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Небесная энциклопедия

Космические корабли и станции, автоматические КА и методы их проектирования, бортовые комплексы управления, системы и средства жизнеобеспечения, особенности технологии производства ракетно-космических систем

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Мониторинг СМИ

Мониторинг СМИ и социальных сетей. Сканирование интернета, новостных сайтов, специализированных контентных площадок на базе мессенджеров. Гибкие настройки фильтров и первоначальных источников.

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Поддерживает ввод нескольких поисковых фраз (по одной на строку). При поиске обеспечивает поддержку морфологии русского и английского языка
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Применить Всего найдено 6668. Отображено 100.
09-02-2012 дата публикации

Method of processing signal, encoding apparatus thereof, decoding apparatus thereof, and signal processing system

Номер: US20120035939A1
Автор: Han-gil Moon, Hyun-Wook Kim, Jong-Hoon Jeong, Nam-Suk Lee
Принадлежит: SAMSUNG ELECTRONICS CO LTD

A method processing a signal, an encoding apparatus, and a decoding apparatus are provided. The method of processing a signal includes restoring a down-mixed original signal using a re-quantized prediction parameter to generate a restored signal in an encoding apparatus; generating mute information indicating whether the down-mixed original signal has been muted, according to a value of the restored signal; and transmitting the mute information and the down-mixed original signal from the encoding apparatus to a decoding apparatus.

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Номер записи: 1
05-04-2012 дата публикации

Multi-channel audio encoding and decoding

Номер: US20120082316A1
Автор: Naveen Thumpudi, Wei-ge Chen
Принадлежит: Microsoft Corp

An audio encoder and decoder use architectures and techniques that improve the efficiency of multi-channel audio coding and decoding. The described strategies include various techniques and tools, which can be used in combination or independently. For example, an audio encoder performs a pre-processing multi-channel transform on multi-channel audio data, varying the transform so as to control quality. The encoder groups multiple windows from different channels into one or more tiles and outputs tile configuration information, which allows the encoder to isolate transients that appear in a particular channel with small windows, but use large windows in other channels. Using a variety of techniques, the encoder performs flexible multi-channel transforms that effectively take advantage of inter-channel correlation. An audio decoder performs corresponding processing and decoding. In addition, the decoder performs a post-processing multi-channel transform for any of multiple different purposes.

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Номер записи: 2
09-08-2012 дата публикации

Method and device for forming a mixed signal, method and device for separating signals, and corresponding signal

Номер: US20120203362A1
Автор: Jean-Marc Brossier, Laurent Girin, Mathieu Parvaix, Sylvain Marchand
Принадлежит: Institut Polytechnique de Grenoble, Universite des Sciences et Tech (Bordeaux 1)

The invention relates to a method of formation of one or more mixed signals (S out ) on the basis of at least two digital source signals (S 1 , S 2 ), in particular audio signals, in which the mixed signal or signals (S out ) are formed by mixing the source signals (S 1 , S 2 ). In particular, a quantity characteristic of a source signal or of the mixing is determined and the value (W 1 , W 2 ) of the said characteristic quantity is watermarked on at least one of the signals (S 1 , S 2 , S out ). The invention also relates to a method of separation intended to separate, at least partially, at least one digital source signal contained in one or more mixed signals comprising a watermarked value of a quantity characteristic of a source signal or of the mixing. According to the method, the watermarked value of the quantity characteristic of the source signal or of the mixing is determined, and then the mixed signal or signals is or are processed as a function of the said value so as to obtain, at least partially, the said source signal. The invention also relates to the corresponding mixed signal (S out ), as well as the corresponding devices.

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Номер записи: 3
18-10-2012 дата публикации

Optimized parametric stereo decoding

Номер: US20120265542A1
Автор: Balazs Kovesi, Stephane Ragot, Thi Minh Nguyet Hoang
Принадлежит: France Telecom SA

A method and decoder are provided for parametrically decoding a stereo digital audio signal. The method includes synthesizing the stereo signal, per frequency sub-band, on the basis of a decoded mono signal ({circumflex over (M)}[j]), arising from a downmix of the stereo signal and from spatial information parameters of the stereo signal, in such a way that the signals obtained have the following form: L ^  [ j ] = c 1  [ j ] · M ^ 1  [ j ] R ^  [ j ] = c 2  [ j ] · M ^ 2  [ j ] , with {circumflex over (L)}[j] and {circumflex over (R)}[j] being channels of the synthesized signal, {circumflex over (M)} 1 [j] and {circumflex over (M)} 2 [j] being signals that are a function of the decoded mono signal and c 1 [j], c 2 [j] being gains, wherein the gains are calculated as follows: c 1  [ j ] = 2  I ^  [ j ] I ^  [ j ] + 1 c 2  [ j ] = 2 I ^  [ j ] + 1 with Î[j] being an amplitude ratio between the two channels of the stereo signal, obtained from the decoded parameters.

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Номер записи: 4
01-11-2012 дата публикации

Processing Stereophonic Audio Signals

Номер: US20120275604A1
Автор: Koen Vos
Принадлежит: Skype Ltd Ireland

Method, apparatus and computer program product for processing an input stereophonic audio signal to thereby generate a converted stereophonic audio signal representing the input stereophonic audio signal, the input stereophonic audio signal comprising a left input audio signal and a right input audio signal, and the converted stereophonic audio signal comprising a first converted audio signal and a second converted audio signal. The first converted audio signal is generated based on the sum of the left input audio signal and the right input audio signal. The second converted audio signal is generated based on the difference between a first function of the left input audio signal and a second function of the right input audio signal. The first and second functions are adjustable to thereby adjust at least one characteristic of the converted stereophonic audio signal.

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Номер записи: 5
13-12-2012 дата публикации

Apparatus and method for extracting a direct/ambience signal from a downmix signal and spatial parametric information

Номер: US20120314876A1
Автор: Bernhard Neugebauer, Jan Plogsties, Juergen Herre, Juha Vilkamo
Принадлежит: Fraunhofer Gesellschaft zur Forderung der Angewandten Forschung eV

An apparatus for extracting a direct and/or ambience signal from a downmix signal and spatial parametric information, the downmix signal and the spatial parametric information representing a multi-channel audio signal having more channels than the downmix signal, wherein the spatial parametric information has inter-channel relations of the multi-channel audio signal, is described. The apparatus has a direct/ambience estimator and a direct/ambience extractor. The direct/ambience estimator is configured for estimating a level information of a direct portion and/or an ambient portion of the multi-channel audio signal based on the spatial parametric information. The direct/ambience extractor is configured for extracting a direct signal portion and/or an ambient signal portion from the downmix signal based on the estimated level information of the direct portion or the ambient portion.

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Номер записи: 6
03-01-2013 дата публикации

Audio encoder, audio encoding method and program

Номер: US20130003980A1
Автор: Jun Matsumoto, Shiro Suzuki, Yasuhiro Toguri, Yuuji Maeda, Yuuki Matsumura
Принадлежит: Sony Corp

There is provided an audio encoder comprising a determination part determining, based on frequency spectra of audio signals of a plurality of channels, a mixing ratio as a ratio, relative to a frequency spectrum after mixing for each channel of the plurality of channels, of the frequency spectrum for another channel, a mixing part mixing the frequency spectra of the plurality of channels for each channel based on the mixing ratio determined by the determination part, and an encoding part encoding the frequency spectra of the plurality of channels after mixing by the mixing part.

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Номер записи: 7
24-01-2013 дата публикации

Binaural decoder to output spatial stereo sound and a decoding method thereof

Номер: US20130022205A1
Автор: Han-gil Moon, In-gyu Chun, Sun-Min Kim
Принадлежит: SAMSUNG ELECTRONICS CO LTD

A binaural decoder for an MPEG surround stream, which decodes an MPEG surround stream into a stereo 3D signal, and a decoding method thereof. The method includes dividing a compressed audio stream and head related transfer function (HRTF) data into subbands, selecting predetermined subbands of the HRTF data divided into subbands and filtering the HRTF data to obtain the selected subbands, decoding the audio stream divided into subbands into a stream of multi-channel audio data with respect to subbands according to spatial additional information, and binaural-synthesizing the HRTF data of the selected subbands with the multi-channel audio data of corresponding subbands.

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Номер записи: 8
31-01-2013 дата публикации

Audio encoder, audio decoder and related methods for processing multi-channel audio signals using complex prediction

Номер: US20130030819A1
Автор: Bernd Edler, Christian Helmrich, Heiko Purnhagen, Johannes Hilpert, Julien Robillard, Lars Villemoes, Matthias Neusinger, Nikolaus Rettelbach, Pontus Carlsson, Sascha Disch
Принадлежит: DOLBY INTERNATIONAL AB, Fraunhofer Gesellschaft zur Forderung der Angewandten Forschung eV

An encoder, based on a combination of two audio channels, obtains a first combination signal as a mid-signal and a residual signal derivable using a predicted side signal derived from the mid signal. The first combination signal and the prediction residual signal are encoded and written into a data stream together with the prediction information. A decoder generates decoded first and second channel signals using the prediction residual signal, the first combination signal and the prediction information. A real-to-imaginary transform may be applied for estimating the imaginary part of the spectrum of the first combination signal. For calculating the prediction signal used in the derivation of the prediction residual signal, the real-valued first combination signal is multiplied by a real portion of the complex prediction information and the estimated imaginary part of the first combination signal is multiplied by an imaginary portion of the complex prediction information.

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Номер записи: 9
28-02-2013 дата публикации

Method and apparatus for frequency domain watermark processing a multi-channel audio signal in real-time

Номер: US20130051564A1
Автор: Michael Arnold, Peter Georg Baum, Ulrich Gries, Xiaoming Chen
Принадлежит: Individual

Digital audio signal watermarking in real-time is difficult in an environment that has limited processing power. According to the invention, the channels in a data block-based audio multi-channel signal are prioritized with respect to watermarking importance, whereby the channel priority can change for different input signal data blocks. For a current input signal block, the most important channel is watermarked and the required processing time is determined. If this required processing time is shorter than a predefined application-dependent threshold, the next most important channel is marked and the additionally required processing time is determined, and so on. Due to the block-based nature of the audio watermarking including block overlap/add and due to the sensitivity of the resulting audio quality against blocking artifacts, several problems are solved in order to lead to acceptable performance and quality.

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Номер записи: 10
28-03-2013 дата публикации

Method and apparatus for down-mixing multi-channel audio

Номер: US20130077793A1
Автор: Chul-Woo Lee, Han-gil Moon
Принадлежит: SAMSUNG ELECTRONICS CO LTD

Provided are a multi-channel audio down-mixing method and apparatus for selecting down-mix target channels based on a calculation of correlations between channels and then down-mixing the down-mix target channels. The method includes: calculating correlations between channels of multi-channel audio; selecting a first channel and a second channel, among the channels of the multi-channel audio, that are to be down-mixed, based on the calculated correlations; and down-mixing the selected first channel and the selected second channel.

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Номер записи: 11
18-04-2013 дата публикации

Method and Apparatus for Generating Sideband Residual Signal

Номер: US20130094655A1
Автор: Hu Chen, Lang Yue, Miao Lei, Wu Wenhai
Принадлежит: Huawei Technologies Co., Ltd.

Embodiments of the present invention provide a method and an apparatus for generating a sideband residual signal. The method includes: comparing energy of a first signal input by a first sound channel with energy of a second signal input by a second sound channel; if the energy of the first signal is greater than the energy of the second signal, generating a sideband residual signal by allocating a monophonic quantization error to the first signal; and if the energy of the first signal is smaller than the energy of the second signal, generating a sideband residual signal by allocating a monophonic quantization error to the second signal. By using the method and apparatus provided in the embodiments of the present invention, it can be avoided that a monophonic quantization error has a greater impact on a signal whose energy is smaller. 1. A method for generating a sideband residual signal comprising:comparing energy of a first signal input by a first sound channel with energy of a second signal input by a second sound channel;generating the sideband residual signal by allocating a monophonic quantization error to the first signal when the energy of the first signal is greater than the energy of the second signal; andgenerating the sideband residual signal by allocating the monophonic quantization error to the second signal when the energy of the first signal is smaller than the energy of the second signal.2. The method according to claim 1 , further comprising generating the sideband residual signal by evenly allocating the monophonic quantization error to the first signal and the second signal when the energy of the first signal is equal to the energy of the second signal.3. The method according to claim 2 , further comprising obtaining a quantized value CLD_Q of a stereophonic parameter CLD before comparing the energy of the first signal input by the first sound channel with the energy of the second signal input by the second sound channel.4. The method according ...

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Номер записи: 12
18-04-2013 дата публикации

AUDIO ADJUSTMENT SYSTEM

Номер: US20130096926A1
Автор: III Robert C., Maling, Noh Daekyoung, Tracey James
Принадлежит: DTS LLC

An audio adjustment system is provided that can output a user interface customized by the provider of the audio system instead of the electronic device manufacturer. Such an arrangement can save both field engineers and manufacturers a significant amount of time. Advantageously, in certain embodiments, such an audio adjustment system can be provided without knowledge of the electronic device's firmware. Instead, the audio adjustment system can communicate with the electronic device through an existing audio interface in the electronic device to enable a user to control audio enhancement parameters in the electronic device. For instance, the audio adjustment system can control the electronic device via an audio input jack on the electronic device. The electronic device can also include decoding features for decoding communications sent by the audio adjustment system. 1. A system for decoding audio enhancement settings with an audio device , the system comprising: wherein the detector causes the audio signal to be decoded in response to detecting the trigger signal in the audio signal, and', 'wherein the detector passes the audio signal to an audio enhancement for audio processing in response to not detecting the trigger signal in the audio signal;, 'a detector implemented in an audio device comprising one or more processors, the detector configured to receive an audio signal and to analyze the audio signal to determine whether a trigger signal is present in the audio signal,'}a decoder configured to, in response to the trigger signal being detected by the detector, decode an instruction in the audio signal; anda configuration module configured to implement the instruction to thereby adjust a characteristic of the audio enhancement.2. The system of claim 1 , wherein the detector is configured to receive the audio signal from one or more of the following: an audio input port in the audio device and a microphone in the audio device.3. The system of claim 1 , wherein the ...

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Номер записи: 13
18-04-2013 дата публикации

AUDIO CODING DEVICE AND AUDIO CODING METHOD, AUDIO DECODING DEVICE AND AUDIO DECODING METHOD, AND PROGRAM

Номер: US20130096927A1
Автор: Maeda Yuuji, MATSUMOTO Jun, MATSUMURA Yuuki, SUZUKI Shiro, Toguri Yasuhiro
Принадлежит:

There is provided an audio coding device including a first windowing part that multiplies an audio signal by a first window function, a second windowing part that multiplies the audio signal by a second window function having a characteristic different from a characteristic of the first window function, a window selecting part that selects the first window function or the second window function as an optimum window function based on the audio signal multiplied by the first windowing part and the audio signal multiplied by the second windowing part, a coding part that codes a frequency spectrum of the audio signal multiplied by the optimum window function, and a transmitting part that transmits the frequency spectrum coded by the coding part and window function information representing the optimum window function. 1. An audio coding device , comprising:a first windowing part that multiplies an audio signal by a first window function;a second windowing part that multiplies the audio signal by a second window function having a characteristic different from a characteristic of the first window function;a window selecting part that selects the first window function or the second window function as an optimum window function based on the audio signal multiplied by the first windowing part and the audio signal multiplied by the second windowing part;a coding part that codes a frequency spectrum of the audio signal multiplied by the optimum window function; anda transmitting part that transmits the frequency spectrum coded by the coding part and window function information representing the optimum window function.2. The audio coding device according to claim 1 , further comprising:a first normalization coefficient determining part that determines a normalization coefficient of a frequency spectrum of the audio signal multiplied by the first windowing part as a first normalization coefficient;a second normalization coefficient determining part that determines a normalization ...

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Номер записи: 14
18-04-2013 дата публикации

Multi-Resolution Switched Audio Encoding/Decoding Scheme

Номер: US20130096930A1
Автор: BAYER Stefan, Fuchs Guillaume, GEIGER Ralf, Gournay Philippe, Grill Bernhard, Lecomte Jeremie, Multrus Markus, NAGEL Frederik, Neuendorf Max, Rettelbach Nikolaus, ROBILLIARD Julien, Salami Redwan
Принадлежит:

An audio encoder for encoding an audio signal has a first coding branch, the first coding branch comprising a first converter for converting a signal from a time domain into a frequency domain. Furthermore, the audio encoder has a second coding branch comprising a second time/frequency converter. Additionally, a signal analyzer for analyzing the audio signal is provided. The signal analyzer, on the hand, determines whether an audio portion is effective in the encoder output signal as a first encoded signal from the first encoding branch or as a second encoded signal from a second encoding branch. On the other hand, the signal analyzer determines a time/frequency resolution to be applied by the converters when generating the encoded signals. An output interface includes, in addition to the first encoded signal and the second encoded signal, a resolution information identifying the resolution used by the first time/frequency converter and used by the second time/frequency converter. 1. Audio encoder for encoding an audio signal , comprising:a first coding branch for encoding an audio signal using a first coding algorithm to acquire a first encoded signal, the first coding branch comprising the first converter for converting an input signal into a spectral domain;a second coding branch for encoding an audio signal using a second coding algorithm to acquire a second encoded signal, wherein the first coding algorithm is different from the second coding algorithm, the second coding branch comprising a domain converter for converting an input signal from an input domain into an output domain, and a second converter for converting an input signal into a spectral domain;a switch for switching between the first coding branch and the second coding branch so that, for a portion of the audio input signal, either the first encoded signal or the second encoded signal is in an encoder output signal;a signal analyzer for analyzing the portion of the audio signal to determine, ...

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Номер записи: 15
25-04-2013 дата публикации

Multi-point sound mixing and distant view presentation method, apparatus and system

Номер: US20130103393A1
Автор: Sun Bo, Wu Mingliang
Принадлежит: ZTE CORPORATION

The disclosure provides a multi-point sound mixing and distant view presentation method, apparatus and system, wherein the multi-point sound mixing and distant view presentation method includes: receiving audio code streams from a plurality of meeting places, wherein each meeting place comprises one or more meeting sections, and each meeting section corresponds to one audio code stream; mixing the audio code streams of the meeting sections which have a corresponding relationship among the plurality of meeting places; and outputting mixed audio code streams to the meeting sections which have the corresponding relationship among the plurality of meeting places. Sounds in different sections of the distant view presentation conference system can be distinguished by technical solutions provided by the disclosure. 1. A multi-point sound mixing and distant view presentation method , comprising:receiving audio code streams from a plurality of meeting places, wherein each meeting place comprises one or more meeting sections, and each meeting section corresponds to one of the audio code streams;mixing the audio code streams of the meeting sections which have a corresponding relationship among the plurality of meeting places; andoutputting mixed audio code streams to the meeting sections which have the corresponding relationship among the plurality of meeting places.2. The method according to claim 1 , wherein each of the meeting sections respectively corresponds to different positions claim 1 , and the step of mixing the audio code streams of the meeting sections which have the corresponding relationship among the plurality of meeting places comprises:mixing the audio code streams of the meeting sections with a same position in each meeting place;the step of outputting the mixed audio code streams to the meeting sections which have the corresponding relationship among the plurality of meeting places comprises:outputting the mixed audio code streams to the meeting sections ...

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Номер записи: 16
25-04-2013 дата публикации

METHOD AND APPARATUS FOR PROCESSING AN AUDIO SIGNAL

Номер: US20130103407A1
Автор: Jeon Hye Jeong, Jeong Gyu Hyeok, Lee Byung Suk, LEE Chang Heon
Принадлежит: LG ELECTRONICS INC.

The present invention relates to a method for processing an audio signal, comprising the following steps: performing a linear predictive analysis on the current frame of an audio signal so as to generate a first target vector, which is a target vector of a first stage, on the basis of a plurality of linear prediction transform coefficients; performing vector quantization on the first target vector so as to acquire a predetermined number of first temporary candidate code vectors of the first stage; calculating first temporary candidate errors, which are errors between the first temporary candidate code vectors and the first target vector; and determining a first number, which is the number of the first candidate code vectors, on the basis of the first temporary candidate errors, and acquiring first final candidate code vectors in the same amount as the first number. 1. An audio signal processing method comprising:generating a first target vector which is a target vector of a first stage based on a plurality of linear predictive conversion coefficients by performing linear predictive analysis on a current frame of an audio signal;acquiring a temporarily determined number of first temporary candidate code vectors of the first stage by vector-quantizing the first target vector;calculating first temporary candidate errors which are errors between the first temporary candidate code vectors and the first target vector; anddetermining a first number which is the number of first candidate code vectors based on the first temporary candidate errors and acquiring the same number of first final candidate code vectors as the first number.2. The audio signal processing method according to claim 1 , further comprising:generating first final candidate errors as target vectors of a second stage based on the first final candidate code vectors;acquiring a temporarily determined number of second temporary candidate code vectors of the second stage by vector-quantizing the second target ...

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Номер записи: 17
09-05-2013 дата публикации

Method and apparatus for estimating interchannel delay of sound signal

Номер: US20130114817A1
Автор: LEI Miao, Wenhai Wu, Yue Lang, Zexin LIU
Принадлежит: Huawei Technologies Co Ltd

A method and an apparatus for estimating an interchannel delay of a sound signal are disclosed, related to the communication field and capable of realizing a stable sound field in a crosstalk. The method includes: calculating an error between an actual interchannel phase difference and a predicted interchannel phase difference of a sound signal, where the predicted interchannel phase difference is predicted according to a predetermined interchannel delay of the sound signal; determining whether the sound signal is a sound signal in a crosstalk according to the error; and if the sound signal is a sound signal in the crosstalk, setting an interchannel delay corresponding to the sound signal to a fixed value

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Номер записи: 18
09-05-2013 дата публикации

SIGNAL CLASSIFICATION METHOD AND DEVICE, AND ENCODING AND DECODING METHODS AND DEVICES

Номер: US20130117029A1
Автор: Liu Zexin, Miao Lei, Taleb Anisse
Принадлежит: Huawei Technologies Co., Ltd.

Embodiments of the present invention provide a signal classification method and device, and encoding and decoding methods and devices. The encoding method includes: dividing a current frame into a low-frequency band signal and a high-frequency band signal; attenuating the high-frequency band signal or a to-be-encoded characteristic parameter of the high-frequency band signal according to an energy attenuation value of the low-frequency band signal, where the energy attenuation value indicates energy attenuation of the low-frequency band signal caused by encoding of the low-frequency band signal; and encoding the attenuated high-frequency band signal or the attenuated to-be-encoded characteristic parameter of the high-frequency band signal. The technical solutions according to the embodiments of the present invention can improve the effect of combining the low-frequency band signal and the high-frequency band signal at the decoder. 1. An encoding method , comprising:dividing a current frame into a low-frequency band signal and a high-frequency band signal;attenuating a one of the group consisting of the high-frequency band signal and a to-be-encoded characteristic parameter of the high-frequency band signal, the attenuating being according to an energy attenuation value of the low-frequency band signal, and wherein the energy attenuation value indicates energy attenuation of the low-frequency band signal caused by encoding of the low-frequency band signal; andencoding the one of the group consisting of the attenuated high-frequency band signal and the attenuated to-be-encoded characteristic parameter of the high-frequency band signal.2. The method according to claim 1 , further comprising:determining a signal class of the high-frequency band signal; andwherein the attenuating the one of the group consisting of the high-frequency band signal and the to-be-encoded characteristic parameter of the high-frequency band signal according to the energy attenuation value of ...

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Номер записи: 19
16-05-2013 дата публикации

DECODING DEVICE, ENCODING DEVICE, AND METHODS FOR SAME

Номер: US20130124201A1
Автор: Oshikiri Masahiro, Yamanashi Tomofumi
Принадлежит: Panasonic Corporation

Disclosed is a decoding device which can efficiently encode/decode spectral data in a high pass section of a broadband signal. In the disclosed device: a sample group extraction unit () partially selects spectral components by means of an ease of selection importance which is the extent that the spectral components come close to the spectral component having the maximum amplitude value, in the spectrum of a high pass estimated by means of first parameters contained in second encoded information and bands most approximated to each of the spectrums of a plurality of sub-bands calculated from the spectrum of a second decode signal; a logarithmic gain application unit () applies second parameters to the partially selected spectral components; and an interpolation processing unit () applies third parameters which are adaptively set according to the value of the second parameters, to the spectral components which were not partially selected. 1. A decoding apparatus comprising:a receiving section that receives first encoded information indicating a low-frequency portion no greater than a predetermined frequency of a speech signal or an audio signal, and second encoded information, the second information containing band information for estimating a spectrum of a high-frequency portion of the speech signal or the audio signal in a plurality of subbands obtained by dividing the high-frequency portion higher than the predetermined frequency, and a first amplitude adjusting parameter that adjusts an amplitude corresponding to a part or all of spectral components in each subband;a first decoding section that decodes the first encoded information to generate a first decoded signal; anda second decoding section that estimates the high-frequency portion of the speech signal or the audio signal from the first decoded signal using the second encoded information and adjusts the amplitude of the spectral component to thereby generate a second decoded signal, wherein a spectral ...

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Номер записи: 20
23-05-2013 дата публикации

Audio Signal Synthesizer

Номер: US20130129096A1
Автор: Faller Christof, Lang Yue, Virette David, Xu Jianfeng
Принадлежит: Huawei Technologies Co., Ltd.

The invention relates to an audio signal synthesizer, the audio signal synthesizer comprises a transformer for transforming the down-mix audio signal into frequency domain to obtain a transformed audio signal; a signal generator for generating a first auxiliary signal, for generating a second auxiliary signal, and for generating a third auxiliary signal upon the basis of the transformed audio signal; a de-correlator for generating a first de-correlated signal, and for generating a second de-correlated signal from the third auxiliary signal, the first de-correlated signal and the second de-correlated signal being at least partly de-correlated; and a combiner for combining the first auxiliary signal with the first de-correlated signal to obtain a first audio signal, and for combining the second auxiliary signal with the second de-correlated signal to obtain the second audio signal, the first audio signal and the second audio signal forming the multi-channel audio signal. 1. Audio signal synthesizer for synthesizing a multi-channel audio signal from a down-mix audio signal , the audio signal synthesizer comprising:a transformer configured to transform the down-mix audio signal into frequency domain to obtain a transformed audio signal, wherein the transformed audio signal represents a spectrum of the down-mix audio signal;a signal generator configured to generate a first auxiliary signal, a second auxiliary signal, and a third auxiliary signal upon the basis of the transformed audio signal;a de-correlator configured to generate a first de-correlated signal and a second de-correlated signal from the third auxiliary signal, wherein the first de-correlated signal and the second de-correlated signal are at least partly de-correlated; anda combiner configured to combine the first auxiliary signal with the first de-correlated signal to obtain a first audio signal, and for combining the second auxiliary signal with the second de-correlated signal to obtain the second audio ...

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Номер записи: 21
23-05-2013 дата публикации

APPARATUS AND METHOD FOR CODING AND DECODING MULTI-OBJECT AUDIO SIGNAL WITH VARIOUS CHANNEL INCLUDING INFORMATION BITSTREAM CONVERSION

Номер: US20130132098A1
Автор: Beack Seung-Kwon, HONG Jin-Woo, JANG Dae-Young, KANG Kyeong-Ok, KIM Jin-Woong, LEE Tae-Jin, LEE Yong-Ju, SEO Jeong-Il
Принадлежит: Electronics and Telecommunications Research Institute

Provided is an apparatus and method for coding and decoding multi-object audio signals with various channels and providing backward compatibility with a conventional spatial audio coding (SAC) bitstream. The apparatus includes: an audio object coding unit for coding audio-object signals inputted to the coding apparatus based on a spatial cue and creating rendering information for the coded audio-object signals, where the rendering information provides a coding apparatus including spatial cue information for audio-object signals; channel information of the audio-object signals; and identification information of the audio-object signals, and used in coding and decoding of the audio signals. 1. An apparatus for coding multi-object audio signals , including:an audio channel coding means for transforming input multi-channel audio signals into audio-object signals and creating rendering information for the multi-channel audio signals; andan audio object coding means for coding the audio-object signals output from the audio channel coding means and input audio-object signals based on spatial cues and creating rendering information for the coded audio-object signals.2. The apparatus of claim 1 , further including:an bitstream generating means for creating an bitstream including the rendering information respectively output from the audio channel coding means and the audio object coding means.3. The apparatus of claim 1 , wherein the audio channel coding means is a Moving Picture Experts Group (MPEG) surround coder. The present invention relates to an apparatus and a method for coding and decoding multi-object audio signals with various channels; and, more particularly, to an apparatus and method for coding and decoding multi-object audio signals with various channels including side information bitstream conversion for transforming side information bitstream and recovering multi-object audio signals with a desired output signal, i.e., various channels, based on transformed ...

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Номер записи: 22
23-05-2013 дата публикации

CODING DEVICE, DECODING DEVICE, AND METHODS THEREOF

Номер: US20130132099A1
Автор: EHARA Hiroyuki, Hori Takako, Oshikiri Masahiro
Принадлежит: Panasonic Corporation

Provided are a coding device, a decoding device, and methods thereof, with which it is possible to implement high sound quality coding and decoding in layered coding (scalable coding or embedded coding) wherein each layer comprises a plurality of bit rates (multi-rate). In the coding device (), a feature analysis unit () extracts feature values of an input signal. Then a bit rate determination unit () determines, on the basis of the feature values of the input signal, a combination of a coding rate (low region coding rate) of a low region signal coding unit () which carries out coding of a low region part of the input signal and a coding rate (high region coding rate) of a high region signal coding unit () which carries out coding of a high region part of the input signal. 1. An encoding apparatus comprising:an analyzing section that analyzes an input signal feature for each of a low-region part and a high-region part of the input signal and that generates feature data that indicates the analysis results;a determining section that, based on a pre-set total encoding rate that is the total of a low-region encoding rate and a high-region encoding rate, and on the feature data, determines a combination of the low-region encoding rate and the high-region encoding rate;a low-region encoding section that encodes the low-region part of the input signal using the determined low-region encoding rate and generates low-region encoded data;a high-region encoding section that encodes the high-region part of the input signal using the determined high-region encoding rate and generates high-region encoded data; anda multiplexing section that multiplexes the low-region encoded data, the high-region encoded data, and the feature data.2. The encoding apparatus according to claim 1 , wherein:the analyzing section takes the results of a comparison between a threshold value and the difference between the energy of the low-region part and the energy of the high-region part as the feature ...

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Номер записи: 23
30-05-2013 дата публикации

SPEECH SIGNAL TRANSMISSION AND RECEPTION APPARATUSES AND SPEECH SIGNAL TRANSMISSION AND RECEPTION METHODS

Номер: US20130138431A1
Автор: CHOI Byung Kwon, KIM Dong Soo, KWON Young Do, Roh Kyung Shik
Принадлежит: SAMSUNG ELECTRONICS CO., LTD.

A speech signal transmission apparatus includes an extractor to extract speech signals from speech source signals collected by a plurality of microphones, a power calculator to calculate powers of speech signals of multiple channels and set any one of the speech signals of the multiple channels as a reference speech signal, a synchronization adjustor to adjust synchronization of the other speech signals based on the reference speech signal, a signal generator to generate extraction signals by offsetting the reference speech signal from the other synchronization-adjusted speech signals, an encryptor to compress and encrypt the reference speech signal and the extraction signals, and a transmitter to transmit the compressed and encrypted reference speech signal and extraction signals. 1. A speech signal transmission apparatus comprising:an extractor to extract a plurality of speech signals from a plurality of speech source signals collected by a plurality of microphones;a power calculator to calculate a plurality of power parameters of the plurality of speech signals and set any one speech signal among the plurality of speech signals of the multiple channels as a reference speech signal;a synchronization adjustor to calculate a plurality of synchronization parameters of the speech signals among the plurality of speech signals excluding the reference speech signal and to adjust synchronization of the speech signals among the plurality of speech signals excluding the reference speech signal based on the reference speech signal;a signal generator to generate a plurality of extraction signals by offsetting the reference speech signal from each synchronization-adjusted speech signal;an encryptor to compress and encrypt the reference speech signal and the plurality of extraction signals; anda transmitter to transmit the compressed and encrypted reference speech signal and the compressed and encrypted plurality of extraction signals.2. The speech signal transmission apparatus ...

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Номер записи: 24
06-06-2013 дата публикации

AUDIO SIGNAL PROCESSING APPARATUS AND AUDIO SIGNAL PROCESSING METHOD

Номер: US20130144631A1
Автор: AKAMATSU Sin, MIYASAKA Shuji, SHINZAKI Satoshi, Yamada Shuhei
Принадлежит: Panasonic Corporation

An audio signal processing apparatus that processes a bit stream generated by coding an audio signal on a frame-by-frame basis, the bit stream including, for each frame, coded data representing the audio signal, additional data and attribute information, the audio signal processing apparatus including a decoding unit configured to decode the coded data to generate a decoded signal, a processing unit configured to process the decoded signal, a detection unit configured to detect whether or not there has been a change in the attribute information, and a storage unit, wherein the processing unit is configured to, when the change is not detected, process the decoded signal by using at least two pieces of additional data stored, and when the change is detected, process the decoded signal by using only either additional data before detection of the change or additional data after detection of the change. 1. An audio signal processing apparatus that processes a bit stream generated by coding an audio signal on a frame-by-frame basis ,the bit stream including, for each frame,coded data representing a coded audio signal,additional data on an amplitude of a decoded signal generated by decoding the coded data, andattribute information indicating a property of the coded data,the audio signal processing apparatus comprising:a decoding unit configured to decode coded data of a target frame to generate the decoded signal;a processing unit configured to process the decoded signal generated by the decoding unit;a detection unit configured to detect whether or not there has been a change in the attribute information between the target frame and an adjacent frame that is consecutive to the target frame; anda storage unit configured to store at least two pieces of additional data including additional data of the target frame,wherein the processing unit is configured to:when the change is not detected by the detection unit, process the decoded signal of the target frame by using the at ...

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Номер записи: 25
13-06-2013 дата публикации

RESAMPLING OUTPUT SIGNALS OF QMF BASED AUDIO CODECS

Номер: US20130151262A1
Автор: CHALUPKA Tobias, GEIGER Ralf, HILDENBRAND Matthias, JANDER Manuel, LOHWASSER Markus, Neuendorf Max, SCHNELL Markus
Принадлежит: Fraunhofer-Gesellschaft zur Foerderung der angewandten Forschung e.V.

An apparatus for processing an audio signal includes a configurable first audio signal processor for processing the audio signal in accordance with different configuration settings to obtain a processed audio signal, wherein the apparatus is adapted so that different configuration settings result in different sampling rates of the processed audio signal. The apparatus furthermore includes an analysis filter bank having a first number of analysis filter bank channels, a synthesis filter bank having a second number of synthesis filter bank channels, a second audio processor being adapted to receive and process an audio signal having a predetermined sampling rate, and a controller for controlling the first number of analysis filter bank channels or the second number of synthesis filter bank channels in accordance with a configuration setting. 1. An apparatus for processing an audio signal , comprising:a configurable first audio signal processor for processing the audio signal in accordance with different configuration settings to acquire a processed audio signal, wherein the apparatus is adapted so that different configuration settings result in different sampling rates of the processed audio signal,an analysis filter bank comprising a first number of analysis filter bank channels,a synthesis filter bank comprising a second number of synthesis filter bank channels,a second audio processor being adapted to receive and process an audio signal comprising a predetermined sampling rate, anda controller for controlling the first number of analysis filter bank channels and the second number of synthesis filter bank channels in accordance with a configuration setting provided to the configurable first audio signal processor, so that an audio signal output of the synthesis filter bank comprises the predetermined sampling rate or a sampling rate being different from the predetermined sampling rate and being closer to the predetermined sampling rate than a sampling rate of an ...

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Номер записи: 26
13-06-2013 дата публикации

METHOD AND DEVICE FOR PROCESSING AUDIO SIGNALS

Номер: US20130151263A1
Автор: Jeong Gyuhyeok, Jeong Hyejeong, Kang Ingyu, Kim Lagyoung, Lee Byungsuk, Lee Changheon
Принадлежит: LG ELECTRONICS INC.

The present invention provides a method for processing audio signals, and the method comprises the steps of: receiving input audio signals corresponding to a plurality of spectral coefficients; obtaining location information that indicates a location of a particular spectral coefficient among said spectral coefficients, on the basis of energy of said input signals: generating a shape vector by using said location information and said spectral coefficients; determining a codebook index by searching for a codebook corresponding to said shape vector; and transmitting said codebook index and said location information, wherein said shape vector is generated by using a part which is selected from said spectral coefficients, and said selected part is selected on the basis of said location information. 1. A method of processing an audio signal , comprising:receiving an input audio signal corresponding to a plurality of spectral coefficients;obtaining location information indicating a location of a specific one of a plurality of the spectral coefficients based on an energy of the input signal;generating a shape vector using the location information and the spectral coefficients;determining a codebook index by searching a codebook corresponding to the shape vector; andtransmitting the codebook index and the location information,wherein the shape vector is generated using a part selected from the spectral coefficients, andwherein the selected part is selected based on the location information.2. The method of claim 1 , further comprising:generating sign information on the specific spectral coefficient; andtransmitting the sign information,wherein the shape vector is generated further based on the sign information.3. The method of claim 1 , further comprising:generating a normalized value for the selected part,wherein the determining comprises generating a normalized shape vector by normalizing the shape vector using the normalized value and determining the codebook index by ...

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Номер записи: 27
04-07-2013 дата публикации

APPARATUS AND METHOD FOR GENERATING PANORAMIC SOUND

Номер: US20130170649A1
Автор: Lee Kang Eun
Принадлежит: SAMSUNG ELECTRONICS CO., LTD.

An apparatus and method for generating a panoramic sound are provided. The panoramic sound generation apparatus may include a panning coefficient calculation unit to calculate a panning coefficient that represents directivity of a sound source using an input signal, a masker determination unit to determine a direction masker that extracts a sound source of a desired direction based on the panning coefficient, and a channel separation unit to separate to be used as an output signal, output to a sound output device, using the direction masker. 1. A panoramic sound generation apparatus comprising:a panning coefficient calculation unit, using at least one processor, to calculate a panning coefficient that represents directivity of a sound source using an input signal;a masker determination unit to determine a direction masker that extracts a sound source of a desired direction based on the panning coefficient; anda channel separation unit to separate the input signal to be used as an output signal, and to output the output signal to a sound output device, using the direction masker.2. The panoramic sound generation apparatus of claim 1 , wherein the panning coefficient calculation unit calculates the panning coefficient using a frequency component based on the input signal.3. The panoramic sound generation apparatus of claim 1 , wherein the masker determination unit comprises:a control coefficient determiner to determine a control coefficient for windowing the panning coefficient using configuration information of the sound output device; anda window processor to process a panning coefficient window based on the sound output device using the control coefficient.4. The panoramic sound generation apparatus of claim 3 , wherein the control coefficient determiner determines the control coefficient using a first angle based on a first sound output device corresponding to a left boundary and a second sound output device corresponding to a right boundary claim 3 , a second ...

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Номер записи: 28
04-07-2013 дата публикации

Front wave field synthesis (wfs) system and method for providing surround sound using 7.1 channel codec

Номер: US20130170652A1
Автор: Jae Hyoun Yoo, Jeong Il Seo, Keun Woo Choi, Kyeong Ok Kang, Tae Jin Lee
Принадлежит: Electronics and Telecommunications Research Institute ETRI

Provided is a system and method for front wave field synthesis (WFS) to provide a surround sound with a reduced number of loudspeakers. An apparatus for encoding a front WFS signal in the front WFS system may include a content receiving unit to receive WFS content including the first channels corresponding to a front array speaker and the second channels corresponding to a rear surround sound speaker, and an encoding unit to encode the first channels and the second channels using a multichannel encoder corresponding to the number of the channels included in the WFS content.

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Номер записи: 29
04-07-2013 дата публикации

APPARATUS FOR DECODING A SIGNAL COMPRISING TRANSIENTS USING A COMBINING UNIT AND A MIXER

Номер: US20130173273A1
Автор: Disch Sascha, Herre Juergen, HILPERT Johannes, KUECH Fabian, KUNTZ Achim
Принадлежит: Fraunhofer-Gesellschaft zur Foerderung der angewandten Forschung e.V.

An apparatus for generating a decorrelated signal including a transient separator, a transient decorrelator, a second decorrelator, a combining unit and a mixer, wherein the transient separator is adapted to separate an input signal into a first signal component and into a second signal component such that the first signal component includes transient signal portions of the input signal and such that the second signal component includes non-transient signal portions of the input signal. The combining unit and the mixer are arranged so that a decorrelated signal from a combination unit is fed into the mixer as an input signal. 1. An apparatus for decoding a signal comprising:a transient separator for separating an apparatus input signal into a first signal component and into a second signal component such that the first signal component comprises transient signal portions of the input signal and such that the second signal component comprises non-transient signal portions of the input signal;a transient decorrelator for decorrelating the first signal component according to a first decorrelation method to acquire a first decorrelated signal component;a further second decorrelator for decorrelating the second signal component according to a second decorrelation method to obtain a second decorrelated signal component, wherein the second decorrelation method is different from the first decorrelation method;a combining unit for combining the first decorrelated signal component and the second decorrelated signal component to acquire a decorrelated combination signal; anda mixer, being adapted to receive mixer input signals and being adapted to generate output signals based on the mixer input signals and a mixing rule;wherein the combining unit and the mixer are arranged so that the decorrelated combination signal is fed into the mixer as a first mixer input signal and that the apparatus input signal or a signal derived from the apparatus input signal is fed into the mixer ...

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Номер записи: 30
18-07-2013 дата публикации

AUDIO CODING DEVICE AND METHOD

Номер: US20130182854A1
Автор: Kishi Yohei, Shirakawa Miyuki, SUZUKI Masanao, Takeuchi Shunsuke
Принадлежит: FUJITSU LIMITED

An audio coding device that uses a first-channel signal, a second-channel signal, and a plurality of channel prediction coefficients included in a code book, according to which predictive coding is performed on a third-channel signal, the first-channel signal, the second-channel signal, and the third-channel signal being included in a plurality of channels of an audio signal, the device includes, a processor; and a memory which stores a plurality of instructions, which when executed by the processor, cause the processor to execute, determining a distribution of error defined by a difference between the third-channel signal before predictive coding and the third-channel signal after predictive coding as a given curved surface according to the first-channel signal, the second-channel signal, and the third-channel signal before predictive coding; and calculating channel prediction coefficients, included in the code book. 1. An audio coding device that uses a first-channel signal , a second-channel signal , and a plurality of channel prediction coefficients included in a code book , according to which predictive coding is performed on a third-channel signal , the first-channel signal , the second-channel signal , and the third-channel signal being included in a plurality of channels of an audio signal , the device comprising:a processor; anda memory which stores a plurality of instructions, which when executed by the processor, cause the processor to execute,determining a distribution of error defined by a difference between the third-channel signal before predictive coding and the third-channel signal after predictive coding as a given curved surface according to the first-channel signal, the second-channel signal, and the third-channel signal before predictive coding; andcalculating channel prediction coefficients, included in the code book, that correspond to the first channel and the second channel from the code book, according to a minimum value of the error, the ...

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Номер записи: 31
18-07-2013 дата публикации

AUDIO ENCODING APPARATUS

Номер: US20130185083A1
Автор: MANO Ryuji
Принадлежит: RENESAS ELECTRONICS CORPORATION

There is provided an audio encoding apparatus that can avoid that audio data becomes irreproducible after fast-forward play. A quantization unit quantizes and buffers audio data into a buffer unit. A stream generating unit puts buffered audio data in a frame where there is a header related to the audio data in a stream and/or in one or plural frames preceding that frame. As for a predetermined frame, the stream generating unit puts in a data field of the frame the whole of an audio data piece related to a header included in that frame and puts audio sample data following that audio sample in a remaining part of the data field. As for a frame not a predetermined one, it puts in a data field of the frame an audio data piece related to a header included in that frame and/or audio data pieces following that audio data piece. 1. An audio encoding apparatus comprising:a quantization unit that quantizes audio data;a buffer unit that buffers the quantized audio data; anda stream generating unit that puts quantized audio data from said buffer unit in a frame where there is a header related to the audio data in a stream and/or in one or plural frames preceding the frame where there is the header,wherein, as for a predetermined frame, said stream generating unit puts in a data field of the frame the whole of an audio data piece related to a header included in that frame and puts audio data pieces following that audio data piece in a remaining part of the data field of the frame and, as for a frame other than the predetermined frame, said stream generating unit puts in a data field of the frame an audio data piece related to a header included in that frame and/or audio data pieces following that audio data piece.2. The audio encoding apparatus according to claim 1 , wherein said predetermined frame exists cyclically as the first one of a given number of successive frames.3. The audio encoding apparatus according to claim 1 , wherein said stream is a stream of MPEG Audio Layer 3 ...

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Номер записи: 32
18-07-2013 дата публикации

SYSTEMS, METHODS, APPARATUS, AND COMPUTER-READABLE MEDIA FOR BIT ALLOCATION FOR REDUNDANT TRANSMISSION

Номер: US20130185084A1
Автор: Krishnan Venkatesh, Rajendran Vivek, Sinder Daniel J.
Принадлежит: QUALCOMM INCORPORATED

Compressibility-based reallocation of initial bit allocations for frames of an audio signal is described. Applications to redundancy-based retransmission of critical frames (e.g., for fixed-bit-rate modes of speech codec operation) are also described. 1. A method of processing an audio signal , said method comprising:calculating at least one value of a decision metric for a second frame of the audio signal that is subsequent in the audio signal to a first frame of the audio signal; andbased on said at least one calculated value of the decision metric, selecting one among a plurality of reallocation candidates,wherein said calculated at least one value is based on a measure of compressibility of the second frame, andwherein said selected reallocation candidate indicates a reallocation of an initial bit allocation for the second frame into a first portion and a second portion.2. The method according to claim 1 , wherein said method includes determining that said first frame is a critical frame of the audio signal.3. The method according to claim 2 , wherein said determining that said first frame is a critical frame is based on information from an encoded version of a frame of the audio signal that is subsequent in the audio signal to the first frame.4. The method according to claim 3 , wherein said encoded version is an encoded version of the second frame.5. The method according to claim 2 , wherein said determining includes comparing a criticality measure to a criticality threshold.6. The method according to claim 5 , wherein said determining includes calculating the criticality threshold based on information relating to a state of a transmission channel.7. The method according to claim 6 , wherein said calculating the criticality threshold includes:comparing a calculated value that is based on the information relating to the state of the transmission channel to a boundary value; andin response to a result of said comparing to the boundary value, selecting the ...

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Номер записи: 33
25-07-2013 дата публикации

METHOD AND APPARATUS FOR DECODING AN AUDIO SIGNAL USING A SHAPING FUNCTION

Номер: US20130191134A1
Автор: LEE Mi-Suk
Принадлежит:

The present invention relates to a method and apparatus for decoding an audio signal using a shaping function. According to one embodiment of the present invention, the method for decoding an audio signal comprises the following steps: taking frame data of the audio signal as an input; restoring a fixed codebook of the frame data using a random function; calculating a shaping function using an adaptive codebook of the frame data; shaping the restored fixed codebook using the shaping function; and synthesizing the audio signal from the frame data using the shaped fixed codebook and adaptive codebook. According to the present invention, the fixed codebook may be restored using the shaping function calculated on the basis of the adaptive codebook upon the occurrence of frame data loss, thus emphasizing a pitch period and reducing the influence of the fixed codebook between the pitch periods so as to reduce the degradation in the quality of the synthesized signal. 1. A method for decoding an audio signal , comprising:receiving frame data of the audio signal;recovering a fixed codebook of the frame data using a random function;calculating a shaping function using an adaptive codebook of the frame data;shaping the recovered fixed codebook using the shaping function; andsynthesizing the audio signal from the frame data by using the shaped fixed codebook and the adaptive codebook.2. The method of claim 1 , wherein the recovering of the fixed codebook includes recovering the fixed codebook in a subframe unit of the frame data.3. The method of claim 1 , wherein the calculating of the shaping function includes:acquiring a maximum value of the adaptive codebook of the subframe of the frame data;normalizing the adaptive codebook of the subframe using the maximum value; andcalculating the shaping function using the normalized adaptive codebook.4. The method of claim 1 , wherein the calculating of the shaping function includes:comparing a function value acquired through the ...

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Номер записи: 34
08-08-2013 дата публикации

REPRODUCTION APPARATUS

Номер: US20130202024A1
Автор: SUZUKI Hiroaki
Принадлежит: Panasonic Corporation

A reproduction apparatus includes a setter configured to set at least one language, a separator configured to separate and output a video signal and an audio signal, a video signal processor configured to decode and encode the video signal, a first decoder configured to output a first center channel signal and a multi-channel signal generated by extraction of the first center channel signal, a second decoder configured to output a second center channel signal, and a selector configured to receive the first and second center channel signals and assign, according to a setting in the setter, the first and second center channel signals respectively to different outputs. 1. A reproduction apparatus for performing multi-channel surround sound reproduction of contents including a video signal and audio signals corresponding respectively to a plurality of languages , the reproduction apparatus comprising:a setter configured to set at least one of the plurality of languages;a separator configured to separate the contents into the video signal and the audio signals corresponding respectively to the plurality of languages set in the setter to output the video signal and the audio signals;a video signal processor configured to decode and encode the video signal output from the separator;a first decoder configured to decode, after receipt of one of the audio signals corresponding respectively to the plurality of languages set in the setter from the separator, the one of the audio signals to generate a multi-channel surround audio signal including a first center channel signal, to extract the first center channel signal from the multi-channel surround audio signal, and to output the first center channel signal and a multi-channel signal generated by extraction of the first center channel signal;a second decoder configured to decode, after receipt of other one of the audio signals different from the one of the audio signals decoded by the first decoder from the separator, the ...

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Номер записи: 35
15-08-2013 дата публикации

QUANTIZATION MATRICES FOR DIGITAL AUDIO

Номер: US20130208901A1
Автор: Chen Wei-ge, Lee Ming-Chieh, Thumpudi Naveen
Принадлежит: MICROSOFT CORPORATION

Quantization matrices facilitate digital audio encoding and decoding. An audio encoder generates and compresses quantization matrices; an audio decoder decompresses and applies the quantization matrices. The invention includes several techniques and tools, which can be used in combination or separately. For example, the audio encoder can generate quantization matrices from critical band patterns for blocks of audio data. The encoder can compute the quantization matrices directly from the critical band patterns, which can be computed from the same audio data that is being compressed. The audio encoder/decoder can use different modes for generating/applying quantization matrices depending on the coding channel mode of multi-channel audio data. The audio encoder/decoder can use different compression/decompression modes for the quantization matrices, including a parametric compression/decompression mode. 144-. (canceled)45. A method for decoding audio in a computing device that implements an audio decoder , the method comprising:with the computing device that implements the audio decoder, decompressing at least one set of weighting factors according to a parametric model to switch between a direct representation and a parametric representation of the at least one set of weighting factors, wherein the parametric representation of the at least one set of weighting factors accounts for audibility of distortion according to a model of human auditory perception; andwith the computing device that implements the audio decoder, outputting a result of the processing, wherein the result is the direct representation.4647-. (canceled)48. The method of wherein the parametric model uses linear predictive coding for the at least one set of weighting factors.49. The method of wherein the at least one set of weighting factors is for a block of audio data claim 48 , and wherein pseudo-autocorrelation values used in the decompressing differ from autocorrelation values for the block due at ...

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Номер записи: 36
15-08-2013 дата публикации

ENCODING DEVICE AND METHOD, DECODING DEVICE AND METHOD, AND PROGRAM

Номер: US20130208902A1
Автор: Chinen Toru, Yamamoto Yuki
Принадлежит: SONY CORPORATION

The present invention relates to an encoding device and method, and a decoding device and method, and a program which enable music signals to be played with higher sound quality by expanding a frequency band. 1. An encoding device comprising:subband diving means configured to divide an input signal into a plurality of subbands, and to generate a low-frequency subband signal made up of a plurality of subbands on the low-frequency side, and a high-frequency subband signal made up of a plurality of subbands on the high-frequency side;feature amount calculating means configured to calculate feature amount that represents features of the input signal based on at least any one of the low-frequency subband signal and the input signal;smoothing means configured to subject the feature amount smoothing;pseudo high-frequency subband power calculating means configured to calculate pseudo high-frequency subband power that is an estimated value of power of the high-frequency subband signal based on the smoothed feature amount and a predetermined coefficient;selecting means configured to calculate high-frequency subband power that is power of the high-frequency subband signal from the high-frequency subband signal, and to compare the high-frequency subband power and the pseudo high-frequency subband rower to select any of a plurality of the coefficients;high-frequency encoding means configured to encode coefficient information for obtaining the selected coefficient, and smoothing information relating to the smoothing to generate high-frequency encoded data;low-frequency encoding means configured to encode a low-frequency signal that is a low-frequency signal of the input signal to generate low-frequency encoded data; andmultiplexing means configured to multiplex the low-frequency encoded data and the high-frequency encoded data to obtain an output code string.2. The encoding device according to claim 1 , wherein the smoothing means subjects the feature amount to smoothing by ...

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Номер записи: 37
29-08-2013 дата публикации

Stereo signal encoding device, stereo signal decoding device, stereo signal encoding method, and stereo signal decoding method

Номер: US20130223633A1
Автор: Hiroyuki Ehara, Masahiro Oshikiri
Принадлежит: Panasonic Corp

Provided is a stereo signal encoding device that enables a lower bitrate without decreasing quality when applying an intermittent transmission technique to a stereo signal. A stereo encoding unit ( 103 ) generates first stereo encoded data by encoding the stereo signal when the stereo signal of the current frame is an audio section A stereo DTX encoding unit ( 104 ) is a means for encoding the stereo signal when the stereo signal of the current frame is a non-audio section, and generates second stereo encoded data by encoding each of: a monaural signal spectral parameter that is a spectral parameter of a monaural signal generated using the first channel signal and the second channel signal; first channel signal information relating to the first channel signal; and second channel signal information relating to the second channel signal.

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Номер записи: 38
29-08-2013 дата публикации

AUDIO ENCODER, AUDIO DECODER, METHOD FOR ENCODING AND AUDIO INFORMATION, METHOD FOR DECODING AN AUDIO INFORMATION AND COMPUTER PROGRAM USING AN OPTIMIZED HASH TABLE

Номер: US20130226594A1
Автор: Fuchs Guillaume, HILDENBRAND Matthias, Multrus Markus, Rettelbach Nikolaus, Subbaraman Vignesh, TRITTHART Arthur, Warmbold Patrick, Weiss Oliver
Принадлежит: Fraunhofer-Gesellschaft zur Foerderung der angewandten Forschung e.V.

An audio decoder includes an arithmetic decoder for providing decoded spectral values on the basis of an arithmetically encoded representation thereof, and a frequency-domain-to-time-domain converter for providing a time-domain audio representation. The arithmetic decoder selects a mapping rule describing a mapping of a code value onto a symbol code representing a spectral value, or a most significant bit-plane thereof, in a decoded form, in dependence on a context state described by a numeric current context value. The arithmetic decoder determines the numeric current context value in dependence on a plurality of previously decoded spectral values. It evaluates a hash table, entries of which define both significant state values amongst the numeric context values and boundaries of intervals of numeric context values, in order to select the mapping rule, wherein the hash table ari_hash_m is defined as given in FIGS. (), (), () and (). 1. An audio decoder for providing a decoded audio information on the basis of an encoded audio information , the audio decoder comprising:an arithmetic decoder for providing a plurality of decoded spectral values on the basis of an arithmetically encoded representation of the spectral values comprised in the encoded audio information; anda frequency-domain-to-time-domain converter for providing a time-domain audio representation using the decoded spectral values, in order to obtain the decoded audio information;wherein the arithmetic decoder is configured to select a mapping rule describing a mapping of a code value of the arithmetically encoded representation of spectral values representing one or more of the spectral values, or a most significant bit-plane of one or more of the spectral values, in an encoded form, onto a symbol code representing one or more of the spectral values, or a most significant bitplane of one or more of the spectral values, in a decoded form, in dependence on a context state described by a numeric current ...

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Номер записи: 39
29-08-2013 дата публикации

APPARATUS AND METHOD FOR LEVEL ESTIMATION OF CODED AUDIO FRAMES IN A BIT STREAM DOMAIN

Номер: US20130226596A1
Автор: DIATSCHUK Marco, GEIGER Ralf, Lutzky Manfred, SCHNELL Markus
Принадлежит: Fraunhofer-Gesellschaft zur Foerderung der angewandten Forschung e.V.

An apparatus for level estimation of an encoded audio signal is provided. The apparatus has a codebook determinator for determining a codebook from a plurality of codebooks as an identified codebook. The audio signal has been encoded by employing the identified codebook. Moreover, the apparatus has an estimation unit configured for deriving a level value associated with the identified codebook as a derived level value and for estimating a level estimate of the audio signal using the derived level value. 1. An apparatus for level estimation of an encoded audio signal , comprising:a codebook determinator for determining a codebook from a plurality of codebooks as an identified codebook, wherein the audio signal has been encoded by employing the identified codebook, andan estimation unit configured for deriving a level value associated with the identified codebook as a derived level value and, for estimating a level estimate of the audio signal using the derived level value.2. The apparatus according to claim 1 , wherein the estimation unit comprises a scaling unit claim 1 ,wherein the scaling unit is adapted to derive a scalefactor relating to the encoded audio signal or to a portion of the encoded audio signal as a derived scalefactor,wherein the scaling unit is adapted to acquire a scaled level value based on the scalefactor and the derived level value,and wherein the estimation unit is adapted to estimate a level estimate of the audio signal using the scaled level value.3. The apparatus according to claim 2 ,wherein the derived level value is a derived energy value, and the scaling unit is adapted to apply the derived scalefactor on the derived energy value to acquire a scaled level value by multiplying derived energy value by the square of the derived scalefactor, orwherein the derived level value is a derived amplitude value, and the scaling unit is adapted to apply the derived scalefactor on the derived amplitude value to acquire a scaled level value by ...

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Номер записи: 40
05-09-2013 дата публикации

Method and an Apparatus for Encoding/Decoding a Multichannel Audio Signal

Номер: US20130230176A1
Автор: Lang Yue, Virette David, Xu Jianfeng
Принадлежит: Huawei Technologies Co., Ltd.

A method and apparatus for decoding a multichannel audio signal comprising the steps of receiving a downmix audio signal and an interchannel cross correlation parameter; deriving an interchannel phase difference parameter from the received interchannel cross correlation parameter; and calculating a decoded multichannel audio signal for the received downmix audio signal depending on the derived interchannel phase difference parameter. 1. A method for decoding a multichannel audio signal comprising the steps of:receiving a downmix audio signal and an interchannel cross correlation parameter;derivingan interchannel phase difference parameter from the received interchannel cross correlation parameter; andcalculating a decoded multichannel audio signal for the received downmix audio signal depending on the derived interchannel phase difference parameter.2. The method according to claim 1 ,wherein said interchannel phase difference parameter is set to a value π for negative values of the received interchannel cross correlation parameter.3. The method according to claim 1 ,wherein said interchannel phase difference parameter is derived from the received interchannel cross correlation parameter in response to a received IPD-activation flag.4. The method according to claims 1 ,wherein for calculating the decoded multichannel audio signal a synthesis matrix is generated by multiplying a rotation matrix with a calculated prematrix.5. The method according to claim 4 ,wherein said prematrix is calculated on the basis of the received interchannel cross connection parameter and a received channel level difference parameter.6. The method according to claim 4 ,wherein said rotation matrix comprises rotation angles which are calculated on the basis of the derived interchannel phase difference parameter and an overall phase difference parameter or which are calculated on the basis of the derived interchannel phase difference parameter and a predetermined angle value.7. The method ...

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Номер записи: 41
05-09-2013 дата публикации

Downmix Limiting

Номер: US20130230177A1
Автор: Michael Ward, Rhonda Wilson, Roger Dressler, Steven Venezia
Принадлежит: Dolby Laboratories Licensing Corp

The invention relates to downmixing techniques by which output audio signals are obtained from input audio signals partitioned into subgroups. A variable common gain limiting factor is applied to all downmix coefficients that govern the contributions from the input signals in a subgroup. While preserving the proportions between signal values within a subgroup, the invention makes it possible to limit the gain of different input signal subgroups to different extents, so that relatively more perceptible signals can be limited relatively less. It then becomes possible to achieve a consistent dialogue level while transitioning in a less perceptible fashion between signal portions with and without gain limiting. Embodiments of the invention include a method, a mixing system and a computer-program product.

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Номер записи: 42
05-09-2013 дата публикации

COMPUTATIONALLY EFFICIENT AUDIO CODER

Номер: US20130231939A1
Автор: Chakravarthy K. P. P. Kalyan, Mondal Bishwarup, Patwardhan Pushkar P., Ruthramoorthy Navaneetha K.
Принадлежит: Sasken Communication Technologies Limited

The present invention provides a computationally efficient technique for compression encoding of an audio signal, and further provides a technique to enhance the sound quality of the encoded audio signal. This is accomplished by including more accurate attack detection and a computationally efficient quantization technique. The improved audio coder converts the input audio signal to a digital audio signal. The audio coder then divides the digital audio signal into larger frames having a long-block frame length and partitions each of the frames into multiple short-blocks. The audio coder then computes short-block audio signal characteristics for each of the partitioned short-blocks based on changes in the input audio signal. The audio coder further compares the computed short-block characteristics to a set of threshold values to detect presence of an attack in each of the short-blocks and changes the long-block frame length of one or more short-blocks upon detecting the attack in the respective one or more short-blocks. 2. The method of claim 1 , wherein the common_scalefac value comprises:a global gain for a given set of spectral values within a frame.3. The method of claim 1 , wherein a start_common_scalefac comprises a theoretical minimum value of the common_scalefac.4. The method of claim 1 , wherein the quantizer_change comprises a step size to arrive at a final value of common_scalefac.5. The method of claim 1 , wherein initializing the common_scalefac value comprises:initializing the common_scalefac value of the current frame with a predicted_common_scalefac.6. The method of claim 1 , wherein initializing the common_scalefac comprises setting the value of common_scalefac to start_common_scalefac+1 when the predicted_common_scalefac is less than the start_common_scalefac.7. The method of wherein initializing the quantizer_change comprises setting the value of quantizer_change to 1.8. The method of claim 1 , wherein computing the counted_bits associated with the ...

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Номер записи: 43
12-09-2013 дата публикации

DEVICE AND METHOD FOR POSTPROCESSING A DECODED MULTI-CHANNEL AUDIO SIGNAL OR A DECODED STEREO SIGNAL

Номер: US20130236022A1
Автор: Lang Yue, Miao Lei, Virette David, Wu Wenhai
Принадлежит: Huawei Technologies Co., Ltd.

According to the invention, a device (′) for postprocessing at least one channel signal of a plurality of channel signals of a multi-channel signal is described, the at least one channel signal being generated from a decoded downmix signal by a low-bit-rate audio coding/decoding system, the device comprising: a receiver (′) for receiving the at least one channel signal generated from the decoded downmix signal, a time envelope of the decoded downmix signal and a classification indication indicating a transient type of the at least one channel signal, wherein the classification indication is associated to the at least one channel signal, and a postprocessor (′) for postprocessing the at least one channel signal based on the time envelope of the decoded downmix signal weighted by a respective weighting factor and in dependence on the classification indication. 1. A device for postprocessing at least one channel signal of a plurality of channel signals of a multi-channel signal , the at least one channel signal being generated from a decoded downmix signal by a low-bit-rate audio coding/decoding system , the device comprising:a receiver for receiving the at least one channel signal generated from the decoded downmix signal, a time envelope of the decoded downmix signal and a classification indication indicating a transient type of the at least one channel signal, wherein the classification indication is associated to the at least one channel signal; anda postprocessor for postprocessing the at least one channel signal based on the time envelope of the decoded downmix signal weighted by a respective weighting factor and in dependence on the classification indication.2. The device of claim 1 , wherein the receiver is adapted to receive the plurality of channel signals and a plurality of classification indications claim 1 , wherein each of the classification indications is associated to a channel signal of the plurality of channel signals claim 1 , and wherein each of the ...

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Номер записи: 44
12-09-2013 дата публикации

PARTIALLY COMPLEX MODULATED FILTER BANK

Номер: US20130238345A1
Автор: EKSTRAND Per, Purnhagen Heiko, VILLEMOES Lars Falck
Принадлежит: DOLBY INTERNATIONAL AB

An apparatus for processing a plurality of real-valued subband signals using a first real-valued subband signal and a second real-valued subband signal to provide at least a complex-valued subband signal comprises a multiband filter for providing an intermediate real-valued subband signal and a calculator for providing the complex-valued subband signal by combining a real-valued subband signal from the plurality of real-valued subband signals and the intermediate subband signal. 1. A method for processing of a discrete-time audio signal , comprising:processing the discrete-time audio signal by a partially complex analysis filter bank to obtain real subbands and complex subbands;modifying real subband samples of the real subbands and complex subband samples of the complex subbands; andprocessing modified complex subband samples and modified real subband samples by a partially complex synthesis filter bank to obtain a processed audio signal.2. The method of claim 1 ,wherein the modifying comprises a shaping of the time-discrete audio signal in time or frequency, an equalization, a spectral envelope adjustment, a frequency selective panning, or a frequency selective spatialization of audio signals.3. The method of claim 1 ,wherein the processing by the partially complex analysis filterbank generates L subbands, wherein complex subband samples for K subbands are created, wherein real subband samples for (L−K) subbands are obtained, wherein L and K are integers, and wherein K is equal to or smaller than L.4. The method of claim 1 , wherein the modifying comprises performing a first modification on the real subband samples and a second modification on the complex subband samples for shaping the audio signal in time and frequency.5. The method of claim 1 ,wherein the processing by the partially complex analysis filter bank comprises:to filtering the discrete-time audio signal by a cosine modulated analysis filter bank to obtain subbands;creating complex subband samples for ...

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Номер записи: 45
19-09-2013 дата публикации

ADAPTIVE PROCESSING WITH MULTIPLE MEDIA PROCESSING NODES

Номер: US20130246077A1
Автор: Farahani Farhad, Pribadi Marvin, Radhakrishnan Regunathan, Riedmiller Jeffrey, Smithers Michael
Принадлежит: DOLBY LABORATORIES LICENSING CORPORATION

Techniques for adaptive processing of media data based on separate data specifying a state of the media data are provided. A device in a media processing chain may determine whether a type of media processing has already been performed on an input version of media data. If so, the device may adapt its processing of the media data to disable performing the type of media processing. If not, the device performs the type of media processing. The device may create a state of the media data specifying the type of media processing. The device may communicate the state of the media data and an output version of the media data to a recipient device in the media processing chain, for the purpose of supporting the recipient device's adaptive processing of the media data. 1109-. (canceled)110. A method , comprising:determining, by a first device in a media processing chain, whether a type of media processing has been performed on an output version of media data; creating or modifying, by a first device, a state of the media data, the state specifying the type of media processing performed on the output version of the media data;', 'communicating, from the first device to a second device downstream in the media processing chain, the output version of the media data and the state of the media data., 'in response to determining, by the first device, that the type of media processing has been performed on the output version of the media data, performing111. The method as recited in claim 110 , further comprising providing claim 110 , to the second device claim 110 , the state of the media data as one or more of: (a) media fingerprints claim 110 , (b) processing state metadata claim 110 , (c) extracted media feature values claim 110 , (d) media class types or sub-type description(s) and/or values (e) media feature class and/or sub-class probability values claim 110 , (f) cryptographic hash value or (f) media processing signaling.112. The method as recited in claim 110 , wherein the ...

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Номер записи: 46
26-09-2013 дата публикации

Audio Encoding Using Adaptive Codebook Application Ranges

Номер: US20130253938A1
Автор: You Yuli
Принадлежит: DIGITAL RISE TECHNOLOGY CO., LTD.

A low bit rate digital audio coding system includes an encoder which assigns codebooks to groups of quantization indexes based on their local properties resulting in codebook application ranges that are independent of block quantization boundaries. The invention also incorporates a resolution filter bank, or a tri-mode resolution filter bank, which is selectively switchable between high and low frequency resolution modes or high, low and intermediate modes such as when detecting transient in a frame. The result is a multichannel audio signal having a significantly lower bit rate for efficient transmission or storage. The decoder is essentially an inverse of the structure and methods of the encoder, and results in a reproduced audio signal that cannot be audibly distinguished from the original signal. 187-. (canceled)88. A method of encoding a digital audio signal , comprising:processing input samples of an audio signal by using an analysis filter bank so as to transform the input audio samples into subband samples that represent the audio signal in a frequency domain;creating quantization indexes by quantizing the subband samples;dividing the quantization indexes into granules, each containing a plurality of quantization indexes;assigning codebooks to individual granules, with each range of contiguous granules that have the same codebook index being an application range for said codebook;replacing codebooks assigned to identified application ranges with the codebook assigned to an immediate neighbor application range, thereby expanding the application ranges of said immediate neighbor codebooks;encoding the quantization indexes using the codebooks applicable within the respective application ranges, including changes made in said replacing step;creating an encoded data stream, including the encoded quantization indexes, indexes for the codebooks and the respective codebook application ranges; andat least one of storing or transmitting the encoded data stream.89. The ...

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Номер записи: 47
03-10-2013 дата публикации

STEREO PARAMETRIC CODING/DECODING FOR CHANNELS IN PHASE OPPOSITION

Номер: US20130262130A1
Автор: Hoang Thi Minh Nguyet, RAGOT STEPHANE
Принадлежит: FRANCE TELECOM

A method and apparatus for the parametric encoding of a stereo digital-audio signal. The method includes encoding a mono signal produced by downmixing applied to the stereo signal and encoding spatialisation information of the stereo signal. Downmixing includes determining, for a predetermined set of frequency sub-bands, a phase difference between two stereo channels; obtaining an intermediate channel by rotating a first predetermined channel of the stereo signal through an angle obtained by reducing the phase difference; determining the phase of the mono signal from the phase of the signal that is the sum of the intermediate channel and the second stereo signal, and from a phase difference between, on the one hand, the signal that is the sum of the intermediate channel and the second channel and, on the other hand, the second channel of the stereo signal. Also provided are a decoding method, an encoder and a decoder. 1. A method for parametric coding of a stereo digital audio signal comprising: determining, for a predetermined set of frequency sub-bands, a phase difference between two stereo channels;', 'obtaining an intermediate channel by rotation of a predetermined first channel of the stereo signal, through an angle obtained by reduction of said phase difference; and', 'determining the phase of the mono signal from the phase of the signal summing the intermediate channel and the second stereo signal and from a phase difference between, on the one hand, the signal summing the intermediate channel and the second channel and, on the other hand, the second channel of the stereo signal., 'wherein the channel reduction processing comprises the following steps, 'a step of coding a mono signal coming from a channel reduction processing applied to the stereo signal and coding information on spatialization of the stereo signal,'}2. The method as claimed in claim 1 , wherein the mono signal is determined according to the following steps:obtaining, by frequency band, an ...

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Номер записи: 48
10-10-2013 дата публикации

MDCT-Based Complex Prediction Stereo Coding

Номер: US20130266145A1
Автор: CARLSSON Pontus, Purnhagen Heiko, VILLEMOES Lars
Принадлежит: Heiko Purnhagen

The invention provides methods and devices for stereo encoding and decoding using complex prediction in the frequency domain. In one embodiment, 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 on the basis of 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. The method comprises performing frequency-domain modifications selectively before or after upmixing. 115-. (canceled)16. A decoder system for providing a stereo signal by complex prediction stereo coding , the decoder system comprising:an upmix stage adapted to generate the stereo signal based on first frequency-domain representations of a downmix signal and a residual signal, each of the first frequency-domain representations comprising first spectral components representing spectral content of the corresponding signal expressed in a first subspace of a multidimensional space, the upmix stage comprising:a module for computing a second frequency-domain representation of the downmix signal based on the first frequency-domain representation thereof, the second frequency-domain representation comprising second spectral components representing spectral content of the signal expressed in a second subspace of the multidimensional space that includes a portion of the multidimensional space not included in the first subspace;a weighted summer for computing a side signal on the basis of the first and second frequency-domain representations of the downmix signal, the first frequency-domain representation of the residual signal and a complex prediction coefficient encoded in the bit stream ...

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Номер записи: 49
24-10-2013 дата публикации

DEVICE AND METHOD FOR POSTPROCESSING A DECODED MULTI-CHANNEL AUDIO SIGNAL OR A DECODED STEREO SIGNAL

Номер: US20130279702A1
Автор: Lang Yue, Miao Lei, Virette David, Wu Wenhai
Принадлежит:

According to the invention, a device for post-processing at least one channel signal of a plurality of channel signals of a multi-channel signal is described, the at least one channel signal being generated from a decoded downmix signal by a low-bit-rate audio coding/decoding system, the device comprising: a receiver for receiving the at least one channel signal generated from the decoded downmix signal, a time envelope of the decoded downmix signal, an interchannel time difference between the channel signal and the downmix signal, and a classification indication indicating a transient type of the downmix signal; and a post-processor for post-processing the at least one channel signal based on the time envelope of the decoded downmix signal weighted by a respective weighting factor and in dependence on the classification indication and the interchannel time difference. 1. A device for post-processing at least one channel signal of a plurality of channel signals of a multi-channel signal , the at least one channel signal being generated from a decoded downmix signal by a low-bit-rate audio coding/decoding system , the device comprising:a receiver for receiving the at least one channel signal generated from the decoded downmix signal, a time envelope of the decoded downmix signal, an interchannel time difference between the at least one channel signal and the downmix signal, and a classification indication indicating a transient type of the downmix signal; anda post-processor for post-processing the at least one channel signal based on the time envelope of the decoded downmix signal weighted by a respective weighting factor and in dependence on the classification indication and the interchannel time difference.2. The device of claim 1 , wherein the receiver is adapted to receive the plurality of channel signals and a plurality of interchannel time differences claim 1 , wherein each of the interchannel time differences is associated to a channel signal of the plurality ...

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Номер записи: 50
24-10-2013 дата публикации

MULTI-CHANNEL ENCODING AND/OR DECODING

Номер: US20130282386A1
Автор: Nikunen Joonas, Vilermo Miikka, Virtanen Tuomas
Принадлежит: Nokia Corporation

A method comprising: receiving input signals for multiple channels; and parameterizing the received input signals into parameters defining multiple different object spectra and defining a distribution of the multiple different object spectra in the multiple channels. 165-. (canceled)66. A method comprising:receiving audio signals for multiple channels, wherein each channel provide separately captured audio signals; andparameterizing the received input signals into parameters defining multiple different object spectra and defining a distribution of the multiple different object spectra in the multiple channels.67. The method as claimed in claim 66 , wherein the parameters comprise tensors including a first tensor representing object spectra claim 66 , a second tensor representing the variation of gain for each object spectra with time claim 66 , and a third tensor representing the variation of gain for each object spectra in respective channels.68. The method as claimed in claim 66 , further comprising:transforming received input signals, from different channels, into a frequency domain and analyzing the transformed input signals to identify a plurality of object spectra; andidentifying object spectra that best match the transformed input signals and time-dependent and channel-dependent gains of the identified object spectra.69. The method as claimed in claim 66 , further comprising performing non-negative tensor factorization claim 66 , wherein object spectra are defined in a first tensor claim 66 , time-dependent gain of the object spectra are defined in a second tensor claim 66 , and channel-dependent gain of the object spectra are defined in a third tensor.70. The method as claimed in claim 66 , further comprising minimizing a cost function claim 66 , that includes a measure of difference between a reference determined from the received input signals and an iterated estimate determined using putative parameters claim 66 , wherein the putative parameters that ...

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Номер записи: 51
07-11-2013 дата публикации

ADVANCED ENCODING OF MUSIC FILES

Номер: US20130294606A1
Автор: HORTON Christopher, SOROKA Howard
Принадлежит: Universal Music Group, Inc.

Example embodiments allow for the creation, distribution, and use of flexible media formats. Example embodiments may allow individual content files to be rendered in multiple formats and versions. In addition, example embodiments may provide for granular rights management, which may allow users to access content files on a feature-by-feature basis. 164-. (canceled)65. A system for the playback of content files , comprising:a memory storing a content file including a plurality of stems, each stem encoding a portion of the audio of a sound recording, multiple stems in the plurality of stems representing different portions of the sound recording for the same time period, the content file also including a set of instructions controlling playback of the stems;a decoder configured to decode the stems according to the set of instructions to create an audio output signal; andan output device configured to output an audio output based on the audio output signal.66. The system of claim 65 , wherein the content file includes multiple different files for the same sound recording claim 65 , and wherein the set of instructions controls the playback of stems from the multiple different files claim 65 , the decoder being further configured to decode the stems from the multiple files according to the set of instructions to create the audio output signal.67. The system of claim 65 , wherein at least a subset of the stems represent instrumental tracks of the sound recording for respective instruments.68. The system of claim 65 , wherein at least a subset of the stems represent spatial arrangement of sound for the sound recording.69. The system of claim 65 , wherein at least a subset of the instructions represent represent spatial arrangement of sound for the sound recording.70. The system of claim 65 , wherein at least a subset of the stems represent multiple versions of the same portion of the sound recording.71. The system of claim 65 , wherein the content file further contains ...

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Номер записи: 52
07-11-2013 дата публикации

BROADCAST TRANSMITTING APPARATUS AND BROADCAST TRANSMITTING METHOD FOR PROVIDING AN OBJECT-BASED AUDIO, AND BROADCAST PLAYBACK APPARATUS AND BROADCAST PLAYBACK METHOD

Номер: US20130294607A1
Автор: Ahn Chie Teuk, KANG Kyeong Ok, Kim Jin Woong, Lee Tae Jin, LEE Yong Ju
Принадлежит: Electronics and Telecommunications Research Institute

A broadcast transmitting apparatus and method, and a broadcast playback apparatus and method for providing an object-based audio by encoding and decoding a multichannel audio signal are provided. The broadcast transmitting apparatus may generate audio identification information used to determine whether the multichannel audio signal is an object-based audio signal. When the multichannel audio signal is determined to be the object-based audio signal, based on the audio identification information, the broadcast playback apparatus may control and output the multichannel audio signal for each channel. 1. A broadcast transmitting apparatus , comprising:an audio encoder to encode a multichannel audio signal; andan audio identification information generator to generate audio identification information, the audio identification information being used to determine whether the multichannel audio signal is an object-based audio signal.2. The broadcast transmitting apparatus of claim 1 , wherein the audio identification information generator generates at least one piece of mixing information including a scheme of mixing channels claim 1 , when the multichannel audio signal is determined to be the object-based audio signal.3. The broadcast transmitting apparatus of claim 1 , wherein the audio identification information generator generates the audio identification information in the form of a descriptor.4. A broadcast transmitting method claim 1 , comprising:encoding a multichannel audio signal; andgenerating audio identification information, the audio identification information being used to determine whether the multichannel audio signal is an object-based audio signal.5. The broadcast transmitting method of claim 4 , wherein the generating comprises claim 4 , when the multichannel audio signal is determined to be the object-based audio signal claim 4 , generating at least one piece of mixing information including a scheme of mixing channels.6. The broadcast transmitting method ...

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Номер записи: 53
14-11-2013 дата публикации

DETERMINING THE INTER-CHANNEL TIME DIFFERENCE OF A MULTI-CHANNEL AUDIO SIGNAL

Номер: US20130301835A1
Автор: Briand Manuel, JANSSON TOFTGÅRD Tomas
Принадлежит: TELEFONAKTIEBOLAGET L M ERICSSON (PUBL)

There is provided a method and device for determining an inter-channel time difference of a multi-channel audio signal having at least two channels. A determination is made, at a number of consecutive time instances, of inter-channel correlation based on a cross-correlation function involving at least two different channels of the multi-channel audio signal. Each value of the inter-channel correlation is associated with a corresponding value of the inter-channel time difference. An adaptive inter-channel correlation threshold is adaptively determined based on adaptive smoothing of the inter-channel correlation in time. A current value of the inter-channel correlation is then evaluated in relation to the adaptive inter-channel correlation threshold to determine whether the corresponding current value of the inter-channel time difference is relevant. Based on the result of this evaluation, an updated value of the inter-channel time difference is determined. 1. A method for determining an inter-channel time difference of a multi-channel audio signal having at least two channels , wherein said method comprises the steps of:determining, at a number of consecutive time instances, an inter-channel correlation based on a cross-correlation function involving at least two different channels of the multi-channel audio signal, wherein each value of the inter-channel correlation is associated with a corresponding value of the inter-channel time difference;adaptively determining an adaptive inter-channel correlation threshold based on adaptive smoothing of the inter-channel correlation in time;evaluating a current value of inter-channel correlation in relation to the adaptive inter-channel correlation threshold to determine whether the corresponding current value of the inter-channel time difference is relevant; anddetermining an updated value of the inter-channel time difference based on the result of this evaluation.2. The method of claim 1 , wherein said step of evaluating a ...

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Номер записи: 54
14-11-2013 дата публикации

Determining the Inter-Channel Time Difference of a Multi-Channel Audio Signal

Номер: US20130304481A1
Автор: Briand Manuel, Jansson Tomas
Принадлежит: TELEFONAKTIEBOLAGET L M ERICSSON (PUBL)

There is provided a method and device for determining an inter-channel time difference of a multi-channel audio signal having at least two channels. A set of local maxima of a cross-correlation function involving at least two different channels of the multi-channel audio signal is determined (S) for positive and negative time-lags, where each local maximum is associated with a corresponding time-lag. From the set of local maxima, a local maximum for positive time-lags is selected as a so-called positive time-lag inter-channel correlation candidate and a local maximum for negative time-lags is selected as a so-called negative time-lag inter-channel correlation candidate (S). When the absolute value of a difference in amplitude between the inter-channel correlation candidates is smaller than a first threshold, it is evaluated whether there is an energy-dominant channel (S). When there is an energy-dominant-channel, the sign of the inter-channel time difference is identified and a current value of the inter-channel time difference is extracted based on either the time-lag corresponding to the positive time-lag inter-channel con-elation candidate or the time-lag corresponding to the negative time-lag inter-channel correlation candidate (S). 118-. (canceled)19. A method for determining an inter-channel time difference of a multi-channel audio signal having at least two channels , wherein said method comprises the steps of:determining a set of local maxima of a cross-correlation function involving at least two different channels of the multi-channel audio signal for positive and negative time-lags, where each local maximum is associated with a corresponding time-lag;selecting, from the set of local maxima, a local maximum for positive time-lags as a so-called positive time-lag inter-channel correlation candidate and a local maximum for negative time-lags is selected as a so-called negative time-lag inter-channel correlation candidate;evaluating, when the absolute value of ...

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Номер записи: 55
28-11-2013 дата публикации

BANDWIDTH EXPANSION METHOD AND APPARATUS

Номер: US20130317831A1
Автор: Liu Zexin, Miao Lei
Принадлежит: Huawei Technologies Co., Ltd.

A bandwidth expansion method and apparatus are disclosed, where the method includes: estimating a bandwidth of at least one decoded frame of a whole-band signal, so as to obtain an estimated bandwidth, where the estimated bandwidth corresponds to a whole-band signal that a decoded lower-band signal needs to be extended into; performing first predictive decoding on a part of the lower-band signal in a band above an effective bandwidth of the lower-band signal and below the estimated bandwidth, so as to obtain the part of the lower-band signal above the effective bandwidth of the lower-band signal and below the estimated bandwidth; and performing second predictive decoding on a part of the lower-band signal in a band above the estimated bandwidth, so as to obtain the part of the lower-band signal above the estimated bandwidth. 1. A bandwidth expansion method , comprising:estimating a bandwidth of at least one decoded frame of a whole-band signal, so as to obtain an estimated bandwidth; wherein the estimated bandwidth corresponds to a whole-band signal that a decoded lower-band signal needs to be extended into;performing first predictive decoding on a part of the lower-band signal in a band above an effective bandwidth of the lower-band signal and below the estimated bandwidth, so as to obtain the part of the lower-band signal above the effective bandwidth of the lower-band signal and below the estimated bandwidth; andperforming second predictive decoding on a part of the lower-band signal in a band above the estimated bandwidth, so as to obtain the part of the lower-band signal above the estimated bandwidth.2. The method according to claim 1 , wherein estimating a bandwidth of a decoded whole-band signal claim 1 , so as to obtain an estimated bandwidth claim 1 , comprises:dividing a high-band signal comprised in each decoded frame of the whole-band signal into N bands in ascending order of frequency, wherein N is an integer greater than 1;for each frame of the whole- ...

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Номер записи: 56
12-12-2013 дата публикации

Method And Apparatus For Delivery Of Aligned Multi-Channel Audio

Номер: US20130329892A1
Автор: Richard Jones Anthony
Принадлежит: Ericsson Television Inc.

There is provided a method of encoding audio and including said encoded audio into a digital transport stream, comprising receiving at an encoder input a plurality of temporally co-located audio signals, assigning identical time stamps per unit time to all of the plurality of temporally co-located audio signals and incorporating the identically time stamped audio signals into the digital transport stream. There is also provided a method decoding said encoded data, and encoding apparatus and decoding apparatus. 1. A method of encoding audio and including said encoded audio into a digital transport stream , comprising:receiving, at an encoder input, a plurality of temporally co-located audio signals;sampling the plurality of temporally co-located audio signals to form a plurality of aligned frames of audio data of a predetermined size;assigning identical time stamps per unit time to the plurality of aligned frames of audio data; andincorporating, into the digital transport stream, the plurality of aligned frames of audio data with identical time stamps.2. The method of claim 1 , further comprising:compressing the plurality of aligned frames of audio data with identical audio encoder configuration settings prior to assigning the identical time stamps; andallocating the plurality of aligned frames to a plurality of mono channels of a transport stream.3. The method of claim 2 , wherein the plurality of mono channels comprises one or more conventional dual mono audio components.4. The method of claim 1 , wherein the predetermined size is the size of an Access Unit in the MPEG standard claim 1 , and the video transport stream is a MPEG-1 or MPEG-2 Transport stream.5. The method of claim 1 , wherein the time stamps are Presentation Time Stamps.6. The method of claim 1 , wherein the step of incorporating further comprises:multiplexing identically time stamped audio data into a transport stream.7. A method of decoding a digital transport stream claim 1 , comprising:receiving ...

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Номер записи: 57
26-12-2013 дата публикации

METHOD AND APPARATUS FOR ENCODING/DECODING STEREO AUDIO

Номер: US20130343551A1
Автор: JEONG Jong-hoon, Lee Chul-Woo, LEE Geon-hyoung, LEE Nam-suk, MOON Han-gil
Принадлежит: SAMSUNG ELECTRONICS CO., LTD.

Provided are a method and apparatus for encoding/decoding stereo audio. In the method for encoding stereo audio, stereo audio is encoded based on at least one of the phase difference between first and second channel audios and information on an angle made by a vector on the intensity of mono-audio and a vector on the intensity of the first channel audio or a vector on the intensity of the second channel audio. Thus, the number of encoded parameters is minimized so that a compression ratio in the encoding of the stereo audio is improved. 1. A method of decoding stereo audio , the method comprising:restoring mono-audio by decoding audio data on the stereo audio;extracting information for determining intensities of first and second channel audios and information on a phase difference between the first and second channel audios in a predetermined frequency band by decoding the audio data; andrestoring the stereo audio in the frequency band based on the restored mono-audio and the extracted information,wherein the mono-audio is generated by adding the first channel audio and a phase-adjusted second channel audio whose phase is adjusted to be the same as a phase of the first channel audio.2. The method of claim 1 , wherein the restoring of the stereo audio comprises:calculating the phase of the second channel audio in the frequency band based on the information on the phase difference and the phase of the mono-audio; andrestoring the stereo audio based on a phase of the mono-audio, the phase of the second channel audio, the intensity of the first channel audio, and the intensity of the second channel audio.3. The method of claim 1 , wherein the information for determining the intensities of the first and second channel audios is information on an angle between a first vector on the intensity of the first channel audio in the frequency band and a third vector on an intensity of the mono-audio or an angle between a second vector on the intensity of the second channel audio ...

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Номер записи: 58
02-01-2014 дата публикации

AUDIO ENCODING DEVICE AND AUDIO ENCODING METHOD

Номер: US20140006035A1
Автор: Kishi Yohei, Shirakawa Miyuki, SUZUKI Masanao, Takeuchi Shunsuke
Принадлежит:

An audio encoding device includes a processor; and a memory which stores a plurality of instructions, which when executed by the processor, cause the processor to execute, calculating first phases indicating phases of a first channel signal and a second channel signal included in audio signals of a plurality of channels; and performing, on the basis of the first phases, either first predictive coding in which a third channel signal included in the audio signals of the plurality of channels is predicted using the first channel signal and the second channel signal or second predictive coding in which the second channel signal is predicted using the first channel signal. 1. An audio encoding device comprising:a processor; anda memory which stores a plurality of instructions, which when executed by the processor, cause the processor to execute,calculating first phases indicating phases of a first channel signal and a second channel signal included in audio signals of a plurality of channels; andperforming, on the basis of the first phases, either first predictive coding in which a third channel signal included in the audio signals of the plurality of channels is predicted using the first channel signal and the second channel signal or second predictive coding in which the second channel signal is predicted using the first channel signal.2. The audio encoding device according to claim 1 ,wherein, when the first phases are other than identical phases or opposite phases, the first predictive coding is performed in the performing, and, when the first phases are the identical phases or the opposite phases, the second predictive coding is performed in the performing.3. The audio encoding device according to claim 1 ,wherein, in the performing, selection information indicating that the first predictive coding or the second predictive coding has been performed as predictive coding is generated.4. The audio encoding device according to claim 1 , further comprising:generating, on ...

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Номер записи: 59
02-01-2014 дата публикации

ENCODING DEVICE, ENCODING METHOD, AND PROGRAM

Номер: US20140006037A1
Автор: Chinen Toru, HATANAKA Mitsuyuki, Honma Hiroyuki, Yamamoto Yuki
Принадлежит: SONG CORPORATION

This technology relates to an encoding device, an encoding method, and a program capable of improving audio quality and more efficiently encoding audio. A first high-frequency encoding circuit encodes a high-frequency range based on a low-frequency subband signal and a high-frequency subband signal and obtains a high-frequency code amount. A low-frequency encoding circuit encodes a low-frequency signal with a code amount determined by the high-frequency code amount and a low-frequency decoding circuit decodes the encoded low-frequency signal. A subband dividing circuit divides a decoded low-frequency signal obtained by decoding into decoded low-frequency subband signals of a plurality of subbands and a second high-frequency encoding circuit generates a high-frequency code string such that a code amount of the high-frequency code string for obtaining a high-frequency component is not larger than the high-frequency code amount based on the decoded low-frequency subband signals and the high-frequency subband signals. The present invention is applicable to the encoding device. 1. An encoding device , comprisinga first high-frequency encoding unit which calculates a high-frequency code amount being a code amount of a high-frequency code string for obtaining a high-frequency component based on a low-frequency component and the high-frequency component of an input signal;a low-frequency encoding unit which encodes the low-frequency component of the input signal to generate a low-frequency code string;a low-frequency decoding unit which decodes the low-frequency code string;a second high-frequency encoding unit which generates the high-frequency code string based on a decoded low-frequency component obtained by decoding the low-frequency code string and the high-frequency component such that the code amount of the high-frequency code string is not larger than the high-frequency code amount; anda multiplexing unit which multiplexes the low-frequency code string and the high- ...

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Номер записи: 60
16-01-2014 дата публикации

AUDIO ENCODER AND DECODER HAVING A FLEXIBLE CONFIGURATION FUNCTIONALITY

Номер: US20140016785A1
Автор: DE BONT Frans, DOEHLA Stefan, Multrus Markus, Neuendorf Max, Purnhagen Heiko
Принадлежит:

An audio decoder for decoding an encoded audio signal, the encoded audio signal including a first channel element and a second channel element in a payload section of a data stream and first decoder configuration data for the first channel element and second decoder configuration data for the second channel element in a configuration section of the data stream, includes: a data stream reader for reading the configuration data for each channel element in the configuration section and for reading the payload data for each channel element in the payload section; a configurable decoder for decoding the plurality of channel elements; and a configuration controller for configuring the configurable decoder so that the configurable decoder is configured in accordance with the first decoder configuration data when decoding the first channel element and in accordance with the second decoder configuration data when decoding the second channel element. 1. Audio decoder for decoding an encoded audio signal , the encoded audio signal comprising a first channel element and a second channel element in a payload section of a data stream and first decoder configuration data for the first channel element and second decoder configuration data for the second channel element in a configuration section of the data stream , comprising:a data stream reader for reading the configuration data for each channel element in the configuration section and for reading the payload data for each channel element in the payload section;a configurable decoder for decoding the plurality of channel elements; anda configuration controller for configuring the configurable decoder so that the configurable decoder is configured in accordance with the first decoder configuration data when decoding the first channel element and in accordance with the second decoder configuration data when decoding the second channel element.2. Audio decoder in accordance with claim 1 ,wherein the first channel element is a ...

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Номер записи: 61
16-01-2014 дата публикации

SYSTEMS, METHODS, APPARATUS, AND COMPUTER-READABLE MEDIA FOR THREE-DIMENSIONAL AUDIO CODING USING BASIS FUNCTION COEFFICIENTS

Номер: US20140016786A1
Автор: Sen Dipanjan
Принадлежит: QUALCOMM INCORPORATED

Systems, methods, and apparatus for a unified approach to encoding different types of audio inputs are described. 1. A method of audio signal processing , said method comprising:encoding an audio signal and spatial information for the audio signal into a first set of basis function coefficients that describes a first sound field; andcombining the first set of basis function coefficients with a second set of basis function coefficients that describes a second sound field during a time interval to produce a combined set of basis function coefficients that describes a combined sound field during the time interval.2. The method according to claim 1 , wherein said audio signal is a frame of a corresponding stream of audio samples.3. The method according to claim 1 , wherein said audio signal is a frame of a pulse-code-modulation (PCM) stream.4. The method according to claim 1 , wherein said spatial information for the audio signal indicates a direction in space.5. The method according to claim 1 , wherein said spatial information for the audio signal indicates a location in space of a source of the audio signal.6. The method according to claim 1 , wherein said spatial information for the audio signal indicates a diffusivity of the audio signal.7. The method according to claim 1 , wherein said audio signal is a loudspeaker channel.8. The method according to claim 1 , wherein said method includes obtaining an audio object that includes said audio signal and said spatial information for said audio signal.9. The method according to claim 1 , wherein said method includes encoding a second audio signal and spatial information for the second audio signal into the second set of basis function coefficients.10. The method according to claim 1 , wherein each basis function coefficient of said first set of basis function coefficients corresponds to a unique one of a set of orthogonal basis functions.11. The method according to claim 1 , wherein each basis function coefficient of ...

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Номер записи: 62
16-01-2014 дата публикации

FRAME ELEMENT LENGTH TRANSMISSION IN AUDIO CODING

Номер: US20140016787A1
Автор: DE BONT Frans, DOEHLA Stefan, Multrus Markus, Neuendorf Max, Purnhagen Heiko
Принадлежит:

Frame elements which shall be made available for skipping may are transmitted more efficiently by arranging that a default payload length information is transmitted separately within a configuration block, with the length information within the frame elements, in turn, being subdivided into a default payload length flag followed, if the default payload length flag is not set, by a payload length value explicitly coding the payload length of the respective frame element. However, if the default payload length flag is set, an explicit transmission of the payload length may be avoided. Rather, any frame element, the default extension payload length flag of which is set, has the default payload length and any frame element, the default extension payload length flag of which is not set, has a payload length corresponding to the payload length value. By this measure, transmission effectiveness is increased. 1. A bitstream comprising a configuration block and a sequence of frames respectively representing consecutive time periods of an audio content , wherein the sequence of frames is a composition of N sequences of frame elements with each frame element being of a respective one of a plurality of element types so that each frame comprises one frame element out of the N sequences of frame elements , respectively , and for each sequence of frame elements , the frame elements are of equal element type relative to each other ,wherein the configuration block comprises, for at least one of the sequences of frame elements, a default payload length information on a default payload length, andwherein each frame element of the at least one of the sequences of frame elements, comprises a length information comprising, for at least a subset of the frame elements of the at least one of the sequences of frame elements, a default payload length flag followed, if the default payload length flag is not set, by a payload length value,wherein any frame element of the at least one of the ...

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Номер записи: 63
30-01-2014 дата публикации

AUDIO DECODING DEVICE AND AUDIO DECODING METHOD

Номер: US20140029752A1
Автор: Kamano Akira, Kishi Yohei, Shirakawa Miyuki, SUZUKI Masanao, Takeuchi Shunsuke
Принадлежит: FUJITSU LIMITED

An audio decoding device includes a processor; and a memory which stores a plurality of instructions, which when executed by the processor, cause the processor to execute, decoding, using a first channel signal and a second channel signal included in a plurality of channels of an audio signal having a first frequency range and a second frequency range, a first prediction coefficient of the first frequency range and a second prediction coefficient of the second frequency range, both selected from a code book when prediction-encoding a third channel signal that is not subjected to prediction encoding and that is included in the plurality of channels; decoding a residual signal included in the first frequency range, the residual signal representing an error occurring in prediction encoding; and prediction-decoding the third channel signal subjected to prediction-encoding in the second frequency range from the first channel signal, the second channel signal. 1. An audio decoding device comprising:a processor; anda memory which stores a plurality of instructions, which when executed by the processor, cause the processor to execute,decoding, using a first channel signal and a second channel signal included in a plurality of channels of an audio signal having a first frequency range and a second frequency range, a first prediction coefficient of the first frequency range and a second prediction coefficient of the second frequency range, both selected from a code book when prediction-encoding a third channel signal that is not subjected to prediction encoding and that is included in the plurality of channels;decoding a residual signal included in the first frequency range, the residual signal representing an error occurring in prediction encoding; andprediction-decoding the third channel signal subjected to prediction-encoding in the second frequency range from the first channel signal, the second channel signal, the third channel signal subjected to prediction encoding, ...

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Номер записи: 64
30-01-2014 дата публикации

METHOD AND APPARATUS FOR PROCESSING AUDIO DATA

Номер: US20140032226A1
Автор: GADDE Raj Narayana, Kim Do Hyung, Lee Kang Eun, RAJU Sandeep, SON Chang Yong
Принадлежит: SAMSUNG ELECTRONICS CO., LTD.

A method and apparatus for processing audio data are provided. When an encoded audio bitstream sampled at a sampling frequency is received, a resampling ratio for processing the encoded audio bitstream is computed. If the the resampling ratio is within the resampling threshold range, then the encoded audio bitstream is processed in frequency domain and a desired number of audio samples per frame are outputted according to the resampling ratio. The encoded audio bitstream is processed in frequency domain using sample rate converter integrated into a filter bank of an audio decoder. If the resampling ratio is outside the resampling threshold range, then the encoded audio bitstream is processed in time domain and a desired number of audio samples per frame are outputted according to the resampling ratio. 1. A method of processing audio data in frequency domain , comprising:partially decoding an encoded audio bitstream to obtain de-quantized spectral data, wherein the encoded audio bitstream is sampled at a first sampling frequency;modifying the de-quantized spectral data based on a resampling ratio; andsynthesizing the modified spectral data according to the resampling ratio to reproduce audio data sampled at a second sampling frequency.2. The method of claim 1 , wherein modifying the de-quantized spectral data based on the resampling ratio comprises:padding the de-quantized spectral data with constant values based on the resampling ratio if the second sampling frequency is greater than the first sampling frequency.3. The method of claim 1 , wherein modifying the de-quantized spectral data based on the resampling ratio comprises:padding the de-quantized spectral data with constant values based on the resampling ratio if the second sampling frequency is less than the first sampling frequency such that audio samples per frame obtained after padding the de-quantized spectral data is integer multiple of desired audio samples per frame.4. The method of and claim 1 , wherein ...

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Номер записи: 65
20-02-2014 дата публикации

DATA EMBEDDING DEVICE, DATA EMBEDDING METHOD, DATA EXTRACTOR DEVICE, AND DATA EXTRACTION METHOD

Номер: US20140050324A1
Автор: Kamano Akira, Kishi Yohei, SUZUKI Masanao, Takeuchi Shunsuke
Принадлежит:

A data embedding device includes a processor; and a memory which stores a plurality of instructions, which when executed by the processor, cause the processor to execute, extracting, from a code book storing a plurality of prediction coefficients, a plurality of candidates of prediction coefficients of a channel signal out of a plurality of channel signals, each candidate having a predictive error falling within a specific range of predictive error of predictive encoding of two other channel signals; and selecting from the extracted candidates a prediction coefficient as a result of the predictive encoding in accordance with a specific data embedding rule and embedding embed target data into the selected prediction coefficient. 1. A data embedding device comprising:a processor; anda memory which stores a plurality of instructions, which when executed by the processor, cause the processor to execute,extracting, from a code book storing a plurality of prediction coefficients, a plurality of candidates of prediction coefficients of a channel signal out of a plurality of channel signals, each candidate having a predictive error falling within a specific range of predictive error of predictive encoding of two other channel signals; andselecting from the extracted candidates a prediction coefficient as a result of the predictive encoding in accordance with a specific data embedding rule and embedding embed target data into the selected prediction coefficient.2. The device according to claim 1 ,wherein the prediction coefficient contains a component of each of the two other channel signals, andwherein the extracting comprises determining a straight line as a set of points, each point having a minimum predictive error on a plane that is defined by the two components of the two prediction coefficients, and extracting the candidate of prediction coefficient in accordance with a positional relationship between the straight line and points on the plane corresponding to the ...

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Номер записи: 66
20-02-2014 дата публикации

MULTI-DIMENSIONAL PARAMETRIC AUDIO SYSTEM AND METHOD

Номер: US20140050325A1
Автор: NORRIS ELWOOD GRANT
Принадлежит:

Systems and methods for producing multi-dimensional parametric audio are provided. The systems and methods can be configured to determine a desired spatial position of an audio component relative to a predetermined listening position; process the audio component for a predetermined number of output channels, wherein the step of processing the audio component comprises determining the appropriate phase, delay, and gain values for each output channel so that the audio component is created at the desired apparent spatial position relative to the listening position; encode two or more output channels of the audio component with the determined phase, delay, and gain values for each output channel; and modulate the encoded output channels onto respective ultrasonic carriers for emission via a predetermined number of ultrasonic emitters 1. A method of producing multi-dimensional parametric audio , comprising:determining a desired spatial position of an audio component relative to a predetermined listening position;processing the audio component for a predetermined number of output channels, wherein the step of processing the audio component comprises determining the appropriate phase, delay, and gain values for each output channel so that the audio component is created at the desired apparent spatial position relative to the listening position;encoding two or more output channels of the audio component with the determined phase, delay, and gain values for each output channel; andmodulating the encoded output channels onto respective ultrasonic carriers for emission via a predetermined number of ultrasonic emitters.2. The method of claim 1 , wherein the step of processing the audio component further comprises determining echo claim 1 , reverb claim 1 , flange claim 1 , and phasor values and the encoding step further comprises encoding two or more output channels with the determined echo claim 1 , reverb claim 1 , flange claim 1 , and phasor values.3. The method of claim 1 , ...

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Номер записи: 67
20-02-2014 дата публикации

Method, medium, and apparatus encoding and/or decoding multichannel audio signals

Номер: US20140052455A1
Автор: Eun-mi Oh, Jung-Hoe Kim
Принадлежит: SAMSUNG ELECTRONICS CO LTD

A method, medium, and apparatus encoding and/or decoding a multichannel audio signal. The method includes detecting the type of spatial extension data included in an encoding result of an audio signal, if the spatial extension data is data indicating a core audio object type related to a technique of encoding core audio data, detecting the core audio object type; decoding core audio data by using a decoding technique according to the detected core audio object type, if the spatial extension data is residual coding data, decoding the residual coding data by using the decoding technique according to the core audio object type, and up-mixing the decoded core audio data by using the decoded residual coding data. According to the method, the core audio data and residual coding data may be decoded by using an identical decoding technique, thereby reducing complexity at the decoding end.

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Номер записи: 68
13-03-2014 дата публикации

APPARATUS AND METHOD FOR RESTORING MULTI-CHANNEL AUDIO SIGNAL USING HE-AAC DECODER AND MPEG SURROUND DECODER

Номер: US20140074487A1
Автор: Beack Seung-Kwon, HONG Jin-Woo, JANG Dae-Young, Jang In-Seon, KIM Jin-Woong, SEO Jeong-Il
Принадлежит: Electronics and Telecommunications Research Institute

Provided is a method for controlling synchronizing downmix signals and MPEG surround side information signals by controlling a delay according to the kind of downmix audio signals in an MPEG surround decoder. When multi-channel audio signals are restored using an HE-AAC decoder and a low-power MPEG surround decoder and complex QMF signals outputted from the HE-AAC decoder are used as downmix signals, a delay unit compensates for a delay caused in a real-to-complex converter. Anther delay unit delays spatial parameters to compensate for a delay caused in QMF and Nyquist banks when time-domain downmix signals are used. Also, when multi-channel audio signals are restored using an HE-AAC decoder and a high-quality MPEG surround decoder and complex QMF signals outputted from the HE-AAC decoder are used as downmix signals, a delay unit compensates for a delay caused in a real-to-complex converter. 1. An apparatus for restoring multi-channel audio signals , comprising:an HE-AAC decoder for outputting downmix signals of at least one among a real QMF domain, a complex QMF domain, and a time domain by decoding downmix signal bitstream; andan MPEG surround decoder for generating multi-channel audio signals by using side information bitstream and the downmix signals,wherein the MPEG surround decoder synchronizes the downmix signals with the side information bitstream by adjusting delay according to a kind of the downmix signals outputted from the HE-AAC decoder.2. An apparatus for restoring multi-channel audio signals , comprising:an HE-AAC decoder for decoding downmix signal bitstream and outputting downmix signals of at least one among a real QMF domain, a complex QMF domain, and time domain; anda high-quality MPEG surround decoder for generating multi-channel audio signals by using side information bitstream and the downmix signals,wherein the high-quality MPEG surround decoder includes a delay unit added to a downmix signal path before a Nyquist analysis filter bank to ...

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Номер записи: 69
20-03-2014 дата публикации

AUDIO ENCODER, AUDIO DECODER, METHOD FOR ENCODING AN AUDIO INFORMATION, METHOD FOR DECODING AN AUDIO INFORMATION AND COMPUTER PROGRAM USING A DETECTION OF A GROUP OF PREVIOUSLY-DECODED SPECTRAL VALUES

Номер: US20140081645A1
Автор: Fuchs Guillaume, Gayer Marc, Griebel Christian, Multrus Markus, Rettelbach Nikolaus, Subbaraman Vignesh, Warmbold Patrick, Weiss Oliver
Принадлежит:

An audio decoder for providing a decoded audio information includes a arithmetic decoder for providing a plurality of decoded spectral values on the basis of an arithmetically-encoded representation of the spectral values and a frequency-domain-to-time-domain converter for providing a time-domain audio representation using the decoded spectral values. The arithmetic decoder is configured to select a mapping rule describing a mapping of a code value onto a symbol code in dependence on a context state. The arithmetic decoder is configured to determine or modify the current context state in dependence on a plurality of previously-decoded spectral values. The arithmetic decoder is configured to detect a group of a plurality of previously-decoded spectral values, which fulfill, individually or taken together, a predetermined condition regarding their magnitudes, and to determine the current context state in dependence on a result of the detection. 1. An audio decoder for providing a decoded audio information on the basis of an encoded audio information , the audio decoder comprising:an arithmetic decoder for providing a plurality of decoded spectral values on the basis of an arithmetically-encoded representation of the spectral values; anda frequency-domain-to-time-domain converter for providing a time-domain audio representation using the decoded spectral values, in order to acquire the decoded audio information;wherein the arithmetic decoder is configured to select a mapping rule describing a mapping of a code value onto a symbol code in dependence on a context state; andwherein the arithmetic decoder is configured to determine the current context state in dependence on a plurality of previously-decoded spectral values,wherein the arithmetic decoder is configured to detect a group of a plurality of previously-decoded spectral values, which fulfill, individually or taken together, a predetermined condition regarding their magnitudes, and to determine or modify the ...

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Номер записи: 70
27-03-2014 дата публикации

SYSTEMS, METHODS, APPARATUS, AND COMPUTER-READABLE MEDIA FOR THREE-DIMENSIONAL AUDIO CODING USING BASIS FUNCTION COEFFICIENTS

Номер: US20140086416A1
Автор: Sen Dipanjan
Принадлежит: QUALCOMM INCORPORATED

Systems, methods, and apparatus for a unified approach to encoding different types of audio inputs are described. 1. An apparatus comprising:a unified encoder configured to produce a unified encoded signal; anda memory configured to store the unified encoded signal.2. The apparatus of claim 1 , wherein the unified encoder comprises:a spherical harmonic analysis module configured to transform an audio coded signal into a first spherical harmonic (SH) based coded signal;a combiner configured to combine the first SH based coded signal with a second SH based coded signal to obtain a combined SH based coded signal; anda unified coefficient set encoder configured to produce the unified encoded signal based on the combined SH based coded signal.3. The apparatus of claim 2 , wherein the unified encoder further comprises:a format detector configured to determine a format of the audio coded signal and producing a corresponding format indicator.4. The apparatus of claim 2 , wherein the audio coded signal is a channel-based audio coded signal.5. The apparatus of claim 2 , wherein the audio coded signal is an object-based audio coded signal.6. The apparatus of claim 2 , wherein the unified encoder further comprises a second spherical harmonic analysis module configured to transform a second audio coded signal into the second spherical harmonic (SH) based coded signal.7. A method comprising:producing, with a unified encoder, a unified encoded signal; andstoring, with a memory, the unified encoded signal.8. The method of claim 7 , wherein producing the unified encoded signal comprises:transforming an audio coded signal into a first spherical harmonic (SH) based coded signal;combining the first SH based coded signal with a second SH based coded signal to generate a combined SH based coded signal; andproducing the unified encoded signal based on the combined SH based coded signal.9. A method of audio signal processing claim 7 , said method comprising:receiving a plurality of ...

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Номер записи: 71
27-03-2014 дата публикации

AUDIO DECODING METHOD AND APPARATUS

Номер: US20140088976A1
Автор: LIU Zhihui, Zhao Yunxuan
Принадлежит: Huawei Device Co.,Ltd.

An audio decoding method and apparatus are disclosed. The audio decoding method includes: receiving data packets; when data packet loss is detected and audio data of an audio frame corresponding to M channels in N channels is lost, if audio data of other channels than the M channels in the N channels, which belongs to the same audio frame as the lost audio data in the audio frame, is not lost, decoding the un-lost audio data; extracting a signal characteristic parameter of the data obtained after decoding; determining whether a correlation exists between a first channel and a second channel; and if the correlation exists, performing packet loss concealment processing on the lost audio data of the audio frame corresponding to the first channel according to the second channel. The audio decoding method and apparatus can effectively improve the effect of packet loss concealment processing in audio decoding. 1. A method for decoding audio data transmitted over multiple channels , the method being performed by an audio decoding apparatus and comprising:receiving audio data packets over the multiple channels, wherein the audio data packets are divided into audio frames, and data of each audio frame are distributed over the multiple channels;determining, according to the received data packets, that a first portion of data of an audio frame carried by a first channel of the multiple channels is lost, and that a second portion of data of the audio frame carried by a second channel of the multiple channels is received by the audio decoded apparatus;decoding the second portion of data;extracting, by the audio decoding apparatus, a signal characteristic parameter of the decoded second portion of data;determining that a data correlation exists between the first channel and the second channel;calculating a signal characteristic parameter of the first portion of data according to the signal characteristic parameter of the second portion of data; andreconstructing the first portion ...

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Номер записи: 72
27-03-2014 дата публикации

METHOD AND APPARATUS FOR ENCODING AND DECODING STEREO SIGNAL AND MULTI-CHANNEL SIGNAL

Номер: US20140088977A1
Автор: CHOO Ki-hyun, Kim Jung-hoe, KUDRYASHOV Boris, Oh Eun-mi, Petrov Sergey
Принадлежит: SAMSUNG ELECTRONICS CO., LTD.

Provided are a method and apparatus for encoding and decoding a stereo signal or a multi-channel signal. According to the method and apparatus, a stereo signal or a multi-channel signal can be encoded and/or decoded by generating parameters based on a mono signal. 1. An apparatus for generating a stereo signal , the apparatus comprising:a mono signal decoding unit to receive a down-mixed mono signal and decode the down-mixed mono signal;a parameter decoding unit to decode at least one parameter that represents characteristics between channels of the stereo signal; andan up-mixing unit to up-mix the decoded down-mixed mono signal by using the at least one decoded parameter, to generate the stereo signal.2. The apparatus of claim 1 , wherein the up-mixing unit performs up-mixing of the decoded down-mixed mono signal by using a decorrelated signal.3. The apparatus for generating a stereo signal claim 1 , the apparatus comprising:a mono signal decoding unit to receive a down-mixed mono signal and decode the down-mixed mono signal;a parameter decoding unit to decode at least one parameter that represents characteristics between channels of the stereo signal;a parameter generating unit to generate a parameter representing a phase difference between one of the stereo signal and the down-mixed mono signal; andan up-mixing unit to up-mix the decoded down-mixed mono signal by using the at least one decoded parameter and the generated parameter, to generate the stereo signal.4. The apparatus of claim 3 , wherein the up-mixing unit performs up-mixing of the decoded down-mixed mono signal by using a decorrelated signal. This application is a Continuation application of prior application Ser. No. 13/366,455, filed on Feb. 6, 2012, which is a divisional application of U.S. Ser. No. 11/876,947, filed Oct. 23, 2007, now U.S. Pat. No. 8,111,829, which claims the benefit of Korean Patent Application No. 10-2007-0037165, filed on Apr. 16, 2007, in the Korean Intellectual Property ...

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Номер записи: 73
03-04-2014 дата публикации

Audio Encoding Method and Apparatus, Audio Decoding Method and Apparatus, and Encoding/Decoding System

Номер: US20140093086A1
Автор: Zhang Jinliang, Zhao Yunxuan
Принадлежит: Huawei Technologies Co., Ltd.

Embodiments of the present invention disclose an audio encoding method. The method includes: obtaining audio data of N channels; and performing channel interleaving and packetization on the obtained audio data of the N channels to obtain data packets, where each data packet includes X*N segments of audio data, where X is a ratio of an amount of audio data included in one data packet to an amount of audio data included in one audio frame, X is an integer greater than or equal to 1, and in the X*N segments of audio data, at least X+1 segments of audio data belong to different audio frames. 1. An audio encoding method , applicable to an audio encoding/decoding system comprising N channels , wherein N is an integer greater than or equal to two , wherein the method comprises:obtaining audio data of the N channels; andperforming channel interleaving and packetization on the obtained audio data of the N channels to obtain data packets,wherein each data packet comprises X*N segments of audio data,wherein X is a ratio of an amount of audio data comprised in one data packet to an amount of audio data comprised in one audio frame,wherein X is an integer greater than or equal to one, andwherein, in the X*N segments of audio data, at least X+1 segments of audio data belong to different audio frames.2. The method according to claim 1 , wherein performing channel interleaving and packetization on the obtained audio data of the N channels to obtain the data packets comprises:{'sup': th', 'th', 'th', 'th, 'composing a data packet by using audio data of an mchannel in an haudio frame and audio data of channels other than an mchannel in an iaudio frame; and'}{'sup': th', 'th', 'th', 'th, 'composing another data packet by using audio data of the mchannel in the iaudio frame and audio data of channels other than the mchannel in the haudio frame,'}wherein h and i are different audio frame numbers, andwherein m is any integer from one to N.3. The method according to claim 2 , wherein the ...

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Номер записи: 74
03-04-2014 дата публикации

APPARATUS AND METHOD FOR CODING AND DECODING MULTI-OBJECT AUDIO SIGNAL WITH VARIOUS CHANNEL

Номер: US20140095178A1
Автор: Beack Seung-Kwon, HONG Jin-Woo, JANG Dae-Young, Jang In-Seon, KANG Kyeong-Ok, KIM Jin-Woong, LEE Tae-Jin, LEE Yong-Ju, SEO Jeong-Il, Yoo Jae-Hyoun
Принадлежит: Electronics and Telecommunications Research Institute

Provided are an apparatus and method for coding and decoding a multi-object audio signal. The apparatus includes a down-mixer for down-mixing the audio signals into one down-mixed audio signal and extracting supplementary information including header information and spatial cue information for each of the audio signals, a coder for coding the down-mixed audio signal, and a supplementary information coder for generating the supplementary information as a bit stream. The header information includes identification information for each of the audio signals and channel information for the audio signals. 1. An apparatus for decoding multi-object audio signals having different channels , comprising:a supplementary information control means for controlling supplementary information extracted from input signal, using control information for downmix audio signal restored from the input signal, wherein the control information includes rendering control information for the restored downmix audio signal; andan output means for outputting the restored downmix audio signal as multi-channel audio signal, using the supplementary information controlled by the supplementary information control means, whereinthe supplementary information includes preset information for the multi-object audio signals.2. The apparatus of claim 1 , wherein the preset information includes:preset mode information for defining a preset mode for the audio signals; andpreset mode support information for defining information required for supporting the preset mode.3. The apparatus of claim 1 , wherein the supplementary information further includes:identification information for each of the audio signals; andchannel information for the audio signals.4. The apparatus of claim 3 , wherein the channel information includes:channel information for each of the audio signals; andinformation of a number of audio objects for each channel of the audio signals.5. The apparatus of claim 1 , wherein the supplementary ...

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Номер записи: 75
03-04-2014 дата публикации

APPARATUS AND METHOD FOR CODING AND DECODING MULTI-OBJECT AUDIO SIGNAL WITH VARIOUS CHANNEL

Номер: US20140095179A1
Автор: Beack Seung-Kwon, HONG Jin-Woo, JANG Dae-Young, Jang In-Seon, KANG Kyeong-Ok, KIM Jin-Woong, LEE Tae-Jin, LEE Yong-Ju, SEO Jeong-Il, Yoo Jae-Hyoun
Принадлежит: Electronics and Telecommunications Research Institute

Provided are an apparatus and method for coding and decoding a multi-object audio signal. The apparatus includes a down-mixer for down-mixing the audio signals into one down-mixed audio signal and extracting supplementary information including header information and spatial cue information for each of the audio signals, a coder for coding the down-mixed audio signal, and a supplementary information coder for generating the supplementary information as a bit stream. The header information includes identification information for each of the audio signals and channel information for the audio signals. 1. An apparatus for decoding multi-object audio signals having different channels , comprising:a supplementary information control means for controlling supplementary information extracted from input signal, using control information for downmix audio signal restored from the input signal, wherein the control information includes rendering control information for the restored downmix audio signal; andan output means for outputting the restored downmix audio signal as multi-channel audio signal, using the supplementary information controlled by the supplementary information control means, whereinthe supplementary information includes spatial cue information for audio object of one of mono channel, stereo channel, and multi-channel of the multi-object audio signal.2. The apparatus of claim 1 , wherein the supplementary information further includes preset information for the audio signals.3. The apparatus of claim 2 , wherein the preset information includes:preset mode information for defining a preset mode for the audio signals; andpreset mode support information for defining information required for supporting the preset mode.4. The apparatus of claim 1 , wherein the supplementary information further includes:identification information for each of the audio signals; andchannel information for the audio signals.5. The apparatus of claim 4 , wherein the channel information ...

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Номер записи: 76
10-04-2014 дата публикации

RECORDING APPARATUS WITH MASTERING FUNCTION

Номер: US20140098961A1
Автор: Adachi Shigeyuki
Принадлежит: TEAC CORPORATION

There is provided a recording apparatus which can effectively execute compression processing regardless of the mixdown level, and which can thereby carry out the mastering process easily. A DSP mixes down audio signals assigned to a plurality of tracks into a stereo audio signal. When performing the mastering process, the DSP normalizes the stereo audio signal before performing the compression processing, subsequently carries out the compression processing, then performs normalization again so as to produce master data, and records the master data in a recorder. 1. A recording apparatus for recording an audio signal , comprising:a pre-normalize unit for amplifying a level of an audio signal to a target reference level; anda compression unit for compressing, from among levels of the audio signal that has been processed by the pre-normalize unit, a level that exceeds a predetermined threshold value.2. The recording apparatus according to claim 1 , further comprisinga mixdown unit for producing a stereo audio signal from audio signals assigned to respective ones of a plurality of tracks,wherein the pre-normalize unit amplifies a level of the stereo audio signal to the reference level.3. The recording apparatus according to claim 1 , further comprising a post-normalize unit for amplifying a level of the audio signal that has been processed by the compression unit to the reference level.4. The recording apparatus according to claim 1 , wherein the compression unit compresses claim 1 , from among the levels of the audio signal that has been processed by the pre-normalize unit claim 1 , the level that exceeds the predetermined threshold value claim 1 , and simultaneously amplifies a level of the audio signal to the reference level.5. The recording apparatus according to claim 2 , further comprisinga filter unit for removing signals in a predetermined frequency band from the stereo signal produced in the mixdown unit, whereinthe pre-normalize unit detects a peak level of ...

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Номер записи: 77
10-04-2014 дата публикации

PARAMETRIC ENCODER FOR ENCODING A MULTI-CHANNEL AUDIO SIGNAL

Номер: US20140098963A1
Автор: Lang Yue, Virette David, Xu Jianfeng
Принадлежит: Huawei Technologies Co., Ltd.

The invention relates to a parametric audio encoder, comprising a parameter generator, the parameter generator being configured to determine a first set of encoding parameters and reference audio signal values, wherein the reference audio signal is another audio channel signal or a downmix audio signal derived from at least two audio channel signals of the plurality of multi-channel audio signals, to determine a first encoding parameter average based on the first set of encoding parameters of the audio channel signal, to determine a second encoding parameter average based on the first encoding parameter average of the audio channel signal and at least one other first encoding parameter average of the audio channel signal, and to determine the encoding parameter based on the first encoding parameter average of the audio channel signal and the second encoding parameter average of the audio channel signal. 1. A parametric audio encoder for generating an encoding parameter (ICC) for an audio channel signal (X[b]) of a plurality of audio channel signals (X[b] , X[b]) of a multi-channel audio signal , each audio channel signal (X[b] , X[b]) having audio channel signal values (X[k] , X[k]) , the parametric audio encoder comprising: [{'sub': 1', '1', '1', '2', '2', '2, 'determine for the audio channel signal (X[b]) of the plurality of audio channel signals a first set of encoding parameters (IPD[b]) from the audio channel signal values (X[k]) of the audio channel signal (X[b]) and reference audio signal values (X[k]) of a reference audio signal (X[b]), wherein the reference audio signal is another audio channel signal (X[b]) of the plurality of audio channel signals or a downmix audio signal derived from at least two audio channel signals of the plurality of multi-channel audio signals,'}, {'sub': 1', 'mean', '1, 'determine for the audio channel signal (X[b]) a first encoding parameter average (IPD[i]) based on the first set of encoding parameters (IPD[b]) of the audio ...

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Номер записи: 78
10-04-2014 дата публикации

APPARATUS AND METHOD FOR CODING AND DECODING MULTI OBJECT AUDIO SIGNAL WITH MULTI CHANNEL

Номер: US20140100856A1
Автор: Beack Seung-Kwon, HONG Jin-Woo, JANG Dae-Young, KANG Kyeong-Ok, KIM Jin-Woong, LEE Tae-Jin, SEO Jeong-Il
Принадлежит: Electronics and Telecommunications Research Institute

Provided are an apparatus and method for coding and decoding a multi object audio signal with multi channel. The apparatus includes: a multi channel encoding means for down-mixing an audio signal including a plurality of channels, generating a spatial cue for the audio signal including the plurality of channels, and generating first rendering information including the generated spatial cue; and a multi object encoding unit for down-mixing an audio signal including a plurality of objects, which includes the down-mixed signal from the multi channel encoding unit, generating a spatial cue for the audio signal including the plurality of objects, and generating second rendering information including the generated spatial cue, wherein the multichannel encoding unit generates a spatial cue for the audio signal including the plurality of objects regardless of a Coder-DECoder (CODEC) scheme the limits the multi channel encoding unit. 1. A transcoding apparatus for generating rendering information to decode an encoded audio signal , comprising:a first matrix means for generating rendering information including information for mapping the encoded audio signal to an output channel of an audio decoding apparatus based on object control information including location and level information of the encoded audio signal and output layout information;a second matrix means for generating channel restoration information for a audio signal including a plurality of channels included in the encoded audio signal based on first rendering information including a spatial cue for the audio signal;a sub-band converting means for converting second rendering information having a spatial cue for an audio signal including a plurality of objects included in the encoded audio signal into rendering information following the CODEC scheme, where the second rendering information includes a spatial cue not limited by a CODEC scheme that limits the first rendering information; andrendering means for ...

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Номер записи: 79
05-01-2017 дата публикации

GENERATING AN AUDIO SIGNAL WITH A CONFIGURABLE DISTANCE CUE

Номер: US20170001561A1
Автор: Christoph Markus
Принадлежит:

Generating an audio signal with a configurable distance cue includes the following: receiving an input audio signal that has at least one signal characteristic that contributes to a listener's auditory distance perception; receiving a distance control signal; and changing at least one characteristic in accordance with a distance control signal that is representative of the perceived distance. 1. A system for generating an audio signal with a configurable distance cue comprising a signal characteristic modification module configured to receive an input audio signal that has at least one signal characteristic; the at least one signal characteristic contributes to a listener's auditory distance perception and changes at least one characteristic in accordance with a distance control signal that is representative of a perceived distance.2. The system of claim 1 , where the at least one signal characteristic affects at least one of phase coherence of harmonics claim 1 , direct/reverberant ratio claim 1 , loudness claim 1 , sound spectrum claim 1 , initial time delay gap claim 1 , movement and level difference.3. The system of claim 2 , where the input audio signal comprises a multiplicity of harmonics and the signal characteristic modification module comprises:a harmonic extraction module configured to receive the input audio signal and to extract the harmonics of the input audio signal to provide signals representative of the harmonics of the input audio signal, each of the signals being representative of the harmonics having respective frequencies and phases;a phase shifting module configured to change the phases of the signals representative of the harmonics in accordance with the distance control signal; anda summer module configured to sum up the signals representative of the harmonics to provide an output audio signal.4. The system of claim 3 , where the signal characteristic modification module further comprises a phase coherence modification module configured to ...

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Номер записи: 80
06-01-2022 дата публикации

AN AUDIO ENCODER AND AN AUDIO DECODER

Номер: US20220005484A1
Автор: FRIEDRICH Tobias, GORLOW Stanislaw, MERPILLAT Celine, Purnhagen Heiko
Принадлежит: DOLBY INTERNATIONAL AB

The present disclosure relates to the field audio coding, an in particular to an audio decoder having at least two decoding modes, and associated decoding methods and decoding software for such audio decoder. In one of the decoding modes, at least one dynamic audio object is mapped to a set of static audio objects, the set of static audio objects corresponding to a predefined speaker configuration. The present disclosure further relates to a corresponding audio encoder, and associated encoding methods and encoding software for such audio encoder. 1. An audio decoder comprising:one or more buffers for storing a received audio bitstream; and to operate in a decoding mode selected from a plurality of different decoding modes for decoding the received audio bitstream into one or more dynamic or static audio objects, a dynamic or static audio object comprising an audio signal associated with either a time-varying or a static spatial position, the plurality of different decoding modes comprising a first decoding mode and a second decoding mode, wherein of the first and second decoding modes only the first decoding mode allows full decoding of one or more encoded dynamic audio objects in the bitstream, into reconstructed individual audio objects; and', to access the received audio bitstream;', 'to determine whether the received audio bitstream includes one or more dynamic audio objects; and', 'responsive at least to determining that the received audio bitstream includes one or more dynamic audio objects, to map at least one of the one or more dynamic audio objects to a set of static audio objects, the set of static audio objects corresponding to a predefined immersive speaker configuration., 'when the selected decoding mode is the second decoding mode], 'a controller coupled to the one or more buffers and configured2. The audio decoder of claim 1 , wherein when the selected decoding mode is the second decoding mode claim 1 , the controller is further configured to render ...

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Номер записи: 81
01-01-2015 дата публикации

NAVIGATION WITH THREE DIMENSIONAL AUDIO EFFECTS

Номер: US20150003616A1
Автор: Chudge Jarnail, McCARTHY Stuart, MIDDLEMISS Simon, Miller Amos, Tsikkos Michael
Принадлежит:

Mechanisms for navigation via three dimensional audio effects are described. A current location of a device and a first point of interest are determined. The point of interest may be determined based on a web service and the current location of the device may be determined via mobile device signals. A zone that includes the point of interest may be determined. A three dimensional audio effect that simulates a sound being emitted from the zone may be generated. The three dimensional audio effect may be transmitted to speakers capable of simulating three dimensional audio effects. The transmitted three dimensional audio effect may aid in navigation from a current location to the point of interest. 1. A computer-implemented method comprising:determining a current location of a device;determining a location of a first point of interest;determining a zone, the zone including the location of the first point of interest;generating a three dimensional audio effect, the three dimensional audio effect to represent a sound emitted from a point in the zone;sending the three dimensional audio effect.2. The computer-implemented method of claim 1 , wherein: sending a web service request with an identifier associated with the first point of interest; and', 'receiving a location associated with the first point of interest., 'determining a location of the first point of interest comprises3. The computer-implemented method of claim 1 , wherein:determining a current location of the device comprises analyzing mobile device signals.4. The computer-implemented method of claim 1 , wherein:sending the three dimensional audio effect further comprises sending a web service message, the web service message including the three dimensional audio effect.5. The computer-implemented method of claim 1 , further comprising:determining a distance between the current location of the device and the first point of interest;wherein the three dimensional audio effect further simulates sound emitted from ...

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Номер записи: 82
07-01-2021 дата публикации

LOUDNESS CONTROL FOR USER INTERACTIVITY IN AUDIO CODING SYSTEMS

Номер: US20210004204A1
Автор: Kratschmer Michael, KÜCH Fabian, MEIER Michael, NEUGEBAUER Bernhard, UHLE Christian
Принадлежит:

An audio processor for processing an audio signal includes: an audio signal modifier for modifying the audio signal in response to a user input; a loudness controller for determining a loudness compensation gain based on a reference loudness or a reference gain and a modified loudness or a modified gain, where the modified loudness or the modified gain depends on the user input; and a loudness manipulator for manipulating a loudness of a signal using the loudness compensation gain. 1. An audio processor for processing an audio signal comprising metadata , the audio processor comprising:an audio signal modifier, wherein the audio signal modifier is configured to modify the audio signal in response to a user input via amplifying or attenuating a group comprising one or more audio elements belonging to the audio signal and according to a selected or a default preset covered by the metadata;a loudness controller, wherein the loudness controller is configured to determine a loudness compensation gain based on the one hand on a reference loudness or a reference gain corresponding to an original audio scene and on the other hand on a modified loudness or a modified gain, wherein the modified loudness or the modified gain depends on the user input, anda loudness manipulator, wherein the loudness manipulator is configured to manipulate a loudness of a signal using the loudness compensation gain and using the metadata,wherein the loudness controller is configured to determine the loudness compensation gain based on the metadata of the audio signal indicating which group is to be used for determining the loudness compensation gain, or indicating which group is not to be used for determining the loudness compensation gain, andwherein a group comprises one or more audio elements, or wherein the loudness controller is configured to determine the loudness compensation gain based on the metadata of the audio signal referring to at least one preset, wherein the preset refers to a ...

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Номер записи: 83
07-01-2016 дата публикации

Methods for Audio Signal Transient Detection and Decorrelation Control

Номер: US20160005405A1
Автор: Grant A. Davidson, Kuan-Chieh Yen, Vinay Melkote
Принадлежит: Dolby Laboratories Licensing Corp

Some audio processing methods may involve receiving audio data corresponding to a plurality of audio channels and determining audio characteristics of the audio data, which may include transient information. An amount of decorrelation for the audio data may be based, at least in part, on the audio characteristics. If a definite transient event is determined, a decorrelation process may be temporarily halted or slowed. Determining transient information may involve evaluating the likelihood and/or the severity of a transient event. In some implementations, determining transient information may involve evaluating a temporal power variation in the audio data. Explicit transient information may or may not be received with the audio data, depending on the implementation. Explicit transient information may include a transient control value corresponding to a definite transient event, a definite non-transient event or an intermediate transient control value.

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Номер записи: 84
07-01-2016 дата публикации

Methods for Controlling the Inter-Channel Coherence of Upmixed Audio Signals

Номер: US20160005406A1
Автор: DAVIDSON Grant A., FELLERS Matthew, MELKOTE Vinay, YEN Kuan-Chieh
Принадлежит: DOLBY LABORATORIES LICENSING CORPORATION

Audio characteristics of audio data corresponding to a plurality of audio channels may be determined. The audio characteristics may include spatial parameter data. Decorrelation filtering processes for the audio data may be based, at least in part, on the audio characteristics. The decorrelation filtering processes may cause a specific inter-decorrelation signal coherence (“IDC”) between channel-specific decorrelation signals for at least one pair of channels. The channel-specific decorrelation signals may be received and/or determined. Inter-channel coherence (“ICC”) between a plurality of audio channel pairs may be controlled. Controlling ICC may involve at receiving an ICC value and/or determining an ICC value based, at least partially, on the spatial parameter data. A set of IDC values may be based, at least partially, on the set of ICC values. A set of channel-specific decorrelation signals, corresponding with the set of IDC values, may be synthesized by performing operations on the filtered audio data. 1100-. (canceled)101. A method , comprising:receiving audio data corresponding to a plurality of audio channels;determining audio characteristics of the audio data, the audio characteristics including spatial parameter data and at least one of tonality information or transient information;determining at least two channel-specific decorrelation filtering processes for the audio data based, at least in part, on the tonality information or the transient information, the channel-specific decorrelation filtering processes causing a specific inter-decorrelation signal coherence (“IDC”), which is a measure of correlation between decorrelation signals, between channel-specific decorrelation signals for at least one pair of channels, each of the channel-specific decorrelation filtering processes comprising applying a decorrelation filter to at least a portion of a corresponding audio channel of the audio data to produce filtered audio data, the channel-specific ...

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Номер записи: 85
07-01-2016 дата публикации

Methods for Parametric Multi-Channel Encoding

Номер: US20160005407A1
Автор: FRIEDRICH Tobias, Linzmeier Karsten, Mueller Alexander, SPENGER Claus-Christian, WAGENBLASS Tobias R.
Принадлежит: DOLBY INTERNATIONAL AB

The present document relates to audio coding systems. In particular, the present document relates to efficient methods and systems for parametric multi-channel audio coding. An audio encoding system () configured to generate a bitstream () indicative of a downmix signal and spatial metadata for generating a multi-channel upmix signal from the downmix signal is described. The system () comprises a downmix processing unit () configured to generate the downmix signal from a multi-channel input signal (); wherein the downmix signal comprises m channels and wherein the multi-channel input signal () comprises n channels; n, m being integers with m Подробнее

Номер записи: 86
07-01-2016 дата публикации

THREE-DIMENSIONAL SOUND COMPRESSION AND OVER-THE-AIR-TRANSMISSION DURING A CALL

Номер: US20160005408A1
Автор: Kim Lae-Hoon, Visser Erik, Xiang Pei
Принадлежит:

A method for encoding three dimensional audio by a wireless communication device is disclosed. The wireless communication device detects an indication of a plurality of localizable audio sources. The wireless communication device also records a plurality of audio signals associated with the plurality of localizable audio sources. The wireless communication device also encodes the plurality of audio signals. 1. A method performed by a wireless communication device , comprising:determining an energy profile of a plurality of audio signals;displaying the energy profiles of each of the plurality of audio signals;detecting an input that selects an energy profile;associating a codec with the input; andincreasing bit allocation to the codec used to compress audio signals based on the input.2. The method of claim 1 , further comprising transmitting the packet over the air.3. The method of claim 1 , further comprising transmitting a channel identification.4. The method of claim 1 , wherein compression of the audio signals results in four packets being transmitted over the air.5. The method of claim 1 , wherein displaying the energy profiles comprises displaying a first energy profile representing a front left audio signal and displaying a second energy profile representing a front right audio signal.6. The method of claim 1 , wherein displaying the energy profiles comprises displaying a first energy profile representing a back left audio signal and displaying a second energy profile representing a back right audio signal.7. The method of claim 1 , wherein the energy profiles are associated with a standard channel format surround sound system.8. A wireless communication device claim 1 , comprising: determine an energy profile of a plurality of audio signals;', 'detect an input that selects an energy profile;', 'associate a codec with the input; and', 'compress the plurality of audio signals based on the codec to generate a packet; and, 'a processor configured toa display ...

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Номер записи: 87
07-01-2021 дата публикации

USING METADATA TO AGGREGATE SIGNAL PROCESSING OPERATIONS

Номер: US20210005211A1
Автор: Boehm Reinhold, Leddy Mark C., Linzmeier Karsten, Mathew Vinay, Plain Simon, Purnhagen Heiko, Sehlstrom Leif, Stahlmann Alexander, Thesing Robin
Принадлежит: DOLBY INTERNATIONAL AB

A technique including receiving and decoding a coded bitstream encoded with audio content including first audio objects corresponding to a first media content type of two consecutive media content types and second audio objects corresponding to a second media content type of the two consecutive media content types, and audio metadata corresponding to the audio content. The audio metadata including first and second audio object gains, for the first and second audio objects, generated in part based on a first fading curve of the first media content type and a second fading curve of the second media content type, respectively. The technique further includes applying the first and second audio object gains to the first and second audio objects, and rendering a sound field represented by the first audio object with the applied first audio object gain and the second audio object with the applied second audio object gain. 1. A method , performed by a downstream audio rendering stage in an end-to-end audio processing chain , comprising:receiving and decoding a coded bitstream generated by an upstream audio processor, wherein the coded bitstream is encoded with audio content and audio metadata corresponding to the audio content;wherein the audio content includes first audio objects corresponding to a first media content type of two consecutive media content types and second audio objects corresponding to a second media content type of the two consecutive media content types;wherein the audio metadata includes first and second audio object gains, respectively for the first and second audio objects, generated at least in part based on a first fading curve of the first media content type and a second fading curve of the second media content type, respectively;applying the first and second audio object gains generated at least in part based on the first and second fading curves to the first and second audio objects, respectively;rendering a sound field represented by the first ...

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Номер записи: 88
04-01-2018 дата публикации

METHOD FOR ENCODING MULTI-CHANNEL AUDIO SIGNAL AND ENCODING DEVICE FOR PERFORMING ENCODING METHOD, AND METHOD FOR DECODING MULTI-CHANNEL AUDIO SIGNAL AND DECODING DEVICE FOR PERFORMING DECODING METHOD

Номер: US20180005635A1
Автор: BEACK Seung Kwon, Choi Jin Soo, Lee Tae Jin, SEO Jeong Il, SUNG Jong Mo
Принадлежит: Electronics and Telecommunications Research Institute

An encoding method for a multi-channel audio signal, an encoding apparatus for performing the encoding method, and a decoding method for a multi-channel audio signal and a decoding apparatus for performing the decoding method are disclosed. A method and apparatus of bypassing an MPEG Surround (MPS) standard operation and using an arbitrary tree when a number of audio signals of N channels exceeds a channel number defined in an MPS standard, is disclosed. 1. An encoding method for a multi-channel audio signal , the method comprising:generating audio signals of N/2 channels by downmixing audio signals of N channels using an MPEG Surround (MPS) encoder; andperforming encoding with respect to a core band of the audio signals of the N/2 channels using a Unified Speech and Audio Codec (USAC) encoder.2. The method of claim 1 , wherein the generating of the audio signals of the N/2 channels comprises generating the audio signals of the N/2 channels by downmixing the audio signals of the N channels using N/2 two-to-one (TTO) coding modules.3. The method of claim 1 , further comprising:converting a sampling rate with respect to an audio signal using a sampling rate converter,wherein the sampling rate converter is disposed before the MPS encoder to convert a sampling rate of the audio signals of the N channels, or disposed after the MPS encoder to convert a sampling rate of the audio signals of the N/2 channels.4. The method of claim 3 , wherein the converting of the sampling rate comprises converting the sampling rate with respect to the audio signal according to a bit rate to be applied to the USAC encoder.5. The method of claim 1 , wherein the generating of the audio signals of the N/2 channels comprises generating the audio signals of the N/2 channels by downmixing the audio signals of the N channels using an arbitrary tree when a number of the N channels exceeds a channel number defined by an MPS standard.6. The method of claim 1 , wherein the generating of the audio ...

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Номер записи: 89
04-01-2018 дата публикации

TRANSMISSION DEVICE, TRANSMISSION METHOD, RECEIVING DEVICE, AND RECEIVING METHOD

Номер: US20180005640A1
Автор: Tsukagoshi Ikuo
Принадлежит: SONY CORPORATION

It is attempted to reduce the processing load of a receiver at the time of integrating plural audio streams. 1. A transmission device comprising:an encoding unit configured to generate a predetermined number of audio streams; anda transmission unit configured to transmit a container of a predetermined format including the predetermined number of audio streams,wherein the audio streams are constituted by an audio frame including a first packet that includes encoded data as payload information and a second packet that includes configuration information representing a configuration of the payload information of the first packet as payload information, andcommon index information is inserted in payloads of the first packet and the second packet that are related.2. The transmission device according to claim 1 , wherein the encoded data that the first packet include as payload information is encoded channel data or encoded object data.3. A transmission method comprising:an encoding step of generating a predetermined number of audio streams; anda transmission step of using a transmission unit to transmit a container of a predetermined format including the predetermined number of audio streams,wherein the audio streams are constituted by an audio frame including a first packet that includes encoded data as payload information and a second packet that includes configuration information representing a configuration of the payload information of the first packet as payload information,and common index information is inserted in payloads of the first packet and the second packet that are related.4. A receiving device comprising:a receiving unit configured to receive a container of a predetermined format including a predetermined number of audio streams,wherein the audio streams are constituted by an audio frame including a first packet that includes encoded data as payload information and a second packet that includes configuration information representing a configuration of ...

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Номер записи: 90
04-01-2018 дата публикации

Method for decoding a higher order ambisonics (hoa) representation of a sound or soundfield

Номер: US20180005641A1
Автор: Alexander Krueger, Sven Kordon
Принадлежит: Dolby Laboratories Licensing Corp

When compressing an HOA data frame representation, a gain control (15, 151) is applied for each channel signal before it is perceptually encoded (16). The gain values are transferred in a differential manner as side information. However, for starting decoding of such streamed compressed HOA data frame representation absolute gain values are required, which should be coded with a minimum number of bits. For determining such lowest integer number (β e ) of bits the HOA data frame representation (C(k)) is rendered in spatial domain to virtual loudspeaker signals lying on a unit sphere, followed by normalisation of the HOA data frame representation (C(k)). Then the lowest integer number of bits is set to β e =┌log 2 (┌log 2 (√{square root over ( K MAX )}·O)┐+ 1 )┐.

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Номер записи: 91
02-01-2020 дата публикации

INTEGRATED RECONSTRUCTION AND RENDERING OF AUDIO SIGNALS

Номер: US20200005801A1
Автор: FRIEDRICH Tobias, PEICHL Klaus, Purnhagen Heiko, Thesing Robin, WOLTERS Martin
Принадлежит: DOLBY INTERNATIONAL AB

A method for rendering an audio output based on an audio data stream including M audio signals, side information including a series of reconstruction instances of a reconstruction matrix C and first timing data, the side information allowing reconstruction of N audio objects from the M audio signals, and object metadata defining spatial relationships between the N audio objects. The method includes generating a synchronized rendering matrix based on the object metadata, the first timing data, and information relating to a current playback system configuration, the synchronized rendering matrix having a rendering instance for each reconstruction instance, multiplying each reconstruction instance with a corresponding rendering instance to form a corresponding instance of an integrated rendering matrix, and applying the integrated rendering matrix to the audio signals in order to render an audio output. 1. A method for rendering an audio output based on an audio data stream , comprising: M audio signals which are combinations of N audio objects, wherein N>1 and M≤N,', {'sub': 'i', 'side information including a series of reconstruction instances cof a reconstruction matrix C and first timing data defining transitions between said instances, said side information allowing reconstruction of the N audio objects from the M audio signals, and'}, {'sub': 'i', 'time-variable object metadata including a series of metadata instances mdefining spatial relationships between the N audio objects and second timing data defining transitions between said metadata instances;'}], 'receiving a data stream including{'sub': sync', 'sync', 'i', 'i, 'generating a synchronized rendering matrix Rbased on the object metadata, the first timing data, and information relating to a current playback system configuration, said synchronized rendering matrix Rhaving a rendering instance rcorresponding in time with each reconstruction instance c;'}{'sub': i', 'i, 'multiplying each reconstruction instance ...

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Номер записи: 92
03-01-2019 дата публикации

TRANSMISSION APPARATUS, TRANSMISSION METHOD, RECEPTION APPARATUS, AND RECEPTION METHOD

Номер: US20190005968A1
Автор: Ichimura Gen
Принадлежит:

To enable to favorably send a compressed digital audio signal at a high data rate. First, second, and third metadata are added to a compressed digital audio signal of a predetermined number of channels. The first metadata is metadata indicating a sending frequency of the compressed digital audio signal. The second metadata is metadata indicating a sampling frequency used for converting an uncompressed digital audio signal of each channel into an analog signal. The third metadata is metadata indicating a ratio of the sending frequency to the sampling frequency. The compressed digital audio signal provided with each type of the metadata is transmitted to an external device through a predetermined sending path. 1. A transmission apparatus comprising:a metadata addition unit that adds, to a compressed digital audio signal of a predetermined number of channels, first metadata indicating a sending frequency of the compressed digital audio signal, second metadata indicating a sampling frequency used for converting an uncompressed digital audio signal of each channel into an analog signal, and third metadata indicating a ratio of the sending frequency to the sampling frequency; anda transmission unit that transmits the compressed digital audio signal provided with each type of the metadata to an external device through a predetermined sending path.2. The transmission apparatus according to claim 1 , whereinthe transmission unit sequentially transmits the compressed digital audio signal on a basis of unit data, andthe metadata addition unit uses a predetermined bit area of a channel status of each block constituted by a predetermined number of continuous pieces of the unit data to add each type of the metadata.3. The transmission apparatus according to claim 1 , whereinan encoding system of the compressed digital audio signal is moving picture experts group-4 adaptive audio coding.4. The transmission apparatus according to claim 3 , whereinthe predetermined number of ...

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Номер записи: 93
03-01-2019 дата публикации

Apparatus and method for stereo filling in multichannel coding

Номер: US20190005969A1
Автор: Dick Sascha, Fueg Richard, HELMRICH Christian, NAGEL Frederik, Rettelbach Nikolaus, Schuh Florian
Принадлежит:

An apparatus for decoding an encoded multichannel signal of a current frame to obtain three or more current audio output channels is provided. A multichannel processor is adapted to select two decoded channels from three or more decoded channels depending on first multichannel parameters. Moreover, the multichannel processor is adapted to generate a first group of two or more processed channels based on the selected channels. A noise filling module is adapted to identify for at least one of the selected channels, one or more frequency bands, within which all spectral lines are quantized to zero, and to generate a mixing channel using, depending on side information, a proper subset of three or more previous audio output channels that have been decoded, and to fill the spectral lines of frequency bands, within which all spectral lines are quantized to zero, with noise generated using spectral lines of the mixing channel. 1. An apparatus for decoding a previous encoded multichannel signal of a previous frame to acquire three or more previous audio output channels , and for decoding a current encoded multichannel signal of a current frame to acquire three or more current audio output channels ,wherein the apparatus comprises an interface, a channel decoder, a multichannel processor for generating the three or more current audio output channels, and a noise filling module,wherein the interface is adapted to receive the current encoded multichannel signal, and to receive side information comprising first multichannel parameters,wherein the channel decoder is adapted to decode the current encoded multichannel signal of the current frame to acquire a set of three or more decoded channels of the current frame,wherein the multichannel processor is adapted to select a first selected pair of two decoded channels from the set of three or more decoded channels depending on the first multichannel parameters,wherein the multichannel processor is adapted to generate a first group of ...

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Номер записи: 94
03-01-2019 дата публикации

ENCODER AND ENCODING METHOD FOR MULTI-CHANNEL SIGNAL, AND DECODER AND DECODING METHOD FOR MULTI-CHANNEL SIGNAL

Номер: US20190005971A1
Автор: BEACK Seung Kwon, JANG Dae Young, KANG Kyeong Ok, Kim Jin Woong, Lee Tae Jin, SEO Jeong Il, SUNG Jong Mo
Принадлежит: Electronics and Telecommunications Research Institute

An encoder and an encoding method for a multi-channel signal, and a decoder and a decoding method for a multi-channel signal are disclosed. A multi-channel signal may be efficiently processed by consecutive downmixing or upmixing. 1. A method of processing a multi-channel signal , the method comprising:identifying the multi-channel signal having four channels horizontally or vertically distributed;performing a joint coding based on a QCE (Quadruple Channel Element) including two consecutive CPE (Channel Pair Element)s.2. The method of claim 1 , wherein the QCE is generated by combining joint stereo coding with complex stereo prediction in horizontal direction and MPEG Surround-based stereo tools in vertical direction.3. The method of claim 1 , wherein the joint coding performs swapping channel.4. The method of claim 1 , wherein the joint coding performs based on qceIndex including stereo CPE claim 1 , QCE without residual claim 1 , and QCE with residual.5. A device of processing a multi-channel signal performed by one or more processor claim 1 , the device comprising:one or more processors are configured to:decode first CPE (Channel Pair Element), and a second bitstream related to second CPE;output cplx_out_dmx_L which is first channel of first CPE after complex prediction stereo coding and cplx_out_dmx_R which is second channel of first CPE after complex prediction stereo coding;output mps_out_L_1 which is first output channel of first MPEG Surround, and mps_out_L_2 which is second output channel of first MPEG surround;output mps_out_R_1 which is first output channel of second MPEG Surround and mps_out_R_2 which is second output channel of second MPEG Surround;output sbr_out_L_1 which is first output channel of first Stereo SBR and sbr_out_R_1 which is second output channel of first Stereo SBR; andoutput sbr_out_L_2 which is first output channel of second Stereo SBR and sbr_out_R_2 which is second output channel of second Stereo SBR.6. The device of claim 5 , ...

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Номер записи: 95
03-01-2019 дата публикации

HIGH-BAND RESIDUAL PREDICTION WITH TIME-DOMAIN INTER-CHANNEL BANDWIDTH EXTENSION

Номер: US20190005973A1
Автор: Atti Venkatraman, CHEBIYYAM Venkata Subrahmanyam Chandra Sekhar
Принадлежит:

A method includes decoding a low-band portion of an encoded mid signal to generate a decoded low-band mid signal. The method also includes processing the decoded low-band mid signal to generate a low-band residual prediction signal and generating a low-band left channel and a low-band right channel based partially on the decoded low-band mid signal and the low-band residual prediction signal. The method further includes decoding a high-band portion of the encoded mid signal to generate a time-domain decoded high-band mid signal and processing the time-domain decoded high-band mid signal to generate a time-domain high-band residual prediction signal. The method also includes generating a high-band left channel and a high-band right channel based on the time-domain decoded high-band mid signal and the time-domain high-band residual prediction signal. 1. A device comprising:a low-band mid signal decoder configured to decode a low-band portion of an encoded mid signal to generate a decoded low-band mid signal;a low-band residual prediction unit configured to process the decoded low-band mid signal to generate a low-band residual prediction signal;an up-mix processor configured to generate a low-band left channel and a low-band right channel based partially on the decoded low-band mid signal and the low-band residual prediction signal;a high-band mid signal decoder configured to decode a high-band portion of the encoded mid signal to generate a time-domain decoded high-band mid signal;a high-band residual prediction unit configured to process the time-domain decoded high-band mid signal to generate a time-domain high-band residual prediction signal; andan inter-channel bandwidth extension decoder configured to generate a high-band left channel and a high-band right channel based on the time-domain decoded high-band mid signal and the time-domain high-band residual prediction signal.2. The device of claim 1 , comprising a receiver configured to receive a bitstream that ...

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Номер записи: 96
03-01-2019 дата публикации

ALIGNMENT OF BI-DIRECTIONAL MULTI-STREAM MULTI-RATE I2S AUDIO TRANSMITTED BETWEEN INTEGRATED CIRCUITS

Номер: US20190005974A1
Автор: HARIHARAN Magesh, Hu Ye, Li Ren, Mehrabi Arash, Mueller Stefan, Rohrer Stefan, Sienko Matthew
Принадлежит:

System, methods and apparatus are described that relate to aligning timing of bi-directional, multi-stream I2S audio transmitted between IC devices, and to support audio streams that are digitized using multiple sampling rates. A method includes time-division multiplexing a first stream of digitized audio data with a second stream of digitized audio data at a primary device to obtain a first multiplexed signal, transmitting the first multiplexed signal over a serial bus to a secondary device is configured to extract the first stream of digitized audio data from the first multiplexed signal and provide the first stream of digitized audio data to a first audio peripheral coupled to the secondary device, extracting the second stream of digitized audio data from the first multiplexed signal at the primary device, and providing the extracted second stream of digitized audio data to a second audio peripheral coupled to the first device 1. A method , comprising:at a primary device, time-division multiplexing a first stream of digitized audio data with a second stream of digitized audio data to obtain a first multiplexed signal;transmitting the first multiplexed signal over a serial bus to a secondary device that is configured to extract the first stream of digitized audio data from the first multiplexed signal and provide the first stream of digitized audio data to a first audio peripheral coupled to the secondary device;at the primary device, extracting the second stream of digitized audio data from the first multiplexed signal to provide an extracted second stream of digitized audio data; andproviding the extracted second stream of digitized audio data to a second audio peripheral coupled to the primary device,wherein the first audio peripheral and the second audio peripheral include digital-to-analog converters configured to produce analog signals from respective digitized audio data.2. The method of claim 1 , wherein time-division multiplexing the first stream of ...

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Номер записи: 97
05-01-2017 дата публикации

MATRIXED AUDIO SETTINGS

Номер: US20170006400A1
Автор: Bourne Nicholas Richard, GALLAGHER David, ROBERTSON Kevin Arthur
Принадлежит:

Methods and systems are provided for matrixed audio settings. In an audio system that comprises at least one audio output element operable to output audio signals, a first user input that comprises a selection of audio mode may be received, and based on the selected audio mode, one or more audio settings applicable during generating, processing, and/or outputting of the audio signals may be determined. At least one user control element may be configured to enable receiving of a second user input that comprises a selection between the determined one or more corresponding audio settings. The audio system (or at least the audio output element thereof) may be a headset. 1. A method comprising receiving a first user input that comprises a selection of an audio mode;', 'determining, based on said selected audio mode, one or more audio settings applicable during generating, processing, and/or outputting of said audio signals;', 'configuring at least one user control element to enable a second user input that comprises a selection between said one or more audio settings; and', 'wherein said determining of said one or more audio settings based on a matrixed mapping between each audio mode supported in said audio system and one or more settings from a plurality of audio settings applicable in said audio system., 'in an audio system that comprises at least one audio output element operable to output audio signals2. The method of claim 1 , comprising pre-programming said matrixed mapping.3. The method of claim 1 , comprising configuring and/or adjusting said matrixed mapping based on one or more of: pre-set data claim 1 , real-time data claim 1 , and/or user input.4. The method of claim 1 , wherein said audio settings comprise equalization (EQ) settings.5. The method of claim 1 , wherein said audio mode comprises one of a plurality of supported audio modes in said audio system.6. The method of claim 5 , wherein said plurality of supported audio modes comprise stereo mode claim ...

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Номер записи: 98
05-01-2017 дата публикации

Method and apparatus for higher order ambisonics encoding and decoding using singular value decomposition

Номер: US20170006401A1
Автор: Holger Kropp, Stefan Abeling
Принадлежит: DOLBY INTERNATIONAL AB

The encoding and decoding of HOA signals using Singular Value Decomposition includes forming ( 11 ) based on sound source direction values and an Ambisonics order corresponding ket vectors (|(Ω5))) of spherical harmonics and an encoder mode matrix (Ξ 0χs ). From the audio input signal (|χ(Ω s ))) a singular threshold value (σ ε ) determined. On the encoder mode matrix a Singular Value Decomposition ( 13 ) is carried out in order to get related singular values which are compared with the threshold value, leading to a final encoder mode matrix rank ( r fin e ). Based on direction values (Ω ι ) of loudspeakers and a decoder Ambisonics order (N ι ), corresponding ket vectors (IY(Ω ι ) ) and a decoder mode matrix (Ψ 0χL ) are formed ( 18 ). On the decoder mode matrix a Singular Value Decomposition ( 19 ) is carried out, providing a final decoder mode matrix rank ( r fin d ). From the final encoder and decoder mode matrix ranks a final mode matrix rank is determined, and from this final mode matrix rank and the encoder side Singular Value Decomposition an adjoint pseudo inverse (Ξ + ) † of the encoder mode matrix (Ξ 0χs ) and an Ambisonics ket vector (Ia′ s ) are calculated. The number of components of the Ambisonics ket vector is reduced ( 16 ) according to the final mode matrix rank so as to provide an adapted Ambisonics ket vector (|a′ ι ). From the adapted Ambisonics ket vector, the output values of the decoder side Singular Value Decomposition and the final mode matrix rank an adjoint decoder mode matrix (Ψ) † is calculated ( 15 ), resulting in a ket vector (|y(Ω ι )

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Номер записи: 99
04-01-2018 дата публикации

METADATA FOR DUCKING CONTROL

Номер: US20180006621A1
Автор: Allamanche Eric A., Baumgarte Frank M., Holman Tomlinson M.
Принадлежит:

An audio encoding device and an audio decoding device are described herein. The audio encoding device may examine a set of audio channels/channel groups representing a piece of sound program content and produce a set of ducking values to associate with one of the channels/channel groups. During playback of the piece of sound program content, the ducking values may be applied to all other channels/channel groups. Application of these ducking values may cause (1) the reduction in dynamic range of ducked channels/channel groups and/or (2) movement of channels/channel groups in the sound field. This ducking may improve intelligibility of audio in the non-ducked channel/channel group. For instance, a narration channel/channel group may be more clearly heard by listeners through the use of selective ducking of other channels/channel groups during playback. 1. A method for encoding a piece of sound program content , comprising:determining a first channel group in the piece of sound program content to emphasize during playback of the piece of sound program content, wherein the piece of sound program content has a plurality of channel groups that include the first channel group and a second channel group, object or stem;while detecting speech in the first channel group during a first time period of the piece of sound program content, generating a set of ducking values to apply to the second channel group, object or stem during the first time period in the piece of sound program content; andassociating the ducking values with the first channel group in an audio asset, wherein the audio asset includes the first channel group, the second channel group, object or stem, and the ducking values that are associated with the first channel group.2. The method of wherein the ducking values vary over time based on activity within the first channel group such that during periods of high activity within the first channel group claim 1 , the ducking values provide more ducking of the ...

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Номер записи: 100