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gerrit-public.fairphone.software / fp2-dev / platform / external / chromium_org / third_party / webrtc / ed019363064d0105f529cbb63a72bc2acdc3fb9f / video
  1. 4068313 Add simulation of network effects to video_loopback tool. by stefan@webrtc.org · 10 years ago
  2. 4a1b3e3 Make sure padding is sent on the first sending RTP module. by mflodman@webrtc.org · 10 years ago
  3. a4dc1ae Fix flaky ramp-up test. by stefan@webrtc.org · 10 years ago
  4. f0a119f Check before send/receive rtp header extensions. by pbos@webrtc.org · 10 years ago
  5. 442dbd4 Make RTCP sender report send media bytes. by pbos@webrtc.org · 10 years ago
  6. 55b0f2e Move pacer to fully use webrtc::Clock instead of webrtc::TickTime. by stefan@webrtc.org · 10 years ago
  7. 09da1a7 Remove the send-side cname getter APIs from voice and video engine. by stefan@webrtc.org · 10 years ago
  8. 51c9def Skip encoding in fake VP8 encoder. by pbos@webrtc.org · 10 years ago
  9. 54f889f Support VP8 encoder settings in VideoSendStream. by pbos@webrtc.org · 10 years ago
  10. 79b66f4 Add full stack test cases with a fake network pipe. by stefan@webrtc.org · 10 years ago
  11. 49927ed Refactor ramp-up tests to have separate help files for the test classes, to make things more reusable. by stefan@webrtc.org · 10 years ago
  12. 6aae61c Some refactoring inside rtp_rtcp/. by pbos@webrtc.org · 10 years ago
  13. f8be3d2 Extract RTP-header SSRC inline in Call. by pbos@webrtc.org · 10 years ago
  14. 2c47316 Add test for VideoEncoder setup/teardown. by pbos@webrtc.org · 10 years ago
  15. 2fd91bd Preserve RTP states for restarted VideoSendStreams. by pbos@webrtc.org · 10 years ago
  16. 38ac032 Fix data races related with traces in bitrate estimator test. by andresp@webrtc.org · 10 years ago
  17. 7f0b309 Remove GetDefaultConfigs() from Call. by pbos@webrtc.org · 10 years ago
  18. 2d4a80c Add boilerplate code for H.264. by stefan@webrtc.org · 10 years ago
  19. 65afbf3 Configure RTX send status on new modules. by pbos@webrtc.org · 10 years ago
  20. c0341b4 Adding pbos as video/ owner and removing persons never working with this folder. by mflodman@webrtc.org · 10 years ago
  21. 88b558f Reserve RTP/RTCP modules in SetSSRC. by pbos@webrtc.org · 10 years ago
  22. eb67a6b Refactor Call-based tests. by pbos@webrtc.org · 10 years ago
  23. 3610f63 GN: Add BUILD.gn files + kjellander to OWNERS by kjellander@webrtc.org · 10 years ago
  24. 4ee6348 Add tests of texture frames in video_send_stream_test. by wuchengli@chromium.org · 10 years ago
  25. f89ce46 Implements start bitrate for new video API. by mflodman@webrtc.org · 10 years ago
  26. 19f89a1 Enable pacing by default and remove the option to disable it from the new API. by stefan@webrtc.org · 10 years ago
  27. 6845de7 Add APIs to enable padding with redundant payloads. by stefan@webrtc.org · 10 years ago
  28. bdfcddf Make VideoSendStream/VideoReceiveStream configs const. by pbos@webrtc.org · 10 years ago
  29. 59a001f Adding back platform specific renderer to video loopback test. by mflodman@webrtc.org · 10 years ago
  30. 903e746 Have RTX be enabled by setting an RTX payload type instead of by setting an RTX SSRC. by stefan@webrtc.org · 10 years ago
  31. 1ef9cee Increase the threshold for CallPerfTest.CaptureNtpTimeWithNetworkDelay to avoid flaky. by wu@webrtc.org · 10 years ago
  32. 99153ba First incoming packet was not accounted for in receive stats. Changed call order for incoming packet to receive statistics class. by asapersson@webrtc.org · 10 years ago
  33. c78d40d Revert "Revert "Remove VideoSendStreamInput::PutFrame."" by pbos@webrtc.org · 10 years ago
  34. f3f2dba Revert "Remove VideoSendStreamInput::PutFrame." by pbos@webrtc.org · 10 years ago
  35. 0c60ee8 Remove VideoSendStreamInput::PutFrame. by pbos@webrtc.org · 10 years ago
  36. b139e8a Disable CallPerfTest.CaptureNtpTimeWithNetworkDelay due to being flaky. by stefan@webrtc.org · 10 years ago
  37. 66b1bf8 Fix Win VideoSendStream::...::ToString() compiles. by pbos@webrtc.org · 10 years ago
  38. 7e68693 Add ToString() to VideoSendStream::Config. by pbos@webrtc.org · 10 years ago
  39. bc57e0f Add DeliveryStatus enum to DeliverPacket(). by pbos@webrtc.org · 10 years ago
  40. ff4e210 Re-enable the BitrateEstimatorTest cases for the Call API. by solenberg@webrtc.org · 10 years ago
  41. 11de507 Move gflags usage to video_loopback. by pbos@webrtc.org · 10 years ago
  42. d2fb259 Move timestamp_extrapolator and rtp_to_ntp to system wrapper so that it can be shared by both audio and video engine. by wu@webrtc.org · 10 years ago
  43. 1cbc360 * Add 100ms network delay to test CaptureNtpTimeWithNetworkJitter. by wu@webrtc.org · 10 years ago
  44. c476e64 Add thread annotations to Call API. by pbos@webrtc.org · 10 years ago
  45. 33d613a Disable flaky CaptureNtpTimeWithNetworkJitter. by pbos@webrtc.org · 10 years ago
  46. fb3f14e Disabling flaky CanReceiveFec. by pbos@webrtc.org · 10 years ago
  47. 093fc0b Calculate local/remote clock delta and capture ntp timestamp in receiver's timebase. by wu@webrtc.org · 10 years ago
  48. 99c0139 Remove TraceCallback use from Call. by pbos@webrtc.org · 10 years ago
  49. 16a058a Rename Start/Stop in Video{Send,Receive}Streams. by pbos@webrtc.org · 10 years ago
  50. 6cee2ba Let A/V sync test use default AudioCoding module by henrik.lundin@webrtc.org · 10 years ago
  51. 98f8320 Make everyone an OWNER for .gyp/.gypi add/delete purposes, non-talk/ edition. by fischman@webrtc.org · 10 years ago
  52. 55bc281 Disable UsesTraceCallback by pbos@webrtc.org · 10 years ago
  53. ea15f8d Implement FEC support in VideoReceiveStream. by pbos@webrtc.org · 10 years ago
  54. 9968131 Convert logs in rtp rtcp module from WEBRTC_TRACE into LOG. by andresp@webrtc.org · 10 years ago
  55. 8d93b11 Clean up traces and logs in RemoteBitrateEstimator. by stefan@webrtc.org · 10 years ago
  56. d7aa228 Re-submit: rev5775 by andresp@webrtc.org · 10 years ago
  57. ca28c29 Revert 5775 "Modify bitrate controller to update bitrate based o..." by andrew@webrtc.org · 10 years ago
  58. 9deb87b Change sprintf format string from %zu to %i by henrik.lundin@webrtc.org · 10 years ago
  59. a0320c2 Modify bitrate controller to update bitrate based on process call and not by andresp@webrtc.org · 10 years ago
  60. fec6b6e VoE changes to allow forwarding of packets from VoE to ViE BWE. by solenberg@webrtc.org · 10 years ago
  61. b9d0acb Add AIMD option to BWE API. by stefan@webrtc.org · 10 years ago
  62. 700d14b Extend perf tests to perform rampup on single stream. by andresp@webrtc.org · 11 years ago
  63. 8abfc7f Disabling SendsSetSimulcastSsrcs. by pbos@webrtc.org · 11 years ago
  64. 45ae9e4 Disable flaky CanSwitchToUseAllSsrcs. by pbos@webrtc.org · 11 years ago
  65. 1d61e3a Simplify pacer interface. by pbos@webrtc.org · 11 years ago
  66. e2a7a77 Remove internal codecs from VideoSendStream. by pbos@webrtc.org · 11 years ago
  67. 0c0c604 Re-comitting r5711: "Fixing a flaky test in video_engine_tests" by henrik.lundin@webrtc.org · 11 years ago
  68. a50f107 Revert 5711 "Fixing a flaky test in video_engine_tests" by turaj@webrtc.org · 11 years ago
  69. 323af23 Fixing a flaky test in video_engine_tests by henrik.lundin@webrtc.org · 11 years ago
  70. c6f6696 Refactor rampup tests: by andresp@webrtc.org · 11 years ago
  71. 96616cb Stopping network threads before tearing down test by henrik.lundin@webrtc.org · 11 years ago
  72. 9376c69 Re-landing "Routing SuspendChange to VideoSendStream::Stats" by henrik.lundin@webrtc.org · 11 years ago
  73. 3f83f9c Implement minimum transmit bitrate. by pbos@webrtc.org · 11 years ago
  74. 691c5b2 Enable all RampUpTest.UpDownUp* tests by henrik.lundin@webrtc.org · 11 years ago
  75. 2a25b6c Replace labs with std::abs. by pbos@webrtc.org · 11 years ago
  76. ee86b90 Remove platform-specific code from new-API tests. by pbos@webrtc.org · 11 years ago
  77. 9759355 Revert "Routing SuspendChange to VideoSendStream::Stats" by henrik.lundin@webrtc.org · 11 years ago
  78. 2ae3c62 Routing SuspendChange to VideoSendStream::Stats by henrik.lundin@webrtc.org · 11 years ago
  79. 5b67882 Adding a link to issue by henrik.lundin@webrtc.org · 11 years ago
  80. 70e2ce9 NetEq4: Changing the behavior of playout_timestamp_ update by henrik.lundin@webrtc.org · 11 years ago
  81. 379c349 Potential deadlock in VideoSendStreamTest::ProducesStats by sprang@webrtc.org · 11 years ago
  82. 0435a83 Use DISABLE_ instead of commenting out tests by henrik.lundin@webrtc.org · 11 years ago
  83. c766098 Adding a new ramp-up-down-up test by henrik.lundin@webrtc.org · 11 years ago
  84. 6a9c344 Fix compilation errors under clang 3.5. by pbos@webrtc.org · 11 years ago
  85. 3e4cdec Incorrect overhead calculation when using FEC + RTP extension headers. by sprang@webrtc.org · 11 years ago
  86. 0feb8fa Make VideoReceiveStream::GetStats() const. by sprang@webrtc.org · 11 years ago
  87. c8ab721 Wire up statistics in video receive stream of new API by sprang@webrtc.org · 11 years ago
  88. 8ef6548 Add configuration for cpu overuse detection to video send stream. by asapersson@webrtc.org · 11 years ago
  89. 2bb7ad5 Fix race when deleting video receive streams in Call. by solenberg@webrtc.org · 11 years ago
  90. 0a29815 Drop early packets when not sending in TransportAdapter. by sprang@webrtc.org · 11 years ago
  91. a370f24 Always initialize Trace in Call TraceDispatcher. by pbos@webrtc.org · 11 years ago
  92. c71929d Wire up RTX in VideoReceiveStream. by pbos@webrtc.org · 11 years ago
  93. e822c84 Set NACKed packet to -1 in TestNackRetransmission. by pbos@webrtc.org · 11 years ago
  94. b4263e0 Add configuration and test for extended RTCP reference time reports to new video api. by asapersson@webrtc.org · 11 years ago
  95. 75e7da3 Permitting double start/stopping of streams. by pbos@webrtc.org · 11 years ago
  96. 0b9d7ce Add another test case for AST/TOF switching. by solenberg@webrtc.org · 11 years ago
  97. 49812e6 Wire up statistics in video send stream of new video engine api by sprang@webrtc.org · 11 years ago
  98. 4f1f5fa Add callbacks for receive channel RTCP statistics. by sprang@webrtc.org · 11 years ago
  99. aacdb9f Integrate fake_network_pipe into direct_transport. by stefan@webrtc.org · 11 years ago
  100. 5e0cbcf cpplint cleaning new API and its implementation files. by mflodman@webrtc.org · 11 years ago