Speech compression system and method

• US 7,593,852 B2
• Filed: 01/30/2007
• Issued: 09/22/2009
• Est. Priority Date: 09/22/1999
• Status: Expired due to Fees

• Current Assignee: DigiMedia Tech, LLC (IP Investments Group LLC)
• Original Assignee: Mindspeed Technologies Inc. (MACOM Technology Solutions Holdings, Inc.)
• Inventors: Gao, Yang, Benyassinc, Adil, Thyssen, Jes, Shlomot, Eyal, Su, Huan-Yu
• Primary Examiner(s): Opsasnick, Michael N
• Application Number:
  US11/700,481
• Publication Number:
  US 20070136052A1
• Time in Patent Office: 966 Days
• Field of Search: 704/223
• US Class Current: 704/233
• 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. A method of encoding an input speech signal to generate an encoded speech, the method comprising:

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

determining a pitch characteristic for each of a plurality of subframes of the frame of the input speech signal, when the second framing structure is selected, wherein the determining the pitch characteristic is performed as a function of prediction coefficients associated with each of the plurality of subframes;

searching a 3-pulse codebook including pulses having relative positions with respect to each other, when the second framing structure is selected, to determine a fixed codebook excitation vector with three pulses having the relative positions with respect to each other;

enhancing the fixed codebook excitation vector using the pitch characteristic to generate an enhanced fixed codebook excitation vector;

determining a fixed codebook gain for each of the plurality of subframes, when the second framing structure is selected, based on the enhanced the fixed codebook excitation vector; and

generating the encoded speech using the fixed codebook gain.

2. The method of claim 1 further comprising:

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

quantizing the pitch gain of each of the plurality of subframes; and

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

3. The method of claim 1, wherein the determining the pitch characteristic comprises applying a third order moving average prediction.

4. The method of claim 1, 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}.

5. A speech compression system for encoding an input speech signal to generate an encoded speech, 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 input speech signal; and

wherein the speech compression system is configured to;

determine a pitch characteristic for each of a plurality of subframes of the frame of the input speech signal, when the second framing structure is selected, wherein the pitch characteristic is determined as a function of prediction coefficients associated with each of the plurality of subframes;

search a 3-pulse codebook including pulses having relative positions with respect to each other, when the second framing structure is selected, to determine a fixed codebook excitation vector with three pulses having the relative positions with respect to each other;

enhance the fixed codebook excitation vector using the pitch characteristic to generate an enhanced fixed codebook excitation vector;

determine a fixed codebook gain for each of the plurality of subframes, when the second framing structure is selected, based on the enhanced the fixed codebook excitation vector;

generate the encoded speech using the fixed codebook gain.

6. The speech compression system of claim 5, wherein the speech compression system is further configured to derive a pitch gain for each of the plurality of subframes during pitch pre-processing, quantize the pitch gain of each of the plurality of subframes, and perform a delayed joint quantization of the fixed codebook gains for each of the plurality of subframes as a function of the stored quantized pitch gain of each of the plurality of subframes.

7. The speech compression system of claim 5, wherein the speech compression system is further configured to determine the pitch characteristic by applying a third order moving average prediction.

8. The speech compression system of claim 5, 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}.