Method for determining the type of coding to be selected for coding at
least two signals
    1.
    发明授权
    Method for determining the type of coding to be selected for coding at least two signals 失效
    用于确定要选择用于编码至少两个信号的编码类型的方法

    公开(公告)号:US5736943A

    公开(公告)日:1998-04-07

    申请号:US557046

    申请日:1996-05-31

    摘要: In the case of coding a plurality of signals which are not independent of e another, a selection of the suitable type of coding is made as a function of a similarity measure. According to one aspect of the invention, the similarity measure is determined by firstly coding one of the signals according to the intensity-stereo method and then decoding it in order to create a signal affected by coding error, whereupon the latter signal and the associated non-coded signal are transformed into the frequency domain. In the frequency domain, a selection or evaluation of the actually audible spectral components, as well as of the signal affected by coding error and of the associated signal not affected by coding error, is undertaken using a listening threshold which is determined by a psycho-acoustic calculation. Intensity-stereo coding is undertaken in the case of a high similarity measure, whereas otherwise a separate coding of the channels is performed.

    摘要翻译: PCT No.PCT / EP94 / 02250 Sec。 371日期:1996年5月31日 102(e)日期1996年5月31日PCT提交1994年7月8日PCT公布。 公开号WO95 / 08227 日期1995年3月23日在对不是彼此独立的多个信号进行编码的情况下,作为相似性度量的函数进行适当的编码类型的选择。 根据本发明的一个方面,通过首先根据强度立体声方法对信号之一进行编码,然后对其进行解码来确定相似性度量,以便产生受编码错误影响的信号,由此产生后一信号和相关联的非信号 编码信号被转换成频域。 在频域中,使用由精神病理学家确定的听力阈值来进行实际可听频谱分量的选择或评估,以及受编码错误影响的信号和不受编码错误影响的相关信号, 声学计算。 在高相似性度量的情况下进行强度立体声编码,否则执行信道的单独编码。

    Process and device for the scalable coding of audio signals
    2.
    发明授权
    Process and device for the scalable coding of audio signals 失效
    用于音频信号的可缩放编码的过程和设备

    公开(公告)号:US6115688A

    公开(公告)日:2000-09-05

    申请号:US51347

    申请日:1998-07-01

    CPC分类号: H04B1/665

    摘要: In coding of an audio signal, coded signals with low quality and bit rate on the one hand and coded signals with high quality and bit rate on the other hand are transmitted to a decoder. At first, the audio signal is coded with low bit rate and is transmitted to the decoder before an additional coded signal is transmitted to the decoder, which either alone or together with the first coded signal upon decoding thereof provides a decoded signal with high quality within the decoder. In this manner, a low-quality decoded signal is generated first in the decoder before decoding of the high-quality signal is possible.

    摘要翻译: PCT No.PCT / EP96 / 03609 Sec。 371日期:1998年7月1日 102(e)1998年7月1日PCT PCT 1996年8月16日PCT公布。 公开号WO97 / 14229 日期1997年04月17日在编码音频信号时,一方面具有低质量和比特率的编码信号和另一方面具有高质量和比特率的编码信号被传送到解码器。 首先,以低比特率对音频信号进行编码,并且在将附加的编码信号发送到解码器之前被传送到解码器,解码器单独或与解码时的第一编码信号一起提供高质量的解码信号 解码器。 以这种方式,在对高质量信号进行解码之前,首先在解码器中产生低质量的解码信号。

    Method and device for detecting a transient in a discrete-time audio signal
    3.
    发明授权
    Method and device for detecting a transient in a discrete-time audio signal 有权
    用于检测离散时间音频信号中的瞬变的方法和装置

    公开(公告)号:US06826525B2

    公开(公告)日:2004-11-30

    申请号:US10183139

    申请日:2002-06-25

    IPC分类号: G10L1900

    CPC分类号: H04B1/665

    摘要: A method for detecting a transient in a discrete-time audio signal is performed completely in the time domain and includes the step of segmenting the discrete-time audio signal so as to generate consecutive segments of the same length with unfiltered discrete-time audio signals xs(T−1). The discrete-time audio signal in a current segment is subsequently filtered. Then either the energy of the filtered discrete-time audio signal in the current segment can be compared with the energy of the filtered discrete-time audio signal in a preceding segment or a current relationship between the energy of the filtered discrete-time audio signal in the current segment and the energy of the unfiltered discrete-time audio signal in the current segment can be formed and this current relationship compared with a preceding corresponding relationship. On the basis of the one and/or the other of these comparisons it is detected whether a transient is present in the discrete-time audio signal.

    摘要翻译: 用于检测离散时间音频信号中的瞬态的方法在时域中完全执行,并且包括分段离散时间音频信号以便生成具有未滤波的离散时间音频信号xs的相同长度的连续片段的步骤 (T-1)。 随后过滤当前片段中的离散时间音频信号。 然后可以将当前段中滤波的离散时间音频信号的能量与前一段中滤波的离散时间音频信号的能量或滤波后的离散时间音频信号的能量之间的当前关系进行比较 可以形成当前段的当前段和未过滤离散时间音频信号的能量,并将该当前关系与先前的对应关系进行比较。 基于这些比较中的一个和/或另一个,检测离散时间音频信号中是否存在瞬态。

    Frequency-domain scalable coding without upsampling filters
    4.
    发明授权
    Frequency-domain scalable coding without upsampling filters 有权
    无上采样滤波器的频域可伸缩编码

    公开(公告)号:US06370507B1

    公开(公告)日:2002-04-09

    申请号:US09319066

    申请日:1999-05-28

    IPC分类号: G10L1902

    CPC分类号: G10L19/24 G10L19/0204

    摘要: In a method of coding discrete time signals (X1) sampled with a first sampling rate, second time signals (x2) are generated using the first time signals having a bandwidth corresponding to a second sampling rate, with the second sampling rate being lower than the first sampling rate. The second time signals are coded in accordance with a first coding algorithm. The coded second signals (X2c) are decoded again in order to obtain coded/decoded second time signals (X2cd) having a bandwidth corresponding to the second sampling frequency. The first time signals, by frequency domain transformation, become first spectral values (X1). Second spectral values (X2cd) are generated from the coded/decoded second time signals, the second spectral values being a representation of the coded/decoded time signals in the frequency domain. To obtain weighted spectral values, the first spectral values are weighted by means of the second spectral values, with the first and second spectral values having the same frequency and time resolution. The weighted spectral values (Xb) are coded in accordance with a second coding algorithm in consideration of a psychoacoustic model and written into a bit stream. Weighting the first spectral values and the second spectral values comprises the subtraction of the second spectral values from the first spectral values in to obtain differential spectral values.

    摘要翻译: 在对以第一采样率采样的离散时间信号(X1)进行编码的方法中,使用具有对应于第二采样率的带宽的第一时间信号来生成第二时间信号(x2),其中第二采样率低于 第一次采样率。 第二时间信号根据第一编码算法进行编码。 再次对编码的第二信号(X2c)进行解码,以获得具有对应于第二采样频率的带宽的编码/解码的第二时间信号(X2cd)。 第一次通过频域变换的信号变为第一个光谱值(X1)。 从编码/解码的第二时间信号产生第二频谱值(X2cd),第二频谱值是频域中编码/解码的时间信号的表示。 为了获得加权光谱值,通过第二光谱值对第一光谱值进行加权,其中第一和第二光谱值具有相同的频率和时间分辨率。 考虑到心理声学模型并将其写入比特流,加权频谱值(Xb)根据第二编码算法进行编码。 加权第一光谱值和第二光谱值包括从第一光谱值减去第二光谱值以获得差分光谱值。

    Apparatus and method of coding a mono signal and stereo information
    5.
    发明授权
    Apparatus and method of coding a mono signal and stereo information 有权
    编码单声道信号和立体声信息的装置和方法

    公开(公告)号:US06629078B1

    公开(公告)日:2003-09-30

    申请号:US09445894

    申请日:1999-12-13

    IPC分类号: H04S100

    摘要: A method of coding a time-discrete stereo signal, the stereo signal having a first and a second channel, permits scalable stereo coding. At first, a mono signal is formed from the stereo signal, which is then coded, whereupon the coded mono signal is transmitted to a bit stream. Thereafter, the coded mono singal is decoded again, whereupon stereo information is formed on the basis of the coded/decoded mono signal and the first and second channels, with such stereo information being coded and being also written into the bit stream in order to obtain a bit stream comprising a complete coded monolayer as well as a layer with coded stereo information.

    摘要翻译: 一种编码时分离立体声信号的方法,具有第一和第二声道的立体声信号允许可缩放的立体声编码。 首先,从立体声信号形成单声道信号,然后将其编码,由此编码的单声道信号被发送到比特流。 此后,再次对编码的单声道进行解码,由此基于编码/解码的单声道信号和第一和第二声道形成立体声信息,这些立体声信息被编码并且也被写入比特流,以便获得 包括完整编码单层的比特流以及具有编码的立体声信息的层。

    Method and a device for coding audio signals and a method and a device for decoding a bit stream
    6.
    发明授权
    Method and a device for coding audio signals and a method and a device for decoding a bit stream 有权
    用于编码音频信号的方法和装置以及用于解码比特流的方法和装置

    公开(公告)号:US06502069B1

    公开(公告)日:2002-12-31

    申请号:US09530001

    申请日:2000-04-20

    IPC分类号: G10L1912

    CPC分类号: H04B1/665 H04B14/046

    摘要: The present invention permits a combination of a scalable audio coder with the TNS technique. In a method for coding time signals sampled in a first sampling rate, second time signals are first generated whose sampling rate is smaller than the first sampling rate. The second time signals are then coded according to a first coding algorithm and written into a bit stream. The coded second time signals are, however, decoded again, and, like the first time signals, transformed into the frequency domain. From a spectral representation of the first time signals, TNS prediction coefficients are calculated. The transformed output signal of the coder/decoder with the first coding algorithm, like the spectral representation of the first time signal, undergoes a prediction over the frequency to obtain residual spectral values for both signals, though only the prediction coefficients calculated on the basis of the first time signals are used. These two signals are evaluated against each other. The evaluated residual spectral values are then coded by means of a second coding algorithm to obtain coded evaluated residual spectral values, which, together with the side information containing the calculated prediction coefficients, are written into the bit stream.

    摘要翻译: 本发明允许可扩展音频编码器与TNS技术的组合。 在对以第一采样率采样的时间信号进行编码的方法中,首先生成采样率小于第一采样率的第二时间信号。 然后根据第一编码算法对第二时间信号进行编码并写入比特流。 然而,编码的第二时间信号被再次解码,并且像第一次信号一样被转换成频域。 根据第一时间信号的频谱表示,计算TNS预测系数。 使用第一编码算法的编码器/解码器的变换输出信号,如第一时间信号的频谱表示,对频率进行预测,以获得两个信号的残差频谱值,尽管仅基于 第一次使用信号。 这两个信号被相互评估。 然后通过第二编码算法对所评估的残差频谱值进行编码,以获得编码的估计残差频谱值,其与包含计算的预测系数的边信息一起写入比特流。

    Method and device for detecting a transient in a discrete-time audiosignal
    7.
    发明授权
    Method and device for detecting a transient in a discrete-time audiosignal 有权
    用于检测离散时间音频信号中的瞬态的方法和装置

    公开(公告)号:US06453282B1

    公开(公告)日:2002-09-17

    申请号:US09424596

    申请日:1999-11-24

    IPC分类号: G10L1900

    CPC分类号: H04B1/665

    摘要: A method for detecting a transient in a discrete-time audio signal is performed completely in the time domain and includes the step of segmenting the discrete-time audio signal as to generate consecutive segments of the same length with unfiltered discrete-time audio signals. The discrete-time audio signal in a current segment is filtered. Either the energy of the filtered discrete-time audio signal in the current segment is compared with the energy of the filtered discrete-time audio signal in a preceding segment or a current relationship between the energy of the filtered discrete-time audio signal in the current segment and the energy of the unfiltered discrete-time audio signal in the current segment is formed and this current relationship compared with a preceding corresponding relationship. Whether a transient is present in the discrete-time audio signal is detected using one and/or the other of these comparisons.

    摘要翻译: 用于检测离散时间音频信号中的瞬态的方法在时域中完全执行,并且包括分段离散时间音频信号以生成具有未滤波离散时间音频信号的相同长度的连续片段的步骤。 当前片段中的离散时间音频信号被过滤。 将当前片段中滤波的离散时间音频信号的能量与先前片段中滤波的离散时间音频信号的能量或电流中滤波后的离散时间音频信号的能量之间的当前关系进行比较 形成当前段中未经滤波的离散时间音频信号的能量,并将该当前关系与先前的对应关系进行比较。 使用这些比较中的一个和/或另一个来检测离散时间音频信号中是否存在瞬态。

    Method and device for determining the tonality of an audio signal
    8.
    发明授权
    Method and device for determining the tonality of an audio signal 失效
    用于确定音频信号的音调的方法和装置

    公开(公告)号:US5918203A

    公开(公告)日:1999-06-29

    申请号:US894844

    申请日:1997-08-13

    CPC分类号: H03G5/005

    摘要: The tonality of an audio signal is determined by a method which includes the steps of blockwise frequency transforming a digital input signal x(n) to create a real positive-value representation X(k) of the input signal, where k designates the index of a frequency line, and determining the tonality T of the signal component for the frequency line k according to the following equation: ##EQU1## where F.sub.1 is the filter function of a first digital filter with a first, differentiating characteristic, F.sub.2 is the filter function of a second digital filter with a second, flat or integrating characteristic or with a characteristic which is less strongly differentiating than the first characteristic, and d.sub.1 and d.sub.2 are integer constants which, depending on the filter parameters, are so chosen that the delays of the filters are compensated for in each case.

    摘要翻译: PCT No.PCT / EP96 / 00550 Sec。 371日期1997年8月13日 102(e)日期1997年8月13日PCT 1996年2月9日PCT公布。 WO96 / 25649 PCT出版物 日期1996年8月22日音频信号的音调由包括对数字输入信号x(n)进行块状频率变换以产生输入信号的实数正值表示X(k)的步骤的方法确定,其中 k表示频率线的索引,并且根据以下等式确定频率线k的信号分量的音调T:其中F1是具有第一微分特性的第一数字滤波器的滤波器函数,F2是 第二数字滤波器的滤波器功能具有第二,平坦或积分特性或具有与第一特性不太强区别的特性,d1和d2是整数常数,其根据滤波器参数被选择为使得延迟 在每种情况下补偿滤波器。

    Process for transmitting and/or storing digital signals of multiple
channels
    9.
    发明授权
    Process for transmitting and/or storing digital signals of multiple channels 失效
    用于发送和/或存储多个信道的数字信号的过程

    公开(公告)号:US5706309A

    公开(公告)日:1998-01-06

    申请号:US428235

    申请日:1995-05-02

    摘要: A process for transmitting and/or storing digital signals of multiple chals. This process is suited, in particular, for transmitting the five channels of 3/2 stereophony as well as for transmitting two stereo channels and three additional commentary channels. In this manner, by way of illustration, television programs with multi-language audio signals can be transmitted. This process is distinguished in that by reduction of the to-be-transmitted data, only a bit rate of 384 kbit/s is required for transmission. The reduction of the data is achieved by the K input channels being imaged in segments onto the N.ltoreq.K virtual spectral data channels, by the spectral data channels being quantized, coded, and transmitted taking into consideration the principles of psychoacoustics, and by K output channels being reproduced from the transmitted bit stream with the aid of a transmitted list from the N.ltoreq.K spectral data channels.

    摘要翻译: PCT No.PCT / DE93 / 01047 Sec。 371日期:1995年5月2日 102(e)日期1995年5月2日PCT提交1993年11月2日PCT公布。 公开号WO94 / 10758 日期1994年5月11日用于发送和/或存储多个信道的数字信号的过程。 该过程特别适用于传输3/2立体声的五个通道,以及用于发送两个立体声通道和三个额外的评论通道。 以这种方式,作为说明,可以发送具有多语言音频信号的电视节目。 该过程的区别在于通过减少要发送的数据,传输需要384kbit / s的比特率。 通过被考虑到心理声学原理的量化,编码和传输的频谱数据信道,由K个输入信道以片段成像到N个K个虚拟频谱数据信道上来实现数据的减少,并且通过 借助于来自N个K个频谱数据信道的发送列表,从发送的比特流再现K个输出信道。