Abstract: Signals of different channels (C1-CN) are combined into one mono signal (x). A set of adaptive filters, preferably one for each channel (C1-CN), is derived in a respective filter adaptation unit (30:1-30:N). When an adaptive filter is applied to the mono signal (x) it reconstructs the signal of the respective channel (C1-CN) under a perceptual constraint. The perceptual constraint is a gain and/or shape constraint. The gain constraint allows the preservation of the relative energy between the channels (C1-CN) while the shape constraint allows more stability by avoiding unnecessary filtering of spectral nulls. The transmitted parameters are the mono signal (x), in encoded form, and the parameters (p1-pN) of the adaptive filters, preferably also encoded. The receiver reconstructs the signal of the different channels by applying the adaptive filters and possibly some additional post-processing.

Abstract: Polyphonic signals are used to create a main signal, typically a mono signal, and a side signal. A number of encoding schemes for the side signal are provided. Each encoding scheme is characterized by a set of sub-frames of different lengths. The total length of the sub-frames corresponds to the length of the encoding frame of the encoding scheme. The encoding scheme to be used on the side signal is selected dependent on the present signal content of the polyphonic signals. In a preferred embodiment, a side residual signal is created as the difference between the side signal and the main signal scaled with a balance factor. The balance factor is selected to minimize the side residual signal. The optimized side residual signal and the balance factor are encoded and provided as encoding parameters representing the side signal.

Abstract: In a radio communication system in a mobile to mobile call with one good (AMR102) and bad (AMR515) radio link, the good radio (AMR102) link is forced by the poor (AMR515) link to use a more robust AMR coded mode (AMR515) and thereby using excessive power (232). A capacity loss in such system is avoided by adjusting the power level (212) for the connection with the good link (AMR102).

Abstract: In adapting the modes of a codec for a communication channel in a mobile telecommunication system, a delay unit is provided to delay the adaptation if communication conditions of the channel are improving, so that a less robust mode having a higher intrinsic quality is selected only after some delay. Otherwise the adapting is not delayed. This may to some extent reduce the quality of the received speech signal, abut the out-of-phase problems due to AMR loop delay in speech codecs are minimized. In particular in speech codecs built according to the AMR standard. The quality of the communication is measured to give quality values for successive measurement occasions. The quality of the communication is measured to give quality values for successive measurement occasions. The quality values are first linearly filtered.

Abstract: In the present invention, the available delay budget for a full-duplex communication is distributed between the links depending on their activity, present and near history. By keeping track on if a link in a full-duplex communication system is active or inactive, i.e. knowing the information value of the transferred data, coding procedures having larger coding delays than normally accepted can be used for active links, if the opposite link simultaneously is inactive. Since the user sensibility for delays is as largest at the moment a link becomes active, coding procedures having smaller coding delays than normally used are assigned at the moment a link becomes active. The coding delay is subsequently increased. Preferably, the round-trip delay is controlled to be kept smaller than a requested maximum value.

Abstract: Techniques for discontinuous transmission (DTX) and fast in-band signaling of configuration changes and protocol messages in speech communications systems provide cost efficiency in terms of radio transmission capacity, in terms of fixed line transmission, and in terms of implementation effort. An exemplary method for performing discontinuous transmission (DTX) in a communications system in which source data is interleaved for transmission from a first component in the system to a second component in the system includes the steps of detecting periods of source data inactivity, and transmitting silence descriptor (SID) frames from the first to the second component during the periods of source data inactivity, certain of the transmitted SID frames being interleaved using a different interleaving algorithm as compared to that used for source data. For example, the source data can be block diagonally interleaved, and certain of the SID frames can be block interleaved.

Abstract: The present invention relates to the concealment of errors in decoded acoustic signals caused by encoded data representing the acoustic signals being partially lost or damaged during transmission over a transmission medium. In case of lost data or received damaged data a secondary reconstructed signal is produced on basis of a primary reconstructed signal. This signal has a spectrally adjusted spectrum (Z4E), such that it deviates less with respect spectral shape from a spectrum (Z3) of a previously reconstructed signal produced from previously received data than a spectrum (Z′4) of the primary reconstructed signal.

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: 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: The invention relates to a method and a device for avoiding interruptions in voice transmission in a cellular communication system. Voice data are thereby divided into segments, and the segments are associated with transmission quanta, so-called bursts. A number of segments is coded in a first transmission mode and is made available for transmission. Said first transmission mode is, for instance, for a transmission at full rate. Additional segments, which follow, are subsequently coded in a second transmission mode and are made available for transmission. The second transmission mode is, for instance, for transmission at half rate. Through the change from a first transmission mode to a second transmission mode, parts of the associated transmission quanta remain unused by applying the so-called interleaving. Said unused transmission quanta are used for performing additional functions, such as initiating a hand-over or FACCH signaling.

Abstract: A communication system supports multiple source coding/channel coding schemes. A mode indicator can be transmitted with payload data to inform the receiver of the particular scheme currently being employed. Similarly, a mode request can be transmitted to inform the receiver of a particular scheme to be employed for information to be transmitted on the return radio link. The rate of change of the mode indications and mode requests can be decimated so that they need not both be transmitted in each frame to thereby reduce the overhead capacity consumption associated with the transmission of mode information. According to one exemplary embodiment, the mode indication and the mode request can be transmitted independently on alternating frames.

Abstract: The present invention relates to the concealment of errors in decoded acoustic signals caused by encoded data representing the acoustic signals being partially lost or damaged during transmission over a transmission medium. In case of lost data or received damaged data a secondary reconstructed signal is produced on basis of a primary reconstructed signal. This signal has a spectrally adjusted spectrum (Z4E), such that it deviates less with respect spectral shape from a spectrum (Z3) of a previously reconstructed signal produced from previously received data than a spectrum (Z′4) of the primary reconstructed signal.

Abstract: The present invention relates to encoding of an acoustic source signal (x) such that a corresponding signal ({circumflex over (z)}) reconstructed on basis of the encoded information has a perceptually high sound quality. The acoustic source signal (x) is encoded into at least one basic coded signal (P) by a coding arrangement in a transmitter (300). The at least one basic coded signal (P) thus represents perceptually significant characteristics of the acoustic signal (x). The coding arrangement in the transmitter (300) may include at least one spectral smoothing unit (305a), which receives at least one of the signal components on which the basic coded signal P is based and generates in response thereto a corresponding smoothed signal component. At least one enhanced coded signal P(E) is then produced from i.a. the corresponding smoothed signal component.

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

Abstract: A communication system supports multiple source coding/channel coding schemes. A mode indicator can be transmitted with payload data to inform the receiver of the particular scheme currently being employed. The mode indicator may be encoded using a relatively weak channel coding to minimize extraneous overhead transmissions. To enhance the likelihood of successfully decoding the mode indicator, which has information that is highly important in successfully decoding the payload data, various value likelihood parameters can be calculated and combined to identify a most likely mode. Included in the likelihood parameters is one parameter which can be calculated based on a model of the mode information's likely value.

Abstract: A process for preparing guanine, in which 2,4-diamino-5-formylamino-6-hydroxypyrimidine (DAFHP) in formic acid is reacted at superatmospheric pressure, optionally in the absence of formamide and/or in the presence of catalytic amounts of an additive which acts as a reducing agent.

Abstract: A call-setup method in a digital cellular radio communication system determines a plurality of parameters defining the current state of the system. By evaluating and combining these parameters into a communication resource request, a communication resource that best matches the request may be allocated in order to optimize system resource usage and/or service quality.

Abstract: A power control system in a TDMA radio communication system has traffic channels which are associated with a set of speech/channel encoding modes. Each mode has a different mix of speech encoder bit rate and data protection bit rate but the same total available gross bit rate. The transmitter includes a power control unit that replaces a mode allocated to a channel by another mode having either a higher or a lower data protection bit rate and either a lower or a higher speech encoder bit rate if the sound to be encoded requires either a lower or a higher speech encoding bit rate. The power control unit also controls a power adjustment unit that either reduces or increases the output power of the transmitter to a lower or a higher level such that an estimated decoded speech quality measure at the receiver is substantially constant.