US 2012/0239389 A1 · AUDIO SIGNAL PROCESSING METHOD AND DEVICE

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.

Claims14

Dependent claims

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.

3 more elements

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.

3 more elements

Dependencies are inferred from the claim text.

Facts

Application

13/511,331Filed 24 Nov 2010

Priority

24 Nov 2009

Published

20 Sep 2012

Term ends

Not estimatedThe source records don’t support an estimate.

Classification5 codes

G10L 19/00
Speech or audio signals analysis-synthesis techniques for redundancy reduction
G10L 19/005
Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals,…:…
G10L 19/12
Determination or coding of the excitation function; Determination or coding of the long-term prediction parameters: the excitation function being…
G10L 2019/0001
Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals,…:…

Family

Application61/264,248Us provisional application · filed 24 Nov 2009

Application61/285,183Us provisional application · filed 10 Dec 2009

Application61/295,166Us provisional application · filed 15 Jan 2010

US 9,020,812 B2Related patent

US 2012/0239389 A1This patent

From the US filing history. Foreign family members aren’t listed.

Audio signal processing method and device

Life of this patent

Audio signal processing method and device

Life of this patent

Abstract

Claims14