Abstract: An audio decoding apparatus decodes high frequency component signals using a band expander that generates multiple high frequency subband signals from low frequency subband signals divided into multiple subbands and transmitted high frequency encoded information. The apparatus is provided with an aliasing detector and an aliasing remover. The aliasing detector detects the degree of occurrence of aliasing components in the multiple high frequency subband signals generated by the band expander. The aliasing remover suppresses aliasing components in the high frequency subband signals by adjusting the gain used to generate the high frequency subband signals. Thus occurrence of aliasing can be suppressed and the resulting degradation in sound quality can be reduced, even when real-valued subband signals are used in order to reduce the number of operations.

Abstract: A wideband, high quality audio signal is decoded with few calculations at a low bitrate. Unwanted spectrum components accompanying sinusoidal signal injection by a synthesis subband filter built with real-value operations are suppressed by inserting a suppression signal to subbands adjacent to the subband to which the sine wave is injected. This makes it possible to inject a desired sinusoid with few calculations.

Abstract: A method of normalizing received digital audio data includes decomposing the digital audio data into a plurality of sub-bands and applying a psycho-acoustic model to the digital audio data to generate a plurality of masking thresholds. The method further includes generating a plurality of transformation adjustment parameters based on the masking thresholds and desired transformation parameters and applying the transformation adjustment parameters to the sub-bands to generate transformed sub-bands.

Abstract: A tri-phosphor light having a reduced emission spectrum is matched with program encoding variables for a video encoder, such that the video encoder substantially encodes only those portions of the visible spectrum corresponding to wavelength bands centered around intensity peaks of the light's output, and substantially does not encode wavelengths which fall above or below the high-wavelength and low-wavelength peaks, respectively, of the light's spectral output.

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: An auditory scene is synthesized from a mono audio signal by modifying, for each critical band, an auditory scene parameter (e.g., an inter-aural level difference (ILD) and/or an inter-aural time difference (ITD)) for each sub-band within the critical band, where the modification is based on an average estimated coherence for the critical band. The coherence-based modification produces auditory scenes having objects whose widths more accurately match the widths of the objects in the original input auditory scene.

Abstract: An envelope generator (20), comprises: an input terminal (20a) for having a signal inputted therein; a first integrator (21) for generating intermediate state of envelopes with a first attack time and a first release time in response to changes in level of said signal inputted through said input terminal (20a) to impart said intermediate state of envelopes to said signal; a second integrator (22) for respectively modifying said intermediate state of envelopes into final state of envelopes with a second attack time and a second release time in response to changes in level of said signal imparted said intermediate state of envelopes; and an output terminal (20d) for outputting said signal with said final state of envelopes therethrough. The envelope generator (20) thus constructed can make gain signal follow rapid fluctuations in level of an audio signal, and can impart a relatively high quality for compressing and expanding level of the audio signal not to break in shape.

Abstract: An MPEG Optimization Software (MOPSW) for maximizing a Video Compression Ratio (VDCR) while maintaining a good output Video Image Quality (VDIQ) of an Input MPEG file (IMPEG) is disclosed. The IMPEG has a set of adjustable MPEG Control Parameters affecting both its VDCR and VDIQ. In a specific embodiment with application to MPEG2 files, the MOPSW employs the following optimized set of adjustable MPEG Control Parameters: Set Video Size=(x)×(Minutes of Video) where x=12 through 18; Maximum BITRATE=2200 through 3300; Maximum Average BITRATE=2200; and Minimum BITRATE=300 and achieved a VDCR that is at least 200% higher than what is typically available from current DVD suppliers in the art while maintaining a good output VDIQ for a representative set of movie titles. An associated method of iteratively determining the optimized set of adjustable MPEG Control Parameters is also presented.

Abstract: A method and apparatus for audio compression receives an audio signal. Transform coding is applied to the audio signal to generate a sequence of transform frequency coefficients. The sequence of transform frequency coefficients is partitioned into a plurality of non-uniform width frequency ranges and then zero value frequency coefficients are inserted at the boundaries of the non-uniform width frequency ranges. As a result, certain of the transform frequency coefficients that represent high frequencies are dropped.

Abstract: In a digital audio broadcast system, to utilize transmission bandwidth efficiently, representations of a stereo audio signal for transmission are generated in accordance with an inventive multidescriptive coding technique. The representations, as generated, are then transmitted through multiple communication channels, respectively. The transmitted representations are received by a receiver where one or more of the representations are selected for recovery of the stereo audio signal. Because of the design of the multidescriptive coding used, the more representations are selected to recover the stereo audio signal, the higher the quality of the recovered signal.

Abstract: A method for transferring real time information on a record carrier, typically bitstream audio on an optical disc, which method comprises encoding consecutive segments of the real time information to compressed real time data in frames, and determining a buffer occupancy for at least one frame, which buffer occupancy is indicative of an amount of compressed real time data to be present in the playback buffer at the start of decoding said frame. A signal is transmitted carrying the compressed real time data and the buffer occupancy, which data are received, stored in a playback buffer and finally decoded. The retrieving and/or the decoding is controlled in dependence on said transferred buffer occupancy. A playback buffer can be used effectively without risk for underflow or overflow. Also a method for recording audio information on a record carrier, a recording device, a record carrier and a playback device are described.

Abstract: A method for summarizing a video first detects audio peaks in a sub-sampled audio signal of the video. Then, motion activity in the video is extracted and filtered. The filtered motion activity is quantized to a continuous stream of digital pulses, one pulse for each frame. If the motion activity is greater than a predetermined threshold the pulse is one, otherwise the pulse is zero. Each quantized pulse is tested with respect to the timing of rising and falling edges. If the pulse meets the condition of the test, then the pulse is selected as a candidate pulse related to an interesting event in the video, otherwise the pulse is discarded. The candidate pulses are correlated, time-wise to the audio peaks, and patterns between the pulses and peaks are examined. The correlation patterns segment the video into uninteresting and interesting portions, which can then be summarized.

Abstract: Waveform data is extracted by referring to an existing waveform dictionary. Regarding the waveform data, a use frequency used for speech synthesis is accumulated and stored. A compression method is gradually changed in accordance with the use frequency, whereby the waveform data is compressed and stored in the waveform dictionary. Furthermore, information on a compression method for each compressed waveform data is stored, and the compressed waveform data is expanded based on information regarding the compression method. Regarding the use frequency of the waveform data, one or a plurality of predetermined threshold values are determined, and in a plurality of use frequency ranges partitioned with threshold values, the waveform data belonging to a use frequency range with a lower use frequency is compressed at a correspondingly increased compression ratio.

Abstract: This invention relates to a signal transmission apparatus and a signal transmission method both for transmitting a plurality of pieces of encoded audio information having a plurality of sampling frequencies of F or 1/N×F, which are encoded by the same encoding method, via a digital interface to a signal reception apparatus.

Abstract: Herein disclosed is an audio signal encoding apparatus comprises initial maximum scale factor band calculation means for calculating an initial maximum scale factor band for an audio signal inputted therein on the basis of the result made by the frame length determining means and the coded mode information inputted from the coded mode information means with reference to the initial maximum scale factor band information and Signal-to-Mask ratio threshold value information stored in the maximum scale factor band table storage means, and maximum scale factor band calculation means for calculating a maximum scale factor band for the audio signal on the basis of the initial maximum scale factor band calculated by the initial maximum scale factor band calculation means in accordance with the Signal-to-Mask ratio information calculated by the psychoacoustic model analyzing means, thereby making it possible to adaptively calculate the maximum scale factor band for the audio signal in accordance with the coded mode in

Abstract: A method for updating multimedia feature information such as weight and reliability is provided. The method according to the present invention first performs a multimedia retrieval based on the previously used weight, receives one or more user feedback on a relevance of the retrieval, calculates a performance of the present retrieval, and updates reliability of the present multimedia feature in consideration of the calculated retrieval performance. On the basis of the updated reliability, the weight of the present multimedia feature is updated.

Abstract: A recording apparatus includes quality test and feedback features for recording speech information of a dictation and subsequently transfers the recorded speech information to a speech recognition device for off-line speech recognition. A recorder records speech information in a recording mode. A transfer unit transfers recorded speech information to the speech recognition device in a transfer mode. The speech recognition device recognizes text information to be assigned to the transferred speech information, the quality of recognized text information depending on the quality of the received speech information. A speech quality tester tests whether the quality of the speech information received in the recording mode is sufficient for obtaining a predefined quality of the recognized text information when the speech information is processed by the speech recognition device.

Abstract: An audio playback/recording apparatus includes an audio input processing section which receives analog audio data, and converts the analog audio data to digital audio data; a playback/recording processing section which compresses digital audio data output from the audio input processing section and stores the compressed digital audio data into a RAM and which decompresses the compressed digital audio data according to attribution data indicating a type of compression; an audio output processing section which receives the decompressed digital audio data, converts the decompressed digital audio data to analog audio data, and outputs the analog audio data to an output apparatus; and an external recording circuit section which records compressed digital audio data stored in the RAM into an external recording medium, reads out the compressed digital audio data, and stores the data into the RAM.

Abstract: An efficient finite length POW10 calculation for MPEG audio encoding. A method for encoding an audio input signal includes storing a plurality of predetermined tonal values corresponding to a plurality of predetermined power levels. The method also includes receiving a plurality of input values each representative of a power level of a spectral component of the audio input signal at a corresponding frequency sub-band and accessing at least one corresponding tonal value of the plurality of predetermined tonal values. The method further includes generating an encoded output signal representative of the audio input signal by using at least one corresponding tonal value for each of the plurality of input values. Further, the storing of the plurality of predetermined tonal values is performed prior to the receiving of the plurality of input values.

Abstract: A digital processing system for monitoring sound effects produced by codecs during a signal processing session is provided. The digital processing system comprises, a sound source for producing signals for processing, a sound monitor having at least two channels for monitoring sound quality of the sound source, a sound recorder for recording the sound source, a playback device for playing a recorded file of sound produced by the sound source, a codec simulator for simulating sound effects produced by codecs, a plurality of codecs for compressing and decompressing sound files and a control interface for sampling, adjusting, and implementing an optimum codec based on monitoring of sound effects produced by codec simulation. A user controlling the system may monitor sound variances produced by any one of the plurality of codecs affecting the quality of sound from the sound source during signal processing of the source sound without interruption of the signal-processing session.

Abstract: Audio tracks or other portions of a particular type of audio material to be encoded are analyzed to determine a value of at least one coding-related parameter suitable for providing optimal encoding of the particular type of audio material. When a given portion of the audio material is to be encoded for transmission in a perceptual audio coder of a communication system, the value of the coding-related parameter is identified and then utilized in conjunction with the encoding of the given portion. The determined value of the coding-related parameter may be at least a portion of a psychoacoustic model utilized in encoding the given portion of the particular type of audio material in the perceptual audio coder. As another example, the value of the coding-related parameter may be a setting of an audio processor utilized to process the given portion of the particular type of audio material prior to encoding the given portion in the perceptual audio coder.

Abstract: Compression of voice and data signals by at least an order of magnitude that permits greater use of the limited bandwidth at the low frequency of operation. A voice recognition system that converts words spoken into a microphone into a sequence of letters and gaps. An encoder codes the letters into a digital message. A transmitter transmits the digital message to a receiver over a communications link. The received digital message is decoded in a decoder and a speech synthesizer converts the decoded message into spoken words that are annunciated on a speaker. In addition, an initial message that identifies a stored voice type that is to be synthesized, or a simultaneous signal that tailors the voice synthesizer in real time as the voice of the speaker changes may be transmitted to cause the speech synthesizer to more accurately resemble a speaker's voice. A speech-to-text processor and a display may be used to display the message at the receiver for hearing impaired users and users located in noisy areas.

Abstract: Voiceband compression techniques are employed in order to enable an RF telecommunications base station to accommodate data signals of high speed voiceband modems and FAX machines. An Ultra-High Speed Codec supports voiceband modem and FAX transmissions up to 14.4 kb/s and operates using four 16-phase RF slots. Because these codecs transmit information over several RF slots which can be contiguous, the slots within RF communication channels are dynamically allocated. The Dynamic Time slot/Bandwidth Allocation feature detects and monitors the data transmission and forms a data channel from the necessary number of slots.

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: Different dynamic range control values are applied to the 2-channel and m-channel outputs without repeating the inverse transform or the windowing of the audio samples. First, m-channel dynamic range control values are applied to audio samples in the frequency domain (“frequency samples” or “frequency coefficients”). The frequency samples are then inverse transformed to generate audio samples in the time domain (“time samples”) and windowed to generate windowed time samples. The windowed time samples are saved and the 2-channel dynamic range control values are applied to the windowed time samples. 2-channel dynamic range control values include 2-channel scale factors that, when multiplied with groups of the windowed time samples, at least partially remove the effects of windowing and the m-ch dynamic range control values applied in the frequency domain and readjust the dynamic range for 2-channel output.

Abstract: Different dynamic range control values are applied to the 2-channel and m-channel outputs without repeating the inverse transform of the audio samples. First, m-channel dynamic range control values are applied to audio samples in the frequency domain (“frequency samples” or “frequency coefficients”). The frequency samples are then inverse transformed to generate audio samples in the time domain (“time samples”). The time samples are duplicated to two sets where the 2-channel dynamic range control values are applied to one set of time samples. 2-channel dynamic range control values include 2-channel final scales that, when multiplied with the first set of time samples, at least partially remove the effects of the m-channel dynamic range control and readjust the dynamic range for 2-channel output. The first set and the second set are then windowed.

Abstract: A digital audio reproducing apparatus including a receiver receiving modulated data, a demodulator demodulating the modulated data received by the receiver, an audio decoder decoding, in a unit of a frame, digital audio information contained in the modulated data demodulated by the demodulator, and an audibility corrector for effecting audibility correction on failing digital audio information contained in a frame that failed to be decoded, when the audio decoder fails to decode the digital audio information.

Abstract: A telephone set on a dialing side includes a voice channel control circuit for, in case that a telephone set on a termination side has a voice recognition function, setting a channel to which a voice signal is sent when the above-described telephone set on a termination side is phoned from the telephone set on a dialing side to a channel for transmitting only a normalized signal in a hybrid coding method.

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 method and apparatus for providing coder independent packet replacement in the presence of frame erasures without requiring that modifications be made to either the encoder or the decoder. An input buffer management process identifies the presence of lost or missing packets and informs a playout buffer management process therof. Then, when packets have been lost, the playout buffer management process advantageously synthesizes an actual signal segment in the absence of a corresponding decoded packet.

Abstract: A method of compressing an audio signal can include accepting input samples of the audio signal wherein the input samples include non-zero input samples. A logarithm of each of the non-zero input samples of the audio signal can be calculated. Compressed output samples for each non-zero input sample can then be determined based on the logarithm of each respective non-zero input sample. Preferably, a linear relationship may exist between logarithms of the non-zero input samples and logarithms of the corresponding compressed output samples. A logarithm of each compressed output sample, corresponding to a non-zero input sample, may be based on a product of a logarithm of each corresponding non-zero input sample and a compression factor. Related devices and computer program products are also discussed.

Abstract: In an exemplary conversion scheme, a frame of a first speech signal comprising a plurality of frames encoded at a plurality of first rates, including a first non-speech rate, is received. The rate of the received frame is determined, and if the received frame is encoded at the first non-speech rate, then the received frame is re-encoded at either a second or third non-speech rate to generate a frame of a second speech signal. Moreover, a system for converting a speech signal comprises a receiver for receiving a frame of a first speech signal and a processor capable of determining the encoding rate of the received frame and re-encoding the received frame at either a second or third non-speech rate if the received frame was originally encoded at a first non-speech rate.

Abstract: Methods and an apparatus for enhancement of source coding systems utilizing high frequency reconstruction (HFR) are introduced. The problem of insufficient noise contents is addressed in a reconstructed highband, by using Adaptive Noise-floor Addition. New methods are also introduced for enhanced performance by means of limiting unwanted noise, interpolation and smoothing of envelope adjustment amplification factors. The methods and apparatus used are applicable to both speech coding and natural audio coding systems.

Abstract: A lossless audio compression scheme is adapted for use in a unified lossy and lossless audio compression scheme. In the lossless compression, the adaptation rate of an adaptive filter is varied based on transient detection, such as increasing the adaptation rate where a transient is detected. A multi-channel lossless compression uses an adaptive filter that processes samples from multiple channels in predictive coding a current sample in a current channel. The lossless compression also encodes using an adaptive filter and Golomb coding with non-power of two divisor.

Abstract: A method and apparatus for an audio scrubbing system for synchronizing audio to an asynchronous clock while preserving pitch utilizes a phase-vocoder to implement time-scaling without pitch-shifting.

Abstract: Method and apparatus for subsampling phase spectrum information by analyzing and reconstructing a prototype of a frame. The prototype is analyzed by correlating phase parameters generated from the prototype with phase parameters generated from a reference prototype in multiple frequency bands. The prototype is reconstructed using linear phase shift values by producing a set of phase parameters of the reference prototype, generating a set of linear phase shift values associated with the prototype, and composing a phase vector from the set of phase parameters and the set of linear phase shift values across multiple frequency bands. The prototype is reconstructed using circular rotation values by producing a set of circular rotation values associated with the prototype, generating a set of bandpass waveforms associated with the phase parameters of the reference prototype in multiple frequency bands, and modifying the set of bandpass waveforms based upon the circular rotation values.

Abstract: In an audio signal encoding apparatus, a first audio signal and a second audio signal are added into an addition-result signal. The first audio signal is subtracted from the second audio signal to generate a subtraction-result signal. A first difference signal is generated which represents a difference in the addition-result signal. A second difference signal is generated which represents a difference in the subtraction-result signal. A plurality of first predictors have different prediction characteristics respectively, and are responsive to the first difference signal for generating first different prediction signals for the first difference signal, respectively. A plurality of first subtracters operate for generating first prediction-error signals representing differences between the first difference signal and the first different prediction signals, respectively. A first minimum prediction-error signal representative of a smallest difference is selected from among the first prediction-error signals.

Abstract: This invention produces data packets that can vary in length and/or data compression ratio. First, an algorithm is employed to transform a data signal into fixed or variable length data packets at variable data compression ratios. If the algorithm produces fixed length data packets, the fixed length data packets are then converted to variable length data packets, which include only the valid data bytes of the fixed length data packets. Finally, the variable compression ratio, variable length data packets are provided with length codes at each end of each data packet to facilitate bidirectional searching and decompression. The transition from fixed to variable length data packets employs a buffer which stores the fixed length data packets until the fixed length data packets are converted to variable length data packets.

Abstract: The invention relates to encoding of broadband and narrowband acoustic source signals (x) such that the perceived sound quality of corresponding reconstructed signals is improved in comparison to the known solutions. An enhancement estimation unit (102), operating in serial or in parallel with the regular encoding/decoding means (101), perceptually enhances a reconstructed acoustic source signal by utilization of an enhancement spectrum (C) comprising a larger number of spectral coefficients than the number of sample values in corresponding frames of the signals carrying the basic encoded representation of the acoustic source signal. The thus extended block length of the enhancement spectrum frame provides a basis for accomplishing the desired improvement of the perceived sound quality.

Abstract: An interpolation device for judging a state of sounds of a frame at which an error or a loss has occurred in the audio data and carrying out the interpolation according to that state is constructed by an input unit for entering the audio data, a detection unit for detecting the error or the loss of each frame of the audio data, an estimation unit for estimating the interpolation information of the frame at which the error or the loss is detected, and an interpolation unit for interpolating the frame at which the error or the loss is detected, by using the interpolation information estimated for that frame by the estimation unit.

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

Abstract: A variable dimension spectral magnitude quantization apparatus and method using a predictive and mel scale binary vector is provided.

Abstract: A fast code length search method for determining the length of a code in a codebook, wherein the method is especially suited for MPEG-compliant audio encoding. A code length table is created which stores pre-calculated code lengths, including any sign bits and linear extension bits necessary, for data value pairs or quadruples. In one embodiment, two code length tables are created, one for determining the code lengths of the codes used for the ones region, and a second code length table for the big values region. When a code length determination is made, the value is simply read from the table, instead of being calculated each time.

Abstract: A method for transferring audio data and audio-related information includes generating second audio data from first audio data, transmitting second audio data and audio-related information associated with the second audio data, and receiving the second audio data and audio-related information which includes information on a sampling frequency of the first audio data.

Abstract: In this speech encoding system, the limiter circuit is input with the delay of adaptive codebook obtained for the previous subframe, and the pitch cycle search range is limited so that the delay of adaptive codebook obtained for the previous subframe is not discontinuous to the delay of adaptive codebook to be obtained for the current subframe, and the pitch cycle search range limited is output to the pitch calculation circuit. The pitch calculation circuit is input with output signal Xw(n) of the perceptual weighting circuit and the pitch cycle search range output from the limiter, calculating the pitch cycle Top, then outputting at least one pitch cycle Top to the adaptive codebook circuit.

Abstract: When a copy of music data recorded in an HDD to another electronic equipment is instructed, whether an accounting process is performed or a sound quality deteriorating process is performed is selected. When the accounting process is performed, after a predetermined accounting procedure was performed, a data copying process is performed and the data is outputted to a copy destination. When the accounting process is performed, a quality of the data is held to be almost identical to that of the original data. When the sound quality deteriorating process is selected, a data conversion is performed by a predetermined sound quality deteriorating process, the quality of the data is deteriorated, and the deteriorated data is outputted to the copy destination. In this case, the accounting is not performed. In case of the move of the data, the accounting process and the sound quality deteriorating process are not performed.

Abstract: Digital audio is transformed using a set of filters derived from the evolving states of a dynamical system (e.g., cellular automata). The ensuing transform coefficients are quantized using a psycho-acoustic model that is a function of a fidelity parameter and the distribution of the transform coefficients in critical bands within the transform space. The technique results in compression of the original audio data. Recovery of a close approximation of the original audio data is obtained via a rapid inverse transformation. An encoding method is provided for accelerating the transmission of audio data through communications networks and storing the data on a digital storage media.

Abstract: Two related voiceband compression techniques are employed in order to enable an RF telecommunications system to accommodate data signals of high speed voiceband modems and FAX machines. A High Speed Codec enables the telecommunications system to pass voiceband modem and FAX transmissions at up to 9.6 kb/s. An Ultra-High Speed Codec supports voiceband modem and FAX transmissions up to 14.4 kb/s. The High Speed Codec operates using three 16-phase RF slots or four 8-phase RF slots, and the Ultra-High Speed Codec operates using four 16-phase RF slots. Because these codecs transmit information over several RF slots which can be contiguous, the slots within RF communication channels are dynamically allocated. The Dynamic Timeslot/Bandwidth Allocation feature detects and monitors the data transmission and forms a data channel from the necessary number of slots.

Abstract: Generating a bit rate scalable audio data stream is applicable in the field of data communications, and is based on the problem of providing a process and a device which can be used in a versatile manner and which have a high degree of flexibility with respect to the available transfer rates. This problem is solved by a process for generating a bit rate scalable audio data stream having the following steps: compression of the audio data stream in a core codec (100) accompanied by determination of core parameters (102); and enhancement of the coding in at least one downstream enhancement stage (110), characterized in that the enhancement in the enhancement stage (110) is controlled by the core parameters (102).

Abstract: Transceiver circuitry 1 comprises a first portion 10,20,30,41,50,100, having a first modulation means 41 operating at a first order of modulation, for transmitting and receiving voice signals; a second portion 20,30,42,50,100, having a second modulation means 42 operating at a second order of modulation, for transmitting and receiving digital signals at a higher data rate than is achievable by the first portion; and a data conversion means 20,30,100 operable to convert from or into voice signals intended for processing by the first portion into or from digital signals for processing by the second portion.