I16 array to sound output

[dwfox1977]

Anyone who can help, 

I am attempting to play a signal to verify the data we have recorded into a binary file.  At present, I have converted the binary data to I16 data and stored each channel into a seperate array for data processing.  However, I would like to 'play' the array using the audio vi's in labview 2009.  Here is what I have...

1)  I16 array

    a) 250k data points representing one second of recording data within the array

    b) recording rate was at 250kHz

2)  I have filtered the data with a Butterworth digital filter and verified the filter with the frequency response curve

Here is what I need:

1)  I need to play the audio using this 'recorded' array.

2)  I need to understand how the vi inputs should be arranged (sound output configure.vi)

    a) sound format (sample rate, number of channels, bits per sample)

    b) number of sample per channel

3) how to format the data to be accepted by the 'sound output write' vi.

As of now, I have attempted to arrange the inputs and configurations as I anticipated they should need to be arranged.  However, this is not working.  If anyone has any knowledge along these lines and willing to share, it will be appreciated.

Thank you for your help,

David

[zenthoef]

Just out of curiosity, is there a reason you aren't using the "Play Waveform" express VI? I've used it before with good results and it was much easier to configure than using the individual sound VIs.

[dwfox1977]

With the express vi, I get the same result.... an annoying buzz.  I am able to save the filtered sound file correctly with Matlab just not Labview.  So, I figure that I have the data bit representations correct since it works in the other program.  Nonetheless, listening to the file would help me verify the filters I am using in Labview and thus remove any speculation.

Unfortunately, I think my troubles are self induced via lack of knowledge.  This is why I was hoping to verify how to wire up the inputs for the parameters of the data I have.  If it is possible to accomplish with the express vi, I am fine with using that as well.

Basically, my program loops around picking off one second of data on each itteration (250k samples).  Then processes the data with filters, sound, trending data, and some correlations.  After several minutes of data are processed the results are then passed into another algorithm that I have developed to show the results based upon the data processed.  The results seem resonable.  However, the only true way to verify the initial processing is to simply listen to it.

David

[Coq_Rouge]

You must down sample your signal to a sample rate that your sound card can axcept. This is typical 44100 samples pr second