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gerrit-public.fairphone.software / platform / external / webrtc / a525c98ca50ae6af35bd4fac027aa10725dd1399
  1. a525c98 Fix parallelizability in ApmTests. by pbos@webrtc.org · 10 years ago
  2. 45db7ee Use Java based audio as default for WebRTC. by henrika@webrtc.org · 10 years ago
  3. 81134d0 Use proxy macro for PeerConnectionFactory instead of sending messages internally in PeerConnectionFactory. by perkj@webrtc.org · 10 years ago
  4. 88a4298 common_audio: Made input vector const in WebRtcSpl_LevinsonDurbin() by bjornv@webrtc.org · 10 years ago
  5. c14e357 common_audio: Made input signal const in WebRtcSplFilterMAFastQ12() by bjornv@webrtc.org · 10 years ago
  6. 19e4e8d Add support for trying alternate server (STUN 300 error message) on TCP by guoweis@webrtc.org · 10 years ago
  7. 0ba1533 Added support for an Origin header in STUN messages. by pthatcher@webrtc.org · 10 years ago
  8. 2693a54 Add WEBRTC_BEAMFORMER define to BUILD.gn by aluebs@webrtc.org · 10 years ago
  9. 8f27fcc Revert 8028 "Support associated payload type when registering Rt..." by andrew@webrtc.org · 10 years ago
  10. 80452d7 Sync Android AppRTCDemo with internal repo. by glaznev@webrtc.org · 10 years ago
  11. 9657265 Revert "Accept incoming pings before remote answer is set to reduce connection latency." by pthatcher@webrtc.org · 10 years ago
  12. f3fd8e7 Add NEON intrinsics version for transform_neon by andrew@webrtc.org · 10 years ago
  13. 1592df7 PRESUBMIT: Add GN trybots for Windows and Mac. by kjellander@webrtc.org · 10 years ago
  14. 2a16964 Support associated payload type when registering Rtx payload type. by pbos@webrtc.org · 10 years ago
  15. 8649fed GN: Fix Windows build. by kjellander@webrtc.org · 10 years ago
  16. 2ead571 Hard define the GUID for AudioEndpoint to avoid conflicts during compile. by decurtis@webrtc.org · 10 years ago
  17. 758d6d4 audio_processing/aecm: Removed usage of macro WEBRTC_SPL_MUL_16_16 by bjornv@webrtc.org · 10 years ago
  18. dec649c audio_processing/ns: Replaced WEBRTC_SPL_MUL_16_16 macro with * by bjornv@webrtc.org · 10 years ago
  19. 5e5b327 audio_processing/agc: Removed usage of macro WEBRTC_SPL_MUL_16_16 in legacy/agc by bjornv@webrtc.org · 10 years ago
  20. 124b9c7 Suppress races in event tracing code. by pbos@webrtc.org · 10 years ago
  21. 5f09564 Suppress AsyncHttpRequestTest.TestCancel leak for LSan by kjellander@webrtc.org · 10 years ago
  22. 823c9b8 Add histograms stats for sent/received fraction loss for a stream: by asapersson@webrtc.org · 10 years ago
  23. d730b28 Remove WebRtcSpl_ScaleAndAddVectorsWithRoundNeon by andrew@webrtc.org · 10 years ago
  24. 59062d5 Rename SendAndReceiveH264SvcQqvga to VP8 instead. by pbos@webrtc.org · 10 years ago
  25. 8af1104 Avoid reading past end of string in GetLine. by decurtis@webrtc.org · 10 years ago
  26. 3663fb0 Reenable dlclose() for InternalUnloadDll on TSan. by pbos@webrtc.org · 10 years ago
  27. bab7995 Convert FileMediaEngineTest to use more expects. by pbos@webrtc.org · 10 years ago
  28. 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
  29. c10ecea Always tag SRTP_PROTECTION_PROFILE and BIO_METHOD as const. by henrike@webrtc.org · 10 years ago
  30. dfef028 Ignore virtual box interfaces. by pthatcher@webrtc.org · 10 years ago
  31. 25dd754 Excluding a flaky test from DrMemory by tina.legrand@webrtc.org · 10 years ago
  32. 7fbf278 Suppress memcheck error in video_engine_tests by kjellander@webrtc.org · 10 years ago
  33. 1777880 Roll gtest-parallel. by pbos@webrtc.org · 10 years ago
  34. 07c83a1 Disable WebRtcVideoMediaChannelSimulcastTest.SimulcastSend tests on Win (take 2) by kjellander@webrtc.org · 10 years ago
  35. 4e5115a RTCPeerConnectionFactory: Explicitly create new worker and signaling threads. by tkchin@webrtc.org · 10 years ago
  36. f6a9714 Remove peer connection and signaling calls from UI thread. by glaznev@webrtc.org · 10 years ago
  37. 2ec50f2 Memcheck suppression for uninitalized memory in WebRtcIsac_Decode by kjellander@webrtc.org · 10 years ago
  38. d95435c Disable WebRtcVideoMediaChannelSimulcastTest.SimulcastSend tests on Win by kjellander@webrtc.org · 10 years ago
  39. cbe7ca8 Roll chromium_revision 8e72e1d..271c6cc (307131:309333) by kjellander@webrtc.org · 10 years ago
  40. 3a63a3c iOS AppRTC: First unit test. by tkchin@webrtc.org · 10 years ago
  41. 4796cb9 Disable flaky RelayServerTest.TestExpiration on all platforms. by andrew@webrtc.org · 10 years ago
  42. fb7a039 Use array geometry in Beamformer by aluebs@webrtc.org · 10 years ago
  43. a37bf2c Hack clock_unittest fix for parallel execution. by andrew@webrtc.org · 10 years ago
  44. c37e72e Make setting identical RTP extensions a no-op. by pbos@webrtc.org · 10 years ago
  45. e5a921a Use tmp files in file_utils_unittests by aluebs@webrtc.org · 10 years ago
  46. 76bc981 Use a temp file in FileLockTest. by pbos@webrtc.org · 10 years ago
  47. 433006a Fixed style issues from lint and got rid of unused fields. by wzh@webrtc.org · 10 years ago
  48. c4ad157 Re-enable test: VideoProcessorIntegrationTest.ProcessNoLossChangeFrameRateFrameDropVP9. by marpan@webrtc.org · 10 years ago
  49. 215bbbd Fix for log typo in ViEExternalCodecImpl::RegisterExternalReceiveCodec. by mflodman@webrtc.org · 10 years ago
  50. aeb0dd3 Disable RelayServerTest.TestExpiration on Mac. by kjellander@webrtc.org · 10 years ago
  51. 8390c27 Add two unit tests for Android AppRTCDemo. by glaznev@webrtc.org · 10 years ago
  52. 896888b Remove min bitrate from simulcast streams. by pbos@webrtc.org · 10 years ago
  53. bac0012 Extend delay estimation window in AEC to 500 ms on all platforms by bjornv@webrtc.org · 10 years ago
  54. 9eacb8c Make P2PTestConductor use VirtualSocketServer. by pbos@webrtc.org · 10 years ago
  55. c62749f Parallelize MediaRecorder unittests. by pbos@webrtc.org · 10 years ago
  56. 3a70625 audio_processing: Added back ATTRIBUTE_UNUSED lost in r7877 by bjornv@webrtc.org · 10 years ago
  57. 27f5317 Use the prod GAE server in AppRTCDemo for iOS. by jiayl@webrtc.org · 10 years ago
  58. 5eb71eb Fix style issues from lint. by jiayl@webrtc.org · 10 years ago
  59. 34ac956 Do not use openmax_dl for MIPS64 platform. by andrew@webrtc.org · 10 years ago
  60. b2bda67 Removing old channel code from a few more places. by glaznev@webrtc.org · 10 years ago
  61. a9b1ec0 Support for DTLS in OpenSSLAdapter by pthatcher@webrtc.org · 10 years ago
  62. c5fd66d Accept incoming pings before remote answer is set to reduce connection latency. by jiayl@webrtc.org · 10 years ago
  63. 84d8447 Minor fixes regarding accumulator usage on MIPS platforms. by andrew@webrtc.org · 10 years ago
  64. b024da3 Add support for audio device selection in AppRTCDemo. by henrika@webrtc.org · 10 years ago
  65. 5ad4178 Move the Jingle-specific network code into webrtc/libjingle. by pthatcher@webrtc.org · 10 years ago
  66. 46d4d29 Add field trial for screenshare bitrates when using temporal layers. by sprang@webrtc.org · 10 years ago
  67. 1be0a78 Removing giles@mozilla.com from WebRTC watchlist. by mflodman@webrtc.org · 10 years ago
  68. 53cb741 Make RelayServerTest use VirtualSocketServer. by pbos@webrtc.org · 10 years ago
  69. 086c8d5 Use a temporary buffer to scale a screencast in OnFrameCaptured by braveyao@webrtc.org · 10 years ago
  70. 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
  71. ed1a48b Fix mac video capture leak. by tkchin@webrtc.org · 10 years ago
  72. 7ce4a58 Add initWithCoder to RTCEAGLVideoView. by tkchin@webrtc.org · 10 years ago
  73. ae643ce Wire up Beamformer in AudioProcessing by aluebs@webrtc.org · 10 years ago
  74. 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
  75. 50f7db8 Remove unneccessary lock causing a potential deadlock. by stefan@webrtc.org · 10 years ago
  76. a6f7ba6 Add a AppRTCDemo setting to change the GAE server. by jiayl@webrtc.org · 10 years ago
  77. 5570769 Remove the last getters from VideoReceiveStream stats. by pbos@webrtc.org · 10 years ago
  78. 742386a Enable payload-based padding by default and remove the API. by stefan@webrtc.org · 10 years ago
  79. aa21f27 Unify the two copies of move.h by kwiberg@webrtc.org · 10 years ago
  80. d16e839 Rtp-Rtcp sender cleanup. by pbos@webrtc.org · 10 years ago
  81. 556caff GN: Fix build for Mac by kjellander@webrtc.org · 10 years ago
  82. 11d8176 Move updating nack bitrate inside UpdateNACKBitRate. by stefan@webrtc.org · 10 years ago
  83. 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
  84. 0c39e91 Merge beamformer by aluebs@webrtc.org · 10 years ago
  85. 1090a6e Remove obsolete target_arch == armv7. by andrew@webrtc.org · 10 years ago
  86. 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
  87. 16a05dd Clean up the Channel code in AppRTCDemo and use GAE prod server for new signaling mode. by jiayl@webrtc.org · 10 years ago
  88. 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
  89. cb79141 Store the received report blocks map (mapped per remote ssrc) in a map per source ssrc. by asapersson@webrtc.org · 10 years ago
  90. ce4e9a3 Refactor some receive-side stats. by pbos@webrtc.org · 10 years ago
  91. 98c04b3 Get avg_delay_ms from DecoderTiming callback. by pbos@webrtc.org · 10 years ago
  92. 9b79197 Suppress REMB in bitrate ctrl if it seems lika a short network glitch. by sprang@webrtc.org · 10 years ago
  93. f832a6d Remove _t from function pointer typedefs. by pbos@webrtc.org · 10 years ago
  94. eed7a22 Make an AudioEncoder subclass for iSAC redundant encoding by henrik.lundin@webrtc.org · 10 years ago
  95. dd8f6f3 Rename rtpDumpPktHdr_t to RtpDumpPacketHeader. by pbos@webrtc.org · 10 years ago
  96. a9cf079 Rename external_hmac_ctx_t to ExternalHmacContext. by pbos@webrtc.org · 10 years ago
  97. e468bc9 Rename _t struct types in audio_processing. by pbos@webrtc.org · 10 years ago
  98. cab1291 Fixing the memory leak in AudioEncoderCopyRedDeathTest.NullSpeechEncoder by henrik.lundin@webrtc.org · 10 years ago
  99. 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
  100. 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