专利快速检索-快速检索全球专利，免费商用专利数据库-IPRDB

11. 发明申请

US20110173009A1 Apparatus and Method for Encoding/Decoding an Audio Signal Using an Aliasing Switch Scheme 有权
标题翻译：使用混叠开关方案对音频信号进行编码/解码的装置和方法
公开(公告)号：US20110173009A1
公开(公告)日：2011-07-14
申请号：US13004351
申请日：2011-01-11
申请人： Guillaume Fuchs , Jeremie Lecomte , Stefan Bayer , Ralf Geiger , Markus Multrus , Gerald Schuller , Jens Hirschfeld
发明人： Guillaume Fuchs , Jeremie Lecomte , Stefan Bayer , Ralf Geiger , Markus Multrus , Gerald Schuller , Jens Hirschfeld
IPC分类号： G10L19/00
CPC分类号： G10L19/20 , G10L19/02 , G10L19/022 , G10L19/04
摘要： An apparatus for encoding an audio signal includes the windower for windowing a first block of the audio signal using an analysis window having an aliasing portion and a further portion. The apparatus furthermore includes a processor for processing the first sub-block of the audio signal associated with the aliasing portion by transforming the sub-block from a domain into a different domain subsequent to windowing the first sub-block to obtain the processed first sub-block, and for processing a second sub-block of the audio signal associated with the further portion by transforming the second sub-block from the domain into the different domain before windowing the second sub-block to obtain a processed second sub-block. The apparatus furthermore includes a transformer for converting the processed first sub-block and the processed second sub-block from the different domain into a further different domain using the same block transform rule to obtain a converted first block which may then be compressed using any of the well-known data compression algorithms. Thus, a critically sampled switch between two coding modes can be obtained, since aliasing portions occurring in two different domains are matched to each other.
摘要翻译：用于编码音频信号的装置包括用于使用具有混叠部分和另一部分的分析窗口对音频信号的第一块进行加窗的加窗器。该装置还包括处理器，用于通过在将第一子块加窗以获得处理的第一子块之后，将子块从域变换为不同的域来处理与混叠部分相关联的音频信号的第一子块，并且用于通过在开启第二子块之前将第二子块从该域变换为不同的域来处理与另一部分相关联的音频信号的第二子块，以获得经处理的第二子块。该装置还包括一个变换器，用于使用相同的块变换规则将处理的第一子块和处理的第二子块从不同的域转换成另一个不同的域，以获得转换的第一块，然后可以使用众所周知的数据压缩算法。因此，由于发生在两个不同域中的混叠部分彼此匹配，因此可以获得两种编码模式之间的批量采样切换。

12. 发明授权

US08862480B2 Audio encoding/decoding with aliasing switch for domain transforming of adjacent sub-blocks before and subsequent to windowing 有权
标题翻译：使用混叠开关进行音频编码/解码，用于在开窗之前和之后对相邻子块进行域变换
公开(公告)号：US08862480B2
公开(公告)日：2014-10-14
申请号：US13004351
申请日：2011-01-11
申请人： Guillaume Fuchs , Jeremie Lecomte , Stefan Bayer , Ralf Geiger , Markus Multrus , Gerald Schuller , Jens Hirschfeld
发明人： Guillaume Fuchs , Jeremie Lecomte , Stefan Bayer , Ralf Geiger , Markus Multrus , Gerald Schuller , Jens Hirschfeld
IPC分类号： G10L19/02 , G10L19/04 , G10L19/20 , G10L19/022
CPC分类号： G10L19/20 , G10L19/02 , G10L19/022 , G10L19/04
摘要： An apparatus for encoding an audio signal includes the windower for windowing a first block of the audio signal using an analysis window having an aliasing portion and a further portion. The apparatus furthermore includes a processor for processing the first sub-block of the audio signal associated with the aliasing portion by transforming the sub-block from a domain into a different domain subsequent to windowing the first sub-block to obtain the processed first sub-block, and for processing a second sub-block of the audio signal associated with the further portion by transforming the second sub-block from the domain into the different domain before windowing the second sub-block to obtain a processed second sub-block. Thus, a critically sampled switch between two coding modes can be obtained.
摘要翻译：用于编码音频信号的装置包括用于使用具有混叠部分和另一部分的分析窗口对音频信号的第一块进行加窗的加窗器。该装置还包括处理器，用于通过在将第一子块加窗以获得处理的第一子块之后，将子块从域变换为不同的域来处理与混叠部分相关联的音频信号的第一子块，并且用于通过在开启第二子块之前将第二子块从该域变换为不同的域来处理与另一部分相关联的音频信号的第二子块，以获得经处理的第二子块。因此，可以获得两种编码模式之间的批量采样开关。

13. 发明申请

US20110173011A1 Audio Encoder and Decoder for Encoding and Decoding Frames of a Sampled Audio Signal 有权
标题翻译：音频编码器和解码器，用于对采样音频信号的帧进行编码和解码
公开(公告)号：US20110173011A1
公开(公告)日：2011-07-14
申请号：US13004475
申请日：2011-01-11
申请人： Ralf Geiger , Bernhard Grill , Bruno Bessette , Philippe Gournay , Guillaume Fuchs , Markus Multrus , Max Neuendorf , Gerald Schuller
发明人： Ralf Geiger , Bernhard Grill , Bruno Bessette , Philippe Gournay , Guillaume Fuchs , Markus Multrus , Max Neuendorf , Gerald Schuller
IPC分类号： G10L19/00
CPC分类号： G10L19/0212 , G10L19/04
摘要： An audio encoder adapted for encoding frames of a sampled audio signal to obtain encoded frames, wherein a frame includes a number of time domain audio samples. The audio encoder includes a predictive coding analysis stage for determining information on coefficients of a synthesis filter and a prediction domain frame based on a frame of audio samples. The audio encoder further includes a time-aliasing introducing transformer for transforming overlapping prediction domain frames to the frequency domain to obtain prediction domain frame spectra, wherein the time-aliasing introducing transformer is adapted for transforming the overlapping prediction domain frames in a critically-sampled way. Moreover, the audio encoder includes a redundancy reducing encoder for encoding the prediction domain frame spectra to obtain the encoded frames based on the coefficients and the encoded prediction domain frame spectra.
摘要翻译：一种音频编码器，适于编码采样的音频信号的帧以获得编码的帧，其中帧包括多个时域音频采样。音频编码器包括用于基于音频样本的帧来确定关于合成滤波器和预测域帧的系数的信息的预测编码分析阶段。音频编码器还包括时间混叠引入变压器，用于将重叠的预测域帧转换到频域以获得预测域帧频谱，其中时间混叠引入变压器适于以重要取样的方式变换重叠的预测域帧。此外，音频编码器包括冗余减少编码器，用于根据系数和编码的预测域帧频谱对预测域帧频谱进行编码以获得编码的帧。

14. 发明授权

US08595019B2 Audio coder/decoder with predictive coding of synthesis filter and critically-sampled time aliasing of prediction domain frames 有权
标题翻译：具有合成滤波器的预测编码和预测域帧的临时采样时间混叠的音频编码器/解码器
公开(公告)号：US08595019B2
公开(公告)日：2013-11-26
申请号：US13004475
申请日：2011-01-11
申请人： Ralf Geiger , Bernhard Grill , Bruno Bessette , Philippe Gournay , Guillaume Fuchs , Markus Multrus , Max Neuendorf , Gerald Schuller
发明人： Ralf Geiger , Bernhard Grill , Bruno Bessette , Philippe Gournay , Guillaume Fuchs , Markus Multrus , Max Neuendorf , Gerald Schuller
IPC分类号： G10L19/00 , G10L19/02 , G10L19/04
CPC分类号： G10L19/0212 , G10L19/04
摘要： An audio encoder adapted for encoding frames of a sampled audio signal to obtain encoded frames, wherein a frame includes a number of time domain audio samples. The audio encoder includes a predictive coding analysis stage for determining information on coefficients of a synthesis filter and a prediction domain frame based on a frame of audio samples. The audio encoder further includes a time-aliasing introducing transformer for transforming overlapping prediction domain frames to the frequency domain to obtain prediction domain frame spectra, wherein the time-aliasing introducing transformer is adapted for transforming the overlapping prediction domain frames in a critically-sampled way. Moreover, the audio encoder includes a redundancy reducing encoder for encoding the prediction domain frame spectra to obtain the encoded frames based on the coefficients and the encoded prediction domain frame spectra.
摘要翻译：一种音频编码器，适于编码采样的音频信号的帧以获得编码的帧，其中帧包括多个时域音频采样。音频编码器包括用于基于音频样本的帧来确定关于合成滤波器和预测域帧的系数的信息的预测编码分析阶段。音频编码器还包括时间混叠引入变压器，用于将重叠的预测域帧转换到频域以获得预测域帧频谱，其中时间混叠引入变压器适于以重要取样的方式变换重叠的预测域帧。此外，音频编码器包括冗余减少编码器，用于根据系数和编码的预测域帧频谱对预测域帧频谱进行编码以获得编码的帧。

15. 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。

16. 发明申请

US20100198586A1 AUDIO TRANSFORM CODING USING PITCH CORRECTION 有权
标题翻译：使用倾斜校正进行音频变换编码
公开(公告)号：US20100198586A1
公开(公告)日：2010-08-05
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： G10L19/02
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。

17. 发明申请

US20070043557A1 METHOD AND DEVICE FOR QUANTIZING AN INFORMATION SIGNAL 有权
标题翻译：用于量化信息信号的方法和装置
公开(公告)号：US20070043557A1
公开(公告)日：2007-02-22
申请号：US11460433
申请日：2006-07-27
申请人： Gerald Schuller , Stefan Wabnik , Jens Hirschfeld , Wolfgang Fiesel
发明人： Gerald Schuller , Stefan Wabnik , Jens Hirschfeld , Wolfgang Fiesel
IPC分类号： G10L19/00
CPC分类号： G10L19/032 , G10L19/26 , G10L19/265
摘要： Quantizing an information signal of a sequence of information values includes frequency-selective filtering the sequence of information values to obtain a sequence of filtered information values and quantizing the filtered information values to obtain a sequence of quantized information values by means of a quantizing step function which maps the filtered information values to the quantized information values and the course of which is steeper below a threshold information value than above the threshold information value.
摘要翻译：量化信息值序列的信息信号包括对信息值序列进行频率选择性滤波以获得滤波信息值序列，并对滤波信息值进行量化，以通过量化步长函数获得量化信息值序列，该量化步长函数将经滤波的信息值映射到量化信息值，并且其过程比阈值信息值之上的阈值信息值更陡。

18. 发明授权

US09754601B2 Information signal encoding using a forward-adaptive prediction and a backwards-adaptive quantization 有权
公开(公告)号：US09754601B2
公开(公告)日：2017-09-05
申请号：US12300602
申请日：2007-02-28
申请人： Jens Hirschfeld , Gerald Schuller , Manfred Lutzky , Ulrich Kraemer , Stefan Wabnik
发明人： Jens Hirschfeld , Gerald Schuller , Manfred Lutzky , Ulrich Kraemer , Stefan Wabnik
IPC分类号： G10L19/06 , G10L19/032 , G10L19/00 , G10L19/035
CPC分类号： G10L19/06 , G10L19/0017 , G10L19/032 , G10L19/035
摘要： A very coarse quantization exceeding the measure determined by the masking threshold without or only very little quality losses is enabled by quantizing not immediately the prefiltered signal, but a prediction error obtained by forward-adaptive prediction of the prefiltered signal. Due to the forward adaptivity, the quantizing error has no negative effect on the prediction on the decoder side.

19. 发明授权

US07716042B2 Audio coding 有权
标题翻译：音频编码
公开(公告)号：US07716042B2
公开(公告)日：2010-05-11
申请号：US11460423
申请日：2006-07-27
申请人： Gerald Schuller , Stefan Wabnik , Jens Hirschfeld , Manfred Lutzky
发明人： Gerald Schuller , Stefan Wabnik , Jens Hirschfeld , Manfred Lutzky
IPC分类号： G10L19/00
CPC分类号： G10L19/02 , G10L19/0017 , G10L19/26 , G10L19/265
摘要： Coding an audio signal of a sequence of audio values into a coded signal includes determining first and second listening thresholds for first and second blocks of audio values of the sequence of audio values; calculating versions of first second parameterizations of the parameterizable filter such that the transfer function thereof roughly corresponds to the inverse of the magnitude of the first and second listening thresholds, respectively; filtering a predetermined block of audio values of the sequence of audio values with the parameterizable filter using a predetermined parameterization which depends on the version of the second parameterization to obtain a block of filtered audio values corresponding to the predetermined block which is quantized; forming a difference between the versions of the first and second parameterizations; integrating information on, inter alias, the difference into the coded signal.
摘要翻译：将音频值序列的音频信号编码为编码信号包括确定音频值序列的音频值的第一和第二块的第一和第二监听阈值; 计算可参数化滤波器的第一第二参数化的版本，使得其传递函数分别大致对应于第一和第二监听阈值的幅度的倒数; 使用预定参数化来过滤具有可参数化滤波器的音频值序列的预定音频值的块，该预定参数取决于第二参数化的版本，以获得对应于被量化的预定块的滤波音频值块; 形成第一和第二参数化的版本之间的差异; 将信息，别名，差异集成到编码信号中。

20. 发明申请

US20090254783A1 Information Signal Encoding 有权
标题翻译：信息信号编码
公开(公告)号：US20090254783A1
公开(公告)日：2009-10-08
申请号：US12300602
申请日：2007-02-28
申请人： Jens Hirschfeld , Gerald Schuller , Manfred Lutzky , Ulrich Kraemer , Stefan Wabnik
发明人： Jens Hirschfeld , Gerald Schuller , Manfred Lutzky , Ulrich Kraemer , Stefan Wabnik
IPC分类号： G06F11/00 , H04B1/66
CPC分类号： G10L19/06 , G10L19/0017 , G10L19/032 , G10L19/035
摘要： A very coarse quantization exceeding the measure determined by the masking threshold without or only very little quality losses is enabled by quantizing not immediately the prefiltered signal, but a prediction error obtained by forward-adaptive prediction of the prefiltered signal. Due to the forward adaptivity, the quantizing error has no negative effect on the prediction on the decoder side.
摘要翻译：通过量化不是立即预过滤信号，而是通过对预过滤信号进行前向自适应预测而获得的预测误差，能够实现超过由掩蔽阈值确定的度量而不仅仅具有非常少的质量损失的非常粗略的量化。由于正向适应性，量化误差对解码器侧的预测没有负面影响。

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