Convert files from the CLI
You just recorded a fabulous
audio file to use as you main voice menu. Then you realize that
Asterisk does not use WAV format audio for the Playback or Background
applications. So what do you do? How can you convert your WAV files
into GSM files that still have good sound quality? (This is partially false, Asterisk can play anything it has a format and codec for, including some wav files. See below.)
Note the differences!
gsm: raw gsm encoding, good for VoIP
wav: MS wav format, 16 bit linear
WAV: MS wav format, gsm encoded (wav49)
Converting to sln format
Starting from Asterisk 1.2.0, the .sln (SLINEAR) format seems to be the preferred format.
To convert wav file to sln, use the following command:
sox foo-in.wav -t raw -r 8000 -s -w -c 1 foo-out.sln
Note that sox v14.3.0 and above (installed in Ubuntu 9.10), the -w option has changed to -2
sox foo-in.wav -t raw -r 8000 -s -2 -c 1 foo-out.sln
If
you have a directory full of .wav files to convert, try this command.
It uses sed to automatically rename the files with the .sln extension
(assuming incoming wav files at a sample rate other than 8khz.)
for a in *.wav; do sox "$a" -t raw -r 8000 -s -w -c 1 `echo $a|sed "s/.wav/.sln/"` resample -ql; done
Converting your WAV files to good GSM files is easier than you might think if you have the program Sox
installed (on Debian systems the libsox-fmt-gsm package is required in
addition to sox). From the shell prompt, enter this command:
sox foo.wav -r 8000 foo.gsm resample -ql
and
hit the <ENTER> key. Note that the sox option '-ql' (lower case
L) modifies the resample option. It is not a number one (1). In a few
moments you will have a new GSM format file in the same directory as the
original WAV file. In this example "foo.wav" is your main voice menu
audio file in WAV format, and "foo.gsm" is the same file converted to
GSM format. If you wanted to, you could use "main-voice-menu.gsm" as
the name in place of "foo.gsm": what matters here is the second file
name you use in this command ends in ".gsm".
If your WAV file was in stereo, add the -c1 option to convert to mono, or the output will sound very strange.
sox foo.wav -r 8000 -c1 foo.gsm resample -ql
You may get better results if you record your WAV file in 16 bit 8000 Hz mono and then run
sox foo.wav foo.gsm
If you have multiple WAV files in one directory and you want to convert them all, use this command:
for a in *.wav; do sox "$a" -r 8000 -c1 "`echo $a|sed -e s/wav//`gsm" resample -ql; done
You can also put a bash script in /usr/bin and name it wav-gsm-convert. The content can be like this
- !/bin/bash
s=`echo $1| sed -e's/\.wav//'|xargs -i{} echo {}.gsm`
sox -t wav $1 -r 8000 -c1 -t gsm $s resample -ql
Next, move your new foo.gsm file to the directory:
/var/lib/asterisk/sounds
Now
you can easily use the applications Playback and Background in your
extensions.conf file to play your fabulous main voice menu. For
example:
exten => s,1,Background(foo)
or
exten => s,1,Background(main-voice-menu)
or
exten => s,1,Playback(foo)
or
exten => s,1,Playback(main-voice-menu)
Playing .sln files from the command line
You can play sln files using sox from the command line (play is part of sox):
play -t raw -r 8000 -s -w -c 1 file.sln
Using WAV files
Asterisk
has codecs for wav (pcm), gsm, g729, g726, and wav49, all of which can
be used for Playback and Background. However, Asterisk does not
understand ADPCM WAV files. To convert your WAV files to a format which
Asterisk can understand, use the following command:
sox foo-in.wav -r 8000 -c 1 -s -w foo-out.wav resample -ql
Note that sox v14.3.0 and above (installed in Ubuntu 9.10), resample is no longer used, remix is used or leave it as:
sox foo-in.wav -r 8k -c 1 -s -w foo-out.wav
All so if your using sox v14.3.foo and above and you are getting check_header errors with play back try this:
sox foo-in.wav -r 8k -c 1 -e gsm foo-out.wav
This also works converting mp3, just make sure you have libsox-fmt-mp3 installed.
Normalizing volume and reducing volume fluxuations
Using
the sox command's "compand" filter, you can reduce or eliminate
flutters in volume level, and you can normalize the volume of your sound
files. This is called dynamic range compression. The effect is
desirable for a PBX system where changes in volume, or voices that are
too quiet to understand would be considered unprofessional. It is
important to perform the compand effect BEFORE you resample it as to
preserve as much quality as possible.
An example command to perform some appropriate dynamic range compression and normalization is shown below:
sox
"foo-in.wav" -r 8000 -c1 "foo-out.gsm" lowpass 4000 compand 0.02,0.05
-60,-60,-30,-10,-20,-8,-5,-8,-2,-8 -8 -7 0.05 resample -ql
In this example, foo-in.wav is a 16-bit mono 44khz uncompressed pcm WAV file.
Converting to a CD writable format
So,
you've decided to do your call recording in GSM format as you don't
care about quality and you don't want to stuff your disks full, but how
do you write that file to an audio CD to send to somebody who wants to
listen to the call?
sox infile.gsm -r 44100 -a outfile.wav
