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11. 发明授权

US08032370B2 Method, apparatus, system and software product for adaptation of voice activity detection parameters based on the quality of the coding modes 有权
标题翻译：用于根据编码模式的质量来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：US08032370B2
公开(公告)日：2011-10-04
申请号：US11431423
申请日：2006-05-09
申请人： Kari Järvinen , Pasi Ojala , Ari Lakaniemi
发明人： Kari Järvinen , 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 for Discontinuous 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 quality of the selected encoding mode and encoding at least the active segments using the selected encoding mode that for a low quality encoding produce a lower number of “active” temporal section detections than for a high quality encoding mode, with comfort noise parameters producing less contrast from background noise for low quality encoding than for high quality modes.
摘要翻译：通过选择用于对信号进行编码的编码模式来编码音频信号，以将信号分类为具有语音活动的活动段和基本上没有语音活动的非活动段，所述信号根据所选择的编码模式的质量使用分类参数并至少编码对于低质量编码，使用所选择的编码模式的活动段对于高质量编码模式产生较少数量的“活动”时间段检测，其中舒适噪声参数从低质量编码的背景噪声产生比对高质量编码模式更低的对比度质量模式

12. 发明申请

US20060088093A1 Packet loss compensation 审中-公开
标题翻译：丢包补偿
公开(公告)号：US20060088093A1
公开(公告)日：2006-04-27
申请号：US11028580
申请日：2005-01-05
申请人： Ari Lakaniemi , Pasi Ojala
发明人： Ari Lakaniemi , Pasi Ojala
IPC分类号： H04N11/04 , H04N11/02 , H04N7/12 , H04B1/66
CPC分类号： G10L19/005 , G10L19/24 , H04L1/0041 , H04L1/0075 , H04L1/0083
摘要： The invention relates to enabling a compensation of packet losses in a packet based transmission of data frames, wherein packets provided for transmission include a first type of frames corresponding to a respective data frame encoded using a first bit rate coding mode and a second type of frames corresponding to a respective data frame encoded using a second bit rate coding mode. In order to limit the processing power in the packet generation, parameters are extracted from a data frame which is to be transmitted in accordance with the first bit rate coding mode. The extracted parameters are quantized in accordance with the first bit rate coding mode to obtain quantized parameters forming a frame of the first type. In addition, a frame of the second type is generated based on the parameters extracted for the frame of the first type and/or on the quantized parameters of the frame of the first type.
摘要翻译：本发明涉及能够对基于分组的数据帧传输中的分组丢失进行补偿，其中提供用于传输的分组包括与使用第一比特率编码模式和第二类型帧编码的相应数据帧相对应的第一类型的帧对应于使用第二比特率编码模式编码的相应数据帧。为了限制分组生成中的处理能力，根据第一比特率编码模式从要发送的数据帧中提取参数。提取的参数根据第一比特率编码模式被量化，以获得形成第一类型的帧的量化参数。此外，基于为第一类型的帧提取的参数和/或第一类型的帧的量化参数来生成第二类型的帧。

13. 发明申请

US20050261900A1 Supporting a switch between audio coder modes 有权
标题翻译：支持音频编码器模式之间的切换
公开(公告)号：US20050261900A1
公开(公告)日：2005-11-24
申请号：US10848971
申请日：2004-05-19
申请人： Pasi Ojala , Jari Makinen , Ari Lakaniemi
发明人： Pasi Ojala , Jari Makinen , Ari Lakaniemi
IPC分类号： G10L19/14 , G10L19/12
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。

14. 发明申请

US20120084082A1 Adaptive Voice Activity Detection 有权
标题翻译：自适应语音活动检测
公开(公告)号：US20120084082A1
公开(公告)日：2012-04-05
申请号：US13248213
申请日：2011-09-29
申请人： Kari Järvinen , Pasi Ojala , Ari Lakaniemi
发明人： Kari Järvinen , Pasi Ojala , Ari Lakaniemi
IPC分类号： G10L11/06
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。

15. 发明授权

US08090588B2 System and method for providing AMR-WB DTX synchronization 有权
标题翻译：提供AMR-WB DTX同步的系统和方法
公开(公告)号：US08090588B2
公开(公告)日：2012-01-03
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。

16. 发明授权

US07983309B2 Buffering time determination 有权
标题翻译：缓冲时间确定
公开(公告)号：US07983309B2
公开(公告)日：2011-07-19
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。

17. 发明授权

US07899191B2 Synthesizing a mono audio signal 有权
标题翻译：合成单声道音频信号
公开(公告)号：US07899191B2
公开(公告)日：2011-03-01
申请号：US10592255
申请日：2004-03-12
申请人： Ari Lakaniemi , Pasi Ojala
发明人： Ari Lakaniemi , Pasi Ojala
IPC分类号： H04R5/00
CPC分类号： G10L19/008
摘要： The invention relates to a method of synthesizing a mono audio signal 3 based on an available encoded multichannel audio signal 2. The encoded multichannel audio signal 2 is assumed to comprise at least for a part of an audio frequency band separate parameter values for each channel of the multichannel audio signal. In order to reduce the processing load in synthesizing the mono audio signal 2, it is proposed that the parameter values of the multiple channels are combined at least for a part of an audio frequency band in the parameter domain. The combined parameter values are then used for synthesizing the mono audio signal. The invention relates equally to a corresponding audio decoder, to a corresponding coding system and to a corresponding software program product.
摘要翻译：本发明涉及一种基于可用编码的多声道音频信号2合成单声道音频信号3的方法。假设编码的多声道音频信号2至少包括音频频带的一部分，每个频道多声道音频信号。为了减少合成单声道音频信号2的处理负荷，提出至少对参数域中的音频频带的一部分组合多个声道的参数值。然后，组合的参数值用于合成单声道音频信号。本发明同样涉及对应的音频解码器，相应的编码系统和相应的软件程序产品。

18. 发明申请

US20080114606A1 Time scaling of multi-channel audio signals 失效
标题翻译：多声道音频信号的时间缩放
公开(公告)号：US20080114606A1
公开(公告)日：2008-05-15
申请号：US11584011
申请日：2006-10-18
申请人： Pasi Ojala , Ari Lakaniemi , Jussi Virolainen
发明人： Pasi Ojala , Ari Lakaniemi , Jussi Virolainen
IPC分类号： G10L19/00
CPC分类号： H04S3/004 , G10L19/008 , G10L21/04 , H04L47/10 , H04L47/22 , H04L47/283 , H04M3/56 , H04M3/568 , H04M7/006 , H04R27/00 , H04R2227/003 , H04S2420/01 , H04S2420/03
摘要： A method and related apparatus comprising: buffering an encoded audio input signal comprising at least one combined signal of a plurality of audio channels and one or more corresponding sets of side information parameters describing a multi-channel sound image; changing the length of at least one audio frame of said combined signal by adding or removing a segment of said combined signal; modifying said one or more sets of side information parameters with a change corresponding to the change in the length of said at least one audio frame of said combined signal; and transferring said at least one audio frame of said combined signal with a changed length and said modified one or more sets of side information parameters to a further processing unit.
摘要翻译：一种方法和相关装置，包括：缓冲编码的音频输入信号，其包括多个音频通道的至少一个组合信号和描述多声道声音图像的一个或多个对应的侧面信息参数组; 通过添加或去除所述组合信号的段来改变所述组合信号的至少一个音频帧的长度; 用对应于所述组合信号的所述至少一个音频帧的长度变化的改变来修改所述一组或多组侧信息参数; 以及将具有改变长度的所述组合信号的所述至少一个音频帧和所述修改的一组或多组侧信息参数传送到另一处理单元。

19. 发明申请

US20080101398A1 Transmission scheme dependent control of a frame buffer 审中-公开
标题翻译：传输方案依赖于帧缓冲器的控制
公开(公告)号：US20080101398A1
公开(公告)日：2008-05-01
申请号：US11591287
申请日：2006-10-31
申请人： Pasi Ojala , Ari Lakaniemi
发明人： Pasi Ojala , Ari Lakaniemi
IPC分类号： H04L12/56 , H04J3/22
CPC分类号： H04L47/10 , H04L47/26 , H04L47/30 , H04L47/36
摘要： This invention relates to a method, a computer program product, apparatuses and a system for controlling a length of a frame buffer. The frame buffer is comprised in a receiver and buffers frames that are transmitted by a transmitter according to a frame transmission scheme and received at the receiver. The length of the frame buffer is controlled under consideration of a change in the frame transmission scheme.
摘要翻译：本发明涉及一种用于控制帧缓冲器的长度的方法，计算机程序产品，装置和系统。帧缓冲器包括在接收机中，并且缓冲由发射机根据帧传输方案传输并在接收机处接收的帧。考虑帧传输方案的变化来控制帧缓冲器的长度。

20. 发明申请

US20080077410A1 SYSTEM AND METHOD FOR PROVIDING REDUNDANCY MANAGEMENT 审中-公开
标题翻译：用于提供冗余管理的系统和方法
公开(公告)号：US20080077410A1
公开(公告)日：2008-03-27
申请号：US11861216
申请日：2007-09-25
申请人： Pasi Ojala , Ari Lakaniemi
发明人： Pasi Ojala , Ari Lakaniemi
IPC分类号： G10L19/00
CPC分类号： G10L19/167 , G10L19/24 , H04L69/24
摘要： A system and method for efficiently implementing redundancy management in speech coding applications. According to various embodiments of the present invention, a sending device selects a redundant transmission level that is suitable for a current transmission channel condition while, at the same time, selecting the most suitable bit rate from the available codec mode set. With embodiments of the present invention, the redundant transmission level that is used is always optimal with regard to the selected codec modes, and no re-negotiation of the codec is needed.
摘要翻译：一种用于在语音编码应用中有效实现冗余管理的系统和方法。根据本发明的各种实施例，发送设备选择适合于当前传输信道条件的冗余传输级别，同时从可用编解码器模式集合中选择最合适的比特率。利用本发明的实施例，所使用的冗余传输级别对于所选择的编解码器模式始终是最优的，并且不需要对编解码器的重新协商。

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