TI DSK C6713 Simulink Model for Realtime Modulation and Demodulation

Download the Real time Modulation and Demodulation Models

You can download from the links below:

Real-time Modulation Model
Real-time Demodulation Model

THEORY

This is the last post of Amplitude and Demodulation using Texas Instrument DSK 6713 and I will try to windup all the important aspects of the term assignment. We have already discussed a simple simulink model of both Amplitude Modulation and Demodulation. You are recommended to see them first. Lets start with the basics.

We will make use of function generator and an oscilloscope. Both will be connected to the TI C6713 on line-in and line-out connectors. Frequency of Carrier Signal is fixed at 15 kHz while the message signal is varried from the function generator. Since DSK C6713 ensures realtime simulation, once the message signal is varied we will see on oscilloscope that modulated signal is also varying. Lets discuss different steps involved.

Block Diagram

C6713 Simulink Model - Amplitude Modulation

• Open a new simulink model and library browser

• Using Embedded Target For TIC6000 DSP menu select C6713 DSK Board Support.

• Using the Embedded Target For TIC6000 DSP tool in the library browser, select C6713 DSK Board Support and add ADC for Line In, DAC for Line Out and Reset Switch. You can extract all these from Simulink Library Browser.

• Implement the simulink model of AM as: You can also download from here.

• Message signal Line in parameters and carrier signal parameters can be set as following. This is very important. Set ADC Source as Line In, Sample Rate 19kHz, Word Length 16-bit. Amplitude :1.

Amplitude of the message signal can be change through function generator (up to 5KHz), whereas, frequency of the carrier signal is 15KHz.

• The output modulated signal observe in oscilloscope is:

C6713 Simulink Model - Amplitude De-Modulation

• Target the device in the similar manner as done above in amplitude modulation.

• A low pass filter has been designed in the FDA Tool in such a way that only the frequency of the message signal (5 KHz) is allow to pass through filter.

• Design the low pass filter using FDA Tool in Matlab.

• Export this filter to the simulink model

• Implement the simulink model of the demodulation. Click to enlarge please or download it from here.

• Run the model and observe the output at the oscilloscope.

CERTIFICATE

Muhammad Ahmed
Jamal Ahmed
Muhammad Faisal

For any queries feel free to contact

11:01 | Posted in Amplitude Demodulation, Amplitude Modulation, c6713, DSP, Electronics Projects, MATLAB, Matlab Simulink | Read More »

MATLAB Simulink Model of Amplitude Demodulation

Before you read this post, see the simulink model of amplitude modulation. Demodulation is the process of extracting the baseband message signal from the carrier so that it may be processed at the receiver. Demodulation is necessary for the massage signal to be received properly at the receiver. For that purpose various methods are used like diode detector method, product detector method, filter detector etc. The same has been implemented on simulink model. That model is then applied to Texas instrument Kit, which will be discussed in the later post. Lets see the block diagram of a basic demodulator :-

BLOCK DIAGRAM OF DEMODULATOR

Low pass filter has been implemented to extract the carrier from the modulated signal. Lowpass filter (LPF) filters out the high frequency component and allows the low frequency component to pass. Since the carrier signal is of relatively much higher frequency than that of message signal, carrier signal is attenuated while the message signal is received at the receiver.

MATLAB SIMULINK MODEL OF DEMODULATION

Here is the snapshot of MATLAB Simulink Model of Amplitude Demodulation which you can also by download it from here.

OUTPUT SHOWING DEMODULATION AT THE SCOPE

CERTIFICATE

This project named “AMPLITUDE MODULATION USING MATLAB SIMULINK AND TEXAS INSTRUMENT KIT C6713”, in all respect is the property of the following personnel who undertake this project as the term project in EE-322 ‘DSP & Filters’ in summer semester 2010. However the copy of the project can be distributed upon the approval of the following members:-

1.Muhammad Ahmed (elprojects@yahoo.com)

2.Jamal Ahmed

3.Muhammad Faisal

19:00 | Posted in Amplitude Demodulation, DSP, MATLAB, Matlab Simulink | Read More »

Amplitude Modulation on MATLAB Simulink

Download the Simulink Model:

Download here

What is Amplitude Modulation?

This topic is the result of Digital Signal Processing term project named Amplitude Modulation and Demodulation on Texas Instrument Kit DSK C6713 with Matlab Simulink. One of the fundamental part of our project is included in this very post. The whole term mini project will be gradually discussed in subsequent posts. Before we proceed, we must know what actually modulation and amplitude modulation is? A modulator alters the carrier waves corresponding to the variation of the modulating signals. Resulting modulated signal thus carries message information. Amplitude modulation is the process of changing the amplitude of a high frequency carrier signal corresponding to the amplitude of the modulating signal (Information). The wave whose amplitude is being varied is called the carrier wave and the signal doing the variation is called the modulating signal i-e message signal or information signal. The carrier is always almost a sinusoidal wave. The modulating or message signal can be a sine wave but it can be arbitrary waveform such as audio signal etc.

Some Mathematics :

For simplicity, suppose both carrier wave and modulating signal are sinusoidal:

vc = Vc sin wc t (c denotes carrier)

and

vm = Vm sin wm t (m denotes modulation)

We want the modulating signal to vary the carrier amplitude, Vc, so that:

vc = (Vc + Vm sin wmt).sin wc t

where (Vc + Vm sin wm t) is the new, varying carrier amplitude.

Expanding this equation gives:

vc = Vc sin ωc t + Vm sin ωc t. sin ωm t

which may be rewritten as:

vc = Vc [sin ωc t + m sin ωc t. sin ωm t]

where m = Vm/Vc and is called the modulation index.

sin ωc t.sin ωm t = (1/2) [cos(ωc - ωm) t - cos(ωc + ωm) t]

so, from the previous equation:

vc = Vc [sin ωc t + m sin ωc t. sin ωm t]

we can express vc as:

vc = Vc sin ωc t + (mVc/2) [cos(ωc - ωm) t] - (mVc/2) [cos(ωc + ωm) t]

The original carrier waveform, at frequency ωc, containing no variations and thus carrying no information.
A component at frequency (ωc - ωm) whose amplitude is proportional to the modulation index. This is called the Lower Side Frequency.
A component at frequency (ωc + ωm) whose amplitude is proportional to the modulation index. This is called the Upper Side Frequency.

AMPLITUDE MODULATION USING MATLAB SIMULINK

Lets discuss step by step the implementation of amplitude modulation on simulink modulation :-

Start the new Simulink model

Open the Simulink library browser

Create the simulation model of the AM

Message Signal:

A Sinusoidal signal of frequency 1 Hz, Amplitude=1
Carrier Signal of frequency 20 Hz and Amplitude=1

The above model is the equivalent of the mathematical expression:

s(t)=[1+m(t)](cos(2πfct))
Where;
m(t) represent the message signal
And cos(2πfct) represent the carrier

Run the Simulink model, the message, carrier and modulated output can be observed on the oscilloscope.

The above diagtam shows both supressed and without supressed carrier.You are recommended to see the following too :-

Amplitude Modulation on Matlab Simulink Model

CERTIFICATE OF OWNERSHIP

Muhammad Ahmed (NUST-ELECTRONICS)
Jamal Ahmed(NUST-ELECTRONICS)
Muhammad Faisal (NUST-ELECTRONICS)

21:48 | Posted in Amplitude Modulation, DSP, MATLAB, Matlab Simulink | Read More »

MATLAB Simulink Model of Music Equalizer using DSK 6713

What is Music Equalizer ?

Music equalizers are devices or software used for amplifying and attenuating predetermined frequency bands. Every music system, including some portable systems as well as professional stereo systems typically has an equalizer to equalize the audio data. Older system has analog equalizer which was tuned manually, but now a day’s digital music equalizer is very common. The advantage of digital equalizer is that we can store the preset gains of frequencies as desired and can be used in future by just pressing button. An audio equalizer typically will adjust the energy levels of the audio data in one or more different frequency bands in order to change the characteristics of the audio data. The equalizer may generate equalized audio data that may then be converted into analog data so that sound may be generated by a sound generating device, such as a speaker or headphone. The center frequencies for the various filters are distributed across an overall bandwidth having an upper and lower boundary. In general, the analog equalizer includes a combination of a simulated inductor and a bridging amplifier which are constructed of operational amplifiers whereas digital equalizer are made with software and processed by digital signal processor. The same has been accomplished on DSK 6713

Block Diagram of Music Equalizer

Audio Input: The Audio input can be taken from any music player. Mobile phone music player is used as the music input which is feed to DSK 6713 kit through line in connector. Mono channel is used for simplicity.

Analog to Digital Conversion: DSK 6713 has the built in A/D converter which converts the analog input signal into digital signal so that digital processing can be done on the input. The sampling frequency is set to 48kHz. Word length is 16-bit, scaling is normalized and samples per frame are 64. Refer to the diagram below :-

Low and Band Pass Filters for Music Equalizer: Filters are used for separating the signal of different frequencies. One low pass and six band pass filters are designed in MATLAB’s tool, FDATOOL, for filtering seven different bands of frequencies. IIR filters are faster than the FIR filter and give smoother amplitude response as compare to FIR filters. So, IIR filters are used. The key to filters the music correctly is that filter’s cut off frequencies must overlap. The cut off frequencies of all filters are given in table below.

Preset Band Gains for Equalizer: Music effects are produced simply by changing the gains of separated signals of different frequencies. For example, the bass effect is achieved by increasing the gain of lower frequency signals. Preset Band Gain block consists of the 4 set of preset gain which produces Flat, Rock, Bass and Opera Effects. Gain setting for Bass effect is shown below.

Preset Selector: Dip switches on the DSK-6713 are used for preset selection. Zero indexing is used so that normal (flat) preset is selected when no button is pressed. Led on the Kit indicates the preset selection. Parameters of DIP switches are set as shown in figure.

Signal Adder: Signal adder combines all separated signals of different frequencies after completing the processing. Simple adder is used for this purpose.

Digital to Analog Conversion: After the combining, the signal is converted in analog. This is done by DSK 6713 built in D/A converter. The sampling frequency of the A/D and D/A should be same for better output.

SIMULINK MODEL FOR MUSIC EQUALIZER ON DSK 6713

You can download the simulink model for the above equalizer from the link given below, however the snapshot of our working simulink model is given below :

Download Simulink Model for Music Equalizer here. In case of any problem. feel free to contact at homeofgadgets@yahoo.com

Important Note: This term project of Digital Signal Processing is supervised by Assistant Professor S K Hasnain and is the property of the following students of Pakistan Navy Engineering College (NUST).

Ahmed Fawad
Waseem Ahmed
Aziz Ahmed
Arslan Amin Dhoraji Wala

For queries please contact at homeofgadgets@yahoo.com

10:34 | Posted in c6713, DSP, Electronics Projects, MATLAB, Matlab Simulink | Read More »

Speech Guard 5 - Speech to Speech translator

Price:US$ 950

Unfortunately the incident which took place in haiti resulted in the vast destruction. Though different countries from different regions are making sure to help them out in the miserable atmosphere. The difficulty faced by the aid providers is the language issue since the Haitians do not understand English and even those who do are found very rare. To solve this problem -a big issue, ECTAO has released English-Haitian speech to speech translator, Speech Guard 5, for assisting medical personnel in communicating with the victims. These hand held translators will no doubt prove to be the perfect solution for interaction between the English Speaking MDs and the victims.

The CEO (Chief executive officer) of ECTACO, David Lubinitsky mentions, "In support of disaster recovery and emergency medical efforts in Haiti, we have readily made available thousands of dedicated units to be shipped expeditiously,"

A maximum of 20 languages are supported by MD-5, you just need to say what you need to translate and translation would be done through the speakers. This way, rescuing volunteers, medical personnel or other English speakers currently aiding the Haitians can easily communicate with the locals.

For more details Click here

17:06 | Posted in DSP, MISCELLANEOUS | Read More »

Blackout Technology - Digital Signal Processing

Cruise Speaker Phone by Jabra

Price: 100$

The modern processing is all based on DSP (Digital Signal Processing), e-g Modern Communication, Image Processing, Digital Filtering are the main subjects of Digital Signal Processing. Noise Blackout technology has also made use of DSP. The idea behind is to help the car drivers especially, because most often the fine imposed by traffic police is due the use of cellular phones while driving. And consequently headphones and blue tooth devices are on their peak in this era. Jabra has just released Cruiser Speaker Phone, featured by Noise Blackout Technology.

The idea is unique in its approach in a way that it caters for the reduction of enviornment noise keeping the user voice clearer. Probably the log amplifier or a filter might have helped them. Noise blackout is the technology that identifies the direction of various sounds – including traffic and backseat chatter – and reduces only surrounding noise without distorting a user’s voice. It uses dual microphones that captures the sound and then intelligently filtering the noise. This technology ensures both sides of caller to heard in natural sound.

The Cruiser speakerphone from Jabra also sports Multiuse connectivity, allowing two Bluetooth-enabled devices (including mobile phones, smartphones, and MP3 players) to be paired simultaneously via a turnkey auto-pairing function for instant usability. MP3 tracks can even be remotely controlled from the speakerphone's built-in play, pause and rewind buttons. In addition, a built-in FM transmitter allows calls and MP3s stored on a mobile device and played via the Cruiser to be heard through a car’s stereo system.

For more details Click here

18:13 | Posted in DSP, MISCELLANEOUS | Read More »