摘要:
A coding apparatus reduces a circuit scale and the amount of coding processing calculation. A frequency domain conversion section performs a frequency analysis of the signal sampled at a sampling rate Fx with an analysis length of 2·Na and calculates first spectrum S1(k)(0≦k
摘要翻译:编码装置减少电路规模和编码处理计算量。 频域转换部对分析长度为2·Na的采样率Fx进行采样的信号的频率分析,计算出第一频谱S1(k)(0≦̸ k
摘要:
An encoding device includes: a frequency region converter which converts an inputted audio signal into a frequency region; a band selector which selects a quantization object band from a plurality of sub bands obtained by dividing the frequency region; and a shape quantizer which quantizes the shape of the frequency region parameter of the quantization object band. When a prediction encoding presence/absence determiner determines that the number of common sub bands between the quantization object band and the quantization object band selected in the past is not smaller than a predetermined value, a gain quantizer performs prediction encoding on the gain of the frequency region parameter of the quantization object band. When the number of common sub bands is smaller than the predetermined value, the gain quantizer non-predictively encodes the gain of the frequency region parameter of the quantization object band.
摘要:
An encoding device improves the sound quality of a stereo signal while maintaining a low bit rate. The encoding device includes: an LP inverse filter which LP-inverse-filters a left signal L(n) by using an inverse quantization linear prediction coefficient AdM(z) of a monaural signal; a T/F conversion unit which converts the left sound source signal Le(n) from a temporal region to a frequency region; an inverse quantizer which inverse-quantizes encoded information Mqe; spectrum division units which divide a high-frequency component of the sound source signal Mde(f) and the left signal Le(f) into a plurality of bands; and scale factor calculation units which calculate scale factors ai and ssi by using a monaural sound source signal Mdeh,i(f), a left sound source signal Leh,i(f), Mdeh,i(f), and right sound source signal Reh,i(f) of each divided band.
摘要翻译:编码装置在维持低比特率的同时提高了立体声信号的声音质量。 编码装置包括:LP逆滤波器,其通过使用单声道信号的逆量化线性预测系数AdM(z)对左信号L(n)进行反滤波; T / F转换单元,其将左声源信号Le(n)从时间区域转换为频率区域; 反量化编码信息Mqe的逆量化器; 将声源信号Mde(f)和左信号Le(f)的高频分量分割为多个频带的频谱分割单元; 以及通过使用单声道声源信号Mdeh,i(f),左声源信号Leh,i(f),Mdeh,i(f)和右声源信号来计算比例因子ai和ssi的比例因子计算单元 Reh,i(f)。
摘要:
A spectrum coding apparatus capable of performing coding at a low bit rate and with high quality is disclosed. This apparatus is provided with a section that performs the frequency transformation of a first signal and calculates a first spectrum, a section that converts the frequency of a second signal and calculates a second spectrum, a section that estimates the shape of the second spectrum in a band of FL≦k
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 are outputted one by one, and an error signal is generated by subjecting the candidate and scale factors outputted from scale factor computing sections to a predetermined operation. A judging section determines a weight vector given to a weighted error computing section depending on the sign of the error signal. The weighted error computing section computes the square of the error signal, multiplies the square of the error signal by the weight vector given from the judging section and computes a weighted squared error E. A search section determines the candidates of the correction scale factor which minimizes the weighted squared error E by a closed loop processing.
摘要:
Disclosed are a encoding device, a decoding device, and encoding and decoding methods, 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 (170) 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 (130) 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:
Provided is an encoding device which divides an input signal into a low-range component and a high-range component and encodes the components in separate encoding units. The encoding device can improve quality of a decoded signal. The encoding device (101) includes: a band division process unit (201) which subjects an input signal to a band division process so as to obtain a lower intermediate-range component lower than a first frequency and a high-range component higher than the first frequency; a low-range encoding unit (202) which suppresses a portion of the lower intermediate-range component higher than a second frequency so as to obtain a low-range component and encodes the low-range component so as to obtain low-range encoded information; an intermediate-range correction unit (203) corrects the intermediate-range component higher than the second frequency among the suppressed lower intermediate-range component so as to obtain a corrected intermediate-range component; an intermediate high-range encoding unit (204) which encodes the corrected intermediate-range component and the high-range component so as to obtain intermediate high-range encoded information; and a multiplexing unit (205) which multiplexes the low-range encoded information and the intermediate high-range encoded information so as to obtain encoded information.
摘要:
Provided is an encoding device which can suppress quality degradation of a decoded signal in a band extension for estimating a high range from a low range of a decoded signal. The encoding device includes: a first layer encoding unit (202) which encodes the low-range portion of an input signal to generate first encoded information; a first layer decoding unit (203) which decodes the first encoded information to generate a decoded signal; a second layer encoding unit (206) which estimates a high-range portion of the input signal from the decoded signal so as to generate an estimated signal and generate second encoded information to obtain the estimated signal; a peak feature analysis unit (207) which obtains a difference in a wave adjustment structure between the high-range portion of the input signal and the estimated signal or the low-range portion of the input signal; and an encoding information multiplexing unit (208) which integrates the first encoded information, the second encoded information, and the difference in the wave adjustment structure.
摘要:
Provided is an encoding device which improves the sound quality of a stereo signal while maintaining a low bit rate. The encoding device includes: an LP inverse filter (121) which LP-inverse-filterS a left signal L(n) by using an inverse quantization linear prediction coefficient AdM(z) of a monaural signal; a T/F conversion unit (122) which converts the left sound source signal Le(n) from a temporal region to a frequency region; an inverse quantizer (123) which inverse-quantizes encoded information Mqe; spectrum division units (124, 125) which divide a high-frequency component of the sound source signal Mde(f) and the left signal Le(f) into a plurality of bands; and scale factor calculation units (126, 127) which calculate scale factors ai and ssi by using a monaural sound source signal Mdeh,i(f), a left sound source signal Leh,i(f), Mdeh,i(f), and right sound source signal Reh,i(f) of each divided band.
摘要:
A coding apparatus capable of reducing a circuit scale and also reducing the amount of coding processing calculation is disclosed. In this apparatus, frequency domain conversion section (103) performs a frequency analysis of the signal sampled at a sampling rate Fx with an analysis length of 2·Na and calculates first spectrum S1(k) (0≦k
摘要翻译:公开了能够减小电路规模并减少编码处理计算量的编码装置。 在该装置中,频域转换部(103)以分析长度为2N··的采样率Fx进行采样的信号的频率分析,计算出第一频谱S1(k)(0≦̸ k