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1.发明授权Method and apparatus for recording compressed variable bitrate audio information 有权用于记录压缩的可变比特率音频信息的方法和装置公开(公告)号：US06320825B1
公开(公告)日：2001-11-20
申请号：US09355380
申请日：1999-07-26
IPC分类号： G11B1722
CPC分类号： G11B20/00007 , G11B7/0045 , G11B7/005 , G11B20/10 , G11B27/034 , G11B27/105 , G11B27/329 , G11B2220/2545
摘要： A method for recording audio information on a record carrier, typically a bitstream signal on an optical disc, is disclosed. In the method the audio information is encoded to compressed audio data having a variable bitrate, in particular using lossless compression. The compressed audio data is recorded at a transfer speed which is controlled in dependence on the variable bitrate. A transfer speed profile (53, 54, 55) is determined for items of audio information, e.g. for each track on a CD, in which profile the average transfer speed is higher than an average transfer speed minimally required to transfer the compressed audio data of said item. Stuffing data is added to the compressed audio data according to the transfer speed profile, and transfer speed profile information is recorded on the record carrier for enabling playback at a transfer speed as indicated by the transfer speed profile information. Also a recording device, a record carrier and a playback device are disclosed.
摘要翻译： 公开了一种在记录载体上记录音频信息的方法，通常是光盘上的比特流信号。 在该方法中，音频信息被编码为具有可变比特率的压缩音频数据，特别是使用无损压缩。 以按照可变比特率进行控制的传送速度记录压缩音频数据。 确定传送速度分布（53,54,55）用于音频信息的项目，例如， 对于CD上的每个轨道，其中平均传送速度高于传送所述项目的压缩音频数据所需的平均传送速度。 根据传送速度分布将填充数据添加到压缩音频数据，并且传送速度分布信息被记录在记录载体上，以便以传送速度分布信息所指示的传送速度进行播放。 还公开了一种记录装置，记录载体和播放装置。
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公开(公告)号：US5687282A
公开(公告)日：1997-11-11
申请号：US585009
申请日：1996-01-11
CPC分类号： H04B1/665
摘要： A method and apparatus is disclosed for determining a masked threshold curve as a function of frequency from an information signal. The method includes (a) a first step of determining for each frequency value a first masking component MC(1,k) at the frequency value f(k) resulting from a frequency component at the frequency value and having a magnitude equal to the magnitude value PV(k) of the magnitude spectrum at the frequency value, (b) a second step of determining for subsequent frequency values when going in one direction through the frequency range of interest a second masking component MC(2,k) at a frequency value, the second masking component for the frequency value being determined from the first masking component MC(1,k-1) and a second masking component MC(2,k-1) at only the frequency value directly preceding the frequency value f(k-1), (c) a third step of determining for subsequent frequency values when going in the reverse direction through the frequency range of interest a third masking component MC(3,k) at a frequency value, the third masking component for the frequency value being determined from masking information that has a relation to at least the first masking component MC(1,k+1) and the third masking component MC(3,k+1) at only the frequency value f(k+1) directly preceding the frequency value, and (d) a fourth step for determining a masking value MV(k) for a frequency value in the masked threshold curve from masking information that has a relation to the first, second and third masking components (if present) for the frequency value.
摘要翻译： 公开了一种用于根据信息信号确定作为频率的函数的掩蔽阈值曲线的方法和装置。 该方法包括：（a）第一步骤，确定每个频率值的频率值f（k）的第一掩蔽分量MC（1，k），该频率值由频率值的频率分量产生，并且具有等于幅度的幅度 在频率值处的幅度谱的值PV（k），（b）当沿着感兴趣的频率范围沿一个方向进行频率的第二掩蔽分量MC（2，k）时，确定随后的频率值的第二步骤 值，仅在频率值f前面的频率值从第一掩蔽分量MC（1，k-1）和第二掩蔽分量MC（2，k-1）确定频率值的第二掩蔽分量 k-1），（c）第三步骤，当频率值为第三掩蔽组件MC（3，k）的频率范围内的相反方向进行后续频率值时，确定第三掩蔽组件， 频率值由掩码确定 至少与第一掩蔽分量MC（1，k + 1）和第三掩蔽分量MC（3，k + 1）有关的信息仅在频率值之前的频率值f（k + 1） ，以及（d）第四步骤，用于根据对于频率值与第一，第二和第三掩蔽分量（如果存在）关系的掩蔽信息确定掩蔽阈值曲线中的频率值的掩蔽值MV（k） 。
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公开(公告)号：US5661755A
公开(公告)日：1997-08-26
申请号：US546436
申请日：1995-10-20
CPC分类号： H04B1/665
摘要： An encoder apparatus is disclosed for encoding a wideband digital information signal. The apparatus comprises an input signal (1) for receiving the wideband digital information signal, a signal splitting unit (2) for splitting the wideband digital information signal into M narrow band sub signals (SB.sub.1 to SB.sub.M). The narrow bands all have a specific constant bandwidth. Further, a scale factor determining unit (6) for determining a scale factor for subsequent signal blocks in each of the sub signals, and a quantization unit (13) for quantizing the samples in a signal block into quantized samples are present. A bit allocation information deriving unit (34,41,48) is present for deriving bit allocation information, the bit allocation information being representative of the number of bits with which samples in a signal block of a sub signal will be represented after quantization in the quantization unit (13). A formatting unit (20) is present for combining the quantized samples in the signal blocks of the quantized sub signals and the scale factors into a digital output signal having a format suitable for transmission or storage. The apparatus further comprises a signal block length determining unit (30) for determining the lengths of the signal block in at least one of the sub signals and for generating block length information, the block length information being representative of the said lengths of the signal blocks of the said at least one sub signal, where the lengths of subsequent signal blocks in said at least one sub signal differ. The scale factor determining unit (6) now determines the scale factors for subsequent signal blocks of varying lengths in response to said block length information, the bit allocation information deriving unit (34,41,48) now derives bit allocation information for subsequent signal blocks of varying lengths in response to said block length information, and the quantization unit (13) now quantize the samples in signal blocks of varying lengths in response to said block length information. The formatting unit (20) further includes the block length information into the digital output signal for transmission or storage.
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4.发明授权Audio signal processing arrangement for deriving a centre channel signal and also an audio visual reproduction system comprising such a processing arrangement 失效用于导出中心声道信号的音频信号处理装置以及包括这种处理装置的音频视频再现系统
公开(公告)号：US5528694A
公开(公告)日：1996-06-18
申请号：US186396
申请日：1994-01-25
CPC分类号： H04S5/00 , H04S2400/05
摘要： An audio signal processing arrangement includes a first filter (20) for splitting off signal components from the left channel signal (L) at least within one frequency band. Signal components are split off from the right channel signal (R) by a second filter (23). The output signals of the filters (20 and 23) are compared with the right channel signal (R) and the left channel signal (L), respectively. The filter parameters of the filters (20 and 23) are adjusted to values at which there is maximum correlation between the compared signals according to a given criterion. The center channel signal is derived in dependence on the filter adjustment. This can be effected by combining the output signals of the filters (20 and 23). In this manner, a center channel signal is obtained formed by the correlating left and right channel signal components, so that the stereo image is hardly disturbed by the addition of the center channel signal, whereas the perceived position of the virtual sources in the stereo image becomes less dependent on the listener's position with respect to the left and right loudspeakers.
摘要翻译： 音频信号处理装置包括：至少在一个频带内分离来自左声道信号（L）的信号分量的第一滤波器（20）。 信号分量通过第二滤波器（23）从右声道信号（R）分离。 滤波器（20和23）的输出信号分别与右声道信号（R）和左声道信号（L）进行比较。 滤波器（20和23）的滤波器参数根据给定的标准被调整为在比较的信号之间存在最大相关性的值。 中心通道信号依赖于滤波器调整而导出。 这可以通过组合滤波器（20和23）的输出信号来实现。 以这种方式，通过相关的左和右声道信号分量获得形成的中心声道信号，使得立体图像几乎不被中心声道信号的增加所扰乱，而立体图像中虚拟源的感知位置 听众对左右扬声器的位置变得较少依赖。
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5.发明授权7-channel transmission, compatible with 5-channel transmission and 2-channel transmission 失效7通道传输，兼容5通道传输和2通道传输
公开(公告)号：US5850456A
公开(公告)日：1998-12-15
申请号：US795820
申请日：1997-02-06
CPC分类号： H04H20/88 , G10L19/0208 , H04B1/667
摘要： A 7-channel encoder and corresponding decoder is disclosed for encoding a 7-channel signal into a transmission signal which is backwards compatible, so that prior art MPEG-1 2-channel decoders are capable of decoding the transmission signal into a compatible stereo signal, prior art MPEG-2 5-channel decoders are capable of decoding the transmission signal into a compatible 5-channel signal. Further, a 7-channel decoder is disclosed for decoding the transmission signal into a 7-channel signal. (FIG. 2).
摘要翻译： 公开了一种用于将7信道信号编码为向后兼容的传输信号的7信道编码器和相应的解码器，使得现有技术的MPEG-1 2信道解码器能够将传输信号解码为兼容的立体声信号， 现有技术的MPEG-2 5信道解码器能够将传输信号解码为兼容的5信道信号。 此外，公开了用于将传输信号解码为7信道信号的7信道解码器。 （图2）。
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公开(公告)号：US5850418A
公开(公告)日：1998-12-15
申请号：US431867
申请日：1995-05-01
CPC分类号： H04B1/667
摘要： An encoding system and an encoding method for encoding a digital signal having at least a first and a second digital signal component. The encoding system includes a splitter unit for dividing the bandwidth of the digital signal components into M successive frequency bands, and generating in response to the digital signal components M sub signals (SB.sub.m1,SB.sub.mr) for each digital signal component, each sub signal of a signal component being associated with one of the frequency bands (m); a bit need determining unit for determining bit needs for time equivalent signal blocks of the sub signals, a signal combination unit for combining, in a number of at least one frequency bands, time equivalent signal blocks of corresponding sub signals of the at least first and second signal component so as to obtain a time equivalent signal block of a composite sub signal in each the at least one frequency bands; quantizing unit for quantizing time equivalent signal blocks of the sub signals in those frequency bands in which no composite sub signal is available and for quantizing the corresponding time equivalent signal blocks of the composite sub signal in the at least one frequency bands in which a composite sub signal is available. Further, a bit allocation unit is available for deriving allocation information (n.sub.mi,n.sub.jc) from bit needs obtained in the bit need determining unit and from a value B, where B corresponds to a number of bits in an available bitpool. The bit need determining unit is adapted to determine a common bit need b.sub.mc for a time equivalent signal block of a composite sub signal SB.sub.mc in a frequency band m from the bit needs b.sub.mi of the time equivalent signal blocks of the corresponding sub signals of the at least two signal components in that frequency band from which the time equivalent signal block of the composite sub signal has been derived, by taking the common bit need b.sub.mc equal to a weighted sum of the bit needs b.sub.mi. The encoding system can be a subband encoding system or a transform encoding system.
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7.发明授权Coder for incorporating an auxiliary information signal in a digital audio signal, decoder for recovering such signals from the combined signal, and record carrier having such combined signal recorded thereon 失效用于在数字音频信号中加入辅助信息信号的编码器，用于从组合信号中恢复这些信号的解码器以及具有记录在其中的这种组合信号的记录载体
公开(公告)号：US5161210A
公开(公告)日：1992-11-03
申请号：US433631
申请日：1989-11-08
CPC分类号： G11B20/00891 , G10L19/018 , G11B20/00086 , G11B20/00884 , G11B20/10527 , H04B1/665 , H04B1/667 , G06F2211/007
摘要： The audio signal is divided into frequency sub-bands, the sub-band samples are quantized according to a predetermined quantizing criterion, and the quantized samples in one or more sub-bands are summed with samples of the auxiliary signal in such sub-bands, the auxiliary signal sample in a sub-band having a maximum amplitude less than half the quantization step used in such sub-band. The combined signal, covering the entire frequency band of the audio signal, can be transmitted or recorded on a record carrier. Upon audio reproduction at a receiver only the audio signal will be audible, the auxiliary signal being masked due to the psychoacoustic characteristics of the human auditory system which are unresponsive to low-level noise in the presence of high amplitude sound. The receiver includes a decoder which analyzes the combined signal into the original frequency sub-bands and re-quantizes the sub-band signals using the same quantizing criterion as at the coder. The auxiliary signal is reconstructed by subtracting the quantized signal sample in each sub-band from the non-quantized signal sample therein and combining the non-quantized sub-band signal sample. The auxiliary signal may be a copy inhibit code which serves to inhibit a recorder from recording the audio signal, thus protecting against unauthorized copying.
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