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gerrit-public.fairphone.software / platform / external / webrtc / 44c608a7a711fad2f9e55fae154516cc50d47d50 / video / video_receive_stream.cc
  1. 38c5d93 Reduce locking for CallStats (preparation for TaskQueue). by Tommi · 7 years ago
  2. 81de14f Reduce implementations of CallStatsObserver by 2. by Tommi · 7 years ago
  3. 0fa82a6 Moved FrameKey to api/video/encoded_frame.h and renamed it to VideoLayerFrameId. by philipel · 7 years ago
  4. 2e1d784 Delete the VideoCodec::plName string. by Niels Möller · 7 years ago
  5. 0a9f6de Removed VCMTiming from RtpVideoStreamReceiver. by philipel · 7 years ago
  6. 132e28e Add thread checks to ReceiveStatisticsProxy that reflect design comments. by Tommi · 7 years ago
  7. fbf3bce Reland "Reduce locking in VideoReceiver and check the threading model." by Tommi · 7 years ago
  8. e7c891f Renamed FrameObject to EncodedFrame. by philipel · 7 years ago
  9. c4f9824 Revert "Reduce locking in VideoReceiver and check the threading model." by Lu Liu · 7 years ago
  10. c75f1e4 Reduce locking in VideoReceiver and check the threading model. by Tommi · 7 years ago
  11. d397a0d Add dropped frames metric on the receive side by Ilya Nikolaevskiy · 7 years ago
  12. d7ae3c3 Reland "Rename stereo video codec to multiplex" by Emircan Uysaler · 7 years ago
  13. 1204448 Revert "Reland "Rename stereo video codec to multiplex"" by Taylor Brandstetter · 7 years ago
  14. 4954a77 Reland "Rename stereo video codec to multiplex" by Emircan Uysaler · 7 years ago
  15. 6bc7bb6 Revert "Rename stereo video codec to multiplex" by Ivo Creusen · 7 years ago
  16. bbdabe5 Rename stereo video codec to multiplex by Emircan Uysaler · 7 years ago
  17. 8e07c13 Optional: Use nullopt and implicit construction in /video by Oskar Sundbom · 7 years ago
  18. a9329db Bug Fix: Peers Cannot Communicate If One With Stereo Codec, One Not by Qiang Chen · 7 years ago
  19. be214a2 Move videosinkinterface.h to common_video to solve a circular dep. by Patrik Höglund · 7 years ago
  20. 49b46e0 Added WEBRTC_VIDEO_CODEC_OK_REQUEST_KEYFRAME decoder return code. by philipel · 7 years ago
  21. 1610f94 Don't cast picture ids (of type int64_t) to int. by philipel · 7 years ago
  22. 0a37547 Add optional stereo codec to SDP negotiation by Emircan Uysaler · 7 years ago
  23. 675513b Stop using LOG macros in favor of RTC_ prefixed macros. by Mirko Bonadei · 7 years ago
  24. c3fa8e1 New method RtpReceiver::GetLatestTimestamps. by Niels Möller · 7 years ago
  25. 48462b6 Continuously request keyframes if decoding does not recover. by philipel · 7 years ago
  26. e21be1d Reland of Fix the video buffer size should take rtt into consideration (patchset #2 id:160001 of https://codereview.chromium.org/3002033002/ ) by philipel · 7 years ago
  27. 7120742 Adding NOLINT for typedefs.h and common_types.h by Mirko Bonadei · 7 years ago
  28. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  29. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/video/video_receive_stream.cc]
  30. ca5706d Reland of Use RtxReceiveStream. (patchset #1 id:1 of https://codereview.webrtc.org/3007303002/ ) by nisse · 7 years ago
  31. 8e7eee0 Revert of Use RtxReceiveStream. (patchset #5 id:320001 of https://codereview.webrtc.org/3006063002/ ) by nisse · 7 years ago
  32. 35713ea Reland of Use RtxReceiveStream. (patchset #1 id:1 of https://codereview.webrtc.org/3010983002/ ) by nisse · 7 years ago
  33. 84f6a3f Move optional.h to webrtc/api/ by kwiberg · 7 years ago
  34. 3c39c01 Revert of Use RtxReceiveStream. (patchset #5 id:80001 of https://codereview.webrtc.org/3008773002/ ) by nisse · 7 years ago
  35. 75204c5 Change reporting of timing frames conditions in GetStats on receive side by ilnik · 7 years ago
  36. 5c0f6c6 Use RtxReceiveStream. by nisse · 7 years ago
  37. 1cdddc9 Make CodecType conversion functions non-optional. by kthelgason · 7 years ago
  38. 3e86e7e Ignore inter-frame delay stats samples when stream is inactive by sprang · 7 years ago
  39. 440b6d9 Move video send/receive stream headers to webrtc/call. by aleloi · 7 years ago
  40. bdbc889 Revert of Fix the video buffer size should take rtt into consideration (patchset #3 id:40001 of https://codereview.chromium.org/2980413002/ ) by philipel · 7 years ago
  41. 3042c2d Reland of quest keyframes more frequently on stream start/decoding error. (patchset #1 id:1 of https://codereview.chromium.org/2995153002/ ) by philipel · 7 years ago
  42. a28122f Change ThreadChecker to SequencedTaskChecker in VideoReceiveStream by eladalon · 7 years ago
  43. 53959fc Revert of quest keyframes more frequently on stream start/decoding error. (patchset #2 id:170001 of https://codereview.webrtc.org/2996823002/ ) by tkchin · 7 years ago
  44. 628ac59 Reland of quest keyframes more frequently on stream start/decoding error. (patchset #1 id:1 of https://codereview.chromium.org/2994043002/ ) by philipel · 7 years ago
  45. 77a9831 Revert of Request keyframes more frequently on stream start/decoding error. (patchset #1 id:1 of https://codereview.webrtc.org/2993793002/ ) by deadbeef · 7 years ago
  46. f1e08d0 Fix the video buffer size should take rtt into consideration by gustavogb · 7 years ago
  47. 26b4804 Request keyframes more frequently on stream start/decoding error. by philipel · 7 years ago
  48. 186d9c3 Renamed fields in common_types.h/RtcpStatistics. by srte · 7 years ago
  49. c0d481a Protected streams report RTP messages directly to the FlexFec streams by eladalon · 7 years ago
  50. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 7 years ago
  51. 2edc684 Report timing frames info in GetStats. by ilnik · 7 years ago
  52. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 7 years ago
  53. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 7 years ago
  54. 0f15f92 Introduce RtpStreamReceiverInterface and RtpStreamReceiverControllerInterface. by nisse · 7 years ago
  55. c8ece43 Minor updates to VideoReceiveStream: by tommi · 7 years ago
  56. b1f2ff9 Rename class RtpStreamReceiver --> RtpVideoStreamReceiver. by nisse · 7 years ago
  57. 3184f8e Dont request keyframes if the stream is inactive or if we are currently receiving a keyframe. by philipel · 7 years ago
  58. d2ef314 Make Call::OnRecoveredPacket parse RTP header and call OnRtpPacket. by nisse · 8 years ago
  59. 0584331 Delete VieRemb class, move functionality to PacketRouter. by nisse · 8 years ago
  60. c337258 Revert of Deliver video frames on Android, on the decode thread. (patchset #7 id:120001 of https://codereview.webrtc.org/2764573002/ ) by guidou · 8 years ago
  61. e3aa88b Deliver video frames on Android, on the decode thread. by tommi · 8 years ago
  62. c964d0b Fixing some case-sensitive codec name comparisons. by deadbeef · 8 years ago
  63. 5f54419 Revert of Don't set the priority of the decoder to 'high' on Android. (patchset #1 id:1 of https://codereview.webrtc.org/2745813003/ ) by tommi · 8 years ago
  64. d0a71ba Updates to VCMDecodedFrameCallback, VideoReceiver and a few related classes/tests. by tommi · 8 years ago
  65. ca37cf6 Don't set the priority of the decoder to 'high' on Android. by tommi · 8 years ago
  66. cb8c146 Add FullStack test for simulcast screenshare mode. by ilnik · 8 years ago
  67. c69385d Add |protected_by_flexfec| flag to VideoReceiveStream::Config. by nisse · 8 years ago
  68. db23ea6 Add performance tracing for PlatformThread and parts of the video code. by tommi · 8 years ago
  69. dea489f Add support for Location (RTC_FROM_HERE) to ProcessThread::RegisterModule. by tommi · 8 years ago
  70. 2dfea3e Avoid busy looping as the VideoReceiveStream is shut down. by philipel · 8 years ago
  71. a45102f Revert of Revert Make the new jitter buffer the default jitter buffer. (patchset #1 id:1 of https://codereview.chromium.org/2682073003/ ) by philipel · 8 years ago
  72. 38cc1d6 Replace RtpStreamReceiver::DeliverRtp with OnRtpPacket. by nisse · 8 years ago
  73. 3dd5ad9 Reland of Added VP8 simulcast tests. Fixed analyzer to correctly infer timestamps. (patchset #2 id:150001 of https://codereview.webrtc.org/2687073002/ ) by ilnik · 8 years ago
  74. cc452e1 Reland of Add QP sum stats for received streams. (patchset #2 id:300001 of https://codereview.webrtc.org/2680893002/ ) by sakal · 8 years ago
  75. e67c59e Revert of Added VP8 simulcast tests. Fixed analyzer to correctly infer timestamps. (patchset #5 id:80001 of https://codereview.webrtc.org/2668763004/ ) by ilnik · 8 years ago
  76. e525d6a Revert Make the new jitter buffer the default jitter buffer. by stefan · 8 years ago
  77. 69fb2cc Revert of Add QP sum stats for received streams. (patchset #10 id:180001 of https://codereview.webrtc.org/2649133005/ ) by skvlad · 8 years ago
  78. 76bc8e8 Delete VideoReceiveStream::Config::pre_render_callback. by nisse · 8 years ago
  79. ff0e72f Add QP sum stats for received streams. by sakal · 8 years ago
  80. 4709e89 Move RemoteBitrateEstimator::RemoveStream calls from receive streams to Call. by nisse · 8 years ago
  81. 5f47126 Added VP8 simulcast tests. Fixed analyzer to correctly infer timestamps. by ilnik · 8 years ago
  82. e5bd702 Reland of Make the new jitter buffer the default jitter buffer. (patchset #2 id:260001 of https://codereview.chromium.org/2656983002/ ) by philipel · 8 years ago
  83. ed01647 Remove bad DCHECK added as part of https://codereview.webrtc.org/2452163004/ by solenberg · 8 years ago
  84. 3ebbcb5 Stop using VoEVideoSync in Call/VideoReceiveStream. by solenberg · 8 years ago
  85. fb45c6c Inform jitter buffer about FlexFEC protection. by brandtr · 8 years ago
  86. 1474212 Reland of Make RTX pt/apt reconfigurable by calling WebRtcVideoChannel2::SetRecvParameters. (patchset #1 id:1 of https://codereview.webrtc.org/2649323010/ ) by brandtr · 8 years ago
  87. 27378f3 Revert of Make the new jitter buffer the default jitter buffer. (patchset #2 id:290001 of https://codereview.chromium.org/2652043005/ ) by philipel · 8 years ago
  88. e497495 Revert of Make RTX pt/apt reconfigurable by calling WebRtcVideoChannel2::SetRecvParameters. (patchset #7 id:160001 of https://codereview.webrtc.org/2646073004/ ) by kjellander · 8 years ago
  89. fe2bef3 Make RTX pt/apt reconfigurable by calling WebRtcVideoChannel2::SetRecvParameters. by brandtr · 8 years ago
  90. 09d6ef0 Reland of Make the new jitter buffer the default jitter buffer. (patchset #1 id:1 of https://codereview.webrtc.org/2638423003/ ) by philipel · 8 years ago
  91. 090c940 Sort method declarations/definitions in VideoReceiveStream. by brandtr · 8 years ago
  92. bfb11b2 Call RtpStreamReceiver.AddReceiveCodec() with codec_params. by johan · 8 years ago
  93. 15389c0 Drop pacer and retransmission_rate_limiter from RtpStreamReceiver constructor. by nisse · 8 years ago
  94. 04926b8 Revert of Make the new jitter buffer the default jitter buffer. (patchset #2 id:230001 of https://codereview.webrtc.org/2642753002/ ) by kjellander · 8 years ago
  95. f20dd00 Reland of Make the new jitter buffer the default jitter buffer. (patchset #1 id:1 of https://codereview.chromium.org/2632123005/ ) by philipel · 8 years ago
  96. c08c191 Revert of Make the new jitter buffer the default jitter buffer. (patchset #7 id:120001 of https://codereview.chromium.org/2627463004/ ) by philipel · 8 years ago
  97. 0f0763d Make the new jitter buffer the default jitter buffer. by philipel · 8 years ago
  98. 022b54e Wire up H264 fmtp sprop-parameter-sets with H264SpsPpsTracker. by philipel · 8 years ago
  99. 721d402 Create VideoReceiver with external VCMTiming object. by philipel · 8 years ago
  100. 07e276c Rename RtpStreamReceiver::SetCodec() to ::AddCodec(). by johan · 8 years ago