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1. 发明授权

US10744322B2 Cochlear implant, device for generating a control signal for a cochlear implant, device for generating a combination signal and combination signal and corresponding methods 有权
公开(公告)号：US10744322B2
公开(公告)日：2020-08-18
申请号：US12067222
申请日：2006-09-05
申请人： Bernd Edler , Andreas Buechner , Waldo Nogueira , Frank Klefenz
发明人： Bernd Edler , Andreas Buechner , Waldo Nogueira , Frank Klefenz
IPC分类号： A61N1/36 , H04R25/00 , A61N1/08
摘要： A cochlear implant for processing signal parameters which are adapted for controlling the cochlear implant, which are based on the audio signal and which enable generating a representation of the audio signal by the cochlear implant includes a receive interface which is implemented to receive the signal parameters and a nerve stimulator for processing the signal parameters to generate nerve cell stimulation signals based on the signal parameters. A device for generating a control signal for a cochlear implant on the basis of an audio signal includes a cochlear parameter extractor for analyzing the audio signal which is implemented to generate signal parameters as input information for the cochlear implant based on an analysis of the audio signal using a human hearing simulation model, and a transmit interface for transmitting the signal parameters to the cochlear implant.

2. 发明申请

US20130117015A1 AUDIO SIGNAL DECODER, AUDIO SIGNAL ENCODER, METHOD FOR DECODING AN AUDIO SIGNAL, METHOD FOR ENCODING AN AUDIO SIGNAL AND COMPUTER PROGRAM USING A PITCH-DEPENDENT ADAPTATION OF A CODING CONTEXT 有权
标题翻译：音频信号解码器，音频信号编码器，用于解码音频信号的方法，用于编码音频信号的方法和使用与编码语境相关的适应性的计算机程序
公开(公告)号：US20130117015A1
公开(公告)日：2013-05-09
申请号：US13608980
申请日：2012-09-10
申请人： Stefan BAYER , Tom BAECKSTROEM , Ralf GEIGER , Bernd EDLER , Sascha DISCH , Lars VILLEMOES
发明人： Stefan BAYER , Tom BAECKSTROEM , Ralf GEIGER , Bernd EDLER , Sascha DISCH , Lars VILLEMOES
IPC分类号： G10L19/00
CPC分类号： G10L19/022 , G10L19/0212 , G10L25/90
摘要： An audio signal decoder includes a context-based spectral value decoder configured to decode a codeword describing one or more spectral values or at least a portion of a number representation thereof in dependence on a context state. The audio signal decoder also includes a context state determinator configured to determine a current context state in dependence on one or more previously decoded spectral values and a time warping frequency-domain-to-time-domain converter configured to provide a time-warped time-domain representation of a given audio frame on the basis of a set of decoded spectral values provided by the context-based spectral value decoder and in dependence on the time warp information. The context-state determinator is configured to adapt the determination of the context state to a change of a fundamental frequency between subsequent audio frames. An audio signal encoder applies a comparable concept.
摘要翻译：音频信号解码器包括基于上下文的频谱值解码器，其被配置为根据上下文状态来解码描述一个或多个频谱值的码字或其数字表示的至少一部分。音频信号解码器还包括上下文状态确定器，其被配置为根据一个或多个先前解码的频谱值确定当前上下文状态，以及时间扭曲频域 - 时域转换器，被配置为提供时间扭曲的时域 - 基于由基于上下文的频谱值解码器提供的解码频谱值的集合并且依赖于时间扭曲信息，给定音频帧的域表示。上下文状态确定器被配置为使上下文状态的确定适应随后的音频帧之间的基频的改变。音频信号编码器应用可比较的概念。

3. 发明申请

US20090022220A1 METHOD AND APPARATUS FOR ENHANCED VIDEO CODING 审中-公开
标题翻译：用于增强视频编码的方法和装置
公开(公告)号：US20090022220A1
公开(公告)日：2009-01-22
申请号：US11911505
申请日：2006-04-13
申请人： Yuri Vatis , Bernd Edler , Ingolf Wassermann , Dieu Thanh Nguyen , Joern Ostermann
发明人： Yuri Vatis , Bernd Edler , Ingolf Wassermann , Dieu Thanh Nguyen , Joern Ostermann
IPC分类号： H04N7/12
CPC分类号： H04N19/59 , H04N19/117 , H04N19/136 , H04N19/174 , H04N19/176 , H04N19/182 , H04N19/187 , H04N19/196 , H04N19/31 , H04N19/33 , H04N19/46 , H04N19/463 , H04N19/51 , H04N19/523 , H04N19/593 , H04N19/61
摘要： Standard video compression techniques apply motion-compensated prediction combined with transform coding of the prediction error. In the context of prediction with fractional-pel motion vector resolution it was shown, that aliasing components contained in an image signal are limiting the prediction efficiency obtained by motion compensation. In order to consider aliasing, quantization and motion estimation errors, camera noise, etc., we analytically developed a two dimensional (2D) non-separable interpolation filter, which is independently calculated for each frame by minimizing the prediction error energy. For every fractional-pel position to be interpolated, an individual set of 2D filter coefficients is determined. Since transmitting filter coefficients as side information results in an additional bit rate, which is almost constant for different image resolutions and total bit rates, the loss in coding gain increases when total bit rates sink. Therefore, we developed an algorithm, which regards the non-separable two-dimensional filter as a polyphase filter. For each frame, predicting the interpolation filter impulse response through evaluation of the polyphase filter, we only have to encode the prediction error of the filter coefficients.
摘要翻译：标准视频压缩技术应用运动补偿预测与预测误差的变换编码相结合。在分数像素运动矢量分辨率的预测的上下文中，显示了包含在图像信号中的混叠分量限制了通过运动补偿获得的预测效率。为了考虑混叠，量化和运动估计误差，相机噪声等，我们分析开发了一种二维（2D）不可分离内插滤波器，它通过最小化预测误差能量对每帧进行独立计算。对于要插值的每个小数像素位置，确定一组二维滤波器系数。由于发送滤波器系数作为侧信息导致附加比特率，对于不同的图像分辨率和总比特率几乎是一定的，所以当总比特率下降时，编码增益的损失增加。因此，我们开发了一种将不可分离的二维滤波器作为多相滤波器的算法。对于每个帧，通过评估多相滤波器来预测内插滤波器脉冲响应，我们只需要对滤波器系数的预测误差进行编码。

4. 发明授权

US09043216B2 Audio signal decoder, time warp contour data provider, method and computer program 有权
标题翻译：音频信号解码器，时间扭曲轮廓数据提供者，方法和计算机程序
公开(公告)号：US09043216B2
公开(公告)日：2015-05-26
申请号：US12935718
申请日：2009-07-01
申请人： Stefan Bayer , Sascha Disch , Ralf Geiger , Guillaume Fuchs , Max Neuendorf , Gerald Schuller , Bernd Edler
发明人： Stefan Bayer , Sascha Disch , Ralf Geiger , Guillaume Fuchs , Max Neuendorf , Gerald Schuller , Bernd Edler
IPC分类号： G10L21/04 , G10L19/00 , G10L19/02 , G10L19/022
CPC分类号： G10L21/04 , G10L19/0212 , G10L19/022 , G10L19/032 , G10L19/167
摘要： An audio signal decoder has a time warp contour calculator, a time warp contour data rescaler and a warp decoder. The time warp contour calculator is configured to generate time warp contour data repeatedly restarting from a predetermined time warp contour start value, based on time warp contour evolution information describing a temporal evolution of the time warp contour. The time warp contour data rescaler is configured to rescale at least a portion of the time warp contour data such that a discontinuity at a restart is avoided, reduced or eliminated in a rescaled version of the time warp contour. The warp decoder is configured to provide the decoded audio signal representation, based on an encoded audio signal representation and using the rescaled version of the time warp contour.
摘要翻译：音频信号解码器具有时间扭曲轮廓计算器，时间扭曲轮廓数据重定标器和扭曲解码器。时间扭曲轮廓计算器被配置为基于描述时间扭曲轮廓的时间演变的时间扭曲轮廓演化信息，从预定时间扭曲轮廓开始值产生重复重新起动的时间扭曲轮廓数据。时间扭曲轮廓数据重定标器被配置为重新缩放时间扭曲轮廓数据的至少一部分，使得在时间扭曲轮廓的重新缩放版本中避免，减少或消除重启时的不连续性。扭曲解码器被配置为基于编码的音频信号表示并使用时间扭曲轮廓的重新缩放版本来提供解码的音频信号表示。

5. 发明授权

US08411731B2 Device and method for processing a real subband signal for reducing aliasing effects 有权
标题翻译：用于处理实际子带信号以减少混叠效应的装置和方法
公开(公告)号：US08411731B2
公开(公告)日：2013-04-02
申请号：US12375728
申请日：2007-06-26
申请人： Bernd Edler , Harald Popp
发明人： Bernd Edler , Harald Popp
IPC分类号： H03K5/159
CPC分类号： H03G3/00 , G10L19/0204 , H03H17/0266 , H03H2218/04
摘要： In order to process a subband signal of a plurality of real subband signals which are a representation of a real discrete-time signal generated by an analysis filter bank, a weighter for weighting a subband signal by a weighting factor determined for the subband signal is provided to obtain a weighted subband signal. In addition, a correction term is calculated by a correction term determiner, the correction term determiner being implemented to calculate the correction term using at least one other subband signal and using another weighting factor provided for the other subband signal, the two weighting factors differing. The correction term is then combined with the weighted subband signal to obtain a corrected subband signal, resulting in reduced aliasing, even if subband signals are weighted to a different extent.
摘要翻译：为了处理作为由分析滤波器组产生的实际离散时间信号的表示的多个实际子带信号的子带信号，提供了对子带信号进行加权以对于子带信号确定的加权因子的加权器以获得加权子带信号。此外，校正项由校正项确定器计算，校正项确定器被实现为使用至少一个其他子带信号来计算校正项，并且使用为另一子带信号提供的另一加权因子，两个加权因子不同。然后将校正项与加权子带信号组合以获得校正的子带信号，导致减少的混叠，即使子带信号被加权到不同程度。

6. 发明申请

US20100150232A1 METHOD FOR CONCEALING A PACKET LOSS 审中-公开
标题翻译：防止包装损失的方法
公开(公告)号：US20100150232A1
公开(公告)日：2010-06-17
申请号：US12446744
申请日：2007-10-31
申请人： Dieu Thanh Nguyen , Bernd Edler , Jorn Ostermann , Nikolce Stefanoski
发明人： Dieu Thanh Nguyen , Bernd Edler , Jorn Ostermann , Nikolce Stefanoski
IPC分类号： H04N7/12
CPC分类号： H04N21/2389 , H04N19/102 , H04N19/139 , H04N19/142 , H04N19/177 , H04N19/188 , H04N19/30 , H04N19/34 , H04N19/44 , H04N19/46 , H04N19/61 , H04N19/895 , H04N21/4385
摘要： A method of concealing a packet loss during video decoding is provided. An input stream having a plurality of network abstraction layer units NAL is received. A loss of a network abstraction layer unit in a group of pictures in the input stream is detected. A valid network abstraction layer unit order from the available network abstraction layer units is outputted. The network abstraction layer unit order is received by a video coding layer (VCL) and data is outputted.
摘要翻译：提供了一种在视频解码期间隐藏分组丢失的方法。接收具有多个网络抽象层单元NAL的输入流。检测输入流中的一组图像中的网络抽象层单元的丢失。输出来自可用网络抽象层单元的有效网络抽象层单元顺序。网络抽象层单元顺序由视频编码层（VCL）接收，数据被输出。

7. 发明申请

US20090319283A1 Apparatus and Method for Generating Audio Subband Values and Apparatus and Method for Generating Time-Domain Audio Samples 有权
标题翻译：用于产生音频子带值的装置和方法以及用于生成时域音频样本的装置和方法
公开(公告)号：US20090319283A1
公开(公告)日：2009-12-24
申请号：US12447126
申请日：2007-10-23
申请人： Markus Schnell , Manfred Lutzky , Markus Lohwasser , Markus Schmidt , Marc Gayer , Michael Mellar , Bernd Edler , Markus Multrus , Gerald Schuller , Ralf Geiger , Bernhard Grill
发明人： Markus Schnell , Manfred Lutzky , Markus Lohwasser , Markus Schmidt , Marc Gayer , Michael Mellar , Bernd Edler , Markus Multrus , Gerald Schuller , Ralf Geiger , Bernhard Grill
IPC分类号： G10L21/00
CPC分类号： G10L19/0204 , G10L19/022 , G10L21/038 , G10L25/45 , H03H17/0266
摘要： An embodiment of an apparatus for generating audio subband values in audio subband channels has an analysis windower for windowing a frame of time-domain audio input samples being in a time sequence extending from an early sample to a later sample using an analysis window function having a sequence of window coefficients to obtain windowed samples. The analysis window function has a first group of window coefficients and a second group of window coefficients. The first group of window coefficients is used for windowing later time-domain samples and the second group of window coefficients is used for windowing an earlier time-domain samples. The apparatus further has a calculator for calculating the audio subband values using the windowed samples.
摘要翻译：用于在音频子频带信道中产生音频子带值的装置的实施例具有分析加窗器，用于使用具有分析窗口功能的时间序列，将时域序列扩展到时间序列，从而将其从早期样本延伸到后面的样本窗系数序列以获得窗口样本。分析窗函数具有第一组窗系数和第二组窗系数。窗口系数的第一组用于对稍后的时域样本进行加窗，第二组窗系数用于对较早的时域采样进行加窗。该装置还具有用于使用窗口样本来计算音频子带值的计算器。

8. 发明授权

US06370507B1 Frequency-domain scalable coding without upsampling filters 有权
标题翻译：无上采样滤波器的频域可伸缩编码
公开(公告)号：US06370507B1
公开(公告)日：2002-04-09
申请号：US09319066
申请日：1999-05-28
申请人： Bernhard Grill , Bernd Edler , Karlheinz Brandenburg
发明人： Bernhard Grill , Bernd Edler , Karlheinz Brandenburg
IPC分类号： G10L1902
CPC分类号： G10L19/24 , G10L19/0204
摘要： In a method of coding discrete time signals (X1) sampled with a first sampling rate, second time signals (x2) are generated using the first time signals having a bandwidth corresponding to a second sampling rate, with the second sampling rate being lower than the first sampling rate. The second time signals are coded in accordance with a first coding algorithm. The coded second signals (X2c) are decoded again in order to obtain coded/decoded second time signals (X2cd) having a bandwidth corresponding to the second sampling frequency. The first time signals, by frequency domain transformation, become first spectral values (X1). Second spectral values (X2cd) are generated from the coded/decoded second time signals, the second spectral values being a representation of the coded/decoded time signals in the frequency domain. To obtain weighted spectral values, the first spectral values are weighted by means of the second spectral values, with the first and second spectral values having the same frequency and time resolution. The weighted spectral values (Xb) are coded in accordance with a second coding algorithm in consideration of a psychoacoustic model and written into a bit stream. Weighting the first spectral values and the second spectral values comprises the subtraction of the second spectral values from the first spectral values in to obtain differential spectral values.
摘要翻译：在对以第一采样率采样的离散时间信号（X1）进行编码的方法中，使用具有对应于第二采样率的带宽的第一时间信号来生成第二时间信号（x2），其中第二采样率低于第一次采样率。第二时间信号根据第一编码算法进行编码。再次对编码的第二信号（X2c）进行解码，以获得具有对应于第二采样频率的带宽的编码/解码的第二时间信号（X2cd）。第一次通过频域变换的信号变为第一个光谱值（X1）。从编码/解码的第二时间信号产生第二频谱值（X2cd），第二频谱值是频域中编码/解码的时间信号的表示。为了获得加权光谱值，通过第二光谱值对第一光谱值进行加权，其中第一和第二光谱值具有相同的频率和时间分辨率。考虑到心理声学模型并将其写入比特流，加权频谱值（Xb）根据第二编码算法进行编码。加权第一光谱值和第二光谱值包括从第一光谱值减去第二光谱值以获得差分光谱值。

9. 发明授权

US09015041B2 Time warp activation signal provider, audio signal encoder, method for providing a time warp activation signal, method for encoding an audio signal and computer programs 有权
标题翻译：时间扭曲激活信号提供器，音频信号编码器，用于提供时间扭曲激活信号的方法，用于编码音频信号的方法和计算机程序
公开(公告)号：US09015041B2
公开(公告)日：2015-04-21
申请号：US13004525
申请日：2011-01-11
申请人： Stefan Bayer , Sascha Disch , Ralf Geiger , Guillaume Fuchs , Max Neuendorf , Gerald Schuller , Bernd Edler
发明人： Stefan Bayer , Sascha Disch , Ralf Geiger , Guillaume Fuchs , Max Neuendorf , Gerald Schuller , Bernd Edler
IPC分类号： G10L19/12 , G10L21/00 , G10L19/00 , G10L21/04 , G10L19/002 , G10L19/028
CPC分类号： G10L21/04 , G10L19/002 , G10L19/0212 , G10L19/022 , G10L19/028 , G10L19/032 , G10L19/265 , G10L21/043 , G10L25/90
摘要： An audio encoder has a window function controller, a windower, a time warper with a final quality check functionality, a time/frequency converter, a TNS stage or a quantizer encoder, the window function controller, the time warper, the TNS stage or an additional noise filling analyzer are controlled by signal analysis results obtained by a time warp analyzer or a signal classifier. Furthermore, a decoder applies a noise filling operation using a manipulated noise filling estimate depending on a harmonic or speech characteristic of the audio signal.
摘要翻译：音频编码器具有窗口功能控制器，风扇，具有最终质量检查功能的时间整形器，时间/频率转换器，TNS阶段或量化器编码器，窗口功能控制器，时间整经器，TNS阶段或附加噪声填充分析仪由时间扭曲分析仪或信号分类器得到的信号分析结果进行控制。此外，解码器使用根据音频信号的谐波或语音特性的操纵噪声填充估计来应用噪声填充操作。

10. 发明授权

US08700388B2 Audio transform coding using pitch correction 有权
标题翻译：使用音调校正的音频变换编码
公开(公告)号：US08700388B2
公开(公告)日：2014-04-15
申请号：US12668912
申请日：2009-03-23
申请人： Bernd Edler , Sascha Disch , Ralf Geiger , Stefan Bayer , Ulrich Kraemer , Guillaume Fuchs , Max Neuendorf , Markus Multrus , Gerald Schuller , Harald Popp
发明人： Bernd Edler , Sascha Disch , Ralf Geiger , Stefan Bayer , Ulrich Kraemer , Guillaume Fuchs , Max Neuendorf , Markus Multrus , Gerald Schuller , Harald Popp
IPC分类号： G10L11/04 , G10L21/00 , G10L15/00 , G10L19/14 , G10L19/02 , G10L13/06
CPC分类号： G10L19/0212 , G10L19/022
摘要： A processed representation of an audio signal having a sequence of frames is generated by sampling the audio signal within first and second frames of the sequence of frames, the second frame following the first frame, the sampling using information on a pitch contour of the first and second frames to derive a first sampled representation. The audio signal is sampled within the second and third frames, the third frame following the second frame in the sequence of frames. The sampling uses the information on the pitch contour of the second frame and information on a pitch contour of the third frame to derive a second sampled representation. A first scaling window is derived for the first sampled representation, and a second scaling window is derived for the second sampled representation, the scaling windows depending on the samplings applied to derive the first sampled representations or the second sampled representation.
摘要翻译：具有帧序列的音频信号的处理表示是通过对帧序列的第一帧和第二帧中的音频信号进行采样，第一帧之后的第二帧，使用关于第一帧的音调轮廓的信息的采样，以及第二帧以获得第一采样表示。音频信号在第二帧和第三帧中采样，第三帧在帧序列中的第二帧之后。采样使用关于第二帧的音调轮廓的信息和关于第三帧的音调轮廓的信息来导出第二采样表示。对于第一采样表示导出第一缩放窗口，并且针对第二采样表示导出第二缩放窗口，缩放窗口取决于应用于导出第一采样表示或第二采样表示的采样。

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