Audio coding method, audio coding apparatus, and data...

Data processing: speech signal processing – linguistics – language – Speech signal processing – For storage or transmission

Reexamination Certificate

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C704S219000, C704S500000

Reexamination Certificate

active

06393393

ABSTRACT:

FIELD OF THE INVENTION
The present invention relates to an audio coding method, an audio coding apparatus, and a data storage medium. More particularly, the present invention relates to an audio coding method and an audio coding apparatus using a subband coding scheme according to an MPEG (Motion Picture Experts Group) standard, and a data storage medium which contains a program for implementing the audio coding method.
BACKGROUND OF THE INVENTION
In recent years, with spread of a multimedia personal computer or internet, it becomes possible to reproduce a moving picture or audio according to MPEG standard by software on the personal computer (PC), and coded data according to MPEG standard has been widely used.
As an encoder for creating coded data, expensive hardware is commonly used. While the coded data is sometimes created by software, since this coding process requires processing time several times as much as real time necessary for playing back a moving picture or audio, a plenty of time and troubles become necessary, and therefore, this has not been widely spread.
In order to make it possible for a PC user to create coded data at a low cost and with ease, it is required that coded data be created in real time by software processing.
Hereinafter, a description will be given of an example of a conventional audio coding method.
FIG. 11
is a block diagram showing an MPEG audio encoder standardized by ISO/IEC11172-3 as a format of coded audio data.
Turning to
FIG. 11
, subband analysis means
202
divides an input digital audio signal into 32 frequency components, and scale factor calculation means
203
calculates scale factors for respective subband signals and makes dynamic ranges for the respective subband signals uniform. The input digital audio signal is also subjected to an FFT (Fast Fourier Transform) process by FFT means
204
. Based on this result, psychoacoustic analysis means
205
derives a relationship model of an SMR (Signal to Mask Ratio) based on a psychoacoustic model utilizing a characteristic of men's auditory sense. Then, using this model, the bit allocation means
206
determines the number of bits to be allocated to each subband signal. According to the number of bits allocated to each subband signal, quantization/encoding means
207
quantizes/encodes each subband signal. Bit stream creating means
209
creates a bit stream comprising quantized/encoded data from the quantization/encoding means
207
and header information and auxiliary information which have been encoded by auxiliary information encoding means
208
, and outputs the bit stream.
In this conventional audio coding method, a coding process is performed for each subband by utilizing the fact that band power is distributed nonuniformly. Therefore, audio quality is determined by bit distribution for each subband signal using the psychoacoustic model. In addition, since the audio coding method has been standardized for the purpose of using a storage medium, it is well suitable for creating high-quality coded data, but is less suitable for a coding process in real time. The psychoacoustic model which determines audio quality requires a large amount of operation.
The conventional audio coding method and audio coding apparatus are so constructed, and are well suitable for creating high-quality coded data for the storage medium, but are less suitable for processing in real time on the PC by software in view of current CPU's processing ability, because use of the psychoacoustic model requires high processing ability. When operation is performed on the PC on which a high-performance CPU which has capability of real-time processing is mounted, if another application occupies a large part of processing by the CPU, processing cannot be performed in real time. As a consequence, discontinuity of audio might occur.
SUMMARY OF THE INVENTION
It is an object of the present invention to provide an audio coding method and an audio coding apparatus, which are capable of creating coded data of high quality and with no discontinuity without being affected by processing ability of a CPU on a personal computer and how much another application occupies processing on the CPU, and a data storage medium which contains aprogram for implementing this coding process.
Other objects and advantages of the invention will become apparent from the detailed description that follows. The detailed description and specific embodiments described are provided only for illustration since various additions and modifications within the spirit and scope of the invention will be apparent to those skill in the art from the detailed description.
According to a 1st aspect of the present invention, in an audio coding method in which a digital audio signal is divided into a plurality of frequency subbands and a coding process is performed for each subband, there are provided plural bit allocation means according to different processing amounts, for generating bit allocation information for each subband, and bit allocation means to be used is changed to perform bit allocation according to external control information such that bit allocation means is selected from the plural bit allocation means and used, whereby the coding process is performed. Therefore, bit allocation means according to an optimum processing amount is always selected and used, and a coding process in which the amount of processing on the CPU which can be occupied by the coding process is not exceeded is realized in an active state. Thereby, when coding the input signal in real time, processing of the input signal will not be delayed. As a result, audio can be reproduced with no discontinuity.
According to a 2nd aspect of the present invention, in the audio coding method of the 1st aspect, a load value indicating a processing amount of a central processing unit which can be occupied by the coding process is used as the external control information, and the bit allocation means is selected such that the processing amount of the central processing unit which can be occupied by the coding process is not exceeded, according to the load value, with reference to a data table which contains respective processing amounts of coding operation by the respective bit allocation means in the coding process on the central processing unit. Therefore, the central processing unit does not accept a request beyond its processing ability, whereby the whole system is controlled smoothly.
According to a 3rd aspect of the present invention, in the audio coding method of the second aspect, processing amount control information from monitoring means for monitoring a processing amount of the central processing unit which can be occupied by the coding process is used as the load value. Therefore, within the highest performance of the central processing unit which can be occupied by the coding process, bit allocation means according to the optimum processing amount is selected. Thereby, when coding the input signal in real time, processing of the signal will not be delayed. As a result, audio can be reproduced with no discontinuity.
According to a 4th aspect of the present invention, in the audio coding method of the 1st aspect, the bit allocation performed by the bit allocation means includes: a process using highly-efficient bit allocation for performing bit allocation with higher efficiency, which realizes high-quality coded data; and a process using low-load bit allocation for performing bit allocation with a lower-load, which performs processing less than the process using highly-efficient bit allocation. Therefore, the encoder carries out a coding process by using processing for higher-quality coded audio data or lower-load processing.
According to a 5th aspect of the present invention, in the audio coding method of the 1st aspect, the bit allocation means to be used in the coding process is changed frame by frame corresponding to a minimum unit decodable into an audio signal. Therefore, in the coding process in real time, when another application which occupies processing on the CPU

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