fuzzmeasure manual german

The frequency response you see is derived by doing an FFT of the portion of the impulse response you select - that way you can select a small portion near the beginning to see predominantly (or entirely, if short enough) the direct signal from the source, or a much wider portion to also see the room's contributions. The shorter the portion you select, however, the less frequency resolution you have and the higher the lowest valid frequency in the response. A single RT60 figure for the whole response is not very meaningful, because the RT60 varies with frequency. It is more common (and useful) to calculate the RT60 figures at octave or one-third octave spacings, which is what FM is showing you. You cannot determine RT60 times from waterfalls. The waterfall is produced by moving a window through the impulse response and plotting an FFT for each position, the apparent slope of levels on the waterfall will depend on the width of the window you are moving along the impulse. Hope that helps.Moving the speaker and getting (repeatable) different results sounds like a room issue, to me. I say repeatable, because if you don't have your measurement parameters (swept sine duration, primarily) set up properly, then you're not getting a sufficient signal to noise ratio in your measurement. You know you got a sufficient SNR by verifying in either the Energy Decay Curve, Log Squared Impulse Response, or now in 3.2b6 (see the latest builds page ), the Envelope Time Curve. I would like to compare the reverberation times of my studio rooms to others, but i just dont know how.RT60 and the waterfall diagram dont correspond at all!!That's the curve proposed by Schroeder for measuring RT60, and the one referenced in the ISO spec for reverberation time. When you use the Reverberation Time plugin, you're simply seeing an octave (or third-octave, an option in 3.2) band decomposition of that impulse being curve-fitted to an ideal reverberation line.

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We strive to respond to all inquiries within 24 working hours when possible. This will help us better handle your request. If you're ever unsure about the software's operation, or need to get an answer about how things work under the hood (within reason!) we're here to help. We are not acoustics consultants and cannot help you treat or design your spaces, sorry! We anticipate introducing a number of new and exciting features to FuzzMeasure later in 2018. We extend a warm welcome to all FuzzMeasure users as they become part of the RODE family! Just send us a message in the form below or call us directly. Learn more Im finalizing the paper now and it has to be done very soon, so i hope i can get some help here.I would like to compare the reverberation times of my studio rooms to others, but i just dont know how.RT60 and the waterfall diagram dont correspond at all!! 3. How much of the the Impulse response should i mark.Markers of the Impuls were set at 0 and 340ms image3: This is the corresponding waterfall diagram.Note the Duration was set to 150 (im guessing ms). The Mode at around 65Hz is around 150ms on this graph, on the RT60 it is over 600ms. Also at 4kHz the reverberation is around 80ms, on the RT60 its over 500ms NikHe's a GS member. --EthanI'm not sure how much you are already familiar with, so I'll try to outline the fundamentals. The impulse response shows how an impulse emitted at the source position would be picked up at the measurement position. Some of that impulse is due to the direct sound from the source, altered by the characteristics and bandwidth of the speaker itself, some is due to the sound bouncing off the various surfaces of the room before arriving at the measurement position. The sound bouncing off the walls etc has to travel further to reach the mic, so it corresponds to later parts of the impulse.

Wrong questions! The FM manual is tight and clear but it does assume that the reader knows quite a bit already. I am somewhat trained and of a technical nature. I have no idea what a Minimum phase version is. Maybe Chris has done a definitions or such, I am stuck on V2, happily. If the manual were to explain that and many more such presumed understood terms it might get scary. Fuzz's forte is immediate ease of use and wonderful intuitive graphics. It will help make accurate decisions even with a light understanding. REW on the other hand has lots of controls and features. A bit like a helicopter in terms of driving it. Maybe I have become used to the Mac ways. John's manual is much much longer. Long enough that I haven't gotten through it. But I will. John does explain the terms in use by the program in a very clear, simple but thorough manner. Altogether a better source to understand the process and the terms in use. So that's the mechanical engineering and the theory of automotion. How to drive the car. I wrote a primer which is a sticky at the top of this forum, which is intended to give a new user a path to a reliable result. It may be time to update it to clarify some issues. e.g. when to drive one speaker, or two. The relative impossibility of getting useful decay figures due to the small rooms. The fortes of each program and so on. I tried throughout to make the point that one should not have inappropriate expectations. Wrong questions! Comparisons are the way to go, not absolutes. Most of us are in houses, not labs. Nix, some answers- Forget EDT and such, they are not really statistically defined in small rooms. You would have more success with REW but why bother. Waterfalls are incredible tools in both programs. Many speaker manufacturers publish frequency response curves with a mic say 1 Metre away, in an anechoic room or even outdoors high above ground. Small movements do change the curve due to imperfect polar response from the speaker, the mic.

In many cases, you're far better off with the waterfall plot to get an idea of the decay behavior in your room. That way you not only see the duration for reverberant peaks, but also the width of those peaks (which helps you determine whether treatment is required, as I understand it.) Note that the waterfall calculation in 3.2b6 (referenced above, with URL) has improved a little so you can offset the beginning of the waterfall period. Whatever you set up in the Waterfall plugin is independent of the frequency response window in the main display, however it operates within that data range. By that, I mean if you window only 100ms of data in the main display, you cannot accurately get 1000ms of waterfall duration in the waterfall plugin. Make sense? Markers of the Impuls were set at 0 and 340msRun an octave band decomposition on your impulse response, and look at the Energy Decay Curve for each band to see what they look like individually. It's possible your recording didn't have sufficient SNR for the measurement at those bands. Note the Duration was set to 150 (im guessing ms). The Mode at around 65Hz is around 150ms on this graph, on the RT60 it is over 600ms. Also at 4kHz the reverberation is around 80ms, on the RT60 its over 500msHope this helps! ChrisAfter the octave band decomposition i get a SNR of over 34db for all bands (most over 40db) except 32hz, which is 18db.Im not planning to use the RT60 for my paper, but instead the waterfall.I also dont really need a RT60 all frequencies put together, as i said before, but i have a few stats of other studios i wanted to compare to.I know its many questions, and they are really really hard to explain, but this is very important to me and im trying hard to be precise.Both Chris and John have done us all a great service in making this software. However, there are issues with the user. Many assume that the software will be easy to use and will deliver all sorts of answers, e.g. where to place traps, and how many. LOL.

You can get a rough idea of the RT60 figure from the impulse by looking at the earliest point at which all subsequent activity in the impulse is at least 60dB down from the peak.After the octave band decomposition i get a SNR of over 34db for all bands (most over 40db) except 32hz, which is 18db.Maybe 10s or 20s to start. Also, in version 2, it might be a good idea to window out just the decay portion (view the Energy Decay Curve to find this), create a copy of the initial measurement, crop it, and then try the reverberation time with that. In V3, the window is now used as part of the reverberation time operation, so this part of the application is much easier to use as a result. Thus, waterfalls just tend to work out better. In version 3.2, this will be a lot less troublesome, as there's now an additional offset parameter you can play with when generating the waterfall graph. I also dont really need a RT60 all frequencies put together, as i said before, but i have a few stats of other studios i wanted to compare to.They are definitely two different views. I should have only told you to look at the EDC, as it's the same schroeder curve used to calculate the T60 value (but just unfiltered). If you use the synchronous averages, then yes. I think it may either use a rectangular window in V2 (ugh), or something different but still sub-optimal. Waterfalls are tricky beasts to tame. That's the approach I took in V2 and V3—select something that should work for most cases. I'm learning that's not the ideal approach, and I'm trying to sort out some kind of more advanced method for allowing more user customization in the future. This is the Impulse response i got from the control room of the studio i did the measurements at. Is it normal, that the decay of the impulse is so strong. This is now a measurement of a livingroom.

In your case, if you are not close to the speaker, the room will intrude. If you are driving two speakers, small movements will have dramatic effects. Drive one speaker for Freg. Two for Waterfalls. DDIf the room continues to be pressurized, the mode will last indefinitely. You need a much bigger computer!;-)))) The rate at which a modal resonance decays is governed by the absorption of the surfaces along the mode's path at its frequency and the length of the path. That rate of decay can be expressed in many ways, but a 60dB decay time is as good as any in an acoustic context. On the more general point you do come across as someone who has discovered an electric screwdriver and is so in love with it thinks it is the tool for every job. Hammers still have their place.There is merely a series of early reflected energy. But then, trees can be clocked going 70 miles an hour with a radar gun.I'm well aware of the relevance or otherwise of RT60 measurements in small spaces, thanks, and have no need to re-read the material. I don't see any mention in my post of RT60 times in any space, large or small, perhaps you should read it again.What i guess i really dont understand is how i can have such a fast falling reverberation.I see many differences in impulse diagramm, such as early reflections and such.So when i use the waterfall diagram i have a few ringing frequencies down low, but they only ring for less than 300ms, the rest stays at around 100ms. i would expect a much longer reverb.Comparably short decay times up high are equally common. FrankI see many differences in impulse diagramm, such as early reflections and such.You need to look at the envelope of the impulse, which is what the ETC shows, and use some smoothing to see the overall trend in the response and not its instantaneous variations. The Schroeder reverse integration is a way of smoothing the plot to show the underlying trend in amplitude and from which the various reverberation times are usually determined.

When looking at a time signal, we are sometimes less interested in the variations of the signal itself than in what its overall envelope is doing - I've attached a plot from wikipedia that shows the difference between a signal and its envelope. It is produced by Schroeder backwards integration of the impulse, but it is important to start the integration at the point the impulse response drops into the noise floor, I'm not sure what methods FM2 uses to generate its EDC. To see where the noise floor of your impulse is, widen the vertical scale and find the level where the noise becomes essentially flat. I suspect, however, that you should easily find staff at the institution where you are doing your thesis who would be able to explain all this, and being professional educators they would likely do a much better job of it than we can Memorials, RIPs and Obituaries Grove Park, Maidenhead, Berkshire SL6 3LW.Hosted by Nimbus Hosting. Probably nobody uses them all so nobody can make such list. You can start poll.Maybe I should try this. (Next Post)There is a large number of tools found online and I'm hoping to separate the really useful and accurate ones from the rest of the crowd. To help organize it, if you would at least say which part of the process you use it for. (ie box, crossover, sim, test etc.) ThanksIf you are designing drivers, COMSOL is most precise, and MOTIV is fastest.I use REW for measuring. I used Visaton's BoxSim (only useful for Visaton drivers) for crossover. Now I would use my SpeakerSim for all speaker design, except measuring and transmission line simulation. When working on SpeakerSim I use other simulators to compare. Like Woofer Box and Circuit Designer, Baffle Diffraction and Boundary Simulator, Xsim, VituixCAD, BoxSim.So bumping this back into view to see if anybody has another to put on the list, anything to take off, comments on relative usefulness etc.Resources saved on this page: MySQL 15.00% vBulletin Optimisation provided by.

There is a hint in the way the question was phrased. I didn’t say time align (and it is not because I am afraid of copyright police). I say phase align because that is precisely what we will do. Simply put, you can’t time align a subwoofer to the mains. Whatever delay time you choose leaves you with a pair of unsettling realities: (a) you are only aligning the timing for a limited ( I repeat LIMITED) frequency range, and (b) you are only aligning the timing for a limited ( I repeat LIMITED) geographical range of the room. So the first thing we need to come to grips is with is the fact that our solution is by no means a global one. There are two decisions to make: what frequency range do we want to optimize for this limited partnership and at what location. What makes the most sense to you? 30 Hz (where the subs are soloists), 100 Hz (where the mains and subs share the load) of 300 Hz (where the mains are soloists). This should be obvious. It should be just as obvious that since we have a moving target in time, that there is not one timing that can fit for all. What time did the train cross the road. The answer spans 5 minutes, depending on whether you count the engine, the middle of the train, or the end. Such it is with the question: when does the subwoofer arrive? (and is also true for when does the main arrive?) How do we couple two time-stretched systems together. In this case it is pretty simple. We will couple the subwoofer train right behind the mains. The rear of the mains is 100 Hz and the front of the subs is the same. We will run the systems in series.You have to figure out who is first and then delay the leader to meet the late speaker. This will depend upon your speaker and mic placement. Nonetheless, it can be done. It’s just a pain. Like using the impulse response to get a nice simple answer directly in milleseconds, instead having to watch the fuzzy phase trace. It is absolutely true that the impulse response method is easier.

Now since there is not very much room treatment and quite a few bare walls and windows, i would expect the reflections to be a lot louder than they are here. What you're asking them to do is not nearly as practical as you seem to think it is. It sounds like you've been doing this for so long (and clearly you have) that you've forgotten that it's not exactly easy like falling off a log. FrankI think I have picked up on a fundamental confusion. Longer sweeps should work similarly. When I refer to averaging I am usually referring to a spatial average. i.e. I take a measurement at various places of interest in the room and 'average' them. I believe FM3 is very advanced in this regard. I believe it can generate a waterfall from an an average generate from multiple spots. This would take into account the different modal responses at the various spots. This is very much in keeping with ISO Building Acoustics measurements where they use multiple speaker and microphone, or even a mic on a large rotating boom. The overall idea is to get a response representing a broad area of the room, not just one spot. I could stretch things by suggesting that EDT and other Decay figures are averages of the decay slopes. Decay is not always linear, and can can be severely twisted by modes, alcoves, an open door to another room, a chimney and so on. I hope I am not insulting you by stating the obvious here. i.e. The word average has a quite a few meanings even within our sphere of discussion here. DDFuzzmeasure lets you average a measurement so it does lets say 8 measurements in a row and averages them then.You need to look at the envelope of the impulse, which is what the ETC shows, and use some smoothing to see the overall trend in the response and not its instantaneous variations. You can get a rough idea of the RT60 figure from the impulse by looking at the earliest point at which all subsequent activity in the impulse is at least 60dB down from the peak.

Fuzzmeasure lets you average a measurement so it does lets say 8 measurements in a row and averages them then.What is your goal here. May I suggest you take a look at the Room Analysis Primer Sticky. It is a simple methodology for getting an assessment of a rooms performance and more. DDWhat do the different types show. They all show some sort of decay of the signal over time (duhh) but they all show it differently.For instance, this EDC.We have two ears for a start. 10 cm this way or that can show large changes. Check it out. IMHO I don't think a single spot assessment has much value. Even Waterfalls and Decay measurements vary with location. I believe the ISO and ASTM have very good reasons their methods. My primer, for instance, although it needs some new work, is deliberately simple, but it does borrow from ISO practice. Regarding your myriad questions and your need to understand all of the technical background for Thesis purposes, may I recommend that you read through the ETF manual and tutorials at Acoustisoft, and as I said earlier, John's manual for REW. At a thesis level you will probably need to go to the original text books also. May I recommend that you redefine your Thesis goal or goals. With respect your questions suggest that you are at the beginning of a learning curve, not at a Thesis writing level. DDIf you wanted to measure reverberation, and see plots that look more like those you may find in acoustics textbooks, you would need a larger space and to make sure the mic is sufficiently far from the source - if you dig out ISO 3382 it explains this in depth, and you really should do that if you want to understand why your measurement attempts are not giving meaningful results. Regarding the types of plots: the impulse itself might be plotted on a linear scale, or on a log scale. Something you need to bear in mind is that your impulse response time signal will pass through zero as it varies between positive and negative.

In my next post I will explain why the easy way lacks sufficient accuracy for me to ever use with a client. The amplitude response answers the question: What would be the level over frequency if we put in a signal that was flat over frequency. This is not hard to get our heads around. If we actually put in a flat signal (pink noise) we would see the response directly in a single channel. If not, we can use two channels and see the same thing as a transfer function.Seen any excitation signals with a flat amplitude AND phase response. You won’t find that in your pink noise. Pink noise achieves its flat amplitude response only by averaging over time. Still the answer is clear: this is what the system under test will do to the phase response over frequency. A waveform with flat amplitude AND phase. That can’t be the pink noise described earlier, because pink noise has random phase. So what is it? A single cycle of every frequency, all beginning at the same time. Ready set, GO, and all frequencies make a single round tripand stop. They all start together, the highest freq finishes first, and the lowest finishes last. If you looked at this on an oscilloscope (amp vs time) you would see the waveform rise vertically from a flat horizontal line, go to its peak and then return back to where is started. The width of the line (in time) will relate to the HF limits of the system. The greater the HF extension, the thinner the impulse. As the HF range diminishes, the shortest round trip takes more time, and as a result the width of the impulse response thickens as the rise and fall reflect the longer timing. A system with a flat phase response has a single perfect rise and fall in its impulse response and a VERY important thing can be said about it: a single value of time can be attributed to it. The train arrives a 12:00 pm. All of it. We do not have to put in a perfect impulse.

We will use a second generation transfer function, the inverse Fourier transform (IFT), which is derived from the xfr frequency and phase responses. This is the answer to the hypothetical question: what would the amplitude vs time response be IF the system were excited by a perfect impulse. Any system that does NOT have a flat and amplitude and phase response will see its impulse response begin to be misshapen. Stretching and ringing, undershoot and overshoot will appear around the vertical peak. Once we are resigned to a non-flat phase response we must come to grips with the fact that a single time value can NOT describe the system. The system is stretched. The time is stretched. The impulse is stretched. We can easily see a high-point on the impulse response, even one that is highly stretched. And here is where we can really get into trouble: we can nudge the analyzer around to get a variety of answers to the same question (e.g. the same speaker) by deciding how we want to filter time and frequency: ALL OF WHICH ARE POTENTIALLY MISLEADING BECAUSE NO SINGLE TIME VALUE CAN DESCRIBE A STRETCHED FUNCTION. So this means something pretty important. The simplistic single number derived from an impulse response can not be used to describe ANY speaker known (to me) especially a subwoofer. This brings us back to the heart of the problem with our original mission: we are trying to link the low frequencies of the main speaker (100 Hz) to the high frequencies of the subwoofer (100 Hz). The peaks of these two respective impulse responses are in totally different worlds. They are both strongly prejudiced toward the HF ranges of their particular devices which means the readings are likely to be the timings of 10 kHz and 100 Hz respectively. Not so good. That’s it for the moment. Next I will describe some of the different ways that impulse responses can be manipulated to give different answers and when and where the impulse response can provide an accurate means of setting delays.

The FFT analyzer can only compute the frequency resp0nse in linear form. The quasi-log display we see is a patchwork of 8 or so linear computations put together into one (almost) seamless picture. Underlying this is the fact that the composite picture is made up of a sequence of DIFFERENT time record lengths. Bear in mind that the editing room floor of our FFT analyzer is littered with unused portions of frequency data. We have clipped and saved only about half the freq response data from any of the individual time records. It is a 2nd generation product of the linear math. The inverse fourier transform (IFT) cannot be derived from the disected and combined slices we use for the freq response.It has to come from a single time event. The importance is this: linear data favors the HF response. If you have 1000 data points, 500 of them are the top octave, 250 the next one down and so on. If you have a leading tweeter, The IR will find it ahead of the pack (in time and level). The mids and lows will appear as lumpy foothills behind (to the right) of the Matterhorn peak. If you have a lagging tweeter, the IR will show the lumpy foothills ahead of the peak (to the left), but the peak will still be the highest point. If we find the arrival time for both we can lock them together. This is a PARALLEL operation. 10kHz is linked to 10 kHz and 1k to 1k and 100 to 100 for as long as they share their range. If the speakers are compatible, one size fits all and the limitations of the IR are even on both sides of the equation.This is also true of aligning a woofer and tweeter in a two-way box. This problem holds for ANY sprectral crossover tuning. Linear frequency math does not have a and fair and balanced perspective over frequency. If you are looking at devices with different ranges they are subject to this distortion. The location of the peak found in our IR is subject to the linear focus. All other freq ranges with appear RELATIVE (leading or lagging) to this range.

If you have a speaker that is similar to 100% of the speakers I have measured in the last 26 years, then one thing is certain: the response at 100 Hz is SUBSTANTIALLY behind the response we just found at 8 kHz. If you have a subwoofer that is similar to 100% of the speakers I have measured in the last 26 years, then one thing is certain: the response at 30 Hz is SUBSTANTIALLY behind the response we just found at its upper region. But it does NOT work that way. You are making a series connection at a specific freq range, not a parellel connection (where bulk might apply).All of them haqve an effect on the shape of the response, how high the peak goes, and where (in time) the peak is found. Without going hard into the math we can look at the most decisive parameters. In the end we have a span of time included in the computation. We could, however choose to display less than the full amount of data we have. The visual may be a cropped version of the computation, or it could be the full length. The capture time also limits how low we can go in frequency. We can’t see 30 Hz if we only have 10ms of data. Most IR response have the option of large amounts of time, so getting low frequencies included will not be a big issue. The finer the slices, the more detail we will see. If we have slice it into.02 ms increments (50 kHz sample rate) we can see up to 25 kHz. If we slice at lower sample rates, the frequency range goes down. This is important. The speaker did not change, but our conclusions about it did. But if we have one speaker with a full HF range and one without the playing field just got tilted. This means that things that go negative will peak downward while positive movement goes upward. Polarity (and its inversion) can be seen. The down side of this is that the linear vertical scaling translating vewry poorly visually toward seeing the details of the IR such as late arrivals, reflections, etc. Worse yet is trying to discern level differences in linear.

The Y axis does not read in dB. It reads in a ratio and this has to be converted. Upward peaks have a positve value and downward have a negative value. Where it strange is when you try to compute positive direct sound to a negative going reflection. But the downside is that there isn’t a downside. Pun intended. What I mean is that the negative side of the impulse is folded over with the positive and these are combined into a single log value. This can now be displayed in dB since everything is going one way. This has various names: Energy-time-curve (ETC) amoung others.If you are going to use the impulse response alone (I say you because it will not be me) to align speakers in different freq ranges you are prone to computational items that will affect the HF and LF sides of the equation differrently. One technique I have seen advocated is the push down the sample freq so low that the upper regions of the HF speaker are filtered out. The idea is this: if the Xover is 100 Hz, then drop the resolution of the analyzer down to filter out the region above 100 in the HF speaker.How important is it to set the delay for your reference signal within your FFT when you’re doing this. Can you just leave it alone so that it’s a common reference against measuring the tops and subs independently, or should you constantly adjust the delay on your reference signal as you adjust the delay on your tops or subs to phase align to two sources. OR do you figure out which sound source is arriving late and set your FFT delay based on that and add delay to the other source until the phase is aligned? The time records are so long down in the subwoofer range that it takes a LOT of error in the propagation time before the data is degraded significantly. In any case there is ONE very critical aspect: whatever internal delay you use, it must remain unchanged through the process.