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gerrit-public.fairphone.software / platform / external / webrtc / b01ce14b13f10b72ea419b269ce9441f77ee2239
  1. b01ce14 add some comments about DEPS lkgr for chromium by fbarchard@google.com · 11 years ago
  2. c9b5072 DrMemory suppression due to r6811. by henrike@webrtc.org · 11 years ago
  3. ee135f7 Memcheck suppression. Re-suppress 3478 suppression after namespace change from talk_base to rtc. by henrike@webrtc.org · 11 years ago
  4. a27342b (Auto)update libjingle 72446860-> 72550257 by buildbot@webrtc.org · 11 years ago
  5. 0040a6e This is a setup to solve https://code.google.com/p/webrtc/issues/detail?id=1906 by minyue@webrtc.org · 11 years ago
  6. 84b9e1e Fix for retransmission. Base layer packets were not retransmitted. by asapersson@webrtc.org · 11 years ago
  7. e0d03f1 (Auto)update libjingle 72443101-> 72446860 by buildbot@webrtc.org · 11 years ago
  8. 6e203d5 (Auto)update libjingle 72442050-> 72443101 by buildbot@webrtc.org · 11 years ago
  9. 52148c2 (Auto)update libjingle 72430895-> 72442050 by buildbot@webrtc.org · 11 years ago
  10. 7cb60cc (Auto)update libjingle 72407428-> 72430895 by buildbot@webrtc.org · 11 years ago
  11. 3bc4824 (Auto)update libjingle 72403605-> 72407428 by buildbot@webrtc.org · 11 years ago
  12. 6955213 (Auto)update libjingle 72389720-> 72403605 by buildbot@webrtc.org · 11 years ago
  13. 42d65ce Fix memory leak in FakeSSLCertificate::GetChain(), discovered by Linux Memcheck build/try bots. by solenberg@webrtc.org · 11 years ago
  14. 1a678c6 (Auto)update libjingle 72320533-> 72380285 by buildbot@webrtc.org · 11 years ago
  15. 6b21b71 (Auto)update libjingle 72205295-> 72320533 by buildbot@webrtc.org · 11 years ago
  16. e1c9caf Fix mistake in rtp/rtcp/BUILD.gn introduced with r6804. by stefan@webrtc.org · 11 years ago
  17. 2ec5606 Add H.264 packetization. by stefan@webrtc.org · 11 years ago
  18. bfe6e08 Add simulation of network effects to video_loopback tool. by stefan@webrtc.org · 11 years ago
  19. d9843da libjingle: stop building files in talk/base as they are no longer used as of r6799 by henrike@webrtc.org · 11 years ago
  20. 48305f5 Disable warning 4702 which affects map, xlist and others on vs2012 and vs2013. by fbarchard@google.com · 11 years ago
  21. 901debd roll libyuv to r1038 from r1035 to add gyp define that makes jpeg optional. by fbarchard@google.com · 11 years ago
  22. d4e598d (Auto)update libjingle 72097588-> 72159069 by buildbot@webrtc.org · 11 years ago
  23. fc8b087 Remove dependency on openssl for android, add dependency on boringssl. Should make Android bots green again. by solenberg@webrtc.org · 11 years ago
  24. fdbe144 Use C functions in aec for MIPS by andrew@webrtc.org · 11 years ago
  25. e75d78d Integrate rtcp packet class to rtcp receiver tests. by asapersson@webrtc.org · 11 years ago
  26. 1e7d60e merge_libs.py: fixes Windows breakage: there should be no space after "lib /OUT:". by henrike@webrtc.org · 11 years ago
  27. 51c5508 (Auto)update libjingle 72016417-> 72097588 by buildbot@webrtc.org · 11 years ago
  28. 8aed945 Remove a disabled test. by pbos@webrtc.org · 11 years ago
  29. 4fe98a9 Remove clang-format rm_binaries.py DEPS entry. by pbos@webrtc.org · 11 years ago
  30. 961293d webrtc/base: FileModifyTime -> OlderThan as that's what it was ever used as. Needed for cl/70828325. by henrike@webrtc.org · 11 years ago
  31. af9e794 Fix compilation on windows with clang, indentation cleanups by sergeyu@chromium.org · 11 years ago
  32. 257e130 Set NACK/REMB when setting receive codecs. by pbos@webrtc.org · 11 years ago
  33. 3155f2b Roll chromium 282879:285412. by fgalligan@google.com · 11 years ago
  34. 185636c Revert of 6778 "Refactor StatsCollector and associated types." Breakes FYI bots. by henrike@webrtc.org · 11 years ago
  35. c7b8f39 Fixes "argument list too long" problem on Linux by using the "find" command instead of re-implementing one in python. by henrike@webrtc.org · 11 years ago
  36. 1ebd2e9 Remove timestamp retreival warning/error. by turaj@webrtc.org · 11 years ago
  37. 2386882 Revert "Fix compilation on windows with clang, indentation cleanups" by sergeyu@chromium.org · 11 years ago
  38. a44fce5 Fix compilation on windows with clang, indentation cleanups by sergeyu@chromium.org · 11 years ago
  39. 190d269 Refactor StatsCollector and associated types. by tommi@webrtc.org · 11 years ago
  40. 06b04ec Fix a crash in statscollector.cc caused by invoking methods on the worker thread which destroys the Transport. by jiayl@webrtc.org · 11 years ago
  41. f946068 Make sure padding is sent on the first sending RTP module. by mflodman@webrtc.org · 11 years ago
  42. 45304ff (Auto)update libjingle 71829282-> 71834788 by buildbot@webrtc.org · 11 years ago
  43. 39f831f Re-revert of 6747 "Refactor StatsCollector and associated types." by henrike@webrtc.org · 11 years ago
  44. 437d57d (Auto)update libjingle 71775619-> 71778545 by buildbot@webrtc.org · 11 years ago
  45. 8c7e329 Revert 6747 "Refactor StatsCollector and associated types." Breakes FYI bots. by henrike@webrtc.org · 11 years ago
  46. 8721f98 Revert 6766 "Temporarily add a default ctor to StatsReport and make |id| non const. As soon as I've updated the chrome side, I'll revert this cl." by henrike@webrtc.org · 11 years ago
  47. e2da234 (Auto)update libjingle 71766184-> 71775619 by buildbot@webrtc.org · 11 years ago
  48. 21b4da8 (Auto)update libjingle 71753329-> 71766184 by buildbot@webrtc.org · 11 years ago
  49. 0f7328c Temporarily add a default ctor to StatsReport and make |id| non const. by tommi@webrtc.org · 11 years ago
  50. 9359cb3 Enable SendAndReceive tests. by pbos@webrtc.org · 11 years ago
  51. f24c4a3 Fix flaky ramp-up test. by stefan@webrtc.org · 11 years ago
  52. 5ff71ab Revert "(Auto)update libjingle 71675033-> 71726409" by pbos@webrtc.org · 11 years ago
  53. 89c833c (Auto)update libjingle 71726409-> 71726772 by buildbot@webrtc.org · 11 years ago
  54. f67f6aa (Auto)update libjingle 71675033-> 71726409 by buildbot@webrtc.org · 11 years ago
  55. 8120353 Implement suspend-below-min-bitrate option. by pbos@webrtc.org · 11 years ago
  56. 543e589 Wire up VideoOptions for payload-based padding. by pbos@webrtc.org · 11 years ago
  57. efe4b9a Add VP8 video decoding hw acceleration support to Java Peerconnection library. by glaznev@webrtc.org · 11 years ago
  58. 6f48f1b Implement encoder options in WebRtcVideoEngine2. by pbos@webrtc.org · 11 years ago
  59. cadd078 Remove unused config.h and math.h includes. by pbos@webrtc.org · 11 years ago
  60. 194fea7 The lastest commit on this file was in by minyue@webrtc.org · 11 years ago
  61. 85f4294 Enable ReceiveStreamReceivingByDefault test. by pbos@webrtc.org · 11 years ago
  62. b0c8228 Remove no longer used SkipEncodingUnusedStreams. by andresp@webrtc.org · 11 years ago
  63. 5ab7616 Remove remains of WEBRTC_NO_STL. by andresp@webrtc.org · 11 years ago
  64. fa5fcd6 (Auto)update libjingle 71599033-> 71605904 by buildbot@webrtc.org · 11 years ago
  65. e69b061 (Auto)update libjingle 71575585-> 71599033 by buildbot@webrtc.org · 11 years ago
  66. ceafa8c MIPS optimizations for ISAC (patch #2) by andrew@webrtc.org · 11 years ago
  67. 908f57e Disable GetStatsForInvalidTrack while I rewrite it. by tommi@webrtc.org · 11 years ago
  68. 756b846 Refactor StatsCollector and associated types. by tommi@webrtc.org · 11 years ago
  69. fd61a1d Revert 6745 "Refactor StatsCollector and associated types." by tommi@webrtc.org · 11 years ago
  70. 647e05c Refactor StatsCollector and associated types. by tommi@webrtc.org · 11 years ago
  71. 3c10758 Check before send/receive rtp header extensions. by pbos@webrtc.org · 11 years ago
  72. 8fdeee6 Implement Base::ConstrainNewCodec2. by pbos@webrtc.org · 11 years ago
  73. 3edbaaf Ignore empty data in DataChannel::Send to match FF's behavior. by jiayl@webrtc.org · 11 years ago
  74. 99f6308 (Auto)update libjingle 71460499-> 71464449 by buildbot@webrtc.org · 11 years ago
  75. a0b929b Revert "Reland r6707 with the fix for callclient.cc." by jiayl@webrtc.org · 11 years ago
  76. 196ae6d (Auto)update libjingle 71456344-> 71456420 by buildbot@webrtc.org · 11 years ago
  77. 3dec81a (Auto)update libjingle 71456173-> 71456344 by buildbot@webrtc.org · 11 years ago
  78. a6e8cf8 Reland r6707 with the fix for callclient.cc. by jiayl@webrtc.org · 11 years ago
  79. f563e85 This is to re-open an earlier CL by minyue@webrtc.org · 11 years ago
  80. 60e65b1 (Auto)update libjingle 71452608-> 71453580 by buildbot@webrtc.org · 11 years ago
  81. 8636fc8 Creates the default track if the remote media content is send-only and there is no stream in the SDP. by jiayl@webrtc.org · 11 years ago
  82. ff50deb Runtime guard for iOS7 property. by tkchin@webrtc.org · 11 years ago
  83. 9343cf6 Fix crash in AudioDeviceUtilityIOS::~AudioDeviceUtilityIOS. by tkchin@webrtc.org · 11 years ago
  84. ba92c52 Disable GetStats on DrMemory. by pbos@webrtc.org · 11 years ago
  85. 026859b This is related to an earlier CL of enabling Opus 48 kHz. by minyue@webrtc.org · 11 years ago
  86. e6f84ae Initial WebRtcVideoEngine2::GetStats(). by pbos@webrtc.org · 11 years ago
  87. e9e4253 Sleep in ThreadTest thread functions. by pbos@webrtc.org · 11 years ago
  88. d1ea06b Restart VideoReceiveStreams in WebRtcVideoEngine2. by pbos@webrtc.org · 11 years ago
  89. c31651d (Auto)update libjingle 71378257-> 71410012 by buildbot@webrtc.org · 11 years ago
  90. e364ac9 AudioBuffer: Optimize const accesses to arrays that autoconvert int16<->float by kwiberg@webrtc.org · 11 years ago
  91. c145668 Reduce runtime of RingBufferTest by a factor of 100. by andrew@webrtc.org · 11 years ago
  92. 4f5da03 Use _numMixedParticipants instead of audioFrameList->size() to determine if there're more than one participants. by wu@webrtc.org · 11 years ago
  93. aa93611 Connect to the turn server if address cannot be resolved by the browser by using by mallinath@webrtc.org · 11 years ago
  94. e5995aa Assigning a priority to TURN server list passed to PeerConnection. First entry in the TURN server list will get the highest priotity and so forth. by mallinath@webrtc.org · 11 years ago
  95. e10d28c fix by jiayl@webrtc.org · 11 years ago
  96. 8b94e3d Fix issue where padding is sent before media with undefined timestamps if not abs-send-time is enabled. by stefan@webrtc.org · 11 years ago
  97. 4065988 Remove unused ExperimentalNS API in AudioProcessing by aluebs@webrtc.org · 11 years ago
  98. 2b6bc8d AudioBuffer: Eliminate the SplitChannelBuffer class by kwiberg@webrtc.org · 11 years ago
  99. 5301b0f Move additional state into WebRtcVideoSendStream. by pbos@webrtc.org · 11 years ago
  100. 2561d52 Simplify AudioBuffer::mixed_low_pass_data API by aluebs@webrtc.org · 11 years ago