Apparatus and method for error concealment in low-delay unified speech and audio coding

An apparatus for generating spectral replacement values for an audio signal has a buffer unit for storing previous spectral values relating to a previously received error-free audio frame. Moreover, the apparatus includes a concealment frame generator for generating the spectral replacement values, when a current audio frame has not been received or is erroneous. The previously received error-free audio frame has filter information, the filter information having associated a filter stability value indicating a stability of a prediction filter. The concealment frame generator is adapted to generate the spectral replacement values based on the previous spectral values and based on the filter stability value.

NOISE GENERATION IN AUDIO CODECS

The spectral domain is efficiently used in order to parameterize the background noise thereby yielding a background noise synthesis which is more realistic and thus leads to a more transparent active to inactive phase switching. 1. An audio encoder comprisinga background noise estimator configured to determine a parametric background noise estimate based on a spectral decomposition representation of an input audio signal so that the parametric background noise estimate spectrally describes a spectral envelope of a background noise of the input audio signal;an encoder for encoding the input audio signal into a data stream during the active phase; anda detector configured to detect an entrance of an inactive phase following the active phase based on the input signal,wherein the audio encoder is configured to encode into the data stream the parametric background noise estimate in the inactive phase,whereinthe background noise estimator is configured to identify local minima in the spectral decomposition representation of the input audio signal and to estimate the spectral envelope of the background noise of the input audio signal using interpolation between the identified local minima as supporting points, orthe encoder is configured to, in encoding the input audio signal, predictively code the input audio signal into linear prediction coefficients and an excitation signal, and transform code a spectral decomposition of the excitation signal, and code the linear prediction coefficients into the data stream, wherein the background noise estimator is configured to use the spectral decomposition of the excitation signal as the spectral decomposition representation of the input audio signal in determining the parametric background noise estimate.2. The audio encoder according to claim 1 , wherein the background noise estimator is configured to perform the determining the parametric background noise estimate in the active phase with distinguishing between a noise component ...

APPARATUS AND METHOD FOR AUDIO ENCODING AND DECODING EMPLOYING SINUSOIDAL SUBSTITUTION

An apparatus for generating an audio output signal based on an encoded audio signal spectrum has a processing unit, a pseudo coefficients determiner, a spectrum modification unit, a spectrum-time conversion unit, a controllable oscillator and a mixer. The pseudo coefficients determiner is configured to determine pseudo coefficients of the decoded audio signal spectrum. The spectrum modification unit is configured to set the pseudo coefficients to a predefined value to acquire a modified audio signal spectrum. The spectrum-time conversion unit is configured to convert the modified audio signal spectrum to a time-domain. The controllable oscillator is configured to generate a time-domain oscillator signal and is controlled by the spectral location and the spectral value of at least one of the pseudo coefficients. The mixer is configured to mix the time-domain conversion signal and the time-domain oscillator signal. 1. An apparatus for generating an audio output signal based on an encoded audio signal spectrum , wherein the apparatus comprises:a processing unit for processing the encoded audio signal spectrum to acquire a decoded audio signal spectrum the decoded audio signal spectrum comprising a plurality of spectral coefficients, wherein each of the spectral coefficients comprises a spectral location within the encoded audio signal spectrum and a spectral value, wherein the spectral coefficients are sequentially ordered according to their spectral location within the encoded audio signal spectrum so that the spectral coefficients form a sequence of spectral coefficients,a pseudo coefficients determiner for determining one or more pseudo coefficients of the decoded audio signal spectrum, each of the pseudo coefficients comprising a spectral location and a spectral value,a spectrum modification unit for setting the one or more pseudo coefficients to a predefined value to acquire a modified audio signal spectrum,a spectrum-time conversion unit for converting the ...

Apparatus, Method or Computer Program for estimating an inter-channel time difference

An apparatus for estimating an inter-channel time difference between a first channel signal and a second channel signal, includes a signal analyzer for estimating a signal characteristic of the first channel signal or the second channel signal or both signals or a signal derived from the first channel signal or the second channel signal; a calculator for calculating a cross-correlation spectrum for a time block from the first channel signal in the time block and the second channel signal in the time block; a weighter for weighting a smoothed or non-smoothed cross-correlation spectrum to obtain a weighted cross correlation spectrum using a first weighting procedure or using a second weighting procedure depending on a signal characteristic estimated by the signal analyzer, wherein the first weighting procedure is different from the second weighting procedure; and a processor for processing the weighted cross-correlation spectrum to obtain the inter-channel time difference. 1. An apparatus for estimating an inter-channel time difference between a first channel signal and a second channel signal , comprising:a signal analyzer for estimating a signal characteristic of the first channel signal or the second channel signal or both signals or a signal derived from the first channel signal or the second channel signal;a calculator for calculating a cross-correlation spectrum for a time block from the first channel signal in the time block and the second channel signal in the time block;a weighter for weighting a smoothed or non-smoothed cross-correlation spectrum to acquire a weighted cross correlation spectrum using a first weighting procedure or using a second weighting procedure depending on a signal characteristic estimated by the signal analyzer, wherein the first weighting procedure is different from the second weighting procedure; anda processor for processing the weighted cross-correlation spectrum to acquire the inter-channel time difference.2. The apparatus of ...

COMFORT NOISE ADDITION FOR MODELING BACKGROUND NOISE AT LOW BIT-RATES

The invention provides a decoder being configured for processing an encoded audio bitstream, wherein the decoder includes: a bitstream decoder configured to derive a decoded audio signal from the bitstream, wherein the decoded audio signal includes at least one decoded frame; a noise estimation device configured to produce a noise estimation signal containing an estimation of the level and/or the spectral shape of a noise in the decoded audio signal; a comfort noise generating device configured to derive a comfort noise signal from the noise estimation signal; and a combiner configured to combine the decoded frame of the decoded audio signal and the comfort noise signal in order to obtain an audio output signal. 1. A decoder being configured for processing an encoded audio bitstream , wherein the decoder comprises:a bitstream decoder configured to derive a decoded audio signal from the bitstream, wherein the decoded audio signal comprises at least one decoded frame;a noise estimation device configured to produce a noise estimation signal comprising an estimation of the level and/or the spectral shape of a noise in the decoded audio signal;a comfort noise generating device configured to derive a comfort noise signal from the noise estimation signal; anda combiner configured to combine the decoded frame of the decoded audio signal and the comfort noise signal in order to acquire an audio output signal.2. The decoder according to claim 1 , wherein the decoded frame is an active frame.3. The decoder according to claim 1 , wherein the decoded frame is an active frame.4. The decoder according to claim 1 , wherein the noise estimating device comprises a spectral analysis device configured to create an analysis signal comprising the level and the spectral shape of the noise in the decoded audio signal and a noise estimation producing device configured to produce the noise estimation signal based on the analysis signal.5. The decoder according to claim 1 , wherein the ...

LOW-COMPLEXITY TONALITY-ADAPTIVE AUDIO SIGNAL QUANTIZATION

The invention provides an audio encoder for encoding an audio signal so as to produce therefrom an encoded signal, the audio encoder including: a framing device configured to extract frames from the audio signal; a quantizer configured to map spectral lines of a spectrum signal derived from the frame of the audio signal to quantization indices, wherein the quantizer has a dead-zone, in which the input spectral lines are mapped to quantization index zero; and a control device configured to modify the dead-zone; wherein the control device includes a tonality calculating device configured to calculate at least one tonality indicating value for at least one spectrum line or for at least one group of spectral lines, wherein the control device is configured to modify the dead-zone for the at least one spectrum line or the at least one group of spectrum lines depending on the respective tonality indicating value. 1. Audio encoder for encoding an audio signal so as to produce therefrom an encoded signal , the audio encoder comprising:a framing device configured to extract frames from the audio signal;a quantizer configured to map spectral lines of a spectrum signal derived from the frame of the audio signal to quantization indices, wherein the quantizer comprises a dead-zone, in which the spectral lines are mapped to quantization index zero; anda control device configured to modify the dead-zone;wherein the control device comprises a tonality calculating device configured to calculate at least one tonality indicating value for at least one spectrum line or for at least one group of spectral lines,wherein the control device is configured to modify the dead-zone for the at least one spectrum line or the at least one group of spectrum lines depending on the respective tonality indicating value.2. Audio encoder according to claim 1 , wherein the control device is configured to modify the dead-zone in such way that the dead-zone at one of the spectral lines is larger than the ...

APPARATUS AND METHOD FOR COMFORT NOISE GENERATION MODE SELECTION

An apparatus for encoding audio information is provided. The apparatus for encoding audio information includes a selector for selecting a comfort noise generation mode from two or more comfort noise generation modes depending on a background noise characteristic of an audio input signal, and an encoding unit for encoding the audio information, wherein the audio information includes mode information indicating the selected comfort noise generation mode. 1. An apparatus for encoding audio information , comprising:a selector for selecting a comfort noise generation mode from two or more comfort noise generation modes depending on a background noise characteristic of an audio input signal, andan encoding unit for encoding the audio information, wherein the audio information comprises mode information indicating the selected comfort noise generation mode,wherein a first one of the two or more comfort noise generation modes is a frequency-domain comfort noise generation mode, and wherein the frequency-domain comfort noise generation mode indicates that the comfort noise shall be generated in a frequency domain and that the comfort noise being generated in the frequency domain shall be frequency-to-time converted.2. The apparatus according to claim 1 ,wherein the selector is configured to determine a tilt of a background noise of the audio input signal as the background noise characteristic, andwherein the selector is configured to select said comfort noise generation mode from two or more comfort noise generation modes depending on the determined tilt.3. The apparatus according to claim 2 ,wherein the apparatus further comprises a noise estimator for estimating a per-band estimate of the background noise for each of a plurality of frequency bands, andwherein the selector is configured to determine the tilt depending on the estimated background noise of the plurality of frequency bands.4. The apparatus according to claim 3 ,wherein, the noise estimator is configured to ...

AUDIO ENCODER AND DECODER USING A FREQUENCY DOMAIN PROCESSOR , A TIME DOMAIN PROCESSOR, AND A CROSS PROCESSING FOR CONTINUOUS INITIALIZATION

An audio encoder for encoding an audio signal includes: a first encoding processor for encoding a first audio signal portion in a frequency domain, wherein the first encoding processor includes: a time frequency converter for converting the first audio signal portion into a frequency domain representation having spectral lines up to a maximum frequency of the first audio signal portion; a spectral encoder for encoding the frequency domain representation; a second encoding processor for encoding a second different audio signal portion in the time domain; a cross-processor for calculating, from the encoded spectral representation of the first audio signal portion, initialization data of the second encoding processor, so that the second encoding processing is initialized to encode the second audio signal portion immediately following the first audio signal portion in time in the audio signal; a controller configured for analyzing the audio signal and for determining, which portion of the audio signal is the first audio signal portion encoded in the frequency domain and which portion of the audio signal is the second audio signal portion encoded in the time domain; and an encoded signal former for forming an encoded audio signal including a first encoded signal portion for the first audio signal portion and a second encoded signal portion for the second audio signal portion. 1. An audio encoder for encoding an audio signal , comprising:{'claim-text': ['a time-frequency converter configured for converting the first audio signal portion into a frequency domain representation comprising spectral lines up to a maximum frequency of the first audio signal portion; and', 'a spectral encoder configured for encoding the frequency domain representation;'], '#text': 'a first encoding processor configured for encoding a first audio signal portion in a frequency domain, wherein the first encoding processor comprises:'}a second encoding processor configured for encoding a second ...

METHOD FOR ESTIMATING NOISE IN AN AUDIO SIGNAL, NOISE ESTIMATOR, AUDIO ENCODER, AUDIO DECODER, AND SYSTEM FOR TRANSMITTING AUDIO SIGNALS

A method is described that estimates noise in an audio signal. An energy value for the audio signal is estimated and converted into the logarithmic domain. A noise level for the audio signal is estimated based on the converted energy value. 1. A method for estimating noise in an audio signal , the method comprising:determining an energy value for the audio signal;converting the energy value into the log 2-domain; andestimating a noise level for the audio signal based on the converted energy value directly in the log 2-domain.2. The method of claim 1 , wherein estimating the noise level comprises performing a predefined noise estimation algorithm claim 1 , like the minimum statistics algorithm.3. The method of claim 1 , wherein determining the energy value comprises acquiring a power spectrum of the audio signal by transforming the audio signal into the frequency domain claim 1 , grouping the power spectrum into psychoacoustically motivated bands claim 1 , and accumulating the power spectral bins within a band to form an energy value for each band claim 1 , wherein the energy value for each band is converted into the logarithmic domain claim 1 , and wherein a noise level is estimated for each band based on the corresponding converted energy value.4. The method of claim 1 , wherein the audio signal comprises a plurality of frames claim 1 , and wherein for each frame the energy value is determined and converted into the logarithmic domain claim 1 , and the noise level is estimated for each band of a frame based on the converted energy value.6. The method of claim 1 , wherein estimating the noise level based on the converted energy value yields logarithmic data claim 1 , and wherein the method further comprises:using the logarithmic data directly for further processing, orconverting the logarithmic data back into the linear domain for further processing.8. A non-transitory digital storage medium having stored thereon a computer program for performing a method for ...

ENCODER FOR ENCODING AN AUDIO SIGNAL, AUDIO TRANSMISSION SYSTEM AND METHOD FOR DETERMINING CORRECTION VALUES

An encoder for encoding an audio signal includes an analyzer for analyzing the audio signal and for determining analysis prediction coefficients from the audio signal. The encoder includes a converter for deriving converted prediction coefficients from the analysis prediction coefficients, a memory for storing a multitude of correction values and a calculator. The calculator includes a processor for processing the converted prediction coefficients to obtain spectral weighting factors. The calculator includes a combiner for combining the spectral weighting factors and the multitude of correction values to obtain corrected weighting factors. A quantizer of the calculator is configured for quantizing the converted prediction coefficients using the corrected weighting factors to obtain a quantized representation of the converted prediction coefficients. The encoder includes a bitstream former for forming an output signal based on the quantized representation of the converted prediction coefficients and based on the audio signal. 1. Encoder for encoding an audio signal , the encoder comprising:an analyzer configured for analyzing the audio signal and for determining analysis prediction coefficients from the audio signal;a converter configured for deriving converted prediction coefficients from the analysis prediction coefficients;a memory configured for storing a multitude of correction values; a processor configured for processing the converted prediction coefficients to obtain spectral weighting factors;', 'a combiner configured for combining the spectral weighting factors and the multitude of correction values to obtain corrected weighting factors; and', 'a quantizer configured for quantizing the converted prediction coefficients using the corrected weighting factors to obtain a quantized representation of the converted prediction coefficients; and, 'a calculator comprisinga bitstream former configured for forming an output signal based on the quantized representation ...

MANAGING NETWORK DEVICE

A network device for managing a call between user terminals checks whether a first user terminal supports usage of a first audio coding mode for the call, and a second user terminal intends to use a second audio coding mode for the call, and, if the first user terminal supports the usage of the first audio coding mode, and the second user terminal intends to use the second audio coding mode, repacks first data of the call sent from the first user terminal to the second user terminal and packetized into first packets referring to the second audio coding mode, into second packets referring to the first audio coding mode; and repacks second data of the call sent from the second user terminal to the first user terminal and packetized into third packets referring to the second audio coding mode, into fourth packets referring to the first audio coding mode. 1. A network device for managing a call between user terminals , configured tocheck whether a first user terminal supports a usage of a first audio coding mode for the call, and a second user terminal intends to use a second audio coding mode for the call; repack first data of the call sent from the first user terminal to the second user terminal and packetized into first packets referring to the first audio coding mode, into second packets referring to the second audio coding mode; and', 'repack second data of the call sent from the second user terminal to the first user terminal and packetized into third packets referring to the second audio coding mode, into fourth packets referring to the first audio coding mode;, 'if the first user terminal supports the usage of the first audio coding mode for the call, and the second user terminal intends to use the second audio coding mode for the call,'}wherein the network device is further configured tocheck whether a first initial call offer sent from an originating user terminal to a terminating user terminal indicates a usage of the second audio coding mode for the call; ' ...

AUDIO ENCODER WITH A SIGNAL-DEPENDENT NUMBER AND PRECISION CONTROL, AUDIO DECODER, AND RELATED METHODS AND COMPUTER PROGRAMS

An audio encoder for encoding audio input data has: a preprocessor for preprocessing the audio input data to obtain audio data to be coded; a coder processor for coding the audio data to be coded; and a controller for controlling the coder processor so that, depending on a first signal characteristic of a first frame of the audio data to be coded, a number of audio data items of the audio data to be coded by the coder processor for the first frame is reduced compared to a second signal characteristic of a second frame, and a first number of information units used for coding the reduced number of audio data items for the first frame is stronger enhanced compared to a second number of information units for the second frame. 1. An audio encoder for encoding audio input data , comprising:a preprocessor for preprocessing the audio input data to acquire audio data to be coded;a coder processor for coding the audio data to be coded; anda controller for controlling the coder processor so that, depending on a first signal characteristic of a first frame of the audio data to be coded, a number of audio data items of the audio data to be coded by the coder processor for the first frame is reduced compared to a second signal characteristic of a second frame, and a first number of information units used for coding the reduced number of audio data items for the first frame is stronger enhanced compared to a second number of information units for the second frame.2. The audio encoder of claim 1 ,wherein the coder processor comprises an initial coding stage and a refinement coding stage,wherein the controller is configured to reduce the number of audio data items encoded by the initial coding stage for the first frame,wherein the initial coding stage is configured to code the reduced number of audio data items for the first frame using a first frame initial number of information units, andwherein the refinement coding stage is configured to use a first frame remaining number of ...

AUDIO ENCODER WITH A SIGNAL-DEPENDENT NUMBER AND PRECISION CONTROL, AUDIO DECODER, AND RELATED METHODS AND COMPUTER PROGRAMS

An audio encoder for encoding audio input data has: a preprocessor for preprocessing the audio input data to obtain audio data to be coded; a coder processor for coding the audio data to be coded; and a controller for controlling the coder processor so that, depending on a first signal characteristic of a first frame of the audio data to be coded, a number of audio data items of the audio data to be coded by the coder processor for the first frame is reduced compared to a second signal characteristic of a second frame, and a first number of information units used for coding the reduced number of audio data items for the first frame is stronger enhanced compared to a second number of information units for the second frame. 1. An audio decoder for decoding encoded audio data , the encoded audio data comprising , for a frame , a frame initial number of information units and a frame remaining number of information units , the audio decoder comprising:a coder processor for processing the encoded audio data, the coder processor comprising an initial decoding stage and a refinement decoding stage; anda controller for controlling the coder processor so that the initial decoding stage uses the frame initial number of information units to acquire initially decoded data items, and the refinement decoding stage uses the frame remaining number of information units,wherein the controller is configured to control the refinement decoding stage to use, when refining the initially decoded data items, at least two information units of the remaining number of information units for refining one and the same initially decoded data item; anda postprocessor for postprocessing refined audio data items to acquire decoded audio data.2. The audio decoder of claim 1 , wherein the frame remaining number of information units comprise calculated values of information units for at least two sequential iterations in a predetermined order claim 1 ,wherein the controller is configured to control the ...

LOW-COMPLEXITY TONALITY-ADAPTIVE AUDIO SIGNAL QUANTIZATION

The invention provides an audio encoder for encoding an audio signal so as to produce therefrom an encoded signal, the audio encoder including: a framing device configured to extract frames from the audio signal; a quantizer configured to map spectral lines of a spectrum signal derived from the frame of the audio signal to quantization indices, wherein the quantizer has a dead-zone, in which the input spectral lines are mapped to quantization index zero; and a control device configured to modify the dead-zone; wherein the control device includes a tonality calculating device configured to calculate at least one tonality indicating value for at least one spectrum line or for at least one group of spectral lines, wherein the control device is configured to modify the dead-zone for the at least one spectrum line or the at least one group of spectrum lines depending on the respective tonality indicating value. 1. Audio encoder for encoding an audio signal so as to produce therefrom an encoded signal , the audio encoder comprising:a framing device configured to extract frames from the audio signal;a quantizer configured to map spectral lines of a spectrum signal derived from the frame of the audio signal to quantization indices, wherein the quantizer comprises a dead-zone, in which the spectral lines are mapped to quantization index zero; anda control device configured to modify the dead-zone;wherein the control device comprises a tonality calculating device configured to calculate at least one tonality indicating value for at least one spectrum line or for at least one group of spectral lines,wherein the control device is configured to modify the dead-zone for the at least one spectrum line or the at least one group of spectrum lines depending on the respective tonality indicating value.2. Audio encoder according to claim 1 , wherein the control device is configured to modify the dead-zone in such way that the dead-zone at one of the spectral lines is larger than the ...

METHOD AND APPARATUS FOR PROCESSING AN AUDIO SIGNAL, AUDIO DECODER, AND AUDIO ENCODER

A method is described that processes an audio signal. A discontinuity between a filtered past frame and a filtered current frame of the audio signal is removed using linear predictive filtering. 1. A method for processing an audio signal , the method comprising:removing a discontinuity between a filtered past frame and a filtered current frame of the audio signal using linear predictive filtering.2. The method of claim 1 , comprising filtering the current frame of the audio signal and removing the discontinuity by modifying a beginning portion of the filtered current frame by a signal acquired by linear predictive filtering a predefined signal with initial states of the linear predictive filter defined on the basis of a last part of the past frame.3. The method of claim 2 , wherein the initial states of the linear predictive filter are defined on the basis of a last part of the unfiltered past frame filtered using the set of filter parameters for filtering the current frame.4. The method of claim 1 , further comprising estimating the linear predictive filter on the filtered or non-filtered audio signal.5. The method of claim 4 , wherein estimating the linear predictive filter comprises estimating the filter based on the past and/or current frame of the audio signal or based on the past filtered frame of the audio signal using the Levinson-Durbin algorithm.6. The method of claim 1 , wherein the linear predictive filter comprises a linear predictive filter of an audio codec.7. The method of claim 1 , wherein removing the discontinuity comprises processing the beginning portion of the filtered current frame claim 1 , wherein the beginning portion of the current frame comprises a predefined number of samples being less or equal than the total number of samples in the current frame claim 1 , and wherein processing the beginning portion of the current frame comprises subtracting a beginning portion of a zero-input-response (ZIR) from the beginning portion of the filtered ...

LINEAR PREDICTION BASED AUDIO CODING USING IMPROVED PROBABILITY DISTRIBUTION ESTIMATION

Linear prediction based audio coding is improved by coding a spectrum composed of a plurality of spectral components using a probability distribution estimation determined for each of the plurality of spectral components from linear prediction coefficient information. The linear prediction coefficient information is available anyway. Accordingly, it may be used for determining the probability distribution estimation at both encoding and decoding side. The latter determination may be implemented in a computationally simple manner by using, for example, an appropriate parameterization for the probability distribution estimation at the plurality of spectral components. The coding efficiency as provided by the entropy coding is compatible with probability distribution estimations as achieved using context selection, but its derivation is less complex. The derivation may be purely analytically and/or does not require any information on attributes of neighboring spectral lines such as previously coded/decoded spectral values of neighboring spectral lines as is the case in spatial context selection. 1. A linear prediction based audio decoder comprising:a probability distribution estimator configured to determine, for each of a plurality of spectral components, a probability distribution estimation from linear prediction coefficient information comprised in a data stream into which an audio signal is encoded;an entropy decoding and dequantization stage configured to entropy decode and dequantize a spectrum composed of the plurality of spectral components from the data stream using the probability distribution estimation as determined for each of the plurality of spectral components; anda filter configured to shape the spectrum according to a transfer function depending on a linear prediction synthesis filter defined by the linear prediction coefficient information,wherein the probability distribution estimator is configured to determine a spectral fine structure from long- ...

ENCODER, DECODER AND METHOD FOR ENCODING AND DECODING AUDIO CONTENT USING PARAMETERS FOR ENHANCING A CONCEALMENT

Described are an encoder for coding speech-like content and/or general audio content, wherein the encoder is configured to embed, at least in some frames, parameters in a bitstream, which parameters enhance a concealment in case an original frame is lost, corrupted or delayed, and a decoder for decoding speech-like content and/or general audio content, wherein the decoder is configured to use parameters which are sent later in time to enhance a concealment in case an original frame is lost, corrupted or delayed, as well as a method for encoding and a method for decoding. 1. An apparatus for encoding speech-like content and/or general audio content ,wherein the apparatus is configured to embed, at least in some frames, parameters in a bitstream, which parameters provide for a guided concealment in case an original frame is lost, corrupted or delayed,wherein the apparatus is configured to create a primary frame and a partial copy, wherein the partial copy is not a low bitrate version of the primary frame but wherein the partial copy comprises the parameters, and wherein the partial copy is transmitted in-band as part of the codec payload,wherein the apparatus is configured to choose between multiple partial copy modes which use different amounts of information and/or different parameter sets, wherein the selection of the partial copy mode is based on parameters,and wherein at least one of the multiple partial copy modes is a frequency domain concealment mode, and at least two of the multiple partial copy modes are different time domain concealment modes,wherein the apparatus is implemented, at least in part, by one or more hardware elements.2. The apparatus according to claim 1 , wherein the apparatus is configured to delay the parameters by some time and to embed the parameters in a packet which is encoded and sent later in time.3. The apparatus according to claim 1 , wherein the apparatus is configured to reduce a primary frame bitrate claim 1 , wherein the primary ...

APPARATUS AND METHOD FOR IMPROVED CONCEALMENT OF THE ADAPTIVE CODEBOOK IN A CELP-LIKE CONCEALMENT EMPLOYING IMPROVED PULSE RESYNCHRONIZATION

An apparatus for reconstructing a frame including a speech signal as a reconstructed frame is provided, the apparatus including a determination unit and a frame reconstructor being configured to reconstruct the reconstructed frame, such that the reconstructed frame completely or partially includes the first reconstructed pitch cycle, such that the reconstructed frame completely or partially includes a second reconstructed pitch cycle, and such that the number of samples of the first reconstructed pitch cycle differs from a number of samples of the second reconstructed pitch cycle. 1. An apparatus for reconstructing a frame comprising a speech signal as a reconstructed frame , said reconstructed frame being associated with at least one available frame , said at least one available frame being at least one of preceding frames of the reconstructed frame and at least one succeeding frame of the reconstructed frame , wherein the at least one available frame comprises at least one pitch cycle as at least one available pitch cycle , wherein the apparatus comprises:a determination unit for determining a sample number difference indicating a difference between a number of samples of one of the at least one available pitch cycle and a number of samples of a first pitch cycle to be reconstructed, anda frame reconstructor for reconstructing the reconstructed frame by reconstructing, depending on the sample number difference and depending on the samples of said one of the at least one available pitch cycle, the first pitch cycle to be reconstructed as a first reconstructed pitch cycle,wherein the frame reconstructor is configured to reconstruct the reconstructed frame, such that the reconstructed frame comprises the first reconstructed pitch cycle, such that the reconstructed frame comprises a second reconstructed pitch cycle, and such that the number of samples of the first reconstructed pitch cycle differs from a number of samples of the second reconstructed pitch cycle, ...

APPARATUS AND METHOD FOR PROCESSING AN AUDIO SIGNAL USING A HARMONIC POST-FILTER

An apparatus for processing an audio signal having associated therewith a pitch lag information and a gain information, includes a domain converter for converting a first domain representation of the audio signal into a second domain representation of the audio signal; and a harmonic post-filter for filtering the second domain representation of the audio signal, wherein the post-filter is based on a transfer function including a numerator and a denominator, wherein the numerator includes a gain value indicated by the gain information, and wherein the denominator includes an integer part of a pitch lag indicated by the pitch lag information and a multi-tap filter depending on a fractional part of the pitch lag. 1. An apparatus for processing an audio signal having associated therewith a pitch lag information and a gain information , comprising:a domain converter for converting a first domain representation of the audio signal into a second domain representation of the audio signal, the second domain representation being a time domain representation; anda harmonic post-filter for filtering the second domain representation of the audio signal, wherein the harmonic post-filter is based on a long-term prediction filter working in the time-domain.2. Apparatus of claim 1 , wherein the long-term prediction filter claim 1 , on which the harmonic post-filter is based claim 1 , is configured to account for an integer part of a pitch lag indicated by the pitch lag information and a fractional part of the pitch lag indicated by the pitch lag information.3. Apparatus of claim 1 , wherein the long-term prediction filter claim 1 , on which the harmonic post-filter is based claim 1 , comprises parameters claim 1 , wherein the parameters are determined from parameters decoded a bitstream comprising the audio signal and the pitch lag information and the gain information.4. Apparatus of claim 3 , wherein the bitstream further comprises a decision bit claim 3 , and wherein the apparatus ...

Apparatus and method for improved concealment of the adaptive codebook in a celp-like concealment employing improved pitch lag estimation

An apparatus for determining an estimated pitch lag is provided. The apparatus includes an input interface for receiving a plurality of original pitch lag values, and a pitch lag estimator for estimating the estimated pitch lag. The pitch lag estimator is configured to estimate the estimated pitch lag depending on a plurality of original pitch lag values and depending on a plurality of information values, wherein for each original pitch lag value of the plurality of original pitch lag values, an information value of the plurality of information values is assigned to the original pitch lag value.

APPARATUS FOR ENCODING A SPEECH SIGNAL EMPLOYING ACELP IN THE AUTOCORRELATION DOMAIN

An apparatus for encoding a speech signal by determining a codebook vector of a speech coding algorithm is provided. The apparatus includes a matrix determiner for determining an autocorrelation matrix R, and a codebook vector determiner for determining the codebook vector depending on the autocorrelation matrix R. The matrix determiner is configured to determine the autocorrelation matrix R by determining vector coefficients of a vector r, wherein the autocorrelation matrix R includes a plurality of rows and a plurality of columns, wherein the vector r indicates one of the columns or one of the rows of the autocorrelation matrix R, wherein R(i, j)=r(|i−j|), wherein R(i, j) indicates the coefficients of the autocorrelation matrix R, wherein i is a first index indicating one of a plurality of rows of the autocorrelation matrix R, and wherein j is a second index indicating one of the plurality of columns of the autocorrelation matrix R. 1. An apparatus for encoding a speech signal by determining a codebook vector of a speech coding algorithm , wherein the apparatus comprises:a matrix determiner for determining an autocorrelation matrix R, anda codebook vector determiner for determining the codebook vector depending on the autocorrelation matrix R, {'br': None, 'i': R', 'i,j', 'r', 'i−j, '()=(||),'}, 'wherein the matrix determiner is configured to determine the autocorrelation matrix R by determining vector coefficients of a vector r, wherein the autocorrelation matrix R comprises a plurality of rows and a plurality of columns, wherein the vector r indicates one of the columns or one of the rows of the autocorrelation matrix R, wherein'}wherein R(i, j) indicates the coefficients of the autocorrelation matrix R, wherein i is a first index indicating one of a plurality of rows of the autocorrelation matrix R, and wherein j is a second index indicating one of the plurality of columns of the autocorrelation matrix R.3. The apparatus according to claim 1 ,wherein the matrix ...

AUDIO ENCODER AND DECODER USING A FREQUENCY DOMAIN PROCESSOR , A TIME DOMAIN PROCESSOR, AND A CROSS PROCESSING FOR CONTINUOUS INITIALIZATION

An audio encoder for encoding an audio signal, includes: a first encoding processor for encoding a first audio signal portion in a frequency domain, wherein the first encoding processor includes: a time frequency converter for converting the first audio signal portion into a frequency domain representation having spectral lines up to a maximum frequency of the first audio signal portion; a spectral encoder for encoding the frequency domain representation; a second encoding processor for encoding a second different audio signal portion in the time domain; a cross-processor for calculating, from the encoded spectral representation of the first audio signal portion, initialization data of the second encoding processor, so that the second encoding processing is initialized to encode the second audio signal portion immediately following the first audio signal portion in time in the audio signal. 1. An audio encoder for encoding an audio signal , comprising: a time frequency converter for converting the first audio signal portion into a frequency domain representation comprising spectral lines up to a maximum frequency of the first audio signal portion;', 'a spectral encoder for encoding the frequency domain representation;, 'a first encoding processor for encoding a first audio signal portion in a frequency domain, wherein the first encoding processor comprisesa second encoding processor for encoding a second different audio signal portion in the time domain,wherein the second encoding processor comprises an associated second sampling rate,wherein the first encoding processor has associated therewith a first sampling rate being different from the second sampling rate; a selector for selecting a portion of a spectrum input into the frequency time converter in accordance with a ratio of the first sampling rate and the second sampling rate,', 'a transform processor comprising a transform length being different from a transform length of the time-frequency converter; and', ' ...

METHOD AND APPARATUS FOR PROCESSING AN AUDIO SIGNAL, AUDIO DECODER, AND AUDIO ENCODER

A method is described that processes an audio signal. A discontinuity between a filtered past frame and a filtered current frame of the audio signal is removed using linear predictive filtering. 1. A method for processing an audio signal , the method comprising:using linear predictive filtering for removing a discontinuity between a filtered past frame and a filtered current frame of the audio signal,wherein the method comprises filtering the current frame of the audio signal and removing the discontinuity by modifying a beginning portion of the filtered current frame by a signal acquired by linear predictive filtering a predefined signal with initial states of the linear predictive filter defined on the basis of a last part of the unfiltered past frame filtered using the set of filter parameters for filtering the current frame.2. The method of claim 1 , further comprising estimating the linear predictive filter on the filtered or non-filtered audio signal.3. The method of claim 2 , wherein estimating the linear predictive filter comprises estimating the filter based on the past and/or current frame of the audio signal or based on the past filtered frame of the audio signal using the Levinson-Durbin algorithm.4. The method claim 1 , wherein the linear predictive filter comprises a linear predictive filter of an audio codec.5. The method of claim 1 , wherein removing the discontinuity comprises processing the beginning portion of the filtered current frame claim 1 , wherein the beginning portion of the current frame comprises a predefined number of samples being less or equal than the total number of samples in the current frame claim 1 , and wherein processing the beginning portion of the current frame comprises subtracting a beginning portion of a zero-input-response (ZIR) from the beginning portion of the filtered current frame.6. The method of claim 5 , comprising filtering the current frame of the audio signal using a non-recursive filter claim 5 , like a FIR ...

METHOD FOR ESTIMATING NOISE IN AN AUDIO SIGNAL, NOISE ESTIMATOR, AUDIO ENCODER, AUDIO DECODER, AND SYSTEM FOR TRANSMITTING AUDIO SIGNALS

A method is described that estimates noise in an audio signal. An energy value for the audio signal is estimated and converted into the logarithmic domain. A noise level for the audio signal is estimated based on the converted energy value. 2. The method of claim 1 , wherein estimating the noise level comprises performing a predefined noise estimation algorithm claim 1 , like the minimum statistics algorithm.3. The method of claim 1 , wherein determining the energy value comprises acquiring a power spectrum of the audio signal by transforming the audio signal into the frequency domain claim 1 , grouping the power spectrum into psychoacoustically motivated bands claim 1 , and accumulating the power spectral bins within a band to form an energy value for each band claim 1 , wherein the energy value for each band is converted into the log 2-domain claim 1 , and wherein a noise level is estimated for each band based on the corresponding converted energy value.4. The method of claim 3 , wherein the audio signal comprises a plurality of frames claim 3 , and wherein for each frame the energy value is determined and converted into the log 2-domain claim 3 , and the noise level is estimated for each band of a frame based on the converted energy value.5. The method of claim 1 , wherein estimating the noise level based on the converted energy value yields logarithmic data claim 1 , and wherein the method further comprises:using the logarithmic data directly for further processing, orconverting the logarithmic data back into the linear domain for further processing.6. The method of claim 5 , whereinthe logarithmic data is converted directly into transmission data, in case a transmission is done in the logarithmic domain, and {'br': None, 'sub': 'n—lin', 'sup': (E', {'sub2': 'n—log'}, '−1)., 'E=2'}, 'converting the logarithmic data directly into transmission data uses a shift function together with a lookup table or an approximation, e.g.,'}9. An audio encoder claim 8 , ...

APPARATUS AND METHOD FOR COMFORT NOISE GENERATION MODE SELECTION

An apparatus for encoding audio information is provided. The apparatus for encoding audio information includes a selector for selecting a comfort noise generation mode from two or more comfort noise generation modes depending on a background noise characteristic of an audio input signal, and an encoding unit for encoding the audio information, wherein the audio information includes mode information indicating the selected comfort noise generation mode. 1. An apparatus for encoding audio information , comprising:a selector for selecting a comfort noise generation mode from two or more comfort noise generation modes depending on a background noise characteristic of an audio input signal, andan encoding unit for encoding the audio information, wherein the audio information comprises mode information indicating the selected comfort noise generation mode,wherein a first one of the two or more comfort noise generation modes is a frequency-domain comfort noise generation mode, and wherein the frequency-domain comfort noise generation mode indicates that the comfort noise shall be generated in a frequency domain and that the comfort noise being generated in the frequency domain shall be frequency-to-time converted.2. The apparatus according to claim 1 ,wherein the selector is configured to determine a tilt of a background noise of the audio input signal as the background noise characteristic, andwherein the selector is configured to select said comfort noise generation mode from two or more comfort noise generation modes depending on the determined tilt.3. The apparatus according to claim 2 ,wherein the apparatus further comprises a noise estimator for estimating a per-band estimate of the background noise for each of a plurality of frequency bands, andwherein the selector is configured to determine the tilt depending on the estimated background noise of the plurality of frequency bands.4. The apparatus according to claim 3 ,wherein, the noise estimator is configured to ...

APPARATUS AND METHOD FOR PROCESSING AN AUDIO SIGNAL USING A HARMONIC POST-FILTER

An apparatus for processing an audio signal having associated therewith a pitch lag information and a gain information, includes a domain converter for converting a first domain representation of the audio signal into a second domain representation of the audio signal; and a harmonic post-filter for filtering the second domain representation of the audio signal, wherein the post-filter is based on a transfer function including a numerator and a denominator, wherein the numerator includes a gain value indicated by the gain information, and wherein the denominator includes an integer part of a pitch lag indicated by the pitch lag information and a multi-tap filter depending on a fractional part of the pitch lag. 1. An apparatus for processing an audio signal having associated therewith a pitch lag information and a gain information , comprising:a domain converter for converting a first domain representation of the audio signal into a second domain representation of the audio signal; anda harmonic post-filter for filtering the second domain representation of the audio signal, wherein the post-filter is based on a transfer function comprising a numerator and a denominator, wherein the numerator comprises a gain value indicated by the gain information, and wherein the denominator comprises an integer part of a pitch lag indicated by the pitch lag information and a multi-tap filter depending on a fractional part of the pitch lag.2. The apparatus of claim 1 , wherein the transfer function of the post-filter comprises claim 1 , in the numerator claim 1 , a further multi-tap FIR filter for a zero fractional part of the pitch lag.3. The apparatus of claim 1 , wherein the denominator comprises a product between the multi-tap filter and the gain value.4. The apparatus of claim 1 , wherein the numerator furthermore comprises a product of a first scalar value and a second scalar value claim 1 , wherein the denominator comprises the second scalar value and not the first scalar ...

Ecosystem and Method for Exchanging Information About ESG Data of a Product Related Entity

A method and system includes subject participants, a verifier participant and a certifier participant of a communication network exchanging information about ESG data, wherein a subject participant provides a proof derived from an ESG credential and the verifier participant receives the proof via the communication network, where the participant receives the ESG credential from the certifier participant, the proof assigns authenticity of the certifier participant providing the ESG credential, wherein the proof additionally assigns integrity of the ESG data, parts or an attribute of the ESG data, wherein the system further includes nodes of a decentralized registry that manages a public key of the certifier participant, where the decentralized registry provides a public key of the certifier participant corresponding to the proof to the verifier participant so that the verifier participant can cryptographically verify the authenticity and integrity with the public key via a computing component of the verifier participant. 1. A system comprising:subject participants, at least one verifier participant, and at least one certifier participant of a communication network each exchanging information about environmental, social and corporate governance (ESG) data of a product related entity;wherein at least one subject participant of the subject participants is configured to provide at least one proof derived from an ESG credential via the communication network and the at least one of the verifier participant is configured to receive the at least one proof via the communication network;wherein the at least one subject participant is configured to receive the ESG credential from the at least one certifier participant;wherein the at least one proof is configured to assign an authenticity of the at least one certifier participant providing the ESG credential;wherein the at least one proof is further configured to assign an integrity of the ESG data, parts of the ESG data or an ...

CONCEPT FOR CODING MODE SWITCHING COMPENSATION

A codec allowing for switching between different coding modes is improved by, responsive to a switching instance, performing temporal smoothing and/or blending at a respective transition. 1. Decoder supporting , and being switchable between , at least two modes so as to decode an information signal , wherein the decoder is configured to , responsive to a switching instance , perform temporal smoothing and/or blending at a transition between a first temporal portion of the information signal , preceding the switching instance , and a second temporal portion of the information signal , succeeding the switching instance , in a manner confined to a high-frequency spectral band.2. Decoder according to claim 1 , wherein the decoder is responsive to a switching of one or more offrom a full-bandwidth audio coding mode to a BWE or sub-bandwidth audio coding mode, andfrom a BWE or sub-bandwidth audio coding mode to a full-bandwidth audio coding mode, andfrom a guided BWE coding mode to a blind BWE coding mode,from a blind BWE coding mode to a guided BWE coding mode, andbetween full-bandwidth audio coding modes with different signal-energy-preserving properties.3. Decoder according to claim 1 , wherein the high-frequency spectral band overlaps with the effective coded bandwidth of both coding modes between which the switching at the switching instance takes place.4. Decoder according to claim 3 , wherein the high-frequency spectral band overlaps with a spectral BWE extension portion of one of the two coding modes between which the switching at the switching instance takes place.5. Decoder according to claim 4 , wherein the high-frequency spectral band overlaps with a spectral BWE extension portion or transform spectrum portion or linear-predictively coded spectral portion of the other of the two coding modes.6. Decoder according to claim 1 , wherein the decoder is configured to perform the temporal smoothing and/or blending additionally depending on an analysis of the ...

ENCODER, DECODER AND METHOD FOR ENCODING AND DECODING AUDIO CONTENT USING PARAMETERS FOR ENHANCING A CONCEALMENT

Described are an encoder for coding speech-like content and/or general audio content, wherein the encoder is configured to embed, at least in some frames, parameters in a bitstream, which parameters enhance a concealment in case an original frame is lost, corrupted or delayed, and a decoder for decoding speech-like content and/or general audio content, wherein the decoder is configured to use parameters which are sent later in time to enhance a concealment in case an original frame is lost, corrupted or delayed, as well as a method for encoding and a method for decoding. 1. An encoder for coding speech-like content and/or general audio content ,wherein the encoder is configured to embed, at least in some frames, parameters in a bitstream, which parameters provide for a guided concealment in case an original frame is lost, corrupted or delayed,wherein the encoder is configured to create a primary frame and a partial copy, wherein the partial copy is not a low bitrate version of the primary frame but wherein the partial copy comprises the parameters, and wherein the partial copy is transmitted in-band as part of the codec payload,wherein the encoder is configured to choose between multiple partial copy modes which use different amounts of information and/or different parameter sets, wherein the selection of the partial copy mode is based on parameters,and wherein at least one of the multiple partial copy modes is a frequency domain concealment mode, and at least two of the multiple partial copy modes are different time domain concealment modes.2. The encoder according to claim 1 , wherein the encoder is configured to delay the parameters by some time and to embed the parameters in a packet which is encoded and sent later in time.3. The encoder according to claim 1 , wherein the encoder is configured to reduce a primary frame bitrate claim 1 , wherein the primary frame bitrate reduction and a partial copy frame coding mechanism together determine a bitrate allocation ...

Apparatus and Method for Estimating an Inter-Channel Time Difference

An apparatus for estimating an inter-channel time difference between a first channel signal and a second channel signal, includes: a calculator for calculating a cross-correlation spectrum for a time block from the first channel signal in the time block and the second channel signal in the time block; a spectral characteristic estimator for estimating a characteristic of a spectrum of the first channel signal or the second channel signal for the time block; a smoothing filter for smoothing the cross-correlation spectrum over time using the spectral characteristic to obtain a smoothed cross-correlation spectrum; and a processor for processing the smoothed cross-correlation spectrum to obtain the inter-channel time difference. 1. An apparatus for estimating an inter-channel time difference between a first channel signal and a second channel signal , comprising:a calculator for calculating a cross-correlation spectrum for a time block from the first channel signal in the time block and the second channel signal in the time block;a spectral characteristic estimator for estimating a characteristic of a spectrum of the first channel signal or the second channel signal for the time block;a smoothing filter for smoothing the cross-correlation spectrum over time using the spectral characteristic to acquire a smoothed cross-correlation spectrum; anda processor for processing the smoothed cross-correlation spectrum to acquire the inter-channel time difference.2. The apparatus of claim 1 ,wherein the processor is configured to normalize the smoothed cross-correlation spectrum using a magnitude of the smoothed cross-correlation spectrum.3. The apparatus of claim 1 ,wherein the processor is configuredto calculate a time-domain representation of the smoothed cross-correlation spectrum or a normalized smoothed cross-correlation spectrum; andto analyze the time-domain representation to determine the inter-channel time difference.4. The apparatus of claim 1 ,wherein the processor is ...

ENCODER FOR ENCODING AN AUDIO SIGNAL, AUDIO TRANSMISSION SYSTEM AND METHOD FOR DETERMINING CORRECTION VALUES

An encoder for encoding an audio signal includes an analyzer for analyzing the audio signal and for determining analysis prediction coefficients from the audio signal. The encoder includes a converter for deriving converted prediction coefficients from the analysis prediction coefficients, a memory for storing a multitude of correction values and a calculator. The calculator includes a processor for processing the converted prediction coefficients to obtain spectral weighting factors. The calculator includes a combiner for combining the spectral weighting factors and the multitude of correction values to obtain corrected weighting factors. A quantizer of the calculator is configured for quantizing the converted prediction coefficients using the corrected weighting factors to obtain a quantized representation of the converted prediction coefficients. The encoder includes a bitstream former for forming an output signal based on the quantized representation of the converted prediction coefficients and based on the audio signal. 1100102100. Encoder () for encoding an audio signal () , the encoder () comprising:{'b': 100', '102', '112', '102, 'an analyzer () configured for analyzing the audio signal () and for determining analysis prediction coefficients () from the audio signal ();'}{'b': 120', '122', '122', '112, 'a converter () configured for deriving converted prediction coefficients (; ′) from the analysis prediction coefficients ();'}{'b': 160', '162, 'a memory () configured for storing a multitude of correction values ();'}{'b': 130', '130, 'claim-text': [{'b': 140', '140', '122', '122', '142', '142, 'a processor (; ′) configured for processing the converted prediction coefficients (; ′) to obtain spectral weighting factors (; ′);'}, {'b': 150', '150', '142', '142', '162', '152', '152, 'a combiner (; ′) configured for combining the spectral weighting factors (; ′) and the multitude of correction values (; a, b, c) to obtain corrected weighting factors (; ′); and ...

AUDIO ENCODER AND DECODER USING A FREQUENCY DOMAIN PROCESSOR WITH FULL-BAND GAP FILLING AND A TIME DOMAIN PROCESSOR

An audio encoder for encoding an audio signal has: a first encoding processor for encoding a first audio signal portion in a frequency domain, having: a time frequency converter for converting the first audio signal portion into a frequency domain representation; an analyzer for analyzing the frequency domain representation to determine first spectral portions to be encoded with a first spectral resolution and second regions to be encoded with a second resolution; and a spectral encoder for encoding the first spectral portions with the first spectral resolution and encoding the second portions with the second resolution; a second encoding processor for encoding a second different audio signal portion in the time domain; a controller for analyzing and determining, which portion of the audio signal is the first audio signal portion encoded in the frequency domain and which portion is the second audio signal portion encoded in the time domain; and an encoded signal former for forming an encoded audio signal having a first encoded signal portion for the first audio signal portion and a second encoded signal portion for the second portion. 1. An audio encoder for encoding an audio signal to generate an encoded audio signal , comprising: a time frequency converter for converting the first audio signal portion into a frequency domain representation comprising spectral lines up to a maximum frequency of the first audio signal portion;', 'an analyzer for analyzing the frequency domain representation up to the maximum frequency to determine first spectral portions to be encoded with a first spectral resolution and second spectral portions to be encoded with a second spectral resolution, the second spectral resolution being lower than the first spectral resolution, wherein the analyzer is configured to determine a first spectral portion from the first spectral portions, the first spectral portion being placed, with respect to frequency, between two second spectral portions ...

Apparatus and Method for Encoding or Decoding a Multi-Channel Signal Using Spectral-Domain Resampling

An apparatus for encoding a multi-channel signal having at least two channels, has: a time-spectral converter for converting sequences of blocks of sample values of the at least two channels into a frequency domain representation having sequences of blocks of spectral values for the at least two channels, wherein a block of sampling values has an associated input sampling rate, and a block of spectral values of the sequences of blocks of spectral values has spectral values up to a maximum input frequency being related to the input sampling rate; a multi-channel processor to obtain at least one result sequence of blocks of spectral values having information related to the at least two channels; a spectral domain resampler to obtain a resampled sequence of blocks of spectral values; a spectral-time converter for converting the resampled sequence of blocks of spectral values into a time domain representation; and a core encoder for encoding the output sequence of blocks of sampling values to obtain an encoded multi-channel signal. 1. An apparatus for encoding a multi-channel signal comprising at least two channels , comprising:a time-spectral converter for converting sequences of blocks of sample values of the at least two channels into a frequency domain representation comprising sequences of blocks of spectral values for the at least two channels, wherein a block of sampling values comprises an associated input sampling rate, and a block of spectral values of the sequences of blocks of spectral values comprises spectral values up to a maximum input frequency being related to the input sampling rate;a multi-channel processor for applying a joint multi-channel processing to the sequences of blocks of spectral values or to resampled sequences of blocks of spectral values to acquire at least one result sequence of blocks of spectral values comprising information related to the at least two channels;a spectral domain resampler for resampling the blocks of the result ...

APPARATUS AND METHOD FOR EFFICIENT SYNTHESIS OF SINUSOIDS AND SWEEPS BY EMPLOYING SPECTRAL PATTERNS

An apparatus for generating an audio output signal based on an encoded audio signal spectrum is provided. The apparatus has a processing unit for processing the encoded audio signal spectrum to obtain a decoded audio signal spectrum having a plurality of spectral coefficients, wherein each of the spectral coefficients has a spectral location within the encoded audio signal spectrum and a spectral value. Moreover, the apparatus has a pseudo coefficients determiner for determining one or more pseudo coefficients. Furthermore, the apparatus has a replacement unit for replacing at least one or more pseudo coefficients by a determined spectral pattern to obtain a modified audio signal spectrum, wherein each of at least two pattern coefficients has a spectral value. Moreover, the apparatus has a spectrum-time-conversion unit for converting the modified audio signal spectrum to a time-domain. 1. An apparatus for generating an audio output signal based on an encoded audio signal spectrum , wherein the apparatus comprises:a processing unit for processing the encoded audio signal spectrum to acquire a decoded audio signal spectrum comprising a plurality of spectral coefficients, wherein each of the spectral coefficients comprises a spectral location within the encoded audio signal spectrum and a spectral value, wherein the spectral coefficients are sequentially ordered according to their spectral location within the encoded audio signal spectrum so that the spectral coefficients form a sequence of spectral coefficients,a pseudo coefficients determiner for determining one or more pseudo coefficients of the decoded audio signal spectrum, wherein each of the pseudo coefficients is one of the spectral coefficients,a replacement unit for replacing at least one or more pseudo coefficients by a determined spectral pattern to acquire a modified audio signal spectrum, wherein the determined spectral pattern comprises at least two pattern coefficients, wherein each of the at least two ...

APPARATUS FOR ENCODING A SPEECH SIGNAL EMPLOYING ACELP IN THE AUTOCORRELATION DOMAIN

An apparatus for encoding a speech signal by determining a codebook vector of a speech coding algorithm is provided. The apparatus includes a matrix determiner for determining an autocorrelation matrix R, and a codebook vector determiner for determining the codebook vector depending on the autocorrelation matrix R. The matrix determiner is configured to determine the autocorrelation matrix R by determining vector coefficients of a vector r, wherein the autocorrelation matrix R includes a plurality of rows and a plurality of columns, wherein the vector r indicates one of the columns or one of the rows of the autocorrelation matrix R, wherein R(i, j)=r(|i−j|), wherein R(i, j) indicates the coefficients of the autocorrelation matrix R, wherein i is a first index indicating one of a plurality of rows of the autocorrelation matrix R, and wherein j is a second index indicating one of the plurality of columns of the autocorrelation matrix R. 1. An apparatus for encoding a speech signal by determining a codebook vector of a speech coding algorithm , wherein the apparatus comprises:a matrix determiner for determining an autocorrelation matrix R, anda codebook vector determiner for determining the codebook vector depending on the autocorrelation matrix R, {'br': None, 'i': R', 'i,j', 'r', 'i−j, '()=(||),'}, 'wherein the matrix determiner is configured to determine the autocorrelation matrix R by determining vector coefficients of a vector r, wherein the autocorrelation matrix R comprises a plurality of rows and a plurality of columns, wherein the vector r indicates one of the columns or one of the rows of the autocorrelation matrix R, wherein'}wherein R(i, j) indicates the coefficients of the autocorrelation matrix R, wherein i is a first index indicating one of a plurality of rows of the autocorrelation matrix R, and wherein j is a second index indicating one of the plurality of columns of the autocorrelation matrix R.3. The apparatus according to claim 1 ,wherein the matrix ...

METHOD FOR ESTIMATING NOISE IN AN AUDIO SIGNAL, NOISE ESTIMATOR, AUDIO ENCODER, AUDIO DECODER, AND SYSTEM FOR TRANSMITTING AUDIO SIGNALS

A method is described that estimates noise in an audio signal. An energy value for the audio signal is estimated and converted into the logarithmic domain. A noise level for the audio signal is estimated based on the converted energy value. 2. The method of claim 1 , wherein estimating the noise level comprises performing a predefined noise estimation algorithm claim 1 , like the minimum statistics algorithm.3. The method of claim 1 , wherein determining the energy value comprises acquiring a power spectrum of the audio signal by transforming the audio signal into the frequency domain claim 1 , grouping the power spectrum into psychoacoustically motivated bands claim 1 , and accumulating the power spectral bins within a band to form an energy value for each band claim 1 , wherein the energy value for each band is converted into the log 2-domain claim 1 , and wherein a noise level is estimated for each band based on the corresponding converted energy value.4. The method of claim 3 , wherein the audio signal comprises a plurality of frames claim 3 , and wherein for each frame the energy value is determined and converted into the log 2-domain claim 3 , and the noise level is estimated for each band of a frame based on the converted energy value.5. The method of claim 1 , wherein estimating the noise level based on the converted energy value yields logarithmic data claim 1 , and wherein the method further comprises:using the logarithmic data directly for further processing, orconverting the logarithmic data back into the linear domain for further processing.6. The method of claim 5 , whereinthe logarithmic data is converted directly into transmission data, in case a transmission is done in the logarithmic domain, and {'br': None, 'sub': n', {'sub2': '—lin'}], 'sup': E', {'sub2': 'n_log'}, '−1), 'E2.'}, 'converting the logarithmic data directly into transmission data uses a shift function together with a lookup table or an approximation, e.g.,'}9. An audio encoder claim ...

APPARATUS AND METHOD FOR PROCESSING AN AUDIO SIGNAL USING A HARMONIC POST-FILTER

An apparatus for processing an audio signal having associated therewith a pitch lag information and a gain information, includes a domain converter for converting a first domain representation of the audio signal into a second domain representation of the audio signal; and a harmonic post-filter for filtering the second domain representation of the audio signal, wherein the post-filter is based on a transfer function including a numerator and a denominator, wherein the numerator includes a gain value indicated by the gain information, and wherein the denominator includes an integer part of a pitch lag indicated by the pitch lag information and a multi-tap filter depending on a fractional part of the pitch lag. 1. An apparatus for processing an audio signal having associated therewith a pitch lag information and a gain information , comprising:a domain converter for converting a first domain representation of the audio signal into a second domain representation of the audio signal; anda harmonic post-filter for filtering the second domain representation of the audio signal, wherein the post-filter is based on a transfer function comprising a numerator and a denominator, wherein the numerator comprises a gain value indicated by the gain information, and wherein the denominator comprises an integer part of a pitch lag indicated by the pitch lag information and a multi-tap filter depending on a fractional part of the pitch lag.2. The apparatus of claim 1 , wherein the transfer function of the post-filter comprises claim 1 , in the numerator claim 1 , a further multi-tap FIR filter for a zero fractional part of the pitch lag.3. The apparatus of claim 1 , wherein the denominator comprises a product between the multi-tap filter and the gain value.4. The apparatus of claim 1 , wherein the numerator furthermore comprises a product of a first scalar value and a second scalar value claim 1 , wherein the denominator comprises the second scalar value and not the first scalar ...

Apparatus and Method for Estimating an Inter-Channel Time Difference

An apparatus for estimating an inter-channel time difference between a first channel signal and a second channel signal, includes: a calculator for calculating a cross-correlation spectrum for a time block from the first channel signal in the time block and the second channel signal in the time block; a spectral characteristic estimator for estimating a characteristic of a spectrum of the first channel signal or the second channel signal for the time block; a smoothing filter for smoothing the cross-correlation spectrum over time using the spectral characteristic to obtain a smoothed cross-correlation spectrum; and a processor for processing the smoothed cross-correlation spectrum to obtain the inter-channel time difference. 1. An apparatus for estimating an inter-channel time difference between a first channel signal and a second channel signal , comprising:a calculator for calculating a cross-correlation spectrum for a time block from the first channel signal in the time block and the second channel signal in the time block;a spectral characteristic estimator for estimating a characteristic of a spectrum of the first channel signal or the second channel signal for the time block;a smoothing filter for smoothing the cross-correlation spectrum over time using the spectral characteristic to acquire a smoothed cross-correlation spectrum; anda processor for processing the smoothed cross-correlation spectrum to acquire the inter-channel time difference.2. The apparatus of claim 1 ,wherein the processor is configured to normalize the smoothed cross-correlation spectrum using a magnitude of the smoothed cross-correlation spectrum.3. The apparatus of claim 1 ,wherein the processor is configuredto calculate a time-domain representation of the smoothed cross-correlation spectrum or a normalized smoothed cross-correlation spectrum; andto analyze the time-domain representation to determine the inter-channel time difference.4. The apparatus of claim 1 ,wherein the processor is ...

APPARATUS FOR ENCODING A SPEECH SIGNAL EMPLOYING ACELP IN THE AUTOCORRELATION DOMAIN

An apparatus for encoding a speech signal by determining a codebook vector of a speech coding algorithm is provided. The apparatus includes a matrix determiner for determining an autocorrelation matrix R, and a codebook vector determiner for determining the codebook vector depending on the autocorrelation matrix R. The matrix determiner is configured to determine the autocorrelation matrix R by determining vector coefficients of a vector r, wherein the autocorrelation matrix R includes a plurality of rows and a plurality of columns, wherein the vector r indicates one of the columns or one of the rows of the autocorrelation matrix R, wherein R(i, j)=r(|i−j|), wherein R(i, j) indicates the coefficients of the autocorrelation matrix R, wherein i is a first index indicating one of a plurality of rows of the autocorrelation matrix R, and wherein j is a second index indicating one of the plurality of columns of the autocorrelation matrix R. 1. An apparatus for encoding a speech signal by determining a codebook vector of a speech coding algorithm , wherein the apparatus comprises:a matrix determiner for determining an autocorrelation matrix R, anda codebook vector determiner for determining the codebook vector depending on the autocorrelation matrix R, {'br': None, 'i': R', 'i,j', 'r', 'i−j, '()=(||),'}, 'wherein the matrix determiner is configured to determine the autocorrelation matrix R by determining vector coefficients of a vector r, wherein the autocorrelation matrix R comprises a plurality of rows and a plurality of columns, wherein the vector r indicates one of the columns or one of the rows of the autocorrelation matrix R, wherein'}wherein R(i, j) indicates the coefficients of the autocorrelation matrix R, wherein i is a first index indicating one of a plurality of rows of the autocorrelation matrix R, and wherein j is a second index indicating one of the plurality of columns of the autocorrelation matrix R.3. The apparatus according to claim 1 ,wherein the matrix ...

Apparatuses and Methods for Encoding or Decoding a Multi-Channel Signal Using Frame Control Synchronization

An apparatus for encoding a multi-channel signal including at least two channels includes a time-spectral converter for converting sequences of blocks of sampling values of the at least two channels into a frequency domain representation having sequences of blocks of spectral values for the at least two channels; a multi-channel processor for applying a joint multi-channel processing to the sequences of blocks of spectral values to obtain at least one result sequence of blocks of spectral values including information related to the at least two channels; a spectral-time converter for converting the result sequence of blocks of spectral values into a time domain representation including an output sequence of blocks of sampling values; and a core encoder for encoding the output sequence of blocks of sampling values to obtain an encoded multi-channel signal. 1. Apparatus for encoding a multi-channel signal comprising at least two channels , comprising:a time-spectral converter for converting sequences of blocks of sampling values of the at least two channels into a frequency domain representation comprising sequences of blocks of spectral values for the at least two channels;a multi-channel processor for applying a joint multi-channel processing to the sequences of blocks of spectral values to acquire at least one result sequence of blocks of spectral values comprising information related to the at least two channels;a spectral-time converter for converting the result sequence of blocks of spectral values into a time domain representation comprising an output sequence of blocks of sampling values; anda core encoder for encoding the output sequence of blocks of sampling values to acquire an encoded multi-channel signal,wherein the core encoder is configured to operate in accordance with a first frame control to provide a sequence of frames, wherein a frame is bounded by a start frame border and an end frame border, andwherein the time-spectral converter or the spectral- ...

ENCODER FOR ENCODING AN AUDIO SIGNAL, AUDIO TRANSMISSION SYSTEM AND METHOD FOR DETERMINING CORRECTION VALUES

An encoder for encoding an audio signal includes an analyzer for analyzing the audio signal and for determining analysis prediction coefficients from the audio signal. The encoder includes a converter for deriving converted prediction coefficients from the analysis prediction coefficients, a memory for storing a multitude of correction values and a calculator. The calculator includes a processor for processing the converted prediction coefficients to obtain spectral weighting factors. The calculator includes a combiner for combining the spectral weighting factors and the multitude of correction values to obtain corrected weighting factors. A quantizer of the calculator is configured for quantizing the converted prediction coefficients using the corrected weighting factors to obtain a quantized representation of the converted prediction coefficients. The encoder includes a bitstream former for forming an output signal based on the quantized representation of the converted prediction coefficients and based on the audio signal. 1. Encoder for encoding an audio signal , the encoder comprising:an analyzer configured for analyzing the audio signal and for determining analysis prediction coefficients from the audio signal;a converter configured for deriving converted prediction coefficients from the analysis prediction coefficients;a memory configured for storing a multitude of correction values; a processor configured for processing the converted prediction coefficients to acquire spectral weighting factors;', 'a combiner configured for combining the spectral weighting factors and the multitude of correction values to acquire corrected weighting factors; and', 'a quantizer configured for quantizing the converted prediction coefficients using the corrected weighting factors to acquire a quantized representation of the converted prediction coefficients; and, 'a calculator comprisinga bitstream former configured for forming an output signal based on the quantized ...

AUDIO ENCODER AND DECODER USING A FREQUENCY DOMAIN PROCESSOR WITH FULL-BAND GAP FILLING AND A TIME DOMAIN PROCESSOR

An audio encoder for encoding an audio signal has: a first encoding processor for encoding a first audio signal portion in a frequency domain, having: a time frequency converter for converting the first audio signal portion into a frequency domain representation; an analyzer for analyzing the frequency domain representation to determine first spectral portions to be encoded with a first spectral resolution and second regions to be encoded with a second resolution; and a spectral encoder for encoding the first spectral portions with the first spectral resolution and encoding the second portions with the second resolution; a second encoding processor for encoding a second different audio signal portion in the time domain; a controller for analyzing and determining, which portion of the audio signal is the first audio signal portion encoded in the frequency domain and which portion is the second audio signal portion encoded in the time domain; and an encoded signal former for forming an encoded audio signal having a first encoded signal portion for the first audio signal portion and a second encoded signal portion for the second portion. 1. An audio encoder for encoding an audio signal , comprising: a time frequency converter for converting the first audio signal portion into a frequency domain representation comprising spectral lines up to a maximum frequency of the first audio signal portion;', 'an analyzer for analyzing the frequency domain representation up to the maximum frequency to determine first spectral portions to be encoded with a first spectral resolution and second spectral portions to be encoded with a second spectral resolution, the second spectral resolution being lower than the first spectral resolution, wherein the analyzer is configured to determine a first spectral portion from the first spectral portions, the first spectral portion being placed, with respect to frequency, between two second spectral portions from the second spectral portions;', 'a ...

GENERATION OF A COMFORT NOISE WITH HIGH SPECTRO-TEMPORAL RESOLUTION IN DISCONTINUOUS TRANSMISSION OF AUDIO SIGNALS

The invention provides an audio decoder being configured for decoding a bitstream so as to produce therefrom an audio output signal, the bitstream including at least an active phase followed by at least an inactive phase, wherein the bitstream has encoded therein at least a silence insertion descriptor frame which describes a spectrum of a background noise, the audio decoder including: a silence insertion descriptor decoder configured to decode the silence insertion descriptor frame; a decoding device configured to reconstruct the audio output signal from the bitstream during the active phase; a spectral converter configured to determine a spectrum of the audio output signal; a noise estimator device configured to determine a first spectrum of the noise of the audio output signal; a resolution converter configured to establish a second spectrum of the noise of the audio output signal; a comfort noise spectrum estimation device; and a comfort noise generator. 1. Audio decoder for decoding a bitstream so as to produce therefrom an audio output signal , the bitstream comprising at least an active phase followed by at least an inactive phase , wherein the bitstream has encoded therein at least a silence insertion descriptor frame which describes a spectrum of a background noise , the audio decoder comprising:a silence insertion descriptor decoder configured to decode the silence insertion descriptor frame so as to reconstruct the spectrum of the background noise;a decoding device configured to reconstruct the audio output signal from the bitstream during the active phase;a spectral converter configured to determine a spectrum of the audio output signal;a noise estimator device configured to determine a first spectrum of the noise of the audio output signal based on the spectrum of the audio output signal provided by the spectral converter, wherein the first spectrum of the noise of the audio output signal comprises a higher spectral resolution than the spectrum of the ...

Audio encoder and decoder using a frequency domain processor, a time domain processor, and a cross processing for continuous initialization

An audio encoder for encoding an audio signal includes: a first encoding processor for encoding a first audio signal portion in a frequency domain, wherein the first encoding processor includes: a time frequency converter for converting the first audio signal portion into a frequency domain representation having spectral lines up to a maximum frequency of the first audio signal portion; a spectral encoder for encoding the frequency domain representation; a second encoding processor for encoding a second different audio signal portion in the time domain; a cross-processor for calculating, from the encoded spectral representation of the first audio signal portion, initialization data of the second encoding processor, so that the second encoding processing is initialized to encode the second audio signal portion immediately following the first audio signal portion in time in the audio signal; a controller configured for analyzing the audio signal and for determining, which portion of the audio signal is the first audio signal portion encoded in the frequency domain and which portion of the audio signal is the second audio signal portion encoded in the time domain; and an encoded signal former for forming an encoded audio signal including a first encoded signal portion for the first audio signal portion and a second encoded signal portion for the second audio signal portion.

LOW-COMPLEXITY TONALITY-ADAPTIVE AUDIO SIGNAL QUANTIZATION

The invention provides an audio encoder for encoding an audio signal so as to produce therefrom an encoded signal, the audio encoder including: a framing device configured to extract frames from the audio signal; a quantizer configured to map spectral lines of a spectrum signal derived from the frame of the audio signal to quantization indices, wherein the quantizer has a dead-zone, in which the input spectral lines are mapped to quantization index zero; and a control device configured to modify the dead-zone; wherein the control device includes a tonality calculating device configured to calculate at least one tonality indicating value for at least one spectrum line or for at least one group of spectral lines, wherein the control device is configured to modify the dead-zone for the at least one spectrum line or the at least one group of spectrum lines depending on the respective tonality indicating value. 1. Audio encoder for encoding an audio signal so as to produce therefrom an encoded signal , the audio encoder comprising:a framing device configured to extract frames from the audio signal;a quantizer configured to map spectral lines of a spectrum signal derived from the frame of the audio signal to quantization indices, wherein the quantizer comprises a dead-zone, in which the spectral lines are mapped to quantization index zero; anda control device configured to modify the dead-zone;wherein the control device comprises a tonality calculating device configured to calculate at least one tonality indicating value for at least one spectrum line or for at least one group of spectral lines,wherein the control device is configured to modify the dead-zone for the at least one spectrum line or the at least one group of spectrum lines depending on the respective tonality indicating value.2. Audio encoder according to claim 1 , wherein the control device is configured to modify the dead-zone in such way that the dead-zone at one of the spectral lines is larger than the ...

Selecting pitch lag

In apparatus, methods, and programs for selecting pitch lag, an encoder obtains a first and a second estimates of a pitch lag for a current frame. A selected value is chosen by selection between the first and the second estimates, based on a first and a second correlation measurements. The second estimate is conditioned by the pitch lag selected at the previous frame. The selection is based on a comparison between: a downscaled version of a first correlation measurement associated to the current frame and obtained at a lag corresponding to the first estimate; and a second correlation measurement associated to the current frame and obtained at a lag corresponding to the second estimate.

APPARATUS AND METHOD FOR IMPROVED CONCEALMENT OF THE ADAPTIVE CODEBOOK IN ACELP-LIKE CONCEALMENT EMPLOYING IMPROVED PULSE RESYNCHRONIZATION

An apparatus for reconstructing a frame including a speech signal as a reconstructed frame is provided, the apparatus including a determination unit and a frame reconstructor being configured to reconstruct the reconstructed frame, such that the reconstructed frame completely or partially includes the first reconstructed pitch cycle, such that the reconstructed frame completely or partially includes a second reconstructed pitch cycle, and such that the number of samples of the first reconstructed pitch cycle differs from a number of samples of the second reconstructed pitch cycle. 1. An apparatus for reconstructing a frame comprising a speech signal as a reconstructed frame , said reconstructed frame being associated with at least one available frame , said at least one available frame being at least one of preceding frames of the reconstructed frame and at least one succeeding frame of the reconstructed frame , wherein the at least one available frame comprises at least one pitch cycle as at least one available pitch cycle , wherein the apparatus comprises:a determination unit for determining a sample number difference indicating a difference between a number of samples of one of the at least one available pitch cycle and a number of samples of a first pitch cycle to be reconstructed, anda frame reconstructor for reconstructing the reconstructed frame by reconstructing, depending on the sample number difference and depending on the samples of said one of the at least one available pitch cycle, the first pitch cycle to be reconstructed as a first reconstructed pitch cycle,wherein the frame reconstructor is configured to reconstruct the reconstructed frame, such that the reconstructed frame comprises the first reconstructed pitch cycle, such that the reconstructed frame comprises a second reconstructed pitch cycle, and such that the number of samples of the first reconstructed pitch cycle differs from a number of samples of the second reconstructed pitch cycle, ...

APPARATUS AND METHOD FOR ENCODING AND DECODING AN AUDIO SIGNAL USING DOWNSAMPLING OR INTERPOLATION OF SCALE PARAMETERS

An apparatus for encoding an audio signal includes: a converter for converting the audio signal into a spectral representation; a scale parameter calculator for calculating a first set of scale parameters from the spectral representation: a downsampler for downsampling the first set of scale parameters to obtain a second set of scale parameters, a second number of scale parameters in the second set of scale parameters being lower than a first number of scale parameters in the first set of scale parameters; a scale parameter encoder for generating an encoded representation of the second set of scale parameters; a spectral processor for processing the spectral representation using a third set of scale parameters, the third set of scale parameters having a third number of scale parameters being greater than the second number of scale parameters, the spectral processor being configured to use the first set of scale parameters or to derive the third set of scale parameters from the second set of scale parameters or from the encoded representation of the second set of scale parameters using an interpolation operation; and an output interface for generating an encoded output signal comprising information on the encoded representation of the spectral representation and information on the encoded representation of the second set of scale parameters.

ENCODER FOR ENCODING AN AUDIO SIGNAL, AUDIO TRANSMISSION SYSTEM AND METHOD FOR DETERMINING CORRECTION VALUES

An encoder for encoding an audio signal includes an analyzer for analyzing the audio signal and for determining analysis prediction coefficients from the audio signal. The encoder includes a converter for deriving converted prediction coefficients from the analysis prediction coefficients, a memory for storing a multitude of correction values and a calculator. The calculator includes a processor for processing the converted prediction coefficients to obtain spectral weighting factors. The calculator includes a combiner for combining the spectral weighting factors and the multitude of correction values to obtain corrected weighting factors. A quantizer of the calculator is configured for quantizing the converted prediction coefficients using the corrected weighting factors to obtain a quantized representation of the converted prediction coefficients. The encoder includes a bitstream former for forming an output signal based on the quantized representation of the converted prediction coefficients and based on the audio signal. 1. Method for determining correction values for a first multitude of first weighting factors each weighting factor adapted for weighting a portion of an audio signal , the method comprising:calculating the first multitude of first weighting factors for each audio signal of a set of audio signals and based on a first determination rule;calculating a second multitude of second weighting factors for each audio signal of the set of audio signals based on a second determination rule, each of the second multitude of weighting factors being related to a first weighting factor;calculating a third multitude of distance values each distance value having a value related to a distance between a first weighting factor and a second weighting factor related to a portion of the audio signal; andcalculating a fourth multitude of correction values adapted to reduce the distance values when combined with the first weighting factors;{'sup': '2', 'wherein the fourth ...

APPARATUS AND METHOD FOR IMPROVED CONCEALMENT OF THE ADAPTIVE CODEBOOK IN A CELP-LIKE CONCEALMENT EMPLOYING IMPROVED PITCH LAG ESTIMATION

An apparatus for determining an estimated pitch lag is provided. The apparatus includes an input interface for receiving a plurality of original pitch lag values, and a pitch lag estimator for estimating the estimated pitch lag. The pitch lag estimator is configured to estimate the estimated pitch lag depending on a plurality of original pitch lag values and depending on a plurality of information values, wherein for each original pitch lag value of the plurality of original pitch lag values, an information value of the plurality of information values is assigned to the original pitch lag value. 1. An apparatus for determining an estimated pitch lag , comprising:an input interface for receiving a plurality of original pitch lag values, anda pitch lag estimator for estimating the estimated pitch lag,wherein the pitch lag estimator is configured to estimate the estimated pitch lag depending on a plurality of original pitch lag values and depending on a plurality of information values, wherein for each original pitch lag value of the plurality of original pitch lag values, an information value of the plurality of information values is assigned to said original pitch lag value.2. An apparatus according to claim 1 , wherein the pitch lag estimator is configured to estimate the estimated pitch lag depending on the plurality of original pitch lag values and depending on a plurality of pitch gain values as the plurality of information values claim 1 , wherein for each original pitch lag value of the plurality of original pitch lag values claim 1 , a pitch gain value of the plurality of pitch gain values is assigned to said original pitch lag value.3. An apparatus according to claim 2 , wherein each of the plurality of pitch gain values is an adaptive codebook gain.4. An apparatus according to claim 2 , wherein the pitch lag estimator is configured to estimate the estimated pitch lag by minimizing an error function.7. An apparatus according to claim 4 , wherein the pitch lag ...

Concept for coding mode switching compensation

A codec allowing for switching between different coding modes is improved by, responsive to a switching instance, performing temporal smoothing and/or blending at a respective transition.

Apparatus and Method for Encoding or Decoding a Multi-Channel Signal Using a Broadband Alignment Parameter and a Plurality of Narrowband Alignment Parameters

The apparatus for encoding a multi-channel signal having at least two channels, includes: a parameter determiner for determining a broadband alignment parameter and a plurality of narrowband alignment parameters from the multichannel signal; a signal aligner for aligning the at least two channels using the broadband alignment parameter and the plurality of narrowband alignment parameters to obtain aligned channels; a signal processor for calculating a mid-signal and a side signal using the aligned channels; a signal encoder for encoding the mid-signal to obtain an encoded mid-signal and for encoding the side signal to obtain an encoded side signal; and an output interface for generating an encoded multi-channel signal including the encoded mid-signal, the encoded side signal, information on the broadband alignment parameter and information on the plurality of narrowband alignment parameters.

Apparatus and Method for Estimating an Inter-Channel Time Difference

An apparatus for estimating an inter-channel time difference between a first channel signal and a second channel signal, includes: a calculator for calculating a cross-correlation spectrum for a time block from the first channel signal in the time block and the second channel signal in the time block; a spectral characteristic estimator for estimating a characteristic of a spectrum of the first channel signal or the second channel signal for the time block; a smoothing filter for smoothing the cross-correlation spectrum over time using the spectral characteristic to obtain a smoothed cross-correlation spectrum; and a processor for processing the smoothed cross-correlation spectrum to obtain the inter-channel time difference. 1. An apparatus for estimating an inter-channel time difference between a first channel signal and a second channel signal , comprising:a calculator for calculating a cross-correlation spectrum for a time block from the first channel signal in the time block and the second channel signal in the time block;a spectral characteristic estimator for estimating a characteristic of a spectrum of the first channel signal or the second channel signal for the time block;a smoothing filter for smoothing the cross-correlation spectrum over time using the spectral characteristic to acquire a smoothed cross-correlation spectrum; anda processor for processing the smoothed cross-correlation spectrum to acquire the inter-channel time difference.2. The apparatus of claim 1 ,wherein the processor is configured to normalize the smoothed cross-correlation spectrum using a magnitude of the smoothed cross-correlation spectrum.3. The apparatus of claim 1 ,wherein the processor is configuredto calculate a time-domain representation of the smoothed cross-correlation spectrum or a normalized smoothed cross-correlation spectrum; andto analyze the time-domain representation to determine the inter-channel time difference.4. The apparatus of claim 1 ,wherein the processor is ...

Apparatus and method for mdct m/s stereo with global ild with improved mid/side decision

An apparatus for encoding a first channel and a second channel of an audio input signal including two or more channels to obtain an encoded audio signal according to an embodiment includes a normalizer configured to determine a normalization value for the audio input signal depending on the first channel of the audio input signal and depending on the second channel of the audio input signal. Moreover, the apparatus includes an encoding unit configured to generate a processed audio signal having a first channel and a second channel. The encoding unit is configured to encode the processed audio signal to obtain the encoded audio signal.

Apparatuses and Methods for Encoding or Decoding a Multi-Channel Signal Using Frame Control Synchronization

An apparatus for encoding a multi-channel signal including at least two channels includes a time-spectral converter for converting sequences of blocks of sampling values of the at least two channels into a frequency domain representation having sequences of blocks of spectral values for the at least two channels; a multi-channel processor for applying a joint multi-channel processing to the sequences of blocks of spectral values to obtain at least one result sequence of blocks of spectral values including information related to the at least two channels; a spectral-time converter for converting the result sequence of blocks of spectral values into a time domain representation including an output sequence of blocks of sampling values; and a core encoder for encoding the output sequence of blocks of sampling values to obtain an encoded multi-channel signal. 1. Apparatus for encoding a multi-channel signal comprising at least two channels , comprising:a time-spectral converter for converting sequences of blocks of sampling values of the at least two channels into a frequency domain representation comprising sequences of blocks of spectral values for the at least two channels;a multi-channel processor for applying a joint multi-channel processing to the sequences of blocks of spectral values to acquire at least one result sequence of blocks of spectral values comprising information related to the at least two channels;a spectral-time converter for converting the result sequence of blocks of spectral values into a time domain representation comprising an output sequence of blocks of sampling values; anda core encoder for encoding the output sequence of blocks of sampling values to acquire an encoded multi-channel signal,wherein the core encoder is configured to operate in accordance with a first frame control to provide a sequence of frames, wherein a frame is bounded by a start frame border and an end frame border, andwherein the time-spectral converter or the spectral- ...

Comfort noise addition for modeling background noise at low bit-rates

The invention provides a decoder being configured for processing an encoded audio bitstream, wherein the decoder includes: a bitstream decoder configured to derive a decoded audio signal from the bitstream, wherein the decoded audio signal includes at least one decoded frame; a noise estimation device configured to produce a noise estimation signal containing an estimation of the level and/or the spectral shape of a noise in the decoded audio signal; a comfort noise generating device configured to derive a comfort noise signal from the noise estimation signal; and a combiner configured to combine the decoded frame of the decoded audio signal and the comfort noise signal in order to obtain an audio output signal.

CONCEPT FOR CODING MODE SWITCHING COMPENSATION

A codec allowing for switching between different coding modes is improved by, responsive to a switching instance, performing temporal smoothing and/or blending at a respective transition. 1. A decoder supporting , and being switchable between , at least two modes so as to decode an information signal , wherein the decoder is configured to , responsive to a switching instance , perform temporal smoothing and/or blending at a transition between a first temporal portion of the information signal , preceding the switching instance , and a second temporal portion of the information signal , succeeding the switching instance , in a manner confined to a high-frequency spectral band.2. The decoder according to claim 1 , wherein the decoder is responsive to a switching of one or more offrom a full-bandwidth audio coding mode to a BWE or sub-bandwidth audio coding mode, andfrom a BWE or sub-bandwidth audio coding mode to a full-bandwidth audio coding mode, andfrom a guided BWE coding mode to a blind BWE coding mode,from a blind BWE coding mode to a guided BWE coding mode, andbetween full-bandwidth audio coding modes with different energy-preserving properties.3. The decoder according to claim 1 , wherein the high-frequency spectral band overlaps with the effective coded bandwidth of both coding modes between which the switching at the switching instance takes place.4. The decoder according to claim 3 , wherein the high-frequency spectral band overlaps with a spectral BWE extension portion of one of the two coding modes between which the switching at the switching instance takes place.5. The decoder according to claim 4 , wherein the high-frequency spectral band overlaps with a spectral BWE extension portion or transform spectrum portion or linear-predictively coded spectral portion of the other of the two coding modes.6. The decoder according to claim 1 , wherein the decoder is configured to perform the temporal smoothing and/or blending additionally depending on an analysis ...

Signal filtering

In methods and systems for filtering an information input signal, a system may have: a first filter unit filtering an input signal at an initial subinterval in a current update interval according to parameters associated to the preceding update interval, the parameters being scaled by a first scaling factor changing towards 0; and a second filter unit filtering a second filter input signal, based on the output of the first filter unit, at the initial subinterval, according to parameters associated to the current update interval, the parameters being scaled by a second scaling factor changing from 0, or a value close to 0, toward a value more distant from 0.

COMFORT NOISE ADDITION FOR MODELING BACKGROUND NOISE AT LOW BIT-RATES

The invention provides a decoder being configured for processing an encoded audio bitstream, wherein the decoder includes: a bitstream decoder configured to derive a decoded audio signal from the bitstream, wherein the decoded audio signal includes at least one decoded frame; a noise estimation device configured to produce a noise estimation signal containing an estimation of the level and/or the spectral shape of a noise in the decoded audio signal; a comfort noise generating device configured to derive a comfort noise signal from the noise estimation signal; and a combiner configured to combine the decoded frame of the decoded audio signal and the comfort noise signal in order to obtain an audio output signal. 1. A decoder being configured for processing an encoded audio bitstream , wherein the decoder comprises:a bitstream decoder configured to derive a decoded audio signal from the bitstream, wherein the decoded audio signal comprises at least one decoded frame;a noise estimation device configured to produce a noise estimation signal comprising an estimation of the level and/or the spectral shape of a noise in the decoded audio signal;a comfort noise generating device configured to derive a comfort noise signal from the noise estimation signal; anda combiner configured to combine the decoded frame of the decoded audio signal and the comfort noise signal in order to acquire an audio output signal, in such way that the decoded frame in the audio output signal comprises artificial noise.2. A decoder according to claim 1 , wherein the decoded frame is an active frame.3. A decoder according to claim 1 , wherein the decoded frame is an inactive frame.4. A decoder according to claim 1 , wherein the noise estimating device comprises a spectral analysis device configured to create an analysis signal comprising the level and the spectral shape of the noise in the decoded audio signal and a noise estimation producing device configured to produce the noise estimation signal ...

Device and method for encoding or decoding multichannel signals using frame control synchronization

【課題】効率的で低遅延を達成する多チャネル信号符号化装置を提供する。【解決手段】多チャネル信号符号化装置は、チャネルのサンプリング値のブロック系列を周波数ドメイン表現へ変換する時間－スペクトル変換部と、スペクトル値のブロック系列にジョイント多チャネル処理を適用してスペクトル値のブロックの結果系列を取得する多チャネル処理部と、スペクトル値のブロックの結果系列をサンプリング値のブロックの出力系列を含む時間ドメイン表現へ変換するスペクトル－時間変換部と、サンプリング値のブロックの出力系列を符号化して符号化済み多チャネル信号を得るコア符号器とを含む。コア符号器は、第１フレーム制御に従って作動してフレーム系列を提供し、１フレームは開始フレーム境界と終了フレーム境界とによって区切られている。時間－スペクトル変換部又はスペクトル－時間変換部は、第１フレーム制御と同期する第２フレーム制御に従って作動する。【選択図】図１

Process for producing a multilayer composite molding

Verfahren (100) zur Herstellung eines mehrschichtigen Verbundformteils (210) in einem aus Unterteil (201) und Oberteil (202) bestehenden Presswerkzeug (200) gekennzeichnet durch folgende Schritte: für eine untere Schicht (211) werden mit Bindemittel versehene Partikel in eine Form im Unterteil (201) des Presswerkzeugs (200) eingefüllt (Schritt 110); auf die erste Schicht (211) wird ein vorgepresstes Formteil (212) eingelegt (Schritt 120), das gröbere Partikel als die der unteren Schicht (211) aufweist; über dem eingelegten Formteil (212) entsprechend zu der unteren Schicht (211) wird mittels des Oberteils (202) des Presswerkzeugs (200) eine obere Schicht (213) gebildet, indem wiederum mit Bindemitteln versehene, feinere Partikel aufgefüllt werden; dann werden unter Druck- und Temperatur-Einwirkung die Schichten (211, 213) zusammen mit dem vorgepressten, eingelegten Formteil (212) zu dem mehrschichtigen Verbundformteil (210) verpresst. Method (100) for producing a multilayer composite molding (210) in a pressing tool (200) consisting of lower part (201) and upper part (202), characterized by the following steps: for a lower layer (211), binder-provided particles are filled in a mold in the lower part (201) of the pressing tool (200) (step 110); on the first layer (211) a pre-pressed molding (212) is inserted (step 120) having coarser particles than that of the lower layer (211); above the inserted molding (212) corresponding to the lower layer (211), an upper layer (213) is formed by means of the upper part (202) of the pressing tool (200), in turn, filling with finer particles provided with binders; then, under pressure and temperature, the layers (211, 213) are pressed together with the pre-pressed, inserted molding (212) to the multilayer composite molding (210).

Device and method for generating a data stream and device and method for reading a data stream

Coding and decoding of audio signals by using intensity stereo and prediction processes

In the coding and decoding of stereo audio spectral values both the intensity stereo process and prediction are used in order to achieve high data compression. If intensity stereo coding is active in one section of scale factor bands, the prediction for the right channel in that range is deactivated, whereby the results of the prediction are not used to form the coded stereo audio spectral values. To allow further adaptation of the prediction for the right channel, the predictor of the right channel is fed with stereo audio spectral values for the channel, which again are intensity stereo decoded.

Method for signalling a noise substitution during audio signal coding

In a method for signalling a noise substitution when coding an audio signal, the time-domain audio signal is first transformed into the frequency domain to obtain spectral values. The spectral values are subsequently grouped together to form groups of spectral values. On the basis of a detection establishing whether a group of spectral values is a noisy group or not, a codebook is allocated to a non-noisy or tonal group by means of a codebook number for redundancy coding of the same. If a group is noisy, an additional codebook number which does not refer to a codebook is allocated to it in order to signal that this group is noisy and therefore does not have to be redundancy coded. By signalling noise substitution by means of a Huffman codebook number for noisy groups of spectral values, which are e.g. sections made up of scale factor bands which do not have to be redundancy coded, an opportunity is provided to indicate the presence of a noise substitution in a scale factor band in the bit stream syntax of the MPEG-2 Advanced Audio Coding (AAC) Standard without having to interfere with the basic coding structure and without having to meddle with the structure of the existing bit stream syntax.

Method for coding and decoding stereo audio spectral values

Verfahren zum Codieren von Stereoaudiospektralwerten, mit folgenden Schritten: Gruppieren der Stereoaudiospektralwerte in Skalenfaktorbänder (28), denen Skalenfaktoren zugeordnet sind; Bilden von Abschnitten, die jeweils aus mindestens einem Skalenfaktorband (28) bestehen; Codieren der Stereoaudiospektralwerte innerhalb wenigstens eines Abschnitts mit einer dem wenigstens einen Abschnitt zugeordneten Codiertabelle aus einer Mehrzahl von Codiertabellen, denen jeweils eine Codiertabellennummer zugeordnet ist, wobei die Codiertabellennummer der verwendeten Codiertabelle als Seiteninformationen zu den codierten Stereoaudiospektralwerten übertragen wird, wobei wenigstens eine zusätzliche Codiertabellennummer vorgesehen ist, die nicht auf eine Codiertabelle verweist, sondern für den Abschnitt, dem dieselbe zugeordnet ist, relevante Informationen anzeigt, wobei einem Abschnitt entweder die Codiertabellennummer oder die zusätzliche Codiertabellennummer zugeordnet ist. Method for coding stereo audio spectral values, comprising the following steps: Grouping the stereo audio spectral values into scale factor bands (28) associated with scale factors; Forming sections each consisting of at least one scale factor band (28); Encoding the stereo audio spectral values within at least a portion having an encoding table associated with the at least one portion from a plurality of encoding tables, each associated with an encoding table number, the encoding table number of the encoding table used being transmitted as side information to the encoded stereo audio spectral values; wherein at least one additional coding table number is provided which does not refer to a coding table but for the section to which it is associated displays relevant information, one section being assigned either the coding table number or the additional coding table number.

Apparatus and method for entropy coding information words and apparatus and method for decoding entropy coded information words

Apparatus and method for efficient synthesis of sinusoids and sweeps by employing spectral patterns

An apparatus for generating an audio output signal based on an encoded audio signal spectrum is provided. The apparatus comprises a processing unit (115) for processing the encoded audio signal spectrum to obtain a decoded audio signal spectrum comprising a plurality of spectral coefficients, wherein each of the spectral coefficients has a spectral location within the encoded audio signal spectrum and a spectral value, wherein the spectral coefficients are sequentially ordered according to their spectral location within the encoded audio signal spectrum so that the spectral coefficients form a sequence of spectral coefficients. Moreover, the apparatus comprises a pseudo coefficients determiner (125) for determining one or more pseudo coefficients of the decoded audio signal spectrum, each of the pseudo coefficients having a spectral value. Furthermore, the apparatus comprises a replacement unit (135) for replacing at least one or more pseudo coefficients by a determined spectral pattern to obtain a modified audio signal spectrum, wherein the determined spectral pattern comprises at least two pattern coefficients, wherein each of the at least two pattern coefficients has a spectral value. Moreover, the apparatus comprises a spectrum-time-conversion unit (145) for converting the modified audio signal spectrum to a time-domain to obtain the audio output signal.

Device and method for entropy encoding of information words and device and method for decoding entropy-encoded information words

A method and a device for entropy encoding and associated decoding make use of a code consisting on the one hand of a code table with reversible code words ( 12 ) and comprising on the other hand an escape region for information words to be coded which are located outside the region ( 14 ) defined by said code table. Said region can be selected in such a way that a major part of the information words is coded with symmetrical code words by the code table. On the one hand, it is thus possible to carry out, in addition to forward decoding, also backward decoding ( 24 ) and on the other hand, use of reversible code words allows for rapid recognition of errors in a code word stream transmitted over a non-ideal channel.

Encoding and decoding audio signals using intensity stereo and prediction

When coding and decoding stereophonic spectral values, both the intensity stereo process and a prediction process are used to achieve a high data compression. When an intensity stereo coding is active in a section composed of scaling factor bands (28), prediction for the right channel (R) is deactivated in this area, so that the prediction results are not used to build the coded stereophonic spectral values. The predictor of the right channel (R) is supplied with stereophonic spectral values for this channel which are in turn decoded by an intensity stereo process, so that prediction for the right channel (R) can continue to adapt itself.

Method for signaling noise substitution when encoding an audio signal

Apparatus and method for encoding and decoding an audio signal using downsampling or interpolation of scale parameters

An apparatus for encoding an audio signal (160), comprises: a converter (100) for converting the audio signal into a spectral representation; a scale parameter calculator (110) for calculating a first set of scale parameters from the spectral representation: a downsampler (130) for downsampling the first set of scale parameters to obtain a second set of scale parameters, wherein a second number of scale parameters in the second set of scale parameters is lower than a first number of scale parameters in the first set of scale parameters; a scale parameter encoder (140) for generating an encoded representation of the second set of scale parameters; a spectral processor (120) for processing the spectral representation using a third set of scale parameters, the third set of scale parameters having a third number of scale parameters being greater than the second number of scale parameters, wherein the spectral processor (120) is configured to use the first set of scale parameters or to derive the third set of scale parameters from the second set of scale parameters or from the encoded representation of the second set of scale parameters using an interpolation operation; and an output interface (150) for generating an encoded output signal (170) comprising information on the encoded representation of the spectral representation and information on the encoded representation of the second set of scale parameters.

Selecting pitch lag

There are provided methods and apparatus capable of performing a low complexity pitch detection procedure, e.g., for long term postfiltering, LTPF, encoding. One inventive apparatus (10, 60a, 110) for encoding an information signal including a plurality of frames, comprises: a first estimator (11) configured to obtain a first estimate (14, T 1 ), the first estimate being an estimate of a pitch lag for a current frame (13); a second estimator (12) configured to obtain a second estimate (16, T 2 ), the second estimate being another estimate of a pitch lag for the current frame (13), a selector (17) configured to choose (S103) a selected value (19, T best ) by performing a selection between the first estimate (14, T 1 ) and the second estimate (16, T 2 ) on the basis of at least one correlation measurement (23, 25), wherein the second estimator (12) is conditioned by the pitch lag (51, 19") selected at the previous frame so as to obtain the second estimate (16, T 2 ) for the current frame (13).

Method for coding an audio signal

A method for coding or decoding an audio signal combines the advantages of TNS processing and noise substitution. A time-discrete audio signal is initially transformed to the frequency domain in order to obtain spectral values of the temporal audio signal. Subsequently, a prediction of the spectral values in relation to frequency is carried out in order to obtain spectral residual values. Within the spectral residual values, areas are detected encompassing spectral residual values with noise properties. The spectral residual values in the noise areas are noise-substituted, whereupon information concerning the noise areas and noise substitution is incorporated into side information pertaining to a coded audio signal. Thus, considerable bit savings in case of transient signals can be achieved.

Process for coding and decoding stereophonic spectral values

A method of coding stereo audio spectral values first carries out grouping of those values in scale factor bands, with which scale factors are associated. Sections are formed next, each comprising at least one scale factor band. The spectral values are coded within at least one section with a code book assigned to the section, out of a plurality of code books each with a code book number assigned to it, the number of the code book used being transmitted as side information to the coded stereo audio spectral values. At least one additional code book number is provided, which does not refer to a code book but shows information relevant to the section to which it is assigned. A method of decoding stereo audio spectral values which are partly coded by the intensity stereo process and which have side information uses the relevant information, showing the additional code book numbers, to cancel the existing coding of the stereo audio spectral values.

Device and method for entropy encoding of information words and device and method for decoding entropy-encoded information words

A method and a device for entropy encoding and associated decoding make use of a code consisting on the one hand of a code table with reversible code words and comprising on the other hand an escape region for information words to be coded which are located outside the region defined by said code table. Said region can be selected in such a way that a major part of the information words is coded with symmetrical code words by the code table. On the one hand, it is thus possible to carry out, in addition to forward decoding, also backward decoding and on the other hand, use of reversible code words allows for rapid recognition of errors in a code word stream transmitted over a non-ideal channel.

Method for coding an audio signal digitalized at a low sampling rate

In a method for coding an audio signal digitalized at a low sampling rate a respective number of successive frequency lines of the digitalized audio signal which are assigned to a scale factor band are coded with the same scale factor, successive scale factor bands forming a region within which all the scale factor bands are coded with the same number of bits, which is determined according to the largest scale factor of the region. The frequency lines of at least the highest region of scale factor bands are coded with the scale factor 0. No scale factor is coded for at least the highest region.

Encoder for encoding an audio signal, audio transmission system and method for determining correction values

An encoder for encoding an audio signal comprises an analyzer configured for analyzing the audio signal and for determining analysis prediction coefficients from the audio signal. The encoder further comprises a converter configured for deriving converted prediction coefficients from the analysis prediction coefficients, a memory configured for storing a multitude of correction values and a calculator. The calculator comprises a processor configured for processing the converted prediction coefficients to obtain spectral weighting factors. The calculator further comprises a combiner configured for combining the spectral weighting factors and the multitude of correction values to obtain corrected weighting factors. A quantizer of the calculator is configured for quantizing the converted prediction coefficients using the corrected weighting factors to obtain a quantized representation of the converted prediction coefficients. The encoder comprises a bitstream former configured for forming an output signal based on the quantized representation of the converted prediction coefficients and based on the audio signal.

Apparatus and procedure to estimate a time difference between channels

Aparato para estimar una diferencia de tiempos entre canales entre una señal de un primer canal y una señal de un segundo canal, que comprende: un calculador (1020) para calcular un espectro de correlación cruzada para un bloque de tiempo de la señal del primer canal en el bloque de tiempo y de la señal del segundo canal en el bloque de tiempo; un estimador de características espectrales (1010) para estimar una característica de un espectro de la señal del primer canal o de la señal del segundo canal para el bloque de tiempo; un filtro de suavizado (1030) para suavizar el espectro de correlación cruzada a lo largo del tiempo usando la característica espectral para obtener un espectro de correlación cruzada suavizado; y un procesador (1040) para tratar el espectro de correlación cruzada suavizado para obtener la diferencia de tiempos entre canales. Apparatus for estimating a time difference between channels between a signal from a first channel and a signal from a second channel, comprising: a calculator (1020) for calculating a cross-correlation spectrum for a time block of the signal from the first channel in the time block and of the second channel signal in the time block; a spectral feature estimator (1010) for estimating a feature of a spectrum of the first channel signal or the second channel signal for the time block; a smoothing filter (1030) to smooth the cross-correlation spectrum over time using the spectral characteristic to obtain a smoothed cross-correlation spectrum; and a processor (1040) for processing the smoothed cross-correlation spectrum to obtain the time difference between channels.

An apparatus for encoding a speech signal employing acelp in the autocorrelation domain

An apparatus for encoding a speech signal by determining a codebook vector of a speech coding algorithm is provided. The apparatus comprises a matrix determiner (110) for determining an autocorrelation matrix R, and a codebook vector determiner (120) for determining the codebook vector depending on the autocorrelation matrix R. The matrix determiner (110) is configured to determine the autocorrelation matrix R by determining vector coefficients of a vector r, wherein the autocorrelation matrix R comprises a plurality of rows and a plurality of columns, wherein the vector r indicates one of the columns or one of the rows of the autocorrelation matrix R, wherein R(i , j)= r(|i~ j|), wherein R(i, j) indicates the coefficients of the autocorrelation matrix R, wherein i is a first index indicating one of a plurality of rows of the autocorrelation matrix R, and wherein j is a second index indicating one of the plurality of columns of the autocorrelation matrix R.

Apparatus and method for improved concealment of the adaptive codebook in acelp-like concealment employing improved pulse resynchronization

An apparatus for reconstructing a frame comprising a speech signal as a reconstructed frame is provided, said reconstructed frame being associated with one or more available frames, said one or more available frames being at least one of one or more preceding frames of the reconstructed frame and one or more succeeding frames of the reconstructed frame, wherein the one or more available frames comprise one or more pitch cycles as one or more available pitch cycles. The apparatus comprises a determination unit (210) for determining a sample number difference indicating a difference between a number of samples of one of the one or more available pitch cycles and a number of samples of a first pitch cycle to be reconstructed. Moreover, the apparatus comprises a frame reconstructor (220) for reconstructing the reconstructed frame by reconstructing, depending on the sample number difference and depending on the samples of said one of the one or more available pitch cycles, the first pitch cycle to be reconstructed as a first reconstructed pitch cycle. The frame reconstructor (220) is configured to reconstruct the reconstructed frame, such that the reconstructed frame completely or partially comprises the first reconstructed pitch cycle, such that the reconstructed frame completely or partially comprises a second reconstructed pitch cycle, and such that the number of samples of the first reconstructed pitch cycle differs from a number of samples of the second reconstructed pitch cycle.

Concept for coding mode switching compensation

A codec allowing for switching between different coding modes is improved by, responsive to a switching instance, performing temporal smoothing and/or blending at a respective transition.

An enhanced audio encoding/decoding device and method

An enhanced audio encoding device, which is consisted of a signal type analyzing module, a psychoacoustical analyzing module, a time-frequency mapping module, a quantization and entropy encoding module, a frequency-domain linear prediction and vector quantization module, and a bit stream multiplexing module, in which the singal type analyzing module is configured to anaylze the signal type of the input audio signal and output to the psychoacoustical analyzing module, the time-frequency mapping module and the bit stream multiplexing module, the frequency-domain linear prediction and vector quantization module is configured to perform a linear prediction of the frequency-domain coefficients and a multi-level vector quantization, and output the residual sequences to the quantization and entropy encoding module while outputting a lateral information to the bit stream multiplexing module. The device is adapted to perform a compressive encoding of the audio signal with high fidelity, which is sampled at multi sampling rates and has multi audio channel configurations, the device can support the audio signal whose sampling rate is between 8kHz and 192kHz and all possible audio channel configurations, and the range of the objective code rate of the audio encode/decode is very wide.

Linear prediction based audio coding using improved probability distribution estimation

Linear prediction based audio coding is improved by coding a spectrum composed of a plurality of spectral components using a probability distribution estimation determined for each of the plurality of spectral components from linear prediction coefficient information. In particular, the linear prediction coefficient information is available anyway. Accordingly, it may be used for determining the probability distribution estimation at both encoding and decoding side. The latter determination may be implemented in a computationally simple manner by using, for example, an appropriate parameterization for the probability distribution estimation at the plurality of spectral components. All together, the coding efficiency as provided by the entropy coding is compatible with probability distribution estimations as achieved using context selection, but its derivation is less complex. For example, the derivation may be purely analytically and/or does not require any information on attributes of neighboring spectral lines such as previously coded/decoded spectral values of neighboring spectral lines as is the case in spatial context selection.

Method of encoding an audio signal digitized at a low sampling frequency

An apparatus for encoding a speech signal employing acelp in the autocorrelation domain

An apparatus for encoding a speech signal by determining a codebook vector of a speech coding algorithm is provided. The apparatus comprises a matrix determiner (110) for determining an autocorrelation matrix R, and a codebook vector determiner (120) for determining the codebook vector depending on the autocorrelation matrix R. The matrix determiner (110) is configured to determine the autocorrelation matrix R by determining vector coefficients of a vector r , wherein the autocorrelation matrix R comprises a plurality of rows and a plurality of columns, wherein the vector r indicates one of the columns or one of the rows of the autocorrelation matrix R, wherein R(i , j) = r(¦i~ j¦), wherein R(i, j) indicates the coefficients of the autocorrelation matrix R, wherein i is a first index indicating one of a plurality of rows of the autocorrelation matrix R, and wherein j is a second index indicating one of the plurality of columns of the autocorrelation matrix R.

Generation of a comfort noise with high spectro-temporal resolution in discontinuous transmission of audio signals.

La invención da a conocer un decodificador de audio que está configurado para decodificar un flujo de bits para producir a partir de éste una señal de salida de audio, donde el flujo de bits comprende por lo menos una fase activa seguida por al menos una fase inactiva, donde el flujo de bits contiene, codificada en el mismo, por lo menos una trama de descriptor de inserción de silencio que describe un espectro de un ruido de fondo, donde el decodificador de audio comprende: un decodificador de descriptores de inserción de silencio configurado para decodificar la trama de descriptor de inserción de silencio a fin de reconstruir un espectro del ruido de fondo; un dispositivo decodificador configurado para reconstruir la señal de salida de audio a partir del flujo de bits durante la fase activa; un conversor espectral configurado para determinar un espectro de la señal de salida de audio un dispositivo estimador de ruido configurado para determinar un primer espectro del ruido de la señal de salida de audio sobre la base del espectro de la señal de salida de audio provista por el conversor espectral, donde el primer espectro del ruido de la señal de salida de audio tiene una resolución espectral más alta que el espectro del ruido de fondo; un conversor de resolución configurado para establecer un segundo espectro del ruido de la señal de salida de audio sobre la base del primer espectro del ruido de la señal de salida de audio, donde el segundo espectro del ruido de la señal de salida de audio tiene la misma resolución espectral que el espectro del ruido de fondo; un dispositivo de estimación del espectro del ruido de confort que consta de un dispositivo de cómputo de factores de escala configurado para calcular factores de escala para un espectro respecto del ruido de confort sobre la base del espectro del ruido de fondo provista por el decodificador de descriptores de inserción de silencio y sobre la base del segundo espectro del ruido de la señal de salida de audio ...

Low-complexity tonality-adaptive audio signal quantization

Apparatus and method for estimating an inter-channel time difference

An apparatus for estimating an inter-channel time difference between a first channel signal and a second channel signal, comprises: a calculator (1020) for calculating a cross- correlation spectrum for a time block from the first channel signal in the time block and the second channel signal in the time block; a spectral characteristic estimator (1010) for estimating a characteristic of a spectrum of the first channel signal or the second channel signal for the time block; a smoothing filter (1030) for smoothing the cross-correlation spectrum over time using the spectral characteristic to obtain a smoothed cross- correlation spectrum; and a processor (1040) for processing the smoothed cross- correlation spectrum to obtain the inter-channel time difference.

Method and apparatus for suppression of a predetermined message

A received TV or radio program is stored in a buffer (12) without being transmitted in response to a control signal (16) identifying the scheduled broadcast. The program so stored, having scheduled transmission no longer, is routed to a TV or radio playback device (14). A buffer read-out occurs at a time staggered with writing the program into a buffer, so that the TV or radio program stored in the buffer is ready for playback, while a later part of the received TV or radio program is stored in a buffer and the predetermined broadcast suppressed in response to the control signal.

Audio coding in the frequency and time domains using a cross processor for continuous initialization

Managing network device

Disclosed is a network device for managing a call between user terminals, that checks whether a first user terminal supports a usage of a first audio coding mode for the cali, and a second user terminal intends to use a second audio coding mode for the call, and, if the first user terminal supports the usage of the first audio coding mode for the call, and the second user terminal intends to use the second audio coding mode for the call, repacks first data of the call sent from the first user terminal to the second user terminal and packetized into first packets referring to the second audio coding mode, into second packets referring to the first audio coding mode; and repacks second data of the call sent from the second user terminal to the first user terminal and packetized into third packets referring to the second audio coding mode, into fourth packets referring to the first audio coding mode.

Engine braking method for an internal combustion engine having two exhaust-gas turbochargers connected in series

In an engine braking method for an internal combustion engine having two exhaust-gas turbochargers (2, 6) which are connected in series, the desired engine braking power is set by regulating the air mass flow rate through a bypass (12), which bypasses the compressor (4) which is closest to the engine, and the exhaust gas mass flow rate through a bypass (10), which bypasses the exhaust-gas turbine (3) which is closest to the engine.

Noise generation in audio codecs

An audio decoder for decoding a data stream to reconstruct an audio signal, includes a background noise estimator to determine a parametric background noise estimate based on a spectral decomposition representation of the audio signal so that the parametric background noise estimate spectrally describes a spectral envelope a background noise of the audio signal. A decoder reconstructs the audio signal from the data stream during an active phase. The audio decoder includes a parametric random generator, and a background noise generator to reconstruct the audio signal during an inactive phase by controlling the parametric random generator with the parametric background noise estimate. The decoder applies shaping a spectral decomposition of an excitation signal transform coded into the data stream according to linear prediction coefficients coded into the data stream. The background noise estimator uses the spectral decomposition of the excitation signal in determining the parametric background noise estimate.

Low-complexity tonality-adaptive audio signal quantization

The invention provides an audio encoder for encoding an audio signal (AS) so as to produce therefrom an encoded signal (ES), the audio encoder (1) comprising: a framing device (2) configured to extract frames (F) from the audio signal (AS); a quantizer (3) configured to map spectral lines (SL 1-32 ) of a spectrum signal (SPS) derived from the frame (F) of the audio signal (AS) to quantization indices (I 0 , I 1 ), wherein the quantizer (3) has a dead-zone (DZ), in which the input spectral lines (SL) are mapped to quantization index zero (I 0 ); and a control device (4) configured to modify the dead-zone (DZ); wherein the control device (4) comprises a tonality calculating device (5) configured to calculate at least one tonality indicating value (TI 5.32 ) for at least one spectrum line (SL 1-32 ) or for at least one group of spectral lines (SL 1-32 ), wherein the control device (4) is configured to modify the dead-zone (DZ) for the at least one spectrum line (SL 1-32 ) or the at least one group of spectrum lines (SL 1-32 ) depending on the respective tonality indicating value (TI 5-32 ).

Audio encoder and decoder using a frequency domain processor, a time domain processor, and a cross processor for continuous initialization

An audio encoder for encoding an audio signal, comprises: a first encoding processor (600) for encoding a first audio signal portion in a frequency domain, wherein the first encoding processor (600) comprises: a time frequency converter for converting the first audio signal portion into a frequency domain representation having spectral lines up to a maximum frequency of the first audio signal portion; a spectral encoder for encoding the frequency domain representation; a second encoding processor for encoding a second different audio signal portion in the time domain; a cross-processor (700) for calculating, from the encoded spectral representation of the first audio signal portion, initialization data of the second encoding processor (610), so that the second encoding processing (610) is initialized to encode the second audio signal portion immediately following the first audio signal portion in time in the audio signal; a controller configured for analyzing the audio signal and for determining, which portion of the audio signal is the first audio signal portion encoded in the frequency domain and which portion of the audio signal is the second audio signal portion encoded in the time domain; and an encoded signal former for forming an encoded audio signal comprising a first encoded signal portion for the first audio signal portion and a second encoded signal portion for the second audio signal portion.

Selecting pitch lag

There are proposed techniques (e.g., in apparatus, methods, programs) for selecting pitch lag. An apparatus (10, 60a, 110) for encoding an information signal including a plurality of frames. The apparatus may comprise a first estimator (11) configured to obtain a first estimate (14, Ti), the first estimate being an estimate of a pitch lag for a current frame (13). The apparatus may comprise a second estimator (12) configured to obtain a second estimate (16, T2), the second estimate being another estimate of a pitch lag for the current frame (13). A selector (17) may be configured to choose (S103) a selected value (19, T bes t) by performing a selection between the first estimate (14, T1) and the second estimate (16, T2) on the basis of a first and a second correlation measurements (23, 25).

Apparatus and method for processing an audio signal using a harmonic post-filter

An apparatus for processing an audio signal having associated therewith a pitch lag information and a gain information, comprises a domain converter (100) for converting a first domain epresentation of the audio signal into a second domain representation of the audio signal; and a harmonic post-filter (104) for filtering the second domain representation of the audio signal, wherein the post-filter is based on a transfer function comprising a numerator and a denominator, wherein the numerator comprises a gain value indicated by the gain information, and wherein the denominator comprises an integer part of a pitch lag indicated by the pitch lag information and a multi-tap filter depending on a fractional part of the pitch lag.

Apparatus and method for error concealment in low-delay unified speech and audio coding (usac)

An apparatus (100) for generating spectral replacement values for an audio signal is provided. The apparatus (100) comprises a buffer unit (110) for storing previous spectral values relating to a previously received error-free audio frame. Moreover, the apparatus (100) comprises a concealment frame generator (120) for generating the spectral replacement values, when a current audio frame has not been received or is erroneous. The previously received error-free audio frame comprises filter information, the filter information having associated a filter stability value indicating a stability of a prediction filter. The concealment frame generator (120) is adapted to generate the spectral replacement values based on the previous spectral values and based on the filter stability value.

Method for coding an audio signal digitized at a low sampling rate

In a method for coding an audio signal digitized at a low sampling rate to obtain time domain audio samples. A frequency domain representation of the time domain audio samples is produced. The frequency domain representation includes successive frequency lines. These frequency lines are grouped into a plurality of scale factor bands. The successive frequency lines in a scale factor band are coded with the same scale factor. A plurality of regions is formed by grouping the scale factor bands, wherein successive scale factor bands form a region within which all the scale factors are coded with the same number of bits, which is determined according to the largest scale factor of the region. The scale factors assigned to scale factor bands within the highest region that includes the higher frequency successive frequency lines are set to zero. The frequency lines in the highest region are coded using the zero-valued scale factors that correspond to a multiplication factor of 1. The scale factors for the highest region, however, are not coded. Thus, the bits that would be required for coding these zero-valued scale factors are saved and can be used for a finer quantization of the rest of the spectrum. Additionally, this coding method when applied to ISO/IEC 13818-3 as a low sampling rate modification thereof only requires minimal changes with respect to this Standard.

Encoder, decoder and method for encoding and decoding audio content using parameters for enhancing a concealment

The present invention concerns an encoder for coding speech-like content and/or general audio content, wherein the encoder is configured to embed, at least in some frames, parameters in a bitstream, which parameters enhance a concealment in case an original frame is lost, corrupted or delayed, and a decoder for decoding speech-like content and/or general audio content, wherein the decoder is configured to use parameters which are sent later in time to enhance a concealment in case an original frame is lost, corrupted or delayed, as well as a method for encoding and a method for decoding.