Scalable coder and decoder performing amplitude flattening for error spectrum estimation

A down-sampler 101 down-samples the sampling rate of an input signal from sampling rate FH to sampling rate FL. A base layer coder 102 encodes the sampling rate FL acoustic signal. A local decoder 103 decodes coding information output from base layer coder 102. An up-sampler 104 raises the sampling rate of the decoded signal to FH. A subtracter 106 subtracts the decoded signal from the sampling rate FH acoustic signal. An enhancement layer coder 107 encodes the signal output from subtracter 106 using a decoding result parameter output from local decoder 103.

TECHNICAL FIELD

The present invention relates to a coding apparatus, decoding apparatus, coding method, and decoding method that perform highly efficient compression coding of an acoustic signal such as an audio signal or speech signal, and more particularly to a coding apparatus, decoding apparatus, coding method, and decoding method that are suitable for scalable coding and decoding that enable decoding of audio or speech even from a part of coding information.

BACKGROUND ART

A sound coding technology that compresses an audio signal or speech signal at a low bit rate is important for efficient utilization of radio in mobile communications and recording media. Methods for speech coding, in which a speech signal is coded, include G726 and G729 standardized by the ITU (International Telecommunication Union). These methods encode narrowband signals (300 Hz to 3.4 kHz), and enable high-quality coding at bit rates of 8 kbits/s to 32 kbits/s.

Standard methods for wideband signals (50 Hz to 7 kHz) include the ITU's G722 and G722.1, and AMR-WB of 3GPP (The 3rd Generation Partnership Project). These methods enable high-quality coding of wideband speech signals at bit rates of 6.6 kbits/s to 64 kbits/s.

An effective method of performing highly efficient coding of speech signals at a low bit rate is CELP (Code Excited Linear Prediction). CELP is a method whereby coding is performed based on a model that simulates through engineering a human voice generation model. To be specific, in CELP, an excitation signal which consists of random values is passed to a pitch filter corresponding to the strength of periodicity and a synthesis filter corresponding to vocal tract characteristics, and coding parameters are determined so that the square error between the output signal and input signal is minimized under auditory characteristic weighting.

In many of the latest standard speech coding methods, coding is performed based on CELP. For example, G729 enables narrowband signal coding at 8 kbits/s, and AMR-WB enables narrowband signal coding at 6.6 kbits/s to 23.85 kbits/s.

Meanwhile, in the case of audio coding that encodes audio signals, methods that convert an audio signal to frequency domain and perform coding using an auditory psychoacoustic model are commonly used, such as the Layer III method and AAC method standardized by MPEG (Moving Picture Experts Group). It is known that with these methods, almost no degradation occurs at 64 kbits/s to 96 kbits/s per channel for a signal with a 44.1 kHz sampling rate.

This audio coding is a method whereby high-quality coding is performed on music. Audio coding can also perform high-quality coding for a speech signal with music or environmental sound in the background as described above, and can handle a signal band of approximately 22 kHz, which is CD quality.

However, when coding is performed using a speech coding method on a signal in which a speech signal is predominant and music or environmental sound is superimposed in the background, there is a problem in that, due to the background music or environmental sound, not only the background signal but also the speech signal degrades, and overall quality deteriorates.

This problem occurs because speech coding methods are based on a method specialized toward a CELP speech model. There is a problem in that speech coding methods can only handle signal bands up to 7 kHz, and a signal that has components in higher bands cannot be handled adequately in terms of composition.

Moreover, with an audio coding method, a high bit rate must be used in order to achieve high-quality coding. With an audio coding method, if coding should be performed with the bit rate held down to 32 kbits/s, there is a problem of a major deterioration of decoded signal quality. There is thus a problem in that use is not possible on a communication network with a low transmission rate.

DISCLOSURE OF INVENTION

It is an object of the present invention to provide a coding apparatus, decoding apparatus, coding method, and decoding method that enable high-quality coding and decoding at a low bit rate even of a signal in which a speech signal is predominant and music or environmental sound is superimposed in the background.

This object is achieved by having two layers, a base layer and an enhancement layer, performing high-quality coding at a low bit rate of an input signal narrowband or wideband frequency region based on CELP in the base layer, and performing coding in the enhancement layer of background music or environmental sound that cannot be represented in the base layer, and also signals with higher frequency components than the frequency region covered by the base layer.

BEST MODE FOR CARRYING OUT THE INVENTION

Essentially, the present invention has two layers, a base layer and an enhancement layer, performs high-quality coding at a low bit rate of an input signal narrowband or wideband frequency region based on CELP in the base layer, and then performs coding in the enhancement layer of background music or environmental sound that cannot be represented in the base layer, and also signals with higher frequency components than the frequency region covered by the base layer, with the enhancement layer having a configuration that enables handling of all signals as with an audio coding method.

By this means, it is possible to perform efficient coding of background music or environmental sound that cannot be represented in the base layer, and also signals with higher frequency components than the frequency region covered by the base layer. A feature of the present invention is that, at this time, enhancement layer coding is performed using information obtained by base layer coding information. By this means, an effect is obtained of being able to keep down the number of enhancement layer coded bits.

With reference now to the accompanying drawings, embodiments of the present invention will be explained in detail below.

FIG. 1is a block diagram showing the configuration of a signal processing apparatus according to Embodiment 1 of the present invention. Signal processing apparatus100inFIG. 1mainly comprises a down-sampler101, base layer coder102, local decoder103, up-sampler104, delayer105, subtracter106, enhancement layer coder107, and multiplexer108.

Down-sampler101down-samples the input signal sampling rate from sampling rate FH to sampling rate FL, and outputs the sampling rate FL acoustic signal to base layer coder102. Here, sampling rate FL is a lower frequency than sampling rate FH.

Base layer coder102encodes the sampling rate FL acoustic signal and outputs the coding information to local decoder103and multiplexer108.

Local decoder103decodes the coding information output from base layer coder102, outputs the decoded signal to up-sampler104, and outputs parameters obtained from the decoded result to enhancement layer coder107.

Up-sampler104raises the decoded signal sampling rate to FH, and outputs the result to subtracter106.

Delayer105delays the input sampling rate FH acoustic signal by a predetermined time, then outputs the signal to subtracter106. By making this delay time equal to the time delay arising in down-sampler101, base layer coder102, local decoder103, and up-sampler104, phase shift is prevented in the following subtraction processing.

Subtracter106subtracts the decoded signal from the sampling rate FH acoustic signal, and outputs the result of the subtraction to enhancement layer coder107.

Base layer coding and enhancement layer coding will now be explained.FIG. 2is a drawing showing an example of input signal components. InFIG. 2, the vertical axis indicates the signal component information amount, and the horizontal axis indicates frequency.FIG. 2shows the frequency bands in which speech information and background music/background noise information contained in the input signal are present.

In the case of speech information, there is a large amount of information in the low frequency region, and the amount of information decreases the higher the frequency region. Conversely, in the case of background music and background noise information, there is comparatively little information in the lower region compared with speech information, and a large amount of information in the higher region.

Thus, a signal processing apparatus of the present invention uses a plurality of coding methods, and performs different coding for each region for which the respective coding methods are appropriate.

FIG. 3is a drawing showing an example of a signal processing method of a signal processing apparatus according to this embodiment. InFIG. 3, the vertical axis indicates the signal component information amount, and the horizontal axis indicates frequency.

Base layer coder102is designed to represent efficiently speech information in the frequency band from0to FL, and can perform good-quality coding of speech information in this region. However, the coding quality of background music and background noise information in the frequency band from0to FL is not high. Enhancement layer coder107encodes portions that cannot be coded by base layer coder102, and signals in the frequency band from FL to FH.

Thus, by combining base layer coder102and enhancement layer coder107, it is possible to achieve high-quality coding in a wide band. Moreover, a scalable function can be implemented whereby speech information can be decoded even with only coding information of at least a base layer coding section.

In this way, a useful parameter from among those generated by coding in local decoder103is supplied to enhancement layer coder107, and enhancement layer coder107performs coding using this parameter.

As this parameter is generated from coding information, when a signal coded by a signal processing apparatus of this embodiment is decoded, the same parameter can be obtained in the sound decoding process, and it is not necessary to add this parameter for transmission to the decoding side. As a result, the enhancement layer coding section can achieve efficient coding processing without incurring an increase in additional information.

For example, there is a configuration whereby, of the parameters decoded by local decoder103, a voiced/unvoiced flag, indicating whether an input signal is a signal with marked periodicity such as a vowel or a signal with marked noise characteristics such as a consonant, is used as a parameter employed by enhancement layer coder107. It is possible to perform adaptation using the voiced/unvoiced flag, such as performing bit allocation stressing the lower region more than the higher region in the enhancement layer in a voiced section, and performing bit allocation stressing the higher region more than the lower region in an unvoiced section.

Thus, according to a signal processing apparatus of this embodiment, by extracting components not exceeding a predetermined frequency from an input signal and performing coding suitable for speech coding, and performing coding suitable for audio coding using the results of decoding the obtained coding information, it is possible to perform high-quality coding at a low bit rate.

For sampling rates FH and FL, it is only necessary for FH to be higher value than FL, and there are no restrictions on the values. For example, coding can be performed with sampling rates of FH=24 kHz and FL=16 kHz.

In this embodiment an example is described in which, of the parameters decoded by local decoder103of Embodiment 1, LPC coefficients indicating the input signal spectrum is used as a parameter utilized by enhancement layer coder107.

A signal processing apparatus of this embodiment performs coding using CELP in base layer coder102inFIG. 1, and performs coding using LPC coefficients indicating the input signal spectrum in enhancement layer coder107.

A detailed description of the operation of base layer coder102will first be given, followed by a description of the basic configuration of enhancement layer coder107. The “basic configuration” mentioned here is intended to simplify the descriptions of subsequent embodiments, and denotes a configuration that does not use local decoder103coding parameters. Thereafter, a description is given of enhancement layer coder107, which uses the LPC coefficients decoded by local decoder103, this being a feature of this embodiment.

FIG. 4is a drawing showing an example of the configuration of base layer coder102. Base layer coder102mainly comprises an LPC analyzer401, weighting section402, adaptive code book search unit403, adaptive gain quantizer404, target vector generator405, noise code book search unit406, noise gain quantizer407, and multiplexer408.

LPC analyzer401obtains LPC coefficients from the input signal sampled at sampling rate FL by down-sampler101, and outputs these LPC coefficients to weighting section402.

Weighting section402performs weighting on the input signal based on the LPC coefficients obtained by LPC analyzer401, and outputs the weighted input signal to adaptive code book search unit403, adaptive gain quantizer404, and target vector generator405.

Adaptive code book search unit403carries out an adaptive code book search with the weighted input signal as the target signal, and outputs the retrieved adaptive vector to adaptive gain quantizer404and target vector generator405. Adaptive code book search unit403then outputs the code of the adaptive vector determined to have the least quantization distortion to multiplexer408.

Adaptive gain quantizer404quantizes the adaptive gain that is multiplied by the adaptive vector output from adaptive code book search unit403, and outputs the result to target vector generator405. This code is then output to multiplexer408.

Target vector generator405performs vector subtraction of the input signal output from weighting section402from the result of multiplying the adaptive vector by the adaptive gain, and outputs the result of the subtraction to noise code book search unit406and noise gain quantizer407as the target vector.

Noise code book search unit406retrieves from a noise code book the noise vector for which distortion relative to the target vector output from target vector generator405is smallest. Noise code book search unit406then supplies the retrieved noise vector to noise gain quantizer407and also outputs that code to multiplexer408.

Noise gain quantizer407quantizes noise gain that is multiplied by the noise vector retrieved by noise code book search unit406, and outputs that code to multiplexer408.

Multiplexer408multiplexes the LPC coefficients, adaptive vector, adaptive gain, noise vector, and noise gain coding information, and outputs the resulting signal to local decoder103and multiplexer108.

Next, the operation of base layer coder102inFIG. 4will be described. First, a sampling rate FL signal output from down-sampler101is input, and LPC coefficients are obtained by LPC analyzer401. The LPC coefficients are converted to a parameter suitable for quantization such as LSP coefficients, and quantized. The coding information obtained by this quantization is supplied to multiplexer408, and the quantized LSP coefficients are calculated from the coding information and converted to LPC coefficients.

By means of this quantization, the quantized LPC coefficients are obtained. Using the quantized LPC coefficients, adaptive code book, adaptive gain, noise code book, and noise gain coding is performed.

Weighting section402then performs weighting on the input signal based on the LPC coefficients obtained by LPC analyzer401. The purpose of this weighting is to perform spectrum shaping so that the quantization distortion spectrum is masked by the spectral envelope of the input signal.

The adaptive code book is then searched by adaptive code book search unit403with the weighted input signal as the target signal. A signal in which a past excitation sequence is repeated on a pitch period basis is called an adaptive vector, and an adaptive code book is composed of adaptive vectors generated at pitch periods of a predetermined range.

If a weighted input signal is designated t(n), and a signal in which an impulse response of a weighted synthesis filter comprising the LPC coefficients is convoluted to the adaptive vector of pitch period i is designated pi(n), then pitch period i of the adaptive vector for which evaluation function D of Equation (1) below is minimized is sent to multiplexer408as a parameter.

D=∑n=0N-1⁢t2⁡(n)-(∑n=0N-1⁢t⁡(n)⁢⁢p⁢⁢i⁡(n))2∑n=0N-1⁢p⁢⁢i2⁡(n)(1)
Here, N indicates the vector length.

Next, quantization of the adaptive gain that is multiplied by the adaptive vector is performed by adaptive gain quantizer404. Adaptive gain β is expressed by Equation (2). This β value undergoes scalar quantization, and the resulting code is sent to multiplexer408.

The effect of the adaptive vector is then subtracted from the input signal by target vector generator405, and the target vector used by noise code book search unit406and noise gain quantizer407is generated. If pi(n) here designates a signal in which the synthesis filter is convoluted to the adaptive vector when evaluation function D expressed by Equation (1) is minimized, and βq designates the quantization value when adaptive vector β expressed by Equation (2) undergoes scalar quantization, then target vector t2(n) is expressed by Equation (3) below.
t2(n)=t(n)−βq·pi(n)  (3)

Aforementioned target vector t2(n) and the LPC coefficients are supplied to noise code book search unit406, and a noise code book search is carried out.

Here, a typical composition of the noise code book with which noise code book search unit406is provided is algebraic. In an algebraic code book, an amplitude 1 pulse is represented by a vector that has only a predetermined extremely small number. Also, with an algebraic code book, positions that can be held for each phase are decided beforehand so as not to overlap. Thus, a feature of an algebraic code book is that an optimal combination of pulse position and pulse code (polarity) can be determined by a small amount of computation.

If the target vector is designated t2(n), and a signal in which an impulse response of a weighted synthesis filter is convoluted to the noise vector corresponding to code j is designated cj(n), then index j of the noise vector for which evaluation function D of Equation (4) below is minimized is sent to multiplexer408as a parameter.

Next, quantization of the noise gain that is multiplied by the noise vector is performed by noise gain quantizer407. Adaptive gain γ is expressed by Equation (5). This γ value undergoes scalar quantization, and the resulting code is sent to multiplexer408.

Multiplexer408multiplexes the sent LPC coefficients, adaptive code book, adaptive gain, noise code book, and noise gain coding information, and outputs the resulting signal to local decoder103and multiplexer108.

The above processing is repeated while there is a new input signal. When there is no new input signal, processing is terminated.

Enhancement layer coder107will now be described.FIG. 5is a drawing showing an example of the configuration of enhancement layer coder107. Enhancement layer coder107inFIG. 5mainly comprises an LPC analyzer501, spectral envelope calculator502, MDCT section503, power calculator504, power normalizer505, spectrum normalizer506, Bark scale normalizer508, Bark scale shape calculator507, vector quantizer509, and multiplexer510.

LPC analyzer501performs LPC analysis on an input signal. And the LPC analyzer501quantizes the LPC coefficients effectively in the domain of LSP or other adequate parameter for quantization, and the LPC analyzer outputs the coding information to multiplexer, and the LPC analyzer outputs the quantized LPC coefficients to spectral envelope calculator502. Spectral envelope calculator502calculates a spectral envelope from the quantized LPC coefficients, and outputs this spectral envelope to vector quantizer509.

MDCT section503performs MDCT (Modified Discrete Cosine Transform) processing on the input signal, and outputs the obtained MDCT coefficients to power calculator504and power normalizer505. Power calculator504finds and quantizes the power of the MDCT coefficients, and outputs the quantized power to power normalizer505and the coding information to multiplexer510.

Power normalizer505normalizes the MDCT coefficients with the quantized power, and outputs the power-normalized MDCT coefficients to spectrum normalizer506. Spectrum normalizer506normalizes the MDCT coefficients normalized according to the power using the spectral envelope, and outputs the normalized MDCT coefficients to Bark scale shape calculator507and Bark scale normalizer508.

Bark scale shape calculator507calculates the shape of a spectrum band-divided at equal intervals by means of a Bark scale, then quantizes this spectrum shape, and outputs the quantized spectrum shape to Bark scale normalizer508, vector quantizer509. And the bark scale shape calculator507outputs the coding information to multiplexer510.

Bark scale normalizer normalizes the normalized MDCT coefficients using quantized bark scale shape, which it outputs to vector quantizer509.

Vector quantizer509performs vector quantization of the normalized MDCT coefficients output from Bark scale normalizer508, finds the code-vector at which distortion is smallest, and outputs the index of the code-vector to multiplexer510as coding information.

Multiplexer510multiplexes all of the coding information, and outputs the resulting signal to multiplexer108.

The operation of enhancement layer coder107inFIG. 5will now be described. The subtraction signal obtained by subtracter106inFIG. 1undergoes LPC analysis by LPC analyzer501. Then the LPC coefficients are calculated by LPC analysis. The LPC coefficients are converted to a parameter suitable for quantization such as LSP coefficients, after which quantization is performed. Coding information related to the LPC coefficients obtained here is supplied to multiplexer510.

Spectral envelope calculator502calculates a spectral envelope in accordance with Equation (6) below, based on the decoded LPC coefficients.

Here, αq denotes the decoded LPC coefficients, NP indicates the order of the LPC coefficients, and M the spectral resolution. Spectral envelope env(m) obtained by means of Equation (6) is used by spectrum normalizer506and vector quantizer509described later herein.

The input signal then undergoes MDCT processing in MDCT section503, and the MDCT coefficients are obtained. A feature of MDCT processing is that frame boundary distortion does not occur because of the use of an orthogonal base whereby the analysis frame of successive frames are completely superimposed one-half at a time, and the first half of the analysis frame is an odd function while the latter half of the analysis frame is an even function. When MDCT processing is performed, the input signal is multiplied by a window function such as a sin window. Designating the MDCT coefficients X(m), the MDCT coefficients are calculated in accordance with Equation (7) below.

X⁡(m)=1N⁢∑n=02⁢N-1⁢x⁡(n)⁢⁢cos⁢{(2⁢n+1+N)·(2⁢m+1)⁢⁢π4⁢N}(7)
Here, x(n) indicates the signal when the input signal is multiplied by a window function.

Next, power calculator504finds and quantizes the power of MDCT coefficients X(m). Power normalizer505then normalizes the MDCT coefficients with the power after that quantization using Equation (8).

Here, M indicates the size of the MDCT coefficients. After MDCT coefficient power pow has been quantized, the coding information is sent to multiplexer510. The power of the MDCT coefficients is decoded using the coding information, and the MDCT coefficients are normalized in accordance with Equation (9) below using the resulting value.

X1⁡(m)=X⁡(m)powq(9)
Here, X1(m) represents the MDCT coefficients after power normalization, and powq indicates the power of the MDCT coefficients after quantization.

Spectrum normalizer506then normalizes the MDCT coefficients that has been normalized according to power using the spectral envelope. Spectrum normalizer506performs normalization in accordance with Equation (10) below.

Next, Bark scale shape calculator507calculates the shape of a spectrum band-divided at equal intervals by means of a Bark scale, then quantizes this spectrum shape. Bark scale shape calculator507sends this coding information to multiplexer510, and also performs normalization of MDCT coefficients X2(m), which is the output signal from spectrum normalizer506, using the decoded value. The correspondence between the Bark scale and Herz scale is given by the conversion expression represented by Equation (11) below.

B=13⁢⁢tan-1⁡(0.76⁢⁢f)=3.5⁢⁢tan-1(f7.5)(11)
Here, B indicates the Bark scale and f the Herz scale. Bark scale shape calculator507calculates a shape in accordance with Equation (12) below for the sub-bands band-divided at equal intervals on the Bark scale.

B⁡(k)=∑m=f⁢⁢l⁢⁢(k)f⁢⁢h⁡(k)⁢X2⁡(m)2⁢⁢0≤k<K(12)
Here, fl(k) indicates the lowest frequency of the k'th sub-band and fh(k) the highest frequency of the k'th sub-band, and K indicates the number of sub-bands.

Bark scale shape calculator507then quantizes Bark scale shape B(k) of each band and sends the coding information to multiplexer510, and also decodes the Bark scale shape and supplies the result to Bark scale normalizer508and vector quantizer509. Using the Bark scale shape after normalization, Bark scale normalizer508generates normalized MDCT coefficients X3(m) in accordance with Equation (13) below.

X3⁡(m)=X2⁡(m)Bq⁡(k)⁢⁢f⁢⁢l⁢⁢(k)≤m≤f⁢⁢h⁢⁢(k)⁢⁢0≤k<K(13)
Here, Bq(k) indicates the Bark scale shape after quantization of the k'th sub-band.

Next, vector quantizer509performs vector quantization of Bark scale normalizer508output X3(m) Vector quantizer509divides X3(m) into a plurality of vectors and finds the code-vector at which distortion is smallest using a code book corresponding to each vector, and sends this index to multiplexer510as coding information.

When performing vector quantization, vector quantizer509determines two important parameters using input signal spectrum information. One of these parameters is quantization bit allocation, and the other is code book search weighting. Quantization bit allocation is determined using spectral envelope env(m) obtained by spectral envelope calculator502.

When quantization bit allocation is determined using spectral envelope env(m), a setting can also be made so that the number of bits allocated in the spectrum corresponding to frequencies0to FL is made small.

One example of implementation of this is a method whereby the maximum number of bits that can be allocated in frequencies0to FL, MAX_LOWBAND_BIT, is set, and a restriction is imposed so that the maximum number of bits allocated in this band does not exceed maximum number of bits MAX_LOWBAND_BIT.

In this implementation example, since coding has already been performed in the base layer at frequencies0to FL, it is not necessary to allocate a large number of bits, and overall quality can be improved by performing quantization with quantization in this band intentionally made coarse and bit allocation kept low, and the extra bits being allocated to frequencies FL to FH. A configuration may also be used where by this bit allocation is determined by combining spectral envelope env(m) and aforementioned Bark scale shape Bq(k).

Vector quantization is performed using a distortion measure employing spectral envelope env(m) obtained by spectral envelope calculator502and weighting calculated from quantized Bark scale shape Bq(k) obtained by Bark scale shape calculator507. Vector quantization is implemented by finding index j of code vector C for which distortion D stipulated by Equation (14) below is minimal.

Weighting function w(m) can be expressed as shown in Equation (15) below using spectral envelope env(m) and Bark scale shape Bq(k).
w(m)=(env(m)·Bq(Herz_to_Bark(m)))p(15)
Here, p indicates a constant between 0 and 1, and Herz_to_Bark( ) indicates a function that converts from the Herz scale to Bark scale.

When weighting function w(m) is determined, it is also possible to make a setting so that the weighting function for bit allocation to the spectrum corresponding to frequencies0to FL is made small. One example of implementation of this is a method whereby the maximum value possible for weighting function w(m) corresponding to frequencies0to FL is set below as MAX_LOWBAND_WGT, and a restriction is imposed so that the value of weighting function w(m) for this band does not exceed MAX_LOWBAND_WGT. In this implementation example, coding has already been performed in the base layer at frequencies0to FL, and overall quality can be improved by intentionally lowering the quantization precision in this band and relatively raising the quantization precision for frequencies FL to FH.

Lastly, multiplexer510multiplexes the coding information and outputs the resultant signal to multiplexer108. The above processing is repeated while there is a new input signal. When there is no new input signal, processing is terminated.

Thus, according to a signal processing apparatus of this embodiment, by extracting components not exceeding a predetermined frequency from an input signal and performing coding using code excited linear prediction, and performing coding by MDCT processing using the results of decoding obtained coding information, it is possible to perform high-quality coding at a low bit rate.

An example has been described above in which the LPC coefficients are analyzed from a subtraction signal obtained by subtracter106, but a signal processing apparatus of the present invention may also perform decoding using the LPC coefficients decoded by local decoder103.

FIG. 6is a drawing showing an example of the configuration of enhancement layer coder107. Parts inFIG. 6identical to those inFIG. 5are assigned the same reference numerals as inFIG. 5and detailed descriptions thereof are omitted.

Enhancement layer coder107inFIG. 6differs from enhancement layer coder107inFIG. 5in being provided with a conversion table601, LPC coefficient mapping section602, spectral envelope calculator603, and transformation section604, and performing coding using the LPC coefficients decoded by local decoder103.

Conversion table601stores base layer LPC coefficients and enhancement layer LPC coefficients with the correspondence therebetween indicated.

LPC coefficient mapping section602references conversion table601, converts the base layer LPC coefficients input from local decoder103to the enhancement layer LPC coefficients, and outputs the enhancement layer LPC coefficients to spectral envelope calculator603.

Spectral envelope calculator603obtains a spectral envelope based on the enhancement layer LPC coefficients, and outputs this spectral envelope to transformation section604. Transformation section604transforms the spectral envelope and outputs the result to spectrum normalizer506and vector quantizer509.

The operation of enhancement layer coder107inFIG. 6will now be described. The base layer LPC coefficients are found for signals in signal band0to FL, and does not coincide with the LPC coefficients used by an enhancement layer signal (signal band0to FH). However, there is a strong correlation between the two. Therefore, in LPC coefficient mapping section602, a conversion table601is separately designed in advance, showing the correspondence between LPC coefficients for signal band0to FL signals and signal band0to FH signals, using this correlation. This conversion table601is used to find the enhancement layer LPC coefficients from the base layer LPC coefficients.

FIG. 7is a drawing showing an example of enhancement layer LPC coefficient calculation. Conversion table601is composed of J candidates {Yj(m)} indicating the enhancement layer LPC coefficients (order M), and candidates {yj(k)} that have the same order (=K) as the base layer LPC coefficients assigned correspondence to {Yj(m)}. {Yj(m)} and {yj(k)} are designed and provided beforehand from large-scale audio and speech data, etc. When base layer LPC coefficients x(k) are input, the sequence of the LPC coefficients most similar to x(k) is found from among {yj(k)}. By outputting enhancement layer LPC coefficients Yj(m) corresponding to index j of the LPC coefficients determined to be most similar, it is possible to implement mapping of the enhancement layer LPC coefficients from base layer LPC coefficients.

Next, spectral envelope calculator603obtains a spectral envelope based on the enhancement layer LPC coefficients found in this way. Then this spectral envelope is transformed by transformation section604. This transformed spectral envelope is then regarded as a spectral envelope of the implementation example described above, and is processed accordingly.

One example of implementation of transformation section604that transforms a spectral envelope is processing whereby the effect of a spectral envelope corresponding to signal band0to FL subject to base layer coding is made small. If the spectral envelope is designated env(m), transformed spectral envelope env′(m) is expressed by Equation (16) below.

env′⁡(m)={env⁡(m)pif⁢⁢0≤m≤Flenv⁡(m)else(16)
Here, p indicates a constant between 0 and 1.

Coding has already been performed in the base layer at frequencies0to FL, and the spectrum of frequencies0to FL of a subtraction signal subject to enhancement layer coding is close to flat. Irrespective of this, such action is not considered in LPC coefficient mapping as described in this implementation example. Quality can therefore be improved by using a technique of correcting the spectral envelope using Equation (16).

Thus according to a signal processing apparatus of this embodiment, by finding the enhancement layer LPC coefficients using the LPC coefficients quantized by a base layer quantizer, and calculating a spectral envelope from enhancement layer LPC analysis, LPC analysis and quantization are made unnecessary, and the number of quantization bits can be reduced.

FIG. 8is a block diagram showing the configuration of the enhancement layer coder of a signal processing apparatus according to Embodiment 3 of the present invention. Parts inFIG. 8identical to those inFIG. 5are assigned the same reference numerals as inFIG. 5and detailed descriptions thereof are omitted.

Enhancement layer coder107inFIG. 8differs from the enhancement layer coder inFIG. 5in being provided with a spectral fine structure calculator801, calculating spectral fine structure using a pitch period coded by base layer coder102and decoded by local decoder103, and employing that spectral fine structure in spectrum normalization and vector quantization.

Spectral fine structure calculator801calculates the spectral fine structure from pitch period T and pitch gain β coded in the base layer, and outputs the spectral fine structure to spectrum normalizer506.

The aforementioned pitch period T and pitch gain β are actually parts of the coding information, and the same information can be obtained by a local decoder (shown inFIG. 1). Thus the bit rate does not increase even if coding is performed using pitch period T and pitch gain β.

Using pitch period T and pitch gain β, spectral fine structure calculator801calculates spectral fine structure har(m) in accordance with Equation (17) below.

har⁡(m)=11-β·ⅇ-j⁢2⁢⁢π⁢⁢m⁢⁢TM(17)
Here, M indicates the spectral resolution. As Equation (17) is an oscillation filter when the absolute value of β is greater than or equal to 1, there is also a method whereby a restriction is set so that the possible range of the absolute value of β is less than or equal to a predetermined set value less than 1 (for example, 0.8).

Spectrum normalizer506performs normalization in accordance with Equation (18) below, using both spectral envelope env(m) obtained by spectral envelope calculator502and spectral fine structure har(m) obtained by spectral fine structure calculator801.

The allocation of quantization bits by vector quantizer509is also determined using both spectral envelope env(m) obtained by spectral envelope calculator502and spectral fine structure har(m) obtained by spectral fine structure calculator801. The spectral fine structure is also used in weighting function w(m) determination in vector quantization. To be specific, weighting function w(m) is defined in accordance with Equation (19) below.
w(m)=(env(m)·har(m)·Bq(Herz_to_Bark(m)))p(19)
Here, p indicates a constant between 0 and 1, and Herz_to_Bark( ) indicates a function that converts from the Herz scale to Bark scale.

Thus, according to a signal processing apparatus of this embodiment, by calculating a spectral fine structure using a pitch period coded by a base layer coder and decoded by a local decoder, and using that spectral fine structure in spectrum normalization and vector quantization, quantization performance can be improved.

FIG. 9is a block diagram showing the configuration of the enhancement layer coder of a signal processing apparatus according to Embodiment 4 of the present invention. Parts inFIG. 9identical to those inFIG. 5are assigned the same reference numerals as inFIG. 5and detailed descriptions thereof are omitted.

Enhancement layer coder107inFIG. 9differs from the enhancement layer coder inFIG. 5in being provided with a power estimation unit901and power fluctuation amount quantizer902, and in generating a decoded signal in local decoder103using coding information obtained by base layer coder102, predicting MDCT coefficients power from that decoded signal, and coding the amount of fluctuation from that predicted value.

InFIG. 1a decoded parameter is output from local decoder103to enhancement layer coder107, but in this embodiment a decoded signal obtained by local decoder103is output to enhancement layer coder107instead of a decoded parameter.

Signal sl(n) decoded by local decoder103inFIG. 5is input to power estimation unit901. Power estimation unit901then estimates the MDCT coefficient power from this decoded signal s1(n). If the MDCT coefficient power estimate is designated powp, powp is expressed by Equation (20) below.

powp=α·∑n=0N-1⁢⁢s⁢⁢l⁡(n)2(20)
Here, N indicates the length of decoded signal sl(n), and α indicates a predetermined constant for correction. In another method that uses spectrum tilt found from the base layer LPC coefficients, an MDCT coefficient power estimate is expressed by Equation (21) below.

powp=α·β·∑n=0N-1⁢⁢s⁢⁢l⁡(n)2(21)
Here, β denotes a variable that depends on the spectrum tilt found from the base layer LPC coefficients, having a property of approaching zero when the spectrum tilt is large (when an amount of spectral energy is big in low band), and approaching 1 when the spectrum tilt is small (when there is power in a relatively high region).

Next, power fluctuation amount quantizer902normalizes the power of the MDCT coefficients obtained by MDCT section503by means of power estimate powp obtained by power estimation unit901, and quantizes the fluctuation amount. fluctuation amount r is expressed by Equation (22) below.

r=powpowp(22)
Here, pow indicates the MDCT coefficient power, and is calculated by means of Equation (23).

pow=∑m=0M-1⁢X⁡(m)2(23)
Here, X(m) indicates the MDCT coefficients, and M indicates the frame length. Power fluctuation amount quantizer902quantizes fluctuation amount r, sends the coding information to multiplexer510, and also decodes quantized fluctuation amount rq. Using quantized fluctuation amount rq, power normalizer505normalizes the MDCT coefficients using Equation (24) below.

Thus, according to a signal processing apparatus of this embodiment, by using the correlation between base layer decoded signal power and enhancement layer MDCT coefficient power, predicting MDCT coefficient power using a base layer decoded signal, and coding the amount of fluctuation from that predicted value, it is possible to reduce the number of bits necessary for MDCT coefficient power quantization.

FIG. 10is a block diagram showing the configuration of a signal processing apparatus according to Embodiment 5 of the present invention. Signal processing apparatus1000inFIG. 10mainly comprises a demultiplexer1001, base layer decoder1002, up-sampler1003, enhancement layer decoder1004, and adder1005.

Demultiplexer1001separates coding information, and generates base layer coding information and enhancement layer coding information. Then demultiplexer1001outputs base layer coding information to base layer decoder1002, and outputs enhancement layer coding information to enhancement layer decoder1004.

Base layer decoder1002decodes a sampling rate FL decoded signal using the base layer coding information obtained by demultiplexer1001, and outputs the resulting signal to up-sampler1003. At the same time, a parameter decoded by base layer decoder1002is output to enhancement layer decoder1004. Up-sampler1003raises the decoded signal sampling frequency to FH, and outputs this to adder1005.

Enhancement layer decoder1004decodes the sampling rate FH decoded signal using the enhancement layer coding information obtained by demultiplexer1001and the parameter decoded by base layer decoder1002, and outputs the resulting signal to adder1005.

Adder1005performs addition of the decoded signal output from up-sampler1003and the decoded signal output from enhancement layer decoder1004.

The operation of a signal processing apparatus of this embodiment will be now described. First, code coded in a signal processing apparatus of any of Embodiments 1 through 4 is input, and that code is separated by demultiplexer1001, generating base layer coding information and enhancement layer coding information.

Next, base layer decoder1002decodes a sampling rate FL decoded signal using the base layer coding information obtained by demultiplexer1001. Then up-sampler1003raises the sampling frequency of that decoded signal to FH.

In enhancement layer decoder1004, the sampling rate FH decoded signal is decoded using enhancement layer coding information obtained by demultiplexer1001and a parameter decoded by base layer decoder1002.

The base layer decoded signal up-sampled by up-sampler1003and the enhancement layer decoded signal are added by adder1005. The above processing is repeated while there is a new input signal. When there is no new input signal, processing is terminated.

Thus, according to a signal processing apparatus of this embodiment, by performing enhancement layer decoder1004decoding using parameters decoded by base layer decoder1002, it is possible to generate a decoded signal from coding information of a sound coding unit that performs enhancement layer coding using decoding parameters in base layer coding.

Base layer decoder1002will now be described.FIG. 11is a block diagram showing an example of base layer decoder1002. Base layer decoder1002inFIG. 11mainly comprises a demultiplexer1101, excitation generator1102, and synthesis filter1103, and performs CELP decoding processing.

Demultiplexer1101separates various parameters from base layer coding information output from demultiplexer1001, and outputs these parameters to excitation generator1102and synthesis filter1103.

Excitation generator1102performs adaptive vector, adaptive vector gain, noise vector, and noise vector gain decoding, generates an excitation signal using these, and outputs this excitation signal to synthesis filter1103. Synthesis filter1103generates a synthesized signal using the decoded LPC coefficients.

The operation of base layer decoder1002inFIG. 11will now be described. First, demultiplexer1101separates various parameters from base layer coding information.

Synthesis filter1103then generates synthesized signal syn(n) in accordance with Equation (26) below, using the decoded LPC coefficients.

syn⁡(n)=ex⁡(n)+∑i=1NP⁢⁢αq⁡(i)·syn⁡(n-i)(26)
Here, αq indicates the decoded LPC coefficients, and NP the order of the LPC coefficients.

Decoded signal syn(n) decoded in this way is output to up-sampler1003, and a parameter obtained as a result of decoding is output to enhancement layer decoder1004. The above processing is repeated while there is a new input signal. When there is no new input signal, processing is terminated. Depending on the CELP configuration, a mode is also possible in which a synthesized signal is output after passing through a post-filter. The post-filter mentioned here has a function of post-processing to make coding distortion less perceptible.

Enhancement layer decoder1004will now be described.FIG. 12is a block diagram showing an example of enhancement layer decoder1004. Enhancement layer decoder1004inFIG. 12mainly comprises a demultiplexer1201, LPC coefficient decoder1202, spectral envelope calculator1203, vector decoder1204, Bark scale shape decoder1205, multiplier1206, multiplier1207, power decoder1208, multiplier1209, and IMDCT section1210.

Demultiplexer1201separates various parameters from enhancement layer coding information output from demultiplexer1001. LPC coefficient decoder1202decodes the LPC coefficients using the LPC coefficients related coding information, and outputs the result to spectral envelope calculator1203.

Spectral envelope calculator1203calculates spectral envelope env(m) in accordance with Equation (6) using the decoded LPC coefficients, and outputs spectral envelope env(m) to vector decoder1204and multiplier1207.

Vector decoder1204determines quantization bit allocation based on spectral envelope env(m) obtained by spectral envelope calculator1203, and decodes normalized MDCT coefficients X3q(m) from coding information obtained from demultiplexer1201and the aforementioned quantization bit allocation. The quantization bit allocation method is the same as that used in enhancement layer coding in the coding method of any of Embodiments 1 through 4.

Bark scale shape decoder1205decodes Bark scale shape Bq(k) based on coding information obtained from demultiplexer1201, and outputs the result to multiplier1206.

Multiplier1206multiplies normalized MDCT coefficients X3q(m) by Bark scale shape Bq(k) in accordance with Equation (27) below, and outputs the result of the multiplication to multiplier1207.
X2q(m)=X3q(m)√{square root over (Bq(k))}fl(k)≦m≦fh(k) 0≦k<K  (27)
Here, fl(k) indicates the lowest frequency of the k'th sub-band and fh(k) the highest frequency of the k'th sub-band, and K indicates the number of sub-bands.

Multiplier1207multiplies normalized MDCT coefficients X2q(m) obtained from multiplier1206by spectral envelope env(m) obtained by spectral envelope calculator1203in accordance with Equation (28) below, and outputs the result of the multiplication to multiplier1209.
X1q(m)=X2q(m)env(m)  (28)

Power decoder1208decodes power powq based on coding information obtained from demultiplexer1201, and outputs the result of the decoding to multiplier1209.

Multiplier1209multiplies normalized MDCT coefficients X1q(m) by decoded power powq in accordance with Equation (29) below, and outputs the result of the multiplication to IMDCT section1210.
Xq(m)=X1q(m)√{square root over (powp)}  (29)

IMDCT section1210executes IMDCT (Inverse Modified Discrete Cosine Transform) processing on the decoded MDCT coefficients obtained in this way, overlaps and adds the signal obtained in half the previous frame and half the current frame, and the resultant signal is an output signal. The above processing is repeated while there is a new input signal. When there is no new input signal, processing is terminated.

Thus, according to a signal processing apparatus of this embodiment, by performing enhancement layer decoder decoding using parameters decoded by a base layer decoder, it is possible to generate a decoded signal from coding information of a coding unit that performs enhancement layer coding using decoding parameters in base layer coding.

FIG. 13is a drawing showing an example of the configuration of enhancement layer decoder1004. Parts inFIG. 13identical to those inFIG. 12are assigned the same reference numerals as inFIG. 12and detailed descriptions thereof are omitted.

Enhancement layer decoder1004inFIG. 13differs from enhancement layer decoder1004inFIG. 12in being provided with a conversion table1301, LPC coefficient mapping section1302, spectral envelope calculator1303, and transformation section1304, and performing decoding using the LPC coefficients decoded by base layer decoder1002.

Conversion table1301stores base layer LPC coefficients and enhancement layer LPC coefficients with the correspondence therebetween indicated.

LPC coefficient mapping section1302references conversion table1301, converts the base layer LPC coefficients input from base layer decoder1002to the enhancement layer LPC coefficients, and outputs the enhancement layer LPC coefficients to spectral envelope calculator1303.

Spectral envelope calculator1303obtains a spectral envelope based on the enhancement layer LPC coefficients, and outputs this spectral envelope to transformation section1304. Transformation section1304transforms the spectral envelope and outputs the result to multiplier1207and vector decoder1204. An example of the transformation method is the method shown in Equation (16) of Embodiment 2.

The operation of enhancement layer decoder1004inFIG. 13will now be described. The base layer LPC coefficients are found for signals in signal band0to FL, and does not coincide with the LPC coefficients used by an enhancement layer signal (signal band0to FH). However, there is a strong correlation between the two. Therefore, in LPC coefficient mapping section1302, a conversion table1301is separately designed in advance, showing the correspondence between LPC coefficients for signal band0to FL signals and signal band0to FH signals, using this correlation. This conversion table1301is used to find the enhancement layer LPC coefficients from the base layer LPC coefficients.

Details of conversion table1301are the same as for conversion table601in Embodiment 2.

Thus according to a signal processing apparatus of this embodiment, by finding the enhancement layer LPC coefficients using the LPC coefficients quantized by a base layer decoder, and calculating a spectral envelope from the enhancement layer LPC coefficients, LPC analysis and quantization are made unnecessary, and the number of quantization bits can be reduced.

FIG. 14is a block diagram showing the configuration of the enhancement layer decoder of a signal processing apparatus according to Embodiment 7 of the present invention. Parts inFIG. 14identical to those inFIG. 12are assigned the same reference numerals as inFIG. 12and detailed descriptions thereof are omitted.

Enhancement layer decoder1004inFIG. 14differs from the enhancement layer decoder inFIG. 12in being provided with a spectral fine structure calculator1401, calculating spectral fine structure using a pitch period decoded by base layer decoder1002, employing that spectral fine structure in decoding, and performing sound decoding corresponding to sound coding whereby quantization performance is improved.

Spectral fine structure calculator1401calculates the spectral fine structure from pitch period T and pitch gain β decoded by base layer decoder1002, and outputs the spectral fine structure to vector decoder1204and multiplier1207.

Using pitch period Tq and pitch gain βq, spectral fine structure calculator1401calculates spectral fine structure har(m) in accordance with Equation (30) below.

har⁡(m)=11-βq·ⅇ-j⁢2⁢⁢π⁢⁢m⁢⁢TqM(30)
Here, M indicates the spectral resolution. As Equation (30) is an oscillation filter when the absolute value of βq is greater than or equal to 1, a restriction may also be set so that the possible range of the absolute value of βq is less than or equal to a predetermined set value less than 1 (for example, 0.8).

The allocation of quantization bits by vector decoder1204is also determined using spectral envelope env(m) obtained by spectral envelope calculator1203and spectral fine structure har(m) obtained by spectral fine structure calculator1401. Then normalized MDCT coefficients X3q(m) is decoded from that quantization bit allocation and coding information obtained from demultiplexer1201. Also, normalized MDCT coefficients X1q(m) is found by multiplying normalized MDCT coefficients X2q(m) by spectral envelope env(m) and spectral fine structure har(m) in accordance with Equation (31) below.
X1q(m)=X2q(m)env(m)har(m)  (31)

Thus, according to a signal processing apparatus of this embodiment, by calculating a spectral fine structure using a pitch period coded by a base layer coder and decoded by a local decoder, and using that spectral fine structure in spectrum normalization and vector quantization, it is possible to perform sound decoding corresponding to sound coding whereby quantization performance is improved.

FIG. 15is a block diagram showing the configuration of the enhancement layer decoder of a signal processing apparatus according to Embodiment 8 of the present invention. Parts inFIG. 15identical to those inFIG. 12are assigned the same reference numerals as inFIG. 12and detailed descriptions thereof are omitted.

Enhancement layer decoder1004inFIG. 15differs from the enhancement layer decoder inFIG. 12in being provided with a power estimation unit1501, power fluctuation amount decoder1502, and power generator1503, and in forming a decoder corresponding to a coder that predicts MDCT coefficient power using a base layer decoded signal, and encodes the amount of fluctuation from that predicted value.

InFIG. 10a decoded parameter is output from base layer decoder1002to enhancement layer decoder1004, but in this embodiment a decoded signal obtained by base layer decoder1002is output to enhancement layer decoder1004instead of a decoded parameter.

Power estimation unit1501estimates the power of the MDCT coefficients from decoded signal sl(n) decoded by base layer decoder1002, using Equation (20) or Equation (21).

Power fluctuation amount decoder1502decodes the power fluctuation amount from coding information obtained from demultiplexer1201, and outputs this to power generator1503. Power generator1503calculates power from the power fluctuation amount.

Multiplier1209finds the MDCT coefficients in accordance with Equation (32) below.
Xq(m)=X1q(m)√{square root over (rq·powp)}(32)
Here, rq indicates the power fluctuation amount, and powp the power estimate. X1q(m) indicates the output signal from multiplier1207.

Thus, according to a signal processing apparatus of this embodiment, by configuring a decoder corresponding to a coder that predicts MDCT coefficient power using a base layer decoded signal and encodes the amount of fluctuation from that predicted value, it is possible to reduce the number of bits necessary for MDCT coefficient power quantization.

FIG. 16is a block diagram showing the configuration of a sound coding apparatus according to Embodiment 9 of the present invention. Sound coding apparatus1600inFIG. 16mainly comprises a down-sampler1601, base layer coder1602, local decoder1603, up-sampler1604, delayer1605, subtracter1606, frequency determination section1607, enhancement layer coder1608, and multiplexer1609.

InFIG. 16, down-sampler1601receives sampling rate FH input data (acoustic data), converts this input data to sampling rate FL lower than sampling rate FH, and outputs the result to base layer coder1602.

Base layer coder1602encodes the sampling rate FL input data in predetermined basic frame units, and outputs the first coding information to local decoder1603and multiplexer1609. Base layer coder1602may code input data using the CELP method, for example.

Local decoder1603decodes the first coding information, and outputs the decoded signal obtained by decoding to up-sampler1604. Up-sampler1604raises the decoded signal sampling rate to FH, and outputs the result to subtracter1606and frequency determination section1607.

Delayer1605delays the input signal by a predetermined time, then outputs the signal to subtracter1606. By making this delay time equal to the time delay arising in down-sampler1601, base layer coder1602, local decoder1603, and up-sampler1604, phase shift is prevented in the following subtraction processing. Subtracter1606performs subtraction between the input signal and decoded signal, and outputs the result of the subtraction to enhancement layer coder1608as an error signal.

Frequency determination section1607determines an area for which error signal coding is performed and an area for which error signal coding is not performed from the decoded signal for which the sampling rate has been raised to FH, and notifies enhancement layer coder1608. For example, frequency determination section1607determines the frequency for auditory masking from the decoded signal for which the sampling rate has been raised to FH, and outputs this to enhancement layer coder1608.

Enhancement layer coder1608converts the error signal to a frequency domain and generates an error spectrum, and performs error spectrum coding based on frequency information obtained from frequency determination section1607. Multiplexer1609multiplexes coding information obtained by coding by base layer coder1602and coding information obtained by coding by enhancement layer coder1608.

The signals coded by base layer coder1602and enhancement layer coder1608respectively will now be described.FIG. 17is a drawing showing an example of acoustic signal information distribution. InFIG. 17, the vertical axis indicates the amount of information, and the horizontal axis indicates frequency.FIG. 17shows how much speech information and background music and background noise information contained in the input signal are present in which frequency bands.

As shown inFIG. 17, in the case of speech information, there is a large amount of information in the low frequency region, and the amount of information decreases the higher the frequency region. Conversely, in the case of background music and background noise information, there is comparatively little information in the lower region compared with speech information, and a large amount of information in the higher region.

Thus, in the base layer, speech signals are coded with high quality using CELP, and in the enhancement layer, background music or environmental sound that cannot be represented in the base layer, and signals with higher frequency components than the frequency region covered by the base layer, are coded efficiently.

FIG. 18is a drawing showing an example of coding regions in the base layer and enhancement layer. InFIG. 18, the vertical axis indicates the amount of information, and the horizontal axis indicates frequency.FIG. 18shows the regions that are the object of information coded by base layer coder1602and enhancement layer coder1608respectively.

Base layer coder1602is designed to represent efficiently speech information in the frequency band from0to FL, and can perform good-quality coding of speech information in this region. However, with base layer coder1602, the coding quality of background music and background noise information in the frequency band from0to FL is not high.

Enhancement layer coder1608is designed to cover portions for which the capability of base layer coder1602is insufficient, as described above, and signals in the frequency band from FL to FH. Thus, by combining base layer coder1602and enhancement layer coder1608, it is possible to implement high-quality coding in a wide band.

As shown inFIG. 18, the first coding information obtained by coding in base layer coder1602contains speech information in the frequency band between0and FL, and therefore a scalable function can be implemented whereby a decoded signal can be obtained even with only at least the first coding information.

Also, raising coding efficiency by using auditory masking in the enhancement layer can be considered. Auditory masking employs the human auditory characteristic whereby, when a certain signal is supplied, a signal in the vicinity of the frequency of that signal cannot be heard (is masked).

FIG. 19is a drawing showing an example of an acoustic (music) signal spectrum. InFIG. 19, the solid line indicates auditory masking, and the dotted line indicates the error spectrum. “Error spectrum” here means the spectrum of an error signal (enhancement layer input signal) for an input signal and base layer decoded signal.

In the error spectrum indicated by shaded areas inFIG. 19, amplitude values are lower than the auditory masking, and therefore sound cannot be heard by the human ear, while in other regions error spectrum amplitude values exceed the auditory masking, and therefore quantization distortion is perceived.

In the enhancement layer, it is only necessary to code the error spectrum included in the white areas inFIG. 19so that quantization distortion of those regions is smaller than the auditory masking. Coefficients belonging to the shaded areas are already smaller than the auditory masking, and so need not be quantized.

In sound coding apparatus1600of this embodiment, a frequency at which a residual error signal is coded according to auditory masking, etc., is not transmitted from the coding side to the decoding side, and the error spectrum frequency at which enhancement layer coding is performed is determined separately by the coding side and the decoding side using an up-sampled base layer decoded signal.

In the case of a decoded signal resulting from decoding of base layer coding information, the same signal is obtained by the coding side and the decoding side, and therefore by having the coding side code the signal by determining the auditory masking frequency from this decoded signal, and having the decoding side decode the signal by obtaining auditory masking frequency information from this decoded signal, it becomes unnecessary to code and transmit error spectrum frequency information as additional information, enabling a reduction in the bit rate to be achieved.

Next, the operation of each block of a sound coding apparatus according to this embodiment will be described in detail. First, the operation of frequency determination section1607, which determines an error spectrum frequency coded in the enhancement layer from an up-sampled base layer decoded signal (hereinafter referred to as “base layer decoded signal”), will be described.FIG. 20is a block diagram showing an example of the internal configuration of the frequency determination section of a sound coding apparatus of this embodiment.

FFT section1901performs orthogonal conversion of base layer decoded signal x(n) output from up-sampler1604, calculates amplitude spectrum P(m), and outputs amplitude spectrum P(m) to estimated auditory masking calculator1902and determination section1903. To be specific, FFT section1901calculates amplitude spectrum P(m) using Equation (33) below.
P(m)=√{square root over (Re2(m)+Im2(m))}{square root over (Re2(m)+Im2(m))}  (33)

Here, Re(m) and Im(m) indicate the real part and imaginary part of Fourier coefficients of base layer decoded signal x(n), and m indicates frequency.

Next, estimated auditory masking calculator1902calculates estimated auditory masking M′(m) using base layer decoded signal amplitude spectrum P(m), and outputs estimated auditory masking M′(m) to determination section1903. Auditory masking is generally calculated based on the spectrum of an input signal, but in this implementation example, auditory masking is estimated using base layer decoded signal x(n) instead of the input signal. This is based on the idea that, since base layer decoded signal x(n) is determined so that there is little distortion with respect to the input signal, adequate approximation will be achieved and there will be no major problem if base layer decoded signal x(n) is used instead of the input signal.

Determination section1903then determines a frequency for which error spectrum coding by enhancement layer coder1608is applicable, using base layer decoded signal amplitude spectrum P(m) and estimated auditory masking M′(m) obtained by estimated auditory masking calculator1902. Determination section1903regards base layer decoded signal amplitude spectrum P(m) as an approximation of the error spectrum, and outputs frequency m for which Equation (34) below holds true to enhancement layer coder1608.
P(m)−M′(m)>0  (34)

In Equation (34), term P(m) estimates the size of the error spectrum, and term M′(m) estimates auditory masking. Determination section1903then compares the value of the estimated error spectrum and estimated auditory masking, and if Equation (34) is satisfied—that is to say, if the value of the estimated error spectrum exceeds the value of the estimated auditory masking—the error spectrum of that frequency is assumed to be perceived as noise, and is made subject to coding by enhancement layer coder1608.

Conversely, if the value of the estimated error spectrum is smaller than the size of the estimated auditory masking, determination section1903considers that the error spectrum of that frequency will not be perceived as noise due to the effects of masking, and determines the error spectrum of this frequency not to be subject to quantization.

The operation of estimated auditory masking calculator1902will now be described.FIG. 21is a drawing showing an example of the internal configuration of the auditory masking calculator of a sound coding apparatus of this embodiment. InFIG. 21, estimated auditory masking calculator1902mainly comprises a Bark spectrum calculator2001, spread function convolution unit2002, tonality calculator2003, and auditory masking calculator2004.

B⁡(k)=∑m=f⁢⁢l⁢⁢(k)f⁢⁢h⁡(k)⁢P2⁡(m)(35)
Here, P(m) indicates an amplitude spectrum, and is found from Equation (33) above, k corresponds to the Bark spectrum number, and fl(k) and fh(k) indicates the lowest frequency and highest frequency respectively of the k'th Bark spectrum. Bark spectrum B(k) indicates the spectral intensity in the case of band distribution at equal intervals on the Bark scale. If the Herz scale is represented by h and the Bark scale by B, the relationship between the Herz scale and Bark scale is expressed by Equation (36) below.

Tonality calculator2003finds spectrum flatness SFM(k) of each Bark spectrum using Equation (38) below.

SFM⁡(k)=μg⁡(k)μ⁢⁢a⁡(k)(3.8)
Here, μg(k) indicates the geometric mean of power spectra in the k'th Bark spectrum, and μa(k) indicates the arithmetic mean of power spectra in the k'th Bark spectrum. Tonality calculator2003then calculates tonality coefficient α(k) from decibel value SFMdB(k) of spectrum flatness SFM(k), using Equation (39) below.

Using Equation (40) below, auditory masking calculator2004finds offset O(k) of each Bark scale from tonality coefficient α(k) calculated by tonality calculator2003.
O(k)=α(k)·(14.5−k)+(1.0−α(k))·5.5  (40)

Auditory masking calculator2004then uses Equation (41) below to calculate auditory masking T(k) by subtracting off set O(k) from C(k) found by spread function convolution unit2002.
T(k)=max(10log10(C(k))−(O(k)/10),Tq(k))  (41)
Here, Tq(k) indicates an absolute threshold value. The absolute threshold value represents the minimum value of auditory masking observed as a human auditory characteristic. Then auditory masking calculator2004converts auditory masking T(k) expressed on the Bark scale to the Herz scale and finds estimated auditory masking M′(m), which it outputs to determination section1903.

Enhancement layer coder1608performs MDCT coefficient coding using frequency m subject to quantization found in this way.FIG. 22is a block diagram showing an example of the internal configuration of an enhancement layer coder of this embodiment. Enhancement layer coder1608inFIG. 22mainly comprises an MDCT section2101and MDCT coefficient quantizer2102.

MDCT section2101multiplies the input signal output from subtracter1606by an analysis window, then performs MDCT (Modified Discrete Cosine Transform) processing to obtain the MDCT coefficients. In MDCT processing, an orthogonal base for analysis is used for successive two frames. And the analysis frame is overlapped one-half, and the first half of the analysis frame is an odd function while the latter half of the analysis frame is an even function. A feature of MDCT processing is that frame boundary distortion does not occur because of addition by overlapping of waveforms after an inverse transform. When MDCT is performed, the input signal is multiplied by a window function such as a sin window. If a sequence of MDCT coefficients is designated X(n), the MDCT coefficients are calculated in accordance with Equation (42) below.

MDCT coefficient quantizer2102quantizes the coefficients corresponding to frequencies from frequency determination section1607. Then MDCT coefficient quantizer2102outputs the quantized MDCT coefficients coding information to multiplexer1609.

Thus, according to a sound coding apparatus of this embodiment, because of determining frequencies for quantization in enhancement layer by using a base layer decoded signal, it is unnecessary to transmit frequency information for quantization from the coding side to the decoding side, and enabling high-quality coding to be performed at a low bit rate.

In the above embodiment, an auditory masking calculation method that uses FFT has been described, but it is also possible to calculate auditory masking using MDCT instead of FFT.FIG. 23is a block diagram showing an example of the internal configuration of an auditory masking calculator of this embodiment. Parts inFIG. 23identical to those inFIG. 20are assigned the same reference numerals as inFIG. 20and detailed descriptions thereof are omitted.

MDCT section2201approximates amplitude spectrum P(m) using the MDCT coefficients. To be specific, MDCT section2201approximates P(m) using Equation (43) below.
P(m)=√{square root over (R2(m))}  (43)
Here, R(m) is the MDCT coefficients found by performing MDCT processing on a signal supplied from up-sampler1604.

Estimated auditory masking calculator1902calculates Bark spectrum B(k) from P(m) approximately. Thereafter, frequency information for quantization is calculated in accordance with the above-described method.

Thus, a sound coding apparatus of this embodiment can calculate auditory masking using MDCT.

The decoding side will now be described.FIG. 24is a block diagram showing the configuration of a sound decoding apparatus according to Embodiment 9 of the present invention. Sound decoding apparatus2300inFIG. 24mainly comprises a demultiplexer2301, base layer decoder2302, up-sampler2303, frequency determination section2304, enhancement layer decoder2305, and adder2306.

Demultiplexer2301separates code coded by sound coding apparatus1600into base layer first coding information and enhancement layer second coding information, outputs the first coding information to base layer decoder2302, and outputs the second coding information to enhancement layer decoder2305.

Base layer decoder2302decodes the first coding information and obtains a sampling rate FL decoded signal. Then base layer decoder2302outputs the decoded signal to up-sampler2303. Up-sampler2303converts the sampling rate FL decoded signal to a sampling rate FH decoded signal, and outputs this signal to frequency determination section2304and adder2306.

Using the up-sampled base layer decoded signal, frequency determination section2304determines error spectrum frequencies to be decoded in enhancement layer decoder2305. This frequency determination section2304has the same kind of configuration as frequency determination section1607inFIG. 16.

Enhancement layer decoder2305decodes the second coding information and outputs the sampling rate of FH decoded signal to adder2306.

Adder2306adds the base layer decoded signal up-sampled by up-sampler2303and the enhancement layer decoded signal decoded by enhancement layer decoder2305, and outputs the resulting signal.

Next, the operation of each block of a sound decoding apparatus according to this embodiment will be described in detail.FIG. 25is a block diagram showing an example of the internal configuration of the enhancement layer decoder of a sound decoding apparatus of this embodiment.FIG. 25shows an example of the internal configuration of enhancement layer decoder2305inFIG. 24. Enhancement layer decoder2305inFIG. 25mainly comprises an MDCT coefficient decoder2401, IMDCT section2402, and overlap adder2403.

MDCT coefficient decoder2401decodes the MDCT coefficients quantized from second coding information output from demultiplexer2301based on frequencies outputted from frequency determination section2304. To be specific, the decoded MDCT coefficients corresponding to the frequencies indicated by frequency determination section2304are positioned, and zero is supplied for other frequencies.

IMDCT section2402executes inverse MDCT processing on the MDCT coefficients output from MDCT coefficient decoder2401, generates a time domain signal, and outputs this signal to overlap adder2403.

Overlap adder2403performs overlap and add operation after windowing with a time domain signal from IMDCT section2042, and it outputs the decoded signal to adder2306. To be specific, overlap adder2403multiplies the decoded signal by a window and overlaps the time domain signal decoded in the previous frame and the current frame, performing addition, and generates an output signal.

Thus, according to a sound decoding apparatus of this embodiment, by determining the frequencies for enhancement layer's decoding by using base layer decoded signal, it is possible to determine the frequencies for enhancement layer's decoding without any additional information, and enabling high-quality coding to be performed at a low bit rate.

In this embodiment an example is described in which CELP is used in base layer coding.FIG. 26is a block diagram showing an example of the internal configuration of a base layer coder of Embodiment 10 of the present invention.FIG. 26shows an example of the internal configuration of base layer coder1602inFIG. 16. Base layer coder1602inFIG. 16mainly comprises an LPC analyzer2501, weighting section2502, adaptive code book search unit2503, adaptive gain quantizer2504, target vector generator2505, noise code book search unit2506, noise gain quantizer2507, and multiplexer2508.

LPC analyzer2501calculates the LPC coefficients of a sampling rate FL input signal, converts the LPC coefficients to a parameter suitable for quantization such as the LSP coefficients, and performs quantization. LPC analyzer2501then outputs the coding information obtained by this quantization to multiplexer2508.

Also, LPC analyzer2501calculates the quantized LSP coefficients from coding information and converts this to the LPC coefficients, and outputs the quantized LPC coefficients to adaptive code book search unit2503, adaptive gain quantizer2504, noise code book search unit2506, and noise gain quantizer2507. LPC analyzer2501also outputs the original LPC coefficients to weighting section2502, adaptive code book search unit2503, adaptive gain quantizer2504, noise code book search unit2506, and noise gain quantizer2507.

Weighting section2502performs weighting on the input signal output from down-sampler1601based on the LPC coefficients obtained by LPC analyzer1501. The purpose of this is to perform spectrum shaping so that the quantization distortion spectrum is masked by the input signal spectral envelope.

The adaptive code book is then searched by adaptive code book search unit2503with the weighted input signal as the target signal. A signal in which a previously determined excitation signal is repeated on a pitch period basis is called an adaptive vector, and an adaptive code book is composed of adaptive vectors generated at pitch periods of a predetermined range.

If a weighted input signal is designated t(n), and a signal in which an impulse response of a weighted synthesis filter comprising the original LPC coefficients and the quantized LPC coefficients is convoluted to the adaptive vector of pitch period i is designated pi(n), then adaptive code book search unit2503outputs pitch period i of the adaptive vector for which evaluation function D of Equation (44) below is minimized to multiplexer2508as coding information.

D=∑n=0N-1⁢⁢t2⁡(n)-(∑n=0N-1⁢⁢t⁡(n)⁢pi⁡(n))2∑n=0N-1⁢pi2⁡(n)(44)
Here, N indicates the vector length. As the first term of Equation (44) is independent of pitch period i, adaptive code book search unit2503actually calculates only the second term.

Adaptive gain quantizer2504performs quantization of the adaptive gain that is multiplied by the adaptive vector. Adaptive gain β is expressed by Equation (45) below. Adaptive gain quantizer2504performs scalar quantization of this adaptive gain β, and outputs the coding information obtained in quantization to multiplexer2508.

Target vector generator2505subtracts the effect of the adaptive vector from the input signal, and generates and outputs the target vector used by noise code book search unit2506and noise gain quantizer2507. In target vector generator2505, if pi(n) designates a signal in which a weighted synthesis filter impulse response is convoluted to the adaptive vector when evaluation function D expressed by Equation (44) is minimized, and βq designates the quantized adaptive gain when adaptive gain β expressed by Equation (45) undergoes scalar quantization, then target vector t2(n) is expressed by Equation (46) below.
t2(n)=t(n)−βq·pi(n)  (46)

Noise code book search unit2506carries out a noise code book search using the aforementioned target vector t2(n), the original LPC coefficients, and the quantized LPC coefficients. Noise code book search unit2506can use random noise or a signal learned using a large-amount speech signal, for example. Also, an algebraic code book can be used. The algebraic codebook consists of some of pulses. A feature of such an algebraic code book is that an optimal combination of pulse position and pulse code (polarity) can be determined by a small amount of computation.

If the target vector is designated t2(n), and a signal in which an impulse response of a weighted synthesis filter is convoluted to the noise vector corresponding to code j is designated cj(n), then noise code book search unit2506outputs to multiplexer2508index j of the noise vector for which evaluation function D of Equation (47) below is minimized.

Noise gain quantizer2507quantizes the noise gain that is multiplied by the noise vector. Noise gain quantizer2507calculates adaptive gain γ using Equation (48) below, performs scalar quantization of this noise gain γ, and outputs the coding information to multiplexer2508.

Multiplexer2508multiplexes the coding information of the LPC coefficients, adaptive vector, adaptive gain, noise vector, and noise gain coding information, and outputs the resultant information to local decoder1603and multiplexer1609.

The decoding side will now be described.FIG. 27is a block diagram showing an example of the internal configuration of a base layer decoder of this embodiment.FIG. 27shows an example of base layer decoder2302. Base layer decoder2302inFIG. 27mainly comprises a demultiplexer2601, excitation generator2602, and synthesis filter2603.

Demultiplexer2601separates first coding information from demultiplexer2301into LPC coefficients, adaptive vector, adaptive gain, noise vector, and noise gain coding information, and outputs the adaptive vector, adaptive gain, noise vector, and noise gain coding information to excitation generator2602. Similarly, demultiplexer2601outputs linear predictive coefficients coding information to synthesis filter2603.

Synthesis filter2603performs LPC coefficient decoding from LPC coefficient coding information, and generates synthesized signal syn(n) from the decoded LPC coefficients using Equation (50) below.

syn⁡(n)=ex⁡(n)+∑i=1NP⁢⁢αq⁡(i)·syn⁡(n-i)(50)
Here, αq indicates the decoded LPC coefficients, and NP the order of the LPC coefficients. Synthesis filter2603then outputs decoded signal syn(n) decoded in this way to up-sampler2303.

Thus, according to a sound coding apparatus of this embodiment, by coding an input signal using CELP in the base layer on the transmitting side, and decoding this coded input signal using CELP on the receiving side, it is possible to implement a high-quality base layer at a low bit rate.

In order to suppress perception of quantization distortion, a coding apparatus of this embodiment can also employ a configuration with subordinate connection of a post-filter after synthesis filter2603.FIG. 28is a block diagram showing an example of the internal configuration of a base layer decoder of this embodiment. Parts inFIG. 28identical to those inFIG. 27are assigned the same reference numerals as inFIG. 27and detailed descriptions thereof are omitted.

Various kinds of configuration may be employed for post-filter2701to achieve suppression of perception of quantization distortion, one typical method being that of using a formant emphasis filter comprising the LPC coefficients obtained by decoding by demultiplexer2601. Formant emphasis filter Hf(z) is expressed by Equation (51) below.

Hf⁡(z)=A⁡(z/γn)A⁡(z/γd)⁢(1-μz-1)(51)
Here, A(z) indicates an analysis filter comprising the decoded LPC coefficients, and γn, γd, and μ indicate constants that determine filter characteristics.

FIG. 29is a block diagram showing an example of the internal configuration of the frequency determination section of a sound coding apparatus according to Embodiment 11 of the present invention. Parts inFIG. 29identical to those inFIG. 20are assigned the same reference numerals as inFIG. 20and detailed descriptions thereof are omitted. Frequency determination section1607inFIG. 29differs from that inFIG. 20in being provided with an estimated error spectrum calculator2801and determination section2802, and in estimating estimated error spectrum E′(m) from base layer decoded signal amplitude spectrum P(m), and determining a frequency of an error spectrum coded by enhancement layer coder1608using estimated error spectrum E′(m) and estimated auditory masking M′(m).

FFT section1901performs Fourier transform of base layer decoded signal x(n) output from up-sampler1604, calculates amplitude spectrum P(m), and outputs amplitude spectrum P(m) to estimated auditory masking calculator1902and estimated error spectrum calculator2801.

Estimated error spectrum calculator2801calculates estimated error spectrum E′(m) from base layer decoded signal amplitude spectrum P(m) calculated by FFT section1901, and outputs estimated error spectrum E′(m) to determination section2802. Estimated error spectrum E′(m) is calculated by executing processing that approximates base layer decoded signal amplitude spectrum P(m) to flatness. To be specific, estimated error spectrum calculator2801calculates estimated error spectrum E′(m) using Equation (52) below.
E′(m)=a·P(m)γ(52)
Here, a and γ are constants of 0 or above and less than 1.

Using estimated error spectrum E′(m) obtained by estimated error spectrum calculator2801and estimated auditory masking M′(m) obtained by estimated auditory masking calculator1902, determination section2802determines frequencies for error spectrum coding by enhancement layer coder1608.

Next, an estimated error spectrum calculated by estimated error spectrum calculator2801of this embodiment will be described.FIG. 30is a drawing showing an example of a residual error spectrum calculated by an estimated error spectrum calculator of this embodiment.

As shown inFIG. 30, the spectrum shape of error spectrum E(m) is smoother than that of base layer decoded signal amplitude spectrum P(m), and its total band power is smaller. Therefore, the precision of error spectrum estimation can be improved by flattening the amplitude spectrum P(m) to the power of γ(0<γ<1), and reducing total band power by multiplying by a (0<a<1).

On the decoding side also, the internal configuration of frequency determination section2304of sound decoding apparatus2300is the same as that of coding-side frequency determination section1607inFIG. 29.

Thus, according to a sound coding apparatus of this embodiment, by smoothing a residual error spectrum estimated from a base layer decoded signal spectrum, the estimated error spectrum can be approximated to the residual error spectrum, and an error spectrum can be coded efficiently in the enhancement layer.

In this embodiment a case has been described in which FFT is used, but a configuration is also possible in which MDCT or other transformation is used instead of FFT, as in above-described Embodiment 9.

FIG. 31is a block diagram showing an example of the internal configuration of the frequency determination section of a sound coding apparatus according to Embodiment 12 of the present invention. Parts inFIG. 31identical to those inFIG. 20are assigned the same reference numerals as inFIG. 20and detailed descriptions thereof are omitted. Frequency determination section1607inFIG. 31differs from that inFIG. 20in being provided with an estimated auditory masking correction section3001and determination section3002, and in that frequency determination section1607, after calculating estimated auditory masking M′(m) by means of estimated auditory masking calculator1902from base layer decoded signal amplitude spectrum P(m), applies correction to this estimated auditory masking M′(m) based on local decoder1603decoded parameter information.

FFT section1901performs Fourier transform of base layer decoded signal x(n) output from up-sampler1604, calculates amplitude spectrum P(m), and outputs amplitude spectrum P(m) to estimated auditory masking calculator1902and determination section3002. Estimated auditory masking calculator1902calculates estimated auditory masking M′(m) using base layer decoded signal amplitude spectrum P(m), and outputs estimated auditory masking M′(m) to estimated auditory masking correction section3001.

It is here assumed that a first order PARCOR coefficient calculated from the decoded LPC coefficients is supplied as base layer coding information. Generally, the LPC coefficients and PARCOR coefficients represent an input signal spectral envelope. Due to the properties of the PARCOR coefficients, as the order of the PARCOR coefficients is lowered, the shape of a spectral envelope is simplified, and when the order of the PARCOR coefficients is 1, the degree of tilt of a spectrum is indicated.

On the other hand, in the spectral characteristics of a audio or speech input signal, there are cases where power is biased toward the lower region as opposed to the higher region (as with vowels, for example), and cases where the converse is true (as with consonants, for example). A base layer decoded signal is susceptible to the influence of such input signal spectral characteristics, and there is a tendency for spectrum power bias to be emphasized more than necessary.

Thus, in a sound coding apparatus of this embodiment, the precision of estimated masking M′(m) can be improved by correcting excessively emphasized spectral bias in estimated auditory masking correction section3001using an aforementioned first order PARCOR coefficient.

Estimated auditory masking correction section3001calculates correction filter Hk(z) from first order PARCOR coefficient k(1) output from base layer coder1602, using Equation (53) below.
Hk(z)=1−β·k(1)·z−1(53)
Here, β indicates a positive constant less than 1. Next, estimated auditory masking correction section3001calculates amplitude characteristic K(m) of correction filter Hk(z) using Equation (54) below.

Then estimated auditory masking correction section3001calculates corrected estimated auditory masking M″(m) from correction filter amplitude characteristic K(m), using Equation (55) below.
M″(m)=K(m)·M′(m)  (55)

Estimated auditory masking correction section3001then outputs corrected estimated auditory masking M″(m) to determination section3002instead of estimated auditory masking M′(m).

Using base layer decoded signal amplitude spectrum P(m), and corrected auditory masking M″(m) output from estimated auditory masking correction section3001, determination section3002determines frequencies for error spectrum coding by enhancement layer coder1608.

Thus, according to a sound coding apparatus of this embodiment, by calculating auditory masking from an input signal spectrum using masking effect characteristics, and performing quantization so that quantization distortion does not exceed the masking value in enhancement layer coding, it is possible to reduce the number of MDCT coefficients subject to quantization without a degradation of quality, and to perform high-quality coding at a low bit rate.

Thus, according to a sound coding apparatus of this embodiment, by applying correction based on base layer coder decoded parameter information to estimated auditory masking, it is possible to improve the precision of estimated auditory masking, and to perform efficient error spectrum coding in the enhancement layer.

On the decoding side also, the internal configuration of frequency determination section2304of sound decoding apparatus2300is the same as that of coding-side frequency determination section1607inFIG. 31.

It is also possible for frequency determination section1607of this embodiment to employ a configuration combining this embodiment and Embodiment 11.FIG. 32is a block diagram showing an example of the internal configuration of the frequency determination section of a sound coding apparatus of this embodiment. Parts inFIG. 32identical to those inFIG. 20are assigned the same reference numerals as inFIG. 20and detailed descriptions thereof are omitted.

FFT section1901performs Fourier transform of base layer decoded signal x(n) output from up-sampler1604, calculates amplitude spectrum P(m), and outputs amplitude spectrum P(m) to estimated auditory masking calculator1902and estimated error spectrum calculator2801.

Estimated auditory masking calculator1902calculates estimated auditory masking M′(m) using base layer decoded signal amplitude spectrum P(m), and outputs estimated auditory masking M′(m) to estimated auditory masking correction section3001.

Estimated error spectrum calculator2801calculates estimated error spectrum E′(m) from base layer decoded signal amplitude spectrum P(m) calculated by FFT section1901, and outputs estimated error spectrum E′(m) to determination section3101.

Using estimated error spectrum E′(m) estimated by estimated error spectrum calculator2801and corrected auditory masking M″(m) output from estimated auditory masking correction section3001, determination section3101determines a frequency subject to error spectrum coding by enhancement layer coder1608.

In this embodiment a case has been described in which FFT is used, but a configuration is also possible in which MDCT or other transform technique is used instead of FFT, as in above-described Embodiment 9.

FIG. 33is a block diagram showing an example of the internal configuration of the enhancement layer coder of a sound coding apparatus according to Embodiment 13 of the present invention. Parts inFIG. 33identical to those inFIG. 22are assigned the same reference numerals as inFIG. 22and detailed descriptions thereof are omitted. The enhancement layer coder inFIG. 33differs from the enhancement layer coder inFIG. 22in being provided with a ordering section3201and MDCT coefficient quantizer3202, and the weighting is performed by frequency on a frequency supplied from frequency determination section1607in accordance with the amount of estimated distortion value D(m).

InFIG. 33, MDCT section2101multiplies the input signal output from subtracter1606by an analysis window, then performs MDCT (Modified Discrete Cosine Transform) processing to obtain MDCT coefficients, and outputs the MDCT coefficients to MDCT coefficient quantizer3202.

Ordering section3201receives frequency information obtained by frequency determination section1607, and calculates the amount by which estimated error spectrum E′(m) of each frequency exceeds estimated auditory masking M′(m) (hereinafter referred to as the estimated distortion value), D(m). This estimated distortion value D(m) is defined by Equation (56) below.
D(m)=E′(m)−M′(m)  (56)

Then ordering section3201performs ordering in high-to-low estimated distortion value D(m) order, and outputs the corresponding frequency information to MDCT coefficient quantizer3202. MDCT coefficient quantizer3202performs quantization, allocating bits proportionally to error spectra E(m) positioned at frequencies in high-to-low distortion value D(m) order based on the estimated distortion value D(m).

As an example, a case will here be described in which frequencies sent from the frequency determination section and estimated distortion values are as shown inFIG. 34.FIG. 34is a drawing showing an example of ranking of estimated distortion values by an ordering section of this embodiment.

Ordering section3201rearranges frequencies in high-to-low estimated distortion value D(m) order based on the information inFIG. 34. In this example, the frequency m order obtained as a result of processing by ordering section3201is: 7, 8, 4, 9, 1, 11, 3, 12. Ordering section3201outputs this ordering information to MDCT coefficient quantizer3202.

At this time, there is allocation of many bits used for error spectrum quantization at the start of the order, and allocation of progressively fewer bits toward the end of the order. That is to say, the larger the estimated distortion value D(m) of a frequency, the greater is the allocation of bits used for error spectrum quantization, and the smaller the estimated distortion value D(m) of a frequency, the smaller is the allocation of bits used for error spectrum quantization.

For example, bit allocation may be executed as follows: 8 bits for E(7), 7 bits for E(8) and E(4), 6 bits for E(9) and E(1), and 8 bits for E(11), E(3), and E(12). Performing adaptive bit allocation according to estimated distortion value D(m) in this way improves quantization efficiency.

When vector quantization is applied, enhancement layer coder1608configures vectors in order from the error spectrum located at the start of the order, and performs vector quantization for the respective vectors. At this time, vector configuration and quantization bit allocation are performed so that bit allocation is greater for an error spectrum located at the start of the order, and smaller for an error spectrum located at the end of the order. In the example inFIG. 34, three vectors—two-dimensional, two-dimensional, and four-dimensional—are configured, with V1=(E(7), E(8)), V2=(E(4), E(9)), and V3=E(1), E(11), E(3), E(12)), and the bit allocations are 10 bits for V1, 8 bits for V2, and 8 bits for V3.

Thus, according to a sound coding apparatus of this embodiment, an improvement in quantization efficiency can be achieved by, in enhancement layer coding, performing coding with a large amount of information allocated to frequencies for which the amount by which the estimated error spectrum exceeds estimated auditory masking is large.

The decoding side will now be described.FIG. 35is a block diagram showing an example of the internal configuration of the enhancement layer decoder of a sound decoding apparatus according to Embodiment 13 of the present invention. Parts inFIG. 35identical to those inFIG. 25are assigned the same reference numerals as inFIG. 25and detailed descriptions thereof are omitted. Enhancement layer decoder2305inFIG. 35differs from that inFIG. 25in being provided with an ordering section3401and MDCT coefficient decoder3402, and in that frequencies supplied from frequency determination section2304are ordered in accordance with the amount of estimated distortion value D(m).

Ordering section3401calculates estimated distortion value D(m) using Equation (56) above. Ordering section3401has the same configuration as above-described ordering section3201. By means of this configuration, it is possible to decode coding information of the above-described sound coding method that enables adaptive bit allocation to be performed and an improvement inquantization efficiency to be achieved.

MDCT coefficient decoder3402decodes second coding information output from demultiplexer2301using frequency information ordered in accordance with the amount of estimated distortion value D(m). To be specific, MDCT coefficient decoder3402positions the decoded MDCT coefficients corresponding to a frequency supplied from frequency determination section2304, and supplies zero for other frequencies. IMDCT section2402then executes inverse MDCT processing on the MDCT coefficients obtained from MDCT coefficient decoder2401, and generates a time domain signal.

Overlap adder2403multiplies the aforementioned signal by a window function for combining, and overlaps the time domain signal decoded in the previous frame and the current frame, performing addition, and generates an output signal. Overlap adder2403outputs this output signal to adder2306.

Thus, according to a sound decoding apparatus of this embodiment, an improvement in quantization efficiency can be achieved by, in enhancement layer coding, performing vector quantization with adaptive bit allocation performed according to the amount by which an estimated error spectrum exceeds estimated auditory masking.

FIG. 36is a block diagram showing an example of the internal configuration of the enhancement layer coder of a sound coding apparatus according to Embodiment 14 of the present invention. Parts inFIG. 36identical to those inFIG. 22are assigned the same reference numerals as inFIG. 22and detailed descriptions thereof are omitted. The enhancement layer coder inFIG. 36differs from the enhancement layer coder inFIG. 22in being provided with a fixed band specification section3501and MDCT coefficient quantizer3502, and in that the MDCT coefficients included in a band specified beforehand is quantized together with the frequencies obtained from frequency determination section1607.

InFIG. 36, a band important in terms of auditory perception is set beforehand in fixed band specification section3501. It is here assumed that “m=15, 16” is set for frequencies included in the set band.

MDCT coefficient quantizer3502categorizes an input signal into coefficients to be quantized and coefficients not to be quantized using auditory masking output from frequency determination section1607in an input signal from MDCT section2101, and encodes the coefficients to be quantized and also the coefficients in a band set by fixed band specification section3501.

Assuming the relevant frequencies to be as shown inFIG. 34, error spectra E(1), E(3), E(4), E(7), E(8), E(9), E(11), E(12), and error spectra E(15), E(16) of frequencies specified by fixed band specification section3501are quantized by MDCT coefficient quantizer3502.

Thus, according to a sound coding apparatus of this embodiment, by forcibly quantizing a band that is unlikely to be selected as an object of quantization but that is important from an auditory standpoint, even if a frequency that should really be selected as an object of coding is not selected, an error spectrum located at a frequency included in a band that is important from an auditory standpoint is quantized without fail, enabling quality to be improved.

The decoding side will now be described.FIG. 37is a block diagram showing an example of the internal configuration of the enhancement layer decoder of a sound decoding apparatus according to Embodiment 14 of the present invention. Parts inFIG. 37identical to those inFIG. 25are assigned the same reference numerals as inFIG. 25and detailed descriptions thereof are omitted. The enhancement layer decoder inFIG. 37differs from the enhancement layer decoder inFIG. 25in being provided with a fixed band specification section3601and MDCT coefficient decoder3602, and in that the MDCT coefficients included in a band specified beforehand is decoded together with a frequency obtained from frequency determination section2304.

InFIG. 37, a band important in terms of auditory perception is set beforehand in fixed band specification section3601.

MDCT coefficient decoder3602decodes an MDCT coefficient quantized from second coding information output from demultiplexer2301based on error spectrum frequencies subject to decoding output from frequency determination section2304. To be specific, MDCT coefficient decoder3602positions decoded MDCT coefficients corresponding to frequencies indicated by frequency determination section2304and fixed band specification section3601, and supplies zero for other frequencies.

IMDCT section2402executes inverse MDCT processing on the MDCT coefficients output from MDCT coefficient decoder3602, generates a time domain signal, and outputs this time domain signal to overlap adder2403.

Thus, according to a sound decoding apparatus of this embodiment, by decoding the MDCT coefficients included in a band specified beforehand, it is possible to decode a signal in which a band that is unlikely to be selected as an object of quantization but that is important from an auditory standpoint has been forcibly quantized, and even if the frequencies that should really be selected as an object of coding on the coding side is not selected, an error spectrum located at the frequencies included in a band that is important from an auditory standpoint is quantized without fail, enabling quality to be improved.

It is also possible for an enhancement layer coder and enhancement layer decoder of this embodiment to employ a configuration combining this embodiment and Embodiment 13.FIG. 38is a block diagram showing an example of the internal configuration of the frequency determination section of a sound coding apparatus of this embodiment. Parts inFIG. 38identical to those inFIG. 22are assigned the same reference numerals as inFIG. 22and detailed descriptions thereof are omitted.

InFIG. 38, MDCT section2101multiplies the input signal output from subtracter1606by an analysis window, then performs MDCT (Modified Discrete Cosine Transform) processing to obtain the MDCT coefficients, and outputs the MDCT coefficients to MDCT coefficient quantizer3701.

Ordering section3201receives frequency information obtained by frequency determination section1607, and calculates the amount by which estimated error spectrum E′(m) of each frequency exceeds estimated auditory masking M′(m) (hereinafter referred to as the estimated distortion value), D(m).

A band important in terms of auditory perception is set beforehand in fixed band specification section3501.

MDCT coefficient quantizer3701performs quantization, allocating bits proportionally to error spectra E(m) positioned at frequencies in high-to-low distortion value D(m) order based on frequency information ordered according to estimated distortion value D(m). MDCT coefficient quantizer3701also encodes the coefficients in a band set by fixed band specification section3501.

The decoding side will now be described.FIG. 39is a block diagram showing an example of the internal configuration of the enhancement layer decoder of a sound decoding apparatus according to Embodiment 14 of the present invention. Parts inFIG. 39identical to those inFIG. 25are assigned the same reference numerals as inFIG. 25and detailed descriptions thereof are omitted.

InFIG. 39, ordering section3401receives frequency information obtained by frequency determination section2304, and calculates the amount by which estimated error spectrum E′(m) of each frequency exceeds estimated auditory masking M′(m) (hereinafter referred to as the estimated distortion value), D(m).

Then ordering section3401performs ordering in high-to-low estimated distortion value D(m) order, and outputs the corresponding frequency information to MDCT coefficient decoder3801. A band important in terms of auditory perception is set beforehand in fixed band specification section3601.

MDCT coefficient decoder3801decodes the MDCT coefficients quantized from second coding information output from demultiplexer2301based on the error spectrum frequencies subject to decoding output from ordering section3401. To be specific, MDCT coefficient decoder3801positions decoded MDCT coefficients corresponding to frequencies indicated by ordering section3401and fixed band specification section3601, and supplies zero for other frequencies.

IMDCT section2402executes inverse MDCT processing on the MDCT coefficients output from MDCT coefficient decoder3801, generates a time domain signal, and outputs this time domain signal to overlap adder2403.

Embodiment 15 of the present invention will now be described with reference to the attached drawings.FIG. 40is a block diagram showing the configuration of a communication apparatus according to Embodiment 15 of the present invention. A feature of this embodiment is that signal processing apparatus3903inFIG. 40is configured as one of the sound coding apparatuses shown in above-described Embodiment 1 through Embodiment 14.

As shown inFIG. 40, a communication apparatus3900according to Embodiment 15 of the present invention comprises an input apparatus3901, A/D conversion apparatus3902, and signal processing apparatus3903connected to a network3904.

A/D conversion apparatus3902is connected to an output terminal of input apparatus3901. An input terminal of signal processing apparatus3903is connected to an output terminal of A/D conversion apparatus3902. An output terminal of signal processing apparatus3903is connected to network3904.

Input apparatus3901converts a sound wave audible to the human ear to an analog signal, which is an electrical signal, and supplies this analog signal to A/D conversion apparatus3902. A/D conversion apparatus3902converts the analog signal to a digital signal, and supplies this digital signal to signal processing apparatus3903. Signal processing apparatus3903encodes the input digital signal and generates code, and outputs this code to network3904.

Thus, according to a communication apparatus of this embodiment of the present invention, effects such as shown in above-described Embodiments 1 through 14 can be obtained in communications, and it is possible to provide a sound coding apparatus that encodes an acoustic signal efficiently with a small number of bits.

Embodiment 16 of the present invention will now be described with reference to the attached drawings.FIG. 41is a block diagram showing the configuration of a communication apparatus according to Embodiment 16 of the present invention. A feature of this embodiment is that signal processing apparatus4003inFIG. 41is configured as one of the sound decoding apparatuses shown in above-described Embodiment 1 through Embodiment 14.

As shown inFIG. 41, a communication apparatus4000according to Embodiment 16 of the present invention comprises a receiving apparatus4002connected to a network4001, a signal processing apparatus4003, a D/A conversion apparatus4004, and an output apparatus4005.

Receiving apparatus4002is connected to network4001. An input terminal of signal processing apparatus4003is connected to an output terminal of receiving apparatus4002. An input terminal of D/A conversion apparatus4004is connected to an output terminal of signal processing apparatus4003. An input terminal of output apparatus4005is connected to an output terminal of D/A conversion apparatus4004.

Receiving apparatus4002receives a digital coded acoustic signal from network4001, generates a digital received acoustic signal, and supplies this received acoustic signal to signal processing apparatus4003. Signal processing apparatus4003receives the received acoustic signal from receiving apparatus4002, performs decoding processing on this received acoustic signal and generates a digital decoded acoustic signal, and supplies this digital decoded acoustic signal to D/A conversion apparatus4004. D/A conversion apparatus4004converts the digital decoded speech signal from signal processing apparatus4003and generates an analog decoded speech signal, and supplies this analog decoded speech signal to output apparatus4005. Output apparatus4005converts the analog decoded speech signal, which is an electrical signal, to air vibrations, and outputs these air vibrations so as to be audible to the human ear as a sound wave.

Thus, according to a communication apparatus of this embodiment, effects such as shown in above-described Embodiments 1 through 14 can be obtained in communications, and it is possible to decode an acoustic signal coded efficiently with a small number of bits, enabling a good acoustic signal to be output.

Embodiment 17 of the present invention will now be described with reference to the attached drawings.FIG. 42is a block diagram showing the configuration of a communication apparatus according to Embodiment 17 of the present invention. A feature of this embodiment is that signal processing apparatus4103inFIG. 42is configured as one of the sound coding apparatuses shown in above-described Embodiment 1 through Embodiment 14.

As shown inFIG. 42, a communication apparatus4100according to Embodiment 17 of the present invention comprises an input apparatus4101, A/D conversion apparatus4102, signal processing apparatus4103, RF modulation apparatus4104, and antenna4105.

Input apparatus4101converts a sound wave audible to the human ear to an analog signal, which is an electrical signal, and supplies this analog signal to A/D conversion apparatus4102. A/D conversion apparatus4102converts the analog signal to a digital signal, and supplies this digital signal to signal processing apparatus4103. Signal processing apparatus4103encodes the input digital signal and generates a coded acoustic signal, and supplies this coded acoustic signal to RF modulation apparatus4104. RF modulation apparatus4104modulates the coded acoustic signal and generates a modulated coded acoustic signal, and supplies this modulated coded acoustic signal to antenna4105. Antenna4105transmits the modulated coded acoustic signal as a radio wave.

Thus, according to a communication apparatus of this embodiment, effects such as shown in above-described Embodiments 1 through 14 can be obtained in radio communications, and it is possible to code an acoustic signal efficiently with a small number of bits.

The present invention can be applied to a transmitting apparatus, transmit coding apparatus, or acoustic signal coding apparatus that uses audio signals. The present invention can also be applied to a mobile station apparatus or base station apparatus.

Embodiment 18 of the present invention will now be described with reference to the attached drawings.FIG. 43is a block diagram showing the configuration of a communication apparatus according to Embodiment 18 of the present invention. A feature of this embodiment is that signal processing apparatus4203inFIG. 43is configured as one of the sound decoding apparatuses shown in above-described Embodiment 1 through Embodiment 14.

As shown inFIG. 43, a communication apparatus4200according to Embodiment 18 of the present invention comprises an antenna4201, RF demodulation apparatus4202, signal processing apparatus4203, D/A conversion apparatus4204, and output apparatus4205.

Antenna4201receives a digital coded acoustic signal as a radio wave, generates a digital received coded acoustic signal, which is an electrical signal, and supplies this digital received coded acoustic signal to RF demodulation apparatus4202. RF demodulation apparatus4202demodulates the received coded acoustic signal from antenna4201and generates a demodulated coded acoustic signal, and supplies this demodulated coded acoustic signal to signal processing apparatus4203.

Signal processing apparatus4203receives the digital demodulated coded acoustic signal from RF demodulation apparatus4202, performs decoding processing and generates a digital decoded acoustic signal, and supplies this digital decoded acoustic signal to D/A conversion apparatus4204. D/A conversion apparatus4204converts the digital decoded speech signal from signal processing apparatus4203and generates an analog decoded speech signal, and supplies this analog decoded speech signal to output apparatus4205. Output apparatus4205converts the analog decoded speech signal, which is an electrical signal, to air vibrations, and outputs these air vibrations so as to be audible to the human ear as a sound wave.

Thus, according to a communication apparatus of this embodiment, effects such as shown in above-described Embodiments 1 through 14 can be obtained in radio communications, and it is possible to decode an acoustic signal coded efficiently with a small number of bits, enabling a good acoustic signal to be output.

The present invention can be applied to a receiving apparatus, receive decoding apparatus, or speech signal decoding apparatus that uses audio signals. The present invention can also be applied to a mobile station apparatus or base station apparatus.

The present invention is not limited to the above-described embodiments, and various variations and modifications may be possible without departing from the scope of the present invention. For example, in the above embodiments a case has been described in which the present invention is implemented as a signal processing apparatus, but the present invention is not limited to this, and this signal processing method can also be implemented as software.

For example, it is also possible for a program that executes the above-described signal processing method to be stored in ROM (Read Only Memory) beforehand, and forth is program to be operated by a CPU (Central Processing Unit).

It is also possible for a program that executes the above-described signal processing method to be stored in a computer-readable storage medium, for the program stored in the storage medium to be recorded in RAM (Random Access Memory) of a computer, and for the computer to be operated in accordance with that program.

In the above description, a case has been described in which MDCT is used as a method of transformation from the time domain to the frequency domain, but the present invention is not limited to this, and any transformation method can be applied as long as it is an orthogonal transformation method. For example, a discrete Fourier transform, discrete cosine transform or wavelet transform method can also be applied.

The present invention can be applied to a receiving apparatus, receive decoding apparatus, or speech signal decoding apparatus that uses audio signals. The present invention can also be applied to a mobile station apparatus or base station apparatus.

As is clear from the above description, according to a coding apparatus, decoding apparatus, coding method, and decoding method of the present invention, by performing enhancement layer coding using information obtained from base layer coding information, it is possible to perform high-quality coding at a low bit rate even in the case of a signal in which speech is predominant and music or environmental sound is superimposed in the background.

This application is based on Japanese Patent Application No. 2002-127541 filed on Apr. 26, 2002, and Japanese Patent Application No. 2002-267436 filed on Sep. 12, 2002, entire content of which is expressly incorporated by reference herein.

INDUSTRIAL APPLICABILITY

The present invention is suitable for use in apparatuses that code and decode speech signals, and communication apparatuses.