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gerrit-public.fairphone.software / fp2-dev / platform / external / chromium_org / third_party / webrtc / refs/tags/FP2-open-16.10.0
  1. a570aa8 modules_unittests: Turned on ApmTest.Process test for Android by bjornv@webrtc.org · 10 years ago
  2. aa50c9a Revert 7266 "WebRtcIsac_Encode and WebRtcIsacfix_Encode: Type en..." by andrew@webrtc.org · 10 years ago
  3. ae889e1 Revert 7266 "WebRtcIsac_Encode and WebRtcIsacfix_Encode: Type en..." by andrew@webrtc.org · 10 years ago
  4. e6c4d20 WebRtcIsac_Encode and WebRtcIsacfix_Encode: Type encoded stream as uint8_t by kwiberg@webrtc.org · 10 years ago
  5. 0d9ffde WebRtcIsac_Encode and WebRtcIsacfix_Encode: Type encoded stream as uint8_t by kwiberg@webrtc.org · 10 years ago
  6. 3f6c663 Revert "Converting five tests to use new AudioCoding interface" (rev 7258). by andresp@webrtc.org · 10 years ago
  7. b43fbd1 Revert "Converting five tests to use new AudioCoding interface" (rev 7258). by andresp@webrtc.org · 10 years ago
  8. 604a2ba Adding test file path as argument of the rtcBot run command's arguments. by houssainy@google.com · 10 years ago
  9. 6fe065f Adding test file path as argument of the rtcBot run command's arguments. by houssainy@google.com · 10 years ago
  10. c747067 Remove Get/SetNetEQPlayoutMode APIs by henrik.lundin@webrtc.org · 10 years ago
  11. 17b5522 Remove Get/SetNetEQPlayoutMode APIs by henrik.lundin@webrtc.org · 10 years ago
  12. 881c38a Adding webrtc_video_streaming test by houssainy@google.com · 10 years ago
  13. 44455c5 Adding webrtc_video_streaming test by houssainy@google.com · 10 years ago
  14. f349bd7 Revert "Convert AcmReceiverTest to new AudioCoding interface" (rev 7258). by andresp@webrtc.org · 10 years ago
  15. 7b79204 Revert "Convert AcmReceiverTest to new AudioCoding interface" (rev 7258). by andresp@webrtc.org · 10 years ago
  16. 49e3622 Convert AcmReceiverTest to new AudioCoding interface by henrik.lundin@webrtc.org · 10 years ago
  17. 898422c Convert AcmReceiverTest to new AudioCoding interface by henrik.lundin@webrtc.org · 10 years ago
  18. 24ad2c8 Converting five tests to use new AudioCoding interface by henrik.lundin@webrtc.org · 10 years ago
  19. 823e8e0 Converting five tests to use new AudioCoding interface by henrik.lundin@webrtc.org · 10 years ago
  20. 913bb23 Clang-format ns_core by aluebs@webrtc.org · 10 years ago
  21. 638953d Clang-format ns_core by aluebs@webrtc.org · 10 years ago
  22. a2970d2 Set number of temporal layers for VideoSendStream. by pbos@webrtc.org · 10 years ago
  23. 2a7ddf5 Set number of temporal layers for VideoSendStream. by pbos@webrtc.org · 10 years ago
  24. 9f91a30 Ensure that NetEq recovers after a large timestamp jump by henrik.lundin@webrtc.org · 10 years ago
  25. 153af05 Ensure that NetEq recovers after a large timestamp jump by henrik.lundin@webrtc.org · 10 years ago
  26. 81b7993 Update makefiles after merge of Chromium at fb34b348eead by Android Chromium Automerger · 10 years ago
  27. d6e65cb Disabled several rtc_unittests so the tests can be turned on in the waterfall by henrike@webrtc.org · 10 years ago
  28. a2f0226 Disabled several rtc_unittests so the tests can be turned on in the waterfall by henrike@webrtc.org · 10 years ago
  29. 65d2bb0 Separate between Analyze and Process in NS by aluebs@webrtc.org · 10 years ago
  30. cf7364b Separate between Analyze and Process in NS by aluebs@webrtc.org · 10 years ago
  31. f46745b Additional disabled tests in rtc_unittests. by kjellander@webrtc.org · 10 years ago
  32. cf1250a Additional disabled tests in rtc_unittests. by kjellander@webrtc.org · 10 years ago
  33. 60b0bb6 Additional disabled tests in rtc_unittests. by kjellander@webrtc.org · 10 years ago
  34. f304de6 Additional disabled tests in rtc_unittests. by kjellander@webrtc.org · 10 years ago
  35. a127c95 base: disabled several base tests on Mac so that rtc_unittests can be turned back on by henrike@webrtc.org · 10 years ago
  36. 6030c8c base: disabled several base tests on Mac so that rtc_unittests can be turned back on by henrike@webrtc.org · 10 years ago
  37. 58b5140 Config struct for VideoEncoder. by pbos@webrtc.org · 10 years ago
  38. 2249a5c Config struct for VideoEncoder. by pbos@webrtc.org · 10 years ago
  39. 0dff1ec Merge from Chromium at DEPS revision 38.0.2125.69 by Torne (Richard Coles) · 10 years ago
  40. db2cf0f Merge third_party/webrtc from https://chromium.googlesource.com/a/external/webrtc/stable/webrtc.git at a6ed0dfa13f606d7f173f740510c2ed5855aa7fd by Torne (Richard Coles) · 10 years ago
  41. d7cb16f Update makefiles after merge of Chromium at 7075322754d5 by Android Chromium Automerger · 10 years ago
  42. bd42003 Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 70861e04b3580c1350c5479c9ee26469f38ff782 by Android Chromium Automerger · 10 years ago
  43. 29c4d87 Re-enable missing android tests disabled due to issue 3770. by andresp@webrtc.org · 10 years ago
  44. 15497d3 Re-enable missing android tests disabled due to issue 3770. by andresp@webrtc.org · 10 years ago
  45. 8d08a85 Clean directx_sdk_path as it is already defined in base/common.gypi by andresp@webrtc.org · 10 years ago
  46. 43ed74e Clean directx_sdk_path as it is already defined in base/common.gypi by andresp@webrtc.org · 10 years ago
  47. d60fc6a Creating a test helper class TimestampJumpRtpGenerator by henrik.lundin@webrtc.org · 10 years ago
  48. 6da7118 Creating a test helper class TimestampJumpRtpGenerator by henrik.lundin@webrtc.org · 10 years ago
  49. 70861e0 Update iOS video capture to use non-deprecated APIs. by tkchin@webrtc.org · 10 years ago
  50. 2884994 Update iOS video capture to use non-deprecated APIs. by tkchin@webrtc.org · 10 years ago
  51. 89fe3ca Remove the 'webrtc_test_video_render_dependencies' target. by pbos@webrtc.org · 10 years ago
  52. 74be5dc Remove the 'webrtc_test_video_render_dependencies' target. by pbos@webrtc.org · 10 years ago
  53. 692b063 Do not require synchronization access on the thread if called from rtc::Thread::WrapCurrent. by jiayl@webrtc.org · 10 years ago
  54. 9ee0166 Do not require synchronization access on the thread if called from rtc::Thread::WrapCurrent. by jiayl@webrtc.org · 10 years ago
  55. c760f53 Trying to fix Chrome FYI bots. by andresp@webrtc.org · 10 years ago
  56. 5568140 Trying to fix Chrome FYI bots. by andresp@webrtc.org · 10 years ago
  57. 6b97015 Expose VP8/H264 defaults through video_encoder.h. by pbos@webrtc.org · 10 years ago
  58. 887fa66 Expose VP8/H264 defaults through video_encoder.h. by pbos@webrtc.org · 10 years ago
  59. 0f0aea0 Fix proper deps in BUILD.gn files. This should make Chrome GN bots happy. by andresp@webrtc.org · 10 years ago
  60. 1d40c7e Fix proper deps in BUILD.gn files. This should make Chrome GN bots happy. by andresp@webrtc.org · 10 years ago
  61. ba6a0c5 Add Analyze API to NS by aluebs@webrtc.org · 10 years ago
  62. 948fdbb Add Analyze API to NS by aluebs@webrtc.org · 10 years ago
  63. 0ab271b Split video_render_module implementation into default and internal implementation. by andresp@webrtc.org · 10 years ago
  64. 8088308 Split video_render_module implementation into default and internal implementation. by andresp@webrtc.org · 10 years ago
  65. 2738d47 Implemented Network::GetBestIP() selection logic as following. by guoweis@webrtc.org · 10 years ago
  66. 96c3a45 Implemented Network::GetBestIP() selection logic as following. by guoweis@webrtc.org · 10 years ago
  67. 223c9c1 The 2x2 black frame on windows when the shared window is minimized caused an assert from vp8 and may lead to memroy corruption. by jiayl@webrtc.org · 10 years ago
  68. a0d3d87 The 2x2 black frame on windows when the shared window is minimized caused an assert from vp8 and may lead to memroy corruption. by jiayl@webrtc.org · 10 years ago
  69. 4d57d89 Modifying NetEqExternalDecoderTest by henrik.lundin@webrtc.org · 10 years ago
  70. fc0c38d Modifying NetEqExternalDecoderTest by henrik.lundin@webrtc.org · 10 years ago
  71. 9f749f3 Refactor VP8 de-packetizer. by stefan@webrtc.org · 10 years ago
  72. 6edef8d Refactor VP8 de-packetizer. by stefan@webrtc.org · 10 years ago
  73. cd896a9 Revert "Disable video_capture_tests for Android." (revision 7023). by andresp@webrtc.org · 10 years ago
  74. fce3a4b Revert "Disable video_capture_tests for Android." (revision 7023). by andresp@webrtc.org · 10 years ago
  75. 36e363e Split video_capture_module specific implementation (external vs internal capture) by andresp@webrtc.org · 10 years ago
  76. f91b519 Split video_capture_module specific implementation (external vs internal capture) by andresp@webrtc.org · 10 years ago
  77. 8d6e944 Split video engine android initialization into each internal module initialization. by andresp@webrtc.org · 10 years ago
  78. 886eea5 Split video engine android initialization into each internal module initialization. by andresp@webrtc.org · 10 years ago
  79. 1c65545 Revert 7151 "Revert 7114 "Expose VideoEncoders with webrtc/video_encoder.h."" by pbos@webrtc.org · 10 years ago
  80. c6fa779 Revert 7151 "Revert 7114 "Expose VideoEncoders with webrtc/video_encoder.h."" by pbos@webrtc.org · 10 years ago
  81. 65b419b webrtc/overrides: add OWNERS-file. by henrike@webrtc.org · 10 years ago
  82. a4efd04 webrtc/overrides: add OWNERS-file. by henrike@webrtc.org · 10 years ago
  83. c72efb4 Narrower include for constructormagic.h in Chromium. by pbos@webrtc.org · 10 years ago
  84. c9f7e1c Narrower include for constructormagic.h in Chromium. by pbos@webrtc.org · 10 years ago
  85. 170b1f3 Implemented Network::GetBestIP() selection logic as following. by guoweis@webrtc.org · 10 years ago
  86. 1a4b462 Implemented Network::GetBestIP() selection logic as following. by guoweis@webrtc.org · 10 years ago
  87. 1e0452c Implemented Network::GetBestIP() selection logic as following. by guoweis@webrtc.org · 10 years ago
  88. ca61389 Implemented Network::GetBestIP() selection logic as following. by guoweis@webrtc.org · 10 years ago
  89. d0cec8b Add a gyp target for producing a voice engine merged library. by andrew@webrtc.org · 10 years ago
  90. 8dc38d5 Add a gyp target for producing a voice engine merged library. by andrew@webrtc.org · 10 years ago
  91. 0d52ec5 gn: Fix cflags usage by pbos@webrtc.org · 10 years ago
  92. 44d4222 gn: Fix cflags usage by pbos@webrtc.org · 10 years ago
  93. 8f32b79 Mark all virtual overrides in the hierarchies of UdpTransportData and by henrikg@webrtc.org · 10 years ago
  94. d13ab1d Mark all virtual overrides in the hierarchies of UdpTransportData and by henrikg@webrtc.org · 10 years ago
  95. a4c22e9 Update makefiles after merge of Chromium at 4adb514cb3ad by Android Chromium Automerger · 10 years ago
  96. 741367c Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 34374d1ea3f56216836788b7378c69a540fe9197 by Android Chromium Automerger · 10 years ago
  97. 34374d1 Fix GN for rtc_base_approved target. by kjellander@webrtc.org · 10 years ago
  98. 86655a5 Fix GN for rtc_base_approved target. by kjellander@webrtc.org · 10 years ago
  99. 99237f4 audio_processing/aec: Ported NEON optimizations of SubbandCoherence() and its sub-functions to SSE2 by bjornv@webrtc.org · 10 years ago
  100. 7c96133 audio_processing/aec: Ported NEON optimizations of SubbandCoherence() and its sub-functions to SSE2 by bjornv@webrtc.org · 10 years ago