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gerrit-public.fairphone.software / fp2-dev / platform / external / chromium_org / third_party / webrtc / refs/tags/FP2-open-16.12.0 / modules
  1. a42684b Update makefiles after merge of Chromium at 39.0.2171.95 by Ben Murdoch · 10 years ago
  2. d0e274e Merge third_party/webrtc from https://chromium.googlesource.com/a/external/webrtc/stable/webrtc.git at 33e74211aee74a1cde7968e646c59ba98337ea2a by Ben Murdoch · 10 years ago
  3. 33e7421 Merge 7729 "Build fix for MIPS Android Webview build." by andrew@webrtc.org · 10 years ago
  4. 41e4e4d Merge 7580 "Build fix for MIPS32R6." by andrew@webrtc.org · 10 years ago
  5. 0cf1796 Merge third_party/webrtc from https://chromium.googlesource.com/a/external/webrtc/stable/webrtc.git at 26a0c4c9568f9e616e9e9fa8652911ddd1f1f70a by Ben Murdoch · 10 years ago
  6. 26a0c4c Merge r7418 to 39 branch by tnakamura@webrtc.org · 10 years ago
  7. db1e40e Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 53545bbfc47f2cddb7038395369a0dcd457c8b34 by Android Chromium Automerger · 10 years ago
  8. 7ce45b5 Revert r7049/r7123, which added unnecessary "u"s to "return 0"s. by henrik.lundin@webrtc.org · 10 years ago
  9. 53545bb Revert r7049/r7123, which added unnecessary "u"s to "return 0"s. by henrik.lundin@webrtc.org · 10 years ago
  10. fc65f4a Fix typo from RtpPacketizerH264. by pbos@webrtc.org · 10 years ago
  11. 05d2a47 Fix typo from RtpPacketizerH264. by pbos@webrtc.org · 10 years ago
  12. 6a4908d Enable render downmixing to mono in AudioProcessing. by andrew@webrtc.org · 10 years ago
  13. e32f060 Enable render downmixing to mono in AudioProcessing. by andrew@webrtc.org · 10 years ago
  14. eb091ef Add missing DesktopConfigurationMonitor Unlock in webrtc::ScreenCapturerMac by jiayl@webrtc.org · 10 years ago
  15. 9073595 Add missing DesktopConfigurationMonitor Unlock in webrtc::ScreenCapturerMac by jiayl@webrtc.org · 10 years ago
  16. 8af00ea Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 291035ed1d8ec308ffbc81e9cd119e2f53f92f86 by Android Chromium Automerger · 10 years ago
  17. 3764658 Call NS AnalyzeCaptureAudio before AEC by aluebs@webrtc.org · 10 years ago
  18. 85b5766 Call NS AnalyzeCaptureAudio before AEC by aluebs@webrtc.org · 10 years ago
  19. 98be0b4 Reduce jitter delay for low fps streams. Enabled by finch flag. by sprang@webrtc.org · 10 years ago
  20. 8fa619d Reduce jitter delay for low fps streams. Enabled by finch flag. by sprang@webrtc.org · 10 years ago
  21. 291035e Moved the filter calculation from analyze to process in ns_core by aluebs@webrtc.org · 10 years ago
  22. 9635698 Moved the filter calculation from analyze to process in ns_core by aluebs@webrtc.org · 10 years ago
  23. 16efee4 audioproc: Now also writes to output file in simulation mode by bjornv@webrtc.org · 10 years ago
  24. 95eaf0f audioproc: Now also writes to output file in simulation mode by bjornv@webrtc.org · 10 years ago
  25. 83a60f3 WebRtcIsac_Encode and WebRtcIsacfix_Encode: Type encoded stream as uint8_t by kwiberg@webrtc.org · 10 years ago
  26. 09c2178 WebRtcIsac_Encode and WebRtcIsacfix_Encode: Type encoded stream as uint8_t by kwiberg@webrtc.org · 10 years ago
  27. d54aa96 Move thread_annotations.h to webrtc/base/. by pbos@webrtc.org · 10 years ago
  28. 8b9a685 Move thread_annotations.h to webrtc/base/. by pbos@webrtc.org · 10 years ago
  29. 6acb36c Use VPX_IMG_FMT_*/VPX_PLANE_* defines by johannkoenig@google.com · 10 years ago
  30. b38f723 Use VPX_IMG_FMT_*/VPX_PLANE_* defines by johannkoenig@google.com · 10 years ago
  31. 08500f0 gn: Hide modules/video_capture:video_capture_internal_impl behind an arg by pbos@webrtc.org · 10 years ago
  32. 437b8f9 gn: Hide modules/video_capture:video_capture_internal_impl behind an arg by pbos@webrtc.org · 10 years ago
  33. e0eadc5 Reland "Converting five tests to use new AudioCoding interface" (r7258) by henrik.lundin@webrtc.org · 10 years ago
  34. 30a7893 Reland "Converting five tests to use new AudioCoding interface" (r7258) by henrik.lundin@webrtc.org · 10 years ago
  35. dabeef3 Reland (rev 7259) "Convert AcmReceiverTest to new AudioCoding interface" by andresp@webrtc.org · 10 years ago
  36. 6f8f514 Reland (rev 7259) "Convert AcmReceiverTest to new AudioCoding interface" by andresp@webrtc.org · 10 years ago
  37. 6712541 audio_processing/agc: Solved building with AGC_DEBUG + few style changes by bjornv@webrtc.org · 10 years ago
  38. debd58e audio_processing/agc: Solved building with AGC_DEBUG + few style changes by bjornv@webrtc.org · 10 years ago
  39. 9865b9a modules_unittests: Turned on ApmTest.Process test for Android by bjornv@webrtc.org · 10 years ago
  40. a570aa8 modules_unittests: Turned on ApmTest.Process test for Android by bjornv@webrtc.org · 10 years ago
  41. aa50c9a Revert 7266 "WebRtcIsac_Encode and WebRtcIsacfix_Encode: Type en..." by andrew@webrtc.org · 10 years ago
  42. ae889e1 Revert 7266 "WebRtcIsac_Encode and WebRtcIsacfix_Encode: Type en..." by andrew@webrtc.org · 10 years ago
  43. e6c4d20 WebRtcIsac_Encode and WebRtcIsacfix_Encode: Type encoded stream as uint8_t by kwiberg@webrtc.org · 10 years ago
  44. 0d9ffde WebRtcIsac_Encode and WebRtcIsacfix_Encode: Type encoded stream as uint8_t by kwiberg@webrtc.org · 10 years ago
  45. 3f6c663 Revert "Converting five tests to use new AudioCoding interface" (rev 7258). by andresp@webrtc.org · 10 years ago
  46. b43fbd1 Revert "Converting five tests to use new AudioCoding interface" (rev 7258). by andresp@webrtc.org · 10 years ago
  47. f349bd7 Revert "Convert AcmReceiverTest to new AudioCoding interface" (rev 7258). by andresp@webrtc.org · 10 years ago
  48. 7b79204 Revert "Convert AcmReceiverTest to new AudioCoding interface" (rev 7258). by andresp@webrtc.org · 10 years ago
  49. 49e3622 Convert AcmReceiverTest to new AudioCoding interface by henrik.lundin@webrtc.org · 10 years ago
  50. 898422c Convert AcmReceiverTest to new AudioCoding interface by henrik.lundin@webrtc.org · 10 years ago
  51. 24ad2c8 Converting five tests to use new AudioCoding interface by henrik.lundin@webrtc.org · 10 years ago
  52. 823e8e0 Converting five tests to use new AudioCoding interface by henrik.lundin@webrtc.org · 10 years ago
  53. 913bb23 Clang-format ns_core by aluebs@webrtc.org · 10 years ago
  54. 638953d Clang-format ns_core by aluebs@webrtc.org · 10 years ago
  55. 9f91a30 Ensure that NetEq recovers after a large timestamp jump by henrik.lundin@webrtc.org · 10 years ago
  56. 153af05 Ensure that NetEq recovers after a large timestamp jump by henrik.lundin@webrtc.org · 10 years ago
  57. 81b7993 Update makefiles after merge of Chromium at fb34b348eead by Android Chromium Automerger · 10 years ago
  58. 65d2bb0 Separate between Analyze and Process in NS by aluebs@webrtc.org · 10 years ago
  59. cf7364b Separate between Analyze and Process in NS by aluebs@webrtc.org · 10 years ago
  60. d7cb16f Update makefiles after merge of Chromium at 7075322754d5 by Android Chromium Automerger · 10 years ago
  61. bd42003 Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 70861e04b3580c1350c5479c9ee26469f38ff782 by Android Chromium Automerger · 10 years ago
  62. 8d08a85 Clean directx_sdk_path as it is already defined in base/common.gypi by andresp@webrtc.org · 10 years ago
  63. 43ed74e Clean directx_sdk_path as it is already defined in base/common.gypi by andresp@webrtc.org · 10 years ago
  64. d60fc6a Creating a test helper class TimestampJumpRtpGenerator by henrik.lundin@webrtc.org · 10 years ago
  65. 6da7118 Creating a test helper class TimestampJumpRtpGenerator by henrik.lundin@webrtc.org · 10 years ago
  66. 70861e0 Update iOS video capture to use non-deprecated APIs. by tkchin@webrtc.org · 10 years ago
  67. 2884994 Update iOS video capture to use non-deprecated APIs. by tkchin@webrtc.org · 10 years ago
  68. c760f53 Trying to fix Chrome FYI bots. by andresp@webrtc.org · 10 years ago
  69. 5568140 Trying to fix Chrome FYI bots. by andresp@webrtc.org · 10 years ago
  70. 6b97015 Expose VP8/H264 defaults through video_encoder.h. by pbos@webrtc.org · 10 years ago
  71. 887fa66 Expose VP8/H264 defaults through video_encoder.h. by pbos@webrtc.org · 10 years ago
  72. 0f0aea0 Fix proper deps in BUILD.gn files. This should make Chrome GN bots happy. by andresp@webrtc.org · 10 years ago
  73. 1d40c7e Fix proper deps in BUILD.gn files. This should make Chrome GN bots happy. by andresp@webrtc.org · 10 years ago
  74. ba6a0c5 Add Analyze API to NS by aluebs@webrtc.org · 10 years ago
  75. 948fdbb Add Analyze API to NS by aluebs@webrtc.org · 10 years ago
  76. 0ab271b Split video_render_module implementation into default and internal implementation. by andresp@webrtc.org · 10 years ago
  77. 8088308 Split video_render_module implementation into default and internal implementation. by andresp@webrtc.org · 10 years ago
  78. 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
  79. 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
  80. 4d57d89 Modifying NetEqExternalDecoderTest by henrik.lundin@webrtc.org · 10 years ago
  81. fc0c38d Modifying NetEqExternalDecoderTest by henrik.lundin@webrtc.org · 10 years ago
  82. 9f749f3 Refactor VP8 de-packetizer. by stefan@webrtc.org · 10 years ago
  83. 6edef8d Refactor VP8 de-packetizer. by stefan@webrtc.org · 10 years ago
  84. cd896a9 Revert "Disable video_capture_tests for Android." (revision 7023). by andresp@webrtc.org · 10 years ago
  85. fce3a4b Revert "Disable video_capture_tests for Android." (revision 7023). by andresp@webrtc.org · 10 years ago
  86. 36e363e Split video_capture_module specific implementation (external vs internal capture) by andresp@webrtc.org · 10 years ago
  87. f91b519 Split video_capture_module specific implementation (external vs internal capture) by andresp@webrtc.org · 10 years ago
  88. 8d6e944 Split video engine android initialization into each internal module initialization. by andresp@webrtc.org · 10 years ago
  89. 886eea5 Split video engine android initialization into each internal module initialization. by andresp@webrtc.org · 10 years ago
  90. 1c65545 Revert 7151 "Revert 7114 "Expose VideoEncoders with webrtc/video_encoder.h."" by pbos@webrtc.org · 10 years ago
  91. c6fa779 Revert 7151 "Revert 7114 "Expose VideoEncoders with webrtc/video_encoder.h."" by pbos@webrtc.org · 10 years ago
  92. 8f32b79 Mark all virtual overrides in the hierarchies of UdpTransportData and by henrikg@webrtc.org · 10 years ago
  93. d13ab1d Mark all virtual overrides in the hierarchies of UdpTransportData and by henrikg@webrtc.org · 10 years ago
  94. 741367c Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 34374d1ea3f56216836788b7378c69a540fe9197 by Android Chromium Automerger · 10 years ago
  95. 99237f4 audio_processing/aec: Ported NEON optimizations of SubbandCoherence() and its sub-functions to SSE2 by bjornv@webrtc.org · 10 years ago
  96. 7c96133 audio_processing/aec: Ported NEON optimizations of SubbandCoherence() and its sub-functions to SSE2 by bjornv@webrtc.org · 10 years ago
  97. 4dc8e4e Add a target for the approved subset of rtc_base. by andrew@webrtc.org · 10 years ago
  98. ec30dc3 Add a target for the approved subset of rtc_base. by andrew@webrtc.org · 10 years ago
  99. 5f5cbf2 Fix memory leak in webrtc::MouseCursorMonitorMac by sergeyu@chromium.org · 10 years ago
  100. 7f0fe3f Fix memory leak in webrtc::MouseCursorMonitorMac by sergeyu@chromium.org · 10 years ago