TECHNICAL FIELD

The present invention relates to a technique for converting a sample sequence derived from a sound signal to a sample sequence compressed or decompressed based on a sample value in the vicinity of the sample sequence in signal processing technology such as sound signal encoding technology.

BACKGROUND ART

Generally, in lossy compression encoding, after a quantizing part17quantizes an input signal, a lossless encoding part18gives a code by lossless encoding such as entropy encoding, based on the quantized signal, and a multiplexing part19outputs a code corresponding to the quantized signal and a code corresponding to a quantization width together as shown inFIG. 1. At the time of decoding, after a demultiplexing part21takes out the signal code and the code corresponding to the quantization width, and a lossless decoding part22performs lossless decoding of the signal code, a dequantizing part23performs dequantization of the quantized signal that has been decoded, to obtain the original signal as shown inFIG. 2.

Especially in lossy compression encoding of a sound signal of voice, music and the like, a method is known in which analysis of the signal by an analyzing part15and a filtering process by a filtering part16are added before the quantization process ofFIG. 1, and a weight appropriate for aural characteristics is given according to the signal so that an error caused by quantization is aurally reduced as shown inFIG. 3(see Non-patent literature 1). In this conventional method, in addition to the code corresponding to the quantized signal and the code corresponding to the quantization width used for quantization, a code corresponding to a filter coefficient used for filtering is also sent to a decoder as auxiliary information, and the decoder obtains the original signal by an inverse filtering part24performing inverse filtering of the weighted signal that has been decoded, as post-processing of the dequantization process ofFIG. 2, as shown inFIG. 4.

PRIOR ART LITERATURE

Non-Patent Literature

Non-patent literature 1: Gerald D. T. Schuller, Bin Yu, Dawei Huang, and Bernd Edler, “Perceptual Audio Coding Using Adaptive Pre-and Post-Filters and Lossless Compression,” IEEE TRANSACTIONS ON SPEECH AND AUDIO PROCESSING, VOL. 10, NO. 6, SEPTEMBER 2002.

SUMMARY OF THE INVENTION

Problems to be Solved by the Invention

In the conventional technique described in Non-patent literature 1, the amount of necessary information increases by the filter coefficient in comparison with simple lossy compression encoding as shown inFIGS. 1 and 2. However, aural weighting is only required to satisfy the following two properties roughly classified, and strict information is often unnecessary. 1. In a frame, a relatively small weight is applied to a large value of a signal or a value of a frequency spectrum of the signal, and a relatively large weight is applied to a small value. 2. In a frame, in the vicinity of a peak of the signal or the frequency spectrum of the signal, a relatively small weight is applied similarly to the peak.

An object of the present invention is to, by having both of the above two properties and converting a sample sequence by pre-processing and post-processing that do not require auxiliary information for the post-processing, enhance aural quality of an encoding process and a decoding process for a sound signal.

Means to Solve the Problems

In order to solve the above problem, a sample sequence converter of a first aspect of the present invention is a sample sequence converter that obtains a weighted frequency domain signal obtained by converting a frequency domain signal corresponding to an input acoustic signal, the weighted frequency domain signal being to be inputted to an encoder encoding the weighted frequency domain signal, or a weighted frequency domain signal corresponding to a weighted time domain signal corresponding to the weighted frequency domain signal obtained by converting the frequency domain signal corresponding to the input acoustic signal, the weighted time domain signal being to be inputted to an encoder encoding the weighted time domain signal, the sample sequence converter comprising: a representative value calculating part calculating, for each frequency section by a plurality of samples fewer than the number of frequency samples of a sample sequence of the frequency domain signal corresponding to the input acoustic signal, from the sample sequence of the frequency domain signal, a representative value of the frequency section from sample values of samples included in the frequency section, for each of predetermined time sections; and a signal companding part obtaining, for each of the predetermined time sections, a frequency domain sample sequence obtained by multiplying a weight according to a function value of the representative value by a companding function for which an inverse function can be defined and each of the samples corresponding to the representative value in the sample sequence of the frequency domain signal, as a sample sequence of the weighted frequency domain signal.

A sample sequence converter of a second aspect of the present invention is sample sequence converter that obtains a frequency domain signal corresponding to a decoded acoustic signal from a weighted frequency domain signal obtained by a decoder or a weighted frequency domain signal corresponding to the weighted time domain signal obtained by the decoder, the sample sequence converter comprising: a companded representative value calculating part calculating, for each frequency section by a plurality of samples fewer than the number of frequency samples of a sample sequence of the weighted frequency domain signal, from the sample sequence of the weighted frequency domain signal, a representative value of the frequency section from sample values of samples included in the frequency section, for each of predetermined time sections; and a signal decompanding part obtaining, for each of the predetermined time sections, a frequency domain sample sequence obtained by multiplying a weight according to a function value of the representative value by a companding function for which an inverse function can be defined and each of the samples corresponding to the representative value in the sample sequence of the weighted frequency domain signal, as a sample sequence of the frequency domain signal corresponding to the decoded acoustic signal.

A sample sequence converter of a third aspect of the present invention is sample sequence converter that obtains a weighted acoustic signal obtained by converting an input acoustic signal, the weighted acoustic signal being to be inputted to an encoder encoding the weighted acoustic signal, or a weighted acoustic signal corresponding to a weighted frequency domain signal corresponding to the weighted acoustic signal obtained by converting the input acoustic signal, the weighted frequency domain signal being to be inputted to an encoder encoding the weighted frequency domain signal, the sample sequence converter comprising: a representative value calculating part calculating, for each time section by a plurality of samples fewer than the number of samples of a sample sequence of the input acoustic signal in a time domain, from the sample sequence of the input acoustic signal, a representative value of the time section from sample values of samples included in the time section, for each of predetermined time sections; and a signal companding part obtaining, for each of the predetermined time sections, a time domain sample sequence obtained by multiplying a weight according to a function value of the representative value by a companding function for which an inverse function can be defined and each of the samples corresponding to the representative value in the sample sequence of the input acoustic signal, as a sample sequence of the weighted acoustic signal.

A sample sequence converter of a fourth aspect of the present invention is a sample sequence converter that obtains a decoded acoustic signal from a weighted acoustic signal in a time domain obtained by a decoder or a weighted acoustic signal in the time domain corresponding to a weighted acoustic signal in a frequency domain obtained by the decoder, the sample sequence converter comprising: a companded representative value calculating part calculating, for each time section by a plurality of samples fewer than the number of samples of a sample sequence of the weighted acoustic signal in the time domain, from the sample sequence of the weighted acoustic signal, a representative value of the time section from sample values of samples included in the time section, for each of predetermined time sections; and a signal decompanding part obtaining, for each of the predetermined time sections, a time domain sample sequence obtained by multiplying a weight according to a function value of the representative value by a companding function for which an inverse function can be defined and each of the samples corresponding to the representative value in the sample sequence of the weighted acoustic signal, as a sample sequence of the decoded acoustic signal.

Effects of the Invention

According to the present invention, it is possible to, by having both of two properties required for aural weighting, and converting a sample sequence by pre-processing and post-processing that do not require auxiliary information for the post-processing, enhance aural quality of an encoding process and a decoding process for a sound signal.

