ago
  8. e0eb13c Remove unnecessary dependency on base. by Patrik Höglund · 7 years ago
  9. 0970851 Reland: Add ability to emulate degraded network in Call via field trial by Erik Språng · 7 years ago
  10. 16cba5c Revert "Add ability to emulate degraded network in Call via field trial" by Ilya Nikolaevskiy · 7 years ago
  11. 31a12c5 Add ability to emulate degraded network in Call via field trial by Erik Språng · 7 years ago
  12. e61bf67 Separate test/fake_audio_device on API and implementation. Step 3. by Artem Titov · 7 years ago
  13. 207a75d Remove unused FrameGeneratorCapturer::Create signature by Emircan Uysaler · 7 years ago
  14. 3faa832 Separate test/fake_audio_device on API and implementation. Step 2. by Artem Titov · 7 years ago
  15. d6fbf2a Tests: Pass codec ID argument to audio codecs by Karl Wiberg · 7 years ago
  16. 03e6ec9 Reland "Add multiplex case to webrtc_perf_tests" by Emircan Uysaler · 7 years ago
  17. dd7e284 Reland "Enable and fix chromium clang warnings in rtp_rtcp test targets" by Danil Chapovalov · 7 years ago
  18. 01aa210 Revert "Enable and fix chromium clang warnings in rtp_rtcp test targets" by Oleh Prypin · 7 years ago
  19. 8fabab1 CNG fuzzer: avoid long fuzzer runs by limiting generator calls by Henrik Lundin · 7 years ago
  20. 9486b11 Enable and fix chromium clang warnings in rtp_rtcp test targets by Danil Chapovalov · 7 years ago
  21. bbf1465 Delete dead code for video quality calculation. by Rasmus Brandt · 7 years ago
  22. 081136f Revert "Reland "Add multiplex case to webrtc_perf_tests"" by Taylor Brandstetter · 7 years ago
  23. 7c5bc1c Reland "Add multiplex case to webrtc_perf_tests" by Emircan Uysaler · 7 years ago
  24. 5aac372 Revert "Add multiplex case to webrtc_perf_tests" by Emircan Uysaler · 7 years ago
  25. d90a7e8 Add multiplex case to webrtc_perf_tests by Emircan Uysaler · 7 years ago
  26. 12edf4c Separate build target for rtc_base/numerics/safe_minmax.h by Karl Wiberg · 7 years ago
  27. 0f03973 Separate test/fake_audio_device on API and implementation. Step 1. by Artem Titov · 7 years ago
  28. 2e1d784 Delete the VideoCodec::plName string. by Niels Möller · 7 years ago
  29. 6723cdc Revert "Separate test/fake_audio_device on API and implementation." by Artem Titov · 7 years ago
  30. 8ea5f9a Separate test/fake_audio_device on API and implementation. by Artem Titov · 7 years ago
  31. 3f693b9 Delete unused method SetPeriodicKeyFrames. by Niels Möller · 7 years ago
  32. 38c15d3 Template argument and corpora for Audio Processing Fuzzer. by Alex Loiko · 7 years ago
  33. 27e8a3e Revert "Adding gtest-spi.h in webrtc/test/gtest.h" by Philip Eliasson · 7 years ago
  34. 68f4904 Adding gtest-spi.h in webrtc/test/gtest.h by Alessio Bazzica · 7 years ago
  35. 0efa941 Move EchoCanceller3Factory to api/auido by Gustaf Ullberg · 7 years ago
  36. 151be2d comfort_noise_decoder_fuzzer: limit the fuzzer input size to avoid timeout by Henrik Lundin · 7 years ago
  37. 06fa153 neteq_rtp_fuzzer: limit the fuzzer input size to avoid timeout by Henrik Lundin · 7 years ago
  38. 2a6d864 neteq_signal_fuzzer: limit the fuzzer input size to avoid timeout by Henrik Lundin · 7 years ago
  39. f35c666 Separate build targets for aec3 and aec3_unittests by Gustaf Ullberg · 7 years ago
  40. 41f16be Silencing warnings in audio send stream unit tests. by Sebastian Jansson · 7 years ago
  41. 64cf731 Roll chromium_revision 2c98648a24..37c4da4be1 (538114:538199) by Mirko Bonadei · 7 years ago
  42. 9a03dd8 Removed new calls on RtpTransportControllerSend. by Sebastian Jansson · 7 years ago
  43. 5d436ac Removed Die mock from MockAudioEncoder by Sebastian Jansson · 7 years ago
  44. 97f61ea Moved bitrate configuration to rtp controller by Sebastian Jansson · 7 years ago
  45. a05ee82 Fixed Digital mode of AGC2 implementation finished. by Alex Loiko · 7 years ago
  46. 9d138fc Drop dependency of common_video on api:libjingle_peerconnection_api. by Niels Möller · 7 years ago
  47. 61405bc Fix infinite loop in rtp packet parsing by Danil Chapovalov · 7 years ago
  48. 0c15a09 Don't use gtest-parallel when running webrtc_perf_tests. by Edward Lemur · 7 years ago
  49. 2b304f1 Simplify CodecSettings helper function. by Rasmus Brandt · 7 years ago
  50. 1e06289 Delete macro RTC_ACCESS_ON, replaced by RTC_GUARDED_BY. by Niels Möller · 7 years ago
  51. c2dd59c Skip oversized rtp header extension when parsing Rtp Packet. by Danil Chapovalov · 7 years ago
  52. cc7125f Sets sending status for active RtpRtcp modules. by Seth Hampson · 7 years ago
  53. 970b088 Reland "Break up rtc_event_log_api to solve circular dependencies." by Qingsi Wang · 7 years ago
  54. edab301 Remove webrtc::test::InitFieldTrialsFromString(const std::string&). by Bjorn Terelius · 7 years ago
  55. 75df728 Revert "Break up rtc_event_log_api to solve circular dependencies." by Mirko Bonadei · 7 years ago
  56. 001546d Break up rtc_event_log_api to solve circular dependencies. by Qingsi Wang · 7 years ago
  57. 2e5966b Store video_quality_loopback_test perf results in Chart JSON format. by Edward Lemur · 7 years ago
  58. 4f6e4f0 Increase rtp_file_reader line length to support ipv6. by Stefan Holmer · 7 years ago
  59. d7ae3c3 Reland "Rename stereo video codec to multiplex" by Emircan Uysaler · 7 years ago
  60. e48c61f Delete unused MediaFile module. by Niels Möller · 7 years ago
  61. 1204448 Revert "Reland "Rename stereo video codec to multiplex"" by Taylor Brandstetter · 7 years ago
  62. 4954a77 Reland "Rename stereo video codec to multiplex" by Emircan Uysaler · 7 years ago
  63. 65ce311 Removing useless dependencies on //testing/gmock. by Mirko Bonadei · 7 years ago
  64. 6bc7bb6 Revert "Rename stereo video codec to multiplex" by Ivo Creusen · 7 years ago
  65. bbdabe5 Rename stereo video codec to multiplex by Emircan Uysaler · 7 years ago
  66. 3a5653a Use FILE* instead of const FILE* in perf_test.h by Edward Lemur · 7 years ago
  67. c9e4522 Add an option to print perf results to a file. by Edward Lemur · 7 years ago
  68. 1723cf9 Get rid of packet loss related stuff from videoprocessor. by Sergey Silkin · 7 years ago
  69. 393e266 Use correct RTP header length in RED generation for ULPFEC packets. by Rasmus Brandt · 7 years ago
  70. 34924c2 Fix warning 4373. by Patrik Höglund · 7 years ago
  71. 46e31ba Reland "Enables/disables simulcast streams by allocating a bitrate of 0 to the spatial layer." by Seth Hampson · 7 years ago
  72. 2ffe3e8 Reland Use runtime enabled features API to enable dual stream mode by Ilya Nikolaevskiy · 7 years ago
  73. 3be2a55 Reland "Updated analysis in videoprocessor." by Sergey Silkin · 7 years ago
  74. c1094eb Revert "Use runtime enabled features API to enable dual stream mode" by Lu Liu · 7 years ago
  75. 6f011df Use runtime enabled features API to enable dual stream mode by Ilya Nikolaevskiy · 7 years ago
  76. a8b7c7f Move remaining traces of VoiceEngine by Fredrik Solenberg · 7 years ago
  77. 18bc3e1 Revert "Updated analysis in videoprocessor." by Sergey Silkin · 7 years ago
  78. 1880c71 Updated analysis in videoprocessor. by Sergey Silkin · 7 years ago
  79. 0f17f9c Revert "Enables/disables simulcast streams by allocating a bitrate of 0 to the spatial layer." by Lu Liu · 7 years ago
  80. 18c4261 Enables/disables simulcast streams by allocating a bitrate of 0 to the spatial layer. by Seth Hampson · 7 years ago
  81. ab20a60 AEC-m and AEC-2 fuzzing. by Alex Loiko · 7 years ago
  82. 3ac67a7 Make aleloi@webrtc.org owner of test/fuzzers by Alex Loiko · 7 years ago
  83. 90ea504 Delete Channel::OnRecoveredPacket. by Niels Möller · 7 years ago
  84. 75baa49 Stop using public_deps in media/. by Mirko Bonadei · 7 years ago
  85. 3f6804d Optional: Use nullopt and implicit construction in /test by Oskar Sundbom · 7 years ago
  86. 8f5787a Move ownership of voe::Channel into Audio[Receive|Send]Stream. by Fredrik Solenberg · 7 years ago
  87. a7f2d84 Revert "Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*""" by Per Kjellander · 7 years ago
  88. c73e1f4 Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*"" by Per Kjellander · 7 years ago
  89. 588c548 GN rtc_* templates: Set default visibility to webrtc_root + "/*" by Karl Wiberg · 7 years ago
  90. 62337e5 Use AudioProcessingBuilder everywhere AudioProcessing is created. by Ivo Creusen · 7 years ago
  91. 3460fa6 Use .empty() instead of '!= ""' by Edward Lemur · 7 years ago
  92. 24722b3 Reland "Wiring the RTCRtpEncodingParameters.priority down to the bitrate allocator." by Seth Hampson · 7 years ago
  93. 9a0a17f Make it possible to change the amplitude of the pulses generated by PulsedNoiseCapturer. by erikvarga@webrtc.org · 7 years ago
  94. e66572b Reland "iOS: Save perf results under Documents/perf_result.json" by Edward Lemur · 7 years ago
  95. 99175c6 Add untracked headers to video_coding. by Patrik Höglund · 7 years ago
  96. c492bf1 Fix JSON format for reporting perf results. by Edward Lemur · 7 years ago
  97. 9e19403 Move videosourceinterface to api. by Patrik Höglund · 7 years ago
  98. be214a2 Move videosinkinterface.h to common_video to solve a circular dep. by Patrik Höglund · 7 years ago
  99. 75f18fc Make building with X11 libraries optional. by Joachim Bauch · 7 years ago
  100. 97cb448 Update Webrtc to new AudioProcessing API. by Alex Loiko · 7 years ago