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    • 2. 发明申请
    • APPARATUS AND METHOD FOR ENCODING AUDIO DATA, AND APPARATUS AND METHOD FOR DECODING AUDIO DATA
    • 用于编码音频数据的装置和方法,以及用于解码音频数据的装置和方法
    • US20070043575A1
    • 2007-02-22
    • US11459513
    • 2006-07-24
    • Takashi OnumaYasuhiro ToguriHideaki WatanabeNoriaki FujitaHaifeng BaoManabu Uchino
    • Takashi OnumaYasuhiro ToguriHideaki WatanabeNoriaki FujitaHaifeng BaoManabu Uchino
    • G10L21/00
    • G10L19/008G10L19/0017G10L19/12G10L19/24
    • A method and apparatus for encoding audio data and a method and apparatus for decoding audio data, which can generate and decode, respectively, scalable lossless streams and which can shorten the time necessary to generate and decode lossless streams. A lossy-core encoder unit performs lossy compression on an input audio signal, generating a core stream. A simplified lossy-core decoding unit decodes only spectral signals of a specified band, e.g., a lower frequency band to generate a lossy decoded audio signal. A subtracter subtracts a lossy decoded audio signal from the input audio signal delayed to generate a residual signal. A rounding-off unit performs a process of rounding off the number of bits constituting the residual signal. A lossless-enhance encoder unit performs lossless compression on the residual signal to generate an enhanced stream. A stream-combining unit combines the core stream and the enhanced stream to generate a scalable lossless stream.
    • 用于对音频数据进行编码的方法和装置以及用于对音频数据进行解码的方法和装置,其可以分别生成和解码可伸缩的无损流,并且可以缩短生成和解码无损流所需的时间。 有损核心编码器单元对输入音频信号执行有损压缩,产生核心流。 简化的有损码解码单元仅解码指定频带(例如较低频带)的频谱信号,以产生有损解码音频信号。 减法器从延迟的输入音频信号中减去有损解码音频信号以产生残留信号。 舍入单元执行舍弃构成残差信号的位数的处理。 无损增强编码器单元对残余信号执行无损压缩以产生增强的流。 流合并单元组合核心流和增强流,以生成可伸缩的无损流。
    • 4. 发明授权
    • Decoding apparatus and method, encoding apparatus and method, and program
    • 解码装置和方法,编码装置和方法以及程序
    • US08972249B2
    • 2015-03-03
    • US13634658
    • 2011-03-15
    • Shiro SuzukiYuuki MatsumuraJun MatsumotoYuuji MaedaYasuhiro Toguri
    • Shiro SuzukiYuuki MatsumuraJun MatsumotoYuuji MaedaYasuhiro Toguri
    • G10L19/02G10L21/038
    • G10L21/038G10L19/0212
    • The present invention relates to a decoding apparatus, a decoding method, an encoding apparatus, an encoding method, and programs that can shorten the delay time caused by the band extension at the time of decoding, and restrain increases in resources on the decoding side.A higher frequency component generating unit (73) generates a pseudo higher frequency spectrum by using a lower frequency spectrum (SP-L) and a higher frequency envelope (ENV-H). A phase randomizing unit (74) randomizes the phase of the pseudo higher frequency spectrum, based on a random flag (RND). An inverse MDCT unit (75) denormalizes the lower frequency spectrum (SP-L) by using a lower frequency envelope (ENV-L), and combines the pseudo higher frequency spectrum supplied from the phase randomizing unit (74) with the denormalized lower frequency spectrum (SP-L). The combination result is used as the spectrum of the entire band. The present invention can be applied to a decoding apparatus that performs band extension decoding, for example.
    • 解码装置,解码方法,编码装置,编码方法和程序技术领域本发明涉及可以缩短在解码时由频带扩展引起的延迟时间的解码装置,解码方法,编码装置,编码方法和程序,并且抑制解码侧的资源增加。 较高频率分量产生单元(73)通过使用较低频谱(SP-L)和较高频率包络(ENV-H)产生伪较高频谱。 相位随机化单元(74)基于随机标记(RND)使伪较高频谱的相位随机化。 逆MDCT单元(75)通过使用较低频率包络(ENV-L)对较低频谱(SP-L)进行非归一化,并且将从相位随机化单元(74)提供的伪高频频谱与非归一化较低频率 光谱(SP-L)。 组合结果用作整个频带的频谱。 本发明可以应用于例如进行频带扩展解码的解码装置。
    • 5. 发明授权
    • Noise shaping for predictive audio coding apparatus
    • 预测音频编码装置的噪声整形
    • US08311816B2
    • 2012-11-13
    • US12639676
    • 2009-12-16
    • Yasuhiro ToguriJun Matsumoto
    • Yasuhiro ToguriJun Matsumoto
    • G10L21/02G10L19/00
    • H04B14/068H03M7/3046
    • An information coding apparatus includes a predictive signal generator that generates a predictive signal; a predictive residual signal generator that generates a predictive residual signal; a quantizer that quantizes a quantization input signal generated based on the predictive residual signal; a quantization error signal generator that generates a quantization error signal; a feedback signal generator that generates a feedback signal for controlling the frequency characteristic of the quantization noise after decoding based on the quantization error signal; and a quantization input signal generator that generates the quantization input signal. The feedback signal generator is configured by a pole-zero filter that includes a filter coefficient of an all-pole filter which is based on spectral envelope information estimated by the input audio signal, a parameter for adjusting a peak level in the frequency characteristic of the quantization noise caused by the all-pole filter, and the predictive filter coefficient.
    • 一种信息编码装置,包括产生预测信号的预测信号发生器; 产生预测残差信号的预测残差信号发生器; 量化器,其量化基于所述预测残差信号产生的量化输入信号; 产生量化误差信号的量化误差信号发生器; 反馈信号发生器,其基于量化误差信号生成用于控制解码之后的量化噪声的频率特性的反馈信号; 以及产生量化输入信号的量化输入信号发生器。 反馈信号发生器由极零滤波器构成,该极零滤波器包括基于由输入音频信号估计的频谱包络信息的全极滤波器的滤波器系数,用于调整频率特性中的峰值电平的参数 由全极滤波器引起的量化噪声和预测滤波器系数。
    • 7. 发明授权
    • Information retrieving method and apparatus
    • 信息检索方法和装置
    • US07747435B2
    • 2010-06-29
    • US12075872
    • 2008-03-15
    • Yasuhiro ToguriMasayuki Nishiguchi
    • Yasuhiro ToguriMasayuki Nishiguchi
    • G10L17/00
    • G10L17/00
    • A speaker of encoded speech data recorded in a semiconductor storage device in an IC recorder is to be retrieved easily. An information receiving unit 10 in a speaker retrieval apparatus 1 reads out the encoded speech data recorded in a semiconductor storage device 107 in an IC recorder 100. A speech decoding unit 12 decodes the encoded speech data. A speaker frequency detection unit 13 discriminates the speaker based on a feature of the speech waveform decoded to find the frequency of conversation (frequency of occurrence) of the speaker in a preset time interval. A speaker frequency graph displaying unit 14 displays the speaker frequency on a picture as a two-dimensional graph having time and the frequency as two axes. A speech reproducing unit 16 reads out the portion of the encoded speech data corresponding to a time position or a time range specified by a reproducing position input unit 15 based on this two-dimensional graph from the storage device 11 and decodes the read-out data to output the decoded data to a speech outputting unit 17.
    • 记录在IC记录器中的半导体存储装置中的编码语音数据的扬声器容易被检索。 扬声器检索装置1中的信息接收单元10读出记录在IC记录器100中的半导体存储装置107中的编码语音数据。语音解码单元12对编码的语音数据进行解码。 扬声器频率检测单元13基于解码的语音波形的特征来区分扬声器,以在预设的时间间隔内找到说话者的会话频率(出现频率)。 扬声器频率图显示单元14将作为具有时间和频率的二维图形的图像上的扬声器频率显示为两个轴。 语音再现单元16基于来自存储装置11的二维图形读出对应于由再现位置输入单元15指定的时间位置或时间范围的编码语音数据的部分,并对读出的数据进行解码 以将解码的数据输出到语音输出单元17。
    • 8. 发明授权
    • Apparatus for performing speaker identification and speaker searching in speech or sound image data, and method thereof
    • 用于在语音或声音图像数据中执行说话者识别和说话者搜索的装置及其方法
    • US07315819B2
    • 2008-01-01
    • US10201069
    • 2002-07-23
    • Yasuhiro ToguriMasayuki Nishiguchi
    • Yasuhiro ToguriMasayuki Nishiguchi
    • G10L17/00
    • G10L17/02G10L17/00G10L19/04
    • A process of identifying a speaker in coded speech data and a process of searching for the speaker are efficiently performed with fewer computations and with a smaller storage capacity. In an information search apparatus, an LSP decoding section extracts and decodes only LSP information from coded speech data which is read for each block. An LPC conversion section converts the LSP information into LPC information. A Cepstrum conversion section converts the obtained LPC information into an LPC Cepstrum which represents features of speech. A vector quantization section performs vector quantization on the LPC Cepstrum. A speaker identification section identifies a speaker on the basis of the result of the vector quantization. Furthermore, the identified speaker is compared with a search condition in a condition comparison section, and based on the result, the search result is output.
    • 通过较少的计算和较小的存储容量来有效地执行识别编码语音数据中的扬声器的处理和搜索扬声器的处理。 在信息搜索装置中,LSP解码部仅从针对每个块读取的编码语音数据提取并解码LSP信息。 LPC转换部将LSP信息转换为LPC信息。 倒谱转换部分将获得的LPC信息转换成表示语音特征的LPC倒频谱。 矢量量化部分对LPC倒频谱进行矢量量化。 扬声器识别部根据矢量量化的结果识别扬声器。 此外,将所识别的扬声器与条件比较部分中的搜索条件进行比较,并且基于该结果,输出搜索结果。