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公开(公告)号：US20100329470A1
公开(公告)日：2010-12-30
申请号：US12823616
申请日：2010-06-25
IPC分类号： H04R29/00
CPC分类号： G10L19/025
摘要： An audio information processing apparatus and method include dividing an audio signal, determining a time period having a power change ratio of an audio signal larger than a first threshold value as an attack candidate, searching the time period of the attack candidate and a time period immediately before the time period of the attack candidate for an attack starting point, correcting a power of an audio signal included in the time period, and determining whether a power change ratio of the audio signal included in the time period is larger than a second threshold value for attack detection which is larger than the first threshold value.
摘要翻译： 音频信息处理装置和方法包括：分割音频信号，确定具有大于第一阈值的音频信号的功率变化率的时间段作为攻击候选者，在攻击候选者的时间周期和紧急时刻 在攻击起始点的攻击候选者的时间段之前，校正包括在该时间段内的音频信号的功率，以及确定该时间段中包括的音频信号的功率变化率是否大于第二阈值 用于大于第一阈值的攻击检测。
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公开(公告)号：US07787500B2
公开(公告)日：2010-08-31
申请号：US10927987
申请日：2004-08-27
CPC分类号： H04L47/10 , H04L47/22 , H04L47/50 , H04L47/564 , H04L65/80
摘要： In a packet receiving method and device which convert a voice packet received into a voice, a receiving packet buffer temporarily stores a voice packet received; a plurality of parameter information monitors respectively determine different buffer adjustment values for determining a buffering amount of the receiving packet buffer based on one or more pieces of parameter information obtained from the voice packet temporarily stored; a buffer adjustment value determiner determines a receiving buffer adjustment value from the plural buffer adjustment values; and a buffer controller controls the buffering amount based on the receiving buffer adjustment value.
摘要翻译： 在将接收到的语音分组转换为语音的分组接收方法和装置中，接收分组缓冲器临时存储接收的语音分组; 多个参数信息监视器分别基于从临时存储的语音分组获得的一个或多个参数信息来确定用于确定接收分组缓冲器的缓冲量的不同缓冲器调整值; 缓冲器调整值确定器从多个缓冲器调整值确定接收缓冲器调整值; 并且缓冲器控制器基于接收缓冲器调整值来控制缓冲量。
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公开(公告)号：US07734053B2
公开(公告)日：2010-06-08
申请号：US11390054
申请日：2006-03-27
CPC分类号： G10L19/0204 , G10L19/002 , G10L19/008
摘要： An encoding apparatus compresses a stereo signal using a sum signal and a difference signal of a left component signal and a right component signal of the stereo signal. The encoding apparatus includes a calculating unit that calculates complexity of the sum signal and complexity of the difference signal; a setting unit that sets, based on the complexity, an allocation rate of bits to be allocated in quantizing the sum signal and the difference signal; and a quantizing unit that quantizes the sum signal and the difference signal based on the allocation rate.
摘要翻译： 编码装置使用和信号和立体声信号的左分量信号和右分量信号的差分信号来压缩立体声信号。 编码装置包括计算和信号的复杂度和差分信号的复杂度的计算单元; 设置单元，其基于复杂度设置在量化所述和信号和所述差信号中要分配的比特的分配速率; 以及量化单元，其基于分配率量化和信号和差分信号。
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公开(公告)号：US20090070120A1
公开(公告)日：2009-03-12
申请号：US12232096
申请日：2008-09-10
IPC分类号： G10L19/00
CPC分类号： G10L19/0204
摘要： According to an aspect of an embodiment, a method for regenerating an audio signal including a low frequency component and a high frequency component by decoding a coded data including a first coded data and a second coded data, the method comprising the steps of: generating the low frequency component; generating the high frequency component; determining whether the low frequency component has transient characteristics or not; generating a low frequency correction component by removing a stationary component when the audio signal has the transient characteristics; generating a corrected high frequency component by correcting the high-frequency component on the basis of the duration of the low frequency correction component when the audio signal has the transient characteristics; and regenerating the audio signal by synthesizing the low frequency component with the corrected high-frequency component.
摘要翻译： 根据实施例的一个方面，一种用于通过对包括第一编码数据和第二编码数据的编码数据进行解码来再生包括低频分量和高频分量的音频信号的方法，所述方法包括以下步骤： 低频分量; 产生高频分量; 确定低频分量是否具有瞬态特性; 当音频信号具有瞬态特性时通过去除固定分量来产生低频校正分量; 当音频信号具有瞬态特性时，通过基于低频校正分量的持续时间校正高频分量来产生校正的高频分量; 以及通过利用所述校正的高频分量合成所述低频分量来再生所述音频信号。
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公开(公告)号：US20090064670A1
公开(公告)日：2009-03-12
申请号：US12279482
申请日：2007-02-19
申请人： Hirofumi Kimura , Tomomi Souda , Masanao Suzuki
发明人： Hirofumi Kimura , Tomomi Souda , Masanao Suzuki
IPC分类号： F01N7/10
CPC分类号： F01N13/1805 , B22C9/24 , F01N13/10 , F01N13/1861 , F01N2450/22 , F01N2530/00
摘要： An exhaust manifold made of heat-resistant cast steel comprising pluralities of flanges each having a hole connected to each exhaust port of a cylinder head of an engine with bolts, pluralities of ports connected to the flanges, and a convergence portion in which the ports are converging, the thickness of the flanges being 80-150% of that of the ports.
摘要翻译： 一种由耐热铸钢制成的排气歧管，包括多个凸缘，每个凸缘均具有连接到具有螺栓的发动机的气缸盖的每个排气口的孔，连接到凸缘的多个端口和端口为 会聚，凸缘的厚度为端口厚度的80-150％。
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公开(公告)号：US20080219344A1
公开(公告)日：2008-09-11
申请号：US12068833
申请日：2008-02-12
IPC分类号： H04B1/66
CPC分类号： G10L21/038 , G10L19/083
摘要： When creating SBR data in a the low-resolution mode, an encoding device divides a high-frequency component of input audio data being encoded by SBR method into a high-frequency band and a low-frequency band, and calculates an average high-frequency power value that indicates the average value of the power in the high-frequency band of the audio data, as well as an average low-frequency power value that indicates the average value of the power in the low-frequency band of the audio data. The encoding device then compares the average high-frequency power value and the average low-frequency power value, selecting the smaller of the two. The encoding device then corrects the power of the high-frequency component of the signal being encoded by the SBR method so that it equals the selected average power value.
摘要翻译： 当以低分辨率模式创建SBR数据时，编码装置将由SBR方法编码的输入音频数据的高频分量划分成高频带和低频带，并计算平均高频 指示音频数据的高频带中的功率的平均值的功率值以及指示音频数据的低频带中的功率的平均值的平均低频功率值。 然后，编码装置比较平均高频功率值和平均低频功率值，选择两者中较小的一个。 然后，编码装置校正由SBR方法编码的信号的高频分量的功率，使得其等于所选择的平均功率值。
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公开(公告)号：US20080154589A1
公开(公告)日：2008-06-26
申请号：US12073276
申请日：2008-03-03
IPC分类号： G10L19/00
CPC分类号： G10L19/035 , G10L25/27
摘要： To alleviate degradation of sound quality which may be caused by pre-echoes and bit starvation. An acoustic analyzer analyzes an audio signal to calculate perceptual entropy indicating how many bits are required for quantization. A coded bit count monitor monitors the number of coded bits produced from the audio signal and calculates the number of available bits for the current frame. Based on the combination of the perceptual entropy and the number of available bits, a frame division number determiner determines a division number N for dividing a frame of the audio signal into N blocks. An orthogonal transform processor divides a frame by the determined division number and subjects each divided block of the audio signal to an orthogonal transform process, thereby obtaining orthogonal transform coefficients. A quantizer quantizes the orthogonal transform coefficients on a divided block basis.
摘要翻译： 以缓解由回声和位饥饿引起的声音质量下降。 声学分析仪分析音频信号以计算感知熵，指示量化所需的位数。 编码比特计数监视器监视从音频信号产生的编码比特数，并计算当前帧的可用比特数。 基于感知熵与可用比特数的组合，帧分割数确定器确定用于将音频信号的帧划分成N个块的分割数N. 正交变换处理器将帧除以确定的分割数，并将音频信号的每个分割块进行正交变换处理，从而获得正交变换系数。 量化器基于分割块量化正交变换系数。
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48.发明授权公开(公告)号：US07089179B2
公开(公告)日：2006-08-08
申请号：US09386824
申请日：1999-08-31
申请人： Yasuji Ota , Masanao Suzuki , Yoshiteru Tsuchinaga
发明人： Yasuji Ota , Masanao Suzuki , Yoshiteru Tsuchinaga
IPC分类号： G10L19/00
CPC分类号： G10L19/10 , G10L2019/0008
摘要： A gain unit scales a code vector Ci output from a configuration variable code book by a gain g after the positions of non-zero samples are controlled according to an index and transmission parameter p. A linear prediction synthesis filter input the multiplication result, and outputs a regenerated signal gACi. A subtracter outputs an error signal E by subtracting the regenerated signal gACi from an input signal X. A error power evaluation unit computes an error power according to an error signal E. The above described processes are performed on all code vectors Ci and gains g. The index i of the code vector Ci and the gain g with which the error power is the smallest are computed and transmitted to the decoder.
摘要翻译： 根据索引和传输参数p，在非零样本的位置被控制之后，增益单元根据增益g缩放从配置变量代码簿输出的代码矢量Ci。 线性预测合成滤波器输入乘法结果，并输出再生信号gACi。 减法器通过从输入信号X中减去再生信号gACi来输出误差信号E.误差功率评估单元根据误差信号E来计算误差功率。对所有码矢量Ci和增益g进行上述处理。 计算代码矢量Ci的索引i和误差功率最小的增益g并将其传送到解码器。
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公开(公告)号：US20050238013A1
公开(公告)日：2005-10-27
申请号：US10927987
申请日：2004-08-27
IPC分类号： H04L12/70 , H04L12/885 , H04L12/56
CPC分类号： H04L47/10 , H04L47/22 , H04L47/50 , H04L47/564 , H04L65/80
摘要： In a packet receiving method and device which convert a voice packet received into a voice, a receiving packet buffer temporarily stores a voice packet received; a plurality of parameter information monitors respectively determine different buffer adjustment values for determining a buffering amount of the receiving packet buffer based on one or more pieces of parameter information obtained from the voice packet temporarily stored; a buffer adjustment value determiner determines a receiving buffer adjustment value from the plural buffer adjustment values; and a buffer controller controls the buffering amount based on the receiving buffer adjustment value.
摘要翻译： 在将接收到的语音分组转换为语音的分组接收方法和装置中，接收分组缓冲器临时存储接收的语音分组; 多个参数信息监视器分别基于从临时存储的语音分组获得的一个或多个参数信息来确定用于确定接收分组缓冲器的缓冲量的不同缓冲器调整值; 缓冲器调整值确定器从多个缓冲器调整值确定接收缓冲器调整值; 并且缓冲器控制器基于接收缓冲器调整值来控制缓冲量。
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50.发明授权Voice encoding and voice decoding using an adaptive codebook and an algebraic codebook 有权使用自适应码本和代数码本的语音编码和语音解码公开(公告)号：US06594626B2
公开(公告)日：2003-07-15
申请号：US10046125
申请日：2002-01-08
申请人： Masanao Suzuki , Yasuji Ota , Yoshiteru Tsuchinaga
发明人： Masanao Suzuki , Yasuji Ota , Yoshiteru Tsuchinaga
IPC分类号： G10L1904
CPC分类号： G10L19/09 , G10L19/04 , G10L19/107 , G10L19/18 , G10L2019/0008
摘要： Disclosed is a voice encoding method having a synthesis filter implemented using linear prediction coefficients obtained by dividing an input signal into frames each of a fixed length, and subjecting the input signal to linear prediction analysis in the frame units, generating a reconstructed signal by driving said synthesis filter by a periodicity signal output from an adaptive codebook and a pulsed signal output from an algebraic codebook, and performing encoding in such a manner that an error between the input signal and said reproduced signal is minimized, wherein there are provided an encoding mode 1 that uses pitch lag obtained from an input signal of a present frame and an encoding mode 2 that uses pitch lag obtained from an input signal of a past frame. Encoding is performed in encoding mode 1 and encoding mode 2, the mode in which the input signal can be encoded more precisely is decided frame by frame and encoding is carried out on the basis of the mode decided.
摘要翻译： 公开了一种语音编码方法，其具有使用通过将输入信号划分成固定长度的帧而获得的线性预测系数实现的合成滤波器，并且对输入信号进行帧单位的线性预测分析，通过驱动所述 通过从自适应码本输出的周期信号和从代数码本输出的脉冲信号的合成滤波器，并且以使输入信号和所述再现信号之间的误差最小化的方式进行编码，其中提供了编码模式1 使用从当前帧的输入信号获得的音调滞后和使用从过去帧的输入信号获得的音调滞后的编码模式2。 在编码模式1和编码模式2中执行编码，逐帧地确定输入信号可以被更精确地编码的模式，并且基于所决定的模式进行编码。
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