6-channel(5.1) wav file to 5.1 mp3 sorround file encoder

English Multimedia
#1

Hi,
I am trying to encode 6- channel wav files to 5.1 mp3 sorround format.The only free encoder which i could get my hands on is Fraunhofer IIS MP# Sorround Command line encoder v1.5 .

To encode the six channel wav file, I type the following command:

$ mp3sEncoder.exe -if 6_channel.wav -of 6_channel.mp3 -br 320000

It gives the following error: Unsupported input sampling rate <96000Hz>
Allowed sampling rate for mono and stereo signals:8000 - 48000Hz
Allowed sampling rate for 5.1 signals: 44100 or 48000 Hz.

Can anyone help me encode the given file to the desired format?

While going through certain audio forums I discovered something which said that one could first convert 6-channel wav file to 6 mono files using a software called “transcode” and then use a mux file to get the 5.1 mp3 sorround. Can someone throw some light on this? What is a mux file? Will converting a 6-channel file to 6 mono wav files affect audio quality?

Thanks!

Regards,
modmath.

#2

you need to resample your WAV file (which has a frequency of 96 kHz) to something that mp3 supports like 48 kHz (48000 Hz). You can do that with Audacity, for example. Then the mp3 encoder will accept it

as for the second method. You deconstruct the 6 channels WAV file into 6 separate Mono files (each carrying one specific channel), then encode them to mp3 and use a program to mux them (joining together) into one file. That’s a bit of tedious process so if you don’t know what you’re doing and how to extract the 6 channels into 6 separate Mono files and then mux together these 6 files in correct channels order, I would not suggest it to you

Also, if I was you, I’d use AAC to encode to multi-channel sound files

#3

Thanks a lot!

#4

Hey microchip8,

  1. Audacity does the downsampling but while storing back the file(export option in file menu) it downmixes 5.1 to stereo. Or it stores all the six channels as separate files. Is there a way out of this?

  2. I tried using sox for downsampling but it ends up “corrupting” the WAV file. The following is a record of what happens:

$ sox clip1_parametric_compressedsum.wav -r 48000 clip1_48_c.wav
sox wav: wave header missing FmtExt chunk
sox sox: resample clipped 12 samples; decrease volume?
sox sox: clip1_parametric_compressedsum.wav: input clipped 39 samples
sox sox: clip1_48_c.wav: output clipped 8 samples; decrease volume?

When I try to run clip1_48_c.wav using foobar, it gives give an error “… .wav file has either a format which is not recognised or is corrupt” or sth to that effect.

#5

Easies way I can think of now (since I just woke up) is to use mplayer

mplayer file.wav -channels 6 -af resample=48000:0:2 -ao pcm:fast:file=output.wav