摘要:
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.
摘要:
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.
摘要:
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.
摘要:
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.
摘要:
There is disclosed a scalable encoding device capable of preventing sound quality deterioration of a decoded signal, reducing the encoding rate, and reducing the circuit size. The scalable encoding device includes: a first layer encoder (100) 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; and a second layer encoder (150) 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.
摘要:
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.
摘要:
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 (110) of this device, a monaural signal generating section (111) generates a monaural signal from first and second-channel sound signal, a monaural signal coding section (112) codes the monaural signal, and a monaural signal decoding section (113) greatest a monaural decoded signal from monaural signal coded data and outputs it to an expansion layer coding block (120). In the expansion layer coding block (120), a first-channel prediction signal synthesizing section (122) 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 (126) synthesizes a second-channel prediction signal from the monaural decoded signal and second-channel prediction filter digitizing parameter.
摘要:
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.
摘要:
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). The CELP encoding units (114, 117) perform the CELP encoding operations of the individual channels by using the individual predicted drive sound source signals.
摘要:
A voice encoding device capable of generating a modulated proper monaural signal enriched in clearness and understandability, when the monaural signal is to be generated from a stereophonic signal. In this device, a weighting unit (11) weights an L-channel signal (XL) and an R-channel signal (XR) individually, and inputs the weighted L-channel signal (XLW) and R-channel signal (XRW) to a monaural signal generating unit (12). This monaural signal generating unit (12) averages the L-channel signal (XLW) and the R-channel signal (XRW), and creates and inputs a monaural signal (XMW) to a monaural signal encoding unit (13). This monaural signal encoding unit (13) encodes the monaural signal (XMW), and outputs an encoding parameter of the monaural signal (XMW) (or a monaural signal encoding parameter).