[linux-audio-dev] Direct Stream Digital / Pulse Density Modulation musing/questions

[Mickael S Vardo]

I'm a professional audio engineer for 15 years, and DSD did convince me. I
could compare the sound quality with TI ADC/DAC.

Conventional PCM techniques are unable to reproduce high frequencies
correctly. And the explanation is very simple. If you record a sound at 44.1
kss, you get a theorical frequency response of 0 - 22050 Hz. BUT to describe
frequencies from 11050 to 22050 Hz, you can only play with a 4-sample long
period.

A 22050 Hz sine could be really accurate (one sample up, one sample down
every 1/22050th second), and so is 11025. But intermediary frequencies
introduces temporal aliasing, some metallic feeling due to temporal
quantization. This is inherent to the very low sampling rate (96 kHz is just
a bit better, but no miracle), which is unable to describe waveforms at high
frequencies.

Bad high frequencies temporal definition means bad transients. Anyone can
notice it when he _actually_ hear and compare PCM and DSD.

Stop speculative talking and try to get some real demo...

-- 
mickael