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    • 35. 发明授权
    • METHOD AND APPARATUS TO ENCODE AND DECODE MULTI-CHANNEL AUDIO SIGNALS
    • FOR编码的方法和装置和解码多声道音频
    • EP1761915B1
    • 2008-12-03
    • EP05746699.7
    • 2005-06-14
    • Koninklijke Philips Electronics N.V.
    • DEN BRINKER, Albertus, C.
    • G10L19/00
    • 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.
    • 37. 发明公开
    • MULTI-CHANNEL ENCODER
    • MEHRKANAL-CODIERER
    • EP1735777A1
    • 2006-12-27
    • EP05718571.2
    • 2005-03-25
    • Koninklijke Philips Electronics N.V.
    • HOTHO, Gerard, H.BREEBAART, Dirk, J.VERBITSKIY, Evgeny, A.DEN BRINKER, Albertus, C.
    • G10L19/02H04S1/00H04S3/00
    • G10L19/02G10L19/008G10L19/0204H04S3/008
    • There is described a method of encoding input signals (CHI to CH3; 400 to 450) in a multi-channel encoder (5; 15) to generate corresponding output data comprising down-mix output signals (610, 620) together with complementary parametric data (600). The method includes a first step of down-mixing input signals (CHI to CH3; 400 to 450) to generate the corresponding down-mix output signals (610, 620), and a second step of processing the input signals (CHI to CH3; 400 to 450) during down-mixing to generate said parametric data (600) complementary to the down-mix output signals (610, 620). Processing of the input signals (CHI to CH3; 400 to 450) involves including information in the down­-mix signals (610, 620) which is useable during subsequent decoding of the down-mix output signals (610, 620) and the parametric data (600) to determine at least some parameter data and thereby enabling representations of the input signals (CHI to CH3; 400 to 450) to be subsequently regenerated. Coders for use in the encoder (5; 15) for performing essential signal processing operations therein are also elucidated.
    • 描述了在多通道编码器(5; 15)中对输入信号(CH1至CH3; 400至450)进行编码的方法,以生成相应的输出数据,包括下混合输出信号(610,620)以及互补参数数据 (600)。 该方法包括将输入信号(CH1至GH3; 400至450)下变频以产生相应的下混合输出信号(610,620)的第一步骤,以及处理输入信号(CH1至CH3; 400至450),以产生与下混合输出信号(610,620)互补的所述参数数据(600)。 输入信号(CH1至CH3; 400至450)的处理涉及将下变频信号(610,620)中的信息包括在下混合输出信号(610,620)的随后解码期间可用的信息和参数数据 (600)以确定至少一些参数数据,从而使得能够随后再生所述输入信号(CH1至CH3; 400至450)的表示。 还阐明了用于在编码器(5; 15)中进行基本信号处理操作的编码器。
    • 38. 发明公开
    • AUDIO CODING
    • 音频编码
    • EP1692688A1
    • 2006-08-23
    • EP04799235.9
    • 2004-11-24
    • Koninklijke Philips Electronics N.V.
    • GERRITS, Andreas, J.DEN BRINKER, Albertus, C.RIERA PALOU, Felip
    • G10L19/14G10L19/08G10L19/10
    • G10L19/093G10L19/10G10L19/24
    • An audio coder is arranged to process a respective set of sampled signal values for each of a plurality of sequential segments of an audio signal (x). The coder comprises an analyser (TSA) arranged to analyse the sampled signal values to provide one or more sinusoidal codes (Cs) corresponding to respective sinusoidal components of the audio signal. A subtractor subtracts a signal corresponding to the sinusoidal components from the audio signal to provide a first residual signal (r1). A modeller (SEG) models the frequency spectrum of the first residual signal (r1) by determining first filter parameters (Ps) of a filter which has a frequency response approximating a frequency spectrum of the first residual signal. Another subtractor subtracts a signal corresponding to the first filter parameters from the first residual signal to provide a second residual signal (r2). Another modeller (RPE) models a component (r2,r3) of the second residual signal with a pulse train coder (RPE) to provide respective pulse train parameters (L0). A bit stream generator (15) generates an encoded audio stream (AS) including the sinusoidal codes (Cs), the first filter parameters (Ps) and the pulse train parameters (L0).
    • 音频编码器被设置为处理音频信号(x)的多个连续段中的每一个的相应的一组采样信号值。 编码器包括分析器(TSA),其被设置为分析采样的信号值以提供与音频信号的相应正弦分量相对应的一个或多个正弦码(Cs)。 减法器从音频信号中减去对应于正弦分量的信号以提供第一残差信号(r1)。 建模器(SEG)通过确定具有近似于第一残差信号的频谱的频率响应的滤波器的第一滤波器参数(Ps)来对第一残差信号(r1)的频谱进行建模。 另一减法器从第一残余信号中减去对应于第一滤波器参数的信号以提供第二残余信号(r2)。 另一个建模器(RPE)用脉冲编码器(RPE)建模第二个剩余信号的分量(r2,r3),以提供相应的脉冲序列参数(L0)。 比特流生成器(15)生成包括正弦码(Cs),第一滤波器参数(Ps)和脉冲串参数(L0)的编码音频流(AS)。
    • 39. 发明授权
    • AUDIO CODING
    • 编码的音频信号
    • EP1514262B1
    • 2006-08-16
    • EP03722975.4
    • 2003-05-16
    • Koninklijke Philips Electronics N.V.
    • DEN BRINKER, Albertus, C.
    • G10L19/04
    • 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 (c 0 . . . 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 (c 0 . . . k) of order K+1 are normalised to provide a set of characteristics (d 1 . . . k) of order K. An encoded audio stream ( 50 ) is generated to include representations (LAR,LSFs) of the normalised set of characteristics (d 1 . . . k) of order K.