Data processing: speech signal processing – linguistics – language – Audio signal bandwidth compression or expansion
Reexamination Certificate
1998-05-28
2002-02-19
Dorvil, Richemond (Department: 2641)
Data processing: speech signal processing, linguistics, language
Audio signal bandwidth compression or expansion
C704S230000
Reexamination Certificate
active
06349284
ABSTRACT:
BACKGROUND OF THE INVENTION
1. Field of the Invention
The present invention relates to audio encoding and decoding, and more particularly, to a scalable audio encoding/decoding method and apparatus using bit-sliced arithmetic coding. The present invention is adopted as ISO/IEC JTC1/SC29/WG11 N1903 (ISO/IEC Committee Draft 14496-3 SUBPART 4).
2. Description of the Related Art
The MPEG audio standards or AC-2/AC-3 method provide almost the same audio quality as a compact disc, with a bitrate of 64~384 Kbps which is one-sixth to one-eighth that of conventional digital coding. For this reason, MPEG audio standards play an important role in storing and transmitting audio signals as in digital audio broadcasting (DAB), internet phone, or audio on demand (AOD).
Research into methods by which clear audio quality in its original sound can be reproduced at a lower bitrate have been ongoing. One method is an MPEG-2 Advanced Audio Coding (AAC) authorized as a new international standard. The MPEG-2 AAC providing the clear audio quality to the original sound at 64 kbps has been recommended by the experts group.
In conventional techniques, a fixed bitrate is given in an encoder, and the optimal state suitable for the given bitrate is searched to then perform quantization and coding, thereby exhibiting considerably better efficiency. However, with the advent of multimedia technology, there is an increasing demand for a coder/decoder (codec) having versatility at a low bitrate. One such demand is a scalable audio codec. The scalable audio codec can make bitstreams coded at a high bitrate into low bitrate bitstreams to then restore only some of them. By doing so, signals can be restored with a reasonable efficiency, with only some of the bitstreams, exhibiting little deterioration in performance due to lowered bitrates, when an overload is applied to the system or the performance of a decoder is poor, or by a user's request.
According to general audio coding techniques such as the MPEG-2 AAC standards, a fixed bitrate is given to a coding apparatus, the optimal state for the given bitrate is searched to then perform quantization and coding, thereby forming bitstreams in accordance with the bitrate. One bitstream contains information for one bitrate. In other words, bitrate information is contained in the header of a bitstream and a fixed bitrate is used. Thus, a method exhibiting the best efficiency at a specific bitrate can be used. For example, when a bitstream is formed by an encoder at a bitrate of 64 Kbps, the best quality sound can be restored by a decoder corresponding to an encoder having a bitrate of 64 Kbps.
According to such methods, bitstreams are formed without consideration of other bitrates, but bitstreams having a magnitude suitable for a given bitrate, rather than the order of the bitstreams, are formed. Actually, if the thus-formed bitstreams are transmitted via a communications network, the bitstreams are sliced into several slots to then be transmitted. When an overload is applied to a transmission channel, or only some of the slots sent from a transmission end are received at a reception end due to a narrow bandwidth of the transmission channel, data cannot be reconstructed properly. Also, since bitstreams are not formed according to the significance thereof, if only some of the bitstreams are restored, the quality is severely degraded. The reconstructed audio data makes sound objectionable to the ear.
In the case of a scalable audio codec for solving the above-described problems, coding for a base layer is performed and then a difference signal between the original signal and the coded signal is coded in the next enhancement layer (K. Brandenburg. Et al., “First Ideas on Scalable Audio Coding”, 97th AES-Conventional, preprint 3924, San Francisco, 1994) and (K. Brandenburg, et al., “A Two- or Three-Stage Bit Rate Scalable Audio Coding System”, 99th AES-Convention, preprint 4132, New York, 1995). Thus, the more layers there are the poorer the performance at a high bitrate. In the case of using a scalable coding apparatus, a signal having good audio quality is reproduced initially. However, if the state of communication channels is worsened or the load applied to the decoder of a receiving terminal is increased, a sound having a low bitrate quality is reproduced. Therefore, the aforementioned encoding method is not suitable for practically attaining scalability.
SUMMARY OF THE INVENTION
To solve the above problems, it is an objective of the present invention to provide a scalable digital audio data encoding method, apparatus, and recording medium for recording the encoding method, using a bit-sliced arithmetic coding (BSAC) technique, instead of a lossless coding module,with all other modules of the conventional coder remaining unchanged.
It is another objective of the present invention to provide a scalable digital audio data decoding method, apparatus, and recording medium for recording the decoding method, using a bit-sliced arithmetic coding (BSAC) technique, instead of a lossless coding module,with all other modules of the conventional audio decoder remaining unchanged.
To achieve the first objective of the present invention, there is provided a scalable audio encoding method for coding audio signals into a layered datastream having a base layer and enhancement layers of a predetermined number, comprising the steps of: signal-processing input audio signals and quantizing the same for each predetermined coding band; and packing the quantized data to generate bitstreams, wherein the bitstream generating step comprises: coding the quantized data corresponding to the base layer; coding the quantized data corresponding to the next enhancement layer of the coded base layer and the remaining quantized data uncoded due to a layer size limit and belonging to the coded layer; and sequentially performing the layer coding steps for all enhancement layers to form bitstreams, wherein the base layer coding step, the enhancement layer coding step and the sequential coding step are performed such that the side information and quantized data corresponding to a layer to be coded are represented by digits of a same predetermined number; and then arithmetic-coded using a predetermined probability model in the order ranging from the MSB sequences to the LSB sequences, the side information containing scale factors and probability models to be used in the arithmetic coding.
The step of coding the scale factors comprises the steps of: obtaining the maximum scale factor; and obtaining differences between the maximum scale factor and the respective scale factors and arithmetic-coding the differences.
When the quantized data is composed of sign data and magnitude data, the coding step comprises the steps of: coding by a predetermined encoding method the most significant bit sequences composed of most significant bits of the magnitude data of the quantized data represented by the same number of bits; coding sign data corresponding to non-zero data among the coded most significant bit sequences; coding the most significant bit sequences among uncoded magnitude data of the digital data by a predetermined encoding method; coding uncoded sign data among the sign data corresponding to non-zero magnitude data among bit sequences; and performing the magnitude data coding step and the sign data coding step on the respective bits of the digital data.
The coding steps are performed by coupling bits composing the respective bit sequences for the magnitude data and sign data, into units of bits of a predetermined number.
A four-dimensional vector coupled in units of bits is divided into two subvectors according to its pre-states in coding the respective samples.
The bitrate of the base layer is 16 kbps and the interlayer bitrate is 8 kbps.
To achieve the second objective of the present invention, there is provided a scalable audio coding apparatus comprising: a quantizing portion for signal-processing input audio signals and quantizing the same for each coding band; and a bit packing portion for generatin
Kim Yeon-bae
Park Sung-hee
Burns Doane Swecker & Mathis L.L.P.
Dorvil Richemond
McFadden Susan
Samsung SDI & Co., Ltd.
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