Adaptive gain reduction to produce fixed codebook target signal

• US 6,104,992 A
• Filed: 09/18/1998
• Issued: 08/15/2000
• Est. Priority Date: 08/24/1998
• Status: Expired due to Term

• Current Assignee: Hanger Solutions, LLC (IP Investments Group LLC)
• Original Assignee: Conexant Systems Incorporated (Synaptics Incorporated)
• Inventors: Gao, Yang, Su, Huan-Yu
• Primary Examiner(s): Hudspeth, David R.
• Assistant Examiner(s): Lerner, Martin
• Application Number:
  US09/154,663
• Time in Patent Office: 697 Days
• Field of Search: 704/219, 704/220, 704/221, 704/224, 704/225
• US Class Current: 704/220
• CPC Class Codes: G10L 19/002 Dynamic bit allocation for perceptual audio coders G10L19/032
• CPC Class Codes: G10L 19/005 Correction of errors induced by the transmission channel, if related to the coding algorithm
• CPC Class Codes: G10L 19/012 Comfort noise or silence coding
• CPC Class Codes: G10L 19/08 Determination or coding of the excitation function; Determination or coding of the long-term prediction parameters
• CPC Class Codes: G10L 19/083 the excitation function being an excitation gain G10L25/90 takes precedence
• 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 19/10 the excitation function being a multipulse excitation
• 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 19/125 Pitch excitation, e.g. pitch synchronous innovation CELP [PSI-CELP]
• CPC Class Codes: G10L 19/18 Vocoders using multiple modes
• CPC Class Codes: G10L 19/265 Pre-filtering, e.g. high frequency emphasis prior to encoding
• CPC Class Codes: G10L 2019/0005 Multi-stage vector quantisation
• CPC Class Codes: G10L 2019/0007 Codebook element generation
• CPC Class Codes: G10L 2019/0011 Long term prediction filters, i.e. pitch estimation
• CPC Class Codes: G10L 21/0364 for improving intelligibility

Abstract

A multi-rate speech codec supports a plurality of encoding bit rate modes by adaptively selecting encoding bit rate modes to match communication channel restrictions. In higher bit rate encoding modes, an accurate representation of speech through CELP (code excited linear prediction) and other associated modeling parameters are generated for higher quality decoding and reproduction. The encoder applies adaptive gain reduction to optimize selection of appropriate gain contributions from the adaptive and fixed codebooks. Specifically, the encoder uses a first target signal to identify a contribution (a best code vector and a gain) from the adaptive codebook. Thereafter, a contribution from the fixed codebook is selected. The gain associated with the adaptive codebook contribution is then reduced by a factor, and the gain contribution from the fixed codebook is searched a second time, permitting fine tuning of the overall contribution. The gain reduction factor applied is adapted by considering both the encoding bit rate and a normalized correlation between the original target signal and the filtered signal from the adaptive codebook.

Claims

1. A speech system using an analysis by synthesis approach on a speech signal, the speech system comprising:

an adaptive codebook;

a fixed codebook;

a processing circuit that sequentially identifies a first gain applied to the adaptive codebook and a second gain applied to the fixed codebook; and

the processing circuit identifies a gain reduction factor applied to the first gain identified, the gain reduction factor is used by the processing circuit to perform the identification of the second gain.

2. The speech system of claim 1 wherein the gain reduction factor comprises an adaptive gain factor.

3. The speech system of claim 2 wherein the processing circuit identifies the adaptive gain factor by considering, at least in part, an encoding bit rate.

4. The speech system of claim 2 wherein the processing circuit identifies the adaptive gain factor by considering a correlation value.

5. The speech system of claim 4 wherein the processing circuit calculates the correlation value based, at least in part, on an original target signal.

6. The speech system of claim 4 wherein the processing circuit calculates the correlation value based, at least in part, on a filtered signal from the adaptive codebook.

7. A speech system using an analysis by synthesis approach on a speech signal, the speech system comprising:

a adaptive codebook;

a fixed codebook;

a processing circuit that generates a first contribution from the adaptive codebook and a second contribution from the fixed codebook; and

the processing circuit applying gain reduction to the first contribution from the adaptive codebook then regenerating the second contribution from the fixed codebook.

8. The speech system of claim 7 wherein the gain reduction comprises application of a gain factor.

9. The speech system of claim 8 wherein the processing circuit identifies the gain factor by considering an encoding bit rate.

10. The speech system of claim 8 wherein the processing circuit identifies the gain factor by considering a correlation value.

11. The speech system of claim 10 wherein the processing circuit calculates the correlation value based, at least in part, on an original target signal.

12. The speech system of claim 10 wherein the processing circuit calculates the correlation value based, at least in part, on a filtered signal from the adaptive codebook.

13. A speech system using an analysis by synthesis approach on a speech signal, the speech system comprising:

an adaptive codebook;

a fixed codebook;

a processing circuit that attempts to minimize a first residual signal using contributions from both the adaptive codebook and the fixed codebook; and

the processing circuit, after attempting to minimize the first residual signal, applying gain reduction to the contribution from the adaptive codebook and then recalculating the contribution from the fixed codebook by attempting to minimize a second residual signal.

14. The speech system of claim 13 wherein the gain reduction comprises use of a gain factor.

15. The speech system of claim 14 wherein the processing circuit identifies the gain factor by considering an encoding bit rate.

16. The speech system of claim 14 wherein the processing circuit identifies the gain factor by considering a correlation value.

17. The speech system of claim 16 wherein the processing circuit calculates the correlation value based, at least in part, on an original target signal.

18. The speech system of claim 16 wherein the processing circuit calculates the correlation value based, at least in part, on a filtered signal from the adaptive codebook.

19. The speech system of claim 13 wherein the second residual signal has a greater contribution from the fixed codebook than in the first residual signal.

20. The speech system of claim 13 wherein, to generate the first residual signal, the processing circuit first selects a contribution from the adaptive codebook and then selects a contribution from the fixed codebook.