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    • 111. 发明授权
    • Method and apparatus for the modeling and synthesis of harmonic distortion
    • 用于建模和合成谐波失真的方法和装置
    • US06504935B1
    • 2003-01-07
    • US09136446
    • 1998-08-19
    • Douglas L. Jackson
    • Douglas L. Jackson
    • H03G300
    • H04B1/0025G10H1/125G10H1/16G10H3/187G10H2250/115G10H2250/191G10H2250/235G10H2250/545H04B1/0003H04B1/28
    • Distortion modeling produces distortion models for use by a distortion synthesizer to synthesize the harmonic distortion effects of audio distortion devices. A sinusoidal waveform is distorted by an audio distortion device and analyzed using a Fourier transform to produce a distortion model comprising harmonic amplitude and phase parameters. A phase correction process compensates for phase shifts induced by the audio distortion device. The distortion synthesizer uses a distortion function that distorts a digital audio signal according to the distortion model. The distortion model can be modified to alter the distortion effect and can be stored in a data-storage device for later retrieval. A frequency bandsplitter and signal mixer allow the distortion effect to be applied only to the low frequency content of the digital audio signal, thus providing spectral headroom to suppress the production of aliasing noise. Aliasing-noise suppression is provided for a full-bandwidth signal by up-converting the sampling rate of the signal before applying the distortion function and down-converting the sampling rate afterwards. A process is provided to remove the direct-current component that may be induced into the signal by the distortion function.
    • 失真建模产生失真模型,由失真合成器使用,以合成音频失真设备的谐波失真效应。 正弦波形由音频失真装置失真,并使用傅立叶变换进行分析,以产生包括谐波幅度和相位参数的失真模型。 相位校正处理补偿由音频失真装置引起的相移。 失真合成器使用根据失真模型扭曲数字音频信号的失真函数。 可以修改失真模型以改变失真效应,并将其存储在数据存储设备中以供以后检索。 频带分配器和信号混频器使得失真效应仅适用于数字音频信号的低频内容,从而提供频谱余量来抑制混叠噪声的产生。 通过在施加失真函数之前对信号的采样率进行上变频,然后对采样率进行下转换,为全带宽信号提供混叠噪声抑制。 提供了通过失真功能去除可能被感应到信号中的直流分量的过程。
    • 113. 发明申请
    • Generation of a note-based code
    • 生成基于笔记的代码
    • US20020035915A1
    • 2002-03-28
    • US09893661
    • 2001-06-29
    • Tero TolonenVille Pulkki
    • G10H001/02
    • G10H1/0025G10G3/04G10H1/36G10H3/125G10H2210/111G10H2210/145G10H2240/056G10H2250/135G10H2250/235
    • A method for generating accompaniment to a musical presentation, the method comprising steps of providing a note-based code representing musical information corresponding to the musical presentation, generating a code sequence corresponding to new melody lines by using said note-based code as an input for a composing method, and providing accompaniment on the basis of the code sequence corresponding to new melody lines. Providing the note-based code representing the musical information comprises steps of receiving the musical information in the form of an audio signal, and applying an audio-to-notes conversion to the audio signal for generating the note-based code representing the musical information, the audio-to-notes conversion comprising the steps of estimating fundamental frequencies of the audio signal for obtaining a sequence of fundamental frequencies, and detecting note events on the basis of the sequence of fundamental frequencies for obtaining the note-based code.
    • 一种用于产生音乐表演伴奏的方法,所述方法包括以下步骤:提供表示对应于音乐演示的音乐信息的基于音符的代码,通过使用所述基于音符的代码作为输入来产生对应于新旋律线的代码序列作为输入 一种组合方法,并且基于与新旋律线对应的代码序列提供伴奏。 提供表示音乐信息的基于音符的代码包括以音频信号的形式接收音乐信息的步骤,以及将音频到音符转换应用于音频信号以产生表示音乐信息的基于音符的代码, 音频到音符转换包括以下步骤:估计用于获得基频序列的音频信号的基本频率,以及基于用于获得基于音符的代码的基本频率序列来检测音符事件。
    • 114. 发明授权
    • Downloading of personalization layers for symbolically compressed objects
    • 下载符号压缩对象的个性化层
    • US6088484A
    • 2000-07-11
    • US745586
    • 1996-11-08
    • Donald C. Mead
    • Donald C. Mead
    • G10H1/00H04N5/14H04N7/26G06K9/36
    • G10H1/0058H04N19/23G10H2240/056G10H2250/221G10H2250/235G10H2250/241H04N5/145
    • A method and apparatus for transferring data signals with personalization layers for symbolically compressed objects comprises a transmitter having an encoder and a receiver having a decoder. The encoder includes a segment selector for identifying signal segments from a group of speech, audio, video, and graphic signals, each segment comprising a representation of a physical waveform. The encoder includes a plurality of encoder libraries, one of the libraries containing the representation of a generic waveform and a symbolic code for the generic object. The decoder has a second plurality of libraries corresponding to the plurality of encoder libraries, and contain the generic waveform, the symbolic code corresponding to the generic physical waveform, and the difference information quantity as a personalization layer of data representing the difference between the generic object code and the waveform from which the generic object was extracted.
    • 用于使用用于符号压缩对象的个性化层传送数据信号的方法和装置包括具有编码器和具有解码器的接收器的发射机。 编码器包括用于从一组语音,音频,视频和图形信号中识别信号段的段选择器,每个段包括物理波形的表示。 编码器包括多个编码器库,其中一个库包含通用波形的表示和通用对象的符号代码。 解码器具有对应于多个编码器库的第二多个库,并且包含通用波形,对应于通用物理波形的符号代码和差异信息量作为表示通用对象之间的差异的数据的个性化层 代码和从中提取通用对象的波形。
    • 115. 发明授权
    • Synthesis of acoustic waveforms based on parametric modeling
    • 基于参数化建模的声波综合
    • US5911170A
    • 1999-06-08
    • US31808
    • 1998-02-27
    • Yinong Ding
    • Yinong Ding
    • G10H1/12G10H7/10G10H1/057
    • G10H7/105G10H1/125G10H2250/081G10H2250/235Y10S84/09
    • A method is disclosed for synthesizing acoustic waveforms, especially musical instrument sounds. The acoustic waveforms are characterized by time-varying amplitudes, frequencies and phases of sinusoidal components. These time-varying parameters, at each analysis frame, are obtained in one embodiment by short term Fourier transforms (STFT). The spectrum envelope at each frame is parameterized with an autoregressive moving average model and applied to a waveform consisting of unit amplitude sinusoids via time-domain filtering. The resulting synthetic waveform preserves the time-varying frequency and phase information and has the same relative energy distribution among different sinusoidal components as that of the original signal. Finally, a general waveform shape for the type of acoustic signal being synthesized is applied. This is particularly useful when musical instrument sounds are being synthesized, where the commonly used four piecewise-linear attack-decay-sustain-release (ADSR) envelope model can be employed.
    • 公开了一种用于合成声波形,特别是乐器声音的方法。 声波的特征在于正弦分量的时变幅度,频率和相位。 在一个实施例中,通过短期傅里叶变换(STFT)在每个分析帧处获得这些时变参数。 每个帧的频谱包络采用自回归移动平均模型进行参数化,并通过时域滤波将其应用于由单位幅度正弦曲线组成的波形。 所得到的合成波形保留时变频率和相位信息,并且在原始信号的不同正弦分量之间具有相同的相对能量分布。 最后,应用正在合成的声信号类型的一般波形形状。 当合成乐器声音时,这是特别有用的,其中可以采用通常使用的四个分段线性攻击 - 衰减 - 持续释放(ADSR)包络模型。
    • 117. 发明授权
    • Signal processing method and sound source data forming apparatus
    • 信号处理方法和声源数据形成装置
    • US5430241A
    • 1995-07-04
    • US438088
    • 1989-11-16
    • Makoto FuruhashiMasakazu SuzuokiKen Kutaragi
    • Makoto FuruhashiMasakazu SuzuokiKen Kutaragi
    • G10H3/12G10H7/08G10K15/02G10L19/02G10H7/00
    • G10K15/02G10H3/125G10H7/08G10L19/02G10H2210/066G10H2250/105G10H2250/235G10H2250/281G10H2250/601Y10S84/09
    • A method for processing a digital signal produced by digitizing an analog signal such as a musical instrument sound signal, and an apparatus for producing sound source data. When the input signal contains a periodically repetitive waveform portion, the fundamental frequency and its high harmonic components of the input signal is extracted by a comb filter prior to signal processing which takes advantage of the periodicity of the input signal. The fundamental frequency or pitch is detected by performing Fourier transform to produce frequency components, phase matching these frequency components and performing inverse Fourier transform. When extracting a repetitive waveform portion or so-called looping domain, such looping domain having the highest similarity in waveform in the vicinity of both ends of the domain is selected. When the bit compression of digital signal data is performed by selecting a filter with blocks each consisting of plural samples as units, a pseudo signal is affixed to the input signal, before the start point of the input signal, which pseudo signal will cause a filter of the lowest order to be selected. The looping domain is set so as to be a whole number multiple of the block which serves as the unit for bit compression, and the parameters of the looping start block are formed on the basis of data of the start and the end blocks. By applying a part or the whole of the signal processing method to a sound source data forming apparatus, sound source data may be formed which is reduced in the looping noise and error caused by data compression and which is of superior sound quality.
    • 一种用于处理通过数字化诸如乐器声音信号的模拟信号产生的数字信号的方法和用于产生声源数据的装置。 当输入信号包含周期性重复的波形部分时,在信号处理之前,由梳状滤波器提取输入信号的基频及其高次谐波分量,这利用了输入信号的周期。 通过执行傅里叶变换以产生频率分量,相位匹配这些频率分量并执行逆傅里叶变换来检测基频或音调。 当提取重复波形部分或所谓的循环域时,选择在该域的两端附近具有最高波形相似度的循环域。 当通过选择具有由多个样本组成的块的滤波器来执行数字信号数据的位压缩时,在输入信号的起始点之前将伪信号附加到输入信号,该伪信号将导致滤波器 的最低订单选择。 循环域被设置为作为用于位压缩的单位的块的整数倍,并且循环开始块的参数基于起始和结束块的数据形成。 通过将一部分或全部的信号处理方法应用于声源数据形成装置,可以形成声源数据,其减少了由数据压缩引起的循环噪声和误差,并且具有优良的音质。
    • 118. 发明授权
    • Electronic musical instrument with memory read sequence control
    • 具有记忆读取序列控制的电子乐器
    • US5298672A
    • 1994-03-29
    • US012978
    • 1993-02-02
    • Rainer Gallitzendorfer
    • Rainer Gallitzendorfer
    • G10H7/02G10H1/06G10H7/12
    • G10H7/02G10H2250/235
    • An electronic musical instrument utilizes a memory unit containing sampled values of waveforms stored in separately addressable memory locations. A first waveform address bus and a second waveform data bus are both connected to the unit. A counter is connected at its output to the first bus and is connected at its input to both a third address bus and a fourth system data bus. Sampled values of waveforms are supplied to the third and fourth buses. Waveform data is read out of the second bus. An arrangement including a central processing unit, a random access memory and a read only memory is connected to the third and fourth buses. This arrangement together with the counter determines, according to the stored sampled values, the sequence in which the individual memory locations and the sound information stored in the locations should be read.
    • 电子乐器利用包含存储在可单独寻址的存储器位置中的波形的采样值的存储器单元。 第一个波形地址总线和第二个波形数据总线均连接到该单元。 计数器在其输出端连接到第一总线,并在其输入端连接到第三地址总线和第四系统数据总线。 波形的采样值被提供给第三和第四总线。 波形数据从第二总线读出。 包括中央处理单元,随机存取存储器和只读存储器的布置被连接到第三和第四总线。 这种配置与计数器一起根据存储的采样值确定应当读取存储在位置中的各个存储位置和声音信息的顺序。
    • 120. 发明授权
    • Harmonic interpolation for producing time variant tones in an electronic
musical instrument
    • 用于在电子乐器中产生时变音调的谐波插值
    • US4677889A
    • 1987-07-07
    • US791631
    • 1985-10-25
    • Ralph Deutsch
    • Ralph Deutsch
    • G10H7/08G10H1/053G10H1/06G10H7/00G10H7/10
    • G10H7/105G10H1/06G10H2250/235
    • A keyboard operated electronic musical instrument is disclosed which has a number of tone generators that are assigned to actuated keyswitches. Musical tones are produced by computing a master data set from an interpolated sequence of harmonic coefficient values. The master data set points are read out sequentially and repetitively from a memory and converted into an audible tone. A plurality of harmonic coefficient memories are used to store preselected sets of harmonic coefficients. In response to a timing clock, the harmonic coefficients from a cyclically chosen pair of harmonic coefficient memories are selected. A tone having a time variant spectra is produced by using a time variant interpolation between the selected pair of harmonic coefficients.
    • 公开了一种键盘操作的电子乐器,其具有分配给致动的钥匙开关的多个音调发生器。 通过从内插的谐波系数值序列计算主数据集来产生乐音。 主数据设定点从存储器顺次读出并转换为可听音。 使用多个谐波系数存储器来存储预选的谐波系数集合。 响应于定时时钟,选择来自循环选择的一对谐波系数存储器的谐波系数。 通过使用所选择的一对谐波系数之间的时间插值来产生具有时变频谱的音调。