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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http://www.machinevision.ch
http://visionindustrielle.ch
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Similar Messages

Make frequency response analyser using frequency generator and counter

Hello
Can we make a frequency response analyser using a Frequency generator and frequency counter?
How to add modulation with it? Modulation frequency is to be varied as per the input to given to the carrier!
The outputs are Frequency, magnitude, and the phase as like solartron FRA.
somebody have an Idea for this
awaiting for the solution
thank you
"Thanks with regards "
by
..........Gireesh..........

Hello Gireesh,
You can use a function generator to generate frequencies and use the modulation tollkit and other tools availabe with LabVIEW to do the modulation part . Or you can use the analog output ports on the daq card to generate different frequency signals for the same purpose .This should pretty much serve your purpose.
http://zone.ni.com/devzone/cda/epd/p/id/5646
http://digital.ni.com/manuals.nsf/websearch/AF3615F31CE9656C862576070020B8F7

MacBook core duo Sept. 2006 - the audio is a mystery. Any analogue out has bass boost with bass distortion. With digital out, by USB or Airport Express Airtunes, the frequency response is normal. Somewhere, Apple put in an analogue bass boost.. why?!

Since new, my Macbook core duo has sent all audio to all analogue outputs with frequency distortion. Some physical hardware or firmware in the analogue section adds a bass boost to the frequency response of the audio.... And, this boost adds bass frequency distortion to most, if not all, of the audio analogue output.
This happens for all audio sources, players, iTunes, videos, streaming audio/video, movies... any sound source available to the Macbook. No-one with whom I have talked about this problem, has ever heard of it. Not elegant ideas for reducing it are to use equalisers on players and iTunes. iTunes even has a preprogrammed "bass reduce" on its equaliser, as if it knows already that this inherent bass boost "feature" has been built into the Macbook, and cannot be defeated by the user.
The digital audio is of sufficiently high quality to play well on a very good to excellent sound-system; so, it's a shame someone has mucked around with the frequency response of the analogue conversion, by designing the frequency distortion right into the computer. This motherboard, (which includes the sound-section), has been replaced, along with the speakers, and everything sounds exactly the same as it was before.
To receive a flat frequency response and no frequency distortion, I listen on one receiver using Airtunes, (it doesn't matter whether or not the analogue or digital output jack is wired analogue or connected via optical cable.... the freq. response is flat); and I listen on a high-end stereo system using an USB output port to an A/D converter, passing the analogue result using long patch-cords connected to the "line-in" jacks of the stereo. The bummer is that most of my audio sources are not derived from iTunes... hence, I cannot use Airtunes with Airport express for them. I've tried using "Airfoil" without luck. The sound becomes distorted with sibilance and other frequency anomalies.
Question: has anyone else discovered this sound imperfection in any Mac product? And, does anyone know if Apple is aware of it? Finally, has anyone found an Apple fix for the problem; or, at least, come up with better solutions than I have?
Thanks heaps for any impute and answers you may supply!! junadowns

Army wrote:
This might not help you a lot, but if you want a stable system, try using packages from the repo wherever possible, look at the news before you update your system and don't mess things up (like bad configuration etc.).
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[repo-ck]
Server = http://repo-ck.com/$arch
(this has to go to the bottom of /etc/pacman.conf)
(use that one which is best for your cpu (in your case this might be the package linux-ck-corex).
Hmmm, Linux-ck-corex doesn't even load.. I am now trying to install the generic one. Hope it works.
Edit: I will first try linux-lqx...
Last edited by exapplegeek (2012-06-26 18:33:31)

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

Frequency response function modal analysis

After reviewing the signal analysis functions in DIAdem I have realized them to be a bit limited for modal analysis. I have a couple hammer impact tests that I need to process a frequency response function for, and since this is brand new to me I'm not seeing anything in the embedded function list that is going to help me. I was wondering if anyone out there has a couple of pointers on generating a FRF plot for modal hammer impact tests. I did notice that the ChnFFT2 command allows me to generate a transfer function, coherance, and FFT Cross Spectrum channels for analysis. Though I might be confused and this may be everything I need. My FFT2 settings are below.
[code]
FFTIndexChn      = 0
FFTIntervUser    = "NumberStartOverl"
FFTIntervPara(1) = 1
FFTIntervPara(2) = 2500
FFTIntervPara(3) = 1
FFTIntervOverl   = 0
FFTNoV           = 0
FFTWndFct        = "Rectangle"
FFTWndPara       = 10
FFTWndChn        = "[1]/Time axis"
FFTWndCorrectTyp = "No"
FFTAverageType   = "No"
FFTAmplFirst     = "Amplitude"
FFTAmpl          = 1
FFTAmplType      = "PSD"
FFTCrossSpectr   = 1
FFTCoherence     = 1
FFTTransFctType = "Spectrum H0"
FFTCrossPhase    = 0
FFTTransPhase    = 0
Call ChnFFT2("[1]/Time axis","'[1]/H_1' - '[1]/H_4'","'[1]/A_1' - '[1]/A_4'") '... XW,ChnNoStr1,ChnNoStr2
[/code]

Standard modal analysis has something denoted as FRF. I have a labview application note "The Fundamentals of FFT-Based Signal Analysis..."
Frequency Response Function
The frequency response function (FRF) gives the gain and phase versus frequency of a network and is typically computed as
where A is the stimulus signal and B is the response signal.
The frequency response function is in two-sided complex form. To convert to the frequency response gain (magnitude) and the frequency response phase, use the Rectangular-To-Polar conversion function. To convert to single-sided form, simply discard the second half of the array.
You may want to take several frequency response function readings and then average them. To do so, average the cross power spectrum, SAB(f), by summing it in the complex form then dividing by the number of averages, before converting it to magnitude and phase, and so forth. The power spectrum, SAA(f), is already in real form and is averaged normally.
Refer to the Frequency Response and Network Analysis topic in the LabVIEW Help (linked below) for the most updated information about the frequency response function.
http://zone.ni.com/devzone/cda/tut/p/id/4278
So the options for FFT2 are
No
DIAdem does not calculate a transfer frequency response.
Spectrum H0
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Spectrum H1
DIAdem specifies the cross spectrum and the auto spectrum for each signal pair. DIAdem calculates the transfer frequency response by dividing the averaged spectra: Middle(cross(A,E))/middle(auto(E)). DIAdem does not average phases, because phases can delete each other.
Spectrum H2
DIAdem specifies the cross spectrum and the auto spectrum for each signal pair. DIAdem calculates the transfer frequency response by dividing the averaged spectra: Middle(auto(A))/middle(cross(E,A))
If you assign the values Spectrum H1 or Spectrum H2 to the variable FFTTransFctType, DIAdem averages and divides the cross spectra and the auto spectra and calculates the amplitudes last.
Which state auto spectrum when FRF is power spectrum.

Intermittent low volume and poor frequency response

I have a 10 month old 30 Gig video IPOD that drops the volume level and loses all its low frequency response at times and then will recover after a while. This happens on a song or songs that I know are properly leveled and that work sometimes and not at other times. The issue seems to be much more prevalent when IPOD is cold or cool. After it has been operating for an hour or so the problem seems to go away. It is very common when first turned on. I am getting ready to return it while it is still under warranty but thought maybe someone has an easy fix.

Returned for warranty repair.

Where can I find the amplitude and phase frequency response when the AC coupling is set on the 4472

My friends,
I need to know the amplitude atenuation but most important, the phase distortion introduced on the low frequency components of the signals when the AC coupling is setted (i.e. when the high pass filter is connected) on the 4472 and 4472B DAQBoard.
I can construct the amplitude frequency response by generating and aquiring a sine waveform of a knew amplitude. But I cannot construct the phase distortion introduced by the circuitry.
However, I assume that this crucial information should be available in the DAQ Manual or in the website of NI, but untill now I haven´t found it.
Thankning in advance,
crimolvic from Chile

crimolvic,
Here are the Specifications and Datasheet for the 4472. They indicate a phase non-linearity of less than 0.5 degrees across all frequencies.
For information on how this varies with frequency, see the attached
spreadsheet. This response was the result of testing on a single
4472. Although this is classified as a "typical" response, it is
not gauranteed.
Have a great, day!
Travis
Attachments:
4472 Phase Linearity.xls ‏21 KB

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.

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.

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 - sound and vibration

Hi,
I need to find the frequency response of the DUT. I am using the NI example from sound and vibration toolkit to do
so (LabVIEW 8.5\examples\Sound and Vibration\Getting Started\SVXMPL_Getting Started with Swept Sine (DAQmx).vi) now my problem is
to tabulate the phase difference between the stimulus and response when i do it i get constant values. Even though i dont give any response
signal to the channel i get the wave same as stimulus, is it suppose to be like that !!!!!!!!!! I will attach my vi so that it gives the idea where
i am measuring the phase difference pls check it and help me with this.....thank you in advance.
Attachments:
DAQ Freq resp (req)_trial.vi ‏121 KB

What is the DAQ card you are using for executing this Frequency Response?
This DAQ error is related to the output frequency of your signal higher than the possible output rate of the card. Basically, you are trying to update at a rate higher than the capability of the card.
For doing Frequency response, generally you need to have a NI DSA card (446x/447x/449x) with one Analog output and two analog inputs (minimum). Generate the swept sine signal from the Analog output channel, give this signal as input to your DUT and also to one of the Analog input channel. This input to your Analog input channel will act as your Reference signal. Then the response signal from your DUT, connect it to another Analog input channel. You would get a very good response results.
The reason why you need to have a DSA board is that, for doing Frequency response, we need to acquire both high frequency and low frequency components without much loss. This is possible only if your DAQ board has a higher dB value (in the range of >110dB) which is present only in DSA boards.
I have completed a Frequency Response Analysis just a week back with the same example programs. So there wont be any problem with that vi.
Regards,
Sundar Ganesh

Calculate frequency response using FFT and inverse FFT

Hi,
Attached is the program using FFT and inverse FFT to filter a time domain signal. The frequency response of the LPF can be obtained by using the chirp signal from 0 to 5kHz. However, I don't know why the signal obtained from a sine wave input is so strange. The amplitude is wrong and has a envelope outside. Please help to point out what's wrong with that.
Bill
Attachments:
fft filter.vi ‏87 KB

If you check the help text for sine wave.vi you'll see that it generates the sine wave based on the following formula:
yi = a*sin(phase[i])
for i = 0, 1, 2, …, n – 1 and where
a is amplitude,
phase[i] = initial_phase + f*360*i
This means that when you input a=1, f=0,1 and initial_phase=0 you will get a sine wave that is based on samples at every n*36 degrees; i.e. at 0, 36, 72 etc...due to this sample rate you never see the full amplitude (+/- 90 degrees), the wave is clipped at the top. If you input an initial phase of 64 degrees you will get the full amplitude, but the wave is still deformed due to digitalization...
The lower the frequency you put in, the closer the digitalized representation will be to the true sine.
Use the Waveform Generator VIs from the analyze palette if you want to have more control over the wave generation (sample rates etc.). (Not available if you have the base package.)
MTO

Generation of test limit masks for frequency response (limit testing in SAV)

Does anyone know a simple way for generating test limit masks from a saved frequency response?
The saved frequency response is from a reference measurement, and I need the limits to follow this response. I must be able to punch in different dB values for upper and lower limit, and for different frequency bands.
Thanks
Truls

Ciao emmanuele,
I think what you are searching for is the Extract Single Tone Information.vi that you can find under Waveform-->Analog Wfm-->Measurements-->Extract Tone.
Have you already taken a look?
Antonios

Low frequency Voltage measurement

Hi,
I am using PCI 6025E to control load bank and to measure voltage, current and frequency of a generator.
While current and frequency measurement is not a problem, voltage measurement is toublesome.
Since on test bench already there are few PCI boards and huge number of sensors, I do not want to add NI 9205 or any other additional boards and clutter the area (other people are also working there).
The problem with voltage measurement is that out of three generators two run from 400 RPM to 1600 RPM i.e.. somewhat 15-55 Hz.
I had two solutions for this:
1, Putting a step down transformer (230 Vac / 5 Vac) and then measure the voltage. But when testing at somthing like 15 Hz I have to deal with magnetic saturation with commercially available transformer. I do not want to take pain to go on built my own transformer.
2, Using high precision voltage Divider. The problem is that what should be the galvanic isolation? Isolation transformer, but then again low frequency problem.
Hence could someone suggest me since there is a bit confusion in my head.
Thanks.
Solved!
Go to Solution.

If you have a sine generator (15Hz migth be hard for the soundcard ) you can do a calibration of your transformer.
(If only the voltage amplitude is of interest , use a isolated DMM as reference and run your generators)
Or you buy a isolation voltage sense amplifier. (weidmueller, phoenix, wago, ...)
Greetings from Germany
Henrik
LV since v3.1
“ground” is a convenient fantasy
'˙˙˙˙uıɐƃɐ lɐıp puɐ °06 ǝuoɥd ɹnoʎ uɹnʇ ǝsɐǝld 'ʎɹɐuıƃɐɯı sı pǝlɐıp ǝʌɐɥ noʎ ɹǝqɯnu ǝɥʇ'

How do I get the line-in to record low frequencies?

The iPhone with GarageBand is potentially one of the most powerful tools for mobile recording ever. Unfortunately there are several tragic yet avoidable drawbacks that reduce it to nothing more than a mere toy. 1. You can't use the camera usb adapter to bring in loss-less digital stereo input. 2. Even if you could, you can't split the stereo into 2 seperate mono tracks. And to take the cake... 3. You can't do much recording with the 3.5 jack's line-in because you get ABSOLUTELY NO LOWS! This is due to the iPhones roll-off low frequency limiter filter, which i'm told since ios6 can possibly be disabled, though it's not in the Settings and i have yet to find an app that will do it in the background while running Garageband. Sure, the Garageband app could do that, but then it would be a powerful incalculably valuable tool in the hands of everyone. No lows means no drums, no bass, and vocals that sound like...well...like they were recorded via telephone. Yes, it's painfully clear you recorded that demo on a phone, because when you play it back in your home or car it sounds like your listening to someone play it for you on the other end of a phone call. There is no adjustment to compensate for the absence of low frequencies. SOMEONE PLEASE TELL ME I AM WRONG and that you have a work-around or know of something I've missed. How do you get lows into Garageband???

Thanks for the info. I am not wanting flat stereo nor digital input from the 3.5. Only wanting flat mono. But before I go the apogee route (which is a $150-200 solution, not to mention a solution that will add more devices to the connection and thus more opportunities for noise), I will probably just resort to purchasing an ipad (can get a mini for $299) . My mixer has usb out, and I can go straight into the ipad digitally with the usb camera kit, albeit I still can't split 1 stereo channel into 2 mono channels. But I digress: my entire purpose for this post is that i would like to stay as mobile as possible by using the iphone, but with all the darn limitations imposed by no one other than Apple, what could be an easy way to record has been nullified. What's frustrating is that all the components are there and I have them, all the hardware is more than capable; but in both cases (the 3.5 mono or the usb stereo) there is a software feature (or lack thereof) blocking me.
Again if anyone knows of an app that will disable the software low-cut, please respond...

Frequency response at low frequency

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