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DAC attenuation before power amp

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#21
I do indeed think the classic analogue pre amplifier has had its day . . . I started with digital volume control on the computer. . . but there is a snag: inadvertently high volume that can blow up your expensive speakers.
Or crashing computer. I am always nervous about my speakers, my ears, and my vintage Yamaha VFET amp (which would be unrepairable).

internet radio, even at moderate bit rates, sounds better than FM radio.
There still are some excellent FM radio stations. IF you have good FM radio stations in your area, and a high-quality / well-aligned tuner, FM radio can provide sublime sonics. Marvels of 20th century engineers pushing the limits of technology.

More FM tuner info at:

http://www.fmtunerinfo.com/
 

Willem

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#22
Together with the retirement yesterday of my Quad 33 pre amplifer, I also disconnected my Quad FM3 tuner that I had not used for more than a year. I agree that you can have pretty decent FM sound, but even in the best of situations, frequency response, distortion, noise and channel separation are miles apart from CD red book. Admittedly low bit rate internet radio is also far from perfect, but the higher resolution radio stations like BBC Radio 3 are now 320 bps (and the BBC is even experimenting with lossless), and that is undoubtedly better than FM . With lower bitrate stations it is of course comparing apples and oranges, but with ever higher bitrates I now find that the imperfections of internet radio are easier to live with than those of FM radio (and bit rates are only going up). The added bonus is, of course, that in Holland I can now listen to the BBC, Radio France or US Public Radio, to mention only some obviously interesting options.
 

hvbias

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#25
This below is an excerpt from the jriver wiki about volume control in their media center.

The precision offered by Media Center's 64bit audio engine is billions of times greater than the best hardware can utilize. In other words, it is bit-perfect on all known hardware.

To demonstrate the incredible precision of 64bit audio, imagine applying 100 million random volume changes (huge changes from -100 to 100 dB), and then applying those same 100 million volume changes again in the opposite direction.


Amazingly, you will have the exact same signal at 32bit after 200 million huge volume changes as when you started.

In other words, this incredible number of changes results in a bit-perfect output at 32bit, which is the highest hardware output bitdepth (most high-end hardware is 24bit).

This also means one volume change or a series of 100 million volume changes that add up to the same net result is bit-identical.

So any software doing 64 bit volume control is going to be effectively beyond reproach. It will be beyond the precision in any mechanical/electrical volume control anyone could possibly manage to build. Now one could end up using a DAC at such a low level the linearity will become a problem perhaps. Even that is unlikely with the sigma delta DACs of quality. Plus electromechanical volume controls have various linearity issues of their own.

Almost any DAC will be 24 bit in how it handles the data. Some are even 32 bit now. Software for playback is usually 32 bit float or 64 bit. The result of lowering the signal is going to be a 24 bit output which isn't really lowering resolution. You loose resolution to noise even with an analog volume control. The current good playback software volume controls are going to create less negative effects than mechanical attenuation.
One of my big issues with headphone amps is too many of them have too much gain for 2 vrms DAC output, and then volume pots (unless you get crazy expensive ones like Alps RK50 or Penny and Giles) have an imbalance right after max attenuation. If I'm listening at 65-70 db average, do you think this is throwing away resolution with the JRiver volume control? Say on a well designed DAC like from Merging or Exasound.
 

bennetng

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#26
One of my big issues with headphone amps is too many of them have too much gain for 2 vrms DAC output, and then volume pots (unless you get crazy expensive ones like Alps RK50 or Penny and Giles) have an imbalance right after max attenuation. If I'm listening at 65-70 db average, do you think this is throwing away resolution with the JRiver volume control? Say on a well designed DAC like from Merging or Exasound.
64-bit float allows more than +/-6165dBFS.
https://www.gearslutz.com/board/showpost.php?p=10582565&postcount=168
http://forum.cakewalk.com/FindPost/2325892

32-bit float allows more than +/-770dBFS.
https://forum.cockos.com/showthread.php?t=215307
https://www.audiosciencereview.com/...attenuation-before-power-amp.5844/post-131431
https://www.audiosciencereview.com/...es-dsd-sound-better-than-pcm.5700/post-135477

Merging and Exasound don't design DAC chips, and DAC chips only understand 24 or 32-bit fixed point data (up to -144.4944 or -192.6592dBFS and zero tolerance above 0dBFS). So bragging about 64-bit float volume control outside of the DAC chip is meaningless at all. The main benefit of floating point processing is avoid clipping above 0dBFS and 32-bit float can do this already.

Therefore regardless of hardware or software digital volume control, the limitation in digital domain will be 24 or 32-bit fixed point. However, all DACs ultimately output analog signal, so real world limitation will be the analog noise floor of the DAC rather than resolution of digital data.

You may benefit from analog volume control if your headphone amp's analog noise floor, measured in analog domain like dBu or dBV, is significantly lower than your DAC's. JDS Labs for example says the Atom amp has -114 dBu noise.
https://www.jdslabs.com/products/190/atom-amp/

After all, it is an unhealthy gain stage if you constantly need to reduce 70dB, it has nothing to do with analog vs digital at all. If you are not planning to replace the headphone amp and worry about channel imbalance, reduce 30dB on the playback software then do the rest on the headphone amp.
 
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#27
I've currently got a Sys between my DAC and DSP -> Crown stage. It works fine, but I think I would be better off with an active preamp. I have to set the gain on the amp pretty high for the 2V signal from the DAC to get the max volume where I want it and that starts to bring the amp's not-great noise floor into play. I am looking forward to trying the JDS Atom in a few days.
 

Azookey

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#28
I currently have a Chromecast audio going optical out to a Modi then into either a Schiit Magni (used as active preamp for speakers) or a Schiit Saga. The amp is the Pass Amp Camp with 10k input impedance. 2 thoughts/questions...

1. Does anyone know what bit depth the chromecast attenuates the digital signal at (again, when using the toslink output)?

2. thoughts on why either the Magni or Saga would be a better choice of preamp. From the measurements of the Saga, an amp with an input impedance below 30k will attenuate the low frequencies in active mode, while any volume level below 50% in passive mode will raise the output impedance too high for the amp I have. Is the Magni, being an active preamp, better for this situation?

Thanks
 

andreasmaaan

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#29
2. thoughts on why either the Magni or Saga would be a better choice of preamp. From the measurements of the Saga, an amp with an input impedance below 30k will attenuate the low frequencies in active mode, while any volume level below 50% in passive mode will raise the output impedance too high for the amp I have. Is the Magni, being an active preamp, better for this situation?
I'd probably use the Saga in passive mode if I were you. Although lowering the volume increases the output impedance, Amir's measurements indicate that the output impedance nevertheless remains stable across the full audio band. Even where you don't meet the recommended 8:1 or 10:1 ratio of output/input impedance, as long as output impedance is uniform across the audio band, there should be no significant impact on frequency response.
 

andymok

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#30
Regarding Merging's NADAC volume control and method, details can be found here

https://confluence.merging.com/plug...ADACvolumecontrolperformed,inAnalogorDigital?

As to pro interfaces, they should be in their manuals (I think)


64-bit float allows more than +/-6165dBFS.
https://www.gearslutz.com/board/showpost.php?p=10582565&postcount=168
http://forum.cakewalk.com/FindPost/2325892

32-bit float allows more than +/-770dBFS.
https://forum.cockos.com/showthread.php?t=215307
https://www.audiosciencereview.com/...attenuation-before-power-amp.5844/post-131431
https://www.audiosciencereview.com/...es-dsd-sound-better-than-pcm.5700/post-135477

Merging and Exasound don't design DAC chips, and DAC chips only understand 24 or 32-bit fixed point data (up to -144.4944 or -192.6592dBFS and zero tolerance above 0dBFS). So bragging about 64-bit float volume control outside of the DAC chip is meaningless at all. The main benefit of floating point processing is avoid clipping above 0dBFS and 32-bit float can do this already.

Therefore regardless of hardware or software digital volume control, the limitation in digital domain will be 24 or 32-bit fixed point. However, all DACs ultimately output analog signal, so real world limitation will be the analog noise floor of the DAC rather than resolution of digital data.

You may benefit from analog volume control if your headphone amp's analog noise floor, measured in analog domain like dBu or dBV, is significantly lower than your DAC's. JDS Labs for example says the Atom amp has -114 dBu noise.
https://www.jdslabs.com/products/190/atom-amp/

After all, it is an unhealthy gain stage if you constantly need to reduce 70dB, it has nothing to do with analog vs digital at all. If you are not planning to replace the headphone amp and worry about channel imbalance, reduce 30dB on the playback software then do the rest on the headphone amp.
 

bennetng

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#31
Regarding Merging's NADAC volume control and method, details can be found here

https://confluence.merging.com/plug...ADACvolumecontrolperformed,inAnalogorDigital?

As to pro interfaces, they should be in their manuals (I think)
I clicked your link but it showed a blank page, I suppose you were linking to the ESS volume control pdf:
https://confluence.merging.com/disp...ADACvolumecontrolperformed,inAnalogorDigital?

I have another post about that ESS volume control pdf file already:
https://www.audiosciencereview.com/...attenuation-before-power-amp.5844/post-131431
 

Daverz

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#32
I've been running XLR cables from my Auralic Vega (ESS 9018 based, 4V output) directly into a Bryston 3B-SST using the Bryston's 23 dB gain setting. I am already using 3 dB of attenuation on the music server using 64-bit precision in brutefir. That cuts the DAC output to 2.83 V, and the Bryston maxes out 150 W @ 8 Ohms for 2.6 V input on the lower gain setting (actual clipping is probably a bit higher). Speaker sensitivity is in the low 80s.

For my usual diet of orchestral recordings, this is close to ideal: the DAC volume is typically between -20 and -10 dBFS. For late night I might go down to -30 dBFS, which still sounds great. There is the very occasional low-level recording where I run out of gas at 0 dBFS and would have to go to the Bryston's 29 dB gain setting.
 
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andymok

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#33

DKT88

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#34
I just installed and Aune X8 dac that has an analogue volume control between the computer and and a Nord power amp. I wanted a hardware volume control for speaker protection.
 

bennetng

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#35
Well... I searched "volume" in the link:
a.PNG


then I clicked the links and they pointed to the ESS volume control pdf.
b.PNG


As mentioned by Merging, the analog gain is just a 6dB switch and intended to be used to match the sensitivity of other equipment. This approach is being used by many other manufacturers as well. For example ADI-2 Pro FS has 4 switchable gains:
http://www.rme-audio.de/en/products/adi_2-pro_be.php#7

Since it is a post DA analog gain, it won't eliminate the risk of digital clipping before entering the DAC.
 
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