New memberUsername: Wintershade
Post Number: 8
I was hoping someone could more clearly explain what the DacMagic Plus' filters are actually doing... in simple terms that an intelligent guy with no technical audio/engineering training could understand. The descriptions that CA provides aren't particularly helpful. (Link below)
Having spent some time comparing them, here are my impressions (using iTunes ALAC files via USB as a source, listening through Sennheisser HD650s). Can someone please help explain WHY and HOW the filters are changing the musical presentation?
Linear-- Pluses: Music sounds extremely tight and cohesive, like the ensemble has been performing together for decades. The performance has drive, the rhythm draws you in. Very engaging. Minuses: Soundstage isn't particularly expansive, the music seems to be radiating from a relatively small area in the middle of my brain, the Sennheisers sound a bit too warm to the point where the bass is almost a bit muddy and highs lack sparkle
Minimum filter-- Pluses: Soundstage is massive and enveloping, voices each have a clear place coming from all around me, my Sennheisers perk up a bit with a pleasing shimmer in the highs. Minuses: Music lacks the drive that I found so engaging with the Linear filter. the once pleasing highs start to wear on me and I find myself wanting to turn the volume down a smidge, but simultaneously I find myself getting bored and sinking into my seat because the individual voices sound less cohesive (like musicians sloppily playing together for the first time?) and the music fails to draw me in
Steep filter-- A bit of a middle ground between the others. For better or worse, it seems to lack the pluses and/or minuses of both of the others. It's like a jack of all trades, pretty good at everything but not spectacular at anything. On one hand it seems like a good compromise; on the other it leaves me feeling like there's something lacking.
I find I'm mostly listening to the Linear filter. I place a high value on whether or not my playback system helps my music speak to me, engage me, draw me to the edge of my seat. The linear filter does just that--and when I crank the volume a bit and give way to the explosion of music between my ears I quickly forget the filter's faults.
Gold MemberUsername: Magfan
Post Number: 2954
I too have the DM+ but don't obsess about the filters. MOST of my use is with XM/Sirius feeds off my small dish which works out to 32k, half of which is 16 khz and below the frequencies at which the filters have most effect.....
Next time I hook up my computer....USB2....and play some ALAC files, I'll make a point of messing with the filters and see if anything 'jumps out' at me.
I'll post back with any notes......
Platinum MemberUsername: Jan_b_vigne
Post Number: 17570
Explaining digital filters in plain language is not an easy task. It's really essential the reader knows a little bit of the lingo that exists within digital audio so that long explanations aren't either overly complex or missing the point completely due to a simple misunderstanding of what is occurring.
Filters, whether digital or analog in nature, exist to perform the task their name imples - to filter some (unwanted) frequencies away from the location where they would otherwise go by their own volition. Crossover networks in loudspeakers are, or example, a single or a group of filters which direct bands of frequencies away from specific drivers.
It is first important you understand digital audio as modern studios use it is predominantly dictated by the Nyquist Theory which was created well back in the 1930's. Well before audio had progressed sufficiently to test the theory in the real world. One of the most important statements in the theory is how upper frequency extension will be determined in the digital domain. Essentially, Nyquist states that the upper limit of the system's frequency response will be one-half the sampling frequency. In practice this means a digital signal sampled at 44.1kHz (Red Book CD standard) will have its high frequency cut off no higher than 22kHz. Above that frequency the sampling rate itself will introduce digital artifacts which would, for most listeners, be disruptive to the musical signal's quality.
As leo states, filters have caused much of the discussion regarding digital audio to center on the nasties which exist due to the almost universally required presence of a filter just slightly above the 20kHz upper limit we consider the minimum frequency response required for "high quality" music playback. Therefore, between the desired and acceptable 20kHz upper limit for music and the 22kHz "cut off" of digital reproduction the signal must be attenuated by at least -20dB (more is better in some ways) if we are to avoid audible problems commonly associated with digital audio playback. These filters can be constructed through either analog or digital means. While a digital filter is cheap and accurate, it still shares many of the same problems found with analog filters such as those found in a multi-way loudspeaker.
These problems exist primarily, but not exclusively, in the areas of time and phase shift. In digital audio these issues exist beneath the desired filter frequency and within the audible bandwidth. I'm going to assume you have some basic understanding of how time and phase shifts affect the audible perception of music. In short, inaccuracies in time and phase result in a signal which exits the circuit not appearing to be identical to how it entered the circuit. As with loudspeakers which are "time aligned" or "phase coherent", the benefits of having these two values remain consistent from input to output will be, among other things, a more accurate presentation of soundstage, imaging, timbre and ambient details.
The audibility of these values will vary from one listener to the next as time and phase errors are common in many parts of the signal chain. Any portion of the signal path which contains a filter (which could roughly be termed any part of the signal path which has either a cap or an inductor in line) will result in some type of time and phase error. It is, therefore, the rather rare system which maintains time and phase information (along with frequency) even part way through the signal chain. It is really more reasonable to say time and phase are always out of synch and we have simply come to accept these errors as a by-product of conventional music playback. Many of our preferences for "this" component or "that" speaker are the result of the specific equipment meeting - or, at least, minimally offending - acceptable limits in these areas for each listener.
However, since we are ideally looking for a reasonably accurate output which bears some passing resemblance to the input, filters are more or less bad joo-joo in high end audio. The trade offs for doing away with (or even minimizing the number and severity of) filters in a typical system are numerous and most listeners will find their elimination an unacceptable swap for more "hi-fi" values. Therefore, most listeners accept the trade offs of filter use as the cost of having a "good" system.
The closer the filter's "action" is to the audible bandwidth, the more dramatic will be the perceptibility of the filter's effects within the music playback. Since filters also represent some additional impedance within the circuit, filters are known to ring when a signal is presented to their input. This will result in (among other things) poor transient and square wave response which can often be heard as a loss of accurate signal decay and quite possibly a softened or smeared intial attack. At its worst, ringing introduces a hardness and a steeliness to the overall sound quality. These are values which make digital stand out from analog reproduction for many listeners by giving CD its traditonally "cool", "bright" and somewhat false sound.
The vexing issue with any digital audio signal sampled at 44.1kHz is the need for relatively steep filters which remove the upper (non-musical) signals rapidly but, in doing so, result in more severe errors of the sort described above as the filter order becomes increasingly steep. Less steep filters have less audible effects in some ways yet more audible effects in others. (There are a few designs which sort of go all hair shirt and do away with high frequency filters all together on the asumption that any manipulation of the signal is detrimental to the music's quality. Such designs are not common in the consumer market.)
Further listening to digitally sampled music results in the detection of what are termed "aliased frequencies" which exist at lower frequencies. "Aliasing" can in some ways be likened to an analog amplifier's intermodulation distortion component. In both cases the dynamic music signal contains frequencies which
"beat" against another frequency. In an amplifier or gain stage this can simply be two dominant frequencies created by two instruments playing at the same time. The IM distortion component is the creation within the gain stage of a third frequency which is commonly related to the two original frequencies. If this is not comprehensible to you, simply do a search engine for "IM distortion". In digital audio the problem of IM still exists but is now coupled to "aliased" frequencies which are the result of audio signal frequencies beating against the sampling frequency. This results in "ghost" frequencies created by the digital playback circuitry. http://www.ni.com/white-paper/3000/en
If you've paid attention, you'll be aware that the flters typically employed to make digital audio listenable also introduce audible problems which are apparent both above, beneath and at the limits of human hearing. The problems are of several types, all of which will be more or less audible to various listeners. Over the years designers have come up with numerous ways to either suppress the audibility of these problems or to simply place a BandAid over the problems. These "fixes" are what your DAC's filter settings represent. By altering the filters' action - or even eliminating the filters altogether in some designs - the listener can select which problems and which fixes they prefer to address.
Here are Cambridge's descriptions of their filters ... "Linear phase is the same as used in the flagship Azur 840C upsampling CD player. It uniquely features â˜constant group delayâ which delays all audio signals at all frequencies by the same amount meaning all audio is fully time-coherent at the output.
Minimum phase meanwhile does not feature constant group delay but rather the co-efficients have been optimized without feed-forward so that the impulse response exhibits no pre-ringing in the time domain. Some commentators have argued that the pre-ringing as seen in nearly all digital filter designs may affect the transient attack of percussive instruments. Minimum phase implementation eliminates this and is a technology only seen previously in some extremely highend CD playback systems.
Lastly, the steep filter removes close in aliasing artefacts by re-calibrating the co-efficients for a very slight roll off at 20kHz but an ultra steep drop to the stop-band just after 20kHz."; http://www.cambridgeaudio.com/content.php?PID=320&COID=128
Here you'll see yet another description (from another manufacturer) of the most commonly used filter types being employed in today's DAC's; "The names of these filters are mentioned in a white paper and in the data sheet of the new WM8742... These filters are:1.Linear phase; â˜soft knee filterâ
2.Minimum phase; â˜soft knee filterâ
3.Linear phase; brickwall filter
4.Minimum phase; apodising filter
5.Linear phase; apodising filter
Filter #3, the linear phase brickwall filter is the traditional/historical filter for digital audio. It is a steep filter right after 20 KHz and has been shown to produce a lot of pre and post ringing from an inpulse response. Wolfson indicates that this filter is to be used with other filters (from other components such as DSP chips) in the digital audio path. Filter #1 is the "slow roll off" filter that more modern DACs and CD players have used to remove some of the "digital hash" that have been observed in the past. The slow roll off filter reduces both the pre and post ringing of an impulse response. These two filters brickwall or fast roll-off and soft knee or slow roll-off have been the mainstay of digital audio reproduction for all these years.
Filter #2 is a new kind of digital filter. Whereas in the past audio engineers have insisted in phase linearity (meaning all frequencies have equal phase or delay), More recent research have shown that a "minimum phase" filter sacrifices some of the phase linearity (adds some phase distortion) for better time response. Specifically, minimum phase filters minimizes the pre-ringing of an impulse response. Audio researchers have argued that pre-ringing is an un-natural effect and therefore the ear is more sensitive to this kind of distortion. They have also argued that phase distortion is not very audible. This filter also incorporates soft-knee or slow roll-off and this reduces post ringing as well.
Filter 4 and 5 are only available through the s/w interface. "Apodising" ("Apodizing" in American English) filters have been equated to minimum phase filters and minimum phase filters with slow roll-off in the literature. But as the name of the filters in the Wofson DAC suggests, "apodizing" is an additional filter technique to that provided by minimum phase soft-knee.
The use of slow roll-off filters allows some of the higher frequency (beyond the Nyquist frequency) energy to be reflected back into the audio band. This is known as "aliasing" an is a source of distortion. An apodizing filter according to the Wolfson white paper, is one where the filter fully attenuates by Fs/2 (the Nyquist frequency) and thus they start attenuating earlier than Fs/2 often sacrificing flat requency response to 20KHz.
Filter #4 appears to combine the best qualities of these filters. "Minimum phase apodizing" signifies that pre-ringing is eliminiated, post-ringing reduced and aliasing distortion eliminated. As the Wolfson white paper indicates, no one filter is the perfect filter but a designer hopes to makes the best trade offs by using multiple filters.
Conceptually Filter #4, minimum phase apodizing filter, is what Meridian is using in their latest CD player."http://hifiduino.blogspot.com/2009/05/wm8741-digital-filters.html
The filters are similar in both cases with the latter DAC having two additional (apodising) options available to the listener. In the end, the DAC Magic's three filters are attempting to address the issues of; time shift (Linear), phase shift and filter ringing (Minimum Phase) and aliasing distortion (Steep). How these changes are accomplished is a bit beyond the scope of a simple answer to your question. It's enough to know each filter has a fairly predictable effect on the music.
The bottom line here is each filter has a specific effect on the audio signal and will, therefore, have some effect on the listener's preferences. Since no two listeners have completely identical priorities in music reproduction and it would be extremely uncommon for any two systems to be entirely identical and exist within identical environments, each user of the DAC Magic is free to select which filter type best suits their priorities, even down to making changes from CD to CD.
Quite often a DAC's designer(s) have made their selections to compensate for basic system ills. One filter might present a bit softer, warmer and not quite as emphatic which might be more suited to a too forward system slightly tipped up in the highs. Still another filter might have a presentation which would move a somewhat warm, laid back system into the more forward and analytical side.
None the less, the various filters will result in subtle changes in presentation which suggest that the old saw of "bit perfect" transport of data is not all there is to digitally reproduced music. Lately, designers are paying more attention to all data packets arriving "in time". In other words, in the end, bits are not just bits and that's not all there is to digital audio.
Gold MemberUsername: Magfan
Post Number: 2955
A very comprehensive test of the DM+ with emphasis on measures.
And yes, most everyone will agree that good measures do not ensure good sound, as bad measures do not assure bad sound.
Please pay particular attention to the section on filters.
Also, at the 'dawn' of the CD era, some players got away from the 44.1 limit by doing multiple 'oversampling'.....this allowed a shallow analogue filter to be used which had few bad artifacts in the audible passband.
My ancient Phillips / Magnevox was such a player.