The Best DAC is no DAC

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DSD utilizes a DAC

In a 1-bit digital signal encoding/decoding system, such as DSD, the system only needs to represent two discrete amplitude states. Call those two states high and low, or 1 and 0, or positive going and negative going, or whatever, but only two states need be recorded, and later reproduced. Because of that, there isn't the need for most of the circuitry (resistor networks and such) required for multibit digital signal conversion. A simple logic gate or some other two-state driver circuit could form the DAC circuit for 1-bit decoding. One consequence of 1-bit encoding/decoding is that is creates a great deal of quantization noise, but that can be moved out of the audible band. In short, functionally, there is a DAC circuit even with bare bones DSD playback.
 
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You still do have a DAC, just its not called a DAC, its your digital signal's buffer as Ken points out just above.

The downside of using just a digital chip for your DAC is that it has about 6dB PSRR on average - meaning your digital power supply's quality is the limitation. This could easily be the cause of your bass lightness. The other issue is jitter - a 'raw' DSD signal is inordinately sensitive to jitter. It can be ameliorated through using a transversal filter - like Miska has shown in one of his DSD DAC schematics.
 
You still do have a DAC, just its not called a DAC, its your digital signal's buffer as Ken points out just above.

The downside of using just a digital chip for your DAC is that it has about 6dB PSRR on average - meaning your digital power supply's quality is the limitation. This could easily be the cause of your bass lightness. The other issue is jitter - a 'raw' DSD signal is inordinately sensitive to jitter. It can be ameliorated through using a transversal filter - like Miska has shown in one of his DSD DAC schematics.
The digital buffer is a buffer. It's input is a digital signal and its output is the same digital signal. So where is the digital to analogue conversion that you are talking about?

The point is this - the digital output from the USB receiver is normally fed to a DAC. I am not feeding it to a DAC. I am trying something different, if it hasn't been tried before. As per Wikipedia (so it must be right!)

"The process of creating a DSD signal is conceptually similar to taking a one-bit delta-sigma analog-to-digital (A/D) converter and removing the decimator, which converts the 1-bit bitstream into multibit PCM. Instead, the 1-bit signal is recorded directly and in theory only requires a lowpass filter to reconstruct the original analog waveform. In reality it is a little more complex, and the analogy is incomplete in that 1-bit sigma-delta converters are these days rather unusual, one reason being that a one-bit signal cannot be dithered properly: most modern sigma-delta converters are multibit."

To re-iterate. I am not using a sigma-delta convertor. Or any other converter. I am feeding a digital signal (DSD, or PDM) into a band-width limited transformer (hence acting as a low pass filter) to reconstruct the original analog waveform. Any conversion (if you want to call it this) happens in the transformer. Same as Class D amp. PWM signal is fed to a low pass filter then to the speaker. Where is the DAC in a Class D amp?

Anyway let's not argue about semantics. Call it a DAC, or not a DAC. Whatever you like. But i am trying to minimise the signal path in my system. Because I feel like it. And I can (well if this works, I can). You point about about power supploies is valid, and I know I need a better p-s. WIll a better p-s improve the bass? That's an interesting point, I can hear noise at the moment that I need to deal with but I hadn't considered the impact of the p-s on bass response.

Also, I don't know what a transversal filter does. Do you have a link?
 
The D/A conversion occurs when you cease to interpret the signal as a digital one and begin to interpret it as an analog signal. So that happens at the output of your buffer because you're not feeding it into a digital circuit downstream, rather an analog one.

I'll have a hunt around and see where I found Miska's circuit - its an implementation of a 32 tap transversal filter.
 
As I understand it, a DSD signal is a form of quantised pulse density modulation. Hence the 'digital' signal already contains the analogue signal, which is why a low pass filter is all you need. So the reason why you don't need a DAC is that the signal was never truly 'digital' at all, but merely used a clever way to record and play back an analogue signal which looks like a digital signal.

Note that 'analogue' does not mean 'music signal represented by voltage level' but 'music signal represented by something variable (but not a number - that would be digital)'. Here it is pulse density.
 
Am I just being stupid, but why do you think there is no DAC involved? Where do you think the analogue audio you are listening to is coming from?

"The process of creating a DSD signal is conceptually similar to taking a one-bit delta-sigma analog-to-digital (A/D) converter and removing the decimator, which converts the 1-bit bitstream into multibit PCM. Instead, the 1-bit signal is recorded directly and in theory only requires a lowpass filter to reconstruct the original analog waveform."

Direct Stream Digital - Wikipedia, the free encyclopedia
 

Seems you would mention WAVE IO especially since one of those you did mention looks a awful lot like a copy of WAVE IO ...

http://www.diyaudio.com/forums/digital-source/188902-xmos-based-asynchronous-usb-i2s-interface.html

"LORIEN" is a true gentleman and a great pleasure to do business with.
 
Hi,
I have an experimental confirmation that you don't need a DAC to convert a DSD signal to analog: http://www.webalice.it/meneghettig/Un%20DSD%20player%20autocostruito.pdf
(sorry, in Italian).
Actually, the DSD signal is discrete in the sense that it can assume only the value high or low, but it is not a numerical representation of a waveform, it is like a PWM signal and for this reason an RC filter can be enough.
 
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.....Hence the 'digital' signal already contains the analogue signal, which is why a low pass filter is all you need. So the reason why you don't need a DAC is that the signal was never truly 'digital' at all, but merely used a clever way to record and play back an analogue signal which looks like a digital signal.
:up: Well put. Thank you.