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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 35471. Отображено 100.
05-01-2012 дата публикации

Full-Band Scalable Audio Codec

Номер: US20120004918A1
Автор: Jinwei Feng, Peter Chu
Принадлежит: Plycom Inc

A scalable audio codec for a processing device determines first and second bit allocations for each frame of input audio. First bits are allocated for a first frequency band, and second bits are allocated for a second frequency band. The allocations are made on a frame-by-frame basis based on the energy ratio between the two bands. For each frame, the codec transform codes both frequency bands into two sets of transform coefficients, which are then packetized based on the bit allocations. The packets are then transmitted with the processing device. Additionally, the frequency regions of the transform coefficients can be arranged in order of importance determined by power levels and perceptual modeling. Should bit stripping occur, the decoder at a receiving device can produce audio of suitable quality given that bits have been allocated between the bands and the regions of transform coefficients have been ordered by importance.

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12-01-2012 дата публикации

Nicam decoder with output resampler

Номер: US20120008724A1
Принадлежит: THAT Corp

A NICAM audio signal re-sampler may include a non-linear interpolator configured to interpolate in a non-linear manner between sequential digital samples that are based on a stream of demodulated NICAM audio samples. A phase differential calculator may be included that compares phase information at different resolutions.

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12-01-2012 дата публикации

Audio processing with time advanced inserted payload signal

Номер: US20120008803A1
Принадлежит: Sony Europe Ltd

An audio processing apparatus for modifying a primary audio signal includes a modulator that increases or decreases a level of a noise signal generated by a noise generator, in response to an increase or a decrease of a detected signal level of the primary audio signal, to generate a modulated noise signal. The apparatus further includes a combiner that combines the primary audio signal and the modulated noise signal. The modulator operates, with respect to a signal delayer, to time-advance a decrease in the level of said noise signal based on a corresponding decrease in the signal level of the primary audio signal.

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19-01-2012 дата публикации

Method and device for audio signal classification

Номер: US20120016677A1
Принадлежит: Huawei Technologies Co Ltd

The present invention discloses a method and a device for audio signal classification, and relates to the field of communications technologies, which solve a problem of high complexity of type classification of audio signals in the prior art. In the present invention, after an audio signal to be classified is received, a tonal characteristic parameter of the audio signal to be classified, where the tonal characteristic parameter of the audio signal to be classified is in at least one sub-band, is obtained, and a type of the audio signal to be classified is determined according to the obtained characteristic parameter. The present invention is mainly applied to an audio signal classification scenario, and implements audio signal classification through a relatively simple method.

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02-02-2012 дата публикации

Systems, methods, apparatus, and computer-readable media for dynamic bit allocation

Номер: US20120029925A1
Принадлежит: Qualcomm Inc

A dynamic bit allocation operation determines a bit allocation for each of a plurality of vectors, based on a corresponding plurality of gain factors, and compares each allocation to a threshold value that is based on a dimensionality of the vector.

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09-02-2012 дата публикации

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

Номер: US20120035939A1
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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16-02-2012 дата публикации

Methods and apparatus for embedding watermarks

Номер: US20120039504A1
Автор: Venugopal Srinivasan
Принадлежит: Individual

Methods and apparatus for embedding a watermark are disclosed. An example method disclosed herein to embed a watermark in a compressed data stream comprises obtaining a set of transform coefficients included in the compressed data stream, the set of transform coefficients having a respective first set of mantissa codes and a respective set of exponents, the first set of mantissa codes associated with a respective set of mantissa step sizes, identifying a first transform coefficient from the set of transform coefficients having a smallest magnitude among the set of transform coefficients, determining a second set of mantissa codes based on the first transform coefficient and the set of step sizes, and replacing the first set of mantissa codes included in the compressed data stream with the second set of mantissa codes to embed the watermark without uncompressing the compressed data stream.

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15-03-2012 дата публикации

Efficient Combined Harmonic Transposition

Номер: US20120065983A1
Принадлежит: DOLBY INTERNATIONAL AB

The present document relates to audio coding systems which make use of a harmonic transposition method for high frequency reconstruction (HFR), and to digital effect processors, e.g. so-called exciters, where generation of harmonic distortion adds brightness to the processed signal. In particular; a system configured to generate a high frequency component of a signal from a low frequency component of the signal is described, The system may comprise an analysis filter bank ( 501 ) configured to provide a set of analysis subband signals from the low frequency component of the signal; wherein the set of analysis subband signals comprises at least two analysis subband signals; wherein the analysis filter bank ( 501 ) has a frequency resolution of Δf, The system further comprises a nonlinear processing unit ( 502 ) configured to determine a set of synthesis subband signals from the set of analysis subband signals using a transposition order P; wherein the set of synthesis subband signals comprises a portion of the set of analysis subband signals phase shifted by an amount derived from the transposition order P; and a synthesis filter bank ( 504 ) configured to generate the high frequency component of the signal from the set of synthesis subband signals; wherein the synthesis filter bank ( 504 ) has a frequency resolution of FΔf; with F being a resolution factor, with F≧1; wherein the transposition order P is different from the resolution factor F.

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29-03-2012 дата публикации

Method and device for frequency compression with selective frequency shifting

Номер: US20120076333A1
Принадлежит: Siemens Medical Instruments Pte Ltd

A method and device for frequency compression of audio signals to reduce the occurrence of artifacts. A component of the audio signal having a plurality of frequency channels is shifted from a first frequency channel into a second frequency channel. A dominant instantaneous frequency is determined in the first frequency channel. During shifting or mapping, first the entire first frequency channel, including the dominant instantaneous frequency, is shifted or mapped into the second frequency channel, wherein the dominant instantaneous frequency obtains an intermediate frequency position. A final frequency position for the dominant instantaneous frequency is determined using a predefined compression characteristic in the second frequency channel, starting from the frequency position of the dominant instantaneous frequency in the first frequency channel. Finally, the dominant instantaneous frequency is shifted or mapped from the intermediate frequency position to the final frequency position.

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05-04-2012 дата публикации

Multi-channel audio encoding and decoding

Номер: US20120082316A1
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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12-04-2012 дата публикации

Btsc encoder

Номер: US20120087502A1
Автор: Christopher M. Hanna
Принадлежит: THAT Corp

The disclosed BTSC encoder includes a left high pass filter means; a matrix means for receiving the digital left and digital right filtered signals, and including means for summing the digital left and digital right filtered signals and thereby generating a digital sum signal, and including means for subtracting one of the digital left and digital right filtered signals from the other of the digital left and digital right filtered signals and thereby generating a digital difference signal; a difference channel processing means for digitally processing the digital difference signal; and a sum channel processing means for digitally processing the digital sum signal.

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03-05-2012 дата публикации

Compensator and Compensation Method for Audio Frame Loss in Modified Discrete Cosine Transform Domain

Номер: US20120109659A1
Принадлежит: ZTE Corp

The invention provides a compensation method for audio frame loss in a MDCT domain, the method comprising: when a frame currently lost is a P th frame, obtaining a set of frequencies to be predicted, and for each frequency in the set, using phases and amplitudes of a plurality of frames before a (P−1) th frame in a MDCT-MDST domain to predict a phase and an amplitude of the P th frame, and using the predicted phase and amplitude to obtain a MDCT coefficient of the P th frame at each corresponding frequency; for a frequency outside the set, using MDCT coefficients of a plurality of frames before the P th frame to calculate a MDCT coefficient value of the P th frame at the frequency; performing an IMDCT for the MDCT coefficients of the P th frame to obtain a time domain signal of the P th frame.

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10-05-2012 дата публикации

Method and apparatus for encoding and decoding high frequency signal

Номер: US20120116757A1
Принадлежит: SAMSUNG ELECTRONICS CO LTD

Provided are a method and apparatus for encoding and decoding a high frequency signal by using a low frequency signal. The high frequency signal can be encoded by extracting a coefficient by linear predicting a high frequency signal, and encoding the coefficient, generating a signal by using the extracted coefficient and a low frequency signal, and encoding the high frequency signal by calculating a ratio between the high frequency signal and an energy value of the generated signal. Also, the high frequency signal can be decoded by decoding a coefficient, which is extracted by linear predicting a high frequency signal, and a low frequency signal, and generating a signal by using the decoded coefficient and the decoded low frequency signal, and adjusting the generated signal by decoding a ratio between the generated signal and an energy value of the high frequency signal.

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17-05-2012 дата публикации

System and method for providing enhanced audio in a video environment

Номер: US20120120270A1
Принадлежит: Cisco Technology Inc

A method is provided in one example and includes receiving audio data at a microphone array that includes a plurality of microphones. The microphone array is provisioned at a first endpoint, which includes a camera element configured to capture video data associated with a video session involving the first endpoint and a second endpoint. The method also includes formatting the audio data into a time division multiplex (TDM) stream, and communicating the stream to a port for a subsequent communication over a network and to the second endpoint.

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31-05-2012 дата публикации

Performing enhanced sigma-delta modulation

Номер: US20120133537A1
Принадлежит: Qualcomm Inc

In general, techniques are described for performing enhanced sigma-delta modulation. For example, an apparatus comprising a predictive filter unit, an amplifier, an oversampling unit and a sigma-delta modulation unit may implement the techniques. The predictive filter unit performs predictive filtering on an input signal to generate a filtered signal and computes an estimate of a predictive gain as a function of an energy of the input signal and an energy of the filtered signal. The amplifier receives the filtered signal and amplifies the filtered signal based on the predictive gain to generate an amplified signal. The oversampling unit receives the amplifies signal and performs oversampling in accordance with an oversampling rate to generate an oversampled signal. The sigma-delta modulation unit receives the oversampled signal and performs sigma-delta modulation to generate a modulated signal.

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21-06-2012 дата публикации

Method And Apparatus For Reducing Rendering Latency For Audio Streaming Applications Using Internet Protocol Communications Networks

Номер: US20120158408A1
Автор: James W. McGowan
Принадлежит: Alcatel Lucent SAS

A method and apparatus for reducing rendering latency in a terminal device which receives audio data from a communication network such as, for example, Voice over Internet Protocol (VoIP) communications networks. Received packets are advantageously decoded “immediately” upon receipt, and the decoded data is placed directly in the rendering buffer at a location corresponding to the time appropriate for rendering, without using any intermediate buffer. Then, in accordance with the principles of the present invention and more particularly in accordance with certain illustrative embodiments thereof, packet loss concealment (PLC) routines are advantageously applied preemptively, without first determining whether or not any subsequent packets have or have not been received by any particular time.

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28-06-2012 дата публикации

Speech Coding

Номер: US20120166189A1
Автор: Koen Bernard Vos
Принадлежит: Skype Ltd Ireland

A method of encoding one or more parent blocks of values, the number of values being the length of each block, the method comprising for each parent block: (a) determining a first sum of values in the parent block; 1(b) splitting the parent block into smaller subblocks; (c) for at least one of the subblocks, determining a second sum of the values in the subblock, selecting a likelihood table from the plurality of likelihood tables based on said first sum of values in the parent block and encoding the second sum using the likelihood table; (d) designating each subblock a parent block; (e) carrying out steps (a), (b), (c) and (d) until at least one parent block reaches a predetermined condition.

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05-07-2012 дата публикации

Apparatus for encoding and decoding an audio signal using a weighted linear predictive transform, and a method for same

Номер: US20120173247A1
Принадлежит: SAMSUNG ELECTRONICS CO LTD

Disclosed is an apparatus for encoding and/or decoding an audio signal having a variable bit rate (VBR). A target bit rate is determined in accordance with characteristics of an audio signal, and a weighted linear predictive transform coding is performed in accordance with the determined target bit rate.

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12-07-2012 дата публикации

Hearing aid with audio codec and method

Номер: US20120177234A1
Принадлежит: Widex AS

A hearing aid comprising a time domain codec. The codec comprises a decoder adapted to generate a decoded output signal based on an input quantization index and an encoder for generating an output quantization index based on an input signal, said encoder comprising said decoder and a predictor receiving an excitation signal derived from said decoder output signal and outputting a prediction signal. The output quantization index is determined by repeated decoding of the quantization indices in order to minimize the error between the input signal and the prediction signal, and the predictor uses a recursive autocorrelation estimate for the error minimization. The invention further provides a method of encoding an audio signal.

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12-07-2012 дата публикации

System and method for efficiently translating media files between formats using a universal representation

Номер: US20120179700A1
Принадлежит: Apple Inc

An apparatus and method are described for reading a file into a universal representation and translating from that universal representation into various file formats. For example, a method according to one embodiment comprises: reading compressed audio data from a first audio file, the first audio file comprising audio data compressed using a first compression algorithm and bookkeeping data having a first format, the bookkeeping data specifying a location of the compressed audio data within the first audio file; and generating a universal representation of the first audio file without decompressing and recompressing the audio data, the universal representation having bookkeeping data of a second format specifying the location of compressed audio data within the universal representation.

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12-07-2012 дата публикации

Digital Watermark Key Generation

Номер: US20120179914A1
Принадлежит: Individual

This disclosure relates to message encoding. One claim recites a digital watermark key generation method in which the key providing security for a plural-bit message. The method comprises: providing a plural-bit seed; randomizing the plural-bit seed; using a programmed electronic processor for encoding the randomized plural-bit seed with convolutional encoding, the encoded seed comprising a key; and transforming an independent message with the key, the independent message to be used in a digital watermark encoding process. Of course, other claims and combinations are provided too.

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02-08-2012 дата публикации

Oversampling in a combined transposer filter bank

Номер: US20120195442A1
Принадлежит: DOLBY INTERNATIONAL AB

The present invention relates to coding of audio signals, and in particular to high frequency reconstruction methods including a frequency domain harmonic transposer. A system and method for generating a high frequency component of a signal from a low frequency component of the signal is described. The system comprises an analysis filter bank ( 501 ) comprising an analysis transformation unit ( 601 ) having a frequency resolution of Δf; and an analysis window ( 611 ) having a duration of D A ; the analysis filter bank ( 501 ) being configured to provide a set of analysis subband signals from the low frequency component of the signal; a nonlinear processing unit ( 502, 650 ) configured to determine a set of synthesis subband signals based on a portion of the set of analysis subband signals, wherein the portion of the set of analysis subband signals is phase shifted by a transposition order T; and a synthesis filter bank ( 504 ) comprising a synthesis transformation unit ( 602 ) having a frequency resolution of QΔf; and a synthesis window ( 612 ) having a duration of D s ; the synthesis filter bank ( 504 ) being configured to generate the high frequency component of the signal from the set of synthesis subband signals; wherein Q is a frequency resolution factor with Q≧1 and smaller than the transposition order T; and wherein the value of the product of the frequency resolution Δf and the duration D A of the analysis filter bank is selected based on the frequency resolution factor Q.

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09-08-2012 дата публикации

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

Номер: US20120203362A1

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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16-08-2012 дата публикации

Audio signal of an fm stereo radio receiver by using parametric stereo

Номер: US20120207307A1
Принадлежит: DOLBY INTERNATIONAL AB

The invention relates to an apparatus for improving a stereo audio signal of an FM stereo radio receiver. The apparatus comprises a parametric stereo (PS) parameter estimation stage. The parameter estimation stage is configured to determine one or more parametric stereo parameters based on the stereo audio signal in a frequency-variant or frequency-invariant manner. Preferably, these PS parameters are time- and frequency-variant. Moreover, the apparatus comprises an upmix stage. The upmix stage is configured to generate the improved stereo signal based on a first audio signal and the one or more parametric stereo parameters. The first audio signal is obtained from the stereo audio signal, e.g. by a downmix operation in a downmix stage. The PS parameter estimation stage may be part of a PS encoder. The upmix stage may be part of a PS decoder.

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16-08-2012 дата публикации

Device and method for quantizing the gains of the adaptive and fixed contributions of the excitation in a celp codec

Номер: US20120209599A1
Автор: Vladimir Malenovsky
Принадлежит: VoiceAge Corp

A device and method for quantizing a gain of a fixed contribution of an excitation in a frame, including sub-frames, of a coded sound signal. The gain is estimated in a sub-frame using a frame classification parameter, and is then quantized in the sub-frame using the estimated gain. The device and method can be used in jointly quantizing gains of adaptive and fixed contributions of an excitation. For retrieving a quantized gain of a fixed contribution of an excitation in a sub-frame, the gain of the fixed excitation contribution is estimated using a frame classification parameter, a gain codebook supplies a correction factor in response to a received, gain codebook index, and a multiplier multiplies the estimated gain by the correction factor to provide the quantized gain.

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23-08-2012 дата публикации

Method and apparatus for mixed dimensionality encoding and decoding

Номер: US20120215525A1
Принадлежит: Huawei Technologies Co Ltd

A method and apparatus for mixed dimensionality encoding and decoding are provided in embodiments of the present invention. The method includes: obtaining at least one variable collection through calculation according to a processed spectral coefficient, determining a processing dimension for a spectral coefficient to be processed, according to a relationship between the at least one variable collection and a corresponding threshold collection, and performing, according to a selected dimension, encoding or decoding under the dimension on the spectral coefficient to be processed. Through the preceding technical means, different processing dimensions are used for different spectral coefficients, improving the encoding and decoding efficiency.

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23-08-2012 дата публикации

Complexity Scalable Perceptual Tempo Estimation

Номер: US20120215546A1
Принадлежит: DOLBY INTERNATIONAL AB

The present document relates to methods and systems for estimating the tempo of a media signal, such as audio or combined video/audio signal. In particular, the document relates to the estimation of tempo perceived by human listeners, as well as to methods and systems for tempo estimation at scalable computational complexity. A method and system for extracting tempo information of an audio signal from an encoded bit-stream of the audio signal comprising spectral band replication data is described. The method comprises the steps of determining a payload quantity associated with the amount of spectral band replication data comprised in the encoded bit-stream for a time interval of the audio signal; repeating the determining step for successive time intervals of the encoded bit-stream of the audio signal, thereby determining a sequence of payload quantities; identifying a periodicity in the sequence of payload quantities; and extracting tempo information of the audio signal from the identified periodicity.

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06-09-2012 дата публикации

Audio coding device, audio coding method, and computer-readable recording medium storing audio coding computer program

Номер: US20120224703A1
Принадлежит: Fujitsu Ltd

An audio coding device includes a time frequency transform unit that, with respect to each of a plurality of channels included in an audio signal, generates a time frequency signal indicating frequency components at each time by performing a time frequency transform on a signal of the channel; a transient detection unit that detects a transient with respect to each of the plurality of channels so as to obtain a transient detection time; a transient time correction unit that, when a difference in transient detection times between an early detection channel in which the transient detection time is earliest and a late detection channel that is a channel other than the early detection channel among the plurality of channels is within a range in which the transient; a grid determination unit that, with respect to each of the plurality of channels, and a coding unit that codes.

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06-09-2012 дата публикации

apparatus for processing a signal and method thereof

Номер: US20120226496A1
Принадлежит: LG ELECTRONICS INC

An apparatus for processing a signal and method thereof are disclosed. The present invention includes receiving coding mode information indicating a speech coding scheme or an audio coding scheme, linear prediction coding degree information indicating a linear prediction coding degree, and the signal including at least one of a speech signal and an audio signal; decoding the signal according to the speech coding scheme or the audio coding scheme based on the coding mode information; decoding linear prediction coding coefficients of the signal based on the linear prediction coding degree information; and generating an output signal by applying the decoded linear prediction coding coefficients to the decoded signal. In this case, the linear prediction coding degree information is determined based on a variation of a value of an LPC residual generated from performing the linear prediction coding on the signal.

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20-09-2012 дата публикации

Integrated Circuit with Conversion Capability for Portable Medial Player

Номер: US20120236928A1
Автор: Alexander G. MacInnis
Принадлежит: Broadcom Corp

Presented herein are system(s), method(s), and apparatus for an integrated circuit with conversion capabilities for transferring data to a portable media player. In one embodiment, there is presented an integrated circuit for providing video data. The integrated circuit comprises at least one input, at least one output, an encoder, and at least another output. At least one input receives video data. At least one output provides the video data to a display screen. The encoder encodes the video data into a particular compressed format. The at least another output for provides the video data in the particular compressed format to an interface.

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20-09-2012 дата публикации

Method and an apparatus for processing an audio signal

Номер: US20120239408A1

A method of processing an audio signal is disclosed. The present invention includes receiving, by an audio processing apparatus, coding identification information indicating whether to apply a first coding scheme or a second coding scheme to a current frame; when the coding identification information indicates that the second coding scheme is applied to the current frame, receiving window type information indicating a particular window for the current frame, from among a plurality of windows; identifying that a current window is a long stop window based on the window type information, wherein the long stop window is followed by only long window of a following frame, wherein the long stop window includes a gentle long stop window and a steep long stop window; and, when the first coding scheme is applied to a previous frame, applying the gentle long stop window to the current frame, wherein: the gentle long stop window comprise an ascending line with first slope, the steep long stop window comprise an ascending line with second slope, and, the first slope is gentler than the second slope.

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04-10-2012 дата публикации

Multi-mode audio codec and celp coding adapted therefore

Номер: US20120253797A1

In an embodiment, bitstream elements of sub-frames are encoded differentially to a global gain value so that a change of the global gain value results in an adjustment of an output level of the decoded representation of the audio content. Concurrently, the differential coding saves bits. Even further, the differential coding enables the lowering of the burden of globally adjusting the gain of an encoded bitstream. In another embodiment, a global gain control across CELP coded frames and transform coded frames is achieved by co-controlling the gain of the codebook excitation of the CELP codec, along with a level of the transform or inverse transform of the transform coded frames. In another embodiment, the gain value determination in CELP coding is performed in the weighted domain of the excitation signal.

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11-10-2012 дата публикации

Method and apparatus for encoding audio data

Номер: US20120259645A1
Принадлежит: Individual

A method for processing audio data includes determining a first common scalefactor value for representing quantized audio data in a frame. A second common scalefactor value is determined for representing the quantized audio data in the frame. A line equation common scalefactor value is determined from the first and second common scalefactor values.

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18-10-2012 дата публикации

Optimized parametric stereo decoding

Номер: US20120265542A1
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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01-11-2012 дата публикации

Method and system for utilizing spread spectrum techniques for in car applications

Номер: US20120274459A1

A method of operating an audio system in an automobile includes identifying a user of the audio system. An audio recording playing on the audio system is identified. An audio setting entered into the audio system by the identified user while the audio recording is being played by the audio system is sensed. The sensed audio setting is stored in memory in association with the identified user and the identified audio recording. The audio recording is retrieved from memory with the sensed audio setting being embedded in the retrieved audio recording as a watermark signal. The retrieved audio recording is played on the audio system with the embedded sensed audio setting being automatically implemented by the audio system during the playing.

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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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15-11-2012 дата публикации

Noise filling and audio decoding

Номер: US20120288117A1
Автор: Eun-mi Oh, Mi-young Kim
Принадлежит: SAMSUNG ELECTRONICS CO LTD

A noise filling method is provided that includes detecting a frequency band including a part encoded to 0 from a spectrum obtained by decoding a bitstream; generating a noise component for the detected frequency band; and adjusting energy of the frequency band in which the noise component is generated and filled by using energy of the noise component and energy of the frequency band including the part encoded to 0.

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15-11-2012 дата публикации

Transform-Domain Codebook In A Celp Coder And Decoder

Номер: US20120290295A1
Автор: Vaclav Eksler
Принадлежит: VoiceAge Corp

Codebook Arrangement for use in coding an input sound signal includes First and Second Codebook Stages. First Codebook Stage includes one of a time-domain CELP codebook and a transform-domain codebook. Second Codebook Stage follows the first codebook stage and includes the other of the time-domain CELP codebook and the transform-domain codebook. Codebook Stage includes an adaptive codebook may be provided before First Codebook Stage. A selector may be provided to select an order of the time-domain CELP codebook and the transform-domain codebook in First and Second Codebook Stages, respectively, as a function of characteristics of the input sound signal. The selector may also be responsive to both the characteristics of the input sound signal and a bit rate of the codec using Codebook Arrangement to bypass Second Codebook Stage. Codebook Arrangement can be used in a coder of an input sound signal.

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22-11-2012 дата публикации

Method, medium, and system encoding/decoding multi-channel signal

Номер: US20120294448A1
Принадлежит: Individual

A multi-channel signal decoding method is provided. A down-mixed signal representative of a multi-channel signal is decoded, and parameters representing characteristic relations between channels of the multi-channel signal are decoded. An additional parameter is estimated by using the decoded parameters, and the decoded down-mixed signal is up-mixed by using the decoded parameters and the estimated parameter so as to decode the multi-channel signal.

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29-11-2012 дата публикации

Audio decoding using variable-length codebook application ranges

Номер: US20120303375A1
Автор: Yuli You
Принадлежит: Digital Rise Technology Co Ltd

Provided are, among other things, systems, methods and techniques for decoding an audio signal from a frame-based bit stream. At least one frame includes processing information pertaining to the frame and entropy-encoded quantization indexes representing audio data within the frame. The processing information includes: (i) code book indexes, and (ii) code book application information specifying ranges of entropy-encoded quantization indexes to which the code books are to be applied. The entropy-encoded quantization indexes are decoded by applying the identified code books to the corresponding ranges of entropy-encoded quantization indexes.

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13-12-2012 дата публикации

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

Номер: US20120314876A1

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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13-12-2012 дата публикации

Parametric joint-coding of audio sources

Номер: US20120314879A1
Автор: Christof Faller

The following coding scenario is addressed: A number of audio source signals need to be transmitted or stored for the purpose of mixing wave field synthesis, multi-channel surround, or stereo signals after decoding the source signals. The proposed technique offers significant coding gain when jointly coding the source signals, compared to separately coding them, even when no redundancy is present between the source signals. This is possible by considering statistical properties of the source signals, the properties of mixing techniques, and spatial hearing. The sum of the source signals is transmitted plus the statistical properties of the source signals which mostly determine the perceptually important spatial cues of the final mixed audio channels. Source signals are recovered at the receiver such that their statistical properties approximate the corresponding properties of the original source signals. Subjective evaluations indicate that high audio quality is achieved by the proposed scheme.

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13-12-2012 дата публикации

Method and apparatus for encoding a signal

Номер: US20120316885A1
Автор: Jonathan A. Gibbs
Принадлежит: MOTOROLA MOBILITY LLC

A method and apparatus for encoding a signal is provided herein. During operation a wideband signal that is to be encoded enters a filter bank. A highband signal and a lowband signal are output from the filter bank. Each signal is separately encoded. During the production of the highband signal, a downmixing operation is implemented after preprocessing, and prior to decimating. The downmixing operation greatly reduces system complexity. In fact, it will be observed that the highest sample rate in the prior-art implementation is 64 kHz whereas the sample rate in the system described above remains at 32 kHz or below. This represents a significant complexity saving, as do the reduced number of processing blocks.

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03-01-2013 дата публикации

Audio encoder, audio encoding method and program

Номер: US20130003980A1
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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03-01-2013 дата публикации

Transform Audio Codec and Methods for Encoding and Decoding a Time Segment of an Audio Signal

Номер: US20130006646A1
Принадлежит: Individual

Methods and devices for efficient encoding/decoding of a time segment of an audio signal. Methods comprise deriving an indicator, z, of the position in a frequency scale of a residual vector associated with the time segment of the audio signal, and deriving a measure, Φ, related to the amount of structure of the residual vector. The methods further comprise determining whether a predefined criterion involving the measure Φ, the indicator z and a predefined threshold Θ, is fulfilled, which corresponds to estimating whether a change of sign of at least some of the non-zero coefficients of the residual vector would be audible after reconstruction of the audio signal time segment. The amplitude of the coefficients of the residual vector is encoded, and the signs of the coefficients of the residual vector are encoded only when it is determined that the criterion is fulfilled, and thus that a change of sign would be audible.

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10-01-2013 дата публикации

Methods and apparatus to facilitate voicemail interaction

Номер: US20130010934A1
Автор: Jon S. Miller
Принадлежит: Individual

Example methods and apparatus to facilitate voicemail interaction are disclosed. A disclosed example method involves, during a call session with a voicemail system, receiving an audio segment from the voicemail system. The example method also involves performing feature recognition on the audio segment and outputting a display element to a user interface based on a recognized feature in the audio segment.

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10-01-2013 дата публикации

Apparatus for processing an audio signal and method thereof

Номер: US20130013321A1
Принадлежит: LG ELECTRONICS INC

A method of processing an audio signal is disclosed. The present invention includes a method for processing an audio signal, comprising: receiving, by an audio processing apparatus, the spectral data including a current block, and substitution type information indicating whether to apply a shape prediction scheme to a current block; when the substitution type information indicates that the shape prediction scheme is applied to the current block, receiving lag information indicating an interval between spectral coefficients of the current block and the predictive shape vector of a current frame or a previous frame; obtaining spectral coefficients by substituting for spectral hole included in the current block using the predictive shape vector.

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17-01-2013 дата публикации

Audio signal coding and decoding method and device

Номер: US20130018660A1
Автор: Fengyan Qi, LEI Miao, Zexin LIU
Принадлежит: Huawei Technologies Co Ltd

Embodiments of the present invention provide an audio signal coding and decoding method and device. The coding method includes: dividing a frequency band of an audio signal into a plurality of sub-bands, and quantifying a sub-band normalization factor of each sub-band; determining signal bandwidth of bit allocation according to the quantified sub-band normalization factor, or according to the quantified sub-band normalization factor and bit rate information; allocating bits for a sub-band within the determined signal bandwidth; and coding a spectrum coefficient of the audio signal according to the bits allocated for each sub-band. According to embodiments of the present invention, during coding and decoding, signal bandwidth of bit allocation is determined according to the quantified sub-band normalization factor and bit rate information. In this manner, the determined signal bandwidth is effectively coded and decoded by centralizing the bits, and audio quality is improved.

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24-01-2013 дата публикации

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

Номер: US20130022205A1
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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31-01-2013 дата публикации

MDCT-Based Complex Prediction Stereo Coding

Номер: US20130028426A1
Принадлежит: DOLBY INTERNATIONAL AB

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 upmixing can be suspended responsive to control data.

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31-01-2013 дата публикации

Audio encoding apparatus and audio encoding method

Номер: US20130030796A1
Автор: Zongxian Liu
Принадлежит: Panasonic Corp

An audio encoding apparatus that allows a decoded signal exhibiting an excellent sound quality to be obtained on a decoding side. In the audio encoding apparatus ( 1000 A), a time-frequency transform unit ( 1001 ) uses a time-frequency transform, such as a discrete Fourier transform (DFT) or a modified discrete cosine transform (MDCT), to transform a time domain signal (S(n)) to a frequency domain signal (spectrum factor) (S(f)). A psychoacoustic model analyzing unit ( 1002 ) performs a psychoacoustic model analysis of the frequency domain signal (S(f)), thereby obtaining a masking curve. An acoustic sense weighting unit ( 1003 ) estimates, based on the masking curve, an importance degree of acoustic sense, and determines and applies the weighting factors of respective spectrum factors to the respective spectrum factors. An encoding unit ( 1004 ) encodes the frequency domain signal (S(f)) as weighted in terms of the acoustic sense. A multiplexing unit ( 1005 ) multiplexes and transmits the encoded parameters.

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31-01-2013 дата публикации

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

Номер: US20130030819A1

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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31-01-2013 дата публикации

Method for secure transactions utilizing physically separated computers

Номер: US20130031005A1
Принадлежит: Posa John G, Schwab Barry H

A secure transaction method involves establishing an electronically accessible verification site authorized by the holder of a credit or debit card, and accessing the verification site by a merchant to determine whether a request for goods or services is authorized. The request for goods or services is based upon the use of the credit or debit card, but the card is not physically presented. The verification site is an electronic mail account which may be established by the merchant, card holder or other authorized person or entity. An authorization message is preferably sent from the site to the merchant in response to the step of accessing the verification site by the merchant. The verification site may also be wirelessly accessible, enabling an authorization message to be delivered through a cellular telephone, personal digital assistant, or other mobile device.

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21-02-2013 дата публикации

Mechanism for dynamic signaling of encoder capabilities

Номер: US20130046534A1
Принадлежит: Telefonaktiebolaget LM Ericsson AB

The present disclosure provides systems and methods for dynamically signaling encoder capabilities of vocoders of corresponding communication nodes. In one embodiment, during a call between a first communication node and a second communication node, a control node (e.g., base station controller or mobile switching center) for the first communication node sends capability information for a voice encoder of a vocoder of the first communication node to a control node for the second communication node. As a result, the second communication node is enabled to select and request a preferred encoder mode for the voice encoder of the vocoder of the first communication node based on the capabilities of the voice encoder of the vocoder of the first communication node.

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21-02-2013 дата публикации

Restoration of high-order Mel Frequency Cepstral Coefficients

Номер: US20130046540A9
Автор: Alexander Sorin
Принадлежит: Individual

A method for estimating high-order Mel Frequency Cepstral Coefficients, the method comprising initializing any of N-L high-order coefficients (HOC) of an MFCC vector of length N having L low-order coefficients (LOC) to a predetermined value, thereby forming a candidate MFCC vector, synthesizing a speech signal frame from the candidate MFCC vector and a pitch value, and computing an N-dimensional MFCC vector from the synthesized frame, thereby producing an output MFCC vector.

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21-02-2013 дата публикации

Periodic Ambient Waveform Analysis for Enhanced Social Functions

Номер: US20130046542A1
Принадлежит: Individual

Client devices periodically capture ambient audio waveforms, generate waveform fingerprints, and upload the fingerprints to a server for analysis. The server compares the waveforms to a database of stored waveform fingerprints, and upon finding a match, pushes content or other information to the client device. The fingerprints in the database may be uploaded by other users, and compared to the received client waveform fingerprint based on common location or other social factors. Thus a client's location may be enhanced if the location of users whose fingerprints match the client's is known. In particular embodiments, the server may instruct clients whose fingerprints partially match to capture waveform data at a particular time and duration for further analysis and increased match confidence.

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28-02-2013 дата публикации

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

Номер: US20130051564A1
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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14-03-2013 дата публикации

Information signal representation using lapped transform

Номер: US20130064383A1

An information signal reconstructor is configured to reconstruct, using aliasing cancellation, an information signal from a lapped transform representation of the information signal including, for each of consecutive, overlapping regions of the information signal, a transform of a windowed version of the respective region, wherein the information signal reconstructor is configured to reconstruct the information signal at a sample rate which changes at a border between a preceding region and a succeeding region of the information signal.

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14-03-2013 дата публикации

Apparatus and method of enhancing quality of speech codec

Номер: US20130066627A1

An apparatus and method of improving the quality of a speech codec are provided. In the method, a first energy of a signal decoded by a low-band codec is calculated, and a second energy of a signal decoded by a low-band enhancement mode is calculated. Then, when the first energy is less than a first threshold value or less than a product of the second energy and a second threshold value, a size of the decoded signal is scaled. Accordingly, generation of a quantization error with respect to a silence segment is reduced.

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14-03-2013 дата публикации

Echo Cancelling-Codec

Номер: US20130066638A1
Принадлежит: QNX Software Systems Ltd

Echo-cancellation is utilized in terminal devices such as speakerphones to compensate for acoustic echoes and interaction of the audio signal with the surrounding environment. An echo-cancelling codec incorporates encoding, decoding and acoustic echo-cancellation in a single device, enabling processing to be utilized that reduces processing and memory resources. The configuration enables processing information to also be shared between encoding, decoding and acoustic echo-cancellation functions to optimize operational characteristics. The acoustic echo cancelling codec interfaces between the amplitude signal domain, speaker and microphone, and an encoded data domain, a data interface, reducing component requirements required to provide echo-cancellation and coding functions.

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21-03-2013 дата публикации

METHOD AND APPARATUS FOR DETECTING WHICH ONE OF SYMBOLS OF WATERMARK DATA IS EMBEDDED IN A RECEIVED SIGNAL

Номер: US20130073065A1
Принадлежит: THOMSON LICENSING

Watermark symbol detection requires a detection metric for deciding at decoder side which candidate symbol is embedded inside the audio or video signal content. The invention provides an improved detection metric processing that achieves a reliable detection of watermarks in the presence of additional noise and echoes, and that is adaptive to signal reception conditions and requires a decreased computational power. This is performed by taking into account the information contained in the echoes of the received audio signal in the decision metric and comparing it with the corresponding metric obtained from decoding a non-marked audio signal, based on recursive calculation of false positive detection rates of Correlations for all REFP Reference Pattern peaks in correlation result values. The watermark symbol corresponding to the reference sequence having the lowest false positive error is selected as the embedded one. 16-. (canceled)7. A method for detecting which one of symbols of watermark data embedded in an original signal—by modifying sections of said original signal in relation to at least two different reference data sequences—is present in a current section of a received version of the watermarked original signal , wherein said received watermarked original signal can include noise and/or echoes , said method including the steps:correlating in each case said current section of said received watermarked signal with candidates of said reference data sequences;based on peak values in the correlation result values for said current signal section, detecting—using related values of false positive probability of detection of the kind of symbol—which one of the candidate symbols is present in said current signal section,wherein that said false positive probability is calculated in a recursive manner, wherein the total false positive probability for a given number of correlation result peak values is evaluated by using initially the false positive probabilities for a ...

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21-03-2013 дата публикации

Audio signal decoder, audio signal encoder, methods and computer program using a sampling rate dependent time-warp contour encoding

Номер: US20130073296A1

An audio signal decoder configured to provide a decoded audio signal representation on the basis of an encoded audio signal representation including a sampling frequency information, an encoded time warp information and an encoded spectrum representation includes a time warp calculator and a warp decoder. The time warp calculator is configured to adapt a mapping rule for mapping codewords of the encoded time warp information onto decoded time warp values describing the decoded time warp information in dependence on the sampling frequency information. The warp decoder is configured to provide the decoded audio signal representation on the basis of the encoded spectrum representation and in dependence on the decoded time warp information.

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21-03-2013 дата публикации

Methods and devices for providing an encoded digital signal

Номер: US20130073297A1

In one embodiment, a method for providing an encoded digital signal is described comprising determining, for each data frame of a plurality of data frames of a digital signal, a plurality of pairs of an encoding data volume and an encoding quality, wherein each pair of an encoding data volume and an encoding quality specifies the encoding data volume required for achieving the encoding quality; determining for each data frame at least one or more interpolations between the plurality of determined pairs; determining a multi-frame relationship between encoding quality and encoding data volume required to encode the plurality of data frames at the encoding quality based on a combination of the at least one or more interpolations for the plurality of data frames; determining an encoding quality for the plurality of data frames based on the relationship; and providing at least one data frame of the plurality of data frames encoded at the determined encoding quality.

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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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04-04-2013 дата публикации

COMMUNICATION SYSTEM, METHOD, AND APPARATUS

Номер: US20130085750A1
Автор: OZAWA Kazunori
Принадлежит: NEC Corporation

A server apparatus acquires content based on instruction information; decodes image data of the acquired content compression encodes captured image data using a predetermined encoding scheme; decodes an audio signal and compression encodes the decoded audio signal using the predetermined encoding scheme, stores the image and the audio signal and sends the packet to a packet forwarding apparatus. A mobile terminal receives the packet, decodes and displays the compression encoded image data stored in the packet; and decodes and reproduces the compression encoded audio signal. 1. A communication system , comprising:first to Nth mobile terminals (N is an integer equal to or larger than 2); anda server apparatus including first to Nth virtual client units, the first to Nth virtual client units being connected respectively to the first to Nth mobile terminals via a packet forwarding apparatus on a mobile network,wherein each of the virtual client unitsreceives instruction information from the mobile terminal via the packet forwarding apparatus,runs an application, based on the instruction information, to generate a screen and compression encodes a part or whole of the screen using an image encoder,once decodes an audio signal, associated with the application or a content file, and compression encodes a part or whole of the decoded audio signal again using a predetermined audio encoder, andstores the compression encoded result in a packet and sends the packet to the packet forwarding apparatus, andwherein the mobile terminal receives the packet from the server apparatus via the packet forwarding apparatus on the mobile network, decodes the compression encoded result stored in the packet, using a screen decoder and displays a screen, and decodes the compression encoded result using an audio decoder and reproduces the decoded result.2. The communication system according to claim 1 ,wherein, for each of the mobile terminals, the server apparatus comprises:a control signal ...

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11-04-2013 дата публикации

Mobile device context information using speech detection

Номер: US20130090926A1
Принадлежит: Qualcomm Inc

Systems and methods for speech detection in association with a mobile device are described herein. A method described herein for identifying presence of speech associated with a mobile device includes obtaining a plurality of audio samples from the mobile device while the mobile device operates in a mode distinct from a voice call operating mode, generating spectrogram data from the plurality of audio samples, and determining whether the plurality of audio samples include information indicative of speech by classifying the spectrogram data.

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11-04-2013 дата публикации

HYBRID AUDIO ENCODER AND HYBRID AUDIO DECODER

Номер: US20130090929A1
Принадлежит:

Provided are a new hybrid audio decoder and a new hybrid audio encoder having block switching for speech signals and audio signals. Currently, very low bitrate audio coding methods for speech and audio signal are proposed. These audio coding methods cause very long delay. Generally, in coding an audio signal, algorithm delay tends to be long to achieve higher frequency resolution. In coding a speech signal, the delay needs to be reduced because the speech signal is used for telecommunication. To balance fine coding quality for these two kinds of input signals with very low bitrate, this invention provides a combination of a low delay filter bank like AAC-ELD and a CELP coding method. 1. A hybrid audio decoder which decodes a coded stream while switching between a speech coding mode in which linear prediction coefficients are used and an audio coding mode in which a low delay orthogonal transform is used , the hybrid audio decoder comprising:a low delay transform decoder which decodes a coded signal in the audio coding mode using an inverse low delay filter bank, to generate a synthesized signal;an audio decoder which decodes, in the speech coding mode, a coded signal including the linear prediction coefficients, to generate an audio synthesized signal; anda block switcher which decodes a signal of a portion of a current frame to be decoded, using a signal of a previous frame preceding the current frame, and combines the decoded signal of the portion of the current frame and the audio synthesized signal of another portion of the current frame generated by the audio decoder, to reconstruct a signal of the current frame, when the current frame is a frame to be decoded immediately before the audio coding mode in which the low delay orthogonal transform is used is switched to the speech coding mode in which the linear prediction coefficients are used.2. The hybrid audio decoder according to claim 1 ,wherein the block switcher decodes the signal of the portion of the ...

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11-04-2013 дата публикации

Apparatus and method for processing an input audio signal using cascaded filterbanks

Номер: US20130090933A1

An apparatus for processing an input audio signal relies on a cascade of filterbanks, the cascade having a synthesis filterbank for synthesizing an audio intermediate signal from the input audio signal, the input audio signal being represented by a plurality of first subband signals generated by an analysis filterbank, wherein a number of filterbank channels of the synthesis filterbank is smaller than a number of channels of the analysis filterbank. The apparatus furthermore has a further analysis filterbank for generating a plurality of second subband signals from the audio intermediate signal, wherein the further analysis filterbank has a number of channels being different from the number of channels of the synthesis filterbank, so that a sampling rate of a subband signal of the plurality of second subband signals is different from a sampling rate of a first subband signal of the plurality of first subband signals.

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11-04-2013 дата публикации

Apparatus and method for generating a synthesis audio signal and for encoding an audio signal

Номер: US20130090934A1

An apparatus for generating a synthesis audio signal using a patching control signal has a first converter, a spectral domain patch generator, a high frequency reconstruction manipulator and a combiner. The first converter is configured for converting a time portion of an audio signal into a spectral representation. The spectral domain patch generator is configured for performing a plurality of different spectral domain patching algorithms, wherein each patching algorithm generates a modified spectral representation having spectral components in an upper frequency band derived from corresponding spectral components in a core frequency band of the audio signal. The spectral domain patch generator is furthermore configured to select a first spectral domain patching algorithm from the plurality of patching algorithms for a first time portion and a second spectral domain patching algorithm from the plurality of patching algorithm for a second different time portion in accordance with the patching control signal to obtain the modified spectral representation.

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18-04-2013 дата публикации

Method and Apparatus for Generating Sideband Residual Signal

Номер: US20130094655A1
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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18-04-2013 дата публикации

METHODS FOR WATERMARKING MEDIA DATA

Номер: US20130096705A1
Принадлежит:

Methods are provided for encoding watermark information into media data containing a series of digital samples in a sample domain. The methods involves: dividing the series of digital samples into a plurality of sections in the sample domain, each section comprising a corresponding plurality of samples; processing the corresponding plurality of samples in each section to obtain a single energy value associated with each section; grouping the sections into groups, each group containing three or more sections; for each group, assigning a nominal bit value according to a bit assignment rule, assigning a watermark bit value and comparing the watermark bit value to the nominal bit value. If the nominal bit value and the watermark bit value do not match, modifying one or more energy values of one or more corresponding sections in the group where re-application of the bit assignment rule would assign the watermark bit value to the group. 1. A method for encoding watermark information into media data containing a series of digital samples in a sample domain , the method comprising:dividing the series of digital samples into a plurality of sections in the sample domain, each section comprising a corresponding plurality of samples;processing the corresponding plurality of samples in each section to obtain a single energy value associated with each section;grouping the sections into groups, each group containing three or more sections;assigning a nominal bit value to each group according to a bit assignment rule, the bit assignment rule based on the energy values of the sections in the group;assigning a watermark bit value to each group;for each group, comparing the watermark bit value to the nominal bit value and, if the nominal bit value and the watermark bit value of the watermark information bit do not match, modifying one or more energy values of one or more corresponding sections in the group such that re-application of the bit assignment rule would assign the watermark ...

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18-04-2013 дата публикации

AUDIO ADJUSTMENT SYSTEM

Номер: US20130096926A1
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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18-04-2013 дата публикации

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

Номер: US20130096927A1
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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18-04-2013 дата публикации

METHOD AND APPARATUS FOR PROCESSING AN AUDIO SIGNAL

Номер: US20130096928A1
Принадлежит:

The present invention relates to a method for processing an audio signal, comprising: determining bandwidth information indicating to which of a plurality of bands the current frame corresponds; determining information on the order corresponding to the present frame on the basis of the bandwidth information; performing a linear predictive analysis of the present frame to generate a first set linear predictive transform coefficient of a first order; performing a vector quantization on the first set linear predictive coefficient to generate a first index; performing a linear predictive analysis of the current frame to generate a second set linear predictive transform coefficient of a second order in accordance with the information on the order; and performing a vector quantization on a second set difference by using the first set index and the second set linear predictive transform coefficient, when the second set linear predictive coefficient is generated. 1. A method of processing an audio signal , comprising the steps of:{'sup': st', 'nd, 'determining bandwidth information indicating that a current frame corresponds to which one among a plurality of bands including a 1band and a 2band by performing a spectrum analysis on the current frame of the audio signal;'}determining order information corresponding to the current frame based on the bandwidth information;{'sup': st', 'st, 'generating a 1set linear-predictive transform coefficient of a 1order by performing a linear-predictive analysis on the current frame;'}{'sup': st', 'st, 'generating a 1set index by vector-quantizing the 1set linear-predictive transform coefficient;'}{'sup': nd', 'nd, 'generating a 2set linear-predictive transform coefficient of a 2order in accordance with the order information by performing the linear-predictive analysis on the current frame; and'}{'sup': nd', 'nd', 'st', 'nd, 'if the 2set linear-predictive transform coefficient is generated, performing a vector-quantization on a 2set ...

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18-04-2013 дата публикации

METHOD AND APPARATUS FOR SEARCHING IN A LAYERED HIERARCHICAL BIT STREAM FOLLOWED BY REPLAY, SAID BIT STREAM INCLUDING A BASE LAYER AND AT LEAST ONE ENHANCEMENT LAYER

Номер: US20130096929A1
Автор: JAX Peter, Kordon Sven
Принадлежит: THOMSON LICENSING

A two-layer hierarchical audio bit stream can have a frame-based structure for the base layer bit stream and can be decoded independently from a higher layer and the decoding can start following every sync header. In the extension layer bit stream the frame structure may not be reflected on bit stream level. To facilitate seek operations with such highly compressed extension-layer data, the header of the extension layer bit stream comprises an FAT table with seek target positions. Because there are fewer entry points in the enhancement layer than sync headers in the base layer, a re-synchronisation and some base layer frames are required to start decoding of the enhancement layer and to generate the full audio quality. Three seeking ways of seeking are described, of which each one offers a different compromise between seeking accuracy, re-synchronisation latency and audio quality. 12-. (canceled)3. A method for searching in a layered hierarchical audio or video hit stream followed by replay , said layered bit stream including a base layer which can be decoded separately starting from base layer entry points , and including at least one enhancement layer which cannot be replayed without re-synchronized data from said base layer and which has fewer entry points than said base layer , said method comprising the steps:from an enhancement layer entry point located directly prior to a desired base layer entry point, starting a partial decoding of the related enhancement layer data, followed by re-synchronization of the related enhancement layer data and, partially in parallel, starting a muted base layer decoding;upon said re-synchronization being carried out, starting from the following base layer entry point, which needs not be an enhancement layer entry point, decoding of the enhancement layer data and decoding of the base layer data, and combining the decoded base layer data and the decoded enhancement layer data so as to output a full-quality audio or video signal.4. ...

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18-04-2013 дата публикации

Multi-Resolution Switched Audio Encoding/Decoding Scheme

Номер: US20130096930A1
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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25-04-2013 дата публикации

METHODS AND APPARATUS FOR AUDIO WATERMARKING A SUBSTANTIALLY SILENT MEDIA CONTENT PRESENTATION

Номер: US20130103172A1
Принадлежит:

Methods and apparatus for audio watermarking a substantially silent media content presentation are disclosed. An example method to audio watermark a media content presentation disclosed herein comprises obtaining a watermarked noise signal comprising a watermark and a noise signal having energy substantially concentrated in an audible frequency band, the watermarked noise signal attenuated to be substantially inaudible without combining with a separate audio signal, associating the watermarked noise signal with a substantially silent content component of the media content presentation, the media content presentation comprising one or more media content components, and outputting the watermarked noise signal during presentation of the substantially silent content component. 1obtaining a watermarked noise signal comprising a watermark and a noise signal having energy substantially concentrated in an audible frequency band, the watermarked noise signal attenuated to be substantially inaudible without combining with a separate audio signal;associating the watermarked noise signal with a substantially silent content component of the media content presentation, the media content presentation comprising one or more media content components; andoutputting the watermarked noise signal during presentation of the substantially silent content component.. A method to audio watermark a media content presentation, the method comprising: This patent arises from a continuation of U.S. application Ser. No. 12/750,359, entitled “METHODS AND APPARATUS FOR AUDIO WATERMARKING A SUBSTANTIALLY SILENT MEDIA CONTENT PRESENTATION” and filed on Mar. 30, 2010, which is hereby incorporated by reference in its entirety.This disclosure relates generally to audio watermarking and, more particularly, to methods and apparatus for audio watermarking a substantially silent media content presentation.Audio watermarking is a common technique used to identify media content, such as television broadcasts, ...

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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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25-04-2013 дата публикации

Device and method for efficiently encoding quantization parameters of spectral coefficient coding

Номер: US20130103394A1
Принадлежит: Panasonic Corp

This invention introduces apparatus and methods to efficiently encode the quantization parameters of split multi-rate lattice vector quantization. In this invention, by doing spectral analysis on the split multi-rate vector quantized spectrum, the spectrum is split to null vectors region and non-null vectors region. For the null vectors region, instead of transmitting series of indication for null vectors, an indication of null vectors region and the quantized value of index of the ending vector in the null vectors region (or the number of the null vectors in the null vectors region) are transmitted. The indication of null vectors region can be designed in many ways, the only requirement is the indication should be distinguishable in the decoder side. The ending index or the number of null vectors can be quantized by an adaptively designed codebook. By applying of the invented method, some bits can be saved from the codebook indications.

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25-04-2013 дата публикации

METHOD AND APPARATUS FOR PROCESSING AN AUDIO SIGNAL

Номер: US20130103407A1
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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02-05-2013 дата публикации

Encoding method, decoding method, encoding device, decoding device, program, and recording medium

Номер: US20130106626A1
Принадлежит: Nippon Telegraph and Telephone Corp

A plurality of samples are vector-quantized to obtain a vector quantization index and quantized values; bits are assigned in a predetermined order of priority based on auditory perceptual characteristics to one or more sets of sample positions among a plurality of sets of sample positions, each set having a plurality of sample positions and being given an order of priority based on the auditory perceptual characteristics, the number of bits not being larger than the number of bits obtained by subtracting the number of bits used for a code corresponding to the vector quantization index from the number of bits assigned for the code corresponding to the vector quantization index; and index information indicating a group of coefficients that minimizes the sum of the error between the value of each sample included in each of the sets of sample positions to which the bits are assigned and the value obtained by multiplying the quantized value of each sample included in the set of sample positions by a coefficient corresponding to the position of the sample, of all the sample positions included in the set of sample positions, is output.

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02-05-2013 дата публикации

Adding Second Enhancement Layer to CELP Based Core Layer

Номер: US20130110507A1
Автор: GAO Yang
Принадлежит: Huawei Technologies Co., Ltd.

In an embodiment, a method of transmitting an input audio signal is disclosed. A first coding error of the input audio signal with a scalable codec having a first enhancement layer is encoded, and a second coding error is encoded using a second enhancement layer after the first enhancement layer. Encoding the second coding error includes coding fine spectrum coefficients of the second coding error to produce coded fine spectrum coefficients, and coding a spectral envelope of the second coding error to produce a coded spectral envelope. The coded fine spectrum coefficients and the coded spectral envelope are transmitted. 1. A method of transmitting an input audio signal with a scalable codec , the method comprising:encoding a low frequency band signal having an inner core layer coding;encoding a first coding error of the inner core layer coding having a first enhancement layer on a same low frequency band;encoding a second coding error of the first enhancement layer by using a second enhancement layer on the same low frequency band after the first enhancement layer, encoding the second coding error comprising coding fine spectrum coefficients of the second coding error to produce coded fine spectrum coefficients, and coding a spectral envelope of the second coding error to produce a coded spectral envelope; andtransmitting the coded fine spectrum coefficients and the coded spectral envelope.2. The method of claim 1 , wherein the scalable codec comprises an inner core layer of code excited linear prediction (CELP) codec.3. The method of claim 1 , wherein:the first enhancement layer comprises a first modified discrete cosine transform (MDCT) enhancement layer; andthe second enhancement layer comprises a second MDCT enhancement layer.4. The method of claim 3 , further comprising compensating missing subbands of the first MDCT enhancement layer before encoding the second coding error using the second MDCT enhancement layer.5. The method of claim 2 , wherein:the first ...

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02-05-2013 дата публикации

Energy lossless-encoding method and apparatus, audio encoding method and apparatus, energy lossless-decoding method and apparatus, and audio decoding method and apparatus

Номер: US20130110522A1
Автор: Eun-mi Oh, Ki-hyun Choo
Принадлежит: SAMSUNG ELECTRONICS CO LTD

A lossless encoding method is provided that includes determining a lossless encoding mode of a quantization coefficient as one of an infinite-range lossless encoding mode and a finite-range lossless encoding mode; encoding the quantization coefficient in the infinite-range lossless encoding mode in correspondence with a result of the lossless encoding mode determination; and encoding the quantization coefficient in the finite-range lossless encoding mode in correspondence with a result of the lossless encoding mode determination.

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02-05-2013 дата публикации

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

Номер: US20130110523A1

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:identification information for each of the audio signals; andchannel information for the audio signals.2. The apparatus of claim 1 , 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.3. The apparatus of claim 1 , wherein the supplementary information further includes preset information for the audio signals.4. The apparatus of claim 3 , 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.5. The apparatus of claim 1 , wherein the supplementary information further ...

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09-05-2013 дата публикации

Method and apparatus for estimating interchannel delay of sound signal

Номер: US20130114817A1
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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09-05-2013 дата публикации

METHODS AND APPARATUS TO PERFORM AUDIO WATERMARKING AND WATERMARK DETECTION AND EXTRACTION

Номер: US20130114831A1
Принадлежит:

Encoding and decoding methods and apparatus as described. An example method of obtaining auxiliary information in an audio signal using a plurality of frequency components residing in a plurality of code bands comprises transforming an audio signal into a frequency domain representation; determining characteristic of frequencies of the frequency domain representation that may contain the auxiliary information; normalizing across the code bands the characteristics of frequencies of the frequency domain representation in a respective one of the code bands that may contain the auxiliary information, wherein the normalization is carried out against a characteristic of a frequency in that code band; summing the normalized characteristics of the frequencies representative of auxiliary information to determine a sum for a frequency representative of auxiliary information; and determining that the sum is representative of the auxiliary information. 1. A method of obtaining auxiliary information in an audio signal using a plurality of frequency components residing in a plurality of code bands , the method comprising:transforming an audio signal into a frequency domain representation;determining characteristic of frequencies of the frequency domain representation that may contain the auxiliary information;normalizing across the code bands the characteristics of frequencies of the frequency domain representation in a respective one of the code bands that may contain the auxiliary information, wherein the normalization is carried out against a characteristic of a frequency in that code band;summing the normalized characteristics of the frequencies representative of auxiliary information to determine a sum for a frequency representative of auxiliary information; anddetermining that the sum is representative of the auxiliary information.2. A method as defined in claim 1 , wherein different sets of frequency components represent respectively different information and wherein one ...

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09-05-2013 дата публикации

Audio signal decoder, audio signal encoder, method for decoding an audio signal, method for encoding an audio signal and computer program using a pitch-dependent adaptation of a coding context

Номер: US20130117015A1

An audio signal decoder includes a context-based spectral value decoder configured to decode a codeword describing one or more spectral values or at least a portion of a number representation thereof in dependence on a context state. The audio signal decoder also includes a context state determinator configured to determine a current context state in dependence on one or more previously decoded spectral values and a time warping frequency-domain-to-time-domain converter configured to provide a time-warped time-domain representation of a given audio frame on the basis of a set of decoded spectral values provided by the context-based spectral value decoder and in dependence on the time warp information. The context-state determinator is configured to adapt the determination of the context state to a change of a fundamental frequency between subsequent audio frames. An audio signal encoder applies a comparable concept.

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09-05-2013 дата публикации

APPARATUS AND METHOD FOR CODING SIGNAL IN A COMMUNICATION SYSTEM

Номер: US20130117028A1
Автор: KIM Hyun-woo

Disclosed are an apparatus for coding a signal in a communication system including: a coding unit configured to code voice and audio signals based on a code excited linear prediction (CELP) coding method; a residual signal calculation unit configured to calculate residual signals of the voice and audio signals; a frequency transform unit configured to transform the residual signal into a signal in a frequency domain; an energy calculation unit configured to use frequency coefficients of the residual signals to calculate frequency energy of the residual signals; an energy concentration calculation unit configured to calculate energy concentrations of each vector dimension of the residual signals from the frequency energy of the residual signals; and a vector dimension determination unit configured to compare the energy concentrations of each vector dimension to determine targeted vector dimensions of the residual signals. 1. An apparatus for coding a signal in a communication system , comprising:a coding unit configured to code voice and audio signals based on a code excited linear prediction (CELP) coding method;a residual signal calculation unit configured to calculate residual signals of the voice and audio signals;a frequency transform unit configured to transform the residual signal into a signal in a frequency domain;an energy calculation unit configured to use frequency coefficients of the residual signals to calculate frequency energy of the residual signals;an energy concentration calculation unit configured to calculate energy concentrations of each vector dimension of the residual signals from the frequency energy of the residual signals; anda vector dimension determination unit configured to compare the energy concentrations of each vector dimension to determine targeted vector dimensions of the residual signals.2. The apparatus of claim 1 , wherein the vector dimension determination unit determines the vector dimensions having a maximum value as the ...

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09-05-2013 дата публикации

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

Номер: US20130117029A1
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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09-05-2013 дата публикации

Signal compression method and apparatus

Номер: US20130117030A1
Принадлежит: Huawei Technologies Co Ltd

A signal compression method and apparatus are provided. The signal compression method includes: multiplying an input signal by a window function; calculating original autocorrelation coefficients of a windowed input signal; calculating a white-noise correction factor or a lag-window according to the original autocorrelation coefficients, and calculating modified autocorrelation coefficients according to the original autocorrelation coefficients, the white-noise correction factor and the lag-window; calculating linear prediction coefficients according to the modified autocorrelation coefficients; and outputting a coded bit stream according to the linear prediction coefficients.

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16-05-2013 дата публикации

Method and apparatus for generating noises

Номер: US20130124196A1
Автор: Jinliang Dai, Libin Zhang
Принадлежит: Huawei Technologies Co Ltd

A method and an apparatus for generating comfortable noises so as to improve user experience are disclosed. The method includes: if a received data frame is a noise frame, calculating a corresponding energy attenuation parameter based on the noise frame and a data frame received earlier than the noise frame; and attenuating noise energy based on the energy attenuation parameter to obtain a comfortable noise signal. An apparatus for generating comfortable noise is also provided.

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16-05-2013 дата публикации

PULSE ENCODING AND DECODING METHOD AND PULSE CODEC

Номер: US20130124199A1
Автор: Ma Fuwei, Zhang Dejun
Принадлежит: Huawei Technologies Co., Ltd.

In a pulse encoding and decoding method and a pulse codec, more than two tracks are jointly encoded, so that free codebook space in the situation of single track encoding can be combined during joint encoding to become code bits that may be saved. Furthermore, a pulse that is on each track and required to be encoded is combined according to positions, and the number of positions having pulses, distribution of the positions that have pulses on the track, and the number of pulses on each position that has a pulse are encoded separately, so as to avoid separate encoding performed on multiple pulses of a same position, thereby further saving code bits. 1. A pulse encoding method , comprising:obtaining pulses that are on T tracks and required to be encoded, wherein T is an integer greater than or equal to 2;{'sub': 't', 'sup': 'th', 'separately collecting, according to positions, statistics about a pulse that is on each track and required to be encoded, to obtain the number Nof positions that have pulses on each track, distribution of the positions that have pulses on the track, and the number of pulses on each position that has a pulse, wherein the subscript t represents a ttrack, and tε[0, T−1];'}{'sub': 0', '1', 'T-1, 'b': 1', '1, 'according to the number {N, N, . . . , N} of positions that have pulses and are on each track, determining a first index I, wherein the first index I corresponds to all possible distribution situations of positions that have pulses and are on each track under the number of the positions having pulses, wherein the number of the positions having pulses is represented by it;'}{'b': 2', '1, 'sub': 't', 'determining a second index Iof each track separately according to distribution of positions that have pulses and are on each track, wherein the second index indicates, among all possible distribution situations corresponding to the first index I, a distribution situation which corresponds to distribution of current positions having pulses on a ...

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16-05-2013 дата публикации

DECODING DEVICE, ENCODING DEVICE, AND METHODS FOR SAME

Номер: US20130124201A1
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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16-05-2013 дата публикации

SIGNAL PROCESSING APPARATUS AND METHOD, AND PROGRAM

Номер: US20130124214A1
Принадлежит:

A method, system, and computer program product for processing an encoded audio signal is described. In one exemplary embodiment, the system receives an encoded low-frequency range signal and encoded energy information used to frequency shift the encoded low-frequency range signal. The low-frequency range signal is decoded and an energy depression of the decoded signal is smoothed. The smoothed low-frequency range signal is frequency shifted to generate a high-frequency range signal. The low-frequency range signal and high-frequency range signal are then combined and outputted. 1. A computer-implemented method for processing an audio signal , the method comprising:receiving an encoded low-frequency range signal corresponding to the audio signal;decoding the encoded signal to produce a decoded signal having an energy spectrum of a shape including an energy depression;performing filter processing on the decoded signal, the filter processing separating the decoded signal into low-frequency range band signals;performing a smoothing process on the decoded signal, the smoothing process smoothing the energy depression of the decoded signal;performing a frequency shift on the smoothed decoded signal, the frequency shift generating high-frequency range band signals from the low-frequency range band signals;combining the low-frequency range band signals and the high-frequency range band signals to generate an output signal; andoutputting the output signal.2. A computer-implemented method as in claim 1 , wherein the encoded signal further comprises energy information for the low-frequency range band signals.3. A computer-implemented method as in claim 2 , wherein performing the frequency shift is based on the energy information for the low-frequency range band signals.4. A computer-implemented method as in claim 1 , wherein the encoded signal further comprises SBR (spectral band replication) information for the high-frequency range bands of the audio signal.5. A computer- ...

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16-05-2013 дата публикации

CODER USING FORWARD ALIASING CANCELLATION

Номер: US20130124215A1

A codec supporting switching between time-domain aliasing cancellation transform coding mode and time-domain coding mode is made less liable to frame loss by adding a further syntax portion to the frames, depending on which the parser of the decoder may select between a first action of expecting the current frame to have, and thus reading forward aliasing cancellation data from the current frame and a second action of not-expecting the current frame to have, and thus not reading forward aliasing cancellation data from the current frame. In other words, while a bit of coding efficiency is lost due to the provision of the new syntax portion, it is merely the new syntax portion which provides for the ability to use the codec in case of a communication channel with frame loss. Without the new syntax portion, the decoder would not be capable of decoding any data stream portion after a loss and will crash in trying to resume parsing. Thus, in an error prone environment, the coding efficiency is prevented from vanishing by the introduction of the new syntax portion. 1. Decoder for decoding a data stream comprising a sequence of frames into which time segments of an information signal are coded , respectively , comprisinga parser configured to parse the data stream , wherein the parser is configured to, in parsing the data stream, read a first syntax portion and a second syntax portion from a current frame; anda reconstructor configured to reconstruct a current time segment of the information signal associated with the current frame based on information acquired from the current frame by the parsing, using a first selected one of a Time-Domain Aliasing Cancellation transform decoding mode and a time-domain decoding mode, the first selection depending on the first syntax portion,wherein the parser is configured to, in parsing the data stream, perform a second selected one of a first action of expecting the current frame to comprise, and thus reading forward aliasing ...

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23-05-2013 дата публикации

Audio Signal Synthesizer

Номер: US20130129096A1
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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23-05-2013 дата публикации

APPARATUS FOR PROCESSING AN AUDIO SIGNAL AND METHOD THEREOF

Номер: US20130132097A1
Автор: Oh Hyen-O
Принадлежит: LG ELECTRONICS INC.

An apparatus for processing an audio signal and method thereof are disclosed. The present invention includes receiving a downmix signal and side information; extracting extension type identifier indicating whether extension area includes a residual signal from the side information; when the extension type identifier indicates that the extension area includes the residual signal, extracting control restriction information for residual using mode from the side information; receiving control information for controlling gain or panning of at least one object signal; estimating modified control information based on the control information and the control restriction information; obtaining at least one of enhanced object signal and one or more regular object signals from the downmix signal using the residual signal; and, generating an output signal using the modified control information and at least one of enhanced object signal and one or more regular object signal, wherein the control restriction information for residual using mode relates to a parameter indicating limiting degree of the control information in case of the residual using mode. 1. A method for processing an audio signal , comprising:receiving a downmix signal and side information;extracting extension type identifier indicating whether extension area includes a residual signal from the side information;when the extension type identifier indicates that the extension area includes the residual signal, extracting control restriction information for residual using mode from the side information;receiving control information for controlling gain or panning of at least one object signal;estimating modified control information based on the control information and the control restriction information;obtaining at least one of enhanced object signal and one or more regular object signals from the downmix signal using the residual signal; and,generating an output signal using the modified control information and at ...

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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

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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23-05-2013 дата публикации

CODING DEVICE, DECODING DEVICE, AND METHODS THEREOF

Номер: US20130132099A1
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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23-05-2013 дата публикации

APPARATUS AND METHOD FOR CODEC SIGNAL IN A COMMUNICATION SYSTEM

Номер: US20130132100A1

The present invention relates to a codec apparatus and method for coding/decoding speech and audio signals in a communication system. In accordance with the present invention, a speech and audio signal in a time domain is transformed into a speech and audio signal in a frequency domain and calculating frequency coefficients of the speech and audio signal, the frequency coefficients are split by a plurality of sub-bands and the sub-band coefficients of the respective sub-bands are calculated from the frequency coefficients, and the sub-band coefficients are quantized depending on a characteristic of the plurality of sub-bands and sub-band quantization indices are calculated by quantizing the sub-band coefficients. 1. A codec apparatus for coding a signal in a communication system , the codec apparatus comprising:a transformer configured to transform a speech and audio signal in a time domain into a speech and audio signal in a frequency domain and calculate frequency coefficients of the speech and audio signal;a band splitter configured to split the frequency coefficients by a plurality of sub-bands and calculate sub-band coefficients of the respective sub-bands from the frequency coefficients; anda sub-band coefficient quantizer configured to quantize the sub-band coefficients depending on a characteristic of the plurality of sub-bands and calculate sub-band quantization indices by quantizing the sub-band coefficients.2. The codec apparatus of claim 1 , wherein the sub-band coefficient quantizer comprises:a mode selector configured to calculate a quantization mode value by taking the characteristic of the plurality of sub-bands into consideration;a first quantizer configured to quantize the sub-band coefficients based on the quantization mode value and generate gain-shape indices as the sub-band quantization indices; anda second quantizer configured to quantize the sub-band coefficients based on the quantization mode value and generate track-pulse indices as the sub ...

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