公开(公告)号：US07184961B2

公开(公告)日：2007-02-27

申请号：US10333471

申请日：2001-06-15

申请人： Yasushi Sato

发明人： Yasushi Sato

IPC分类号： G10L19/02 ,  G10L21/00 ,  H03M7/00

CPC分类号： G10L19/0208 ,  G10L19/06 ,  H04B1/667 ,  H04B14/048

摘要： A device and a method for compressing signal information by removing (thinning out) the signal component of a signal in a specific frequency band. Firstly, an input time-series signal (e.g., a PCM signal) is converted by an analyzer (11) into a spectrum signal. Next, of the bands obtained by dividing the spectrum equally into bands, the band having a predetermined or higher correlation in the spectrum distribution with the lower frequency band is specified as a harmonic band by a frequency band masking unit (12). Then, a removal band from which the spectrum is to be removed is determined from the harmonic band, and the spectrum signal of this removal band, from which the spectrum component has been removed (namely the frequency component has been thinned out), is fed to a synthesizer (13).

摘要翻译： 通过去除（稀疏）特定频带中的信号的信号分量来压缩信号信息的装置和方法。 首先，输入时间序列信号（例如，PCM信号）由分析器（11）转换为频谱信号。 接下来，通过将频谱等分成频带而获得的频带中，在频带分布中具有较低频带的具有预定或更高相关性的频带被频带掩蔽单元（12）指定为谐波频带。 然后，从谐波频带确定要去除频谱的去除频带，并且馈送频谱分量已被去除的该去除频带的频谱信号（即，频率分量已经被稀疏），被馈送 到合成器（13）。

公开(公告)号：US20050025251A1

公开(公告)日：2005-02-03

申请号：US10660504

申请日：2003-09-12

申请人： Yuh-Chin Chang ,  Hui-Ya Chou ,  Peng-Cheng Chen ,  Hong-Tsan Lee

发明人： Yuh-Chin Chang ,  Hui-Ya Chou ,  Peng-Cheng Chen ,  Hong-Tsan Lee

IPC分类号： H03M7/38 ,  H04B14/04 ,  H04B14/06

CPC分类号： H03M7/3046 ,  H04B14/048 ,  H04B14/066

摘要： A method of optimizing the compression rate in Adaptive Differential Pulse Code Modulation (ADPCM) is disclosed. The modified pulse code modulation technique employs a prognostic code converter to generate variable length codes on top of the ADPCM coding, based on the probability of occurrence of data bits in a data sample. This variable-length coding is able to further reduce the compressed data size by increasing the compression rate of the conventional ADPCM coding.

摘要翻译： 公开了一种在自适应差分脉冲编码调制（ADPCM）中优化压缩率的方法。 基于在数据样本中发生数据位的概率，经修改的脉码调制技术采用预测码转换器在ADPCM编码之上生成可变长度码。 该可变长度编码能够通过增加常规ADPCM编码的压缩率来进一步减小压缩数据大小。

公开(公告)号：US06775297B1

公开(公告)日：2004-08-10

申请号：US09553998

申请日：2000-04-20

申请人： James C. Shelby ,  Thomas D. McKay ,  Robert G. Pecina

发明人： James C. Shelby ,  Thomas D. McKay ,  Robert G. Pecina

IPC分类号： H04J322

CPC分类号： H04B14/048 ,  H03M7/3046 ,  H04B14/068

摘要： An apparatus for transforming digital data into encoded digital data describing analog signals representing the digital data. The apparatus comprises a CPU, memory, and control logic for the data translation. The control logic of the apparatus employs mathematical formulae to produce encoded digital data which describes the characteristics of analog signals representing digital data without converting the digital data into analog signals. The apparatus further performs the function of transforming encoded digital data describing analog signals representing digital data into digital data.

摘要翻译： 一种用于将数字数据变换成描述表示数字数据的模拟信号的编码数字数据的装置。 该装置包括用于数据转换的CPU，存储器和控制逻辑。 该装置的控制逻辑采用数学公式来产生编码的数字数据，其描述表示数字数据的模拟信号的特性，而不将数字数据转换成模拟信号。 该装置还执行将表示数字数据的模拟信号变换为数字数据的编码数字数据变换的功能。

公开(公告)号：US06751260B1

公开(公告)日：2004-06-15

申请号：US09527009

申请日：2000-03-16

申请人： Zhenyu Wang

发明人： Zhenyu Wang

IPC分类号： H04B166

CPC分类号： H04B14/048

摘要： A first transceiver transmits a set of test levels to a second transceiver as a signal through a communication channel as encoded samples and subjected to one or more of a plurality of line encoding algorithms. An information channel is superimposed in the signal transmitted through the communication channel. The second transceiver determines line encoding with, and conversion between, the companding laws present in the communication channel based on the received set of test signals. The set of test levels are signals having levels determined based on the difference between the normalized amplitude, vertex, or energy curves for the types of companding laws, with or without accounting for other sources of network distortion. Encoded samples representing the transmitted test levels are reconstructed by the second transceiver in accordance with the one or more detected line encoding algorithms, the encoded samples for each of the set of test levels packed into a corresponding sample cell. The encoded samples in a sample cell are compared with one another to form a tentative decision of the sample cell for the presence or absence of the superimposed information channel; and each tentative decision for different sample cells are compared to detect the presence or absence of the superimposed information channel.

摘要翻译： 第一收发器将作为经过通信信道的信号的一组测试电平发送到第二收发器作为经编码的采样并经受多个线路编码算法中的一个或多个。 信息信道叠加在通过通信信道发送的信号中。 第二收发机基于所接收的一组测试信号来确定在通信信道中存在的压扩法的线路编码和转换。 测试级别的集合是具有基于压缩法的类型的归一化幅度，顶点或能量曲线之间的差确定的等级的信号，具有或不考虑其他网络失真源。 代表发送的测试级别的编码样本根据一个或多个检测到的线路编码算法由第二收发器重建，所述编码样本针对被包装到对应的采样小区中的每组测试级别。 将样本单元中的编码样本彼此进行比较，以形成样本单元的叠加信息信道的存在或不存在的暂定决定; 并对不同样本单元的每个暂定决定进行比较，以检测叠加信息信道的存在或不存在。

公开(公告)号：US20030167164A1

公开(公告)日：2003-09-04

申请号：US10333471

申请日：2003-01-21

发明人： Yasushi Sato

IPC分类号： G10L019/14 ,  H03M007/00

CPC分类号： G10L19/0208 ,  G10L19/06 ,  H04B1/667 ,  H04B14/048

摘要： A device and a method for compressing signal information by removing (thinning out) the signal component of a signal in a specific frequency band. Firstly, an input time-series signal (e.g., a PCM signal) is converted by an analyzer (11) into a spectrum signal. Next, of the bands obtained by dividing the spectrum equally into bands, the band having a predetermined or higher correlation in the spectrum distribution with the lower frequency band is specified as a harmonic band by a frequency band masking unit (12). Then, a removal band from which the spectrum is to be removed is determined from the harmonic band, and the spectrum signal of this removal band, from which the spectrum component has been removed (namely the frequency component has been thinned out), is fed to a synthesizer (13). Lastly, the synthesizer (13) converts the spectrum signal having the thinned frequency component into a time-series signal. As a result, the compression of the information signal can be highly efficiently realized, and the high quality of an audio signal is retained in the case of a compression ration as high as that of the prior art.

摘要翻译： 通过去除（稀疏）特定频带中的信号的信号分量来压缩信号信息的装置和方法。 首先，输入时间序列信号（例如，PCM信号）由分析器（11）转换为频谱信号。 接下来，通过将频谱等分成频带而获得的频带中，在频带分布中具有较低频带的具有预定或更高相关性的频带被频带掩蔽单元（12）指定为谐波频带。 然后，从谐波频带确定要去除频谱的去除频带，并且馈送频谱分量已被去除的该去除频带的频谱信号（即，频率分量已经被稀疏），被馈送 到合成器（13）。 最后，合成器（13）将具有减薄频率分量的频谱信号转换为时间序列信号。 结果，可以高效地实现信息信号的压缩，并且在压缩比高达现有技术的情况下，保持音频信号的高质量。

公开(公告)号：US20020025004A1

公开(公告)日：2002-02-28

申请号：US09938378

申请日：2001-08-24

申请人： Koninklijke Philips Electronics N.V.

发明人： Ronaldus Maria Aarts

IPC分类号： H04B014/06

CPC分类号： G11B20/10527 ,  H04B14/048

摘要： Method and signal processing apparatus for reducing the number of bits of a digital input signal (Mi) comprising the steps of adding a pseudo-random noise signal (Na) to the digital input signal (Mi) to obtain an intermediate signal (Di), the pseudo-random noise signal (Na) being defined by noise parameters (Np), and quantizing the intermediate signal (Di) having a word length of n bits to a reduced word length signal (Me) having a word length of m bits, n being larger than or equal to m. The method further comprises the step of quantizing the intermediate signal (Di) comprises a first transfer function which is non-linear, the first transfer function being defined by non-linear device parameters (NLDp). Also, the present invention relates to a method and signal decoding apparatus for recovering an output signal (Mo) from a reduced word length signal (Me) provided by the method according to the invention.

摘要翻译： 用于减少数字输入信号（Mi）的位数的方法和信号处理装置包括以下步骤：将伪随机噪声信号（Na）加到数字输入信号（Mi）上以获得中间信号（Di） 所述伪随机噪声信号（Na）由噪声参数（Np）定义，并将具有n位字长的中间信号（Di）量化为具有m位字长的减小字长信号（Me） n大于或等于m。 该方法还包括量化中间信号（Di）的步骤包括非线性的第一传递函数，第一传递函数由非线性设备参数（NLDp）定义。 此外，本发明涉及一种用于从根据本发明的方法提供的缩小字长信号（Me）中恢复输出信号（Mo）的方法和信号解码装置。

公开(公告)号：US5862179A

公开(公告)日：1999-01-19

申请号：US807955

申请日：1997-03-04

申请人： Yuri Goldstein ,  Yuri Okunev

发明人： Yuri Goldstein ,  Yuri Okunev

IPC分类号： H03M7/00 ,  H04B14/02 ,  H04B14/04 ,  H04L25/49 ,  H04L27/00 ,  H04L27/02 ,  H04L27/34

CPC分类号： H04B14/023 ,  H04B14/048 ,  H04L25/4927 ,  H04L27/3411

摘要： A pulse amplitude modulated (PAM) mapper includes a constellation matrix memory which stores indications of a plurality of different constellations, wherein at least one of the different stored constellations is of different dimension than another of the stored constellations. The constellations are used individually or together to support a plurality of different modem data rates. In a preferred embodiment, the mapper also includes a logic block, a constellation controller, a PAM code generation block, and an output register. The logic block receives incoming bits and groups the bits as a function of the desired or agreed upon bit rate as indicated by the constellation controller, and provides a plurality of each group of bits to the PAM code generation block, and one or more sign bits to the output register. The PAM code generation block uses the provided bits to choose at least one point from one of the constellations, and uses each chosen constellation point to generate a PAM code (typically .mu.-law or A-law code level) word. Each PAM code word is provided to the output register, and together with associated sign bits generates output bytes. Algorithms are provided for choosing multiple points from the 2D and higher dimensional constellations from provided groups of bits, and for limiting power by substituting combinations of high-power constellation points with otherwise unused lower-power combinations.

摘要翻译： 脉冲幅度调制（PAM）映射器包括存储多个不同星座的指示的星座矩阵存储器，其中不同存储的星座中的至少一个与所存储的星座中的另一个具有不同的维度。 星座单独或一起用于支持多种不同的调制解调器数据速率。 在优选实施例中，映射器还包括逻辑块，星座控制器，PAM码生成块和输出寄存器。 逻辑块接收输入比特并将比特分组为由星座控制器指示的所需或约定的比特率的函数，并且向PAM码生成块提供多个每组比特，以及一个或多个符号比特 到输出寄存器。 PAM代码生成块使用提供的位从一个星座中选择至少一个点，并且使用每个选择的星座点来生成PAM码（通常为μ-或A-law码级）字。 每个PAM码字被提供给输出寄存器，并且与关联的符号位一起产生输出字节。 提供算法用于从提供的位组的2D和更高维度星座中选择多个点，并且通过用其他未使用的较低功率组合替代高功率星座点的组合来限制功率。

公开(公告)号：US5812075A

公开(公告)日：1998-09-22

申请号：US801973

申请日：1997-02-19

申请人： William Lewis Betts ,  Keith Alan Souders

发明人： William Lewis Betts ,  Keith Alan Souders

IPC分类号： H03M13/00 ,  H04B14/04 ,  H04J3/12 ,  H04L12/00 ,  H04L25/49 ,  H04M11/06

CPC分类号： H04B14/048 ,  H04J3/125

摘要： A compensation system is configured to improve the accuracy of digital signals that are communicated through a digital network by reducing loss from digital attenuation quantization (DAQ; digital pad quantization) and rob bit signaling (RBS). The combined DAQ/RBS compensation system can be employed within a transmitting modem connected to the digital network and is constructed as follows. In a first embodiment, a first adjustment mechanism combines a DAQ compensation quantity with each segment of the digital data, prior to transmission, in order to enhance accuracy of the received digital data. The value of the DAQ compensation quantity depends on feedback that is provided to the compensation system during a series of test transmissions. Next, the word is communicated to a linear-mu-linear converter, which is configured to simulate a digital transmission by mu-law encoding each digital data word into a code word and then subsequently mu-law decoding each code word back into a linear digital data word, while taking into account the compensation quantity during the encoding/decoding process. In addition, the linear-mu-linear converter includes an RBS compensation system that causes an RBS compensation quantity to be mathematically combined with each segment to be tainted by RBS in order to enhance accuracy of the RBS segments, which typically occur periodically. A second adjustment mechanism is connected to the linear-mu-linear converter. The second adjustment mechanism combines the reciprocal of the DAQ compensation quantity with the linear data from the linear-mu-linear converter. Finally, the linear digital data word is passed from the linear-mu-linear converter to a linear-mu converter for conversion into a mu-law code word and transmission to the network. In a second embodiment of the combined DAQ/RBS system, the RBS compensation system is not implemented within, but after, the linear-mu-linear converter.

公开(公告)号：US5237591A

公开(公告)日：1993-08-17

申请号：US747038

申请日：1991-08-19

申请人： Kimerie W. Javitt ,  Kannan Ramchandran

发明人： Kimerie W. Javitt ,  Kannan Ramchandran

IPC分类号： H04B14/04

CPC分类号： H04B14/048

摘要： Disclosed is a circuit for adding loss to a signal in a loop carrier transmission system. As data is being transmitted, the circuit determines the digital output needed for an appropriate amount of loss for each channel unit at the end user interface.

公开(公告)号：US5007086A

公开(公告)日：1991-04-09

申请号：US552767

申请日：1990-07-16

申请人： Kishan Shenoi ,  Helena S. Ho

发明人： Kishan Shenoi ,  Helena S. Ho

IPC分类号： H04B14/04

CPC分类号： H04B14/048

摘要： A method and apparatus for generating low level noise signals are provided. Two random numbers X.sub.1 and Y.sub.1 are first generated and bits 0-4 of the first random number X.sub.1 are extracted to produce a number X.sub.2. A number X.sub.3 is further computed by the formula: X.sub.3 =2.sup.-N (X.sub.2 .multidot.I), where N is a predetermined number and 1.ltoreq.I.gtoreq.2.sup.N -1. Bit 7 of the second random number Y.sub.1 is then extracted and combined as a sign bit with the computed number X.sub.3 to produce a sign-magnitude eight bit number W representing a sample of low level noise encoded in accordance with .mu.-law.

摘要翻译： 提供了一种用于产生低电平噪声信号的方法和装置。 首先生成两个随机数X1和Y1，并且提取第一随机数X1的位0-4产生数X2。 通过以下公式进一步计算X3：X3 = 2-N（X2×I），其中N是预定数，1 / N = 2N-1。 然后提取第二随机数Y1的位7并将其组合为具有计算出的数量X3的符号位，以产生表示根据μ-lo编码的低电平噪声样本的符号幅度八位数W。