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gerrit-public.fairphone.software / platform / external / webrtc / c18957e8773e558de869a809df97ec6ca924794b
  1. c18957e Make Git ignore in resources more fine-grained by kjellander@webrtc.org · 9 years ago
  2. 354becf Remove Git ignore exclusion of .sha1 files by kjellander@webrtc.org · 9 years ago
  3. 7cc92aa Use WebRtcVideoRenderFrame for texture frames. by pbos@webrtc.org · 9 years ago
  4. 62f6e75 Refactoring WebRTC Java/JNI audio recording in C++ and Java. by henrika@webrtc.org · 9 years ago
  5. c2d0473 Switch to using AudioEncoderPcm16B instead of ACMPCM16B by henrik.lundin@webrtc.org · 9 years ago
  6. f58fe0a Rename GYP and GN targets for video capture+render. by kjellander@webrtc.org · 9 years ago
  7. 2c29c2e C++ readability review for ajm. by andrew@webrtc.org · 9 years ago
  8. 5d60895 Fix bug when there are no blocks in a chunk in Beamformer by aluebs@webrtc.org · 9 years ago
  9. bc35703 Add a method to remove an existing renderer from the internal list of Android renderers. by glaznev@webrtc.org · 9 years ago
  10. bc40324 Merge fixes and changed for Android AppRTCDemo from internal repo. by glaznev@webrtc.org · 9 years ago
  11. d35a5c3 Make ChannelBuffer aware of frequency bands by aluebs@webrtc.org · 9 years ago
  12. d7472b5 base/arraysize.h: We use size_t, so need to include stddef.h by kwiberg@webrtc.org · 9 years ago
  13. 91ba79a Make sure that the norms are positive in Beamformer by aluebs@webrtc.org · 9 years ago
  14. b6856d2 Apply mask smoothing in Beamformer by aluebs@webrtc.org · 9 years ago
  15. 8da96ac Switch to using AudioEncoderIlbc instead of ACMILBC by henrik.lundin@webrtc.org · 9 years ago
  16. 1a072f9 Address comments from previous review round for rtc::Event. by tommi@webrtc.org · 9 years ago
  17. f4c10d2 Always use DeliverI420Frame in WebRtcVideoEngine. by pbos@webrtc.org · 9 years ago
  18. 027e113 Introduce PacketReceiver and remove configuration of simulations via the BweTestConfig. by stefan@webrtc.org · 9 years ago
  19. 30015e3 Fix bug in EventPosix where we'd miss a set event. by tommi@webrtc.org · 9 years ago
  20. 648f5d6 pcm16b: Make input arrays const and use uint8_t[] for byte arrays by kwiberg@webrtc.org · 9 years ago
  21. 948d617 Create a separate thread for pacing. by mflodman@webrtc.org · 9 years ago
  22. c11348b Fixing a bug in expand_rate calculation for stereo signal. by minyue@webrtc.org · 9 years ago
  23. 8e612ab Remove voice_engine_ member variable and GetVoiceEngine() from ViEChannelManager. by tommi@webrtc.org · 9 years ago
  24. 5b8f3e0 Roll chromium_revision 598c3e9..601e6f3 by kjellander@webrtc.org · 9 years ago
  25. 44ae4c8 Support using VP9 video codec in AppRTCDemo. by glaznev@webrtc.org · 9 years ago
  26. f7e6cfd Add CHECK to EventWrapper to see if there's a subtle bug there or not. by tommi@webrtc.org · 9 years ago
  27. 669bc7e Modify default field trial implementation to allow by glaznev@webrtc.org · 9 years ago
  28. 11c5db0 Revert 8273 "Temporarily change ThreadPosix to CHECK (crash) if ..." by tommi@webrtc.org · 9 years ago
  29. 0d852d5 Use VideoReceiveStream as an ExternalRenderer. by pbos@webrtc.org · 9 years ago
  30. d6e25a5 Revert r8297 "Introduce PacketReceiver and remove configuration of simulations via the BweTestConfig." by stefan@webrtc.org · 9 years ago
  31. 03c1c10 Introduce PacketReceiver and remove configuration of simulations via the BweTestConfig. by stefan@webrtc.org · 9 years ago
  32. 53d9012 Clean kForever from basictypes and move it to the interfaces that actually have it. by andresp@webrtc.org · 9 years ago
  33. e01bae2 Fixing a nit by henrik.lundin@webrtc.org · 9 years ago
  34. 1c6239a G711: Make input arrays const and use uint8_t[] for byte arrays by kwiberg@webrtc.org · 9 years ago
  35. d0165c6 Use a manual reset event in PosixThread. by tommi@webrtc.org · 9 years ago
  36. 4c0fd96 Move rtc::Event to rtc_base_approved. We need an event implementation in WebRTC that allows us to specify whether it's manually reset or automatically. EventWrapper currently doesn't support it and it adds a heap allocation + vtable, so rtc::Event is the lighter of the two. by tommi@webrtc.org · 9 years ago
  37. 8cf9bdb Remove USE_WEBRTC_DEV_BRANCH. by pbos@webrtc.org · 9 years ago
  38. 2b69eab Restructure GYP for vp9, opus and direct trace by kjellander@webrtc.org · 9 years ago
  39. f31f56d Remove default arguments in EncodedImageCallback. by changbin.shao@webrtc.org · 9 years ago
  40. 6c930c7 Cleanup: unify rotation to be enum based instead of int for degree. by guoweis@webrtc.org · 9 years ago
  41. 7a57f8f Reland 8203 "Reducing locking in OveruseFrameDetect..." by tommi@webrtc.org · 9 years ago
  42. 103f328 Fix the binary layout of ProcessThreadImpl. by tommi@webrtc.org · 9 years ago
  43. ec499be Increase testclient timeout from 1 to 5 seconds by jlmiller@webrtc.org · 9 years ago
  44. fe19699 Revert 8260 "Base RWLockWrapper on rtc::SharedExclusiveLock." by tommi@webrtc.org · 9 years ago
  45. 2eb1660 Switch ThreadCheckerImpl over to using PlatformThreadRef. by tommi@webrtc.org · 9 years ago
  46. 2bf0e90 Revert 8275 "This CL adds an API to the SSL stream adapters and ..." by tommi@webrtc.org · 9 years ago
  47. 1d4830a Disable ProcessThread tests that are dependent on timing. by tommi@webrtc.org · 9 years ago
  48. 95a32ec Revert 8271 "VirtualSocketServer out-of-order issue with closing..." by bjornv@webrtc.org · 9 years ago
  49. 2a44be9 Normalize delay-and-sum mask in Beamformer by aluebs@webrtc.org · 9 years ago
  50. 799e667 Add high frequency correction to Beamformer by aluebs@webrtc.org · 9 years ago
  51. 0c7ec77 Cleanup: unify rotation to be enum based instead of int for degree. by guoweis@webrtc.org · 9 years ago
  52. 110443a Cleanup: unify rotation to be enum based instead of int for degree. by guoweis@webrtc.org · 9 years ago
  53. 1d11c82 This CL adds an API to the SSL stream adapters and transport channels to get the SSL cipher that was negotiated with the remote peer. by pthatcher@webrtc.org · 9 years ago
  54. 63da1dd audio_processing: Now records mic volume level also when using new AGC by bjornv@webrtc.org · 9 years ago
  55. ccd7e99 Temporarily change ThreadPosix to CHECK (crash) if we ever spend more than 30 seconds waiting for thread shutdown. There are cases on build bots where it looks like we're hitting this problem, but reproducing locally has been a struggle. by tommi@webrtc.org · 9 years ago
  56. 13a0e18 Temporarily disable a couple of ThreadChecker tests on Mac. by tommi@webrtc.org · 9 years ago
  57. 4770437 VirtualSocketServer out-of-order issue with closing TCP sockets by pthatcher@webrtc.org · 9 years ago
  58. 9baa9ca Add libjingle_peerconnection_so.so to Java test dependencies. by perkj@webrtc.org · 9 years ago
  59. b5a1252 Hack to work around the current issues with rolling WebRTC into chromium. by tommi@webrtc.org · 9 years ago
  60. 751a365 Switch to using AudioEncoderPcmU/A instead of ACMPCMU/A by henrik.lundin@webrtc.org · 9 years ago
  61. 02270cd Implementing a packet router class, used to route RTP packets to the by mflodman@webrtc.org · 9 years ago
  62. 10a9e92 Fix delete of stack allocated object causing test crashes. by stefan@webrtc.org · 9 years ago
  63. 4b320cf Revert "Cleanup: unify rotation to be enum based instead of int for degree." by magjed@webrtc.org · 9 years ago
  64. fb609a1 Wire up new feedback format by introducing a FeedbackPacket type. by stefan@webrtc.org · 9 years ago
  65. 353c8b8 audio_processing/agc: Changed to correct include path in agc_unittests by bjornv@webrtc.org · 9 years ago
  66. bc3241a Update ProcessCallAfterXms to better match the performance of our faster bots. Previously I had made sure these tests didn't flake out on our slow trybots, but apparently I need to do the same for the fast bots :) by tommi@webrtc.org · 9 years ago
  67. 0c3e12b Revamp the ProcessThreadImpl implementation. by tommi@webrtc.org · 9 years ago
  68. 7502543 Base RWLockWrapper on rtc::SharedExclusiveLock. by pbos@webrtc.org · 9 years ago
  69. 5e05731 Roll chromium_revision cd35af6..598c3e9 by kjellander@webrtc.org · 9 years ago
  70. 57ac2c8 Default destination used by c line should be IPv4 only to avoid parsing error in legacy client. by guoweis@webrtc.org · 9 years ago
  71. 3e733a4 Cleanup: unify rotation to be enum based instead of int for degree. by guoweis@webrtc.org · 9 years ago
  72. 74d2788 Remove defined(__cplusplus) tests in C++ code. by jan.skoglund@webrtc.org · 9 years ago
  73. f45c8ca Reland r8248 "Introduce ACMGenericCodecWrapper" by henrik.lundin@webrtc.org · 9 years ago
  74. ec4521c Clean up Beamformer initialization by aluebs@webrtc.org · 9 years ago
  75. e69220c Fix the value of the first byte of nal unit generated by fake H.264 encoder. by glaznev@webrtc.org · 9 years ago
  76. f693229 Fix Android video renderer to support video frames with stride > width. by glaznev@webrtc.org · 9 years ago
  77. cc64a9c voice_engine: Updates GetEcDelayMetrics() w.r.t. new metric by bjornv@webrtc.org · 9 years ago
  78. 4b9622f Roll gtest-parallel. by pbos@webrtc.org · 9 years ago
  79. 3a87630 Revert r8248 "Introduce ACMGenericCodecWrapper" by henrik.lundin@webrtc.org · 9 years ago
  80. af8c13f Introduce ACMGenericCodecWrapper by henrik.lundin@webrtc.org · 9 years ago
  81. 5d32f43 Disable CondVarTest.InitFunctionsWork. by tommi@webrtc.org · 9 years ago
  82. 877ac76 Cleanup and prepare for bundling. by pthatcher@webrtc.org · 9 years ago
  83. cf7efeb Add new AudioEncoderOpusTest by henrik.lundin@webrtc.org · 9 years ago
  84. 520a69e Revert 8238 "Add RefCounting for TransportProxies" by bjornv@webrtc.org · 9 years ago
  85. 875c97e Remove SetNotAlive method from the thread class. by tommi@webrtc.org · 9 years ago
  86. c5f6971 Revert 8237 "Cleanup and prepare for bundling." by bjornv@webrtc.org · 9 years ago
  87. dc096f2 system_wrappers: Disabled flaky test CondVarTest.PassBatonMultipleTimes by bjornv@webrtc.org · 9 years ago
  88. 4414939 Add method for incrementing RtpPacketCounter. Removes duplicate code. by asapersson@webrtc.org · 9 years ago
  89. e250667 Add RefCounting for TransportProxies by decurtis@webrtc.org · 9 years ago
  90. af01d93 Cleanup and prepare for bundling. by pthatcher@webrtc.org · 9 years ago
  91. 322a564 Fix datachannel stats id and timestamp. by decurtis@webrtc.org · 9 years ago
  92. d43bdf5 Rewrite ThreadPosix. by tommi@webrtc.org · 9 years ago
  93. bfdee69 Roll chromium_revision 9070a80..cd35af6 (313233:314322) by kjellander@webrtc.org · 9 years ago
  94. 0ec50be Changing include guard in frame_callback.h. by mflodman@webrtc.org · 9 years ago
  95. 200ac00 Remove temp files in audio_processing_unittest.cc. by pbos@webrtc.org · 9 years ago
  96. 0e8bf6c Enable bitrate probing by default. by stefan@webrtc.org · 9 years ago
  97. b1786db audio_processing: Added a new AEC delay metric value that gives the amount of poor delays by bjornv@webrtc.org · 9 years ago
  98. 0e81fdf Avoid implicit type truncations by inserting explicit casts or modifying prototypes to avoid needless up- and then down-casting. by pkasting@chromium.org · 9 years ago
  99. 19f3f71 Fix apparent typo: int -> char. by pkasting@chromium.org · 9 years ago
  100. 946ad76 Switched lists of packets to lists of packet pointers. Allows Packet polymorphism. by stefan@webrtc.org · 9 years ago