Speech or audio signal analysis-synthesis techniques for redundancy reduction, e.g. in vocoders and coding or decoding of speech or audio signals, e.g. for compression or expansion, source-filter models or and psychoacoustic analysis (G10L19)

G   Physics(388509)
G10L19                     Speech or audio signal analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; coding or decoding of speech or audio signals, e.g. for compression or expansion, source-filter models or; psychoacoustic analysis(589)

Watermark generator, watermark decoder, method of generating watermark signal, method of generating binary message data depending on watermarked signal and computer program based on improved synchronisation concept // 2614855
FIELD: data processing.SUBSTANCE: invention relates to means of generating, encoding and decoding a watermark. Watermark generator, which generates an electronic watermark signal based on binary message data, including in its structure an information spreader configured to spread an information unit to a plurality of time-frequency-domain values, to obtain a spread information representation. Watermark generator also includes a synchronization inserter configured to multiplicatively combine spread information representation with a synchronization spread sequence to obtain a combined information-synchronization representation. Watermark generator also includes watermark signal provider configured to provide watermark signal on basis of combined information-synchronization representation. Also described are a watermark decoder, methods and computer program.EFFECT: technical result consists in ensuring reliable synchronisation when transmitting watermark.19 cl, 41 dwg

Advanced stereo coding based on combination of adaptively selectable left/right or mid/side stereo coding and of parametric stereo coding // 2614573
FIELD: information technology; data processing.SUBSTANCE: invention relates to means of encoding and decoding audio signals. Downmix signal and a residual signal are generated based on a stereo signal. Difference in intensity between channels and cross-correlation between channels is determined. Preferably, parametric stereo coding parameters are time- and frequency-dependent. Transform stage generates a pseudo left/right stereo signal by performing a transform based on downmix signal and residual signal. Pseudo stereo signal is processed by a perceptual stereo encoder. For stereo coding left/right encoding or mid/side coding may be used. Preferably selection between left/right coding and mid/side coding is depending on time and frequency.EFFECT: technical result consists in improvement of quality of signal coding.20 cl, 26 dwg
Decorrelation of signals in audio data processing system // 2614381
FIELD: data processing.SUBSTANCE: invention relates to signal processing. Method of processing audio data includes receiving from a bit stream audio data, corresponding to a row of audio channels, application of decorrelation process to some of audio data, wherein decorrelation process involves using a decorrelation algorithm, active completely on real-valued coefficients.EFFECT: reduced complexity of coding and decoding algorithms owing to decorrelation of only real-valued coefficients.16 cl, 37 dwg

Data encoding method, data decoding method, encoder, decoder and codec // 2613031
FIELD: physics, computer engineering.SUBSTANCE: invention relates to encoding and decoding means. Correspondence of one or more portions of source data with one or more elements within one or more database is set up said one or more elements represent a respective one or more data block, and reference values which connect said one or more portions of input data with said one or more respective element is recorded. Reference values are included in the encoded data together with said one or more databases and/or information identifying said one or more databases. Encoded data including reference values and informaion on area identifiers, and information on one or more database is received. Reference values are decoded from the encoded data. One or more element from said one or more databases is extracted according to said reference values, wherein one or more said elements represents one or more corresponding data block. One or more said data block is formed for collection of corresponding outputting decoded data.EFFECT: invention improves the efficiency of data encoding/decoding.30 cl, 5 tbl, 3 dwg

Frequency emphasizing for lpc-based encoding in frequency domain // 2612589
FIELD: information technology.SUBSTANCE: invention relates to encoding and decoding audio signals. Audio encoder comprises a combination of a linear predictive coding filter having multiple linear predictive coding coefficients (LC) and a converter from the time domain into the frequency domain, herewith the said combination is configured with the ability of filtration and conversion of frame (FI) of audio signal (AS) into the frequency domain in order to output spectrum (SP) basing on the said frame (FI) and linear predictive coding coefficients (LC); a low-frequency emphasizing unit configured able to calculate processed spectrum (PS) basing on spectrum (SP), herewith spectral lines (SL) of processed spectrum (PS) representing a lower frequency than reference spectral line (RSL) are emphasized; and a control device made able to control the calculation of processed spectrum (PS) by means of the low-frequency emphasizing unit depending upon linear predictive coding coefficients (LC) of the linear predictive coding filter.EFFECT: technical result is ensuring the minimum of audible artefacts of encoding in the output audio signal at low frequencies by low-frequency emphasizing at the encoder side and deemphasizing at the decoder side.28 cl, 10 dwg

Control over phase coherency for harmonic signals in perceptual audio codecs // 2612584
FIELD: acoustics.SUBSTANCE: invention relates to means for control over phase coherency for harmonic signals in perceptual audio codecs. Decoder comprises a decoding unit and a phase adjustment unit. Decoding unit is adapted for decoding an encoded audio signal to obtain a decoded audio signal. Phase adjustment unit is adapted for adjustment of the decoded audio signal to obtain an adjusted in phase audio signal. Phase adjustment unit is configured able to receive control information depending on the vertical phase coherence of the encoded audio signal. Besides, the phase adjustment unit is adapted for adjustment of the decoded audio signal basing on control information.EFFECT: technical result is improvement the audio signal quality.18 cl, 9 dwg

Audio encoding device, audio encoding method, audio encoding software, audio decoding device, audio decoding method and audio decoding software // 2612581
FIELD: acoustics; information technologies.SUBSTANCE: invention relates to audio encoding and decoding devices. Audio signal transmission device for audio signal encoding includes audio encoding unit, which encodes audio signal, and supplementary information encoding unit, which calculates and encodes supplementary information from predicted signal. Audio signal receiving device for audio code decoding and audio signal output includes audio code buffer, which detects packet loss based on audio packet reception status, audio parameters decoding unit, which decodes audio code, when audio packet is received correctly, supplementary information decoding unit, which decodes supplementary information code, when audio packet is received correctly, supplementary information collecting unit, which collects supplementary information obtained by supplementary information code decoding, missing audio parameters processing unit, which outputs audio parameter, when audio packet loss is detected, and audio synthesis unit, which synthesizes decoded audio from audio parameter.EFFECT: technical result consists in reduction of delay in audio recovery after packet loss when encoding.8 cl, 43 dwg

Signal processor, window provider, coded media signal, signal processing method and method of forming windows // 2611986
FIELD: information technology.SUBSTANCE: invention relates to signal encoding and is intended for processing a signal with window weight coefficients. Signal processor for providing a processed version of an input signal depending on input signal comprises a windower, configured to window a portion of input signal, or of a pre-processed version thereof, in dependence on a signal processing window, described by signal processing window values for a plurality of window value index values, in order to obtain processed version of input signal. Signal processor also comprises a window provider for providing signal processing window values for a plurality of window value index values in dependence on one or more window shape parameters.EFFECT: technical result is high efficiency of encoding by adaptation of window characteristics to characteristics of input spectrum.6 cl, 16 dwg

Transform encoding/decoding of harmonic audio signals // 2611017
FIELD: information technology.SUBSTANCE: invention relates to means of encoding and decoding of harmonic audio signals. Encoder for encoding frequency transform coefficients of a harmonic audio signal includes following elements: a peak locator, configured to locate spectral peaks having magnitudes exceeding a predetermined frequency dependent threshold. Peak region encoder configured to encode peak regions including and surrounding located peaks. Low-frequency set encoder configured to encode at least one low-frequency set of coefficients outside the peak regions and below a crossover frequency that depends on ] number of bits used to encode peak regions. Noise-floor gain encoder, configured to encode a noise-floor gain of at least one high-frequency set of not yet encoded coefficients outside peak regions.EFFECT: technical result consists in improvement of quality of encoded harmonic audio signal.20 cl, 23 dwg, 1 tbl

Calculation of converter signal-noise ratio with reduced complexity // 2610588
FIELD: sound.SUBSTANCE: present invention relates to sound encoding and decoding facilities. Audio encoder includes conversion module, configured to determine set of spectral ratios based on said audio signal frame. Besides, encoder comprises encoding module with floating point, made with possibility to determine set of scaling factors and set of scaled values based on said set of spectral ratios; and encoding said set of scaling factors to obtain encoded set scaling factors. In addition, encoder comprises bits distribution and quantization module, configured to determine total number of available bits for set of scaled values quantization based on data transfer first target rate and based on number of bits used for set of coded scaling factors; determination of first control parameter, which serves as sign of total number of available bits distribution for quantization of scaled values from set of scaled values; and for set of scaled values quantization.EFFECT: technical result consists in reduction of computational complexity of bits distribution process used in audio encoding/decoding.32 cl, 12 dwg

ultichannel audio playback // 2610416
FIELD: acoustics.SUBSTANCE: invention relates to playback of multichannel audio and is used, in particular, in a home theater/surround sound playback system using wireless speakers units. Multichannel audio playback system comprises an audio playback module, which generates audio signals for multiple audio channels. Multiple interchangeable speakers units comprises a storage battery and an audio converter to playback an audio signal. Charging unit is connected to a specific audio channel and has a charging source, which can charge a storage battery connected to a speaker unit. Binding circuit can bind the speaker unit when it is connected to the charging unit with the first audio channel. System can continuously associate specific channels with charging units (or passive bases of speakers, which can not perform charging), and the interchangeable speaker units can be adapted depending on their connection. This approach may enable to perform charging a speaker unit (possibly, a wireless one) just by replacing this speaker unit with one of those connected to the charging unit.EFFECT: technical result is increased flexibility and easier operation by reducing the need for wire connections.15 cl, 16 dwg

Harmonic audio frequency band expansion // 2610293
FIELD: physics.SUBSTANCE: plurality of the gain values associated with the frequency band b and the plurality of the adjacent frequency bands for band b are received. It is determined whether the reconstructed corresponding frequency band b' contains a spectral peak. When band b' contains a spectral peak, the gain value associated with band b' is set as the first value based on the received plurality of the gain values; and otherwise, the gain value is set as the second value based on the received plurality of the gain values.EFFECT: improving quality of the harmonic audio frequency band expansion.12 cl, 10 dwg

ethod of detecting low-rate encoding protocols // 2610285
FIELD: physics, acoustics.SUBSTANCE: invention is intended for detecting low-rate speech encoding (LRSE) protocols . The technical result is achieved by increasing the number of dimensions of a measured vector of redundancy factors Z to L, L=Z+2 and accounting for the shifting effect of vector elements L by forming a quadratic reference matrix Lj et for all J known LRSE protocols, j=1, 2 , J. To this end, a digital stream Y is received during a given time interval T. A rectangular information matrix YKL is formed, rows of which are information units arranged in series one below the other. The vector of redundancy factors L is calculated, followed by element by element comparison of the measured vector L with rows Lj et(l) of all J quadratic reference matrices Lj et, determining deviation Pj(l) between the measured vector L and rows of all J reference matrices Lj et, making a decision in favour of the j-th LRSE protocol, for which is ensured minimum deviation Pj(l)min of the measured vector L from the l-th row of the j-th quadratic reference matrix Lj et.EFFECT: method of improving accuracy of detecting LRSE protocols.3 cl, 9 dwg

Device and method of spatial audio encoding streams combining based on geometry // 2609102
FIELD: acoustics.SUBSTANCE: invention relates to audio data composite stream generating means. Device contains demultiplexer to produce plurality of audio data single-level streams, where demultiplexer is adapted to receive one or more audio data input streams, where each audio data input stream contains one or more levels, where demultiplexer is adapted for demultiplexing of each of audio data input streams, having one or more levels, on two or more demultiplexed audio data streams, having exactly one level, so, that said two or more demultiplexed audio streams together contain said one or more levels of audio data input stream. Besides, device also contains matching module for generating of audio data composite stream, having one or more levels, based on said plurality of single-level audio streams.EFFECT: technical result consists in providing possibility of audio data composite stream generating.18 cl, 44 dwg

Device and methods for adaptation of audio information at spatial encoding of audio objects // 2609097
FIELD: acoustics.SUBSTANCE: invention relates to means for adaptation of input audio information encoding one or more audio objects. Input audio information includes two or more input downmix audio channels and additionally contains input parametric supplementary information. Adapted audio information includes one or more adapted downmix audio channels and additionally contains adapted parametric supplementary information. Device contains a downmix signal modifier for adaptation, depending on the adaptation information, of two or more input downmix audio channels to obtain one or more adapted downmix audio channels. Besides, the device comprises a means to adapt parametric supplementary information for adaptation, depending on the adaptation information, of the input parametric supplementary information to obtain the adapted parametric supplementary information.EFFECT: technical result is improvement of efficiency of adaptation of audio information to a specific target application scenario.13 cl, 9 dwg

Comfortable noise generation // 2609080
FIELD: acoustics.SUBSTANCE: invention relates to comfortable noise generating devices. Predetermined size buffer is configured, to store CN-parameters for SID (Silence Insertion Descriptor) frames and tightening active frames. Subset selecting device is configured to determine CN-parameters subset relevant for SID-frames, based on stored CN-parameters age and based on residual energies. Comfortable noise control parameters extraction device is configured, in order to use certain subset of CN-parameters for determining of CN control parameters for first SID-frame, following active signal frame.EFFECT: technical result consists in increase in perceptible sound quality.18 cl, 12 dwg

Level adjustment in time domain for decoding or encoding audio signals // 2608878
FIELD: acoustics.SUBSTANCE: invention relates to encoding, decoding and processing audio signals. Audio signal decoder for providing a decoded audio signal presentation based on an encoded audio signal presentation includes a cascade of preliminary processing of the decoder to obtain a plurality of frequency band signals from the encoded audio signal presentation, a clipping estimation module, a level shift module, a converter of the frequency domain into the time one and a level shift compensator. Clipping estimation module analyses the encoded audio signal representation and/or an additional information regarding amplification of frequency band signals to determine the current level shift coefficient. Level shift module shifts the frequency band signals levels in accordance with the level shift coefficient. Converter of the frequency domain into the time one converts the frequency band signals with shifted levels into the time domain presentation. Level shift compensator affects the time domain presentation for partial compensation of the corresponding level shift and for producing a significantly compensated presentation of the time domain.EFFECT: technical result is the possibility of the signal level adjustment within the dynamic range without loss of the data accuracy.16 cl, 17 dwg

Audio scenes encoding // 2608847
FIELD: acoustics.SUBSTANCE: invention relates to sound encoding and decoding. Exemplary embodiments offer encoding and decoding methods and corresponding coders and decoders for audio scene encoding and decoding, which contains at least one or more audio objects. Encoder generates bit stream, which contains step-down mixing signals and additional information, which comprises separate matrix elements for recovery matrix, which enables possibility for recovery of one or more audio objects in decoder.EFFECT: technical result is provision of less complex and more flexible recovery of audio objects.33 cl, 9 dwg

Effective attenuation of leading echo signals in digital audio signal // 2607418
FIELD: radio engineering.SUBSTANCE: invention relates to means of attenuation of leading echo signals in a digital audio signal. Attenuated are leading echo signals in a digital audio signal obtained by encoding by means of conversion. In the decoded signal a position of attack is detected. Determined is a zone of the leading echo signal preceding the position of attack detected in the decoded signal. Attenuation coefficients are calculated for each subunit of the zone of the leading echo signal depending on at least the frame, in which the attack was detected, and on the previous frame. Performed is the leading echo signal attenuation in subunits of the zone of the leading echo signal with the help of appropriate attenuation coefficients. Method of attenuating a leading echo signal additionally includes the step of using adaptive filtration to provide a spectral shape for the zone of the leading echo signal on the current frame prior to the detected position of attack.EFFECT: technical result is the possibility of attenuation of high frequencies and parasitic leading echo signals when decoding without transmitting by the encoder any auxiliary information.13 cl, 12 dwg

Device for providing upmix signal representation based on downmix signal representation, device for providing bitstream representing multichannel audio signal, methods, computer programs and bitstream representing multichannel audio signal using linear combination parameter // 2607267
FIELD: acoustics.SUBSTANCE: invention relates to devices for providing upmix signal representation based on downmix signal representation. Device includes distortion limiter formed to obtain modified imaging matrix using user-defined imaging matrix linear combination and specified imaging matrix depending on linear combination parameter. Device also includes signal processor, generated, to obtain upmix signal representation based on downmix signal representation and associated with parametric information object using modified imaging matrix.EFFECT: technical result consists in provision of high sound quality even in case of audio encoding matrix user selection while maintaining low level of computational efficiency on audio encoder side.21 cl, 19 dwg

Apparatus, method and computer program for providing adjusted parameters for provision of upmix signal representation on basis of a downmix signal representation and parametric side information associated with downmix signal representation, using an average value // 2607266
FIELD: acoustics; data processing.SUBSTANCE: invention relates to mixing. Apparatus for providing one or more adjusted parameters for a provision of an upmix signal representation, on basis of a downmix signal representation and parametric side information associated with downmix signal representation, using an average value of parameters, including a parameter adjuster configured to receive one or more parameters and to provide, on basis thereof, one or more adjusted parameters, wherein the parameter adjuster is configured to provide one or more adjusted parameters depending on an average value of a plurality of parameter values, such that a distortion of upmix signal representation caused by use of non-optimal parameters for provision of upmix signal representation is reduced, at least for one or more parameters deviating from optimum parameters is greater than a predetermined deviation.EFFECT: elimination of audible distortions in signal.22 cl, 10 dwg

Audio signal decoder, audio signal encoder, method of decoding audio signal, method of encoding audio signal and computer program using pitch-dependent adaptation of coding context // 2607264
FIELD: acoustics.SUBSTANCE: invention relates to means of encoding and decoding an audio signal. Audio signal decoder comprises a context-based spectral value decoder configured to decode a codeword, describing one or more spectral values or at least a portion of a number representation of one or more spectral values in dependence on a context state, to obtain decoded spectral values. Audio signal decoder also includes a context state determinator configured to determine a current context state in dependence on one or more previously decoded spectral values. Audio signal decoder also includes a time warping frequency-domain-to-time-domain converter configured to provide a time-warped time-domain representation of a given audio frame on basis of a set of decoded spectral values, associated with given audio frame and provided by context-based spectral value decoder.EFFECT: technical result is improvement of efficiency of encoding in presence of oscillations of fundamental frequency.18 cl, 51 dwg

Device and method for encoding and decoding an encoded audio signal using a temporary noise/overlays generating // 2607263
FIELD: data processing.SUBSTANCE: invention relates to means of encoding and decoding an encoded audio signal. Device for decoding an encoded signal comprises: audio decoder in a spectral region to generate a first decoded representation of a first set of first spectral parts representing residual spectral prognostic values; repeated frequency generating module for generating a restored second spectral part using the first spectral part from the first set of the first spectral parts, restored second spectral part additionally comprises residual spectral prognostic values; and backward predictive filter for inverse frequency prediction using residual spectral values for the first set of the first spectral parts and the restored second spectral part using information of the predictive filter included in an encoded audio signal.EFFECT: technical result is improved encoding/decoding principle, allowing reducing bit rate.20 cl, 41 dwg

Device and method for reproducing an audio signal, device and method for generating encoded audio signal, computer program and encoded audio signal // 2607262
FIELD: acoustics.SUBSTANCE: invention relates to means for generating an audio signal and audio playing. Device comprises a first reproducing means to reproduce the first part of the audio signal based on the first data. Providing means is configured to provide a signal-patch in the second frequency band, wherein the signal-patch is at least partially uncorrelated relative to the first part of the audio signal or is at least partially decorrelated version of the first part of the audio signal, which is shifted to the second frequency band. Second reproducing means is configured to reproduce the second part of the audio signal in the second frequency band based on the second data and signal-patch. Joining means is configured to join the reproduced first part of the audio signal and signal-patch prior to reproducing the second part of the audio signal by the second reproducing means.EFFECT: technical result is generating and reproducing the audio signal with reducing available data transmission speed.15 cl, 13 dwg

Systems and methods for determining set of interpolation coefficients // 2607260
FIELD: information technology.SUBSTANCE: invention relates to systems and methods of determining a set of interpolation coefficients. Method for determining a set of interpolation coefficients by means of electronic device includes determination of a value based on property of current frame and property of previous frame, determination of whether or not value is outside a range, determination of set of interpolation coefficients based on value and prediction mode indicator, if value is outside range.EFFECT: technical result consists in optimisation of throughput, thereby balancing desired average bit rate of recovered voice signal.32 cl, 16 dwg
Adaptation of weighing analysis or synthesis windows for encoding or decoding by conversion // 2607230
FIELD: acoustics; information technology.SUBSTANCE: invention relates to processing an audio signal and/or a video signal in the form of a sequence of samples and is intended for encoding and decoding a digital audio signal. For this purpose encoding or decoding is carried out by converting the digital audio signal using weighing analysis windows (ha) or synthesis windows (hs) used for samples frames, herewith the method includes non-uniform sampling (E10) of the initial window provided for a transformant of preset initial size N in order to use secondary conversion size M, which is different from N. Device includes a sampling module configured able to perform non-uniform sampling of the initial window provided for the transformant of given initial size N in order to use secondary conversion size M, which is different from N.EFFECT: saving memory resources and computational resources when encoding and decoding digital an audio signal by converting using weighing analysis and synthesis windows.15 cl, 7 dwg, 3 tbl

Device for quantization of linear predictive coding coefficients, sound encoding device, device for dequantization of linear predictive coding coefficients, sound decoding device and electronic device to this end // 2606552
FIELD: electronics.SUBSTANCE: invention relates to quantization of linear predictive coding coefficients. Device for quantization of a speech or an audio signal comprises: selection module configured to select basing on a prediction error one of the first sampling unit and the second unit of sampling by open circuit; the first sampling unit is configured to input signal sampling, including, at least, one of the speech signal or the audio signal without an interframe prediction; the second sampling unit is configured to input signal sampling with the interframe prediction.EFFECT: technical result is higher efficiency of quantization of an audio or a voice signal by selecting the optimum quantization module.20 cl, 38 dwg, 9 tbl

Audio encoder, audio decoder, method of encoding audio information, method of decoding audio information and computer program using iterative reduction of size of interval // 2605677
FIELD: data processing.SUBSTANCE: group of inventions relates to audio information encoding/decoding audio techniques. Disclosed is an audio decoder for providing decoded audio information based on encoded audio information. Audio decoder comprises an arithmetic decoder and frequency-domain to time domain converter. Arithmetic decoder is designed to provide a plurality of decoded spectral values based on an arithmetically encoded representation of spectral values. Arithmetic decoder is configured to select a mapping rule describing mapping of a code value onto a symbol code depending on numerical value of current context, which describes current state of context. Arithmetic decoder is also configured to determine numeric value of current context depending on a plurality of previously decoded spectral values.EFFECT: high efficiency of encoding/decoding audio information.15 cl, 48 dwg

Delay-optimised overlap transform, coding/decoding weighting windows // 2604994
FIELD: information technology.SUBSTANCE: invention relates to coding/decoding of a digital signal, consisting of successive blocks of samples. Coding includes application of a weighting window to two blocks of M successive samples. In particular, this weighting window is asymmetric and comprises four distinct portions extending successively over two aforesaid blocks, with a first portion, increasing over a first interval, a second portion, constant at a weighting value over a second interval, a third portion, decreasing over a third interval, and a fourth portion, constant at a weighting value over a fourth interval.EFFECT: technical result is improved quality of encoded sound.17 cl, 10 dwg

System and method of exciting mixed codebook for speech coding // 2604425
FIELD: information technology.SUBSTANCE: invention relates to means of exciting a mixed codebook for speech coding. Method of encoding an audio/speech signal involves determining a vector of the mixed codebook basing on an incoming audio/speech signal, herewith the mixed codebook vector includes a sum of the first codebook record from the first codebook and the second codebook record from the second codebook. Method additionally includes generating an encoded audio signal basing on the defined vector of the mixed codebook and transfer of a coded excitation index of the defined mixed codebook vector.EFFECT: technical result is increased perceptible quality of the speech signal in comparison with coding systems using only pulse excitation or only noise excitation.26 cl, 17 dwg

Signal processing device, method and program // 2604338
FIELD: communication.SUBSTANCE: invention relates to signal processing device. Envelope information generation module can generate information on envelope that represents enveloping form of high-frequency components of audio signal intended for encoding. Sine-wave information generating module can select sine wave signal from high-frequency components of audio signal and generate information on sine wave, which represents position of appearing sine wave signal. Encoding flow generation module can multiplex information on envelope, information on sine wave, and low-frequency components of audio signal, which were encoded, and outputs encoded stream, obtained in result. As result, high-frequency components included in sine wave signal, can be predicted with higher accuracy from information on envelope and information on sine wave at side of receiving encoded stream.EFFECT: technical result is possibility of obtaining sound of higher quality when decoding audio signal.14 cl, 25 dwg

Decoder and method of multi-instance spatial encoding of audio objects using parametric concept for cases of the multichannel downmixing/upmixing // 2604337
FIELD: acoustics.SUBSTANCE: invention relates to audio systems and is intended for generating output audio signal. Decoder comprises router of input channel to receive three or more channels of down-mixing and to receive auxiliary information, as well as two units of channel processing for generating of two treated channels to obtain output of audio channels. Router of feed channel is made with ability of input of each down-mixing channel in one of two units for processing channel, each of which receives one or more down-mixing channels, at that, each of two units for processing channel takes less than the total number of three or more down-mixing channels. Each unit of processing channel is configured with ability to generate one or more of the two treated channels depending on the auxiliary information and depending of the mentioned one or more down-mixing channels received by said unit of processing channel from a router input channel.EFFECT: technical result - increasing of accuracy of reproducing an audio signal.14 cl, 4 dwg

Audio encoder and decoder // 2602988
FIELD: information technology.SUBSTANCE: invention relates to multichannel audio encoding. Method of decoding in multichannel audio signal processing system for reconstruction of M encoded channels, where M>2 comprises following steps: reception of N down-mixing signals with coding shape, comprising spectral coefficients, corresponding to frequencies between first and second crossover frequencies, where 1<N<M; receiving M signals with coding shape, comprising spectral coefficients, corresponding to frequencies of up to first crossover frequency, wherein each of M signals with coding shape corresponds to corresponding one of M encoded channels; down-mixing M signals with coding shape in N down-mixing signals, containing spectral coefficients which correspond to frequencies of up to first crossover frequency; combination of each of N down-mixing signals with coding shape, comprising spectral coefficients, corresponding to frequencies between first and second crossover frequencies with corresponding one of N down-mixing signals, comprising spectral coefficients, corresponding to frequencies of up to first crossover frequency, in N combined down-mixing signals; expansion of each of N combined down-mixing signal into a frequency range above second crossover frequency by means of high frequency reconstruction, as a result of which each expanded down-mixing signal comprises spectral coefficients, corresponding to a range passing below first crossover frequency and higher than second crossover frequency; performing parametric up-mixing of N combined down-mixing signals with extended frequency range in M up-mixing signals, comprising spectral coefficients, corresponding to frequencies higher than first crossover frequency, each of M up-mixing signals corresponds to one of M encoded channels; and combination of M up-mixing signals comprising spectral coefficients, corresponding to frequencies higher than first crossover frequency, with M signals with coded shape, comprising spectral coefficients which correspond to frequencies of up to first crossover frequency.EFFECT: high quality of encoded and decoded audio signal.29 cl, 8 dwg

etadata conversion // 2602332
FIELD: information technology.SUBSTANCE: invention relates to metadata processing and is intended for metadata conversion with reduced computational complexity. For this purpose, a code converter can convert the inlet bit stream, containing a content inlet frame and associated metadata inlet frame in the outlet bit stream, containing the content outlet frame and associated metadata outlet frame. Code converter comprises a decoder to convert an inlet frame of the content in a set of decoded discrete values of the PCM signal, extraction of metadata from the metadata inlet frame, generation of signature value for the set of decoded discrete values of the PCM and the extracted metadata using cryptographic decoder key, encoder, made to receive the set of discrete values of the PCM and associated metadata, receiving the signature value, checking the received signature value for validity by means of cryptographic coder key, generation of content outlet frame and generation of associated metadata outlet frame of the outlet bit stream.EFFECT: technical result is high accuracy of metadata conversion.25 cl, 13 dwg, 5 tbl

ethods and systems for efficient recovery of high frequency audio content // 2601188
FIELD: information technology.SUBSTANCE: invention relates to encoding, decoding and processing an audio signal, in particular, it relates to means of recovering high-frequency content of an audio signal from low-frequency content of same audio signal. Method comprises determining a first banded tonality value for a first frequency subband. First banded tonality value is used for approximating a high frequency component of audio signal based on a low frequency component of audio signal. Determining a set of transform coefficients in a corresponding set of frequency bins based on a block of samples of audio signal. Determining a set of bin tonality values for set of frequency bins using set of transform coefficients, respectively. Forming a first subset of two or more of set of bin tonality values for two or more corresponding adjacent frequency bins of set of frequency bins lying within first frequency subband, thereby yielding first banded tonality value for first frequency subband.EFFECT: technical result is reducing complexity of calculations in audio encoding based on systems with spectral expansion.29 cl, 15 dwg, 2 tbl

ultistage iir filter and parallel data filtration by such a filter // 2599970
FIELD: information technology.SUBSTANCE: present group of inventions relates to multi-stage filters and may be used for filtering data using such filters. In one embodiment the device contains a buffer memory, at least two cascades of biquadratic filters and a controller connected to the cascades of biquadratic filters and configured to approve a single instruction flow in the first and second cascades of biquadratic filter; the said cascades operate independently and in parallel in response to the instruction flow; the first cascade is connected to the memory and is configured to perform biquadratic filtration of the block of N input discrete values in response to the flow of instructions for generating intermediate values and for approval of intermediate values in the memory, which include a filtrated version of each subset of input discrete values, the next cascade of the filter is connected to the memory and is configured to perform biquadratic filtration of buffered values, extracted from the memory in response to the flow of instructions for generating unit output values, wherein the output values contain an output value corresponding to each of the input discrete values in the block of N input discrete values and buffered values contain some of the intermediate values generated in the first cascade in response to a block of N input discrete values.EFFECT: to enable parallel data processing.36 cl, 11 dwg

Speech decoder, speech encoder, speech decoding method, speech encoding method, speech decoding program and speech encoding program // 2599966
FIELD: information technology.SUBSTANCE: invention relates to speech decoder and encoder speech. Speech decoder includes a multiplexing unit, a unit for decoding the low-frequency band, a block of filter bank for frequency band separation, an encoded sequence, a decoding unit and de-quantiser encoded sequence, a unit for generating high-frequency band, a unit for calculating the time envelope of the low-frequency band, a unit for correction of time bending components of high-frequency band.EFFECT: technical result-reduction of distortion of the reproduced signal.19 cl, 40 dwg

Selective bass post-filter // 2599338
FIELD: acoustics.SUBSTANCE: invention relates to encoding digital sound, specifically to means of encoding audio signals having components of different nature. Audio encoding method is characterised by a decision whether a device which will decode resultant bit stream must use post-filtering, including attenuation of inter-harmonic noise. Thus, decision on whether to use a post-filter, encoded inn bit stream, is made separately from decision on most suitable encoding mode. Audio decoding method, where a decoding step is followed by a post-filtering step, including inter-harmonic noise attenuation and characterised by presence of step of switching off post-filter in accordance with information on post-filtering, encoded in bit stream signal. Said method is well suited for audio signals of mixed origin due to its ability to deactivate post-filter depending only on information on post-filtering and, therefore, independently of such factors as current encoding mode.EFFECT: technical result is improvement of accuracy of reproducing sound.18 cl, 11 dwg

Audio encoder, audio decoder, method of encoding audio information, method of decoding audio information and computer program using range-dependent arithmetic encoding mapping rule // 2596596
FIELD: acoustics.SUBSTANCE: invention relates to audio encoding. Audio decoder includes an arithmetic decoder to provide a plurality of decoded spectral values based on an arithmetically encoded representation of spectral values; and frequency-domain converter to time domain; arithmetic decoder is configured to select a mapping rule describing mapping of a code value arithmetically encoded display on a symbol code which displays one or more decoded spectral value or at least part of one or more of decoded spectral values, depending on state of context; arithmetic decoder is configured to determine numeric current context value describing state of current context depending on a plurality of previously decoded spectral values, as well as depending on where there is a spectral value to decode-in first preset frequency domain or in second preset frequency domain.EFFECT: technical result consists in improvement of efficiency of encoding at low computational costs.17 cl, 48 dwg

Audio signal encoder, audio signal decoder, method for encoded representation of audio content, method for decoded representation of audio and computer program for applications with small delay // 2596594
FIELD: computer engineering.SUBSTANCE: invention relates to computer engineering. Audio signal encoder comprises a transform-domain path configured to obtain a set of spectral coefficients and noise-shaping information on the basis of a time-domain representation of a portion of the audio content to be encoded in a transform-domain mode. Transform-domain path comprises a time-domain-to-frequency-domain converter which performs window weighing in time domain of audio representation and outputs a set of spectral coefficients using time-domain-to-frequency-domain conversion window-weighted time representation of audio. Audio signal encoder includes a code-excited linear-prediction-domain path (CELP), which extracts information on code excitation and parameters of field of linear prediction of fragment audio encoded in CELP mode. Audio signal encoder allows selective formation of anti-aliasing information, when current fragment of audio follows fragment of audio coded by a CELP mode.EFFECT: technical result consists in improvement of efficiency of encoding successive fragments of audio.28 cl, 32 dwg

Spatial audio processor and method of providing spatial parameters based on acoustic input signal // 2596592
FIELD: acoustics.SUBSTANCE: invention relates to means for obtaining spatial parameters based on acoustic input signal. Spatial audio processor for providing spatial parameters based on an input audio signal comprises a module for determining signal characteristics and a controlled module. A module for determining signal characteristics is configured to determine signal characteristic of input audio signal. Controlled module to calculate spatial parameters of input audio signal in accordance with formula for calculation of variable spatial parameter is configured to modify formula for calculation of variable spatial parameter in accordance with certain signal characteristic.EFFECT: technical result consists in obtaining spatial parameters for input audio signal with minimum differences with model associated with changes in time or time instability of input audio signal.15 cl, 10 dwg

Coding of generalised audio signals at low bit rates and low delay // 2596584
FIELD: information technology.SUBSTANCE: invention relates to mixed time-domain/frequency-domain coding means for coding an input sound signal. Cut-off frequency for time-domain excitation contribution is also calculated in response to input sound signal, and a frequency extent of time-domain excitation contribution is adjusted in relation to said cut-off frequency. Following calculation of a frequency-domain excitation contribution in response to input sound signal, adjusted time-domain excitation contribution and frequency-domain excitation contribution are added to form a mixed time-domain/frequency-domain excitation constituting a coded version of input sound signal. In calculation of time-domain excitation contribution, input sound signal may be processed in successive frames of input sound signal and a number of sub-frames to be used in a current frame may be calculated.EFFECT: faster processing of delay in classification of sound signal and its conversion into frequency domain.54 cl, 6 dwg

Speech encoding device, speech decoding device, speech encoding method, speech decoding method, speech encoding program and speech decoding program // 2595951
FIELD: information technology.SUBSTANCE: present invention relates to the means of encoding and decoding a speech signal. A linear prediction coefficient of a signal presented in a frequency domain is obtained by performing linear prediction analysis in a frequency direction by using a covariance method or an autocorreletion method. Once the filter strength of the obtained linear prediction coefficient is adjusted, filtering may be performed in the frequency direction on the signal by using the adjusted coefficient, whereby the temporal envelope of the signal is formed. This reduces the occurence of pre-echo and post-echo and improves the subjective quality of the decoded signal, without significantly increasing the bit rate in a band extension technique in the frequency domain, represented by a spectral band replication (SBR).EFFECT: to reduce pre-echo and post-echo and improve the quality of the decoded signal without increasing the bit rate.4 cl, 50 dwg

Audio system and method for operation thereof // 2595943
FIELD: sound.SUBSTANCE: invention relates to an audio system, particularly to virtual spatial reproduction of audio signals. Audio system comprises a receiver for receiving an audio signal, such as an audio object or a signal of a channel of a spatial multi-channel signal. Binaural circuit generates a binaural output signal by processing audio signal. Processing is representative of a binaural transfer function providing a virtual sound source position for audio signal. Measurement circuit generates measurement data indicative of a characteristic of acoustic environment, and a determining circuit determines an acoustic environment parameter in response to measurement data. Acoustic environment parameter may typically be a reverberation parameter, such as a reverberation time. Adaptation circuit adapts binaural transfer function dynamically in response to acoustic environment parameter. For example, adaptation may modify a reverberation parameter to more closely resemble reverberation characteristics of acoustic environment.EFFECT: technical result is providing complete spatial audio perception on basis of binaural signals.14 cl, 7 dwg

Speech encoding device, speech decoding device, speech encoding method, speech decoding method, speech encoding program and speech decoding program // 2595915
FIELD: information technology.SUBSTANCE: present invention relates to the means of encoding and decoding a speech signal. A linear prediction coefficient of a signal presented in a frequency domain is obtained by performing linear prediction analysis in a frequency direction by using a covariance method or an autocorrelation method. Once the filter power of the obtained linear prediction coefficient is corrected, the signal is frequency filtered using the corrected coefficient, thereby forming a signal time envelope. This reduces the occurence of pre-echo and post-echo and improves the subjective quality of the decoded signal, without significantly increasing the bit rate in a band extension technique in the frequency domain, represented by a spectral band replication (SBR).EFFECT: to reduce pre-echo and post-echo and improve the quality of the decoded signal without increasing the bit rate.5 cl, 50 dwg

Speech encoding device, speech decoding device, speech encoding method, speech decoding method, speech encoding program and speech decoding program // 2595914
FIELD: information technology.SUBSTANCE: present invention relates to the means of encoding and decoding a speech signal. A linear prediction coefficient of a signal presented in a frequency domain is obtained by performing linear prediction analysis in a frequency direction by using a covariance method or an autocorreletion method. Once the filter strength of the obtained linear prediction coefficient is adjusted, filtering may be performed in the frequency direction on the signal by using the adjusted coefficient, whereby the temporal envelope of the signal is formed. This reduces the occurence of pre-echo and post-echo and improves the subjective quality of the decoded signal, without significantly increasing the bit rate in a band extension technique in the frequency domain, represented by a spectral band replication (SBR).EFFECT: to reduce pre-echo and post-echo and improve the quality of the decoded signal without increasing the bit rate.4 cl, 50 dwg

Audio system and method therefor // 2595912
FIELD: data processing; acoustics.SUBSTANCE: invention relates to audio systems. Audio system comprises a receiver which receives the input audio signal. Decomposition unit decomposes the audio signal into the transient component signal and the nontransient component signal. Then the output circuit generates the first output audio signal in response to balanced integration of the transient component signal and the nontransient component signal. Within the integration transient component signal balancing differs from nontransient component signal balancing. A new signal can be obtained with another emphasizing of the sound specific characteristics. Approach can be used to form new spatial audio channels from the existing spatial audio channel as forming a raised channel of the lower channel audio signals.EFFECT: providing improved spatial perception of the audio signal reproduced by the audio system, wider range of available audio effects.13 cl, 7 dwg

Audio signal processor for processing encoded multi-channel audio signals and method therefor // 2595910
FIELD: communication.SUBSTANCE: invention relates to simultaneous rendering of multi-channel signals. An audio signal processor receives a plurality of encoded multi-channel audio signals. A multi-channel decoder decodes a first encoded multichannel signal, to generate the first decoded multi-channel signal. A generator generates an encoded further audio signal selecting audio encoding data. A further decoder generates a further decoded signal by decoding the further encoded audio signal. A combiner combines the first decoded multi-channel signal and the further decoded signal to generate a multi-channel output signal.EFFECT: technical result is improved processing, maintaining low complexity and and/or reduced computational load, improved audio quality and facilitated operation.15 cl, 4 dwg

Encoding device and method, decoding device and method and program // 2595544
FIELD: information technology.SUBSTANCE: invention relates to means of encoding and decoding of sound. Subband power calculation module calculates subband signal power for high-frequency subband among multiple sub-bands, components of input signal. High-frequency subband power calculation module performs weighing of power subband, having high power, as for subband, having plurality of high frequency subband, for calculation of power of high frequency subband for subband. Multiplexing scheme multiplexes encoded data and encoded data of low frequency for output. Encoded data of high frequency is selected based on high-frequency subband power and is obtained by encoding information used for producing high-frequency component of input signal by estimating, and coded data of low frequency is obtained by encoding low frequency components of input signal.EFFECT: technical result consists in improvement of accuracy of audio signal, obtained as result of decoding.17 cl, 6 dwg

ethods and apparatus for encoding and decoding signals // 2592412
FIELD: communication. SUBSTANCE: invention relates to communication engineering and is intended for encoding and decoding signals. Signal encoding method includes obtaining a frequency domain signal according to an input signal; allocating predetermined bits to frequency domain signal according to a predetermined allocation rule; adjusting bit allocation for frequency domain signal when a highest frequency of frequency domain signal to which bits are allocated is greater than a predetermined value; and encoding frequency domain signal according to bit allocation for frequency domain signal. EFFECT: technical result is high accuracy of encoding and decoding signals. 20 cl, 9 dwg
 
2551397.
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