Speech sampling and companding device

Abstract

An analog speech signal is sampled of a nominal rate of 6 kilohertz and digitized in a Mu-Law Encoder. The digital output of the Mu-Law Encoder is converted by a microprocessor performing table look-up to linearized pulse code modulation (PCM) samples nominally of eight bits per sample. Using a BSPCM (Block Scaled Pulse Code Modulation) method, in each block of nominally 246 eight-bit PCM samples (representing approximately 41 milliseconds), the maximum and minimum sample values are found and used to calculate a scale factor equal to the maximum sample value minus the minimum sample value, with the difference being then divided by a constant number nominally equaling 16. Then the BSPCM samples are generated from the PCM samples each as a corresponding one PCM sample minus the minimum PCM sample value, the difference being then divided by the scale factor. In effect, the bit rate is reduced by adjusting the step size to follow the local block dynamic range. The BSPCM samples so created are susceptible to signal processing operations like as to PCM samples. When the BSPCM encoded words plus the minimum, PCM encoded, sample plus the range increment scale factor are stored as a data block, then such data block can, at a later time, be decoded, or reconstituted, into linear PCM data. A silence interval is encoded as zero amplitude using run length coding of the number of blocks. Such digital PCM data can be converted to an analog audio signal for voice output across a telephone system.