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webrtc / src / a941970d4afe5fec3042e24bef7070e7119a8ffd
  1. a941970 Change explicit static cast from int to uint16_t to implicit cast of 0u. by fbarchard@google.com · 11 years ago
  2. 9fe1101 Fix the RTC+Chromium GN build. by brettw@chromium.org · 11 years ago
  3. 54cf150 ValidateFrame, When dumping the first 4 samples of a frame, first copy it to a temporary buffer that is zero padded, them use that. by fbarchard@google.com · 11 years ago
  4. 22406fc TurnPort should retry allocation with a new address on error STUN_ERROR_ALLOCATION_MISMATCH. by jiayl@webrtc.org · 11 years ago
  5. 04b853b Bot Browser files moved to /bot/browser/ by houssainy@google.com · 11 years ago
  6. 3d81b1b Relanding https://code.google.com/p/webrtc/source/detail?r=7093, after it got by mallinath@webrtc.org · 11 years ago
  7. 4bbd3c8 fix a bug in the logic when new Networks are merged. This happens when by guoweis@webrtc.org · 11 years ago
  8. 1b088ee More suppressions, uninitialized read in cricket::VideoFrame::Validate by sprang@webrtc.org · 11 years ago
  9. 4d19e05 Peerconnection_jni to use webrtc/base/checks.h instead of implementing its own. by andresp@webrtc.org · 11 years ago
  10. b420191 Expose VideoEncoders with webrtc/video_encoder.h. by pbos@webrtc.org · 11 years ago
  11. 641bda6 Initialize ChannelBuffer's memory to avoid uninitialized reads. by andrew@webrtc.org · 11 years ago
  12. 8b0b211 Revert 7093: "Implementing ICE Transports type handling in libjingle transport." by henrike@webrtc.org · 11 years ago
  13. 519c9e2 Convert GN visibility to be a list. by brettw@chromium.org · 11 years ago
  14. 7118e61 Finish work queue in SctpDataMediaChannelTest. by pbos@webrtc.org · 11 years ago
  15. 0e52772 Fix a bot-breaking memory leak from early returning in ParseMediaDescription. by jiayl@webrtc.org · 11 years ago
  16. c172320 Revert "Fixes two issues in how we handle OfferToReceiveX for CreateOffer:" because it broke content_browsertests on Android. by jiayl@webrtc.org · 11 years ago
  17. 17454f7 Add ctors to ChannelBuffer to enable copying on construction. by andrew@webrtc.org · 11 years ago
  18. fd42f9d (Auto)update libjingle 74955991-> 75042522 by buildbot@webrtc.org · 11 years ago
  19. 1272ee5 Suppress uninitialized read warning in cricket::VideoFrame::Validate by sprang@webrtc.org · 11 years ago
  20. c64246f Set a default speech type in iSAC wrapper by henrik.lundin@webrtc.org · 11 years ago
  21. ed8bcd3 Starting to implement the new ACM API by henrik.lundin@webrtc.org · 11 years ago
  22. 9600519 Adding the ability to test on Chrome for Android. by houssainy@google.com · 11 years ago
  23. 37c39f3 audio_processing: Removed use of macro WEBRTC_SPL_UMUL_16_16 by bjornv@webrtc.org · 11 years ago
  24. 0d394f3 video_processing: Removed usage of WEBRTC_SPL_UMUL_16_16 by bjornv@webrtc.org · 11 years ago
  25. c77e4d6 - Adding AndroidDeviceManager to botManager.js to help in selecting devices, in case running test on Android devices. by houssainy@google.com · 11 years ago
  26. 142bb9d Roll chromium_revision 94532b1..ea769fd by kjellander@webrtc.org · 11 years ago
  27. fe16167 Fix RTT calculations for send-only channels. by stefan@webrtc.org · 11 years ago
  28. c30e9e2 Ignore FEC packet in stats, if it is first packet on ssrc. by sprang@webrtc.org · 11 years ago
  29. 6d08ca6 GN: Prefix WebRTC specific variables with "rtc_" by kjellander@webrtc.org · 11 years ago
  30. f68cf93 Add video_capture_tests_apk_target by kjellander@webrtc.org · 11 years ago
  31. 7256d31 Implementing ICE Transports type handling in libjingle transport. by mallinath@webrtc.org · 11 years ago
  32. a781f68 Fix rm command for class cleanup in r7091 by kjellander@webrtc.org · 11 years ago
  33. 9510022 Cleanup temporary class files for OpenSlDemo by kjellander@webrtc.org · 11 years ago
  34. cc06056 Remove unnecessary include from testutils.cc. by thorcarpenter@google.com · 11 years ago
  35. 992febb (Auto)update libjingle 74873066-> 74873164 by buildbot@webrtc.org · 11 years ago
  36. a3344cf Fix webrtcvideoframe tests. by thorcarpenter@google.com · 11 years ago
  37. ddb85ab Updated SCTP SDP attributes according to draft-ietf-mmusic-sctp-sdp-07 by jiayl@webrtc.org · 11 years ago
  38. 8f073c5 Create a new interface for AudioCodingModule by henrik.lundin@webrtc.org · 11 years ago
  39. af5fa952 (Auto)update libjingle 74857067-> 74860820 by buildbot@webrtc.org · 11 years ago
  40. 7e3bd3d (Auto)update libjingle 74851128-> 74857067 by buildbot@webrtc.org · 11 years ago
  41. bc6fa18 (Auto)update libjingle 74825992-> 74851128 by buildbot@webrtc.org · 11 years ago
  42. 287e961 Disable TestDrain test on memcheck bots. by pbos@webrtc.org · 11 years ago
  43. cdb48db Enable VideoAdapterTest.BlackOutput on DrMemory. by pbos@webrtc.org · 11 years ago
  44. fed47dc Drop buildbot_tests.py script by kjellander@webrtc.org · 11 years ago
  45. a2da031 Remove use_relative_paths from DEPS by kjellander@webrtc.org · 11 years ago
  46. bcf75e3 Modifying audio_coding/codecs/OWNERS by henrik.lundin@webrtc.org · 11 years ago
  47. c2c4117 common_audio: Replaced WEBRTC_SPL_LSHIFT_U32 with << in audio_processing by bjornv@webrtc.org · 11 years ago
  48. 2c03a97 Roll chromium_revision f0a439d..94532b1 by kjellander@webrtc.org · 11 years ago
  49. 818b7b3 (Auto)update libjingle 74825084-> 74825992 by buildbot@webrtc.org · 11 years ago
  50. dfbcf81 Fix an issue in MediaStreamSignaling that a remotely create DataChannel is added to the list twice. by jiayl@webrtc.org · 11 years ago
  51. f1427c6 Revert 7070 "TurnPort should retry allocation with a new address on error by henrike@webrtc.org · 11 years ago
  52. 4b23404 Reduce maximum video resolution for Android. by glaznev@webrtc.org · 11 years ago
  53. 574f2f6 TurnPort should retry allocation with a new address on error STUN_ERROR_ALLOCATION_MISMATCH. by jiayl@webrtc.org · 11 years ago
  54. 021e76f Add support for WAV output in audioproc by aluebs@webrtc.org · 11 years ago
  55. 52055a2 Fixes two issues in how we handle OfferToReceiveX for CreateOffer: by jiayl@webrtc.org · 11 years ago
  56. afa77cd Add direct_dependent_config to desktop_capture in GN build. by brettw@chromium.org · 11 years ago
  57. ceb956b Abort Negotiate() if DoCreateOffer() fails. by pbos@webrtc.org · 11 years ago
  58. d57c95f Change Chromium .gclient to not use Managed mode. by kjellander@webrtc.org · 11 years ago
  59. fa822b9 Fix strange owners files with comments that crashs "git cl presubmit" by andresp@webrtc.org · 11 years ago
  60. 79ee97c [MIPS] Fix gn gen failure for MIPS in webrtc by kjellander@webrtc.org · 11 years ago
  61. 38ef664 Moving the api.js and bot.js to /rtcbot/bot/ to be shared between by houssainy@google.com · 11 years ago
  62. 262e676 Reland rev 7041 with BUILD.gn files. by andresp@webrtc.org · 11 years ago
  63. 3cbd6c2 Fix MSVC warnings about value truncations, webrtc/common_audio/ edition. by bjornv@webrtc.org · 11 years ago
  64. f6ab6f8 Rename Audio[Multi]Vector.CopyFrom to .CopyTo by henrik.lundin@webrtc.org · 11 years ago
  65. 3c0aae1 Change gflags and gmock includes to be full paths. by kjellander@webrtc.org · 11 years ago
  66. 51bb33c ACMOpus: Remove useless member variable fec_enabled_ by kwiberg@webrtc.org · 11 years ago
  67. 7825b1a Add support for multi-channel DTMF tone generation by henrik.lundin@webrtc.org · 11 years ago
  68. bcb6bcf Remove HybridVideoEngine. by pbos@webrtc.org · 11 years ago
  69. 9d45393 Change return value for number of discarded packets to be int. by asapersson@webrtc.org · 11 years ago
  70. 01581da Fix audio/video sync when FEC is enabled. by stefan@webrtc.org · 11 years ago
  71. bfd7a8c Fix compile errors on webrtc/base. by andresp@webrtc.org · 11 years ago
  72. 0229cba Remove ambiguous call to MakeCheckOpString. by andresp@webrtc.org · 11 years ago
  73. 95c2458 * Move test data assests required by video frame tests to be in libjingle instead of elsewhere and co-located with other libjingle test data files. by thorcarpenter@google.com · 11 years ago
  74. 9328f39 cast return values in uint16_t RTPFile::Read() to uint16_t to avoid compile error by fbarchard@google.com · 11 years ago
  75. 5b83af4 Fix leak of NSAutoreleasePool. by tkchin@webrtc.org · 11 years ago
  76. 609f987 (Auto)update libjingle 74696326-> 74723281 by buildbot@webrtc.org · 11 years ago
  77. 1b8b4c4 Revert 7041 " Audio codecs to include webrtc/typedefs.h" by henrike@webrtc.org · 11 years ago
  78. fa4535b (Auto)update libjingle 74694022-> 74696326 by buildbot@webrtc.org · 11 years ago
  79. 26c0c41 Network up/down signaling in Call. by pbos@webrtc.org · 11 years ago
  80. ebee401 Remove flake in SendsLowerResolutionOnSmallerFrames. by pbos@webrtc.org · 11 years ago
  81. c4175b9 Set resolution based on incoming VideoFrames. by pbos@webrtc.org · 11 years ago
  82. 9730d3a Audio codecs to include webrtc/typedefs.h by andresp@webrtc.org · 11 years ago
  83. 0372b93 Partial revert of r7014 (Android APK refactor) by kjellander@webrtc.org · 11 years ago
  84. bac0726 Use the sample rate as a temporary solution to unpack aecdumps with wrong sizes by aluebs@webrtc.org · 11 years ago
  85. adee8f9 Renaming SetOpusMaxBandwidth to SetOpusMaxPlaybackRate by minyue@webrtc.org · 11 years ago
  86. 0a214ff Setting marker bit on DTMF correctly by stefan@webrtc.org · 11 years ago
  87. 74cf916 Fix issues in audioproc for float aecdumps by aluebs@webrtc.org · 11 years ago
  88. 48f2568 audio_processing/nsx: Bug fix that could cause divide by zero by bjornv@webrtc.org · 11 years ago
  89. d944a68 Suppressing VideoAdapterTest.AdaptResolutionWide and VideoAdapterTest.AdaptResolutionNarrow on DrMemory by minyue@webrtc.org · 11 years ago
  90. 72e4485 (Auto)update libjingle 74628537-> 74648573 by buildbot@webrtc.org · 11 years ago
  91. 9075048 Remove deprecated RTCVideoRenderer constructor. by tkchin@webrtc.org · 11 years ago
  92. 34a6764 Remove the checks.h dependence on logging.h in a standalone build. by andrew@webrtc.org · 11 years ago
  93. 8e24d87 Fix race in Voice Engine's Channel where it accesses RemoteNtpTimeEstimator from both the audio playback thread and the network thread without locking. by stefan@webrtc.org · 11 years ago
  94. 9f34128 Remove WebRtcVideoEngine::default_codec_format(). by pbos@webrtc.org · 11 years ago
  95. 03655143 Remove files from talk/PRESUBMIT.py. by pbos@webrtc.org · 11 years ago
  96. d72a759 Create a copy of talk/xmllite under webrtc/xmllite. by henrike@webrtc.org · 11 years ago
  97. 6f729e8 Disable video_engine_tests and webrtc_perf_tests on Android. by kjellander@webrtc.org · 11 years ago
  98. ee0fb18 Divide-by-zero problem in NetEq's Normal::Process fixed by henrik.lundin@webrtc.org · 11 years ago
  99. 94da203 Remove retired android_apk[_rel] trybots from PRESUBMIT.py by kjellander@webrtc.org · 11 years ago
  100. 324b72d Disable video_capture_tests for Android. by kjellander@webrtc.org · 11 years ago