Room frequency response measurements

Does anyone know of a Shareware program available that use your PC's sound card to play pink noise while recording from the microphone, and then display the room's frequency response?
I have a high-end stereo system and am looking for something that will show me what my room modes are doing.
Thanks in advance.

thanks but I'm not looking for pink noise to drown out others. It's to use as a measurement. see my post again.
1.33ghz PowerBook g4, 17-inch Mac OS X (10.4.7) 1.5gb ram

Similar Messages

Frequency response measurements with pxi-5922

I’ am using signal express and the pxi-5922 digitizer together with the AWG pxi-5441 to analyse the frequency response of a buffer amplifier. See the attached signal express file. Many different ways to measure the frequency response have been tried and this is the best I came up with. It is basically two tone extract steps in a sweep loop. But I’ am still uncertain if this is the best way to do this kind of measurement. The fact that the detected frequency differs between the two channels worries me, even when the two channels of the pxi-5922 are looped. Is there a more accurate way to determine the frequency response?
Best wishes
Stefan Johansson, SP
Attachments:
sweep.JPG ‏397 KB
Frequency Sweep funkar.seproj ‏81 KB

Claudia-
Thanks for the response.
Regarding the CJC- When I switch it on, the temperature readings I get are very random, roughly negative 1 degrees. (I am operating right now at room temperature, and will be using J-type TC's to measure ~43 degrees C). Also, when I use the built-in CJC, the aquisition rate seems to slow down considerably. When I use the "user specified" everything seems to be ok, including the aquisition rate.
I measured the resitance of the Thermistor on the TBX-68T and it was about 5000 Ohms, as expected.
Just to make sure: When using the TBX-68T, do I need to hard-wire a thermocouple to Channel 1/auto-zero and another to channel 0/CJC? Because I connected a TC to channel 0 right now, but I wasn't 100%
sure.
I've attached my main vi and two sub vi's that I am using for the voltage aquisition part of my project. (Note:the current measurements are just voltage measurements multiplied by the recipricol of the resistance it was measured across, ie. 10).
I would like to keep this file as is because it writes to a file exactly the way I want it to. I'd like to have the temperature aquisition with the 4351 in the same vi as the 6030E so that they both stop and start at the same time. I am just not sure how and where to log the temperature data since there will be fewer data points than the voltage data. Any suggestions? Should I write two separate files? can I somehow append them?
Thanks again. Hope to here from you soon.
Attachments:
EBlackMainDAQ.vi ‏107 KB
Save_Data8.vi ‏45 KB
Build_String_Array5.vi ‏33 KB

Reflected power frequency response measurement

My goal is to measure the reflected power from a DUT. I am using a Transmission/Reflection test set from Agilent to sample the reflected power and pass it to the PXI-5660. I sweep the signal over a range of 100 MHz to 500 Mhz and measure the reflected power signal at each point. A calibration sweep is taken first with a short at DUT end of the cable. I also sample the incident power with the T/R test set. The output signal is typically 0 dBm. My question is: What is the formula to calculate the reflected power from the signal measured? The signal I see on my application is not what it should look like.

Hello Brian!
Your best bet at finding that formula is probably Google. I did a search and got many different relection formulas and how to derive them.
Here is a derivation...
http://www.physics.cornell.edu/sethna/teaching/sss/pythag/pythag04.htm
Here is a discussion and definition..
http://www.eagle-1st.com/notes/RPM/body.htm
If neither of these are helpful if there something specifically with the 5660 I can help you with. Let me know and have a great day!
Allan S.
National Instruments

Flat Frequency Response

I probably shouldn't have to be asking this question since I charge people for my obviously amateurish recording abilities but it's one that I've never had explained to me and one I need to know the answer to......
Let me set the question up this way:
When you get in the car and pop in a professionly produced cd, most people crank the treble control up to 6-10 (on a scale of 10) and the bass up to (4-10) depending on the factory speakers and type of material and whether or not they care what their music sounds like. When you get home and you're listening to the much higher end home stereo still listening to that professionaly produced album, you still reach for those treble and bass knobs and crank them up several notches or if you have a graphic eq you tweak out a smiley face .
There's so much emphasis in the recording world about getting a flat frequency response out of your room with absorbers and bass traps and spreading around the reflections with deflectors, etc...etc..etc..., that we spend thousands of dollars on this stuff and some measuring software to make sure that it's flat. Then we use that flat response to produce music that sounds great and expect that to translate to those cd players and home stereo systems.
(I'll additonally preface my question by saying that I've had no problems getting my music to translate from my home studio to any other playback system, but I'm a little confused about what's going on.)
Now finally my question(s), when we reach for those treble and bass knobs on our car and home playback systems, are we really just trying to make up for the lack of bass and top end in those systems so that we too can achieve a flat frequency response and make the music sound good on whatever system? or
Do we as listeners actually prefer the smiley face frequency response in music and are we taking a cd that itself has a flat frequncy response and making a smiley face out of it so that it sounds good to our ears? (Please don't give me a material/genre answer.)
The reason I ask is because I have to put a graphic eq on my Truth 2031A monitors to make the professional stuff sound good through them, and then I in turn mix my music to sound the same for whatever material/genre of course. (I'm not really interested in any monitor bashers or I would've asked this over at Gearslutz.)
So again rephrased...Do the masses think music sounds good when it has a flat frequency response or the smiley face and if it's the lattter of those, how are we supposed to achieve that when our home studio setup is producing a flat frequency response, do we tune our monitors with eqs like me?
Additonaly I understand that when were talking about flat frequency responses in rooms we're talking about throwing sine waves through a system and ranging their frequencies, measuring them out so that we can detect any over emphasis/deficiencies in the room so maybe this question is a little more towards monitor tuning.

If you look up Fletcher and Munson in Google, you might begin to get a bit of the start of an idea of why this isn't quite as straightforward as it seems.... and I'm not sure that I can give you a complete answer either, although I can give you a few connected but slightly random things to ponder, wearing my acoustician's hat:
The fundamental problem is that when things are quieter (and less distorted, incidentally) our ears get more sensitive to the midrange frequencies, and if we listen to music that way, it invariably sounds as though the bass and treble are out of balance. In cars it's slightly different though; the frequency response of whatever's in there generally tends to be anything but flat - and often over-emphasises the midrange anyway. Treble tends to get absorbed very easily in upholstered cars, and since most car speakers don't have anything like acceptable tweeters in as a rule, it's not surprising that people want to increase the treble. As for the bass - well I'm always turning that down personally, but I know what you mean in principle!
If by a 'commercial' CD you mean one where the vocal is prominent, then yes I can easily imagine why you might as a matter of course want to increase the response at the extremes - it makes sense if you think about it. The mid-range vocal is prominent and probably compressed, so its average level is louder than the backing - this helps it to stand out. But also it distorts the overall time-based response - the backing may well be balanced so it's okay on its own, but that doesn't always translate if you have the wrong vocal settings applied, or at a minimum, applied unsympathetically. And some voices make this significantly worse; for instance Sealion Dying (AKA Celine Dion) makes the most appalling racket in the midrange, and you'd definitely need less of that!
So really I'd say that it's not a Bass/Treble issue, but a midrange one. If you look at commercial CDs in general, you tend to find that the energy distribution is pretty even over the whole audio band, which implies that it falls off at 6dB/octave if you look at it in Audition (this is an energy/Hz thing), but in reality most CDs these days are mixed a little brighter than that - more like a -3dB/octave slope down from about 1kHz, and that's partly to compensate for a lot of things - some of which are cars... You do have to watch out for the distortion issue though - most people don't realise, but you are able to tolerate rather higher levels of non-distorted sound than anything with significant distortion levels in it, and if that distortion is in the midrange, then you'll want it quieter anyway. So decent, over-rated under-run PA systems always sound cleaner and louder but you should beware - they can damage your ears just as much, if not more.
Do car interiors themselves increase the chances of midrange boost occurring? I think it's a pretty safe bet that they do, as a rule, simply because of the size of them, and the treble problem I already mentioned. And if you are trying to compensate for too much midrange, then the rest follows. Most domestic replay systems these days seem to be midrange heavy to me as well - I haven't heard anything cheap recently that had anything like a flattish response - and they really don't suit the rooms they are in either.
If you want to listen to material as it really should be, then you need to experience it live first, I'd say, and then do a direct comparison with what you can hear in your monitors. I'm fortunate - I can do this quite regularly with a variety of material. Do I tend to leave things as flat as they are recorded? Well, it depends on what it is. If it's in any way classical, then sometimes I look carefully at the bass balance, but generally I leave the rest alone. Everything else these days I just get to sound good - and that can mean all sorts of tweaks, depending on all sorts of things. More and more though, I've come to the conclusion that too much midrange isn't necessarily a good thing - but that's mainly because of the general lack of good reproduction equipment around these days.
As for monitors and flatness - well that's not really an issue for most people, compared to getting their listening environment correct. If you have a pair of cheap monitors in a good room, the chances are that the results will sound better than an expensive pair in an uncorrected room - despite what all the monitor freaks on gearslutz might say. These days, even the cheaper ones can sound quite respectable. But flatness isn't an issue with monitors really - a decent impulse response, and low distortion are far more important. Chances are that if a manufacturer has got this right, the monitor is going to be suitably 'revealing' anyway - which is what a monitor is supposed to do.
The one thing you do not do though, is EQ the feed to your monitors - that went out of fashion almost as soon as it came in - fortunately. You fix the room so that it's more truthful. If you EQ the monitor feed it will inevitably only sound good in one place in the room, and that's no use to man nor beast. The only decent things that proper room correction systems can do is equalise the immediate time response to take account of what's actually between the monitors and you - which if done properly can improve stereo imaging no end.
So answers? Well not really. But at least I've explained a few (but not all) of the issues.

Loudspeaker Frequency Response

Hi all! I'm making a program to measure the frequency response of a loudspeaker. I'll be using a sweep to test the loudspeaker.
Particularly i would like the program to be like this, which is real time. How can i implement this in LabVIEW. This is the screenshot
Details about the plot.
First graph:
Shows a plot in real-time of a frequency sweep with a constant sine sweep amplitude of 1 V. When sweep is started, the graph shows a plot of FFT moving from left to right, with peak of FFT at maximum amplitude of 1 at corresponding frequency of the sweep.
Second graph:
Shows the plot of the Sound Pressure Level in dB versus freqeuncy.
Please refer to the picture and video link below.
https://www.youtube.com/watch?v=sKC3ioWXG38, skip to 4:10

Assuming your idea is to sweep a frequency into an amplifier connected to the loudspeaker, and measure the frequency response with a microphone:
You need to monitor the signal at the loudspeaker terminals to account for any non-linearity's in the signal generator and amplifier.
You need to know the frequency response of the microphone, this is difficult, and that is why calibrated microphones are expensive.
You need an anechoic chamber so that the results are not affected by any room resonances.
Your sound level plot is in dB (A). My understanding of the A weighting is so that the human perceived loudness is constant across the audio frequency range. If you are concerned about loudspeaker performance, is it worth discarding the complexity of this additional frequency response curve?
This will be a difficult project. Please let us know how you get on.

Frequency response requirements for headphones with CMSS on XFi ???

Hi,
I would like to know if someone could tell me what kind of heaphones are suitable for the CMSS mode with the XFi.
I mean between : flat response/free-field correction/diffuse-field correction.
Applying HRTF filtering should mean that headphones with flat response is the best option ( same configuration as binaural recordings).
But I have a big doubt that Creative team expects costumers to possess such a pair of headphones, as it is rather for scientific uses (psychoacoustics, audiology etc...).
So, if we look at the technical solutions for wide audience we have two options (FF correction and DF correction). Here is a trick because these corrections intend to reproduce some of the effects from HRTF (for two different environment configuration of HRTF measurements). It is why the frequency response of most of the headphones have a notch between 4Hz and 0 kHz.
To simplify, if we listen binaural sounds with classical headphones the effect of outer pinna is reproduced twice.
So I guess Creative have implemented a kind of normalization/equalization/correction process to deal with the non-flat frequency response of headphones, but do someone know if they have chosen diffuse field or free field correction ?
This post might seem a detail but the issue can be very important for the accurate localisation and the coloration? of 3D sounds with headphones.
Thank you, and please forgive my english!

The only possibility that I can think of is that 2/2. mode is NOT as simple as headphone mode with crosstalk cancelation. Perhaps the HRTF only kicks in for sound sources outside of the arc directly in front of the listener. If that were the case, you wouldn't percei've any distortion for sound sources in front of you.
Also, you are wrong regarding DirectSound3D. Keep in mind that Direct3D and DirectSound3D are not the same. The whole point of OpenAL and DirectSound3D is that they present an API to the programmer through which there is NO specification of the number of speakers. When using OpenAL or DirectSound3D, the only thing a programmer can do is specify the location of a mono sound source in 3D space relati've to the listener. The speaker settings for your DirectSound3D or OpenAL device will then determine how this sound is "rendered" by the soundcard. It is not under control of the game. For example, if you have 5. speakers and the 3D position is behind you, the SOUND CARD will make the decision to use the rear speakers. If you use headphones, the SOUND CARD will decide to apply an HRTF to create the illusion of a rear sound source. The point is that the game does not have control over how many speakers you will get sound from.
However, to further complicate the situation, there are SOME games (HL2 is an example) where DirectSound3D is used, BUT the sound output of the game itself IS a function of the Windows speaker settings. This is not how programmers are SUPPOSED to use DirectSound3D. I've written about this countless times. There is a good post on [H]ard|Forum about this. Do an "advanced" search with my username (thomase) looking for the terms "hl2" and "cmss".

Frequency Response Function & FFT & Inverse FFT (problem of unit Volts-RMS)

Hello everyone,
I am currently working on a VI in order to compare two analog signals : the first one corresponds to the output signal (my reference) which is sent by my data acquisition card to a shaker and the second one corresponds to the input signal recorded by an accelerometer fixed on the same shaker. The final goal of the VI is to correct the analog output signal by using the analog input recorded signal in order to have the vibrations on the shaker which corresponds to what we really want.
To summary, I have a problem of unit with the Volts-RMS...
So this is my method for the VI :
First, I have to calculate the Frequency Response Function between the two analog signals (output and input). For it, I use the " Frequency Response Function (Real-Im).vi " which returns the complex values of the FRF in Volts-RMS (but I don't want to use this unit).
Then, I want to calculate the FFT of the analog output signal (my reference). There are two different blocs which can be used : " FFT Spectrum (Real-Im).vi " and " FFT.vi ".
The " FFT Spectrum (Real-Im).vi " returns the FFT complex values of the signal in Volts-RMS and the " FFT.vi " returns the FFT complex values in Volts (or say me if I am wrong, thank you). I really would like to use the second one because of the unit.
Then, I divide the FFT just calculated with the Frequency Response Function calculated just before.
For the end, I calculate the inverse FFT of that with the " Inverse FFT.vi " which use the complex values with the same unit than for the " FFT.vi ".
I don't want to use the Volts-RMS unit because I absolutly want to use the blocs " FFT.vi " and " Inverse FFT.vi ".
The problem is that I don't find a bloc which use the same unit for the Frequency Response Function. The " Frequency Response Function (Real-Im).vi " returns only the complex values in Volts-RMS unit. Maybe it is possible to convert it correctly? Or maybe there is an other bloc which can be used in order to calculate the Frequency Response Function with the same init than for the FFT and Inverse FFT ? Because I can't mix everything for the moment...
Thank you for your help,
Best regards,
Sebastien

Hello Preston,
No, I have not use the Sound and Vibration toolkit. I have only used the signal processing toolkit with the two toolboxes " Waveform measurement " and " Transforms ".
But I think that what I have done for the moment in my VI is correct (I have finished the complete VI). But I am not sure of the units (Volts, Volts-RMS...) and I would like to understand.
I have tried with the Sound and Vibration toolkit for the frequency response function (because you say me that it deals with all the unit conversion) and I can obtain the same results than with the " Frequency Response Function.vi " of the toolbox " Waveform measurement ".
But I would like to understand the units (see my previous post please). For example, for the FFT (the result is a complex), why sometimes it is in Volts, sometimes it is in Volts-RMS ? Is it possible to convert it ? How ?
If you want, I can attach on the forum my VI and that will maybe help you to explain me. Maybe it will help other people interested.
And if someone else can give me other precisions or advices about it, do not hesitate.
Thank you for your help,
Sebastien

Frequency Response VS FFT for measring frequency response of a audio ouput signal.

We have purchased the Sound and Vibration Toolkit and I have some questions.
From the frequency response example I looked at you measure the input then the ouput and the Vi gives you the difference.
I want to measure the audio frequency response a radio.
So all I have is the output I dont have the audio input to use as a reference.
How would I measure frequency response with the frequency rasponse VI with only the output audio signal?
I was also looking at the FFT example to measure the audio signal frequency response.
But from what I can tell so far this only does 1 channel I want to do both channles at the same time.
Is there a way to do FFT on 2 channels at once and have them output on the same graph?
Thanks for any help you can provide.

Hi,
I was looking through the examples and If you look at the SVXMPL_Multichannel FFT (simulated) examples, this shows how to take an FFT with multiple signals. You just pass in an array of the waveforms. If you need to put your signals together you can use the merge signals VI, which is located in the Express»SignalManipulation Pallette. Let me know if you need any further help with this issue.
Have a great day,
Michael D
Applications Engineering
National Instruments

Buil-in microphone frequency response graph

I'm curious as to where I can find a frequency response graph for the built-in mic in the 15" MBP 2.8. I know it's a silly thing to want, but... Sometimes I don't have the time or resources to set-up an external mic with a flat frequency response, and would just like to use the built-in mic very quickly to "EQ" or "ring out" rooms. I can't find the info on the built-in mic anywhere. Any suggestions?

Have you ever used the microphone? Have you ever actually used a laptop to do this?
I don't have a graph, but from years of working with audio I can tell you that it doesn't have the frequency response, sensitivity and directionality to do what you want.
A nice external mic and portable interface like the iMac would not only do a better job but would look a lot more professional than running around trying to point the cheap microphone of a laptop in the right direction.

FS7 audio Frequency Response 50Hz - 20KHz?

Hello guys, I'd like to point out an issue that I couldn't find on the forum regarding the audio frequency response.I've always worked with pro camcorders that recorded 20Hz to 20KHz, which is the standard frequency us humans can hear. So I was surprised when I saw in the FS7 manual that the audio Frequency Response is 50Hz to 20KHz. Is this a typo? Or is the camera actually limited to 50Hz for the lower freqencies?I would assume that the internal mic would be limited to that range, yes, but not the internal recording of the sound? Why limit the camera to 50Hz? Any light on the subject would be appreciated. Thanks!

The tests I posted are for the mic inputs, not line inputs. I did test both, but since I often run & gun and cannot support all the extra gear to make the FS7 mic inputs on a par with Sony's EX3 mic inputs, I'm requesting that the high pass filter added with firmware 2.0 be removed in futre updates. We have the wind filter already. No additional filter is needed. While Genelec monitors are pretty good, they're not noted for deep bass response. Most studios use the ubiquitous NS10, which are only good to 80hz, which is why few people notic the deficiency. The custom built system in my screening room has a -3dB point of 7hz. I noticed that even piano sounds thin on this camera. that doesn't surprise me, since the rolloff starts at 200hz. All I'm asking for is the same quality I'm used to with my EX3's mic inputs. Only 0.5dB down @ 20hz and no thinning of the midrange. When I do large budget orchestra shoots (rare) I use 24/96 sampling, 8 channels and large diaphragm studio condensers. My recordings earned critical praise from Peter Aczel at The Audio Critic. I was a sound engineer for 4 decades before I got my first digital camera. In addition to that, I was consulted as an independent expert on infrasonics for chapter 17 of Ethan Winer's THE AUDIO EXPERT, published by Focal Press. I was responsible for the information on subwoofers and how they operate. Acoustic and speaker design share a long history with me, and my 'day job' is amplifier design, modification and repair. I'm too intimately aware of what's going on in the signal path of an audio system and when modern digital systems deviate intentionally from DAT-quality, it really bothers me. I made a big deal about this in 2007 with the Sony HVR-V1U, and Sony must have listened, because they got it right with the EX1 and EX3 cameras. But sadly, they've reverted to these games with the FS7, though not as badly as 2007.

Frequency response at low frequency

I'm working on a bandpass filter, and I'd like to get the frequency response showing that the frequencies outside the lower and higher cutoff frequencies are being cut off. However, the correct plot is shown only when my cutoff frequencies are high (roughly from 1000-8000 hertz). When I use low cutoff frequences(roughly 4-5 hertz), the plot is incorrect. So how can I get the frequency response to my low cutoff frequencies? Thanks.
P.S. In the code, some parts are irrelevent. In the front panel, the only relevent part is the frequency response plot at the lower right corner, and the specs above it; in the block diagram, only the upper half(with IIR and FRF) is relevent. Thanks.
Attachments:
BME_Pressure_Sensor_V1.00.vi ‏591 KB

Hi Manson
There is a bug in your diagram since you connected the number of samples where you should have connected the sampling frequency.
The sampling frequency is related to the pace at which you take the measurement.
Usually, Fs = 1 / dT
where Fs is the sampling frequency and dT is the time interval.
It should work better.
In any case, to have a better resolution in the low frequency range of your spectrum computations, you have to increase the number of points of your data because there exist the following relationship between dF (space between 2 points in you spectrum), dT, and N (number of data points) :
dF = 1 / (2 x dT x N)
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How to equalize an analog output by a known frequency response?

I'd like the analog output of my system have a flat response. What I'm going to do is first measure the stimulus signal, then using a filter to compensate the frequency response to make the following output signal flat. The difficulty is how to build a filter according to the frequency response. I know it's easy to do by using digital filter design of signal processing toolkit. But I need to do it by LabVIEW and the response is changed frequently. Any suggestions?
Bill

The filter does the signal "adjusting." Filters are typically characterized in the frequency domain, but the work on the signal fed them, which usually occurs sequentially in time. The lookup table is just to select the appropriate filter so you do not have to do a lot of calculations at run time. As an example, suppose you have just treble and bass and only one cut and one boost setting for each. You measure the stimulus and find the bass is too high and the treble too low. You select the bass cut and the treble boost filters for this run. If this is expanded to octave (or third octave) filters and 16 gain/attenuation settings in each band the lookup table approach saves time and may also provide a compact means of recording the equalization settings with your test data (rather than filter coefficients which do not actually indicate the response without characterizing the filter).
Lynn

Frequency response of real analog filter using mydaq

Hello!
I am trying to find the frequency response of a analog filter using the NI myDAQ. There are several examples of this in Labview, but they are usually analyzing digital filters, not real world filters. There is one example, see attached vi, that simulates measuring a realworld signal using a simulated frequency generator and a simulated multimeter. I believe that this could be done for real using the mydaq.
Can someone please help me convert this VI to run off of the mydaq?
Thanks!
Attachments:
Frequency Response.vi ‏21 KB

Thank you Kyle, this is exactly what I was looking for. I searched the examples that came with labview 2010, the signal processing module, the 8.6 DSP module and the signal anaysis toolkit, but I had not yet installed the elvis cd so I didn't run across these VIs. Thank you once again!

TS-11 Frequency Response At 1/8" Jack

Hi,
I tried to play back sound from my new Pavilion TS-11-e100sr using 1/8" jack. The sound was audibly equalized for laptop speakers or maybe earphones and was terrible when it played back through outer sound system. I have measured output voltage at 1/8" jack playing back generated sine waveforms of constant level. The result is shown at the picture attached. As you see, there is an enormous 6dB boost down 100Hz and -3dB cut at mid range around 300-400Hz. At high frequencies there is also progressive 3dB boost up to 16kHz. I tried to find where I can switch off this EQ but I couldn't. Can you please instruct me what should I do to obtain flat frequency response at the output?
This question was solved.
View Solution.

I myself found a solution. I've replaced two Realtec drivers with Microsoft and this gives flat response close to AC'97 specs.

Frequency response formula

Hello!
I have done the following experiment to find out the frequency response of a "box":
I have introduced a sound speaker inside a box emitting pink noise-->I have measured the noise using the "sound and vibration assistant" and a microphone( the microphone is also inside the box)
Then I have put the microphone outside the box and I have emmited pink noise again (the sound speaker is inside the box)
With these datas I want to calculate the frequency response of my box so I am using the "frequency response step" of the sound and vibration assistant software.
My question is: The unit of the amplitude graph of is "dB ref 1E+0" Could someone tell me which is the formula to calculate the frequency response of my box and obtain the units of "dB ref 1E+0"???
Thank you!

Hi dSierra,
Sorry but I do not understand when you say : "If you want to reconvert to natural units, you just have to invert the equation 20 log (FreqResponse/1 unit (stimulus/response))". Could you explain a little bit more please
Thank you!
Best regards

Room frequency response measurements

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