[B] NEED HELP - Diganosing audio differences between source components [/B] - Page 9 - AVS Forum
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post #241 of 361 Old 05-24-2012, 06:20 PM
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Originally Posted by hd_newbie View Post

I thought you were only carrying high-end equipment.

So now there are two of you who think we should stop discussing OP's topic and talk about equipment my company carries. Is that right?

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post #242 of 361 Old 05-25-2012, 12:26 AM
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It is unfortunate that the avs members argue back and forth so much on this site that a lot of threads get derailed. One person trying to prove that they are smarter than the next. Not always the most mature route to take.

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post #243 of 361 Old 05-25-2012, 02:31 AM
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Originally Posted by amirm View Post

So now there are two of you who think we should stop discussing OP's topic and talk about equipment my company carries. Is that right?

When you are in the thread, we are always talking about the equipment your company carries!

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

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post #244 of 361 Old 05-25-2012, 03:26 AM
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Quote:
Originally Posted by arnyk View Post

Quote:
Originally Posted by amirm View Post

The album I used is Reference Recording's Exotic Dances from the Opera, Eiji Oue by Minnesota Orchestra. Here is the data:

Track: Bacchanale from Samson and Delilah:
Code:
Left Right
Peak Amplitude: 0 dB -.97 dB
Minimum RMS Power: -112.81 dB -111.06 dB
Maximum RMS Power: -2.26 dB -4.24 dB
Average RMS Power: -25.91 dB -26.61 dB
Total RMS Power: -21.94 dB -22.67 dB
Actual Bit Depth: 24 Bits 24 Bits


Please provide the track length and timing of the point at which the minimum power you show was obtained.

A common rookie mistake is to include artificial fade-ins, and fade-outs as part of the analysis. To be relevant, the minimum must be obtained when the microphones were actually running with normal gain settings.

Also, what means were used for extracting the digtial information from the disc?

I have to admit that I was excited by the possibility of a musical track that "broke the rules". It turns out that the Samson and Delilah "Bacchanalle" track is availble from HDTracks, so I downloaded it. The other track Amir mentioned is only available if you download the entire album.

It is probable that the HDTracks version of this track is an reasonably accurate representation of all other versions of it, and it is probable that Amir used the same methodology to analyze both tracks.

Here are the Cool Edit statistics for that track with obvious fade-ins and fade-outs excluded from the analysis:


Left Right
Min Sample Value: -30026.1 -29381.16
Max Sample Value: 31237.16 28701.73
Peak Amplitude: -.42 dB -.95 dB
Possibly Clipped: 0 0
DC Offset: 0 0
Minimum RMS Power: -70.68 dB -68.08 dB
Maximum RMS Power: -6.37 dB -7.94 dB
Average RMS Power: -28.47 dB -29.08 dB
Total RMS Power: -25.16 dB -25.45 dB
Actual Bit Depth: 24 Bits 24 Bits

(Using RMS Window of 20 ms)

A minimum RMS value is located at 7:39.114 into the track, and is isolated.

20 msec is 1/50th of a second, far, far less than human reaction time. It is improbable that one would sense the difference between total silence and -60 dB noise for this short of a period of time. My preferred window for perceptual purposes might be as much as half a second.

Bottom line is that with minimum RMS power of about -70 dB this is subjectively a reasonably quiet recording, but it does not deviate from the pattern that I have observed with every other recording that I have checked.

I've seen far better!
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post #245 of 361 Old 05-25-2012, 04:37 AM
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Originally Posted by amirm View Post

Before explaining the results, let me preface this with the fact it is non-trivial to determine such stats as what is music and what is noise is hard to separate.

Kind of a problem don't you think? Are you measuring the dynamic range of the equipment or the dynamic range of the music?
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post #246 of 361 Old 05-25-2012, 05:23 AM
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Originally Posted by derrickdj1 View Post

It is unfortunate that the avs members argue back and forth so much on this site that a lot of threads get derailed. One person trying to prove that they are smarter than the next. Not always the most mature route to take.

If the back and forth is 100% pesronal, then it is a waste.

If the back and forth brings to light relevant data, particulaly if it is new, then it may have some value.
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post #247 of 361 Old 05-25-2012, 05:31 AM
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Originally Posted by arnyk View Post

Here are the Cool Edit statistics for that track with obvious fade-ins and fade-outs excluded from the analysis:


Left Right
Min Sample Value: -30026.1 -29381.16
Max Sample Value: 31237.16 28701.73
Peak Amplitude: -.42 dB -.95 dB
Possibly Clipped: 0 0
DC Offset: 0 0
Minimum RMS Power: -70.68 dB -68.08 dB
Maximum RMS Power: -6.37 dB -7.94 dB
Average RMS Power: -28.47 dB -29.08 dB
Total RMS Power: -25.16 dB -25.45 dB
Actual Bit Depth: 24 Bits 24 Bits

...Bottom line is that with minimum RMS power of about -70 dB this is subjectively a reasonably quiet recording, but it does not deviate from the pattern that I have observed with every other recording that I have checked.

I've seen far better!

Just like his bogus claim that 7000ps is "typical" of HDMI jitter, it looks like amirm was cooking the books with his claim of extraordinary dynamic range of the HDtracks recording.

And he wonders why his assertions are taken to task. You can't trust anything he says.
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post #248 of 361 Old 05-25-2012, 05:36 AM
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Quote:
Originally Posted by audiophilesavant View Post


Just like his bogus claim that 7000ps is "typical" of HDMI jitter, it looks like amirm was cooking the books with his claim of extraordinary dynamic range of the HDtracks recording.

I have to stand up for Amir at this point. His original 7000 ps jitter data was an accurate rendition of the web site he cited.

The data he presented here about the opera recording appears to be a true and genuine rookie mistake.

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And he wonders why his assertions are taken to task. You can't trust anything he says.

On balance Amir gets a lot of things right, but there is that other stuff! ;-)
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post #249 of 361 Old 05-25-2012, 05:55 AM
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Originally Posted by arnyk View Post

I have to stand up for Amir at this point. His original 7000 ps jitter data was an accurate rendition of the web site he cited.;-)

So we have a total of six HDMI jitter data points:

5 ps
50 ps
1582 ps
3700 ps
3860 ps
7660 ps

Assuming that the sample size is sufficient, which is a huge leap (six out of hundreds), how do you arrive at 7000ps being "typical" from the data presented? Five of the six are lower, with the next closest being nearly half that amount. He just chose the highest and called it typical. If that's okay, why not choose the lowest and call that typical.

It's not the data he presents that's the problem. It's the interpretation. It's that it's cherry-picked and/or interpreted with whatever spin his finds useful to support his argument. He is not in search of the truth; he is in search of any data that will support his position, while ignoring the rest. Hence the push-back. The HDMI jitter statement is just one example.

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Originally Posted by arnyk View Post

On balance Amir gets a lot of things right, but there is that other stuff! ;-)

It's that "other stuff" that taints everything else, and why I am wary of anything he has to say.
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post #250 of 361 Old 05-25-2012, 06:00 AM
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Originally Posted by mcnarus View Post


When you are in the thread, we are always talking about the equipment your company carries!

It is difficult not to when he keeps posting pictures of equipment he is selling on a daily basis. I end up seeing pictures of his ML amps in almost every thread that exceeds few pages.
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post #251 of 361 Old 05-25-2012, 07:00 AM
 
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Originally Posted by hd_newbie View Post

It is difficult not to when he keeps posting pictures of equipment he is selling on a daily basis. I end up seeing pictures of his ML amps in almost every thread that exceeds few pages.

What else should we expect from someone who sells those gears and uses this forum as advertising outlet?
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post #252 of 361 Old 05-25-2012, 08:45 AM
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Quote:
Originally Posted by arnyk View Post

I have to admit that I was excited by the possibility of a musical track that "broke the rules". It turns out that the Samson and Delilah "Bacchanalle" track is availble from HDTracks, so I downloaded it. The other track Amir mentioned is only available if you download the entire album.

I have to say, I am exceptionally impressed that you spent the money to investigate this track. We argue a lot. But you do stand above others in trying to figure things out .

Quote:


It is probable that the HDTracks version of this track is an reasonably accurate representation of all other versions of it, and it is probable that Amir used the same methodology to analyze both tracks.

We used the same technology but not the same track. Mine was given to me by the producer/recording engineer, Keith Johnson (through a mutual friend) and is 24-bit/176 KHz. The version on HD Tracks is marked 24/96 Khz. So resampling is used. So we are not quite looking at the same bits.

Quote:


Here are the Cool Edit statistics for that track with obvious fade-ins and fade-outs excluded from the analysis:

Ah. I did not cut out part of it. I loaded the track and let it analyze it.

Quote:


Left Right
Min Sample Value: -30026.1 -29381.16
Max Sample Value: 31237.16 28701.73
Peak Amplitude: -.42 dB -.95 dB
Possibly Clipped: 0 0
DC Offset: 0 0
Minimum RMS Power: -70.68 dB -68.08 dB
Maximum RMS Power: -6.37 dB -7.94 dB
Average RMS Power: -28.47 dB -29.08 dB
Total RMS Power: -25.16 dB -25.45 dB
Actual Bit Depth: 24 Bits 24 Bits

(Using RMS Window of 20 ms)

A minimum RMS value is located at 7:39.114 into the track, and is isolated.

I cut off the beginning and the end arbitrary (i.e. to allow for fade in and fade out assuming as you did that they do exist as such), and I get similar numbers to your minimum now. The max for me though stays near clipping at -1 db for left channel and -3 db for right. If I use these numbers I get 1.4 (70-1) = 96 dB assuming sine wave and 69 assuming square wavefor our peak computation which is what we need here. So yes, looks like you have busted this as far as it being way out there.

Quote:


20 msec is 1/50th of a second, far, far less than human reaction time. It is improbable that one would sense the difference between total silence and -60 dB noise for this short of a period of time. My preferred window for perceptual purposes might be as much as half a second.

The problem is that Audition uses this number to average the values it computes. It is a sliding window across the whole track. To know which number is right, requires knowing the precise nature of what you are averaging. If I had silence and then an impulse and I averaged the two because I used too long of an interval, clearly I am not seeing the right value for the lowest and highest numbers anymore. The manual page for Audition explains the same thing: that the right value is content dependent. This is why I said this is back of the envelop stuff. The true analysis can only be done by the person recording the material or someone sitting there analyzing the track manually. I could get higher numbers if I shrunk the value below 20 msec but figured that was a good balanced position as a thumb in the air kind of thing.

Quote:


Bottom line is that with minimum RMS power of about -70 dB this is subjectively a reasonably quiet recording, but it does not deviate from the pattern that I have observed with every other recording that I have checked.

I've seen far better!

Can you post some? Per above, for the purposes of this discussion, we care about peak values, not RMS. The DAC doesn't see RMS. It has to produce the actual voltages represented by the peak and valleys, not RMS which attempts to provide an averaging function (root *mean* square). As I noted above, the sample we looked at with your modification is already at the 96 db limit of CD (non noise-shaped) for sinewaves. So if you have lower noise floor than that, then we have discovered such tracks.

Amir
Founder, Madrona Digital
"Insist on Quality Engineering"

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post #253 of 361 Old 05-25-2012, 09:08 AM
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Originally Posted by audiophilesavant View Post

Just like his bogus claim that 7000ps is "typical" of HDMI jitter, it looks like amirm was cooking the books with his claim of extraordinary dynamic range of the HDtracks recording.

You know, I have a policy of not ever reporting anyone on this forum. But it is posts like this that come dangerously close to changing my policy. You couldn't have waited five minutes to see my response before running off with the accusation that I am not ethical and cooked the numbers? Why do you think I provided the full data on the album/track? It was done so that others could verify what I say. I am meticulous about doing that. I provide back up to where I even had lunch if that were the topic. In this case, as I noted to Arny, his assumptions of where I got the track is wrong. This is the version I have: http://www.elusivedisc.com/prodinfo.asp?number=RRDR71



As I mentioned, it is at different sampling rate and hence I could easily claim that Arny and I had different tracks and hence his analysis doesn't apply to mine. But what is right is right. He was correct in taking out the fade in and fade out and I had not done that. So I re-ran and once I confirmed the same difference on mine, I presented the same.

Quote:


And he wonders why his assertions are taken to task. You can't trust anything he says.

Earlier I asked you what your position is with respect to dynamic range that we should strive for as the topic is for the thread. You can't be bothered to answer that relevant question but thought you should stir up trouble like this and calling my ethics into question? This, when I repeatedly couched my analysis as ad-hoc with more caveats than I could possibly put in?

Someone asked why these threads are so bad in tone. This is why. At least Arny and I try to contribute to the topic technically. We honestly don't need cheerleaders egging us on to have a fight, especially when they are grown men who won't win a beauty contest to save their lives.

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post #254 of 361 Old 05-25-2012, 09:41 AM
 
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Originally Posted by amirm View Post

and I try to contribute to the topic technically.

Deception is contribution now?
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post #255 of 361 Old 05-25-2012, 10:43 AM
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Originally Posted by amirm View Post

I asked you what your position[/url] is with respect to dynamic range that we should strive for as the topic is for the thread. You can't be bothered to answer that relevant question...

With noise shaping, we have dynamic range which exceeds the threshold of human hearing with 16 bits. You have posted charts which demonstrate that. Professional recording engineers record using DSD or 24 bit PCM. Professional recording engineers also use noise shaping when they dither down from DSD/24 bits to 16 bits when producing CDs. CDs are therefore capable of delivering all the dynamic range that is required by human hearing.

Do I think 120db of dynamic range is required for playback in our listening rooms? No. The background noise in the recording venue, the actual dynamic range of the music, the noise in the recording equipment from microphone to hard disk (or worse yet analog tape), the noise in the mixing and mastering equipment, which often includes multiple digital to analog to digital processes, the noise in the playback equipment, and the background noise in even the best listening rooms all mitigate against it.

What is the actual dynamic range required for a playback medium. That is determined by the levels of background noise in the recording and playback environments, the music itself, and the weakest link in the recording and playback chain. It varies from recording to recording, and system to system, but it is nowhere near 120db, and even if it were, CDs are capable of delivering it.

The file you analyzed and the file of the same recording Arny analyzed were both 24 bit; therefore, although they were of different sampling rates, they both have the same theoretical dynamic range (without noise shaping) of 144db. (Why HDtracks downsampled it to 96k instead of 88.2k is beyond me. Okay, it was for marketing purposes). Arny determined, and you have now agreed, that the actual dynamic range of the recording is in the range of 70db. It was recorded by one of the acknowledged master recording engineers of all time, one, who unlike most others, does not employ compression, which would have further reduced the dynamic range of the recording. Yet it is 50db shy of 120db. Although Arny alleges that he knows of recordings with greater dynamic range, I think you would be hard pressed to find them. If audiophile recordings made by master recording engineers only have a dynamic range of 70db, I think the hand-wringing over the necessity of 120db is misplaced.

What is the level of noise in your reference theater, which, having been designed by Keith Yates, is undoubtedly among the best - and quietest - listening rooms available? Subtract that number from 120db and let me know what you get.
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post #256 of 361 Old 05-25-2012, 11:48 AM
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Quote:
Originally Posted by audiophilesavant View Post

With noise shaping, we have dynamic range which exceeds the threshold of human hearing with 16 bits. You have posted charts which demonstrate that. Professional recording engineers record using DSD or 24 bit PCM. Professional recording engineers also use noise shaping when they dither down from DSD/24 bits to 16 bits when producing CDs. CDs are therefore capable of delivering all the dynamic range that is required by human hearing.

Who says "professional engineers" are using noise shaping? You have spectrum analysis showing that vast majority of titles have noise shaping? Or are you asking us to take your word for it?

Here is the UI for conversion of the track we are talking about to 16/44.1:



You see the default I have circled? No noise shaping? You think the typical creative type would naturally gravitate toward overriding the default and selecting something called "noise shaping?" How many do you think know what noise shaping is? Why didn't Adobe make that the default?

But sure, let's have you show us a sampling of spectrum analysis that you have done yourself to arrive at that conclusion or market research that indicates noise shaping is common in production of CD.

Quote:


Do I think 120db of dynamic range is required for playback in our listening rooms? No. The background noise in the recording venue, the actual dynamic range of the music, the noise in the recording equipment from microphone to hard disk (or worse yet analog tape), the noise in the mixing and mastering equipment, which often includes multiple digital to analog to digital processes, the noise in the playback equipment, and the background noise in even the best listening rooms all mitigate against it.

All of that layman logic has been debunked with the research papers I presented earlier: http://www.avsforum.com/avs-vb/showt...5#post22061565

The confusion consistently comes from assuming noise is noise. As Arny rightfully said, numbers can just be numbers. You have to correlate them to what we hear. And what we hear is not a flat frequency response. So single number noise dB means nothing. This includes the analysis we just did for the audio track, and your read of amplifier S/N. Do you know what a-weighting is? It is a simplified model overlaid on measurements to make them somewhat realistic relative to what we actually hear. This is what it looks like:



You see that shape? Now look at the threashold of hearing:



You see how the a-weighting takes down the amplitude of noise in low frequencies very similarly to where the ear is least sensitive?

I have even quoted the specific sections from research that talks about all of this so you don't even need AES subscription to read it or ability to find the right section and the point is still missed. Yet folks keep talking about "oh, but do we hear it?" Well to know that you better use psychoacoustics and not blind metrics devoid of that. And don't substitute your common sense for top research that has already done this.

Proper analysis of the high-res track we just did would have to analyze the spectrum of background noise and determine how much of it is in the mid-frequencies. That is what matters the most. The dynamic range there may be far higher. Simple ultrasonic (> 20 Khz) noise could have set the noise floor for us yet we know such things are not audible and there must be excluded. But the dumb analysis tool we have can't do that. This is why I put all the caveats there. Please don't confuse anecdotal info with real research data.

Quote:


What is the actual dynamic range required for a playback medium. That is determined by the levels of background noise in the recording and playback environments, the music itself, and the weakest link in the recording and playback chain. It varies from recording to recording, and system to system, but it is nowhere near 120db, and even if it were, CDs are capable of delivering it.

Well, you are clearly mistaken based on data in front of you. Go and read the Fielder papers and tell me where he didn't know what he was doing when he measured a) peak levels in live venues and recording equipment and b) how quiet ordinary and professional listening rooms are.

Quote:


The file you analyzed and the file of the same recording Arny analyzed were both 24 bit; therefore, although they were of different sampling rates, they both have the same theoretical dynamic range (without noise shaping) of 144db. (Why HDtracks downsampled it to 96k instead of 88.2k is beyond me. Okay, it was for marketing purposes). Arny determined, and you have now agreed, that the actual dynamic range of the recording is in the range of 70db.

No, I didn't agree with that. You didn't understand the technical explanation I provided not once, but twice. Audiotion reported *RMS* values. Those are averages. The DAC has to represent real voltage values at every instance in time, not an average value. That means peak values. You must back out the conversion from peak to RMS that Audition performed. Doing that gets us to 96 dB for sine wave which is he default in Audition. Arny says he has material that is quieter than that. Therefore with the data in front of us as of this moment, there is nothing that says 70 dB is enough. This is of course on top of other reasons why this is not the number to be trusted.

Quote:


It was recorded by one of the acknowledged master recording engineers of all time, one, who unlike most others, does not employ compression, which would have further reduced the dynamic range of the recording. Yet it is 50db shy of 120db. Although Arny alleges that he knows of recordings with greater dynamic range, I think you would be hard pressed to find them.

I would be? How would you know that? Based on what data? Gut feeling? Are you an industry expert with substantial experience in this area so we should trust your gut? And Keith released the tracks as 24 bit for what reason if he knows what he is doing and all that was there was 70 dB?

Quote:


If audiophile recordings made by master recording engineers only have a dynamic range of 70db, I think the hand-wringing over the necessity of 120db is misplaced.

Let's be clear: the 120 dB did NOT come from me. It came from top experts in the industry who presented papers at Audio Engineering Society based on real research and measurements. Maybe you have a record of them being laughed out of the room when they knew the reality was 70 dB but I don't. Here is the quote again from the research:

"Dynamic-Range Issues in the Modern Digital Audio Environment

The peak sound levels of music performances are combined with the audibility of noise in sound reproduction circumstances to yield a dynamic-range criterion for noise-free reproduction of music. This criterion is then examined in light of limitations due to microphones, analog-to-digital conversion, digital audio storage, low-bit-rate coders, digital-to-analog conversion, and loudspeakers. A dynamic range of over 120 dB is found to be necessary in the most demanding circumstances, requiring the reproduction of sound levels of up to 129 dB SPL. Present audio systems are shown to be challenged to yield these values."


In your next post, please don't address me. Address Fielder. Give us your credentials and body of data that would say he is mistaken.

Quote:


What is the level of noise in your reference theater, which, having been designed by Keith Yates, is undoubtedly among the best listening rooms available? Subtract that number from 120db and let me know what you get.

If I did that, it would show that you don't understand psychoacoustics. Would you really like me to go there? Here is the measurements from Fielder of consumer listening rooms:


You see that someone has built a quieter room than threshold of hearing. Importantly note how he has the spectrum of the noise and he compares it one for one against threshold of hearing. You cannot trust a meter with a dumb number thinking it means the audible noise level of the room. You would be ignoring the way our brain and ear work together. A rumble from a freeway five miles away at 18 Hz would register on that meter yet can be totally inaudible. The fact that it shows up at -60 db means absolutely nothing.

Let's cap this off with this: http://www.stereophile.com/content/m...r-measurements

"The No.532H is also superbly quiet; its wideband, unweighted signal/noise ratio, ref. 2.83V output with the input short-circuited, measured 93dB in both channels. (The A-weighted ratio was 105.4dB.) With its high output power, the No.532H is one of the few amplifiers that can match the dynamic range of true high-resolution recordings."

You see the a-weighting? You see how much higher it is than none, which at 93 db is 8 db higher than your assumed "85 dB?" You talk about spins. You think you did OK by changing a "> 85 dB" to 85 dB and calling it done?

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post #257 of 361 Old 05-25-2012, 12:13 PM
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You see the a-weighting? You see how much higher it is than none, which at 93 db is 8 db higher than your assumed "85 dB?" You talk about spins. You think you did OK by changing a "> 85 dB" to 85 dB and calling it done?

So spinning is quoting the >85db from the specifications published by Harman? For both the No. 532H and the No. 53.

You are quoting the measurement of the No. 532H that Stereophile measured. If Stereophile measured 93db, then maybe they got a better than average sample.

Perhaps you could explain why Harman published a >85db specification, instead of something like say, >86db, >87db, >88db...

Seems like JA is even stating that 93db (unweighted) "can match the dynamic range of true high-resolution recordings."

I'll ask again:

What is the level of noise in your reference theater, which, having been designed by Keith Yates, is undoubtedly among the best - and quietest - listening rooms available? Subtract that number from 120db and let me know what you get.
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The problem is that Audition uses this number to average the values it computes. It is a sliding window across the whole track. To know which number is right, requires knowing the precise nature of what you are averaging. If I had silence and then an impulse and I averaged the two because I used too long of an interval, clearly I am not seeing the right value for the lowest and highest numbers anymore.

There is 20 mSec of binary zeros at the end of the track we've been talking about. It is way out of place because it is preceded by what looks like room noise at about -70 dB. The jump from 70 dB down to more than 144 dB down in less than a millisecond is clearly artificial.


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The manual page for Audition explains the same thing: that the right value is content dependent. This is why I said this is back of the envelop stuff. The true analysis can only be done by the person recording the material or someone sitting there analyzing the track manually. I could get higher numbers if I shrunk the value below 20 msec but figured that was a good balanced position as a thumb in the air kind of thing.

Audition is a tool, and like any good tool the results depends on the skill of the person using it. One strength of its Statistics Tool is that the calculations are run over the area of the file you select, from just a few samples to the entire file.

Of course, you have to know which is real music and which is edited in, in this case probably by mistake or at least unintentional.

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Can you post some?

If memory serves I first posted this information at AVS in this post: http://72.9.159.100/avs-vb/showthrea...6#post15928896 The date is 02-27-09.

Check out the Beethoven Symphonies on the Swedish BIS label. I don't know about all of them, but at least one is exceptional - dynamic range around 85 dB. This one is perfectly compatible with the CD format, particularly if noise shaping is used as it should be on any modern recording with pretensions to high dynamic range.

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Per above, for the purposes of this discussion, we care about peak values, not RMS.

We are actually interested in the difference between the quietest passage that is perceptually meaningful, and the loudest.


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The DAC doesn't see RMS.

True, but the power amp's power supply and heat sinks do. ;-)
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Originally Posted by audiophilesavant View Post

Seems like JA is even stating that 93db (unweighted) "can match the dynamic range of true high-resolution recordings."

No, that's not what I stated. You are misunderstanding my S/N measurements, which are referenced to 2.83V, ie, 1W into 8 ohms. My statement that the No.532H's dynamic range capability matches that of hi-rez recordings is due to the fact that to that 93dB, you need to add another 25.5dB, the difference between 2.83V and 53.5V, the voltage at which the amplifier clips.

This gives the No.532H a dynamic range capability of 118.5dB, across the audioband, which is close both to Louis Fielder's 120dB and to 20 bits of resolution.

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So spinning is quoting the >85db from the specifications published by Harman? For both the No. 532H and the No. 53.

You are quoting the measurement of the No. 532H that Stereophile measured. If Stereophile measured 93db, then maybe they got a better than average sample.

Perhaps you could explain why Harman published a >85db specification, instead of something like say, >86db, >87db, >88db...

Seems like JA is even stating that 93db (unweighted) "can match the dynamic range of true high-resolution recordings."

Ooops!

One other hidden agenda is the fact that a power amp cannot be modeled as a fixed noise source with a variable signal. As the output signal goes up, the noise increases. Some of this is due to the fact that the ripple on the power amp's power supply increases dramatically as power goes up towards full unclipped output. At full output, power amp power supply ripple is often on the order of several volts.

This effect may have adverse effects on JA's post immediately before this one (post 259). His calculations appear to assume that the amps output noise is invariant with power level, and in general this is not true. Amp noise tends to increase with power output and may increase more rapidly as we approach the clipping point.

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I'll ask again:

What is the level of noise in your reference theater, which, having been designed by Keith Yates, is undoubtedly among the best - and quietest - listening rooms available? Subtract that number from 120db and let me know what you get.

Don't hold your breath waiting for that answer! ;-)
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No, that's not what I stated. You are misunderstanding my S/N measurements, which are referenced to 2.83V, ie, 1 into 8 ohms. My statement that the No.532H's dynamic range capability matches that of hi-rez recordings is due to the fact that to that 93dB, you need to add another 25.5dB, the difference between 2.83V and 53.5V, the voltage at which the amplifier clips.

This gives the No.532H a dynamic range capability of 118.5dB, across the audioband, which is close both to Louis Fielder's 120dB and to 20 bits of resolution.

John Atkinson
Editor, Stereophile

A technical question: is an amplifier approaching clipping when it is reproducing a signal that is -118.5dbfs?
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A technical question: is an amplifier approaching clipping when it is reproducing a signal that is -118.5dbfs?

My apologies. I don't understand your question. The topic was dynamic range. I was pointing out that an unweighted S/N ratio of 93dB ref. 2.83V was 118.5dB ref. the voltage at which the amplifier clipped and therefore would be incapable of amplifying the signal further without modifying its waveform. This is thus the amplifier's dynamic range capability.

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This effect may have adverse effects on JA's post immediately before this one (post 259). His calculations appear to assume that the amps output noise is invariant with power level, and in general this is not true. Amp noise tends to increase with power output and may increase more rapidly as we approach the clipping point.

I have only very rarely seen such noise modulation with amplifiers, provided they have an adequate power supply for the specified power. Amir referred to the Stereophile review of the Mark Levinson No.532H; if you look at fig.8 in my measurements accompanying that review - http://www.stereophile.com/content/m...r-measurements - which shows the spectrum of a 50Hz sinewave 2.6dB below clipping, you can see that the random noise components lie below -150dB and supply-related spuriae are below -125dB. There is no noise modulation worth mentioning.

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My apologies. I don't understand your question. The topic was dynamic range. I was pointing out that an unweighted S/N ratio of 93dB ref. 2.83V was 118.5dB ref. the voltage at which the amplifier clipped and therefore would be incapable of amplifying the signal further without modifying its waveform. This is thus the amplifier's dynamic range capability.

John Atkinson
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Well, it just struck me that if an amplifier was reproducing the really soft sounds close to the threshold of audibility, it would be using closer to 1W (or even less) where the SNR of the amp was 93db (or worse), in the case of the ML 532H you measured, rather than approaching clipping, where you say the SNR would be 118.5db, and as a result the noise of the amplifier might be in excess of the soft sound it was attempting to reproduce.
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My apologies. I don't understand your question. The topic was dynamic range. I was pointing out that an unweighted S/N ratio of 93dB ref. 2.83V was 118.5dB ref. the voltage at which the amplifier clipped and therefore would be incapable of amplifying the signal further without modifying its waveform. This is thus the amplifier's dynamic range capability.

Except that the above extrapolation may not reflect the true performance of the power amp.



The IM plot above suggests relatively large signals at low frequencies, but the frequency scale is so compressed at that point that it is impossible to resolve them.

I do not believe that the test report before us does more than suggest that nonlinear distortion and noise (mixed together) contains any power supply noise due to the heavy load on it.

This plot suggests rapidly rising THD+N at high powers well below clipping:



Is the rise due to nonlinear distortion, is it due to noise? Nothing in the test appears to answer this question.

JA has the equipment to study rapidly deteriorating SNR at high powers but he does not appear to exploit that capability to shed light on issues like dynamic range at high powers.
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You see the default I have circled? No noise shaping? You think the typical creative type would naturally gravitate toward overriding the default and selecting something called "noise shaping?" How many do you think know what noise shaping is?

Bit of a slur on both professional recording engineers and professional "creative types" in general don't you think. The dithering down from 24-bit to 16-bit occurs at the last stage of the mastering process by the mastering engineer. I am willing to give professional mastering engineers the benefit of the doubt that they know what they are doing and are not making rookie mistakes like truncating to 16 bits on the conversion. But perhaps your friend Bruce at Pugut Sound Studios would be in a better to position to speak to the technical knowledge of professional mastering engineers in general. Why don't you inquire of him instead of just broadly casting aspersions on the recording profession. But it really doesn't matter, does it, if the maximum dynamic range of the recording is in the neighborhood of 70db as was the case with Keith Johnson's 176.4/24 effort, or even the 85db on the BIS recordings arnyk referred to.

We both respect JA. I know he used noise shaping on his Stereophile recordings twenty years ago because he was using the long discontinued Meridian 518. Speaking of which, perhaps JA could enlighten us as to the maximum dynamic range he has achieved on the recordings he has done.
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I have only very rarely seen such noise modulation with amplifiers, provided they have an adequate power supply for the specified power. Amir referred to the Stereophile review of the Mark Levinson No.532H; if you look at fig.8 in my measurements accompanying that review - http://www.stereophile.com/content/m...r-measurements - which shows the spectrum of a 50Hz sinewave 2.6dB below clipping, you can see that the random noise components lie below -150dB and supply-related spuriae are below -125dB. There is no noise modulation worth mentioning.

I am having difficulty finding a textual or graphic reference that supports this claim on that web page. What I do see is is a figure 8 (apparently from a FFT) which only goes down to -140 dB. I am also aware of the fallacy of reading noise floors off of FFTs, because of the well known fact that they drop when the number of data points is increased arbitrarily..

http://www.analog.com/static/importe...gnal_Sect5.pdf

"The number of FFT points also determines the noise floor of the FFT with respect to the broadband noise level, and this may also be a consideration. Figure 5.25 shows the relationships between the system fullscale signal level, the broadband noise level (measured over the bandwidth DC to fs/2), and the FFT noise floor. Notice that the FFT processing gain is determined by the number of points in the FFT. The FFT acts like an analog spectrum analyzer with a sweep bandwidth of fs/N. Increasing the number of points increases the FFT resolution and narrows its bandwidth, thereby reducing the noise floor. This analysis neglects noise caused by the FFT
round off error."

John please provide better documentation supporting your claim or provide a better explanation of what you have provided...
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post #268 of 361 Old 05-26-2012, 08:16 AM
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Bit of a slur on both professional recording engineers and professional "creative types" in general don't you think. The dithering down from 24-bit to 16-bit occurs at the last stage of the mastering process by the mastering engineer. I am willing to give them the benefit of the doubt that they know what they are doing and are not making rookie mistakes.

Benefit of doubt? That is your back up for what you said and I quote, "Professional recording engineers also use noise shaping when they dither down from DSD/24 bits to 16 bits when producing CDs."

Not even one sample you have run on your own music to show this being there? No surveys of it being common practice? Just your hunch that "they know what they are doing?" Is this the standard by which we demonstrate our points? Hoping that an entire industry "know what they are doing" and therefore it becomes the foundation to make a technical argument here?

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But perhaps your friend Bruce at Pugut Sound Studios would be in a better to position to speak to the technical knowledge of professional mastering engineers in general.

If you believe in his expertise and you should, then you should go and ask about such things before asserting them to then go on to tell us that CD is routinely preserving 120 dB of dynamic range because professional "know" to use noise shaping.

As to me needing to ask him, I don't. I have read the research and I have debated the point with recording engineers who think anything called "noise" must be a bad thing. No amount of explaining the science goes anywhere. And lest you think subjectivists only live here, they also live there and they will often tell you adding dither made the sound worse, not better. And there are endless debates as what to add. Here is some random googles into their forum: http://www.gearslutz.com/board/post-...-question.html

"Okay, question about something that I feel like I should know but I don't. I understand dithering and noise shaping, but my question is about the use of noise shaping without dithering. The file is going to be 16bit wav in 16bit wav out so I don't need to dither, but I was wondering what effect noise shaping could have without dithering. In theory, could noise shaping lower the perceived white noise present in the raw file? Or would noise shaping cause more problems? Thanks!"

Translation: he thinks noise shaping is for shaping noise that is already in his recording! Not as an aide to lower the distortion that gets added when you reduce bit depth.

http://www.gearslutz.com/board/maste...at-dither.html

"I know it should be the contrary, but in my experience noise shaped dither is more audible then TDPF dither."

"Noise shaped dither is just asking for trouble. Who really thinks that the product is always going to be played back on a flat system? No, its often played back with the treble boosted."

"Doesn't noise shaping add high frequency noise that is capable of distorting certain playback systems? If that's the case, then I would rather just go with flat dither noise. I don't mind a little subtle noise in my recordings."

"Leave the noise-shaping off, and personally I would focus my energy (Har!) elsewhere."

"If you think it may go through another process, then TPDF dither with no noise shaping is probably a good bet, since it results in pretty well white noise. If it's for final delivery, just listen, pick one you like, and stop worrying about it."

'Just experience ... every dither option sound different so if you are happy with your master before dithering you should take 2-3 minutes to decide and test which one you would like to use on the given material. It does affect the sound though of course this is a very little step in the whole process"

And the final post:
"If, as you say, you like your sound before dithering, the dither that will have the least effect on that sound will be plain old flat vanilla TPDF. That's why many mastering engineers use it more often than the shaped dithers."

So no, what you said has no foundation. It is a prayer. You also keep confusing the research to mean 16 bits = CD. That is not the same thing. CD has a limited bandwidth of 22 Khz. For noise shaping to work well, you want to push the noise well into ultrasonics so that we know we can't hear them and they readily get filtered out. Pushing them in frequencies above mid audio band as the previous graph showed is not the most comfortable place to put it. That is why the research paper that produces that graph, has the recommendation of 58 Khz as the lowest sampling rate (among other reasons). As a minimum you like to have 48 Khz. This *may* explain the dissatisfaction some of the Pros have with regards to its audibility.

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But it really doesn't matter does it, if the maximum dynamic range of the recording is in the neighborhood of 70db as was the case with Keith Johnson's 176.4/24 effort?

Didn't I already explain that? That you can't use the 70 dB because it is averaged value and not actual voltage samples that get us dynamic range? And how it is a single dumb number that is blind to spectrum? Yet it is the spectrum that we need to determine how audible the noise is? Didn't I explain that if you had a 33 Khz ultransonic noise that was high in level it would set the "minimum" noise level yet it is totally inaudible and hence has to be taken out and that tool cannot since it uses blind computation of sample values?

How do you expect this discussion to go forward when I can't communicate with you using basic engineering terms involving the difference between RMS and peak values?

Your argument style is random assumptions that are convenient to your cause that you can't substantiate which you then turn into my homework ("You go and ask Bruce"). And continuing to repeat the same incorrect technical points even though they have been addressed. And not just by text here but with research papers. I suggest leaving these arguments to Arny. You are losing more points than gaining .

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Benefit of doubt? That is your back up for what you said and I quote, "Professional recording engineers also use noise shaping when they dither down from DSD/24 bits to 16 bits when producing CDs."

Not even one sample you have run on your own music to show this being there? No surveys of it being common practice? Just your hunch that "they know what they are doing?" Is this the standard by which we demonstrate our points? Hoping that an entire industry "know what they are doing" and therefore it becomes the foundation to make a technical argument here?








How do you expect this discussion to go forward when I can't communicate with you using basic engineering terms involving the difference between RMS and peak values?

Your argument style is random assumptions that are convenient to your cause that you can't substantiate which you then turn into my homework ("You go and ask Bruce"). And continuing to repeat the same incorrect technical points even though they have been addressed. And not just by text here but with research papers. I suggest leaving these arguments to Arny. You are losing more points than gaining .

Quoted for future reference.
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Didn't I already explain that? That you can't use the 70 dB because it is averaged value and not actual voltage samples that get us dynamic range? And how it is a single dumb number that is blind to spectrum? Yet it is the spectrum that we need to determine how audible the noise is? Didn't I explain that if you had a 33 Khz ultransonic noise that was high in level it would set the "minimum" noise level yet it is totally inaudible and hence has to be taken out and that tool cannot since it uses blind computation of sample values?

If it is not 70db, what is the dynamic range of the recording in question?

As for the technical knowledge of professional recording engineers, you seem perfectly content to assume that they have the technical know how to achieve a recording with a 120db dynamic range; why am I not allowed to assume that they have the technical know how to apply dither when they convert from 24 bits to 16 bits? Why don't we just assume that the "creative types" don't have the skill to achieve 120db dynamic range.

Let me rephrase your statement and ask of you:

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Not even one sample you have run on your own music to show this being there? No surveys of it being common practice? Just your hunch that "they know what they are doing?" Is this the standard by which we demonstrate our points? Hoping that an entire industry "know what they are doing" and therefore it becomes the foundation to make a technical argument here?

Not even one sample you have run on your own music to show [noise shaping dither not being applied in a commercial recording converted by a professional mastering engineer from 24 bits to 16 bits] being there? No surveys of [professional mastering engineers not applying noise shaping dither] being common practice? Just your hunch that "they don't know what they are doing?" Is this the standard by which we demonstrate our points? Hoping that an entire industry "don't know what they are doing" and therefore it becomes the foundation to make a technical argument here?

As to who is "losing points", I leave that for the reader to decide.
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