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WAV recorder shifts frequency DAQmx

DAQmx connected to microphone and 1kHz calibration signal.
Real time information frequency and amplitude is correct using all SVXMPL_XXXX.vi 's
but the SVXMPL_WAV Recorder introduces a frequency drop.
 
It is like the 50 kS/s is being written as 44,100 S/s.
 
Editing the sub-vi's to match the write frequency to match a whole division of the DAQms frequency fails.
 
50,000 => 44,100
25,000 => 22,050
12,500 => 11,025
 
Editing the sample rate on the DAQ to match the write frequency to sub-vi frequency also fails since it reverts to the "actual sample rate".
 
This should be really obvious!
 
Please help
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Message 1 of 5
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I found this older forum thread:

"If you really need to have the data at 44.1 kS/s (for example, to output to a standard audio file) try using the "Resample Waveforms" VI in LabVIEW under the "Signal Processing >> Waveform Conditioning" pallette. On this VI you need to choose the interpolation mode; you should choose either "Spline" or "FIR".  Don't use "Linear".  "

The VI that I am using is a SV Toolkit standard item with no editing.

Does the reply infer that LabVIEW will always shift a 50 kS/s sample to a 44.1 kHz Audio RIFF unless you interpolate and resample; that can't be right why not just write to 50 kHz RIFF?

 

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The input

Realtime Input Spectrum

 

The output

Wave Playback Spectrum

 

Message Edited by Midlothian on 10-27-2006 07:22 AM

Message Edited by Midlothian on 10-27-2006 07:26 AM

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I have tested the resampling method and this fails to correct the problem.

It is as if the Signal Conditioning resampling input "dt" is adjusting to maintain the original time x-axis.

The DAQmx remains 50,000 kS/s based  and all Audio Wav vi's and the sub-vi's are 44,100 kHz based with no obvious method of correcting the mis-match.

I have raised an NI problem report by telephone and asked for urgent work-around and eventually a new release of the corrected SVXMPLS WAV  Recorder (DAQmx).vi .

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