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21. 发明申请

US20150206541A1 Method and Apparatus for Allocating Bits of Audio Signal 有权
标题翻译：用于分配音频信号位的方法和装置
公开(公告)号：US20150206541A1
公开(公告)日：2015-07-23
申请号：US14675031
申请日：2015-03-31
申请人： Huawei Technologies Co., Ltd.
发明人： Fengyan Qi , Zexin Liu , Lei Miao
IPC分类号： G10L19/002
CPC分类号： G10L19/002 , G10L19/0204 , G10L19/032 , G10L19/035
摘要： A method and an apparatus for allocating bits of an audio signal. The method includes dividing a frequency band of an audio signal into multiple sub-bands, and quantizing a sub-band normalization factor of each sub-band; classifying the multiple sub-bands into multiple groups, and acquiring a sum of intra-group sub-band normalization factors of each group; performing initial inter-group bit allocation to determine the initial number of bits of each group; performing secondary inter-group bit allocation to allocate coding bits of the audio signal to at least one group; and allocating the bits of the audio signal to sub-bands in the group. The present invention can, by means of grouping, ensure relatively stable allocation in a previous frame and a next frame and reduce an impact of global allocation on local discontinuity in a case of low and medium bit rates.
摘要翻译：一种用于分配音频信号的位的方法和装置。该方法包括将音频信号的频带划分成多个子带，并量化每个子带的子带归一化因子; 将多个子带分为多个组，并获取每组的组内子带归一化因子之和; 执行初始组间比特分配以确定每组的初始比特数; 执行次要组间比特分配以将音频信号的编码比特分配给至少一个组; 并将音频信号的比特分配给组中的子带。本发明可以通过分组确保在先前帧和下一帧中的相对稳定的分配，并且在低和中比特率的情况下减少全局分配对局部不连续性的影响。

22. 发明授权

US09031852B2 Data compression apparatus, computer-readable storage medium having stored therein data compression program, data compression system, data compression method, data decompression apparatus, data compression/decompression apparatus, and data structure of compressed data 有权
标题翻译：数据压缩装置，存储有数据压缩程序，数据压缩系统，数据压缩方法，数据解压缩装置，数据压缩/解压缩装置和压缩数据的数据结构的计算机可读存储介质
公开(公告)号：US09031852B2
公开(公告)日：2015-05-12
申请号：US13598826
申请日：2012-08-30
申请人： Tomokazu Abe
发明人： Tomokazu Abe
IPC分类号： G10L19/00 , G10L19/02 , G10L19/035
CPC分类号： G10L19/00 , G10L19/02 , G10L19/035
摘要： A data compression/decompression apparatus, for example, acquires sampling data obtained by sampling an audio signal with a predetermined period, and converts the sampling data into frequency domain data. The data compression/decompression apparatus divides a data sequence of the converted frequency domain data into a plurality of blocks such that the number of pieces of data included in each block is variable, and compresses each block.
摘要翻译：例如，数据压缩/解压缩装置获取通过以预定周期对音频信号进行采样而获得的采样数据，并将采样数据转换成频域数据。数据压缩/解压缩装置将转换的频域数据的数据序列划分成多个块，使得每个块中包括的数据块数是可变的，并压缩每个块。

23. 发明授权

US08983851B2 Noise filer, noise filling parameter calculator encoded audio signal representation, methods and computer program 有权
标题翻译：噪声滤波器，噪声填充参数计算器编码音频信号表示，方法和计算机程序
公开(公告)号：US08983851B2
公开(公告)日：2015-03-17
申请号：US13004493
申请日：2011-01-11
申请人： Nikolaus Rettelbach , Bernhard Grill , Guillaume Fuchs , Stefan Geyersberger , Markus Multrus , Harald Popp , Juergen Herre , Stefan Wabnik , Gerald Schuller , Jens Hirschfeld
发明人： Nikolaus Rettelbach , Bernhard Grill , Guillaume Fuchs , Stefan Geyersberger , Markus Multrus , Harald Popp , Juergen Herre , Stefan Wabnik , Gerald Schuller , Jens Hirschfeld
IPC分类号： G10L19/00 , G10L19/032 , G10L19/02 , G10L19/028 , G10L19/035 , G10L25/18
CPC分类号： G10L19/035 , G10L19/008 , G10L19/02 , G10L19/0204 , G10L19/028 , G10L19/032 , G10L25/18
摘要： A noise filler for providing a noise-filled spectral representation of an audio signal on the basis of an input spectral representation of the audio signal has a spectral region identifier configured to identify spectral regions of the input spectral representation spaced from non-zero spectral regions of the input spectral representation by at least one intermediate spectral region, to obtain identified spectral regions, and a noise inserter configured to selectively introduce noise into the identified spectral regions to obtain the noise-filled spectral representation of the audio signal. A noise filling parameter calculator for providing a noise filling parameter on the basis of a quantized spectral representation of an audio signal has a spectral region identifier, as mentioned above, and a noise value calculator configured to selectively consider quantization errors of the identified spectral regions for a calculation of the noise filling parameter. Accordingly, an encoded audio signal representation representing the audio signal can be obtained.
摘要翻译：用于基于音频信号的输入频谱表示来提供音频信号的噪声填充频谱表示的噪声填充器具有频谱区域标识符，其被配置为识别从非零频谱区域间隔开的输入频谱表示的频谱区域通过至少一个中间光谱区域的输入光谱表示，以获得所识别的光谱区域，以及噪声插入器，被配置为选择性地将噪声引入所识别的光谱区域中，以获得音频信号的噪声填充的频谱表示。用于根据音频信号的量化频谱表示提供噪声填充参数的噪声填充参数计算器具有如上所述的频谱区域标识符，以及噪声值计算器，被配置为选择性地考虑所识别的频谱区域的量化误差噪声填充参数的计算。因此，可以获得表示音频信号的编码音频信号表示。

24. 发明授权

US08892429B2 Encoding device and encoding method, decoding device and decoding method, and program 有权
标题翻译：编码装置和编码方法，解码装置及解码方法及程序
公开(公告)号：US08892429B2
公开(公告)日：2014-11-18
申请号：US13583994
申请日：2011-03-08
申请人： Shiro Suzuki , Yuuki Matsumura , Yasuhiro Toguri , Yuuji Maeda
发明人： Shiro Suzuki , Yuuki Matsumura , Yasuhiro Toguri , Yuuji Maeda
IPC分类号： G10L19/02 , G10L19/035
CPC分类号： G10L19/035 , G10L19/0212
摘要： The present invention relates to an encoding device and an encoding method, a decoding device and a decoding method, and a program that reduce deterioration of sound quality due to encoding of audio signals.An envelope emphasis part (51) emphasizes an envelope (ENV). A noise shaping part (52) divides an emphasized envelope (D) formed by emphasis of the envelope (ENV) by a value larger than 1, and subtracts noise shaping (G) specified by information (NS) from a result of the division. A quantization part (14) sets a result of the subtraction as a quantization bit count (WL), and quantizes a normalized spectrum (S1) formed by normalization of a spectrum (S0) based on the quantization bit count (WL). A multiplexing part (53) multiplexes the information (NS), a quantized spectrum (QS) formed by quantization of the normalized spectrum (S1), and the envelope (ENV). The present invention can be applied to an encoding device encoding audio signals, for example.
摘要翻译：编码装置和编码方法，解码装置和解码方法技术领域本发明涉及一种减少音频信号编码导致的音质劣化的程序。信封重点部分（51）强调信封（ENV）。噪声整形部分（52）将由包络（ENV）的强调形成的强调包络（D）除以大于1的值，并从分割结果中减去由信息（NS）指定的噪声整形（G）。量化部分（14）将减法的结果设置为量化位计数（WL），并且通过基于量化位计数（WL）对通过频谱归一化形成的归一化频谱（S1）进行量化。复用部分（53）复用信息（NS），通过归一化频谱（S1）的量化形成的量化频谱（QS）和信封（ENV）。例如，本发明可以应用于编码音频信号的编码装置。

25. 发明授权

US08738369B2 Enhancing performance of spectral band replication and related high frequency reconstruction coding 有权
标题翻译：提高频带复制性能和相关高频重构编码
公开(公告)号：US08738369B2
公开(公告)日：2014-05-27
申请号：US13973193
申请日：2013-08-22
申请人： Dolby International AB
发明人： Lars G. Liljeryd , Kristofer Kjoerling , Per Ekstrand , Fredrik Henn
IPC分类号： G10L19/00 , G10L19/14
CPC分类号： G10L19/028 , G10L19/035 , G10L19/06 , G10L19/26 , G10L19/265 , G10L21/038 , G10L25/18
摘要： The present proposes new methods and an apparatus for enhancement of source coding systems utilizing high frequency reconstruction (HFR). It addresses the problem of insufficient noise contents in a reconstructed highband, by Adaptive Noise-floor Addition. It also introduces new methods for enhanced performance by means of limiting unwanted noise, interpolation and smoothing of envelope adjustment amplification factors. The present invention is applicable to both speech coding and natural audio coding systems.
摘要翻译：本发明提出了利用高频重建（HFR）来增强源编码系统的新方法和装置。它通过自适应噪声层加法解决了重构高频带中噪声含量不足的问题。它还通过限制不需要的噪声，包络调整放大因子的插值和平滑来介绍新的增强性能的方法。本发明适用于语音编码和自然音频编码系统。

26. 发明授权

US08706509B2 Method and a decoder for attenuation of signal regions reconstructed with low accuracy 有权
标题翻译：用于以低精度重建的信号区域衰减的方法和解码器
公开(公告)号：US08706509B2
公开(公告)日：2014-04-22
申请号：US13379054
申请日：2011-12-15
申请人： Sebastian Näslund , Volodya Grancharov , Erik Norvell
发明人： Sebastian Näslund , Volodya Grancharov , Erik Norvell
IPC分类号： G10L19/00 , G10L21/00
CPC分类号： G10L19/02 , G10L19/0212 , G10L19/035 , G10L19/038 , G10L19/10
摘要： The embodiments of the present invention improves conventional attenuation schemes by replacing constant attenuation with an adaptive attenuation scheme that allows more aggressive attenuation, without introducing audible change of signal frequency characteristics.
摘要翻译：本发明的实施例通过用允许更积极衰减的自适应衰减方案代替恒定衰减来改进常规衰减方案，而不引入信号频率特性的可听见的改变。

27. 发明申请

US20140006790A1 METHOD AND APPARATUS FOR WATERMARKING AN AC-3 ENCODED BIT STREAM 有权
标题翻译：用于水印AC-3编码位流的方法和装置
公开(公告)号：US20140006790A1
公开(公告)日：2014-01-02
申请号：US13928142
申请日：2013-06-26
申请人： THOMSON LICENSING
发明人： Xiaoming CHEN , Peter Georg Baum , Michael Arnold , Ulrich Gries
IPC分类号： H04N21/845
CPC分类号： H04N21/845 , G10L19/018 , G10L19/0212 , G10L19/035
摘要： For watermark embedding, without full Dolby AC-3 decoding, mantissa bits in the AC-3 bit stream are changed by exploiting the fact that during AC-3 encoding more bits than required by the perceptual masking curve are used for the quantization of the MDCT coefficient mantissa values. In addition to the bap values, truly required bap values are determined and the mantissa values are changed accordingly, controlled by the watermark information. The advantages are efficient watermark embedding, other bit stream parameters and values remain unchanged except for CRC checks, and the audio quality of the watermarked signal remains perceptually unchanged.
摘要翻译：对于水印嵌入，在没有完全杜比AC-3解码的情况下，AC-3比特流中的尾数位被改变，这是因为在AC-3编码期间，比感知掩蔽曲线所需的更多的比特用于MDCT的量化系数尾数值。除了bap值之外，确定真正需要的bap值，并且相应地改变尾数值，由水印信息控制。优点是有效的水印嵌入，其他比特流参数和值保持不变，除了CRC校验，水印信号的音频质量保持感知不变。

28. 发明申请

US20130282382A1 Audio Encoder and Decoder 有权
标题翻译：音频编码器和解码器
公开(公告)号：US20130282382A1
公开(公告)日：2013-10-24
申请号：US13901960
申请日：2013-05-24
申请人： DOLBY INTERNATIONAL AB
发明人： Per Hedelin , Pontus Carlsson , Leif Jonas Samuelsson , Michael Schug
IPC分类号： G10L19/26
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。

29. 发明授权

US08484019B2 Audio encoder and decoder 有权
公开(公告)号：US08484019B2
公开(公告)日：2013-07-09
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： G10L19/00
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.

30. 发明授权

US08468026B2 Audio decoding using variable-length codebook application ranges 有权
标题翻译：使用可变长度码本应用范围进行音频解码
公开(公告)号：US08468026B2
公开(公告)日：2013-06-18
申请号：US13568705
申请日：2012-08-07
申请人： Yuli You
发明人： Yuli You
IPC分类号： G10L11/00 , G10L21/00 , G10L19/14 , G10L19/00 , G10L19/12 , G10L21/04 , H04B1/66 , H04N11/02
CPC分类号： G10L19/025 , G10L19/008 , G10L19/0208 , G10L19/022 , G10L19/035 , G10L19/038 , G10L19/167 , G10L19/24
摘要： Provided are, among other things, systems, methods and techniques for decoding an audio signal from a frame-based bit stream. At least one frame includes processing information pertaining to the frame and entropy-encoded quantization indexes representing audio data within the frame. The processing information includes: (i) code book indexes, and (ii) code book application information specifying ranges of entropy-encoded quantization indexes to which the code books are to be applied. The entropy-encoded quantization indexes are decoded by applying the identified code books to the corresponding ranges of entropy-encoded quantization indexes.
摘要翻译：尤其提供了用于从基于帧的比特流解码音频信号的系统，方法和技术。至少一帧包括处理与该帧有关的信息和表示该帧内的音频数据的熵编码的量化索引。处理信息包括：（i）代码簿索引，以及（ii）指定要应用代码簿的熵编码量化索引的范围的代码本申请信息。熵编码的量化索引通过将所识别的码本应用于熵编码的量化索引的对应范围来解码。

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