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gerrit-public.fairphone.software / platform / external / webrtc / a8b97373d5d3154357cc6589ff949ee9f6f99d8d
  1. a8b9737 Add tests and modify tools for new float deinterleaved interface. by andrew@webrtc.org · 10 years ago
  2. 3046b84 Adding new data files for audio classifier unit testing on Android try bots by jan.skoglund@webrtc.org · 10 years ago
  3. d3d6bce (Auto)update libjingle 62865357-> 62871616 by henrike@webrtc.org · 10 years ago
  4. d32797f Add a float interface to PushSincResampler. by andrew@webrtc.org · 10 years ago
  5. bc206ea iOS video_render: omit no-op setNeedsDisplay by fischman@webrtc.org · 10 years ago
  6. f792d17 AppRTCDemo(iOS): video support; part 1 of 2: webrtc/. by fischman@webrtc.org · 10 years ago
  7. 0537634 (Auto)update libjingle 62713454-> 62865357 by henrike@webrtc.org · 10 years ago
  8. 4a47be0 Disable CallTest.ReceivesAndRetransmitsNack for TSan by kjellander@webrtc.org · 10 years ago
  9. 36b6221 Adding a link to issue by henrik.lundin@webrtc.org · 10 years ago
  10. 6b0cbcb Roll chromium_revision 249215:255773 by kjellander@webrtc.org · 10 years ago
  11. 9b5f4d8 Fix build breakage introduce with r5665. by stefan@webrtc.org · 10 years ago
  12. f9e7c9d Add option to bwe_rtp_to_text to output arrival times only in nanoseconds. by stefan@webrtc.org · 10 years ago
  13. a01daf0 RTCPeerConnectionTest(objc): deflake by ignoring ICECompleted. by fischman@webrtc.org · 10 years ago
  14. 13320ea PeerConnectionTest(objc): expect ICE Completed state post 61460797-p10 by fischman@webrtc.org · 10 years ago
  15. 7811469 Roll libvpx 251850:254609 by marpan@webrtc.org · 10 years ago
  16. 11aab0e Populate VoiceReceiverInfo::delay_estimate_ms, jitter_buffer_ms, and jitter_buffer_preferred_ms to getStats. by jiayl@webrtc.org · 10 years ago
  17. 64e0405 Avoid crash in ViEEncoder::DeRegisterExternalEncoder(). by fischman@webrtc.org · 10 years ago
  18. cc08e3f Moves WEBRTC_POSIX define from header file to gyp-settings. by henrike@webrtc.org · 10 years ago
  19. 3ecc162 Remove std:: prefixes from C functions in webrtc/. by pbos@webrtc.org · 10 years ago
  20. 371243d Remove std:: prefixes from C functions in talk/. by pbos@webrtc.org · 10 years ago
  21. 46509c8 adding FEC support to WebRTC Opus wrapper and tests. by minyue@webrtc.org · 10 years ago
  22. 0454688 This CL is to add Opus complexity knob and to test it. by minyue@webrtc.org · 10 years ago
  23. ebdb0e3 Help to land 7969005 on behalf of solenberg. The review and try is done in 7969005. by wu@webrtc.org · 10 years ago
  24. 79047f9 (Auto)update libjingle 62691533-> 62713454 by henrike@webrtc.org · 10 years ago
  25. 2d213e4 (Auto)update libjingle 62550414-> 62691533 by henrike@webrtc.org · 10 years ago
  26. f714e7f Remove abs() use in PseudoTcp::process. by pbos@webrtc.org · 10 years ago
  27. 4584697 Fixes a bug in the simulation framework where the time offset is accumulating as the packet trace is repeated, causing increasingly large gaps with no packets being transmitted. by stefan@webrtc.org · 10 years ago
  28. ed865b5 NetEq4: Changing the behavior of playout_timestamp_ update by henrik.lundin@webrtc.org · 10 years ago
  29. 60ad5fd Potential deadlock in VideoSendStreamTest::ProducesStats by sprang@webrtc.org · 10 years ago
  30. 998cb8f Use DISABLE_ instead of commenting out tests by henrik.lundin@webrtc.org · 10 years ago
  31. 845862f Adding a new ramp-up-down-up test by henrik.lundin@webrtc.org · 10 years ago
  32. a0d11da Remove upper check for number of cores in VCM, I didn't find any good reasons for checking this. by mflodman@webrtc.org · 10 years ago
  33. cf85f1c Reorganize libjingle path variables. by kjellander@webrtc.org · 10 years ago
  34. 9f4d212 adding sha1 files for audio classifier test by jan.skoglund@webrtc.org · 10 years ago
  35. 3e0b60f Switch to correct interpretation of int and float input data in audio_processing_unittest by bjornv@webrtc.org · 10 years ago
  36. 17e4064 Add a deinterleaved float interface to AudioProcessing. by andrew@webrtc.org · 10 years ago
  37. b90991d Update libjingle 62472237->62550414 by henrike@webrtc.org · 10 years ago
  38. 7bd4a27 VideoCaptureAndroid: don't deliver frames after stopCapture(). by fischman@webrtc.org · 10 years ago
  39. be50ab6 Including algorithm header to avoid VS2013 breakage by henrik.lundin@webrtc.org · 10 years ago
  40. 52e898d Add .bin and .rx files to svn:ignore in resources by kjellander@webrtc.org · 10 years ago
  41. 24dae94 Add pthatcher@webrtc.org to talk/OWNERS. by pbos@webrtc.org · 10 years ago
  42. a25a92e Add third_party dependencies to svn:ignore by kjellander@webrtc.org · 10 years ago
  43. db41b4d Remove the deprecated GetStats method from PeerConnectionInterface. by jiayl@webrtc.org · 10 years ago
  44. 80bbf4c Enable test SSLStreamAdapterTestDTLS.TestDTLSConnectWithSmallMtu since it does not fail anymore. by jiayl@webrtc.org · 10 years ago
  45. 40b3b68 Update libjingle 62364298->62472237 by henrike@webrtc.org · 10 years ago
  46. 1bbfb57 Rollback of r5629 "(Auto)update libjingle 62364298-> 62368661". by henrike@webrtc.org · 10 years ago
  47. 0117d1c Fix compilation errors under clang 3.5. by pbos@webrtc.org · 10 years ago
  48. 31413dc (Auto)update libjingle 62364298-> 62368661 by henrike@webrtc.org · 10 years ago
  49. 10adbef Exclude /out* instead of just /out from pylint checks. by fischman@webrtc.org · 10 years ago
  50. 2bd5944 Re-enable libjingle_peerconnection_java_unittest since bug 2952 is fixed. by fischman@webrtc.org · 10 years ago
  51. d3dc424 Remove posting of ICE messages from WebRTCSession in PeerConnection to signaling thread. by mallinath@webrtc.org · 10 years ago
  52. bcfc167 AppRTCDemo(android): don't send local SDP until it's set. by fischman@webrtc.org · 10 years ago
  53. b898ce9 Revert of r5622 "disable unit tests" as it should be fixed in r5623. by henrike@webrtc.org · 10 years ago
  54. b8395eb (Auto)update libjingle 62293974-> 62364298 by henrike@webrtc.org · 10 years ago
  55. eec3843 TSAN only disable of two of libjingle's tests for atomic ops as they are failing for TSAN-bot. by henrike@webrtc.org · 10 years ago
  56. 9fd8d87 Adds APIs for reporting pacer queuing delay. by jiayl@webrtc.org · 10 years ago
  57. 56e4a05 Remove ProcessingComponent's dependence on AudioProcessingImpl. by andrew@webrtc.org · 10 years ago
  58. 806768a (Auto)update libjingle 62281784-> 62293974 by henrike@webrtc.org · 10 years ago
  59. 704bf9e (Auto)update libjingle 62063505-> 62278774 by henrike@webrtc.org · 10 years ago
  60. f0fc72f Call PrintWindow for the first time of capturing to capture the window frames correctly. by jiayl@webrtc.org · 10 years ago
  61. 00073aa Clean up CPU detection defines in SincResampler a little. by andrew@webrtc.org · 10 years ago
  62. 0231e80 Invalidate the whole screen when the frame size is changed. by jiayl@webrtc.org · 10 years ago
  63. 2038920 Use scoped_ptr<T[]> in SincResampler to avoid .get()[] weirdness. by andrew@webrtc.org · 10 years ago
  64. c0e9aeb Add SetConfig method to FakeNetworkPipe and to DirectTransport by henrik.lundin@webrtc.org · 10 years ago
  65. eaadeca iOS, AppRTCDemo: Fixes exception due to JSON for ice using "urls" instead of "url", which is introduced by r5599. by braveyao@webrtc.org · 10 years ago
  66. 90173e1 Roll libvpx 248011:251850 by marpan@webrtc.org · 10 years ago
  67. bc1d224 Add experimental noise suppression flag to audioproc test by aluebs@webrtc.org · 10 years ago
  68. 050892a Missing include in experiments.h by sprang@webrtc.org · 10 years ago
  69. 7f52a6e Split the implementation of VP8Encoder|Decoder::Create into a seperated file by wu@webrtc.org · 10 years ago
  70. 79a1cff Android, AppRTCDemo: Fixes java exception due to JSON for ice using "urls" instead of "url". by henrike@webrtc.org · 10 years ago
  71. bf88ecc Added turn-prober.sh: a super-simple prober for TURN servers & candidates. by fischman@webrtc.org · 10 years ago
  72. 78ea3d5 Check pcConfig (which can be null) before use. by wu@webrtc.org · 10 years ago
  73. 91cbaa4 (Auto)update libjingle 61966318-> 62063505 by henrike@webrtc.org · 10 years ago
  74. 23caa2d Fix to get total number of sent and received rtcp packets. by asapersson@webrtc.org · 10 years ago
  75. 4f0801b AviRecorder is missing a critical section. by braveyao@webrtc.org · 10 years ago
  76. bc0470f AppRTC Sample: Switch AppRTC to use RTCIceServer.urls. by braveyao@webrtc.org · 10 years ago
  77. 55fcd71 Disable libjingle_peerconnection_java_unittest by kjellander@webrtc.org · 10 years ago
  78. 33af96c Removed unused mock methods in audio_processing by bjornv@webrtc.org · 10 years ago
  79. d43aa9d Update libjingle 61901702->61966318 by henrike@webrtc.org · 10 years ago
  80. a7b9818 Unrevert 5590 "description"(=(Auto)update libjingle 61834300->61901702). by henrike@webrtc.org · 10 years ago
  81. 125a66a Memory and Tsan tests: Turn off the new-ACM tests by tina.legrand@webrtc.org · 10 years ago
  82. ef22151 Revert 5590 "description" by xians@webrtc.org · 10 years ago
  83. 0f2809a Add RTCP packet class. Adds packet types: sr, rr, bye, fir. by asapersson@webrtc.org · 10 years ago
  84. c0907ef MIPS optimizations for AEC audio processing module by andrew@webrtc.org · 10 years ago
  85. 2643805 description by henrike@webrtc.org · 10 years ago
  86. 3f170dd Updated WebRTC version to 3.50 TBR= wu@webrtc.org by elham@webrtc.org · 10 years ago
  87. d617a44 Add an AlignedFreeDeleter and remove scoped_ptr_malloc. by andrew@webrtc.org · 10 years ago
  88. d4d5be8 Minor improvement in RoundToInt16 implementation. by turaj@webrtc.org · 10 years ago
  89. a0a6df3 Modified overuse detection thresholds. by asapersson@webrtc.org · 10 years ago
  90. 04a691a Removing a variable that was never read by henrik.lundin@webrtc.org · 10 years ago
  91. 6606199 ifdef the alsa code based on macro USE_X11 by fbarchard@google.com · 10 years ago
  92. 056176b Presubmit script that prohibits cls to both trunk/webrtc and trunk/talk. by henrike@webrtc.org · 10 years ago
  93. 78f0db4 Fix the break caused by r5579. by turaj@webrtc.org · 10 years ago
  94. 571df2d Update libjingle 61759961->61834300 by henrike@webrtc.org · 10 years ago
  95. c2d69d3 Resolves memcheck issue in AudioCodingModuleTest. The issue is coditional jumnp based on uninitialized variable. by turaj@webrtc.org · 10 years ago
  96. 97e7a64 Make WindowCapturerLinux handling window resize events. by jiayl@webrtc.org · 10 years ago
  97. 2421025 Added architecture for compiling under chrome NaCl. by andresp@webrtc.org · 10 years ago
  98. 056287e This CL separate all ACM tests with new and old implementation of ACM and NetEq. The reason is to debug an issue with failure on Android try bots. We need to see if the error only occurs with the new ACM/NetEq, or if it is a flakiness that affects both. by tina.legrand@webrtc.org · 10 years ago
  99. 8098e07 Add RTCP packet type counter (for getting statistics such as sent/received NACK and FIR). by asapersson@webrtc.org · 10 years ago
  100. b7a91fa Removes VoERTP_RTCP::InsertExtraRTPPacket. by henrika@webrtc.org · 10 years ago