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1. 发明授权

US08311816B2 Noise shaping for predictive audio coding apparatus 有权
标题翻译：预测音频编码装置的噪声整形
公开(公告)号：US08311816B2
公开(公告)日：2012-11-13
申请号：US12639676
申请日：2009-12-16
申请人： Yasuhiro Toguri , Jun Matsumoto
发明人： Yasuhiro Toguri , Jun Matsumoto
IPC分类号： G10L21/02 , G10L19/00
CPC分类号： H04B14/068 , H03M7/3046
摘要： An information coding apparatus includes a predictive signal generator that generates a predictive signal; a predictive residual signal generator that generates a predictive residual signal; a quantizer that quantizes a quantization input signal generated based on the predictive residual signal; a quantization error signal generator that generates a quantization error signal; a feedback signal generator that generates a feedback signal for controlling the frequency characteristic of the quantization noise after decoding based on the quantization error signal; and a quantization input signal generator that generates the quantization input signal. The feedback signal generator is configured by a pole-zero filter that includes a filter coefficient of an all-pole filter which is based on spectral envelope information estimated by the input audio signal, a parameter for adjusting a peak level in the frequency characteristic of the quantization noise caused by the all-pole filter, and the predictive filter coefficient.
摘要翻译：一种信息编码装置，包括产生预测信号的预测信号发生器; 产生预测残差信号的预测残差信号发生器; 量化器，其量化基于所述预测残差信号产生的量化输入信号; 产生量化误差信号的量化误差信号发生器; 反馈信号发生器，其基于量化误差信号生成用于控制解码之后的量化噪声的频率特性的反馈信号; 以及产生量化输入信号的量化输入信号发生器。反馈信号发生器由极零滤波器构成，该极零滤波器包括基于由输入音频信号估计的频谱包络信息的全极滤波器的滤波器系数，用于调整频率特性中的峰值电平的参数由全极滤波器引起的量化噪声和预测滤波器系数。

2. 发明授权

US08300711B2 Adaptive differential pulse code modulation/demodulation system and method 有权
标题翻译：自适应差分脉码调制/解调系统及方法
公开(公告)号：US08300711B2
公开(公告)日：2012-10-30
申请号：US12690029
申请日：2010-01-19
申请人： Chu-Feng Lien
发明人： Chu-Feng Lien
IPC分类号： H04B14/06
CPC分类号： H04L25/085 , H03M3/042 , H04B14/068
摘要： An adaptive differential pulse-code modulation-demodulation system and method thereof is provided. The method includes steps of modulating an analog audio input signal into a data packet, including a plurality of digital data through adaptive differential pulse-code modulation, an initial value and a scale factor associated with the digital data, to be sent to the communication network, and demodulating the data packet according to the digital data, the initial value and the scale factor, thereby reconstructing the data packet to an analog audio output signal.
摘要翻译：提供了一种自适应差分脉码调制解调系统及其方法。该方法包括以下步骤：将模拟音频输入信号调制成数据分组，包括通过自适应差分脉冲编码调制的多个数字数据，与数字数据相关联的初始值和比例因子发送到通信网络，并且根据数字数据，初始值和比例因子对数据分组进行解调，从而将数据分组重建为模拟音频输出信号。

3. 发明申请

US20100183092A1 ADAPTIVE DIFFERENTIAL PULSE CODE MODULATION/DEMODULATION SYSTEM AND METHOD 有权
标题翻译：自适应差分脉冲代码调制/解调系统及方法
公开(公告)号：US20100183092A1
公开(公告)日：2010-07-22
申请号：US12690029
申请日：2010-01-19
申请人： Chu-Feng Lien
发明人： Chu-Feng Lien
IPC分类号： H04L27/04 , H04J3/24
CPC分类号： H04L25/085 , H03M3/042 , H04B14/068
摘要： An adaptive differential pulse-code modulation-demodulation system and method thereof is provided. The method includes steps of modulating an analog audio input signal into a data packet, including a plurality of digital data through adaptive differential pulse-code modulation, an initial value and a scale factor associated with the digital data, to be sent to the communication network, and demodulating the data packet according to the digital data, the initial value and the scale factor, thereby reconstructing the data packet to an analog audio output signal.
摘要翻译：提供了一种自适应差分脉码调制解调系统及其方法。该方法包括以下步骤：将模拟音频输入信号调制成数据分组，包括通过自适应差分脉冲编码调制的多个数字数据，与数字数据相关联的初始值和比例因子发送到通信网络，并且根据数字数据，初始值和比例因子对数据分组进行解调，从而将数据分组重建为模拟音频输出信号。

4. 发明授权

US5839100A Lossless and loss-limited compression of sampled data signals 失效
标题翻译：采样数据信号的无损和有限压缩
公开(公告)号：US5839100A
公开(公告)日：1998-11-17
申请号：US636019
申请日：1996-04-22
申请人： Albert William Wegener
发明人： Albert William Wegener
IPC分类号： H04B14/06 , G10L3/02 , G10L5/02 , G10L9/00
CPC分类号： H04B14/068
摘要： An efficient method for compressing audio and other sampled data signals without loss, or with a controlled amount of loss, is described. The compression apparatus contains a subset selector, an approximator, an adder, two derivative encoders, a header encoder, and a compressed block formatter. The decompression apparatus contains a compressed block parser, a header decoder, two integration decoders, an approximator, and an adder. The compressor first divides each block of input samples into a first subset and a second subset. The approximator uses the first subset samples to approximate the second subset samples. An error signal is created by subtracting the approximated second subset samples from the actual second subset samples. The first subset samples and error signal are separately encoded by the derivative encoders, which select the signal's derivative that requires the least amount of storage for a block floating point representation. A compressed block formatter combines the compression control parameters, encoded subset array, and encoded error array into a compressed block. The decompression apparatus first parses the compressed block into a header, an encoded first subset array, and an encoded error array. The header decoder recovers the compression control parameters from the header. Using the compression control parameters, the integration decoders reconstruct the first subset and error arrays from their block floating point representations. The approximator uses the first subset samples to approximate the original second subset samples. The adder combines the subset samples, the error samples, and the approximated second subset samples to identically re-create the original, uncompressed signal. An indexing method is described which allows random access to specific uncompressed samples within the stream of compressed blocks.
摘要翻译：描述了一种用于压缩音频和其他采样数据信号而无损失或受控量损失的有效方法。压缩装置包含子集选择器，近似器，加法器，两个导数编码器，头编码器和压缩块格式化器。解压缩装置包含压缩块解析器，报头解码器，两个积分解码器，近似器和加法器。压缩器首先将每个输入样本块划分成第一子集和第二子集。近似器使用第一子集样本近似第二子集样本。通过从实际的第二子集样本中减去近似的第二子集样本来产生误差信号。第一子集采样和误差信号由导数编码器单独编码，导数编码器选择对块浮点表示需要最少存储量的信号导数。压缩块格式化器将压缩控制参数，编码子集阵列和编码错误阵列组合成压缩块。解压缩装置首先将压缩块解析成报头，编码的第一子集阵列和编码的误差阵列。报头解码器从报头中恢复压缩控制参数。使用压缩控制参数，积分解码器从其块浮点表示重建第一个子集和错误数组。近似器使用第一子集样本近似原始的第二子集样本。加法器组合了子集样本，误差样本和近似的第二子集样本，以相同地重新创建原始的未压缩信号。描述了一种索引方法，其允许随机访问压缩块流内的特定未压缩样本。

5. 发明授权

US5687189A Method of noise reduction for an ADPCM signal 失效
标题翻译： ADPCM信号降噪方法
公开(公告)号：US5687189A
公开(公告)日：1997-11-11
申请号：US308185
申请日：1994-09-19
申请人： Sybren D. Smith , Paul D. Marko , Craig P. Wadin
发明人： Sybren D. Smith , Paul D. Marko , Craig P. Wadin
IPC分类号： H04B14/06 , H04L1/00
CPC分类号： H04L1/0059 , H04B14/068
摘要： Noise in an ADPCM signal is muted or suppressed by modifying the magnitude of the ADPCM information during noisy conditions. In one method, the ADPCM data is changed to a predetermined magnitude value. In another method, the magnitude is reduced by a predetermined amount.
摘要翻译：通过在噪声条件下修改ADPCM信息的大小，ADPCM信号中的噪声被抑制或抑制。在一种方法中，ADPCM数据被改变为预定的幅度值。在另一种方法中，幅度减小预定量。

6. 发明授权

US5550837A Adaptive differential pulse code modulation system with transmission error compensation 失效
标题翻译：具有传输误差补偿的自适应差分脉码调制系统
公开(公告)号：US5550837A
公开(公告)日：1996-08-27
申请号：US248143
申请日：1994-05-24
申请人： Hyokang Chang
发明人： Hyokang Chang
IPC分类号： H04B14/06 , H04L1/24 , G06F11/00
CPC分类号： H04L1/24 , H04B14/068
摘要： A codeword forcing technique is applied which provides gradual degradation of an audio output during fading such that analog degradation in a fading signal environment is simulated. The system may be used in conjunction with digital cordless telephones where gradual degradation of a 32 Kbit/s ADPCM signal is obtained by indirectly detecting a transmission bit error rate, The bit error rate detection is accomplished by sensing the occurrence of an invalid all-zero code, Codeword forcing techniques are then applied to compensate for the inaccurate transmission and associated noise level,
摘要翻译：应用码字强制技术，其在衰落期间提供音频输出的逐渐劣化，从而模拟衰落信号环境中的模拟劣化。该系统可以与数字无绳电话一起使用，其中通过间接检测传输误码率获得32Kbit / s ADPCM信号的逐渐劣化。误码率检测通过感测无效全零的发生来实现代码，然后应用码字强制技术来补偿不准确的传输和相关的噪声水平，

7. 发明授权

US5317522A Method and apparatus for noise burst detection in a signal processor 失效
标题翻译：用于信号处理器中噪声突发检测的方法和装置
公开(公告)号：US5317522A
公开(公告)日：1994-05-31
申请号：US088944
申请日：1993-07-12
申请人： Luis A. Bonet , Carlos A. Greaves , Jose G. Corleto
发明人： Luis A. Bonet , Carlos A. Greaves , Jose G. Corleto
IPC分类号： H03M7/38 , H04B1/10 , H04B14/06 , H04M1/725
CPC分类号： H04B1/1027 , H04B14/068 , H04M1/72502 , H04M1/72505
摘要： A signal processor such as an ADPCM decoder (128b) receives an input signal. As part of the CCITT Recommendation G.726 algorithm, an inverse adaptive quantizer (41) processes the input signal to provide a quantized difference signal d.sub.q (k). When enabled, a noise detector (50) samples signal d.sub.q (k) once for each of a predetermined number of received samples. The noise detector (50) adds the absolute value of signal d.sub.q (k) to a total energy estimate. At the end of the predetermined number of samples, the noise detector (50) compares the total energy estimate to a product of a noise threshold and the predetermined number. If the total energy estimate exceeds this product, then a noise indication is provided. In another embodiment (228b) a noise detector (250) compares an existing energy estimate signal d.sub.ml (k) computed by an adaptation speed control block (48) as part of the G.726 algorithm to an energy threshold to save processing time.
摘要翻译：诸如ADPCM解码器（128b）的信号处理器接收输入信号。作为CCITT建议G.726算法的一部分，逆自适应量化器（41）处理输入信号以提供量化差分信号dq（k）。当使能时，噪声检测器（50）对于预定数量的接收样本中的每一个对信号dq（k）进行一次采样。噪声检测器（50）将信号dq（k）的绝对值与总能量估计相加。在预定数量的样本结束时，噪声检测器（50）将总能量估计与噪声阈值和预定数量的乘积进行比较。如果总能量估计超过该产品，则提供噪声指示。在另一实施例（228b）中，噪声检测器（250）将作为G.726算法的一部分的自适应速度控制块（48）计算出的现有能量估计信号dml（k）与节能处理时间进行比较。

8. 发明授权

US4768203A Signal-to-noise ratio testing in adaptive differential pulse code modulation 失效
标题翻译：自适应差分脉冲编码调制中的信噪比测试
公开(公告)号：US4768203A
公开(公告)日：1988-08-30
申请号：US98296
申请日：1987-09-18
申请人： James F. Ingle
发明人： James F. Ingle
IPC分类号： H04B14/06 , H04B3/46
CPC分类号： H04B14/068
摘要： The adaptive and predictive capabilities of Adaptive Differential Pulse Code Modulation (ADPCM) equipment enable a telecommunication system to maintain acceptable signal/noise levels in voice transmission while utilizing a significantly lower encoding bit rate than that of conventional Pulse Code Modulation (PCM). ADPCM, however, has a deleterious effect on high-speed voiceband data transmission, yet due to its adaptive capabilities cannot readily be identified or evaluated by means of conventional ANSI/IEEE standard test signals and methods. The procedure of the present invention enables such identification and evaluation by imposing upon an ADPCM system a multiple-tone test signal which spans the voiceband and has amplitude characteristics similar to white noise. This signal thereby effectively overloads the adaptive and predictive capabilities of the system and causes the generation of a notably high level of quantizing noise. The resulting multitone signal-with-noise output from the system is processed in a spectrum analyzer where the accumulation of the signal levels in the distinct and narrow input tone bands is compared with the remainder of the accumulated signal power to obtain an accurate signal/noise measurement which, in addition to providing substantive analytical data, yields an indication of the presence of ADPCM, as distinguished even from tandem PCM, encoding equipment in the system.
摘要翻译：自适应差分脉码调制（ADPCM）设备的自适应和预测能力使电信系统能够在语音传输中保持可接受的信号/噪声电平，同时利用比常规脉码调制（PCM）低得多的编码比特率。然而，ADPCM对高速语音带数据传输具有有害影响，但由于其自适应能力不能通过传统的ANSI / IEEE标准测试信号和方法容易地识别或评估。本发明的过程通过对ADPCM系统施加跨越语音带并且具有类似于白噪声的振幅特性的多音测试信号来实现这种识别和评估。因此，该信号有效地使系统的自适应和预测能力过载，并导致产生显着高水平的量化噪声。在频谱分析仪中对来自系统的所得到的多音频信号噪声输出进行处理，其中将不同和窄的输入音频带中的信号电平的累积与累积信号功率的剩余部分进行比较以获得精确的信号/噪声除了提供实质的分析数据之外，测量还产生ADPCM的存在的指示，即使是来自串联PCM，系统中的编码设备也是如此。

9. 发明授权

US4569058A Transmission system 失效
标题翻译：传输系统
公开(公告)号：US4569058A
公开(公告)日：1986-02-04
申请号：US596659
申请日：1984-04-04
申请人： Hans-Joachim Grallert
发明人： Hans-Joachim Grallert
IPC分类号： H04B14/06 , H04N7/32 , H04N19/124 , H04N19/50 , H04N7/13
CPC分类号： H04B14/068 , H04N19/124 , H04N19/50
摘要： A transmission system wherein PCM code words are converted into DPCM code words at the transmit side and are reconverted into PCM code words at the receive side. A DPCM coder comprising a respective quantizer for variable-length code words, a quantizer for code words having a fixed, below-average length, and a DPCM decoder comprising a code converter for variable-length code words and a code converter for code words having a fixed, below-average length are provided for this purpose. The type of conversion for the next PCM code word present at the input is selected after each conversion on the basis of the number of memory locations that still exist. This transmission system is suitable for data reduction, particularly of digital video signals.
摘要翻译：一种传输系统，其中PCM码字在发送侧被转换成DPCM码字，并在接收侧被重新转换成PCM码字。 DPCM编码器，其包括用于可变长度码字的相应量化器，用于具有固定的，低于平均长度的码字的量化器和包括用于可变长度码字的码转换器的DPCM解码器和用于具有为此目的提供固定的低于平均长度的长度。存在于输入端的下一个PCM码字的转换类型是根据仍然存在的存储器位置的数量在每个转换之后被选择的。该传输系统适用于特别是数字视频信号的数据简化。

10. 发明授权

US4360713A Echo detector particularly for speech interpolation communication systems 失效
标题翻译：回波检波器特别适用于语音插值通信系统
公开(公告)号：US4360713A
公开(公告)日：1982-11-23
申请号：US128384
申请日：1980-03-10
申请人： Soumagne Joel
发明人： Soumagne Joel
IPC分类号： H04B3/20 , H04B14/06 , H04J3/17
CPC分类号： H04B14/068 , H04B3/20
摘要： An echo detector which is able to distinguish a speech signal echo on the outgoing channel without reference to the signal received at the incoming channel. The echo detector is comprised of a circuit for digital differential coding in which the error signal between the incoming signal and the predicted signal is divided by the average value of a certain number of samples of previously received incoming signals, having sign according to the second-last received sample, the quotient obtained in this first division being again divided by the average value of a certain number of such quotients previously computed, the logarithm to base 2 of the average value for the entering signal samples and the logarithm to base 2 of the average value of the quotients, for which only the integer portion is kept, forming for each treated sample two values defining one point of a matrix of points, a first counter counting for a number of sets of two values for which the corresponding points are above or on the secondary diagonal of the matrix, a second counter counting simultaneously the number of sets of values for which the corresponding points are below or on the above-mentioned secondary diagonal, a divider carrying out the quotient of the counts from the first and the second counter, this quotient being compared to a predetermined value to decide, if it is greater than this value, that the entering signal is an echo and not an information signal.
摘要翻译：回波检测器，其能够区分输出信道上的语音信号回波，而不参考在输入信道处接收到的信号。回波检测器由用于数字差分编码的电路组成，其中输入信号和预测信号之间的误差信号除以先前接收到的输入信号的一定数量样本的平均值，最后接收的样本，将在该第一分割中获得的商再次除以先前计算的一定数量的这样的商的平均值，对于进入信号样本的平均值的基数2和对于输入信号样本的基数2的对数对于每个处理的样本，仅形成整数部分的商的平均值，形成定义点矩阵的一个点的两个值，对应于相应点的上述两个值的集合的数量的第一个计数器或在矩阵的次对角线上，第二计数器同时计数对应点低于o的值的集合的数量 r在上述次对角线上，分频器执行来自第一和第二计数器的计数商，该商与预定值进行比较，以确定如果大于该值，则输入信号为回声而不是信息信号。

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