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Digital Sound FAQ

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I hate to say it MattMan, but I'm also going there is _zero_ difference in sound quality of a digital source. The thing about a digital signal is that there is usually a set protocol to go along with it. Source devices have to stick to that protocol. It doesn't matter if it's a $400 laptop or another $4,000 high end audio source, if the manufactures of those devices want them hooked up to something like a receiver using some form of digital audio, they both have to meet that protocol, which means sending the exact same signal, or else the receiver will not know what to do with it. It is not a matter of good audio or bad audio, it is a matter of THE audio or NO audio.

Also, disabling the analogue audio on the laptop will have zero effect. There is no "reduced signal path", because it is all the same signal. It gets sent as a digital signal to the laptops audio controller, which then, if needed converts it to analog and sends it along one set of traces to the analog jacks, whatever those may be (on a laptop, more than likely that's your standard 3.5mm jack). Or, if digital, converts it to whatever protocol is needed (HDMI, TOSLINK, there is no signal loss in a digital to digital conversion so it doesn't matter) and sends it along a different set of traces. Chances are you won't ever have analog and digital signals using the same traces.

Now, if you still claim that your squeeze box is somehow putting out a different digital signal than the laptop, that must mean there must be some sort of EQ or something on the software side that is manipulating the source before it gets sent out. If you're still thinking that one digital source can put out a digital signal "better" than another. you might as well go on and say that the squeeze box is simply reading data from the hard drive "better", which not only is the same line of thinking, but it would mean the laptop is reading that signal wrong. This means you'd get nothing off the drive, because to the laptop, the hard drive would be speaking gibberish.
Protocols are used to establish the common connection between devices. Doesn't dictate the actual data being passed between them.

You've never heard of modding audio devices to reduce the signal path for the electrical signal? Pretty common mod on a lot of audio components, like the Jolida phono pre. People move a few cap positions on the PCB boards to reduce the electrical signal path and has been pretty much proven to make improvements.

Nope, my laptop was using Foobar2000, a very streamlined piece of playback software, hooked up via ASIO to my DAC. No EQing there. The laptop was inducing noise into the signal. Not a problem with how it was reading the files, but just wasn't transmitting them cleanly. And to say that all computers have an equal audio controller in terms of quality, down to the hardware level, is far from true.

I've heard the argument about digital sources dozens and dozens of times and I gotta say after being into audio for many years, and speaking about this exact topic numerous times over those years, I disagree with mostly of what you said. The proof is in the pudding. At the end of the day as long as you're happy with your system, all that matters.
Yep agreed. Fiber is the safe option.

I've actually seen that before. It's a cool experiment. The laptop was/is bit perfect. The big problem I had with it that others have as well, were electrical grounding. Some power supplies give off noise into the USB connection that cause some crap noise when the signal is still considered bit perfect. Even after I fixed this, Squeezebox was better.
All I know id if I get withing 5 feet of the ole Trip Z's with it cranked up playing anything atm it will blow the hair on your legs and sounds clean, have many speakers I've modified myself hooked up and work well also.

I just enjoy full blown concerts in my living room though I don't crank it up all the way that often and even if I do it doesn't distort with 700 watts.

More than enough and sounds clean to me.
Here's a (long!) video explaining how Delta Sigma works:

In a digital amplifier, you take the PCM coming from the PC or whatever and send it through a digital input Delta Sigma modulator in order to convert it to a bitstream. Unlike ordinary digital signals, that bitstream has the interesting property that despite being only two levels, low pass filter it and you get the analog signal. But before doing that, a digital amplifier uses the bitstream to drive some MOSFETs or IGBTs to boost the power to a usable level, then low pass filters it with a simple LC filter. Hence why digital amplifiers are sometimes called power DACs.

And that explains why HVICs and surface mount MOSFETs/IGBTs are superior to IPMs and through hole MOSFETs/IGBTs as far as audio quality goes. By keeping stray capacitances and inductances to a minimum, the operating frequency of the power stage (and thus the clock frequency of the modulator) can be increased, some able to go into the low MHz. (If you're thinking of overclocking your amplifier to boost audio quality, don't. There are many reasons why that is not possible. Even if you could, it would almost certainly decrease quality as it's all tuned to work at that specific clock frequency.)