Re: [Audiodo-develop] New user

[Stefano]

Ciao Sayeed Ahmad,
 il giorno Fri, Feb 15, 2002 at 09:40:32PM -0000 hai scritto:
> I am here my brother
> I want to get the software which converts bad 34kbps songs to 128kbps by 
> changing its quality.

I'm sorry, but I've not understand what does it means "changing its quality".

If I'm not wrong you are speaking about MP3s.
If you aren't speaking about MP3, please let me better understand what you
are speaking about... ;-)

Actually  you can only DECODE an MP3 song to a format suitable by
cdrecord/cdrdao. Near future I'll implement encoding too (if someone is
interested), but you must to know that if you start from a bad encoded MP3
file and then you try to decode it, change its bitrate and then re-encode
it, you will loose something. (MP3 is a loss-speach compression algorithm)

If you are speaking about "bitrate" it's value is 32Kbps. This script does
already something like this:

$ file muppets.mp3
mupptes.mp3: MPEG 1.0 layer 3 audio stream data,  32 kBit/s, 44.1 kHz, jstereo

$ MP3do -s muppets.mp3
[....]

$ file 01-muppets.wav
01-mupptes.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, 
stereo 44100 Hz

$ play -V 01-muppets.wav
sox: Reading Wave file: Microsoft PCM format, 2 channels, 44100 samp/sec
sox:         176400 byte/sec, 4 block align, 16 bits/samp, 9971712 data
bytes
sox: Input file: using sample rate 44100
        size shorts, encoding signed (2's complement), 2 channels
sox: Input file: comment "01-mupptes.wav"

sox: Output file: using sample rate 44100
        size shorts, encoding signed (2's complement), 2 channels
sox: Output file: comment "01-mupptes.wav"


So a 32 kbBit/s file (but encoded at 44.1kHz) is "automagically" converted to
a 176400 Bit/s at 44100 samp/sec. This is done with sox's "rate" method, so
it is LOOSING quality (it add noise).
I don't know how to convert bitrate and/or sampling freq without loose
quality. 
If you know more about this please let me know! :-) 

Actually if you have a mp3 encoded with 32.0kHz of sampling frequence
anf bitrate at 32kbps my script does'nt does a well work.

Firstly because my script does'nt well calculate sampling rate and then for
another problem on stdout output. 
I've modified mpg321 and mpg123 plugins to correct these behavoirs. Check 
CVS repository for these two updated plugins.

Please tell me if something goes wrong with these two plugins.
 


Thank you for your interest! :-)

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