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[cfastner]

I have upgraded MediaCenter 9 to version 113 as suggested by JimH but converting a .WAV file to a 64kbit .MP3 file speeds up the audio, sounding like the chipmunks.  I am trying to record some old audiobook cassettes encoding them to 64kbit .MP3 files.   The recordings work fine and encode without a problem but if I don't make it back to the computer by the time that the audio is finished, I want to open the file in MediaEditor trim off the "dead air" at the end of the track.  When I attempt to save the .WAV file to 64kbit .MP3 files the audio speeds up.  I have heard of speed reading but speed listening is not what I am trying to accomplish!

Any ideas would be greatly appreciated!

[Autoelph]

Some things to look for:

when listening to the .wav (b4 encoding) does it sound OK?

What sample rate is your audio card recording at? The default can be found in the control panel/sound-audio section (depending on OS) or sometimes in the audio cards' properties or setup panel. The default sample rate for wav to MP3 is 16bit-44k...if you recorded the wav at 8bit-22k, the encode would speed up the original to 4x faster. When using encoding defaults, record audio at 16bit-44k, then convert to MP3.

[Soundman]

Hello.

I have this problem too.  I think the problem happens if the .WAV is not the standard 44.1KHz stereo wav.  In my case, I record spoken audio books from cassette tape at 22.1kHz, 16-bit mono.  Then, I "custom" encode to mp3 using these options:

--cbr -b 24 --resample 16

That is, 24kbit, at a sample rate of 16kHz, which I found the best quality/bit rate tradeoff after much experimentation.

Anyway, this always results in 4X playback speed after encoding!  This does not happen if I manually encode it using lame.exe on the command line with the same parameters.

It seems like MC9 is somehow mangles the data as it is pushed to the lame DLL.

Note that all works OK if the original WAV is the standard CD format 44.1K, 16-bit stereo.

Please fix!

[Autoelph]

Well, I'm not sure I totally follow your logic:

I have this problem too.  I think the problem happens if the .WAV is not the standard 44.1KHz stereo wav.  In my case, I record spoken audio books from cassette tape at 22.1kHz, 16-bit mono.  Then, I "custom" encode to mp3 using these options:

--cbr -b 24 --resample 16

That is, 24kbit, at a sample rate of 16kHz, which I found the best quality/bit rate tradeoff after much experimentation.

Save yourself the hassles, record at 16-44 stereo, then encode and this will give you the best encoding quality possible. Don't be fooled, ANY conversion will cause loss of quality. Definitely bit depth, resampling and channel (mono to stereo) conversions are the biggest offenders. And doing these conversions in the encode process is the absolute worst time to do them.