Efficient speech stream conversion
    1.
    发明授权
    Efficient speech stream conversion 有权
    有效的语音流转换

    公开(公告)号:US08543388B2

    公开(公告)日:2013-09-24

    申请号:US12095709

    申请日:2005-11-30

    IPC分类号: G10L19/00

    CPC分类号: G10L19/012 G10L19/173

    摘要: Speech frames of a first speech coding scheme are utilized as speech frames of a second speech coding scheme, where the speech coding schemes use similar core compression schemes for the speech frames, preferably bit stream compatible. An occurrence of a state mismatch in an energy parameter between the first speech coding scheme and the second speech coding scheme is identified, preferably either by determining an occurrence of a predetermined speech evolution, such as a speech type transition, e.g. an onset of speech following a period of speech inactivity, or by tentative decoding of the energy parameter in the two encoding schemes followed by a comparison. Subsequently, the energy parameter in at least one frame of the second speech coding scheme following the occurrence of the state mismatch is adjusted. The present invention also presents transcoders and communications systems providing such transcoding functionality.

    摘要翻译: 第一语音编码方案的语音帧被用作第二语音编码方案的语音帧,其中语音编码方案对于语音帧使用类似的核心压缩方案,优选地与比特流兼容。 识别出第一语音编码方案和第二语音编码方案之间的能量参数中的状态失配的发生,优选地通过确定诸如语音类型转换的预定语音演进的发生,例如, 在语音不活动的时期之后的语音开始,或者通过对两个编码方案中的能量参数的暂时解码进行比较。 随后,调整在发生状态失配之后的第二语音编码方案的至少一帧中的能量参数。 本发明还提供了提供这种代码转换功能的代码转换器和通信系统。

    EFFICIENT SPEECH STREAM CONVERSION
    2.
    发明申请
    EFFICIENT SPEECH STREAM CONVERSION 有权
    有效的语音流转换

    公开(公告)号:US20100223053A1

    公开(公告)日:2010-09-02

    申请号:US12095709

    申请日:2005-11-30

    IPC分类号: G10L19/04

    CPC分类号: G10L19/012 G10L19/173

    摘要: Speech frames of a first speech coding scheme are utilized as speech frames of a second speech coding scheme, where the speech coding schemes use similar core compression schemes for the speech frames, preferably bit stream compatible. An occurrence of a state mismatch in an energy parameter between the first speech coding scheme and the second speech coding scheme is identified, preferably either by determining an occurrence of a predetermined speech evolution, such as a speech type transition, e.g. an onset of speech following a period of speech inactivity, or by tentative decoding of the energy parameter in the two encoding schemes followed by a comparison. Subsequently, the energy parameter in at least one frame of the second speech coding scheme following the occurrence of the state mismatch is adjusted. The present invention also presents transcoders and communications systems providing such transcoding functionality.

    摘要翻译: 第一语音编码方案的语音帧被用作第二语音编码方案的语音帧,其中语音编码方案对于语音帧使用类似的核心压缩方案,优选地与比特流兼容。 识别出第一语音编码方案和第二语音编码方案之间的能量参数中的状态失配的发生,优选地通过确定诸如语音类型转换的预定语音演进的发生,例如, 在语音不活动的时期之后的语音开始,或者通过对两个编码方案中的能量参数的暂时解码进行比较。 随后,调整在发生状态失配之后的第二语音编码方案的至少一帧中的能量参数。 本发明还提供了提供这种代码转换功能的代码转换器和通信系统。

    Method and apparatus for increasing perceived interactivity in communications systems
    3.
    发明申请
    Method and apparatus for increasing perceived interactivity in communications systems 审中-公开
    增加通信系统中感知交互性的方法和装置

    公开(公告)号:US20050227657A1

    公开(公告)日:2005-10-13

    申请号:US10819376

    申请日:2004-04-07

    IPC分类号: G10L13/06 H04L12/56 H04Q7/38

    CPC分类号: G10L21/04

    摘要: Perceived interactivity in user communications is achieved by reducing a perceived delay switching the active transmitter in the communication without having to reduce actual transmission and setup delays associated with a communication exchange. A sound signal is identified in the user communication. The sound signal is analyzed to identify or estimate a sound signal segment. The sound signal segment is preferably (though not necessarily) located at the beginning or the end of the sound signal. The sound signal segment may be selected directly from the sound signal itself, from a modified version of the sound signal, or from a signal associated with the sound signal. A determination is made that a length or duration of the sound signal segment should be or can be modified. One or more modifications for the sound signal segment are determined and are provided to one or more processing units to perform the modification(s).

    摘要翻译: 用户通信中的感知交互性通过减少在通信中切换有源发射机的感知延迟而不用减少与通信交换相关联的实际传输和建立延迟来实现。 在用户通信中识别声音信号。 分析声音信号以识别或估计声音信号段。 声音信号段优选地(尽管不一定)位于声音信号的开始或结束处。 可以从声音信号本身,声音信号的修改版本或与声音信号相关联的信号直接选择声音信号段。 确定声音信号段的长度或持续时间应该是或可以被修改。 确定声音信号段的一个或多个修改,并将其提供给一个或多个处理单元以执行修改。

    Method And Arrangement For Adapting Transmission Of Encoded Media
    4.
    发明申请
    Method And Arrangement For Adapting Transmission Of Encoded Media 有权
    适应传输媒体的方法和布置

    公开(公告)号:US20100232297A1

    公开(公告)日:2010-09-16

    申请号:US12438443

    申请日:2007-05-28

    IPC分类号: H04L12/26 H04L29/02

    摘要: The present invention is to select an adaptation scheme for the transmission of the encoded media that results in a satisfactory performance of the transmitted encoded media. A difference from the prior art is that each adaptation scheme defines a set of different transmission formats, wherein each transmission formats is a combination of at least two of the parameters the source codec bit rate, the packet rate, the number of frames of each packet (referred to as frame aggregation), and the level of redundancy. By using the different transmission formats, the transmission can be adapted to different operating scenarios and the performance is hence improved.

    摘要翻译: 本发明是选择用于传输编码媒体的适应方案,其导致所传送的编码媒体的令人满意的性能。 与现有技术的不同之处在于每个适配方案定义了一组不同的传输格式,其中每个传输格式是源编解码器比特率,分组速率,每个分组的帧数的至少两个参数的组合 (简称帧聚合)和冗余级别。 通过使用不同的传输格式,传输可以适应不同的操作场景,从而改善性能。

    CELP encoding/decoding method and apparatus
    5.
    发明授权
    CELP encoding/decoding method and apparatus 有权
    CELP编码/解码方法和装置

    公开(公告)号:US07146311B1

    公开(公告)日:2006-12-05

    申请号:US09395909

    申请日:1999-09-14

    IPC分类号: G10L19/08

    CPC分类号: G10L19/22 G10L2019/0005

    摘要: A multi-codebook fixed bitrate CELP signal block encoder/decoder includes a codebook selector (22) for selecting, for each signal block, a corresponding codebook identification in accordance with a deterministic selection procedure that is independent of signal type. Included are also means for encoding/decoding each signal block by using a codebook having the selected codebook identification.

    摘要翻译: 多码本固定位速率CELP信号块编码器/解码器包括码本选择器(22),用于根据独立于信号类型的确定性选择过程为每个信号块选择相应的码本识别。 包括也是通过使用具有所选码本识别的码本来对每个信号块进行编码/解码的装置。

    Method to determine the excitation pulse positions within a speech frame
    6.
    发明授权
    Method to determine the excitation pulse positions within a speech frame 失效
    确定语音帧内的激励脉冲位置的方法

    公开(公告)号:US6064956A

    公开(公告)日:2000-05-16

    申请号:US930951

    申请日:1998-01-05

    申请人: Jonas Svedberg

    发明人: Jonas Svedberg

    IPC分类号: G10L19/10

    CPC分类号: G10L19/10

    摘要: A determination is made of the positions within a speech frame for a given number of excitation pulses in a linear predictive speech encoder. A combination of two known methods is used. The positions of the excitation pulses are calculated in a number of calculation stages according to the first known method. The positions of the excitation pulses are then calculated in a number of calculation stages in accordance with the second method to obtain one of a number of pulse placements. Each calculation according to the second method begins at a starting point from one of a number of positions calculated in accordance with the first method. The proportion between the number of calculation stages in the first method and the second method is chosen so as to obtain the least calculation complexity for a certain given speech quality.

    摘要翻译: PCT No.PCT / SE96 / 00465 Sec。 371日期:1998年1月5日 102(e)日期1998年1月5日PCT提交1996年4月10日PCT公布。 第WO96 / 32712号公报 日期1996年10月17日对线性预测语音编码器中给定数量的激励脉冲的语音帧内的位置进行确定。 使用两种已知方法的组合。 根据第一已知方法,在多个计算阶段中计算激励脉冲的位置。 然后根据第二种方法在多个计算阶段中计算激励脉冲的位置,以获得多个脉冲放置中的一个。 根据第二种方法的每个计算都是从根据第一种方法计算出的多个位置之一开始的起始点。 选择第一种方法中的计算级数与第二种方法之间的比例,以便为某种给定的语音质量获得最小的计算复杂度。

    Interoperability for wireless user devices with different speech processing formats
    8.
    发明申请
    Interoperability for wireless user devices with different speech processing formats 有权
    具有不同语音处理格式的无线用户设备的互操作性

    公开(公告)号:US20060034260A1

    公开(公告)日:2006-02-16

    申请号:US11197768

    申请日:2005-08-05

    摘要: Interoperability is achieved between wireless user communication devices that have different speech processing formats and/or attributes. A first wireless user communication device includes a primary speech codec that encodes a first speech message using a first speech encoding format. The encoded speech is then sent to a second wireless user communications device that includes a primary speech codec supporting a second speech encoding format. The first user device receives from the second user device a second speech message encoded using the second speech encoding format. The second speech message is then decoded by the first user device using a second speech decoder supporting decoding of the second speech encoding format. But the first communication device does not support speech encoding using the second speech encoding format—regardless of whether the first communication device includes or does not includes an encoder for encoding speech using the first speech encoding format.

    摘要翻译: 在具有不同语音处理格式和/或属性的无线用户通信设备之间实现互操作性。 第一无线用户通信设备包括使用第一语音编码格式对第一语音消息进行编码的主要语音编解码器。 然后将经编码的语音发送到包括支持第二语音编码格式的主要语音编解码器的第二无线用户通信设备。 第一用户设备从第二用户设备接收使用第二语音编码格式编码的第二语音消息。 第二语音消息然后由支持第二语音编码格式解码的第二语音解码器由第一用户设备解码。 但是第一通信设备不支持使用第二语音编码格式的语音编码,而不管第一通信设备是否包括或不包括使用第一语音编码格式来编码语音的编码器。

    Method and arrangement for speech coding in wireless communication systems
    10.
    发明授权
    Method and arrangement for speech coding in wireless communication systems 有权
    无线通信系统中语音编码的方法和装置

    公开(公告)号:US08438018B2

    公开(公告)日:2013-05-07

    申请号:US12278529

    申请日:2006-02-06

    IPC分类号: G10L19/00

    摘要: The present invention relates to speech coding in wireless and wireline communication systems. The present invention provides a method of saving bandwidth by a controlled dropping of speech frames at an encoder in a sending communication device. The dropping is controlled in a manner to minimize the effects on the speech quality after the decoding in the receiving communication device, by assuring that the state mismatch between the encoder and the decoder is removed or at least significantly reduced. This is achieved by letting the encoder run an ECU algorithm with a similar behavior as the one running in the decoder in the receiving communication device.

    摘要翻译: 本发明涉及无线和有线通信系统中的语音编码。 本发明提供了一种通过在发送通信设备中的编码器处控制丢弃语音帧来节省带宽的方法。 通过确保去除或至少显着地减少编码器和解码器之间的状态不匹配,在接收通信设备中解码之后,以最小化对语音质量的影响的方式来控制丢弃。 这通过使编码器运行具有与在接收通信设备中的解码器中运行的ECU算法类似的行为的ECU算法来实现。