专利快速检索-快速检索全球专利，免费商用专利数据库-IPRDB

1. 发明申请

WO0115144A8 VOICE ENCODER AND VOICE ENCODING METHOD 审中-公开
标题翻译：语音编码器和语音编码方法
公开(公告)号：WO0115144A8
公开(公告)日：2001-04-26
申请号：PCT/JP0005621
申请日：2000-08-23
申请人： MATSUSHITA ELECTRIC IND CO LTD , YASUNAGA KAZUTOSHI , MORII TOSHIYUKI
发明人： YASUNAGA KAZUTOSHI , MORII TOSHIYUKI
IPC分类号： G10L19/083 , G10L19/09 , G10L19/16 , G10L19/04
CPC分类号： G10L19/16 , G10L19/083 , G10L19/09
摘要： A vector code book (1094) where representative samples of vectors to be quantized are stored is created. Each vector is made up of three elements: an AC gain, a value corresponding the logarithm of an SC gain, and an adjustment coefficient of the prediction coefficient of SC. Coefficients for predictive coding are stored in a prediction coefficient storage section (1095). The coefficients are the prediction coefficients of MA, and two kinds of coefficients, AC and SC for the order of prediction are stored. A parameter calculating section (1091) calculates a parameter necessary for distance calculation from an auditory sensation weighting input voice, an adaptive sound source subjected to auditory weighting LPC synthesis, a probabilistic sound source subjected to auditory sensation weighting LPC synthesis, a decoded vector (AC, SC, adjustment coefficient) stored in a decoded vector storage section (1096), and the prediction coefficients (AC, SC) stored in the prediction coefficient storage section (1095).
摘要翻译：创建存储待量化向量的代表性样本的向量码本（1094）。每个矢量由三个元素组成：AC增益，对应于SC增益的对数的值，以及SC的预测系数的调整系数。用于预测编码的系数被存储在预测系数存储部分（1095）中。这些系数是MA的预测系数，并且存储用于预测顺序的两种系数AC和SC。参数计算部分（1091）根据听觉加权输入语音，进行听觉加权LPC合成的自适应声源，进行听觉加权LPC合成的概率声源，解码矢量（AC 存储在解码矢量存储部（1096）中的预测系数（SC，调整系数）和存储在预测系数存储部（1095）中的预测系数（AC，SC）。

2. 发明申请

WO2007099244A2 METHOD FOR LIMITING ADAPTIVE EXCITATION GAIN IN AN AUDIO DECODER 审中-公开
标题翻译：在音频解码器中限制自适应激励增益的方法
公开(公告)号：WO2007099244A2
公开(公告)日：2007-09-07
申请号：PCT/FR2007050779
申请日：2007-02-13
申请人： FRANCE TELECOM , KOVESI BALAZS , VIRETTE DAVID
发明人： KOVESI BALAZS , VIRETTE DAVID
IPC分类号： G10L19/00 , G10L19/005 , G10L19/08 , G10L19/083
CPC分类号： G10L19/083 , G10L19/005
摘要： The invention concerns a decoder for an audio signal coded by an encoder comprising a long-term predictive filter. According to the invention, said decoder comprises: a block (211) for detecting losses of transmission frames, a module (222) for calculating values of an error indicating function, representing the accumulated error in decoding on the adaptive excitation following said transmission frame loss, an arbitrary value being assigned to said adaptive excitation for the lost frame, a module (213) for calculating an error indicating parameter based on said values of the error indicating function, a comparator (214) of said error indicating parameter with at least one given threshold, a discriminator (215) for determining based on of the result provided by the comparator (214) a value of at least one adaptive excitation gain to be used by the decoder. The invention is applicable to encoding and decoding digital signals such as audiofrequency signals.
摘要翻译：本发明涉及由编码器编码的音频信号的解码器，该编码器包括长期预测滤波器。根据本发明，所述解码器包括：用于检测传输帧丢失的块（211）;用于计算错误指示函数的值的模块（222），表示解码所述传输帧丢失之后的自适应激励的累积错误，为丢失帧分配给所述自适应激励的任意值，用于基于所述误差指示函数的值计算误差指示参数的模块（213），所述误差指示参数的比较器（214）具有至少一个给定阈值;鉴别器（215），用于基于由比较器（214）提供的结果来确定解码器要使用的至少一个自适应激励增益的值。本发明适用于对诸如音频信号的数字信号进行编码和解码。

3. 发明申请

WO9211627A3 SPEECH CODING 审中-公开
标题翻译：语音编码
公开(公告)号：WO9211627A3
公开(公告)日：1992-10-29
申请号：PCT/GB9102291
申请日：1991-12-20
申请人： BRITISH TELECOMM
发明人： FREEMAN DANIEL KENNETH , WONG WING-TAK-KENNETH , DAVIS ANDREW GORDON
IPC分类号： G10L19/083 , G10L19/12 , G10L9/14
CPC分类号： G10L19/083 , G10L19/12
摘要： In a codebook excited speech coder, speech is analysed (200) to produce coefficients of a synthesis filter and (203, 204) the parameters of a long-term prediction filter (LTP) and a codeword indication one of a set of excitations (stored in 205), the results being transmitted to a receiver where they can be used to resynthesise the speech. The LTP and excitation analysis involve computation (224) of impulse response products (stored in 301, 302). Computation of new products is performed by adding additional terms to products already formed. Multiplication of these products by excitation terms is performed using a pointer table (303) storing precalculated addresses of locations in the store (301, 302). If some excitations are shifted versions of others, some addressed can be obtained by modifying other addresses. The LTP analysis (203) may include selection between a simple delay prediction and a prediction consisting of the sum of two differently delayed terms, to provide improved predictor delay resolution.

4. 发明申请

WO2015057680A1 METHOD, APPARATUS, DEVICE, COMPUTER-READABLE MEDIUM FOR BANDWIDTH EXTENSION OF AN AUDIO SIGNAL USING A SCALED HIGH-BAND EXCITATION 审中-公开
标题翻译：方法，装置，设备，使用标准高频带激励的音频信号带宽扩展的计算机可读介质
公开(公告)号：WO2015057680A1
公开(公告)日：2015-04-23
申请号：PCT/US2014/060448
申请日：2014-10-14
申请人： QUALCOMM INCORPORATED
发明人： ATTI, Venkatraman, S. , KRISHNAN, Venkatesh , VILLETTE, Stephane, Pierre , RAJENDRAN, Vivek
IPC分类号： G10L21/038 , G10L19/083
CPC分类号： G10L19/035 , G10L19/0204 , G10L19/08 , G10L19/083 , G10L21/038
摘要： 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.
摘要翻译：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的高频带信号和音频信号的高频带部分确定增益参数。

5. 发明申请

WO2009096717A3 METHOD AND APPARATUS FOR ENCODING AND DECODING AUDIO SIGNAL 审中-公开
标题翻译：编码和解码音频信号的方法和装置
公开(公告)号：WO2009096717A3
公开(公告)日：2009-09-24
申请号：PCT/KR2009000435
申请日：2009-01-29
申请人： SAMSUNG ELECTRONICS CO LTD
发明人： LEE GEON-HYOUNG , LEE CHUL-WOO , JEONG JONG-HOON , LEE NAM-SUK , MOON HAN-GIL
IPC分类号： G10L19/00 , G10L19/08 , G10L19/083
CPC分类号： G10L19/083
摘要： Disclosed are a method and an apparatus for efficiently encoding and decoding a signal of high frequency band that is higher than a critical frequency among audio signals. The disclosed method and apparatus encode a high frequency signal with a small number of bits to improve voice quality by encoding both a linear prediction coding coefficient and the gain information of a residual signal. The linear prediction coding coefficient is created through a linear prediction coding analysis for a high frequency signal.
摘要翻译：公开了一种高效编码和解码音频信号中高于临界频率的高频信号的方法和装置。所公开的方法和装置对具有少量位的高频信号进行编码，以通过对线性预测编码系数和残差信号的增益信息进行编码来提高语音质量。线性预测编码系数是通过对高频信号的线性预测编码分析产生的。

6. 发明申请

WO00016315A2 LINEAR PREDICTIVE ANALYSIS-BY-SYNTHESIS ENCODING METHOD AND ENCODER 审中-公开
标题翻译：线性预测分析 - 合成编码方法和编码器
公开(公告)号：WO00016315A2
公开(公告)日：2000-03-23
申请号：PCT/SE1999/001433
申请日：1999-08-24
IPC分类号： H03M7/30 , G10L19/083 , H03M7/36 , G10L19/14
CPC分类号： G10L19/083
摘要： A linear predictive analysis-by-synthesis encoder includes a search algorithm block (50) and a vector quantizer (58) for vector quantizing optimal gains from a plurality of subframes in a frame. The internal encoder states are updated using (50, 52, 54, 56) the vector quantized gains.
摘要翻译：线性预测分析合成编码器包括搜索算法块（50）和矢量量化器（58），用于矢量量化来自帧中的多个子帧的最佳增益。使用（50,52,54,56）矢量量化增益更新内部编码器状态。

7. 发明申请

WO2021207825A1 METHOD AND DEVICE FOR SPEECH/MUSIC CLASSIFICATION AND CORE ENCODER SELECTION IN A SOUND CODEC 审中-公开
公开(公告)号：WO2021207825A1
公开(公告)日：2021-10-21
申请号：PCT/CA2021/050465
申请日：2021-04-08
申请人： VOICEAGE CORPORATION
发明人： MALENOVSKY, Vladimir
IPC分类号： G10L19/20 , G10L19/008 , G10L19/083 , G10L25/15
摘要： Two-stage speech/music classification device and method classify an input sound signal and select a core encoder for encoding the sound signal. A first stage classifies the input sound signal into one of a number of final classes. A second stage extracts high-level features of the input sound signal and selects the core encoder for encoding the input sound signal in response to the extracted high-level features and the final class selected in the first stage.

8. 发明申请

WO01084541A1 METHOD FOR IMPROVING SPEECH QUALITY IN SPEECH TRANSMISSION TASKS 审中-公开
标题翻译：方法提高语音质量进行语音传输的任务
公开(公告)号：WO01084541A1
公开(公告)日：2001-11-08
申请号：PCT/EP2001/002603
申请日：2001-03-08
IPC分类号： G10L19/083 , G10L19/09 , G10L25/78 , G10L19/14
CPC分类号： G10L19/083 , G10L19/09 , G10L25/78
摘要： The invention relates to a method for calculating the amplication factor, which co-determines the volume, for a speech signal transmitted in encoded form. Said speech signal is divided into short temporal signal segments. The individual signal segments are encoded and transmitted separately from each other, and the amplication factor for each signal segment is calculated, transmitted and used by the decoder to reconstruct the signal. The amplication factor is determined by minimizing the value E(g_opt2) = (1-a) *f>1 2
摘要翻译：本发明涉及一种方法，用于计算体积共同确定用于编码发送的语音信号，其中，所述语音信号被划分成时间短的信号部分，并且彼此独立地编码和发送的各个信号部分的增益因子，并计算放大因子的每一个信号部分，发射和从解码器到所述信号的重建被使用，其中，所述增益通过最小化E（g_opt2）=（1-α）* F + F> 2的尺寸被确定> 1 <（g_opt2）的a * <（g_opt2），其中，所述加权因子的确定考虑到发生时，周期性和编码的语音信号的平稳性两者。

9. 发明申请

WO99041737A1 METHOD AND APPARATUS FOR HIGH SPEED DETERMINATION OF AN OPTIMUM VECTOR IN A FIXED CODEBOOK 审中-公开
标题翻译：用于在固定的代码簿中快速确定最佳矢量的方法和装置
公开(公告)号：WO99041737A1
公开(公告)日：1999-08-19
申请号：PCT/RU1998/000041
申请日：1998-02-17
IPC分类号： G10L19/083 , G10L19/12 , H03M7/30 , G10L9/14
CPC分类号： G10L19/083 , G10L19/12 , G10L2019/0002 , G10L2019/0013
摘要： A method for a CELP algorithm including the steps of pre-processing (101) a sampled speech s{n} in a signal pre-processor so as to output at least a noise filtered speech output vector and a channel noise estimate, model parameter estimation (102) of the noise filtered speech output vector so as to output a prediction residual and a long term prediction gain, encoding (104 - 120) the prediction residual so as to output an adaptive codebook vector including an index of impulse response functions of a filter and a vector gain, formatting (121) the encoded speech packets, is proposed wherein the step of encoding (104 - 120) comprises in the following order the steps of determination (104 - 109) of the gain by choosing a start value close to a theoretical optimal value, and vector optimisation (110 - 120) by successive searching for an extremum of an estimate function based on a recursively corrected correlation vector. Further, a digital signal processor for processing electrical signals to determine a codebook vector and a gain of said codebook vector is provided that operates correspondingly to the method according to the invention.
摘要翻译：一种用于CELP算法的方法，包括以下步骤：在信号预处理器中预处理（101）采样语音s {n}，以便至少输出噪声滤波语音输出向量和信道噪声估计，模型参数估计（102），以输出预测残差和长期预测增益，对预测残差进行编码（104-120），以便输出包括脉冲响应函数的指标的自适应码本向量滤波器和向量增益，格式化（121）编码的语音分组，其中编码步骤（104-120）按以下顺序包括通过选择开始值关闭的增益的确定（104-109）的步骤通过基于递归校正的相关向量连续搜索估计函数的极值，获得理论最优值和向量优化（110-120）。此外，提供了一种用于处理电信号以确定码本矢量和所述码本矢量的增益的数字信号处理器，其对应于根据本发明的方法进行操作。

10. 发明申请

WO2015055531A1 CONCEPT FOR ENCODING AN AUDIO SIGNAL AND DECODING AN AUDIO SIGNAL USING SPEECH RELATED SPECTRAL SHAPING INFORMATION 审中-公开
标题翻译：使用语音相关光谱形状信息编码音频信号和解码音频信号的概念
公开(公告)号：WO2015055531A1
公开(公告)日：2015-04-23
申请号：PCT/EP2014/071767
申请日：2014-10-10
申请人： FRAUNHOFER-GESELLSCHAFT ZUR FÖRDERUNG DER ANGEWANDTEN FORSCHUNG E.V.
发明人： FUCHS, Guillaume , MULTRUS, Markus , RAVELLI, Emmanuel , SCHNELL, Markus
IPC分类号： G10L19/20 , G10L19/083
CPC分类号： G10L19/12 , G10L19/032 , G10L19/083 , G10L19/20 , G10L25/93
摘要： According to an aspect of the present invention an encoder for encoding an audio signal comprises an analyzer configured for deriving prediction coefficients and a residual signal from a frame of the audio signal. The encoder comprises a formant information calculator configured for calculating a speech related spectral shaping information from the prediction coefficients, a gain parameter calculator configured for calculating a gain parameter from an unvoiced residual signal and the spectral shaping information and a bitstream former configured for forming an output signal based on an information related to a voiced signal frame, the gain parameter or a quantized gain parameter and the prediction coefficients.
摘要翻译：根据本发明的一个方面，用于对音频信号进行编码的编码器包括被配置为从音频信号的帧导出预测系数和残差信号的分析器。所述编码器包括：共振峰信息计算器，被配置为从所述预测系数计算与语音有关的频谱整形信息;增益参数计算器，被配置为从无声残差信号和所述频谱整形信息计算增益参数;以及比特流形成器，被配置用于形成输出基于与有声信号帧相关的信息，增益参数或量化增益参数以及预测系数的信号。

你已经成功收藏专利！

检索式保存成功!

IPRDB

热门服务

关于我们

友情链接

联系方式