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Sbr bitstream parameter downmix Invention relates to means of decoding and/or transcoding audio. A first and a second source set of spectral band replication (SBR) parameters are merged into a target set of SBR parameters. The first and second source set comprise a first and second frequency band partitioning, respectively, which are different from one another. The first source set comprises a first set of energy related values associated with frequency bands of the first frequency band partitioning. The second source set comprises a second set of energy related values associated with frequency bands of the second frequency band partitioning. The target set comprises a target set of energy related values associated with an elementary frequency band. The method comprises steps of breaking up the first and the second frequency band partitioning into a joint grid comprising the elementary frequency band; assigning a first value of the first set of energy related values to the elementary frequency band; assigning a second value of the second set of energy related values to the elementary frequency band; and combining the first and second value to yield the target energy related value for the elementary frequency band. |
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At transmitting side, each video signal code processing channel includes a "code 2n-code 2n-1" converter and each audio code processing channel includes a "sound-code" converter and at the receiving side, each screen matrix element is made from one emitting cell. |
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Mdct-based complex prediction stereo coding Invention relates to means of stereo encoding and decoding using complex prediction in the frequency domain. A decoding method for obtaining an output stereo signal from an input stereo signal encoded by complex prediction coding and comprising first frequency-domain representations of two input channels, comprises the upmixing steps of: (i) computing a second frequency-domain representation of a first input channel; and (ii) computing an output channel based on the first and second frequency-domain representations of the first input channel, the first frequency-domain representation of the second input channel and a complex prediction coefficient. |
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Invention relates to digital broadcasting which provides an audio indicator of link quality. After receiving a digital radio signal using a digital radio receiver, the quality of the received digital radio transmission is determined. Then an audio message from the received digital radio transmission is decoded. Then an audio indicator is superimposed onto the audio message, to form a composite audio signal. Finally, the amplitude of the audio indicator is dynamically adjusted relative to the amplitude of the audio message depending on the quality of the received digital radio transmission. |
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Invention relates to an information system for delivering different types of information to an end device through acoustic waves. A transmitter capable of generating acoustic waves for transmitting information, which are almost inaudible to the human ear, is required in a medium which enables to transmit information through acoustic waves. The transmitter is a device for converting different types of information into an acoustic wave in a sound spectrum, and transmission, having a microphone for receiving ambient sound at the point from where the acoustic wave is emitted, which serves as the input signal of the ambient sound; a peak frequency detector for determining in the ambient sound signal the peak frequency of the main component of ambient sound; a carrier generator for generating carriers, having a plurality of frequencies equal to the product of the peak frequency and a natural number and can be used to mask ambient sound; and a modulator for modulating the plurality of carriers of the baseband. |
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Apparatus for generating output spatial multichannel audio signal Invention relates to means of generating an output spatial multichannel audio signal based on an input audio signal. The input audio signal is decomposed based on an input parameter to obtain a first signal component and a second signal component that are different from each other. The first signal component is rendered to obtain a first signal representation with a first semantic property and the second signal component is rendered to obtain a second signal representation with a second semantic property different from the first semantic property. The first and second signal representations are processed to obtain an output spatial multichannel audio signal. |
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Audio signal processing device and method Invention relates to audio signal transmission and is intended for processing an audio signal by varying the phase of spectral values of the audio signal, realised in a bandwidth expansion scheme. The audio signal processing method and device comprise a window processing module for generating a plurality of successive sampling units, a plurality of successive units including at least one added audio sampling unit, an added unit having added values and audio signal values, a first converter for converting the added unit into a spectral representation having spectral values, a phase modifier for varying the phase of spectral values and obtaining a modified spectral representation and a second converter for converting the modified spectral representation into a time domain varying audio signal. |
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Invention relates to audio encoding and decoding technology, particularly to hierarchical audio encoding and decoding and hierarchical audio encoding and decoding for transient signals. The hierarchical audio encoding method comprises performing a transient detection on an audio signal of a current frame; performing a time-frequency transform; quantising and encoding amplitude envelope values of core layer encoding sub-bands and extended layer encoding sub-bands; quantising and encoding core layer frequency-domain coefficients; inversely quantising the frequency-domain coefficients in the core layer which are performed with a vector quantisation; performing a difference calculation with original frequency-domain coefficients to obtain a core layer difference signal; and calculating amplitude envelope quantisation indices of the core layer difference signals; quantising and encoding the extended layer encoding signals; multiplexing and packeting the amplitude envelope encoded bits of the core layer encoding sub-bands and the extended layer encoding sub-bands, the encoded bits of the core layer frequency-domain coefficients and the encoded bits of the extended layer coding signals, and then transmitting to a decoding end. |
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Multi-resolution switched audio encoding/decoding scheme Invention relates to audio encoding technologies. An audio encoder for encoding an audio signal has a first coding channel for encoding an audio signal using a first coding algorithm. The first coding channel has a first time/frequency converter for converting an input signal into a spectral domain. The audio encoder also has a second coding channel for encoding an audio signal using a second coding algorithm. The first coding algorithm differs from the second coding algorithm. The second coding channel has a domain converter for converting an input signal from an input domain into an output domain audio signal. |
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Invention relates to means of encoding and decoding audio signals. A downmix signal and a residual signal are generated based on a stereo signal. The difference in intensity between channels and cross-correlation between channels are determined. Preferably, parametric stereo coding parameters are time- and frequency-dependent. A transform stage generates a pseudo left/right stereo signal by performing a transform based on the downmix signal and the residual signal. The pseudo stereo signal is processed by a perceptual stereo encoder. For stereo encoding, left/right encoding or mid/side encoding can be selected. Preferably, the selection between left/right stereo encoding and mid/side stereo encoding is time- and frequency-dependent. |
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Invention relates to an audio format transcoder (100) for transcoding an input audio signal. The input audio signal has at least two directional audio components. The audio format transcoder (100) comprises a converter (110) for converting the input audio signal into a converted signal, having a converted signal representation and a converted signal direction of arrival. The audio format transcoder (100) further comprises a position determiner (120) for determining at least two spatial positions of at least two spatial audio sources and a processor (130) for processing the converted signal representation using the at least two spatial positions to obtain at least two separated audio source measurements. |
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Invention relates to means of encoding and decoding an audio stream based on transformation of an input audio signal. An audio stream is obtained, which contains information describing the frequency band of the audio content and information describing a multi-band quantisation error. The multi-band quantisation error is determined for a plurality of frequency bands of the input audio signal in which there is gain information for separate bands. The average quantisation error for the plurality of frequency bands of the input audio signal is calculated. Frequency bands whose spectral components are completely quantised to zero are excluded. Noise is input into the spectral components for the plurality of frequency bands, wherein gain information in separate frequency bands is associated with the general value of intensity of multi-band noise. |
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Using multichannel decorrelation for improved multichannel upmixing Invention relates to means of multichannel upmixing using multichannel decorrelation. A system of linear equations is used to upmix a number N of audio signals to generate a larger number M of audio signals that are psychoacoustically decorrelated with respect to each other and that can be used to improve representation of a diffuse sound field. The linear equations are defined by a matrix which specifies a set of vectors in an M dimensional space that are substantially orthogonal to each other. Methods of deriving the system of linear equations are disclosed. |
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Vector quantiser, vector inverse quantiser and methods therefor Invention relates to computer engineering. The vector quantiser comprises a first selecting section that selects a classification code vector indicating a type of a feature correlated with a quantisation target vector, from a plurality of classification code vectors; a second selecting section that selects a first codebook corresponding to the selected classification code vector from a plurality of first codebooks; a first quantisation section that quantises the quantisation target vector using a plurality of first code vectors forming the selected first codebook, to produce a first code; a third selecting section that selects a first matrix corresponding to the selected classification code vector from a plurality of matrices; and a second quantisation section that quantises a first residual vector which is the difference between the first code vector, said first code and the quantisation target vector, using a plurality of second code vectors and the selected first matrix, to produce a second code. |
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Hardware unit, method and computer programme for expanding compressed audio signal Invention relates to computer engineering. The hardware unit for expanding a compressed audio signal, having one or more compressed audio channels, into an expanded audio signal, having a plurality of expanded audio channels, wherein the hardware unit includes an expansion unit tuned to use current values of variable expansion parameters for expanding the compressed audio signal and obtaining the expanded audio signal; as well as a parameter interpolation module adapted to obtain one or more current interpolated expansion parameters to be used in the expansion unit based on information describing a first complex-valued expansion parameter and a subsequent second complex-valued expansion parameter, wherein the parameter interpolation module is adapted for independent interpolation between the magnitude of the first complex-valued expansion parameter and the magnitude of the second complex-valued expansion parameter, and between the phase of the first complex-valued expansion parameter and the phase (256) of the second complex-valued expansion parameter, to obtain one or more current interpolated complex-valued expansion parameters. |
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Method and device for audio signal processing Invention relates to audio signal processing. Proposed method comprises audio signal filtration for division into two frequency bands and generation of multiple sub bands for signal of every frequency band. Note here that for signal in one frequency band multiple signals of sub bands are generated by conversion from time band to frequency band. For another frequency band, multiple signals of sub bands are generated with the help of bank of sub band filters. Proposed device comprises one processor and one memory device with computer program code. Note also that one memory device and one computer program code are configured to make at least one processor control over process implementation. |
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Audio encoder and audio decoder for encoding and decoding audio signal readings Audio encoder (100) for encoding audio signal readings includes a first encoder with time superposition (aliasing) (110) for encoding audio readings in a first encoding region according to a first windowing rule, with attachment of a start window and a stop window. The audio encoder (100) further includes a second encoder (120) for encoding readings in a second encoding region, which processes a frame format-set number of audio readings and comprising a series of audio readings of an encoding mode stabilisation interval, which applies a different, second encoding rule, wherein the frame of the second encoder (120) is an encoded representation of time-consecutive audio signals, the number of which is set by the frame format. The audio encoder (100) also includes a controller (130) which performs switching from the first encoder (110) to the second encoder (120) according to the characteristics of the audio readings and corrects the second windowing rule when switching from the first encoder (110) to the second encoder (120) or modifies the start window or stop window of the first encoder (110) while keeping the second windowing rule unchanged. |
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Angle-dependent operating device or method for generating pseudo-stereophonic audio signal Invention relates to audio signals and devices or methods for generation, transmission, conversion and reproduction thereof. A monophonic audio signal of arbitrary directional characteristic is subjected to targeted propagation time difference (1210, 1211) and loudness corrections (derived from 1212 and 1213), while parameterising the angle phi (1205) included by the main axis (1203) and the direction of impingement of the sound source (1204), an imaginary left opening angle alpha (1206), and an imaginary right opening angle beta (1207), and the directional characteristic of the monophonic signal to be stereophonicised (represented in polar coordinates). The result is an M-signal and an S-signal allowing MS matrix formation (and thereby the stereophonic reproduction of the originally monophonic audio signal). |
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Noise filler for creating a noise-filled spectral representation of an audio signal based on an input spectral representation of the audio signal consists of a spectral region identifier configured to identify spectral regions of the input spectral representation spaced from non-zero spectral regions of the input spectral representation by at least one intermediate spectral region, to obtain identified spectral regions, and a noise inserter configured to selectively introduce noise into the identified spectral regions to obtain the noise-filled spectral representation of the audio signal. A noise filling parameter calculator for calculating a noise filling parameter based on a quantised spectral representation of an audio signal comprises a spectral region identifier, as mentioned above, and a noise value calculator configured to selectively consider quantisation errors in the identified spectral regions for calculation of the noise filling parameter. Accordingly, an encoded audio signal representation representing the audio signal can be obtained. |
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Encoding method, decoding method, encoder, decoder, programme and recording medium Invention relates to an encoding method and more specifically to a method of encoding a fundamental tone period. Encoding involves calculating the fundamental tone period for time sequence signals in a predefined time interval and outputting a code corresponding thereto. In said encoding, resolution for expressing fundamental tone periods and/or fundamental tone period encoding mode is switched according to whether an index which indicates the level of periodicity and/or stationarity of the time sequence signals satisfies a condition which indicates high or low periodicity and/or stationarity. In said decoding, in accordance with whether the index which indicates the level of periodicity and/or stationarity, an index included in the input code or obtained based on the input code, which corresponds to the predefined time interval, satisfies the condition which indicates high periodicity and/or stationarity; the decoding mode for the code included in the input code, which corresponds to fundamental tone periods, is switched for decoding a code which corresponds to fundamental tone periods in order to obtain fundamental tone periods which correspond to the predefined time interval. |
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Method and apparatus for synchronising highly compressed enhancement layer data Invention relates to data formats of multimedia applications which use hierarchical data layers. The method for encoding an audio or video signal has a base layer bit stream and an enhancement layer bit stream relating to the base layer bit stream. The base layer data and the enhancement layer data are structured into packets and packets of the base layer bit stream have corresponding packets of the enhancement layer bit stream. The method involves calculating a checksum of a packet of the base layer bit stream and a corresponding packet of the enhancement layer bit stream, as well as entropy encoding the packet of the base layer bit stream to obtain an entropy encoded, byte-aligned base layer packet starting with a synchronised word. |
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Input spectrum is broken into a plurality of subbands. A representative value is calculated for each subband using an arithmetic mean and a geometric mean. Nonlinear conversion is performed with respect to each representative value. The nonlinear conversion characteristic is amplified as the value increases. The representative value, which was subjected to nonlinear conversion for each subband, is smoothed in the frequency domain. |
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Audio signal decoder, temporary deformation loop data provider, method and computer program Audio signal decoder designed to provide a decoded representation of an audio signal based on an encoded representation of the audio signal, which includes information on evolution of a temporary deformation loop, includes a temporary deformation loop computer, a device for changing the scale of the temporary deformation loop data and a deformation decoder. The temporary deformation loop computer is designed to generate temporary deformation loop data through multiple restarting from a predefined starting value of the temporary deformation loop based on information on evolution of the temporary deformation loop, which describes time evolution of the temporary deformation loop. The device for changing the scale of temporary deformation loop data is designed to change the scale of at least part of temporary deformation loop data to avoid, reduce or eliminate non-uniformity during restart in a scaled version of the temporary deformation loop. The deformation decoder is designed to provide a decoded representation of an audio signal based on an encoded representation of the audio signal and by using the scaled version of the temporary deformation loop. |
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Method and apparatus for hierarchical encoding and decoding audio Method for hierarchical encoding audio includes dividing frequency domain coefficients of an audio signal after modified discrete cosine transform (MDCT) into a plurality of coding subbands, quantising and encoding values of the amplitude envelope of the coding subbands; distributing bits into each coding subband of the basic level, quantising and encoding frequency domain coefficients of the basic level to obtain encoded bits of the frequency domain coefficients of the basic level; calculating the amplitude envelope value of each encoding subband of the residual signal of the basic level; distributing bits into each encoding subband of the extended level, quantising and encoding the encoding signal of the extended level to obtain encoded bits of the ecoding signal of the extended level; multiplexing and packing the encoded bits of the amplitude envelope value of each encoding subband, which consists of frequency domain coefficients of the basic level and the extended level, encoded bits of the frequency coefficients of the basic level and encoded bits of the encoding signal of the extended level, followed by transmission to the decoding side. |
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Device and method for quantising and inverse quantising lpc filters in super-frame Invention relates to means of quantising, in a super-frame including a sequence of frames, LPC filters calculated during the frames of the sequence. The LPC filter quantising device comprises: an absolute quantiser for first quantising one of the LPC filters using absolute quantisation; and at least one quantiser of the other LPC filters using a quantisation mode selected from a group consisting of absolute quantisation and differential quantisation relative to at least one previously quantised filter among the LPC filters. For inverse quantising, at least the first quantised LPC filter is received and an inverse quantiser inverse quantises the first quantised LPC filter using absolute inverse quantisation. If any quantised LPC filter other than the first quantised LPC filter is received, an inverse quantiser inverse quantises this quantised LPC filter using one of the following inverse quantisation modes: absolute inverse quantisation and differential inverse quantisation relative to at least one previously received quantised LPC filter. |
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Method and apparatus for generating equalised multichannel audio signal Multiple time-superimposed audio signals are received at the input of an encoder. The time-superimposed signals are discretised to generate equalised frames of audio data of a predetermined size. Identical timestamps per unit time are assigned to all of the multiple superimposed audio signals. The stamped audio signals are included in a digital transport data stream. |
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Method of embedding digital information into audio signal Method of embedding digital information into an audio signal involves performing the following operations: dividing digital information into high-priority and low-priority streams, wherein the high-priority data are embedded via frequency-selective echo modulation, and the low-priority data are embedded through noise-like signals or using multi-carrier digital modulation; dividing the initial audio signal into a first frequency portion and a second frequency portion. The first frequency portion of the initial audio signal is modulated via frequency-selective echo modulation with different delay and echo signal amplitude values, and the second frequency portion of the initial audio signal is transmitted to a unit for psycho-acoustic analysis based on a psycho-acoustic model which takes into account a frequency and/or time masking effect, wherein the psycho-acoustic analysis unit generates at each analysis interval a spectral mask which reflects the audibility threshold of distortions, and said spectral mask is applied to a multi-carrier signal or to a noise-like signal with subsequent addition of the obtained signal in the psycho-acoustic analysis unit to the second frequency portion of the initial audio signal; combining the two modulated frequency portions of the acoustic signal. |
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Apparatus for generating output spatial multichannel audio signal Apparatus (100) for generating an output spatial multichannel audio signal is based on an input audio signal and an input parameter. The apparatus (100) includes a decomposer (110) for breaking down the input audio signal based on the input parameter to obtain a first signal component and a second signal component, different from each other. The apparatus (100) also consists of a rendering unit (110) for rendering the first signal component to obtain a first rendered signal with a first semantic property and for rendering the second signal component to obtain a second rendered signal with a second semantic property different from the first semantic property. The apparatus (100) includes a processor (130) for processing the first and second rendered signals to obtain an output spatial multichannel audio signal. |
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Method of creating codebook and search therein during vector quantisation of data Method can be used to reduce consumption of computational resources and the required size of storage devices when creating codebooks and executing reference vector search algorithms therein, including when performing low-speed speech signal coding. The technical result of the disclosed method is reducing the required size of storage devices and reducing consumption of computing computational resources when performing search in a codebook during vector quantisation. The set task is achieved by constructing a special codebook structure based on neural networks using training algorithms with adjustment. Search is performed in form of step-by-step hierarchical vector quantisation. The resultant vector is a sum of code vectors found at each step. The disclosed method can be used to reduce consumption of computational resources and the required size of storage devices when executing reference vector search algorithms in a codebook. |
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Method and apparatus for selective signal coding based on core encoder performance In a selective signal encoder, an input signal is first encoded (1004)using a core layer encoder to produce a core layer encoded signal. The core layer encoded signal is decoded (1006) to produce a reconstructed signal, and an error signal is generated (1008) as the difference between the reconstructed signal and the input signal. The reconstructed signal is compared (1010) with the input signal. One of two or more enhancement layer encoders is selected (1014, 1016) depending on the comparison and is used to encode the error signal. The core layer encoded signal, the enhancement layer encoded signal and a selection indicator are output (1018) to a channel (e.g., for transmission or storage). |
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Encoding device, decoding device and method Disclosed is an encoding device which can accurately specify a band having a large error among all bands by using a small calculation amount. The device includes: a first position identification unit (201) which uses a first layer error conversion coefficient indicating an error of decoding signal for an input signal so as to search for a band having a large error in a relatively wide bandwidth in all the bands of the input signal and generates first position information indicating the identified band; a second position identification unit (202) which searches for a target frequency band having a large error in a relatively narrow bandwidth in the band identified by the first position identification unit (201) and generates second position information indicating the identified target frequency band; and an encoding unit (203) which encodes a first layer decoding error conversion coefficient contained in the target frequency band. The first position information, the second position information, and the encoding unit are transmitted to a communication partner. |
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Encoder, decoder, encoding method and decoding method In the encoder, spectrum residue form vector candidates are stored in a spectrum residue form codebook (305), spectrum residue gain candidates are stored in a spectrum residue gain codebook (307), and the spectrum residue form vector and the spectrum residue gain are successively output from the candidates in accordance with an instruction from a search unit (306). A multiplier (308) multiplies the spectrum residue form vector candidate by the spectrum residue gain candidate and sends the result to a filtration unit (303). Using the internal status of the filter, the filtration unit (303) filters the fundamental tone given by the filter status setting unit (302), lag T which is output by a lag setting unit (304), and the spectrum residue form vector and the controlled gain. |
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Audio encoder and audio decoder for encoding frames presented in form of audio signal samples Audio encoder (100) for encoding frames presented in form of audio signal samples to obtain encoded frames, wherein a frame consists of a plurality of time domain audio signals, including a predictive coding analysis stage (110) and determining information on coefficients of a synthesis filter and prediction domain frame information based on a frame of audio samples. The audio encoder (100) further includes a domain converter (120) for converting a frequency domain audio sample frame and obtaining a frame spectrum and an encoding domain computer (130) for making a decision on encoded data for a frame based on information on coefficients and information on a prediction domain frame, or based on the frame spectrum. The audio encoder (100) includes a controller (140) for determining information on a switching coefficient for cases when the encoding domain computer decides that encoded data of the current frame are based on information on coefficients and information on a prediction domain frame, and [for cases] when data of a previous frame were encoded based on the spectrum of the previous frame and redundancy reducing encoder (150) for encoding information on the prediction domain frame, information on coefficients, information on the switching coefficient and/or frame spectrum. |
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Parametric stereophonic upmix apparatus (300, 400) generating a left signal (206) and a right signal (207) from a monophonic downmix signal (204) based on spatial parameters (205). Said parametric stereophonic upmix is characterised by that it comprises a means (310) for predicting a difference signal (311) comprising a difference between the left signal (206) and the right signal (207) based on the monophonic downmix signal (204) scaled with a prediction coefficient (321). Said prediction coefficient is derived from the spatial parameters (205). Said parametric stereophonic upmix apparatus (300, 400) further comprises an arithmetic means (330) for deriving the left signal (206) and the right signal (207) based on a sum and a difference of the monophonic downmix signal (204) and said difference signal (311). |
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Concealment of transmission error in digital audio signal in hierarchical decoding structure Method is provided for concealing a transmission error in a digital signal broken down into a plurality of successive frames associated with different time intervals in which, at reception, the signal may contain erased frames and normal frames, the normal frames containing information (inf.) relating to the concealment of frame loss. The method is implemented during hierarchical decoding using core decoding and transform-based decoding using low-delay windows introducing a time delay of less than a frame with respect to the core decoding. To replace at least the last frame erased before a normal frame, the method comprises: a step (23) of concealing a first set of missing samples for the erased frame, implemented in a first time interval; a step (25) of concealing a second set of missing samples for the erased frame taking into account information of said normal frame and implemented in a second time interval; and a step (29) of switching between the first set of missing samples and the second set of missing samples so as to obtain at least a part of the missing frame. |
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Method of reducing transmission rate of linear prediction low bit rate voders Objective set in a linear prediction vocoder is achieved by avoiding transmission of drive signal information over a communication channel. The drive signal is identified immediately at reception from data on parameters of the synthesising model using a neural network. Information on coefficients of the forming model, gain factor, parameters which characterise the encoded speech signal which are calculated on each quasi-stationary analysis segment of the speech signal are transmitted over the communication channel. |
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Sound encoding device includes: a downmixing signal generating unit (410) which generates in the time domain a first downmixing signal which is one of a 1-channel sound signal and a 2-channel sound signal from an input multi-channel sound signal; a downmixing signal encoding unit (404) which encodes the first downmixing signal; a first conversion unit t-f (401), which converts the input multi-channel sound signal into a frequency-domain multi-channel sound signal; and a spatial information computing unit (409), which generates spatial information for generating a multi-channel sound signal from the downmixing signal. |
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Invention relates to lossless multi-channel audio codec which uses adaptive segmentation with random access point (RAP) and multiple prediction parameter set (MPPS) capability. The lossless audio codec encodes/decodes a lossless variable bit rate (VBR) bit stream with random access point (RAP) capability to initiate lossless decoding at a specified segment within a frame and/or multiple prediction parameter set (MPPS) capability partitioned to mitigate transient effects. This is accomplished with an adaptive segmentation technique that fixes segment start points based on constraints imposed by the existence of a desired RAP and/or detected transient in the frame and selects a optimum segment duration in each frame to reduce encoded frame payload subject to an encoded segment payload constraint. RAP and MPPS are particularly applicable to improve overall performance for longer frame durations. |
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When a frame immediately preceding a target encoding frame to be encoded by a first encoding unit operating according to a linear predictive coding scheme is encoded by a second encoding unit operating according to a coding scheme different from the linear predictive coding scheme, the target encoding frame can be encoded according to the linear predictive coding scheme by initialising the internal state of the first encoding unit. Consequently, encoding processing performed according to a plurality of coding schemes including the linear predictive coding scheme and a coding scheme different from the linear predictive coding scheme can be realised. |
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When a frame immediately preceding a target encoding frame to be encoded by a first encoding unit operating according to a linear predictive coding scheme is encoded by a second encoding unit operating according to a coding scheme different from the linear predictive coding scheme, the target encoding frame can be encoded according to the linear predictive coding scheme by initialising the internal state of the first encoding unit. Consequently, encoding processing performed according to a plurality of coding schemes including the linear predictive coding scheme and a coding scheme different from the linear predictive coding scheme can be realised. |
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Described is a method and a system for generating a converted output signal from an input signal using a conversion coefficient T. The system includes an analysis window with length La, which extracts an input signal frame, and a unit which analyses transformation of the order M, which transforms discrete values into M complex coefficients. M depends on the conversion coefficient T. The system also includes a nonlinear processing unit, which changes the phase of complex coefficients using the conversion coefficient T, a unit which synthesises transformation of the order M, which transforms the changed coefficients into M changed discrete values, and a synthesis window with length Ls, which generates an output signal va(n) frame. |
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Apparatus for providing a set of spatial indicators associated with an upmix audio signal, having more than two channels, based on a double-channel microphone signal, comprises a signal analyser and an additional spatial information generator. The signal analyser is configured to receive component energy information and direction information based on the double-channel microphone signal such that the component energy information describes an estimate of energy of the direct sound component of the double-channel microphone signal, and such that the direction information describes an estimate of the direction from which the direct sound component of the double-channel microphone signal arrives. The additional spatial information generator is configured to compare component energy information and direction information with spatial indicator information which describes the set of spatial indicators associated with an upmix audio signal, having more than two channels. |
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Apparatus and method for encoding/decoding audio signal using aliasing switch scheme Apparatus for encoding an audio signal includes a windower (11) for windowing a first block of the audio signal using an analysis window, the analysis window having an aliasing portion and a further portion; a processor (12) for processing a first sub-block of the audio signal associated with the aliasing portion by transforming the first sub-block from the domain to another domain, subsequent to windowing the first sub-block to obtain a processed first sub-block, and for processing a second sub-block of the audio signal associated with the further portion by transforming the second sub-block from the domain to another domain before windowing the second sub-block to obtain a processed second sub-block. The apparatus also includes a transformer (13) for converting the processed first sub-block and the processed second sub-block from the different domain into a further domain using the same block transform rule to obtain a converted first block, which is compressed using any of the known data compression algorithms. Therefore, critically sampled switching between two encoding modes is obtained since aliasing portions of the analysis window that appear in two different domains are suited to each other. |
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Efficient use of stepwise transmitted information in audio encoding and decoding Audio signal can be derived using correlation information indicating a correlation between first and second input audio signals, when a signal characterisation information, indicating at least a first or a second different characteristic of the input audio signal, is additionally considered. Phase information indicating a phase relation between the first and the second input audio signals is derived, when the input audio signals have the first characteristic. The phase information and a correlation measure are included into the encoded representation when the input audio signals have the first characteristic, and only the correlation information is included into the encoded representation when the input audio signals have the second characteristic. |
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Audio signal decoder and method of controlling audio signal decoder balance To support stereo perception, localisation vibration of a decoded signal is suppressed. A selection unit (220) selects balance parameters if balance parameters are input data from a gain decoding unit (210), or selects input data of balance parameters from a gain computation unit (223) if there are no input data of the balance parameter from the gain decoding unit (210), and outputs the selected balance parameters to a multiplier unit (221). The multiplier unit (221) multiplies gain input data from the selection unit (220) by decoded input data of a monophonic signal from a monophonic decoding unit (202), to process balance control. |
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Method is proposed to compensate for losses of sound signal frames in the MDCT area, including: a step a, during which, when the current lost frame is the P frame, a set of forecast frequencies is received, for each frequency in this set they use phases and amplitudes of multiple frames before the (P-1) frame in the area MDCT-MDST to forecast phase and amplitude of the P frame, and the forecast phase and amplitude are used for production of MDCT coefficients of the P frame, corresponding to each frequency; a step b, at which for frequencies outside the set of forecast frequencies the MDCT coefficients of multiple frames before the P frame are used for calculation of MDCT coefficient values of P frame on these frequencies; a step c, during which they perform inverse modified discrete cosine transformation (IMDCT) for MDCT coefficients of the P frame at all frequencies for production of the signal in the time area for the P frame. Also a compensator is proposed for losses of frames. The invention has advantages of no delay, low volume of calculations, low volume of memory space and simplicity of realisation. |
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Coding/decoding based on transformation with adaptive windows Invention provides for coding/decoding of a digital signal, in particular, using transformation with closure by means of weighing windows. According to the invention, two serial units of signal count with equal size may be weighed by appropriate different serial windows. These two windows may be selected independently on each other in accordance with the criterion corresponding to signal characteristics (entropy, ratio of data transfer speed/distortion), which are determined for each of two units. |
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Coding device, decoding device and method Speech coding device comprises a coding section of the first level, which performs processing by coding in respect to the input speech signal; a decoding section of the first level, which performs processing by decoding using coded data of the first level; a section for calculation of erroneous transformation coefficients of the first level, which converts the erroneous signal of the first level into a frequency area for calculation of erroneous coefficients of transformation of the first level; and a coding section of the second level, which performs processing by coding in respect to erroneous transformation coefficients of the first level, besides, the coding section of the second level comprises: a setting section; a selection section; a connected strip configuration section, which connects a strip selected from the low-frequency strip, and a fixed strip from the high-frequency strip, in order to configure the connected strip; and a section of coded data generation, which codes erroneous transformation coefficients of the first level, included into the connected strip, in order to generate coded data of the second level. |
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Mixing of incoming information flows and generation of outgoing information flow Device (500) is used for mixing of multiple incoming information flows (510), in which each of incoming information flows (510) comprises a frame (540) of audio data in a spectral area, the frame (540) of the incoming information flow (510), comprising spectral information for multiple spectral components. The device comprises a unit of data processing (520), arranged to compare frames (540) of multiple incoming information flows (510). The data processing unit (520) is also arranged to determine, on the basis of comparison, for a spectral component of an outgoing frame (550) of an outgoing information flow (530) only one incoming information flow (510) from multiple incoming information flows (510). The data processing unit (520) is further arranged to generate an outgoing information flow (530) by means of copying of at least a part of information of the appropriate spectral component of the frame of the certain information flow (510), in order to describe the spectral component of the outgoing frame (550) of the outgoing information flow (530). |
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Audio encoding device and audio decoding device Audio encoding device (100) for encoding coefficient segments, having different time or frequency resolution of the selected audio signal includes a processor (110) for obtaining encoding context for the current encoded coefficient of the current segment based on the pre-encoded coefficient of the previous segment of a pre-encoded coefficient, having a different time or frequency resolution from the current encoded coefficient. The audio encoding device (100) also has an entropy (120) encoding device which encodes the current coefficient based on the encoding context to obtain an encoded audio stream. |
Another patent 2513645.