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(393877)
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)
Adaptive generation of scattered signal in upmixer // 2642386
FIELD: physics.SUBSTANCE: upmixer can be configured to detect cases of transient states of the audio signal. In the cases of transient states of the audio signal, the upmixer can be configured to add signal-adaptive control to the expansion process of the scattered signal, in which M audio signals are output. The upmixer can change the expansion process of the scattered signal over time in such a way that in cases of transient states of the audio signal, the scattered parts of the audio signals can be distributed mainly only to the output channels spatially close to the input channels. In cases of intransitive states of the audio signal, the scattered parts of the audio signals can be distributed in a substantially uniform manner.EFFECT: possibility of dividing the scattered and non-scattered parts of N input audio signals.42 cl, 12 dwg

Audio signal processing method, signal processing unit, stereophonic render, audio coder and audio decoder // 2642376
FIELD: physics.SUBSTANCE: in this audio signal processing method, according to the room impulse response, the audio signal is processed using an early portion of the room impulse response separately from the late room reverberation of the room impulse response, wherein the late reverberation processing comprises forming a scaled reverberant signal, wherein the scaling depends on the audio signal. The processed early portion of the audio signal and the scaled reverberant signal are combined.EFFECT: identity of the late reverberation of the room impulse response to the result of convolution of an audio signal with a full impulse response.19 cl, 10 dwg, 2 tbl

Principle for audio coding and decoding for audio channels and audio objects // 2641481
FIELD: physics.SUBSTANCE: audio encoder for encoding the input audio data to receive the output audio data comprises an input interface for receiving a plurality of audio channels, a plurality of audio objects, and metadata associated with one or more of the plurality of audio objects; a mixer for reducing a plurality of objects and a plurality of channels in such a way as to obtain a plurality of pre-merged channels, wherein each pre-merged channel containes audio data of the channel and audio data of, at least, one object; a basic encoder for basic encoding of the input data of the base encoder; and a metadata compression module for compressing metadata associated with one or more of a plurality of audio objects.EFFECT: increasing the compression efficiency with high sound quality.24 cl, 11 dwg

ethod and device for processing signals // 2641466
FIELD: physics.SUBSTANCE: method includes: determining the total number of bits to be allocated corresponding to the sub-bands of the current frame; allocating the primary bits for the subbands according to the total number of bits; determining the number of primary information units for each subband that is allocated to the primary bits to obtain the total number of redundant bits of the current frame and the number of information units corresponding to each subband; selecting a subband for extracting secondary bits from the subbands according to a secondary bit allocation parameter comprising one of the total number of redundant bits or a subband characteristic for each subband; allocating secondary bits for the subbands to extract the redundant bits and obtaining the number of secondary bits for each subband; and determining the number of secondary information units for each subband in order to retrieve the number of information units corresponding to each subband from the subbands to extract the secondary bits.EFFECT: improving the quality of encoding and decoding audio signals and eliminating the bit loss.20 cl, 14 dwg
ethod, device and system for processing audio data // 2641464
FIELD: physics.SUBSTANCE: noise frame of an audio signal is received. The current noise frame is expanded to the noise signal of the low frequency band and the noise signal of the high frequency band. The noise signal of the low frequency band is encoded and transmitted by using the first intermittent transmission mechanism. The noise signal of the high frequency band is encoded and transmitted by using the second intermittent transmission mechanism.EFFECT: decrease in the bandwidth and an improvement in the quality of the audio data encoding.26 cl, 9 dwg

Decorrelator structure for parametric recovery of sound signals // 2641463
FIELD: physics.SUBSTANCE: coding system encodes multiple audio signals (X) as a downmix signal (Y) together with the coefficients (P, C) of the controlled and forward up-mix. In the decoding system, the pre-multiplying unit calculates the intermediate signal (W) by linear mapping the downmix signal in accordance with the first set of coefficients (Q); the decorrelating section outputs the decorrelated signal (Z) based on the intermediate signal; the controllable up-mixer section calculates a controlled up-mix signal; the forward-up-mixing section calculates the up-mix signal; the summation section provides a multidimensional reconstructed signal ( ) by summing the signals of the controlled and forward up-mix; and the converter calculates the first set of coefficients based on the coefficients of the controlled and forward up mix and feeding it to the pre-multiply block.EFFECT: improving the accuracy of the audio signal recovery.18 cl, 4 dwg

Audio encoder, audio decoder, method of providing coded audio information, method of providing decoded audio information, computer program and coded presentation using signal-adaptive bandwidth extension // 2641461
FIELD: physics.SUBSTANCE: audio encoder comprises a low-frequency encoder for obtaining an encoded representation of the low-frequency portion and a unit for providing bandwidth extension information based on the input audio information. The audio encoder is also configured to selectively include the bandwidth extension information in the encoded audio information. The audio decoder comprises a low-frequency decoder to obtain a decoded representation of the low-frequency portion and an extension of the bandwidth to obtain a blind-bandwidth expansion signal for portions of audio content, for which the bandwidth expansion parameters are not included in the coded audio information and to obtain a bandwidth extension signal for portions of audio content, for which the parameters of the bandwidth extension are included in the coded audio information.EFFECT: providing bandwidth expansion to improve the tradeoff between the bit rate and the sound quality.38 cl, 8 dwg
Sound coding device and decoding device // 2641265
FIELD: physics.SUBSTANCE: method of decoding a coded bit of the audio stream in the processing system of audio signals is disclosed, wherein the method comprises: extracting the first shape encoding signal from the encoded bit audio stream, containing the spectral coefficients corresponding to the frequencies to the first channel separation frequency; performing parametric decoding on the second channel separation frequency to generate the reconstructed signal. The second channel separation frequency is higher than the first channel separation frequency and the parametric decoding uses reconstruction parameters derived from the encoded bitstream to generate the reconstructed signal; extracting the second shape encoding signal from the encoded bit audio stream, comprising spectral coefficients corresponding to a subset of frequencies above the first channel separation frequency; alternating the second shape encoding signal with the reconstructed signal to generate an interlaced signal and combining the interlaced signal with the first shape encoding signal.EFFECT: decoding the encoded bit audio stream in the audio signal processing system.16 cl, 8 dwg

Device and method for processing sound signal using error signal due to spectrum aliasing // 2641253
FIELD: physics.SUBSTANCE: device for processing audio signal, containing a sequence of blocks of spectral values, includes a processor for calculating a signal under the influence of aliasing using at least one first modification to the first block of a sequence of blocks, and using at least one different second modification value for the second block of a sequence of blocks and for estimating the error signal due to aliasing, representing the error due to aliasing in the signal influenced by aliasing. The combination module combines the signal influenced by aliasing and the error signal due to aliasing.EFFECT: increasing the efficiency of audio signal processing.13 cl, 16 dwg

Adaptive band extension and device therefor // 2641224
FIELD: physics.SUBSTANCE: method includes decoding the flow of audio bytes in order to form a decoded audio signal of the low frequency band and the excitation spectrum in the low frequency band corresponding to the low frequency band. The frequency subband area is selected from the low frequency bands by using the parameter that specifies information of the energy spectral envelope of the decoded audio signal of the low frequency band. Excitation spectrum in the high frequency band is formed for high frequency band by duplicating the excitation spectrum in the frequency subband from the selected area of the frequency subband to the area of the high frequency band corresponding to the high frequency band. Using the formed excitation spectrum in the high frequency band, the audio signal of the extended high frequency band is formed by applying a spectral envelope of the high frequency band. The audio signal of the extended high frequency band is summed with the decoded audio signal of the low frequency band in order to form an output audio signal having an extended frequency band.EFFECT: ensuring the formation of an extended frequency band in the decoder.19 cl, 18 dwg
Audio encoding device, audio encoding method, audio encoding programme, audio decoding device, audio decoding method and audio decoding programme // 2640743
FIELD: physics.SUBSTANCE: audio signal transmission device for encoding an audio signal includes an audio encoding unit that encodes an audio signal and a side information encoding unit that calculates and encodes side information from the prediction signal. The receiving device of the audio signal to decode the audio-code and audio signal output includes a buffer of the audio-code that detects a packet loss based on the reception status of the audio pack, a decoding unit of audio-parameters that decodes the audio-code, when the audio pack is adopted correctly, a decoding unit of side information, which decodes the side information, when the audio pack is adopted correctly, a collection unit of side information, which collects the side information obtained by decoding side information, a processing unit of missing audio-parameters and an audio synthesis unit.EFFECT: restoration of sound quality without increasing the algorithmic delay, when there is a loss of the packet when encoding audio.43 dwg

Extraction of reverberative sound using microphone massives // 2640742
FIELD: physics.SUBSTANCE: method contains the stages, on which the spatial coherence between the first phase of the scattered sound in the first microphone signal and the second phase of the ambient sound in the second microphone signal is estimated. The first microphone signal is captured by the first microphone, and the second microphone signal is captured by the second microphone, which is located at a distance from the first microphone. The method additionally contains a stage, at which the linear restrictions of the filtration coefficients of the ambient sound filter is determined, and the linear constraint is based on spatial coherence. The method also comprises the step of calculating at least one of the signal statistics and noise statistics with respect to the first microphone signal and the second microphone signal. The method also contains a stage, at which the coefficients of the filter filtration of ambient sound filter is determined by solving the optimisation problem against at least one of signal statistics and noise statistics, given the linear restrictions of the filtration coefficients.EFFECT: ensuring the capture of scattered sound coming to the microphone array from all directions, by obtaining a better directional pattern of the scattered sound filter.15 cl, 8 dwg

Improved quantizer // 2640722
FIELD: physics.SUBSTANCE: quantization module configured to quantize the first coefficient from the coefficient block. This coefficient block contains a number of coefficients for a number of corresponding frequency resolution elements. The quantization module is configured to create a set of quantizers. This set of quantizers contains a number of different quantizers associated with a number of different signal-to-noise ratios, respectively, called SNRs. This series of different quantizers contains a quantizer with noise coverage; one or more quantizers with the addition of pseudorandom noise; and one or more quantizers without the addition of pseudorandom noise. The quantization module is also configured to determine the SNR pointer, serving as a sign of SNR assigned to a specified first coefficient, and to select the first quantizer from a set of quantizers based on the SNR index. In addition, the quantization module is configured to quantize the said first coefficient using the said first quantizer.EFFECT: increasing the flexibility in assuming different data rates and different levels of distortion.20 cl, 19 dwg

Device for coding audio signal having plurality of channels // 2640650
FIELD: physics.SUBSTANCE: device comprises a reception unit for receiving phase information, a transient process separating unit, a transient decorrelator, the second decorrelator, and a combiner unit, in which the transient process separating unit is adapted to divide the input signal into the first signal component and the second signal component such that the first signal component contains parts of the input signal transient process, and so that the second signal component contains signal parts without the input signal transient process. A decorrelator of the transition process is adapted to apply the phase information received by the reception unit to the transient process signal component.EFFECT: increasing the data transfer rate due to different processing of the transient process signal and the signal that does not contain the transient process.7 cl, 6 dwg

Device and method of transforming first and second input channels, at least, in one output channel // 2640647
FIELD: physics.SUBSTANCE: spatial encoding of audio begins with a number of source inputs, such as five or seven input channels, which are identified by placing them in the layout for playback as a left channel, a central channel, a right channel, a left surround channel, a right surround channel, and a low frequency enhancing channel (LFE). In the device, each input channel and each output channel has a direction, in which the associated loudspeaker is located relative to the center position of the listener, wherein the device is configured to convert the first input channel to the first output channel from the configuration of the output channels.EFFECT: improved audio reproduction in case of format conversion between different speaker channel configurations.4 cl, 14 dwg

Device and method for decoding coded audio with filter for separating around transition frequency // 2640634
FIELD: physics.SUBSTANCE: device for decoding an encoded audio signal containing the encoded base signal, contains: a base decoder for decoding the coded reference signal to obtain the decoded primary signal; a module for generation of fragments for the formation of one or more spectral fragments having frequencies not included in the decoded reference signal, using the spectral part of the decoded reference signal; and a separation filter for the spectral filtration for separating the decoded reference signal and the first frequency of the fragment, with the frequencies going from the interval filling frequency of the signal absence to the upper boundary frequency, or for the spectral filtration for separating the first frequency part and the second frequency part.EFFECT: providing the possibility of encoding audio signals over a wide range of bit rates.15 cl, 35 dwg

Hybrid speech amplification with signal form coding and parametric coding // 2639952
FIELD: physics.SUBSTANCE: hybrid speech amplifying method uses parametric encoding amplification for some signal states and waveform encoding amplification for the remaining signal states. Other aspects are methods for generating a bitstream indicating a sound program including speech and other content such that hybrid speech amplification can be performed in relation to the program, a decoder including a buffer that stores, at least, one segment of the encoded audio bitstream generated by any embodiment implementing the inventive method, and a system or a device configured to perform any embodiment of the inventive method. At least, some of the speech amplifying operations are performed by the receiving audio decoder using the medium/side channel speech metadata generated by the upstream audio encoder.EFFECT: increasing the audibility of the audio signal speech contents with respect to the non-speech audio content.37 cl, 11 dwg

Device and method for coding/decoding for expansion of high-frequency range // 2639694
FIELD: physics.SUBSTANCE: encoder can downsample the input signal, perform basic encoding of the input signal with reduced sampling, perform the frequency conversion of the input signal and perform the bandwidth extension encoding using the base signal of the input signal in the frequency domain.EFFECT: extending the high-frequency range by extracting the main signal of the input signal and adjusting the energy of the input signal using the tonality of the high-frequency range of the input signal and the key tone.7 cl, 38 dwg

ethod and device for normalized playing audio mediadata with embedded volume metadata and without them on new media devices // 2639663
FIELD: physics.SUBSTANCE: device decodes a bitstream containing audio data and volume metadata containing a reference loudness value to generate an audio output signal. The device comprises a signal processor comprising a gain control device configured to adjust the output audio signal level. The gain control device includes a reference volume decoder configured to generate a loudness value, a gain calculator configured to calculate the gain value based on the loudness value and based on the sound power control value, and a volume processor configured to control the volume of the audio output signal based on the gain value.EFFECT: providing the ability to normalize the playback volume both the content containing metadata of loudness, and the content that does not contain volume metadata.16 cl, 5 dwg

Coder, decoder and methods for backward compatible dynamic adaptation of time/frequency authorization for spatial coding of audio objects // 2639658
FIELD: physics.SUBSTANCE: decoder for generating an output audio signal comprises one or more audio output channels, a downmix signal including a plurality of time domain downmix samples, a window sequence generator for determining a plurality of analysis windows, each of the analysis windows comprising a plurality of downmix samples of the time domain of the downmix signal. The decoder contains a module of t/f-analysis for converting the said plurality of samples of down-mixing the temporal region of each analysis window of the above-mentioned plurality of analysis windows from time domain in frequency-time domain depending on the window length of the said analysis window to obtain a transformed downmix. In addition, the decoder comprises an up-mixer.EFFECT: expanding the possibilities of multi-channel reproduction of individual audio content in order to improve the auditory sense.17 cl, 22 dwg

Device for transforming linear prediction coefficients and method for transforming linear prediction coefficients // 2639656
FIELD: physics.SUBSTANCE: converting device converts the first linear prediction coefficients computed at the first sampling frequency into the second linear prediction coefficients at the second sampling frequency different from the first sampling frequency and comprises: a means for calculating, on the real axis of the unit circle, a power spectrum corresponding to the second linear prediction coefficients at the second sampling frequency, based on the first linear prediction coefficients; a means for calculating, on the real axis of the unit circle, the autocorrelation coefficients from the power spectrum; and a means for converting the autocorrelation coefficients to the second linear prediction coefficients at the second sampling frequency.EFFECT: effective reduction in the amount of computation, when converting the linear prediction coefficients.2 cl, 6 dwg

Device and method for coding signals // 2638752
FIELD: physics.SUBSTANCE: method is implemented by predicting the comfort noise that is generated by the decoder according to the currently input frame in the case, in which the currently input frame is encoded into a silence descriptor (SID) frame and by determining the actual silence signal. The currently input frame is a silence frame, determining the degree of deviation between the comfort noise and the actual silence signal, determining that the encoding method of the currently input frame is a coding technique with a tightening frame, according to the degree of deviation and coding the currently inputted frame according to the encoding technique with a snap frame.EFFECT: expanding the arsenal of technical means for coding signals.21 cl, 15 dwg

Device and method for reducing quantization noise in decoder of temporal area // 2638744
FIELD: physics.SUBSTANCE: decoded excitation in the time domain is converted into an excitation in the frequency domain. A weight mask is formed to reconstruct the spectral information lost in the quantization noise. The excitation in the frequency domain is modified in order to increase the dynamics of the spectrum by applying a weight mask. The modified excitation in the frequency domain is converted into a modified excitation in the time domain. The method and device can be used to improve reproduction of music content by codecs based on linear prediction (LP). Optionally, the synthesis of the decoded excitation in the time domain can be classified into one of the first set of excitation categories and the second set of excitation categories.EFFECT: improving the quality of the encoded speech signal.31 cl, 4 tbl, 4 dwg

Coding of spectral coefficients of audio signal spectrum // 2638734
FIELD: physics.SUBSTANCE: in this device, the adjustment of the relative spectral distance between the pre-encoded/decoded spectral coefficient and the currently-encoded spectral coefficient depends on information regarding the spectrum shape. Information regarding the spectrum shape may contain a measure of the pitch or periodicity of the audio signal, the measure of inter-harmonic distance spectrum of the audio signal and/or the relative location of the formant and/or troughs of the spectral envelope of the spectrum, and based on this knowledge, the spectral neighbourhood, which is used in order to form the context of spectral coefficients to be coded/decoded in a given time, can be adapted to the particular form of the spectrum.EFFECT: increasing the efficiency of coding the spectral coefficients of the audio signal by encoding, decoding the spectral coefficient to be encoded, decoding at a given time, by entropy encoding, decoding.22 cl, 22 dwg

Transforming coding/decoding of harmonic sound signals // 2637994
FIELD: physics.SUBSTANCE: encoder for encoding coefficients (Y(k)) of frequency conversion of a harmonic audio signal includes the following elements: a location determinant for spectral peaks having values greater than a predetermined frequency dependent threshold. An encoder of the peak areas including and surrounding the detected peaks. A low-frequency coefficient set encoder is outside the peak areas and below the transition frequency, which depends on the number of bits used to encode the peak areas. A noise level enhancement encoder configured to encode a noise gain coefficient of, at least, one high frequency set of still unencoded coefficients outside the peak areas.EFFECT: improving the quality of the encoded harmonic sound signal.10 cl, 23 dwg

ethod and device for predicting signal of excitation of upper band // 2637885
FIELD: physics.SUBSTANCE: set of spectral frequency parameters, which are arranged in the order of the frequencies, are received according to the received bit stream of the lower frequency band. The parameters of the spectral frequency contain the parameters of the linear spectral frequency, LSF, the lower frequency band or the parameters of the spectral frequency of the immittance, ISF, the lower frequency band. Differences in the spectral frequency parameters between each two spectral frequency parameters that have the same position interval in some or all of the said spectral frequency parameters are calculated. The search range for searching for the minimum difference in the spectral frequency parameters is determined. The search range indicates the part of the calculated spectral frequency parameter differences. The portions of the calculated spectral frequency parameter differences are corrected using a correction factor to obtain a plurality of corrected spectral frequency parameter differences.EFFECT: improving the quality of the excitation signal of the upper band.20 cl, 11 dwg

Device and method for coding // 2636697
FIELD: physics.SUBSTANCE: this group of inventions involves performing the correct allocation of quantization bits for the spectral coefficients of the audio signal, thereby improving the quality of the signal received by the decoder by decoding. The method includes: after splitting the spectral coefficients of the current data frame into subbands, obtaining the values of the quantized frequency envelopes of the subbands; changing the values of the quantized frequency envelopes of the subbands in the first number in the subbands; allocating quantization bits to subbands according to the changed values of the quantized frequency envelopes of the subbands in the first quantity; quantizing the spectral coefficient of the subband, to which the quantization bit is allocated, on the subbands; and recording the quantized spectral subband coefficient, to which the quantization bit is allocated, into a bitstream.EFFECT: increasing the efficiency of compression coding and improving the signal quality.28 cl, 7 dwg
Presentation of multichannel sound using interpolated matrices // 2636667
FIELD: physics.SUBSTANCE: method for encoding an N-channel audio program comprises the steps of: determining the first cascade of elementary matrices N×N, which, when applied to discrete values of N encoded signal channels, realizes the first mixing of audio content in M output channels. The first mixing corresponds to a time-varying mixing A(t); determining the interpolation values, which, together with the first cascade of elementary matrices and the interpolation function defined in the subinterval, indicate a sequence of stages of the updated elementary matrices N×N, so that each cascade of the updated elementary matrices, when applied to discrete values of N coded signal channels, implements updated mixing, N coded channel signals in M output channels; generating an encoded bit stream that indicates coded audio content, interpolation values, and the first cascade of elementary matrices.EFFECT: elimination of unwanted artifacts in the encoding, decoding of the sound program.59 cl, 8 dwg

Speech signal encoding device using acelp in autocorrelation area // 2636126
FIELD: physics.SUBSTANCE: speech signal encoding device by determining the codebook vector of the speech encoding algorithm comprises a matrix determining module for determining the autocorrelation matrix R and a codebook determining module for determining the codebook vector, depending on the autocorrelation matrix R. The matrix determining module is configured to determine an autocorrelation matrix R by determining the vector coefficients for the vector r. The autocorrelation matrix R contains a plurality of rows and a plurality of columns. The vector r denotes one of the columns or one of the rows of the autocorrelation matrix R, where R(i, j)=r(|i-j|), where R(i, j) denotes the coefficients of the autocorrelation matrix R, where i is the first index denoting one from the plurality of rows of the autocorrelation matrix R, and where j is the second index denoting one of the plurality of columns of the autocorrelation matrix R.EFFECT: increasing the encoding efficiency.24 cl, 3 dwg
Prediction based on model in filter set with critical discreteization // 2636093
FIELD: physics.SUBSTANCE: signal of the first audio subband is determined using a set of analyzing filters comprising a series of analyzing filters that generate a series of subband signals from the audio signal, respectively, in a row of subbands. A method for estimating the first discrete signal value of the first subband signal in the first subband of an audio signal includes determining a model parameter for a signal model; determining a prediction coefficient to be applied to the previous discrete value of the decoded signals of the first subband obtained from the first subband signal based on the signal model based on the model parameter and based on the set of analyzing filters. The time interval of the previous discrete value precedes the time interval of the first discrete value; and determining the estimation of the first discrete value by applying the prediction coefficient to the previous discrete value.EFFECT: providing a low data rate with a low level of spurious frequencies.33 cl, 9 dwg

ethod and signal processor for converting plurality of input channels from configuration of input channels to output channels from configuration of output channels // 2635903
FIELD: radio engineering, communication.SUBSTANCE: method comprises providing a set of rules associated with each input channel from a plurality of input channels, the rules specifying different conversions between the associated input channel and a set of output channels. For each input channel from the plurality of input channels, the rule associated with the input channel is accessed, a determination is made as to, whether there is a set of output channels specified in the access rule in the configuration of the output channels and the rule is selected, which is accessed if the set of the output channels specified in the access rule is present in the configuration of the output channels. Input channels are converted to output channels according to the selected rule.EFFECT: improving the sound quality.23 cl, 9 dwg, 6 tbl

Device and method for coding or decoding sound signal with intelligent filling of intervals in spectral area // 2635890
FIELD: physics.SUBSTANCE: device for decoding the encoded audio signal comprises an audio decoder in the spectral domain to form the first decoded representation of the first set of the first spectral portions. The decoded representation has the first spectral resolution; a parametric decoder for generating the second decoded representation of the second set of the second spectral portions having the second spectral resolution lower than the first spectral resolution; a frequency re-forming unit for re-forming each composed second spectral portion having the first spectral resolution using the first spectral portion and the spectral envelope information for the second spectral portion; and a time-spectral converter for converting the first decoded representation and the reconstructed second spectral portion to the temporal representation.EFFECT: increasing the efficiency of encoding and decoding video data.26 cl, 41 dwg

Device and method for delivering improved characteristics of direct downmixing for three-dimensional audio // 2635884
FIELD: physics.SUBSTANCE: device for downmixing three or more input audio channels in order to receive two or more audio output channels, comprises a receiving interface for receiving three or more input audio channels and for receiving auxiliary information. In addition, the device includes a downmixer for downmixing three or more audio input channels depending on the side information in order to obtain two or more audio output channels. The number of output audio channels is less than the number of audio input channels. The auxiliary information indicates a characteristic of, at least, one of three or more input audio channels or a characteristic of one or more sound waves recorded in one or more input audio channels or a characteristic of one or more sound sources that emit one or more sound waves recorded in one or more input audio channels.EFFECT: increasing the audio signal encoding efficiency.10 cl, 9 dwg

Encoding and decoding positions of spectral peaks // 2635876
FIELD: physics.SUBSTANCE: audio signal segment encoding method comprises: determining, which of the two nondissipative coding circuits for spectral peak positions is used, wherein the first circuit is suitable for periodic or semi-periodic spectral peak position distributions, and the second circuit is suitable for sparse spectral peak position distributions; wherein the definition is based on the maximum distance dmax between two spectral peaks in the audio signal segment and the comparison of the number of bits required for the corresponding circuit after encoding the audio signal segment using two circuits; selecting the second coding circuit for spectral peak positions when the maximum distance dmax between two spectral peaks in the audio signal segment exceeds the threshold T; and selecting a spectral peak position encoding circuit that requires the least number of bits to encode the positions of the spectral peaks of the audio signal segment when the maximum distance dmax does not exceed the threshold T.EFFECT: increasing the efficiency of encoding/decoding spectral peaks.22 cl, 17 dwg, 1 tbl

Device and method for spatial coding of audio object using hidden objects for impacting on signal mixture // 2635244
FIELD: physics.SUBSTANCE: device includes a downmixer for downmixing one or more audio objects to produce one or more raw downmix signals, a processing unit for processing one or more raw downmix signals to produce one or more processed downmix signals, a signal calculating unit for calculating one or more additional signals, the signal calculating unit being configured to calculate each from one or more additional signals based on the difference between one of the one or more processed downmixed signals and one of one or more raw downmixed signals, an object information generator for generating parametric information of audio objects for one or more audio objects and parametric additional information for the additional signal.EFFECT: improving the playback quality of target audio scenes.16 cl, 11 dwg

Device and method of quantizing vectors of envelope frequencies // 2635069
FIELD: physics.SUBSTANCE: method includes: dividing N envelope frequencies in one frame into N1 vectors, where each vector in N1 vectors includes M envelopes; quantizing the first vector in N1 vectors by using the first codebook to obtain a codeword corresponding to the quantized first vector, wherein the said first codebook is divided into 2B1 sites; determining, according to the codeword corresponding to the quantized first vector, that the quantized first vector is associated with i-th site of 2B1 sites of the said first codebook; determining the second codebook according to the codebook of the i-th site; and quantizing the second vector in N1 vectors based on the said second codebook. In the embodiments of the present invention, the frequency envelopes are divided into a plurality of vectors with smaller dimensions, so that the quantization of the vectors can be performed with respect to the envelope vectors by using a codebook with fewer bits.EFFECT: increasing the efficiency of quantizing the vectors of envelope frequencies.8 cl, 3 dwg
Effective encoding of sound scenes containing sound objects // 2634422
FIELD: physics.SUBSTANCE: encoding method includes, inter alia, computing the M downmix signals by forming combinations of the N audio objects, where M≤N, and calculating parameters allowing the retrieval of a set of audio objects generated based on the N audio objects, starting from the M downmix signals. The M downmix signals are computed in accordance with a criterion independent of any speaker configuration.EFFECT: increasing the efficiency of encoding and decoding of audio objects.36 cl, 11 dwg

Device and method for effective synthesis of synusoid and swip-synusoid by using spectral patterns // 2633136
FIELD: physics.SUBSTANCE: device for generating an output audio signal comprises a processing unit for processing a spectrum of the coded audio signal to obtain a spectrum of a decoded audio signal comprising a plurality of spectral coefficients, each of the spectral coefficients having a spectral arrangement in the spectrum of the coded sound signal and the spectral value. The spectral coefficients are sequentially ordered according to their spectral location in the spectrum of the coded sound signal so that the spectral coefficients form a sequence of spectral coefficients. In addition, the device comprises a replacement unit for replacing, at least, one or more pseudo-coefficients with a specific spectral pattern to produce an altered audio spectrum. The determined spectral pattern comprises, at least, two template coefficients, each of, at least, two of the pattern coefficients having a spectral value.EFFECT: increasing the efficiency of audio coding with low latency and low data rate.23 cl, 17 dwg, 4 tbl

Device and method for forming plurality of parametric sound flows and device and method for forming plurality of acoustic system signals // 2633134
FIELD: physics.SUBSTANCE: device for forming a plurality of parametric audio flows from an input spatial audio signal obtained from the record of the recording space includes a segmentation device and a driver. The segmentation device is configured to provide, at least, two input segmented audio signals from an input spatial audio signal, wherein, at least, two input segmented audio signals are associated with respective segments of the recording space. The shaper is configured to generate a parametric audio flow for each of, at least, two input segmented audio signals to produce a plurality of parametric audio flows.EFFECT: improving the quality of spatial sound.14 cl, 12 dwg

Digital switching signal sequence for switching purposes, device for including related digital switching signal sequence into digital information audio signal and device for receiving information signal supplied with switching signal sequence // 2633108
FIELD: radio engineering, communication.SUBSTANCE: digital switching signal sequence is realized as a pre-digitized, high-pass filtered white noise signal of the predetermined duration T, with an upper cutoff frequency lying above the frequency, at which the loudness threshold characteristic in silence for human hearing has the greatest possible sensitivity. The switching signal sequence is used for switching purposes, for example switching between two information signals.EFFECT: increasing the accuracy of detecting the digital switching signal sequence.11 cl, 10 dwg

Adding comfort noise for modeling background noise at low data transmission rates // 2633107
FIELD: physics.SUBSTANCE: decoder is configured to process the encoded audio bitstream. The decoder comprises: a bitstream decoder configured to extract a decoded audio signal from the bitstream, the decoded audio signal comprising, at least, one decoded frame; a noise estimating device configured to generate a noise estimate signal including a level and/or spectral noise estimate in the decoded audio signal; a comfort noise generating unit configured to extract a comfort noise signal from the noise estimation signal; and a combiner configured to combine the decoded frame of the decoded audio signal and the comfort noise signal to obtain an audio output signal.EFFECT: increasing the natural sound of the coded audio signal.26 cl, 6 dwg

System and method of excitating mixed codebook for speech coding // 2633105
FIELD: physics.SUBSTANCE: audio/speech coding method includes determining a mixed codebook vector based on an incoming audio/speech signal, wherein the mixed codebook vector includes a recording amount of the first codebook from the first codebook and the second codebook recording from the second codebook. The method further includes generating a coded audio signal based on the specific mixed codebook vector and transmitting an index of the coded excitation of the specific mixed codebook vector.EFFECT: increasing the perceived quality of the speech signal in comparison with the coding systems using only pulse excitation or only noise excitation.26 cl, 17 dwg

ethods and devices for signal coding and decoding // 2633097
FIELD: physics.SUBSTANCE: it is determined, in accordance with the number of available bits and the predetermined first saturation threshold i, the number k of subbands to be encoded, where i is a positive number and k is a positive integer. In accordance with the quantized envelopes of all subbands, k subbands from all subbands are selected or k subbands from all subbands are selected in accordance with the psychoacoustic model. The first coding operation is performed on the spectral coefficients of k subbands. In embodiments of the present invention, the number of k subbands to be encoded is determined according to the number of available bits and the predetermined first saturation threshold, and the coding is performed on k subbands that are selected from all subbands and not over the full frequency range.EFFECT: improving the decoded signal quality.18 cl, 8 dwg

ethod and device for obtaining spectral coefficients for replacement audio frame, audio decoder, audio receiver and audio system for audio transmission // 2632585
FIELD: physics.SUBSTANCE: method for obtaining the spectral coefficients for the replacing audio frame is performed as follows: detecting the tonal components of the audio signal spectrum based on the peak that is present in the spectra of the frames preceding the replacing frame, the prediction of the spectral coefficients for the peak and its surroundings in the spectrum of the replacing frame is carried out for the tonal component of the spectrum and the non-predicted spectral coefficients for the replacing frame or of the corresponding spectral coefficient of the frame preceding the replacing frame is used for the non-tonal component of the spectrum. The spectral coefficients for the peak and its environment in the spectrum of the replacing frame are predicted based on the amplitude of the complex spectrum of the frame preceding the replacing frame and the predicted phase of the complex spectrum of the replacing frame, and the phase of the complex spectrum of the replacing frame is predicted based on the phase of the complex spectrum of the frame preceding the replacing frame, and the phase shift between the frames preceding the replacing frame.EFFECT: improving the accuracy of decoding.39 cl, 8 dwg

Device and method of selection of one of first coding algorithm and second coding algorithm by using harmonic reduction // 2632151
FIELD: physics.SUBSTANCE: device for selecting one of the first coding algorithm and the second coding algorithm to encode a portion of the audio signal to obtain an encoded version of a portion of the audio signal comprises a filter configured to receive an audio signal, reduce the amplitude of harmonics in the audio signal, and output a filtered version of the audio signal. The first evaluation module is provided to use the filtered version of the audio signal when estimating an SNR or segmental SNR portion of an audio signal as a first quality score for a portion of an audio signal that is associated with the first coding algorithm without actually encoding and decoding a portion of the audio signal using the first coding algorithm. The second evaluation module is provided for estimating the SNR or segment SNR as the second quality score for the portion of the audio signal that is associated with the second encoding algorithm.EFFECT: reducing the complexity of the choice between the first coding algorithm and the second coding algorithm.15 cl, 5 dwg

Noise filling in audio coding with perception transformation // 2631988
FIELD: physics.SUBSTANCE: audio decoder with perceptual transformation comprises: a noise filling module, a noise generating module of frequency area, wherein the noise generating module of frequency area is configured to: determine the spectral perceptual weighting function from the information of linear prediction coefficients, signaled in the data stream, in which the spectrum is encoded, or to determine the perceptual spectral weighting function of the scaling coefficients relating to the range of the scaling coefficients signaled in the data stream, in which the spectrum is encoded. The noise filling module is configured to: generate an intermediate noise signal; to identify continuous spectral zero parts of the audio signal spectrum; to determine the function for each continuous spectral zero part depending on the width of the corresponding continuous spectral zero part, the spectral position of the corresponding continuous spectral zero part; and to generate an intermediate noise signal.EFFECT: improving the audio quality after the spectrum is filled with noise.26 cl, 23 dwg

ethod of low-speed coding and decoding speech signal // 2631968
FIELD: radio engineering, communication.SUBSTANCE: in the vocoder based on linear prediction, the excitation signal vector is searched on the basis of vector quantization using an analysis procedure through synthesis on previously trained small code books that are statistically related to the initial parameter vector describing the state of the voice path. The index of the vocal path parameter vector, the codebook subspace vector of the small dimension of the excitation signal parameters statistically associated with the vocal path parameter vector and the corresponding scaling coefficient of the excitation signal are transmitted on the communication channel to synthesise the speech signal on each quasi-stationary segment of the speech signal analysis.EFFECT: improving the quality of the synthesised speech signal in low-speed vocoder with linear prediction with limitations on the data transfer rate.4 dwg

Audiodecoding device, device for audio coding, audiodecode method, audio coding method, audiodecoding program and audio code program // 2631155
FIELD: physics.SUBSTANCE: audio decoding device decodes the encoded audio signal and outputs the audio signal. The decoding unit decodes the encoded sequence containing the encoded audio signal and obtains the decoded signal. The selective shaping unit of the temporal envelope forms a temporal envelope of the decoded signal in the frequency band based on decoding related information regarding the decoding of the encoded sequence.EFFECT: reducing the distortion of a component of the frequency range encoded with a small number of bits in the time domain.16 cl, 24 dwg

ethod and device for determining the coding mode, method and device for coding audio signals and a method and device for decoding audio signals // 2630889
FIELD: physics.SUBSTANCE: encoding mode determination method includes determining one of a plurality of coding modes including the first coding mode and the second coding mode as the initial coding mode in accordance with the audio signal characteristics and if there is an error in determining the initial coding mode, generating the corrected the encoding mode by correcting the initial encoding mode to the third encoding mode.EFFECT: reducing delays caused by frequent change of coding mode.2 cl, 9 dwg
Sound coding device and decoding device // 2630887
FIELD: physics.SUBSTANCE: speech encoder contains a framing module configured to receive a set of blocks; wherein mentioned set of blocks comprises a number of consecutive blocks of MDCT transformation coefficients; wherein mentioned set of blocks serves as a sign of discrete values of the speech signal. The transform coefficient block contains a number of transform coefficients for the corresponding series of frequency resolution elements. In addition, the encoder comprises an envelope estimation module configured to determine the current envelope based on a series of successive blocks of transform coefficients. This current envelope serves as a sign of a number of spectral energy values for a corresponding series of frequency resolution elements. In addition, the encoder comprises an envelope interpolation module configured to determine a number of interpolated envelopes based on a series of successive blocks of transform coefficients, respectively, based on the current envelope.EFFECT: improving the quality of encoding by providing a smooth transition between time-domain coding and frequency-domain coding.20 cl, 10 dwg
 
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