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• FFMPEG stereo track stops capturing at random times during a capture session

  26 mai 2022, par mrwassen

  I am currently working on building a workflow to capture and archive a large stash of family and friends PAL and NTSC VHS tapes. The hardware setup is as follows :

  


  

  • JVC HR-7860S VCR

  


  • s-video / RCA audio >

  


  • ADVC-3000 converter

  


  • SDI / BNC cable >

  


  • Blackmagic Decklink Mini Recorder 4K PCIe card

  


  • installed in a fairly hi-spec windows machine : AMD Ryzen 9 5900X 3.7 Ghz base 12 core, GEFORCE RTX 3060 12 gB, 32 gB ram

  


  


  The plan is to capture to lossless AVI, then drop into an NLE (Vegas Pro v.16) to do a minimal amount of cleanup / trimming, then render to a more compressed video format (TBD) for upload to AWS S3 accessible through a family website.

  


  The issue I am having is that when I run the capture using ffmpeg/directshow e.g. for a perfectly fine 90 min. PAL tape, at some random point of time during the capture one of the 2 stereo channels just stops capturing. This has happened with all of the tapes I have tested so far, and it happens at different times during the same video. I have examined the frames surrounding points in time when this happens, and it doesn't correlate to any transitions or jitter, but often just randomly in the middle of a perfectly smooth scene. Once the one channel stops capturing it never starts back up again during that capture session.

  


  The ADVC-3000 and the VCR are both showing both stereo channels playing normally throughout the capture. The windows machine running the capture hardly breaks a sweat at any time, and the transfer easily keeps up constantly showing a speed = 1x which I assume means nothing lagging. Also there are no video/audio sync issues at any point in time even towards the end of long tapes e.g. 90 mins.

  


  I am fairly new at ffmpeg, so I have spent extensive amounts of time reading up on forum posts and experimenting and have ended up with the following syntax :

  


  ffmpeg -y -f dshow -rtbufsize 2000M -i video="Blackmagic WDM Capture":audio="Blackmagic WDM Capture" -codec:v v210 -pix_fmt yuv422p -codec:a pcm_s16le -b:a 128k -t 02:00:00 -r 25 -threads 4 -maxrate 2500k -filter:a "volume=1.5" output_v210_audio.avi


  


  The capture runs without a single dropped frame, the only error I am getting when launching (and perhaps this is a smoking gun ?) is :

  


  


  "Non-monotonous DTS in output stream 0:1 ; previous : 0, current : -30 ;
changing to 1. This may result in incorrect timestamps in the output
file."

  


  


  I have tried to troubleshoot this in the hopes that it is tied to my issue but so far without luck.

  


  Hoping somebody can help correct or modify my command line or perhaps other ideas to help resolve the issue.

  


• tests : do not use AVFrame accessor

  22 avril 2017, par Muhammad Faiz

  tests : do not use AVFrame accessor
  Reviewed-by : wm4 <nfxjfg@googlemail.com>
  
  Signed-off-by : Muhammad Faiz <mfcc64@gmail.com>
  

  • [DH] tests/api/api-flac-test.c
  • [DH] tests/api/api-h264-test.c
  • [DH] tests/api/api-seek-test.c
  • [DH] tests/api/api-threadmessage-test.c

• fate/libavcodec : add codec_desc test

  10 février 2018, par Muhammad Faiz

  fate/libavcodec : add codec_desc test
  Remove runtime check at codec_desc.c
  Reviewed-by : wm4 <nfxjfg@googlemail.com>
  
  Reviewed-by : Michael Niedermayer <michael@niedermayer.cc>
  
  Signed-off-by : Muhammad Faiz <mfcc64@gmail.com>
  

  • [DH] libavcodec/Makefile
  • [DH] libavcodec/codec_desc.c
  • [DH] libavcodec/tests/codec_desc.c
  • [DH] tests/fate/libavcodec.mak