公开(公告)号：US09947329B2

公开(公告)日：2018-04-17

申请号：US14830484

申请日：2015-08-19

申请人： Fraunhofer-Gesellschaft zur Foerderung der angewandten Forschung e.V.

发明人： Christian Helmrich ,  Jeremie Lecomte ,  Goran Markovic ,  Markus Schnell ,  Bernd Edler ,  Stefan Reuschl

IPC分类号： G10L19/02 ,  G10L19/025 ,  H04N19/172 ,  H04N19/176 ,  H04N19/44 ,  G10L19/022

CPC分类号： G10L19/025 ,  G10L19/0212 ,  G10L19/022 ,  H04N19/172 ,  H04N19/176 ,  H04N19/44

摘要： An apparatus for encoding an audio or image signal, includes: a controllable windower for windowing the audio or image signal to provide the sequence of blocks of windowed samples; a converter for converting the sequence of blocks of windowed samples into a spectral representation including a sequence of frames of spectral values; a transient location detector for identifying a location of a transient within a transient look-ahead region of a frame; and a controller for controlling the controllable windower to apply a specific window having a specified overlap length to the audio or image signal in response to an identified location of the transient, wherein the controller is configured to select the specific window from a group of at least three windows, wherein the specific window is selected based on the transient location.

公开(公告)号：US20180018981A1

公开(公告)日：2018-01-18

申请号：US15718524

申请日：2017-09-28

申请人： Waymo LLC

发明人： Matthew Sharifi ,  Dominik Roblek

IPC分类号： G10L19/06 ,  G10L25/51 ,  G10L19/022 ,  G08B29/18 ,  G08B3/10 ,  H04R29/00 ,  G06F17/30

CPC分类号： G10L19/06 ,  G06F17/30743 ,  G08B3/10 ,  G08B29/185 ,  G10L19/022 ,  G10L25/51 ,  H04R29/00

摘要： The present disclosure provides methods and apparatuses that enable an apparatus to identify sounds from short samples of audio. The apparatus may capture an audio sample and create several audio signals of different lengths, each containing audio from the captured audio sample. The apparatus my process the several audio signals in an attempt to identify features of the audio signal that indicate an identification of the captured sound. Because shorter audio samples can be analyzed more quickly, the system may first process the shortest audio samples in order to quickly identify features of the audio signal. Because longer audio samples contain more information, the system may be able to more accurately identify features in the audio signal in longer audio samples. However, analyzing longer audio signals takes more buffered audio than identifying features in shorter signals. Therefore, the present system attempts to identify features in the shortest audio signals first.

公开(公告)号：US09858945B2

公开(公告)日：2018-01-02

申请号：US15644983

申请日：2017-07-10

申请人： Dolby International AB

发明人： Lars Villemoes

IPC分类号： G10L19/00 ,  G10L21/038 ,  G10L19/032 ,  G10L25/18 ,  G10L19/02 ,  G10L19/022 ,  G10L21/04

CPC分类号： G10L21/038 ,  G10L19/0204 ,  G10L19/022 ,  G10L19/032 ,  G10L21/04 ,  G10L25/18

摘要： The present document relates to audio source coding systems which make use of a harmonic transposition method for high frequency reconstruction (HFR), as well as to digital effect processors, e.g. exciters, where generation of harmonic distortion add brightness to the processed signal, and to time stretchers where a signal duration is prolonged with maintained spectral content. A system and method configured to generate a time stretched and/or frequency transposed signal from an input signal is described. The system comprises an analysis filterbank configured to provide an analysis subband signal from the input signal; wherein the analysis subband signal comprises a plurality of complex valued analysis samples, each having a phase and a magnitude. Furthermore, the system comprises a subband processing unit configured to determine a synthesis subband signal from the analysis subband signal using a subband transposition factor Q and a subband stretch factor S. The subband processing unit performs a block based nonlinear processing wherein the magnitude of samples of the synthesis subband signal are determined from the magnitude of corresponding samples of the analysis subband signal and a predetermined sample of the analysis subband signal. In addition, the system comprises a synthesis filterbank configured to generate the time stretched and/or frequency transposed signal from the synthesis subband signal.

公开(公告)号：US09852722B2

公开(公告)日：2017-12-26

申请号：US15118044

申请日：2015-02-18

申请人： DOLBY INTERNATIONAL AB

发明人： Arijit Biswas

IPC分类号： G10H1/40 ,  G10H7/00 ,  G10H1/00 ,  G10L19/008 ,  G10L19/16 ,  G10L25/03 ,  G10L19/022

CPC分类号： G10H1/0008 ,  G10H1/40 ,  G10H2210/076 ,  G10L19/008 ,  G10L19/022 ,  G10L19/167 ,  G10L25/03

摘要： The invention relates to estimating tempo information directly from a bitstream encoding audio information, preferably music. Said tempo information is derived from at least one periodicity derived from a detection of at least two onsets included in the audio information. Such onsets are detected via a detection of long to short block transitions (in the bitstream) or/and via a detection of a changing bit allocation (change of cost) regarding encoding/transmitting the exponents of transform coefficients encoded in the bitstream.

公开(公告)号：US20170364479A1

公开(公告)日：2017-12-21

申请号：US15696091

申请日：2017-09-05

申请人： Huawei Technologies Co., Ltd.

发明人： Deming Zhang ,  Haiting Li ,  Anisse Taleb ,  Jianfeng Xu

IPC分类号： G06F17/14 ,  G10L19/02 ,  G10L19/022

CPC分类号： G06F17/142 ,  G06F17/141 ,  G06F17/147 ,  G10L19/0212 ,  G10L19/022

摘要： A method for processing an audio signal, including: sound is converted to an analog audio input signal and converted into a digital audio signal; a windowed time domain signal is obtained and then a twiddled signal is obtained; the twiddled signal is pre-rotated and then an FFT is performed; an in-place fixed rotate compensation is performed on the FFT signal and then an post-rotated is performed; a quantized signal is obtained and then wrote into a bitstream for transmitting or storing.

公开(公告)号：US09799343B2

公开(公告)日：2017-10-24

申请号：US15372130

申请日：2016-12-07

申请人： HUAWEI TECHNOLOGIES CO., LTD.

发明人： Zexin Liu ,  Lei Miao

IPC分类号： G10L19/00 ,  G10L21/00 ,  G10L19/022 ,  G10L19/135 ,  G10L19/20 ,  G10L25/45 ,  G10L21/038 ,  G10L19/032 ,  G10L19/12

CPC分类号： G10L19/022 ,  G10L19/032 ,  G10L19/12 ,  G10L19/135 ,  G10L19/20 ,  G10L21/038 ,  G10L25/45

摘要： 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, M 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.

公开(公告)号：US20170278522A1

公开(公告)日：2017-09-28

申请号：US15621938

申请日：2017-06-13

申请人： Fraunhofer-Gesellschaft zur Foerderung der angewandten Forschung e.V.

发明人： Andreas NIEDERMEIER ,  Stephan WILDE ,  Daniel FISCHER ,  Matthias HILDENBRAND ,  Marc GAYER ,  Max NEUENDORF

IPC分类号： G10L19/22 ,  G10L19/022 ,  G10L19/24 ,  G10L21/038 ,  G10L19/20 ,  G10L19/16

CPC分类号： G10L19/22 ,  G10L19/008 ,  G10L19/022 ,  G10L19/167 ,  G10L19/20 ,  G10L19/24 ,  G10L21/038

摘要： An apparatus for decoding an encoded audio signal including bandwidth extension control data indicating either a first harmonic bandwidth extension mode or a second non-harmonic bandwidth extension mode, includes: an input interface for receiving the encoded audio signal including the bandwidth extension control data indicating either the first harmonic bandwidth extension mode or the second non-harmonic bandwidth extension mode; a processor for decoding the audio signal using the second non-harmonic bandwidth extension mode; and a controller for controlling the processor to decode the audio signal using the second non-harmonic bandwidth extension mode, even when the bandwidth extension control data indicates the first harmonic bandwidth extension mode for the encoded signal.

公开(公告)号：US09773504B1

公开(公告)日：2017-09-26

申请号：US15289792

申请日：2016-10-10

申请人： Digimarc Corporation

发明人： Ravi K. Sharma ,  Adnan M. Alattar

IPC分类号： G10L21/00 ,  G10L19/00 ,  G10L19/018 ,  G10L19/26 ,  G10L21/055 ,  G10L19/022 ,  G10L15/00 ,  G10L25/00 ,  H04N7/167 ,  G06K9/00 ,  G06K9/62 ,  G06K9/46 ,  H04L9/00 ,  H04R29/00

CPC分类号： G10L19/018 ,  G10L15/08 ,  G10L19/0019 ,  G10L19/022 ,  G10L19/12 ,  G10L19/265 ,  G10L21/055

摘要： Spectral encoding methods are more robust when used with improved weak signal detection and synchronizations methods. Further robustness gains are achieved by using informed embedding, error correction and embedding protocols that enable signal to noise enhancements by folding and pre-filtering the received signal.

公开(公告)号：US09716962B2

公开(公告)日：2017-07-25

申请号：US15379830

申请日：2016-12-15

申请人： AMX LLC

发明人： Fawad Nackvi

IPC分类号： H04S7/00 ,  G10L19/022

CPC分类号： H04S7/305 ,  G10L19/022 ,  H04S7/301 ,  H04S7/307 ,  H04S2420/07

摘要： Disclosed are an apparatus and method of processing an audio signal to optimize audio for a room environment. One example method of operation may include recording the audio signal generated within a particular room environment and processing the audio signal to create an original frequency response based on the audio signal. The method may also include creating at least two iterative filters based on at least two separate frequency ranges of the original frequency response, calculating an error difference between the frequency response modified by the at least two iterative filters and the original frequency response, and applying the error difference to the audio signal.

公开(公告)号：US20170200453A1

公开(公告)日：2017-07-13

申请号：US15469429

申请日：2017-03-24

申请人： Lester F. LUDWIG

发明人： Lester F. LUDWIG

IPC分类号： G10L19/022 ,  G10L19/16 ,  G10L19/26

CPC分类号： G10L19/022 ,  G10L13/08 ,  G10L19/167 ,  G10L19/26 ,  G10L25/48

摘要： A computer numerical processing method for encoding and decoding audio information for use in conjunction with human hearing is described. The method comprises approximating an eigenfunction equation representing a model of human hearing, calculating the approximation to each of a plurality of eigenfunctions from at least one aspect of the eigenfunction equation, and storing the approximation to each of a plurality of eigenfunctions for use in encoding and decoding. The approximation to each of a plurality of eigenfunctions represents a perception-oriented basis functions for mathematically representing audio information in a Hilbert-space representation of an audio signal space. The model of human hearing can include a bandpass operation with a bandwidth having the frequency range of human hearing and a time-limiting operation approximating the time duration correlation window of human hearing. In an embodiment, the approximated eigenfunctions comprise a convolution of a prolate spheroidal wavefunction with a trigonometric function.