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stardust wrote:This is again the kind of magic and vodoo that usually only trained ears will hear and typically also only in A/B comparisons.

http://emusician.com/mag/emusic_debunking_digitalaudio_myths/ wrote: Analog audio has a theoretically infinite frequency range, whereas digital audio (software and hardware) has a hard limit on high frequencies, as determined by the sample rate.

Many analog processes take “infinite” frequency range for granted, but they can't do the same in the digital domain. For example, a standard analog peaking EQ has a bell curve that is symmetrical around the center frequency; one side slopes to 0 Hz, and the other slopes toward infinite Hz.

You can implement the same EQ in the digital domain, but infinity is suddenly much closer. In fact, what was infinite Hz is now the Nyquist frequency (half the sampling rate) or 22.05 kHz at a sampling rate of 44.1 kHz. This difference in the proximity of infinity results in an EQ curve with a dramatically lopsided shape (see Fig. 6).

Things can get stranger as the EQ's center frequency approaches the Nyquist frequency. At those high frequencies, I've seen digital EQs that started to take on weird globular shapes and even go down in actual frequency as I turned up the frequency knob. I've even encountered a peaking EQ that looked much more like a resonant highpass filter.

Those problems can be avoided or at least minimized by clever programming. The degree to which the programmer is successful defines, to a great extent, the differences between good and bad digital EQs.

...

Compressors and limiters also have frequency-related issues. You're probably familiar with aliasing — it causes audio artifacts when sampled audio contains frequencies higher than the Nyquist frequency. Aliasing doesn't occur only during sampling, however; it can also happen entirely within the digital domain.

For instance, compression and limiting work by modulating one audio-rate signal (the input) with another audio-rate signal (the compressor or limiter's automatic gain control, which operates in the audio range when the attack and release envelope times are fast). When you modulate one audio-rate signal with another, it has the effect of adding the two signals' frequencies; if the total exceeds the Nyquist frequency, you'll get some aliasing.

A full-bandwidth audio signal processed with a limiter or a compressor with fast attack or decay times falls into that category; the faster the attack or release and the greater the compression or limiting amount, the more aliasing you hear. That is the cause of the crunchiness many people hear in digital-dynamics processors. Again, clever programming, especially oversampling, can minimize these aliasing artifacts.

You'll find similar predicaments in synths. Resonant filters suffer from the same infinity-is-much-too-close syndrome as digital EQs, and various synths differ widely in their success at addressing the problem. For instance, standard Chamberlin digital filters (the most common type) only work correctly to about one-sixth of the sampling rate. For a synthesizer running at 44.1 kHz, that means the resonance tops out at about 7 kHz.

Oscillators have problems similar to compressors. For example, a square wave at, say, 4 kHz is actually generating frequencies well above the Nyquist limit, because of the waveform's sharp edges. Untamed, that can cause excruciating aliasing, especially toward the top of the keyboard (as you can hear in some popular products). Similar aliasing can also happen when samples are transposed above their original pitches. Techniques for dealing with these complications vary and account for some of the sonic differences among synthesizers.

stardust wrote:Yes, for the generation part this is undoubted as said by pollux...

Neutron wrote:... what you might notice, when you have a very high note. even though we can not hear over a certain frequency, at 44.1 you can tell the difference between a sine wave and a square wave at the highest pitch you can hear, not because your ears could hear the difference with a natural sound, but because the samples do not "line up" perfectly with the waveform.

... you would hear a subharmonic "beat frequency" (aliasing) which can not be completely gotten rid of except with a perfect (non existant) analog filter on the DAC output. ...

Neutron wrote:i have always noticed more difference with bit depth than bitrate, i use 44.1 so i can get more out of my cards.

stardust wrote:I found these excellent demos for what resampling aliasing can do with high frequency harmonics and in A/B comparisons you can hear it.