ProjectMekonnen

Digital Signal Processing Applications Using C6713 DSK

Communications Laboratory – University of Kassel

Project Work  by Zemene Walle Mekonnen

Supervisor: Dipl.–Ing. Thomas Edlich

Project Description DIGITAL SIGNAL PROCESSING APPLICATIONS USING C6713 DSK Digital Signal Processors (DSP) are designed for real-time digital signal processing. Using DSPs offers great flexibility, reconfigurability and easy debugging capabilities. In this student project, a first introduction to simple DSP applications shall be given to enable future users quickly getting familiar with the DSP. The work is based on the DSP Starter Kit C6713 DSK by Texas Instruments, which already contains a high-quality 24-bit stereo codec including several audio connectors and an onboard JTAG programming interface via USB. The development environment includes the programming and debugging tool Code Composer Studio. The first part of this student project is to get familiar with the DSP Starter Kit and the development

environment.

This

contains

the

usage

of

pre-programmed

modules

(eXpressDSP) and the configuration of the real-time operating system DSP/BIOS. After this, several baseband signals shall be generated with the DSP and put out using the audio interfaces. Different visualization and interaction capabilities of the integrated development environment shall be introduced. Finally, examples of real-time signal processing shall be implemented. A detailed test of the generated programs and a well-documented description of  the implementation complete this work. Activities:

30% theory, 50% programming, 20% tests and measurements

Prerequisites:

basic knowledge of programming in C/C++, willingness to work  autonomously, knowledge of programming microprocessors may be helpful

Supervisor:

Thomas Edlich

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Declaration To the best of my knowledge I do hereby declare that this documentation is my own work. It has not been submitted in any form of another degree or diploma to any other university or institution of education. Information derived from the published or unpublished work of  others has been acknowledged acknowledged in the text and a list l ist of references is given. _____________________ Zemene Walle Mekonnen th

Kassel, 27 February 2009.

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Table of Contents Abstract ........................................... .................................................................. ............................................. ............................................ ............................................ ...................... vi Acknowledgments Acknowledgments.......................................... ................................................................ ............................................. ............................................. ............................ ...... vii 1

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Introduction Introduction ........................................... ................................................................. ............................................. ............................................. .............................. ........ 1 1.1

Why DSP? ........................................... ................................................................. ............................................. ............................................. .......................... .... 1

1.2

Basic Elements of a Real-Time DSP System .......................................... ............................................................. ................... 2

1.3

Required Tools ........................................... .................................................................. ............................................. ......................................... ................... 3

1.4

Hardware and Software Installation ............................................. ................................................................... .............................. ........ 5

Sampling and Reconstruction Reconstruction ...................................................... ............................................................................. ................................... ............ 6 2.1 Talk-through Using Polling........................................... ................................................................. ............................................. ....................... 6 2.1.1 Creating a Project .......................................... ................................................................. ............................................. .............................. ........ 7 2.1.2 Code Generation and Options .......................................... ................................................................ .................................. ............ 9 2.1.3 Building and Running a Project ........................................... ................................................................. ............................ ...... 10 2.1.4 Plotting with CCS.......................................... ................................................................. ............................................. ............................ ...... 12 2.2

3

Support Program/Files Consideration .......................................... ................................................................ ............................ ...... 14

Periodic Signal Generation......................................... ................................................................ ............................................. ............................ ...... 19 3.1 Sine Wave Generation........................................... .................................................................. ............................................. ............................ ...... 19 3.1.1 Variable Amplitude Sine Wave Generation ........................................... ..................................................... .......... 21 3.1.2 Variable Amplitude and Frequency Sine Wave Generation ............................ ............................ 23 3.2

4

Triangular Waveform Generation ............................................. ................................................................... ............................... ......... 25

AM Signal Generation ............................................ .................................................................. ............................................. ................................. .......... 27 4.1

Amplitude Modulation (AM) ............................................ .................................................................. ....................................... ................. 27

4.2

DSB-WC Signal Generation ............................................. ................................................................... ....................................... ................. 30

4.3

DSB-SC Signal Generation ........................................... ................................................................. ........................................... ..................... 32

4.4

SSB Signal Generation .......................................... ................................................................. ............................................. ............................ ...... 34

Conclusion and Further Work ........................................... .................................................................. ............................................. ............................ ...... 40 Appendix ............................................. ................................................................... ............................................. ............................................. ....................................... ................. 41 FIR Filter Coefficient Generation Using Matlab SPTool .................................................... .................................................... 41 References ........................................... ................................................................. ............................................. ............................................. ....................................... ................. 43

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List of Figures Figure 1.1: Basic functional blocks of real-time DSP system [3]. .............................. ............................................. ............... 3 Figure 1.2: TMS320C6713 DSK board. .............................................. ..................................................................... ...................................... ............... 4 Figure 1.3: “C6713 DSK CCS” and “C6713 DSK diagnostic utility” shortcuts. ...................... ...................... 5 Figure 2.1: A talk-through system [2]. .................................. ......................................................... ............................................. .............................. ........ 6 Figure 2.2: Talk_through.c using polling [1-2]. ........................................... ................................................................. .............................. ........ 6 Figure 2.3: CCS project window for new project creation. ....................................................... ....................................................... 7 Figure 2.4: Project file window. ............................................ ................................................................... ............................................. .............................. ........ 8 Figure 2.5: Project Build Options window for the compile. .......................................... ...................................................... ............ 9 Figure 2.6: Program compilation, linking, and execution [3]. ............. .................................... .................................... ............. 10 Figure 2.7: Modified talk_through.c [1]. .......................................... ................................................................ ....................................... ................. 12 Figure 2.8: CCS Graph Property Dialog (a) time-domain plot; (b) Frequency-domain plot. .. 13 Figure 2.9: Partial listing of c6713dskinit.c [1]........................................................... ....................................................................... ............. 15 Figure 2.10: Partial listing of  Vectors_intr.asm [1]. .......................................... ............................................................... ..................... 17 Figure 2.11: Generic linker command file (c6713dsk.cmd ) [1]. .............................................. .............................................. 18 Figure 3.1: Source file for sine wave generation using lookup table ( sine_lookup.c) [1]. ...... 20 Figure 3.2: Slider GEL file that represents gain.gel [1]........................................................... .......................................................... 21 Figure 3.3: Modified sine_lookup.c [1]. ............................................ .................................................................. ....................................... ................. 22 Figure 3.4: A slider representing Amplitude (Gain). ................................... ......................................................... ............................ ...... 23 Figure 3.5: gainFreq.gel [1]. .......................................... ................................................................ ............................................. .................................... ............. 23 Figure 3.6: Source file for variable frequency generation (sine_varFreq.c ) [1]. .................... 24 Figure 3.7: Triangular wave form with period T. .............................................................. .................................................................... ...... 25 Figure 3.8 Source code that generates a triangular t riangular waveform (triangle.c) (tr iangle.c) [1] ......................... ......................... 26 Figure 4.1: Amplitude modulation (a) modulating signal, (b) carrier signal, (c) modulated signal ............................................ .................................................................. ............................................. ............................................. ....................................... ................. 28 Figure 4.2 SSB signal generation using filtering method. .................................. ....................................................... ..................... 30 Figure 4.3: Amplitude modulation source code ( DSBWC.c) [1]............................................ ............................................. .. 31 Figure 4.4: AM.gel [1]. ............................................ .................................................................. ............................................ ........................................... ..................... 31 Figure 4.5: Program file for DSB-SC signal generation ( DSBSC.C ). ). ...................................... ...................................... 33 Figure 4.6: A snapshot of the spectrum of the (a) input i nput signal ( inBuffer ) , (b) output signal (outBuffer ). ). .......................................... ................................................................. ............................................. ............................................ ................................ .......... 34 Figure 4.7: Functional sketch of FIR filter. .......................................... ................................................................. .................................... ............. 35 Figure 4.8: Source code for SSB signal generation using filtering method ( SSB.C ). ). .............. 36 iv

Figure 4.9: (a)Coefficients for low pass FIR filter ( LPfilt.cof ), (b) Spectrum of the filter. ..... 37 Figure 4.10: (a) Coefficients for band pass FIR filter ( BPfilt.cof ), (b) Spectrum of the filter. 38 Figure 4.11: (a) Generated DSB-SC signal (b) SSB signal generated after a band pass filter. 39 Figure A.1: statup.spt . .............................................................................................................. 41 Figure A.2: Low pass filter design specifications. ................................................................... 42

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Abstract This manuscript is intended to introduce the reader to the Texas Instruments TMS320C6713 DSP Starter Kit (called C6713 DSK for short), which can be used to implement real-time DSP algorithms. It is assumed that the reader has basic knowledge of C/C++ Programming language. The documentation starts with a brief introduction about DSP, followed by necessary software and hardware installations. In Chapter 2 the tools (hardware and software) that are accompanied with the C6713 DSK package are introduced with simple programming examples. In Chapter 3 the generation of periodic signals using the C6713 DSK is discussed. Specifically the generation of sinusoidal and triangular waveforms is considered. Chapter 4 covers the generation of different amplitude modulated (AM) signals. The AM signals that are discussed include: double-sideband with carrier (DSB-WC), double sideband-suppressed carrier (DSB-SC), and signal-sideband (SSB).

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Acknowledgments First of all I would like to thank my project advisor Dipl.-Ing. Thomas Edlich, for giving me the opportunity to work on the C6713 DSK, for his feedback, comments, suggestions and most of all for his kindness. I would also like to extend my sincere thanks to all of the members of the Communications Laboratory (Comlab) at the University of Kassel for their assistance. I would also like to express my appreciation to the members of “DSP Related Discussion Group”, who have helped me in tackling the problems that I faced when I was preparing this manuscript. Finally, I would like to thank my family and friends for their help and support.

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1 Introduction Signals can be divided in to three categories: continuous-time (analog) signals, discretetime signals, and digital signals.   Analog signals are signals that represent physical quantities with a function, which is continuous both in time and amplitude.  Discrete-time signals model physical quantities with a function, which is continuous in amplitude but

defined only at some specific time instances. On the other hand  Digital signals are defined for a particular time instances as discrete-time signals but they take only discrete amplitude values, so they are discrete both in time and amplitude. Signal processing systems may also be classified along the same line as signals . Digital systems are those for which the input and output of the system are digital signals [4].   Digital Signal Processing (DSP) is therefore concerned with representation of physical

quantities using digital signals and the processing of these signals to extract information from them.

1.1 Why DSP? There are lots of reasons that Digital Signal Processing technique is preferred over analog signal processing technique, which uses active and/or passive devices to process a signal [3]. Some of the advantages of a DSP system over the analog counterpart are:

Flexibility: It is easy to change and/or upgrade a DSP system by modifying the software that has been used to implement the specific algorithm on the same hardware. But in the analog counterpart we need to change the whole/part of the circuitry that implements the signal-processing algorithm if we want to modify or upgrade the system. DSP systems, due to these flexibility characteristics, can also be used to simulate signal processing that will be implemented on an analog system before constructing the analog hardware. This enables the analog system designer to have experimental flexibility without wasting engineering or economic resources.

Complexity: Sophisticated applications can be implemented by using power efficient (less complex) DSP algorithms.

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Reliability: The DSP hardware components (e.g. memory, logic) are resistant to changing environmental conditions or aging of electronic components than the analog counterparts.

Cost: Due to the advent of technology, which results in reduction of cost of digital computers, it is possible to implement a cost effective DSP systems. 1

Due to these and other reasons Digital Signal Processors (DSPs) are used for a wide range of applications from communication and control to speech and image processing. Application embedded DSPs are found in cellular phones, fax/modem, disk drives, radio, printers, MP3 players, high-definition television (HDTV), digital cameras, and so on [1]. Having said some the advantages of DSP systems, it is worthwhile to mention some of  their limitations. For example, the bandwidth of the DSP system is dependent on the sampling rate and hardware peripherals, initial costs of the DSP system may be expensive when large bandwidth signals are involved [3].

1.2 Basic Elements of a Real-Time DSP System DSP applications can be divided as non-real-time and real-time. Non-real-time applications are those which process signals that have been digitized and stored before the signal processing action. In other words the result of non-real-time applications is not needed at the time the signal collection and digitization is performed. But in real-time signal processing it is necessary to finish a given task with in a certain time frame work; if  not, the system couldn’t operate appropriately. The basic functional blocks of a real-time DSP is illustrated in figure 1.1. The Analog to Digital Converter (ADC), in the input channel, converts an analog input signal in to a digital signal so that it can be processed digitally using the DSP hardware. The Digital to Analog Converter (DAC), the counterpart for ADC in the output channel, changes the output of the DSP hardware, which is a digital signal, into an analog signal. The amplifier at the input channel represents proper electronic sensor that converts pressure, temperature, or sound into an electrical signal [3]. An example of the input

1

The acronym DSP is used for Digital Signal Processing and Digital Signal Processor interchangeably, the meaning should be clear from the context.

2

amplifier is a microphone. The amplifier at the output channel performs the reverse action of the input channel amplifier. An example of the output channel amplifier is a headphone. The anti-aliasing filter at the input channel is used to cutout signals with a frequency component above the Nyquist frequency (one half of the sampling frequency). Otherwise we can’t distinguish signals having frequency above the Nyquist frequency with signals having lower frequency (aliasing effect), as our sampling rate is too small to take enough sample of the signal above the Nyquist frequency. For a detailed description of the functional blocks of figure 1.1 the reader is referred to [3].

Input

Anti-aliasing Filter

Other digital systems

ADC

Amplifier Input channel DSP Hardware Output channel

Output

Reconstruction Filter

DAC

Other digital systems

Amplifier Figure 1.1: Basic functional blocks of real-time DSP system [3].

1.3 Required Tools The main objective of this manuscript is to enable the reader to get to know the DSP Starter Kit (DSK), which can be used to implement a real-time DSP. Specifically the TMS320C6713 DSK (called C6713 DSK in short) is used.

3

The following tools are required to perform the experiments listed in the manuscript: •

TI’s DSK package. It includes:

- A board that contains the floating point DSP (C6713) and built in 32-bit stereo codec (TLV320AIC23) for input and output (I/O) support. See figure 1.2 for the photo of C6713 DSK.

- Code Composure Studio (CCS): an Integrated Development Environment (IDE), which contains compiler, assembler, linker, and debugger that provide necessary software support.

- A Universal Synchronous Bus (USB), which is used to connect the DSK board to a PC.

- An adaptor with 5V output that is used to power up the DSK board. •

A Personal Computer (PC): make sure that you have Windows 2000 or Windows XP operating system running on to your PC.

•

An oscilloscope, signal generator, Microphones, and speakers.

•

The CD included with this manuscript that contains the soft copy of the examples that are listed in the manuscript.

Figure 1.2: TMS320C6713 DSK board.

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1.4 Hardware and Software Installation Install the CCS from the CD-ROM included with the DSK package on to your PC. Throughout it is assumed that you have installed it in the folder C:\C6713. You can install it in any folder you like it to be, but take this into account in following the instructions that are listed in the manuscript. After successful installation two shortcuts “C6713 DSK CCS” and “C6713 DSK diagnostic utility”, shown in figure 1.3, will appear on your desktop (If  you haven’t disabled the option “put a desktop shortcut” during installation). Copy the files from the CD, which is included with this manuscript, to the directory C:\C6713\myprojects.

Figure 1.3: “C6713 DSK CCS” and “C6713 DSK diagnostic utility” shortcuts.

When you are ready: •

Plug in the 5V power supply, which is included with the DSK package, on the DSK board to power it up. Verify that the +5V LED is “ON”.

•

Plug the USB cable into the DSK board and plug the other end of the cable to your PC’s USB port.

•

Launch the DSK diagnostic utility from the icon on the desktop to test your connection. From the diagnostic utility window, press the start button to run the diagnostics. If the connection is correct, in approximately 30 seconds all the test indicators shall turn to green and the text “PASS” shall be displayed on the diagnostic status text box. Incase the test fails, check your connection and retry until you see a green light.

•

Launch the CCS by double clicking on its icon.

•

If you are using CCS version 3.1, “No target connected” icon will be displayed on the bottom-left of CCS IDE screen. Go to “Debug” pull down menu and select “connect”, this will connect the DSP to the CCS and display “the target is now connected” icon.

•

Up on successful completion you can move on to the next chapter.

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2 Sampling and Reconstruction In this chapter a very simple DSP application, which is depicted on figure 2.1, is considered. As shown in the figure we simply pass a signal through an ADC and we feed this signal to a DAC without any further processing, we call this process a talk-through. With this simple DSP system, the primary focus is to be familiar with the software and hardware tools of the DSK package not the functional blocks. It is assumed that the reader has a basic knowledge of C/C++. Readers who are not familiar with C/C++ programming language are referred to [9-11]. Analog signal

ADC

DAC

Reconstructed analog signal

Figure 2.1: A talk-through system [2].

2.1 Talk-through Using Polling A program that implements the talk-through process is shown in figure 2.2.

1 #include "DSK6713_AIC23.h" //codec-DSK file support 2 Uint32 fs=DSK6713_AIC23_FREQ_8KHZ; //set sampling rate 3 void main() 4 { 5 short sample_data; 6 comm_poll(); //init DSK, codec, McBSP 7 while(1) //infinite loop 8 { 9 sample_data = input_sample(); //sample input 10 output_sample(sample_data); //output sample 11 } 12 }

Figure 2.2: Talk_through.c using polling [1-2].

Line 2 in the program listing sets the sampling frequency of the codec to F s = 8 kHz. So samples are taken from the inputs of the codec every T s = 0.125 ms. Inside the  main

1

function  comm_poll, a function that is located on the communication and support file  2

c6713dskinit.c  , is called. This function initializes the DSK, the AIC23 codec, and other

necessary hardware for the process. For now think the file c6713dskinit.c as a black-box 1

Throughout this documentation function names are written in  bold-italics font, and file and directory names are written in italics font. 2

This file is inside the CD that is included with this manuscript (or in the myprojects folder if you have copied the files from the CD to this folder) under the subfolder support.

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that initialize DSK components and provide necessary functionalities to enable them communicate each other, we will uncover it in the next section. The statement while(1) on line 7 creates an infinite loop that enables the DSK to listen for an incoming signal all the time (of course periodically every T s seconds). Inside the infinite loop a function input_sample, which is inside c6713dskinit.c, is called. This function takes samples of the input signal every T s seconds. The function output_sample, which is also inside c6713dskinit.c, then feeds the sampled data to the DAC. Now let’s create a project which will run the talk_through program on the DSK.

2.1.1 Creating a Project •

Launch the CCS

•

To create a project named talk_through.pjt, select Project → New. As shown in figure 2.3, type the name for the project “talk_through” and select the target TMS320C67XX. This will create a project named “talk_through” under the folder, C:\C6713\myprojects\talk_through.

Figure 2.3: CCS project window for new project creation.

•

In order to perform the talk-through task we need to equip the project with necessary functionalities by adding files into it. Let’s add first the source file talk_through.c in to our project. Select Project → Add Files to Project. From the

folder C:\C6713\myprojects\talk_through, double click on the C source file talk_through.c to add it to the project. To verify that the file is added, click on the

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“+” sign in front of  talk_through.pjt  inside the Project window with in the CCS; again click on the “+” sign in front of the source folder. Under the source folder talk_through.c shall appear. •

It is not only the talk_through.c file that is needed to perform the talk-through task; we also need other support files (e.g. c6713dskinit.c). Following the step above, add the following files that are on the folder C:\C6713\myproject\support : c6713dskinit.c, vectors_poll.asm, C6713dsk.cmd . We will discuss the function of 

these files later in this chapter. •

We also need to add library support files that contain functions that are used by source files. As we have added source files add the run-time support library file rts6700.lib (from the folder C:\C6713\C6000\cgtools\lib), the board support

library file dsk6713bsl.lib (from the folder C:\C6713\C6000\dsk6713\lib), and the chip support library file csl6713.lib (from the folder C:\C6713\C6000\csl\lib). •

By now your project window should look like figure 2.4.

Figure 2.4: Project file window.

Except the library files any of the files from the file window can be displayed by double clicking them. There is no need to add the include files they will be added automatically

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when we build the project (or if you right click on the project and select Scan All File Dependencies). But we need to tell the CCS where to find them as discussed below.

2.1.2 Code Generation and Options When a project is created, a number of default options will be assigned to it. These options greatly affect the execution of the project, so care must be taken in assigning project options. Select Project → Build Options to view the project options. Figure 2.5 shows the Project Build Options window for the compiler.

Figure 2.5: Project Build Options window for the compile.

Compiler Options Select the following compiler option under the “Basic” category: C671x (-mv6710) for “Target version” and leave the default option for “Generate Debug info”, “Opt Speed Vs Size”, “Opt Level”, and “Program Level Opt”. Under the “Advanced” category change the memory models option from “Far Aggregate” to “Far (--mem_model:data=far)”. Under “Processor” category: type CHIP_6713 instead of _DEBUG for “Define Symbol option”

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and type the path for dsk6713 header file, i.e. C:\C6713\C6000\dsk6713\include, for 1

“Include path option” . Then press OK. Linker Option Click on the “Linker” tab of the project build option window to see the linker options. When the project is build the output of this operation will be stored on the file indicated by the “Output File Name” option under the “Basic” category. The “.” in the path represents the folder where the project resides. This file, which will be loaded onto the DSK board, is a machine language translation of the source code that was written in C or assembly language. We can add library support files as discussed above or we can type the paths of each library file directly within “Include Libraries” option under the “Basic” category, separated by a comma. This will be the fastest way to add libraries as you get used to the CCS, but if the library files are added in this way they will not appear under the project file window.

2.1.3 Building and Running a Project Programs written in C, which are categorized as high-level languages even though they allow access to low level routines, need to be translated in to machine language that will be executed on the DSP hardware. The process of compilation, linking, loading, and execution is summarized in figure 2.6. Libraries

C program (Source)

C compiler

Machine code (Object)

Linker/Loader

Under the PC

Data

Execution

Program output

Under the DSK

Figure 2.6: Program compilation, linking, and execution [3].

Building a process refers to the compilation and linking operation, i.e. the source program is compiled and linked with necessary library files.

1

The files (header files) under this folder will be added to the projects automatically when we build the project (or when “Scan all File Dependencies” is selected).

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Let’s build and load the talk_through project: •

Select Project → Rebuild All. This compiles and assembles all the C files using c16x and assembles the assembly file vectors_poll.asm using asm6x [1]. The resulting machine code (object files) are then linked with the library files to give the executable file talk_through.out  that will be loaded and executed on the DSP hardware.

•

Select File → Load Program to load the executable file talk_through.out , which is in C:\C6713\myprojects\talk_through\Debug, on to the DSK. There is an option in the CCS that enables us to load a program automatically after build operation. Select Options → Customize Dialog, on the “Customize” dialog box click on the “Program Load Options” tab, select the option “Load Program After Build”, and press “OK”.

After building and loading the last step is to run the project. Select Debug → Run to run the project. Connect the LINE IN port of the DSK with a sinusoidal signal waveform of 2 Vpp and frequency 1 kHz. Connect the output port of the DSK, LINE OUT, to an oscilloscope and verify that a sinusoidal waveform of the same frequency but delayed and attenuated version of the input signal is produced. It can be clearly seen from the output of the DSK that the output signal is the sampled and reconstructed form of the input signal. First the input signal is sampled every T s seconds (0.125 ms) during ADC operation, and during DAC operation an over sampling (10 times in this case) is introduced to smoothen the output of the DSK. Try to zoom in the oscilloscope display so that you can clearly see the stairs of length 0.0125 ms (T s /oversampling_factor). Change the sampling frequency of the codec by modifying line 1

2 of the talk_through.c to 16, 24, 32, 44.1, 48, and 96 kHz and confirm that the length of  the stairs is always T s /oversampling_factor. Let’s now try to determine the amplitude and frequency limits of t he DSK: •

Increase the amplitude of the input signal until you encounter a distortion on the output signal. Confirm that the output signal begins to distort when the input signal amplitude goes beyond 6 Vpp.

1

These are sampling frequencies that are supported by AIC23 codec.

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•

Now reset the amplitude of the input signal to 2 Vpp and increase its frequency. Observe that the output signal gets distorted beyond the frequency F s  /2, as expected.

So in designing any DSP system with the aforementioned DSK setup these amplitude and frequency limits need to be taken into account.

2.1.4 Plotting with CCS We can plot signals both in frequency and time domain using the CCS. To see how it works we need to modify the talk-through source program of figure 2.2. Open talk_through.c and edit it to have the form shown in figure 2.7. 1 #include "DSK6713_AIC23.h" //codec-DSK file support 2 Uint32 fs=DSK6713_AIC23_FREQ_8KHZ; //set sampling rate 3 shor output_buffer[256]; 4 const short BUFLEN = 256; 5 int index = 0; 6 void main() 7 { 8 short sample_data; 9 comm_poll(); //init DSK, codec, McBSP 10 while(1) //infinite loop 11 { 12 sample_data = input_sample(); //sample input 13 output_buffer[index] = sample_data; // copy of the data 14 // that is feed to the DAC 15 index++; 16 index = index%BUFLEN; // if index is greater than BUFLEN 17 // reinitialize it to zero 18 output_sample(sample_data); //output sample 19 } 20 }

Figure 2.7: Modified talk_through.c [1].

Build and Run the project.

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To plot output_buffer, select View → Graph → Time/Frequency. Change the graph property dialog so that the options are as shown in figure 2.8 (a) to plot the output signal in time domain, or as shown in figure 2.8 (b) to have frequency-domain plot of the signal, and leave the default for the other options.

(a)

(b) Figure 2.8: CCS Graph Property Dialog (a) time-domain plot; (b) Frequency-domain plot.

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2.2 Support Program/Files Consideration In the previous section the support files c6713dskinit.c, vectors_poll.asm, and c6713dsk.cmd  were used for the talk-through project without knowing what was inside.

These files are almost used to support all of the projects that are listed here in the manuscript so it is worthwhile to understand their functionalities. Discussing how to construct them will be however out of the scope of this manuscript, so the main emphasis is on how to use them. Before we start to discuss these files we need to distinguish between two types of  programming techniques, i.e. polling-based and interrupt-driven. While polling-based techniques provide continues testing procedure for an occasion to happen, interrupt-driven techniques, as the name implies, use hardware or software interrupts to inform them for an occasion. A simple analogy might help to make things clear. Think of you want to talk to a manager of a company and you went to his office but he wasn’t there by the time you arrived and you know that he will comeback but you don’t know when. Now there are two ways to meet the person: the first is to wait for him in front of his office until he shows up, and the second one is to tell the secretary to give you a call by the time he comebacks and go to your work. The first method corresponds to the polling-based technique and the second one corresponds to the interrupt-driven technique. It is clear from the analogy that the polling-based methods are simple to implement but they are less efficient as compared to their interrupt-driven counterpart. Initialization/Communication file ( c6713dskinit.c) The main functionality of the file c6713dskinit.c is to initialize the DSK and provide necessary input-output functionalities for communication with the outside world. The partial listing of the program is shown in figure 2.9.

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In the talk-through project, what we have done first was to call the function  comm_poll . This function and  comm_intr (for interrupt-driven programs) in turn call the function  c6713_dsk_init that initializes the DSK. When interrupt-driven technique is used, i.e. 1

when comm_intr is called, interrupt #11 (INT11 ) is configured and enabled [1].

 //C6713dskinit.c Includes functions from TI in the C6713 CSL and C6713DSK BSL ... void c6713_dsk_init() //dsp-peripheral initialization { DSK6713_init(); //call BSL to init DSK-EMIF,PLL) hAIC23_handle=DSK6713_AIC23_openCodec(0, &config);//pointer to codec DSK6713_AIC23_setFreq(hAIC23_handle, fs); //set sample rate ... } void comm_poll() //added for communication/init using polling { poll=1; //1 if using polling c6713_dsk_init(); //init DSP and codec } void comm_intr() //for communication/init using interrupt { poll=0; // 0 since not polling IRQ_globalDisable(); //disable interrupts c6713_dsk_init(); // init DSP and codec … IRQ_map(CODECEventId, 11); //map McBSP1 Xmit to INT11 IRQ_reset(CODECEventId); //reset codec INT 11 IRQ_globalEnable(); //globally enable interrupts IRQ_nmiEnable(); //enable NMI interrupt IRQ_enable(CODECEventId); //enable CODEC eventXmit INT11 output_sample(0); //start McBSP interrupt outputting a sample } void output_sample(int out_data) //for out to Left and Right channels { short CHANNEL_data; AIC_data.uint=0; //clear data structure AIC_data.uint=out_data; //32-bit data -->data structure … if (poll) while(!MCBSP_xrdy(DSK6713_AIC23_DATAHANDLE)); //if ready to transmit MCBSP_write(DSK6713_AIC23_DATAHANDLE,AIC_data.uint); //write data } Uint32 input_sample() //for 32-bit input { short CHANNEL_data; if (poll) while(!MCBSP_rrdy(DSK6713_AIC23_DATAHANDLE));//if ready to receive AIC_data.uint=MCBSP_read(DSK6713_AIC23_DATAHANDLE);//read data … return(AIC_data.uint); } … Figure 2.9: Partial listing of c6713dskinit.c [1].

1

INT11 is chosen randomly, interrupts INT4 through INT15 can be chosen.

15

… void output_right_sample(short out_data) //for output from right channel { AIC_data.uint=0; //clear data structure AIC_data.channel[RIGHT]=out_data; //data from Right channel -->data structure if (poll) while(!MCBSP_xrdy(DSK6713_AIC23_DATAHANDLE)); MCBSP_write(DSK6713_AIC23_DATAHANDLE,AIC_data.uint); } void output_left_sample(short out_data) //for output from left channel … short input_left_sample() //for input to left channel … short input_right_sample() //for input to right channel Figure 2.9: (Continued).

The functions input_sample and output_sample are used to read and write data from/onto the codec respectively. If the program is polling-based, then poll = 1 and the code segment, if(poll) while(!MCBSP_xrdy(DSK6713_AIC23_DATAHANDLE)) ; //ready to transmit   MCBSP_write(DSK6713_AIC23_DATAHANDLE,AIC_data.uint) ; //write data,

will enter the while loop. The function  MCBSP_xrdy returns the value of the transmit1

ready (xrdy) register bit (= 1 if ready to transmit, or = 0 if not). While the McBSP is not ready to transmit (i.e. xrdy bit is 0) the loop condition is true, so the “while” loop becomes an infinite loop. When the McBSP is ready to transmit (i.e. xrdy bit is 1) the loop condition becomes false, so the loop will be jumped and the data is written on to the codec. If the program was interrupt-driven (poll = 0), the condition for “if” will become false so the data will be written without testing the xrdy bit. The same logic holds for the input_sample function, but in this case it tests the receive-ready (rrdy) bit. The function input_sample captures a 32-bit data, 16-bits from the left channel and 16-bit from the right-channel [1]. All of the examples in this manuscript use the codec in mono format, i.e. we treat both of the channels as a single channel, but the codec has a stereo capability in which two different input signals can be applied on the left and right channel. The functions input_left_sample and input_right_sample , which can capture 16 bit data from the left and right channel of the codec respectively, can be used in DSP applications that need stereo capability of the codec. Similarly the functions  output_left_sample and 1

McSBP = Multi-channel Buffered Serial Port

16

 output_right_sample can be used to write a 16 bit data at the left and right channel of the codec respectively. Vector File (vectors_poll.asm/vectors_intr.asm ) When a DSP receives an enabled interrupt, it will jump to a location in the vector table corresponding to the interrupt and begin execution [2]. The files vectors_poll.asm and vectors_intr.asm are used to fill the DSP interrupt vector table for polling-based and

interrupt-driven programs respectively. The partial listing of  vectors_intr.asm is shown in figure 2.10.

*Vectors_intr.asm Vector file for interrupt INT11 .global _vectors ;global symbols .global _c_int00 .global _vector1 .global _vector2 … .global _vector10 .global _c_int11 ;for INT11 .global _vector12 … .ref _c_int00 ;entry address … _vectors: _vector0: VEC_ENTRY _c_int00 ;RESET _vector1: VEC_ENTRY _vec_dummy ;NMI _vector2: VEC_ENTRY _vec_dummy ;RSVD _vector3: VEC_ENTRY _vec_dummy … _vector10: VEC_ENTRY _vec_dummy _vector11: VEC_ENTRY _c_int11 ;ISR address _vector12: VEC_ENTRY _vec_dummy … Figure 2.10: Partial listing of  Vectors_intr.asm [1].

As it can be seen from the program listing interrupt #11 (INT11), which was configured and enabled inside the file c6713dskinit.c when an interrupt-driven technique is used, is 1

associated with an ISR  c_int11. So, when an interrupt occurs the DSP will jump to this ISR, executes the commands inside the ISR, and returns to the place where it was before the interrupt occurs and continue its execution. We will see an example that utilizes this ISR in the next chapter. Polling-based programs don’t listen to interrupts, so for vectors_poll.asm (the vector file for polling-based programs) the branch instruction to the ISR for interrupt INT11 shall be replaced by NOP (No Operation) [1]. 1

ISR = Interrupt Service Routine

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Linker Command File ( c6713dsk.cmd ) The object files that result from the compilation of  *.c and *.asm files need to be linked together to produce the executable file that will be loaded into the DSK. To do this, the linker command file indicates where the physical memory is located, and how code sections are to be placed into the possible physical locations [2]. The linker command file c6713dsk.cmd  is shown in figure 2.11.  /*C6713dsk.cmd Linker command file*/  MEMORY { IVECS: org=0h, len=0x220 IRAM: org=0x00000220, len=0x0002FDE0 /*internal memory*/  SDRAM: org=0x80000000, len=0x00100000 /*external memory*/  FLASH: org=0x90000000, len=0x00020000 /*flash memory*/  } SECTIONS { .EXT_RAM :> SDRAM .vectors :> IVECS /*in vector file*/  .text :> IRAM .bss :> IRAM .cinit :> IRAM .stack :> IRAM .sysmem :> IRAM .const :> IRAM .switch :> IRAM .far :> IRAM .cio :> IRAM .csldata :> IRAM } Figure 2.11: Generic linker command file (c6713dsk.cmd ) [1].

The linker allocates the program in memory using default allocation algorithm [1]. But as it is shown in the program listing with a generic linker command file user defined memory allocations are possible. So the generic linker command file c6713dsk.cmd  is used to indicate the location of the physical memories (IRAM, SDRAM, …) and place different sections into these physical memory locations (e.g. section “.text” is placed inside IRAM).

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3 Periodic Signal Generation The C6713 DSK can be used to generate different types of signals that have a wide range of applications in communication systems. It can be used to generate sinusoidal waveforms for oscillators, alerting signals (e.g. telephone rings and emergency alert signals), signaling tones (e.g. DTMF) to mention the few. In this chapter the generation of periodic signals will be considered.

3.1 Sine Wave Generation The main thing that you, as a DSK programmer, need to take in to account for generating a waveform is in writing the sample values of the waveform on to the codec every sampling period. During DAC operation, as it was discussed in chapter two, an oversampling will be introduced to smoothen the sample values and then they will be fed to a reconstruction filter which even smoothes the sample values by cutting out frequency contents beyond the band of interest. Due to these summed effects you will be able to measure the intended waveform at the output of the codec. With this framework, the generation of periodic waveforms can be performed using a lookup table, where the table entries contain the sample values of the waveform to be generated. An interrupt-driven program that generates a sine wave using a lookup table is shown in figure 3.1. Line 3 in program listing creates a sine wave lookup table, each table entry representing sine wave values scaled by 1000 for angles loop*45o, where loop = {0, 1, 2, ... 7}. In the  main, the function  comm_intr is called. As we discussed in the previous chapter this function initializes the DSK, the codec and McBSP. Apart from this initialization it configures and enables interrupt #11 (INT11). The infinite while(1) loop enables the DSK to listen for interrupts all the time using INT11. When an interrupt occurs the DSP will jump to the  c_int11 (the ISR corresponding to INT11 in Vectors_intr.asm). In the ISR a sine wave value corresponding to the table index loop is written on to the codec. If the table index hasn’t reached its limit, it will be

incremented to output the next sample of the sine wave. If it exceeds the limit, it implies that all the samples belonging to a single period has been written to the codec, so the table index shall be reinitialized to zero to output the samples for the next period of the

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sinusoidal waveform. After performing all these operations the DSP will return to the place where it was before the interrupt occurs, the infinite while(1) loop, to listen for the next interrupt.  // Sine generation using lookup table 1 #include "dsk6713_aic23.h" //support file for codec,DSK 2 Uint32 fs=DSK6713_AIC23_FREQ_8KHZ; //set sampling rate 3 short sine_table[8]={0,707,1000,707,0,-707,-1000,-707}; //8 sine values for one period 4 int loop = 0; //table index 5 const short TABLE_LEN = 8; 6 interrupt void c_int11() 7{ 8 output_sample(sine_table[loop]); 9 if (loop < TABLE_LEN - 1) 10 ++loop; 11 else 12 loop = 0; 13 return; 14 }

//interrupt service routine //output sine values //check for end of table //reinit table index //return from interrupt

15 void main() 16 { 17 comm_intr(); 18 while(1); 19 }

//init DSK, codec, McBSP //infinite loop

Figure 3.1: Source file for sine wave generation using lookup table ( sine_lookup.c) [1].

Since the sampling rate for the codec is set to 8 kHz, the sample values will be written to the codec every 0.125 ms. With a lookup table of length 8, i.e. 8 samples per period, this will generate a sinusoidal waveform of frequency 1 kHz. Create a project named “sine_lookup”. Add the source files sine_lookup.c (which can be found in the folder myprojects/Sine_lookup), c6713dskinit.c, and Vectors_intr.asm. Be careful not to add Vectors_poll.asm because sine_lookup.c is an interrupt driven program. Add the library files rts6700.lib (run-time support library file), dsk6713bsl.lib (board support library file), and csl6713.lib (chip support library file). This time try to add them by typing the full path of each library file separated by a comma in the linker option under the category “Basic” on the text box “Include Libraries (-l)”. Following the procedures in the previous chapter set the project options the same as the “talk_through” project options. Then build and run the project. By connecting the codec output to an oscilloscope confirm that a 1 kHz sinusoidal wave is generated. You can also plot the output of the codec using the CCS graphics capability as

20

we have done it for the “talk_through” project. To perform this modifying the source file sine_lookup.c to include a buffer that accumulates the output of the codec and plot the

content of this buffer using the CCS.

3.1.1 Variable Amplitude Sine Wave Generation Representing the variables of a source program with interfaces will enable the user of the program to control the DSK output during execution. For example, the user can control the amplitude of the waveform generated by the project “sine_lookup” if some of the variables in the source file sine_lookup.c are represented by an interface. A GEL (General Execution Language), a scripting language supported by the CCS, can be used to create sliders, menu boxes and other interfaces to make our program more userinteractive. Since these interfaces represent variables that are being used in the source program, every time a change takes place on the interfaces the DSP will halt the program execution while the corresponding variables are being updated. Despite of this undesirable effect, GEL files are being used to represent variables because they can be created easily using the CCS. To create a GEL file that will construct a slider that represents the amplitude (gain) of the waveform generated by the source program sine_lookup.c: •

Open a blank file by selecting File → New → Source File.

•

Edit the file to have the form shown in figure 3.2.

•

Save it as gain.gel in the folder sine_lookup, where the project sine_lookup.pjt  resides.

1 /*gain.gel Create slider and vary amplitude (gain) of sine wave*/  2 menuitem "Sine Gain" 3 slider Gain(10,30,5,1,gain_parameter) 4{ 5 gain = gain_parameter; 6}

/*incr by 5,up to 35*/  /*vary gain of sine*/ 

Figure 3.2: Slider GEL file that represents gain.gel [1].

Line 2 in the program listing creates a menu item “Sine Gain” that will appear under the pull down menu “GEL” of the CCS. Lines 3-6 create a submenu item “Gain” under the 21

newly created pull down menu item “Sine Gain” and a slider described by “gain_parameter”. The description “gain_parameter” represents a slider labeled “Gain” with a staring value of 10, ending value of 30, incremental step of 5, and a page scroll of  1. Variables that are represented by GEL interfaces (“gain” in our case) need to be defined in the source program. Open the source file sine_lookup.c and edit it as shown in figure 3.3 so that the slider can control the amplitude of output. Load the GEL file gain.gel by selecting File → Load GEL. •

Build and run the project.

•

Open the slider “Gain” by selecting GEL → Sine Gain → Gain. This shall open the slider shown in figure 3.4.

 // Sine generation using lookup table 1 #include "dsk6713_aic23.h" //support file for codec, DSK 2 Uint32 fs=DSK6713_AIC23_FREQ_8KHZ; //set sampling rate 3 short sine_table[8]={0,707,1000,707,0,-707,-1000,-707}; //8 sine values for one period 4 int loop = 0; //table index 5 const short TABLE_LEN = 8; 6 short gain =10; 7 interrupt void c_int11() 8{ 9 output_sample(sine_table[loop]*gain); 10 if (loop < TABLE_LEN - 1) 11 ++loop; 12 else 13 loop = 0; 14 return; 15 }

//interrupt service routine  //output sine values //check for end of table //reinit table index //return from interrupt

16 void main() 17 { 18 comm_intr(); 19 while(1); 20 }

//init DSK, codec, McBSP //infinite loop

Figure 3.3: Modified sine_lookup.c [1].

By pressing the up-arrow key verify that the amplitude of the codec output increases as the value of the variable “gain” increases with step size of 5.

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Figure 3.4: A slider representing Amplitude (Gain).

3.1.2 Variable Amplitude and Frequency Sine Wave Generation The GEL file that generates the two sliders representing the gain and frequency of the codec output is shown in figure 3.5. •

Remove the existing GEL file gain.gel from the GEL file list in the Project View Window.

•

Load the GEL file gainFreq.gel from the folder myprojects/sine_lookup.

•

Remove the source file sine_lookup.c from the project files list.

•

Add the source file sine_varFreq.c, which is shown in figure 3.6, to the project. This file can be found in the folder myprojects/sine_lookup.  /*gainFreq.gel Two sliders to vary gain and frequency*/  menuitem "Sine Parameters" slider Gain(1,8,1,1,gain_parameter) { gain = gain_parameter; }

/*incr by 1,up to 8*/  /*vary gain*/ 

slider Frequency(2,8,2,2,frequency_parameter) { frequency = frequency_parameter; }

/*incr by 2,up to 8*/  /*vary frequency*/ 

Figure 3.5: gainFreq.gel [1].

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As shown in figure 3.6, the lookup table sine_table of the source file sine_varFreq.c contains 32 samples of a single period of a sinusoid scaled by 1000. Every time an interrupt occurs the value in the lookup table indexed by the variable loop is written on to the codec. Then the index loop is incremented by an amount equal to  frequency (the variable represented by one of the sliders). For example when  frequency = 2, the values that are indexed by loop will be sine_table[0], sine_table[2] , sine_table[4], sine_table[6], … sine_table[30] , which in effect generates a sinusoidal waveform with frequency 500

Hz. By setting the variable  frequency represented by the slider “Frequency” to 2, 4, 6, and 8 a sinusoidal waveform with frequency 500 Hz, 1 kHz, 1.5 kHz, 2 kHz can be generated.  //  sine_varFreq.c, Sine generation with different # of points 1 #include "dsk6713_aic23.h" //support file for codec, DSK 2 Uint32 fs=DSK6713_AIC23_FREQ_8KHZ; //set sampling rate 3 short sine_table[32]={0,195,383,556,707,831,924,981, 1000,981,924,831,707,556,383,195, 0,-195,-383,-556,-707,-831,-924, -981,-1000,-981,-924,-831,-707,-556,-383,-195}; 4 short loop = 0; 5 const short TABLE_LEN = 32; 6 short gain =1; 7 short frequency = 2;

//table index

8 interrupt void c_int11() 9{ 10 output_sample(sine_table[loop]*gain); 11 loop += frequency; 12 loop = loop%TABLE_LEN; 13 return; 14 }

//interrupt service routine //output sine values //increment frequency index //reinitialize table index if it goes beyond limit //return from interrupt

15 void main() 16 { 17 comm_intr(); 18 while(1); 19 }

//init DSK, codec, McBSP //infinite loop

Figure 3.6: Source file for variable frequency generation ( sine_varFreq.c ) [1].

Build and run the project. Open the two sliders “Gain” and “Frequency” under the submenu GEL → Sine Parameters. By varying the values of the two sliders confirm that the amplitude and frequency of the codec output vary in a fashion as discussed above.

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3.2 Triangular Waveform Generation A periodic triangular waveform can be generated using the DSK in the same manner as we have generated a periodic sinusoidal waveform. A source code that generates a periodic triangular waveform, which is under the folder myprojects/triangle, is shown in figure 3.8. The two  for  loops inside the main generate a lookup table representing the sample values for a single period of a triangular waveform, which is shown in figure 3.7. The first  for  loop generates the sample values for the first half of the period and the second loop generates the samples the second half of the period. When an interrupt occurs (i.e. every T s seconds) the code segment inside the ISR  c_int11 will be executed, which in effect

results a periodic triangular waveform at the output of the codec. •

Create a project named “ triangle.pjt” inside the folder myprojects.

•

Add necessary source and library files to the project.

•

Set the “Build Options” the same as the previous examples.

•

Build and Run the project.

•

By measuring the output of the codec confirm that a triangular waveform with a fundamental frequency 125 Hz is generated. Amplitude

T First half 

second half 

Figure 3.7: Triangular wave form with period T.

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time

 //triangle.c Generates a ramp using a look-up table #include "dsk6713_aic23.h" Uint32 fs=DSK6713_AIC23_FREQ_8KHZ; #define table_size 64 short data_table[table_size]; //lookup table int i; // table index interrupt void c_int11() { output_sample(data_table[i]); if (i < table_size-1) i++; else i = 0; return; }

//interrupt service routine //output value for each Ts //if table size is reached //reinitialize counter

void main() { short step = 2 * 0x7FFF/table_size; for(i=0; i < table_size/2; i++) // Fill the lookup table representing the first { // half of the triangular waveform data_table[i] = -0x8000 + i * step; } for(i=table_size/2; i < table_size; i++) // Fill the lookup table representing the second { // half of the period data_table[i] =data_table[i-1] - step; } i = 0; comm_intr(); while (1);

//reinit index //init DSK, codec, McBSP //infinite loop

}

Figure 3.8 Source code that generates a triangular waveform (triangle.c) [1]

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4 AM Signal Generation Amplitude modulated signals can be generated by following the same procedure that we used for periodic signal generation. Before discussing the generation of these signals, it is worthwhile to review some fundamentals of amplitude modulation.

4.1 Amplitude Modulation (AM) One of the possible ways to share the available frequency spectrum among computing users is to apply frequency multiplexing, where messages of different users occupy different frequency bins. This multiplexing can be achieved by varying the amplitude, frequency, and/or phase of an auxiliary waveform (carrier signal) according to the signal representing the message of each user (baseband signal). This process is called modulation. In amplitude modulation, as the name implies the amplitude of the carrier signal is varied according to the baseband signal. DSB-WC Signals One method to achieve an AM signal is to add the product of the carrier signal and the baseband signal to the carrier signal itself. This can be represented mathematically as: V (t ) = Ac [1 + m(t )] cos(2π f ct )

With, m(t ): baseband (modulating) signal  Accos(2π f ct ): carrier signal V (t ): modulated signal.

An example of amplitude modulation is shown in figure 4.1. Figure 4.1 (a) shows a sinusoidal modulating signal of frequency 400 Hz and Vpp = 1.6. Modulating a carrier signal having a frequency 2 kHz, figure 4.1 (b), with this modulating signal will result in the modulated signal shown in figure 4.1 (c). These kinds of AM signals are sometimes called double sideband with carrier (DSB-WC). The name came from the fact that both the lower and upper side bands of the modulated signal are transmitted along with the carrier signal.

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(a)

(b)

(c)

Figure 4.1: Amplitude modulation (a) modulating signal, (b) carrier signal, (c) modulated signal

The other thing that needs to be noted is that for an AM signal, generated with the method mentioned above, to be perfectly demodulated at the receiver the amplitude of the modulating signal (modulation index) shouldn’t exceed the amplitude of the carrier signal. DSB-SC Signals For receivers that implement phase-coherent demodulation it is necessary to maintain phase synchronization between the transmitter and the receiver local oscillators for perfect signal demodulation. One way to achieve phase synchronization between the transmitter and the receiver is to transmit a pilot (carrier) signal along with the information-bearing signal (modulated signal). With such a transmission the receiver can extract the pilot signal by employing a narrowband filter, and synchronize its local oscillator to the frequency and phase of the received signal. Clearly, the DSB-WC modulation scheme implements this method as it transmits the carrier signal along with the informationbearing signal. The other approach to deal with synchronization at the receiver is to estimate the carrier phase directly from the received signal. This approach is more efficient because the total transmit power can be allocated for the transmission of the information-bearing signal [8]. A detailed description of different carrier phase estimation techniques can be found in [8].

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So for receivers that perform the second carrier synchronization technique the pilot signal can be suppressed from the AM signal considered above, i.e. it is enough to transmit the product of the modulating signal and the carrier signal given by: U (t ) = Ac m(t ) cos(2π f ct ).

An AM signal, U (t ), generated in this way is called double-sideband suppressed carrier (DSB-SC) signal. SSB Signals Since the baseband signal can be recovered at the receiver from one of the sidebands, it is of great interest to transmit only one of the sidebands so that the available spectrum can be utilized more efficiently. Signals that contain only one of the sidebands (upper or lower) of the DSB-SC signal are called single sideband (SSB) signals. Two of the methods that are used in practice to generate SSB signals are: •

Filtering, and

•

Phasing.

In this manuscript we will consider SSB signal generation using filtering technique. For a detailed description of the two signal generation techniques the reader is referred to [6]. The processes of generating an SSB signal from a DSB-SC signal using a band pass filter is summarized in figure 4.2. In the figure it is assumed that the DSB-SC signal is generated from the modulation of a carrier signal having frequency  f c with a baseband signal having a lower frequency component  f l and a higher frequency component  f h. This DSB-SC signal is fed to a band pass filter having a center frequency  f c + (f h + f l)/2 and a passband of  f h – f l. With such specifications the filter will pass only the upper side band of  the DSB-SC signal, which in effect will result in the SSB signal shown in the figure.

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 f h - f l  H 1( f )

 f   f c + ( f h + f l)/2

DSB-SC signal

Spectral density

BP filter  H 1( f )

SSB signal Spectral densitSpectral y density

 f c - f h

 f c  f c – f l

f c + f h

freq.  f c - f h

 f c + f l

 f c + f h

freq.

 f c + f l

Figure 4.2 SSB signal generation using filtering method.

4.2 DSB-WC Signal Generation A polling based program that generates a DSB-WC signal is shown in figure 4.3. Inside the  main two lookup tables are created. The first lookup table, baseband , contains the sample values of a modulating signal (cosine wave with frequency 400 Hz). The second lookup table, carrier , contains the sample value of a carrier signal (cosine wave with frequency 2 kHz). Note that the contents of the lookup table baseband  are the sample values of the modulating signal for one period, but the contents of the lookup table carrier  are the sample values of the carrier signal for 5 periods. In fact only the first four samples (samples for a single period) can generate the same carrier signal that could be generated by the lookup table containing samples of 5 periods. This kind of repetition is introduced to match the length of both lookup tables, which eases mathematical calculations.

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 //DSBWC.c DSBWC signal generation using table for carrier and baseband signals #include "DSK6713_AIC23.h" //codec-dsk support file Uint32 fs=DSK6713_AIC23_FREQ_8KHZ; //set sampling rate short modIndex = 1; //index for modulation short output[20]; void main() { short baseband[20]={1000,951,809,587,309,0,-309, -587,-809,-951, -1000,-951,-809, -587,-309,0,309,587,809,951}; //400-Hz baseband short carrier[20] ={1000,0,-1000,0,1000,0,-1000, 0,1000,0, -1000,0,1000,0,-1000, 0,1000,0,-1000,0}; //2-kHz carrier short k; comm_poll(); //init DSK, codec, McBSP while(1) //infinite loop { for (k=0; k<20; k++) { output[k]= carrier[k] + ((modIndex*baseband[k]*carrier[k]/10)>>12); output_sample(20*output[k]); //scale output } } } Figure 4.3: Amplitude modulation source code ( DSBWC.c) [1].

Inside the  for  loop a DSB-WC signal sample is written on to the codec every sampling interval. A GEL file shown in figure 4.4 constructs a slider representing the variable modIndex, which controls the modulation index. Since only integer variables can be

represented by a slider, it is necessary to translate non-integer variables into an integer to represent them using a GEL slider. So in our case the slider “ModIndex” represents the modulation index multiplied by 10. To compensate this we have divided the variable modIndex by 10 in the DSB-WC equation.

The product of the modulating signal and the carrier signal in the DSB-WC equation is scaled by 2

12

to emphasis the modulation.  /*AM.gel To control modulation index*/  menuitem "Modulation Characteristics" slider ModIndex(1,10,1,1,index) { modIndex = index;

/*incr by 1,from 1 up to 10*/  /*vary modulation index*/ 

} Figure 4.4: AM.gel [1] .

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•

Create a project named “DSBWC”.

•

Add necessary files to the project including the source file  DSBWC.c, which is under the folder myprojects/DSBWC .

•

After setting the usual build options, build and run the project.

•

Load the GEL file AM.gel from the folder myprojects/DSBWC .

•

Measure the output of the DSK using a scope or plot the content of the buffer output  using the CCS graphics capability and confirm that a DSB-WC signal is

generated.

• Open the slider representing the modulation index by selecting GEL → Modulation Characteristics → ModIndex. By varying the slider value confirm that the depth of modulation changes accordingly.

4.3 DSB-SC Signal Generation A DSB-SC signal can be generated using the DSK by following the same procedure that we used for DSB-WC signal generation. Figure 4.5 shows a polling based program that generates a DSB-SC signal. Inside the main function a lookup table representing a carrier signal wit h frequency 3 kHz is generated. In this example we will take an audio signal as an input to the DSK that can be used as a baseband signal to generate a DSB-SC signal. To do so, inside the infinite while(1) loop the function input_sample is called to capture the samples of an input signal from the output of the DAC every sampling period. These sample values are then multiplied by the carrier signal and written on to the codec to produce the DSB-SC signal. Create a project named “DSBSC” and include the source file  DSBSC.c, which can be found in the folder myprojects/DSBSC . Also include other necessary library and support files. After setting the usual build options, build and run the project. A wave file THEFORCE.wav, which is in the folder myprojects/DSBSC, can be used as an input to the

DSK. Play the wave file and connect your PC's sound card output to the input of the DSK.

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 //DSBSC.c DSB-SC signal generation using table for carrier signal and  // an audio signal as a baseband signal #include "DSK6713_AIC23.h" Uint32 fs=DSK6713_AIC23_FREQ_16KHZ; #define tableLen 16 #define buffLen 256 short outBuffer[256]; short inBuffer[256];

//codec-dsk support file //set sampling rate

void main() { short carrier[16] = {1000, 383, -707, -924, 0, 924, 707, -383, -1000, -383, 707, 924, 0, -924, -707, 383}; short index = 0; short k = 0; comm_poll(); //init DSK, codec, McBSP while(1) //infinite loop { inBuffer[k]=input_sample(); outBuffer[k]= ((inBuffer[k]*carrier[index])>>12); output_sample(outBuffer[k]); k++; index++; k = k% buffLen; index = index% tableLen;

// carrier signal with  // frequency 3 kHz

//input new sample data // generate a DSB-SC signal

} } Figure 4.5: Program file for DSB-SC signal generation ( DSBSC.C ).

By plotting the input buffer inBuffer  and the output buffer outBuffer  in the frequency domain (the magnitude of the FFT) using the CCS graphics capability, confirm that a DSB-SC signal is generated. The snapshot of the contents of the two buffers, for an input wave file THEFORCE.wav, is shown in figure 4.6. As it can be seen from the figure a DSB-SC signal that have two sidebands centered at 3 kHz is generated. From the figure we also note that the upper sideband contains the exact copy of the input spectrum (only shifted by 3 kHz), and the lower sideband is the mirror image of the upper sideband (with the mirror positioned at frequency equal to 3 kHz).

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(a)

(b) Figure 4.6: A snapshot of the spectrum of the (a) input signal (inBuffer ) , (b) output signal (outBuffer ).

4.4 SSB Signal Generation As discussed earlier an SSB signal can be generated from a DSB-SC signal using filtering method. So it is worth revising some of the concepts of digital filters to facilitate our discussion about SSB signal generation. The two main categories of digital filters are: finite impulse response (FIR) and infinite impulse response (IIR). Where the main difference between the two categories lies on the 34

fact that the FIR filter output depends only on the history of input but the IIR filter output depends on the history of both the input and output. In this manuscript we will consider only FIR filters. A functional sketch of an FIR filter that has  N coefficients h(0), h(1), …, h( N -1) is shown in figure 4.7. As shown in the figure the filtering operation is a convolution, which can be expressed as: N −1

y(n) = ∑ h ( k ) x (n  − k ) , k = 0

with x(n) is the filter input and y(n) is the filter output at time n. h(0  x(n -1

X h(1

Z  x(n-1)

-1

Σ

h(2

Z  x(n-2

 y(n

X

. . . -1

Z  x(n-(N -1

h(N -1

X

Figure 4.7: Functional sketch of FIR filter.

There are several software packages that can be used to generate the coefficients of an FIR using different windowing functions, such as Hamming, Hanning, Blackman and Kaiser. For a detailed description of the FIR filter and different windowing functions the reader is referred to [4]. For the SSB signal generation example we will use filter coefficients generated using Matlab SPTool (Signal Processing Toolbox). The generation of FIR filter coefficients using Matlab SPTool is discussed in the appendix. A polling based program that generates an SSB signal is shown in figure 4.8. In the program listing two files that contain the coefficients of a low pass filter ( LPfilt.cof ) and band pass filter ( BPfilt.cof ) are included. The coefficient file  LPfilt.cof , generated using Matlab SPTool and shown in figure 4.9 (a), represents a low pass filter with cutoff  frequency at 3 kHz. This low pass filter is used to band limit the input audio signal with in the band 0 to 3 kHz so that aliasing can be avoided. This assumes that the information35

bearing signal is band limited between 0 and 3 kHz, which is the case for human speech. The spectrum of the filter coefficients is shown in figure 4.9 (b). The coefficient file  BPfilt.cof , which is also generated using Matlab SPTool and shown in figure 4.10 (a), contains the coefficients of a band pass filter centered at 4.5 kHz with a passband of 3 kHz. As discussed below, this filter is used to select the upper sideband of  the DSB-SC signal. The spectrum of the filter coefficients is shown in figure 4.10 (b).  // SSB.C SSB signal generation using filtering method #include "dsk6713_aic23.h" Uint32 fs=DSK6713_AIC23_FREQ_16KHZ; //set sampling rate #include "LPfilt.cof" //low pass filter coefficient file #include "BPfilt.cof" //band pass filter coefficient file #define carrierBufLen 16 #define sigLength 256 static short LPbuff[N], BPbuff[N]; short input[256], lpfiltOut[256], dsbscSig[256], bpfiltOut[256]; short filter(short newinp, short *dlyinp, short filtcoff [ ]); // function declaration void main() { short i; short sigIndex; short carrier[16] = {1000, 383, -707, -924, 0, 924, 707, -383, -1000, -383, 707, 924, 0, -924, -707, 383}; // carrier signal with frequency 3 kHz comm_poll();

 //init DSK using polling

for (i=0; i< N; i++) { LPbuff[i] = 0;  //init the LP filter buffer BPbuff[i] = 0; //init the BP filter buffer } i = 0; sigIndex = 0; while(1) { input[sigIndex]=input_sample(); //input new sample data lpfiltOut[sigIndex] = filter(input[sigIndex], Lpbuff, LPfiltCoff ); dsbscSig[sigIndex] = (lpfiltOut[sigIndex]*carrier[i])>>15; // scale the product of the  // baseband signal and  // the carrier signal bpfiltOut[sigIndex] = filter(dsbscSig[sigIndex], Bpbuff, BpfiltCoff); output_sample(bpfiltOut[sigIndex]); i++; sigIndex++; i = i%carrierBufLen; //reint table index if it exceeds the limit sigIndex = sigIndex%sigLength; } }

... Figure 4.8: Source code for SSB signal generation using filtering method (SSB.C ).

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... short filter(short newinp, short *dlyinp, short filtcoff [ ]) //implement FIR { short i; int yn ; dlyinp[N-1] = newinp; //insert the newest sample into the buffer yn = dlyinp[0] * filtcoff[N-1]; //y(0)=x(n-(N-1))*h(N-1) for (i = 1; i < N; i++) //loop for the rest { yn += dly[i] * filtcoff[N-(i+1)]; //y(n)=x[n-(N-1-i)]*h[N-1-i] dly[i-1] = dly[i]; //update the filter buffer } yn = (yn >>15); //filter's output return yn; //return y(n) at time n } Figure 4.8: (continued).

Inside the main, in the program SSB.C , a lookup table representing a carrier signal with frequency 3 kHz is generated. Then within the  for  loop the buffers for the two filters that contain the input history of the respective filter are initialized to zero. Inside the infinite while(1) loop first the input sample is captured from the ADC, then the captured sample is passed through the low pass filter, which is implemented by the function filter. As shown in the program listing the function filter, which takes newinp, dlyinp,  filtcoff as its input, returns the output of a convolution operation with newly arriving input sample newinp, delayed input samples dlyinp and filter coefficients  filtcoff . Apart from the

convolution operation the contents of the filter buffer are updated since we have passed the array Lpbuff to the function using a pointer (passing by reference). The low pass filter output is then multiplied with the carrier signal t o produce the DSB-SC signal sample. This DSB-SC signal sample is then fed to the band pass filter to cut out the lower sideband. The output of the band pass filter is then fed to the DAC to produce the SSB signal. #define N 81 // number of coefficients short LPfiltCoff[N] = {0, 12, 12, -8, -28, -13, 29, 45, 0, -63, -57, 36, 107, 47, -99, -147, 0, 188, 162, -98, -288, -123, 255, 372, 0, -464, -398, 241, 708, 306, -640, -955, 0, 1285, 1169, -771, -2550, -1314, 3667, 9623, 12288, 9623, 3667, -1314, -2550, -77 1, 1169, 1285, 0, -955, -640, 306, 708, 241, -398, -464, 0, 372, 255, -123, -288, 98, 162, 188, 0, -147, -99, 47, 107, 36, -57, -63, 0, 45, 29, -13, -28, -8, 12, 12, 0}; (a) Figure 4.9: (a)Coefficients for low pass FIR filter ( LPfilt.cof ), (b) Spectrum of the filter.

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(b) Figure 4.9: (continued).

short BPfiltCoff[81] = {0, -21, 5, -7, 28, 37, -70, -11, 0, 15, 137, -101, -107, 40, -41, 260, 0, 332, 67, -83, 288, 352, -616, -87, 0, 109, 960, -687, -708, 259, -265, 1686, 0, -2269, 484, -653, 2550, 3742, -8853, -2258, 12288, -2258, 8853, 3742, 2550, -653, 484, -2269, 0, 1686, -265, 259, -708, -687, 960, 109, 0, -87, -616, 352, 288, -83, 67, -332, 0, 260, -41, 40, -107, 101, 137, 15, 0, -11, -70, 37, 28, -7, 5, -21, 0}; (a)

(b)

Figure 4.10: (a) Coefficients for band pass FIR filter  (BPfilt.cof ), (b) Spectrum of the filter.

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Create a project named “SSB” and all necessary files to it including the source file SSB.C , which can be found in the folder myprojects/SSB. After setting the usual build options, build and run the project. By plotting the spectrum of the output buffer bpfiltOut and input buffer input , confirm that a SSB signal having the same shape as the input spectrum but shifted in frequency is generated. The plot of the DSB-SC signal sample buffer dsbscSig and the SSB signal sample buffer bpfiltOut for a particular time is shown in figure 4.11.

(a)

(b) Figure 4.11: (a) Generated DSB-SC signal (b) SSB signal generated after a band pass filter.

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Conclusion and Further Work With this introductory documentation it has been tried to discuss that the Texas Instrument TMS320C6713 DSP Starter Kit (C6713 DSK) can be used to implement real time DSP algorithms. As it was discussed with examples the Code Composer Studio (CCS), an IDE 1 accompanied with the C6713 DSK package, is a powerful tool that helps to implement DSP algorithms with high level programming language (C/C++). After compilation and linking, the machine code translation of the DSP algorithms t hat was written in C/C++ can be loaded and run on the DSK. The built-in AIC23 stereo audio codec performs all the tasks that are necessary to connect the DSP with the outside world. The operations include ADC, DAC, sampling and low pass filtering. The codec is designed with sigma-delta technology, where the main design objective is to achieve higher resolution for audio applications. The higher resolution is achieved by introducing higher oversampling ratio with the cost of reduction in maximum achievable sampling rate. The AIC23 codec supports a sampling rate of 44.1 kHz with CD quality. So if we use this sampling rate, with Nyquist criterion, the signal that can be recovered from the sampled values are those that have frequency components less than half the sampling rate (22.05 kHz). Frequency components above the allowed frequency will be filtered out by the antialiasing filter. To work with signals that have frequency components higher than the frequency that is allowed by the AIC23 stereo codec one need to interface an external daughter card that enhance the built-in codec. One possible solution is to interface DAC8581EVM daughter card with the DSK, which can work with signals in the radio frequency range. The C6713 DSK is connected with the host PC through the JTAG emulation port (the USB port). Although this interface allows us to load a machine code translation of a DSP algorithm that is written using high level programming language, its bandwidth is too limited to allow a communication between the DSK and a software application running on the host PC. This problem can be alleviated by interfacing a daughter card, which uses the Host Port Interface (HPI) pins of the DSK to access the internal memory of the DSP.

1

IDE = Integrated Development Environment.

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Appendix FIR Filter Coefficient Generation Using Matlab SPTool The coefficients of a finite impulse response (FIR) can be generated by the Matlab SPTool. In this appendix the generation of FIR filter design using a Kaiser window is considered. Enter the command "SPTool" inside the Matlab command window. This shall open a startup window startup.spt , shown in figure A.1.

Figure A.1: statup.spt.

•

Click on the button "New" on the startup window to open the "Filter Designer" environment.

•

In the environment set the sampling frequency to 16 kHz and select "Kaiser Window FIR" as a window algorithm.

•

Edit the "Specification" category to look like the specifications shown in figure A.2. With these specifications a low pass filter with cutoff frequency 3 kHz is generated.

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•

To change the name of the filter, go to the startup window and select Edit → Name → "Default name of the designed filter". On the popup box type the desired name

of the filter (e.g. lp3000).

Figure A.2: Low pass filter design specifications.

•

On the startup window ( startup.spt ) select File → Export. On the window "Export from SPTool" click on the button "Export to workspace". By doing so the filter can be accessed from the Matlab command window.

•

Enter the command "lp3000.tf.num" on the command window. "lp3000" represent the filter, "tf" will access the transfer function of the filter, and "num" will return the numerators of the transfer function of the filter in a vector form. So the result of the command is the coefficients of the low pass filter that we have designed.

The coefficients of a band pass, high pass and band stop filters can be generated by following the same procedure.

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