Originally posted by RD Do I interpret this correctly to mean that it is not that case that ALL frequencies at 18.67 Khz are cut out, but rather like Tangent said there is a gentle cutoff slope (or maybe not so gentle) beginning with 18671 Hz and ending with 19205 Hz?
And therefore that we do not have full attenuation until sometime after 19 Khz?
Originally posted by tangent Well, I believe that MP3 does lowpass filtering in the frequency domain rather than the time domain, which I believe doesn't cause the phase distortion which normal filtering methods cause.
Originally posted by RD A while back on the r3mix board there was a big controversy over this and I remember Julius calling me stupid because I advocated --lowpass 19 over --lowpass 19.5
Originally posted by RD Have you come across a file while testing --alt-preset normal that you felt --lowpass 19 was not enough for you?
Originally posted by niktheblak Actually you can't do a lowpass but in frequency domain. In time domain there is no direct way to alter frequencies since the samples have no "frequency"-value.You can however do a lowpass in time-domain by mathematical process called convolution. You can convolve the source signal with a lowpass-like impulse response signal and get a lowpassed signal. But these impulse responses are designed in frequency domain to begin with and IFFT'd (Inverse Fast Fourier Transform) into time-domain for the use of convolution.
In frequency domain we can "draw" a perfect brickwall lowpass filter (with transition band of 0, looking like a square) but it cannot be perfectly transformed into time-domain. In complex DFT (Discrete Fourier Transform) the IFFT'd lowpassed signal would have imaginary values as well as real values. Discarding these imaginary values would mean loss of phase information. So we have to use only frequency responses which do not result imaginary output. These signals are "sinusoidal like" smooth gentle slopes.