There are people who would answer this question entirely using Algebra, but unfortunately, my Algebra is not up to standard, specifically when applied to Fourier Transforms. Yet, I can often visualize such problems and reason them out, which can provide a kind of common-sense answer, even to this type of a question.

If a DCT is fed a time-domain sine-wave, the frequency of which exactly corresponds to an odd-numbered frequency coefficient, but which is 90 degrees out of phase with that coefficient, the fact stands, that the coefficient in question remains zero for the current sampling interval.

But in that case, the even-numbered coefficients, *and not only the two directly adjacent to this center frequency*, will alternate between positive and negative values. When the coefficients are then laid out, a kind of **decaying** wave-pattern becomes humanly discernible, which happens to have its zero-crossings, directly at the odd coefficients.

Also, in this case, if we were just to add all the coefficients, we should obtain zero, which would also be what the time-domain sample at n=0 should be equal to, consistently with a sine wave and not a cosine wave.

And this is why, if a ~~DCT~~ is applied to the coefficients, and if the phase information of this chosen *IDCT* is correct, the original sine wave can be reconstructed.

Note: If the aim is to compress and then reproduce sound, we normalize the DCT, but do not normalize the IDCT. Hence, with the Inverse, if a coefficient stated a certain magnitude, then that one coefficient by itself is also expected to produce a ‘sine-wave’, with the corresponding amplitude. ( :1 )

I think that it is a kind of slip which people can make, to regard a Fourier Transform ‘as if it was a spectrum analyzer’, the ideal behavior of which, in response to an analog sine-wave of one frequency, was just to display one line, which represents a single non-zero data-point, in this case corresponding to a frequency coefficient. In particular because Fourier Transforms are often computed for finite sampling intervals, the latter can behave differently. And the DCT seems to display this the most strongly.

While it would be tempting to say, that a DFT might be better behaved, the fact is that when computers crunch complex numbers, they represent those as pairs of real numbers. So while there is a ‘real’ component that results from the cosine-multiplication, and an ‘imaginary’ component that results from the sine-multiplication, each of these components could leave a human viewer equally confused as a DCT might, because again, each of these is just an orthogonal component vector.

So even in the case of the DFT, each number is initially not yet an amplitude. We still need to square each of these, and to add them. Only then, depending on whether we take the square root or not, we are left with an amplitude, or a signal energy, finally.

When using a DFT, it can be easy to forget, that if we feed it a time-domain single-pulse, what it will yield in the frequency-domain, is actually a series of complex numbers, the absolutes of which do not change, but which do a rotation in the complex plane, when plotted out along the frequency-domain. And then, if all we could see was either their real or their imaginary component, we would see that the DFT also produces a fringing effect initially.

The fact that these numerical tools are *not* truly spectrographs, *can* make them *unsuitable* for direct use in Psychoacoustics, especially if they have not been adapted in some special way for that use.

Dirk

1: ) This latter observation also has a meaning, for when we want to entropy-encode a (compressed) sound file, and when the time-domain signal was white noise. If we can assume that each frame states 512 coefficients, and that the maximum amplitude of the simulated white noise is supposed to be +/- 32768, Then the amplitude of our ‘small numbers’, would really only need to reach 64, so that when they interfere constructively and destructively over an output interval, they will produce this effect.

Now, one known fact about musical sounds which are based on white noise is, that they are likely to be ‘colored’, meaning that the distribution of signal energy is usually not uniform over the entire audible spectrum. Hence, If we wanted just 1/8 of the audible spectrum to be able to produce a full signal strength, Then we would need for the entropy-encoded samples to reach 512. And, we might not expect the ‘small numbers’ to be able to reproduce white noise at full amplitude, since the length of the big numbers is ‘only’ 15 bits+ anyway. One entropy-encoded value might already have a length of ~3 bits. So it could also be acceptable, if as many as 1/6 of the coefficients were encoded as ‘big numbers’, so that again, the maximum amplitude of the ‘small numbers’ would not need to carry the sound all by itself…

And yet, some entropy-encoding tables with high amplitudes might be defined, just in case the user asks for the lowest-possible bit-rates.