Sampling frequency do not match

Similar Questions

• 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!

• Why my sample rate does not match the output of timestamps in the waveform?

  Hello

  I run a simple application to read the data of two pressure sensors output signals 0 - 5V to a NI9215 module, and one connected to the 9237 module load cell. They are housed in the 9172 chassis.

  I am new to DAQ and labview, and I find it difficult to reconcile the sampling frequency that I put in the sample clock and the apparent rate data (according to the timestamps in the waveform that I output to a text file). For example, if I ask 100 Hz rate (and 10 samples to read), the data appear to sample at 1612,9 Hz. If I ask the sampling frequency of 1000 Hz, outgoing data is 1612,9 Hz to 20 kHz, the data came out to 25 kHz.

  Can someone tell me to trust the timestamps given in the waveform that is written in the text file, and if there is a way to check this?  If this timestamp is correct, how can I force the application of sample data at the requested speed?

  As a secondary issue, in my attached VI, you can see that I have an attached to an array of construction shift register. I can't understand how to initialize the array outside of the loop as it clears the table before the next time I run the program. Any advice?

  My VI is attached.

  Thank you

  Claire.

  Hi Marc, thanks for the quick response and the right explanation. It's all much more clear now.

  Have a great weekend,

  Claire.

• Sampling frequency of input/output does not match.

  Hi all

  I recently moved to audacity to audition and I have a small question: I have a Zoom h2n who worked well with Audacity, but when I tried my first recording with audition, I was told that my input and output sample rates do not match. I've already looked around and discovered that I need to change the format of my speakers to match the entrance to 44.1 I my h2n zoom fixed to however I was wondering if there was when even just ignore it. What the output is important in any case, my fault for my speakers is quality studio 24 bit 48000. Did I miss an option in the settings of the hearing? What I really need to drop out of the enclosure to 16 bit, 44100?

  Thanks in advance for any help.

  Peace.

  If you use the Zoom as an interface to audition then you really must match your session, the sampling frequency of the aufdition otherwise interface is going to be on-the-fly resampling

  So when you create a new session you assign at the same rate as the Zoom or vice versa

  You can also try to edit > Preferences > Audio hardware and check or uncheck the "attempt to force material to document sampling frequency.

• DT of a waveform is not equal to the sampling frequency

  I wrote the attached VI to take data from two different channels, 1 a cell, and the other being an encoder. I have my calendar synchronized with the entry load cell. The question I have is that at different sampling rates high for the program, does not change my dt in my waveform data.

  If I understand correctly the dt of the waveform should coincide with the sampling frequency. Am I missing something?

  Thanks in advance for your help

  
  

• Audio samples of input and output rates do not match.

  If I get an error that supposedly get a lot of people, I think I know the problem of fixing, but I think that it is impossible with the microphone that I use but I want to make sure it's just the microphone that I use and nothing else. So I went in "reading devices" and "recording devices" check if they were both on the same sampling frequency, but I could change the sampling frequency of spekers in properties, but not the MIC... the microphone that I use is the PS2 eye cam, it works very well I had to download "CL eye test" to get it running, but it works , by default the format its deadlock is: 4 channels, 16000 Hz (tape recorder).

  The box is grayed out and I have no idea how to do to change the frequency of sampling for it... so now I can't use the program pretty at all to save

  I am sure that you will not be able to change because it is defined by the hardware built into the microphone. If you want to use with the hearing, you should make sure that windows sounds and hearing are configured for 16 k. Otherwise if there is any software that came with that allows you to record must be able to transfer the audio to hearing to change.

  However, Eye Cam is really a toy. If you are seriously in the audio recording, you'd be better off getting a good microphone and audio interface.

• Error #3606: Sampler 0 format does not match the format of texture.

  Hello

  I try to use a cube texture but I got this error:

  Error #3606: Sampler 0 format does not match the format of texture.

  with this fragment of AGAL:

  ft0.w MOV, FC0 .x

  FT1.xyzw Tex, v0 .xzy, fs0
  Mul ft0.xyz, fc1.xyz, ft1.xyz
  MOV OC.xyzw, ft0.xyzw

  Apparently glsltoagal does not write the < parameters > fs0 is a problem or it has default values?

  Can I create textures to Cube in a specific way?

  I did a mistake using a faulty method, it was

  createCubeTexture

• Xincrement is not agree with sampling frequency

  I use two PXI-5114 scanners that are being synchronized. I am taking 5 seconds worth of data.  I have = 1e3 sample rate and record length = 5e3. I get only about 1.3 seconds worth of data.  I looked at the actual length of the record and the actual sampling frequency outputs and they said the same thing as the façade made. I looked at and then the info out of the Cluster.vi of Fetch NiScope Multi Wfm and he had the same record length, but the value of Xincrement is 262e-6. This fits with what I get.  Any ideas?

  Thanks in advance.

  Hi AT1,.

  Thanks for posting. What you see, this due to the fact that the minimum sampling rate of the PXI-5114 is 3.81 kech. / s. The digitizer will force any value less than 3.81 kech. / s up to this value, so a record of 5000 samples take 1.31 seconds to acquire. If you are looking to acquire 5 seconds worth of data, I recommend to increase the length of record about 19 050 samples, which should be about 5 seconds worth of data. Let me know if this would work for your application.

  Kind regards

  Joe S.

• Alternating - Signal Express sampling frequency

  Hi, Im trying to get data using Signal Express + NOR USB-6211, I put the sample rate to 44100 Hz. and everything seems to be ok, but when I checked the data I realized this time column does not increase in a period time constant (ie. sample FREQ not fixed).

  For example:

  1 / (Time (1) - Time (2)) = 43478,26 Hz.

  1 / (Time (2) - Time (3)) = 45454,54 Hz.

  and it constantly alternates between the two frequencies. I have to perform an FFT and Im not sure what is the actual frequency and which continues.

  I appreciate any help on this issue.

  Kind regards.

  This seems to be a bug that NEITHER should be aware of if they are not already.

  If you set the frequency of sampling to 44100, the actual sampling frequency will be (20 MHz / 453) or ~44.150 kHz.  The USB-6211 case has a basis of TIME 20 MHz (±50 ppm) and this clock is divided by an integer to derive the sample clock.  The sampling frequency is always be forced to match to the top if you choose a rate that can only be achieved with a wide divider.  Regardless of what the file shows, map DAQ itself samples at a uniform rate (unless you use an external clock, or something like that...).

  As for the behavior in the column duration... I know exactly how you look at it but I get the same behavior is written my data using 'absolute time' for the time column to an ASCII file (or by using a file LVM, which always uses the absolute time).  With the help of 'on time' however gives me a correct result.  I can't speculate on the cause... it seems likely to me that there is a comma floating rounding error (absolute time is in terms of seconds since 1904, so the software works with a large number at the same time requiring great precision).

  Best regards

• How to acquire with NiScope at different sampling frequencies and lengths Records?

  I need to acquire the data of 2 channels of the NI PXI-5114 map two different sampling frequencies high, at the same time. Also, I put 2 different record length. Is this possible?

  I understand that 'Vertical' settings can be configured for individual chains because the function 'Vertical niScope Configure' has 'channels of entry with which we can assign the desired channel. But for horizontal settings such as "min sampling rate" and the record min length, I could not find such an option to specify the channel. Would it not common to both channels?

  I hope that the device is capable of simultaneous sampling and therefore channels can be configured individually to different sampling rate.

  Hi AJ_CS,

  Why do you have to be distinct from sampling frequencies on channels separated from the digitizer even?  What different sampling rate do you want?

  But for horizontal settings such as "min sampling rate" and the record min length, I could not find such an option to specify the channel. Would it not common to both channels?

  You do not have an option to configure the settings of hoirizontal on a channel by channel basis because this concept does not exist in the traditional sense of the use of a scope.  Compatible with the concept of IVI, an oscilloscope traditional benchtop will have only a button or a set of buttons for setting the parameters of synchronization of the unit.  There is therefore no horizontal configuration to separate channels on the scanners NOR.

  I hope that the device is capable of simultaneous sampling and therefore channels can be configured individually to different sampling rate.

  Similar to a traditional benchtop oscilloscpe, the device is capable of simultaneous sampling.  But as mentioned above, the channels can not be configured for different sampling frequencies high.

  However, you can ignore data that you think is not relevant.  For example, if you assign 100MS/s CH0 and CH1 to 50 MS/s, then you throw all other samples.

  Alternatively, you can use separate scanners (a channel on each digitizer) and configure them to taste at different rates.  You can set frequencies of sampling on scanners NOR separated and even synchronize them with TClk.

  -Andrew

• How to acquire the signal to very high sampling frequency

  Hello world

  My name is Luke Ho. I am trying to acquire the signal with Labview (Sthelescope). The signal comes from sensor acoustics, then filters and amplifiers to adapt to ADC rank (0 - 5V). Thus, the maximum frequency of the signal is 40 kHz.

  According to the Nyquist theorem, I sampled at least 80 Khz signal.

  Is there a sampling frequency devices like that? or y at - it another way of better? I used the Arduino before, but it was about 10 kHz.

  I need your advice.
  Thank you all and have a nice day.

  holucbme wrote:

  Thanks for your recommendation

  But is it possible without USB Data Acquisition, it is quite expensive for me.

  This is the cheapest option to NEITHER. I tried to look for options to other companies, but more I found in the same price range, or not answering is not your condition of sample rate.

• AI PXI-6255 sampling frequency

  Hello

  We use a simulated PXI-6255 device and with the LabVIEW example, apparently, that we can achieve a sampling rate of 20 kHz to 50 channels without warnings or errors.

  The spec is the number max of 20 kHz channels must be approximately 38 channels (750000 / 20000).

  
  
  
  
  
  
  
  

  1.25 single channel of MECH. / s multi-channel of 750 ksps / s (aggregation)

  The simulation cannot not representative of the unit or the real PXI-6255 does support a higher rate that we expect?

  Hi jharris66,

  It is not a limitation of the driver to the sampling frequency, so a simulated device reports any error in this case.

  In this example, I think that the limit of 750 kHz comes to the multiplexer.  I don't expect to see the same error messages with the real device, but your accuracy will probably not to spec because not allowing enough time settling between the channels.

  The limit of 1.25 MHz is the maximum ADC sampling frequency.  You can probably exceed this number somewhat, but once you go too far beyond 1.25 MHz you will eventually receive a synchronization from the hardware error when the ADC cannot follow.

  Thus, the specification is still technically 750 kHz for channels multiple acquisitions.  You can probably run at a higher rate than this (even on real hardware), but I wouldn't say that it is supported.

  Best regards

• How to change the DAQmx sampling frequency

  Hello

  I'm trying to: record streaming channels (acceleration 21 and 1 tension) using a DAQmx task, then convert the data to a PDM file. The program files and output to the TDMS file very well. The issue I'm having is that I can't change the sampling frequency. I want to record 500samples/s and I can not get the "real sampling rate" of change of 1651.61samples / s. I am trying to use the clock to do this and I succumbed. I also tried to change the settings of "Timing" in the task without a bit of luck. Here is a screenshot of the .VI I created. I've also attached a copy of the file VI. Any help would be greatly appreciated!

  

  Thank you

  Tony

  You will need to provide the model of your device. You can also look in the sheet/manual to see what the real supported sampling rates. Some devices have limited rates.

• How to specify the sampling frequency? Must use "measurement &amp; Automation Explorer '?

  I use to measure the input current analog OR cDAQ-9171 (chassis only location USB) and NOR-9207. I have 2010 NOR-installed DAQmx and LabVIEW.

  How can I specify the sampling frequency?

  If I use M & A Explorer to create the task, I can specify the flow rate (Hz) on the Configuration tab-> sync settings.

  For the acquisition of data NOR, it is mandatory to use M & A exploring?

  If I don't want to use M & A Explorer, how can I specify the rate (Hz)?

  Hello

  You can specify the sampling frequency with "DAQmx Timing.vi" located in the function palette DAQmx (read context-sensitive help on how to use wisely).

  You do not have to use M & A exploring (MAX) to create a task.

  A simple and quick way is to use DAQ Assistant (same configuration as in MAX) to configure your measurement.

  Another is to use blocks of DAQmx function to manually build your application code.

  In my experience Assistant DAQ is ideal for simple tasks (one measure), with regard to the more complex measures (synchronized the analog and digital inputs).

  I tend to use function blocks because they give you more freedom about code execution.

  Note: You can also build DAQmx code from a wizard configured DAQ task.

  Best regards

  Matej

• Elapsed time on graph does not match indicator sound

  Hello

  I created a vi that displays three waveforms on a graph whose axis x shows the time elapsed since the vi was launched. The data is written to an excel spreadsheet. Everything works as expected with the exception that column of time in my file does not match the time that is displayed on the x-axis of my table. The time in my file only shows increments of 0.1 second, with the same sample for every ten samples, so that the time on the graph increases in increments of 0.001 (accuracy I had put in) and is unique for each data point in amplitude. I have attached my output file for reference. The column 'X' is both.

  I am collecting data at 100 Hz and 10 samples of reading both using the continuous sampling mode. Without my loop for the values of x, I get a sample of time on every ten - what I understand why, given my sampling rate; with it, I get this same sample for all ten samples (again, I think I understand why, but foolishly hoped that might solve the problem). What I don't understand is why the time on my table is incrementing differently time data, I wrote to my folder, since both are from the same source. Anyone would be able to help me understand that, if I can try to solve my problem?

  I have also included my vi. I use USB 6229 OEM.

  This is because you write 10 copies of t0 to the worksheet

  Try this little maodification and see if that helps