Please use this identifier to cite or link to this item: http://localhost:8081/xmlui/handle/123456789/3170
Authors: Mahawar, Kapil
Issue Date: 2012
Abstract: In secure communications (especially in military) and communications in special channels (e.g., underwater), the transmission bandwidth is usually limited. Moreover in digital communication system, the quality of the entire communication system has a direct relationship with the performance of speech communication. So we increasingly tend to compress the voice signal as much as possible in order to conserve the channel resource and promote the communication capacity, of course the compression of the speech signal must be within a certain quality. It has also become imperative to provide IP based solutions instead of using traditional user interfaces like RS-232, RS-422 etc. Thus, in line with the current trends, armed forces also continuously demand for network enabled solutions. Keeping in view the above requirement, a Network enabled Voice Codec Unit is to be designed which can bridge analog speech I/O (compressed speech well within the perceived speech quality) to an Ethernet network and thereby result in a VoIP (Voice over IP) based solution which can serve as a terminal controller for the baseband in military based communication. For the above said purpose a custom FPGA based board is designed primarily around Altera Cyclone II, AMBE Vocoder and SMSC LAN IC. The final objective is to enable one such unit to convert the input analog speech into digital speech samples, encode the speech using the selected AMBE Vocoder mode and then send the compressed bit stream out as TCP packets over the Ethernet interface. Simultaneously, the compressed bit stream of TCP packets from the other similar unit is to be read in from the Ethernet interface and decoded back in to digital speech samples. The decoded samples are to be converted back into analog speech via the codec whose output is sent to both the handset and line-level output connections. The proposed work was divided in two phases. In phase I, it was desired to design the Voice Codec Module (VCM) around the AMBE-2000 and codec AD73311 over the designed FPGA platform which was completed during the Aug 2011-Dec 2011 period (Illyd Sem). A VCM is one which is capable of accepting the analog speech either from Telephone/Microphone and delivering the analog speech signal back either to Telephone/Speaker. In this particular VCM, the A/D-D/A chip samples and quantizes the input voice data by microphone, then the AMBE-2000 runs the speech compression coding. The encoded voice data is sent to FPGA where it is processed and transmitted to the channel. Process of receiving voice data is just opposite of transmitting. During the phase I of the project, a generic code based on FPGA was developed for codec and Vocoder configuration which can process the voice data in any desired way. In phase II, the work achieved in the development of the voice codec module is utilized to complete the desired objective of developing the complete Computer Network enabled Voice Codec Unit (VCU) which can bridge analog speech I/O to an Ethernet network and thereby realize a Voice over IP (VoIP) terminal. Here basically I have relied on the functionalities of ordinary telephones to generate voice and signaling information and an embedded application over NIOS II soft processor platform employing Niche TCP/IP stack and MicroC-OSII RTOS in the same FPGA to transfer packet based data to the destination. - Two such Network enabled Voice Codec Units are developed that communicated directly with each other and not through intermediary servers. This work basically presents the IP net telephony system using computer network for... realizing a FPGA telephone terminal. In order to control the telephone terminal, the system uses a FPGA chip with NIOSII soft processor. residing over it with the TCP/IP protocol. The system uses AMBE-2000 based on AMBE compression algorithm which gives call voice clear and legible at the rate of 2.0-2.4Kbps which is very low compression rate. The software for IP net telephony system management is programmed in embedded C which is cross-compiled and the final executable code is stored in the flash memory of the FPGA telephone terminal. Here basically an insight into the integration of Ethernet, voice compression and phone communication network is given
Other Identifiers: M.Tech
Research Supervisor/ Guide: Gupta, H. O.
Kumar, Vishal
metadata.dc.type: M.Tech Dessertation
Appears in Collections:MASTERS' THESES (Electrical Engg)

Files in This Item:
File Description SizeFormat 
EEDG22145.pdf13.99 MBAdobe PDFView/Open

Items in DSpace are protected by copyright, with all rights reserved, unless otherwise indicated.