EEVblog Electronics Community Forum
Products => Test Equipment => Topic started by: DaJMasta on January 03, 2017, 08:47:33 pm
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I know there are a bunch of audio types around here and while I think I know what I want and what sort of specs to be looking for, I don't have any immediate experience using audio analyzers so I'd like some input to help me figure things out.
I'm a musician with a bit of a fascination with audio gear, and it's been a fairly longstanding goal of mine to do some measurements to characterize the sound of different instruments, techniques, spaces, etc.. I'm also interested in being able to characterize the performance of my gear - frequency response, THD+N, etc - both as a reference for further experiments and so I can see the effect of different configurations or modifications.
To those ends, I want an analyzer that has:
A high resolution FFT analysis mode that can update at least several times a second (minimum) and which all data can be saved
A built in audio signal generator for characterizing equipment or spaces
Software to compute distortion, snr, mark fft peaks, visualize the input in real time (or near to it), etc.
Two input and two output channels
Some degree of portability - I don't care if it's heavy and takes a bit to set up, but it can't be half a dozen bits of gear that require a dolly to move around
Digital audio signal measurement does not concern me, I'm working entirely with analog signals
I don't get the impression that a DSA will give me the signal generation and software options I want. I don't think the HP8903 is an option either because best I can tell, it doesn't offer FFT capability. While something like a very nice sound card and some software could potentially do the job, I don't know what software that would be and I do want it to be just one piece of software - I want the input and output to be run through the same stuff so it can handle the math in the output as I don't want to be programming my own stuff for the standard measurements. I also know a lot of test equipment related software can be quite pricey. I've seen the QA401 suggested a few times and it looks promising, but I don't know if the software is on the level of other brands - ease of use is important and built in features are important, but I also know nothing about the update or capture rate for the visualizations or data.
If the QA401 is on the inexpensive side, of what I've seen, the Rohde & Schwarz UPL and the Audio Precision ATS-2 are in the same price ballpark (at least buying used) and have both excellent performance figures and the reputation of their respective brands behind the quality and usability of their software (all I really have to go on without personal experience). On the very high end of prices I'd consider, there's the R&S UPV and the Stanford Research SR1.
Having the complete system integrated without a computer has its benefits in terms of usability and portability, but I get the impression that a computer interface will make it easier to store data and will increase visualization performance (for example, the UPL uses a pre-USB era computer, so while it's performance is adequate, it's probably not fast and it's probably a bit of a pain to get the data out and on to a modern computer).
So does anyone have experience with the software on any of these devices? Know of something else I should consider? I'm not in a rush to buy, I'm comfortable taking at least a few months to decide and watch for deals, but I'd appreciate some input.
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Have you looked at Audiomatica? http://www.audiomatica.com (http://www.audiomatica.com)
or Linearx? http://www.audiomatica.com (http://www.audiomatica.com)
The back of the Loudspeaker Design Cookbook has lots of ads for stuff like this...
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As I did some time ago in an other thread concerning a similar question I will recommend again "hpw-works" [http://hpw-works.com/ (http://hpw-works.com/)].
This is a very powerful software for exactly all the measurements you want to do. It also generates the clean test-signals you must have to achieve reliable results and it is able to work with two channels (and also with two independent interfaces). It is a really professional solution and so it is not really cheap but worth its price.
You need to have a good audio-interface to gain the maximum profit from hpw-works like RME or similar high-quality interfaces. But if you will make recordings and other audio-stuff at a high level, you will have such an interface anyway.
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For sure no DSA or Scope provides sufficient dynamic range for audio analysis.
Since you are into music production, if you have a capture card/ADC, you can start with that and free software from Rightmark: http://audio.rightmark.org/index_new.shtml. (http://audio.rightmark.org/index_new.shtml.) PC is quite powerful so you can show you real-time FFT depending on length and sample rate.
If you have a few thousand dollars, then dSound from Prism Sound is a good option. They are small and pretty portable to carry and the software interface is very good.
Step up from that is Audio Precision and there we are talking $15K to $25K with better analog front-end and specs.
Rhode and Schwartz is also up there with an embedded display and high prices to go with it.
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What would people think about something like the PicoScope 4262. I've never used it. But, it might be appealing as a more general piece of test equipment for audio circuit use (instead of just directly plugging in mics)? 16bit scope. AWG. I've heard the PicoScope software permits very long FFT record lengths for finer freq resolution. Thoughts?
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I think even "high res" scopes are out simply because that high resolution or high resolution mode is still actually somewhat low for audio sampling. Sure they can take many more measurements in the same time frame, but it seems that the enhanced resolution of the ADCs in audio circuits gives them better dynamic range and lower noise floors by a good margin.
Anyways, I've been taking a look at the suggested stuff and there seems to be some real promise in there, the CLIO seems to be about right in terms of specs and while there isn't a ton of info on the software that I've found, it does seem like it's built to do the kind of things I'm looking it, it's just that the interface is firewire |O Certainly not insurmountable, but of all the interfaces to choose... I know it's better for low latency audio than USB 2, but even thunderbolt would be preferable to me ???
The LinearX LX500 seems capable and the software suite seems very comprehensive, I wouldn't be going for it but the 3d analysis stuff is pretty impressive too. Price is a bit higher, but it's worth further looking into I think.
The Prisim Sound dScope seems like it's also hardware that will do the job admirably, and the vote of confidence in the software is a good sign, but the price could be high... no immediate indicators on the site :p
For clarification's sake, yes I know a new UPV is probably still significantly more than that, but for these pricier bits of gear I'm looking for second hand stuff and the going rate seems to be a stretch, but potentially manageable.
I do have a question about the software-only options though, and that would be what sort of interface would actually qualify as one good enough to do the sort of characterization I'm looking for? I do have a decent ADC built into my mixer (that's at least comparable with some of the dedicated interfaces I've used in the past), a pretty good USB DAC for my normal listening, and I always have the option to get something better for better measurements, but while I don't doubt that studio quality gear will be able to match the fidelity requirements, are they really suitable for the measurements? I think my main areas of concern would be absolute voltage references (if you put a 1V sinewave out of your generator, run it through a follower amp and into an input, is studio gear going to be able to guarantee that it spat out a 1V sine and is reading a 1V sine or is it just going to be able to give you accurate measurements relative to wherever the reference point is at?) and clock jitter. I haven't taken apart high end interfaces before, so maybe I'm crazy, but I would think jitter plays a pretty decent part in getting low-noise FFT measurements without lots of averaging, and I don't know of any studio gear that would use, for example, an ovenized oscillator, or accept a reference frequency input. Basically, I'm concerned that gear just designed to sound really good won't actually be precise enough to be measuring and characterizing gear that also sounds good, but I don't know enough about what interfaces are available to be able to say for sure.
If there are audio interfaces that aren't specialized analyzer gear, then I think some of the recommended software is definitely up to the task, but I don't know if they exist.... and if they do and they're just real expensive, I sort of would prefer a unit designed for measurement specifically though that's really only based on a 'gut feeling'.
For others' info, I also heard from a member with a UPL and they said that usability and update rate were no problem even on the older hardware, which is great to hear.
Anyways, I'll keep the research going, thanks for the input so far!
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Your studio gear is fine. With FFT and fast PC you can use long number of samples which gives you very nice "process gain." In simple English, you can hugely lower the noise level in FFTs using even average ADCs.
The above allows you then to see jitter as sidebands of your signal.
As for voltage measurement, you can do that too by playing files with some voltage and seeing what your capture shows. Then compensate for the level difference.
You can see a sample of software analysis in blogger Archmiago's write ups. Here is an example: http://archimago.blogspot.com/2016/10/measurements-raspberry-pi-3-hifiberry.html (http://archimago.blogspot.com/2016/10/measurements-raspberry-pi-3-hifiberry.html)
You can certainly start at this level with no money out of pocket to learn the topic itself.
That said, I am not a fan of software solutions as it is not a system where results can be easily duplicated and compared.
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What about the QuantAsylum qa401.
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Clio has a USB "pocket" version (pocket - as in very big pockets). Not sure it does everything you need. My guess is with these types of systems is to try a few. Depends on their slant (loudspeaker, acoustic analysis, vibration analysis...) - sure - some are so task specific they are clearly out (like red rock acoustics SpeaD and reverse SpeaD) - but others may or may not satisfy your requirements.
Liberty Instruments (Praxis, Liberty audiosuite, etc) now has a $299 package with Parts Express: https://www.parts-express.com/dayton-audio-omnimic-v2-computer-based-precision-room-measurement-system--390-792 (https://www.parts-express.com/dayton-audio-omnimic-v2-computer-based-precision-room-measurement-system--390-792) From the picture it looks like the old Behringer measurement microphone (remember that? used to be supplied with Velodyne subwoofers).
Even for iPhones there are pretty capable packages (like AudioTools) - that with a calibrated microphone can be pretty cool - not perfect but can replace most uses for "inexpensive" SPL meters (but probably not for work environment certification / noise pollution). Pretty expensive, but they used to have 70% off discounts once in a while.
ETF was a favorite 15 years ago (seems like they have a new product but the site states 2013 so don't know if they went the way of the Dodo): http://etfacoustic.com/ (http://etfacoustic.com/)
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For acoustical jobs, I would always choose for a software only option, even if cost was not an issue. A half decent sound card will have at least 10x less distortions than any acoustic environment or speaker you will ever measure. But the advantage is you can use more software tools (no dedicated hardware) and much more types of testsignals and much more ways to visualize and report the measurements (very important to make sense of the measurements)
TIP: always use constant latency ASIO drivers for your sound card to avoid timing issues
For measuring audio equipment (electronics), it is different (because you need much better specs, you want to avoid ground loops, you’re working with low and high voltage signals, you want very low noise...) so here a decent hardware solution (AP, R&S, Prismsound...) can make things a lot easier, but it is not absolutely necessary.
In my opinion, recent audio equipment (electronics) are nowadays more than good enough not to invest your time in that (avoid so called “high end” gear, as these sometimes deliberately distort to sound “different” than the competition).
Speakers and acoustics however still have lots and lots of issues (due to the laws of physics), so most benefit is to be had here. So if I were you, I would buy a calibrated mic, a sound card with recent and stable ASIO drivers, and start learning to work with software tools like ARTA, Holmimpulse, REW, Abec3 AND read lots and lots of books on acoustics and speaker design…
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I'm currently finishing up a headphone amplifier build and I got the QuantAsylum QA401 when I started developing the amp. With todays extremely low noise and distortion figures on op amps, You'll hit the bottom of what the analyser can measure very quickly. I had to build a twin-T notch filter and get hold of a extremely low THD signal generator to be able to measure below -108dB THD. Getting a industry standard Audio Precision is the dream, but they are way to expensive for me...
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I'm currently finishing up a headphone amplifier build and I got the QuantAsylum QA401 when I started developing the amp. With todays extremely low noise and distortion figures on op amps, You'll hit the bottom of what the analyser can measure very quickly. I had to build a twin-T notch filter and get hold of a extremely low THD signal generator to be able to measure below -108dB THD. Getting a industry standard Audio Precision is the dream, but they are way to expensive for me...
But the question is, why would we need to see lower than that? It can be fun as a technical exercise, but it will not improve the sound quality we hear any more…
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I mentioned the QA401 in the original post and I just don't have any experience with the software, which would probably be the big component of it. A 32 bit DAC and ADC sound fancy, but it doesn't look like the specs are any better than the 24 bit competitors, and I think it just comes down to physical restrictions of the design and parasitics at that point. Since that is very software dependent too, I'm not sure how much performance would actually differ from just using my existing equipment.
The CLIO pocket seems like a similar little device, but the big drawback is that it's only a single channel. Sure you could get two, but it seems like there's several options for this sort of USB box with some converters so it may be easier just to pick another if I go that route. I think the measurement mic pack is probably not worth it for me - just the mic itself isn't that expensive and I've got interfaces that are probably better than a built in USB thing, so then it comes down to the software and it does seem like there are affordable options. I've also got an older dedicated SPL meter, a CEL-231 with a calibrator box. Sure, the cal seal on the inside is from 1992, but the spec is down to +-1dB , which isn't really common with most inexpensive meters :)
I think in the short term, getting rightmark and experimenting with my current interfaces is basically a no-brainer, but I agree with the liking a hardware solution sentiment. It also seems to afford you some more input protection, better filtering, more EMI rejection, and a cleaner power supply without having to put everything on an external power conditioner. Maybe a good interface can give you performance good enough that you can't really hear a difference, higher performance will still get you more fidelity in the visualization and the same amount of accuracy with a shorter sample time, and since I do want close to real time FFT visualization, the extra dynamic range will help keep the visualization from getting too noisy when turning up the responsiveness. Another concern is that I'm trying to be as scientific as possible with my main use - the measurement of instrument sounds - mostly because I don't know what results I'll find. Having better equipment will let you see more detail in the minor differences and reduce the chances of running into hardware artifacts or limitations in the measurements that are difficult to identify. I'm looking for an instrument that can beat the specs on my current hardware because my current hardware seems to edge out my own ear across the board (I've been honing my listening skills for a while now as part of my trade), and I don't want to run into a situation where my measurements can't match that level or my ears can match it with another couple decades of experience. I want measurements that are good enough that no one can hear beyond them, because some of the people who may eventually be interested in my data may literally be the most discerning listeners in the world.
Couple new questions from reading through datasheets and such:
Are there audio analyzers that provide phantom power? Are commercial phantom injectors also noise sources? I assume it's a concern with ultra low noise gear, but is that a common issue?
Do you think it's valuable for the generator side of the instrument to produce more than sine waves? It seems like some of the lowest distortion generators are pure sine only, but I've already been using some AM and added harmonics in signals on my existing function generator... is that something that's likely to be useful? I assume all the normal audio rating specs are done with sine only, but it's definitely not a "realistic" test signal.
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Although human ears and human hearing are fantastic and amazing in respect of its construction and function, there will be no doubt that any reasonably decent technical solution will be far better in terms of precision, reliability and reproducibility of its results.
The frequency response of any decent high-quality (omni-)microphone will be easily outplay any adult ear and there are for example some new Sennheiser Studio-Microphones which will reach up to 50 KHz. Also the dynamic range of a really good microphone/amplifier will exceed the total amount of our ears because there are microphones which will deliver a low distortion output up to 140 dB and a decent amplifying circuit will also let pass the signal unspoiled.
And the great amount of the dynamic range of our ears from 0 to nearly 120 dB (normally the maximum pain threshold) will only be achieved by an time-consuming adaptive process. If your ear has been exposed some time to a strepitous environement you will afterwards not be able for some time to hear really quite sound for some time. This is due to the internal protective mechanism of our ear.
And all the functionality will only be completely there, if our ears are sane and undamaged.
But the most erratic part will be our brain itself which must interpret all the signals coming from our ears. And the brain will interpret continuous - because this is the task of our brain. And there will be expectations, mistakes, misconceptions and so on.
Mostly the expectations will colour our perceptions.
So I will underline that any (half-)decent microphone, amplifier, interface and software will give you the all the reliable results you want to have measuring audio sources like musical instruments, voices and so on. And today it is not necessary to pay a fortune for having professional results.
Even a simple and dirt cheap app like "Speedy spectrum analyser" will show astonishing results.
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Couple new questions from reading through datasheets and such:
Are there audio analyzers that provide phantom power? Are commercial phantom injectors also noise sources? I assume it's a concern with ultra low noise gear, but is that a common issue?
XLR inputs are for line level capture and not microphone. So they neither provide phantom power no mic amplification.
Do you think it's valuable for the generator side of the instrument to produce more than sine waves? It seems like some of the lowest distortion generators are pure sine only, but I've already been using some AM and added harmonics in signals on my existing function generator... is that something that's likely to be useful? I assume all the normal audio rating specs are done with sine only, but it's definitely not a "realistic" test signal.
The common way to do this testing is to use a synthetically created test file and using that through a DAC for the source. These are mathematically produced and with dither can go down to any depth you want. No need to use any analog generator. And for testing things like DACs, you just feed them the digital stream eliminating.
You can even create the files using your DAW. Just make sure you add dither (TPDF) so that you don't see the distortions caused there in your analysis. YOu can create single frequency and sweeps as needed.
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Ah ok, it seems like some of the built in generators are specified as sine only, but not by any means all of them. Having the capability to do effectly "true arb" patterns by just playing an audio file seems like a good way to do it.
If it's line level inputs, would you also want an external preamp for connecting a mic directly, or is the gain high enough and noise floor low enough on the inputs that they can interface directly with the mic (given phantom and AC coupled as needed)?
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I used an Audio Precision SYS 2722 for years (not mine unfortunately, at a client). Don't know what you want to pay, you can get it used on eBay for $6000. It is worth the price.
The software is easy to use. See the manual here (http://exodus.poly.edu/~kurt/manuals/manuals/Other/AUDIO%20PRECISION%20Series%202700%20User%20also%20for%20SYS%2020226.pdf) for all the details. Can do FFT, too. It is heavy, but portable in the way you described it, only one big device, plus the required PC or laptop. You can enhance the software with own Basic scripts. Once I implemented a complete calibrate and test script for production for another high priced audio gear. It really has all you ever need, like balanced and unbalanced analog inputs / outputs (with phantom power) and digital in/outs (AES/EBU, up to 192 kHz sampling rate). In the software you can configure the outputs with signal generators and configure which inputs you want to measure and you can save setup configurations. Can output all kind of useful test signals, like sin, noise, arbitrary waveform (IIRC the length was a bit limited) or many different kind of pulses and bursts (useful for loudness measurement tests, or verifying peak meters).
And very useful if you develop hardware or want to do in-depth tests: the analog features of the digital inputs, like jitter measurement, see e.g. this video:
https://www.youtube.com/watch?v=s_Cma3cvD7o (https://www.youtube.com/watch?v=s_Cma3cvD7o)
I don't like their software of the newer devices, like for the APx500. The GUI looks more polished, but in my opinion it is less flexible and less useful, because you can't organize all the measurements and generators in one easy to access configuration and I don't know if it is scriptable, too.
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If it's line level inputs, would you also want an external preamp for connecting a mic directly, or is the gain high enough and noise floor low enough on the inputs that they can interface directly with the mic (given phantom and AC coupled as needed)?
I am not a recording guy but I am pretty sure the mic will need a pre-amp to get up to the line level inputs. You can feed it lower level signal but then you lose a ton of your dynamic range.
What will you be doing with the mic anyway? If it is for room measurements then that is an entirely different topic. There, I 100% recommend computer software and any microphone you can hook up to your computer. The weapon of choice is free software called Room EQ Wizard. It is a very sophisticated piece of software, outperforming many commercial ones. The down side is that despite what it calls itself, it is hard to learn to use. At least initially. To that end, I have written two tutorials to get people started:
Part 1: http://www.audiosciencereview.com/forum/index.php?threads/room-measurement-tutorial-for-dummies-part-1.4/ (http://www.audiosciencereview.com/forum/index.php?threads/room-measurement-tutorial-for-dummies-part-1.4/)
Part 2: http://www.audiosciencereview.com/forum/index.php?threads/room-measurement-tutorial-for-dummies-part-2.5/ (http://www.audiosciencereview.com/forum/index.php?threads/room-measurement-tutorial-for-dummies-part-2.5/)
I plan to finish the series by showing what to measure and what to do to improve room acoustics. It will include much of the research into psychoacoustics as opposed to simple cookbooks out there.
Anyway, tell us more and we can give more relevant advice. :)
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I'm racing out the door to the studio here but I would advise the O.P. to slow way down and walk before you can run. You are talking about some very high end test equipment but making very basic mistakes in your understanding of audio, electronics and measurements.
I totally understand the idea of doing things right and getting the best but understanding the fundamentals and how things are done will help you make better choices when you buy higher end gear later.
I use REW (free) and a Prism Lyra 1 interface for acoustics work and on the road electronic testing, on the bench I use a Prism dScope. You can go a long way with something like REW or audio tester and an interface for a couple of hundred dollars. Start learning the fundamentals - for example why we test with tones and not music, then buy gear as you know more about what you really need.
Can you tell us more about your application?
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Maybe the issue is that I have two applications: There's the testing application that involves the generator and the low noise floor and distortion, and the more primary real time FFT visualization with recording (of the FFT frames, I can get the audio however if needed).
The idea was to try to get them both on the same instrument, since most of the testers which had enough visualization power to generate a Bode plot (not a Keithley 2015 or an HP 8903) also had a good FFT mode and the frontend to keep it nice and clean. I'm not trying to use music to test the performance of gear, but I've used AM and tone + harmonics modes on my current function generator for some audio stuff before and I'd like to have the capability (my current gen is only .075% THD in the audio band, so it wouldn't be suitable for use with these instruments).
So I'm looking for, specifically:
Good FFT mode (reasonably quick updates, lots of bins available for detailed analysis, wide dynamic range, real time visualization, recording of each frame)
Ability to test studio grade gear for distortion, noise, and frequency response
The rest is much less important, but if it's fascinating and it doesn't cost a ton more, I'd certainly like it. I don't plan on characterizing the acoustics of any halls, but it would be great if I could pick up an omnidirectional measurement mic and dabble. Measuring timing differences or harmonic differences between input signals would also be really cool to play with, but it would be just for that, I don't have a use for those features I know of. Something like setting up a mic and a speaker in a room and evaluating how performance changes with different furniture or mic/speaker placement is definitely an extra but would be fascinating.
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I'm racing out the door to the studio here but I would advise the O.P. to slow way down and walk before you can run. You are talking about some very high end test equipment but making very basic mistakes in your understanding of audio, electronics and measurements.
I totally understand the idea of doing things right and getting the best but understanding the fundamentals and how things are done will help you make better choices when you buy higher end gear later.
I have to agree with this. Good audio measurements are much more about understanding WHAT and HOW you are measuring, then about having very good measurement gear. I would recommend to read a few good books on the subject. I can recommend Floyd Toole "Sound Reproduction", Joseph d'appolito "Testing loudspeakers" (a bit dated, but explains the basics really wel)and Bob Cordell "Audio Amplifiers" (contains a good section about measuring amplifier and electronics).
To understand "how" we hear, I can recommend JanSchnupp "auditory neuroscience making sense of sound". This book will give you a better idea what parameter a important, and which need to be only good enough.
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I'm currently finishing up a headphone amplifier build and I got the QuantAsylum QA401 when I started developing the amp. With todays extremely low noise and distortion figures on op amps, You'll hit the bottom of what the analyser can measure very quickly. I had to build a twin-T notch filter and get hold of a extremely low THD signal generator to be able to measure below -108dB THD. Getting a industry standard Audio Precision is the dream, but they are way to expensive for me...
But the question is, why would we need to see lower than that? It can be fun as a technical exercise, but it will not improve the sound quality we hear any more…
The numbers one gets from an analyzer are completely artificial - nobody listens to sine waves. So, stating that such and such distortion levels are audible or not is hard to defend 'a priori'. It's reasonable to expect that, given a complex musical signal, the distortion products are greater than with a simple sine wave.
Despite this uncertainty, I have found that these simple sine wave tests can be used to expose the behavior of a circuit very reliably, and that knowledge can help a designer to avoid specific distortion mechanisms, or optimize the circuit to minimize these distortions. Something as simple as changing the frequency compensation of an amplifier might have a small effect or no effect at all, and relying on a listening test to say whether the change improved performance, worsened it, or had no effect might be extremely difficult. This is the beauty of an analyzer: if one's goal is a clean circuit, an analyzer will provide a tireless, repeatable, and honest assessment of a circuit. With an analyzer, some circuit "tweak" that proves to have no effect can be understood as "irrelevant" right away, without the vagaries of a series of tedious listening tests.
It is still up to the operator to choose the test and understand the meaning of the results, but given a high resolution distortion measurement, it's my opinion that one can more accurately and rapidly evaluate a circuit's behavior using a high resolution distortion analyzer than is possible with listening tests.
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The numbers one gets from an analyzer are completely artificial - nobody listens to sine waves. So, stating that such and such distortion levels are audible or not is hard to defend 'a priori'. It's reasonable to expect that, given a complex musical signal, the distortion products are greater than with a simple sine wave.
You are right, because just measuring the THD (or THD+N) doesn't say anything about the interactions between different signals. This is one reason Audio Precision has the multitone test:
https://www.ap.com/technical-library/using-multitones-in-audio-test/ (https://www.ap.com/technical-library/using-multitones-in-audio-test/)
Additionally, any amplifier exhibits some filtering. This means that the phase shift of signals at different frequencies can be different (depending of the kind of filter, even if not intentionally). For this it is useful if you can do a bode-plot:
https://www.ap.com/technical-library/using-bode-plots-in-apx/ (https://www.ap.com/technical-library/using-bode-plots-in-apx/)
Disclaimer: I'm not associated with Audio Precision, but it is just the best for audio measurement :)
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The numbers one gets from an analyzer are completely artificial - nobody listens to sine waves. So, stating that such and such distortion levels are audible or not is hard to defend 'a priori'. It's reasonable to expect that, given a complex musical signal, the distortion products are greater than with a simple sine wave.
You are right, because just measuring the THD (or THD+N) doesn't say anything about the interactions between different signals. This is one reason Audio Precision has the multitone test:
https://www.ap.com/technical-library/using-multitones-in-audio-test/ (https://www.ap.com/technical-library/using-multitones-in-audio-test/)
Additionally, any amplifier exhibits some filtering. This means that the phase shift of signals at different frequencies can be different (depending of the kind of filter, even if not intentionally). For this it is useful if you can do a bode-plot:
https://www.ap.com/technical-library/using-bode-plots-in-apx/ (https://www.ap.com/technical-library/using-bode-plots-in-apx/)
Disclaimer: I'm not associated with Audio Precision, but it is just the best for audio measurement :)
I'm a big fan of AP gear as well :-)
Still, while multitone and IM tests seem useful, I have found that, for my work, the APx-555 high precision analog generator has a lower residual for simple 1-2kHz sine wave tests, and thus seems to more honestly 'sort' and differentiate different devices under test, especially when the devices are so clean that they approach the AP residual.
For gear that is, for example, worse than 1 ppm distortion, the multitone and other DAC based stimulus signals can be OK, and can do some useful, quick characterization. But, if you're like me and you're scraping the residual with 32-64 FFT averages, the IM and DAC based signals won't match the low residual of the high resolution generator / analyzer, and they fail to properly 'sort' DUTs according to their real errors.
It'd be neat to have an IM test set that could match this performance, but keep in mind that a (basically linear, time invariant) device that fails an IM test also has to fail a simple sine wave test. One thing that an IM test can do is torture the device with high frequencies, but that can also be done with a sine wave test, measured at several frequencies. This can show an indication of the distortion rise with frequency, which can point to the mechanism for the circuit's behavior - e.g. 6dB/octave rise or 12dB/octave can point to different causes etc.
Well, that's my world, and the APx-555 helps me out. But still, I could use another 20dB of distortion floor, just to be able to measure silly things accurately. I suspect that this will be possible in a decade or so, and I'll queue up for that box when I can!
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Well, that's my world, and the APx-555 helps me out. But still, I could use another 20dB of distortion floor, just to be able to measure silly things accurately. I suspect that this will be possible in a decade or so, and I'll queue up for that box when I can!
Well, personally I think anything better than -120 dB THD+N (like some DACs and ADCs can do, meaning 20 bits effective resolution) doesn't make a difference and it gets really expensive. Be careful with additional 20 dB, because if you move while measure something, the induced current caused by your body-electricity, or if a truck is moving outside and then the piezoelectric effect of the vibrating capacitors might cause measureable distortions :D
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For FFT if your doing more than testing analog amplifiers accuracy (i.e. looking at musical waveforms) there is software http://www.qsl.net/dl4yhf/spectra1.html (http://www.qsl.net/dl4yhf/spectra1.html) Spectrum Lab that will show much more than a typical FFT. if you are looking at acoustics there are several with ARTA being more versatile for both electronics and acoustics. REW is a good room and speaker equalizer package and there are many more. You can do well with any of a number of soundcards, the best are in the $600-1K range or something like the new standalone stereo boxes from Lynx or RME for around $1500.
I have a QA401 and find it does what its intended to do quite well. You can get about 10 dB better for about 50X the price with the latest APX555 but that seems like a small ROI. Any of these are far beyond anything you can encounter acoustically.
You can find more on these specifics at http://www.diyaudio.com/forums/design-build/ (http://www.diyaudio.com/forums/design-build/)
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Spectrum Lab looks like a really great visualizer for FFTs, and I'm similarly surprised that it's free!
I think my aim is gradually being swayed towards a really high quality audio interface over a dedicated audio analyzer. The analyzer will have its specified software suite, will have better input protection, and will probably have more output level flexibility whereas going with an interface means wider software compatibility (much more likely to just be an ASIO driver or similar), potentially built in mic preamps, and a lower noise floor for a given price.
While I do intend to be testing some of my own designs, I won't be testing in an industrial environment so input protection may not be a top concern, and having a preamp and phantom supply built in takes another part out of the chain for doing the microphone measurements. Something like the RME Babyface Pro, for as ridiculous as its name is, offers very impressive noise floor, on par with the AP 2700 series analyzers according to their numbers, and has built in preamps for something like $750 - which may be still a bit high vs the QA401 given that the a preamp box probably won't cost the remainder, and either is considerably under a "full on" analyzer's price.
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Look up on the old audio analysers made by Panasonic and ShibaSoku, there are a few models that hold up today in terms of residual THD+N like the Panasonic VP-7722A which gets about -112dB(0.00023%) THD+N @ 1kHz 2.5Vrms, 30kHz BW, you'll have a tough time approaching that with a studio audio interface + software combo due to the limitations of our current ADCs.
Those old instruments can be controlled by GPIB so you can get some pretty graphs that way and they have a monitor port in which you can perform FFT on the post-notch signal, I've attached an example below, the second harmonic is at -139dBr but I couldn't quite figure out how to make it scale properly in ARTA. :P
Great thing about these old analysers is that if you only need quick and dirty distortion results, there's no need to boot up a PC or anything and they can read individual harmonics up to the 5th at the touch of a button, too easy.
The ShibaSoku 725 series don't include the generator, but in return you can get -120dB THD+N @ 30kHz BW performance with some easy modding so they are still very capable machines today. I think unmodded they did around -116dB, which still eats anything alive except the AP flagships, not too bad for maybe $500 used eh?
The dScope started at around $3000 or so IIRC, it'll probably do everything you need but it's a biiit out of reach for hobbyists IMO (which I'm assuming this is more for hobby/learning use as otherwise you'd already have gone for an Audio Precision ahahah). ^-^
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The OP wrote:
"I'm a musician with a bit of a fascination with audio gear, and it's been a fairly longstanding goal of mine to do some measurements to characterize the sound of different instruments, techniques, spaces, etc.. I'm also interested in being able to characterize the performance of my gear - frequency response, THD+N, etc - both as a reference for further experiments and so I can see the effect of different configurations or modifications."
The way I read this, is he wants to tweak his equipment to a get better sound, and understand the effect of the tweaks. His goal is not to get better numbers only (he does not work for a commercial business where it can be interesting for marketing to build the first 32 ENOB-dac with an SN of 194db :-) so all audiophiles consider this the next Walhalla and sell their house to buy this new gear).
So we should advice gear that is :
- capable of measuring “audible” differences (and a bit below audioble so the cumulative effect of multiple changes is also possible)
- relatively easy to use and to understand (he will not use this 8 hours a days, 5 days a week…)
- affordable (so the OP can spent the rest of his available budget on other nice gear for an electronics lab :-) )
So I think the recommended QA401 will be perfect for measuring the electronics (differential inputs are a very big plus), and use a sound card with ARTA or equivalent software (he can play around with as many free tools he wants) will get him going on the acoustics side.
I recommended this as an “ex-audiophile”, as once upon a time I also really believed in the benefits of using ultra expensive audiophile capacitors, ultra-low jitter clocks, buying expensive signal cables… :-[ and wasted quite some money doing that until I started reading.
One of the things that cured me from my “audiophile illness” was the following software tool:http://www.foobar2000.org/components/view/foo_abx
It allows you perform blind ABX comparison between two music tracks. For example you can test if you can hear the difference between a compression less music and MP3@320, the effect of 0.01% THD added, the effect of some phase distortions,the effect of adding some reflections …) For manipulating the audio tracks (THD distortion, phase distortion, low and high pass filters, adding reflections) use the typical musician tools which I suspect will be not an issue for the OP.
By doing lots and lot of ABX testing (use a good headphone) you will be amazed how difficult it is to actually hear some differences. For example I was really sure I would be able on any music to recognize MP3 (even at 320kbps) from compression less, but this proved to be not so easy. :o
Combine this with a good book on psychoacoustics and acoustics, and you soon realize you will have been wasting your time on the wrong things (and the wrong audiophile gear). Your new goals will be to manage early reflections in your listing room, having a speaker with constant directivity and having multiple subs in many positions to get a reasonably flat frequency response in this modal sound region. All of these are dictated by the way sound waves travel (and the wavelengths involved) from the speaker to your ear, and the way our hearing works. The electronics cannot alter these new goals (only exception, having a speaker with steers the sound beams using DSP and multiple speakers). So once you know your electronics are “good enough” (which should not be a problem for modern affordable gear), you can start tinkering on how the fool the laws of physics…
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In the interest of learning how to crawl before you can run... have you considered an Analog Discovery (2)? All the instruments mentioned before in this thread are several order of magnitude better than this but you may find it easier to guage what you need when you have worked with something that fails your criteria...
http://store.digilentinc.com/analog-discovery-2-100msps-usb-oscilloscope-logic-analyzer-and-variable-power-supply/ (http://store.digilentinc.com/analog-discovery-2-100msps-usb-oscilloscope-logic-analyzer-and-variable-power-supply/)
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What also helped me in my cure for audiophile illness was something our friend Albert Einstein once said: If the fact’s don’t fit the theory, change the facts!
What I mean by this, if the following:
Once I was invited by our local high end hifi store for a listening session to hear the sound difference between a reasonably prices solid state amp, and a high end valve amp, both on the same high end speaker. And for sure the high end valve amp sound much better, so after that I knew for a FACT that valve amps were much better, and I wanted (lusted) for a high end valve amp from then on.
What I did not realize, is that the high end speaker used during the demo had a minimalistic passive crossover specifically tuned for an amp with a high output impedance (=valve amp), which caused the solid state amp to sound worse. This was not the fault of the solid state amp (in the contrary, the solid state amp showed the faults of the high end speaker more clearly). This kind of thing unfortunately happens a lot in the business of high end audio, making it very difficult to convince people some differences are not real.
Just take this into account whenever you “hear” a difference, there might be another reason then the better “quality” of the equipment…
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I'm currently finishing up a headphone amplifier build and I got the QuantAsylum QA401 when I started developing the amp. With todays extremely low noise and distortion figures on op amps, You'll hit the bottom of what the analyser can measure very quickly. I had to build a twin-T notch filter and get hold of a extremely low THD signal generator to be able to measure below -108dB THD. Getting a industry standard Audio Precision is the dream, but they are way to expensive for me...
But the question is, why would we need to see lower than that? It can be fun as a technical exercise, but it will not improve the sound quality we hear any more…
Going down the rabbit hole is part of the fun. I want it to measure as good as possible.
Sent from my iPad using Tapatalk
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Haven't been considering the Analog Discovery because I have a nice scope and a nice function gen - that level of generation and data capture I can already achieve. Specifically with the FFT performance, I can get much better results with a sound card's extra resolution than with the scope, even in high res mode with as many bins as possible.
I think I have to agree with the last post, I enjoy being able to measure precisely and I really like being able to notice things I couldn't without the hardware. For the sake of being sure I won't be able to notice any difference on my own, I want something with significantly better fidelity than I can discern. Most of the testing I have in mind is either strictly electrical or acoustic with as much of the room taken out as possible (a source and a mic in a small isolation chamber in a quiet room), so my goal isn't trying to setup a perfect listening room or evaluating speaker setups. I don't consider myself an audiophile, I'm perfectly satisfied with high bitrate mp3s for listening, and I don't believe in tube, capacitor, or oxygen-free copper voodoo. If I'm going to buy an expensive audio cable, it's because I want one with lots of EMI rejection because I have it routed through a nest of digital signal cables and want it to effect my noise floor an absolute minimum (and for whatever reason I couldn't just untangle the nest).
The QA401 seems like the right approach, but I don't know if it's the ideal unit for my applications. The BNC interfaces just mean I need a bunch of adapters and no mic preamps mean extra stuff in the signal chain and extra expense. While the software may be good, it seems there's no shortage of good audio analysis software that will run with sound card devices, so i don't think that's a particular selling point. It's price point is good, but if I can get a deal on a good audio interface, I can get the connectors and preamp integrated and can get a bit lower THD+N baseline on everything. I don't think the QA401 offers any special input protection that studio gear wouldn't match.
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The advatnage of using an audio interface is its relatively low initial cost, the con is the opamp many of them have in the signal chain were not made specifically for sound and vibration measurement - but "good sound", whatever that means to the designer.
Bruel & Kjaer makes measurement mics, some people recorded music with them (including Mark Levinson at one point) and subjectively, they were never the equal of Neumann. The same goes for audio interfaces. It's the combination of the input opamps, ADC's and PSU that make it all work, or not. The one that sits in this sweet spot is the E-MU 1616m, which uses good quality NJM2068s and near state-of-the-art AK5394 ADCs. Using the supplied PatchMix software, a 24 192 session could be setup to feed analysis software through ASIO drivers.
Like many good things, they are out of production and since E-MU has been acquired by Creative, they no longer evolve drivers for the new OSs coming on stream or supply schematics for repairs. The 1616m itself does show up on eBay but since these are all at least 10 years old, a thorough recap is likely needed, even if it's "new" - since this was made to sell at a pricepoint, the caps they used were not the best and there are many of them. Attached are pictures of the bottom PCBs, before and after modications. The top PCB is modded too, a little less complicated but not much.
This route is definitely for the curious and patient. It's a fascinating trip and very educational. But the dollars and cents do add up in the end.
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Going down the rabbit hole is part of the fun. I want it to measure as good as possible.
I also agree this can be a lot of fun, and if this is the goal, then buy all means buy the gear necesarry for it. But I just wanted to avoid people thinking it is essential to get good sound...
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If you have a audio-interface like the RME Babyface or you intend to buy it or a similar high-quality interface like the Fireface UC, then I will consider again "hpw-works". I am musician too and my approach to electronics is to control, maintain and if necessary repair my audio gear. Therefore I have some signal-generators, a Rigol scope (DS1104Z) and of course some multimeters. But if I want to check or verify the performance of modern half-decent and decent preamps, ADC´s and so on more in deep, this stuff will not be accurate and sensitive enough.
But I never would pay a fortune for one of the (really superb!) Audio-Precision devices. If you will buy expensive stuff like this, you must make money with it.
So I looked for am reasonable and payable alternative. The QA401 looked also good to me, but I had already some very good RME-Interfaces and finally I discovered the software "hpw-works" and I considered that this software-solution would give me not only a more than sufficient but also a really professional test-suite together with my RME´s. So i purchased it.
You can download a 14-day test-version of hpw-works here: http://hpw-works.com/index.php/download/evaluation-edition-sw-kit (http://hpw-works.com/index.php/download/evaluation-edition-sw-kit)
There you will find also the pdf-tutorial of hpw-works. The fact that it has an amount of 475 pages may give a hint of the capabilitys of this software.
My experience is, that if troubles, failures and bugs will become audible, you never have to look as deep as -120 dB into a spectrum, audible trash will be located far above this level. But if your test-suite is capable to look as deep as -120 dB (or even deeper how my suite can do) and you will not find any "dirtiness", than you can put your mind at rest and you can assume that you audio-gear will work perfectly.
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For the sake of being sure I won't be able to notice any difference on my own, I want something with significantly better fidelity than I can discern.
I think this holds true for a QA401 also, and even for a sound-card based system.
Most of the testing I have in mind is either strictly electrical or acoustic with as much of the room taken out as possible (a source and a mic in a small isolation chamber in a quiet room)
Depends on what frequencies you want to measure, but going below 1000Hz it becomes increasingly more difficult to have “an isolation chamber” and take out the room, and you can forget about”small”. Outside is than the most feasible option, and for low frequencies you can use ground plane measurements.
I don't consider myself an audiophile, I'm perfectly satisfied with high bitrate mp3s for listening, and I don't believe in tube, capacitor, or oxygen-free copper voodoo. If I'm going to buy an expensive audio cable, it's because I want one with lots of EMI rejection because I have it routed through a nest of digital signal cables and want it to effect my noise floor an absolute minimum (and for whatever reason I couldn't just untangle the nest).
Than you are already on the right track ?
The QA401 seems like the right approach, but I don't know if it's the ideal unit for my applications. The BNC interfaces just mean I need a bunch of adapters and no mic preamps mean extra stuff in the signal chain and extra expense. While the software may be good, it seems there's no shortage of good audio analysis software that will run with sound card devices, so i don't think that's a particular selling point. It's price point is good, but if I can get a deal on a good audio interface, I can get the connectors and preamp integrated and can get a bit lower THD+N baseline on everything. I don't think the QA401 offers any special input protection that studio gear wouldn't match.
That is a reasonable point to which I can fully agree , but I think buying AP or dScope like gear will still be a lot more expensive unfortunately . But if you can catch a nice deal on one of these, it is for sure more convenient to play around with, and the extra THD+N headroom is a nice bonus…
It is indeed unfortunate that the QA401 still has no ASIO drivers available (it on their to-do list), that would make this device absolutely fantastic.
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The advatnage of using an audio interface is its relatively low initial cost, the con is the opamp many of them have in the signal chain were not made specifically for sound and vibration measurement - but "good sound", whatever that means to the designer.
This is not true, I have this card, and it frequency responce is rules flat, so no sound tailoring. See meausurment I made in loopback (output connected to input)
Edit: addad summary
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Although the EMU-1616m is very good performance wise, the big disadvantage are its drivers, which reguraly cause a BSOD and other nasty issues
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The advatnage of using an audio interface is its relatively low initial cost, the con is the opamp many of them have in the signal chain were not made specifically for sound and vibration measurement - but "good sound", whatever that means to the designer.
This is not true, I have this card, and it frequency responce is rules flat, so no sound tailoring. See meausurment I made in loopback (output connected to input)
Edit: addad summary
I think there is a misunderstanding. What I said is "The one that sits in this sweet spot is the E-MU 1616m". This means that the 1616m is good, not bad. My comment is about 2 points:
1. Not every audio interface is a good candidate for objective measurement. The circuit, parts and implementation must be carefully evaluated. The 1616m is a winner, many are not.
2, Many audio interfaces cost well under $1000, some costing just $100 to $200 retail. At these prices, parts quality are not always the best. And that means they do not last, or they are not always linear. Even if the circuit, parts and implementation are all correct, if the parts quality do not stand up to critical use, then it's only as strong as the weakest link. In the case of the 1616m, the problems are in the caps count and their low quality. You can see bulges in many of the caps in the pictures listed in the post.
As to the drivers, there has been no issues with Win7 X64 or Win10 Pro. Touch wood - I have not seen a BSOD ever.
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The advatnage of using an audio interface is its relatively low initial cost, the con is the opamp many of them have in the signal chain were not made specifically for sound and vibration measurement - but "good sound", whatever that means to the designer.
This is not true, I have this card, and it frequency responce is rules flat, so no sound tailoring. See meausurment I made in loopback (output connected to input)
Edit: addad summary
I think there is a misunderstanding. What I said is "The one that sits in this sweet spot is the E-MU 1616m". This means that the 1616m is good, not bad. My comment is about 2 points:
1. Not every audio interface is a good candidate for objective measurement. The circuit, parts and implementation must be carefully evaluated. The 1616m is a winner, many are not.
2, Many audio interfaces cost well under $1000, some costing just $100 to $200 retail. At these prices, parts quality are not always the best. And that means they do not last, or they are not always linear. Even if the circuit, parts and implementation are all correct, if the parts quality do not stand up to critical use, then it's only as strong as the weakest link. In the case of the 1616m, the problems are in the caps count and their low quality. You can see bulges in many of the caps in the pictures listed in the post.
As to the drivers, there has been no issues with Win7 X64 or Win10 Pro. Touch wood - I have not seen a BSOD ever.
Ok, I misunderstood, sorry.
About point 1, I agree they are not all as good as the EMU, but many are more then good enough.
About point 2, haven't checked the caps, but so far the card perform normal
The drivers, I run on win7 32, and gives sometimes touble on pc, and from googling for a solution, I was for sure not alone (but havent found a solution)
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The drivers, I run on win7 32, and gives sometimes touble on pc, and from googling for a solution, I was for sure not alone (but havent found a solution)
This works for me - but be careful these are the exact drivers you need, which may not be the ones on the Creative support site.
1. Install first the driver (EmuPMX_PCDrv_US_2_30_00_BETA.exe),
2. Reboot,
3. Then the app (EmuPMX_PCApp_US_2_20_00.exe)
Good luck. It may or may not work, so YMMV.
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This works for me - but be careful these are the exact drivers you need, which may not be the ones on the Creative support site.
1. Install first the driver (EmuPMX_PCDrv_US_2_30_00_BETA.exe),
2. Reboot,
3. Then the app (EmuPMX_PCApp_US_2_20_00.exe)
Good luck. It may or may not work, so YMMV.
Thanks. If it start acting up again, will give that a try.
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DajMasta,
A $200 soundcard, $100 measurement mic and free software will not be the limiting factor in your tests. Measuring audio equipment and running acoustical testing accurately is not a trivial business. Relating measurements to subjective observations is even trickier.
Take for example what you said below.
"Most of the testing I have in mind is either strictly electrical or acoustic with as much of the room taken out as possible (a source and a mic in a small isolation chamber in a quiet room)"
The space you measure in will dominate any tests at real audio frequencies. Are there things you can do? Sure, windowing for example is very powerful and can create quasi anechoic measurements limited in low frequency range only by the size of your room (which determines when surface reflections make it back to your mic).
Get an affordable sound card, a cheap calibrated mic and start measuring. You will learn a lot by doing.
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I am in a similar situation but my budget is currently limited around 1K. So far these are the unit I'm considering:
- QA401
- AP SYS-222
- JensH Audio Analyzer (http://www.diyaudio.com/forums/equipment-tools/277808-diy-audio-analyzer-ak5397-ak5394a-ak4490.html)
The QA401 seems a very nice product. Yes it comes with its own software but as far as i know it works well and the software team seem responsive to user request. I don't like the BNC connectors and the fact you need to make your own adapters cable but it is probably the best bang per buck. I'm very tempted to buy it. The AP system one is a very old unit but it is the only option if you want to bring the best brand audio analyzer in you shop. The performance I guess are in the same neighborhood of the QA401 (?) but you need an old pc with ISA card and that doesn't seem very attractive to me.
The last one is not yet a product available in the market but it seems the more promising to me. I haven't followed the entire discussion on diyaudio.com but it looks like will give better performance compared to a QA401 and also ASIO support. I'd like to hear your thoughts.
Alessio
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I haven't used the Quant, but will probably pick one up for fun this year.
I have owned an AP system one and it is a wonderful, huge and powerful beast. But it is unsupported, and not trivial to set up. FFT is very powerful and missing on the older S1.
Jens' project looks great but again learning what to do is more important than waiting for "better".
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- JensH Audio Analyzer (http://www.diyaudio.com/forums/equipment-tools/277808-diy-audio-analyzer-ak5397-ak5394a-ak4490.html)
The last one is not yet a product available in the market but it seems the more promising to me. I haven't followed the entire discussion on diyaudio.com but it looks like will give better performance compared to a QA401 and also ASIO support. I'd like to hear your thoughts.
Looks at first glance like a very nice produc, but would like to see a performance summary first. The hardware is certainly up to the task, and the fact that is supports ASIO will make it much more attractable then the QA401... Thanks for pointing this unit out to us.
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I'm currently finishing up a headphone amplifier build and I got the QuantAsylum QA401 when I started developing the amp. With todays extremely low noise and distortion figures on op amps, You'll hit the bottom of what the analyser can measure very quickly. I had to build a twin-T notch filter and get hold of a extremely low THD signal generator to be able to measure below -108dB THD. Getting a industry standard Audio Precision is the dream, but they are way to expensive for me...
But the question is, why would we need to see lower than that? It can be fun as a technical exercise, but it will not improve the sound quality we hear any more…
Going down the rabbit hole is part of the fun. I want it to measure as good as possible.
Sent from my iPad using Tapatalk
There is also the QA aspect of this - if you designed your channels to measure very low, any noisy component or problems in the amplification stages will pop up in the THD+N. They may not be audible, but you production run has a problem. It is no longer a Blameless amp.
I think that is the reason companies like Bryston measure very low and add a cert sheet to every amp - QA. I do not have the ability to measure (nor hear) that low - but their amplifiers are very nice (every bit as nice as every other blameless amplifier).....
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Going back to the OP's 2 applications, monitoring a FFT from complex audio signals in real time is quite a different matter than using an ultra pure sine wave to measure the THD+N of a circuit. These are very different needs, if only one devices is used to satisfy, that device would need to be of instrument grade and capable of doing so.
For the first applicataion, B&K at the high end and AudioControl at "everyday" price levels come to mind. The signals come from microphones and go into a device that "analyzes" and displays the level in each frequency band. They respond fast but depending on the sophistication of the device, price varies. AudioControl RTA boxes sold for years around $200 with a microphone, but there are no bins or frames to poke into. B&K, the Danish sound & vibration expert that supplies everyone from NASA to Mercedes Benz made very competent microphones, mic pres and analyzers, but again, their output reflects the need to characterize a complex audio signal and sort out that hairball in a meaningful way. Their tools weed out the chafe.
For developing circuits, every "chafe" is meaningful, because some glitch that happens at -100dB has a root cause, perhaps originating from somewhere else at -110dB. In the time domain, scopes that see reliably see these glitches are highly prized, and many tools have been developed over the years to catch these elusive glitches more reliably and visibly, precisely because they need to find the root cause. In the frequency domain, the need and technique is no different - just because it's not evident at -60dB where it is clearly audible does not mean there is no problem at -100dB. Simply putting on a set of good headphones will reveal significantly more normally inaudible glitches - that's why recording studios use all 3 ways to monitor a session - the soffit-mounted 15" monitors, the mini's on the console meter bridge and headphones. Any engineer who lets a mix go to tape without first getting all 3 right would not have a job for long.
An expensive analyzer like an AP or Rohde & Schwarz can do both, but they are not cheap and is not an casual purchase, unlike a USB soundcard or a freeware FFT. But these instruments are made by seriously competent people with very demanding customers, and their findings are not compromised by a budgetary constraint or makeshift construction. Of course, there is a component of brand value in their price, but in the long run, it really comes down on how much the user needs to rely on their findings, and what value they place on these findings.
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Going back to the OP's 2 applications, monitoring a FFT from complex audio signals in real time is quite a different matter than using an ultra pure sine wave to measure the THD+N of a circuit. These are very different needs, if only one devices is used to satisfy, that device would need to be of instrument grade and capable of doing so.
Why are they different? REW, a free and very well supported piece of software can do both with it's signal generator and FFT. It won't do stereo but I find that it's rarely a limitation.
Your point does get to the heart of something though - that everyone has a slightly different approach to measurement. I for example am not very interested in THD+N as a metric, with out proper filtering PSU hum (especially in tube gear) can be a contributor that doesn't have a huge subjective effect. I'm more interested in the harmonics, when I see higher order harmonics I am more concerned.
The O.P. needs to start his journey of measuring to understanding what it is that he needs.
Simply putting on a set of good headphones will reveal significantly more normally inaudible glitches - that's why recording studios use all 3 ways to monitor a session - the soffit-mounted 15" monitors, the mini's on the console meter bridge and headphones. Any engineer who lets a mix go to tape without first getting all 3 right would not have a job for long.
This is not the case. I know many very famous engineers who never use headphones and who never check on the mains (lots of newer studios don't have mains and in older spaces they often sounded dreadful)."
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Why are they different?
They are different because soundcards typically do not include DSPs for front-end signal processing, which are almost always implemented in high-end analyzers to improve accuracy. The incoming signal is scrubbed then passed onto 32-bit or higher-precision DSPs to do the processing in the digital domain. Soundcards may have analog "processing" but the intent is very different hence the design would have to follow suit.
This is not the case. I know many very famous engineers who never use headphones and who never check on the mains (lots of newer studios don't have mains and in older spaces they often sounded dreadful)."
If you are Al Schmidt or Ed Cherney, sure, you would probably skip not only headphones, but a few other steps because you have 40+ years under your belt. But if you were hired by Al Schmidt or Ed Cherney to assistant-produce at the board and you left a few glitches in a pricey session by not triple-checking before the talent got back on their private jet heading to their next gig, you'd be toast on the spot - guaranteed. Perhaps "engineers" only tracking and mixing on Yamaha NS-10s are the real reasons why recordings sound the way they do now, despite all the much better tools available in modern rooms.
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ci11,
There's no question that a high end analyzer has extras that a sound card does not, auto ranging being the most important, input impedance switching is another, on some units there are analog notch filters pre A/D. I don't really understand your point re dsp. Sure, the Apx555 has some tricks up it's sleeve to lower distortion in the analyzer section but typically analyzers are looking to digitize the signal with the least distortion, then analyze.
My point is that you can do very real work with a sound card and software once you understand the principles involved. Would I prefer a real audio analyzer? Sure, but I can go very far with a sound card and REW. The O.P. will find the complexity of something like an A.P. daunting and will be limited by his understanding and technique until he has some experience measuring acoustically and electrically. I've measured power amps, very large SSL mixing consoles, many many 6 figure speaker systems and lots of audio hardware with an interface and free software. This is real work, in the field for professional clients.
Let's not argue about your second point. I live in that world here in L.A. and work with engineers and producers all day long at every level from beginner to the top level. You paint a picture from a very different era.
Why are they different?
They are different because soundcards typically do not include DSPs for front-end signal processing, which are almost always implemented in high-end analyzers to improve accuracy. The incoming signal is scrubbed then passed onto 32-bit or higher-precision DSPs to do the processing in the digital domain. Soundcards may have analog "processing" but the intent is very different hence the design would have to follow suit. Perhaps "engineers" only tracking and mixing on Yamaha NS-10s are the real reasons why recording sound the way they do now, despite all the much better tools available in modern rooms.
This is not the case. I know many very famous engineers who never use headphones and who never check on the mains (lots of newer studios don't have mains and in older spaces they often sounded dreadful)."
If you are Al Schmidt or Ed Cherney, sure, you would probably skip not only headphones, but a few other steps because you have 40+ years under your belt. But if you were hired by Al Schmidt or Ed Cherney to assistant-produce at the board and you left a few glitches in a pricey session by not triple-checking before the talent got back on their private jet heading to their next gig, you'd be toast on the spot - guaranteed.
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I've measured power amps, very large SSL mixing consoles, many many 6 figure speaker systems and lots of audio hardware with an interface and free software. This is real work, in the field for professional clients.
What is your interface of choice for such a job ?
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I'm using a Prism Sound Lyra 1. In the interests of full disclosure I do some work with Prism's U.S. distributor, mostly bench repair work for the Maselec line of outboard gear. I've used a variety of interfaces in the past though.
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There's no question that a high end analyzer has extras that a sound card does not, auto ranging being the most important, input impedance switching is another, on some units there are analog notch filters pre A/D. I don't really understand your point re dsp. Sure, the Apx555 has some tricks up it's sleeve to lower distortion in the analyzer section but typically analyzers are looking to digitize the signal with the least distortion, then analyze.
My point is that you can do very real work with a sound card and software once you understand the principles involved. Would I prefer a real audio analyzer? Sure, but I can go very far with a sound card and REW. The O.P. will find the complexity of something like an A.P. daunting and will be limited by his understanding and technique until he has some experience measuring acoustically and electrically. I've measured power amps, very large SSL mixing consoles, many many 6 figure speaker systems and lots of audio hardware with an interface and free software. This is real work, in the field for professional clients.
Let's not argue about your second point. I live in that world here in L.A. and work with engineers and producers all day long at every level from beginner to the top level. You paint a picture from a very different era.
I hope my points are helpful to address the OP's applications. These comments were intended to urge the OP - or anyone - to thoroughly understand their application needs and pro's can con's of each option before being lured into decisions based on a seemingly low entry threshold, be it price or complexity.
Not many people discuss the real differences between soundcard-based solutions and current, purpose-built audio analyzers costing much more, probably because few think they could ever afford them so it is easy to deny their significance. But these differences are there not only for those who need them and are willing to pay for them, but help define or inspire future requirements for hardware and software developers, and they warrant discussion. Whether they are necessary is an individual decision, but the state of the art is moving forward, albeit slowly as they start bumping into limits imposed by physics.
"I don't really understand your point re dsp" Attached are 2 pictures lifted from AP and R&S literature that describe the front-end processing of their analyzers. These have nothing to do lowering THD+N, that's something different implemented elsewhere. Compare their approach to soundcard-based solutions, and the difference is obvious. These circuits simply do not exist on soundcards or analog-based outboards. They help the user see better and analyze more. Again, perhaps this difference is not needed or justified, but the current capabilities are clear and deserve to be mentioned. Even if some aspiring developer writes this pre-processing for a Raspberry Pi and put the software on Github for free, a tracking analog notch filter setup and input protection is not a trivial add-on to a soundcard.
The "picture I painted" is that of successful engineers who have a backlog of work because they deliver good results every time, on time and with no excuses or budget overrun drama becasue they run a tight ship. Many people don't want to or even need to work that way, today, yesterday or tomorrow in LA, Nashville or NYC. That's is not the point relevant to this discussion - the point is that headphones reveal more details that can easily be lost, and those details may point to problems that started somewhere else, and hence the need for better-performing instruments to chase after those root causes - if so desired.
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ci11,
Let's not derail the thread by arguing. I agree with you that anything we can do to help the O.P. understand the issues is useful to him and possibly others. Again I state that the limiting factor for now will be the O.P.'s understanding of the issues and techniques, not the gear. Investing is higher end gear makes little sense (and I love high end test gear!). I don't mean that in a demeaning way at all, I expect that when I finally get my hands on an APx555 I will be the limiting factor too :-)
As far as I can tell, all the DSP I see outlined in both of your examples below is handled in software in a typical sound card based solution. The hardware tracking analog filter you mentioned was already discussed in a previous post of mine and is part of yielding the higher performance dedicated gear can give.
There are real differences between dedicated audio test gear and bodged together solutions, some of which will matter to the O.P. and some of which may not. The NWavguy blog does a pretty good job of talking through some of these issues - http://nwavguy.blogspot.com (http://nwavguy.blogspot.com)
The DIYaudio forum is another great resource.
I agree that quality matters. I pursue it relentlessly in my mastering work, the gear I use, room I work in etc. I want and need gear that can measure lower levels of distortion, and as you say others will not need that or care.
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Attached are 2 pictures lifted from AP and R&S literature that describe the front-end processing of their analyzers. These have nothing to do lowering THD+N, that's something different implemented elsewhere.
These have almost exclusively to do with improving THD+N. By notching out the test signal (sine), and amplifiing the remaining signal, one can see the lower harmonic distortions much better, because the distortion signals are now amplified, and the test signal is not (otherwise it would be impossible to amplify the input, because the ADC will be overloaded).
If I want to go really low in THD measurements, I do exactly the same by putting Bob Cordell distortion magnifier (http://www.cordellaudio.com/instrumentation/distortion_magnifier.shtml (http://www.cordellaudio.com/instrumentation/distortion_magnifier.shtml)) in front of my sound card. This unit subtracts the test signal from the return signal of the DUT. This distortion magnifier can be bought in a kit from 128 from pilghamaudio (see link at cordellaudio site).
As discussed before, sometimes you need to be a little more creative than having an all in one solution, but very very good results can be obtained using affordable gear. As also said before, the main difficulty is understanding what to measure, and interpreting the result correctly, and the audio precision tools want help in that department
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There are real differences between dedicated audio test gear and bodged together solutions, some of which will matter to the O.P. and some of which may not. The NWavguy blog does a pretty good job of talking through some of these issues - http://nwavguy.blogspot.com (http://nwavguy.blogspot.com)
Agreed.
The goals dictate the means. With a plethora of available options, many who read these posts do not get a full pictures. Sometimes, it helps to expand, and other times to focus. In this thread, I hope it is clear that there are as many ways to skin the cat as there are opinions, but some very smart people have made important advances to bring more light to the matter. I offered my comments based on this exact spirit.
I wish you well.
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These have almost exclusively to do with improving THD+N.
Yes, this processing may possibly lower the THD+N of an input signal but its primary purpose is to clean up a "dirty" signal for the ADC before sending it to the DSP for processing. The THD+N improvement circuit for the sine wave output of the internal generator in an APx555 (to lower it from -117db to <-122dB) is a separate matter, and that is what I was referring to.
As discussed before, sometimes you need to be a little more creative than having an all in one solution, but very very good results can be obtained using affordable gear. As also said before, the main difficulty is understanding what to measure, and interpreting the result correctly, and the audio precision tools want help in that department
Agreed. There is quite a learning curve, and it would be great if analyzers can be rented so those interested can use it on real projects to see if it is their "cup of tea" and it can do their job.
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THD+N is perceptually blind. That is, its value does not correlate with how we hear (i.e. ignores masking thresholds). For that reason I don't use THD analysis in my AP. Just do a spectrum analysis with the original signal still there and look at the distortions spurs.
Same thing can be done in software and computer power is superior there to all-in-one-boxes. The down side is knowing what you are doing with respect to selecting windowing function and such.
BTW, I just met with Prism Sound folks at CES and because of Brexit, they said the retail price has dropped $2000 and you can now buy their analyzer for $6K! Pretty cheap in my book. :D
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For what its worth I use regularly a Boonton 1120, a modified Shibasoku 725, a QA401 and 4 different soundcards. Its not the tools but knowing what to look for. Any of these can show you pretty much any aberration if you know what to look for.
The QA401 + a handful of BNC-RCA or BNC-Banana adapters makes it compatible with 90% of anything you may want to do. There was a possible promise of a standard audio driver for the QA401 but not yet. And there is an API if you want to create specific tests or test suites. I think its the best value simple because it will just work. A sound card + HPw works or Arta or REW or RMAA or . . . will work fine after you sort it out which could take some time even if you have done it before. Even RMAA which is simple can lead you astray if you don't have the right interfaces selected etc.
Jen's analyzer will be at least 2X a QA401. And its slightly better than an Emu 1616M or 1212M on the ADC (better analog) but the EMU DAC is not as good. (I mostly use the EMU in XP for stability but Win 10 is OK with a clean install). The QA401 has an automatic input attenuator but is still limited on its peak input. Same for most audio stuff. Very few won't smoke when measuring AC line distortion for example.
However I need to measure everything from headphones and speakers to SOTA DAC's so many options are needed. Some which can't be done with an APx555 without an additional $10K of addons.
The RME babyface would be a great starting point with software (RMAA?) and some interface cable/adapters. Learn how to make some measurements and how to see when its all lying to you. When you need more look at the options.
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Thanks for all the input so far, the discussion of the differences between the frontends of an audio analyzer vs. that of a soundcard have been helpful too.
I think part of the appeal of a purpose-built analyzer, at least initially, was that it was so well characterized. If I make a mistake in taking a measurement and realize it, then the finger can almost never be pointed at an analyzer because all the specs are laid out and built to be stable. It's been amazing to me how many high-end interfaces there are available which don't even have a manufacturer-specified noise floor or THD rating, but every dedicated analyzer will have data on that, evenness of frequency response, and many other aspects. Looking around, there has been some helpful info in the form of consumer run tests of some of these itnerfaces, but the "guarantee" of reputable test equipment has it's appeal.
That said, most of the cheaper, but still well performing, dedicated analyzers aren't as well suited for my FFT tasks as a soundcard is. All-in-one units like the UPL make extended data capture a pain, and computer-attached units like the ATS-2 have options, but without ASIO or other sound drivers, don't have easy ways of getting that data into other software, some of which seems ideal for the kinds of recording and visualization I want to do.
Not being in an industrial environment or testing totally unknown systems means the input protection and attenuation options are less important, and if I really wanted absolutely quantifiable numbers for signal strength and whatnot.... I can just use my multimeter or scope to get the voltage of a sine at X output level and do a little math to convert the axes to volts or what have you.
I've also been sort of insisting on the THD+N figure specifically for the +N, something I've run into issues with before. I've had pretty significant hum on supposedly good internal PC sound cards, and have only been satisfied with the noise floor for normal recording levels with my latest refresh of equipment - both the mics and the interfaces I've been using before have been audible. While I don't expect to be hearing noise under -100dB or something, I have heard some in the -90s, so I wanted to aim as low as possible to avoid issues with it, even if the parts of the signal I probably care about are still well above even the audible noise floors.
The EMU cards sound like great choices, and they're pretty cheap on ebay right now, but they're internal and I don't want to be hauling around a desktop to take measurements... plus my previous issues with in-case emi or power conditioning related hum. Looking at lots of external interfaces, the newer Motu 624 looks like a good contender. Notably lower advertised noise floor than the babyface pro at only slightly more price, a compact size, and the signal paths seem to be the same hardware as the Motu 1248, which has been characterized by some end users and seems to meet its spec and have very flat frequency response. A Prism dScope III ticks all the boxes across the board and seems like a great unit for the $6000 or so price tag..... but that's still quite a price tag, and I think a lot of its capability would be lost on my applications.
I agree the QA401 seems like a clear winner for general audio testing in terms of value for performance, but if I can get a good preamp with phantom, the right connectors, standard sound drivers to use whatever software I want, and a lower noise floor out of an interface, I think it's a better choice for me.
So that's the plan for now, a good interface coupled with rightmark and spectrum lab because they're free. Once I get a better feel for the software side and the measurements/data recording I want to do, I can invest as needed in a software suite.
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I think part of the appeal of a purpose-built analyzer, at least initially, was that it was so well characterized. If I make a mistake in taking a measurement and realize it, then the finger can almost never be pointed at an analyzer because all the specs are laid out and built to be stable.
This is by far the reason I use my Audio Precision. I bought a TI EVM with their nice ADCs and AES capture but I had no idea what it was supposed to have spec-wise. I publish my results so I need others to be able to trust and/or repeat them. With sound card you just can't do that. There are also potential issues around ground loops and such with PC cards which are reduced if not avoided by using an external analyzer.
Still, the cost is just way too high for satisfying one's curiosity and or casual measurements. So going with the PC solution as you mention is the right path for you for sure for now.
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Not being in an industrial environment or testing totally unknown systems means the input protection and attenuation options are less important, and if I really wanted absolutely quantifiable numbers for signal strength and whatnot.... I can just use my multimeter or scope to get the voltage of a sine at X output level and do a little math to convert the axes to volts or what have you.
Probably a workable idea but the linearity and flatness of frequency response of the DMM and especially a scope you intend to use should be checked since accuracy at audio frequency bandwidth is not what the designers of these devices usually have in mind. Many DMMs can only measure Vrms out several hundred Hz and not beyond.
The EMU cards sound like great choices, and they're pretty cheap on ebay right now, but they're internal and I don't want to be hauling around a desktop to take measurements... plus my previous issues with in-case emi or power conditioning related hum.
The E-MU 1616m was made with 3 different interfaces: PCI, PCIe for desktop and a Cardbus interface for use with laptops. All function equally well. And the E-MU 1616m "Microdock", the breakout box where all connections are made to analog and digital I/O, supplies 48V Phantom on both mic input channels.
Also attached is a plot from the noise floor plot of my 1616m using ARTA. Input was shorted but plot used the Hanning window but not averaged. Averaged noise floor would be around -150dB for this setup. The 1616m was plugged into wall AC and there does not appear to be an issue with EMI or hum, even at 60/120Hz here.
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Very impressive! I think all I had seen on ebay was the internal card versions, and PCMCIA would need an adapter for modern laptops (I've had a couple laptops since I had one with a PCMCIA slot!), but adapters certainly do exist.
I remember a little after when Creative bought them, I was looking at sound cards including a couple of E-MU options. I remember them not actually being too expensive, it's neat to think they were well enough designed to stand up so well for so long. Given that they were priced to compete with high end consumer options, I wouldn't have expected the longevity.
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Very impressive! I think all I had seen on ebay was the internal card versions, and PCMCIA would need an adapter for modern laptops (I've had a couple laptops since I had one with a PCMCIA slot!), but adapters certainly do exist.
I remember a little after when Creative bought them, I was looking at sound cards including a couple of E-MU options. I remember them not actually being too expensive, it's neat to think they were well enough designed to stand up so well for so long. Given that they were priced to compete with high end consumer options, I wouldn't have expected the longevity.
They are out there so patience will be rewarded. They are not always well kept or maintained by previous owners, and they do run hot. So buy carefully and be prepared to re-cap the entire box. Attached are the before- and after- of the re-cap of the top PCB on the 1616m to complement the bottom PCB pictures that had already been listed earlier in this thread. The number of popped caps can be clearly seen in the before- picture.
Good luck.
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They are out there so patience will be rewarded. They are not always well kept or maintained by previous owners, and they do run hot. So buy carefully and be prepared to re-cap the entire box. Attached are the before- and after- of the re-cap of the top PCB on the 1616m to complement the bottom PCB pictures that had already been listed earlier in this thread. The number of popped caps can be clearly seen in the before- picture.
Good luck.
What values of the caps have you soldered to A+ A- B+ B- of akm IC and to 2068 between V+ AGND and V-?
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They are out there so patience will be rewarded. They are not always well kept or maintained by previous owners, and they do run hot. So buy carefully and be prepared to re-cap the entire box. Attached are the before- and after- of the re-cap of the top PCB on the 1616m to complement the bottom PCB pictures that had already been listed earlier in this thread. The number of popped caps can be clearly seen in the before- picture.
Good luck.
What values of the caps have you soldered to A+ A- B+ B- of akm IC and to 2068 between V+ AGND and V-?
Please download the picture from the post, put red circles where you want the caps identified and I will get the type and values for you.
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Please download the picture from the post, put red circles where you want the caps identified and I will get the type and values for you.
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Please download the picture from the post, put red circles where you want the caps identified and I will get the type and values for you.
Here are Mouser PNs for the caps used:
1. Blue tops - 661-APSE6R3L561MF08S
2. Red tops - 647-RNE1C101MDS1 - these are also used near the 2068s.
Attached is a closer look at them from another angle.
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FWIW here are directions for getting the EMU 1616/1212 running in current Win 10: https://www.kvraudio.com/forum/viewtopic.php?t=529349 (https://www.kvraudio.com/forum/viewtopic.php?t=529349) I can vouch that they do work, at least in my system.
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Thanks for all the input so far, the discussion of the differences between the frontends of an audio analyzer vs. that of a soundcard have been helpful too.
So that's the plan for now, a good interface coupled with rightmark and spectrum lab because they're free. Once I get a better feel for the software side and the measurements/data recording I want to do, I can invest as needed in a software suite.
I think you have been asking good questions. Fwiw, I think someone could download the free version of something like this and learn a lot with a sound card in a PC plus a minimal investment in a mic and phantom power supply.
https://trueaudio.com/rta_selection_guide.htm
Just follow the hardware suggestions. The free version will give way to a paid version and the little bit of entry hardware will cost still more, so this path is not free for long but it can be a good starting point.
In a few days the test signal generation, sweeps, equalizations, and lots of measuring would likely be enlightening - which in turn would help navigate the rabbit hole.
What I think you might find near the bottom of the rabbit whole is that you can have test equipment that would make Mr. Hewlett and Mr. Packard envious and you can design and build or buy exquisite hifi gear (speakers, amp, preamp, source players, etc) and you can work your butt off to get the measurements to coincide with what you are hearing and vice versa, but you will never fully get “there” until you address the listening room acoustics. All the gear inside the system is not the full system; the room is a MAJOR part of the system. And when you realize the dimensions and construction techniques needed to address the room the cost for all the test gear and hifi gear might start to look like just a moderate % of the room treatment costs.
This is not to say there aren’t clever practical cost-effective solutions but rather that the sooner you see the impact the room has on the speakers (in concert with speaker and listening position placement) the sooner you will start to recognize the full scope of the rabbit hole.
Once you settle in on speaker design that will inform some electronics requirements, and it can be an iterative process, of course. Good electronics and a good source player can reveal some pretty exciting nuances but the frequency response and spatial imaging (including the sense of “air”, “transparency”, “definition”, “detail”, “transient response”, “slam”, “boom”, yada yada) are all a result of eliminating distortions (especially distortive interactions) that have to be first managed at the speaker/room level before the electronics can fully do their magic on a more subtle level - all of which will be beholden to the reflections and nulls that will dictate timing interactions that will have both a coarse and a subtle impact on what reaches your ears - so the room must (should) be addressed before all the nuances can fully surface. If you get this far it will make you marvel at how all the original detailed content ever got captured on the source media in the first place.
Overall it can be a very fun and rewarding journey.
Of course hifi can be intrinsically rewarding without addressing any of this. Just install whatever hifi gear in some room and you can hear the Beatles and Beethoven. It’s just a matter of how accurately you want to reproduce the sound of what was recorded - how much high fidelity (faithfulness to the original) you want, and how much you want to measure and understand what causes what.
Net, net: The suggestion on starting with the sound card, software, and mic is that it will (should) quickly show an experimenting user the impact of the room that is almost certainly going to be a major part of the system and the system’s performance.
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Hello guys, newbie here (but lurking around for years), first post.
Recently I aquired a complete 1616m for quite cheap.
Looks good from the outside. Seems hardly used.
Plugged in the power supply (which BTW get`s freaking hot, is this normal?) and seems to work too.
Currrently I am about to set up a laptop system for the EMU cardbus which came with it.
I want to use it as an audio measurement system as discussed here.
Took off the hood of the 1616m Microdock. Just so far that I can peak inside but have not (yet) completely disassembled it.
Caps seem to look still okay too, at least as far as I can see but regardless I would want to change them ALL as they are ALL crap and at least those in the power-supply will ALL fail (sooner or later).
I still don`t get it, how in an otherwise fine instrument, one could put in such a load of junk (and there are MANY caps inside).
As some of You have already done excactly what I still intend to do, I would like to ask if someone could post a list of all the caps in the EMU-1616m Microdock with values (µF/voltage-rating) and if possible lead spacing as well?
Would be really grateful as it would spare me to disassemble (+ figure this out myself) and reassemble the thing twice.
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The 1616M is a remarkable product but having inadvertantly fried one its not easy to get back together. There are two independent sets of pin headers from top to bottom that must be correct or it will become toast.
The caps are not great, however if it has seen little use changing them all out is a lot of hassle for little return. I'm sure the BOM for all those caps ($) had a lot of influence into the decisions.
I think I collected most of the available reverse engineering on it. If asked I'll look for it. Someone worked out how to hack the firmware to get it to work on Win10. I have those notes somewhere. Finding a computer with a cardbus interface will not be easy. I use a desktop with the desktop interface.
It has 6 channels of AK5394A ADC, the best performing ADC from AKM. It also has a plethora of JRC opamps that seem to be pretty good.
The power supply should not be that hot. I got some generic 48V supplies that work fine as a substitute.
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Hello 1audio,
I have read Your posts over at diyaudio about the EMU 1616 and took quite a few notes. :-+
Thanks for that and thanks for Your reply!
Good to know, to have in mind and to take extra care about what You mentioned regarding the pin headers!
Already yesterday I was almost about to dismantle the EMU - but wimped out...
Yes, judged by the cosmetic appearance I think my EMU 1616 has seen little use but that does not mean much IMO. Could be it was just handled gently.
And only a few days ago I had a failure with another oldie but goodie which I know was hardly used. A Terratec Phase X24 FW I still use occasionally.
Although it is really old as well, I still like it a lot not only for it`s headamp and use it with my desktop PC (win7).
However, have not used the Phase X24 a couple of years. Switched it on recently, it came on but did not work properly anymore (sound just stopped after a few minutes).
Took it apart and the first thing I noticed was one cap in the power-supply already bulging. That was definitely not the case before, because when I got it, I opened it up, of course.
That means the cap has gone bad meanwhile, even without using the gear. While I was at it, measured a few other caps. Some more dead or at least highly suspicious.
IMO does not make much sense to change just the already obviously bad ones.
Same as with the EMU, the only ones which could be kept in are the coupling caps, maybe, I believe.
I am aware that changing the caps is a lot of work and it won`t be easy. More so as I don`t have a vacuum desoldering station.
Would have to do it the "old way" with desoldering braid and/or manual desoldering pump. Maybe it is time to invest in a proper desoldering tool eventually...
From what I have read, changing anything else besides the caps does not bring noticeable improvement, if at all. So I will leave it at that.
>> Cardbus. Not long ago I bought a second Dell Latitude E6500 notebook just for the EMU and older software I still want to use.
My first DELL I bought second hand >>10 years ago. Still in use and still going strong (for what I use it anyway). I love these DELLs.
Today they are dirt cheap and go for only a couple of bucks. The EMU 1616m (+ Cardbus + 2 PCI cards + 2 Power supplies) + the DELL did cost me only slightly >100€ together.
Not bad for something what could potentially almost rival an AP for audio stuff (minus of course the attenuators etc., etc. the AP has to offer).
Installed dual-boot WinXP and Win7 on the DELL and deactivated all the stuff I don`t need. So no need for me to hassle around with Win10 (I really hate anything >Win7).
I still need a lot of TRS cables for the 1616m though, which will cost me quite somewhat, even when soldering the cables myself.
My 1616m came with 2 MEAN-WELL 18W/48V power-supplies. Both get hot. Measured the power consumption: 14-15W. So just shy of the rated 18W.
Seems EMU cheaped out not only on caps but on the power-supply as well.
Or maybe this is a sign that already something in the EMU Microdock does not work as it should (albeit it seems to work okay, as far as I can tell)?
As I don`t like things running so hot, I will replace the Mean-Well with something more powerful and somewhat better specs wise, if possible.
Maybe a PS with better specs (output ripple) can have an impact on overall quality as well. Maybe not.
So, if nobody comes up for the EMU 1616m with something similar as on the attached picture (or a list of caps, so I can order parts, without having to disassemble the EMU first)
I have done (precautionary) for the Phase X24-FW years ago, (unfortunately) I have to do it myself.....
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I posted in this thread during 2016/2017 after deciding to find an E-Mu 1616m as a part of a DIY audio analyzer. I completely recapped the "dock" portion of the unit, which connected successfully to a Thinkpad of mine that has a PCMCIA slot. After the hardware was in order, I also successfully installed the E-MU proprietary software under Win10 after a number of misses and gyrations. Expending all that effort required leaning on many who followed this path previously for help,, including @1audio who had been an invaluable resource. Thank you again, 1audio.
The question is - would I do it again? The answer is a resolute NO. Here's why:
1. The dock unit runs uncomfortably warm due to having a lot of parts on 2 PCBs inside an unvented enclosure. Its stability has not been verified for instrumentation purposes.
2. Its audio performance is no better than a modern, purpose-made audio analyzer such as a QuantAsylum QA401 which is very reasonably priced for what it does, and it has an up-to-date software package with an easy to use UI.
3. The E-MU software install was VERY tricky, requiring many unsupported, back-level versions that may eventually be crippled as Windows Update continue to unapologetically disable older software. Even though it worked, it was held together with Band-Aids. Moreover, the functionality and ease of use of the audio interface software was anything but intuitive. E-MU was sold to Creative a long time ago and Creative has long since dropped support for anything E-MU.
4. The hardware upgrade was not difficult but certainly tedious. Some caps have 2.5mm lead spaclng so a desoldering iron will be very helpful.. Attached is my Mouser order for parts that I determined should be changed. The part number cross references should be verified again as there may be mistakes I did not correct. These parts went into the "after" picture I posted from my refurbished PCB in post# 65 on this thread.
Even with a perfectly restored E-MU 1616m - or any other competent sound card - a complete audio analyzer requires much more to be actually considered "competent". Functions like ultra-pure test tone generation, input filtering and attenuation, notch filtering, post-ADC digitization for DSP digital filtering all help to ensure measurement results can be relied on.
All the best in your quest.
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Hi ci11,
thanks a lot for the list! Very helpful and very much appreciated!
The QuantAsylum QA401 You mentioned I already had an eye on and (not only after what You wrote) I indeed have to contemplate some more if this would not be a (better) option for me.
Lastly this is the only one of the dedicated analyzers for audio in the price bracket of what I could still justify to spend.
Regarding the EMU, I`m not so concerned about software and installation. As mentioned, I already have a dedicated DELL laptop with WinXP and Win7 for it.
So no need to hassle with Win10 installation, band-aids to get it running and/or worrying about (Win) udpates to keep it up running.
The EMU with Win7 should not be a problem, even less so with WinXP.
However, the EMU PatchMix software is indeed very confusing to me but I have not yet made any serious efforts to understand how it works.
I`m convinced, once necessary and when I really get into it, I will handle it.
What I like about the 1616m is that I could use it not only for electrical measurements but with various software (ARTA for example) also for speaker measurements.
I do have a dedicated PC based speaker measurement system though - but to use it, I have to carry around the clunky, heavy, loud 19" PC case with an even older ISA mainboard in it.
With a 12V to 48V power-supply + battery for the EMU and the laptop also on battery, the system could be running even completely independent from mains voltage (thinking car audio or outdoor speaker measurements).
What I`m also not much concerned about is the heat issue. I`m convinced, once recapped with quality caps, the EMU would last a long time, despite the heat (as I`m already of what could be called somewhat "advanced" age, it might even outlive me....).
Especially as I intend to use it for measurement purposes exclusively. It would be comparatively short times it will be actually in use.
Moreover the heat issue could be solved with some more case vents or maybe even a small fan.
I do not worry about "verified instrumentation purposes". This is just for occasional hobby usage.
What is indeed the biggest drawback with a soundcard are missing attenuators and to some extend a low-distortion generator (but I have an old Krohn-Hite which I think would be good enough) and finally a notch-filter.
As always back at the very basic two options: soundcard + DIY periphery or spending the bucks on something like the QuantAsylum (+ accessories). Hmmm...
There are worse things in live than this kind of "luxury" decisions.. 8)
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If I can find the "smoking hulks" I'll take good photos of the boards and share them. Not sure where in my piles of stuff they are but good to use as a starting point.
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According to the QuantAsylum forum the QA404 will be available June/July 2023.
https://forum.quantasylum.com/t/qa404-and-qa403-update/795/24