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公开(公告)号：US08756056B2
公开(公告)日：2014-06-17
申请号：US12496880
申请日：2009-07-02
IPC分类号： G10L19/032 , G10L19/002
CPC分类号： G10L19/032 , G10L2019/0005
摘要： For determining a quantizer step size for quantizing a signal including audio or video information, a first quantizer step size as well as an interference threshold are provided. Then, the actual interference introduced by the first quantizer step size is determined and compared with the interference threshold. Despite the fact that the comparison reveals that the actually introduced interference exceeds the threshold, a second, coarser quantizer step size is nevertheless used, which will then be used for quantization if it turns out that the interference introduced by the coarser, second quantizer step size falls below the threshold or falls below the interference introduced by the first quantizer step size. Thus, the quantization interference is reduced while the quantization is coarsened and, thus, the compression gain is increased.
摘要翻译： 为了确定用于量化包括音频或视频信息的信号的量化器步长，提供第一量化器步长以及干扰阈值。 然后，确定由第一量化器步长引入的实际干扰并将其与干扰阈值进行比较。 尽管比较显示实际引入的干扰超过阈值，但是仍然使用第二较粗略的量化器步长，然后将其用于量化，如果证明由较粗的第二量化器步长引入的干扰 低于阈值或低于由第一量化器步长引入的干扰。 因此，量化干扰减小，而量化粗大，因此压缩增益增加。
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2.发明申请公开(公告)号：US20120328115A1
公开(公告)日：2012-12-27
申请号：US13581453
申请日：2011-03-07
IPC分类号： H03G7/00
摘要： The present document relates to processing of multimedia data, notably the encoding, the transmission, the decoding and the rendering of multimedia data, e.g. audio files or bitstreams. In particular, the present document relates to the implementation of loudness control in multimedia players. A method for providing loudness related data to a media player is described. The method comprises the steps of providing a first loudness related value associated with an audio signal; wherein the first loudness related value has been determined according to a first procedure; of converting the first loudness related value into a second loudness related value using a reversible relation; wherein the second loudness related value is associated with a second procedure for determining loudness related values; of storing the second loudness related value in metadata associated with the audio signal; and of providing the metadata to the media player.
摘要翻译： 本文件涉及多媒体数据的处理，特别是多媒体数据的编码，传输，解码和呈现，例如， 音频文件或比特流。 特别地，本文件涉及多媒体播放器中的响度控制的实现。 描述了向媒体播放器提供响度相关数据的方法。 该方法包括以下步骤：提供与音频信号相关联的第一响度相关值; 其中所述第一响度相关值已经根据第一过程确定; 使用可逆关系将第一响度相关值转换为第二响度相关值; 其中所述第二响度相关值与用于确定响度相关值的第二过程相关联; 将第二响度相关值存储在与音频信号相关联的元数据中; 并向媒体播放器提供元数据。
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3.发明申请System and Method for Non-destructively Normalizing Loudness of Audio Signals Within Portable Devices 有权便携式设备内音频信号响度的非破坏性系统和方法公开(公告)号：US20120310654A1
公开(公告)日：2012-12-06
申请号：US13576386
申请日：2011-02-03
IPC分类号： G10L19/00
CPC分类号： G10L19/22 , G10L19/02 , G10L19/0208 , G10L19/167 , G10L19/26 , H03G3/3089 , H03G3/32 , H03G7/007
摘要： Many portable playback devices cannot decode and playback encoded audio content having wide bandwidth and wide dynamic range with consistent loudness and intelligibility unless the encoded audio content has been prepared specially for these devices. This problem can be overcome by including with the encoded content some metadata that specifies a suitable dynamic range compression profile by either absolute values or differential values relative to another known compression profile. A playback device may also adaptively apply gain and limiting to the playback audio. Implementations in encoders, in transcoders and in decoders are disclosed.
摘要翻译： 许多便携式播放设备不能解码和播放具有宽带宽和宽动态范围的编码音频内容，具有一致的响度和清晰度，除非编码音频内容已经为这些设备专门准备。 通过使用编码的内容包含一些通过相对于另一已知压缩简档的绝对值或差分值来指定合适的动态范围压缩简档的元数据来克服该问题。 播放装置还可以自适应地对播放音频应用增益和限制。 公开了在编码器，代码转换器和解码器中的实现。
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公开(公告)号：US20140072120A1
公开(公告)日：2014-03-13
申请号：US14113362
申请日：2012-05-07
申请人： Michael Schug , Harald H. Mundt
发明人： Michael Schug , Harald H. Mundt
IPC分类号： H04S1/00
CPC分类号： H04S1/007 , G10L19/008 , G10L19/03
摘要： The invention discloses a method and an encoder for processing a digital audio stereo signal. A digital audio encoder for coding such audio signal comprises a predictive Temporal Noise Shaping (TNS) filter, a Mid-/Side (M/S) coding unit, a control unit for determining a first prediction gain related to the unmodified L/R signal processed by the TNS filter and for determining a second prediction gain related to the M/S-coded L/R signal processed by the TNS filter, wherein the control unit is adapted to disable TNS-filtering—i.e. to bypass the TNS filter—for a current signal frame, if the first and second prediction gains differ by more than a pre-determined mismatch range. Preferably, the first and second prediction gains are determined from signal energy ratios calculated for each channel of the stereo signal including the signal energies of both the TNS-processed (unmodified) L- respectively (unmodified) R-signal and the TNS-processed M/S coded L- respectively M/S coded R-signal divided by the respective signal energies before TNS processing. Furthermore, the control unit is preferably adapted to overrule the disabling of the TNS filter, if the input signal is a near-mono audio signal exhibiting only low energy either in its M- or S-band. In that case, operation of the TNS filter on the stereo audio signal is maintained.
摘要翻译： 本发明公开了一种用于处理数字音频立体声信号的方法和编码器。 用于对这种音频信号进行编码的数字音频编码器包括预测时间噪声整形（TNS）滤波器，中/侧（M / S）编码单元，用于确定与未修改的L / R信号相关的第一预测增益的控制单元 由TNS滤波器处理并确定与由TNS滤波器处理的M / S编码的L / R信号相关的第二预测增益，其中该控制单元用于禁用TNS滤波 如果第一和第二预测增益相差超过预定的不匹配范围，则绕过TNS滤波器以获得当前信号帧。 优选地，第一和第二预测增益是根据对包括TNS处理（未修改）L信号和TNS处理的M信号的两个信号能量的立体声信号的每个信道计算的信号能量比确定的 / S编码的L-分别M / S编码的R信号除以TNS处理之前的各个信号能量。 此外，如果输入信号是在其M波段或S波段中仅表现出低能量的近乎单声道的音频信号，则控制单元优选地适用于推翻TNS滤波器的禁用。 在这种情况下，维持TNS滤波器对立体声音频信号的操作。
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公开(公告)号：US08655652B2
公开(公告)日：2014-02-18
申请号：US12446164
申请日：2007-09-25
申请人： Michael Schug
发明人： Michael Schug
CPC分类号： G10L19/032
摘要： An apparatus for encoding an information signal having discrete values includes a quantizer having a quantizer border, wherein the quantizer is adapted so that a discrete value above the quantization border is quantized to a quantization index, which is different from a quantization index obtained by quantizing a discrete value below the quantization border, a controller for modifying the quantization border, wherein the quantizer having a first quantization border setting is adapted to generate a first set of quantization indices for the discrete values, and wherein the quantizer having a second modified quantization border setting is adapted to generate a second set of quantization indices, and an output interface for outputting an encoded information signal which is either based on the first set of quantization indices or the second set of quantization indices dependent on a decision function.
摘要翻译： 用于对具有离散值的信息信号进行编码的装置包括具有量化器边界的量化器，其中量化器适于使得高于量化边界的离散值被量化为量化指标，其不同于通过量化 离散值低于量化边界，用于修改量化边界的控制器，其中具有第一量化边界设置的量化器适于生成用于离散值的第一组量化索引，并且其中量化器具有第二修改量化边界设置 适于产生第二组量化索引，以及输出接口，用于输出基于第一组量化索引的编码信息信号或者取决于决策函数的第二组量化索引。
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公开(公告)号：US20100286991A1
公开(公告)日：2010-11-11
申请号：US12811421
申请日：2008-12-30
申请人： Per Hedelin , Pontus Carlsson , Jonas Samuelsson , Michael Schug
发明人： Per Hedelin , Pontus Carlsson , Jonas Samuelsson , Michael Schug
IPC分类号： G10L21/00
CPC分类号： G10L19/26 , G10L19/008 , G10L19/032 , G10L19/035
摘要： The present invention teaches a new audio coding system that can code both general audio and speech signals well at low bit rates. A proposed audio coding system comprises linear prediction unit for filtering an input signal based on an adaptive filter; a transformation unit for transforming a frame of the filtered input signal into a transform domain; and a quantization unit for quantizing the transform domain signal. The quantization unit decides, based on input signal characteristics, to encode the transform domain signal with a model-based quantizer or a non-model-based quantizer. Preferably, the decision is based on the frame size applied by the transformation unit.
摘要翻译： 本发明教导了一种新的音频编码系统，其可以以低比特率良好地对一般音频和语音信号进行编码。 所提出的音频编码系统包括用于基于自适应滤波器对输入信号进行滤波的线性预测单元; 变换单元，用于将经滤波的输入信号的帧变换为变换域; 以及用于量化变换域信号的量化单元。 量化单元基于输入信号特性来决定用基于模型的量化器或非基于模型的量化器对变换域信号进行编码。 优选地，该决定基于由变换单元应用的帧大小。
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公开(公告)号：US20070033056A1
公开(公告)日：2007-02-08
申请号：US11464315
申请日：2006-08-14
申请人： Juergen Herre , Michael Schug , Alexander Groeschl
发明人： Juergen Herre , Michael Schug , Alexander Groeschl
CPC分类号： G10L19/03 , G10L19/008
摘要： An apparatus for processing a multi-channel signal includes a means for determining a similarity between a first one of two channels and a second one of the two channels. Furthermore, a means for performing a prediction filtering of the spectral coefficients is provided, which is formed to perform a prediction filtering with only a single prediction filter for both channels in case of high similarity between the first and the second channel, and to perform a prediction filtering with two separate prediction filters in case of a dissimilarity between the first and the second channel. With this, an introduction of stereo artifacts and a deterioration of the coding gain in stereo coding techniques are avoided.
摘要翻译： 用于处理多信道信号的装置包括用于确定两个信道中的第一个信道和两个信道中的第二信道之间的相似性的装置。 此外，提供了一种用于执行频谱系数的预测滤波的装置，其被形成为在第一和第二信道之间具有高相似性的情况下仅对两个信道执行预测滤波，并且执行 在第一和第二信道之间具有不相似性的情况下，使用两个单独的预测滤波器进行预测滤波。 由此，避免立体声伪影的引入和立体声编码技术中的编码增益的恶化。
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8.发明授权公开(公告)号：US07050972B2
公开(公告)日：2006-05-23
申请号：US09987657
申请日：2001-11-15
申请人： Fredrik Henn , Andrea Ehret , Michael Schug
发明人： Fredrik Henn , Andrea Ehret , Michael Schug
IPC分类号： G10L21/02
CPC分类号： G10L19/18 , G10L21/038
摘要： An apparatus for encoding an audio signal to obtain an encoded audio signal to be used by a decoder having a high frequency reconstruction module for performing a high frequency reconstruction for a frequency range above a crossover frequency includes, a core encoder for encoding a lower frequency band of the audio signal up to the crossover frequency, the crossover frequency being variable, and the core encoder being operable on a block-wise frame by frame basis, and a crossover frequency control module for estimating, dependent on a measure of the degree of difficulty for encoding the audio signal by the core encoder and/or a boarder between a tonal and a noise-like frequency range of the audio signal, the crossover frequency to be selected by the core encoder for a frame of a series of subsequent frames, so that the crossover frequency is variable adaptively over time for the series of subsequent frames.
摘要翻译： 一种用于对音频信号进行编码以获得要由具有用于在高于交越频率的频率范围执行高频重建的高频重建模块的解码器使用的编码音频信号的装置包括：用于编码较低频带的核心编码器 所述音频信号直到所述交叉频率，所述交叉频率是可变的，并且所述核心编码器可以逐帧地逐帧操作;以及交叉频率控制模块，用于根据难度测量值来估计 用于通过核心编码器和/或边缘在音频信号的音调和类似噪声的频率范围之间编码音频信号，由核心编码器为一系列后续帧的帧选择的交叉频率，因此 交叉频率随着时间的推移随后续帧的变化而自适应变化。
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9.发明授权Apparatus for generating and interpreting a data stream with segments having specified entry points 有权用于生成和解释具有指定入口点的段的数据流的装置公开(公告)号：US07809018B2
公开(公告)日：2010-10-05
申请号：US11611290
申请日：2006-12-15
IPC分类号： H04J3/24
CPC分类号： G10L19/167 , H04L1/0045
摘要： An apparatus for generating a data stream having a series of segments using data organized in subsequent data frames. The apparatus comprises a packetiser for packetising data from a data frame into the series of segments having a first segment and a second segment, where the second segment has interpretable data entities and has a data entity fragment, the data entity fragment including only a part of an interpretable data entity preceding an interpretable data entity. The apparatus comprises furthermore an information block adder for adding an information block associated with the second segment, the information block indicating an entry point into the second segment, the entry point indicating a start of the interpretable data entity following the data entity fragment.
摘要翻译： 一种用于使用在后续数据帧中组织的数据来生成具有一系列段的数据流的装置。 该装置包括分组器，用于将数据从数据帧打包成具有第一段和第二段的段，其中第二段具有可解释数据实体并具有数据实体段，数据实体段仅包括一部分 在可解释数据实体之前的可解释数据实体。 该装置还包括用于将与第二段相关联的信息块相加的信息块加法器，将指示入口点的信息块添加到第二段中，该入口点指示跟随数据实体段落的可解释数据实体的开始。
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公开(公告)号：US07574355B2
公开(公告)日：2009-08-11
申请号：US11514006
申请日：2006-08-30
CPC分类号： G10L19/032 , G10L2019/0005
摘要： For determining a quantizer step size for quantizing a signal including audio or video information, a first quantizer step size as well as an interference threshold are provided. Then, the actual interference introduced by the first quantizer step size is determined and compared with the interference threshold. Despite the fact that the comparison reveals that the actually introduced interference exceeds the threshold, a second, coarser quantizer step size is nevertheless used, which will then be used for quantization if it turns out that the interference introduced by the coarser, second quantizer step size falls below the threshold or falls below the interference introduced by the first quantizer step size. Thus, the quantization interference is reduced while the quantization is coarsened and, thus, the compression gain is increased.
摘要翻译： 为了确定用于量化包括音频或视频信息的信号的量化器步长，提供第一量化器步长以及干扰阈值。 然后，确定由第一量化器步长引入的实际干扰并将其与干扰阈值进行比较。 尽管比较显示实际引入的干扰超过阈值，但是仍然使用第二较粗略的量化器步长，然后将其用于量化，如果证明由较粗的第二量化器步长引入的干扰 低于阈值或低于由第一量化器步长引入的干扰。 因此，量化干扰减小，而量化粗大，因此压缩增益增加。
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