公开(公告)号：US20050071153A1

公开(公告)日：2005-03-31

申请号：US10498254

申请日：2002-12-13

Applicant: Mikko Tammi ,  Milan Jelinek ,  Claude LaFlamme ,  Vesa Ruoppila

Inventor： Mikko Tammi ,  Milan Jelinek ,  Claude LaFlamme ,  Vesa Ruoppila

IPC: G10L19/12 ,  G10L19/10

CPC classification number: G10L19/08

Abstract: For determining a long-term-prediction delay parameter characterizing a long term prediction in a technique using signal modification for digitally encoding a sound signal, the sound signal is divided into a series of successive frames, a feature of the sound signal is located in a previous frame, a corresponding feature of the sound signal is located in a current frame, and the long-term-prediction delay parameter is determined for the current frame while mapping, with the long term prediction, the signal feature of the previous frame with the corresponding signal feature of the current frame. In a signal modification method for implementation into a technique for digitally encoding a sound signal, the sound signal is divided into a series of successive frames, each frame of the sound signal is partitioned into a plurality of signal segments, and at least a part of the signal segments of the frame are warped while constraining the warped signal segments inside the frame. For searching pitch pulses in a sound signal, a residual signal is produced by filtering the sound signal through a linear prediction analysis filter, a weighted sound signal is produced by processing the sound signal through a weighting filter, the weighted sound signal being indicative of signal periodicity, a synthesized weighted sound signal is produced by filtering a synthesized speech signal produced during a last subframe of a previous frame of the sound signal through the weighting filter, a last pitch pulse of the sound signal of the previous frame is located from the residual signal, a pitch pulse prototype of given length is extracted around the position of the last pitch pulse of the sound signal of the previous frame using the synthesized weighted sound signal, and the pitch pulses are located in a current frame using the pitch pulse prototype.

Abstract translation: 为了确定在使用用于数字编码声音信号的信号修改的技术中表征长期预测的长期预测延迟参数，声音信号被分成一系列连续的帧，声音信号的特征位于 前一帧，声音信号的对应特征位于当前帧中，并且为当前帧确定长期预测延迟参数，同时长期预测将前一帧的信号特征与 当前帧的相应信号特征。 在用于实现用于对声音信号进行数字编码的技术的信号修改方法中，声音信号被分成一系列连续的帧，声音信号的每个帧被划分为多个信号段，并且至少一部分 框架的信号段扭曲，同时约束框架内的翘曲的信号段。 为了在声音信号中搜索音调脉冲，通过线性预测分析滤波器对声音信号进行滤波来产生残留信号，通过加权滤波器处理声音信号产生加权声音信号，加权声音信号表示信号 通过对通过加权滤波器的声音信号的先前帧的最后一个子帧产生的合成语音信号进行滤波，产生合成加权声音信号，将前一帧的声音信号的最后音调脉冲从剩余的位置 信号，使用合成的加权声音信号在前一帧的声音信号的最后音调脉冲的位置周围提取给定长度的音调脉冲原型，并且使用音调脉冲原型将音调脉冲位于当前帧中。

公开(公告)号：US08121833B2

公开(公告)日：2012-02-21

申请号：US12288592

申请日：2008-10-21

Applicant: Mikko Tammi ,  Milan Jelinek ,  Claude LaFlamme ,  Vesa Ruoppila

Inventor： Mikko Tammi ,  Milan Jelinek ,  Claude LaFlamme ,  Vesa Ruoppila

IPC: G10L19/00

CPC classification number: G10L19/08

Abstract: The exemplary embodiments of the invention provide at least a method and an apparatus to perform operations including dividing a sound signal into a series of successive frames, dividing each frame into a number of subframes, producing a residual signal by filtering the sound signal through a linear prediction analysis filter, locating a last pitch pulse of the sound signal of a previous frame from the residual signal, extracting a pitch pulse prototype of given length around a position of the last pitch pulse of the previous frame using the residual signal, and locating pitch pulses in a current frame using the pitch pulse prototype.

Abstract translation: 本发明的示例性实施例至少提供了一种执行操作的方法和装置，包括将声音信号划分为一系列连续的帧，将每个帧划分成多个子帧，通过线性化滤波声音信号产生残余信号 预测分析滤波器，从剩余信号定位前一帧的声音信号的最后音调脉冲，使用剩余信号提取在前一帧的最后音调脉冲的位置周围的给定长度的音调脉冲原型，以及定位音调 使用音调脉冲原型在当前帧中的脉冲。

公开(公告)号：US07680651B2

公开(公告)日：2010-03-16

申请号：US10498254

申请日：2002-12-13

Applicant: Mikko Tammi ,  Milan Jelinek ,  Claude LaFlamme ,  Vesa Ruoppila

Inventor： Mikko Tammi ,  Milan Jelinek ,  Claude LaFlamme ,  Vesa Ruoppila

IPC: G10L19/00

CPC classification number: G10L19/08

Abstract: In accordance with the exemplary embodiments of the invention there is disclosed at least a method and apparatus for determining a long-term-prediction delay parameter characterizing a long term prediction in a technique using signal modification for digitally encoding a sound signal, the sound signal is divided into a series of successive frames, a feature of the sound signal is located in a previous frame, a corresponding feature of the sound signal is located in a current frame, and the long-term-prediction delay parameter is determined for the current frame while mapping, with the long term prediction, the signal feature of the previous frame with the corresponding signal feature of the current frame. Each divided frame of the sound signal is partitioned into a plurality of signal segments, and at least a part of the signal segments of the frame are warped while constraining the warped signal segments inside the frame.

Abstract translation: 根据本发明的示例性实施例，至少公开了一种用于在使用用于数字编码声音信号的信号修改的技术的技术中确定表征长期预测的长期预测延迟参数的方法和装置，声音信号是 分为一系列连续帧，声信号的特征位于先前帧中，声信号的对应特征位于当前帧中，并且为当前帧确定长期预测延迟参数 同时用长期预测将前一帧的信号特征与当前帧的对应信号特征进行映射。 声音信号的每个分割帧被划分成多个信号段，并且框架的信号段的至少一部分变形，同时约束框架内的扭曲信号段。

公开(公告)号：US06606593B1

公开(公告)日：2003-08-12

申请号：US09371332

申请日：1999-08-10

Applicant: Kari Jarvinen ,  Pekka Kapanen ,  Vesa Ruoppila ,  Jani Rotola-Pukkila

Inventor： Kari Jarvinen ,  Pekka Kapanen ,  Vesa Ruoppila ,  Jani Rotola-Pukkila

IPC: G10L1908

CPC classification number: G10L19/012

Abstract: An improved method for generating comfort noise (CN) in a mobile terminal operating in a discontinuous transmission (DTX) mode. In one embodiment the invention provides an improved method for comfort noise generation, in which a random excitation is modified by a spectral control filter so that the frequency content of comfort noise and background noise become similar. In another embodiment the transmitter identifies speech coding parameters that are not representative of the actual background noise, and replaces the identified parameters with parameters having a median value. In this manner the non-representative parameters do not skew the result of an averaging operation.

Abstract translation: 一种用于在以不连续传输（DTX）模式操作的移动终端中产生舒适噪声（CN）的改进方法。 在一个实施例中，本发明提供了一种用于舒适噪声产生的改进方法，其中随机激励由频谱控制滤波器修改，使得舒适噪声和背景噪声的频率含量变得相似。 在另一个实施例中，发射机识别不代表实际背景噪声的语音编码参数，并用具有中值的参数替换所识别的参数。 以这种方式，非代表性参数不会使平均运算的结果偏斜。

公开(公告)号：US20090063139A1

公开(公告)日：2009-03-05

申请号：US12288592

申请日：2008-10-21

Applicant: Mikko Tammi ,  Milan Jelinek ,  Claude LaFlamme ,  Vesa Ruoppila

Inventor： Mikko Tammi ,  Milan Jelinek ,  Claude LaFlamme ,  Vesa Ruoppila

IPC: G10L11/04

CPC classification number: G10L19/08

Abstract: For determining a long-term-prediction delay parameter characterizing a long term prediction in a technique using signal modification for digitally encoding a sound signal, the sound signal is divided into a series of successive frames, a feature of the sound signal is located in a previous frame, a corresponding feature of the sound signal is located in a current frame, and the long-term-prediction delay parameter is determined for the current frame while mapping, with the long term prediction, the signal feature of the previous frame with the corresponding signal feature of the current frame. In a signal modification method for implementation into a technique for digitally encoding a sound signal, the sound signal is divided into a series of successive frames, each frame of the sound signal is partitioned into a plurality of signal segments, and at least a part of the signal segments of the frame are warped while constraining the warped signal segments inside the frame. For searching pitch pulses in a sound signal, a residual signal is produced by filtering the sound signal through a linear prediction analysis filter, a weighted sound signal is produced by processing the sound signal through a weighting filter, the weighted sound signal being indicative of signal periodicity, a synthesized weighted sound signal is produced by filtering a synthesized speech signal produced during a last subframe of a previous frame of the sound signal through the weighting filter, a last pitch pulse of the sound signal of the previous frame is located from the residual signal, a pitch pulse prototype of given length is extracted around the position of the last pitch pulse of the sound signal of the previous frame using the synthesized weighted sound signal, and the pitch pulses are located in a current frame using the pitch pulse prototype.

Abstract translation: 为了确定在使用用于数字编码声音信号的信号修改的技术中表征长期预测的长期预测延迟参数，声音信号被分成一系列连续的帧，声音信号的特征位于 前一帧，声音信号的对应特征位于当前帧中，并且为当前帧确定长期预测延迟参数，同时长期预测将前一帧的信号特征与 当前帧的相应信号特征。 在用于实现用于对声音信号进行数字编码的技术的信号修改方法中，声音信号被分成一系列连续的帧，声音信号的每个帧被划分为多个信号段，并且至少一部分 框架的信号段扭曲，同时约束框架内的翘曲的信号段。 为了在声音信号中搜索音调脉冲，通过线性预测分析滤波器对声音信号进行滤波来产生残留信号，通过加权滤波器处理声音信号产生加权声音信号，加权声音信号表示信号 通过对通过加权滤波器的声音信号的先前帧的最后一个子帧产生的合成语音信号进行滤波，产生合成加权声音信号，将前一帧的声音信号的最后音调脉冲从剩余的位置 信号，使用合成的加权声音信号在前一帧的声音信号的最后音调脉冲的位置周围提取给定长度的音调脉冲原型，并且使用音调脉冲原型将音调脉冲位于当前帧中。

公开(公告)号：US06915257B2

公开(公告)日：2005-07-05

申请号：US09740826

申请日：2000-12-21

Applicant: Ari Heikkinen ,  Samuli Pietila ,  Vesa Ruoppila

Inventor： Ari Heikkinen ,  Samuli Pietila ,  Vesa Ruoppila

IPC: G10L25/93 ,  G10L11/06 ,  G10L11/04

CPC classification number: G10L25/93

Abstract: This invention presents a voicing determination algorithm for classification of a speech signal segment as voiced or unvoiced. The algorithm is based on a normalized autocorrelation where the length of the window is proportional to the pitch period. The speech segment to be classified is further divided into a number of sub-segments, and the normalized autocorrelation is calculated for each sub-segment if a certain number of the normalized autocorrelation values is above a predetermined threshold, the speech segment is classified as voiced. To improve the performance of the voicing determination algorithm in unvoiced to voiced transients, the normalized autocorrelations of the last sub-segments are emphasized. The performance of the voicing decision algorithm can be enhanced by utilizing also the possible lookahead information.

Abstract translation: 本发明提出了一种用于将语音信号段分类为有声或无声的语音确定算法。 该算法基于归一化的自相关，其中窗口的长度与音调周期成比例。 要分类的语音段被进一步划分为多个子段，并且如果一定数量的归一化自相关值高于预定阈值，则针对每个子段计算归一化的自相关，该语音段被分类为有声 。 为了提高无声至浊音瞬态中的发音确定算法的性能，强调了最后一个子段的归一化自相关。 可以通过利用可能的前瞻信息来增强语音决策算法的性能。

公开(公告)号：US5960389A

公开(公告)日：1999-09-28

申请号：US965303

申请日：1997-11-06

Applicant: Kari Jarvinen ,  Pekka Kapanen ,  Vesa Ruoppila ,  Jani Rotola-Pukkila

Inventor： Kari Jarvinen ,  Pekka Kapanen ,  Vesa Ruoppila ,  Jani Rotola-Pukkila

IPC: G10L19/00 ,  G10C3/02

CPC classification number: G10L19/012

Abstract: An improved method for generating comfort noise (CN) in a mobile terminal operating in a discontinuous transmission (DTX) mode. In one embodiment the invention provides an improved method for comfort noise generation, in which a random excitation is modified by a spectral control filter so that the frequency content of comfort noise and background noise become similar. In another embodiment the transmitter identifies speech coding parameters that are not representative of the actual background noise, and replaces the identified parameters with parameters having a median value. In this manner the non-representative parameters do not skew the result of an averaging operation.