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• ffmpeg merge separate .webm audio and video files using pts_time possible ?

  4 août 2021, par Zach

  I have a number of audio and video files with different start times and end times. They're all generated from piped input streams (node.js) with .webm format, so the audio files have gaps where there is no audio in the piped stream.

  


  I'm trying to :

  


  

  1. Merge the audio files together with wall-clock correct start/end times

  


  2. Merge the video files using hstack.

  


  3. Combine the merged audio and merged video into 1 final video with all video/audio.

  


  


  Right now I'm still stuck on step 1 - Merge the audio

  


  My command that generates separate audio files is :

  


          '-protocol_whitelist',
        'pipe,udp,rtp',
        '-fflags',
        '+genpts',
        '-f',
        'sdp',
        '-use_wallclock_as_timestamps',
        'true',
        '-i',
        'pipe:0'
        '-copyts',
        '-map',
        '0:a:0',
        '-strict'
        '-2',
        '-c:a',
        'copy'


  


  I'd love to combine them somehow using the timestamps of each packet fill the empty space with silence. Right now, I'm putting them together with offsets using the time that I initiate the ffmpeg process from node.js, but these times are incorrect, as it takes a moment for ffmpeg to start up.

  


  Any assistance or a push in the right direction for time sensitive merging of audio/video .webm files with ffmpeg would be outstanding.

  


  Thanks !

  


  PS, here's what I'm currently doing and running into a problems with :

  


     '-i',
  './recordings/audio_1.webm',
  '-i',
  './recordings/audio_2.webm',
  '-filter_complex',
  '[1]adelay=6384|6384[b];[0][b]amix=2',
  './recordings/merged_audio.webm'


  


  The delays are inaccurate (because they're based on an estimate of when the first packet starts) and doesn't account for gaps in the audio files :(

  


• Set the filename downloaded via youtube-dl to a variable [closed]

  22 septembre 2020, par Jim Jamil

  This is the current script, it's a Windows batch file that prompts for a Youtube url and then downloads the best audio in m4a. It's basically cobbled together and uses aria2 to manage the download.

  


  @echo off
SETLOCAL ENABLEDELAYEDEXPANSION
(set /p var1="Url? " && youtube-dl -f bestaudio[ext=m4a] --external-downloader aria2c --external-downloader-args "-j 16 -s 16 -x 16 -k 5M" --restrict-filenames -o "%%(title)s.%%(ext)s" --add-metadata --embed-thumbnail !Var1!)
ENDLOCAL
pause


  


  After asking for the url, I want to also prompt the user to input the start and end times to trim the audio, which would be done by ffmpeg post download.

  


  Something like :

  


  ffmpeg -i file.m4a -ss 00:00:20 -to 00:00:40 -c copy file-2.m4a


  


  Based on this : https://unix.stackexchange.com/questions/182602/trim-audio-file-using-start-and-stop-times/302469#302469

  


  The beginning and end times would need to be variables set by user input in 00:00:00 format, but not sure how to add the ffmpeg post-processing at the end or how it would all fit together. I want to add this trimming feature to remove some of the preamble on podcasts and get straight to the guest part of the show.

  


  The --embed-thumbnail is optional, and won't work anyway unless Atomic Parsley is present. FFmpeg often has trouble with Album Art anyway so I usually just use -vn on the final output file.

  


• Streaming raw h264 video from Raspberry PI to server for capture and viewing [closed]

  24 juin 2024, par tbullers

  This is really an optimization question - I have been able to stream h264 from a raspberry pi 5 to a linux system and capture the streams and save them to .mp4 files.

  


  But I intend to run the video capture and sending on a battery powered Pi Zero 2 W and want to use the least amount of power to maximize battery life and still providing good video quality.

  


  I've explored many different configuration settings but am getting lost in all the options.

  


  This is what I run on the pi :

  


  rpicam-vid -t 30s --framerate 30 --hdr --inline --listen -o tcp://0.0.0.0:5000


  


  I retrieve this video from the more powerful Ubuntu server with :

  


  ffmpeg -r 30 -i tcp://ralph:5000 -vcodec copy video_out103.mp4


  


  It generally works but I receive lots of errors on the server side like this :

  


  [mp4 @ 0x5f9aab5d0800] pts has no valuee= 975.4kbits/s speed=1.19x
Last message repeated 15 times
[mp4 @ 0x5f9aab5d0800] pts has no valuee=1035.3kbits/s speed=1.19x
Last message repeated 15 times
[mp4 @ 0x5f9aab5d0800] pts has no valuee=1014.8kbits/s speed=1.18x
Last message repeated 9 times
[mp4 @ 0x5f9aab5d0800] pts has no valuee=1001.1kbits/s speed=1.17x
Last message repeated 7 times
[mp4 @ 0x5f9aab5d0800] pts has no value
Last message repeated 1 times
[out#0/mp4 @ 0x5f9aab5ad5c0] video:3546kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead : 0.120360%
size= 3550kB time=00:00:27.50 bitrate=1057.5kbits/s speed=1.18x

  


  Any suggestions on how to correct these errors ?

  


  Also any suggestions on how to make the video capture side more efficient ? Should I use a different codec ? (yuv instead of h264 ?) Would using UDP decrease overhead ? Can I improve video quality with the mode or hdr options ? What does denoise do ?

  


  With all the options available with these tools I think it's unlikely that I have a well thought out approach to capture and streaming. I'm hoping that people who are more familiar with this space might be able to provide some suggestions.

  


  Thank you !

  


  -tom