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公开(公告)号:DE60129327T2
公开(公告)日:2008-03-20
申请号:DE60129327
申请日:2001-05-22
Applicant: ERICSSON TELEFON AB L M
Inventor: SUNDQVIST JIM , LENNESTAL HAKAN , NOHLGREN ANDERS , LINDQVIST MORGAN , WANG WEI
IPC: H04L12/64
Abstract: The present invention relates to a receiver system in a communication system supporting packet-based communicatior (e.g., an IP-network), including a receiver, speech decoder ( 40 ) and a jitter buffer ( 20 ) for handling delay variations in the reception of a speech signal consisting of packets containing frames with encoded speech. A jitter buffer controller ( 50 ) is provided for keeping information about the functional size of the jitter buffer ( 20 ) and for providing the speech decoder ( 40 ) with control information, such that the speech decoder ( 40 ), based on that information. provides a dynamic adaptation of the size of the jitter buffer ( 20 ) using the received encoded, packetized speech signal. The invention also relates to a method of adapting the functional size of the jitter buffer of a receiver system.
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公开(公告)号:DE60027875T2
公开(公告)日:2006-11-30
申请号:DE60027875
申请日:2000-02-25
Applicant: ERICSSON TELEFON AB L M
Inventor: WESTBERG LARS , SVANBRO KRISTER , SUNDQVIST JIM
Abstract: The soft state of a header compression scheme in a communication system carrying packet traffic including a real time communication signal can be updated (63) during periods of communication signal inactivity (62), during which there is no need to transmit the communication signal. The header compression soft state can also be updated by stealing bits (83, 84) from the communication signal to carry the header update information (73). If the communication signal includes source encoded data, the header compression soft state can be updated selectively (126) based on the bit rate (122, 124) of a codec that produced the source encoded data.
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公开(公告)号:AT328431T
公开(公告)日:2006-06-15
申请号:AT00946673
申请日:2000-06-29
Applicant: ERICSSON TELEFON AB L M
Inventor: SUNDQVIST JIM
Abstract: A source encoder schedules its access to a network IP stack. During that scheduled period, the source encoder blocks the transport of non-real-time TCP and UDP packets. After at least one real-time packet has been transported through the stack, the source encoder allows the non-real-time packets to access the physical layer until the beginning of the next scheduled period. Consequently, the packet transport delay is reduced significantly, which results in a substantial increase in throughput.
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公开(公告)号:AU778401B2
公开(公告)日:2004-12-02
申请号:AU6744100
申请日:2000-08-11
Applicant: ERICSSON TELEFON AB L M
Inventor: SUNDQVIST JIM , HARASZTI ZSOLT , SVANBRO KRISTER , JONSSON LARS-ERIK , AYADURAI VICKNESAN
IPC: H04L29/06 , H04L12/56 , H04L12/801 , H04L12/00
Abstract: In packet communication paths that include header compression, header fields of packets to be communicated are altered. The alteration of the header fields does not disturb their functionality, and is transparent to the header compression scheme of the packet communication path. The altered header fields are provided for compression by the header compression scheme, resulting in improved performance of the header compression scheme. Performance improvements can also be achieved in packet communication paths that do not use header compression, by violating the integrity of header fields in packets to be transmitted over the packet communication path.
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公开(公告)号:AU6282801A
公开(公告)日:2001-12-11
申请号:AU6282801
申请日:2001-05-22
Applicant: ERICSSON TELEFON AB L M
Inventor: SUNDQVIST JIM , LENNESTAL HAKAN , NOHLGREN ANDERS , LINDQVIST MORGAN , WANG WEI
IPC: H04L12/64
Abstract: The present invention relates to a receiver system in a communication system supporting packet-based communicatior (e.g., an IP-network), including a receiver, speech decoder ( 40 ) and a jitter buffer ( 20 ) for handling delay variations in the reception of a speech signal consisting of packets containing frames with encoded speech. A jitter buffer controller ( 50 ) is provided for keeping information about the functional size of the jitter buffer ( 20 ) and for providing the speech decoder ( 40 ) with control information, such that the speech decoder ( 40 ), based on that information. provides a dynamic adaptation of the size of the jitter buffer ( 20 ) using the received encoded, packetized speech signal. The invention also relates to a method of adapting the functional size of the jitter buffer of a receiver system.
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公开(公告)号:AU5897301A
公开(公告)日:2001-11-20
申请号:AU5897301
申请日:2001-05-10
Applicant: ERICSSON TELEFON AB L M
Inventor: SVEDBERG JONAS , SUNDQVIST JIM , UVLIDEN ANDERS , NOHLGREN ANDERS , WESTERLUND MAGNUS
Abstract: An improved forward error correction (FEC) technique for coding speech data provides an encoder module which primary-encodes an input speech signal using a primary synthesis model to produce primary-encoded data, and redundant-encodes the input speech signal using a redundant synthesis model to produce redundant-encoded data. A packetizer combines the primary-encoded data and the redundant-encoded data into a series of packets and transmits the packets over a packet-based network, such as an Internet Protocol (IP) network. A decoding module primary-decodes the packets using the primary synthesis model, and redundant-decodes the packets using the redundant synthesis model. The technique provides interaction between the primary synthesis model and the redundant synthesis model during and after decoding to improve the quality of a synthesized output speech signal. Such "interaction," for instance, may take the form of updating states in one model using the other model.
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公开(公告)号:DE69937613D1
公开(公告)日:2008-01-03
申请号:DE69937613
申请日:1999-12-07
Applicant: ERICSSON TELEFON AB L M
Inventor: SOERQVIST PATRIK , SVENSSON TOMAS , SUNDQVIST JIM , ERIKSSON ANDERS
Abstract: Adaptive gain control techniques provide correctly adjusted audio signal levels during the entirety of an Internet telephony conversation and are resilient to background noise and loudspeaker echo. Disclosed techniques can account for multiple near-end speakers, as well as changes in near-end environment. In an exemplary embodiment, an adaptive gain controller includes a gain control processor configured to adjust an analog gain for a microphone output signal based on measurements of the microphone output signal and on measurements of a loudspeaker input signal.
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公开(公告)号:DE60027177T2
公开(公告)日:2006-12-28
申请号:DE60027177
申请日:2000-09-01
Applicant: ERICSSON TELEFON AB L M
Inventor: SUNDQVIST JIM , FRANKKILA TOMAS , NOHLGREN ANDERS
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公开(公告)号:ES2262323T3
公开(公告)日:2006-11-16
申请号:ES99927021
申请日:1999-04-28
Applicant: ERICSSON TELEFON AB L M
Inventor: EKUDDEN ERIK , SVENSSON TOMAS , SUNDQVIST JIM
Abstract: Un método de ocultación de paquetes retrasados en una red de paquetes que usa los parámetros de la trama retrasada para mejorar la ocultación de los paquetes retrasados, caracterizado por realizar los siguientes pasos cuando se retrasa un paquete: guardar una copia del estado inicial del decodificador que existe después de decodificar el último paquete no retrasado (S8); decodificar y disimular el paquete retrasado usando los parámetros de trama predichos y dicho estado inicial del decodificador, produciendo por ello un estado inicial del decodificador erróneo (S4, S5); decodificar usando los parámetros de trama actualizados de dicho paquete retrasado y dicho estado inicial del decodificador guardado, produciendo por ello un estado inicial del decodificador corregido (S11); decodificar usando un conjunto de parámetros de trama y dicho estado inicial erróneo del decodificador, produciendo por ello una primera señal de salida y además un estado inicial del decodificador erróneo (S12); decodificar usando dicho conjunto de parámetros de trama y dicho estado inicial del decodificador corregido, produciendo por ello una segunda señal de salida (S13); y combinar dichas primera y segunda señales de salida (S14).
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公开(公告)号:ES2261216T3
公开(公告)日:2006-11-16
申请号:ES00946673
申请日:2000-06-29
Applicant: ERICSSON TELEFON AB L M
Inventor: SUNDQVIST JIM
Abstract: Un método para programar transporte de paquetes en una pila de red de telecomunicaciones, que comprende las etapas de: seleccionar un intervalo de tiempo predeterminado; durante dicho intervalo de tiempo predeterminado seleccionado, bloquear una transmisión de paquetes en tiempo virtual; y sensible a una determinación que al menos un paquete a tiempo real ha sido transportado a través de dicha pila de red durante dicho intervalo de tiempo predeterminado seleccionado, que permite al menos que uno de dichos paquetes en tiempo virtual acceda a una capa física de dicha pila de red.
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