Question about filters - MQA vs Hi-Res?

[awdeeoh]

Is it really so even after truncation of inaudible parts?

mqa @ 16/24 44.1/48 is larger than a plain non mqa equivalent (~1-3mb difference). so they don't save space at that rate.

regarding about the filter differences, My earsss can hear the differences and I prefer the mqa filters used.

the difference is not substantial

the list of available music in mqa is not enough for me to buy mqa-dac.

software decode is enough.

 

[BDWoody]

regarding about the filter differences, My earsss can hear the differences and I prefer the mqa filters used.

Any listening controls?

 

[MalinYamato]

mqa @ 16/24 44.1/48 is larger than a plain non mqa equivalent (~1-3mb difference). so they don't save space at that rate.
software decode is enough.

pls proivde a source where this is proven.

About filtering: As far as I understand, MQA is unfiltered as this is one of the rationale behind MQA.

 

[mansr]

If an engineering student was given an assignment to design a resampling filter and they handed in the MQA version, they'd get a failing grade.

 

[bhobba]

What MQA does is not that hard. It is known, but some will argue, that as you increase sampling frequency from 48k to 96k to 192k to 384k etc., the recording sounds better. I have tried it and think it does, but as I say, opinions do vary. A theorem says the maximum frequency that engineers can record is half the sampling frequency. But since the most we can hear is 20khz, 48k should be more than good enough. So why does it sound better? MQA conjecture it is what is called time smear:

MQA: Questions and Answers Tutorial: Temporal Errors In Audio

Tutorial: Temporal Errors In Audio In our Hierarchical paper [2] section 2.3 Temporal Limits, we explain:

www.stereophile.com

The question then is how do we convey the audible 20kz region but have the lower time smear of high sample rates? Here is how MQA does it. Let us imagine a high sample rate say 4XDSD. Recorders are readily available to do that. Now let us use a triangle function to downscale that to 96k as per the following:

MQA Time-domain Accuracy & Digital Audio Quality

Even at high sample rates, standard PCM audio ‘smears’ important timing information. A new digital format, MQA, promises vastly improved time-domain accuracy — without the huge file sizes.

www.soundonsound.com

This has the effect of introducing a very slow roll-off above 20kz - which being above 20kz is inaudible and chucking samples above 96k away. The result is as in the following:

MQA 192k / 96k There and back again.

The issue is these aliasing components that are in the audio band. But here we are in luck. Analysis of recordings shows with the slow roll-off, they are below the notice floor. Secondly, the research also indicates that recordings have nothing but noise above 50kz, so 96 kHz is not a concern. We then transmit the 96 kHz data (MQA uses a tricky way of doing this you can read about by chopping off the bottom bits, recording at 48k, but replacing those chopped off bits with the 24-48khz information - but for simplicity, we assume it is transmitted at 96k). When received, we want to get back the data chucked away. Simple - we linear interpolate - it's all noise anyway, so it will not make any difference if it is about right.

Not that hard - and we now have a very high sampling rate recording with much less time smear to feed into our DAC.

In practice, MQA uses something more sophisticated than triangles and linear interpolation called Bessel functions. In effect, the resampling and reconstruction functions are filters, and they use downsampling-up sampling filters matched to each other. Now, does doing that improve the sound over simple triangles and linear interpolation? Beats me. MQA think it does.

Thanks
Bill

 

[mansr]

the resampling and reconstruction functions are filters, and they use downsampling-up sampling filters matched to each other

Why is that supposed to be an advantage? Both are low-pass filters, and as long as they are close enough to an ideal one, the end result will also be good enough, even if they differ. The MQA filters are, of course, anything but good, what with them hardly filtering anything at all.

The fact that MQA content can sound good at all only proves one thing: that high-res audio (and by extension MQA) is pointless.

 

[bhobba]

The fact that MQA content can sound good at all only proves one thing: that high-res audio (and by extension MQA) is pointless.

I think each person can judge that for themselves via a simple blind test.

Thanks
Bill

 

[MalinYamato]

What MQA does is not that hard. It is known, but some will argue, that as you increase sampling frequency from 48k to 96k to 192k to 384k etc., the recording sounds better. I have tried it and think it does, but as I say, opinions do vary. A theorem says the maximum frequency that engineers can record is half the sampling frequency. But since the most we can hear is 20khz, 48k should be more than good enough. So why does it sound better? MQA conjecture it is what is called time smear:

MQA: Questions and Answers Tutorial: Temporal Errors In Audio

Tutorial: Temporal Errors In Audio In our Hierarchical paper [2] section 2.3 Temporal Limits, we explain:

www.stereophile.com

The question then is how do we convey the audible 20kz region but have the lower time smear of high sample rates? Here is how MQA does it. Let us imagine a high sample rate say 4XDSD. Recorders are readily available to do that. Now let us use a triangle function to downscale that to 96k as per the following:

MQA Time-domain Accuracy & Digital Audio Quality

Even at high sample rates, standard PCM audio ‘smears’ important timing information. A new digital format, MQA, promises vastly improved time-domain accuracy — without the huge file sizes.

www.soundonsound.com

This has the effect of introducing a very slow roll-off above 20kz - which being above 20kz is inaudible and chucking samples above 96k away. The result is as in the following:

MQA 192k / 96k There and back again.

The issue is these aliasing components that are in the audio band. But here we are in luck. Analysis of recordings shows with the slow roll-off, they are below the notice floor. Secondly, the research also indicates that recordings have nothing but noise above 50kz, so 96 kHz is not a concern. We then transmit the 96 kHz data (MQA uses a tricky way of doing this you can read about by chopping off the bottom bits, recording at 48k, but replacing those chopped off bits with the 24-48khz information - but for simplicity, we assume it is transmitted at 96k). When received, we want to get back the data chucked away. Simple - we linear interpolate - it's all noise anyway, so it will not make any difference if it is about right.

Not that hard - and we now have a very high sampling rate recording with much less time smear to feed into our DAC.

In practice, MQA uses something more sophisticated than triangles and linear interpolation called Bessel functions. In effect, the resampling and reconstruction functions are filters, and they use downsampling-up sampling filters matched to each other. Now, does doing that improve the sound over simple triangles and linear interpolation? Beats me. MQA think it does.

Thanks
Bill

thank you for an excellent review of a topic known by very few. I listen to MQA and spend all my time upgrading amps and phones rather than wasting time in arguing between the inaudible diff between high-res uncompressed FLAC and MQA.