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Digital audio takes more power?


Guest Anonymous

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Guest Anonymous

According to my receiver's volume scale, to obtain the same volume/loudness, there is a significant amount of more power required to produce digital sound than analog sound.

1. I am assuming this is normal and not unique to my set-up?

2. If this is true, I am guessing (In layman's terms) the reason is because in digital sound there is a lot more information to decode/produce per channel? Thus, more power per channel is required to drive this more complex sound?

Experts, can you verify or revise. (Keep it in layman's terms)

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Guest Anonymous

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On 4/1/2005 11:53:53 AM Champagne taste beer budget wrote:

It probably has more to do with the output voltages of your cd player, or whatever digital source you're using, verses the voltage output of your turntable.

That's my opinion, others may vary.

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You most likely know more than me, but let me explain a little more specifics that may or may not support or refute our differing theories.

1. Recently, went from a super cheap samsung dvd/cd player (had about 2 years) to the Sony I currently have listed and both were played at the approximate same amp volume control level. I have the Sony is more efficient in power consumption?

2. A few weeks ago, went from a "Digital" Cable box with analog outputs to a Digital Cable box with digital outputs. Noticed to obtain the same level of sound/volume (even watching a news show), I had to significantly turn up the amp's volume control when in digital audio.

3. A few days ago, I upgraded from a Pioneer VSX-D914 to a Pioneer VSX-1014TX (FYI, both have same power output ratings and basically the 1014 just has more decoding capabilities.) All things being equal, in just comparing the difference between each amp's volume output settings when playing the SONY DVD and the cable box with digital audio out, there is a noticable difference. For example to equal the approximate same volume levels, the 914 was played at -55 and -35 for cable and dvd respectively. In comparison, to obtain the same volume levels, the 1014 is played at -30 and -10 respectively.

Why such a change from amp to amp? I have to imagine the 1014 uses power more efficiently? Or is it because it's decoders are more sophisticated and it requires more power to create more complex sounds?

Are digital decoders the culprit?

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Are these surround sound receivers? If so you have to program them to set up gain on each channel. If the baseline gain is set high you will need less on the volume knob to produce the same actual sound level. None of this has anything to do with digital sound containing more data. In fact, all digital sound sources contain less data than analog since it is digitized (where's the info from the space between discrete time points?, It's' lost!) There can be different output levels from different D to A converters so that could be a factor also. Don't go to the issue of 5 or 7 channels versus 2 for analog, this is a processor speed issue and has nothing to do with line out signal levels.

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"According to my receiver's volume scale, to obtain the same volume/loudness, there is a significant amount of more power required to produce digital sound than analog sound. "

Your receivers volume scale is a lousy indicator of power output.

If the decoded digital signal is at a lower amplitude then the analog signal you would have to turn your volume up higher to play at the same level. Because you would need to attenuate the signal less before it hits the power amp section of your receiver compared against the analog signal that was at a higher amplitude.

If you are playing at the same level you are using the same amount of power... no matter where your volume control is set.

Shawn

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" (where's the info from the space between discrete time points?, It's' lost!)"

No, it isn't. The waveform is reconstructed exactly as it was input as long as the wave was sampled at more then 2 points per cycle.

If the wave is reconstructed the same as it was input nothing is lost.

However, in the case of DD/DTS encoded/decoded material then there absolutely is some material lost during the lossy encoding of those formats.

Shawn

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"That's like saying that digital pictures have the same resolution as film.  It just isn't so."

Actually digital is getting to the point of having more resolution then film. Film is limited by its grain structure, digital has no such inherent limitation. Digital also has more exposure latitude with good equipment.

"Analog is continuous, digital is discrete."

Once digital is converted back to analog it is continuous again too and exactly matches the input. People that think digital is a 'connect the dots' don't understand how it works.

Shawn

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I gotta go with yaffstone on this one. Regardless of the sampling rate, there will always be "steps" in a digital signal. They may be small, seemingly insignificant steps, but they're there nonetheless. Imagine looking at a pyramid from a distance, it looks like one continuous slope on each side. But once you get closer, you can see that there are steps to each level as it rises. Get closer, you may see that at some point along the slope, there is a stairway that is comprised of very small steps in relation to the large blocks that make up the walls. That would be digital, with a different sampling rate. To make it analog, there would be a smooth surfaced wheelchair ramp going up the side of the surface. If you bent down to examine the surface of the ramp, there would be slight imperfections, rises and falls, that you may feel if you were actually going up the slope in a wheelchair, but you'd never notice by walking up the ramp. When you walk, you take steps, and skip over any imperfections in the surface between strides. When you are in the wheelchair, you are in constant contact with the surface and feel every nuance.

Sorry for the diversion, back on topic, I think that a digital input of "X" mV will give you the same volume, at the same knob setting, as an analog input of an equal mV. Two channel only, I'm not familiar enough with all the intricacies of surround decoders to tell you how they divide the input to the seperate channels.

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"They may be small, seemingly insignificant steps, but they're there nonetheless. Imagine looking at a pyramid from a distance, it looks like one continuous slope on each side. But once you get closer, you can see that there are steps to each level as it rises."

That is all fine and good as a flawed analogy but the reality is you can zoom in as much as you want on the analog output from a digital source and you will NOT see any such thing. That is the reality. The reconstruction filter makes the waveform look just like what was input into the digital system in the first place. No stair steps, just a smooth waveform that matches the input.

A higher sampling rate (such as 96kHz) does NOTHING more for material below 22kHz compared to traditional redbook CDs sampling rate of 44.1kHz. All the higher sampling rate does is extend the range of what can be captures to just under 48kHz.

Shawn

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You're right, an analog output won't have the steps. I was simply comparing a pure digital output to an analog output. One from a digital source, say CD player with digital outputs, one from an analog, say turntable. You probably have more experience than I re the effects of analog convertors, but in my mind, which may be completly off base, if, in between one of the "steps", there was a small bump, the digital would miss it.

Again, I may be all wet here, but I am certainly more than willing to learn. That's why I frequent this forum.

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" I was simply comparing a pure digital output to an analog output."

The digital signal that feeds the dac looks nothing like the analog waveform. If you watch it on a scope it literally looks just like binary in action...for 16 bits you see 16 bits toggling high or low. If anything it sort of looks like a VU meter and it does represent signal level. The more bits toggling on/off the louder the encoded signal is.

" but in my mind, which may be completly off base, if, in between one of the "steps", there was a small bump, the digital would miss it."

This is *absolutely* correct... but think about that. If you could have a complete cycle of a wave fit in between the two samples what do you now know about that wave?

CD samples at 44,100 samples per second.... that is its sampling frequency. For a complete cycle of a wave to fit between two samples then that wave *must* have a frequency greater then 44,100/2..... or 22,500 hz.

You just described exactly why the analog high frequency limit of a digital system is just less then half the sampling frequency.

Any wave below 22,500hz can not fit between two samples of a 44.1kHz sampling frequency system.

That is how it works.

Shawn

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It sounds different partly because even though you can't hear above 20khz or so the second, third, and higher harmonics are lost due to the limited sampling rate. They still exist in analog. In audiophile terms, you may perceive a lack of depth in the digital representation. The newer digital techniques are clearly better than back in the 80's but there is an intrinsic limit by the act of digitizing. BTW, I'm not an analog snob, I listen to both records and CDs.

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Isn't the loss of detail at low signal levels greater at 44.1 than at 96? Also (I know almost nothing about this), is some sort of dithering introduced to deal with this?

Sfogg is right about grain limitations in photography. In small and medium sized prints, digital photos using large file sizes can look fully as detailed as good film pics, and I think we've seen examples on the forum.

Larry

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On 4/1/2005 7:34:09 PM Champagne taste beer budget wrote:

Sorry for the diversion, back on topic, I think that a digital input of "X" mV will give you the same volume, at the same knob setting, as an analog input of an equal mV.

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You aren't dealing with millivolts on a digital input. You are dealing with ones and zeros, all of the same amplitude (on or off). If the are over a certain value, they are ones, if below a certain value they are zeros. The issue will be the analog output of the DAC, and how it matches the analog source where they both connect to the summing buss (or source selector, etc.).

Marvel

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Larry,

"Isn't the loss of detail at low signal levels greater at 44.1 than at 96?  Also (I know almost nothing about this), is some sort of dithering introduced to deal with this?"

The sampling rate has nothing to do with low signal level. What does have to do with that is the bit depth. More bits gives more dyanamic range... greater signal to noise ratio. CDs (16bit) have theoretically 96db of S/N ratio. In theory 24 bit DVD-A has 144db S/N but with the self noise in DACs and such I think the realistic levels achieve is around 100 to 120db S/N.

Dithering is used to increase linearity at low signal levels like you said and can be used with CD or DVD-A material.

Shawn

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"It sounds different partly because even though you can't hear above 20khz or so the second, third, and higher harmonics are lost due to the limited sampling rate.  "

One of the biggest reasons a new stereo DVD-A of an old recording sounds different then an earlier CD release is because the DVD-A has been remastered/remixed. You aren't listening to the identical material delivered between to different formats... you are hearing different material delivered between two different formats.

Shawn

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On 4/2/2005 9:55:56 AM sfogg wrote:

"It sounds different partly because even though you can't hear above 20khz or so the second, third, and higher harmonics are lost due to the limited sampling rate. "

One of the biggest reasons a new stereo DVD-A of an old recording sounds different then an earlier CD release is because the DVD-A has been remastered/remixed. You aren't listening to the identical material delivered between to different formats... you are hearing different material delivered between two different formats.

Shawn

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At the same time though, when working in the studio with the original source material, the 192 still sounds better...a lot of the typical digital grainy sound is gone. As mentioned, it's certainly not because the DAC is producing a "stepped curve." The output of a DAC is just as analog as any other analog source. I personally think the difference in sound is simply due to the fact that a 192 DAC is simply better than a 44.1 DAC (less jitter perhaps?). There is also the phenomenon of "combination tones." When the ear detects two frequencies, the ear also hears 2 more; the sum and difference of the original two frequencies: F1+F2 and F1-F2. So if you have a 25,000 Hz tone and a 20,000 tone (say with a flute), then the ear is going to be creating a 5,000 tone. Now record the flute in 44.1 and the 25,000 Hz gets cut out. Now you also no longer have the 5,000 Hz combination tone and thus the flute will sound different. Even though you can't "hear" the 25kHz, the combination tones are still going to be created. That 5,000 combination tone is going to also cause other harmonics to be created by the ear as well. It is these undertones and harmonics that help make something sound more full (aka, less harsh or less grainy). Spend some time in the studio recording cymbals, flutes, voices, etc etc and the 192 bit rate becomes so much better. The phenomenon is a minor one and it wouldn't be that hard to measure the highest frequency produced by any instrument and then just pick the right bitrate accordingly. I can't imagine needing anything much more than 192 though...

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getting back to the man's original question i have to wonder if it's simply a function of the equipment that he is observing. in my case, when i switch from normal listening mode to the dts and other digital modes when playing either vhs or dvd, i have to crank up the volume a touch to maintain the same db's. i always attributed this to the possibility that its using more of the decoding circuitry or some such and needed to draw more power to do so thus also reducing the volume.

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