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1. 发明授权

US06658382B1 Audio signal coding and decoding methods and apparatus and recording media with programs therefor 有权
标题翻译：音频信号编码和解码方法以及具有程序的设备和记录介质
公开(公告)号：US06658382B1
公开(公告)日：2003-12-02
申请号：US09534297
申请日：2000-03-23
申请人： Naoki Iwakami , Takehiro Moriya , Akio Jin , Kazuaki Chikira , Takeshi Mori
发明人： Naoki Iwakami , Takehiro Moriya , Akio Jin , Kazuaki Chikira , Takeshi Mori
IPC分类号： G10L1902
CPC分类号： G10L19/0212
摘要： An input signal is time-frequency transformed, then the frequency-domain coefficients are divided into coefficient segments of about 100 Hz width to generate a sequence of coefficient segments, and the sequence of coefficient segments is split into subbands each consisting of plural coefficient segments. A threshold value is determined based on the intensity of each coefficient segment in each subband. The intensity of each coefficient segment is compared with the threshold value, and the coefficient segments are classified into low- and high-intensity groups. The coefficient segments are quantized for each group, or they are flattened respectively and then quantized through recombination.
摘要翻译：输入信号被时间频率变换，频域系数被划分为大约100Hz宽度的系数段，以产生一系列系数段，并将系数段序列分成各个由多个系数段组成的子带。基于每个子带中的每个系数段的强度来确定阈值。将每个系数段的强度与阈值进行比较，将系数段分为低强度组和高强度组。每个组对系数段进行量化，或者分别对其进行平坦化，然后通过重组进行量化。

2. 外观设计

USD657815S1 Banknote counter 有权
标题翻译：钞票柜台
公开(公告)号：USD657815S1
公开(公告)日：2012-04-17
申请号：US29370590
申请日：2010-08-09
申请人： Naoki Iwakami
设计人： Naoki Iwakami

3. 发明授权

US06345246B1 Apparatus and method for efficiently coding plural channels of an acoustic signal at low bit rates 失效
标题翻译：用于以低比特率有效地编码声信号的多个声道的装置和方法
公开(公告)号：US06345246B1
公开(公告)日：2002-02-05
申请号：US09018042
申请日：1998-02-03
申请人： Takehiro Moriya , Takeshi Mori , Kazunaga Ikeda , Naoki Iwakami
发明人： Takehiro Moriya , Takeshi Mori , Kazunaga Ikeda , Naoki Iwakami
IPC分类号： G10L1904
CPC分类号： G10L19/008
摘要： In multichannel acoustic signal coding and decoding, left- and right-channel signals are alternately interleaved for each sample to generate a one-dimensional signal sample sequence. The one-dimensional signal sample sequence is subjected to coding based on correlation. In coding, the left- and right-channel signals may preferably be interleaved after reducing an imbalance in power between input channels. In such an instance, a power imbalance is introduced between the decoded left- and right-channel signal sample sequences
摘要翻译：在多声道声信号编码和解码中，左和右声道信号被交替地交替地用于每个采样以产生一维信号采样序列。一维信号采样序列经过相关的编码。在编码中，在减少输入通道之间的功率不平衡的情况下，左通道信号和右声道信号可优选地交错。在这种情况下，在解码的左声道和右声道信号样本序列之间引入功率不平衡

4. 发明授权

US5651090A Coding method and coder for coding input signals of plural channels using vector quantization, and decoding method and decoder therefor 失效
标题翻译：用于使用向量量化对多个信道的输入信号进行编码的编码方法和编码器及其解码方法和解码器
公开(公告)号：US5651090A
公开(公告)日：1997-07-22
申请号：US433962
申请日：1995-05-04
申请人： Takehiro Moriya , Naoki Iwakami
发明人： Takehiro Moriya , Naoki Iwakami
IPC分类号： G06T9/00 , G10L19/00 , G10L19/008 , G10L19/02 , G10L19/038 , G10L19/12 , H04B1/66 , H04B14/04 , G10L3/02
CPC分类号： G10L19/038 , G06T9/008 , G10L19/008 , H04B1/667 , H04B14/046 , G10L19/00 , G10L19/0212 , G10L19/12
摘要： Power normalization parts calculate the average powers of input signals of a plurality of channels for each frame and divide the signals by the calculated average powers to generate normalized signals and, at the same time, generate weights corresponding to the normalization gains. The normalized signals of the plurality of channels are combined in a combining part into predetermined sequences and outputted therefrom as one or more interleaved signal vectors. The combining part combines the weights from the power normalization part into the same sequences of the normalized signal and outputs one or more interleaved weight vectors. In a vector quantization part the signal vectors are vector quantized by the interleaved weight vectors corresponding thereto, respectively, and quantization indexes and normalization indexes are outputted as results of coding.
摘要翻译：功率归一化部件计算每个帧的多个信道的输入信号的平均功率，并将信号除以计算出的平均功率以产生归一化信号，并且同时产生对应于归一化增益的权重。多个信道的归一化信号在组合部分中组合成预定的序列，并作为一个或多个交织的信号向量输出。组合部分将来自功率归一化部分的权重组合成归一化信号的相同序列，并输出一个或多个交织权重向量。在矢量量化部分中，信号矢量分别由与其对应的交织权重向量进行矢量量化，作为编码结果输出量化索引和归一化指标。

5. 发明授权

US5732188A Method for the modification of LPC coefficients of acoustic signals 失效
标题翻译：用于修改声信号的LPC系数的方法
公开(公告)号：US5732188A
公开(公告)日：1998-03-24
申请号：US612797
申请日：1996-03-11
申请人： Takehiro Moriya , Kazunori Mano , Satoshi Miki , Hitoshi Ohmura , Shigeaki Sasaki , Naoki Iwakami
发明人： Takehiro Moriya , Kazunori Mano , Satoshi Miki , Hitoshi Ohmura , Shigeaki Sasaki , Naoki Iwakami
IPC分类号： G10L19/00 , G10L19/04 , G10L19/06 , G10L19/12 , H03H17/02 , G10L9/04
CPC分类号： G10L19/06 , G10L25/24
摘要： In a CELP coding scheme, p-order LPC coefficients of an input signal are transformed into n-order LPC cepctrum coefficients c.sub.j (S.sub.2), which are modified into n-order modified LPC cepstrum coefficients c.sub.j ' (S.sub.3). Log power spectral envelopes of the input signal and a masking function suited thereto are calculated (FIGS. 3B, C), then they are subjected to inverse Fourier transform to obtain n-order LPC cepstrum coefficients, respectively, (FIGS. 3D, E), then the relationship between corresponding orders of the LPC cepstrum coefficients is calculated, and the modification in step S.sub.3 is carried out on the basis of the relationship. The modified coefficients c.sub.j are inversely transformed by the method of least squares into m-order LPC coefficients for use as filter coefficients of a perceptual weighting filter. This concept is applicable to a postfilter as well.
摘要翻译：在CELP编码方案中，输入信号的p阶LPC系数被变换为n阶LPC位移系数cj（S2），其被修改为n阶修正LPC倒谱系数cj'（S3）。计算输入信号的对数功率谱包络和适合其的屏蔽功能（图3B，C），然后对它们进行傅里叶逆变换以分别获得n阶LPC倒谱系数（图3D，E）然后计算LPC倒谱系数的相应次序之间的关系，并且基于该关系来执行步骤S3中的修改。经修正的系数cj通过最小二乘法的方法被逆变换为m阶LPC系数，用作感知加权滤波器的滤波器系数。这个概念也适用于后置过滤器。

6. 发明授权

US5684920A Acoustic signal transform coding method and decoding method having a high efficiency envelope flattening method therein 失效
标题翻译：声信号变换编码方法及其中具有高效率包络平滑化方法的解码方法
公开(公告)号：US5684920A
公开(公告)日：1997-11-04
申请号：US402660
申请日：1995-03-13
申请人： Naoki Iwakami , Takehiro Moriya , Satoshi Miki
发明人： Naoki Iwakami , Takehiro Moriya , Satoshi Miki
IPC分类号： G01R23/16 , G10L19/02 , G10L25/12 , G10L25/27 , G10L9/16
CPC分类号： G10L19/0212 , G10L19/0204 , G10L25/12 , G10L25/27
摘要： An input acoustic signal is subjected to modified discrete cosine transform processing to obtain its spectrum characteristics. Linear prediction coefficients are derived from the input acoustic signal in a linear prediction coding analysis part, and the prediction coefficients are subjected to Fourier transform in a spectrum envelope calculation part to obtain the envelope of the spectrum characteristics of the input acoustic signal. In a normalization part the spectrum characteristics are normalized by the envelope thereof to obtain residual coefficients. Another normalization part normalizes the residual coefficients by a residual-coefficients envelope predicted in a residual-coefficients envelope calculation part, thereby obtaining fine structure coefficients, which are vector-quantized in a quantization part. A de-normalization part de-normalizes the quantized fine structure coefficients. The residual-coefficients envelope calculation part uses the reproduced residual coefficients to predict the envelope of residual coefficients of the subsequent frame.
摘要翻译：对输入声信号进行修正的离散余弦变换处理以获得其频谱特性。线性预测系数从线性预测编码分析部中的输入声信号导出，并且在频谱包络计算部中对预测系数进行傅里叶变换，以获得输入声信号的频谱特性的包络。在归一化部分中，光谱特性通过其包络进行归一化以获得残差系数。另一个归一化部分通过在残差系数包络计算部分中预测的残差系数包络对残差系数进行归一化，从而获得在量化部分中矢量量化的精细结构系数。去归一化部分使量化的精细结构系数解规范化。剩余系数包络计算部分使用再现的残差系数来预测后续帧的残差系数的包络。

7. 发明授权

US06549147B1 Methods, apparatuses and recorded medium for reversible encoding and decoding 有权
标题翻译：用于可逆编码和解码的方法，装置和记录介质
公开(公告)号：US06549147B1
公开(公告)日：2003-04-15
申请号：US09572789
申请日：2000-05-17
申请人： Takehiro Moriya , Naoki Iwakami , Akio Jin , Takeshi Mori , Kazuaki Chikira
发明人： Takehiro Moriya , Naoki Iwakami , Akio Jin , Takeshi Mori , Kazuaki Chikira
IPC分类号： H03M700
CPC分类号： H04N19/00 , H03M7/30 , H04N19/90
摘要： An object of the invention is to provide a method for compressing digital input signals at high compression efficiency and reproducing the input data perfectly. The method includes the steps of: converting a digital input signal in each frame to bitstreams according to a sign-magnitude format; deblocking the bitstreams into individual bits; joining each bit in a time sequence while retaining an identical chronological order of bits in all the frames; and reversibly encoding each bitstream obtained by joining the bits. And, the reversible decoding method includes the steps of: reversibly decoding a reversible code sequence in each frame; deblocking the bitstreams obtained by reversible decoding into individual bits; joining each bit in a time sequence while retaining an identical chronological order of bits in all the frames; and joining successive frames obtained by joining the bits.
摘要翻译：本发明的目的是提供一种以高压缩效率压缩数字输入信号并且完美地再现输入数据的方法。该方法包括以下步骤：根据符号幅度格式将每帧中的数字输入信号转换为比特流; 将比特流解块为单独的比特; 以时间顺序连接每个比特，同时保留所有帧中相同的时间顺序的比特; 并且可逆地编码通过连接比特获得的每个比特流。并且，可逆解码方法包括以下步骤：对每帧中的可逆码序列进行可逆解码; 将通过可逆解码获得的比特流解块为单独的比特; 以时间顺序连接每个比特，同时保留所有帧中相同的时间顺序的比特; 并连接通过连接比特获得的连续帧。

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