公开(公告)号：US20110137643A1

公开(公告)日：2011-06-09

申请号：US13057454

申请日：2009-08-07

申请人： Tomofumi Yamanashi ,  Masahiro Oshikiri ,  Toshiyuki Morii ,  Hiroyuki Ehara

发明人： Tomofumi Yamanashi ,  Masahiro Oshikiri ,  Toshiyuki Morii ,  Hiroyuki Ehara

IPC分类号： G10L19/02

CPC分类号： G10L19/0204 ,  G10L19/0212 ,  G10L19/032 ,  G10L19/24 ,  G10L21/02

摘要： Disclosed is a spectral smoothing device with a structure whereby smoothing is performed after a nonlinear conversion has been performed for a spectrum calculated from an audio signal, and with which the amount of processing calculation is significantly reduced while maintaining excellent audio quality. With this spectral smoothing device, a sub band division unit (102) divides an input spectrum into multiple sub bands; a representative value calculation unit (103) calculates a representative value for each sub band using an arithmetic mean and a geometric mean; with respect to each representative value, a nonlinear conversion unit (104) performs a nonlinear conversion the characteristic of which is further emphasized as the value increases; and a smoothing unit (105) that smoothes the representative value which has undergone the nonlinear conversion for each sub band, at the frequency domain.

摘要翻译： 公开了一种光谱平滑装置，其具有在对从音频信号计算出的频谱进行非线性转换之后进行平滑处理，并且在保持优良音频质量的同时显着减少处理计算量的结构。 利用该频谱平滑装置，子带分割单元（102）将输入频谱划分成多个子频带; 代表值计算单元（103）使用算术平均值和几何平均值计算每个子带的代表值; 对于每个代表值，非线性转换单元（104）执行非线性转换，其特性随着值的增加被进一步强调; 以及平滑化单元（105），其以频域对每个子带进行了非线性转换的代表值进行平滑化。

公开(公告)号：US07949518B2

公开(公告)日：2011-05-24

申请号：US11587495

申请日：2005-04-22

申请人： Masahiro Oshikiri

发明人： Masahiro Oshikiri

IPC分类号： G06F15/00 ,  G10L11/00 ,  G10L11/06 ,  G10L19/14

CPC分类号： G10L19/24

摘要： A hierarchy encoding apparatus capable of calculating appropriate delay amounts and also capable of suppressing increase in the bit rate. In this apparatus, a first layer encoding part (101) encodes the input signal of the n-th frame to produce a first layer encoded code. A first layer decoding part (102) generates a first layer decoded signal from the first layer encoded code and applies it to a delay amount calculating part (103) and a second layer encoding part (105). The delay amount calculating part (103) uses the first layer decoded signal and input signal to calculate the delay amount to be added to the input signal, and applies the calculated delay amount to a delay part (104). The delay part (104) delays the input signal by the delay amount applied from the delay amount calculating part (103) and then applied it to a second layer encoding part (105). The second layer encoding part (105) uses the first layer decoded signal and the input signal from the delay part (104) for encoding.

摘要翻译： 一种能够计算适当的延迟量并且还能够抑制比特率的增加的层次编码装置。 在该装置中，第一层编码部分（101）对第n帧的输入信号进行编码以产生第一层编码。 第一层解码部分（102）从第一层编码代码生成第一层解码信号并将其应用于延迟量计算部分（103）和第二层编码部分（105）。 延迟量计算部（103）使用第一层解码信号和输入信号来计算要添加到输入信号的延迟量，并将所计算的延迟量应用于延迟部（104）。 延迟部分（104）将输入信号延迟由延迟量计算部分（103）施加的延迟量，然后将其施加到第二层编码部分（105）。 第二层编码部分（105）使用第一层解码信号和来自延迟部分（104）的输入信号进行编码。

公开(公告)号：US20100332221A1

公开(公告)日：2010-12-30

申请号：US12918575

申请日：2009-03-13

申请人： Tomofumi Yamanashi ,  Masahiro Oshikiri

发明人： Tomofumi Yamanashi ,  Masahiro Oshikiri

IPC分类号： G10L11/04 ,  G10L21/00

CPC分类号： G10L21/038 ,  G10L19/18 ,  G10L19/24

摘要： It is possible to improve quality of a decoding signal in a band spread for estimating a high band from a low band of a decoding signal. A first layer encoding unit (202) encodes a lower band portion below a predetermined frequency of an input signal so as to generate first layer encoded information. A first layer decoding unit (203) decodes the first layer encoded information so as to generate a first layer demodulated signal. A second layer encoding unit (206) divides a high band portion higher than a predetermined frequency of an input signal into a plurality of sub-bands and estimates each of the sub-bands from the input signal or the first layer decoded signal by using the estimation result of the sub-band adjacent to the lower band side so as to generate second encoded information including the estimation results of the sub-bands.

摘要翻译： 可以提高从解码信号的低频带估计高频带的频带扩展中的解码信号的质量。 第一层编码单元（202）对输入信号的预定频率以下的低频带部分进行编码，以产生第一层编码信息。 第一层解码单元（203）对第一层编码信息进行解码，以产生第一层解调信号。 第二层编码单元（206）将高于输入信号的预定频率的高频带部分划分成多个子带，并且通过使用第二层编码单元从输入信号或第一层解码信号估计每个子带 估计与低频带侧相邻的子带的结果，以产生包括子频带的估计结果的第二编码信息。

公开(公告)号：US20100228541A1

公开(公告)日：2010-09-09

申请号：US12095548

申请日：2006-11-29

申请人： Masahiro Oshikiri

发明人： Masahiro Oshikiri

IPC分类号： G10L19/02

CPC分类号： G10L19/0204

摘要： A subband coding apparatus carries out subband coding which prevents deterioration in coding performance and improves audio quality of decoded signals. The subband coding apparatus includes a low-band coding section (103) to code a low-band spectrum (S13). A low-band decoding section (106) decodes a low-band coded data (S14) and outputs a decoded low-band spectrum (S18) to a high-band coding section (107). A spectrum rearranging section (105) rearranges to make each frequency component of a high-band spectrum (S16) in reverse order on the frequency axis and outputs a modified high-band spectrum (S17) after rearranging to a high-band coding section (107). The high-band coding section (107) uses the decoded low-band spectrum (S18) output from the low-band decoding section (106) to code the modified high-band spectrum (S17) output from the spectrum rearranging section (105).

摘要翻译： 子带编码装置执行防止编码性能恶化并改善解码信号的音频质量的子带编码。 子带编码装置包括用于编码低频带频谱的低频编码部分（S13）（S13）。 低频段解码部（106）解码低频带编码数据（S14），将解码后的低频带（S18）输出到高频编码部（107）。 频谱重排部分（105）重新排列以在频率轴上以相反的顺序使高频谱（S16）的每个频率分量重新排列，并将其重新排列后的修改的高频带频谱（S17）输出到高频带编码部分 107）。 高频编码部（107）使用从低频带解码部（106）输出的解码低频带（S18）对频谱重排部（105）输出的修正后的高频谱（S17）进行编码， 。

公开(公告)号：US20100106496A1

公开(公告)日：2010-04-29

申请号：US12528877

申请日：2008-02-29

申请人： Toshiyuki Morii ,  Masahiro Oshikiri ,  Tomofumi Yamanashi

发明人： Toshiyuki Morii ,  Masahiro Oshikiri ,  Tomofumi Yamanashi

IPC分类号： G10L19/00

CPC分类号： G10L19/032 ,  G10L19/0212 ,  G10L19/10

摘要： Provided is an encoding device which can reduce the encoding distortion as compared to the conventional technique and can obtain a preferable sound quality for auditory sense. In the encoding device, a shape quantization unit (111) quantizes the shape of an input spectrum with a small number of pulse positions and polarities. The shape quantization unit (111) sets a pulse amplitude width to be searched later upon search of the pulse position to a value not greater than the pulse amplitude width which has been searched previously. A gain quantization unit (112) calculates a gain of a pulse searched by the shape quantization unit (111) for each of bands.

摘要翻译： 提供了一种与传统技术相比可以减少编码失真并且可以获得用于听觉的优选声音质量的编码装置。 在编码装置中，形状量化单元（111）以少量的脉冲位置和极性量化输入频谱的形状。 形状量化单元（111）将脉冲位置搜索后的脉冲幅度宽度设定为以前已经搜索到的脉冲幅度宽度以下的值。 增益量化单元（112）针对每个频带计算由形状量化单元（111）搜索的脉冲的增益。

公开(公告)号：US20100057446A1

公开(公告)日：2010-03-04

申请号：US12529219

申请日：2008-02-29

申请人： Toshiyuki Morii ,  Masahiro Oshikiri ,  Tomofumi Yamanashi

发明人： Toshiyuki Morii ,  Masahiro Oshikiri ,  Tomofumi Yamanashi

IPC分类号： G10L19/00 ,  G10L19/08

CPC分类号： G10L19/038 ,  G10L19/10 ,  G10L25/18

摘要： Provided is an encoding device which can obtain a sound quality preferable for auditory sense even if the number of information bits is small. The encoding device includes a shape quantization unit (111) having: a section search unit (121) which searches for a pulse for each of bands into which a predetermined search section is divided; and a whole search unit (122) which performs search for a pulse over the entire search section. The shape of an input spectrum is quantized by a small number of pulse positions and polarities. A gain quantization unit (112) calculates a gain of the pulse searched by the shape quantization unit (111) and quantizes the gain for each of the bands.

摘要翻译： 提供一种编码装置，即使信息比特数较少，也可以获得听觉优先的声音质量。 编码装置包括形状量化单元（111），具有：搜索单元（121），搜索预定搜索区被分割成的每个频带的脉冲; 以及整个搜索单元（122），其执行搜索整个搜索部分的脉冲。 输入频谱的形状由少量的脉冲位置和极性量化。 增益量化单元（112）计算由形状量化单元（111）搜索的脉冲的增益，并对每个频带的增益进行量化。

公开(公告)号：US20100017197A1

公开(公告)日：2010-01-21

申请号：US12447667

申请日：2007-11-01

申请人： Masahiro Oshikiri

发明人： Masahiro Oshikiri

IPC分类号： G10L19/02 ,  G10L21/00

CPC分类号： G10L19/0204

摘要： It is an object to disclose a voice coding device, etc. in which the deterioration of a voice quality of a decoded signal can be reduced in the case that low frequency domain components of a spectrum are used for coding high frequency domain components and that no low frequency domain components exist. In this voice coding device, a frequency domain transform unit (101) generates an input spectrum from an input voice signal, a first layer coding unit (102) codes a lower frequency domain portion of the input spectrum to generate first layer coded data, a first layer decoding unit (103) decodes the first layer coded data to generate a first layer decoded spectrum, a lower frequency domain component judging unit (104) judges if there are low frequency domain components of the first layer decoded spectrum, and a second decoding unit (105); codes high frequency domain components of the input spectrum to generate second layer coded data in the case that the low frequency domain components exist and codes high frequency domain components by using a predetermined signal disposed in the low frequency domain components to generate second layer coded data in the case that the low frequency domain components do not exist.

摘要翻译： 本发明的目的是公开一种语音编码装置等，其中在使用频谱的低频域分量用于编码高频域分量的情况下，可以降低解码信号的语音质量的劣化，并且不存在 存在低频域分量。 在该语音编码装置中，频域变换部（101）从输入声音信号生成输入频谱，第一层编码部（102）对输入频谱的低频域部进行编码，生成第一层编码数据， 第一层解码单元（103）对第一层编码数据进行解码以生成第一层解码频谱，低频域分量判定单元（104）判断是否存在第一层解码频谱的低频域分量，以及第二解码频谱 单位（105）; 编码输入频谱的高频域分量，以在存在低频域分量的情况下生成第二层编码数据，并通过使用布置在低频域分量中的预定信号对高频域分量进行编码，以产生第二层编码数据 低频域分量不存在的情况。

公开(公告)号：US20090248407A1

公开(公告)日：2009-10-01

申请号：US12295338

申请日：2007-03-29

申请人： Masahiro Oshikiri

发明人： Masahiro Oshikiri

IPC分类号： G10L21/02

CPC分类号： G10L19/26 ,  G10L21/0232

摘要： A sound encoder enabling prevention of deterioration of the sound quality of a reproduced signal even if the harmonic structure is broken in a part of the sound signal. The filter state position determining section (111) of the sound encoder judges the noise characteristic of the first-layer decoding spectrum and thereby determines the band of the first-layer decoding spectrum to be used to set the filter state. A filter state setting section (112) sets the first-layer decoding spectrum contained in the determined band out of the first-layer decoding spectrum as the filter state. A filtering section (113) performs filtering of the first-layer decoding spectrum according to the set filter state and the pitch coefficient and computes an estimate spectrum of the input spectrum. An optimal pitch coefficient is determined by a closed loop processing from the filtering section (113) through a search section (114) to a filter information setting section (115).

摘要翻译： 声音编码器，即使声音信号的一部分中的谐波结构被破坏，也能够防止再生信号的音质恶化。 声音编码器的滤波器状态位置确定部分（111）判断第一层解码频谱的噪声特性，从而确定要用于设置滤波器状态的第一层解码频谱的频带。 滤波器状态设置部分（112）将包含在第一层解码频谱中的确定频带中的第一层解码频谱设置为滤波状态。 滤波部（113）根据设定的滤波状态和音调系数对第一层解码频谱进行滤波，并计算输入频谱的估计频谱。 通过从滤波部分（113）通过搜索部分（114）到滤波器信息设置部分（115）的闭环处理来确定最佳的音调系数。

公开(公告)号：US5819213A

公开(公告)日：1998-10-06

申请号：US791741

申请日：1997-01-30

申请人： Masahiro Oshikiri ,  Tadashi Amada ,  Masami Akamine ,  Kimio Miseki

发明人： Masahiro Oshikiri ,  Tadashi Amada ,  Masami Akamine ,  Kimio Miseki

IPC分类号： G10L19/08 ,  G10L9/14

CPC分类号： G10L19/08

摘要： A speech encoding method and apparatus including analyzing, using a codebook expressing speech parameters within a predetermined search range, an input speech signal in an audibility weighting filter corresponding to a pitch period longer than the search range of the codebook, and searching, from the codebook, on the basis of the analysis result, a combination of speech parameters by which the distortion of the input speech signal is minimized, and encoding the combination. The apparatus uses an adaptive codebook of pitch and a noise codebook. The codebooks search a group formed by extracting vectors of predetermined length from one original code vector, while sequentially shifting position so that the vectors overlap each other. The search group is further restricted and another preselection is made before the final search. Search is based on inversely convoluted, orthogonally transformed vectors.

摘要翻译： 一种语音编码方法和装置，包括使用在预定搜索范围内表达语音参数的码本来分析与音码周期相对应的音频周期的声音加权滤波器中的输入语音信号，并且从码本中搜索 基于分析结果，将输入语音信号的失真最小化的语音参数的组合以及编码该组合。 该装置使用音调和噪声码本的自适应码本。 码本搜索通过从一个原始码矢量提取预定长度的矢量而形成的组，同时依次移位位置使得矢量彼此重叠。 搜索组进一步限制，并在最终搜索之前进行另一预选。 搜索基于逆卷积正交变换载体。

公开(公告)号：US09514757B2

公开(公告)日：2016-12-06

申请号：US13882750

申请日：2011-10-17

申请人： Masahiro Oshikiri ,  Hiroyuki Ehara

发明人： Masahiro Oshikiri ,  Hiroyuki Ehara

IPC分类号： H04R5/00 ,  G10L19/008 ,  G10L19/22

CPC分类号： G10L19/008 ,  G10L19/22

摘要： A stereo signal encoding device is provided that enables a lower bitrate without decreasing quality when applying an intermittent transmission technique to a stereo signal. A stereo encoding unit generates first stereo encoded data by encoding the stereo signal when the stereo signal of the current frame is an audio section. A stereo DTX encoding unit is a means for encoding the stereo signal when the stereo signal of the current frame is a non-audio section. The stereo DTX encoding unit generates second stereo encoded data by encoding each of a monaural signal spectral parameter that is a spectral parameter of a monaural signal generated using the first channel signal and the second channel signal, first channel signal information relating to the first channel signal, and second channel signal information relating to the second channel signal.

摘要翻译： 提供一种立体声信号编码装置，当将间歇传输技术应用于立体声信号时，能够实现较低的比特率而不降低质量。 当当前帧的立体声信号是音频部分时，立体声编码单元通过对立体声信号进行编码来产生第一立体声编码数据。 立体声DTX编码单元是当当前帧的立体声信号是非音频部分时对立体声信号进行编码的装置。 立体声DTX编码单元通过对作为使用第一声道信号和第二声道信号产生的单声道信号的频谱参数的单声道信号频谱参数进行编码，生成第二立体声编码数据，与第一声道信号有关的第一声道信号信号 ，以及与第二信道信号相关的第二信道信号信息。