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gerrit-public.fairphone.software / fp2-dev / platform / external / chromium_org / third_party / webrtc / 5c28a0a23c4b04c44d7b1a5db4903f4e5ee18ff9
  1. 5c28a0a Update makefiles after merge of Chromium at 277521 by Android Chromium Automerger · 10 years ago
  2. e5a0f26 Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 847dfa535730a30d57cf26d788d31070b70a02af by Android Chromium Automerger · 10 years ago
  3. cb4fdd1 Update makefiles after merge of Chromium at 277428 by Android Chromium Automerger · 10 years ago
  4. c7fcada Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 5fcef2b6df45ceab39ee96a616ab0a4d3c63b83a by Android Chromium Automerger · 10 years ago
  5. 847dfa5 Add SDES, APP, IJ, SLI and PLI packet types to RTCP packet class. by asapersson@webrtc.org · 10 years ago
  6. e82b71d Remove ivinnichenko from webrtc/test/OWNERS by kjellander@webrtc.org · 10 years ago
  7. d3a2886 Importing ThreadChecker class from Chromium by henrik.lundin@webrtc.org · 10 years ago
  8. d998689 Adds aluebs@webrtc.org as owner to audio_processing by bjornv@webrtc.org · 10 years ago
  9. c1a2a43 common_audio: Removes macro WEBRTC_SPL_LSHIFT_U16 by bjornv@webrtc.org · 10 years ago
  10. f89ce46 Implements start bitrate for new video API. by mflodman@webrtc.org · 10 years ago
  11. 0c14539 Add thread annotations to parts of ACMGenericCodec by henrik.lundin@webrtc.org · 10 years ago
  12. d05de74 Add missing sources to webrtc/base/base.gyp by kjellander@webrtc.org · 10 years ago
  13. bd98cef Add glaznev@ to OWNERS for webrtc/modules/video_capture and talk/app/webrtc. by glaznev@webrtc.org · 10 years ago
  14. 9257c64 Neon version of OverdriveAndSuppress() by bjornv@webrtc.org · 10 years ago
  15. 00dffd7 Pass GYP DEPTH variable to isolate. by kjellander@webrtc.org · 10 years ago
  16. dd32ef8 Revert 6415 "Update generated asm offsets scripts." by wu@webrtc.org · 10 years ago
  17. 555f957 json.h include different header files depending on WEBRTC_CHROMIUM_BUILD being defined or not. Since json.h/cc is not even used in chromium it is the wrong flag to use. Instead add WEBRTC_EXTERNAL define. Also added OWNERS for base which is a copy of system_wrappers owners as the two folders are being merged. by henrike@webrtc.org · 10 years ago
  18. 19f89a1 Enable pacing by default and remove the option to disable it from the new API. by stefan@webrtc.org · 10 years ago
  19. 0e43e6f Update generated asm offsets scripts. by fgalligan@google.com · 10 years ago
  20. 5fcef2b Revert 6411 "Revert 6407 "Revert 6405 "Update generated asm offs..." by kjellander@webrtc.org · 10 years ago
  21. 4150d6e Revert 6407 "Revert 6405 "Update generated asm offsets scripts."" by minyue@webrtc.org · 10 years ago
  22. f6eaabf Increased kMaxRampUpDelayMs (120 to 240s). by asapersson@webrtc.org · 10 years ago
  23. 4121fcd Revert 6405 "Update generated asm offsets scripts." by henrike@webrtc.org · 10 years ago
  24. 88417a9 Update generated asm offsets scripts. by fgalligan@google.com · 10 years ago
  25. 6298c29 Re-land "Create a joint encoder/decoder wrapper for iSAC in ACM" by henrik.lundin@webrtc.org · 10 years ago
  26. 3acaa1f Reland: Making WebRTC able to play and record audio to files for tests. by phoglund@webrtc.org · 10 years ago
  27. 6845de7 Add APIs to enable padding with redundant payloads. by stefan@webrtc.org · 10 years ago
  28. 3fe0d4f Revert 6395 "Making WebRTC able to play and record audio to file..." by minyue@webrtc.org · 10 years ago
  29. 994f778 Making WebRTC able to play and record audio to files for tests. by phoglund@webrtc.org · 10 years ago
  30. caf328c Revert r6377 "Create a joint encoder/decoder wrapper for iSAC in ACM" by henrik.lundin@webrtc.org · 10 years ago
  31. ae4a452 common_audio/signal_processing: Removes macro WEBRTC_SPL_RSHIFT_U16 by bjornv@webrtc.org · 10 years ago
  32. 6e6b951 common_audio/signal_processing: Moves WEBRTC_SPL_UMUL_16_16_RSFT16 to iSAC fix by bjornv@webrtc.org · 10 years ago
  33. b96d9c7 modules/audio_processing: Adds a config for reported delays by bjornv@webrtc.org · 10 years ago
  34. adda09e Update makefiles after merge of Chromium at 276202 by Android Chromium Automerger · 10 years ago
  35. 604ba6f Delete last file in neteq4 folder by henrik.lundin@webrtc.org · 10 years ago
  36. bc9c195 MIPS optimizations for ISAC (patch #1) by andrew@webrtc.org · 10 years ago
  37. d70e23e Noise suppression: Change signature to work on floats instead of ints by kwiberg@webrtc.org · 10 years ago
  38. 9cd8281 Add additional metric (relative standard deviation of encode time) for overuse detection. by asapersson@webrtc.org · 10 years ago
  39. f006e8d Add kjellander@webrtc.org as OWNER for *.isolate by kjellander@webrtc.org · 10 years ago
  40. bdcde22 Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at bdfcddf7091e92134143e9a2d9ccce908e43979e by Android Chromium Automerger · 10 years ago
  41. eaaba5a Create a joint encoder/decoder wrapper for iSAC in ACM by henrik.lundin@webrtc.org · 10 years ago
  42. 6da16b3 Add thread annotations to AcmReceiver by henrik.lundin@webrtc.org · 10 years ago
  43. 8097a46 Update makefiles after merge of Chromium at 275833 by Android Chromium Automerger · 10 years ago
  44. b999e11 Make some methods in Clock class const declared by henrik.lundin@webrtc.org · 10 years ago
  45. c05ab94 Remove unused test_env.py from isolate files + fix nss path. by kjellander@webrtc.org · 10 years ago
  46. 377e7fd Enables DelayCorrection tests by bjornv@webrtc.org · 10 years ago
  47. f03a4a6 Updated conformance tests and w3c-ified them. by phoglund@webrtc.org · 10 years ago
  48. 6d7c6e6 Multi-threaded unit test for Audio Coding Module using iSAC by henrik.lundin@webrtc.org · 10 years ago
  49. 4b50adf audio_processing: Forces extended filter to be used in splitting filter test. by bjornv@webrtc.org · 10 years ago
  50. e5abc85 Rename neteq4 folder to neteq by henrik.lundin@webrtc.org · 10 years ago
  51. c50f06d Re-enable AudioCodingModuleMtTest again by henrik.lundin@webrtc.org · 10 years ago
  52. 20d9f00 Update makefiles after merge of Chromium at 275661 by Android Chromium Automerger · 10 years ago
  53. 9cfa46c Revert r6358 "AppRTCDemo(Android): only stop the cameraThread's looper after stopping the camera." by fischman@webrtc.org · 10 years ago
  54. 00035af Use XErrorTrap in MouseCursorMonitorX11 to catch the error if the shared window has been closed. by jiayl@webrtc.org · 10 years ago
  55. fea960e AppRTCDemo(Android): only stop the cameraThread's looper after stopping the camera. by fischman@webrtc.org · 10 years ago
  56. d3d2598 Unbreak NDEBUG compile by RTC_UNUSED()ing an assert()d variable. by fischman@webrtc.org · 10 years ago
  57. daf186d ViEAutoTestAndroid: Unbreak compile by casting void* to jobject. by fischman@webrtc.org · 10 years ago
  58. 5101f84 AppRTCDemo(android): support app (UI) & capture rotation. by fischman@webrtc.org · 10 years ago
  59. 5befd8b VideoCaptureImpl::IncomingFrame(): avoid deadlock by acquiring _apiCs. by fischman@webrtc.org · 10 years ago
  60. 431772f Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 81f8df9af96c6b4bf43234f2a0162146a5da6112 by Android Chromium Automerger · 10 years ago
  61. bdfcddf Make VideoSendStream/VideoReceiveStream configs const. by pbos@webrtc.org · 10 years ago
  62. 6e6292d Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 00d9c49cb076626f711988332749a0ebe8d2a32f by Android Chromium Automerger · 10 years ago
  63. 64027c5 Rebase webrtc/base with r6345 version of talk/base: by henrike@webrtc.org · 10 years ago
  64. 81f8df9 Fix the chain that propagates the audio frame's rtp and ntp timestamp including: by wu@webrtc.org · 10 years ago
  65. 553b68f Opus send rate overflows if over 65 kbps by tina.legrand@webrtc.org · 10 years ago
  66. 0e7d6a6 Revert 6341 "Fixes and enables SystemDelayTests." by bjornv@webrtc.org · 10 years ago
  67. 52e8925 Fixes and enables SystemDelayTests. by bjornv@webrtc.org · 10 years ago
  68. 14ac552 NetEq: Add thread annotation to const scoped_ptrs by henrik.lundin@webrtc.org · 10 years ago
  69. 59a001f Adding back platform specific renderer to video loopback test. by mflodman@webrtc.org · 10 years ago
  70. 39dec2c The correct fix of workaround in r6261. by bjornv@webrtc.org · 10 years ago
  71. c806290 common_audio/signal_processing: Removed macro WEBRTC_SPL_MUL_16_16_RSFT_WITH_FIXROUND by bjornv@webrtc.org · 10 years ago
  72. 903e746 Have RTX be enabled by setting an RTX payload type instead of by setting an RTX SSRC. by stefan@webrtc.org · 10 years ago
  73. 00d9c49 Android: cleanup gtest_target_type conditions. by henrike@webrtc.org · 10 years ago
  74. fc94634 Make it possible to build webrtc for arm64. by solenberg@webrtc.org · 10 years ago
  75. 2b51241 Disables SystemDelayTest.CorrectDelayDuringDrift on Android by bjornv@webrtc.org · 10 years ago
  76. 625c309 Disables some modules_unittests on Android. by bjornv@webrtc.org · 10 years ago
  77. d442cf9 Moved verbose logging in rtcp_receiver.cc to LS_VERBOSE. by andresp@webrtc.org · 10 years ago
  78. fef1e23 Adding missing break in media_file_utility.cc. by mflodman@webrtc.org · 10 years ago
  79. bb2058b Enable videoprocessor_integrationtest tests on android. by marpan@webrtc.org · 10 years ago
  80. 7115998 Revert 6312 "Re-enable AudioCodingModuleMtTest" by turaj@webrtc.org · 10 years ago
  81. 6e83888 Re-enable AudioCodingModuleMtTest by henrik.lundin@webrtc.org · 10 years ago
  82. 6038f4c Update makefiles after merge of Chromium at 274467 by Android Chromium Automerger · 10 years ago
  83. baec5e7 Reformat integer accessors to look like their float counterparts by kwiberg@webrtc.org · 10 years ago
  84. 78dec3f Remove an optimization that's no longer worth the extra complexity it causes by kwiberg@webrtc.org · 10 years ago
  85. cdbeb1d - Get rid of 'using' from .h by solenberg@webrtc.org · 10 years ago
  86. bb57de4 Disable MouseCursorMonitorTest by henrik.lundin@webrtc.org · 10 years ago
  87. 1a3e45b Disable MouseCursorMonitorTest.FromScreen by henrik.lundin@webrtc.org · 10 years ago
  88. 81000de Adding thread annotations to parts of Audio Coding Module by henrik.lundin@webrtc.org · 10 years ago
  89. f4d3760 Re-enables CommonFormats test for Android. by bjornv@webrtc.org · 10 years ago
  90. 57019f2 VideoCaptureAndroid: don't synchronized on camera thread. by fischman@webrtc.org · 10 years ago
  91. 35af59e Add a Reset() method to AudioFrame. by andrew@webrtc.org · 10 years ago
  92. ab4f5eb Disable AudioCodingModuleMtTest due to memcheck and tsan failures. by andrew@webrtc.org · 10 years ago
  93. c05f045 Update makefiles after merge of Chromium at 273839 by Android Chromium Automerger · 10 years ago
  94. 58f48bb Multi-threaded test for Audio Coding Module by henrik.lundin@webrtc.org · 10 years ago
  95. 858611f Add native_test dependency to webrtc_perf_tests. by pbos@webrtc.org · 10 years ago
  96. e0beaf4 Fix bug where RTP headers in the packet history were replaced with the RTX wrapped headers. by stefan@webrtc.org · 10 years ago
  97. 4260aa2 Fixing a bug regarding VOE packet loss rate feedback to ACM by minyue@webrtc.org · 10 years ago
  98. 27de386 Revert 6272 "Update generated asm offsets scripts." by sprang@webrtc.org · 10 years ago
  99. b15df23 Increase VPMVideoDecimator's initial max_frame_rate_ to 60, which allow us potentially do 60fps. by wu@webrtc.org · 10 years ago
  100. 6f3ce73 * Revert clock.cc changes made in 6178, but keep the changes to the test. by wu@webrtc.org · 10 years ago