Abstract: In an image forming device management system and method, a failure in any of a plurality of image forming devices on a first LAN is detected by a management apparatus. A failure message is transmitted from the management apparatus to a center system via a public switched telephone network when the failure is detected, the failure message including a failure code provided to identify the failure. The failure message, transmitted by the management apparatus, is received at the center system. A database of the center system is accessed by using the failure code of the received failure message so as to produce results of the accessing at the center system. A service department ID is extracted from the database based on the accessing results. A service request message is transmitted from the center system to a service department indicated by the service department ID.

Abstract: The invention relates to a method for encoding voice signals, especially so-called voice onset sections. By establishing the first amplification factor, the data quantity for representing the whole of the first or adaptive amplification factor and adaptive code book entry is reduced, whereby other parameters which occur during the voice encoding can be represented in a more precise manner. The invention also relates to a method for transmitting voice signals which are encoded in such a way.

Abstract: A normalizer (100, 300) of the accent of accented speech modifies (210, 410) the characteristics of input signals that represent the speech spoken in an individual voice with an accent to form output signals that represent the speech spoken in the same voice but with less or no accent.

Abstract: In order to enhance the quality of a communication signal comprising speech signal components due to speech and noise signal components due to noise, a filter divides the communication signal into a plurality of frequency band signals representing the speech signal components and the noise signal components in a plurality of frequency bands. A calculator generates a plurality of weighting signals having weighting values corresponding to the frequency band signals. The weighting values represent at least approximations of the normalized powers of the noise signal components in the frequency band signals. The frequency band signals are altered in response to the weighting signals to generate weighted frequency band signals which are combined to generate a communication signal with enhanced quality.

Abstract: A system and method for interfacing with a component located in a network environment is provided. A user in a network environment can connect to a device on the network and automatically learn at least one detail regarding the device software image details. Examples of the software image details may include software version number, size in bytes, device model/family name, software filename, interface hardware details, and supported software feature set such as Internet Protocol (IP), Internet Packet Exchange (IPX), and AppleTalk. The invention provides capability of determining whether the software image version or feature set is supported by a product which the user desires to use, suggesting an upgrade to an appropriate software version or feature set to accommodate the product if the current version is not supported by the product, and automatically upgrading the software if the user approves of such action.

Abstract: A method of determining the topology of a network comprising: transmitting a signal comprised of a sequence of bursts of packets formed of orthogonal signals, monitoring devices in the network including the destination device for reception of the signal, and defining a sequence of devices within the network by sensing a sequence of reception of the signal in the devices from the source device toward the destination device.

Abstract: An information processing apparatus includes a feature parameter detector for detecting feature parameters based on a plurality of input data, a normalizer for normalizing the feature parameters detected by the feature parameter detector while maintaining their feature components, and an integration unit for integrating the feature parameters normalized by the normalizer. In the information processing apparatus, feature parameters from a plurality of input data are normalized based on learning normalization coefficients, and distances from each of the normalized feature parameters and to a normal parameter are calculated. Based on the calculated distances, time-series normalization coefficients for performing speech recognition are determined for the feature parameters. Therefore, optimal normalization coefficients for recognizing the feature parameters at each point of time can be obtained.

Abstract: An indication of the loudness of an audio signal containing speech and other types of audio material is obtained by classifying segments of audio information as either speech or non-speech. The loudness of the speech segments is estimated and this estimate is used to derive the indication of loudness. The indication of loudness may be used to control audio signal levels so that variations in loudness of speech between different programs is reduced. A preferred method for classifying speech segments is described.

Abstract: In a voice pitch normalization device equipped in a voice recognition device VRAp for recognizing an incoming command voice Sva uttered by any speaker, and used to normalize the incoming command voice to be in an optimal pitch for voice recognition, a target voice generator produces a target voice signal by changing the incoming command voice Svd on the basis of a predetermined degree. A probability calculator calculates a probability indicating a degree of coincidence among the target voice signal and a plurality of words in sample data. A voice pitch changer repeatedly changes the target voice signal in voice pitch until a maximum probability becomes a predetermined probability or greater.

Abstract: A received voice processing apparatus is provided, in which the received voice processing apparatus includes: a target spectrum calculation part for calculating, for each frequency band, a target spectrum on the basis of a compression ratio for a voice spectrum; a gain calculation part for calculating a gain value for amplifying the voice spectrum to the target spectrum; a filter coefficient calculation part for calculating a filter coefficient from the gain value; and a filer part for processing a received voice signal by using the filter coefficient.

Abstract: A system and method facilitating training machine learning systems utilizing sequential conditional generalized iterative scaling is provided. The invention includes an expected value update component that modifies an expected value based, at least in part, upon a feature function of an input vector and an output value, a sum of lambda variable and a normalization variable. The invention further includes an error calculator that calculates an error based, at least in part, upon the expected value and an observed value. The invention also includes a parameter update component that modifies a trainable parameter based, at least in part, upon the error. A variable update component that updates at least one of the sum of lambda variable and the normalization variable based, at least in part, upon the error is also provided.

Abstract: Methods and systems for filtering synthesized or reconstructed speech are implemented. A filter based on a set of linear predictive coding (LPC) coefficients is constructed by transforming the LPC coefficients to the pseudo-cepstrum, a domain existing between LPC domain and the line spectral frequency (LSF) domain. The resulting filter can emphasize spectral frequencies associated with various formants, or spectral peaks, of an inverse transfer function relating to the LPC coefficients, and can de-emphasize spectral frequencies associated with various spectral minima, or spectral valleys, of the inverse transfer function relating to the LPC coefficients.

Abstract: An input signal is time-frequency transformed, then the frequency-domain coefficients are divided into coefficient segments of about 100 Hz width to generate a sequence of coefficient segments, and the sequence of coefficient segments is split into subbands each consisting of plural coefficient segments. A threshold value is determined based on the intensity of each coefficient segment in each subband. The intensity of each coefficient segment is compared with the threshold value, and the coefficient segments are classified into low- and high-intensity groups. The coefficient segments are quantized for each group, or they are flattened respectively and then quantized through recombination.

Abstract: An adaptive noise suppression system includes an input A/D converter, an analyzer, a filter, and a output D/A converter. The analyzer includes both feed-forward and feedback signal paths that allow it to compute a filtering coefficient, which is input to the filter. In these paths, feed-forward signal are processed by a signal to noise ratio estimator, a normalized coherence estimator, and a coherence mask. Also, feedback signals are processed by a auditory mask estimator. These two signal paths are coupled together via a noise suppression filter estimator. A method according to the present invention includes active signal processing to preserve speech-like signals and suppress incoherent noise signals. After a signal is processed in the feed-forward and feedback paths, the noise suppression filter estimator then outputs a filtering coefficient signal to the filter for filtering the noise out of the speech and noise digital signal.

Abstract: Encoding an acoustic source signal such that a signal {circumflex over (z)} reconstructed from the encoded information has a perceptually high sound quality. The acoustic source signal is encoded into at least one basic coded signal that represents perceptually significant characteristics of the acoustic signal. The encoder can include at least one spectral smoothing unit which receives at least one of the signal components on which the basic coded signal is based and generates in response thereto a corresponding smoothed signal component. At least one enhanced coded signal is then produced from the corresponding smoothed signal. component for transmission. A receiver receives at least one estimate {circumflex over (P)}E of the transmitted signal(s), and a spectral smoothing unit in the receiver produces, on basis of a primary spectrum Ŷ decoded from the at least one received estimate {circumflex over (P)}E, a smoothed primary decoded spectrum ŶE.

Abstract: A voice signal processing system with multiple parallel control paths, each of which address different problems, such as the high peak-to-RMS signal ratios characteristic of speech, wide variations in RMS speech levels, and high background noise levels. Different families of input-output control curves are used simultaneously to achieve efficient peak limiting and dynamic range compression as well as low-level dynamic expansion to prevent excessive amplification of background noise. In addition, a delay in the audio path relative to the control path makes it possible to employ an effective look-ahead in the control path, with FIR filtering smoothing-matched to the look-ahead. Digital domain peak interpolators are used for estimating the peaks of the input signal in the continuous time domain.

Abstract: A method and an apparatus for processing a sound signal are provided, which process an input sound signal including degraded sound such as quantization noise so as to make the degraded sound subjectively unperceptible. A transformation strength controller calculates a spectrum of a decoded speech after perceptually weighting the decoded speech as the input sound signal, and calculates transformation strength based on the extent of the amplitude and the continuity of the spectrum. A signal transformer obtains a spectrum of the decoded speech, smoothes the amplitude and disturbs the phase based on the transformation strength, and the obtained signal is returned back to a signal region as a transformed decoded speech. A signal evaluator obtains background noise likeness by analyzing the decoded speech and the obtained value is made to be an addition control value.

Abstract: A method for upsampling a digital audio signal is described. The method includes receiving a first digital audio signal including samples and having a first sampling rate. The method also includes outputting at least one sample from the first digital audio signal as part of a second digital audio signal, the second digital audio signal having a desired second sampling rate, the second sampling rate being higher than the first sampling rate. The method also includes incrementing a counter for each sample from the first digital audio signal that is output as part of the second digital audio signal. The method also includes, when the counter exceeds a threshold number, inserting at least one synthetic sample as part of the second digital audio signal. The method also includes repeating the outputting, incrementing, and inserting until all the samples in the first digital audio signal have been output.

Abstract: An apparatus for normalizing speech feature elements in a signal derived from a spoken utterance. The apparatus includes an input, a processing unit and an output. The input receives speech feature elements transmitted over a channel that induces a channel specific distortion in the speech feature elements. The processing unit is coupled to the input and is operative for altering the speech feature elements to generate normalized speech feature elements. The normalized speech feature elements simulate a transmission of the speech feature elements over a reference channel that is other than the channel over which the transmission actually takes place. The apparatus can be used as a speech recognition pre-processing unit to reduce channel related variability in the signal on which speech recognition is to be performed.

Abstract: A method and system that simplify the management of enterprise network devices and information through the use of scripts and a scripting object model. An API is provided that transforms scripts passed from a scripting engine into the existing “low-level” COM syntax required for accessing system management object information. A scripting engine interprets a script and works with the API to translate script instructions into the COM method calls needed to directly access properties and methods of CIMOM objects from a script. Other aspects related to scripting are handled, including collections, events, monikers and security. Collections enable a set of objects to be serviced iteratively, for example, to manipulate or retrieve properties for a set of resources in simple loop. Events enable queries to be made asynchronously, such that calls return immediately and complete via event notifications.

Abstract: An apparatus and a method for encoding an input signal on the time base through orthogonal transform involves removing the correlation of signal waveform based on parameters obtained by linear predictive coding (LPC) analysis and pitch analysis of the input signal on the time base prior to the orthogonal transform. A normalization circuit section removes the correlation of the signal waveform and takes out the residue by an LPC inverse filter and pitch inverse filter and sends the residue to an orthogonal transform circuit section. The LPC parameters and the pitch parameters are sent to a bit allocation calculation circuit. A coefficient quantization section quantizes the coefficients from the orthogonal transform circuit section according to the number of allocated bits from the bit allocation calculation section.

Abstract: An adaptive noise suppression system includes an input A/D converter, an analyzer, a filter, and a output D/A converter. The analyzer includes both feed-forward and feedback signal paths that allow it to compute a filtering coefficient, which is input to the filter. In these paths, feed-forward signal are processed by a signal to noise ratio estimator, a normalized coherence estimator, and a coherence mask. Also, feedback signals are processed by a auditory mask estimator. These two signal paths are coupled together via a noise suppression filter estimator. A method according to the present invention includes active signal processing to preserve speech-like signals and suppress incoherent noise signals. After a signal is processed in the feed-forward and feedback paths, the noise suppression filter estimator then outputs a filtering coefficient signal to the filter for filtering the noise out of the speech and noise digital signal.

Abstract: Methods and apparatus, in the context of speech recognition, for compensating in the cepstral domain for the effect of an interfering signal by using a reference signal.

Abstract: A method for tracking the phase of a quasi-periodic signal includes the steps of estimating the phase of the signal for frames during which the signal is periodic, monitoring the performance of the estimated phase with a closed-loop performance measure, and measuring the phase of the signal for frames during which the signal is periodic and performance of the estimated phase falls below a predefined threshold level. In estimating the phase, the initial phase value is set equal to the estimated final phase value of the previous frame if the previous frame was periodic. The initial phase value is set equal to a measured phase value of the previous frame if the previous frame was nonperiodic, or if the previous frame was periodic and performance of the estimated phase for the previous frame fell below the predefined threshold level. For frames during which the signal is nonperiodic, the phase of the signal is measured.

Abstract: According to a broad aspect of a preferred embodiment of the invention, a Customer Quality of Service Management system is provided. First, a hybrid network event is received which may include customer inquiries, required reports, completion notification, quality of service terms, service level agreement terms, service problem data, quality data, network performance data, and/or network configuration data. Next, the system determines customer reports to be generated and generates the customer reports accordingly based on the event received.

Abstract: As the application of a variance normalization (VN) to a speech signal (S) may be advantageous as well as disadvantageous with respect to the recognition rate in a speech recognizing process in dependence of the degree of the signal disturbance it is suggested to calculate a degree (ND) of variance normalization strength in dependence of the noise level of the signal, thereby skipping the step of variance normalization in the case of an undisturbed or clean signal.

Abstract: Perceptually relevant non-speech information can be preserved during encoding of an audio signal by determining whether the audio signal includes such information. If so, a speech/noise classification of the audio signal is overriden to prevent misclassification of the audio signal as noise.

Abstract: The audio source is spectrally shaped by filtering in the time domain to approximate or emulate a standardized or target microphone input channel. The background level is adjusted by adding noise to the time domain signal prior to the onset of speech to set a predetermined background noise level based on a predetermined target. The audio source is then monitored in real time and the signal-to-noise ratio is adjusted by adding noise to the time domain signal, in real time, to maintain a signal-to-noise ratio based on a predetermined target value. The normalized audio signal may be applied to both training speech and test speech. The resultant normalization minimizes the mismatch between training and testing and also improves other speech processing functions, such as speech endpoint detection.

Abstract: A method for upsampling a digital audio signal is described. The method includes receiving a first digital audio signal including samples and having a first sampling rate. The method also includes outputting at least one sample from the first digital audio signal as part of a second digital audio signal, the second digital audio signal having a desired second sampling rate, the second sampling rate being higher than the first sampling rate. The method also includes incrementing a counter for each sample from the first digital audio signal that is output as part of the second digital audio signal. The method also includes, when the counter exceeds a threshold number, inserting at least one synthetic sample as part of the second digital audio signal. The method also includes repeating the outputting, incrementing, and inserting until all the samples in the first digital audio signal have been output.

Abstract: A code transform apparatus performs fast data transform by enabling intercode data transform. The code transform apparatus includes a code-string decomposing unit for inputting a first code string obtained by coding a spectral signal which has been transformed with a first block length after a time-series information signal had been divided into a first band group. A signal-component decoding unit decodes the input first code string into a spectral signal. A spectral-signal transform unit transforms the decoded spectral signal into a spectral signal which is transformed with a second block length after being divided into a second band group. A signal-component coding unit and a code-string generating unit code the transformed spectral signal into a second code string.

Abstract: A system and method are disclosed for partitioning a real-value windowed attribute into ranges, wherein the values within each range generally correspond to a particular class of results associated with runs of a process. The system and method determines a low range having attribute values generally corresponding to a first class, a middle range having attribute values generally corresponding to a second class, and an upper range having attribute values generally corresponding to the first class. The system and method may be used in a system that produces an induction tree useful in developing an indication of a cause of a particular result of a process from values associated with at least one real-valued, windowed attribute that arises during the runs of the process.

Abstract: A method for performing noise suppression and channel equalization of a noisy voice signal comprising the steps of sampling the noisy voice signal at a predetermined sampling rate fs; segmenting the sampled voice signal into a plurality of frames having a predetermined number of samples per frame, over a predetermined temporal window; generating an N-point spectral sample representation of each of the sample signal frames; determining the magnitude of each of the N-point spectral samples and generating a histogram of the energy associated with each of the N-point spectral samples at a particular frequency; detecting a peak amplitude of the histogram which corresponds to a noise threshold Nf associated with the particular frequency; determining a channel frequency response Cf associated with the particular frequency by determining a geometric mean over all the spectral samples having magnitude exceeding the noise threshold Nf; subtracting from each of the magnitudes of the N point spectral samples the noise th

Abstract: A method of separating high-level prosodic behavior from purely articulatory constraints so that timing information can be extracted from human speech is presented. The extracted timing information is used to construct duration templates that are employed for speech synthesis. The duration templates are constructed so that words exhibiting the same stress pattern will be assigned the same duration template. Initially, the words of input text segmented into phonemes and syllables, and the associated stress pattern is assigned. The stress assigned words are then assigned grouping features by a text grouping module. A phoneme cluster module groups the phonemes into phoneme pairs and single phonemes. A static duration associated with each phoneme pair and single phoneme is retrieved from a global static table. A normalization module generates a normalized syllable duration value based upon the retrieved static durations associated with the phonemes that comprise the syllable.

Abstract: Either or both the calling and called parties to a telephone call carried by a telecommunications network may invoke normalization of their speech to enhance intelligibility. In response to such a request, a speech normalization platform determines the manner in which the speech should be normalized. The platform does so by selecting from among a set of rules that specify the manner in which the speech should be modified, the rule that most closely corresponds with a set of parameters indicative of the party's speech. Having selected the rule, the platform then implements the rule to modify the party's speech to enhance its intelligibility.

Abstract: An automobile radio telephone apparatus capable of detecting a drop of power source voltage to prevent a telephone channel from being occupied wastefully and to thereby enhance efficient use of channels with regard to the operation of an automobile radio telephone system. When the power source voltage is lower than a predetermined voltage, the apparatus informs the user of such an occurrence. On the lapse of a predetermined period of time and if a conversation is under way, the apparatus warns the user that it will execute a forcible conversation ending procedure and then executes it.

Abstract: A method for encoding speech wherein an input speech signal is separated by a component separator into a first component mainly constituted by speech and a second component mainly constituted by a background noise at each predetermined unit of time, a bit allocation selector selects bit allocation for each component based on the first and second components from among a plurality of predetermined candidates for bit allocation, a speech encoder and a noise encoder encode the first and second components from the component separator based on the bit allocation according to predetermined different methods for encoding, and a multiplexer multiplexes encoded data of the first and second components and information on the bit allocation and outputs them as transmitted encoded data.

Abstract: A radio communication system includes a voice recognition system (221) for converting (400) a caller's voice message to a textual speech message. The textual speech message is then transmitted to an intended selective call radio (122). To perform these functions, the radio communication system includes a caller interface circuit (218), a transmitter (116), and a processor (222). To perform voice-to-text conversion, the processor is adapted to cause the caller interface circuit to sample a voice signal generated by the caller during a plurality of frame intervals, and to apply a Fourier transform thereto, thereby generating spectral data. The spectral data is subdivided into a plurality of bands. The spectral envelope of the spectral data is then filtered out to generate filtered spectral data. A Fourier transform is applied thereto to generate an autocorrelation function for each band.

Abstract: A signal processing apparatus and method decodes codes generated by encoding decomposed frequency components, and combines frequency components obtained in the decoding to form a waveform signal. The apparatus and method further monitors a processing condition in the decoding, and controls a processed band of the waveform signal, formed in the combining, in accordance with the monitoring result obtained in the monitoring.

Abstract: An audio decoder which includes a coefficient memory and an arithmetic logic unit (ALU) can implement an efficient method for calculating a gain value specified by a range control field. In one embodiment, the audio decoder comprises coefficient memory, an ALU, frame control logic, and ALU control logic. The frame control logic extracts a range control field value from an audio packet header and provides it to the ALU control logic. The ALU control logic takes the binary representation of the range control field value and uses it to provide a sequence of addresses to the coefficient memory. In response to the sequence of addresses, the coefficient memory provides a sequence of pre-calculated factors to the ALU. The ALU control logic further directs the ALU to determine the product of the pre-calculated factors in the sequence. As a final step in finding the gain value, the ALU control logic may provide a shift instruction to the ALU.

Abstract: A method to detect a normalized data field of all zeros or all ones includes receiving a control field and a data field, dividing the data field into segments, and performing detections on each segment. Each segment undergoes all zeros detection, all ones detection, modified zeros detection, and modified ones detection. The modified zeros detection and modified ones detection are both done based on the control field. Each detection for each segment generates a response. Then, a pair of the four responses, or a clear responses signal, is selected for each of the segments based on the control field. From the selected responses, the method determines if the normalized data field is all zeros or all ones.

Abstract: A communication apparatus of the present invention has a hands-free communication capability and is applicable to, but not limited to, a motor vehicle. Only if the apparatus is connected to, e.g., an on-board cigar-lighter, hands-free communication can be held. This eliminates the need for a hands-free unit, an outside microphone and other extra parts conventionally mounted on the vehicle to implement a hands-free feature.

Abstract: A multi-rate speech codec supports a plurality of encoding bit rate modes by adaptively selecting encoding bit rate modes to match communication channel restrictions. In higher bit rate encoding modes, an accurate representation of speech through CELP (code excited linear prediction) and other associated modeling parameters are generated for higher quality decoding and reproduction. The encoder applies adaptive gain reduction to optimize selection of appropriate gain contributions from the adaptive and fixed codebooks. Specifically, the encoder uses a first target signal to identify a contribution (a best code vector and a gain) from the adaptive codebook. Thereafter, a contribution from the fixed codebook is selected. The gain associated with the adaptive codebook contribution is then reduced by a factor, and the gain contribution from the fixed codebook is searched a second time, permitting fine tuning of the overall contribution.

Abstract: A speech processor for processing speech signals in a manner that minimizes the peak to average ratio of a vocal tract response waveform of the speech signal with minimal loss of intelligibility of speech reproduced from the processed waveform. This is accomplished, in general terms, by providing a speech processor for providing an approximately constant peak level within periods of a vocal tract response waveform. The speech processor may include a feedback-controlled signal compressor multiplier and an input signal delay means. The attack, hang and decay parameters of the speech processor are determined in accordance with typical vocal tract response characteristics to optimize the balance between compression of the vocal tract response waveform and introduction of harmonics into the resulting signal.

Abstract: A speech encoding apparatus capable of averting the deterioration of synthesis speech quality in encoding the input speech and of generating a high-quality synthesis output speech through small quantities of computation. The apparatus includes a target speech generation part for generating from the input speech a target speech vector of a vector length corresponding to a delay parameter; an adaptive codebook for generating from previously generated excitation signals an adaptive vector of the vector length corresponding to the delay parameter; an adaptive code search part for evaluating the distortion of a synthesis vector obtained from the adaptive vector with respect to the target speech vector so as to search for the adaptive vector conducive to the least distortion; and a frame code generation part for generating an excitation signal of a frame length from the adaptive vector conducive to the least distortion.

Abstract: Quantizing optimizes compression encoding for transmission of audio data. A working set of most frequent sound levels in a sequence of audio samples is determined from a histogram. Sound levels from the working set are then substituted for original sound levels in the subject audio data. This filters the audio data by increasing redundancy and decreasing granularity/resolution.

Abstract: For coding human speech for subsequent audio reproduction thereof, a plurality of speech segments is derived from speech received, and systematically stored in a data base for later concatenated readout. After the deriving, respective speech segments are fragmented into temporally consecutive source frames, similar source frames as governed by a predetermined similarity measure thereamongst that is based on an underlying parameter set are joined, and joined source frames are collectively mapped onto a single storage frame. Respective segments are stored as containing sequenced referrals to storage frames for therefrom reconstituting the segment in question.

Abstract: An audio coder/decoder ("codec") that is suitable for real-time applications due to reduced computational complexity, and a novel adaptive sparse vector quantization (ASVQ) scheme and algorithms for general purpose data quantization. The codec provides low bit-rate compression for music and speech, while being applicable to higher bit-rate audio compression. The codec includes an in-path implementation of psychoacoustic spectral masking, and frequency domain quantization using the novel ASVQ scheme and algorithms specific to audio compression. More particularly, the inventive audio codec employs frequency domain quantization with critically sampled subband filter banks to maintain time domain continuity across frame boundaries. The input audio signal is transformed into the frequency domain in which in-path spectral masking can be directly applied. This in-path spectral masking usually results in sparse vectors.

Abstract: The object of the invention is to provide a coding/decoding method in which degradation of sound quality perceptible by the listener does not occur at an low bit rate. A shift number calculation section of a decoding device divides a frequency domain into at least two sub-bands, and approximates each of normalized transform coefficients in the sub-band whose allocated bit value is less than a predetermined threshold using a quantized value of the transform coefficient in a predetermined sub-band other than the sub-band so as to obtain information concerning the approximation, and a multiplexer multiplexes the information and another signal and transmits them. A de-multiplexer of a decoding device separates the code of information concerning the approximation, and a shift number restore section restores the information based thereon.

Abstract: Portions of recorded speech waveform (e.g., corresponding to phonemes) are combined to synthesize words. In order to provide a smoother delivery, each voiced portion of a waveform portion has its amplitude adjusted to a predetermined reference level. The scaling factor used is varied gradually over a transition region between such portions and between voiced and unvoiced portions.

Abstract: A signal encoding method and apparatus for encoding input digital signals by so-called high efficiency encoding, and a recording medium having the encoded signals. An attack portion and a release portion of audio signals are detected and a gain control function is selected at least for waveform elements (waveform signals) of a signal portion ahead of the attack portion and waveform elements of the release portion from among plural gain control functions responsive to characteristics of the waveform signals. At least the waveform elements (waveform signals) ahead of the attack portion and the waveform elements of the release portion are gain controlled. The resulting gain-controlled audio signals are transformed into plural spectral components which are encoded along with the control information for gain control.