Abstract: Provided is a communication apparatus for direct communication between networks of different types. The communication apparatus includes a transmission data selector determining whether or not data input from a first communication network is speech data, a data processor digitizing and packetizing the data transferred from the transmission data selector, and a modem for converting the digitized and packetized data into analog data and then directly transmitting the analog data to a second communication network different from the first communication network through a speech channel.

Abstract: A system and method may be configured to reconstruct an audio signal from transformed audio information. The audio signal may be resynthesized based on individual harmonics and corresponding pitches determined from the transformed audio information. Noise may be subtracted from the transformed audio information by interpolating across peak points and across trough points of harmonic pitch paths through the transformed audio information, and subtracting values associated with the trough point interpolations from values associated with the peak point interpolations. Noise between harmonics of the sound may be suppressed in the transformed audio information by centering functions at individual harmonics in the transformed audio information, the functions serving to suppress noise between the harmonics.

Abstract: In a pulse encoding and decoding method and a pulse codec, more than two tracks are jointly encoded, so that free codebook space in the situation of single track encoding can be combined during joint encoding to become code bits that may be saved. Furthermore, a pulse that is on each track and required to be encoded is combined according to positions, and the number of positions having pulses, distribution of the positions that have pulses on the track, and the number of pulses on each position that has a pulse are encoded separately, so as to avoid separate encoding performed on multiple pulses of a same position, thereby further saving code bits.

Abstract: A method and apparatus are provided for training a transformation matrix of a feature vector for an acoustic model. The method includes training the transformation matrix of the feature vector. The transformation matrix maximizes an objective function having a regularization term. The method further includes transforming the feature vector using the transformation matrix of the feature vector, and updating the acoustic model stored in a memory device using the transformed feature vector.

Abstract: For a bandwidth extension of an audio signal, in a signal spreader the audio signal is temporally spread by a spread factor greater than 1. The temporally spread audio signal is then supplied to a demicator to decimate the temporally spread version by a decimation factor matched to the spread factor. The band generated by this decimation operation is extracted and distorted, and finally combined with the audio signal to obtain a bandwidth extended audio signal. A phase vocoder in the filterbank implementation or transformation implementation may be used for signal spreading.

Abstract: An apparatus for determining a plurality of local center-of-gravity frequencies of a spectrum of an audio signal includes an offset determiner, a frequency determiner and an iteration controller. The offset determiner determines an offset frequency for each iteration start frequency of a plurality of iteration start frequencies based on the spectrum of the audio signal, wherein a number of discrete sample values of the spectrum is larger than a number of iteration start frequencies. The frequency determiner determines a new plurality of iteration start frequencies by increasing or reducing each iteration start frequency of the plurality of iteration start frequencies by the corresponding determined offset frequency. The iteration controller provides the new plurality of iteration start frequencies to the offset determiner for further iteration or provides the plurality of local center-of-gravity frequencies, if a predefined termination condition is fulfilled.

Abstract: A device and method for estimating a tonal stability of a sound signal include: calculating a current residual spectrum of the sound signal; detecting peaks in the current residual spectrum; calculating a correlation map between the current residual spectrum and a previous residual spectrum for each detected peak; and calculating a long-term correlation map based on the calculated correlation map, the long-term correlation map being indicative of a tonal stability in the sound signal.

Abstract: A method for improving decomposition of digital signals using training sequences is presented. A method for improving decomposition of digital signals using initialization is also provided. A method for sorting digital signals using frames based upon energy content in the frame is further presented. A method for utilizing user input for combining parts of a decomposed signal is also presented.

Abstract: In a pulse encoding and decoding method and a pulse codec, more than two tracks are jointly encoded, so that free codebook space in the situation of single track encoding can be combined during joint encoding to become code bits that may be saved. Furthermore, a pulse that is on each track and required to be encoded is combined according to positions, and the number of positions having pulses, distribution of the positions that have pulses on the track, and the number of pulses on each position that has a pulse are encoded separately, so as to avoid separate encoding performed on multiple pulses of a same position, thereby further saving code bits.

Abstract: A method of operating an audio processing device to improve a user's perception of an input sound includes defining a critical frequency fcrit between a low frequency range and a high frequency range, receiving an input sound by the audio processing device, and analyzing the input sound in a number of frequency bands below and above the critical frequency. The method also includes defining a cut-off frequency fcut below the critical frequency fcrit, identifying a source frequency band above the cut-off frequency fcut, and extracting an envelope of the source band. Further, the method identifying a corresponding target band below the critical frequency fcrit, extracting a phase of the target band, and combining the envelope of the source band with the phase of the target band.

Abstract: Disclosed is an audio encoding device which removes unnecessary inter-channel parameters from the subject to be encoded, improving the encoding efficiency thereby. In this audio encoding device, a principal component analysis unit (301) converts an inputted left signal {Lsb(f)} and an inputted right signal {Rsb(f)} into a principal component signal {PCsb(f)} and an ambient signal {Asb(f)} and calculates for each sub-band, a rotation angle which indicates the degree of conversion; a monophonic encoding unit (303) encodes the principal component signal {Pcsb(f)}; a rotation angle encoding unit (302) encodes the angle of rotation {?b}; a local monophonic decoding unit (603) creates a decoded principal component signal; and a redundant parameter elimination unit (604) identifies the redundant parameters by analyzing the encoding quality of the decoded principal component signal and eliminates the redundant parameters from the signal to be encoded.

Abstract: An audio encoder for encoding segments of coefficients, the segments of coefficients representing different time or frequency resolutions of a sampled audio signal, the audio encoder including a processor for deriving a coding context for a currently encoded coefficient of a current segment based on a previously encoded coefficient of a previous segment, the previously encoded coefficient representing a different time or frequency resolution than the currently encoded coefficient. The audio encoder further includes an entropy encoder for entropy encoding the current coefficient based on the coding context to obtain an encoded audio stream.

Abstract: A method of converting speech from the characteristics of a first voice to the characteristics of a second voice, the method comprising: receiving a speech input from a first voice, dividing said speech input into a plurality of frames; mapping the speech from the first voice to a second voice; and outputting the speech in the second voice, wherein mapping the speech from the first voice to the second voice comprises, deriving kernels demonstrating the similarity between speech features derived from the frames of the speech input from the first voice and stored frames of training data for said first voice, the training data corresponding to different text to that of the speech input and wherein the mapping step uses a plurality of kernels derived for each frame of input speech with a plurality of stored frames of training data of the first voice.

Abstract: An audio encoder has a first information sink oriented encoding branch, a second information source or SNR oriented encoding branch, and a switch for switching between the first encoding branch and the second encoding branch, wherein the second encoding branch has a converter into a specific domain different from the spectral domain, and wherein the second encoding branch furthermore has a specific domain coding branch, and a specific spectral domain coding branch, and an additional switch for switching between the specific domain coding branch and the specific spectral domain coding branch. An audio decoder has a first domain decoder, a second domain decoder for decoding a signal, and a third domain decoder and two cascaded switches for switching between the decoders.

Abstract: A method for decoding an audio signal in a decoder having a CELP-based decoder element including a fixed codebook component, at least one pitch period value, and a first decoder output, wherein a bandwidth of the audio signal extends beyond a bandwidth of the CELP-based decoder element. The method includes obtaining an up-sampled fixed codebook signal by up-sampling the fixed codebook component to a higher sample rate, obtaining an up-sampled excitation signal based on the up-sampled fixed codebook signal and an up-sampled pitch period value, and obtaining a composite output signal based on the up-sampled excitation signal and an output signal of the CELP-based decoder element, wherein the composite output signal includes a bandwidth portion that extends beyond a bandwidth of the CELP-based decoder element.

Abstract: A method for coding and decoding an audio signal or speech signal and an apparatus adopting the method are provided.

Abstract: A computer-implemented method and apparatus are disclosed for decoding an encoded data signal. In one embodiment, the method includes accessing, in a memory, a set of signal elements. The encoded data signal is received at a computing device. The signal includes signal fragments each having a projection value and an index value. The projection value has been calculated as a function of at least one signal element of the set of signal elements and at least a portion of the data signal. The index value associates its respective signal fragment with the at least one signal element used to calculate the projection value. The computing device determines amplitude values based on the projection values in the signal fragments. The decoded signal is determined using the amplitude values and the signal elements associated with the at least some of the signal fragments.

Abstract: In a pulse encoding and decoding method and a pulse codec, more than two tracks are jointly encoded, so that free codebook space in the situation of single track encoding can be combined during joint encoding to become code bits that may be saved. Furthermore, a pulse that is on each track and required to be encoded is combined according to positions, and the number of positions having pulses, distribution of the positions that have pulses on the track, and the number of pulses on each position that has a pulse are encoded separately, so as to avoid separate encoding performed on multiple pulses of a same position, thereby further saving code bits.

Abstract: A parameter estimation method for inactive voice signals and a system thereof and comfort noise generation method and system are disclosed. The method includes: for an inactive voice signal frame, performing time-frequency transform on a sequence of time domain signals containing the inactive voice signal frame to obtain a frequency spectrum sequence, calculating frequency spectrum coefficients according to the frequency spectrum sequence, performing smooth processing on the frequency spectrum coefficients, obtaining a smoothly processed frequency spectrum sequence according to the smoothly processed frequency spectrum coefficients, performing inverse time-frequency transform on the smoothly processed frequency spectrum sequence to obtain a reconstructed time domain signal, and estimating an inactive voice signal parameter according to the reconstructed time domain signal to obtain a frequency spectrum parameter and an energy parameter.

Abstract: A voice quality conversion device including: a target vowel vocal tract information hold unit holding target vowel vocal tract information of each vowel indicating target voice quality; a vowel conversion unit (i) receiving vocal tract information with phoneme boundary information of the speech including information of phonemes and phoneme durations, (ii) approximating a temporal change of vocal tract information of a vowel in the vocal tract information with phoneme boundary information applying a first function, (iii) approximating a temporal change of vocal tract information of the same vowel held in the target vowel vocal tract information hold unit applying a second function, (iv) calculating a third function by combining the first function with the second function, and (v) converting the vocal tract information of the vowel applying the third function; and a synthesis unit synthesizing a speech using the converted information.

Abstract: Content signal recognition is based on a multi-axis filtering of the content signal. The signatures are calculated, formed into data structures and organized in a database for quick searching and matching operations used in content recognition. For content recognition, signals are sampled and transformed into signatures using the multi axis filter. The database is searched to recognize the signals as part of a content item in the database. Using the content identification, content metadata is retrieved and provided for a variety of applications. In one application, the metadata is provided in response to a content identification request.

Abstract: Provided are a method and an apparatus for encoding and decoding an audio signal. A method for encoding an audio signal includes receiving a transformed audio signal, dividing the transformed audio signal into a plurality of subbands, performing a first sinusoidal pulse coding operation on the subbands, determining a performance region of a second sinusoidal pulse coding operation among the subbands on the basis of coding information of the first sinusoidal pulse coding operation, and performing the second sinusoidal pulse coding operation on the determined performance region, wherein the first sinusoidal pulse coding operation is performed variably according to the coding information. Accordingly, it is possible to further improve the quality of a synthesized signal by considering the sinusoidal pulse coding of a lower layer when encoding or decoding an audio signal in an upper layer by a layered sinusoidal pulse coding scheme.

Abstract: Signal separation techniques based on frequency dependency are described. In one implementation, a blind signal separation process is provided that avoids the permutation problem of previous signal separation processes. In the process, two or more signal sources are provided, with each signal source having recognized frequency dependencies. The process uses these inter-frequency dependencies to more robustly separate the source signals. The process receives a set of mixed signal input signals, and samples each input signal using a rolling window process. The sampled data is transformed into the frequency domain, which provides channel inputs to the inter-frequency dependent separation process. Since frequency dependencies have been defined for each source, the process is able to use the frequency dependency to more accurately separate the signals.

Abstract: A method for decoding an audio signal having a bandwidth that extends beyond a bandwidth of a CELP excitation signal in an audio decoder including a CELP-based decoder element. The method includes obtaining a second excitation signal having an audio bandwidth extending beyond the audio bandwidth of the CELP excitation signal, obtaining a set of signals by filtering the second excitation signal with a set of bandpass filters, scaling the set of signals using a set of energy-based parameters, and obtaining a composite output signal by combining the scaled set of signals with a signal based on the audio signal decoded by the CELP-based decoder element.

Abstract: An apparatus for encoding an audio signal includes the windower for windowing a first block of the audio signal using an analysis window having an aliasing portion and a further portion. The apparatus furthermore includes a processor for processing the first sub-block of the audio signal associated with the aliasing portion by transforming the sub-block from a domain into a different domain subsequent to windowing the first sub-block to obtain the processed first sub-block, and for processing a second sub-block of the audio signal associated with the further portion by transforming the second sub-block from the domain into the different domain before windowing the second sub-block to obtain a processed second sub-block. Thus, a critically sampled switch between two coding modes can be obtained.

Abstract: An electronic device for determining a set of pitch cycle energy parameters is described. The electronic device includes a processor and executable instructions stored in memory. The electronic device obtains a frame, a set of filter coefficients and a residual signal based on the frame and the set of filter coefficients. The electronic device determines a set of peak locations based on the residual signal and segments the residual signal such that each segment includes one peak. The electronic device determines a first set of pitch cycle energy parameters based on a frame region between two consecutive peak locations and maps regions between peaks in the residual signal to regions between peaks in a synthesized excitation signal to produce a mapping. The electronic device determines a second set of pitch cycle energy parameters based on the first set of pitch cycle energy parameters and the mapping.

Abstract: Methods and systems to facilitate communications between users via different modalities. A method includes identifying, by a first user device, a voice call originating from a second user device, and presenting a user interface to a user of the first user device, where the user interface provides an option to respond to the voice call by voice and an option to respond to the voice call in a text form. The method further includes detecting that the user of the first user device has selected the option to respond to the voice call in the text form, and causing a user response to the voice call to be converted into voice data for the second user device.

Abstract: A method of encoding an audio signal, where signals including two or more channel signals are downmixed to a mono signal, the mono signal is divided into a low-frequency signal and a high-frequency signal, the low-frequency signal is encoded through algebraic code excited linear prediction (ACELP) or transform coded excitation (TCX), and the high-frequency signal is encoded using the low-frequency signal. A method of decoding of an audio signal, a low-frequency signal encoded through ACELP or TCX is decoded, a high-frequency signal is decoded using the low-frequency signal, the low-frequency signal and the high-frequency signal are combined to generate a mono signal, and the mono signal is upmixed by decoding spatial parameters regarding signals including two or more channel signals.

Abstract: An original loudness level of an audio signal is maintained for a mobile device while maintaining sound quality as good as possible and protecting the loudspeaker used in the mobile device. The loudness of an audio (e.g., speech) signal may be maximized while controlling the excursion of the diaphragm of the loudspeaker (in a mobile device) to stay within the allowed range. In an implementation, the peak excursion is predicted (e.g., estimated) using the input signal and an excursion transfer function. The signal may then be modified to limit the excursion and to maximize loudness.

Abstract: A frequency spectrum processing apparatus and method using a source filter are disclosed. The frequency spectrum processing apparatus may include a first excitation spectrum generation unit to generate a first excitation spectrum using a tonal excitation spectrum according to an input signal and a gain of the tonal excitation spectrum, a second excitation spectrum generation unit to generate a second excitation spectrum using a non-tonal excitation spectrum according to the input signal and a gain of the non-tonal excitation spectrum, and an output spectrum generation unit to generate an output spectrum using the first excitation spectrum and the second excitation spectrum.

Abstract: The present invention relates to a method and apparatus for processing a voice signal, and the voice signal encoding method according to the present invention comprises the steps of: generating transform coefficients of sine wave components forming an input voice signal by transforming the sine wave components; determining transform coefficients to be encoded from the generated transform coefficients; and transmitting indication information indicating the determined transform coefficients, wherein the indication information may include position information, magnitude information, and sign information of the transform coefficients.

Abstract: A method and an apparatus for encoding and decoding audio signals using adaptive sinusoidal coding are provided. The audio signal encoding method includes the steps of dividing a synthesized audio signal into a plurality of sub-bands, calculating the energy of each sub-band, selecting a predetermined number of sub-bands having a relatively large amount of energy from the sub-bands, and performing sinusoidal coding with regard to the selected sub-bands. Application of sinusoidal coding based on consideration of the amount of energy of each sub-band of the synthesized signal improves the quality of the synthesized signal more efficiently.

Abstract: Provided are a method and an apparatus for encoding and decoding an audio signal. A method for encoding an audio signal includes receiving a transformed audio signal, dividing the transformed audio signal into a plurality of subbands, performing a first sinusoidal pulse coding operation on the subbands, determining a performance region of a second sinusoidal pulse coding operation among the subbands on the basis of coding information of the first sinusoidal pulse coding operation, and performing the second sinusoidal pulse coding operation on the determined performance region, wherein the first sinusoidal pulse coding operation is performed variably according to the coding information. Accordingly, it is possible to further improve the quality of a synthesized signal by considering the sinusoidal pulse coding of a lower layer when encoding or decoding an audio signal in an upper layer by a layered sinusoidal pulse coding scheme.

Abstract: A method for encoding an audio signal including: processing a selected subset of a lower series of samples forming a lower frequency spectral band of the audio signal and a higher series of samples forming a higher frequency spectral band of the audio signal to parametrically encode the higher series of samples forming the higher frequency spectral band by identifying a sub-series of the lower series of samples.

Abstract: A method and apparatus for generating synthesis audio signals are provided. The method includes decoding a bitstream; splitting the decoded bitstream into n sub-band signals; generating n transformed sub-band signals by transforming the n sub-band signals in a frequency domain; and generating synthesis audio signals by respectively multiplying the n transformed sub-band signals by values corresponding to synthesis filter bank coefficients.

Abstract: The present technology provides robust, high quality dereverberation of an acoustic signal which can overcome or substantially alleviate the problems associated with the diverse and dynamic nature of the surrounding acoustic environment. The present technology utilizes acoustic signals received from a plurality of microphones to carry out a multi-faceted analysis which accurately identifies reverberation based on the correlation between the acoustic signals. Due to the spatial distance between the microphones and the variation in reflection paths present in the surrounding acoustic environment, the correlation between the acoustic signals can be used to accurately determine whether portions of one or more of the acoustic signals contain desired speech or undesired reverberation. These correlation characteristics are then used to generate signal modifications applied to one or more of the received acoustic signals to preserve speech and reduce reverberation.

Abstract: Provided are a multi-stage speech recognition apparatus and method. The multi-stage speech recognition apparatus includes a first speech recognition unit performing initial speech recognition on a feature vector, which is extracted from an input speech signal, and generating a plurality of candidate words; and a second speech recognition unit rescoring the candidate words, which are provided by the first speech recognition unit, using a temporal posterior feature vector extracted from the speech signal.

Abstract: The invention concerns a method for trained discrimination and attenuation of echoes of a digital audio signal generated from a transform coding, which consists, for each current frame of the signal. In comparing (A) in real time, in at least one frequency band a variable derived from one characteristic of the echo generating signal with that of a non-echo generating signal at a threshold value, and deducing therefrom (B) the existence or non-existence (C) of an echo derived from the transform coding, discriminating the existence of the echo and defining (D) a false alarm zone in the high-energy parts of the digital audio signal, determining an initial processing and attenuating the echoes (E) in the parts complementary to the low-energy false alarm zone and inhibiting (F) the attenuation of echoes in the false alarm zone. The invention is applicable to the technology of coders/decoders in particular hierarchical coders/decoders.

Abstract: An adaptive time/frequency-based encoding mode determination apparatus including a time domain feature extraction unit to generate a time domain feature by analysis of a time domain signal of an input audio signal, a frequency domain feature extraction unit to generate a frequency domain feature corresponding to each frequency band generated by division of a frequency domain corresponding to a frame of the input audio signal into a plurality of frequency domains, by analysis of a frequency domain signal of the input audio signal, and a mode determination unit to determine any one of a time-based encoding mode and a frequency-based encoding mode, with respect to the each frequency band, by use of the time domain feature and the frequency domain feature.

Abstract: A method and device are provided for coding or decoding a digital audio signal by transform using analysis or synthesis weighting windows applied to sample frames. The method includes an irregular sampling of an initial window provided for a transform of given initial size N, to apply a secondary transform of size M different from N.

Abstract: Disclosed is a spectral smoothing device with a structure whereby smoothing is performed after a nonlinear conversion has been performed for a spectrum calculated from an audio signal, and with which the amount of processing calculation is significantly reduced while maintaining excellent audio quality. With this spectral smoothing device, a sub band division unit (102) divides an input spectrum into multiple sub bands; a representative value calculation unit (103) calculates a representative value for each sub band using an arithmetic mean and a geometric mean; with respect to each representative value, a nonlinear conversion unit (104) performs a nonlinear conversion the characteristic of which is further emphasized as the value increases; and a smoothing unit (105) that smoothes the representative value which has undergone the nonlinear conversion for each sub band, at the frequency domain.

Abstract: A system and method for merging audio data streams receive audio data streams from separate inputs, independently transform each data stream from the time to the frequency domain, and generate separate feature data sets for the transformed data streams. Feature data from each of the separate feature data sets is selected to form a merged feature data set that is output to a decoder for recognition purposes. The separate inputs can include an ear microphone and a mouth microphone.

Abstract: The invention provides a compensation method for audio frame loss in a MDCT domain, the method comprising: when a frame currently lost is a Pth frame, obtaining a set of frequencies to be predicted, and for each frequency in the set, using phases and amplitudes of a plurality of frames before a (P?1)th frame in a MDCT-MDST domain to predict a phase and an amplitude of the Pth frame, and using the predicted phase and amplitude to obtain a MDCT coefficient of the Pth frame at each corresponding frequency; for a frequency outside the set, using MDCT coefficients of a plurality of frames before the Pth frame to calculate a MDCT coefficient value of the Pth frame at the frequency; performing an IMDCT for the MDCT coefficients of the Pth frame to obtain a time domain signal of the Pth frame.

Abstract: A mobile wireless system and method are described for distributing emergency alert messages to mobile wireless devices in multiple languages. The emergency alert system receives an alert message including a geographic area identification, and a text alert. The emergency alert system renders one or more translations of the text alert from the text alert. The one or more translated versions of the text alert are provided in particular foreign languages based upon designated foreign languages corresponding to the geographic area identification. The emergency alert system transmits the text alert and one or more translated text alerts commercial mobile wireless service provider networks for broadcasting the text alert and the translated text alert(s) via mobile wireless transmitters having a coverage area falling within a region corresponding to the geographic area identification.

Abstract: In this invention, the design of the Huffman table can be done offline with a large input sequence database. The range of the quantization indices (or differential indices) for Huffman coding is identified. For each value of range, all the input signal which have the same range will be gathered and the probability distribution of each value of the quantization indices (or differential indices) within the range is calculated. For each value of range, one Huffman table is designed according to the probability. And in order to improve the bits efficiency of the Huffman coding, apparatus and methods to reduce the range of the quantization indices (or differential indices) are also introduced.

Abstract: An input signal, in the form of a sequence of feature vectors, is transformed to an output signal by first storing parameters of a model of the input signal in a memory. Using the vectors and the parameters, a sequence of vectors of hidden variables is inferred. There is at least one vector hn of hidden variables hi,n for each feature vector xn, and each hidden variable is nonnegative. The output signal is generated using the feature vectors, the vectors of hidden variables, and the parameters. Each feature vector xn is dependent on at least one of the hidden variables hi,n for the same n. The hidden variables are related according to h i , n = ? j , l ? ? c i , j , l ? ? l , n ? h j , n - 1 , where j and l are summation indices. The parameters include non-negative weights ci,j,l, and ?l,n are independent non-negative random variables.

Abstract: The signal processing is based on the concept of using a time-domain aliased frame as a basis for time segmentation and spectral analysis, performing segmentation in time based on the time-domain aliased frame and performing spectral analysis based on the resulting time segments. The time resolution of the overall “segmented” time-to-frequency transform can thus be changed by simply adapting the time segmentation to obtain a suitable number of time segments based on which spectral analysis is applied. The overall set of spectral coefficients, obtained for all the segments, provides a selectable time-frequency tiling of the original signal frame.

Abstract: A hierarchical audio coding, decoding method and system are provided. The method includes dividing frequency domain coefficients of an audio signal after MDCT into a plurality of coding sub-bands, quantizing and coding amplitude envelope values of coding sub-bands; allocating bits to each coding sub-band of the core layer, quantizing and coding core layer frequency domain coefficients to obtain coded bits of core layer frequency domain coefficients; calculating the amplitude envelope value of each coding sub-band of the core layer residual signal; allocating bits to each coding sub-band of the extended layer, quantizing and coding the extended layer coding signal to obtain coded bits of the extended layer coding signal; multiplexing and packing amplitude value envelope coded bits of each coding sub-band composed by core layer and extended layer frequency domain coefficients, core layer frequency coefficients coded bits, and extended layer coding signal coded bits, then transmitting to the decoding end.

Abstract: The present disclosure relates to a signal analyzer for processing an overlapped input signal frame comprising 2N subsequent input signal values. The signal analyzer comprises: a windower adapted to window the overlapped input signal frame to obtain a windowed signal, wherein the windower is adapted to zero M+N/2 subsequent input signal values of the overlapped input signal frame, wherein M is equal or greater than 1 and smaller than N/2; and a transformer adapted to transform the remaining 3N/2?M subsequent windowed signal values of the windowed signal using N?M sets of transform parameters to obtain a transformed-domain signal comprising N?M transformed-domain signal values.

Abstract: The invention relates to the analysis of characteristics of audio and/or video signals for the generation of audio-visual content signatures. To determine an audio signature a region of interest for example of high entropy—is identified in audio signature data. This region of interest is then provided as an audio signature with offset information. A video signature is also provided.