INTERACT FORUM
More => Old Versions => Media Center 11 (Development Ended) => Topic started by: Shadowen on September 03, 2005, 05:05:24 pm
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Hello,
I am having problems recording from the Line-in input directly into a mono 8kb/s file using the MP3 Encoder in MC11.
I have gone into Encoding->Sound Recorder settings and specified a custom command-line argument with the MP3 Encoder to be -m m -b 8.
According to the documentation (http://lame.sourceforge.net/USAGE) for the LAME encoder, the -m m option forces the mode to be monaural and the -b 8 option forces a constant bitrate of 8.
However, MC does not appear to be adhering to these command-line options during the encoding process. Could there be an issue with MC in this regard?
I am confident that these options are correct, because I have made the exact same settings under the Encoder->File Conversion settings and they work as expected when converting files.
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I never have gotten Lame to record in mono correctly
I don't know if it is Lame Or MC
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Try this command line.
--preset voice --resample 16 -b 8 -a -m m -h --lowpass 4 --highpass 1200
I'm guessing you are trying to record voice from your explanation.
That was to show how to set sampling, cutoffs, etc.
A simple way is to just use the --preset voice option which will result in a mono recording as well. The preset may work where other switches do not in recording. But if switches are not being passed, then it is a problem of course.
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That still did not work.
As with my settings, when made under the Encoder->File Conversion section it works as desired and when it is set under the Encoder->Sound Recorder it does not work correctly.
Knowing this limitation, I realize that I can always re-encode it after I capture it, I was just really hoping that MC would be able to do it during the capture process to eliminate one extra step.
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It seems that JRiver might be aware of a BUG with the Sound Recorder after all. I came across this post (http://yabb.jriver.com/interact/index.php?topic=27198.0) on the site today.
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Sorry, but the sound recorder only outputs 16 bit 44.1khz files currently.
I wonder why resampling doesn't work then?
I understand that is what it is sending to the record process.
Why can't LAME resample as it is told?
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this problem is as old as rune stones
most people never see it