AVS/AIX High-Resolution Audio Test: Next Step - Page 8 - AVS Forum
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post #211 of 232 Old 06-16-2014, 03:09 PM
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Quote:
Originally Posted by mcnarus View Post
(By the way, has anyone at AVS noticed that the quote function doesn't really work anymore?)
it works but it's pretty hard to read them but they change a lot in the past days the "thread subscriptions" is back!
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post #212 of 232 Old 06-16-2014, 04:06 PM
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By three sources I meant individual input envelopes from a source. Voice, piano, guitar, drums, keyboard. Even an organ/keyboard uses more than one single note to achieve the tone desired. Many devices do derive their inputs via a moving diaphram like a microphone, others from an electromagnetic coil that picks up vibrations of metal strings. All of those devices creates it own sinusoid, mixed with others gives even more harmonics.
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post #213 of 232 Old 06-16-2014, 06:18 PM
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[quote]By three sources I meant individual input envelopes from a source. Voice, piano, guitar, drums, keyboard. Even an organ/keyboard uses more than one single note to achieve the tone desired. Many devices do derive their inputs via a moving diaphram like a microphone, others from an electromagnetic coil that picks up vibrations of metal strings. All of those devices creates it own sinusoid, mixed with others gives even more harmonics.[/quote]
But it still all sums to one sinusoid before it hits the ADC—and that happens in the analog domain. Encoding that sinusoid is no different than encoding a pure tone.

If you can't explain how it works, you can't say it doesn't.—The High-End Creed

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post #214 of 232 Old 06-17-2014, 03:20 AM
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Originally Posted by SiGGy View Post
Yes, I have used ABX comparator and that player before.
Apparently not with these comparisons, or otherwise you'd say so.

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I have also done my homework and time matched, level matched
Apparently not blind with these comparisons, or otherwise you'd say so.

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the files then put it through this app as well.

http://www.libinst.com/Audio%20DiffMaker.htm
Irrlevant.

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There are many ways to do a comparison.
Your apparent ability to avoid the relevant ones, is interesting.

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Just because ABX made some software/hardware doesn't mean is the only/proper way to do it.
Avoding double blind testing says it all.

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Your tone/attitude is kinda off putting here.

I get lectured about attitude by someone who pooh-poohs DBTs?

LOL!

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The calibrated level of the monitors being used can also hide or enhance any differences. usually best to have the artists listen for differences IMO. Most casual listeners are not that in tune with how instruments should sound. Especially solo'd.
Irrelevant to the 500 pound gorilla in the room which is bias control.

People hear what they want to hear. As long as they avoid true blind testing, that's what they say they hear.
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post #215 of 232 Old 06-17-2014, 06:09 AM
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Quote:
Originally Posted by arnyk View Post
Apparently not with these comparisons, or otherwise you'd say so.



Apparently not blind with these comparisons, or otherwise you'd say so.



Irrlevant.



Your apparent ability to avoid the relevant ones, is interesting.



Avoding double blind testing says it all.




I get lectured about attitude by someone who pooh-poohs DBTs?

LOL!



Irrelevant to the 500 pound gorilla in the room which is bias control.

People hear what they want to hear. As long as they avoid true blind testing, that's what they say they hear.
If you didn't argue/speak in the tone/talking down I'd continue to have a discussion with you. I'm done at this point.

The point of this thread is for normal folks to be able to do their own comparisons and brainstrom the how's. I think the software you mentioned is one way for regular folks to do this; especially with it's cost.

However...

1. there are multiple versions/implementations of the ABX software; do we assume they are all implemented the same?

That won't be the case...

2. Depending on the implementation will it level match a 44khz and 96khz file? using what?
a) so, does this need to be done in advance?
1a) if it does need to be matched in advance might as well just roll your own single file.

Generally ABX is used for codec/bitrate differences not sampling rate changes.

Like I said, it would just be easier to level match and up-convert a 44khz sample to 96khz. Put both the untouched 96k and the up-converted in the same FLAC/WAV and have it switch back and forth between the 96k and 44khz up-sampled.

Way less involved... many ways to do this. In fact almost all of the ABX disclaims also state "ABX is not intended to be a replacement for a properly setup A/B listening"

Anyway, if anyone else has thought I'd love to hear them.

-SiGGy
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post #216 of 232 Old 06-17-2014, 05:56 PM
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Guys, I have no problem with the discussion at hand, as that is what forums are all about. However, when it starts to get personal, that's when I step in. Please keep all of your posts focused on the current topic, and do it in a friendly, non combative manner, or I will be forced to close this thread.

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post #217 of 232 Old 06-17-2014, 06:10 PM
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Quote:
Originally Posted by SiGGy View Post
The point of this thread is for normal folks to be able to do their own comparisons and brainstrom the how's.

1. there are multiple versions/implementations of the ABX software; do we assume they are all implemented the same?
I have no clue. How could I possibly know the details of every implementation of ABX software including those that I may have never seen before?

Quote:
T2. Depending on the implementation will it level match a 44khz and 96khz file? using what?
Again I have no clue. How could I possibly know the details of every implementation of ABX software including those that I may have never seen before?

The *right* way to do this kind of comparison is

(1) Start out with a high sample rate file

(2) Downsample it

(3) Upsample it

(4) Compare the two high sample rate files.
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post #218 of 232 Old 06-17-2014, 06:33 PM
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Originally Posted by arnyk View Post
I have no clue. How could I possibly know the details of every implementation of ABX software including those that I may have never seen before?



Again I have no clue. How could I possibly know the details of every implementation of ABX software including those that I may have never seen before?

The *right* way to do this kind of comparison is

(1) Start out with a high sample rate file

(2) Downsample it

(3) Upsample it

(4) Compare the two high sample rate files.
Its one way of doing it, i even mentioned it prior. But I don't think it's the "right" way. Oh wait sorry the *right* way.

Down sampling from 96k is not the same as recording at 44k. Any good algorithm down sampling will look at all of the data and make a logic decision based on the 96k wave form. This will enhance the 44k from the extra data that existed in the 96k sample.

Ask a good dsp programmer if you know one...

Same applies to applying effects to tracks. Some effects applied at 96k then down covered to 44k sound better then if you applied them at 44k. There are subtleties you would not hear otherwise.

This is the similar to scanning images on a computer. You always scan at double the dpi you need then scale down using an algorithm. Image comes out clearer because it has more data to choose what is "right" when scaling.

I am done now though, love to continue this discussion with others on here.
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-SiGGy

Last edited by SiGGy; 06-17-2014 at 06:42 PM.
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post #219 of 232 Old 06-18-2014, 02:38 AM
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Quote:
Originally Posted by SiGGy View Post
Its one way of doing it, i even mentioned it prior. But I don't think it's the "right" way. Oh wait sorry the *right* way.

Down sampling from 96k is not the same as recording at 44k. Any good algorithm down sampling will look at all of the data and make a logic decision based on the 96k wave form. This will enhance the 44k from the extra data that existed in the 96k sample.

Ask a good dsp programmer if you know one...

Same applies to applying effects to tracks. Some effects applied at 96k then down covered to 44k sound better then if you applied them at 44k. There are subtleties you would not hear otherwise.

This is the similar to scanning images on a computer. You always scan at double the dpi you need then scale down using an algorithm. Image comes out clearer because it has more data to choose what is "right" when scaling.

I am done now though, love to continue this discussion with others on here.
As anyone with minimal Photoshop experience knows, you're right, it't better to apply the effect on the higher resolution, then downscale, as opposed to the other way around.

But the issue in question is whether 16/44.1 (some even argued as low as 12 bits ) is an appropriately transparent encoding to deliver to consumers, not whether it's appropriate for mastering.

I find that downscaling 24/96 to 16/44 and then upsampling is perfectly adequate, when trying to test for audible differences between the formats.
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post #220 of 232 Old 06-18-2014, 05:18 AM
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Quote:
Originally Posted by SiGGy View Post
Its one way of doing it, i even mentioned it prior. But I don't think it's the "right" way. Oh wait sorry the *right* way.

Down sampling from 96k is not the same as recording at 44k. Any good algorithm down sampling will look at all of the data and make a logic decision based on the 96k wave form. This will enhance the 44k from the extra data that existed in the 96k sample.

Actually downsampling and upsampling is the right way.


Recording at 16/44.1 and 24/96 may be different due to implementation of brickwall filter and downsampling in the AD converter.
If you hear a difference you can be sure it's the implementation and not the difference in sampling rate.


A down sampling algorithm doesn't 'look at all the data'. Simply put it applies a brickwall filter and after this it drops samples. After this it adds dither noise at the appropriate level and then truncates to 16 bit. There is no 'intelligence' by looking at the samples and making decisions basad on the content.


Quote:
Originally Posted by SiGGy View Post
Ask a good dsp programmer if you know one...

Same applies to applying effects to tracks. Some effects applied at 96k then down covered to 44k sound better then if you applied them at 44k. There are subtleties you would not hear otherwise.

The discussion is not about the best format for processing and editing. It's about the final delivery format after the final mastering.


Quote:
Originally Posted by SiGGy View Post
This is the similar to scanning images on a computer. You always scan at double the dpi you need then scale down using an algorithm. Image comes out clearer because it has more data to choose what is "right" when scaling.

I am done now though, love to continue this discussion with others on here.
Images and audio are vastly different and you can't compare image processing and audio processing.

Last edited by Frank Derks; 06-18-2014 at 05:21 AM. Reason: quote tag problems
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post #221 of 232 Old 06-18-2014, 07:28 AM
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Originally Posted by Frank Derks View Post
Actually downsampling and upsampling is the right way.


Recording at 16/44.1 and 24/96 may be different due to implementation of brickwall filter and downsampling in the AD converter.
If you hear a difference you can be sure it's the implementation and not the difference in sampling rate.


A down sampling algorithm doesn't 'look at all the data'. Simply put it applies a brickwall filter and after this it drops samples. After this it adds dither noise at the appropriate level and then truncates to 16 bit. There is no 'intelligence' by looking at the samples and making decisions basad on the content.





The discussion is not about the best format for processing and editing. It's about the final delivery format after the final mastering.




Images and audio are vastly different and you can't compare image processing and audio processing.
No, this is incorrect on down sampling. Some down sampling does apply logical dithering (quantization) from the high set of data. The best ones do not just drop the bits and hit a wall. We do this in our signal processing (DSP) for our products, so I know it has to be done by some audio manufacturers. I know I have seen different dithering implementations mentioned on Protools/Sonar in fact you can usually choose which one you want.

I see what you are saying. I can see how there could be differences depending on how it is approached in the hardware too. However most DAC lock at a clock, they don't sample high then output at a lower rate. However internally there has to be some level of dithering in the DAC.

yes, I agree. So it's really post mastering were discussing I was solely focused on 44k vs 96k raw samples which I have A<->B many times (correctly). I was thinking I don't want to give the 44k any more advantage then it would normally had if sampled at 44k, but in reality most productions would be 96k down converted to 44k anyway.

-SiGGy

Last edited by SiGGy; 06-18-2014 at 07:37 AM.
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post #222 of 232 Old 06-18-2014, 09:56 AM
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Originally Posted by SiGGy View Post
Its one way of doing it, i even mentioned it prior. But I don't think it's the "right" way. Oh wait sorry the *right* way.

Down sampling from 96k is not the same as recording at 44k.
That is true and is also good news, it turns out.

Quote:
Any good algorithm down sampling will look at all of the data and make a logic decision based on the 96k wave form. This will enhance the 44k from the extra data that existed in the 96k sample.
Enhancing down sampling that way is impossible if you understand Shannon's Information Theory. But downsampling can potentially do as good of job as is possible because it doesn't have to happen in real time.

If you use one of the many very fine downsampling programs,

http://src.infinitewave.ca/

then you are comparing sampling resolution as such which has more generality. If you make separate recordings at different sample rates then you are comparing different pieces of equipment which has reduced generality and you are also making a lot of added work for yourself.

The downsampling approach is very democratic, just about anybody can do it for themselves with only a good so-called high rez recording on hand. After all of these reports including yours of mind blowing improvements due to high sample rates, why now all of the sudden is so much correct mouth holding required?

Quote:
Ask a good dsp programmer if you know one...
Thanks for talking down to me after falsely accusing me of doing the same thing to you. ;-)
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post #223 of 232 Old 06-18-2014, 10:04 AM
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Originally Posted by SiGGy View Post
No, this is incorrect on down sampling. Some down sampling does apply logical dithering (quantization) from the high set of data. The best ones do not just drop the bits and hit a wall.
It's not just the best ones that apply dither when reducing resolution, its the ones that are competently designed, which in this day and age is most of them.

Try this reference:

http://src.infinitewave.ca/

Some of these SRCs show obvious evidence of poor design. For example the one by Acon Digital.

It looks like the majority do things right as far as decreasing resolution, but quite a bit fewer do the frequency domain part correctly (example of frequency domain performance that is not so good - all flavors of Abletron).
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post #224 of 232 Old 06-18-2014, 10:11 AM
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Originally Posted by arnyk View Post
That is true and is also good news, it turns out.



Enhancing down sampling that way is impossible if you understand Shannon's Information Theory. But downsampling can potentially do as good of job as is possible because it doesn't have to happen in real time.

If you use one of the many very fine downsampling programs,

http://src.infinitewave.ca/

then you are comparing sampling resolution as such which has more generality. If you make separate recordings at different sample rates then you are comparing different pieces of equipment which has reduced generality and you are also making a lot of added work for yourself.

The downsampling approach is very democratic, just about anybody can do it for themselves with only a good so-called high rez recording on hand. After all of these reports including yours of mind blowing improvements due to high sample rates, why now all of the sudden is so much correct mouth holding required?



Thanks for talking down to me after falsely accusing me of doing the same thing to you. ;-)
Actually if you read my threads you'd see I was going to use the same identical 12 channel DACs (I have multiple) at the same time splitting the mics to output to both identical converters. Thus recording at both 44k and 96k at the same time. So it would be identical different pieces of equipment. However I agree with earlier poster that this comparison is really about post production not simply raw recordings.

You are putting words in my mouth. I just said it was audible especially with dynamic instruments and gave examples. However I was also talking about raw tracks, not dithered down-sampled audio. Care to quote where I said "mind blowing improvements". That would be a personal subjective opinion, one of which I never said. I don't post like you do so I know I never said that

Dithering and quantization is anything but simple and there are MANY approaches to doing it.

Spoke down to you? It appears from your point of view everyone is down...

-SiGGy

Last edited by SiGGy; 06-18-2014 at 10:21 AM.
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post #225 of 232 Old 06-18-2014, 10:20 AM
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Quote:
Originally Posted by arnyk View Post
It's not just the best ones that apply dither when reducing resolution, its the ones that are competently designed, which in this day and age is most of them.

Try this reference:

http://src.infinitewave.ca/

Some of these SRCs show obvious evidence of poor design. For example the one by Acon Digital.

It looks like the majority do things right as far as decreasing resolution, but quite a bit fewer do the frequency domain part correctly (example of frequency domain performance that is not so good - all flavors of Abletron).
That's interesting, yes some are quite bad. Took me a bit to figure out how to interpret what I was looking at. Lots of data to go through in there.

-SiGGy
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post #226 of 232 Old 06-18-2014, 11:13 PM
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Originally Posted by SiGGy View Post
That's interesting, yes some are quite bad.
Luckily quite a few are excellent, so there is hardly any excuse for not using a near-perfect SRC. These tests were done in the digital domain. You can see that artifacts are at a very low level, which makes it hard to believe that there is any audible difference between the 96 and 44.1 kHz format, unless something goes terribly wrong in the monitoring chain (DAC, amp, speakers).
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post #227 of 232 Old 06-19-2014, 03:26 AM
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Originally Posted by SiGGy View Post

Spoke down to you? It appears from your point of view everyone is down...
Specifically, the idea that I don't know people who write DSP code or other signal processing code is totally ludicrous.
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post #228 of 232 Old 06-19-2014, 05:57 PM
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Please keep all of your posts focused on the current topic, and do it in a friendly, non combative manner, or I will be forced to close this thread.
I thought it was rather tame for these neck of the woods


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post #229 of 232 Old 06-21-2014, 03:56 PM
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<p>A couple of weeks ago, I started <a href="http://www.avsforum.com/t/1531590/avs-aix-high-res-audio-test-proposed-methodology">a thread</a> asking for comments about a methodology to create audio files that AVS members could use to investigate whether or not high-resolution audio can be audibly discerned from CD audio. The response has been great; thanks to all who contributed to that discussion!</p>
<p> </p>
<p>After carefully considering those comments and continuing to think about the best way to proceed, AIX Records founder Mark Waldrep (aka Dr. AIX) and I have made a few decisions. First of all, it's pretty clear that most consumer audio systems are not capable of reproducing the ultrasonic frequencies and expanded dynamic range that 24/96 audio can capture. If you try to compare a high-resolution 24/96 track that contains additional fidelity to a 16/44.1 downsampled version of the same track on most consumer systems, which top out at 20 kHz, both tracks will deliver the same acoustic information.</p>
<p> </p>
<p>Still, I'm willing to bet that some AVS members have systems that are capable of reproducing ultrasonic frequencies and wider dynamic range than a Redbook CD, so we've decided to limit our data collection to those with such systems. Of course, anyone can download the files we create and give them a listen, but unless your system meets or exceeds the required specs, we can't include your observations about which files are high-res and which are CD audio in our data.</p>
<p> </p>
<p>What are the required specs? Let's start with system configuration. Basically, the fewer components in the signal chain, the better, so we think the best setup is a computer as the source device connected via USB to a high-quality DAC (digital-to-analog converter). The DAC would be connected directly to a power amp, which would, of course, drive the speakers.</p>
<p> </p>
<p><a class="H-lightbox-open" href="http://www.avsforum.com/content/type/61/id/439786/"><img alt="" class="lightbox-enabled" data-id="439786" data-type="61" src="http://www.avsforum.com/content/type/61/id/439786/width/1000/height/1000/flags/LL" style="; width: 900px; height: 183px"></a></p>
<p><em>The best system configuration for the AVS/AIX high-res audio test is a computer sending 24/96 data to a high-performance DAC that is connected directly to a power amp feeding the speakers. This uses the fewest possible components, minimizing the potential for mucking up the signal.</em></p>
<p> </p>
<p>Obviously, the computer needs to be able to output native 24/96 via USB, and the DAC needs to be able to convert that to analog without downsampling (or oversampling, for that matter). The analog output of the DAC must be able to support ultrasonic frequencies up to near 48 kHz and a dynamic range greater than 93 dB (CD's effective maximum after dithering to deal with quantization noise), and the power amp must have similar frequency-response and dynamic-range specs from input to output. Finally, the speakers must be able to reproduce those ultrasonic frequencies and wide dynamic range with minimal distortion.</p>
<p> </p>
<p>What about using headphones? A few headphones claim ultrasonic response—for example, the Oppo PM-1 is spec'd to 50 kHz in a free field, though the attenuation at that frequency is not specified. But I know of no independent measurements that verify these claims beyond 22 kHz. And even if a set of headphones can reproduce ultrasonic frequencies, it's possible that the acoustic coupling between the headphones and ears might affect these frequencies in some significant way—notice that the Oppo spec is in a free field.</p>
<p> </p>
<p>It's also possible that human perception of ultrasonic frequencies is not auditory, but perhaps involves some other sensory mechanism; if so, headphones would not allow that mechanism to contribute to the overall perception. Other theories of ultrasonic perception suggest that the auditory system is involved, so ultrasonic-capable headphones might work, but we don't know for sure.</p>
<p> </p>
<p>Several commenters suggested that we provide two files for each clip—one native 24/96 with verified ultrasonic content and wide dynamic range and the other a 16/44.1 downsampled version of the same clip that is then upsampled to 24/96—and have participants use an ABX program such as foobar2000 for Windows or ABXTester for the Mac. I was originally against this idea, thinking that I wanted listeners to be able to participate without having to use a computer, so I devised an alternate approach. But it looks like a computer is the best source device to use in this exercise, so using an ABX program to compare two files seems like a good way to go.</p>
<p> </p>
<p>For me, the biggest problem is keeping the test honest—which, let's face it, is impossible if we're going to rely on participants to inform us about their systems and observations of which file is which without independent verification. Someone could easily mislead us about their system and/or take a peek at the spectrum of the test files to see which is which. As a result, this experiment cannot be considered a rigorous scientific test. Still, we think it's an interesting exercise, and I can only hope that those who end up participating will do so honestly.</p>
<p> </p>
<p>To start the process, we will pre-quality the audio systems of those who wish to participate. If that includes you, please post a comment here with a list of your system components and specs (and room measurements if you have them); alternatively, you can PM me with this info with "AVS/AIX HRA Test" in the subject line. Once we have a number of qualified participants, we will make the test files available for download. As I said earlier, anyone is welcome to download those files and give a listen, but unless you have a system such as I've described here, there will likely be little or no acoustic difference to perceive.</p>
<p> </p>
<p>What do you think of our approach? I'd love to read your comments about it here. If you want to discuss the general notion of whether or not high-res audio can be distinguished from CD, please do so in <a href="http://www.avsforum.com/t/1532092/debate-thread-scotts-hi-res-audio-test">Amir's debate thread</a> or <a href="http://www.avsforum.com/t/1529833/is-high-resolution-audio-irrelevant">my "Is High-Res Audio Irrelevant?" thread</a>. Let's keep this thread focused on the experiment we're proposing.</p>
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I've been messing around with HD music and reading with some interest the Dr AIX posts. I'd like to take part in the AB comparison that you are setting up.

I have a Mac Mini running Amarra, outputting to a NAD M51 via USB, connected to a Linn Chakra 2200 power amp and Sonus Faber Cremona floorstanding speakers.

I think this system is well up to the job of showing differences between tracks - the listening room is a typical non-dedicated living room with assorted pets and children, so I'll have to pick my moments...
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post #230 of 232 Old 06-25-2014, 08:17 PM
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What is the ETA of the rubber hitting the road? When does the action/results thread get created?


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post #231 of 232 Old 06-30-2014, 01:10 PM - Thread Starter
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Originally Posted by BassThatHz View Post
What is the ETA of the rubber hitting the road? When does the action/results thread get created?
It's here.

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post #232 of 232 Old 07-04-2014, 10:56 PM
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sweet!

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