Abstract: There is disclosed a stereo encoding device capable of accurately encoding a stereo signal at a low bit rate and suppressing delay in audio communication. The device performs monaural encoding in its first layer (110). In a second layer (120), a filtering unit (103) generates an LPC (Linear Predictive Coding) coefficient and generates a left channel drive sound source signal. A time region evaluation unit (104) and a frequency region evaluation unit (105) perform signal evaluation and prediction in both of their regions. A residual encoding unit (106) encodes a residual signal. A bit distribution control unit (107) adaptively distributes bits to the time region evaluation unit (104), the frequency region evaluation unit (105), and the residual encoding unit (106) according to a condition of the audio signal.

Abstract: A spectrum modifying method and the like wherein the efficiencies of the signal estimation and prediction can be improved and the spectrum can be more efficiently encoded. According to this method, the pitch period is calculated from an original signal, which serves as a reference signal, and then a basic pitch frequency (f0) is calculated. Thereafter, the spectrum of a target signal, which is a target of spectrum modification, is divided into a plurality of partitions. It is specified here that the width of each partition be the basic pitch frequency. Then, the spectra of bands are interleaved such that a plurality of peaks having similar amplitudes are unified into a group. The basic pitch frequency is used as an interleave pitch.

Abstract: A scalable encoding device capable of reducing an encoding rate to reduce a circuit scale while preventing sound quality deterioration of a decoded signal. An extension layer is coarsely divided into a system for processing a first channel and a system for processing a second channel. A sound source predictor for processing the first channel predicts a drive sound source signal of the first channel from a drive sound source signal of a monaural signal, and outputs the predicted drive sound source signal through a multiplier to a first CELP encoder. A sound source predictor for processing the second channel predicts the drive sound source signal of the second channel from the drive sound source signal of the monaural signal and the output from the first CELP encoder, and outputs the predicted drive sound source signal through a multiplier to a second CELP encoder.

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: A scalable encoding device prevents sound quality deterioration of a decoded signal, reduces the encoding rate, and reduces the circuit size. The scalable encoding device includes a first layer encoder for generating a monaural signal by using a plurality of channel signals (L channel signal and R channel signal) constituting a stereo signal and encoding the monaural signal to generate a sound source parameter. The scalable encoding device also includes a second layer encoder for generating a first conversion signal by using the channel signal and the monaural signal, generating a synthesis signal by using the sound source parameter and the first conversion signal, and generating a second conversion coefficient index by using the synthesis signal and the first conversion signal.

Abstract: A pulse allocating method capable of coding stereophonic voice signals efficiently. In the fixed code note retrievals of this pulse allocating method, for individual subframes, the stereophonic voice signals are compared to judge similarity between channels, and are judged on their characteristics. On the basis of the similarity between the channels and the characteristics of the stereophonic signals, the pulse numbers to be allocated to the individual channels are determined. Pulse retrievals are executed to determine the pulse positions for the individual channels, so that the pulses determined are coded.

Abstract: A stereo signal generating apparatus capable of obtaining stereo signals that exhibit a low bit rate and an excellent reproducibility. In this stereo signal generating apparatus (90), an FT part (901) converts a monaural signal (M?t) of time domain to a monaural signal (M?) of frequency domain. A power spectrum calculating part (902) determines a power spectrum (PM?). A scaling ratio calculating part (904a) determines a scaling ratio (SL) for a left channel, while a scaling ratio calculating part (904b) determines a scaling ratio (SR) for a right channel. A multiplying part (905a) multiplies the monaural signal (M?) of frequency domain by the scaling ratio (SL) to produce a left channel signal (L?) of a stereo signal, while a multiplying part (905b) multiplies the monaural signal (M?) of frequency domain by the scaling ratio (SR) to produce a right channel signal (R?) of the stereo signal.

Abstract: A sound coding device having a monaural/stereo scalable structure and capable of efficiently coding stereo sound. even when the correlation between the channel signals of a stereo signal is small. In a core layer coding block of this device, a monaural signal generating section generates a monaural signal from first and second-channel sound signal, a monaural signal coding section codes the monaural signal, and a monaural signal decoding section greatest a monaural decoded signal from monaural signal coded data and outputs it to an expansion layer coding block. In the expansion layer coding block, a first-channel prediction signal synthesizing section synthesizes a first-channel prediction signal from the monaural decoded signal and a first-channel prediction filter digitizing parameter and a second-channel prediction signal synthesizing section synthesizes a second-channel prediction signal from the monaural decoded signal and second-channel prediction filter digitizing parameter.

Abstract: A scalable encoding device for realizing scalable encoding by CELP encoding of a stereo sound signal and improving the encoding efficiency. In this device, an adder and a multiplier obtain an average of a first channel signal CH1 and a second channel signal CH2 as a monaural signal M. A CELP encoder for a monaural signal subjects the monaural signal M to CELP encoding, outputs the obtained encoded parameter to outside, and outputs a synthesized monaural signal M? synthesized by using the encoded parameter to a first channel signal encoder. By using the synthesized monaural signal M? and the second channel signal CH2, the first channel signal encoder subjects the first channel signal CH1 to CELP encoding to minimize the sum of the encoding distortion of the first channel signal CH1 and the encoding distortion of the second channel signal CH2.

Abstract: A stereo audio encoding apparatus capable of preventing degradation of the sound quality of a decoded signal, while reducing the encoding bit rate. A spatial information analyzer analyzes spatial information for each of a L channel signal and an R channel signal. A similarity raiser corrects, based on an analysis result of the spatial information analyzer, a difference between the spatial information of the L channel signal and that of the R channel signal, to raise a similarity between the L and R channel signals. A channel signal encoder uses a sound source common to the two channels to encode the L and R channel signals as raised in similarity and output the resultant single encoded information. A spatial information encoder encodes the analysis result of the spatial information analyzer to output the resultant encoded information.

Abstract: The conventional error conceal processing generates a greatly fluctuating irregular sound which is unpleasant to ears and causes a remarkable echo effect and click noise. A notification signal detection unit (301) judges processing for an input frame. In case of an error frame, a sound detection unit (303) makes judgment whether a preceding non-error data frame is a sound signal. If it is a sound frame, a sound copying unit (304) generates a replacing frame. If it is a non-sound frame, a transient signal detection unit (305) judges whether it is an attack signal by the transient signal detection and selects an appropriate area from the preceding non-error frame.

Abstract: There is provided an audio encoding device capable of generating an appropriate monaural signal from a stereo signal while suppressing the lowering of encoding efficiency of the monaural signal. In a monaural signal generation unit (101) of this device, an inter-channel prediction/analysis unit (201) obtains a prediction parameter based on a delay difference and an amplitude ratio between a first channel audio signal and a second channel audio signal; an intermediate prediction parameter generation unit (202) obtains an intermediate parameter of the prediction parameter (called intermediate prediction parameter) so that the monaural signal generated finally is an intermediate signal of the first channel audio signal and the second channel audio signal; and a monaural signal calculation unit (203) calculates a monaural signal by using the intermediate prediction parameter.

Abstract: A speech coder and so forth for preventing deterioration of the quality of a reproduced speech signal while reducing the coding rate. In a speech signal modifying section (101) of the coder, a masking threshold calculating section (114) calculates a masking threshold M(f)) of the spectrum S(f) of an input speech signal, an ACB sound source model spectrum calculating section (117) calculates an adaptive codebook sound source model spectrum SACB(f), an input spectrum shape modifying section (112) refers to both values of the masking threshold M(f) and the adaptive code book sound source model spectrum S?ACB(f) having an LPC spectral envelope and carries out a preprocessing of the spectrum S(f) so that the shape of the spectrum S(f) is modified to match a CELP coding section (102) of the succeeding stage. The CELP coding section (102) carries out CELP coding of the preprocessed speech signal and outputs a coded parameter.

Abstract: Disclosed is a stereo audio encoding device capable of reducing a bit rate. In this device, a stereo audio encoding unit (103) performs LPC analysis on an L channel signal and an R channel signal so as to obtain an L channel LPC coefficient and an R channel LPC coefficient. An LPC coefficient adaptive filter (105) obtains an LPC coefficient adaptive filter parameter to minimize the mean square error between the L channel LPC coefficient and the R channel LPC coefficient. An LPC coefficient reconfiguration unit (106) reconfigures the R channel LPC coefficient by using the L channel LPC coefficient and the LPC coefficient adaptive filter parameter. A route calculation unit (107) calculates a polynomial route indicating the safety of the R channel reconfigured LPC coefficient. A selection unit (108) selects and outputs the LPC coefficient adaptive filter parameter or the R channel LPC coefficient according to the safety of the R channel reconfigured LPC coefficient.

Abstract: Multichannel signal coding equipment is provided for presenting a high quality sound at a low bit rate. In the multichannel signal coding equipment (2), a down mix part (10) generates monaural reference channel signals for N number of channel signals. A coding part (11) codes the generated reference channel signal. A signal analyzing part (12) extracts parameters indicating characteristics of each of the N number of channel signals. An MUX part (13) multiplexes the coded reference channel signal with the extracted parameters.

Abstract: Disclosed is a stereo speech decoding device and others capable of reducing a stereo speech encoding bit rate and suppressing degradation of speech quality. In this device, a section 0 where only an L-channel signal SL(n) exists is identified, a monaural signal of the section 0 transmitted from the stereo speech encoding side is made to be an L-channel signal of section 0 SL(0)(n), and the L-channel signal SL(0)(n) of the section 0 is scale-adjusted so as to predict an R-channel signal SR(1)(n) of a section 1. A contribution of the R-channel signal SR(1)(n) of the predicted section 1 is subtracted from the monaural signal of the section 1 so as to isolate the L-channel signal SL(1)(n) of the section 1. This device continuously repeats the aforementioned scale adjustment and isolation process so as to obtain the L-channel signal SL(n) and the R-channel signal SR(n) of all the sections.

Abstract: Disclosed is a scalable encoding device capable of reducing an encoding rate thereby to reduce a circuit scale while preventing sound quality deterioration of a decoded signal. In this device, an extension layer is coarsely divided into a system for processing a first channel and a system for processing a second channel. A sound source prediction unit (112) for processing the first channel predicts the drive sound source signal of the first channel from the drive sound source signal of a monaural signal, and outputs the predicted drive sound source signal through a multiplier (113) to a CELP encoding unit (114). A sound source prediction unit (115) for processing the second channel predicts the drive sound source signal of the second channel from the drive sound source signal of the monaural signal and the output from the CELP encoding unit (114), and outputs the predicted drive sound source signal through a multiplier (116) to a CELP encoding unit (117).

Abstract: There is disclosed a stereo encoding device capable of accurately encoding a stereo signal at a low bit rate and suppressing delay in audio communication. The device performs monaural encoding in its first layer (110). In a second layer (120), a filtering unit (103) generates an LPC (Linear Predictive Coding) coefficient and generates a left channel drive sound source signal. A time region evaluation unit (104) and a frequency region evaluation unit (105) perform signal evaluation and prediction in both of their regions. A residual encoding unit (106) encodes a residual signal. A bit distribution control unit (107) adaptively distributes bits to the time region evaluation unit (104), the frequency region evaluation unit (105), and the residual encoding unit (106) according to a condition of the audio signal.

Abstract: A stereo audio encoding apparatus capable of preventing degradation of the sound quality of a decoded signal, while reducing the encoding bit rate. In the apparatus, a spatial information analyzing part (101) analyzes the spatial information for each of L and R channel signals. A similarity raising part (102) corrects, based on an analysis result of the spatial information analyzing part (101), a difference between the spatial information of the L channel signal and that of the R channel signal to raise the similarity between the L and R channel signals. A channel signal encoding part (103) uses a sound source common to the two channels to encode the L and R channel signals as raised in similarity and output the resultant single encoded information. A spatial information encoding part (104) encodes the analysis result of the spatial information analyzing part (101) to output the resultant encoded information.

Abstract: A prediction performance between the individual channels of a stereo signal is improved to improve the sound quality of a decoded signal. An LPF (101-1) interrupts the high-range component of an S1, and outputs an S1? (a low-range component). An LPF (101-2) interrupts the high-range component of an S2, and outputs an S2? (a low-range component). A prediction unit (102) predicts the S2? from the S1?, and outputs a prediction parameter composed of a delay time difference (t) and an amplitude ratio (g). A first channel encoding unit (103) encodes the S1. A prediction parameter encoding unit (104) encodes the prediction parameter. The encoded parameters of the encoded parameter of the S1 and the prediction parameter are finally outputted.