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• 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.

  


• pydub unable to find the selected files

  4 février 2021, par big_Z_0909

  my code is as below

  


  import pydub
from pathlib import Path
import os, sys

pydub.AudioSegment.converter = r"D\\Anaconda3\\Scripts\\ffmpeg-2021-02-02-git-2367affc2c-essentials_build\\bin\\ffmpeg.exe"
pydub.AudioSegment.ffprobe = r"D\\Anaconda3\\Scripts\\ffmpeg-2021-02-02-git-2367affc2c-essentials_build\\bin\\ffprobe.exe"

my_file = "D:\\Anaconda3\\Scripts\\outcall\\luyinwenjian\\20210203.mp3"
song = pydub.AudioSegment.from_mp3(my_file)


  


  i downloaded the essential windows eddition of ffmpeg from the official website. i still cannot load the mp3 file from the document, got the FileNotFoundError : [WinError 2]. In the warning, it said that it could find ffprobe and ffmpeg, but in fact the path is correct in the code. Does anyone point out what exactly i went wrong here ?

  


• FFMPEG Audio Optimisation with speex

  2 février 2021, par ThinkAdvantage

  I am trying to cleanup the Audio of my Paragliding Videos down to my own voice mostly by using ffmpeg. I stumbled upon speex codec and as far as I understood it I should be able to encode my Audio Stream with Libspeex and this should cleanup the audio stream down to my voice. I am working on a current windows 10 and downloaded a precompiled Windows ffmpeg build which includes libspeex. (i could verify that) Now i tried to input my mp4 and output it with libspeex encoded audio but i always get :

  


  


  Could not find tag for codec speex in stream #1, codec not currently supported in container
Could not write header for output file #0 (incorrect codec parameters ?) : Invalid argument
Error initializing output stream 0:1 —

  


  


  ffmpeg -i D :\Cyclevision\2021.01.24\20210124-short.mp4 -c:v copy -acodec libspeex -vad 1 D :\Cyclevision\2021.01.24\20210124-shortSPEEX.mp4

  


  I also tried outputting only the Audio to mp4, mp3, aac but everything failed.

  


  My question now - how can I use libspeex properly to re-encode my Audio stream of an MP4 ?

  


  Is it even a good idea to use libspeex or is it only for Audio and not Audio/Video combinations ?
Thanks in Advance for your Answers.

  


  P.S : I am no proper Programmer - just an IT Infrastructure System engineer trying to use FFMPEG with Powershell.