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1. 发明申请

WO2011082047A4 HYBRID ARITHMETIC-COMBINATORIAL ENCODER 审中-公开
公开(公告)号：WO2011082047A4
公开(公告)日：2011-07-07
申请号：PCT/US2010/061714
申请日：2010-12-22
申请人： MOTOROLA MOBILITY, INC. , MITTAL, Udar , ASHLEY, James, P. , RAMABADRAN, Tenkasi, V.
发明人： MITTAL, Udar , ASHLEY, James, P. , RAMABADRAN, Tenkasi, V.
IPC分类号： H03M7/30 , H03M7/40
摘要： Hybrid range coding /combinatorial coding (FPC) encoders and decoders are provided. Encoding and decoding can be dynamically switched between range coding and combinatorial according to the ratio of ones to the ratio of bits in a partial remaining sequence in order to reduce the computational complexity of encoding and decoding.

2. 发明申请

WO2010021804A1 METHOD AND APPARATUS TO FACILITATE DETERMINING SIGNAL BOUNDING FREQUENCIES 审中-公开
标题翻译：方法和装置，以确定信号接合频率
公开(公告)号：WO2010021804A1
公开(公告)日：2010-02-25
申请号：PCT/US2009/051331
申请日：2009-07-22
申请人： MOTOROLA, INC. , RAMABADRAN, Tenkasi, V. , JASIUK, Mark, A.
发明人： RAMABADRAN, Tenkasi, V. , JASIUK, Mark, A.
IPC分类号： G10L21/02
CPC分类号： G10L21/0388
摘要： A signal processing platform (300) presents (101) a signal to be processed and identifies (102) signal portions with specific characteristics that are used (103) to automatically determine at least one bounding frequency that can be used to facilitate bandwidth extension for the signal. Identifying these signal portions can comprise identifying signal portions that exhibit at least a predetermined level of energy. The step of determining the bounding frequency can comprise computing a magnitude spectrum for each of the identified signal portions that can be used to determine a corresponding measure of flatness within a pass band as pertains to a corresponding normalized signal portion to thereby provide corresponding vetted signal portions. Determining the bounding frequency can then comprise accumulating the magnitude spectrum for these vetted signal portions and using the resultant accumulation to estimate a corresponding signal envelope. This signal envelope can then be used to determine the at least one bounding frequency.
摘要翻译：信号处理平台（300）呈现（101）要处理的信号，并识别（102）具有特定特征的信号部分（103），以自动确定可用于促进带宽扩展的至少一个边界频率信号。识别这些信号部分可以包括识别表现出至少预定水平能量的信号部分。确定边界频率的步骤可以包括计算每个识别的信号部分的幅度谱，所述信号部分可被用于确定与通过频带内的对应的对应的测量平均度，以对应于对应的归一化信号部分，从而提供相应的被检测的信号部分。确定边界频率可以包括累积这些被检测的信号部分的幅度谱，并且使用所得到的累积来估计相应的信号包络。然后可以使用该信号包络来确定至少一个边界频率。

3. 发明申请

WO2007067837A2 VOICE QUALITY CONTROL FOR HIGH QUALITY SPEECH RECONSTRUCTION 审中-公开
标题翻译：高品质语音重建的语音质量控制
公开(公告)号：WO2007067837A2
公开(公告)日：2007-06-14
申请号：PCT/US2006/060935
申请日：2006-11-15
申请人： MOTOROLA INC. , MA, Changxue, C. , CHENG, Yan, M. , NOWLAN, Steven, J. , RAMABADRAN, Tenkasi, V.
发明人： MA, Changxue, C. , CHENG, Yan, M. , NOWLAN, Steven, J. , RAMABADRAN, Tenkasi, V.
IPC分类号： G10L15/04
CPC分类号： G10L25/69 , G10L15/26
摘要： A method and apparatus are provided for reproducing a speech sequence of a user through a communication device of the user. The method includes the steps of detecting a speech sequence from the user through the communication device, recognizing a phoneme sequence within the detected speech sequence and forming a confidence level of each phoneme within the recognized phoneme sequence. The method further includes the steps of audibly reproducing the recognized phoneme sequence for the user through the communication device and gradually highlighting or degrading a voice quality of at least some phonemes of the recognized phoneme sequence based upon the formed confidence level of the at least some phonemes.
摘要翻译：提供了一种用于通过用户的通信设备再现用户的语音序列的方法和装置。该方法包括以下步骤：通过通信设备检测来自用户的语音序列，识别检测到的语音序列内的音素序列，并形成识别的音素序列内每个音素的置信度。该方法还包括以下步骤：通过通信设备可听地再现用户的识别音素序列，并且基于形成的至少一些音素的置信水平逐渐突出或降低所识别的音素序列的至少一些音素的语音质量。

4. 发明申请

WO2007047037A2 AN ADAPTIVE EQUALIZER FOR A CODED SPEECH SIGNAL 审中-公开
标题翻译：用于编码语音信号的自适应均衡器
公开(公告)号：WO2007047037A2
公开(公告)日：2007-04-26
申请号：PCT/US2006/037408
申请日：2006-09-26
申请人： MOTOROLA, INC. , JASIUK, Mark, A. , RAMABADRAN, Tenkasi, V.
发明人： JASIUK, Mark, A. , RAMABADRAN, Tenkasi, V.
IPC分类号： G10L19/00
CPC分类号： G10L19/26
摘要： A speech communication system provides a speech encoder that generates a set of coded parameters representative of the desired speech signal characteristics. The speech communication system also provides a speech decoder that receives the set of coded parameters to generate reconstructed speech. The speech decoder includes an equalizer that computes a matching set of parameters from the reconstructed speech generated by the speech decoder, undoes the set of characteristics corresponding to the computed set of parameters, and imposes the set of characteristics corresponding to the coded set of parameters, thereby producing equalized reconstructed speech.
摘要翻译：语音通信系统提供语音编码器，其生成表示所需语音信号特性的一组编码参数。语音通信系统还提供语音解码器，其接收编码参数集合以产生重构语音。语音解码器包括一个均衡器，该均衡器从语音解码器产生的重构语音中计算匹配的一组参数，取消对应于所计算的参数集合的一组特征，并施加与编码的参数集相对应的特征集合，从而产生均衡的重建语音。

5. 发明申请

WO2004090865A2 SYSTEM AND METHOD FOR COMBINED FREQUENCY-DOMAIN AND TIME-DOMAIN PITCH EXTRACTION FOR SPEECH SIGNALS 审中-公开
标题翻译：用于语音信号的组合频域和时域音调提取的系统和方法
公开(公告)号：WO2004090865A2
公开(公告)日：2004-10-21
申请号：PCT/US2004/010119
申请日：2004-03-31
申请人： MOTOROLA, INC. , INTERNATIONAL BUSINESS MACHINES CORPORATION , RAMABADRAN, Tenkasi, V. , SORIN, Alexander
发明人： RAMABADRAN, Tenkasi, V. , SORIN, Alexander
IPC分类号： G10L
CPC分类号： G10L25/90
摘要： A system, computer readable medium, and method for sampling a speech signal; dividing the sampled speech signal into overlapped frames; extracting first pitch information from a frame using frequency domain analysis; providing at least one pitch candidate, each being associated with a spectral score, from the first pitch information, each of the at least one pitch candidate representing a possible pitch estimate for the frame; extracting second pitch information from the frame using a time domain analysis; providing a correlation score for the at least one pitch candidate from the second pitch information; and selecting one of the at least one pitch candidate to represent the pitch estimate of the frame. The system, computer readable medium, and method are suitable for speech coding and for distributed speech recognition.
摘要翻译：用于采样语音信号的系统，计算机可读介质和方法; 将采样的语音信号分成重叠的帧; 使用频域分析从帧中提取第一音调信息; 从所述第一音高信息提供至少一个音高候选者，每个音高候选者与频谱分数相关联，所述至少一个音高候选者中的每一个表示所述帧的可能音高估计值; 使用时域分析从帧中提取第二音调信息; 从第二音高信息提供至少一个音高候选者的相关分数; 以及选择所述至少一个音调候选中的一个以表示所述帧的音调估计。该系统，计算机可读介质和方法适用于语音编码和分布式语音识别。

6. 发明申请

WO2004072948A2 CLASS QUANTIZATION FOR DISTRIBUTED SPEECH RECOGNITION 审中-公开
标题翻译：分类语音识别的类别量化
公开(公告)号：WO2004072948A2
公开(公告)日：2004-08-26
申请号：PCT/US2004/003419
申请日：2004-02-05
申请人： MOTOROLA, INC. , INTERNATIONAL BUSINESS MACHINES CORPORATION , RAMABADRAN, Tenkasi, V. , SORIN, Alexander
发明人： RAMABADRAN, Tenkasi, V. , SORIN, Alexander
IPC分类号： G10L
CPC分类号： G10L25/93 , G10L15/30 , G10L25/90 , G10L2025/935
摘要： A system, method and computer readable medium for quantizing class information and pitch information of audio is disclosed. The method on an information processing system includes receiving audio and capturing a frame of the audio. The method further includes determining a pitch of the frame and calculating a codeword representing the pitch of the frame, wherein a first codeword value indicates an indefinite pitch. The method further includes determining a class of the frame, wherein the class is any one of at least two classes indicating an indefinite pitch and at least one class indicating a definite pitch. The method further includes calculating a codeword representing the class of the frame, wherein the codeword length is the maximum of the minimum number of bits required to represent the at least two classes and the minimum number of bits required to represent the at least one class.
摘要翻译：公开了用于量化音频的类信息和音调信息的系统，方法和计算机可读介质。信息处理系统中的方法包括接收音频并捕获音频的帧。该方法还包括确定帧的音高并计算表示帧的音调的码字，其中第一码字值指示不确定音高。所述方法还包括确定所述帧的类别，其中所述类别是指示不确定音调的至少两个类别中的任何一个，以及指示确定音调的至少一个类别。所述方法还包括计算表示所述帧类别的码字，其中所述码字长度是表示所述至少两个类所需的最小比特数的最大值以及表示所述至少一个类所需的最小比特数。

7. 发明申请

WO2007062287A3 METHOD AND APPARATUS FOR DOUBLE-TALK DETECTION IN A HANDS-FREE COMMUNICATION SYSTEM 审中-公开
公开(公告)号：WO2007062287A3
公开(公告)日：2007-05-31
申请号：PCT/US2006/060656
申请日：2006-11-08
申请人： MOTOROLA INC. , CRUZ-ZENO, Edgardo, M. , PIKET, James, B. , RAMABADRAN, Tenkasi, V.
发明人： CRUZ-ZENO, Edgardo, M. , PIKET, James, B. , RAMABADRAN, Tenkasi, V.
IPC分类号： H04M9/08
摘要： An echo canceling circuit comprising a double talk detector, an upper band signal filter configured to pass only near-end upper band signals to the double talk detector and remove lower band signals, an adaptive filter circuit, a control circuit operatively coupled to the double talk detector and to the adaptive filter circuit, and a threshold estimator configured to iteratively calculate an upper adaptive decision threshold value and a lower adaptive decision threshold value. The double talk detector declares near-end speech to be present if an estimated power level of the upper band signals exceeds the upper adaptive decision threshold value, and declares the near-end speech to be absent if the estimated power level of the upper band signals falls below the lower adaptive decision threshold value for a predetermined number of iterative cycles.

8. 发明申请

WO2004072949A2 PITCH QUANTIZATION FOR DISTRIBUTED SPEECH RECOGNITION 审中-公开
标题翻译：用于分布式语音识别的定量定量
公开(公告)号：WO2004072949A2
公开(公告)日：2004-08-26
申请号：PCT/US2004/003425
申请日：2004-02-05
申请人： MOTOROLA, INC. , INTERNATIONAL BUSINESS MACHINES CORPORATION , RAMABADRAN, Tenkasi, V. , SORIN, Alexander
发明人： RAMABADRAN, Tenkasi, V. , SORIN, Alexander
IPC分类号： G10L
CPC分类号： G10L19/09 , G10L15/30
摘要： A system, method and computer readable medium for quantization pitch information of audio is disclosed. The method includes capturing audio representing a numbered frame of a plurality of numbered frames. The method further includes calculating a class of the frame, wherein a class is any one of a voiced or unvoiced class. If the frame is an even numbered frame and a voiced class, a codeword of a first length is calculated by absolutely quantizing the frame pitch. If the frame is an odd numbered frame and a voiced class and a reliable frame is available, a codeword of a second length is calculated by differentially quantizing the frame pitch. If there is no reliable frame available, a codeword of the second length is calculated by absolutely quantizing the frame pitch.
摘要翻译：公开了一种音频量化音调信息的系统，方法和计算机可读介质。该方法包括捕获表示多个编号帧的编号帧的音频。该方法还包括计算帧的类别，其中类是有声或无声类中的任何一个。如果帧是偶数帧和浊音类，则通过绝对量化帧间距来计算第一长度的码字。如果帧是奇数帧，并且有声类和可靠帧可用，则通过对帧间距进行差分量化来计算第二长度的码字。如果没有可靠的帧可用，则通过绝对量化帧间距来计算第二长度的码字。

9. 发明公开

EP2316118A1 METHOD AND APPARATUS TO FACILITATE DETERMINING SIGNAL BOUNDING FREQUENCIES 有权转让
标题翻译：方法定义信号极限频率
公开(公告)号：EP2316118A1
公开(公告)日：2011-05-04
申请号：EP09790695.2
申请日：2009-07-22
申请人： Motorola Mobility, Inc.
发明人： RAMABADRAN, Tenkasi, V. , JASIUK, Mark, A.
IPC分类号： G10L21/02
CPC分类号： G10L21/0388
摘要： A signal processing platform (300) presents (101) a signal to be processed and identifies (102) signal portions with specific characteristics that are used (103) to automatically determine at least one bounding frequency that can be used to facilitate bandwidth extension for the signal. Identifying these signal portions can comprise identifying signal portions that exhibit at least a predetermined level of energy. The step of determining the bounding frequency can comprise computing a magnitude spectrum for each of the identified signal portions that can be used to determine a corresponding measure of flatness within a pass band as pertains to a corresponding normalized signal portion to thereby provide corresponding vetted signal portions. Determining the bounding frequency can then comprise accumulating the magnitude spectrum for these vetted signal portions and using the resultant accumulation to estimate a corresponding signal envelope. This signal envelope can then be used to determine the at least one bounding frequency.

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