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Digital Signal Processing and Filtering-GSM Codec


The conversion of analogue speech waveform into digital form is usually called speech coding. A major benefit of using speech coding is the possibility to compress the signal, that is, to reduce the bit rate of the digital speech. In other words speech codec represents speech with as few bits as possible, while maintaining a speech quality that is acceptable. The efficient digital representation of the speech signal makes it possible to achieve bandwidth efficiency both in the transmission of the signal, e.g. between two antennas in a GSM system, as well as in storing the signal, e.g. on a magnetic media such as a GSM telephone memory. In GSM systems this kind of functionality is of critical importance, because in mobile communication the channel bandwidth is limited. What comes to the quality criterion, in transferring speaking voice over a GSM media the quality of the sound doesn’t have to be nearly as good as in the case of e.g. listening Mozart’s symphony from a CD-player. When two people are speaking on the phone the quality of speech doesn’t need to be perfect in order for mutual understanding to happen. In the near future, however, even Mozart’s symphonies might be transferred over some mobile media. Accordingly, the importance of speech coding will probably increase with the arrival of multimedia services. In the subsequent chapters the speech creation process is first explained in order to gain understanding of the basic principles in speech coding. After that different speech codec types will be introduced and explained in a bit more detail. Last, codec used in GSM will be examined with a bit more detail. Issues concerning delay and complexity of different codecs are excluded from this paper. The codec used in GSM systems is presented based on the original 13 kbits/s full-rate RPE codec. Newest standardisation is also excluded from this paper

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