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1. 发明公开

EP3561808A1 METHOD AND APPARATUS FOR CONTROLLING AUDIO FRAME LOSS CONCEALMENT 有权
公开(公告)号：EP3561808A1
公开(公告)日：2019-10-30
申请号：EP19178384.4
申请日：2014-01-22
申请人： Telefonaktiebolaget LM Ericsson (publ)
发明人： BRUHN, Stefan , SVEDBERG, Jonas
IPC分类号： G10L19/005
摘要： In accordance with an example embodiment of the present invention, disclosed is a method and an apparatus thereof for adaptation of a frame loss concealment method in audio decoding. The method comprises detecting a transient in a previously received and reconstructed audio signal and modifying the concealment method by selectively adjusting a magnitude of a substitution frame spectrum. The method further comprises detecting a burst loss and further modifying the concealment method by selectively adjusting a magnitude of a substitution frame spectrum.

2. 发明公开

EP3155616A1 BURST FRAME ERROR HANDLING 审中-公开
标题翻译： FEHLERHANDHABUNG EINES BURST-RAHMENS
公开(公告)号：EP3155616A1
公开(公告)日：2017-04-19
申请号：EP15733938.3
申请日：2015-06-08
申请人： Telefonaktiebolaget LM Ericsson (publ)
发明人： BRUHN, Stefan
IPC分类号： G10L19/005
CPC分类号： G10L19/005 , G10L19/028
摘要： There is provided mechanisms for frame loss concealment. A method is performed by a receiving entity. The method comprises adding, in association with constructing a substitution frame for a lost frame, a noise component to the substitution frame. The noise component has a frequency characteristic corresponding to a low-resolution spectral representation of a signal in a previously received frame.
摘要翻译：提供了帧丢失隐藏机制。一种方法由接收实体执行。该方法包括将与丢失帧的替换帧相关联地添加到替代帧的噪声分量。噪声分量具有对应于先前接收的帧中的信号的低分辨率频谱表示的频率特性。

3. 发明公开

EP3143620A1 AUDIO SIGNAL CLASSIFICATION AND CODING 审中-公开
标题翻译： KLASSIFIZIERUNG UND CODIERUNG VON TONSIGNALEN
公开(公告)号：EP3143620A1
公开(公告)日：2017-03-22
申请号：EP15726394.8
申请日：2015-05-12
申请人： Telefonaktiebolaget LM Ericsson (publ)
发明人： NORVELL, Erik , BRUHN, Stefan
IPC分类号： G10L25/18 , G10L19/20
CPC分类号： G10L19/20 , G10L25/18
摘要： The invention relates to a codec and a signal classifier and methods therein for signal classification and selection of a coding mode based on audio signal characteristics. A method embodiment to be performed by a decoder comprises, for a frame m: determining a stability value D(m) based on a difference, in a transform domain, between a range of a spectral envelope of frame m and a corresponding range of a spectral envelope of an adjacent frame m−1. Each such range comprises a set of quantized spectral envelope values related to the energy in spectral bands of a segment of the audio signal. The method further comprises selecting a decoding mode, out of a plurality of decoding modes, based on the stability value D(m); and applying the selected decoding mode.
摘要翻译：本发明涉及一种编解码器和信号分类器及其方法，用于信号分类和基于音频信号特性的编码模式的选择。由解码器执行的方法实施例包括：对于帧m：在变换域中，基于差异来确定帧m的频谱包络的范围与相应的帧的范围之间的稳定值D（m）相邻帧m-1的频谱包络。每个这样的范围包括与音频信号的段的频谱带中的能量相关的一组量化频谱包络值。该方法还包括基于稳定性值D（m）从多个解码模式中选择解码模式; 以及应用所选择的解码模式。

4. 发明公开

EP3086319A1 METHODS AND APPARATUSES FOR DTX HANGOVER IN AUDIO CODING 有权
标题翻译：音频编码中用于DTX吊挂的方法和设备
公开(公告)号：EP3086319A1
公开(公告)日：2016-10-26
申请号：EP16173655.8
申请日：2013-12-12
申请人： Telefonaktiebolaget LM Ericsson (publ)
发明人： BRUHN, Stefan , SEHLSTEDT, Martin , JANSSON TOFTGÅRD, Tomas
IPC分类号： G10L19/012
CPC分类号： G10L19/012 , G10L25/84
摘要： Transmitting node and receiving node for audio coding and methods therein. The nodes being operable to encode/decode speech and to apply a discontinuous transmission (DTX) scheme comprising transmission/reception of Silence Insertion Descriptor (SID) frames during speech inactivity. The method in an encoder comprising determining a number N of hangover frames being representative of background noise, wherein the number N of hangover frames is variable, and further transmitting the N hangover frames to a decoder. The method further comprises transmitting a first SID frame to the after the transmission of the N hangover frames, where the first SID frame comprises information indicating the determined number N of hangover frames to the decoder. The method enables the decoder to generate comfort noise based on the hangover frames most adequate for the purpose.
摘要翻译：发送节点和接收节点用于音频编码及其中的方法。所述节点可操作以对语音进行编码/解码并且应用包括在语音不活动期间发送/接收静默插入描述符（SID）帧的不连续传输（DTX）方案。编码器中的方法包括确定代表背景噪声的拖尾帧的数量N，其中拖尾帧的数量N是可变的，并且进一步将N个拖尾帧发送到解码器。该方法进一步包括将第一SID帧发送到传输N个拖尾帧之后，其中第一SID帧包括指示向解码器确定的拖尾帧的数目N的信息。该方法使得解码器能够基于最适合于该目的的宿醉框架产生舒适噪声。

5. 发明公开

EP2954516A1 ENHANCED AUDIO FRAME LOSS CONCEALMENT 审中-公开
标题翻译： VERBESSERTE AUDIO-RAHMENVERLUSTÜBERBRÜCKUNG
公开(公告)号：EP2954516A1
公开(公告)日：2015-12-16
申请号：EP14704703.9
申请日：2014-01-22
申请人： Telefonaktiebolaget LM Ericsson (PUBL)
发明人： BRUHN, Stefan
IPC分类号： G10L19/005
CPC分类号： G10L19/005 , G10L19/04 , G10L25/69
摘要： Concealing a lost audio frame of a received audio signal by performing a sinusoidal analysis (81) of a part of a previously received or reconstructed audio signal, wherein the sinusoidal analysis involves identifying frequencies of sinusoidal components of the audio signal, applying a sinusoidal model on a segment of the previously received or reconstructed audio signal, wherein said segment is used as a prototype frame in order to create a substitution frame for a lost audio frame, and creating the substitution frame (84) for the lost audio frame by time-evolving sinusoidal components of the prototype frame, up to the time instance of the lost audio frame, in response to the corresponding identified frequencies. The method further comprising performing (82)at least one of an enhanced frequency estimation and an adaptation of the creating of the substitution frame in response to the tonality of the audio signal.
摘要翻译：提供了一种通过对先前接收或重构的音频信号的一部分执行正弦分析来隐藏接收到的音频信号的丢失音频帧的方法。正弦分析涉及识别音频信号的正弦分量的频率，以及在先前接收或重构的音频信号的段上应用正弦模型。该片段用作原型帧，以便为丢失的音频帧创建替换帧。该方法包括响应于相应的识别频率，通过时变演化的原型帧的正弦分量创建丢失音频帧的替换帧，直到丢失音频帧的时间实例。该方法还包括响应于音频信号的音调来执行增强频率估计和替代帧的创建中的至少一个。

6. 发明授权

EP2132733B1 NON-CAUSAL POSTFILTER 有权
标题翻译：不是因果后滤波器
公开(公告)号：EP2132733B1
公开(公告)日：2012-03-07
申请号：EP07852271.1
申请日：2007-12-14
申请人： Telefonaktiebolaget LM Ericsson (publ)
发明人： BRUHN, Stefan
IPC分类号： G10L19/14 , G10L19/08 , G10L19/04 , G10L19/12 , G10L21/02
CPC分类号： G10L19/26
摘要： A decoder arrangement comprises a receiver input (40) for parameters (4) of frame-based coded signals and a decoder (20), arranged to provide frames of decoded audio signals (5) based on the parameters (4). The receiver input (40) and/ or the decoder (20) is arranged to establish a time difference between the occasion when parameters of a first frame is available at the receiver input (40) and the occasion when a decoded audio signal of the first frame is available at an output of the decoder (20), which time difference corresponds to at least one frame. A postfilter (30) is connected to the output of the decoder (20) and to the receiver input (40). The postfilter (30) is arranged to provide a filtering of the frames of decoded audio signals into an output signal (6) in response to parameters (4) of a respective subsequent frame.

7. 发明授权

EP2132732B1 POSTFILTER FOR LAYERED CODECS 失效
标题翻译：后置滤波器用于分层编解码器
公开(公告)号：EP2132732B1
公开(公告)日：2012-03-07
申请号：EP07852270.3
申请日：2007-12-14
申请人： Telefonaktiebolaget LM Ericsson (publ)
发明人： BRUHN, Stefan
IPC分类号： G10L19/14 , G10L19/04
CPC分类号： G10L19/26
摘要： A scalable decoder device (50) for signals representing audio comprises a primary decoder (21) connected to an input (40). The primary decoder (21) is arranged to provide a primary decoded signal (23) based on received parameters (4). A primary postfilter (31) is connected to the primary decoder (23) to provide a primary postfiltered signal (32). A secondary enhancement decoder (45) is connected to the input (40) and arranged to provide a secondary decoded enhancement signal (44). The device further comprises a combiner arrangement (55), arranged for combining the primary postfiltered signal (32) and a signal (53) based on the secondary decoded enhancement signal (44) into an output signal (6) to be provided at an output (6). The combining is made with an adaptable strength relation between contributions from the two signals. A method for decoding coded signals representing audio operates in analogy with the scalable decoder device (50).

8. 发明公开

EP2118889A1 METHOD AND ARRANGEMENT FOR CONTROLLING SMOOTHING OF STATIONARY BACKGROUND NOISE 有权
标题翻译：方法及控制装置EVENING平稳的背景噪音
公开(公告)号：EP2118889A1
公开(公告)日：2009-11-18
申请号：EP08712848.4
申请日：2008-02-27
申请人： Telefonaktiebolaget LM Ericsson (publ)
发明人： BRUHN, Stefan
IPC分类号： G10L19/00 , G10L21/02
CPC分类号： G10L19/087 , G10L19/012 , G10L19/06 , G10L19/26 , G10L21/02 , G10L21/0308 , G10L25/84 , G10L2021/02168
摘要： In a method of smoothing stationary background noise in a telecommunication speech session, initially receiving and decoding S10 a signal representative of a speech session, where the signal comprises both a speech component and a background noise component. Subsequently, providing S20 a noisiness measure for the signal, and adaptively S30 smoothing the background noise component based on the provided noisiness measure.

9. 发明公开

EP1982328A1 VARIABLE FRAME OFFSET CODING 审中-公开
标题翻译： CODIERUNG MIT VARIABLEM RAHMENOFFSET
公开(公告)号：EP1982328A1
公开(公告)日：2008-10-22
申请号：EP06701856.4
申请日：2006-02-06
申请人： Telefonaktiebolaget LM Ericsson (publ)
发明人： BRUHN, Stefan
IPC分类号： G10L19/02 , H04N7/52 , H04L1/12
CPC分类号： G10L19/022 , H04N21/2368 , H04N21/4341
摘要： An encoding/decoding method, an coder/decoder (codec) and a radio communication device utilize a variable offset coding technique. In accordance with the technique, the start of processing of a first frame is time offset in relation to the end of the processing of the frame that precedes the first frame, the time offset bringing about a time gap between the end of the preceding frame and the start of processing the first frame. A substitution signal is inserted in the time gap.
摘要翻译：编码/解码方法，编码器/解码器（编解码器）和无线电通信装置利用可变的偏移编码技术。根据该技术，第一帧的处理开始是与第一帧之前的帧的处理结束相关的时间偏移，时间偏移导致前一帧的结束与时间间隔的时间间隔开始处理第一帧。在时间间隙中插入替换信号。

10. 发明公开

EP1797553A1 METHODS AND ARRANGEMENTS FOR ADAPTIVE THRESHOLDS IN CODEC SELECTION 有权
标题翻译：方法和自适应阈值安排，编解码器选择
公开(公告)号：EP1797553A1
公开(公告)日：2007-06-20
申请号：EP05784634.7
申请日：2005-09-21
申请人： TELEFONAKTIEBOLAGET LM ERICSSON (publ)
发明人： HÅKANSSON, Stefan , BRUHN, Stefan , LUNDBERG, Tomas
IPC分类号： G10L19/14
CPC分类号： H04L1/0021 , G10L19/22 , H04L1/0014 , H04L1/0025 , H04L1/0026
摘要： A method and arrangement for dynamically adapting thresholds used for selecting a codec mode to be used is presented. Thresholds are adapted (240) in response to the current received signal quality. An estimate of actual prevailing received signal quality is obtained (230) on which the adaptation is based. The present invention can be applied either on the mobile terminal side or on the network side, working on the uplink and /or the downlink. The thresholds can be modified on the receiving side, or, when operating in the network and working on the downlink, the threshold adaptation can be initiated in the terminal.

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