It all depends on what is meant by a "passive attenuator". If you take the full-scale analog output of your audio interface, put it through a genuine passive attenuator (e.g. the SM Pro Audio M-Patch2) and then into an active loudspeaker, you have active components driving the attenuator and active components after the attenuator (the amp in the loudspeaker), but the actual element that controls the volume is a passive unit.

In my mind, there are two main instances where an attenuation function is not passive. The first is in the gain control of a pre-amp, where at least part of the control is universally performed by varying the feedback round an amplifier. In this case, the amplifier transfer function is modified by varying the feedback, so the tonality is affected, even if it's not particularly audible. Note, however, that most high-quality pre-amps have a combination of passive attenuation on the front end and active feedback control to avoid too great a change of tonal characteristics over the pre-amp's gain range. The second case is the one you referred to, and that is changing the amplitude of a signal digitally before it is converted to an analog signal by a D-A converter. In this case, the analog path has a fixed gain and the digital amplitude control does not change that.

The equipment designer has a problem when it comes to needing a blend control in an audio interface. The blend control adjusts a headphone monitor output by varying the ratio of signal from the input channels to outgoing signal, and this can be used to avoid latency effects during tracking. Leaving aside for a moment the problem of what to do when you have more than two input channels, the question comes as to where and how the blend function is implemented, given that it needs to be taken in conjunction with an output volume control. As it happens, I've been a design consultant for several of these audio interface units from different manufacturers, and blend/monitoring/volume is a topic that often produces the most differences of opinion in the design teams.

So the answer to your question is that different audio interfaces will do it different ways, depending on the design constraints including cost and functionality. What you can normally tease out is whether in any given unit the knob generates a d.c. voltage that is used to control volume and/or blend operations digitally with a fixed analog output gain, or whether there is a stereo potentiometer with real analog signals on it. I can't give you a list of which interfaces do what.

Boswell, post: 408185 wrote: It all depends on what is meant by a "passive attenuator". If you take the full-scale analog output of your audio interface, put it through a genuine passive attenuator (e.g. the SM Pro Audio M-Patch2) and then into an active loudspeaker, you have active components driving the attenuator and active components after the attenuator (the amp in the loudspeaker), but the actual element that controls the volume is a passive unit.

In my mind, there are two main instances where an attenuation function is not passive. The first is in the gain control of a pre-amp, where at least part of the control is universally performed by varying the feedback round an amplifier. In this case, the amplifier transfer function is modified by varying the feedback, so the tonality is affected, even if it's not particularly audible. Note, however, that most high-quality pre-amps have a combination of passive attenuation on the front end and active feedback control to avoid too great a change of tonal characteristics over the pre-amp's gain range. The second case is the one you referred to, and that is changing the amplitude of a signal digitally before it is converted to an analog signal by a D-A converter. In this case, the analog path has a fixed gain and the digital amplitude control does not change that.

The equipment designer has a problem when it comes to needing a blend control in an audio interface. The blend control adjusts a headphone monitor output by varying the ratio of signal from the input channels to outgoing signal, and this can be used to avoid latency effects during tracking. Leaving aside for a moment the problem of what to do when you have more than two input channels, the question comes as to where and how the blend function is implemented, given that it needs to be taken in conjunction with an output volume control. As it happens, I've been a design consultant for several of these audio interface units from different manufacturers, and blend/monitoring/volume is a topic that often produces the most differences of opinion in the design teams.

So the answer to your question is that different audio interfaces will do it different ways, depending on the design constraints including cost and functionality. What you can normally tease out is whether in any given unit the knob generates a d.c. voltage that is used to control volume and/or blend operations digitally with a fixed analog output gain, or whether there is a stereo potentiometer with real analog signals on it. I can't give you a list of which interfaces do what.

Thanks, appreciate the detailed reply. I'd certainly like to know more about the last points you made in relation to the audio interfaces I'm considering for my new setup.

Your quote: "The second case is the one you referred to, and that is changing the amplitude of a signal digitally before it is converted to an analog signal by a D-A converter. In this case, the analog path has a fixed gain and the digital amplitude control does not change that."

Yes, it's more this that I want to try to get right in my new setup, in particular, from the computer & audio interface to the studio monitors.

It's more what the software is doing before it gets to the audio interfaces output D-A converter that I'd like to know.

There's a detailed paper on this and Windows somewhere but I couldn't find it today (been a while since I looked at it). However, what I'm meaning is hinted at on this page - [[url=http://[/URL]="http://msdn.microso…"]Session Volume Controls (Windows)[/]="http://msdn.microso…"]Session Volume Controls (Windows)[/] (though I'm referring to Windows 7 pro 64 bit specifically)

So the analog path has a fixed gain. But, if you take the E-MU 1212m, the control in the mixer software which they call "monitor" just controls the windows system volume. But this isn't non-destructive, as the API shows. When reduced to a lower volume level that way, it's fewer bits. And that's taking away from what I'd be trying to hear, if you see what I'm getting at. So, yea, it's going down the analog path at a fixed level, but fewer bits are going into the d-a converter, as defined by the APIs implementation.

I first picked up on this by accident after hearing output from the EMU card direct to monitors vs EMU -> Mackie Big Knob -> Monitors. I'd had EMU volume at 100% going via the Mackie, but when going directly from card to monitors, I noticed how I could no longer hear certain parts or nuances in the music at lower volumes.

I decided to change the Mackie, though, as it did colour the sound, and although it wasn't too bad, I needed it out of my signal/sound path.

I used the passive M Patch instead and compared the two again. And sure enough, at lower volumes setup directly from card to monitors, there are aspects I could not hear, which bothered me, as I like to mix at lower volumes too.

Going EMU Card -> MPatch -> Monitors, I set the EMU volume (or system volume!) at 100% and put the volume down low on the MPatch and found more detail. With some songs etc, you really have to listen carefully but I found it more obvious with certain music. There was a clear difference.

I can obviously just have the same setup again with a new interface, go New Interface -> MPatch -> Monitors and run the audio interface at 100% volume, but I'd ideally like to remove the MPatch if I don't need it in the chain.

I've noticed too that some software (e.g. Focusrite) has a monitor control in the software but then a dial/knob on the external device too. When I tried this in a store, the two did not seem to be linked... I could turn the one in the software right up and then turn the one on the box down independently, so what is happening there/how are they implemented etc?

I can see how it might be seen as a non-point, but actually, I think lacking bits is just as bad as coloration, and that this missing audio detail at low volume levels is important, especially if you mix at low - mid levels

I have to admit... I've no idea which interface to go for now :-(

I own a Focusrite Liquid Saffire 56. Altought I don't have de schematics that shows if the knob affect digital levels or an analog circuit. I tend to think it affects the digital because the software knob can be affected independently. but when you touch the physical know the software follows.

BTW, This is stated in the manual : (All digital output levels are unaffected by Liquid Saffire 56 control. Use the output levels of the DAW to control digital output levels.)

In any case, I always mix at low level and just spot check at hi level... It always sound better when it's louder, so if you make it sound nice at low volume you're in a win situation !