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    • 1. 发明授权
    • Forward error correction in speech coding
    • 语音编码中的前向纠错
    • US06757654B1
    • 2004-06-29
    • US09569312
    • 2000-05-11
    • Magnus WesterlundAnders NohlgrenJonas SvedbergAnders UvlidenJim Sundqvist
    • Magnus WesterlundAnders NohlgrenJonas SvedbergAnders UvlidenJim Sundqvist
    • G10L1302
    • G10L19/005
    • 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.
    • 用于编码语音数据的改进的前向纠错(FEC)技术提供了一种编码器模块,其使用主要合成模型对输入语音信号进行一次编码以产生初始编码数据,并且使用冗余合成模型对输入语音信号进行冗余编码 以产生冗余编码数据。 分组器将主编码数据和冗余编码数据组合成一系列分组,并通过诸如因特网协议(IP)网络的基于分组的网络传送分组。 解码模块使用主合成模型对分组进行主要解码,并使用冗余合成模型对分组进行冗余解码。 该技术在解码期间和之后提供主要合成模型和冗余合成模型之间的交互以提高合成输出语音信号的质量。 例如,这种“交互”可以采用其他模型在一个模型中更新状态的形式。
    • 3. 发明授权
    • Method and apparatus in a telecommunications system
    • 电信系统中的方法和装置
    • US06873954B1
    • 2005-03-29
    • US09655326
    • 2000-09-05
    • Jim SundqvistTomas FrankkilaAnders Nohlgren
    • Jim SundqvistTomas FrankkilaAnders Nohlgren
    • H04J3/06G10L13/02
    • H04J3/0632
    • Audio artifacts due to overrun or underrun in a playout buffer caused by the sampling rates at a sending and receiving side not being at the same rate are reduced. An LPC-residual is modified on a sample-by-sample basis. The LPC-residual block, which includes N samples, is converted to a block comprising N+1 or N−1 samples. A sample rate controller decides whether samples should be added to or removed from the LPC-residual. The exact position at which to add respective remove samples is either chosen arbitrarily or found by searching for low energy segments in the LPC-residual. A speech synthesiser module then reproduces the speech. By using the proposed sample rate conversion method the playout buffer can be continuously controlled. Furthermore, since the method works on a sample-by-sample basis the buffer can be kept to a minimum and hence no extra delay is introduced.
    • 由于发送和接收侧的采样率不在同一速率引起的播放缓冲器中的超载或欠载的音频伪影减少。 LPC-残差在逐个样本的基础上修改。 包括N个采样的LPC残差块被转换为包括N + 1或N-1个采样的块。 采样率控制器决定是否应将样本添加到LPC残差中或从LPC残差中去除。 添加相应去除样本的确切位置是任意选择的或通过搜索LPC残差中的低能量段来找到。 语音合成器模块然后再现语音。 通过使用提出的采样率转换方法,可以连续控制播放缓冲器。 此外,由于该方法在逐个采样的基础上工作,因此缓冲器可以保持最小,因此不会引入额外的延迟。
    • 4. 发明授权
    • Method for sending information in a telecommunication system
    • 在电信系统中发送信息的方法
    • US06870876B1
    • 2005-03-22
    • US09612132
    • 2000-07-07
    • Anders NohlgrenJim Sundqvist
    • Anders NohlgrenJim Sundqvist
    • H04L29/08H04B14/04H04J3/06H04L7/00H04L12/64H04Q11/04H04B1/38H04L5/16
    • H04L12/64H04J3/0632
    • The invention is concerned with a method and an apparatus, wherein information data is sent between at least two transceivers in a telecommunication system. The information data is transmitted from the sender of a transceiver to the receiver of one or more other transceivers in form of digital signals having a given sampling frequency. The signals are played out by the receiver in a controlled way. The invention is mainly characterized by estimation of the sender's sampling rate at the sending side of a transceiver, transmitting the estimation to the receiving side of an another transceiver, and controlling the playout of the information data at the receiving side by means of the sampling rate estimated at the sending side to avoid delays and/or interrupts in the presentation. The invention is especially suitable in connection with packet based networks wherein the information data is sent between the transceivers in the telecommunication system in form of packet data frames, such as audio frames.
    • 本发明涉及一种方法和装置,其中信息数据在电信系统中的至少两个收发器之间发送。 信息数据从收发器的发送器以具有给定采样频率的数字信号的形式发送到一个或多个其他收发器的接收器。 接收机以受控的方式播放信号。 本发明的主要特征在于估计收发机发送侧的发送者采样率,将估计发送到另一收发机的接收侧,并通过采样率控制接收侧信息数据的播出 估计在发送端避免演示中的延迟和/或中断。 本发明特别适用于基于分组的网络,其中以诸如音频帧的分组数据帧的形式在电信系统中的收发机之间发送信息数据。
    • 7. 发明申请
    • Multimedia Session Management
    • 多媒体会话管理
    • US20070266122A1
    • 2007-11-15
    • US11666891
    • 2005-11-24
    • Torbjorn EinarssonAnders NohlgrenMagnus WesterlundMagnus Andersson
    • Torbjorn EinarssonAnders NohlgrenMagnus WesterlundMagnus Andersson
    • G06F15/177
    • H04N21/6408H04N21/4383H04N21/4384H04N21/643
    • A multimedia provider (100) generates, based on a request for a media channel from a user terminal (10), a session set-up description that is returned to the user terminal (10). The description defines, when processed in the terminal (10), a session window (20) that is displayed on a screen (12) of the terminal and comprises a region (22) for displaying media data of a requested channel and a channel region (24) comprising information on alternative available media channels. The description also defines a binding between user inputs (14) of the terminal (10) and identifiers of the alternative channels. The set-up description allows for a user-friendly channel switch by automatically transmitting a channel switch request to the multimedia provider (100) upon a triggering of the user input (14) that is associated with the identifier of the requested media channel. The description, thus, provides a visually appealing graphical interface and simultaneously allows for a user-friendly media channel switch.
    • 多媒体提供者(100)根据来自用户终端(10)的对媒体频道的请求生成返回给用户终端(10)的会话建立说明。 该描述定义了在终端(10)中处理的会话窗口(20),其显示在终端的屏幕(12)上,并且包括用于显示所请求的频道的媒体数据的区域(22)和信道区域 (24)包括关于替代可用媒体信道的信息。 该描述还定义了终端(10)的用户输入(14)和替代频道的标识符之间的绑定。 所述设置描述允许在触发与所请求的媒体信道的标识符相关联的用户输入(14)时,自动向多媒体提供商(100)发送频道切换请求,从而进行用户友好的频道切换。 因此,该描述提供了视觉吸引人的图形界面,同时允许用户友好的媒体频道切换。
    • 8. 发明授权
    • Multimedia session management
    • 多媒体会话管理
    • US09003041B2
    • 2015-04-07
    • US11666891
    • 2005-11-24
    • Torbjörn EinarssonAnders NohlgrenMagnus WesterlundMagnus Andersson
    • Torbjörn EinarssonAnders NohlgrenMagnus WesterlundMagnus Andersson
    • G06F15/16H04N21/6408H04N21/438H04N21/643
    • H04N21/6408H04N21/4383H04N21/4384H04N21/643
    • A multimedia provider (100) generates, based on a request for a media channel from a user terminal (10), a session set-up description that is returned to the user terminal (10). The description defines, when processed in the terminal (10), a session window (20) that is displayed on a screen (12) of the terminal and comprises a region (22) for displaying media data of a requested channel and a channel region (24) comprising information on alternative available media channels. The description also defines a binding between user inputs (14) of the terminal (10) and identifiers of the alternative channels. The set-up description allows for a user-friendly channel switch by automatically transmitting a channel switch request to the multimedia provider (100) upon a triggering of the user input (14) that is associated with the identifier of the requested media channel. The description, thus, provides a visually appealing graphical interface and simultaneously allows for a user-friendly media channel switch.
    • 多媒体提供者(100)根据来自用户终端(10)的对媒体频道的请求生成返回给用户终端(10)的会话建立说明。 该描述定义了在终端(10)中处理的会话窗口(20),其显示在终端的屏幕(12)上,并且包括用于显示所请求的频道的媒体数据的区域(22)和信道区域 (24)包括关于替代可用媒体信道的信息。 该描述还定义了终端(10)的用户输入(14)和替代频道的标识符之间的绑定。 所述设置描述允许在触发与所请求的媒体信道的标识符相关联的用户输入(14)时,自动向多媒体提供商(100)发送频道切换请求,从而进行用户友好的频道切换。 因此,该描述提供了视觉吸引人的图形界面,同时允许用户友好的媒体频道切换。