Abstract: A transmission system wherein a speech signal is represented by a plurality of prediction parameters updated once per frame. Each frame comprises a plurality of sub-frames in which an excitation signal generated by a fixed codebook and an adaptive codebook is updated. In order to enhance the reconstructed speech quality the prediction coefficients are interpolated at the decoder by an LPC coefficient interpolator to obtain interpolated prediction coefficients for each sub-frame. According to the present invention the interpolation of the prediction coefficients is not based on the prediction coefficients used for transmission, such as reflection coefficients or Log Area Ratios, but on Line Spectral Frequencies. This reduces degradation of speech quality due to interpolation.

Abstract: The concatenative speech synthesizer employs demi-syllable subword units to generate speech. The synthesizer is based on a source-filter model that uses source signals that correspond closely to the human glottal source and that uses filter parameters that correspond closely to the human vocal tract. Concatenation of the demi-syllable units is facilitated by two separate cross fade techniques, one applied in the time domain to the demi-syllable source signal waveforms, and one applied in the frequency domain by interpolating the corresponding filter parameters of the concatenated demi-syllables. The dual cross fade technique results in natural sounding synthesis that avoids time-domain glitches without degrading or smearing characteristic resonances in the filter domain.

Abstract: An information delivery system for delivery of audible information to a plurality of end-users comprising, in accordance with the preferred embodiments of an apparatus of the present invention, a master controller connected to a plurality of remotely-located information sources and to a plurality of remotely-located end-user information devices. The master controller connects to a plurality of distant local controllers via a data transport network, including a public switched telecommunications network and a broadcast data transport network. Each local controller includes a sound synthesizer which connects to an end-user audio device, such as a loudspeaker system, a tape recorder, or earphone.

Abstract: A pitch synchronous sinusoidal synthesizer for multi-band excitation vocoders will produce excitation signals necessary to artificially mimic speech from input data. The input data will contain the pitch frequencies for current and previous synthesizing frame samples, starting phase information for all harmonics within the current synthesizing frame sample, magnitudes for each of the harmonics present within the current synthesizing frame sample, the voiced/unvoiced decisions for each of the harmonics within the current frame sample, and an energy description for the harmonics of the current synthesizing frame sample. The pitch synchronous sinusoidal synthesizer will produce the synthetic speech with a minimum of the distortion caused by the sampling and regeneration of the speech excitation signals. The pitch synchronized sinusoidal synthesizer has a plurality of pitch interpolators.

Abstract: A method is described for short-time Fourier-converting a speech signal and for resynthesizing an output speech signal from the modulus of its short-time Fourier transform and from an initial phase. In particular, after the Fourier converting the signal is subjected to a phase-specifying operation. Subsequently speech duration is affected by systematically maintaining, periodically repeating or periodically suppressing result intervals of the successive Fourier converting and phase affecting. Finally, a resynthesizing operation is executed. Speech pitch can likewise be affected through systematically excising or inserting signal intervals. Finally, the two strategies can be combined, so that ultimately, pitch and duration can be affected independently from each other.

Abstract: This invention relates to converting the characteristics of sounds such as oices, musical tones, natural sounds, and so on, and more specifically to facilitating the conversion operation, and also to sound-label association suitable for the characteristic conversion. Various embodiments of the invention comprise several of the following elements to provide useful results: sound-label data holding means, display control means, conversion means, sound-label dividing means, label-data dividing means, association forming means, data inputting means, and communication means. Other embodiments of the invention may be practiced as processes or articles of manufacture.

Abstract: An encoding unit 2 divides speech signals provided to an input terminal 10 into frames and encodes the divided signals on the frame basis to output encoding parameters such as line spectral pair (LSP) parameters, pitch, voiced(V)/unvoiced (UV) or spectral amplitude A.sub.m. The modified encoding parameter calculating unit 3 interpolates the encoding parameters for calculating modified encoding parameters associated with desired time points. A decoding unit 6 synthesizes sine waves and the noise based upon the modified encoding parameters and outputs the synthesized speech signals at an output terminal 37. Speed control can be achieved easily at an arbitrary rate over a wide range with high sound quality with the phoneme and the pitch remaining unchanged.

Abstract: An information delivery system for delivery of audible information to a plurality of end-users comprising, in accordance with the preferred embodiments of an apparatus of the present invention, a master controller connected to a plurality of remotely-located information sources and to a plurality of remotely-located end-user information devices. The master controller connects to a plurality of distant local controllers via a data transport network, including a public switched telecommunications network and a broadcast data transport network. Each local controller includes a sound synthesizer which connects to an end-user audio device, such as a loudspeaker system, a tape recorder, or earphone.

Abstract: A method (400), device and system (300) provide, in response to linguistic information, efficient generation of a parametric representation of speech using a neural network.

Abstract: A low-complexity method and apparatus for performing waveform interpolation in a low bit-rate WI speech decoder, wherein interpolation between received waveforms is performed with use of spline coefficients generated based thereupon. Specifically, two signals are received from a WI encoder, each comprising a set of frequency domain parameters representing a speech signal segment of a corresponding pitch period. Then, spline coefficients are generated from each of the received signals, wherein each set of spline coefficients comprises a spline representation of a time domain transformation of the corresponding set of frequency domain parameters. Finally, the decoder interpolates between the spline representations to generate interpolated time domain data which is used to synthesize a reconstructed speech signal. In certain embodiments of the present invention, the time scale of at least one of the spline representations is modified to enable the interpolation therebetween.

Abstract: In the disclosed speech decoding device, activation of a postfilter process is halted during unvoiced sections. However, the updating process of the internal states of the postfilter continues even though the postfilter process is not activated during unvoiced sections. At changes between voiced and unvoiced sections, output signals outputted during voiced sections that have been subjected to a postfilter process and output signals outputted during unvoiced sections that have not been subjected to a postfilter process are interpolated and outputted. In one embodiment, a prefilter controller activates a prefilter state updater for unvoiced sections to update the internal state of the filter of a prefilter section based on excited signals to decrease any perception of noncontinuity in an output signal switching between activation and deactivation of the prefilter when changing between voiced and unvoiced sections.

Abstract: A speech synthesis apparatus includes; a memory for storing a plurality of typical waveforms corresponding to a plurality of frames, the typical waveforms each previously obtained by extracting in units of at least one frame from a prediction error signal formed in predetermined units, a voiced speech source generator including an interpolation circuit for performing interpolation between the typical waveforms read out from the memory means to obtain a plurality of interpolation signals each having at least one of an interpolation pitch period and a signal level which changes smoothly between the corresponding frames, a superposition circuit for superposing the interpolation signals obtained by the interpolation circuit to form a voiced speech source signal, an unvoiced speech source generator for generating an unvoiced speech source signal, and a vocal tract filter selectively driven by the voiced speech source signal outputted from the voiced speech source generator and the unvoiced speech source signal fro

Abstract: Apparatus for simultaneously decompressing and interpolating compressed audio data. The compressed audio data is stored in differential log format, meaning that the difference between each two consecutive data points is taken and the log of the difference calculated to form each compressed data point. To efficiently decompress and interpolate the compressed data, advantage is taken of the fact that addition of logs is equivalent to multiplication of linear values. Thus the log of an interpolation factor is added to each compressed data point prior to taking the inverse log of the sum. An integrator block completes the interpolation and decompression of the data.

Abstract: A speech coding system providing reconstructed voiced speech with a smoothly evolving pitch-cycle waveform. A speech signal is represented by isolating and coding prototype waveforms. Each prototype waveform is an exemplary pitch-cycle of voiced speech. A coded prototype waveform is transmitted at regular intervals to a receiver which synthesizes (or reconstructs) an estimate of the original speech segment based on the prototypes. The estimate of the original speech signal is provided by a prototype interpolation process which provides a smooth time-evolution of pitch-cycle waveforms in the reconstructed speech. Illustratively, a frame of original speech is coded by first filtering the frame with a linear predictive filter. Next a pitch-cycle of the filtered original is identified and extracted as a prototype waveform. The prototype waveform is then represented as a set of Fourier series (frequency domain) coefficients.

Abstract: A processing system time compresses a voice message before transmission, and processing system time expands the message after reception. To process the message the processing systems perform at least one of: (a) randomizing the order of a sequence of samples form a silent portion of the message after reception thereof before blending the sequence with a last portion of the expanded message; (b) selecting the sequence of samples from the silent portion of the message after reception thereof, the sequence selected being poorly correlated with the last portion of the expanded message, before blending the sequence with the last portion of the message; and (c) compressing the dynamic range of the message before transmission, by an amount dependent upon the signal-to-noise ratio of the message, and aggressively expanding the dynamic range of the message after reception, by a fixed amount.

Abstract: In a transmission system for a quasi-periodic signal, an encoder (12) transmits to a decoder (28) no more than a single period from a total number of periods of the quasi-periodic signal in encoded form. In this decoder (28) the lacking periods of the quasi-periodic signal are reconstructed by means of interpolation. For obtaining an enhanced quality of the reconstructed signal, a signal segment is transmitted that is representative of two successive periods of the quasi-periodic signal. This signal segment may contain two successive periods, but may also be a signal segment having the length of a single period and being determined by a weighted sum of two successive periods.

Abstract: A speech synthesis apparatus in which spectrum emphasis characteristics can be set easily taking into account the frequency response and psychoacoustic hearing sense and in which the degree of freedom in setting the response is larger. An excitation signal ex(n) is synthesized by a synthesis filter 12 to give a synthesized speech signal which is sent to a spectrum emphasis filter 13. The spectrum emphasis filter 13 spectrum-emphasizes the synthesized speech signal and outputs the resulting spectrum-emphasized signal. The vocal tract parameters from an input terminal 21 are converted by a parameter conversion circuit 23 into linear spectral pair (LSP) frequencies which are interpolated by an LSP interpolation circuit 24 with equal-interval line spectral pair frequencies to produce interpolated LSP frequencies. The transfer function of the spectrum emphasis filter 13 is determined on the basis of the interpolated LSP frequencies.

Abstract: Voice quality is improved in low-rate dynamic bit allocation sub-band coding (DBASC) by inserting synthetic filler signals for non-transmitted signal bands. Large spectral holes may exist in an output voice speech spectrum of DBASC with a low transmission rate. The holes are caused by low energy signal sub-bands that are not transmitted and create noticeable artifacts in the received voice signal. To avoid these artifacts, the spectral holes in the received signal are filled with synthetic signals generated from the received signal. The synthetic filler signals are signals from transmitted energy bands that are scaled to the energy level of the non-transmitted band.

Abstract: A method and apparatus which allows the transmission of the perceptually important features of a speech-coding parameter at a low bit rate. The speech coding parameter may, for example, comprise the signal power of the speech. The parameter is processed on a block by block basis. The parameter value at the block boundaries is transmitted by conventional methods such as, for example, by means of differential quantization. The shape of the reconstructed parameter contour within block boundaries is based on a classification. The classification determines perceptually important features of the parameter contour within a block. The classification can be performed either at the transmitting end of the coder (using, for example, the original parameter contour with high time resolution and possibly other speech parameters as well) or at the receiving end of the coder (using, for example, the transmitted parameter values, and possibly other transmitted speech parameters as well).

Abstract: An efficient method for compressing audio and other sampled data signals without loss, or with a controlled amount of loss, is described. The compression apparatus contains a subset selector, an approximator, an adder, two derivative encoders, a header encoder, and a compressed block formatter. The decompression apparatus contains a compressed block parser, a header decoder, two integration decoders, an approximator, and an adder. The compressor first divides each block of input samples into a first subset and a second subset. The approximator uses the first subset samples to approximate the second subset samples. An error signal is created by subtracting the approximated second subset samples from the actual second subset samples. The first subset samples and error signal are separately encoded by the derivative encoders, which select the signal's derivative that requires the least amount of storage for a block floating point representation.

Abstract: The present invention comprises an interpolation circuit for performing a linear interpolation operation for a voice synthesizer receiving a plurality of frames of voice data wherein each frame is subdivided into a plurality of sub frames with each sub frame being assigned with a predefined interpolation weighting factor. The interpolation circuit comprises an input port for receiving a plurality of sub frame voice data to be interpolated wherein said sub frame data are received in a predefined order. The interpolation circuit further includes a read-only-memory (ROM) for storing a plurality of multiplications of each of the plurality of sub frame voice data to each of the weighting factors and a read-only-memory (ROM) address generator capable of determining a address in ROM utilizing the sub frame voice data received from the input port and the predefined order of the frame and the sub frame thereof when the voice data is being received.

Abstract: The voice synthesis of the invention analyzes a voice signal by orthogonal breakdown on a basis of wavelets with compact support, preferably Daubechies wavelets. The synthesis is carried out on the basis of coefficients which are stored and selected during the analysis, according to the same algorithm as that used for the analysis.

Abstract: In a LPC speech signal decoder, background noise is simulated during periods of silence at the transmitting end based upon a background noise frame containing information about the background noise at the sending end. When the silence persists, the transmitter periodically updates the background noise frame previously send by transmitting an updating background noise frame. When an update background noise frame is received, an interpolation is performed so as to make the simulated background noise sound natural to the listener. The interpolation process includes a step of selecting between interpolation spectrum parameters which are produced by the interpolation process and the updated spectrum parameters which are based solely upon the most recent updated background noise frame.

Abstract: A quantizer and a low bit rate communication system using the quantizer is described. The quantizer includes a 3-bit and 5-bit encoder where the 3-bit encoder provides the encoded gain for a first half of a sampled frame of speech and the second encoder for the second half of the frame of speech. A special 3-bit code is provided when a steady state is determined by comparing the 3-bit code and neighboring 5-bit codes. The decoder in the system when detecting the special code provides an average of the 5-bit codes if the decoded 5-bit code is within 5 dB of the previous 5-bit code.

Abstract: A text-to-voice audio output unit includes a storage section for storing analyzed information pertaining to words, boundaries between articulations, and accents obtained by analyzing an input character list, a voice synthesis rule section for changing a reduction or damping characteristic of a phrase component of a fundamental frequency of an output voice, and a voice synthesizing section for generating a composite tone based on the analyzed information from the storage section. The reduction or damping characteristic, calculated for each phrase component, is overdamped, critically damped, or underdamped and is based on speech rate, syntactic information, number of articulations, and positional information. When a prosodic phrase is short, the reduction or damping characteristic causes a decrease in the fundamental frequency for a meaningfully-delimited portion, and when a prosodic phrase is long, the reduction or damping characteristic is controlled over the entire prosodic phrase.

Abstract: In the first step of a sound morphing process, each sound which forms the basis for the morph is converted into one or more quantitative representations, such as spectrograms. After the representations have been obtained, the temporal axes of the two sounds are matched, so that similar components of the two sounds, such as onsets, harmonic regions and inharmonic regions, are aligned with one another. Other characteristics of the sounds, such as pitch, formant frequencies, or the like, are then matched. Once the energy in each of the sounds has been accounted for and matched to that of the other sound, the two sounds are cross-faded, to produce a representation of a new sound. This representation is then inverted, to generate the morphed sound.

Abstract: A sound recording and reproducing device codes a sound signal by a code excited linear predictive coding means including an adaptive code book created on the basis of past excitation signals, a stochastic code book having a plurality of different stochastic signals and a pulse code book having a plurality of pulse signals. When decoding the sound data coded frame by frame by code excited linear predictive coding, each frame comprising a plurality of sub-frames, by using an adaptive code book created on the basis of a past excitation signal, and a stochastic code book having a plurality of different stochastic signals, a switch controller controls switches such that the coded data in a predetermined sub-frame of each frame is skipped in accordance with the rate of change of reproduction speed to decode the coded data in the remaining sub-frame or sub-frames without decoding the coded data in the skipped sub-frame, and the contents of the adaptive code book are renewed.

Abstract: A sinusoidal model for acoustic waveforms is applied to develop a new analysis/synthesis technique which characterizes a waveform by the amplitudes, frequencies, and phases of component sine waves. These parameters are estimated from a short-time Fourier transform. Rapid changes in the highly-resolved spectral components are tracked using the concept of "birth" and "death" of the underlying sine waves. The component values are interpolated from one frame to the next to yield a representation that is applied to a sine wave generator. The resulting synthetic waveform preserves the general waveform shape and is perceptually indistinguishable from the original. Furthermore, in the presence of noise the perceptual characteristics of the waveform as well as the noise are maintained. The method and devices are particularly useful in speech coding, time-scale modification, frequency scale modification and pitch modification.