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Apparatus and techniques for audio and voice transmission over noisy wireless links.

机译：在嘈杂的无线链路上进行音频和语音传输的设备和技术。

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摘要

The widespread of audio and voice transmission over wireless networks imposes several challenges upon the design and implementation of efficient transmission schemes. Some of these challenges include reducing the packetization overhead of small audio and voice frames as well as protecting their content against both bit errors introduced by wireless fading channels and packet erasures introduced by network buffering. In this dissertation, we present several optimization frameworks that address the afore-mentioned challenges.;First, we looked into the problem of transmitting a stored audio sequence. To that end, we present an optimization framework designed to optimally protect a stored audio sequence transmitted over a wireless link while considering the packetization overhead of audio frames. Utilizing rate compatible punctured RS codes and dynamic programming, it identifies the optimal assignment of parity to audio frames according to their perceptual importance such that the Segmented SNR of the received audio sequence is maximized. Our framework covers two cases. In the first case, a frame grouping technique is proposed to packetize audio frames and protect them against temporarily correlated bit errors introduced by a fading wireless channel. In this case, each packet is treated as a channel coding codeword. In the second case, a one-dimensional RS coder is applied vertically to a sequence of horizontally formed packets associated with an audio sequence in order to protect the sequence against both bit errors introduced by fading wireless channels and packet erasures introduced by network buffering.;Second, we looked into the problem of transmitting live voice sequences over tandem links, we present two novel frameworks that aim at unequal protection of voice frames according to their perceptual importance such that the expected distortion of a received voice sequence is minimized. The first framework proposes a hybrid media dependent and media independent error correction scheme to mitigate tandem loss. The main objective of the framework is to reduce the buffering delay at both transmitter and receiver, in turn reducing the overall end-to-end delay. The second framework utilizes a rate-adaptation mechanism identifying the optimal allocation of source- and channel-coding components for each voice frame. While the first frame-work offers a better end-to-end delay profile and a better performance in moderate as well as good channel conditions, the second framework provides a stronger error correction capability making it the preferred choice for bad channel conditions and for voice applications that can tolerate a higher transmission delay. Our simulation results showed that the proposed audio and voice transmission schemes have better performance compared to their literature counterpart.
机译：无线网络上音频和语音传输的广泛普及对有效传输方案的设计和实现提出了一些挑战。这些挑战中的一些挑战包括减少小型音频和语音帧的打包开销，以及保护它们的内容不受无线衰落信道引入的误码和网络缓冲引入的分组擦除的影响。本文提出了几种优化框架来应对上述挑战。首先，我们研究了传输存储的音频序列的问题。为此，我们提出了一种优化框架，该框架旨在考虑到音频帧的打包开销，从而最佳地保护通过无线链路传输的存储音频序列。利用速率兼容的穿孔RS码和动态编程，它可以根据音频帧的感知重要性识别奇偶校验的最佳分配，从而使接收到的音频序列的分段SNR最大化。我们的框架涵盖两种情况。在第一种情况下，提出了一种帧分组技术来对音频帧进行打包，并保护它们免受衰落无线信道引入的暂时相关的误码。在这种情况下，每个分组被视为信道编码码字。在第二种情况下，将一维RS编码器垂直应用于与音频序列关联的水平形成的数据包序列，以保护该序列免受无线信道衰落引起的误码和网络缓冲引起的数据包擦除的影响。其次，我们研究了在串联链路上传输实时语音序列的问题，我们提出了两个新颖的框架，旨在根据语音帧的感知重要性来对语音帧进行不平等的保护，以使接收到的语音序列的预期失真最小化。第一个框架提出了一种混合媒体相关和媒体独立错误纠正方案，以减轻串联损失。该框架的主要目标是减少发送器和接收器的缓冲延迟，进而减少总的端到端延迟。第二个框架利用速率自适应机制，为每个语音帧识别源代码和信道编码组件的最佳分配。虽然第一个框架在中等以及良好的信道条件下提供了更好的端到端延迟曲线和更好的性能，但是第二个框架提供了更强的纠错能力，使其成为恶劣的信道条件和语音的首选可以承受更高传输延迟的应用。我们的仿真结果表明，与文献中的同类相比，所提出的音频和语音传输方案具有更好的性能。

著录项

  • 作者

    Khalifeh, Ala'.;

  • 作者单位

    University of California, Irvine.;

  • 授予单位 University of California, Irvine.;
  • 学科 Engineering Electronics and Electrical.
  • 学位 Ph.D.
  • 年度 2010
  • 页码 118 p.
  • 总页数 118
  • 原文格式 PDF
  • 正文语种 eng
  • 中图分类
  • 关键词

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