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: The invention relates to a method for the scalable improvement of the quality of an encoding method according to IT-U Recommendation G.722, including the following steps: —a digital error signal (E) derived from an input signal to be encoded and a prognosis signal is compared in sections to a number of M*LN different reference signals in an iterative process having a number of repeated steps depending on the scope of the expansion, and the reference signal having a minimum error signal of a prescribed error criteria is derived therefrom, —the reference signals are each made up of equidistant Dirac impulses ?(n) according to (I), wherein off=[0 . . . M?1], indicates the distance of the first impulse from a zero time point, ??{?, ?, . . . , ?} indicates the amplitude value, M the distance between the individual pulses, N the number of pulses, and L the number of different levels, —the information about the reference signal having the minimum error signal is transmitted.

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: In one aspect, a noise suppression process for a decoded signal comprising a first decoded signal portion and a second decoded signal portion is provided. A first energy envelope generating curve and a second energy envelope generating curve of the first signal portion and of the second decoded signal portion are determined. An identification number depending on a comparison of the first and second energy envelope generating curves is formed. An amplification factor which depends on the identification number is derived. Multiplying the second decoded signal portion by the amplification factor, reduces pre-echo and post-echo interference noises.

Abstract: The invention relates to a method for the vector quantization of a feature vector, in particular with respect to a data compression of a signal to be transmitted or to be stored, particularly a voice signal or a video signal, wherein at least one codebook from a plurality of codebook vectors is searched for a code vector representing the feature vector. During the search, a sequence of codebook vectors is examined for the suitability thereof to represent the feature vector. In the course of the search for the code vector, a set of neighboring vectors is dedicated to at least one of the codebook vectors potentially to be examined, preferably prior to the search. The search for the code vector includes at least part of the neighboring vectors.

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: The invention relates to a method for the vector quantization of a feature vector, in particular with respect to a data compression of a signal to be transmitted or to be stored, particularly a voice signal or a video signal, wherein at least one codebook from a plurality of codebook vectors is searched for a code vector representing the feature vector. During the search, a sequence of codebook vectors is examined for the suitability thereof to represent the feature vector. In the course of the search for the code vector, a set of neighboring vectors is dedicated to at least one of the codebook vectors potentially to be examined, preferably prior to the search. The search for the code vector includes at least part of the neighboring vectors.

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: The invention relates to a method for the scalable improvement of the quality of an encoding method according to IT-U Recommendation G.722, including the following steps:-a digital error signal (E) derived from an input signal to be encoded and a prognosis signal is compared in sections to a number of M*LN different reference signals in an iterative process having a number of repeated steps depending on the scope of the expansion, and the reference signal having a minimum error signal of a prescribed error criteria is derived therefrom,-the reference signals are each made up of equidistant Dirac impulses ?(n) according to (I), wherein off=[0 . . . M?1], indicates the distance of the first impulse from a zero time point, ??{?,?, . . . ,?} indicates the amplitude value, M the distance between the individual pulses, N the number of pulses, and L the number of different levels,-the information about the reference signal having the minimum error signal is transmitted.

Abstract: An analog signal divided into time frames is encoded and a synthetic signal is formed on the model thereof in a time frame manner via a synthesis filter which is excited by an excitation signal. The excitation signal is formed by at least one adaptive code list containing a plurality of scanning values provided with a defined scanning space. For the actual excitation signal, a segment corresponding to the time frame length is selected from the plurality of scanning values via a speech-based frequency parameter which can take non-integer values and, in such a case, the values intermediate to the scanning values defined by the speech-based frequency parameter are formed in such a way that the time space between the intermediate values and the scanning values is reduced and the totality of the intermediate and the scanning values is used for forming the excitation signal.

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: An analog signal divided into time frames is encoded and a synthetic signal is formed on the model thereof in a time frame manner via a synthesis filter which is excited by an excitation signal. The excitation signal is formed by at least one adaptive code list containing a plurality of scanning values provided with a defined scanning space. For the actual excitation signal, a segment corresponding to the time frame length is selected from the plurality of scanning values via a speech-based frequency parameter which can take non-integer values and, in such a case, the values intermediate to the scanning values defined by the speech-based frequency parameter are formed in such a way that the time space between the intermediate values and the scanning values is reduced and the totality of the intermediate and the scanning values is used for forming the excitation signal.

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: 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 one aspect, a noise suppression process for a decoded signal comprising a first decoded signal portion and a second decoded signal portion is provided. A first energy envelope generating curve and a second energy envelope generating curve of the first signal portion and of the second decoded signal portion are determined. An identification number depending on a comparison of the first and second energy envelope generating curves is formed. An amplification factor which depends on the identification number is derived. Multiplying the second decoded signal portion by the amplification factor, reduces pre-echo and post-echo interference noises.