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    • 1. 发明授权
    • Audio processing device
    • 音频处理设备
    • US08634275B2
    • 2014-01-21
    • US12910182
    • 2010-10-22
    • Kazunobu KondoKenichi YamauchiHideki Kemmochi
    • Kazunobu KondoKenichi YamauchiHideki Kemmochi
    • H04B1/06
    • H04B1/06H04R1/406H04R3/005
    • In an audio processing device, a target sound emphasizer generates a target sound emphasized component by emphasizing a target sound component contained in a plurality of audio signals generated by a plurality of sound receiving devices. A stereo processor generates a stereo component of a plurality of channels from the plurality of audio signals. A first adjuster adjusts a sound pressure level of the target sound emphasized component according to a first adjustment value, and a second adjuster adjusts a sound pressure level of the stereo component according to a second adjustment value. A variable setter variably sets a zoom value which is changeable between a wide angle side and a telephoto side relative to a target. An adjustment controller controls the first adjustment value according to the zoom value such that the sound pressure level of the target sound emphasized component exponentially decreases as the zoom value changes toward the wide-angle side and controls the second adjustment value according to the zoom value such that the sound pressure level of the stereo component increases as the zoom value changes toward the wide-angle side.
    • 在音频处理装置中,目标声音增强器通过强调由多个声音接收装置产生的多个音频信号中包含的目标声音成分来产生目标声音强调分量。 立体声处理器从多个音频信号产生多个声道的立体声分量。 第一调节器根据第一调整值调整目标声音增强分量的声压级,并且第二调节器根据第二调整值调节立体声分量的声压级。 可变设定器可变地设置相对于目标物在广角侧和长焦侧之间可变的变焦值。 调节控制器根据变焦值控制第一调整值,使得目标声音强调成分的声压级随着变焦值向广角侧变化而指数地减小,并且根据变焦值控制第二调整值, 立体声分量的声压级随着变焦值向广角侧变化而增加。
    • 2. 发明授权
    • Sound processing apparatus and method, and program therefor
    • 声音处理装置及方法及其程序
    • US07945446B2
    • 2011-05-17
    • US11372812
    • 2006-03-09
    • Hideki KemmochiYasuo YoshiokaJordi Bonada
    • Hideki KemmochiYasuo YoshiokaJordi Bonada
    • G10L21/00G10L13/06G10L13/00
    • G10H1/366G10H1/10G10H5/005G10H2210/251G10H2250/031G10L13/033
    • Spectrum envelope of an input sound is detected. In the meantime, a converting spectrum is acquired which is a frequency spectrum of a converting sound comprising a plurality of sounds, such as unison sounds. Output spectrum is generated by imparting the detected spectrum envelope of the input sound to the acquired converting spectrum. Sound signal is synthesized on the basis of the generated output spectrum. Further, a pitch of the input sound may be detected, and frequencies of peaks in the acquired converting spectrum may be varied in accordance with the detected pitch of the input sound. In this manner, the output spectrum can have the pitch and spectrum envelope of the input sound and spectrum frequency components of the converting sound comprising a plurality of sounds, and thus, unison sounds can be readily generated with simple arrangements.
    • 检测输入声音的频谱包络。 同时,获取转换频谱,其是包括多个声音(例如一致声音)的转换声音的频谱。 通过将检测到的输入声音的频谱包络赋予所获取的转换频谱来产生输出频谱。 声音信号是根据产生的输出频谱进行合成的。 此外,可以检测输入声音的音调,并且可以根据检测到的输入声音的音调来改变所获取的转换频谱中的峰值频率。 以这种方式,输出频谱可以具有包括多个声音的转换声音的输入声音和频谱频率分量的音调和频谱包络,从而可以以简单的布置容易地产生一致的声音。
    • 4. 发明申请
    • Apparatus for and program of processing audio signal
    • 用于处理音频信号的装置和程序
    • US20060111903A1
    • 2006-05-25
    • US11273749
    • 2005-11-14
    • Hideki KemmochiJordi Bonada
    • Hideki KemmochiJordi Bonada
    • G10L15/06
    • G10H1/0091G10H1/366G10H2210/251G10H2250/455G10L13/04G10L19/26
    • In an audio signal processing apparatus, a generation section generates an audio signal representing a voice. A distribution section distributes the audio signal generated by the generation section to a first channel and a second channel, respectively. A delay section delays the audio signal of the first channel relative to the audio signal of the second channel for creating a phase difference between the audio signal of the first channel and the audio signal of the second channel such that the created phase difference has a duration corresponding to either an added value of a first duration which is approximately one half of a period of the audio signal generated by the generation section and a second duration which is set shorter than the first duration, or a difference value of the first duration and the second duration. An addition section adds the audio signal of the first channel and the audio signal of the second channel with one another, between which the phase difference is created by the delay section, and outputs the added audio signal which represents natural voice with various characteristics.
    • 在音频信号处理装置中,生成部生成表示声音的音频信号。 分配部将由生成部生成的音频信号分别分配到第一信道和第二信道。 延迟部分相对于第二通道的音频信号延迟第一通道的音频信号,以产生第一通道的音频信号和第二通道的音频信号之间的相位差,使得所创建的相位差具有持续时间 对应于第一持续时间的附加值,其大约是由生成部分生成的音频信号的周期的一半,以及被设置为短于第一持续时间的第二持续时间,或者第一持续时间和 第二个持续时间。 加法部分将第一信道的音频信号和第二信道的音频信号彼此相加,在延迟部分之间产生相位差,并输出表示具有各种特性的自然语音的附加音频信号。
    • 5. 发明授权
    • Voice synthesizer of multi sounds
    • 多声音合成器
    • US07613612B2
    • 2009-11-03
    • US11345023
    • 2006-01-31
    • Hideki KemmochiJordi Bonada
    • Hideki KemmochiJordi Bonada
    • G10L13/04
    • G10L13/06G10L25/18G10L2021/0135
    • In a voice synthesizer, an envelope acquisition portion obtains a spectral envelope of a reference frequency spectrum of a given voice. A spectrum acquisition portion obtains a collective frequency spectrum of a plurality of voices which are generated in parallel to one another. An envelope adjustment portion adjusts a spectral envelope of the collective frequency spectrum obtained by the spectrum acquisition portion so as to approximately match with the spectral envelope of the reference frequency spectrum obtained by the envelope acquisition portion. A voice generation portion generates an output voice signal from the collective frequency spectrum having the spectral envelope adjusted by the envelope adjustment portion.
    • 在语音合成器中,包络获取部分获得给定语音的参考频谱的频谱包络。 频谱获取部分获得彼此并行产生的多个语音的集体频谱。 信封调整部分调整由频谱获取部分获得的集体频谱的频谱包络,以便与由包络获取部分获得的参考频谱的频谱包络近似匹配。 声音产生部分从具有通过包络线调节部分调整的频谱包络的​​集体频谱产生输出声音信号。
    • 7. 发明授权
    • Apparatus for and program of processing audio signal
    • 用于处理音频信号的装置和程序
    • US08170870B2
    • 2012-05-01
    • US11273749
    • 2005-11-14
    • Hideki KemmochiJordi Bonada
    • Hideki KemmochiJordi Bonada
    • G10L11/04G10L13/00G10H1/06
    • G10H1/0091G10H1/366G10H2210/251G10H2250/455G10L13/04G10L19/26
    • In an audio signal processing apparatus, a generation section generates an audio signal representing a voice. A distribution section distributes the audio signal generated by the generation section to a first channel and a second channel, respectively. A delay section delays the audio signal of the first channel relative to the audio signal of the second channel for creating a phase difference between the audio signal of the first channel and the audio signal of the second channel such that the created phase difference has a duration corresponding to either an added value of a first duration which is approximately one half of a period of the audio signal generated by the generation section and a second duration which is set shorter than the first duration, or a difference value of the first duration and the second duration. An addition section adds the audio signal of the first channel and the audio signal of the second channel with one another, between which the phase difference is created by the delay section, and outputs the added audio signal which represents natural voice with various characteristics.
    • 在音频信号处理装置中,生成部生成表示声音的音频信号。 分配部将由生成部生成的音频信号分别分配到第一信道和第二信道。 延迟部分相对于第二通道的音频信号延迟第一通道的音频信号,以产生第一通道的音频信号和第二通道的音频信号之间的相位差,使得所创建的相位差具有持续时间 对应于第一持续时间的附加值,其大约是由生成部分生成的音频信号的周期的一半,以及被设置为短于第一持续时间的第二持续时间,或者第一持续时间和 第二个持续时间。 加法部分将第一信道的音频信号和第二信道的音频信号彼此相加,在延迟部分之间产生相位差,并输出表示具有各种特性的自然语音的附加音频信号。
    • 8. 发明授权
    • Singing voice synthesizing apparatus, singing voice synthesizing method and program for singing voice synthesizing
    • 唱歌语音合成装置,歌唱合成方法和歌唱合成程序
    • US07135636B2
    • 2006-11-14
    • US10375272
    • 2003-02-27
    • Hideki KemmochiYasuo YoshiokaJordi Bonada
    • Hideki KemmochiYasuo YoshiokaJordi Bonada
    • G10H1/06G10H7/00
    • G10L13/00G10H7/00G10H2240/056G10H2250/235G10H2250/455
    • A method for synthesizing a natural-sounding singing voice divides performance data into a transition part and a long sound part. The transition part is represented by articulation (phonemic chain) data that is read from an articulation template database and is outputted without modification. For the long sound part, a new characteristic parameter is generated by linearly interpolating characteristic parameters of the transition parts positioned before and after the long sound part and adding thereto a changing component of stationary data that is read from a constant part (stationary) template database. An associated apparatus for carrying out the singing voice synthesizing method includes a phoneme database for storing articulation data for the transition part and stationary data for the long sound part, a first device for outputting the articulation data, and a second device for outputting the newly-generated characteristic parameter of the long sound part.
    • 用于合成自然发声的歌声的方法将演奏数据分成转换部分和长音部分。 过渡部分由从关节运动模板数据库读取并且没有修改地输出的关节(音素链)数据表示。 对于长音部分,通过线性内插位于长声部分之前和之后的过渡部分的特征参数,并且向其添加从恒定部分(静止)模板数据库读取的静止数据的变化分量,生成新的特征参数 。 用于执行歌唱声合成方法的相关装置包括用于存储用于转换部分的发音数据的音素数据库和用于长音部分的固定数据,用于输出关节数据的第一装置,以及用于输出新音符的第二装置, 生成长音部分的特征参数。
    • 9. 发明授权
    • Voice synthesis apparatus and method
    • 语音合成装置及方法
    • US07552052B2
    • 2009-06-23
    • US11180108
    • 2005-07-13
    • Hideki Kemmochi
    • Hideki Kemmochi
    • G10L13/02
    • G10L13/06G10L13/033G10L13/04
    • A plurality of voice segments, each including one or more phonemes are acquired in a time-serial manner, in correspondence with desired singing or speaking words. As necessary, a boundary is designated between start and end points of a vowel phoneme included in any one of the acquired voice segments. Voice is synthesized for a region of the vowel phoneme that precedes the designated boundary vowel phoneme, or a region of the vowel phoneme that succeeds the designated boundary in the vowel phoneme. By synthesizing a voice for the region preceding the designated boundary, it is possible to synthesize a voice imitative of a vowel sound that is uttered by a person and then stopped to sound with his or her mouth kept opened. Further, by synthesizing a voice for the region succeeding the designated boundary, it is possible to synthesize a voice imitative of a vowel sound that is started to sound with the mouth opened.
    • 根据期望的歌唱或说话字,以时间序列方式获取每个包括一个或多个音素的多个语音段。 根据需要,在包含在所获取的语音片段中的任一个中的元音音素的起点和终点之间指定边界。 在元音音素之前的元音音素区域合成声音,或是在元音音素中成为指定边界的元音音素的区域。 通过合成指定边界之前的区域的声音,可以合成一个模仿由一个人发出的元音的声音,然后在他或她的嘴保持打开的情况下停止发声。 此外,通过合成在指定边界之后的区域的声音,可以合成开口的开始发声的元音的模仿语音。
    • 10. 发明授权
    • Singing voice synthesizing apparatus with selective use of templates for attack and non-attack notes
    • 唱歌语音合成装置,可选择使用模板进行攻击和非攻击笔记
    • US07383186B2
    • 2008-06-03
    • US10792265
    • 2004-03-03
    • Hideki Kemmochi
    • Hideki Kemmochi
    • G10L19/00
    • G10L13/033G10H5/00G10H2250/455
    • In an apparatus for synthesizing a singing voice of a song, a storage section stores template data in correspondence to various expressions applicable to music notes. The template data includes first and second template data differently defining a temporal variation of a characteristic parameter for applying the corresponding expression to an attack note and a non-attack note, respectively. An input section inputs voice information representing a sequence of vocal elements and specifying expressions in correspondence to the respective vocal elements. A synthesizing section synthesizes the singing voice from the sequence of the vocal elements based on the inputted voice information. When the vocal element is of an attack note, the first template data is applied to the vocal element. Otherwise, when the vocal element is of a non-attack note, the second template data is applied to the vocal element.
    • 在用于合成歌曲的歌声的装置中,存储部分对应于适用于音乐音符的各种表达来存储模板数据。 模板数据包括不同地限定用于将相应表达式应用于攻击记录和非攻击记录的特征参数的时间变化的第一和第二模板数据。 输入部分输入表示声乐元素序列的声音信息,并且对应于各声部元素指定表达式。 合成部根据所输入的语音信息,根据声音元素的序列合成唱歌声音。 当声音元素是攻击记录时,第一个模板数据被应用于声音元素。 否则,当声音元素是非攻击音符时,第二个模板数据被应用于声音元素。