摘要:
With respect to audio signal coding and decoding apparatuses, there is provided a coding apparatus that enables a decoding apparatus to reproduce an audio signal even through it does not use all of data from the coding apparatus, and a decoding apparatus corresponding to the coding apparatus. A quantization unit constituting a coding apparatus includes a first sub-quantization unit comprising sub-quantization units for low-band, intermediate-band, and high-band; a second sub-quantization unit for quantizing quantization errors from the first sub-quantization unit; and a third sub-quantization unit for quantizing quantization errors which have been processed by the first sub-quantization unit and the second sub-quantization unit.
摘要:
The speech detection apparatus comprises: a reference model maker for extracting a plurality of parameters for a speech detection from training data, and for making a reference model based on the parameters; a parameter extractor for extracting the plurality of parameters from each frame of an input audio signal; and a decision device for deciding whether or not the audio signal is speech, by comparing the parameters extracted from the input audio signal with the reference model. The reference model maker makes the reference model for each phoneme. The decision devices includes: a similarity computing unit for comparing the parameters extracted from each frame of the input audio signal with the reference model, and for computing a similarity of the frame with respect to the reference model; a phoneme decision unit for deciding a phoneme of each frame of the input audio signal based on the similarity computed for each phoneme; and a final decision unit for deciding whether or not a specific period of the input audio signal including a plurality of frames is speech, based on the result of the phoneme decision for the plurality of frames.
摘要:
With respect to audio signal coding and decoding apparatuses, there is provided a coding apparatus that enables a decoding apparatus to reproduce an audio signal even through it does not use all of data from the coding apparatus, and a decoding apparatus corresponding to the coding apparatus. A quantization unit constituting a coding apparatus includes a first sub-quantization unit comprising sub-quantization units for low-band, intermediate-band, and high-band; a second sub-quantization unit for quantizing quantization errors from the first sub-quantization unit; and a third sub-quantization unit for quantizing quantization errors which have been processed by the first sub-quantization unit and the second sub-quantization unit.
摘要:
An audio signal compression apparatus for compressively coding an input audio signal comprises a time-to-frequency transformation unit for transforming the input audio signal to a frequency domain signal; a spectrum envelope calculation unit for calculating a spectrum envelope having different resolutions for different frequencies, from the input audio signal, using a weighting function on frequency based on human auditory characteristics; a normalization unit for normalizing the frequency domain signal using the spectrum envelope to obtain a residual signal; a power normalization unit for normalizing the residual signal by the power; an auditory weighting calculation unit for calculating weighting coefficients on frequency, based on the spectrum of the input audio signal and human auditory characteristics; and a multi-stage quantization device having plural stages of vector quantizers connected in series, to which the normalized residual signal is input, and at least one of the vector quantizers quantizing the residual signal using the weighting coefficients. Therefore, a low frequency band, which is auditively important, can be analyzed with a higher frequency resolution as compared with a high frequency band, whereby efficient signal compression utilizing human auditory characteristics is realized.
摘要:
An audio signal compression apparatus for compressively coding an input audio signal comprises a time-to-frequency transformation unit for transforming the input audio signal to a frequency domain signal; a spectrum envelope calculation unit for calculating a spectrum envelope having different resolutions for different frequencies, from the input audio signal, using a weighting function on frequency based on human auditory characteristics; a normalization unit for normalizing the frequency domain signal using the spectrum envelope to obtain a residual signal; a power normalization unit for normalizing the residual signal by the power; an auditory weighting calculation unit for calculating weighting coefficients on frequency, based on the spectrum of the input audio signal and human auditory characteristics; and a multi-stage quantization device having plural stages of vector quantizers connected in series, to which the normalized residual signal is input, and at least one of the vector quantizers quantizing the residual signal using the weighting coefficients. Therefore, a low frequency band, which is auditively important, can be analyzed with a higher frequency resolution as compared with a high frequency band, whereby efficient signal compression utilizing human auditory characteristics is realized.
摘要:
With respect to audio signal coding and decoding apparatuses, there is provided a coding apparatus that enables a decoding apparatus to reproduce an audio signal even through it does not use all of data from the coding apparatus, and a decoding apparatus corresponding to the coding apparatus. A quantization unit constituting a coding apparatus includes a first sub-quantization unit comprising sub-quantization units for low-band, intermediate-band, and high-band; a second sub-quantization unit for quantizing quantization errors from the first sub-quantization unit; and a third sub-quantization unit for quantizing quantization errors which have been processed by the first sub-quantization unit and the second sub-quantization unit.
摘要:
Apparatus for expanding the bandwidth of speech signals such that a narrowband speech signal is input and digitized, the spectral envelope information and residual information are extracted from the digitized signal by linear predictive coding analysis, the spectral envelope information is expanded into wideband information by a spectral envelope converter, the residual information is expanded into wideband information by a residual converter, the converted spectral envelope information and residual information are combined to produce a wideband speech signal, frequency information not contained in the input signal is extracted from the obtained wideband speech signal by a filter, and the resulting signal is added to the original digitized input signal, and the obtained signal is converted into an analog signal as the output signal of the apparatus. The apparatus comprises a linear mapping function codebook used for converting spectral parameters, and a weights calculator and an adder for weighing and summing function outputs.
摘要:
An audio decoding apparatus comprises: a plurality of decoding units; a band replicating unit which processes a decoded signal obtained when a corresponding decoding unit decodes a coded signal, according to a scheme specified by transmitted information; and an information transmitting unit which transmits, to a signal processing unit, information identifying the corresponding decoding unit from among the plurality of decoding units.
摘要:
A hybrid sound signal decoder decodes a bitstream including audio frames encoded by an audio encoding process using a low delay filter bank and speech frames encoded by a speech encoding process using linear prediction coefficients. When a current frame to be decoded is an ith frame which is an initial speech frame after switching from an audio frame to a speech frame, the hybrid sound signal decoder generates sub-frames which are a signal corresponding to an i−1th frame before being encoded, using a sub-frame which is a signal generated using a signal of the i−1th frame before being encoded, the signal of the i−1th frame being obtained by decoding the ith frame.
摘要:
A signal processing device includes a generation unit that generates a second signal from a first signal that is obtained by down mixing two signals; a mixing coefficient determination unit that determines, based on a value L and a value θ, a mixing degree for mixing the first signal and the second signal; and a mixing unit that mixes the first signal and the second signal based on the mixing degree determined by the mixing coefficient determination unit. The generation unit includes a first filter that generates a low frequency band signal in the second signal, from a low frequency band signal in the first signal; and a second filter that generates a high frequency band signal in the second signal, from a high frequency band signal in the first signal. The first filter is a filter unit which, for a complex-number signal, de-correlates an input signal and adds a reverberation component by using a delay unit and an all pass filter, and the processing unit is a filter unit different from the first filter.