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Measuring the "sound signature" of two different integrated amplifiers.

GXAlan

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@KSTR figured out that the left and right channels were swapped during recording! Presumably this must have been the case when listening too! The calculations had been done in stereo.

This changes the numbers and shows smaller changes between amps when fed directly but still has a change when the tape loop is fed through.


See update after reading this first post: https://audiosciencereview.com/foru...rent-integrated-amplifiers.37963/post-1342936


-----------------------------
Measuring the "sound signature" of two different integrated amplifiers.
Extraordinary claims require extraordinary evidence.

I have two high-performance integrated amps. The identities of these amps are not important and both reach "blue region" status for 1 kHz test tones for SINAD at 5 watts/4 ohms. I am presenting these measurements to showcase how @pkane's Deltawave tools can be used to correlate subjective listening impressions with actual measurements/science. Thank you @pkane for reviewing my methodology.

1. Background
When comparing two integrated amps at my normal quiet listening levels, I felt as if one system had better "attack" or "PRAT" to the music and I wanted to try to measure it. My subjective impression is what started this process. I had no measurements beyond sighted bias of having two units with very high 1 kHz tone performance. The TLDR is that I was successful in proving that differences can be measured and it turned out to be some sort of volume compression for transients that would be within the threshold of audibility.

Music on test: Concerto symphonique No. 4 in D minor, Op. 102: Scherzo from SFS0060 - Masterpieces in Miniature (physical SACD)

System A: SACD player -- balanced--> Integrated Amp 1
"subjective greater attack to the piano notes"

System B: SACD player -- balanced--> Integrated Amp 1 --> tape out --unbalanced--> Integrated Amp 2

System C: SACD player --unbalanced--> Integrated Amp 2.

Recording system:
E1DA Cosmos ADC Grade A with 4.48 ohm Dale Vishay 1% NH-250 resistors, 32-bit/176 kHz
Windows Laptop on battery power for recordings

Classical musical content was used for measuring the sound signature.

2. Matching the Volume

Using System B, I attached real bookshelf speakers (~87 db/2.83V) and set the volume to the actual level I prefer listening to. It's a small room and this was around 68-70 dB at the listening position for the soft portions of the music. Normal piano practice is quoted as 60-70 dB. System B was chosen as the reference volume setting since Integrated Amp 2 has no display indicating the volume, so once it's set, it's set. I then detached the speakers and replaced it with an E1DA Cosmos ADC setup and made a recording of the first minute. I switched to System A with real speakers, set to the same volume by ear, made a recording and then evaluated peaks at various frequencies from Audacity (not just 1 kHz). The volume difference was not linear in Audacity and then based upon my best estimate, I adjusted the volume of System A to get proper volume matching to System B.

System A and System B were matched to 0.056 dB for the SPL peaks in analog (prior to any digital correction).

3. Validation of test environment precision

System A was measured twice, one day apart.
System B was measured twice, one day apart.
System C was measured twice, 3 hours apart.
Deltawave was used to compare the run-to-run variability, trimmed at start and end of the music for the middle ~50 seconds of analysis.

The PK Metric was created by Paul K, the inventor of DeltaWave, and is designed to "more directly answer the question of whether the difference between two devices is likely to be audible or not."

These results show that the test environment (the amplifiers being tested, the cables, the ADC, etc.) were all very precise and are able to generate reproducible results.

PK Metric showed very high precision of the test environment
System A repeated one day later. -120.4 dbFS
System B repeated one day later. -117.2 dbFS
System C repeated 3 hours later. -117.8 dbFS

1664785774338.png


1664785798767.png

1664785815914.png




4. System A vs. System B
Initial peak values Reference: -28.017dB Comparison: -28.073dB
Initial RMS values Reference: -50.67dB Comparison: -49.022dB

Final peak values Reference: -28.017dB Comparison: -29.707dB
Final RMS values Reference: -50.67dB Comparison: -50.647dB

Recall that I matched volume in analog by making a recording and then using the Peak DB tool for various frequencies in Audacity. That's the initial value. I got my peaks within 0.056 dB of each other yet the RMS values were off by more than 1.5 dB. DeltaWave digitally corrects the level based upon RMS and got the two recordings to <0.03 dB matching. In doing so, the peak values are different by about 1.7 dB.

The differences in volume are non-linear. Sometimes System A (blue) is louder than System B (white) and sometimes it's not. It's not consistent to a single channel either.
1664785832905.png

1664785848819.png

1664785879173.png



Subjectively, I liked System A better because it had better "attack." When analyzing the two recordings, we see that
System B actually has a subtle compression effect relative to System A
or
System A was adding artificial impulse relative to System B.


The opposite could be true. System A could be adding artificial volume to impulse. We don't know what is actually more correct. It just shows that there is a measurable difference between these two systems in a manner that makes sense that it could be audible.

The PK Metric is -48.6 dBFS.

5. System A vs. System C

System B has the limitation of running through the tape out of the integrated amplifier in System A. Is the compression effect from the circuitry in the tape out, or from the amplifier? To answer this question, the SACD Player was connected directly to Integrated Amp B. The resulting recording was quieter and so the volume knob on Integrated Amp 2 was raised to attempt to match the volumes. These recordings were a day apart!

Initial peak values Reference: -28.017dB Comparison: -28.571dB
Initial RMS values Reference: -50.605dB Comparison: -49.532dB

Final peak values Reference: -28.017dB Comparison: -29.563dB
Final RMS values Reference: -50.605dB Comparison: -50.443dB

The matching wasn't as good in the analog realm with the peak values differing by around 0.5 dB. However, once Deltawave corrected the RMS values to <0.2 dB deviation, the peaks are still >1.5 dB different.

The tape out circuitry was not responsible for any of the perceived compression effect.
The PK Metric is -48.6 dbFS.


1664785897328.png

1664785919943.png


6. System B vs. System C
The difference between B and C is the tape loop. We have the same integrated amplifier being used in both. Once the two recordings were calibrated by DeltaWave, the RMS was ultra-precise at 0.004 dB and the peaks really aren't different (<0.01 dB). These recordings were a day apart!

Initial peak values Reference: -28.073dB Comparison: -28.571dB
Initial RMS values Reference: -49.02dB Comparison: -49.532dB

Final peak values Reference: -28.073dB Comparison: -28.137dB
Final RMS values Reference: -49.02dB Comparison: -49.024dB

Going straight from the SACD player to an integrated amplifier versus having a tape out in the signal chain had very little difference subjectively or objectively. The PK Metric is -93.0 dbFS.


1664785943759.png

1664785959331.png


7. Conclusions
"Attack" isn't a precise description. "PRAT" (pace, rhythm and timing) isn't a precise description. These are simply words to describe subjective experiences where we like one audio system more than another and cannot articulate the differences with more precise language. What you consider "PRAT," I might consider "attack."

But what we can agree upon are volume differences and how differences in volume that are non-linear can change the sound. I won't be able to convince everyone that I heard these differences between the two amplifiers. I'm sure there are those who will say these measured differences are not audible. All I will say is that I heard a difference which is why I embarked on this test and my original "matching volume by ear" got me to nearly to ~1 dB matching which I attribute to the extensive formal training I've had as a classical pianist.

I won't be able to convince everyone that I ran these tests perfectly or setup each component in the chain to its very best. This is why I have separated these as "systems" rather than naming specific components. Running the E1DA Cosmos without optimized gains should actually make it harder to detect differences between the systems. The E1DA Cosmos does not have an input buffer, and maybe that makes a difference from one system to the other.

I won't be able to convince everyone that my selected music or preferred volume is universal.

I'm not trying to convince you that these differences are the most efficient/meaningful uses of your money. There is clear consensus that speaker placement/furniture placement is the best ban for the buck (free), and when it comes to gear, speaker/subwoofers make the biggest impact to sound.

Anything that can be heard can be measured.
Maybe 30-40 years ago, the test equipment wasn't good enough to capture everything audiophiles thought they could hear. In 2022, hobbyist level ADCs are so good that you should expect/demand claims to be backed by measurements. I very easily can substitute my term "attack" with "PRAT." We have seen time and time again that a lot of audiophile tweaks prove to be useless.

Everything measured cannot be heard.
We're still human. The PK Error Metric is something I just learned about this week. If someone says they can hear something -300 dB away from reference, that's not really believable. Or better stated to be generous, that difference is not going to be meaningful to you unless you have a genetic mutation allowing you to hear something most humans cannot. The threshold for audibility of the PK Error metric has been stated as -50 dB and my reported differences met this threshold.

Two high-performance amplifiers seem to have measurable differences in one real-world condition that also meets the threshold for audibility.

Both of these amps are rated into the triple-digit watts into 4 ohms and I was pretty much running ~50 milliwatt to ~5W peaks given that the volume was originally set with an ~87 db/2.83V stereo, 5-6 ohm speaker pair near a wall, 7 ft listening distance, resulting in 68~70 dB RMS with presumably 90-92 dB peaks based upon the analysis of the recordings.

Consider adding Track 1 from Masterpieces in Miniature when testing gear.
It's the kind of classical music that can bring a smile to aficionados of the symphony while still being a piece that can be appreciated by those who rarely listen to classical music. I was very surprised that I thought I heard a difference on this track between two integrated amplifiers and maybe this happens to be a very good test.



EDIT: And for comparison…

EDIT: And now revealing what’s behind the curtain.
https://audiosciencereview.com/foru...rent-integrated-amplifiers.37963/post-1341620
 

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staticV3

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Nice work! Sadly, your graphs don't load for me
Screenshot_20221003-102108.jpg
 

dualazmak

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For me too, the php file cannot be seen. You would better to past all the diagram images in jpg format, I assume.
 

restorer-john

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The tape outs in 99% of amplifiers are merely the switched signal from the inputs, via a rec-out selector, or the source switch itself. Most integrated amplifiers and preamplifiers will pad the tape outs via some resistors of varying values. Some (very few like the Holman, some expensive Yamahas etc) will place active buffers (opamps) specifically for the tape rec outs in order to drive a low impedance and not load down the main signal. You could have any combination of the above.

After the tape outs, you have your balance, volume, (buffer) tone (if any) and perhaps another buffer stage before the signal passes to the power amplifier stage.

In short, testing anything, or sourcing anything via the tape outs should be limited to Phono stage testing only (where you are testing the RIAA active stage before the tape out). No conclusions can be drawn where tape outs are used on line stage signals as compared to the full path to speaker outs.

Why don't you level match electrically at the speaker terminals of each amplifier with a DMM?
 

McFly

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Yeah I don't get the point of the tape out but very interesting results. Were both integrated amps capable of the same power into 4ohms? I would guess not... System A looks more dynamic. I'm dying to know what the amps are. I'm with John, level match the amps with voltage into your loads and go again.
 

-Matt-

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Why don't you level match electrically at the speaker terminals of each amplifier with a DMM?
I'm also a bit concerned about the level matching.

Maybe I didn't follow your procedure exactly but I think you made recordings (that differed somewhat in peak and rms levels) then used software to normalise the peaks. Is that correct?

If so, I would argue that we shouldn't rely on software normalisation, the actual level of the recordings will still be important. My concern would be that the recordings have used a different amount of the dynamic range available in the recording system. Taking this to the extreme cases (not saying this applies here)... if the level is too high for the recording system then peaks will be clipped - this will exactly result in nonlinear compression. If the recording level is too low you will make a lower bit depth replica of the waveform (possibly with some details falling below the noise floor).

I'd say great care in level matching (with electrical checks as suggested by @restorer-john) would be a good idea to avoid any chance of such differences in the dynamic range of the recordings. Some additional details on this part of the procedure would be most welcome.
 
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GXAlan

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Yeah I don't get the point of the tape out but very interesting results. Were both integrated amps capable of the same power into 4ohms? I would guess not...

System B and System C are identically with and without the tape out.

This is just how I had it set up for convenience at the time I initially was doing my listening tests and once I got the volume matching to a crazy 0.0056 0.056 dB, I dared not mess with anything until I had a full set of tests. System B is measuring the combination of the tape out (the exact opposite of "shortest signal paths") and it's nice to see that the really convoluted signal path does make an audible difference. The B vs. C has a PK metric around -90 as opposed to -120 so there is a difference -- just not audible. It's also nice to see how that measured difference is inaudible and cannot be seen while the other is.

System A is advertised as having more power.

I'm also a bit concerned about the level matching.

Maybe I didn't follow your procedure exactly

Not at all. The level matching INITIAL = actual matching. FINAL = software normalization

I was matched to 0.056 dB without any software modification for System A vs B.
The software normalization got it tighter to 0.023 dB matching by focusing on RMS.


Both the raw measurement and the software normalized data show that RMS vs peak is different for the two.
 
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McFly

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The E1DA should be capable of 43Vrms peaks, if set as such, would one amp have been hitting that here, I doubt it.
 
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GXAlan

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The E1DA should be capable of 43Vrms peaks, if set as such, would one amp have been hitting that here, I doubt it.

That's right. You can see that the measurements that I was at -28 dB, so I had plenty of headroom. The danger is that detail is buried into the noise of the E1DA but that is good for this comparison because it will make it harder to show a difference.
 

restorer-john

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With amplifiers or preamplifiers, the volume should be set at maximum or in the case of a digital (0dB).

The input signal should be adjusted in level to obtain the output level in volts you want to test at. The connected load should be identical for both amplifiers.

The volume control/s should not be used to match levels, as the response and performance of pretty much all conventional amplifiers/preamplifiers varies massively across the volume pot range. Both FR and noise vary and there is pretty much always a maximum and minimum noise point on a pot rotation or encoder/digital level control. That's why maximum volume is always specified and used by all manufacturers for their main specifications.

If that means you have to split your source signal and pad it down via a few 10turn pots to get the outputs right, so be it. You then have a reference.

1kHz sine should be used as it has been the reference frequency since Adam was a boy. Normalize to that.

Once you have the front end set up, you can play music, tones, whatever, with or without loads or speakers and obtain useful data to throw into the software. The sampling must be done from the speaker terminals- forget tape outs or pre-outs- nothing remotely useful comes from using line stage tape outs as a reference source.
 

-Matt-

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Not at all. The level matching INITIAL = actual matching. FINAL = software normalization

I was matched to 0.056 dB without any software for System A vs B by focusing on PEAKS.

The software normalization got it to 0.023 dB matching by focusing on RMS.
Thanks for the clarification.

Edit: Wait, how many zeros? In some places you say 0.0056 and others 0.056 dB.
 

McFly

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With amplifiers or preamplifiers, the volume should be set at maximum or in the case of a digital (0dB).
Not applicable with this test as source is SACD player, unless it had variable out, and still you'd need a disc with 1khz 0db on it to calibrate. But yes, could make source a dac w vol control and rip the cd to achieve
 
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GXAlan

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With amplifiers or preamplifiers, the volume should be set at maximum or in the case of a digital (0dB).

The input signal should be adjusted in level to obtain the output level in volts you want to test at. The connected load should be identical for both amplifiers.

The volume control/s should not be used to match levels, as the response and performance of pretty much all conventional amplifiers/preamplifiers varies massively across the volume pot range. Both FR and noise vary and there is pretty much always a maximum and minimum noise point on a pot rotation or encoder/digital level control. That's why maximum volume is always specified and used by all manufacturers for their main specifications.

If that means you have to split your source signal and pad it down via a few 10turn pots to get the outputs right, so be it. You then have a reference.

1kHz sine should be used as it has been the reference frequency since Adam was a boy.

Once you have the front end set up, you can play music, tones, whatever, with or without loads or speakers and obtain useful data to throw into the software. The sampling must be done from the speaker terminals- forget tape outs or pre-outs.

That's the thing. This wasn't about testing the gear at specific volts. It was me sitting down to listen to music in my office on one system, then testing an amplifier my brother sent me and listening to the same music and then hearing a difference and being very surprised.

All sampling was done from speaker terminals.

Literally, this is what a real person would do. Attach a CD/SACD player to an integrated amp, and then a set of speakers and then they'd listen to the music at the level they want to.

System A and C are exactly that scenario.
System B is the weird one, but once I had volumes matched at 0.056 dB I dared not change anything since there was a real risk I'd never be able to match it again. It was just how I hooked up the amp when I first tried it.
 

dualazmak

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Thanks for your fixing, now I (we) can see all the diagrams!
 

-Matt-

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The matching wasn't as good in the analog realm with the peak values differing by around 0.5 dB. However, once Deltawave corrected the RMS values to <0.2 dB deviation, the peaks are still >1.5 dB different.
In one place you lost a further factor of 10. I assume there may be some typos. Would be good if you could check through the op and correct if necessary.
 
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GXAlan

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I think the key is why I am calling this a system and not naming the gear.

@restorer-john is exactly right.

I don't know why there's a difference. There is a difference that is reproducible and as far as I can tell, accurately determined.

@restorer-john has brought up the point that maybe I'm just seeing a difference is in the volume control since he points out that there are differences in noise, FR, etc.
I don't know how the volume control is implemented in these two setups.

But as I said. This is how an ordinary person would listen to music. Disc player -> int amp -> speaker.

No variable out on the disc player, I can only turn the knob to the level that sounds right to me :)
 

Sokel

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I love real world measurements,just love them!
Thank you for the time and effort!
 
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