I am not an audiophile expert so I am wondering what people's opinions are on this subject.

It seems like having a digital out on your digital audio player would be better than an analog line out. The player would have to convert the digital signal to analog and it seems that being able to to do that conversion would be better on a high quality reciever.

Do you think that a digital audio player such as the Karma should have one, or does that not really make a difference?

Digital, as you probably know, will help elminate ground loop problems. the empeg does not have digital out by default, but see here and the link therein for details. A thread in the topic is here

It seems like having a digital out on your digital audio player would be better than an analog line out.

If you happen to have an amplifier with a digital input, yeah.

Do you think that a digital audio player such as the Karma should have one, or does that not really make a difference?

Since the Karma is a portable hand-held player and designed to be played with headphones, then no it doesn't really make a difference.

It seems like having a digital out on your digital audio player would be better than an analog line out.

If you happen to have an amplifier with a digital input, yeah.

I don't know that I inherently agree with that. It assumes that the amplifier's DAC is better than the player's, and I don't know that's the case. I don't think we're losing much signal in the cable going from the HU to the amp, so that's probably irrelevant.

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It assumes that the amplifier's DAC is better than the player's, and I don't know that's the case.

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Unless of course you are interfacing with a switching amp with digital inputs straight to the digital circuitry, in which case there would be no DAC per se.

If the actual amp circuitry is completely analog internally, I would go for a separate DAC dedicated to the conversion process from digital to analog. Of course the quality of the DAC depends on many things including the DAC chip itself, power supply, component selection, I/V stage where applicable, board layout and analog filtering.

Not all DAC sound quality issues are measurable. Each DAC chip is said by many to have a sound signature of sorts, meaning that two DACs with identical specs may sound different. Some purists insist on using the older R2R DAC topology for most pleasing sound, even at the expense of lower performance than can economically be achieved with the latest delta sigma topologies. Using an external DAC will let you listen to the "flavor" of sound you prefer most.

Stu

Well, at some point, there's always going to be a DAC of some nature, as actual ``travel in a physical medium via compression waves and push and pull my eardrums'' must inherently be analog.

Naah. Just get one of those neural interface things embedded in your skull

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Well, at some point, there's always going to be a DAC of some nature, as actual ``travel in a physical medium via compression waves and push and pull my eardrums'' must inherently be analog.

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Of course, that was the per se part of my explanation.

Stu

Could you give me an example of the type of amp (without DAC) you are refering to ?

Check out the Harmon Kardon DRP series. As best I can tell, it's a switching amplifier where you modulate the switching frequency via Pulse Width Modulated wafeform, thus changing the gain???

http://www.harmankardon.com/product_detail.asp?cat=REC&prod=DPR%201001&sType=C

Best part about switching supplies is that they are very efficient (80-90 pct vs ~50 percent for a linear supply), and thus can be small and light (i.e. no heat sinks)

-Ted

This might be a dumb question, but indulge me - with a PWM amp, does the speaker itself act as the DAC?

AIUI (and CMIIW) the digital amp provides a nice constant voltage (if it's any good) and does PWM at a frequency too high to hear (and too high for the speaker to keep up), meaning that the speaker smoothes the PWM to pass the audible frequencies and not the inaudible harmonics.

The PWM is then effectively analog still subject to noise pickup, unless it's re-generated nearer to the speaker. IOW, it may be treated as a digital signal or as an analogue one.

The D/A conversion is happening in the amplifier. By changing the pulse width, you are changing the output voltage of the amp. At least that's how I think it works. I guess there'd be no need for a LPF on the output. Hmmm.. I'm not really sure how it works, so take it with a grain of salt.

-Ted

I guess there'd be no need for a LPF on the output. Hmmm.. I'm not really sure how it works

Well, the output transistors in a class D or T amp are pulswidth modulated, ie fully on or fully off, for varying times - this is what makes them so power efficient compared to A or A/B class amps. By not running "somewhere in the middle" where you have both fairly high current through the transistor and a fairly large voltage drop over it, you get much smaller losses in the output stage - P=UxI.

It's effectively the mass of the moving parts of the speaker that reconstructs the low frequency signal out of the PWM high frequency switched signal. The cone can't follow each on/off transition due to its mass, but instead "follows the trend" in the PWM signal, ie the low frequency signal that was PWM'ed

/Michael

Are you saying the PWM waveform is what gets sent to the speakers????

-T

That is my understanding, yes.

The reason they're so efficient is that the output transistors are always fully on (lots of current, little voltage through/over the transistors) or fully off (no current, high voltage) which makes the UxI losses low.

Switch the transistors fast enough and the speaker has no chance to follow the actual signal, but only the averages. But to accurately reproduce higher notes you have to switch very fast, which is the reason fullrange D or T class amps are rare; mainly they're meant for subs.

Hmm, looking at some links it appears that at least some have an inductor/cap filter at the output (to which the speaker probably adds a fair bit considering its inductance) to help smooth the signal.

Good tech paper, and another one.

/Michael

The latest from TI called Equibit, uses a PCM-PWM converter followed by a digital amplifier stage to take digital signals directly. The output of the amp is still PWM, but before arriving at the speakers, the signal is sent through a simple LC circuit which TI calls a "demodulator" circuit. The speakers take care of the rest.

To input analog audio into these amps, you actually have to use a analog to digital converter first.

Sharp makes 1-bit digital amps (SM-SX1 and SM-SX100) that take a DSD stream as opposed to PCM. I'm not sure what they are doing to get the PWM out of the DSD, but it's probably interesting.

Stu