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51. 发明授权

US08996388B2 Method and an apparatus for processing an audio signal 有权
标题翻译：用于处理音频信号的方法和装置
公开(公告)号：US08996388B2
公开(公告)日：2015-03-31
申请号：US13960540
申请日：2013-08-06
申请人： Industry-Academic Cooperation Foundation, Yonsei University
发明人： Hyen-O Oh , Chang Heon Lee , Hong Goo Kang , Jung Wook Song
IPC分类号： G10L19/04 , G10L19/00 , G10L19/18 , G10L19/022 , G10L19/16 , G11B20/00 , G11B20/10
CPC分类号： G10L19/00 , G10L19/022 , G10L19/04 , G10L19/167 , G10L19/18 , G11B20/00007 , G11B2020/00028 , G11B2020/10546
摘要： 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.
摘要翻译：公开了一种用于处理音频信号的装置及其方法。本发明包括由音频处理设备接收包括用矩形编码方案编码的第一块的第一数据和用非矩形编码方案编码的第二块的第二数据的音频信号; 接收对应于第二块的补偿信号; 使用所述第一数据估计混叠部分的预测; 并且基于第二数据，补偿信号和混叠部分的预测来获得第二块的重构信号。

52. 发明授权

US08892427B2 Method and an apparatus for processing an audio signal 有权
标题翻译：用于处理音频信号的方法和装置
公开(公告)号：US08892427B2
公开(公告)日：2014-11-18
申请号：US13387248
申请日：2010-07-27
申请人： Hyen-O Oh , Hong Goo Kang , Chang Heon Lee , Jeong Ook Song
发明人： Hyen-O Oh , Hong Goo Kang , Chang Heon Lee , Jeong Ook Song
IPC分类号： G10L21/00 , G10L19/00 , G10L25/45 , G10L19/04 , G10L19/18 , G10L19/022 , G10L19/005
CPC分类号： G10L19/04 , G10L19/005 , G10L19/022 , G10L19/18 , G10L25/45
摘要： 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.
摘要翻译：公开了一种处理音频信号的装置及其方法。本发明包括由音频处理设备接收包括用矩形编码方案编码的第一块的第一数据和用非矩形编码方案编码的第二块的第二数据的音频信号; 接收对应于第二块的补偿信号; 使用所述第一数据估计混叠部分的预测; 并且基于第二数据，补偿信号和混叠部分的预测来获得第二块的重构信号。

53. 发明授权

US08831961B2 Preprocessing method, preprocessing apparatus and coding device 有权
标题翻译：预处理方法，预处理装置和编码装置
公开(公告)号：US08831961B2
公开(公告)日：2014-09-09
申请号：US13914206
申请日：2013-06-10
申请人： Huawei Technologies Co., Ltd.
发明人： Lei Miao , Fengyan Qi , Jianfeng Xu , Dejun Zhang , Qing Zhang
IPC分类号： G10L19/00 , G10L19/18 , G10L21/06 , G10L19/06 , G10L19/09
CPC分类号： G10L21/06 , G10L19/06 , G10L19/09 , G10L19/18
摘要： The present disclosure relates to coding and decoding technologies, and discloses a preprocessing method, a preprocessing apparatus, and a coding device. The preprocessing method includes: obtaining characteristic information of a current frame signal; identifying whether the current frame signal requires no coding operation of removing LTC according to the characteristic information of the current frame signal and preset information; and if identifying that the current frame signal requires no coding operation of removing LTC, performing the coding operation of removing STC for the current frame signal; and if identifying that the current frame signal requires the coding operation of removing LTC, performing the coding operations of removing both LTC and STC for the current frame signal. Through the technical solution provided herein, the coding operation of removing LTC is performed for only part of the input frame signals.
摘要翻译：本公开涉及编码和解码技术，并且公开了一种预处理方法，预处理装置和编码装置。预处理方法包括：获得当前帧信号的特征信息; 识别当前帧信号是否不需要根据当前帧信号和预设信息的特征信息去除LTC的编码操作; 并且如果识别出当前帧信号不需要去除LTC的编码操作，则执行去除当前帧信号的STC的编码操作; 并且如果识别当前帧信号需要去除LTC的编码操作，则执行去除当前帧信号的LTC和STC的编码操作。通过本文提供的技术方案，仅对部分输入帧信号执行去除LTC的编码操作。

54. 发明申请

US20050240415A1 Method and apparatus for encoding signals 有权
标题翻译：用于编码信号的方法和装置
公开(公告)号：US20050240415A1
公开(公告)日：2005-10-27
申请号：US11143011
申请日：2005-06-02
申请人： Detlef Wiese , Joerg Rimkus
发明人： Detlef Wiese , Joerg Rimkus
IPC分类号： G10L19/18 , H04B1/66 , H03D1/00
CPC分类号： G10L19/18 , H04B1/665
摘要： The invention concerns a method of encoding signals, in particular digitised audio signals, with an encoding device for encoding the signal in an encoding format and a processing device for processing the encoded signal. Methods of that kind are known for example from European patent specification No 290 581. In that case, in the bit rate-reducing encoding of audio signals which are already present in digitized form, for example 48 kHz sampling frequency/16-bit resolution, psycho-acoustic phenomena of the perception of audio signals are used in such a way that the original bit rate of the audio signals is considerably reduced. Such methods are also familiar and standardised under the heading of ‘source encoding’ (ISO 11172 and 11318). The object of the invention is to provide a method of the kind set forth in the opening part of this specification, which resolves the above-indicated problems and in which re-coding operations, once encoding has been effected, are very substantially avoided.
摘要翻译：本发明涉及一种使用编码装置编码信号，特别是数字化音频信号的方法，用于以编码格式对信号进行编码，以及用于处理编码信号的处理装置。这种方法例如从欧洲专利说明书No.250581中已知。在这种情况下，在以数字化形式存在的音频信号的比特率降低编码中，例如48kHz采样频率/ 16位分辨率，使用音频信号感知的心理声学现象，使得音频信号的原始比特率显着降低。这种方法在“源编码”（ISO 11172和11318）的标题下也是熟悉和标准化的。本发明的目的是提供一种在本说明书的开头部分中阐述的那种解决上述问题的方法，并且其中一旦编码已被实现，重新编码操作被非常基本地避免。

55. 发明申请

US20050219073A1 Method and device for code conversion between audio encoding/decoding methods and storage medium thereof 失效
标题翻译：音频编码/解码方法及其存储介质之间的代码转换方法和装置
公开(公告)号：US20050219073A1
公开(公告)日：2005-10-06
申请号：US10515168
申请日：2003-05-22
申请人： Atsushi Murashima
发明人： Atsushi Murashima
IPC分类号： G10L19/12 , G10L19/18 , H03M7/30 , H03M7/36 , H03M7/34
CPC分类号： G10L19/173
摘要： When performing audio communication by using different encoding/decoding methods, a code obtained by encoding audio by a certain method is converted into a code decodable by another method with a high audio quality and a small calculation amount. In a code conversion device for converting a first code string into a second code string, an audio decoding circuit acquires a first linear prediction coefficient and excitation signal information from the first code string and drives the filter having the first linear prediction coefficient by the excitation signal obtained from the excitation signal information, thereby creating a first audio signal. A fixed codebook code generation circuit uses the fixed codebook information and minimizes the distance between the second audio signal generated from the information obtained from the second code string and the first audio signal, thereby obtaining the fixed codebook information in the second code string.
摘要翻译：当通过使用不同的编码/解码方法执行音频通信时，通过某种方法对音频进行编码而获得的代码被转换成具有高音频质量和小计算量的另一方法可解码的代码。在用于将第一代码串转换为第二代码串的代码转换装置中，音频解码电路从第一代码串获取第一线性预测系数和激励信号信息，并通过激励信号驱动具有第一线性预测系数的滤波器从激励信号信息获得，从而产生第一音频信号。固定码本代码生成电路使用固定码本信息，并且使从第二代码串获得的信息产生的第二音频信号与第一音频信号之间的距离最小化，从而获得第二代码串中的固定码本信息。

56. 发明授权

US6134518A Digital audio signal coding using a CELP coder and a transform coder 失效
标题翻译：使用CELP编码器和变换编码器的数字音频信号编码
公开(公告)号：US6134518A
公开(公告)日：2000-10-17
申请号：US34931
申请日：1998-03-04
申请人： Gilad Cohen , Yossef Cohen , Doron Hoffman , Hagai Krupnik , Aharon Satt
发明人： Gilad Cohen , Yossef Cohen , Doron Hoffman , Hagai Krupnik , Aharon Satt
IPC分类号： G10L19/02 , G10L19/04 , G10L19/18 , G10L11/02
CPC分类号： G10L19/18 , G10L19/0212 , G10L19/04
摘要： Apparatus is described for digitally encoding an input audio signal for storage or transmission. A distinguishing parameter is measure from the input signal. It is determined from the measured distinguishing parameter whether the input signal contains an audio signal of a first type or a second type. First and second coders are provided for digitally encoding the input signal using first and second coding methods respectively and a switching arrangement directs, at any particular time, the generation of an output signal by encoding the input signal using either the first or second coders according to whether the input signal contains an audio signal of the first type or the second type at that time. A method for adaptively switching between transform audio coder and CELP coder, is presented. In a preferred embodiment, the method makes use of the superior performance of CELP coders for speech signal coding, while enjoying the benefits of transform coder for other audio signals. The combined coder is designed to handle both speech and music and achieve an improved quality.
摘要翻译：描述了用于对用于存储或传输的输入音频信号进行数字编码的装置。一个区别的参数是从输入信号测量。根据测量的区分参数确定输入信号是否包含第一类型或第二类型的音频信号。提供了第一和第二编码器，用于分别使用第一和第二编码方法对输入信号进行数字编码，并且切换装置在任何特定时间通过使用根据第一或第二编码器对输入信号进行编码来引导输出信号的产生此时输入信号是否包含第一类型或第二类型的音频信号。提出了一种在转换音频编码器和CELP编码器之间进行自适应切换的方法。在优选实施例中，该方法利用CELP编码器用于语音信号编码的优越性能，同时享受用于其他音频信号的变换编码器的优点。组合编码器旨在处理语音和音乐，并实现改进的质量。

57. 发明授权

US6058359A Speech coding including soft adaptability feature 失效
标题翻译：语音编码包括软适应性特征
公开(公告)号：US6058359A
公开(公告)日：2000-05-02
申请号：US34590
申请日：1998-03-04
申请人： Roar Hagen , Erik Ekudden
发明人： Roar Hagen , Erik Ekudden
IPC分类号： G10L19/002 , G10L19/12 , G10L19/18 , G10L9/00
CPC分类号： G10L19/12 , G10L19/18 , G10L19/002
摘要： Adaptive speech coding includes receiving an original speech signal, performing on the original speech signal a current coding operation, and adapting the current coding operation in response to information used in the current coding operation. Adaptive speech decoding includes receiving coded information, performing a current decoding operation on the coded information, and adapting the current decoding operation in response to information used in the current decoding operation.
摘要翻译：自适应语音编码包括接收原始语音信号，对原始语音信号执行当前编码操作，并根据当前编码操作中使用的信息调整当前编码操作。自适应语音解码包括接收编码信息，对编码信息执行当前解码操作，以及响应当前解码操作中使用的信息调整当前解码操作。

58. 发明授权

US5781881A Variable-subframe-length speech-coding classes derived from wavelet-transform parameters 失效
标题翻译：从小波变换参数导出的可变子帧长度语音编码类
公开(公告)号：US5781881A
公开(公告)日：1998-07-14
申请号：US734657
申请日：1996-10-21
申请人： Joachim Stegmann
发明人： Joachim Stegmann
IPC分类号： G10L19/18 , G10L25/27 , G10L25/78 , G10L25/93 , G10L7/02 , H03M7/30
CPC分类号： G10L19/18 , G10L2025/786 , G10L25/27 , G10L25/93
摘要： A method and a device are described for classifying speech on the basis of the wavelet transformation for low-bit-rate speech coding processes. The method and the device permit a more robust classifier of speech signals for signal-matched control of speech coding processes in order to reduce the bit rate without affecting the speech quality or to increase the quality at the same bit rate. The method provides that, after segmenting the speech signal, a wavelet transformation is calculated for each frame, from which a set of parameters is determined with the help of adaptive thresholds. The parameters control a finite-state model, which subdivides the frames into shorter subframes if required, and classifies each subframe into one of several classes typical for speech coding. The speech signal is classified on the basis of the wavelet transformation for each time frame. Thus both a high time resolution (location of pulses) and frequency resolution (good mean values) can be achieved. This method and the classifier are therefore especially well suited for the control and selection of code books in a low-bit-rate speech coder. They also have a low sensitivity to background noise and low complexity.
摘要翻译：描述了一种基于用于低比特率语音编码处理的小波变换来分类语音的方法和装置。所述方法和装置允许更强大的语音信号分类器用于语音编码处理的信号匹配控制，以便在不影响语音质量的情况下降低比特率或者以相同比特率提高质量。该方法提供了在对语音信号进行分段之后，针对每个帧计算小波变换，借助自适应阈值确定一组参数。参数控制有限状态模型，如果需要，将帧细分为较短的子帧，并将每个子帧分类为语言编码典型的几个类别之一。基于每个时间帧的小波变换对语音信号进行分类。因此，可以实现高时间分辨率（脉冲位置）和频率分辨率（良好的平均值）。因此，该方法和分类器特别适用于低比特率语音编码器中的代码簿的控制和选择。它们对背景噪声的敏感性低，复杂度低。

59. 发明授权

US5765127A High efficiency encoding method 失效
标题翻译：高效编码方法
公开(公告)号：US5765127A
公开(公告)日：1998-06-09
申请号：US150082
申请日：1993-12-06
申请人： Masayuki Nishiguchi , Jun Matsumoto , Shinobu Ono
发明人： Masayuki Nishiguchi , Jun Matsumoto , Shinobu Ono
IPC分类号： G10L19/02 , G10L19/038 , G10L19/04 , G10L19/10 , G10L19/12 , G10L19/18 , G10L25/27 , G10L25/90 , G10L25/93 , G10L3/00
CPC分类号： G10L25/90 , G10L19/0212 , G10L19/038 , G10L19/12 , G10L19/18 , G10L25/93 , G10L19/04 , G10L19/10 , G10L2025/937 , G10L25/27
摘要： A high efficiency encoding method for encoding data on frequency axis obtained by dividing an input audio signal on block-by-block basis and converting the signal onto the frequency axis, wherein V bands are searched for a band B.sub.VH with the highest center frequency if it is decided that there are one or more shift points of voiced (V)/unvoiced (UV) decision data of all bands on the frequency axis, and wherein the number of V bands N.sub.V up to the band B.sub.VH is found, so as to decide whether proportion of the V bands is equal to or higher than a predetermined threshold N.sub.th, thereby deciding one V/UV boundary point. Thus, it is possible to replace the V/UV decision data for each band by information on one demarcation in all bands, thereby to reduce data volume and to reduce bit rate. Also, by using two-stage hierarchical vector quantization in quantizing the data on the frequency axis, operation volume for codebook search and memory capacity of the codebook are reduced.
摘要翻译： PCT No.PCT / JP93 / 00323 Sec。 371日期：1993年12月6日 102（e）日期1993年12月6日PCT提交1993年2月18日PCT公布。第WO93 / 19459号公报日期1993年9月30日一种用于编码通过逐块分割输入音频信号而获得的频率轴上的数据的高效编码方法，并将信号转换到频率轴上，其中V带被搜索带BVH，如果确定频率轴上所有频带的有声（V）/清音（UV）判定数据都有一个或多个位移点，则发现最高中心频率，并且其中找到频带BV的数量，直到频带BVH 以确定V波段的比例是否等于或高于预定阈值Nth，从而确定一个V / UV边界点。因此，可以通过关于所有频带中的一个分界的信息来替换每个频带的V / UV判定数据，从而减少数据量并降低比特率。此外，通过在量化频率轴上的数据时使用两级分层矢量量化，减少码本搜索的操作量和码本的存储容量。

60. 发明公开

US20240144940A1 MDCT-BASED COMPLEX PREDICTION STEREO CODING 审中-公开
公开(公告)号：US20240144940A1
公开(公告)日：2024-05-02
申请号：US18502973
申请日：2023-11-06
申请人： DOLBY INTERNATIONAL AB
发明人： Heiko Purnhagen , Pontus Carlsson , Lars Villemoes
IPC分类号： G10L19/008 , G10L19/02 , G10L19/06 , G10L19/16 , G10L19/18 , H04S3/00
CPC分类号： G10L19/008 , G10L19/0212 , G10L19/06 , G10L19/167 , G10L19/18 , H04S3/008 , G10L25/12
摘要： 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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