公开(公告)号：WO2021104623A1

公开(公告)日：2021-06-03

申请号：PCT/EP2019/082802

申请日：2019-11-27

Applicant: FRAUNHOFER-GESELLSCHAFT ZUR FÖRDERUNG DER ANGEWANDTEN FORSCHUNG E.V. ,  FRIEDRICH-ALEXANDER-UNIVERSITAET ERLANGEN-NUERNBERG

Inventor： GUO, Ning ,  EDLER, Bernd

IPC: G10L19/09 ,  G10L19/02

Abstract: An encoder (100) for encoding a current frame of an audio signal depending on one or more previous frames of the audio signal according to an embodiment is provided. The one or more previous frames precede the current frame, wherein each of the current frame and the one or more previous frames comprises one or more harmonic components of the audio signal, wherein each of the current frame and the one or more previous frames comprises a plurality of spectral coefficients in a frequency domain or in a transform domain. To generate an encoding of the current frame, the encoder (100) is to determine an estimation of two harmonic parameters for each of the one or more harmonic components of a most previous frame of the one or more previous frames. Moreover, the encoder (100) is to determine the estimation of the two harmonic parameters for each of the one or more harmonic components of the most previous frame using a first group of three or more of the plurality of spectral coefficients of each of the one or more previous frames of the audio signal.

公开(公告)号：WO2010079167A4

公开(公告)日：2010-10-14

申请号：PCT/EP2010050057

申请日：2010-01-05

Applicant: SKYPE LTD ,  VOS KOEN BERNARD ,  JENSEN SOREN SKAK

Inventor： VOS KOEN BERNARD ,  JENSEN SOREN SKAK

IPC: G10L19/12 ,  G10L19/09

CPC classification number: G10L19/12 ,  G10L19/09

Abstract: A method, system and computer program for encoding speech according to a source-filter model. The method comprises deriving a spectral envelope signal representative of a modelled filter and a first remaining signal representative of a modelled source signal, and deriving a second remaining signal from the first remaining signal by, at intervals during the encoding: exploiting a correlation between approximately periodic portions in the first remaining signal to generate a predicted version of a later portion from a stored version of an earlier portion, and using the predicted version of the later portion to remove an effect of said periodicity from the first remaining signal. The method further comprises, once every number of intervals, transforming the stored version of the earlier portion of the first remaining signal prior to generating the predicted version of the respective later portion.

Abstract translation: 一种根据源滤波器模型对语音进行编码的方法，系统和计算机程序。 该方法包括导出表示建模过滤器的频谱包络信号和表示建模源信号的第一剩余信号，以及在编码期间间隔从第一剩余信号导出第二剩余信号：利用近似周期性的相关性 第一剩余信号中的部分，以从较早部分的存储版本生成稍后部分的预测版本，并​​且使用稍后部分的预测版本来从第一剩余信号中去除所述周期性的影响。 该方法还包括：每产生一次间隔之后，在生成相应较后部分的预测版本之前变换第一剩余信号的较早部分的存储版本。

公开(公告)号：WO02017301A1

公开(公告)日：2002-02-28

申请号：PCT/EP2001/009618

申请日：2001-08-13

IPC: G10L19/005 ,  G10L19/09 ,  G10L19/00

CPC classification number: G10L19/005 ,  G10L19/09

Abstract: An audio transmission system comprises: a decoder for converting a frame organized bitstream into an audio output representation; and a bad frame processing means arranged for detecting bad or disturbed frames in the bitstream. The audio transmission system further comprises a pitch period estimator coupled to said decoder audio output for estimating the pitch period of the audio representation and the pitch period estimator is further coupled to the bad frame processing means for replacing the audio output during a detected bad frame by a pitch period determined representation of said audio output. As a consequence no smoothing is necessary at the edges of neighboring frames.

Abstract translation: 音频传输系统包括：用于将帧组织的比特流转换成音频输出表示的解码器; 以及布置成用于检测比特流中的不良或干扰帧的坏帧处理装置。 音频传输系统还包括音调周期估计器，耦合到所述解码器音频输出，用于估计音频表示的音调周期，并且音调周期估计器进一步耦合到坏帧处理装置，用于在检测到的坏帧期间替换音频输出 音调周期确定所述音频输出的表示。 因此，在相邻帧的边缘处不需要平滑。

公开(公告)号：WO2014202784A1

公开(公告)日：2014-12-24

申请号：PCT/EP2014/063171

申请日：2014-06-23

Applicant: FRAUNHOFER-GESELLSCHAFT ZUR FÖRDERUNG DER ANGEWANDTEN FORSCHUNG E.V.

Inventor： SCHNABEL, Michael ,  MARKOVIC, Goran ,  SPERSCHNEIDER, Ralph ,  LECOMTE, Jérémie ,  HELMRICH, Christian

IPC: G10L19/005 ,  G10L25/90 ,  G10L19/09

CPC classification number: G10L19/005 ,  G10L19/002 ,  G10L19/012 ,  G10L19/0212 ,  G10L19/06 ,  G10L19/07 ,  G10L19/083 ,  G10L19/09 ,  G10L19/12 ,  G10L19/22 ,  G10L2019/0002 ,  G10L2019/0011 ,  G10L2019/0016

Abstract: An apparatus for decoding an audio signal is provided. The apparatus comprises a receiving interface (110), wherein the receiving interface (110) is configured to receive a first frame comprising a first audio signal portion of the audio signal, and wherein the receiving interface (110) is configured to receive a second frame comprising a second audio signal portion of the audio signal. Moreover, the apparatus comprises a noise level tracing unit (130), wherein the noise level tracing unit (130) is configured to determine noise level information depending on at least one of the first audio signal portion and the second audio signal portion, wherein the noise level information is represented in a tracing domain. Furthermore, the apparatus comprises a first reconstruction unit (140) for reconstructing, in a first reconstruction domain, a third audio signal portion of the audio signal depending on the noise level information, if a third frame of the plurality of frames is not received by the receiving interface (110) or if said third frame is received by the receiving interface (110) but is corrupted, wherein the first reconstruction domain is different from or equal to the tracing domain. Moreover, the apparatus comprises a transform unit (121) for transforming the noise level information from the tracing domain to a second reconstruction domain, if a fourth frame of the plurality of frames is not received by the receiving interface (110) or if said fourth frame is received by the receiving interface (110) but is corrupted, wherein the second reconstruction domain is different from the tracing domain, and wherein the second reconstruction domain is different from the first reconstruction domain. Furthermore, the apparatus comprises a second reconstruction unit (141) for reconstructing, in the second reconstruction domain, a fourth audio signal portion of the audio signal depending on the noise level information being represented in the second reconstruction domain, if said fourth frame of the plurality of frames is not received by the receiving interface (110) or if said fourth frame is received by the receiving interface (110) but is corrupted.

Abstract translation: 提供一种用于解码音频信号的装置。 所述装置包括接收接口（110），其中所述接收接口（110）被配置为接收包括所述音频信号的第一音频信号部分的第一帧，并且其中所述接收接口（110）被配置为接收第二帧 包括音频信号的第二音频信号部分。 此外，该装置包括噪声电平跟踪单元（130），其中噪声电平跟踪单元（130）被配置为根据第一音频信号部分和第二音频信号部分中的至少一个来确定噪声电平信息，其中， 噪声电平信息在跟踪域中表示。 此外，该装置包括第一重建单元（140），用于在第一重建域中根据噪声电平信息重建音频信号的第三音频信号部分，如果多个帧中的第三帧未被 接收接口（110）或者如果所述第三帧被接收接口（110）接收但被破坏，其中第一重构域不同于或等于跟踪域。 此外，如果多个帧的第四帧没有被接收接口（110）接收，或者如果所述第四帧被接收，则该装置包括用于将噪声级信息从跟踪域变换到第二重建域的变换单元（121） 帧由接收接口（110）接收但被破坏，其中第二重构域与跟踪域不同，并且其中第二重建域与第一重建域不同。 此外，该装置包括第二重构单元（141），用于在第二重建域中根据在第二重建域中表示的噪声电平信息重建音频信号的第四音频信号部分，如果所述第二重构单元 多个帧不被接收接口（110）接收，或者如果所述第四帧被接收接口（110）接收但被破坏。

公开(公告)号：WO2007106638A2

公开(公告)日：2007-09-20

申请号：PCT/US2007062277

申请日：2007-02-16

Applicant: MOTOROLA INC ,  GIBBS JONATHAN ,  FIKRETLER HALIL

Inventor： GIBBS JONATHAN ,  FIKRETLER HALIL

IPC: G10L19/00 ,  G10L19/08 ,  G10L19/09

CPC classification number: G10L19/09

Abstract: A speech communication unit (100) comprises a speech encoder (134) capable of representing an input speech signal. The speech encoder (134) comprises long-term prediction (LTP) logic having memory (215) operably coupled to quantization logic, wherein the quantization logic is arranged to quantize a memory state of the LTP logic. On the decoder side, a first speech decoder (260) receives the speech encoded bitstream and has a conventional long term predictor (LTP) memory element (265) that is driven by one or more previous codebook decisions made by the speech encoder. A second decoder (275) also receives the speech encoded bitstream and has an LTP memory element (280) that is updated by quantized values of a memory state of an LTP logic of the speech encoder.

Abstract translation: 语音通信单元（100）包括能够表示输入语音信号的语音编码器（134）。 语音编码器（134）包括具有可操作地耦合到量化逻辑的存储器（215）的长期预测（LTP）逻辑，其中量化逻辑被设置为量化LTP逻辑的存储器状态。 在解码器侧，第一语音解码器（260）接收语音编码比特流，并具有由语音编码器做出的一个或多个先前的码本决定驱动的常规长期预测器（LTP）存储元件（265）。 第二解码器（275）还接收语音编码比特流并且具有通过语音编码器的LTP逻辑的存储器状态的量化值更新的LTP存储器元件（280）。

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: G10L19/083 ,  G10L19/09 ,  G10L25/78 ,  G10L19/14

CPC classification number: G10L19/083 ,  G10L19/09 ,  G10L25/78

Abstract: 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

Abstract translation: 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。

公开(公告)号：WO2020005699A1

公开(公告)日：2020-01-02

申请号：PCT/US2019/038185

申请日：2019-06-20

Applicant: CYPRESS SEMICONDUCTOR CORPORATION

Inventor： WADA, Ted ,  PANDEY, Ashutosh

IPC: G10K11/16 ,  G10K11/178 ,  G10L19/06 ,  G10L19/08 ,  G10L19/09 ,  G10L21/0208 ,  G10L21/0216 ,  G10L21/0264 ,  H04M9/08

Abstract: Techniques for acoustic echo cancellation are described herein. In an example embodiment, a system comprises a speaker, a microphone array with multiple microphones, a beamformer (BF) logic and an acoustic echo canceller (AEC) logic. The speaker is configured to receive a reference signal. The BF logic is configured to receive audio signals from the multiple microphones and to generate a beamformed signal. The AEC logic is configured to receive the beamformed signal and the reference signal. The AEC logic is also configured to compute a vector of bias coefficients multiple times per time frame, to compute a background filter coefficient based on the vector of bias coefficients, to apply a background filter to the reference signal and the beamformed signal based on the background filter coefficient, to generate a background cancellation signal, and to generate an output signal based at least on the background cancellation signal.

公开(公告)号：WO2016016053A1

公开(公告)日：2016-02-04

申请号：PCT/EP2015/066677

申请日：2015-07-21

Applicant: FRAUNHOFER-GESELLSCHAFT ZUR FÖRDERUNG DER ANGEWANDTEN FORSCHUNG E.V.

Inventor： RAVELLI, Emmanuel ,  MULTRUS, Markus ,  DÖHLA, Stefan ,  GRILL, Bernhard ,  JANDER, Manuel

IPC: G10L19/22 ,  G10L19/02 ,  G10L19/08 ,  G10L19/09

CPC classification number: G10L19/22 ,  G10L19/0212 ,  G10L19/032 ,  G10L19/08 ,  G10L19/09 ,  G10L19/12 ,  G10L19/265 ,  G10L2019/0002 ,  G10L2019/0011

Abstract: An apparatus for selecting one of a first encoding algorithm having a first characteristic and a second encoding algorithm having a second characteristic for encoding a portion of an audio signal to obtain an encoded version of the portion of the audio signal, comprises a filter configured to receive the audio signal, to reduce the amplitude of harmonics in the audio signal and to output a filtered version of the audio signal. A first estimator is provided for using the filtered version of the audio signal in estimating a SNR or a segmented SNR of the portion of the audio signal as a first quality measure for the portion of the audio signal, which is associated with the first encoding algorithm, without actually encoding and decoding the portion of the audio signal using the first encoding algorithm. A second estimator is provided for estimating a SNR or a segmented SNR as a second quality measure for the portion of the audio signal, which is associated with the second encoding algorithm, without actually encoding and decoding the portion of the audio signal using the second encoding algorithm. The apparatus comprises a controller for selecting the first encoding algorithm or the second encoding algorithm based on a comparison between the first quality measure and the second quality measure.

Abstract translation: 一种用于选择具有第一特性的第一编码算法和具有用于对音频信号的一部分进行编码以获得音频信号的一部分的编码版本的第二特性之一的装置，包括被配置为接收 音频信号，以减小音频信号中的谐波的幅度并输出音频信号的滤波版本。 提供了第一估计器，用于在将音频信号的该部分的SNR或分段的SNR估计为音频信号的与第一编码算法相关联的部分的第一质量度量时，使用滤波版本的音频信号 ，而不使用第一编码算法实际地对音频信号的部分进行编码和解码。 提供第二估计器，用于将SNR或分段SNR估计为与第二编码算法相关联的音频信号部分的第二质量度量，而不使用第二编码实际编码和解码音频信号的部分 算法。 该装置包括：控制器，用于基于第一质量测量和第二质量测量之间的比较来选择第一编码算法或第二编码算法。

公开(公告)号：WO2014077254A1

公开(公告)日：2014-05-22

申请号：PCT/JP2013/080589

申请日：2013-11-12

Applicant: 株式会社NTTドコモ

Inventor： 堤　公孝 ,  菊入　圭 ,  山口　貴史

IPC: G10L19/005 ,  G10L19/00 ,  G10L19/09 ,  H03M7/30

CPC classification number: G10L19/125 ,  G10L19/005 ,  G10L19/09

Abstract: 　音声符号化におけるパケットロスにおいて、アルゴリズム遅延を増加させずに音声品質を回復することを目的とする。音声信号を符号化する音声信号送信装置は、音声信号を符号化する音声符号化部と、先読み信号から補助情報を算出して符号化する補助情報符号化部と、を備える。一方、音声符号を復号して音声信号を出力する音声信号受信装置は、音声パケットの受信状態からパケットロスを検出する音声符号バッファと、音声パケット正常受信時に音声符号を復号する音声パラメータ復号部と、音声パケット正常受信時に補助情報符号を復号する補助情報復号部と、補助情報符号を復号して得られる補助情報を蓄積する補助情報蓄積部と、音声パケットロス検出時に音声パラメータを出力する音声パラメータ紛失処理部と、音声パラメータから復号音声を合成する音声合成部と、を備える。

Abstract translation: 本发明的目的是当音频编码中发生分组丢失时，在不增加算法延迟的情况下恢复音频质量。 编码音频信号的音频信号发送装置包括对音频信号进行编码的音频编码单元，以及从预取计算并编码补充信息的补充信息编码单元。 相反，对音频编码进行解码并输出音频信号的音频信号接收装置包括：音频编码缓冲器，其基于音频分组的接收状态来检测分组丢失; 音频参数解码单元，其在成功接收到音频分组时对音频编码进行解码; 补充信息解码单元，其在成功接收到音频分组之后解码补充信息编码; 补充信息累积单元，其累积通过解码补充信息编码获得的补充信息; 音频参数丢失处理单元，其在检测到音频分组丢失时输出音频参数; 以及音频合成单元，其从音频参数复合解码的音频。

公开(公告)号：WO01015144A1

公开(公告)日：2001-03-01

申请号：PCT/JP2000/005621

申请日：2000-08-23

IPC: G10L19/083 ,  G10L19/09 ,  G10L19/16 ,  G10L19/04

CPC classification number: G10L19/16 ,  G10L19/083 ,  G10L19/09

Abstract: 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).

Abstract translation: 创建存储要量化的向量的代表性样本的向量代码簿（1094）。 每个向量由三个元素组成：AC增益，对应于SC增益的对数的值和SC的预测系数的调整系数。 用于预测编码的系数存储在预测系数存储部分（1095）中。 系数是MA的预测系数，存储了预测顺序的两种系数AC和SC。 参数计算部（1091）从听觉加权输入声音，经听觉加权LPC合成的自适应声源，经受听觉感觉加权LPC合成的概率声源，解码矢量（AC ，SC，调整系数）以及存储在预测系数存储部（1095）中的预测系数（AC，SC）。