公开(公告)号：US06470470B2

公开(公告)日：2002-10-22

申请号：US09019656

申请日：1998-02-06

申请人： Kari Jarvinen ,  Matti Kajala ,  Janne Vainio

发明人： Kari Jarvinen ,  Matti Kajala ,  Janne Vainio

IPC分类号： H03M1300

CPC分类号： H04L1/20 ,  H04L1/0009 ,  H04L1/201

摘要： Focused error correction and/or focused error detection is used in the information coding system. A speech encoding method, in which the number of speech parameter bits on which error correction coding and/or error detection coding focuses is automatically adjusted in relation to the number of total speech parameter bits as a function of the quality of the information transfer connection. There is no need to reduce the number of bits used for speech encoding. Thus the voice quality of the speech remains high. The error correction and/or error detection is focused on the bits most important for the voice quality e.g., as a function of the C/I (Channel to Interference)13 parameter describing the quality of the information transfer connection. The muting of speech synthesizing occuring in prior systems on poor information transfer connection is reduced by using focused error detection.

摘要翻译： 在信息编码系统中使用聚焦误差校正和/或聚焦误差检测。 一种语音编码方法，其中以与信息传送连接的质量的函数相关的总话音参数位数来自动调整纠错编码和/或错误检测编码对焦的语音参数位数。 不需要减少用于语音编码的位数。 因此语音的语音质量仍然很高。 错误校正和/或错误检测集中于对于语音质量最重要的位，例如，作为描述信息传输连接的质量的C / I（信道到干扰）13参数的函数。 通过使用聚焦误差检测来减少在现有系统中发生的信息传递不良连接上的语音合成静音。

公开(公告)号：US06606593B1

公开(公告)日：2003-08-12

申请号：US09371332

申请日：1999-08-10

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

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

IPC分类号： G10L1908

CPC分类号： G10L19/012

摘要： 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.

摘要翻译： 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：US20070265842A1

公开(公告)日：2007-11-15

申请号：US11431423

申请日：2006-05-09

申请人： Kari Jarvinen ,  Pasi Ojala ,  Ari Lakaniemi

发明人： Kari Jarvinen ,  Pasi Ojala ,  Ari Lakaniemi

IPC分类号： G10L11/06

CPC分类号： G10L25/93 ,  G10L19/18 ,  G10L25/78

摘要： Encoding audio signals with selecting an encoding mode for encoding the signal categorizing the signal into active segments having voice activity and non-active segments having substantially no voice activity by using categorization parameters depending on the selected encoding mode and encoding at least the active segments using the selected encoding mode.

摘要翻译： 通过选择编码模式来编码音频信号，用于对信号进行编码，将信号分类为具有语音活动的活动段和基本上不具有语音活动的非活动段，其通过使用根据所选择的编码模式的分类参数，并且至少使用 选择编码模式。

公开(公告)号：US5432884A

公开(公告)日：1995-07-11

申请号：US36544

申请日：1993-03-22

申请人： Pekka Kapanen ,  Yrjo Neuvo ,  Kari Jarvinen

发明人： Pekka Kapanen ,  Yrjo Neuvo ,  Kari Jarvinen

IPC分类号： G10L19/06 ,  G10L9/00

CPC分类号： G10L19/06

摘要： Disclosed herein are methods and apparatus for improving the quality of synthesized speech that is transmitted through a channel that is susceptible to transmission errors. In a presently preferred embodiment of the invention a speech signal is assumed to be first encoded using a Linear Predictive Coding (LPC) technique prior to transmission. The parameters that describe the short-term spectral behavior of the speech signal are received and then applied to and processed by a non-linear median processing block only on an occurrence of a predetermined number of transmission errors in the received LPC speech signal. The median-processed short term speech parameters are subsequently employed, together with a received excitation signal, in a synthesis filter to synthesize a speech signal of improved quality over what would be obtained if the short term speech parameters were not median processed to compensate for the transmission errors.

摘要翻译： 这里公开了用于改善通过易于传输错误的信道传输的合成语音质量的方法和装置。 在本发明的当前优选实施例中，假定在传输之前使用线性预测编码（LPC）技术首先编码语音信号。 描述语音信号的短期频谱行为的参数仅在所接收的LPC语音信号中发生预定数量的传输错误的情况下才被非线性中值处理块应用并被处理。 随后将中值处理的短期语音参数与接收到的激励信号一起用于合成滤波器中，以合成具有改善质量的语音信号，如果短期语音参数未被中值处理以补偿 传输错误。

公开(公告)号：US08374860B2

公开(公告)日：2013-02-12

申请号：US13248213

申请日：2011-09-29

申请人： Kari Jarvinen ,  Pasi Ojala ,  Ari Lakaniemi

发明人： Kari Jarvinen ,  Pasi Ojala ,  Ari Lakaniemi

IPC分类号： G10L11/02 ,  G10L19/00

CPC分类号： G10L25/93 ,  G10L19/18 ,  G10L25/78

摘要： Encoding audio signals with selecting an encoding mode for encoding the signal categorizing the signal into active segments having voice activity and non-active segments having substantially no voice activity by using categorization parameters depending on the selected encoding mode and encoding at least the active segments using the selected encoding mode.

摘要翻译： 通过选择编码模式来编码音频信号，用于对信号进行编码，将信号分类为具有语音活动的活动段和基本上不具有语音活动的非活动段，其通过使用根据所选择的编码模式的分类参数，并且至少使用 选择编码模式。

公开(公告)号：US06170073A

公开(公告)日：2001-01-02

申请号：US08821537

申请日：1997-03-21

申请人： Kari Jarvinen ,  Janne Vainio ,  Petri Haavisto ,  Tero Honkanen

发明人： Kari Jarvinen ,  Janne Vainio ,  Petri Haavisto ,  Tero Honkanen

IPC分类号： H03M1300

CPC分类号： H04L1/0061 ,  H03M13/47 ,  H04L1/0045 ,  H04L1/007

摘要： An encoder encodes digital signals representative of data by classifying the digital signals into first and second classes indicative of their influence on data quality and subjects them to error detection encoding capable of generating at least two error detection codes which respectively correspond to the first and second classes. A decoder receives the encoded digital signals classified into first and second digital signal classes, decodes the error detection codes, and generates error signals, corresponding to the respective digital signal classes, from which the quality of the received digital signals is estimated and the utility of the received digital signals is determined.

摘要翻译： 编码器通过将数字信号分类为表示其对数据质量的影响的第一和第二类别来对代表数据的数字信号进行编码，并使它们能够产生能够产生分别对应于第一和第二类的至少两个错误检测码的错误检测编码 。 解码器接收分类为第一和第二数字信号类别的编码数字信号，对错误检测码进行解码，并产生对应于相应数字信号类别的误差信号，接收数字信号的质量由此从其估计， 确定接收到的数字信号。

公开(公告)号：US6144936A

公开(公告)日：2000-11-07

申请号：US849361

申请日：1997-08-11

申请人： Kari Jarvinen ,  Janne Vainio

发明人： Kari Jarvinen ,  Janne Vainio

IPC分类号： G10L19/005 ,  G10L19/012 ,  H03G3/34 ,  H04B1/10 ,  H04B7/26 ,  G10L21/02

CPC分类号： G10L19/012 ,  H03G3/341 ,  H04B1/1027 ,  G10L19/005

摘要： When several successive bad frames are received, a speech signal applied to an audio speaker is muted after a certain period of time, whereby the user may switch the phone off even though it is possible for the transmission channel to regain its good quality. Instead of the muting of the phone, the user is, according to the invention, given a specific indication, a weak noise signal, for example, indicating that the call synthesization has been discontinued on purpose. The noise signal may be generated in a noise generator prior to a speech decoder, and the noise signal is summed with the attenuated parameter values of the frames substituted during subsitution process.

摘要翻译： PCT No.PCT / FI95 / 00667 Sec。 371日期1997年8月11日 102（e）日期1997年8月11日PCT提交1995年12月5日PCT公布。 WO96 / 18251 PCT出版物 日期1996年6月13日当接收到几个连续的坏帧时，施加到音频扬声器的语音信号在一定时间段之后被静音，由此用户可以切换电话，即使传输信道可以重新获得 高品质。 除了电话的静音之外，根据本发明，用户给出了特定的指示，例如指示呼叫合成已经故意停止的弱噪声信号。 噪声信号可以在语音解码器之前的噪声发生器中产生，并且噪声信号与在归属过程期间被替换的帧的衰减参数值相加。

公开(公告)号：US5526366A

公开(公告)日：1996-06-11

申请号：US375550

申请日：1995-01-19

申请人： Kari Jarvinen ,  Janne Vainio

发明人： Kari Jarvinen ,  Janne Vainio

IPC分类号： G10L19/00 ,  H03M13/00 ,  H04B14/04 ,  H04L1/00 ,  H04L1/20 ,  G06F11/00

CPC分类号： G10L19/005 ,  H03M13/00 ,  H04L1/20 ,  G10L25/69

摘要： The invention relates to a method for processing the parameters of a speech coder according to the quality of the transmission connection. The quality of the transmission connection is determined in a quality monitoring member (409) on the basis of erroneous/error-free classifications of single received frames so that for determining the quality of the transmission connection the classifications of single frames performed during several subsequent speech frames are scanned. On the basis of the quality of the transmission connection both the replacement procedure of erroneous speech frames carried out in the replacement member (402) and the processing of the frames detected as error-free carried out in the processing member (406) of error-free frames are controlled so that the effect of the transmission errors on the quality of the decoded speech signal remains insignificant (FIG. 4).

摘要翻译： 本发明涉及一种根据传输连接质量来处理语音编码器的参数的方法。 基于单个接收帧的错误/无差错分类，在质量监视构件（409）中确定传输连接的质量，以便为了确定传输连接的质量，在几个后续语音中执行的单个帧的分类 扫描帧。 基于传输连接的质量，在更换部件（402）中执行的错误语音帧的替换过程和在错误发送处理部件（406）中执行的检测为无错误的帧的处理， 控制自由帧，使得传输错误对解码语音信号的质量的影响保持不显着（图4）。

公开(公告)号：US6029128A

公开(公告)日：2000-02-22

申请号：US662991

申请日：1996-06-13

申请人： Kari Jarvinen ,  Tero Honkanen

发明人： Kari Jarvinen ,  Tero Honkanen

IPC分类号： G10L19/04 ,  G10L19/26 ,  H03M7/30 ,  G10L3/00

CPC分类号： G10L19/26 ,  G10L19/04

摘要： A post-processor 317 and method substantially for enhancing synthesised speech is disclosed. The post-processor 317 operates on a signal ex(n) derived from an excitation generator 211 typically comprising a fixed code book 203 and an adaptive code book 204, the signal ex(n) being formed from the addition of scaled outputs from the fixed code book 203 and adaptive code book 204. The post-processor operates on ex(n) by adding to it a scaled signal pv(n) derived from the adaptive code book 204. A gain or scale factor p is determined by the speech coefficients input to the excitation generator 211. The combined signal ex(n)+pv(n) is normalised by unit 316 and input to an LPC or speech synthesis filter 208, prior to being input to an audio processing unit 209.

摘要翻译： 公开了一种基本上用于增强合成语音的后处理器317和方法。 后处理器317对从通常包括固定码本203和自适应码本204的激励发生器211导出的信号ex（n）进行操作，信号ex（n）是从固定的 码本203和自适应码本204.后处理器通过向自适应码本204导出的缩放信号pv（n）加上ex（n）来操作。增益或比例因子p由语音系数 输入到激励发生器211.组合信号ex（n）+ pv（n）在输入到音频处理单元209之前由单元316归一化并输入到LPC或语音合成滤波器208。

公开(公告)号：US06021385A

公开(公告)日：2000-02-01

申请号：US809675

申请日：1997-03-19

申请人： Kari Jarvinen ,  Janne Vainio

发明人： Kari Jarvinen ,  Janne Vainio

IPC分类号： G10L13/00 ,  G10L19/005 ,  H04B7/26 ,  H04L1/00 ,  H04L1/20 ,  H04Q7/38 ,  G01L9/00 ,  H04B13/02

CPC分类号： H04L1/20 ,  G10L19/005

摘要： A method for detecting defective speech frames in a receiver of a speech communication system includes a coded speech signal consisting of speech frames conveying speech coding parameters received through a transmission link. The detection method for defective speech frames depends on the quality of the transmission link. If the quality of the transmission link is "poor", unusual parameter values of a speech signal in a speech frame are interpreted as transmission errors, and the speech frame is classified as defective. During a transmission link of "good" quality, all received parameter values possible for speech are accepted and forwarded to a speech decoder. In the latter mode of operation, the detection of defective speech frames is solely based on conventional error detection methods employed in the receiver.

摘要翻译： PCT No.PCT / FI95 / 00510 Sec。 371日期1997年3月19日 102（e）1997年3月19日PCT PCT 1995年9月19日PCT公布。 出版物WO96 / 09704 日期1996年3月28日一种用于检测语音通信系统的接收机中的有缺陷的语音帧的方法包括由传送通过传输链路接收的语音编码参数的语音帧组成的编码语音信号。 缺陷语音帧的检测方法取决于传输链路的质量。 如果传输链路的质量“差”，语音帧中的语音信号的异常参数值被解释为传输错误，并且语音帧被分类为有缺陷的。 在“良好”质量的传输链路期间，可接收所有接收到的用于语音的参数值并将其转发到语音解码器。 在后一种操作模式中，缺陷语音帧的检测仅基于在接收机中使用的常规错误检测方法。