ISSN (Online) : 2278-1021 ISSN (Print) : 2319-5940 International Journal of Advanced Research in Computer and Communication Engineering Vol. 3, Issue 10, October 2014 Copyright to IJARCCE www.ijarcce.com 8105 Study and Performance of AMR Codecs for GSM Divya Choudhary 1 , Abhinav Kumar 2 Student B.Tech Final Year, ECE, MIET, Jodhpur, India 1 Assistant Professor, ECE, MIET, Jodhpur, India 2 Abstract: In wireless communication system, limited bandwidth and power is the primary restriction. The existing wireless systems involved in transmission of speech visualized that efficient and effective methods be developed to transmit and receive the same while maintaining quality of speech, especially at the receiving end. Speech coding technique is a material of research for the scientific and academic community since the era of digitization (digital). Amongst all elements of the communication systems (transmitter, channel and receiver), transmission channel is the most critical and plays a key role in the transmission and reception of information. The quality of speech at receiver end decides by channel conditions. Modelling a channel is a multifarious task. A number of techniques are adopted to alleviate the effect of the channel. Adaptive Multi Rate is one of the techniques that neutralize the deleterious effect of the channel on speech. This technique utilizes variable bit rate that dynamically switches to specific modes of operation depending upon the channel conditions. For example, Low bit rate mode of operation is selected in adverse channel conditions, this helps to provide more error protection bits for channel coding and vice versa. Therefore, in this paper, application of Code Excited Linear Prediction (CELP) source codec on speech followed by AMR codec is studied. Further, higher the bit rate used, the better is the quality of speech. In this paper apart from speech codec about AMR is also studied that why the AMR is proposed for the GSM, how the bits rates are reduced in AMR, operation of AMR and other applications of AMR. Keywords: AMR, LPC, CELP, GSM I. INTRODUCTION Speech coding still is a major issue in the area of digital speech processing. Speech coding is transforming the speech signal to a more compacted form, which can then be transmitted with a significantly smaller memory. It is not possible to access infinite bandwidth. Hence, there is a need to code and compact speech signals. Speech compression is necessary in long-distance communication, high-quality speech storage, and message encryption. For instance, in digital cellular technology many consumers require to share the same frequency bandwidth. Utilizing speech compression makes it potential for more consumers to share the available system. Another example where speech compression is desirable is in digital voice storage. For a set quantity of available memory, compression makes it possible to store longer messages. Speech coding is a lossy type of coding, which means that the output signal does not closely sound like the input. The input and the output signal may perhaps distinguish to be different. Several techniques of speech coding such as Linear predictive Coding (LPC), Waveform Coding and Sub-bands Coding exist. The speech signals that require to be coded are wideband signals with frequencies from 0 to 8 kHz. Speech coding could be explained as the conversion of an analog speech signal into a digital signal, but such an explanation better suits the usual meaning of an analog to digital (AD) converter. Speech coding means the conversion of a speech signal which has been already digitalized, into another digital signal featuring a lower bit rate than the original signal. it is also referred to as transcoding and also as compression or bit rate decrement. Time and again, speech coding does not for only mean encoding a speech signal, but somewhat the complete process including decoding. The words speech codec by and large refers in a similar way to the decoder as well. At times, also the word “codec" is used which is a concatenation of the words coder and decoder, and not of the words encoder and decoder, as one would anticipate. A codec encodes a data stream/signal for transmission, storage, or decodes it for playback or editing [2][3]. Speech coding is essential to the operation of the public switched telephone network (PSTN), digital cellular communications, videoconferencing systems and emerging voice over Internet protocol (VoIP) applications. The objective of speech coding is to represent speech in digital form with as few bits as possible while retaining the clearness and quality required for the particular application. Interest in speech coding is motivated by the evolution to digital communications and the need to minimize bit rate, and therefore, conserve bandwidth. There is always an exchange between lowering the bit rate and preserving the delivered voice quality and clearness, however, depending on the application, many other restraints also must be considered, such as intricacy, delay, and performance with bit errors or packet losses. Two networks that have been developed primarily with voice communications in mind are the public switched telephone network (PSTN) and digital cellular networks. Furthermore, with the popularity of the Internet, voice over the Internet Protocol (VoIP) is growing swiftly and is anticipated to do so for the near future [1].