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11. 发明授权

US5890110A Variable dimension vector quantization 失效
标题翻译：可变维矢量量化
公开(公告)号：US5890110A
公开(公告)日：1999-03-30
申请号：US411436
申请日：1995-03-27
申请人： Allen Gersho , Amitava Das , Ajit Venkat Rao
发明人： Allen Gersho , Amitava Das , Ajit Venkat Rao
IPC分类号： G10L19/00 , G10L19/02 , G10L5/06 , H04B1/66
CPC分类号： G10L19/02
摘要： A variable dimension vector quantization method that uses a single "universal" codebook. The method can be given the interpretation of sampling full-dimensioned codevectors in the universal codebook and generating subcodevectors of the same dimension as input data subvector, which dimension may vary in time. A subcodevector is selected from the codebook to have minimum distortion between it and the input data subvector. The subcodevector with minimum distortion corresponds to the representative, full-dimensioned codevector in the codebook. The codebook is designed by inverse sampling of training subvectors to obtain full-dimension vectors, then iteratively clustering the training set until a stable centroid vector is obtained.
摘要翻译：使用单个“通用”码本的可变维度向量量化方法。该方法可以给出通用码本中采样全尺寸代码矢量的解释，并生成与输入数据子向量相同维度的子代码矢量，该维度可能随时间变化。从码本中选择一个子代码向量，使其与输入数据子向量之间具有最小的失真。具有最小失真的子码矢量对应于码本中的代表性的，全尺寸的码矢量。该码本是通过对训练子向量进行逆采样来设计的，以获得全维向量，然后迭代地聚类训练集，直到获得稳定的质心向量。

12. 发明授权

US5889891A Universal codebook vector quantization with constrained storage 失效
标题翻译：具有约束存储的通用码本矢量量化
公开(公告)号：US5889891A
公开(公告)日：1999-03-30
申请号：US561251
申请日：1995-11-21
申请人： Allen Gersho , Kenneth Rose , Sangeeta Ramakrishnan
发明人： Allen Gersho , Kenneth Rose , Sangeeta Ramakrishnan
IPC分类号： H04N7/18
CPC分类号： H04N19/94
摘要： Vectors associated with multiple sources are quantized with a codebook. A mapping function associates each source with a subset of codevectors in the codebook, each subset being defined prior to quantization. Each vector from a source is matched to a codevector in the subset corresponding to the source. The universal codebook is designed by first providing a codebook and a mapping function. Using the codebook, the mapping function is updated by identifying subsets that reduce distortion when vectors from a source are quantized using a corresponding subset.
摘要翻译：与多个源相关联的向量用码本量化。映射函数将每个源与码本中的码矢量子集相关联，每个子集在量化之前被定义。来自源的每个向量与对应于源的子集中的码矢匹配。通用码本是首先提供码本和映射功能而设计的。使用码本，通过识别使用相应子集量化来自源的向量的减少失真的子集来更新映射函数。

13. 发明授权

US07315815B1 LPC-harmonic vocoder with superframe structure 有权
标题翻译：具有超帧结构的LPC谐波声码器
公开(公告)号：US07315815B1
公开(公告)日：2008-01-01
申请号：US09401068
申请日：1999-09-22
申请人： Allen Gersho , Vladimir Cuperman , Tian Wang , Kazuhito Koishida
发明人： Allen Gersho , Vladimir Cuperman , Tian Wang , Kazuhito Koishida
IPC分类号： G10L19/12
CPC分类号： G10L19/173 , G10L19/087
摘要： An enhanced low-bit rate parametric voice coder that groups a number of frames from an underlying frame-based vocoder, such as MELP, into a superframe structure. Parameters are extracted from the group of underlying frames and quantized into the superframe which allows the bit rate of the underlying coding to be reduced without increasing the distortion. The speech data coded in the superframe structure can then be directly synthesized to speech or may be transcoded to a format so that an underlying frame-based vocoder performs the synthesis. The superframe structure includes additional error detection and correction data to reduce the distortion caused by the communication of bit errors.
摘要翻译：一种增强的低比特率参数语音编码器，其将来自基础帧的声码器（例如MELP）的多个帧分组成超帧结构。从底层帧中提取参数并量化到超帧中，这允许在不增加失真的情况下减少底层编码的比特率。然后，可以将在超帧结构中编码的语音数据直接合成为语音，或者将其转码为格式，使得基础的基于帧的声码器进行合成。超帧结构包括附加的错误检测和校正数据，以减少由位错误的通信引起的失真。

14. 发明授权

US06625226B1 Variable bit rate coder, and associated method, for a communication station operable in a communication system 有权
标题翻译：可变比特率编码器和相关联的方法，用于在通信系统中可操作的通信站
公开(公告)号：US06625226B1
公开(公告)日：2003-09-23
申请号：US09455012
申请日：1999-12-03
申请人： Allen Gersho , Vladimir Cuperman , Jan Linden , Ajit V. Rao , Sassan Ahmadi , Fenghua Liu , Ryan Heidari
发明人： Allen Gersho , Vladimir Cuperman , Jan Linden , Ajit V. Rao , Sassan Ahmadi , Fenghua Liu , Ryan Heidari
IPC分类号： H04B1500
CPC分类号： G10L19/24
摘要： A variable bit rate coder, and an associated method, for encoding a frame of speech, such as frames of data generated during operation of a communication station operable in a cellular communication system. Selection of the coding rate is made responsive to indicia of actual coding performance of a coder at more than one coding rate.
摘要翻译：一种用于编码语音帧的可变比特率编码器和相关联的方法，诸如在蜂窝通信系统中可操作的通信站操作期间生成的数据帧。编码速率的选择对编码器的编码速度的编码速度的编码性能进行响应。

15. 发明授权

US06233550B1 Method and apparatus for hybrid coding of speech at 4kbps 有权
标题翻译：用于4kbps语音混合编码的方法和装置
公开(公告)号：US06233550B1
公开(公告)日：2001-05-15
申请号：US09143265
申请日：1998-08-28
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，结构和强度表达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。

16. 发明授权

US5778335A Method and apparatus for efficient multiband celp wideband speech and music coding and decoding 失效
标题翻译：用于高效多频段宽带语音和音乐编码和解码的方法和装置
公开(公告)号：US5778335A
公开(公告)日：1998-07-07
申请号：US605509
申请日：1996-02-26
申请人： Anil Wamanrao Ubale , Allen Gersho
发明人： Anil Wamanrao Ubale , Allen Gersho
IPC分类号： G10H1/12 , G10H7/00 , G10L11/02 , G10L19/00 , G10L19/02 , G10L19/12 , G10L19/14 , G10L9/14 , H04B1/66
CPC分类号： G10L19/18 , G10H1/125 , G10H7/00 , G10L19/12 , G10H2210/046 , G10H2240/251 , G10H2250/581 , G10H2250/585 , G10L19/002 , G10L19/0204 , G10L19/10 , G10L2019/0005 , G10L2025/783
摘要： A method of digitally compressing speech and music by use of multiple band ("multiband") fixed excitations stored in codebooks. The use of multiband fixed excitations, along with a coupling method for interconnecting the excitation codebooks and adaptive codebooks and for generating the composite excitation signal, improve the long-term and short-term prediction, and the use of voice-music classification allows the coding structure to be adapted to the statistical character of the audio signal.
摘要翻译：通过使用存储在码本中的多个频带（“多频带”）固定激励数字压缩语音和音乐的方法。使用多频段固定激励以及用于互连激励码本和自适应码本并用于产生复合激励信号的耦合方法，改善了长期和短期预测，并且使用语音分类允许编码结构适应于音频信号的统计特性。

17. 发明授权

US3931596A Adaptive quantizer apparatus using training mode 失效
标题翻译：自适应量化仪器采用训练模式
公开(公告)号：US3931596A
公开(公告)日：1976-01-06
申请号：US510412
申请日：1974-09-30
申请人： Allen Gersho , David Joel Goodman
发明人： Allen Gersho , David Joel Goodman
IPC分类号： H03M3/04 , H03K13/22
CPC分类号： H03M3/042
摘要： Ideally, when a signal is quantized, the range of the quantizer should be matched to the power of the signal. Disclosed herein is adaptive quantizer apparatus which utilizes a particular quantizing scheme for those communications where the power of the signal to be quantized is unknown, but remains constant for the duration of a communication. The amplitude range of an adaptive quantizer is adjusted during an initial training period to a value appropriate to the power of the applied input signal. This is accomplished by successively altering the amplitude range by a multiplicative quantity that depends upon the amplitude of the quantizer output signal and upon the time elapsed since the onset of the training period.
摘要翻译：理想情况下，当量化信号时，量化器的范围应与信号的功率相匹配。这里公开了一种自适应量化器装置，其利用特定量化方案用于那些其中待量化信号的功率未知的通信，但在通信的持续时间内保持恒定。自适应量化器的振幅范围在初始训练周期内被调整到适合所施加的输入信号的功率的值。这是通过依赖于量化器输出信号的幅度和从训练周期开始以来经过的时间的乘法量连续地改变幅度范围来实现的。

18. 发明授权

US07124077B2 Frequency domain postfiltering for quality enhancement of coded speech 有权
公开(公告)号：US07124077B2
公开(公告)日：2006-10-17
申请号：US11045907
申请日：2005-01-28
申请人： Hong Wang , Vladimir Cuperman , Allen Gersho , Hosam A. Khalil
发明人： Hong Wang , Vladimir Cuperman , Allen Gersho , Hosam A. Khalil
IPC分类号： G10L19/04 , G10L21/02
CPC分类号： G10L19/26 , G10L21/0364
摘要： A method and system of performing postfiltering in the frequency domain to improve the quality of a speech signal, especially for synthesized speech resulting from codecs of low bit-rate, is provided. The method comprises LPC tilt computation and compensation methods and modules, a formant filter gain computation method and module, and an anti-aliasing method and module. The formant filter gain calculation employs an LPC representation, an all-pole modeling, a non-linear transformation and a phase computation. The LPC used for deriving the postfilter may be transmitted from an encoder or may be estimated from a synthesized or other speech signal in a decoder or receiver. The invention may be implemented in a linked decoder and encoder. A separate LPC evaluation unit that is responsible for processing and or deriving the LPC may be implemented within the invention.

19. 发明授权

US07010482B2 REW parametric vector quantization and dual-predictive SEW vector quantization for waveform interpolative coding 有权
公开(公告)号：US07010482B2
公开(公告)日：2006-03-07
申请号：US09811187
申请日：2001-03-16
申请人： Oded Gottesman , Allen Gersho
发明人： Oded Gottesman , Allen Gersho
IPC分类号： G10L19/14
CPC分类号： G10L19/097
摘要： An enhanced analysis-by-synthesis waveform interpolative speech coder able to operate at 2.8 kbps. Novel features include dual-predictive analysis-by-synthesis quantization of the slowly-evolving waveform, efficient parametrization of the rapidly-evolving waveform magnitude, and analysis-by-synthesis vector quantization of the rapidly evolving waveform parameter. Subjective quality tests indicate that it exceeds G.723.1 at 5.3 kbps, and of G.723.1 at 6.3 kbps.

20. 发明申请

US20050131696A1 Frequency domain postfiltering for quality enhancement of coded speech 有权
公开(公告)号：US20050131696A1
公开(公告)日：2005-06-16
申请号：US11045907
申请日：2005-01-28
申请人： Hong Wang , Vladimir Cuperman , Allen Gersho , Hosam Khalil
发明人： Hong Wang , Vladimir Cuperman , Allen Gersho , Hosam Khalil
IPC分类号： G10L19/02 , G10L11/00 , G10L19/14 , G10L21/02 , H03M7/30 , G10L13/06
CPC分类号： G10L19/26 , G10L21/0364
摘要： A method and system of performing postfiltering in the frequency domain to improve the quality of a speech signal, especially for synthesized speech resulting from codecs of low bit-rate, is provided. The method comprises LPC tilt computation and compensation methods and modules, a formant filter gain computation method and module, and an anti-aliasing method and module. The formant filter gain calculation employs an LPC representation, an all-pole modeling, a non-linear transformation and a phase computation. The LPC used for deriving the postfilter may be transmitted from an encoder or may be estimated from a synthesized or other speech signal in a decoder or receiver. The invention may be implemented in a linked decoder and encoder. A separate LPC evaluation unit that is responsible for processing and or deriving the LPC may be implemented within the invention.

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