Summary: You will write code to compute the autocorrelation or autocovariance of an input signal. Then you connect a microphone to the DSP and write code to detect the beginning of a speech segment. Finally, you will combine the two programs and compare results with MATLAB.
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The sample rate on the 6-channel
DSP boards is fixed at
Compute the autocorrelation or
autocovariance coefficients of
The next step is to use a speech signal as the input to your system. Use a microphone as input to the original thru6.asm code and adjust the gains in your system until the output uses most of the dynamic range of the system without saturating. Now, to capture and analyze a small segment of speech, write code that determines the start of a speech signal in the microphone input, records a few seconds of speech, and computes the autocorrelation or autocovariance coefficients. The start of a speech signal can be determined by comparing the input to some noise threshold; experiment to find a good value. For recording large segments of speech, you may need to use external memory. Refer to Core File: Accessing External Memory on TI TMS320C54x for more information.
Finally, incorporate your code which computes autocorrelation or autocovariance coefficients with the code which takes speech input and compare the results seen on the oscilloscope to those generated by MATLAB.
"Doug course at UIUC using the TI C54x DSP has been adopted by many EE, CE and CS depts Worldwide "