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    • 1. 发明申请
    • Spectrum modeling
    • 频谱建模
    • US20060129389A1
    • 2006-06-15
    • US11345993
    • 2006-02-02
    • Albertus Den BrinkerArnoldus Oomen
    • Albertus Den BrinkerArnoldus Oomen
    • G10L19/04
    • G10L19/06G10L21/0208G10L25/12G10L25/18H03H17/0258
    • Modeling a target spectrum (S) is provided by determining (21) filter parameters (pi,qi) of a filter which has a frequency response approximating the target spectrum (S), wherein the target spectrum is split in at least a first part and a second part, a first modeling operation is used on the first part of the target spectrum to obtain auto-regressive parameters, a second modeling operation is used on the second part of the target spectrum to obtain moving-average parameters, and the auto-regressive parameters and the moving-average parameters are combined to obtain the filter parameters. The invention is preferably applied in audio coding, wherein a spectrum of a noise component (S) in the signal (A) is modeled.
    • 通过确定具有近似于目标光谱的频率响应的滤波器的滤波器参数(21),来提供对目标频谱(S)的建模( S),其中目标光谱在至少第一部分和第二部分中分裂,在目标光谱的第一部分上使用第一建模操作以获得自回归参数,在第二部分使用第二建模操作 获得移动平均参数的目标频谱的一部分,并且组合自回归参数和移动平均参数以获得滤波器参数。 本发明优选地应用于音频编码,其中对信号(A)中的噪声分量(S)的频谱进行建模。
    • 3. 发明申请
    • Low bit-rate audio encoding
    • 低比特率音频编码
    • US20070112560A1
    • 2007-05-17
    • US10564656
    • 2004-07-08
    • Andreas GerritsAlbertus Den Brinker
    • Andreas GerritsAlbertus Den Brinker
    • G10L19/14
    • G10L19/093
    • In a sinusoidal audio encoder a number of sinusoids are estimated per audio segment. A sinusoid is represented y frequency, amplitude and phase. Normally, phase is quantised independent of frequency The invention uses a frequency dependent quantisation of phase, and in particular the low frequencies are quantised using smaller quantisation intervals than at higher frequencies. Thus, the unwrapped phases of the lower frequencies are quantised more accurately, possibly with a smaller quantisation range, than the phases of the higher frequencies. The invention gives a significant improvement in decoded signal quality, especially for low bit-rate quantisers
    • 在正弦音频编码器中,每个音频段估计出多个正弦波。 正弦曲线表示y频率,振幅和相位。 通常,相位与频率无关地量化本发明使用相位的频率相关量化,特别是使用比较高频率更小的量化间隔量化低频。 因此,与较高频率的相位相比,更准确地量化更低频率的展开相位,可能具有较小的量化范围。 本发明给出了解码信号质量的显着改进,特别是对于低比特率量化器
    • 6. 发明申请
    • Audio encoding
    • 音频编码
    • US20070100639A1
    • 2007-05-03
    • US10575428
    • 2004-10-04
    • Albertus Den BrinkerAndreas Gerrits
    • Albertus Den BrinkerAndreas Gerrits
    • G10L21/00
    • G10L19/032G10L19/093
    • Coding of an audio signal (x) represented by a respective set of sampled signal values (x(t)) for each of a plurality of sequential time segments is disclosed. The sampled signal values are analyzed to determine one or more sinusoidal components for each of the plurality of sequential segments. The sinusoidal components are linked across a plurality of sequential segments to provide sinusoidal tracks, where each track comprises a number of frames. An encoded signal (AS) is generated, including sinusoidal codes (Cs) comprising a representation level (r) for each frame or including sinusoidal codes (Cs) where some of these codes comprise a phase (φ), a frequency (ω) and a quantization table (Q) for a given frame when the given frame is designated as a random-access frame. The invention allows random access in a track while avoiding long adaptation of the quantization accuracy in a quantizer and/or the need for a large bit stream while still maintaining improved audio quality.
    • 公开了对于多个连续时间段中的每一个对由相应的一组采样信号值(x(t))表示的音频信号(x)的编码。 分析采样的信号值以确定多个顺序段中的每一个的一个或多个正弦分量。 正弦分量连接在多个顺序段上以提供正弦曲线,其中每个轨道包括多个帧。 生成编码信号(AS),包括包括每帧的表示级(r)或包括正弦代码(C SUB)的正弦码(C SUB),其中一些 的这些代码包括当给定帧被指定为随机接入帧时给定帧的相位(phi),频率(ω)和量化表(Q)。 本发明允许在轨道中随机访问,同时避免量化器中的量化精度的长时间适应性和/或对大比特流的需要,同时仍然保持改善的音频质量。
    • 7. 发明申请
    • Sinusoid selection in audio encoding
    • 音素编码中的正弦选择
    • US20060212501A1
    • 2006-09-21
    • US10539311
    • 2003-11-19
    • Andreas GerritsAlbertus Den Brinker
    • Andreas GerritsAlbertus Den Brinker
    • G06F17/10
    • G10L19/093
    • A method of encoding (1) an audio signal (x(t)) by representing (12) at least part of the audio signal by a plurality of sinusoids, the method comprising the steps of performing an analysis on a first segment of said audio signal, selecting candidate sinusoids based on said analysis, determining for at least one of the candidate sinusoids a phase consistency defined by an extent to which a phase of said candidate sinusoid at a certain moment in time can be predicted from a phase of said candidate sinusoid determined at another moment in time, and selecting said candidate sinusoid as a selected sinusoid when its phase consistency is above a predetermined threshold. The selection of sinusoids according to the invention will result in a smaller number of sinusoids to be encoded for a given audio quality, which is advantageous in terms of bit-rate for a given audio quality.
    • 一种通过用多个正弦曲线表示(12)至少部分音频信号来对(1)音频信号(x(t))进行编码的方法,所述方法包括以下步骤:对所述音频的第一段进行分析 信号,基于所述分析来选择候选正弦波,确定候选正弦曲线中的至少一个候选正弦曲线,其相位一致性定义在从所述候选正弦曲线的相位可以预测在某一特定时刻的所述候选正弦曲线的相位的程度 在时间上的另一时刻确定,并且当其相位一致性高于预定阈值时,将所述候选正弦波选择为选择的正弦曲线。 根据本发明的正弦曲线的选择将导致对于给定音频质量而要编码的较少数量的正弦曲线,这在给定音频质量的比特率方面是有利的。
    • 8. 发明申请
    • Audio coding
    • 音频编码
    • US20050228656A1
    • 2005-10-13
    • US10515746
    • 2003-05-16
    • Albertus Den Brinker
    • Albertus Den Brinker
    • G10L19/06G10L19/04G10L25/12G10L19/14
    • G10L19/04G10L25/12
    • A method of encoding (14) an audio signal (x(n)) is disclosed. The method comprises the step of modelling (16) the audio signal in accordance with a frequency sensitizing parameter (( ) to provide a set of infinite impulse response (IIR) filter type characteristics ((0 . . . k−1) of an order K and capable of being linearly combined with the sensitizing parameter (( ) to provide an estimate ( ) for the audio signal (x(n)), the IIR type filter model satisfying the requirements of a minimum phase filter. The set of characteristics ((0 . . . k−1) of order K are transformed as a function of the sensitizing parameter (( ) to provide a set of characteristics (c0 . . . k) of order K+1 compatible with finite impulse response (FIR) filter type characteristics satisfying the requirements of a minimum phase filter. The set of characteristics (c0 . . . k) of order K+1 are normalised to provide a set of characteristics (d1 . . . k) of order K. An encoded audio stream (50) is generated to include representations (LAR,LSFs) of the normalised set of characteristics (d1 . . . k) of order K.
    • 公开了一种编码(14)音频信号(x(n))的方法。 该方法包括根据频率敏化参数(())对音频信号进行建模(16)的步骤,以提供一组无限脉冲响应(IIR)滤波器类型特征((...,k-1)) K,并且能够与敏化参数(())线性组合以提供音频信号(x(n))的估计(),满足最小相位滤波器要求的IIR型滤波器模型。 顺序K的(0 ... k-1)作为敏化参数(()的函数被变换,以提供与有限脉冲响应兼容的阶数K + 1的一组特征(c 0。...) )滤波器类型特性满足最小相位滤波器的要求,将K + 1级的特性集(c 0。。k)归一化,以提供一个K阶特性(d 1。k)。 生成编码音频流(50)以包括归一化的一组chara的表示(LAR,LSF) 手段(d 1。 。 。 k)。
    • 9. 发明申请
    • Method and Apparatus to Encode and Decode Multi-Channel Audio Signals
    • 编码和解码多声道音频信号的方法和装置
    • US20070248157A1
    • 2007-10-25
    • US11570522
    • 2005-06-14
    • Albertus Den Brinker
    • Albertus Den Brinker
    • H04B1/66
    • G10L19/008G10L19/04G10L25/12G10L25/27
    • An encoder (100) for encoding a multi-channel audio signal comprises a prediction processor (101) which generates two residual signals for two signal components of the multi-channel signal by linear prediction which is associated with psycho-acoustic prediction filters. A rotation processor (105) rotates the combined signal of the two residual signals to generate a main signal and a side signal. Preferably, the energy of the main signal is maximized and the energy of the side signal is minimized. An encoding processor (109) encodes the main and preferably the side signal and an output processor (111) generates an output signal comprising the encoded main data and preferably the side data, prediction parameters and rotation parameters. The combination of linear prediction, use of psycho-acoustic characteristics and the general encoder (100) for encoding a multi-channel signal comprises a prediction processor (101) which generates two residual signals for two signal components of the multi-channel signal by linear prediction which is associated with psycho-acoustic characteristics and which specifically uses psycho-acoustic prediction filters. A rotation processor (105) rotates the combined signal of the two residual signals to generate a main signal and a side signal. Preferably, the energy of the main signal is maximized and the energy of the side signal is minimized. An encoding processor (109) encodes the main and preferably the side signal and an output processor (111) generates an output signal comprising the encoded main data and preferably the side data, prediction parameters and rotation parameters. The combination of linear prediction, use of psycho-acoustic characteristics and the generation of a main and side signal improves encoding and enhances the flexibility of the encoder for different data rates.
    • 用于对多声道音频信号进行编码的编码器(100)包括预测处理器(101),其通过与心理声学预测滤波器相关联的线性预测产生用于多声道信号的两个信号分量的两个残余信号。 旋转处理器(105)旋转两个残差信号的组合信号以产生主信号和侧信号。 优选地,主信号的能量最大化并且侧信号的能量被最小化。 编码处理器(109)对主要信号和优选的侧信号进行编码,并且输出处理器(111)产生包括编码的主数据和优选的侧数据,预测参数和旋转参数的输出信号。 线性预测,心理声学特性的使用和用于编码多声道信号的通用编码器(100)的组合包括预测处理器(101),其通过线性方式为多声道信号的两个信号分量产生两个残余信号 与心理声学特征相关的预测,其具体使用心理声学预测滤波器。 旋转处理器(105)旋转两个残差信号的组合信号以产生主信号和侧信号。 优选地,主信号的能量最大化并且侧信号的能量被最小化。 编码处理器(109)对主要信号和优选的侧信号进行编码,并且输出处理器(111)产生包括编码的主数据和优选的侧数据,预测参数和旋转参数的输出信号。 线性预测,心理声学特性的使用以及主要和侧面信号的产生的组合改善了编码并提高了编码器对不同数据速率的灵活性。
    • 10. 发明申请
    • Quality of decoded audio by adding noise
    • 通过增加噪声对解码音频的质量
    • US20070124136A1
    • 2007-05-31
    • US10562359
    • 2004-06-25
    • Albertus Den BrinkerFrancois Myburg
    • Albertus Den BrinkerFrancois Myburg
    • G10L21/00
    • G10L21/038
    • The present invention relates to a method of encoding and decoding an audio signal. The invention further relates to an arrangement for encoding and decoding an audio signal. The invention further relates to a computer-readable medium comprising a data record indicative of an audio signal and a device for communicating an audio signal having been encoded according to the present invention. By the method of encoding, a double description of the signal is obtained, where the encoding comprises two encoding steps, a first standard encoding and an additional second encoding. The second encoding is able to give a coarse description of the signal, such that a stochastic realization can be made and appropriate parts can be added to the decoded signal from the first decoding. The required description of the second encoder in order to make the realization of a stochastic signal possible requires a relatively low bit rate, while other double/multiple descriptions require a much higher bit rate.
    • 本发明涉及一种对音频信号进行编码和解码的方法。 本发明还涉及用于对音频信号进行编码和解码的装置。 本发明还涉及包括指示音频信号的数据记录的计算机可读介质和用于传送根据本发明已被编码的音频信号的设备。 通过编码的方法,获得对信号的双重描述,其中编码包括两个编码步骤,第一标准编码和附加的第二编码。 第二编码能够给出信号的粗略描述,使得可以进行随机实现,并且可以从第一解码将合适的部分添加到解码信号。 为了使随机信号的实现可能需要第二编码器的所需描述需要相对低的比特率,而其他双重/多重描述需要高得多的比特率。