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    • 3. 发明申请
    • SPEECH CONVERTER UTILIZING PREPROGRAMMED VOICE PROFILES
    • 语音转换器使用预编译语音配置文件
    • WO2003071523A1
    • 2003-08-28
    • PCT/US2003/005232
    • 2003-02-19
    • QUALCOMM, INCORPORATED
    • BI, NingDEJACO, Andrew, P.
    • G10L21/00
    • G10L21/00G10L2021/0135
    • A speech processing system modifies various aspects of input speech according to a user-selected one of various preprogrammed voice fonts. Initially, the speech converter receives a formants signal representing an input speech signal and a pitch signal representing the input signal's fundamental frequency. One or both of the following may also be received: a voicing signal comprising an indication of whether the input speech signal is voiced, unvoiced, or mixed, and/or a gain signal representing the input speech signal's energy. The speech converter also receives user selection of one of multiple preprogrammed voice fonts, each specifying a manner of modifying one or more of the received signals (i.e., formants, voicing, pitch, gain). The speech converter modifies at least one of the formants, voicing, pitch, and/or gain signals as specified by the selectedvoice font.
    • 语音处理系统根据用户选择的各种预编程语音字体中的一种来修改输入语音的各个方面。 最初,语音转换器接收表示输入语音信号的共振峰信号和表示输入信号的基频的音调信号。 还可以接收以下中的一个或两个:包括输入语音信号是有声,无声或混合的指示的语音信号和/或表示输入语音信号的能量的增益信号。 语音转换器还接收多个预编程语音字体之一的用户选择,每个语音字体指定修改接收信号中的一个或多个(即,共振峰,发音,音高,增益)的方式。 语音转换器修改由所选择的发音字体指定的共振峰,发音,音调和/或增益信号中的至少一个。
    • 4. 发明申请
    • METHOD AND APPARATUS FOR REDUCING UNDESIRED PACKET GENERATION
    • 用于减少不需要的分组产生的方法和设备
    • WO2002065459A2
    • 2002-08-22
    • PCT/US2002/003728
    • 2002-02-06
    • QUALCOMM INCORPORATED
    • CHOY, Eddie-Lun, TikANANTHAPADMANABHAN, Arasanipalai, K.DEJACO, Andrew, P.
    • G10L19/14
    • G10L19/167
    • A method and apparatus for enhancing coding efficiency by reducing illegal or other undesirable packet generation while encoding a signal. The probiability of generating illegal or other undesirable packets while encoding a signal is reduced by first analyzing a history of the frequency of codebook values selected while quantizing speech parameters. Codebook entries are then reordered so that the index/indices that create illegal or ther undesirable packets contain the least frequently used entry/entries. Reordering multiple codebooks for various parameters further reduces the probability, that an illegal or ther undesirable packet will be created during signal encoding. The method and apparatus may be applied to reduce the probability of generating illegal null traffic channel data packets while encoding eight rate speech.
    • 通过在编码信号时减少非法或其他不希望的分组产生来增强编码效率的方法和设备。 通过首先分析在量化语音参数的同时选择的码本值的频率的历史来减少在编码信号时产生非法或其他不合需要的包的可能性。 然后将码本条目重新排序,以便创建非法或不期望的包的索引/索引包含最不经常使用的条目/条目。 对各种参数重新排序多个码本进一步降低了在信号编码期间产生非法或不期望的分组的概率。 该方法和装置可以用于减少在编码八种速率语音的同时产生非法空业务信道数据分组的概率。
    • 6. 发明申请
    • METHOD AND APPARATUS FOR APPLYING A USER SELECTED FREQUENCY RESPONSE PATTERN TO AUDIO SIGNALS PROVIDED TO A CELLULAR TELEPHONE SPEAKER
    • 将用户选择的频率响应模式应用于提供给蜂窝电话扬声器的音频信号的方法和装置
    • WO1998005150A1
    • 1998-02-05
    • PCT/US1997013593
    • 1997-07-31
    • QUALCOMM INCORPORATED
    • QUALCOMM INCORPORATEDDEJACO, Andrew, P.
    • H04M01/60
    • H04M1/6016H03G5/025H04M1/72519
    • The cellular telephone (10) includes an equalization filter (54) for adjusting the frequency response pattern of an audio signal provided to the speaker (24). The equalization filter (54) operates in response to user control to allow the user to adjust the frequency response pattern as desired. In one specific embodiment, the cellular telephone (10) includes an equalization filter table (56) for storing sets of audio frequency filter parameters, and the user merely selects one of the sets of filter parameters by pressing a corresponding button on a front control panel (11) of the cellular telephone (10). In other embodiments, the cellular telephone (10) includes an equalizer scroll bar allowing a large number of sets of filter parameters to be accessed. The equalization filter (54) and the filter table (56) may form part of a digital signal processing unit (42) also including vocoder encoders (50) and decoders (52). By providing an equalization filter (54), a cellular telephone (10) user may adjust the frequency response pattern of received signals to compensate, for example, for local noise or for hearing abnormalities to thereby allow the user to hear the other party to a telephone call more clearly. Even in the absence of any significant noise and even for a user having normal hearing, the user may still gain at least a perceived listening improvement.
    • 蜂窝电话(10)包括用于调节提供给扬声器(24)的音频信号的频率响应模式的均衡滤波器(54)。 均衡滤波器(54)响应于用户控制而操作,以允许用户根据需要调整频率响应模式。 在一个具体实施例中,蜂窝电话(10)包括用于存储音频滤波器参数集的均衡滤波器表(56),并且用户仅通过按下前控制面板上的对应按钮来选择滤波器参数之一 (11)。 在其他实施例中,蜂窝电话(10)包括允许访问大量滤波器参数的均衡器滚动条。 均衡滤波器(54)和滤波器表(56)可以形成也包括声码器编码器(50)和解码器(52)的数字信号处理单元(42)的一部分。 通过提供均衡滤波器(54),蜂窝电话(10)用户可以调整接收信号的频率响应模式,以补偿例如本地噪声或用于听到异常,从而允许用户听到另一方 电话更清晰。 即使没有任何明显的噪音,甚至对于具有正常听力的用户,用户仍然可以获得至少一个感知到的听觉改善。
    • 10. 发明申请
    • METHOD AND APPARATUS FOR DETECTION AND BYPASS OF TANDEM VOCODING
    • 用于TANDEM VOCODING的检测和旁路的方法和装置
    • WO1996023297A1
    • 1996-08-01
    • PCT/US1996001166
    • 1996-01-25
    • QUALCOMM INCORPORATED
    • QUALCOMM INCORPORATEDWEAVER, Lindsay, A., Jr.LAM, S., KatherineJACOBS, Paul, E.GARDNER, Willima, R.DEJACO, Andrew, P.SIH, Gilbert, C.
    • G10L05/00
    • H04W88/181G10L19/16
    • A first remote vocoder (15) receives analog voice (170) and produces packetized vocoder data (190) which is transmitted over a wireless link (20). A first local vocoder (35) receives the packetized vocoder data (190) from the wireless link (20). The first local vocoder (35) converts the packetized data (190) to a multibit PCM output (120). The first local vocoder (35) also adds a detection code to one of the least significant bits (LSB) of the PCM output (210). The first local vocoder (35) passes the PCM signal (210) to the PSTN (40). The first local vocoder (210) also receives PCM input (120) over the PSTN (40). The first local vocoder (35) constantly monitors the least significant bit of the PCM input (120) for a detection code indicating that a second local vocoder (55) is connected at the receiving end. If the first local vocoder (35) detects the detection code from the second local vocoder (55), it begins to substitute packetized data and a redundancy check for a second one of the LSB's of the outgoing PCM (210). The first local vocoder (35) also begins to monitor the second one of the LSB's of the incoming PCM (120). If the redundancy check indicates that valid packetized data has been received, the first local vocoder (35) stops converting the PCM output (120) into packetized data and simply passes the packetized data on the second one of the LSB's to the first remote vocoder (15) as packets (100). If at any time the redundancy check fails and the detection code is not detected, the first local vocoder (35) returns to converting the incoming PCM (190) to packetized data. In this way, the tandem vocoding arrangement is avoided.
    • 第一远程声码器(15)接收模拟语音(170)并产生通过无线链路(20)发送的分组化声码器数据(190)。 第一本地声码器(35)从无线链路(20)接收分组声码器数据(190)。 第一本地声码器(35)将分组化数据(190)转换成多位PCM输出(120)。 第一本地声码器(35)还将检测码添加到PCM输出(210)的最低有效位(LSB)之一。 第一本地声码器(35)将PCM信号(210)传递到PSTN(40)。 第一本地声码器(210)还通过PSTN(40)接收PCM输入(120)。 第一本地声码器(35)经常监视PCM输入(120)的最低有效位,用于检测码,指示在接收端连接第二本地声码器(55)。 如果第一本地声码器(35)检测到来自第二本地声码器(55)的检测码,则它开始替代分组数据和对输出PCM(210)的LSB中的第二个的冗余校验。 第一本地声码器(35)也开始监视输入PCM(120)的LSB的第二个。 如果冗余检查指示已经接收到有效的分组化数据,则第一本地声码器(35)停止将PCM输出(120)转换成分组化数据,并且将LSB的第二个上的分组化数据简单地传递给第一远程声码器( 15)作为分组(100)。 如果在任何时候冗余检查失败并且未检测到检测码,则第一本地声码器(35)返回以将输入的PCM(190)转换为分组数据。 以这种方式,避免了串联声音编码布置。