Adaptive codebook gain control for speech coding

• US 9,190,066 B2
• Filed: 01/26/2009
• Issued: 11/17/2015
• Est. Priority Date: 09/18/1998
• Status: Expired due to Term

• Current Assignee: MACOM Technology Solutions Holdings, Inc.
• Original Assignee: Mindspeed Technologies Inc. (MACOM Technology Solutions Holdings, Inc.)
• Inventors: Gao, Yang
• Primary Examiner(s): Opsasnick, Michael N
• Application Number:
  US12/321,934
• Publication Number:
  US 20090157395A1
• Time in Patent Office: 2,486 Days
• Field of Search: 704219-221
• US Class Current: 1/1
• CPC Class Codes: G10L 19/0204 using subband decomposition
• CPC Class Codes: G10L 19/09 Long term prediction, i.e. removing periodical redundancies, e.g. by using adaptive codebook or pitch predictor
• CPC Class Codes: G10L 19/12 the excitation function being a code excitation, e.g. in code excited linear prediction [CELP] vocoders
• CPC Class Codes: G10L 19/18 Vocoders using multiple modes
• CPC Class Codes: G10L 19/20 using sound class specific coding, hybrid encoders or object based coding
• CPC Class Codes: G10L 2019/0002 Codebook adaptations
• CPC Class Codes: G10L 2019/0016 Codebook for LPC parameters
• CPC Class Codes: G10L 25/90 Pitch determination of speech signals

Abstract

In accordance with one aspect of the invention, a selector supports the selection of a first encoding scheme or the second encoding scheme based upon the detection or absence of the triggering characteristic in the interval of the input speech signal. The first encoding scheme has a pitch pre-processing procedure for processing the input speech signal to form a revised speech signal biased toward an ideal voiced and stationary characteristic. The pre-processing procedure allows the encoder to fully capture the benefits of a bandwidth-efficient, long-term predictive procedure for a greater amount of speech components of an input speech signal than would otherwise be possible. In accordance with another aspect of the invention, the second encoding scheme entails a long-term prediction mode for encoding the pitch on a sub-frame by sub-frame basis. The long-term prediction mode is tailored to where the generally periodic component of the speech is generally not stationary or less than completely periodic and requires greater frequency of updates from the adaptive codebook to achieve a desired perceptual quality of the reproduced speech under a long-term predictive procedure.

Claims

1. A method of encoding a speech signal, the method comprising:

converting the speech signal from an analog form to a digitized speech signal;

applying a high-pass filter to the digitized speech signal to generate a high-pass filtered speech signal;

applying a perceptual weighting filter to the high-pass filtered speech signal to generate a first target signal;

identifying an adaptive codebook vector from an adaptive codebook using the first target signal;

filtering the adaptive codebook vector to generate a filtered adaptive codebook vector;

calculating an adaptive codebook gain for the adaptive codebook vector;

selectively reducing the adaptive codebook gain based on a correlation between the first target signal and the filtered adaptive codebook vector to generate a reduced adaptive codebook gain;

generating a second target signal based on the first target signal, the filtered adaptive codebook vector and the reduced adaptive codebook gain; and

generating an encoded speech based on the second target signal;

wherein the selectively reducing of the adaptive codebook gain reduces the adaptive codebook gain more when the encoding bit rate is a first encoding bit rate of a plurality of encoding bit rates than when the encoding bit rate is a second encoding bit rate of the plurality of encoding bit rates, and wherein the first encoding bit rate is lower than the second encoding bit rate.

2. The method of claim 1, wherein the selectively reducing the adaptive codebook gain is further based on an encoding bit rate selected from a plurality of encoding bit rates for the converting of the input speech signal to the encoded speech.

3. A speech encoding device for encoding a speech signal, the device comprising:

an adaptive codebook; and

a processing circuitry configured to;

convert the speech signal from an analog form to a digitized speech signal;

apply a high-pass filter to the digitized speech signal to generate a high-pass filtered speech signal;

apply a perceptual weighting filter to the high-pass filtered speech signal to generate a first target signal;

identify an adaptive codebook vector from the adaptive codebook using the first target signal;

filter the adaptive codebook vector to generate a filtered adaptive codebook vector;

calculate an adaptive codebook gain for the adaptive codebook vector;

selectively reduce the adaptive codebook gain based on a correlation between the first target signal and the filtered adaptive codebook vector to generate a reduced adaptive codebook gain;

generate a second target signal based on the first target signal, the filtered adaptive codebook vector and the reduced adaptive codebook gain; and

generate an encoded speech based on the second target signal;

wherein the selectively reducing of the adaptive codebook gain reduces the adaptive codebook gain more when the encoding bit rate is a first encoding bit rate of a plurality of encoding bit rates than when the encoding bit rate is a second encoding bit rate of the plurality of encoding bit rates, and wherein the first encoding bit rate is lower than the second encoding bit rate.

4. The speech encoding device of claim 3, wherein the processing circuitry is further configured to selectively reduce the adaptive codebook gain based on an encoding bit rate selected from a plurality of encoding bit rates to convert the input speech signal to the encoded speech.

5. A method of encoding a speech signal, the method comprising:

converting the speech signal from an analog form to a digitized speech signal;

applying a high-pass filter to the digitized speech signal to generate a high-pass filtered speech signal;

applying a perceptual weighting filter to the high-pass filtered speech signal to generate a first target signal;

identifying an adaptive codebook vector from an adaptive codebook using the first target signal;

calculating an adaptive codebook gain for the adaptive codebook vector;

calculating a gain factor based on a long term prediction gain;

selectively reducing the adaptive codebook gain using the gain factor to generate a reduced adaptive codebook gain;

filtering the adaptive codebook vector to generate a filtered adaptive codebook vector;

applying the reduced adaptive codebook gain to the filtered adaptive codebook vector to generate an adaptive codebook contribution;

generating a second target signal as a difference of the first target signal and the adaptive codebook contribution; and

generating an encoded speech based on the second target signal;

wherein the selectively reducing of the adaptive codebook gain reduces the adaptive codebook gain more when the encoding bit rate is a first encoding bit rate of a plurality of encoding bit rates than when the encoding bit rate is a second encoding bit rate of the plurality of encoding bit rates, and wherein the first encoding bit rate is lower than the second encoding bit rate.

6. The method of claim 5, wherein the selectively reducing the adaptive codebook gain is further based on an encoding bit rate selected from a plurality of encoding bit rates for the converting of the input speech signal to the encoded speech.

7. A speech encoding device for encoding a speech signal, the device comprising:

an adaptive codebook; and

a processing circuitry configured to;

convert the speech signal from an analog form to a digitized speech signal;

apply a high-pass filter to the digitized speech signal to generate a high-pass filtered speech signal;

apply a perceptual weighting filter to the high-pass filtered speech signal to generate a first target signal;

identify an adaptive codebook vector from the adaptive codebook using the first target signal;

calculate an adaptive codebook gain for the adaptive codebook vector;

calculate a gain factor based on a long term prediction gain;

selectively reduce the adaptive codebook gain using the gain factor to generate a reduced adaptive codebook gain;

filter the adaptive codebook vector to generate a filtered adaptive codebook vector;

apply the reduced adaptive codebook gain to the filtered adaptive codebook vector to generate an adaptive codebook contribution;

generate a second target signal as a difference of the first target signal and the adaptive codebook contribution; and

generate an encoded speech based on the second target signal;

wherein the selectively reducing of the adaptive codebook gain reduces the adaptive codebook gain more when the encoding bit rate is a first encoding bit rate of a plurality of encoding bit rates than when the encoding bit rate is a second encoding bit rate of the plurality of encoding bit rates, and wherein the first encoding bit rate is lower than the second encoding bit rate.

8. The speech encoding device of claim 7, wherein the processing circuitry is further configured to selectively reduce the adaptive codebook gain based on an encoding bit rate selected from a plurality of encoding bit rates to convert the input speech signal to the encoded speech.