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gerrit-public.fairphone.software / fp2-dev / platform / external / chromium_org / third_party / webrtc / f47d0f86cab3e8f1df029cfadf549e986cbff20e
  1. f47d0f8 Remove unneeded *_NOT_SUPPORTED from VoEAudioProcessing. by andrew@webrtc.org · 11 years ago
  2. 7533659 Fixes some pacer/padding issues found while testing. by stefan@webrtc.org · 11 years ago
  3. c9cb798 Use 3 threads for higher than 720p resolutions by fbarchard@google.com · 11 years ago
  4. 5b9adb0 Add a log message to see video delay break down by hclam@chromium.org · 11 years ago
  5. b82ee51 Make ScreenCapturerMac work in versions of OSX before Lion. by sergeyu@chromium.org · 11 years ago
  6. 9ca71b1 Enable ScreenCapturer unittests by sergeyu@chromium.org · 11 years ago
  7. ba458e2 Use intptr_t to represent window IDs on all platforms. by sergeyu@chromium.org · 11 years ago
  8. 69f7605 Wire up pacer-based padding. by stefan@webrtc.org · 11 years ago
  9. 67ca2b4 Revert r4145 "Revert 4127 "Switch frame list implementation to std::map."" by stefan@webrtc.org · 11 years ago
  10. 376ae3e Revert r4146 "Revert 4104 "Refactor jitter buffer to use separate lists for de..."" by stefan@webrtc.org · 11 years ago
  11. b06dd93 Fix AV sync issue by hclam@chromium.org · 11 years ago
  12. 9540e2a Log current and target AV delay in ViESyncModule by hclam@chromium.org · 11 years ago
  13. 8dbc8ab Merge more tests into modules_{unit,integration}tests. by kjellander@webrtc.org · 11 years ago
  14. 6828566 WebRTCDemo: ensures that using front and back camera work as expected. by henrike@webrtc.org · 11 years ago
  15. 19b339a Fixes linker issue with no op trace. by henrike@webrtc.org · 11 years ago
  16. 7b2c430 Risk of division by zero. by turaj@webrtc.org · 11 years ago
  17. cefb004 Revert 4211 "Build all java files into jar for each module on An..." by fischman@webrtc.org · 11 years ago
  18. 8ed5369 Disable ViEExtendedIntegrationTest.RunsCodecTestWithoutErrors test. by kjellander@webrtc.org · 11 years ago
  19. 83163e0 Disable ViEExtendedIntegrationTest.RunsCodecTestWithoutErrors test. by kjellander@webrtc.org · 11 years ago
  20. 789e98b Fix breakage due to test_fec conversion to gtest. by kjellander@webrtc.org · 11 years ago
  21. f04f54a Convert test_fec to gtest by kjellander@webrtc.org · 11 years ago
  22. f09f7b2 Disable ViEStandardIntegrationTest.RunsRtpRtcpTestWithoutErrors test. by kjellander@webrtc.org · 11 years ago
  23. 6ab2b1f G722_1/G722_1C codecs won't instantiate by tina.legrand@webrtc.org · 11 years ago
  24. 46cec2a Reorganize test targets in WebRTC by kjellander@webrtc.org · 11 years ago
  25. 0604490 Build all java files into jar for each module on Android by fischman@webrtc.org · 11 years ago
  26. e8c9ecd Allow the screen capturer to capture oversized cursors and cursors without alpha channel (Windows). by alexeypa@chromium.org · 11 years ago
  27. a463a4b Landing binary cursor image files to be used in a follow up CL. by alexeypa@chromium.org · 11 years ago
  28. f1bcae0 Updated WebRTC version to 3.33 by elham@webrtc.org · 11 years ago
  29. b4c89a4 Making no NACK mode work again in VideoEngine. by mflodman@webrtc.org · 11 years ago
  30. 63988b2 RW lock access to ssrc maps in VideoCall. by pbos@webrtc.org · 11 years ago
  31. 203d656 Add back the WEBRTC_DIRECT_TRACE flag. by solenberg@webrtc.org · 11 years ago
  32. 0a86a9f AudioDeviceAndroidOpenSLES: NULL variables might be referenced in StopPlayout() by braveyao@webrtc.org · 11 years ago
  33. 1125d89 Revert some variables to uint32_t to fix compile errors on Mac gcc. by andrew@webrtc.org · 11 years ago
  34. 64e2651 Allow audio devices with up to 64 channels on Mac. by andrew@webrtc.org · 11 years ago
  35. 0e7cd85 Fixed Rtp/Rtcp tests by pwestin@webrtc.org · 11 years ago
  36. cfe2a74 Fix relative path to .gitignore and other minor changes. by andrew@webrtc.org · 11 years ago
  37. a18c6e5 Removing functionality for inserting pre-encoded frames instead of raw by mflodman@webrtc.org · 11 years ago
  38. 029c3f4 Add script for appending entries to .gitignore. by andrew@webrtc.org · 11 years ago
  39. 9aeef32 Fix size_t to int conversion error on Win64. by andrew@webrtc.org · 11 years ago
  40. a653113 Remove fake screen capturer because it's not used anywhere. by sergeyu@chromium.org · 11 years ago
  41. 5f545ff Fix for STL vector function data not available. by pwestin@webrtc.org · 11 years ago
  42. 4aa9f1a Connect ACM with RTP module for audio NACK. by pwestin@webrtc.org · 11 years ago
  43. d0631e3 Nack for audio. by turaj@webrtc.org · 11 years ago
  44. 31fad0b Fix leaks in DesktopRegion by sergeyu@chromium.org · 11 years ago
  45. 69bab25 Implement DetectNumberOfCores on Android and make it consistent on Linux and Android by fischman@webrtc.org · 11 years ago
  46. b8171ff Wire up Nack for Voe by pwestin@webrtc.org · 11 years ago
  47. 6fb2ca3 Fix init list for VideoSendStream::Config::Rtp. by pbos@webrtc.org · 11 years ago
  48. 6f1c3ef Stats+Config moved into VideoSend/ReceiveStreams. by pbos@webrtc.org · 11 years ago
  49. 53304e8 Merge webrtc_utility_unittests into modules_unittests. by kjellander@webrtc.org · 11 years ago
  50. 266fc69 Restore relative include paths to libyuv. by andrew@webrtc.org · 11 years ago
  51. 9b82368 Issue 1847, memcopy is wrong and unnecessary, it is sufficient to store the pointer before clearing the instance, and write back the pointer. by turaj@webrtc.org · 11 years ago
  52. ae05178 resolve b9050210. Avoid pushing sync packet before any packet received. Do not turn on AV-sync if initial delay is zero. by turaj@webrtc.org · 11 years ago
  53. 6c82a7e Move screen capturers from chromium to webrtc. by sergeyu@chromium.org · 11 years ago
  54. 12bce3b Refactor padding and rtp header functionality. by stefan@webrtc.org · 11 years ago
  55. d8ecee5 Update the remote bitrate estimator before passing the packet to the RTP module. by stefan@webrtc.org · 11 years ago
  56. e54928f Remove XvRenderer. by pbos@webrtc.org · 11 years ago
  57. 751253d Fix build error introduced with r4168. by stefan@webrtc.org · 11 years ago
  58. 695ff2a Add support for padding in pacer. by stefan@webrtc.org · 11 years ago
  59. 026d1ce Include files from webrtc/.. paths in common_video/ by pbos@webrtc.org · 11 years ago
  60. cff5c03 Include files from webrtc/.. paths in tools/ by pbos@webrtc.org · 11 years ago
  61. 1cc4ed7 Disable neteq_unittests on Win x64 in code. by kjellander@webrtc.org · 11 years ago
  62. ec5caf3 Disable audio_decoder_unittests on Win x64 in code. by kjellander@webrtc.org · 11 years ago
  63. f4fc8ba Disable audio_coding_unittests on Win x64 in code. by kjellander@webrtc.org · 11 years ago
  64. 3b35ec6 Do not hold a lock when calling VCMReceiveCallback::FrameToRender. by fischman@webrtc.org · 11 years ago
  65. aec1bc8 Optimized DesktopRegion implementation. by sergeyu@chromium.org · 11 years ago
  66. ad9ee0d Removed unused class members to enable clang=1 android build. by fischman@webrtc.org · 11 years ago
  67. 0016110 Setting SSRC in vie_loopback_test by mikhal@webrtc.org · 11 years ago
  68. 915ca75 Fix error in mixing test for supported sample rates. by andrew@webrtc.org · 11 years ago
  69. a80d94b Change SetRTPAudioLevelIndicationStatus to ignore the id in the case of disabling. by wu@webrtc.org · 11 years ago
  70. 92bfbbd Replace the old resampler with SincResampler in the voice engine signal path. by andrew@webrtc.org · 11 years ago
  71. 379dce7 Remove ancient and unused CNG test. by andrew@webrtc.org · 11 years ago
  72. 2753b76 Add dummy audio NACK APIs by niklas.enbom@webrtc.org · 11 years ago
  73. 40954f0 Prevent excessive logging in jitter buffer by hclam@chromium.org · 11 years ago
  74. 8bf7456 Revert 4104 "Refactor jitter buffer to use separate lists for de..." by tnakamura@webrtc.org · 11 years ago
  75. 884ff69 Revert 4127 "Switch frame list implementation to std::map." by tnakamura@webrtc.org · 11 years ago
  76. 1aa406c MIPS optimizations for the following functions: by andrew@webrtc.org · 11 years ago
  77. e477574 VCM/Timing: Setting clear names to members & methods by mikhal@webrtc.org · 11 years ago
  78. 9f62516 Fixes the frameRate stats by grouping the frames by timestamp. by jiayl@webrtc.org · 11 years ago
  79. e3e4615 Use int for FPS instead of size_t. by pbos@webrtc.org · 11 years ago
  80. cbd78ae Include files from webrtc/.. paths in rtp_rtcp/ by pbos@webrtc.org · 11 years ago
  81. 54b6ebc Correctly set SSRCs for extra send RTP modules. by stefan@webrtc.org · 11 years ago
  82. 23e3f44 Remove assert for aborting FrameGeneratorCapturer. by pbos@webrtc.org · 11 years ago
  83. c1506a2 Fake VideoCapturer based on FrameGenerator by pbos@webrtc.org · 11 years ago
  84. 4e5f983 Fix a return value mismatch introduced in r4129. by stefan@webrtc.org · 11 years ago
  85. 50a4d9f Remove #pragma once by pbos@webrtc.org · 11 years ago
  86. 6696fba Breaking out RTP header parsing from the RTP module. by stefan@webrtc.org · 11 years ago
  87. 4988d94 Break video_engine/new_include/common.h into smaller parts. by pbos@webrtc.org · 11 years ago
  88. afe587e Switch frame list implementation to std::map. by stefan@webrtc.org · 11 years ago
  89. 5221d1c Rename voice_engine_core -> voice_engine and move targets to voice_engine.gyp. by andrew@webrtc.org · 11 years ago
  90. 2dcf742 Add comment about test_packet_masks_metrics. by marpan@webrtc.org · 11 years ago
  91. 3990df2 Updated WebRTC version to 3.32 TBR=wu@webrtc.org by elham@webrtc.org · 11 years ago
  92. e3b52e6 Don't return an estimated receive BW for channels not receiving video. by mflodman@webrtc.org · 11 years ago
  93. bb6bef5 Include gflags with "gflags/gflags.h" instead of <> by pbos@webrtc.org · 11 years ago
  94. 8838f68 Include "gtest/gtest.h", not by full path, on WEBRTC_ANDROID_PLATFORM_BUILD by pbos@webrtc.org · 11 years ago
  95. 39784c4 Improve vie_autotest_rtp_rtcp by reenabling important tests and reducing flakiness. by stefan@webrtc.org · 11 years ago
  96. 0c836bf Include files from webrtc/.. paths in audio_conference_mixer/ by pbos@webrtc.org · 11 years ago
  97. 9fb1613 Include files from webrtc/.. paths in audio_processing/ by pbos@webrtc.org · 11 years ago
  98. b2d1a40 Default constructors for new VideoEngine structs. by pbos@webrtc.org · 11 years ago
  99. 5437a2c 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
  100. f40e9b6 - Created RemoteBitrateEstimator wrapper for use internally in (ViE) ChannelGroup. by solenberg@webrtc.org · 11 years ago