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gerrit-public.fairphone.software / platform / external / webrtc / 5f09564354871bfbdb5ce24f7a95c6348a8e2c7b
  1. 5f09564 Suppress AsyncHttpRequestTest.TestCancel leak for LSan by kjellander@webrtc.org · 10 years ago
  2. 823c9b8 Add histograms stats for sent/received fraction loss for a stream: by asapersson@webrtc.org · 10 years ago
  3. d730b28 Remove WebRtcSpl_ScaleAndAddVectorsWithRoundNeon by andrew@webrtc.org · 10 years ago
  4. 59062d5 Rename SendAndReceiveH264SvcQqvga to VP8 instead. by pbos@webrtc.org · 10 years ago
  5. 8af1104 Avoid reading past end of string in GetLine. by decurtis@webrtc.org · 10 years ago
  6. 3663fb0 Reenable dlclose() for InternalUnloadDll on TSan. by pbos@webrtc.org · 10 years ago
  7. bab7995 Convert FileMediaEngineTest to use more expects. by pbos@webrtc.org · 10 years ago
  8. 69472e7 Add a dummy implemenation of SChannelAdapter::SetMode that makes sure that StartSSL fails if the mode is set to DTLS. by pthatcher@webrtc.org · 10 years ago
  9. c10ecea Always tag SRTP_PROTECTION_PROFILE and BIO_METHOD as const. by henrike@webrtc.org · 10 years ago
  10. dfef028 Ignore virtual box interfaces. by pthatcher@webrtc.org · 10 years ago
  11. 25dd754 Excluding a flaky test from DrMemory by tina.legrand@webrtc.org · 10 years ago
  12. 7fbf278 Suppress memcheck error in video_engine_tests by kjellander@webrtc.org · 10 years ago
  13. 1777880 Roll gtest-parallel. by pbos@webrtc.org · 10 years ago
  14. 07c83a1 Disable WebRtcVideoMediaChannelSimulcastTest.SimulcastSend tests on Win (take 2) by kjellander@webrtc.org · 10 years ago
  15. 4e5115a RTCPeerConnectionFactory: Explicitly create new worker and signaling threads. by tkchin@webrtc.org · 10 years ago
  16. f6a9714 Remove peer connection and signaling calls from UI thread. by glaznev@webrtc.org · 10 years ago
  17. 2ec50f2 Memcheck suppression for uninitalized memory in WebRtcIsac_Decode by kjellander@webrtc.org · 10 years ago
  18. d95435c Disable WebRtcVideoMediaChannelSimulcastTest.SimulcastSend tests on Win by kjellander@webrtc.org · 10 years ago
  19. cbe7ca8 Roll chromium_revision 8e72e1d..271c6cc (307131:309333) by kjellander@webrtc.org · 10 years ago
  20. 3a63a3c iOS AppRTC: First unit test. by tkchin@webrtc.org · 10 years ago
  21. 4796cb9 Disable flaky RelayServerTest.TestExpiration on all platforms. by andrew@webrtc.org · 10 years ago
  22. fb7a039 Use array geometry in Beamformer by aluebs@webrtc.org · 10 years ago
  23. a37bf2c Hack clock_unittest fix for parallel execution. by andrew@webrtc.org · 10 years ago
  24. c37e72e Make setting identical RTP extensions a no-op. by pbos@webrtc.org · 10 years ago
  25. e5a921a Use tmp files in file_utils_unittests by aluebs@webrtc.org · 10 years ago
  26. 76bc981 Use a temp file in FileLockTest. by pbos@webrtc.org · 10 years ago
  27. 433006a Fixed style issues from lint and got rid of unused fields. by wzh@webrtc.org · 10 years ago
  28. c4ad157 Re-enable test: VideoProcessorIntegrationTest.ProcessNoLossChangeFrameRateFrameDropVP9. by marpan@webrtc.org · 10 years ago
  29. 215bbbd Fix for log typo in ViEExternalCodecImpl::RegisterExternalReceiveCodec. by mflodman@webrtc.org · 10 years ago
  30. aeb0dd3 Disable RelayServerTest.TestExpiration on Mac. by kjellander@webrtc.org · 10 years ago
  31. 8390c27 Add two unit tests for Android AppRTCDemo. by glaznev@webrtc.org · 10 years ago
  32. 896888b Remove min bitrate from simulcast streams. by pbos@webrtc.org · 10 years ago
  33. bac0012 Extend delay estimation window in AEC to 500 ms on all platforms by bjornv@webrtc.org · 10 years ago
  34. 9eacb8c Make P2PTestConductor use VirtualSocketServer. by pbos@webrtc.org · 10 years ago
  35. c62749f Parallelize MediaRecorder unittests. by pbos@webrtc.org · 10 years ago
  36. 3a70625 audio_processing: Added back ATTRIBUTE_UNUSED lost in r7877 by bjornv@webrtc.org · 10 years ago
  37. 27f5317 Use the prod GAE server in AppRTCDemo for iOS. by jiayl@webrtc.org · 10 years ago
  38. 5eb71eb Fix style issues from lint. by jiayl@webrtc.org · 10 years ago
  39. 34ac956 Do not use openmax_dl for MIPS64 platform. by andrew@webrtc.org · 10 years ago
  40. b2bda67 Removing old channel code from a few more places. by glaznev@webrtc.org · 10 years ago
  41. a9b1ec0 Support for DTLS in OpenSSLAdapter by pthatcher@webrtc.org · 10 years ago
  42. c5fd66d Accept incoming pings before remote answer is set to reduce connection latency. by jiayl@webrtc.org · 10 years ago
  43. 84d8447 Minor fixes regarding accumulator usage on MIPS platforms. by andrew@webrtc.org · 10 years ago
  44. b024da3 Add support for audio device selection in AppRTCDemo. by henrika@webrtc.org · 10 years ago
  45. 5ad4178 Move the Jingle-specific network code into webrtc/libjingle. by pthatcher@webrtc.org · 10 years ago
  46. 46d4d29 Add field trial for screenshare bitrates when using temporal layers. by sprang@webrtc.org · 10 years ago
  47. 1be0a78 Removing giles@mozilla.com from WebRTC watchlist. by mflodman@webrtc.org · 10 years ago
  48. 53cb741 Make RelayServerTest use VirtualSocketServer. by pbos@webrtc.org · 10 years ago
  49. 086c8d5 Use a temporary buffer to scale a screencast in OnFrameCaptured by braveyao@webrtc.org · 10 years ago
  50. 4c0544a Move Jingle-specific files from talk/session/media to webrtc/libjingle/session/media. This is part of an ongoing effort to remove Jingle-specific files from the WebRTC repository. by pthatcher@webrtc.org · 10 years ago
  51. ed1a48b Fix mac video capture leak. by tkchin@webrtc.org · 10 years ago
  52. 7ce4a58 Add initWithCoder to RTCEAGLVideoView. by tkchin@webrtc.org · 10 years ago
  53. ae643ce Wire up Beamformer in AudioProcessing by aluebs@webrtc.org · 10 years ago
  54. 8817256 Fix the ramp-up-down-up test which was using ts-offset extension with the abs-send-time estimator. by stefan@webrtc.org · 10 years ago
  55. 50f7db8 Remove unneccessary lock causing a potential deadlock. by stefan@webrtc.org · 10 years ago
  56. a6f7ba6 Add a AppRTCDemo setting to change the GAE server. by jiayl@webrtc.org · 10 years ago
  57. 5570769 Remove the last getters from VideoReceiveStream stats. by pbos@webrtc.org · 10 years ago
  58. 742386a Enable payload-based padding by default and remove the API. by stefan@webrtc.org · 10 years ago
  59. aa21f27 Unify the two copies of move.h by kwiberg@webrtc.org · 10 years ago
  60. d16e839 Rtp-Rtcp sender cleanup. by pbos@webrtc.org · 10 years ago
  61. 556caff GN: Fix build for Mac by kjellander@webrtc.org · 10 years ago
  62. 11d8176 Move updating nack bitrate inside UpdateNACKBitRate. by stefan@webrtc.org · 10 years ago
  63. 5647877 Breakup Transports and TransportParsers and move TransportParsers into webrtc/libjingle. This is part of an ongoing effort to move Jingle-specific code out of WebRTC and into its own repository. by pthatcher@webrtc.org · 10 years ago
  64. 0c39e91 Merge beamformer by aluebs@webrtc.org · 10 years ago
  65. 1090a6e Remove obsolete target_arch == armv7. by andrew@webrtc.org · 10 years ago
  66. aacc234 Split up (Jingle)Session from BaseSession. This is part of an ongoing effort to move Jingle-specific code out of WebRTC and into its own repository. by pthatcher@webrtc.org · 10 years ago
  67. 16a05dd Clean up the Channel code in AppRTCDemo and use GAE prod server for new signaling mode. by jiayl@webrtc.org · 10 years ago
  68. f5847d7 Move session/tunnel to webrtc/libjingle. This is part of the ongoing effort to move Jingle-specific things out of WebRTC and into its own repository. I won't submit this until all other projects have moved off of compiling this as well. by pthatcher@webrtc.org · 10 years ago
  69. cb79141 Store the received report blocks map (mapped per remote ssrc) in a map per source ssrc. by asapersson@webrtc.org · 10 years ago
  70. ce4e9a3 Refactor some receive-side stats. by pbos@webrtc.org · 10 years ago
  71. 98c04b3 Get avg_delay_ms from DecoderTiming callback. by pbos@webrtc.org · 10 years ago
  72. 9b79197 Suppress REMB in bitrate ctrl if it seems lika a short network glitch. by sprang@webrtc.org · 10 years ago
  73. f832a6d Remove _t from function pointer typedefs. by pbos@webrtc.org · 10 years ago
  74. eed7a22 Make an AudioEncoder subclass for iSAC redundant encoding by henrik.lundin@webrtc.org · 10 years ago
  75. dd8f6f3 Rename rtpDumpPktHdr_t to RtpDumpPacketHeader. by pbos@webrtc.org · 10 years ago
  76. a9cf079 Rename external_hmac_ctx_t to ExternalHmacContext. by pbos@webrtc.org · 10 years ago
  77. e468bc9 Rename _t struct types in audio_processing. by pbos@webrtc.org · 10 years ago
  78. cab1291 Fixing the memory leak in AudioEncoderCopyRedDeathTest.NullSpeechEncoder by henrik.lundin@webrtc.org · 10 years ago
  79. 4fba293 Workaround for issue 3927 to allow localhost IP even if it doesn't match the local turn port by guoweis@webrtc.org · 10 years ago
  80. 4cb3856 Revert "Split up (Jingle)Session from BaseSession. This is part of an ongoing effort to move Jingle-specific code out of WebRTC and into its own repository." by pthatcher@webrtc.org · 10 years ago
  81. 536f999 Split up (Jingle)Session from BaseSession. This is part of an ongoing effort to move Jingle-specific code out of WebRTC and into its own repository. by pthatcher@webrtc.org · 10 years ago
  82. c51fb93 Fix an assert failure caused by race condition by guoweis@webrtc.org · 10 years ago
  83. 0ab42bc Make safe_conversions suitable for rtc_base_approved. by andrew@webrtc.org · 10 years ago
  84. bc03192 Move jingle examples from talk/ into webrtc/libjingle. This is part of the effor to move Jingle out of WebRTC and into its own repository. by pthatcher@webrtc.org · 10 years ago
  85. 0eb6eec Move VirtualSocket into the .h file to allow unit tests more control over behavior. by guoweis@webrtc.org · 10 years ago
  86. 6f10ae2 Support block_size greater than chunk_size in Blocker by aluebs@webrtc.org · 10 years ago
  87. eb54446 Rename _t struct types in audio_coding. by pbos@webrtc.org · 10 years ago
  88. 209df9b Change MockStatsObserver to grab values inside of OnComplete. by tommi@webrtc.org · 10 years ago
  89. e728ee0 Remove or rename typedefs with _t prefixes. by pbos@webrtc.org · 10 years ago
  90. 5263c58 Add a little utility to capture cpu graphs. by tommi@webrtc.org · 10 years ago
  91. 70f74f3 Add overshoot of target bitrate for screenshare with temporal layers. by sprang@webrtc.org · 10 years ago
  92. 45a272a Change aggregated fraction loss to be calculated from the cumulative loss and extended sequence number diff between the current and the last report block of two get stats calls. by asapersson@webrtc.org · 10 years ago
  93. e102e81 Enable the iSACfix AudioDecoder test (and make it work again) by kwiberg@webrtc.org · 10 years ago
  94. 38881be If one of the bundled content is missing in SDP, return false to MaybeEnalbeMuxingSupport(). by braveyao@webrtc.org · 10 years ago
  95. 950c518 Add adapter_type into Candidate object. by guoweis@webrtc.org · 10 years ago
  96. 971bf55 Fix path to mock_agc.h by andrew@webrtc.org · 10 years ago
  97. f050791 Revert "Split up (Jingle)Session from BaseSession. This is part of an ongoing effort to move Jingle-specific code out of WebRTC and into its own repository." by pthatcher@webrtc.org · 10 years ago
  98. 4afb599 Split up (Jingle)Session from BaseSession. This is part of an ongoing effort to move Jingle-specific code out of WebRTC and into its own repository. by pthatcher@webrtc.org · 10 years ago
  99. e2b7585 Move ViewRequest and MediaStreams to streamparams.h, and remove dependency on mediasessionclient.h and mediamessages.h. This is part of the effort to remove Jingle-specific code from WebRTC and into its own repository. by pthatcher@webrtc.org · 10 years ago
  100. a32487f Disable AudioEncoderCopyRedDeathTest.NullSpeechEncoder by henrik.lundin@webrtc.org · 10 years ago