Abstract: An audio processing device is described comprising an energy distribution determiner configured to determine an energy distribution of a sound and an acoustical environment determiner configured to determine based on the energy distribution whether the sound includes a sound caused by the acoustical environment.

Abstract: An audio processing device is described comprising an energy distribution determiner configured to determine an energy distribution of a sound and an acoustical environment determiner configured to determine based on the energy distribution whether the sound includes a sound caused by the acoustical environment.

Abstract: A method, system, and computer program product for managing noise in a noise reduction system, comprising: receiving a first signal at a first microphone; receiving a second signal at a second microphone; identifying noise estimation in the first signal and the second signal; identifying a transfer function of the noise reduction system using a ratio of a power spectral density of the second signal minus the noise estimation to a power spectral density of the first signal, wherein the noise estimation is removed from only the power spectral density of the second signal; and identifying a gain of the noise reduction system using the transfer function.

Abstract: The invention relates to a method for outputting a speech signal. Speech signal frames are received and are used in a predetermined sequence in order to produce a speech signal to be output. If one speech signal frame to be received is not received, then a substitute speech signal frame is used in its place, which is produced as a function of a previously received speech signal frame. According to the invention, in the situation in which the previously received speech signal frame has a voiceless speech signal, the substitute speech signal frame is produced by means of a noise 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: In a method for embedding steganographic information into the signal information of a signal encoder, a solution is to be created, which enables steganographic information being embedded into the signal information of a signal encoder such that a reduction of the voice quality is largely avoided.

Abstract: A method, system, and computer program product for managing noise in a noise reduction system, comprising: receiving a first signal at a first microphone; receiving a second signal at a second microphone; identifying noise estimation in the first signal and the second signal; identifying a transfer function of the noise reduction system using a ratio of a power spectral density of the second signal minus the noise estimation to a power spectral density of the first signal, wherein the noise estimation is removed from only the power spectral density of the second signal; and identifying a gain of the noise reduction system using the transfer function.

Abstract: The invention pertains to a method and apparatus of efficient encoding and decoding of vector quantized data. The method and system explores and implements sub-division of a quantization vector space comprising class-leader vectors and representation of the class-leader vectors by a set of class-leader root-vectors facilitating faster encoding and decoding, and reduced storage requirements.

Abstract: A more efficient method reduces reverberation in binaural hearing systems. This has been done by developing a method for obtaining a reduced-reverberation, binaural output signal, for a binaural hearing apparatus. First of all, a left input signal and a right input signal are provided. The two input signals are combined to form a reference signal. The reference signal is used to ascertain spectral weights, or these weights are provided in another way, in order to use them to reduce late reverberation. To this end, the two input signals have the spectral weight applied to them. Furthermore, a coherency for signal components of the weighted input signals is ascertained. Non-coherent signal components of both weighted input signals are then attenuated in order to reduce early reverberation.

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: The invention relates to a method for outputting a speech signal. Speech signal frames are received and are used in a predetermined sequence in order to produce a speech signal to be output. If one speech signal frame to be received is not received, then a substitute speech signal frame is used in its place, which is produced as a function of a previously received speech signal frame. According to the invention, in the situation in which the previously received speech signal frame has a voiceless speech signal, the substitute speech signal frame is produced by means of a noise signal.

Abstract: The invention pertains to a method and apparatus of efficient encoding and decoding of vector quantized data. The method and system explores and implements sub-division of a quantization vector space comprising class-leader vectors and representation of the class-leader vectors by a set of class-leader root-vectors facilitating faster encoding and decoding, and reduced storage requirements.

Abstract: In a method for embedding steganographic information into the signal information of a signal encoder, a solution is to be created, which enables steganographic information being embedded into the signal information of a signal encoder such that a reduction of the voice quality is largely avoided.

Abstract: A wideband audio coding concept is presented that provides good audio quality at bit rates below 3 bits per sample with an algorithmic delay of less than 10 ms. The concept is based on the principle of Linear Predictive Coding (LPC) in an analysis-by-synthesis framework. A spherical codebook is used for quantisation at bit rates which are higher in comparison to low bit rate speech coding for improved performance for audio signals. For superior audio quality, noise shaping is employed to mask the coding noise. In order to reduce the computational complexity of the encoder, the analysis-by synthesis framework has been adapted for the spherical codebook to enable a very efficient excitation vector search procedure. Furthermore, auxiliary information gathered in advance is employed to reduce a computational encoding and decoding complexity at run time significantly. This auxiliary information can be considered as the SCELP codebook.

Abstract: In one aspect a coding of stereophonic audio signals based on inter-channel linear prediction is provided. Each of the two channels is predicted by filtering the center stereo image of both the channels. Optimal filter coefficients are calculated for both channels is a generalization of Mid/Side and Left/Right joint-stereo coding.

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 wideband audio coding concept is presented that provides good audio quality at bit rates below 3 bits per sample with an algorithmic delay of less than 10 ms. The concept is based on the principle of Linear Predictive Coding (LPC) in an analysis-by-synthesis framework. A spherical codebook is used for quantisation at bit rates which are higher in comparison to low bit rate speech coding for improved performance for audio signals. For superior audio quality, noise shaping is employed to mask the coding noise. In order to reduce the computational complexity of the encoder, the analysis-by synthesis framework has been adapted for the spherical codebook to enable a very efficient excitation vector search procedure. Furthermore, auxiliary information gathered in advance is employed to reduce a computational encoding and decoding complexity at run time significantly. This auxiliary information can be considered as the SCELP codebook.

Abstract: A method of encoding a signal, particularly a message signal for speech and/or image transmission, in which signal-matched encoding of the signal with M bits is performed using Q quantization levels, with Q<2M, and in which at least part of the remaining bit rate or the entire redundancy of M−Id(Q) bits is used to protect the signal against errors. The error protection uses a nonlinear code, for example. Furthermore, additional, signal-matched code redundancy can be added to an arbitrary source-coded signal. To decode the redundant signals in both cases, parameter estimation is performed at the receiving end using at least part of the redundancy.

Abstract: With a method and a device for error masking in digital transmission systems whereby a distribution function for estimating transmitted parameters is determined at the receiving end, the method provides that a distribution function is adjusted around the output signal value of a predictor and is integrated by sections into a new distribution function; this new distribution function is multiplied by a distribution function which takes the received quality into consideration and the result of an a posteriori distribution which can be used with conventional estimation methods for the final parameter estimation.