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Небесная энциклопедия

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

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Мониторинг СМИ

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

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Поддерживает ввод нескольких поисковых фраз (по одной на строку). При поиске обеспечивает поддержку морфологии русского и английского языка
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03-05-2012 дата публикации

Adaptive audio transcoding

Номер: US20120109643A1
Автор: Huisheng Wang, Vijnan Shastri, Xiaoquan Yi
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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Номер записи: 1
03-05-2012 дата публикации

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

Номер: US20120109659A1
Автор: Guoming Chen, Hao Yuan, Jiali Li, Jing Lu, Kaiwen Liu, KE PENG, Ming Wu, Xiaojun Qiu, Zheng Deng, Zhibin LIN
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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Номер записи: 2
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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Номер записи: 3
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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Номер записи: 4
12-07-2012 дата публикации

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

Номер: US20120179700A1
Автор: Kazuhisa Ohta, Takazuki Mizuno, Walker J. Eagleston
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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Номер записи: 5
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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Номер записи: 6
23-08-2012 дата публикации

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

Номер: US20120215329A1
Автор: Claus Bauer, Regunathan Radhakrishnan, Wenyu Jiang
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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Номер записи: 7
06-09-2012 дата публикации

apparatus for processing a signal and method thereof

Номер: US20120226496A1
Автор: Hyun Kook Lee, Sung Yong Yoon, Tack Sung Choi
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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Номер записи: 8
20-09-2012 дата публикации

Method and an apparatus for processing an audio signal

Номер: US20120239408A1
Автор: Chang Heon Lee, Hong Goo Kang, Hyen-O Oh, Jeongook Song
Принадлежит: Industry Academic Cooperation Foundation of Yonsei University, LG ELECTRONICS INC

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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Номер записи: 9
04-10-2012 дата публикации

Multi-mode audio codec and celp coding adapted therefore

Номер: US20120253797A1
Автор: Bernhard Grill, Guillaume Fuchs, Markus Multrus, Ralf Geiger
Принадлежит: Fraunhofer Gesellschaft zur Forderung der Angewandten Forschung eV

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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Номер записи: 10
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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Номер записи: 11
22-11-2012 дата публикации

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

Номер: US20120294448A1
Автор: Eun-mi Oh, Jung-Hoe Kim, Ki-hyun Choo, Konstantin Osipov
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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Номер записи: 12
31-01-2013 дата публикации

MDCT-Based Complex Prediction Stereo Coding

Номер: US20130028426A1
Автор: Heiko Purnhagen, Lars Villemoes, Pontus Carlsson
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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Номер записи: 13
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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Номер записи: 14
14-03-2013 дата публикации

Information signal representation using lapped transform

Номер: US20130064383A1
Автор: Eleni FOTOPOULOU, Emmanuel Ravelli, Markus Schnell, Ralf Geiger
Принадлежит: Fraunhofer Gesellschaft zur Forderung der Angewandten Forschung eV

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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Номер записи: 15
11-04-2013 дата публикации

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

Номер: US20130090934A1
Автор: Frederik Nagel, Guillaume Fuchs, Jeremie Lecomte, Johannes Hilpert, Julien Robilliard, Markus Multrus, Stefan Bayer
Принадлежит: Fraunhofer Gesellschaft zur Forderung der Angewandten Forschung eV

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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Номер записи: 16
18-04-2013 дата публикации

VOIP DEVICE, VOIP CONFERENCING SYSTEM, AND RELATED METHOD

Номер: US20130094653A1
Автор: Bathurst Tracy A., Braithwaite Michael, Bryson Paul R., Ericksen Russel S., Graham Derek, Harris Brett, Kalra Sandeep, Lambert David K., Manley Peter H., Pandey Ashutosh, Shaw Bryan, Tilelli Michael
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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Номер записи: 17
18-04-2013 дата публикации

SELECTIVE BASS POST FILTER

Номер: US20130096912A1
Автор: Kjorling Kristofer, Resch Barbara, VILLEMOES Lars
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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Номер записи: 18
25-04-2013 дата публикации

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

Номер: US20130103394A1
Автор: Masahiro Oshikiri, Zongxian Liu
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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Номер записи: 19
02-05-2013 дата публикации

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

Номер: US20130110506A1
Автор: Bruhn Stefan, Norvell Erik, Pobloth Harald
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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Номер записи: 20
09-05-2013 дата публикации

Signal compression method and apparatus

Номер: US20130117030A1
Автор: Dejun Zhang, Fengyan Qi, Jianfeng Xu, LEI Miao, Marcel Taddei Herve, Qing Zhang
Принадлежит: Huawei Technologies Co Ltd

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

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Номер записи: 21
06-06-2013 дата публикации

Communication device with reduced noise speech coding

Номер: US20130143618A1
Автор: Nambirajan Seshadri
Принадлежит: Broadcom Corp

A communication device includes memory, an input interface, a processing module, and a transmitter. The processing module receives a digital signal from the input interface, wherein the digital signal includes a desired digital signal component and an undesired digital signal component. The processing module identifies one of a plurality of codebooks based on the undesired digital signal component. The processing module then identifies a codebook entry from the one of the plurality of codebooks based on the desired digital signal component to produce a selected codebook entry. The processing module then generates a coded signal based on the selected codebook entry, wherein the coded signal includes a substantially unattenuated representation of the desired digital signal component and an attenuated representation of the undesired digital signal component. The transmitter converts the coded signal into an outbound signal in accordance with a signaling protocol and transmits it.

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Номер записи: 22
06-06-2013 дата публикации

Half-Rate Vocoder

Номер: US20130144613A1
Автор: Hardwick John C.
Принадлежит: DIGITAL VOICE SYSTEMS, INC.

Encoding a sequence of digital speech samples into a bit stream includes dividing the digital speech samples into one or more frames, computing model parameters for a frame, and quantizing the model parameters to produce pitch bits conveying pitch information, voicing bits conveying voicing information, and gain bits conveying signal level information. One or more of the pitch bits are combined with one or more of the voicing bits and one or more of the gain bits to create a first parameter codeword that is encoded with an error control code to produce a first FEC codeword that is included in a bit stream for the frame. The process may be reversed to decode the bit stream. 1. A speech coder configured to encode a sequence of digital speech samples into a bit stream , the speech coder being operable to:divide the digital speech samples into one or more frames;compute model parameters for a frame;quantize the model parameters to produce pitch bits conveying pitch information, voicing bits conveying voicing information, and gain bits conveying signal level information, wherein the pitch bits, the voicing bits and the gain bits are included in quantizer bits for the frame;combine one or more of the pitch bits with one or more of the voicing bits and one or more of the gain bits to create a first parameter codeword that includes less than all of the quantizer bits for the frame;encode the first parameter codeword with an error control code to produce a first FEC (“forward error control”) codeword; andinclude the first FEC codeword in a bit stream for the frame.2. The speech coder of claim 1 , wherein the speech coder is operable to compute the model parameters for the frame by computing a fundamental frequency parameter claim 1 , one or more of voicing decisions claim 1 , and a set of spectral parameters.3. The speech coder of claim 1 , wherein the speech coder is operable to compute the model parameters for a frame using the Multi-Band Excitation speech model.4. The ...

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Номер записи: 23
04-07-2013 дата публикации

AUDIO ENCODING DEVICE AND AUDIO DECODING DEVICE

Номер: US20130173275A1
Автор: Chong Kok Seng, Liu Zongxian, Oshikiri Masahiro
Принадлежит: Panasonic Corporation

Provided is an audio encoding device that can suppress degradation of audio quality. Spectral coefficients of synthesized signal from CELP core layer are utilized to fulfill spectral gaps in error signal spectrum coefficients from a transform coding layer. By both spectral coefficients, decoded signal spectral coefficients are generated. The decoded signal spectral coefficients and the input signal spectral coefficients are divided into a plurality of sub bands. In each sub band, the energy of the input signal spectral coefficient corresponding to a zero decoded error signal spectral coefficient is calculated, and the energy of the decoded signal spectral coefficient corresponding to the zero decoding error signal spectral coefficient is calculated, and their energy ratio is calculated and is quantized and transmitted. 1. An audio coding apparatus comprising:a first coding section that codes an input signal and generates first coded data;a first local decoding section that decodes the first coded data, and generates a first decoded signal;a subtractor section that subtracts the first decoded signal from the input signal, and generates an error signal;a second coding section that codes only a portion of spectral coefficients of the error signal, and generates second coded data;a spectral envelope shaping parameter calculation section that calculates a spectral envelope shaping parameter; anda quantization section that quantizes the spectral envelope shaping parameter, and generates third coded data.2. The audio coding apparatus according to claim 1 , wherein the spectral envelope shaping parameter calculation section comprises:a second local decoding section that generates, from the second coded data, decoded error signal spectral coefficients comprising zero decoded error signal spectral coefficients and non-zero decoded error signal spectral coefficients;an adder section that adds spectral coefficients of the first decoded signal and the decoded error signal ...

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Номер записи: 24
11-07-2013 дата публикации

SYSTEM AND METHOD FOR MULTI LEVEL TRANSCRIPT QUALITY CHECKING

Номер: US20130179171A1
Автор: Howes Simon L.
Принадлежит: NUANCE COMMUNICATIONS, INC.

Methods and systems for multi level quality checking of transcripts are disclosed. The method includes the steps of searching subsets of metadata associated with the transcripts, identifying a group of transcripts having at least one particular subset of metadata, selecting a number of transcripts from the group of identified transcripts corresponding to a predetermined percentage, identifying a group of correctionists having a proper set of characteristics to correct the selected transcripts by matching the identified subsets of metadata associated with the transcripts with characteristics of correctionists, providing the transcripts and any voice files from which the transcripts derive to the selected correctionists, and, following correction, updating the subsets of metadata associated with the transcripts to include subsets of metadata pertaining to the voice files from which the transcripts were derived, any transcriptionist who transcribed the transcripts, or any correctionist who corrected the transcripts. 1. A method of multi level quality checking of transcripts of voice files comprising steps of:searching, via at least one computer system, metadata associated with said transcripts of voice files, wherein the metadata for each transcript comprises an indication of a confidence associated with a transcriptionist who prepared that transcript;using the metadata associated with said transcripts, identifying, via the at least one computer system, a group of transcripts having a particular quality assurance level;selecting, via the at least one computer system, a subset of said group of identified transcripts to be corrected;identifying, via the at least one computer system, a group of correctionists having a proper set of characteristics to correct said selected subset of transcripts;providing, via the at least one computer system, said subset of transcripts and any voice files from which the subset of transcripts derive to correctionists within said group of ...

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Номер записи: 25
18-07-2013 дата публикации

Systems, methods, apparatus, and computer-readable media for criticality threshold control

Номер: US20130185062A1
Автор: Daniel J. Sinder, Venkatesh Krishnan, Vivek Rajendran
Принадлежит: Qualcomm Inc

Systems, methods, and apparatus as disclosed herein may be implemented to adjust criticality thresholds for speech frames, based on channel conditions. Such a threshold may be used to control retransmission frequency in response to changes in channel state.

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Номер записи: 26
18-07-2013 дата публикации

Audio Signal Encoding Method, Audio Signal Decoding Method, Encoding Device, Decoding Device, Audio Signal Processing System, Audio Signal Encoding Program, and Audio Signal Decoding Program

Номер: US20130185075A1
Автор: KIKUIRI Kei, Naka Nobuhiko, Tsujino Kosuke
Принадлежит: NTT DOCOMO, INC.

When a frame immediately preceding an encoding target frame to be encoded by a first encoding unit operating under a linear predictive coding scheme is encoded by a second encoding unit operating under a coding scheme different from the linear predictive coding scheme, the encoding target frame can be encoded under the linear predictive coding scheme by initializing the internal state of the first encoding unit. Therefore, encoding processing performed under a plurality of coding schemes including the linear predictive coding scheme and a coding scheme different from the linear predictive coding scheme can be realized. 1. An audio signal encoding method for encoding an audio signal including a plurality of frames , using a first encoder operating under a linear predictive coding scheme and a second encoder operating under a coding scheme which is different from the linear predictive coding scheme , the audio signal encoding method comprising:a switching step of switching, to encode a second frame immediately succeeding a first frame, from the second encoder to the first encoder after the first frame of the audio signal is encoded by the second encoder; andan initialization step of initializing an internal state of the first encoder according to a predetermined method after the switching step,wherein, in the initialization step, an internal state of the first encoder is initialized by setting a residual signal as a content of an adaptive codebook of the first encoder, wherein the residual signal is obtained by applying a linear predictive inverse filter to either the first frame yet to be encoded by the second encoder or a signal obtained by decoding an encoded result of the first frame generated by the second encoder, andwherein linear predictive coefficients of the first frame are included in codes of the second frame, and, in the initialization step, the linear predictive coefficients is utilized for the linear predictive inverse filter.2. An audio signal decoding ...

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Номер записи: 27
18-07-2013 дата публикации

Audio Signal Encoding Method, Audio Signal Decoding Method, Encoding Device, Decoding Device, Audio Signal Processing System, Audio Signal Encoding Program, and Audio Signal Decoding Program

Номер: US20130185085A1
Автор: Kei Kikuiri, Kosuke Tsujino, Nobuhiko Naka
Принадлежит: NTT DOCOMO INC

When a frame immediately preceding an encoding target frame to be encoded by a first encoding unit operating under a linear predictive coding scheme is encoded by a second encoding unit operating under a coding scheme different from the linear predictive coding scheme, the encoding target frame can be encoded under the linear predictive coding scheme by initializing the internal state of the first encoding unit. Therefore, encoding processing performed under a plurality of coding schemes including the linear predictive coding scheme and a coding scheme different from the linear predictive coding scheme can be realized.

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Номер записи: 28
25-07-2013 дата публикации

HANDHELD DEVICE WIRELESS MUSIC STREAMING FOR GAMEPLAY

Номер: US20130190088A1
Автор: Booth Jesse Nathaniel, Oberg Gregory Keith
Принадлежит: Activision Publishing, Inc.

Systems and methods are disclosed for streaming of audio data of separate streams in at least two different formats. In some embodiments handheld game devices are in wireless communication and a first of the handheld game devices streams audio data during game play to a second of the handheld game devices. In some embodiments the audio data includes audio data from a plurality of streams of audio data. In some embodiments the streams of audio data include streams of audio data in different formats, generally different compressed formats, some of which may be selected based on whether a device includes circuitry specifically configured to decompress audio data in a specific data format. 16.-. (canceled)7. A method for use in communicating audio information from a first handheld device to a second handheld device , the method comprising:providing a first audio stream in a first data format;providing a second audio stream in a second data format, the second data format being different from the first data format;combining the first audio stream and the second audio stream into an output stream; andtransmitting the output stream from the first handheld device to the second handheld device.8. The method of claim 7 , wherein:the first data format is Ogg; andthe second data format is ADPCM.9. The method of claim 7 , wherein the first handheld device and the second handheld device are each a handheld game device.10. The method of claim 7 , wherein the transmitting the output stream from the first handheld device to the second handheld device is performed in accordance with an 802.11 standard.11. The method of claim 7 , wherein:the providing a first audio stream in a first data format comprises loading data of the first audio stream from a memory cartridge.12. A method for use in communicating audio information from a first handheld device to a second handheld device claim 7 , the method comprising:receiving an input stream at the second handheld device from the first ...

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Номер записи: 29
25-07-2013 дата публикации

CONSTRAINED SOFT DECISION PACKET LOSS CONCEALMENT

Номер: US20130191120A1
Автор: Chen Juin-Hwey, Zopf Robert W.
Принадлежит: BROADCOM CORPORATION

Methods, systems, and apparatuses for performing packet loss concealment are disclosed. In response to determining that an encoded frame representing a segment of a signal is bad, an encoded parameter within the encoded frame is decoded based on bit information (such as soft bit information) associated with the encoded parameter to obtain a decoded parameter. Whether the decoded parameter violates a parameter constraint is determined. If a parameter constraint violation is detected, an estimate of the decoded parameter is generated. Either the decoded parameter or estimate of the decoded parameter is passed to a decoder for use in decoding the encoded frame. 1. A method for performing packet loss concealment , comprising:determining that an encoded frame is bad, the encoded frame representing a segment of a signal; decoding an encoded parameter within the encoded frame based on bit information associated with the encoded parameter to obtain a decoded parameter;', 'determining if the decoded parameter violates a parameter constraint;', 'in response to determining that the decoded parameter does not violate the parameter constraint, passing the decoded parameter to a decoder for use in decoding the encoded frame; and', 'in response to determining that the decoded parameter does violate the parameter constraint, generating an estimate of the decoded parameter, and passing the estimate of the decoded parameter to the decoder for use in decoding the encoded frame., 'in response to determining that the encoded frame is bad2. The method of claim 1 , wherein the encoded parameter comprises one of: gain claim 1 , pitch claim 1 , line spectral frequencies claim 1 , pitch gain claim 1 , fixed codebook gain claim 1 , and fixed codebook excitation.3. The method of claim 1 , wherein the bit information comprises soft bit information obtained from a channel decoding process.4. The method of claim 1 , wherein decoding the encoded parameter based on the bit information comprises: ...

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Номер записи: 30
08-08-2013 дата публикации

Signal processing apparatus and signal processing method, encoder and encoding method, decoder and decoding method, and program

Номер: US20130202118A1
Автор: Hiroyuki Honma, Toru Chinen, Yuhki Mitsufuji, Yuki Yamamoto
Принадлежит: Sony Corp

The present invention relates to a signal processing apparatus and a signal processing method, an encoder and an encoding method, a decoder and a decoding method, and a program capable of reproducing music signal having a better sound quality by expansion of frequency band. A sampling frequency conversion unit converts a sampling frequency of an input signal, and a sub-band division circuit divides the input signal after the sampling conversion into sub-band signals of sub-bands having the number corresponding to the sampling frequency. A pseudo high band sub-band power calculation circuit calculates pseudo high band sub-band powers based on low band signals of the input signal and coefficient tables having coefficients for the respective high band sub-bands. A pseudo high band sub-band power difference calculation circuit compares high band sub-band powers and the pseudo high band sub-band powers to each other and selects a coefficient table from plural coefficient tables. In addition, a coefficient index which specifies the coefficient table is encoded and set as high band encoded data. The present invention can be applied to an encoder.

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Номер записи: 31
08-08-2013 дата публикации

METHOD FOR SPEECH CODING, METHOD FOR SPEECH DECODING AND THEIR APPARATUSES

Номер: US20130204615A1
Автор: YAMAURA Tadashi
Принадлежит: RESEARCH IN MOTION LIMITED

A high quality speech is reproduced with a small data amount in speech coding and decoding for performing compression coding and decoding of a speech signal to a digital signal. In speech coding method according to a code-excited linear prediction (CELP) speech coding, a noise level of a speech in a concerning coding period is evaluated by using a code or coding result of at least one of spectrum information, power information, and pitch information, and various excitation codebooks are used based on an evaluation result 12.-. (canceled)3. A speech decoding method , the method comprising:decoding a gain from a received gain code;obtaining a time series vector with a number of samples with zero amplitude-value from an excitation codebook; andmodifying the time series vector based on the decoded gain such that the number of samples with zero amplitude-value is changed.4. The method of claim 3 , further comprising:receiving a coded speech signal including a linear prediction parameter code, a gain code, and an adaptive code; andobtaining an adaptive code vector from an adaptive codebook based on the received adaptive code.5. The method of claim 4 , further comprising:weighting the adaptive code vector and the modified time series vector; andadding together the weighted adaptive code vector and the weighted time series vector.6. The method of claim 5 , further comprising:decoding a linear prediction parameter from the received linear prediction parameter code; andsynthesizing a speech signal using the linear prediction parameter and the added weighted adaptive code vector and weighted time series vector.7. The method of claim 6 , wherein the decoded linear prediction parameter corresponds to coefficients of a synthesis filter.8. The method of claim 4 , wherein the gain is decoded in a decoding period corresponding to the coded speech signal.9. The method of claim 4 , wherein the adaptive codebook is based on a past excitation.10. The method of claim 3 , further ...

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Номер записи: 32
22-08-2013 дата публикации

VOICE PROCESSING APPARATUS AND VOICE PROCESSING METHOD

Номер: US20130218556A1
Автор: ISHIHARA Hiroyuki, NAKA Yukio, SUWA Yutaka, TOYOMURA Norihiro
Принадлежит: Panasonic Corporation

A voice processing apparatus is provided in an ADPCM (Adaptive Differential Pulse Code Modulation) voice transmission system in which voice data that is differentially quantized through an ADPCM scheme is transmitted. The voice processing apparatus includes an error detector which detects whether or not an error occurs in a transmission frame containing voice data that indicates a differential value, and an error determiner which determines a level of the error detected by the error detector when the error detector detects the error. The voice processing apparatus also includes a voice processor which corrects the voice data with a correction value depending on the level of the error detected b the error detector and an ADPCM decoder which decodes the voice data corrected by the voice processor. 116.-. (canceled)17. A voice processing apparatus in an ADPCM (Adaptive Differential Pulse Code Modulation) voice transmission system in which voice data that is differentially quantized through an ADPCM scheme is transmitted , the voice processing apparatus comprising:an error detector which detects whether or not an error occurs in a transmission frame containing voice data that indicates a differential value;an error determiner which determines a level of the error detected by the error detector when the error detector detects the error;a voice processor which corrects the voice data with a correction value depending on the level of the error detected by the error detector; andan ADPCM decoder which decodes the voice data corrected by the voice processor.18. The voice processing apparatus according to claim 17 ,wherein the voice processor corrects the voice data from a head bit for a given time length depending on the level of the error by replacing with a value that indicates a mute so that a volume of the voice data is attenuated.19. The voice processing apparatus according to claim 17 ,wherein the voice processor subtracts or adds a value depending on the level of the ...

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Номер записи: 33
29-08-2013 дата публикации

APPARATUS AND METHOD FOR PROCESSING AN AUDIO SIGNAL AND FOR PROVIDING A HIGHER TEMPORAL GRANULARITY FOR A COMBINED UNIFIED SPEECH AND AUDIO CODEC (USAC)

Номер: US20130226570A1
Автор: Bessette Bruno, Fuchs Guillaume, Gournay Philippe, Grill Bernhard, Lefebvre Roch, Multrus Markus, Neuendorf Max, Rettelbach Nikolaus, Wilde Stephan
Принадлежит:

An apparatus for processing an audio signal is provided. The apparatus has a signal processor and a configurator. The configurator is adapted to configure the signal processor based on configuration information such that a configurable upsampling factor is equal to a first upsampling value when a first ratio of the second configurable number of samples to a first configurable number of samples has a first ratio value. Moreover, the configurator is adapted to configure the signal processor such that the configurable upsampling factor is equal to a different second upsampling value, when a different second ratio of the second configurable number of samples to the first configurable number of samples has a different second ratio value. The first or the second ratio value is not an integer value. 1. An apparatus for processing an audio signal , comprising:a signal processor being adapted to receive a first audio signal frame comprising a first configurable number of samples of the audio signal, being adapted to upsample the audio signal by a configurable upsampling factor to acquire a processed audio signal, and being adapted to output a second audio signal frame comprising a second configurable number of samples of the processed audio signal; anda configurator being adapted to configure the signal processor,wherein the configurator is adapted to configure the signal processor based on configuration information such that the configurable upsampling factor is equal to a first upsampling value when a first ratio of the second configurable number of samples to the first configurable number of samples comprises a first ratio value, and wherein the configurator is adapted to configure the signal processor such that the configurable upsampling factor is equal to a different second upsampling value, when a different second ratio of the second configurable number of samples to the first configurable number of samples comprises a different second ratio value, and wherein the ...

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Номер записи: 34
19-09-2013 дата публикации

System and Method for Post Excitation Enhancement for Low Bit Rate Speech Coding

Номер: US20130246055A1
Автор: Yang Gao
Принадлежит: Huawei Technologies Co Ltd

In accordance with an embodiment, a method of decoding an audio/speech signal includes decoding an excitation signal based on an incoming audio/speech information, determining a stability of a high frequency portion of the excitation signal, smoothing an energy of the high frequency portion of the excitation signal based on the stability of the high frequency portion of the excitation signal, and producing an audio signal based on smoothing the high frequency portion of the excitation signal.

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Номер записи: 35
19-09-2013 дата публикации

METHOD AND APPARATUS FOR DECODING AN AUDIO SIGNAL USING AN ADPATIVE CODEBOOK UPDATE

Номер: US20130246068A1
Автор: LEE Mi-Suk
Принадлежит: Electronics and Telecommunications Research Institute

Disclosed are a method and apparatus for decoding a an audiospeech signal using an adaptive codebook update. The method for decoding speechan audio signal includes: receiving an N+1-th normal frame data that is a normal frame transmitted after an N-th frame that is a loss frame data loss; determining whether an adaptive codebook of a final subframe of the N-th frame is updated or notby using the N-th frame and the N+1-th frame; updating the adaptive codebook of the final subframe of the N-th frame by using athe pitch index of the N+1-the frame; and synthesizing an audio a speech signal of by using the N+1-th frame. 1. A method for decoding an speech signal , comprising:receiving an N+1-th normal frame data after an N-th frame data loss;determining whether an adaptive codebook of a final subframe of the N-th frame is updated by using the parameters of the N-th frame and the N+1-th frame;updating the adaptive codebook of the final subframe of the N-th frame by using the parameter of the N+1-the frame; andsynthesizing speech signal of the N+1-th frame,2. The method of claim 1 , wherein the determining whether the adaptive codebook of the final subframe of the N-th frame is updated includes;determining whether a pitch of the final subframe of the N-th frame and a pitch of a first subframe of the N+1-th frame are different from each other;determining whether a difference between a pitch of the first subframe and a second subframe of the N+1-th subframe is smaller than a predetermined reference value; andif it is determined that both of the determination results are affirmative, determining that the adaptive codebook of the final subframe of the N-th frame is updated.3. The method of claim 1 , wherein the updating of the adaptive codebook of the final subframe of the N-th frame includes updating the adaptive codebook of the final subframe of the N-th frame by using the pitch index of the first subframe of the N+1-th frame.4. An apparatus for decoding speech signal claim 1 ...

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Номер записи: 36
19-09-2013 дата публикации

Coding device, coding method, decoding device, decoding method, and storage medium

Номер: US20130246073A1
Автор: Goro Sakata
Принадлежит: Casio Computer Co Ltd

For respective sampling data of waveform data of sounds to be coded, a prediction residual value is calculated as sampling residual data, and an effective bit length is calculated from this residual waveform data. Then, for the effective bit length data, a maximum effective bit length among processing targets is generated as common effective actual data, and coded data in which this common effective actual data and information indicating the common effective bit length are arranged in a predetermined configuration format are generated. The information included in the coded data is analyzed and each of the plurality of the common effective bit information is extracted. Then, waveform data of the sounds are decoded by performing inverse linear prediction processing from an analysis result on the residual waveform data decompressed by performing bit extension which adds a portion other than the common effective bit length.

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Номер записи: 37
26-09-2013 дата публикации

PARAMETER DECODING APPARATUS AND PARAMETER DECODING METHOD

Номер: US20130253922A1
Автор: EHARA Hiroyuki
Принадлежит: Panasonic Corporation

A parameter decoding apparatus includes a prediction residue decoder that finds a quantized prediction residue based on encoded information included in a current frame subject to decoding and a moving-average predictor produces a predicted parameter by multiplying a predictive coefficient with a past quantized prediction residue. An adder decodes a parameter by adding the quantized prediction residue and the predicted parameter, wherein the prediction residue decoder, when the current frame is erased, finds a current-frame quantized prediction residue from a weighted linear sum of a parameter decoded in the past and a future-frame quantized prediction residue. 17-. (canceled)8. A parameter decoding apparatus comprising:a prediction residue decoder that finds a quantized prediction residue based on encoded information included in a current frame subject to decoding,a moving-average predictor that produces a predicted parameter by multiplying a predictive coefficient with a past quantized prediction residue; andan adder that decodes a parameter by adding said quantized prediction residue and said predicted parameter,wherein said prediction residue decoder, when said current frame is erased, finds a current-frame quantized prediction residue from a weighted linear sum of a parameter decoded in the past and a future-frame quantized prediction residue.10. A parameter decoding method comprising:finding a quantized prediction residue based on encoding information included in a current frame subject to decoding,producing a predicted parameter by multiplying a predictive coefficient with a past quantized prediction residue; anddecoding a parameter by adding said quantized prediction residue and said predicted parameter,wherein, in the finding, when said current frame is erased, a current-frame quantized prediction residue is found from a weighted linear sum of a parameter decoded in the past and a future-frame quantized prediction residue. This application is a continuation ...

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Номер записи: 38
26-09-2013 дата публикации

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

Номер: US20130253939A1
Автор: KIKUIRI Kei, Tsutsumi Kimitaka
Принадлежит: NTT DOCOMO, INC.

An audio packet error concealment system includes an encoding unit for encoding an audio signal consisting of a plurality of frames, and an auxiliary information encoding unit for estimating and encoding auxiliary information about a temporal change of power of the audio signal. The auxiliary information is used in packet loss concealment in decoding of the audio signal. The auxiliary information about the temporal change of power may contain a parameter that functionally approximates a plurality of powers of subframes shorter than one frame, or may contain information about a vector obtained by vector quantization of a plurality of powers of subframes shorter than one frame. 1. An audio encoding device for encoding an audio signal consisting of a plurality of frames , the encoding device comprising:a processor;an audio encoding unit executable by the processor to encode the audio signal; andan auxiliary information encoding unit executable by the processor to estimate and encode auxiliary information about a temporal change of power of the audio signal, the auxiliary information used in packet loss concealment in decoding of the audio signal.2. The audio encoding device according to claim 1 , wherein the auxiliary information about the temporal change of power comprises a parameter that functionally approximates a plurality of powers of subframes shorter than one frame.3. The audio encoding device according to claim 1 , wherein the auxiliary information about the temporal change of power of the audio signal comprises information about a vector obtained by vector quantization of powers of subframes shorter than one frame.4. The audio encoding device according to claim 1 , wherein the auxiliary information about the temporal change of power comprises parameters which functionally approximate claim 1 , for respective subbands claim 1 , a plurality of powers of subframes shorter than one frame claim 1 , where the one frame is calculated for the respective subbands ...

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Номер записи: 39
10-10-2013 дата публикации

METHOD AND DEVICE FOR PROCESSING AUDIO SIGNAL

Номер: US20130268265A1
Автор: JEON Hyejeong, Jeong Gyuhyeok, Kang Ingyu, Kim Lagyoung, Lee Byungsuk
Принадлежит:

The present invention relates to a method for processing an audio signal, and the method comprises the steps of: receiving an audio signal; determining a coding mode corresponding to a current frame, by receiving network information for indicating the coding mode; encoding the current frame of said audio signal according to said coding mode; and transmitting said encoded current frame, wherein said coding mode is determined by the combination of a bandwidth and bitrate, and said bandwidth includes two or more bands among narrowband, wideband, and super wideband. 1. An audio signal processing method comprising:receiving an audio signal;receiving network information indicative of a coding mode;determining the coding mode corresponding to a current frame;encoding the current frame of the audio signal according to the coding mode; and,transmitting the encoded current frame, whereinthe coding mode is determined based on a combination of bandwidths and bitrates, and the bandwidths comprise at least two of narrowband, wideband, and super wideband,wherein the bitrates comprise two or more predetermined support bitrates for each of the bandwidths.2. The method according to claim 1 , whereinthe super wideband is a band that covers the wideband and the narrowband, andthe wideband is a band that covers the narrowband.3. The method according to claim 1 , further comprising:determining whether or not the current frame is a speech activity section by analyzing the audio signal,wherein the determining and the encoding are performed if the current frame is the speech activity section.4. The method according to claim 1 , further comprising:determining whether the current frame is a speech activity section or a speech inactivity section by analyzing the audio signal;if the current frame is the speech inactivity section, determining one of a plurality of types including a first type and a second type as a type of a silence frame for the current frame based on bandwidths of one or more ...

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Номер записи: 40
17-10-2013 дата публикации

APPARATUS AND METHOD FOR CONCEALING FRAME ERASURE AND VOICE DECODING APPARATUS AND METHOD USING THE SAME

Номер: US20130275127A1
Автор: Choi Seungho, Lee Kangeun, Sung Hosang
Принадлежит:

An apparatus and method for concealing frame erasure and a voice decoding apparatus and method using the same. The frame erasure concealment apparatus includes: a parameter extraction unit determining whether there is an erased frame in a voice packet, and extracting an excitement signal parameter and a line spectrum pair parameter of a previous good frame; and an erasure frame concealment unit, if there is an erased frame, restoring the excitement signal and line spectrum pair parameter of the erased frame by using a regression analysis from the excitement signal and line spectrum pair parameter of the previous good frame. According to the method and apparatus, by predicting and restoring the parameter of the erased frame through the regression analysis, the quality of the restored voice signal can be enhanced and the algorithm can be simplified. 1. A method for decoding a transmission packet , the method comprising:determining whether there is an erased frame in the transmission packet;predicting a spectral parameter of the erased frame, by applying a regression analysis to a spectral parameter of at least one previous good frame, if it is determined that there is an erased frame in the transmission packet;concealing, by way of a processor, the erased frame using the predicted spectral parameter; anddecoding the transmission packet using the concealed erased frame.2. The method of claim 1 , wherein the regression analysis uses a linear function.3. The method of claim 1 , wherein the regression analysis uses a non-linear function.4. The method of claim 1 , wherein the spectral parameter corresponds to a gain parameter.5. A non-transitory computer readable medium storing instructions that control at least one processor to implement the method of .6. The method of claim 4 , further comprising:deriving a function by way of the regression analysis using gain parameters of the at least one previous good frame; andpredicting a gain parameter of the erased frame based on ...

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Номер записи: 41
24-10-2013 дата публикации

APPARATUS AND METHOD FOR ENCODING/DECODING FOR HIGH FREQUENCY BANDWIDTH EXTENSION

Номер: US20130282368A1
Автор: CHOO Ki-hyun, Oh Eun-mi, Sung Ho-Sang
Принадлежит: SAMSUNG ELECTRONICS CO., LTD.

A method and apparatus for performing coding and decoding for high-frequency bandwidth extension. The coding apparatus may down-sample an input signal, perform core coding on the down-sampled input signal, perform frequency transformation on the input signal, and perform bandwidth extension coding by using a base signal of the input signal in a frequency domain. 143-. (canceled)44. A coding apparatus comprising:a signal classification unit configured to determine a coding mode of an input signal, based on characteristics of the input signal;a code excited linear prediction (CELP) coder configured to perform CELP coding on a low-frequency signal of the input signal when a coding mode of the input signal is determined to be a CELP coding mode;a time-domain (TD) extension coder configured to perform extension coding on a high-frequency signal of the input signal when CELP coding is performed on the low-frequency signal of the input signal;a frequency transformer configured to perform frequency transformation on the input signal when the coding mode of the input signal is determined as to be a frequency-domain (FD) mode; andan FD coder configured to perform FD coding on the transformed input signal.45. The coding apparatus of claim 44 , wherein the FD coder comprises:a normalization coder configured to extract energy from the transformed input signal for each frequency band and further configured to quantize the extracted energy;a factorial pulse coder configured to perform factorial pulse coding (FPC) on a value obtained by scaling the transformed input signal by using a quantized normalization value; andan additional noise information generator configured generate additional noise information according to performing of the FPC,wherein the transformed input signal input to the FD coder is a transient frame.46. The coding apparatus of claim 44 , wherein the FD coder comprises:a normalization coder configured to extract energy from the transformed input signal for each ...

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Номер записи: 42
24-10-2013 дата публикации

Audio Encoder and Decoder

Номер: US20130282382A1
Автор: CARLSSON Pontus, Hedelin Per, Samuelsson Leif Jonas, Schug Michael
Принадлежит: DOLBY INTERNATIONAL AB

The present invention teaches a new audio coding system that can code both general audio and speech signals well at low bit rates. A proposed audio coding system comprises linear prediction unit for filtering an input signal based on an adaptive filter; a transformation unit for transforming a frame of the filtered input signal into a transform domain; and a quantization unit for quantizing the transform domain signal. The quantization unit decides, based on input signal characteristics, to encode the transform domain signal with a model-based quantizer or a non-model-based quantizer. Preferably, the decision is based on the frame size applied by the transformation unit. 1. Audio coding system comprising:a linear prediction unit for filtering an input signal based on an adaptive filter;a transformation unit for transforming a frame of the filtered input signal into a transform domain; anda quantization unit for quantizing the transform domain signal;wherein the quantization unit decides, based on input signal characteristics to encode the transform domain signal with a model-based quantizer or a non-model-based quantizer.2. Audio coding system according to claim 1 , wherein the model in the model-based quantizer is adaptive and variable over time.3. Audio coding system according to claim 1 , wherein the quantization unit decides how to encode the transform domain signal based on the frame size applied by the transformation unit.4. Audio coding system according to claim 1 , wherein the quantization unit comprises a frame size comparator and is configured to encode a transform domain signal for a frame with a frame size smaller than a threshold value by means of a model-based entropy constrained quantization.5. Audio coding system according to claim 1 , comprising a quantization step size control unit for determining the quantization step sizes of components of the transform domain signal based on linear prediction and long term prediction parameters.6. Audio coding ...

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Номер записи: 43
24-10-2013 дата публикации

Audio Encoder and Decoder

Номер: US20130282383A1
Автор: CARLSSON Pontus, Hedelin Per, Samuelsson Leif Jonas, Schug Michael
Принадлежит: DOLBY INTERNATIONAL AB

The present invention teaches a new audio coding system that can code both general audio and speech signals well at low bit rates. A proposed audio coding system comprises linear prediction unit for filtering an input signal based on an adaptive filter; a transformation unit for transforming a frame of the filtered input signal into a transform domain; and a quantization unit for quantizing the transform domain signal. The quantization unit decides, based on input signal characteristics, to encode the transform domain signal with a model-based quantizer or a non-model-based quantizer. Preferably, the decision is based on the frame size applied by the transformation unit. 1. Audio coding system comprising:{'b': '201', 'a linear prediction (LP) unit () for filtering an audio signal based on a LP filter, the LP unit operating on a first frame length of the audio signal;'}{'b': '202', 'an adaptive length transformation unit () for transforming a frame of the audio signal into a transform domain, the transformation being a Modified Discrete Cosine Transform (MDCT) operating on a variable second frame length;'}{'b': '203', 'a quantization unit () for quantizing a MDCT-domain signal;'}{'b': '1070', 'a gain curve generation unit () for generating MDCT-domain gain curves based on magnitude responses of the LP filter; and'}{'b': '1100', 'a mapping unit () for mapping LP parameters to corresponding frames of the MDCT-domain signal.'}2. Audio coding system of claim 1 , comprising:a window sequence control unit for determining, for a block of the audio signal, the second frame lengths for overlapping MDCT windows.3. Audio coding system according to claim 1 , comprising a perceptual modeling unit that modifies a characteristic of the LP filter by chirping and/or tilting an LPC polynomial generated by the linear prediction unit for an LPC frame.4. Audio coding system according to claim 1 , comprising:a frequency splitting unit for splitting the audio signal into a lowband ...

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Номер записи: 44
31-10-2013 дата публикации

Low-delay sound-encoding alternating between predictive encoding and transform encoding

Номер: US20130289981A1
Автор: Balazs Kovesi, Pierre Berthet, Stephane Ragot
Принадлежит: France Telecom SA

An encoder and a method for encoding a digital signal are provided. The method includes encoding a preceding frame of samples of the digital signal according to a predictive encoding process, and encoding a current frame of samples of the digital signal according to a transform encoding process. The method is implemented such that a first portion of the current frame is also encoded by predictive encoding that is limited relative to the predictive encoding of the preceding frame by reusing at least one parameter of the predictive encoding of the preceding frame and only encoding the parameters of said first portion of the current frame that are not reused. A decoder and a decoding method are also provided, which correspond to the described encoding method.

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Номер записи: 45
31-10-2013 дата публикации

Method and apparatus for encoding and decoding high frequency for bandwidth extension

Номер: US20130290003A1
Автор: Ki-hyun Choo
Принадлежит: SAMSUNG ELECTRONICS CO LTD

Disclosed are a method and apparatus for encoding and decoding a high frequency for bandwidth extension. The method includes: estimating a weight; and generating a high frequency excitation signal by applying the weight between random noise and a decoded low frequency spectrum.

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Номер записи: 46
07-11-2013 дата публикации

ERROR CONCEALMENT METHOD AND APPARATUS FOR AUDIO SIGNAL AND DECODING METHOD AND APPARATUS FOR AUDIO SIGNAL USING THE SAME

Номер: US20130297322A1
Автор: CHOO Ki-hyun, Kim Jung-hoe, Lee Kang Eun, Oh Eun-mi, SON Chang-yong, Sung Ho-Sang
Принадлежит: Samsung Electronics Co., Ltd

An error concealment method and apparatus for an audio signal and a decoding method and apparatus for an audio signal using the error concealment method and apparatus. The error concealment method includes selecting one of an error concealment in a frequency domain and an error concealment in a time domain as an error concealment scheme for a current frame based on a predetermined criteria when an error occurs in the current frame, selecting one of a repetition scheme and an interpolation scheme in the frequency domain as the error concealment scheme for the current frame based on a predetermined criteria when the error concealment in the frequency domain is selected, and concealing the error of the current frame using the selected scheme. 1. An error concealment apparatus for an audio signal , the apparatus comprising:an error detection unit to determine whether an error occurs in a current frame; andan error concealment unit, when the error occurs in the current frame, to conceal the error of the current frame by applying at least one of a plurality of schemes comprising a repetition scheme as an error concealment scheme for the current frame, based on characteristics information obtained from a plurality of frames comprising at least one previous frame of the current frame.2. A decoding apparatus for an audio signal , the apparatus comprising:an error detection unit to determine whether an error occurs in a current frame;a decoding unit to decode the current frame when the error does not occur in the current frame; andan error concealment unit, when the error occurs in the current frame, to conceal the error of the current frame by applying at least one of a plurality of schemes comprising a repetition scheme as an error concealment scheme for the current frame, based on characteristics information obtained from a plurality of frames comprising at least one previous frame of the current frame.3. The error concealment apparatus of claim 1 , wherein the ...

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Номер записи: 47
28-11-2013 дата публикации

Efficient Encoding/Decoding of Audio Signals

Номер: US20130317811A1
Автор: Grancharov Volodya, Norvell Erik, Sverrisson Sigurdur
Принадлежит: TELEFONAKTIEBOLAGET L M ERICSSON (PUBL)

A method for encoding of an audio signal comprises performing () of a transform of the audio signal. An energy offset is selected () for each of the first subbands. An energy measure of a first reference band within a low band of an encoding of a synthesis signal is obtained (). The first high band is encoded () by providing quantization indices representing a respective scalar quantization of a spectrum envelope in the first subbands of the first high band relative to the energy measure of the first reference band by use of the selected energy offset. An encoder apparatus comprises means for carrying out the steps of the method. Corresponding decoder methods and apparatuses are also described. 142-. (canceled)43. A method for encoding of an audio signal , the method comprising:obtaining a low band synthesis signal of an encoding of said audio signal;obtaining a first energy measure of a first reference band within a low band (LB) in said low band synthesis signal;performing a transform of said audio signal into a transform domain;{'b': 1', '1, 'selecting an energy offset from a set of at least two predetermined energy offsets for each of a plurality of first subbands of a first high band (HB-) of said audio signal in said transform domain, said first high band (HB-) being situated at higher frequencies than said low band (LB); and'}{'b': 1', '1', '1', '1, 'encoding said first high band (HB-), wherein said encoding of said first high band (HB-) comprises providing a first set of quantization indices representing a respective scalar quantization of a spectrum envelope in said plurality of first subbands of said first high band (HB-) relative to said first energy measure, said first set of quantization indices being given with a respective said selected energy offset, and wherein said encoding of said first high band (HB-) further comprises providing a parameter defining the used energy offset;'}obtaining a second energy measure of a second reference band within said ...

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Номер записи: 48
28-11-2013 дата публикации

SPECTRAL ENVELOPE CODING OF ENERGY ATTACK SIGNAL

Номер: US20130317813A1
Автор: GAO Yang
Принадлежит: Huawei Technologies Co., Ltd.

MDCT or FFT-based audio coding algorithms often have the problem named here spectral pre-echoes when coding an energy attack signal. This invention presents several possibilities to avoid the spectral pre-echoes existing in decoded signal segment before the energy attack point. The spectral envelope before the attack point can be improved by performing spectrum smoothing, replacing the segment of having spectral pre-echoes or filtering the segment with a combined filter obtained by doing LPC analysis. 1. A signal processing method , comprising:receiving, by an access device, an encoded energy attack signal of an audio signal in a frequency domain, wherein the encoded energy attack signal is encoded from an energy attack signal in a time domain by performing a transformation with a current transform window, and wherein the current transform window covers a significant energy portion of the energy attack signal;decoding the encoded energy attack signal into the time domain by performing an inverse-transformation;detecting an energy attack point of the decoded energy attack signal in the time domain;performing LPC analysis on signal segment with spectral pre-echoes before the decoded energy attack point to obtain a LPC predictor A1(z);performing LPC analysis on signal segment without spectral pre-echoes covered by a previous transform window to obtain a LPC predictor A2(z);filtering the signal segment before the energy attack point with combined filter A1(z)/A2(z).2. The method of claim 1 , wherein the energy attack point is a time point at which energy of the decoded energy attack signal suddenly increases.31. The method of claim 1 , wherein the combined filter is expressed in weighted domain: A1(z/α)/A2(z/α) or A1(z/α)/A2(z/β) claim 1 , 0<α≦1 claim 1 , 0<β≦.4. An access device claim 1 , comprising:a receiver, configured to receive an encoded energy attack signal of an audio signal in a frequency domain, wherein the encoded energy attack signal is encoded from an ...

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Номер записи: 49
28-11-2013 дата публикации

ENCODING METHOD, DECODING METHOD, ENCODER, DECODER, PROGRAM, AND RECORDING MEDIUM

Номер: US20130317814A1
Автор: Fukui Masahiro, Harada Noboru, Hiwasaki Yusuke, Kamamoto Yutaka, Moriya Takehiro
Принадлежит: Nippon Telegraph and Telephone Corporation

In encoding, the number of bits to be assigned to codes corresponding to noise or a pulse sequence obtained according to prediction analysis applied to time series signals included in a predetermined time interval is switched according to whether an index that indicates a level of periodicity and/or stationarity of input time series signals satisfies a condition that indicates high periodicity and/or high stationarity or a condition that indicates low periodicity and/or low stationarity, to acquire the codes corresponding to the noise and the pulse sequence. In decoding, a decoding mode for codes corresponding to noise or a pulse sequence included in codes corresponding to a predetermined time interval is switched according to the same criterion as that described above to decode the codes corresponding to the noise or the pulse sequence to acquire noise or a pulse sequence corresponding to the predetermined time interval. 1. An encoding method comprising acquiring codes corresponding to prediction residuals obtained according to prediction analysis applied to time series signals included in a predetermined time interval of input time series signals , with the number of bits to be assigned to the codes corresponding to the prediction residuals being switched according to whether an index that indicates a level of periodicity and/or stationarity corresponding to the time series signals in the predetermined time interval or time series signals in an interval before the predetermined time interval of the input time series signals satisfies a condition that indicates high periodicity and/or high stationarity or a condition that indicates low periodicity and/or low stationarity.2. The encoding method according to claim 1 ,wherein the number of bits of the codes corresponding to the prediction residuals, obtained when the index that indicates the level of periodicity and/or stationarity satisfies the condition that indicates high periodicity and/or high stationarity, is ...

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Номер записи: 50
05-12-2013 дата публикации

ENCODING APPARATUS, DECODING APPARATUS, ENCODING METHOD AND DECODING METHOD

Номер: US20130325457A1
Автор: Morii Toshiyuki, Oshikiri Masahiro, Yamanashi Tomofumi
Принадлежит: Panasonic Corporation

An encoding apparatus includes a first layer encoder that encodes a signal, a first layer decoder that decodes first layer encoded data, a first layer error transform coefficient calculator that transforms a first layer error signal into a frequency domain and a second layer encoder that encodes the first layer error transform coefficient to acquire second layer encoded data. The second layer encoder includes a band determiner that determines a band to be encoded by the second layer encoder, and a first shape vector encoder that refers the first layer error transform coefficient included in the band to generate a first shape vector and first shape encoded information, a target gain calculator calculates target gain per subband, a gain vector generator generates a gain vector using a plurality of target gains, and a gain vector encoder encodes the gain vector to acquire gain encoded information. 1. An encoding apparatus comprising:a first layer encoder that encodes an input signal to acquire first layer encoded data;a first layer decoder that decodes the first layer encoded data to acquire a first layer decoded signal;a first layer error transform coefficient calculator that transforms a first layer error signal that is a difference between the input signal and the first layer decoded data into a frequency domain to calculate a first layer error transform coefficient; anda second layer encoder that encodes the first layer error transform coefficient to acquire second layer encoded data,wherein the second layer encoder comprises:a band determiner that determines a band which is a target to be encoded by the second layer encoder, based on tonality or energy of the input signal;a first shape vector encoder that refers the first layer error transform coefficient included in the band which is determined by the band determiner and has a predetermined first bandwidth, to generate a first shape vector by arranging a predetermined number of pulses in the band, and to generate ...

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Номер записи: 51
05-12-2013 дата публикации

METHOD AND AN APPARATUS FOR PROCESSING AN AUDIO SIGNAL

Номер: US20130325487A1
Автор: Kang Hong Goo, LEE Chang Heon, Oh Hyen-O, Song Jung Wook
Принадлежит: INDUSTRY-ACADEMIC COOPERATION FOUNDATION YONGSEI UNIVERSITY

An apparatus for processing an audio signal and method thereof are disclosed. The present invention includes receiving, by an audio processing apparatus, an audio signal including a first data of a first block encoded with rectangular coding scheme and a second data of a second block encoded with non-rectangular coding scheme; receiving a compensation signal corresponding to the second block; estimating a prediction of an aliasing part using the first data; and, obtaining a reconstructed signal for the second block based on the second data, the compensation signal and the prediction of aliasing part. 116-. (canceled)17. A method for processing an audio signal , comprising:receiving, by an audio processing apparatus, an audio signal including a first data of a first block and a second data of a second block;receiving a compensation signal corresponding to the second block; andobtaining a reconstructed signal for the second block based on the second data, the compensation signal and a window of the second block,wherein, when the first data is encoded with a rectangular coding scheme and the window of the second block belongs to transition window class, the window of the second block has ascending line with a first slope,wherein the first slope is gentler than a second slope.18. The method of claim 17 , wherein claim 17 , when the first data is encoded with a non-rectangular coding scheme and the window of the second block belongs to the transition window class claim 17 , the window of the second block has ascending line with the second slope.19. The method of claim 17 , wherein claim 17 , when the transition window class comprises long_stop window and stop_start window claim 17 , andthe long_stop window and the stop_start window are horizontal-asymmetry, and have a zero part in a left half.20. The method of claim 17 , wherein the compensation signal is received claim 17 , when the first data is encoded with the rectangular coding scheme.21. The method of claim 17 , ...

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Номер записи: 52
12-12-2013 дата публикации

Apparatus and method for encoding and decoding an audio signal using an aligned look-ahead portion

Номер: US20130332148A1
Автор: Bernhard Grill, Christian Helmrich, Emmanuel Ravelli, Guillaume Fuchs, Markus Schnell, Ralf Geiger, Tom Baeckstroem, Vesa Ruoppila
Принадлежит: Fraunhofer Gesellschaft zur Forderung der Angewandten Forschung eV

An apparatus for encoding an audio signal having a stream of audio samples has: a windower for applying a prediction coding analysis window to the stream of audio samples to obtain windowed data for a prediction analysis and for applying a transform coding analysis window to the stream of audio samples to obtain windowed data for a transform analysis, wherein the transform coding analysis window is associated with audio samples within a current frame of audio samples and with audio samples of a predefined portion of a future frame of audio samples being a transform-coding look-ahead portion, wherein the prediction coding analysis window is associated with at least the portion of the audio samples of the current frame and with audio samples of a predefined portion of the future frame being a prediction coding look-ahead portion, wherein the transform coding look-ahead portion and the prediction coding look-ahead portion are identically to each other or are different from each other by less than 20%; and an encoding processor for generating prediction coded data or for generating transform coded data.

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Номер записи: 53
12-12-2013 дата публикации

APPARATUS AND METHOD FOR PROCESSING A DECODED AUDIO SIGNAL IN A SPECTRAL DOMAIN

Номер: US20130332151A1
Автор: DOEHLA Stefan, Fuchs Guillaume, GEIGER Ralf, RAVELLI Emmanuel, SCHNELL Markus
Принадлежит:

An apparatus for processing a decoded audio signal including a filter for filtering the decoded audio signal to obtain a filtered audio signal, a time-spectral converter stage for converting the decoded audio signal and the filtered audio signal into corresponding spectral representations, each spectral representation having a plurality of subband signals, a weighter for performing a frequency selective weighting of the filtered audio signal by a multiplying subband signals by respective weighting coefficients to obtain a weighted filtered audio signal, a subtractor for performing a subband-wise subtraction between the weighted filtered audio signal and the spectral representation of the decoded audio signal, and a spectral-time converter for converting the result audio signal or a signal derived from the result audio signal into a time domain representation to obtain a processed decoded audio signal. 1. Apparatus for processing a decoded audio signal , comprising:a filter for filtering the decoded audio signal to acquire a filtered audio signal;a time-spectral converter stage for converting the decoded audio signal and the filtered audio signal into corresponding spectral representations, each spectral representation comprising a plurality of subband signals;a weighter for performing a frequency selective weighting of the spectral representation of the filtered audio signal by multiplying subband signals by respective weighting coefficients to acquire a weighted filtered audio signal;a subtractor for performing a subband-wise subtraction between the weighted filtered audio signal and the spectral representation of the audio signal to acquire a result audio signal; anda spectral-time converter for converting the result audio signal or a signal derived from the result audio signal into a time domain representation to acquire a processed decoded audio signal.2. Apparatus in accordance with claim 1 , further comprising a bandwidth enhancement decoder or a mono-stereo ...

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Номер записи: 54
12-12-2013 дата публикации

Apparatus and method for error concealment in low-delay unified speech and audio coding

Номер: US20130332152A1
Автор: Jeremie Lecomte, Martin Dietz, Michael Schnabel, Ralph Sperschneider
Принадлежит: Fraunhofer Gesellschaft zur Forderung der Angewandten Forschung eV, Technische Universitaet Ilmenau

An apparatus for generating spectral replacement values for an audio signal has a buffer unit for storing previous spectral values relating to a previously received error-free audio frame. Moreover, the apparatus includes a concealment frame generator for generating the spectral replacement values, when a current audio frame has not been received or is erroneous. The previously received error-free audio frame has filter information, the filter information having associated a filter stability value indicating a stability of a prediction filter. The concealment frame generator is adapted to generate the spectral replacement values based on the previous spectral values and based on the filter stability value.

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Номер записи: 55
12-12-2013 дата публикации

Bandwidth Extension via Constrained Synthesis

Номер: US20130332171A1
Автор: Athineos Marios, Avendano Carlos, Duni Ethan
Принадлежит:

Audio signal bandwidth extension may be performed on a narrow bandwidth signal received from a remote source over the audio communication network. The narrow band signal bandwidth may be extended such that the bandwidth is greater than that of the audio communication network. The signal may be extended by synthesizing an audio signal having spectral values within an extended bandwidth from synthetic components. The synthetic components may be generated using parameters derived from original narrowband audio signal. The audio signal may be synthesized in the form of an excitation signal and vocal tract envelope. The excitation signal and vocal tract may be extended independently. In various embodiments, excitation components may be derived from constrained synthesis using a constraint filter with nulls in regions where the extension is desired. 1. A method for extending bandwidth of an audio signal , the method comprising:receiving, by a processor, an audio signal having spectral values within a narrow bandwidth;determining, via instructions stored in a memory and executed by the processor, synthetic components of an audio signal having spectral values within an extended bandwidth; andsynthesizing, via instructions stored in the memory and executed by the processor and based on the synthetic components, an extended audio signal having spectral values within an extended bandwidth.2. The method of claim 1 , wherein the extended bandwidth includes a frequency outside the narrow bandwidth.3. The method of claim 1 , wherein the synthetic components are divided into a spectral envelope and excitation components.4. The method of claim 3 , wherein the spectral envelope and the excitation components are estimated independently.5. The method of claim 3 , wherein the spectral envelope for the extended bandwidth signal is estimated based on information derived from the spectral envelope of the narrow bandwidth signal.6. The method of claim 3 , wherein the spectral envelope for ...

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Номер записи: 56
12-12-2013 дата публикации

Audio codec using noise synthesis during inactive phases

Номер: US20130332175A1
Автор: Konstantin Schmidt, Panji Setiawan, Stephan Wilde
Принадлежит: Fraunhofer Gesellschaft zur Forderung der Angewandten Forschung eV

A parametric background noise estimate is continuously updated during an active or non-silence phase so that the noise generation may immediately be started with upon the entrance of an inactive phase following the active phase. In accordance with another aspect, a spectral domain is very efficiently used in order to parameterize the background noise thereby yielding a background noise synthesis which is more realistic and thus leads to a more transparent active to inactive phase switching.

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Номер записи: 57
19-12-2013 дата публикации

CODING DEVICE, COMMUNICATION PROCESSING DEVICE, AND CODING METHOD

Номер: US20130339009A1
Автор: Morii Toshiyuki, Yamanashi Tomofumi
Принадлежит: Panasonic Corporation

Provided are a coding device, a communication processing device, and a coding method, whereby processing operation load (computational load) is significantly reduced for a configuration which computes either frame energy or sub-frame energy of an input signal, using auto-correlation operations, without causing a decline in the precision of either the frame energy or the sub-frame energy. In a coding device (), a sub-frame energy computation unit () computes the sub-frame energy by substituting the sum of input signal auto-correlation operations in a first range with the sum of auto-correlation operations in a second range which differs at least partially from the first range. 1. A coding apparatus comprising:an energy calculation section that calculates one of frame energy and subframe energy of an input signal using an auto-correlation operation of the input signal; anda coding section that encodes the input signal using one of the frame energy and the subframe energy, and generates encoded information, whereinthe energy calculation section calculates one of the frame energy and the subframe energy by substituting the sum of auto-correlation operations in a first range of the input signal with the sum of auto-correlation operations in a second range which differs at least partially from the first range.2. The coding apparatus according to claim 1 , wherein the energy calculation section performs control so as to increase the frequency with which the sum of auto-correlation operations in the first range is substituted with the sum of auto-correlation operations in the identical second range as a delay time during auto-correlation operations in the first range increases.3. The coding apparatus according to claim 1 , wherein the energy calculation section substitutes the sum of auto-correlation operations in the first range with auto-correlation operations in the second range based on whether or not a sample in which the amplitude of the input signal is equal to or ...

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Номер записи: 58
19-12-2013 дата публикации

Speech/audio encoding apparatus, speech/audio decoding apparatus, and methods thereof

Номер: US20130339012A1
Автор: Masahiro Oshikiri, Takuya Kawashima
Принадлежит: Panasonic Corp

Provided is a speech/audio encoding apparatus with which it is possible to code a significant frequency domain region with high precision, and to enable high audio quality. A speech/audio encoding apparatus codes a linear prediction coefficient. A significant frequency domain region detection unit identifies a frequency domain region which is aurally significant from the linear prediction coefficient. A frequency domain region repositioning unit repositions the significant frequency domain region which is identified by the significant frequency domain region detection unit. A bit allocation computation unit determines a coding bit allocation on the basis of the significant frequency domain region which is repositioned by the frequency domain region repositioning unit.

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Номер записи: 59
19-12-2013 дата публикации

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

Номер: US20130339034A1
Автор: Kazuhisa Ohta, Takayuki Mizuno, Walker J. Eagleston
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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Номер записи: 60
09-01-2014 дата публикации

DECODING WIRELESS IN-BAND ON-CHANNEL SIGNALS

Номер: US20140012570A1
Автор: BI DONGSHENG, Haddad Bassel, Ravindran Bomiraj
Принадлежит:

Described herein are systems, methods and apparatus for decoding in-band on-channel signals and extracting audio and data signals. Memory requirements are reduced by selectively filtering a bit stream of data in the signal so that services of interest which are encoded therein are processed. A single pool of memory may be shared between physical layer and data link layer processing. Memory in this pool may be allocated dynamically between processing of data at the physical and data link layers. When the available memory is not sufficient to support the required services, the dynamic allocation allows for graceful degradation. 132.-. (canceled)33. An apparatus for decoding wireless signals , the apparatus comprising:a memory; a dynamic memory management module;', 'a demodulation module coupled to the dynamic memory management module;', 'a physical layer segment planner module coupled to the dynamic memory management module;', 'a deinterleaver module coupled to the physical layer segment planner module;', 'an address filter module coupled to the physical layer segment planner module and the deinterleaver module;', 'a convolutional decoding module coupled to the physical layer segment planner module; and', 'a service boundary predictor module coupled to the dynamic memory management module and the convolutional decoding module., 'a decoder coupled to the memory, the decoder comprising34. The apparatus of claim 33 , the memory configured to store at least data link layer buffers claim 33 , audio pulse code modulated data claim 33 , deinterleaver buffers claim 33 , or a combination thereof.35. The apparatus of claim 33 , the dynamic memory management module configured to:receive service boundary information from the service boundary predictor module, the service boundary information comprising addresses of segments for a service of interest in a bit stream of data provided by the demodulation module;receive channel condition information from the demodulation module, the ...

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Номер записи: 61
16-01-2014 дата публикации

ENCODING METHOD, DECODING METHOD, ENCODER, DECODER, PROGRAM, AND RECORDING MEDIUM

Номер: US20140019145A1
Автор: Fukui Masahiro, Harada Noboru, Hiwasaki Yusuke, Kamamoto Yutaka, Moriya Takehiro
Принадлежит: Nippon Telegraph and Telephone Corporation

In encoding, a frequency-domain sample sequence derived from an acoustic signal is divided by a weighted envelope and is then divided by a gain, the result obtained is quantized, and each sample is variable-length encoded. The error between the sample before quantization and the sample after quantization is quantized with information saved in this variable-length encoding. This quantization is performed under a rule that specifies, according to the number of saved bits, samples whose errors are to be quantized. In decoding, variable-length codes in an input sequence of codes are decoded to obtain a frequency-domain sample sequence; an error signal is further decoded under a rule that depends on the number of bits of the variable-length codes; and from the obtained sample sequence, the original sample sequence is obtained according to supplementary information. 1. An encoding method for encoding , with a predetermined number of bits , a frequency-domain sample sequence derived from an acoustic signal in a predetermined time interval , the encoding method comprising:an encoding step of encoding, by variable-length encoding, an integer corresponding to the value of each sample in the frequency-domain sample sequence to generate a variable-length code;an error calculation step of calculating a sequence of error values each obtained by subtracting the integer corresponding to the value of each sample in the frequency-domain sample sequence from the value of the sample; andan error encoding step of encoding the sequence of error values with the number of surplus bits obtained by subtracting the number of bits of the variable-length code from the predetermined number of bits to generate error codes.2. The encoding method according to claim 1 , wherein claim 1 , among error samples constituting the sequence of error values claim 1 , error samples whose corresponding integers are not 0 are encoded with priority with the number of surplus bits in the error encoding step.3. ...

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Номер записи: 62
30-01-2014 дата публикации

ADAPTIVE TIME/FREQUENCY-BASED AUDIO ENCODING AND DECODING APPARATUSES AND METHODS

Номер: US20140032213A1
Автор: CHOO Kihyun, KIM Junghoe, OH Eunmi, SON Changyong
Принадлежит: Samsung Electronics Co., Ltd

Adaptive time/frequency-based audio encoding and decoding apparatuses and methods. The encoding apparatus includes a transformation & mode determination unit to divide an input audio signal into a plurality of frequency-domain signals and to select a time-based encoding mode or a frequency-based encoding mode for each respective frequency-domain signal, an encoding unit to encode each frequency-domain signal in the respective encoding mode, and a bitstream output unit to output encoded data, division information, and encoding mode information for each respective frequency-domain signal. In the apparatuses and methods, acoustic characteristics and a voicing model are simultaneously applied to a frame, which is an audio compression processing unit. As a result, a compression method effective for both music and voice can be produced, and the compression method can be used for mobile terminals that require audio compression at a low bit rate. 1. An audio decoding apparatus , comprising:a first decoding unit to decode first encoded data, by using a code excited linear prediction (CELP) with at least a long-term prediction, in a first domain, based on a mode information of encoded data in a bitstream;a second decoding unit to decode second encoded data by using an advanced audio coding (AAC), in a second domain, based on the mode information of the encoded data in the bitstream;an inverse-transform unit to inverse-transform data decoded in the second domain; anda signal generation unit to generate a signal including the inverse-transformed data and the result of decoding in the first domain.2. The apparatus of claim 1 , wherein the first and second domains comprise a frequency domain.3. The apparatus of claim 1 , wherein the first and second domains are different from each other.4. An audio decoding apparatus claim 1 , comprising:a first decoding unit to decode first encoded data, by using at least a long term prediction, in a linear prediction coding domain, based on a ...

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Номер записи: 63
30-01-2014 дата публикации

Bit error management methods for wireless audio communication channels

Номер: US20140032227A1
Автор: Michael S. Gustafson, Robert W. Zopf, VIVEK Kumar
Принадлежит: Broadcom Corp

Systems and methods are described for managing bit errors present in an encoded bit stream representative of a portion of an audio signal, wherein the encoded bit stream is received via a channel in a wireless communications system. The channel may comprise, for example, a Synchronous Connection-Oriented (SCO) channel or an Extended SCO (eSCO) channel in a Bluetooth® wireless communications system.

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Номер записи: 64
13-02-2014 дата публикации

Multistage IIR Filter and Parallelized Filtering of Data with Same

Номер: US20140046673A1
Автор: Rathi Khushbu P.
Принадлежит: DOLBY LABORATORIES LICENSING CORPORATION

In some embodiments, a multistage filter whose biquad filter stages are combined with latency between the stages, a system (e.g., an audio encoder or decoder) including such a filter, and methods for multistage biquad filtering. In typical embodiments, all biquad filter stages of the filter are operable independently to perform fully parallelized processing of data. In some embodiments, the inventive multistage filter includes a buffer memory, at least two biquad filter stages, and a controller coupled and configured to assert a single stream of instructions to the filter stages. Typically, the multistage filter is configured to perform multistage filtering of a block of input samples in a single processing loop with iteration over a sample index but without iteration over a biquadratic filter stage index. 1. An audio encoder configured to generate encoded audio data in response to input audio data , said encoder including at least one multistage filter coupled and configured to filter the audio data , wherein the multistage filter includes:a buffer memory;at least two biquad filter stages, including a first biquad filter stage and a subsequent biquad filter stage; anda controller, coupled to the biquad filter stages and configured to assert a single stream of instructions to both the first biquad filter stage and the subsequent biquad filter stage, wherein said first biquad filter stage and said subsequent biquad filter stage operate independently and in parallel in response to the stream of instructions,wherein the first biquad filter stage is coupled to the memory and configured to perform biquadratic filtering on a block of N input samples in response to the stream of instructions to generate intermediate values, and to assert the intermediate values to the memory, wherein the intermediate values include a filtered version of each of at least a subset of the input samples, andwherein the subsequent biquad filter stage is coupled to the memory and configured to ...

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Номер записи: 65
13-03-2014 дата публикации

Method and apparatus for encoding/decoding speech signal using coding mode

Номер: US20140074461A1
Автор: Eun Mi Oh, Ho Sang Sung, Jung Hoe Kim, Ki Hyun Choo
Принадлежит: SAMSUNG ELECTRONICS CO LTD

An apparatus and a method to encode and decode a speech signal using an encoding mode are provided. An encoding apparatus may select an encoding mode of a frame included in an input speech signal, and encode a frame having an unvoiced mode for an unvoiced speech as the selected encoding mode.

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Номер записи: 66
06-01-2022 дата публикации

METHODS OF ENCODING AND DECODING AUDIO SIGNAL USING NEURAL NETWORK MODEL, AND DEVICES FOR PERFORMING THE METHODS

Номер: US20220005488A1
Автор: BEACK Seung Kwon, JANG Inseon, LEE Mi Suk, Lee Tae Jin, LIM Woo-taek, SUNG Jongmo
Принадлежит: Electronics and Telecommunications Research Institute

The encoding method includes computing the first feature information of an input signal using a recurrent encoding model, quantizing the first feature information and producing the first feature bitstream, computing the first output signal from the quantized first feature information using a recurrent decoding model, computing the second feature information of the input signal using a nonrecurrent encoding model, quantizing the second feature information and producing the second feature bitstream, computing the second output signal from the quantized second feature information using a nonrecurrent decoding model, determining an encoding mode based on the input signal, the first and second output signals, and the first and second feature bitstreams, and outputting an overall bitstream by multiplexing an encoding mode bit and one of the first feature bitstream and the second feature bitstream depending on the encoding mode. 1. An encoding method comprising:computing the first feature information of an input signal using a recurrent encoding model;quantizing the first feature information and producing the first feature bitstream;computing the first output signal from the quantized first feature information using a recurrent decoding model;computing the second feature information of the input signal using a nonrecurrent encoding model;quantizing the second feature information and producing the second feature bitstream;computing the second output signal from the quantized second feature information using a nonrecurrent decoding model;determining an encoding mode based on the input signal, the first output signal, the second output signal, the first feature bitstream, and the second feature bitstream; andoutputting an overall bitstream by multiplexing an encoding mode bit and one of the first feature bitstream and the second feature bitstream depending on the encoding mode.2. The encoding method of claim 1 , wherein the recurrent encoding model is configured to encode the ...

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Номер записи: 67
01-01-2015 дата публикации

MAX SOUND AUDIO PROGRAM

Номер: US20150003633A1
Автор: Trammell Lloyd
Принадлежит: Max Sound Corporation

A process and system for enhancing and customizing sound comprises receiving an input audio sound and processing the input by one or more filter blocks to dynamically reshape the dynamic, phase and frequency content of the input audio sound. The parallel processed audio is combined in a sound mixer and tube harmonics are added to the output from the mixer. The tube harmonics added audio is fed to a highpass filter which sets a frequency limit of audio passing through this block. A low pass filter movines the audio in a plus or minus direction with a dynamic envelope control. A frequency divider is provided for shifting audio sound below a selected frequency down an octave. Output from this stage is edited by feeding a certain amount of each side of the audio to a corresponding opposite side. The frequency balance is adjusted by setting a band phase coherent equalizer for frequency adjustments. The original, unprocessed sound is then mixed with the processed audio. Gain is adjusted and the processed sound is outputted for use. 1. A process and system for enhancing and customizing sound comprising:Receiving an input audio sound;Processing the input by one or more filter blocks to dynamically reshape the dynamic, phase and frequency content of the input audio sound;Combining the processed input audio sounds in a mixer;Adding tube harmonics to the output from the mixer;Providing the tube harmonics added audio to a highpass filter which sets a frequency limit of audio passing through this block;A low pass filter for moving the audio in a plus or minus direction with a dynamic envelope control;A frequency divider for shifting audio sound below a selected frequency down an octave;Editing sound output by feeding a certain amount of each side of the audio to a corresponding opposite side;Adjusting a sound frequency balance by setting a band phase coherent equalizer for frequency adjustments;Mix the original unprocessed audio with the processed audio;Adjust Gain;Output the ...

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Номер записи: 68
01-01-2015 дата публикации

SOUND ENHANCEMENT FOR MOVIE THEATERS

Номер: US20150006180A1
Автор: Trammell Lloyd
Принадлежит: Max Sound Corporation

A process and system for enhancing and customizing movie theatre sound includes receiving an input audio sound and enhancing the voice audio input in two or more harmonic and dynamic ranges by re-synthesizing the audio into a full range PCM wave. The enhancement includes the parallel processing the input audio via a low pass filter with dynamic offset, an envelope controlled bandpass filter, a high pass filter, adding an amount of dynamic synthesized sub bass to the audio and combining the four treated audio signals in a summing mixer with the original audio. 1. A process and system for enhancing and customizing a movie theater sound comprising:Receiving an input audio sound;Enhancing the voice audio input in two or more harmonic and dynamic ranges by re-synthesizing the audio into a full range PCM wave;Outputting the enhanced audio sound.2. The process of claim 1 , wherein the enhancement includes the parallel processing the input audio as follows:A module that is a low pass filter with dynamic offset;An envelope controlled bandpass filter;A high pass filter;Adding an amount of dynamic synthesized sub bass to the audio;Combining the four treated audio signals in a summing mixer with the original audio. Embodiments of the present invention relate to U.S. Provisional Application Ser. No. 61/767,737, filed Feb. 21, 2013, entitled “MOVIE THEATER SOUND SYSTEM”, the contents of which are incorporated by reference herein and which is a basis for a claim of priority.Movie theater sound systems are conventionally 5.1, 7.1, or more in their configuration. While these systems sound good, they are usually very loud in order to fill a theater with audio. There is a “Sweet Spot” where the convergence of the system audio is the best sounding, meaning that anything out of the small “Sweet Spot” has a less desirable seat. Meaning that if user sits directly under the Rear Left speaker, then that is the predominant audio source user will hear.The disadvantage of the conventional ...

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Номер записи: 69
07-01-2021 дата публикации

AUDIO ENCODER FOR ENCODING AN AUDIO SIGNAL, METHOD FOR ENCODING AN AUDIO SIGNAL AND COMPUTER PROGRAM UNDER CONSIDERATION OF A DETECTED PEAK SPECTRAL REGION IN AN UPPER FREQUENCY BAND

Номер: US20210005210A1
Автор: Multrus Markus, NEUKAM Christian, SCHNELL Markus, SCHUBERT Benjamin
Принадлежит:

An audio encoder for encoding an audio signal having a lower frequency band and an upper frequency band includes: a detector for detecting a peak spectral region in the upper frequency band of the audio signal; a shaper for shaping the lower frequency band using shaping information for the lower band and for shaping the upper frequency band using at least a portion of the shaping information for the lower band, wherein the shaper is configured to additionally attenuate spectral values in the detected peak spectral region in the upper frequency band; and a quantizer and coder stage for quantizing a shaped lower frequency band and a shaped upper frequency band and for entropy coding quantized spectral values from the shaped lower frequency band and the shaped upper frequency band. 1a detector for detecting a peak spectral region in the upper frequency band of the audio signal;a shaper for shaping the lower frequency band using shaping information for the lower band and for shaping the upper frequency band using at least a portion of the shaping information for the lower frequency band, wherein the shaper is configured to additionally attenuate spectral values in the detected peak spectral region in the upper frequency band; anda quantizer and coder stage for quantizing a shaped lower frequency band and a shaped upper frequency band and for entropy coding quantized spectral values from the shaped lower frequency band and the shaped upper frequency band.. Audio encoder for encoding an audio signal comprising a lower frequency band and an upper frequency band, comprising: This application is a continuation of U.S. patent application Ser. No. 16/143,716, filed Sep. 25, 2018, which is a continuation of copending International Application No. PCT/EP2017/058238, filed Apr. 6, 2017, which is incorporated herein by reference in its entirety, which additionally claimed priority from European Application No. EP 16 164 951.2, filed Apr. 12, 2016, which is incorporated herein by ...

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Номер записи: 70
07-01-2021 дата публикации

USING METADATA TO AGGREGATE SIGNAL PROCESSING OPERATIONS

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

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

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Номер записи: 71
04-01-2018 дата публикации

METHOD AND APPARATUS FOR PROCESSING TEMPORAL ENVELOPE OF AUDIO SIGNAL, AND ENCODER

Номер: US20180005638A1
Автор: Liu Zexin, Miao Lei
Принадлежит: Huawei Technologies CO.,Ltd.

A method and an apparatus for processing a temporal envelope of an audio signal, and an encoder are disclosed. When multiple temporal envelopes are solved, continuity of signal energy can be well maintained, and in addition, complexity of calculating a temporal envelope is reduced. The method includes: obtaining a high-band signal of the current frame audio signal according to the received current frame audio signal; dividing the high-band signal of the current frame signal into M subframes according to a predetermined temporal envelope quantity M, where M is an integer that is greater than or equal to 2; calculating a temporal envelope of each of the subframes; performing windowing on the first subframe of the M subframes and the last subframe of the M subframes by using an asymmetric window function; and performing windowing on a subframe except the first subframe and the last subframe of the M subframes. 1. A method for processing an audio signal , comprising:obtaining a high-band signal of a current frame signal;dividing the high-band signal of the current frame signal into M subframes, wherein M is an integer that is greater than or equal to 2; performing windowing on a first subframe of the M subframes and a last subframe of the M subframes by using a first asymmetric window function; andperforming windowing on a subframe except the first subframe and the last subframe of the M subframes.2. The method according to claim 1 , wherein before the performing windowing on the first subframe of the M subframes and the last subframe of the M subframes by using the first asymmetric window function claim 1 , the method further comprises:determining the first asymmetric window function according to a lookahead buffer length of the high-band signal of the current frame signal.3. The method according to claim 1 , wherein the performing windowing on the subframe except the first subframe and the last subframe of the M subframes comprises:performing windowing on the subframe ...

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Номер записи: 72
04-01-2018 дата публикации

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

Номер: US20180005640A1
Автор: Tsukagoshi Ikuo
Принадлежит: SONY CORPORATION

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

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Номер записи: 73
02-01-2020 дата публикации

Audio processing device and audio playback system thereof

Номер: US20200005800A1
Автор: Lichao Shi
Принадлежит: Lichao Shi

The present disclosure is provided an audio processing device and an audio playback system thereof. The audio processing device includes a receiving module configured to receive an audio signal and identify a transmission mode of the audio signal, with the transmission mode at least including a Bluetooth transmission mode and a WIFI transmission mode; a processing module configured to decode the audio signal into an analog audio signal and a digital audio signal; an output module configured to receive the analog audio signal and the digital audio signal and then output the analog audio signal to a conventional audio via an AUX analog output port and output the digital audio signal to an HiFi audio via an optical fiber output port. The present disclosure can receive audio signals with different transmission types and output the audio signals of different types, which enriches audio selectivity and is of high interest.

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Номер записи: 74
03-01-2019 дата публикации

HYBRID CONCEALMENT METHOD: COMBINATION OF FREQUENCY AND TIME DOMAIN PACKET LOSS CONCEALMENT IN AUDIO CODECS

Номер: US20190005967A1
Автор: Lecomte Jeremie, TOMASEK Adrian
Принадлежит:

Embodiments of the invention relate to an error concealment unit for providing an error concealment audio information for concealing a loss of an audio frame in an encoded audio information. The error concealment unit provides a first error concealment audio information component for a first frequency range using a frequency domain concealment. The error concealment unit also provides a second error concealment audio information component for a second frequency range, which includes lower frequencies than the first frequency range, using a time domain concealment. The error concealment unit also combines the first error concealment audio information component and the second error concealment audio information component, to obtain the error concealment audio information. Other embodiments of the invention relate to a decoder including the error concealment unit, as well as related encoders, methods, and computer programs for decoding and/or concealing. 1. An error concealment unit for providing an error concealment audio information for concealing a loss of an audio frame in an encoded audio information ,wherein the error concealment unit is configured to provide a first error concealment audio information component for a first frequency range using a frequency domain concealment,wherein the error concealment unit is further configured to provide a second error concealment audio information component for a second frequency range, which comprises lower frequencies than the first frequency range, using a time domain concealment, andwherein the error concealment unit is further configured to combine the first error concealment audio information component and the second error concealment audio information component, to acquire the error concealment audio information.2. The error concealment unit according to claim 1 ,wherein the error concealment unit is configured such that the first error concealment audio information component represents a high frequency portion of a ...

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Номер записи: 75
03-01-2019 дата публикации

TIME-DOMAIN INTER-CHANNEL PREDICTION

Номер: US20190005970A1
Автор: Atti Venkatraman, CHEBIYYAM Venkata Subrahmanyam Chandra Sekhar, SINDER Daniel Jared
Принадлежит:

A method includes decoding a low-band portion of an encoded mid channel to generate a decoded low-band mid channel. The method also includes filtering the decoded low-band mid channel according to one or more filter coefficients to generate a low-band filtered mid channel. The method also includes generating an inter-channel predicted signal based on the low-band filtered mid channel and the inter-channel prediction gain. The method further includes generating a low-band left channel and a low-band right channel based on an up-mix factor, the decoded low-band mid channel, and the inter-channel predicted signal. 1. A device comprising:a receiver configured to receive a bitstream that includes an encoded mid channel and an inter-channel prediction gain;a low-band mid channel decoder configured to decode a low-band portion of the encoded mid channel to generate a decoded low-band mid channel;a low-band mid channel filter configured to filter the decoded low-band mid channel according to one or more filter coefficients to generate a low-band filtered mid channel;an inter-channel predictor configured to generate an inter-channel predicted signal based on the low-band filtered mid channel and the inter-channel prediction gain;an up-mix processor configured to generate a low-band left channel and a low-band right channel based on an up-mix factor, the decoded low-band mid channel, and the inter-channel predicted signal;a high-band mid channel decoder configured to decode a high-band portion of the encoded mid channel to generate a decoded high-band mid channel;an inter-channel prediction mapper configured to generate a predicted high-band side channel based on the inter-channel prediction gain and a filtered version of the decoded high-band mid channel; andan inter-channel bandwidth extension decoder configured to generate a high-band left channel and a high-band right channel based on the decoded high-band mid channel and the predicted high-band side channel.2. The device ...

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Номер записи: 76
03-01-2019 дата публикации

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

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

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

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Номер записи: 77
05-01-2017 дата публикации

PERSONAL AUDIO STUDIO SYSTEM

Номер: US20170006402A1
Автор: Park Ji Hoon
Принадлежит:

One embodiment of the present invention provides technology which enables a user to process non-compressed input content or compressed input content according to settings of the user, and technology capable of selectively supporting adding, editing, and eliminating an object from the compressed input content on the basis of various coding methods. 1. A personal audio studio system , the system comprising:a selector configured to select one of non-compressed input content and compressed input content including a plurality of object signals;a first object control module configured to compress the non-compressed input content; anda second object control module configured to remove an object signal from the compressed input content, to edit the object signal for the compressed input content, or to insert the object signal into the compressed input content.2. The system of claim 1 , wherein the first object control module selectively uses one of a spatial audio object coding (SAOC) method claim 1 , a vocal harmonic coding (VHC) method claim 1 , and a residual coding (RC) method.3. The system of claim 2 , wherein the first object control module uses the RC method for outputting a down-mix signal claim 2 , an object level difference (OLD) claim 2 , and a residual signal for each object signal.4. The system of claim 1 , wherein the second object control module removes an object using one of object removal based on an SAOC method claim 1 , object removal based on a VHC method claim 1 , and object removal based on an RC method.5. The system of claim 4 , wherein the second object control module generates a weighted factor based on a removed object signal claim 4 , modifies a down-mix signal based on the weighted factor claim 4 , and modifies an OLD for each of a plurality of object signals to perform the object removal based on the SAOC method.6. The system of claim 4 , wherein the second object control module generates a weighted factor based on a removed object signal claim ...

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Номер записи: 78
02-01-2020 дата публикации

Coding Method, Decoding Method, Coder, and Decoder

Номер: US20200007153A1
Автор: Ma Fuwei, Zhang Dejun
Принадлежит:

A coding method, a decoding method, a coder, and a decoder, where the coding method includes obtaining the pulse distribution, on a track, of the pulses to be encoded on the track, determining a distribution identifier for identifying the pulse distribution according to the pulse distribution, and generating a coding index that includes the distribution identifier. The decoding method includes receiving a coding index, obtaining a distribution identifier from the coding index, wherein the distribution identifier is configured to identify the pulse distribution, on a track, of the pulses to be encoded on the track, determining the pulse distribution, on a track, of all the pulses to be encoded on the track according to the distribution identifier, and reconstructing the pulse order on the track according to the pulse distribution. 2. The coder of claim 1 , wherein the instructions further cause the processor to be configured to:obtain pulse sign information indicating positive and negative features of the pulses; anddetermine a pulse sign index corresponding to the pulse sign information, and the coding index further comprising the pulse sign index corresponding to the pulse sign information for each pulse.3. The coder of claim 1 , wherein the instructions further cause the processor to be configured to:overlay information about the second and third indices with the first index used as a first value, a value of the first index corresponding to an independent value range of the coding index, using, in the overlaid information, a combination of the second index and the third index according to I3×W(N)+I2, the I2 representing the second index, the I3 representing the third index, and the W(N) representing a total quantity of all possible distributions of the pulse positions on the track.5. The method of claim 4 , further comprising:obtaining, by the coder, pulse sign information indicating positive and negative features of each pulse; anddetermining, by the coder, a ...

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Номер записи: 79
07-01-2021 дата публикации

5G TV, VIDEO PLAYBACK METHOD BASED ON 5G TV, AND READABLE STORAGE MEDIUM

Номер: US20210006865A1
Автор: HONG Wensheng, MA Wanle, WAN Qiao, WAN Yongqi, WANG Dechuang, YU Minghuo
Принадлежит:

Disclosed are a 5G TV, a video playback method based on 5G TV and a computer readable storage medium. The 5G TV includes a 5G communication module, a decode module, and an audio and video playback module; the 5G communication module is connected to the decode module, and is configured to receive 5G audio and video signals and send the 5G audio and video signals to the decode module; and the decode module is connected to the audio and video playback module, and is configured to decode the received 5G audio and video signals to obtain audio data and video data, and is configured to send the audio data and the video data to the audio and video playback module, to make the audio and video playback module synchronously play the audio data and the video data. 1. A 5G TV , comprising: a 5G communication module , a decode module , and an audio and video playback module , wherein:the 5G communication module is connected to the decode module, and is configured to receive 5G audio and video signals and send the 5G audio and video signals to the decode module; andthe decode module is connected to the audio and video playback module, and is configured to decode the received 5G audio and video signals to obtain audio data and video data, and is configured to send the audio data and the video data to the audio and video playback module, to make the audio and video playback module synchronously play the audio data and the video data.2. The 5G TV of claim 1 , wherein:the 5G communication module comprises a 5G communication component, a 5G communication antenna, and a SIM card interface for connecting with a SIM card;a first terminal of the 5G communication component is connected to the decode module as an output terminal of the 5G communication module, a second terminal of the 5G communication component is connected to the SIM card through the SIM card interface, the 5G communication component is configured to send an online request comprising identification information after ...

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Номер записи: 80
07-01-2021 дата публикации

ADAPTING AUDIO STREAMS FOR RENDERING

Номер: US20210006918A1
Автор: Munoz Isaac Garcia, Peters Nils Günther, Swaminathan Siddhartha Goutham
Принадлежит:

In general, techniques are described for adapting audio streams for rendering. A device comprising a memory and one or more processors may be configured to perform the techniques. The memory may store a plurality of audio streams that include one or more sub-streams. The one or more processors may determine, based on the plurality of audio streams, a total number of the one or more sub-streams for all of the plurality of audio streams, and adapt, when the total number of the sub-streams is greater than a render threshold, the plurality of audio streams to decrease the number of the one or more sub-streams and obtain an adapted plurality of audio streams. The one or more processors may also apply the renderer to the adapted plurality of audio streams to obtain the one or more speaker feeds, and output the one or more speaker feeds to one or more speakers. 1. A device configured to play one or more of a plurality of audio streams , the device comprising:a memory configured to store a plurality of audio streams, each of the plurality of audio streams representative of a soundfield and include one or more sub-streams; andone or more processors coupled to the memory, and configured to:determine, based on the plurality of audio streams, a total number of the one or more sub-streams for all of the plurality of audio streams;adapt, when the total number of the one or more sub-streams is greater than a render threshold indicative of a total number of sub-streams a renderer supports when rendering the plurality of audio streams to one or more speaker feeds, the plurality of audio streams to decrease the number of the one or more sub-streams and obtain an adapted plurality of audio streams including a reduced total number of the one or more sub-streams that is equal to or less than the render threshold;apply the renderer to the adapted plurality of audio streams to obtain the one or more speaker feeds; andoutput the one or more speaker feeds to one or more speakers.2. The ...

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Номер записи: 81
03-01-2019 дата публикации

Switching Binaural Sound

Номер: US20190007776A1
Автор: Lyren Philip Scott, Norris Glen A.
Принадлежит:

A method provides binaural sound to a person through electronic earphones. The binaural sound localizes to a sound localization point (SLP) in empty space that is away from but proximate to the person. When an event occurs, the binaural sound switches or changes to stereo sound, to mono sound, or to altered binaural sound. 120.-. (canceled)21. A method executed by one or more electronic devices in a computer system to switch binaural sound to one of stereo sound and mono sound during an electronic communication between a person and a user , the method comprising:executing, by the one or more electronic devices in the computer system, the electronic communication that provides a voice of the user in binaural sound to the person such that the voice of the user in the binaural sound externally localizes to the person to a sound localization point (SLP) that is at least three feet away from a head of the person;determining, by the one or more electronic devices in the computer system during the electronic communication, when an object enters an area of the SLP;switching, by the one or more electronic devices in the computer system during the electronic communication, the binaural sound to the one of stereo sound and mono sound when the object enters the area of the SLP; andproviding, by the one or more electronic devices in the computer system during the electronic communication and in response to the switching, the voice of the user to the person in the one of stereo sound and mono sound.22. The method of claim 21 , further comprising:determining, by the one or more electronic devices in the computer system during the electronic communication, when a packet loss is above a threshold; andswitching, by the one or more electronic devices in the computer system during the electronic communication and in response to the determining that the packet loss is above the threshold, the binaural sound to the one of stereo sound and mono sound.23. The method of claim 21 , wherein ...

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Номер записи: 82
12-01-2017 дата публикации

AUDIO ENCODER DEVICE AND AN AUDIO DECODER DEVICE HAVING EFFICIENT GAIN CODING IN DYNAMIC RANGE CONTROL

Номер: US20170011749A1
Автор: Kratschmer Michael, KUECH Fabian, MEIER Michael, NEUGEBAUER Bernhard, Schreiner Stephan, UHLE Christian
Принадлежит:

An audio encoder device includes an audio encoder configured for producing an encoded audio bitstream from an audio signal having consecutive audio frames; a dynamic range control encoder configured for producing an encoded dynamic range control bitstream from an dynamic range control sequence corresponding to the audio signal and having consecutive dynamic range control frames, wherein each dynamic range control frame of the dynamic range control frames has one or more nodes, wherein each node of the one or more nodes has gain information for the audio signal and time information indicating to which point in time the gain information corresponds. 1. An audio encoder device comprising:an audio encoder configured for producing an encoded audio bitstream from an audio signal comprising consecutive audio frames;a dynamic range control encoder configured for producing an encoded dynamic range control bitstream from an dynamic range control sequence corresponding to the audio signal and comprising consecutive dynamic range control frames, wherein each dynamic range control frame of the dynamic range control frames comprises one or more nodes, wherein each node of the one or more nodes comprises gain information for the audio signal and time information indicating to which point in time the gain information corresponds;wherein the dynamic range control encoder is configured in such way that the encoded dynamic range control bitstream comprises for each dynamic range control frame of the dynamic range control frames a corresponding bitstream portion;wherein the dynamic range control encoder is configured for executing a shift procedure, wherein one or more nodes of the nodes of one reference dynamic range control frame of the dynamic range control frames are selected as shifted nodes, wherein a bit representation of each of the one or more shifted nodes of the one reference dynamic range control frame is embedded in the bitstream portion corresponding to the dynamic range ...

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Номер записи: 83
12-01-2017 дата публикации

APPARATUS AND METHOD FOR SURROUND AUDIO SIGNAL PROCESSING

Номер: US20170011750A1
Автор: Liu Zongxian, Tanaka Naoya
Принадлежит:

An apparatus for decoding surround audio signal, includes a Bitstream De-multiplexer for unpacking a bitstream into spatial parameters and core parameters, a set of Core Decoder for decoding the core parameters into a set of core signal, a matrix derivation unit for deriving the rendering matrix from the spatial parameters and playback speaker layout information, a renderer for rendering of the decoded core signal to playback signals using the rendering matrix. 1. An apparatus for decoding a surround audio signal , comprising:a Bitstream De-multiplexer for unpacking a bitstream into predominant sound parameters, ambiance parameters, channel assignment parameters and core parameters;a set of Core Decoders for decoding the core parameters into a set of core signals;a predominant sound ambiance switch for assigning the decoded core signal to predominant sound and ambiance according to the channel assignment parameters;a matrix derivation unit for deriving a predominant sound rendering matrix from the predominant sound parameters and playback speaker layout information;a matrix derivation unit for deriving an ambiance rendering matrix from the ambiance parameters and playback speaker layout information;a predominant sound renderer for rendering of the predominant sound to playback signals using the predominant sound rendering matrix;an ambiance renderer for rendering of ambient sound to the playback signals using the ambiance rendering matrix; andan output signal composition unit for composing the playback signals using the rendered predominant sound and the rendered ambient sound.2. An apparatus according to claim 1 , wherein said core decoder corresponds to MPEG-1 Audio Layer III or AAC or HE-AAC or Dolby AC-3 or MPEG USAC standard.3. An apparatus according to claim 1 , wherein said surround audio signal is High Order Ambisonics signal.4. An apparatus according to claim 1 , wherein said spatial parameters comprising of Principal Component Analysis (PCA) or Singular ...

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Номер записи: 84
12-01-2017 дата публикации

SIGNAL CLASSIFYING METHOD AND DEVICE, AND AUDIO ENCODING METHOD AND DEVICE USING SAME

Номер: US20170011754A1
Автор: CHOO Ki-hyun, OSIPOV Konstantin Sergeevich, POROV Anton Viktorovich
Принадлежит: SAMSUNG ELECTRONICS CO., LTD.

The present invention relates to an audio encoding and, more particularly, to a signal classifying method and device, and an audio encoding method and device using the same, which can reduce a delay caused by an encoding mode switching while improving the quality of reconstructed sound. The signal classifying method may comprise the operations of: classifying a current frame into one of a speech signal and a music signal; determining, on the basis of a characteristic parameter obtained from multiple frames, whether a result of the classifying of the current frame includes an error; and correcting the result of the classifying of the current frame in accordance with a result of the determination. By correcting an initial classification result of an audio signal on the basis of a correction parameter, the present invention can determine an optimum coding mode for the characteristic of an audio signal and can prevent frequent coding mode switching between frames. 1. A signal classification method comprising:classifying a current frame as one of a speech signal and a music signal;determining whether there is an error in a classification result of the current frame, based on feature parameters obtained from a plurality of frames; andcorrecting the classification result of the current frame in response to a result of the determination.2. The signal classification method of claim 1 , wherein the correcting is performed based on a plurality of independent state machines.3. The signal classification method of claim 2 , wherein the plurality of independent state machines include a music state machine and a speech state machine.4. The signal classification method of claim 1 , wherein the feature parameters are obtained from the current frame and a plurality of previous frames.5. The signal classification method of claim 1 , wherein the determining comprises determining that there is an error in the classification result when it is determined that the classification result of ...

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Номер записи: 85
11-01-2018 дата публикации

TRANSMISSION-AGNOSTIC PRESENTATION-BASED PROGRAM LOUDNESS

Номер: US20180012609A1
Автор: KOPPENS Jeroen, Norcross Scott Gregory
Принадлежит:

This disclosure falls into the field of audio coding, in particular it is related to the field of providing a framework for providing loudness consistency among differing audio output signals. In particular, the disclosure relates to methods, computer program products and apparatus for encoding and decoding of audio data bitstreams in order to attain a desired loudness level of an output audio signal. 1. A method comprising:obtaining, by a decoding device, an encoded bitstream;extracting, by the decoding device, an audio signal and metadata from the encoded bitstream, the metadata including compression curve data and loudness data;generating, by the decoding device, loudness values using the loudness data;mapping, by the decoding device, the loudness values to dynamic range compression (DRC) gains using the compression curve data; andapplying, by the decoding device, the DRC gains to the audio signal.2. The method of claim 1 , wherein the audio signal includes at least a dialog content stream and a non-dialog content stream claim 1 , and applying the DRC gains to the audio signal comprises:applying the DRC gains to a time segment of the non-dialog content stream of the audio signal to increase a loudness of the dialog content stream.3. The method of claim 1 , wherein the DRC data applies to groups of channels.4. The method of claim 3 , wherein at least some of the loudness data is associated with a specific channel in the groups of channels.5. The method of claim 1 , wherein the DRC data comprises multiple DRC profiles corresponding to DRC modes claim 1 , each DRC profile tailored to a particular audio signal to which the DRC gains can be applied.6. The method of claim 1 , wherein mapping the loudness values to DRC gains comprises a smoothing operation of the DRC gains.7. The method of claim 6 , wherein the metadata includes time-constants for use in the smoothing operation.8. The method of claim 7 , wherein the time-constants are different depending on properties ...

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Номер записи: 86
11-01-2018 дата публикации

AUDIO ENCODER AND DECODER WITH DYNAMIC RANGE COMPRESSION METADATA

Номер: US20180012610A1
Автор: Riedmiller Jeffrey, Ward Michael
Принадлежит: DOLBY LABORATORIES LICENSING CORPORATION

An audio processing unit (APU) is disclosed. The APU includes a buffer memory configured to store at least one frame of an encoded audio bitstream, where the encoded audio bitstream includes audio data and a metadata container. The metadata container includes a header and one or more metadata payloads after the header. The one or more metadata payloads include dynamic range compression (DRC) metadata, and the DRC metadata is or includes profile metadata indicative of whether the DRC metadata includes dynamic range compression (DRC) control values for use in performing dynamic range compression in accordance with at least one compression profile on audio content indicated by at least one block of the audio data. 1. An audio processing method , comprising:receiving a block of encoded audio data;receiving encoded audio metadata associated with the block of audio data;performing a cryptographic hash on the audio metadata and on at least a portion of the audio data, to produce a currently-computed hash;retrieving, from a data field associated with the encoded audio metadata, a previously-computed hash; andperforming an authentication process that involves comparing the currently-computed hash with the previously-computed hash.2. The audio processing method of claim 1 , further comprising disabling or altering at least one operation to be performed on the audio data if the authentication process succeeds.3. The audio processing method of claim 1 , wherein the metadata comprises dynamic range compression metadata.4. The audio processing method of claim 1 , wherein the metadata comprises loudness processing state metadata.5. The audio processing method of claim 1 , further comprising decoding at least a portion of the audio metadata or the audio data.6. A non-transitory medium having software stored thereon claim 1 , the software including instructions for performing an audio processing method claim 1 , the audio processing method comprising:receiving a block of encoded ...

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Номер записи: 87
11-01-2018 дата публикации

SYSTEMS AND METHODS FOR IMPLEMENTING CROSS-FADING, INTERSTITIALS AND OTHER EFFECTS DOWNSTREAM

Номер: US20180012611A1
Автор: Christine Charles W., Lowe Raymond, Ward Christopher
Принадлежит:

Systems and methods are presented for cross-fading (or other multiple clip processing) of information streams on a user or client device, such as a telephone, tablet, computer or MP3 player, or any consumer device with audio playback. Multiple clip processing can be accomplished at a client end according to directions sent from a service provider that specify a combination of (i) the clips involved; (ii) the device on which the cross-fade or other processing is to occur and its parameters; and (iii) the service provider system. For example, a consumer device with only one decoder, can utilize that decoder (typically hardware) to decompress one or more elements that are involved in a cross-fade at faster than real time, thus pre-fetching the next element(s) to be played in the cross-fade at the end of the currently being played element. The next elements(s) can, for example, be stored in an input buffer, then decoded and stored in a decoded sample buffer, all prior to the required presentation time of the multiple element effect. At the requisite time, a client device component can access the respective samples of the decoded audio clips as it performs the cross-fade, mix or other effect. Such exemplary embodiments use a single decoder and thus do not require synchronized simultaneous decodes. 131-. (canceled)32. An audio playback device , comprising:an input buffer for storing audio clips;a decoder connected to the input buffer and configured to decode the audio clips stored in the input buffer;a decoded audio buffer connected to the decoder and configured to store the audio clips decoded by the decoder; download the audio clips to be stored by the input buffer; and', 'play back the decoded audio clips stored in the decoded audio buffer with at least one fade or transition effect at a boundary between successive audio clips; and, 'a playout controller configured to determine network conditions and performance of hardware of the audio playback device; and', 'adjust ...

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Номер записи: 88
11-01-2018 дата публикации

DEVICE AND METHOD FOR CALCULATING LOUDSPEAKER SIGNALS FOR A PLURALITY OF LOUDSPEAKERS WHILE USING A DELAY IN THE FREQUENCY DOMAIN

Номер: US20180012612A1
Автор: FRANCK Andreas, Rath Michael, SLADECZEK Christoph
Принадлежит:

A device for calculating loudspeaker signals for a plurality of loudspeakers while using a plurality of audio sources, an audio source including an audio signal, includes a forward transform stage for transforming each audio signal, block-by-block, to a spectral domain so as to obtain for each audio signal a plurality of temporally consecutive short-term spectra, a memory for storing a plurality of temporally consecutive short-term spectra for each audio signal, a memory access controller for accessing a specific short-term spectrum among the plurality of short-term spectra for a combination consisting of a loudspeaker and an audio signal on the basis of a delay value, a filter stage for filtering the specific short-term spectrum for the combination of the audio signal and the loudspeaker by using a filter provided for the combination of the audio signal and the loudspeaker, so that a filtered shot-term spectrum is obtained for each combination of an audio signal and a loudspeaker, a summing stage for summing up the filtered short-term spectra for a loudspeaker so as to obtain summed-up short-term spectra for each loudspeaker, and a backtransform stage for backtransforming, block-by-block, summed-up short-term spectra for the loudspeakers to a time domain so as to obtain the loudspeaker signals. 1. A device for calculating loudspeaker signals for a plurality of loudspeakers while using a plurality of audio sources , each audio source comprising an audio signal , said device comprising:a forward transform stage configured to transform each audio signal, block-by-block, to a spectral domain so as acquire for each audio signal a plurality of temporally consecutive short-term spectra;a memory configured to store a plurality of temporally consecutive short-term spectra for each audio signal;a memory access controller configured to access a specific short-term spectrum among the plurality of temporally consecutive short-term spectra for a combination comprising a ...

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Номер записи: 89
10-01-2019 дата публикации

AUDIO OBJECT SEPARATION FROM MIXTURE SIGNAL USING OBJECT-SPECIFIC TIME/FREQUENCY RESOLUTIONS

Номер: US20190013031A1
Автор: Disch Sascha, Kastner Thorsten, Paulus Jouni
Принадлежит:

An audio decoder is proposed for decoding a multi-object audio signal including a downmix signal X and side information PSI. The side information includes object-specific side information PSIfor an audio object sin a time/frequency region R(t,f), and object-specific time/frequency resolution information TFRIindicative of an object-specific time/frequency resolution TFRof the object-specific side information for the audio object sin the time/frequency region R(t,f). The audio decoder includes an object-specific time/frequency resolution determiner configured to determine the object-specific time/frequency resolution information TFRIfrom the side information PSI for the audio object s. The audio decoder further includes an object separator configured to separate the audio object sfrom the downmix signal X using the object-specific side information in accordance with the object-specific time/frequency resolution TFRI. A corresponding encoder and corresponding methods for decoding or encoding are also described. 1. An audio decoder device for decoding a multi-object audio signal comprising a downmix signal and side information , the side information comprising object-specific side information for at least one audio object in at least one time/frequency region , and object-specific time/frequency resolution information indicative of an object-specific time/frequency resolution of the object-specific side information for the at least one audio object in the at least one time/frequency region , the audio decoder device comprising:an object-specific time/frequency resolution determiner configured to determine the object-specific time/frequency resolution information from the side information for the at least one audio object; andan object separator configured to separate the at least one audio object from the downmix signal using the object-specific side information in accordance with the object-specific time/frequency resolution,wherein the object-specific side information ...

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Номер записи: 90
12-01-2017 дата публикации

APPARATUS AND METHOD FOR AUDIO RENDERING EMPLOYING A GEOMETRIC DISTANCE DEFINITION

Номер: US20170013388A1
Автор: FUEG Simone, Grill Bernhard, Herre Juergen, Neuendorf Max, PLOGSTIES Jan
Принадлежит:

An apparatus for playing back an audio object associated with a position includes a distance calculator for calculating distances of the position to speakers or for reading the distances of the position to the speakers. The distance calculator is configured to take a solution with a smallest distance. The apparatus is configured to play back the audio object using the speaker corresponding to the solution. 1. An apparatus for playing back an audio object associated with a position , comprising:a distance calculator for calculating distances of the position to speakers,wherein the distance calculator is configured to take a solution with a smallest distance, andwherein the apparatus is configured to play back the audio object using the speaker corresponding to the solution,wherein the distance calculator is configured to calculate the distances depending on a distance function which returns a great-arc distance, or which returns weighted absolute differences in azimuth and elevation angles, or which returns a weighted angular difference.2. The apparatus according to claim 1 ,wherein the distance calculator is configured to calculate the distances of the position to the speakers only if a closest speaker playout flag, being received by the apparatus, is enabled,wherein the distance calculator is configured to take a solution with a smallest distance only if the closest speaker playout flag is enabled, andwherein the apparatus is configured to play back the audio object using the speaker corresponding to the solution only of the closest speaker playout flag is enabled.3. The apparatus according to claim 2 , wherein the apparatus is configured to not conduct any rendering on the audio object claim 2 , if the closest speaker playout flag is enabled.4. The apparatus according to claim 1 , wherein the distance function is defined according to{'br': None, 'i': a', 'az', 'el, 'diffAngle=cos(cos(Diff)*cos(Diff)),'}wherein azDiff indicates a difference of two azimuth angles, ...

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Номер записи: 91
09-01-2020 дата публикации

MDCT-Domain Error Concealment

Номер: US20200013413A1
Автор: Biswas Arijit, FRIEDRICH Tobias, PEICHL Klaus
Принадлежит: DOLBY INTERNATIONAL AB

An error-concealing audio decoding method comprises: receiving a packet comprising a set of MDCT coefficients encoding a frame of time-domain samples of an audio signal; identifying the received packet as erroneous; generating estimated MDCT coefficients to replace the set of MDCT coefficients of the erroneous packet, based on corresponding MDCT coefficients associated with a received packet directly preceding the erroneous packet; assigning signs of a first subset of MDCT coefficients of the estimated MDCT coefficients, wherein the first subset comprises such MDCT coefficients that are associated with tonal-like spectral bins, to coincide with signs of corresponding MDCT coefficients of said preceding packet; randomly assigning signs of a second subset of MDCT coefficients of the estimated MDCT coefficients, wherein the second subset comprises MDCT coefficients associated with noise-like spectral bins; replacing the erroneous packet by a concealment packet containing the estimated MDCT coefficients and the signs assigned. 1. A method for concealing errors in packets of data that are to be decoded in a modified discrete cosine transform (MDCT) based audio decoder arranged to decode a sequence of packets into a sequence of decoded frames , the method comprising:receiving, from an MDCT based audio encoder arranged to encode an audio signal, a packet comprising N/2 MDCT coefficients associated with N windowed time-domain samples of the audio signal;identifying the packet to be an erroneous packet in that the packet comprises one or more errors;estimating a first subset comprising N/4 windowed time-domain aliased samples of a first half of an intermediate frame comprising N windowed time-domain aliased samples associated with the erroneous packet, the estimation being based on relations between windowed time-domain aliased samples of the first subset and windowed time-domain samples of the N windowed time-domain samples of the audio signal;estimating a second subset ...

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Номер записи: 92
09-01-2020 дата публикации

EMBEDDING ENHANCED AUDIO TRANSPORTS IN BACKWARD COMPATIBLE AUDIO BITSTREAMS

Номер: US20200013414A1
Автор: KIM Moo Young, Peters Nils Günther, Sen Dipanjan, Thagadur Shivappa Shankar, Walters Richard Paul
Принадлежит:

In general, techniques are described by which to embed enhanced audio transports in backward compatible bitstreams. A device comprising a memory and one or more processors may be configured to perform the techniques. The memory may store the backward compatible bitstream, which conforms to a legacy transport format. The processor(s) may obtain, from the backward compatible bitstream, legacy audio data that conforms to a legacy audio format, and obtain, from the backward compatible bitstream, extended audio data that enhances the legacy audio data. The processor(s) may also obtain, based on the legacy audio data and the extended audio data, enhanced audio data that conforms to an enhanced audio format, and output the enhanced audio data to one or more speakers. 1. A device configured to process a backward compatible bitstream , the device comprising:one or more memories configured to store at least a portion of the backward compatible bitstream, the backward compatible bitstream conforming to a legacy transport format; andone or more processors configured to:obtain, from the backward compatible bitstream, legacy audio data that conforms to a legacy audio format;obtain, from the backward compatible bitstream, extended audio data that enhances the legacy audio data;obtain, based on the legacy audio data and the extended audio data, enhanced audio data that conforms to an enhanced audio format; andoutput the enhanced audio data to one or more speakers.2. The device of claim 1 , wherein the legacy transport format comprises a psychoacoustic codec transport format.3. The device of claim 2 , wherein the psychoacoustic coded transport format comprises an Advanced Audio Coding (AAC) transport format or an AptX transport format.4. The device of claim 1 ,wherein the legacy transport format comprises an Advanced Audio Coding transport format or an AptX transport format,wherein the one or more processors are configured to obtain the enhanced audio data from one or more fill ...

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Номер записи: 93
09-01-2020 дата публикации

Synchronizing enhanced audio transports with backward compatible audio transports

Номер: US20200013426A1
Автор: Dipanjan Sen, Ferdinando Olivieri, Nils Günther Peters, Shankar Thagadur Shivappa
Принадлежит: Qualcomm Inc

In general, techniques are described by which to synchronize enhanced audio transports with backward compatible audio transports. A device comprising a memory and one or more processors may be configured to perform the techniques. The memory may store a backward compatible bitstream conforming to a legacy transport format. The processor may obtain, from the backward compatible bitstream, a first audio transport stream, and obtain, from the backward compatible bitstream, a second audio transport stream. The processor(s) may also obtain, from the backward compatible bitstream, indications representative of synchronization information for the first audio transport stream and the second audio transport stream. The processor(s) may synchronize, based on the indications, the first audio transport stream and the second audio transport to obtain synchronized audio data stream. The processor(s) may obtain, based the synchronized audio data, enhanced audio data, and output the enhanced audio data to one or more speakers.

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Номер записи: 94
11-01-2018 дата публикации

AUDIO METADATA PROVIDING APPARATUS AND METHOD, AND MULTICHANNEL AUDIO DATA PLAYBACK APPARATUS AND METHOD TO SUPPORT DYNAMIC FORMAT CONVERSION

Номер: US20180014136A1
Автор: Lee Seok Jin, Lee Tae Jin, YOO Jae Hyoun
Принадлежит:

An audio metadata providing apparatus and method and a multichannel audio data playback apparatus and method to support a dynamic format conversion are provided. Dynamic format conversion information may include information about a plurality of format conversion schemes that are used to convert a first format set by an author of multichannel audio data into a second format that is based on a playback environment of the multichannel audio data and that are each set for corresponding playback periods of the multichannel audio data. The audio metadata providing apparatus may provide audio metadata including the dynamic format conversion information. The multichannel audio data playback apparatus may identify the dynamic format conversion information from the audio metadata, may convert the first format of the multichannel audio data into the second format based on the identified dynamic format conversion information, and may play back the multichannel audio data in the second format. 1. An audio metadata providing method comprising:identifying conversion information for multichannel audio data from a first format to a second format, the first format being set by an author of the multichannel audio data and the second format being based on a playback environment of the multichannel audio data; andgenerating audio metadata based on format conversion information,wherein the playback environment is determined based on a layout of speakers, which the multichannel audio data is played back,wherein the layout is associated with a position of each of the speakers or the number of the speakers.2. The audio metadata providing method of claim 1 , wherein the conversion information comprises a matrix to convert the first format into the second format.3. The audio metadata providing method of claim 1 , wherein the speaker corresponds to each channel of the multichannel audio data.4. The audio metadata providing method of claim 1 , wherein the conversion information is applied to ...

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Номер записи: 95
15-01-2015 дата публикации

MEASURING AND IMPROVING SPEECH INTELLIGIBILITY IN AN ENCLOSURE

Номер: US20150019213A1
Автор: Nongpiur Rajeev Conrad
Принадлежит:

A method for accurately estimating and improving the speech intelligibility from a loudspeaker (LS) is disclosed. A microphone is placed in a desired position and using an adaptive filter, an estimate of the clean speech signal at the microphone is generated. By using the adaptive-filter estimate of the clean speech signal and measuring the background noise in the enclosure an accurate Speech Intelligibility Index (SII) or Articulation Index (AI) measurement at the microphone position is obtained. On the basis of the estimated speech intelligibility measurement, a decision can be made if the LS signal needs to be modified to improve the intelligibility. 1. A method for determining a shape of a spectral mask , comprising:determining a prescribed distortion level of the spectral mask;determining a prescribed average gain of the spectral mask; andmaximizing an index indicative of speech intelligibility of an audio signal subject to the prescribed distortion level of the spectral mask and the prescribed average gain of the spectral mask.2. The method of claim 1 , wherein maximizing the index includes:estimating, for each of a plurality of subband components of a measured signal, a signal-to-noise ratio;determining, for each of the plurality of subband components of the measured signal, a corresponding subband gain of the spectral mask;multiplying the signal-to-noise ratios with the corresponding subband gains; andsumming the result of the multiplication.3. The method of claim 2 , further comprising:performing the estimating, determining, multiplying, and summing a plurality of times, each resulting in a candidate mask;searching over the space of candidate masks for a particular candidate mask that satisfies the prescribed distortion level and the prescribed average gain.4. The method of claim 2 , further comprising:identifying a weight for each subband component of the measured signal; andapplying the weight to the corresponding subband gains.5. The method of claim 4 , ...

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Номер записи: 96
21-01-2016 дата публикации

OPTIMIZED PARTIAL MIXING OF AUDIO STREAMS ENCODED BY SUB-BAND ENCODING

Номер: US20160019902A1
Автор: Kovesi Balazs, Lamblin Claude, Nagle Arnault
Принадлежит:

The invention relates to a method for combining a plurality of audio streams encoded by frequency sub-band encoding, comprising the following steps: decoding (E) a portion of the encoded streams over at least one frequency sub-band; combining (E) the streams thus encoded to form a mixed stream; selecting (E), from among the plurality of encoded audio streams, at least one encoded replication stream, over at least one frequency sub-band that is different from that of the decoding step. The method is such that the selection of the at least one encoded replication stream is carried out according to a criterion which takes into consideration the presence of a predetermined frequency band in the encoded stream (E). The invention also relates to a device which implements the described method and can be integrated into a conference bridge, a communication terminal or a communication gateway. 1. A method for combining a plurality of audio streams coded according to a frequency sub-band coding , comprising the following steps:{'b': '301', 'decoding (E) of a part of the streams coded on at least one frequency sub-band;'}{'b': '302', 'addition (E) of the streams thus decoded to form a mixed stream;'}{'b': '303', 'claim-text': {'b': '304', 'the method being characterized in that the selection of the at least one replication coded stream is effected according to a criterion taking into account the presence of a predetermined frequency band in the coded stream (E).'}, 'selection (E), from among the plurality of coded audio streams, of at least one replication coded stream, on at least one frequency sub-band different from that of the decoding step;'}2. The method as claimed in claim 1 , characterized in that it furthermore comprises a step of preselecting the coded audio streams according to a predetermined criterion.3. The method as claimed in claim 1 , characterized in that claim 1 , in the case where several coded streams are selected in the selection step claim 1 , an ...

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Номер записи: 97
21-01-2016 дата публикации

OPTIMIZED MIXING OF AUDIO STREAMS ENCODED BY SUB-BAND ENCODING

Номер: US20160019903A1
Автор: Kovesi Balazs, Lamblin Claude, Nagle Arnault
Принадлежит:

The invention relates to a method for mixing a plurality of audio streams coded according to a frequency sub-band coding, comprising the steps for decoding (E) a part of the coded streams over at least a first frequency sub-band, for summing (E) the streams thus decoded so as to form at least a first mixed stream. The method is such that it comprises the steps for detection (E), over at least a second frequency sub-band different from the at least first sub-band, of the presence of a predetermined frequency band in the plurality of coded audio streams and for summing (E) the decoded audio streams (E) for which the presence of the predetermined frequency band has been detected, over said at least a second sub-band, so as to form at least a second mixed stream. 1. A method for mixing a plurality of coded audio streams according to a coding by frequency sub-bands , comprising the following steps:{'b': '201', 'decoding (E) of a part of the coded streams over at least a first frequency sub-band;'}{'b': '202', 'summing (E) of the streams thus decoded so as to form at least a first mixed stream;'}the method being characterized in that it comprises the steps for:{'b': '203', 'detection (E), over at least a second frequency sub-band different from the at least first sub-band, of the presence of a predetermined frequency band within the plurality of coded audio streams;'}{'b': 205', '204, 'summing (E) of the decoded audio streams (E) for which the presence of the predetermined frequency band has been detected, over said at least second sub-band, so as to form at least a second mixed stream.'}2. The method as claimed in claim 1 , characterized in that it furthermore comprises a step for pre-selection of the coded audio streams according to a predetermined criterion claim 1 , prior to the detection step.3. The method as claimed in claim 1 , characterized in that it furthermore comprises a step of re-coding the mixed streams.4. The method as claimed in claim 1 , characterized in ...

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Номер записи: 98
19-01-2017 дата публикации

Audio Encoder and Decoder for Interleaved Waveform Coding

Номер: US20170018279A1
Автор: KJOERLING Kristofer, MUNDT Harald, Purnhagen Heiko, ROEDEN Karl Jonas, Thesing Robin
Принадлежит: DOLBY INTERNATIONAL AB

There is provided methods and apparatuses for decoding and encoding of audio signals. In particular, a method for decoding includes receiving a waveform-coded signal having a spectral content corresponding to a subset of the frequency range above a cross-over frequency. The waveform-coded signal is interleaved with a parametric high frequency reconstruction of the audio signal above the cross-over frequency. In this way an improved reconstruction of the high frequency bands of the audio signal is achieved. 1. A method for decoding an audio signal in an audio processing system , the method comprising:receiving a first waveform-coded signal having a spectral content up to a first cross-over frequency,receiving a second waveform-coded signal having a spectral content corresponding to a subset of the frequency range above the first cross-over frequency, wherein the subset of the frequency range above the first cross-over frequency includes an isolated frequency interval not contiguous with the spectral content of the first waveform-coded signal,receiving high frequency reconstruction parameters,performing high frequency reconstruction using at least a portion of the first waveform-coded signal and the high frequency reconstruction parameters so as to generate a frequency extended signal having a spectral content above the first cross-over frequency, andinterleaving the frequency extended signal with the second waveform-coded signal,wherein the audio processing system is implemented at least in part with hardware.2. The decoding method of claim 1 , wherein the spectral content of the second waveform-code signal has a time-variable upper bound.3. The decoding method of claim 1 , further comprising combining the frequency extended signal claim 1 , the second waveform-coded signal claim 1 , and the first wave-form coded signal to form a full bandwidth audio signal.4. The decoding method of claim 1 , wherein the step of performing high frequency reconstruction comprises ...

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Номер записи: 99
19-01-2017 дата публикации

Method and apparatus for encoding/decoding an audio signal

Номер: US20170018280A1
Автор: Hyun-Wook Kim, Nam-Suk Lee
Принадлежит: SAMSUNG ELECTRONICS CO LTD

Provided are a method and apparatus for encoding an audio signal and a method and apparatus for decoding an audio signal, in which errors generated during encoding and decoding of the audio signal are reduced to enhance the audio quality of a reconstructed audio signal. The method of encoding the audio signal includes detecting a pitch of the audio signal, determining a filter coefficient based on the detected pitch, performing second filtering on the audio signal, based on the determined filter coefficient; and encoding an audio signal resulting from the second filtering.

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Номер записи: 100