Nå er det FPGA og ikke chip baserte DAC'er som er i støtet.
Dette gir en helt annen fleksibilitet og mindre kompromisser.
Jeg er opptatt av hvordan det lyder og ikke om det er inn eller ut.
Isåfall er også Playback Design og en drøss med andre teknologiske dinausaurer.
Skal man bare se helt bort fra lydkvalitetskriteriet for at noen papers sier at dette ikke er tingen?
Legg merke til at de paper'ne viser at lydkvaliteten fra en en-bits DAC vil ha noen svakheter sammenlignet med en multibits DAC. Den kan sikkert låte bra, men vil ha noen iboende problemer med støy etc. Om det er implementert i en reprogrammerbar FPGA eller i en tradisjonell chip enndrer ikke på det. For all del, entusiasme er fint, men jeg ser ikke helt hva som skal være den store nyvinningen i designet.
Regner med at det er noise shaping du refererer til:
Noise shaping - Wikipedia, the free encyclopedia
Her må man selvsagt være nøye, slik at dette ikke folder seg ned i det hørbare området.
Specs ang. S/N ble jo vedlagt på side 3 i mitt innlegg og dette er jo meget gode tall.
Skal finne en kurve som viser noise floor.
Man kan også justere volum praktisk talt uten tap, som innebærer at man kan vurdere å droppe forforsterker hvis inngangene på DAC'en er tilstrekkelige.
Ellers, så sier vel ikke PS Audio at dette er revolusjonerende ny teknologi - det de sier er: 'DSD done right'.
Edit:
Her er støygulv måling med kommentarer fra Principal designer (klippet fra PSA community forum).
Y-aksen er i kHz.
Problemet er at etter migrering til det nye forumet så har oppløsningen nærmest blitt redusert til thumbnail størrelse og dermed uleselig, men han sier ihvertfall at '
As you see in the plots the noise is completely insignificant (-90dB) compared to the high frequency material that should be in a 192kHz recording.'
Tho the noise gets shifted to higher frequencies the output filter should tame it down quickly.
Some modified players might not filter the highs as much as they should, but by spec the ultrasonic noise on an SACD player shouldn't rise above -40dBFS. This is 1/100 of full scale. Many (most?) players filter more out and as the frequency rises the noise drops off quickly.
Anyway here's a plot with my noisy scope of the noise floor of the
DirectStream. The noise floor and the spikes are from the scope not the
DirectStream. Still you can clearly see the DSD bump and you can see that it's not too big and that the output filter is starting to bringing it back down. That's one reason I upsample to double rate DSD.
I have definitely felt the pressure you mention, in one system it was from the Nordost Valhalla cables. I don't know what they did to the ultrasonics but they sucked in at least that system.
The purpose of shaping noise is exactly to keep the level of any noise proper to a low enough level to not be a problem. What you can't see in my plot because of the scope's floor is that the noise is well below what you see in the plot (continue the line down to the left from what it looks like at 57k to get an idea of how low it is in the digital signal.)
Aliasing can happen when you change sample rates up and haven't bandlimited your input to 1/2 the sampling frequency. In this case I filter out everything over 1/2 of 2 * 64 *44100kHz before upsampling to 10 * 64 * 44100kHz and then back to 2 * 64 * 44100 so there's no aliasing.
They are two independent things.
Noen kommentarer til noise shaping metoden:
Noise shaping is simple at the 30,000 ft view:
Think of the rectangle defined by bandwidth (1/2 of the sample rate for PCM) and the S/N (approx. bit width * 6 for PCM). That box has a certain "area".
When you use noise shaping you shift some of the area from the higher frequencies that you don't care about on to the lower frequencies that you do.
For example you can (roughly) take the top 1/2 of the frequency and trade it for increasing the resolution. You can do that over and over.
Another way of looking at things is that a sigma delta A/D can be thought of as comparing a low pass filtered version of the digital output signal (e.g. what a sigma delta DAC does) to the incoming signal. If it's too high a zero is output, if it's too low a one is output. When the DAC in the A/D (and the DAC in your playback system) get a one they head for the positive rail as fast as the output filter allows, when they get a zero they head for the negative rail as fast as the output filter allows. The bandwidth of this process is set by that filter. But really it's the high frequency component of the input and the noise from the single bit DAC that's filtered away. By nuking the high frequency noise the A/D is only paying attention to the part of the signal we care about, the lower frequencies. The ones and zeros are trying to track the low frequencies and ignoring the high frequencies so you've traded off low frequency accuracy for getting gobs of noise in the high frequencies.
J.J. Johnson has a good slide deck out there somewhere. I'll see if I can find it.
-Ted