Specific sampling rate?
So maybe this is a stupid question, but I need to know because I train for a specific sound. Is there a way [to logic] to shoot/change of a certain frequency sampling rates. I can imitate the sound I'm looking for with a low pass filter, reverb and a distortion. But I don't want to 'emulate', this sound, I want to create. Then I can put my own effects and play with it like I want to. If I have to use a bunch of effects to make it sound like I want that also the addition of said effects remove the sound and sound horrible. as to where pulling the sampling frequency of the high frequency and no downs will make me THE noise that I need and always allow to add nice effects to make MY sound instead of someone else. I hope you know what I mean. Let me to you specific real once more. I want to pull or carry a certain frequency sampling rates for a sound under water. I don't want to use filters to make the sound. So can you please help me. I invited everyone locally on how to do it and nothing works. Also if this is not possible in the logic of tell me if there are third party plug ins or maybe even a different DAW that could do like komplete Kontrol or audacity.
See if this thread is helpful at all...
Tags: Professional Applications
Similar Questions
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I'm new to myRIO and use it to measure sine wave (0V to 5V) of up to 10 Hz 20 KHz. I also quickly transformed of Fourier (FFT) of the signal measured in real time.
Sideways FPGA of things, I try to keep things pretty simple, just read 2 channels of AI (connector B: AI0 and AI1), therefore potentially able to read each HAVE 250 kech. / s (as the unit has a capacity of 500kS/s). Does that mean this program gets a two analog inputs data exactly every 4 microsecond? If this is not the case, how can I make sure that the data is acquired through a fixed sampling rate?
I realized that we can add to the FFT in FPGA function, but I wanted to manipulte the acquired data of analog inputs before it is sent to the FFT, which I don't know how to do now. Can someone explain me how do the arithmetic data (muliplication, division and so) on the acquired data and analog inputs to reducde the 12-bit resolution 10-bit to program FPGAS.
Later, I created a myRIO program to read analog data 2 FPGA program which continues to turn in timed loop. In the program myRIO, the timed loop is configured to 1 MHz clock source type by a delay of 25 microseconds.
This configuration means that the loop runs exactly every 25 microsecond?
When I set up the less than 10 micro second time, myRIO has stopped working. Why is it so?
Is it because myRIO cannot run as fast as FPGA?
It is advisable to make the FFT of myRIO side analog data or FPGA?
When I tried to do FFT using the power spectrum of myRIO side, he asked for waveform data. What I acquire is data analog. How can I convert in waveform data?
If I read in the forum for help, I couldn't have the full answer to my doubts
Discussions at the Forum I did reference:
A lot of good questions here, I will try to answer as much as I can so as to offer a bit of advice.
First of all, if you are looking to acquire data at a very specific rate on the FPGA, you'll want to use the Timer VI. You are also going to use a FIFO of DMA to transfer data of FPGA in real time. A node read-write using as you do now means you'll run out of samples, or read the sample even several times. The link below is a very good tutorial on how to do what I described above.
http://www.NI.com/Tutorial/4534/en/
Later, I created a myRIO program to read analog data 2 FPGA program which continues to turn in timed loop. In the program myRIO, the timed loop is configured to 1 MHz clock source type by a delay of 25 microseconds.
This configuration means that the loop runs exactly every 25 microsecond?
When I set up the less than 10 micro second time, myRIO has stopped working. Why is it so?
Is it because myRIO cannot run as fast as FPGA?
In general, you should not run a timed loop much faster than 1 kHz. Using timed inside loop knots, you can monitor the real rate of loop during execution to see if f you meet your needs of the moment.
The portion of your myRIO RT is slower than an FPGA in the sense where it cannot manage the rates of lines 40 MHz (he makes up for it by being able to work with much better pictures) and it is important to remember that it is just a computer. The advantage of a real-time operating system, is that you have more control on the Scheduler, not that he is faster (less jitter, not faster code). There is more good reading below.
http://www.NI.com/white-paper/3938/en/
It is advisable to make the FFT of myRIO side analog data or FPGA?
When I tried to do FFT using the power spectrum of myRIO side, he asked for waveform data. What I acquire is data analog. How can I convert in waveform data?
I would say that it is generally advisable to treat your FFT on the side FPGA as long as you have the resources available, but for many applications probably little matter ultimately.
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Hi all
I use a module 9237 for certain measures of the load. My experiences last over time and so I'm generating a lot of data due to the minimum sampling frequency. I can't define an external time base so I can lower my sampling rate to something easier to manage? Even just a sample rate of 500 s/s would make a huge difference.
Thank you
Hi cannisbellum,
9237 specifications frequency range of minimum data (fs) using the internal master time base is 1,613 kech. / s and external use master timebase is 391 s/s. The simplest would be to sample at 2kS/s and decimate your data by 4. This can be done by using 'Decimate 1 table D' or ".vi Decimate (continuous).
Rates valid for the NI 9233 OR 9234 sampling and NI 9237 - http://digital.ni.com/public.nsf/allkb/593CC07F76B1405A862570DE005F6836?OpenDocument
Best,
CARISA
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using a NI4070 multimeter and I see the max connection is 300 kHz by respect it. But I don't understand how to set the min and max, acFrequency according to the sampling frequency or speed reading.
6 1/2 digits resolution, the speed can vary from 0.25 s/s to 100 s/s and this range corresponds to a lower end on the connection (minimum acFreq) from 1 Hz to 400 Hz.
(Q1a) - is the playback speed, controlled by the minimum setting of IviDmm_ConfigureACBandwidth? or vice versa?
Otherwise, I do not see how to control the rate of reading or the sampling frequency. IviDmm_ConfigureMeasurement only allows you to control the range and resolution.
(Q1b) - is there a way to directly control the sample rate (digitizer) or playback speed (dmm)?
(T2) - the upper limit of the bandwidth of AC always seems to be at 300 kHz... is there still a reason to reduce this maximum value?
(T3) - Finally, unlike the traditional niDmm function, the resolution via the IVI configuration should be passed as absolute value; does directly when number of digits and the beach? For example if I want to 6 1/2 digit to 300V range, I guess that by the specifications that the resolution should be set at 0.001 V... followign, if I want 5 1/2 digits to 1V range, the resolution should be set to 0.00001 V?
Hi Rjohnson,
I'll try to answer your questions as best as I can:
Q1A. The ConfigurACBandwidth function is used by the driver OR DMM to calculate the good aperautre for the measure. So yes, by adjusting your minimum frequency, you will affect your reading speed.
Q1B. Your reading rate will depend largely on your measuring cycle. To get a fast measuring cycle, there are a few things that you can adjust. You can programmatically control your time aperature, as well as your time to settle.
Q2. I can't find a reason to change. This parameter is only used for error-checking and verifies that the value of
This setting is less than the maximum frequency of the device.Q2B. I think what you say is right, but I'll need to check on that - I'll let know you as soon as.
Hope that helps. "" "I would recommend checking the explanation of the Cycle of the DMM measurement in DMM help' devices ' NI 4070" DMM Measuments "DMM measurement Cycle.
Take care!!
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fast sampling rate question...
Hello
I use USB-6009 and max sampling rate is about 48 K samples/s according to
the specification...
Question 1.
48 K samples/s means... only when you receive 1 analog input?
If I have 2 analog inputs then forge would be just half of the 48K?
Question 2.
using the daq assistant.
I would like to get about 50 samples between 10ms
If I do the math I get 5 K samples/s, which is enough for me
However, I played with samples to read and throughout the day, the sampling rate,
do not get this rate... (I'm outputing in file with LVM)
I searched on the sampling frequency, and people here said
samples read and sample rate do not havea correlation...
but I see clearly that they are relevant. When I change a setting
I get a different number of acquisition... I do N smaples.
Please help:)
Q1. Yes, except that the switching of channels takes awhile so the net price per channel is slightly less than half the rate of single channel. The USB-6009 specification document does not indicate what is the switching time. You should be able to get 5 kHz on both channels. 20 kHz might be close to the upper limit, but that's just a guess.
Q2. The DAQ Assistant is often not the best choice for maximum performance. I do not have the DAQ Assistant, so I can't be more specific. If you get the data as an array of DBL, rather than dynamic data type, it can be recorded directly, without conversion. The other thing that can make a big difference is a loop two architecture of producer/consumer. This allows the acquisition of data and save it to the file to run it at different speeds so that each can be optimized separately. If you are trying to acquire 50 ms of data at a time and then, he writes to the file, you write to the file twenty times per second. The first time, the operating system must reallocate some file space or do something else what delays write the file, your timing loop is disrupted.
Lynn
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on the sample rate of PCI-6723
According to the specification of PCI-6723, faster sampling rate is 45kSample/s, 32 channels working simultaneously. But he's always fine when I put the sampling frequency to 200kSample/s, 32 channels working simultaneously. This configuration will damage the material?
Ok. I assume you mean update rate of sampling frequency not. As long as you use the onboard buffer you can reach 204Ks/s 32 channels. Since you do not get an error the device and things work it is probably what you're doing.
You had asked the rate could only be achieved you would have been a mistake. And the material would not be damaged by incorrect update rate adjustment. You're ready to go!
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We are looking to buy a card PCI-6259 usable on a Linux machine. We would use NIDAQmx to access the card. If we were to use only a few channels to increase the sampling rate, do I need to select specific channels?
For example, the card is 1 MHz. If I select the channel 1 and channel 2, I can taste each channel at 500 KHz. could I choose 8 channel and channel 13 and still be able to sample each channel to 500KHz? Or need of specific channels to use when a subset of channels are selected?
Thank you
-Tom
Thanks for the quick response.
-Tom
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How to change resolution/sampling rate a project?
Is it possible to change the video resolution and audio sampling rates a project first CS5.5? I don't see how once a project is launched.
A project is not a specific resolution, frame rate and so on. sequences of only. That said, you cannot change the settings of the sequence once it has been created. The workaround is to create a new sequence with the settings that you want, and then copy and paste the content of your original sequence.
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Hello
If I scan documents to Audition 2.0 48 kHz and then convert to 44.1 using software for recording on CD, would there be loss of quality?
Thank you.
If it has voice recordings and you need an original CDs can just master at 44.1 k and have done with it. Since there is no content in your files anywhere near the limit of Nyquist (this is half of the sampling frequency and represents the highest frequency at this rate, you can save), then all you need to do is an extra oversampling when the format of distribution requires.
The speech contained in frequencies up to 12 kHz (hopefully) and even if you add music the answer will not exceed 20 kHz - and in any case, this is where the human ear cup (in children - rather less than that of adults). Therefore recording at 44.1 k means that you will capture all this, and you will be able to register without creating large files by scanning of a plu top noise load audio - which is all that recording at a higher sample rate will reach.
If you need pace to a different distribution format, then you have lost nothing - just do it when you need to.
I record a lot of original acoustic music and unless specifically requested to (which is never arrived, I might add...) I always master to 44.1 k. It's been proven by research academics caution that no one can tell the difference, then what is the point of a high sampling rate, besides wasting disk space unnecessarily?
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Currently, I am trying to log readings of DC voltage with an AA battery in an ASCII using LabVIEW 2009 of SingalExpress file and the USB-4065 digital multimeter (DMM). I have two stages:
(1) acquisition of Signal > voltage DC using DMM
-resolution 4.5 with 3.333333E the value-5 sampling period
2) save in ASCII
-The value to add to the file, delete the file after each race
Faster reading, I can get is a data point written in the ASCII file every eighth of a second.
Furthermore, I am new and software OR LabVIEW, the LabVIEW SignalExpress software I have is only for evaluation as it was included in the CD of the driver for the USB 4065 DMM.
- Max (30 000 samples per second) sampling rate is only achievable by a LabVIEW VI?
- Don't I have the wrong settings for DMM step?
- Is it because I haven't activated SignalExpress and am only using the evaluation version?
Thanks in advance for any help!
Hello Lukos,
You are assuming that you need access to lower level functions in order to obtain the higher sampling rates. In order to get these speeds, we need to disable some settings that are not accessible via Signal Express. You can create a VI and then use a step VI call in Signal Express to stay in the same environment.
Kind regards
-Travis E
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Hello
What is the sample rate max 5154 PCI for two channel inputs? The manual States the 2GS/s is for one channel only. So, am I not able to get a bandwidth of 1 GHz for the simultaneous measurement of two channels? Thank you!
Hi gbhaha,
First of all, TIS mode up to 20 GECH. / s using an ADC, while your real time sampling uses two converters a/n at the same time to a single channel. Take a look at these diagrams that I linked in my first post for more details on this architects.
About the difference in the bandwidth between the 5153 and 5154 - the 5153 has 500 MHz of bandwidth in its circuits, even when acquiring at faster sampling rates. The 5154 1 GHz of bandwidth, this is why it is more expensive.
Kind regards
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sample rate real vs min sampling rate
I'm sure it's an obvious answer, but here goes.
I have a USB-5132 ' scope and using niScope horizontal configuration Timing.vi I put, among other things, the minimum sampling rate. In my case, I chose 20 MHz, which of course gives a sampling of 50 ns period.
I use niScope reading (poly) .vi with the WDT variant to read waveform data. I noticed something very strange - waveform limit testing throw error 1802 "signals have a dt of different values '-if I put a waveform components unclusterizer Get on the wire of waveform and looked at the value of dt of the wave." He told me that my dt is 40 ns, which of course is of 25 MHz. I also plead for only 2000 samples.
So what causes this shift? Why the digitizer does not accept everything just my desired sampling frequency?
Austin Walton wrote:
Andy,
The setting of minimum sampling frequency is the frequency at which digitized
the samples are stored, expressed in samples per second. This setting is rounded
up to and including the next legal collection that supports your device. Ownership of the actual sampling rate calculates the actual sample used for the acquisition rate.Unless you specify another source of the clock, the digitizer uses an internal oscillator as clock source. For the 5132, this oscillator is clocked at 50 Mhz. When using the oscillator internal as the sample, the digitizer clock source can use versions split to the bottom of this clock, for certain sampling frequencies are not possible.
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How to create a waveform from an array with arbitrary sampling rates
Hi I know that sounds a little silly,
Suppose I created a simple table of figures DBL with a structure For, Say size 16. now, I want to create a waveform DBL with these 16 numbers at an arbitrary sample rate. so if I use 1 kech. / s to the sampling frequency, I want to have a waveform with a duration of 16 milliseconds.
Please help me, I need it too
TNX
Hello
You must use the wave to build function as shown here: http://zone.ni.com/reference/en-XX/help/371361G-01/lvwave/build_waveform/ . Wire you your Board at the entrance Y and then wire the dt of entry in your sampling rate.
-Zach
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I am using a cDAQ 9172 with modules NI 9219, NI 9264 and three NI 9211. I'm looking to acquire signals out of the acquisition of data within a loop under continuous sampling. My program works fine if I set the number of samples to read 1-2 Hz, but I need to go faster than that. If I change the sampling rate, the loop is executed at this speed but sensors still read only in samples at 2 Hz and then duplicating over and over again. I was wondering if it was possible to read on 1 sample at the time of the acquisition of data at a faster rate. I know that the frequency of sampling on the sensors and data acquisition are much higher than that. 1 sample at the time of the Board of Directors has the limitatioins of being only able to run at 2 Hz? Please let me know
Thank you
Craig
Hi Craig,.
I don't know exactly what you describe. Are you feeding the DAQmx Read output in an express VI? Or are you using the express VI DAQ Assistant for the analog input task?
If you use the DAQ Assistant, you can set the ADC synchronization mode without changing your code:
If not, use the 'Active channels (if subset)' property to control the subset of channels on which your VI defines AI. ADCTimingMode.
For example, the following code snippet creates 8 virtual channels named myVoltage0 by myVoltage7 and sets HAVE them. ADCTimingMode on myVoltage4 of virtual channels through myVoltage7. These are in the cDAQ1Mod2/ai0 physical channels via cDAQ1Mod2/ai3:
If you leave off of the entry "name" on the string to create VI, then the virtual channel names are the same as the names of physical channel, so it's the equivalent:
And by the way, a right-click on the property and selecting "create > Constant ' context menu saves you from having to hardcode a number like 14712.
Brad
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Meter in a loop and read reduced sampling rate
Chassis: 9188
AI: 9219
CI: 9401
As pictured, without reading of CI, I can adjust the sampling rate of metered software. But reading of CI, the maximum rate is around 5 Hz. I already changed 9219 high-resolution property to high speed. What is the problem?
Hi, Carlos, thanks for your response. I acutally has solved this problem by using the connection series I and CI (i.e. connect error off HAVE error in the CI) but not parallel as the pic shows.
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