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Volume Control between DAC and Power AMP

I am too lazy to audition multiple amplifiers driving my heavy floor standing speakers around town.
That would not enlighten you anyway!

You need the amps on a short list to be in your own home and system for a week or so to assess their suitability. The amps I tried were either bought new (including Benchmark, Micromega) or were demo units from dealers (Lyngdorf, Mark Levinson, Bakoon, etc), or bought used but nearly new (GamuT, Accuphase, NAD, etc).

What surprised me was the fact that these amps sounded so different, despite doing the "simple" job of amplifying an existing analogue signal. Although this testing was a bit of a hassle, it turned out a very worthwhile exercise as it really did demonstrate how much variety in sound delivery there is in 10 solid state amps within a price range where all should sound good. They were of Class A, AB or D, but their class was never a consideration - just how much musical satisfaction they delivered. And all the unwanted ones bought were resold at little or no loss, so the eventual cost was no more than choosing a single amp based on opinions of others or reviews. I'll never do that again, much as I may respect their opinions, although these may well influence my short list.
 
I can change foobar output to use WASAPI, but I rather stay with Benchmark supplied ASIO drivers.

Just to be clear, the device doesn't perform any bit manipulation itself - it is merely an alternative to changing the volume with an attached keyboard (I guess it is mascarding as a keyboard) or did I missed something?
--Edit--
I checked the product spec and indeed it is registering itself as a keyboard and you can even configure which key sequence to send for each operation.
Assuming I leave foobar2000 in focus then all keystrokes will arrive to it so I can configure the gadget to send foobar key binding for vol up/down/mute and next/prev track easily.
Come to think about it, the same could be done with Microsoft surface dial...

I could imagine a bridge device acting as multifunction USB target and initiator- taking the USB audio stream, applying digital attenuation and sending the attenuated data downstream.
You could probably implement the whole thing in a small device embedded in the USB cable like some of the entry level DACs.
It should be powered by USB and can even have a BT receiver to allow controlling it with a remote knob like the surface-dial or something like nuimo (RIP)
I understand. At least what happens in JRiver is that you use the ASIO driver of your device in its settings to choose the audio output, and then use the wasapi on top to redirect everything in Windows through that. You may not need it, but I do as I use Roon but need the audio processing of JRiver.
 
After some more reading I reached the conclusion that I can add miniDSP Flex Digital between the computers and Benchmark DAC3 and it could be used to control volume (digital attenuation) with a physical knob, switch between digital sources and it has a nice display and a remote.
It costs $500 which is less that what will cost me to change the DAC3B to DAC3L and the Display looks much nicer!

Starting on the miniDSP path I opened a can of worms and now thinking about doing XO and maybe even room correction with the FLEX or even SHD...
 
I use a TC Electronic Pilot X to control volume for my PC audio (specs in sig). The Pilot X is compact, but very solid and well built. You can get them new for $60 - $70 from Sweetwater, Amazon, Reverb, among others. Used models can be found on Reverb or eBay for $30 - $50.

907470_PXL_20230911_213539698.png
 
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I use a TC Electronic Pilot X to control volume for my PC audio (specs in sig). The Pilot X is compact, but very solid and well built. You can get them new for $60 - $70 from Sweetwater, Amazon, Reverb, among others. Used models can be found on Reverb or eBay for $30 - $50.

View attachment 384564
Thanks!
It looks real nice, but I decided to add a miniDSP between the music-server and the DAC which will get me the digital attenuation with a physical knob, switch between digital sources and a path for adding subwoofer and doing XO
 
Thanks!
It looks real nice, but I decided to add a miniDSP between the music-server and the DAC which will get me the digital attenuation with a physical knob, switch between digital sources and a path for adding subwoofer and doing XO
Just got to ask... are you absolutely set on having the Benchmark DAC? From what measurements show, the Flex is functionally transparent sound wise...

Reread and saw it's the Flex digital... enjoy your choice! :)
 
After some more reading I reached the conclusion that I can add miniDSP Flex Digital between the computers and Benchmark DAC3 and it could be used to control volume (digital attenuation) with a physical knob, switch between digital sources and it has a nice display and a remote.
It costs $500 which is less that what will cost me to change the DAC3B to DAC3L and the Display looks much nicer!

Starting on the miniDSP path I opened a can of worms and now thinking about doing XO and maybe even room correction with the FLEX or even SHD...
I've tried for several years to incorporate DSP into a Benchmark stack (DAC3B/LA4/AHB2) without adding a second volume control (2.1 system). Many email conversations with Rory eventually led to the Studio/2 DAC 3Bs/LA4/AHB2 with one of the DACs dedicated to the sub (my only source is streaming). I still had two volume controls and it is silly IMO to spend so much $ when a Flex balanced + a well measuring amp of adequate power is likely state of the art and audibly indistinguishable. I like many of us admire the quality and integration of Benchmark components but until they come out with a DSP capable device that integrates into their system I think there are better (more sensible) choices. One of the problems with audiodisease is the tendency to get lost in the weeds and spend $$ on gear with no audible benefit.
PS- would love to hear how you like your new setup
 
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I've tried for several years to incorporate DSP into a Benchmark stack (DAC3B/LA4/AHB2) without adding a second volume control (2.1 system). Many email conversations with Rory eventually led to the Studio/2 DAC 3Bs/LA4/AHB2 with one of the DACs dedicated to the sub (my only source is streaming). I still had two volume controls and it is silly IMO to spend so much $ when a Flex balanced + a well measuring amp of adequate power is likely state of the art and audibly indistinguishable. I like many of us admire the quality and integration of Benchmark components but until they come out with a DSP capable device that integrates into their system I think there are better (more sensible) choices. One of the problems with audiodisease is the tendency to get lost in the weeds and spend $$ on gear with no audible benefit.
PS- would love to hear how you like your new setup

I agree completely with everything you wrote.
I wish Benchmark will add the basic DSP functions to the DAC3 (probably DAC4) with a nice interface like miniDSP/Okto, but I don't see it happening any time soon which means for now miniDSP is the way to go.

Unfortunately, miniDSP doesn't sell in my country which means I need to wait for my next visit to the US to get one (hopefully before XMAS)
 
I've been trying to answer a similar question myself, recently.
I don't think digital attenuation or passive attenuation are good options compared with a preamp, which can be very good value.
The last few years have seen some cracking Chinese headphone / pre amps come to the market.
I'd suggest the SMSL SP-400 or Topping L70, both of which have relay-switched resistor network volume controls.
 
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When I use JRiver with my DACs ASIO driver, both the volume control within JRiver and the Windows volume control are disabled, and a USB volume control won't work for that.

I imagine that's the same for all ASIO drivers.
 
When I use JRiver with my DACs ASIO driver, both the volume control within JRiver and the Windows volume control are disabled, and a USB volume control won't work for that.

I imagine that's the same for all ASIO drivers.
That's not necessary. USB volume control has worked for me with JRiver and various ASIO drivers. There is some variability in the application as all will permit VC within Jriver but only a few will support control of the VC within the DAC (e.g., exaSound).
 
I've been trying to answer a similar question myself, recently.
I don't think digital attenuation or passive attenuation are good options compared with a preamp, which can be very good value.
The last few years have seen some cracking Chinese headphone / pre amps come to the market.
I'd suggest the SMSL SP-400 or Topping L70, both of which have relay-switched resistor network volume controls.
Digital attenuation is fine, if done properly. Roon, for example resamples to 64-bit floating before doing any DSP or digital volume control. That allows a lot of attenuation before there's any risk to audio quality.

That said, relay volume control with precision resistors is also very good.
 
Does Roon resample to 64-bit?
Or just loads the sample into 64-bit format?
 
Hi,
I got Benchmark DAC3-B and recently I ordered Benchmark AHB2 Power Amplifier to replace my Roksan Caspian2 integrated AMP.
I'm using a single source a dedicated music PC running foobar2000->DAC3B->AHB2->Gamut PHI-5
I looking for an advice on how to control the volume:
  1. Control the volume digitally with foobar (the vast majority of my content is 16bit so that leaves 8 bit to control the volume)
  2. Resale my DAC3B and get the DAC3-L with a *digital* PRE function
  3. Use my old Antelope Zodiac+ PRE function with XLR IN/OUT or use the Roksan Caspian2 PRE function (RCA IN / XLR OUT)
  4. Get a dedicated PRO volume control like Goldpoint SA1X/SPL Vol2/Khozmo/...
  5. Get a fully functioning PRE like Benchmark LA4
Here is the system noise performance of various Benchmark Systems:
SystemNoise Performance - DAC3_LA4_AHB2.png

The green curve is the AHB2 power amplifier running in stereo mode with the inputs shorted. This is the output noise of the amplifier with no noise from any upstream devices.

The horizontal axis is the volume control setting. "0" is unity gain between the output of the DAC3 and the input of the AHB2. The AHB2 amplifier will be driven to full output at exactly -2 dB on this particular scale. We designed the system such that the DAC3 would have 2 dB of headroom remaining when driving the AHB2 to full output. So, just remember -2 dB on the horizontal axis is the clip point of the amplifier.

The vertical axis is the noise at the output of the AHB2 power amplifier. It is expressed in dBW. This is dB relative to 1 watt into 8 ohms. It is also dB relative to 2.828 Vrms. The reason for these units is that these are the same units that are used to specify speaker sensitivity. Add the numbers on the vertical scale to your speaker sensitivity and the result is the sound pressure level (SPL) of the noise at 1 meter from your speakers in an anechoic room. This number will be very close to the noise at your listening position in a typical room with two speakers. So just add the vertical scale to your speaker sensitivity to determine the noise SPL at your listening position.

If you have speakers with a 95 dB SPL@1W1m sensitivity, then the very top of the graph would represent the 0 dB SPL threshold of normal hearing. In other words, if your speakers have a sensitivity of 95 dB or less, nothing on this graph will exceed the threshold of hearing. But, if you have very efficient 105 dB horn-loaded loudspeakers, then you will encounter the blue curve and portions of the orange curve.

Now lets compare the various Benchmark system options:
  1. DAC3 HGC connected directly to an AHB2: This is the dark blue curve. Notice that the noise produced by the DAC3 does not change when the volume control is rotated. The DAC3 HGC has a 32-bit digital volume control in front of the D/A converter. Like all digital volume controls, the signal is attenuated but the noise of the D/A converter stays at a constant level. With this combined system, the noise is -102.7 dBW. If you have 95 dB speakers, the noise will be about -7.7 dB SPL at your listening position. This is about 7.7 dB below the threshold of normal hearing, so this noise is absolutely inaudible. If your speaker efficiency exceeds 102.7 dB, you may want to consider adding an LA4 between the DAC3 and the AHB2.
  2. DAC3 B with volume controlled by an upstream device that has a 24-bit digital volume control: This option is also represented by the dark blue curve. The noise at the output of the DAC3 will not be increased if a 24-bit digital volume control is applied before the DAC3. The noise performance when using an external 24-bit volume control is the same as when using the internal 32-bit volume control. The noise is limited by the D/A converter and not by the 24-bit DSP process.
  3. DAC3 with volume controlled by an upstream 16-bit device: This will raise the system noise by about 29 dB. The noise of a 16-bit volume control is 29 dB higher than the noise produced by the DAC3! This 29 dB increase means that the noise at the output of the output of the amplifier would be about -74 dBW (which is 21 dB above the top of the graph). This 16-bit volume control would create noise at a level of 21 dB SPL if you are using 95 dB speakers. If you use a volume control ahead of 16-bit Apple AirPlay or bluetooth, then this is the noise level that you will encounter. Avoid 16-bit volume controls.
  4. DAC3 to LA4 to AHB2: This option is represented by the orange curve. At low volume settings, the LA4 can reduce the system noise by about 8 dB (difference between dark blue curve and orange curve at a gain of -30 dB). Near unity gain, the DAC3 noise starts to dominate the system noise. Note that the LA4 only adds about 0.25 dB of noise to the DAC3 when the LA4 is set to 0dB. As stated above, adding an LA4 may be important if your speaker efficiency exceeds 102.7 dB.
For most users, the real advantage of the LA4 is the exact control of gain that it provides. The display shows exactly where you are relative to your maximum system playback level (assuming a 2 dB amplifier overdrive). It also gives you the very-best volume control available for controlling analog sources (such as the output of a phono preamplifier). The LA4 can be programmed to display the names of connected devices and volume control offsets can normalize all of the inputs to achieve a consistent playback level from all signal sources.

For those of you who were about to ask, the light blue curve is the noise produced by the LA4 and AHB2 combination where the inputs to the LA4 are shorted. Notice that the LA4 only increases the output noise of the AHB2 by 0.6 dB at low gain settings and by just 1.5 dB at unity gain. No other preamplifer is this quiet.
 
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AK10 Astell&Kern has a digital volume control and an LED to indicate optimal pairing with the amplifier.. bought from a Japanese for 80 on eBay
 
Лучше сразу заменить батарею на новую.
 
The nearest to the concept of the discontinued Nuimo controller is potentially the Fosi VOL20 or YouTube.

Maybe not what is needed but an interesting product.
 
The nearest to the concept of the discontinued Nuimo controller is potentially the Fosi VOL20 or YouTube.

Maybe not what is needed but an interesting product.
Thanks! It looks real nice.
I don't think it will work with Foobar2000/Volumio as it manipulating the OS volume directly.
They even say that "...the system volume must not be locked".

If they will supply an open source library for the commands it should be easy to tie them to Foobar/Volumio ...
 
Just for OP @GabrieI's possible reference and interest, especially when you would go into multi-amplifier multichannel world...

I dare to use four HiFi "integrated amplifiers", according to my policy of "right-person-in-right-place", in my DSP-Based multichannel multi-SP-driver multi-amplifier fully active audio setup, where top-upstream JRiver MC acts as master volume controller. You can find the details of my latest setup here #931 on my project thread.

I rather intensively wrote there the pros and merits of analog level relative gain control by "integrated amplifiers", as shown under this spoiler cover;
Here in this post, please let me emphasize again about the pros and merits of relative gain (i.e. tone) control not only in digital domain but also in analog domain using pre-amplifiers or integrated-amplifiers (in my setup). I recently wrote again in my post #56 on a remote thread like these;
Yes, as for safe and flexible tone controls (or I can say "relative gain controls among the multiple SP drivers"), my stance (policy) at least, is that we are encouraged to utilize the "best combination" of "DSP configuration in digital domain" and "analog domain tone controls using HiFi-grade preamplifiers and/or integrated amplifiers".

We need to note (and to respect for) that analog domain tone controls (relative gain controls among the multiple SP drivers) give no effect nor influence at all on the upstream DSP configuration (XO/EQ/Gain/Phase/Polarity/Group-Delay). I believe that this is a great merit of flexible tone controls in analog domain. We know well, on the other hand, in case if we would like to do the "tone/gain controls" only within DSP configurations, such DSP gain controls always affect more-or-less on "XO" "EQ" "phase" and "delay" of the DSP settings which will leads you to possible endless DSP tuning spirals every time; within DSP configurations, XO EQ Gain Phase and Delay are always not independent with each other, but they are always interdependent/on-interaction.

Just for your possible reference, my DSP-based multichannel multi-SP-driver multi-amplifier active system has flexible and safe analog level on-the-fly relative gain controls (in addition to upstream on-the-fly DSP gain controls) for L&R subwoofers, woofers, midrange-squawkers, tweeters, and super-tweeters, all independently and remotely. My post here shows you a typical example case for such safe and flexible on-the-fly analog-level tone controls. This my post (as well as
this post) would be also of your interest.

Of course, I know well that I (we) can also perform such relative gain control using DAC8PRO’s 8-channel output gain controllers. I do not like, however, to change the DAC8PRO’s output levels frequently on-the-fly (while listening to music) due to safety and inconvenience concerns; I like to keep DAC8PRO’s analog out gain level always at constant -4 dB which should remain to be usually “untouchable” in my case.

One of the very unique aspects/features of my multichannel audio rig is that I fully utilize four HiFi-grade “integrated amplifiers” plus L&R active subwoofers, each of them have its own gain (volume) controller for safe and flexible relative gain (tone) control in analog domain even on-the-fly i.e. while listening to music.

In this perspective, my posts #438 and #643 should also give you better understandings. Furthermore, my posts #317(remote thread), #313(remote thread) would be also of your reference and interest.
For safety concerns, my post here #204(remote thread) should be also well noted and understood.
As shown in my diagram, all the parameters in EKIO can be modified/tuned on-the-fly while listening to music.
Fig03_WS00007533 (2) (1).JPG

EKIO also have ABX comparator functionality for comparison of two set of saved configuration files.

I usually do not like, however, to modify the DSP parameter(s) on-the-fly mainly for safety reasons; if I would unintentionally mistype 20 dB gain-up by keyboard instead of correct 2.0 dB, it may harm/destroy my treasure SP drivers, especially the Beryllium tweeters and/or metal horn tweeters; in this case, it is better to use mouse wheel up/down, not by keyboard numeric input, for on-the-fly gain tuning.

After my decision of "the simpler, the better" EKIO configuration (fine EQs included, if needed), therefore, I usually use four HiFi integrated amplifiers for further "coarse" but excellent enough relative gain (tone) controls.

Furthermore, you would please visit my post here #311 for the summary of my long journey of amplifier selection exploration.
 
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