[App_rpt-users] Playing mp3 files in app_rpt

[Tim Sawyer]

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 at 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 at cs.com] 
> Sent: Tuesday, May 27, 2014 5:50 PM
> To: app_rpt-users at ohnosec.org
> Cc: kk6ecm at 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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