Sample Rate vs Frequency

"The increments across the x-axis of the graph are controlled by the sample rate.  In this example, 8000 units along the x-axis represents
one second of audio. "
Actually not. One should sample at twice the highest possible input frequency.
Renee
"MODERN PROGRAMMING is deficient in elementary ways BECAUSE of problems INTRODUCED by MODERN PROGRAMMING." Me

You see basically an FFT describes the frequencies and amplitudes of samples or events. If one undersamples one cannot hope to describe the event and that's what it's all about. If one samples at a rate less than twice the maximum frequency one cant cant
hope to describe the event because one cant be certain that data is take fast enough to describe it. I'll give you a worse case example.
Consider an electrical sine wave occurring at exactly 1 khz. The signal oscillates from +1 volts to -1 volts at the rate of one khz, Let us consider what would happen if we sampled this signal at 1 khz, We would see a constant voltage at the input although
the signal is changing sinusoidally at a 1 khz rate, If we sampled at 1.5 times the signal we'd see variations. And if we sampled at twice the maximum frequency we could be sure we're sampling at a rate which will allow description of the input.
Renee
"MODERN PROGRAMMING is deficient in elementary ways BECAUSE of problems INTRODUCED by MODERN PROGRAMMING." Me
Not quite.
When you're taking a true analog signal and "digitizing it", it will never have the resolution of the analog signal ever again.
The faster the sampling rate, the closer the approximation but the actual digital result is and will always be a square wave (not triangle wave, not sine wave, a square wave).
The faster it samples it, the closer the "stair" to the next one, thus more closely approximating the original sine wave, but it can never be completly true to the original signal.
This isn't related only to sound but to any true analog waveform being "digitized".
Please call me Frank :)
Well maybe you could do something like a Bezier curve or
Bézier Spline that would only need 4 points for each half of the wave in order to match the wave or something digitally? I don't know much about it but it seems to me you could somehow match an analog waveform digitally.
Please BEWARE that I have NO EXPERIENCE and NO EXPERTISE and probably onset of DEMENTIA which may affect my answers! Also, I've been told by an expert, that when you post an image it clutters up the thread and mysteriously, over time, the link to the image
will somehow become "unstable" or something to that effect. :) I can only surmise that is due to Global Warming of the threads.

Similar Messages

  • Parallel port sample rate

    Hi,
    I`m desperately in need of your help.
    For my school project, I`m trying to write a labview code in order to acquire air pressure data by using 8-bit analog digital converter via parallel port. Although my signal frequency very low -100 Hz-I think  I experience sampling rate problem. I  want to put sample rate and frequency data in my code but  how can I put it in my code?I `m giving square waveform via function generator which I need to see it on my graph in Labview. But there is something wrong in my signal in waveform chart. You can see screenshot and code attached.  . Also there are some noises and amplitude is not stable . How can I fix this?
    Would you please have a look at to my code and tell me how can I set sample rate and frequency?
    Attachments:
    CAN1.VI ‏21 KB

    Hi LW-s
    There isn't really a way to set the sampling rate for these VIs. You will have to use software timing. The only way to do this is with 'wait' functions inside your loop. You can also implement some logic to smooth/adjust your waveforms.  For a square wave the logic is pretty simple. If the input value is above a threshold store it as the max and if it is below the threshold store it as the min. (See the example below). For sine waves this is a bit more difficult and if I were you I would probably try to implement some sort of averaging. 
    Best Regards,
    Chris J
    Message Edited by ChrisJ on 08-23-2005 05:31 PM
    Attachments:
    example.JPG ‏86 KB

  • NI-DAQmx frequency sampling rate

    Hi there!
    I'm working on setting up a data acquisition Labview VI, to measure different signals on a test rig.
    I'm using the NI-DAQmx assistance (the Express VI?) to continously measure analog signals (Variable current, voltage and temperatures). This is working just fine, and i can change the sampling rate by writing to the express VI. The idea is, that the user can change the sampling rate from around 1 to 500 Hz. 
    We do however have a sensor that transmittes digital signals (a frequency), and are using a NI-9423 module to "read" it. As this is a digital signal, another NI-DAQmx express VI is needed to handle it (that's ok), but so far we can't figure out how to alter the sampling rate - it's apperently locked at 1kHz. 
    Being that we want to merge the analog and digital signals to one array, we are recieving overflow errors from the "analog" DAQ, if it's not set at exactly 1kHz. 
    So, in short - is it possible to change the sampling rate of a DAQmx recieving frequencies? So that we to DAQ assistences have the same sampling rate?
    Help would be greatly appreciated!
    - Nicklas
    Attachments:
    DAQissue.PNG ‏64 KB

    Unlike voltage measurements, which tend to be (more or less) instantaneous, frequency measurements take a finite (and often variable) amount of time.
    If it is a slow signal then you measure the number of counts of your reference clock that occur in one period of your input signal. As your input signal varies in frequency, so does the measurement rate. If it is a fast signal, you can either measure how long it takes to get n cycles or your input (again variable) or you could count how many cycles of your input occur in a fixed time period.
    The NI help on frequency measurements describes three different ways you can configure a counter to measure frequency.
    The long and short of this is that generally counter measurements come at variable measurement rates which can be problematic to fit in with a fixed rate loggin system. If the measurement period is much smaller than your desired rate then you can wait and trigger a measurement at regular intervals. If not, you can let the counter run at its own rate, placing the latest result on a notifier, and in another loop just read the latest measurement from the notifier each time you want to record a result. Depending if you counter is running faster or slower than your desired logging rate you will end up with either missed samples or repeated samples. There are inherant timing inaccuracies in both approaches because, unlike analog measurements, the counter measurement is not made at 'that exact time, now!' but over a period of time which may be long or short compared to your logging rate.

  • Is it possible to detect low frequency signals with a high sampling rate?

    Hello everyone,
    I'm having an issue detecting low frequency signals with a high sampling rate.  Shouldn't I be able to detect the frequencies as long as the sampling rate is at least 2 times the highest frequency I will measure?  The frequency range I am measuring is 5-25 Hz, and I use Extract Single Tone.vi to measure the frequency.  The sampling setting I am using is 2 samples at 10 kHz.  Is there a method I can use to make this work?
    Attached is the vi.
    Attachments:
    frequencytest.vi ‏21 KB

    You are sampling at 10Ks/S, but only taking two samples. What do you expect to see? If your signals are binary (On or Off) you would only see either an on or an off, or if the rise/fall time was fast and you were Extremely lucky, one of each. If you want to see a waveform you have to sample for at least the period of a waveform. So you should take samples for at least 0.2 seconds to capture an entire waveform at 5Hz, ideally longer.   Think of looking at a tide change at a dock. If you want to see the entire tide change you will probably have to measure repeatedly over 24 hours, not just run out on the dock, measure the height twice and leave. That wouldn't tell you anything other than at that precise moment the tide height was X, but not that it was at high tide, low tide, in between, etc.
    I type too slowly, I see that a more technical answer has been given, so mine will be the philosophical one!
    Putnam
    Certified LabVIEW Developer
    Senior Test Engineer
    Currently using LV 6.1-LabVIEW 2012, RT8.5
    LabVIEW Champion

  • IIR Filtering and response .vi: Butterwort​h filter magnitude response depends on sampling rate -why?

    Hi folks,
    I am not expert in filter design, only someone applying them, so please can someone help me with an explanation?
    I need to filter very low-frequent signals using a buttherwoth filter 2. or 3. order as bandpass 0.1 to 10 Hz .
    Very relevant amplitudes are BELOW 1 Hz, often below 0.5 Hz but there will be as well relevant amplitudes above 5 Hz to be observed.
    This is fixed and prescribed for the application.
    However, the sampling rate of the measurement system is not prescribed. It may be between say between 30 and 2000 Hz. This will depend on whether the same data set is used for analysing higher frequencies up to 1000 Hz of the same measurement or this is not done by the user and he chooses a lower sampling rate to reduce the file sizes, especially when measuring for longer periods of several weeks.
    To compare the 2nd and 3rd order's magnitude response of the filter I used the example IIR Filtering and response .vi:
    I was very astonished when I the found that the magnitude response is significantly influenced by the SAMPLING RATE I tell the signal generator in this example vi.
    Can you please tell me why - and especially why the 3rd order filter will be worse for the low frequency parts below 1 Hz of the signal. I was told by people experienced with filters that the 3rd oder will distort less the amplitudes which is not at all true for my relevant frequencies below 1 Hz.  
    In the attached png you see 4 screenshots for 2 or 3 order and sampling rate 300 or 1000 Hz to show you the varying magnitude responses without opening labview.
    THANK YOU for your ANSWERS!!!
    chris
    Solved!
    Go to Solution.
    Attachments:
    butterworth-filter-differences.png ‏285 KB

    Hello Lynn,
    thanks for the answer. You are right that there are few points "behind" the curve in the graph, see png.
    However, this is the filter response which Labview (2009) provides to me directly out of the "IIR Filter for 1 Channel. vi" in the "filter information" output cluster. Where up to now I do not know how to influence it - apart from adjusting the input parameters "IIR filter specifications". OK, I assume I have to gain more knowledge of this. The curve of the magnitude resonse dies not change when I change the number of samples of the input signal of the signal generator, only wehn I change the sampling rate.
    I used directly the example vi from Labview with the name indicated in my first post "IIR Filtering and Response.vi".
    So I assumed that everybody has it in his/her examples shipped with LV and it is not necessary to post it.
    I just adjusted the size of the diagram of magnitude response to see the curves better as you see in the attached vi.
    So I did no changes to the vital parts of signal generation and filter of the example. The screenshots are like they come from the example when using the option "one waveform" where I as user assume that this which is behind is quality-controlled by NI.
    I was also astonished that the filter magnitude response is different to the one I copied out of graphs 1 year ago - but I unfortunately cannot reconstruct which example I used there...
    Thanks for any further comments
    chris
    Attachments:
    IIR Filtering and Response_CH.vi ‏55 KB
    butterworth2nd_order_bandpass_0p1to10Hz_mag_response.PNG ‏18 KB

  • How to use on board counter to change sample rate dynamically on pci-6134

    Hi,
    I am relatively new in LabView.
    I am making power quality measurement system and I need to vary the sampling rate of my pci-6134 dynamically (all channels simultaneously). What I need is to have a constant amount of samples in each period of measured signal (grid voltage), which changes slightly all the time. Therefore I will have to measure the voltage, find its exact frequency and then adjust the sampling rate of daq accordingly. I know that there will always be some delay, but I would rather like not to go into any predictive algorithms...
    I have found an information in the Forum that one of possible solutions is to use an onboard counter to change the sampling frequency but I have no idea how to make that. Can someone help me or possibly show an example? 
    Is there a simple way to solve that problem?
    Thanks in advance
    Andrzej

    At least at a glance, the code generally looks like it ought to work.  Two thoughts:
    1. Instead of getting into PFI3 vs PFI8 routing stuff, can't you just specify "Dev1/Ctr0InternalOutput" as the AI Sample Clock source?  (You may need to right-click the terminal to get at the menu that exposes the so-called "advanced terminals").
    2. Try writing both the freq AND duty cycle  properties when you want to update the freq.  Or try using the DAQmx Write vi instead of a property node.  My past experience suggests that writing only the freq property *should* still work, but writing both isn't hard to try and may turn out to help if the behavior of your version of DAQmx differs somehow.
    -Kevin P.
    P.S. Bonus 3rd thought.  I just went back to reread the thread more carefully, including your first screenshot.  I'm now thinking that maybe the hardware actually WAS behaving properly, and that you just weren't aware of it.   When you query the AI task for it's sampling rate, all the task can know is whatever rate you told it when you configured it outside the loop.  So even as you change the counter freq to change the actual hardware sampling rate on the fly, the AI task will continue to report its orig freq.  After all, how is *it* supposed to know?
           Try an experiment:   Set your original freq very low so that the AI task produces a timeout error without getting all the requested samples within the 10 sec timeout window.  Run and verify the timeout.  Then run again, but after 3-5 seconds set a new frequency that will produce all those samples in another 1 sec or less.  Verify that you get the samples rather than timing out.  That should demonstrate taht the counter freq change really *does* produce a change to the hardware sample rate, even though the task property node remains unaware.

  • Low sample-rat​e measuremen​ts on the PCI-6115 DAQ card

    I need to measure an analog signal at a sampling rate of a few tens to hundreds of Hz in sync with the rising edge of an external clock. I have a PCI-6115 DAQ card w/ Labview 6.1 and NI-DAQ 6.9.2. The PCI-6115 is a high speed card and has a minimum sample rate of 10 KS/s. Is there any way of implementing a low sampling rate measurement using the PCI-6115 in sync with an external clock?
    Thanks in advance.

    Kuldeep,
    It is possible to do what you are describing above (in fact I don't think an external clock is required to do this), however, bear in mind that the reason for this minimum sampling frequency is due to the ADCs used on this high speed board. The ADCs used are pipelined ADCs, meaning that when a signal is digitized, it is digitized in distinct stages within the ADC (in the case of the 6115, I think there are 3 stages involved). Data is moved from one stage of the ADC to the next each time a sample clock pulse is recieved. If too much time elapses between these clock edges, the signal to be digitized can actually 'leak' off of the ADC. This can result in improper digitization, which can lead to less accurate measurements. So, while it is possible to mak
    e the device sample below it's minimum rate, it may be advisable to sample faster than the rates required by your application, and either average multiple data points per measurement, or throw away extra points taken.
    I hope this helps,
    Dan

  • IPhone: AudioQueue - is it possible to change the sample rate?

    I've been playing around with the AudioQueue stuff for a few days and it's all working fine.
    I was trying to build a low-latency playback system by making the streaming buffers the same size as the audio file and pre-loading the buffers (which works fine) but I've hit a snag.
    I've been trying to get the streaming to work at different sample rates so that I can play back the same sample at different pitches. I managed to do it by modifying the sample rate in the AudioStreamBasicDescription structure but in order to actually make the stream playback at the new rate it seems you have to create a new output, reload the audio file into the buffers and re-enqueue the output queue before starting playback again, otherwise the sample rate change has no effect.
    There is a method to set queue properties; AudoQueueSetProperty() but unfortunately the sample rate Property (kAudioQueueDeviceProperty_SampleRate) is read-only
    Can anyone suggest a way to achieve this with AudioQueue or do I need to move over to OpenAL?
    Thanks,
    Neil

    Dan,
    there is one point in your understanding, which i am not sure what you think about when talking about it: I understand E-series devices do not support this property change while the VI is running.
    infact, you cannot change the sample clock rate during acquisition. but
    this does not mean that you cannot change it while the VI is running.
    you have only to interrupt the acquisition. since you want to acquire
    continuous, this would have the same effect as stopping the vi, i asume.
    so the best way to accomplish this task is to use an external clock.
    this is e.g. often used for acquistion on rotating shafts. the
    acquistionrate is always e.g. 24 points per revolution regardless of
    the rotational speed of the shaft, except for a maximum frequency of
    course.
    Norbert B.
    NI - Germany
    Message Edited by Norbert B on 09-14-2005 04:16 AM
    CEO: What exactly is stopping us from doing this?
    Expert: Geometry
    Marketing Manager: Just ignore it.

  • Setting the sampling rate in SignalExpress

    I am using a cDAQ-9172 with a strain gauge module and a thermocouple module and using SignalExpress.  I want to acquire data at a pretty low frequency rate (2Hz), but I am unable to use 1 sample on demand in the acquisition setup.  I get the following error.
    Error -201087 occurred at DAQ Assistant
    Possible Reason(s):
    Measurements: Task contains physical channels on one or more devices that require you to specify the Sample Clock Rate.
    Specify a Sample Clock Rate.
    Device: cDAQ1
    I am unable to setup a sample clock rate as it is 'blacked' out by the software in the 1 sample on demand mode.
    When I try to use continuous or 'N' sample mode, the cDaq is sampling at a rate of around 1600 Hz, even though I am putting in a value of 1Hz on the seti[ screen.  The large amount of data will prevent me from downloading the data to Excel for further reporting.  I obviouisly don't need to sample my TC module at that rate either.   
    Is there anything I can 'easily' do to decrease the sampling rate down to a low level?  Everything I have tried doesn't seem to work and I don't really want to go away from SignalExpress.
    Jay 

    Hi Jay
    I am assuming that you are using the NI-9237.  This strain module has only a specific set of sampling rates.  If you specify a rate that is not supported it will corers the rate to the next higher sampling rate.  Also since you are using the cDAQ chassis, there is only sample clock so it must choose the highest sampling rate.  So it is not possible to sample at a lower rate.  Please see the following link.
     If you have access to LabVIEW then I would suggest taking your readings at the fast rate and then average the samples before you write them to your file.  The following link has some more information on this. 
    If you are limited to Signal Express, this task is tricky.  If you select a N-Sample acquisition, set the samples to read to greater than two, and set the post acquisition delay  under the execution control tab to 1000, you acquisition will acquire the two samples at the very fast rate, but it will wait 1000mS between each acquisition.  This will slow the overall sample rate.
    Chris_K
    National Instruments
    Applications Engineer

  • How to adjust sample rate of data?

    I have some data collected at 1683 Hz (yes, that was what I had!) and would like to reduce the sampling rate to some meaningful number, say 1024, 500, 400, or similar.
    What should I do?

    Well, the calculation is the approximation of your channel (variables with index 0) to a new one (index 1).
    The freq(0) and freq(1) are the sampling frequencies for the channels for the case you have waveform channels.
    The n(0) and n(1) are the numbers of the data points inside the channels. The new created channel should have the number n(1), calculated from n(0) with regard to different sampling ratios.
    The real code is the line Call ChnSplineXYCalc(..... The properties swapping can be commented out, but then the new channel would have a "system name", something like "Approximated XY", and the same for description and units... Probably one can avoid it by changing of settings, but I use to do it by code.
    In short, here you copy the properties from the "old" channel and paste them to the new one.

  • [Q] How to set the sampling rate separately?

    Hello,
    I'm using LV5.1 in Windows98 with SCXI-1200.
    I want to set different sampling rate in each input channel.
    I've ever used "AI acquire waveforms.vi" only.
    Anybody can help me?
    Example codes are highly appreciated.
    Regards,
    Hyun-ho Lee
    [email protected]

    The inputs of the SCXI-1200 are multiplexed to a single Analog-to-Digital converter. Because of this, the only difference in sampling rates that are achievable would be integer divisions of a common high frequency. This is functionally identical to acquiring all channels at the highest rate, and decimating (throwing away) data from channels that need lower data rates.

  • Highest audio sampling rate in CS4?

    Hello,
    I apologize if this has already been asked, but I have been searching everywhere and I simply cannot find the answer to this.
    What is the highest audio sampling rate that can be utilized in Premiere Pro CS4? Can it import and export 192kHz 24-bit audio?
    Thanks in advance

    Hey Hacienda,
    I might not have the experience in audio work you have since I've only been doing this for the past 6 years or so.  But I've been a musician for far longer than that, and I've learned A LOT mostly from really smart people in the industry.  So, I'm not gonna lie to you and say that I've done extensive testing in this area because I simply do not have the equipment, nor the money to buy it (WAY too expensive).  But we do share the neophyte status when it comes to video editing :-P
    Anyways, the Nyquist Theorem is not a theory, which is what people are led to believe.  It is a theorem, meaning it's already mathematically proven.  It is proven that, as long as you follow the premise of capturing twice the highest frequency of the sound source, you'll get a perfect reproduction of it.  To capture more than that is a waste of bandwith specially because most people won't even hear above 18KHz, nor do they have the equipment to reproduce such frequencies.  Most consumer systems and audio gear, including those found in professional studios, go up to about 22KHz.  You need to spend BIG dolars for anything that goes beyond that.  So, who are we really making music for here?  The super rich?  Dolphins?
    Now, I know you're not just talking about higher frequencies, but the amount of samples needed to recronstruct a perfect copy of the original waveform.  OK, well, this is the kind of snake oil marketing BS I was talking about.  The biggest one being that 1bit DSD crap that Sony/Phillips is pushing.  Adding more samples to the recording will not make any difference on how faithfully you can reproduce a sound.  It'll just make the files bigger for no reason.  Again, the Nyquist Theorem already proves this.  This is FACT!  Here's a link I found interesting regarding these audio industry lies, maybe you will too: http://theaudiocritic.com/back_issues/The_Audio_Critic_26_r.pdf It starts on page 5, but the one pertaining this discussion is lie #3 on page 6. :-D
    Don't forget that modern converters already sample at much higher frequencies than the target sampling rate.  I believe my RME Fireface 400 samples at 5.6MHz, which is twice the amount of samples compared to DSD technology, before going back down to the target rate.  But, like I said, it does so for other reasons and NOT because it needs that many samples in order to faithfully reproduce a waveform.  Of more importance are the quality of the FIR (Finite Impulse Response) filter and the clock inside the converters.  These components are what make a converter high grade, among others.  The converter chips themselves are very inexpensive (in the tens of dolars) which why you hear some companies advertizing having the same converter chip as a ProTools HD rig (not the best example I know).
    By the way, I didn't say humans can only hear up to 20KHz.  I'm sure there are people who can hear above that.  My point was that the 20Hz - 20KHz range is what's generally accepted as an average for humans (which implies that there are people who can hear avobe/below that).  Also, the reason why modern-day pop records causes headaches and sound horrible is because of a totally different issue known as "The Loudness War" (I'm sure you know about it so I won't go into details).  However, I do agree with you as far as compressed audio goes.  Unfortunately, there's a reason for that and there's nothing we can do about it until the day Internet bandwith becomes more accessible and cheaper.  Eventually it'll get to the point where uncompressed audio can be streamed reliably through the net.  But, until then, we're stuck with MP3, AAC, DTS and other audio compression formats.  As far as digital media distribution goes, it's the future and companies are seeing that.  More and more people download music rather than buying CDs, so I do believe those numbers are accurate.  Just look at sales from iTunes and even games like Guitar Hero and Rock Band.  It's just a matter of time.
    Take care!

  • Higher Sampling rate... in logig

    no one know certainly the tecnical limit of time recording at 96 or 192Khz in logic? In the next update It can be fixed?... I can record only 62 minutes at 192!!!!

    the difference between a 48khz to 192khz is not only in the possibility to reach a 50khz frequencies (a man whit good hear can reach 18khz... good hear),
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    (I have SOME space on the HD, is enought 900Gb of free space?)

  • ADjusting the sampling rate on a FIR filter?

    How do I adjust the sampling rate on a digital FIR filter? Thanks in advance.
    -David

    You should really start a new thread instead of posting to one that is 5 years old.
    To answer your question, it depends on your data. I don't use the DFD but with the filter functions in LabVIEW, if you pass a waveform data type to the function, then the waveform data type contains a dt value. So, set the DAQmx Read to return waveform data. If you are using low level filter functions where the input is a 1D DBL array, then the filter has to be configured. With the low level functions in LabVIEW, you use the various coefficients functions that have a sampling frequency input.

  • Audition 3 seeing a different sample rate setting than what the device shows

    Hi,
    I have just installed Adobe Audition 3, along with the 3.01 patch, on a brand new system running Windows 7 64 bit. The mother board is an Asus Sabertooth X58 using Realtek High Definition Audio. The device drivers show that the audio sampling rate for line input is set to 24 bit 192K. I wanted to set it to the maximum that the sound card would allow to test performance and audio quality.
    The problem is when I bring up Audition 3 and hit record, I get the message "We do not support recording when your file does not match your hardware sample rate. Your current hardware sample rate is 44100Hz". Clearly this is not the case since the Line In Properties - Advanced tab is displaying "2 channel, 24 bit, 192000 Hz (Studio Quality).
    Under Audition's Audio Hardware Setup it shows only one choice for Audio Driver: Audition 3.0 Windows Sound. It also displays Sample Rate: 44100Hz, Clock Source: Internal, Buffer Size: 2048 samples with no way to change these values.
    If I click on the Control Panel button I get:
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    Device Name: Line In (High Definition Audio Device
    Audio Channels: 2
    Bits per Sample: 16
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    Dale

    DaleChamberlain wrote: Anyone know of how I can change these settings to get Audition to agree with the device settings?
    I'm afraid that life is nowhere near that simple. The main issue here is that Audition, in common with most audio software, uses a driver system called ASIO to talk to the sound device - this cuts out a lot of the OS and reduces the latency of the system considerably. There are several problems with ASIO though - the first being that it only supports a single device per system (or sometimes multiple identical devices if the manufacturer can make them look like a single device), and with software designed to use this driver, then to use any other driver (like a native Windows one) you have to use a converter stage like ASIO4ALL. This will convert the ASIO streams to WDM, and let you use multiple sound devices - but with increased latency.
    It's the second problem that's really going to stuff you though - and that is that quite reasonably, ASIO is limited by its inventors to run only at three sample rates; 44.1k, 48k and 96k. So there's no way you can run at what you think might be a higher quality setting. All settings above even 48k are making your sound device work much harder, and for what? All that happens is that you increase the potential frequency response to way beyond the human hearing range - to no purpose at all. You don't have sources that can produce useful output at these frequencies, and you certainly don't have the means to reproduce them. This has all been well documented and explained before, so I'm not going over all that again. In a nutshell, Nyquist points out that any digital sampling device has a frequency response limited to a maximum of half of the sample rate, so for 48k that gives us a frequency response up to 24kHz - comfortably higher than any adult can hear by quite a long way. Anything you sample and record beyond this by using even 96k is nothing but noise as far as humans are concerned, and unpercievable noise at that.
    So what the line input properties tab is saying is, if you have a non-ASIO driver designed to support all potential rates, possible. You don't have an ASIO driver available, because it's a built-in sound device, and anyway you've already pointed out that it's using the Audition Windows driver (a cut-down version of ASIO4ALL, effectively), so a conversion is already taking place. What Realtek refer to as 'High Definition Audio' is no such thing - all on-board sound devices of this nature are of universally low quality, and to improve this you'd need an external device - of which there are many available, usually with dedicated ASIO drivers. But none of them will work with ASIO beyond 96k, simply because the standard doesn't support any higher rates.
    If you download and use ASIO4ALL (it's free), then you will get an additional control panel which will show you exactly what your sound device is capable of doing as far as Audition or any other ASIO software is concerned, and this is a useful diagnostic tool anyway, so it's worth doing. You just select this option when installed, instead of the Audition Windows Driver.
    I'm sorry to be the bearer of what seems like bad news, but actually, it isn't. You will percieve no quality difference at all running at anything beyond 48k sample rates; all you will be doing is wasting your computer's resources unnecessarily. You waste both processing resources and hard drive space by processing at ridiculously high sample rates, and there are zero returns.

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