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webrtc / src.git / c6273b53dd78a7deaa78589ad8590e0631e7723d
  1. c6273b5 DrMemory suppresssions, likely from r6811. by marpan@webrtc.org · 11 years ago
  2. c3d2bd2 Fix GetStats() crash. by pbos@webrtc.org · 11 years ago
  3. 3d53f61 .gitignore removed openssl by henrike@webrtc.org · 11 years ago
  4. aa2344e talk/third_party: removes the empty directory. by henrike@webrtc.org · 11 years ago
  5. 8d57f08 (Auto)update libjingle 73072800-> 73072800 by buildbot@webrtc.org · 11 years ago
  6. 40995c7 Fixing uninitialized variable in file_audio_device.cc. by phoglund@webrtc.org · 11 years ago
  7. 0a3cbb3 common_audio/signal_processing: Removes macro WEBRTC_SPL_MUL_32_32_RSFT32 by bjornv@webrtc.org · 11 years ago
  8. cf8f33a Removes mismatching signs in signal_processing_unittests by bjornv@webrtc.org · 11 years ago
  9. 6aac93b Adding SetOpusMaxBandwidth in VoE and ACM by minyue@webrtc.org · 11 years ago
  10. c98ce3b modules/audio_processing: Updates output_data_fixed.pb test file by bjornv@webrtc.org · 11 years ago
  11. 6ac22e6 Remove more dependencies on openssl, add dependency on boringssl. Continues on r6798 by henrike@webrtc.org · 11 years ago
  12. 820f8e9 modules/audio_processing: Moves declaration of kDelayDiffOffsetSamples by bjornv@webrtc.org · 11 years ago
  13. 4e4b098 Merge NetEqDecodingTest.TestBitExactnesst and .TestNetworkStatistics by henrik.lundin@webrtc.org · 11 years ago
  14. 065247b Rebase webrtc/base with r6863 version of talk/base: by henrike@webrtc.org · 11 years ago
  15. 730bf30 Refactor StatsCollector and associated types. by tommi@webrtc.org · 11 years ago
  16. 1c83912 Use test::Packet test::PacketSource classes in neteq_rtpplay by henrik.lundin@webrtc.org · 11 years ago
  17. 96d8b0e Revert 6860 "SSE2 version of SubbandCoherence()" by bjornv@webrtc.org · 11 years ago
  18. 0db82f3 SSE2 version of SubbandCoherence() by bjornv@webrtc.org · 11 years ago
  19. 7ec3f9f Fix a bug in parsing IceCandidate with IPV6 address. by jiayl@webrtc.org · 11 years ago
  20. 9eabe5e (Auto)update libjingle 72931377-> 72931377 by buildbot@webrtc.org · 11 years ago
  21. 2d60c5e Encoding and Decoding of TCP candidates as defined in RFC 6544. by mallinath@webrtc.org · 11 years ago
  22. 8c01e59 Allow root build dependencies to be overridden. by harryjin@google.com · 11 years ago
  23. 53df88c (Auto)update libjingle 72847605-> 72850595 by buildbot@webrtc.org · 11 years ago
  24. 65b98d1 (Auto)update libjingle 72839629-> 72847605 by buildbot@webrtc.org · 11 years ago
  25. 3763b9b webrtc/base: removes linkage of crypto by henrike@webrtc.org · 11 years ago
  26. c8554be Support for TURN/TLS. by tkchin@webrtc.org · 11 years ago
  27. cb46de2 Add new OWNERS file to talk/examples. by tkchin@webrtc.org · 11 years ago
  28. 5b1ebac (Auto)update libjingle 72820109-> 72822008 by buildbot@webrtc.org · 11 years ago
  29. d509678 (Auto)update libjingle 72819313-> 72820109 by buildbot@webrtc.org · 11 years ago
  30. 94b996c (Auto)update libjingle 72785516-> 72819313 by buildbot@webrtc.org · 11 years ago
  31. 59a2f9f Remove the old H264 code now that a new H.264 packetizer has been implemented. by stefan@webrtc.org · 11 years ago
  32. 9d74f7c Fix single nalu packetization bug. by stefan@webrtc.org · 11 years ago
  33. e8c84bf Fix so video_replay logs aren't spammed. by pbos@webrtc.org · 11 years ago
  34. 1d956dd Since the packet loss rate cannot be estimated accurately, there is always a mismatch between the estimated packet loss rate and the true one. Such a mismatch will make Opus FEC suboptimal. by minyue@webrtc.org · 11 years ago
  35. ea25784 Change how background noise mode in NetEq is set by henrik.lundin@webrtc.org · 11 years ago
  36. 476efa2 (Auto)update libjingle 72785180-> 72785516 by buildbot@webrtc.org · 11 years ago
  37. 4f0d401 (Auto)update libjingle 72682155-> 72785180 by buildbot@webrtc.org · 11 years ago
  38. aaecefe Revert 6839 "Allow root build dependencies to be overridden." by harryjin@google.com · 11 years ago
  39. e34abfb Allow root build dependencies to be overridden. by harryjin@google.com · 11 years ago
  40. 4b5625e RTP video playback tool using Call APIs. by pbos@webrtc.org · 11 years ago
  41. 1ccff349 Fix crashing fake network pipe tests. by stefan@webrtc.org · 11 years ago
  42. 2a8df7c Fixing two bugs in voe_cmd_test. by minyue@webrtc.org · 11 years ago
  43. 79c3359 Add end-to-end H.264 packetization test. by stefan@webrtc.org · 11 years ago
  44. e415864 GN: Add PRESUBMIT.py check for GN changes + default bots. by kjellander@webrtc.org · 11 years ago
  45. 8b033ad Change the way we reference enumerators in H.264 packetization code to be standard C++ compliant. by stefan@webrtc.org · 11 years ago
  46. 56d8e05 A followup to r6828 to fix a condition check in mediasession.cc. by jiayl@webrtc.org · 11 years ago
  47. d7b4dea initialize packet len in NETEQTEST_DummyRTPpacket.cc and NETEQTEST_RTPpacket.cc to fix build error on vs2013 by fbarchard@google.com · 11 years ago
  48. dde16f1 Fix some code styles. by pbos@webrtc.org · 11 years ago
  49. 624a504 (Auto)update libjingle 72659510-> 72673987 by buildbot@webrtc.org · 11 years ago
  50. e7d47a1 Maintain the order of the m-lines in CreateOffer and CreateAnswer. by jiayl@webrtc.org · 11 years ago
  51. e086af0 Fix implicite cast from signed int to unsigned int in unittest.cc by fbarchard@google.com · 11 years ago
  52. 923db6a Remove remove_old_gn_binaries DEPS entry. by pbos@webrtc.org · 11 years ago
  53. fdcb42d Fix potential crash when depacketizing VP8. by stefan@webrtc.org · 11 years ago
  54. 8e88599 (Auto)update libjingle 72566057-> 72591796 by buildbot@webrtc.org · 11 years ago
  55. d654285 Unbreaks linux.cc in Chromium. by henrike@webrtc.org · 11 years ago
  56. b18bf5e Adds the support of RTCOfferOptions for PeerConnectionInterface::CreateOffer. by jiayl@webrtc.org · 11 years ago
  57. b01ce14 add some comments about DEPS lkgr for chromium by fbarchard@google.com · 11 years ago
  58. c9b5072 DrMemory suppression due to r6811. by henrike@webrtc.org · 11 years ago
  59. ee135f7 Memcheck suppression. Re-suppress 3478 suppression after namespace change from talk_base to rtc. by henrike@webrtc.org · 11 years ago
  60. a27342b (Auto)update libjingle 72446860-> 72550257 by buildbot@webrtc.org · 11 years ago
  61. 0040a6e This is a setup to solve https://code.google.com/p/webrtc/issues/detail?id=1906 by minyue@webrtc.org · 11 years ago
  62. 84b9e1e Fix for retransmission. Base layer packets were not retransmitted. by asapersson@webrtc.org · 11 years ago
  63. e0d03f1 (Auto)update libjingle 72443101-> 72446860 by buildbot@webrtc.org · 11 years ago
  64. 6e203d5 (Auto)update libjingle 72442050-> 72443101 by buildbot@webrtc.org · 11 years ago
  65. 52148c2 (Auto)update libjingle 72430895-> 72442050 by buildbot@webrtc.org · 11 years ago
  66. 7cb60cc (Auto)update libjingle 72407428-> 72430895 by buildbot@webrtc.org · 11 years ago
  67. 3bc4824 (Auto)update libjingle 72403605-> 72407428 by buildbot@webrtc.org · 11 years ago
  68. 6955213 (Auto)update libjingle 72389720-> 72403605 by buildbot@webrtc.org · 11 years ago
  69. 42d65ce Fix memory leak in FakeSSLCertificate::GetChain(), discovered by Linux Memcheck build/try bots. by solenberg@webrtc.org · 11 years ago
  70. 1a678c6 (Auto)update libjingle 72320533-> 72380285 by buildbot@webrtc.org · 11 years ago
  71. 6b21b71 (Auto)update libjingle 72205295-> 72320533 by buildbot@webrtc.org · 11 years ago
  72. e1c9caf Fix mistake in rtp/rtcp/BUILD.gn introduced with r6804. by stefan@webrtc.org · 11 years ago
  73. 2ec5606 Add H.264 packetization. by stefan@webrtc.org · 11 years ago
  74. bfe6e08 Add simulation of network effects to video_loopback tool. by stefan@webrtc.org · 11 years ago
  75. d9843da libjingle: stop building files in talk/base as they are no longer used as of r6799 by henrike@webrtc.org · 11 years ago
  76. 48305f5 Disable warning 4702 which affects map, xlist and others on vs2012 and vs2013. by fbarchard@google.com · 11 years ago
  77. 901debd roll libyuv to r1038 from r1035 to add gyp define that makes jpeg optional. by fbarchard@google.com · 11 years ago
  78. d4e598d (Auto)update libjingle 72097588-> 72159069 by buildbot@webrtc.org · 11 years ago
  79. fc8b087 Remove dependency on openssl for android, add dependency on boringssl. Should make Android bots green again. by solenberg@webrtc.org · 11 years ago
  80. fdbe144 Use C functions in aec for MIPS by andrew@webrtc.org · 11 years ago
  81. e75d78d Integrate rtcp packet class to rtcp receiver tests. by asapersson@webrtc.org · 11 years ago
  82. 1e7d60e merge_libs.py: fixes Windows breakage: there should be no space after "lib /OUT:". by henrike@webrtc.org · 11 years ago
  83. 51c5508 (Auto)update libjingle 72016417-> 72097588 by buildbot@webrtc.org · 11 years ago
  84. 8aed945 Remove a disabled test. by pbos@webrtc.org · 11 years ago
  85. 4fe98a9 Remove clang-format rm_binaries.py DEPS entry. by pbos@webrtc.org · 11 years ago
  86. 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
  87. af9e794 Fix compilation on windows with clang, indentation cleanups by sergeyu@chromium.org · 11 years ago
  88. 257e130 Set NACK/REMB when setting receive codecs. by pbos@webrtc.org · 11 years ago
  89. 3155f2b Roll chromium 282879:285412. by fgalligan@google.com · 11 years ago
  90. 185636c Revert of 6778 "Refactor StatsCollector and associated types." Breakes FYI bots. by henrike@webrtc.org · 11 years ago
  91. 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
  92. 1ebd2e9 Remove timestamp retreival warning/error. by turaj@webrtc.org · 11 years ago
  93. 2386882 Revert "Fix compilation on windows with clang, indentation cleanups" by sergeyu@chromium.org · 11 years ago
  94. a44fce5 Fix compilation on windows with clang, indentation cleanups by sergeyu@chromium.org · 11 years ago
  95. 190d269 Refactor StatsCollector and associated types. by tommi@webrtc.org · 11 years ago
  96. 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
  97. f946068 Make sure padding is sent on the first sending RTP module. by mflodman@webrtc.org · 11 years ago
  98. 45304ff (Auto)update libjingle 71829282-> 71834788 by buildbot@webrtc.org · 11 years ago
  99. 39f831f Re-revert of 6747 "Refactor StatsCollector and associated types." by henrike@webrtc.org · 11 years ago
  100. 437d57d (Auto)update libjingle 71775619-> 71778545 by buildbot@webrtc.org · 11 years ago