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1999 IEEE Workshop on Speech Coding Proceedings. Model, Coders, and Error Criteria (Cat. No.99EX351)最新文献

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Voice activity detection for GSM adaptive multi-rate codec GSM自适应多速率编解码器语音活动检测
A. Vahatalo, I. Johansson
This paper describes the VAD (voice activity detection) for controlling DTX (discontinuous transmission) of the GSM AMR (adaptive multi-rate) speech codec. The algorithm is based on spectral estimation and periodicity detection. The VAD contains a 9-band IIR filter bank, which divides input signals into frequency bands. The signal level at each band is calculated. Background noise is estimated in each sub-band. The VAD decision is computed by comparing input signal level and background noise estimate. The algorithm incorporates novel methods to estimate background noise and to detect periodic components based on open-loop pitch gain. A new method is also derived to detect correlated complex signals like music.
本文介绍了用于控制GSM自适应多速率语音编解码器DTX(不连续传输)的VAD(语音活动检测)技术。该算法基于频谱估计和周期性检测。VAD包含一个9波段IIR滤波器组,它将输入信号划分为多个频段。计算每个频段的信号电平。在每个子带中估计背景噪声。通过比较输入信号电平和背景噪声估计来计算VAD决策。该算法采用了新的方法来估计背景噪声和基于开环螺距增益的周期分量检测。提出了一种检测音乐等相关复杂信号的新方法。
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引用次数: 14
An improved background noise coding mode for variable rate speech coders 一种改进的可变速率语音编码器的背景噪声编码模式
K. El-Maleh, P. Kabal
In this paper, we present a novel background noise coding scheme for variable rate speech coders. Existing approaches to noise coding at very low bit rates (i.e. below 1 kbps) fail to faithfully reproduce background noise resulting in a degradation of the overall perceptual quality. In our approach, classification of the noise type is used to select the type of excitation to be used at the receiver. To illustrate the benefits of our scheme, we have modified the noise coding mode of the CDMA enhanced variable rate codec (EVRC) to include the proposed class-dependent noise excitation model. Evaluation tests have shown that we have improved the overall quality with the proposed noise coding scheme without an increase in bit rate.
本文提出了一种新的用于可变速率语音编码器的背景噪声编码方案。现有的非常低比特率(即低于1kbps)的噪声编码方法不能忠实地再现背景噪声,从而导致整体感知质量的下降。在我们的方法中,噪声类型的分类用于选择在接收器上使用的激励类型。为了说明该方案的优点,我们修改了CDMA增强可变速率编解码器(EVRC)的噪声编码模式,以包含所提出的类相关噪声激励模型。评估测试表明,我们在不增加比特率的情况下提高了噪声编码方案的整体质量。
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引用次数: 2
A perturbation-based pre-processing algorithm for CELP-coders 基于微扰的celp编码器预处理算法
J. Jensen, S. H. Jensen, E. Hansen
A novel pre-processing algorithm for CELP-coders is proposed. The algorithm aims at perturbing the original signal slightly, such that the perturbed signal is subjectively indistinguishable from the original but can be coded more effectively. A key feature of the algorithm is the possibility of controlling the frequency domain properties of the perturbations. Preliminary simulations with the proposed algorithm in combination with a CELP-like coder indicate improvements in terms of segmental SNR and subjective speech quality.
提出了一种新的celp编码器预处理算法。该算法旨在对原始信号进行轻微扰动,使扰动后的信号在主观上与原始信号无法区分,但可以更有效地进行编码。该算法的一个关键特点是可以控制扰动的频域特性。将该算法与类似celp的编码器结合进行的初步模拟表明,在分段信噪比和主观语音质量方面有所改善。
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引用次数: 4
Coding distortion caused by a phase difference between the LP filter and its residual 由低电平滤波器与其残差之间的相位差引起的编码失真
M. Tammi, V.T. Ruoppila, S. Kuusisto, J. Saarinen
Several speech coding algorithms modify the time scale of the residual signal to facilitate efficient coding of pitch information. Time scaling, however, results in a phase difference between the coded residual signal and the time-variant linear prediction (LP) filter used for synthesis in the decoder. In this paper, we examine the coding distortion induced by this phase difference. Moreover, we show that it may cause audible artifacts to the synthesized speech even if lossless coding of all parameters is employed. These artifacts occur particularly at onsets when the frequency response of successive LP filters changes rapidly. A waveform interpolation coder is used to illustrate the effects of the phase mismatch.
有几种语音编码算法通过修改残差信号的时间尺度来实现对音高信息的有效编码。然而,时间尺度导致编码残差信号与解码器中用于合成的时变线性预测(LP)滤波器之间存在相位差。本文研究了这种相位差引起的编码失真。此外,我们表明,即使采用所有参数的无损编码,也可能对合成语音造成可听伪影。当连续低电压滤波器的频率响应迅速变化时,这些伪影尤其会发生。波形插值编码器用来说明相位失配的影响。
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引用次数: 8
SD optimization of spectral coders 频谱编码器的SD优化
P. Hedelin, F. Nordén, J. Skoglund
In spectral coding of speech, several different criteria are in use for designing and evaluating quantizers. One measure, spectral distortion (SD), has become dominant for comparisons between coders. At run-time, a coder normally quantizes vectors according to other measures, e.g. line spectrum frequency (LSF) distance, in order to keep computational complexity down. In this study, we adopt the SD criterion both in coder design and for quantizer operation. The quantizer is optimized to give minimal average SD scores, This allows us to address the question, is average SD measure really a good criterion, matching subjective ratings. We perform a few objective and subjective tests based on SD optimized coding and some versions thereof. Our tests imply that minimizing average SD may not lead to the best subjective scoring.
在语音的频谱编码中,量化器的设计和评价采用了几种不同的标准。光谱失真(SD)这一指标已成为编码器之间比较的主要指标。在运行时,编码器通常根据其他度量(例如线谱频率(LSF)距离)对矢量进行量化,以降低计算复杂度。在本研究中,我们在编码器设计和量化器操作中都采用了SD准则。量化器被优化为给出最小的平均SD分数,这使我们能够解决这个问题,平均SD测量是否真的是一个很好的标准,匹配主观评分。我们基于SD优化编码和一些版本进行了一些客观和主观的测试。我们的测试表明,最小化平均SD可能不会导致最佳的主观得分。
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引用次数: 16
Joint speech codec parameter and channel decoding of parameter individual block codes (PIBC) 参数分组码(PIBC)的联合语音编解码参数和信道解码
T. Fingscheidt, S. Heinen, P. Vary
In digital mobile speech transmission usually the most important (class la) bits provided by the speech coding scheme are protected by a CRC for error detection. As a consequence all parameters spanned by the class la bits have to be marked at the receiver either as reliable or as unreliable. In contrast to this somewhat coarse approach we propose the usage of what we call parameter individual block codes (PIBC) for the most important codec parameters. This allows joint speech codec parameter and PIBC decoding taking advantage of the error concealing properties of soft-bit speech decoding.
在数字移动语音传输中,通常由语音编码方案提供的最重要的(la类)位由CRC保护以进行错误检测。因此,在接收端,所有由la类比特所跨越的参数都必须被标记为可靠或不可靠。与这种略显粗糙的方法相反,我们建议对最重要的编解码器参数使用我们称之为参数单个块码(PIBC)的方法。这允许联合语音编解码器参数和PIBC解码,利用软位语音解码的错误隐藏特性。
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引用次数: 20
Closed-loop tracking of sinusoids for speech and audio coding 用于语音和音频编码的正弦波闭环跟踪
R. Taori, R. Sluijter
A well recognised problem in low bit rate representation of audio and speech signals, based on the sinusoidal model, is that of tracking the sinusoidal components. Imperfections in the analysis process and the presence of components over a limited duration of time gives rise to ambiguities in the tracking process. As a solution to this problem, we propose a mechanism to achieve closed-loop tracking by means of using analysis-by-synthesis incorporating phase prediction. A simple implementation of such an algorithm is discussed by considering an overlap-add synthesizer. Finally, the results are presented using a voiced speech segment as an example.
在基于正弦模型的音频和语音信号的低比特率表示中,一个公认的问题是跟踪正弦分量。分析过程中的缺陷和在有限时间内存在的组件会导致跟踪过程中的模糊性。为了解决这一问题,我们提出了一种利用结合相位预测的合成分析来实现闭环跟踪的机制。通过考虑重叠加合成器,讨论了这种算法的一个简单实现。最后,以一个浊音段为例给出了结果。
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引用次数: 2
Radio link parameter based speech quality index-SQI 基于无线链路参数的语音质量指标- sqi
A. Karlsson, G. Heikkila, T. B. Minde, M. Nordlund, B. Timus
Measurement of cellular speech quality has applications from equipment installation to daily network maintenance and benchmarking. The area is under development, driven by cost and lead-time of subjective listening tests. Lately, field tests has shown usability for objective speech quality methods. The new SQI-measure, based on radio link parameters is one of these methods. It is independent of transmitted signal and can provide better performance than PSQM and much better performance then RxQual, when used for network tuning. In this paper, the SQI measure is described and categorized, performance comparison figures are presented, and a motivation that speech quality is possible to estimate given radio link status is given.
蜂窝语音质量的测量从设备安装到日常网络维护和基准测试都有应用。由于主观听力测试的成本和交付时间,这一领域正在开发中。最近,现场测试显示了客观语音质量方法的可用性。基于无线电链路参数的sqi测量就是其中一种方法。它独立于传输的信号,当用于网络调优时,可以提供比PSQM更好的性能,比RxQual更好的性能。本文对SQI测量进行了描述和分类,给出了性能比较数据,并给出了在给定无线电链路状态下语音质量可以估计的动机。
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引用次数: 9
How to deflate polynomials in LSP computation 如何消除LSP计算中的多项式
B. Dumitrescu, I. Tabus
In this paper we propose a new deflation algorithm for line spectral pair (LSP) computation in speech coding. This algorithm is much more reliable than other methods based on deflation.
本文提出了一种新的用于语音编码中线谱对(LSP)计算的压缩算法。该算法比其他基于通货紧缩的方法可靠得多。
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引用次数: 4
A time warper for speech signals 语音信号的时间扭曲器
R. Sluijter, A.J.E.M. Janssen
A parabolic time warper designed to enhance the stationarity of voiced speech segments, is presented. It is shown how, for a harmonic signal segment, the parabolic time warping function can remove the part of the frequency variation which progresses linearly with time, without changing the time duration of that segment. In the actual implementation of the time warping system, the linear part of the pitch frequency variation in a segment is removed on the basis of maximization of the pitch-related autocorrelation peak of the warped signal. As a by-product, the time warper yields a very reliable pitch estimation. An example on real speech is discussed.
提出了一种抛物线型时间失真器,用于提高浊音段的平稳性。对于谐波信号段,抛物线时间翘曲函数可以去除随时间线性发展的频率变化部分,而不改变该段的持续时间。在时间整波系统的实际实现中,在使被整波信号的音高相关自相关峰值最大化的基础上,去除一段音高频率变化的线性部分。作为副产品,时间扭曲产生了非常可靠的基音估计。最后讨论了一个真实语音的例子。
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引用次数: 13
期刊
1999 IEEE Workshop on Speech Coding Proceedings. Model, Coders, and Error Criteria (Cat. No.99EX351)
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