What is the minimum response of analog input, through DSP online, output analog time?

Hello experts!

I want to know if it is possible to get a very quick response latency (~ 1 ms) sound recording (analog input), through online registration (DSP online), the presentation of his (analog output) processing, by using the DAQmx programming codes. My system of NEITHER includes NOR SMU 8135, SMU 6358 DAQ Multifunction controller and SMU 5412 arbitrary signal generator. I also have access to the latest version of Labview (2015 Version) software.

My project is on auditory disturbances, which inovles record vocalizations, manipulating the recorded vocalziations and then present the manipulated vocalizations. My current idea of how to achieve this fact triggered output voltage after reading the input using DAQmx Read samples. DAQmx Read output is filtered online and then passed as input for the DAQmx writing for analog output. For purposes of illustration, examples of code are presented below. Note for simplisity, codes for the trigger part are not presented here. It's something to work in the future.

My question here is If the idea above should be reaching ~ 1ms delay? Or I have to rely on a totally different programming module, the FPGA? I am very new to Labview so as to NEITHER. After reading some documentation on FPGA, I realized that my current hardware is unable to do so because I do not have the FPGA signals processing equipment. Am I wrong?

Something might be important to mention, I'm tasting with network (approximately 16 microphones) microphones at very high sampling rate (250 kHz), which is technically very high speed. Natually, these records must be saved on hard drive. Here again, a single microphone is shown.

I have two concerns that my current approach could achieve my goal.

First, for the DAQmx Read function in step 2, I put the samples to be read as 1/10 of the sampling frequency. It's recommended by Labview and so necessary to avoid buffer overflow when a smaller number is used. However, my concern here associated with the latency of the answer is that it might already cause a delay of 100 ms response, i.e. the time to collect these samples before reading. Is this true?

Secondly, every interaction while the loop takes at least a few tens of milliseconds (~ 30 ms). He is originally a State 30 late?

Hey, I've never used or familiar with the hardware you have. So I can't help you there.

On the side of RT, again once I don't know about your hardware, but I used NOR myRIO 1900, where he has a personality of high specific speed for the RT where I can acquire the kHz Audio @44 and process data. Based image processing is ultimately do the treatment on a wide range of audio data you have gathered through high sampling frequency and number number of samples as permitted by latency, please check this .

I lost about 2 weeks to understand host-side does not work and another 2 weeks to understand the even side of RT does not work for online processing (real time). Then, finally now I'm working on FPGA, where the sampling rate is 250 kHz (of course shared by multiple channels).

The complex thing with FPGA is coding, please check if the filter you want is given below as labview automatically generates some codes of some filters.

Most of them will work in 1 SCTL IE if your target has 40 MHz clock algorithm will run in 25 ns. That's what I was looking for, I hope you

See you soon... !

Tags: NI Software

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