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webrtc / src / da79008ab4e2d1f652199ea2f927892291e28f5e
  1. da79008 Disabling crashing or flaky tests in peerconnection_unittest. by stefan@webrtc.org · 12 years ago
  2. 32d640e Fix typo in r4765. by pbos@webrtc.org · 12 years ago
  3. da2c4ce Fix dangling pointer _encoder in video_sender.cc. by pbos@webrtc.org · 12 years ago
  4. be63fd6 Initialize CodecInst structs in test_api_audio.cc. by pbos@webrtc.org · 12 years ago
  5. d1fc5d4 Dedicated speed test for NetEq4 by henrik.lundin@webrtc.org · 12 years ago
  6. 28a331e Add support for multiple report blocks. by stefan@webrtc.org · 12 years ago
  7. fc10c5c This is related to https://code.google.com/p/webrtc/issues/detail?id=1341 by sjlee@webrtc.org · 12 years ago
  8. e6ac163 This is related to https://code.google.com/p/webrtc/issues/detail?id=846 by sjlee@webrtc.org · 12 years ago
  9. c3e51ac To use the channel_transport on the iOS platform, some #if directives are changed. by sjlee@webrtc.org · 12 years ago
  10. 15e979b Call AllowCommandLineReparsing in unit tests. by andrew@webrtc.org · 12 years ago
  11. b3af8ae Verify local and remote transport description before negotiation. by mallinath@webrtc.org · 12 years ago
  12. f6ae62f Add Win TSan exclude and Dr Memory suppressions by kjellander@webrtc.org · 12 years ago
  13. eddbfb8 Add more Dr Memory suppressions for common_audio_unittests by kjellander@webrtc.org · 12 years ago
  14. e401c2e Split video coding module unit tests into sender and receiver unit tests. by andresp@webrtc.org · 12 years ago
  15. ab800f6 Disable flaky libjingle tests under tsan and memcheck. by stefan@webrtc.org · 12 years ago
  16. 5860de0 Implement NACK over RTX for VideoSendStream. by pbos@webrtc.org · 12 years ago
  17. 8fa436b Remove use of vcm->ResetDecoder from modules/utility. by andresp@webrtc.org · 12 years ago
  18. 62b816a Fixed pylint warnings. by phoglund@webrtc.org · 12 years ago
  19. 15b8871 Allocate float_buffer_ in the initializer list. by andrew@webrtc.org · 12 years ago
  20. 8a14489 Disable WebRtcSessionTest.TestCreateOfferWithSctpEnabledWithoutStreams by sergeyu@chromium.org · 12 years ago
  21. f7eb75b Split VideoCodingModuleImpl into VideoSender and VideoReceiver. by andresp@webrtc.org · 12 years ago
  22. a59696b Update libjingle to 52300956 by sergeyu@chromium.org · 12 years ago
  23. 48af652 Prepare to compile ACM1 and ACM2. by turaj@webrtc.org · 12 years ago
  24. bc189fb * Prefer to send ISAC on clank. by wu@webrtc.org · 12 years ago
  25. 6ab45b9 Implement DesktopRegion subtraction. by sergeyu@chromium.org · 12 years ago
  26. 1f09dbe Moving test-only code (stream_generator) out of vcm implemention. by andresp@webrtc.org · 12 years ago
  27. 2553450 Fix win trybot errors due to r4729. by andrew@webrtc.org · 12 years ago
  28. 6a5cc9d Fix crash in the window capturer on windows by sergeyu@chromium.org · 12 years ago
  29. 7959e16 ACM2 integration with NetEq 4. by turaj@webrtc.org · 12 years ago
  30. 82a846f Adding Ami to the video renderer and capturer modules. by mallinath@webrtc.org · 12 years ago
  31. 36cf4d2 The video render module for iOS. by fischman@webrtc.org · 12 years ago
  32. e509f94 This issue is related to https://chromereviews.googleplex.com/9908014/ by minyue@webrtc.org · 12 years ago
  33. 8fa03a1 Make PCM16 available in Chromium builds. by andrew@webrtc.org · 12 years ago
  34. 89df092 Make the destructor of AudioCodingModule public. by andrew@webrtc.org · 12 years ago
  35. 5eb997a Fix unsigned/signed comparison error due to r4729. by andrew@webrtc.org · 12 years ago
  36. 8f94013 Reduce frequency of high audio delay warning logs. by andrew@webrtc.org · 12 years ago
  37. 256b831 Removes function that is not used anywhere but somehow still causing library load issues on Android Release build. by henrike@webrtc.org · 12 years ago
  38. 5c678ea Implement 'abs-send-time' extension in VideoSendStream. by pbos@webrtc.org · 12 years ago
  39. 6138c5c OpenSl: fixes crashes externally reported in issue 2361 and 2362. by henrike@webrtc.org · 12 years ago
  40. 036b743 Adding APIs. These APIs are not implemented yet, they are to help developement of ACM. by turaj@webrtc.org · 12 years ago
  41. a80ee74 AppRTC: using a footer element instead of div#footer in CSS. by braveyao@webrtc.org · 12 years ago
  42. d4d59ac Remove FrameForStorage:Follow up on r4688 by mikhal@webrtc.org · 12 years ago
  43. 2902328 Implement 'toffset' extension in VideoSendStream. by pbos@webrtc.org · 12 years ago
  44. 554d158 Reset jitter buffer and timing if frames are getting too much delay. by stefan@webrtc.org · 12 years ago
  45. 835ef67 Remove repeated conditions key. by andrew@webrtc.org · 12 years ago
  46. 82f014a OpenSL (not default): Enables low latency audio on Android. by henrike@webrtc.org · 12 years ago
  47. 6413409 Show the signaling state and ice connection state in AppRTC by hooking up the peerconnections .onsignalingstatechange and .oniceconnectionstatechange events. by braveyao@webrtc.org · 12 years ago
  48. 319c98d Fix format string in video_quality_analysis.cc. by pbos@webrtc.org · 12 years ago
  49. 182d025 Remove include_dirs from voice_engine.gyp. by pbos@webrtc.org · 12 years ago
  50. df531a2 Test that VideoSendStream responds to NACK. by pbos@webrtc.org · 12 years ago
  51. f880f86 Convert printing in video quality tests to Chromium's perf format. by kjellander@webrtc.org · 12 years ago
  52. e07049f Lock RTPSender statistics. by pbos@webrtc.org · 12 years ago
  53. 744fbc7 Split up EngineTests and RampupTests. by pbos@webrtc.org · 12 years ago
  54. eda189b Remove redundant STR_CASE_CMP macro definitions. by andrew@webrtc.org · 12 years ago
  55. a19c9f4 Updated WebRTC version to 3.41 by elham@webrtc.org · 12 years ago
  56. 021c42b Lock use of _packetRequestCallback in VCM. by pbos@webrtc.org · 12 years ago
  57. 7ebf0e7 Remove include_dirs from video_engine_core.gypi. by pbos@webrtc.org · 12 years ago
  58. 59f20bb Break out RTCPSender dependency on ModuleRtpRtcpImpl. by pbos@webrtc.org · 12 years ago
  59. 26b0d77 Suppress RTPSender race regardless of codec. by pbos@webrtc.org · 12 years ago
  60. 841c8a4 Rename VideoCall to Call. by pbos@webrtc.org · 12 years ago
  61. 86136a0 Re-enable tests for Remote Bitrate Estimator by solenberg@webrtc.org · 12 years ago
  62. 0181b5f ExternalVideoDecoder for new VideoEngine API. by pbos@webrtc.org · 12 years ago
  63. 30e055c Handle empty RTP video packets agnostic to codec. by pbos@webrtc.org · 12 years ago
  64. 1b476d9 Disabling channelmanager unittest. This test is causing by mallinath@webrtc.org · 12 years ago
  65. ab5a091 Fixing the build error on Windows. Problem is in coversion from size_t to int. by mallinath@webrtc.org · 12 years ago
  66. 1b15f42 Update talk to 51960985. by mallinath@webrtc.org · 12 years ago
  67. b159c2e Reduce cost of PushSincResampler::Resample(). by andrew@webrtc.org · 12 years ago
  68. c7f7086 Clamp camera id to legal values. by fischman@webrtc.org · 12 years ago
  69. b2c8a95 Improving padding rules and breaking out bw allocation to ViEEncoder. by stefan@webrtc.org · 12 years ago
  70. 7bb8f02 Adds support for combining RTX and FEC/RED. by stefan@webrtc.org · 12 years ago
  71. 5500d93 Add temporal layer factory. by andresp@webrtc.org · 12 years ago
  72. 016eec0 Unbreak build by adding new mandatory ICE username param. by fischman@webrtc.org · 12 years ago
  73. f1e807c Removing FrameForStorage by mikhal@webrtc.org · 12 years ago
  74. c31d4d0 AppRTCDemo(iOS): prefer ISAC as audio codec by fischman@webrtc.org · 12 years ago
  75. aa3d1c8 Make unittest log printouts opt-in with a --logs flag. by andrew@webrtc.org · 12 years ago
  76. bebf399 Pre-multiply images for MouseCursorShape. by alexeypa@chromium.org · 12 years ago
  77. 31b4a5a Recognize armv7 target_arch for ios support in webrtc common.gyp by fischman@webrtc.org · 12 years ago
  78. be588f9 Apprtc Demo: calling createOffer/Answer without failureCallback is deprecated in FF by braveyao@webrtc.org · 12 years ago
  79. 9080518 Restore severity precondition to logging.h. by andrew@webrtc.org · 12 years ago
  80. 95e51f5 Remove send and receive streams when destroyed. by pbos@webrtc.org · 12 years ago
  81. 164c4f7 Add clockdrift to RtpGenerator by henrik.lundin@webrtc.org · 12 years ago
  82. 7e1bf31 Allow unknown flags in test_main.cc. by pbos@webrtc.org · 12 years ago
  83. 36439bf NetEq4: Small change to reduce allocs in AudioMultiVector by henrik.lundin@webrtc.org · 12 years ago
  84. e2d4da6 Enable EngineTest.ReceivesPliAndRecoversWithNack and fix memcheck suppression filter. by mflodman@webrtc.org · 12 years ago
  85. be23b32 Adding tsan suppression for BUG 2349. by mflodman@webrtc.org · 12 years ago
  86. 77bf5c2 Clean capture timestamp code. by andresp@webrtc.org · 12 years ago
  87. 06f1f74 Disable EngineTest.ReceivesPliAndRecoversWithNack. by mflodman@webrtc.org · 12 years ago
  88. b21e528 Protecting Bitrate to avoid data race found by tsan. by mflodman@webrtc.org · 12 years ago
  89. 65abb6b Revert 4671 "Enable SetInitialPlayoutDelay on Android." by mflodman@webrtc.org · 12 years ago
  90. 310ac91 Enable SetInitialPlayoutDelay on Android. by dwkang@webrtc.org · 12 years ago
  91. 3abb82d Suppress video engine test by mikhal@webrtc.org · 12 years ago
  92. 3c5a924 Don't force cont' when enabling kWithErrors by mikhal@webrtc.org · 12 years ago
  93. 635b2b8 Removing some TODO's from libyuv by mikhal@webrtc.org · 12 years ago
  94. 2b810bf Removing non decodable count from session info: This value isn't used, and therfore can be removed. This is a step towards the refactor of the session info to use maps. by mikhal@webrtc.org · 12 years ago
  95. ccf8b56 AppRTCDemo(android): prefer ISAC for audio codec. by fischman@webrtc.org · 12 years ago
  96. 8788167 PeerConnection Java: explicitly cast DataChannel* to jlong for Java. by fischman@webrtc.org · 12 years ago
  97. c8c3263 Remove JpegEncoder suppression as jpeg is now removed. by kjellander@webrtc.org · 12 years ago
  98. f5f5da0 Adding TSAN suppression for test posix udp transport. by mflodman@webrtc.org · 12 years ago
  99. 3a6ff41 Document the source of test scenarios for Dummynet wrapper script. by kjellander@webrtc.org · 12 years ago
  100. cac7325 Adding critsect for child_modules_ in ModuleRtpRtcpImpl::Process() to avoid race with ModuleRtpRtcp::RegisterChildModule. by mflodman@webrtc.org · 12 years ago