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• ffmpeg recording audio / video sync issues [closed]

  1er novembre 2022, par sling jones

  I'm using ffmpeg to record NTSC analog video on Linux Fedora 36 using a Blackmagic Intensity Pro 4K for video and a Scarlett 2i2 for audio. I'm using a TBC to avoid dropped frames and to ensure a constant S-Video Y/C framerate on the analog end.

  


  The problem I'm running into is that on playback the audio will start out relatively in sync with the video at the beginning of the captured file but will eventually run ahead of the video eventually becoming many seconds off.

  


  Nothing I do seems to change this or change the degree to which it happens. The audio and video stay in sync throughout the entire video as I'm monitoring the source so I don't understand how they can diverge so much once encoded into a digital file ?

  


  Here is the command I am using :

  


  ffmpeg -hwaccel cuda -fflags +igndts -format_code ntsc -f decklink -raw_format auto -vsync passthrough -rtbufsize 1500M -thread_queue_size 512 -i 'Intensity Pro 4K' -f pulse -rtbufsize 500M -thread_queue_size 512 -i 'Scarlett 2i2 Camera Analog Stereo' -c copy -map 0:1 -map 1:0 "/tmp/ffmpeg-raw/file-raw.avi"


  


  here's the ffprobe output from one of my files :

  


  Input #0, avi, from 'test2-raw.avi':
  Metadata:
    software        : Lavf59.33.100
  Duration: 00:11:42.87, start: 0.000000, bitrate: 169341 kb/s
  Stream #0:0: Video: rawvideo (UYVY / 0x59565955), uyvy422, 720x486, 167801 kb/s, 59.94 fps, 59.94 tbr, 59.94 tbn
  Stream #0:1: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz, 2 channels, s16, 1536 kb/s



  


  As you can see in my code snippet, I've been throwing things at the wall for a bit. I've tried different rtbufsizes, adding -copyts, and going through the different -vsync options. I've tried it with and without hardware acceleration ( I do have a NVIDIA card), +igdts did get rid of a warning but did not help with the sync, as did changing the thread queue sizes.

  


  OBS can do this, why can't I ?

  


• What is the best solution to convert old videos to newer more optimised formats ? [on hold]

  9 septembre 2019, par Jack

  Not too sure if this is the wrong place - please move it as I couldn’t find a more suitable network

  I have loads of media (Movies, TV Shows) as well as home videos (old VHS stuff ripped using some awful VHS to digital kit)

  Most of the movies/TV shows are in H264 (MP4/MKV containers) format, however some older ones are in AVI and WMV - I’d like to convert these into either H264 or a newer format (HEVC ?) To save some disk space and also because WMVs and AVIs are getting harder to deal with nowadays. I’m concerned about losing quality and am wondering what would be the best compromise in terms of converting these to HEVC/MPEG4’s encoder quality settings as compared to the data savings.

  The media collection of TV shows/movies, I don’t mind too much about losing some quality but the home videos/VHS tapes I have in old file formats, the storage factor for these is less important but I was wondering what I’d need to do to convert old AVI’s/MPEG2’s to MPEG4/HEVC - mainly if it is possible to convert one of these old video files to a newer format, without loss of quality, I thought the newer video encoding’s had lossless and lossy compression, but I could be completely wrong and don’t know much about video codecs.

  I was more curious on the best solution to do this as, Googling it gives me loads of commercial software and I’d rather have something which I can use command line/programmatically against my entire libraries. I also couldn’t find anything on these commercial sites about the technicals of re-encoding video so, was wondering if anyone had any experience with any command line applications/have an understanding of the video codecs.

• change wav, aiff or mov audio sample rate of MOV or WAV WITHOUT changing number of samples

  6 mars 2013, par John Pilgrim

  I need a very precise way to speed up audio.
  I am preparing films for OpenDCP, an open-source tool to make Digital Cinema Packages, for screening in theaters.
  My source files are usually quicktime MOV files at 23.976fps with 48.000kHz audio.
  Sometimes my audio is a separate 48.000kHz WAV.
  (FWIW, the video frame rate of the source is actually 24/100.1 frames per second, which is a repeating decimal.)

  The DCP standard is based around a 24.000fps and 48.000kHz program, so the audio and video of the source need to be sped up.
  The image processing workflow inherently involves converting the MOV to a TIF sequence, frame-per-frame, which is then assumed to be 24.000fps, so I don't have to get involved in the internals of the QT Video Media Handler.

  But speeding up the audio to match is proving to be difficult. Most audio programs cannot get the number of audio samples to line up with the retimed image frames. A 0.1% speed increase in Audacity results in the wrong number of samples. The only pathway that I have found that works is to use Apple Cinema Tools to conform the 23.976fps/48.000kHz MOV to 24.000fps/48.048kHz (which it does by changing the Quicktime headers) and then using Quicktime Player to export the audio from that file at 48.000kHz, resampling it. This is frame accurate.

  So my question is : are there settings in ffmpeg or sox that will precisely speed up the audio in a MOV or in a WAV or AIFF precisely ? I would like a cross platform solution, so I am not dependent on Cinema Tools, which is only MacOS.

  I know this is a LOT of background. Feel free to ask clarifying questions !