摘要:
An information extraction unit extracts spectral envelope information of L-dimension from each frame of speech data by discrete Fourier transform. The spectral envelope information is represented by L points. A basis storage unit stores N bases (L>N>1). Each basis is differently a frequency band having a maximum as a peak frequency in a spectral domain having L-dimension. A value corresponding to a frequency outside the frequency band along a frequency axis of the spectral domain is zero. Two frequency bands of which two peak frequencies are adjacent along the frequency axis partially overlap. A parameter calculation unit minimizes a distortion between the spectral envelope information and a linear combination of each basis with a coefficient for each of L points of the spectral envelope information by changing the coefficient, and sets the coefficient of each basis from which the distortion is minimized to a spectral envelope parameter of the spectral envelope information.
摘要:
A phoneme sequence corresponding to a target speech is divided into a plurality of segments. A plurality of speech units for each segment is selected from a speech unit memory that stores speech units having at least one frame. The plurality of speech units has a prosodic feature accordant or similar to the target speech. A formant parameter having at least one formant frequency is generated for each frame of the plurality of speech units. A fused formant parameter of each frame is generated from formant parameters of each frame of the plurality of speech units. A fused speech unit of each segment is generated from the fused formant parameter of each frame. A synthesized speech is generated by concatenating the fused speech unit of each segment.
摘要:
A phoneme sequence corresponding to a target speech is divided into a plurality of segments. A plurality of speech units for each segment is selected from a speech unit memory that stores speech units having at least one frame. The plurality of speech units has a prosodic feature accordant or similar to the target speech. A formant parameter having at least one formant frequency is generated for each frame of the plurality of speech units. A fused formant parameter of each frame is generated from formant parameters of each frame of the plurality of speech units. A fused speech unit of each segment is generated from the fused formant parameter of each frame. A synthesized speech is generated by concatenating the fused speech unit of each segment.
摘要:
A speech processing apparatus according to an embodiment of the invention includes a conversion-source-speaker speech-unit database; a voice-conversion-rule-learning-data generating means; and a voice-conversion-rule learning means, with which it makes voice conversion rules. The voice-conversion-rule-learning-data generating means includes a conversion-target-speaker speech-unit extracting means; an attribute-information generating means; a conversion-source-speaker speech-unit database; and a conversion-source-speaker speech-unit selection means. The conversion-source-speaker speech-unit selection means selects conversion-source-speaker speech units corresponding to conversion-target-speaker speech units based on the mismatch between the attribute information of the conversion-target-speaker speech units and that of the conversion-source-speaker speech units, whereby the voice conversion rules are made from the selected pair of the conversion-target-speaker speech units and the conversion-source-speaker speech units.
摘要:
According to one embodiment, a first storage unit stores n band noise signals obtained by applying n band-pass filters to a noise signal. A second storage unit stores n band pulse signals. A parameter input unit inputs a fundamental frequency, n band noise intensities, and a spectrum parameter. A extraction unit extracts for each pitch mark the n band noise signals while shifting. An amplitude control unit changes amplitudes of the extracted band noise signals and band pulse signals in accordance with the band noise intensities. A generation unit generates a mixed sound source signal by adding the n band noise signals and the n band pulse signals. A generation unit generates the mixed sound source signal generated based on the pitch mark. A vocal tract filter unit generates a speech waveform by applying a vocal tract filter using the spectrum parameter to the generated mixed sound source signal.
摘要:
A voice conversion apparatus stores, in a parameter memory, target speech spectral parameters of target speech, stores, in a voice conversion rule memory, a voice conversion rule for converting voice quality of source speech into voice quality of the target speech, extracts, from an input source speech, a source speech spectral parameter of the input source speech, converts extracted source speech spectral parameter into a first conversion spectral parameter by using the voice conversion rule, selects target speech spectral parameter similar to the first conversion spectral parameter from the parameter memory, generates an aperiodic component spectral parameter representing from selected target speech spectral parameter, mixes a periodic component spectral parameter included in the first conversion spectral parameter with the aperiodic component spectral parameter, to obtain a second conversion spectral parameter, and generates a speech waveform from the second conversion spectral parameter.
摘要:
A voice conversion apparatus stores, in a parameter memory, target speech spectral parameters of target speech, stores, in a voice conversion rule memory, a voice conversion rule for converting voice quality of source speech into voice quality of the target speech, extracts, from an input source speech, a source speech spectral parameter of the input source speech, converts extracted source speech spectral parameter into a first conversion spectral parameter by using the voice conversion rule, selects target speech spectral parameter similar to the first conversion spectral parameter from the parameter memory, generates an aperiodic component spectral parameter representing from selected target speech spectral parameter, mixes a periodic component spectral parameter included in the first conversion spectral parameter with the aperiodic component spectral parameter, to obtain a second conversion spectral parameter, and generates a speech waveform from the second conversion spectral parameter.
摘要:
A speech processing apparatus according to an embodiment of the invention includes a conversion-source-speaker speech-unit database; a voice-conversion-rule-learning-data generating means; and a voice-conversion-rule learning means, with which it makes voice conversion rules. The voice-conversion-rule-learning-data generating means includes a conversion-target-speaker speech-unit extracting means; an attribute-information generating means; a conversion-source-speaker speech-unit database; and a conversion-source-speaker speech-unit selection means. The conversion-source-speaker speech-unit selection means selects conversion-source-speaker speech units corresponding to conversion-target-speaker speech units based on the mismatch between the attribute information of the conversion-target-speaker speech units and that of the conversion-source-speaker speech units, whereby the voice conversion rules are made from the selected pair of the conversion-target-speaker speech units and the conversion-source-speaker speech units.
摘要:
According to one embodiment, an apparatus for supporting reading of a document includes a model storage unit, a document acquisition unit, a feature information extraction, and an utterance style estimation unit. The model storage unit is configured to store a model which has trained a correspondence relationship between first feature information and an utterance style. The first feature information is extracted from a plurality of sentences in a training document. The document acquisition unit is configured to acquire a document to be read. The feature information extraction unit is configured to extract second feature information from each sentence in the document to be read. The utterance style estimation unit is configured to compare the second feature information of a plurality of sentences in the document to be read with the model, and to estimate an utterance style of the each sentence of the document to be read.
摘要:
A speech synthesis system in a preferred embodiment includes a speech unit storage section, a phonetic environment storage section, a phonetic sequence/prosodic information input section, a plural-speech-unit selection section, a fused-speech-unit sequence generation section, and a fused-speech-unit modification/concatenation section. By fusing a plurality of selected speech units in the fused speech unit sequence generation section, a fused speech unit is generated. In the fused speech unit sequence generation section, the average power information is calculated for a plurality of selected M speech units, N speech units are fused together, and the power information of the fused speech unit is so corrected as to be equalized with the average power information of the M speech units.