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gerrit-public.fairphone.software / platform / external / webrtc / 994d0b7229a18b255d81979c2bedaf8ecfae9bd7
  1. 994d0b7 Refactor Call-based tests. by pbos@webrtc.org · 11 years ago
  2. 35d46fb Roll chromium_revision 277350:280149 by kjellander@webrtc.org · 11 years ago
  3. c8e9818 Receiver bit-exactness test for AudioCoding Module by henrik.lundin@webrtc.org · 11 years ago
  4. 7ea71de clock.h: Removed GUARDED_BY annotation as it breaks som builds. by henrike@webrtc.org · 11 years ago
  5. 1d1e40f Add Chromium's src/buildtools to DEPS. by kjellander@webrtc.org · 11 years ago
  6. 19db3e3 Don't forward declare RWLockWrapper in clock.h by henrik.lundin@webrtc.org · 11 years ago
  7. aa0e56e Fixes a bug causing NACKs to be dropped excessively at the send-side. by stefan@webrtc.org · 11 years ago
  8. 269605c Implement SetSendCodecs() unit tests for WebRtcVideoChannel2. by pbos@webrtc.org · 11 years ago
  9. 420ca43 (Auto)update libjingle 69860953-> 70002228 by buildbot@webrtc.org · 11 years ago
  10. a2142ca Bump version number to 3.55 by tnakamura@webrtc.org · 11 years ago
  11. fe526ff fix after r6472 in rtp_sender, comparison between signed and unsigned integer expressions. by henrike@webrtc.org · 11 years ago
  12. 4ddcc40 pkg-config-wrapper should not be run when build_nss is disabled (=0). by henrike@webrtc.org · 11 years ago
  13. 3b84b3a Add RTCP packet types to packet builder: by asapersson@webrtc.org · 11 years ago
  14. 6568e97 This is to compare NetEq with various codecs under a shared packet loss pattern. by minyue@webrtc.org · 11 years ago
  15. d5075bd Neon version of FilterFar() by bjornv@webrtc.org · 11 years ago
  16. 1ed1af9 Remove payload duplication in AudioDecoderTest by henrik.lundin@webrtc.org · 11 years ago
  17. ec9f5fb Change SdpSerializeCandidate to output candidate line without the "a=" and without the leading \r\n", i.e. candidate-attribute as defined in section 15.1 of [ICE]. by wu@webrtc.org · 11 years ago
  18. 1da152d talk/base and webrtc/base suppression had the same names for their suppressions which is not allowed. Renamed the talk/base ones as they are going away. by henrike@webrtc.org · 11 years ago
  19. eecf5e6 Removing neteq decode lock and friends by henrik.lundin@webrtc.org · 11 years ago
  20. 05f1464 Exclude AsyncWriteTest.TestWrite from Win DrMemory Full bot and suppress the reported errors by aluebs@webrtc.org · 11 years ago
  21. 04fbc38 Neon version of ScaleErrorSignal() by bjornv@webrtc.org · 11 years ago
  22. 9a4f651 Disable PhysicalSocketTest.TestUdpReadyToSendIPv4 for TSAN2 by aluebs@webrtc.org · 11 years ago
  23. 71dffb7 (Auto)update libjingle 69648312-> 69830415 by buildbot@webrtc.org · 11 years ago
  24. b338ca6 Annotating the rest of AcmGenericCodec by henrik.lundin@webrtc.org · 11 years ago
  25. f6d37de Fix array declarations in aec_core.c by andrew@webrtc.org · 11 years ago
  26. ceb5a1d Annotating the rest of AudioCodingModuleImpl by henrik.lundin@webrtc.org · 11 years ago
  27. 1227ab8 GN: Add BUILD.gn files + kjellander to OWNERS by kjellander@webrtc.org · 11 years ago
  28. c00ca62 Rebase webrtc/base with r6521 version of talk/base: by henrike@webrtc.org · 11 years ago
  29. 948f768 Roll libvpx 269083:278497 by fgalligan@google.com · 11 years ago
  30. b6ebe75 Disables tests that breaks Android bots by bjornv@webrtc.org · 11 years ago
  31. a36a259 TSan v2 deadlock suppressions. by kjellander@webrtc.org · 11 years ago
  32. a97f6f3 Exclude flaky libjingle_peerconnection_unittest test for Memcheck. by kjellander@webrtc.org · 11 years ago
  33. c70b2f9 Add third_party/colorama to DEPS by kjellander@webrtc.org · 11 years ago
  34. 27ab19d Roll chromium_revision 272489:277350 + fix sanitizer options by kjellander@webrtc.org · 11 years ago
  35. 78f440c GN: BUILD.gn for system_wrappers by kjellander@webrtc.org · 11 years ago
  36. ff1b1bf When creating an answer, takes the codec preference from the offer. by wu@webrtc.org · 11 years ago
  37. 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 · 11 years ago
  38. 0d15159 (Auto)update libjingle 69634309-> 69640360 by buildbot@webrtc.org · 11 years ago
  39. b43c99d Limits the send and receive buffer by bytes, not by packets. by jiayl@webrtc.org · 11 years ago
  40. db397e5 Re-evalutes the ICE role on ICE restart. Also unifies the logic of ICE restart. by jiayl@webrtc.org · 11 years ago
  41. 0b893b1 Do not hold the critical section in VideoCaptureAndroid::SetCaptureRotation since it would case possible deadlock with OS Camear thread. by braveyao@webrtc.org · 11 years ago
  42. bb2d658 (Auto)update libjingle 69617317-> 69623266 by buildbot@webrtc.org · 11 years ago
  43. 75ce920 (Auto)update libjingle 69600065-> 69617317 by buildbot@webrtc.org · 11 years ago
  44. f425b55 Add tests of texture frames in video_send_stream_test. by wuchengli@chromium.org · 11 years ago
  45. 83785d3 Remove unused ALLOCATE_DELAY constant. by pbos@webrtc.org · 11 years ago
  46. 4c25c67 (Auto)update libjingle 69589535-> 69600065 by buildbot@webrtc.org · 11 years ago
  47. 58e7c86 (Auto)update libjingle 69588980-> 69589535 by buildbot@webrtc.org · 11 years ago
  48. 0970dd8 (Auto)update libjingle 69588608-> 69588980 by buildbot@webrtc.org · 11 years ago
  49. 8563ef4 (Auto)update libjingle 69587333-> 69588608 by buildbot@webrtc.org · 11 years ago
  50. 1ef789d (Auto)update libjingle 69568113-> 69587333 by buildbot@webrtc.org · 11 years ago
  51. 594aefa Do not call CaptureCursor in ScreenCapturerWinGdi if no MouseShapeObserver. by jiayl@webrtc.org · 11 years ago
  52. df9bbbe (Auto)update libjingle 69567902-> 69568113 by buildbot@webrtc.org · 11 years ago
  53. fbd1328 (Auto)update libjingle 69555283-> 69567902 by buildbot@webrtc.org · 11 years ago
  54. 21794f9 (Auto)update libjingle 69543894-> 69555283 by buildbot@webrtc.org · 11 years ago
  55. 304ca76 Revert 6481 and 6482 by fgalligan@google.com · 11 years ago
  56. 8de8c91 Maintain constantness of the input to iSAC-fix decoder, and prevent heap-buffer overflow. by turaj@webrtc.org · 11 years ago
  57. 9158df2 Adding an empty constructor implementation to the AudioSink class by henrik.lundin@webrtc.org · 11 years ago
  58. 84f8ec1 Changes to tests and tools in audio_processing. by bjornv@webrtc.org · 11 years ago
  59. 077593b Ensure that the start bitrate can be set multiple times. by stefan@webrtc.org · 11 years ago
  60. 496a984 Adding test::AudioSink interface and derived classes by henrik.lundin@webrtc.org · 11 years ago
  61. 5c3f4e3 Fixes and re-enables tests disabled on Android by bjornv@webrtc.org · 11 years ago
  62. d27d9ae (Auto)update libjingle 69506154-> 69515138 by buildbot@webrtc.org · 11 years ago
  63. 6ce1d58 Exclude flaky test PeerConnectionEndToEndTest.CreateDataChannelAfterNegotiate on memcheck. by jiayl@webrtc.org · 11 years ago
  64. acede34 Fix a memory leak in SctpDataMediaChannelTest. by jiayl@webrtc.org · 11 years ago
  65. 85b19a1 Exclude SctpDataMediaChannelTest on Win DrMemory for third_party/usrsctp issues. by jiayl@webrtc.org · 11 years ago
  66. f8063d3 Properly shut down the SCTP stack. by jiayl@webrtc.org · 11 years ago
  67. a19b930 Update webrtc to fix unpack_lib expansion. by fgalligan@google.com · 11 years ago
  68. 8f06a8a Update generated asm offsets scripts. by fgalligan@google.com · 11 years ago
  69. b947d95 Neon version of FilterAdaptation() by bjornv@webrtc.org · 11 years ago
  70. 12396ab Update PacketSource and RtpFileSource by henrik.lundin@webrtc.org · 11 years ago
  71. d8de066 Revert "Restore ptypes.txt file" by henrik.lundin@webrtc.org · 11 years ago
  72. ec869bf Revert 6473 "Update generated asm offsets scripts." by turaj@webrtc.org · 11 years ago
  73. e398954 Update usrsctp to r8875 by jiayl@webrtc.org · 11 years ago
  74. 32196de Update generated asm offsets scripts. by fgalligan@google.com · 11 years ago
  75. a15fbfd Add round-robin selection of send stream to pad on. by stefan@webrtc.org · 11 years ago
  76. 9c09e6e Add high perf mode to VP8 by niklas.enbom@webrtc.org · 11 years ago
  77. 26eaf7c Add a check to all.gyp to respect the include_tests variable. by andrew@webrtc.org · 11 years ago
  78. 2eaac18 Makes the sid of a closed DataChannel available to reuse per the spec. by jiayl@webrtc.org · 11 years ago
  79. a685c9d base: Renaming + conforming: post commit review changes for https://webrtc-codereview.appspot.com/17699005/ by henrike@webrtc.org · 11 years ago
  80. 5654b30 Rebase webrtc/base with r6464 version of talk/base: by henrike@webrtc.org · 11 years ago
  81. d469443 Rolling new version of opus.gyp by tina.legrand@webrtc.org · 11 years ago
  82. ed3e0d8 Increasing tolerances quite a bit to fight flakes. by phoglund@webrtc.org · 11 years ago
  83. ae740dd (Auto)update libjingle 69359922-> 69365993 by buildbot@webrtc.org · 11 years ago
  84. d42da54 Revert 6458 "Since NetEq4 is ready to handle 48 kHz codec, it is..." by minyue@webrtc.org · 11 years ago
  85. 851a09e Initial GN work for WebRTC by kjellander@webrtc.org · 11 years ago
  86. 2ca2188 Restore ptypes.txt file by henrik.lundin@webrtc.org · 11 years ago
  87. 6b06142 Updated W3C getusermedia tests to the latest version of the spec. by phoglund@webrtc.org · 11 years ago
  88. 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 · 11 years ago
  89. 44a317a (Auto)update libjingle 69337301-> 69359922 by buildbot@webrtc.org · 11 years ago
  90. 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 · 11 years ago
  91. 53f5793 (Auto)update libjingle 69306183-> 69323802 by buildbot@webrtc.org · 11 years ago
  92. 587ef60 Implement RTP extension support in WebRtcVideoEngine2. by pbos@webrtc.org · 11 years ago
  93. d054bff (Auto)update libjingle 69292418-> 69293749 by buildbot@webrtc.org · 11 years ago
  94. d980307 Add max limit of number for overuses. When limit is reached always apply the rampup delay. by asapersson@webrtc.org · 11 years ago
  95. 88d9fa6 (Auto)update libjingle 69291002-> 69292418 by buildbot@webrtc.org · 11 years ago
  96. 4b12d40 Add SDES, APP, IJ, SLI and PLI packet types to RTCP packet class. by asapersson@webrtc.org · 11 years ago
  97. 27626a6 (Auto)update libjingle 69278008-> 69291002 by buildbot@webrtc.org · 11 years ago
  98. d6e2213 Remove ivinnichenko from webrtc/test/OWNERS by kjellander@webrtc.org · 11 years ago
  99. 1e3c5c2 Importing ThreadChecker class from Chromium by henrik.lundin@webrtc.org · 11 years ago
  100. b099a6f Adds aluebs@webrtc.org as owner to audio_processing by bjornv@webrtc.org · 11 years ago