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gerrit-public.fairphone.software / platform / external / webrtc / fae640003cf028477819309777ecd29a74c25eff / video / rtp_video_stream_receiver_unittest.cc
  1. cebdbf6 switch RtpVideoStreamReceiver to use VideoRtpDepacketizer interface by Danil Chapovalov · 4 years, 8 months ago
  2. d08bb1e Propagate absolute capture time through video receiving side. by Ruslan Burakov · 4 years, 8 months ago
  3. aa3f5da Fork VCMPacket for PacketBuffer into own struct by Danil Chapovalov · 4 years, 9 months ago
  4. 7b46e17 In rtc::ByteBuffer drop support for ORDER_HOST as unused by Danil Chapovalov · 4 years, 9 months ago
  5. c71d85b Pass full RtpPacket to RtpVideoStreamReceiver::OnReceivedPayload by Danil Chapovalov · 4 years, 10 months ago
  6. 51bf200 Reduce number of RTPVideoSender::SendVideo parameters by Danil Chapovalov · 4 years, 10 months ago
  7. 1b57541 Always pass arguments to INSTANTIATE_TEST_SUITE_P. by Mirko Bonadei · 4 years, 10 months ago
  8. 317a1f0 Use std::make_unique instead of absl::make_unique. by Mirko Bonadei · 4 years, 10 months ago
  9. 0d210ee Change return type of of ReceiveStatistics::Create to unique_ptr. by Niels Möller · 5 years ago
  10. bd3f305 Request a new key frame if packet buffer is cleared by Johannes Kron · 5 years ago
  11. a4d8737 Format almost everything. by Jonas Olsson · 5 years ago
  12. f00bf42 Add plumbing of RtpPacketInfos to each VideoFrame as input for SourceTracker. by Chen Xing · 5 years ago
  13. ef09c5b Buffer RTCP feedback messages in RtpVideoStreamReceiver by Elad Alon · 5 years ago
  14. 2f5554d Make KeyFrameRequestSender injectable in RtpVideoStreamReceiver by Niels Möller · 5 years ago
  15. 479c055 Let RtpVideoStreamReceiver implement KeyFrameRequestSender by Niels Möller · 5 years ago
  16. fe68daa Add option to configure raw RTP packetization per payload type. by Mirta Dvornicic · 5 years ago
  17. 60f4e29 Delete configuration of unused transport_sequence_number_allocator by Niels Möller · 5 years ago
  18. 0fb0bd8 Delete WebRtcRTPHeader, this struct is no longer used. by Niels Möller · 5 years ago
  19. abbc50e Move frame_type member from RtpDepacketizer::ParsedPayload to RTPVideoHeader by Niels Möller · 5 years ago
  20. d51ec58 Parse color space only in last packet of key frame by Johannes Kron · 5 years ago
  21. 6a489f2 Fully qualify googletest symbols. by Mirko Bonadei · 5 years ago
  22. 8f7ce22 Make VideoFrameType an enum class, and move to separate file and target by Niels Möller · 5 years ago
  23. 125b5d6 Refactor RtpVideoStreamReceiver::OnReceivedPayloadData without WebRtcRTPHeader by Niels Möller · 5 years ago
  24. 8026d60 Injecting Clock in video receive. by Sebastian Jansson · 5 years ago
  25. ccb9b75 Create version 01 of Generic Frame Descriptor - with discardability flag by Elad Alon · 5 years ago
  26. c84f661 Stop using Googletest legacy APIs. by Mirko Bonadei · 5 years ago
  27. 067dc86 Make SetFirstSubFrameInFrame and SetLastSubFrameInFrame protected by Elad Alon · 6 years ago
  28. 10542f2 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries by Steve Anton · 6 years ago
  29. 9c84390 Delete VCMEncodedFrame methods Buffer and MutableBuffer by Niels Möller · 6 years ago
  30. f0eee00 Move size() method to EncodedImage base class by Niels Möller · 6 years ago
  31. 648a7ce Delete method EncodedFrame::GetBitstream, part 1 by Niels Möller · 6 years ago
  32. a32d7e2 Add default values for PlayoutDelay in RTPVideoHeader. by Niels Möller · 6 years ago
  33. 192eeec Enable End-to-End Encrypted Video Frames. by Benjamin Wright · 6 years ago
  34. fab9129 Get frame type, width and height from the generic descriptor. by philipel · 6 years ago
  35. 17f4878 Remove deprecated field_trial_default and metrics_default. by Mirko Bonadei · 6 years ago
  36. dabfcae Use the generic descriptor information in the RtpFrameReferenceFinder. by philipel · 6 years ago
  37. ef615ea Added is_last_packet_in_frame to match is_first_packet_in_frame. by philipel · 6 years ago
  38. 7d745e5 Reland "Remove RTPVideoHeader::h264() accessors." by philipel · 6 years ago
  39. 5daeff9 Revert "Remove RTPVideoHeader::h264() accessors." by JT Teh · 6 years ago
  40. dfbced6 Remove RTPVideoHeader::h264() accessors. by philipel · 6 years ago
  41. a715f28 Fix handling invalid empty red packets by Danil Chapovalov · 6 years ago
  42. 5ab67a5 Add accessors to the types in the RTPVideoTypeHeader in RTPVideoHeader. by philipel · 6 years ago
  43. 918f50c Use absl::make_unique and absl::WrapUnique directly by Karl Wiberg · 6 years ago
  44. cb96ad8 Add ParsedPayload::video_header() accessor. by philipel · 6 years ago
  45. 665174f Reformat the WebRTC code base by Yves Gerey · 6 years ago
  46. 520ca4e Delete enum RtpVideoCodecTypes, replaced with VideoCodecType. by Niels Möller · 6 years ago
  47. 2e1d784 Delete the VideoCodec::plName string. by Niels Möller · 6 years ago
  48. 0a9f6de Removed VCMTiming from RtpVideoStreamReceiver. by philipel · 6 years ago
  49. e7c891f Renamed FrameObject to EncodedFrame. by philipel · 6 years ago
  50. d8f6c16 Avoid infinite recursion if a RED packet encapsulate a RED packet. by philipel · 7 years ago
  51. 88f080a Move SPS/PPS/IDR requirement from RtpFrameObject to PacketBuffer. by Rasmus Brandt · 7 years ago
  52. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  53. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/video/rtp_video_stream_receiver_unittest.cc]
  54. ca5706d Reland of Use RtxReceiveStream. (patchset #1 id:1 of https://codereview.webrtc.org/3007303002/ ) by nisse · 7 years ago
  55. 8e7eee0 Revert of Use RtxReceiveStream. (patchset #5 id:320001 of https://codereview.webrtc.org/3006063002/ ) by nisse · 7 years ago
  56. 35713ea Reland of Use RtxReceiveStream. (patchset #1 id:1 of https://codereview.webrtc.org/3010983002/ ) by nisse · 7 years ago
  57. c0d481a Protected streams report RTP messages directly to the FlexFec streams by eladalon · 7 years ago
  58. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 7 years ago
  59. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 7 years ago
  60. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 7 years ago
  61. 83c97da Only append SPS/PPS to bitstream if supplied out of band. by philipel · 7 years ago
  62. b1f2ff9 Rename class RtpStreamReceiver --> RtpVideoStreamReceiver. by nisse · 7 years ago[Renamed (84%) from webrtc/video/rtp_stream_receiver_unittest.cc]
  63. 2c53b13 Request keyframe if the first received frame is not a keyframe. by philipel · 7 years ago
  64. 0584331 Delete VieRemb class, move functionality to PacketRouter. by nisse · 7 years ago
  65. 54ca919 Allow padding packet in video streams. by philipel · 7 years ago
  66. 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 · 7 years ago
  67. e525d6a Revert Make the new jitter buffer the default jitter buffer. by stefan · 7 years ago
  68. 4709e89 Move RemoteBitrateEstimator::RemoveStream calls from receive streams to Call. by nisse · 7 years ago
  69. e5bd702 Reland of Make the new jitter buffer the default jitter buffer. (patchset #2 id:260001 of https://codereview.chromium.org/2656983002/ ) by philipel · 7 years ago
  70. 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
  71. 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
  72. 62d02c3 Unit test out of band H264 SPS,PPS within RtpStreamReceiver. by johan · 8 years ago