John Atkinson: Yes, MQA IS Elegant...

[crenca]

2 hours ago, Jud said:

 

The fact that MQA uses minimum phase to eliminate pre-ringing doesn't really affect whether it's good or bad when used by anyone else.  Regarding "distorting," unless your speakers have linear phase crossovers, which makes it quite difficult for them to have flat frequency response, I'm not sure that will be audible, at least to someone without specific training.  @Archimago offered blind tests at his site, and the folks who took the test did no better than chance.

 

I do have speakers with linear phase crossovers, and I fancy I might be able to hear the difference.  I've even passed a couple of (literally two) self-administered blind tests.  But I'm not confident I could pass additional blind tests. 

 

1 hour ago, Jud said:

 

That's interesting, since one might think the opposite - it's one of the reasons minimum phase filters are used, to avoid pre-ringing that is purported to "smear" transient response.

 

Most of my listening (must be >90% now) is HP's, so no crossover (though I tend to use software crossfeed - a complicating factor?).  I am liking the sound of upper frequencies like cymbal, brass, etc. now that I am using  linear or "zero" phase most of the time, they seem to bring these out in a more accurate way, whereas there is a "smear", or softness when I use minimum phase.  I also think strings seem more subtly accurate as well. By upsampling everything to DXD and using the quality filters in HQPlayer, I am probably getting just about the best out of linear phase.  All this is andoctotal only...

[Jud]

41 minutes ago, crenca said:

 

 

Most of my listening (must be >90% now) is HP's, so no crossover (though I tend to use software crossfeed - a complicating factor?).  I am liking the sound of upper frequencies like cymbal, brass, etc. now that I am using  linear or "zero" phase most of the time, they seem to bring these out in a more accurate way, whereas there is a "smear", or softness when I use minimum phase.  I also think strings seem more subtly accurate as well. By upsampling everything to DXD and using the quality filters in HQPlayer, I am probably getting just about the best out of linear phase.  All this is andoctotal only...

 

Yeah, I can't tell with headphones (though admittedly I haven't tried a lot).  The differences I hear or imagine I hear with speakers have to do with soundstaging and location within the soundstage, and that's totally different with my headphones.

[vl]

47 minutes ago, crenca said:

 

 

Most of my listening (must be >90% now) is HP's, so no crossover (though I tend to use software crossfeed - a complicating factor?).  I am liking the sound of upper frequencies like cymbal, brass, etc. now that I am using  linear or "zero" phase most of the time, they seem to bring these out in a more accurate way, whereas there is a "smear", or softness when I use minimum phase.  I also think strings seem more subtly accurate as well. By upsampling everything to DXD and using the quality filters in HQPlayer, I am probably getting just about the best out of linear phase.  All this is andoctotal only...

 

I experience the same with my LCD2 headphones.  A good pair of HPs usually have better amplitude and phase characteristics than a two way monitor of standard design.  With Dirac Live my speakers and HPs have very similar sonic characteristics.  

 

Good upsampling goes beyond multiplying the sampling rate.  The better upsamplers use polynomial curve fitting, probably for both upping the sampling rate and extrapolating 16 bits to 24 bits.  There is some art involved.  Most upsamplers, regardless of cost, sound a little sterile.  I have come across two upsamplers that are noticeably more musical.  They come in the Auralic Vega and the Cambridge Audio DACMagic Plus.  The latter is about 10% the price of the former but it gets 90% of the sonic quality when playing CDs.  The Vega upsamples to 1500 MHz and the DACMagic Plus to 384.  The higher upsampling help suppress the aliased byproducts in D to A conversion.  A very powerful DSP is needed to do a good job.

[vl]

22 minutes ago, Jud said:

 

Yeah, I can't tell with headphones (though admittedly I haven't tried a lot).  The differences I hear or imagine I hear with speakers have to do with soundstaging and location within the soundstage, and that's totally different with my headphones.

 

Yes.  HPs have different imaging and sound staging than speakers in a normal room, as the room tends to impart its characteristics on the sound.  However the HPs and speakers can sound similar in frequency response, transient, details, reverb, etc.

[Fokus]

11 hours ago, Jud said:

 Minimum phase filtering is widely used in audio, including by nearly all room equalization software.

 

That is because room equalisation aims at correcting (mostly) minimum phase problems. You don't want to use linear phase for that.

 

[mcgillroy]

14 hours ago, vl said:

 

Compare the sound from the Sony Classical CDs to the classical music CDs from Decca and EMI.  The upper two octaves are usually far closer to live sound in the Sony while the other two brands have their characteristic house sound that is often bright and harsh.  

 

 

Slightly OT but can you demonstrate this? Which CDs to compare and do you have any graphs too?

[Le Concombre Masqué]

12 hours ago, Jud said:

 

Here - a linear phase and a minimum phase filter.  See how there's no pre-ringing to possibly smear the initial transient in the minimum phase filter?

 

 

I'm not sure definitions are the same for Room correction softwares ; if I recall correctly minimum phase is there defined kind of path of least resistance with the shortest transfer time through the system. What I know for sure is that when you flatten your phase curve to 0 you can flatten (preferably with a downward slope) your frequency curve and that's awesome. 

I think that RePhase outputs linear phase convolution filters ; I very recently discovered I prefer linear phase versions of HQP filters in conjunction with my rePase convolution filters that yield to this :

[Jud]

53 minutes ago, Le Concombre Masqué said:

I'm not sure definitions are the same for Room correction softwares ; if I recall correctly minimum phase is there defined kind of path of least resistance with the shortest transfer time through the system.

 

That is indeed the same as minimum phase in the sample rate conversion software I showed.  The shortest possible phase delay varies by frequency, with the result that minimum phase filters are "dispersive" and have "group delay," meaning the time for the components of a given signal (such as music) to pass through the filter varies by frequency.  A linear phase filter means all frequencies have the same delay, but the time to pass through the filter can only be as short as the minimum time for the frequency that takes the longest.

 

Minimum phase filters have the property I described and showed in the graphs with respect to ringing, that is, they push all ringing energy to a point after impulses such as those used in the test.

 

Dirac's philosophy, insofar as they have revealed it and that can provide any enlightenment, can be found on pages 15 -16 here:

 

https://static1.squarespace.com/static/55c8a274e4b09cb562cd3ea0/t/57d159a69de4bbd9d306ac8f/1473337780806/Dirac+Room+Correction+(Audio+Engineering+Society%2C+AES+Sweden+lecture).pdf

[crenca]

46 minutes ago, Jud said:

 

That is indeed the same as minimum phase in the sample rate conversion software I showed.  The shortest possible phase delay varies by frequency, with the result that minimum phase filters are "dispersive" and have "group delay," meaning the time for the components of a given signal (such as music) to pass through the filter varies by frequency.  A linear phase filter means all frequencies have the same delay, but the time to pass through the filter can only be as short as the minimum time for the frequency that takes the longest.

 

Minimum phase filters have the property I described and showed in the graphs with respect to ringing, that is, they push all ringing energy to a point after impulses such as those used in the test.

 

Dirac's philosophy, insofar as they have revealed it and that can provide any enlightenment, can be found on pages 15 -16 here:

 

https://static1.squarespace.com/static/55c8a274e4b09cb562cd3ea0/t/57d159a69de4bbd9d306ac8f/1473337780806/Dirac+Room+Correction+(Audio+Engineering+Society%2C+AES+Sweden+lecture).pdf

 

That is how I understand it as well.  The "time to pass through the filter can only be as short as the minimum time for the frequency that takes the longest" as I understand it is technically called "latency", and the only folks who seem to worry about it are HT types who want perfect alignment of the sound with the image on the screen.  Perhaps I am missing something, but in an otherwise normal musical playback situation I don't believe latency matters (am I wrong?) 

 

"On paper" then, what is the argument for "minimum" phase filter?  Why would you want a filter that had the effect of causing the phase of the frequencies (varies by frequency - I believe upper frequencies are "delayed" compared to lower...or do I have this backwards?) to be misaligned?  The two arguments as I understand it are that you are (A)  using the phase misalignment of the filter to "correct" or counteract phase misalignment in some other part of the playback chain (e.g. caused by room or speakers).  Then there is (B), which is "ringing" and its alleged audible effects.  

 

For myself with my normal HP listening (A) should normally be a non-factor.  (B) is "played up" in Audiophiledom.  Thing is, as near as I can tell, with what we know about "good" filter design (anyone can point to a situation where ringing becomes audible in a extreme EQ adjustment, or a poor filter design, or an impulse that by definition is a "step" function outside of the audioband and thus "breaks" the filters normal behavior), it rests on dubious assertions.  

 

As someone upstream said so much attention is given to this phenomenon in Audiophiledom, this "transient" behavior.  Yet, as we all know Audiophiledom has a habit (indeed is built upon) the ghostly edge phenomenons where the small becomes the large technically (allegedly) and important to $sells$.  In other words, "ringing" while real in a sense, appears to be leveraged in the audiophile confidence game for its own ends.  What am I missing?

[Shadders]

14 minutes ago, crenca said:

 

That is how I understand it as well.  The "time to pass through the filter can only be as short as the minimum time for the frequency that takes the longest" as I understand it is technically called "latency", and the only folks who seem to worry about it are HT types who want perfect alignment of the sound with the image on the screen.  Perhaps I am missing something, but in an otherwise normal musical playback situation I don't believe latency matters (am I wrong?) 

 

"On paper" then, what is the argument for "minimum" phase filter?  Why would you want a filter that had the effect of causing the phase of the frequencies (varies by frequency - I believe upper frequencies are "delayed" compared to lower...or do I have this backwards?) to be misaligned?  The two arguments as I understand it are that you are (A)  using the phase misalignment of the filter to "correct" or counteract phase misalignment in some other part of the playback chain (e.g. caused by room or speakers).  Then there is (B), which is "ringing" and its alleged audible effects.  

 

For myself with my normal HP listening (A) should normally be a non-factor.  (B) is "played up" in Audiophiledom.  Thing is, as near as I can tell, with what we know about "good" filter design (anyone can point to a situation where ringing becomes audible in a extreme EQ adjustment, or a poor filter design, or an impulse that by definition is a "step" function outside of the audioband and thus "breaks" the filters normal behavior), it rests on dubious assertions.  

 

As someone upstream said so much attention is given to this phenomenon in Audiophiledom, this "transient" behavior.  Yet, as we all know Audiophiledom has a habit (indeed is built upon) the ghostly edge phenomenons where the small becomes the large technically (allegedly) and important to $sells$.  In other words, "ringing" while real in a sense, appears to be leveraged in the audiophile confidence game for its own ends.  What am I missing?

Hi,

From the miniDSP sweb site :

REW uses IIR filters, while Dirac Live uses mixed-phase filtering - in effect, a combination of IIR and FIR filters. FIR filters are more powerful than IIR filters, but more expensive to implement. See the app note FIR vs IIR filtering for information on the differences between these two different types of digital filter.

(https://www.minidsp.com/applications/digital-room-correction/dirac-live-vs-rew)

 

I have not studied this, but one aspect that immediately occurs is that to implement room correction in regards to amplitude you will be splitting up the audio band into many smaller bands - so the gain of each band can be modified to achieve an overall flat response.

 

The band pass filters will always have out of band energy applied to them - and this will always be in the hearing frequencies (20Hz to 20kHz).

 

Therefore filters with ringing are a bad idea - you will need filters with no ringing.

 

Someone else may be able to confirm this.

 

Regards,

Shadders.

[Rt66indierock]

6 minutes ago, Shadders said:

Hi,

From the miniDSP sweb site :

REW uses IIR filters, while Dirac Live uses mixed-phase filtering - in effect, a combination of IIR and FIR filters. FIR filters are more powerful than IIR filters, but more expensive to implement. See the app note FIR vs IIR filtering for information on the differences between these two different types of digital filter.

(https://www.minidsp.com/applications/digital-room-correction/dirac-live-vs-rew)

 

I have not studied this, but one aspect that immediately occurs is that to implement room correction in regards to amplitude you will be splitting up the audio band into many smaller bands - so the gain of each band can be modified to achieve an overall flat response.

 

The band pass filters will always have out of band energy applied to them - and this will always be in the hearing frequencies (20Hz to 20kHz).

 

Therefore filters with ringing are a bad idea - you will need filters with no ringing.

 

Someone else may be able to confirm this.

 

Regards,

Shadders.

 

I got to disagree with you as would Rob Watts and others. Preringing is necessary to reconstruct transits properly. 

[Jud]

30 minutes ago, crenca said:

 

That is how I understand it as well.  The "time to pass through the filter can only be as short as the minimum time for the frequency that takes the longest" as I understand it is technically called "latency", and the only folks who seem to worry about it are HT types who want perfect alignment of the sound with the image on the screen.  Perhaps I am missing something, but in an otherwise normal musical playback situation I don't believe latency matters (am I wrong?) 

 

"On paper" then, what is the argument for "minimum" phase filter?  Why would you want a filter that had the effect of causing the phase of the frequencies (varies by frequency - I believe upper frequencies are "delayed" compared to lower...or do I have this backwards?) to be misaligned?  The two arguments as I understand it are that you are (A)  using the phase misalignment of the filter to "correct" or counteract phase misalignment in some other part of the playback chain (e.g. caused by room or speakers).  Then there is (B), which is "ringing" and its alleged audible effects.  

 

For myself with my normal HP listening (A) should normally be a non-factor.  (B) is "played up" in Audiophiledom.  Thing is, as near as I can tell, with what we know about "good" filter design (anyone can point to a situation where ringing becomes audible in a extreme EQ adjustment, or a poor filter design, or an impulse that by definition is a "step" function outside of the audioband and thus "breaks" the filters normal behavior), it rests on dubious assertions.  

 

As someone upstream said so much attention is given to this phenomenon in Audiophiledom, this "transient" behavior.  Yet, as we all know Audiophiledom has a habit (indeed is built upon) the ghostly edge phenomenons where the small becomes the large technically (allegedly) and important to $sells$.  In other words, "ringing" while real in a sense, appears to be leveraged in the audiophile confidence game for its own ends.  What am I missing?

 

I don't know if you're missing anything, though I do have some questions.  I was going to post Part II of "Elementary Fundamental Questions About Digital Audio I Don't Know the Answers To" about this anyway, so look for it in the next day or two.

 

[Jud]

People are mixing up filter length, IIR/FIR, and phase, which just confuses things.  Phase is a filter characteristic on its own.  It isn't necessarily related to filter length.  An IIR filter can't be linear phase because it would then be infinitely long, but an FIR filter can be minimum phase, linear phase, or something in between.

[Jud]

6 minutes ago, Rt66indierock said:

 

I got to disagree with you as would Rob Watts and others. Preringing is necessary to reconstruct transits properly. 

 

As you noted, there is disagreement.  Some folks (including people not from the Meridian/MQA school) who actually do have some education and training in filter design use minimum phase filters because they think pre-ringing makes transients sound incorrect (since pre-ringing doesn't occur in nature).  Other folks agree with you and Rob Watts.  I hope to figure out a little more regarding what folks are disagreeing about by asking questions. 

[Shadders]

Hi,

My point was - that splitting the audio band up into frequency bands, 20Hz to 100Hz, 100Hz to 200Hz, 200Hz to 400Hz, and so on, means that if each band pass filter has ringing, and it is the energy at the ringing frequency that causes the actual ringing, then each band pass filter will ring, and this ringing will be audible.

 

Whether the ringing in the upper edge of the band pass is cancelled out by the next band pass filters lower edge ringing, i do not know. (Assuming that they have the same cut off frequency......)

 

Regards,

Shadders.

[Jud]

31 minutes ago, Shadders said:

you will need filters with no ringing

 

Not sure that is possible if implementing filtering in the audible range.  You would presumably want the filter to cut frequencies below and/or above the range, and would presumably be sending it frequencies at (as well as above and below) those cutoff points, else there'd be no reason for a filter.  If these things are true, then the filter is going to ring.  You can minimize the ringing by minimizing steepness, but that lets through more of the frequencies you don't want.  You can push more or all of the ringing energy to after the signal that excites the ringing by using intermediate or minimum phase filters, respectively.

 

Whether upsampling (or just sampling at sufficiently high rates) works to minimize or eliminate ringing in this circumstance is beyond my knowledge.

[Shadders]

1 minute ago, Jud said:

 

Not sure that is possible if implementing filtering in the audible range.  You would presumably want the filter to cut frequencies below and/or above the range, and would presumably be sending it frequencies at (as well as above and below) those cutoff points, else there'd be no reason for a filter.  If these things are true, then the filter is going to ring.  You can minimize the ringing by minimizing steepness, but that lets through more of the frequencies you don't want.  You can push more or all of the ringing energy to after the signal that excites the ringing by using intermediate or minimum phase filters, respectively.

 

Whether upsampling (or just sampling at sufficiently high rates) works to minimize or eliminate ringing in this circumstance is beyond my knowledge.

Hi,

Depends on the order of the filter - 8th order will have minimal ringing if linear phase, and even less effect if minimum phase ?

 

For room correction to work, the bands of the audio signal will have to be split up - so you will have band pass filters in the audio band 20Hz to 20kHz.

 

Regards,

Shadders.

[Jud]

2 minutes ago, Shadders said:

Hi,

Depends on the order of the filter - 8th order will have minimal ringing if linear phase, and even less effect if minimum phase ?

 

For room correction to work, the bands of the audio signal will have to be split up - so you will have band pass filters in the audio band 20Hz to 20kHz.

 

Regards,

Shadders.

 

Remember that phase doesn't change the *amount* of ringing energy, just its location in time.  Minimum phase filters eliminate *pre-ringing* at the cost of adding it to the post-ringing.

[Shadders]

1 minute ago, Jud said:

 

Remember that phase doesn't change the *amount* of ringing energy, just its location in time.  Minimum phase filters eliminate *pre-ringing* at the cost of adding it to the post-ringing.

Hi,

Sorry, my mistake, not constipating enough.

Regards,

Shadders.

[mansr]

19 minutes ago, Shadders said:

From the miniDSP sweb site :

REW uses IIR filters, while Dirac Live uses mixed-phase filtering - in effect, a combination of IIR and FIR filters. FIR filters are more powerful than IIR filters, but more expensive to implement.

That doesn't make much sense. The output of an IIR filter is a linear combination of input samples and previous output samples. FIR filters use only input samples to create the output, making them a subset of IIR filters. If anything IIR filters are more "powerful" since they can achieve things impossible with FIR. For example, an impulse response consisting of a step function is trivial as an IIR (current input + previous output). IIR filters can also be unstable and oscillate or increase without bound. Some FIR filters have an equivalent IIR filter with fewer taps, which is cheaper to realise. This probably what they are referring to in the last bit.

[Jud]

 

13 minutes ago, Shadders said:

Depends on the order of the filter - 8th order will have minimal ringing...?

Same as I've been saying - greater cut, more ringing.  Less, cut, less ringing, but more frequencies you don't want get through.  Here's Wikipedia:

 

Quote

In electronic filters, the trade-off between frequency domain response and time domain ringing artifacts is well-illustrated by the Butterworth filter: the frequency response of a Butterworth filter slopes down linearly on the log scale, with a first-order filter having slope of −6 dB per octave, a second-order filter –12 dB per octave, and an nth order filter having slope of -6n dB per octave – in the limit, this approaches a brick-wall filter. Thus, among these the first-order filter rolls off slowest, and hence exhibits the fewest time domain artifacts, but leaks the most in the stopband, while as order increases, the leakage decreases, but artifacts increase.

 

 

[Jud]

Just now, mansr said:

That doesn't make much sense. The output of an IIR filter is a linear combination of input samples and previous output samples. FIR filters use only input samples to create the output, making them a subset of IIR filters. If anything IIR filters are more "powerful" since they can achieve things impossible with FIR. For example, an impulse response consisting of a step function is trivial as an IIR (current input + previous output). IIR filters can also be unstable and oscillate or increase without bound. Some FIR filters have an equivalent IIR filter with fewer taps, which is cheaper to realise. This probably what they are referring to in the last bit.

 

Even with what little I (think I) know, the degree to which marketing gets in the way of clear explanation is frustrating.

[Jud]

9 minutes ago, Jud said:

Same as I've been saying - greater cut, more ringing.  Less, cut, less ringing, but more frequencies you don't want get through.

 

By the way - the potential importance of this, in the context of MQA, is that a filter with very low ringing is likely going to be very leaky, and so you have to wonder whether what people who enjoy MQA like about it is less (pre-)ringing, or a little THD with their cornflakes (i.e., the ultrasonics leaking through the filter intermodulating and creating low level potentially audible distortion).

[Rt66indierock]

1 hour ago, Jud said:

 

As you noted, there is disagreement.  Some folks (including people not from the Meridian/MQA school) who actually do have some education and training in filter design use minimum phase filters because they think pre-ringing makes transients sound incorrect (since pre-ringing doesn't occur in nature).  Other folks agree with you and Rob Watts.  I hope to figure out a little more regarding what folks are disagreeing about by asking questions. 

 

Actually Jud I believe and have stated most of the time linear filers work best, sometimes a minimum phase works and sometimes an intermediate filter like Archimago's is best. I'm in the category of agreeing with Julius O Smith of Stanford. Charles Hansen and I were friends so I heard a lot about why minimum phase filters worked but he didn't live long enough to experiment with intermediate phase filters. And Rob Watts has made compelling arguments to me in our discussions as have several others. I keep saying one size does not fit all.

 

But I have a lot doubt about anything but linear filters when you are assembling songs from tracks.  And I would rather listen to an old iPod than my iPhone X with a minimum phase filter. 

 

The doesn't occur in nature isn't a convincing argument to me. 

[Hifi Bob]

47 minutes ago, Shadders said:

For room correction to work, the bands of the audio signal will have to be split up - so you will have band pass filters in the audio band 20Hz to 20kHz.

 

Regards,

Shadders.

Not so—have a read of http://www.dspguide.com/ch17.htm