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1.发明授权CELP speech synthesizer with epoch-adaptive harmonic generator for pitch harmonics below voicing cutoff frequency 失效CELP语音合成器，具有时代自适应谐波发生器，用于音调谐波低于浊音截止频率
公开(公告)号：US06138092A
公开(公告)日：2000-10-24
申请号：US114661
申请日：1998-07-13
IPC分类号： G10L19/12
CPC分类号： G10L19/12
摘要： A speech coding system and associated method relies on a speech encoder (15) and a speech decoder (20). The speech decoder (20) includes an LPC synthesis filter (90), a Gaussian noise generator (80) for generating unvoiced excitation, an epoch-adaptive harmonic generator (70) for generating voiced excitation for pitch harmonics below voicing cutoff frequency, and an excitation summer (72) for summing the voiced and unvoiced excitation generated by the Gaussian noise generator (80) and the harmonic generator (70). The output of the excitation summer (72) is provided to the LPC synthesis filter (90) to generate synthesized speech. The system and method provides natural sounding synthesized speech at a low bit rate.
摘要翻译： 语音编码系统和相关方法依赖于语音编码器（15）和语音解码器（20）。 语音解码器（20）包括LPC合成滤波器（90），用于产生无声激励的高斯噪声发生器（80），用于产生低于声音截止频率的音调谐波的有声激励的时代自适应谐波发生器（70） 用于对由高斯噪声发生器（80）和谐波发生器（70）产生的有声和无声激励求和的激励加法器（72）。 激励加法器（72）的输出被提供给LPC合成滤波器（90）以产生合成语音。 该系统和方法以低比特率提供自然发声合成语音。
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公开(公告)号：US06393072B1
公开(公告)日：2002-05-21
申请号：US09340102
申请日：1999-06-25
申请人： John Anderson Fergus Ross , Nick Andrew Van Stralen , Mark Lewis Grabb , Steven Robert Koch , Richard Louis Zinser, Jr. , John Bailey Anderson
发明人： John Anderson Fergus Ross , Nick Andrew Van Stralen , Mark Lewis Grabb , Steven Robert Koch , Richard Louis Zinser, Jr. , John Bailey Anderson
IPC分类号： H04L2706
CPC分类号： H04L1/0047 , H04L25/061 , H04L25/067
摘要： A method for decoding (voice)data where the data is encoded by a finite-state data encoder (216) for transmission over a data channel (18), includes the step of procuring a table of joint statistics representing the probability of occurrence, in a frame of source data of each of the bits of the frame. The joint statistics may be determined ahead of time. The method includes the step of calculating intermediate gamma signals in response to the decoded data of the preceding frame and the joint statistics. The joint statistics source distribution signals represent the likelihood that, for a given logic level of the preceding bit, the “current” bit takes on a particular state; for uncorrelated bits, this value is 0.5. State probability signals and transition probability signals are generated from the gamma signals. The state probability and transition probability signals are processed to produce bit probability signals indicative of the probability that the current bit is in a given state, from which hard bit decisions can be made.
摘要翻译： 一种用于解码（语音）数据的方法，其中数据由有限状态数据编码器（216）编码，用于通过数据信道（18）进行传输，包括以下步骤：采购表示发生概率的联合统计表， 帧的每个比特的源数据帧。 联合统计可以提前确定。 该方法包括响应于前一帧的解码数据和联合统计来计算中间伽马信号的步骤。 联合统计源分布信号表示对于前一比特的给定逻辑电平，“当前”比特处于特定状态的可能性; 对于不相关的位，此值为0.5。 从伽马信号产生状态概率信号和转移概率信号。 处理状态概率和转移概率信号以产生指示当前比特处于给定状态的概率的比特概率信号，从中可以进行硬比特决定。
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3.发明授权Speech coding system and method including harmonic generator having an adaptive phase off-setter 失效包括具有自适应相位偏移器的谐波发生器的语音编码系统和方法
公开(公告)号：US6119082A
公开(公告)日：2000-09-12
申请号：US114663
申请日：1998-07-13
摘要： A speech coding system (10) and associated method rely on a speech encoder (15) and a speech decoder (20). The speech decoder (20) includes a Linear Predictive Coding (LPC) filter (90) having an input and an output. The LPC filter (90) provides synthesized speech at its output in response to voiced and unvoiced excitation provided at its input. A harmonic generator (70) for providing voiced excitation to the input of the LPC filter includes a pitch-related randomized adaptive phase off-setter. The adaptive phase off-setter off-sets the harmonic phases of the voiced excitation in accordance with the fundamental pitch frequency of respective frames of speech. The system and method thereby reduce perceived buzziness of synthesized speech provided at the output of the LPC filter.
摘要翻译： 语音编码系统（10）和相关联的方法依赖于语音编码器（15）和语音解码器（20）。 语音解码器（20）包括具有输入和输出的线性预测编码（LPC）滤波器（90）。 LPC滤波器（90）响应于在其输入处提供的有声和无声激励而在其输出端提供合成语音。 用于向LPC滤波器的输入端提供有声激励的谐波发生器（70）包括音调相关的随机自适应相位偏移器。 自适应相位偏移器根据各个语音帧的基本音调频率来偏移有声激励的谐波相位。 该系统和方法从而减少了在LPC滤波器的输出处提供的合成语音的感知嗡嗡声。
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4.发明授权Speech coding system and method including adaptive finite impulse response filter 失效语音编码系统及方法包括自适应有限脉冲响应滤波器
公开(公告)号：US6081776A
公开(公告)日：2000-06-27
申请号：US114658
申请日：1998-07-13
摘要： A speech coding system and associated method relies on a speech coder and a speech decoder. The speech coder provides speech spectrum information, including Line Spectral Frequency (LSF) information, for respective frames of speech. The LSF information is provided to the speech decoder. The speech decoder includes an LSF smoothing filter comprising an adaptive finite impulse response filter for averaging consecutive LSFs which are similar. Consecutive LSFs which are dissimilar are not averaged. The system and method causes a reduction in the shaky quality of synthesized speech without causing a corresponding decrease in the clarity of the synthesized speech.
摘要翻译： 语音编码系统和相关方法依赖于语音编码器和语音解码器。 语音编码器为各个语音帧提供语音频谱信息，包括线谱频率（LSF）信息。 LSF信息被提供给语音解码器。 语音解码器包括LSF平滑滤波器，其包括用于平均相似的连续LSF的自适应有限脉冲响应滤波器。 不相似的连续LSF不是平均的。 该系统和方法导致合成语音的抖动质量的降低，而不会导致合成语音的清晰度的相应降低。
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公开(公告)号：US6098036A
公开(公告)日：2000-08-01
申请号：US114664
申请日：1998-07-13
CPC分类号： G10L19/26 , G10L25/15 , H04R2225/43
摘要： A speech coding system and associated method rely on a speech encoder and a speech decoder. The speech decoder includes a Linear Predictive Coding (LPC) filter having an input and an output. The LPC filter provides synthesized speech at the output in response to voiced and unvoiced excitation provided at the input. A harmonic generator for providing voiced excitation to the input of the LPC filter includes a spectral formant enhancer for attenuating the amplitude of harmonics generate by the harmonic generator in spectral valleys between format peaks of respective frames of voiced speech. The system and method reduce perceived buzziness while increasing perceived spectral depth of synthesized speech at the output of the LPC filter.
摘要翻译： 语音编码系统和相关方法依赖于语音编码器和语音解码器。 语音解码器包括具有输入和输出的线性预测编码（LPC）滤波器。 LPC滤波器响应于在输入处提供的有声和无声激励而在输出端提供合成语音。 用于向LPC滤波器的输入端提供有声激励的谐波发生器包括频谱共振峰增强器，用于衰减由有声语音的各个帧的格峰之间的频谱谷中的谐波发生器产生的谐波振幅。 该系统和方法减少感知的嗡嗡声，同时在LPC滤波器的输出端增加合成语音的感知频谱深度。
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6.发明授权Speech coding system and method including voicing cut off frequency analyzer 失效语音编码系统及方法，包括语音切断频率分析仪
公开(公告)号：US6078880A
公开(公告)日：2000-06-20
申请号：US114660
申请日：1998-07-13
CPC分类号： G10L25/93 , G10L2025/937
摘要： A speech coding system and associated method relies on a speech encoder (15) and a speech decoder (20). The speech encoder (15) includes a voicing cut off frequency analyzer (60). Voicing cut off frequency analyzer (60) includes voicing cut off frequency estimator (61) and voicing cut off frequency quantizer (62). Voicing cut off frequency estimator (61) estimates a voicing cut off frequency value for respective samples of an input speech waveform (1). To accomplish this, voicing cut off frequency estimator (61) utilizes a bandpass filter to estimate a frequency above which a sample of speech is voiced and below which the sample of speech is unvoiced. Voicing cut off frequency quantizer (62) quantizes the estimated voicing cut off frequency value and provides, for respective samples, a voicing cut off frequency index signal (6) which may be stored or transmitted. Voicing cut off frequency index signal (6) may comprise as few as 1 bit, and in a preferred embodiment, as few as 3 bits.
摘要翻译： 语音编码系统和相关方法依赖于语音编码器（15）和语音解码器（20）。 语音编码器（15）包括语音切断频率分析器（60）。 声音切断频率分析器（60）包括发声切断频率估计器（61）和发声切断频率量化器（62）。 声音截止频率估计器（61）估计输入语音波形（1）的各样本的声音截止频率值。 为了实现这一点，发声截止频率估计器（61）利用带通滤波器来估计一个频率，在该频率上，语音样本被发声，并且低于该频率的语音样本是清音的。 声音截止频率量化器（62）量化估计的声音截止频率值，并为各个采样提供可存储或发送的发声切断频率指标信号（6）。 切断频率索引信号（6）可以包括少至1位，并且在优选实施例中，少至3位。
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7.发明授权LPC speech synthesis using harmonic excitation generator with phase modulator for voiced speech 失效LPC语音合成使用谐波激发发生器和相位调制器进行有声语音
公开(公告)号：US06067511A
公开(公告)日：2000-05-23
申请号：US114662
申请日：1998-07-13
摘要： A speech coding system (10) and associated method relies on a speech encoder (15) and a speech decoder (20). The speech decoder (20) includes a harmonic generator (70) which modulates the phase of each generated harmonic with a low frequency, low bandwidth signal to remove the buzzy quality of the speech and to produce natural sounding speech. The amplitude of the phase modulating signal is adjusted in accordance with the harmonic magnitude. For harmonics residing in a spectral valley the amplitude of the modulating signal is relatively large and for harmonics residing near spectral peaks, the amplitude of the modulation signal is relatively small.
摘要翻译： 语音编码系统（10）和相关联的方法依赖于语音编码器（15）和语音解码器（20）。 语音解码器（20）包括谐波发生器（70），其利用低频，低带宽信号调制每个产生的谐波的相位，以消除语音的嗡嗡声质量并产生自然的声音语音。 相位调制信号的幅度根据谐波幅度进行调整。 对于驻留在频谱谷中的谐波，调制信号的幅度相对较大，并且对于驻留在频谱峰附近的谐波，调制信号的幅度相对较小。
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公开(公告)号：US07421388B2
公开(公告)日：2008-09-02
申请号：US11451290
申请日：2006-06-12
IPC分类号： G10L19/00
CPC分类号： H04W88/181 , G10L19/00 , G10L19/0204 , G10L19/06 , G10L19/173 , G10L25/78
摘要： The system and method of the present invention comprises a compressed domain voice activity detector that detects the presence or absence of voice activity in a digital input signal. The method includes converting a digital input signal into parametric data. The parametric data is subsequently analyzed, and then compared against a background noise threshold to determine if voice activity is present.
摘要翻译： 本发明的系统和方法包括：压缩域语音活动检测器，其检测数字输入信号中是否存在语音活动。 该方法包括将数字输入信号转换为参数数据。 随后分析参数数据，然后与背景噪声阈值进行比较，以确定是否存在语音活动。
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公开(公告)号：US6094629A
公开(公告)日：2000-07-25
申请号：US114659
申请日：1998-07-13
IPC分类号： G10L19/06
CPC分类号： G10L19/07
摘要： A speech coding system and associated method relies on a speech encoder and a speech decoder. The encoder includes a spectral quantizer for computing line spectral frequencies (LSFs) for respective frames of speech and for quantizing the LSFs to obtain a minimum bit representation of a spectral envelope of each respective frame of speech. For even numbered frames of speech the LSFs are quantized using a vector quantization technique. For odd numbered frames of speech samples the LSFs are quantized using a dynamic bit allocation (DBA) method. The dynamic bit allocation method determines an interpolation factor for interpolating between the LSFs of the previous and next frames. According to the dynamic bit allocation method the most perceptually important LSFs are represented by relatively more bits, while the least perceptually important LSFs are represented by relatively fewer bits. The system and associated method thereby reduces an overall bit rate required to represent, transmit or store the speech samples.
摘要翻译： 语音编码系统和相关方法依赖于语音编码器和语音解码器。 编码器包括用于计算各个语音帧的频谱频率（LSF）的频谱量化器，并用于量化LSF以获得每个相应的语音帧的频谱包络的最小比特表示。 对于甚至编号的语音帧，使用矢量量化技术对LSF进行量化。 对于奇数帧的语音采样，使用动态位分配（DBA）方法对LSF进行量化。 动态位分配方法确定用于在前一帧和下一帧的LSF之间进行插值的内插因子。 根据动态位分配方法，最感知重要的LSF由相对较多的比特表示，而最不感知重要的LSF由相对较少的比特表示。 因此，系统和相关联的方法减少了表示，发送或存储语音样本所需的总比特率。
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