公开(公告)号：US20170076728A1

公开(公告)日：2017-03-16

申请号：US15358184

申请日：2016-11-22

Applicant: PANASONIC INTELLECTUAL PROPERTY CORPORATION OF AMERICA

Inventor： Takuya KAWASHIMA ,  Masahiro OSHIKIRI

IPC: G10L19/002 ,  G10L19/035 ,  G10L19/02

CPC classification number: G10L19/002 ,  G10L19/0208 ,  G10L19/035 ,  G10L19/06 ,  G10L19/12

Abstract: A speech/audio encoding device for selectively allocating bits for higher precision encoding. The speech/audio encoding device receives a time-domain speech/audio input signal, transforms the speech/audio input signal into a frequency domain, and quantizes an energy envelope corresponding to an energy level for a frequency spectrum of the speech/audio input signal. The speech/audio encoding device further groups quantized energy envelopes into a plurality of groups, determines a perceptual significant group including one or more significant bands and a local-peak frequency, and allocates bits to a plurality of subbands corresponding to the grouped quantized energy envelopes, in which each of the subbands is obtained by splitting the frequency spectrum of the speech/audio input signal. The speech/audio encoding device encodes the frequency spectrum using the bits allocated to the subbands.

Abstract translation: 用于选择性地分配比特以用于更高精度编码的语音/音频编码装置。 语音/音频编码装置接收时域语音/音频输入信号，将语音/音频输入信号变换成频域，并量化对应于语音/音频输入信号的频谱的能级的能量包络 。 语音/音频编码装置进一步将量化的能量包络分组成多个组，确定包括一个或多个有效频带和局部峰值频率的感知有效组，并将比特分配给对应于分组的量化能量包络的多个子带 ，其中通过分割语音/音频输入信号的频谱来获得每个子带。 语音/音频编码设备使用分配给子带的比特对频谱进行编码。

公开(公告)号：US20170061977A1

公开(公告)日：2017-03-02

申请号：US15352729

申请日：2016-11-16

Applicant: Telefonaktiebolaget LM Ericsson (publ)

Inventor： Sebastian Näslund ,  Volodya Grancharov ,  Erik Norvell

IPC: G10L19/038 ,  G10L19/02 ,  G10L19/035 ,  G10L19/10

CPC classification number: G10L19/02 ,  G10L19/0212 ,  G10L19/035 ,  G10L19/038 ,  G10L19/10

Abstract: The embodiments of the present invention improves conventional attenuation schemes by replacing constant attenuation with an adaptive attenuation scheme that allows more aggressive attenuation, without introducing audible change of signal frequency characteristics.

Abstract translation: 本发明的实施例通过用允许更积极衰减的自适应衰减方案代替恒定衰减来改进常规衰减方案，而不引入信号频率特性的可听见的改变。

公开(公告)号：US20170025131A1

公开(公告)日：2017-01-26

申请号：US15222283

申请日：2016-07-28

Applicant: Intel Corporation

Inventor： Dmitry N. Budnikov ,  Igor Igor Chikalov ,  Sergey N. Zheltov

IPC: G10L19/035 ,  G10L19/02 ,  G10L19/012

CPC classification number: G10L19/035 ,  G10L19/012 ,  G10L19/0204 ,  G10L21/04

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

Abstract translation: 用于处理音频数据的方法包括：确定用于表示帧中的量化音频数据的第一公共比例因子值。 确定用于表示帧中的量化音频数据的第二公共比例因子值。 根据第一和第二公共比例因子值确定线方程公式比例因子值。

公开(公告)号：US20170004839A1

公开(公告)日：2017-01-05

申请号：US15266862

申请日：2016-09-15

Applicant: Fraunhofer-Gesellschaft zur Foerderung der Angewandten Forschung e.V.

Inventor： Nikolaus RETTELBACH ,  Bernhard GRILL ,  Guillaume FUCHS ,  Stefan GEYERSBERGER ,  Markus MULTRUS ,  Harald POPP ,  Juergen HERRE ,  Stefan WABNIK ,  Gerald SCHULLER ,  Jens HIRSCHFELD

IPC: G10L19/035 ,  G10L19/028 ,  G10L19/02 ,  G10L19/008

CPC classification number: G10L19/035 ,  G10L19/008 ,  G10L19/02 ,  G10L19/0204 ,  G10L19/028 ,  G10L19/032 ,  G10L25/18

Abstract: An encoder for providing an audio stream on the basis of a transform-domain representation of an input audio signal includes a quantization error calculator configured to determine a multi-band quantization error over a plurality of frequency bands of the input audio signal for which separate band gain information is available. The encoder also includes an audio stream provider for providing the audio stream such that the audio stream includes information describing an audio content of the frequency bands and information describing the multi-band quantization error.A decoder for providing a decoded representation of an audio signal on the basis of an encoded audio stream representing spectral components of frequency bands of the audio signal includes a noise filler for introducing noise into spectral components of a plurality of frequency bands to which separate frequency band gain information is associated on the basis of a common multi-band noise intensity value.

Abstract translation: 用于基于表示音频信号的频带的频带的频谱分量的编码音频流提供音频信号的解码表示的解码器包括噪声填充器，用于将噪声引入多个频带的频谱分量中，分频频带 增益信息是根据共同的多频带噪声强度值相关联的。

公开(公告)号：US20160329056A1

公开(公告)日：2016-11-10

申请号：US15110356

申请日：2014-01-13

Applicant: NOKIA TECHNOLOGIES OY

Inventor： Adriana VASILACHE ,  Lasse Juhani LAAKSONEN ,  Anssi Sakari RÄMÖ

IPC: G10L19/008 ,  G10L25/18 ,  G10L19/035 ,  G10L19/02

CPC classification number: G10L19/008 ,  G10L19/0204 ,  G10L19/035 ,  G10L19/22 ,  G10L25/03 ,  G10L25/18 ,  H04R2499/11 ,  H04S1/007 ,  H04S3/008 ,  H04S2420/01 ,  H04S2420/07

Abstract: It is disclosed inter alia a method comprising: estimating a value of entropy for a multi-channel audio signal; determining a channel configuration of the multi-channel audio signal from the value of entropy; and encoding the multi-channel audio signal, wherein the mode of encoding is dependent on the channel configuration.

Abstract translation: 特别地公开了一种方法，包括：估计多声道音频信号的熵值; 从熵值确定多声道音频信号的信道配置; 并对多声道音频信号进行编码，其中编码模式取决于信道配置。

公开(公告)号：US20160196826A1

公开(公告)日：2016-07-07

申请号：US14916808

申请日：2013-09-05

Applicant: Michael Anthony STONE ,  SAMSUNG ELECTRONICS CO., LTD.

Inventor： Michael Anthony STONE ,  Han-gil MOON ,  Nam-suk LEE ,  Eun-mi OH

IPC: G10L19/005 ,  G10L19/035 ,  G10L15/12 ,  G10L19/022

CPC classification number: G10L19/005 ,  G10L15/12 ,  G10L19/022 ,  G10L19/032 ,  G10L19/035

Abstract: Provided are an apparatus and a method for encoding and decoding audio signals, in which when determining a masking threshold according to a psychoacoustic model, accurate results may be obtained for a short window-based audio signal as well as for a long window-based audio signal. The apparatus for encoding audio signals according to the present invention comprises a masking threshold determining unit configured to determine, on the basis of a frame length of a first window having a divided audio signal, a masking threshold for a second window that has a different frame length from that of the first window.

Abstract translation: 提供了一种用于对音频信号进行编码和解码的装置和方法，其中当根据心理声学模型确定掩蔽阈值时，可以获得针对基于短窗口的音频信号以及基于长窗口的音频的准确结果 信号。 根据本发明的用于编码音频信号的装置包括：掩蔽阈值确定单元，被配置为基于具有分割音频信号的第一窗口的帧长度确定具有不同帧的第二窗口的掩蔽阈值 从第一个窗口的长度。

公开(公告)号：US09384746B2

公开(公告)日：2016-07-05

申请号：US14512892

申请日：2014-10-13

Applicant: QUALCOMM Incorporated

Inventor： Venkatraman S. Atti ,  Venkatesh Krishnan ,  Stephane Pierre Villette ,  Vivek Rajendran

IPC: G10L19/035 ,  G10L21/038 ,  G10L19/08 ,  G10L19/083 ,  G10L19/02

CPC classification number: G10L19/035 ,  G10L19/0204 ,  G10L19/08 ,  G10L19/083 ,  G10L21/038

Abstract: A method includes determining a first modeled high-band signal based on a low-band excitation signal of an audio signal, where the audio signal includes a high-band portion and a low-band portion. The method also includes determining scaling factors based on energy of sub-frames of the first modeled high-band signal and energy of corresponding sub-frames of the high-band portion of the audio signal. The method includes applying the scaling factors to a modeled high-band excitation signal to determine a scaled high-band excitation signal and determining a second modeled high-band signal based on the scaled high-band excitation signal. The method includes determining gain parameters based on the second modeled high-band signal and the high-band portion of the audio signal.

Abstract translation: 一种方法包括基于音频信号的低频带激励信号来确定第一建模的高频带信号，其中音频信号包括高频带部分和低频带部分。 该方法还包括基于第一建模高频带信号的子帧的能量和音频信号的高频带部分的对应子帧的能量来确定缩放因子。 该方法包括将缩放因子应用于建模的高频带激励信号以确定缩放的高频带激励信号，并且基于经缩放的高频带激励信号确定第二建模的高频带信号。 该方法包括基于第二建模高频带信号和音频信号的高频带部分确定增益参数。

公开(公告)号：US09299357B2

公开(公告)日：2016-03-29

申请号：US14157157

申请日：2014-01-16

Applicant: SAMSUNG ELECTRONICS CO., LTD.

Inventor： Kang Eun Lee ,  Do Hyung Kim ,  Chang Yong Son ,  Shi Hwa Lee

IPC: G10L19/002 ,  G10L19/16 ,  G10L19/035

CPC classification number: G10L19/173 ,  G10L19/035

Abstract: An apparatus and method for decoding audio data. The apparatus for decoding the audio data may perform block data unpacking by preferring a channel order to a block order from a bitstream, and perform dithering through preferring a block order to a channel order. Complexity in decoding may be reduced through integrating bitstream searching and the bock data unpacking, and a dithering error may be prevented through processing the block data unpacking and the dithering separately.

Abstract translation: 一种用于解码音频数据的装置和方法。 用于对音频数据进行解码的装置可以通过优选从比特流的频道顺序到块顺序来执行块数据解包，并且通过优选块顺序到频道顺序来执行抖动。 可以通过集成比特流搜索和块状数据解包来减少解码的复杂度，并且可以通过分别处理块数据解包和抖动来防止抖动错误。

公开(公告)号：US09153240B2

公开(公告)日：2015-10-06

申请号：US13939931

申请日：2013-07-11

Applicant: Telefonaktiebolaget L M Ericsson (publ)

Inventor： Manuel Briand ,  Anisse Taleb

IPC: G10L19/02 ,  G10L19/035

CPC classification number: G10L19/0204 ,  G10L19/0212 ,  G10L19/035

Abstract: In a method of perceptual transform coding of audio signals in a telecommunication system, performing the steps of determining transform coefficients representative of a time to frequency transformation of a time segmented input audio signal; determining a spectrum of perceptual sub-bands for said input audio signal based on said determined transform coefficients; determining masking thresholds for each said sub-band based on said determined spectrum; computing scale factors for each said sub-band based on said determined masking thresholds, and finally adapting said computed scale factors for each said sub-band to prevent energy loss for perceptually relevant sub-bands.

Abstract translation: 在电信系统中对音频信号的感知变换编码的方法中，执行以下步骤：确定表示时间分段输入音频信号的时间到频率变换的变换系数; 基于所述确定的变换系数确定所述输入音频信号的感知子带的频谱; 基于所述确定的频谱确定每个所述子带的掩蔽阈值; 基于所述确定的掩蔽阈值计算每个所述子带的比例因子，并且最后适应每个所述子带的所述计算的比例因子以防止感知相关子带的能量损失。

公开(公告)号：US09135923B1

公开(公告)日：2015-09-15

申请号：US14605571

申请日：2015-01-26

Applicant: Chengjun Julian Chen

Inventor： Chengjun Julian Chen

IPC: G10L19/02 ,  G10L19/125 ,  G10L19/038 ,  G10L19/035 ,  G10L25/90 ,  G10L19/00

CPC classification number: G10L25/90 ,  G10L19/0212 ,  G10L19/20

Abstract: A pitch-synchronous method and system for speech coding using timbre vectors is disclosed. On the encoder side, speech signal is segmented into pitch-synchronous frames without overlap, then converted into a pitch-synchronous amplitude spectrum using FFT. Using Laguerre functions, the amplitude spectrum is transformed into a timbre vector. Using vector quantization, each timbre vector is converted to a timbre index based on a timbre codebook. The intensity and pitch are also converted into indices respectively using scalar quantization. Those indices are transmitted as encoded speech. On the decoder side, by looking up the same codebooks, pitch, intensity and the timbre vector are recovered. Using Laguerre functions, the amplitude spectrum is recovered. Using Kramers-Kronig relations, the phase spectrum is recovered. Using FFT, the elementary waves are regenerated, and superposed to become the speech signal.

Abstract translation: 公开了一种使用音色矢量进行语音编码的音调同步方法和系统。 在编码器侧，语音信号被分割为音调同步帧而不重叠，然后使用FFT转换为音调同步幅度谱。 使用Laguerre函数，将幅度谱转换为音色矢量。 使用矢量量化，基于音色码本将每个音色矢量转换为音色索引。 强度和音调也分别使用标量量化转换成索引。 这些索引作为编码语音传输。 在解码器侧，通过查找相同的码本，恢复音高，强度和音色矢量。 使用Laguerre函数，恢复幅度谱。 使用Kramers-Kronig关系，恢复相位谱。 使用FFT，基本波被再生，叠加成语音信号。