Abstract: A greeting message is recorded and stored at a first rate for use in a system. The system includes a command indicating the desired rate to be used in coding on the system. The greeting message is then reconstructed and re-encoded based on the desired rate. Alternatively, the greeting message may be stored in a plurality of rate formats. The appropriate rate format may then be selected by the system.

Abstract: A encoded-sound-code decoding method comprises a first step of performing a decoding process on a predetermined number of sample codes starting from a sound-reproduction start position at some midpoint of a sequence of codes based on a predetermined initial value of a sound parameter; a second step of making comparison between a judgment parameter corresponding to a decoding result and a predetermined threshold value thereby determining whether the decoding result is proper or not; a third step in which, in response to a determination that the decoding result is not proper, the initial value of the sound parameter used at the first step is modified and then the processes of the first and second steps are performed; and a fourth step in which, after repetitions of the process of the third step until the decoding result is determined to be proper, the codes are sequentially decoded from the sound-reproduction start position at some midpoint of the code sequence.

Abstract: To facilitate the reversion of a communication to tandem operation (200), a first transcoder (20), having previously changed back to tandem operation (206), inverts bits (208) of a double-encoded frame format that correspond to synchronisation bits in a non-tandem, single-encoded frame format to generate errors, in relation to the synchronisation bits, at a second transcoder (34). Upon detection (210) of a predetermined number of errors in the synchronisation bits during a predetermined time, the second transcoder (34) reverts to tandem operation, as shown in the flow diagram of FIG. 2.

Abstract: The present invention relates to a method and an arrangement for reconstruction of a received speech signal (r), which has been transmitted over a radio channel that has been subjected to disturbances, such as, e.g., noise, interference or fading. A speech signal (r.sub.rec), where the effects from these disturbances are minimized, is generated by an estimated speech signal (r), corresponding to expected future values of the received speech signal (r), produced according to a linear predictive reconstruction model in a signal modelling circuit. The received speech signal (r) and the estimated speech signal (r) are combined in a signal combination circuit according to a variable ratio, which ratio is determined by a quality parameter (q).

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: An audio decoder converts a linear PCM audio data packet into two concurrently provided digital audio sample sequences: a high-quality sequence and a decimated sequence. In one embodiment, the audio decoder is part of an audio system that further includes two audio devices. The first audio device is configured to produce an audio signal from a 96 kHz sequence, and the second audio device expects a 48 kHz sequence. The audio decoder includes an input interface, an arithmetic logic unit (ALU), and two output buffers. The input interface is configured to receive a linear PCM audio data packet and to reconfigure bytes as necessary to reconstruct a sequence of unscaled audio samples. The ALU multiplies each of the unscaled audio samples by a gain factor and buffers the resulting scaled audio sample sequence in a first output buffer.

Abstract: The present invention relates to a speech encoder/decoder system employing digital transmission in which the encoding and decoding operations are complimentary, and these operations make use of sets of parameters which may be optimized for a speaker and for a particular digital radio link. A number of sets of parameters are determined experimentally, for example, by employing human sample groups in which perceived audio and transmission quality are tested. The encoder/decoder system then employs a group, or number, of sets of parameters serving all speakers rather than employing one fixed set of parameters. The particular set of parameters for a speaker in the encoder of a first transceiver is determined by a processor which receives values based on an analysis of the input audio signal, and then a parameter set identifier is sent within the digital signal for use by a decoder of a second transceiver.

Abstract: A sound data decoder is provided which includes a decode portion, a PCM output buffer having a plurality of fractional banks, a bank management portion supplying an address indicating a location of a writable fractional bank to the decode portion, and a PCM output portion reading and outputting, in response to the address, data from a fractional bank corresponding to that address and supplying to the bank management portion an address indicating a location of a fractional bank which is made writable.

Abstract: A voice detector that detects whether an input voice signal is voiced or unvoiced, the detector has a long-term averaging circuit calculating a long-term weighted average value, a short-term averaging circuit calculating a short-term weighted average value, a noise level discriminator discriminating based on the long-term weighted average value and the short-term weighted average value and a voice discriminator determining voiced/unvoiced terms based on comparison of the long-term weighted average value and the discriminated noise level.

Abstract: Method and apparatus for encoding sub information is provided for synchronizing the main information of a digital audio disk with the sub information thereof. A sub signal encoding unit is composed as part of a unit for recording a signal of a digital audio disk. The sub signal encoding unit is inputted with a main signal having a format of the digital audio disk and a sub signal for improving the sound quality of the audio signal. A data cutting section is provided with the main signal and takes out part of the main signal as a synchronous signal. A Q channel generating section operates to replace the format of the synchronous signal with the Q channel data format so that the synchronous signal may be recorded in the Q channel of the subcode of the digital audio disk. A subcode encoding section is provided with the sub signal and the Q channel data and encodes the sub signal according to the format of the subcode.

Abstract: An audio decoder is described that can concurrently produce two synchronized outputs of a digital audio stream at different sampling rates and can provide for seamless switching between the rates. In one embodiment, the audio decoder includes a first output buffer, an arithmetic logic unit (ALU), a second output buffer, and a control module. The first audio buffer is configured to buffer a sequence of digital audio samples and to provide the first sequence of digital audio samples to an output device at 96 kHz. The arithmetic logic unit (ALU) is coupled to the first output buffer to retrieve the first sequence of digital audio samples and to convert the first sequence of digital audio samples into a decimated sequence of digital audio samples. The second output buffer is coupled to the ALU to buffer the decimated sequence of digital audio and to provide the decimated sequence of digital audio samples to a second output device at 48 kHz.

Abstract: The present invention provides a sinusoidal transform vocoder based on the Bark spectrum, which has high quality and low bit rate for coding. The present invention includes the steps of transforming a harmonic sine wave from a frequency spectrum to a perception-based Bark spectrum. An equal-loudness pre-emphasis and the loudness to a subjective loudness transformation are also involved in the method. Last, a pulse code modulation (PCM) is used to quantize the subjective loudness to obtain quantized subjective loudness. In synthesis, the Bark spectrum is inversely processed to obtain the excitation pattern following the sone-to-phone conversion and equal-loudness deemphasis. Then, the sine wave amplitudes can be estimated from the excitation pattern by assuming that the amplitudes belonging to the same critical band are equal.

Abstract: A method and system for producing speech signals is disclosed. The speech signals are produced by sequentially reproducing a series of stored speech signal segments. The speech signals may be used to generate a voice message. The signals are formed by reproducing signal segments for system announcement messages and beginning, end and word-pair fragments for "key" words. The system announcement messages and "key" words define a dictionary for the system. The signal segments for word pair fragments correspond to the end portion of one word, a transition to another word and a beginning portion of that other word. The transition between sequentially produced "key" words in a voice message generated from a signal produced from word pair fragments is audibly smooth. The resulting speech signal may correspond to any sequences of words from the dictionary. The method and system are suited for telephony applications, such as voice mail or directory assistance applications.

Abstract: A data compression apparatus for data compressing a digital information signal obtained from a digital audio signal. The digital information signal includes p-bit samples, where p is an integer larger than 1. The apparatus has an input (16) for receiving the digital information signal, and a lossless compression unit (18) for carrying out a substantially lossless compression step on the digital information signal so as to obtain a data compressed digital information signal. The lossless compression unit includes a Rice encoder, which is distinguishable by a code parameter m. Further, an output terminal (22) is available for supplying the data compressed digital information signal. The Rice encoder has a generator unit (30) for generating the code parameter m from N samples of the digital information signal, in accordance with a formula which optimizes the value of m for each frame of N samples.

Abstract: A method for controlling the point of time when a bypass mode operation is begun, based on the format of pulse code modulation (PCM) data. A switching unit, which is coupled between outgoing-end and incoming-end mobile stations, receives signals output from vocoders respectively associated with the outgoing-end and incoming-end mobile stations, thereby checking respective operation modes of the vocoders. Based on the result of the checking, the switching unit controls the vocoders so that communications between the outgoing-end and incoming-end mobile stations can be enabled when both the mobile stations operate in a bypass mode. Accordingly, it is possible to achieve smooth communications without a degradation in speech quality, as compared to conventional communications methods which do not take into consideration operation modes of outgoing-end and incoming-end mobile stations.

Abstract: A method and apparatus for discriminating between voice and voiceband data (fax/modem data) in an input signal from a voiceband channel, which is available by blocks (packets) of samples. Said discrimination is based upon the computation of two characteristics of the input signal: an autocorrelation function and a power variation function, the combination of which provides a discrimination factor which is highly accurate while requiring a low computing power.

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: There is provided a decoder and a data recovery apparatus which has enabled power saving by setting the band to which series of processes are not carried out among the N bands to vary the processing time required for the decoding operation within the predetermined time interval and extend non-operable period, on the occasion of independently executing the predetermined decoding operation in every band for the unit data obtained by dividing the compressed data into N (integer N.gtoreq.2) bands for every predetermined time interval.

Abstract: A first remote vocoder receives analog voice and produces packetized vocoder data which is transmitted over a wireless link. A first local vocoder receives the packetized vocoder data from the wireless link. The first local vocoder converts the packetized data to a multibit PCM output. The first local vocoder also adds a detection code to one of the least significant bits (LSB) of the PCM output. The first local vocoder passes the PCM signal to the PSTN from the second end user. The first local vocoder also receives PCM input over the PSTN. The first local vocoder constantly monitors the least significant bit of the PCM input for a detection code indicating that a second local vocoder is connected at the receiving end. If the first local vocoder detects the detection code from the second local vocoder, it begins to substitute packetized data and a redundancy check for a second one of the LSB's of the outgoing PCM. The first local vocoder also begins to monitor the second one of the LSB's of the incoming PCM.

Abstract: A small-sized audio signal reproducing apparatus for hearing reproduced audio signals with the aid of a headphone, wherein digitized and compression encoded audio signals, stored in a semiconductor memory, are read out so as to undergo a decoding operation, which is an inverse operation to compression encoding, to reproduce the audio signals, and the reproduced signals are heard by the headphone. The apparatus may be significantly reduced in size and weight as compared to the apparatus in which a tape or a disk is used as the recording medium.

Abstract: When one vocoding system is coupled to another vocoding system, a tandem arrangement results. The tandem configuration results in voice quality degradation as speech is encoded and decoded, then encoded and decoded again. One reason for the degradation is that postfiltering performed at the output of the speech decoding process introduces distortions in the spectral content of the reconstructed speech as compared to the original speech. The present invention prevents the degradation due to the use of postfilters by modifying the postfiltering within the vocoders where a tandem configuration exists. A detection code is embedded within the data signal to indicate the existence of a tandem configuration. If the detection code is received at a vocoder, modified vocoding is established within the vocoders to prevent the degradation due to the postfiltering.

Abstract: A digital signal processing method and apparatus processes a signal obtained on splitting the entire frequency band of an input digital signal into a plurality of sub-bands. The entire frequency band of the input digital signal is split into a plurality of sub-bands, and signals of each sub-band are allocated to a plurality of sub-words divided from a word of the input digital signal. In allocating the signals, the information for reducing the noise in at least one of the sub-bands is allocated to the sub-word different from the sub-word to which is allocated the signal of the sub-band whose noise is to be reduced.

Abstract: A data processing device uses a portion of random access memory 121 as an output buffer 124 for holding a portion of a stream of PCM data which is to be output to a digital to analog converter 530. D/A 530 forms a left analog channel and a right analog channel for speaker subsystems 814 and 815. The PCM data stream is stored in the output buffer so that PCM data samples which pertain to the left channel are stored at even address and PCM data samples which pertain to the right channel are stored at odd address. Control circuitry 145 monitors direct memory access (DMA) transfers which transfer PCM data samples to PCM serializer 142. By comparing the address of each DMA transfer to a left/right channel signal from the D/A, the control circuitry can verify that channel synchronization is correct. If a synchronization error is detected, an channel synchronization error correction procedure is invoked.

Abstract: An electronic method and apparatus for signal encoding and decoding to provide ultra low distortion reproduction of analog signals, while remaining compatible with industry standardized signal playback apparatus not incorporating the decoding features of the invention, and wherein the improved system provides an interplay of gain, slew rate and wave synthesis operations to reduce signal distortions and improve apparent resolution, all under the control of concealed control codes for triggering appropriate decoding signal reconstruction compensation complementing the signal analysis made during encoding. In addition, signals lacking the encoding process features of the invention are likewise compatible with playback decoders which do embody the invention, to provide some overall restoration enhancement.

Abstract: A pitch determination device which separates at least each frame of the input speech signal into separate, lower resolution portions is provided. The pitch determination device includes a resolution lowering unit, a signal selecting unit and a pitch determination device. The resolution lowering unit has an input line on which the input speech signal is provided and K output lines, on each of which output lines, one of K lower resolution input signals is provided. The signal selecting unit has K input lines connected to the K output lines of the resolution lowering unit and has an output line on which is provided one of the K lower resolution signals which fulfill a predetermined quality criterion. The criterion is typically based on the energy content of the lower resolution signals. The pitch determination device has an input line connected to the output line of the signal selecting unit and an output line which provides a pitch value for the selected lower resolution input signal.

Abstract: An MPEG audio decoder has an irregular-pattern processing circuit for detecting irregular patterns in the bit stream input to the decoder, and altering these irregular patterns, or altering data or signals derived from these irregular patterns, so that the irregular patterns do not cause annoying defects in the audio signal output from the decoder. The alteration may take the form of replacement by a minimum value, or interpolation of a preceding value.

Abstract: A computer system having a control unit, a rounding circuit, and an ADPCM decoder. The control unit generates ADPCM sound data. The rounding circuits rounds a calculating value of the ADPCM sound data so that the ADPCM decoder reproduces original sound from the rounding value of the ADPCM sound data provided by the rounding circuit.

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

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

Abstract: A data processing system (10) is disclosed which comprises a microprocessor host (12) coupled to a decoding system (14). A host interface block (18) receives a bit stream and passes bit stream on to a system decoder block (20). The system decoder block (20) extracts the appropriate data from the bit stream and loads an input buffer (24) or an optional external buffer (26). An audio decoder block (28) retrieves the data from the input buffer (24) and generates scale factor indices, bit per code word values and subband samples which are stored in an arithmetic unit buffer (30). A hardware filter arithmetic unit block (32) retrieves the information from the arithmetic unit buffer (30) and dequantizes, transforms and filters the data to generate PCM output data which is loaded into a PCM buffer (34). The data within the PCM buffer (34) is output by a PCM output block (36) to a digital-to-analog converter (16).

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

Abstract: A noise reducing method and apparatus for reducing quantization error or noise generated on quantizing input signals, such as digital audio signals, wherein a dither signal concentrated in signal energy in the low frequency range is summed to the input signal and the resulting sum signal is sent to a quantizer. The quantizer error generated in the quantizer is fed back via a noise filter to the input side of the quantizer. The resulting signal has a smoothed noise floor and has noise components suppressed in a frequency range to which the human hearing mechanism exhibits high sensitivity, thus realizing the sound quality comparable to that of the input signal which prevailed prior to noise shaping.