[Asterisk-Users] Why does * only work with an ancient mpg123?

Christopher Dobbs crdobbs at eracew.net
Mon Dec 20 15:06:38 MST 2004


I have:

High Performance MPEG 1.0/2.0/2.5 Audio Player for Layer 1, 2 and 3.
Version 0.59s-mh4 (2000/Oct/27). Written and copyrights by Michael Hipp.

It works verry Well.

--
Christopher Dobbs


Remco Barende wrote:

> On Mon, 20 Dec 2004, Eric Wieling aka ManxPower wrote:
>
>> Remco Barende wrote:
>>
>>> Hi list!
>>>
>>> Just wondering, why is * sticking with an mpg123 version from the 
>>> stoneage?
>>>
>>> Gentoo comes with 0.59s-r8 and this version doesn't even start.
>>>
>>> Ik know I could forcibly unmerge mpg123 and install the old version 
>>> but I guess some day newer versions will have to be supported?
>>
>>
>> Asterisk sets the following mpg123 options:
>>
>> "mpg123 -q -s --mono -r 8000 -b 2048 -f 4096"
>>
>> -q, --quiet  Quiet.  Suppress diagnostic messages.
>>
>> -s, --stdout  The decoded  audio  samples  are  written  to  standard 
>> output, instead  of  playing them through the audio device.  This option
>> must be used if your audio hardware is not supported by  mpg123.
>> The  output format is raw (headerless) linear PCM audio data, 16
>> bit, stereo, host byte order.
>>
>> -r rate, --rate rate  Set sample rate (default: automatic).  You may 
>> want  to change this  if  you  need  a  constant  bitrate independed 
>> of the mpeg stream rate. mpg123 automagically converts the rate. You 
>> should then combine this with --stereo or --mono.
>>
>> -b size, --buffer size  Use an audio output buffer of size Kbytes.  
>> This  is useful  to bypass  short periods of heavy system activity, 
>> which would normally cause the audio output  to  be  interrupted.   
>> You should specify  a buffer size of at least 1024 (i.e. 1 Mb, which 
>> equals about 6 seconds of audio data) or more; less than about 300 
>> does not make  much  sense.  The default is 0, which turns buffering 
>> off.
>>
>> -f factor, --scale factor  Change scale factor (default: 32768).
>>
>> Pretty much any program that accepts these options to generate raw 
>> (headerless) linear PCM audio data, 16 bit, mono, host byte order, at 
>> 8khz to stdout will work.  At this time the only one that does this 
>> that I know is mpg123 0.59r
>
>
> Thanks! But when I look at the output of mpg123 0.59s-r8 all these 
> commandline switches are still supported, why it it only the old 
> version that is supported, not the newer ones?
>
>
> asterisk # mpg123
> High Performance MPEG 1.0/2.0/2.5 Audio Player for Layer 1, 2 and 3.
> Version 0.59s-r8 (2000/Oct/27). Written and copyrights by Michael Hipp.
> Uses code from various people. See 'README' for more!
> THIS SOFTWARE COMES WITH ABSOLUTELY NO WARRANTY! USE AT YOUR OWN RISK!
>
> usage: mpg123 [option(s)] [file(s) | URL(s) | -]
> supported options [defaults in brackets]:
>    -v    increase verbosity level       -q    quiet (don't print title)
>    -t    testmode (no output)           -s    write to stdout
>    -w <filename> write Output as WAV file
>    -k n  skip first n frames [0]        -n n  decode only n frames [all]
>    -c    check range violations         -y    DISABLE resync on errors
>    -b n  output buffer: n Kbytes [0]    -f n  change scalefactor [32768]
>    -r n  set/force samplerate [auto]    -g n  set audio hardware 
> output gain
>    -os,-ol,-oh  output to built-in speaker,line-out connector,headphones
>                                         -a d  set audio device
>    -2    downsample 1:2 (22 kHz)        -4    downsample 1:4 (11 kHz)
>    -d n  play every n'th frame only     -h n  play every frame n times
>    -0    decode channel 0 (left) only   -1    decode channel 1 (right) 
> only
>    -m    mix both channels (mono)       -p p  use HTTP proxy p 
> [$HTTP_PROXY]
>    -@ f  read filenames/URLs from f
>    -z    shuffle play (with wildcards)  -Z    random play
>    -u a  HTTP authentication string     -E f  Equalizer, data from file
> See the manpage mpg123(1) or call mpg123 with --longhelp for more 
> information.
>
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