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49 条记录 (耗时 0.019 秒)

    METHOD AND ARRANGEMENT FOR ADAPTING TRANSMISSION OF ENCODED MEDIA
    31.
    发明授权
    METHOD AND ARRANGEMENT FOR ADAPTING TRANSMISSION OF ENCODED MEDIA 有权
    方法和系统调整发送编码的媒体

    公开(公告)号：EP2055036B1

    公开(公告)日：2012-05-23

    申请号：EP07748525.8

    申请日：2007-05-28

    申请人： Telefonaktiebolaget LM Ericsson (publ)

    发明人： JOHANSSON, Ingemar ,  FRANKKILA, Tomas ,  SVEDBERG, Jonas

    IPC分类号： H04L1/00

    CPC分类号： H04L1/0003 ,  H04L1/0009 ,  H04L1/0014 ,  H04L1/0015 ,  H04L1/0016

    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.

    METHOD AND ARRANGEMENT FOR ADAPTING TRANSMISSION OF ENCODED MEDIA
    32.
    发明公开
    METHOD AND ARRANGEMENT FOR ADAPTING TRANSMISSION OF ENCODED MEDIA 有权
    方法和系统调整发送编码的媒体

    公开(公告)号：EP2055036A1

    公开(公告)日：2009-05-06

    申请号：EP07748525.8

    申请日：2007-05-28

    申请人： TELEFONAKTIEBOLAGET LM ERICSSON (publ)

    发明人： JOHANSSON, Ingemar ,  FRANKKILA, Tomas ,  SVEDBERG, Jonas

    IPC分类号： H04L1/00

    CPC分类号： H04L1/0003 ,  H04L1/0009 ,  H04L1/0014 ,  H04L1/0015 ,  H04L1/0016

    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.

    METHOD AND ARRANGEMENT FOR SPEECH CODING IN WIRELESS COMMUNICATION SYSTEMS
    33.
    发明公开
    METHOD AND ARRANGEMENT FOR SPEECH CODING IN WIRELESS COMMUNICATION SYSTEMS 有权
    VERFAHREN UNDOORRENEN ZOR SPRACHCERIUNGUNG DRAHTLOSEN KOMMUNIKATIONSSYSTEMEN

    公开(公告)号：EP1982331A1

    公开(公告)日：2008-10-22

    申请号：EP06704638.3

    申请日：2006-02-06

    申请人： Telefonaktiebolaget LM Ericsson (publ)

    发明人： JOHANSSON, Ingemar ,  SVEDBERG, Jonas

    IPC分类号： G10L19/14 ,  H04L12/56

    CPC分类号： G10L19/24 ,  G10L19/005 ,  H04L47/10

    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 (125) in a sending communication device (105). The dropping is controlled in a manner to minimize the effects on the speech quality after the decoding in the receiving communication device (110), by assuring that the state mismatch between the encoder (125) and the decoder (145) is removed or at least significantly reduced. This is achieved by letting the encoder (125) run an ECU algorithm with a similar behavior as the one running in the decoder (145) in the receiving communication device (110).

    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与在接收通信设备中的解码器中运行的ECU算法具有相似行为的ECU算法来实现。

    FIDELITY-OPTIMISED VARIABLE FRAME LENGTH ENCODING
    35.
    发明授权
    FIDELITY-OPTIMISED VARIABLE FRAME LENGTH ENCODING 有权
    FIDELITY优化的编码可变帧长度

    公开(公告)号：EP1623411B1

    公开(公告)日：2007-08-29

    申请号：EP04820553.8

    申请日：2004-12-15

    申请人： TELEFONAKTIEBOLAGET LM ERICSSON (publ)

    发明人： BRUHN, Stefan ,  JOHANSSON, Ingemar ,  TALEB, Anisse ,  ENSTRÖM, Daniel

    IPC分类号： G10L19/00

    CPC分类号： G10L19/008

    摘要： Polyphonic signals are used to create a main signal, typically a mono signal, and a side signal (Xside). A number of encoding schemes (81) for the side signal (Xside) are provided. Each encoding scheme (81) is characterised by a set of sub-frames (90) of different lengths. The total length of the sub-frames (90) corresponds to the length of the encoding frame (80) of the encoding scheme (81). The encoding scheme (81) to be used on the side signal (Xside) is selected dependent on the present signal content of the polyphonic signals. In a preferred embodiment, a side residual signal is created as the difference between the side signal and the main signal scaled with a balance factor. The balance factor is selected to minimise the side residual signal. The optimised side residual signal and the balance factor are encoded and provided as encoding parameters representing the side signal.

    SPEECH CODING
    36.
    发明授权
    SPEECH CODING 有权
    语音编码

    公开(公告)号：EP1112568B1

    公开(公告)日：2007-02-21

    申请号：EP99951312.0

    申请日：1999-09-10

    申请人： TELEFONAKTIEBOLAGET LM ERICSSON (publ)

    发明人： JOHANSSON, Ingemar ,  SVEDBERG, Jonas ,  UVLIDEN, Anders

    IPC分类号： G10L19/00 ,  G10L19/14

    CPC分类号： G10L19/012 ,  G10L19/083

    摘要： In producing an approximation of an original speech signal from encoded information about the original speech signal, current parameters (EnPar(i)) associated with a current segment of the original speech signal are determined from the encoded information. Reproduction of a noise component of the original speech signal is improved by using at least one of the current parameters and corresponding previous parameters respectively associated with previous segments of the original speech signal (31, 37, 39) to produce a modified parameter (EnPar(i) mod ). The modified parameter is then used (25, 40) to produce an approximation of the current segment of the original speech signal.

    摘要翻译： 在根据关于原始语音信号的编码信息产生原始语音信号的近似值时，从编码信息确定与原始语音信号的当前片段相关联的当前参数（EnPar（i））。 原始语音信号的噪声分量的再现通过使用分别与原始语音信号（31,37,39）的先前分段相关联的当前参数和对应的先前参数中的至少一个来产生经修改的参数（EnPar（ ⅰ）MOD）。 然后使用修改的参数（25,40）来产生原始语音信号的当前段的近似值。