It’s real simple to convert .mp3 to a .wav file: sox newsline.mp3 -r 8000 temp.wav The trick is installing SOX with LAME. I found the attached procedure worked for me. -- Tim :wq On May 27, 2014, at 9:15 PM, Bob <kk6ecm@gmail.com> wrote:
You make a good point, and I should have said more about my application.
My issue just happens to be with the ARRL Audio News which uses mp3 format, and the desire to play it once per week via the repeater.
I am experimenting with both the add-on (thanks for the head-up regarding the output level), and a script to convert the mp3 to ulaw format.
Thanks, r/Bob
-----Original Message----- From: Ken [mailto:ke2n@cs.com] Sent: Tuesday, May 27, 2014 5:50 PM To: app_rpt-users@ohnosec.org Cc: kk6ecm@gmail.com Subject: Re: [App_rpt-users] Playing mp3 files in app_rpt
Asterisk/app_rpt will play MP3 files alright, using the add-on, although, you may need to change the output level.
In format_mp3.c
#define OUTSCALE 32768
Seems to be about right (I think the original was 8192 but YMMV as they say)
I used mp3 for bulletins for a while - I even bought a copy of Adobe Audition CS5.5 (now there is version 6) so I could make "real" MP3 files. MP3's made by Audacity sound *really* weird.
MP3 files contain a frequency spectrum that depends on the sampling rate of the recording. This is a separate rate from the Bit Rate of the MP3 playback. If you reduce the playback bitrate of an MP3 file, you may still have frequency components up through perhaps 12 or 15 kHz, you just get more distortion and artifacts. All those frequency components above about 3.6 KHz will be ignored by Asterisk, which runs its sound system at 8 k-samples/sec. This may produce some strange effects since you are playing back stuff that Asterisk can't "hear". The distortion below 3.6 kHz comes through of course...
To get a file like ARRL Audio News (44 kHz sample rate, in stereo, with an MP3 bit rate of 192k) to play properly on asterisk, you need some software that can re-sample the original file down to 8 k sampling rate (and 8k bit rate). That way the playback spectrum of the MP3 will have components only up through about 3.4 kHz, which is quite compatible with asterisk. Adobe Audition 5.5 (and later) can do this.
Audacity will also re-sample (at the "Project Rate"), but it is best used to make ulaw files, not MP3's; but it's free.
If the original recording was frequency-limited then no re-sampling of the recording is needed. For example, my NWS weather podcasts are a computer voice designed for telephone bandwidth, but distributed as MP3's. Those MP3's played through asterisk sound just like the original played on any MP3 player.
Lots of fun
73 Ken
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