Speech compression system and method

• US 20070136052A1
• Filed: 01/30/2007
• Published: 06/14/2007
• Est. Priority Date: 09/22/1999
• Status: Active Grant

• Current Assignee: DigiMedia Tech, LLC (IP Investments Group LLC)
• Original Assignee: Mindspeed Technologies Inc. (MACOM Technology Solutions Holdings, Inc.)
• Inventors: Shlomot, Eyal, Gao, Yang, Thyssen, Jes, Benyassine, Adil, Su, Huan-yu
• Granted Patent: US 7,593,852 B2
• Time in Patent Office: Days
• US Class Current: 704/207
• CPC Class Codes: G10L 19/00 Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis in musical instruments G10H
• CPC Class Codes: G10L 19/167 Audio streaming, i.e. formatting and decoding of an encoded audio signal representation into a data stream for transmission or storage purposes
• CPC Class Codes: G10L 19/20 using sound class specific coding, hybrid encoders or object based coding
• CPC Class Codes: G10L 19/22 Mode decision, i.e. based on audio signal content versus external parameters
• CPC Class Codes: G10L 19/24 Variable rate codecs, e.g. for generating different qualities using a scalable representation such as hierarchical encoding or layered encoding
• CPC Class Codes: G10L 2019/0001 Codebooks
• CPC Class Codes: H03G 3/00 Gain control in amplifiers or frequency changers

Abstract

The invention improves the encoding and decoding of speech by focusing the encoding on the perceptually important characteristics of speech. The system analyzes selected features of an input speech signal, and first performing a common frame based speech coding of an input speech signal. The system then performs a speech coding based on either a first speech coding mode or a second speech coding mode. The selection of a mode is based on characteristics of the input speech signal. The first speech coding mode uses a first framing structure and the second speech coding mode uses a second framing structure.

Claims

1. (canceled)

2. A method of processing a speech signal, the method comprising:

selecting one of a first framing structure and a second framing structure for encoding parameters of a frame of the speech signal;

predicting a fixed codebook characteristic for each of a plurality of subframes when the second framing structure is selected, wherein the predicting is performed as a function of prediction coefficients associated with each subframe and a fixed codebook characteristic from each of plurality of subframes of a previous frame; and

representing a fixed codebook gain for each of the subframe with respective predicted fixed codebook characteristic when the second framing structure is selected.

3. The method of claim 2 further comprising:

deriving a pitch gain for each of a plurality of subframes during pitch pre-processing;

quantizing the pitch gain of each of the subframes; and

performing a delayed joint quantization of the fixed codebook gains for each of the subframes as a function of the stored quantized pitch gain of each of the subframes.

4. The method of claim 2, wherein the predicting the fixed codebook characteristic comprises applying a third order moving average prediction.

5. The method of claim 2, wherein the prediction coefficients comprise a first subframe predictor coefficient represented as {0.6, 0.3, 0.1}, a second subframe predictor coefficient represented as {0.4, 0.25, 0.1}, and a third subframe predictor coefficient represented as {0.3, 0.015, 0.075}.

6. The method of claim 2, wherein the fixed codebook characteristic comprises fixed codebook energy.

7. A speech compression system for processing a speech signal, the speech compression system comprising:

a mode selection module configured to selected one of a first framing structure and a second framing structure for encoding parameters of a frame of the speech signal; and

a prediction module configured to predict a fixed codebook characteristic for each of a plurality of subframes when the second framing structure is selected, the prediction module predicts the fixed codebook characteristic as a function of prediction coefficients associated with each subframe and a fixed codebook characteristic from each of plurality of subframes of a previous frame;

wherein a fixed codebook gain for each of the subframe is represented with respective predicted fixed codebook characteristic when the second framing structure is selected.

8. The speech compression system of claim 7, wherein the speech compression system derives a pitch gain for each of a plurality of subframes during pitch pre-processing, quantizes the pitch gain of each of the subframes, and performs a delayed joint quantization of the fixed codebook gains for each of the subframes as a function of the stored quantized pitch gain of each of the subframes.

9. The speech compression system of claim 7, wherein the prediction module applies a third order moving average prediction.

10. The speech compression system of claim 7, wherein the prediction coefficients comprise a first subframe predictor coefficient represented as {0.6, 0.3, 0.1}, a second subframe predictor coefficient represented as {0.4, 0.25, 0.1}, and a third subframe predictor coefficient represented as {0.3, 0.015, 0.075}.

11. The speech compression system of claim 7, wherein the fixed codebook characteristic comprises fixed codebook energy.