[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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