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• ffmpeg http slower than downloading first

  9 août 2017, par Krylic

  When trying to get a clip of a video from a remote source
  Input source :
  

  ffprobe version 3.3.2-static http://johnvansickle.com/ffmpeg/  Copyright (c) 2007-2017 the FFmpeg developers
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'http://website.com/video.mp4':
 Metadata:
   major_brand     : mp42
   minor_version   : 0
   compatible_brands: mp42mp41
   creation_time   : 2017-07-13T15:44:58.000000Z
 Duration: 00:57:32.42, start: 0.000000, bitrate: 1939 kb/s
   Stream #0:0(eng): Video: h264 (Main) (avc1 / 0x31637661), yuv420p(tv), 720x480 [SAR 10:11 DAR 15:11], 1745 kb/s, 29.97 fps, 29.97 tbr, 29970 tbn, 59.94 tbc (default)
   Metadata:
     creation_time   : 2017-07-13T15:44:58.000000Z
     handler_name    : Mainconcept MP4 Video Media Handler
     encoder         : AVC Coding
   Stream #0:1(eng): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 189 kb/s (default)
   Metadata:
     creation_time   : 2017-07-13T15:44:58.000000Z
     handler_name    : Mainconcept MP4 Sound Media Handler

  Current command :
  ffmpeg version 3.3.2-static
  ffmpeg.linux -threads 2 -y -ss 3273 -i "http://website.com/video.mp4" -an -movflags +faststart -preset veryfast -codec copy /outputfolder/trimmed_video.mp4
  This 5m35.102s to create a 45mb 2min file.

  If I download the file using wget it takes 28s and using ffmpeg only takes 0.243s

  If I add -vn OR -an to the output portion of the command it completes the trim in about 2.101s. Meaning it’s faster to download the two stream and merge them myself.

  Can anyone explain this behaviour and why my first command takes so long when on a lot of other video files it’s very fast ?

• lavc : Use hardware config information in ff_get_format()

  26 octobre 2017, par Mark Thompson

  lavc : Use hardware config information in ff_get_format()
  This removes the dependency that hardware pixel formats previously had on
  
  AVHWAccel instances, meaning only those which actually do something need
  
  exist after this patch.
  Also updates avcodec_default_get_format() to be able to choose hardware
  
  formats if either a matching device has been supplied or no additional
  
  external configuration is required, and avcodec_get_hw_frames_parameters()
  
  to use the hardware config rather than searching the old hwaccel list.
  The FF_CODEC_CAP_HWACCEL_REQUIRE_CLASS mechanism is deleted because it
  
  no longer does anything (the codec already contains the pointers to the
  
  matching hwaccels).
  

  • [DH] libavcodec/avcodec.h
  • [DH] libavcodec/cuviddec.c
  • [DH] libavcodec/decode.c
  • [DH] libavcodec/internal.h

• 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 ?