Abstract: The purpose of the present invention is to more efficiently extend, using a low bit rate, the bandwidth of input signals having a harmonics structure, in order to obtain better audio quality. The present invention is installed in a device that extends bandwidth for audio signal encoding and decoding. This novel bandwidth extension encoding identifies a low-frequency spectrum component having the highest correlation to a high-frequency bandwidth signal among input signals, duplicates a high-frequency spectrum by energy adjustment of said component, and maintains the harmonic relationship between the low-frequency spectrum and the duplicated high-frequency spectrum by adjusting the spectral peak position of the duplicated high-frequency spectrum, on the basis of a harmonic frequency estimated from a composite low-frequency spectrum.

Abstract: This invention introduces apparatus and methods to efficiently encode the quantization parameters of split multi-rate lattice vector quantization. In this invention, by doing spectral analysis on the split multi-rate vector quantized spectrum, the spectrum is split to null vectors region and non-null vectors region. For the null vectors region, instead of transmitting series of indication for null vectors, an indication of null vectors region and the quantized value of index of the ending vector in the null vectors region (or the number of the null vectors in the null vectors region) are transmitted. The indication of null vectors region can be designed in many ways, the only requirement is the indication should be distinguishable in the decoder side. The ending index or the number of null vectors can be quantized by an adaptively designed codebook. By applying of the invented method, some bits can be saved from the codebook indications.

Abstract: Provided are a voice audio encoding device, voice audio decoding device, voice audio encoding method, and voice audio decoding method that efficiently perform bit distribution and improve sound quality. Dominant frequency band identification unit identifies a dominant frequency band having a norm factor value that is the maximum value within the spectrum of an input voice audio signal. Dominant group determination units and non-dominant group determination unit group all sub-bands into a dominant group that contains the dominant frequency band and a non-dominant group that contains no dominant frequency band. Group bit distribution unit distributes bits to each group on the basis of the energy and norm variance of each group. Sub-band bit distribution unit redistributes the bits that have been distributed to each group to each sub-band in accordance with the ratio of the norm to the energy of the groups.

Abstract: Disclosed is an audio encoding device which removes unnecessary inter-channel parameters from the subject to be encoded, improving the encoding efficiency thereby. In this audio encoding device, a principal component analysis unit (301) converts an inputted left signal {Lsb(f)} and an inputted right signal {Rsb(f)} into a principal component signal {PCsb(f)} and an ambient signal {Asb(f)} and calculates for each sub-band, a rotation angle which indicates the degree of conversion; a monophonic encoding unit (303) encodes the principal component signal {Pcsb(f)}; a rotation angle encoding unit (302) encodes the angle of rotation {?b}; a local monophonic decoding unit (603) creates a decoded principal component signal; and a redundant parameter elimination unit (604) identifies the redundant parameters by analyzing the encoding quality of the decoded principal component signal and eliminates the redundant parameters from the signal to be encoded.

Abstract: This invention introduces audio/speech encoding apparatus audio/speech decoding apparatus, audio/speech encoding method and audio/speech decoding method to efficiently encode the quantization parameters of split multi-rate lattice vector quantization. In this invention, the position of the sub-vector whose codebook indication consumes the most bits is firstly located, and then the value of the codebook is estimated based on the total number of bits available and the bits usage information for other sub-vectors. The difference value is calculated between the actual value and estimated value. Finally, instead of transmitting the codebook indication which consumes the most bits, the position of the sub-vector whose codebook indication consumes the most bits and the difference value between the actual value and the estimated value are transmitted. By applying of the invented method, bits can be saved by the codebook indications.

Abstract: An encoder whereby the bit efficiency of encoding can be improved, thereby improving the qualities of signals as decoded. In the encoder: a time-frequency converting unit (101) converts signals, which are to be encoded, to frequency domain signals; an adaptive spectrum formation encoding unit (102) determines an effective range in the frequency band of the frequency domain signals; and a pulse vector encoding unit (103) pulse vector encodes only the signal components within the effective range.

Abstract: An audio encoding apparatus capable of reducing the bit rate even if a codebook having a larger codebook number is selected in a split multi-rate lattice vector quantization is provided. Sub-vector determining unit (121) determines, in the spectrum of an input signal having been divided into a predetermined number of sub-vectors, a sub-vector using the largest number of bits. Positional information encoding unit (122) encodes the positional information of the determined sub-vector. Codebook indication value estimating unit (124) estimates a number of used bits for a codebook indication value of the largest number of used bits by use of the (N?1) other codebook indication values, and generates a number-of-used-bits estimation value. Difference calculating unit (125) calculates a difference by subtracting the number-of-used-bits estimation value from the actual value of the codebook indication value of the largest number of used bits. Difference encoding unit (126) encodes the difference information.

Abstract: In this invention, the design of the Huffman table can be done offline with a large input sequence database. The range of the quantization indices (or differential indices) for Huffman coding is identified. For each value of range, all the input signal which have the same range will be gathered and the probability distribution of each value of the quantization indices (or differential indices) within the range is calculated. For each value of range, one Huffman table is designed according to the probability. And in order to improve the bits efficiency of the Huffman coding, apparatus and methods to reduce the range of the quantization indices (or differential indices) are also introduced.

Abstract: Disclosed is a stereo acoustic signal encoding apparatus in which the signal quality does not deteriorate if there are a plurality of sound sources. A peak tracing unit (401) splits frames of a right channel signal and a left channel signal into a plurality of sub frames; detects the peaks of wave shapes of the split sub frames; and estimates a frame delay time D for each frame of the right channel signal and the left channel signal by comparing the positions of the detected peaks. A time adjusting unit (402) adjusts the time of the right channel signal on the basis of the frame time delay D. A down-mix operation is carried out using the right channel signal which has been subjected to the time adjustment and the left channel signal to generate a mono signal and a sub signal. A mono signal encoding unit (403) encodes the mono signal. A sub signal encoding unit (404) encodes the sub signal. The time delay encoding unit (405) encodes the frame time delay D.

Abstract: Provided is an audio encoding device that can suppress degradation of audio quality. Spectral coefficients of synthesized signal from CELP core layer are utilized to fulfill spectral gaps in error signal spectrum coefficients from a transform coding layer. By both spectral coefficients, decoded signal spectral coefficients are generated. The decoded signal spectral coefficients and the input signal spectral coefficients are divided into a plurality of sub bands. In each sub band, the energy of the input signal spectral coefficient corresponding to a zero decoded error signal spectral coefficient is calculated, and the energy of the decoded signal spectral coefficient corresponding to the zero decoding error signal spectral coefficient is calculated, and their energy ratio is calculated and is quantized and transmitted.

Abstract: Provided is an encoder which can decode a high-quality stereo signal while keeping the amount of information in the bit allocation information to a minimum when a scalable coding technique is used for a stereo signal. In the encoder, a principal component analysis (PCA) converter converts the left signal and the right signal of the stereo signal and generates the main signal of the first layer and the sub-signal of the first layer. In the first layer to the M-th layer (where M is a natural number, 2 or greater), an adaptive residual encoder compares the importance of the main signal of the m-th layer, where m is a natural number from 1 to M, and the importance of the sub-signal of the m-th layer, selects the signal having the higher importance, encodes the selected signal, and generates the encoded data of the m-th layer.

Abstract: This invention introduces apparatus and methods to efficiently encode the quantization parameters of split multi-rate lattice vector quantization. In this invention, by doing spectral analysis on the split multi-rate vector quantized spectrum, the spectrum is split to null vectors region and non-null vectors region. For the null vectors region, instead of transmitting series of indication for null vectors, an indication of null vectors region and the quantized value of index of the ending vector in the null vectors region (or the number of the null vectors in the null vectors region) are transmitted. The indication of null vectors region can be designed in many ways, the only requirement is the indication should be distinguishable in the decoder side. The ending index or the number of null vectors can be quantized by an adaptively designed codebook. By applying of the invented method, some bits can be saved from the codebook indications.

Abstract: An encoder improves inter-channel prediction (ICP) performance in scalable stereo sound encoding using an ICP. In the encoder, ICP analysis units use, as reference signal candidates, a frequency coefficient in the low-band portion of a side residual signal, a frequency coefficient in each sub-band portion of a monaural residual signal, and a frequency coefficient in the low-band portion of the monaural residual signal, respectively, and perform an ICP analysis between the these respective candidates and a frequency coefficient in each sub-band portion of the side residual signal to generate first, second, and third ICP coefficients.

Abstract: An audio encoding apparatus that allows a decoded signal exhibiting an excellent sound quality to be obtained on a decoding side. In the audio encoding apparatus (1000A), a time-frequency transform unit (1001) uses a time-frequency transform, such as a discrete Fourier transform (DFT) or a modified discrete cosine transform (MDCT), to transform a time domain signal (S(n)) to a frequency domain signal (spectrum factor) (S(f)). A psychoacoustic model analyzing unit (1002) performs a psychoacoustic model analysis of the frequency domain signal (S(f)), thereby obtaining a masking curve. An acoustic sense weighting unit (1003) estimates, based on the masking curve, an importance degree of acoustic sense, and determines and applies the weighting factors of respective spectrum factors to the respective spectrum factors. An encoding unit (1004) encodes the frequency domain signal (S(f)) as weighted in terms of the acoustic sense. A multiplexing unit (1005) multiplexes and transmits the encoded parameters.

Abstract: Disclosed is an audio encoding device which removes unnecessary inter-channel parameters from the subject to be encoded, improving the encoding efficiency thereby. In this audio encoding device, a principal component analysis unit (301) converts an inputted left signal {Lsb(f)} and an inputted right signal {Rsb(f)} into a principal component signal {PCsb(f)} and an ambient signal {Asb(f)} and calculates for each sub-band, a rotation angle which indicates the degree of conversion; a monophonic encoding unit (303) encodes the principal component signal {Pcsb(f)); a rotation angle encoding unit (302) encodes the angle of rotation {?b}; a local monophonic decoding unit (603) creates a decoded principal component signal; and a redundant parameter elimination unit (604) identifies the redundant parameters by analyzing the encoding quality of the decoded principal component signal and eliminates the redundant parameters from the signal to be encoded.

Abstract: An encoder whereby the bit efficiency of encoding can be improved, thereby improving the qualities of signals as decoded. In the encoder: a time-frequency converting unit (101) converts signals, which are to be encoded, to frequency domain signals; an adaptive spectrum formation encoding unit (102) determines an effective range in the frequency band of the frequency domain signals; and a pulse vector encoding unit (103) pulse vector encodes only the signal components within the effective range.

Abstract: Disclosed is a stereo acoustic signal encoding apparatus in which the signal quality does not deteriorate if there are a plurality of sound sources. A peak tracing unit (401) splits frames of a right channel signal and a left channel signal into a plurality of sub frames; detects the peaks of wave shapes of the split sub frames; and estimates a frame delay time D for each frame of the right channel signal and the left channel signal by comparing the positions of the detected peaks. A time adjusting unit (402) adjusts the time of the right channel signal on the basis of the frame time delay D. A down-mix operation is carried out using the right channel signal which has been subjected to the time adjustment and the left channel signal to generate a mono signal and a sub signal. A mono signal encoding unit (403) encodes the mono signal. A sub signal encoding unit (404) encodes the sub signal. The time delay encoding unit (405) encodes the frame time delay D.

Abstract: Provided is an encoder which can decode a high-quality stereo signal while keeping the amount of information in the bit allocation information to a minimum when a scalable coding technique is used for a stereo signal. In the encoder, a principal component analysis (PCA) converter (101) PCA converts the left signal and the right signal of the stereo signal and generates the main signal of the first layer and the sub-signal of the first layer. In the first layer to the M-th layer (where M is a natural number, 2 or greater), an adaptive residual encoder (102-m) (where m is a natural number from 1 to M) compares the importance of the main signal of the m-th layer and the importance of the sub-signal of the m-th layer, selects the signal having the higher importance, encodes the selected signal, and generates the encoded data of the m-th layer.

Abstract: There is provided an encoder capable of improving inter-channel prediction (ICP) performance in scalable stereo sound encoding using an ICP. In the encoder, ICP analysis units (113, 114, 115) use, as reference signal candidates, a frequency coefficient (sL?(f)) in the low-band portion of a side residual signal, a frequency coefficient (mM,i(f)) in each sub-band portion of a monaural residual signal, and a frequency coefficient (mL(f)) in the low-band portion of the monaural residual signal, respectively, and perform an ICP analysis between the respective these candidates and a frequency coefficient (sM,i(f)) in each sub-band portion of the side residual signal to generate first, second, and third ICP coefficients.