Sampling rate of change

Similar Questions

• 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.

• 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.

  Effect under water

  See if this thread is helpful at all...

• DMM (NI 4070), how to correctly set AC Freq (bandwidth) by the sampling rate

  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!!

• Sampling rate 9172 OR cDAQ

  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

• 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.

• 6255 sampling rate causes the dc offset

  I see a dc offset in the measures of analog input I select different sampling frequencies.

  I have USB-6255 (mass termination) multifunction data acquisition and I use measurement and Automation Explorer to put in place my entries.

  My raw analog input is-0, 6250 volts dc, I have set up a task that uses 4 differential channels with no custom scale.

  I have defined the scope of the input signal to +/-0 .8v for you sure I get good resolution.

  Acquisition mode is continuous, samples of read is 1 k and I play with the order of 10kS/s rates 50kS/s.

  While this task runs in the MAX, I can put my cursor in the rate field and use the top and down arrow keys to change the sampling frequency. As I do, I can see the light changes reported as much as 150 MV rate from one to the other.

  It is a significant change when the total time of entry is lower to +/-1v.

  The direction of movement is independent of the increase or decrease of the sampling frequency.

  For example,.

  23kS/s, the declared value is - 0.540v,

  24kS/s, she moves to-0.620v.

  25kS/s, she moves to-0.690v.

  26kS/s, she moves back to-0665v.

  27kS/s, she moves back to-0.625v.

  and 28kS/s, she moves to - 0.535v.

  At first, I thought that the sampling change made a change of the input impedance and change the load on my source, however, all the time, my dc signal source remained at the - 0.625v (as measured with a multimeter fluke at the connection point to data acquisition).

  Why this is happening and what can I do about it? I want to give my users the ability to choose their desired sampling frequency.

  My guess is that I need to add an amplifier to fixed gain with a gain of 5 to 10 to make the input signal to use the maximum of the analog input level (+ / 10v).

  I use MAX version 5.0.0f1

  Thanks for any help,

  Tobin

  Hi Tobin,

  What do you use to generate the signal-. 625 volt? If you are using a switching power supply, you can experience aliasing where the power supply is turned on and stop.

  In addition, are see you the same tensions at the same sampling rate? See you always - .540v to 23kS/s or vary over time?

  Finally, you have a second 6255 you can try to replicate this on? It could be that the unit is defective.

  N

• 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

• . VI filtering IIR and response: response of Butterworth filter size depends on sampling rate - why?

  Hi people,

  I'm not an expert in the design of the filter, only a person in applying them, so please can someone help me with an explanation?

  I need to filter signals very infrequent using a buttherwoth filter 2. or 3. order of the bandpass 0.1 to 10 Hz.

  Very relevant amplitudes are BELOW 1 Hz, often less than 0.5 Hz, but there is as well the amplitudes beyond 5 Hz to observe.

  It's fixed and prescribed for the application.

  However, the sampling rate of the measuring system is not prescribed. It may be between say between 30 and 2000 Hz. Depends on the question of whether the same set of data is used for analysis of the higher up to 1000 Hz frequencies on the same measure or this is not done by the user and he chooses a lower sampling rate to reduce the size of files, especially when measuring for longer periods of several weeks.

  To compare the response amplitude of 2nd and 3rd order filter, I used the example of IIR filtering .vi and response:

  I was very surprised when I found that the response of greatness is considerably influenced by the SAMPLING RATE I say the signal generator in this example vi.

  Can you please tell me why - and especially why the filter of order 3 will be worse for the parts of low frequency below 1 Hz signal. Told me of people experienced with filters that the 3rd oder will less distort the amplitudes which does nothing for my the frequencies below 1 Hz.

  In the attached png you see 4 screenshots for 2 or 3 command and sampling rate of 300 or 1000 Hz to show you the answers of variable magnitude without opening labview.

  THANK YOU very much for your ANSWERS!

  Chris

  Hello Cameron and thanks for my lenses of compensation.

  I can now proudly present the solution of my problem.

  It seems to be purely a problem of the visualistion information filters through the cluster of the scale.

  After looking in the front panel of the IIR, I suddenly noticed that the "df" of the pole size is changing with the Fs of the input signal.

  For a Fs to 30 Hz, the "df" is 0.03 Hz so you see the curve of the filter with more points, see png.

  For a Fs 300 Hz "df" is 0.3 Hz, so the curve is larger with only 3 points between 0 and 1 Hz.

  For a 1 kHz Fs the df is 0,976 Hz, so there is no point in the graph between 0 and 1 Hz.

  It's strange that for constant Fs, df of this cluster NOT reduced with the increase in the number of samples, as it does in an FFT.

  However, I hope now the filter used now for the curves obtained with the proposed Lynn way and the response of greatness from the filter information fit together.

  Thank you for your support.

  Merry Christmas and a happy new year to all.

  Chris

• NI 6552 Signal Express sampling rate

  I'm generating multiple signals for Signal Express. When I run them, some work well and others not, in other words, for some changes in sampling rate signals! Thera are two options in Signal Express: 1. read file waveform 2 sampling rate. to manually set the sampling frequency. In both cases, the rate is changed when I run the waveform of my 90 MHz to 100 MHz.

  Any ideas?

  Hello

  The internal clock on the 6552 is generated from 200 MHz time base; Is that you can generate the frequency of 200 MHz/N where N is from 2 to a large number which takes up to 47 Hz. The driver for the Board of Directors will force the neerest value, so if you swipe from 90 Mhz to 100 MHz frequency, you will get 100 MHz in all cases. There is more information about the synchronization on the specs here:

  http://digital.NI.com/manuals.nsf/WebSearch/E4C93B141B71ED93862573CC005E8EA1

  Once, you run your project, Signal Express should show you the corced value, you can use this to read the actual frequency generated.

  I hope this helps.

  Juan Carlos

• How to increase the sampling rate in this VI?

  Hi all

  I have recently inherited this mess of a VI and can not figure out how to increase the sampling rate. I tried to change the "ms of waiting ' clock, but it does not add more data points." The main VI, as well as the Subvi, which contains a Daq Assistant to a load cell and LVDT is attached. Any ideas on how to improve the sampling without a complete overhaul would be greatly appreciated!

  Thanks in advance!

  If you are grateful, feel free to give congratulations and mark the topic as resolved.

• With the NI 9205 module Max sampling rate - problems

  Dear friends,

  I develop a project of lv, which makes and control system of engine dyno. The material is CRio-9022 with other cards and also 9205 for AI. There is an encoder for angle attached to the motor shaft with 3600 chatted by Tower as well as an index to indicate the end of a revolution. the output of the encoder is measured by card 9411. The speed of the motor is 1500 rpm. I measure pressure data and couple when I receive a 'tick' of the wheel. This means my sampling rate for pressure and torque each is 90KO/s.

  but I was not successful to lead it. The program is great and I can show them, but I believe that there is a problem in the choice of material for the task. With the data of pressure and torque of the 9205, I also measure other channels for the controller output mass flow and temperatures. So in all I use 8 channels of the 32 available. But only the pressure and torque are acquired at the wheel-driven sampling rate. the rest are acquired about 5 times per second.

  Since the 9025 is a multiplexing ADC, 250K sampling frequency is divided by the number of channels accessed = 250 K/8 = 31 K samples/channel. With this in mind, I decided to acquire data of pressure and torque with each beat 3rd rotary encoder, essentially on 30K samples/s sampling. However, I see a large amount of noise.

  So I decide to average more than 1 second cycles (so the engine runs at about 25 cycles/sec, I averaged over this issue). The resulting pressure and torque graphics do not match with those measured by an oscilloscope in terms of amplitude but the frequency and shape is correct.

  I noticed an interesting feature in the charts. When I pass interpolation between the points, I see several curves made by points instead of a continuous locus of points. Accordingly, I find that the acquisition is slower than necessary, and so there are less number of points sampled as required. These points are not synchronized 25 cycles I have on average and therefore the separate "curves". It is because of the possibility that some points receive a higher number of 'contributions' several times (when you add), the neighbouring points.

  so I conculde that the 9205 is not fast enough to do the job. also noise, perhaps due to crosstalk or gosting when the mux changes channels. the impdences output pressure and the couple are of the order of 10 K ohms.

  the Labview code outline: well, there is a vi FPGA, which takes the rotary encoder ticks and sends a signal to the case of each 3rd tick. The signal contains a 16-bit integer, indicating the number of ticks. This signal is sent to a 1 element FIFO. This fifo is read in a parallel while loop, where it remains awaiting a new element. The while loop bed fifo, where data are available, takes a measure of pressure channel. A node memory of the method is called to provide data according to contained in the index number equal to the number of ticks to signal fifo. Then he adds the current pressure reading to the reading of the memory and stores the sum in the same memory location.  Thus, an array of elements of 1200 is formed, where each elemnt is a sum of the values taken of more than 25 cycles. This memory is transferd to a dma fifo and reading side host. is done similarly to involved couple. host-side the fifo is read and divided by 25 to get the average. This average is displayed on a waveform graph.

  Please check the attached file to get an idea of the problem. Sorry for the long post.

  Please suggest if you understand the problem and suggesstions or solutions.

  
  

• Conflict to sample rate (specified real vs)

  I use NI 9234 to acquire my sensor data using labVIEW 8.6. I have been using labView for only the past two weeks, so please bear with me as my knowledge is so fundamental. I'm reading several channels over time. My problem is when I finished my VI, I discovered the whenever I change my bit in VI code rate, she even more fast (several sampling rate than what I said).

  I've never used a time base external with a data acquisition card, so it's a bit outside my field of knowledge. You may consult the manual to see if it's possible.

  Personally, I wouldn't bother. If you want a lower sampling rate 1,652 kHz, you could always decimating up to a lower rate. For example, if you enjoy at 1,652 kHz and then take each sample 16, you would then end up with an effective 103,25 Hz sampling rate.

  If you want to exactly 100 Hz you could make, because they suggest in the link and use the 'resample waveforms (continuous) .vi"to re - sample data.

• Audition 3.0 how to disable ASIO and the default Sample Rate recording?

  Hi people,

  New here, but I hope someone can help me with a few questions, I'll have with Audition 3.

  Firstly, some background questions.

  I use hearing parallel to a broadcast audio broadcast program called SpotOn,

  This software requires that I run the sound card, a RME Madiface XT in 48 k mode, and that all the outputs that it uses are defined as WDM Windows so that the windows kernel mixer can control them.

  This means that when I use the hearing at the same time, I have to configure it to use the "Audition 3.0 Windows Audio" driver to stop him from taking control of the sound card directly and change setting which prevent SpotOn to see its output.

  The problems I encounter are that hearing itself seems to randomly change mode in the edit window ASIO driver, I suspect that this happens when I import audio data from a key which is from 44.1 to modify for use in SpotOn. This often seems to not only make the outputs of the card its invisible to the windows kernel mixer but also change the sampling frequency of 44.1 sound card and stops work SpotOn.

  The second question I have is that the sampling frequency of default record when I record in edit mode is always 44.1 and if used it again change the map sound 44.1 and causes the same problems, I'd be very keen to know how to change this default to be 48 k if possible.

  Then.

  What I ultimatly looking is...

  1. a way to disable the ASIO drivers in hearing so that it is not only this option is available, and cannot use the Audition 3.0 Windows Sound Drivers.

  2. a way to make the sampling rate 48 k to stop people choosing 44.1 mistakenly when saving default record.

  Any help or advice that anyone can give would be much appreciated.

  Thanks in advance for your comments

  What other pilots ASIO sees your installation of 3 AA? If it's the Madiface one and you use only WDM drivers you can just uninstall the ASIO RME driver?

• Getting the error «for input and output sample rate does not match...» »

  I am trying to create an audio file, but when I click on the record button I get this error:

  «The sampling rate of the audio output devices and audio don't match.» Audio cannot be recorded until this problem is corrected.

  Use the appropriate operating system or control panel of audio device to adjust the sampling frequency of the output devices and audio entries to use the same settings. »

  I have the preferences set to use operating system controls and it seems to work because when I change in the Win 7 Pro CP it adjusts at the hearing.

  So I do not know which need to be addressed. I use the video card as the source and not the speakers thus mean the output must be la carte too?

  Please help, I'm confused.

  Thanks for your time!

  Got an answer to a previous question asking how to record and thought, this could be useful for others.

  rothmannnyc,

  

  February 18, 2013 16:33 in response to talbot

  It takes a Windows setting and a setting of the hearing.

  You will need to go into your settings from Windows recording devices - if you see a speaker in the lower right of your screen, right-click on it and choose recording devices.  Find your audio input source, select Properties, go to advanced.  You should see a drop down menu with options for the quality and frequency of sampling, mono/stereo.  In general, you'll want to 44.1 kHz for audio; 48 Hz for the video.

  At the hearing, go to Edition/Preferences/Audio Hardware and select the same frequency of sampling for output.  I went through the same question a few months ago, and it worked for me.  Good luck!