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Abstract---We review the design of three audio effect

using TMS320 C6713 DSK. The effect has different

types, which are fuzz, echo, and reverb. The

applications of the effects are widely used in music and

entertainment industry. Our design has effectively

implemented the algorithm on a low cost platform DSP

board using C programming. The results are analyzed

using FFT method and to be assesed for meeting the

design purpose. The results shows that TMS320 C6713

DSK designs, can be develop into a real-time applicationfor audio effects in electronic devices and acoustic.

Keyword---Audio effect, Algorithm

I. INTRODUCTION

Audio effects are one of many applications ofDSP. The principle of an audio effect is to manipulate

input signals to a new output signal just as expected. In

daily life, audio effects are commonly used inentertainment industry. Before digital era, audio effects

are created using analog circuit, but the use of analog

circuit for multiple audio effects in one circuit needs avery complex circuit. This complex circuit refers to time,

budget, and also inflexibility. Audio effects using a DSPprocessor is not as complex as the audio effects build

using an analog circuit. Even though using a DSP

processor can create multiple audio signals in one project,the circuit is not significantly change or added, but the

program will be much more complex. The DSP processor

that is use in this audio effects design is the DSK

TMS320C6713. In the DSK, applications build in

algorithms using the C language, or the Assembler

language. So the first thing to do is to do the coding. Thenafter the coding is done, the algorithm can be implicated

directly to the DSK, and then the output and theperformance of the program can be analyzed.In general, audio effects are classified into three

types, which are Drive/Distortion, Modulation, and

Ambience. Before implementing these effects in the DSKTMS320C6713, we have to know how these audio effects

works. Knowing how it works will help us to make the

algorithm of the audio effects.

ANALYSIS OF REAL TIME AUDIO EFFECT DESIGN

USING TMS320 C6713 DSK

Rio Harlan, Fajar Dwisatyo, Hafizh Fazha, M. Suryanegara, Dadang Gunawan

Departemen Elektro Fakultas Teknik Universitas IndonesiaKampus Baru UI Depok

rioharlan@gmail.com,

msurya@ee.ui.ac.id

This paper will review about the performance ofa real-time audio effects design for fuzz, echo, and reverb.

The DSK TMS320C6713 is used for the design process.

The result shows that designing using the DSP board, can

be develop into a real-time application for audio effects inelectronic devices and acoustic. So then the design can be

written in the C6713 chip device and can be fabricated.

II. AUDIO EFFECT IN DSP BOARD

A. DSP BOARD TMS320 C6713

DSK TMS320 C6713 is a DSP board, a platform

where someone can build a signal processing application

by writing a program in the DSP board [1].

The process of the DSP board is very simple, the

analog signal coming from port mic in/line in then, it issampled to digital signal by the codec. After that, the

signal will be processed digitally by the DSP. How the

DSP process this signal depends on how the user programthe DSP board. Then, the output signal will be release

thru line out/headphone.

As mentioned before, signal processing by DSPdepends on program algorithm given from user to theDSP processor. Writing a program in the DSP processor

is as same as writing a program in other processors, which

is using the machine language or assembler. As an

alternative, we can also use C language to program theDSP processor. To development an application with DSK,

Texas Instrument provides software called Code

Composer Studio (CCS). CCS is software that is used for

interfacing user and the DSP board. CCS provides a workenvironment that contains all of the tools needed to

program the DSP processor [3]. The tools are integrated

with the CCS. It is also flexible software that can work

together with popular software, such as MATLAB,LabView, Visual Basic, etc.

B. FUZZ

Basic principle of fuzz is making a harmonic

distortion from a signal, it means occurrence of anotherfrequency component other than fundamental frequency.

mailto:msurya@ee.ui.ac.idmailto:msurya@ee.ui.ac.id

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The method used to make harmonic distortion is clipping

method. In this method, output value is limited in some

range. When the output value surpasses this limit, output

signal will automatically clip.

0 2 4 6 8 10 12 14-1

-0.8

-0.6

-0.4

-0.2

0

0.2

0.4

0.6

0.8

1Grafik input

Waktu

Amplitudo

Fig. 1. Input signal

0 2 4 6 8 10 12 14-0.8

-0.6

-0.4

-0.2

0

0.2

0.4

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0.8Grafik output

Waktu

Amplitudo

Fig. 2. Clipped input signal

Fig. 1 show input signal with assumed amplitude 1 volt.

After the fuzz effect applied to the input, the result will

showed in Fig. 2. It can be seen from this figure that

signal is clipped at 0.7 volt changing the signal shape.

C. ECHO

Echo is a sound reflection that comes to thelistener in a period amount of time after the original sound.

Generally, echo has a time delay that is relative long,which is more than 1 second. So that in echo effect, the

true sound and the artificial sound are clearly separated,

so that human hearings can tell the difference. The

mechanism of echo can be seen below.

Fig. 3. Echo mechanism

As seen in Fig. 3, the signal goes from the sourceto the listener in two paths [2]. First, the signal from the

source goes directly to the listener. Second, the signal

goes to the wall and then reflected to the user. The secondprocess will takes more time than the first process, so the

listener will hear two sounds in a different period of time.

The signal power from the second process will be

attenuated due to the reflection process.

D. REVERBERATION

Reverb is the sum of all sound reflections. The

principle of reverberations is more like echo, but in reverb

the sound reflections comes very often in a short period of

time. In reverb effect, usually the listener can not tell the

difference between the original sound and the reflectedsound. The mechanism can be seen below.

Fig.4. Reverb mechanism

Fig. 4 shows the mechanism of reverb effect.Every reflection from the source can be a new sound

source. There are two important parameters in

reverberation [5], which are:- Predelay, is the period amount of time of the first

sound reflection

- Reverb decay, is the period amount of time ofreverb since the input stops.

III. DESIGN

There are a few steps taken working in thisproject. The first step is to do an initialization on the

board. There are many types of initialization. We

initialize the board components that we use for the project.For example, in this audio effects project, the codec for

audio input and output jack in this board must be

initialized. If the other components like the switches and

LEDs are used, its must be initialize first. The

initialization can be written in C language.The next step is to write the main program. In

this section, the input signal will be manipulated with a

certain algorithm so that the output will create signals asexpected.

With the help of CCS features and external

analysis tools, the performance of the output such as

attenuation, delay, and noise can be analyzed. From here,the performance of DSP also can be seen, and it

represents the efficiency of the algorithm created by the

user.

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A. FUZZ

From the principles explained above, the

algorithm of fuzz effect can be made using C language.

int fuzz(int input)

{

int output;

output = gain * input;

if (output > batasclip)

{

output = batasclip;

}

else if (output < -batasclip)

{

output = -batasclip;

}

return(output);

}

From the program above, the gain parameter and

batasclip are the parameters that have been defined before.

The use of the batasclip parameter is for the limit where a

signal will be clipped. The gain parameter is needed tostrengthen the signal that is located under the clipping

limit, so the signal will also be clipped.

B. ECHO

The algorithm of this effect can be seen below.

int echo(int input){

int temp,output;

static int i=0;

temp = buffer[i];

buffer[i] = input;

i++;

if (i==buffer_size) i=0;

output = temp * 0.5;

output = output + input;

return(output);

}

The main line of the algorithm is located in theend of the algorithm. In the end of the algorithm, the latest

output will come together with the new output. The outputwill be attenuated, just like the theory of this effect. The

attenuation value used in this algorithm is 50%. The

buffer[i] parameter is an array that helps the delay process.In echo, the delay process is done in the input.

C. REVERB

The algorithm to implement this effect in DSK

TMS320C6713 can bee seen below

int reverb(int input)

{

int temp,output;static int i=0;

temp = buffer1[i];

output = temp * 0.5;

output = output + input;buffer1[i] = output;

i++;

if (i==buffer_length) i=0;

return(output);}

The difference between the reverb algorithm andthe echo algorithm is not significant. The difference takes

place in the program structure, and the delay process. Inreverb, the delay process is applied in the output. These

differences create a different effect character.

IV. RESULT AND DISCUSSION

The result of this application is analyzed with

audio editor software. A sinusoid signal with 1 kHzfrequency applied as the input to the board. This signal is

generated using computer software and sent to the line in

on the board. Then output from the board is sent to thecomputer to be analyzed in the same software.

Fig. 6. Input signal in frequency domain

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Fig. 7. FFT graph of output signal without effect

Fig. 6 and Fig. 7 show the FFT (Fast Fourier

Transform) graph of input and output signal without usingaudio effect algorithm where the axis represents the

frequency of a signal in Hertz and ordinate represent the

signal power in dB. It can be seen from the graph that inthe output, there are some other frequency component

beside the fundamental frequency. These frequency

components are not wanted and known as noise. But if welook from signal intensity, the noise signals have a very

low power compare to the main signal, which is causingthe noise effect not significant. Next, if the input and

output is compared, it will be found that there are 9,64 dB

of attenuation between these signal. This attenuationoccurs because of the characteristic and setting from the

board. But the attenuation will not be a big problem,

because when the application running, the output can be

amplified using an external amplifier.When fuzz effect algorithm is implemented to

the input, the harmonic distortion will occur as can be

seen in Fig. 8.

Fig. 8. FFT graph of output signal from fuzz effect

As it mention in the theory, that the harmonic

distortion should occur in the odd number of harmonic. Inthe figure above, there are two main frequency

components, 1 kHz and 3 kHz, which is match with the

theory. However, there are other frequency components

with lower power and unwanted around two main

frequencies. These unwanted frequencies can occurbecause of the noises that already occur before, when the

input is not using fuzz effect. Noise occurs because of

some factor, for example a poor audio transmission cable,noise from other instrument that is used, etc. However,

the signal that comes out from this fuzz effect is still can

be heard as the original signal, it means the frequency that

is heard is still 1 kHz, just in a distorted form. It can be

proved from the graph where the signal that has thehighest power is 1 kHz.

In delay based audio effect, the parameter thatcan be analyzed is the delay time. The delay time must be

analyzed in time domain.

Fig. 9. Echo output signal in time domain

Fig. 9 show two output signals from echo effect. The

original signal is in the left side while the reflected signal

is in the right side. The delay time is measured from the

beginning of the first signal to the second signal, resulting1,024 second. The attenuation between those signals can

be measure by comparing the FFT graph.

Fig. 10. FFT graph of original signal from echo effect

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Fig. 11. FFT graph of reflected signal from echo effect

From Fig. 10 and Fig. 11 show the attenuation between

original signal and reflected signal is 6,19 dB. Noise in

this graph is the same noise as discussed earlier.

Fig. 12. Reverb output signal in t ime domain

Fig. 12 shows the output result from reverb effect. As

mention before, there are two parameter that can bemeasured in reverb effect, predelay and reverb decay,

which are 320 ms and 2,239 s respectively. Theseparameter can be vary depend on buffer variable that is

used in reverb algorithm. Same as echo effect, the

attenuation between original signal and reflected signalcan be seen in FFT (Fig. 13 and Fig. 14). The attenuation

is 6,09 dB and constant for every reflected signal.

Fig. 13. FFT graph of original output signal from reverb effect

Fig. 14. FFT graph of reverb reflected signal, showing the signalattenuation

The time delay that is measured in echo and

reverb effect depends on buffer value in its algorithm. Afurther analysis can be done by finding mathematical

expression that shows relationship between these variablewith iteration method. With this function, we can easily

control the time delay.

0.3840.512

0.64

0.768

0.896

1.021

1.1521.28

1.408

1.536

0

5000

10000

15000

20000

25000

30000

0 0.5 1 1.5 2

Delay (s)

Buffer

Fig. 15. Iteration graph for echo effect

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0.32

0.448

0.575

0.704

0.832

0.96

1.088

1.216

1.344

1.472

0

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10000

15000

20000

25000

0 0.5 1 1.5 2

Delay (s)

Buffer

Fig. 16. Iteration graph for reverb effect

Fig. 15 and Fig. 16 show the iteration graph from

echo and reverb respectively. Because both graph are look

similar, we may conclude that buffer value have sameimpact in echo and reverb algorithm. The relation

between buffer and time delay can be written in

mathematic expression, resulting y = 15627x + 2.6468and y = 15621x + 4,8843 for echo and reverb respectively.

As we can see in both graph, y is defined as the buffervalue and x is defined as time delay.

However, we found that there is a weakness inboth algorithms. The value of buffer is limited in some

constant. If the value surpasses this limit, overflow can

occur. This weakness shows that the algorithm is not

perfect yet and has to be developed further.

V. CONCLUSION

The design of audio effect can be applied to the

DSP board successfully. It also can be used in real time

application. The input signal of audio effect can be a

music instrument or voice. The measured performances inevery effect give a good result and the purpose of thedesign is met. Noises still occur in this application, but the

effect is not significant so this noise will not disturb the

process of application.

VI. REFERENSI[1] TMS320C6713 DSK Technical Reference,

Spectrum Digital, Inc., Mei 2003

[2] Sikora R.,Implementing Echo and Reverberationusing the Audio Daughter Card and the

TMS320C5402 DSK and the TMS320C6711 DSK,

22 July 2001

[3] Code Composer Studio Help, Texas Instrument,2003

[4] Chassaing R., Digital Signal Processing andApplications with C6713 and C6416 DSK, John

Wiley & Sons, Hoboken, New Jersey, 2005

[5] Lehman,S.,Reverberation, http://www.harmony-

central.com/Effects/Articles/Reverb/, 5 Juni 2006