I recently bought a sound card (Xonar D1). I'm using Linux (3.2.0.) The drivers of the D1 offer a mixer control element titled "DAC Filter" and it has two settings: "Slow Roll-off" and "Fast Roll-off". Does anyone have an idea of that is and what it does?
Judging from the name of this setting, it selects between two different modes of operation for the DAC's internal interpolation filter. Technically, there should be no audible difference, since this setting only affects filter response in the ultrasonic range. Virtually the only case when this could result in an audible difference is when the reproduced recording contains high amount of ultrasonics and analog part of the reproduction chain is highly non-linear.
From theoretical standpoint, fast roll-off is the best option.
In practice, you have to perform double-blind listening test to determine whether this option audibly affects the sound in your particular setup.
DACs need a lowpass filter at fs/2 to remove qantization noise. This analog filter can have different shapes. A very steep filter is what is normaly used. Some DAC designers offer a "Slow" version that promises better impulse response. My DAC has such and option and claims -3dB at 18.2khz while the standard filter has no drop to 20khz. One may argue that if you hear a difference between filters it simply is that absense of some highs and not the bettered shown on paper impulse response. The impulse response only betters above the cutoff frequency and shows less so called ringing there AFAIK so won´t be of any benefit.
Thanks for the info. I can't hear any difference whatsoever between slow and fast. So I'll just leave it at the default (which is the "fast" setting.)
I take this as an excuse to rave about transient response (http://en.wikipedia.org/wiki/Transient_response)...
Audio is normally upsampled before being fed into the DAC so as to make the conversion easier. This requires a low pass filter, and that filter has some kind of roll-off characteristic between the pass band and the stop band.
Some idealise the filter as having a perfect brick-wall cutoff -- an instantaneous transition from pass to stop levels -- and the best approximation of that is the steepest roll-off you can manage. This maximises the possible width of the pass band by allowing you to start rolling off much closer to the start of the stop band (close to the Nyquist frequency). However, this kind of transfer function can ring (http://en.wikipedia.org/wiki/Ringing_artifacts#sinc_filter) near transients; both before and after.
If you don't want ringing, then you can use a shallower roll-off (http://en.wikipedia.org/wiki/Ringing_artifacts#Low-pass_filter). The transition band becomes wider and that necessarily eats into the pass band, so you get a smaller pass band and more high-frequency loss.
These trade-offs are actually the other way around from the more general fact that a wider pass band improves transient response by allowing ringing to settle more quickly. Here I'm saying that you can improve transient response with, effectively, a narrower pass band. Just not too much narrower. It's all about compromise.
You can also mess around with phase, but I'm not going to touch the psychoacoustic debate around that. Presumably both your options have linear phase.
Slower roll-off is also achievable with less computation, and it introduces less delay. I doubt either of these matter here.
For 44kHz content the ringing (when present) will be around 22kHz, and probably not directly audible; but audio hardware will still try to reproduce it and then you might worry about nonlinearities and/or your dog. In any case, it depends on the content, as you'll need content capable of triggering the ringing. I've been told that such content was endemic in the early days of CD.
Conversely, even with a wide transition band the region of activity should still all be ultrasonic, and so the information you save with a steep roll-off is not necessarily more important than the noise you eliminate with a shallow roll-off.
If the drivers allow you to play back 8kHz audio without resampling it then you can bring the artefacts down into the audible band and try both settings. When you are dealing with a perceptible Nyquist frequency the problem is highly subjective and debatable, much like image sharpening.
See also:
- http://sox.sourceforge.net/sox.html#EFFECTS (http://sox.sourceforge.net/sox.html#EFFECTS) (see: rate effect, after "NB")
- http://en.wikipedia.org/wiki/Gibbs_phenomenon (http://en.wikipedia.org/wiki/Gibbs_phenomenon)
[edit]delete probable myth
That was way more information that I hoped to get. Thanks The card (or the driver) doesn't accept 8kHz (32kHz is the lowest it can do), so I guess I will never be able to hear ringing artifacts.
For 44kHz content the ringing (when present) will be around 22kHz, and probably not directly audible; but audio hardware will still try to reproduce it and then you might worry about nonlinearities and/or your dog. In any case, it depends on the content, as you'll need content capable of triggering the ringing. I've been told that such content was endemic in the early days of CD.
With the word "content" here do you refer to the actual stream of samples, which could, in those early days, be affected by sampling frequency jitter due to the lower build precision of first generation AD converters? So does this imply that a large amount of jitter could trigger ringing? And so, might we deduct that the different roll off of the filter owned by the OP is aimed at the rather audiophile quest of reduction of jitter injected along the decoding chain?
P. S. just joking a little before bedtime...
The card (or the driver) doesn't accept 8kHz (32kHz is the lowest it can do), so I guess I will never be able to hear ringing artifacts.
32kHz might be close enough; that's a 16kHz cutoff. However, you can't do the test blind. As Wombat points out, the perceptible difference may simply be the loss of some high-frequency. If you could hear that then you'd know which was which, and then you're required to make a subjective determination on which is more pleasant without the benefit of science goggles.
With the word "content" here do you refer to the actual stream of samples, which could, in those early days, be affected by sampling frequency jitter due to the lower build precision of first generation AD converters? So does this imply that a large amount of jitter could trigger ringing? And so, might we deduct that the different roll off of the filter owned by the OP is aimed at the rather audiophile quest of reduction of jitter injected along the decoding chain?
You know what? I'm going to take you seriously.
Jitter wouldn't trigger the ringing because steep filters are a digital matter. However, if the ringing were present then jitter would effectively frequency-modulate it, and that could spray the noise all over the audible spectrum.
The card (or the driver) doesn't accept 8kHz (32kHz is the lowest it can do), so I guess I will never be able to hear ringing artifacts.
32kHz might be close enough; that's a 16kHz cutoff. However, you can't do the test blind. As Wombat points out, the perceptible difference may simply be the loss of some high-frequency. If you could hear that then you'd know which was which, and then you're required to make a subjective determination on which is more pleasant without the benefit of science goggles.
So I guess that for all practical intents and purposes, it doesn't matter which filter setting is used (since I set 44.1kHz for music listening and 48kHz for watching DVD material; I don't see a need for 32kHz for anything.)
Unless we consider the neighbor's dog. So I've set it to "Slow Roll-off" in case it's an audiophile dog
DACs need a lowpass filter at fs/2 to remove qantization noise. This analog filter can have different shapes. A very steep filter is what is normaly used. Some DAC designers offer a "Slow" version that promises better impulse response. My DAC has such and option and claims -3dB at 18.2khz while the standard filter has no drop to 20khz. One may argue that if you hear a difference between filters it simply is that absense of some highs and not the bettered shown on paper impulse response. The impulse response only betters above the cutoff frequency and shows less so called ringing there AFAIK so won´t be of any benefit.
These re-construction/anti-imaging filters are normally analog Butterworth filters? I think the OP said his digital mixer has the option of the two different settings, so a digital filter would be used, then the analog filter after the output DAC?
DACs need a lowpass filter at fs/2 to remove qantization noise. This analog filter can have different shapes. A very steep filter is what is normaly used. Some DAC designers offer a "Slow" version that promises better impulse response. My DAC has such and option and claims -3dB at 18.2khz while the standard filter has no drop to 20khz. One may argue that if you hear a difference between filters it simply is that absense of some highs and not the bettered shown on paper impulse response. The impulse response only betters above the cutoff frequency and shows less so called ringing there AFAIK so won´t be of any benefit.
These re-construction/anti-imaging filters are normally analog Butterworth filters? I think the OP said his digital mixer has the option of the two different settings, so a digital filter would be used, then the analog filter after the output DAC?
My DAC has this "slow" option for its digital filter, yes. These are simply set over a register. The analog output stage itself acts like another, very slow LPF.