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gerrit-public.fairphone.software / fp2-dev / platform / external / chromium_org / third_party / webrtc / c9faf10f99da606e2d58b0b4a79c03c232b7c50f / modules
  1. c9faf10 Add thread_annotations for clang targets. by andresp@webrtc.org · 11 years ago
  2. 4f1f5fa Add callbacks for receive channel RTCP statistics. by sprang@webrtc.org · 11 years ago
  3. 9edcdb0 Refactoring MediaOptimization so it can easily be turned into a thread-safe class. by andresp@webrtc.org · 11 years ago
  4. eb9ce11 Make MouseCursor mutable by sergeyu@chromium.org · 11 years ago
  5. 8d14e06 audio_processing_unittest: unbreak clang compilation. by fischman@webrtc.org · 11 years ago
  6. d138166 JNI Audio: remove dead members. by fischman@webrtc.org · 11 years ago
  7. 432e574 Revert "Make MouseCursor mutable" by sergeyu@chromium.org · 11 years ago
  8. 865be14 Make MouseCursor mutable by sergeyu@chromium.org · 11 years ago
  9. 202d38d Added method for getting default module state and protect agains a by mflodman@webrtc.org · 11 years ago
  10. d60137f Modify video_render/ to allow a single old frame. by pbos@webrtc.org · 11 years ago
  11. 5041831 Revert 5285 "Revert 5228 "Use the RTT from RtcpRttStats class if..." by asapersson@webrtc.org · 11 years ago
  12. 0443f6c Simplification of histogram normalization in delay estimator. by bjornv@webrtc.org · 11 years ago
  13. 46f7288 Revert r5294 to re-roll r5293. by pbos@webrtc.org · 11 years ago
  14. 1b3b8cb Adds robust validation functionality to the delay estimator by bjornv@webrtc.org · 11 years ago
  15. 1eb1008 Incorrect iterator++ in ModuleRtpRtcpImpl::RegisterVideoBitrateObserver by sprang@webrtc.org · 11 years ago
  16. c5a5713 Revert 5293 "Auto instantiate RBE depending on whether AST or TO..." by turaj@webrtc.org · 11 years ago
  17. 3bbc91e Auto instantiate RBE depending on whether AST or TOF is available in incoming packet stream. by solenberg@webrtc.org · 11 years ago
  18. afceaca Measure pacer queue size based on when packets are inserted rather than captured. by stefan@webrtc.org · 11 years ago
  19. 79d6daf Update talk to 58174641 together with http://review.webrtc.org/4319005/. by wu@webrtc.org · 11 years ago
  20. 1465cef Revert 5228 "Use the RTT from RtcpRttStats class if provided whe..." by asapersson@webrtc.org · 11 years ago
  21. b70db6d Callback for send bitrate estimates - new roll by sprang@webrtc.org · 11 years ago
  22. c49a3fa Making RemoteRateControl::min_configured_bit_rate_ configurable by henrik.lundin@webrtc.org · 11 years ago
  23. da4d59e ACM 2 compatibility with ACM 1. by turaj@webrtc.org · 11 years ago
  24. 27f0841 Enables mixing and matching Java and native audio. It is used for getting best of both worlds capabilities (AEC and low latency). by henrike@webrtc.org · 11 years ago
  25. c33d37c Add SwapFrame() to VideoSendStreamInput. by pbos@webrtc.org · 11 years ago
  26. ffea4ce Revert 5259 "Callback for send bitrate estimates" by sprang@webrtc.org · 11 years ago
  27. ca63ad9 Roll chromium_revision 232627:238260 by kjellander@webrtc.org · 11 years ago
  28. 1430bc3 Callback for send bitrate estimates by sprang@webrtc.org · 11 years ago
  29. b46e68d Removed unnecessary Pulse init from VoE startup. by fischman@webrtc.org · 11 years ago
  30. 2cafda4 Change uses of the obsolete armv7 setting to arm_version==7. by kjellander@webrtc.org · 11 years ago
  31. 5424c16 Fix compilation errors on Fedora 20. by fischman@webrtc.org · 11 years ago
  32. 241103f Ensure WEBRTC_MODULE_UTILITY_VIDEO is undefined for enable_video==0. by andrew@webrtc.org · 11 years ago
  33. 935c8c7 Add shape in DesktopFrame. by sergeyu@chromium.org · 11 years ago
  34. 8beba83 Add new method to MockAudioProcessing. by andrew@webrtc.org · 11 years ago
  35. 7b72264 Allow opening an AEC dump from an existing file handle. by henrikg@webrtc.org · 11 years ago
  36. 539670c Fix use of uninitialized memory in RtpSenderTest::StreamDataCountersCallbacks by sprang@webrtc.org · 11 years ago
  37. 0b16527 Use the RTT from RtcpRttStats class if provided when sending/receiving NACK. by asapersson@webrtc.org · 11 years ago
  38. b113981 Add callbacks for send channel rtp statistics by sprang@webrtc.org · 11 years ago
  39. ee234be Add API to query video engine for the send-side delay. by stefan@webrtc.org · 11 years ago
  40. 9b30fd3 Add callbacks for send channel rtcp statistics by sprang@webrtc.org · 11 years ago
  41. f03505e Make RTPSender::SendPadData public. by stefan@webrtc.org · 11 years ago
  42. 6d1a71b Remove unused ThreadData struct. by andrew@webrtc.org · 11 years ago
  43. 5fdd10a Add send frame rate statistics callback by sprang@webrtc.org · 11 years ago
  44. c0d6b2d Fixes a crash in fullstack tests introduced with r5209. by stefan@webrtc.org · 11 years ago
  45. a5be230 Small fixes to plot_neteq_delay.m by henrik.lundin@webrtc.org · 11 years ago
  46. 47f0c41 Adds support for sending redundant payloads over RTX. by stefan@webrtc.org · 11 years ago
  47. 76c6ac4 Fix a typo in neteq.gypi by henrik.lundin@webrtc.org · 11 years ago
  48. 3f0b77f Compile-out functions only used by the bit-exact test. by andrew@webrtc.org · 11 years ago
  49. 38aa817 Don't HANDLE_EINTR(close). Use IGNORE_EINTR(close). by fischman@webrtc.org · 11 years ago
  50. af00735 Add baseline generation/verification to BWE test framework. by solenberg@webrtc.org · 11 years ago
  51. 09299b0 Utility class for reading/writing network-byte-ordered integers. by sprang@webrtc.org · 11 years ago
  52. 7374da3 Change BitrateStats to more generalized RateStatistics by sprang@webrtc.org · 11 years ago
  53. cc19dce Do not use recursive calling in NetEq test tools by henrik.lundin@webrtc.org · 11 years ago
  54. db085fa Fixing NetEq tests for new Opus version by tina.legrand@webrtc.org · 11 years ago
  55. cd8c2b1 This CL adds an API to enable robust validation of delay estimates. by bjornv@webrtc.org · 11 years ago
  56. 1003b7d Fixes a crash in the pacer where it fails to find a normal prio packet if there are no high prio packets, given that the queue has grown too large. by stefan@webrtc.org · 11 years ago
  57. c3cce18 Recommit CL5184 by bjornv@webrtc.org · 11 years ago
  58. cda3cf3 Refactor Remote Estimators Test into a more reusable form. by solenberg@webrtc.org · 11 years ago
  59. 85ef326 Revert 5184 "Small refactoring change in delay_estimator." by bjornv@webrtc.org · 11 years ago
  60. cd96113 Small refactoring change in delay_estimator. by bjornv@webrtc.org · 11 years ago
  61. 04d6593 Ensure that no packet stays in the pacer queue for longer than 2 seconds. by stefan@webrtc.org · 11 years ago
  62. 2e98d45 Implement and test EncodedImageCallback in new ViE API. by sprang@webrtc.org · 11 years ago
  63. 66fba2b Faster implementation of BitRateStats. by mikhal@webrtc.org · 11 years ago
  64. b1d7931 Add include stdlib.h to files using abs. by stefan@webrtc.org · 11 years ago
  65. 98d217d Clear empty video frames in unittest so DrMemory will allow them to be read without an uninitialized read error. by fbarchard@google.com · 11 years ago
  66. 01a09ac Don't register iSAC-swb and iSAC-fb in NetEqDecodingTest. by turaj@webrtc.org · 11 years ago
  67. c5a2831 Fix MouseCursor to MouseCursorShape conversion in ScreenCapturerWin. by sergeyu@chromium.org · 11 years ago
  68. 936fd5f Fix issues with sequence number wrap-around in jitter statistics. by turaj@webrtc.org · 11 years ago
  69. f1fccf7 Distinguish instances of ACM1 from ACM2 by a version string. This is fpr testing purposes and will be removed when the experiment is done and ACM1 is fade out. by turaj@webrtc.org · 11 years ago
  70. 50293f5 Replace VideoFrameI420 with I420VideoFrame. by pbos@webrtc.org · 11 years ago
  71. 84d69d4 Don't reset the AEC filter in extended mode. by andrew@webrtc.org · 11 years ago
  72. c41f11b Increase size of pacer window to 500 ms as that better matches the encoder. by stefan@webrtc.org · 11 years ago
  73. c3f827c Lock access to ModuleRtpRtcpImpl::simulcast_. by pbos@webrtc.org · 11 years ago
  74. 31bd97d Fix issues with sequence number wrap-around in jitter statistics by henrik.lundin@webrtc.org · 11 years ago
  75. 3dc7ff3 Added API for enabling/disabling RTCP Receiver Reference Time extension. by asapersson@webrtc.org · 11 years ago
  76. 7950b98 Fixes a crash in VoE when unregistering JNI hooks. by henrike@webrtc.org · 11 years ago
  77. c4af4cf Add possibility to get the last processed RTT from the call stats class (to be used by RTP/RTCP module). by asapersson@webrtc.org · 11 years ago
  78. 09b40ec Add experimental noise suppression dummy API. by aluebs@webrtc.org · 11 years ago
  79. 8b0791c Fix DesktopAndCursorComposer to restore frames to the original state. by sergeyu@chromium.org · 11 years ago
  80. eb45a20 Adding back main() to the test. Now it is possible to choose between ACM1 and ACM2, furthermore, the test can simulate a channel with packet loss and FEC can be activated. Packet loss pattern is based on channel implementation in Channel{.cc,.h}, which currently is a determenistic pattern with 1 every 3rd packet is discarded. by turaj@webrtc.org · 11 years ago
  81. 4590177 Rename AutoMute to SuspendBelowMinBitrate by henrik.lundin@webrtc.org · 11 years ago
  82. e028410 Hook up audio/video sync to Call. by stefan@webrtc.org · 11 years ago
  83. 69b1aa4 Increment RTP timestamps for padding packets by henrik.lundin@webrtc.org · 11 years ago
  84. b748c9d Fix for RTX in combination with pacing. by stefan@webrtc.org · 11 years ago
  85. 591be3b Reimplementing NetEq4's AudioVector by henrik.lundin@webrtc.org · 11 years ago
  86. f4fbef3 Parse next RTCP XR report block after an unsupported block type. by asapersson@webrtc.org · 11 years ago
  87. 26f5492 Reducing opus_test runtime to pass Android test by minyue@webrtc.org · 11 years ago
  88. ff4fc2b MIPS optimizations for AECM audio processing module by andrew@webrtc.org · 11 years ago
  89. 731a87b Move audio_processing dependencies to a variable. by andrew@webrtc.org · 11 years ago
  90. 9965e3a Remove ".." from include_dirs in build/common. by pbos@webrtc.org · 11 years ago
  91. 3051ff7 Remove unnecessary include_dirs from audio_processing. by andrew@webrtc.org · 11 years ago
  92. 7e97e4c Fix for making sure that the packet in order checks are done prior to updating the last received packet state. by stefan@webrtc.org · 11 years ago
  93. b4d7835 Fix for video_processor_intergration_tests to run in parallel. by marpan@webrtc.org · 11 years ago
  94. 93bf70f Add missing dependencies to .isolate files by kjellander@webrtc.org · 11 years ago
  95. 690a03c Fix broken build on x86 Android by fischman@webrtc.org · 11 years ago
  96. 1bd9a7b Removed unused code. by asapersson@webrtc.org · 11 years ago
  97. af92d3e Revert "Remove extra copy in VideoCaptureImpl::IncomingFrameI420" by sheu@chromium.org · 11 years ago
  98. a191cb0 Remove extra copy in VideoCaptureImpl::IncomingFrameI420 by sheu@chromium.org · 11 years ago
  99. 6baaf30 Revert "Remove extra copy in VideoCaptureImpl::IncomingFrameI420" by sheu@chromium.org · 11 years ago
  100. 7773eec Remove extra copy in VideoCaptureImpl::IncomingFrameI420 by sheu@chromium.org · 11 years ago