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• ffmpeg : improving MP4 to webm ogg conversions

  14 juillet 2017, par Randy

  (Edited to include some of the things I’ve tried)

  I’m a musician, and occasional web coder. I’ve been using video editing software (old version of Roxio Videowave from 2011) to build promotional videos from clips of some of my performances, and I’d like to put some of them on my own web pages in HTML5 video format. So that currently means I need MP4, WEBM, and OGG conversions. Fortunately the editing software churns out some very nice MP4 (H264) files, and has plenty of options for doing so. I purposely output the output size about 2X the likely display size, in hopes of offering more detail for better conversions. Specifically, the video output was AVC/H.264, 800 x 450, 30fps, variable bit rate, but with 600000 as a base line (that was the default for this setting anyway).

  Now I’m nowhere near expert at this stuff, and I probably left out some important data. But bottom line, the resulting MP4 looked very good. Unfortunately, to put it on my own web page means at least converting to WEBM and OGG formats. It would be nice if all browsers just supported MP4, but then there would be licensing fees, so conversions are needed. Sadly, I’ve been wasting days now trying to do this with ffmpeg. Its easy to do, its doing it WELL that is a mystery to me. Just letting ffmpeg work using its defaults (meaning I just specify an input and output file) results in pretty terrible video. But I’ve also tried most of the settings for better quality available, and the resulting conversions are nowhere near as good as youtube’s conversions.

  Based on the info about my original MP4 file, can someone suggest some better settings for ffmpeg conversions to WEBM and OGG ? Am I going about this all wrong ? The best I’ve done so far was with a string like this, which specified a high quality and a fairly robust bit rate...

  ffmpeg -i input-file.mp4 -c:v libvpx -crf 10 -b:v 1M -c:a libvorbis output-file.webm

  That was much better than the default settings, but still nowhere near the quality of YOUTUBE conversions. In my resulting WEBM video, you can pretty plainly see how the picture degrades, and will snap into focus every few seconds when a "key frame" comes up. These artifacts should not be so obvious. Thanks for any help.

• FFMPEG : Remove packets based on PTS/DTS

  9 mai 2018, par stevendesu

  I have a video which contains some audio packets beyond the end of the video data :

  $> ffprobe -show_packets video.mp4
  
  ...
  
  ...
  
  ...
  
  [PACKET]
  
  codec_type=video
  
  stream_index=0
  
  pts=5653648
  
  pts_time=235.568667
  
  dts=5653648
  
  dts_time=235.568667
  
  duration=1001
  
  duration_time=0.041708
  
  convergence_duration=N/A
  
  convergence_duration_time=N/A
  
  size=1030
  
  pos=25233684
  
  flags=__
  
  [/PACKET]
  
  [PACKET]
  
  codec_type=audio
  
  stream_index=1
  
  pts=11310080
  
  pts_time=235.626667
  
  dts=11310080
  
  dts_time=235.626667
  
  duration=1024
  
  duration_time=0.021333
  
  convergence_duration=N/A
  
  convergence_duration_time=N/A
  
  size=284
  
  pos=25234714
  
  flags=K_
  
  [/PACKET]
  
  [PACKET]
  
  codec_type=audio
  
  stream_index=1
  
  pts=11311104
  
  pts_time=235.648000
  
  dts=11311104
  
  dts_time=235.648000
  
  duration=1024
  
  duration_time=0.021333
  
  convergence_duration=N/A
  
  convergence_duration_time=N/A
  
  size=285
  
  pos=25234998
  
  flags=K_
  
  [/PACKET]
  
  [PACKET]
  
  codec_type=audio
  
  stream_index=1
  
  pts=11312128
  
  pts_time=235.669333
  
  dts=11312128
  
  dts_time=235.669333
  
  duration=992
  
  duration_time=0.020667
  
  convergence_duration=N/A
  
  convergence_duration_time=N/A
  
  size=290
  
  pos=25235283
  
  flags=K_
  
  [/PACKET]
  
  $>

  The last video packet in the video has a PTS time of 235.568667 and a duration of 0.041708 - meaning all video data ends at 235.610375. However there are audio packets beginning at 235.626667 and later.

  Is there an easy way to strip these audio packets from the file so that the audio and video end simultaneously ?

• How to read raw audio data using FFmpeg ?

  6 juin 2020, par Yousef Alaqra

  I'm trying to use this command to get the audio stream using UDP :

  



  ffmpeg -i udp://192.168.1.1:6980 -acodec copy


  



  I got an error when I execute it, which says :

  



  [udp @ 00000157a76b9a40] bind failed: Error number -10048 occurred
udp://192.168.1.1:6980: I/O error


  



  What's the meaning of this error ?

  



  Update :

  



  I was able to read raw audio data using FFmpeg and output into a wave file, using the following command :

  



  ffmpeg -f u16be -ar 44100 -ac 2 -i 'udp://127.0.0.1:1223' output.wav


  



  The problem now, Sine there is surrounding metadata in the network packets being received, it needs to be stripped out or it will result in noise.

  



  In C# I used Skip() to trim the first 28 bytes of the received packet, how would I achieve this using FFmpeg ?

  



  Update :

  



  I was able to read the raw bytes from UDP packets using by executing child process in node js :

  



  var http = require("http");
var port = 8888;
var host = "localhost";
var children = require("child_process");

http
  .createServer(function (req, res) {
    //ffmpeg -f s16le -ar 48000 -ac 2 -i 'udp://192.168.1.230:65535' -b:a 128k -f webm -
    var ffm = children.spawn(
      "ffmpeg",
      "-f s16le -ar 48000 -ac 2 -i udp://192.168.1.230:65535 -b:a 128k -f webm -".split(
        " "
      )
    );

    res.writeHead(200, { "Content-Type": "audio/webm" });
    ffm.stdout.on("data", (data) => {
      console.log(data);
      res.write(data);
    });
  })
  .listen(port, host);

console.log("Server running at http://" + host + ":" + port + "/");


  



  As you can see in the code sample above, I'm trying to pipe the output of the child process into the response, so I would be able to hear the audio in the browser.

  



  I'm receiving the data, after executing the child process, but the browser unable to play audio for some reason that I need to figure it out.

  



  Do you have an idea of what am I missing ?