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DAC attenuation before power amp

bennetng

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Hello bennetng,

I finally got round to doing this test. To my surprise the volume control in the MiniDSP NanoAVR did not pass the test.

This was a big surprise, because I assumed that the 32bit floating point SHARC processor would not work this way.
I don't have any MiniDSP product, but please don't think that you are being cheated by the company.

Every step in the signal chain needed to be in float until volume change, so if the file decoder, playback software, driver, firmware etc... even if one of them is not working in float before reaching the float capable DSP processor, then clipping will occur.

Therefore I always recommend adjusting digital volume in software as early as possible.
 

andreasmaaan

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Every step in the signal chain needed to be in float until volume change, so if the file decoder, playback software, driver, firmware etc... even if one of them is not working in float before reaching the float capable DSP processor, then clipping will occur.

@bennetng would you mind please explaining why this is? I'm not sure I understand, I would have thought that as long as no clipping occurs prior to the SHARC processor, there would be no need to keep the whole chain in 32-bit float. Could you explain why that's not the case please?
 

Nick Laslett

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Just to satisfy my curiosity, I opened up the NanoAVR Bass Management app on my laptop, to see if this would make any changes to the clipping in the file.

In my simple understanding I assumed lowering the dB in the routing table might have an effect. This is not something that can be easily accessed, so it is not really viable as a volume control. This had no effect to the clipping.

I’m obviously out of my depth here. Just re-reading your reply I wonder what the Oppo 203 is doing before sending the audio onto the NanoAVR. Could the clipping be baked in here? I will try some other sources.
 

bennetng

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@bennetng would you mind please explaining why this is? I'm not sure I understand, I would have thought that as long as no clipping occurs prior to the SHARC processor, there would be no need to keep the whole chain in 32-bit float. Could you explain why that's not the case please?
Try this:
https://www.audiosciencereview.com/...attenuation-before-power-amp.5844/post-131431

Here is an updated version of the calculator, in the attachment:
https://www.audiosciencereview.com/...hould-i-set-my-dac-to.8956/page-4#post-255104

Search for "Realize that even if we don't use any DSP features" in this article:
https://archimago.blogspot.com/2019/06/guest-post-why-we-should-use-software.html

Actually, many lossy files have 0dBFS+ data, and they are clipped during playback, but the clipped amount is not as dramatic as my intentionally tweaked file so it is less noticeable or unnoticeable. So some people can ABX something dosen't automatically mean the bitrate is not high enough or the psychoacoustic algorithm is not good enough, or someone has better hearing than others.

Lossy formats, as well as lossless float-capable formats like wavpack can store data way above 0dBFS, a fixed point data path will clip those data, as long as the MSB is aligned to 0dBFS.
 

bennetng

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Just to satisfy my curiosity, I opened up the NanoAVR Bass Management app on my laptop, to see if this would make any changes to the clipping in the file.

In my simple understanding I assumed lowering the dB in the routing table might have an effect. This is not something that can be easily accessed, so it is not really viable as a volume control. This had no effect to the clipping.

I’m obviously out of my depth here. Just re-reading your reply I wonder what the Oppo 203 is doing before sending the audio onto the NanoAVR. Could the clipping be baked in here? I will try some other sources.
I believe the driver is usually the weakest link, for example, see RMAA's website:
http://audio.rightmark.org/index_new.shtml
The headline:
added Pro Tools HD/HDX compatibility, that uses Float32LSB ASIO I/O format

Which is in fact, very rare. You can visit iXBT, which has a deep connection with the development of RMAA, for example this review:
https://prosound.ixbt.com/interfaces/rme-ufx-plus.shtml

Scroll down to the ASIO capability table, you will see a lot of "Int32LSB", which means fixed point.

WASAPI perhaps can't do such a thing as well.
 

pma

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This is my result of analog x digital volume control. DacMagic Plus used in the test. The volume set to -40dB once as a digital attenuation (red), and second time by volume control of a low noise linear preamp (DAC set at full volume this time). Output signal amplified by a 30dB gain measuring amplifier in both cases to get above measuring system intrinsic noise. The result is below. Analog volume control has both less noise and better linearity.

1581754425951.png
 

bennetng

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This is my result of analog x digital volume control. DacMagic Plus used in the test. The volume set to -40dB once as a digital attenuation (red), and second time by volume control of a low noise linear preamp (DAC set at full volume this time). Output signal amplified by a 30dB gain measuring amplifier in both cases to get above measuring system intrinsic noise. The result is below. Analog volume control has both less noise and better linearity.

View attachment 50250
We are talking about digital clipping prevention here. Analog volume control can't prevent digital clipping. I don't object the use of analog volume control per se.
 

Nick Laslett

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I reached a bit of a mental block. I have to feed the NanoAVR a PCM signal over HDMI. So trying another source seems a bit pointless. My assumption is that the decode to PCM bakes in the clipping. In another thread you discussed how opening the file in Audacity is not relevant to the specific test due to potential conversion issues.

So the volume on the NanoAVR could be okay.
 

Panelhead

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I purchased an integrated amp recently. It has analog and digital inputs. After the input selector, a hardware electronic switch, the input signal is routed to a driver to a MAS6116 digital volume control. From this output to a driver for the power amp section.
The MAS6116 looks to function as an eight bit dac. It has 256 steps of 0.5 dB. But has analog input and output. Read the data sheet, Google searching yielded a Schiit preamp and a couple other products.
The data sheet specs look good. Is this a better option than the volume control built into the dac chip of the interface?
I apply a -6d attenuation in Pure Music to address potential digital clipping in the interface. Benchmark explains this well in a tech sheet.
 

yodog

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Would a -3db, -6db, or 12db attenuator be optimal if the power amp has 29db of grain level and needs 1.5vrms to reach full power with it being fed a full 2vrms signal?
 

Panelhead

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You should not need any attenuation. Just do not go into 0 dB output on the dac.
 

yodog

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You should not need any attenuation. Just do not go into 0 dB output on the dac.

Would that be like full volume?

I have a bluesound node 2i and I use a separate preamp that’s fed it’s signal via the rca preouts on the node 2i which means it uses its internal DAC. I set it to fixed output mode and I believe that’s literally max volume. Is this not ok?
 

bennetng

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Would that be like full volume?

I have a bluesound node 2i and I use a separate preamp that’s fed it’s signal via the rca preouts on the node 2i which means it uses its internal DAC. I set it to fixed output mode and I believe that’s literally max volume. Is this not ok?
Depends on the audio files. For example the track on the right may clip if you are sending max volume to the DAC.
Image1.png
 
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