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Real-world comparisons of 24-bit and 16-bit music from Linn Records -
2Bdecided
post Aug 24 2009, 15:47
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It's supposedly a direct-to-stereo acoustically / naturally mixed recording.

There's obviously at least a gain change - you'd never record that close to 0dB FS live and be lucky enough not to clip!

Cheers,
David.
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WernerO
post Sep 9 2009, 09:10
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I am sorry, but it appears to me that many of you don't know what you are
doing here.

When starting from a 96kHz master it is impossible to compare 96kHz with 44.1kHz.

What you are comparing is the 96kHz source with [[[ the combination of a 44.1kHz sampled signal space with the particular anti-alias filter used when converting from 96kHz to 44.1kHz ]]].

So before starting you have to agree on which AA filter you'll include.

"Generate your own Redbook master from Linn's HiRez WAV (VHQ SRC with intermediate phase low-pass"

SoX with intermediate phase is, obviously, non-linear-phase and thus will add phase distortion in the 10-22kHz band of the 44.1kHz version. You introduce a difference before you even started ... (hence the failing nulls).

Maybe that is what you wanted to test, but I doubt it.

What AA filter to use then during downsampling? (edit: typo)

One possible choice would be a linear phase half-band filter with not too much of transition band attenuation, mimicking the type of filter that is part of the typical recording ADC.
It injects pre- and post-ringing at 22.05kHz and it has some aliasing. Upon replay such a signal will force the typical DAC to pre-ring too, on top of the AA's ringing.

Another possible choice would be a linear phase filter with cut-off slightly below 22kHz, say 21kHz, and a very steep transition band rolloff. It injects pre- and post-ringing at 21kHz, but it has no aliasing. Upon replay such a signal will not force the typical DAC to pre-ring too. There's only the AA's ringing.

The minimum phase and intermediate phase AA filters exist to get rid of the pre-ringing, at the cost of phase distortion. Whether upon replay there still is pre-ringing depends entirely on the particular configuration of the AA filter and the choice of replay filter in the DAC (or software oversampler).

"I didnīt do any upsampling - that would have changed the results. "

Quite the contrary, in fact. Not doing oversampling in a controlled way in the software, prior to sending the 44.1kHz to the DAC leaves all oversampling, including the reconstruction filter, at the mercy of the DAC. And typical DAC oversampling/reconstruction filters are linear-phase half-band, i.e. with pre- and post-ringing at 22.05kHz and imperfect stop band rejection.
Such a reconstructor violates Shannon's theorem, which prescribes the Sinc(t) function for replay. So, in order to do a valid test, take the DAC out of the equation and use SoX for oversampling, with settings that approximate a Sinc function (linear phase, highest quality, highest steepness, 22kHz or slightly less).

This post has been edited by WernerO: Sep 9 2009, 09:11
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MLXXX
post Sep 13 2009, 07:57
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QUOTE (WernerO @ Sep 9 2009, 18:10) *
"I didnīt do any upsampling - that would have changed the results. "

Quite the contrary, in fact. Not doing oversampling in a controlled way in the software, prior to sending the 44.1kHz to the DAC leaves all oversampling, including the reconstruction filter, at the mercy of the DAC. And typical DAC oversampling/reconstruction filters are linear-phase half-band, i.e. with pre- and post-ringing at 22.05kHz and imperfect stop band rejection. ...

Thanks WernerO for your various observations.

I think we are left with the fact that if we are to attempt any comparisons between 24/96 and 16/44.1, extreme care is needed in creating and playing the 16/44.1 files. An interesting article I came across today (which many HA members may have already seen) on the topic of ant-aliasing filtering for 44.1KHz, Ringing False: Digital Audio's Ubiquitous Filter, by Keith Howard in stereophile January 2006, can be found here.

If there are any audible differences, they appear to be likely to be extremely small.
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krabapple
post Sep 13 2009, 18:29
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Howard is a white hat but alas for the usual Stereophile pandering to subjectivism, placing its 'findings' on par with those of formal tests:


QUOTE
listening tests were undertaken to establish formally that the anti-alias filtering required for a nominally flat response to 20kHz, even at the 44.1kHz sampling rate—the sternest test because the transition band from 20kHz to 22.05kHz is so narrow, demanding an extremely steep filter rolloff—has no audible effect. Many informal listening tests were conducted too, often using Sony's PCM-F1, because it was the first 16-bit digital recorder that most people were able to lay hands on. The outcomes of this testing, formal and informal, were overwhelmingly positive. Many PCM-F1 users claimed that a signal passed through the machine's A-to-D and D-to-A stages was indistinguishable from the feed, and some still cite that experience as proof that 16-bit/48kHz audio, properly realized, is essentially perfect.

But even in the earliest days of domestic digital audio there were dissenting voices. Many hi-fi writers, myself included, were thoroughly underwhelmed by our initial experiences of Compact Disc, and so were some influential audio professionals, such as Doug Sax*. Over a period of some years the intensity of this opposition to CD decreased somewhat, but many commentators and ordinary audio consumers concluded that there was something fundamentally amiss with 16/44.1 and 16/48 audio. Many of them voted with their feet, continuing to prefer the sound of the "obsolescent" LP.


AND, the 'various experiments' he cites to support response to >20kHz playback, are all from the unreplicated work of Oohashi et al.

rolleyes.gif



*who uses Shakti Stones.

This post has been edited by krabapple: Sep 13 2009, 18:55
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Arnold B. Kruege...
post Sep 14 2009, 03:04
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QUOTE (WernerO @ Sep 9 2009, 04:10) *
I am sorry, but it appears to me that many of you don't know what you are
doing here.

When starting from a 96kHz master it is impossible to compare 96kHz with 44.1kHz.

What you are comparing is the 96kHz source with [[[ the combination of a 44.1kHz sampled signal space with the particular anti-alias filter used when converting from 96kHz to 44.1kHz ]]].

So before starting you have to agree on which AA filter you'll include.

"Generate your own Redbook master from Linn's HiRez WAV (VHQ SRC with intermediate phase low-pass"


This sounds like a red herring argument to me. No matter what wider-bandpass source you start out with including a wide-bandwidth analog source, you're converting from a wider-bandwidth source to a lower-bandwidth source.

Before one looses too much sleep over this, they ought to try the experiment with whatever down-conversion product they have, and see if they get a positive result. Then if you get positive results. we can do a post-mortum.

Back in 2002 or so I posted a goodly number of downsampled files, made from 24/96 recordings that I made with a wide-bandwidth production system including mics that were reasonably flat past 40 KHz. These recordings were arranged so that they naturally containe exceptional amounts of energy > 20 KHz. I did the resampling with CoolEdit Pro. Nver heard any postive results involving anything but some training files that involved downsampling to some rediculous sampling rate like 28 KHz.


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Kees de Visser
post Sep 14 2009, 10:02
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QUOTE (Arnold B. Krueger @ Sep 14 2009, 04:04) *
they ought to try the experiment with whatever down-conversion product they have, and see if they get a positive result. Then if you get positive results. we can do a post-mortum.
According to this HA thread it might be time for a post-mortem. There seems to be a positive ABX for a 24/96 to 16/44.1 conversion.
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Arnold B. Kruege...
post Sep 14 2009, 13:55
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QUOTE (Kees de Visser @ Sep 14 2009, 05:02) *
QUOTE (Arnold B. Krueger @ Sep 14 2009, 04:04) *
they ought to try the experiment with whatever down-conversion product they have, and see if they get a positive result. Then if you get positive results. we can do a post-mortum.
According to this HA thread it might be time for a post-mortem. There seems to be a positive ABX for a 24/96 to 16/44.1 conversion.


So you didn't note or follow up on the coment in the heading: "Successful ABX of TPDF white dither vs. noise-shaping at normal listen, Editorial note: Seems there are other issues besides dither at play."
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Kees de Visser
post Sep 14 2009, 14:27
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QUOTE (Arnold B. Krueger @ Sep 14 2009, 14:55) *
Seems there are other issues besides dither at play."
Sure, and it's a pity that the OP doesn't seem to see the importance of separating the issues of SRC and dithering/re-quantization. However, if he can hear a difference between a 24/96 and a 16/44.1 source, that deserves some attention IMO since according to HA belief this should not be possible unless something has been done wrong.
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Arnold B. Kruege...
post Sep 14 2009, 17:25
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QUOTE (Kees de Visser @ Sep 14 2009, 09:27) *
QUOTE (Arnold B. Krueger @ Sep 14 2009, 14:55) *
Seems there are other issues besides dither at play."
Sure, and it's a pity that the OP doesn't seem to see the importance of separating the issues of SRC and dithering/re-quantization. However, if he can hear a difference between a 24/96 and a 16/44.1 source, that deserves some attention IMO since according to HA belief this should not be possible unless something has been done wrong.



Hence the interest in the origional commerically-available track.

True scientific skepticism does not deny that the possibility that there may be some audio some place that actually sounds different when downsampled right.

Our present findings are due to us not having the good fortune of actually finding that audio, given many energetic efforts to find it. While we might suspect that such a piece of audio does not exist, we have to remain open to the possibility that it may exist.

There is also the possiblity that some errors in the prepartion of the test may explain its results. One way to convince ourselves of the adquacy of the test results that have been reported is for us to replicate them ourselves, starting with the origional source material.
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Nick.C
post Sep 14 2009, 18:13
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QUOTE (Kees de Visser @ Sep 14 2009, 14:27) *
....this should not be possible unless something has been done wrong.
Until the original is provided and processed in the prescribed manner we would be foolish to make any assumption regarding the correctness of the processing of the dithered samples that have been provided.


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rpp3po
post Oct 7 2009, 12:08
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QUOTE (Cavaille @ Aug 19 2009, 11:16) *
QUOTE (rpp3po @ Aug 19 2009, 11:03) *
As far as I know the E-MU 0202 USB does not switch sample rates automatically, but employs Windows' mediocre quality realtime-SRC to convert to the rate set in the E-MU USB Audio Control Panel.
Oh, but it does. Completely automatic.


The following comment by Cavaille in another thread made me look at our only claimed positive 24/96 vs. 16/44.1 ABX result in this thread in a different light:

QUOTE (Cavaille @ Oct 7 2009, 10:27) *
I use the 0202 USB [...] Both interfaces also have one big problem: they donīt work very well with foobarīs ASIO output. It has to be configured anew after every restart of the host-pc.


Looks like it might not have been as "automatic" as claimed and it might have been flawed switching after all and not golden ears that made the difference. I restate: prior upsampling using a high quality converter is a much cleaner approach to evaluate wether 16/44.1 discards audible content that 24/96 would preserve.

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Cavaille
post Oct 20 2009, 00:00
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QUOTE (rpp3po @ Oct 7 2009, 13:08) *
QUOTE (Cavaille @ Oct 7 2009, 10:27) *
I use the 0202 USB [...] Both interfaces also have one big problem: they donīt work very well with foobarīs ASIO output. It has to be configured anew after every restart of the host-pc.


Looks like it might not have been as "automatic" as claimed and it might have been flawed switching after all and not golden ears that made the difference. I restate: prior upsampling using a high quality converter is a much cleaner approach to evaluate wether 16/44.1 discards audible content that 24/96 would preserve.
My friend, you try to punch holes into something where nothing is contradicting. I suspect you do that just so you can keep your opinions alive - whatever the costs and whatever they are.
So letīs read the following sentence very carefully: The E-MU 0202 USB switches sample rates automatically (it doesnīt have an internal resampler). BUT: when you re-start your PC youīll have to configure foobar2000s ASIO output. Letīs read again: foobar2000īs ASIO output. Not the interface itself.

Reason: foobar2000 somehow uses the hardware-ID for assigning the ASIO output (there is a thread for this somewhere on HA). Since the interface always gets a different Hardware-ID after every re-start due to the special synchronized USB connection, foobar2000 forgets all the information. So this is not a failure of the interface, it is a failure of... well, it should be obvious by now. Several users complained about this but apparently nothing will be done - and I donīt want to indulge about the reasons for this.

Every other ASIO-capable program I use (WaveLab, iZotope, Sound Forge, hell, even WinAmp (!)) works every time with the interface. It is just a problem of foobar2000.

I donīt know what youīre talking about here but I had to correct this.

This post has been edited by Cavaille: Oct 20 2009, 00:13


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2Bdecided
post Oct 20 2009, 09:51
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QUOTE (2Bdecided @ Aug 24 2009, 10:36) *
I ordered the http://www.soundkeeperrecordings.com/ disc - the music was quite nice too, which is more than you can say for many "audiophile" recordings.
As a follow up, a friend was saying that he thought mp3 players were just toys (though he had and used one), and he thought music was best from vinyl.

So I played him the Dragon Boats track from my Sansa Clip through HD580 headphones.

I can't say that I "converted" him there and then, but he was very impressed! He was amazed that you could really hear the space in which it was recorded.

Cheers,
David.
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rpp3po
post Oct 20 2009, 12:45
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QUOTE (Cavaille @ Oct 20 2009, 01:00) *
So letīs read the following sentence very carefully: The E-MU 0202 USB switches sample rates automatically (it doesnīt have an internal resampler). BUT: when you re-start your PC youīll have to configure foobar2000s ASIO output. Letīs read again: foobar2000īs ASIO output. Not the interface itself.


Technically, that's not accurate. Since the 0202 doesn't employ an ASRC the sample rate has to be set in software (and ASIO has provisions to do that automatically). A hardware sample rate detection would mean a considerable amount of samples lost during rate detection.

And yes, I don't believe in your results (or at least conclusions). It is against anything known, that has been conducted in a controlled environment. There is no audible difference for 24/96 for humans at the listening levels we have had here. The only advantage of 96 kHz is that cheaper filters can accomplish what takes some effort to be inaudible at 44.1 kHz. If you live in the Stuttgart region, how about we repeat the exact same test at the Max Planck Institute in Tübingen? Only condition would be that a 2nd control test is conducted with pre-converted files (96->44.1->96 vs. 96) to rule out side effects of the 0202.

QUOTE (Cavaille @ Oct 20 2009, 01:00) *
Reason: foobar2000 somehow uses the hardware-ID for assigning the ASIO output (there is a thread for this somewhere on HA). Since the interface always gets a different Hardware-ID after every re-start due to the special synchronized USB connection, foobar2000 forgets all the information. So this is not a failure of the interface, it is a failure of... well, it should be obvious by now.


And BTW, the 0202 probably doesn't use "synchronized" (isochronous) USB, that's about the worst mode there is for audio (common for budget gear), but asynchronous (isochronous) USB, which can be designed to have very high jitter tolerance. With asynchronous USB the audio device signals the exact rate at which the host should push audio frames. Since the audio hardware is in total control over the clock and the host just "listens & obeys", the sample rate must be known by the audio device in advance, it cannot be "detected". This is either done through a driver control panel or, in the case of ASIO, sent as side information of an audio stream to the driver, which extracts it and sends a command to the device to switch the pulling rate.

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krabapple
post Aug 4 2010, 17:21
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More thread fodder --

Diament has now posted three versions of a track, in 16/44, 24/96, and 24/192

http://www.soundkeeperrecordings.com/format.htm
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2Bdecided
post Aug 4 2010, 17:26
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There are three new samples up on the Soundkeeper Recordings website:

http://www.soundkeeperrecordings.com/format.htm

44.1kHz vs 96kHz vs 192kHz this time.

They're from a soon-to-be-released album. Beautifully recorded, just like the first. I wish music that I like was recorded with such care.

(I'm surprised to see a laptop present in the recording room itself).

Cheers,
David.

EDIT - DOH! I guess you're on his mailing list too then krab! wink.gif

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krabapple
post Aug 5 2010, 03:40
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QUOTE (2Bdecided @ Aug 4 2010, 12:26) *
EDIT - DOH! I guess you're on his mailing list too then krab! wink.gif



Nope...i have other informants. wink.gif
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2Bdecided
post Aug 6 2010, 12:31
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One instrument on this new recording has "lots" of energy above 20kHz. Everything else would be basically untouched by a 20kHz low pass filter, even if humans could hear up to 40kHz.

Still, you've gotta love Stereophile (talking about the previous release) "...It's not that there's so much going on in those extended high frequencies, but it's obvious that they're not there at all on the "Red Book" CD. It's not so much a question of night and day as one of life and death..."

Now you'd think life vs death would be ABXable, wouldn't you? wink.gif When confronted with an ABX test, can Stereophile journalists no longer tell whether they're dead or alive? wink.gif

Cheers,
David.
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Northpack
post Aug 6 2010, 13:23
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QUOTE (2Bdecided @ Aug 6 2010, 11:31) *
Now you'd think life vs death would be ABXable, wouldn't you? wink.gif

Well, do you really think you could ABX beeing dead?

Hm... maybe we are all going to hell for our outrageous scepticism and our punishment shall be to do 16/44.1 vs. 24/96 ABX testing for the rest of eternity.

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Wombat
post Aug 6 2010, 14:02
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QUOTE (Northpack @ Aug 6 2010, 14:23) *
QUOTE (2Bdecided @ Aug 6 2010, 11:31) *
Now you'd think life vs death would be ABXable, wouldn't you? wink.gif

Well, do you really think you could ABX beeing dead?


You may concentrate on the sound of breathing, not breathing.

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krabapple
post Aug 9 2010, 07:58
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It appears this thread has sparked some interest over on Steve Hoffman's board.....

http://www.stevehoffman.tv/forums/showpost...p;postcount=606

Start there and read ahead to savor this classic bit of audiophilery:

QUOTE ("LeeS")
The Hydrogen guys are inexperienced. Anyone who has done even a modest amount of hirez recording understands the value of 24/96 over 16/44. It's simply not subtle, no matter how good the 16/44 playback is.
"
It seems to me we have two audiophiles - those that listen to music, seek out the best, and enjoy it and those that debate the scientific merits endlessly.

I don't want to shut down the technical debate if it is done in an open-minded way and subjective assessments are included...after all, you cannot measure everything we hear. Sadly I see little open-mindedness at Hydrogen.


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Northpack
post Aug 9 2010, 08:27
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QUOTE ("LeeS")
It seems to me we have two audiophiles - those that listen to music, seek out the best, and enjoy it and those that debate the scientific merits endlessly.

Funny, but I thought there are two kind of "audiophiles" indeed. Those who enjoy the music on equipment sufficient for the task, and those who listen to their equipment and not to the music.
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2Bdecided
post Aug 9 2010, 11:07
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More of the usual "of course you can easily hear a difference, it's obvious" - which would, if pressed, magically turn into "I can't ABX a difference".

I like the honest post that says "the differences are subtle". I think that's the conclusion that any rational person must come to - if there are any audible differences, they must be very subtle, because positive test results are so rare.


Consider this...

1. People say "it's all about having revealing gear" - now, I can see why that could be true in theory.
2. People say "it's all about having golden ears" - again, I can see why that could be true in theory.*

But if both of these things were true in practice for 24/96, people selling "revealing" gear would do tests to prove that the 16/44 vs 24/96 difference was provably genuinely audible through their equipment (at least for the "golden ears" who could hear it).

How great a piece of marketing would this be? "Buy our speakers - you can really hear the benefits of 24/96 - here are the test results to prove it".

Such evidence and marketing is notable by its absence. Draw your own conclusions.

Cheers,
David.

P.S. * - number 2 is true in practice for many provably real, audible, measurable but subtle differences. Psychoacoustic test results (measuring the ability to hear the barely audible) prove this all the time - results are determined by 1) raw ability and 2) training. No one would argue with this.
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Arnold B. Kruege...
post Aug 9 2010, 12:16
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QUOTE (2Bdecided @ Aug 9 2010, 06:07) *
More of the usual "of course you can easily hear a difference, it's obvious" - which would, if pressed, magically turn into "I can't ABX a difference".

I like the honest post that says "the differences are subtle". I think that's the conclusion that any rational person must come to - if there are any audible differences, they must be very subtle, because positive test results are so rare.


Consider this...

1. People say "it's all about having revealing gear" - now, I can see why that could be true in theory.
2. People say "it's all about having golden ears" - again, I can see why that could be true in theory.*


If you understand all of the theory, then you conclude that its not all about just any one thing.

If you want the most sensitive test possible, its about the source material, *and* the equipment, *and* the room, *and* the listener training, *and* the listening procedures. The best complete statement about all of that is AFAIK ITU Recommendation BS 1116.

Anybody who goes off on just one or two of the above is basically telling us about how incomplete his knowlege and experience are.
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Cavaille
post Aug 9 2010, 12:20
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The post from the stevehoffman forum is almost insulting! Some people appear to be thinking that thereīs a war going on between either side. Such unnecessary, time consuming bulls**t. However, as stupid as that comment is I always was under the impresson that "both sides" need each other: the fools for making stupid propositions and the realistic guys to prove them wrong or right. Just imagine, there could be coming something good from it after all. It doesnīt appear to be right now but maybe in the future... I also try to think up an example from the past - right now it slips my mind.

BTW, David could it be argued that higher samplerate is more important after all? This may be farfetched but I draw conclusions from your listening tests of brickwall limiting where I found the differences subtle but existing.


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