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1. 发明申请

WO1998020482A1 TIME-DOMAIN TIME/PITCH SCALING OF SPEECH OR AUDIO SIGNALS, WITH TRANSIENT HANDLING 审中-公开
标题翻译：语音时钟/音频信号的时域/时间调整，瞬时处理
公开(公告)号：WO1998020482A1
公开(公告)日：1998-05-14
申请号：PCT/US1997020310
申请日：1997-11-06
申请人： CREATIVE TECHNOLOGY LTD. , LAROCHE, Jean
发明人： CREATIVE TECHNOLOGY LTD.
IPC分类号： G10L03/02
CPC分类号： G10L21/01
摘要： Method and apparatus for time-scaling and/or pitch shifting by discarding and/or repeating segments of a signal (302). The signal (302) is stored as a series of samples on a memory (124) where it is readable by one or more read pointers (304, 306). Periodicity of segments of the signal (302) is determined by evaluating normalized cross-correlation over a range of possible periods. Transients are detected by monitoring changes in rms signal value. To achieve time compression or time stretching, a segment is skipped/discarded whenever a maximum time-discrepancy between the current output and an ideal output is reached or a high periodicity is detected, a jump of the optimal length would not make this time discrepancy too high, and no transient is present in the segment to be skipped/discarded.
摘要翻译：用于通过丢弃和/或重复信号的段来对时间缩放和/或音调移位的方法和装置（302）。信号（302）作为一系列样本存储在存储器（124）上，其中它可被一个或多个读指针（304,306）读取。信号（302）的段的周期性通过在可能的周期的范围内评估归一化的互相关来确定。通过监测有效值信号值的变化检测瞬态。为了实现时间压缩或时间延伸，每当达到当前输出和理想输出之间的最大时间差异或检测到高周期性时，跳过/丢弃一个段，最佳长度的跳跃也不会使此时间差异高，并且在要跳过/丢弃的段中不存在瞬态。

2. 发明申请

WO1997041548A1 METHOD AND APPARATUS FOR PEAK-TO-AVERAGE RATIO CONTROL IN AN AMPLITUDE MODULATION PAGING TRANSMITTER 审中-公开
标题翻译：用于调幅调制发射机中的峰值平均比例控制的方法和装置
公开(公告)号：WO1997041548A1
公开(公告)日：1997-11-06
申请号：PCT/US1997006949
申请日：1997-04-25
申请人： GLENAYRE ELECTRONICS, INC.
发明人： GLENAYRE ELECTRONICS, INC. , BHATTACHARYA, Bhaskar , KAVANAGH, Patricia, F. , HUSAIN, Mohammad, A.
IPC分类号： G10L03/02
CPC分类号： H03G7/007 , H04L5/06
摘要： A method and apparatus for reducing peak-to-average power ratio in a paging transmitter broadcasting and independent sideband amplitude modulation protocol is disclosed. A digital signal (xl(n), xu(n)), for example a digital voice signal, is provided to a control gain generator (103) which generates a digital gain sequence. The digital gain sequence is such that during peak values in the digital signal the gain is low and during low values in the digital signal the gain is high. The digital gain sequence is multiplied (104) with the digital signal to compress the dynamic range of the digital signal.
摘要翻译：公开了一种用于降低寻呼发射机广播和独立边带幅度调制协议中的峰值与平均功率比的方法和装置。将数字信号（x1（n），xu（n）），例如数字语音信号提供给产生数字增益序列的控制增益发生器（103）。数字增益序列使得在数字信号的峰值期间增益为低，而在数字信号的低值期间增益为高。数字增益序列与数字信号相乘（104），以压缩数字信号的动态范围。

3. 发明申请

WO1997031367A1 MULTI-STAGE SPEECH CODER WITH TRANSFORM CODING OF PREDICTION RESIDUAL SIGNALS WITH QUANTIZATION BY AUDITORY MODELS 审中-公开
标题翻译：具有通过审计模型量化的预测残留信号变换编码的多级语音编码器
公开(公告)号：WO1997031367A1
公开(公告)日：1997-08-28
申请号：PCT/US1997002898
申请日：1997-02-26
申请人： AT & T CORP. , CHEN, Juin-Hwey
发明人： AT & T CORP.
IPC分类号： G10L03/02
CPC分类号： G10L19/002 , G10L19/0212 , G10L19/12
摘要： A speech compression system called "Transform Predictive Coding" or TPC, provides encoding for 7 kHz band speech at 16 kHz sampling at a target bit-rate of 16 or 32 kb/s one or two bits per sample. The system uses short and long term prediction to remove redundancy. The prediction residual is transformed and coded in the frequency domain as shown on the figure by (110) after accepting time domain data from (60) and parameter input from (100), which corrects the spectrum for auditory perception. The TPC coder uses only open-loop quantization as shown by (70) and therefore has low complexity. The speech quality is transparent at 32 kb/s, is very good at 24 kb/s, and is acceptable at 16 kb/s.
摘要翻译：称为“变换预测编码”或TPC的语音压缩系统以16kHz采样的16kHz或32kb / s的目标比特率提供7kHz频带语音的编码，每个样本一个或两个比特。该系统使用短期和长期预测来消除冗余。在接收来自（60）的时域数据和（100）的参数输入后，预测残差在频域中如图所示变换和编码，如图110所示，其校正了听觉感知的频谱。 TPC编码器仅使用如（70）所示的开环量化，因此具有低复杂度。语音质量在32kb / s下是透明的，在24kb / s下是非常好的，并且在16kb / s下是可接受的。

4. 发明申请

WO1996028809A1 ARRANGEMENT AND METHOD RELATING TO SPEECH TRANSMISSION AND A TELECOMMUNICATIONS SYSTEM COMPRISING SUCH ARRANGEMENT 审中-公开
标题翻译：与语音传输相关的安排和方法以及包含这种安排的电信系统
公开(公告)号：WO1996028809A1
公开(公告)日：1996-09-19
申请号：PCT/SE1996000311
申请日：1996-03-11
申请人： TELEFONAKTIEBOLAGET LM ERICSSON , HALLKVIST, Per , GALYAS, Peter , JUNG, Stefan , ANDERSSON, Johan
发明人： TELEFONAKTIEBOLAGET LM ERICSSON
IPC分类号： G10L03/02
CPC分类号： G10L25/78 , G10L19/005
摘要： The present invention relates to an arrangement and a method in a speech transmission system particularly with a speech frame stucture. Means (VAD) are provided for detecting if a signal comprises speech information and detecting means are provided for detecting the presence of a bad or lost frame. When it has been detected that a speech frame has been corrupted or lost during transmission, it is replaced by a frame representing mainly background noise or a combination of at least one such frame and at least one correctly received speech frame.
摘要翻译：本发明涉及特别是具有语音帧结构的语音传输系统中的布置和方法。提供用于检测信号是否包括语音信息的装置（VAD），并且提供检测装置以检测是否存在不良或丢失的帧。当检测到语音帧在传输期间已被破坏或丢失时，它被主要代表背景噪声的帧或者至少一个这样的帧和至少一个正确接收到的语音帧的组合所代替。

5. 发明申请

WO1996011467A1 METHOD, DEVICE, AND SYSTEMS FOR DETERMINING A MASKING LEVEL FOR A SUBBAND IN A SUBBAND AUDIO ENCODER 审中-公开
标题翻译：用于确定子带音频编码器中的子带的掩蔽级别的方法，设备和系统
公开(公告)号：WO1996011467A1
公开(公告)日：1996-04-18
申请号：PCT/US1995009303
申请日：1995-07-24
申请人： MOTOROLA INC.
发明人： MOTOROLA INC. , FIOCCA, James, Leonard
IPC分类号： G10L03/02
CPC分类号： G10L19/0208 , G10L25/18
摘要： The masking level for a particular subband in a subband audio encoder is efficiently calculated by the present invention. The first step is calculating a signal level for each of the subbands based on an audio frame (102). Then, the masking level is calculated for the particular subband based on the signal levels, an offset function, and a weighting function (104).
摘要翻译：通过本发明有效地计算子带音频编码器中的特定子带的掩蔽电平。第一步是基于音频帧（102）计算每个子带的信号电平。然后，基于信号电平，偏移函数和加权函数来计算针对特定子带的屏蔽电平（104）。

6. 发明申请

WO1995022817A1 METHOD AND APPARATUS FOR MITIGATING AUDIO DEGRADATION IN A COMMUNICATION SYSTEM 审中-公开
标题翻译：在通信系统中减轻音频降级的方法和装置
公开(公告)号：WO1995022817A1
公开(公告)日：1995-08-24
申请号：PCT/US1994014751
申请日：1994-12-22
申请人： MOTOROLA INC.
发明人： MOTOROLA INC. , KOTZIN, Michael, D.
IPC分类号： G10L03/02
CPC分类号： G10L19/005 , G10L19/18
摘要： Audio degradation is minimized in scenarios where tandem coding occurs. One such scenario is in the environment of voice mail service. Characteristics of an audio information signal are determined, and the signal is classified (303) as to whether further coding (306) should be performed and, if so, which rate/type of coding should be performed. Characteristics of the audio signal which are determined are, inter alia, quality characteristics, rate of previous coding, type of previous coding and the source of previous coding of the audio information signal. The source of previous coding determined may further include, inter alia, an analog network, a digital network, a PSTN or a wireless communication system. Based on this information, the voice mail service will either choose not to further code the audio information signal or code the audio information signal with the best coding algorithm available.
摘要翻译：在发生串联编码的情况下，音频降级最小化。一种这样的情况是在语音邮件服务的环境中。确定音频信息信号的特征，并且对于是否应当执行进一步的编码（306）并且如果是的话应该执行哪个速率/编码类型来对信号进行分类（303）。确定的音频信号的特征尤其包括质量特性，先前编码的速率，先前编码的类型和音频信息信号的先前编码的来源。确定的先前编码的来源还可以包括模拟网络，数字网络，PSTN或无线通信系统。基于该信息，语音邮件服务将选择不进一步编码音频信息信号或使用可用的最佳编码算法对音频信息信号进行编码。

7. 发明申请

WO1995014297A1 DEVICE FOR PROCESSING A SOUND SIGNAL AND APPARATUS COMPRISING SUCH A DEVICE 审中-公开
标题翻译：用于处理声音信号的装置和包含这种装置的装置
公开(公告)号：WO1995014297A1
公开(公告)日：1995-05-26
申请号：PCT/FR1993001134
申请日：1993-11-18
申请人： LEFEVRE, Frank , GUILLARM, Gilles
IPC分类号： G10L03/02
CPC分类号： G10L21/04 , G09B19/04 , G09B21/009 , G10L2021/065
摘要： Device for processing a sound signal (10) comprising an event detector, capable of recognizing different sound phenomena and comprising a time structure analyzer (12) of each event for distinguishing continuous-type event sequences from explosion-type event sequences, a signal energy analyzer (11) analysing the signal for each event and a frequential structure analyzer (13) for distinguishing sequences in which the event has a harmonic structure composed of formants from random structure-type sequences. The device also comprises an event modifier (21) comprising a modification unit (21c) for modifying the time structure, a modification unit (21a) for altering the energy of at least one of the events and a modification unit (21b) for modifying the frequential structure of at least one of the events.
摘要翻译：用于处理包括事件检测器的声音信号（10）的装置，其能够识别不同的声音现象，并且包括用于区分连续型事件序列与爆炸型事件序列的每个事件的时间结构分析器（12），信号能量分析器（11）分析每个事件的信号，以及频率结构分析器（13），用于区分事件具有来自随机结构类型序列的共振峰组成的谐波结构的序列。该装置还包括一个事件修改器（21），包括用于修改时间结构的修改单元（21c），用于改变至少一个事件的能量的修改单元（21a）和修改单元至少有一个事件的频率结构。

8. 发明申请

WO1998003965A1 METHOD OF REDUCING VOICE SIGNAL INTERFERENCE 审中-公开
标题翻译：法语音信号的减少干扰
公开(公告)号：WO1998003965A1
公开(公告)日：1998-01-29
申请号：PCT/EP1997003482
申请日：1997-07-02
申请人： DAIMLER-BENZ AG , SCHRÖGMEIER, Peter , HAULICK, Tim , LINHARD, Klaus
发明人： DAIMLER-BENZ AG
IPC分类号： G10L03/02
CPC分类号： G10L21/0208 , G10L21/0264
摘要： The invention concerns a method of reducing voice signal interference using a noise-reducing method. According to the invention, a masking curve is determined both for the input signal and the output signal of the noise reduction. By comparing the signal portions exceeding the respective masking curve, newly audible portions can be detected in the form of interference in the output signal, according to the type of musical tone, and subsequently damped selectively.
摘要翻译：用于减少使用降噪方法的语音信号的干扰的方法，提出了以确定输入和根据掩蔽曲线上的噪声减小器的输出信号。通过在输出信号中的相应的掩蔽曲线超过信号成分的比较新的可听部分可被检测为对乐音的类型的干扰，并随后选择性地阻尼。

9. 发明申请

WO1997046999A1 NON-UNIFORM TIME SCALE MODIFICATION OF RECORDED AUDIO 审中-公开
标题翻译：记录音频的非均匀时间尺度修改
公开(公告)号：WO1997046999A1
公开(公告)日：1997-12-11
申请号：PCT/US1997007646
申请日：1997-05-12
申请人： INTERVAL RESEARCH CORPORATION
发明人： INTERVAL RESEARCH CORPORATION , COVELL, Michele , WITHGOTT, M., Margaret
IPC分类号： G10L03/02
CPC分类号： G10L21/04
摘要： To modify the temporal scale of recorded speech, relative stress and relative speaking rate terms are computed for individual sections, or frames, of the speech. These terms are then combined into a single value denoted as audio tension. For a nominal time-scale modification rate, the audio tension is employed to adjust the modification rate of the individual frames of speech in a non-uniform manner, relative to one another. With this approach, compressed speech can be reproduced at a relatively fast rate, while remaining intelligible to the listener.
摘要翻译：为了修改记录的语音的时间尺度，针对语音的单个部分或帧计算相对应力和相对语速率项。然后将这些术语组合成表示为音频张力的单个值。对于标称时标修改率，采用音频张力来相对于彼此以不均匀的方式调整各个语音帧的修改率。利用这种方法，可以以相对较快的速率再现压缩语音，同时对收听者仍然可理解。

10. 发明申请

WO1997028527A1 A NOISY SPEECH PARAMETER ENHANCEMENT METHOD AND APPARATUS 审中-公开
标题翻译：噪音演讲参数增强方法和设备
公开(公告)号：WO1997028527A1
公开(公告)日：1997-08-07
申请号：PCT/SE1997000124
申请日：1997-01-27
申请人： TELEFONAKTIEBOLAGET LM ERICSSON (publ)
发明人： TELEFONAKTIEBOLAGET LM ERICSSON (publ) , HÄNDEL, Peter , SÖRQVIST, Patrik
IPC分类号： G10L03/02
CPC分类号： G10L21/0208
摘要： Noisy speech parameters are enhanced by determining (22, 26) a background noise PSD estimate, determining (18) noisy speech parameters, determining (20) a noisy speech PSD estimate from the speech parameter, subtracting (30) a background noise PSD estimate from the noisy speech PSD estimate, and estimating (32) enhanced speech parameters from the enhanced speech PSD estimate.
摘要翻译：通过确定（22,26）背景噪声PSD估计，确定（18）噪声语音参数，从语音参数确定（20）噪声语音PSD估计，减去（30）背景噪声PSD估计，从而增强噪声语音参数嘈杂语音PSD估计，并从增强语音PSD估计估计（32）增强语音参数。

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