Steve williams
Site Founder, Site Owner, Administrator
And it's exactly what the work I do in tweaking components addresses
you mean you do this as your job?
And it's exactly what the work I do in tweaking components addresses
No, not at all, I meant work as in activity, or focus of one's attention. Could be doing it in the future, though ...you mean you do this as your job?
Putting all kidding aside.Keeping our egos in check. Subverting our financial interest. Acknowledging that there is no magic or alchemy. Accounting for our prejudices. A long winded way of saying "all things being equal."
I believe that measurements are very revealing! Whew! How could they not be? Despite all the name calling, all designers measure. Clearly they only take us part of the way home. Frank I get your point about car measurements. The fact is if we could not predict how a car behaves in a curve there would be a lot more dead people.
It all boils down to the fact there are so many variables and so many conflicting influences that it is difficult to reach any consensus about the parameters. Even when there is a consensus those who know what is correct are afraid to risk retaliation from their fellow manufacturers, reviewers, and most importantly their customers. I recall quizzing Alon Wolf. You would think he was taking his oral exam for his engineering degree. What we are left with is a hodge podge of rules and products with no real way for the end user to sort it out.
What we end up doing is applying our(the average audiophile) necessarily limited knowledge to the available products. Hopefully we come up with something that allows us to come close to the music. So far the results are as varied as those attempts to create it.
Old Listener, I think we are on the same page. Yelp is about sharing subjective experiences of restaurants, etc., while the WBF is about sharing subjective audio experiences.
I don't understand why those that believe that measurements are the end all be all don't follow the path blazed by Ethan Winer - buy a $49 cd player, cheap studio monitors, and room treatments, and listen to good music until the end of time.
Exactly the result I would have expected: this is a macro measurement and would have picked up extremely little.With that amplifier, adding large storage capacitors to the power supply improved dynamic ability and produced a sound that may be described subjectively as "more muscular", "beefy sound", "more meat on the bones", as opposed to "thin". Frequency response into either a dummy resistive load, or a real loudspeaker was +/- 0.1dB from 22 to 18k. I measured at 1W, 10W, and even 20W and 100W into the dummy load.
I figured that with "beefy", I could measure THD using bass frequencies. So, I plotted THD from 20Hz to 20kHz at 20W and then added or took away large capacitors in the power supply. No statistically significant difference
Possibly. Were you looking at spectrum results?Adding more small, fast capacitors to the power supply allowed more resolution and detail. I thought that I might be able to find the measurement for resolution and detail at the very low power IMD, but the difference was so tiny that it could have been due to the hook-up I used (alligator clips). Increasing the line resolution from 1 pixel to 3 pixels effectively erased that difference on screen.
This could go somewhere, and since you're prepared to put some real energy and time into this exercise, may I ask, as a starting point, the following: if you repeated the 80Hz test can you do a spectrum analysis of the result, we're not interested in any total figures like THD, and if so, what precision does your hardware and software setup allow, in terms of measuring a low level peak adjacent to a high level peak?Measuring THD and IMD using a real speaker load and a resistor did reveal some difference, but not enough to make any sense of, but it was larger than the difference in IMD adding speedy capacitors to the power supply.
I also tried different IMD frequencies and mixes. I can't remember all the combinations, but SMPTE which specifies 60Hz and 7kHz was one of them. I mixed the two tones at various ratios from 4:1 to 1:1. (I guess that satisfies fas42's test of a bass frequency below 100Hz) DIN spec measured a hair lower at higher power because it's 250Hz and 8kHz.
I also looked at TIM (transient intermodulation distortion) at high power and low power.
After failing using the suite of measurements available on CLIO (the software I use), I tried to derive something different. To measure how well an amplifier resolves micro-dynamic detail (the quiet breathing of the singer while the rest of the band is doing a solo), I figured that if I injected a 80Hz large signal into the amplifier, and then measured the THD of a 1kHz signal, we might be able to see something.
The 80Hz would have to be filtered out somehow, so I built a 4th order passive filter at 80Hz, and ran the amplifier into a dummy load. I compared the THD measurement (the passive crossover increased the THD at the load by about 0.5%) with and without the 80Hz signal. I measured the THD at 1kHz, subtracted the 0.5% from the crossover, and subtracted the 80Hz THD measurement - no statistically significant difference.
Actually, what you demonstrated is that the ears are an excellent measuring tool, extremely sophisticated in terms of filtering away extraneous detail, and that beat the pants of any device you stick into a point in the wall in getting an accurate assessment of progress.After about a month of that, I figured that I was a better listener than a measurer, and did the rest of that part of my design with my ears.
3. Spectral fusion
Recently, Mike McNabb, a composer and researcher at our center, discovered a perceptual phenomenon while experimenting with a vocal synthesis technique. He obtained the Fourier Transform of a soprano tone that was recorded and digitized at the center. He then synthesized tone, using additive synthesis, such that the spectral balance was the same as indicated by the Fourier Transform. At first, the frequency for each harmonic was kept constant; however, the tone did ? sound vocal at all. In fact, it didn't even sound natural. When some vibrato was added, such that; harmonics were affected synchronously, the percept was strikingly realistic.
John Chowning explored this phenomenon even further. He synthesized a tone such that ea< harmonic (a sine tone) began one after another, but remaining sustained. Again, the spectral balance of the partials corresponded to the levels obtained from a Fourier Transform of a recorded soprano tone. With all harmonics playing, it was very easy to hear each harmonic separately, as if there we many sources, or voices (each source being a sine tone). But as soon as a common vibrato was add< to all the harmonics, the sound fused into a percept of a single source ~ that of a sung soprano tone.
These examples show that temporal aspects of a tone are important features of its timbre, even during its so-called steady-state portion. In other words, spectral balance alone cannot determine timbre because a constant spectrum may not fuse, and one can't really have timbre without fusion. Ti examples also raise this question: What are the characteristics of a sound that cause it to fuse into percept of a single source with a particular timbre?
Elizabeth Cohen has investigated the role of harmonicity (or the lack thereof) in the fusion of complex tones. She has found that fusion depends on temporal envelope, degree of inharmonicity, and spectral content (Cohen 1979a,b,d and 19S0a). Stephen McAdams, a graduate student at our center, is al investigating fusion and source identification for his doctoral research. Results of his preliminary work appeared in (McAdams and Bregman, 1979). The work of Cohen and McAdams will be valuable aid in determining the parameters for timbre.
D. Timbre and perceived onset
Finally, in this section we would like to cover some of the interests we have in looking at the relationships between timbre and other aspects of tone, such as perceived onset time, duration and loudness. We would expect rather direct and strong relationships between timbre and these other ton attributes because they are effects of similar acoustical dimensions. Spectral shape is a determinant ? timbre as well as of loudness. Similarly, the temporal envelope of a signal is a determinant of onset duration and timbre. Combining the two, a spectral shape that changes with time is a complex description of timbre, and also provides the material for making loudness, onset and duration judgments on naturalistic signals, such as actual musical timbres.
In past research, we have found, for example, that a model of loudness perception for steady-state spectral distributions (Zwicker & Scharf, 1965) was useful in modeling the timbral dimension relating perceived spectral brightness (Grey & Cordon, 1977). The recent derivation of a model for loudness perception for time-varying tones (Zwicker, 1977) presents encouraging possibilities for extending t\ model to include various other aspects of the perception of time-varying tones. Already, this model hi been useful in analyzing various aspects of the perception of timbre.
12 I. Perceived onset time
We have long been interested in formal models for the temporal properties of tone. One possible ? related to the above, concerns modeling the relationships actually found along temporally-related dimensions of timbre perception uncovered in our multidimensional scaling studies. Various subject correlates to these temporal dimensions have been noted, one being the "hardness" or "explosiveness" the attack (found with D. L. Wessel, 1977). In the hopes that this attribute may have something to with perceived onset time for tones, Cordon and Crey have run an experiment to equalize the on times for the timbres used in the original studies.
The procedure (like that of J. Vos and R. Rasch, 19S0) involved the setting of two tones (e.g. A and to be locked into rhythmic phase such that their alternating series (i.e. ? ? ? ? ? ? „.) made perceptually isochronous rhythm and the perceived onset of the ? tones perfectly bisected the duration between the A tones. By adjustment, the listener set the temporal delay between A and B, where I physical delay between all A's and all B's was equal. In looking at the relative delays for the differ stimulus tones, all of which were taken from the set used in the multidimensional scaling studies, independent measure of onset time has been achieved.
Regardless of the success of the experiment in terms of modeling our temporal axes from the scale research, we still have data for the onset relationships among a set of timbres. We hope to be able mode! These onset relationships taking advantage of the temporal features of the new model; loudness perception mentioned above (Cordon, 191).
Orb, I would just say you're over-complicating the picture of what an audio signal looks like: no matter how complex the sound, it is ultimately nothing more than a meandering "wiggle", a pure sine wave is just a wiggle that meanders in a very perfect, smooth pattern. At the level that the electronics have to look at what's going on, the audio signal is really very easy to deal with, the problems arise because something disturbs the precise path the meandering takes, or the electronics run out of puff, the power supplies can't keep up with what the circuitry is demanding from them in a clean enough fashion.
Frank
I've read through all 24 pages twice this evening hoping to find something useful. Going back to Mark's original post, I've been trying to correlate measurements to what *I* hear for the past few years, and haven't had any luck - once the measurements are "good enough". Let's take 2 examples of something that we can hear.....
I am assuming that for an amplifier the frequency response and phase are flat to +/- 1 dB from 20Hz to 20kHz (and even beyond), THD, IMD are below 0.1%. I think that we know that we already know that we can measure competence. I've got a competent amp - see this post I put up in the beginning of the year:
http://www.whatsbestforum.com/showt...Analog-Dialectic&p=35495&viewfull=1#post35495
The PNWAS members have heard it, and none of them have said that it sounded incompetently designed.
With that amplifier, adding large storage capacitors to the power supply improved dynamic ability and produced a sound that may be described subjectively as "more muscular", "beefy sound", "more meat on the bones", as opposed to "thin". Frequency response into either a dummy resistive load, or a real loudspeaker was +/- 0.1dB from 22 to 18k. I measured at 1W, 10W, and even 20W and 100W into the dummy load.
I figured that with "beefy", I could measure THD using bass frequencies. So, I plotted THD from 20Hz to 20kHz at 20W and then added or took away large capacitors in the power supply. No statistically significant difference.
Adding more small, fast capacitors to the power supply allowed more resolution and detail. I thought that I might be able to find the measurement for resolution and detail at the very low power IMD, but the difference was so tiny that it could have been due to the hook-up I used (alligator clips). Increasing the line resolution from 1 pixel to 3 pixels effectively erased that difference on screen.
Measuring THD and IMD using a real speaker load and a resistor did reveal some difference, but not enough to make any sense of, but it was larger than the difference in IMD adding speedy capacitors to the power supply.
I also tried different IMD frequencies and mixes. I can't remember all the combinations, but SMPTE which specifies 60Hz and 7kHz was one of them. I mixed the two tones at various ratios from 4:1 to 1:1. (I guess that satisfies fas42's test of a bass frequency below 100Hz) DIN spec measured a hair lower at higher power because it's 250Hz and 8kHz.
I also looked at TIM (transient intermodulation distortion) at high power and low power.
After failing using the suite of measurements available on CLIO (the software I use), I tried to derive something different. To measure how well an amplifier resolves micro-dynamic detail (the quiet breathing of the singer while the rest of the band is doing a solo), I figured that if I injected a 80Hz large signal into the amplifier, and then measured the THD of a 1kHz signal, we might be able to see something.
The 80Hz would have to be filtered out somehow, so I built a 4th order passive filter at 80Hz, and ran the amplifier into a dummy load. I compared the THD measurement (the passive crossover increased the THD at the load by about 0.5%) with and without the 80Hz signal. I measured the THD at 1kHz, subtracted the 0.5% from the crossover, and subtracted the 80Hz THD measurement - no statistically significant difference.
After about a month of that, I figured that I was a better listener than a measurer, and did the rest of that part of my design with my ears.
If anyone can come up with a set of measurements that can measure the resolution and dynamic ability of an amplifier, I'm all ears
On the other hand, I *think* that I know what measurement you can make for "image density" or "image sharpness" of an amplifier. Measure the group delay of each channel across frequency and with respect to power. If the group delay tracks well between the two channels, you get a sharp image. If the group delay wavers with power, you lose focus. If the group delay wavers with respect to frequency, you lose PRaT (but only if you believe in PRaT).
If anyone can come up with a set of measurements that can measure the resolution and dynamic ability of an amplifier, I'm all ears
On the other hand, I *think* that I know what measurement you can make for "image density" or "image sharpness" of an amplifier. Measure the group delay of each channel across frequency and with respect to power. If the group delay tracks well between the two channels, you get a sharp image. If the group delay wavers with power, you lose focus. If the group delay wavers with respect to frequency, you lose PRaT (but only if you believe in PRaT).
Gary while we have your attention would you care to weigh on why speakers for the most part don't publish distortion figures. Is it that speakers don't lend themselves to easy to those type of specs. like amps? Is the distortion so bad that it's just not a good marketing idea? Or do you feel that distortion can be extrapolated form the present batetry of measurements ? (Hope there are not strawman arguments in there. Smile)
To put this into perspective Frank,
simple task please describe accurately how a piano chord will sound using existing measurements shown on Stereophile for two different amps (1 class A and 1 AB), description in terms of texture,tonality, musicality-how it flows, timbre brightness, lean-cool,etc.
You cannot, and that is what this thread is about; what we hear relating to audio and that is specifically inharmonic and harmonic sounds generated by voices and instruments.
Thanks
Orb
because we know that the transmission - the cable - is perfect even when it is a just zip cord. I've never been able to measure even a smidgen of distortion in any wire.
Just to recap this thread is about correlating measurements to what we hear; therefore it is how we hear the music or instrument that has been reproduced by an audio system.
With that in mind, you would need to be able to take the engineering type measurements you mention and then model those to reflect the music's/instrument's waveform.
Because this is an example of what we hear (actual instrument is a trumpet):
Therefore looking at the measurements used for amps-DAC-etc and those you mention, could you honestly say how it would affect the original waveform in the above example?