摘要:
A scalable decoding apparatus capable of providing decoded audio signals of high quality having less degradation of a high frequency spectrum even when decoding audio signals by generating the high frequency spectrum by use of a low frequency spectrum. In the apparatus, an amplitude adjusting part (1211) uses different adjustment coefficients in accordance with the characteristic of first layer spectrum information to adjust the amplitude of a first layer decoded signal spectrum, and then outputs the amplitude-adjusted first layer decoded signal spectrum to a pseudo-spectrum generating part (1012). Using amplitude-adjusted first layer decoded signal spectrum received from the amplitude adjusting part (1211), the pseudo-spectrum generating part (1012) generates and outputs a pseudo-spectrum of high frequencies to a scaling part (1013). The scaling part (1013) scales the spectrum received from the pseudo-spectrum generating part (1012) and then outputs it to an adder (B).
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 stores candidates of spectrum residue shape vectors, a spectrum residue gain codebook 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. A multiplier 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. The filtering section performs filtering by using a pitch filter internal state set by a filter state setting section, a lag T outputted by a lag setting section, and a spectrum residue shape vector which has undergone gain adjustment.
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.
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:
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:
There is disclosed a scalable encoding device capable of increasing the conversion performance from a narrow-band LSP to a wide-band LSP (prediction accuracy when predicting the wide-band LSP from the narrow-band LSP) and realizing a high-performance band scalable LSP encoding. The device includes a conversion coefficient calculation unit (109) for calculating a conversion coefficient by using a narrow-band quantization LSP which has been outputted from a narrow-band LSP encoding unit (103) and a wide-band quantization LSP which has been outputted from a wide-band LSP encoding unit (107). The wide-band LSP encoding unit (107) multiplies the narrow-band quantization LSP with the conversion coefficient inputted from the conversion coefficient calculation unit (109) so as to convert it into a wide-band LSP. The wide-band LSP is multiplied by a weight coefficient to calculate a prediction wide-band LSP. The wide-band LSP encoding unit (107) encodes an error signal between the obtained prediction wide-band LSP and the wide-band LSP so as to obtain a wide-band quantization LSP.
摘要:
Disclosed is a spectral smoothing device with a structure whereby smoothing is performed after a nonlinear conversion has been performed for a spectrum calculated from an audio signal, and with which the amount of processing calculation is significantly reduced while maintaining excellent audio quality. With this spectral smoothing device, a sub band division unit (102) divides an input spectrum into multiple sub bands; a representative value calculation unit (103) calculates a representative value for each sub band using an arithmetic mean and a geometric mean; with respect to each representative value, a nonlinear conversion unit (104) performs a nonlinear conversion the characteristic of which is further emphasized as the value increases; and a smoothing unit (105) that smoothes the representative value which has undergone the nonlinear conversion for each sub band, at the frequency domain.
摘要:
There is disclosed an audio decoding device capable of improving audio quality of a decoded signal by considering the energy change of a past signal in eracure concealment processing. In this device, an energy change calculation unit (143) calculates an average energy of an audio source signal of one-pitch cycle from the end of the ACB vector outputted from an adaptive codebook (106). Moreover, the energy change calculation unit (143) calculates a ratio of the average energy of the current sub-frame and the sub-frame immediately before and outputs the ratio to an ACB gain generation unit (135). The ACB gain generation unit (135) outputs a conceal processing ACB gain defined by the ACB gain decoded in the past or information on the energy change ratio outputted from the energy change calculation unit (143) to a multiplier (132).
摘要:
The objective of the present invention is to suppress deterioration of call quality caused by transcoding without interrupting a call even if a codec used by one of the terminals during communication is changed. A modification determination unit, in the case of detecting a modification of a codec used by one terminal of two terminals, determines whether or not to constrain the bandwidth of the first codec using a first codec of the other terminal and a second codec after modification by the first-mentioned one of the terminals. A signaling generation unit transmits, to the other terminal, signaling for limiting the bandwidth if the bandwidth is to be limited.
摘要:
A decoding device reduces abrupt changes in the number of channels in a decoded signal when transmission errors occur as a result of lost frames in an encoding/decoding system for multichannel signals. In the device, a demultiplexer receives an encoded monaural signal and an encoded differential signal and detects change over time in the received encoded differential signal. An M signal decoder decodes the encoded monaural signal and obtains a decoded monaural signal. An S signal decoder decodes the encoded differential signal and obtains a decoded differential signal. A smoothing unit performs smoothing on the decoded differential signal by means of a computation involving the decoded differential signal and coefficients corresponding to the change over time detected by the demultiplexer. An L/R signal computation unit computes a decoded stereo signal from the decoded monaural signal and the smoothed decoded differential signal.
摘要翻译:由于多通道信号的编码/解码系统中的丢失帧的结果,当传输错误发生时,解码装置减少解码信号中的信道数量的突然变化。 在该装置中,解复用器接收经编码的单声道信号和经编码的差分信号,并检测接收到的编码差分信号中随时间的变化。 M信号解码器解码编码的单声道信号并获得解码的单声道信号。 S信号解码器对编码的差分信号进行解码并获得解码的差分信号。 平滑单元通过涉及解码的差分信号的计算和对应于由解复用器检测到的随时间变化的系数对解码的差分信号进行平滑处理。 L / R信号计算单元从解码的单声道信号和平滑的解码的差分信号计算解码的立体声信号。