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Preamp Design Fundamentals
we are lucky on this forum to have folks like scott dorsey, john
hardy, dan kennedy, john lagrou, and others who are top pros in their field. i would like to take advantage of this, and request that you guys please give the rest of us a "primer" on preamp electronics design basics. 1. what is the simplest preamp design? ie, what exactly is required in the circuit and why? 2. what are the basic problems with the simplest possible design? (noise, fidelity, whatever) 3. what are the approaches to turning the basic design into a reasonably clean and useable mic preamp? (better parts, additions to the basic circuit, etc.) 4. what are the primary philosophies of top-end preamp design? ie, why does a grace preamp sound so different from an HV3 when they are both designed to be clean and transparent, or why tubes vs transformers for more "colored" preamp dsigns? |
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Preamp Design Fundamentals
jnorman wrote:
4. what are the primary philosophies of top-end preamp design? ie, why does a grace preamp sound so different from an HV3 You can read some of Michael Grace's thoughts here http://www.gracedesign.com/support/tech3.pdf |
#3
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Preamp Design Fundamentals
Harvey Gerst has an excellent discussion on just that topic over at
PSW, with some of the designers you mentioned participating. 14 pages of it: http://recpit.prosoundweb.com/viewtopic.php?t=7774 jp (jnorman) wrote in message . com... we are lucky on this forum to have folks like scott dorsey, john hardy, dan kennedy, john lagrou, and others who are top pros in their field. i would like to take advantage of this, and request that you guys please give the rest of us a "primer" on preamp electronics design basics. *snip* |
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Preamp Design Fundamentals
"Kurt Albershardt" wrote in message
jnorman wrote: 4. what are the primary philosophies of top-end preamp design? ie, why does a grace preamp sound so different from an HV3 You can read some of Michael Grace's thoughts here http://www.gracedesign.com/support/tech3.pdf I'm not saying it's all snake oil, but there's some really hilarious stuff the "By terminating the destination end of the output cable with a 300 Ohm resistor, the load resistance matches the output source resistance to create a 600 Ohm balanced transmission line. Matching the impedance at or near the characteristic impedance of the cable eliminates signal reflections in the wire which greatly improves transient performance and preserves harmonic integrity. This type of termination is very advantageous for driving very long lines because it provides maximum power transfer through the cable and virtually eliminates the effects of cable capacitance and inductance." |
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Preamp Design Fundamentals
"John Penovich" wrote in message
om Harvey Gerst has an excellent discussion on just that topic over at PSW, with some of the designers you mentioned participating. 14 pages of it: http://recpit.prosoundweb.com/viewtopic.php?t=7774 Most significantly it lays the groundwork for this excellent link: http://www.mil-media.com/docs/articles/preamps.shtml |
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Preamp Design Fundamentals
jnorman wrote:
we are lucky on this forum to have folks like scott dorsey, john hardy, dan kennedy, john lagrou, and others who are top pros in their field. i would like to take advantage of this, and request that you guys please give the rest of us a "primer" on preamp electronics design basics. 1. what is the simplest preamp design? ie, what exactly is required in the circuit and why? It has to have a differential input and it has to have gain. I think you can do that with a single long-tailed pair, or with a step-up transformer and a single transistor stage. 2. what are the basic problems with the simplest possible design? (noise, fidelity, whatever) You get noise, you get poor linearity, you don't get enough gain. 3. what are the approaches to turning the basic design into a reasonably clean and useable mic preamp? (better parts, additions to the basic circuit, etc.) Constant current sources, for one thing. Trying to get less gain per stage and getting more stages. Using higher voltage rails for more headroom. Using follower outputs for more current drive. Take a look at a typical tube preamp, though. Two, sometimes three gain stages. A lot of the gain is from the input transformer. It's pretty simple, and you can do the same thing with FETs if you want. 4. what are the primary philosophies of top-end preamp design? ie, why does a grace preamp sound so different from an HV3 when they are both designed to be clean and transparent, or why tubes vs transformers for more "colored" preamp dsigns? You can make very transparent tube preamps too. The problem is that there are widely varying philosophies. Since you cannot make anything completely transparent (especially given the widely varying loading requirements of different mikes), you are stuck with making tradeoffs between different kinds of coloration. And different people make different tradeoffs. --scott -- "C'est un Nagra. C'est suisse, et tres, tres precis." |
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Preamp Design Fundamentals
Arny Krueger wrote:
http://www.mil-media.com/docs/articles/preamps.shtml Thanks, couldn't find that one earlier. |
#9
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Preamp Design Fundamentals
jnorman wrote:
1. what is the simplest preamp design? ie, what exactly is required in the circuit and why? I think the simplest possible preamp would be an input transformer followed by a single unity-gain active stage. The transformer would have to have a high turns ratio in order to provide all the gain necessary. The active device would have to have a nearly infinite input impedance and an output impedance near zero. 2. what are the basic problems with the simplest possible design? (noise, fidelity, whatever) The transformer will have a limit to its operating level and bandwidth, will cause distortion, and will be bulky and expensive. The high ratio required to obtain the necessary gain will compound every one of these limitations. The active device will require peripheral components including a power source, and will have limits to its linearity and bandwidth, in addition to adding noise. There is no control over the gain, and the output signal is unbalanced. There are no utility features such as input pad to prevent overload, polarity switching, phantom power, etc. The preamp will be susceptible to interference from environmental noise, be it sound pressure, electrical, or magnetic. 3. what are the approaches to turning the basic design into a reasonably clean and useable mic preamp? (better parts, additions to the basic circuit, etc.) Adding more active stages to allow for a lower-ratio transformer. Adding a gain control or an input attenuator. Adding still more gain stages to compensate for losses in the attenuator or gain control. Adding a differential amplifier or transformer for balanced output. Adding contol over polarity, an input pad, and phantom power. Housing the whole thing in an enclosure to protect it from damage and noise. Adding a control surface to allow the user to easily and effectively manipulate all the features you just added. Some connectors to allow quick and easy interfacing with different microphones and destinations. 4. what are the primary philosophies of top-end preamp design? ie, why does a grace preamp sound so different from an HV3 when they are both designed to be clean and transparent, or why tubes vs transformers for more "colored" preamp dsigns? Dunno. ulysses |
#10
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Preamp Design Fundamentals
I've decided that, for me at least, as long as the mike preamp is
considerably better than the microphone, there's little to gain from using an expensive one. The preamps that come with a Mackie mixer are better than my mikes. Are yours? Cheers, Norm Strong "Harvey Gerst" wrote in message ... (John Penovich) wrote: Harvey Gerst has an excellent discussion on just that topic over at PSW, with some of the designers you mentioned participating. 14 pages of it: http://recpit.prosoundweb.com/viewtopic.php?t=7774 Some of the participants in that discussion of preamp design include Geoff Tanner, Dan Kennedy, Mark McQuilken, John Le Grou, Tim Ferrant, and John Hardy. Harvey Gerst Indian Trail Recording Studio http://www.ITRstudio.com/ |
#11
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Preamp Design Fundamentals
"normanstrong" wrote:
I've decided that, for me at least, as long as the mike preamp is considerably better than the microphone, there's little to gain from using an expensive one. The preamps that come with a Mackie mixer are better than my mikes. Are yours? Cheers, Norm, Generally speaking, we use the board's preamps for most things. For certain mics and/or certain sounds, we use outboard preamps to get what we want. I'm not a big fan of Mackie preamps. When we were looking around for our first decent board for the small studio, we passed on the Mackie, Behringer, Soundcraft, and several other boards, in favor of the Soundtrac Topaz. Not only because the Topaz had more features and a better EQ (IMHO)than the other boards, but because it had better sounding preamps, to my tired ears, anyway. As far as your statement that "The preamps that come with a Mackie mixer are better than my mikes", I don't understand that at all. The Shure SM57 comes alive thru a really nice preamp. The Mackie does nothing for the SM57. Are you saying that you don't even have an SM57 that would benefit from a better preamp? Or are you saying you don't even have a mic as good as an SM57 in your mic collection, so it doesn't matter to you? Harvey Gerst Indian Trail Recording Studio http://www.ITRstudio.com/ |
#12
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Preamp Design Fundamentals
normanstrong wrote:
I've decided that, for me at least, as long as the mike preamp is considerably better than the microphone, there's little to gain from using an expensive one. The preamps that come with a Mackie mixer are better than my mikes. Are yours? Compare the sound of an SM-57 into a Mackie and then into the Great River. Very substantial difference, mostly due to loading it's true, but a very substantial difference. --scott -- "C'est un Nagra. C'est suisse, et tres, tres precis." |
#13
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Preamp Design Fundamentals
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#14
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Preamp Design Fundamentals
"Harvey Gerst" wrote in message ... "normanstrong" wrote: I've decided that, for me at least, as long as the mike preamp is considerably better than the microphone, there's little to gain from using an expensive one. The preamps that come with a Mackie mixer are better than my mikes. Are yours? Cheers, Norm, Generally speaking, we use the board's preamps for most things. For certain mics and/or certain sounds, we use outboard preamps to get what we want. I'm not a big fan of Mackie preamps. When we were looking around for our first decent board for the small studio, we passed on the Mackie, Behringer, Soundcraft, and several other boards, in favor of the Soundtrac Topaz. Not only because the Topaz had more features and a better EQ (IMHO)than the other boards, but because it had better sounding preamps, to my tired ears, anyway. As far as your statement that "The preamps that come with a Mackie mixer are better than my mikes", I don't understand that at all. The Shure SM57 comes alive thru a really nice preamp. The Mackie does nothing for the SM57. Are you saying that you don't even have an SM57 that would benefit from a better preamp? Or are you saying you don't even have a mic as good as an SM57 in your mic collection, so it doesn't matter to you? I have 3 AKG C451's, 2 Nak CM-300's and 3 RadShack PZM's. No Shure mikes. So far as I can tell, mikes vary in noise level, frequency response and output impedance. As long as the preamp has the right impedance level for the mike used, has a smoother and more extended frequency response than the mike, and adds little noise to the self noise of the mike, it should be good enough. Anything else falls into the category of mystery; I don't do mystery. If you feel I've missed something pertinent, please feel free to correct me. Norm Strong |
#15
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Preamp Design Fundamentals
"normanstrong" wrote in
news:gg9tb.151521$ao4.482861@attbi_s51: I have 3 AKG C451's, 2 Nak CM-300's and 3 RadShack PZM's. No Shure mikes. So far as I can tell, mikes vary in noise level, frequency response and output impedance. As long as the preamp has the right impedance level for the mike used, has a smoother and more extended frequency response than the mike, and adds little noise to the self noise of the mike, it should be good enough. Anything else falls into the category of mystery; I don't do mystery. If you feel I've missed something pertinent, please feel free to correct me. I have 451's and traded up from Mackie to Great River preamps. There is an audible difference, particularly at quiet levels. Things that better preamps do better (that you can easily measure): * More headroom. Lets you stretch beyond 20 dB gain without breaking up * Better loading. Get more out of dynamic and ribbon mics. * Better phantom power. Produces more gain and runs more mics. If you're recording a loud, noisy source on a single condenser mic close up, the Mackie will sound almost as good as a Great River. What you can't easily measure is the lower distortion and noise levels that let you pull more detail out of the microphone. Anything with subtle nuances, like a single voice in a quiet studio, benefits from the clarity a better preamp produces. The difference is more noticeable when the source being recorded is familiar to the listener (again, like the sound of a human voice). Accuracy counts. |
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Preamp Design Fundamentals
Carey Carlan wrote:
"normanstrong" wrote in news:gg9tb.151521$ao4.482861@attbi_s51: I have 3 AKG C451's, 2 Nak CM-300's and 3 RadShack PZM's. No Shure mikes. So far as I can tell, mikes vary in noise level, frequency response and output impedance. As long as the preamp has the right impedance level for the mike used, has a smoother and more extended frequency response than the mike, and adds little noise to the self noise of the mike, it should be good enough. Anything else falls into the category of mystery; I don't do mystery. If you feel I've missed something pertinent, please feel free to correct me. I have 451's and traded up from Mackie to Great River preamps. There is an audible difference, particularly at quiet levels. In part this is because of the output transformer in the 451. It needs to see just the right input impedance to be happy. Things that better preamps do better (that you can easily measure): * More headroom. Lets you stretch beyond 20 dB gain without breaking up * Better loading. Get more out of dynamic and ribbon mics. * Better phantom power. Produces more gain and runs more mics. I agree that if a preamp has the right impedance for the mike use, a smooth and extended frequency response, and adds little noise, AND has low distortion (and therefore little coloration), it should be good enough. The problem is that very few mike preamps fall into that category, if any., What you can't easily measure is the lower distortion and noise levels that let you pull more detail out of the microphone. Anything with subtle nuances, like a single voice in a quiet studio, benefits from the clarity a better preamp produces. The difference is more noticeable when the source being recorded is familiar to the listener (again, like the sound of a human voice). Accuracy counts. Yup, that stuff CAN be measured, and it will show up very easily on a straightwire test. It is AMAZING to see the difference between a Mackie and a Great River in terms of distortion spectra. The Great River has a few big peaks, the Mackie has lots of little ones going way up the chart. --scott -- "C'est un Nagra. C'est suisse, et tres, tres precis." |
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Preamp Design Fundamentals
jnorman wrote:
we are lucky on this forum to have folks like scott dorsey, john hardy, dan kennedy, john lagrou, and others who are top pros in their field. i would like to take advantage of this, and request that you guys please give the rest of us a "primer" on preamp electronics design basics. 1. what is the simplest preamp design? ie, what exactly is required in the circuit and why? I'm going to rule out transformers because of cost. The circuitry may be simple but it won't find its way into a practical 'simple preamp' design. You need a differential input, typically a long tailed pair and a means of converting the differential output at the collector loads to a single ended signal - usually an op-amp suitably configured. 2. what are the basic problems with the simplest possible design? (noise, fidelity, whatever) Common general purpose transistors are noisy w.r.t. 200 ohm sources. Best choice is typically pnp ( higher carrier mobility in the base region results in lower flicker noise ) and large geometry ( lower intrinsic resistance has lower thermal noise ). Devices originally designed for moving coil pre-amp designs work very well. The input stage also needs to be run at several milliamps to get the benefit of the noise improvements because of the emitter dynamic impedance ( 27 / Ie ( mA ) Ohms, true up to a few milliamps ). Single transistor gain stages have poor linearity and generate plenty of THD esp at high signal levels. This is overcome by 'compound' arrangements or indeed using the op-amp stage to close a feedback loop to the input transistor emitters. There are more elaborate methods too. 3. what are the approaches to turning the basic design into a reasonably clean and useable mic preamp? (better parts, additions to the basic circuit, etc.) Better parts for the input transistors especially for noise. More advanced circuit design ( more bits ). 4. what are the primary philosophies of top-end preamp design? ie, why does a grace preamp sound so different from an HV3 when they are both designed to be clean and transparent, Don't know them so can't comment. or why tubes vs transformers for more "colored" preamp dsigns? That's easy. Some ppl like technically inaccurate designs since they produce a 'warm sound'. Valves produce little gain compared to transistors per stage, and rarely have much feedback applied around them, so a valve design will produce plenty of nice 2nd harmonic distortion. Transformers do all sorts of stuff to the signal. Kinda where do you want me to begin ? Regds, Graham |
#18
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Preamp Design Fundamentals
Arny Krueger wrote:
"Kurt Albershardt" wrote in message jnorman wrote: 4. what are the primary philosophies of top-end preamp design? ie, why does a grace preamp sound so different from an HV3 You can read some of Michael Grace's thoughts here http://www.gracedesign.com/support/tech3.pdf I'm not saying it's all snake oil, but there's some really hilarious stuff the "By terminating the destination end of the output cable with a 300 Ohm resistor, the load resistance matches the output source resistance to create a 600 Ohm balanced transmission line. Matching the impedance at or near the characteristic impedance of the cable eliminates signal reflections in the wire which greatly improves transient performance and preserves harmonic integrity. This type of termination is very advantageous for driving very long lines because it provides maximum power transfer through the cable and virtually eliminates the effects of cable capacitance and inductance." As classicly used by telephone companies ;-) Graham |
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Preamp Design Fundamentals
Carey Carlan wrote:
(Scott Dorsey) wrote in : What you can't easily measure is the lower distortion and noise levels that let you pull more detail out of the microphone. Anything with subtle nuances, like a single voice in a quiet studio, benefits from the clarity a better preamp produces. The difference is more noticeable when the source being recorded is familiar to the listener (again, like the sound of a human voice). Accuracy counts. Yup, that stuff CAN be measured, and it will show up very easily on a straightwire test. It is AMAZING to see the difference between a Mackie and a Great River in terms of distortion spectra. The Great River has a few big peaks, the Mackie has lots of little ones going way up the chart. --scott Do you have a preference on the few big peaks vs many small peaks? Yes. High order even harmonic distortion is what gave transistor amps the reputation for harsh and glassy sound back in the seventies. --scott -- "C'est un Nagra. C'est suisse, et tres, tres precis." |
#21
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Preamp Design Fundamentals
The Great
River has a few big peaks, the Mackie has lots of little ones going way up the chart. --scott Do you have a preference on the few big peaks vs many small peaks? Yes. High order even harmonic distortion is what gave transistor amps the reputation for harsh and glassy sound back in the seventies. High-order harmonic distortion is bad, period, but in general the high-order odd harmonics are worse than the evens. Ideally, you'd have a little bit of 2nd harmonic distortion, somewhat less 3rd harmonic, and nothing else measurable, at all levels up to clipping. (Well, semi-ideally; ideally, of course, there'd be none at all. But that doesn't happen in the real world, and in practice, what outlined above produces a nice clean-sounding preamp. You can do this with tubes, or discrete transistors, or -- somewhat more difficultly -- with ICs.) Peace, Paul |
#22
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Preamp Design Fundamentals
In article ,
Pooh Bear wrote: As classicly used by telephone companies ;-) Graham Yeah, over thousands of feet where it actually is a transmission line for audio frequencies. All audio is lumped parameters. Certainly anything to do with mic preamps. But it's a pretty common mistake. DC |
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Preamp Design Fundamentals
Dave Collins wrote:
In article , Pooh Bear wrote: As classicly used by telephone companies ;-) Graham Yeah, over thousands of feet where it actually is a transmission line for audio frequencies. All audio is lumped parameters. Certainly anything to do with mic preamps. But it's a pretty common mistake. And if it *was* a transmission line, it sure looks like a 300-ohm (not 600) line to me. |
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Preamp Design Fundamentals
S O'Neill wrote:
Dave Collins wrote: In article , Pooh Bear wrote: As classicly used by telephone companies ;-) Graham Yeah, over thousands of feet where it actually is a transmission line for audio frequencies. All audio is lumped parameters. Certainly anything to do with mic preamps. But it's a pretty common mistake. And if it *was* a transmission line, it sure looks like a 300-ohm (not 600) line to me. Uhuh, that too ! Graham |
#25
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Preamp Design Fundamentals
"Scott Dorsey" wrote in message
Yes. High order even harmonic distortion is what gave transistor amps the reputation for harsh and glassy sound back in the seventies. Two recent relevant AES papers are and Audio Engineering Society Convention Paper 5890 Auditory Perception of Nonlinear Distortion - Theory Earl R. Geddes 1 and Lidia W. Lee Audio Engineering Society Convention Paper 5891 Auditory Perception of Nonlinear Distortion Lidia W. Lee 1 and Earl R. Geddes Basically, they propose weighting nonlinear coefficients (e.g., harmonics) based on the cosine of the amplitude times order squared. The claim is made that the audibility of nonlinear distortion is inversely related to amplitude (nonlinearities at low levels are more audible) and also related to order squared (nonlinearities that generate spurious responses that are more broadly dispersed from the original signal are far more audible). They back their new metric up with listening test results showing that this criteria does a better job of fitting a variety of synthetic distortion sources to perceptions of degraded sound quality. In short, the Geddes/Lee papers show that the way most people formally characterize audio gear nonlinear distortion today, which focuses on high outputs and unweighted harmonics, is about as close to irrelevant as one could imagine. The papers report experimental studies of current metrics that support the idea that they are irrelevant or at least uncorrelated with human perceptions of sound quality. My own simplistic research into the subject and the scientific literature was highly supportive of Geddes/Lee general thinking before they got the far more polished and complete results that are published in the articles. My point here is that it seems reasonable to view the Geddes/Lee results as being orthodox and reasonable, as far as they go. To their credit they seem to have found a far better metric, but maybe not the best metric. I'll take better as long as it is the best we have! Today, most audible nonlinear distortion comes from loudspeakers and other electromechanical/electroacoustic transducers. Nonlinear analysis of speakers is in its infancy, but two methods for characterizing speaker nonlinear distortion are currently being used pretty widely. One is due to Klippel, and one is due to Clark and Geddes. AFAIK the Klippel methodology only produces results for the second and third order, which would appear to be a significant failing. The Clark & Geddes methodology (AKA Dumax) produces results for higher orders as well which would appear to be a significant strength. |
#26
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Preamp Design Fundamentals
Two recent relevant AES papers are
and Audio Engineering Society Convention Paper 5890 Auditory Perception of Nonlinear Distortion - Theory Earl R. Geddes 1 and Lidia W. Lee Arne says: My own simplistic research into the subject and the scientific literature was highly supportive of Geddes/Lee general thinking before they got the far more polished and complete results that are published in the articles. Today, most audible nonlinear distortion comes from loudspeakers and other electromechanical/electroacoustic transducers. Arne, this is very interesting, and I like it because it agrees :-) with my feeling/observation of 'distortion' effects in lower-level signals. Some systems seem to have artifacts discernable in the low-level passages of material with high dynamic range, while the 'loud' parts sound fine, and typical distortion measurements give good numbers. How do you think this may relate to preamps?? What can we learn about testing preamps that is relevant to these findings? Can you suggest testing approaches that we little guys could use in evaluating our own preamp designs?? Hey, good new information always generates LOTS of questions! -- Regards, Terry King ...In The Woods In Vermont The one who Dies With The Most Parts LOSES!! What do you need? |
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Preamp Design Fundamentals
Terry King wrote:
Two recent relevant AES papers are and Audio Engineering Society Convention Paper 5890 Auditory Perception of Nonlinear Distortion - Theory Earl R. Geddes 1 and Lidia W. Lee Arne says: My own simplistic research into the subject and the scientific literature was highly supportive of Geddes/Lee general thinking before they got the far more polished and complete results that are published in the articles. Today, most audible nonlinear distortion comes from loudspeakers and other electromechanical/electroacoustic transducers. Arne, this is very interesting, and I like it because it agrees :-) with my feeling/observation of 'distortion' effects in lower-level signals. Some systems seem to have artifacts discernable in the low-level passages of material with high dynamic range, while the 'loud' parts sound fine, and typical distortion measurements give good numbers. Crossover distortion in class B amplifiers is this way. It produces a "dead band" at the zero-crossing point which is very large in comparison with a low-level signal, but rather small in comparison with a large one. How do you think this may relate to preamps?? What can we learn about testing preamps that is relevant to these findings? Can you suggest testing approaches that we little guys could use in evaluating our own preamp designs?? Preamps are hard because they have two problems: first you have whatever distortion the system has in isolation (which is easy to detect with a straight wire test), and secondly you have whatever distortion is produced due the microphone/preamp interaction and loading (which is hard to quantify easily). --scott -- "C'est un Nagra. C'est suisse, et tres, tres precis." |
#28
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Preamp Design Fundamentals
"Terry King" wrote in message
Two recent relevant AES papers are and Audio Engineering Society Convention Paper 5890 Auditory Perception of Nonlinear Distortion - Theory Earl R. Geddes 1 and Lidia W. Lee Arne says: My own simplistic research into the subject and the scientific literature was highly supportive of Geddes/Lee general thinking before they got the far more polished and complete results that are published in the articles. Today, most audible nonlinear distortion comes from loudspeakers and other electromechanical/electroacoustic transducers. Arne, this is very interesting, and I like it because it agrees :-) with my feeling/observation of 'distortion' effects in lower-level signals. Some systems seem to have artifacts discernable in the low-level passages of material with high dynamic range, while the 'loud' parts sound fine, and typical distortion measurements give good numbers. How do you think this may relate to preamps?? I have to regrettably plead ignorance. Right now I class myself as a mic preamp newbie. I have a working theory of sorts about why mic pres sound different that revolves around: Noise at low levels Clipping at high levels Mic loading effects EMI/RFI Not on the list is any kind of low-level nonlinear distortion, but that might just be ignorance on my part. I've had my nose rubbed in most of the other items on the list. I await further practical experience and time on the test bench. What can we learn about testing preamps that is relevant to these findings? In all seriousness, show me some current online technical tests of mic preamps that have levels of completeness that PCAVTech provides for sound cards and power amps. I don't even see a starting point... So I don't see a lot that is readily available to even learn from. Learning on your own is a slow process, whether its you, me, or somebody else. Can you suggest testing approaches that we little guys could use in evaluating our own preamp designs?? Well one thing I'm trying to say is that w/r/t mic preamps I'm very much a little guy. But I don't see anybody who really seems to be running with the ball. This is the sort of thing that got me into PCAVTech in the first place... Hey, good new information always generates LOTS of questions! I think the Geddes/Lee results are going to make a lot of people feel good about their past reservations about current means of characterizing nonlinear distortion. However, I don't see a final answer or even a really workable interim answer. I see the beginning of a potentially productive line of questioning. |
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#30
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Preamp Design Fundamentals
Two good starting points:
1) Harmonic spectra at a variety of levels (near-maximum, nominal, and, say, 50 dB below nominal), done using bass, midrange and high-frequency tones. One looks for higher-order harmonics, and also for odd changes in spectrum from one level to another. 2) Intermodulation tests, again at a variety of levels. These can be 2-tone (19kHz and 19.5kHz, for example) or multitone. Most of these tests can be done, assuming adequately clean source and converters (big assumption!), using the spectrum-analyzer features of popular audio editing programs such as CoolEdit. The hard one is high-frequency harmonic spectra, since the higher harmonics will be chopped off by anti-aliasing filters. I've found that HF intermod. measurements help separate sheep from goats in the case of ICs. They can also show up quirks, such as the typical rise in distortion of FET-input circuits when the impedance feeding a non-inverting input is high. Peace, Paul |
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In article ,
says... Most of these tests can be done, assuming adequately clean source and converters (big assumption!), using the spectrum-analyzer features of popular audio editing programs such as CoolEdit. The hard one is high-frequency harmonic spectra, since the higher harmonics will be chopped off by anti-aliasing filters. Paul, This is a good approach. This reminds me that I shouldn't mind no longer having the HP Distortion Analyzer I had at IBM.. It's more important to know what the (Not MY Fundamental!) residuals ARE, and that means decent spectrum Analysis. I can do 96Khz at 24 bits with my 1010LT's; Can I assume I can SEE higher-order products, such as from the high-frequency IM tests? Maybe we can figure out what guys without a lot of gear can do. I have an HP function generator, but I THINK I could put together a 2-tone signal in Cool Edit or N-Track at 24Bits/96Khz that would be a respectable test source for IM tests. What do you and the other guys here think?? Has anyone done what they consider 'good' tests like this using computer-based 24bit/96Khz equipment? Sorry, I've been away for a while. I feel a little like Rip Van Winkle meets Robert Pease... -- Regards, Terry King ...In The Woods In Vermont "The one who dies with the most parts LOSES! What do you need??" |
#32
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Preamp Design Fundamentals
"Terry King" wrote in message
In article , says... Most of these tests can be done, assuming adequately clean source and converters (big assumption!), using the spectrum-analyzer features of popular audio editing programs such as CoolEdit. The hard one is high-frequency harmonic spectra, since the higher harmonics will be chopped off by anti-aliasing filters. Paul, This is a good approach. This reminds me that I shouldn't mind no longer having the HP Distortion Analyzer I had at IBM.. It's more important to know what the (Not MY Fundamental!) residuals ARE, and that means decent spectrum Analysis. I can do 96Khz at 24 bits with my 1010LT's; Can I assume I can SEE higher-order products, such as from the high-frequency IM tests? You betcha. If you want to see a whole lot of equipment tests based on FFT analysis, please visit www.pcavtech.com. Some of the more complete tests include: http://www.pcavtech.com/pwramp/macrot-5000VZ/index.htm http://www.pcavtech.com/soundcards/LynxTWO/index.htm Please note that all of the numeric test results are hyperlinks. Click them to see the supporting FFT-based graphical analysis. Maybe we can figure out what guys without a lot of gear can do. I have an HP function generator, but I THINK I could put together a 2-tone signal in Cool Edit or N-Track at 24Bits/96Khz that would be a respectable test source for IM tests. This is exactly how the tests at www.pcavtech.com were done. The FFT analysis tool used was Spectra Lab, which can be had free for 30 days by downloading from http://www.soundtechnology.com/download-center.htm Has anyone done what they consider 'good' tests like this using computer-based 24bit/96Khz equipment? Sorry, I've been away for a while. I feel a little like Rip Van Winkle meets Robert Pease... There is also a handy freebie FFT-based tool that is highly automated, You can download it free from http://audio.rightmark.org/ Here are results from it that you can compare to the ones mentioned above: http://audio.rightmark.org/test/lynx-two-b-32192.htm http://audio.egregious.net/lynx_righ..._96/24_96.html http://audio.egregious.net/lynx_righ...4_48/24_48.htm And here are some delta 1010lt results from audio rightmark: http://www.alfainfocom.ru/delta-rmaa/Comparison.htm |
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Preamp Design Fundamentals
I can do 96Khz at 24 bits with my 1010LT's; Can I assume I can SEE
higher-order products, such as from the high-frequency IM tests? You betcha. If you want to see a whole lot of equipment tests based on FFT analysis, please visit www.pcavtech.com. Some of the more complete tests include: Arny, That is a great bunch of info; I'm very encouraged and digging into that stuff. Thanks for saving us a lot of WWW research time. I'm amazed at the SNR that both M-Audio and SB have achieved on a card physically inside a PC, with Unbal RCA connectors. I worked on the IBM Mwave stuff (DSP Based sound/modem cards), and we were fighting like heck to get to -80. Fortunately, back then (1994 or so) SB was a lot worse!! Now if I could just read Russian.... -- Regards, Terry King ...In The Woods In Vermont The one who Dies With The Most Parts LOSES!! What do you need? |
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Preamp Design Fundamentals
I can do 96Khz at 24 bits with my 1010LT's; Can I assume I can SEE
higher-order products, such as from the high-frequency IM tests? Yes. In fact, you can see many of the products with a 44.1kHz A/D, since they're down in the audio band. Maybe we can figure out what guys without a lot of gear can do. I have an HP function generator, but I THINK I could put together a 2-tone signal in Cool Edit or N-Track at 24Bits/96Khz that would be a respectable test source for IM tests. What do you and the other guys here think?? Certainly. I've generated 2- and 3-tone signals in CoolEdit. Has anyone done what they consider 'good' tests like this using computer-based 24bit/96Khz equipment? I've done quite a few using 16bit 44.1kHz equipment. I burned a CD of the aforementioned test signals, played it on a decent-quality but not spectacular CD player (Philips/Magnavox), recorded the results using a decent-quality DAT recorder (Sony 2000ES) and loaded it into my computer via S/PDIF. I then used DC-ART to lowpass filter the signal at 2400Hz and raise the level of the filtered signal by 20dB, and used their spectrum analyzer to look at the results. Very clean, almost entirely noise with perhaps just a hint of a distortion product around -96dB (that's *after* raising by 20dB). I tested a lot of IC's that way, before I got a 24/96 setup, and it's good enough to at least see some of the differences, such as the IM increase with higher source impedance on FET-input circuits I mentioned earlier. Must see what the fancier setup can measure now that I've got it. But with my quite modest former setup I got much usable information. Such as what the output section of a Mackie does to the signal. Trust me, you don't want to know. Peace, Paul |
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Preamp Design Fundamentals
P Stamler wrote:
I've done quite a few using 16bit 44.1kHz equipment. I burned a CD of the aforementioned test signals, played it on a decent-quality but not spectacular CD player (Philips/Magnavox), recorded the results using a decent-quality DAT recorder (Sony 2000ES) and loaded it into my computer via S/PDIF. I then used DC-ART to lowpass filter the signal at 2400Hz and raise the level of the filtered signal by 20dB, and used their spectrum analyzer to look at the results. Very clean, almost entirely noise with perhaps just a hint of a distortion product around -96dB (that's *after* raising by 20dB). I was thinking...If you were to do this again with modern equipment, you could play the test tone out through a channel of a DAW interface and simultaneously record the result back into an input on said same DAW. The resulting recording would be sample-synchronized to the original test tone file due to both converters running off the same clock. You could therefore use a mathematical subtraction of the original test tone so that the removal of the original test tones would be complete, and without artifacts. Is this what everybody else already does? I suppose you could do this with the equipment you described IF you synchronize the DAT machine and the CD player. ulysses |
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In article ,
says... I was thinking...If you were to do this again with modern equipment, you could play the test tone out through a channel of a DAW interface and simultaneously record the result back into an input on said same DAW. The resulting recording would be sample-synchronized to the original test tone file due to both converters running off the same clock. You could therefore use a mathematical subtraction of the original test tone so that the removal of the original test tones would be complete, and without artifacts. If this works, it would enable the kind of real-time circuit tweaking that some of us feel defines "Hands-On" electronics. Fooling with -90 signals might require shielded gloves, though... -- Regards, Terry King ...In The Woods In Vermont "The one who dies with the most parts LOSES! What do you need??" |
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#38
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Preamp Design Fundamentals
I was thinking...If you were to do this again with modern equipment,
you could play the test tone out through a channel of a DAW interface and simultaneously record the result back into an input on said same DAW. The resulting recording would be sample-synchronized to the original test tone file due to both converters running off the same clock. You could therefore use a mathematical subtraction of the original test tone so that the removal of the original test tones would be complete, and without artifacts. Is this what everybody else already does? Probably not; old devil latency would get in the way. However, you can filter digitally with few artifacts generated, depending of course on what you're looking for. I suppose you could do this with the equipment you described IF you synchronize the DAT machine and the CD player. Which'd probably be almost impossible. But filtering's easy enough. Peace, Paul |
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Preamp Design Fundamentals
"Justin Ulysses Morse" wrote in message
m P Stamler wrote: I've done quite a few using 16bit 44.1kHz equipment. I burned a CD of the aforementioned test signals, played it on a decent-quality but not spectacular CD player (Philips/Magnavox), recorded the results using a decent-quality DAT recorder (Sony 2000ES) and loaded it into my computer via S/PDIF. I then used DC-ART to lowpass filter the signal at 2400Hz and raise the level of the filtered signal by 20dB, and used their spectrum analyzer to look at the results. Very clean, almost entirely noise with perhaps just a hint of a distortion product around -96dB (that's *after* raising by 20dB). I was thinking...If you were to do this again with modern equipment, you could play the test tone out through a channel of a DAW interface and simultaneously record the result back into an input on said same DAW. The resulting recording would be sample-synchronized to the original test tone file due to both converters running off the same clock. You could therefore use a mathematical subtraction of the original test tone so that the removal of the original test tones would be complete, and without artifacts. Is this what everybody else already does? I've done it, and it ain't a bed of roses. One serious problem is that the granularity of the alignment is one sample due to its basic crudity. The next step of finesse is to upsample and do the alignment in a higher sample rate domain where the sample size is far smaller. Been there done that. In fact that's how plots like http://www.pcavtech.com/soundcards/L...192-xfus10.gif were made. Note that I used upsampling to do a more precise alignment, and then I turned around and did the analysis by means of FFTs, not subtraction or nulling. I suppose you could do this with the equipment you described IF you synchronize the DAT machine and the CD player. The problem with nulling or subtraction approaches is that the difference signal is very sensitive to variations in both the frequency domain and amplitude domain. FFT analysis is a lot more straight-forward IMO. |
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Preamp Design Fundamentals
P Stamler wrote:
Is this what everybody else already does? Probably not; old devil latency would get in the way. However, you can filter digitally with few artifacts generated, depending of course on what you're looking for. But you can compensate for latency after you get your signal back into the computer, and before you add the inverted original test tone. I suppose you could do this with the equipment you described IF you synchronize the DAT machine and the CD player. Which'd probably be almost impossible. But filtering's easy enough. I think it would be pretty easy IF your DAT machine allows clocking to SPDIF input while recording the analog input. I know I can do this with my AudioMedia II card, which costs less than a DAT machine, but I don't recall if my DA30 does this or not. I think yes. ulysses |
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