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    • 1. 发明授权
    • Musical tone apparatus
    • 音乐设备
    • US07678988B2
    • 2010-03-16
    • US11987503
    • 2007-11-30
    • Takuya SatoKatsuhiko ToriiKoji Yoshida
    • Takuya SatoKatsuhiko ToriiKoji Yoshida
    • G10H1/32G10H3/00
    • G10H3/146H04R1/26H04R3/00H04S3/00
    • Sounds in different frequency range are generated from the rear, top and/or front face of the musical tone apparatus, sounds are generated and spread in all directions around the musical tone apparatus, and the tones sound very realistically acoustic. Vibration of the sound board does not resonate, tones generated from the sound board are not changed unintentionally, and tones of real acoustic musical instruments are realized. Vibration from the sound board is not directly transferred to the whole musical tone apparatus, by means of not allowing the sound board to touch the body of the musical tone apparatus and of pressing and fixing the soundboard to the attachment component with the thickness of the attachment component compressed.
    • 从音乐装置的后面,顶部和/或正面产生不同频率范围的声音,在音乐装置周围的各个方向上产生声音并扩展声音,并且音调非常实际地声音。 声音板的振动不会发生共鸣,声音板产生的声音不会无意地改变,实现了真正的声学乐器的音调。 声音板的振动不直接传递到整个乐音设备,借助于不允许声音板接触音乐设备的主体并且按照附件的厚度将音板按压并固定到附件上 组件压缩。
    • 2. 发明申请
    • Musical tone apparatus
    • 音乐设备
    • US20080127798A1
    • 2008-06-05
    • US11987503
    • 2007-11-30
    • Takuya SatoKatsuhiko ToriiKoji Yoshida
    • Takuya SatoKatsuhiko ToriiKoji Yoshida
    • G10D13/08
    • G10H3/146H04R1/26H04R3/00H04S3/00
    • Sounds in different frequency range are generated from the rear, top and/or front face of the musical tone apparatus, sounds are generated and spread in all directions around the musical tone apparatus, and the tones sound very realistically acoustic.Vibration of the sound board does not resonate, tones generated from the sound board are not changed unintentionally, and tones of real acoustic musical instruments are realized.Vibration from the sound board is not directly transferred to the whole musical tone apparatus, by means of not allowing the sound board to touch the body of the musical tone apparatus and of pressing and fixing the soundboard to the attachment component with the thickness of the attachment component compressed.
    • 从音乐装置的后面,顶部和/或正面产生不同频率范围的声音,在音乐装置周围的各个方向上产生声音并扩展声音,并且音调非常实际地声音。 声音板的振动不会发生共鸣,声音板产生的声音不会无意地改变,实现了真正的声学乐器的音调。 声音板的振动不直接传递到整个乐音设备,借助于不允许声音板接触音乐设备的主体并且按照附件的厚度将音板按压并固定到附件上 组件压缩。
    • 3. 发明授权
    • Audio encoding device using concealment processing and audio decoding device using concealment processing
    • 使用隐藏处理的音频编码装置和使用隐藏处理的音频解码装置
    • US09129590B2
    • 2015-09-08
    • US12528671
    • 2008-02-29
    • Takuya KawashimaHiroyuki EharaKoji Yoshida
    • Takuya KawashimaHiroyuki EharaKoji Yoshida
    • G10L19/00G10L19/005G10L19/12
    • G10L19/005G10L19/12
    • Disclosed are an audio encoding device and an audio decoding device which reduce degradation of subjective quality of a decoding signal caused by power mismatch of a decoding signal which is generated by a concealing process upon disappearance of a frame. When a frame is lost, a past encoding parameter is used to obtain a concealed LPC of the current frame and a concealed sound source parameter. A normal CELP decoding is performed from the obtained concealed sound source parameter. Correction is performed by using a conceal parameter on the obtained concealed LPC and the concealed sound source signal. The power of the corrected concealed sound source signal is adjusted to match a reference sound source power. A filter gain of the synthesis filter is adjusted so as to adjust the power of a decoded sound signal to the power of a decoded sound signal during an error-free state. Moreover, a synthesis filter gain adjusting coefficient is calculated by using an estimated normalized residual power so that a filter gain of a synthesis filter formed by using a concealed LPC is a filter gain during an error-free state.
    • 公开了一种音频编码装置和音频解码装置,其减少由于帧消失时由隐藏处理产生的解码信号的功率失配引起的解码信号的主观质量的劣化。 当帧丢失时,使用过去的编码参数来获得当前帧的隐藏的LPC和隐藏的声源参数。 从获得的隐藏声源参数执行正常的CELP解码。 通过对所获得的隐蔽LPC和隐藏的声源信号使用隐藏参数来执行校正。 校正的隐蔽声源信号的功率被调整以匹配参考声源功率。 调整合成滤波器的滤波器增益,以便在无错误状态期间将解码声音信号的功率调整为解码声音信号的功率。 此外,通过使用估计的归一化剩余功率来计算合成滤波器增益调整系数,使得通过使用隐藏的LPC形成的合成滤波器的滤波器增益在无错误状态期间是滤波器增益。
    • 5. 发明授权
    • Speech decoding and encoding apparatus for lost frame concealment using predetermined number of waveform samples peripheral to the lost frame
    • 语音解码和编码装置,用于使用丢失帧外围的预定数量的波形样本丢帧隐藏
    • US08812306B2
    • 2014-08-19
    • US12307974
    • 2007-07-11
    • Takuya KawashimaHiroyuki EharaKoji Yoshida
    • Takuya KawashimaHiroyuki EharaKoji Yoshida
    • G10L21/00H04L1/00G10L19/005G10L25/90H04L12/56
    • G10L19/005G10L25/90G10L2019/0011H04L47/10
    • An audio decoding device capable of suppressing an information amount for a lost flame compensation process and encoding efficiency is provided. A decoded sound source generator generates a lost frame's CELP decoded sound source signal. A pitch pulse information decoder CELP decodes a pitch pulse position information and a pitch pulse amplitude information. A pitch pulse waveform learner learns a pitch pulse learning waveform in a past frame in advance from the lost frame. A convolution adjuster amplitude-adjusts the pitch pulse learning waveform according to the pitch pulse amplitude information by considering a predetermined number of waveforms peripheral to a peak position of the lost frame's CELP decoded excitation signal, and convolutes a pitch pulse waveform into a time axis which has been amplitude-adjusted according to the pitch pulse position information. A sound source signal corrector adds or replaces the pitch pulse waveform convoluted into the time axis to the lost flame decoded sound source signal.
    • 提供能够抑制失火补偿处理和编码效率的信息量的音频解码装置。 解码声源发生器产生丢失帧的CELP解码声源信号。 音调脉冲信息解码器CELP解码音调脉冲位置信息和音调脉冲幅度信息。 音调脉冲波形学习者从丢失的帧中预先学习过去帧中的音调脉冲学习波形。 卷积调整器通过考虑丢失帧的CELP解码的激励信号的峰值位置外围的预定数量的波形,根据音调脉冲幅度信息来振幅调整音调脉冲学习波形,并将音调脉冲波形卷曲成时间轴, 已经根据音调脉冲位置信息进行了幅度调整。 声源信号校正器将与时间轴卷积的音调脉冲波形相加或替换为失火解码声源信号。
    • 8. 发明授权
    • Audio encoding/decoding apparatus and method providing multiple coding scheme interoperability
    • 音频编码/解码装置和方法提供多种编码方案的互操作性
    • US08670988B2
    • 2014-03-11
    • US11658150
    • 2005-06-29
    • Koji Yoshida
    • Koji Yoshida
    • G10L19/00A61K38/16
    • G10L19/012G10L25/78
    • There is provided an audio encoding device capable of causing a decoding side to freely select an audio decoding mode corresponding to a control method used for audio encoding and capable of generating data which can be decoded even when the decoding side does not correspond to the control method. The audio encoding device (100) outputs encoded data corresponding to an audio signal containing an audio component and encoded data corresponding to an audio signal containing no audio component. An audio encoding unit (102) encodes the input audio signal in a predetermined section unit and generates encoded data. An audio present/absent judgment unit (106) decides whether the input audio signal contains an audio component for each predetermined section. A bit embedding unit (104) performs synthesis of noise data only for those generated from the input audio signal of the voice absent section in the encoded data generated by the audio encoding unit (102), thereby acquiring encoded data corresponding to an audio signal containing an audio component and encoded data corresponding to an audio signal containing no audio component.
    • 提供了一种音频编码装置,其能够使解码侧自由地选择与用于音频编码的控制方法相对应的音频解码模式,并且即使当解码侧不对应于控制方法时也能够生成即使解码的数据 。 音频编码装置(100)输出对应于包含音频分量的音频信号的编码数据和对应于不包含音频分量的音频信号的编码数据。 音频编码单元(102)以预定的部分单位对输入音频信号进行编码,并生成编码数据。 音频存在/不存在判断单元(106)判定输入音频信号是否包含每个预定部分的音频分量。 位嵌入单元(104)仅对由音频编码单元(102)生成的编码数据中的语音缺失部分的输入音频信号产生的噪声数据进行合成,从而获取对应于包含 音频分量和对应于不包含音频分量的音频信号的编码数据。
    • 10. 发明授权
    • Quantizer, encoder, and the methods thereof
    • 量化器,编码器及其方法
    • US08473288B2
    • 2013-06-25
    • US12990697
    • 2009-06-18
    • Toshiyuki MoriiHiroyuki EharaKoji Yoshida
    • Toshiyuki MoriiHiroyuki EharaKoji Yoshida
    • G10L19/00
    • G10L19/008G10L25/27
    • Disclosed are a quantizer, encoder, and the methods thereof, wherein the computational load is reduced when the values related to the transform coefficients of the principal component analysis transform are quantized when a principal component analysis transform is applied to code stereo. A quantizer includes a power correlation calculator which calculates the power of the left channel signal, the power of the right channel signal, and the correlation between the left channel signal and the right channel signal; an intermediate value calculator which calculates the intermediate value which is the difference between left channel signal the power and the right channel signal power; a codebook which holds a plurality of sets of the coefficients related to the transform coefficients of the principal component analysis transform and the code; and a quantizer which calculates the sum of the first multiplication result obtained by multiplying the coefficient by the correlation value and the second multiplication result obtained by multiplying the coefficient by the intermediate value as the cost function E, selects the coefficients where the cost function E becomes the maximum, and fetches the code related to the selected coefficients as the quantized code.
    • 公开了一种量化器,编码器及其方法,其中当将主分量分析变换应用于代码立体声时,当与主成分分析变换的变换系数相关的值被量化时,计算负荷减小。 量化器包括功率相关计算器,其计算左声道信号的功率,右声道信号的功率,以及左声道信号和右声道信号之间的相关性; 中间值计算器,其计算作为左声道信号的功率和右声道信号功率之间的差的中间值; 码本,其保存与主成分分析变换和代码的变换系数相关的多个系数集合; 以及量化器,其计算通过将系数乘以相关值而获得的第一相乘结果与通过将系数乘以中间值而获得的第二乘法结果作为成本函数E的和,选择成本函数E变为 最大值,并且将与所选择的系数相关的代码作为量化代码获取。