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(391283)
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(641)

Coder, decoder, system and method using concept of balance for parametric coding of audio objects // 2628900
FIELD: physics.SUBSTANCE: decoder comprises a parametric decoding unit for generating a plurality of first estimated audio object signals by upmixing three or more downmix signals. The parametric decoding unit is configured to upmix the three or more downmix signals depending on the parametric side information indicating information about the plurality of original audio object signals. Moreover, the decoder comprises a residue processing unit for generating a plurality of second estimated audio object signals by modifying one or more of the first estimated audio object signals. The residual processing unit is configured to modify mentioned one or more of mentioned first estimated audio object signals depending on one or more residual signals.EFFECT: increasing the coding efficiency.26 cl, 18 dwg
Irregular quantization of parameters for improved connection // 2628898
FIELD: physics.SUBSTANCE: group of inventions provides for irregular quantization of parametres related to parametric spatial coding of audio signals, and also provides restoration of an audio signal in a sound decoding system that takes into account irregular quantized parameters. According to the invention, such an approach makes it possible to reduce bit costs without significantly reducing the quality of the reconstructed audio signal.EFFECT: increasing the efficiency and quality of the encoding of the audio signal.21 cl, 7 dwg 4 tbl

Real-in-complex transformation with a small delay in bank filters with overlapping for partially integrated treatment // 2628219
FIELD: physics.SUBSTANCE: configuration of the filter banks with overlapping contains a synthesis cascade and an analysis cascade. The synthesis stage receives the first signal segmented into time blocks and outputs on this basis an intermediate signal to be received by the cascade of analysis, forming the basis for calculating the second signal segmented into time frames. In an embodiment, the synthesis stage is configured to output an approximate value of an intermediate signal in a time block located at L-1 time blocks ahead of its output block, the approximate value being calculated based on any available time blocks of the first signal such that the approximate value contributes to the second signal in the cascade of analysis. The delay is usually reduced by L-1 blocks. Applications include the processing of an audio signal in general and, in particular, a real-to-complex transformation.EFFECT: decreasing the delay of conversion.22 cl, 12 dwg, 1 tbl

asking errors in pictures // 2628197
FIELD: physics.SUBSTANCE: tracking (S11) sign changes between the corresponding conversion factors of predefined sub-vectors of successive good stationary frames; accumulating (S12) the number of sign changes in the respective sub-vectors of a predetermined number of successive good stationary frames; reconstructing (S13) the error frame with the help of the last good stationary frame, but with the reverse signs of the transformation coefficients in the sub-vectors having the accumulated number of sign changes that exceeds the predetermined threshold.EFFECT: improved masking of errors in frames due to the lack of transmission of additional third-party parameters or the formation of additional delays required by interpolation.12 cl, 16 dwg, 1 tbl

Decoder and method of parametric generalized concept of the spatial coding of digital audio objects for multi-channel mixing decreasing cases/step-up mixing // 2628195
FIELD: physics.SUBSTANCE: decoder to generate audio output contains one or more audio output from mixing with decreasing signal containing one or more channels of decreasing-mixing. Mixing with decreasing signal encodes one or more signals of digital audio objects. Decoder contains the determinant of a threshold to determine the threshold depending on the power signal and/or energy noise, at least one of the one or more signals of digital audio objects and/or depending on the power signal and/or energy noise, at least one of the one or more channels of decreasing-mixing. Moreover, the decoder contains a processing unit for generating of referred one or more output audio channels mentioned one or more channels of mixing ground depending on the threshold.EFFECT: improving the quality of encoding audio objects.14 cl, 4 dwg
ethods of coding and decoding sound, corresponding machine-readable media and corresponding coding device and device for sound decoding // 2628177
FIELD: physics.SUBSTANCE: according to the disclosure, this is achieved by calculating and applying two weighting factors, one for the approximated sound object and the other for the decorrelated audio object, for decorrelating the sound objects into the sound encoding system.EFFECT: providing the opportunity to improve the sound object recovery, less complex and more flexible control of the encoded decorrelation sound.28 cl, 7 dwg

Audio encoder, audio decoder, method of coding and decoding audio information and computer program, determining value of context sub-adaption based on norm of the decoded spectral values // 2628162
FIELD: information technology.SUBSTANCE: encoder, an audio decoder, method of encoding and decoding audio information, and a computer program are proposed to obtain a frequency value based on the previously normal decoded spectral values. The audio decoder for providing decoded audio information based on the encoded audio information includes an arithmetic decoder for providing a plurality of decoded spectral values based on the arithmetic encoded spectral value representation, and a time-frequency converter for providing audio representation of a time domain using the decoded spectral values to obtain decoded audio information, the arithmetic decoder selects a mapping rule that describes the mapping of the code values for the symbol code, depending on the context state, which is described by the numerical value of the current context, determines the numerical value of the current context, depending on the set of the previously decoded spectral values.EFFECT: reducing the encoding errors of the input audio information spectral values.19 cl, 24 dwg

Decoding method and decoding device // 2628159
FIELD: physics.SUBSTANCE: decoding method comprises the stages whereat: if it is determined that the current frame of the speech signal is a lost frame, the high frequency band signal is synthesized according to the decoding result of the previous frame with respect to the current frame; subframe amplification of at least two subframes of the current frame is determined according to the subframe amplifications of the subframes of at least one frame, previous to the current frame, and the amplification gradient between the subframes of at least one frame; global amplification of the current frame is determined and the synthesized high frequency band signal is adjusted according to the global amplification and the subframe amplifications of at least two subframes so as to receive the high frequency band signal of the current frame.EFFECT: preventing or reducing the phenomenon of noise during the processing of frame losses, thereby improving the quality of speech.15 cl, 12 dwg

ethod and device for controlling audio frame loss masking // 2628144
FIELD: physics.SUBSTANCE: transient condition is detected in the property of the previously received and restored audio signal (which indicates a signal surge or decay) that can lead to a suboptimal quality of recovery, when the original masking method is used to create a substitute frame. The original masking method is modified by selectively adjusting the amplitude of the substitute frame spectrum, when the transient condition is detected. Additionally, the second condition is observed in the statistical property of observed frame losses, which can lead to a non-optimal quality of recovery, when the original masking method is used to create a substitute frame. Additionally, the original masking method is modified by selectively adjusting the amplitude of the substitute frame spectrum, when the second condition is detected. The second condition is the loss of several frames in a row.EFFECT: increasing the quality of masking in case of audio frame loss.27 cl, 15 dwg

Device and method for generating coded signal or decoding encoded audio signal by using site with multiple overlap // 2626666
FIELD: physics.SUBSTANCE: device for generating an encoded signal comprises: a controller of the window sequence for generating information about the window sequence for framing an audio signal, wherein the information about the window sequence indicates the first window function for generating the first frame of the spectral values, a preprocessor for framing the second block of samples corresponding to the second window function and one or more third window functions, a spectrum converter for receiving the first and the second frames of the spectral values and for receiving the first and the second portions of the spectral values of the second frame, a processor for processing the first frame and the second frame to obtain encoded frames of an audio signal.EFFECT: providing an improved concept of audio encoding, decoding, which provides an improved efficiency of the device.41 cl, 44 dwg

Per-segment spatial audio installation to another loudspeaker installation for playback // 2625953
FIELD: physics.SUBSTANCE: device is proposed for adaptation of the spatial audio signal intended for initial speaker installation, to the speaker installation for playback, which differs from the reference loudspeaker installation. The device comprises a unit of decomposing to the direct sound and the surround sound, which is adapted to decompose the channel signals in the reference loudspeaker installation segment to the components of the direct sound and the surround sound, and to determine the arrival direction of the direct sound components. The unit of presenting the direct sound receives the loudspeaker installation information for the playback and adjusts the direct sound components using the loudspeaker installation information for the playback so that the perceived arrival direction for the direct sound components in the loudspeaker installation for the playback is identical to the arrival direction for the direct sound components.EFFECT: preserving the audio spatial image, while resetting the audio signal to another loudspeaker installation.16 cl, 9 dwg

Coder, decoder and methods of scale transformation dependent on signal in spatial audio object coding // 2625939
FIELD: physics.SUBSTANCE: decoder for generating an output audio signal comprising one or more audio output channels from the downmix signal. The decoder comprises the first analysis module for converting the downmix signal to obtain the first converted downmix containing a plurality of channels of the first sub-range. The decoder comprises the second analysis module for generating the second transformed downmix by converting, at least, one of the channels of the first sub-range, to obtain a plurality of channels of the second sub-range, wherein the second transformed downmix includes the first sub-range channels, that were not converted by the second analysis module, and the channels of the second sub-range. The decoder comprises an upmix configured to upmix the second transformed downmix based on the parametric additional information on the said one or more audio object signals, to obtain an output audio signal.EFFECT: improving the encoded audio signal quality.16 cl, 22 dwg

Principle for coding mode switch compensation // 2625561
FIELD: information technology.SUBSTANCE: in response to a switch event, a temporary smoothing and/or mixing is implemented at transition between the first temporary portion of information signal before a switch event and the second temporary portion of information signal after a switch event in high-frequency spectrum band. Response is given to switching of one or more of: an audio encoding mode in the full bandwidth to a BWE audio coding mode, and a BWE audio coding mode to an audio encoding mode in the full bandwidth, wherein the high frequency spectrum band overlaps a portion of BWE-extension spectrum of a BWE audio encoding mode and a portion of spectrum of transformation or linear prediction encoding with a portion of audio encoding mode spectrum in the full bandwidth.EFFECT: increased audio quality when switching between different encoding modes with different bandwidths by smoothing and/or mixing sound at appropriate transition.16 cl, 21 dwg

Device and method for encoding or decoding audio signal with overlap depending on transition location // 2625560
FIELD: physics.SUBSTANCE: device comprises: a controlled framing module for framing an audio signal to provide a sequence of the sample framed blocks; a converter for converting the sequence of the sample framed blocks into the spectral representation comprising a sequence of the spectral value frames; a transition location detector to identify the transition location in the anticipation transferring area in the frame; and a controller to control the controlled framing module, to apply a particular window having a predetermined overlapping length, to the audio signal in response to the identified transition location, where the controller is configured to select the particular window of the group of, at least, three windows, including the first window having the first overlapping length, the second window having the second overlapping length, and the third window having the third overlapping or non-overlapping length.EFFECT: providing an improved audio encoding, decoding, providing the increase of the device performance.35 cl, 44 dwg

Audio processing system // 2625444
FIELD: physics.SUBSTANCE: audio processing system comprises a pre-processing component, that receives the quantized spectral components and performs the inverse quantization, thereby obtaining a time domain representation for the intermediate signal, a processing stage in the frequency domain, configured to provide the time domain representation to the processed audio signal, and a sampling rate converter providing the reconstructed audio signal sampled with the target sampling frequency. Corresponding internal sampling frequencies of the time domain representation for the intermediate audio signal and the time domain representation for the processed audio signal. In specific embodiments, the processing stage comprises a parametric upmixing stage, which operates in two different modes, and is associated with the delay stage, which provides a constant total delay.EFFECT: sampled sound wave restoration in the single-channel, stereo or multi-channel format.15 cl, 34 dwg, 4 tbl

Watermark signal generation and embedding watermark // 2624549
FIELD: physics.SUBSTANCE: watermark signal generator generating a watermark signal carrying the watermark data for hidden embedding into an audio signal in the reduction of the watermark signal and the audio signal. The watermark signal feeding circuit includes a psychoacoustic processor, that determines the audio signal masking threshold, and a modulator, that generates a watermark signal based on the superposition of the reference forming functions mutually spaced by the temporary time reference amount of the watermark data discrete-time representation, from which each function, forming the time reference, has an amplitude weighted by using the respective time reference of the discrete-time representation multiplied by the corresponding amplitude weight depending on the masking threshold, wherein the time reference is shorter than the time interval, for which the time reference forming functions are defined; and an individual amplitude weight depends on the time reference of the discrete-time representation bordering this time reference.EFFECT: optimal balancing the ability to calculate the watermark signal and the lack of its effect on hearing.12 cl, 39 dwg

Device and method for generation of expanded by signal frequency, using the formation of extension signal // 2624104
FIELD: radio engineering, communication.SUBSTANCE: device for generating a signal with an improved range comprises: a calculator to calculate a value, which describes the energy distribution on the frequency in the main signal; and a signal generator for generating expansion signal comprising a frequency band expansion is not included in the main signal from the main signal. A signal generator configured to generate extension signal or the main signal so that the spectral envelope of the extension signal or the main signal dependent on the value describing the distribution of energy versus frequency in the main signal.EFFECT: reduction in the required amount of side information as compared with full parametric description of the missing frequencies.17 cl, 18 dwg

Audio encoder and audio decoder with metadata information on program or substreams structure // 2624099
FIELD: information technology.SUBSTANCE: buffer memory is a non-volatile carrier configured to store at least one frame of encoded audio bitstream. Wherein encoded audio bitstream contains audio data and a metadata container. Metadata container includes a header and one or more metadata contents after a header. One or more metadata contents comprise dynamic range compression (DRC) metadata, and DRC metadata represents or contains profile metadata indicating whether DRC metadata contain control values of dynamic range compression for use in dynamic range compression in accordance with one compression profile for audio content. If profile metadata indicate that DRC metadata contain DRC control values for use in dynamic range compression, DRC metadata also contains DRC control values set generated in accordance with a compression profile.EFFECT: increased efficiency of detection and correction of errors of audio stream metadata identification due to presence of audio stream content metadata in audio stream.14 cl, 8 dwg, 2 tbl

ethod and device for compressing and restoring representation of high-order ambisonic system for sound field // 2623886
FIELD: physics.SUBSTANCE: HOA representation is analyzed for the presence of the dominant sound sources, and their directions are evaluated. The HOA representation is then decomposed into a number of dominant directional signals and a residual component. This residual component is transformed into a discrete spatial domain in order to obtain functions of common plane waves in uniform sampling directions that are predicted from dominant directed signals. Finally, the prediction error is converted back to the HOA domain and represents the residual surrounding HOA component, for which the order is reduced, followed by perceptual coding of the dominant directional signals and the residual component.EFFECT: improving the quality of the compressed representation of the HOA sound field.14 cl, 8 dwg

Audio encoder and decoder for encoding on interleaved waveform // 2622872
FIELD: physics.SUBSTANCE: these methods and devices for encoding and decoding audio signals, in particular, the method for endecoding includes receiving the encoded signal on the waveform, having a spectral content corresponding to the subset of the frequency range above the transition frequency. The encoded signal on the waveform is interleaved with the parametric high frequency audio signal reconstruction above the transition frequency, thereby an improved reconstruction of the audio signal high-frequency bands is achieved.EFFECT: providing an improved reconstruction of the transient processes and the tonal components in the high-frequency bands.25 cl, 9 dwg

Effective pre-echo attenuation in digital audio signal // 2622863
FIELD: radio engineering, communication.SUBSTANCE: method comprises: decoded signal decomposition to at least two subsignals according to a predetermined decomposition criterion; calculating attenuation coefficients relating to a sub-signal and a discrete value from a predetermined pre-echo zone; pre-echo attenuation in the pre-echo zone of each sub-signal by applying attenuation coefficients to these sub-signals; receiving the signal subjected to attenuation by combining the sub-signals subjected to attenuation.EFFECT: reduction of pre-echo irritating effect.15 cl, 13 dwg

Device and method for encoded signal processing and encoder and method for encoded signal generating // 2622860
FIELD: information technology.SUBSTANCE: encoded signal comprises an encoded audio signal with information regarding to pitch delay or pitch gain, and a postfilter management parameter for low audio frequencies, comprising: an audio signal decoder for encoded audio signal decoding using information concerning pitch delay or pitch gain to obtain a decoded audio signal; a controlled post-filter of low sound frequencies to filter the decoded audio signal to obtain a processed signal, wherein a controlled post-filter of low sound frequencies has a variable characteristic of post-filter of low sound frequencies controlled by a management parameter of a post-filter of low sound frequencies; and a controller for setting of a variable characteristic of post-filter of low sound frequencies according to a management parameter of a post-filter of low sound frequencies, that is included to an encoded signal.EFFECT: improved quality of encoded audio signal.18 cl, 13 dwg

Transmitter of activation signal with the time-deformation, acoustic signal coder, method of activation signal with time deformation converting, method of acoustic signal encoding and computer programs // 2621965
FIELD: information technology.SUBSTANCE: encoder comprises the window function controller (504), window organizing unit (502), device for time-deformation (506) with the final functional quality control ability, time / frequency inverter (508), TNS stage (510) or encoder quantizer (512 ), window function controller (504), device for time-deformation (506), TNS stage (510) or an additional noise filling analyzer (524) are controlled by the results of the signal analyzing obtained by time-deformation analyzer (516) or the signal categorizer (520). The decoder applies the noise filling operation using the adjusted noise filling estimation depending on a harmonic or speech sound signal characteristic.EFFECT: increased efficiency of coding, decoding of the acoustic signal with the harmonic or speech content.8 cl, 37 dwg

Adaptive tone quantization of low complexity audio signals // 2621003
FIELD: physics.SUBSTANCE: audio encoder comprises: a framing device configured to extract frames (F) from the audio signal (AS); a quantization module configured to display the spectral lines (SL1-32) of the spectrum signal (SPS) extracted from the frame (F) of the audio signal (AS), in the quantization index (I0, I1). The quantization module has a dead zone (DZ), in which the input spectral lines (SL) are displayed in the zero quantization index (I0); and a control device configured to modify the dead zone (DZ). The control device comprises a tonality calculating device configured to calculate a single value (TI5-32) indicating the tone for one spectral line (SL1-32) or for one group of the spectral lines (SL1-32). The control unit is configured to modify the dead zone (DZ) for one said spectral line (SL1-32) or one said group of the spectral lines (SL1-32) depending on the respective values (TI5-32) of specifying the tonality.EFFECT: providing the improved concepts for adaptive quantization of the audio signals.17 cl, 2 dwg

Improving sound signal using estimated spatial parameters // 2620714
FIELD: physics.SUBSTANCE: received audio data may include the first set of the frequency coefficients and the second set of the frequency coefficients. The spatial parameters of, at least, a part of the second set of the frequency coefficients may be estimated, at least, partially based on the first set of the frequency coefficients. The estimated spatial parameters can be applied to the second set of the frequency coefficients to generate the second modified set of the frequency coefficients. The first set of the frequency coefficients may correspond to the first frequency band, and the second set of the frequency coefficients may correspond to the second frequency band. The combined frequency coefficients of the composite binding channel may be based on the frequency coefficients of two or more channels. It is possible to calculate a mutual correlation between the frequency coefficients of the first channel and the combined frequency coefficients.EFFECT: creating means that provide improved audio processing.17 cl, 37 dwg

ethod of encoding coefficient quantization with linear prediction, sound encoding method, method of decoding coefficient quantization with linear prediction, sound decoding method and record medium // 2619710
FIELD: physics, computer engineering.SUBSTANCE: invention relates to a method of quantization. For this purpose, there is a selection based on prediction error of one of the first quantization module without interframe prediction and the second quantization module with interframe prediction without feedback and quantization of the input signal using the selected quantization module.EFFECT: invention allows to increase quantization performance of quantization module.21 cl, 38 dwg, 9 tbl

Audio encoder and audio decoder with metadata information on program or substreams structure // 2619536
FIELD: information technology.SUBSTANCE: encoded audio bitstream that comprises audio data and metadata is received. Said metadata and audio data are extracted from said encoded audio bitstream. Metadata represent or contain metadata information on program and metadata of substreams structure. Coded audio bitstream contains a sequence of frames serving as features of at least one audio program, said metadata information on program and metadata of substream structure serve as features of said program. Each of frames comprises at least one segment of audio data. Each said segment of audio data comprises at least some of audio data, each frame of at least a subset of frames comprises a segment of metadata, and each said segment of metadata comprises at least some of metadata information on program and at least some of metadata of substream structure. Segment of metadata is located in reserved data field selected from group consisting of ignore data field, addbsi field, auxdata fields or combinations thereof.EFFECT: improved quality of encoded audio data.20 cl, 8 dwg, 2 tbl

Device and method for audio synthesizing, decoder, encoder, system and computer program // 2618919
FIELD: physics.SUBSTANCE: method and device for synthesizing an audio signal, and an audio decoder are described. The spectral tilt is applied to the coding book code used to synthesize the current audio signal frame. The spectral tilt is based on the spectral tilt of the current audio signal frame.EFFECT: improved synthesizing the audio signal both at low and at higher transmission rates by using the audio signal spectral tilt by synthesizing the signal to increase the achievable coding efficiency and thus enhancing the quality of the reproduced signal.27 cl 5 dwg

The device and method for selecting one of the first audio encoding algorithm and the second audio encoding algorithm // 2618848
FIELD: information technology.SUBSTANCE: device for selecting one of the first encoding algorithm with the first characteristic and the second encoding algorithm with the second characteristic for the audio signal part encoding to obtain an encoded version of the audio signal part comprises the first estimation unit for estimating of the audio signal part of the first quality measure, which is linked with the first encoding algorithm, without actual encoding and decoding of the audio signal part with the use of the first encoding algorithm, the second estimation unit is provided for evaluation of the audio signal parts of the second quality measure, which is linked with the second encoding algorithm, without actual encoding and decoding of the audio signal part with the use of the second encoding algorithm, a controller for selection of the first encoding algorithm or the second encoding algorithm basing on comparing between the first quality measure and the second quality measure.EFFECT: provision of encoding algorithm with good quality selection.23 cl, 5 dwg

Encoding and decoding of audio objects // 2618383
FIELD: radio engineering, communication.SUBSTANCE: audio objects encoder includes a receiver which receives N audio objects. The mixer mixes N audio objects into M audio channels, and channels circuit receives K audio channels from M audio channels, K = 1, 2, and K<M. The parameters circuit generates parameters of upmixing of audio objects for at least a part of each of N audio objects relative to K audio channels, and the output circuit generates output data stream having parameters of upmixing of audio objects and M audio channels. The decoder of audio objects receives the data stream and includes a channel circuit receiving K audio channels from downmixing of M channels; and the decoder of objects for forming at least part of each of N audio objects by upmixing of K audio channels on the basis of parameters of upmixing of audio objects.EFFECT: increase of efficiency of coding and decoding of audio objects.14 cl, 12 dwg

ethod, device and system for processing audio data // 2617926
FIELD: physics, instrument-making.SUBSTANCE: invention relates to means of processing audio data and communication engineering. The method includes: obtaining a noise frame of an audio signal and decomposing the current noise frame into a low-frequency band noise signal and a high-frequency band noise signal; and encoding and transmitting the low-frequency band noise signal using a first discontinuous transmission mechanism and encoding and transmitting the high-frequency band noise signal using a second discontinuous transmission mechanism. According to the present invention, various processing methods are employed for the high-frequency band signal and the low-frequency band signal, the computational load and encoded bits can be saved by allowing non-decrease of subjective codec quality, and bits which are saved can help achieve the goal of reducing transmission bandwidth or improving overall coding quality.EFFECT: invention is characterised by high coding efficiency.8 cl, 9 dwg

Signal processor, window provider, encoded media signal, method for processing signal and method for providing window // 2616863
FIELD: information technology.SUBSTANCE: invention relates to data transmission and is intended for processing an input signal with window weight coefficients. Signal processor to provide a processed version of an input signal depending on input signal, includes 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: high efficiency of encoding by adaptation of window characteristics to characteristics of input spectrum of signal.35 cl, 17 dwg

Audiodecoder for decoding bit audio performance, audiocoder for encoding sound signal and method of decoding frame of encoded sound signal // 2616774
FIELD: physics.SUBSTANCE: method for decoding a frame of the encoded audio signal, wherein the encoded audio signal is generated by the audio encoder, the method comprises: extracting the first parameter from the encoded audio signal, wherein the first parameter represents the encoding mode for the frame; extracting the second parameter from the encoded audio signal; starting either the first decoding mode or the second decoding mode based on the first parameter value; generating a preliminary audio signal from the encoded audio signal when operating in either the first decoding mode or in the second decoding mode; and filtering the preliminary audio signal with a pitch filter. Thus when filtering, the second parameter is used for selective connecting or disconnecting the pitch filter, and the second parameter is different from the said first parameter.EFFECT: attenuating the interharmonic noise.20 cl, 11 dwg

Device and method for processing speech /audio signal // 2616557
FIELD: physics.SUBSTANCE: method for processing a speech/audio signal includes the stages, at which the starting high frequency signal is received upon switching the transmission band of the speech/audio signal corresponding to the current frame of the speech/audio signal; the global gain parameter of the high frequency source signal is received in the time domain; weight processing is performed for the ratio of energy and the global gain parameter in the time domain, and the obtained weighted value is used as a predicted global gain parameter, wherein the energy ratio is a ratio between the energy of the high frequency signal archival frame in the time domain and the energy of the current high-frequency source signal frame; the high frequency source signal is corrected with the predicted global gain parameter to obtain the corrected high-frequency signal in the time domain; the current frame of the narrowband signal in the time domain and the corrected high-frequency signal in the time domain are synthesized, and the synthesized signal is output.EFFECT: providing comfortable sound reproduction while switching the transmission band of the speech/audio signal.23 cl, 10 dwg

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
 
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