Fyp Final Presentation E1 Tapping
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The document describes a project to simulate and implement E1 tapping, which is the process of intercepting and decoding an E1 communication stream. It involves three main parts: 1. An E1 simulator created in MATLAB that generates an E1 stream from audio inputs. 2. A C# application that extracts the payload from the E1 stream, separates it into 30 channels, and writes each channel to a separate file. 3. A voice recovery unit that prepends headers to the channel files to make the voices playable.
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In this paper real time system identification is implemented on TMS320C6713 for speech signal using standard IEEE sentence (SP23) of NOIZEUS database with different types of real world noises at different level SNR taken from AURORA data base. The performance is measured in terms of signal to noise ratio (SNR) improvement. The implementation is done with “C’ program for least mean square (LMS) and recursive least square (RLS) algorithms and processed using DSP processor with Code composer studio.
My Project
This document summarizes a project on robotic control through speech recognition. The project uses a speech recognition module to recognize spoken commands and control a motorized robot with two DC motors. The speech recognition system is trained during a training phase by having registered speakers provide speech samples. During the testing phase, input speech is matched to the stored reference models and command recognition is observed through the operation of a stepper motor and DC motor.
My Project
This document summarizes a project on robotic control through speech recognition. The project uses a speech recognition module to recognize spoken commands and control a motorized robot with two DC motors. The speech recognition system is trained during a training phase by having registered speakers provide speech samples. During the testing phase, input speech is matched to the stored reference models and command recognition is observed through the operation of a stepper motor and DC motor.
CDR2(Sajjad Tarahomi)
The document describes the design and development of an RDS (Radio Data System) encoder that the author worked on from 2011-2012 at TAKTA Company. Some key points:
- The RDS encoder allows digital information like time, advertising, and text messages to be transmitted alongside FM radio broadcasts.
- The author's role included designing circuits and programming, assembling prototypes, testing, and managing the project.
- Technical specifications of the RDS encoder include supporting all RDS groups and protocols, interfaces for serial, TCP/IP, and I/O, and features for transmitting information like station name, program type, radio text, and traffic messages.
Radio Astronomy and electronics
This document describes an amplifier-integrator circuit with an ADC that is useful for amateur radio astronomy applications. It can amplify and average detected signals from a radio receiver, allowing the user to set gain, integration time constant, and output offset. The circuit includes analog and digital display/output and can automatically log measurement data on a PC via an RS232 serial interface. Software is also provided to control sampling and data storage.
Digital Tuner
The document discusses the design and testing of a musical instrument tuner. It covers:
- How musical notes are distinguished by their frequencies in different octaves
- How a tuner can identify the frequency of an input signal using a Fourier transform
- The process of tuning the tuner to detect specific note frequencies like C and output high or low
- Testing the tuner using a function generator and then a microphone for real instruments
Ilovepdf merged
The document provides an overview of source coding in digital communication systems. It discusses the key elements of a communication system including the transmitter, receiver, and channel. It then describes how an analog information source is converted to a digital signal through sampling, quantization, and coding. Source coding aims to remove redundancy in the information so as to minimize the bandwidth required for transmission. Channel coding adds extra bits to help detect and correct errors. Line coding represents the digital bit stream as voltage or current variations suited for the transmission channel. Key techniques discussed include pulse code modulation (PCM), companding, and various line codes.
Voice over IP (VoIP)
Overview of VoIP (Voice over IP) and FoIP (Fax over IP) technologies like Session Initiation Protocol and H.323.
Even though voice over IP (VoIP) was hailed as a technological innovation, the idea to transport real-time traffic over TCP/IP networks was not new back in the 1990s when VoIP started being deployed in networks. Chapter 2.5 of the venerable RFC793 (TCP) shows both data oriented application traffic as well as voice being transported over IP based networks.
Nevertheless, VoIP puts high demands on signal and protocol processing capabilities so it became possible at reasonable costs only in the 1990s.
VoIP can be roughly split into two main functions. Signaling protocols like SIP (Session Initiation Protocol), H.323 and MGCP/H.248 are used to establish a conference session and the data path for transporting real-time voice data packets. SIP has largely supplanted H.323 in recent years to its simpler structure and packet sequences. MGCP and H.248 are mostly used in carrier backbone networks.
Protocols like RTP (Real Time Protocol) transport voice packets and provide the necessary information for receivers to equalize packet flow variations to provide a smooth playback of the original voice signal.
Voice codecs are one of the core functions of the data path. Voice compression reduces the bandwidth required to transport voice over an IP based network. Compression may be less of a concern in local area networks with gigabit speeds, on slower links like 3G (UMTS, LTE) it still makes a lot of sense.
The algorithms used in different codecs make use of various characteristics of the characteristics of human speech recognition. Redundant information is removed from the signals thus slightly reducing the quality, but greatly reducing the required bandwidth.
In VoIP networks, the echo problem is typically compounded by the increased delay incurred by packetization of voice signals. To counteract the echo problem, VoIP gear (hard phones, soft phones, gateways) include echo cancelers to remove echo signals from the transmit signal.
To transport facsimile over an IP based network, even more technology is needed. Facsimile protocols are very susceptible to delay and delay variation and thus need more compensation algorithms. Protocols like T.38 terminate facsimile protocols like T.30 (analog facsimile) and transport the fax images as digitized pictures over IP based networks.
VoCoRoBo: Remote Speech Recognition and Tilt Sensing Multi-Robotic System
This work is based on the implementation of real-time speech recognition using DSP algorithms such as Chebyshev IIR filters, accelerometer for tilt-sensing and establishment of short-range wireless secure link with ARC4 cipher, all using low-cost 8-bit ATmega microcontrollers. The robot implements a simple but effective algorithm for comparing the spoken word with a dictionary of fingerprints using a modified Euclidean distance calculation. It also includes the ability to securely control the navigation of multiple robots located at remote locations wirelessly from the Control Module and also gather the various environmental data collected by the Robot Modules and display them in the back to Control. Considering the time-critical algorithms actually requiring large computations as well as a variety of sensors interfaced in the system, this project can demonstrate how one can build an expansible multi-robotic system from cheap and ubiquitous electronics.
Analog to digital conversion
1. Analog-to-digital conversion (ADC) allows computers to interact with analog signals by sampling and quantizing analog signals from devices like CD players.
2. During recording, an ADC converts an analog audio signal into a digital format by repeatedly measuring and assigning a binary number to the signal's amplitude at set intervals defined by the sample rate.
3. During playback, a digital-to-analog converter (DAC) reconverts the digital numbers back into an analog signal by combining the amplitude information from each sample to rebuild the original wave.
Similar to Fyp Final Presentation E1 Tapping (20)
DSP Based Speech Operated Home Appliances UsingZero Crossing Features
DSP Based Speech Operated Home Appliances UsingZero Crossing Features
Technical Elements of the Public Switched Telephone Network
Technical Elements of the Public Switched Telephone Network
Real Time System Identification of Speech Signal Using Tms320c6713
Real Time System Identification of Speech Signal Using Tms320c6713
VoCoRoBo: Remote Speech Recognition and Tilt Sensing Multi-Robotic System
VoCoRoBo: Remote Speech Recognition and Tilt Sensing Multi-Robotic System
Fyp Final Presentation E1 Tapping
- 1. DATA COMMUNICATION RECOVERY (E1 Tapping) USMAN TARIQ s470.05 M. SHAHZAIB ABBAS s471.05 NOMAN MUMTAZ ALI s481.05 S.M. ALI ABBAS RIZVI s488.05 A Project in collaboration with Snoopy Devices & Systems
- 16. E1-Simulator
- 17. C# E1 Tapping Application
- 23. E1-Tapping Application Stream Viewer Module 1 Module 2 Module 3 E1 Stream E1 Frames Payload . . . File # 01 File # 02 File # 30 Voice File # 01 Voice File # 02 Voice File # 30 . . . Voice Outputs Headers Ts # 0/Ts # 16 Frame Count Count No. of Frames Final Design
- 24. Simulation and Results E1-Simulator
- 29. Simulation Results E1-Tapping Application