>
> When encoding 44.1 kHz audio to a 128kbit/sec mp3, lame by default cuts
> off the high end with a transition band of 15115 Hz - 15648 Hz. (32 kHz
> audio to 128kbit/sec mp3 has a slightly lower cut-off by default with a
> transition band of 14065 Hz - 14452 Hz.)
>
> If someone was encoding 128kbit/sec mp3s from their soundcard (ie from an
> analog source like a mixer instead of a CD rip) and was okay with these
> default low pass filter frequencies, should they probably use 32kHz as the
> sample rate instead of 44.1 kHz? Seems to me about the same frequencies
> would be represented ( ~20 Hz up to around 15 or 16 kHz) but the
> compression ratio wouldn't be as big for 32kHz. Would that lower
> compression ratio result in better sound quality? Or does more samples a
> second result in better sound quality at all frequencies?
>
If you test this out, let us know the results. FhG's encoder
will automatically downsample to 32khz if you encode stereo 96kbs,
but at higher bitrates they leave the samplerate at 44.1khz.
Mark
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