Abstract

This chapter describes the basics of sound and audio signals, the way data is presented to the processor from a variety of audio converters, and the formats in which audio data is stored and processed. It also discusses some software building blocks for embedded audio systems and examines data buffering as it applies to audio algorithms. Finally, it covers some fundamental audio algorithms and concludes with a discussion of audio and speech compression. In order to create an analog signal that represents a sound wave, a transducer must be used to convert the mechanical pressure energy into electrical energy. Digitization of the analog signals is accomplished with an analog-to-digital converter (ADC). In systems where noise is present, dynamic range is described as the ratio of the maximum signal level to the noise floor. The two main processor architectures used for audio processing are fixed-point and floating-point. A processor connected to an audio codec usually uses a direct memory access (DMA) engine to transfer the data from the codec link (like a serial port) to some memory space available to the processor.

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