Abstract: A system and method for voice transmission over high level network protocols. On the Internet and the World Wide Web, such high level protocols are HTTP/TCP. The restrictions imposed by firewalls and proxy servers are avoided by using HTTP level connections to transmit voice data. In addition, packet delivery guarantees are obtained by using TCP instead of UDP. Variable compression based on silence detection takes advantage of the natural silences and pauses in human speech, thus reducing the delays in transmission caused by using HTTP/TCP. The silence detection includes the ability to bookend the voice data sent with small portions of silence to insure that the voice sounds natural. Finally, the voice data is transmitted to each client computer, independently from a common circular list of voice data, thus insuring that all clients will stay current with the most recent voice data. The combination of these features enables simple, seamless, and interactive Internet conferencing.

Abstract: A method for transmitting speech of a first person communicating with a second person via a packet switched network comprising: generating a stream of samples of the first person's speech during the communication; parsing the sample stream into audio frames; determining which audio frames correspond to periods when the first person is speaking and which correspond to periods when the first person is silent; transmitting audio frames corresponding to silent periods and speaking periods of the first person's speech; and transmitting at least some of the audio frames corresponding to speaking periods, but none of the audio frames corresponding to silent periods, at least twice.

Abstract: Speech recording is effected in a GSM phone handset (100) by storing in a memory (116) speech frames during the presence of speech, one or more SID frames during the absence of speech, and data representative of the duration of the absence of speech. In this way memory (116) does not store silent speech frames, and utilisation of memory space is therefore particularly efficient. In addition, items such as a voice activity detector and a comfort noise estimator, which are already provided in the handset as part of the GSM system, are “re-used” by the invention, thereby making efficient use of already-provided hardware/software.

Abstract: An apparatus for automatically generating punctuation marks in a continuous speech recognition system, comprises means (1,2,3,5) for recognizing user speech and converting the user speech into words, characterized in that means (1,2,3,5) for recognizing user speech is further used to recognize pseudo noises in the user speech; and the apparatus characterized by further comprising: means (9) for marking pseudo noises in output results of means (1,2,3,5) for recognizing user speech; means (10,14,13) for generating punctuation marks by finding most likely pseudo punctuation marks at locations of pseudo noises marked by the means (9) for marking pseudo noises based on a language model containing pseudo punctuation marks.

Abstract: In an acoustic signal encoding apparatus (100), a tonal noise verification unit (110) verifies whether the input acoustic time-domain signals are tonal or noisy. If the input acoustic time-domain signals are tonal, tonal component signals are extracted by a tonal component extraction unit (121), and tonal component parameters are normalized and quantized in a normalization/quantization unit (122). The residual time-domain signals, obtained on extracting the tonal component signals from the acoustic time-domain signals, are transformed by an orthogonal transforming unit (131) into the spectral information, which spectral information is normalized and quantized by a normalization/quantization unit (132). A code string generating unit (140) generates a code string from the quantized tonal component parameters and the quantized residual component spectral information.

Abstract: The invention relates to a method for transmitting background noise information including a silence descriptor identifier and background noise parameters in a communication system in which the information to be transmitted is formed into data frames. The data frames are subjected to channel coding to form channel-coded frames. The channel-coded frames are interleaved to be transmitted in two or more data transmission frames, and information of two channel-coded frames is transmitted in each data transmission frame. A first silence descriptor frame is formed provided with the silence descriptor identifier. The first silence descriptor frame is subjected to channel coding to form a channel-coded silence descriptor frame. The channel-coded silence descriptor frame is transmitted in two or more data transmission frames, and at least one data transmission frame transmitting part of the channel-coded silence descriptor frame is also used to transmit at least the background noise parameters.

Abstract: A digital speech signal processed by successive frames is subjected to noise suppression taking account of estimates of the noise included in the signal, updated for each frame in a manner dependent on at least one degree of vocal activity. A priori noise suppression is applied to the speech signal of each frame on the basis of estimates of the noise obtained on processing at least one preceding frame, and the energy variations of the a priori noise-suppressed signal are analyzed to detect the degree of vocal activity of said frame.

Abstract: According to one embodiment of the invention, an apparatus is provided which includes an encoder to encode input speech signals. The speech signals contain frames of talk spurts and silence gaps. The apparatus further includes a voice activity detector coupled to the encoder, the voice activity detector to detect whether a current frame of the input speech signals is the first active frame of a talk spurt. In response to the voice activity detector detecting that the current frame is the first active frame of a talk spurt, the encoder is reset and the encoder states are initialized.

Abstract: A communications transmitter which operates as a mobile telephone incorporates a noise suppressor (100, FIG. 1) which reduces the background noise in the transmitted voice signal. An external voice activity detector (150), which operates in conjunction with a noise suppressor (100) estimates the signal power of the incoming voice signal and compares this to an estimated noise floor. As a result of this comparison, a voice activity factor is applied to an updated noise floor estimate to create a voice activity threshold estimate. The voice activity threshold estimate is then used to decide whether or not to the force noise suppressor (100) to perform an update of a noise content estimate of the incoming voice signal.

Abstract: An overflow problem of LSF quantization in G.729 Annex B speech encoding. Preferred embodiments fix the problem with default or limited random variable assignments or flagging the overflow and adjusting the frame encoding such as by limiting spectral components or changing quantization targets.

Abstract: A medical record processor and method include a voice processor for receiving unprompted and unstructured free dictation, for processing the dictation, and for generating voice data comprising a plurality of terms therefrom. A key term comparator identifies a key medical term in the generated voice data by comparing each term in the generated voice data with a plurality of key medical terms. The identified key medical terms are separated from non-key medical terms in the generated voice data, and a non-key medical term is eliminated from the generated voice data. The identified key medical term is matched with a patient condition, and additional data are extracted from a knowledge database on the patient condition. A draft medical record is generated that includes the identified key medical term and the additional data.

Abstract: A method and system for utilizing information content in speech and a transition hangover between speech and noise to generate comfort noise on the decoder side. This adaptation to noise may be accomplished using various algorithms of estimating the spectrum of color noise. According to an embodiment of the present invention, an adaptation algorithm may be implemented that adapts with time, rather than a block based algorithm to prevent the repeated generation of artifacts present in the block that are being adapting to. The method and system of the present invention generates comfort noise in the absence of silent insertion descriptions containing spectrum information thereby saving bandwidth and generating colored comfort noise that reflects the spectrum of the actual noise.

Abstract: A buffering process for real-time digital audio is provided to effect of network “jitter” from inconsistent network packet delivery rates. The buffering algorithm is particularly useful for audio data including distinct bursts separated by silence, such as speech. The process holds incoming audio packets in a queue until either: (a) the buffer contents meet a predetermined threshold; or (b) the end packet of a burst is received. The result is that silent periods between bursts may expand or decrease relative to the original audio pattern, allowing cumulative jitter to be played out as silence. The threshold is sized such that the deviation in silence is unnoticeable by a listener. In an optional embodiment, the process periodically adjusts the threshold to adapt to network conditions.

Abstract: With respect to data having periodicity to be compressed, windows of the same size are set for every two sections according to an interval of peaks appearing substantially periodically and processing for sorting sample data alternately among the set windows of the same size is sequentially performed, whereby a frequency of data having periodicity is replaced with an approximately half frequency without damaging reproducibility to original data at all to make it possible to apply compression processing to data of the replaced low frequency. If this sorting processing is applied to compression processing having a characteristic that a compression ratio is not increased in a high-frequency region, it becomes possible to improve a compression ratio without damaging a quality of reproduced data by decompression at all.

Abstract: A first CN code (silence code) obtained by encoding a silence signal, which is contained in an input signal, by a silence compression function of a first speech encoding scheme is transcoded to a second CN code of a second speech encoding scheme without decoding the first CN code to a CN signal. For example, the first CN code is demultiplexed into a plurality of first element codes by a code demultiplexer, the first element codes are each transcoded to a plurality of second element codes that constitute the second CN code, and the second element codes obtained by this transcoding are multiplexed to output the second CN code.

Abstract: A method and apparatus for detecting silence in voice packets. A packet energy calculator calculates a smoothed energy value for each packet of voice data to be transmitted. A noise level detector adaptively calculates noise values during periods of said silence. A silent packet detector compares the energy value to the noise value and if it is less than the noise value and less than a predetermined silence ceiling value then silence is indicated. Also, if the energy value is less than a predetermined silence noise value then silence is also indicated.

Abstract: Disclosed is a five-step process for producing closed captions for a television program, subtitles for a movie or other uses for time-aligned transcripts. An operator transcribes the audio track while listening to the recorded material. The system helps him/her to work efficiently and produce precisely aligned captions. The first step consists of identifying the portions of the input audio that contain spoken text. Only the spoken parts are further processed by the invention system. The other parts may be used to generate non-spoken captions. The second step controls the rate of speech depending on how fast the operator types. While the operator types, the third module records the time the words were typed in. This provides a rough time alignment for the transcribed text. Then the fourth module realigns precisely the transcribed text on the audio track. A final module segments the transcribed text into captions, based on acoustic clues and natural language constraints.

Abstract: The invention relates to a voice activity detecting device and a voice activity detecting method. An object of the invention is to adapt to various characteristics of noise which may possibly be superimposed on an aural signal to thereby reliably discriminate between an active voice segment and a non-active voice segment. For this purpose, the voice activity detecting device comprises: a speech-segment inferring section 11 for determining the probability that each of active voice frames given in order of time sequence belongs to the active voice segment, based on the statistical characteristic of the aural signal; a quality monitoring section 12 for monitoring the quality of the aural signal for each active voice frame, and a speech-segment determining section 13 for weighting the determined probability with the above quality to obtain for each active voice frame the accuracy that the active voice frame belongs to the active voice segment.

Abstract: A half duplex switching device includes an input connection for receiving an input audio signal, and classification module coupled to the input connection. The classification module provides an output which indicates a classification of the input signal based upon a density of the input audio signal, an energy level of the input audio signal, and classification data provided with the input audio signal. A switching device is coupled to the classification module and determines if the received input audio signal contains speech signals based upon the output of the classification module. The communication receiving device can be used in both communication systems which provide continuous speech signals, and communication systems which remove silence and only provide speech signals.

Abstract: In a voice communication system having a transmitter and receiver on opposing sides of an interface, the transmitter is switched on to transmit speech components and is switched off during speech pauses. To provide comfort noise at the receiver, and thus avoid annoying effects caused by continual switching of the transmitter, a comfort noise generator disposed to produce comfort noise of an adjustable amplitude is located on the receiver side of the interface. A first subsystem responsive to operation of the transmitter provides a flag to the receiver to commence operation of the comfort noise generator, when the transmitter discontinues transmission in response to a speech pause. A second subsystem transmits a succession of amplitude parameters through the interface to selectively adjust the amplitude of the generator in corresponding relationship with a noise level at the transmitter.

Abstract: A method and apparatus for detecting and transmitting voice signals in a packet voice network system. The method and apparatus make use of a voice activity detection (VAD) unit at a transmitter, for determining if an input signal contains active audio information or passive audio information, where the input signal includes a plurality of frames. For one or more frames of the input signal containing active audio information, the VAD computes a hangover time period. This computation includes determining whether the hangover time period has a fixed duration or a variable duration on the basis of characteristics of the active audio information contained in the one or more frames. When the VAD detects a frame containing passive audio information subsequent to the one or more frames containing active audio information, the input signal is suppressed after the expiry of the computed hangover time period from the detection of the passive audio information.

Abstract: A system and method employing an access terminal for maintaining discontinuous communications including a gateway receiver for receiving the discontinuous information, a radio frequency (RF) communication link via geosynchronous earth orbit satellite for conveying multiple communication channels using time division multiple access (TDMA), the access terminal initiating information communication with the receiver via at least one of the multiple communication channels. The access terminal further includes a memory for storing protocol processing information and a transmitter for establishing the radio frequency communication link to the receiver of the terrestrial gateway system. The access terminal memory provides for destroying of a signal pattern or protocol assigned to the access terminal by the gateway receiver or transmission of keep-alive bursts by the transmitter during periods of inactivity to maintain information communication with the receiver.

Abstract: Frame power of an input signal is calculated to discriminate speech frame intervals from non-speech intervals, by thresholding current frame power using an adaptive speech-detection threshold based on the past maximum frame power value and the difference between past maximum and the minimum frame power values, adaptively updated using a predetermined number of frames prior to the current one.

Abstract: Various methods for effectively detecting the discontinuous transmission (DTX) mode are provided both from the physical layer perspective and from the medium access layer (MAC) perspective. The DTX mode transmission on reverse supplemental channels is detected by comparing a transmitted and a received power ratio of a pilot channel and a supplemental channel. For a reverse link communication, the DTX mode transmission is detected on dedicated control channels. The DTX mode transmission on a forward link supplemental channel can also be detected by monitoring both the transmitted and received power ratios of a primary power control channel and the supplemental channel. As an alternative to or in combination with the above mentioned methods, in order to monitor the DTX mode, a new network parameter is introduced which requires a change in the IS-2000-3 standard. This parameter acts as an indicator signaling to a receiver, from the MAC layer, that the transmitter has entered into the DTX mode.

Abstract: A technique is used in a speech encoder (107) that reduces non-speech activity of a low bit rate digital voice message. Speech model parameters that include quantized speech spectral parameter vectors are generated in a sequence of frames. A determination is made as to which frames of the sequence of frames are voiced frames and which frames are unvoiced frames. A consecutive sequence of frames of unvoiced frames is identified (2330) as an unvoiced burst when a length, NUV, of the consecutive sequence of frames exceeds a predetermined length, Ns. A non-speech activity portion of the unvoiced burst is identified (2335-2365) and removed.

Abstract: A method is disclosed for reducing the power during discontinuous transmission in a digital cellular systems. In order to reduce battery usage and decrease interference to other users, a method is disclosed which employs the transmission of a silence descriptor frame and the following reduction of power once the frame is properly received in the receiver. While the power is reduced there is still the ability to receive and decode inactivity frames.

Abstract: A first filter (2061 in FIG. 1) calculates a long-time average of first change quantities based on a difference between a line spectral frequency of an input voice signal and a long-time average thereof. A second filter (2062 in FIG. 1) calculates a long-time average of second change quantities based on a difference between a whole band energy of the input voice signal and a long-time average thereof. A third filter (2063 in FIG. 1) calculates a long-time average of third change quantities based on a difference between a low band energy of the input voice signal and a long-time average thereof. A fourth filter (2064 in FIG. 1) calculates a long-time average of fourth change quantities based on a difference between a zero cross number of the input voice signal and a long-time average thereof. A voice/non-voice determining circuit (1040 in FIG.

Abstract: An input voice signal is passed from a voice signal encoding unit to a voice signal compressing unit so that it is encoded and compressed. No transmission of a silence signal within the voice signal to a multiplexing unit is performed in a silence signal eliminating unit. A multiplexing unit multiplexes the voice signal with a data to be transmitted which has been passed via a data terminal control unit and a data terminal control unit from a data terminal and is transmitted to a communication line via the transmission control unit. Lapse of a time-out period of time which is present in a timer is detected in a silence signal inserting unit so that a silence signal is generated and is reproduced in a voice signal decompressing unit and a voice signal decoding unit.

Abstract: An apparatus for automatically generating punctuation marks in a continuous speech recognition system, comprises means (1,2,3,5) for recognizing user speech and converting the user speech into words, characterized in that means (1,2,3,5) for recognizing user speech is further used to recognize pseudo noises in the user speech; and the apparatus characterized by further comprising: means (9) for marking pseudo noises in output results of means (1,2,3,5) for recognizing user speech; means (10,14,13) for generating punctuation marks by finding most likely pseudo punctuation marks at locations of pseudo noises marked by the means (9) for marking pseudo noises based on a language model containing pseudo punctuation marks.

Abstract: A method of reducing undesired components from a signal that includes a desired component and undesirable components utilizes an autoregressive model technique. An autoregressive module determines a power spectral density approximation of the signal. An error component of the power spectral density approximation includes the desired component. Portions of the error component having frequencies outside of the expected range of the desired component preferably are filtered so that the result is the desired component with the undesired component removed. The invention is useful, for example, for reducing undesirable noise components from sound signals.

Abstract: An apparatus and method for forming medical records and a medical record formed thereby are provided. The apparatus preferably includes a voice input device for inputting free dictation of medical personnel, a voice processor responsive to the voice input device for processing the inputted voice and generating voice data therefrom, and a key term recognizer responsive to the voice processor for recognizing key medical terms freely dictated by the medical personnel. The key term recognizer preferably includes a key medical term database for storing a plurality of key medical terms therein. The apparatus preferably also includes a medical term matcher responsive to the key word recognizer for matching recognized key medical terms used by the medical personnel at least with patient conditions to thereby add additional data thereto and a medical record creator responsive to the medical term matcher for creating an actual patient medical record therefrom.

Abstract: Silence description coding for multi-rate speech coding systems that employ discontinued transmission. Speech coding systems include multi-rate speech codecs having an encoder and a decoder. The silence description coding is performed in either the encoder or the decoder of the multi-rate speech codec. It may also be performed in a distributed manner wherein it is performed partially in the encoder and partially in the decoder. The silence description coding is performed on a speech signal having a substantially non-speech-like characteristic. Voice activity detection classifies the speech signal as being either substantially speech-like or substantially non-speech-like. The silence description coding is selected from a plurality of coding modes. In certain embodiments of the invention, the silence description coding is a source coding mode that operates at a bit rate that fits within a bit rate budget as determined by all of the available source coding modes within the plurality of coding modes.

Abstract: A comfort noise block, that include a hangover period and comfort noise parameters, is transmitted in such a manner that it is not interrupted by other messages, such as FACCH messages. This is accomplished in a mobile station by a determination of whether any FACCH messages are required to be transmitted. If such FACCH messages exist, a further determination may be made as to which transmission can be made in the shortest time (i.e., the FACCH message or messages or the comfort noise parameters message), and this transmission is made first. In any event the comfort noise parameters block is transmitted without interruption. In a further embodiment of this invention the comfort noise parameters message is transmitted by being concatenated with another message, such as a neighbor channel measurement results message, so as to reduce overhead, conserve bandwidth, and reduce power consumption.

Abstract: A method of separating and identifying verbal announcements in a wireless transmission according to the GSM standard, in which verbal announcements from the audio stream can be associated with digital signals with the help of an ID number, so that the announcements can be separated.

Abstract: A method and apparatus for eighth-rate random number generation for speech coders includes a random number generator configured to generate values of a first random variable. A lookup table is used to store values of a second random variable. The lookup table is addressed with the values of the first random variable. The second random variable is an inverse transform of a cumulative distribution function of the first random variable. An codec encodes input silence frames with the values of the first and second random variables, and regenerates the silence frames with the values of the first and second random variables. The speech coder may be an enhanced variable rate coder, and the silence frames may be encoded at eighth rate. The random variables are advantageously Gaussian random variables with values that are uniformly distributed between zero and one.

Abstract: A voice recording and/or reproducing device includes a plurality of coders having different bit rates for coding voice to provide coded voice data, a voice recording mode change over switch for selecting one of the plurality of coders, and a system controller. The system controller stores coding selection data obtained by the change over of the voice recording mode and coded voice data obtained from the selected coder, to a storing medium, and reduces a deterioration of the voice due to the change over. The voice recording and/or reproducing device also includes a detector for detecting the coding selection data, and a plurality of decoders for decoding the coded voice data at the bit rate corresponding to the detected coding selection data.

Abstract: A method for playback of speech in an audio recording. The method comprises performing full word-level recognition of the speech including recognition of silent pauses and filled pauses, suppressing playback of the filled pauses and silent pauses, alerting a listener of the audio recording to locations of suppressed filled pauses and silent pauses during playback of the audio recording, and accepting a user command to disable suppression of any filled pause or silent pause during playback of the audio recording.

Abstract: A digital signal processor (100) receives a digitally vocoded signal (102), and calculates a staggered average value (404) from the frame energy of each received frame, or the product of the frame energy and a voicing value. While the staggered average value is above a threshold voice indicator value, speech is declared present.

Abstract: This invention relates to a method by means of which more than one audio signal can be recorded in compressed form in a memory element, and to a system implementing such a method. In the system according to the invention, audio signal samples are recorded only when voice is detected in the audio signals. The system according to the invention saves memory capacity required by the recording by combining the audio signal samples when voice is detected in samples of more than one audio signal. Furthermore, an audio signal is not recorded when no voice is detected in the signal. The invention also reduces the average computing capacity needed and thus power consumption, since signal combination, or mixing, is advantageously performed only when voice is detected in the samples of more than one audio signal.

Abstract: The present invention provides a sound compression/decompression system and method which processes sound data, frame by frame, while transmitting information on special processing at the same time without degrading the bit rate. When a frame for special processing is transmitted in the form of a bit stream, this system sets the index value of the bit stream to a special value not used for normal processing, thus preventing the bit rate from being degraded. This system also contains information on special-processing in the non-index bits of the bit stream to implement various types of special processings. For example, a plurality of consecutive unvoiced frames may be transmitted in a one-frame bit stream.

Abstract: It is judged before encoding sound data whether one frame of the sound data for transmission is silent, and if judged as silent, switching circuits 202 and 203, which is for stopping/resuming electric power supply to a circuit 201 for encoding the sound data, are turned off by a power control circuit 30A during an encoding period of one frame. The power control circuit 30A consists of a counter 32 initialized by a frame synchronization pulse FSYNC and counting a sound sampling clock CLKS, a constant setting part 34, a comparator 33 activating a coincident signal EQ when a count of the counter 32 coincides with a set value, and a flip-flop 31 set by a no-voice detect pulse NOV and reset when EQ is activated.

Abstract: A method and apparatus for discriminating non-sounds and voiceless sounds of speech signals, recorded on a recording medium, from each other when playing back the speech signals at a varied play-back speed. The method includes the steps of setting, as a reference voltage level, an optional value between a voltage level corresponding to non-sounds and a voltage level corresponding to voiceless sounds, detecting a pitch component of each waveform of the speech signals, comparing the absolute value of a voltage level of the detected pitch component with the reference voltage level, and distinguishing and outputting a portion of the speech signal associated with the detected pitch component based on the result of the comparison.

Abstract: A silence compression system that improves data compression in a digital speech storage device, such as a digital telephone answering machine, without undue clipping of voice signals. Instead of employing only real-time compression, the inventive silence system analyzes and compresses or re-compresses digital speech samples stored previously, when the voice messaging system is off-line or otherwise in a low priority state. A method of silence compression comprises receiving real-time speech samples, storing the same in memory, and analyzing the stored speech samples at a later time to determine thresholds for periods of silence. The periods of silence are then compressed, and the silence compressed voice message is restored in memory. In this fashion, the processor is not required to make a silence period determination on-the-fly simultaneous with encoding and compression of the real-time voice message, and thus is not subjected to heavy processor loads typically encountered in real time.

Abstract: A signal communication apparatus and method enables direct communication between communication systems of the silence compression and the non-silence compression type. The transmission and reception can each discriminate whether an audio signal is in a sound-present period or in a sound-absent period and this discrimination is output as period identification data. The audio signal is encoded, and the encoded data is selected and transmitted when the period identification data represents a sound-present period. Blank data prepared in advance are selected when the period identification data represents a sound-absent period. Encoded audio signal data of a variable bit rate are received, and the encoded data is selected, decoded, and output at a fixed bit rate when the period identification data represents a sound-present period. Data prepared in advance are outputted when the period identification data represents a sound-absent period.

Abstract: An audio data compression method improves over existing standards because of its encoding strategy for silence. The method analyzes the audio input to an encoder. If the audio is for an analyzed time frame is silence, a single byte output is generated by the encoder. If the next frame is silence, no output is generated. When a receiver receives the compressed data, and detects a one-byte silence signal, it can capture that signal and repeat it to a decoder. When the compressed signal reaches the decoder, it is decompressed into an analog signal.

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: In a voice coding section 1, a digital voice signal coded in a voice coder 10, a linear predictive coefficient used as a filter coefficient in a short-term predictive filter 102, a pitch period and a pitch predictive coefficient used, respectively, as a tap coefficient and a filter coefficient in a long-term predictive filter 103, and voice/no-voice status information of an input voice, are multiplexed in a multiplexer 12. Only when the voice/no-voice status information indicate the voice state is a cell assembled and transmitted. In a voice decoding section 2, the received cell is disassembled to provide multiplexed coded data. The voice signal is decoded by a short-term synthesis filter and a long term synthesis filter. The short term synthesis filter uses a linear predictive coefficient as a filter coefficient that is decoded from multiplexed coded data.

Abstract: An improved method for generating comfort noise (CN) in a mobile terminal operating in a discontinuous transmission (DTX) mode. In one embodiment the invention provides an improved method for comfort noise generation, in which a random excitation is modified by a spectral control filter so that the frequency content of comfort noise and background noise become similar. In another embodiment the transmitter identifies speech coding parameters that are not representative of the actual background noise, and replaces the identified parameters with parameters having a median value. In this manner the non-representative parameters do not skew the result of an averaging operation.

Abstract: Improved telecommunication apparatus is realized with a structure that is tailored to interact with the telecommunication network strictly in packet format. The apparatus includes circuitry for creating either control or information packets, where each packet comprises a header portion and a payload portion. The apparatus also includes an identifier module that enables the apparatus to identify itself to the telecommunication network with an identifier signal that is unique to it. The apparatus further includes encoding and decoding circuitry to create, and decode, highly compressed digital representations of audio signals and, optionally, encryption and decryption features to enhance security of communication. Still further, the apparatus includes circuitry to enable users to interact with the network and partake of telephony service.