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    • 2. 发明授权
    • Method and apparatus for improving the voice quality of tandemed vocoders
    • 提高串联声码器语音质量的方法和装置
    • US5995923A
    • 1999-11-30
    • US883353
    • 1997-06-26
    • Paul MermelsteinRafi RabipourWilliam NavarroPaul Coverdale
    • Paul MermelsteinRafi RabipourWilliam NavarroPaul Coverdale
    • G10L19/02G10L19/14H03M7/30H04W88/18H03M9/00
    • G10L19/16H04W88/181
    • In recent years, the telecommunications industry has witnessed the proliferation of a variety of digital vocoders in order to meet bandwidth demands of different wireline and wireless communication systems. The rapid growth in the diversity of networks and the number of users of such networks is increasing the number of instances where two vocoders are placed in tandem to serve a single connection. Such arrangements of low bit-rate codecs can degrade the quality of the transmitted speech. To overcome this problem the invention provides a novel method and an apparatus for transmitting digitized voice signals in the wireless communications environment. The apparatus is capable of converting a compressed speech signal from one format to another format via an intermediate common format, thus avoiding the necessity to successively de-compress voice data to a PCM type digitization and then recompress the voice data.
    • 近年来,电讯业目睹了各种数字声码器的扩散,以满足不同有线和无线通讯系统的带宽需求。 网络的多样性的快速增长和这种网络的用户数量正在增加两个声码器串联放置以服务于单个连接的情况的数量。 低比特率编解码器的这种布置可能会降低发送语音的质量。 为了克服这个问题,本发明提供了一种用于在无线通信环境中发送数字化语音信号的新颖方法和装置。 该装置能够通过中间公共格式将压缩的语音信号从一种格式转换成另一格式,从而避免必须将语音数据连续解压缩到PCM型数字化,然后再重新压缩语音数据。
    • 3. 发明授权
    • Apparatus and method for coding speech signals by making use of an adaptive codebook
    • 通过使用自适应码本对语音信号进行编码的装置和方法
    • US06345255B1
    • 2002-02-05
    • US09621959
    • 2000-07-21
    • Paul Mermelstein
    • Paul Mermelstein
    • G10L1900
    • G10L19/18
    • An audio signal encoding device is provided including an input for receiving a sub-frame of an audio signal to be encoded, an adaptive codebook and a processing unit. The adaptive codebook stores at least one prior knowledge entry which includes a data element representative of characteristics of at least a portion of a previously generated audio signal sub-frame. The processing unit generates a set of parameters allowing for synthesization of the audio signal sub-frame received at the input on the basis of at least the sub-frame of the audio signal received at the input and the data element stored in the adaptive codebook. A corresponding decoding device for synthesizing an audio signal on the basis of a set of parameters is also provided.
    • 提供一种音频信号编码装置,包括用于接收要编码的音频信号的子帧的输入,自适应码本和处理单元。 自适应码本存储至少一个先验知识条目,其包括表示先前产生的音频信号子帧的至少一部分的特性的数据元素。 处理单元生成一组参数,其允许基于至少在输入端接收的音频信号的子帧和存储在自适应码本中的数据元素在输入处接收到的音频信号子帧进行合成。 还提供了一种用于基于一组参数来合成音频信号的相应解码装置。
    • 4. 发明授权
    • Speech recognition
    • 语音识别
    • US4956865A
    • 1990-09-11
    • US191824
    • 1988-05-02
    • Matthew LennigPaul MermelsteinVishwa N. Gupta
    • Matthew LennigPaul MermelsteinVishwa N. Gupta
    • G10L11/02G10L15/02
    • G10L25/87G10L15/02
    • In a speech recognizer, for recognizing unknown utterances in isolated-word speech or continuous speech, improved recognition accuracy is obtained by augmenting the usual spectral representation of the unknown utterance with a dynamic component. A corresponding dynamic component is provided in the templates with which the spectral representation of the utterance is compared. In preferred embodiments, the representation is mel-based cepstral and the dynamic components comprise vector differences between pairs of primary cepstra. Preferably the time interval between each pair is about 50 milliseconds. It is also preferable to compute a dynamic perceptual loudness component along with the dynamic parameters.
    • 在语音识别器中,为了识别孤立词语音或连续语音中的未知语音,通过用动态分量增加未知语音的常规频谱表示来获得改进的识别精度。 在模板中提供相应的动态分量,与之对比发音的频谱表示。 在优选实施例中,该表示是基于mel的倒频谱,并且动态分量包括主要cepstra对之间的矢量差异。 优选地,每对之间的时间间隔约为50毫秒。 还优选地计算动态感知响度分量以及动态参数。
    • 8. 发明授权
    • Apparatus and method for coding speech signals by making use of voice/unvoiced characteristics of the speech signals
    • 通过利用语音信号的语音/清音特征对语音信号进行编码的装置和方法
    • US06249758B1
    • 2001-06-19
    • US09107385
    • 1998-06-30
    • Paul Mermelstein
    • Paul Mermelstein
    • G10L1900
    • G10L19/18
    • An audio signal encoding device is provided comprising an input for receiving a sub-frame of an audio signal, a voiced audio signal synthesis stage, an unvoiced audio signal synthesis stage, and a processing unit. The voiced audio signal synthesis stage is operative for producing a first synthetic audio signal approximating the sub-frame of an audio signal received at the input on the basis of a first set of parameters. The unvoiced audio signal synthesis stage is operative for producing a second synthetic audio signal approximating the sub-frame of an audio signal received at the input on the basis of a second set of parameters. The processing unit is operative for releasing a set of parameters allowing to generate a selected one of the first synthetic audio signal and the second synthetic audio signal.
    • 提供了一种音频信号编码装置,包括用于接收音频信号的子帧的输入,有声音频信号合成级,无声音频信号合成级和处理单元。 有声音频信号合成级用于产生基于第一组参数来逼近在输入端接收的音频信号的子帧的第一合成音频信号。 无声音频信号合成级用于产生基于第二组参数近似在输入端接收的音频信号的子帧的第二合成音频信号。 处理单元用于释放允许生成第一合成音频信号和第二合成音频信号中所选择的一个参数的一组参数。