Speech processing system using adaptive vector quantization

• US 4,972,483 A
• Filed: 09/23/1988
• Issued: 11/20/1990
• Est. Priority Date: 09/24/1987
• Status: Expired due to Fees

• Current Assignee: Newbridge Networks Corporation (Nokia Corporation)
• Original Assignee: Newbridge Networks Corporation (Nokia Corporation)
• Inventors: Carey, Michael J.
• Primary Examiner(s): NOT, DEFINED
• Assistant Examiner(s): NOT, DEFINED
• Application Number:
  US07/248,379
• Time in Patent Office: 788 Days
• Field of Search: 381/29-41, 364/513.5, 375/122, 375/25-31
• US Class Current: 704/222
• CPC Class Codes: H03M 7/3082 Vector coding for television signals, see H04N19/94

Abstract

A speech processing system having an encoder comprising apparatus for receiving successive samples of PCM (pulse code modulated) encoded speech signals and apparatus, for applying sequential groups of the PCM encoded speech signals as primary vector signals to an encoder code book memory for selecting code words stored in the memory most closely approximating the vector signals. Apparatus is included for outputting to an output line the selected code words at a first bit rate. Further apparatus connects the selected code words to converted vector signals. The primary vector signals and converted vector signals are compared and difference signals result. The difference signals are quantized error signals are provided thereby. The quantized error signals are applied to the output line at the first or at a second bit rate to enable the processing system to transmit the speech signals at an effective bit rate constituting double the first bit rate or the sum of the first and second bit rate, or the quantized error signals is inhibited from being applied to the line. The processing system can thereby transmit the speech signals at an effective rate of the first bit rate thus incressing the traffic carrying capacity of the line.

Claims

1. A speech processing system having an encoder comprising:

(a) means for receiving successive samples of PCM (pulse code modulated) encoded speech signals,

(b) means for applying sequential groups of said PCM encoded speech signals as primary vector signals to an encoder code book memory for selecting code words stored in said memory most closely approximating said vector signals,

(c) means for outputting to an output line said selected code words at a first bit rate,

(d) means for converting said selected code words to converted vector signals,

(e) means for comparing the primary vector signals and converted vector signals and for providing difference signals therefrom,

(f) means for quantizing the difference signals and for providing quantized error signals therefrom,

(g) means for applying the quantized error signals to the output line at the first or at a second bit rate for enabling the processing system to transmit said speech signals at an effective bit rate constituting either double the first bit rate or the sum of the first and second bit rate, or for inhibiting the quantized error signals from being applied to the line whereby the processing system can transmit said speech signals at an effective rate of the first bit rate thus increasing traffic carrying capacity of the line.

2. A speech processor as defined in claim 1 in which the receiving means is comprised of an adaptive analysis filter having its transfer characteristics defined by a plurality of coefficient signals, and further including means for generating converted PCM encoded speech signals from the converted vector signals comprising an adaptive synthesis filter which is the analog of the adaptive analysis filter, for receiving the converted vector signals, filter coefficient signal generating means for receiving the converted vector signals and the converted PCM encoded speech signals and for generating and applying corresponding filter coefficients to the synthesis and analysis filters whereby the synthesis and analysis filters are rendered as analogous as possible.

3. A speech processor as defined in claim 2 in which the synthesis and analysis filters are lattice filters, the filter coefficient signal generating means including means for generating a new set of filter coefficient signals after each sample of said PCM encoded speech signal.

4. A speech processor as defined in claim 3 in which the receiving means is comprised of means for receiving the PCM encoded speech signals in linear form, pre-emphasis filter means for pre-emphasizing the linear form of PCM encoded speech signals, said adaptive analysis filter for receiving the pre-emphasized signals, flattening their spectra and reducing their average amplitudes, and smoothing filter means for smoothing the adaptive analysis filtered signals, and for providing the smoothed signals to the sequential group applying means.

5. A speech processor as defined in claim 4 further including means for normalizing the gain of the samples of the smoothed signals interposed between the smoothing filter means and the sequential group applying means.

6. A speech processor as defined in claim 1, 2 or 5 in which the comparing means includes means for determining the difference between each sample of each primary vector signal and each sample of a corresponding converted vector signal to provide said deference signal.

7. A speech processing system as defined in claim 5, in which the comparing means includes means for determining the difference between each sample of each primary vector signal and each sample of a corresponding converted vector signal to provide said difference signals, and a decoder comprising:

(a) means for receiving said selected code words and said quantized error signals from the output line,

(b) a decoder code book memory containing a table of code words identical to that of the encoder code book memory,

(c) means for applying the selected code words to the decoder code book memory and for obtaining best matched vector signals stored herein corresponding to the selected code words, p1 (d) means for converting the best matched vector signals into sequential samples of PCM encoded speech signals for output from the processing system.

(e) means for dequantizing the quantized error signals and adding the dequantized error signals to the best matched vector signals in the event of receipt of said error signals to form corrected vector signals,

(f) means for filtering the corrected vector signals for restoring the gain of the corrected signals to correspond to that of the PCM encoded speech signals, and

(g) means for encoding the gain corrected signals into PCM speech signals and for outputting the encoded signals from the decoder.

8. A speech processor as defined in claim 3, 4 or 7 in which the comparing means includes means for determining the difference between each sample of each primary vector signal and each sample of a corresponding converted vector signal to provide said difference signal, the synthesis filter and the analysis filter including means for generating backward error signals, and means for combining the backward error signals generated in the synthesis filter with the backward error signals generated in the analysis filter, thereby cancelling part of the noise caused by quantization and flattening the spectrum of the selected code words applied to the line relative to the received PCM encoded speech signals.

9. A speech processing system as defined in claim 3 or 4, in which the comparing means includes means for determining the difference between each sample of each primary vector signal and each sample of a corresponding converted vector signal to provide said difference signals, and a decoder comprising:

(a) means for receiving said selected code words and said quantized error signals from the output line,

(b) a decoder code book memory containing a table of code words identical to that of the encoder code book memory,

(c) means for applying the selected code words to the decoder code book memory and for obtaining best matched vector signals stored therein corresponding to the selected code words,

(d) means for converting the best matched vector signals into sequential samples of PCM encoded speech signals for output from the processing system.

10. A speech processor as defined in claim 4 further including means for normalizing the gain of the samples of the smoothed signals interposed between the smoothing filter means and the sequential group applying means, and in which the comparing means includes means for determining the difference between each sample of said gain normalized primary vector signal and each sample of a corresponding converted vector signal to provide said difference signals.

11. A speech processor as defined in claim 10 in which the synthesis and analysis filters are lattice filters, the filter coefficient signal generating means including means for generating a new set of filter coefficient signals after selection of each vector signal.

12. A speech processor as defined in claim 11 in which the receiving means is comprised means for receiving the PCM encoded speech signals in linear form, pre-emphasis filter means for pe-emphasizing the linear form of PCM encoded speech signals, said adaptive analysis filter for receiving the pre-emphasized signals, flattening their spectra and reducing their average amplitudes, and smoothing filter means for smoothing the adaptive analysis filtered signals, and for providing the smoothed signals to the sequential group applying means.

13. A speech processor as defined in claim 12 further including means for normalizing the gain of the samples of the smoothed signals interposed between the smoothing filter means and the sequential group applying means.

14. A speech processor as defined in claim 13 in which the comparing means includes means for determining the difference between each sample of each primary vector signal and each sample of a corresponding converted vector signal to provide said difference signal, the synthesis filter and the analysis filter including means for generating backward error signals, and means for combining the backward error signals generated in the synthesis filter with the backward error signals generated in the analysis filter, thereby cancelling part of the noise caused by quantization and flattening the spectra of the selected code words applied to the line relative to the received PCM encoded speech signals.

15. A speech processing system for receiving successive samples of PCM (pulse code modulated) input signals and corresponding scalar quantized error signals comprising:

(a) a plurality of speech encoders for vector quantization of said PCM input signals, each providing output signals comprising a code book entry signal corresponding to a group of samples of said PCM input signals and corresponding scalar quantized error signals,

(b) a multiplexer for multiplexing all of the code book entry signals and error signals for transmission to a decoder,

(c) means for sensing demands for transmission of additional digital signals, and p1 (d) means for progressively eliminating the error signals from being multiplexed in the multiplexer and for substituting said additional signals in place thereof as said demands increase,

whereby the capacity of the processing system to transmit code book entry signals and said additional signals is progressingly increased while decreasing the resolution of multiplexed transmission of the PCM input signals.

16. A speech processor as defined in claim 16, in which each of the speech encoders is comprised of:

(a) means for receiving successive samples of PCM (pulse code modulated) encoded speech signals,

(b) means for applying sequential groups of said PCM encoded speech signals as primary vector signals to an encoder code book memory for selecting code words stored in said memory most closely approximating said vector signals,

(c) means for outputting to an output line said selected code words at a first bit rate,

(d) means for converting said selected code words to converted vector signals,

(e) means for comparing the primary vector signals and converted vector signals and for providing difference signals therefrom,

(f) means for quantizing the difference signals and for providing quantized error signals therefrom,

(g) means for applying the quantized error signals to the output line at the first or at a second bit rate for enabling the processing system to transmit said speech signals at an effective bit rate constituting either double the first bit rate or the sum of the first and second bit rate, or for inhibiting the quantized error signals from being applied to the line whereby the processing system can transmit said speech signals at an effective rate of the first bit rate thus increasing traffic carrying capacity of the line.

18. A speech processor as defined in claim 16, in which each of the speech encoders is comprised of means for digitally encoding a plurality of input signals into a corresponding plurality of first low resolution signals for transmission in a first group of channels, and for encoding a corresponding group of error signals for transmission in a second group of channels for subsequent receipt by a decoder whereby the error signals can be combined with corresponding low resolution signals into high resolution (high equivalent bit rate) output signals, means for substituting additional low resolution encoded input signals in one or more of said second group of channels in place of said error signals for subsequent decoding of the additional low resolution signals into additional output signals, and substituting of the original low resolution signals corresponding to the error signals into low resolution output signals.

19. A speech processor system comprising means for digitally encoding a plurality of input signals into a corresponding plurality of first low resolution (low bit rate) signals for transmission in a first group of channels, and for encoding a corresponding group of error signals for transmission in a second group of channels for subsequent receipt by a decoder whereby the error signals can be combined with corresponding low resolution signals into high resolution (high equivalent bit rate) output signals, means for substituting additional low resolution encoded input signals in one or more of said second group of channels in place of said error signals for subsequent decoding of the additional low resolution signals into additional output signals, and substituting of the original low resolution signals corresponding to the error signals into low resolution output signals.

17. A speech processor as defined in claim 19 further including means for normalizing the gain of the samples of the smoothed signals interposed between the smoothing filter means and the sequential group applying means, and in which the comparing means includes means for determining the difference between each sample of said gain normalized primary vector signal and each sample of a corresponding converted vector signal to provide said difference signals.