摘要:
In a formant emphasis method of emphasizing the formant as the spectral peak of an input speech signal and attenuating the spectral valley of the input speech signal, a spectrum emphasis filter performs processing for emphasizing the formant of the input speech signal and attenuating the valley of the input speech signal. A first-order variable characteristic filter whose characteristic adaptively changes in accordance with the characteristic of the input speech signal and a first-order fixed characteristic filter compensate a spectral tilt included in an output signal from the spectrum emphasis filter.
摘要:
The coding apparatus comprises an adaptive codebook storing excitation signals as vectors, a synthesis filter for forming a synthesis signal, referring to the vectors stored in the adaptive codebook, a similarity computation circuit for computing a similarity between the synthesis signal obtained by the synthesis filter and a target signal, and a coding scheme determining circuit for deciding one coding scheme from a plurality of coding schemes respectively having coding bit rates different from each other, on the basis of the similarity obtained by the similarity computation circuit.
摘要:
A learning-type speech encoding apparatus comprises an adaptive code book storing driving signal vectors, a minimum distortion searching circuit for searching the adaptive code book for an optimum driving signal vector on the basis of the input speech signal, a synthesizing filter for synthesizing a speech signal using the optimum driving signal vector retrieved, a buffer for storing the optimum driving signal vector retrieved, a training vector creating section for producing a training vector by segmenting the stored driving signal vector in units of a specified length, and a learning section for learning by constantly updating the driving signal vectors in the code book on the basis of the training vector.
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.
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:
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。
摘要:
A voice encoding device accurately encodes a spectrum shape of a signal having a strong tonality such as a vowel. The device includes: a sub-band divider which divides a first layer error conversion coefficient to be encoded into M sub-bands so as to generate M sub-band conversion coefficients; a shape vector encoder which performs encoding on each of the M sub-band conversion coefficients so as to obtain M shape encoded information and calculates a target gain of each of the M sub-band conversion coefficients; a gain vector former which forms one gain vector by using M target gains; a gain vector encoder which encodes the gain vector so as to obtain gain encoded information; and a multiplexer which multiplexes the shape encoded information with the gain encoded information.
摘要:
Provided is a stereo signal encoding device that enables a lower bitrate without decreasing quality when applying an intermittent transmission technique to a stereo signal. A stereo encoding unit (103) generates first stereo encoded data by encoding the stereo signal when the stereo signal of the current frame is an audio section A stereo DTX encoding unit (104) is a means for encoding the stereo signal when the stereo signal of the current frame is a non-audio section, and generates second stereo encoded data by encoding each of: a monaural signal spectral parameter that is a spectral parameter of a monaural signal generated using the first channel signal and the second channel signal; first channel signal information relating to the first channel signal; and second channel signal information relating to the second channel signal.