Re: [expert] Streaming MP3-wav conversion problem

2003-06-09 Thread Rob Blomquist
On Sunday 08 June 2003 09:20 pm, Rob Blomquist wrote:
 I am trying to convert a streaming MP3 file to a wav file for editing in
 Audacity. The mp3 stream was caught by XMMS with the MPEG Layer 1/2/3
 plugin. XMMS and Noatun play it prefectly, so I bet that I can convert it
 to a wav file somehow.

 I have tried lame, bladeenc, and mpg123 to try to either output it as a
 wav, raw, or standard output (so I could pipe it back to an encoder to
 rebuild the mp3 into something converable).

 Here is the output from lame showing the problem it is having with the
 bitstream, variable frequency, and the number of channels.

Once again, I find my own answers, after asking. I guess I just have to look 
for the answers harder before asking.

I upgraded to Audacity 1.1.3 to find the crashing problem gone away, now was 
the new one. My mp3 was 128Mb, and Audacity filled my /tmp with 3Gb of au 
files! 

I found mp3splt, and using XMMS to get times, I am able to chop the files into 
managable chunks, and it can handle the variable bit rate sampling in the 
file, creating nice new ones that are much smaller, while preserving the 
source file.

Now, only if I can keep Audacity from crashing. I guess its time to chop some 
more.

Rob

-- 

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[expert] Streaming MP3-wav conversion problem

2003-06-08 Thread Rob Blomquist
I am trying to convert a streaming MP3 file to a wav file for editing in 
Audacity. The mp3 stream was caught by XMMS with the MPEG Layer 1/2/3 plugin. 
XMMS and Noatun play it prefectly, so I bet that I can convert it to a wav 
file somehow.

I have tried lame, bladeenc, and mpg123 to try to either output it as a wav, 
raw, or standard output (so I could pipe it back to an encoder to rebuild the 
mp3 into something converable).

Here is the output from lame showing the problem it is having with the 
bitstream, variable frequency, and the number of channels. 

Any ideas?

$ lame --decode incoming/KEXPraw.mp3
input:  incoming/KEXPraw.mp3  (48 kHz, 2 channels, MPEG-1 Layer III)
output: incoming/KEXPraw.mp3.wav  (16 bit, Microsoft WAVE)
skipping initial 1105 samples (encoder+decoder delay)
Frame# 1/289994 452 kbps bitstream problem: resyncing...
Frame# 2/289994 356 kbps  L  R   bitstream problem: resyncing...
Frame# 3/289994 452 kbps bitstream problem: resyncing...
Frame# 4/289994 388 kbps bitstream problem: resyncing...
Frame# 5/289994 448 kbps bitstream problem: resyncing...
Frame# 6/289994 128 kbps  LMSR  ibitstream problem: resyncing...
Frame# 7/289994 416 kbps bitstream problem: resyncing...
Error: sample frequency has changed in MP3 file - not supported
Frame# 8/289994 160 kbps bitstream problem: resyncing...
.
.
.
Frame#   137/289994 193 kbps bitstream problem: resyncing...
Error: number of channels has changed in MP3 file - not supported
Error: sample frequency has changed in MP3 file - not supported
Frame#   138/289994 299 kbps fatal error.  MAXFRAMESIZE not large 
enough.
Segmentation fault

-- 

Linux: For the people, by the people.

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Go to http://www.mandrakestore.com


Re: [expert] Streaming MP3-wav conversion problem

2003-06-08 Thread Michael Holt
This is a script I use for batch conversion; works on one or more mp3's. 
If this doesn't work, maybe there is a problem with the xmms plugin you
mentioned?

#!/bin/bash
# allmp3wav
for i in *.mp3; do
 echo $i
  tgt=$(echo $i | sed -e s/mp3/wav/)
   mpg123 -b 1 -s -r 44100 $i | sox -t raw -r 44100 -s
-w -c2 - $tgt
done

mike


 I am trying to convert a streaming MP3 file to a wav file for editing in
 Audacity. The mp3 stream was caught by XMMS with the MPEG Layer 1/2/3
 plugin.
 XMMS and Noatun play it prefectly, so I bet that I can convert it to a wav
 file somehow.

 I have tried lame, bladeenc, and mpg123 to try to either output it as a
 wav,
 raw, or standard output (so I could pipe it back to an encoder to rebuild
 the
 mp3 into something converable).

 Here is the output from lame showing the problem it is having with the
 bitstream, variable frequency, and the number of channels.

 Any ideas?

 $ lame --decode incoming/KEXPraw.mp3
 input:  incoming/KEXPraw.mp3  (48 kHz, 2 channels, MPEG-1 Layer III)
 output: incoming/KEXPraw.mp3.wav  (16 bit, Microsoft WAVE)
 skipping initial 1105 samples (encoder+decoder delay)
 Frame# 1/289994 452 kbps bitstream problem: resyncing...
 Frame# 2/289994 356 kbps  L  R   bitstream problem: resyncing...
 Frame# 3/289994 452 kbps bitstream problem: resyncing...
 Frame# 4/289994 388 kbps bitstream problem: resyncing...
 Frame# 5/289994 448 kbps bitstream problem: resyncing...
 Frame# 6/289994 128 kbps  LMSR  ibitstream problem: resyncing...
 Frame# 7/289994 416 kbps bitstream problem: resyncing...
 Error: sample frequency has changed in MP3 file - not supported
 Frame# 8/289994 160 kbps bitstream problem: resyncing...
 .
 .
 .
 Frame#   137/289994 193 kbps bitstream problem: resyncing...
 Error: number of channels has changed in MP3 file - not supported
 Error: sample frequency has changed in MP3 file - not supported
 Frame#   138/289994 299 kbps fatal error.  MAXFRAMESIZE not large
 enough.
 Segmentation fault

 --

 Linux: For the people, by the people.

 Want to buy your Pack or Services from MandrakeSoft?
 Go to http://www.mandrakestore.com



-- 
Michael Holt
Snohomish, WA  (o_
[EMAIL PROTECTED](o_  (o_  //\
www.holt-tech.net(/)_ (/)_ V_/_www.mandrake.com




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