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• Trim off N bytes from audio file using SoX / FFmpeg etc, on Windows ?

  17 novembre 2020, par Rinaldo Jonathan

  My team accidentally on purpose clicked NO when Audacity asked to save the recording. So I left with bunch of *.au files, after using recovery program.

  


  Some of them did have header and still open-able with audacity itself (example : this one), and some other are just complete nonsense, sometimes having the header filled with text from random javascript or HTML code (like this one). Probably hard disk half overwritten with browser cache ? I don't know. And at this point, I almost don't care.

  


  The audacity is on default settings, with sample rate 44100Hz. I can open a-113.au using audacity, from standard open files. I also was able to achieve open files using "Open RAW files" from Audacity, using this settings :

  


  

  


  so I assume header takes 12384 bytes.

  


  Now, how do I trim 12384 bytes from the file when opened as RAW, with either SoX or ffmpeg ? because if I open it as RAW with 0 offset (default settings), it will add the header as a noise.

  


  Current ffmpeg command I used : ffmpeg.exe  -f f32le -ar 44.1k -ac 1 -i  source destination
  
Current sox command I used : sox -t raw --endian little --rate 44100 -b 1 -b 32 --encoding floating-point  %%A "converted/%%~nxA.wav"
  
Both still have header as a noise in the converted files.

  


• Fixing "RTP : dropping old packet received too late" in FFMPEG

  25 juin 2014, par user985030

  I have been using FFMPEG 0.6 for years with no problems and recently ported much of my code to 2.2 ; however, there is still a problem that I cannot resolve after fixing many of the deprecated functions. I am generating simulated video and then using RTSP to unicast this generated stream. The problem is that when I change the height and width of my video data, I basically recreate a new stream to send the subscribed client. The algorithm I used to do this in 0.6 worked like a charm, never had any problems. Now that I have upgraded, I get "RTP : dropping old packet received too late" as soon as I change my frame size. I think I have been dropping packets all along, but the new code is causing connection issues for me. The packets being dropped in the past were negligible and I really didn’t care if I missed them as long as the stream eventually corrected itself. Is there a flag that I can set to not drop these packets ? Or at least recover more quickly ? I believe that this has something to do with receiving packets out of order. There is a section of code that does a diff in FFMPEG in the rtpdec.c file in the function rtp_parts_one_packet. The only reference I found similar to my issue is here :

  http://en.it-usenet.org/thread/16949/6708/#post6707. Any tips would be greatly appreciated. In the meantime, I am just going to patch the FFMPEG code to not do the following check :

  if (diff < 0) {
  
  
  
  /* Packet older than the previously emitted one, drop */
  
  
  
  av_log(s->st ? s->st->codec : NULL,        AV_LOG_WARNING,
  
  
  
  "RTP: dropping old packet received too late\n");
  
  
  
  return -1;
  
  
  
  }

  By commenting out the above code, I am able to run my streaming application like I used to, but I have a feeling that I am not doing something correctly but I am not sure what it is. Thanks for any advice !

• macos - Batch create 'samples' with multiple cuts from videos [closed]

  17 février, par Thiago

  I'm on macOS, and have ffmpeg and python installed through homebrew. Bash solutions are also welcome - though I have no experience with bash

  


  I have folders with many videos, most (if not all) in either mp4 or mkv. I want to generate quick samples for each video, and each sample should have multiples slices/cuts from the original video.

  


  I don't care if the resulting encode, resolution and bitrate are the same as the original, or if it's fixed for example : mp4 at 720. Whatever is easier to write on the script or faster do execute.

  


  Example result : folder has 10 videos of different duration. Script will result in 10 videos titled "ORIGINALTITLE-sample.EXT". Each sample will be 2 minutes long, and it'll be made of 24 cuts - 5 seconds long each - from the original video.

  


  The cuts should be approximately well distributed during the video, but doesn't need to be precise.

  


  Edit

  


  someone on Reddit suggested the script below, but the result has some issues, like images blinking in and out (see it here https://youtu.be/FZC3aIvugpI). Maybe it's related to errors like this I saw ? [hevc @ 0x11c631a30] Could not find ref with POC -43

  


  I was also not able to change the 1-second duration of each clip for something longer, and would still need to make this loop on every video in the folder.

  


  #!/bin/bash
f="original.mp4"
dur=$(ffprobe -v 16 -show_entries format=duration -of csv=p=0 "$f")
cnt=$(echo "scale=0; ${dur} * 0.95 / 8" | bc -l)
echo $dur $cnt

ffmpeg -i "$f" -c copy -f segment -segment_time $cnt -reset_timestamps 1 "/tmp/out_%03d.${f##*.}" -y -hide_banner

echo "#list">/tmp/1.txt
for g in /tmp/out_*; do
    echo "file $g" >> /tmp/1.txt
    echo "outpoint 1" >> /tmp/1.txt
done

o="/tmp/out.${f##*.}"
ffmpeg -f concat -safe 0 -i /tmp/1.txt -c copy "$o" -y -v error -stats test.mp4