公开(公告)号：US07454335B2

公开(公告)日：2008-11-18

申请号：US11385553

申请日：2006-03-20

申请人： Yang Gao ,  Eyal Shlomot

发明人： Yang Gao ,  Eyal Shlomot

IPC分类号： G10L21/02 ,  G10L19/06

CPC分类号： H03G3/3089 ,  G10L21/045 ,  H03G3/341

摘要： There is provided a method of reducing effect of noise producing artifacts in silence areas of a speech signal for use by a speech decoding system. The method comprises obtaining a plurality of incoming samples of a speech subframe; summing an absolute value of an energy level for each of the plurality of incoming samples to generate a total input level (gain_in); smoothing the total input level to generate a smoothed level (Level_in_sm); determining that the speech subframe is in a silence area based on the total input level, the smoothed level and a spectral tilt parameter; defining a gain using k1*(Level_in_sm/1024)+(1−k1), where K1 is a function of the spectral tilt parameter; and modifying an energy level of the speech subframe using the gain.

摘要翻译： 提供了一种减少由语音解码系统使用的语音信号的静音区域中产生噪声的噪声的影响的方法。 该方法包括获得语音子帧的多个输入样本; 将多个输入样本中的每一个的能级的绝对值求和以产生总输入电平（gain_in）; 平滑总输入电平以产生平滑电平（Level_in_sm）; 基于总输入电平，平滑电平和频谱倾斜参数，确定语音子帧在静音区域中; 使用k1 *（Level_in_sm / 1024）+（1-k1）定义增益，其中K1是频谱倾斜参数的函数; 以及使用所述增益来修改所述语音子帧的能级。

公开(公告)号：US20070219791A1

公开(公告)日：2007-09-20

申请号：US11385553

申请日：2006-03-20

申请人： Yang Gao ,  Eyal Shlomot

发明人： Yang Gao ,  Eyal Shlomot

IPC分类号： G10L15/20

CPC分类号： H03G3/3089 ,  G10L21/045 ,  H03G3/341

摘要： There is provided a method of reducing effect of noise producing artifacts in silence areas of a speech signal for use by a speech decoding system. The method comprises obtaining a plurality of incoming samples of a speech subframe; summing an absolute value of an energy level for each of the plurality of incoming samples to generate a total input level (gain_in); smoothing the total input level to generate a smoothed level (Level_in_sm); determining that the speech subframe is in a silence area based on the total input level, the smoothed level and a spectral tilt parameter; defining a gain using k1*(Level_in_sm/1024)+(1-k1), where K1 is a function of the spectral tilt parameter; and modifying an energy level of the speech subframe using the gain.

摘要翻译： 提供了一种减少由语音解码系统使用的语音信号的静音区域中产生噪声的噪声的影响的方法。 该方法包括获得语音子帧的多个输入样本; 将多个输入样本中的每一个的能级的绝对值求和以产生总输入电平（gain_in）; 平滑总输入电平以产生平滑电平（Level_in_sm）; 基于总输入电平，平滑电平和频谱倾斜参数，确定语音子帧在静音区域中; 使用k1 *（Level_in_sm / 1024）+（1-k1）定义增益，其中K1是频谱倾斜参数的函数; 以及使用所述增益来修改所述语音子帧的能级。

公开(公告)号：US20070136052A1

公开(公告)日：2007-06-14

申请号：US11700481

申请日：2007-01-30

申请人： Yang Gao ,  Adil Benyassine ,  Jes Thyssen ,  Eyal Shlomot ,  Huan-yu Su

发明人： Yang Gao ,  Adil Benyassine ,  Jes Thyssen ,  Eyal Shlomot ,  Huan-yu Su

IPC分类号： G10L11/04

CPC分类号： G10L19/00 ,  G10L19/167 ,  G10L19/20 ,  G10L19/22 ,  G10L19/24 ,  G10L2019/0001 ,  H03G3/00

摘要： The invention improves the encoding and decoding of speech by focusing the encoding on the perceptually important characteristics of speech. The system analyzes selected features of an input speech signal, and first performing a common frame based speech coding of an input speech signal. The system then performs a speech coding based on either a first speech coding mode or a second speech coding mode. The selection of a mode is based on characteristics of the input speech signal. The first speech coding mode uses a first framing structure and the second speech coding mode uses a second framing structure.

公开(公告)号：US20060217976A1

公开(公告)日：2006-09-28

申请号：US11342130

申请日：2006-01-26

申请人： Yang Gao ,  Eyal Shlomot ,  Adil Benyassine

发明人： Yang Gao ,  Eyal Shlomot ,  Adil Benyassine

IPC分类号： G10L15/20

CPC分类号： G10L25/78 ,  G10L2025/786

摘要： There is provided a method of updating a noise state of a voice activity detector (VAD) for indicating an active voice mode and an inactive voice mode. The method comprises receiving an input signal having a plurality of frames, determining an elapsed time since the last update of the noise state, updating the noise state of the VAD if the elapsed time exceeds a predetermined time, determining an average minimum energy based on two or more of the plurality of frames, determining a current minimum energy based on a current frame of the plurality of frames, updating the noise state of the VAD if the average minimum energy is less than the current minimum energy, and updating the noise state of the VAD if the average minimum energy is greater than the current minimum energy plus a first predetermined value.

摘要翻译： 提供了一种更新用于指示主动语音模式和无效语音模式的语音活动检测器（VAD）的噪声状态的方法。 该方法包括接收具有多个帧的输入信号，确定自上次更新噪声状态以来经过的时间，如果经过时间超过预定时间，则更新VAD的噪声状态，基于二次确定平均最小能量 或更多个帧，基于多个帧的当前帧确定当前最小能量，如果平均最小能量小于当前最小能量，则更新VAD的噪声状态，并且更新噪声状态 VAD，如果平均最小能量大于当前最小能量加上第一预定值。

公开(公告)号：US06604070B1

公开(公告)日：2003-08-05

申请号：US09663734

申请日：2000-09-15

申请人： Yang Gao ,  Adil Benyassine ,  Jes Thyssen ,  Eyal Shlomot ,  Huan-yu Su

发明人： Yang Gao ,  Adil Benyassine ,  Jes Thyssen ,  Eyal Shlomot ,  Huan-yu Su

IPC分类号： G10L1912

CPC分类号： G10L19/00 ,  G10L19/167 ,  G10L19/24 ,  H03G3/00

摘要： A speech compression system capable of encoding a speech signal into a bitstream for subsequent decoding to generate synthesized speech is disclosed. The speech compression system optimizes the bandwidth consumed by the bitstream by balancing the desired average bit rate with the perceptual quality of the reconstructed speech. The speech compression system comprises a full-rate codec, a half-rate codec, a quarter-rate codec and an eighth-rate codec. The codecs are selectively activated based on a rate selection. In addition, the full and half-rate codecs are selectively activated based on a type classification. Each codec is selectively activated to encode and decode the speech signals at different bit rates emphasizing different aspects of the speech signal to enhance overall quality of the synthesized speech.

公开(公告)号：US06475245B2

公开(公告)日：2002-11-05

申请号：US09777424

申请日：2001-02-05

申请人： Allen Gersho ,  Eyal Shlomot ,  Vladimir Cuperman ,  Chunyan Li

发明人： Allen Gersho ,  Eyal Shlomot ,  Vladimir Cuperman ,  Chunyan Li

IPC分类号： G10L1106

CPC分类号： G10L19/18 ,  G10L19/02 ,  G10L19/09 ,  G10L19/10 ,  G10L2025/935

摘要： A method and apparatus for encoding speech for communication to a decoder for reproduction of the speech where the speech signal is classified into steady state voiced (harmonic), stationary unvoiced, and “transitory” or “transition” speech, and a particular type of coding scheme is used for each class. Harmonic coding is used for steady state voiced speech, “noise-like” coding is used for stationary unvoiced speech, and a special coding mode is used for transition speech, designed to capture the location, the structure, and the strength of the local time events that characterize the transition portions of the speech. The compression schemes can be applied to the speech signal or to the LP residual signal.

摘要翻译： 用于将语音编码的语音的方法和装置用于再现语音的语音的解码器，其中语音信号被分类为稳态浊音（谐波），固定无声和“暂时”或“转换”语音，以及特定类型的编码 方案用于每个类。 谐波编码用于稳态浊音，“静噪”编码用于固定无声语音，特殊编码方式用于转换语音，用于捕获本地时间的位置，结构和强度 表征演讲过渡部分的事件。 压缩方案可以应用于语音信号或LP残差信号。

公开(公告)号：US06256606B1

公开(公告)日：2001-07-03

申请号：US09200624

申请日：1998-11-30

申请人： Jes Thyssen ,  Huan-yu Su ,  Adil Benyassine ,  Eyal Shlomot

发明人： Jes Thyssen ,  Huan-yu Su ,  Adil Benyassine ,  Eyal Shlomot

IPC分类号： G10L2100

CPC分类号： G10L19/012

摘要： Silence description coding for multi-rate speech coding systems that employ discontinued transmission. Speech coding systems include multi-rate speech codecs having an encoder and a decoder. The silence description coding is performed in either the encoder or the decoder of the multi-rate speech codec. It may also be performed in a distributed manner wherein it is performed partially in the encoder and partially in the decoder. The silence description coding is performed on a speech signal having a substantially non-speech-like characteristic. Voice activity detection classifies the speech signal as being either substantially speech-like or substantially non-speech-like. The silence description coding is selected from a plurality of coding modes. In certain embodiments of the invention, the silence description coding is a source coding mode that operates at a bit rate that fits within a bit rate budget as determined by all of the available source coding modes within the plurality of coding modes. The silence description coding is also accompanied with signaling coding and channel coding of the speech signal. Error checking is performed using an unused portion of a bandwidth of the multi-rate speech codec's bit rate. This error checking involves majority voting in certain embodiments of the invention.

摘要翻译： 使用中断传输的多速率语音编码系统的静音描述编码。 语音编码系统包括具有编码器和解码器的多速率语音编解码器。 在多速率语音编解码器的编码器或解码器中执行静音描述编码。 它也可以以分布式方式执行，其中部分地在编码器中执行，部分地在解码器中执行。 对具有基本上非语音的特征的语音信号执行静音描述编码。 语音活动检测将语音信号分类为基本上是语音的或基本上非语音的。 从多种编码模式中选择静音描述编码。 在本发明的某些实施例中，静默描述编码是以适合在多个编码模式内的所有可用源编码模式所确定的比特率预算中的比特率操作的源编码模式。 静音描述编码也伴随着语音信号的信令编码和信道编码。 使用多速率语音编解码器的比特率的带宽的未使用部分来执行错误检查。 在本发明的某些实施例中，该错误检查涉及多数投票。

公开(公告)号：US5774849A

公开(公告)日：1998-06-30

申请号：US589509

申请日：1996-01-22

申请人： Adil Benyassine ,  Eyal Shlomot

发明人： Adil Benyassine ,  Eyal Shlomot

IPC分类号： G10L11/02 ,  G01L3/00 ,  G10L19/00 ,  H03M7/30 ,  G10L5/06

CPC分类号： G01L3/00

摘要： A method is disclosed for generating frame voicing decisions for an incoming speech signal having periods of active voice and non-active voice for a speech encoder in a speech communication system. The method first extracts a predetermined set of parameters from the incoming speech signal for each frame and then makes a frame voicing decision of the incoming speech signal for each frame according to a set of difference measures extracted from the predetermined set of parameters. The predetermined set of extracted parameters comprises a description of the spectrum of the incoming speech signal based on line spectral frequencies ("LSF"). Additional parameters may include full band energy, low band energy and zero crossing rate. The way to make a frame voicing decision of the incoming speech signal for each frame according to the set of difference measures is by finding a union of sub-spaces with each sub-space being described by a linear function of at least a pair of parameters from the predetermined set of parameters.

摘要翻译： 公开了一种用于在语音通信系统中为语音编码器的有效语音和非有效语音周期的输入语音信号生成帧发声决定的方法。 该方法首先从每个帧的输入语音信号中提取一组预定参数，然后根据从预定参数集中提取的一组差异度量对每帧进行入口语音信号的帧发声决定。 提取的参数的预定组包括基于线谱频率（“LSF”）的输入语音信号的频谱的描述。 附加参数可以包括全带能量，低频带能量和零交叉率。 根据差分测量集合对每个帧进行入站语音信号的帧发声决定的方法是通过找到每个子空间的并集，其中每个子空间由至少一对参数的线性函数描述 从预定的一组参数。

公开(公告)号：US20230084443A1

公开(公告)日：2023-03-16

申请号：US17898494

申请日：2022-08-30

申请人： Eyal Shlomot

发明人： Eyal Shlomot

IPC分类号： A63F9/24 ,  G06F3/0488 ,  A63H33/26

摘要： This invention describes an interactive computerized toy that provides light, audio and vibration entertaining patterns in response to touch stimuli from all directions and where the light patterns may be displayed in all directions. In particular, the present invention describes an interactive computerized toy in the shape of a cube with six faces with a unique versatility in providing endless programming options, in a fashion similar to loading and playing different games on the screens of handheld devices such as smartphones.

公开(公告)号：US08468017B2

公开(公告)日：2013-06-18

申请号：US12772190

申请日：2010-05-01

申请人： Eyal Shlomot ,  Jiliang Dai ,  Fuliang Yin ,  Xin Ma ,  Jun Zhang

发明人： Eyal Shlomot ,  Jiliang Dai ,  Fuliang Yin ,  Xin Ma ,  Jun Zhang

IPC分类号： G10L19/12 ,  G10L19/00

CPC分类号： G10L19/06 ,  G10L19/07 ,  G10L2019/0005 ,  G10L2019/0007 ,  H03M7/3082

摘要： The invention discloses a multi-stage quantization method, which includes the following steps: obtaining a reference codebook according to a previous stage codebook; obtaining a current stage codebook according to the reference codebook and a scaling factor; and quantizing an input vector by using the current stage codebook. The invention also discloses a multi-stage quantization device. With the invention, the current stage codebook may be obtained according to the previous stage codebook, by using the correlation between the current stage codebook and the previous stage codebook. As a result, it does not require an independent codebook space for the current stage codebook, which saves the storage space and improves the resource usage efficiency.

摘要翻译： 本发明公开了一种多级量化方法，包括以下步骤：获得根据前一级码本的参考码本; 根据参考码本和缩放因子获得当前阶段码本; 并通过使用当前阶段码本量化输入向量。 本发明还公开了一种多级量化装置。 利用本发明，可以通过使用当前阶段码本和前一级码本之间的相关性，根据前一级码本获得当前级码本。 因此，不需要当前级码本的独立码本空间，可以节省存储空间，提高资源使用效率。