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gerrit-public.fairphone.software / fp2-dev / platform / external / chromium_org / third_party / webrtc / b3afc181a05250e034a48a821b6cb45bcb7fc5de
  1. b3afc18 Remove #pragma once by pbos@webrtc.org · 11 years ago
  2. 028ec72 Breaking out RTP header parsing from the RTP module. by stefan@webrtc.org · 11 years ago
  3. 266d5f6 Break video_engine/new_include/common.h into smaller parts. by pbos@webrtc.org · 11 years ago
  4. f378d7c Switch frame list implementation to std::map. by stefan@webrtc.org · 11 years ago
  5. 251c209 Rename voice_engine_core -> voice_engine and move targets to voice_engine.gyp. by andrew@webrtc.org · 11 years ago
  6. 9162bbc Add comment about test_packet_masks_metrics. by marpan@webrtc.org · 11 years ago
  7. 65a5f3f Updated WebRTC version to 3.32 TBR=wu@webrtc.org by elham@webrtc.org · 11 years ago
  8. 3bf9290 Don't return an estimated receive BW for channels not receiving video. by mflodman@webrtc.org · 11 years ago
  9. cab277d Include gflags with "gflags/gflags.h" instead of <> by pbos@webrtc.org · 11 years ago
  10. ef32e73 Include "gtest/gtest.h", not by full path, on WEBRTC_ANDROID_PLATFORM_BUILD by pbos@webrtc.org · 11 years ago
  11. d927a4d Improve vie_autotest_rtp_rtcp by reenabling important tests and reducing flakiness. by stefan@webrtc.org · 11 years ago
  12. 464f9be Include files from webrtc/.. paths in audio_conference_mixer/ by pbos@webrtc.org · 11 years ago
  13. 382c8b3 Include files from webrtc/.. paths in audio_processing/ by pbos@webrtc.org · 11 years ago
  14. a6ca12e Default constructors for new VideoEngine structs. by pbos@webrtc.org · 11 years ago
  15. 66b8717 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
  16. 8a3b04d - Created RemoteBitrateEstimator wrapper for use internally in (ViE) ChannelGroup. by solenberg@webrtc.org · 11 years ago
  17. 9bacbf4 Adding Mac test renderer, some test refactoring and made cpplint pass. by mflodman@webrtc.org · 11 years ago
  18. a960785 Include files from webrtc/.. paths in system_wrappers/ by pbos@webrtc.org · 11 years ago
  19. 6c9726a Include files from webrtc/.. paths in test/channel_transport/ by pbos@webrtc.org · 11 years ago
  20. 519d7cf Include files from webrtc/.. paths in video_processing/ by pbos@webrtc.org · 11 years ago
  21. d8f7f53 Include files from webrtc/.. paths in remote_bitrate_estimator/ by pbos@webrtc.org · 11 years ago
  22. abf0cd8 Include files from webrtc/.. paths in common_audio/ by pbos@webrtc.org · 11 years ago
  23. 104218e Disabling a flaky expectation in vie_autotest_rtp_rtcp.cc. by stefan@webrtc.org · 11 years ago
  24. 56041ab Include files from webrtc/.. paths in test/ by pbos@webrtc.org · 11 years ago
  25. 9e0d3ec Refactor jitter buffer to use separate lists for decodable and incomplete frames. by stefan@webrtc.org · 11 years ago
  26. 3f2091a Fix bugs in DesktopRegion::IntersectWith() and DesktopRect::IntersectWith(). by sergeyu@chromium.org · 11 years ago
  27. 087f8c6 Remove dead testRateControl.cc by pbos@webrtc.org · 11 years ago
  28. 2f30cee Removed dead testH263Parser.cc by pbos@webrtc.org · 11 years ago
  29. 2ea6127 Remove dead bitstreamTest.cc. by pbos@webrtc.org · 11 years ago
  30. 77fa22e Make sure GlxRenderer frees its resources. by pbos@webrtc.org · 11 years ago
  31. 1a37edd Adds integration test for RTX and fixes bugs found. by stefan@webrtc.org · 11 years ago
  32. 06721fc Fix regression where retransmission bitrate is no longer estimated. by stefan@webrtc.org · 11 years ago
  33. c0dba24 CreateEmptyFrame casts from size_t to int. by pbos@webrtc.org · 11 years ago
  34. d9f9185 FrameGenerator class for future fake capture device. by pbos@webrtc.org · 11 years ago
  35. 3d6a8bf Control new VideoEngine tests with gflags. by pbos@webrtc.org · 11 years ago
  36. 2660072 Adds print out of incoming resolution. by henrike@webrtc.org · 11 years ago
  37. 53e452d Log the type of recycled frames. by stefan@webrtc.org · 11 years ago
  38. 6bd2847 Log a message when a key frame packet is received by hclam@chromium.org · 11 years ago
  39. 1286255 Fix failing tests on 32 bit Linux. by solenberg@webrtc.org · 11 years ago
  40. 6de406d 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
  41. d822fe4 - Changed RemoteBitrateEstimator::IncomingPacket() to include a const WebRtcRTPHeader& and remove ssrc, rtp_timestamp. by solenberg@webrtc.org · 11 years ago
  42. d921161 Disable WindowCapturer tests on OSX and Linux by sergeyu@chromium.org · 11 years ago
  43. e52fbdd Add direct_dependent_settings in common.gypi. by sergeyu@chromium.org · 11 years ago
  44. 2e87970 Refactor VCM/Timing. No changes in functionality. by mikhal@webrtc.org · 11 years ago
  45. 1da8866 Consolidate GetFrame and InsertPacket and move NACK list processing to after a packet has been successfully inserted. by stefan@webrtc.org · 11 years ago
  46. be72fe4 Include files from webrtc/.. paths in voice_engine/ by pbos@webrtc.org · 11 years ago
  47. c9a4463 Make sure VoiceEngine tests only include one test framework. by pbos@webrtc.org · 11 years ago
  48. 08dbe39 Remove <iostream> usage from loopback.cc by pbos@webrtc.org · 11 years ago
  49. 9a15fc3 Suffix VcmCapturer's privates with underscore_ by pbos@webrtc.org · 11 years ago
  50. 4602e44 Log timestamp of the frame when it's dropped from the render module by hclam@chromium.org · 11 years ago
  51. e37160f Log error in ViESender::SendRTCPPacket by hclam@chromium.org · 11 years ago
  52. 5f600c8 Revert 4000 "Reverting r3978" by andrew@webrtc.org · 11 years ago
  53. 04958f7 Revert 4001 "Revert 3977" by andrew@webrtc.org · 11 years ago
  54. 3ec8ef6 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
  55. 02175b6 Recalibrate point sample expectation by fbarchard@google.com · 11 years ago
  56. 967320b Add functions to ViE API to enable/disable the absolute send time header extension. by solenberg@webrtc.org · 11 years ago
  57. 091158d Window capturer implementation for Windows. by sergeyu@chromium.org · 11 years ago
  58. 0313a3a Avoid NPE crash on Android platforms that don't support getting preview framerate. by fischman@webrtc.org · 11 years ago
  59. ecd1591 Include gflags properly and X11 include order in VideoEngine. by pbos@webrtc.org · 11 years ago
  60. f2e6fb3 Include files from webrtc/.. paths in video_engine/ by pbos@webrtc.org · 11 years ago
  61. 329c951 Improve wraparound handling in the render time extrapolator. by stefan@webrtc.org · 11 years ago
  62. 3f3dcd1 Moved command line parsing to internal tools and moved back the mic volume thingie. by phoglund@webrtc.org · 11 years ago
  63. db9d0be Enable WebRTC demo application on x86 Android by fischman@webrtc.org · 11 years ago
  64. db298d5 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
  65. 49ba1dc Add one unit test for NACKing a key frame by hclam@chromium.org · 11 years ago
  66. fc8382b Cleanup traces in WebRTC by hclam@chromium.org · 11 years ago
  67. 5ff68ae Avoid resetting encoder on identical settings. by pbos@webrtc.org · 11 years ago
  68. b421849 Bugfix: VCM would report wrong sentBitrate by marpan@webrtc.org · 11 years ago
  69. 4df4e2c Formatted FEC stuff. by phoglund@webrtc.org · 11 years ago
  70. 1d76489 Moved force_volume_max to its own gyp file to avoid a circular dependency. by phoglund@webrtc.org · 11 years ago
  71. f20975f Wrote a small portable tool for forcing the mic volume to 100%. by phoglund@webrtc.org · 11 years ago
  72. dc8c883 New VideoEngine API implementation on top of old one, first steps. by pbos@webrtc.org · 11 years ago
  73. 3932563 Log too long non-decodable duration events. by stefan@webrtc.org · 11 years ago
  74. 864f9d7 Remove SetOverUseDetectorOptions and cleaned ViESharedData. by mflodman@webrtc.org · 11 years ago
  75. 1e77b3b Add handling of the absolute send time header extension to the rtp_rtcp module. by solenberg@webrtc.org · 11 years ago
  76. ba4ccdd Updating NACK RTX test by mikhal@webrtc.org · 11 years ago
  77. dfdfaf5 VCM/JB: Bug fix in ExtractAndSetDecode BUG=1771 R=stefan@webrtc.org by mikhal@webrtc.org · 11 years ago
  78. 8197221 RemoteBitrateEstimatorTest::TestRateIncreaseReordering sent in arrival timestamps in non monotonically increasing order. Fixed. by solenberg@webrtc.org · 11 years ago
  79. e07cbc5 CoreAudio Win: release resources safely under certain rare circumstances. by braveyao@webrtc.org · 11 years ago
  80. 5fa31f7 Linux support for typing detection by niklas.enbom@webrtc.org · 11 years ago
  81. 356329b Address sanitizer out of bounds read in iSAC by turaj@webrtc.org · 11 years ago
  82. 1a25618 Remove const for plain data types in common_video/ by pbos@webrtc.org · 11 years ago
  83. 2b2e78c Adding a factory to remote bitrate estimator and allow it to be set via config. by andresp@webrtc.org · 11 years ago
  84. 1aecacb Fixes a bug where the render buffer size (and indirectly the non-continuous duration) was computed incorrectly. by stefan@webrtc.org · 11 years ago
  85. 28a9f65 Removed Mac capture crash and memory leak. by mflodman@webrtc.org · 11 years ago
  86. 6bbd8b1 Add script for comparing video quality by kjellander@webrtc.org · 11 years ago
  87. b2a298c Reformatted FEC tables. by phoglund@webrtc.org · 11 years ago
  88. 60003b2 Remove const for plain data types in common_audio/ by pbos@webrtc.org · 11 years ago
  89. 07a1c11 Remove const for plain data types in voice_engine/ by pbos@webrtc.org · 11 years ago
  90. 1e8424f Replace ExtraCodecOptions with new Config class that supports multiple settings at once. by andresp@webrtc.org · 11 years ago
  91. 4981e61 Fix typo in log statement. witdh should be width. by fbarchard@google.com · 11 years ago
  92. 302f731 Add more tracing for key frames. by justinlin@chromium.org · 11 years ago
  93. f308b75 Increased the limit for KViEMaxCaptureDevices from 10 to 256. See issue 1343. by vikasmarwaha@webrtc.org · 11 years ago
  94. 34d0fec Updated WebRTC version to 3.31 TBR=wu@webrtc.org by elham@webrtc.org · 11 years ago
  95. aedb73b Revert 4008 "Avoid resetting video encoder for similar configs." by phoglund@webrtc.org · 11 years ago
  96. 196ed2e Disabled flaky codec test (RunsCodecTestWithoutErrors) by phoglund@webrtc.org · 11 years ago
  97. 16dfb75 Avoid resetting video encoder for similar configs. by pbos@webrtc.org · 11 years ago
  98. ad2b368 Wiring down config from video engine until video coding and remote bitrate estimator modules instantiation. by andresp@webrtc.org · 11 years ago
  99. af6696e Remove TEXT(x) for BUILDINFO macros. by pbos@webrtc.org · 11 years ago
  100. 93219bb Added a config class to ease passing a set of options across webrtc. by andresp@webrtc.org · 11 years ago