METHOD AND APPARATUS FOR AUDIO CODING AND DECODING

• US 20130030798A1
• Filed: 07/26/2011
• Published: 01/31/2013
• Est. Priority Date: 07/26/2011
• Status: Active Grant

• Current Assignee: Google Technology Holdings LLC (Alphabet Inc.)
• Original Assignee: Motorola Mobility, Inc. (Lenovo Group Ltd.)
• Inventors: Mittal, Udar, Ashley, James P., Gibbs, Jonathan A.
• Granted Patent: US 9,037,456 B2
• Time in Patent Office: Days
• US Class Current: 704/219
• CPC Class Codes: G10L 19/20 using sound class specific coding, hybrid encoders or object based coding

Abstract

An encoder and decoder for processing an audio signal including generic audio and speech frames are provided herein. During operation, two encoders are utilized by the speech coder, and two decoders are utilized by the speech decoder. The two encoders and decoders are utilized to process speech and non-speech (generic audio) respectively. During a transition between generic audio and speech, parameters that are needed by the speech decoder for decoding frame of speech are generated by processing the preceding generic audio (non-speech) frame for the necessary parameters. Because necessary parameters are obtained by the speech coder/decoder, the discontinuities associated with prior-art techniques are reduced when transitioning between generic audio frames and speech frames.

Claims

1. A method for decoding audio frames, the method comprising the steps of:

decoding a first audio frame with a first decoder to produce a first reconstructed audio signal;

determining a filter state for a second decoder from the first reconstructed audio signal;

initializing a second decoder with the filter state determined from the first reconstructed audio signal; and

decoding speech frames with the second decoder initialized with the filter state,

wherein determining the filter state for the second decoder comprises determining an inverse of the filter state that is initialized in the second decoder.

2. The method of claim 1 wherein:

the step of determining the filter state comprises performing at least one of LPC analysis on the reconstructed audio signal, down sampling of the reconstructed audio signal, and pre-emphasis of the reconstructed audio signal; and

the step of initializing the second decoder with the filter state is accomplished by receiving at least one of an LPC synthesis state, an upsampling filter state, and a de-emphasis filter state.

3. The method of claim 1 wherein the filter state comprises at least one of

a Re-sampling filter state memory

a Pre-emphasis/de-emphasis filter state memory

a Linear prediction (LP) coefficients for interpolation

a Weighted synthesis filter state memory

a Zero input response state memory

an Adaptive codebook (ACB) state memory

an LPC synthesis filter state memory

a Postfilter state memory

a Pitch pre-filter state memory

4. The method of claim 1 wherein the first decoder comprises a generic-audio decoder encoding less speech-like frames.

5. The method of claim 4, wherein the first decoder comprises a Modified Discrete Cosine Transform (MDCT) decoder.

6. The method of claim 4 wherein the second decoder comprises a speech decoder decoding more speech-like frames.

7. The method of claim 6, wherein the second decoder comprises Code Excited Linear Predictive (CELP) coder.

17. The method of claim 1 wherein the filter states comprise at least one of

a Re-sampling filter state memory

a Pre-emphasis/de-emphasis filter state memory

a Linear prediction (LP) coefficients for interpolation

a Weighted synthesis filter state memory

a Zero input response state memory

an Adaptive codebook (ACB) state memory

an LPC synthesis filter state memory

a Postfilter state memory

a Pitch pre-filter state memory

8. An apparatus comprising:

a first coder encoding generic audio frames;

a state generator outputting filter states for a generic audio frame m;

a second encoder for encoding speech frames, the second encoder receiving the filter states for the generic audio frame m and using the filter states for the generic audio frame m to encoder a speech frame m+1.

9. The apparatus of claim 8 wherein the state generator receives reconstructed audio and determines the filter states from the reconstructed audio.

10. The apparatus of claim 9 further comprising a decoder decoding frame m to produce the reconstructed audio.

11. The apparatus of claim 8 wherein the filter states comprise at least one of

a Re-sampling filter state memory

a Pre-emphasis/de-emphasis filter state memory

a Linear prediction (LP) coefficients for interpolation

a Weighted synthesis filter state memory

a Zero input response state memory

an Adaptive codebook (ACB) state memory

an LPC synthesis filter state memory

a Postfilter state memory

a Pitch pre-filter state memory

12. The apparatus of claim 8 wherein the first encoder comprises a generic-audio encoder encoding less speech-like frames.

13. The apparatus of claim 12 wherein the first coder comprises a Modified Discrete Cosine Transform (MDCT) coder.

14. The apparatus of claim 12 wherein the second encoder comprises a speech encoder encoding more speech-like frames.

15. The apparatus of claim 14, wherein the second encoder comprises Code Excited Linear Predictive (CELP) coder.

16. A method for decoding audio frames, the method comprising the steps of:

decoding generic audio frames with a first decoder;

determining filter states for a second decoder from a generic audio frame;

initializing a second decoder with the filter states determined from the generic-audio frame; and

decoding speech frames with the second decoder initialized with the filter states.

18. An apparatus comprising:

a first decoder decoding generic audio frames;

a state generator outputting filter states for a generic audio frame m;

a second decoder for decoding speech frames, the second decoder receiving the filter states for the generic audio frame m and using the filter states for the generic audio frame m to decode a speech frame m+1.

19. The apparatus of claim 18 wherein the state generator receives reconstructed audio and determines the filter states from the reconstructed audio.

20. The apparatus of claim 18 wherein the filter states comprise at least one of

a Re-sampling filter state memory

a Pre-emphasis/de-emphasis filter state memory

a Linear prediction (LP) coefficients for interpolation

a Weighted synthesis filter state memory

a Zero input response state memory

an Adaptive codebook (ACB) state memory

an LPC synthesis filter state memory

a Postfilter state memory

a Pitch pre-filter state memory