Abstract: A multimodal code-excited linear prediction (CELP) speech coder determines a pitch-lag-periodicity-independent peakiness measure from the input speech. If the measure is greater than a peakiness threshold the encoder classifies the speech in a first coding mode. In one embodiment only frames having an open-loop pitch prediction gain not greater than a threshold, a zero-crossing rate not less than a threshold, and a peakiness measure not greater than the peakiness threshold will be classified as unvoiced speech. Accordingly, the beginning or end of a voiced utterance will be properly coded as voiced speech and speech quality improved. In another embodiment, gain-match scaling matches coded speech energy to input speech energy. A target vector (the portion of input speech with any effects of previous signals removed) is approximated using the precomputed gain for excitation vectors while minimizing perceptually-weighted error.

Abstract: A voice encoder using a VOX (voice operated transmission) control has a pitch analyzer and a high-efficiency encoder. When a voiced state is detected in an input audio signal, the input audio signal and pitch information extracted therefrom are encoded by the high-efficiency encoder and transmitted to a voice decoder. When an unvoiced state is detected, the high-efficiency encoder encodes the input audio signal without a gain of the pitch information. The encoded data without using the gain information is transmitted after a post-amble signal to obtain natural background noise.

Abstract: In a speech encoder (4), a speech signal is encoded using a voiced speech encoder (16) and an unvoiced speech encoder (14). Both speech encoders (14,16) use analysis coefficients to represent the speech signal. The analysis coefficients are determined more frequently when a transition from voiced to unvoiced speech or vice versa is detected. This has been found to achieve significantly improved quality of speech reproduced from the encoded speech signal.

Abstract: Vocoder bypass is provided using in-band signaling. In preferred embodiments of the present invention, three signaling channels are arranged for transmission within the compressed speech. Each of the signaling channels are communicated at a preferred rate to permit fast, reliable detection of conditions indicating vocoder bypass mode of operation, to negotiate suitable vocoder type if necessary, and to synchronize and communicate compressed speech in a vocoder bypass mode of operation.

Abstract: The present invention can obtain a clear velocity converted sound in a sound signal which is recorded in recording media, without changing an interval of the sound signal. An input sound signal (1a) is transmitted from a sound signal storage memory (1) to a voiced sound/unvoiced sound deciding portion (2). In the voiced sound/unvoiced sound deciding portion (2), it is decided whether the input sound signal (1a) is a voiced sound or an unvoiced sound. A decision result is transmitted to a speech velocity converter (4) as a switching flag (1b). The speech velocity converter (4) outputs the unvoiced sound as it is. A predetermined windowing and adding processing is performed to the voiced sound, a time compression is carried out so as to output the voiced sound. An output signal (1e) from the speech velocity converter (4) is output as a frame output signal (1g) through an output sound signal frame buffer (8). In another mode, a switch and an adder may be used.

Abstract: Vocoder bypass is provided using a combination of out-of-band and in-band signaling. In preferred embodiments of the present invention, two signaling channels are arranged for transmission within the compressed speech. Each of the signaling channels are communicated at a preferred rate to permit fast, reliable detection of conditions indicating vocoder bypass mode of operation and to synchronize and communicate compressed speech in a vocoder bypass mode of operation.

Abstract: For transmitting voice between subscriber stations of a radio system, particularly a mobile radio telephone system, whereby an analog voice signal is digitally coded by voice frames (S), the voice frames (S) are first converted into voice-coded voice frames (S1). Subsequently, a plurality of these voice frames (S1) is combined to form a multi-voice frame (MSF1). These multi-voice frames (MSF1) are transmitted with error-protection. At the reception side, the multi-voice frames (MSF4) are in turn divided into voice-coded voice frames (S4) and, subsequently, the digital voice information contained in the transmitted voice frames is in turn converted into an analog voice signal.

Abstract: A MBE synthesizer (2200) for generating speech from information received by a receiver (114) includes a voiced signal generator (2280) for generating voiced signal components in the time domain using an IDFT in a pitch wave generator (2210) and and a pitch wave resampler (2232) and an unvoiced signal generator (2290) for generating unvoiced signal components in the time domain. The MBE synthesizer also includes a voicing processor (2218) responsive to band voicing flags within the excitation information for controlling selection of a voiced spectral component or an unvoiced spectral component from a harmonic amplitude spectrum.

Abstract: A barge-in detector for use in connection with a speech recognition system forms a prompt replica for use in detecting the presence or absence of user input to the system. The replica is indicative of the prompt energy applied to an input of the system. The detector detects the application of user input to the system, even if concurrent with a prompt, and enables the system to quickly respond to the user input.

Abstract: Speech is distinguished from noise by a spectral comparison of an input signal with a stored noise estimate. Updating of the noise estimate stored in a buffer is permitted during periods when speech is absent under control of an auxiliary detector. In order to improve operation in the presence of signals with strong harmonic components, e.g., signaling tones, an LPC prediction gain is computed from the input (x(i)) and a residual (y(i)) obtained from the input following filtering by a filter having a response complementary to the frequency spectrum of the input, and if the gain exceeds a threshold, buffer updating is suppressed.

Abstract: Adaptive speech coding includes receiving an original speech signal, performing on the original speech signal a current coding operation, and adapting the current coding operation in response to information used in the current coding operation. Adaptive speech decoding includes receiving coded information, performing a current decoding operation on the coded information, and adapting the current decoding operation in response to information used in the current decoding operation.

Abstract: A method and apparatus for voiced/unvoiced decision for judging whether an input speech signal is voiced or unvoiced. The input parameters for performing the voiced/unvoiced (V/UV) decision are comprehensively judged in order to enable high-precision V/UV decision by a simplified algorithm. Parameters for the voiced/unvoiced (V/UV) decision include the frame-averaged energy of the input speech signal lev, the normalized autocorrelation peak value r0r, the spectral similarity degree pos, the number of zero crossings nZero, and the pitch lag pch. If these parameters are denoted by x, these parameters are converted by function calculation circuits using a sigmoid function g(x) represented byg(x)=A/(1+exp (-(x-b)/a))where A, a, and b are constants differing with each input parameter. Using the parameters converted by this sigmoid function g(x), the voiced/unvoiced decision is made a V/UV decision circuit.

Abstract: A transmission system with a transmitter and a receiver. The transmitter has a speech encoder with analysis means, has calculation means, and has control means. The receiver has a speech decoder. Through a transmission medium, the transmitter transmits frames of data to the receiver. The analysis means determine analysis coefficients from a speech signal. From a bitrate setting, the calculation means calculate a fraction of the frames of data to carry more information about the analysis coefficients than a remaining number of the frames of data. The control means control the transmitter to transmit the fraction of the frames of data and to transmit the remaining number of the frames of data. The receiver receives the frames of data. The receiver derives a reconstructed speech signal from the received frames of data.

Abstract: A code printing apparatus includes a speech input unit for inputting speech information, a code output unit for printing the speech information input by the speech input unit on a medium as an optically readable code pattern image, and a speech code printing control unit for controlling the code output unit to print the speech information input by the speech input unit on the medium as a corresponding code pattern image on the basis of the input operation at an operation unit controlled by the user. The user can thus easily achieve a function of printing speech data or the like as an optically readable code.

Abstract: A speech coding apparatus which allows a speech decoding apparatus to output a more familiar background noise. The speech coding apparatus includes a voice presence/absence discrimination section, a coding section, a unique word production section, and a data switching section which selectively outputs one of outputs of the coding section and the unique word production section as an output of the speech coding apparatus in response to a result of discrimination of the voice presence/absence discrimination section. The speech coding apparatus further includes an amplitude level discrimination section, a clip processing section and an input switching section. The input switching section selects, when the input speech signal includes voice, the input speech signal, but when the input speech signal includes no voice and a code for updating background noise is to be produced, the input switching section selects the input speech signal after clip processing.

Abstract: A voice speed converter comprising a speech classifying unit for classifying an input speech signal into an unvoiced part and another part, a pitch frequency extracting unit for extracting a pitch frequency from the input speech signal and supplying it, a quasi-pitch frequency supplying unit for supplying a quasi-pitch frequency of fixed length, a voice speed converter for performing voice speed conversion processing on the input speech signal by the use of the pitch frequency or the quasi-pitch frequency, and a switch for controlling switching operations according to the classification result by the speech classifying unit, so as to send the quasi-pitch frequency to the voice speed converter when the input speech signal belongs to the unvoiced part, or so as to send the pitch frequency to the voice speed converter when the input speech signal belongs to another part.

Abstract: The system and method of the invention relates to voice detection technology for determining instants of time at which a snapshot of noise characteristics results in improved adaptation of noise floors used in voice detection. The approach is based on the "lower envelope" of the smoothed input signal power. Incorporation of this approach in a simple time domain VAD (Voice Activity Detector) results in an effective low-complexity system which, on the basis of simulations, gives good performance down to SNR values of about 0 dB. In the invention the lower envelope also provides the updated value of the noise threshold during the presence of speech. The invention can also be embedded in other, more complex (e.g., frequency domain) VADs at low computational cost.

Abstract: A transcoder which prevents tandem coding of speech in a mobile-to-mobile call within a mobile communication system uses a speech coding method for reducing transmission rate on the radio path. The transcoder includes a speech coder, which encodes the speech signal into speech parameters for transmission to a mobile station, and decodes the speech parameters received from the mobile station into a speech signal according to the speech coding method, as well as a PCM coder for transmitting an uplink speech signal to and for receiving a downlink speech signal from a PCM interface in the form of PCM speech samples. In addition to the normal operation, the transcoder transmits and receives speech parameters through a PCM interface in a subchannel formed by least significant bits of the PCM speech samples. Thus, it is possible to prevent tandem coding while maintaining the standard PCM interface, and the signaling and services associated thereto.

Abstract: To conduct pitch control of a voiced speech signal that is to be coded or decoded, the voiced signal is subjected to sinusoidal analysis coding for each coding unit obtained by dividing the voiced signal on the time axis at a predetermined coding unit. A linear predictive residual of the voiced signal is taken out, and resultant voiced signal coded data are processed. A pitch component of the voiced signal coded data coded by the sinusoidal analysis coding is altered without changing the phonemes by a predetermined computation processing in a pitch conversion unit.

Abstract: A method for reducing noise in an input speech signal by adaptively controlling a maximum likelihood filter that is provided to calculate speech components based on a probability of speech occurrence and on a calculated signal-to-noise ratio based on the input speech signal. The characteristics of the maximum likelihood filter are smoothed along both the frequency axis and along the time axis. In the case of the frequency axis, smoothing filtering is based upon a median value of characteristics of the filter in the frequency range under consideration and on the characteristics of the filter in neighboring left and right frequency ranges, and in the case of smoothing filtering along the time axis, smoothing is done both for signals of a speech part and of a noise part.

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: A voice presence/absence discriminator can accurately determine the presence or absence of a voice in a frame that includes an uttered syllable head portion of an input voice and avoids performing erroneous determinations in bad environments such as those where background noise is of a high magnitude. In a sub-frame power calculation section, a sub-frame power Pm is calculated in units of sub-frames prepared by dividing a frame into four sub-frame portions. Based on this sub-frame power Pm, in a frame maximum power production section, a moving average (short-period average value) of the power of a sub-frame and the power of a sub-frame that precedes this sub-frame by one unit are calculated in units of a sub-frame and the short-period average values are compared with each other among the sub-frames that constitute the same frame to thereby select a maximum one of them as the frame maximum power Pf of this frame.

Abstract: A system and a method is disclosed for verifying a voice of a user conducting a telephone transaction. The system and method includes a mechanism for prompting the user to speak in a limited vocabulary. A feature extractor converts the limited vocabulary into a plurality of speech frames. A pre-processor is coupled to the feature extractor for processing the plurality of speech frames to produce a plurality of processed frames. The processing includes frame selection, which eliminates each of the plurality of speech frames having an absence of words. A Viterbi decoder is also coupled to said feature extractor for assigning a frame label to each of the plurality of speech frames to produce a plurality of frame labels. The processed frames and frame labels are then combined to produce a voice model, which includes each of the plurality of frame labels that correspond to the number of plurality of processed frames.

Abstract: A speech frame is converted to speech bursts and interleaved over a predetermined number of TDMA frames for transmission. At a receive site, the transmitted signal is equalized and a quality signal indicating the quality of each burst signal is produced from the equalized signal. The equalized signal is de-interleaved, and then Vitrerbi-decoded speech samples are produced. A CRC circuit performs an error check operation on the decoded speech samples to produce an error bit if a speech frame failed the check. A decision circuit compares the quality signal with a first reference value and produces a first disable signal when it is lower than the first reference value. The decision circuit compares a total sum of the quality signals produced from the predetermined number of burst signals with a second reference value and produces a second disable signal when the total sum of the quality signals is lower than the second reference value.

Abstract: A method and apparatus for the selection of an encoding mode for speech frames in a variable rate encoding system. For each speech frame, the method and apparatus selects the encoding mode which provides for rate efficient coding. A mode measurement element receives a speech signal and a signal derived from the same speech signal, and generates a set of parameters which are ideally suited for operational mode selection. Rate determination logic receives the set of parameters and selects an encoding rate using predetermined selection rules. The selection rules further distinguish between unvoiced speech and temporally masked speech, which are encoded at the same rate but with different encoding strategies.

Abstract: If the same parameter is repeatedly used in an unvoiced frame inherently devoid of pitch, there is produced a pitch of the frame length period, thus producing an extraneous feeling. This can be prevented from occurring by evading repeated use of excitation vectors having the same waveform shape. To this end, when decoding an encoded speech signal obtained on waveform encoding an encoding-unit-based time-axis speech signal obtained on splitting an input speech signal in terms of a pre-set encoding unit on the time axis, input data is checked by CRC by a CRC and bad frame masking circuit 281, which processes a frame corrupted with an error with bad frame masking of repeatedly using parameters of a directly previous frame. If the error-corrupted frame is unvoiced, an unvoiced speech synthesis unit 220 adds the noise to an excitation vector from a noise codebook or randomly selects the excitation vector of the noise codebook.

Abstract: A speech packet transmission system of the present invention has a speech coding device and a speech decoding device. In the coding device a speech coder (21) codes a PCM (Pulse Code Modulation) speech and outputs the resulting speech coded data and prediction coefficients. A speech detector (22) determines whether the input PCM speech is voiced or unvoiced. A transmission prediction coefficient memory (23) memorizes the prediction coefficients. A delay circuit (24) delays the speech coded data by a preselected delay time. On the transition from an unvoiced state to a voiced state, a transmitter (25) sends the prediction coefficients as a leading packet and then sends the delayed speech coded data as the following packet. In the decoding device, a receiver (31) received the packets from the coding device separates the packets into the prediction coefficients and the coded data.

Abstract: A high efficiency encoding method for encoding data on frequency axis obtained by dividing an input audio signal on block-by-block basis and converting the signal onto the frequency axis, wherein V bands are searched for a band B.sub.VH with the highest center frequency if it is decided that there are one or more shift points of voiced (V)/unvoiced (UV) decision data of all bands on the frequency axis, and wherein the number of V bands N.sub.V up to the band B.sub.VH is found, so as to decide whether proportion of the V bands is equal to or higher than a predetermined threshold N.sub.th, thereby deciding one V/UV boundary point. Thus, it is possible to replace the V/UV decision data for each band by information on one demarcation in all bands, thereby to reduce data volume and to reduce bit rate. Also, by using two-stage hierarchical vector quantization in quantizing the data on the frequency axis, operation volume for codebook search and memory capacity of the codebook are reduced.

Abstract: A method and apparatus for reproducing speech signals at a controlled speed and for synthesizing speech includes a dividing unit that divides the input speech into time segments and an encoding unit that discriminates whether each of the speech segments is voiced or unvoiced. Based on the results of the discrimination, the encoding unit performs sinusoidal synthesis and encoding for voiced segments and vector quantization by closed-loop search for an optimum vector using an analysis-by-synthesis method for unvoiced segments in order to find encoded parameters. A period modification unit modifies the length of time associated with each signal segment and calculates a set of modified encoded parameters.

Abstract: An improved voice activity detection system and method is provided for use in speakerphones and other voice activated systems. To facilitate switching between various operating modes, the voice activity detection scheme utilizes a new voice energy term which is based on an integral of the absolute value of a derivative of a speech signal. Voice activity is detected during a silence mode by comparing a first ratio of a current voice energy value to a background noise value with a voice activity threshold value. Voice activity is detected when the first ratio is greater than the voice activity threshold value. Another step involves identifying a direction of the voice activity during a transmit and receive mode by comparing a second ratio of a transmit path voice energy value to a receive path voice energy value with a transmit threshold value and a receive threshold value. When the second ratio is greater than the transmit threshold value, voice activity is present in the transmit path.

Abstract: A speech coder, formed with a digital speech encoder and a digital speech decoder, utilizes fast excitation coding to reduce the computation power needed for compressing digital samples of an input speech signal to produce a compressed digital speech datastream that is subsequently decompressed to synthesize digital output speech samples. Much of the fast excitation coding is furnished by an excitation search unit in the encoder. The search unit determines excitation information that defines a non-periodic group of excitation pulses The optimal location of each pulse in the non-periodic pulse group is chosen from a corresponding set of pulse positions stored in the encoder. The search unit ascertains the optimal pulse positions by maximizing the correlation between (a) a target group of filtered versions of digital input speech samples provided to the encoder for compression and (b) a corresponding group of synthesized digital speech samples.

Abstract: A speed-variable speech signal reproduction apparatus and method for playing back speech signals stored in a storage medium at an adjusted speed while preventing any degradation in tone or loss of the speech signals from occurring. The method includes the steps of detecting the pitch of input digital speech signals using an average magnitude difference function, separating voice and voiceless sounds of the speech signals from each other based on the result of the detecting step, temporarily storing the separated voiceless sound, modulating the lengths of the speech signals by copying or eliminating a part of the separated voice sound, and synthesizing the modulated voice sound step with the voiceless sound temporarily stored in the storing step.

Abstract: A signal encoding system A1 includes a bark spectrum calculating device 2 for calculating a bark spectrum as a parameter based on an auditory model, a bark spectrum encoding device 3 for encoding the bark spectrum, a sound source calculating device 4 and a sound source encoding device 5. The bark spectrum calculating device 2 includes a power spectrum calculating device 6, a critical band integrating device 7, an equal loudness compensating device 8 and a loudness converting device 9. These devices are formed by engineering the functions and effects which are similar to those of the auditory model. The decoding process perform the conversion in the opposite direction. As a result, the signals can be encoded and decoded through less calculation in a manner well matching the human auditory characteristics. When speech signals are to be encoded, it can be realized through less calculation and memory while suppressing noise components other than the speech signal.

Abstract: A signal detector for detecting the presence of a intermittent signal component in a signal. The signal detector receives each of the signal strength samples during a corresponding iteration, and compares a threshold value with the received signal sample. The signal detector sets a counter to a pre-determined number if the sample compared is greater than the threshold value. The signal detector decrements the persistence counter if a corresponding sample is not greater than the threshold value. If the persistence counter is greater than a trigger value, the detector indicates the presence of a intermittent signal component or otherwise declares the absence of a intermittent signal component. The detector may indicate the presence of a intermittent signal component by a logical value of 1 and the absence by a logical value of 0. The threshold value is composed of two components; the intermittent signal component and the background signal component.

Abstract: A receiving unit receives input speech data on a frame-by-frame basis. An error detection unit checks whether errors exist in each frame, and outputs a signal indicative thereof to a first switch circuit. The first switch circuit outputs the input speech data to a second switch circuit if an error is detected, while it outputs the input speech data to a speech decoder unit if no error is detected. A data memory stores the input speech data after delaying the data by one frame, and outputs the delayed data to a bad frame masking unit for voiced frame, and a bad frame masking unit for unvoiced frame. The speech decoder unit decodes the input speech data by using spectral parameter data, delay of an adaptive codebook, an index of an excitation codebook, gains of the adaptive and excitation codebooks, and the amplitude of the input speech signal. The speech decoder unit outputs a decoding result to a voiced/unvoiced frame judging unit, as well as to an output terminal.

Abstract: A speech encoding method and apparatus for encoding an input speech signal on a block-by-block or frame-by-frame basis wherein short-term prediction residuals are found and then sinusoidal analytic encoding parameters are produced based on those short-term prediction residuals. Perceptually weighted vector quantization is performed for voiced blocks or frames by encoding their sinusoidal frequency or analytic harmonic magnitudes and, in the case of unvoiced blocks or frames, the time waveforms of the unvoiced blocks are encoded.

Abstract: A method an apparatus for modifying the play time of digital audio tracks. For each data block of samples taken from a plurality of such blocks comprising an audio stream or track, the original block data plus a time shifted copy of the block data are superimposed, in a manner dependent on either desired expansion or desired contraction of play time, to create an overlap region. The overlap region is selected at a position of best match as resulting from a normalized correlation over the overlap, the super position being weighted by a linear cross-fading function. All signal crossovers (signal transitions of opposite signal sign) in a block are located. A position at which to superimpose a copy of the data block with the data block, and the length of the data block, are determined based upon the location of said crossovers within a given data block.

Abstract: In a radio communication system such as a digital automobile telephone, a VOX controller stops a transmission of a frame having no voice data to perform power savings, and a VOX device discriminates whether the voice data are present or absent in each frame. In a reference power measuring circuit, all or selected pilot signals of a received frame are oppositely rotated using theoretical values of the pilot signals to uniform their phases, and in-phase vector synthesis is conducted. The vector synthesis result is averaged by the number of the synthesis symbols, and an absolute value of the averaged vector value becomes the desired signal reference power X. In a fading compensation circuit, by using the previous and next pilot signals of a data part, each fading vector is calculated. In a power measuring circuit, the data upon which the fading compensation has been conducted are oppositely rotated to uniform their phases, and in-phase vector synthesis is carried out.

Abstract: A audio reproducing apparatus includes an audio decoder and a voice speed converting unit. The audio decoder decodes an audio data stream to produce an audio signal. The voice speed converting unit converts the audio signal in such a manner that when a bit rate is higher than a normal bit rate, a pitch of a reproduced sound interval is the same as the pitch of the sound interval in a normal playback mode and a voice speed in the reproduced sound interval approaches a voice speed in a sound interval in the normal playback mode. The voice speed converting unit further performs voice speed conversion on the audio signal in such a manner that when the bit rate is lower than the normal bit rate, the sound interval is not noticeably interrupted.

Abstract: A method and a device for discriminating a voiced sound from an unvoiced sound or background noise in speech signals are disclosed. Each block or frame of input speech signals is divided into plural sub-blocks and the standard deviation, effective value or the peak value is detected in a detection unit for detecting statistical characteristics from one sub-block to another. A bias detection unit detects a bias on the time scale of the standard deviation, effective value or the peak value to decide whether the speech signals are voiced or unvoiced from one block to another.

Abstract: The periodicity of a signal from a voice channel is determined by sampling the signal, computing the log power spectrum, optionally thresholding and differencing the power spectrum, and then performing an autocorrelation function of limited order to confine the search for periodicity to spans of up to about 400 Hz to 500 Hz.

Abstract: A speech encoding communication system for use with a land mobile radio telephone system which decreases an unfamiliar feeling to a sound output caused by a cyclic tone variation of background noise. On the transmission side, an audition weighting filter selectively receives a sound signal or an output of a low-pass filter for the sound signal in response to VOX mode information. In a sound absent condition based on the VOX mode information, a power quantizer outputs a power index calculated by an average of the power over a long period, and an LPC analyzer outputs a unique value as an LPC and an LSP quantizer outputs a quantized LSP index and a quantized LPC obtained when the LPC has the unique value in the sound absent condition. Further, an adaptive codebook search unit controls an adaptive codebook index to a unique value without performing searching processing. On the reception side, a power controller receives a quantized power, VOX mode information and a sound signal.

Abstract: An encoding/decoding system employing vector quantization realizes a high quality encoding and decoding with decreased quantizing errors, employing a small sized codebook which faithfully represents each of the inputted waveform vectors. An encoding/decoding system includes an encoding apparatus and a decoding apparatus, each having a codebook for storing information vectors representative of a predetermined number of signal patterns and index that determine the information vectors. The encoding apparatus compares a vector representing an object signal to be quantized with each information vector in the codebook, selects an information vector that is closest to the vector and outputs an index for the information vector. The decoding apparatus obtains an information vector corresponding to the index obtained at the encoding apparatus side by referring to the codebook and decodes the object signal. The codebook utilizes a temporary memory connected thereto.

Abstract: A method and a device are described for classifying speech on the basis of the wavelet transformation for low-bit-rate speech coding processes. The method and the device permit a more robust classifier of speech signals for signal-matched control of speech coding processes in order to reduce the bit rate without affecting the speech quality or to increase the quality at the same bit rate. The method provides that, after segmenting the speech signal, a wavelet transformation is calculated for each frame, from which a set of parameters is determined with the help of adaptive thresholds. The parameters control a finite-state model, which subdivides the frames into shorter subframes if required, and classifies each subframe into one of several classes typical for speech coding. The speech signal is classified on the basis of the wavelet transformation for each time frame. Thus both a high time resolution (location of pulses) and frequency resolution (good mean values) can be achieved.

Abstract: A speech coding device capable of delivering a speech signal of excellent sound quality at a low bit rate is disclosed. The disclosed device is characterized by a method of calculating lag corresponding to pitch period and a speech signal coding method. Lag is calculated as follows: A speech signal is divided into frames; one frame is divided into a plurality of subframes; for each frame, subframes in which lag of a speech signal is expressed in the form of a differential relative to lag of a previous subframe and subframes in which lag is expressed in the form of an absolute value, i.e.

Abstract: A method is disclosed for generating frame voicing decisions for an incoming speech signal having periods of active voice and non-active voice for a speech encoder in a speech communication system. The method first extracts a predetermined set of parameters from the incoming speech signal for each frame and then makes a frame voicing decision of the incoming speech signal for each frame according to a set of difference measures extracted from the predetermined set of parameters. The predetermined set of extracted parameters comprises a description of the spectrum of the incoming speech signal based on line spectral frequencies ("LSF"). Additional parameters may include full band energy, low band energy and zero crossing rate.

Abstract: An apparatus synchronizes a voice coder and a voice decoder which are of the vector-coding type in order to prevent a false synchronization even when a signal having the same period as a string of synchronizing bits is inputted. A noise component adding unit adds a noise component to an input voice signal. Therefore, even if the input voice signal has the same period as that of a string of synchronizing bits and is completely periodic, the periodicity of the input voice signal is lost by the added noise component. Based on the input voice signal which is no longer periodic, a vector-coding unit, a quantizing signal vector generating unit, and a code book index transmitting unit generate code book indexes and transmit the generated code book indexes to a voice decoder. Therefore, the voice decoder is prevented from developing a false synchronization.

Abstract: A high efficiency encoding method for encoding data on frequency axis obtained by dividing an input audio signal on block-by-block basis and converting the signal onto the frequency axis, wherein V bands are searched for a band B.sub.VH with the highest center frequency if it is decided that there are one or more shift points of voiced (V)/unvoiced (UV) decision data of all bands on the frequency axis, and wherein the number of V bands N.sub.V up to the band B.sub.VH is found, so as to decide whether proportion of the V bands is equal to or higher than a predetermined threshold N.sub.th, thereby deciding one V/UV boundary point. Thus, it is possible to replace the V/UV decision data for each band by information on one demarcation in all bands, thereby to reduce data volume and to reduce bit rate. Also, by using two-stage hierarchical vector quantization in quantizing the data on the frequency axis, operation volume for codebook search and memory capacity of the codebook are reduced.

Abstract: Voice compression is performed in multiple stages to increase the overall compression between the incoming analog voice signal and the resulting digitized voice signal over that which would be obtained if only a single stage of compression were to be used. A first type of compression is performed on a voice signal to produce an intermediate signal that is compressed with respect to the voice signal, and a second, different type of compression is performed on the intermediate signal to produce an output signal that is compressed still further. As a result, compression better than 1920 bits per second (and approaching 960 bits per second) are obtained without sacrificing the intelligibility of the subsequently reconstructed analog voice signal. Voice compression is also performed by recognizing redundant portions of said voice signal, such as silence, and replacing such redundant portions with a special code in said compressed signal.

Abstract: A CELP speech decoder includes a first portion comprising an adaptive codebook and a second portion comprising a fixed codebook. The CS-ACELP decoder generates a speech excitation signal selectively based on output signals from said first and second portions when said decoder fails to receive reliably at least a portion of a current frame of compressed speech information. The decoder does this by classifying the speech signal to be generated as periodic (voiced) or non-periodic (unvoiced) and then generating an excitation signal based on this classification. If the speech signal is classified as periodic, the excitation signal is generated based on the output signal from the first portion and not on the output signal from the second portion. If the speech signal is classified as non-periodic, the excitation signal is generated based on the output signal from said second portion and not on the output signal from said first portion.