Re: [expert] Streaming MP3-wav conversion problem
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 -- Linux: For the people, by the people. Want to buy your Pack or Services from MandrakeSoft? Go to http://www.mandrakestore.com
[expert] Streaming MP3-wav conversion problem
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
Re: [expert] Streaming MP3-wav conversion problem
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 Want to buy your Pack or Services from MandrakeSoft? Go to http://www.mandrakestore.com