公开(公告)号：US11996816B2

公开(公告)日：2024-05-28

申请号：US17006036

申请日：2020-08-28

申请人： Peter Graham Craven ,  John Robert Stuart

发明人： Peter Graham Craven ,  John Robert Stuart

IPC分类号： H03H17/02 ,  G11B20/10 ,  H03H17/00 ,  H03H17/04 ,  H03H17/06

CPC分类号： H03H17/0261 ,  G11B20/10 ,  H03H17/04 ,  G10H2250/071 ,  G10H2250/121 ,  G11B2020/10546 ,  H03H2017/009 ,  H03H2017/0472 ,  H03H17/0664

摘要： Methods and devices are described for reducing the audible effect of pre-responses in an audio signal. The pre-responses are effectively delayed by employing a digital non-minimum-phase filter, which includes a zero lying outside the unit circle in its z-transform response. This zero is not paired with another zero at a reciprocal position inside the unit circle, as this would linearise the phase modification. The filtering can introduce a greater group delay at the pre-response frequency than at a low frequency, such as 500 Hz or even 0 Hz. The technique can be used to reduce pre-responses in an existing audio signal and also to pre-empt pre-responses that would be introduced to the audio signal by subsequent processing.

公开(公告)号：US6111181A

公开(公告)日：2000-08-29

申请号：US72400

申请日：1998-05-04

申请人： Michael W. Macon ,  Wai-Ming Lai ,  Alan V. McCree ,  Vishu R. Viswanathan

发明人： Michael W. Macon ,  Wai-Ming Lai ,  Alan V. McCree ,  Vishu R. Viswanathan

IPC分类号： G10H1/12 ,  G10H7/00

CPC分类号： G10H1/125 ,  G10H7/00 ,  G10H2230/255 ,  G10H2250/065 ,  G10H2250/071 ,  G10H2250/075 ,  G10H2250/235 ,  G10H2250/601

摘要： A synthesis of percussion musical instruments sounds is provided using a microprocessor (17) that implements an all pole lattice filter and applying either a single impulse signal to the filter or N samples of an excitation signal sequence to the filter by a memory (19). The coefficients of the filter are determined by storing digital samples (501) of desired musical note from a desired percussion instrument, generating a Fourier transform to get a spectrum (502), picking the peaks of the spectrum (503) to select the most prominent components in the spectrum and determining wanted frequencies for decaying sine waves and for the frequencies finding the time envelope and estimating therefrom the pole radius.

摘要翻译： 使用实现全极点格子的微处理器（17）提供打击乐器声音的合成，并且通过存储器（19）向滤波器施加单个脉冲信号或向滤波器施加激励信号序列的N个样本。 通过从期望的打击乐器存储所需音符的数字样本（501）来确定滤波器的系数，产生傅立叶变换以获得频谱（502），拾取频谱的峰值（503）以选择最突出的 确定频谱中的分量，并确定衰减正弦波的有用频率以及找到时间包络的频率，并从其估计极半径。

公开(公告)号：US5248845A

公开(公告)日：1993-09-28

申请号：US854554

申请日：1992-03-20

申请人： Dana C. Massie ,  David P. Rossum

发明人： Dana C. Massie ,  David P. Rossum

IPC分类号： G10H1/12 ,  G10H5/00

CPC分类号： G10H1/125 ,  G10H5/007 ,  G10H2210/066 ,  G10H2230/181 ,  G10H2230/241 ,  G10H2250/071 ,  G10H2250/075 ,  G10H2250/081 ,  G10H2250/125 ,  G10H2250/235 ,  G10H2250/251 ,  G10H2250/255 ,  G10H2250/445 ,  G10H2250/451 ,  G10H2250/491 ,  G10H2250/581 ,  Y10S84/09 ,  Y10S84/10

摘要： An electronic music system which imitates acoustic instruments addresses the problem wherein the audio spectrum of a a recorded note is entirely shifted in pitch by transposition. The consequence of this is that unnatural formant shifts occur, resulting in the phenomenon known in the industry as "munchkinization." The present invention eliminates munchkinization, thus allowing a substantially wider transposition range for a single recording. Also, the present invention allows even shorter recordings to be used for still further memory improvements. An analysis stage separates and stores the formant and excitation components of sounds from an instrument. On playback, either the formant component or the excitation component may be manipulated.

摘要翻译： 模仿声学仪器的电子音乐系统解决了一个问题，其中所记录的音符的音频频谱通过转置完全偏移。 这样做的结果是发生了不自然的共振变化，导致了业界已知的“混沌”现象。 本发明消除了复合，从而允许单个记录的基本更宽的转置范围。 此外，本发明允许将更短的记录用于进一步的存储器改进。 分析阶段分离和存储来自仪器的声音的共振峰和激发分量。 在播放时，可以操纵共振峰分量或激励分量。

公开(公告)号：US5872727A

公开(公告)日：1999-02-16

申请号：US752014

申请日：1996-11-19

申请人： Chih-Chung Kuo

发明人： Chih-Chung Kuo

IPC分类号： G10H1/12 ,  G10H1/20 ,  G06F17/10 ,  G10H1/06 ,  G10H7/00

CPC分类号： G10H1/20 ,  G10H1/125 ,  G10H2250/071 ,  G10H2250/075 ,  G10H2250/255 ,  G10H2250/601 ,  G10H2250/621 ,  G10H2250/631

摘要： An improved method for shifting the pitches of a tone is disclosed. It comprises: (a) subjecting a digitized original waveform to a whitening process using an all-zero filter (AZF) to obtain a whitened waveform; (b) resampling the whitened waveform at a desired scaling ratio to obtain a scaled and whitened waveform; (c) subjecting the scaled and whitened waveform to a coloring process using an all-pole filter (APF) to obtain a synthesized waveform. In a preferred embodiment, the all-zero filter performs the transformation function of: ##EQU1## and the all-pole filter performs the transformation function of: ##EQU2## wherein the a.sub.i 's and b.sub.i 's are linear predictive coefficients. The whitened waveforms can be compressed and stored as wavetables, which can be subsequently retrieved and decompressed before resampling.

摘要翻译： 公开了一种改变音调音高的方法。 它包括：（a）使用全零滤波器（AZF）对数字化的原始波形进行白化处理以获得白化波形; （b）以期望的比例比对白化的波形进行重采样以获得标度和白化的波形; （c）使用全极滤波器（APF）对经缩放和白化的波形进行着色处理，以获得合成波形。 在一个优选实施例中，全零滤波器执行以下变换函数：＆lt; IMAGE＆gt;全极滤波器执行以下变换函数：＆lt; IMAGE＆gt;其中，ai和bi是线性预测系数。 白化的波形可以压缩并存储为波形图，可以在重新采样之前随后检索和解压缩。

公开(公告)号：US5698807A

公开(公告)日：1997-12-16

申请号：US611014

申请日：1996-03-05

申请人： Dana C. Massie ,  David P. Rossum

发明人： Dana C. Massie ,  David P. Rossum

IPC分类号： G10H1/12 ,  G10H5/00 ,  G01H1/12 ,  G10L9/04 ,  G10L9/14

CPC分类号： G10H1/125 ,  G10H5/007 ,  G10H2210/066 ,  G10H2230/181 ,  G10H2230/241 ,  G10H2250/071 ,  G10H2250/075 ,  G10H2250/081 ,  G10H2250/125 ,  G10H2250/235 ,  G10H2250/251 ,  G10H2250/255 ,  G10H2250/445 ,  G10H2250/451 ,  G10H2250/491 ,  G10H2250/581 ,  Y10S84/09 ,  Y10S84/10

摘要： An electronic music system which imitates acoustic instruments addresses the problem wherein the audio spectrum of a a recorded note is entirely shifted in pitch by transposition. The consequence of this is that unnatural formant shifts occur, resulting in the phenomenon known in the industry as "munchkinization." The present invention eliminates munchkinization, thus allowing a substantially wider transposition range for a single recording. Also, the present invention allows even shorter recordings to be used for still further memory improvements. An analysis stage separates and stores the formant and excitation components of sounds from an instrument. On playback, either the formant component or the excitation component may be manipulated.

摘要翻译： 模拟声学仪器的电子音乐系统解决了一个问题，其中所记录的音符的音频频谱通过转置完全偏移。 这样做的结果是发生了不自然的共振变化，导致了业界已知的“混沌”现象。 本发明消除了复合，从而允许单个记录的基本更宽的转置范围。 此外，本发明允许将更短的记录用于进一步的存储器改进。 分析阶段分离和存储来自仪器的声音的共振峰和激发分量。 在播放时，可以操纵共振峰分量或激励分量。

公开(公告)号：US20180293969A1

公开(公告)日：2018-10-11

申请号：US15762841

申请日：2017-03-16

申请人： TENCENT TECHNOLOGY (SHENZHEN) COMPANY LIMITED

发明人： Weifeng ZHAO

IPC分类号： G10H1/36 ,  G10L25/18 ,  G10L25/12 ,  G10L25/21 ,  G10L25/30

CPC分类号： G10H1/361 ,  G10H1/125 ,  G10H1/36 ,  G10H2210/005 ,  G10H2210/041 ,  G10H2210/056 ,  G10H2230/025 ,  G10H2250/071 ,  G10H2250/275 ,  G10H2250/311 ,  G10L25/12 ,  G10L25/18 ,  G10L25/21 ,  G10L25/30

摘要： An audio information processing method and apparatus are provided. The method includes decoding a first audio file to acquire a first audio subfile corresponding to a first sound channel and a second audio subfile corresponding to a second sound channel; extracting first audio data from the first audio subfile; extracting second audio data from the second audio subfile; acquiring a first audio energy value of the first audio data; acquiring a second audio energy value of the second audio data; and determining an attribute of at least one of the first sound channel and the second sound channel based on the first audio energy value and the second audio energy value.

公开(公告)号：US5744742A

公开(公告)日：1998-04-28

申请号：US808676

申请日：1997-02-28

申请人： Eric Lindemann ,  Jeffrey Barish

发明人： Eric Lindemann ,  Jeffrey Barish

IPC分类号： G10H5/00 ,  G10H7/10 ,  G10H1/00 ,  G10H1/12

CPC分类号： G10H5/007 ,  G10H7/10 ,  G10H2250/015 ,  G10H2250/071 ,  G10H2250/075 ,  G10H2250/081 ,  G10H2250/095 ,  G10H2250/101 ,  G10H2250/235 ,  G10H2250/285 ,  G10H2250/481 ,  G10H2250/545 ,  G10H2250/611 ,  Y10S84/09

摘要： A parametric signal modeling musical tone synthesizer utilizes a multidimensional filter coefficient space consisting of many sets of filter coefficients to model an instrument. These sets are smoothly interpolated over pitch, intensity, and time. The filter excitation for a particular note is derived from a collection of single period excitations, which form a multidimensional excitation space, which is also smoothly interpolated over pitch, intensity and time. The synthesizer includes effective modeling of attacks of tones, and the noise component of a tone is modelled separately from the pitched component. The input control signals may include initial pitch and intensity, or the intensity may be time-varying. A variety of instruments may be specified.

摘要翻译： 参数信号建模乐音合成器利用由许多滤波器系数组成的多维滤波器系数空间来对仪器建模。 这些组在音高，强度和时间上顺利插值。 用于特定音符的滤波器激励源自单周期激励的集合，其形成多维激励空间，其也在音调，强度和时间上平滑地内插。 合成器包括对音调的有效建模，并且音调的噪声分量与俯仰分量分开建模。 输入控制信号可以包括初始音高和强度，或者强度可以是时变的。 可以指定各种仪器。