Abstract: There is disclosed a scalable encoding device capable of increasing the conversion performance from a narrow-band LSP to a wide-band LSP (prediction accuracy when predicting the wide-band LSP from the narrow-band LSP) and realizing a high-performance band scalable LSP encoding. The device includes a conversion coefficient calculation unit (109) for calculating a conversion coefficient by using a narrow-band quantization LSP which has been outputted from a narrow-band LSP encoding unit (103) and a wide-band quantization LSP which has been outputted from a wide-band LSP encoding unit (107). The wide-band LSP encoding unit (107) multiplies the narrow-band quantization LSP with the conversion coefficient inputted from the conversion coefficient calculation unit (109) so as to convert it into a wide-band LSP. The wide-band LSP is multiplied by a weight coefficient to calculate a prediction wide-band LSP.

Abstract: A sound decoding device is capable of improving the lost frame compensation performance and improving quality of the decoded sound. A rise frame sound source compensation unit generates a compensation sound source signal when the current frame is a lost frame and a rise frame. An average sound source pattern update unit updates the average sound source pattern held in an average sound source pattern holding unit over a plurality of frames. When a frame is lost, an LPC synthesis unit performs LPC synthesis on a decoded sound source signal by using the compensation sound source signal inputted via a switching unit and a decoded LPC parameter from an LPC decoding unit and outputs the compensation decoded sound signal.

Abstract: There is disclosed an encoder apparatus whereby, when a band expanding technique for encoding, based on the spectral data of a lower frequency portion, the spectral data of a higher frequency portion is applied to a lower layer in a hierarchical encoding/decoding system, an efficient encoding can be performed in an upper layer as well, thereby improving the decoded-signal quality. In an encoder apparatus (101), a second layer decoder unit (207) calculates a spectrum (differential spectrum), which is to be encoded in a third layer encoder unit (210) that is an upper layer of the second layer decoder unit (207), by applying such an ideal gain (first gain parameter a1) that minimizes the energy of the differential spectrum.

Abstract: Disclosed is an audio encoding device capable of adjusting a spectrum inclination of a quantized noise without changing the Formant weight. The device includes: an HPF (131) which extracts a high-frequency component of the frequency region from an input audio signal; a high-frequency energy level calculation unit (132) which calculates an energy level of the high-frequency component in a frame unit; an LPF (133) which extracts a low-frequency component of the frequency region from the input audio signal; a low-energy level calculation unit (134) which calculates an energy level of a low-frequency component in a frame unit; an inclination correction coefficient calculation unit (141) multiplies the difference between SNR of the high-frequency component and SNR of the low-frequency component inputted from an adder (140) by a constant and adds a bias component to the product so as to calculate an inclination correction coefficient ?3.

Abstract: There is provided a wide-band LSP prediction device and others capable of predicting a wide-band LSP from a narrow-band LSP with a high quantization efficiency and a high accuracy while suppressing the size of a conversion table correlating the narrow-band LSP to the wide-band LSP. In this device, a non-linear prediction unit (102) performs non-linear prediction by using a converted wide-band LSP inputted from a narrow-band/wide-band conversion unit (101) and inputs the non-linear prediction result to an amplifier (103). The converted wide-band LSP is inputted to an amplifier (104). An adder (122) adds multiplication results (vectors) inputted from the amplifiers (103, 104).

Abstract: An encoding apparatus and method for generating low-frequency-band encoding information and high-frequency-band encoding information from an original signal. The encoding apparatus includes a first spectrum calculator that calculates a first spectrum of a low frequency band from a decoded signal of the low-frequency-band encoding information, a second spectrum calculator that calculates a second spectrum from the original signal, an estimator that divides a high frequency band of the second spectrum into a plurality of bands and estimates the second spectrum included in each band, using the first spectrum, and a first error component encoder that encodes a first error component between the high frequency band of the second spectrum and an estimated spectrum. A corresponding decoding apparatus and method provides decoding.

Abstract: A scalable decoder capable of avoiding deterioration in subjective quality of a listener. The scalable decoder for decoding core layer encoding data and extension layer encoding data including an extension layer gain coefficient, wherein a voice analysis section detects variation in power of a core layer decoding voice signal being obtained from the core layer encoding data, a gain attenuation rate calculating section (140) sets the attenuation intensity variable depending on variation in power, and a gain attenuation section (143) attenuates the extension layer gain coefficient in a second period preceding a first period according to a set attenuation intensity when extension layer encoding data in the first period is missing, thus interpolating the extension layer gain coefficient in the first period.

Abstract: An audio decoding device performs frame loss compensation capable of obtaining a decoded audio which is natural for ears with little noise. The audio decoding device includes a non-cyclic pulse waveform detection unit for detecting a non-cyclic pulse waveform section in a n?1-th frame, which is repeatedly used with a pitch cycle in the n-th frame upon compensation of loss of the n-th frame. The audio coding device also includes a non-cyclic pulse waveform suppression unit for suppressing a non-cyclic pulse waveform by replacing an audio source signal existing in the non-cyclic pulse waveform section in the n?1-th frame by a noise signal. The audio coding device further includes a synthesis filter for using a linear prediction coefficient decoded by an LPC decoding unit to perform synthesis by a synthesis filter by using the audio source signal of the n?1-th frame from the non-cyclic pulse waveform suppression unit as a drive audio source, thereby obtaining the decoded audio signal of the n-th frame.

Abstract: A scalable decoder capable of preventing degradation of the quality of the decoded signal in a disappeared data interpolation in band scalable coding. A core layer decoding section (101) acquires a core layer decoded signal and narrow band spectrum information by decoding. A narrow band spectrum slope computing section (103) computes the slope of an attenuation line of a narrow band spectrum from the narrow band spectrum information. An extended layer disappearance detection section (104) detects whether extended layer coded data has disappeared or not. An extended layer decoding section (105) normally decodes the extended layer coded data. If the extended layer disappears, a parameter required for decoding is interpolated and synthesizes an interpolation decoded signal by the interpolated parameter. The gain of the interpolated data is controlled according to the results of the computation, by the narrow band spectrum slope computing section (103).

Abstract: Provided is a decoding device that can reduce abrupt changes in the number of channels in a decoded signal when transmission errors occur as a result of lost frames in an encoding/decoding system for multichannel signals. Said decoding device is also capable of per-sample smoothing and can reduce degradation of audio quality. In the provided device, a demultiplexer (301) receives an encoded monaural signal and an encoded differential signal and detects change over time in the received encoded differential signal. An M signal decoder (302) decodes the encoded monaural signal and obtains a decoded monaural signal. An S signal decoder (303) decodes the encoded differential signal and obtains a decoded differential signal. A smoothing unit (304) performs smoothing on the decoded differential signal by means of a computation involving the decoded differential signal and coefficients corresponding to the change over time detected by the demultiplexer (301).

Abstract: An encoder, decoder, encoding method, and decoding method enabling acquisition of high-quality decoded signal in scalable encoding of an original signal in first and second layers even if the second or upper layer section performs low bit-rate encoding. In the encoder, a spectrum residue shape codebook stores candidates of spectrum residue shape vectors, a spectrum residue gain codebook stores candidates of spectrum residue gains, and a spectrum residue shape vector and a spectrum residue gain are sequentially outputted from the candidates according to the instruction from a search section. A multiplier multiplies a candidate of the spectrum residue shape vector by a candidate of the spectrum residue gain and outputs the result to a filtering section. The filtering section performs filtering by using a pitch filter internal state set by a filter state setting section, a lag T outputted by a lag setting section, and a spectrum residue shape vector which has undergone gain adjustment.

Abstract: Disclosed is a voice decoding apparatus wherein the processor may be continuously employed for other applications for a prescribed time but, in response to an urgent interrupt, the processor can generate synthesised sound even when being used for other applications, without interruption. In this apparatus, a packet receiving section (101) receives packets of the layers of a plurality of frames and extracts code from the received packets. A state/code storage section (103) stores the code and decoding state of the code. A layer selection section (104) selects a layer number and a frame number corresponding to the code to be initially decoded, based on the decoding state. A decoding section (105) decodes the code of the selected frame number and layer number.

Abstract: A prediction performance between individual channels of a stereo signal is improved to improve a sound quality of a decoded signal. A first low pass filter LPF interrupts a high-range component of a first channel signal S1, and outputs a first low-range component S1?. A second low pass filter LPF interrupts a high-range component of a second channel signal S2, and outputs a second low-range component S2?. A predictor predicts the S2? from the S1?, and outputs a prediction parameter composed of a delay time difference t and an amplitude ratio g. first channel encoder encodes the S1. A prediction parameter encoder encodes the prediction parameter. The encoded parameters of the encoded parameter of the S1 and the prediction parameter are then outputted.

Abstract: Provided is a speech encoding device that is capable of performing encoding in an extension encoder even when the core encoder and core decoder of each layer have been interchanged, and that is also capable of performing high precision encoding by using the appropriate codec for each situation. The speech encoding device (100) performs hierarchical encoding of a speech signal by using the information of a lower layer in a higher layer. A core encoder (102) in the speech encoding device (100) generates a code by encoding the speech signal. A core decoder (104) generates a decoded signal by decoding the code generated by the core encoder (102). An adding unit (106) detects the encoding residual between the speech signal and the decoded signal generated by the core decoder (104). An auxiliary analyzing unit (107) inputs the decoded signal and generates lower layer information by conducting analysis processing and adjustment processing.

Abstract: Provided is an encoder which can effectively encode/decode spectrum data of a broad frequency signal in a high frequency range, can dramatically reduce the number of the arithmetic operations to be performed, and can improve the quality of the decoded signal. The encoder comprises a first layer coding unit (202) which encodes an input signal in a low frequency range below a predetermined frequency to generate first coded information, a first layer decoding unit (203) which decodes the first coded information to generate a decoded signal, and a second layer coding unit (206) which splits the input signal in a high frequency range above a predetermined frequency, into a plurality of sub-bands, presumes the respective sub-hands from the input signal or decoded signal, partially selects a spectrum component within each sub-band, and calculates an amplitude adjustment parameter used to adjust the amplitude of the selected spectrum component to thereby generate second coding information.

Abstract: Disclosed is a tone determination device that determines the tonality of an input signal using correlations between the frequency components of a current frame with the frequency components of the preceding frame, such that the tone determination device is able to decrease the calculation complexity. In the device, a vector coupling unit (104) couples some of the SDFT coefficients of the preceding frame with some of the down-sampled SDFT coefficients of the preceding frame to generate new SDFT coefficients, and also couples some of the SDFT coefficients of the current frame with some of the down-sampled SDFT coefficients of the current frame to generate new SDFT coefficients. A correlation analysis unit (105) finds correlations for the SDFT coefficients between frames, and also finds the power of the current frame for each specific band.

Abstract: An encoding apparatus and method for generating low-frequency-band encoding information and high-frequency-band encoding information from an original signal. The encoding apparatus includes a first spectrum calculator that calculates a first spectrum of a low frequency band from a decoded signal of the low-frequency-band encoding information, a second spectrum calculator that calculates a second spectrum from the original signal, an estimator that divides a high frequency band of the second spectrum into a plurality of bands and estimates the second spectrum included in each band, using the first spectrum, and a first error component encoder that encodes a first error component between the high frequency band of the second spectrum and an estimated spectrum. A corresponding decoding apparatus and method provides decoding.

Abstract: A CELP speech decoder includes an adaptive codebook that generates an adaptive code vector and a random codebook that generates a random code vector. The random codebook includes an input vector provider that provides an input vector including at least one pulse, each pulse having a position and a polarity, a fixed waveform storage that stores at least one fixed waveform, and a selector that selects at least one of a first process and a second process based on a value of an adaptive codebook gain. The random codebook further includes a convolution section that generates the random code vector by convoluting the at least one fixed waveform with the input vector when the first process is selected. A synthesis filter outputs synthesized speech by performing linear prediction coefficient synthesis on a signal based on the adaptive code vector and the random code vector.

Abstract: There is provided a scalable encoding device capable of realizing a bandwidth scalable LSP encoding with high performance by improving the conversion performance from narrow band LSPs to wide band LSPs.

Abstract: There is disclosed a speech switching device capable of improving quality of a decoded signal. In the device, a weighted addition unit outputs a mixed signal of a narrow-band speech signal and a wide-band speech signal when switching the speech signal band. A mixing unit formed by an extended layer decoded speech amplifier and an adder mixes the narrow-band speech signal with the wide-band speech signal while changing the mixing ratio of the narrow-band speech signal and the wide-band speech signal as the time elapses, thereby obtaining a mixed signal. An extended layer decoded speech gain controller variably sets the degree of change of the mixing ratio by the time.