What contributes most to a change in how an amplifier sounds?
Amplifiers include tubes (if not solid state), big transformers, lots of internal wiring, Power supply, cabinet, gain controls if you're lucky, connections for incoming and outgoing cables, Computer chips, Control panels, semiconductor boards, design choices, age, etc.
Of all this stuff, what contributes the most to a change in how an amplifier sounds?
amidst all of these really good comments there were no comments from people who actually build amplifiers that I could see. It would seem to me an amplifier designer is putting together an amplifier he would need to choose between lots of parts to create something that was good. I would love to hear what these parts are contained within an amplifier that impact sound which is really what it’s all about.
What is it about the transformer?? What is it about the wiring? What is it about the chassis? What is it about the chips? What is it about the capacitors? What is it about the power of the amplifier? etc
"What contributes most to a change in how an amplifier sounds?"
amp designer, manufacturer, components incl. tubes, speakers, listening room acoustics, preamp incl. tone control feature etc., listener, music program and source, cables, RFI environment.
It all sounds amazing. It's so complicated. Very elegant prose. SAT verbal score was kinda low so forgive me I don't understand this too well. Some scary things related to a D amplifier. Wish I would've pursued an advanced degree maybe this would've helped.
A lower level of Dynamics, flatter line, is expected when impedance is lower?
@emergingsoulDynamics comes from the signal not the amp. When it seems like the amp is more 'dynamic' the chances are extremely high that what you are hearing is actually just distortion masquerading as 'dynamics' due to how the distortion interacts with the human ear.
Class D operation sidesteps the annoying Class AB artifacts, but then you get deep into designing pulse-width modulators that are unconditionally stable, resist transient upsets, have good phase margin, and also have low distortion, even under dynamic conditions. Basically all the challenges of designing a state-of-the-art ADC and DAC that can also deliver power into complex and nonlinear loads.
@lynn_olson If you design a self-oscillating class D amp then you satisfy all these requirements. In a self oscillating amp you intentionally exceed the phase margin by adding so much feedback the amp goes into oscillation as soon as its powered up. The feedback loop is designed to only allow one solution for the oscillation, which is used as the switching frequency. This has the benefit of allowing much higher feedback without the problems caused by lessor amounts and having it poorly applied. It also solves the problem of noise caused when the switching frequency drifts. So this allows the amp to be dead silent even on horns.
An amplifier's "sound" is only apparent when connected to cables and a speaker. So it could depend on the capacitance of the speaker cable, or any steep phase angles the speaker has.
And, as mentioned, the power supply, which is why the average receiver will not drive say, a Magnepan or some other type speaker (hybrids, electrostatics and other exotic types) that places demands on the amp it usually can't handle.
Sorry to disappoint, but not anything a user can alter, or even measure. It takes intimate knowledge of the circuit (and the design philosophy) to make these changes.
First, a high-current driver section that can rapidly charge and discharge the capacitance of the output section. These are often under-designed in both transistor and tube amps. The driver current affects slew rate, cleanness of Class AB transitions, and distortion above 1 kHz.
Secondly, avoiding current fluctuations in the power supply (caused by Class AB switching transitions in the output section) from affecting the input and driver sections (which are typically in Class A). Power supply variations affect the subjective quality of dynamic impact and clarity with dense program material.
Thirdly, in feedback amplifiers, adequate phase margin, preferably well in excess of "textbook" stability minimums. This prevents reactive speaker loads from degrading and stretching out the settling time of the amplifier. Settling time, by the way, is the time required to recover from a slew-transient overload. This affects the amp-speaker interface, and why some speakers don’t "get on" with some amplifiers.
All of these aspects are audible to the amplifier designer, and appear on internal measurements of the amplifier’s functions. They do NOT appear on external measurements of a "black box" under test. Unless you have designed amplifiers yourself, you, or a reviewer, are not likely to hear what changing these parameters sounds like. If you have, though, they are pretty obvious.
P.S. Reading between the lines, Class A operation and multiple independent power supplies makes for clean and stable amplifiers that have good subjective results. But ... true thermal Class A operation greatly restricts power output, and multiple independent power supplies also raise the price.
Class D operation sidesteps the annoying Class AB artifacts, but then you get deep into designing pulse-width modulators that are unconditionally stable, resist transient upsets, have good phase margin, and also have low distortion, even under dynamic conditions. Basically all the challenges of designing a state-of-the-art ADC and DAC that can also deliver power into complex and nonlinear loads.
I agree with those identifying the listening environment. It is critical factor that greatly affects sound quality. Obviously, the quality of the equipment, interconnects, power cords, speaker wiring, quality of AC power, conditioners, etc., all have an effect on the ultimate SOUND! I believe Bill Duddleston's work on developing equipment to address speaker room interactions is very important. To quote:
Loudspeakers and The Listening Room
A quick trip to your local high-end dealer can often leave one in bewilderment. If all these speakers are supposed to be accurate, why do they sound so different?
The answer lies primarily in the way the loudspeaker couples with the listening room. How it couples is a function of the power response and the physical properties of the listening room. The power response is related to the dispersion pattern of the loudspeaker and its amplitude response. Room properties include such factors as geometry, speaker, and listener placements, reflectivity of surfaces, and even ambient noise levels.
Reflections
Today, much attention is given to the way a speaker measures in an anechoic (reflection free) environment. While an on-axis pulse, FFT analyzed, can tell a speaker designer a great deal about how his speaker will behave at a distance of 2 meters in an anechoic chamber, what about the real world? Clearly these anechoic measurements weigh only the DIRECT ARRIVAL path to the listener.
A real listening room with walls, floor, and ceiling will have an infinite number of paths from the speaker to the listener. Most of the reflections from these surfaces have long path lengths, are diffuse and exhibit random phase, amplitude, and directionality cues. These reflections are usually not detrimental, and in most cases add to the “air” or “ambiance” in the recording. Reflections of this type are termed LATE REFLECTIONS.
But what about the short path reflections such as the inevitable floor reflection? These EARLY REFLECTIONS also tend to be the strongest of the reflections. Their single bounce pattern leaves very little opportunity for absorption or randomization, particularly problematic is the frequency range from 250 Hz to 1500 Hz. Such frequencies possess wavelengths whose dimension is similar to that of the reflected path- length, thus causing strong response anomalies. These frequencies also fall into the range where the human ear is most sensitive to the phase anomalies. (Current research indicates that human phase acuity diminishes sharply above 2 kHz. It appears that the hair-like transmitters (cilia) within the ear begin to scramble phase information when forced to change direction thousands of times per second. Amplitude information ultimately reaches the brain at these frequencies, but with unintelligible phase relationships.)
Early reflections can effect tonal balance, clarity, and image localization. The brain has a tendency to fuse these EARLY REFLECTIONS with DIRECT ARRIVALS into one smeared arrival. Unfortunately these troubling reflections are only about 6 dB weaker than the direct sound arrivals. In fact, most listening rooms will have average level differences of early to late sound of only 6 to 8 dB. Even more surprising is data demonstrating that only 15% of the reflections typically reaching the listener are from the sidewalls. The remainder consists primarily of the floor and ceiling reflections.
How do we deal with the dreadful floor reflection? Most of you have already done something by carpeting the floor of the listening rooms. Unfortunately, carpet is of little benefit below 1500 Hz.
We have found, as did Roy Allison a long time ago, that by elevating the lower midrange driver the proper distance off the floor, one can reduce mid-bass anomalies significantly. Then by establishing a crossover point to the woofer in the mid bass range, the responsive dip is virtually eliminated! This is owing to the averaging effect caused by the unique path lengths from the woofer and mid-woofer to the listener. This technique is effective in the Signature III, Focus and Whisper speakers, where multiple mid-bass drivers share the load.
What about the sidewall reflections? We have found that the biggest reason sidewalls cause a lack of clarity is that they are typically reflecting a signal that is inaccurate. Simply put, most loudspeakers exhibit horizontal dispersion that is not uniform.
How Does a Loudspeaker’s Dispersion Pattern Influence what we Hear?
Consider the classic 2-way speaker system utilizing a 1” dome and 8” woofer with a 2800 Hz crossover. While this speaker could appear near ideal on axis, when measurements are taken 30 degrees off axis laterally, it will in fact exhibit an 8 dB suckout near 2800 Hz, and an abrupt roll-off above 13.5 kHz. This undesirable beaming effect applies to all drivers and becomes a limitation when the wavelength of the radiated sound is smaller than the width of the diaphragm itself.
Such traditional design neglects the audible effects on power response and thus results in strong colorations in the reverbant field. In contrast each model of the Legacy speaker line is designed for specific application. Whether a corner sub, an on-wall rear speaker, a downward directed center channel or a forward firing tower, Legacy has taken great care in addressing directivity pattern and power response.
His basic design and his choice on the quality of: power supply, transformers, capacitors, resistors, wiring, connectors, potentiometers, remote volume control, the tube and/or transistor choices, and basically all components
Being turned on, or off....everything else subject to budget and/or taste....the latter goes deep end of the pool real quickly....
Always preferred flexibility vs. performance in extremis of late...the latter I can't hear without my aids and the variety of speakers driven for their varied qualities....
"...normally react to my given name (and a few others, done in vain or pain...), but Frank N. Stein re audio distractions will suffice..." 😏
Paraphrasing, "Not so normal as to be Normal, I intended to be a freak for the rest of my life..."
Miss-ion in progress, Please Stand By....and Back. *L* ;)
I inserted ‘real life speaker load’ into Google search and got this fascinating AI explanation. I was intrigued by this term that I read above. The AI info is less than helpful and doesn't really answer things very well.
A real-life speaker load is more complex than a resistive load, which is what amplifiers are usually measured against. Here are some things to consider about speaker loads:
Impedance
A speaker’s impedance varies depending on the frequency it’s playing. The stated impedance is usually measured at 400 Hz, but a speaker’s impedance can vary widely across the frequency range. For example, a speaker’s impedance can be up to five times its rated impedance at its resonant peak.
Acoustic load
The acoustic load a speaker experiences depends on how it’s coupled to the surrounding structures and cabinet. For example, a speaker placed in a horn will have a higher acoustic impedance than a speaker in free space.
Power handling
Using an amplifier that’s too small and driving it to distortion levels can damage a speaker.
Matching speakers to amplifier
It’s important to match the power output of your amplifier to your speakers. If your speakers are underpowered, the sound will be thin and hollow. If your speakers are overpowered, they could distort and become damaged over time.
Calculating speaker load
When speakers are connected in series, you can add the nominal impedance of each speaker together to calculate the total impedance.
To be honest I can't hear much difference when I swap between my 3 power amps. My 600 watter definitely has more oomph. Maybe its just nostalgia but I do like my old Phase Linear 400, however my ears are now, low-fi.
The peanut gallery chiming in with my $0.02... I’m going to say power cord and power conditioning. The Audioquest Niagara 7000 is no joke in the way it makes the background black, and the transient in all its variable forms, leap out at you.
Next, the Niagara is sensitive to power cords. I use an AQ Hurricane HC (high current) power cord, it has 3 loosely braided shielded conductors. I **think** the Hurricane "high current" cord is the top range in the Storm series of cords, and then next comes the BIG bucks cord series in the "Mythical Creatures," category consisting of (again, I think) Thunderbird, Firebird, and the big daddy, Dragon; these have copper + silver blends for the first two, and the Dragon uses all silver conductors.
The Hurricane HC (high current) is good. It’s very sopen and smooth. Then I inserted an older Shunyata power cord in the Python Zitron HC. With this cord, all the sudden, it sounded like bass got added and there was greater "jump" to music. Then I tried a Anaconda Zitron HC, which yielded an even warmer tone and gave everything greater separation, shape and dimensionality. My last cord is a more current cord in the Sigma XC (same as HC) V2 power cord. I think, Sigma is on top, then Alpha V2 and then Delta on the bottom; after that are the big bucks cords in the Omega XC and QR cords. Greater clarity and for the most part, bit better bass except a bit of warmth (which I do like) was lost. There something to those older Shunyata Python/Anaconda/King Cobra cords and the bit of warmth they add that I find welcoming.
Not only does the main power cord to the entire rig make a difference as the first cord in the chain, that changes everything up stream. It makes everything sound better, not surprisingly. If had throw away cash, I run out and buy a Shunyata Omega XC (QC?) cord or that really killer Synergistic Galileo SX or SRX power cord that is nose bleed costly, at least for humble me. Have a prosperous and Happy New Year’s all! 😁
No the first graph is better as the power level, and hence volume, of the sound does not change with frequency.
In the second graph, the power goes up and speakers will get louder ~75 Hz and 1-2kHz and have a dip at ~5kHz so you would not attain a flat frequency response with that amp with that speaker load.
Note that the same "real life" speaker load is used for both amps.
So the second graph is better with respect to the line that’s more wavy. Which means it’s more responsive with a more accurate delivery of Dynamics as the music demands. A lower level of Dynamics, flatter line, is expected when impedance is lower?
If you people would read the measurements section of Stereophile, you would see that I’m correct.
Look at the frequency response for the Moon 861. Remember, we are talking power amps here, so no preamp functions (e.g., loudness) enter into the discussion.
And the discussion:
The output impedance in stereo mode was extremely low, at 0.008 ohms at 20Hz and 1kHz, rising slightly to 0.03 ohms at 20kHz. As the two output stages are in series in mono mode, the output impedances were twice the stereo values. In both stereo and mono modes, the variation in the frequency response with our standard simulated loudspeaker (fig.1, gray trace) was negligible.
Now look at the BAT REX 500 frequency response:
And the discussion:
The REX 500 amplifier’s output impedance, including the series impedance of 6’ of spaced-pair cable, was relatively high, at 0.45 ohms at low and middle frequencies and 0.6 ohms at the top of the audioband. As a result, the variation in the frequency response with our standard simulated loudspeaker (fig.1, gray trace) was ±0.3dB.
So while the response is flat into a load resistor, it varies by ~+-0.3 dB into a real life speaker load. That’s enough of a difference to hear as a difference. And of course every speaker is going to have a different load, so the amp is going to have a different response for each speaker dependent on that load.
Know that a lot of amps show considerably more variation than this.
I love it when you guys talk harmonics. I think you guys should get together for a play date.
I think the harmonics area is underserved when people evaluate amplifiers. I like tube amplifiers because they seem to emphasize upper and lower harmonic areas that are underserved by the solid state amps.
I wish I better understood output impedance. But these days most components get along fine. Would love to hear examples where they don’t get along fine and maybe I should learn more about impedance.
@emergingsoulThe secondary thing is the distortion signature of the amp, which audiophiles call the 'sonic' signature. This is in turn caused by something called the 'transfer characteristic' of the amp. The transfer characteristic has something to say about how the amp responds to transients, in particular those that overload the amp. It also says what sort of distortion will be produced, for example if based on a quadratic exponent, the 2nd harmonic or a cubic exponent, resulting in a dominant 3rd harmonic.
These two harmonics are treated by the ear much the same way in that they are innocuous. But intermodulation distortion is not and also plays an important role that is defined by the transfer function.
If you want to know more about this, read this article which starts at page 35 at the link. It might be more than you want to know, but it does answer your question correctly.
The frequency response of an amplifier will vary with the output impedience when pushing a real life speaker load, and not some fixed value resistor used to determine the power.
The lower the output impedience, the flatter the frequency response across the band. An amp with a high output impedience may vary by several decibels across the band when pushing a real life load. You can't get around Ohm's Law.
You will hear a difference of several decibels across the band far more than the difference between say 0.01% and 0.001% distortion, or...
Take two SS amplifiers that produce a perfectly linear response from 20Hz to 20kHz. How is it they can sound different? My hypothesis is the answer lies in the amount and type of harmonic distortion, and what mix of harmonics dominates the distortion profile. Conventional class AB SS amps tend to be low in distortion overall, with the third harmonic the highest in level. That’s why it’s often difficult to tell them apart in a blind, level-matched comparison.
The less conventional sliding bias and zero global feedback designs (Threshold, Coda, Ayre etc) that produce more “bloom” in the midrange tend to have a more complex distortion profile in which the second harmonic is often nearly as high or even higher in amplitude than the third. I think it’s for that reason that such amps can sound remarkably similar to a push-pull AB tube amp. The same applies to some class D amps. A couple class D amps I recently had in house (Bel Canto E1X and Axxess Forté 1), sounded remarkably similar in the high freqs to an average KT88 or EL34 push-pull amplifier. Where they differed from a typical tube amp is they didn’t have the elevated midrange and upper bass response. Turns out those amps have higher 2nd order distortion than a typical SS amp, especially the Forté 1, which happens to be 2nd order dominant according to one magazine’s bench test.
Parasound’s John Curl amps also tend to be second-order dominant. IME, some of that midrange bloom comes through in their sound when matched with a transparent preamp. Though they are generally low in distortion and noise, they sound considerably warmer than an amp like the Benchmark AHB2. After living with an AHB2 and many other designs, I have come to the conclusion that our ears (if still healthy) are far more sensitive to distortion and noise than was claimed by many industry objectivists. That isn’t to say it’s the best sounding amp subjectively, rather, that you can hear the difference when A/Bd against more conventional designs.
Personally, I find that relatively affordable amps can be remarkably good these days, when they have the right synergy with the paired speakers. The amplifier sound quality/cost ratio is quickly catching up to that of DACs and sources. We are already past the days where one need spend the equivalent of a good used car to get true high performance amplification.
@davetheoilguy, the heart perfuses all of the organs, including itself and the liver; the liver will die if it is not perfused for a long enough period of time (cardiac muscle will die a lot quicker from absence of perfusion).
When heart rhythm that produces a pulse ceases, one is clinically dead. Liver function can cease but that does not meet the criteria for clinical death. Liver cancer can have a favorable prognosis, and if it doesn’t you will still have time to get your affairs in order as quality of life declines. On the other hand, if you go into cardiac arrest, if you are not in the right place and around the right people your life expectancy usually drops to under ten minutes.
All the organs were deciding who should be the boss....
"I should be in charge," said the brain , "I run all the body's systems, without me nothing would happen."
"I should be in charge," said the heart , "I circulate oxygen and nutrients all over."
"No! I should be in charge," said the stomach, "I process the food that gives us energy."
"I should be in charge," said the legs, "without me the body couldn't go anywhere."
"I should be in charge," said the eyes, "I allow the body to see where it goes."
"I should be in charge," said the asshole, "I am responsible for waste removal."
All of the other body parts laughed at the asshole and insulted him. So he shut down. Within a few days, the brain had a terrible headache, the stomach was bloated, the legs got wobbly, the eyes got watery, and the heart pumped toxic blood. They all decided that the asshole should be the boss.
What is the moral of the story? Even though everybody else does all of the work some asshole is usually in charge.
Reminds me a bit of the old story where organs of the body are having a fight over who was most important. Can’t remember what the plot twist was — something like the little toe saving the day.
But the moral was the organs make up a team, all of which is “most important” depending on what is going on.
@davetheoilguy when you go into cardiac arrest, I think the answer to the above will be clearer.
The cathodes are one of very important issues in SET tube amplifiers.
Some people use a fixed bias solution for the SET output stage. But there is the same issue in the input and driver stage and most designers do nothing to solve this problem. The problem is there are 2 signal paths (input and output) that go through each cathode. There is a Lynn Olson article about it.
There are 3 solutions: using fixed bisa, using batteries and using huge value capacitors (DIY enthusiast Yuri Makarov is another of this 3rd solution).
I tried the huge value capacitor solution (100,000uF with AN Kiasei NP 50uf bypass) in my SET amplifier first stage and driver stage.
The result exceeded my expectations. Much bigger 3D soundstage, better instrument separation, deeper and more controlled bass.
The most difficult was the solution for the output tube cathode. I afraid to use fixed bias. I think it is not safe. I also can't use 100,000uF because it is a strong output tube current for a long period of time after switching on while big cathode capacitors are charging. So I put 7000uf + 50uf AN Kaisei + 50uF Kemet DC-Link. It worked but I could hear the drawbacks of Kaisei capacitors. So I added Siemensk MKV 5uf + Duelund Cu-Sn bypass + 6000ps SGM3 silver mica.
In addition to my cathode experience,
I assembled a new 300B amplifier with external power supply. Most parts were reused from my previous amplifier but the basic break-in took 2 month. After that I started to recognize whatever I tried to do and tweek the sound of my SET amplifier was always sterile. And it was very strange for a 300B set amplifier.
In the end I decided to change the driver tube from 6f6 to 6v6. And I have to change the cathode resistor to increase 6v6 idle current. I bought a new AN tantalum resistor for it. At the same time I decided to change the input stage (6sn7) cathode resistor from Vishay Z-foil to the Shinkoh tantalum resistor that I used previously in the driver (the value of the resistor is matched). The difference between driving tubes 6f6, 6v6, 6L6 is another story, BUT since I changed Vishay Z-foil to Shinkoh, my amplifier no longer sounds sterile anymore.
About twenty years ago I bought a pair of old, used Altec-Lansing 1568A mono amplifiers in nice shape, they had spenent their lives in a rack.
Those 1568A’s are simple, lightly stressed designs, made to run 24 hours a day, using two EL34 tubes in a push-pull configuration to deliver a clean 40 watts.
They also have some of the finest transformers ever put in an amplifier, second only to the legendary output trannys found in the Harman-Kardon Citation II.
Any tube amplifier made today, regardless of price, except for some insanely expensive Ongakus, has transformers that cannot hold a candle to those on those Altecs and the H-K Citation.
But the Altecs also had anemic and primitive power supplies. Too simple.
Upgrading a tube amplifier power supply is really not too difficult, and these had plenty of room to work in.
So I went to town on these, replacing the PS caps with new oversize electrolytics with poyptopylene bypass caps.
I also also added a filter choke. A slight hum disappeared completely, All the signal caps were updated with Polypropes. All the resistors were replaced with new carbon films
Finally two NOS sets of Philips EL-34’s finished the job.
Those old Altecs put my McIntosh MC-60’s to shame. Greater slam in the bass, smoother mids, and the highs were just plan gorgeous, silky and detaled, into my JBL C50/S7 speakers.
I soon sold the Macs for $2000, and the total cost of those Altecs including the mods was about $700.00.
Downside: These Altec 1568A amplifiers are just plain ugly to look at.
Just put them somewhere you don’t have to see them.
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