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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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Electronic computer has random access memory, the output of which is connected to an arithmetic logic unit, as well as rows of photocells which respond to red light and are connected through switches to the random access memory. The output of the arithmetic logic unit is connected through switches to thirty comparison units. Outputs of the thirty comparison units are connected to control electrodes of thirty switches, respectively. A pulse generator is connected to inputs of the thirty switches, outputs of which are connected to inputs of the thirty switches, respectively. Outputs of the thirty switches are connected to the random access memory of a bit-map display. |
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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 refers to medical equipment, namely to ultrasound therapeutic apparatuses. An ultrasonic transducer contains at least one ultrasonic radiating element with ultrasonic wave front sets radiated by the element representing spherical surfaces of the same radius, while the ultrasonic radiating element performs a function of ultrasound reflection and forms a spherical resonant cavity. If using more than one ultrasonic radiating element, they are configured to form a combined spherical resonant cavity. An internal space of the spherical resonant cavity is shaped as a full spherical envelope or a truncated spherical envelope with a sphere centre inside, while the ultrasonic waves generated by at least one ultrasonic radiating element are focused in the area which comprises the sphere centre of the spherical resonant cavity. |
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Audio encoder and bandwidth extension decoder Invention relates to means of generating an equalised multichannel audio signal. An audio encoder for obtaining an output signal using an input audio signal comprises a patch generator, a comparator and an output interface. The patch generator generates at least one bandwidth extension signal, having a high-frequency band. The high-frequency band of the bandwidth extension signal is based on a low frequency band of the input audio signal. A comparator calculates a plurality of comparison parameters. A comparison parameter is calculated based on a comparison of the input audio signal and a generated bandwidth extension signal. Each comparison parameter of the plurality of comparison parameters is calculated based on a different offset frequency between the input audio signal and a generated high bandwidth signal. Further, the comparator determines a comparison parameter from the plurality of comparison parameters, wherein the determined comparison parameter satisfies a predefined criterion. |
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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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Piano double repetition mechanism Piano double repetition mechanism comprises at least one spring 22, one end of which interacts with the upper part of a jack 8. The mechanism further includes at least one bar mounted on at least two pins 23 which pass through openings in the bar and are mounted on the main rail 4. The second end of the spring 22 is connected to the board. |
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Invention relates to means of filtering a multichannel audio signal, having a speech channel and at least one non-speech channel. The method includes determining at least one attenuation control value which serves as a feature of the extent of similarity between speech-related content which is defined by the speech channel and speech-related content which is defined by the non-speech channel; attenuating the non-speech channel in response to at least one attenuation control value; scaling the raw attenuation control signal (e.g. a gain control signal with suppression of a weak signal with a stronger signal) for the non-speech channel in response to at least one attenuation control value. |
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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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Acoustic echo suppressing device and front-end conference call device Invention relates to acoustic echo suppressing means. An acoustic echo suppressor includes an input interface (230) means for extracting a downmix signal (310) from an input signal (300) which contains downmix (310) and overhead parametric information (320), collectively representing a multichannel signal; and also includes a computer (220) for calculating transmission factors of an adaptive filter (240) based on the downmix signal (310) and a microphone signal (340) or a signal derived from a microphone signal; and an adaptive filter (240) for a microphone signal (340) or a signal derived from a microphone signal, using assigned transmission factors for suppressing echo excited by a multichannel signal in the microphone signal (340). |
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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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Electronic musical keyboard instrument "maxbox" Electronic musical keyboard instrument includes a system of interconnected sound-extraction units, wherein each unit is designed to generate one fundamental audio frequency and is matched on that frequency with audio frequencies generated by other units, wherein the number of sound-extraction units, the principle of their matching on generated audio frequencies, the labelling of sound-extraction units on the instrument and spatial arrangement thereof are defined by a multi-step musical pitch, formed from a discrete set of monochromatic sounds, the instrument being characterised by that the system of sound-extraction units consists of independent keyboards with control units and is configured to use an octave with the step 2, 3, 4, 5, 6, 7, 8, 9, 10, wherein the number of tones in an octave varies from 4 to 25, the structure of the octave itself has the capacity for a combinational set of a series of non-uniform tones: tone, 1/2, 1/3, 1/4, 1/5, 1/6, 1/7, 1/8, 1/9 and 1/10 tone, wherein the keyboard are arranged in two levels. |
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Piano double repetition mechanism Piano double repetition mechanism comprises at least one spring (22), at least one adjustment screw (23) connected to one end of the spring (22), and at least one angle section (24) in form of interconnected shelves, which is mounted with the outer surface of one shelf on the top surface of the jack damper. In the other shelf of the angle section there is at least one opening in which said adjustment screw (23), which is connected to one end of the spring, is inserted. The spring (22) exerts pressure through its other end on the top part of the jack (8). |
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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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Improved subband block based harmonic transposition Invention relates to means of encoding a source audio signal to form an equalised multichannel audio signal. A harmonic transposition method for high frequency reconstruction is employed. A signal of the analysed subband is generated from an input signal, where the signal of the analysed subband includes a series of complex-valued analysed discrete values, each having a phase and an amplitude. A signal of the synthesised subband is determined from the signal of the analysed subband using a subband transposition factor and a subband stretch factor. Block-based nonlinear processing is performed, where the amplitude of the discrete values of the signal of the synthesised subband is determined from the amplitude of corresponding discrete values of the signal of the analysed subband and a pre-determined discrete value of the signal of the analysed subband. A time-stretched and/or frequency-transposed signal is generated from the signal of the synthesised subband. |
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Structure with cellular filler for application in turbojet nacelle bearing panel Invention relates to the structure with cellular filler for application in aircraft turbojet nacelle bearing panel that makes an acoustic panel. The structure comprises the block with cellular filler including central part containing mid cellular cells and two lateral parts, each including multiple cellular cells. One part of connection cellular cells has one extra wall to make a connection. Block or blocks are interconnected by one connection zone produced by punching two overlapped extra walls of connection cellular cells pertaining to different lateral parts. |
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Invention relates to means of estimating the quality of an audio signal for a multimedia telecommunication service. The method involves calculating the audio packet loss frequency if at least one audio packet to be estimated exists in once or constantly generated IP packet losses, wherein calculation of the audio packet loss frequency is based on information from the received IP packets by calculating packet loss; calculating the average exposure time/average duration of the audio packet based on information of the received IP packets, wherein the average exposure time serves as the average time during which the quality of the audio signal has an effect at loss frequency of audio packets contained in one-time loss of audio packets; estimating subjective quality based on audio packet loss frequency and either the average exposure time or average duration of the audio packet; calculating audio data transmission speed for calculating the audio data transmission speed based on information from the received IP packets. The subjective quality estimate is calculated based on the quality of the encoded audio signal, audio packet loss frequency and average exposure time. |
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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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Automotive integral noise killing module Invention relates to transport engineering. integral noise-killing module is composed by battery of acoustic resonators integrated with standard devices. Said battery consists of acoustic resonators and/or volume expansion chambers. Open necks of the latter are communicated with the underhood space. Said battery features the following overall dimensions: length L, width B and height H. Said dimensions are composed by standard parts, that is, hood, front board and wheels mud flaps, front bumper, engine room bottom shield, i.e. engine mud flap, frame, grate, cooling system fan case and facing. Said parts make the engine room front upper and lower vent openings. Battery of acoustic resonators is composed by quarter-wave acoustic resonators and/or half-wave acoustic resonators and/or Helmholtz resonators and/or expansion chambers. Frequencies of oscillations used by said resonators fR equal or differ from those of lower acoustic modes fmL, fmB, fmH of engine room space by not over 1.2 times. Overall dimensions and expansion chamber cavities ratio multiple thereof are not equal and are not multiple of overall dimensions and engine room L:B:H ratio multiple thereof. Mutual radio of expansion chambers sizes and those of said spaces multiple thereof differ by at least 1.2 times. |
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Binaural rendering of multi-channel audio signal Binaural rendering of a multi-channel audio signal into a binaural output signal is described. The multi-channel audio signal includes a stereo downmix signal (18) into which a plurality of audio signals are downmixed; and side information includes downmix information (DMG, DCLD), indicating for each audio signal, to what degree the corresponding audio signal was mixed in the first channel and second channel of the stereo downmix signal (18), respectively, as well as object level information of the plurality of audio signals and inter-object cross correlation information, describing similarity between pairs of audio signals of the plurality of audio signals. Based on a first rendering prescription, a preliminary binaural signal (54) is computed from the first and second channels of the stereo downmix signal (18). A decorrelated signal ( X d n , k ) is generated as an perceptual equivalent to a mono downmix (58) of the first and second channels of the stereo downmix signal (18) being, however, decoded to the mono downmix (58). |
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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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Apparatus and method of generating wide bandwidth signal Invention relates to audio signal processing devices. An input signal is presented for a first band with first resolution data and for a second band with second resolution data, the second resolution being lower than the first resolution. A patch generator generates a first patch from the first band of the input signal according to a first patch generation algorithm and generates a second patch from the first band of the input signal according to a second patch generation algorithm. Spectral density of the second patch generated according to the second patch generation algorithm is higher than the spectral density of the first patch generated according to the first patch generation algorithm. A coupler merges the first patch, the second patch and the first band of the input signal to obtain a wide bandwidth signal. The apparatus for generating a wide bandwidth signal scales the input signal according to the first patch generation algorithm and according to the second patch generation algorithm or scales the first patch and the second patch such that the wide bandwidth signal satisfies the spectrum envelope criterion. |
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Muffler comprises a cylindrical body, rigidly connected to the end inlet and outlet nozzles, with a central partition, the body from inside is lined with a sound-absorbing structure, and the central partition is made in the form of a sound-absorbing element, having a carcass, which at two sides is lined with a sound-absorbing material, besides, the carcass of the sound absorbing element may rotate in the plane perpendicular to the direction of movement of an aerodynamic flow, and the sound-absorbing structure is made from three layers of the sound-absorbing material, at the same time the first layer, a stiffer one, is made as solid and profiled and is fixed on the smooth surface, and the second layer, a softer one, compared to the first one, is made as interrupted and is arranged in the focus of the sound-reflecting surfaces of the first layer, and the third layer of the sound-absorbing element is made of a foamed sound-absorbing material, for instance, construction sealing foam, and is arranged between the first stiffer layer and perforated surface of the sound-absorbing element. |
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Invention relates to design of string musical instruments, e.g. bowed or plucked electrical instruments, having a resonance chamber. The string musical instrument has a neck joined to the body and consisting of two longitudinal parts, the neck having in its middle part a rod which is arched to the base, the rod having end fixing members, a string holder, a sound pickup, a peg mechanism, strings consisting of two sides and two decks of the body with a sound chamber made under the string holder, the open part of which faces the lower end of the body, wherein the sound chamber of the string musical instrument is extended to the point of attachment of the neck to the upper part of the body, the string holder is arched and attached to the sides, and the upper deck of the instrument has a variable thickness which falls from the open part of the sound chamber to the point of attachment of the neck to the body. |
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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 of detecting emotions from voice Invention relates to means for recognition of human emotions from voice. Intensity of the voice and tempo, defined by the rate at which the voice appears, are detected, respectively, and intonation which reflects the picture of intensity variation in each word pronounced by the voice is detected based on the input voice signal in form of a time value. A first variation value, indicating intensity variation of the detected voice in the direction of the time axis, a second variation value, indicating tempo variation of the voice in the direction of the time axis, and a third variation value indicating intonation variation of the voice in the direction of the time axis are obtained. The voice signal of a Russian-speaking subscriber is input and intensity of the voice and tempo is then detected. Once the third variation value is obtained, the base frequency of the voice signal is detected and a fourth variation value which indicates base frequency variation in the direction of the time axis is obtained; signals expressing the emotional state of anger, fear, grief and pleasure are generated, respectively, based on said first, second, third and fourth variation values. |
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Method of re-sounding audio materials and apparatus for realising said method Method and apparatus improve the quality of the teaching phase, improves match of the voice of a user (target speaker) in a converted speech signal and facilitates a one-time teaching phase for different audio materials. Said technical result is achieved due to that a program-controlled electronic information processing device (PCEIPD) generates an acoustic base of initial audio materials (ABIA) and acoustic teaching base (ATB). Data are transmitted from the ABIA to display a list of initial audio materials on a monitor screen. Upon selecting at least one audio material from the list of ABIA, data on said material are transmitted to PCEIPD random-access memory for storage. Files are selected from the ATB of teaching phrases of the speaker, said files being converted to audio phrases and transmitted to the user at an audio playback device. Through a microphone, the user repeats audio phrases, during playback of which the monitor screen displays text of the played back phrase and a cursor which moves along the text of the phrase in accordance with how the user should repeats it. Files are created in accordance with the played back phrases, which are stored according to the order of playing back phrases in the target speaker acoustic base (TSAB) formed. The PCEIPD monitors the rate of the played back phrase and its volume. A conversion function is created. Using the conversion function, ABIA files are converted for storage in the acoustic base of converted audio materials (ABCA) and providing the user with data on the converted audio materials on the monitor screen. |
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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 and discriminator for classifying different signal segments Method and discriminator for classifying different segments of a signal designed to classify different segment of a signal which comprises segments of at least a first type and second type, e.g. musical and speech segments, a short- term classification (150) signal based on at least one short-term feature extracted from the signal and a short- term classification result (152); a long-term classification (154) signal based on at least one short-term feature and at least one long-term feature extracted from the signal and a long-term classification result (156). The short-term classification result (152) and the long-term classification result (156) are combined (158) to provide an output sampling signal (160) indicating whether a segment of the signal is of the first type or of the second type. |
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Apparatus for processing an audio signal to obtain control information for a speech enhancement filter (12) comprises a feature extractor (14) for extracting at least one feature in the frequency band of a plurality of frequency bands of a short-time spectral representation of a plurality of short-time spectral representations, where the at least one feature represents a spectral shape of the short-time spectral representation in the frequency band. The apparatus further comprises a feature combiner (15) for combining the at least one feature for each frequency band using combination parameters to obtain the control information for the speech enhancement filter for a time portion of the audio signal. The feature combiner can use a neural network regression method, which is based on combination parameters determined in a training phase for the neural network. |
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Synthesiser with accompaniment and vocal-instrument processor Musical device enables to create automatic accompaniment and harmonise vocal performed by one musician singing and playing a polyphonic instrument. The musical instrument has a linear output and includes a main unit and a chord recognition unit in which sound of the polyphonic musical instrument is converted to notes of the struck chord. Sound of a chord is generated using a tone generator. The device also has a first foot-operated controller for controlling the module of the automatic accompaniment and a second foot-operated controller for controlling the vocal and instrument processor. |
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Complexity scalable perceptual tempo estimation Method and system for extracting tempo information of an audio signal from an encoded bit stream of the audio signal comprising spectral band replication data are described. The method comprises steps of determining a payload quantity associated with the amount of spectral band replication data contained in the encoded bit stream for a time interval of the audio signal; repeating the determining step for successive time intervals of the encoded bit stream of the audio signal, thereby determining a sequence of payload quantities; identifying periodicity in the sequence of payload quantities; and extracting tempo information of the audio signal from the identified periodicity. |
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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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Method and apparatus for suppressing narrow-band noise in passenger cabin of vehicle Apparatus for suppressing noise in the passenger cabin of a vehicle has at least one converter, a programmable computer and at least one acoustic sensor. The computer is configured to apply the electroacoustic model of the passenger cabin to the model of an adjustment system, having a fixed coefficient master controller which is connected to a variable coefficient unit, having a Yule parameter in form of a Yule Q unit. In the method, the first step involves determining and calculating the electroacoustic model and control law for at least one predetermined noise frequency. At the second step, the computer applies the control law to the electroacoustic model in real time in accordance with the current noise frequency to be suppressed. |
Another patent 2513823.