Abstract: The invention relates to a method and means for encoding background noise information during voice signal encoding methods. A basic idea of the invention is to provide the scalability known for transmitting voice information in a similar manner when forming an SID frame. The invention provides encoding of a narrowband first component and of a broadband second component of a piece of background noise information and formation of an SID frame which describes the background noise with separate areas for the first and second components.

Abstract: The inventive method provides for an encoder in a voice codec to be designed such that after a particular idle time (“Idle Period”) it recalculates the averaged energy and the autocorrelation function. Administrative points in the network inform the encoder about the idle time which has been set in the transmission network.

Abstract: In an embodiment, a method of receiving a digital audio signal, using a processor, includes correcting the digital audio signal from lost data. Correcting includes copying frequency domain coefficients of the digital audio signal from a previous frame, adaptively adding random noise coefficients to the copied frequency domain coefficients, and scaling the random noise coefficients and the copied frequency domain coefficients to form recovered frequency domain coefficients. Scaling is controlled with a parameter representing a periodicity or harmonicity of the digital audio signal. A corrected audio signal is produced from the recovered frequency domain coefficients.

Abstract: To form an audio signal, frequency components of the audio signal which are allotted to a first subband are formed by means of a subband decoder using supplied fundamental period values which respectively indicate a fundamental period for the audio signal. Frequency components of the audio signal which are allotted to a second subband are formed by exciting an audio synthesis filter using an excitation signal which is specific to the second subband. To produce this excitation signal, an excitation signal generator derives a fundamental period parameter from the fundamental period values. The fundamental period parameter is used by the excitation signal generator to form pulses with a pulse shape which is dependent on the fundamental period parameter at an interval of time which is determined by the fundamental period parameter and to mix them with a noise signal.

Abstract: A method for the artificial extension of the bandwidth of speech signals involves: a) Provision of a wideband input speech signal (swbi(k)); b) Determination of the signal components (seb(k)) of the wideband input speech signal (swbi(k)) required for the bandwidth extension from an extension band from the wideband input speech signal (swbi(k)); c) Determination of the temporal envelopes of the signal components (seb(k)) determined for the bandwidth extension; d) Determination of the spectral envelopes of the signal components (seb(k)) determined for bandwidth extension; e) Encoding of the information for the temporal envelopes and the spectral envelopes, and provision of the encoded information by carrying out the extension of the bandwidth; f) Decoding of the encoded information and generation of the temporal envelopes and the spectral envelopes from the encoded information for the production of a bandwidth-extended output speech signal (swbo(k)).

Abstract: A basic idea of the invention is to ascertain information on the course of the bit rate switching during an active speech phase. According to the invention, during the speech phase, information on the percentage proportion of broadband active speech frames in comparison to narrowband active speech frames is compiled on the part of the decoder. A high percentage proportion of broadband active speech frames indicates that a broadband use is preferred on the part of the codec and therefore a need exists for synthesizing noise information in broadband form during a DTX phase.

Abstract: According to the invention, an excitation signal is generated as a result of sampled excitation values in order to excite an audio synthesis filter, the generated sampled excitation values being continuously stored in an adaptive codebook. A noise generator is provided which continuously generates random sampled values. A sequence of the stored sampled excitation values is selected from the adaptive codebook based on a fed audio fundamental frequency parameter by means of which a time gap between the sequence that is to be selected and the actual time reference is predefined. The excitation signal is generated by mixing the selected sequence with a random sequence encompassing actual random sampled valued of the noise generator.

Abstract: In on aspect, user data is decoded which has been coded as base data and refining data, wherein the base data is decoded using a first delay and the refining data is decoded using a second delay which is longer than the first delay. Furthermore, a check establishes whether the refining data is available in accordance with a time parameter. If the check result is positive, the decoded base data is additionally delayed, the additionally delayed decoded base data is mixed with the decoded refining data, and resulting mixed data is output as user data. If the check result is negative, however, the decoded base data is output as user data.

Abstract: A basic idea of the invention is to ascertain information on the course of the bit rate switching during an active speech phase. According to the invention, during the speech phase, information on the percentage proportion of broadband active speech frames in comparison to narrowband active speech frames is compiled on the part of the decoder. A high percentage proportion of broadband active speech frames indicates that a broadband use is preferred on the part of the codec and therefore a need exists for synthesizing noise information in broadband form during a DTX phase.

Abstract: The inventive method provides for an encoder in a voice codec to be designed such that after a particular idle time (“Idle Period”) it recalculates the averaged energy and the autocorrelation function. Administrative points in the network inform the encoder about the idle time which has been set in the transmission network.

Abstract: The invention relates to a method and means for encoding background noise information during voice signal encoding methods. A basic idea of the invention is to provide the scalability known for transmitting voice information in a similar manner when forming an SID frame. The invention provides encoding of a narrowband first component and of a broadband second component of a piece of background noise information and formation of an SID frame which describes the background noise with separate areas for the first and second components.

Abstract: In an embodiment, a method of receiving a digital audio signal, using a processor, includes correcting the digital audio signal from lost data. Correcting includes copying frequency domain coefficients of the digital audio signal from a previous frame, adaptively adding random noise coefficients to the copied frequency domain coefficients, and scaling the random noise coefficients and the copied frequency domain coefficients to form recovered frequency domain coefficients. Scaling is controlled with a parameter representing a periodicity or harmonicity of the digital audio signal. A corrected audio signal is produced from the recovered frequency domain coefficients.

Abstract: To form an audio signal, frequency components of the audio signal which are allotted to a first subband are formed by means of a subband decoder using supplied fundamental period values which respectively indicate a fundamental period for the audio signal. Frequency components of the audio signal which are allotted to a second subband are formed by exciting an audio synthesis filter using an excitation signal which is specific to the second subband. To produce this excitation signal, an excitation signal generator derives a fundamental period parameter from the fundamental period values. The fundamental period parameter is used by the excitation signal generator to form pulses with a pulse shape which is dependent on the fundamental period parameter at an interval of time which is determined by the fundamental period parameter and to mix them with a noise signal.

Abstract: A method of encoding a speech signal for transmission in a communications network involves transforming the signal into a sequence of frames, each frame including a plurality of coefficients; dividing the frame into a set of sub-bands each containing a sub-set of the plurality of coefficients; applying an optimization function to calculate respective test values corresponding to respective candidate sets of pulses representing a coded form of at least some of the coefficients; and selecting a set of pulses having a test value which meets a selectability criterion. If the optimisation function is an error function, the selectability criterion is minimization of the function. If the optimization function is an iterative function, the selectability criterion is selecting an iteration in which a certain condition is reached.

Abstract: According to the invention, an excitation signal is generated as a result of sampled excitation values in order to excite an audio synthesis filter, the generated sampled excitation values being continuously stored in an adaptive codebook. A noise generator is provided which continuously generates random sampled values. A sequence of the stored sampled excitation values is selected from the adaptive codebook based on a fed audio fundamental frequency parameter by means of which a time gap between the sequence that is to be selected and the actual time reference is predefined. The excitation signal is generated by mixing the selected sequence with a random sequence encompassing actual random sampled valued of the noise generator.

Abstract: A method for the artificial extension of the bandwidth of speech signals involves: a) Provision of a wideband input speech signal (swbi(k)); b) Determination of the signal components (seb(k)) of the wideband input speech signal (swbi(k)) required for the bandwidth extension from an extension band from the wideband input speech signal (swbi(k)); c) Determination of the temporal envelopes of the signal components (seb(k)) determined for the bandwidth extension; d) Determination of the spectral envelopes of the signal components (seb(k)) determined for bandwidth extension; e) Encoding of the information for the temporal envelopes and the spectral envelopes, and provision of the encoded information by carrying out the extension of the bandwidth; f) Decoding of the encoded information and generation of the temporal envelopes and the spectral envelopes from the encoded information for the production of a bandwidth-extended output speech signal (swbo(k)).

Abstract: In on aspect, user data is decoded which has been coded as base data and refining data, wherein the base data is decoded using a first delay and the refining data is decoded using a second delay which is longer than the first delay. Furthermore, a check establishes whether the refining data is available in accordance with a time parameter. If the check result is positive, the decoded base data is additionally delayed, the additionally delayed decoded base data is mixed with the decoded refining data, and resulting mixed data is output as user data. If the check result is negative, however, the decoded base data is output as user data.

Abstract: Disclosed is a method for suppressing (10) echo (z(t)) in uplink data (y(t), 12-16) originating from a terminal (2, 3), said method allowing a delay and a decrease in quality during echo suppression of uplink data originating from a wireless terminal (2, 3). According to the inventive method, original or copied downlink data (12-16) and uplink data (19-21) are analyzed in order to prepare echo suppression (10), and uplink data (19-21) is modified in the compressed state in order to reduce echo (10) by using the results of said analysis (9) of the downlink data (12-16) and uplink data (19-21).

Abstract: The invention relates to a method for encoding voice signals, especially so-called voice onset sections. By establishing the first amplification factor, the data quantity for representing the whole of the first or adaptive amplification factor and adaptive code book entry is reduced, whereby other parameters which occur during the voice encoding can be represented in a more precise manner. The invention also relates to a method for transmitting voice signals which are encoded in such a way.

Abstract: Source-coded information from a multi-mode source encoder is channel coded for transmission in a communication system. A multi-mode channel encoder associates a different channel coding error protection profile with each of the modes of the multi-mode source encoder. The channel encoder determines a mode used by the multi-mode source encoder to source code a given frame or other designated portion of the information, and channel codes the source-coded portion of the information utilizing an error protection profile identified at least in part based on the determined mode. A multi-mode channel decoder is operative to hypothesize that a particular one of the modes of the multi-mode source encoder was used to source code the designated portion of the information.