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webrtc / src / / ca3c8017e5fc272e6320d2de47befb3915c2297e / video / rtp_video_stream_receiver_unittest.cc
  1. c84f661 Stop using Googletest legacy APIs. by Mirko Bonadei · 6 years ago
  2. 067dc86 Make SetFirstSubFrameInFrame and SetLastSubFrameInFrame protected by Elad Alon · 6 years ago
  3. 10542f2 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries by Steve Anton · 6 years ago
  4. 9c843906 Delete VCMEncodedFrame methods Buffer and MutableBuffer by Niels Möller · 6 years ago
  5. f0eee00 Move size() method to EncodedImage base class by Niels Möller · 6 years ago
  6. 648a7ce Delete method EncodedFrame::GetBitstream, part 1 by Niels Möller · 6 years ago
  7. a32d7e2 Add default values for PlayoutDelay in RTPVideoHeader. by Niels Möller · 6 years ago
  8. 192eeec Enable End-to-End Encrypted Video Frames. by Benjamin Wright · 6 years ago
  9. fab9129 Get frame type, width and height from the generic descriptor. by philipel · 6 years ago
  10. 17f4878 Remove deprecated field_trial_default and metrics_default. by Mirko Bonadei · 7 years ago
  11. dabfcae Use the generic descriptor information in the RtpFrameReferenceFinder. by philipel · 7 years ago
  12. ef615ea Added is_last_packet_in_frame to match is_first_packet_in_frame. by philipel · 7 years ago
  13. 7d745e5 Reland "Remove RTPVideoHeader::h264() accessors." by philipel · 7 years ago
  14. 5daeff9 Revert "Remove RTPVideoHeader::h264() accessors." by JT Teh · 7 years ago
  15. dfbced6 Remove RTPVideoHeader::h264() accessors. by philipel · 7 years ago
  16. a715f28 Fix handling invalid empty red packets by Danil Chapovalov · 7 years ago
  17. 5ab67a5 Add accessors to the types in the RTPVideoTypeHeader in RTPVideoHeader. by philipel · 7 years ago
  18. 918f50c Use absl::make_unique and absl::WrapUnique directly by Karl Wiberg · 7 years ago
  19. cb96ad8 Add ParsedPayload::video_header() accessor. by philipel · 7 years ago
  20. 665174f Reformat the WebRTC code base by Yves Gerey · 7 years ago
  21. 520ca4e Delete enum RtpVideoCodecTypes, replaced with VideoCodecType. by Niels Möller · 7 years ago
  22. 2e1d784 Delete the VideoCodec::plName string. by Niels Möller · 7 years ago
  23. 0a9f6de Removed VCMTiming from RtpVideoStreamReceiver. by philipel · 7 years ago
  24. e7c891f Renamed FrameObject to EncodedFrame. by philipel · 7 years ago
  25. d8f6c16 Avoid infinite recursion if a RED packet encapsulate a RED packet. by philipel · 7 years ago
  26. 88f080a Move SPS/PPS/IDR requirement from RtpFrameObject to PacketBuffer. by Rasmus Brandt · 7 years ago
  27. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 8 years ago
  28. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 8 years ago[Renamed from webrtc/video/rtp_video_stream_receiver_unittest.cc]
  29. ca5706d Reland of Use RtxReceiveStream. (patchset #1 id:1 of https://codereview.webrtc.org/3007303002/ ) by nisse · 8 years ago
  30. 8e7eee0 Revert of Use RtxReceiveStream. (patchset #5 id:320001 of https://codereview.webrtc.org/3006063002/ ) by nisse · 8 years ago
  31. 35713ea Reland of Use RtxReceiveStream. (patchset #1 id:1 of https://codereview.webrtc.org/3010983002/ ) by nisse · 8 years ago
  32. c0d481a Protected streams report RTP messages directly to the FlexFec streams by eladalon · 8 years ago
  33. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 8 years ago
  34. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 8 years ago
  35. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 8 years ago
  36. 83c97da Only append SPS/PPS to bitstream if supplied out of band. by philipel · 8 years ago
  37. b1f2ff9 Rename class RtpStreamReceiver --> RtpVideoStreamReceiver. by nisse · 8 years ago[Renamed (84%) from webrtc/video/rtp_stream_receiver_unittest.cc]
  38. 2c53b13 Request keyframe if the first received frame is not a keyframe. by philipel · 8 years ago
  39. 0584331 Delete VieRemb class, move functionality to PacketRouter. by nisse · 8 years ago
  40. 54ca919 Allow padding packet in video streams. by philipel · 8 years ago
  41. 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
  42. e525d6a Revert Make the new jitter buffer the default jitter buffer. by stefan · 8 years ago
  43. 4709e89 Move RemoteBitrateEstimator::RemoveStream calls from receive streams to Call. by nisse · 8 years ago
  44. 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
  45. 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
  46. 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
  47. 62d02c3 Unit test out of band H264 SPS,PPS within RtpStreamReceiver. by johan · 8 years ago