I am trying to save TDMS files that ideally contain the following:
- 3 xy graphs (each containing two 1d arrays)
- 1 waveform
The problem i'm running into is that when I convert the xy graphs to waveforms, the x-axis is converted to time, which isn't real or useful to me. I've attached screenshots of what the XY graph should look like VS what it ends up looking like with the waveform.
How to I make sure the x-axis is preserved so that I can save to TDMS?
Edit: VI is included & pictures have been updated to better represent my code.
TDMS Waveform Example.vi
I'm working on a project that requires sound output via the Windows audio subsystem, and so I've recently found myself using the LabVIEW "Sound" pallet vi's for the first time in.....well...ever!
The sound output ("Play") vi's allow you to setup and configure an output "task". As part of that configuration, you define a buffer size. Once that sound output task is configured, you effectively write your sound samples into this buffer over time, periodically refreshing this buffer with new audio sample data.
That's all fine and good, but unfortunately it seems like there is no mechanism to query the buffer status and find out if the buffer is about to overflow or underflow. While this might not seem to matter if you're playing a sound file for a few minutes on a machine with lots of RAM, it definitely does matter if you're streaming live audio continuously through the system for several days or weeks.
If you refresh the buffer with new audio sample data at rate that is just slightly faster than the audio card's configured sample rate, then it seems like the buffer will eventually use up all available PC RAM. if you are refreshing that buffer just slightly slower than the audio card's configured sample rate, then it seems like the buffer will eventually underflow and create a glitch in the audio output. And since there is no way to monitor whether the buffer is trending toward overflow or underflow, there is no way to figure out how to adjust the rate at which you feed new audio sample data into the buffers.
Am I missing something here? Is the audio subsystem doing some sample rate conversion that I don't know about? What is the proper way to ensure that the sound playback buffers do not overflow or underflow over an extended period of time?
Since LabVIEW uses only Windows MME driver for sound in and out, I'm looking for an other way to get sound data into LabVIEW with much less latency.
Has someone use ASIO4ALL for audio input?
For sound out I use Midi via MME and the Windows own synthesizer. Its latency is really not high. But the input data comes with some hundred milliseconds lateness.
With MME: I tested on 2 systems. About 350ms latency and not really depending of the system load. I tried also with small sample packages of 600 and 1200 byte/chn and a sound card acquisition rate of 96kS/s. In order to collect this data amount the time what the PC needs should be theoretical 6,25/12,5ms plus data transfer time and some reaction times of the system modules. But the response time of MME seems in-depending on it. It needs always around 320...380ms. And the processor load was very low.
Maybe someone can help me to use a faster software interface (ASIO?).
I have been using a Developers Suite for quite some time and have found some of the VI's in the signal processing tool kit very helpful. I am not involved in video processing at this point; lots of underwater acoustics. There are many VI's I have not used and there are limited examples in many cases. I have not found current resources that discuss DSP applications for most of the Vi's. What resorces are available?