专利快速检索-快速检索全球专利，免费商用专利数据库-IPRDB

1. 发明授权

US08069034B2 Method and apparatus for encoding an audio signal using multiple coders with plural selection models 有权
标题翻译：使用具有多个选择模型的多个编码器对音频信号进行编码的方法和装置
公开(公告)号：US08069034B2
公开(公告)日：2011-11-29
申请号：US11126380
申请日：2005-05-06
申请人： Jari Mäkinen , Ari Lakaniemi , Pasi Ojala
发明人： Jari Mäkinen , Ari Lakaniemi , Pasi Ojala
IPC分类号： G10L19/00 , G10L19/14 , G10L11/06 , G10L21/04
CPC分类号： G10L19/20
摘要： A method for supporting an encoding of an audio signal is shown, 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 in which this method is implemented, to a device and a system comprising such a module and to a software program product including a software code for realizing the proposed method.
摘要翻译：示出了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。本发明同样涉及实现该方法的模块，涉及包括这种模块的装置和系统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。

2. 发明授权

US07596486B2 Encoding an audio signal using different audio coder modes 有权
标题翻译：使用不同的音频编码器模式对音频信号进行编码
公开(公告)号：US07596486B2
公开(公告)日：2009-09-29
申请号：US10848971
申请日：2004-05-19
申请人： Pasi Ojala , Jari Mäkinen , Ari Lakaniemi
发明人： Pasi Ojala , Jari Mäkinen , Ari Lakaniemi
IPC分类号： G10L19/00
CPC分类号： G10L19/22
摘要： The invention relates to a method for supporting an encoding of an audio signal, wherein a first coder mode and a second coder mode are available for encoding a respective section of an audio signal. The second coder mode enables a coding of a respective section based on a first coding model, which requires for an encoding of a respective section only information from the section itself, and based on a second coding model, which requires for an encoding of a respective section in addition an overlap signal with information from a preceding section. After a switch from the first coder mode to the second coder mode, always the first coding model is used for encoding a first section of the audio signal. This section can then be employed to generate an artificial overlap signal for a subsequent section, which is possibly to be encoded with the second coding model.
摘要翻译：本发明涉及一种用于支持音频信号的编码的方法，其中第一编码器模式和第二编码器模式可用于对音频信号的相应部分进行编码。第二编码器模式使得能够基于第一编码模型来编码相应部分，该第一编码模型需要对来自部分本身的相应部分信息的编码，并且基于第二编码模型，其需要对相应部分进行编码再加上具有来自前一部分的信息的重叠信号。在从第一编码器模式切换到第二编码器模式之后，始终使用第一编码模型对音频信号的第一部分进行编码。然后可以使用该部分来产生可能用第二编码模型编码的后续部分的人造重叠信号。

3. 发明申请

US20090154495A1 SCALABLE ERROR DETECTION AND CROSS-SESSION TIMING SYNCHRONIZATION FOR PACKET-SWITCHED TRANSMISSION 有权
标题翻译：用于分组交换传输的可扩展错误检测和交叉时序同步
公开(公告)号：US20090154495A1
公开(公告)日：2009-06-18
申请号：US12234512
申请日：2008-09-19
申请人： Pasi Ojala , Ari Lakaniemi
发明人： Pasi Ojala , Ari Lakaniemi
IPC分类号： H04J3/22
CPC分类号： H04L1/0083 , H04L1/0061 , H04L1/008
摘要： 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.
摘要翻译：一种用于为分组交换传输提供改进的可伸缩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。

4. 发明申请

US20090063165A1 SYSTEM AND METHOD FOR PROVIDING AMR-WB DTX SYNCHRONIZATION 有权
标题翻译：用于提供AMR-WB DTX同步的系统和方法
公开(公告)号：US20090063165A1
公开(公告)日：2009-03-05
申请号：US12199735
申请日：2008-08-27
申请人： Pasi Ojala , Ari Lakaniemi
发明人： Pasi Ojala , Ari Lakaniemi
IPC分类号： G10L21/00
CPC分类号： G10L19/012 , G10L19/005
摘要： 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.
摘要翻译：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前的零时，接收到的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。

5. 发明申请

US20070186145A1 Controlling a time-scaling of an audio signal 有权
标题翻译：控制音频信号的时间缩放
公开(公告)号：US20070186145A1
公开(公告)日：2007-08-09
申请号：US11349745
申请日：2006-02-07
申请人： Pasi Ojala , Ari Lakaniemi
发明人： Pasi Ojala , Ari Lakaniemi
IPC分类号： G06F17/00
CPC分类号： G10L21/04 , H04J3/0632
摘要： For controlling a time-scaling of an audio signal, the audio signal being distributed to a sequence of frames that are received via a packet switched network, a change in a delay of received frames is detected. Moreover, an amount of time scaling that is to be applied to received frames for compensating for the detected change is determined. Further, a kind of the change is determined. Further, a length of a time window within which a time scaling of the determined amount is to be completed is determined depending on the determined kind of the change.
摘要翻译：为了控制音频信号的时间缩放，音频信号被分配到经由分组交换网络接收到的帧序列，检测到接收帧的延迟的改变。此外，确定要应用于接收到的帧以补偿检测到的变化的时间缩放量。此外，确定了一种变化。此外，根据确定的改变的种类来确定要在其中完成所确定的量的时间缩放的时间窗口的长度。

6. 发明申请

US20050246164A1 Coding of audio signals 审中-公开
标题翻译：音频信号的编码
公开(公告)号：US20050246164A1
公开(公告)日：2005-11-03
申请号：US11107334
申请日：2005-04-15
申请人： Pasi Ojala , Jari Makinen , Ari Lakaniemi
发明人： Pasi Ojala , Jari Makinen , Ari Lakaniemi
IPC分类号： G10L20060101 , G10L19/24 , G10L25/18 , G10L19/14
CPC分类号： G10L19/24 , G10L25/18
摘要： An encoder comprises an input for inputting frames of an audio signal in a frequency band, an analysis filter dividing the frequency band into lower and higher frequency bands, a first encoding block for encoding the audio signals of the lower frequency band, a second encoding block for encoding the audio signals of the higher frequency band, and a mode selector for selecting an operating mode for the encoder among at least a first mode where signals only on the lower frequency band are encoded, and a second mode where signals on both the lower and higher frequency band are encoded. The encoder has a scaler to gradually change the encoding properties of the second encoding block in connection with a change in the operating mode of the encoder. The invention also relates to a device, a decoder, a method, a module, a computer program product, and a signal.
摘要翻译：编码器包括用于输入频带中的音频信号的帧的输入，将频带划分为较低和较高频带的分析滤波器，用于对较低频带的音频信号进行编码的第一编码块，第二编码块用于对较高频带的音频信号进行编码的模式选择器，以及用于在至少第一模式中选择编码器的操作模式的模式选择器，其中仅在较低频带上的信号被编码，以及第二模式，并且对频带进行编码。编码器具有缩放器，以与编码器的操作模式的改变相结合地逐渐改变第二编码块的编码属性。本发明还涉及一种设备，解码器，方法，模块，计算机程序产品和信号。

7. 发明授权

US08570869B2 Scalable error detection and cross-session timing synchronization for packet-switched transmission 有权
标题翻译：分组交换传输的可扩展错误检测和跨会话定时同步
公开(公告)号：US08570869B2
公开(公告)日：2013-10-29
申请号：US12234512
申请日：2008-09-19
申请人： Pasi Ojala , Ari Lakaniemi
发明人： Pasi Ojala , Ari Lakaniemi
IPC分类号： H04J3/14
CPC分类号： H04L1/0083 , H04L1/0061 , H04L1/008
摘要： Systems, methods and computer program codes are provided to facilitate error detection and timing synchronization of scalable data transmissions. To this end, checksum error detection is applied to the core layer and enhancement layers of the scalable payload data in such a way that dropping one or several enhancement layers from the payload data 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.
摘要翻译：提供系统，方法和计算机程序代码以便于可伸缩数据传输的错误检测和定时同步。为此，将校验和错误检测应用于可扩展有效载荷数据的核心层和增强层，使得从有效载荷数据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。

8. 发明授权

US08566108B2 Synchronization of multiple real-time transport protocol sessions 失效
标题翻译：多个实时传输协议会话的同步
公开(公告)号：US08566108B2
公开(公告)日：2013-10-22
申请号：US12746076
申请日：2007-12-03
申请人： Pasi Ojala , Ari Lakaniemi
发明人： Pasi Ojala , Ari Lakaniemi
IPC分类号： G10L19/00 , G10L21/04 , H04B1/66
CPC分类号： H04L65/607 , G10L19/167 , G10L19/24 , H04L65/608
摘要： A packet generator for generating packets from an input signal configured to: generate at least one first signal, dependent on the input signal, the first signal comprising a first relative time value; generate at least one second signal, dependent on the input signal and associated with the at least one first signal; and generate at least one indicator associated with each of the at least one second signal, each indicator dependent on the first relative time value.
摘要翻译：一种分组生成器，用于根据输入信号产生分组，所述分组生成器被配置为：根据所述输入信号生成至少一个第一信号，所述第一信号包括第一相对时间值; 根据输入信号产生至少一个第二信号并与至少一个第一信号相关联; 并且产生与所述至少一个第二信号中的每一个相关联的至少一个指示符，每个指示符取决于所述第一相对时间值。

9. 发明申请

US20080175276A1 Buffering Time Determination 有权
标题翻译：缓冲时间确定
公开(公告)号：US20080175276A1
公开(公告)日：2008-07-24
申请号：US11624836
申请日：2007-01-19
申请人： Ari Lakaniemi , Pasi Ojala
发明人： Ari Lakaniemi , Pasi Ojala
IPC分类号： H04J3/06
CPC分类号： H04J3/0632
摘要： This invention relates to methods, a computer program product and apparatuses in the context of frame buffering. A buffering time for one or more frames received by a frame buffer is determined based on a specific buffering time associated with a specific frame and on information representative of a specific amount of data stored in the frame buffer.
摘要翻译：本发明涉及在帧缓冲的上下文中的方法，计算机程序产品和装置。基于与特定帧相关联的特定缓冲时间和表示存储在帧缓冲器中的特定数据量的信息来确定由帧缓冲器接收的一个或多个帧的缓冲时间。

10. 发明授权

US08374860B2 Method, apparatus, system and software product for adaptation of voice activity detection parameters based oncoding modes 有权
标题翻译：基于编码模式适应语音活动检测参数的方法，装置，系统和软件产品
公开(公告)号：US08374860B2
公开(公告)日：2013-02-12
申请号：US13248213
申请日：2011-09-29
申请人： Kari Jarvinen , Pasi Ojala , Ari Lakaniemi
发明人： Kari Jarvinen , Pasi Ojala , Ari Lakaniemi
IPC分类号： G10L11/02 , G10L19/00
CPC分类号： G10L25/93 , G10L19/18 , G10L25/78
摘要： Encoding audio signals with selecting an encoding mode for encoding the signal categorizing the signal into active segments having voice activity and non-active segments having substantially no voice activity by using categorization parameters depending on the selected encoding mode and encoding at least the active segments using the selected encoding mode.
摘要翻译：通过选择编码模式来编码音频信号，用于对信号进行编码，将信号分类为具有语音活动的活动段和基本上不具有语音活动的非活动段，其通过使用根据所选择的编码模式的分类参数，并且至少使用选择编码模式。

你已经成功收藏专利！

检索式保存成功!

IPRDB

热门服务

关于我们

友情链接

联系方式