US 4,550,425 A · 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.

Claims13

Dependent claims

A method for the digital encoding of an analog electrical signal representing speech comprising:

digitalizing said analog electrical signal at a sampling rate in order to produce upon an interval of time a multiplicity, or block, of linear pulse code modulated samples;
computing a scale factor for said multiplicity of pulse code modulated samples as a first quantity divided by a second quantity, wherein siad first quantity equals the maximum valued one of said multiplicity of samples minus the minimum valued one of said multiplicity of samples while said second quantity equals a constant number;
and
generating, as said digital encoding of said analog electrical signal, a multiplicity of successive ones of digital code words wherein each successive one digital code word is formulated as a third quantity divided by said scale factor, wherein said third quantity equals a corresponding one of said multiplicity of pulse code modulated samples minus said minimum valued one of said multiplicity of samples;
whereby said method of digital encoding is called Block-Scaled Pulse Code Modulation because a multiplicity, or Block, of Pulse Code Modulated samples are Scaled by a factor, a scale factor, which scale factor is computed in consideration of the dynamic range of said multiplicity, or Block, of Pulse Code Modulated Samples.

2 more elements

A method for decoding Block-Scaled Pulse Code Modulated (BSPCM) digitalized information consisting of a multiplicity of BSPCM digital code words, plus a minimum valued one of that like multiplicity of linear pulse code modulated (PCM) sample quantities from which said BSPCM digital code words were originally encoded, plus a range increment sale factor, in order to thereby said decoding reconstitute a multiplicity of linear PCM digital sample words, said decoding method comprising:

receiving, all as information input into a first buffer, a multiplicity of BSPCM digital code words, plus a minimum valued one of that like multiplicity of linear pulse code modulated (PCM) sample quantities from which said BSPCM digital code words were originally encoded, plus a range increment scale factor which is that quantity equaling the maximum valued one minus the minimum valued one of said multiplicity of linear PCM sample quantities from which said BSPCM digital code words were originally encoded entirely divided by a constant number;
computing from and upon the contents of said buffer a multiplicity of linear PCM digital sample words wherein each successive one is formulated as ((the corresponding successive one of said multiplicity of BSPCM digital code words) times (said range increment scale factor)) plus (said minimum valued one of said multiplicity of linear PCM sample quantities);
and
storing said computed multiplicity of linear PCM digital sample words in a second buffer;
whereby said receiving, said computing, and said storing, digitalized information of a BSPCM nature which, most notably, is both more compact than PCM information but which BSPCM information is not directly susceptible to digitalized signal processing has been decoded into PCM digitalized information which is, most notably, susceptible to digitalized signal processing operations.

2 more elements

Dependencies are inferred from the claim text.

Speech sampling and companding device

Life of this patent

Speech sampling and companding device

Life of this patent

Abstract

Claims13