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公开(公告)号：US20230328434A1
公开(公告)日：2023-10-12
申请号：US18043133
申请日：2020-08-27
发明人： Jianwen ZHENG , Shao-Fu SHIH
CPC分类号： H04R3/02 , G10H1/366 , H04R1/406 , H04R3/005 , H04R3/04 , G10H2250/121 , G10H2210/005 , H04R2201/401
摘要： A low complexity howling suppression system and method for portable karaoke system are provided. In the howling suppression, at least one infinite impulse response (IIR) filters are introduced for estimating the acoustic feedback picked up by the microphone from the real environment, and thereby to cancel out the acoustic feedback from the microphone input signal.
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公开(公告)号：US07405499B2
公开(公告)日：2008-07-29
申请号：US10516819
申请日：2003-06-27
IPC分类号： H03K3/00
CPC分类号： G10H7/12 , G10H7/10 , G10H2250/121 , G10H2250/225 , H03B28/00
摘要： Amplitude, phase and frequency of a sine wave to be generated are calculated on the basis of feature quantity s1 delivered to feature quantity detecting means (2), and are sent to initialization means (3). The initialization means (3) calculates first two points of the sine wave to send the points thus calculated to oscillator (sine wave generating means) (4) as initial value s4. The oscillator (4) sequentially calculates values of respective sample points of waveform by using recurrence formula in accordance with initial value or values instructed from the initialization means (3) to thereby generate a sine wave signal. Thus, sine wave generation is performed without performing modulo-addressing.
摘要翻译： 基于递送到特征量检测装置（2）的特征量s 1计算要产生的正弦波的幅度，相位和频率，并将其发送到初始化装置（3）。 初始化装置（3）计算正弦波的前两点，将由此计算的点发送到振荡器（正弦波产生装置）（4）作为初始值s 4。 振荡器（4）通过使用根据从初始化装置（3）指示的初始值或值产生正弦波信号的递归公式，顺序地计算各波形采样点的值。 因此，在不执行模寻址的情况下执行正弦波生成。
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公开(公告)号：US20030233930A1
公开(公告)日：2003-12-25
申请号：US10602845
申请日：2003-06-24
发明人： Daniel Ozick
IPC分类号： G10H001/36
CPC分类号： G10H1/361 , G10H1/366 , G10H2210/066 , G10H2240/141 , G10H2250/091 , G10H2250/121
摘要： A song-matching system, which provides real-time, dynamic recognition of a song being sung and providing an audio accompaniment signal in synchronism therewith, includes a song database having a repertoire of songs, each song of the database being stored as a relative pitch template, an audio processing module operative in response to the song being sung to convert the song being sung into a digital signal, an analyzing module operative in response to the digital signal to determine a definition pattern representing a sequence of pitch intervals of the song being sung that have been captured by the audio processing module, a matching module operative to compare the definition pattern of the song being sung with the relative pitch template of each song stored in the song database to recognize one song in the song database as the song being sung, the matching module being further operative to cause the song database to download the unmatched portion of the relative pitch template of the recognized song as a digital accompaniment signal; and a synthesizer module operative to convert the digital accompaniment signal to the audio accompaniment signal that is transmitted in synchronism with the song being sung.
摘要翻译： 歌曲匹配系统提供对正在唱歌的歌曲的实时动态识别并与其同步地提供音频伴奏信号，包括具有歌曲曲目的歌曲数据库，数据库的每首歌曲被存储为相对音高 模板，音频处理模块，响应于所演唱的歌曲而被演奏，以将被演唱的歌曲转换为数字信号;响应于数字信号操作的分析模块，以确定表示歌曲的音调间隔的序列的定义模式 已经被音频处理模块捕获的唱片，匹配模块，用于将所播放的歌曲的定义模式与存储在歌曲数据库中的每首歌曲的相对音调模板进行比较，以在歌曲数据库中识别歌曲中的一首歌曲 所述匹配模块进一步操作以使歌曲数据库下载该相对音调模板的不匹配部分 识别歌曲作为数字伴奏信号; 以及合成器模块，用于将数字伴奏信号转换为与所唱歌曲同步地发送的音频伴奏信号。
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公开(公告)号：US5877971A
公开(公告)日：1999-03-02
申请号：US947289
申请日：1997-10-07
CPC分类号： G10H1/125 , H03H17/0294 , H03H17/04 , G10H2250/101 , G10H2250/121 , G10H2250/545
摘要： Digital signal processing (e.g. filtering) apparatus comprises pipelined digital signal processing means for generating each output sample of an output digital signal from input samples of an input digital signal by combining the input samples with a plurality of subsets of a set of filter coefficients associated with that output sample during a respective plurality of sample periods of the input digital signal; and coefficient generating means for supplying sets of filter coefficients to the signal processing means in response to the current state of a filter control signal. The coefficient generating means is operable to supply the plurality of subsets of each set of filter coefficients to the signal processing means during the respective plurality of sample periods of the input digital signal.
摘要翻译： 数字信号处理（例如滤波）装置包括流水线数字信号处理装置，用于通过将输入样本与一组与滤波系数相关联的多个子集组合而从输入数字信号的输入样本产生输出数字信号的每个输出样本 在输入数字信号的相应多个采样周期期间该输出采样; 以及系数产生装置，用于响应于滤波器控制信号的当前状态向信号处理装置提供滤波器系数集合。 系数产生装置可操作以在输入数字信号的相应多个采样周期期间将每组滤波器系数的多个子集提供给信号处理装置。
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公开(公告)号：US20100077910A1
公开(公告)日：2010-04-01
申请号：US12571181
申请日：2009-09-30
申请人： Yoji KANEKO
发明人： Yoji KANEKO
IPC分类号： G10H1/12
CPC分类号： G10H1/125 , G10H7/02 , G10H2210/221 , G10H2250/095 , G10H2250/121 , G10H2250/615
摘要： In a filter device, a filter coefficient calculation circuit has a parameter table. The parameter table stores a plurality of sets of filter coefficients associated with a first parameter based on a frequency and a second parameter based on respective plurality of levels representing a degree of attenuation or enhancement of a gain of a filter in filter characteristics. The filter coefficient calculation circuit extracts a set of filter coefficients from a parameter table with the use of the first parameter and the second parameter determined according to a frequency and a strength of a musical sound signal, and outputs the extracted set of filter coefficients to the filter. The filter circuit performs filter processing for the musical sound signal, based on the filter characteristics determined by the set of filter coefficients.
摘要翻译： 在滤波器装置中，滤波器系数计算电路具有参数表。 基于表示滤波器特性中的滤波器的增益的衰减程度或增强度的相应多个级别，基于频率和第二参数，参数表存储与第一参数相关联的多个滤波器系数集合。 滤波器系数计算电路使用根据音乐声音信号的频率和强度确定的第一参数和第二参数从参数表提取一组滤波器系数，并将所提取的滤波器系数集合输出到 过滤。 滤波器电路基于由滤波器系数集合确定的滤波器特性来对音乐声音信号进行滤波处理。
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公开(公告)号：US20040088169A1
公开(公告)日：2004-05-06
申请号：US10284854
申请日：2002-10-30
IPC分类号： G10L019/00
CPC分类号： G10H7/004 , G10H1/0091 , G10H2210/026 , G10H2210/281 , G10H2210/301 , G10H2240/275 , G10H2240/285 , G10H2240/305 , G10H2250/121
摘要： Efficient recursive audio processing of one or more input data streams using a multistage processor for performing one or more predetermined functions and programmable audio effects. A first stage performs a first predetermined function, such as frequency shifting function. Intermediate results are preferably mixed. The second stage applies programmable audio effects to the mixed data, such as a reverberation effect, and stores the second stage output in a destination mix bin. The second stage output is preferably transferred to a main memory accessible to a primary processor. The second stage output is directed back to the first stage of the multistage processor to perform a second predetermined function, such as three dimensional spatialization. The primary processor modifies parameters of the first predetermined function to efficiently perform dynamic operations, such as Doppler shifts and volume transitions between multiple sound sources and a mixture of those sounds as a single point source.
摘要翻译： 使用多级处理器执行一个或多个预定功能和可编程音频效果的一个或多个输入数据流的有效递归音频处理。 第一级执行第一预定功能，例如频移功能。 中等结果优选混合。 第二阶段将可编程音频效果应用于混合数据，例如混响效果，并将第二级输出存储在目的地混音箱中。 第二级输出优选地被传送到主处理器可访问的主存储器。 第二级输出被引导回到多级处理器的第一级以执行第二预定功能，例如三维空间化。 主处理器修改第一预定功能的参数以有效地执行动态操作，例如多声道偏移和多个声源之间的音量转换以及将这些声音作为单点源的混合。
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7.发明授权公开(公告)号：US06426456B1
公开(公告)日：2002-07-30
申请号：US10012273
申请日：2001-10-26
申请人： Jean Khawand , Radu Frangopol
发明人： Jean Khawand , Radu Frangopol
IPC分类号： G10H1057
CPC分类号： G10H1/12 , G10H2230/305 , G10H2250/121 , G10H2250/475 , Y10S84/09 , Y10S84/12
摘要： A percussive sound contains both harmonic and non-harmonic frequency spectral content. To reproduce a particular percussive sound, such as the sound of a drum or cymbal or hand clap, for example, the harmonic and non-harmonic content are determined empirically. Also, and tendency of the harmonic content to change over time, and the temporal aspects of attack, sustain, and decay are likewise determined empirically. These attributes are represented in the invention in a percussive sound file which includes a harmonic content profile (502), noise shape filter (504), Doppler shift profile (506), and a time wave shaping profile (508). The harmonic content profile is used by an FM generator (114) to generate a frequency modulated signal (116). The noise shape profile is used by a noise generator (134) to generate and shape the non-harmonic spectral content. While the sound is being generated, the Doppler shift profile is used to adjust the base frequency of the FM signal. The harmonic and non-harmonic signals are scaled (142, 144) and summed (146). The summed signal is then shaped (150) in time to substantially simulate the attack, sustain, and decay properties of the sound. The shaped, summed signal is then played by an audio circuit and converted to an acoustic signal.
摘要翻译： 冲击声包含谐波和非谐波频谱内容。 为了再现特定的打击乐音，例如鼓或钹或手拍的声音，例如，经验地确定谐波和非谐波含量。 此外，谐波含量随时间变化的趋势，以及攻击，维持和衰减的时间方面同样由经验确定。 这些属性在本发明中在包括谐波内容轮廓（502），噪声形状滤波器（504），多普勒频移分布（506）和时间波形整形轮廓（508）的冲击声音文件中表示。 谐波含量分布由FM发生器（114）使用以产生调频信号（116）。 噪声形状轮廓由噪声发生器（134）使用以产生和形成非谐波频谱内容。 在产生声音的同时，多普勒频移分布用于调整FM信号的基频。 谐波和非谐波信号被缩放（142,144）并求和（146）。 然后将加和的信号及时成形（150），以基本上模拟声音的攻击，维持和衰减特性。 然后由音频电路播放形状的求和信号，并转换成声信号。
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公开(公告)号：US5712808A
公开(公告)日：1998-01-27
申请号：US510167
申请日：1995-08-02
CPC分类号： G10H1/125 , H03H17/0294 , H03H17/04 , G10H2250/101 , G10H2250/121 , G10H2250/545
摘要： Digital signal processing (e.g. filtering) apparatus comprises pipelined digital signal processing means for generating each output sample of an output digital signal from input samples of an input digital signal by combining the input samples with a plurality of subsets of a set of filter coefficients associated with that output sample during a respective plurality of sample periods of the input digital signal; and coefficient generating means for supplying sets of filter coefficients to the signal processing means in response to the current state of a filter control signal. The coefficient generating means is operable to supply the plurality of subsets of each set of filter coefficients to the signal processing means during the respective plurality of sample periods of the input digital signal.
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公开(公告)号：US5210806A
公开(公告)日：1993-05-11
申请号：US598380
申请日：1990-10-16
申请人： Hisashi Kihara , Shinjiro Kato , Fumio Tamura , Shuichi Mori
发明人： Hisashi Kihara , Shinjiro Kato , Fumio Tamura , Shuichi Mori
CPC分类号： G10H1/125 , G10K15/02 , G10H2250/121
摘要： Disclosed herein is a digital audio signal processing apparatus of a type wherein a graphic equalizer including a plurality of filters connected in series to one another is subjected to arithmetic operation processing to be defined to thereby output the result of its processing as data therefrom. When a change-over command is generated, one filter out of the plurality of filters except for filters positioned at both ends is supplied to one of two output terminals with output data of a filter immediately before said one filter, the stored data is applied to the input of a filter immediately after said one filter and output data of a final filter is applied to the other of said two output terminals so as to define two graphic equalizers. Thus, where it is desired to carry out a change in the mode from arithmetic operation processing which defines a graphic equalizer comprising a plurality of bands to arithmetic operation processing which defines two-separated graphic equalizers comprising a plurality of bands or to the contrary, where it is desired to carry out a change in the mode contrary to the mode referred to above, the change in the arithmetic operation processing can be completed in a relatively short time.
摘要翻译： 这里公开了一种数字音频信号处理装置，其中包括彼此串联连接的多个滤波器的图形均衡器经受定义的算术运算处理，从而将其处理结果作为数据输出。 当生成转换命令时，除了位于两端的滤波器之外的多个滤波器中的一个滤波器被提供给紧邻所述一个滤波器之前的滤波器的输出数据的两个输出端子中的一个输出端子，所存储的数据被应用于 紧接在所述一个滤波器之后的滤波器的输入和最终滤波器的输出数据被施加到所述两个输出端子中的另一个，以便限定两个图形均衡器。 因此，在期望从定义包括多个频带的图形均衡器的算术运算处理中进行模式改变到定义包括多个频带的两分离图形均衡器的算术运算处理或相反地，其中 期望与上述模式相反地执行模式的改变，可以在较短的时间内完成算术运算处理的改变。
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公开(公告)号：US08067684B2
公开(公告)日：2011-11-29
申请号：US12571181
申请日：2009-09-30
申请人： Yoji Kaneko
发明人： Yoji Kaneko
CPC分类号： G10H1/125 , G10H7/02 , G10H2210/221 , G10H2250/095 , G10H2250/121 , G10H2250/615
摘要： In a filter device, a filter coefficient calculation circuit has a parameter table. The parameter table stores a plurality of sets of filter coefficients associated with a first parameter based on a frequency and a second parameter based on respective plurality of levels representing a degree of attenuation or enhancement of a gain of a filter in filter characteristics. The filter coefficient calculation circuit extracts a set of filter coefficients from a parameter table with the use of the first parameter and the second parameter determined according to a frequency and a strength of a musical sound signal, and outputs the extracted set of filter coefficients to the filter. The filter circuit performs filter processing for the musical sound signal, based on the filter characteristics determined by the set of filter coefficients.
摘要翻译： 在滤波器装置中，滤波器系数计算电路具有参数表。 基于表示滤波器特性中的滤波器的增益的衰减程度或增强度的相应多个级别，基于频率和第二参数，参数表存储与第一参数相关联的多个滤波器系数集合。 滤波器系数计算电路使用根据音乐声音信号的频率和强度确定的第一参数和第二参数从参数表提取一组滤波器系数，并将所提取的滤波器系数集合输出到 过滤。 滤波器电路基于由滤波器系数集合确定的滤波器特性来对音乐声音信号进行滤波处理。
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