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• Why the converted videos file size is greater than original file size ?

  6 décembre 2011, par svk

  I am using ffmpeg to convert the videos into mp4.Its working fine and its playing with high quality.No problem.But the worst case is I uploaded 14Mb file and after converting it goes to 30 Mb file.I am using the following the script to convert

  exec("ffmpeg -i videowithaudio.flv -vcodec libx264 -vpre hq -vpre ipod640 -b 250k -bt 50k -acodec libfaac -ab 56k -ac 2 -s 480x320 video_out_file.mp4 > output1.txt 2> apperror1.txt");   //webkit compatible

  I am using PHP for executing this command.Could you please help me how to reduce the file size from this 30Mb (nearly to uploaded file size is ok) with same quality.

• correct way to download a mp4 [Audio only] file as a mp3 file via php

  3 février 2018, par Kapil

  I am asking this question because i found it impossible as far my knowledge stands. However i believe here on stackoverflow a lot of genius persons visits so maybe someone can give a good advice / trick.

  My problem is, I am downloading a audio/mp4 file of youtube hosted on googlevideo.com’s server.

  My PHP code for this purpose :
  $mp3path is url of video

  header('Content-Description: File Transfer');
  
  if(strpos($mp3path, "https://") === false) {
  
      header('Content-length: ' . size($mp3path)); //size is custom function
  
  } 
  
      header("Content-Type: audio/MP4A-LATM, audio/MP4A, audio/m4a, audio/mp4, audio/mp4a, audio/mp4-audio, audio/mpeg"); 
  
      header('Content-Type: application/force-download');
  
      header("Accept-Ranges: bytes");
  
      header('Content-Transfer-Encoding: binary'); 
  
      header('Content-disposition: attachment; filename="'.$title.'.mp3"'); 
  
      header('Connection: Keep-Alive');
  
      header('Expires: 0');
  
      header('Cache-Control: must-revalidate, post-check=0, pre-check=0');
  
      header('Last-Modified: ' . gmdate('D, d M Y H:i:s', time()) . ' GMT');
  
      header("Cache-Control: private", false);
  
      header('Pragma: no-cache');
  
      readfile($mp3path);
  
      exit;

  I can download this audio file but few mp3 players are not able to play it when i did some research on by using mp3val [mp3val.sourceforge.net]. I found this file do not contain sample-rate, bit-rate or some other required codecs inside the file.
  Error received from mp3val : Unknown file format

  I know one possible way of doing this which is ffmpeg but i am looking for a less time consuming option of doing this because first ffmpeg download the whole file and then convert it to mp3 and save it on server. It takes a lot of time.

  I am looking for a easiest solution in which, i dont want to save the whole file on my server. I want to call this file from remote server and want to add sample rate, bit-rate in the file and then just somehow with php i want to start download on browser.

  All i mean to say i need a faster solution. I have seen 2 yt to mp3 converter, They are doing the same thing, giving instant download. i dont know how ?

  Please if you think its a stupid question or not possible then dont report to stackoverflow. I am trying to figure it out that’s why i am asking this question here. I hope you’ll understand my curiosity.

  Thanks,

• FFMPEG decode from RTP dump file into mp3 file

  5 février 2021, par pingvincible

  I'm trying to save RTP stream into mp3 file. I use this command :

  


  ffmpeg -loglevel debug -protocol_whitelist file -f rtp -i microphone.rtpdump -f mp3 microphone.mp3


  


  I get this result :

  


  user@pc:~/$ ffmpeg-amrnb -loglevel debug -protocol_whitelist file -f rtp -i microphone.rtpdump -f mp3 microphone.mp3
ffmpeg version N-100958-g4f3d8cb554 Copyright (c) 2000-2021 the FFmpeg developers
  built with gcc 9 (Ubuntu 9.3.0-17ubuntu1~20.04)
  configuration: --enable-gpl --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-librtmp --enable-libtheora --enable-libvorbis --enable-nonfree --enable-version3
  libavutil      56. 64.100 / 56. 64.100
  libavcodec     58.120.100 / 58.120.100
  libavformat    58. 65.101 / 58. 65.101
  libavdevice    58. 11.103 / 58. 11.103
  libavfilter     7.102.100 /  7.102.100
  libswscale      5.  8.100 /  5.  8.100
  libswresample   3.  8.100 /  3.  8.100
  libpostproc    55.  8.100 / 55.  8.100
Splitting the commandline.
Reading option '-loglevel' ... matched as option 'loglevel' (set logging level) with argument 'debug'.
Reading option '-protocol_whitelist' ... matched as AVOption 'protocol_whitelist' with argument 'file'.
Reading option '-f' ... matched as option 'f' (force format) with argument 'rtp'.
Reading option '-i' ... matched as input url with argument 'microphone.rtpdump'.
Reading option '-f' ... matched as option 'f' (force format) with argument 'mp3'.
Reading option 'microphone.mp3' ... matched as output url.
Finished splitting the commandline.
Parsing a group of options: global .
Applying option loglevel (set logging level) with argument debug.
Successfully parsed a group of options.
Parsing a group of options: input url microphone.rtpdump.
Applying option f (force format) with argument rtp.
Successfully parsed a group of options.
Opening an input file: microphone.rtpdump.
[rtp @ 0x556947200580] Unable to receive RTP payload type 97 without an SDP file describing it
microphone.rtpdump: Invalid data found when processing input


  


  It looks like that microphone.rtpdump file format is correct as ffmpeg can find RTP payload type 97. The problem is that I don't understand how to use SDP file in this situation.

  


  I have an SDP file for this payload type which I use, when I send data over network. It looks like this :

  


  v=0
o=- 0 0 IN IP4 127.0.0.1
s=No Name
c=IN IP4 127.0.0.1
t=0 0
a=tool:libavformat 58.65.101
m=audio 1234 RTP/AVP 97
b=AS:12
a=rtpmap:97 AMR/8000/1
a=fmtp:97 octet-align=1


  


  And now I want to decode RTP stream from file, not by receiving it from network.

  


  How to adapt my SDP file to read RTP stream from file ?

  


  UPDATE : My rtpdump file is not a real rtpdump file format. It is just payloads from UDP packets written into file without any additional headers.