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    • 1. 发明申请
    • AUDIO ENCODING WITH DIFFERENT CODING MODELS
    • 使用不同编码模式的音频编码
    • WO2005112004A1
    • 2005-11-24
    • PCT/IB2004/001579
    • 2004-05-17
    • NOKIA CORPORATIONMÄKINEN, JariLAKANIEMI, AriOJALA, Pasi
    • MÄKINEN, JariLAKANIEMI, AriOJALA, Pasi
    • G10L19/14
    • G10L19/20
    • The invention relates to a method for supporting an encoding of an audio signal, wherein at least a first and a second coder mode are available for encoding a section of the audio signal. The first coder mode enables a coding based on two different coding models. A selection of a coding model is enabled by a selection rule which is based on signal characteristics which have been determined for a certain analysis window. In order to avoid a misclassification of a section after a switch to the first coder mode, it is proposed that the selection rule is activated only when sufficient sections for the analysis window have been received. The invention relates equally to a module 2,3 in which this method is implemented, to a device 1 and a system comprising such a module 2,3, and to a software program product including a software code for realizing the proposed method.
    • 本发明涉及一种用于支持音频信号的编码的方法,其中至少第一和第二编码器模式可用于对音频信号的一部分进行编码。 第一编码器模式能够进行基于两种不同编码模型的编码。 编码模型的选择通过基于已经为特定分析窗口确定的信号特性的选择规则来实现。 为了避免在切换到第一编码器模式之后对一个部分进行错误分类,建议仅在接收到用于分析窗口的足够部分时激活选择规则。 本发明同样涉及其中实施方法的装置1和包括这种模块2,3的系统的模块2,3以及包括用于实现所提出的方法的软件代码的软件程序产品。
    • 3. 发明申请
    • SCALABLE ERROR DETECTION AND CROSS-SESSION TIMING SYNCHRONIZATION FOR PACKET-SWITCHED TRANSMISSION
    • 用于分组交换传输的可扩展错误检测和交叉时序同步
    • WO2009037672A2
    • 2009-03-26
    • PCT/IB2008/053823
    • 2008-09-20
    • NOKIA CORPORATIONNOKIA INCOJALA, PasiLAKANIEMI, Ari
    • OJALA, PasiLAKANIEMI, Ari
    • H04L1/0083H04L1/0061H04L1/008
    • A system and method for providing improved scalable error detection and cross- timing synchronization for packet-switched transmission. In one embodiment, checksum error detection is applied for the core layer and for enhancement layers of the scalable payload in such a way that dropping one or several enhancement layers from the payload does not change the value of the checksum. Only one checksum is transmitted, e.g., in the payload or in the header of the lower-layer protocol. The transmitter modifies the encoded bit stream in such a manner that the entity in the network deploying the scalable payload and removing layers from the packet does not need to recalculate the checksum placed in the payload or packet header, even when the payload size is changed. A prefix/tail bit field is added in the beginning/end of each enhancement layer to make the checksum check match with the common checksum. In another embodiment, the receiver may check the correctness of each received layer simultaneously and, if desired, utilize data only from correctly received layers.
    • 一种用于为分组交换传输提供改进的可缩放错误检测和交叉定时同步的系统和方法。 在一个实施例中,以核心层和可扩展有效负载的增强层应用校验和错误检测,使得从有效载荷丢弃一个或几个增强层不会改变校验和的值。 仅传送一个校验和,例如在有效载荷或低层协议的报头中。 发射机以这样的方式修改编码比特流,使得即使当有效载荷大小改变时,部署可扩展有效载荷并从分组移除层的实体也不需要重新计算放置在有效载荷或分组报头中的校验和。 在每个增强层的开始/结束处添加前缀/尾部位字段,以使校验和校验与公共校验和匹配。 在另一个实施例中,接收机可以同时检查每个接收层的正确性,并且如果需要,仅利用来自正确接收的层的数据。
    • 7. 发明申请
    • SPEECH DECODER WITH ERROR CONCEALMENT
    • 具有错误隐含的语音解码器
    • WO2010000303A1
    • 2010-01-07
    • PCT/EP2008/058400
    • 2008-06-30
    • NOKIA CORPORATIONOJALA, PasiLAKANIEMI, Ari
    • OJALA, PasiLAKANIEMI, Ari
    • G10L19/00
    • G10L19/005
    • An apparatus for decoding a speech signal configured to: detect in a bitstream comprising a plurality of speech data frames the absence of at least one speech data frame; determine the absent at least one speech data frame is associated with a transition from a silence signal type frame to a speech signal type frame; adapt at least one speech parameter associated with at least one speech data frame succeeding the absent at least one speech data frame dependent on the determination of the absent at least one speech data frame is associated with the transition from the silence signal type frame to the speech signal type frame.
    • 一种用于解码语音信号的装置,被配置为:在包含多个语音数据帧的比特流中检测不存在至少一个语音数据帧; 确定缺少的至少一个语音数据帧与从静默信号类型帧到语音信号类型帧的转换相关联; 与至少一个语音数据帧相关联的至少一个语音参数与至少一个语音数据帧相关联的至少一个语音参数与从无声信号类型帧到语音的转换相关联,所述至少一个语音参数与不存在的至少一个语音数据帧相关, 信号类型帧。
    • 8. 发明申请
    • SYSTEM AND METHOD FOR PROVIDING AMR-WB DTX SYNCHRONIZATION
    • 用于提供AMR-WB DTX同步的系统和方法
    • WO2009027936A2
    • 2009-03-05
    • PCT/IB2008/053459
    • 2008-08-28
    • NOKIA CORPORATIONNOKIA INCOJALA, PasiLAKANIEMI, Ari
    • OJALA, PasiLAKANIEMI, Ari
    • G10L19/00H04J3/06
    • G10L19/012G10L19/005
    • A system and method for providing improved adaptive multi-rate wideband (AMR- WB) discontinuous transmission (DTX) synchronization. According to various embodiments, an indication on the start of the inactive speech period is signalled to the decoder via a voice activity detection (VAD) flag a predetermined number of frames before the DTX period will start, i.e., before the SID_FIRST frame is received. When the VAD flag indicates active speech, or when the VAD flag has been set to zero less than the predetermined number of frames ago, the received NO DATA frame can be classified with a high degree of reliability as active speech, i.e., considered as transmitter, network or terminal-initiated signalling, and can be substituted by a SPEECH_LOST frame. When the VAD flag was set to zero eight frames ago or earlier, the NO_DATA frame is classified as DTX.
    • 一种用于提供改进的自适应多速率宽带(AMR-WB)不连续传输(DTX)同步的系统和方法。 根据各种实施例,在DTX周期开始之前,即在接收到SID_FIRST帧之前,通过语音活动检测(VAD)标志将预定数量的帧在无效语音周期开始时的指示信号通知解码器。 当VAD标志指示有效语音时,或者当VAD标志已被设置为小于预定数目的帧以前的零时,所接收的NO DATA帧可以以高可靠性分类为活动语音,即被认为是发射机 ,网络或终端发起的信令,并且可以被SPEECH_LOST帧替代。 当VAD标志在八帧前或更早版本设置为零时,NO_DATA帧被分类为DTX。
    • 9. 发明申请
    • ADAPTIVE JITTER MANAGEMENT CONTROL IN DECODER
    • 解码器中的自适应抖动管理控制
    • WO2007132377A1
    • 2007-11-22
    • PCT/IB2007/051622
    • 2007-05-02
    • Nokia CorporationOJALA, PasiLAKANIEMI, Ari
    • OJALA, PasiLAKANIEMI, Ari
    • H04J3/06
    • H04J3/0632G10L19/005G10L19/012
    • A method, a chipset, a receiver, a transmitter, an electronic device and a system for enabling a control of jitter management of an audio signal is described, wherein the audio signal is distributed to a sequence of frames that are received via a packet switched network, the received frames comprising active audio frames and non-active audio frames, wherein a concatenation of subsequent active audio frames represents an active audio burst, wherein a discrete information of audio activity of the audio signal via the packet switched network is received, the end of an active audio burst is determined based on the received discrete information of audio activity, and wherein jitter compensation of the received frames is controlled on the basis of the determined end of an active audio burst. The invention further relates to a corresponding software program product storing a software code for controlling jitter management of an audio signal.
    • 描述了一种方法,芯片组,接收器,发射器,电子设备和用于使得能够控制音频信号的抖动管理的系统,其中音频信号被分配到经由分组交换接收的帧序列 网络,所接收的帧包括有源音频帧和非有效音频帧,其中随后的活动音频帧的级联表示活动音频突发,其中接收到经由分组交换网络的音频信号的音频活动的离散信息, 基于接收到的音频活动的离散信息来确定活动音频突发的结束,并且其中基于所确定的有效音频突发的结束来控制所接收的帧的抖动补偿。 本发明还涉及存储用于控制音频信号的抖动管理的软件代码的相应的软件程序产品。