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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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Robust voice activity detection for DTX operation of speech coders 语音编码器DTX操作的鲁棒语音活动检测
F. Basbug, S. Nandkumar, K. Swaminathan
Robust detection of voice activity for short-term speech frames is essential for discontinuous transmission (DTX) mode of operation of vocoders such as IS-641. A reference VAD for the IS-641 coder has been chosen for such a purpose and is based on the GSM-EFR (enhance full rate) VAD. We show by developing a comprehensive evaluation procedure that the reference VAD is sensitive to speech level variations. For example, a significant increase is seen in frames falsely classified as active at speech levels of 10 dB above or below nominal level. We propose a solution based on automatic gain control to reduce level sensitivity. Objective performance measures confirm the robustness of our proposed VAD.
短期语音帧的语音活动鲁棒检测对于is -641等声码器的不连续传输(DTX)操作模式至关重要。为此,选择了is -641编码器的参考VAD,它基于GSM-EFR(增强全速率)VAD。我们通过开发一个综合评估程序表明,参考VAD对语音水平变化很敏感。例如,当语音水平高于或低于标称水平10 dB时,被错误地分类为活动的帧显著增加。我们提出了一种基于自动增益控制的方案来降低电平灵敏度。客观性能测量证实了我们提出的VAD的鲁棒性。
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引用次数: 6
LSP quantization in wideband speech coders 宽带语音编码器中的LSP量化
M. Ferhaoui, S. Van Gerven
This paper deals with multi-stage vector quantization of line spectrum pair (LSP) parameters in wideband speech coders and discusses commonly used spectral distortion measures and their relation to the perceptual quality of the speech coding.
本文研究了宽带语音编码中线谱对(LSP)参数的多级矢量量化,讨论了常用的频谱失真度量及其与语音编码感知质量的关系。
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引用次数: 14
Quantization of SEW and REW components for 3.6 kbit/s coding based on PWI 基于PWI的3.6 kbit/s编码中SEW和REW分量的量化
U. Bhaskar, S. Nandkumar, K. Swaminathan, G. Zakaria
The design of a prototype waveform interpolation (PWI) based codec, operating at 3.6 kbit/s, is presented with main focus on the quantization of the slowly evolving waveform (SEW) and rapidly evolving waveform (REW) components. The SEW magnitude component is quantized using a hierarchical mean-shape-gain predictive vector quantization approach. SEW phase is derived using a phase model, based on a measure of voice periodicity. The REW magnitude is quantized using a gain and a sub-band based shape. The REW phase is obtained by high pass filtering a weighted combination of the SEW and a white noise process.
设计了一种基于波形插值(PWI)的编码解码器,其工作速度为3.6 kbit/s,主要研究了慢演变波形(SEW)和快速演变波形(REW)分量的量化。SEW幅度成分量化使用分层平均形状增益预测矢量量化方法。SEW相位是使用相位模型,基于测量语音周期性。利用增益和基于子带的形状对REW幅度进行量化。REW相位是通过对SEW和白噪声加权组合进行高通滤波得到的。
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引用次数: 3
Performance of current perceptual objective speech quality measures 当前感知客观语音质量测量的性能
L. Thorpe, Wonho Yang
This paper describes the performance of current objective speech quality measures designed to estimate subjective quality. We examined perceptual objective quality measures using a wide range of distortions including speech compression, wireless channel impairments, VoIP channel impairments, and modifications to the signal from features such as AGC. The results of this study indicate the range of conditions to which these objective measures may be applied, the validity of the estimates they provide, and the general maturity of the field.
本文描述了目前用于估计主观语音质量的客观语音质量度量的性能。我们检查了感知客观质量测量使用广泛的失真,包括语音压缩,无线信道损伤,VoIP信道损伤,和修改信号的特征,如AGC。本研究的结果表明,这些客观措施可能适用的条件范围,它们提供的估计的有效性,以及该领域的总体成熟度。
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引用次数: 64
The adaptive multi-rate speech coder 自适应多速率语音编码器
E. Ekudden, R. Hagen, I. Johansson, J. Svedberg
In this paper, we describe the adaptive multi-rate (AMR) speech coder currently under standardization for GSM systems as part of the AMR speech service. The coder is a multi-rate ACELP coder with 8 modes operating at bit-rates from 12.2 kbit/s down to 4.75 kbit/s. The coder modes are integrated in a common structure where the bit-rate scalability is realized mainly by altering the quantization schemes for the different parameters. The coder provides seamless switching on 20 ms frame boundaries. The quality when used on GSM channels is significantly higher than for existing services.
在本文中,我们描述了目前正在GSM系统标准化中的自适应多速率(AMR)语音编码器作为AMR语音服务的一部分。该编码器是一种多速率ACELP编码器,具有8种模式,工作速率从12.2 kbit/s到4.75 kbit/s。编码模式集成在一个共同的结构中,主要通过改变不同参数的量化方案来实现比特率的可扩展性。编码器在20毫秒帧边界上提供无缝切换。在GSM信道上使用时,质量明显高于现有业务。
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引用次数: 65
Advances in objective estimation of perceived speech quality 感知语音质量客观估计的研究进展
S. Voran
We present two techniques that can be used to enhance objective estimators of perceived speech quality. Frame normalization and frame-energy plane partitioning are described and applied to a log-spectral-error-based estimator. The resulting estimators are compared with each other and with two established estimators. This is done through correlation with MOS values from 17 formal subjective tests. We find that the proposed techniques significantly improve the log-spectral-error-based estimator.
我们提出了两种技术,可用于增强感知语音质量的客观估计。描述了帧归一化和帧-能量平面划分,并将其应用于基于对数频谱误差的估计器。将得到的估计量相互比较,并与两个已建立的估计量进行比较。这是通过与17个正式主观测试的MOS值的相关性来完成的。我们发现所提出的技术显著改善了基于对数光谱误差的估计。
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引用次数: 42
Study and subjective evaluation on MPEG-4 narrowband CELP coding under mobile communication conditions 移动通信条件下MPEG-4窄带CELP编码的研究与主观评价
K. Ozawa, T. Nomura, M. Serizawa, H. Ehara, K. Yoshida, N. Tana
This paper evaluates MPEG-4 narrowband (NB) CELP speech coding under various mobile communication conditions, such as clean, background noise and transmission errors. In order to make the codec robust against the errors with minimum increase of redundant bits, a CRC error correction code is attached into the codec as well as an error concealment is included in the decoder. Subjective evaluation results demonstrate that the speech quality for MPEG-4 speech coding at above 8.3 kb/s is higher than that for the ITU-T G.726 ADPCM at 32 kb/s in the clean speech condition. Further, the speech quality degradation is less than 0.1 in MOS under 10/sup -3/ bit error conditions, and still comparable to or higher than that for G.726 at 32 kb/s without error.
本文对MPEG-4窄带(NB) CELP语音编码在清洁、背景噪声和传输误差等不同移动通信条件下的性能进行了评价。为了使编解码器在最小冗余位增加的情况下对错误具有鲁棒性,在编解码器中附加了CRC纠错码,并在解码器中包含了错误隐藏。主观评价结果表明,在干净语音条件下,8.3 kb/s以上的MPEG-4语音编码的语音质量高于32 kb/s的ITU-T G.726 ADPCM。此外,在10/sup -3/比特误差条件下,MOS的语音质量退化小于0.1,并且仍然与32 kb/s无错误的G.726相当或更高。
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引用次数: 0
MVDR based all-pole modeling: properties, enhancements, and comparisons 基于MVDR的全极建模:属性、增强和比较
M. Murthi, B. Rao
In this paper, we present several features of minimum variance distortionless response (MVDR) based all-pole filters which are suitable for modeling all types of speech. In particular, we demonstrate how the MVDR all-pole spectrum, based upon time-domain correlations, can provide high quality spectral envelope modeling of voiced speech. Simulation results are included showing that the MVDR all-pole spectrum's modeling of voiced speech harmonics improves as the model order increases, leading to a monotonically decreasing spectral distortion. Furthermore, we show how the MVDR all-pole envelope can be enhanced by using forward-backward linear prediction. In addition, low order (10-14) MVDR based all-pole filters are examined and compared with other all-pole spectral envelopes. The reduced order MVDR all-pole spectrum is shown to compare favorably with linear prediction (LP) and LP cubic spline spectral envelopes in terms of spectral modeling and complexity.
在本文中,我们提出了基于最小方差无失真响应(MVDR)的全极滤波器的几个特征,这些特征适用于所有类型的语音建模。特别是,我们展示了基于时域相关性的MVDR全极谱如何为浊音语音提供高质量的频谱包络建模。仿真结果表明,随着模型阶数的增加,MVDR全极谱对浊音谐波的建模能力得到改善,频谱失真呈单调递减趋势。此外,我们还展示了如何使用正向向后线性预测来增强MVDR全极包络。此外,研究了基于低阶(10-14)MVDR的全极滤波器,并与其他全极谱包膜进行了比较。降阶MVDR全极谱在光谱建模和复杂性方面优于线性预测(LP)和LP三次样条谱包络。
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引用次数: 3
Embedded WI coding between 2.0 and 4.8 kbit/s 嵌入式WI编码2.0 ~ 4.8 kbit/s
Hong-Goo Kang, D. Sen
This paper describes an embedded speech coder based on waveform interpolation (WI) techniques. Since the quantization of line spectral frequency (LSF) parameters is fairly orthogonal to the quantization of excitation information, designing an embedded system with WI is much easier than that of other approaches. By using a hierarchical bit-allocation of excitation signals that consist of a slowly evolving waveform (SEW) and a rapidly evolving waveform (REW), the proposed system works well at the bit-rate of 2.0, 2.4, 3.0, 4.0 and 4.8 kbit/s. Listening tests indicate that the performance of the new system is comparable to an optimized fixed-rate WI coder, and the quality degrades gracefully as the bit-rate decreases.
介绍了一种基于波形插值(WI)技术的嵌入式语音编码器。由于线谱频率(LSF)参数的量化与激励信息的量化是完全正交的,因此用WI设计嵌入式系统要比用其他方法容易得多。通过使用由慢速波形(SEW)和快速波形(REW)组成的激励信号的分层位分配,该系统可以在2.0、2.4、3.0、4.0和4.8 kbit/s的比特率下良好地工作。监听测试表明,新系统的性能可与优化的固定速率WI编码器相媲美,并且随着比特率的降低,质量会优雅地下降。
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引用次数: 4
期刊
1999 IEEE Workshop on Speech Coding Proceedings. Model, Coders, and Error Criteria (Cat. No.99EX351)
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