Lab: Digital Signal Processing with LabVIEW and SPEEDY-33 DSP Board, Lab Reports of Digital Signal Processing

Download Lab: Digital Signal Processing with LabVIEW and SPEEDY-33 DSP Board and more Lab Reports Digital Signal Processing in PDF only on Docsity! EE-289, DIGITAL SIGNAL PROCESSING LAB November 2005 Introduction To LabVIEW and the DSP Board 1 Overview The purpose of this lab is to familiarize you with the DSP development system by looking at sampling, analysis, and reconstruction of continuous-time signals. We start with the sampling and reconstruction of continuous time signals. We will then look at displays in the time and frequency domains. We also examine the importance of the sampling frequency and its effects on aliasing. 2 DSP development overview 2.1 PC Details The PC is required to run the LabVIEW DSP module which is required to design and implement and download code to (run on) the DSP. 2.2 DSP Board Highlights A self-contained, high-performance, programmable DSP board for signal processing applications. The SPEEDY-33 comes equipped with a Texas Instruments DSP for high speed DSP calculations and Analog to Digital (A/D) and Digital to Analog (D/A) converters. • High-performance 32-bit floating-point digital signal processor (DSP) • 150 million floating-point operations per second (MFLOPS) • 13-ns Instruction Cycle Time • 16/32-bit integer and 32/40-bit floating-point operations • 32-bit instruction word, 24-Bit Addresses • Parallel arithmetic/logic unit (ALU) and multiplier execution in a single cycle • Supports standalone operation The SPEEDY-33, which connects to a PC through a USB host port is programmed in LabVIEW. The program as well as other data such as tables, sounds etc. are stored in the flash memory. Once the FLASH is programmed, the SPEEDY-33 can run in standalone mode, without the PC. The TMS320C33 is one of the SM320C3x generation of floating-point DSPs from TI. The SPEEDY-33 is a self-contained, programmable board for signal processing applications. 2.3 Software The software used in the lab is National Instruments LabVIEW DSP Module; a block diagram-based graph- ical programming package for digital signal processing that will be used for: data acquisition, time domain data display, frequency domain calculation and display, filter implementation, convolution and correlation 1 Figure 1: SPEEDY 33 calculations, speech/signal modeling, and real-time filtering. Using LabVIEW DSP Module along with the LabVIEW Digital Filter Design Toolkit and the DSP board allows you to perform data acquisition, analysis, and filtering, it will also give them the ability to listen to the filtered output. 3 System Setup Follow the instructions below to set up the system correctly. • Log on to the computer • Connect a USB cable between the PC USB Port of the SPEEDY-33 (Figure 2-7) and the USB port of your PC. The SPEEDY-33 is powered by the USB port. Once you connect the cable, the green Power LED (Figure 2-2) on the SPEEDY-33 will light. • Confirm that the Power LED on the SPEEDY-33 is lit. • Connect the speakers to the Audio Stereo Output Port (Figure 2-4) on the SPEEDY-33. • The on-board microphones will be the input to the DSP board. 4 A/D and D/A Conversion The NI LabVIEW DSP module is a block diagram-based DSP development platform that allows the user to quickly set up complex DSP algorithms. The true power of LabVIEW lies in its ability to interface with external DSP devices and / or internal sound cards that are installed on the PC. The developed algorithms are downloaded to the DSP board, which then runs the algorithm in real time environment. In this lab, we look at how LabVIEW DSP interfaces with the A/D and D/A converters on the DSP board. Specifically, we simply connect the A/D converter to the D/A converter so that the DSP plays back audio signal sent to its input. 4.1 Simple Sampling/Reconstruction System 1. Keeping the Surround Mixer open, start LabVIEW Embedded Edition. On the startup screen from the Execution Target dropdown menu select your hardware device. The Execution Target specifies where the code will run. Refer to Figure 3. 2 Figure 5: Functions Palette with the Elemental I/O blocks 5 Figure 6: Finished VI • Sample Rate in Hz: 48000 5. Before we finish, the Analog Input and Output blocks need to be wired together. Hover your mouse over the Left(Right) channel output of the Analog Input block, the mouse curser will change to a wire spindle. Left click and drag the wire to the Left(Right) channel input of the Analog Output block. 6. We require the VI to run continously. To do this, we use a While Loop. Place the While Loop (in the FunctionsStructures Palette) around the two nodes by left clicking on the top left corner and dragging it to the bottom right. The final VI should look similar to Figure 6.The Stop button that appears is wired to the conditional terminal of the loop. This also appears on the front panel. Once the VI is run, this button aborts execution. 7. Save your VI by using the File>>Save As dialog. 8. Your audio input will be the on-board microphones. 4.3 Run LabVIEW Code On The DSP 1. When you press Run, you should see something like Figure 7. 2. What happens when the framesize and/or sampling frequency are changed? 3. Does sampling freq. more affect music or speech? 6 Figure 7: Running Code on the DSP 5 Time and Frequency Displays LabVIEW has the ability to display data in the time and/or frequency domains. 5.1 Time Domain When an analog signal is passed through an A/D converter, its samples are stored as an integer. If we use a 16-bit system, there are 216 = 65,536 possible values that can be stored. These stored values can be displayed either as discrete samples or as a smooth line connecting the samples. We use the Waveform display to set up a time domain display. Switch to the Front Panel and right click anywhere to bring up the Controls palette. From the Graph palette up the Waveform Chart and place it on the Front Panel, as shown in step 1 of Figure 8. Right click on the Waveform Chart and select Properties from the popup menu as shown in step 2 of Figure 8. In the Chart Properties window on the Appearance tab change the Label to Left Channel Time Domain. Specify the settings as shown in step 4 of Figure 8. Deselecting Autoscale will ensure that the Y-axis of the chart does not change with the data values. Minimum and Maximum values of -32000 and 32000 will cover all the values for a 16-bit A/D channel. The Waveform Chart maintains a buffer (History) of values that are plotted on it. The number of points stored can be adjusted by right clicking on the chart and selecting Chart History Length from the popup menu. 1. Connect the left output of the Analog Input block to the Waveform Chart as shown in Figure 9. 2. Run the program and speak into the microphones. You should be able to see your voice, in addition to hearing it. 3. Stopping the program will freeze data on the Waveform Chart. 5.2 Displaying the magnitude of the Fourier Transform 1. To display the frequency content of the audio signal use the Fast Fourier Transform (FFT). Add a second Waveform Chart Display to the Front Panel. Change the label of this Chart to Left Channel Frequency Domain and enable Autoscale for the Y-axis by checking the box in the Waveform Chart Properties window on the Scales Tab. On the Scales Tab switch over to the X axis properties by selecting it in the dropdown menu and change the X-axis range to 0-255. 7