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gerrit-public.fairphone.software / platform / external / webrtc / bcfc1670d6454e547a79b983162e363a3a54f1dd
  1. bcfc167 AppRTCDemo(android): don't send local SDP until it's set. by fischman@webrtc.org · 11 years ago
  2. b898ce9 Revert of r5622 "disable unit tests" as it should be fixed in r5623. by henrike@webrtc.org · 11 years ago
  3. b8395eb (Auto)update libjingle 62293974-> 62364298 by henrike@webrtc.org · 11 years ago
  4. eec3843 TSAN only disable of two of libjingle's tests for atomic ops as they are failing for TSAN-bot. by henrike@webrtc.org · 11 years ago
  5. 9fd8d87 Adds APIs for reporting pacer queuing delay. by jiayl@webrtc.org · 11 years ago
  6. 56e4a05 Remove ProcessingComponent's dependence on AudioProcessingImpl. by andrew@webrtc.org · 11 years ago
  7. 806768a (Auto)update libjingle 62281784-> 62293974 by henrike@webrtc.org · 11 years ago
  8. 704bf9e (Auto)update libjingle 62063505-> 62278774 by henrike@webrtc.org · 11 years ago
  9. f0fc72f Call PrintWindow for the first time of capturing to capture the window frames correctly. by jiayl@webrtc.org · 11 years ago
  10. 00073aa Clean up CPU detection defines in SincResampler a little. by andrew@webrtc.org · 11 years ago
  11. 0231e80 Invalidate the whole screen when the frame size is changed. by jiayl@webrtc.org · 11 years ago
  12. 2038920 Use scoped_ptr<T[]> in SincResampler to avoid .get()[] weirdness. by andrew@webrtc.org · 11 years ago
  13. c0e9aeb Add SetConfig method to FakeNetworkPipe and to DirectTransport by henrik.lundin@webrtc.org · 11 years ago
  14. eaadeca iOS, AppRTCDemo: Fixes exception due to JSON for ice using "urls" instead of "url", which is introduced by r5599. by braveyao@webrtc.org · 11 years ago
  15. 90173e1 Roll libvpx 248011:251850 by marpan@webrtc.org · 11 years ago
  16. bc1d224 Add experimental noise suppression flag to audioproc test by aluebs@webrtc.org · 11 years ago
  17. 050892a Missing include in experiments.h by sprang@webrtc.org · 11 years ago
  18. 7f52a6e Split the implementation of VP8Encoder|Decoder::Create into a seperated file by wu@webrtc.org · 11 years ago
  19. 79a1cff Android, AppRTCDemo: Fixes java exception due to JSON for ice using "urls" instead of "url". by henrike@webrtc.org · 11 years ago
  20. bf88ecc Added turn-prober.sh: a super-simple prober for TURN servers & candidates. by fischman@webrtc.org · 11 years ago
  21. 78ea3d5 Check pcConfig (which can be null) before use. by wu@webrtc.org · 11 years ago
  22. 91cbaa4 (Auto)update libjingle 61966318-> 62063505 by henrike@webrtc.org · 11 years ago
  23. 23caa2d Fix to get total number of sent and received rtcp packets. by asapersson@webrtc.org · 11 years ago
  24. 4f0801b AviRecorder is missing a critical section. by braveyao@webrtc.org · 11 years ago
  25. bc0470f AppRTC Sample: Switch AppRTC to use RTCIceServer.urls. by braveyao@webrtc.org · 11 years ago
  26. 55fcd71 Disable libjingle_peerconnection_java_unittest by kjellander@webrtc.org · 11 years ago
  27. 33af96c Removed unused mock methods in audio_processing by bjornv@webrtc.org · 11 years ago
  28. d43aa9d Update libjingle 61901702->61966318 by henrike@webrtc.org · 11 years ago
  29. a7b9818 Unrevert 5590 "description"(=(Auto)update libjingle 61834300->61901702). by henrike@webrtc.org · 11 years ago
  30. 125a66a Memory and Tsan tests: Turn off the new-ACM tests by tina.legrand@webrtc.org · 11 years ago
  31. ef22151 Revert 5590 "description" by xians@webrtc.org · 11 years ago
  32. 0f2809a Add RTCP packet class. Adds packet types: sr, rr, bye, fir. by asapersson@webrtc.org · 11 years ago
  33. c0907ef MIPS optimizations for AEC audio processing module by andrew@webrtc.org · 11 years ago
  34. 2643805 description by henrike@webrtc.org · 11 years ago
  35. 3f170dd Updated WebRTC version to 3.50 TBR= wu@webrtc.org by elham@webrtc.org · 11 years ago
  36. d617a44 Add an AlignedFreeDeleter and remove scoped_ptr_malloc. by andrew@webrtc.org · 11 years ago
  37. d4d5be8 Minor improvement in RoundToInt16 implementation. by turaj@webrtc.org · 11 years ago
  38. a0a6df3 Modified overuse detection thresholds. by asapersson@webrtc.org · 11 years ago
  39. 04a691a Removing a variable that was never read by henrik.lundin@webrtc.org · 11 years ago
  40. 6606199 ifdef the alsa code based on macro USE_X11 by fbarchard@google.com · 11 years ago
  41. 056176b Presubmit script that prohibits cls to both trunk/webrtc and trunk/talk. by henrike@webrtc.org · 11 years ago
  42. 78f0db4 Fix the break caused by r5579. by turaj@webrtc.org · 11 years ago
  43. 571df2d Update libjingle 61759961->61834300 by henrike@webrtc.org · 11 years ago
  44. c2d69d3 Resolves memcheck issue in AudioCodingModuleTest. The issue is coditional jumnp based on uninitialized variable. by turaj@webrtc.org · 11 years ago
  45. 97e7a64 Make WindowCapturerLinux handling window resize events. by jiayl@webrtc.org · 11 years ago
  46. 2421025 Added architecture for compiling under chrome NaCl. by andresp@webrtc.org · 11 years ago
  47. 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 · 11 years ago
  48. 8098e07 Add RTCP packet type counter (for getting statistics such as sent/received NACK and FIR). by asapersson@webrtc.org · 11 years ago
  49. b7a91fa Removes VoERTP_RTCP::InsertExtraRTPPacket. by henrika@webrtc.org · 11 years ago
  50. e384104 Fix DesktopAndCursorComposer not to crash by sergeyu@chromium.org · 11 years ago
  51. 5cf3e8f (Auto)update libjingle $LAST_P10_REVISION-> $NEW_P10_REVISION by henrike@webrtc.org · 11 years ago
  52. 27c6980 Move the volume quantization workaround from VoE to AGC. by andrew@webrtc.org · 11 years ago
  53. 00844d7 Remove obsolete voe_unit_test. by solenberg@webrtc.org · 11 years ago
  54. 358e336 PeerConnection(java): enable HW encoder on N5 for standalone build. by fischman@webrtc.org · 11 years ago
  55. c2d75e0 PeerConnection(java): account for thread shutdown vagaries. by fischman@webrtc.org · 11 years ago
  56. c320027 Don't print a warning if RTPPacketHistory::SetStorePacketStatus is called by mflodman@webrtc.org · 11 years ago
  57. 2086e0f Remove unnecessary warnings. by turaj@webrtc.org · 11 years ago
  58. a079233 Remove external encryption API for VoE. by solenberg@webrtc.org · 11 years ago
  59. 0a9d822 Change mime type to text/html for multiple-relay.html by kjellander@webrtc.org · 11 years ago
  60. 346094c Incorrect overhead calculation when using FEC + RTP extension headers. by sprang@webrtc.org · 11 years ago
  61. b60346e Reset estimate if no frame has been seen for a certain time (to avoid large jitter if stop sending). by asapersson@webrtc.org · 11 years ago
  62. 92fdfeb Update talk to 61699344. by mallinath@webrtc.org · 11 years ago
  63. e384289 Adding tsan suppression for error introduced in r5555, causing libjingle_unittest to fail on TSan bot. by mflodman@webrtc.org · 11 years ago
  64. 340746a Misc small nits in NetEq by henrik.lundin@webrtc.org · 11 years ago
  65. 1009798 Demo of multi-pass encode - used for testing limits. by hta@webrtc.org · 11 years ago
  66. f92aaff AudioProcessing is not a Module. by andrew@webrtc.org · 11 years ago
  67. b8c254a (Auto)update libjingle 61549749-> 61608469 by henrike@webrtc.org · 11 years ago
  68. e2fc13e Refactoring common_audio/signal_processing: Removed two macros used by isac only. by bjornv@webrtc.org · 11 years ago
  69. c5d506a AppRTCDemo(android): clarified README on how to launch app using adb. by fischman@webrtc.org · 11 years ago
  70. 505f2a0 Disabling WebRtcSessionTest.TestIceStatesBundle under memcheck. by stefan@webrtc.org · 11 years ago
  71. 9075d51 Adding a critical section missing in r5543. by stefan@webrtc.org · 11 years ago
  72. a3708ec PeerConnectionTest(java): unbreak following 61460797-p10 by fischman@webrtc.org · 11 years ago
  73. 385857d Update talk to 61549749. by mallinath@webrtc.org · 11 years ago
  74. b9a088b Update talk to 61538839. by wu@webrtc.org · 11 years ago
  75. 0de2950 Revert 5545 "Update libjingle to 61514460" by wu@webrtc.org · 11 years ago
  76. 38bf249 Initialize output_will_be_muted_. by andrew@webrtc.org · 11 years ago
  77. e749c9e Update libjingle to 61514460 by xians@webrtc.org · 11 years ago
  78. 8f690bc Increase overuse and normal use thresholds for Mac. by asapersson@webrtc.org · 11 years ago
  79. ae2563a Fixes a race when writing to send_padding_. by stefan@webrtc.org · 11 years ago
  80. 12cb88c Add check to verify tree is open to PRESUBMIT.py. by kjellander@webrtc.org · 11 years ago
  81. fcfc6a9 Small refactoring of NetEq unittest for CNG with clock drift by henrik.lundin@webrtc.org · 11 years ago
  82. 3eda643 PeerConnection(java): added MediaConstraints support to AudioSource, now fed to AudioTrack. by fischman@webrtc.org · 11 years ago
  83. 540acde PeerConnection(java): use MediaCodec for HW-accelerated video encode where available. by fischman@webrtc.org · 11 years ago
  84. 17342e5 Add a method to inform AudioProcessing that its output will be muted. by andrew@webrtc.org · 11 years ago
  85. de78218 Change the type of propagation delta from int64 to int. by jiayl@webrtc.org · 11 years ago
  86. 07b5950 Initialize key_pressed_. by andrew@webrtc.org · 11 years ago
  87. ce8e077 Add a keypress field to the audioproc debug proto. by andrew@webrtc.org · 11 years ago
  88. 8118f18 Set pacing bitrates in SetEncoder. by pbos@webrtc.org · 11 years ago
  89. 67e7044 Remove unused and not working voe_extended_test. by solenberg@webrtc.org · 11 years ago
  90. 5591046 .gitignore: + /third_party/{clang_format,usrcsctp} by pbos@webrtc.org · 11 years ago
  91. 14d8079 PeerConnectionClient needs to initialize SSL. BUG=2911 R=fischman@webrtc.org by jiayl@webrtc.org · 11 years ago
  92. b659e28 Reduce mixing threshold in test to avoid flakiness. by andrew@webrtc.org · 11 years ago
  93. 75dd288 Add an interface for accepting keypress signals to AudioProcessing. by andrew@webrtc.org · 11 years ago
  94. aa1278d Rename merged webrtc lib to libwebrtc_merged.a. by andrew@webrtc.org · 11 years ago
  95. 8685af7 Remove "Too long processing time of Incoming frame" logspam. by fischman@webrtc.org · 11 years ago
  96. a80be4b Add boundary checking to supress gcc 4.8.3 warning. by turaj@webrtc.org · 11 years ago
  97. fc32046 Remove ViE external encryption API. by solenberg@webrtc.org · 11 years ago
  98. 82ebb46 Use libvpx's obj_int_extract and unpack_lib_posix to generate offset header file. by michaelbai@google.com · 11 years ago
  99. dd82fa7 Revert 5516 "Thread annotation of talk_base::CriticalSection." by wjia@webrtc.org · 11 years ago
  100. 16c08f0 Restore mixing integration tests. by andrew@webrtc.org · 11 years ago