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Wireless speech and audio communications

机译:无线语音和音频通信

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摘要

The limited applicability of Shannon’s separation theorem in practical speech/audio systems motivates the employment of joint source and channel coding techniques. Thus, considerableefforts have been invested in designing various types of joint source and channel coding schemes. This thesis discusses two different types of Joint Source and Channel Coding(JSCC) schemes, namely Unequal Error Protection (UEP) aided turbo transceivers as well as Iterative Source and Channel Decoding (ISCD) exploiting the residual redundancyinherent in the source encoded parameters.More specifically, in Chapter 2, two different UEP JSCC philosophies were designed for wireless audio and speech transmissions, namely a turbo-detected UEP scheme usingtwin-class convolutional codes and another turbo detector using more sophisticated Irregular Convolutional Codes (IRCC). In our investigations, the MPEG-4 Advanced Audio Coding (AAC), the MPEG-4 Transform-Domain Weighted Interleaved Vector Quantization (TwinVQ) and the Adaptive MultiRate WideBand (AMR-WB) audio/speech codecs were incorporated in the sophisticated UEP turbo transceiver, which consisted of a threestage serially concatenated scheme constituted by Space-Time Trellis Coding (STTC), Trellis Coded Modulation (TCM) and two different-rate Non-Systematic Convolutionalcodes (NSCs) used for UEP. Explicitly, both the twin-class UEP turbo transceiver assisted MPEG-4 TwinVQ and the AMR-WB audio/speech schemes outperformed their corresponding single-class audio/speech benchmarkers by approximately 0.5 dB, in terms of the required Eb/N0, when communicating over uncorrelated Rayleigh fading channels. By contrast, when employing the MPEG-4 AAC audio codec and protecting the class-1 audio bits using a 2/3-rate NSC code, a more substantial Eb/N0 gain of about 2 dB was achieved. As a further design alternative, we also proposed a turbo transceiver employing IRCCs for the sake of providing UEP for the AMR-WB speech codec. The resultant UEP schemes exhibited a better performance when compared to the corresponding EqualError Protection (EEP) benchmark schemes, since the former protected the audio/speech bits according to their sensitivity. The proposed UEP aided system using IRCCs exhibitsan Eb/N0 gain of about 0.4 dB over the EEP system employing regular convolutional codes, when communicating over AWGN channels, at the point of tolerating a SegSNR degradation of 1 dB. In Chapter 3, a novel system that invokes jointly optimised ISCD for enhancing the error resilience of the AMR-WB speech codec was proposed and investigated. The resultant AMR-WB coded speech signal is protected by a Recursive Systematic onvolutional (RSC) code and transmitted using a non-coherently detected Multiple-Input Multiple-Output (MIMO) Differential Space-Time Spreading (DSTS) scheme. To further enhance the attainable system performance and to maximise the coding advantage of the proposed transmission scheme, the system is also combined with multi-dimensional Sphere Packing (SP) modulation. The AMR-WB speech decoder was further developed for the sake of accepting the a priori information passed to it from the channel decoder as extrinsic information,where the residual redundancy inherent in the AMR-WB encoded parameters was exploited.Moreover, the convergence behaviour of the proposed scheme was evaluated with the aid of both Three-Dimensional (3D) and Two-Dimenstional (2D) EXtrinsic Information Transfer (EXIT) charts. The proposed scheme benefitted from the exploitation of the residual redundancy inherent in the AMR-WB encoded parameters, where an approximately 0.5 dB Eb/N0 gain was achieved in comparison to its corresponding hard speechdecoding based counterpart. At the point of tolerating a SegSNR degradation of 1 dB, the advocated scheme exhibited an Eb/N0 gain of about 1.0 dB in comparison to the benchmark scheme carrying out joint channel decoding and DSTS aided SP-demodulation in conjunction with separate AMR-WB decoding, when communicating over narrowband temporally correlated Rayleigh fading channels.In Chapter 4, two jointly optimized ISCD schemes invoking the soft-output AMRWB speech codec using DSTS assisted SP modulation were proposed. More specifically, the soft-bit assisted iterative AMR-WB decoder’s convergence characteristics were further enhanced by using Over-Complete source-Mapping (OCM), as well as a recursive precoder. EXIT charts were used to analyse the convergence behaviour of the proposed turbo transceivers using the soft-bit assisted AMR-WB decoder. Explicitly, the OCM aided AMR-WB MIMO transceiver exhibits an Eb/N0 gain of about 3.0 dB in comparison to the benchmark scheme also using ISCD as well as DSTS aided SP-demodulation, but dispensing with the OCM scheme, when communicating over narrowband temporally correlated Rayleigh fading channels. Finally, the precoded soft-bit AMR-WB MIMO transceiver exhibits an Eb/N0 gain of about 1.5 dB in comparison to the benchmark scheme dispensing with the precoder, when communicating over narrowbandtemporally correlated Rayleigh fading channels.
机译:香农分离定理在实际语音/音频系统中的适用性有限,这促使人们采用联合源和通道编码技术。因此,在设计各种类型的联合源和信道编码方案上已经投入了相当大的努力。本文讨论了两种不同类型的联合源和信道编码(JSCC)方案,即不等错误保护(UEP)辅助Turbo收发器以及利用源编码参数中固有的剩余冗余的迭代源和信道解码(ISCD)。在第2章中,针对无线音频和语音传输设计了两种不同的UEP JSCC原理,即使用双级卷积码的Turbo检测UEP方案和使用更复杂的不规则卷积码(IRCC)的Turbo检测器。在我们的研究中,将MPEG-4高级音频编码(AAC),MPEG-4变换域加权交织矢量量化(TwinVQ)和自适应多速率宽带(AMR-WB)音频/语音编解码器合并到复杂的UEP Turbo中收发器,由空时网格编码(STTC),网格编码调制(TCM)和两个用于UEP的不同速率非系统卷积码(NSC)构成的三级串行串联方案组成。明确地说,在通信时,就所需的Eb / N0而言,双级UEP Turbo收发器辅助的MPEG-4 TwinVQ和AMR-WB音频/语音方案均比其相应的单级音频/语音基准测试仪高出约0.5 dB。在不相关的瑞利衰落信道上。相比之下,当采用MPEG-4 AAC音频编解码器并使用2/3速率NSC码保护1类音频比特时,可获得大约2 dB的更大Eb / N0增益。作为进一步的设计替代方案,我们还提出了采用IRCC的Turbo收发器,以便为AMR-WB语音编解码器提供UEP。与相应的EqualError Protection(EEP)基准方案相比,所得UEP方案表现出更好的性能,因为前者根据其灵敏度来保护音频/语音位。当在AWGN信道上通信时,在容许1 dB的SegSNR衰减时,使用IRCC的拟议UEP辅助系统在采用常规卷积码的EEP系统上的Eb / N0增益大约为0.4 dB。在第三章中,提出并研究了一种新颖的系统,该系统调用联合优化的ISCD来增强AMR-WB语音编解码器的错误恢复能力。所得的AMR-WB编码语音信号受递归系统进化(RSC)码保护,并使用非相干检测的多输入多输出(MIMO)差分时空扩展(DSTS)方案进行传输。为了进一步增强可获得的系统性能并最大程度地提高建议的传输方案的编码优势,该系统还与多维球体打包(SP)调制相结合。 AMR-WB语音解码器的进一步开发是为了接受从信道解码器传递给它的先验信息作为外部信息,其中利用了AMR-WB编码参数固有的剩余冗余。借助三维(3D)和二维(2D)外部信息传递(EXIT)图对所提出的方案进行了评估。所提出的方案得益于AMR-WB编码参数中固有的剩余冗余的利用,与基于对应的基于硬语音解码的对等方相比,该方法可实现约0.5 dB Eb / N0的增益。在可以承受1 dB的SegSNR衰减的点上,与执行联合信道解码和DSTS辅助SP解调结合单独的AMR-WB解码的基准测试方案相比,所提倡的方案的Eb / N0增益约为1.0 dB在第四章中,提出了两种共同优化的ISCD方案,它们采用DSTS辅助SP调制来调用软输出AMRWB语音编解码器。更具体地说,通过使用超完备的源映射(OCM)和递归预编码器,进一步增强了软比特辅助迭代AMR-WB解码器的收敛特性。 EXIT图用于使用软位辅助AMR-WB解码器分析建议的Turbo收发器的收敛性能。明确地说,与使用ISCD以及DSTS辅助SP解调的基准方案相比,OCM辅助AMR-WB MIMO收发器的Eb / N0增益约为3.0 dB,但是在临时通过窄带通信时放弃了OCM方案相关瑞利衰落信道。最后当在窄带与时间相关的瑞利衰落信道上进行通信时,与放弃预编码器的基准方案相比,预编码的软比特AMR-WB MIMO收发器具有约1.5 dB的Eb / N0增益。

著录项

  • 作者

    Othman Noor Shamsiah;

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  • 年度 2008
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  • 原文格式 PDF
  • 正文语种 {"code":"en","name":"English","id":9}
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