AUDIO SIGNAL PROCESSING METHOD AND DEVICE

• US 20120239389A1
• Filed: 11/24/2010
• Published: 09/20/2012
• Est. Priority Date: 11/24/2009
• Status: Active Grant

• Current Assignee: Industry Academic Cooperation Foundation, LG Electronics, Inc. (LG Corporation)
• Original Assignee: LG Electronics, Inc. (LG Corporation)
• Inventors: Jeon, Hye Jeong, Kim, Dae Hwan, Kang, Hong Goo, Lee, Min Ki, Lee, Byung Suk, Jeong, Gyu Hyeok
• Granted Patent: US 9,020,812 B2
• Time in Patent Office: Days
• US Class Current: 704/208
• 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 19/005 Correction of errors induced by the transmission channel, if related to the coding algorithm
• 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 2019/0001 Codebooks

Abstract

Disclosed is an audio signal processing method comprising the steps of: receiving an audio signal containing current frame data; generating a first temporary output signal for the current frame when an error occurs in the current frame data, by carrying out frame error concealment with respect to the current frame data a random codebook; generating a parameter by carrying out one or more of short-term prediction, long-term prediction and a fixed codebook search based on the first temporary output signal; and memory updating the parameter for the next frame; wherein the parameter comprises one or more of pitch gain, pitch delay, fixed codebook gain and a fixed codebook.

Claims

1. An audio signal processing method comprising:

receiving an audio signal including data of a current frame;

performing, when an error has occurred in the data of the current frame, frame error concealment on the data of the current frame using a random codebook to generate a first temporary output signal of the current frame;

performing at least one of short term prediction, long term prediction, and fixed codebook search based on the first temporary output signal to generate a parameter; and

updating a memory with the parameter for a next frame,

wherein the parameter includes at least one of a pitch gain, a pitch delay, a fixed codebook gain, and a fixed codebook.

2. The audio signal processing method according to claim 1, further comprising:

performing, when an error has occurred in the data of the current frame, extrapolation on a past input signal to generate a second temporary output signal; and

selecting the first temporary output signal or the second temporary output signal according to speech characteristics of a previous frame,

wherein the parameter is generated by performing at least one of short term prediction, long term prediction, and fixed codebook search on the selected temporary output signal.

3. The audio signal processing method according to claim 2, wherein the speech characteristics of the previous frame are associated with whether voiced sound characteristics or unvoiced sound characteristics of the previous frame are greater and the voice sound characteristics are greater when the pitch gain is high and the pitch delay changes little.

4. The audio signal processing method according to claim 1, wherein the memory includes a memory for long term prediction and a memory for short term prediction and includes a memory used for parameter quantization of a prediction scheme.

5. The audio signal processing method according to claim 1, further comprising generating a final output signal of the current frame by performing at least one of fixed codebook acquisition, adaptive codebook synthesis, and short term synthesis using the parameter.

6. The audio signal processing method according to claim 5, further comprising updating the memory with the final output signal and an excitation signal acquired through the long term synthesis and fixed codebook synthesis.

7. The audio signal processing method according to claim 1, further comprising performing at least one of long term synthesis and short term synthesis on a next frame based on the memory when no error has occurred in data of the next frame.

8. An audio signal processing device comprising:

a demultiplexer for receiving an audio signal including data of a current frame and checking whether or not an error has occurred in the data of the current frame;

an error concealment unit for performing, when an error has occurred in the data of the current frame, frame error concealment on the data of the current frame using a random codebook to generate a first temporary output signal of the current frame;

a re-encoder for performing at least one of short term prediction, long term prediction, and fixed codebook search based on the first temporary output signal to generate a parameter; and

a decoder for updating a memory with the parameter for a next frame,

wherein the parameter includes at least one of a pitch gain, a pitch delay, a fixed codebook gain, and a fixed codebook.

9. The audio signal processing device according to claim 8, wherein the error concealment unit includes:

an extrapolation unit for performing, when an error has occurred in the data of the current frame, extrapolation on a past input signal to generate a second temporary output signal; and

a selector for selecting the first temporary output signal or the second temporary output signal according to speech characteristics of a previous frame,

wherein the parameter is generated by performing at least one of short term prediction, long term prediction, and fixed codebook search on the selected temporary output signal.

10. The audio signal processing device according to claim 9, wherein the speech characteristics of the previous frame are associated with whether voiced sound characteristics or unvoiced sound characteristics of the previous frame are greater and the voice sound characteristics are greater when the pitch gain is high and the pitch delay changes little.

11. The audio signal processing device according to claim 8, wherein the memory includes a memory for long term prediction and a memory for short term prediction and includes a memory used for parameter quantization of a prediction scheme.

12. The audio signal processing device according to claim 8, wherein the decoder generates a final output signal of the current frame by performing at least one of fixed codebook acquisition, adaptive codebook synthesis, and short term synthesis using the parameter.

13. The audio signal processing device according to claim 12, wherein the decoder updates the memory with the final output signal and an excitation signal acquired through the long term synthesis and fixed codebook synthesis.

14. The audio signal processing device according to claim 8, wherein the decoder performs at least one of long term synthesis and short term synthesis on a next frame based on the memory when no error has occurred in data of the next frame.