Slew rate and rise time

Hi, I just wanna ask if slew rate is the same as rise time. Which of the parameters is used to describe how fast an amp responses to input signal? Is there a minimal slew rate or rise time value which is still compatible with good sonics, or are sonics not depending on these values?
They are two different ways of looking at the same thing.

Slew rate is a measure of how fast a signal is changing. How many volts it changes in a given period of time. Usually stated in volts per microsecond.

Rise time tells you how long it takes to get from 10% of the peak to 90% of the peak and ignores the beginning and end of the signal. If you had a signal that went from 0 to 10 volts, the rise time would be how long it took to get from 1 to 9 volts.

To use an automobile analogy: Slew rate would equal mph and rise tine would be the time it took to travel a certain distance. You can calculate one from the other.

Both are related to the maximum frequency a device can reproduce. If the slew rate is too low it will not be able to reproduce the higher frequencies. This is rarely a limiting factor in audio devices although some feel that the higher the slew rate the better the device.
Herman makes some good points, but i'd like to take his explanation a bit further.

Think of an electrical circuit as a hose for electrons. Then think of the music as a fire. Different size hoses can deliver different volumes of signal at different pressure levels. Volume and pressure are different things, so don't confuse the two. Fires also burn at different rates in different areas, making some harder to control than others. Music is much like a fire as you have different intensities in different parts of the audible spectrum, all taking place simultaneously.

Think of how fast the signal can make it through the hose ( circuit ) from one end to the other on demand. This is a measure of accelleration and is equivalent to the rise time. In other words, we are clocking how fast the circuit can go from just above a trickle of flow to almost wide open flow. Obviously, when we need water ( electrons ) to put out a fire ( reproduce the signal ), we need to get it there fast. The faster that the circuit can deliver maximum flow, the higher the rise time. The higher the rise time, the more responsive we are to putting out the fire. There's one problem here though.

What happens when we have a hose of a given speed, but different size fires? Hmmm.... Obviously, a garden hose could be used to put out a fire, but whether or not it could keep up with the demand of flow required for a very large fire that changes intensity or direction rapidly is another story. This is where slew rate comes into play.

Slew rate is equivalent to how much flow the hose ( circuit ) is capable of. The bigger the hose ( higher the slew rate ), the more volume we can flow. The more volume that we can flow, the larger the fire ( signal ) that we can put out. After all, it is possible to get water onto a fire in rapid fashion ( fast rise time ), but if the fire ( signal ) is bigger than the volume of water ( limited slew rate ) that we can deliver, that fire ( signal ) goes at least partially unquenched.

The end result would be that our fast response time ( fast rise time ) wouldn't be enough in itself to handle the amount of fire ( signal ) due to a lack of volume of flow ( slew rate ). In effect, we not only need to be able to deliver the signal fast, but we must also be able to deliver the quantity of signal needed as the situation varies.

With all of that in mind, rise time and slew rates are not one in the same and shouldn't be thought of as being directly tied together. While one can derive horsepower figures by looking at the torque rating of a motor at a given rpm, the horsepower and torque curves do not run parallel to each other at any given time. Such is the same with slew rate and rise time. Obviously, rise times and slew rates are two different measures of a circuit i.e. speed and capacity. Having good performance in one area without the other means a limitation in performance somewhere down the line though.

In most cases, a circuit that has a faster rise time will demonstrate a higher slew rate by design. That is, if the engineer / designer is smart enough to take both aspects of circuit performance into account. The faster the rise time and the higher the slew rate, the more responsive the circuit is to any given signal and the less challenged it will be in trying to reproduce that signal. Combining great speed and agility ( fast rise time ) with a capacity for brute force ( high slew rate ) makes for a very well rounded performer.

There are many other spec's that are taken for granted when it comes to circuit speed and finesse that never get mentioned. That is, just as a circuit has a rise time ( how fast it can go up the hill ), there is a fall time to ( how fast it can come down from the peak ). Most circuits offer pretty symmetrical levels of performance here, but not always.

If symmetry is lacking in this area, the peaks will tend to be reproduced TOO avidly, resulting in overshoot. When you get a high peak that wasn't meant to be that sharp, the sound gets "edgy". This has to do with a lack of damping i.e. too slow of a fall or "recovery" time. To continue on with our hose analogy, we can get the water to the fire in both the time and quantity needed, we just couldn't regulate the quantity of flow from the hose before the water itself created further damage. This is sometimes referred to as excessive leading edge energy when one can find a qualified reviewer. They might not understand the what's & why's, but they can hear it taking place : )

Another overlooked spec is Td or the Time Delay of the circuit. Like the hose mentioned above, different circuits have different length paths to them. A shorter hose or circuit can deliver what we need faster with less potential for delay whereas a longer hose or circuit will take longer to deliver the goods. Much like a hose, a longer circuit may end up having more losses along the way with greater variances in flow due to all of the various connections made.

For best results, we want to keep the path short and as simple as is feasible. This results in mininal Time Delay with the least potential for loss or smearing. Most circuits don't do this and the end result is not only a loss in signal, but also timing. The fact that a circuit can have different loss rates at different frequencies along that path can really play games with what we hear. That's because not only can the timing be altered as frequency varies, but the amount of loss can differ as frequency is varied too.

This has to do with dielectric absorption of the parts and why changing passive parts like resistors, capacitors, inductors, wiring, component jacks, etc... can change the sonic presentation that we hear. Not only in terms of tonal balance, but transient response, speed, coherency, spectral purity, etc... These effects also take place in active parts like transistors, diodes, tubes, etc, too. That's why "doping" or the application of chemicals to the external parts of the device can change the electrical and sonic characteristics, etc...

While one could conclude that a longer path is both slower and more lossy ( logical conclusion ), there are also other factors here too. Not only does negative feedback increase the complexity and parts count of the circuit, it also increases the length of the path and the response time of the circuit. The more negative feedback that you have, the more time delay that you'll have. The more time delay that you have, the more loss that you have of various time and frequency related artifacts.

This is why high negative feedback designs sound less "liquid" as they suck the life out of the music. That is, due to the above mentioned non-linear losses with increased parts count and circuit complexity, the timing, harmonic structure and transient response are all mangled to a much higher degree. This makes the music sound lifeless, even though most of the distortion type spec's look great on paper.

This is where the design prowess and listening skills of the manufacturer come into play. It is also the reason why products that are built only to measure well don't sound all that good i.e. the "distortion wars" that resulted in all of the horribly hard, bright and sterile sounding SS gear of the past. That is, the engineers didn't count on all of the negative sonic aspects of electrical error correction and circuit complexity that are involved, nor the non-linearities of the low quality parts that they were using. They assumed that the end would justify the means, which in reality, it really did. Unfortunately, the end sums up the means and the means weren't well thought out or properly executed to begin with. That's why they had to rely on so much negative feedback to get the job done.

This is also why products that can measure similarly don't always sound alike i.e. two different design routes can produce different complexities of circuits with different parts used. As such, the sonics of the circuit are even more important than the measurable accuracy that most engineers base their designs on. This is why many circuits that measure poorly actually sound pretty good. That is, the distortions and non-linearities that they introduce are actually not as detrimental as the means used to achieve some of the better measuring, but poorer sounding gear.

In effect, the "PRAT" or "musicality" of the circuit are directly correlated to all of the above. While there is no real way to quantify the term, "PRAT" is a term that takes into account both the musical accuracy and the electrical accuracy of the circuit. That is, musical notes don't sound like musical notes, whether it is due to non-linear circuit design, using low grade parts, relying on error correction circuitry to compensate for lossy parts and / or non-linear circuitry or any combo of the above. The end result is always audible.

Obviously, there are different levels of PRAT based on how well the circuit does all of the above and preserves the signal being fed into it. With that in mind, once you've heard a system that has even a smidgen of PRAT, you'll know the difference between a "sound system" and a "music reproduction system". One reproduces sounds that emulate music and the other reproduces music. The former is what i refer to as measurable accuracy and the latter is what i refer to as accurate musicality. Sean

PS... I did this at 5 in the morning after waking up in the middle of the night. Please cut me some slack if it's a little lacking in flow and / or specificity : )

Thx for explanation Herman.
Sean, excellent "Article." I'm going to keep it for a reference. I have never seen all of this subject matter put together in an understandable format with the relationships such as this. You have again increased my knowledge of what I hear vs design parameters. I knew what I have heard but the why has been in some question.
Bigtee: There have been questions here that i've wanted to respond to in this fashion and i've had private requests to try and explain this too. This specific thread caught me at the right time and in the right mood to attempt this type of "article". I was able to write all of this in about 45 minutes ( including additions and revisions ), so it wasn't as bad as i had thought it was going to be. Obviously, it is not all inclusive, but it should help explain a few things and give people that are interested in learning something to work with.

This might be a good thread to keep going with other technical questions pertaining to spec's, measurements and what we hear. This would give newcomers a point of reference as to what spec's are, how to interpret them, how they interact with each other, etc... Only problem is, getting people to post the pertinent questions & comments within this thread and then keeping those that shouldn't be posting comments & questions in this thread away from it.

I don't have a problem with people presenting alternative points of view, but i do have a problem with those that present information in dogmatic form that they've read elsewhere without having any first hand knowledge on the subject and / or those who refuse to either consider alternative points of view and / or won't learn on their own.

Having said that, most all points of view have some form of validity to them and the key factor is being able to fit all the different pieces that make up the puzzle together in a cohesive manner. Too many people want to grab hold of one aspect of operation and / or theory at the expense of how that part of the equation factors into the sum product. The end result is a partial understanding that is based on fact, which is what makes it believable, but may not be completely applicable when one considers all the factors involved in the grander scheme of things.

This is why i've said that spec's can be used as a very useful tool, but only if the spec's are valid and the type and quantity of spec's are suitable for a meaningful interpretation.

The only variables beyond the spec's would be the actual parts used and the lay-out of the circuit, which can surely present some variables to the sonic outcome of any given product. None the less, and as i've shown on several different occassions, "reasonable guesstimates" of what to expect sonically are possible even with those variables entering into the equation.

Part of this has to do with common sense. If one starts off with a junk platform, even the best parts and excellent execution of the circuit can't make it an awesome performer. On the other hand, even an excellent design that is saddled with poor parts and / or less than optimum execution isn't going to perform the best either. That's why we end up with so many mediocre products i.e. all the factors aren't considered evenly and the end result is a "happy medium" that the bean-counters can live with. Sean
These numbers won't tell you squat about how it sounds.
Thank you Sean for your descriptive analogy. I'll keep it also. You know, another issue for discussion is also what the relationship is between gain, volume and amount of power that an amp delivers. Let's say an amp is capable of delivering 200wpc into 8 Ohms. Is this with the volume knob wide open or doesn't it relate to the position of the volume knob? But this is as I say another issue to discuss. I'll start a thread about this.
Sean, your analogies are very colorful, one might say even fiery, but I am afraid they are not exactly right. Given the fact that part of what you say is correct and some of it isn't makes it hard to follow.

"Think of how fast the signal can make it through the hose ( circuit ) from one end to the other on demand. This is a measure of accelleration and is equivalent to the rise time." No, how fast it makes through the hose would be propogation delay. The signal could take all day to make it from one end to the other and still have a very fast rise time once it got to the output. The two are unrelated.

"Slew rate is equivalent to how much flow the hose ( circuit ) is capable of." You also refer to it as "volume of water." No, this would be the equivalent of electrical current.

Slew rate, by definition, is simply the maximum rate of change of the output voltage for all possible inputs. In simple terms, how fast the voltage can change.

Mathematically it is the slope of the line on a graph plotting voltage versus time. Take the amount the voltage has changed, divide it by the time it took to change, and you have the slew rate, usually expresed in volts per microsecond.

Since music is made up of sine waves, and higher frequency sine waves have faster rates of change (the fastest being where it crosses zero) at some frequency this rate of change will exceed the slew rate of the amplifier. It will distort this frequency and all of those higher.

Therefore, we can say that the bandwidth of an amplifier is slew rate limited.

Rise time is how long it takes the voltage to get from 10 to 90 percent of it's peak as I stated above. You are correct that there is also a fall time and it can be different.

I admit I may have oversimplified the relationship between risetime and slew rate. Even though you can calculate a rate of change based on the risetime it is not the same as the slew rate. It is possible that a circuit can have a high enough slew rate to handle a signal but still be unable to do so because it is limited by the rise time, and vice versa.

From a practical point of view it makes more sense to talk about rise time with digital circuits since they are always switching between 2 voltage levels. For instance, a zero represented by zero volts and a one represented by 5 volts. The rise time would be how long it takes to get from .5V to 4.5V. The falltime how long it takes to get from 4.5V to .5V. In an analog circuit that has an infinite number of different levels it probably makes more sense to describe it in terms of slew rate.

I could go into a long winded mathematical explanation of the two and show how they are both related to the RC time constants of a circuit, and therefore both are limiting the bandwidth of a circuit, but that is probably more than is needed and it is very difficult to do mathematical equations on this forum.
Sean, I think Herman is right:
But I also understand your analogy, Sean. Is slew rate analogous to a car's acceleration speed and rise time to the time a car needs to go from 0 to 90 mph? The slew rate and rise time are a bit related to each other, but if plotted a device with a certain rise time can have another slew rate if compared to another device with the same rise time.
The term acceleration is not used correctly in Sean's analogy. The correct term would be velocity, but the analogy is still wrong. Velocity is how fast something is moving, how much distance you can travel in a given period of time. Acceleration is how fast the velocity is changing, you are moving but either speeding up or slowing down. Neither slew rate nor rise time have anything to do with speeding up or slowing down, only moving at a constant rate.

Velocity is a rate of change.
Acceleration is a changing rate of change.

If you want to use a speed analogy then slew rate is the velocity of the voltage. It is not related to the "volume" of the signal, at least I don't think it is because in 25 years I've never heard anyone talk about the volume of a signal except in terms of current.
Herman: Iin the grander scheme of things, the hose analogy was not perfect and your comments pertaining to propogation delay were correct. I think that most people that lacked such understanding would get the basic idea though, or at least i think that they would. For sake of reference, i did try to explain things further when i said that rise time referred to the transition time from small to large signal flow. I'm glad that you pointed this out though for sake of clarity and better understanding.

Herman's clarification about propogation of delay from the beginning to the end of the circuit ( i referred to it as Td or Time Delay ) vs rise time also brings another matter to light. When trying to discuss technical matters in a non-technical manner, it is easy to make things more confusing to the lay person. As i think that most of you know, that is not my goal at all. It is quite evident that i'm not a professional teacher though, so my wording or ability to convey some ideas may not be the best that could have been chosen for any given subject.

As such, i would encourage questions, comments and clarification as one feels the need. I would rather make sure that everyone feels comfortable and has a grasp on what we've covered. While we can't expect to cover every aspect of operation in a simplistic forum like this, i don't want to gloss over what might be important details and leave one confused about the subject for life. As such, those having questions and / or corrections should PLEASE post them in a timely basis. The longer that someone believes "mis-information" to be correct, the harder it is to get them to unlearn it. On top of that, further clarification and / or correction will typically lead to an even wider coverage of the subject, furthering the educational value of this thread and the resultant questions asked.

I ran into this problem with my business partner many years ago. His college professor used an extremely poor analogy that caused him to stumble when trying to understand a certain part of circuit design & troubleshooting. This analogy, and therefore this stumbling block, has stuck with him for life. I have had a helluva hard time trying to get him to understand why it was wrong and a better way to look at things. The fact that i'm "self-educated" had made it hard for him to believe that i could know more about the subject or how to teach it than his college professor did. None the less, the only way that i found out about this analogy was because he couldn't figure out what was wrong with a circuit. As such, i tried to explain how to find the problem using a different analogy than what he had been taught, which is what caused us to get into the semantics of the analogy and why he couldn't track the problem.

With that in mind, i don't want to be the one that teaches you folks the wrong thing that you have to "unlearn" at a later date, so please ask questions / add comments as different subjects arise. I and the others that typically contribute to such threads will do our best to try and explain things in an easy to understand but correct manner. So long as we stay on course and avoid personal conflict, i think that this thread could be very enlightening. Sean
I personally have read enough to understand Sean's analogy and even though "technically" it needed a "Slight" bit of refinement, I thought it was an excellent post allowing these parameters to be interpreted in an overall scheme. I also enjoyed Herman's post(s).
I would like to ask this, when looking at the effects of negative feedback in an amplifier, how does local feedback vs global feedback effect the overall musicality of the amp or overall performance (is there a compromise in here somewhere?)
I have seen a few amps that featured no global feedback but used local feedback and then, according to Charlie Hansen at Ayre, use no feedback, either local or global. Which type of feedback would be most audible or does it all equal up to about the samething?
I feel there is something to this feedback thing. Even though the no negative feedback amps do not offer up the "Slam" of feedback amps, they do seem to sound(for lack of a better description) more tube like. However, I can certainly see why someone might pass on the zero feedback amp if running it full range. The bass is a little light compared to some.
A really good example was when I compared the Parasound JC1's(39db of global feedback) to the 0 feedback Ayre V5xe amp (yes, I owned both and still have the Ayre.) Yes, the JC1 sounded more visceral but it also sounded more "Hi-fi-ish" overall. The mids and highs portrayed by the JC1 just didn't sound totally realistic. The Ayre was more liquid and harmonically rich. Also, the portrayal of the soundstaging cues seemed more fully developed through the Ayre (deeper stage cues vs a somewhat flatter stage presentation.)
I use subs, so the bottom end is not that much of a concern with me. Soundstaging, natural mids/highs and liquidity is.
Can this be directly related to the feedback or is it more of my personal preference to one sound over another or do I need to look at other parameters? I kinda would like to know why I hear what I hear or if it is truly just a personal thing and has nothing to do with design.
For general information, the amps were both broken in and used on 20 amp dedicated lines(2 in the case of the JC1.) I listened to both at around 85-90db or lower and my impressions stayed the same. I guess what I'm trying to say is the JC1 was kind of like a GTO, Camaro or whatever (lots of power but a touch short on refinement) where the Ayre was more BMW like(The Theta Dreadnaught also seemed this way-another zero feedback amp.)
BTW, this is not to disparage anyone who owns the JC1's but is related in context to design and its effects. This was my take over an extended period in my room with my equipment.
Local feedback is a correction network for only a small segment of the circuitry whereas global feedback compares the signal at the output of the amp to the input of the amp and tries to correct accordingly. As such, there is less to monitor and less to subjectively correct in the local feedback design. On top of that, by taking care of each mess individually at a local level rather than trying to fix everything collectively at one time, less total feedback can be used and that amount is spread out amongst the various gain stages. In effect, local would be deemed "faster" and "less detrimental" overall. To my ears, this is both audible and desirable from a sonic standpoint.

The drawback here is that the output impedance of the amp will always be higher when using less or no feedback in the output stage. The output impedance being higher directly relates to a reduced damping factor. While some think that damping factor corresponds to how much control the amp has over the driver, that isn't really technically correct.

Damping factor corresponds to how much the speaker can modulate ( temporarily distort or even take control of ) the output of the amp. A higher damping factor means that this potential is reduced. While some would say that this equates to increased amplifier control, and it does in an indirect sort of way, the only thing that keeps the amplifier in control of the speaker is the ability for it to load more voltage and current into the driver than what the driver is generating in reflected EMF ( Electro-Motive Force or "voltage" ).

All drivers generate some amount of "electrical backlash" when a signal is applied to them, hence the term "reflected EMF". Larger drivers with more mass and motor structure and / or drivers that are making longer excursions generate more reflected EMF. That's why the theory goes that with big speakers, one needs a big amp. A big amp typically has more voltage and current potential than a smaller amp, so it theoretically should offer greater driver control too.

The thing with tube amps is that they typically have a very high rail voltage, which means more resistance to the reflected voltage from the amp. The only problem is that most tubed amps suffer from a lack of current capacity and have a higher output impedance. The lack of current can cause the output rails to sag, lowering the amount of voltage available to "ward off" or "fight back against" the reflected emf. On top of that, the higher output impedance makes the amp far more susceptable to having the output stage "modulated" by that same reflected emf. As mentioned above, the more excursion that a driver takes, the more reflected emf that you have to deal with. This is why many tubed amps sound "warm & rich" at low volumes but "loose & flabby" as spl & excursion are increased. As a side note, the more that the output stage of the amp is modulated by the reflected emf, the more likely distortion is to occur further up in the driver stages too. In amps with poor stereo separation, the one output stage can actually modulate the other channel of the amp, from beginning to end.

There are ways to play games with the output impedance on any given amp without introducing gobs of negative feedback though. The simplest and most common method is to use more outputs. Since the outputs are in parallel with each other, the output impedance of each device is reduced by a significant percentage each time another device is added to share the load. Only problem is, now you've got a ton of output devices, making it harder to get them all to work in unison at the same time.

In this type of situation, the poorer the matching of output devices, the more smearing of signal with a reduction in "blackness" between notes. Various types of distortion can climb too, so many amp designers will simply tack more negative feedback into the circuit to help keep things under control. In effect, one bad implimentation in the design requires more detrimental circuitry to cover it up. This is one of the main reasons why many people think that smaller amps sound better. They can have shorter signal paths, require less error correction and there aren't as many parts to worry about.

Another approach to lowering the output impedance of an amp is to lower the value of the emitter resistor. This increases the current flow through the output stage and develops more heat though. On top of that, if the output devices are running out of balance due to poor parts matching, the increased current will aggravate the situation and cause an early death. Like most things in life, free lunches are hard to come by and most everything requires some type of decision / design compromise.

As one can see, there is somewhat of a reason why we have the sonic differences that we do between no / low negative feedback amps and those that use quite a bit of it. Obviously, the best way to do this would be to build and impliment the best circuit that you could using all hand-matched parts of extremely high quality and then use feedback only as needed in very small quantities at the local level.

Wanna guess what the drawback is here ??? Cost and labor. This is why the amps that sound REALLY good tend to cost a LOT of money. While i'm NOT trying to justify the massive prices on some of this gear, you simply can't make a hand built and tuned component as cheaply as you can a mass produced component.

In terms of spaciousness of presentation, my experience is that low / no negative feedback amps ALWAYS sound more spacious. As mentioned above, feedback tends to slow down the response of the amp. That reduction in signal processing speed comes in to play when trying to reproduce the harmonic overtones of the original signal. The reduced harmonic overtones not only changes the timbre of the instruments, our ears use the time differences between the primary note and the harmonics to place the origin of each note within the soundstage. If we lose the harmonic structure of the note, we also lose spatial cues too.

One of the ways to get better bass out of any amp is to use an iron core transformer, a very short signal path and gobs of power supply reserve ( filter capacitance ). On amps that use output inductors, increasing the gauge of that inductor also allows it to pass more instantaneous current, giving it better bass slam. Having said that though, high frequency response can suffer, so you have to juggle the gauge vs desired bandwidth. This is why some amps are great for bass but less desirable for treble, etc... Many low / no feedback amps do not use an output inductor, which can be both beneficial in some areas and detrimental in others.

It has been my experience that amps that have all of these design attributes ( with a reasonably low output impedance ) all have pretty solid bass response. Lowering the cut-off frequency of the amp or DC coupling it is also a good thing in this regards, as the higher the cut-off frequency of the amp in the bass region, the less articulate the bass is due to ringing from the filters occuring.

Obviously, a lot of the above information is subjective at best and how much of it applies to any given amp or circuit depends on how that amp or circuit is designed, implimented and constructed. There are so many variables involved here that it could make your head spin. As such, i'm hoping that this has covered some of the basics of that area and will give you folks more to think about and digest. There is no "magic circuit" at this point in time, only design variations that each designer / manufacturer thinks is worth giving a chance. Sean

Makes sense. I know in the Ayre amp, Hansen does use a lot of handmatched output devices. On the other hand, I don't know how he gets the amps noise floor so low and the "Blackness" between the notes with this particular zero feedback design. I do know this is one of its strong suits along with a simply holographic soundstage. He recommends against using any power conditioner.
The other point is it clips like a tube amp, fairly gradual and the distortion figures are relatively high for a SS amp at max output (something along =>1% at max power.) The amp does double down and is stable into low impedances(a good thing for me since my speakers do dip near 3 ohms at a few points.)
Now, another question, I have heard that feedback alters the "Phase and time" relationships of the input signal(global feedback) if you look at what arrives at the output in comparison. Is this true? Is this what you are refering to above talking about delay? Would no neg. feedback amps pass a truer duplicate of the input signal(based on other parameters remaining the same?)
Just in passing, I was in the smaller amp camp until a year or so ago. I biamped using smaller amps. They just sounded better overall(provided they were high quality to start with.) It seems bigger amps have caught up now ( at reasonable prices) and some are truly excellent.