HydrogenAudio

http://musicischanging.com/ (http://musicischanging.com/)

I see more descriptions on 3rd party sites.

http://www.whathifi.com/news/meridian-audi...h-res-streaming (http://www.whathifi.com/news/meridian-audio-mqa-paves-way-high-res-streaming)

http://www.stuff.tv/meridian/meridian-s-mq...lity-music/news (http://www.stuff.tv/meridian/meridian-s-mqa-format-allows-streaming-studio-quality-music/news)


From the description, it looks like a hybrid approach - a normal CD-quality stream with Meridian's algorithm/information called MQA. If the receiver cannot decode the algorithm, it will be just played as a CD-quality stream. Otherwise, it will be restored back to high-res music.

I am not even sure if this whole process is lossless or lossy. But I really do not like to have a new proprietary format.

All I'm getting from this is yet another hymn to hi-res music, to be honest. The usual "it sounds so much better than regular CD!".
The only "advantage" is that it supposedly takes as much space as a regular audio CD, so you get "more" for "free". Kind of like HDCD?

Just a guess: redbook lossywav with a lossily compressed difference file (from the hi-res) stored in the freed-up space.

Just a guess: redbook lossywav with a lossily compressed difference file (from the hi-res) stored in the freed-up space.

The correction file will be huge, because lossyWAV will make it plainly apparent that actual dynamic range of most music material is way below 16 bits.

reading http://www.realhd-audio.com/?p=3851 (http://www.realhd-audio.com/?p=3851:), this was quoted from an actual paper by Stuart!:

When recording, the ideal is to capture a performance so that the highest possible sound quality can be recovered from the archive. While an archive has no hard limit on the quantity of data assignable to that information, in distribution the data deliverable depends on application-specific factors such as storage, bandwidth or legacy compatibility. Recent interest in high-resolution digital audio has been accompanied by a trend to higher and higher sampling rates and bit depths, yet the sound quality improvements show diminishing returns and so fail to reconcile human auditory capability with the information capacity of the channel. By bringing together advances in sampling theory with recent findings in human auditory science, our approach aims to deliver extremely high sound quality through a hierarchical distribution chain where sample rate and bit depth can vary at each link but where the overall system is managed from end-to-end, including the converters. Our aim is an improved time/frequency balance in a high-performance chain whose errors, from the perspective of the human listener, are equivalent to no more than those introduced by sound traveling a short distance through air.

According to this Meridian graph (http://musicischanging.com/) I should stick to my Studer tape recorder, if I can live with its low convenience level.

Maybe they are going to build a network open reel machine which can download analog audio from internet and record to tapes.

According to this Meridian graph (http://musicischanging.com/) I should stick to my Studer tape recorder, if I can live with its low convenience level.

Wow - interesting graph. From it, I see that:

• LP is higher quality and more convenient than DVD-A
• DVD-A is extremely inconvenient. (How so? Is it because not everyone has a DVD-A capable player?)

I always had a fair amount of respect for Meridian - they've done some genuinely useful things in the domestic audio playback arena. (In particular, promoting the idea of digital active speakers).

But honestly - what have they been smoking?

And reel-to-reel is the best quality.
I think my parents still have their reel-to-reel machine somewhere. The tapes looked like this:

This may be based on the ideas revealed in a recent AES paper...
http://www.hydrogenaud.io/forums/index.php...mp;#entry882550 (http://www.hydrogenaud.io/forums/index.php?showtopic=107124&st=700&p=882550&#entry882550)


When I saw that quality/convenience graph I nearly wept. I assume it means studio quality reel-to-reel machines running at 15 or 30ips with professional tape stock, not the 7.5ips rubbish sold to consumers in the 1960s, but even so...

Cheers,
David.

And reel-to-reel is the best quality.
I think my parents still have their reel-to-reel machine somewhere. The tapes looked like this:

Analog tape is an audibly flawed medium. Even one generation on very high quality analog tape gear is audible in ABX testing.

What's this?
http://patentscope.wipo.int/search/en/deta...p;maxRec=599628 (http://patentscope.wipo.int/search/en/detail.jsf?docId=WO2013186561&recNum=132&docAn=GB2013051548&queryString=nano%20OR%20filter%20OR%20ceramic&maxRec=599628)

Not had time to read it (and the ref. to a newer patent) properly yet, but this (http://www.meridianunplugged.com/ubbthreads/ubbthreads.php?ubb=showflat&Number=225238#Post225238) may be relevant.

What's this?
http://patentscope.wipo.int/search/en/deta...p;maxRec=599628 (http://patentscope.wipo.int/search/en/detail.jsf?docId=WO2013186561&recNum=132&docAn=GB2013051548&queryString=nano%20OR%20filter%20OR%20ceramic&maxRec=599628)

The file format looks like a fixed-bitdepth lossyWAV, with correction data stored in the LSBs of the signal. The approach seems to be similar to HDCD. Incompatible hardware will be audibly OK with most data since the correction info is in the LSBs beyond the upper 16 bits. Fidelity will be slightly worse than true 16 bit files because of the 3 bit lossy part in the stream. Compatible hardware will be able to restore the original file. Have I deduced that correctly from the graph in the patent?

Anyway, I wonder what the point is. If you deliver 24 bit files anyway, why do this compression in the first place. The argument that you can truncate the LSBs to get lower bitdepth files in low bandwidth cases is still possible with plain 24bit PCM.

My suspicion is the following: The filesize and bitrate will still be huge enough, so that audiophiles can be content about huge numbers on their displays. Audio quality will be worse than true 16 bit delivery, and maybe just bad enough to be noticeable on selected material, so that there is an incentive for people to buy the expensive meridian decoder hardware. This format only exists to intentionally deliver low fidelity music to consumers without Meridian hardware, to sell them expensive Meridian decoders.

Anyway, I wonder what the point is. If you deliver 24 bit files anyway, why do this compression in the first place. The argument that you can truncate the LSBs to get lower bitdepth files in low bandwidth cases is still possible with plain 24bit PCM.

.... because it is a new bespoke format that *must be better* than those that have gone before, however if simple bit-depth reduction on playback were to be used, uers would try to notice the difference between a 24-bit PCM stream and a (rounded or truncated and dithered) 16-bit PCM stream and would presumably be unable to at normal listening levels?

Does the new format allow any form of DRM to be included?

Not had time to read it (and the ref. to a newer patent) properly yet, but this (http://www.meridianunplugged.com/ubbthreads/ubbthreads.php?ubb=showflat&Number=225238#Post225238) may be relevant.

I'm sure it is.

For those who haven't bough the other Meridian paper from the recent AES conference, the patent related to it might be helpful...
http://worldwide.espacenet.com/publication...677A1&KC=A1 (http://worldwide.espacenet.com/publicationDetails/biblio?DB=EPODOC&II=1&ND=3&adjacent=true&locale=en_EP&FT=D&date=20140717&CC=WO&NR=2014108677A1&KC=A1)

I don't know whether MQA is based on the patent co-authored with Peter Craven alone (http://worldwide.espacenet.com/publicationDetails/originalDocument?CC=WO&NR=2014108677A1&KC=A1&FT=D&ND=3&date=20140717&DB=EPODOC&locale=en_EP), the one co-authored with Malcolm Law and Peter Craven (http://worldwide.espacenet.com/publicationDetails/originalDocument?CC=GB&NR=2503110A&KC=A&FT=D&ND=3&date=20131218&DB=EPODOC&locale=en_EP), or both.

There's also
http://www.theabsolutesound.com/articles/r...o-meridian-mqa/ (http://www.theabsolutesound.com/articles/robert-harley-listens-to-meridian-mqa/)

Cheers,
David.

This may be based on the ideas revealed in a recent AES paper...
http://www.hydrogenaud.io/forums/index.php...mp;#entry882550 (http://www.hydrogenaud.io/forums/index.php?showtopic=107124&st=700&p=882550&#entry882550)


When I saw that quality/convenience graph I nearly wept. I assume it means studio quality reel-to-reel machines running at 15 or 30ips with professional tape stock, not the 7.5ips rubbish sold to consumers in the 1960s, but even so...

Cheers,
David.

Do you mean this paper.
http://www.aes.org/e-lib/browse.cfm?elib=17501 (http://www.aes.org/e-lib/browse.cfm?elib=17501)

I stumped uo the $20 and downloaded it. I have to confess that I found it very confusing.  I have only an enthusiasitc amateur understanding of sampling based on the standard  undergraduate texts, but I have to say I found bits of it very difficult to follow, especailly the stuff about time blur. Also the explanantion of the limitation of time resolution using inter sample diracs seemed very odd. Obviously Staurt and Craven are clever and expert, and the paper was peer reviewed. But somehow talking about trasnients which start and finish between samples (as opposed to locatign the peak of an event) seemed a bit dubious to me. Do these exist in music, or in the world?(assuming 44khz sample rate) Isn't an analysis of sampling a dirac "against the law" as it is of course not band-limited?

"Can a sampled system convey time differences that are
shorter than the periods between successive samples? An
intuitive answer might be ‘no’ [60], but we note that even
when convolved with a sinc function, an arbitrarily small
displacement of an impulse can be detecof waveform comparison, assuming one has sufficient
signal-to-noise ratio.
Instantaneous sampling without any filtering is not
recommended, for the sampling would then be vulnerable
to high-frequency noise (even to the megahertz region).

Further, a Dirac impulse would not be registered at all if
it happened to occur between the sampling instants.
Intuitively one would at least integrate over one sample
period, as illustrated in Figure 11 (upper). Here a
transient falling entirely within the sample period
corresponding to Sample 0 will be integrated and the
value of Sample 0 will represent the area of the transient

If the transient moves to the right, there will be no change
in the sample values until the transient crosses into the
adjacent territory of Sample 1. Positional information
has been lost, indeed quantized, so the above ‘intuitive’
answer was correct for this case.

The information loss can be avoided by using an
integration kernel in the form of a triangle or dual ramp
that spans two sample periods, as shown in Figure 11
(lower). By comparing the values of Sample 0 and
Sample 1, both the area and the position of the transient
can now be unambiguously determined."

Can someone explain to me where they are going with this?

There's also
http://www.theabsolutesound.com/articles/r...o-meridian-mqa/ (http://www.theabsolutesound.com/articles/robert-harley-listens-to-meridian-mqa/)

the result is, according to Meridian, a vast improvement in sound quality over even the highest-resolution formats extant.

[/size]

Can someone explain to me where they are going with this?

It seems to me he's saying that there exist acoustical signals (dirac) that can be missed during sampling. Haven't we heard that before from Dr. Kunchur (https://www.gearslutz.com/board/4252198-post14.html) ?

Unless a different interpretation of minimal temporal separation is taken, it is completely fallacious to assert that a CD can resolve less than 5 microseconds when its individual samples are separated by periods of 23 microseconds. (Note that it is true that small alterations in temporal profiles can be indirectly encoded through variations in adjacent levels and that this is certainly aided by having more bits; however, a true translation in time of a temporal feature can only take place in quantized sample periods.)

[/size]It would be helpful if someone could demonstrate with a test how to generate such a signal and how it completely escapes the AD converter.

Perhaps Amir, who is an industry insider and from his posts has worked in this field, will have an opportunity to read the patents and other announcements, and offer his synopsis of what this Meridian technology does.

Do you mean this paper.
http://www.aes.org/e-lib/browse.cfm?elib=17501 (http://www.aes.org/e-lib/browse.cfm?elib=17501)

Yes.

AFAICT they're proposing what most people would call non-ideal downconversion. Compared to what's normally done, they are radically shortening the anti-alias / anti-image filters, which means these filters now allow some aliasing and introduce some time-invariance to the sampling. The benefit they claim is that it allows them to shorten the impulse response dramatically. Because of the first two effects, you "have" to use a higher sampling rate than CD, otherwise you would get some nasty aliasing in the audible band. They claim to ensure it's below the noise floor.

Cheers,
David.

AFAICT they're proposing what most people would call non-ideal downconversion. Compared to what's normally done, they are radically shortening the anti-alias / anti-image filters, which means these filters now allow some aliasing and introduce some time-invariance to the sampling. The benefit they claim is that it allows them to shorten the impulse response dramatically. Because of the first two effects, you "have" to use a higher sampling rate than CD, otherwise you would get some nasty aliasing in the audible band. They claim to ensure it's below the noise floor.

So it seems it all comes down again to eliminate ringing. I wonder if this ringing only is used because this impulse response picture meanwhile is that strong manifested it must be the holy grail.

I have to confess that I found it very confusing.

From reading your quote of the paper I'm also confused.

If you properly lowpass a signal which contains an "inter sample dirac", wouldn't this dirac ripple out into the neighboring samples in the form of a sinc? No need for any other magic. Correct?

Quote from: adamdea on 2014-12-05 14:53:31

I have to confess that I found it very confusing.

From reading your quote of the paper I'm also confused.

If you properly lowpass a signal which contains an "inter sample dirac", wouldn't this dirac ripple out into the neighboring samples in the form of a sinc? No need for any other magic. Correct?

That is my understanding- at least that it would spread out -after all a band limited signal cannot be time limited- but the shape of the remaining signal would I suppose depend on the filter. Then again I am not an expert.

Quote from: adamdea on 2014-12-05 14:53:31

Do you mean this paper.
http://www.aes.org/e-lib/browse.cfm?elib=17501 (http://www.aes.org/e-lib/browse.cfm?elib=17501)

Yes.

AFAICT they're proposing what most people would call non-ideal downconversion. Compared to what's normally done, they are radically shortening the anti-alias / anti-image filters, which means these filters now allow some aliasing and introduce some time-invariance to the sampling. The benefit they claim is that it allows them to shorten the impulse response dramatically. Because of the first two effects, you "have" to use a higher sampling rate than CD, otherwise you would get some nasty aliasing in the audible band. They claim to ensure it's below the noise floor.

Cheers,
David.

There seem to be a number of different points floating around one seems to be a sort of perceptual coding which changes the amount of information for the higher frequencies allowing some like 24/96 or possibly 24/192 at a data rate of only about 1 mbps. The other seems to be the filtering. But there also seems to be some idea about the "sampling kernel". I have to confess that I don't really understand this part which seems to include the passage I quoted. 

In my ignorance I srt of thought that if you were sampling by avaeraging across samples you would be reducing the time resolution and I've never quite understood what was supposed to be so good about shortening the impulse response (except for reducing latency). Is the triangular kernel thing orthodox?

As I understand it the system if I can call it that is intended to be applied end to end from A/D to storage to D/A

That makes no sense.

Make the A/D filter steep, non-aliasing. What matters for playback is the D/A or resampling filter used, which can allow imaging, be a lot less steep, even be minimum phase ...

I also wonder does this new format do anything high sample rate Ogg Vorbis wouldn't accomplish?  It will give somewhere around a 4 to 1 reduction in file size while maintaining bandwidth at highest quality settings.  If you used your own filter at something like 30 khz just to be sure you can knock down the file size by another factor of two or three.  Not that I believe we need the extra bandwidth myself.

According to this Meridian graph (http://musicischanging.com/) I should stick to my Studer tape recorder, if I can live with its low convenience level. 

The most startling one to me is LPs being said to be better than CD. Since CD is better in terms of rotational wow, warp wow, flutter, speed accuracy, hiss, dynamic range, hum, crackle, skips, pops/ticks, frequency response deviation, distortion, groove wear, inner groove distortion, channel separation, etc., I wonder what specific audio parameter they deem more important than all of these so as to give LP the edge by >2 "units" of quality?

Maybe if they don't say vinyl is better than CD they'll lose cred with their intended customer base.

When I saw that quality/convenience graph I nearly wept.

I felt a wave of nausea, and then sadness. To see such a bizarre graph being published in 2014 is quite saddening.

The most startling one to me is LPs being said to be better than CD. Since CD is better in terms of rotational wow, warp wow, flutter, speed accuracy, hiss, dynamic range, hum, crackle, skips, pops/ticks, frequency response deviation, distortion, groove wear, inner groove distortion, channel separation, etc., I wonder what specific audio parameter they deem more important than all of these so as to give LP the edge by >2 "units" of quality?

Well, CDs involve sampled audio, don't you know, rather than recording the wave continuously in a "proper analogue way". Not convinced? Ok, here's a specific: vinyl can be operated further into the ultrasonic realm than CDs, theoretically. (That's of highly questionable value of course.) I think it boils down to greynol's point immediately above.

There's a certain percentage of audiophiles who have been "educated" to believe that the standard CD format is inferior, and who are accordingly looking for an improvement over CD sound.

Maybe if they don't say vinyl is better than CD they'll lose cred with their intended customer base.

Ah yes, good point.

...and to your point, it doesn't help to push something as just an incremental improvement over 30 year old technology;

an incremental improvement that requires going to extraordinary and dubious lengths just to show a tepidly noticeable difference.

Depressing.

Maybe if they don't say vinyl is better than CD they'll lose cred with their intended customer base.

Don't forget that Meridian has to explain why it's safe to use digital speakers, even when listening to vinyl or reel to reel.

Yeah, Meridian did early go digital-only - I use their speakers BTW, with a cheap USB-to-SPDIF between them and fb2k.
I do not understand why they want to present this mumbojumbo, it just doesn't fit someone who used to promote a fully digital reproduction line.

That reel-to-reel is "better" is in some sense justifiable from the point of view that most likely that is the original source. It is of course grossly inconvenient to remaster the album every time you need to listen to it, but it is not unlikely better than the DR = 3 CD

I wonder:

- would a "Dirac" - which, I figure, must be an energy discontinuity, i.e. an infinitely loud infinitely short spike or something practically very close - be anything you wanted to feed into your playback equipment?

- assuming it would be reproduced: would you want to feed it into your ears?

I wonder:

- would a "Dirac" - which, I figure, must be an energy discontinuity, i.e. an infinitely loud infinitely short spike or something practically very close - be anything you wanted to feed into your playback equipment?

- assuming it would be reproduced: would you want to feed it into your ears?

Very good point. Even ignoring the idea that it requires all of the energy in the universe (which is how it was explained to me) I'm not sure I understand what it means in sound terms though. After all we are supposed to be talking about recorded sound here not x rays

Surely in order to be a dirac, the air molecules would have to move to one side and then stay there. If they do any compressing and uncompressing then the event would not be a dirac any longer. Is there not an inevitably finite rise time for a sound wave. No idea what is might be though. Any ideas? My intuitive instinct was that it might have something to do with the speed of sound. (don't hit me)

Quote from: 2Bdecided on 2014-12-05 10:16:17

This may be based on the ideas revealed in a recent AES paper...
http://www.hydrogenaud.io/forums/index.php...mp;#entry882550 (http://www.hydrogenaud.io/forums/index.php?showtopic=107124&st=700&p=882550&#entry882550)

When I saw that quality/convenience graph I nearly wept. I assume it means studio quality reel-to-reel machines running at 15 or 30ips with professional tape stock, not the 7.5ips rubbish sold to consumers in the 1960s, but even so...

Cheers,
David.

Do you mean this paper.
http://www.aes.org/e-lib/browse.cfm?elib=17501 (http://www.aes.org/e-lib/browse.cfm?elib=17501)

I stumped uo the $20 and downloaded it. I have to confess that I found it very confusing.  I have only an enthusiasitc amateur understanding of sampling based on the standard  undergraduate texts, but I have to say I found bits of it very difficult to follow, especailly the stuff about time blur. Also the explanantion of the limitation of time resolution using inter sample diracs seemed very odd. Obviously Staurt and Craven are clever and expert, and the paper was peer reviewed.

Conference paper, not a journal paper.

But somehow talking about trasnients which start and finish between samples (as opposed to locatign the peak of an event) seemed a bit dubious to me.

In general there is nothing special about an event starting on sample boundaries. 

Do these exist in music, or in the world?(assuming 44khz sample rate)

In general they don't exist.

(1) Everything in the real world that we interact with has been subject to any number of low pass filters.

(2) In the world of audio the rule of low pass filtering is especially relevant to musical instruments and microphones. But most real world audio gear has builtin low pass filters starting as low as 15 KHz (FM Stereo) 22 KHz (44 KHz sampled audio) 50 -100 KHz - (most analog gear).

Isn't an analysis of sampling a dirac "against the law" as it is of course not band-limited?

One might think...

"Can a sampled system convey time differences that are
shorter than the periods between successive samples? An
intuitive answer might be ‘no’ [60], but we note that even
when convolved with a sinc function, an arbitrarily small
displacement of an impulse can be detecof waveform comparison, assuming one has sufficient
signal-to-noise ratio.
Instantaneous sampling without any filtering is not
recommended, for the sampling would then be vulnerable
to high-frequency noise (even to the megahertz region).

Further, a Dirac impulse would not be registered at all if
it happened to occur between the sampling instants.
Intuitively one would at least integrate over one sample
period, as illustrated in Figure 11 (upper). Here a
transient falling entirely within the sample period
corresponding to Sample 0 will be integrated and the
value of Sample 0 will represent the area of the transient

If the transient moves to the right, there will be no change
in the sample values until the transient crosses into the
adjacent territory of Sample 1. Positional information
has been lost, indeed quantized, so the above ‘intuitive’
answer was correct for this case.

The information loss can be avoided by using an
integration kernel in the form of a triangle or dual ramp
that spans two sample periods, as shown in Figure 11
(lower). By comparing the values of Sample 0 and
Sample 1, both the area and the position of the transient
can now be unambiguously determined."

Can someone explain to me where they are going with this?

Out of the known world. ;-)

Quote from: 2Bdecided on 2014-12-05 10:16:17

This may be based on the ideas revealed in a recent AES paper...
http://www.hydrogenaud.io/forums/index.php...mp;#entry882550 (http://www.hydrogenaud.io/forums/index.php?showtopic=107124&st=700&p=882550&#entry882550)


When I saw that quality/convenience graph I nearly wept. I assume it means studio quality reel-to-reel machines running at 15 or 30ips with professional tape stock, not the 7.5ips rubbish sold to consumers in the 1960s, but even so...

Cheers,
David.

Do you mean this paper.
http://www.aes.org/e-lib/browse.cfm?elib=17501 (http://www.aes.org/e-lib/browse.cfm?elib=17501)

I stumped uo the $20 and downloaded it. I have to confess that I found it very confusing.  I have only an enthusiasitc amateur understanding of sampling based on the standard  undergraduate texts, but I have to say I found bits of it very difficult to follow, especailly the stuff about time blur. Also the explanantion of the limitation of time resolution using inter sample diracs seemed very odd. Obviously Staurt and Craven are clever and expert, and the paper was peer reviewed. But somehow talking about trasnients which start and finish between samples (as opposed to locatign the peak of an event) seemed a bit dubious to me. Do these exist in music, or in the world?(assuming 44khz sample rate) Isn't an analysis of sampling a dirac "against the law" as it is of course not band-limited?

"Can a sampled system convey time differences that are
shorter than the periods between successive samples? An
intuitive answer might be ‘no’ [60], but we note that even
when convolved with a sinc function, an arbitrarily small
displacement of an impulse can be detecof waveform comparison, assuming one has sufficient
signal-to-noise ratio.
Instantaneous sampling without any filtering is not
recommended, for the sampling would then be vulnerable
to high-frequency noise (even to the megahertz region).

Further, a Dirac impulse would not be registered at all if
it happened to occur between the sampling instants.
Intuitively one would at least integrate over one sample
period, as illustrated in Figure 11 (upper). Here a
transient falling entirely within the sample period
corresponding to Sample 0 will be integrated and the
value of Sample 0 will represent the area of the transient

If the transient moves to the right, there will be no change
in the sample values until the transient crosses into the
adjacent territory of Sample 1. Positional information
has been lost, indeed quantized, so the above ‘intuitive’
answer was correct for this case.

The information loss can be avoided by using an
integration kernel in the form of a triangle or dual ramp
that spans two sample periods, as shown in Figure 11
(lower). By comparing the values of Sample 0 and
Sample 1, both the area and the position of the transient
can now be unambiguously determined."

Can someone explain to me where they are going with this?

I'll take a guess. 

This is how they will market this as the first digital music that doesn't "sound digital".

Alongside the nice Convenience/Quality graph will be another stair step digital waveform graph.

"See" how digital sounds?
I mean .... just look at the graph!
MQA has the ability to measure between the digital steps and capture that missing emotion.
Finally you can have all the convenience of digital files with all the quality of reel to reel.

... or something like that  ...

Quote from: Porcus on 2014-12-06 12:04:22

I wonder:

- would a "Dirac" - which, I figure, must be an energy discontinuity, i.e. an infinitely loud infinitely short spike or something practically very close - be anything you wanted to feed into your playback equipment?

- assuming it would be reproduced: would you want to feed it into your ears?

Very good point. Even ignoring the idea that it requires all of the energy in the universe (which is how it was explained to me)

I have not seen the definitions, but I think they mean it only requires infinite power - over zero time. That is, if an amp delivers 1 watt continuously - meaning 1 J per second - this signal added would mean that e.g. precisely at 30.000004 seconds , the total energy dissipated since the start, jumps abruptly from 30.000004 to 30.000104 and then goes on 1 J / second so that at 31 seconds it has delivered 31.0001 joules. This jump would correspond to "infinite" frequency and would therefore not be captured adequately by an ADC nor digital formats. I would be surprised though, if it could be played back on any music equipment (could air actually take such a waveform?).

Now. What if this discontinuity occurs once every 1/100 second? Is it then a 100 Hz tone? Is it sampleable? Is it possible to reproduce with amps and speakers? Will the human ear detect it? Does it even exist?


(My gut feeling says that someone was first and foremost looking for a signal that won't be encoded to a digital format by an ADC, and any question regarding usefulness of the example (like e.g. "does such one even exist on planet Earth?" was of secondary importance.)

http://hiddenwires.co.uk/2014/meridian-aud...technology-mqa/ (http://hiddenwires.co.uk/2014/meridian-audio-launches-new-audio-technology-mqa/)

Meridian Audio Launches New Audio Technology: MQA
09/12/2014
Bob Stuart, founder of Meridian Audio, recently launched MQA (Master Quality Authenticated), a new British technology, which the company says “is poised to change the way people enjoy music all over the world”. The launch, hosted in The Shard, was attended by key music industry executives, artists and commentators.

Developed by Meridian as a way of combating what the company sees as the trend of sacrificing sound quality for convenience, MQA is said to capture and preserve nuances and vital information that current music files obscure or discard, but in a file that is small and convenient to download or stream.

“Music lovers need no longer be shortchanged; finally we can all hear exactly what the musicians recorded,” says Bob Stuart, the pioneer behind MQA technology. “MQA gives a clear, accurate and authentic path from the recording studio all the way to any listening environment—at home, in the car or on the go. And we didn’t sacrifice convenience.”

Stuart also advises that “the announcement of MQA is really about the future of recorded music. Music is important to us all. When the sound is authentic it is more involving, we understand it better and enjoy it longer. MQA is already receiving broad support from the music industry, artists, recording and mastering engineers and record labels.

MQA will be available early 2015.

Quote from: adamdea on 2014-12-05 14:53:31

I have to confess that I found it very confusing.

From reading your quote of the paper I'm also confused.

If you properly lowpass a signal which contains an "inter sample dirac", wouldn't this dirac ripple out into the neighboring samples in the form of a sinc? No need for any other magic. Correct?

Correct.

Manners requires that I do not speak plainly of what I see elsewhere in this thread.

That is my understanding- at least that it would spread out -after all a band limited signal cannot be time limited- but the shape of the remaining signal would I suppose depend on the filter. Then again I am not an expert.

While that's true, one can assure that the spectral content outside of the passband can be reduced to any arbitrary level, in particular one can assure that the contributions from outside the passband are smaller than the quanitzation noise.

Well, CDs involve sampled audio, don't you know, rather than recording the wave continuously in a "proper analogue way". Not convinced? Ok, here's a specific: vinyl can be operated further into the ultrasonic realm than CDs, theoretically. (That's of highly questionable value of course.) I think it boils down to greynol's point immediately above.

CD's also fail to have a variety of euphonic distortion mechanisms. 'nuff said?

And right on schedule, lookit what the hi-rez cheerleaders are claiming (or claiming that Meridian claims).  Another *vast improvement*!:

When an MQA-encoded recording is played back through a consumer-audio product with MQA decoding, the result is, according to Meridian, a vast improvement in sound quality over even the highest-resolution formats extant.

  - Robert Harley, TAS


http://www.theabsolutesound.com/articles/r...o-meridian-mqa/ (http://www.theabsolutesound.com/articles/robert-harley-listens-to-meridian-mqa/)

Of course you know Robert Hartley is just full of it when he writes stuff like this:

The treble was totally unlike any other digital I’d heard, completely free from the metallic hardness and artifacts we assume are part-and-parcel of digital audio.

Hard to take anything in the article seriously, but it does give hints at what MQA actually is, although really much more about Meridian's marketing slant.

You should have put the emph on according to Meridian ... journalism schmournalism.

According to me your happiness in life will increase if you transfer big bucks to my account.

Quote from: Gecko on 2014-12-05 17:36:28

If you properly lowpass a signal which contains an "inter sample dirac", wouldn't this dirac ripple out into the neighboring samples in the form of a sinc? No need for any other magic. Correct?

Correct.

Manners requires that I do not speak plainly of what I see elsewhere in this thread.

Thanks. Could you help with the question of whether there is a minimum period of a sonic event (at a particular place) in air. I simply can't get my mind round how a dirac (or anything like a dirac) could be produced by the propagation of sound in air. Or am I wrong in imagining that air must vibrate for a while when a pressure wave passes through it.

My anxiety on this subject is caused by the idea that we might for some reason need to be able to locate the leading edge of  a sonic event with greater precision than 16/44 or even 24/96 (conventionally filtered) will allow. I understand that 16/44 allows something like 60 ps of resolution in terms of locating an intersample peak of an  event extending across sampling instants but Stuart and Craven seem to know this and yet for some reason still insist that we migth need to be able to locate an event which rises to a peak from nothing imbetween samples. This sounds suspiciously like something you can't hear to me, but I'm just trying to understand what this might mean in relation to a real world transient

I have been trying to find a recording of a short sound event. I think Bob Stuart mentions twigs snapping somewhere.
I managed to find this recording but I think its only at 32000 khz 
http://soundbible.com/48-Branch-Break.html (http://soundbible.com/48-Branch-Break.html)
Still snapping seems to take a long time to form -at least 2 ms for the main hump (if that's the right way to describe it) and ignoring the  longer build up from scratch. I can see though that this seems to be band limited to about 11 and a bit Khz, but even then it doesn't seem to me that this event could have slipped through the net if it weren't band limited. Nevertheless it would be useful to see a higher sample rate recording to understand how the time domain representation of this event would have differed. I'm simply trying to understand why according to Stuart and Craven my perception of the timing of the event would be different

Perhaps the castanets here would be a better example as they seem to have been sampled at 44.1Khz
http://www.scalatech.co.uk/bsac.htm (http://www.scalatech.co.uk/bsac.htm)
Even here though I can't see how the sound could have fitted imbetween two samples if it had not been low pass filtered. 

I find it frustrating that  in Stuart's paper supposedly explaining the time domain requirement for >44khz sampling and the problems of a conventional anti alias filter at 48khz or so he nevr shows a real world time domain illustration of the "time blur" on a real world sonic event explaining the different times at which we will (according to the theory) perceive the sonic event at 44.1 , 96khz with an ordinary filter and 96Khz with the magic filter.

Stuart is proposing a theory. It could be proven or disproven in a rigorous psychoacoustic experiment, but to my knowledge no such experiment has been performed. His recent paper is not proof. It contains too many other variables.

In showing the impulse response of the filter, the paper is pretty much showing you what will happen at the leading edge of any wide bandwidth sharp transient. The waveform changes significantly, but all that horrible looking stuff is at 22kHz.

Cheers,
David.

the paper is pretty much showing you what will happen at the leading edge of any wide bandwidth sharp transient. The waveform changes significantly, but all that horrible looking stuff is at 22kHz.

http://www.hydrogenaud.io/forums/index.php...st&p=880450 (http://www.hydrogenaud.io/forums/index.php?s=&showtopic=107124&view=findpost&p=880450)

Of course you know Robert Hartley is just full of it when he writes stuff like this:

Quote

The treble was totally unlike any other digital I’d heard, completely free from the metallic hardness and artifacts we assume are part-and-parcel of digital audio.

Hard to take anything in the article seriously, but it does give hints at what MQA actually is, although really much more about Meridian's marketing slant.

the technical aspects are described in the other AES convention paper, "A Hierarchical Approach to Archiving and Distribution"

http://www.aes.org/e-lib/browse.cfm?elib=17501 (http://www.aes.org/e-lib/browse.cfm?elib=17501)


NB there are a lot of 'could bes' and 'can bes' and even a 'we imagine' in that paper.

You should have put the emph on according to Meridian ... journalism schmournalism.

According to me your happiness in life will increase if you transfer big bucks to my account.

To be fair, I have not seen any quotes so far directly from Meridian or Stuart that use that phrase 'vast improvement'.  I've only seen it in Harley's article.


But I haven't looked very hard either.

Stuart is proposing a theory. It could be proven or disproven in a rigorous psychoacoustic experiment, but to my knowledge no such experiment has been performed. His recent paper is not proof. It contains too many other variables.

In showing the impulse response of the filter, the paper is pretty much showing you what will happen at the leading edge of any wide bandwidth sharp transient. The waveform changes significantly, but all that horrible looking stuff is at 22kHz.

Cheers,
David.

Thanks. I do appreciate that there is no psychoacoustic evidence to support Stuart's theory, and of course that is arguably that.

However I think that there may be some benefit in trying to understand what his claim might even mean

It is my understanding that only energy in the transition band of a filter is distributed through the impulse response. If there isn't very much energy in that transition band then there won;t be very much to distribute. On that basis I find questionable the assumption that a shallow filter with a wide transition band (and short ringing)is preferable to a filter with a very narrow transition band and longer (but very low amplitude) ringing 

Returning though to my earlier hobby horse. It seems to me very likely that the entire concept of a sharp transient sonic event (occurring between the samples relative to 44.1khz  sampling) is very dubious. 
As I understand it air is not a very friendly medium to ultrasound. I have been struggling to find references but this paper here fig 2.2 suggests that air at 20 C and ordinary pressure and humidity
absorbs 100Khz sound at 2.2dB/m , 500kHz sound at 40dB/m and 1MHz sound at 160dB/m
 
http://www.ktu.lt/ultra/journal/pdf_50_1/5...ladisauskas.pdf (http://www.ktu.lt/ultra/journal/pdf_50_1/50-2004-Vol.1_09-A.Vladisauskas.pdf)
and also here which suggests 7-8dB/m for 200khz
http://www.ndt.net/article/ultragarsas/63-...-jakevicius.pdf (http://www.ndt.net/article/ultragarsas/63-2008-no.1_03-jakevicius.pdf)

If this is correct then surely we can rule out any sonic event you are likely to hear at 5m* away from having any 500Khz content.
btw if you are at higher temperature or humidity it seems the absorption is much greater.
Unfortunately the article does not give figures for frequencies between 200khz to 500 Khz.

Ah no I have found this:
http://www.sengpielaudio.com/calculator-air.htm (http://www.sengpielaudio.com/calculator-air.htm) (does not work with google chrome) which enables one to calculate absorption for any frequency. (about 11.dB/m at 250Khz). 350 Khz seems a reasonable maximum at 5m

I am happy to be shot down in flames if I have misunderstood this, but it seems to me that this is sufficient to demonstrate that you simply can't have a dirac or sonic square wave- or even anything that would look a bit like say a sonic  40Khz square wave or it seems a rise time of much less than a microsecond if the event occurs 5m away. It seems to me that one ought to start, even when theorising about sound recording, with a possible sound event.   

It would be interesting to look at the effect of this air filter throughout the frequency range if we were to consider the inherent time smear in any sonic transient (assuming there were any point in considering inaudible frequencies)


* I'm assuming that we are interested in recording some sort of event one might watch, although I suppose some people might be interested in capturing the sound of a trumpet next to their ear. And it seems to me that the "our bodies have adapted to be able to localise the snapping of a twig" must refer to something a little way away or it would not be very useful.

Returning though to my earlier hobby horse. It seems to me very likely that the entire concept of a sharp transient sonic event (occurring between the samples relative to 44.1khz  sampling) is very dubious.

His theory doesn't require that. His sharp-ish filter takes a few samples to get 40dB down, and hundreds of samples to get 60dB down. If you Nyquist filter with something like a sinc filter, energy leaks out from any sharp rising edge a very long way. But only at 22kHz, and only if the rising edge has energy at 22kHz.

Hang on, you've said the same thing already, so you understand all this. But then you've gone on to do calculations that require an order of magnitude higher frequency because you've been distracted by the idea of needing sub-sample rise times for this to be significant. You don't.

Cheers,
David.

I am happy to be shot down in flames if I have misunderstood this, but it seems to me that this is sufficient to demonstrate that you simply can't have a dirac or sonic square wave- or even anything that would look a bit like say a sonic  40Khz square wave or it seems a rise time of much less than a microsecond if the event occurs 5m away. It seems to me that one ought to start, even when theorising about sound recording, with a possible sound event.

In 2009 I've done some microphone testing with electrical spark discharges. Since the spark isn't a perfect dirac the recorded impulse isn't just the IR of the microphone alone, but it did allow me to compare different microphones.
From AES paper 7065 (http://www.aes.org/e-lib/browse.cfm?elib=14050):

An acoustic test impulse which approximates the Dirac (t) distribution can be generated with a pistol shot or an electrical spark discharge. The former approach is poorly reproducible and provides single impulses only. Spark discharges can be reproduced periodically, and thus better captured. A spark discharge between two electrodes has the appearance of a heavily damped period close to the aperiodic boundary [2]. The short positive overpressure peak is followed by underpressure as the suddenly-expanded air flows back toward the center of ‘explosion.’ Consequently there is almost no acoustic wave propagation, since the real part of the acoustic radiation resistance approaches zero.

[/size]
These are the wav file and the waveform.
Spark Recording 24bit96kHz wav (20kB) (http://www.galaxyclassics.com/public/SparkDischargeImpulseRecording.wav)


The spark was recorded at 24bit 96kHz from about 5 cm distance. I could redo it at 24/192 if anyone thinks that's useful.
The B&K4006 (now DPA4006) microphone is a "typical" high quality small diaphragm condenser microphone.
Note that one sample period in the graph is about 10 µs.

Quote from: adamdea on 2014-12-11 14:48:53

Returning though to my earlier hobby horse. It seems to me very likely that the entire concept of a sharp transient sonic event (occurring between the samples relative to 44.1khz  sampling) is very dubious.

His theory doesn't require that. His sharp-ish filter takes a few samples to get 40dB down, and hundreds of samples to get 60dB down. If you Nyquist filter with something like a sinc filter, energy leaks out from any sharp rising edge a very long way. But only at 22kHz, and only if the rising edge has energy at 22kHz.

Hang on, you've said the same thing already, so you understand all this. But then you've gone on to do calculations that require an order of magnitude higher frequency because you've been distracted by the idea of needing sub-sample rise times for this to be significant. You don't.

Cheers,
David.

Thanks, I did go a bit off on one. I think I was distracted by the concept (in the passage I quoted) of the inter sample dirac. His theory about the importance of ringing  may not require it, but he goes on about it when he taks about time resolution. I didn;t make that bit up.

I do get though that there will be pre-ringing if there is energy in the transition band of the rising edge, so we only need 22Khz energy to excite the ringign in 16/44

Staurt maintains that people can hear the effect of a conventional (normally linear phase, but he implies even minimum phase) filter at 48kHz.  But how much energy?

What confuses me is that he applies a significance test for aliasing by asking how big the product is relative to the noise level, but shouldn't we be asking the same question of the pre-ringing which might matter. I have never seen an analysis of a real world trasnient showing the amount of energy which is distributed at a given time relative to the noise floor- after all the MQA system depends in its coding on the point that there isn't actually any significant signal over 48Khz and the peak to floor level isn't very great in the octave below.

In 2009 I've done some microphone testing with electrical spark discharges. Since the spark isn't a perfect dirac the recorded impulse isn't just the IR of the microphone alone, but it did allow me to compare different microphones.
From AES paper 7065 (http://www.aes.org/e-lib/browse.cfm?elib=14050):

Quote

An acoustic test impulse which approximates the Dirac (t) distribution can be generated with a pistol shot or an electrical spark discharge. The former approach is poorly reproducible and provides single impulses only. Spark discharges can be reproduced periodically, and thus better captured. A spark discharge between two electrodes has the appearance of a heavily damped period close to the aperiodic boundary [2]. The short positive overpressure peak is followed by underpressure as the suddenly-expanded air flows back toward the center of ‘explosion.’ Consequently there is almost no acoustic wave propagation, since the real part of the acoustic radiation resistance approaches zero.

[/size]
These are the wav file and the waveform.


The spark was recorded at 24bit 96kHz from about 5 cm distance. I could redo it at 24/192 if anyone thinks that's useful.
The B&K4006 (now DPA4006) microphone is a "typical" high quality small diaphragm condenser microphone.
Note that one sample period in the graph is about 10 µs.

Wow thanks, that's the sort of thing I was looking for, although I'm not sure what the limiting factor is.

Would it be a fair summary that the impulse response is dominated by that of the microphone.? There seems to be maybe 70us of pre ringing visible. What is the frequency response of the mic?  Is is safe to assume that its stopband is much lower than 48Khz? I was guessing that the pre ringign was at about 30Khz. but would be grateful for fonrimation of how one measures this.

does the A/D use a linear phase filter?

I'm intrigued by what it would look like at 24/192 unless you think it's obvious that it wouldn't make a difference

DPA4006A

Frequency range, ± 2 dB: 10 Hz to 20 kHz

4006 (http://www.dpamicrophones.com/en/products.aspx?c=Item&category=234&item=24385#specifications)

DPA4006A

Frequency range, ± 2 dB: 10 Hz to 20 kHz

4006 (http://www.dpamicrophones.com/en/products.aspx?c=Item&category=234&item=24385#specifications)

Thanks- I did look at those figures, but I was wondering at what point it reaches what one might call a stop band. If for the sake of argument it reached -80 dB at 30 Khz, then the impulse response of a conventional A/D filter for 24/96 would be irrelevant (as would the air filtering effect too). If however the mic has a very slow roll off above 20Khz then the impulse response of the A/d filter at 24/96 might have an impact.

In order to see the effect that the A/D filter has at 24/96 it would be useful to see a recording of a mic which could definitely capture frequencies over 40Khz (ideally 100 khz). ideally with flattish response but i imagine that might be tricky

Assuming that the DPA mic was flattish to 22.05, It would be interesting to compare a 44/1 recording where one really would be looking at the response of the  filter to the spark

Assuming that the DPA mic was flattish to 22.05, It would be interesting to compare a 44/1 recording where one really would be looking at the response of the filter to the spark

Most (studio) recordings are made at 96 kHz (or higher) sampling rate and then downconverted to 44.1, so that's what I did here. The original spark signal was sample rate converted to 44.1kHz and back to 96kHz (with iZotope RX SRC). I've assembled a short stereo file with the 24/96 original in the left channel and the 44.1 version in the right channel, followed by the same but with L/R swapped. It allows to listen for differences (looped playback can be handy).
Hope this helps.

Spark sample 24/96 versus 24/44.1 (http://www.galaxyclassics.com/public/Spark96vs44.1.wav)

Well, MQA has some DXD competition (http://www.stereophile.com/content/promates-worlds-first-dxd-download-store), though I'm not sure where it falls on the Meridian "Quality vs Convenience" graph.

From the Probates website (https://www.promates.com/music-store/hd-audio#prettyPhoto)


and



and of course:

For me, it is clear that there is an undeniable relationship between resolution and quality. Each format seems to have has its own expression and conversely, its own limitations. To reproduce classical music, for example, I have always preferred the quality of DSD.

Yet, it was not until I experienced DXD that I was no longer able to hear the signal as a digital reproduction. In contrast, DXD is calm and warm, with a deep, resonant and well defined stereo perspective. So much so that it evokes memories of the heady days of analog.

Peter Scheelke

cheers,

AJ

I'm sure Nyquist will move up one octave again as soon as the technology allows it.

From the Probates website (https://www.promates.com/music-store/hd-audio#prettyPhoto)

Can I have some of what they are smoking?

Quote from: ajinfla on 2014-12-13 20:59:50

From the Probates website (https://www.promates.com/music-store/hd-audio#prettyPhoto)
. Then those amplitudes could be anything.


People have identified this possible problem in the past. I've only actually heard it with sound blaster cards and test signals. Even so, you can solve it during playback by using a reconstruction filter that cuts at about 20kHz, not 22kHz. You can solve it during recording by using an anti-alias filter that cuts "early" like that too. Looking at the spectrum of CDs, such a thing is quite common. Not all, but many CDs have almost no spectral content approaching Nyquist, which implies there was nothing left above Nyquist to alias.

Cheers,
David.

a most interesting firsthand account of Meridian's award-winning paper presentation at the last AES conference.  Apparently not everyone in the audience was convinced...

http://www.avsforum.com/forum/286-latest-i...ml#post29832714 (http://www.avsforum.com/forum/286-latest-industry-news/1793162-meridian-mqa-promises-revolution-recorded-music-quality-convenience.html#post29832714)

I have read the paper and was also present when it was presented. In the question and answer period following the presentation, Analog Devices' Bob Adams (designer of the first successful IC asynchronous sample rate converter for AD) pointed out that the shapes of the filters used in the study were pathologically selective compared to normal commercial practice, having a much longer impulse response (by about a factor of 4x) than filters commonly used. So the take-away from this paper, at least for me, is that one can design an audible linear-phase digital filter whose transition region is above 20 kHz if you make the impulse response long enough.

As for the use of RFDF [sic] dither in the 16-bit tests, this just seems bizarre to me because it is so contrary to good engineering practice.

From the paper:

"The frequencies of the transition bands were 23500-24000 Hz and 21591-22050 Hz, corresponding to the
standard sample rates of 48 kHz and 44.1 kHz respectively."

In normal commercial practice, the transition region is allowed to start at 20 kHz.

(Actually maybe this belongs in a different thread, but that thread is closed.  New thread, mods?)

(Actually maybe this belongs in a different thread, but that thread is closed.  New thread, mods?)

I think it was well established in the closed thread that the authors of this paper (intentionally?) used inappropriate practices to push Meridian's agenda to sell expensive gear and their new format. Is there anything new to say to this?

I'm slightly confused: is the great contention of Meridian that we can hear the ringing introduced by conventional A/D and D/A conversion? This thread hasn't quite clarified things for me, and I already bought two of their papers (and deeply regret doing so ). Many thanks for any advice.

Quote from: krabapple on 2014-12-17 18:44:09

(Actually maybe this belongs in a different thread, but that thread is closed.  New thread, mods?)

I think it was well established in the closed thread that the authors of this paper (intentionally?) used inappropriate practices to push Meridian's agenda to sell expensive gear and their new format. Is there anything new to say to this?

Given the sheer amount of noise on that thread compared to signal, perhaps adding your note above to the end of it would help the newcomer (or better, pin it to the start)?

Given the sheer amount of noise on that thread compared to signal, perhaps adding your note above to the end of it would help the newcomer (or better, pin it to the start)?

I agree, better yet would be to bisect the thread into signal and noise, which will be a humongous task for the holidays.

I'm slightly confused: is the great contention of Meridian that we can hear the ringing introduced by conventional A/D and D/A conversion? This thread hasn't quite clarified things for me, and I already bought two of their papers (and deeply regret doing so ). Many thanks for any advice.

I also bought both papers and FWIW and based them and on other documents and ads of theirs, Meridian and Dolby are definitely of the belief that many people can hear the ringing introduced by conventional A/D and D/A conversion, and that tihs is a major reason for consumers being dissatisfied with the sound quality of their audio systems of all kinds except of course those blessed with Meridian's patented technology (which Dolby has licensed).

IME Meridian has always played this game, but with Dolby it is a clear case of Sic Transit Gloria.

I haven't seen any link reported between MQA and Dolby - have you read something I haven't?

Obviously Meridian licensed MLP to Dolby who sell it as Dolby True HD, but MQA is a different thing.

Cheers,
David.

He's referring to Advanced 96k Upsampling.

He's referring to Advanced 96k Upsampling.

Thanks.

I was answering the question that was asked:

Is it the "...great contention of Meridian that we can hear the ringing introduced by conventional A/D and D/A conversion?"

And it is exactly that which is the contention of both Meridian and Dolby Labs

More specifically:

http://www.dolby.com/us/en/technologies/do...white-paper.pdf (http://www.dolby.com/us/en/technologies/dolby-truehd-lossless-audio-performance-white-paper.pdf)

They are describing "...applying an advanced Apodizing that masks pre-ringing..."

Given Dolby's apparent influence in the AES, this makes that  best paper award some kind of a slam dunk.  IMO it is just a case of one dirty hand buffing the dirt on another. :-(

Ah, I see - thank you.

Quote from: krabapple on 2014-12-17 21:12:25

Given the sheer amount of noise on that thread compared to signal, perhaps adding your note above to the end of it would help the newcomer (or better, pin it to the start)?

I agree, better yet would be to bisect the thread into signal and noise, which will be a humongous task for the holidays.

I would call that a Christmas miracle.

Just what I thought. Revolutionary developments in psychoacoustics my arse.

The funny thing is that, as I understand it, for 44.1ksps you can make a faint case for the potential audibility of said ringing: if you carefully pick your content, and then get some kids with exceptional HF hearing, it's not completely out of the realms of possibility that the ringing might be very barely possibly audible (recalling a post by JJ on SkepticForum).

Of course - and I'm preaching to the choir here, I realise - by the time you get to 88.2ksps ringing is unconditionally inaudible to whatever freak combination of hypothetical listeners and material you could devise. So Meridian's format helps with...what, exactly?

But seriously, is that it? I am bemused as to how people can take this seriously - does nobody in the audience laugh when they try to pull this crap? I'm very, very far from an expert on the subject, but as far as I can tell this is reasonably clear-cut BS. It's not even new BS: issue 16 of "The Audio Critic" (http://www.theaudiocritic.com/back_issues/The_Audio_Critic_16_r.pdf) records Wadia trying to shovel similar shit in 1991 (only their ingenious solution was simply rolling off earlier enough to screw with the audioband, but very slowly...).

if you carefully pick your content, and then get some kids with exceptional HF hearing, it's not completely out of the realms of possibility that the ringing might be very barely possibly audible

But how many of those kids would care?

But seriously, is that it? I am bemused as to how people can take this seriously - does nobody in the audience laugh when they try to pull this crap?

Well, if they take audiophile ethernet cables seriously...

I need help understanding something about this top octave "ringing problem"; perhaps a person who has examined this themselves from a real time display can step in to answer my question aboiut it: Is it a static alteration to the frequency response or does it ripple and fluctuate over time? When I see graphs of it, frozen images of the ripple, would the image be any different if taken say a moment later?

Whatever it is I'd assume the level changing is smaller than the JND for level discrimination at these top frequencies (which only kids can hear anyways). I bet Bob Stuart knows this but strategically ignores it.

Just what I thought. Revolutionary developments in psychoacoustics my arse.

The funny thing is that, as I understand it, for 44.1ksps you can make a faint case for the potential audibility of said ringing: if you carefully pick your content, and then get some kids with exceptional HF hearing, it's not completely out of the realms of possibility that the ringing might be very barely possibly audible (recalling a post by JJ on SkepticForum).

Of course - and I'm preaching to the choir here, I realise - by the time you get to 88.2ksps ringing is unconditionally inaudible to whatever freak combination of hypothetical listeners and material you could devise. So Meridian's format helps with...what, exactly?

But seriously, is that it? I am bemused as to how people can take this seriously - does nobody in the audience laugh when they try to pull this crap? I'm very, very far from an expert on the subject, but as far as I can tell this is reasonably clear-cut BS. It's not even new BS: issue 16 of "The Audio Critic" (http://www.theaudiocritic.com/back_issues/The_Audio_Critic_16_r.pdf) records Wadia trying to shovel similar shit in 1991 (only their ingenious solution was simply rolling off earlier enough to screw with the audioband, but very slowly...).

Oddly enough, Meridian published this photo from the MQA launch event on their website (https://www.meridian-audio.com/news-events/meridian-audio-launches-mqa-master-quality-authenticated/):



The facial expression of the man in the third row is particularly appropriate.

But seriously, is that it? I am bemused as to how people can take this seriously - does nobody in the audience laugh when they try to pull this crap?

I experienced that people are polite at those conferences and don't do that sort of thing, even though sometimes it is warranted. It's more constructive to talk to the speaker later in a constructive manner.

Then again I only attended science and not homeopathy conferences. Many people in AES seem to have pecuniary interests, so dog won't eat dog.

I need help understanding something about this top octave "ringing problem"; perhaps a person who has examined this themselves from a real time display can step in to answer my question aboiut it: Is it a static alteration to the frequency response or does it ripple and fluctuate over time? When I see graphs of it, frozen images of the ripple, would the image be any different if taken say a moment later?

Whatever it is I'd assume the level changing is smaller than the JND for level discrimination at these top frequencies (which only kids can hear anyways). I bet Bob Stuart knows this but strategically ignores it.

And how often do people confuse the Gibbs phenomenon (http://'http://en.wikipedia.org/wiki/Gibbs_phenomenon") for ringing?

Just what I thought. Revolutionary developments in psychoacoustics my arse.

To play devil's advocate and jump to the other side of the argument for a minute:

The AES paper showed that under (what many here would describe as) extreme conditions, people could just about hear a difference, and proved this with statistically significant double blind test results.

This isn't the night and day difference that audiophiles attribute to hi-res or that marketing departments may proclaim, but it's not "nothing" either.

If I was designing these filters, I would aim for a gentle roll off and minimised pre-ring. Even if I couldn't hear the difference myself. It doesn't hurt, and in some circumstances it might help.


Compare ultrasonic filters with dither. In theory, with clean signals, dither is important. In practice, with real-world dirty signals, dither is rarely audible.
In theory, with clean transducers and most of what we currently know about hearing, ultrasonic filters are inaudible. In practice, with real-world transducers and human ears it seems that the characteristics of ultrasonic filters might be just audible.

Everyone understands dither because you can generate quiet sine waves in almost any audio editor, convert the audio to a lower bitdepth without dither, and see harmonic distortion appear. Yet most audible demos of dither use 8-bits, or vastly boosted listening levels, because it's hard to hear the effect when used normally, and undetectable (un ABX-able) with almost any real audio source at a reasonable listening level.

If we can "cheat" in this way to make dither easily audible for a demonstration, why not "cheat" in a comparable way to make ultrasonic filters easily audible for a demonstration? Imagine we run the linear phase vs appodizing filter test at 10kHz, or even 5kHz. That's equally cheating (if not more so) for a filter test as raising the level by 46dB is for a test of a small amplitude effect (dither), but if you did this almost everyone would hear the difference. Then we extrapolate that the same thing happens ~22kHz, but we don't hear it in the same way (or at all) unless something we don't really understand yet happens. But because it's audible in a non-representative case, we decide it's a good thing do anyway.



For me, there's not really sufficient rigorous evidence to start claiming this as fact yet. But it's not an unreasonable theory, and gentle filters with minimised pre-ringing are not bad in themselves, so why not?

The questions I would like answering are:
1. Is the ABXable effect reported in the AES paper due to pre-ringing, or the lack of ultrasonic content?
2. If pre-ringing, is this detected due to artefacts in the amp and/or speakers, or air transmission, or some mechanism in the human ear?
3. If pre-ringing (by whatever mechanism), what is the threshold of audibility?

Cheers,
David.

P.S. remember krabapple's point: even if people can detect a difference, this isn't the difference "heard" by audiophiles in sighted tests or the difference heard when hi-res release use different masters/mastering.

One more thing to me is that in the listening test result they used words like 'round' sound for descriping the difference.
I don't have the paper. Maybe the lowpassed versions even sound better in a way

Quote from: Willakan on 2014-12-18 19:15:11

Just what I thought. Revolutionary developments in psychoacoustics my arse.

To play devil's advocate and jump to the other side of the argument for a minute:

All good and reasonable (and thanks for the shout-out).  Sure, if we *can* design to account for corner cases, and especially if it can done cheaply,  why not?  But for heaven's sake just stop (not you, David, I mean 'audiophiles and marketing departments' and audio journalists/bloggers)  making outlandish claims about the importance and audibility of these effects.  The demonstrable fact is, they are typically miniscule if not inaudible, under normal (or even 'audiophile') home audio conditions.  And STOP implying that 'hi rez'  of itself, means we 'finally get what the musicians heard in the studio'.  That *only* happens if engineers don't radically remaster the audio for consumer delivery (and of course it actually *never* happens, because we are never in the control booth listening to the same system the musicians did). 

There are comparatively *major* audible problems consumers face with home audio that by any rational evaluation deserve more attention than 'audibility of digital filters'.  Hi rez cheerleaders need to switch teams!

Quote from: Willakan on 2014-12-18 19:15:11

Just what I thought. Revolutionary developments in psychoacoustics my arse.

To play devil's advocate and jump to the other side of the argument for a minute:



For me, there's not really sufficient rigorous evidence to start claiming this as fact yet. But it's not an unreasonable theory, and gentle filters with minimised pre-ringing are not bad in themselves, so why not?

The questions I would like answering are:
1. Is the ABXable effect reported in the AES paper due to pre-ringing, or the lack of ultrasonic content?
2. If pre-ringing, is this detected due to artefacts in the amp and/or speakers, or air transmission, or some mechanism in the human ear?
3. If pre-ringing (by whatever mechanism), what is the threshold of audibility?

Cheers,
David.

P.S. remember krabapple's point: even if people can detect a difference, this isn't the difference "heard" by audiophiles in sighted tests or the difference heard when hi-res release use different masters/mastering.

David
These are interesting and thoughtful points, but I think the "not bad in themselves" assertion is problematic. I'm sure you know all this much better than me , but .....
Unlike dithering whose adverse effects are confined to raising the overall noise level which  can be comfortably dismissed as insignificant in most cases, any filter which is non-linear phase and slow roll off has inevitable downsides in terms of aliasing, phase distortion and non- flat frequency response. Surely all of these can be shown to be detectable in corner cases, just like ringing.
So unfortunately in the absence of established evidence of audibility, there is no agreed compromise. You only have to look at the MQA paper to see Stuart and Craven selectively asserting that aliasing can be ignored if the products are below a certain level, in order to justify their choice of filter (whose benefit is ?). Applying a consistent approach to each of the potential downsides would leave them floundering.

Many people still think that applying an orthodox sharp linear phase filters make sense.

That takes us to the list of questions needing answering, which of course they do. I would suggest that c. ( threshold of audibility) bears some refinement- it seems often to be assumed that what matters is the length of pre- ringing in time ( as shown by the impulse response) , but surely the amplitude and frequency matter. If the energy in the pre-ringing s the energy in the transition band, then surely a really steep filter will have less energy in the ringing- even if it goes on longer. If so surely that might at least sometimes be less audible. Apologies if I have got this wrong. Perhaps a real world example might help here

On your last point I don't think it's coincidental that the supposed time domain benefits of relaxed filters never seem to be demonstrated or even illustrated with real examples, because if they were it would serve to show implicitly how rarely they would apply in most music.

That takes us to the list of questions needing answering, which of course they do. I would suggest that c. ( threshold of audibility) bears some refinement- it seems often to be assumed that what matters is the length of pre- ringing in time ( as shown by the impulse response) , but surely the amplitude and frequency matter. If the energy in the pre-ringing s the energy in the transition band, then surely a really steep filter will have less energy in the ringing- even if it goes on longer. If so surely that might at least sometimes be less audible. Apologies if I have got this wrong. Perhaps a real world example might help here

Agreed. I was discussing this with a friend who is a long time audio engineer and more of a perfectionist than I. My argument is that there are three reasons why 44 KHz low pass filter ringing should be inadible which seems to amount to be a paraphrase of the previous paragraph:

(1) The ringing takes place at a frequency that is generally considered to be ultrasonic, which is to say too high of a frequency to be audible

(2) The ringing has a relatively low amplitude compared to the impulse which further decreases its potential audibility.

(3) The ringing is subject to temporal masking which most authorities show (albeit at far more audible frequencies but also a longer masker) to extend several milliseconds before the loud noise, and even longer afte rit.

On your last point I don't think it's coincidental that the supposed time domain benefits of relaxed filters never seem to be demonstrated or even illustrated with real examples, because if they were it would serve to show implicitly how rarely they would apply in most music.

It appears to me that the ringing can be "tuned" by adjusting the width of the transition band, the steepness of the filter,  and the maximum/linear/minimum phase properties of the low pass filter.

The concept of the Apodizing filter seems to me to be a technological step backward. Why not just tune the low pass filter itself for the best possible audible performance (presuming as has not been proven, that the current practice of linear phase filters is somehow audibly suboptimal).  The Apodizing filter is just an add on that seems to be designed to correct something that can be done correctly (if this is even warranted) for no additional cost in terms of complexity. 

These are digital filters and making them right seems like an easy enough thing to do. Again, it seems to be controversial to claim that they are currently suboptimal.  I don't see any compelling evidence to support that claim.

If I were a skeptic, I'd say that the Apodizing filter was contrived to create a patentable device, given that prior art probably precludes patenting just doing it right.

My argument is that there are three reasons why 44 KHz low pass filter ringing should be inadible which seems to amount to be a paraphrase of the previous paragraph:

(1) The ringing takes place at a frequency that is generally considered to be ultrasonic, which is to say too high of a frequency to be audible

(2) The ringing has a relatively low amplitude compared to the impulse which further decreases its potential audibility.

(3) The ringing is subject to temporal masking which most authorities show (albeit at far more audible frequencies but also a longer masker) to extend several milliseconds before the loud noise, and even longer afte rit.

Apart from your 3 points, I have my doubts as to whether actual music excites that ringing in practice. When people demonstrate the pre-ringing, they invariably use either a step function or a sharp pulse of one sample to excite it, and show a plot or oscillogram of it. The latter presumably to approximate a Dirac pulse. Since digital audio is a bandlimited system that cannot represent frequency content beyond the Nyquist limit, such signals can not mean what people apparently think they mean. They are neither a step function, nor a Dirac impulse. They are something with a spectrum that is limited to 1/2 the sampling rate. If you try to come up with a waveform that obeys this bandwidth limit, and at the same time touches all sample values from the original signal, you will invariably arrive at something that shows pre- and post-ringing. That is not an artefact created by the converter, it is something that must have been in the signal to start with.

In one word: The Gibbs-Phenomenon. Something that is excited by a step function can not occur in real-world music, because real-world music doesn't contain step functions, and digital audio signals can not encode step functions. If that's what people mean with pre-ringing, it is a red herring.

However, if we assume that nonlinear processing happened in the digital domain, particularly clipping, then the ringing might be the result of that. But that's another issue.

Depending on where and what we are filtering, of course musical content can make ultrasonic filters ring...
http://www.hydrogenaud.io/forums/index.php?showtopic=68524 (http://www.hydrogenaud.io/forums/index.php?showtopic=68524)
I agree that just showing the impulse response of a filter, and moanng about the ringing, is misleading.
Cheers,
David.

As for whether these gentle filters are ok in themselves, it depends. Flat + linear phase-ish to 20k, dead by 22k is fine. The stuff in the second AES paper is another thing entirely. How close MQA is to that paper I do not know.

Cheers,
David.

Quote from: Arnold B. Krueger on 2014-12-20 12:53:32

My argument is that there are three reasons why 44 KHz low pass filter ringing should be inadible which seems to amount to be a paraphrase of the previous paragraph:

(1) The ringing takes place at a frequency that is generally considered to be ultrasonic, which is to say too high of a frequency to be audible

(2) The ringing has a relatively low amplitude compared to the impulse which further decreases its potential audibility.

(3) The ringing is subject to temporal masking which most authorities show (albeit at far more audible frequencies but also a longer masker) to extend several milliseconds before the loud noise, and even longer afte rit.

Apart from your 3 points, I have my doubts as to whether actual music excites that ringing in practice. When people demonstrate the pre-ringing, they invariably use either a step function or a sharp pulse of one sample to excite it, and show a plot or oscillogram of it. The latter presumably to approximate a Dirac pulse. Since digital audio is a bandlimited system that cannot represent frequency content beyond the Nyquist limit, such signals can not mean what people apparently think they mean. They are neither a step function, nor a Dirac impulse. They are something with a spectrum that is limited to 1/2 the sampling rate. If you try to come up with a waveform that obeys this bandwidth limit, and at the same time touches all sample values from the original signal, you will invariably arrive at something that shows pre- and post-ringing. That is not an artefact created by the converter, it is something that must have been in the signal to start with.

In one word: The Gibbs-Phenomenon. Something that is excited by a step function can not occur in real-world music, because real-world music doesn't contain step functions, and digital audio signals can not encode step functions. If that's what people mean with pre-ringing, it is a red herring.

However, if we assume that nonlinear processing happened in the digital domain, particularly clipping, then the ringing might be the result of that. But that's another issue.

Nonlinear processing in the digital domain still seems to be subject to some limitations that keep it from being as strong of a stimulus as the idealistic stimuli favored by proponents of this alleged problem.  For one thing, any signal in the digital domain still has spectral limits that are imposed by the sample rate. the common situation is that the nonlinear processing "Tries" to create energy that is out of band, but it gets inherently mirrored or aliased down into the system's normal Nyquist Frequency imposed limits.  Secondly, Most real world nonlinear processing whether intentional or accidental does not create the theoretical maximum possible amount of energy.

So, this creates a 4th limiting condition in addition to the 3 listed above:

(4) Real world stimuli generally excite the system far less energetically than a theoretically ideal maximized impulse.

In short, advocates of the audibility of this alleged ringing are not asking people to experience the worst day of their life, but rather to concurrently experience the 4 worst days of their life on the same day. Two words: Mission Impossible.

Nonlinear processing in the digital domain still seems to be subject to some limitations that keep it from being as strong of a stimulus as the idealistic stimuli favored by proponents of this alleged problem.  For one thing, any signal in the digital domain still has spectral limits that are imposed by the sample rate. the common situation is that the nonlinear processing "Tries" to create energy that is out of band, but it gets inherently mirrored or aliased down into the system's normal Nyquist Frequency imposed limits.  Secondly, Most real world nonlinear processing whether intentional or accidental does not create the theoretical maximum possible amount of energy.

So, this creates a 4th limiting condition in addition to the 3 listed above:

(4) Real world stimuli generally excite the system far less energetically than a theoretically ideal maximized impulse.

In short, advocates of the audibility of this alleged ringing are not asking people to experience the worst day of their life, but rather to concurrently experience the 4 worst days of their life on the same day. Two words: Mission Impossible.

Even though the energy of such artefacts is going to be rather low, that doesn't mean they must be inaudible. However, it would be unfair to blame the filters or converters for that. After all, it is the spectral effects of the nonlinear processing that creates those artefacts. Garbage in, garbage out.

Given the mistakes and the propaganda that circle around those issues, my conclusion has been, that each and every time someone uses impulses and/or rectangles in an argument about digital audio, heightened suspicion is in order. Quite frequently the argument contains a more or less well hidden violation of a basic precondition of sampling. The connection between the time-domain view and the frequency-domain view of digital signals is simply too tricky for most people, and even experts sometimes get caught. This makes it very easy to show suggestive diagrams that are bound to be interpreted in a manner that's both false and intentional.

Some kinds of diagrams, which are used frequently even by experts with no hidden agenda, contribute to the false intuition. For example, I have come to dislike the stairstep diagrams of sampling. They are a wrong depiction of what's happening. They are the output of a DAC with zero order hold. As such they can be a signal that might be observed at some point in a circuit, but it is most definitely not a correct rendering of the digital signal, and hence such a diagram shouldn't be used in places where a faithful depiction of the digital signal is expected.

Given the mistakes and the propaganda that circle around those issues, my conclusion has been, that each and every time someone uses impulses and/or rectangles in an argument about digital audio, heightened suspicion is in order. Quite frequently the argument contains a more or less well hidden violation of a basic precondition of sampling. The connection between the time-domain view and the frequency-domain view of digital signals is simply too tricky for most people, and even experts sometimes get caught. This makes it very easy to show suggestive diagrams that are bound to be interpreted in a manner that's both false and intentional.

Some kinds of diagrams, which are used frequently even by experts with no hidden agenda, contribute to the false intuition. For example, I have come to dislike the stairstep diagrams of sampling. They are a wrong depiction of what's happening. They are the output of a DAC with zero order hold. As such they can be a signal that might be observed at some point in a circuit, but it is most definitely not a correct rendering of the digital signal, and hence such a diagram shouldn't be used in places where a faithful depiction of the digital signal is expected.

I agree with your points.

My distrust of square wave testing goes back to long before the advent of digital audio.  I saw it being used to support what seemed to me to be illogical conclusions about tubed power amplifiers in the 60s for example.

OTOH the concept of impulse response has a very solid theoretical footing, but eyeballing the response of gear to impulses is really not any part of its effective use. Furthermore, most impulse response based evaluation of audio gear and listening environments is not based on the actual use of impulses!

I've Heard the Future of Streaming: Meridian's MQA (http://www.stereophile.com/content/ive-heard-future-streaming-meridians-mqa)

JA
- Returning to the triangle of musical information in fig.1, the old question is why do we need to preserve and reproduce frequencies above the limit of human hearing, even if we can do it? Bob spent some time discussing this in his presentation and it comes down to the fact that the ear-brain doesn't just operate as a frequency analyzer. Evolution has fine-tuned the system to be able to detect temporal differences that are equivalent to a bandwidth considerably greater than 20kHz and that the anti-aliasing filters in A/D converters and reconstruction filters in D/A converters introduce temporal "smearing" that it is considerably greater than what our ear-brains are tuned to expect from natural sounds: this "smearing" is, I believe, responsible for so-called "digital" sound.

The MQA encoder and decoder together have been designed to have a transient response of the same form and order as that of the temporal sensitivity of the ear-brain. And if at the MQA-encoding stage, the temporal effect of the A/D converter can be compensated for, the complete system offers a transparent window into the original musical event. Meridian describes this as "taking an original master further, toward the original performance, in an analogous way to the processes expert antique picture restorers use to clean the grime and discolored varnish from an Old Master to reveal the original color and vibrancy of the work."

This, from someone who attended this what can really be done with "Hi Rez" and compressed streaming demo (http://www.onhifi.com/features/20010615.htm) 

It gets better: Meridian's MQA: One Listener's Impression (http://www.stereophile.com/content/meridians-mqa-one-listeners-impression)

JS
- In Audio High's exceedingly dry listening room, we began with an 24/88.2k file of Hilary Hahn playing what I believe was a movement from J. S. Bach's Violin Concerto No.2 in E, BWV1042. As compelling as the untreated hi-res file sounded, I literally laughed at the difference when the MQA version began. Not only did it feel as though a veil had been lifted, with far more color to the sound, but instruments also possessed more body. With more meat on dem bones, I also noticed less of a digital edge on the violin. I've heard Hahn in concert several times, and this was the closest to real I've ever heard her violin sound on recording.

So there you have it. MQA sounds better that the original (24/88) Hi Rez file.

cheers,

AJ

It gets better: Meridian's MQA: One Listener's Impression (http://www.stereophile.com/content/meridians-mqa-one-listeners-impression)

Quote

JS
- In Audio High's exceedingly dry listening room, we began with an 24/88.2k file of Hilary Hahn playing what I believe was a movement from J. S. Bach's Violin Concerto No.2 in E, BWV1042. As compelling as the untreated hi-res file sounded, I literally laughed at the difference when the MQA version began. Not only did it feel as though a veil had been lifted, with far more color to the sound, but instruments also possessed more body. With more meat on dem bones, I also noticed less of a digital edge on the violin. I've heard Hahn in concert several times, and this was the closest to real I've ever heard her violin sound on recording.

So there you have it. MQA sounds better that the original (24/88) Hi Rez file.

cheers,

AJ

Everything they stand by is spot on IMO. I'm glad they're doing this, just what industry needs.

It gets better: Meridian's MQA: One Listener's Impression (http://www.stereophile.com/content/meridians-mqa-one-listeners-impression)

Quote

JS
... ... ... I literally laughed at the difference when the MQA version began. Not only did it feel as though a veil had been lifted, with far more color to the sound, but instruments also possessed more body. With more meat on dem bones, ... ... ...

Is anybody else thinking, "That's exactly what I experience when I  turn up the volume a touch" ?

I'd like to think that would be just too obvious for an experienced reviewer, but...

Everything they stand by is spot on IMO.

Nothing wrong with capitalism, though some might question the morality of legal fraud.

I'm glad they're doing this, just what industry needs.

What "industry"?
The grime and discolored varnish master restorer? The audio$cam one, or...??

cheers,

AJ

The thing that I find amazing is the list of superlatives that people use to describe the improvement. I've been lucky enough to hear studio master recordings several times. It would be a lie to say they always sound great, but they often sound better than the released versions. Other than where the released versions are obviously butchered, I've never heard a large enough difference to justify the kind of words people use to describe the improvements. People did this with SACD, 24/96, and now MQA.

I can usually hear and appreciate the difference when people take more care in bringing the sound of the master tapes to the release (first generation tape, properly adjusted tape machine, scrupulous care to avoid over-use of compression and EQ, etc), but the CD release is then so close to the master tape, that it doesn't leave room for the quantity of improvement suggests by these superlatives.

I'm obviously deaf, or enjoying music is a completely different way. I find being happy and/or emotional is a much greater contributor to musical enjoyment than better speakers.

Cheers,
David.

I find being happy and/or emotional is a much greater contributor to musical enjoyment than better speakers.

Hear, hear!

It's a real shame so many people posing as intellectuals, seem to forget that life works more or less on a logarythmic scale; i.e, after reaching a certain quality standard, it takes a huge ammount of money and effort to actually move on to the next stage and reach any real, tangible difference in terms of quality.

Quote

JS
- In Audio High's exceedingly dry listening room, we began with an 24/88.2k file of Hilary Hahn playing what I believe was a movement from J. S. Bach's Violin Concerto No.2 in E, BWV1042. As compelling as the untreated hi-res file sounded, I literally laughed at the difference when the MQA version began. Not only did it feel as though a veil had been lifted, with far more color to the sound, but instruments also possessed more body. With more meat on dem bones, I also noticed less of a digital edge on the violin. I've heard Hahn in concert several times, and this was the closest to real I've ever heard her violin sound on recording.

So there you have it. MQA sounds better that the original (24/88) Hi Rez file.

cheers,

AJ

let me guess without clicking through: JS = Jason Serinus. 

Quote from: giro1991 on 2015-01-14 11:38:40

Everything they stand by is spot on IMO.

What "industry"?
The grime and discolored varnish master restorer? The audio$cam one, or...??

cheers,

AJ

you know...*the business*  Amir kept telling us about, that we don't understand but he does.

Meridian and Stereophile are just giving us *the business*.

Serinus in full flower :

Not only did it feel as though a veil had been lifted, ....

*Another* one? How many f*cking veils are there to lift, anyway, before audio reproduction becomes 'transparent'.

It seems like a new one is found every few years, by Stereophile , TAS, and their paymasters.

I'm obviously deaf, or enjoying music is a completely different way. I find being happy and/or emotional is a much greater contributor to musical enjoyment than better speakers.

Here is something to enhance subjective sound quality: http://porknwhiskey.com/wp-content/uploads...D_5872-Edit.jpg (http://porknwhiskey.com/wp-content/uploads/2012/01/Talisker-20120124-RCD_5872-Edit.jpg)
Right now I am lubricating my hearing equipment (i.e. my brain) with just a tad of it.

... I actually can ABX out quite a few of these, but I do make mistakes under extended sighted evaluation without any other reference than the age and price tag.

+1 for bourbon signal processing.

On the other side of the coin, when my commute home from work puts me in a bad mood, there is some definite coloration of the music. As soon as someone cuts me off or fails to merge properly, I want to turn everything off. I immediately stop enjoying whatever it is I was rocking out to! Fix this Meridian!

Quote from: ajinfla on 2015-01-13 22:11:58

Quote

JS
- In Audio High's exceedingly dry listening room, we began with an 24/88.2k file of Hilary Hahn playing what I believe was a movement from J. S. Bach's Violin Concerto No.2 in E, BWV1042. As compelling as the untreated hi-res file sounded, I literally laughed at the difference when the MQA version began. Not only did it feel as though a veil had been lifted, with far more color to the sound, but instruments also possessed more body. With more meat on dem bones, I also noticed less of a digital edge on the violin. I've heard Hahn in concert several times, and this was the closest to real I've ever heard her violin sound on recording.

So there you have it. MQA sounds better that the original (24/88) Hi Rez file.

cheers,

AJ

let me guess without clicking through: JS = Jason Serinus. 

Confirmed: Jason Seronis' Stereophile Article on MQA (http://www.stereophile.com/content/meridians-mqa-one-listeners-impression)

At the risk of being TOSed, I'd be very curious how an encoding format can increase the size of an instrument after recording ("instruments also possessed more body") and influencing synesthesia ("far more color to the sound").

At the risk of being TOSed, I'd be very curious how an encoding format can increase the size of an instrument after recording ("instruments also possessed more body") and influencing synesthesia ("far more color to the sound").

Begs the question what would happen if someone devised a perceptual coder that did this sort of thing. 

We've long had recording formats such as vinyl that intentionally and unintentionally played with the spectral contents and imaging of recordings in potentially audible ways.

Adding color to the sound is just a matter of audible boosting and/or cutting spectral bands. Increasing or decreasing imaging related perceptions is just matter of adding or subrtacting cross feed between the channels.  You can mix the two as you desire.

Thought it would be more appropriate to post this here rather than create a new thread, the BBC just posted a video attempting to explain the state and future of "hi-res;" a lot of cringeworthy and groanworthy stuff as you'd expect (all this time I had no idea they were Wahvs... ), though in fairness they did at least attempt to address the matter of transparency (with the video analogy) and raise the question as to whether anything is to be gained from exorbitant file sizes and devices. 

http://www.bbc.com/news/technology-30880509 (http://www.bbc.com/news/technology-30880509)

And again there's not a single mention of where the bad sound quality actually comes from: loudness war, terrible mixing and mastering ..

Apart from equating WAV to CD quality, and FLAC to HiRes, I thought that was pretty good.

EDIT: xnor: well, that can apply equally well to HiRes, so...

It appears when they demo MQA, it's a comparison now (http://www.avsforum.com/forum/173-2-channel-audio/1844737-revisiting-meridian-mqa-ces-2015-a.html).
Still trying to figure out what time "smearing" JA is talking about. Didn't he use to post here?

cheers,

AJ

@2Bdecided: Exactly, that is my point!

Wow, all they offered for comparison was 128 CBR mp3? They must be really confident in the superiority of MQA ...

And again there's not a single mention of where the bad sound quality actually comes from: loudness war, terrible mixing and mastering ..

I'm slowly realizing that's HA's fault, too. Bring it out to the public! Who's our PR guy?

Quote from: xnor on 2015-01-20 15:40:52

And again there's not a single mention of where the bad sound quality actually comes from: loudness war, terrible mixing and mastering ..

I'm slowly realizing that's HA's fault, too. Bring it out to the public! Who's our PR guy?

I can read whatever you put on the teleprompter!

And again there's not a single mention of where the bad sound quality actually comes from: loudness war, terrible mixing and mastering ..

That's because MQA can fix that griminess:

Meridian describes this as "taking an original master further, toward the original performance, in an analogous way to the processes expert antique picture restorers use to clean the grime and discolored varnish from an Old Master to reveal the original color and vibrancy of the work."

Just compare it to 128k mp3 and you too will believe.

cheers,

AJ

Who's our PR guy?

Is Amir too busy these days?
Whoever it is, make sure they have science-ish looking graphs to point at.

But no lab coats, hand held meters or putty in their ears.

cheers,

AJ

I wonder why audiophiles suddenly should think that noise can be left out. I had to read for many years how important the maybe last 8 bit of noise are for good sound leave alone the harmonics burried there at 50kHz+

Quote from: Kohlrabi on 2015-01-20 21:37:56

Who's our PR guy?

Is Amir too busy these days?
Whoever it is, make sure they have science-ish looking graphs to point at.

But no lab coats, hand held meters or putty in their ears.

About that graph (it's one of several slides, supposedly from Stuart's PP presentation, and associated with the Dec 2014 Stereophile article here: http://www.stereophile.com/content/ive-hea...g-meridians-mqa (http://www.stereophile.com/content/ive-heard-future-streaming-meridians-mqa) ) ...

Fig.1 is a variation on what is called a "Shannon Diagram"; it shows the information space required by an audio signal with level shown as the vertical scale in dB, frequency as the horizontal scale in kilohertz.

While it's not wrong to create an ad hoc info-graphic (the Stuart graph may be okay if their premises are sound), confused nomenclature is not ideal. I Google'd  "Shannon Diagram" and only came up with something very different-looking here:
http://en.wikipedia.org/wiki/A_Mathematica...f_Communication (http://en.wikipedia.org/wiki/A_Mathematical_Theory_of_Communication)

More to the point ... I've never seen info-graphics in the style of Stuart's Origami-like depiction.

The current issue of "the absolute sound" (Issue #253 - May/June 2015) has a 6 page article:

'Beyond High Resolution: Meridian Audio’s Master Quality Authenticated (MQA)'
by Robert Harley

The current issue of "the absolute sound" (Issue #253 - May/June 2015) has a 6 page article:

'Beyond High Resolution: Meridian Audio’s Master Quality Authenticated (MQA)'
by Robert Harley

You honestly suggest we should buy this?

It's not a bad science fiction magazine.
And it has lot's of pretty pictures.

Quote from: Speedskater on 2015-04-17 14:57:59

The current issue of "the absolute sound" (Issue #253 - May/June 2015) has a 6 page article:

'Beyond High Resolution: Meridian Audio’s Master Quality Authenticated (MQA)'
by Robert Harley

You honestly suggest we should buy this?

Everyone should from time-to-time.  I buy 3 per year...just to see what people will spend on.  Even my kids cannot believe some of the stuff that is printed.  Very good to see.  Utter bunk mostly...

After further inspection:
The current issue of "the absolute sound" (Issue #253 - May/June 2015) contains:
6 pages : main article
1 page : MQA and the industry
6 pages : interview with J. Robert Stuart of Meridian

After further inspection:
The current issue of "the absolute sound" (Issue #253 - May/June 2015) contains:
6 pages : main article
1 page : MQA and the industry
6 pages : interview with J. Robert Stuart of Meridian

Waiting for your summary.

The current issue of "the absolute sound" (Issue #253 - May/June 2015) has a 6 page article:

'Beyond High Resolution: Meridian Audio’s Master Quality Authenticated (MQA)'
by Robert Harley

I'd have preferred the article be titled "Never mind the bollocks, here's the Meridian Audio’s Master Quality Authenticated (MQA)".

But that's just me.

Quote from: Speedskater on 2015-04-17 14:57:59

The current issue of "the absolute sound" (Issue #253 - May/June 2015) has a 6 page article:

'Beyond High Resolution: Meridian Audio’s Master Quality Authenticated (MQA)'
by Robert Harley

I'd have preferred the article be titled "Never mind the bollocks, here's the Meridian Audio’s Master Quality Authenticated (MQA)".

But that's just me.

Not only you. If ringing distortion at its low level can in any way influence sound in the audible band you should better have no content at all above. I always wonder if people state hearing the ringing in PCM at 192kHz. This marketing BS has gone to far already.

Not only you. If ringing distortion at its low level can in any way influence sound in the audible band you should better have no content at all above. I always wonder if people state hearing the ringing in PCM at 192kHz. This marketing BS has gone to far already.

I come from the school that if this is the only way they can make their argument, it's actually a sign of progress.

All I'm getting from this is yet another hymn to hi-res music, to be honest. The usual "it sounds so much better than regular CD!".
The only "advantage" is that it supposedly takes as much space as a regular audio CD, so you get "more" for "free". Kind of like HDCD?

So how does a MQA file size compare to a 24/192 FLAC file?

I know that a lot of high rez FLAC files compress like crazy because after all, there's nothing but zeros in a lot of that 24/192 file.

The obvious comparison would be to a 16/44 uncompressed .wav file which is what the MQA blurb seems to be using as its baseline.

Anyone been able to find MQA files to download to try with the Meridian Explorer2 MQA DAC?

As far as I can see Tidal has not started streaming MQA data yet.

The esteemed Robert Harley, never one to miss an opportunity to demonstrate he has no idea how sampling theorem works (he even manages to bring up "a sampled system cannot convey time differences shorter than two sample periods" as an "inviolable law" of pre-MQA sampling), has excreted a great deal of words on MQA.

http://www.theabsolutesound.com/articles/b...igh-resolution/ (http://www.theabsolutesound.com/articles/beyond-high-resolution/)

Truly, digital filtering is the new jitter, as far as incoherent audiophile ramblings are concerned.

The esteemed Robert Harley, never one to miss an opportunity to demonstrate he has no idea how sampling theorem works (he even manages to bring up "a sampled system cannot convey time differences shorter than two sample periods" as an "inviolable law" of pre-MQA sampling), has excreted a great deal of words on MQA.

http://www.theabsolutesound.com/articles/b...igh-resolution/ (http://www.theabsolutesound.com/articles/beyond-high-resolution/)

Truly, digital filtering is the new jitter, as far as incoherent audiophile ramblings are concerned.

What Harley actually says is:

"Once inviolable “laws” of sampling theory, such as “a sampled system cannot convey time differences shorter than two sample periods,” are exposed as merely the conventional wisdom of an earlier age."

So what he says is in a sense a little bit different than affirming the false idea. More properly stated, it is a straw man, since it has never been true, but more than a few golden ears who should know better have asserted it, most (in)famously Kuncher (sp?).

One of the differences with MQA appears to be that it is capable of turning on an indicator when it is in play. This is obviously, a sighted cue. The efficacy of this refinement goes all the way back to the advent of FM Stereo, where FM stations discovered that they could vastly increase listener perceived satisfaction by broadcasting the 19 KHz FM stereo pilot signal, no actual stereo recordings required. ;-)

It seems MQA will stay a lot longer than I thought....

http://www.twice.com/news/audio/pioneer-s-...table-mqa/58527 (http://www.twice.com/news/audio/pioneer-s-xdp-100r-1st-high-res-portable-mqa/58527)


To be honest, I am utterly surprised to see this technology appearing from non-Meridian companies. It seems Meridian is pushing MQA really hard.

It seems MQA will stay a lot longer than I thought....

http://www.twice.com/news/audio/pioneer-s-...table-mqa/58527 (http://www.twice.com/news/audio/pioneer-s-xdp-100r-1st-high-res-portable-mqa/58527)


To be honest, I am utterly surprised to see this technology appearing from non-Meridian companies. It seems Meridian is pushing MQA really hard.

I'm surprised you're surprised that a nearly $800 portable (http://www.custom-cable.co.uk/pioneer-xdp-100r-portable-player.html) wouldn't have (need?) nonsense bling like MQA.
The question remains whether nonsense bling like this will sell, but there is only one way to answer that.

cheers,

AJ

http://www.twice.com/news/audio/pioneer-s-...table-mqa/58527 (http://www.twice.com/news/audio/pioneer-s-xdp-100r-1st-high-res-portable-mqa/58527)

I'm surprised you're surprised that a nearly $800 portable (http://www.custom-cable.co.uk/pioneer-xdp-100r-portable-player.html) wouldn't have (need?) nonsense bling like MQA.
The question remains whether nonsense bling like this will sell, but there is only one way to answer that.

Finaly dsd up to 11.2MHz on a portabe!!

Finaly dsd up to 11.2MHz on a portabe!!

Time will tell if hi-res audio is a lossy business or not

The bit I like best on that page....

Want to read more stories like this?

Well, no, actually.  But something tells me that I probably will.

I'm not sure if this thing comes with headphones that have the high frequency extension necessary to "appreciate" the added range, but for that matter does anyone here have links to third party measurements of the high frequency response of the Pono's headphone output?

I'm not sure if this thing comes with headphones that have the high frequency extension necessary to "appreciate" the added range, but for that matter does anyone here have links to third party measurements of the high frequency response of the Pono's headphone output?

Relevance?

I was wrong to specify Pono. Sorry. I should have written: "Does anyone have links to third-party measurements of the headphone outputs' high frequency response of any of these various portable Hi-Res music players?"

Last I heard Pono was said to be MQA capable with a firmware upgrade. Since I have little interest in these for my own use, I don't follow them closely, and  I'm not sure if that ever came to fruition. Did it?

If these various MQA portable players claim their sound is better at the higher frequencies, yet their electronics don't have much HF output because most typical, off the shelf headphone amp chips don't support much above 20kHz or so (at least I'd think), it would be interesting.

I didn't know Meridian had specified that reproduction of frequencies >20k was necessary.

I think they are playing word games with lossy audio re-categorizing it as lossless so they can create an industry and make a lot of money via patents and licensing and so forth. 
I don't trust it.  I am putting my faith in future improvements of web streaming going way of AAC or OPUS.  We already have 256 kBps AAC internet radio and it sounds pretty good without cloggin up nets with 96 kHz nonsense.  And if websites would just adopt that for streams, most people would be happy.  And they could still provide lossless downloads like SoundCloud does. 

I used to offer 48 kHz 24-bit FLAC's on SoundCloud for free download.  Now I offer 44.1 kHz 16-bit FLAC's for more compatibility with portable players.  Most people occasionally complain about the sound quality of their conversions down to 128 kBps MPEG streaming. 

My point is, there's not a true demand for this technology which is must trying to pull the wool over people's eyes.  We don't need more gear.  Musicians and engineers already have plenty of hardware and software tools and formats to choose from.  And most of us record 24-bit or 32-bit float and render down to 16-bit but still at 44.1 kHz or 48 kHz.  Most people in the studio really aren't into 96 kHz at all.  And they would be the gatekeepers of the master encodings.  It's fine as it is now.  I use my home computer and gear to create my own masters and deliver the formats conveniently without having to go to a "specialist" with Meridian.  Seriously geez!

i'm the guy who used to have a miniDisc recorder with 24-bit ADC's for goodness sake... that was a big deal back then because not everything was 24-bit yet, but it was still utilized for the ATRAC compression which was about as good as humble MP3 encoding.  Yes, the input was less noisey which is good, but still 90 % (or whatever percent it is) of the audio was thrown out to get the remainder onto those tiny discs!  My point is, this Meridian junk is probably similar in terms of being a fancy high quality encoding front end for a lousy=lossy back end. 

I don't trust it!  Just let me keep my FLAC's and WavPacks and low-brow MP3's there's no need for another middle man.  And now they say they are going to disguise their product within ALAC and FLAC containers?  Don't do it, man!  corruption of formats that work!

It is a shame how this HiBit business tries to push several things lately. I bet you will get some MQA releases exclusively mastered to sound better as the releases before. That way they always faked improvement. Even so called audiophile labels makes fools out of their customers when a CD suddenly can't hold enough dynamics. See the last Al Di Meola as only one out of many. (http://dr.loudness-war.info/album/list?artist=al+di+meola&album=elysium)The CD has DR6 while the 24/96 has DR12. In this case DR tells us something.
This dsd rebirth is even more fascinating! Marketed as the one and only digital format able to catch the magic of analog mixing gear.
These weird things atm make me wonder if the business goes really that bad so they have to push such crap to survive.

These weird things atm make me wonder if the business goes really that bad so they have to push such crap to survive.

yes. this.

It is a shame how this HiBit business tries to push several things lately. I bet you will get some MQA releases exclusively mastered to sound better as the releases before. That way they always faked improvement. Even so called audiophile labels makes fools out of their customers when a CD suddenly can't hold enough dynamics. See the last Al Di Meola as only one out of many. (http://dr.loudness-war.info/album/list?artist=al+di+meola&album=elysium)The CD has DR6 while the 24/96 has DR12. In this case DR tells us something.
This dsd rebirth is even more fascinating! Marketed as the one and only digital format able to catch the magic of analog mixing gear.
These weird things atm make me wonder if the business goes really that bad so they have to push such crap to survive.

Well it makes me hope that the business may push better masterings as a survival strategy. Of course annoying that they try to fool people into thinking the difference is in the format, but worth it we finally get over the loudness war.
On the other hand ... chances are that they will release hypercompressed masterings to the rest of us, only to act as proof of their (selling) point.

I am not sure if it has been discussed around here, but: http://www.meridianunplugged.com/ubbthread...p;Number=227413 (http://www.meridianunplugged.com/ubbthreads/ubbthreads.php?ubb=showflat&Number=227413)
It seems that an MQA source will ask the DAC to identify itself and then feed the 'appropriate' signal. Because the identification will be encrypted certificate-based (most likely not so much flawed as CSS on DVD, and most likely not so many geeks trying to hack it) one can only speculate whether they can choose to deliver different-sounding signals depending on how they identify.

(Don't think I will rush to my Meridian dealer to have my speaker firmware upgraded.)

Well it makes me hope that the business may push better masterings as a survival strategy. Of course annoying that they try to fool people into thinking the difference is in the format, but worth it we finally get over the loudness war.
On the other hand ... chances are that they will release hypercompressed masterings to the rest of us, only to act as proof of their (selling) point.

Hasn't this happened already many years ago when SACD was introduced? It failed, and one could argue why. Of course this doesn't mean that it will fail again.

I think it is fair to say that the content industry has tried to kill off the "normal" CD with all their might. It started when CDs became copiable with the advent of CD-R. This gave rise to CD copy protection schemes, some of which worsened the sound. It was in vain and the CD copy protection schemes died out.

Next try was the SACD. A number of releases indeed had better mastering than the equivalent CD. However, they needed to be CD compatible, which means that they had to provide this compatibility layer. And that means that you look bad when the CD and the DSD layer has significantly different levels. And that is a hindrance to making both layers significantly different in quality. Furthermore, you would have to master twice for a single product. And the customers were already wary of the copy protection schemes of the CD before, and (rightly) saw the SACD as a trick to establish copy protection once more.

Then iPod & Co. came along and showed everyone that copy protection really was unacceptable, as it made transferring your legally acquired titles to the mobile player a steeplechase. Effectively, if you got your material legally, you were more restricted in what you could do with it, than those who had illegal rips.

Now we have hi-res streaming, and we can already see how they try to rip off the customer again. There are numerous cases where the alleged high-res material is really an upconverted CD-quality original. People have started to obfuscate this by adding some fuzz that makes the higher frequences a bit busy for people who look at spectrograms. This is starting to become so widespread that formerly respectable companies latch on to it. See the Clari-Fi stuff from Harman with their expensive promo video last year. I recently became aware that something similar is being sold with the newest smartphones by Samsung. We really have the next wave of bullshit rolling in full force.

I see MQA as another attempt at establishing a form of copy control. Its benefit for the customer is basically nonexistent, but it might seem to some business players that it can serve as a trojan horse to get more control over copying. In this regard, it is nothing new. The industry keeps on trying the same thing over and over again, apparently in the belief that if it didn't work, you need to try harder.

Isn't MQA supposed to be completely backwards compatible?
Isn't it just putting aliased content below levels of the original content? So if it should gain acceptance then I don't see how implementing the unfolding algorithm would be that hard, but that of course would be illegal because then they don't own their license fee....

Maybe we should implement an open and free alternative?


Anyway, I'm wondering why audiophiles are not complaining about the lower bits being "abused" to store something other than was recorded. That was the whole point of going to 24 bits in "high-resolution" audio files, right?

People have started to obfuscate this by adding some fuzz that makes the higher frequences a bit busy for people who look at spectrograms.

Wasn't that just tape hiss?

Anyway, I'm wondering why audiophiles are not complaining about the lower bits being "abused" to store something other than was recorded.

Oh, but it stores reconstructions of what was recorded, but lost when converting to digital, right? Isn't that the point Bob S. is touting about knowing which ADCs were used?

(Why does one not just remaster it properly?)

I'm not sure I understand.

I assumed that tracks were recorded normally e.g. into 24/192, edited, mixed ... and finally the high frequency content is attenuated and aliased into e.g. 48 kHz sampling rate. Isn't that basically it?

This wouldn't work if the signal content at high frequencies wasn't low level and noise-like to begin with. ADCs could produce a rising noise floor at higher frequencies, which of course needs to be dealt with.

It is a shame how this HiBit business tries to push several things lately. I bet you will get some MQA releases exclusively mastered to sound better as the releases before. That way they always faked improvement.

Has anything like this surfaced with the Pono, yet? And I doubt you will get "better" masters, because people in charge don't realize what's "wrong" with the old ones.

Has anything like this surfaced with the Pono, yet? And I doubt you will get "better" masters, because people in charge don't realize what's "wrong" with the old ones.

Now i don't know what you mean. These new HiBit releases mostly for sure sound different as the CD remaster before and the DVD version and the older remaster and the original realease...
Now maybe less compressed as the last attempt. There are funny examples. At Acoustic Sounds they sell for example a Blues album. (http://store.acousticsounds.com/d/94461/Jimmy_Rogers-Blue_Bird-FLAC_96kHz24bit_Download)The 24/48 is mastered by Doug Sax, the 24/96 is mastered by Kevin Grey and the DSD version is no idea but most likely a bit more huggle bass. If there was a MQA version i bet it will be different again.

One more thing Porcus, regarding (re)mastering. If you've recorded at 48 or even 44.1 kHz then you cannot create a genuine "hi-res" track from it. MQA doesn't help here either.

I think what they want to achieve is some minimal recording format (say 24/96 or even 24/192) with MQA'd tracks, so that the consumer doesn't have to worry about fake "hi-res". But from a purely technical pov, I don't think anything prevents one from creating a fake "hi-res" MQA file...

I assumed that tracks were recorded normally e.g. into 24/192, edited, mixed ... and finally the high frequency content is attenuated and aliased into e.g. 48 kHz sampling rate. Isn't that basically it?

Not really, if the buzz is to be believed (which is certainly some of an "if"). Seems that they want to reverse ADC artifacts from recordings that were certainly not recorded into 24/192.
http://www.audiostream.com/content/tidal-s...mqa-music-files (http://www.audiostream.com/content/tidal-successfully-streams-meridian-mqa-music-files)

One more thing Porcus, regarding (re)mastering. If you've recorded at 48 or even 44.1 kHz then you cannot create a genuine "hi-res" track from it. MQA doesn't help here either.

Not a "genuine", but if we for the sake of the argument buy into the claim that anti-aliasing filters with artifacts way down in the audible range were employed upon digitizing, then you can certainly try to improve. As you point out, you cannot (re-) create a "genuine" lossless version of what was filtered off, but you can in principle use knowledge of the filter to improve.

(Again, I do not want to sound like I endorse their claims, but there could be more to it than ultrasound mumbojumbo.)

I assumed that tracks were recorded normally e.g. into 24/192, edited, mixed ... and finally the high frequency content is attenuated and aliased into e.g. 48 kHz sampling rate. Isn't that basically it?

IIRC they downsample from 192 to 96 with a very shallow filter (very wide transition band, very narrow impulse response), even allowing some aliasing above 30kHz or so.

Then they take this 96k recording and fold the 24-48k band into the lower bits of the baseband, obtaining a 48k encoded recording with '96k bandwidth' and '192k temporal accuracy'.

Not really, if the buzz is to be believed (which is certainly some of an "if"). Seems that they want to reverse ADC artifacts from recordings that were certainly not recorded into 24/192.

This only makes sense for CD-rate recordings where at least two major classes of recording-time anti-aliasing filters can be distinguished:
-PCM1630-style analogue (minimum phase, starting at 20kHz, and -surprisingly- quite a lot of attenuation at Nyquist)
-half-band FIR (linear phase, -6dB or so at Nyquist)

Presumably Meridian would want to replay the former with a half-band linear-phase FIR, leaving the result dominated by the PCM1630, and the latter by a minimum phase apodising filter, leaving the result dominated by the MQA filter.

One more thing Porcus, regarding (re)mastering. If you've recorded at 48 or even 44.1 kHz then you cannot create a genuine "hi-res" track from it. MQA doesn't help here either.

I think what they want to achieve is some minimal recording format (say 24/96 or even 24/192) with MQA'd tracks, so that the consumer doesn't have to worry about fake "hi-res". But from a purely technical pov, I don't think anything prevents one from creating a fake "hi-res" MQA file...

Some would argue 16/44.1 is hi-res...

Not really, if the buzz is to be believed (which is certainly some of an "if"). Seems that they want to reverse ADC artifacts from recordings that were certainly not recorded into 24/192.
http://www.audiostream.com/content/tidal-s...mqa-music-files (http://www.audiostream.com/content/tidal-successfully-streams-meridian-mqa-music-files)

[...]

Not a "genuine", but if we for the sake of the argument buy into the claim that anti-aliasing filters with artifacts way down in the audible range were employed upon digitizing, then you can certainly try to improve. As you point out, you cannot (re-) create a "genuine" lossless version of what was filtered off, but you can in principle use knowledge of the filter to improve.

(Again, I do not want to sound like I endorse their claims, but there could be more to it than ultrasound mumbojumbo.)

I'm really unsure about this.
On the reversal of ADC artifacts:
As I said, one thing could be handling the high frequency noise produced by ADCs that would only be a waste of space. Especially to PCM converted DSD comes to mind.
Another thing could be that they seem to dislike steep and linear-phase low pass filters, so they could use a slow filter with a lower cutoff frequency to reduce evil ringing? to theoretically get nice looking impulse responses.

But we'd need to hear a technical person talk about what actually happens instead of this marketing stuff.


I don't think that anything changes about digitizing. MQA just seems to compress a high sampling rate PCM file to a low sampling rate one by aliasing the high frequency content and storing it in the "noise bits". It's not lossless if you consider the "noise bits" information.
This is different from aliasing happening during A/D conversion, e.g. a small portion (until the filter reaches its stopband) of signal energy above 96 kHz (given 192 kHz sampling) "folding back". This is usually just pure noise and MQA would have to go to 384 kHz anyway to reproduce it, so I don't see that happening or any good reason for it. You also cannot "unfold" this. Information is lost during sampling.

I could be mistaken though.

Quote from: xnor on 2015-09-14 13:05:52

One more thing Porcus, regarding (re)mastering. If you've recorded at 48 or even 44.1 kHz then you cannot create a genuine "hi-res" track from it. MQA doesn't help here either.

Not a "genuine", but if we for the sake of the argument buy into the claim that anti-aliasing filters with artifacts way down in the audible range were employed upon digitizing, then you can certainly try to improve. As you point out, you cannot (re-) create a "genuine" lossless version of what was filtered off, but you can in principle use knowledge of the filter to improve.

Try...and fail.

Even if you knew the exact ADC used and the response of its anti-aliasing filter, you simply cannot reliably recreate frequency content above Nyquist.

As an example, suppose you have content sampled at 48kHz and there is a tone at 22k.  How do you know this tone was originally 26k and not simply 22k all along?  What if there were both 22k and 26k in the source, how would you know how much of which?

There is no "in principle" unless you know some law that gets you around Shannon.

The way I understand it, they convey extra information in a data channel that is hidden in the noise. That's an old trick that people have found many applications for, for example surround information, watermarking, etc.

It is not a great stretch of imagination to use such a data channel for conveying high frequency information. You are not bound by Nyquist here in its strict sense, because the data must first be decoded and processed before it can be used to modify the sound output. It sort of requires that the levels you want to encode in this extra channel are rather low, otherwise you need too much extra data, but I understand that this is a prerequisite that Stuart et.al. have mentioned themselves.

If you want a simple to understand graphical picture, you'd imagine a sound wave at a fundamental frequency, with some superimposed small ripples at higher frequencies, which would go lost when sampled at standard rates. Rather than sampling the entire thing at higher sampling rates, you could sample it at the standard rate and note the difference to the normal sample at several instants before you take the next sample. Since the difference will be small, it can be encoded with only a few bits, rather than spending an entire wordlength.

To put it in another way, it would be oversampling where the in-between-samples are kept and encoded with an efficient difference code. I hope you can understand my attempt at describing it without an actual picture.

I'm not saying that this is what they are doing. It is just meant as a relatively simple example of a sampling method that can capture small amounts of high frequencies while sampling at the base rate. The effect of the entire thing is a kind of tradeoff that surrenders some dynamic range for a small amount of high frequency. This can be made backwards compatible in that the extra data (and with it the high frequencies) are lost when playing it back with a normal DAC that is unaware of the scheme. The lost dynamic range remains lost, however.

It does provide the possibilty to sell (limited) high resolution, in the sense that higher frequencies are contained with restricted levels, at the expense of some dynamic range, while at the same time being compatible with the CD format. In other words, it offers the content providers the opportunity to make a single product that is both "high-res" and can be played back with CD-format equipment, instead of having to produce the product in two separate formats. That could be attractive to some, but it basically kills the possibility to do separate masterings for the two quality levels. And the normal customer with standard playback equipment loses some dynamic range, so the CD-format quality will suffer (again).

Personally, I would use such a data channel for supporting additional headroom instead of higher frequencies. I have no clue whether that can be pulled of in a satisfactory way, but if it can, it would tackle a much more important problem than including high frequency content that a few deluded people believe they can hear.

You are not bound by Nyquist here in its strict sense, because the data must first be decoded and processed before it can be used to modify the sound output.

As it relates to Porcus's suggestion that you can somehow improve a non-mqa-encoded digitization that is plagued by aliasing, you are bound by Nyquist of the original sample rate.

Even if a down-sampled version is mqa-encoded, you are still bound by Nyquist of the sample rate of the source that was encoded.

Personally, I would use such a data channel for supporting additional headroom instead of higher frequencies.

Something like HDCD, right?  Well, I hope we don't need to have another discussion over the woeful* inadequacy of 16 bits to deliver legitimate content.  I'll also point back to Kohlrabi's last post (https://www.hydrogenaud.io/forums/index.php?s=&showtopic=107666&view=findpost&p=907062).

(*) I'm being sarcastic.

Why on earth do people talk about lossless compression then? It's not.

And why not just take noise-shaped 16 bit, 192 kHz with some real lossless codec to get the same "temporal resolution" and possibly higher dynamic range? That is actually lossless, doesn't cost license fees, doesn't require special hardware and has about the bitrate of CD audio.
For lower bitrates streaming you could add lossyWAV processing.

Even if you knew the exact ADC used and the response of its anti-aliasing filter, you simply cannot reliably recreate frequency content above Nyquist.

Do they claim only to work above Nyquist?

As an example, suppose you have content sampled at 48kHz and there is a tone at 22k.  How do you know this tone was originally 26k and not simply 22k all along?

I would suppose a good vinyl pop removal algorithm could by and large improve, despite not possessing the 100 percent sure knowledge of what is noise and what was intended. Wouldn't you?

Why on earth do people talk about lossless compression then?

Which people, the shysters of the audiophile world?

Quote from: greynol on 2015-09-14 17:59:10

Even if you knew the exact ADC used and the response of its anti-aliasing filter, you simply cannot reliably recreate frequency content above Nyquist.

Do they claim only to work above Nyquist?

I assume you mean Meridian or those who are speculating about mqa.  Either way, I don't know.  Do they?

I would suppose a good vinyl pop removal algorithm could by and large improve, despite not possessing the 100 percent sure knowledge of what is noise and what was intended. Wouldn't you?

All sorts of processing can be used to make subjective improvements, but I thought you were talking about removing aliasing or restoring aliased information, in which case you are only guessing about information that has been permanently lost.  If you try to apply some filter using what you know about the ADC, you may get an audible difference and that difference may result in a subjective improvement for one signal, but the same filter may also result in a subjective worsening for another signal.

As it relates to Porcus's suggestion that you can somehow improve a non-mqa-encoded digitization that is plagued by aliasing, you are bound by Nyquist of the original sample rate.

Even if a down-sampled version is mqa-encoded, you are still bound by Nyquist of the sample rate of the source that was encoded.

Well, sure. The MQA encoding requires compatible gear at both ends to pull off the trick. You can't magically after-guess spectral content that was lost in sampling. There are techniques which try that, but I consider them a fraud.

Something like HDCD, right?  Well, I hope we don't need to have another discussion over the woeful* inadequacy of 16 bits to deliver legitimate content.  I'll also point back to Kohlrabi's last post (https://www.hydrogenaud.io/forums/index.php?s=&showtopic=107666&view=findpost&p=907062).

(*) I'm being sarcastic.

HDCD doesn't seem to to do much on the headroom side; it improves the noise floor (but I admit that I don't remember much about it).

I share your sarcasm here, because there's an implicit and tacit admission in the MQA scheme: That 16-bit audio has enough dynamic range to sacrifice some of it for other purposes. I wonder how much of that has registered with the audiophiles who are praising MQA.

And why not just take noise-shaped 16 bit, 192 kHz with some real lossless codec to get the same "temporal resolution" and possibly higher dynamic range? That is actually lossless, doesn't cost license fees, doesn't require special hardware and has about the bitrate of CD audio.

That's not backward compatible with the CD. I think that's the major point here which makes their scheme attractive to content providers who would have to offer a CD-compatible version anyway.

Which people, the shysters of the audiophile world?

Like Lavorgna.

That's not backward compatible with the CD. I think that's the major point here which makes their scheme attractive to content providers who would have to offer a CD-compatible version anyway.

Ironic, isn't it?

HDCD doesn't seem to to do much on the headroom side; it improves the noise floor (but I admit that I don't remember much about it).

IIRC, peak extension provides an increase in dynamic range at the expense of a little noise.

That's not backward compatible with the CD. I think that's the major point here which makes their scheme attractive to content providers who would have to offer a CD-compatible version anyway.

The way I understood it the main problem is the doubling of bitrate with doubling of sampling rate. Streaming actually seems to be their main point, which requires decoding software anyway..
Also, 24/48 (which seems to be the preferred format) is not actually CD-compatible. Even with 44.1 kHz you have to do a conversion.

And lastly, you can always offer resampled CD versions. That is how it's done actually for quite some time now. If you additionally compressed that losslessly, you will get even lower bitrate.

Yes, but you might lose fidelity that can potentially be detected 56% of the time with cherry-picked hardware and also assuming you are using a "typical" digital filter.

No no, with 16/192 and proper lossless compression you wouldn't lose anything. Actually I think you'd achieve higher quality than MQA but without a proprietary format or license fees.


As for the argument "If the receiver cannot decode the algorithm, it will be just played as a CD-quality stream": in our day and age a receiver is a computer, smartphone or portable audio player. Most of them can decode something like FLAC.
And nobody, except some crazy audiophiles, stores music in an uncompressed PCM format nor is music streamed in such a format.

Ironic, isn't it?

I'd call it devious.

Lets for a moment assume the stance of a content provider who doesn't care about quality, but rather about his revenues. What would his interests be? I'd say:

• He wants copy control (he'll call it "rights management", because it sounds rather more harmless)
• He wants compatibility with the CD format, because that's still the most widespread; a sort of least common denominator that suits the largest number of people.
• He doesn't want several different formats, because that causes cost and problems.
• He realizes that he can't copy control the CD-format; that's been tried and it failed. Repeatedly.
• Hence he's prepared to surrender copy control of a limited-quality version of his product.
• As this is unavoidable, he's going to want the lowest quality the market tolerates for the CD-format "common denominator".
• The native CD format is much too good for that, see the success of the much inferior LP.
• That means that the CD has spare data capacity, which might be used intelligently.
• So why not limit the baseline CD to the performance of the LP, mainly by raising the noise floor some 20 dB (or more if it is spectrally shaped), and convey extra data in the noise?
• Now the question is how to intelligently use the extra data channel. Copy control is a given, but that requires only a small fraction of the capacity.
• The question is how to use the spare capacity to convey "extra quality" the consumer is likely to crave for hard enough to swallow the bait.

No no, with 16/192 and proper lossless compression you wouldn't lose anything. Actually I think you'd achieve higher quality than MQA but without a proprietary format or license fees.

Tongue in cheek.

I can claim improvements all day long if you aren't allowed to hold me down to something specific.

Anyway, I was answering the second portion of your post:

And lastly, you can always offer resampled CD versions. That is how it's done actually for quite some time now. If you additionally compressed that losslessly, you will get even lower bitrate.

Sorry that I didn't quote it.

• The question is how to use the spare capacity to convey "extra quality" the consumer is likely to crave for hard enough to swallow the bait.

[/li][/list]...for hf content no one will ever hear at the expense of noise which someone might very well be able to hear.

...for hf content no one will ever hear at the expense of noise which someone might very well be able to hear.

That's the problem. Is it going to be enough when they can be made to believe they hear it?

Quote from: Porcus on 2015-09-14 19:55:40

Quote from: greynol on 2015-09-14 17:59:10

Even if you knew the exact ADC used and the response of its anti-aliasing filter, you simply cannot reliably recreate frequency content above Nyquist.

Do they claim only to work above Nyquist?

I assume you mean Meridian or those who are speculating about mqa.  Either way, I don't know.  Do they?

I do not know either, I asked because you used the "above" word. Myself I just hadn't even imagined they would express a reservation that it only has effect above Nyquist.

Quote from: Porcus on 2015-09-14 19:55:40

I would suppose a good vinyl pop removal algorithm could by and large improve, despite not possessing the 100 percent sure knowledge of what is noise and what was intended. Wouldn't you?

All sorts of processing can be used to make subjective improvements, but I thought you were talking about removing aliasing or restoring aliased information, in which case you are only guessing about information that has been permanently lost.  If you try to apply some filter using what you know about the ADC, you may get an audible difference and that difference may result in a subjective improvement for one signal, but the same filter may also result in a subjective worsening for another signal.

1) ... which is fine (though not necessarily so by the standards which apply in a market segment which has decided that expensive cables constitute a good solution and a decent EQ will never do ... but I digress), if they are by and large improving or not changing anything, and an algorithm may always provide for letting some signals through unaltered.  I assume there will be many signals where there is a theoretical case for improvement (though not necessarily any gain in practice, given the frequency range where it will kick in) - for example, if software can identify fairly reliably a string quartet, then one has a certain knowledge of where one should expect to find overtones (and thus, given the digitizing algorithm, their aliases).

2) And I wasn't restricting to that (basically because I have not read through all the marketing claims in detail); if back in the day the signal was digitized using a filter that started rolling off below 22.05 (e.g. http://xiph.org/~xiphmont/demo/neil-young.html#toc_o (http://xiph.org/~xiphmont/demo/neil-young.html#toc_o) , but you know all that ...) then at least the frequency response part could easily be corrected (I didn't claim it would be audible ... and don't tell the audiophiles it is just an EQ).

if back in the day the signal was digitized using a filter that started rolling off below 22.05 (e.g. http://xiph.org/~xiphmont/demo/neil-young.html#toc_o (http://xiph.org/~xiphmont/demo/neil-young.html#toc_o) , but you know all that ...) then at least the frequency response part could easily be corrected (I didn't claim it would be audible ... and don't tell the audiophiles it is just an EQ).

Agreed.

there's an implicit and tacit admission in the MQA scheme: That 16-bit audio has enough dynamic range to sacrifice some of it for other purposes.

Do they really only deliver 16 bits including whatever extra information? Even if they use SPDIF, there is enough room for 24-bit audio in that 32-bit wordlength. And that is uncompressed - compressing they can effectively get more.

Do they really only deliver 16 bits including whatever extra information?

I think they're more flexible than that. The question is what their customers want, i.e. the content providers.

Anyway, I was answering the second portion of your post

Oh I see.
Well, I see no reason why someone shouldn't be able to use any filter they like to resample.

--

I don't think they only use 16 bits because the HF needs a couple of bits, so you would audibly degrade the freaking main audible content. It could only work with extremely loud and compressed music ... which kinda doesn't fit the purpose of the format.

So again, you cannot put this on a CD. I guess what they meant to say is that a limited receiver that can only operate at 44.1 or 48 kHz would probably accept 20 or 24 bit words but just throw away the lower bits. That's all the compatibility I see.

I guess what they meant to say is that a limited receiver that can only operate at 44.1 or 48 kHz would probably accept 20 or 24 bit words but just throw away the lower bits. That's all the compatibility I

need.

No, it's not needed. And it's actually inefficient.

Would you rather transfer 24 bits to any consumer which will potentially throw away 1/3rd of that transferred information, or let a few audiophiles opt-in to receive the extra bits or higher sampling rate which most likely have hardware that can handle it?

No, it's not needed. And it's actually inefficient.

Would you rather transfer 24 bits to any consumer which will potentially throw away 1/3rd of that transferred information, or let a few audiophiles opt-in to receive the extra bits or higher sampling rate which most likely have hardware that can handle it?

That is S/PDIF? Wordlength: 32 bits. Header/subchannel etc.: 8 bits. Of the remaining 24 bits, 1/3rd is thrown away (4 through 12 are always zero with 16-bit audio).

That is S/PDIF?

S/PDIF is a consumer digital audio interconnect. It has nothing to do with streaming audio over the Internet which costs bandwidth.

And show me an MQA ready DAC that uses S/PDIF and is limited to less than 192 kHz.
Because instead of being able to use your current DAC (192 kHz compatible ones have been produced for years) which may or may not use S/PDIF, you have to buy a new MQA DAC anyway! Otherwise you are just playing 16/44.1, throwing away a lot of information streamed over the Internet...

So it makes no sense.

Because instead of being able to use your current DAC (192 kHz compatible ones have been produced for years) which may or may not use S/PDIF, you have to buy a new MQA DAC anyway! Otherwise you are just playing 16/44.1, throwing away a lot of information streamed over the Internet...

Do you know that the streaming part of it will transfer the full signal if there is no MQA-aware DAC identified? (I don't.)

Source?

No but if that were the case then it would make even less sense.

If you do it the non-dumb way and stream only the format compatible with the user's hardware (like I've explained above), then why would you need to package "hi-res" with probably some loss into a proprietary format with low sampling rate that costs license fees?

"hi-res" with probably some loss

And do you know that?

Well yes, it isn't lossless data compression by definition.

Well yes, it isn't lossless data compression by definition.

I agree.  This is the key problem.  And what are the actual numbers for it's equivalent bitrate?  We know that Redbook Audio (44.1 kHz 16-bit linear stereo PCM) is 1411 kBps. 
So what kind of actual numbers does MQA actually have?  And I mean of AUDIO data, not meta-data?  They can't have it both ways.  All the bits are necessary from a hi-fidelity lossless
perspective.  You can't just suddenly re-appropriate the lower bits to some other purpose and still consider it lossless.  Either you are representing the waveform with those bits or you
aren't.  And if if they are represented by something else, how accurate is that? 

The format thus far brings more questions than answers.  And that's not transparency in a codec; it's convoluted or it's kind of a technical hoax. 
That's my educated guess anyways.  If it's perceptually encoded, it's lossy by definition.  They can't have it both ways.

They can't have it both ways.

They can have losslessness to 1411 and whatever beyond that.

If you look at the patent and just the first 16 bits for 96 kHz input the output has 13 bits LF + 3 bits lossy HF.

If you look at the patent and just the first 16 bits for 96 kHz input the output has 13 bits LF + 3 bits lossy HF.

I can't remember, but I bet it'll have noise shaping too, which at 96k can get the noise floor below the equivalent of 44.1/16-bit flat or even noise shaped, even with "just" 13-bits.

Cheers,
David.

Btw, they say in the patent: "We can conclude that a 16-bit 96 kHz channel with appropriate noise shaping is entirely adequate as a distribution format, meeting audiophile requirements with some margin to spare."

FLAC compressed that would be roughly 1.2 Mbps. A 16/96 stream "compressed/packed" into a 24/48 PCM would be 2.3 Mbps.

But how much would a MQA 24/48 PCM stream compress using FLAC (or another other lossless audio codec)?

EDIT: I mean, if a MQA "24/48" stream compressed to smaller than a real 16/96 stream, it would be of some value.

I have a few 24/48 FLACs and they are roughly 1.5 Mbps. This should get worse if the lower bits are randomized due to other compression.

If you look at the patent and just the first 16 bits for 96 kHz input the output has 13 bits LF + 3 bits lossy HF.

Oh, thanks!

Then recalculating to lossless Red Book equivalent to compare to my collection, they need either a pass-through (that kills the HF completely), or compress down to 1146-ish kb/s. Of my rock/metal collection encoded to FLAC -8, the latter leads to a loss in 1 out of 300 tracks, but when streaming variable bitrate with less than a full track of buffering, one should exceed 1146 more often than that.

I have a few 24/48 FLACs and they are roughly 1.5 Mbps. This should get worse if the lower bits are randomized due to other compression.

When are the lower bits of 24 bit material ever NOT random?

If you look at the patent and just the first 16 bits for 96 kHz input the output has 13 bits LF + 3 bits lossy HF.

Yes, and the patent tells all. IMO a considerably different story than the one that might be discerned from the high end audio press:

Google Patents Link (http://www.google.com/patents/WO2013186561A3?cl=en)

"
An encoder for digital audio signals at a higher sample rate creates a stream for consumer distribution at a lower sampling rate, with compatibility for standard PCM players without a decoder. In conjunction with a suitable decoder, two enhanced playback options are supported, the first option allowing full lossless reconstruction of a noise-shaped higher sampling rate signal, the second option allowing lossy bandwidth extension even if an intervening transmission chain has truncated the least-significant-bits of the encoder's output signal.
"

Link To MQA Encoding Graphic (https://www.hydrogenaud.io/forums/index.php?s=&showtopic=107570&view=findpost&p=907256)

When are the lower bits of 24 bit material ever NOT random?

There are varying degrees of randomness. The more lower bits you scramble, the higher the entropy, the lower the later compression ratio.

Quote from: xnor on 2015-09-17 01:58:55

I have a few 24/48 FLACs and they are roughly 1.5 Mbps. This should get worse if the lower bits are randomized due to other compression.

When are the lower bits of 24 bit material ever NOT random?

Depends on the program material. Also depends on which lower bits that we are talking about.

Take a 24 bit file with power supply or deterministic HVAC noise that is 96 dB or more down, for example.  If it is HVAC noise, it is usually less than 96 dB down.  Eithe way, not uncommon in my investigations.

If you look at the patent and just the first 16 bits for 96 kHz input the output has 13 bits LF + 3 bits lossy HF.

Thanks for replying with some science instead of just telling me "it's lossless" over and over again.  Seriously, thanks.  I will maybe take a look at the patent later too.  Good idea.

About 24-bit audio... not all audio is recorded from microphones or through consoles.  There's a heck of a lot of electronic music that is generated and recorded entirely inside of computer software.  Every bit of every byte of that is ostensibly audio.  There are no "free" bits at the LSB end if the end result is rendered to a high-resolution audio file.  The audio is really pristine for some genres such as ambient. 

Personally, i wouldn't want just 13 bits for the bulk (lower end) of my audio signals.  That could start sounding like bitcrusher effects, which I just don't like the sound of. 
They need to be upfront about this format... MQA is LOSSY.  If they can't be honest and upfront about that, what other aspects of their business are they hiding?

Personally, i wouldn't want just 13 bits for the bulk (lower end) of my audio signals.  That could start sounding like bitcrusher effects, which I just don't like the sound of.

13-bits, dithered and noise shaped, does not sound like intentionally lo-fi audio. The noise floor goes up a bit.

I wouldn't assume that the product is the same as the patent. I wouldn't assume that most/much MQA will be distributed as 44.1/16 anyway, hence the worry about 13-bits may be misplaced.

Cheers,
David.

In the patent they seem to compress 96/24 into 48/24, but their real product is supposed to compress 192/24 to the same output (?) ... so I don't see how it would be better. If anything, it is more "lossy".

And raising the noise floor is lossy by definition. Else lossyWAV could be renamed to losslessWAV.

Well, it looks like MQA is not going to wither away, at least what I saw in CES. The push from Meridian is quite incredible. They really want it to be succeeded, even if that means literally burning cash.

The problem is they expect to get those cash back by encouraging/forcing people to buy new DACs and stuffs.

The problem is they expect to get those cash back by encouraging/forcing people to buy new DACs and stuffs.

It is among the obvious benefits of the licensing stranglehold business.

I don't see the CD dying any time soon.  So, no new DACs needed here.

The placebophiles will be all over this new format.  But, I think MQA faces an uphill battle.  FLAC is entrenched as the audiophile codec of choice right now.  Hard drive space is cheap and high speed internet is everywhere.  I feel like thsi solves a problem that existed last decade.

Yes, you can cram more music on a device if it's lossy compressed, but if there's one thing placebophiles have proven, it's that they're willing to spend money.  Most of them won't hesitate to buy a FiiO X5 and cram 2 256 GB MicroCD cards in there, and then buy a second X5 for all the music the first player couldn't hold.

And you also face the placebophile belief that lossless is always superior to lossy.  They'll probably say that it's better than MP3, but not as good as FLAC.

Did anyone dig deeper into the MQA fog? Some things seem to become clear when reading around.
Very likely it is only about selling the latest remaster again in a way you need certified MQA hardware to decode it.
New A/D conversions will be sold as being superior only because of the patended technology.
So far nobody has decoded digital data to test anything with value.
No way you can handle the digital decoded data for anything because you won't get it over digital out or a software that makes it usable PCM data.
No room correction, no own digital speakers and much other things. Leave alone own editing, EQ, cutting, mixing...

My possibilities are very limited with analysis so i may be pretty off.
2L offers some MQA encoded testfiles that contain the compatible playable content.
There is a piano sample in MQA "2L-120_01_stereo-44k-24b.mqa.wav" and non MQA "2L-120_stereo-44k-16b_01.wav"
Since it is piano there is not much content above 10kHz and a delta file of both creates a dip in the noisefloor at ~12kHz.
Can this be a phase shift kicking in at 12kHz because of some apodizing going from linear to non linear?

I see noise shaped dither as plus while others already claim degration because they can 'see' the noise raising above 12kHz as degration already. Must be the species without ATH :)
At the end of one sample "2L-048_14_stereo_96kHz_FLAC.mqa.flac" i find ~100ms dithered silence that i try to visualize.
Wavosaur isn't the most scientific tool i guess.
May this be the minimum noisefloor of the MQA process? How to quantify the shape to real bit-depth?
I have a pic showing the shape of these 100ms above against iZotope 16bit dither but don't know what setting exactly.
(http://www.directupload.net/file/d/4238/yyfkrjzm_png.htm)

There is a bit of file analysis here

http://www.computeraudiophile.com/blogs/miska/some-analysis-and-comparison-mqa-encoded-flac-vs-normal-optimized-hires-flac-674/ (http://www.computeraudiophile.com/blogs/miska/some-analysis-and-comparison-mqa-encoded-flac-vs-normal-optimized-hires-flac-674/)


and some critique here, amidst the cries of the fan boys. You'd better work backwards from the thread.

http://www.computeraudiophile.com/f8-general-forum/mqa-ces-27127/index28.html (http://www.computeraudiophile.com/f8-general-forum/mqa-ces-27127/index28.html)

Many of those 2L recordings are so good that you could convert them to 128kbps mp3 and they'd still sound better than 99% of the recordings in the world. If people are just listening cold to those recordings (MQA or not!), they'll think they sound great, because they do. Without a comparison, it's not a helpful demo.

I think we need to be very careful to have a technically accurate discussion here on HA. It's great that some of the details are being analysed. Remember that marketing hype will always be that. Cry BS on the BS, but don't claim it's all BS. Truth is, there was an (endlessly debated) test that showed avoiding "CD filtering" could produce an ABX-able improvement (caveats: filtering that you didn't really have to use even for CDs; improvement was very very small), and MQA is supposed to bring that improvement without bloating the file size, with a check of bit-perfect delivery of what was sent.

That's all fine, in and of itself. Some of the other problems will probably be worked around too. Doubtless when this technology is licensed to receiver manufacturers, they will have access to the digital data internally to do room correction, EQ etc. If it gets really popular, there's bound to be a not-very-legal reverse-engineered open source decoder appear.

The two problems I see are 1) the marketing is a bit confused, and 2) it's lossy. For downloading today (and streaming in a few years), why wouldn't audiophiles just pick the 192kHz version? I guess a family of formats (44.1, 96k, 192k) would make sense, where the 192k is authenticated but lossless, the 96kHz is lossless to 24kHz and authenticated, etc, and they all use the specified filtering. That would make sense. That could be their plan.

I quite like the idea of authenticated. I'm not claiming that any of the not-bit-perfect issues I've seen over the years have been audible, but it's annoying to see people's incompetence in inadvertently changing digital audio signals, and this would put an end to that. Until someone takes the output of an MQA decoder, runs it through a $20 radioshack analogue equaliser, and feeds it into an MQA encoder. The result is still "authenticated" but is a long way from the original ;-) Nothing to stop people selling brick-walled masters in MQA either.

Cheers,
David.

The argument I'm seeing repeatedly is that MQA sounds *better* than even the most outlandishly encoded PCM. For instance, over at AudioStream the resident clown declares:

"I'll just add that the MQA encoded file sounded a lot better than the 24/352.8 original, which sounded just lovely to begin with."

This seems to be the consensus among the placebophile cognoscenti, its insanity notwithstanding...

There is a bit of file analysis here
http://www.computeraudiophile.com/blogs/miska/some-analysis-and-comparison-mqa-encoded-flac-vs-normal-optimized-hires-flac-674/ (http://www.computeraudiophile.com/blogs/miska/some-analysis-and-comparison-mqa-encoded-flac-vs-normal-optimized-hires-flac-674/)

I did read around there already and in miskas analysis "MQA data appearing as high frequency noise" is simply the dithershape visible in spectral view i picture above.
I wrote elsewhere already for the unencrypted part:
"Now take well shaped dither plus proper downsampling and the claim better as standard cd is not very far fetched. You seldom see noise shaped dither to maximize the dynamicrange on CDs."

There's an old AES paper (from Peter Craven, I think) which talks about using noise-shaped dither to reduce the real signal on a CD to about 12-bits and then hide a lossy encoded 5.1 version of the same content in the (now unused) least significant bits (also noise shaped).

It sounds like the idea, in a different form, is being used here.

Cheers,
David.
P.S. Apologies if I have misremembered this - the paper is in a box somewhere in my loft, and I'm not about to go and find it!

When i watch the realtime FFT while playing it really seems it has a low resolution and the noisefloor is pretty high.
My first feeling was it can be maximum 14bit noise shaped but these 100ms silence i found above seem to be 16bit.
That was the point why i asked if someone with more knowledge can analyse it.

The nice people at MQA have seen fit to put a nice, compact explanation of their silliness together. Apologies if this has already been linked:

http://www.audiostream.com/images/MQA_AES_HighResolutionAudio.pdf

. For downloading today (and streaming in a few years), why wouldn't audiophiles just pick the 192kHz version?

I suppose it could be worse (i.e., even sillier than 192kHz as the 'standard').  The woo merchants could have gone the route of touting even higher sample rate PCM.

Nothing to stop people selling brick-walled masters in MQA either.

Sigh.  So many 'solutions' to the wrong problem.

So many 'solutions' to the wrong problem.

Not if you consider that it is an attempt to solve a supply-side problem rather than a demand-side problem.

The argument I'm seeing repeatedly is that MQA sounds *better* than even the most outlandishly encoded PCM. For instance, over at AudioStream the resident clown declares:

"I'll just add that the MQA encoded file sounded a lot better than the 24/352.8 original, which sounded just lovely to begin with."

This seems to be the consensus among the placebophile cognoscenti, its insanity notwithstanding...

So, the author is claiming that a file that LOSSY compressed with MQA sounds better than the lossless original it came from.  If this is more than just placebophile BS (meaning you can actually ABX the difference),, then there has got to be some kind of EQ or other adjustment going on there by the decoder.

If this is more than just placebophile BS (meaning you can actually ABX the difference),, then there has got to be some kind of EQ or other adjustment going on there by the decoder.

1) ABX is a test for difference, not preference.

2) It is entirely possible that a listener may have a preference for artifacts (vinyl, anyone???).  I recall a discussion about this as it relates to 128kbit mp3, though I can't be bothered to look for it.

So, the author is claiming that a file that LOSSY compressed with MQA sounds better than the lossless original it came from.  If this is more than just placebophile BS (meaning you can actually ABX the difference),, then there has got to be some kind of EQ or other adjustment going on there by the decoder.

It's always entertaining to watch these people maneuvering themselves into these dilemmas.

Either there is no audible difference and they have proven for the millionth time that they are not trustworthy and listen with their eye$,
or MQA is not transparent either due to being lossy (quantization distortion) or due to additional processing causing distortion.

I think some commenters are being far too sensible and rational. MQA is being marketed, and has been seized upon by audiophiles, as not just a better way to distribute compressed music, nor even a better form of digital audio.

MQA is the 'third way': a completely new way of recording and playing back music. There is no conflict in the idea of MQA-encoded audio sounding better than the original PCM file from which it was taken because the MQA process "de-blurs" it. And it doesn't end there, because in future, the recordings can made with MQA from end to end - there are official MQA ADCs in the pipeline.

MQA should not be thought of as "digital" at all. All those vinyl-o-philes who condemn all digital audio for its "silver sheen" and "lack of musicality" can now confidently take their first steps in a brave new world. etc. etc.

MQA is the 'third way': a completely new way of recording and playing back music.

Ah, right. The penny has dropped. MQA is being positioned in the audiophile's mind to take up where DSD leaves off.
And to think most of them have only just bought their DSD-capable DACs.
They must be wetting themselves at the thought of being pickpocketed again so soon.

Quote from: krabapple on 2016-01-19 19:01:52

So many 'solutions' to the wrong problem.

Not if you consider that it is an attempt to solve a supply-side problem rather than a demand-side problem.

From the demand side (me) it looks like the supply side (mastering) is the problem.

...and I'm suggesting you shift your paradigm. ;)

People buy and listen to and "enjoy" crap. It's as simple as that .. in a capitalist world. ;)
But, in defense of consumers, see the loudness war. Producers are just as stupid as consumers, I guess.

Quote from: apastuszak on 2016-01-19 20:19:07

If this is more than just placebophile BS (meaning you can actually ABX the difference),, then there has got to be some kind of EQ or other adjustment going on there by the decoder.

1) ABX is a test for difference, not preference.

2) It is entirely possible that a listener may have a preference for artifacts (vinyl, anyone???).  I recall a discussion about this as it relates to 128kbit mp3, though I can't be bothered to look for it.

True.  But if they're claiming that MQA sound better than FLAC, then we have some DSP going on also.  I kinda object to that idea.  And it's gonna put more work on the process.  Now they have to see how it sounds for PCM release (FLAC, MP3/ACC, CD) and how it's gonna sound when MQA DSP is applied.

DSP going on

That has always been Meridian's schtick.  I take it you've read about their magical apodizing filters?

Quote from: apastuszak on 2016-01-20 21:30:26

DSP going on

That has always been Meridian's schtick.  I take it you've read about their magical apodizing filters?

I have not.  Meridian Audio is new to me.  Off to do some Googling.

We don't know if there is any extra DSP going on. We already know that MQA cannot be lossless despite their claims, but doing some hidden DSP like EQ or dynamics processing really would take the biscuit.

DSP like EQ or dynamics processing

Why is this even being entertained?

Why is this even being entertained?

Because of the claim that it sounds different than the source?
Here:

True.  But if they're claiming that MQA sound better than FLAC, then we have some DSP going on also.

Quote from: greynol on 2016-01-25 16:44:55

Why is this even being entertained?

Because of the claim that it sounds different than the source?
Here:

Quote from: apastuszak on 2016-01-20 21:30:26

True.  But if they're claiming that MQA sound better than FLAC, then we have some DSP going on also.

Not just different.  They're claiming it's better than the source.

...but if the source used a brick wall (sinc) filter, and if the process removes the ringing from this filter without adding any more, then it's not factually incorrect to claim an improvement.

With 192kHz sampling, it's several miles beyond the limit of credibility to claim this is an audible improvement, unless someone can prove different. But in signal processing terms, if you define ringing as a bad thing to have, then by that definition this is an improvement. (Reality check: there's usually little or no original signal content up there to excite the ringing.)

I'm still getting over the idea that ringing at 22kHz has reportedly been ABXed. If anyone is going to repeat the trick at 96kHz (i.e. the cut off frequency for 192kHz sampling) then I shall probably need therapy :)

EDIT: I take it as read that all claims of "night and day" differences can be dismissed out of hand. I automatically ignore these. Even in the world of true "professionals", 90%+ of it is just imagined. I am open to the possibility that some people have concentrated on certain specific issues so long that they really do hear them as night and day differences, while the rest of us can barely hear them at all. You can see that some of us do that right here on HA with non-transparent (but really rather good) audio coding: some of us ABX it readily enough, but grade it as 4.9; others hear the same thing and give it a 3.0 because it's their life. If some people really can hear different anti-alias filters, it's got to be in this category, but 1000x more subtle.

Cheers,
David.

Someone's been uncovering a bit more info:
http://archimago.blogspot.com.au/2016/01/measurements-mqa-observations-and-big.html

They're claiming it's better than the source.

...and that's supposed to justify the following nonsense:

if they're claiming that MQA sound better than FLAC, then we have some DSP going on also.

??

Please understand that I will expect you to provide some substantiation should you tell me the above quote isn't anything more than an attempt at an educated guess.

Shy of substantiation, my response to this...

733 date=1453234747]If this is more than just placebophile BS (meaning you can actually ABX the difference),, then there has got to be some kind of EQ or other adjustment going on there by the decoder.

For the sake of argument, let's say a difference is ABX-able, that certainly does not mean there is some kind of manipulation outside of the technology as it is currently understood by *knowledgeable* third parties.

We already know that MQA cannot be lossless despite their claims

I think that Meridian want MQA to be considered as an integral part of the mastering process.

So after mastering with MQA  ‘encapsulation’ (avoiding the term compression and the inherent question of loss), the distribution and playback process is lossless.

Of course, you could say the same about any other lossy codec employed at the end of the mastering chain.

@2Bdecided

Is there really even vaguely compelling evidence for the audibility of ringing with 44.1ksps sampling? I mean, call me stubborn but it seems like the Stuart paper (and I think there's one other recent AES candidate?) isn't really much to support what would be an astonishing capacity for detecting heavily masked, low level ultrasonic content. I mean, I can cite more vigorously controlled papers that 'prove' ESP...

@bandpass well yeah, but no
MQA is as much 'encapsulation' instead of compression as is lossyWAV ... it's compression, and lossy.
24/192 PCM input => MQA => output does not equal input. That's lossy.

@2Bdecided
Is there really even vaguely compelling evidence for the audibility of ringing with 44.1ksps sampling?

Not if done right according to Meridians owners manual.
But because it can be done wrong, we now have an elixir, MQA. Now everything needs to be "cured".
yay

cheers,

AJ

@2Bdecided

Is there really even vaguely compelling evidence for the audibility of ringing with 44.1ksps sampling? I mean, call me stubborn but it seems like the Stuart paper (and I think there's one other recent AES candidate?) isn't really much to support what would be an astonishing capacity for detecting heavily masked, low level ultrasonic content.

For me, it needs to be both independently repeated and understood. Saying "this is why it's detectable" is quite different from proving that is why it's detectable (if it is).

Remember the results were only 5% better than chance - i.e. one time out of 20 someone did better than guessing when trying to detect the ringing. There were enough trials to make this statistically significant, but it's a tiny effect.

I can't hear it, and I've tried. IIRC there were a couple of positive ABX results in a thread looking at the audibility of 20kHz ringing that I started years ago. Even in my own extreme samples, I could hear nothing wrong. https://hydrogenaud.io/index.php/topic,68524.0.html (the last positive ABX result is facecious ;) )

Cheers,
David.

I still hesitate to accept they heard the filter. They may have heard simply a piece of the used hardware acting different without HF content. My bet is on the tweeter.

I remember a test Archimago created 176.4kHz music files with very strong ringing settings for linear and mimimum phase from 44.1kHz sources.
http://archimago.blogspot.ca/2015/04/internet-blind-test-linear-vs-minimum.html
A CA member that has for sure a minimum of 100 posts about the working and finetuning of upsampling filters only tried to find excuses why he can't hear it, no usable result.
The fear of ringing is really strong and works best sighted. For MQA a lightbulb indicator alone surely will do wonders.
In several forums there also seem to be a growing interest in doing custom filters for DACs.
Many claim all kind of things with these also.
So MQA is good at satisfying a trend.

I wonder if really something new could have been a 16bit/64kHz srongly noise-shaped format that in the end is smaller compressed as MQA.

I wonder if really something new could have been a 16bit/64kHz srongly noise-shaped format that in the end is smaller compressed as MQA.

I agree. It might not even have needed to be 16/64, 14/64 maybe with a very modest increase in file size over 16/44

@2Bdecided

Thanks for the links. The test you conducted was particularly interesting: I'd expect a number of people to be able to hear it (ringing a mere 10dB down at a relatively low frequency). I'd put some of that down to people actually hearing the ringing, and some of it down to the surprisingly poor distortion performance exhibited by a lot of integrated audio solutions at high frequencies. Hell, I was looking at TomsHardware, and almost all the audio solutions they tested - and these were on expensive boards - had high frequency IMD (RMA, not sure of the exact test protocol) at about 45-50dB down.

And of course we have Meridian's...study. Their literature (looking at their other papers too) is a mess - their citations are frequently FUD. They make a lot of perfectly reasonable claims about the ear's temporal resolution (albeit poorly cited - I particularly enjoyed when they cited, in one paper, the entire Oxford Handbook of Psychoacoustics with no fucking page references), and then stitch that to their "temporal smearing" bullshit via such luminaries as Kunchur, of "I don't know how sampling theorem works" fame. Taken as a whole, their arguments are somewhere between incompetent and wilfully misleading.

I suspect I'm preaching to the choir here. I feel like one shouldn't be too open minded when it comes to stuff like this. The ability to make a vaguely plausible argument for the super-marginal audibility of ringing at 44ksps completely disappears once you push the sample rate just a little higher...and seeing as an aggressively noise-shaped 16/88.2ksps file is about the same size as their stupid proprietary format, even when one is scrupulously open-minded I can't help but see this as really, really crappy.

Once you get past the argument that conceivably justifies slightly higher sampling rates so we can shift the filters well out of the audible range (which isn't one they're actually making), to justify the rest of it (MQA: better than even 32/350ksps+ PCM!) they rely on an outrageous collection of innuendos, insinuations and half-truths.

One question to the experts when we are at it. This apodizing seems to work linear phase up to a frequency and minimum phase for higher frequencies up from there.
Lets assume the linear part ends at 12kHz. Isn't the pre-ringing of that part of the filter not at 12kHz? So you may have less ringing but much nearer or inside the audible band?
I remember a spectral pic of iZotopes intermediate phase filter with an impulse exactly showing this.

Mixing linear and min phase is not different from just filtering an initially linear phase filtered file again with a min phase filter or vice versa.
The linear phase filter will just ring at its frequency. If the min phase filter used has some attenuation at that frequency then the "previous" ringing will also be attenuated accordingly. Of course the min phase filter will also ring.

Thanks! So using changing phase behaviours in one process can lead to other problems but may produce nicer pics in some areas you want to look better on funny graphs.

There's always a tradeoff. With an added min phase filter it's phase shift and/or roll-off.

Got one, thanks! Above iZotope pre-ringing 50 that must be linear to ~15kHz and below SoX b92 -a
(http://www.directupload.net/file/d/4249/fa9ez2tl_png.htm)

Edit: Somehow the embedding does not always work. Direct link:
http://www.directupload.net/file/d/4249/fa9ez2tl_png.htm

There's always a tradeoff. With an added min phase filter it's phase shift and/or roll-off.

But the official story is that phase shift is inaudible, even at loudspeaker crossover frequencies. Which is why no one bothers about it. Verging on the ultrasonic, surely it must be of academic interest only..?

But the official story is that phase shift is inaudible, even at loudspeaker crossover frequencies. Which is why no one bothers about it.

It is best to read more (http://www.aes.org/e-lib/browse.cfm?elib=3824), post less (nonsense).

1) Even quite small midrange phase nonlinearities can be audible on suitably chosen signals.

I believe JJ once mentioned "percussive" sounds, maybe acoustic guitar, but I'm too lazy to search and that's your job anyway.

And now back to the miracles of MQA...


cheers

AJ

Charming, as ever.

Possibly cut-and-pasting from the same web site you looked at:

Dr Floyd Toole:

" It turns out that, within very generous tolerances, humans are insensitive to phase shifts. Under carefully contrived circumstances, special signals auditioned in anechoic conditions, or through headphones, people have heard slight differences. However, even these limited results have failed to provide clear evidence of a 'preference' for a lack of phase shift. When auditioned in real rooms, these differences disappear.. ."

I am intrigued that one set of industry gurus say that microsecond-level timing differences are highly audible (MQA), and others say that we can heavily distort the signal with phase shifts (in non-corrected speakers for example) without it being audible.

Possibly cut-and-pasting from the same web site you looked at

My quotation is directly from the AES site/paper I linked. Please spend more time there.
I know Dr Tooles opinions and summaries quite well thank you. As well as how often they are misunderstood and misquoted.

Dr Floyd Toole:

" It turns out that, within very generous tolerances, humans are insensitive to phase shifts. Under carefully contrived circumstances, special signals auditioned in anechoic conditions, or through headphones, people have heard slight differences. However, even these limited results have failed to provide clear evidence of a 'preference' for a lack of phase shift. When auditioned in real rooms, these differences disappear.. ."

Quote from: Green Marker on 2016-01-30 23:56:01

But the official story is that phase shift is inaudible

One of these things is not like the other, which one is it, can you tell?
(Hint, you've misunderstood the last part).

Sigh.

Sigh.

What are you complaining about?

If you're not familiar, he's spent a lot of time worrying about phase shifts, but very little (if any!) time trying to determine whether they're audible.  Since our rules won't allow him to espouse the immense virtues of his work, he's left doing what every other person who resents TOS8 does: try to pit one expert or popular belief against another.

With music you can usually get away with large phase shifts.... but we are not even talking about that, but relatively small and smooth phase shifts with music.
Since aj has to know better he decided to throw in the red herring of special synthetic test signals that when distorted can sound different.

But anyway, if someone is worried about pre-ringing at >21 kHz then simply resample (can be done during playback) by x2 with a min phase filter.

If anyone missed that one:
The Linear vs. Minimum Phase Upsampling Filters Test (http://archimago.blogspot.ca/2015/07/the-linear-vs-minimum-phase-upsampling.html)
Even when we have very steep settings in that test there seems to be something to it even when Archimagos test not meets anyones academic expectations.

If anyone missed that one:
The Linear vs. Minimum Phase Upsampling Filters Test (http://archimago.blogspot.ca/2015/07/the-linear-vs-minimum-phase-upsampling.html)
Even when we have very steep settings in that test there seems to be something to it even when Archimagos test not meets anyones academic expectations.

As Archimago notes,  ' the calculated p-values are not impressive ".  Only one comparison, for one musical sample  showed a significant difference.

Whatever the something is, it's not much.

I am intrigued that one set of industry gurus say that microsecond-level timing differences are highly audible (MQA), and others say that we can heavily distort the signal with phase shifts (in non-corrected speakers for example) without it being audible.

The problem is that most proponents of the audibility of such small timing differences are too vague in their prose. I have no doubts that some are deliberately vague, while others don't see through it. It is quite a subtle matter.

Hardly anybody denies the significance of relative timing between the channels in a stereo or surround signal. That's where the figure of around 6..10 µs apparently originates. However, there's no problem with meeting this requirement with CD-type signals, or with signals of even lower resolution (even 8kHz telephone-type encoding would suffice for that). It may not be obvious for people who have difficulties with understanding the finer points of sampling, but it is demonstrably so.

But that's not the same as temporal resolution in a single signal channel. You would have to define precisely what you actually mean with this term. You may well find that what you call "such small timing differences" mean something rather different between the different authors you refer to. In Toole's case it is almost certainly the single-channel case, where the phase shift is between different frequency components in a single signal, as can be caused by filtering, or by the construction of a multi-way loudspeaker, just to state a few examples.

In the case of MQA, I have tried to figure out exactly what they mean with temporal resolution, and they are being too vague to tell. I am convinced that this is deliberate. In their 2014 AES paper (Hierarchical Archiving and Distribution), they have a large number of literature references, but fail to clear up the fog. I quote:

"By exploiting population coding, temporal resolution can approach 8 μs and this precision reflects neural processing, rather than being strictly proportional to our 18 kHz bandwidth (an estimate of the upper ‘bin’ of the cochlea and upper limit of pitch perception) [56][58][4][5][6][7][70][71]."

Some of the references given there clearly relate to interaural time differences, which is the multichannel case that doesn't need higher sampling resolution. Some of the references are papers by Dr. Kunchur who doesn't seem to understand this point, and produces erroneous conclusions based on his misunderstanding, as has been discussed elsewhere in this forum. I don't see a paper there that specifically relates to the single channel case, but I haven't been able to get hold of all of them (has anyone here read those papers?).

Later, in the same paper, they say (after discussing inter-aural timing, i.e. the multi-channel case):

"Other mechanisms have been investigated that hint to similar discrimination limits within a channel, i.e. monaurally, including: temporal fine structure in pitch perception, the comprehension of speech against a fluctuating background [91] and other cues [34]."

I haven't got access to the two references given there, but I note that the second poses the thesis as a question, and in any case Stuart et.al. use prose that doesn't suggest that the investigations have produced an affirmative result. So we should be careful not to assume that the papers give sufficient evidence. You would certainly wish that something like MQA would be based on something more solid than some "hints".

So the quoted experts may not actually be in any disagreement, because they're not talking about the exact same thing. The apparent discrepancy is the result of a mixup of different concepts.

As far as my comprehension goes, the references given by Stuart et. al. in defense of MQA do not (yet) support the conclusion, that higher sampling rates than 48 kHz are needed. A 48 kHz sampling rate provides enough temporal resolution for the multi-channel case, where phase differences between channels can be represented that are much smaller than the required 6..10 µs. It also provides enough phase fidelity among the different frequencies in a single signal channel, even when taking reconstruction filters and antialiasing filters into account (temporal resolution is not a useful term here, since such phase shifts can be continuous even in a digital signal).

There's a real danger that by perpetually citing each other's flawed or misunderstood articles, the proponents of high-res create the appearance of something that's well researched and underpinned with evidence, when in reality it is not.

Quote from: Green Marker on 2016-01-31 15:03:18

I am intrigued that one set of industry gurus say that microsecond-level timing differences are highly audible (MQA), and others say that we can heavily distort the signal with phase shifts (in non-corrected speakers for example) without it being audible.

The problem is that most proponents of the audibility of such small timing differences are too vague in their prose. I have no doubts that some are deliberately vague, while others don't see through it. It is quite a subtle matter.

He's deeply (or willfully) confused about the issues of multi-driver loudspeaker phase audibility, which as Toole notes, is very difficult detect (though not impossible) in typical usage, despite often having XOs in the area where our hearing sensitivity is highest...vs filtering at or beyond hearing limits.
In desperation he sees "phase" and is getting his hopes up because he doesn't understand the issues.

I find the whole idea of MQA retroactively "fixing" filtering because it is possible, though remote, that "audible" filtering was used somewhere in the recording chain, to be absurd.
The BS test apparently showed it's possible to concoct such filtering, not that it actually exists to any extent in actual recordings.
It's clearly aimed at audiophiles who want the 5th version of the same 2ch construct. They're always starving for more.

cheers,

AJ

In desperation he sees "phase" and is getting his hopes up because he doesn't understand the issues.

He would certainly not be alone with that. I'd say that a large part of consumer land doesn't understand it, and it seems that there is a significant number of professionals who have their problems with it, too.

If you read the paper by Stuart et.al. which I was quoting from above, you can't help thinking that they actually do understand it, and that they are quite conscious of the different contexts and meanings of the term. I wouldn't have expected otherwise from them. But that means that they must be obfuscating the matter deliberately. They tiptoe back and forth between notions of bandwidth and resolution which have no simple relation with each other and can not be translated in the way they implicitly suggest.

I find the whole idea of MQA retroactively "fixing" filtering because it is possible, though remote, that "audible" filtering was used somewhere in the recording chain, to be absurd.
The BS test apparently showed it's possible to concoct such filtering, not that it actually exists to any extent in actual recordings.

I thought the idea was to have MQA everywhere in the chain, so that there's no need to remove such artefacts. That doesn't make it any more sensible, however. Its necessity for transparency hasn't been demonstrated, neither by their use of scientific references, nor by their own listening tests. We're still (and that's after decades of propaganda) at the stage of "hints".

@Pemazo, et al.:
You've omitted what is possibly the most important part of AJ's post:

He's deeply (or willfully) confused about the issues of multi-driver loudspeaker phase audibility, which as Toole notes, is very difficult detect (though not impossible) in typical usage, despite often having XOs in the area where our hearing sensitivity is highest...vs filtering at or beyond hearing limits.

There's been a lot of wanking going on in this discussion; little of which is actually on-topic.

Maybe I should re-open the other BS topic and let you all have back at it.

...of course it still doesn't open the door for those who are butt-hurt because they can't talk about their still-off-topic pet interests.

No confusion on my part, but thanks for your concern, AJ.

I am just trying to cut through the woolliness of this whole timing argument. We have detailed discussions on the minutiae of the timing implications of sample rates, bit depths etc. probably playing the MQA people's 'game' for them, but then listen to the results on speakers which distort the phase and/or don't have time-aligned drivers. Presumably many of the people being suckered by MQA are doing the same thing. It would be nice to point out the absurdity of it without getting drawn into a cycle of competitive 'fact' regurgitation.

(why not save your blood pressure, AJ, and simply ignore my comments?)

I am just trying to cut through the woolliness..
..but then listen to the results on speakers which "distort" the phase and/or don't have "time-aligned" drivers.
No (self assessed) confusion on my part

Sigh

I am just trying to cut through the woolliness of this whole timing argument.

By complaining about nothing to people who do not care?   This is the least effective way you could possibly do that. 

It would be nice to point out the absurdity of it without getting drawn into a cycle of competitive 'fact' regurgitation.

If you don't like the existing evidence, I'd think it would be fairly straightforward to replicate any timing difference you like and produce your own data.  Not sure it would be interesting, but you could certainly do it. 

Archimago did some, pictures, captures, text and even abx :)
MEASUREMENTS / IMPRESSIONS: Meridian Explorer2 Analogue Output - 24/192 PCM vs. Decoded MQA (http://archimago.blogspot.ca/2016/02/measurements-impressions-meridian.html)

^From that link:

Note that even though I was aware that the MQA decoded file had a higher noise floor, I did not purposely take advantage of this by listening to a quiet portion of the file with volume pumped up to listen to the hiss. That would be cheating.

-Archimago

So there is an audible difference, the background hiss, however he assures us he didn't consciously listen for it. And what assurance do we have that he didn't subconsciously hear it and/or key on that hiss difference without being consciously aware that was doing so?

Just like we can't accept data from people who say, "Well, there was an audible difference in level between A and B, but I assure you, I'm special and can hear passed that", we also can't accept data from people who say, "Well, there was an audible difference in background hiss, but I assure you I only listened to song sections where it shouldn't matter, and I can't possibly be mistaken on that point because I'm special."

It is the first source i trust about any of these MQA results. When you read older posts of Archimago he often failed to abx things he could have easily cheated, especially with his nice equipment around.
Maybe ask him for short samples at his blog if you are curious?

Just to be clear, I never accussed anyone of consciously cheating.

We know that often when there is a small fraction of a dB difference in level, say around .25 to .5 dB, many if not most people are unaware that the sound difference they can successfully differentiate (ABX) is nothing more than this difference in level. They however often attribute it to "clarity, detail,. de-blurring" etc..  It is an uncanny illusion and just like the McGurk Effect being well educated on the illusion doesn't make one immune to it. Even the world's top expert on the McGurk Effect, Dr. Laurence Rosenblum, says he himself is still fully susceptible to the illusion even though he knows better than anyone how it works. [reference available upon request]

So it seems there is some ABX'able difference, maybe just the noise floor. And it's just different, not necessarily better. I'm sure that's pretty much what we all expected, should there be an actual audible difference.

His statement that he did not feel very confident in his choices and that he could only identify any difference by switching back and forth and presumably listening to the same short section over and over again, echoes my own experiences with ABXing MP3 close to my personal limit of transparency. Extremely subtle differences that are very hard to pin down or put into words, but still significant enough for a (near) 100% success rate. I would never be able to identify the tested tracks as FLAC or MP3 in ordinary listening, though. Not even very focused and concentrated listening.

Talk about diminishing returns. With the files nulling out to -74dB, the differences can only be very subtle. And the much-vaunted "de-blurring" can be applied to any PCM signal in the studio, so there's absolutely no reason why you couldn't do it to everything, without having to distribute it in MQA format.

^From that link:

Quote

Note that even though I was aware that the MQA decoded file had a higher noise floor, I did not purposely take advantage of this by listening to a quiet portion of the file with volume pumped up to listen to the hiss. That would be cheating.

-Archimago

So there is an audible difference, the background hiss, however he assures us he didn't consciously listen for it. And what assurance do we have that he didn't subconsciously hear it and/or key on that hiss difference without being consciously aware that was doing so?

Just like we can't accept data from people who say, "Well, there was an audible difference in level between A and B, but I assure you, I'm special and can hear passed that", we also can't accept data from people who say, "Well, there was an audible difference in background hiss, but I assure you I only listened to song sections where it shouldn't matter, and I can't possibly be mistaken on that point because I'm special."

This seems like carping on your part.

His ABX results indicate he heard a difference, and he went looking for what it might be .   In the end his most reasonable candidate is the 'de blurring' step.  He's on solid ground in presuming that a small difference in noise floor  (not overall level) would be masked during music played at normal levels.  Even then he doesn't rule out that the changing noise floor with frequency , audible in a difference file played at high level, might account for his ABX

 He's not a fan of MQA.  

Realizing that Meridian/MQA has provided essentially no technical details or objective results, if I am correct about what is going on as described above, I am personally not interested in MQA as a format I feel I would want. There’s no “magic” here and there are evident compromises when trying to be everything to everyone as MQA seems to be aiming for.

I apply the exact same standards to everyone, regardless of what camp they fall in, including myself. If there is an audible difference in level or background hiss between A and B, then I don't even bother to run a test on myself.

But you understand that 'can be audible' doesn't mean 'is always audible', right?

He measures a difference in noise at -74dB. This is not hiss as i understand it and worth an abx.

... So there is an audible difference, the background hiss, ...

He didn't say it was audible. He said that there was a measurable difference, but he had not tried to unrealistically boost the level to see if he could hear it.

I invited our forum member Archimago, I assume the same person, to join us and post his samples. I also attempted to do so at his blog but after registering through my Google account, it seemed to get stuck in an endless loop and didn't work. Could someone else please try?

I was curious what version of ABX he was using, meaning if it was current, so I opened a test of my own just to see what version I had. [I don't keep track of the version numbers; they are Greek to me.] One of the songs I happened to have ready to go on my playlist you'll see happens to be the same name as his song, since I wanted to see what kind of a song it was, so I downloaded a 1 minute sample. The other song, B,  is just dead air on Arny's setup. [Where is he by the way?]

Note my time entries. This is from just clicking "This is X", "Next trail", ten times, and I beat his score just by random chance!  What a weird fluke!

foo_abx 2.0.1 report
foobar2000 v1.3.9
2016-02-15 18:27:58

File A: Blagutten 1 minute.flac
SHA1: f60ec14800aaf64032491a6d13e3bf9df15a76c6
File B: digital_black_2496.wav
SHA1: 7965703e5d9eea3f3d60ab5399a1423727a4e992

Output:
DS : Primary Sound Driver
Crossfading: NO

18:27:58 : Test started.
18:28:17 : 01/01
18:28:18 : 01/02
18:28:19 : 02/03
18:28:21 : 03/04
18:28:22 : 04/05
18:28:24 : 05/06
18:28:25 : 06/07
18:28:27 : 07/08
18:28:29 : 08/09
18:28:30 : 09/10
18:28:30 : Test finished.

 ---------- 
Total: 9/10
Probability that you were guessing: 1.1%

 -- signature -- 
2fe5385a85cd5e851b057697d3e46ee534c31f32

Considering he only made one test and didn't even hit a p-value of .05 or better, it is odd how many seem convinced he heard a difference. Stranger things have happened. . .in fact one just did: my 9/10 score by random chance!

Are other people seeing my ABX data in multi colors? I just cut and paste the data. I have no idea where the colors came from.

Colored, yes. The way you argue foobar abx means nothing then, never when 9/10 are hit. So don't wonder if he won't come here to proof anything.

Archimago used foo_abx 2.0 and foobar 1.3, as can be seen from the screen capture in his post.

I've notified him in comments there that 8/10, at p=0.055, is still shy of 'significant' by standard criteria.

Is that hostility I am sensing, Wombat?
ABX is a wonderful test I love, but a single,  so far never replicated test, of one person, with only ten trails, with a p-value not quite reaching the  .05 level? Perhaps we feel differently about the significance of this.

I also may be reading what he meant by being 'aware' of the noise floor differently than others, in his blog:

I can hear the musical residue but shifted to a higher pitch when playing the "difference" file. If I pump up the volume, the noise tonally is higher with a rising noise level above 5kHz as shown in the FFT. Perhaps this accounts for the tonality change I heard when ABX'ing.

[emphasis mine]
So he himself admits the noise may be the culprit! His notion that we don't always hear noise as noise, but rather it may, for example, alter perceived tonality, jives with my own experience.

and

Note that even though I was aware that the MQA decoded file had a higher noise floor, I did not purposely take advantage of this by listening to a quiet portion of the file with volume pumped up to listen to the hiss.

So he didn't focus on the hiss purposely. Good. But what assurance do we have that he didn't accidentally do so on a subconscious level?
----

Have you all listened to the tune? It's a great tune for hearing differences in noise floor:
It's quiet and yet dynamic, so when we say "the noise is 74 dB down [from 0dBFS]" it doesn't mean the same thing as with a loud and compressed song. If for example this quiet song has its (only occasional) loudest peaks 13dB down and the average RMS level is 26 dB down, then this noise floor is much closer to the music level than is normal/typical of compressed pop music.

Did he consciously listen for the noise? No. Might he have subconsciously though? Hmm....

If he heard the difference in an ABX test well enough to pass the test, it doesn't matter one bit whether he heard it consciously or unconsciously. He still passed that one particular test, so it is very likely that there is a actual audible difference.

More testing is needed.

But he didn't pass the test under the very common criteria of needing to reach a p-val = .05 or better. Heck, I happened to do even better by fluke, without even listening to the music in my test yesterday of other material and just randomly guessing. Single tests, by just one person, with a small number of trials, which don't even hit p-val .05 don't mean much in my book, but sure, let's do more tests, and with more people.

Plus,  if we were to later determine, through more testing [which I too encourage] the only reason MQA is discernably different from non-MQA music is because of it's added high frequency [but <20kHz] noise, as I suspect, is that something anyone would really want to purchase? Not me.
---

So today he has added a pic. Although, as he explains, he has offset the levels by 20 dB to make the graph easier to read, the relative noise floors of the different formats remain intact when compared against each other. At frequencies which matter, under 20kHz, there is clearly a lot more high frequency noise with MQA both decoded and undecoded. Look at 15 kHz, for example, a frequency some of you younger than me might still be able to hear: MQA is not just a little bit noisier, it is a whopping 12 dB noisier! [Not that I'm claiming it is necessarily always an audible difference.]


Even when not listening to a dead silent passage where at elevated playback this sticks out like a sore thumb as a jump in hiss level or tonality, added high frequency noise can add more "air", "openness", and "sheen" to certain instruments in certain recordings, in fact I believe he mentioned the sound he heard he described as "more sparkly".

Meridian is marketing nothing more than added high frequency noise* if you ask me. The emperor has no clothes!

*in the audible, < 20 kHz frequency band

But he didn't pass the test under the very common criteria of needing to reach a p-val = .05 or better. Heck, I happened to do even better by fluke, without even listening to the music in my test yesterday of other material and just randomly guessing. Single tests, by just one person, with a small number of trials, which don't even hit p-val .05 don't mean much in my book, but sure, let's do more tests.

Plus,  if we were to later determine, through more testing [which I too encourage] the only reason MQA is discernably different from non-MQA music is because of it's added high frequency [but <20kHz] noise, as I suspect, is that something anyone would really want to purchase? Not me.
---

Not Archimago, either.

So today he has added a pic. Although, as he explains, he has offset the levels by 20 dB to make the graph easier to read, the relative noise floors of the different formats remain intact when compared against each other. At frequencies which matter, under 20kHz, there is clearly a lot more high frequency noise with MQA both decoded and undecoded. Look at 15 kHz, for example, a frequency some of you younger than me might still be able to hear. MQA is not just a little bit noisier, it is a whopping 12 dB noisier! [Not that I'm claiming it is necessarily always an audible difference.]

Even after that whopping increase.... it's still more than -72dB down in the audible band. 

Even when not listening to a dead silent passage where at elevated playback this sticks out like a sore thumb as hiss, added high frequency noise can add more "air", "openness", and "sheen" to certain instruments in certain recordings, in fact I believe he mentioned the sound he heard was "more sparkly".

Even when it's -72dB or more  down?  I rather expect such effect would be minor at best and difficult to ABX, at normal listening levels, ...which, not surprisingly, is what he reports. 

So I'm not sure what battle you think you are fighting here.

I guess Meridian shapes the noisefloor this way by purpose. It may be one of the 'new found psycho acoustic elements' they use. In short: shaped noise
If you can hear that as easily as hiss as you claim there is no way to bit reduce with noise-shaped dither 24bit to 16bit. For that you can hear the noise when amplifying silent parts but not when music plays.

Yes, maybe, krabapple. Remember the song he used is quiet and dynamic and the (only very occasional) peaks are far away [down] from 0dBFS from the get go (13dB), not to mention the typical, average RMS level which is even farther still, like around 26 dB down. Did he set his volume by listening to a loud track and then left it there and moved on to Blagutten without touching the volume knob? I bet not. He probably set the volume while listening to this quiet track itself [i.e. he cranked it up]. Also, he listens in an environment not all of us can easily replicate where hearing faint things is easier than for most of us:

very quiet room using a silent fanless computer setup

The reason we aren't usually troubled by noise in recordings during quiet passages, at such low levels, is not because it is too faint to detect, it is because it is almost always masked by typical environmental room noise [computer, HVAC, street noise, wind noise against the windows, etc.]. So his listening environment has some clear advantages for hearing faint noise.

In the right environment humans can hear sounds which are about 120 dB down from their pain threshold or 85 dB down from what this chart (https://homestudiorecording.files.wordpress.com/2011/10/soundspl.gif) calls "average home hi-fi level".

I remember back in the days of cassette decks that even Dolby's stronger type C NR, which could provide a noise level of say -72 dBA wasn't good enough for me to eliminate noise under all conditions. That's why switching on dbx instead made a difference (http://i50.tinypic.com/2repoci.gif), albeit with incompatibility issues for playback in some scenarios.

When I hastily recorded the Blagutten song yesterday from the web's sample [I did not download it because I am cheap] it was merely to assess what sort of a song it was. I didn't think I would be posting it or images of its dynamics so I did not attempt to record it at a level with fairly accurate absolute levels, in fact I didn't set anything I just hit the record button in Audacity (which I barely know how to use). So don't make any judgements about the absolute levels from this image. Still, I think this image says a lot about the dynamic nature of this song compared to typical compressed music these days, so I thought to post it for all to see:

The song is quiet, especially the opening 17 seconds, yet gets very dynamic too, with peaks which are markedly higher than the average level. There is also a moment or two within this first minute of the song where it fades down to almost complete silence, albeit only briefly.

You would hope that decoded MQA vs undecoded MQA would be a little more different. Maybe it is, in the time domain. Maybe it is for other content. Looking at that spectral view, taking it at face value, decoding seems a little pointless.

It would be interesting to send some impulses through a complete chain and see what comes out.

Don't TOS8 me. I'm not talking about what it sounds like. I'm talking about seeing what the algorithm is doing.

Cheers,
David.

The whole MQA paradigm makes no sense;  and I don't think it's really supposed to make sense. 
the more people are confused they less they can criticize it technically for all it's LOSSY flaws. 
It's pretty much just a ploy to create demand, infrastructure, support, and money for a product nobody needed. 

The same is true for anything greater than 48kHz, 18-bit* as a delivery format.  This is just another effort to convince the masses who will always "want the best" that they need to have "more."

That it can be ABX-ed is just short of proof that the entire thing is smoke and mirrors since at least some of the difference between the original source and the undecoded output can be captured at 16/44.1.  That some of the difference between the original source and the decoded output can also be captured at 16/44.1 really demonstrates the absurdity of the whole thing!

It shouldn't come as a surprise that the sample presented by mzil was obviously not subjected to the standard dynamic range mutilation, either.

(*) 44.1kHz, 16-bit if we're talking about something other than specialized signals presented to people with >20 kHz sensitivity and/or highly dynamic content played in the quietest of environments which is likely combined with gain-riding.

The whole MQA paradigm makes no sense;  and I don't think it's really supposed to make sense.

I think that's a bit unfair. Clearly it's supposed to make money, but it's also supposed to solve a (just about ABXable under pretty extreme conditions) problem with filtering at 44.1kHz, without having to transmit higher sample rate content.

If you look back at the AES papers from Bob Stuart, Peter Craven - and also some papers from Malcolm Hawksford and even Michael Gerzon - you can see some of these ideas coming for a very long time. Burying "correction" data within 16/44.1 audio isn't a new idea, and neither is the suggestion that the filters used with CD quality audio need some work.

I'm sure the technology is supposed to make sense*, and work. I'm surprised at that spectral plot.

* ignoring the level of audibility, and other non-proprietary ways of achieving a similar effect.

Cheers,
David.

I think that's a bit unfair. Clearly it's supposed to make money, but it's also supposed to solve a (just about ABXable under pretty extreme conditions) problem with filtering at 44.1kHz, without having to transmit higher sample rate content.

It is not even clear that the "problem" still exists at 48 kHz, which is what MQA uses.

If you look back at the AES papers from Bob Stuart, Peter Craven - and also some papers from Malcolm Hawksford and even Michael Gerzon - you can see some of these ideas coming for a very long time. Burying "correction" data within 16/44.1 audio isn't a new idea, and neither is the suggestion that the filters used with CD quality audio need some work.

It is all well to look into these filtering issues, I don't think anyone would have a problem with that. If we understand exactly what matters in the design of reconstruction filters, we can all benefit. If there are mistakes that can be made with potentially audible effects, we can learn how to avoid them.

I was and still am prepared to go along with this apodizing filtering thing, and was hoping for some convincing arguments and blind tests to confirm that there's really a valid point. I'm not yet convinced that there is, but I'm prepared to change my mind given sufficient evidence.

I am rather less impressed by the timing-related arguments, which are supposed to point towards the need for higher sampling rates. They are either hopelessly vague or in ignorance of basic properties of sampling. The scientific papers by Stuart et.al. which I've seen, tiptoe around the problem in a way that makes me believe that they know what they are talking about, but studiously shape their prose to make the reader suspect that there is something substantial, without giving any tangible evidence. I have to say that I have a rather dim view of that.

Clearly it's supposed to make money

Ok, we agree on one thing.

but it's also supposed to solve a (just about ABXable under pretty extreme conditions) problem with filtering at 44.1kHz, without having to transmit higher sample rate content.

Right. The BS paper apparently showed pathological 44.1k is detectable. Hurray.
Meridians own manual states non-pathological 44.1k is "transparent".
What's it gonna be mate?
How's MiracleQA going to fix the zillion CDs out there, imagined to have embedded innumerable filter "problems"?
Wait, remasters you say? Why not with per Meridian manual, "transparent" 44.1k filtering this time? Or are you suggesting their manual is wrong, no such thing as transparent 44.1k?

cheers,

AJ

Bob Stuart has answered the questions that were collected by the ComputerAudiophile forum:

http://www.computeraudiophile.com/content/694-comprehensive-q-mqa-s-bob-stuart/ (http://www.computeraudiophile.com/content/694-comprehensive-q-mqa-s-bob-stuart/)

Having skimmed through it, I couldn't avoid some blood pressure increase. Stuart really has converted bullshitting into an art form.

To pick just one example of his style of arguing, here's how he deals with Archimago's lengthy blog post (http://archimago.blogspot.de/2016/01/measurements-mqa-master-quality.html) from January (the "Editor" of Stuart's response seems to have missed that the URL has changed, see Q82):

• He cryptically suggests in footnote 13 that Archimago prevented him from posting corrections. Yet, there's a comment section following Archimago's article, and there's no reason given why he couldn't have used this means to post his objections. Archimago obviously looks at those comments, and he appends new information to his article in the form of updates. I regard Archimago's behavior as exemplary, and if Stuart has any issue with it, he ought to make clear what his accusations are, and provide some evidence.
• He paraphrases the assertions such that he can more easily dismiss them and make the asserter look silly. Try reading Archimago's blog post and find where he writes what Stuart has paraphrased.
• He criticises Archimago for assuming that the Nielsen example was from a 16-bit master (A82 part iii). This is a particularly telling example. The Nielsen piece was listed (http://www.2l.no/hires/) among the free examples for comparing MQA to the original, which was described to be a 44/16 DAT master. As it turns out, you needed to read the narrative given on the detailed page (https://shop.klicktrack.com/2l/468051) for this release to realize that we are actually dealing with two separate masterings. This was far from obvious, yet Archimago became aware of it not long after he wrote his article, as evidenced by his Addendum 3 at the bottom of the page. That's more than two months ago, but Stuart doesn't seem to have noticed it, yet he has no problem leaving Archimago in a bad light for not immediately becoming aware of the issue. Anyway, note that comparing two separate masterings with each other is not particularly useful when one wants to see what MQA does, so this example puts a shadow over the "HiRes Download - test bench" list page provided by 2L.
• Even when the Nielsen piece was remastered to 24 bits before MQA encoding, it still obviously originates from a 16-bit source. Such remastering can't magically remove the noise that was there before. They allegedly did some processing derived from the measured behavior of the converter that was originally used for recording, in order to change the temporal behavior (their time-smear thing), but Stuart doesn't explain how this makes the lower 8 bits not noise. The result, as Archimago points out, compresses less well with FLAC, in addition to being more data to begin with. I don't see anything wrong with Archimago's comments.

There's a lot more (don't get me started on Stuart's diagrams), but what I described already suffices to demonstrate the ethics in force here.

Bob Stuart has answered the questions that were collected by the ComputerAudiophile forum:

http://www.computeraudiophile.com/content/694-comprehensive-q-mqa-s-bob-stuart/ (http://www.computeraudiophile.com/content/694-comprehensive-q-mqa-s-bob-stuart/)

That's an awful lot of words to simply say "You're too stupid to understand our great format, and even if you weren't I'm not going to tell you what it does".

Too mindlessly moronic to read it through to the end

Too mindlessly moronic to read it through to the end

As I mentioned earlier in the thread, this is par for the course. I am surprised that others haven't really laid into the technical sections of their papers, which consist of the most outrageous half-truths, insinuations and semi-citations in an effort to push this ridiculous "revolution in psychoacoustics proves audiophiles were right all along" nonsense. It's like nailing snake oil jelly to a wall.

Much as I respect the technical expertise of members of this forum, my issue is so much more low-end, naive, newbee, etc.  When Robert Hartley of Absolute Sound goes so far out on a limb in favor of MQA, I feel I need to judge the sound for myself.  I'm sure there are better ways, but I bought a Meridian Explorer2, hooked it up, downloaded a host of MQA-encoded tracks, and let it rip.  The sound was OK, but the lights on my Explorer2 told me that the output of the DAC was CD-level PCM.  I called Meridian in England, and they recommended Foobar 2000, which would not convert the MQA signal to PCM (or similar evil) before sending it to the Explorer2.  I followed the few instructions they gave me, and the result was:  failure.  The three little lights remained stark white, indicating a PCM signal with no MQA encoding.

As always, I'm eager to take the blame.  I must have set up Foobar 2000 incorrectly.  Can anyone tell me whether there's a method, even a trick, for configuring Foobar 2000 to send the MQA signal from my computer hard disk to the DAC?  I would be extremely grateful.

Jeffrey Steingarten
Food Critic, VOGUE magazine, NYC

I am not familiar with their instructions.
All you should need is bit-perfect playback. The easiest way to ensure that is by using the wasapi output plugin (https://www.foobar2000.org/components/view/foo_out_wasapi). On foobar's preferences - output page select "WASAPI (event): your device" with highest supported output bit depth supported by the device.
Make sure you have no active DSPs on the DSP manager preferences page and the volume slider in the main windows set to 0 dB (100%).

Depending on how "great" the Explorer2 is you may even have to set the system volume to 100%.

If that doesn't work you could also try ASIO output ...

Much as I respect the technical expertise of members of this forum, my issue is so much more low-end, naive, newbee, etc.  When Robert Hartley of Absolute Sound goes so far out on a limb in favor of MQA, I feel I need to judge the sound for myself. 

That's an admirable sentiment, but: Robert Harley is not a credible authority on the nuts and bolts of audio, or on performing listening tests.  He's a pseudoscientist at best and a huckster at least.  I wouldn't embark on any investigations of sound quality on the basis of his recommendation (or disrecommendation).

I am not familiar with their instructions.
All you should need is bit-perfect playback. The easiest way to ensure that is by using the wasapi output plugin (https://www.foobar2000.org/components/view/foo_out_wasapi). On foobar's preferences - output page select "WASAPI (event): your device" with highest supported output bit depth supported by the device.
Make sure you have no active DSPs on the DSP manager preferences page and the volume slider in the main windows set to 0 dB (100%).

He might have to install the WASAPI plugin first... if so here it is:

https://www.foobar2000.org/components/view/foo_out_wasapi

That's MQA doing one of the things it was designed to do: proving that you have a bit perfect copy of what was created in the studio arriving at your DAC (or proving that you don't, in this case!).

For all the rest of it, I like that idea.

(Yep, I know it can be a bit perfect copy of a nasty brickwalled (or otherwise hopeless) recording ;)  )

Cheers,
David.

That's MQA doing one of the things it was designed to do: proving that you have a bit perfect copy of what was created in the studio arriving at your DAC (or proving that you don't, in this case!).

As does any other proprietary, patented, closed, or obscure format or even just watermark. Heck, a simple checksum that a studio publishes would be enough and wouldn't require tampering with the audio data at all.
Bit-perfect playback is also easy to set up.

In the end this also means all MQA DACs, always and ever, no matter what vendor must sound absolutely the same.

When Robert Hartley of Absolute Sound goes so far out on a limb in favor of MQA, I feel I need to judge the sound for myself.

A guy who can "hear" the improvements wrought by "high end" cabling...on a boombox (http://www.theabsolutesound.com/articles/audioquest-wind-interconnect/)??
Okey dokey. Well, I suppose audiophiles are always starving for the latest greatest nonsense like MQA and "higher" stereo "resolution".

Well, whatever we think of MQA's pros and cons, Warner Music Group have just announced they'll be using it for their downloads. If one of the biggest forces in the market have adopted it, it has now become something we will have to live with.
http://www.digitaltrends.com/music/warner-mqa-sign-deal-to-offer-hi-res-music-downloads/

.... which is why I still buy CDs and rip them myself.

Well, whatever we think of MQA's pros and cons, Warner Music Group have just announced they'll be using it for their downloads. If one of the biggest forces in the market have adopted it, it has now become something we will have to live with.
http://www.digitaltrends.com/music/warner-mqa-sign-deal-to-offer-hi-res-music-downloads/

Goodness... I thought Meridian was a niche audiophile equipment supplier; had no clue that they could have so much clout.

I can't speak about the hardware, but I find their licensing business to be parasitic in nature.

Bob Stuart could have never existed and the world of audio wouldn't have suffered any. Some other highly educated and well intentioned fluffer* would have filled the void, no doubt.

(*) it's figurative.  Heaven forbid a team of lawyers would step in (http://www.realhd-audio.com/?p=5659) to defend businessmen who peddle marshmallow.

EDIT: I grossly underestimated how much of their business is selling hardware.

Quote from: 2Bdecided on 2016-05-06 22:38:20

That's MQA doing one of the things it was designed to do: proving that you have a bit perfect copy of what was created in the studio arriving at your DAC (or proving that you don't, in this case!).

As does any other proprietary, patented, closed, or obscure format or even just watermark. Heck, a simple checksum that a studio publishes would be enough and wouldn't require tampering with the audio data at all.
Bit-perfect playback is also easy to set up.

Not wishing to argue for the sake of it, or suggest I would pay for it, or even that it matters - but you can't easily check the checksum at the DAC, and bitperfect playback is often broken, sometimes even unknowingly after you have set it up.

Whereas a system which adds something in the studio and puts a light on in the DAC does do this (assuming it works as advertised).

Cheers,
David.

I wonder

- if one could intentionally and audibly diminish the source material, but only when it is being played back in its compressed state.

- whether a competently-created 16/44.1 presentation couldn't be audibly equivalent, otherwise.
  (I don't really wonder about this part)

This shouldn't be a comfort* a consumer has to pay for** by way of an officially licensed indicator light.

(*) or implied "promise" of a superior mastering.

(**) twice: once for the licensing that is more than covered by the price paid by the consumer for the playback device; once for the markup of the encoded content.

Goodness... I thought Meridian was a niche audiophile equipment supplier; had no clue that they could have so much clout.

Meridian "supplies" the codec for Dolby TrueHD as well, with a market share of about 13 percent of Blu-Rays (http://www.blu-raystats.com/Stats/Stats.php) (thanks to this post (https://hydrogenaud.io/index.php/topic,107415.msg880360.html#msg880360) for the source.) Paramount uses Meridian Lossless on some 45 percent of their releases. Making the guess that most Blu-Ray actually use MLP from time to time, how does that userbase compare to the FLAC/ALAC/WMAL?

(Myself, I have been using their speakers for thirteen years or so. One coax in, amplification after x-overs.)

Some eye opening insights on MQA strategy hi-fi+ Bob Stuart of MQA and Morten Lindberg of 2L (http://www.hifiplus.com/articles/meet-your-maker-bon-stuart-of-mqa-and-morten-lindberg-of-2l/)
Some hard to misinterpret facts what it is really meant for.

One of my favourite sentences
"BS: We didn’t want to do public A-B tests, because they are completely uncontrolled. You know what it’s like – you put three audiophiles in a room and you get nine opinions..."

Think about that for a moment. This can mean many things. How about that one? "We better don't do public A-B tests because people suffering ringiphobia (fear of ringing) in reality don't hear a sh** so we better tell them what they hear and give it 3 magic letters."

Some hard to misinterpret facts what it is really meant for.

It also gives additional evidence that this whole intellectual property and licensing thing is behind it.

Quote from: Thad E Ginathom on 2016-05-07 19:02:05

Goodness... I thought Meridian was a niche audiophile equipment supplier; had no clue that they could have so much clout.

Meridian "supplies" the codec for Dolby TrueHD as well, with a market share of about 13 percent of Blu-Rays (http://www.blu-raystats.com/Stats/Stats.php) (thanks to this post (https://hydrogenaud.io/index.php/topic,107415.msg880360.html#msg880360) for the source.) Paramount uses Meridian Lossless on some 45 percent of their releases. Making the guess that most Blu-Ray actually use MLP from time to time, how does that userbase compare to the FLAC/ALAC/WMAL?

(Myself, I have been using their speakers for thirteen years or so. One coax in, amplification after x-overs.)

Good grief: Spotlight on my ignorance! Well, nothing new there, I suppose :-[

"BS: We didn’t want to do public A-B tests, because they are completely uncontrolled. You know what it’s like – you put three audiophiles in a room and you get nine opinions..."

Well, he's got a point. The magic wire guys blew it by having a few folks with >2 functional brain cells observe their "bump the volume on tracks" scam in public "tests" and it got exposed. So now what? Use a metal dome tweeter with a huge resonance peak >20k and drive the heck out of it with origami, or...?? Even a 70+ yr old daydream believer might hear that.

cheers

AJ

Quote from: Wombat on 2016-07-24 02:27:30

Some hard to misinterpret facts what it is really meant for.

It also gives additional evidence that this whole intellectual property and licensing thing is behind it.

That was the point. Japan must have one of the most strict copyright laws. No wonder they like it. Indirectly he accusses most others out there breaking the copyright.

We're all just a bunch of conspiracy (https://hydrogenaud.io/index.php/topic,112204.msg924496.html#msg924496) theorists.

Man the lifeboats, the BS "Answers" are finally here (albeit in the safety cocoon of an audiophile magazine):
http://www.stereophile.com/content/mqa-questions-and-answers#ugE36qSjYCoW3HKL.97 (http://www.stereophile.com/content/mqa-questions-and-answers#ugE36qSjYCoW3HKL.97)

It is now widely, although not universally, accepted that "hi-rez" digital audio, with increased sampling rate or bit-depth, delivers improved sound quality....

In the universe where SACD and DVDA, etc. didn't crash and burn, the fringe actually believe this.  ::)

But I want archival quality studio masters! And I want all of the original stems! So I can compose my own personal mixes in my fully licensed copy of Logic Pro X that I bought totally on a whim, and never share them with anyone else because I know they'll suck anyway, and it wouldn't be legal since all I'd be purchasing is personal listening rights.

Man the lifeboats, the BS "Answers" are finally here

"BS" is an abbreviation that needs disambiguating. Or... well, perhaps not.

Who's the guy in the picture with the wig and the bottle? Is that the actual eponymous? The man himself?

  

Who's the guy in the picture with the wig and the bottle?

Not a wig, Bob Stuart himself, hopefully no kool-aid in bottle.

On this occasion, I read the comments rather than the article. I like the one about the Donald Trump of Audio.

(And, even after so many years of reading audio fora, I still do a double take when I come across somebody looking for "BS Speakers." I mean, why would anybody?  :)) )

It looks like some albums are being remixed for MQA, which is further going to confuse the issue, since people will be comparing an MQA remix vs the original CD/Digital Download and attribute the changes to the superiority of the MQA format.

Check out this Mozart recording for sale on 7Digital:

https://us.7digital.com/artist/marianne-thorsen-and-trondheimsolistene/release/mozart-violin-concertos-mqa-remix-2016-5317973?f=20%2C19%2C12%2C16%2C17%2C9%2C2

I find it pretty funny that this "MQA remix" is available in MP3 format for under $9.00

The MQA version is for sale here (https://us.7digital.com/artist/marianne-thorsen-and-trondheimsolistene/release/mozart-violin-concertos-5080928?f=5317973) and costs a whopping $24.00.

I'm curious if we'll ever be able to convert MQA to another format and do a true ABX test.

I suppose the mp3 version is encoded from the undecoded files.

One should hope the mp3 wasn't transcoded from lossy origami.

Maybe someone can help me here. MQA is lossy but undecoded MQA is even more lossy. What is a MQA Remix exactly?

Presumably, a mix aimed at mitigating the effects of MQA’s lossy compression; i.e. to mask audibility of artefacts in the sound, and to minimise the visibility of artefacts in the spectrogram. Perhaps even to ‘enhance’ the appearance of the spectrogram, e.g. by boosting the ultra-sonics.

Presumably, a mix aimed at mitigating the effects of MQA’s lossy compression; i.e. to mask audibility of artefacts in the sound, and to minimise the visibility of artefacts in the spectrogram. Perhaps even to ‘enhance’ the appearance of the spectrogram, e.g. by boosting the ultra-sonics.

It might also just be a remix that sounds better overall, and they're just calling it an MQA remix because it was remixed to be released as MQA, but may sound better than the original mix if it was released as a standard 16/44.1 FLAC.  I'm not quite sure how you're going to get someone to drop $25 on an MQA release of an album.  But then again, people are buying HDTracks, SACD and DVD Audio, so someone is going to buy into this.

Eurgh, that Stereophile article makes me want to stick a fork up my nose. It's so calculatedly misleading - the farrago of citations, half-truths and innuendoes that characterises their 'papers' is even more in evidence here. It's rare to see so much in the way of mathematics, footnotes (and repetitions of the word 'neuroscience') and so little goddamn substance. It's a veritable work of art - it reads like it was written by their legal department.

So would you buy a used audio format from that guy, then?

Interesting write up from Dr Jim Lesurf
http://www.audiomisc.co.uk/MQA/origami/ThereAndBack.html (http://www.audiomisc.co.uk/MQA/origami/ThereAndBack.html)

BS explains how MQA finaly will fix digital audio of all labels. Everything ever sold in digital audio was simply done wrong.
MQA and Warner Stereophile link (http://www.stereophile.com/content/mqa-and-warner-real-scoop#1I3IXZYrxXZmwakv.97)

Too bad our savior-in-chief didn't get to the tapes before time  did?

Is this like a Monsanto-model type thing? Making money from every earful of music, rather than every mouthful of food, consumed by everyone in the world?

Thanks to Wombat for the link.

Quote:
"If a recording is important enough, and all there is is a 78, that's where we start. . . We're really concerned about producing the definitive thing," not the thing with the highest bit depth or sampling rate."

So it's just DRM then. With the advantage that it's the "definitive" drm'd version.

There is no definitive thing in pretty everything ever released. If you follow remasters of the last 30 years the EQ choice alone from version to version makes me believe this authenticated thing is hogwash. The MQA man only pushes the knob once again with a more stylish EQ until MQA+ will use another one in some years.
Funnily it often believed these HiBit releases are "flat" transfers.

MQA cures 24 bit ADC "smearing" also.
Next, it will be used for gout.

And the part before it:

Stuart indicates that MQA is not about high resolution in the usual sense; it's about authenticity. "As far as we're concerned, anything from a cylinder forward is legitimate as long as it's the definitive statement about a recording,

I would want this, if only for unmastered pre-compression-applied stuff from the last twenty-five-or-so years to be released in a better version. Not holding my breath though ...

Quote:
"If a recording is important enough, and all there is is a 78, that's where we start. . . We're really concerned about producing the definitive thing," not the thing with the highest bit depth or sampling rate."

So it's just DRM then. With the advantage that it's the "definitive" drm'd version.

Yeah, my placebo receptors already smell enough DRM for a significantly negative bias.

We were able to fingerprint and reverse-engineer the A/D out of the recording."

- BS

He's apparently saying they went back to the original digital recording and, using the original ADC as a guide,  'reverse engineered out' anything he deemed was distortion due to the ADC. 

IOW, a fancy version of noise modelling+denoising, which we know audiophiles just love, right?

Stereophile's craven sucking up to BS is pathetic even by its low standards.

Has anyone done an ABX test with an MQA release vs a FLAC?

No real abx i heard of. The MQA'd versions sound different to their older available versions. People say the new MQA version played over an MQA capable DAC and its captured output by a modern ADC are sounding indistinguishable.

why would somebody with Bob Stuart's legacy produce a fraud?

why would somebody with Bob Stuart's legacy produce a fraud?

Retirement fund?

https://patentscope.wipo.int/search/en/detail.jsf?docId=WO2013186561&recNum=132&docAn=GB2013051548&queryString=nano%20OR%20filter%20OR%20ceramic&maxRec=599628 (https://patentscope.wipo.int/search/en/detail.jsf?docId=WO2013186561&recNum=132&docAn=GB2013051548&queryString=nano%20OR%20filter%20OR%20ceramic&maxRec=599628)

Although it is necessary to have a decoder to achieve lossless reproduction of an original high-sample-rate signal, the signal provided to the legacy listener thus being described as 'lossy', the reduction to lossy is carried out in a manner that is described as 'benign' in audiophile circles

You can't get more scientifically valid than "being described as 'benign' in audiophile circles", now can you?

cheers,

AJ

Quote

it is necessary to have a decoder to achieve lossless reproduction of an original high-sample-rate signal

Without further qualification, I'm being kind by only saying this is misleading at best.

I think this would be an appropriate first track to be encoded in MQA:

:-)

https://www.youtube.com/watch?v=aLEhh_XpJ-0

Yeah, I know you can buy MQA encoded music now.  I just think the video is very appropriate for MQA.

Highresaudio.com has decided to stop offering music encoded with MQA, as they consider it a lossy format:

http://www.hifiplus.com/articles/highresaudio-to-stop-offering-mqa/#comment-3207552914

The comment section is... Interesting, to say the least. I have no idea who Peter Jasz is, but he seems extremely invested in MQA.

He's apparently saying they went back to the original digital recording and, using the original ADC as a guide,  'reverse engineered out' anything he deemed was distortion due to the ADC. 

IOW, a fancy version of noise modelling+denoising, which we know audiophiles just love, right?

Correcting phase and amplitude errors in the frequency response of the ADC is a straightforward linear process, and if done correctly would make the result more objectively correct.

However, trying to correct for aliasing, noise, etc would be as you suggest.

Cheers,
David.

Correcting phase and amplitude errors in the frequency response of the ADC is a straightforward linear process, and if done correctly would make the result more objectively correct.

Maybe, but the mastering engineer already performs all corrections/adjustments to get the best possible result—if MQA changes it audibly, then it’s worse.

Highresaudio.com has decided to stop offering music encoded with MQA, as they consider it a lossy format ... ... ...

Alternative headline: Highresaudio.com not all bad!

... ... ... I have no idea who Peter Jasz is, but he seems extremely invested in MQA.

He seems extremely interested in you! :o

.

Quote from: KozmoNaut on 2017-03-21 08:36:14

... ... ... I have no idea who Peter Jasz is, but he seems extremely invested in MQA.

He seems extremely interested in you! :o

If you read through his Disqus comment history, he's a very hotheaded audiophile true believer with very little social grace. He goes straight for the insults every time.

I just read the block close to that link and skimmed a bit more. I can easily believe it is typical. People that behave like that on the internet tend to... behave like that on the internet.

Quote from: krabapple on 2016-10-31 23:10:27

He's apparently saying they went back to the original digital recording and, using the original ADC as a guide,  'reverse engineered out' anything he deemed was distortion due to the ADC. 

IOW, a fancy version of noise modelling+denoising, which we know audiophiles just love, right?

Correcting phase and amplitude errors in the frequency response of the ADC is a straightforward linear process, and if done correctly would make the result more objectively correct.

Yes, but are those normally in need of correcting? Are ADCs often that flawed?

Quote from: Thad E Ginathom on 2017-03-21 12:04:01

Quote from: KozmoNaut on 2017-03-21 08:36:14

... ... ... I have no idea who Peter Jasz is, but he seems extremely invested in MQA.

He seems extremely interested in you! :o

If you read through his Disqus comment history, he's a very hotheaded audiophile true believer with very little social grace. He goes straight for the insults every time.

He's a nutter.

Are ADCs often that flawed?

That's the question I'm asking: http://www.stereophile.com/content/chesky-release-mqa-cds-may#4g45Cykdy5aid1Is.97 (http://www.stereophile.com/content/chesky-release-mqa-cds-may#4g45Cykdy5aid1Is.97)
More MQA Cds  ::)

Some comprehensive analysis and thoughts about MQA working. Archimago does measurements of audioquest-dragonfly-black (http://archimago.blogspot.ca/2017/06/measurements-audioquest-dragonfly-black_24.html)

Some comprehensive analysis and thoughts about MQA working. Archimago does measurements of audioquest-dragonfly-black (http://archimago.blogspot.ca/2017/06/measurements-audioquest-dragonfly-black_24.html)

I like Archimago and post on his forum, but the relevance of what he finds is limited by the fact that there don't seem to be any reliable listening tests to back it up. Many of the things he tests are among the easiest kinds of things to DBT.

Since I'm 70 and suffering from additional  neural damage related to the extensive chemotherapy that was part of  my recent (apparently successful) bout with Stage 3 colorectal cancer, this is a game I can't directly help with. 

Most of his measured results seem to relate to things that I wouldn't expect to sound much different, one way or the other.

Archimago provided some samples for MQA blind tests.
http://archimago.blogspot.hk/2017/07/internet-blind-test-mqa-core-decoding.html

But I found some problems in the samples. Read the comment section, I posted under another nickname with an ABX log. Other members here can also download the samples to find potential problems in order to make the test more reliable.

Nice finding. Still it will be interesting how many select the godlike deblurred one correctly in the end. I doubt many will submit abx logs.

This audio business never disappoints.
When your hardware lacks MQA decoding just beat the opponents with their own weapons. Use some upsample magic and tell your customers what to hear.
digitalaudioreview mqas-missing-link (http://www.digitalaudioreview.net/2017/07/kih-46-mqas-missing-link/)
SoX upsampling the crippped in bits MQA file without restoring any origami hidden HF content to a file sounding as perfectly good as the DXD original so better as MQA!
It sure will be fun to watch the smearfilterringingblur dominating high-end business in the future :)

Nice finding. Still it will be interesting how many select the godlike deblurred one correctly in the end. I doubt many will submit abx logs.

Regardless of the results, after the test expired and Archimago revealed the answers, people can always leave a comment in his blog and say they knew File B was real MQA because of the air, the realism, the inner details blah blah blah...

Regardless of the results, after the test expired and Archimago revealed the answers, people can always leave a comment in his blog and say they knew File B was real MQA because of the air, the realism, the inner details blah blah blah...

Absolutely. The comments section already has some pearls like 'MQA does best with pre 1960 recordings' ;)

This audio business never disappoints.
When your hardware lacks MQA decoding just beat the opponents with their own weapons. Use some upsample magic and tell your customers what to hear.
digitalaudioreview mqas-missing-link (http://www.digitalaudioreview.net/2017/07/kih-46-mqas-missing-link/)
SoX upsampling the crippped in bits MQA file without restoring any origami hidden HF content to a file sounding as perfectly good as the DXD original so better as MQA!
It sure will be fun to watch the smearfilterringingblur dominating high-end business in the future :)

Is it really this easy to unfold the mess?

digitalaudioreview mqas-missing-link (http://www.digitalaudioreview.net/2017/07/kih-46-mqas-missing-link/)
SoX upsampling the crippped in bits MQA file without restoring any origami hidden HF content to a file sounding as perfectly good as the DXD original so better as MQA!

Interesting. Though the previous article on that site about the "sound" of audiophile USB cables was funnier.

Quote from: Wombat on 2017-07-19 17:05:54

This audio business never disappoints.
When your hardware lacks MQA decoding just beat the opponents with their own weapons. Use some upsample magic and tell your customers what to hear.
digitalaudioreview mqas-missing-link (http://www.digitalaudioreview.net/2017/07/kih-46-mqas-missing-link/)
SoX upsampling the crippped in bits MQA file without restoring any origami hidden HF content to a file sounding as perfectly good as the DXD original so better as MQA!
It sure will be fun to watch the smearfilterringingblur dominating high-end business in the future :)

Is it really this easy to unfold the mess?

Remember that the whole article reads like the usual Placebophile...

I always thought you paid more for MQA so the little light would turn on.

Quote from: ron spencer on 2017-07-20 03:18:32

Quote from: Wombat on 2017-07-19 17:05:54

This audio business never disappoints.
When your hardware lacks MQA decoding just beat the opponents with their own weapons. Use some upsample magic and tell your customers what to hear.
digitalaudioreview mqas-missing-link (http://www.digitalaudioreview.net/2017/07/kih-46-mqas-missing-link/)
SoX upsampling the crippped in bits MQA file without restoring any origami hidden HF content to a file sounding as perfectly good as the DXD original so better as MQA!
It sure will be fun to watch the smearfilterringingblur dominating high-end business in the future :)

Is it really this easy to unfold the mess?

Remember that the whole article reads like the usual Placebophile...

So I think I will try this test on my own using their methodology, I have already downloaded the files. I guess after I convert the MQA one in SoX I should get my wife to rename the files and keep a record of what they are without me seeing. Then I can listen. Not fully scientific, but I guess it will do.

So I think I will try this test on my own using their methodology, I have already downloaded the files. I guess after I convert the MQA one in SoX I should get my wife to rename the files and keep a record of what they are without me seeing. Then I can listen. Not fully scientific, but I guess it will do.

You upsample with minimum phase, very steep filter without restoring the lower bits or the encrypted HF portion.
A negative abx may tell you MQA is never needed to begin with and DXD is also worthless. Good luck!
All this reasoning of the article is pretty weird imho.

Quote from: ron spencer on 2017-07-21 01:17:20

So I think I will try this test on my own using their methodology, I have already downloaded the files. I guess after I convert the MQA one in SoX I should get my wife to rename the files and keep a record of what they are without me seeing. Then I can listen. Not fully scientific, but I guess it will do.

You upsample with minimum phase, very steep filter without restoring the lower bits or the encrypted HF portion.
A negative abx may tell you MQA is never needed to begin with and DXD is also worthless. Good luck!
All this reasoning of the article is pretty weird imho.

Well I did as the article said, and had my wife use a random file name generator on them. From a casual listening perspective, I could not hear any differences. Using Focal headphones or Sennheiser IEMs on FiioX5 G3.

I figured I would try to ABX via foobar, but I get the "critical ABX error The parameter is incorrect.   (0x80070057)". I can play these files fine, but as soon as I choose any Play Button the error pops up. Doesn't happen for DSD files or any other flac/pcm. Any way to fix this?

So far undecoded MQA sounds like DXD when upsampled to you. This means low bit, standard samplerate done correctly sounds like DXD.
Can't help with that foobar error but try linear standard SoX upsampling please and try again.

So used the following to upsample

sox -S 2L-087_06_stereo_DXD_FLAC.mqa.flac MQA_upsampled.flac rate -vsL 352800

Note the L by the -vsL, as opposed to the M from the original link.

It still sounds good to me. Damn good. I really can't hear any differences, even in the silent fade outs.  Very good.

I would like to do a proper ABX test, but the ABX Comparator does not like DXD files, even though Foobar will play them; stupid error. If anyone knows a fix I'd like to ABX properly. In any case though, the linear upsampled MQA to DXD sounds awesome and I can't seem to tell the difference from the ORIGINAL DXD file I downloaded.

What if I downsampled the original DXD file to 192/24, then upsampled the MQA to 192/24 (to avoid the crashing of the ABX plugin with DXD files), and then use the Foobar ABX test. Would this tell us anything?

So odd that pcm and DSD will ABX fine with the plugin and not DXD.

The ABX component doesn't crash. You get the error from Windows as the maximum sample rate supported by DirectSound is 200 kHz. When you play tracks like this outside ABX session foobar2000 automatically enables a resampler to allow playback without errors.

Your workaround suggestion of resampling all files to 192 kHz is valid. You could also keep the files as they are and add a resampler to your foobar2000 playback chain and tick the "Use DSP" checkbox in the ABX component.

I would like to do a proper ABX test, but the ABX Comparator does not like DXD files, even though Foobar will play them; stupid error. If anyone knows a fix I'd like to ABX properly. In any case though, the linear upsampled MQA to DXD sounds awesome and I can't seem to tell the difference from the ORIGINAL DXD file I downloaded.

We shouldn't make the conclusion one needs upsampling to get good sound. We have enough of this nonsense in other forums.
When i think about the findings you made so far i guess the only thing that becomes clear is todays resampling is transparent.

Quote from: ron spencer on 2017-07-21 03:58:14

I would like to do a proper ABX test, but the ABX Comparator does not like DXD files, even though Foobar will play them; stupid error. If anyone knows a fix I'd like to ABX properly. In any case though, the linear upsampled MQA to DXD sounds awesome and I can't seem to tell the difference from the ORIGINAL DXD file I downloaded.

We shouldn't make the conclusion one needs upsampling to get good sound. We have enough of this nonsense in other forums.
When i think about the findings you made so far i guess the only thing that becomes clear is todays resampling is transparent.

Yah I understand about reaching for conclusions, but once the work day is over I am going to try this. I really am astounded as to just how good the upsampled MQA file is when comparing to the original DXD file. At least from my ears. Not sure what value I would get from MQA then.

You listen to upsampled PCM, not MQA.

I really am astounded as to just how good the upsampled MQA file is when comparing to the original DXD file. At least from my ears. Not sure what value I would get from MQA then.

Like it should sound with any correct downsampled version of the DXD file without MQA.

Seems like more vendors of hard and software that are not certified start to claim MQA sounds better as everything before with their own way of handling it. The main reason seems to come from not to loose potential buyers to MQA certified vendors.
Still this whole happening is facinating. People that already suffer of the 'fear of ringing' are happy to expand their psychogram with the newest ache 'phantom blur'
There is no cure but MQA!

Seems like more vendors of hard and software that are not certified start to claim MQA sounds better as everything before with their own way of handling it.

Care to clarify?

First my Reply #357 (https://hydrogenaud.io/index.php/topic,107666.msg942224.html#msg942224) and only an hour after posting this from a DAC/Sofware designer without MQA support How to handle MQA (https://www.computeraudiophile.com/forums/topic/30381-mqa-is-vaporware/?do=findComment&comment=732096)
Nobody wants to argue to much against MQA because it also may damage sales of own non MQA products.

Breaking news, there is proof ... at least it is "planned": https://www.computeraudiophile.com/forums/topic/30381-mqa-is-vaporware/?do=findComment&comment=731942

Breaking news, there is proof ... at least it is "planned": https://www.computeraudiophile.com/forums/topic/30381-mqa-is-vaporware/?do=findComment&comment=731942

"Planned", indeed.
https://www.computeraudiophile.com/forums/topic/30381-mqa-is-vaporware/?do=findComment&comment=732411

I won't be holding my breath waiting for it. I hope the good doctor does... :-)

“everyone has a plan until they get punched in the mouth”
Confucius Tyson

Actually, I think McGill will come through with a study

Archimago provided some samples for MQA blind tests.
http://archimago.blogspot.hk/2017/07/internet-blind-test-mqa-core-decoding.html

But I found some problems in the samples. Read the comment section, I posted under another nickname with an ABX log. Other members here can also download the samples to find potential problems in order to make the test more reliable.

Has anybody done any difference testing using either subtraction in an audio editor or Diffmaker on Archimago's before and after MQA files? 

I did and am looking at a lot of weird stuff. Since this should be pretty quick and easy to do, I'd like to see what happens when an independent party tried it.

In the interest of preserving independence, I'm not going to say more, but I'd like to...

I did before but it was not conclusive to me.  I doubt an average person could make a conclusion but listening.
If you want a working cheat just isolate the lower bits and the difference should be obvious like with any other bithdepth comparison. MQA has no real correlated info there.

I did some subtraction tests and there were indeed some differences apart from the fades, may be due to the fact that Archimago's simulated filter has some differences with the MQA renderer's filter, or some hard-to-align phase differences. However I found no hint to tell which is which.

Anyway, the test is over and I really hope Archimago will write some other topics since I am getting bored. I read some pages in the CA threads and it seems that some shills were banned. Thanks to the culture of HA the thread here is not as chaotic.

I read some pages in the CA threads and it seems that some shills were banned.

No difference when others are seamless replaced by an even more obvious sent out jerk to spoil threads.

“everyone has a plan until they get punched in the mouth”
Confucius Tyson

Actually, I think McGill will come through with a study

Whether that is a good thing or a bad thing, remains to be seen in the details of the actual experiment(s). The proposal presented to the AES seems to be very non-specific.

Today Mandy Parnell, a MQA hired mastering engineer is cited in one of the most popular online computer pages in germany that MQA encryption is a good thing for fighting the loudness war because streaming-service can't apply own additional compression.
This is the first time i read about it here.
heise.de Ableton Loop... (https://www.heise.de/newsticker/meldung/Ableton-Loop-Musik-Streaming-Dienste-sind-ein-Minenfeld-3888126.html)

I am not sure if Google is translating correctly (in Chinese or English), but the translated text say

With MQA, we get mastering engineers back in control, we deliver the encoded songs so no streaming provider can spoil them with their compression

That's mostly untrue.

[1] Most streaming services just decrease the volume of low DR songs, they don't compress high DR songs.
http://productionadvice.co.uk/online-loudness/

[2] Loudness normalization can be disabled/customized.
https://community.spotify.com/t5/Desktop-Linux-Windows-Web-Player/Disable-loudness-normalization/td-p/1024596
https://smartphones.gadgethacks.com/how-to/tidal-101-enable-loudness-normalization-0180380/

In fact, when loudness normalization is correctly implemented, it can reduce the chance of getting lossy codec/DAC induced clipping. Also, if MQA doesn't support loudness normalization, it just means it has yet another demerits:

[1] it lacks a function that other formats have
[2] it is not immune to loudness war since louder songs won't get turned down

The translation is good and why many Tidal MQA releases are more compressed as older CDs is a mistery then ;)

"we get mastering engineers back in control" - pure bullshit, and that's exactly the cause of loudness war.

http://fairhedon.com/2017/11/05/an-interview-with-mastering-engineer-brian-lucey/ (http://fairhedon.com/2017/11/05/an-interview-with-mastering-engineer-brian-lucey/)

Sounds like more of the same lossy compression is the cause of loudness compression idiocy.

"If I want that distortion in the master I would’ve put it there in the first place. The results of MQA I would call fatal to the source material even as they are very subtle." - Brian Lucey

The artist intent isn't to make lossy sounding noise.

"we get mastering engineers back in control" - pure bullshit, and that's exactly the cause of loudness war.

well...not necessarily.  Some report that they simple are doing what their client wants, when they pump up the loudness.

http://fairhedon.com/2017/11/05/an-interview-with-mastering-engineer-brian-lucey/ (http://fairhedon.com/2017/11/05/an-interview-with-mastering-engineer-brian-lucey/)

"However I’m always open minded and am not a crusty cynic like some, so I gave it an open minded listen.  Not bad, not great was my impression.  It’s definitely a lossy codec, that was clear.

Where as mastered for iTunes is harmonically cold and loses some low volume/low end information, actually altering the groove to make everything sound like a nerdy white wedding band, MQA brightens the high-mids in the Mid section while thinning the low-mids on the Sides. There’s also some harmonic distortion which some people could find pleasing,  If I want that distortion in the master I would’ve put it there in the first place. The results of MQA I would call fatal to the source material even as they are very subtle.
[/b] "

Gonna bet that his audition method wouldn't pass HA standards, though.  I'm skeptical whenever listeners report this sort of 'very subtle' yet 'clear difference' between a lossy encoding, and lossless.  Even if I agree that MQA is bunk.

http://fairhedon.com/2017/11/05/an-interview-with-mastering-engineer-brian-lucey/ (http://fairhedon.com/2017/11/05/an-interview-with-mastering-engineer-brian-lucey/)

From further reading Brian Lucey likes to master in 24bit and his releases are 44.1kHz. Modern music for the typical human.
In this case i believe him more as if it came from an audiophile studio dweller who palavers about DSD and DXD.
Like with many things in audio one can only speculate what the real intention the different audio engeneers may have because of things happening in the background.
What is staggering is that Lucey really registered at CA to stand for his findings. There the fun part just seems to start because he is told that the golden eared species doesn't suffer of things like ears frequency limits and physics. I didn't read everything yet but will do :)

Quote from: bennetng on 2017-11-11 18:41:20

"we get mastering engineers back in control" - pure bullshit, and that's exactly the cause of loudness war.

well...not necessarily.  Some report that they simple are doing what their client wants, when they pump up the loudness.

Likely because their clients still don't know about what loudness normalization does.

It should be fine if they don't mind their songs got turned down then. Merzbow still sounds like Merzbow when played at 14dB quieter.

What is staggering is that Lucey really registered at CA to stand for his findings.

Link?
Yes, I didn't posit his statements about MQA "sound" as gospel, but rather as an example of a mastering engineer who hasn't drank the MQA koolaid and isn't hopping on board and peddling the scam for $$ like Ludwig et al.

Brian Lucey comment at CA (https://www.computeraudiophile.com/forums/topic/30381-mqa-is-vaporware/?do=findComment&comment=742658)
Brian Lucey on mastering as pro (https://www.computeraudiophile.com/forums/topic/37552-mqa-off-topic-spinoff/)

Authenticated? Brian Lucey: "Many of my records currently sold as MQA, I had nothing to do with any of them" (https://www.computeraudiophile.com/forums/topic/30381-mqa-is-vaporware/?page=231&tab=comments#comment-748543)

Seeing Chris being made the fool never gets tiring.

Nice ban. Now Brian Lucey can concentrate on making music instead of wasting his time on a shilling forum.

Nice ban. Now Brian Lucey can concentrate on making music instead of wasting his time on a shilling forum.

Chris ConMan needs the MQA money to feed his kids

The latest examination of "MQA claims": https://www.stereophile.com/content/mqa-some-claims-examined (https://www.stereophile.com/content/mqa-some-claims-examined)

Anyone seen this talk from the chaos communication congress?

Trigger warning: Contains some moderate amount of audiophilia. Nonetheless this is worth watching. MQA explained from a Hacker's viewpoint.

MQA - A clever stealth DRM-Trojan (https://media.ccc.de/v/34c3-9113-mqa_-_a_clever_stealth_drm-trojan)

By the way, the "authenticated" part of MQA strikes me as a single benefit of this otherwise useless format. And no, it doesn't mean we should buy into it.

The cryptographic authentication process - at least in theory - would keep the rights owners and download/streaming services from applying digital watermarks further down the supply chain. Or at least consumers could notice immediately that the audio has been altered. Audible or not, i see these watermarks as a way to spy on consumer behavior and i'd really like to get rid of these. Sad thing that meridian implemented this idea first.

Maybe someone could hack up an open source solution to cryptographically "authenticate" audio using an existing format such as flac. That would be pretty awesome.

*edit*
Maybe even in a way that "survives" resampling and/or bitdepth reduction, by not considering the single "bits and bytes", but the actual resulting signal, allowing for just the right margin of error (i.e. everything that cannot be represented in redbook resolution / that doesn't fall in the audible frequency range - when playing back redbook resolution audio).

Another thought...
Maybe this is the whole reason MQA exists in the first place. I'm sure meridian holds patents on the authentication stuff and will take licensing fees from anyone who is implementing some similar technology, which i'm sure someone will want to do in the future.

Someone was able to let his own files distributed on Tidal getting the full MQA treatment including test tones.
He summarizes the debacle well in a video.
GoldenSound on youtube I published music on Tidal to test MQA - MQA Review (https://www.youtube.com/watch?v=pRjsu9-Vznc)
By accident he also stumbled across the typical Watermark pattern at ~10.20  :)

Someone was able to let his own files distributed on Tidal getting the full MQA treatment including test tones.
He summarizes the debacle well in a video.
GoldenSound on youtube I published music on Tidal to test MQA - MQA Review (https://www.youtube.com/watch?v=pRjsu9-Vznc)
By accident he also stumbled across the typical Watermark pattern at ~10.20  :)

This video is excellent, and something everyone should watch.

The slurpers won't care. The blue Pavlov light is suffice.

Just had a quick nosey at their website. Still pretending to be lossless...

https://www.mqa.co.uk/newsroom/qa/is-mqa-lossless

Still pretending to be lossless...

It is lossless AND better than lossless!!!oneoneone (where "oneoneone" unfolds to 1101 which has higher resolution).

Eff, i am using Bob Stuart's loudspeakers ...

MQA perfectly illustrates the ineffectiveness of development solely for profit versus development in pursuit of effective design.

MQA your wallet comes up lossless.  😂

Just had a quick nosey at their website. Still pretending to be lossless...

https://www.mqa.co.uk/newsroom/qa/is-mqa-lossless

Back when Meridian still owned MQA, they did not claim to be lossless. Now I guess they're trying to change their image. I remember seeing audiophiles clearly state that MQA is lossy. But that didn't matter. Whatever secret sauce they were using was making things sounds BETTER than lossless.

I think one of the more disturbing things is that Tidal is removing 16/44.1 FLAC versions of their music and ONLY offering MQA as an option.

I think one of the more disturbing things is that Tidal is removing 16/44.1 FLAC versions of their music and ONLY offering MQA as an option.

In for an edit war at https://en.wikipedia.org/wiki/Tidal_(service)#Plans against MQA shills then?

MQA shill army already is out on ASR for example with 10 posts fresh members ignoring all obvious findings and spreading disturbing marketing crap all over again. MQA is good for you (https://www.audiosciencereview.com/forum/index.php?threads/mqa-deep-dive-i-published-music-on-tidal-to-test-mqa.22549/page-15#post-749388)
This MQA plague has the most obvious machinery doing exactly these things since i read about it.

Today there was an edit referencing that video, to the  MQA wikipedia page (https://en.wikipedia.org/wiki/Master_Quality_Authenticated) ... (not by me).
Let's see what happens.

On YouTube a video was posted showing how the MQA process alters the sound and how the claims made by MQA do not hold true.

😂

MQA=Mangles the Quality of Audio

MQA=Mangles the Quality of Audio

Master Quality Aliasing

Master Quality Aliasing

LOL, what about
Master Quality Annihilated

Mastered by QAnon

Making Quirky Assertion

Meridians Questionable Audio

Mystery Quality Audio

This one should be a concern for everyone thinking MQA does no harm to me as long they don't use Tidal.
GoldenOne is the person who created the video linked above also hinted to another problem: MQA flagging removed (https://www.audiosciencereview.com/forum/index.php?threads/mqa-deep-dive-i-published-music-on-tidal-to-test-mqa.22549/post-754693)
This means there may be MQA processed files out there simply not being identifiable as such.
In this process the lower bits are pure and hard to compress garbage. It reduces the available bits you purchase.
Labels may prepare one and the same data but only flag the files used for explicit MQA certified enviroments.
For 16bit MQA butchered CD files you may loose up to 3 bits if i understood this right.

He said that the 2 files were bit-identical, but one identified as MQA and the other didn't. Since the MQA control bitstream is embedded in the audio, both claims can't be correct?

No idea where the MQA is exactly embedded but it doesn't seem so easy. There is mqascan and MQA identifier for detecting it but MQA identifier seems to miss some files where mqascan does detect it.
Maybe you can find a post Mans Rullgard the mqascan dev describes its working.

Mans handed me the clue I was missing... Apparently Tidal indicate whether a file is MQA or not in the metadata. But both files actually contain MQA encoded audio. Shady AF...

Now to the next disturbing thing one can learn from following the different active MQA threads here and there.
The Roon music software pays money to MQA for every decode they do in their player.
At Audiophile Style someone linked to the Roon forum post of a COO posting "MQA bills us based on Core Decodes"
This MQA vermin is a clever beast indeed. Wonder how much money the artist sees for every Core Decode.
Oh Wait! The artist already has the MQA encrypted Crown Jewels rescued so this is obviously only fair...

Now to the next disturbing thing one can learn from following the different active MQA threads here and there.
The Roon music software pays money to MQA for every decode they do in their player.
At Audiophile Style someone linked to the Roon forum post of a COO posting "MQA bills us based on Core Decodes"
This MQA vermin is a clever beast indeed. Wonder how much money the artist sees for every Core Decode.
Oh Wait! The artist already has the MQA encrypted Crown Jewels rescued so this is obviously only fair...

That can't possibly be true, surely!? Who the hell would sign up to that business model? Maybe that's why Roon is charging soft heads $700 for a software media player...

May Hydrogenaud.io and AudioScienceReview save us all.

Link to the roonlabs thread MQA bills us based on Core Decodes (https://community.roonlabs.com/t/why-do-manufacturers-support-mqa/155965/135?u=kenneth_jonge)
Imagine if Roon adds localy played MQA files to the count a Roon user pirated. MQA Bob still gets his money.