公开(公告)号：US06691085B1

公开(公告)日：2004-02-10

申请号：US09691323

申请日：2000-10-18

申请人： Jani Rotola-Pukkila ,  Hannu Mikkola ,  Janne Vainio

发明人： Jani Rotola-Pukkila ,  Hannu Mikkola ,  Janne Vainio

IPC分类号： G10L2102

CPC分类号： G10L21/0364 ,  G10L25/78

摘要： A method and system for encoding and decoding an input signal, wherein the input signal is divided into a higher frequency band and a lower frequency band in the encoding and decoding processes, and wherein the decoding of the higher frequency band is carried out by using an artificial signal along with speech related parameters obtained from the lower frequency band. In particular, the artificial signal is scaled before it is transformed into an artificial wideband signal containing colored noise in both the lower and the higher frequency band. Additionally, voice activity information is used to define speech periods and non-speech periods of the input signal. Based on the voice activity information, different weighting factors are used to scale the artificial signal in speech periods and non-speech periods.

摘要翻译： 一种用于对输入信号进行编码和解码的方法和系统，其中在编码和解码处理中将输入信号划分为较高频带和较低频带，并且其中通过使用 人工信号以及从较低频带获得的语音相关参数。 特别地，在将人造信号变换成包含较低和较高频带内的彩色噪声的人造宽带信号之前进行缩放。 此外，语音活动信息用于定义输入信号的语音周期和非语音周期。 基于语音活动信息，使用不同的加权因子在语音周期和非语音周期中缩放人造信号。

公开(公告)号：US06662155B2

公开(公告)日：2003-12-09

申请号：US09970091

申请日：2001-10-02

申请人： Jani Rotola-Pukkila ,  Hannu Mikkola ,  Janne Vainio

发明人： Jani Rotola-Pukkila ,  Hannu Mikkola ,  Janne Vainio

IPC分类号： G10L2102

CPC分类号： G10L19/012 ,  G10L25/78

摘要： A method and system for providing comfort noise in the non-speech periods in speech communication. The comfort noise is generated based on whether the background noise in the speech input is stationary or non-stationary. If the background noise is non-stationary, a random component is inserted in the comfort noise using a dithering process. If the background noise is stationary, the dithering process is not used.

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：US06230125B1

公开(公告)日：2001-05-08

申请号：US08894676

申请日：1998-01-05

申请人： Janne Vainio

发明人： Janne Vainio

IPC分类号： G10L1100

CPC分类号： H04L1/0072 ,  G10L19/005 ,  H04L1/0061 ,  H04L1/007 ,  H04L2001/0098

摘要： The present invention relates to processing speech coding parameters in a telecommunication system. The speech coding parameters of a speech frame, produced by a speech encoder, are divided into groups, i.e. so-called virtual channels, in which speech parameter error correction, channel coding and processing of error-free or erroneous speech parameters are performed independently. At the receiving end, the processing (505) of erroneous and error-free speech parameters can thus be controlled independently on each virtual transmission channel (502) according to the quality of each virtual transmission channel. The speech parameters of the high-quality virtual channels of a speech frame can thus be processed as error-free, replacing the speech coding parameters of the low-quality virual channels only. The independently processed (505) speech parameters of the virtual channels are thus reassembled (507) into a speech frame, which is applied to decoding. Since part of the information of also erroneous speech frames is utilized, the use of speech information received from a transmission channel can be increased in speech decoding, which reduces for instance interruptions occurring in speech as compared with a situation where all speech frames erroneous even to a slight degree were discarded. The increased and more focused error indication also reduces the number of undetected errors and thus reduces significantly the worst audible disturbances.

摘要翻译： 本发明涉及在电信系统中处理语音编码参数。 由语音编码器产生的语音帧的语音编码参数被分成几组，即所谓的虚拟频道，其中独立地执行语音参数纠错，信道编码和无错误或错误语音参数的处理。 因此，在接收端，根据每个虚拟传输信道的质量，可以在每个虚拟传输信道（502）上独立地控制错误和无错误语音参数的处理（505）。 因此，语音帧的高质量虚拟信道的语音参数可以被无错处理，仅代替低质量虚拟信道的语音编码参数。 因此虚拟通道的独立处理（505）语音参数被重新组合（507）到被应用于解码的语音帧中。 由于利用了错误语音帧的信息的一部分，所以在语音解码中可以增加从传输信道接收到的语音信息的使用，这减少了例如在语音中发生的中断，与所有语音帧甚至是 轻微的程度被丢弃。 增加和更聚焦的错误指示也减少了未检测到的错误的数量，从而显着降低了最差的可听到的干扰。

公开(公告)号：US07483830B2

公开(公告)日：2009-01-27

申请号：US09797115

申请日：2001-03-01

申请人： Jani Rotola-Pukkila ,  Janne Vainio ,  Hannu Mikkola

发明人： Jani Rotola-Pukkila ,  Janne Vainio ,  Hannu Mikkola

IPC分类号： G10L19/04

CPC分类号： G10L19/16 ,  G10L19/0212

摘要： A speech decoder comprises a decoder (103) for converting a linear prediction encoded speech signal into a first sample stream having a first sampling rate and representing a first frequency band. Additionally it comprises a vocoder (105) for converting an input signal into a second sample stream having a second sampling rate and representing a second frequency band, and combination means (107) for combining the first and second sample streams in processed form. It comprises also means (301) for generating a second linear prediction filter, to be used by the vocoder (105) on the second frequency band, on the basis of a first linear prediction filter used by the decoder (103) on the first frequency band. Extrapolation through an infinite impulse response filter is the preferable method of generating the second linear prediction filter.

摘要翻译： 语音解码器包括用于将线性预测编码语音信号转换为具有第一采样率并表示第一频带的第一采样流的解码器（103）。 另外，它包括用于将输入信号转换为具有第二采样率并表示第二频带的第二采样流的声码器（105），以及用于以处理形式组合第一和第二采样流的组合装置（107）。 它还包括用于产生由第二频带上的声码器（105）使用的第二线性预测滤波器的装置（301），该第二线性预测滤波器基于第一频率由解码器（103）使用的第一线性预测滤波器 带。 通过无限脉冲响应滤波器的外推是产生第二线性预测滤波器的优选方法。

公开(公告)号：US20080120098A1

公开(公告)日：2008-05-22

申请号：US11562067

申请日：2006-11-21

申请人： Jari M. Makinen ,  Juha Marila ,  Hannu J. Mikkola ,  Janne Vainio ,  Tuomas Vaittinen ,  Sakari Himanen ,  Kai K. Samposalo

发明人： Jari M. Makinen ,  Juha Marila ,  Hannu J. Mikkola ,  Janne Vainio ,  Tuomas Vaittinen ,  Sakari Himanen ,  Kai K. Samposalo

IPC分类号： G10L19/00

CPC分类号： G10L19/22

摘要： The present invention provides, methods, computer-readable media, and apparatuses for tuning and adjusting the computational complexity of algorithm that is executed by a signal encoder. The signal encoder may comprise a speech encoder. When a resource shortage on a computer platform is detected, a degree of the resource shortage and a corresponding complexity adjustment for a speech encoder are determined. The speech encoder is then tuned to adjust the computational complexity of an executed speech processing algorithm. The resource shortage may correspond to a computational capability, audio buffer memory, or battery of a mobile device. A speech process being executed by the mobile device is tuned to adjust the computational demands in accordance with a complexity adjustment. A number of iteration rounds may be adjusted while the speech encoder is executing a speech processing algorithm. The iterations may correspond to an algebraic codebook search.

摘要翻译： 本发明提供了用于调整和调整由信号编码器执行的算法的计算复杂度的方法，计算机可读介质和装置。 信号编码器可以包括语音编码器。 当检测到计算机平台上的资源短缺时，确定了语音编码器的资源短缺程度和对应的复杂度调整。 然后调谐语音编码器以调整执行的语音处理算法的计算复杂度。 资源短缺可能对应于移动设备的计算能力，音频缓冲存储器或电池。 调整由移动设备执行的语音过程以根据复杂性调整来调整计算需求。 当语音编码器执行语音处理算法时，可以调整多个迭代轮。 迭代可以对应于代数码本搜索。

公开(公告)号：US20070011009A1

公开(公告)日：2007-01-11

申请号：US11177250

申请日：2005-07-08

申请人： Jani Nurminen ,  Sakari Himanen ,  Anssi Ramo ,  Janne Vainio

发明人： Jani Nurminen ,  Sakari Himanen ,  Anssi Ramo ,  Janne Vainio

IPC分类号： G10L13/08

CPC分类号： G10L13/06

摘要： The invention relates to a support of a concatenative TTS synthesis. In order to generate a speech database as a basis for the TTS synthesis, first, a speech processing including a segmental parametric speech encoding of speech data based on a parametric modeling of speech is performed, which results in compressed parameterized speech segments. Then, the compressed parameterized speech segments are assembled in a speech database. In order to synthesize output speech, compressed parameterized speech segments are selected from the speech database based on an available text and decompressed to regain parameterized speech segments. The parameterized speech segments are then concatenated in a parameter domain. The output speech is synthesized based on these concatenated parametric speech segments.

摘要翻译： 本发明涉及一种级联TTS合成的支持。 为了生成语音数据库作为TTS综合的基础，首先，执行包括基于语音的参数建模的语音数据的分段参数语音编码的语音处理，这导致压缩的参数化语音段。 然后，压缩的参数化语音段被组合在语音数据库中。 为了合成输出语音，基于可用文本从语音数据库中选择压缩的参数化语音段，并且解压缩以重新获得参数化语音段。 参数化语音段然后在参数域中连接。 基于这些连接的参数语音段来合成输出语音。

公开(公告)号：US20060267931A1

公开(公告)日：2006-11-30

申请号：US11433090

申请日：2006-05-12

申请人： Janne Vainio ,  Hannu Mikkola ,  Hannu Korhonen ,  Sakari Himanen ,  Toni Nieminen ,  Tuomas Vaittinen ,  Juha Marila

发明人： Janne Vainio ,  Hannu Mikkola ,  Hannu Korhonen ,  Sakari Himanen ,  Toni Nieminen ,  Tuomas Vaittinen ,  Juha Marila

IPC分类号： G09G5/00

CPC分类号： G06F3/0233 ,  G06F3/167 ,  H04M1/23 ,  H04M2250/70

摘要： According to an aspect of the invention, an enhanced audible feedback solution has been invented for electronic devices using an input device facilitating navigation though a plurality of available user interface input options and confirmation of a selected input option. The electronic device is arranged to define, as a response to detecting a selection of a character on the basis of a detection of a first input to an input device of the electronic device, an audio segment specific to the character. The electronic device is arranged to output the defined audio segment via the audio output means prior to a confirmation by a second input to the input device, the second input being associated with a function adding the character as part of a character sequence entered by the user.

公开(公告)号：US07031926B2

公开(公告)日：2006-04-18

申请号：US09918300

申请日：2001-07-30

申请人： Jari Mäkinen ,  Hannu Mikkola ,  Janne Vainio ,  Jani Rotola-Pukkila

发明人： Jari Mäkinen ,  Hannu Mikkola ,  Janne Vainio ,  Jani Rotola-Pukkila

IPC分类号： G10L19/00

CPC分类号： G10L19/005 ,  G10L19/06 ,  G10L25/93

摘要： A method for use by a speech decoder in handling bad frames received over a communications channel a method in which the effects of bad frames are concealed by replacing the values of the spectral parameters of the bad frames (a bad frame being either a corrupted frame or a lost frame) with values based on an at least partly adaptive mean of recently received good frames, but in case of a corrupted frame (as opposed to a lost frame), using the bad frame itself if the bad frame meets a predetermined criterion. The aim of concealment is to find the most suitable parameters for the bad frame so that subjective quality of the synthesized speech is as high as possible.

公开(公告)号：US06981202B2

公开(公告)日：2005-12-27

申请号：US10040885

申请日：2002-01-02

申请人： Hannu Mikkola ,  Janne Vainio ,  Jani Rotola-Pukkila

发明人： Hannu Mikkola ,  Janne Vainio ,  Jani Rotola-Pukkila

IPC分类号： H03M13/29 ,  H03M13/00 ,  H03M13/23 ,  H03M13/27 ,  H03M13/35 ,  H04L1/00 ,  H03M13/03 ,  G01R31/28

CPC分类号： H04L1/0068 ,  H03M13/23 ,  H03M13/27 ,  H03M13/35 ,  H04L1/0059 ,  H04L1/0071 ,  H04L1/0083

摘要： A method and corresponding apparatus for encoding a sequence of bits for transmission as symbols, some of the bit positions of the symbols having a higher bit error rate than other bit positions. A plurality of sequences of bits is provided using a convolutional encoder, in response to a sequence of input bits, each sequence of bits being defined by a predetermined generator polynomial having a predetermined level of sensitivity to puncturing. Then the bits of each sequence of bits are mapped to symbol positions based on the level of sensitivity of the generator polynomial defining the sequence of bits. With interleaving, the mapping of bits of each sequence of bits to symbol positions can precede a symbol interleaving step, or it can follow a bit interleaving step.

摘要翻译： 一种用于将用于传输的比特序列编码为符号的方法和相应的装置，所述符号的一些比特位置具有比其他比特位置更高的比特错误率。 使用卷积编码器提供多个比特序列，响应于输入比特序列，每个比特序列由具有对穿孔的预定灵敏度水平的预定生成多项式定义。 然后，基于定义比特序列的生成多项式的灵敏度级别，将每个比特序列的比特映射到符号位置。 通过交织，每个比特序列的比特映射到符号位置可以在符号交织步骤之前，或者可以跟随比特交织步骤。