Nippon Telegraph and Telephone Corporation

Iwatsu Electric Company Ltd.

Masanobu Abe

Yuki Yoshida

Yasuhiro Sota

Syohei Suzuki

Keiichi Nakayama

Hironori Ohata

Kimihito Tanaka

Tsutomu Irishima

Conference telephone

A speech signal modification and concatenation method is provided, in which spoken messages having different voice characteristics can be concatenated without causing a sense of incompatibility, and it is possible to efficiently perform addition or modification of spoken messages. In the speech signal modification and concatenation method, when two speech signals having different voice characteristics are concatenated, the speech signals are concatenated by modifying a parameter indicating a character of speech signals in a manner such that the parameter is gradually changed from a value indicating a feature of one of the speech signals to a value indicating a feature of the other speech signal over a predetermined period. Accordingly, a time-scaled change of a feature amount of spoken sounds can be performed; thus, even if two speech signals of different speakers are concatenated, it is possible to avoid an abrupt change of voice characteristics in the concatenation section, and thus possible to concatenate speech signals without causing a sense of incompatibility to listeners.

Speech signal modification and concatenation method by gradually changing speech parameters

Data in the same range of the fundamental frequency F.sub.0 as speech segments are used as learning data to prepare a reference codebook CB.sub.M for a spectrum envelope. The same learning data for a higher range than F.sub.0 and the same learning data for a lower range are subject to a linear stretch matching with respect to the learning data for the range F.sub.0. For each vector code in the reference codebook CB.sub.M, the spectrum envelope is clustered to prepare a high range codebook CB.sub.H and a low range codebook CB.sub.L. The spectrum envelope of input speech segments are fuzzy vector quantized (S402) with the reference codebook, and depending on the synthesized F.sub.0, a high, middle or low codebooks is selected. The selected codebook is used to decode the fuzzy vector quantized code, and the decoded output is subject to the inverse FFT. Alternatively, codebooks CM.sub.MH and CB.sub.ML each comprising differential vectors for corresponding code vectors between CB.sub.M and CB.sub.H and between CB.sub.M and CB.sub.L are prepared. The quantized code is decoded using either CB.sub.MH or CB.sub.ML, and the decoded differential vector is stretched in accordance with a difference in the fundamental frequency between the synthesized speech and the original speech for CB.sub.M. The stretched differential vector is added to the code vector which was used for the fuzzy vector quantization.

Method and apparatus for speech synthesis and program recorded medium

In a method and apparatus which use actual speech as auxiliary information and synthesize speech by speech synthesis by rule, prosodic information for a phoneme sequence of each word of a word sequence obtained by an analysis of an input text is set by referring to a word dictionary, and a speech waveform sequence is obtained from the phoneme sequence of each word by referring to a speech waveform dictionary. Additional prosodic information is extracted from input actual speech, and at least one of the set prosodic information or at least one of the extracted prosodic information is selected and used to control the speech waveform sequence to create synthesized speech.

Speech synthesis method utilizing auxiliary information, medium recorded

A wideband speech signal (8 kHz, for example) of high quantity is reconstructed from a narrowband speech signal (300 Hz to 3.4 kHz). The input narrowband speech signal is LPC-analyzed to obtain spectrum information parameters, and the parameters are vector-quantized using a narrowband speech signal codebook. For each code number of the narrowband speech signal codebook, the wideband speech waveform corresponding to the codevector concerned is extracted by one pitch for voiced speech and by one frame for unvoiced speech and prestored in a representative waveform codebook. Representative waveform segments corresponding to the respective output codevector numbers of the quantizer are extracted from the representative waveform codebook. Voiced speech is synthesized by pitch-synchronous overlapping of the extracted representative waveform segments and unvoiced speech is synthesized by randomly using waveforms of one frame length. By this, a wideband speech signal is produced. Then, frequency components below 300 Hz and above 3.4 kHz are extracted from the wideband speech signal and are added to an up-sampled version of the input narrowband speech signal to thereby reconstruct the wideband speech signal.

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Yokohama, Japan

Masanobu Abe