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webrtc / src / / a261e6136655af33f283eda8e60a6dd93dd746a4 / webrtc / test
  1. ba7dc72 Add rotation to EncodedImage and make sure it is passed through encoders. by Per · 9 years ago
  2. 06176e4 Added new VideoFrameBuffer methods Data[YUV]() etc. by nisse · 9 years ago
  3. e532aec Add isolate files for Android tests by kjellander · 9 years ago
  4. 26acec4 Delete method webrtc::VideoFrame::native_handle. by nisse · 9 years ago
  5. 90a1351 Fixed rtcp rpsi parsing of invalid packets. by danilchap · 9 years ago
  6. 95177d1 GN: Fix some build errors for iOS. by kjellander · 9 years ago
  7. 4a206a9 Remove webrtc::ScopedVector by kwiberg · 9 years ago
  8. 7cc9cc0 New method I420Buffer::Copy. by nisse · 9 years ago
  9. 7ade7b3 Delete class webrtc::VideoRenderer and its header file. by nisse · 9 years ago
  10. 1d19441 Replace RefCountImpl with rtc::RefCountedObject. by Peter Boström · 9 years ago
  11. eb83a1a This is an initial cleanup step, aiming to delete the by nisse · 9 years ago
  12. b4c8247 Added function for parsing single rtcp packet in tests. by Danil Chapovalov · 9 years ago
  13. 94a23f0 Reland "Add check_deps rules in DEPS files." by kjellander@webrtc.org · 9 years ago
  14. 56cf60e Revert of Add check_deps rules in DEPS files. (patchset #2 id:60001 of https://codereview.webrtc.org/1796413002/ ) by kjellander · 9 years ago
  15. 086f851 Add check_deps rules in DEPS files. by kjellander@webrtc.org · 9 years ago
  16. 8842c3e Relanding https://codereview.webrtc.org/1715883002/ in pieces. by solenberg · 9 years ago
  17. 3ecb5c8 Revert of - Clean up unused voice engine DTMF code. (patchset #4 id:60001 of https://codereview.webrtc.org/1722253002/ ) by solenberg · 9 years ago
  18. 8886c81 - Clean up unused voice engine DTMF code following removal of VoEDtmf APIs. by solenberg · 9 years ago
  19. 622d895 Remove the VoEDtmf interface. by solenberg · 9 years ago
  20. 0e73934 Remove webrtc/test/webrtc_test_common.gyp by kjellander · 9 years ago
  21. c891eb4 Replace scoped_ptr with unique_ptr in webrtc/common_video/ by kwiberg · 9 years ago
  22. 739fcb9 Cleanup of webrtc::VideoFrame. by Niels Möller · 9 years ago
  23. 54ebfca Revert of Cleanup of webrtc::VideoFrame. (patchset #6 id:100001 of https://codereview.webrtc.org/1679323002/ ) by kjellander · 9 years ago
  24. 2080196 Cleanup of webrtc::VideoFrame. by nisse · 9 years ago
  25. ba3e25e Simple RTCP receiver fuzzer. by Peter Boström · 9 years ago
  26. 66a9928 Roll chromium_revision 1d144ca..fa5d546 (375480:376142) by kjellander@webrtc.org · 9 years ago
  27. b7f89d6 Replace scoped_ptr with unique_ptr in webrtc/voice_engine/ by kwiberg · 9 years ago
  28. 0206000 iOS: Add resource files for tests and implement OutputPath by kjellander · 9 years ago
  29. 9c6a0c7 Added A/V sync tests with drifting clocks. by danilchap · 9 years ago
  30. fd2be27 Fuzzer tests for AudioDecoder's DecodeRedundant and IncomingPacket by henrik.lundin · 9 years ago
  31. f6b5509 Fix GYP and GN references that are invalid in Chromium builds. by kjellander · 9 years ago
  32. 988d31e Move gtest_prod_util.h out of webrtc/test tree. by kjellander · 9 years ago
  33. 2ab8157 Remove implicit downcast in producer_fec_fuzzer.cc. by Peter Boström · 9 years ago
  34. ed3277b Deprecate VideoDecoder::Reset() and remove calls. by Peter Boström · 9 years ago
  35. bba9dec Use separate rtp module lists for send and receive in PacketRouter. by stefan · 9 years ago
  36. f5b804b Fix implicit bool casts in producer_fec_fuzzer.cc. by Peter Boström · 9 years ago
  37. bab934b H.264 video codec support using OpenH264 (http://www.openh264.org/) for encoding and FFmpeg (https://www.ffmpeg.org/) for decoding. by hbos · 9 years ago
  38. a2c5523 Allow packets to be reordered in the fake network pipe. by philipel · 9 years ago
  39. 5ad935c Remove mutable from rtc::CriticalSection members. by pbos · 9 years ago
  40. 693a114 Add stefan@webrtc.org to webrtc/test/OWNERS. by Peter Boström · 9 years ago
  41. 3313ec9 Enable transport seq num extension on receive channel to suppress log warning. by stefan · 9 years ago
  42. 72c08ed Reenables several NetEq unittests on android. by ivoc · 9 years ago
  43. f01ea4f Remove use of ConditionVariableWrapper and CriticalSectionWrapper from UdpSocket2Windows. by Tommi · 9 years ago
  44. 2067826 Remove dependency on ConditionVariableWrapper and CriticalSectionWrapper in UdpSocketPosix. by Tommi · 9 years ago
  45. 04cb763 Add tests for verifying transport feedback for audio and video. by Stefan Holmer · 9 years ago
  46. 688e308 Re-land: "Use an explicit identifier in Config" by aluebs · 9 years ago
  47. ff2a635 Add ramp-up tests for transport sequence number with and w/o audio. by Stefan Holmer · 9 years ago
  48. fca54f4 Revert of Use an explicit identifier in Config (patchset #4 id:60001 of https://codereview.webrtc.org/1538643004/ ) by tommi · 9 years ago
  49. 25249d9 Use an explicit identifier in Config by aluebs · 9 years ago
  50. e74eef1 Add CreateSend/ReceiveTransport() methods to CallTest. by stefan · 9 years ago
  51. 37ebcf0 Reland "Add APK targets to build libjingle tests for Android." by phoglund · 9 years ago
  52. 9fea80f Add audio streams to CallTest and a first A/V call test. by Stefan Holmer · 9 years ago
  53. 1fe48a5 Add implementation in metrics.h that uses atomic pointer. by asapersson · 9 years ago
  54. e2976c8 Remove DISABLED_ON_ macros. by Peter Boström · 9 years ago
  55. 13f61df Move fake-handle frame creation into test target. by Peter Boström · 9 years ago
  56. f6975f4 [rtp_rtcp] Lint errors cleaned from rtp_utility by danilchap · 9 years ago
  57. ff48361 Step 1 to prepare call_test.* for combined audio/video tests. by stefan · 9 years ago
  58. b7d9a97 Expose codec implementation names in stats. by Peter Boström · 9 years ago
  59. 1e0cfd9 Add VP8 and H264 depacketizer fuzzers. by Peter Boström · 9 years ago
  60. 3514cbe Add DrFuzz support to webrtc fuzzers. by pbos · 9 years ago
  61. 7cae30c Disable warnings failing when using Clang on Windows. by kjellander · 9 years ago
  62. 78315b9 Reland of Base webrtc fuzzers on a template. (patchset #1 id:1 of https://codereview.webrtc.org/1528043002/ ) by Peter Boström · 9 years ago
  63. 5e0218c Revert of Base webrtc fuzzers on a template. (patchset #1 id:1 of https://codereview.webrtc.org/1524993002/ ) by tommi · 9 years ago
  64. 5ea3da2 Base webrtc fuzzers on a template. by Peter Boström · 9 years ago
  65. bc14164 Revert of Add APK targets to build libjingle tests for Android. (patchset #10 id:180001 of https://codereview.webrtc.org/1511633002/ ) by stefan · 9 years ago
  66. a78c021 Add APK targets to build libjingle_peerconnection_unittests for Android. by perkj · 9 years ago
  67. 4c1093b Add FEC producer fuzzing and a unittest for one of the issues found. by Stefan Holmer · 9 years ago
  68. 5811a39 Replace EventWrapper in video/, test/ and call/. by Peter Boström · 9 years ago
  69. 84e78f9 Rewrote the PRNG using an xorshift* algorithm and moved the files from test/ to base/. by terelius · 9 years ago
  70. 7623ce4 Reland of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:300001 of https://codereview.webrtc.org/1507903005/ ) by Peter Boström · 9 years ago
  71. d3c9447 Nuke TickTime::UseFakeClock. by Peter Boström · 9 years ago
  72. 8237abf Revert of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:20001 of https://codereview.webrtc.org/1506773002/ ) by kjellander · 9 years ago
  73. fe32a76 Create fuzzer tests for audio decoders by Henrik Lundin · 9 years ago
  74. 03ef053 Merge webrtc/video_engine/ into webrtc/video/ by Peter Boström · 9 years ago
  75. b86d4e4 Prepare the AudioSendStream to be hooked up to send-side BWE. by Stefan Holmer · 9 years ago
  76. b572768 - Remove calls to VoEDtmf from WVoE/MC. by Fredrik Solenberg · 9 years ago
  77. 358057b Use ChannelProxy for most calls on voe::Channel in Audio[Receive|Send]Stream. by solenberg · 9 years ago
  78. def5820 Default to LS_INFO logging for release builds. by Peter Boström · 9 years ago
  79. 8c38e8b Clean up PlatformThread. by Peter Boström · 9 years ago
  80. ad113e5 Fix bug in calculation of averge queue time in paced sender. by Erik Språng · 9 years ago
  81. 871c419 Add fuzzing of VP8 QP parsing. by Peter Boström · 9 years ago
  82. 89d658f Fix fuzzer breakage in Chromium. by Peter Boström · 9 years ago
  83. 1372508 Open backdoor in VoiceEngineImpl to get at the actual voe::Channel objects from an ID. by solenberg · 9 years ago
  84. 12411ef Move ThreadWrapper to ProcessThread in base. by pbos · 9 years ago
  85. 62e9bda Implement fuzzing of VP9 depacketization. by Peter Boström · 9 years ago
  86. 2557b86 modules/video_coding refactorings by Henrik Kjellander · 9 years ago
  87. 6f8ce06 common_video: rename interface -> include by kjellander · 9 years ago
  88. 3a94154 Move some send stream configuration into webrtc::AudioSendStream. by solenberg · 9 years ago
  89. 0b9e29c Remove include dirs from modules/{media_file,pacing} by Henrik Kjellander · 9 years ago
  90. 5dda80a Remove webrtc/modules/video_{capture,render}/include by Henrik Kjellander · 9 years ago
  91. 1323fc3 Remove webrtc/test/channel_transport/include by Henrik Kjellander · 9 years ago
  92. 56b1128 Change to use local Random object instead of global rand() in the RtcEventLog unit test. by terelius · 9 years ago
  93. c4a1c37 Removed vie_defines.h by mflodman · 9 years ago
  94. ff761fb modules: more interface -> include renames by Henrik Kjellander · 9 years ago
  95. 0ccae13 Changed FakeVoiceEngine into a MockVoiceEngine. by Fredrik Solenberg · 9 years ago
  96. ce4aef1 Adding support for simulcast and spatial layers into VideoQualityTest by sprang · 9 years ago
  97. 74f0f35 Delete a chain of methods in ViE, VoE and ACM by henrik.lundin · 9 years ago
  98. 69ccb33 Remove redudant encoder rate calls. by Peter Boström · 9 years ago
  99. 1295297 Register header extensions in RtpRtcpObserver to avoid log spam. by Stefan Holmer · 9 years ago
  100. 95192fb Create a 'webrtc_nonparallel_tests' target. by Peter Boström · 9 years ago