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gerrit-public.fairphone.software / platform / external / webrtc / 1227ab89a7c08e4e5af051a63daba889ea0d2da7
  1. 1227ab8 GN: Add BUILD.gn files + kjellander to OWNERS by kjellander@webrtc.org · 10 years ago
  2. c00ca62 Rebase webrtc/base with r6521 version of talk/base: by henrike@webrtc.org · 10 years ago
  3. 948f768 Roll libvpx 269083:278497 by fgalligan@google.com · 10 years ago
  4. b6ebe75 Disables tests that breaks Android bots by bjornv@webrtc.org · 10 years ago
  5. a36a259 TSan v2 deadlock suppressions. by kjellander@webrtc.org · 10 years ago
  6. a97f6f3 Exclude flaky libjingle_peerconnection_unittest test for Memcheck. by kjellander@webrtc.org · 10 years ago
  7. c70b2f9 Add third_party/colorama to DEPS by kjellander@webrtc.org · 10 years ago
  8. 27ab19d Roll chromium_revision 272489:277350 + fix sanitizer options by kjellander@webrtc.org · 10 years ago
  9. 78f440c GN: BUILD.gn for system_wrappers by kjellander@webrtc.org · 10 years ago
  10. ff1b1bf When creating an answer, takes the codec preference from the offer. by wu@webrtc.org · 10 years ago
  11. a24d366 - Exit from a camera thread lopper loop() method only after all camera release calls are completed. This fixes camera exceptions observed from time to time when calling camera functions on a terminated looper. by glaznev@webrtc.org · 10 years ago
  12. 0d15159 (Auto)update libjingle 69634309-> 69640360 by buildbot@webrtc.org · 10 years ago
  13. b43c99d Limits the send and receive buffer by bytes, not by packets. by jiayl@webrtc.org · 10 years ago
  14. db397e5 Re-evalutes the ICE role on ICE restart. Also unifies the logic of ICE restart. by jiayl@webrtc.org · 10 years ago
  15. 0b893b1 Do not hold the critical section in VideoCaptureAndroid::SetCaptureRotation since it would case possible deadlock with OS Camear thread. by braveyao@webrtc.org · 10 years ago
  16. bb2d658 (Auto)update libjingle 69617317-> 69623266 by buildbot@webrtc.org · 10 years ago
  17. 75ce920 (Auto)update libjingle 69600065-> 69617317 by buildbot@webrtc.org · 10 years ago
  18. f425b55 Add tests of texture frames in video_send_stream_test. by wuchengli@chromium.org · 10 years ago
  19. 83785d3 Remove unused ALLOCATE_DELAY constant. by pbos@webrtc.org · 10 years ago
  20. 4c25c67 (Auto)update libjingle 69589535-> 69600065 by buildbot@webrtc.org · 10 years ago
  21. 58e7c86 (Auto)update libjingle 69588980-> 69589535 by buildbot@webrtc.org · 10 years ago
  22. 0970dd8 (Auto)update libjingle 69588608-> 69588980 by buildbot@webrtc.org · 10 years ago
  23. 8563ef4 (Auto)update libjingle 69587333-> 69588608 by buildbot@webrtc.org · 10 years ago
  24. 1ef789d (Auto)update libjingle 69568113-> 69587333 by buildbot@webrtc.org · 10 years ago
  25. 594aefa Do not call CaptureCursor in ScreenCapturerWinGdi if no MouseShapeObserver. by jiayl@webrtc.org · 10 years ago
  26. df9bbbe (Auto)update libjingle 69567902-> 69568113 by buildbot@webrtc.org · 10 years ago
  27. fbd1328 (Auto)update libjingle 69555283-> 69567902 by buildbot@webrtc.org · 10 years ago
  28. 21794f9 (Auto)update libjingle 69543894-> 69555283 by buildbot@webrtc.org · 10 years ago
  29. 304ca76 Revert 6481 and 6482 by fgalligan@google.com · 10 years ago
  30. 8de8c91 Maintain constantness of the input to iSAC-fix decoder, and prevent heap-buffer overflow. by turaj@webrtc.org · 10 years ago
  31. 9158df2 Adding an empty constructor implementation to the AudioSink class by henrik.lundin@webrtc.org · 10 years ago
  32. 84f8ec1 Changes to tests and tools in audio_processing. by bjornv@webrtc.org · 10 years ago
  33. 077593b Ensure that the start bitrate can be set multiple times. by stefan@webrtc.org · 10 years ago
  34. 496a984 Adding test::AudioSink interface and derived classes by henrik.lundin@webrtc.org · 10 years ago
  35. 5c3f4e3 Fixes and re-enables tests disabled on Android by bjornv@webrtc.org · 10 years ago
  36. d27d9ae (Auto)update libjingle 69506154-> 69515138 by buildbot@webrtc.org · 10 years ago
  37. 6ce1d58 Exclude flaky test PeerConnectionEndToEndTest.CreateDataChannelAfterNegotiate on memcheck. by jiayl@webrtc.org · 10 years ago
  38. acede34 Fix a memory leak in SctpDataMediaChannelTest. by jiayl@webrtc.org · 10 years ago
  39. 85b19a1 Exclude SctpDataMediaChannelTest on Win DrMemory for third_party/usrsctp issues. by jiayl@webrtc.org · 10 years ago
  40. f8063d3 Properly shut down the SCTP stack. by jiayl@webrtc.org · 10 years ago
  41. a19b930 Update webrtc to fix unpack_lib expansion. by fgalligan@google.com · 10 years ago
  42. 8f06a8a Update generated asm offsets scripts. by fgalligan@google.com · 10 years ago
  43. b947d95 Neon version of FilterAdaptation() by bjornv@webrtc.org · 10 years ago
  44. 12396ab Update PacketSource and RtpFileSource by henrik.lundin@webrtc.org · 10 years ago
  45. d8de066 Revert "Restore ptypes.txt file" by henrik.lundin@webrtc.org · 10 years ago
  46. ec869bf Revert 6473 "Update generated asm offsets scripts." by turaj@webrtc.org · 10 years ago
  47. e398954 Update usrsctp to r8875 by jiayl@webrtc.org · 10 years ago
  48. 32196de Update generated asm offsets scripts. by fgalligan@google.com · 10 years ago
  49. a15fbfd Add round-robin selection of send stream to pad on. by stefan@webrtc.org · 10 years ago
  50. 9c09e6e Add high perf mode to VP8 by niklas.enbom@webrtc.org · 10 years ago
  51. 26eaf7c Add a check to all.gyp to respect the include_tests variable. by andrew@webrtc.org · 10 years ago
  52. 2eaac18 Makes the sid of a closed DataChannel available to reuse per the spec. by jiayl@webrtc.org · 10 years ago
  53. a685c9d base: Renaming + conforming: post commit review changes for https://webrtc-codereview.appspot.com/17699005/ by henrike@webrtc.org · 10 years ago
  54. 5654b30 Rebase webrtc/base with r6464 version of talk/base: by henrike@webrtc.org · 10 years ago
  55. d469443 Rolling new version of opus.gyp by tina.legrand@webrtc.org · 10 years ago
  56. ed3e0d8 Increasing tolerances quite a bit to fight flakes. by phoglund@webrtc.org · 10 years ago
  57. ae740dd (Auto)update libjingle 69359922-> 69365993 by buildbot@webrtc.org · 10 years ago
  58. d42da54 Revert 6458 "Since NetEq4 is ready to handle 48 kHz codec, it is..." by minyue@webrtc.org · 10 years ago
  59. 851a09e Initial GN work for WebRTC by kjellander@webrtc.org · 10 years ago
  60. 2ca2188 Restore ptypes.txt file by henrik.lundin@webrtc.org · 10 years ago
  61. 6b06142 Updated W3C getusermedia tests to the latest version of the spec. by phoglund@webrtc.org · 10 years ago
  62. 8f8503d Since NetEq4 is ready to handle 48 kHz codec, it is good to remove the 48-to-32kHz downsampling of Opus output. This facilitates webrtc to make full use of Opus's bandwidth and eliminates unneeded computation in resampling. by minyue@webrtc.org · 10 years ago
  63. 44a317a (Auto)update libjingle 69337301-> 69359922 by buildbot@webrtc.org · 10 years ago
  64. 9f36c08 Makes it possible to prevent some third party libraries (jsoncpp and openssl) from being linked. This makes it possible to link webrtc with external implementations of those libraries in case the project depending on webrtc requires another version of those libraries. by henrike@webrtc.org · 10 years ago
  65. 53f5793 (Auto)update libjingle 69306183-> 69323802 by buildbot@webrtc.org · 10 years ago
  66. 587ef60 Implement RTP extension support in WebRtcVideoEngine2. by pbos@webrtc.org · 10 years ago
  67. d054bff (Auto)update libjingle 69292418-> 69293749 by buildbot@webrtc.org · 10 years ago
  68. d980307 Add max limit of number for overuses. When limit is reached always apply the rampup delay. by asapersson@webrtc.org · 10 years ago
  69. 88d9fa6 (Auto)update libjingle 69291002-> 69292418 by buildbot@webrtc.org · 10 years ago
  70. 4b12d40 Add SDES, APP, IJ, SLI and PLI packet types to RTCP packet class. by asapersson@webrtc.org · 10 years ago
  71. 27626a6 (Auto)update libjingle 69278008-> 69291002 by buildbot@webrtc.org · 10 years ago
  72. d6e2213 Remove ivinnichenko from webrtc/test/OWNERS by kjellander@webrtc.org · 10 years ago
  73. 1e3c5c2 Importing ThreadChecker class from Chromium by henrik.lundin@webrtc.org · 10 years ago
  74. b099a6f Adds aluebs@webrtc.org as owner to audio_processing by bjornv@webrtc.org · 10 years ago
  75. 721f970 common_audio: Removes macro WEBRTC_SPL_LSHIFT_U16 by bjornv@webrtc.org · 10 years ago
  76. eb16b81 Implements start bitrate for new video API. by mflodman@webrtc.org · 10 years ago
  77. 0a1e7e0 (Auto)update libjingle 69276003-> 69278008 by buildbot@webrtc.org · 10 years ago
  78. 63e4607 Add thread annotations to parts of ACMGenericCodec by henrik.lundin@webrtc.org · 10 years ago
  79. 249211e Disable flaky test (WebRtcVideoMediaChannelTest.GetStats) on DrMemory Full. by asapersson@webrtc.org · 10 years ago
  80. d159140 (Auto)update libjingle 69260070-> 69276003 by buildbot@webrtc.org · 10 years ago
  81. 2bae321 Add missing sources to webrtc/base/base.gyp by kjellander@webrtc.org · 10 years ago
  82. 117afee (Auto)update libjingle 69188577-> 69260070 by buildbot@webrtc.org · 10 years ago
  83. ab23d49 Add glaznev@ to OWNERS for webrtc/modules/video_capture and talk/app/webrtc. by glaznev@webrtc.org · 10 years ago
  84. c6c1dfd Add extra logging and latency restriction to VP8 HW encoder. by glaznev@webrtc.org · 10 years ago
  85. a6764ab (Auto)update libjingle 69144530-> 69164179 by buildbot@webrtc.org · 10 years ago
  86. af6f02f Neon version of OverdriveAndSuppress() by bjornv@webrtc.org · 10 years ago
  87. db56390 (Auto)update libjingle 69143161-> 69144530 by buildbot@webrtc.org · 10 years ago
  88. f99c2f2 Add NACK feedback parameter to WebRtcVideoEngine2. by pbos@webrtc.org · 10 years ago
  89. e322a17 Implement RTX tests+fixes in WebRtcVideoEngine2. by pbos@webrtc.org · 10 years ago
  90. 9fbb717 Remove engine_codecs_ cache from unittests. by pbos@webrtc.org · 10 years ago
  91. d54ec12 Fix GYP DEPTH for libjingle isolate files by kjellander@webrtc.org · 10 years ago
  92. a1bfc50 Pass GYP DEPTH variable to isolate. by kjellander@webrtc.org · 10 years ago
  93. c800c1c (Auto)update libjingle 69131548-> 69132244 by buildbot@webrtc.org · 10 years ago
  94. 1c8223c Initial owners file for talk/media/webrtc/. by pbos@webrtc.org · 10 years ago
  95. 7e71b77 (Auto)update libjingle 69102234-> 69116997 by buildbot@webrtc.org · 10 years ago
  96. 8e256ee Revert 6415 "Update generated asm offsets scripts." by wu@webrtc.org · 10 years ago
  97. 1a6c628 Revert r6420 'Revert r6390 "Adds end to end DataChannel tests." Flaky on linux_memcheck' by jiayl@webrtc.org · 10 years ago
  98. 3c13ed3 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
  99. ddeec04 Revert r6390 "Adds end to end DataChannel tests." Flaky on linux_memcheck by jiayl@webrtc.org · 10 years ago
  100. 3f3f428 (Auto)update libjingle 69097619-> 69099564 by buildbot@webrtc.org · 10 years ago