公开(公告)号：WO2002099787A1

公开(公告)日：2002-12-12

申请号：PCT/US2002/017037

申请日：2002-05-31

Applicant: QUALCOMM INCORPORATED

Inventor： KANDHADAI, Ananthapadmanabhan ,  DEJACO, Andrew, P. ,  MANJUNATH, Sharath

IPC: G10L19/10

CPC classification number: G10L19/10 ,  G10L2019/0013

Abstract: Methods and apparatus for quickly selecting an optimal excitation waveform from a codebook are presented herein. In encoding schemes that use forward and backward pitch enhancement, storage and processor load is reduced by approximating a two-dimensional autocorrelation matrix with a one-dimensional autocorrelation vector. The approximation is possible when a cross-correlation element is configured to determine the autocorrelation matrix of an impulse response and a pulse energy determination element is configured to determine the energy of a pulse code vector that incorporates secondary pulse positions.

Abstract translation: 本文给出了从码本快速选择最佳激励波形的方法和装置。 在使用前向和后向间距增强的编码方案中，通过用一维自相关向量逼近二维自相关矩阵来减少存储和处理器负载。 当互相关元件被配置为确定脉冲响应的自相关矩阵并且脉冲能量确定元件被配置为确定包含次级脉冲位置的脉冲码矢量的能量时，近似是可能的。

公开(公告)号：WO1996004646A1

公开(公告)日：1996-02-15

申请号：PCT/US1995009780

申请日：1995-08-01

Applicant: QUALCOMM INCORPORATED

Inventor： QUALCOMM INCORPORATED ,  DEJACO, Andrew, P.

IPC: G10L05/00

CPC classification number: G10L19/18 ,  G10L19/002

Abstract: It is an objective of the present invention to provide an optimized method of selection of the encoding mode that provides rate efficient coding of input speech. A rate determination logic element (14) selects a rate at which to encode speech. The rate selected is based upon the target matching signal to noise ration computed by a TMSNR computation element (2), normalized autocorrelation computed by a NACF computation element (4), a zero crossings count determined by a zero crossings counter (6), the prediction gain differential computed by a PGD computation element (8) and the interframe energy differential computed by a frame energy differential element (10).

Abstract translation: 本发明的目的是提供一种提供输入语音的速率有效编码的编码模式的优化方法。 速率确定逻辑元件（14）选择编码语音的速率。 选择的速率基于由TMSNR计算元件（2）计算的目标匹配信号与噪声比，由NACF计算元件（4）计算的归一化自相关，由过零点计数器（6）确定的过零点数， 由PGD计算元件（8）计算的预测增益差分和由帧能量差分元件（10）计算的帧间能量差。

公开(公告)号：WO2003071523A1

公开(公告)日：2003-08-28

申请号：PCT/US2003/005232

申请日：2003-02-19

Applicant: QUALCOMM, INCORPORATED

Inventor： BI, Ning ,  DEJACO, Andrew, P.

IPC: G10L21/00

CPC classification number: G10L21/00 ,  G10L2021/0135

Abstract: 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.

Abstract translation: 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。

公开(公告)号：WO2002065459A2

公开(公告)日：2002-08-22

申请号：PCT/US2002/003728

申请日：2002-02-06

Applicant: QUALCOMM INCORPORATED

Inventor： CHOY, Eddie-Lun, Tik ,  ANANTHAPADMANABHAN, Arasanipalai, K. ,  DEJACO, Andrew, P.

IPC: G10L19/14

CPC classification number: G10L19/167

Abstract: 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.

Abstract translation: 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。

公开(公告)号：WO2002050504A3

公开(公告)日：2002-06-27

申请号：PCT/US2001/047761

申请日：2001-12-13

Applicant: QUALCOMM INCORPORATED

Inventor： GARUDADRI, Harinath ,  DEJACO, Andrew, P. ,  CHANG, Chienchung

IPC: G10L15/28

Abstract: A method and an accompanying apparatus provide for a distributed voice recognition (VR) capability in a remote device (201). Remote device (201) decides and controls what portions of the VR processing may take place at remote device (201) and what other portions may take place at a base station (202) in wireless communication with remote device (201).

公开(公告)号：WO1998005150A1

公开(公告)日：1998-02-05

申请号：PCT/US1997013593

申请日：1997-07-31

Applicant: QUALCOMM INCORPORATED

Inventor： QUALCOMM INCORPORATED ,  DEJACO, Andrew, P.

IPC: H04M01/60

CPC classification number: H04M1/6016 ,  H03G5/025 ,  H04M1/72519

Abstract: 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.

Abstract translation: 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。

公开(公告)号：WO2002099788A1

公开(公告)日：2002-12-12

申请号：PCT/US2002/017816

申请日：2002-06-05

Applicant: QUALCOMM INCORPORATED

Inventor： KANDHADAI, Ananthapadmanabhan ,  DEJACO, Andrew, P. ,  MANJUNATH, Sharath

IPC: G10L19/10

CPC classification number: G10L19/10 ,  G10L2019/0013

Abstract: Methods and apparatus for quickly selecting an optimal excitation waveform from a codebook are presented herein. To reduce the number of computations required to choose the optimal codebook vector, a subset of codevectors are selected based upon optimal pulse locations (425), wherein the subset of codevectors form a subcodebook. Rather than searching the entire codebook, only the entries of the subcodebook are searched (400).

Abstract translation: 本文给出了从码本快速选择最佳激励波形的方法和装置。 为了减少选择最佳码本矢量所需的计算次数，基于最佳脉冲位置（425）选择码矢量的子集，其中码矢量子集形成子码本。 搜索整个码本，而不是搜索子码本的条目（400）。

公开(公告)号：WO2002076123A1

公开(公告)日：2002-09-26

申请号：PCT/US2002/007696

申请日：2002-03-15

Applicant: QUALCOMM INCORPORATED

Inventor： DEJACO, Andrew, P. ,  EL-MALEH, Shaled

IPC: H04Q7/30

CPC classification number: H04W88/181

Abstract: A call setup procedure is presented to permit vocoder bypass, which will allow the transmission of wideband speech packets between wideband terminals over narrowband transmission constraints. In addition, methods and apparatus are presented that allow the conversion between a wideband tandem-free operation, a narrowband tandem-free operation, and a standard tandem operation.

Abstract translation: 提出了呼叫建立过程以允许声码器旁路，这将允许通过窄带传输约束在宽带终端之间传输宽带语音分组。 另外，提出了允许在宽带无串联操作，窄带无串联操作和标准串联操作之间进行转换的方法和装置。

公开(公告)号：WO1998005030A1

公开(公告)日：1998-02-05

申请号：PCT/US1997013594

申请日：1997-07-31

Applicant: QUALCOMM INCORPORATED

Inventor： QUALCOMM INCORPORATED ,  DEJACO, Andrew, P. ,  BI, Ning

IPC: G10L09/00

CPC classification number: G10L19/12 ,  G10L25/06

Abstract: Method and apparatus for selecting a code vector in an algebraic codebook wherein the analysis window for the coder is extended beyond the length of the target speech frame. An input signal is filtered by a perceptual weighting filter (76). Then, the filter is set to ring out for a number of samples equal to the length of the perceptual weighting filter (76), while a zero input vector is applied as input. By extending the analysis window, the two dimensional impulse response matrix can be stored as a one dimensional autocorrelation matrix in memory (60, 80), greatly saving on the computational complexity and memory required for the search.

Abstract translation: 用于选择代数码本中的码矢量的方法和装置，其中用于编码器的分析窗口被扩展到目标语音帧的长度之外。 输入信号由感知加权滤波器（76）滤波。 然后，将滤波器设置为响应等于感知加权滤波器（76）的长度的多个样本，同时施加零输入向量作为输入。 通过扩展分析窗口，可以将二维脉冲响应矩阵作为一维自相关矩阵存储在存储器中（60,80），大大节省了搜索所需的计算复杂度和存储空间。

公开(公告)号：WO1996023297A1

公开(公告)日：1996-08-01

申请号：PCT/US1996001166

申请日：1996-01-25

Applicant: QUALCOMM INCORPORATED

Inventor： QUALCOMM INCORPORATED ,  WEAVER, Lindsay, A., Jr. ,  LAM, S., Katherine ,  JACOBS, Paul, E. ,  GARDNER, Willima, R. ,  DEJACO, Andrew, P. ,  SIH, Gilbert, C.

IPC: G10L05/00

CPC classification number: H04W88/181 ,  G10L19/16

Abstract: 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.

Abstract translation: 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。