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Quality Enhancement of Packet Audio with Time-Scale Modification

机译:具有时间尺度修改的数据包音频的质量增强

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In traditional packet voice or the emerging 2.5G and 3G wireless data services, smooth and timely delivery of audio is an essential requirement in Quality of Service (QoS) provision. It has been shown in our previous work that, by adapting time-scale modification to audio signals, an adaptive play-out algorithm can be designed to minimize packet dropping at the receiver end. By stretching the audio frame duration up and down, the proposed algorithm could adapt quickly to accommodate fluctuating delays including delay spikes. In this paper, we will address the packet audio QoS with emphasis on end-to-end delay, packet loss, and delay jitter. The characteristics of delay and loss will be discussed. Adaptive playback will enhance the audio quality by adapting to the transmission delay jitter and delay spike. Coupled with Forward Error Correction (FEC) schemes, the proposed delay and loss concealment algorithm achieves less overall application loss rate without sacrificing on the average end-to-end delay. The optimal solution of such algorithms will be discussed. We also investigate the stretching-ratio transition effect on perceived audio quality by measuring the objective Perceptual Evaluation of Speech Quality (PESQ) Mean Opinion Score (MOS).
机译:在传统的数据包语音或新兴2.5G和3G无线数据服务中,音频的平滑和及时交付是服务质量(QoS)提供的必要要求。在我们之前的工作中已显示,通过调整对音频信号的时间尺度修改,可以设计自适应播放算法以最小化接收器端的数据包。通过上下拉伸音频帧持续时间,该算法可以快速适应以适应波动延迟,包括延迟尖峰。在本文中,我们将以重点介绍端到端延迟,数据包丢失和延迟抖动来解决数据包音频QoS。将讨论延迟和损失的特征。自适应播放将通过适应传输延迟抖动和延迟尖峰来提高音频质量。耦合前向纠错(FEC)方案,所提出的延迟和损耗隐藏算法在不牺牲平均端到端延迟的情况下实现了较少的整体应用损耗速率。将讨论这种算法的最佳解决方案。我们还通过测量语音质量(PESQ)平均意见评分(MOS)的客观感知评估来研究对感知音频质量的拉伸比转换效应。

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