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gerrit-public.fairphone.software / fp2-dev / platform / external / chromium_org / third_party / webrtc / 0d8474dd557e512fd45e3612ab4b50aa32e5f7f9
  1. 0d8474d Remove metrics_unittests by kjellander@webrtc.org · 11 years ago
  2. ef1f6c3 Remove media_file from VideoEngine dependencies. by pbos@webrtc.org · 11 years ago
  3. 2a4595a cpplint cleaning new API and its implementation files. by mflodman@webrtc.org · 11 years ago
  4. b409d78 Reduced execution time for CallTest::ReceivesPliAndRecovers, by dropping only one packet and made it predictable by removing rand(). by mflodman@webrtc.org · 11 years ago
  5. f22f12a Speeding up CallTest.ReceivesAndRetransmitsNack and removed the random packet loss. by mflodman@webrtc.org · 11 years ago
  6. cc407fd Make MouseCursor mutable by sergeyu@chromium.org · 11 years ago
  7. 32a0f69 audio_processing_unittest: unbreak clang compilation. by fischman@webrtc.org · 11 years ago
  8. 6b89cba JNI Audio: remove dead members. by fischman@webrtc.org · 11 years ago
  9. 32705ce Revert "Make MouseCursor mutable" by sergeyu@chromium.org · 11 years ago
  10. 0c7efa2 Make MouseCursor mutable by sergeyu@chromium.org · 11 years ago
  11. 4db3691 Stop transport in test SuspendBelowMinBitrate. by pbos@webrtc.org · 11 years ago
  12. 6f43aa7 Added method for getting default module state and protect agains a by mflodman@webrtc.org · 11 years ago
  13. 620d9e5 Modify video_render/ to allow a single old frame. by pbos@webrtc.org · 11 years ago
  14. 4494516 Delete capturers after destroying streams in test. by pbos@webrtc.org · 11 years ago
  15. f3aed2f Revert 5285 "Revert 5228 "Use the RTT from RtcpRttStats class if..." by asapersson@webrtc.org · 11 years ago
  16. b06cca3 Simplification of histogram normalization in delay estimator. by bjornv@webrtc.org · 11 years ago
  17. 39139dc Revert r5294 to re-roll r5293. by pbos@webrtc.org · 11 years ago
  18. 2ec8a62 Adds robust validation functionality to the delay estimator by bjornv@webrtc.org · 11 years ago
  19. beb643b Incorrect iterator++ in ModuleRtpRtcpImpl::RegisterVideoBitrateObserver by sprang@webrtc.org · 11 years ago
  20. 0af1d21 Revert 5293 "Auto instantiate RBE depending on whether AST or TO..." by turaj@webrtc.org · 11 years ago
  21. ee867fa Auto instantiate RBE depending on whether AST or TOF is available in incoming packet stream. by solenberg@webrtc.org · 11 years ago
  22. b8dc2e2 Disabled tests on Android. The issue 2723 is filed to investigate the reason for tests failing. by turaj@webrtc.org · 11 years ago
  23. f34e39b Measure pacer queue size based on when packets are inserted rather than captured. by stefan@webrtc.org · 11 years ago
  24. b50a841 Fix jitter buffer delay estimate. by turaj@webrtc.org · 11 years ago
  25. 7f0519e Update talk to 58174641 together with http://review.webrtc.org/4319005/. by wu@webrtc.org · 11 years ago
  26. ab6ccbc Adding REMB to receive stream configuration, the send side will always by mflodman@webrtc.org · 11 years ago
  27. 9b3321f Revert 5228 "Use the RTT from RtcpRttStats class if provided whe..." by asapersson@webrtc.org · 11 years ago
  28. 9d9f138 Merge metrics_unittests into video_engine_tests. by pbos@webrtc.org · 11 years ago
  29. d1dd1d2 Move realtime tests to webrtc_perf_tests. by pbos@webrtc.org · 11 years ago
  30. 0e4512b Callback for send bitrate estimates - new roll by sprang@webrtc.org · 11 years ago
  31. e4d538a Make sure channels in the same call are in the same channel group. by mflodman@webrtc.org · 11 years ago
  32. e6dc4ff Making RemoteRateControl::min_configured_bit_rate_ configurable by henrik.lundin@webrtc.org · 11 years ago
  33. 3a4fc4b Update talk to 58127566 together with by wu@webrtc.org · 11 years ago
  34. b9cf1de ACM 2 compatibility with ACM 1. by turaj@webrtc.org · 11 years ago
  35. 9b3d2bf Revert 5274 "Update talk to 58113193 together with https://webrt..." by wu@webrtc.org · 11 years ago
  36. cde78d6 Update talk to 58113193 together with https://webrtc-codereview.appspot.com/5309005/. by wu@webrtc.org · 11 years ago
  37. a4670a1 Complete rewrite of demo application. by henrike@webrtc.org · 11 years ago
  38. 0ceb51f Remove overloaded CpuOveruseMeasure function. by asapersson@webrtc.org · 11 years ago
  39. 2f70422 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
  40. f49d16c Fix common_video_unittests in apk_tests.gyp. by pbos@webrtc.org · 11 years ago
  41. 7123a80 Add SwapFrame() to VideoSendStreamInput. by pbos@webrtc.org · 11 years ago
  42. 66e84b0 Revert 5259 "Callback for send bitrate estimates" by sprang@webrtc.org · 11 years ago
  43. 894dab9 Roll chromium_revision 232627:238260 by kjellander@webrtc.org · 11 years ago
  44. f1d22d4 Callback for send bitrate estimates by sprang@webrtc.org · 11 years ago
  45. 72b0d40 Removed unnecessary Pulse init from VoE startup. by fischman@webrtc.org · 11 years ago
  46. e8ca064 Correctly define OVERRIDE when building with g++ 4.7 and C++11 support by andrew@webrtc.org · 11 years ago
  47. 090f37f Improve VideoSendStreamTest::MaxPacketSize by sprang@webrtc.org · 11 years ago
  48. ba8b32c Change uses of the obsolete armv7 setting to arm_version==7. by kjellander@webrtc.org · 11 years ago
  49. 934be30 Fix compilation errors on Fedora 20. by fischman@webrtc.org · 11 years ago
  50. 4adc7ad Ensure WEBRTC_MODULE_UTILITY_VIDEO is undefined for enable_video==0. by andrew@webrtc.org · 11 years ago
  51. e681a01 Add shape in DesktopFrame. by sergeyu@chromium.org · 11 years ago
  52. e8dd108 Add new method to MockAudioProcessing. by andrew@webrtc.org · 11 years ago
  53. e4d591a Revert 5229 "Make VideoSendStreamTest::MaxPacketSize test a whol..." by andrew@webrtc.org · 11 years ago
  54. c8bd975 Allow opening an AEC dump from an existing file handle. by henrikg@webrtc.org · 11 years ago
  55. 9e40eba Stop video capturers in multi-stream test. by pbos@webrtc.org · 11 years ago
  56. 5b23ce6 Fix use of uninitialized memory in RtpSenderTest::StreamDataCountersCallbacks by sprang@webrtc.org · 11 years ago
  57. ed8c496 Fraction lost statistics not being reported by sprang@webrtc.org · 11 years ago
  58. 6dccf13 Make VideoSendStreamTest::MaxPacketSize test a whole range of frame sizes, to make sure all corner cases are covered. by sprang@webrtc.org · 11 years ago
  59. 2de68d6 Use the RTT from RtcpRttStats class if provided when sending/receiving NACK. by asapersson@webrtc.org · 11 years ago
  60. cf5c552 Add callbacks for send channel rtp statistics by sprang@webrtc.org · 11 years ago
  61. f0d9b20 Remove CallTest dependency on voice_engine/test/. by pbos@webrtc.org · 11 years ago
  62. 8db148e Add API to query video engine for the send-side delay. by stefan@webrtc.org · 11 years ago
  63. adc238a Fixing the android build by henrik.lundin@webrtc.org · 11 years ago
  64. 3d70641 Move implementation files out of the webrtc/ root. by pbos@webrtc.org · 11 years ago
  65. b669e60 Remove default implementations for SuspendBelowMinBitrate by henrik.lundin@webrtc.org · 11 years ago
  66. 3bcea52 Fix bug where fraction_lost is always set to 0 when getting received RTCP statistics. by stefan@webrtc.org · 11 years ago
  67. 8911937 Add callbacks for send channel rtcp statistics by sprang@webrtc.org · 11 years ago
  68. d90bebb Make RTPSender::SendPadData public. by stefan@webrtc.org · 11 years ago
  69. 991d58c Remove unused ThreadData struct. by andrew@webrtc.org · 11 years ago
  70. 5459e0b Remove the long disabled WEBRTC_SVNREVISION define. by andrew@webrtc.org · 11 years ago
  71. 382cfdd Removing DropDeltaAfterKey functionality which is unused. by andresp@webrtc.org · 11 years ago
  72. 9435a17 Add send frame rate statistics callback by sprang@webrtc.org · 11 years ago
  73. f2c136b Added a delay measurement, measures the time between an incoming captured frame until the frame is being processed. Measures the delay per second. by asapersson@webrtc.org · 11 years ago
  74. 0cb8020 Fixes a crash in fullstack tests introduced with r5209. by stefan@webrtc.org · 11 years ago
  75. 29a9669 Small fixes to plot_neteq_delay.m by henrik.lundin@webrtc.org · 11 years ago
  76. da3ae7c Adds support for sending redundant payloads over RTX. by stefan@webrtc.org · 11 years ago
  77. c101a27 Fix a typo in neteq.gypi by henrik.lundin@webrtc.org · 11 years ago
  78. c749348 Compile-out functions only used by the bit-exact test. by andrew@webrtc.org · 11 years ago
  79. b1f4a72 Don't HANDLE_EINTR(close). Use IGNORE_EINTR(close). by fischman@webrtc.org · 11 years ago
  80. adaa8b5 Add baseline generation/verification to BWE test framework. by solenberg@webrtc.org · 11 years ago
  81. 4ed6832 Utility class for reading/writing network-byte-ordered integers. by sprang@webrtc.org · 11 years ago
  82. 397aae0 Change BitrateStats to more generalized RateStatistics by sprang@webrtc.org · 11 years ago
  83. 309b2c8 Set local SSRC for VideoReceiveStream. by pbos@webrtc.org · 11 years ago
  84. f91a14f Do not use recursive calling in NetEq test tools by henrik.lundin@webrtc.org · 11 years ago
  85. 02817f8 Fixing NetEq tests for new Opus version by tina.legrand@webrtc.org · 11 years ago
  86. 169a27a Disable check for all sent SSRCs being valid. by pbos@webrtc.org · 11 years ago
  87. 758ef4c This CL adds an API to enable robust validation of delay estimates. by bjornv@webrtc.org · 11 years ago
  88. c8918cb 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
  89. 0e2571d Lock frame in ViECapturer::IncomingFrameI420. by pbos@webrtc.org · 11 years ago
  90. 9105cbd Set up SSRCs correctly after switching codec. by pbos@webrtc.org · 11 years ago
  91. b8b2a23 Recommit CL5184 by bjornv@webrtc.org · 11 years ago
  92. 151cd25 Refactor Remote Estimators Test into a more reusable form. by solenberg@webrtc.org · 11 years ago
  93. 66d634f Revert 5184 "Small refactoring change in delay_estimator." by bjornv@webrtc.org · 11 years ago
  94. 2c75d4e Small refactoring change in delay_estimator. by bjornv@webrtc.org · 11 years ago
  95. 801822c Ensure that no packet stays in the pacer queue for longer than 2 seconds. by stefan@webrtc.org · 11 years ago
  96. 8f9da30 Create default implementation to fix issue in libjingle by sprang@webrtc.org · 11 years ago
  97. 4a9843f Implement and test EncodedImageCallback in new ViE API. by sprang@webrtc.org · 11 years ago
  98. 2622be1 Added measure of encode time. Added encode time to the ViE CpuOveruseMeasure api. by asapersson@webrtc.org · 11 years ago
  99. 58b912b Remove const in vie_rtp_rtcp, where there is conflict with by sprang@webrtc.org · 11 years ago
  100. e7270f5 Faster implementation of BitRateStats. by mikhal@webrtc.org · 11 years ago