摘要:
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. The device includes: an autocorrelation coefficient conversion unit (301) for converting the narrow band LSPs of Mn order to an autocorrelation coefficients of Mn order; an inverse lag window unit (302) for applying a window which has an inverse characteristic of a lag window supposed to be applied to the autocorrelation coefficients; an extrapolation unit (303) for extending the order of the autocorrelation coefficients to (Mn+Mi) order by extrapolating the inverse lag windowed autocorrelation coefficients; an up-sample unit (304) for performing an up-sample process in the autocorrelation domain which is equivalent to an up-sample process in a time domain for the autocorrelation coefficients of the (Mn+Mi) order so as to obtain autocorrelation coefficients of Mw order; a lag window unit (305) for applying a lag window to the autocorrelation coefficients of Mw order; and an LSP conversion unit (306) for converting the lag windowed autocorrelation coefficients into LSPs.
摘要:
The total number of entries of an algebraic codebook is decreased by liming a random code vector generated from the algebraic codebook, and entries of a random codebook with a large number of pulses are assigned to a decreased portion. Further, the number of entries of the decreased portion is adaptively switched according to a mode.
摘要:
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.
摘要:
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.
摘要:
A quantizing device for more efficient quantization realized by lessening the computational complexity of quantization of a balance weighting factor. The device includes a power/correlation calculating unit (201), an intermediate value calculating unit (202), a codebook (203), a searching unit (204), and a decoding unit (205). The power/correlation calculating unit (201) determines the value of the correlation between an L signal and an M signal and the value of the correlation between an R signal and the M signal and calculates the power of the M signal. The intermediate value calculating unit (202) determines two intermediate values by using the power of the M signal and the values of the correlations. The codebook (203) holds scalar values. The searching unit (204) selects a coefficient for balance adjustment of the amplitude of the M signal with respect to the L signal from among the scalar values according to the two intermediate values. The decoding unit (205) determines the coefficient for balance adjustment of the M signal with respect to the R signal by using the selected coefficient for balance adjustment of the M signal with respect to the L signal on the basis of the quantitative relation between the amplitudes of the signals of when the M signal is generated by down-mixing the L and R signals.
摘要:
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).
摘要:
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. The inclination correction coefficient is used for adjusting the spectrum inclination of a quantized noise.
摘要:
A CELP type voice encoding device and a CELP type encoding device. Both the CELP type encoding device and the CELP type encoding device have a noise code book that can be searched in two modes in accordance with linear predictive analysis results, a pitch gain and a pitch cycle, all of which are obtained as analysis results of an input voice. Also the number of pulses forming a noise code vector is switched between a first case where a variation in pitch cycle is small througtout continuous sub-frames and in a second case where the variation is not small througtout continuous sub-frames.
摘要:
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. The device includes: an autocorrelation coefficient conversion unit (301) for converting the narrow band LSPs of Mn order to an autocorrelation coefficients of Mn order; an inverse lag window unit (302) for applying a window which has an inverse characteristic of a lag window supposed to be applied to the autocorrelation coefficients; an extrapolation unit (303) for extending the order of the autocorrelation coefficients to (Mn+Mi) order by extrapolating the inverse lag windowed autocorrelation coefficients; an up-sample unit (304) for performing an up-sample process in the autocorrelation domain which is equivalent to an up-sample process in a time domain for the autocorrelation coefficients of the (Mn+Mi) order so as to obtain autocorrelation coefficients of Mw order; a lag window unit (305) for applying a lag window to the autocorrelation coefficients of Mw order; and an LSP conversion unit (306) for converting the lag windowed autocorrelation coefficients into LSPs.
摘要:
Disclosed are a quantizer, encoder, and the methods thereof, wherein the computational load is reduced when the values related to the transform coefficients of the principal component analysis transform are quantized when a principal component analysis transform is applied to code stereo. A quantizer (110) is comprised of a power correlation calculation unit (111) which calculates the power (C11) of the left channel signal, the power (C22) of the right channel signal, and the correlation (C12) between the left channel signal and the right channel signal; an intermediate value calculation unit (112) which calculates the intermediate value (C1122) which is the difference between the power (C11) and the power (C22); a codebook (113) which holds a plurality of sets of the coefficients ?1,n,?2,n related to the transform coefficients of the principal component analysis transform and the code; and a quantizer (114) which calculates the sum of the first multiplication result obtained by multiplying the coefficient ?1,n by the correlation value C12 and the second multiplication result obtained by multiplying the coefficient ?1,n by the intermediate value C1122 as the cost function E, selects the coefficients ?1,n,?2,n where the cost function E becomes the maximum, and fetches the code related to the selected coefficients ?1,n,?2,n as the quantized code.