Implementation of a Modem for Narrow Bandwidth Channel Using 6713 DSK

As communication plays an important role in day to day life, the effective and efficient voice transmission is to be maintained. This paper mainly deals with voice transmission over a channel and implemented using 6713 DSK. For this purpose, some modulation schemes and voice coders are implemented. So two points of view are developed. First, a static point of view, using a prototype on MATLAB, estimates the different combinations\u27 performances, using a stored speech sample. Then, a more dynamic point of view tests the system in real time, using C code adapted from MATLAB and embedded on DSPs, with the actual transmission channel being emulated by another DSP. In MATLAB the voice signal using different techniques are simulated and the outputs for modulation and demodulation signal are obtained which are shown in this paper for random bits operation of signals. An optimal transmission/reception scheme intended for voice transmission on DSP Processor TMS320C6713 is done using hardware and the results are compared in MATLAB by maintaining proper accuracy

Real-time digital signal processing system for normal probe diffraction technique

Ultrasonic systems are widely used in many fields of non-destructive testing. The increasing requirement for high quality steel product stirs the improvement of both ultrasonic instruments and testing methods. The thesis indicates the basics of ultrasonic testing and Digital Signal Processing (DSP) technology for the development of an ultrasonic system. The aim of this project was to apply a new ultrasonic testing method - the Normal Probe Diffraction method to course grained steel in real-time and investigate whether the potential of probability of detection (POD) has been improved. The theories and corresponding experiment set-up of pulse-echo method, TOFD and NPD method are explained and demonstrated separately. A comparison of these methods shows different contributions made by these methods using different types of algorithms and signals. Non-real-time experiments were carried out on a VI calibration block using an USPC 3100 ultrasonic testing card to implement pulse-echo and NPD method respectively. The experiments and algorithm were simulated and demonstrated in Matlab. A low frequency Single-transmitter-multi-receiver ultrasonic system was designed and built with a digital development board and an analogue daughter card to transmit or receive signals asynchronously. A high frequency high voltage amplifier was designed to drive the ultrasonic probes. A Matlab simulation system built with Simulink indicates that the Signal to Noise Ratio (SNR) can be improved with an increment of up to 3dB theoretically based on the simulation results using DSP techniques. The DSP system hardware and software was investigated and a real-time DSP hardware system was supposed to be built to implement the high frequency system using a rapid code generated system based on Matlab Simulink model and the method was presented. However, extra effort needs to be taken to program the hardware using a low-level computer language to make the system work stably and efficiently

Real-Time Digital Modeling of Analog Circuitry for Audio Applications

The goal of this project was to develop a scalable digital signal processing platform capable of modeling analog audio circuits using state-space modeling techniques. Using circuit theory as a foundation, the analog models were built around time-domain solution of circuit analysis. The resultant platform was capable of indistinguishably modeling variable analog filter circuits, with order being only restricted by hardware capabilities. Various continuous-to-discrete time conversion methods were investigated to determine the optimal sounding and performing algorithm

Real time speaker recognition using MFCC and VQ

Speaker Recognition is a process of automatically recognizing who is speaking on the basis of the individual information included in speech waves. Speaker Recognition is one of the most useful biometric recognition techniques in this world where insecurity is a major threat. Many organizations like banks, institutions, industries etc are currently using this technology for providing greater security to their vast databases.Speaker Recognition mainly involves two modules namely feature extraction and feature matching. Feature extraction is the process that extracts a small amount of data from the speaker’s voice signal that can later be used to represent that speaker. Feature matching involves the actual procedure to identify the unknown speaker by comparing the extracted features from his/her voice input with the ones that are already stored in our speech database.In feature extraction we find the Mel Frequency Cepstrum Coefficients, which are based on the known variation of the human ear’s critical bandwidths with frequency and these, are vector quantized using LBG algorithm resulting in the speaker specific codebook. In feature matching we find the VQ distortion between the input utterance of an unknown speaker and the codebooks stored in our database. Based on this VQ distortion we decide whether to accept/reject the unknown speaker’s identity. The system I implemented in my work is 80% accurate in recognizing the correct speaker.In second phase we implement on the acoustic of Real Time speaker ecognition using mfcc and vq on a TMS320C6713 DSP board. We analyze the workload and identify the most timeconsuming operations

Real-Time Digital Signal Processing Demonstration Platform

In order to demonstrate various digital signal processing (DSP) algorithms to students or potential students, a program was developed that runs in real-time on low cost, commercially available hardware. The program includes several common DSP algorithms such as lowpass filter, highpass filter, echo, reverb, quantization, aliasing, simple speech recognition, and fast Fourier transform (FFT). The program allows the user to easily switch between algorithms, to adjust the parameters of the algorithms, and to immediately hear the results. The demonstration hardware consists of the TMS320C5515 eZdsp USB Stick, a powered microphone, an audio source such as an MP3 player or cellphone, and speakers. Undergraduate electrical engineering students were shown the demonstration and were surveyed to determine which algorithms they found most interesting. The C language source code for the software is available from the author for free, so this program can be modified by instructors who wish to make their own demonstrations or used as a convenient starting point for student projects

An analysis of frequency recognition algorithms and implementation in realtime

Frequency recognition is an important task in many engineering fields, such as audio signal processing and telecommunications engineering. There are numerous applications where frequency recognition is absolutely necessary like in Dual-Tone Multi-Frequency (DTMF) detection or the recognition of the carrier frequency of a Global Positioning System (GPS) signal. Furthermore, frequency recognition has entered many other engineering disciplines such as sonar and radar technology, spectral analysis of astronomic data, seismography, acoustics and consumer electronics. Listening to electronic music and playing electronic musical instruments is becoming more and more popular, not only among young musicians. This dissertation details back groundinformation and a preliminary analysis of a musical system, the Generic Musical Instrument System (GMIS), which allows composers to experiment with electronic instruments without actually, learning how to play them.This dissertation gives background information about frequency recognition algorithms implemented in real time. It analyses state-of-the-art techniques, such as Dual- Tone Multiple Frequency (DTMF) implementations and MIDI-based musical systems, in order to work out their similarities. The key idea is to adapt well-proven frequency recognition algorithms of DTMF systems, which are successfully and widely used in telephony. The investigations will show to what extent these principles and algorithms can be applied to a musical system like the GMIS. This dissertation presents results of investigations into frequency recognition algorithms implemented on a Texas Instruments (TI) TMS320C6713 Digital Signal Processor (DSP) core, in order to estimate the frequency of an audio signal in real time. The algorithms are evaluated using selected criteria in terms of speed and accuracy with accomplishing over 9600 single measurements. The evaluations are made with simple sinusoids and musical notes played by instruments as input signals which allows a solid decision, which of these frequency recognition algorithms is appropriate for audio signal processing and for the constraints of the GMIS in real time

DSP implementation of OFDM acoustic modem

The success of multicarrier modulation in the form of OFDM in radio channels illuminates a path one could take towards high-rate underwater acoustic communications,and recently there are intensive investigations on underwater OFDM. Processing power has increased to a point where orthogonal frequency division multiplexing (OFDM) has become feasible and economical. Since many wireless communication systems being developed use OFDM, it is a worthwhile research topic. Some examples of applications using OFDM include Digital subscriber line (DSL), Digital Audio Broadcasting (DAB),High definition television (HDTV) broadcasting, IEEE 802.11 (wireless networking standard).OFDM is a strong candidate and has been suggested or standardized in high speed communication systems. In this Thesis in first phase ,we analyzes the factor that affects the OFDM performance. The performance of OFDM was assessed by using computer simulations performed using Matlab7.2 .it was simulated under Additive white Gaussian noise (AWGN) ,Exponential Multipath channel and Carrier frequency offset conditions for different modulation schemes like binary phase shift keying (BPSK), Quadrature phase shift keying (QPSK),16 Quadrature amplitude modulation (16-QAM),64-Quadrature amplitude modulation(64-QAM)which are used for achieving high data rates.In second phase we implement the acoustic OFDM transmitter and receiver design of [4,5] on a TMS320C6713 DSP board. We analyze the workload and identify the most timeconsuming operations. Based on the workload analysis, we tune the algorithms and optimize the code to substantially reduce the synchronization time to 0.2 seconds and the processing time of one OFDM block to 2.7235 seconds on a DSP processor at 225 MHz. This experimentation provides guidelines on our future work to reduce the per-block processing time to be less than the block duration of 0.23 seconds for real time operations