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    • 4. 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。
    • 5. 发明申请
    • Method and arrangements in an IP network
    • IP网络中的方法和布置
    • US20070147344A1
    • 2007-06-28
    • US11315293
    • 2005-12-23
    • Jim SundqvistErik Lundgren
    • Jim SundqvistErik Lundgren
    • H04L12/66
    • H04L65/4076H04L12/1877H04L12/1881H04L29/06027H04L47/15H04L47/2416H04L47/70H04L47/806H04L47/822
    • An IP network includes a network resource manager having a resource utilization map adapted to manage network resources in an IP network and an application framework having elements for receiving a request for a multicast distribution from a client in the IP network, and elements for requesting network resources from a network resource manager for a Media Quick Start, to start the requested multicast distribution. Further, the network resource manager includes elements for providing the application framework with feedback information relating to network resource availability from the network resource manager, and the application framework includes elements for receiving the feedback information and elements for allowing the Media Quick start or to use another behavior to start the requested multicast distribution based on the received feedback information.
    • IP网络包括网络资源管理器,其具有适于管理IP网络中的网络资源的资源利用率映射图,以及具有用于从IP网络中的客户端接收对多播分发的请求的元素的应用框架,以及用于请求网络资源的元素 从媒体快速启动的网络资源管理器启动请求的多播分发。 此外,网络资源管理器包括用于向应用框架提供与网络资源管理器相关的网络资源可用性的反馈信息的元件,并且应用框架包括用于接收反馈信息的元件和用于允许媒体快速启动或使用另一个 基于接收到的反馈信息来启动所请求的多播分发的行为。
    • 6. 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。
    • 7. 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。