摘要:
A coding apparatus capable of coding a spectrum at a low bit rate and with high quality without producing any disturbance in a harmonic structure of the spectrum. In this apparatus, internal state setting section sets an internal state of a filtering section using a first spectrum S1(k). A pitch coefficient setting section outputs a pitch coefficient T by gradually changing it. The filtering section calculates an estimated value S′2(k) of a second spectrum S2(k) based on a pitch coefficient T. A search section calculates the degree of similarity between S2(k) and S′2(k). At this time, pitch coefficient T′ corresponding to the maximum calculated degree of similarity is given to a filter coefficient calculation section. The filter coefficient calculation section determines a filter coefficient βi using this pitch coefficient T′.
摘要:
A transform coder leading to reduction of degradation of perceptual sound quality even if an adequate number of bits is not assigned. Candidates of a correction scale factor stored in a correction scale factor codebook (123) are outputted one by one, and an error signal is generated by subjecting the candidate and scale factors outputted from scale factor computing sections (121, 122) to a predetermined operation. A judging section (126) determines a weight vector given to a weighted error computing section (127) depending on the sign of the error signal. The weighted error computing section (127) computes the square of the error signal, multiplies the square of the error signal by the weight vector given from the judging section (126), and computes a weighted squared error E. A search section (128) determines the candidates of the correction scale factor which minimizes the weighted squared error E by a closed loop processing.
摘要:
A coding device is provided with features in which optimum coding in a higher layer is flexibly carried out based on a coding result of a lower layer and a quality audio signal in limited circumstances is served to users. In this coding device, a basic layer coding unit codes an input signal to generate a basic layer information source code and outputs a linear prediction coefficient (LPC) and a quantum LPC, which are parameters calculated at coding, to an expanded layer control unit. A basic layer decoding unit decodes the basic layer information source code. An adding unit reverses a polarity of a basic layer decoded signal, adds the same to the input signal, and calculates a difference signal. The expanded layer control unit generates expanded layer mode information indicative of a coding mode in an expanded layer based on the LPC and the quantum LPC. An expanded layer coding unit codes the difference signal obtained from the adding unit under control of the expanded layer control unit.
摘要:
There is disclosed an encoding device capable of improving similarity between the high frequency band spectrum of the original signal and a new spectrum to be generated while realizing a low bit rate when encoding a wide-band signal spectrum. The encoding device has sub-band amplitude calculation units (122, 128) for calculating the amplitude of the respective sub-bands for the high frequency band spectrum obtained from the wide-band signal. A search unit (124) and a gain codebook (125) select some sub-bands from a plurality of sub-bands and only the gain of the selected sub-bands is subjected to encoding. An interpolation unit (126) expresses the gain of the sub-band not selected, by mutually interpolating the selected gains.
摘要:
A sound encoding device enabling the amount of delay to be kept small and the distortion between frames to be mitigated. In the sound encoding device, a window multiplication part (211) of a long analysis section (21) multiplies a long analysis frame signal of analysis length M1 by an analysis window, the resultant signal multiplied by the analysis window is outputted to an MDCT section (212), and the MDCT section (212) performs MDCT of the input signal to obtain the transform coefficients of the long analysis frame and outputs it to a transform coefficient encoding section (30). The window multiplication part (221) of a short analysis section (22) multiplies a short analysis frame signal of analysis length M2 (M2
摘要:
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 (305) stores candidates of spectrum residue shape vectors, a spectrum residue gain codebook (307) 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 (306). A multiplier (308) 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 (303). The filtering section (303) performs filtering by using a pitch filter internal state set by a filter state setting section (302), a lag T outputted by a lag setting section (304), and a spectrum residue shape vector which has undergone gain adjustment.
摘要:
Processing for producing encoded output representing information about a pitch period of an input speech signal is performed. The pitch period of a previously entered speech signal is stored in a buffer. A search range-determining portion determines a range in which a current pitch period is analyzed, according to the pitch period of the previously entered speech signal. A presently entered speech signal is applied from a speech input terminal. A pitch analysis portion makes a pitch analysis of candidates for the pitch period contained in the determined search range. Information about the pitch period is delivered from an output terminal and stored in the buffer for subsequent processing. The pitch period of the speech signal can be calculated with a small amount of calculation and represented with a small amount of information.
摘要:
Provided is a speech/audio encoding apparatus with which it is possible to code a significant frequency domain region with high precision, and to enable high audio quality. A speech/audio encoding apparatus codes a linear prediction coefficient. A significant frequency domain region detection unit identifies a frequency domain region which is aurally significant from the linear prediction coefficient. A frequency domain region repositioning unit repositions the significant frequency domain region which is identified by the significant frequency domain region detection unit. A bit allocation computation unit determines a coding bit allocation on the basis of the significant frequency domain region which is repositioned by the frequency domain region repositioning unit.
摘要:
An encoding device, a decoding device, and encoding and decoding methods are provided, wherein when a multi-channel signal is encoded with high efficiency, using an adaptive filter, the number of arithmetic operations to update a filter coefficient of the adaptive filter can be reduced. An update range determination unit determines the range of a filter coefficient order (update order range) of a filter coefficient to be updated, among filter coefficients gk(n) of the adaptive filter, on the basis of a mutual correlation function between an input (L) signal and an input (R) signal. The adaptive filter updates the filter coefficient gk(n) of the filter coefficient order (n) to be updated, using a decoding (L) signal and a decoding error (R) signal.
摘要:
An encoding device enables the amount of processing operations to be significantly reduced while minimizing deterioration in the quality of an output signal. This encoding device (101) encodes an input signal by determining the correlation between a first signal generated by using the input signal and a second signal generated by a predetermined method. An importance assessment unit (202) sets the importance of each of a plurality of processing units obtained by dividing the frames of the input signal. A CELP coder (203) performs sparse processing in which the amplitude value of a predetermined number of samples among multiple samples constituted by the first signal and/or the second signal in each processing unit is set to zero according to the importance that was set for each processing unit, and calculates the correlation between the first signal and the second signal, either of which was subjected to sparse processing.