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Adaptive predictive coding with transform domain quantization using block size adaptation and high-resolution spectral modeling 采用块大小自适应和高分辨率光谱建模的变换域量化自适应预测编码
B. Bhaskar
The adaptive predictive coding with transform domain quantization (APC-TQ) technique was proposed by Bhaskar (1991) for the compression of audio signals. Since then, significant developments have taken place leading to a reduction in the coding rate. While enhancing the audio quality. These developments include (i) the use of block size adaptation to exploit the variations in the stationarity of the signal, (ii) high resolution spectral modeling using LPC analysis orders up to 64, and (iii) an adaptive bit-allocation procedure to minimize coding noise power as well as minimize the perception of coding noise. The result is a near transparent quality compression of 5 kHz bandwidth audio at a rate of 17 kbit/s. This technology will find applications in the distribution and transmission of AM quality audio programming over low rate channels such as the INMARSAT Standard A, B and aeronautical systems.<>
Bhaskar(1991)提出了带变换域量化的自适应预测编码(APC-TQ)技术,用于音频信号的压缩。从那时起,发生了重大发展,导致编码率降低。同时增强音频质量。这些发展包括(i)使用块大小自适应来利用信号平稳性的变化,(ii)使用高达64阶的LPC分析阶数进行高分辨率频谱建模,以及(iii)自适应比特分配程序来最小化编码噪声功率并最小化编码噪声的感知。其结果是以17kbit /s的速率对5 kHz带宽的音频进行近乎透明的质量压缩。该技术将在低速率信道(如INMARSAT标准A、B和航空系统)上的AM质量音频节目的分发和传输中得到应用。
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引用次数: 1
Hearing aids for profoundly deaf people based on a new parametric concept 基于新参数概念的深度聋人助听器
K. Hermansen, F. K. Fink, U. Hartmann, V.M. Hansen
People with severe hearing loss only have a minor part of the frequency range available for reception of information in speech signals. These people do not benefit from normal hearing aids as the information in high frequency parts of the speech is not available. To overcome this problem the authors have developed a new method enabling to present information from the frequency range of interest in the frequency range available for the hearing disabled. By means of parametric modeling of the speech production system, transforming the speech production model to match the available frequency range, and then finally resynthesize the speech using this transformed model, one can present the speech information of interest in a frequency range at choice. This concept is believed to reduce wideband background noise which is a problem for hearing disabled as well as for people with normal hearing ability.<>
严重听力损失的人只有一小部分频率范围可用于接收语音信号中的信息。这些人不能从正常的助听器中受益,因为无法获得讲话中高频部分的信息。为了克服这个问题,作者开发了一种新的方法,可以在听力残疾人可用的频率范围内从感兴趣的频率范围内呈现信息。通过对语音产生系统进行参数化建模,对语音产生模型进行变换,使其与可用的频率范围相匹配,最后利用变换后的模型重新合成语音,可以在任意频率范围内呈现出感兴趣的语音信息。这一概念被认为可以减少宽带背景噪音,这对听力障碍者和听力正常的人来说都是一个问题。
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引用次数: 3
A simplified source/filter model for percussive sounds 一个简化的打击声源/滤波器模型
J. Laroche, J. Meillier
This paper deals with source-filter models of percussive instruments. A 'multi-channel excitation/filter model' is presented in which a single excitation is used to generate several sounds, for example six piano tones belonging to the same octave. Techniques for estimating the model parameters are presented and applied to the sound of a real piano. Our experiments demonstrate that it is possible to calculate a single excitation signal which when fed into different filters, generates very accurate synthetic tones. Finally, a low-cost synthesis method is proposed that can be used to generate natural sounding percussive tones.<>
本文讨论了打击乐器的源-滤波器模型。提出了一种“多通道激励/过滤模型”,其中单个激励用于产生几种声音,例如属于同一八度的六个钢琴音调。给出了模型参数的估计方法,并将其应用于实际钢琴的声音中。我们的实验表明,可以计算出一个单一的激励信号,当输入到不同的滤波器时,产生非常精确的合成音调。最后,提出了一种低成本的合成方法,可以产生自然的打击乐。
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引用次数: 7
Constraint based audio interpolators 基于约束的音频插值器
D. Rossum
In audio digital signal processing, interpolators are used for a variety of functions, including sample rate conversion. Linear interpolation is commonly used, but has serious signal quality problems for signals with significant high frequency content. Higher order interpolators based on sine functions or other conventional lowpass filter design techniques offer somewhat better performance, but are not optimal in terms of perceived audio performance for a given degree of computational complexity. We present a new technique for the design of higher order interpolators for digital audio which provides improved performance at a given degree of complexity. Also presented are new methods for evaluating audio interpolators.<>
在音频数字信号处理中,插值器用于各种功能,包括采样率转换。线性插值是一种常用的插值方法,但对于高频含量较高的信号,线性插值存在严重的信号质量问题。基于正弦函数或其他传统低通滤波器设计技术的高阶插值器提供了更好的性能,但在给定计算复杂性程度的感知音频性能方面并不是最佳的。我们提出了一种设计数字音频高阶插值器的新技术,它在给定的复杂程度下提供了更好的性能。本文还介绍了评价音频插值器的新方法。
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引用次数: 7
Interpolation of forced structural responses from non-uniform sparse measurements 非均匀稀疏测量的强迫结构响应插值
Larry Heck, K. Naghshineh, J. Stach
This paper presents a method for interpolating a sparse set of nonuniformly spaced velocity measurements on the surface of a vibrating structure. The method utilizes knowledge of the physical nature of the vibrating structure specified in terms of a given bound on the energy of the excitation forces, estimated mobilities of the structure and a known set of sparse velocity measurements. To minimize the maximum possible error of the estimated surface velocities. The method employs an estimation approach derived from the theory of optimal signal recovery. Results are presented which demonstrate the performance of the method on interpolating surface velocities of a rectangular plate. With only four randomly selected point velocity measurements out of 209 possible locations. The method estimates the structural surface velocity with a normalized error of only -45 dB. The ability to achieve this performance with a small number of sensors makes this method important for many active noise control applications where an accurate measure of structural surface velocity is required to predict the radiated acoustic field.<>
本文提出了一种在振动结构表面插值非均匀间隔速度测量稀疏集的方法。该方法利用了振动结构的物理性质的知识,这些知识是根据激励力的能量的给定界限、结构的估计移动性和已知的一组稀疏速度测量值来指定的。使估计表面速度的最大可能误差最小化。该方法采用了一种由最优信号恢复理论导出的估计方法。结果表明,该方法对矩形板表面速度的插值是有效的。在209个可能的地点中,只有四个随机选择的测速点。该方法估计结构表面速度的归一化误差仅为-45 dB。使用少量传感器就能实现这种性能,这使得该方法对于许多主动噪声控制应用非常重要,这些应用需要精确测量结构表面速度来预测辐射声场
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引用次数: 0
A comparison of gradient-based algorithms for echo compensation with decorrelating properties 具有去相关特性的基于梯度的回波补偿算法比较
M. Rupp
Cancelling echoes by using the normalized least mean square (NLMS) algorithm has been state of the art for many years. In acoustical echo compensation, however, it is common to estimate more than 1000 parameters resulting in a too slow convergence when driven by speech signals. In order to overcome this drawback, a lot of modifications have been published in the last years, all having one goal: to decorrelate the driving process. Beginning with a deterministic approach we show that all these different ideas can be arranged in one scheme, allowing a uniform normalization. The different properties of the several algorithms are then obvious. A comparison of some algorithms with 2N-4N complexity is presented. Surprisingly, all algorithms do not work perfectly for a large compensator filter length and speech as input process.<>
使用归一化最小均方(NLMS)算法消除回波已经有很多年的历史了。然而,在声学回波补偿中,通常估计超过1000个参数,导致在语音信号驱动下收敛太慢。为了克服这个缺点,在过去的几年里发布了许多修改,所有这些修改都有一个目标:去关联驾驶过程。从确定性方法开始,我们展示了所有这些不同的想法可以安排在一个方案中,允许统一的规范化。这几种算法的不同性质是显而易见的。比较了几种复杂度为2N-4N的算法。令人惊讶的是,所有的算法都不能完美地工作于一个大的补偿器滤波器长度和语音作为输入过程。
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引用次数: 1
Improving joint stereo audio coding by adaptive inter-channel prediction 利用自适应信道间预测改进联合立体声音频编码
H. Fuchs
A method for exploiting inter-channel redundancies of stereophonic or multichannel audio signals is presented. In contrast to known stereo redundancy reduction techniques used in joint stereo audio coding. Where only the statistical dependencies between two concurrent samples of the left and right channel signals are considered, the adaptive inter-channel prediction also takes into account possible phase or time delay between the channels and exploits more than only one value of the cross-correlation function. The analysis of subjective listening test results has shown that this technique is especially effective for a class of test sequences which has proven to be most critical for the ISO MPEG Layer II and Layer III codecs at bit rates of 2/spl times/64 kbit/s. For these signals the gain due to the stereo redundancy reduction technique used in Layer III joint stereo coding is less than 5-10 dB, while in Layer II joint stereo coding no specific stereo redundancy reduction technique is used. In a first step, the adaptive inter-channel prediction has been applied to an ISO MPEG Layer II codec. The simulation results show that a prediction gain up to 30-40 dB can be achieved for large parts of the above mentioned signals.<>
提出了一种利用立体声或多声道音频信号的信道间冗余的方法。对比已知的立体声冗余减少技术用于联合立体声音频编码。在仅考虑左右信道信号的两个并发样本之间的统计相关性的情况下,自适应信道间预测还考虑了信道之间可能的相位或时间延迟,并利用了不止一个互相关函数的值。主观聆听测试结果的分析表明,该技术对一类测试序列特别有效,这些序列已被证明是ISO MPEG第二层和第三层编解码器在2/spl次/64 kbit/s的比特率下最关键的。对于这些信号,第三层联合立体声编码中使用的立体声冗余降低技术所带来的增益小于5-10 dB,而在第二层联合立体声编码中没有使用特定的立体声冗余降低技术。在第一步,自适应信道间预测已应用于ISO MPEG第二层编解码器。仿真结果表明,对上述大部分信号的预测增益可达30 ~ 40db。
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引用次数: 31
Current and future standardization of high-quality digital audio coding in MPEG MPEG中高质量数字音频编码的当前和未来标准化
R. van der Waal, K. Brandenburg, G. Stoll
Since 1988 ISO/IEC JTCI/SC29 WG11 (MPEG) is working on the standardization of video and audio signals. The Audio subgroup of MPEG is working on bit rate reduction systems for high quality digital audio. Since the first phase of this standardization effort has been finished, MPEG/Audio is extending its work to multichannel audio coding systems as well as to medium quality coding at lower sampling frequencies and lower bit rates. Future standardization work aims at next-generation coder suitable for high quality audio transmission and storage at bit rates of 64 kb/s per channel and well below.<>
自1988年以来,ISO/IEC JTCI/SC29 WG11 (MPEG)一直致力于视频和音频信号的标准化。MPEG的音频小组正在研究用于高质量数字音频的比特率降低系统。由于该标准化工作的第一阶段已经完成,MPEG/Audio正在将其工作扩展到多通道音频编码系统以及低采样频率和低比特率的中质量编码。未来的标准化工作旨在下一代编码器适合高质量的音频传输和存储,比特率为每通道64 kb/s或远低于64 kb/s。
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引用次数: 6
Developments in transaural stereo 跨耳立体声的发展
Jerry Bauck, Duane H. Cooper
Transaural stereo achieves precision 3-D imaging by compensating for spectral distortions in the loudspeaker-to-car signal paths. The heart of transaural stereo, signal processing for crosstalk cancellation, is herein generalized to accommodate any number of loudspeakers and listeners in any layout. Transaural equations are written and then solved using standard algebraic methods. Worked-out examples are shown and several applications are proposed.<>
通过补偿扬声器到汽车信号路径中的频谱失真,跨声立体声实现了精确的三维成像。跨声立体声的核心,串声消除的信号处理,在这里被概括为适应任何数量的扬声器和听众在任何布局。先写出跨耳方程,然后用标准代数方法求解。给出了算例,并提出了几种应用。
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引用次数: 2
Generalized overlap-add sinusoidal modeling applied to quasi-harmonic tone synthesis 广义叠加正弦建模在拟谐音合成中的应用
E. George, M.J.T. Smith
Analysis-by-synthesis/overlap-add (ABS/OLA) sinusoidal modeling has been successfully demonstrated as an accurate, flexible, and computationally tractable representation for the purposes of speech modification and harmonic tone synthesis; however, the model formulation used to synthesize these signals does not take full advantage of the structure of quasi-harmonic music signals. This paper describes a generalized overlap-add sinusoidal model formulation that accounts for the time-frequency behavior of quasi-harmonic tones and which reduces to the previous formulation as a special case.<>
合成分析/重叠添加(ABS/OLA)正弦建模已被成功证明是一种准确、灵活和计算易于处理的表示,用于语音修改和谐波合成;然而,用于合成这些信号的模型公式并没有充分利用准谐波音乐信号的结构。本文描述了一种广义的叠加正弦模型公式,该公式考虑了准谐波的时频行为,并作为一种特殊情况简化为前面的公式。
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引用次数: 5
期刊
Proceedings of IEEE Workshop on Applications of Signal Processing to Audio and Acoustics
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