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1. 发明授权

US06370500B1 Method and apparatus for non-speech activity reduction of a low bit rate digital voice message 失效
标题翻译：用于低比特率数字语音消息的非语音活动减少的方法和装置
公开(公告)号：US06370500B1
公开(公告)日：2002-04-09
申请号：US09409187
申请日：1999-09-30
申请人： Jian-Cheng Huang , Sunil Satyamurti , Floyd Simpson , Kenneth Finlon
发明人： Jian-Cheng Huang , Sunil Satyamurti , Floyd Simpson , Kenneth Finlon
IPC分类号： G10L1900
CPC分类号： G10L19/012 , G10L25/78
摘要： A technique is used in a speech encoder (107) that reduces non-speech activity of a low bit rate digital voice message. Speech model parameters that include quantized speech spectral parameter vectors are generated in a sequence of frames. A determination is made as to which frames of the sequence of frames are voiced frames and which frames are unvoiced frames. A consecutive sequence of frames of unvoiced frames is identified (2330) as an unvoiced burst when a length, NUV, of the consecutive sequence of frames exceeds a predetermined length, Ns. A non-speech activity portion of the unvoiced burst is identified (2335-2365) and removed.
摘要翻译：在语音编码器（107）中使用技术来减少低比特率数字语音消息的非语音活动。包括量化语音频谱参数矢量的语音模型参数在帧序列中生成。确定帧序列的哪些帧是浊音帧，哪些帧是清音帧。当连续帧序列的长度NUV超过预定长度Ns时，确定无声帧的连续序列（2330）为无声突发。确定清音突发的非语音活动部分（2335-2365）并移除。

2. 发明授权

US06772126B1 Method and apparatus for transferring low bit rate digital voice messages using incremental messages 失效
标题翻译：用于使用增量消息传送低比特率数字语音消息的方法和装置
公开(公告)号：US06772126B1
公开(公告)日：2004-08-03
申请号：US09410006
申请日：1999-09-30
申请人： Floyd Simpson , Jian-Cheng Huang , Sunil Satyamurti , Kenneth Finlon , Robert Schwendeman
发明人： Floyd Simpson , Jian-Cheng Huang , Sunil Satyamurti , Kenneth Finlon , Robert Schwendeman
IPC分类号： G10L2104
CPC分类号： G10L19/24
摘要： A system controller (106) is for transferring a low bit rate digital voice message. The system controller generates from an analog voice signal representing the voice message a set of speech model parameters, and generates a first derived set of speech model parameters from a first subset of the set of speech model parameters, the first derived set encoding the voice signal at a second voice quality and second vocoder rate that are less, respectively, than a first voice quality and vocoder rate. The system controller transmits (3610) the low bit rate-digital voice message comprising the first derived set of speech model parameters to a communication receiver (114). The communication receiver requests (3640) an incremental message when the quality of the voice message is unsatisfactory. The system controller generates and transmits (3555, 3650) an incremental message-and the communication receiver uses (3660) the incremental message to generate a higher quality voice message.
摘要翻译：系统控制器（106）用于传送低比特率数字话音消息。系统控制器从表示语音消息的模拟语音信号产生一组语音模型参数，并且从语音模型参数集合的第一子集生成第一导出的语音模型参数集，编码语音信号的第一导出集合分别具有比第一语音质量和声码率更小的第二语音质量和第二声码器速率。系统控制器将包括第一导出的语音模型参数集合的低比特率数字话音消息（3610）发送到通信接收机（114）。当语音消息的质量不令人满意时，通信接收器请求（3640）增量消息。系统控制器生成并发送增量消息（3555,3650），通信接收器使用增量消息（3660）生成更高质量的语音消息。

3. 发明授权

US06418405B1 Method and apparatus for dynamic segmentation of a low bit rate digital voice message 失效
标题翻译：低比特率数字语音消息的动态分割方法和装置
公开(公告)号：US06418405B1
公开(公告)日：2002-07-09
申请号：US09410140
申请日：1999-09-30
申请人： Sunil Satyamurti , Jian-Cheng Huang , Floyd Simpson , Kenneth Finlon
发明人： Sunil Satyamurti , Jian-Cheng Huang , Floyd Simpson , Kenneth Finlon
IPC分类号： G10L1900
CPC分类号： G10L15/04 , G10L19/0018
摘要： A system controller (106) includes a speech encoder (107) that dynamically segments frames of a low bit rate digital voice message. Speech model parameters have been generated in a sequence of frames. The speech model parameters include quantized speech spectral parameter vectors. The speech encoder selects (1820) a first quantized speech spectral parameter vector as a current anchor vector, selects (1820, 1830) a second quantized speech spectral parameter vector located a predetermined number of frames (LMAX) from the current anchor vector as a target speech parameter vector, and perturbs (1840) the target speech parameter vector to derive a plurality (K) of perturbed speech parameter vectors.
摘要翻译：系统控制器（106）包括动态地分段低比特率数字语音消息的帧的语音编码器（107）。语音模型参数已经在一系列帧中生成。语音模型参数包括量化语音频谱参数向量。语音编码器选择（1820）作为当前锚矢量的第一量化语音频谱参数矢量，选择（1820,1830）从位于当前锚矢量的预定数量帧（LMAX）的第二量化语音频谱参数矢量作为目标语音参数向量，并扰动（1840）目标语音参数向量，以导出多个（K）扰动语音参数向量。

4. 发明授权

US06496798B1 Method and apparatus for encoding and decoding frames of voice model parameters into a low bit rate digital voice message 失效
标题翻译：用于将语音模型参数帧编码和解码为低比特率数字话音消息的方法和装置
公开(公告)号：US06496798B1
公开(公告)日：2002-12-17
申请号：US09409183
申请日：1999-09-30
申请人： Jian-Cheng Huang , Floyd Simpson , Sunil Satyamurti , Oleg Andric , Kenneth Finlon
发明人： Jian-Cheng Huang , Floyd Simpson , Sunil Satyamurti , Oleg Andric , Kenneth Finlon
IPC分类号： G10L1900
CPC分类号： G10L19/167
摘要： A system controller (106) includes a speech encoder (107) that encodes a low bit rate digital voice message. The speech encoder sets values of words of a header of the encoded message. The values of the words define a quantity of frames in the voice message, N, and define a vocoder rate used for the encoded message. The speech encoder sets a state of each indicator in each frame status field of N frame status fields that are transmitted after the header of the encoded message. The speech encoder assembles N frame data fields, wherein each of the frame data fields comprises a set of data words. The N frame data fields follow the N frame status fields. Each set of data words conforms to at least one of the vocoder rate and the states of the indicators. A decoder (3310) decodes the encoded low bit rate digital message.
摘要翻译：系统控制器（106）包括编码低比特率数字语音消息的语音编码器（107）。语音编码器设置编码消息的报头的字的值。这些字的值定义了语音消息N中的帧数量，并且定义了用于编码消息的声码器速率。语音编码器在编码消息的头部之后发送的N个帧状态字段的每个帧状态字段中设置每个指示符的状态。语音编码器组合N个帧数据字段，其中每个帧数据字段包括一组数据字。 N帧数据字段遵循N帧状态字段。每组数据字符合至少一个声码器速率和指示器的状态。解码器（3310）对编码的低比特率数字消息进行解码。

5. 发明授权

US06418407B1 Method and apparatus for pitch determination of a low bit rate digital voice message 失效
标题翻译：用于音高确定低比特率数字语音消息的方法和装置
公开(公告)号：US06418407B1
公开(公告)日：2002-07-09
申请号：US09410007
申请日：1999-09-30
申请人： Jian-Cheng Huang , Floyd Simpson , Sunil Satyamurti , Kenneth Finlon
发明人： Jian-Cheng Huang , Floyd Simpson , Sunil Satyamurti , Kenneth Finlon
IPC分类号： G10L1104
CPC分类号： G10L25/90 , G10L19/09 , G10L19/10
摘要： A pitch determiner (931) of a system controller (106) that generates a smoothed pitch value for a current frame of a low bit rate voice message includes a pitch function generator (955) that generates a pitch detection function (PDF) for each frame of digital samples of a voice signal, a pitch candidate selector (960) that selects a future frame pitch candidate from a pitch detection function (PDF), and a pitch adjuster (978) that generates the smoothed pitch value. The pitch adjuster includes a subharmonic pitch corrector (965) that determines a future frame pitch value by performing pitch subharmonic correction of a future frame pitch candidate using a roughness factor of the frequency transformed window.
摘要翻译：产生低比特率语音消息的当前帧的平滑的音调值的系统控制器（106）的音调确定器（931）包括：产生用于每个帧的音调检测功能（PDF）的音调函数发生器（955）语音信号的数字样本，从音调检测功能（PDF）中选择未来的帧间距候选的音调候选选择器（960）以及产生平滑的音调值的音调调节器（978）。音调调节器包括次谐波音调校正器（965），其通过使用频率变换窗口的粗糙度因子执行对未来帧音调候选的音调次谐波校正来确定未来帧音调值。

6. 发明授权

US5703570A Method and apparatus for indicating undelivered messages in a communication device 失效
标题翻译：用于在通信设备中指示未传送消息的方法和装置
公开(公告)号：US5703570A
公开(公告)日：1997-12-30
申请号：US515845
申请日：1995-08-16
申请人： Paul Edward Gorday , Xuan-Khanh Tran Gorday , Sunil Satyamurti
发明人： Paul Edward Gorday , Xuan-Khanh Tran Gorday , Sunil Satyamurti
IPC分类号： H04W88/18 , G08B5/22
CPC分类号： H04W88/185
摘要： A method and apparatus is for use in a radio communication system (100) for delivering a message from a system controller (102) to a selective call device. In the system controller (102) the method includes the steps of generating a first message intended for the selective call device (106), changing a delivery state of the first message to undelivered after the first message has been transmitted at least once from the system controller (102), and storing the first message in the system controller (102) when the delivery state is changed to undelivered. In the selective call device the method includes the steps of determining that the first message is stored in the system controller (102) and presenting user with information which indicates that an undeliverable message is being stored by the system controller (102) in response to determining that the first message (102) is stored in the system controller (102).
摘要翻译：一种用于无线电通信系统（100）中的方法和装置，用于将消息从系统控制器（102）传送到选择性呼叫装置。在所述系统控制器（102）中，所述方法包括以下步骤：产生用于所述选择性呼叫设备（106）的第一消息，在从所述系统发送所述第一消息至少一次之后，将所述第一消息的传送状态改变为未传送控制器（102），并且当传送状态改变为未传送时，将第一消息存储在系统控制器（102）中。在选择性呼叫设备中，该方法包括以下步骤：响应于确定，确定第一消息被存储在系统控制器（102）中并且向用户呈现指示系统控制器（102）正在存储不可递送消息的信息第一消息（102）被存储在系统控制器（102）中。

7. 发明授权

US5699404A Apparatus for time-scaling in communication products 失效
标题翻译：通信产品时间缩放的装置
公开(公告)号：US5699404A
公开(公告)日：1997-12-16
申请号：US494662
申请日：1995-06-26
申请人： Sunil Satyamurti , Salvador Sibecas
发明人： Sunil Satyamurti , Salvador Sibecas
IPC分类号： G10L21/04 , H04W88/18 , H04M11/00 , G10L3/02
CPC分类号： H04W88/185 , G10L21/04
摘要： A communication system using voice compression includes a transmitter base station (113) and a selective call receiver (112). The transmitter base station (113) includes an input device (204) to receive an audio voice message which is stored in a memory (209). A processing device (208) digitizes the audio voice message to provide an input signal which is divided into a sequence of equivalent time frames, and differences in a short-term frequency spectrum between adjacent time frames is determined to derive distance measurements. A speed factor is computed as an average of the distance measurements, and the input signal is time-scales in accordance with the speed factor and then transmitted by a transmitter (102). The selective call receiver (112) includes a receiver (105) which receives the time-scaled signal, a processing device (115, 106) which demodulates and expands the time-scaled signal in accordance with the speed factor to provide a reconstructed signal which is amplified by an amplifier (108) into an audio signal.
摘要翻译：使用语音压缩的通信系统包括发射机基站（113）和选呼接收机（112）。发射机基站（113）包括用于接收存储在存储器（209）中的音频语音消息的输入设备（204）。处理装置（208）将音频语音消息数字化，以提供被分成等效时间帧序列的输入信号，并确定相邻时间帧之间的短期频谱中的差异以导出距离测量。速度因子被计算为距离测量的平均值，并且输入信号是根据速度因子的时间尺度，然后由发射机（102）发送。选择呼叫接收机（112）包括接收时间标度信号的接收机（105），根据速度因子解调和扩展时标信号的处理装置（115,106），以提供重构信号，被放大器（108）放大成音频信号。

8. 发明授权

US06707826B1 Method and apparatus for wireless bandwidth efficient multi-way calling 有权
标题翻译：无线带宽高效多路呼叫的方法和装置
公开(公告)号：US06707826B1
公开(公告)日：2004-03-16
申请号：US09531292
申请日：2000-03-20
申请人： Paul Edward Gorday , Sunil Satyamurti
发明人： Paul Edward Gorday , Sunil Satyamurti
IPC分类号： H04J316
CPC分类号： H04W88/181 , H04W4/00
摘要： A bandwidth efficient method of wireless communication among simultaneous multiple users includes the steps of monitoring (102) for voice activity from a plurality of sources (302 or 304) on a channel and detecting voice activity (107) among the plurality of sources during a predetermined time period. If voice activity is detected from only a first source, then code the voice activity from the first source and transmit a full-rate packet of data (109) to a wireless subscriber (50). If voice activity is detected from at least a first source and a second source amounting to N sources, then code such N sources using a 1/N-rate vocoder to obtain N different 1/N rate data packets (112) and combining (114) the N 1/N-rate data packets from at least the first source and the second source before transmitting (116) a full packet of data to the wireless subscriber.
摘要翻译：在同时多个用户之间进行无线通信的带宽有效的方法包括以下步骤：在信道上监测（102）来自多个源（302或304）的语音活动，并且在预定的时间段内检测多个源中的语音活动（107）时间段。如果仅从第一来源检测到语音活动，则从第一来源编码语音活动，并向无线用户（50）发送全速率数据包（109）。如果从至少源于N个源的第一源和第二源检测到语音活动，则使用1 / N速率声码器对这样的N个源进行编码以获得N个不同的1 / N速率数据分组（112）并组合（114 ）将来自至少第一源和第二源的N 1 / N速率数据分组发送（116）完整的数据分组到无线用户。

9. 发明授权

US6078793A High precision controlled gain voice processing system and method 失效
标题翻译：高精度可控增益语音处理系统及方法
公开(公告)号：US6078793A
公开(公告)日：2000-06-20
申请号：US965301
申请日：1997-11-06
申请人： Sunil Satyamurti , Thomas Victor D'Amico
发明人： Sunil Satyamurti , Thomas Victor D'Amico
IPC分类号： H01Q11/12 , H04B1/04
CPC分类号： H01Q11/12 , H04B1/04
摘要： A processing system (150) that generates a voice output signal having high precision controlled gain from a voice input signal by generating (225) a slow gain signal from the voice message by controlling a signal level gain of the voice signal using a setup gain and a first slow time constant, and then generating (230) a high precision controlled gain voice signal from the slow gain signal by controlling a signal level gain of the slow gain signal using at least one of a fast attack time constant and a slow release time constant.
摘要翻译：一种处理系统（150），通过使用建立增益控制语音信号的信号电平增益，通过从语音消息产生（225）慢增益信号，从语音输入信号生成具有高精度受控增益的语音输出信号;以及第一慢时间常数，然后通过使用快速攻击时间常数和缓慢释放时间中的至少一个来控制慢增益信号的信号电平增益，从慢增益信号产生（230）高精度可控增益话音信号不变。

10. 发明授权

US5828995A Method and apparatus for intelligible fast forward and reverse playback of time-scale compressed voice messages 失效
标题翻译：用于时间尺度压缩语音消息的可理解的快进和反向重放的方法和装置
公开(公告)号：US5828995A
公开(公告)日：1998-10-27
申请号：US953271
申请日：1997-10-17
申请人： Sunil Satyamurti , Paul Edward Gorday , Radha Khan
发明人： Sunil Satyamurti , Paul Edward Gorday , Radha Khan
IPC分类号： G10L19/00 , G10L21/04 , G10L3/02
CPC分类号： G10L21/04 , G10L19/00 , G10L19/24
摘要： A method and apparatus for providing intelligible fast forward and reverse playback of messages which include time-scale compressed speech using a time scale modification technique. A receiver (2604) receives (2902) a message including speech compressed at a predetermined compression rate and a message rate identifier which identifies the predetermined compression rate which are then stored in a memory by a processor (2610). The processor (2610) processes (2912) the stored message to time-scale expand the compressed speech at an expansion rate that is equal to the predetermined compression defined by the message rate identifier so as to produce a normal speech playback speed. The processor (2610) also processes (3000, 3100) the stored message to time-scale expand the compressed speech at an expansion rate that is lower than the predetermined compression rate defined by the message rate identifier, so as to produce a perceptibly increased speech playback speed.
摘要翻译：一种用于使用时间尺度修改技术提供包括时间尺度压缩语音的消息的可理解的快进和反向重放的方法和装置。接收器（2604）接收（2902）包含以预定压缩率压缩的语音的消息，以及识别由处理器（2610）存储在存储器中的预定压缩率的消息速率标识符。处理器（2610）处理（2912）存储的消息以按照等于由消息速率标识符定义的预定压缩率的扩展速率来扩展压缩语音，以便产生正常语音播放速度。处理器（2610）还处理（3000,3100）存储的消息以按照低于由消息速率标识符定义的预定压缩率的扩展速率对压缩语音进行时间尺度扩展，从而产生可察觉地增加的语音播放速度。

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