公开(公告)号：US06930618B2

公开(公告)日：2005-08-16

申请号：US10483105

申请日：2003-04-30

申请人： Keisuke Touyama ,  Shiro Suzuki ,  Minoru Tsuji

发明人： Keisuke Touyama ,  Shiro Suzuki ,  Minoru Tsuji

IPC分类号： G10L19/00 ,  G10L19/02 ,  H03M7/30 ,  H03M7/40 ,  H04N7/24 ,  H04N19/00 ,  H04N19/625 ,  H04N19/91 ,  H03M7/38

CPC分类号： G10L19/032 ,  G10L19/18 ,  H03M7/40 ,  H04N7/24

摘要： In an encoding apparatus (10), a code table group exchange judgment unit (12) selects a group of code tables from a plurality of groups of code tables according to the property such as tonality of a spectrum signal D11, and a quantization unit (15) encodes quantization coefficients using a code table included in the selected group. Then, a multiplexer (18) multiplexes a group index D12 together with coefficient data D17. Furthermore, in case resources for an encoder is small, and sound quality may have to be somewhat degraded so as to realize practical encoding speed, a code table number change judgment unit (17) changes or reduces the number of code tables to be used.

摘要翻译： 在编码装置（10）中，代码表组交换判定部（12）根据频谱信号D11的音调，量化部（12）等的属性从多组代码表中选择一组代码表 （15）使用包括在所选择的组中的代码表对量化系数进行编码。 然后，多路复用器（18）将组索引D12与系数数据D17一起复用。 此外，在编码器的资源小的情况下，为了实现实际的编码速度，可能必须稍微降低声音质量，码表号码变更判断单元（17）改变或减少要使用的代码表的数量。

公开(公告)号：US07428489B2

公开(公告)日：2008-09-23

申请号：US10483088

申请日：2003-04-30

申请人： Keisuke Touyama ,  Shiro Suzuki ,  Minoru Tsuji

发明人： Keisuke Touyama ,  Shiro Suzuki ,  Minoru Tsuji

IPC分类号： G10L19/00 ,  G10L19/14 ,  G10L21/00

CPC分类号： G10L19/083 ,  G10L19/028

摘要： In a decoding apparatus (30), power compensation spectrum generation/composition units (371 to 374) adjust power of power compensation spectrums PCSP based on quantization accuracy information, normalization coefficients, gain control information, and power adjustment information. Then, power of the spectrums SP is compensated by replacing spectrums SP being equal to or smaller than a threshold with the power-adjusted power compensation spectrums PCSP, or by adding the power-adjusted power compensation spectrums PCSP to the spectrums SP.

摘要翻译： 在解码装置（30）中，功率补偿频谱产生/合成单元（37 1〜37 4）根据量化精度信息调整功率补偿频谱PCSP的功率，归一化 系数，增益控制信息和功率调整信息。 然后，通过用功率调整功率补偿频谱PCSP替换等于或小于阈值的频谱SP，或者通过将功率调整的功率补偿频谱PCSP加到频谱SP来补偿频谱SP的功率。

公开(公告)号：US07933417B2

公开(公告)日：2011-04-26

申请号：US11324633

申请日：2006-01-03

申请人： Keisuke Toyama ,  Shiro Suzuki ,  Minoru Tsuji

发明人： Keisuke Toyama ,  Shiro Suzuki ,  Minoru Tsuji

IPC分类号： H04R5/00 ,  G10L19/00 ,  G10L19/14 ,  H03M7/00 ,  H03M7/34 ,  H04B14/04

CPC分类号： H04B1/667 ,  G10L19/0208

摘要： The present invention relates to an encoding device for saving the number of bits of codes. In step S11, the differential value between a normalization coefficient Bi to be encoded and a normalization coefficient Bi-1 for an encoding unit Ai-1 in a band adjacent to the lower side of an encoding unit Ai corresponding to the normalization coefficient Bi is computed. In step S12, reference is made to a table in which a differential value having a high frequency of occurrence is associated with a code having a small number of bits, and a code corresponding to the computed differential value is read. In step S13, it is determined whether or not all normalization coefficients B have been encoded. If it is determined that all normalization coefficients B have been encoded, in step S14, the code read in step S12 is output. The present invention is applicable to an audio recorder.

摘要翻译： 本发明涉及一种用于节省代码比特数的编码装置。 在步骤S11中，计算要编码的归一化系数Bi与与归一化系数Bi相对应的编码单元Ai的下侧的频带中的编码单位Ai-1的归一化系数Bi-1的差分值 。 在步骤S12中，参考具有高出现频率的差分值与具有少量位的代码相关联的表，并且读取与所计算出的差分值相对应的代码。 在步骤S13中，确定所有归一化系数B是否已被编码。 如果确定所有归一化系数B已被编码，则在步骤S14中，输出在步骤S12中读取的代码。 本发明可应用于音频记录器。

公开(公告)号：US07366661B2

公开(公告)日：2008-04-29

申请号：US10203733

申请日：2001-12-14

申请人： Minoru Tsuji ,  Shiro Suzuki ,  Keisuke Toyama

发明人： Minoru Tsuji ,  Shiro Suzuki ,  Keisuke Toyama

IPC分类号： G10L11/00 ,  G10L13/00 ,  G10L19/00

CPC分类号： G10L19/093 ,  G10L19/022

摘要： The present invention relates to an information extraction apparatus capable of analyzing an acoustic signal with accuracy and high efficiency.An amplitude analysis section 32 determines whether or not an attack or release is contained on the basis of an amplitude value for each small region of an input time-series signal. When it is determined that there is an attack or release, an analysis region setting section 33 sets the portion from an attack position to a release position as an analysis region. A frequency analysis section 34 analyzes the input time-series signal by generalized harmonic analysis and outputs extracted waveform information. An extracted waveform synthesis section 35 synthesizes the extracted waveform information and outputs the information to a time-series compensation section 36. The time-series compensation section 36 compensates the signal of the synthesized result with a signal outside the analysis region and outputs an extracted waveform time-series signal to a subtraction unit 37. The subtraction unit 37 generates a residual time-series signal from the input time-series signal and the extracted waveform time-series signal. The present invention can be applied to various audio apparatuses, voice recognition apparatuses, voice synthesis apparatuses, etc., for processing an acoustic signal.

摘要翻译： 信息提取装置技术领域本发明涉及能够精确，高效地分析声音信号的信息提取装置。 振幅分析部分32基于输入时间序列信号的每个小区域的幅度值来确定是否包含攻击或释放。 当确定存在攻击或释放时，分析区域设置部33将来自攻击位置的部分设置为释放位置作为分析区域。 频率分析部分34通过广义谐波分析分析输入的时间序列信号并输出​​提取的波形信息。 提取的波形合成部分35合成提取的波形信息并将其输出到时间序列补偿部分36.时间序列补偿部分36用分析区域外的信号补偿合成结果的信号，并输出提取的波形 时间序列信号发送到减法单元37.减法单元37根据输入时间序列信号和提取的波形时间序列信号产生残留时间序列信号。 本发明可以应用于用于处理声信号的各种音频装置，语音识别装置，语音合成装置等。

公开(公告)号：US07212973B2

公开(公告)日：2007-05-01

申请号：US10344624

申请日：2002-06-11

申请人： Keisuke Toyama ,  Shiro Suzuki ,  Minoru Tsuji ,  Masayuki Nishiguchi

发明人： Keisuke Toyama ,  Shiro Suzuki ,  Minoru Tsuji ,  Masayuki Nishiguchi

IPC分类号： G10L19/02 ,  G06K9/38

CPC分类号： G10L19/032 ,  G10L19/02 ,  G10L19/06 ,  H03M7/30

摘要： In a quantization step information encoding unit, an average value of the quantization step information is found in an approximate shape extraction unit (20), first of all, from one set of a given number of unitary quantization units to another. In an approximate shape encoding unit (21), the approximate shape information is vector-quantized. In a residual signal computing unit (22), the residual signals between the quantization step information and the quantized approximate shape vector are computed. In a residual signal encoding unit (23), the residual signals are variable length encoded, and the so encoded residual signals and the vector quantized approximate shape information are output.

摘要翻译： 在量化步长信息编码单元中，在近似形状提取单元（20）中找到量化步长信息的平均值，首先，从一组给定数量的单位量化单元到另一组。 在近似形状编码单元（21）中，近似形状信息被矢量量化。 在残差信号计算单元（22）中，计算量化步长信息和量化近似形状矢量之间的残差信号。 在剩余信号编码单元（23）中，剩余信号是可变长度编码的，并且输出如此编码的残差信号和矢量量化的近似形状信息。

公开(公告)号：US20060115092A1

公开(公告)日：2006-06-01

申请号：US11324633

申请日：2006-01-03

申请人： Keisuke Toyama ,  Shiro Suzuki ,  Minoru Tsuji

发明人： Keisuke Toyama ,  Shiro Suzuki ,  Minoru Tsuji

IPC分类号： H04R5/00 ,  G10L21/00 ,  G10L19/00

CPC分类号： H04B1/667 ,  G10L19/0208

摘要： The present invention relates to an encoding device for saving the number of bits of codes. In step S11, the differential value between a normalization coefficient Bi to be encoded and a normalization coefficient Bi-1 for an encoding unit Ai-1 in a band adjacent to the lower side of an encoding unit Ai corresponding to the normalization coefficient Bi is computed. In step S12, reference is made to a table in which a differential value having a high frequency of occurrence is associated with a code having a small number of bits, and a code corresponding to the computed differential value is read. In step S13, it is determined whether or not all normalization coefficients B have been encoded. If it is determined that all normalization coefficients B have been encoded, in step S14, the code read in step S12 is output. The present invention is applicable to an audio recorder.

摘要翻译： 本发明涉及一种用于节省代码比特数的编码装置。 在步骤S11中，要编码的归一化系数B i i i与编码单元A i-1的归一化系数B i-1 <1的差分值< 计算与归一化系数B＆lt; I＆gt;对应的编码单元A的下侧的频带中的/ SUB。 在步骤S12中，参考具有高出现频率的差分值与具有少量位的代码相关联的表，并且读取与所计算出的差分值相对应的代码。 在步骤S13中，确定所有归一化系数B是否已被编码。 如果确定所有归一化系数B已被编码，则在步骤S14中，输出在步骤S12中读取的代码。 本发明可应用于音频记录器。

公开(公告)号：US07016502B2

公开(公告)日：2006-03-21

申请号：US10204298

申请日：2001-12-21

申请人： Keisuke Toyama ,  Shiro Suzuki ,  Minoru Tsuji

发明人： Keisuke Toyama ,  Shiro Suzuki ,  Minoru Tsuji

IPC分类号： H04R5/00

CPC分类号： H04B1/667 ,  G10L19/0208

摘要： The present invention relates to an encoding device for saving the number of bits of codes. In step S11, the differential value between a normalization coefficient Bi to be encoded and a normalization coefficient Bi-1 for an encoding unit Ai-1 in a band adjacent to the lower side of an encoding unit Ai corresponding to the normalization coefficient Bi is computed. In step S12, reference is made to a table in which a differential value having a high frequency of occurrence is associated with a code having a small number of bits, and a code corresponding to the computed differential value is read. In step S13, it is determined whether or not all normalization coefficients B have been encoded. If it is determined that all normalization coefficients B have been encoded, in step S14, the code read in step S12 is output. The present invention is applicable to an audio recorder.

摘要翻译： 本发明涉及一种用于节省代码比特数的编码装置。 在步骤S11中，要编码的归一化系数B i i i与编码单元A i-1的归一化系数B i-1/1之间的差分值 计算与对应于归一化系数B i i i的编码单元A的下侧相邻的频带中的。 在步骤S12中，参考具有高发生频率的差分值与具有少量位数的代码相关联的表，并且读取与所计算的差分值相对应的代码。 在步骤S13中，确定所有归一化系数B是否已被编码。 如果确定已经对所有归一化系数B进行编码，则在步骤S14中，输出在步骤S12中读取的代码。 本发明可应用于音频记录器。

公开(公告)号：US07912731B2

公开(公告)日：2011-03-22

申请号：US10484230

申请日：2003-05-12

申请人： Minoru Tsuji ,  Shiro Suzuki ,  Keisuke Toyama

发明人： Minoru Tsuji ,  Shiro Suzuki ,  Keisuke Toyama

IPC分类号： G10L19/14 ,  G10L21/04

CPC分类号： G10L19/008

摘要： A method for encoding sound signals on multiple channels includes extracting an arbitrary number of sine waves from each of the sound signals. The sine waves include at least a first sine wave, extracted from a first one of the channels and having first-channel information, and a second sine wave, extracted from a second one of the channels and having second-channel information. Using the first-channel information and one of the second-channel information and sine wave information corresponding to a predetermined sine wave, one of the second-channel information and the sine wave information corresponding to the predetermined sine wave is selected as a to-be-correlated object for encoding in a correlation with the first-channel information. The correlation includes a frequency-based absolute value of a difference between frequency information included in the first-channel information and frequency information included in the second-channel information and is used to encode the first- and second-channel information.

摘要翻译： 用于在多声道上对声音信号进行编码的方法包括从每个声音信号中提取任意数量的正弦波。 正弦波包括从第一通道提取并具有第一通道信息的至少第一正弦波和从第二通道提取并具有第二通道信息的第二正弦波。 使用第一声道信息和对应于预定正弦波的第二声道信息和正弦波信息之一，将与预定正弦波相对应的第二声道信息和正弦波信息中的一个选择为待定 相关对象，用于与第一信道信息相关联地进行编码。 相关性包括第一信道信息中包括的频率信息与包括在第二信道信息中的频率信息之间的差的基于频率的绝对值，并且用于对第一和第二信道信息进行编码。

公开(公告)号：US07447640B2

公开(公告)日：2008-11-04

申请号：US10362007

申请日：2002-06-11

申请人： Minoru Tsuji ,  Shiro Suzuki ,  Keisuke Toyama

发明人： Minoru Tsuji ,  Shiro Suzuki ,  Keisuke Toyama

IPC分类号： G10L21/00 ,  G10L21/02

CPC分类号： G10L19/032 ,  G10L19/087 ,  G10L19/20 ,  G10L25/93

摘要： In an acoustic signal encoding apparatus (100), a tonal noise verification unit (110) verifies whether the input acoustic time-domain signals are tonal or noisy. If the input acoustic time-domain signals are tonal, tonal component signals are extracted by a tonal component extraction unit (121), and tonal component parameters are normalized and quantized in a normalization/quantization unit (122). The residual time-domain signals, obtained on extracting the tonal component signals from the acoustic time-domain signals, are transformed by an orthogonal transforming unit (131) into the spectral information, which spectral information is normalized and quantized by a normalization/quantization unit (132). A code string generating unit (140) generates a code string from the quantized tonal component parameters and the quantized residual component spectral information.

摘要翻译： 在声信号编码装置（100）中，音调噪声验证单元（110）验证输入的声时域信号是音调还是噪声。 如果输入的声时域信号是音调的，则由音调分量提取单元（121）提取音调分量信号，并在归一化/量化单元（122）中对音调分量参数进行归一化和量化。 通过正交变换单元（131）将从声时域信号中提取音调分量信号获得的剩余时域信号变换为频谱信息，该频谱信息由归一化/量化单元进行归一化和量化 （132）。 代码串生成单元（140）从量化的音调分量参数和量化残差分量频谱信息生成代码串。

公开(公告)号：US20080082325A1

公开(公告)日：2008-04-03

申请号：US11951056

申请日：2007-12-05

申请人： Minoru Tsuji ,  Shiro Suzuki ,  Keisuke Toyama

发明人： Minoru Tsuji ,  Shiro Suzuki ,  Keisuke Toyama

IPC分类号： H03M7/30

CPC分类号： G10L19/008

摘要： A sound signal encoder for high efficiency encoding of sound signals from a plurality of channels is provided which includes a to-be-correlated object setter (52), to-be-correlated object selector (56) and a variable-length encoder (58). The to-be-correlated object setter (52) sets, on the basis of left-channel frequency information held in a left-channel frequency information holder (50) and right-channel frequency information held in a right-channel frequency information holder (51), index [i] indicating which ones of sine waves on the left channel are to be correlated with, namely, are to be subtracted from, sine waves on the right channel. The to-be-correlated object selector (56) selects a default value read from a storage unit (55) or index [i]-th amplitude information read from a left-channel amplitude information holder (53) as an object to be subtracted from the i-th amplitude information on the right channel according to the index [i]. The variable-length encoder (58) makes variable-length encoding of a difference resulted from subtraction of the left-channel amplitude information or default value as the to-be-correlated object from the amplitude information on the right channel.

摘要翻译： 提供了一种用于对来自多个通道的声音信号进行高效编码的声音信号编码器，其包括待相关对象设定器（52），待相关对象选择器（56）和可变长度编码器（58​​） ）。 要相关对象设定器（52）基于保持在左声道频率信息保持器（50）中的左声道频率信息和保持在右声道频率信息保持器（50）中的右声道频率信息， 51），指示左侧信道上哪个正弦波与哪个相关联的索引[i]，即从右声道上的正弦波中减去。 待关联对象选择器（56）从作为要减去的对象的从左信道幅度信息保持器（53）读取的从存储单元（55）读取的默认值或从第一信号振幅信息保持器（53）读取的索引[i] 根据索引[i]从右声道上的第i幅度信息。 可变长度编码器（58​​）对从右声道的振幅信息中减去左声道幅度信息或默认值作为要相关对象产生的差的可变长度编码。