Abstract: A method of controlling a terminal of an integrated circuit includes determining a frequency ratio between a frequency of a signal and a frequency of another signal received by an integrated circuit. A selected signal appearing at a selected terminal of the integrated circuit is selectively interpreted in accordance with an operating mode when the frequency ratio is below a selected value and in accordance with another operating mode when the frequency of the signal is above a selected value.

Abstract: A method for operating a voice function of a dual-mode mobile communication apparatus including a speaker's voice recognition function and a voice output function of stored information while the mobile communication apparatus is operating in an analog mode is disclosed. The method comprises the step of determining whether a voice function request signal is input or not, switching a vocoder into a digital mode for operating the voice function, and operating the voice function in digital mode.

Abstract: Methods and apparatuses for encoding speech signals in the presence of interference that accurately establishes a speech signal value subsequent to lost transmission packets. For one embodiment of the present invention, the initial bits of a speech transmission packet are encoded using a PCM encoding scheme and the remaining bits are encoded using a CVSD encoding scheme. Upon encoding, the initial bits of each packet, the instantaneous value of the voltage as derived from CVSD coder/decoder at the transmitter is encoded using PCM coding rather than CVSD coding. At the receiver, each packet is decoded independently using the PCM-encoded bits, rather than the terminal value of a preceding packet, to define a starting value. The PCM encoded bits of a valid packet are used to reestablish the signal value, thus avoiding packet-to-packet error extension in the presence of burst interference.

Abstract: In a digital cordless telephone system, a different speech coding scheme is used in the audio path from the base unit to its remote handset than that which is used in the audio path in the opposite direction from the remote handset to the base unit. It is found that this unbalanced coding scheme optimizes overall system cost and performance. In particular, two different types, quality and/or bit rates of speech encoders are implemented in opposite directions of the same full-duplex audio path, providing an unbalanced coding in a digital cordless telephone. Implementation of different types of speech encoders in a common full duplex path optimizes system cost and performance. It is recognized by the invention that the communications link in a first direction from a base unit to its remote handset in a digital cordless telephone system potentially requires better codec and audio performance than the communications link in the opposite direction from the remote handset to base unit.

Abstract: A system for determining a data converter operating mode includes measurement circuitry operable to measure a master clock frequency of a master clock signal and measure a frequency ratio between a data clock frequency of a data clock signal and the master clock frequency. A mapping system maps the measurements of the master clock frequency and the frequency ratio to an operating mode of the data converter utilizing an explicit formula. In a further embodiment, the mapping system maps the measurements of the master clock frequency and the frequency ratio to an operating mode of the data converter utilizing a lookup table. In an additional embodiment, the mapping system tests an available set of operating modes, independent of any previous tests, to determine a suitable operating mode for the data converter.

Abstract: An audio signal encoding apparatus includes a device for compressing multiple-channel digital audio signals into compression-resultant multiple-channel signals respectively. The multiple-channel digital audio signals relate to a sampling frequency and a quantization bit number. The compression-resultant multiple-channel signals, a signal representative of the sampling frequency, and a signal representative of the quantization bit number are formatted into a formatting-resultant signal. The formatting-resultant signal contains a sub packet and a sync information portion. The sub packet contains at least portions of the compression-resultant multiple-channel signals. The sync information portion contains the signal representative of the sampling frequency and the signal representative of the quantization bit number.

Abstract: A communication system is disclosed that encodes multiple bits of digital data on a single analog signal cycle. The communication system includes a digital data encoding system that receives the multiple bits of digital data and looks up the digital data in a Digital-to-Analog (D/A) conversion table. The D/A conversion table correlates the multiple bits of digital data to amplitudes of an analog signal and yields amplitudes values. The digital data encoding system then generates the analog signal cycle based on the amplitude values. The digital data encoding system advantageously increases the bandwidth available to customers, which is particularly important to help solve “last mile” bandwidth problems. The communication system also includes a digital data decoding system on the receiver side that decodes the multiple bits of digital data from the analog signal cycle using an Analog-to-Digital (A/D) conversion table.

Abstract: In a MPEG audio decoding process, an IDCT (Inverse Discrete Cosine Transform) process that generates time domain samples from frequency domain samples using a very limited number of prestored cosine coefficients is performed. Only the cosine coefficients that satisfy cos(?*(i/64)) where i=0-32 are prestored. The cosine coefficients for i=33-63 are calculated using the prestored coefficients by changing a sign of a corresponding symmetrical one of the stored coefficients, respectively. Then, sixty-four time domain samples (Vi) are generated from thirty-two frequency domain samples (Sk) according to the equation V i = ? k = 0 31 ? cos ? ( ( ? / 64 ) ? ( i + 16 ) ? ( 2 ? k + 1 ) ) × S k where i=0 to 63, using only the prestored cosine coefficients and the calculated cosine coefficients.

Abstract: In transmission of digital audio data usingIEEE1394, for example, when the audio data is changed from linear PCM to nonlinear PCM, identifier adding means of the transmitting apparatus inserts a silent identifier and nearly zero data for a specific time, and first identifier distinguishing means of the receiving apparatus changes over the output of data processing selecting means from linear PCM processing side to nonlinear processing side when detecting the silent identifier, thereby preventing generation of noise when changing over the data.

Abstract: Provided is a code correction circuit for presuming an error pattern of the received code when a received ADPCM code contains a click noise and correcting such error. A device for improving voice signal comprises a click noise detector (13) which judges that a received ADPCM code contains a click noise when a short interval average value of the absolute value of the received ADPCM code exceeds a predetermined value, and a code correction circuit (11) which presumes an error position of the bit of the received ADPCM code and corrects such error so as to minimize the variation in the short interval average value of the absolute value of the received ADPCM code when a transmission error is generated in the receiving ADPCM code judged to be containing a click noise.

Abstract: A flexible variable rate vocoder and related method of operation. The vocoder selects a target average data rate responsive to at least one network parameter and at least one external parameter.

Abstract: This invention is related to tandem free operation (TFO) in mobile cellular systems. The present invention implements a tandem free operation by using a special feedback loop which makes the decoded parameters available, performs the comfort noise insertion and bad frame handling operations, produces the parameter quantisation indices corresponding to the output of these operations, and synchronises the speech encoders and the speech decoders in the transmission path from the uplink mobile station to the downlink mobile station. This functionality is realized by partly decoding and re-encoding the parameters and synchronising and resetting the quantiser prediction memories in a specific manner. A basic idea of the invention is, that during BFH and CNI processes, a re-encoding block produces models of encoded speech parameters from the BFH/CNI processed speech parameters. These models of encoded speech parameters are then transmitted to the receiving end.

Abstract: The data stream between the transcoders (TCE1, TCE2) of a mobile wireless system is subdivided into a first data stream with samples for transmission and a second data stream with signal parameters for reconstruction of user data and/or for signaling. Both data streams are transmitted at the same time in particular in a handshake phase. The invention permits an improvement in the quality of transmitted data, e.g., speech data in a GSM network in tandem operation between mobile subscribers, in particular during a handshake phase.

Abstract: An input speech signal to an input terminal is supplied to a speech synthesizer section through a speech analyzer section and frequency parameter quantizer section to form a synthesis filter, and the input speech signal is expressed by quantized LPC coefficients representing the characteristics of the synthesis filter and an excitation signal for exciting the synthesis filter. In this case, in a pulse excitation section, a pulse position selector selects pulse position candidates from the integer pulse positions and non-integer pulse positions stored in a pulse position codebook, and an integer position pulse generator and non-integer position pulse generator respectively generate integer position pulses set at sampling points of the excitation signal and non-integer position pulses set at positions located between sampling points. These pulses are synthesized into a pulse train serving as a source of an excitation signal.

Abstract: The present invention relates to a method of determining quantization parameters which is N scalefactors (SF (I), I=1˜N) needed for quantizing N frequence subbands of an audio frame. The Ith scalefactor of the N scalefactors corresponds to the Ith frequence subband of the N frequence subbands. Each of the scalefactors has a psychoacoustic masking value (PM (I), I=1˜N), wherein N and I are natural numbers. The method comprises (a) calculating a Ith offset (O(I), I=1˜N)of the Ith frequence subband, (b) inputting the Ith psychoacoustic masking value (PM (I), I=1˜N) and the Ith offset into the first projection formula to generate an Ith first projection value (FPV (I), I=1˜N), and (c) repeating the above steps until the N scalefactors are determined.

Abstract: Methods and apparatus for quickly selecting an optimal excitation waveform from a codebook are presented herein. In encoding schemes that use forward and backward pitch enhancement, storage and processor load is reduced by approximating a two-dimensional autocorrelation matrix with a one-dimensional autocorrelation vector. The approximation is possible when a cross-correlation element is configured to determine the autocorrelation matrix of an impulse response and a pulse energy determination element is configured to determine the energy of a pulse code vector that incorporates secondary pulse positions.

Abstract: An audio signal encoding apparatus includes a device for compressing multiple-channel digital audio signals into compression-resultant multiple-channel signals respectively. The multiple-channel digital audio signals relate to a sampling frequency and a quantization bit number. The compression-resultant multiple-channel signals, a signal representative of the sampling frequency, and a signal representative of the quantization bit number are formatted into a formatting-resultant signal. The formatting-resultant signal contains a sub packet and a sync information portion. The sub packet contains at least portions of the compression-resultant multiple-channel signals. The sync information portion contains the signal representative of the sampling frequency and the signal representative of the quantization bit number.

Abstract: A system is provided for wavelet-based compression of an audio sample set including multiple audio samples. For each of the audio samples, the system receives the audio sample and, according to a psychoacoustic model, determines perceptually important information in the audio sample. The system decomposes the audio sample into multiple sub-bands according to a Wavelet Packet Transform and allocates bits to each of the sub-bands of the audio sample according to the determined perceptually important information in the audio sample. The system compresses the audio sample according to the allocation of bits to the sub-bands. The plurality of compressed audio samples includes a compressed audio sample set usable to generate a plurality of synthesized audio signals.

Abstract: The present invention relates to an audio waveform reproduction apparatus for reproducing a recorded audio waveform at a reproduction tempo that can be specified as desired, and its object is to achieve that the reproduction does not deviate from the tempo when performed at a tempo that is different from the tempo at the time of recording of the audio waveform.

Abstract: A sub-band adaptive differential pulse code modulation/encoding apparatus includes means (102, 103, 104, 105) having a predetermined asymmetric impulse response for receiving an audio signal and band-dividing the received audio signal into a predetermined number of sub-bands, so as to obtain a plurality of band-divided sub-band audio signals, a plurality of quantization means (110, 111, 112, 113) for quantizing the band-divided sub-band audio signals with the predetermined number of sub-bands, and encoding means (115) for performing adaptive differential pulse code modulation/encoding of the quantized sub-band audio signals.

Abstract: A speech compression system with a special fixed codebook structure and a new search routine is proposed for speech coding. The system is capable of encoding a speech signal into a bitstream for subsequent decoding to generate synthesized speech. The codebook structure uses a plurality of subcodebooks. Each subcodebook is designed to fit a specific group of speech signals. A better way is used to calculate a criterion value, minimizing an error signal in a minimization loop as part of the coding system. An external signal sets a maximum bitstream rate for delivering encoded speech into a communications system. The speech compression system comprises a full-rate codec, a half-rate codec, a quarter-rate codec and an eighth-rate codec. Each codec is selectively activated to encode and decode the speech signals at different bit rates to enhance overall quality of the synthesized speech at a limited average bit rate.

Abstract: A distribution unit 1 divides PCM audio data in plurality of divided data. Each divided data has overlapping sections overlapping with previous and following divided data. An MP3 encoding unit 2 encodes each divided data individually into MP3 data. A filtering process at the time of encoding is executed by using the overlapping sections as similar to before the division of the data. An analyzing unit 3 analyzes the overlapping sections of each divided data encoded into the MP3 data and searches a frame where main data (bit storage values) are not overlapping. A combiner unit 4 combines the adjoining divided data at the searched frames as combining frames. Although the audio data is divided and the divided data are encoded to MP3 data by parallel processes, a continuity of the data and compressing quality can be maintained.

Abstract: A method and apparatus for effectively encoding an audio signal into a Moving Picture Experts Group (MPEG)-1 layer III audio signal of a low-speed bitrate. In the audio encoding method, harmonic components are extracted using fast Fourier transformation (FFT) result information that is obtained by applying psycho-acoustic model 2 to received pulse code modulation (PCM) audio data. Then, the extracted harmonic components are removed from the received PCM audio data. Thereafter, the PCM audio data from which the extracted harmonic components are removed is subjected to a modified discrete cosine transform (MDCT) and quantization. Accordingly, efficient encoding can be achieved even using a small number of allocated bits.

Abstract: A digital audio signal processing method includes establishing a first frequency response corresponding to an audio signal, generating a second frequency response based on the first frequency response, and using the second frequency response to process the audio signal. The first frequency response has a plurality of first sampling points and a plurality of first amplitudes corresponding to the first sampling points respectively. The second frequency response is generated from selecting a portion of the first sampling points as second sampling points and corresponding first amplitudes as second amplitudes, in which the number of the second sampling points is less than the first sampling points. The second frequency response is used to process the audio signal through a convolution algorithm.

Abstract: In a method of lossless processing of an integer value signal in a prediction filter which includes a quantiser, a numerator of the prediction filter is implemented prior to the quantiser and a denominator of the prediction filter is implemented recursively around the quantiser to reduce the peak data rate of an output signal. In the lossless processor, at each sample instant, an input to the quantiser is jointly responsive to a first sample value of a signal input to the prediction filter, a second sample value of a signal input to the prediction filter at a previous sample instant, and an output value of the quantiser at a previous sample incident. In a preferred embodiment, the prediction filter includes noise shaping for affecting the output of the quantiser.

Abstract: The invention enables FAX signals to be transmitted transparently in a CDMA-WLL system. Data of many FAX signal waveforms are prestored in a vocoder in a terminal, and when a FAX signal is input, the index value of the waveform data having the smallest difference relative to the input signal is transmitted out.

Abstract: Methods and apparatuses for encoding speech signals in the presence of interference that accurately establish a speech signal value subsequent to lost transmission packets. For one embodiment of the present invention, the initial bits of a speech transmission packet are encoded using a PCM encoding scheme and the remaining bits are encoded using a CVSD encoding scheme. Upon encoding, the initial bits of each packet, the instantaneous value of the voltage as derived from CVSD coder/decoder at the transmitter is encoded using PCM coding rather than CVSD coding.

Abstract: The present invention discloses a voice signal synthesizing method and device, wherein voice signals are sampled at a relatively lower sampling frequency. During the reproduction of the signals, interpolation is used to calculate values of voice signals between two sampled periods and the calculated values are filled between the two sampled periods, whereby lower distortion rate may be obtained in the reproduced voice. This invention provides a low distortion rate and low sampling frequency voice signal synthesizing method and device.

Abstract: An input digital audio signal is divided into sub-band signals in respective sub-bands. Scale factors of the respective sub-bands are determined on the basis of the sub-band signals for every frame. Calculation is made as to differences between the determined scale factors for a first frame and the determined scale factors for a second frame preceding the first frame. Absolute values of the calculated scale-factor differences are calculated, and data representative of the calculated absolute values are generated. The data representative of the calculated absolute values are encoded into data of a Huffman code. Sign bits are generated which represent signs of the calculated scale-factor differences. The sub-band signals are quantized in response to the determined scale factors for every frame to generate quantized samples of the sub-band signals. The Huffman-code data, the generated sign bits, and the quantized samples of the sub-band signals are combined into a bit stream.

Abstract: Original digital audio signals are represented as PCM sample values wherein the distance between the values corresponds to the sampling frequency. Digital signals can have a length that is an integer multiple only of this time element. In particular coded digital audio signals are processed block-based, leading to a total length that is a multiple only of the block unit. According to the invention, information about the exact length of the original signal is transferred together with the encoded audio information. Additionally, an information value can be transferred that represents the total encoder and/or decoder delay. The decoder extracts these items of information and adjusts the total length of the decoded signal by cutting off samples from the decoded program or track.

Abstract: A speech coding apparatus and speech decoding apparatus to improve audio quality. The dequantized value obtained in dequantizing section 135 is input to adaptive bit assigner 140 per a predetermined number of frames such as a pitch period basis. Adaptive bit assigner 140 calculates an energy of the dequantized value, i.e., square sum of the dequantized value as a sample, output from each of ADPCM quantizers 130a to 130d, and based on the calculated energy of the dequantized value, determines the number of bits assigned to each residual signal to be quantized in respective one of ADPCM quantizers 130a to 130d.

Abstract: The invention provides a system and method for enabling DRR that supports multiple, simultaneous voice call events. In addition the invention provides for DRR that support the activation and deactivation of bit synchronous PCM timeslots within a well-defined period of time (e.g. not to exceed 100 ms) that can accommodate any voice call control signaling procedures, DRR that provides for robust communication of the DRR signaling protocol between the line terminal (LT) and the network terminal (NT), DRR that supports periodically refreshing the status of the bit synchronous PCM timeslots and DRR that aligns with proposed DSL bearer requirements.

Abstract: A speech compression system with a special fixed codebook structure and a new search routine is proposed for speech coding. The system is capable of encoding a speech signal into a bitstream for subsequent decoding to generate synthesized speech. The codebook structure uses a plurality of subcodebooks. Each subcodebook is designed to fit a specific group of speech signals. A criterion value is calculated for each subcodebook to minimize an error signal in a minimization loop as part of the coding system. An external signal sets a maximum bitstream rate for delivering encoded speech into a communications system. The speech compression system comprises a full-rate codec, a half-rate codec, a quarter-rate codec and an eighth-rate codec. Each codec is selectively activated to encode and decode the speech signals at different bit rates to enhance overall quality of the synthesized speech at a limited average bit rate.

Abstract: A data stream between transcoders of a mobile wireless system is subdivided into a first data stream with samples for transmission and a second data stream with signal parameters for reconstruction of user data and/or for signaling. Both data streams are transmitted at the same time permitting an improvement in the quality of transmitted data, e.g. speech data in a GSM network in tandem operation between mobile subscribers.

Abstract: A device for detecting a predetermined waveform in a received signal and synchronizing the detected waveform to the predetermined waveform is disclosed. The device includes a memory element for storing a reference set of encoded values. The reference set representing an encoded version of the predetermined waveform. An encoder is used to PCM encode the signal to obtain sets of encoded values representing the received signal. A processor calculates the statistical correlation coefficient of the reference set and the signal sets. The processor then determines the maximum statistical correlation coefficient. The predetermined waveform is detected in the signal if the maximum statistical correlation coefficient is greater than or equal to a predetermined threshold value. The device provides a compact, inexpensive, and fast method for detecting a known reference waveform in a received signal.

Abstract: The present invention relates to a network unit, an internet access device or gateway, a computer program product and a method for modifying a coded digital signal being represented by a set of parameter values of a speech or audio synthesis model. The coded digital signal is modified in the coded domain by modifying at least one of the parameter values. An application is acoustic echo and/or noise reduction of the coded digital signal.

Abstract: An apparatus and method for data processing that improves estimation of spectral parameters of speech data and reduces algorithmic delay in a data coding operation. Estimation of spectral parameters is improved by adaptively adjusting a gain function used to enhance data based on whether the data contains information speech and noise or noise only. Delay is reduced by extracting coding parameters using incompletely processed data. This data is formed by multiplying a less current portion of an input data frame with a synthesis window and a more current portion of the data frame with an inverse analysis window, and performing an overlap-add process on the data frame and a similarly processed previous data frame.

Abstract: The method of the present invention utilizes machine-learning techniques, particularly Support Vector Machines in combination with a neural network, to process a unique machine-learning enabled representation of the audio bitstream. Using this method, a classifying machine is able to autonomously detect characteristics of a piece of music, such as the artist or genre, and classify it accordingly. The method includes transforming digital time-domain representation of music into a frequency-domain representation, then dividing that frequency data into time slices, and compressing it into frequency bands to form multiple learning representations of each song. The learning representations that result are processed by a group of Support Vector Machines, then by a neural network, both previously trained to distinguish among a given set of characteristics, to determine the classification.

Abstract: The invention relates to a digital radio system and to a method for correcting a synchronization error in a digital radio system comprising at least one base station (100) communicating with terminals (102, 104) in its coverage area, and a mobile telephone exchange (108) communicating with the base station and controlling the operation of the base stations. The information to be transmitted is coded and decoded in a transcoder unit (200) into a form suitable for the transmission. The base station sends information frames to the transcoder at a certain pace, and, correspondingly, the transcoder sends information frames to the base station at a certain pace. To ensure easy transmission of information and to increase flexibility, the base station (100) indicates in an information frame sent to the transcoder (200) the synchronization error present in the information frames coming from the transcoder, and the transcoder corrects its synchronization after receiving said message.

Abstract: A deciding apparatus and method for deciding an audio signal coding system. A digital signal processor receives a coded audio signal, selects a specific coding system for the coded audio signal based on a predetermined portion of a data sequence of additional data of the audio signal, and decodes the audio signal using the selected coding system. A memory stores decoded programs for decoding the coded audio signal.

Abstract: A computer system having a PSG, a volume control circuit, a sound data output unit containing two channels of an ADPCM decoder each using a 32 kHz sampling frequency, a 32 bits CPU, an output control unit, and memories connected to the sound data output unit, the CPU, and the output control unit, respectively. So that the computer system has a high quality sound output function and can reproduce a sound data at high quality.

Abstract: A telecommunication system routs real-time information traffic from an originating digital radio unit served by an originating network to a terminating unit served by a terminating network via an intermediate network interconnecting the originating and terminating networks. The originating digital radio unit has an encoder/decoder (e.g., a vocoder) for generating digital wireless frames from information that is input thereto. The originating network includes an originating node with an encoder/decoder for performing wireless-specific conversion of the digital wireless frames to digital wireline (e.g., PCM) traffic. The intermediate network includes an originating-end interface node with an encoder/decoder for compressing the digital wireline traffic for transport across the intermediate network.

Abstract: An apparatus for decoding encoded voice data comprises a demodulator (101) which demodulates the encoded voice data (RF) and provides a demodulated encoded voice data (APO, RD), an adaptive differential pulse code modulation decoder (102) which decodes the demodulated encoded voice data and provides a pulse code modulation data (PO), an error detector (103) which detects whether error is present in the encoded voice data based on the demodulated encoded voice data and outputs a detection result (CRCERR) and a limiter (104) which outputs either the pulse code modulation data (POL) or a limit data (POL) in accordance with the detection result (CRCERR).

Abstract: A speech compression system with a special fixed codebook structure and a new search routine is proposed for speech coding. The system is capable of encoding a speech signal into a bitstream for subsequent decoding to generate synthesized speech. The codebook structure uses a plurality of subcodebooks. Each subcodebook is designed to fit a specific group of speech signals. A better way is used to calculate a criterion value, minimizing an error signal in a minimization loop as part of the coding system. An external signal sets a maximum bitstream rate for delivering encoded speech into a communications system. The speech compression system comprises a full-rate codec, a half-rate codec, a quarter-rate codec and an eighth-rate codec. Each codec is selectively activated to encode and decode the speech signals at different bit rates to enhance overall quality of the synthesized speech at a limited average bit rate.

Abstract: A device and method are provided for merging one or more secondary audio channels, received from either a streaming application or from memory, with a primary stream of audio data output from a main audio decoder. In addition to the audio decoder, the device/method employs a controller for processing received secondary audio data in a streaming application, and retrieving from system memory audio PCM data once stored. An audio playback device is provided for formatting the PCM data for mixing with the primary stream of audio data output from the audio decoder. Multiple digital-to-analog converters convert the multiple streams of audio data into analog signals which are then mixed into a single merged audio signal for presentation.

Abstract: A data processing apparatus includes an input terminal for receiving an audio signal, a 1-bit A/D converter for A/D converting the audio signal so as to obtain a bitstream signal, and a prediction unit for carrying out a prediction step on the bitstream signal so as to obtain a predicted bitstream signal. The data processing apparatus further includes a signal combination unit for combining the bitstream signal and the predicted bitstream signal so as to obtain a residue bitstream signal, and an output terminal for supplying the residual bitstream signal. A recording apparatus or a transmitter apparatus can use the data processing apparatus.

Abstract: A digital cordless phone system and a communication method for improving the distance of speech communication using error concealment are provided. The digital cordless phone system has a base unit and a remote unit, and each of the base and remote units has a first codec, a data processing unit, a controller, a modem and a switch. The first codec encodes voice data in a first encoding mode, and decodes the data encoded in the first encoding mode in a first decoding mode.

Abstract: The present invention provides a method of generating a signal that may be used to determine the characteristic response of a communication channel that utilizes the public Digital Telephone Network (DTN). The channel includes the DTN, which may have Network Digital Attenuators (NDA) and/or Robbed Bit Signalling (RBS), and a Digital-to-Analog Converter (DAC), (also known as a codec), as well as the analog characteristics of the local loop, typically a twisted pair of copper wires. The present invention provides a method and apparatus to determine the optimal sampling instant of the received data stream. The present invention provides a probing signal that is well-suited for use in determining the channel's response to a known sequence of PCM codes used as data symbols. This is especially useful in so-called PCM modulation schemes that utilize the DTN, where knowledge of network and DAC distortion predicates the selection of available PCM codes used to represent data.

Abstract: To decode voice data coded in a coding system on a specific standard (for example, ADPCM or a different standard), the voice data is previously sorted by a data sorter 102 and is stored in a second data storage 103. Voice data read from a third data storage 105 is decoded by a decoder 107 and is converted from parallel data into serial data by a parallel/serial data converter 109. The resultant data is transferred to a D/A converter, whereby the voice data coded on one standard and the voice data coded on a different standard can be decoded and transferred to the D/A converter by circuit change on a small scale without the need for a separate voice control system for each coding system.

Abstract: A telephony device includes a base station and a wireless handset. The base station has a responder with a memory. A processor of base station processes digital speech signals having a first code on signal lines connecting the input and output of the base station. The same processor also controls a converter which is located between the memory and the signal lines. Prior to storing in the memory, the converter codes the digital speech signals into a second code for storage in the memory, where the second code is the same code of the digital signals exchanged between the base station and the handset. Further, prior to reading from the memory, the converter decodes the digital speech signals of the second code stored in the memory into the first code.