On the audio recording problems

Hi all

I have a problem with regard to save audio files in Labview.

In my program, I acquire the audio signals of the soundcard of the PC constantly and save them in a queue. After that certain prerequisites are activated, these audio signals in the queue is stored in a .wav file. This program works well, except a problem, which is the time lag in this .wav file. For example, I save a sound segment for about 30 seconds, but after the backup, this .wav file is just 21 long s. It contains all of the original recorded segment, but the time axis is compressed, which sounds like played fast. I searched all day and could not find any information about it.

These sounds record and save functions are part of a comprehensive program, and I'm sorry that I can't give a single vi on it. I add the numbers below and I hope it helps. I am also willing to provide any additional information.

PS: I have reviewed the incompatibility caused by approximation of the "t0" (published in http://forums.ni.com/t5/LabVIEW/Time-mismatch-with-Sound-Input-Read-VI/td-p/1385404 ) and compensate for that during the savings phase. (see figure 2)

Thank you very much for the help!

Hi, Vans,

Thank you for your response.

For your questions.

1. I'm using Labview 2011 sp1 on Windows 7/Xp. I tested the program on desktop computers and laptops. I don't know about the types of sound card, but given that this problem occurs on several different computers, I think that this is not the problem of sound cards.

2. for the sound card sampling frequency, in fact I checked that. I read signals of waveform of the 'read.vi of sound entry' and the term 'dt' is equal to 1 on the sampling frequency. So I think that the sampling rate of the sound card is correct.

In fact I have just solved this problem by myself, but the solution is pretty weird. I think there are two reasons that caused this problem.

1. when I receive tones of "read.vi of sound input", I first save them in a queue. Once the acquisition is completed, I save all the signals in the queue to a .wav file. But I'm going to do to overcome the overlapping sound and the gap caused by the "t0" reconciliation, a waveform signal re - organize before saving it, as below. According to my knowledge, I think that this 'dt' in the resulting waveform (indicated by the arrow) must be the same as before, but in fact it is not, and it becomes 1.

2. Thus, to solve the first problem, I manually change the term "dt" once again as below. After the change, I'm sure that the "dt" is back to the correct value. But then is the strange thing is I saved the same signal waveform in two different ways, using "his write.vi file" and "file its writing simple.vi" respectively. The first method is the one I have used before and it caused the problem I posted here. The second works. In my program, I put as 16 kHz, 16 bits/sample, and 1-channel sampling rate. The file .wav from the first method has a strange flow 705 bps and the wave from the second method file has the right bitrate which is 256 bits/s.

I tested the program of hundreds of times and each time he acted like I described above. I don't know if anyone has ever faced the same problem as me, but I hope that my solution can be useful for others. And I also hope to be useful to improve your LabView products.

Thank you for your time.

Best regards

Sun87

Tags: NI Software

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