公开(公告)号：US5848391A

公开(公告)日：1998-12-08

申请号：US678666

申请日：1996-07-11

申请人： Marina Bosi ,  Grant Davidson ,  Charles Robinson ,  Martin Dietz ,  Uwe Gbur ,  Oliver Kunz ,  Karlheinz Brandenburg

发明人： Marina Bosi ,  Grant Davidson ,  Charles Robinson ,  Martin Dietz ,  Uwe Gbur ,  Oliver Kunz ,  Karlheinz Brandenburg

IPC分类号： G10L19/02 ,  H04B1/66 ,  G10L5/00

CPC分类号： G10L19/022 ,  H04B1/665

摘要： A method of encoding time-discrete audio signals comprises the steps of weighting the time-discrete audio signal by means of window functions overlapping each other so as to form blocks, the window functions producing blocks of a first length for signals varying weakly with time and blocks of a second length for signals varying strongly with time. A start window sequence is selected for the transition from windowing with blocks of the first length to windowing with blocks of the second length, whereas a stop window sequence is selected for the opposite transition. The start window sequence is selected from at least two different start window sequences having different lengths, whereas the stop window sequence is selected from at least two different stop window sequences having different lengths. A method of decoding blocks of encoded audio signals selects a suitable inverse transformation as well as a suitable synthesis window as a reaction to side information associated with each block.

摘要翻译： 对时间离散音频信号进行编码的方法包括以下步骤：通过彼此重叠的窗口函数对时间离散音频信号进行加权，以形成块，窗口函数产生用于随时间变化的信号的第一长度的块， 对于随时间强烈变化的信号的第二长度的块。 选择起始窗口序列，用于从具有第一长度的块的窗口到具有第二长度的块的窗口的转换，而针对相反的转换选择停止窗口序列。 起始窗口序列从具有不同长度的至少两个不同的开始窗口序列中选择，而停止窗口序列从具有不同长度的至少两个不同的停止窗口序列中选择。 对编码音频信号的块进行解码的方法选择合适的逆变换以及合适的合成窗口作为与每个块相关联的侧信息的反应。

公开(公告)号：US07260225B2

公开(公告)日：2007-08-21

申请号：US10149248

申请日：2000-12-07

申请人： Bodo Teichmann ,  Oliver Kunz ,  Juergen Herre ,  Klaus Peichl ,  Michael Beer

发明人： Bodo Teichmann ,  Oliver Kunz ,  Juergen Herre ,  Klaus Peichl ,  Michael Beer

IPC分类号： H04H5/00

CPC分类号： H04S1/00 ,  G10L19/008

摘要： In a device for processing a stereo audio signal having a first channel and a second channel the stereo signal is at first analyzed to obtain a measure for a quantity of bits required by a coder to code the stereo audio signal using a coding algorithm. The first channel and the second channel are then modified when the measure for the quantity of bits is larger than a predetermined value, the modification being performed in such a way that the energy of a sum signal of the first and the second modified channel is in a predetermined relation to the energy of a sum signal of the first and the second channel and that a difference signal of the first and the second modified channel is attenuated in contrast to the difference signal of the first and the second channel. Especially for audio coders requiring a constant output bit rate the side channel is attenuated in the case of stereo audio signals, the coding of which cannot meet the output bit rate of the coder, by which a stereo channel separation is abandoned for the benefit of an increased audio bandwidth or a reduction of quantizing disturbances, respectively.

摘要翻译： 在用于处理具有第一通道和第二通道的立体声音频信号的设备中，首先分析立体声信号以获得编码器使用编码算法编码立体声音频信号所需的位数量度。 然后当比特量的测量大于预定值时修改第一信道和第二信道，修改是以第一和第二修改信道的和信号的能量处于 与第一和第二信道的和信号的能量的预定关系，并且与第一和第二信道的差分信号相比，第一和第二修改信道的差分信号被衰减。 特别是对于需要恒定输出比特率的音频编码器，在立体声音频信号的情况下，侧信道被衰减，其编码不能满足编码器的输出比特率，通过该编码器，放弃了立体声信道分离以有利于 分别增加音频带宽或减少量化干扰。

公开(公告)号：US06766293B1

公开(公告)日：2004-07-20

申请号：US09367775

申请日：1999-08-18

申请人： Jürgen Herre ,  Uwe Gbur ,  Andreas Ehret ,  Martin Dietz ,  Bodo Teichmann ,  Oliver Kunz ,  Karlheinz Brandenburg ,  Heinz Gerhäuser

发明人： Jürgen Herre ,  Uwe Gbur ,  Andreas Ehret ,  Martin Dietz ,  Bodo Teichmann ,  Oliver Kunz ,  Karlheinz Brandenburg ,  Heinz Gerhäuser

IPC分类号： G10L2102

CPC分类号： G10L19/028 ,  H04B1/665

摘要： In a method for signalling a noise substitution when coding an audio signal, the time-domain audio signal is first transformed into the frequency domain to obtain spectral values. The spectral values are subsequently grouped together to form groups of spectral values. On the basis of a detection establishing whether a group of spectral values is a noisy group or not, a codebook is allocated to a non-noisy or tonal group by means of a codebook number for redundancy coding of the same. If a group is noisy, an additional codebook number which does not refer to a codebook is allocated to it in order to signal that this group is noisy and therefore does not have to be redundancy coded. By signalling noise substitution by means of a Huffman codebook number for noisy groups of spectral values, which are e.g. sections made up of scale factor bands which do not have to be redundancy coded, an opportunity is provided to indicate the presence of a noise substitution in a scale factor band in the bit stream syntax of the MPEG-2 Advanced Audio Coding (AAC) Standard without having to interfere with the basic coding structure and without having to meddle with the structure of the existing bit stream syntax.

摘要翻译： 在对音频信号编码时用于发信号通知的方法中，首先将时域音频信号变换成频域以获得频谱值​​。 光谱值随后被分组在一起以形成光谱值组。 基于确定一组频谱值是否为噪声组的检测，通过用于冗余编码的码本号将码本分配给非噪声或色调组。 如果组噪声，则分配不附加码本的附加码本号，以便发信号通知该组噪声，因此不必进行冗余编码。 通过用于噪声组的频谱值的霍夫曼码本号对信号进行信号替换， 由不必冗余编码的比例因子带组成的部分提供了一种机会，以指示在MPEG-2高级音频编码（AAC）标准的比特流语法中的比例因子频带中存在噪声替换 而不必干扰基本编码结构，而不必介入现有比特流语法的结构。

公开(公告)号：USRE44897E1

公开(公告)日：2014-05-13

申请号：US13897221

申请日：2013-05-17

申请人： Oliver Kunz ,  Martin Dietz ,  Rainer Buchta ,  Jurgen Zeller ,  Karlheinz Brandenburg ,  Martin Sieler ,  Heinz Gerhauser

发明人： Oliver Kunz ,  Martin Dietz ,  Rainer Buchta ,  Jurgen Zeller ,  Karlheinz Brandenburg ,  Martin Sieler ,  Heinz Gerhauser

IPC分类号： H04B1/66

摘要： In a method for coding an audio signal digitized at a low sampling rate to obtain time domain audio samples. A frequency domain representation of the time domain audio samples is produced. The frequency domain representation includes successive frequency lines. These frequency lines are grouped into a plurality of scale factor bands. The successive frequency lines in a scale factor band are coded with the same scale factor. A plurality of regions is formed by grouping the scale factor bands, wherein successive scale factor bands form a region within which all the scale factors are coded with the same number of bits, which is determined according to the largest scale factor of the region. The scale factors assigned to scale factor bands within the highest region that includes the higher frequency successive frequency lines are set to zero. The frequency lines in the highest region are coded using the zero-valued scale factors that correspond to a multiplication factor of 1. The scale factors for the highest region, however, are not coded. Thus, the bits that would be required for coding these zero-valued scale factors are saved and can be used for a finer quantization of the rest of the spectrum. Additionally, this coding method when applied to ISO/IEC 13818-3 as a low sampling rate modification thereof only requires minimal changes with respect to this Standard.

摘要翻译： 在以低采样率对数字化的音频信号进行编码以获得时域音频样本的方法中， 产生时域音频样本的频域表示。 频域表示包括连续的频率线。 这些频率线被分组成多个比例因子频带。 缩放因子带中的连续频率线以相同的比例因子进行编码。 通过对比例因子频带进行分组来形成多个区域，其中，连续的比例因子波段形成一个区域，在该区域内，所有比例因子都以相同的比特数进行编码，这是根据该区域的最大比例因子确定的。 分配给包括较高频率的连续频率线的最高区域内的比例因子频带的比例因子被设置为零。 最高区域中的频率线使用与乘法因子1对应的零值比例因子进行编码。然而，最高区域的比例因子未被编码。 因此，编码这些零值比例因子所需的位被保存，并且可以用于其余频谱的更精细的量化。 此外，当将其应用于ISO / IEC 13818-3作为其低采样率修改时，该编码方法仅需要相对于本标准的最小变化。

公开(公告)号：US06441755B1

公开(公告)日：2002-08-27

申请号：US09786614

申请日：2001-05-07

申请人： Martin Dietz ,  Ali Nowbakht-Irani ,  Ralph Sperschneider ,  Oliver Kunz

发明人： Martin Dietz ,  Ali Nowbakht-Irani ,  Ralph Sperschneider ,  Oliver Kunz

IPC分类号： H03M700

CPC分类号： H03M7/40

摘要： A method and a device for entropy encoding and associated decoding make use of a code consisting on the one hand of a code table with reversible code words (12) and comprising on the other hand an escape region for information words to be coded which are located outside the region (14) defined by said code table. Said region can be selected in such a way that a major part of the information words is coded with symmetrical code words by the code table. On the one hand, it is thus possible to carry out, in addition to forward decoding, also backward decoding (24) and on the other hand, use of reversible code words allows for rapid recognition of errors in a code word stream transmitted over a non-ideal channel.

摘要翻译： 用于熵编码和相关解码的方法和装置利用一方面由具有可逆码字（12）的码表组成的码，另一方面，包括位于所要被编码的信息字的转义区域 在由所述代码表定义的区域（14）之外。 所述区域可以以这样一种方式进行选择，使得主要部分的信息字通过码表用对称码字编码。 一方面，除了正向解码之外，还可以进行反向解码（24），另一方面，可逆的代码字的使用允许快速识别通过 非理想渠道

公开(公告)号：US06424939B1

公开(公告)日：2002-07-23

申请号：US09402684

申请日：1999-10-06

申请人： Jürgen Herre ,  Uwe Gbur ,  Andreas Ehret ,  Martin Dietz ,  Bodo Teichmann ,  Oliver Kunz ,  Karlheinz Brandenburg ,  Heinz Gerhauser

发明人： Jürgen Herre ,  Uwe Gbur ,  Andreas Ehret ,  Martin Dietz ,  Bodo Teichmann ,  Oliver Kunz ,  Karlheinz Brandenburg ,  Heinz Gerhauser

IPC分类号： G10L1900

CPC分类号： H04B1/665 ,  G10L19/028

摘要： A method for coding or decoding an audio signal combines the advantages of TNS processing and noise substitution. A time-discrete audio signal is initially transformed to the frequency domain in order to obtain spectral values of the temporal audio signal. Subsequently, a prediction of the spectral values in relation to frequency is carried out in order to obtain spectral residual values. Within the spectral residual values, areas are detected encompassing spectral residual values with noise properties. The spectral residual values in the noise areas are noise-substituted, whereupon information concerning the noise areas and noise substitution is incorporated into side information pertaining to a coded audio signal. Thus, considerable bit savings in case of transient signals can be achieved.

摘要翻译： 用于对音频信号进行编码或解码的方法结合了TNS处理和噪声替换的优点。 时间离散音频信号最初被变换到频域以获得时间音频信号的频谱值。 随后，进行与频率相关的频谱值的预测，以获得谱残差值。 在光谱残差值内，检测到包含具有噪声特性的光谱残差值的区域。 噪声区域中的频谱残差值被噪声替代，因此关于噪声区域和噪声替换的信息被并入与编码音频信号有关的侧面信息中。 因此，可以实现在瞬态信号的情况下相当可观的位节省。

公开(公告)号：US06185539B2

公开(公告)日：2001-02-06

申请号：US09077395

申请日：1998-05-26

申请人： Oliver Kunz ,  Martin Dietz ,  Rainer Buchta ,  Jürgen Zeller ,  Karlheinz Brandenburg ,  Martin Sieler ,  Heinz Gerhäuser

发明人： Oliver Kunz ,  Martin Dietz ,  Rainer Buchta ,  Jürgen Zeller ,  Karlheinz Brandenburg ,  Martin Sieler ,  Heinz Gerhäuser

IPC分类号： H04B166

CPC分类号： H04B1/665

摘要： In a method for coding an audio signal digitized at a low sampling rate to obtain time domain audio samples. A frequency domain representation of the time domain audio samples is produced. The frequency domain representation includes successive frequency lines. These frequency lines are grouped into a plurality of scale factor bands. The successive frequency lines in a scale factor band are coded with the same scale factor. A plurality of regions is formed by grouping the scale factor bands, wherein successive scale factor bands form a region within which all the scale factors are coded with the same number of bits, which is determined according to the largest scale factor of the region. The scale factors assigned to scale factor bands within the highest region that includes the higher frequency successive frequency lines are set to zero. The frequency lines in the highest region are coded using the zero-valued scale factors that correspond to a multiplication factor of 1. The scale factors for the highest region, however, are not coded. Thus, the bits that would be required for coding these zero-valued scale factors are saved and can be used for a finer quantization of the rest of the spectrum. Additionally, this coding method when applied to ISO/IEC 13818-3 as a low sampling rate modification thereof only requires minimal changes with respect to this Standard.

摘要翻译： 在以低采样率对数字化的音频信号进行编码以获得时域音频样本的方法中， 产生时域音频样本的频域表示。 频域表示包括连续的频率线。 这些频率线被分组成多个比例因子频带。 缩放因子带中的连续频率线以相同的比例因子进行编码。 通过对比例因子频带进行分组来形成多个区域，其中，连续的比例因子波段形成一个区域，在该区域内，所有比例因子都以相同的比特数进行编码，这是根据该区域的最大比例因子确定的。 分配给包括较高频率的连续频率线的最高区域内的比例因子频带的比例因子被设置为零。 最高区域中的频率线使用与乘法因子1对应的零值比例因子进行编码。然而，最高区域的比例因子未被编码。 因此，编码这些零值比例因子所需的位被保存，并且可以用于其余频谱的更精细的量化。 此外，当将其应用于ISO / IEC 13818-3作为其低采样率修改时，该编码方法仅需要相对于本标准的最小变化。