A listening test of two power amps


Hello, 

It's my first post here. I've been using two power amp setups for my main stereo and I've been curious to see if I can really discern any acoustic difference between the two. One setup involves a bi-wired high-powered stereo power amp and the other uses a pair of identical lower-powered amps with which the speakers, a pair of Tannoy System 12 DMT II monitors, are vertically bi-amped.

I decided to devise a listening test involving a mono acoustic recording made with a valve-condenser mic positioned at my usual listening position. I've used a relatively simple method to ensure that the recordings are level-matched. I've chosen a mono recording method since my goal is, principally, to evaluate the "tone" of the two recordings. I've been inspired to do this test after reading W. A. James' eBook "High end audio on a budget". The aim of the listening test is to try and discern which power amp setup provides the most realistic rendering of acoustic instruments. I thought that a mono recording might help the listener concentrate on the tone. After listening, I think it does. It's less distracting, especially on piano, where stereo or other multi-mic recording setups tend to splay out the notes across the stereo field.

I made two recordings for the test and will place links below so that the audio can be downloaded. I won't at this point give the make and model of the power amps involved, but this is the method used:

Method

1. I created an audio file with white noise at -10dB RMS and put the file on a Logitech Media Server so that I could play it on my stereo using a Raspberry Pi 3 with Audio Injector Pro card and RCA interface (192kHz 24bit DAC).

2. I then put on an LP on a Pro-ject 1.2 and set the volume to my usual listening level on a Quad 34 preamp. Following this, I then played the white noise and used a decibel meter, positioned next to the mic, to measure the level. It measured 67.3 dB.

3. Still playing the noise, I set the record level on a portable Tascam digital recorder arbitrarily to somewhere above -15dB. The microphone used was a large diaphragm valve condenser mic. The Tascam was set to record at 192kHz 24bit.

4. I then recorded the first track of the LP on the Tascam.

5. After that, I wired up the other amp configuration. I played the white noise and adjusted the volume of the preamp such that the decibel meter again measured 67.3dB at the position of the mic. The volume control on the Quad 34 is stepped, so I was lucky it matched!

6. I then recorded the same track on the LP as before, leaving the Tascam record levels unchanged.

7. I tidied the two recordings in Ardour (trimming start and finish only) and exported each as a 192kHz 24bit Flac file. I did not adjust the gain on either recording.

8. I listened to the recordings on the computer with a pair of AKG K501 headphones and Focusrite Scarlett interface.

Results

At first, I could distinguish a marked difference between the two. But now, I'm uncertain of the first qualitative difference that I'd noticed but I have noticed other more subtle differences (for the moment anyway). And that's why I'm here!

It would be wonderful if some people here could listen to the recordings and say which recording produces the most realistic rendering of the three instruments therein, and why. The instruments being piano, drums and string bass.

I've given the two files nondescript names: e.flac and t.flac. If anyone needs a different format or for me to down-sample, please let me know.

Finally, here are the files:

https://escuta.org/webtmp/e.flac

https://escuta.org/webtmp/t.flac

Cheers,

128x128surdo

Normalizing should be done to peak amplitude, not RMS. Both of your files have a peak amplitude that is clipped at between 1’08”  and 1’09”. I generally normalize to -1.0 dB of peak. I used values in your original samples to come up with the values I used when initially normalizing them since I wanted to add a little variable as I could to the samples. 

The rest of the values from the waveform indicate the min and max of both the RMS absolute dB are very closely matched.

Haven’t listened to them yet, but noticed that the sonogram for the Quad has some strong tones around 8KHz that is much less in the Sunfire.

@budjoe Thanks again for your help with this. Yes, the last set of recordings was done with the same cables, speakers (Tannoy), preamp (Quad 34) and recording device. Only the power amps were changed.

I wonder if normalising the peak values is the right way to go. I imagine that the two amps will potentially render peaks quite differently. And if that’s the case, wouldn’t that produce audio mismatched for loudness and with the background noise exaggerated in the recording with the lower peak?

I’ve gone back to my Tannoy/Quad34 recordings and normalised the files, constraining the RMS to -5.0dB. There was no clipping and the two files sound reasonably equal in loudness to me in headphones.

Yes the Quad recording has noise and a thin, grainy sounding hum, but the hum of the Sunfire is louder and deeper, to my ears at least. On the Quad the hum is louder in the right channel. On the Sunfire the hum is louder in the left but more balanced.

All the same...

More important to me than the noise, is the distortion of the piano sound in the Sunfire recording. Are you (or anyone else) able to hear this distortion?

Here are the two files normalised for loudness:

https://escuta.org/webtmp/QuadPiano.flac

https://escuta.org/webtmp/SunfireVoltagePiano.flac

Cheers,

 

In the samples I sent to you, the waveforms were normalized to -3.0 dB, then the zoomed portion was normalized to -0.3 dB.

In the zoomed section, the noise on the quiet part of the j-piano-zoom was very similar on both the left and right channel. Some high frequency noise, but very low level. In the p-piano-zoom, the left channel (top) has almost no high frequency noise, but the right channel has the most noise of all the waveforms. That is what leads me to believe there is something else going on.

Connections, grounding, or impedance, or some problem in one of the amps. I can’t say from here. I think you should explore some tests with the minimum number of components in each chain. Try with all components in the same power circuit. If any component used has only a 2 conductor power cord, try turning that cable over to see if it changes the waveform and/or the sound. Swap left and right channel interconnects one at a time and see if the noise moves from one channel to the other. Swap the speaker wires from left to right again to see if the noise moves from one channel to the other.

From what you are hearing, and the wave forms; i don’t think you need to have any input to chase the issue. I would use headphones at the amplifier headphone jack (if there is one) and with just the minimal signal path connected and no audio playing turn up the volume and listen just for the noise and try to swap interconnects and speaker cables to look for any changes. You can add in your recording set up to see if the noise you are hearing shows up in the waveforms. That’s how I chase noise.

@surdo "The Sunfire’s speaker outputs produce a low level, but audible on headphones, 60Hz hum (or a harmonic of 60Hz). The Quad, while perhaps more noisy (with hiss) has no 60Hz tone. I tested both the Sunfire’s unbalanced inputs with the Quad 34 preamp and the balanced inputs with the sound coming from a mixing desk. Both produce the hum and the amp produces a hum even with no inputs connected and with the laptop, that’s monitoring the terminals, running on batteries. I’ve tried disconnecting the Sunfire’s earth and reversing the pins, but the hum is unchanged.”

Just went back and read this part again. I don’t see much 60Hz on any of the waveforms, but as I said, there is 120Hz on all of them. Using just the Quad 34 preamp and connected to the 2 amps and no input look for the noise. I am also not clear if your tests were done with 2 different sets of speakers and speaker cables, but to compare the amps, it should be done with the same speakers and speaker cables from either amp. If that is not the case, the speakers and cables cannot be separated from what the amps are doing.

 

budjoe, I normalised the peak of the j-piano and p-piano recordings to -0.3dB (which is what you did?). The above is Ardour’s spectrum analysis of the 1st note of the two recordings. The j-piano recording seems to have a lot more noise. The p-piano part seems to have stronger high frequency transients between 1 and 5kHz.

Also, just above the 1st orange band above 100Hz, there’s what seems like a pulsing partial that’s not present in the p recording. This was evident before normalising both recordings and in which there was less apparent noise but still very weak upper partials (between 1 and 5kHz) in the j recording.

 

"I played both the normalized and originals thru 2 different playback apps and my observations hold across both the original and normalized thru both apps. If you want the full analysis values from the originals and the normalized files, I will send them in a direct message. Let me know."

Thanks a lot budjoe. I really appreciate your comments. Yes, please send the analysis. I’m interested to see exactly what you’re looking at.

"Were listening to these files through the speakers or headphones? Possible you were hearing issues with the headphones and possibly being overdriven?"

I don’t think the phones were overdriven. I hear this sound through the speakers and on headphones (AKG K501 and K246). What I don’t hear on the speakers is the 60Hz buzz. This I only hear when "tapping" the speaker output of the amp and listening on headphones, either directly or on a recording. Although that doesn’t necessarily mean that the sound is not there - there’s a bit of other noise here always.

The more I listen to the sound, the more it sounds like a type of ring-modulation of the piano and I wonder if some sort of cross-modulation is occurring. If this is actually the case, there would be additional side bands appearing in the analysis and not necessarily a distortion.

Listening now to the 3 piano files on phones and at the same volume, p-piano.flac does not have this ring-modulation sound in the first half of the recording, but y and j certainly do.

The original recording is of tracks 8 and 9 ("Var. V. Lento" and "Var. VI Poco movendo") of Alessandro Simonetto’s album "Erik Satie: Works for Piano". I streamed them into a DAW from Qobuz and used this recording for the three amp tests.

Just read your post about distortion on the j and y piano files. I think you may have something else going on. Looking at the wave form of the j and p files, there is almost no ripple in the quiet parts on either file. Certainly not enough to be audible. (Wish I could record something from lp to digital with that low a signal, but maybe your source was not lp). I pushed those files (j and p) to the -0.3db normalization and again see almost no ripple in the quiet parts and very little difference between the files.

Were listening to these files through the speakers or headphones? Possible you were hearing issues with the headphones and possibly being overdriven?

Listened to all nine as originally provided, and with a modification (more on that later).

In general across the 3 different versions of each I liked j the most, y second, and p the least. Differences are slight. I like the attack on j and y the best, and think that p tends to round the sound off more; both on the attack and the decay. Most differences showed up in the trio recordings where the lead guitar has a much more natural sound to me. The plucking of the strings and the resonance of them is better and more natural. Also most noticeable in the voice where her voice cracks in the beginning at ”and you know darn well”.

Had a hard time telling much difference in the piano samples except that once again I felt the p was more rounded. That is the only way I know how to describe it. Bass was also very hard to tell across all samples. You may need someone with better sense of piano and bass to hear the differences in those samples.

Here is the more later.

I loaded all the samples in an app that lets me analyze the waveform, and also see there that the j and y samples are very close to each other, and the p ends up being pretty close to 1 dB lower in both average RMS, max RMS, and peak amplitude. across all samples. Within that app I normalized all the samples. Used -3.0 dB peak for the voice and piano and -0.3 dB peak for the trio as that was the max peak in the originals.

I played both the normalized and originals thru 2 different playback apps and my observations hold across both the original and normalized thru both apps.

If you want the full analysis values from the originals and the normalized files, I will send them in a direct message. Let me know.

 

The problem is in the recordings j-piano.flac and y-piano.flac which are recordings of the Sunfire’s "Voltage source" and "Current source" speaker outs respectively. The piano, in its quieter passages, produces a distinct distortion, especially on certain notes. The recording p-piano.flac, that of the Quad 303, produces no such distortion.

I have a theory on what the problem is:

The Sunfire’s speaker outputs produce a low level, but audible on headphones, 60Hz hum (or a harmonic of 60Hz). The Quad, while perhaps more noisy (with hiss) has no 60Hz tone. I tested both the Sunfire’s unbalanced inputs with the Quad 34 preamp and the balanced inputs with the sound coming from a mixing desk. Both produce the hum and the amp produces a hum even with no inputs connected and with the laptop, that’s monitoring the terminals, running on batteries. I’ve tried disconnecting the Sunfire’s earth and reversing the pins, but the hum is unchanged.

My theory is that when the piano is played softly, certain notes seem to interact with the 60Hz hum, producing what seems like distortion, so if the hum can be removed, perhaps that distortion will disappear.

Is some kind of a mains filter likely to solve the problem or is there something that can be done to the Sunfire to make it less susceptible to this 60Hz hum?

And yes, I do hear the distortion through the speakers. I had noticed this sound before on occasions, but since I was using the Sunfire in my studio, I always suspected the speakers (a pair of JBL 4312A monitors), which have had a long hard life.

 

Listening to these now, I think I’ve found an unpleasant problem and not just a difference, in one of the amps....

Back with a new test as per erik_squires’s suggestion

This time the tannoys high and low frequency terminals were jumper-ed and there was no bi-amping or bi-wiring done.

Three recordings were made of three pieces of music with two amps. One amp has two speaker out options, so that’s why there’s three recordings. The three amp configurations are name "j", "y" and "p". The three recordings are of a sparse piano track with a change in dynamic in the middle: "piano". A samba jazz trio: "trio". And a voice with orchestra track (great production but the master recording could be better, i think): "voice". So the 9 edited recordings are:

https://escuta.org/webtmp/j-trio.flac

https://escuta.org/webtmp/y-trio.flac

https://escuta.org/webtmp/p-trio.flac

https://escuta.org/webtmp/j-piano.flac

https://escuta.org/webtmp/y-piano.flac

https://escuta.org/webtmp/p-piano.flac

https://escuta.org/webtmp/j-voice.flac

https://escuta.org/webtmp/y-voice.flac

https://escuta.org/webtmp/p-voice.flac

Steps.

1. A 60Hz sine tone was recorded at approximately -1.0dB

2. The tone was played with each amp setup with the voltage of the amp adjusted with the preamp to measure 0.5V. Fine adjustments of the voltage were made with the volume control of the digital source (Logitech Media Server)

3. For each amp, the three musics were played at the adjusted amplitude.

4. Recordings were made at 48kHz with a Behringer UCA222 interface on a laptop running the DAW Ardour.

5. Nine edited ecordings exported as 48kHz 24bit Flac files.

If anyone has time to listen and give some feedback, please do!

 

Thank you very much budjoe for taking the time to listen to the recordings and for the detailed account. Yes, it’s a very good point about environmental sound affecting the recording and the listener’s ability to distinguish between the two. There was perhaps a bit more wind in the first recording too along with those angry birds at the end. I think the Tascam and the valve mic had been on for over half an hour and the second test was done shortly after the first. So probably not much of an imbalance there. But good point, also.

jji666. I agree!

Thanks too, erik_squires. I think it’s a great idea to do this test bi-passing the speakers and consequently any unwanted ambient noise and irregular room interactions. I’m hoping I can do this later today or on Wednesday. I’ll likely do the recording at the amp end of the speaker wire. Some recent surgical work has left me a bit shy of shifting equipment!

@surdo 

Finally got to due a little more compare of the e & t files. My first try was at a lower volume than I think should be used for comparison. First pass at that low volume, file e was much better than t, but on replay, t was closer to e than first pass of either. Maybe my electronics were not fully warmed up.

Today played at more reasonable listening level, and also played through headphones. Both files very close overall. Occasional portions sound a little better on one or the other. In general, I like the piano sound (tibre, resonance, attack) better on e than on t, but that is mostly from middle C and above. Lower octaves may be slightly better on t. Bass very close on both. Drums and Drum kit too low in volume and too far in behind piano and bass to judge.

In general, I think most of the differences may be more environmental and microphone, rather than effects of either amplifier and connection. Note especially squeaky wheel/bird song audible on file e from about 4’43’’ to 4’51”, but missing on file t.

So in all, I think you tried an interesting experiment, but I can only tell you about what I can hear differently on the 2 files, and I think there may be more environmental artifacts in the two files than you were hoping for. Some may even be from the microphone and tascam reaching a thermal equilibrium, as well as the electronics and the room. Not sure I would hear the same things if I were in the room listening without the microphone and Tascam involved, so I think any of my comments about the files should be discarded.

@erik_squires "It’s your experiment, but as someone who worked quite a bit with microphones and speakers and electronics, the advice I gave you is solid."

As I said, I am ambivalent regarding matching amplifier out at speaker via 60HZ for amplifier compare. 60Hz sine voltage matching thru the rest of the audio chain is a really good way to compare different pieces of the chain. It is also a good reference for speaker efficiency. For all that is going on between and amplifier and speakers I am not sure it takes everything into account. 

I would love to see a graphing/charting of 60Hz voltage matched amps to the same speakers done in an anechoic chamber with dbA and dbC with some wide frequency music to see the active perceptual effects. Know anyone that can perform this kind of testing? Will see if I can find anything regarding similar testing.

OP:

It’s your experiment, but as someone who worked quite a bit with microphones and speakers and electronics, the advice I gave you is solid.

Measuring the SPL of quasi random white or pink noise is hard to get precisely accurate, it’s even hard with a multi-meter, which is why level matching should be done with a multimeter and single signal.  60 Hz is a frequency which is convenient as any $10 multimeter will read it, and all are more relatively accurate IMHO than an audio meter, given they are not subject to random room noise or the periodic variability of white/pink noise sources. Level matched experiments comparing DAC’s for instance should be level matched this way and with a decent meter will give an accuracy well better than 0.1 dB.

Now, once we agree to use a multi-meter, and a standard sine wave, whether they measure RMS voltage accurately, peak or peak to peak is irrelevant so long as they are absolutely RELATIVELY accurate. That is, you can trust that 3V now will be 3V in an hour after you switch amps.

Also, with a 60 Hz sine wave signal, again, any $10 meter will measure the voltage pretty accurately and almost perfectly relatively accurately. A nice meter will give you more zeroes.

The anti-testing and anti-measurement crowd certainly makes an impression. It's a good thing those folks aren't doctors or pharmacologists, or we wouldn't have gotten any further than "which medicine do I think makes me feel better..." [which, back in the day, is how people chose which, literally, snake oil product they'd consume]

I think it's a great post and even if not scientifically perfect, this is the type of thing that can actually validate all these claims of magnificent differences in amps, cables, whatever. So I have to assume they don't want to know what the emperor is wearing, or not. 

@erik_squires "1. Match level by multimeter instead of SPL. Get a 60 Hz signal and check the output on either speaker. You can get really accurate this way."

I am a little ambivalent about using 60Hz sine and voltage to verify output level balance. It can tell you something about what an amplifier is doing, and something about speaker efficiency, but I am not sure it can translate into anything about sound quality. There is also the problem of ensuring the meter you are using is measuring true RMS versus average AC voltage. If you are trying to use that method to balance output level, I think you would be better to use pink noise which has a full frequency range. Measurement becomes more of a problem, but it more closely resembles what you may hear when converted to sound using real music. In other words, since speakers are a reactive load, frequency and duration affect response and the amplifier’s drive of voltage across a frequency and time response will vary and may affect actual perceived sound volume.

I don’t really understand why SPL would be frowned upon. It is specifically an agreed measurement of perceptual sound volume reference. It is tailored for a specific frequency range and can be A or C balanced for perceptual preference favoring broad frequency response (A) or lower frequency weighted (C). The only caution for use in this testing is that the ambient noise level is maintained the same, and the SPL is taken from the same location in the same environment. If you are within 0.1 or 0.2 db that is considered well matched.

BTW, I don't believe in static tests very much.

That is, as a builder I use static tests such as frequency sweeps, voltage, etc. to guide my work, but I also know that these types of static tests may not cover all possible dynamic situations.  By this I mean frequency response, output, distortion.

We use these static tests and basic models because they are damn convenient and accurate for what they are, but the testing I'd want to see done, which I never see done, is to compare the output at the speaker or across the speaker cable with actual music.  With demanding speakers or loads we may very well see and measure dynamic behavior which is more complicated and unexpected based on the traditional static measurements say taken by Stereophile. 

OP:  Should be about the same.  Another test is to connect your line inputs to either side of the HOT cable.  This will record any and all variations caused by the cable alone. :)

I read this and listened to this on my ipad. Not interested in responses related to the ipad. I heard a difference. I thought recording two was, a little crisper, analytical, forward, livelier, compared to the first. I thought it fun and i did it without foreknowledge of amps (not that it would have mattered for i have never listened to either but knowing the brands could have made a guess according to my readings not listening, which would have confirmed what i heard). Very imteresting and a fun experiment. Thank you.

Here’s the switching program that I did in the graphical audio scripting language "puredata". You can jump between up to 3 different audio files and a quick equal-power crossfade of 5 milliseconds is applied when the switch occurs, so no click. It only works on Wav or Aiff files, not flac. If anyone wants to try it, please let me know and I’ll give more details.

"It's am interesting quest, certainly do not be deterred from pursuing your methodology. I do find fault with the logic of searching for what sounds most like live instruments, as if there is some absolute. And there lies the fallacy in the premise.

If we can agree that all devices color the sound with distortion, I believe that some of us find some distortions consonant with the fabric of the music, or "pleasurable" if you prefer, and others find different distortion spectra to be closer to the sound of real music. What works for you will not work for me. And this is supported by the wide variety of amplification topologies that sell well in the marketplace. Like finding a mate, there are some faults that one can live with and some that one cannot. Thank goodness we dont all find fault in the same things."

viridian Thanks for your message. I don't believe its a fallacy that one can be able to distinguish an excellent reproduction from another. It's just that some people are better at it than others. An excellent recording engineer will be somewhere way up on the scale of this ability, especially when they are working closely with musicians to get the sound that the musician wishes (as they should). An excellent musician will also know and even an excellent audiophile that keeps regular contact with live acoustic performance of music. I think it's our ears that get estranged from the absolute sound of instruments because of our distance from them and because we mostly hear instruments played in amplified settings and via reproduction - and in the case of the listener of recorded music near-exclusively, the ears can be corrupted in feedback loop of "taste" - with his or her preferred sound redefining mentally and incrementally what they believe an instrument should sound like. Certainly, every musician has his or her playing technique/embouchure, etc. which affects the sound as does the sound of the instrument itself and the room, but those familiar with the instruments, room acoustics and with the particular player where possible, playing live unamplified, will be able to get closer to an absolute recognition of natural live performance.

So when someone above joked asking if I lived above Carnegie Hall, that would really great and would help me as listener a lot. I have to rely more on memory. But if we want to be good listeners, we do need to know what it is that we're listening to. And I'm not saying what amplifier.

It's interesting, I sent the links of the two recordings to a nephew who is also a drummer, but has younger ears than mine. He heard the opposite to me in terms of body and fullness but agreed with me when it came to the cymbals, choosing e.flac as more realistic in this sense.

I'm still reserving my final decision on which amp I consider more natural until I can write this little program that lets me switch instantly from one recording to another. I'm also really keen to try eric_squires suggestion.

"If you attempt to record off the speaker terminals though you have to be sure you keep your voltages low, probably 2V or less at all times."

erik_squires Thanks Erik. Yes, I won't use one of my good recorders for this or the laptop input, but I have a cheap Behringer dongle interface that I can risk. Would it matter if I connect the wires at the amplifier end, as long as the speakers are attached? I can keep the wires shorter that way and won't have to shift heavy speakers around.

OP:

Yes, use the speaker terminals to create a line-level input, which you feed to your PC's inputs.  It's a little dangerous because your amp CAN fry the inputs.

2.83V is ~ 1 Watt if fed to 8 Ohm speakers. 

For your listening tests, it doesn't matter what reference level you set, as the multimeter will take several hundred volts, but whatever you pick, use the meter to get you to near millivolt accuracy in terms of level matching.

If you attempt to record off the speaker terminals though you have to be sure you keep your voltages low, probably 2V or less at all times.

A strange post....can't say I learned anything. All I care about is which amp sounds best, and how much.

Post removed 

Very interesting post!  Some on this forum do not support single or double blind testing, however I appreciate your efforts.  I had a listen to both files driven from my laptop using Sennheiser 58X headphones - not the best chain, but what I had readily available.  I used Audacity to playback, which allowed me to quickly switch between the two files.  I think you did a great job of level and timing matching.  I have to say I heard little to no difference between the files.  There were small differences visible in the waveforms, however using Audacity to calculate to overall spectra of the files they were essentially identical.  I tried to post them, however couldn't figure out the process for posting images.  In any case, well done on an interesting presentation.

What a great post! Thoughtful and challenging and, wow! thorough.
And you're absolutely right about naming the amps! That would just trigger the haters to... well, hate.

"How can a $30 op-amp sound better than a 200lb amp with a 1/4" front panel?"

(Okay, it almost certainly doesn't, but... hyperbole.) 

Some people made useful comments and I see you took them in.

You are very welcome here! At least, by me. 

Have fun!

I have run the DAC gamut. I've had a Schiit Bifrost, RME ADI2, a host of Topping's, IFI K9 PRO, Denafrips ARES II, leading up to a Cary Audio DMS-550. My current DAC/Streamer is by far the best I've had. Eversolo DMP-A8. I have friends that have DACs costing over $20k say they are shocked at the quality of the Eversolo. My system is Cary SLP-05 preamp, MF M8-700m monoblock amps and Sonus Faber Serafino speakers. My $2k DAC fits right in.

"1. Match level by multimeter instead of SPL. Get a 60 Hz signal and check the output on either speaker. You can get really accurate this way.

2. Consider recording AT the speaker connections. You have to be careful to keep the peak voltages ~ 1V but this would let you get a noise-free recording of what the amp is actually offering the speakers. Then you can do some very interesting analysis such as output by frequency, compression by frequency, etc.

This option also makes it easier to listen to any differences directly as opposed to via another microphone, though the max signal to noise will be different."

That’s brilliant erik_squires, thanks for your input!

Am I understanding correctly that you’re talking about getting the signal to be recorded by attaching a cable to the terminals on the the speaker? ie. connect positive and negative terminals to the wire and shield respectively of a short unbalanced cable (one cable for each speaker)?

I think I’d have to test with the speakers wired traditionally - ie. not bi-wired or bi-amped, correct?

I know very little about electronics, but do have a meter, soldering iron, etc. What sort of voltage might I expect at "normal" (not too loud) listening levels, seated just a bit more than two meters away from each speaker? They’re 8 ohm speakers.

"All amps of low distortion, low output impedance and flat response will sound indistinguishable operated below clipping. Decades ago Peter Walker of Quad did a test using RTR tape copies of classical master tapes between two Quad amps - a transistor and a tube amp. Speakers used were the then new Quad 63’s. The listening audience were British HiFi press. Nobody could tell the two amps apart!"
 
jasonbourne71 I’ve heard about that experiment and if it’s really the case with my two amps, I’ll be very happy. Two good amps!

OP:

I see where you are going with this and I have a couple of suggestions:

 

1. Match level by multimeter instead of SPL. Get a 60 Hz signal and check the output on either speaker. You can get really accurate this way.

2. Consider recording AT the speaker connections. You have to be careful to keep the peak voltages ~ 1V but this would let you get a noise-free recording of what the amp is actually offering the speakers. Then you can do some very interesting analysis such as output by frequency, compression by frequency, etc.

This option also makes it easier to listen to any differences directly as opposed to via another microphone, though the max signal to noise will be different.

 

All amps of low distortion, low output impedance and flat response will sound indistinguishable operated below clipping. Decades ago Peter Walker of Quad did a test using RTR tape copies of classical master tapes between two Quad amps - a transistor and a tube amp. Speakers used were the then new Quad 63's. The listening audience were British HiFi press. Nobody could tell the two amps apart!

"You are your own enemy in this over-technical quest"

I think it was pretty simple test and reasonably quick to do, as I mentioned above

"Answer is obvious or it isn’t. It’s preference, not better."

Sorry, but once again, I don’t think it’s preference. I’ve listened to many systems, good and bad, and I’m never every fully convinced that there’s a musician sitting somewhere to my front, playing an instrument. But some systems approach this ideal better than others. They reproduce the reality of an instrumental sound and its performance, better, through their rendering of timbre, spatial qualities, etc. (although the recording itself also plays a huge part in this). So that’s what I’m looking to do in this, yes, experimental test, choosing to concentrate on the tone and timbre of the instruments rather than the spatial aspect.

"Forget the earphones, listen to the speakers in the space."

I of course do this all the time, as we all do. The problem though is in making a clear comparison of two systems when you need to switch amps on and off and mess around with cables in between sittings. My listening memory is not good enough to make this type of comparison when the two systems are of a comparable quality. For those that can do this with confidence, good for them.

At some point, on the weekend, hopefully, I’ll use some software to devise a switch, where I can jump, with a single click of the mouse, between one recording and another - like you can toggle two solo buttons on a mixing desk with two hands. As it is, I have to stop one recording on a certain player, go to another player, and switch that on. It’s not as awkward as messing around with cables, but it could be bettered. So the test will be more technical still, but I think more advantageous.

You are your own enemy in this over-technical quest

Answer is obvious or it isn’t. It’s preference, not better.

"One setup involves a bi-wired high-powered stereo power amp and the other uses a pair of identical lower-powered amps with which the speakers, a pair of Tannoy System 12 DMT II monitors, are vertically bi-amped."

......................................................

Forget the earphones, listen to the speakers in the space.

You could use a SPL meter or your phone to volume match. Practice 1st, find two equal volume positions, mark with pencil.

High Power/Low Power, instantaneous peaks: difference?

bi-wired/bi-amped, difference? low volume difference? higher volume difference?

.....................................

btw: how are you matching the mono block's volumes? measure with SPL meter

I was blessed with two smart daughters (now both PhDs) who were picked on because they were smart. I would tell them early on That the world is full of A__holes these are young ones and you will run into them all your life. Pay them no mind.  This site has its share.

I have read a lot of books, mainly just the first page. Thanks to people like you you were able to get to page two and three and beyond. If left to me, there would still be wagon wheel repair on every corner. Way to keep the evolution chain evolving for the better

@surdo   Great post and listening exercise.  I listen with my ears and not my eyes or wallet.  That being said, there is no wrong answer because everyone hears differently and prefers a different sound profile.  I will listen this weekend on my main system.  

It's like listening to music over the telephone.  You expect a valid comparison?

Just listen to music you are very familiar with and compare them directly. You will know pretty quickly. Good luck. 

I was hoping (like the OP) to read some OPINIONS or IMPRESSIONS comparing the 2 recordings. I don't see ONE (maybe hidden in all the other non relevant comments). Neither my ears (listening skill) nor may system are good enough to make a judgement, so my "I can't hear a difference" verdict should not count. 

Sure seems like a lot of work to come to a subjective conclusion. Only thing that matters is which one sounds best to you. Admire your dedication and tenacity!

"surdoYou actually played it well. This forum has it's share of haters and naming your gear will usually open you up to a kick in the nads from at least one or two of them. You're still getting a good taste of that even without brands. I for one am ashamed that your first visit has been so contentious."

No worries. I'm having fun and I'm here to learn. I also occasionally frequent bicycle forums. They're far worse. People here will have to try harder to get me wound up. :-)

"Haven’t listened to the files yet, but hope to later today."

Great, thanks

"I find your wiring configuration of the Sunfire amp confusing. To my understanding, current drive would have the most effect on the lower frequencies. The tweeters should not need for much current regulation or drive. Have you tried the Sunfire with the current/voltage switched. I think that might change the sound presentation significantly."

Yes, I see you what you mean. On the back of the amp, the suggestion for bi-wiring is to use the current source on the tweeter and mid-driver and voltage source on the sub. Since the 12" driver on the monitors handles the mids as well as the lower frequencies, I should probably swap it around. I guess there's nothing much to warm up in the sound coming from the tweeter alone. Thanks for the suggestion, will try soon.

surdoYou actually played it well. This forum has it's share of haters and naming your gear will usually open you up to a kick in the nads from at least one or two of them. You're still getting a good taste of that even without brands. I for one am ashamed that your first visit has been so contentious.

Haven’t listened to the files yet, but hope to later today.

I find your wiring configuration of the Sunfire amp confusing. To my understanding, current drive would have the most effect on the lower frequencies. The tweeters should not need for much current regulation or drive. Have you tried the Sunfire with the current/voltage switched. I think that might change the sound presentation significantly.

"@surdo  you have to measure voltage at the outputs using test tones."

Ah, yes, that makes sense. Thanks

I didn’t realise that this would be such a contentious post! :-) Starting with the most recent:

"What is the "real instrument" reference? Do you have an orchestra or maybe a jazz combo around? Maybe you live in an apartment above Carnegie Hall. In any case it’s personal taste that drives the "reference" myth as all recorded music is an illusion filtered through our addled brains."

Yes, the reference is the sound of real unamplified musical instruments, played by musicians. In my case, I know well the sound of a piano. I played violin (still play a bit) and have been around quite a few double basses. I also played drums for a number of years and I still play other instruments. And actually, in the recordings, it’s the snare drum and the cymbals that have helped me detect realism. So those are my references. Anyone else that’s heard an instrument played unamplified can use that as a reference too. When you hear a recording, it might (and often does) sound too bright, for example, and that might be the fault of the stereo that’s playing back the sound or the way the instrument was recorded (with mics) or equalised. And yes, that’s a big issued when you’re trying to compare gear - one piece of equipment might be rendering the rotten recording perfectly, when another smooths out the rough edges. But the listener can tell whether or not the sound appears "natural" through his or her experience of listening to real instruments. I’m sure most people on this forum have listened to instruments played with no amplification.

I don’t think it has anything to do with personal taste. A real instrument sounds like a real instrument.

"Which recording produces the most realistic rendering determined by what ? Not the device that rendered it ? but by ?? I am easily confused not being rude but I am in need of little more guidance in order to participate please. You have provided a recording engineer’s stats and it is obvious you know what they mean."

As above, but re, the stats: I just put those there because someone mentioned that my method of level matching wouldn’t work. I don’t doubt that there’s a better way, but I printed the stats to show that my recordings were fairly evenly matched, at the least with any difference being imperceptible, I think. The peak of any audio file is the digital sample of the file with the greatest amplitude and the highest sample amplitude possible in a digital system is 0 dB, when using the logarithmic decibel scale of intensity. Other values are negative and the more negative the value, the less amplitude it has. The trouble with these peaks is that they occur rapidly, so fast that our ears can fail to perceive their intensity. RMS stands for root mean square and this is a type of average of adjacent values (so those amplitude values that occur around the same time). These "RMS" values relate much more closely to our perception of "loudness" (the human perception of the acoustic property, intensity). So since the two recordings had RMS peak values -16.77 dB and -16.72 dB (0.05 dB "louder"), the difference is so low, that I believe the loudness of the two recording is reasonably well matched.

"I’m not aiming to deride your efforts, OP, just sharing objectively: by limiting the samples to 2 and stating each sample is from a different amp, you’ve set the stage for expectation bias in responses. Repeated measures can make for a challenging experimental design online."

I agree. But that’s all I had time to do and I didn’t think anyone here would have time or inclination to listen to more, let alone 2 recordings;

"Much a do about nothing IMHO...sorry."

Don’t feel bad about it, humble one. That’s ok.

"What an incredible amount of trouble to go to to make a comparison between two amplifiers that does not necessarily align with real world sq and whether or not one sounds better to the listener or not."

The test and editing only took an hour. Writing and responding here is taking longer. Although I’m enjoying it.

"I would venture to say few here have the desire to dive into the minutiae the way you seem to enjoy. So which amp sounds better to you? Don’t over analyze it..."

That’s true, but it would surprise me that people don’t put this effort (and concentration) into their own gear selections and configurations, especially with such large sums of money involved. I’ll get to my opinion soon.

"I can’t read the flac files and don’t care to load the software to enable it. Put them on YouTube for a bigger audience."

If you’d like me to put a CD format wav file online, please let me know.

"Tracks actually loaded easily via copy/paste into a url search field on my Mac. Listening through Sennheiser headphones and a very decent dongle Dac. e.flac seems to have more body, realism relative to the room of the performance. t.flac gets more detail, but seems a bit colder in tonality. Level match seems close enough to render the important qualities to me? But maybe try a couple more…"

Thanks for listening @riccitone! And since we have a listener, I’ll reveal the amps:

A pair of Quad 303 amps used to bi-amp the monitors and a Sunfire Signature 600 (early version) bi-wiring the speakers with the "voltage source" speaker outputs attached to the main driver and the "current source" outs connected to the tweeter.

What I did to listen, was to load the two files into two VLC media players. I have a way on my computer that i can shuffle the two players, with one on top of the other, so that I don’t know which is on top after shuffling. I then separate them, so they can be played, and conceal the title bar to make them anonymous until satisfied enough to check. I did this a whole lot of times.

My conclusions were very similar to yours and the first thing that struck me was the full body-ness of e.flac. I also paid a lot of attention to the snare drum in the opening passage and that appeared to have the airiness of a real drum in e.flac despite t,flac sounding a tad cooler. I also found the bass a little clearer in t.flac and I think the piano masks it more in e.flac. I have to admit it’s really difficult though. I think both sound good. And after some repetitions of listening and revealing the file name, I didn’t always reach the same conclusion. But mostly did.

Which amp corresponds to which file?

e.flac is the Quads
t.flac is the Sunfire

Without going through the rigmarole of setting up a mic and listening to the recording, I think I hear the same tonal differences and instrument separations when sitting in front of the system, listening in stereo. I think there, the Sunfire has the advantage of slightly better imaging and I suspect that is due to its slightly cooler sound and with the higher frequencies (usually easier to locate) taking predominance. But the thing with the snare (and cymbals) sounding more real and airy on the Quad still confuses me. i.e. why wouldn’t this help with imaging? Perhaps my two "identical" 303s aren’t so identical, there might be a mismatch. I have tried using just one of course. It didn’t sound so dynamic and I didn’t at the time compare it with the Sunfire.

"GREAT version of U.M.M.G btw 👍🏼👍🏼👍🏼"

Well spotted! I didn’t give the full name of the track and the musicians because I think there might be copyright issues. But it’s a Brazilian trio residing in New York. Recorded two years ago and released on vinyl last year. The pianist was 83 when the recording was made and he was one of the founders of the Brazilian jazz trio movement in the sixties. A cult figure. Pioneer of samba-funk too.

I think there have been other replies since I started writing this message.

What is the "real instrument" reference? Do you have an orchestra or maybe a jazz combo around? Maybe you live in an apartment above Carnegie Hall. In any case it's personal taste that drives the "reference" myth as all recorded music is an illusion filtered through our addled brains.

Which recording produces the most realistic rendering determined by what ? Not the device that rendered it ? but by  ?? I am easily confused not being rude but I am in need of little more guidance in order to participate please. You have provided a recording engineer's stats and it is obvious you know what they mean. I don't.  I am very impressed with your knowledge and your  detailed test. 

I’m not aiming to deride your efforts, OP, just sharing objectively:

by limiting the samples to 2 and stating each sample is from a different amp, you’ve set the stage for expectation bias in responses.

Repeated measures can make for a challenging experimental design online.