John Atkinson: Yes, MQA IS Elegant...

[mansr]

10 minutes ago, Shadders said:

Thanks. So does everyone use the kaiser window ?

I can't speak for everybody, but it generally gives good results. See also the kaiserord function.

[mansr]

1 hour ago, Shadders said:

OK, just did a quick check, the values of the filter coefficients have a dynamic range from 2x10^-17 to 0.25. So not appropriate for fixed point arithmetic in IC's.

 

How would you implement this filter with similar performance using fixed point arithmetic ?

Use enough bits.

[vl]

On 8/26/2018 at 2:56 PM, John_Atkinson said:

 

Because there is no spectral content above the new Nyquist frequency of 22.05kHz, due to the SRC's high-order low-pass filter. (Actually, spectral content above 22.05kHz analyzed in the digital domain after resampling to 96kHz, lies at approximately -127dB ref. the peak pulse level.) It is thus a "legal" 44.1kHz signal.

 

John Atkinson

Editor, Stereophile

 

The definition of a "legal" signal deserves some discussion.

 

From Wikipedia we have the definition of the Nyquist-Shannon sampling theory:

 

If a function x(t) contains no frequencies higher than B hertz, it is completely determined by giving its ordinates at a series of points spaced 1/(2B) seconds apart.

 

There is also a modern clarification in a following paragraph:

 

Modern statements of the theorem are sometimes careful to explicitly state that x(t) must contain no sinusoidal component at exactly frequency B, or that B must be strictly less than ½ the sample rate. 

 

Both statements are technically correct.  The sampling theory implies that when a signal with limited bandwidth B is sampled at 1/(2B) second intervals,  all the information in the original signal is contained in the sampled signal.  This sampling theorem does not however tell us how to retrieve the original information, without unwanted byproducts.

 

In the case of the sampling rate being exactly 2x of the bandwidth, an unwanted byproduct is the ringing excitation of the anti aliasing or reconstruction filter.  If the sampling rate is more than 2x the bandwidth of the signal, ringing of the anti aliasing or reconstruction will not occur.  

 

Anytime we limit the bandwidth of the of a signal by LP filtering it, we harm its transient property.  If we use a gentle filter, there will be less ringing.  If we use a brick wall filter, the ringing will be more concentrated at the "wall".  We can choose the characteristics of our filters.  If we do not go for "audiophile" specifications, we can use a gentle LP anti aliasing (AA) filter that may be a couple dB down at 20 KHz.  This will minimize the concentrated ringing at the brick wall frequency.  This is a choice of the designer of the ADC or the recording engineer.

 

In the case of the DAC, the first D to A operation usually occurs in software in the upsampler.  The reconstruction filter is first applied there.  If the signal is "legal" in the sense that the AA filter creates a condition of oversampling in sampling rate, The DAC will not reconstruct any ringing near the audio band in the upsampler.  If the "legal" signal excites the brick wall filter in the ADC, the DAC will reconstruct and reproduce it.  Downstream the DAC will produce more ringing far above the audio band and it will not be audible.  

 

My message is that if we choose the AA LP filter wisely, we can probably band limit a musical signal to fit into a CD with very good (but not perfect) results.  A brick wall AA LP filter should probably be used with care, with good understanding of the spectrum of the music being recording.  Jokingly speaking if we have musical instrument or human voices that are like impulses, we probably do not want to use a brick wall AA filter.  

 

If we do not want to compromise the transient of the music, than we should not band limit it at the ADC.  Just use a high enough sampling rate.  

 

Finally a word on MQA.  As a compression scheme, and a lossy one at that, it is quite elegant.  I can also say the same for our good friend MP3.  While MQA tries to provide large bandwidth in a small bucket and it appears to be quite good in doing so, it may have some unwanted byproducts that deserves more attention.  Since it uses minimum phase filters, how does the phase distortion affection the sound, compared to linear phase filters?  Since the MQA filters are gentle, leaky filters, there will be alising, the effect of that may be quite audible.  

 

One effective challenge for MQA is to compare an MQA encoded track with a 24/192 PCM track.  MQA wins if both tracks sound the same.  If the MQA track sounds different, even if it is better, that may be a sign of byproducts.

 

vl

[Rt66indierock]

12 minutes ago, vl said:

 

The definition of a "legal" signal deserves some discussion.

 

From Wikipedia we have the definition of the Nyquist-Shannon sampling theory:

 

If a function x(t) contains no frequencies higher than B hertz, it is completely determined by giving its ordinates at a series of points spaced 1/(2B) seconds apart.

 

There is also a modern clarification in a following paragraph:

 

Modern statements of the theorem are sometimes careful to explicitly state that x(t) must contain no sinusoidal component at exactly frequency B, or that B must be strictly less than ½ the sample rate. 

 

Both statements are technically correct.  The sampling theory implies that when a signal with limited bandwidth B is sampled at 1/(2B) second intervals,  all the information in the original signal is contained in the sampled signal.  This sampling theorem does not however tell us how to retrieve the original information, without unwanted byproducts.

 

In the case of the sampling rate being exactly 2x of the bandwidth, an unwanted byproduct is the ringing excitation of the anti aliasing or reconstruction filter.  If the sampling rate is more than 2x the bandwidth of the signal, ringing of the anti aliasing or reconstruction will not occur.  

 

Anytime we limit the bandwidth of the of a signal by LP filtering it, we harm its transient property.  If we use a gentle filter, there will be less ringing.  If we use a brick wall filter, the ringing will be more concentrated at the "wall".  We can choose the characteristics of our filters.  If we do not go for "audiophile" specifications, we can use a gentle LP anti aliasing (AA) filter that may be a couple dB down at 20 KHz.  This will minimize the concentrated ringing at the brick wall frequency.  This is a choice of the designer of the ADC or the recording engineer.

 

In the case of the DAC, the first D to A operation usually occurs in software in the upsampler.  The reconstruction filter is first applied there.  If the signal is "legal" in the sense that the AA filter creates a condition of oversampling in sampling rate, The DAC will not reconstruct any ringing near the audio band in the upsampler.  If the "legal" signal excites the brick wall filter in the ADC, the DAC will reconstruct and reproduce it.  Downstream the DAC will produce more ringing far above the audio band and it will not be audible.  

 

My message is that if we choose the AA LP filter wisely, we can probably band limit a musical signal to fit into a CD with very good (but not perfect) results.  A brick wall AA LP filter should probably be used with care, with good understanding of the spectrum of the music being recording.  Jokingly speaking if we have musical instrument or human voices that are like impulses, we probably do not want to use a brick wall AA filter.  

 

If we do not want to compromise the transient of the music, than we should not band limit it at the ADC.  Just use a high enough sampling rate.  

 

Finally a word on MQA.  As a compression scheme, and a lossy one at that, it is quite elegant.  I can also say the same for our good friend MP3.  While MQA tries to provide large bandwidth in a small bucket and it appears to be quite good in doing so, it may have some unwanted byproducts that deserves more attention.  Since it uses minimum phase filters, how does the phase distortion affection the sound, compared to linear phase filters?  Since the MQA filters are gentle, leaky filters, there will be alising, the effect of that may be quite audible.  

 

One effective challenge for MQA is to compare an MQA encoded track with a 24/192 PCM track.  MQA wins if both tracks sound the same.  If the MQA track sounds different, even if it is better, that may be a sign of byproducts.

 

vl

 

Very little doubt MQA changes things. I asked Bob at the LAAS in 2017 about plugins for DAWs so we could tell what MQA was changing in studios. The conversation continued at the AES convention later that year. 

[vl]

8 minutes ago, Rt66indierock said:

 

Very little doubt MQA changes things. I asked Bob at the LAAS in 2017 about plugins for DAWs so we could tell what MQA was changing in studios. The conversation continued at the AES convention later that year. 

I agree.  That is to be expected.  With the use of "leaky" filters that allow aliasing and minimum phase filters that have significant phase distortion, the sound has to be different.  

vl

[Rt66indierock]

Just now, vl said:

I agree.  That is to be expected.  With the use of "leaky" filters that allow aliasing and minimum phase filters that have significant phase distortion, the sound has to be different.  

vl

 

There seem to be other changes created in the encoding process as well.

[Archimago]

17 hours ago, Rt66indierock said:

 

There seem to be other changes created in the encoding process as well.

 

Would be interesting to get a sense of what these are as well @Rt66indierock.

 

This ongoing talk about filters, showing yet more impulse response results, etc... IMO has clearly reached the end of the road and isn't going to convince anyone that MQA has any secret sauce in the recipe. If intellectually that's all it is, it's clearly bankrupt and impossible to significantly improve fidelity - and likely worsens it.

 

I'm sure audiophiles are "all ears" to know if there's some super duper DSP system in the MQA process that actually does something to make music sound better or in the mastering process...

 

[vl]

23 minutes ago, Archimago said:

 

I'm sure audiophiles are "all ears" to know if there's some super duper DSP system in the MQA process that actually does something to make music sound better or in the mastering process...

 

 

It appears that MQA's leaky filters and their associated impulse responses are the only remaining technical smoke screens MQA and its proponents are repeating over and over again in their apparent publicity effort to influence those who are less technically informed.  

 

To eliminate the brick wall filter ringing one can choose to use a non brick wall anti aliasing that has enough attenuation at the Nyquist frequency.  MQA is not required.  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.  CDs with its modest sampling rate and bit depth can sound very good, when the format is used properly.  MQA is not needed.

 

 

[Jud]

19 hours ago, vl said:

I agree.  That is to be expected.  With the use of "leaky" filters that allow aliasing and minimum phase filters that have significant phase distortion, the sound has to be different.  

vl

 

Keeping things accurate: "Leaky" is bad.  Minimum phase filtering is widely used in audio, including by nearly all room equalization software.

[Shadders]

19 hours ago, mansr said:

Use enough bits.

Hi,

Not to labour the point, here is the worst case filter from the PCM1792 - Texas Instruments DAC :

Stop band attenuation (-100dB to -130dB) is not that good compared to the kaiser window method (-300dB) of filter design.

Regards,

Shadders.

[Rt66indierock]

1 hour ago, Archimago said:

 

Would be interesting to get a sense of what these are as well @Rt66indierock.

 

This ongoing talk about filters, showing yet more impulse response results, etc... IMO has clearly reached the end of the road and isn't going to convince anyone that MQA has any secret sauce in the recipe. If intellectually that's all it is, it's clearly bankrupt and impossible to significantly improve fidelity - and likely worsens it.

 

I'm sure audiophiles are "all ears" to know if there's some super duper DSP system in the MQA process that actually does something to make music sound better or in the mastering process...

 

 

I agree but all I have is a sense of a few changes from approved masters that can't be accounted for by changes to filters when run through the MQA process. Most of these changes are the same or similar what is done when albums are remastered.

 

As for some super duper DSP. I've encountered too many things changed I didn't like or in the case of one of the best MQA files Roberta Flack's "Killing Me Softly" there are differences but they can all be explained with conventional mastering. I won't believe there is any secret sauce in MQA until I look inside an Encoder.

[crenca]

1 hour ago, Jud said:

 

Keeping things accurate: "Leaky" is bad.  Minimum phase filtering is widely used in audio, including by nearly all room equalization software.

 

Just a subjective note:  I have gravitated back to linear (zero) phase, away from "minimum" phase distorting filtering lately.  So now for me minimum phase is "bad".  I don't want MQA or any other scheme imposing it on me.

 

Perhaps subjective audiophiledom will be the very thing that kills off MQA in the end!  ?

[NOMBEDES]

2 hours ago, crenca said:

 

(Edit)

 

Perhaps subjective audiophiledom will be the very thing that kills off MQA in the end!  ?

 

Well, I hope SOMETHING kills it off!

[vl]

4 hours ago, Jud said:

 

Keeping things accurate: "Leaky" is bad.  Minimum phase filtering is widely used in audio, including by nearly all room equalization software.

 

I agree that many room EQ SW use minimum phase filters.  The better ones, like Dirac Live, improves on the impulse response of the system.  This means the phase distortion of its filters is low.

[Jud]

1 minute ago, vl said:

 

I agree that many room EQ SW use minimum phase filters.  The better ones, like Dirac Live, improves on the impulse response of the system.  This means the phase distortion of its filters is low.

 

No. It very much depends on what is meant by "improves impulse response."  It could very likely mean any ringing energy is moved to after the impulse, which is simply another way of saying the filter is minimum phase. Any minimum phase filter will do this, not just those used by Dirac. Go to http://src.infinitewave.ca/ and have a look at the impulse response of any minimum phase filter there. 

[Jud]

3 hours ago, crenca said:

 

Just a subjective note:  I have gravitated back to linear (zero) phase, away from "minimum" phase distorting filtering lately.  So now for me minimum phase is "bad".  I don't want MQA or any other scheme imposing it on me.

 

Perhaps subjective audiophiledom will be the very thing that kills off MQA in the end!  ?

 

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. 

[vl]

25 minutes ago, Jud said:

 

No. It very much depends on what is meant by "improves impulse response."  It could very likely mean any ringing energy is moved to after the impulse, which is simply another way of saying the filter is minimum phase. Any minimum phase filter will do this, not just those used by Dirac. Go to http://src.infinitewave.ca/ and have a look at the impulse response of any minimum phase filter there. 

 

I should clarify.  I was referring to the band limited impulse response of my Dirac Live system, which is limited by my preference to 17 KHz on the high end and 30 Hz on the low end.  The attached screen shot shows the impulse response of the audio system including the speakers.  The left impulse is before Dirac Live correction.  Please note the slight dispersion (spread out) of the impulse in the first mS after the impulse.  The right impulse is with Dirac Live correction.  This dispersion is gone and the pulse is taller.  This indicates better phase behavior after correction.  

 

The Dirac Live unit, a miniDSP model, operates at a sampling rate of 96 KHz.  So its filter ringing is not visible in this band limited presentation.  It uses a blend of FIR and IIR filters.

 

[vl]

16 minutes ago, Jud said:

 

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. 

 

My main speakers are KEF LS50s.  I do not know what crossovers they have.  Looking at the impulse response of the system before Dirac Live correction, you can see that they are quite time coherent.  I crossover the LS50s to two woofers at 200 Hz using linear phase FIR filters running at 96 KHz.  You can see from the right impulse that the corrected impulse response is well behaved.  

 

Most two way monitors do not achieve this level of time coherence.  I find with these monitors, it is hard to tell a minimum phase or linear phase reconstruction filter apart.  With my system I can easily tell.  The linear phase filter has better transient response, is sharper and has more details.  The minimum phase filter smears.

[Jud]

2 minutes ago, vl said:

The linear phase filter has better transient response, is sharper and has more details.  The minimum phase filter smears.

 

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.

[vl]

2 minutes 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.

 

The minimum phase filter has phase distortion, which may affect the perceived transient response.  The ringing of the filter at its Nyquist frequency should be inaudible to most people, even at the 44.1 K sampling rate.  If this ringing is from the AA filter at the ADC and if it is perfectly rendered by the reconstruction filter in the upsampler of the DAC, it should be inaudible, pre ringing or not.  If the reconstruction filter does not have enough computation power, this ringing may be reconstructed with some distortion.  Distortion means non linearity.  Non linearity means intermodulation, which may fall into the audio band and become audible. 

[Rt66indierock]

23 minutes 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.

 

Rob Watts of Chord talked about this last year at RMAF. Beyond Off the Self Dac chips about 17 minutes in.

[asdf1000]

32 minutes 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.

 

Per Jussi @Miskajust one page back:

 

"looked from two extremes, both Chord talking about transient accuracy with extremely long filters and MQA talking about transient accuracy with extremely short filters are both right in a way, but only looking at things from one point of view while ignoring others. As usual in life, truth is somewhere between the extremes..."

 

https://www.computeraudiophile.com/forums/topic/49609-john-atkinson-yes-mqa-is-elegant/?do=findComment&comment=866897

 

[Jud]

1 minute ago, vl said:

The minimum phase filter has phase distortion, which may affect the perceived transient response. 

 

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?

 

 

[vl]

13 minutes 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?

 

 

 

We were talking about two different things.  Your impulse response is that of the anti aliasing or reconstruction filter.  The ringing frequency is at the Nyquist frequency, which is 22.05 KHz for CDs.  This single frequency ringing is above the audio band.  I have yet to understand how such inaudible ringing can lead to transient degradation of audio signals that are audible.

 

My impulse response is that of the entire audio system, included the speakers, limited to a bandwidth of 30 Hz to 17 KHz by my design.  It is a band limited impulse.  Since it is in the audio band, it represents what we hear.  In my system the Dirac Live DSP is done at 96 KHz, so its filter ringing, at 48 KHz, should be inaudible.

[vl]

23 minutes ago, Em2016 said:

 

Per Jussi @Miskajust one page back:

 

"looked from two extremes, both Chord talking about transient accuracy with extremely long filters and MQA talking about transient accuracy with extremely short filters are both right in a way, but only looking at things from one point of view while ignoring others. As usual in life, truth is somewhere between the extremes..."

 

https://www.computeraudiophile.com/forums/topic/49609-john-atkinson-yes-mqa-is-elegant/?do=findComment&comment=866897

 

 

There is a distinction.  Long filters (meaning very steep) are very good for the reconstruction of information contained in the digitized signal.  Short filters are very good for letting transients pass through them, but they are very poor for anti aliasing or reconstruction, as these leaky filters allow aliased signals to reach the audio components downstream, and the ears of the listeners.