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gerrit-public.fairphone.software / platform / external / webrtc / 3ee13e4ac219338c861a5d08212ffba88001cdb1
  1. 3ee13e4 Optimized DesktopRegion implementation. by sergeyu@chromium.org · 11 years ago
  2. 34a7735 Removed unused class members to enable clang=1 android build. by fischman@webrtc.org · 11 years ago
  3. 6eb0f6a Setting SSRC in vie_loopback_test by mikhal@webrtc.org · 11 years ago
  4. 0a38432 Fix error in mixing test for supported sample rates. by andrew@webrtc.org · 11 years ago
  5. fa64a59 Change SetRTPAudioLevelIndicationStatus to ignore the id in the case of disabling. by wu@webrtc.org · 11 years ago
  6. c1eb560 Replace the old resampler with SincResampler in the voice engine signal path. by andrew@webrtc.org · 11 years ago
  7. 31c5f1c Remove ancient and unused CNG test. by andrew@webrtc.org · 11 years ago
  8. 2b3a865 Revert 4149 "bug fixes for extremely large images - 10000x10000 ..." by mikhal@webrtc.org · 11 years ago
  9. b35d2e3 Add dummy audio NACK APIs by niklas.enbom@webrtc.org · 11 years ago
  10. b1bba16 Prevent excessive logging in jitter buffer by hclam@chromium.org · 11 years ago
  11. 85f2865 bug fixes for extremely large images - 10000x10000 and 100000 pixel wide. by fbarchard@google.com · 11 years ago
  12. a6494e6 roll libyuv to r711 for scaler fix to webrtc unittests that scale up and down and check for fairly similar results. by fbarchard@google.com · 11 years ago
  13. 694cdc6 Revert 4104 "Refactor jitter buffer to use separate lists for de..." by tnakamura@webrtc.org · 11 years ago
  14. 4d9c07a Revert 4127 "Switch frame list implementation to std::map." by tnakamura@webrtc.org · 11 years ago
  15. 5ed7051 Apprtc: not to start the call until we get Turn response. by braveyao@webrtc.org · 11 years ago
  16. f9f39d5 Add a drover.properties file for reference. by andrew@webrtc.org · 11 years ago
  17. eed919d MIPS optimizations for the following functions: by andrew@webrtc.org · 11 years ago
  18. adc64a7 VCM/Timing: Setting clear names to members & methods by mikhal@webrtc.org · 11 years ago
  19. fddf6be Updated apprtc to use new TURN format for chrome versions M28 & above. by vikasmarwaha@webrtc.org · 11 years ago
  20. 046bc44 Fixes the frameRate stats by grouping the frames by timestamp. by jiayl@webrtc.org · 11 years ago
  21. 4213633 Use int for FPS instead of size_t. by pbos@webrtc.org · 11 years ago
  22. a048d7c Include files from webrtc/.. paths in rtp_rtcp/ by pbos@webrtc.org · 11 years ago
  23. eea2622 Correctly set SSRCs for extra send RTP modules. by stefan@webrtc.org · 11 years ago
  24. 7bdfff3 Remove assert for aborting FrameGeneratorCapturer. by pbos@webrtc.org · 11 years ago
  25. 26d1210 Fake VideoCapturer based on FrameGenerator by pbos@webrtc.org · 11 years ago
  26. 08994cc Fix a return value mismatch introduced in r4129. by stefan@webrtc.org · 11 years ago
  27. 9aca5b3 Remove #pragma once by pbos@webrtc.org · 11 years ago
  28. a5cb98c Breaking out RTP header parsing from the RTP module. by stefan@webrtc.org · 11 years ago
  29. 1ecee9a Break video_engine/new_include/common.h into smaller parts. by pbos@webrtc.org · 11 years ago
  30. ace7ad2 Switch frame list implementation to std::map. by stefan@webrtc.org · 11 years ago
  31. f791b1c Rename voice_engine_core -> voice_engine and move targets to voice_engine.gyp. by andrew@webrtc.org · 11 years ago
  32. a6ae644 Add comment about test_packet_masks_metrics. by marpan@webrtc.org · 11 years ago
  33. fe6a75e Updated WebRTC version to 3.32 TBR=wu@webrtc.org by elham@webrtc.org · 11 years ago
  34. a066cbf Don't return an estimated receive BW for channels not receiving video. by mflodman@webrtc.org · 11 years ago
  35. 4079c31 Include gflags with "gflags/gflags.h" instead of <> by pbos@webrtc.org · 11 years ago
  36. 8c34cee Include "gtest/gtest.h", not by full path, on WEBRTC_ANDROID_PLATFORM_BUILD by pbos@webrtc.org · 11 years ago
  37. 3496ef1 Improve vie_autotest_rtp_rtcp by reenabling important tests and reducing flakiness. by stefan@webrtc.org · 11 years ago
  38. 15c1c61 Include files from webrtc/.. paths in audio_conference_mixer/ by pbos@webrtc.org · 11 years ago
  39. 7fad4b8 Include files from webrtc/.. paths in audio_processing/ by pbos@webrtc.org · 11 years ago
  40. eceb532 Default constructors for new VideoEngine structs. by pbos@webrtc.org · 11 years ago
  41. 68c05f4 Remove libvpx_intrinsics_sse4_1.a in Android.mk since this target is no longer generated in libvpx by fischman@webrtc.org · 11 years ago
  42. a6db54d - Created RemoteBitrateEstimator wrapper for use internally in (ViE) ChannelGroup. by solenberg@webrtc.org · 11 years ago
  43. 7f944f3 Adding Mac test renderer, some test refactoring and made cpplint pass. by mflodman@webrtc.org · 11 years ago
  44. acaf3a1 Include files from webrtc/.. paths in system_wrappers/ by pbos@webrtc.org · 11 years ago
  45. 1e50231 Include files from webrtc/.. paths in test/channel_transport/ by pbos@webrtc.org · 11 years ago
  46. 6f3d8fc Include files from webrtc/.. paths in video_processing/ by pbos@webrtc.org · 11 years ago
  47. 47ce120 Include files from webrtc/.. paths in remote_bitrate_estimator/ by pbos@webrtc.org · 11 years ago
  48. aa30bb7 Include files from webrtc/.. paths in common_audio/ by pbos@webrtc.org · 11 years ago
  49. 0afd840 Disabling a flaky expectation in vie_autotest_rtp_rtcp.cc. by stefan@webrtc.org · 11 years ago
  50. 34741c8 Include files from webrtc/.. paths in test/ by pbos@webrtc.org · 11 years ago
  51. 7f3f8bc Refactor jitter buffer to use separate lists for decodable and incomplete frames. by stefan@webrtc.org · 11 years ago
  52. ead3c6d Fix bugs in DesktopRegion::IntersectWith() and DesktopRect::IntersectWith(). by sergeyu@chromium.org · 11 years ago
  53. 8665da8 Remove dead testRateControl.cc by pbos@webrtc.org · 11 years ago
  54. a01f7f6 Removed dead testH263Parser.cc by pbos@webrtc.org · 11 years ago
  55. c1f0eb2 Remove dead bitstreamTest.cc. by pbos@webrtc.org · 11 years ago
  56. 28556f5 Make sure GlxRenderer frees its resources. by pbos@webrtc.org · 11 years ago
  57. c74c3c2 Adds integration test for RTX and fixes bugs found. by stefan@webrtc.org · 11 years ago
  58. 5c58f63 Fix regression where retransmission bitrate is no longer estimated. by stefan@webrtc.org · 11 years ago
  59. d445d22 CreateEmptyFrame casts from size_t to int. by pbos@webrtc.org · 11 years ago
  60. 9b30348 FrameGenerator class for future fake capture device. by pbos@webrtc.org · 11 years ago
  61. 771cdcb Control new VideoEngine tests with gflags. by pbos@webrtc.org · 11 years ago
  62. 191c596 Adds print out of incoming resolution. by henrike@webrtc.org · 11 years ago
  63. a7dc37d Log the type of recycled frames. by stefan@webrtc.org · 11 years ago
  64. 8c49c1e Log a message when a key frame packet is received by hclam@chromium.org · 11 years ago
  65. 46db413 Fix failing tests on 32 bit Linux. by solenberg@webrtc.org · 11 years ago
  66. e46c8d3 API to control target delay in NetEq jitter buffer. NetEq maintains the given delay unless channel conditions require a higher delay. by turaj@webrtc.org · 11 years ago
  67. 561990f - Changed RemoteBitrateEstimator::IncomingPacket() to include a const WebRtcRTPHeader& and remove ssrc, rtp_timestamp. by solenberg@webrtc.org · 11 years ago
  68. 6ec2507 Disable WindowCapturer tests on OSX and Linux by sergeyu@chromium.org · 11 years ago
  69. 6ebfd34 Add direct_dependent_settings in common.gypi. by sergeyu@chromium.org · 11 years ago
  70. 5f8f112 Not to request to TURN server for local tests. Follow-up work to issue1197. by braveyao@webrtc.org · 11 years ago
  71. 106afff Roll libvpx to 196669. -pick up libvpx roll to 9981006d by marpan@webrtc.org · 11 years ago
  72. 2eaf98b Refactor VCM/Timing. No changes in functionality. by mikhal@webrtc.org · 11 years ago
  73. 3417eb4 Consolidate GetFrame and InsertPacket and move NACK list processing to after a packet has been successfully inserted. by stefan@webrtc.org · 11 years ago
  74. 956aa7e Include files from webrtc/.. paths in voice_engine/ by pbos@webrtc.org · 11 years ago
  75. 8a025e2 Make sure VoiceEngine tests only include one test framework. by pbos@webrtc.org · 11 years ago
  76. d2541e8 Remove <iostream> usage from loopback.cc by pbos@webrtc.org · 11 years ago
  77. 375deb4 Suffix VcmCapturer's privates with underscore_ by pbos@webrtc.org · 11 years ago
  78. 0d540c3 Log timestamp of the frame when it's dropped from the render module by hclam@chromium.org · 11 years ago
  79. 69bb348 Log error in ViESender::SendRTCPPacket by hclam@chromium.org · 11 years ago
  80. ac0ef48 Revert 4067 "libyuv roll to r698 for Core Media fourccs for OSX ..." by andrew@webrtc.org · 11 years ago
  81. f9825e5 Revert 4000 "Reverting r3978" by andrew@webrtc.org · 11 years ago
  82. 225f2b8 Revert 4001 "Revert 3977" by andrew@webrtc.org · 11 years ago
  83. c0352d5 Fix assertions in rtp_header_extension.h caused by not handling the AudioLevel extension. Added unit tests to do basic checks of the AudioLevel extension. by solenberg@webrtc.org · 11 years ago
  84. e5794cb Recalibrate point sample expectation by fbarchard@google.com · 11 years ago
  85. a58d729 libyuv roll to r698 for Core Media fourccs for OSX camtwist support and performance improvements in ARGB scaler. by fbarchard@google.com · 11 years ago
  86. cb9cff0 Add functions to ViE API to enable/disable the absolute send time header extension. by solenberg@webrtc.org · 11 years ago
  87. b10ccbe Window capturer implementation for Windows. by sergeyu@chromium.org · 11 years ago
  88. 5e2a1bb AppRTC: make requestTurn() failure non-fatal to call establishment. by fischman@webrtc.org · 11 years ago
  89. 8d6eb56 Avoid NPE crash on Android platforms that don't support getting preview framerate. by fischman@webrtc.org · 11 years ago
  90. 5a602d7 Enable WebRTC demo application on x86 Android by fischman@webrtc.org · 11 years ago
  91. 2163212 Include gflags properly and X11 include order in VideoEngine. by pbos@webrtc.org · 11 years ago
  92. f5d4cb1 Include files from webrtc/.. paths in video_engine/ by pbos@webrtc.org · 11 years ago
  93. 9f557c1 Improve wraparound handling in the render time extrapolator. by stefan@webrtc.org · 11 years ago
  94. 14d7700 Moved command line parsing to internal tools and moved back the mic volume thingie. by phoglund@webrtc.org · 11 years ago
  95. e874a8f Enable WebRTC demo application on x86 Android by fischman@webrtc.org · 11 years ago
  96. 8630cfe Guarding certain operations, e.g. bandwidth estimation, RTCP statistics update etc., not to be run on sync RTPS. by turaj@webrtc.org · 11 years ago
  97. fe307e1 Add one unit test for NACKing a key frame by hclam@chromium.org · 11 years ago
  98. b3e5acf Cleanup traces in WebRTC by hclam@chromium.org · 11 years ago
  99. b9bb3d1 Avoid resetting encoder on identical settings. by pbos@webrtc.org · 11 years ago
  100. 890f609 Bugfix: VCM would report wrong sentBitrate by marpan@webrtc.org · 11 years ago