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gerrit-public.fairphone.software / platform / external / webrtc / 241a9b0b65937660f07066cd802c6aab78b3cd94
  1. 241a9b0 Fixing compile error. by phoglund@webrtc.org · 10 years ago
  2. 22292df Adding explicit check for using dummy file devices. by phoglund@webrtc.org · 10 years ago
  3. 33d110d Tight data race suppressions around thread_posix. by andresp@webrtc.org · 10 years ago
  4. af38f4e Extract RTP-header SSRC inline in Call. by pbos@webrtc.org · 10 years ago
  5. a70be68 Disabling shared socket mode for TURN ports. This is done as currently when by mallinath@webrtc.org · 10 years ago
  6. 3c637cd Clean data races from system_wrappers_unittests. by andresp@webrtc.org · 10 years ago
  7. 285e9bc Fix potential deadlock in webrtc/system_wrappers/source/logging_unittest.cc. by andresp@webrtc.org · 10 years ago
  8. 5f2c81c webrtc/base: Fixes miss in base.gyp for windows. See https://code.google.com/p/webrtc/source/browse/trunk/talk/libjingle.gyp?r=6503#764 for the corresponding condition. by henrike@webrtc.org · 10 years ago
  9. ba93f9a drmemory flaky: EndToEndTest.RestartingSendStreamPreservesRtpState[WithRtx] suppressed on drMemory. by henrike@webrtc.org · 10 years ago
  10. 161f808 Add test for VideoEncoder setup/teardown. by pbos@webrtc.org · 10 years ago
  11. 2bb1bda Preserve RTP states for restarted VideoSendStreams. by pbos@webrtc.org · 10 years ago
  12. 73823ca Add initial gn build files for video_coding and video_processing. by stefan@webrtc.org · 10 years ago
  13. 03c817e Fix pacer to accept duplicate sequence numbers on different SSRCs. by pbos@webrtc.org · 10 years ago
  14. b941fe8 Fix data races related with traces in bitrate estimator test. by andresp@webrtc.org · 10 years ago
  15. bd249bc Remove GetDefaultConfigs() from Call. by pbos@webrtc.org · 10 years ago
  16. 7832648 Add missing break introduced in r6603. by stefan@webrtc.org · 10 years ago
  17. bee164a Fix test issues and a win compile error introduced with r6605. by stefan@webrtc.org · 10 years ago
  18. 875ad49 Revert conversion from TickTime to int64_t in paced sender. by stefan@webrtc.org · 10 years ago
  19. 8faa5db Add pbos@webrtc.org as owner for webrtc/test/. by pbos@webrtc.org · 10 years ago
  20. b9f5453 Add boilerplate code for H.264. by stefan@webrtc.org · 10 years ago
  21. d8440f7 Have Opus follow Chromium revisions by tina.legrand@webrtc.org · 10 years ago
  22. 20c1f56 Configure RTX send status on new modules. by pbos@webrtc.org · 10 years ago
  23. 88e0dda Introduces PacedVideoSender to test framework and moves the Pacer to use Clock. by stefan@webrtc.org · 10 years ago
  24. 614000d Adding pbos as video/ owner and removing persons never working with this folder. by mflodman@webrtc.org · 10 years ago
  25. c5e53dd Revert 6597 "Roll chromium_revision 280876:281094" by kjellander@webrtc.org · 10 years ago
  26. cb1df98 Roll chromium_revision 280876:281094 by kjellander@webrtc.org · 10 years ago
  27. 720964f Fix memcheck error in r6594. by marpan@webrtc.org · 10 years ago
  28. 11bea89 GN: Implement BUILD.gn for common_video. by kjellander@webrtc.org · 10 years ago
  29. c836453 Fix for FEC decoding with sequence number wrap-around. by marpan@webrtc.org · 10 years ago
  30. 69ef991 delay_estimator: Allows dynamically used history sizes by bjornv@webrtc.org · 10 years ago
  31. 224a140 Make experimental NS API not purely virtual by aluebs@webrtc.org · 10 years ago
  32. c0ba439 common_audio: Removes macro WEBRTC_SPL_SHIFT_W16 by bjornv@webrtc.org · 10 years ago
  33. 38214d5 EchoCancellationImpl::ProcessRenderAudio: Use float samples directly by kwiberg@webrtc.org · 10 years ago
  34. a82f9a2 Add Tsan2 to .gitignore by andresp@webrtc.org · 10 years ago
  35. dfdaeb9 Removed old code and default implementations. by asapersson@webrtc.org · 10 years ago
  36. 9c89e93 WebRTCDemo: set local SSRC for loopback test, otherwise receiver would reset it due to ssrc clash, which would cause delayed remote rendering. by braveyao@webrtc.org · 10 years ago
  37. 3ffa1f9 (Auto)update libjingle 70422491-> 70424781 by buildbot@webrtc.org · 10 years ago
  38. b25b08b Remove tools/resources by kjellander@webrtc.org · 10 years ago
  39. 93426cd Implement BUILD.gn for desktop_capture. by jiayl@webrtc.org · 10 years ago
  40. 33586c8 Make deadlock suppressions less generic. by andresp@webrtc.org · 10 years ago
  41. 1295dc6 Possibly fix deadlock happening due to unregister/register modules as switching between AST and TSO estimators. by andresp@webrtc.org · 10 years ago
  42. 0bb9fac (Auto)update libjingle 70343444-> 70394475 by buildbot@webrtc.org · 10 years ago
  43. 8956980 Roll chromium 280149:280876. by marpan@webrtc.org · 10 years ago
  44. d8a9069 (Auto)update libjingle 70340027-> 70343444 by buildbot@webrtc.org · 10 years ago
  45. 74bf7a6 Add tkchin@ to OWNERS. by tkchin@webrtc.org · 10 years ago
  46. 974bbbb Fix uninitialized value in DtlsTransport and TransportDescription. by jiayl@webrtc.org · 10 years ago
  47. 0856454 roll libyuv to r1025 for mips n32 support, arm nacl port, psnr tool jpeg support. by fbarchard@google.com · 10 years ago
  48. 6335645 (Auto)update libjingle 70329914-> 70330023 by buildbot@webrtc.org · 10 years ago
  49. 37b4e1b webrtc/base: add dependent setting for gtest include directory that was missed when creating base_tests.gyp. Same as https://code.google.com/p/webrtc/source/browse/trunk/talk/libjingle_tests.gyp?r=6484#39 by henrike@webrtc.org · 10 years ago
  50. 0402515 Implement command line flags for peerconnection client example on Windows by kjellander@webrtc.org · 10 years ago
  51. 9138eb6 Fix compile error introduced with r6571. by stefan@webrtc.org · 10 years ago
  52. 5779ca4 Fixes a potential BWE clock mismatch bug. by stefan@webrtc.org · 10 years ago
  53. 6d21ddc audio_processing/aec: Refactors NonLinearProcessing to prepare for NEON optimizations by bjornv@webrtc.org · 10 years ago
  54. d5a0506 Use X509_NAME, not struct X509_name_st. by henrike@webrtc.org · 10 years ago
  55. 59adb1d Neon version of cftmdl_128() by bjornv@webrtc.org · 10 years ago
  56. 9825afc Add ExperimentalNs support in Config by aluebs@webrtc.org · 10 years ago
  57. 2be53a3 Disable CanSwitchToUseAllSsrcs on DrMemory. by pbos@webrtc.org · 10 years ago
  58. be9d2a4 Reserve RTP/RTCP modules in SetSSRC. by pbos@webrtc.org · 10 years ago
  59. cd9b90a Neon version of cft1st_128() by bjornv@webrtc.org · 10 years ago
  60. e9b9ec5 Removing W3C conformance tests after move to web-platform-tests. by phoglund@webrtc.org · 10 years ago
  61. ae7cfd7 Make MediaOptimization thread-safe. by wuchengli@chromium.org · 10 years ago
  62. 62711f8 GN: Fix build by disabling compiler warning in base. by kjellander@webrtc.org · 10 years ago
  63. 7497fa7 GN: Refactor base/BUILD.gn and fix dbus-glib error. by kjellander@webrtc.org · 10 years ago
  64. b3c188f Use the libvpx rev from Chromium's DEPS, not the Chromium rev. by andrew@webrtc.org · 10 years ago
  65. ee4e466 Roll libvpx: follow the Chromium revision. by marpan@webrtc.org · 10 years ago
  66. 6f833c3 Rebase webrtc/base with r6555 version of talk/base: by henrike@webrtc.org · 10 years ago
  67. bfa758a (Auto)update libjingle 70004190-> 70103367 by buildbot@webrtc.org · 10 years ago
  68. 680555f constructormagic.h macros are duplicated in several repositories. undef them in webrtc to prevent conflict for some build configurations. by henrike@webrtc.org · 10 years ago
  69. f4d6d7c Add DrMemory suppression for AsyncWriteTest by aluebs@webrtc.org · 10 years ago
  70. 767d98e TSan: Move suppressions to source file. by kjellander@webrtc.org · 10 years ago
  71. 994d0b7 Refactor Call-based tests. by pbos@webrtc.org · 10 years ago
  72. 35d46fb Roll chromium_revision 277350:280149 by kjellander@webrtc.org · 10 years ago
  73. c8e9818 Receiver bit-exactness test for AudioCoding Module by henrik.lundin@webrtc.org · 10 years ago
  74. 7ea71de clock.h: Removed GUARDED_BY annotation as it breaks som builds. by henrike@webrtc.org · 10 years ago
  75. 1d1e40f Add Chromium's src/buildtools to DEPS. by kjellander@webrtc.org · 10 years ago
  76. 19db3e3 Don't forward declare RWLockWrapper in clock.h by henrik.lundin@webrtc.org · 10 years ago
  77. aa0e56e Fixes a bug causing NACKs to be dropped excessively at the send-side. by stefan@webrtc.org · 10 years ago
  78. 269605c Implement SetSendCodecs() unit tests for WebRtcVideoChannel2. by pbos@webrtc.org · 10 years ago
  79. 420ca43 (Auto)update libjingle 69860953-> 70002228 by buildbot@webrtc.org · 10 years ago
  80. a2142ca Bump version number to 3.55 by tnakamura@webrtc.org · 10 years ago
  81. fe526ff fix after r6472 in rtp_sender, comparison between signed and unsigned integer expressions. by henrike@webrtc.org · 10 years ago
  82. 4ddcc40 pkg-config-wrapper should not be run when build_nss is disabled (=0). by henrike@webrtc.org · 10 years ago
  83. 3b84b3a Add RTCP packet types to packet builder: by asapersson@webrtc.org · 10 years ago
  84. 6568e97 This is to compare NetEq with various codecs under a shared packet loss pattern. by minyue@webrtc.org · 10 years ago
  85. d5075bd Neon version of FilterFar() by bjornv@webrtc.org · 10 years ago
  86. 1ed1af9 Remove payload duplication in AudioDecoderTest by henrik.lundin@webrtc.org · 10 years ago
  87. 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 · 10 years ago
  88. 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 · 10 years ago
  89. eecf5e6 Removing neteq decode lock and friends by henrik.lundin@webrtc.org · 10 years ago
  90. 05f1464 Exclude AsyncWriteTest.TestWrite from Win DrMemory Full bot and suppress the reported errors by aluebs@webrtc.org · 10 years ago
  91. 04fbc38 Neon version of ScaleErrorSignal() by bjornv@webrtc.org · 10 years ago
  92. 9a4f651 Disable PhysicalSocketTest.TestUdpReadyToSendIPv4 for TSAN2 by aluebs@webrtc.org · 10 years ago
  93. 71dffb7 (Auto)update libjingle 69648312-> 69830415 by buildbot@webrtc.org · 10 years ago
  94. b338ca6 Annotating the rest of AcmGenericCodec by henrik.lundin@webrtc.org · 10 years ago
  95. f6d37de Fix array declarations in aec_core.c by andrew@webrtc.org · 10 years ago
  96. ceb5a1d Annotating the rest of AudioCodingModuleImpl by henrik.lundin@webrtc.org · 10 years ago
  97. 1227ab8 GN: Add BUILD.gn files + kjellander to OWNERS by kjellander@webrtc.org · 10 years ago
  98. c00ca62 Rebase webrtc/base with r6521 version of talk/base: by henrike@webrtc.org · 10 years ago
  99. 948f768 Roll libvpx 269083:278497 by fgalligan@google.com · 10 years ago
  100. b6ebe75 Disables tests that breaks Android bots by bjornv@webrtc.org · 10 years ago