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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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Форма поиска

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 42789. Отображено 100.
10-12-2003 дата публикации

УСТРОЙСТВО ДЛЯ АНАЛИЗА КЛИППИРОВАННЫХ РЕЧЕВЫХ СИГНАЛОВ

Номер: RU0000034795U1

Устройство для анализа клиппированных речевых сигналов, содержащее первый канал, включающий в себя первый счетчик и оперативное запоминающее устройство, информационный вход которого соединен с выходом первого счетчика, вход управления записью оперативного запоминающего устройства является первым управляющим входом канала, отличающееся тем, что в него дополнительно введены второй канал идентичный первому, два блока статистического анализа, элементы И и ИЛИ, триггер, инвертор, в состав каждого канала дополнительно введены второй счетчик, элемент задержки, одновибратор и два канальных элемента И, первый вход первого канального элемента И которого является вторым управляющим входом канала, входом обнуления которого является обнуляющий вход второго счетчика, суммирующий вход которого объединен с первым входом второго канального элемента И, входом одновибратора и является информационным входом канала, разрешающим входом канала является второй вход второго канального элемента И, выход которого соединен с суммирующим входом первого счетчика, обнуляющий вход которого через элемент задержки подключен к выходу одновибратора, который соединен со вторым входом первого канального элемента И, выход которого соединен со входом выбора кристалла оперативного запоминающего устройства, адресный вход которого соединен с выходом второго счетчика, выход переноса которого является управляющим выходом канала, информационным выходом которого является выход оперативного запоминающего устройства, тактовым входом канала является третий вход второго канального элемента И, информационные выходы каналов подключены к единой шине данных, к которой также подключены информационные входы двух блоков статистического анализа, тактовые входы первого и второго каналов объединены и являются тактовым входом устройства, разрешающие входы первого и второго каналов объединены и подключены к выходу триггера, обнуляющий вход которого соединен с выходом элемента ИЛИ, первый и второй входы которого соединены с ...

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27-04-2008 дата публикации

ДЕКОДИРУЮЩЕЕ УСТРОЙСТВО

Номер: RU0000072780U1

Декодирующее устройство, содержащее блок коррекции высокочастотных предыскажений, частотный детектор, блок коррекции низкочастотных предыскажений, блок задержки на строку и коммутатор, отличающееся тем, что, с целью расширения полосы сигнала яркости при заданной степени подавления спектра сигнала цветности, оно дополнительно содержит режекторный фильтр с двумя управляющими входами, один из которых соединен через фильтр нижних частот с выходом частотного детектора, а второй вход соединен с выходом того же частотного детектора через дифференциатор, вычислитель модуля и фильтр нижних частот, и, кроме того, в канал цветности дополнительно введены гребенчатые вертикальные фильтры, состоящие из блоков задержки на строку и сумматоров, а в канал яркости введены фильтры нижних и верхних частот, блок подавления колебательности, гребенчатый вертикальный фильтр второго порядка, состоящий из блоков задержки на строку и сумматора, а также выходной сумматор канала яркости. РОССИЙСКАЯ ФЕДЕРАЦИЯ (19) RU (11) 72 780 (13) U1 (51) МПК G10L 19/00 (2006.01) ФЕДЕРАЛЬНАЯ СЛУЖБА ПО ИНТЕЛЛЕКТУАЛЬНОЙ СОБСТВЕННОСТИ, ПАТЕНТАМ И ТОВАРНЫМ ЗНАКАМ (12) ОПИСАНИЕ ПОЛЕЗНОЙ МОДЕЛИ К ПАТЕНТУ (21), (22) Заявка: 2007148313/22 , 26.12.2007 (24) Дата начала отсчета срока действия патента: 26.12.2007 (45) Опубликовано: 27.04.2008 (73) Патентообладатель(и): ОАО Всероссийский научно-исследовательский институт телевидения и радиовещания (RU) U 1 7 2 7 8 0 R U Ñòðàíèöà: 1 U 1 Формула полезной модели Декодирующее устройство, содержащее блок коррекции высокочастотных предыскажений, частотный детектор, блок коррекции низкочастотных предыскажений, блок задержки на строку и коммутатор, отличающееся тем, что, с целью расширения полосы сигнала яркости при заданной степени подавления спектра сигнала цветности, оно дополнительно содержит режекторный фильтр с двумя управляющими входами, один из которых соединен через фильтр нижних частот с выходом частотного детектора, а второй вход соединен с выходом того же частотного ...

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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

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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12-01-2012 дата публикации

Audio adjusting device

Номер: US20120010737A1
Автор: Kei Sakagami, Shiro Suzuki
Принадлежит: Pioneer Corp

The audio adjusting device is preferable applied to an audio system, and includes an audio analyzing unit, a control unit and an adjusting unit. The audio analyzing unit analyzes audio based on the inputted audio signals on the real time basis. The control unit generates the control signal for adjusting the audio signals based on the analysis information of the audio analyzed by the analyzing unit and the volume level instructed by the user. The adjusting unit adjusts and outputs the audio signals based on the control signal in terms of at least one of spreading feeling, speech clearness and bass volume feeling. The control unit varies the adjusting amount by the adjusting unit based on the signal level of the audio signals and the volume level.

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26-01-2012 дата публикации

Data transferring method

Номер: US20120023275A1
Автор: Akiyoshi Yamashita
Принадлежит: Panasonic Corp

A first device ( 1 ) includes a first interface ( 4 ) which transmits and receives, to and from a rotating recording medium, data in data blocks in a size equal to the integral multiple of the sector size of the recording medium, and transmits and receives, to and from a second device ( 2 ), in packets, the data transferred from or to the recording medium. The second device ( 2 ) includes a second interface ( 5 ) which is connected to the first interface ( 4 ) and transmits and receives data in packets to and from the first device ( 1 ). In transferring data between the first device ( 1 ) and the second device ( 2 ), data is transferred in the integral ratio of sectors to packets set such that the number of packets is smaller than the number of packets required for data transfer for one sector.

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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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23-02-2012 дата публикации

Apparatus and method for improving communication quality in mobile terminal

Номер: US20120046943A1
Принадлежит: SAMSUNG ELECTRONICS CO LTD

An apparatus and a method for voice communication of a mobile terminal are provided. More particularly, an apparatus and a method for clearly receiving a counterpart user's voice signal in a mobile terminal positioned at a place where a noise occurs are provided. The apparatus includes an input unit, an extension signal generator, and an adder. The input unit receives a voice signal. The extension signal generator generates, based on a voice signal received via the input unit, a harmonics signal corresponding to a frequency band that represents a reaction sensitive to a sense of hearing. The adder merges the generated harmonics signal with the received voice signal.

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08-03-2012 дата публикации

Apparatus for determining a spatial output multi-channel audio signal

Номер: US20120057710A1
Принадлежит: Individual

An apparatus for determining a spatial output multi-channel audio signal based on an input audio signal and an input parameter. The apparatus includes a decomposer for decomposing the input audio signal based on the input parameter to obtain a first decomposed signal and a second decomposed signal different from each other. Furthermore, the apparatus includes a renderer for rendering the first decomposed signal to obtain a first rendered signal having a first semantic property and for rendering the second decomposed signal to obtain a second rendered signal having a second semantic property being different from the first semantic property. The apparatus comprises a processor for processing the first rendered signal and the second rendered signal to obtain the spatial output multi-channel audio signal.

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15-03-2012 дата публикации

Mobile electronic device and sound playback method thereof

Номер: US20120063607A1
Автор: Hann-Shi Tong, Lei Chen
Принадлежит: HTC Corp

A mobile electronic device and a sound playback method thereof are provided. The mobile electronic device includes a sensor, a speaker, and a controller coupled to the sensor and the speaker. The sensor detects whether the speaker is blocked or not. When the speaker is blocked, the controller multiplies a sound signal by a transfer function and then outputs the multiplied sound signal. The speaker plays the sound signal outputted by the controller. The transfer function changes the direction in which the speaker plays the sound signal.

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22-03-2012 дата публикации

Terminal device, mobile terminal, and navigation program

Номер: US20120069711A1
Принадлежит: Fujitsu Ltd

A terminal device includes an orientation calculating unit that calculates the orientation of a device with respect to the target. Furthermore, the terminal device also includes a degree-of-processing determining unit that determines the degree of processing related to an attribute of a sound that indicates the target in accordance with the orientation calculated by the orientation calculating unit. Furthermore, the terminal device also includes an output control unit that controls an output of a sound in accordance with the degree of processing determined by the degree-of-processing determining unit.

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22-03-2012 дата публикации

Optical disc recorder and buffer management method thereof

Номер: US20120072660A1
Принадлежит: MediaTek Inc

A buffer management method is provided. A host issues a read command requesting access for a read data block and a write command requesting recording of a write data block. A write buffer is dedicated to store the write data block. A read buffer is dedicated to store the read data block. The method comprises entering the optical disc recorder into a write loop. During the write loop, the optical disc recorder triggering a write command handling procedure in response to the write command; triggering a read command handling procedure in response to the read command; and triggering a pre-recording procedure to prepare the write data block in the write buffer for recording. Wherein contents between the write buffer and read buffer are exchangeable during the write handling procedure, the read handling procedure or the pre-recording procedure.

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29-03-2012 дата публикации

Apparatus, method, and program product for presenting moving image with sound

Номер: US20120076304A1
Автор: Kaoru Suzuki
Принадлежит: Toshiba Corp

According to one embodiment, an apparatus for presenting a moving image with sound includes an input unit, a setting unit, a main beam former unit, and an output control unit. The input unit inputs data on a moving image with sound including a moving image and a plurality of channels of sounds. The setting unit sets an arrival time difference according to a user operation, the arrival time difference being a difference in time between a plurality of channels of sounds coming from a desired direction. The main beam former unit generates a directional sound in which a sound in a direction having the arrival time difference set by the setting unit is enhanced, from the plurality of channels of sounds included in the data on the moving image with sound. The output control unit outputs the directional sound along with the moving image.

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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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03-05-2012 дата публикации

Adaptive audio transcoding

Номер: US20120109643A1
Принадлежит: Google LLC

A system and method provide an audio/video coding system for adaptively transcoding audio streams based on content characteristics of the audio streams. An audio stream metadata extraction module of the system is configured to extract metadata of a source audio stream. An audio stream classification module of the system is configured to classify the source audio stream into one of the several audio content categories based on the metadata of the source audio stream. An adaptive audio encoder of the system is configured to determine one or more transcoding parameters including target bitrate and sampling rate based on the metadata and classification of the source audio stream. An adaptive audio transcoder of the system is configured to transcode the source audio stream into an output audio stream using the transcoding parameters.

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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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14-06-2012 дата публикации

Telephone or other device with speaker-based or location-based sound field processing

Номер: US20120150542A1
Автор: Wei Ma
Принадлежит: National Semiconductor Corp

A method includes obtaining audio data representing audio content from at least one speaker. The method also includes spatially processing the audio data to create at least one sound field, where each sound field has a spatial characteristic that is unique to a specific speaker. The method further includes generating the at least one sound field using the processed audio data. The audio data could represent audio content from multiple speakers, and generating the at least one sound field could include generating multiple sound fields around a listener. The spatially processing could include performing beam forming to create multiple directional beams, and generating the multiple sound fields around the listener could include generating the directional beams with different apparent origins around the listener. The method could further include separating the audio data based on speaker, where each sound field is associated with the audio data from one of the speakers.

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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

Immersive audio rendering system

Номер: US20120170757A1
Принадлежит: SRS Labs Inc

A depth processing system can employ stereo speakers to achieve immersive effects. The depth processing system can advantageously manipulate phase and/or amplitude information to render audio along a listener's median plane, thereby rendering audio along varying depths. In one embodiment, the depth processing system analyzes left and right stereo input signals to infer depth, which may change over time. The depth processing system can then vary the phase and/or amplitude decorrelation between the audio signals over time to enhance the sense of depth already present in the audio signals, thereby creating an immersive depth effect.

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05-07-2012 дата публикации

Speech decoding apparatus for producing an excitation signal and a synthesis filter

Номер: US20120173230A1
Автор: Kimio Miseki
Принадлежит: Toshiba Corp

A wideband speech coding method comprising identifying whether an input speech signal is a narrowband signal or a wideband signal, and coding the input speech signal by controlling a predetermined parameter of a wideband speech coding process based on the identification result.

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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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16-08-2012 дата публикации

Speech signal restoration device and speech signal restoration method

Номер: US20120209611A1
Принадлежит: Mitsubishi Electric Corp

A synthesis filter 106 synthesizes a plurality of wide-band speech signals by combining wide-band phoneme signals and sound source signals from a speech signal code book 105 , and a distortion evaluation unit 107 selects one of the wide-band speech signals with a minimum waveform distortion with respect to an up-sampled narrow-band speech signal output from a sampling conversion unit 101 . A first bandpass filter 103 extracts a frequency component outside a narrow-band of the wide-band speech signal and a band synthesis unit 104 combines it with the up-sampled narrow-band speech signal.

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23-08-2012 дата публикации

Alignment and Re-Association of Metadata for Media Streams Within a Computing Device

Номер: US20120215329A1
Принадлежит: Dolby Laboratories Licensing Corp

Techniques for re-associating dynamic metadata with media data are provided. A media processing system creates, with a first media processing stage, binding information comprising dynamic metadata and a time relationship between the dynamic metadata and media data. The binding information may be derived from the media data. While the first media processing stage delivers the media data to a second media processing stage in a first data path, the first media processing stage passes the binding information to the second media processing stage in a second data path. The media processing system re-associates, with the second media processing stage, the dynamic metadata and the media data using the binding information.

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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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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

Directional Electroacoustical Transducing

Номер: US20120224729A1
Принадлежит: Bose Corp

A multichannel audio system for radiating sound to a listening area that includes a plurality of listening spaces. The audio system includes directional audio devices, positioned in a first of the listening spaces, close to a head of a listener, for radiating first sound waves corresponding to components of one of the channels and nondirectional audio devices, positioned inside the listening area and outside the listening space, distant from the listening space, for radiating sound waves corresponding to components of a second of the channels.

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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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13-09-2012 дата публикации

Bandwidth extension of a low band audio signal

Номер: US20120230515A1
Принадлежит: Telefonaktiebolaget LM Ericsson AB

Estimation of a high band extension of a low band audio signal includes the following steps: extracting (S 1 ) a set of features of the low band audio signal; mapping (S 2 ) extracted features to at least one high band parameter with generalized additive modeling; frequency shifting (S 3 ) a copy of the low band audio signal into the high band; controlling (S 4 ) the envelope of the frequency shifted copy of the low band audio signal by said at least one high band parameter.

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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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11-10-2012 дата публикации

Method of decoding content data blocks, corresponding computer program product and decoding device

Номер: US20120260143A1
Принадлежит: Canon Inc

When decoding a set of symbols to be decoded, several data blocks representative of the set of symbols to be decoded are received by a decoding node of a communications network. The data blocks are encoded using an error correction code enabling a decoding by erasure. The decoding node performs the following steps: first selecting at least one of the data blocks, first determining first erasures, and checking whether the number of the first erasures is below a given threshold. In a case the check is positive, the decoding node performs first decoding by erasure of the set of symbols to be decoded. In a case the check is negative, the decoding node performs second selecting of at least one of the data blocks, second determining second erasures, and second decoding by erasure of the set of symbols to be decoded from the second erasures.

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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

Method and a system for an acoustic curtain that reveals and closes a sound scene

Номер: US20120288122A1
Автор: Morten Lydolf
Принадлежит: Bang and Olufsen AS

A system and a method of operating the system which has at least two groups of sound providers, where the virtual position of the second group of sound providers is moved from the position of the first group of speakers to that of the second group of speakers. Also a third group of speakers may be used, the position of which is virtually moved from that of the first group via the position of the second group to that of the third group of speakers.

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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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22-11-2012 дата публикации

Systems, methods, and apparatus for wideband encoding and decoding of inactive frames

Номер: US20120296641A1
Принадлежит: Qualcomm Inc

Speech encoders and methods of speech encoding are disclosed that encode inactive frames at different rates. Apparatus and methods for processing an encoded speech signal are disclosed that calculate a decoded frame based on a description of a spectral envelope over a first frequency band and the description of a spectral envelope over a second frequency band, in which the description for the first frequency band is based on information from a corresponding encoded frame and the description for the second frequency band is based on information from at least one preceding encoded frame. Calculation of the decoded frame may also be based on a description of temporal information for the second frequency band that is based on information from at least one preceding encoded frame.

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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

Method and apparatus for reproducing three-dimensional sound

Номер: US20130010969A1
Принадлежит: SAMSUNG ELECTRONICS CO LTD

Stereophonic sound is reproduced by acquiring image depth information indicating a distance between at least one object in an image signal and a reference location, acquiring sound depth information indicating a distance between at least one sound object in a sound signal and a reference location based on the image depth information, and providing sound perspective to the at least one sound object based on the sound depth information.

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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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07-02-2013 дата публикации

Stereophonic sound apparatus for vehicle

Номер: US20130034253A1
Автор: Toshiaki Nakayama
Принадлежит: Denso Corp

A stereophonic sound apparatus for a vehicle includes ultrasonic speakers and a dual channel reproduction unit. The dual channel reproduction unit modulates a sound signal into an ultrasonic modulated sound having ultrasonic wave frequency, and provides the ultrasonic modulated sound toward a passenger through the reproduction ultrasonic speakers for generating a three-dimensional sound. The dual channel reproduction unit is configured to generate a sub localization sound through the reproduction ultrasonic speakers when generating a front localization sound, the front localization sound being a sound perceived to be generated from a position in front of a seat, and the sub localization sound being a sound perceived to be generated from a position different from the position in front of the seat.

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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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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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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

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 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

VOIP DEVICE, VOIP CONFERENCING SYSTEM, AND RELATED METHOD

Номер: US20130094653A1
Принадлежит: CLEARONE COMMUNICATIONS, INC.

Voice over internet protocol (VoIP) devices and conferencing systems may include a spatial encoder associated with a first endpoint and a spatial renderer associated with a second endpoint. The spatial renderer may configured to receive audio data. The audio data may be rendered among a plurality of speakers based on a first set of spatial information for a plurality of microphones associated with the first endpoint, and a second set of spatial information for the plurality of speakers associated with the second endpoint. A method for generating a sound field may include determining spatial information for a plurality of microphones in a local room, determining spatial information for a plurality of speakers in a remote room, mapping the spatial information for the plurality of microphones and the spatial information for the plurality of speakers, and generating a sound field in the remote room based on the mapping. 1. A voice over Internet protocol (VoIP) device , comprising:a spatial renderer associated with a second endpoint, the spatial renderer configured to receive audio data from a first endpoint, and render the audio data among a plurality of speakers based, at least in part, on a first set of spatial information for a plurality of microphones associated with the first endpoint, and a second set of spatial information for the plurality of speakers associated with the second endpoint.2. The VoIP device of claim 1 , wherein the audio data includes raw audio data.3. The VoIP device of claim 1 , wherein the audio data includes mixed audio data from a spatial encoder of the first endpoint.4. The VoIP device of claim 3 , wherein the mixed audio data includes a plurality of audio streams.5. The VoIP device of claim 1 , wherein a quantity of the plurality of microphones and a quantity of the plurality of speakers are not equal.6. The VoIP device of claim 1 , wherein a spatial configuration of the plurality of microphones and a spatial configuration of the plurality ...

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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

SELECTIVE BASS POST FILTER

Номер: US20130096912A1
Принадлежит: DOLBY INTERNATIONAL AB

In one aspect, the invention provides an audio encoding method characterized by a decision being made as to whether the device which will decode the resulting bit stream Bitstream should apply post filtering including attenuation of interharmonic noise. Hence, the decision whether to use the post filter, which is encoded in the bit stream, is taken separately from the decision as to the most suitable coding mode. In another aspect, there is provided an audio decoding method with a decoding step followed by a post-filtering step, including interharmonic noise attenuation, and being characterized in a step of disabling the post filter in accordance with post filtering information encoded in the bit stream signal. Such a method is well suited for mixed-origin audio signals by virtue of its capability to deactivate the post filter in dependence of the post filtering information only, hence independently of factors such as the current coding mode. 1. An interharmonic noise attenuation post filter adapted to receive an input signal , which comprises a preliminary audio signal decoded according to one of a plurality of decoding modes , wherein post-filter activity is conventionally associated with particular decoding modes , and to supply an output audio signal ,characterized by a control section for selectively operating the post filter in one of the following modes:i) a filtering mode, wherein it filters the preliminary audio signal to obtain a filtered signal and supplies this as output audio signal; andii) a pass-through mode, wherein it supplies the preliminary audio signal as output audio signal,said control section being configured to enter the pass-through mode in response to the value of a post-filtering signal, whereby a conventionally filtered decoding mode is applied unfiltered.2. The post filter of claim 1 , wherein the post-filtering signal is included in the input signal.3. The post filter of claim 1 , further comprising a decision module adapted to estimate ...

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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

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

Method and device for producing a downward compatible sound format

Номер: US20130108054A1
Автор: Jens Groh
Принадлежит: Institut fuer Rundfunktechnik GmbH

In order to reduce the disturbing background noises that may arise during the summation with weighting of the spectral coefficients using a correction factor in a downmix method, the proposition is made that the correction factors m(k) are computed as follows: eA ( k )=Real( A ( k ))Real( A ( k ))+Imag( A ( k ))·Imag( A ( k )) eB ( k )=Real( B ( k ))·Real( B ( k ))+Imag( B ( k ))·Imag( B ( k )) x ( k )=Real( A ( k ))·Real( B ( k ))+Imag( A ( k ))·Imag( B ( k )) w ( k )= D·x ( k )/( eA ( k )+ L·eB ( k )) m ( k )=( w ( k ) 2 +1 ) (1/2) −w ( k ) wherein m is the k th correction factor; and A(k) is the k th spectral value of the signal to be prioritized; and B(k) is the k th spectral value of the signal not to be prioritized; and D is the degree of compensation; and L is the degree of the limitation of the compensation.

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02-05-2013 дата публикации

Audio Encoder and Decoder and Methods for Encoding and Decoding an Audio Signal

Номер: US20130110506A1
Принадлежит: TELEFONAKTIEBOLAGET L M ERICSSON (PUBL)

The present invention relates to a frequency domain based method of encoding and decoding an audio signal, wherein an adaptive spectral code book is updated with synthesized frequency domain representations of a time domain signal segment. A frequency analysis is performed of a received time domain signal segment in order to obtain a frequency domain representation, and the adaptive spectral code book is searched for a first approximation of the frequency domain representation. A fixed spectral code book is searched for an approximation of the residual frequency representation. A synthesized frequency domain representation may be generated from the two approximations. 148-. (canceled)49. A method of encoding an audio signal , the method comprising:receiving, in an audio encoder, a time domain signal segment originating from the audio signal;performing, in the audio encoder, a frequency analysis of the time domain signal segment so as to obtain a frequency domain representation of the signal segment;searching an adaptive spectral code book of the audio encoder for an adaptive spectral code book vector which provides a first approximation of the frequency domain representation, the adaptive spectral code book comprising a plurality of adaptive spectral code book vectors;selecting the adaptive spectral code book vector providing a first approximation;generating a residual frequency representation from a difference between the frequency domain representation and the selected adaptive spectral code book vector;searching a fixed spectral code book of the audio encoder for a fixed spectral code book vector which provides an approximation of the residual frequency representation, the fixed spectral code book comprising a plurality of fixed spectral code book vectors;selecting the fixed spectral code book vector providing an approximation of the residual frequency representation;updating the adaptive spectral code book of the audio encoder by including a vector obtained as a ...

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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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