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研究生:林敬雄
研究生(外文):Lin Jing Shyong
論文名稱:AMR編碼效能分析及Viterbi解碼器於數位訊號處理器之實現
論文名稱(外文):AMR Coding Performance Analysis and DSP Implementation of Viterbi Decoder
指導教授:杭學鳴杭學鳴引用關係
指導教授(外文):Hang Hsueh-Ming
學位類別:碩士
校院名稱:國立交通大學
系所名稱:電子工程系
學門:工程學門
學類:電資工程學類
論文種類:學術論文
論文出版年:2002
畢業學年度:90
語文別:中文
論文頁數:110
中文關鍵詞:AMRViterbi Decoder
外文關鍵詞:AMRViterbi Decoder
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近年來,多媒體與無線通訊已成為市場的主要發展方向,對消費者而言,面對不同的應用需求,也衍生出不同的傳輸品質要求。因此,如何在既有的傳輸通道上,將各種不同的應用與網路的傳輸保護作整合性的規範,並保持傳輸品質的穩定,已成為研究趨勢之一。在另一方面,因應資訊產品週期日益縮短,如何將不同的應用利用現有的數位訊號處理器來實現也成為另一個值得探討的課題。
在論文中,我們將介紹第三代無線通訊系統所採用的語音編碼標準「適應性多速率編碼」,在這新一代的語音標準中引進了訊源與通道聯合編碼的概念。我們將簡單的描述標準內部演算法與編碼的架構,並詳述訊源與通道編碼的整合方式,包括非等同性保護(Unequal Error Protection)、多重比例保護碼(Multi- rate protection)、對角式交錯排序(diagonal block interleaver)等,並針對不同特性的通道條件進行效能評估。另外我們選擇頻譜差異分析作為語音品質的量度參考,並根據量測數據指出現有標準未盡周詳的疏漏之處,最後並將針對其中表現較為特殊的模式進行理論上的分析和深入的解釋。
除此之外,我們利用德州儀器公司所發展的數位訊號處理器為平台,在之上進行Viterbi 解碼器的實現,在這裡我們的焦點主要集中在整體演算法的執行速度及程式所需要的記憶體空間。我們會先簡單的描述整體硬體架構的設計及運作模式,另外,我們也將介紹程式編譯器所採用的編譯技術。我們同時也將整個程式最佳化的流程作詳細的舉例說明,並提供在程式寫作風格(coding style)上需要注意的地方,並得到相當程度的效能提升。我們也將從編譯器所提供的相關資訊作進一步的程式分析,並瞭解整體程式如何實際在數位訊號處理器上運作。最後我們證明能夠有效的提升整體執行速度為原先的二十倍。

Multimedia and wireless communication have been the two high tech main streams in the recent years. Consumers are demand ing for better QoS (Quality of Service) in a variety of applications. A related research topic is the possibility of joint source/channel coding that provides a way to overcome the channel fading while it still keeps the service quality above a certain level. DSP implementation is another issue in practical
applications. It is a quick time-to-market solution for industry; however, we need to tune the DSP algorithms to overcome its speed and structure limitations.
One focus of this thesis is AMR (Adaptive Multi Rate), the speech standard of 3GPP. One of its features is that it takes the channel condition into account in its design.
We will describe the basic structure of the AMR speech and channel coder and how they are combined together. We then show the simulation results of the AMR under various channel conditions. Here, SD (Spectral Distance) is chosen to be our
methodology for speech quality assessment. We identified the problems of some AMR modes (12.2 and 7.95 kbit/s) operated in a noisy environment. Further, we try to give the reasons why these codecs do not have robust performance under noisy channel
conditions. Finally, we conclude that the simulation results indicate that all the other modes have similar reconstructed speech quality except for the modes of 12.2 and 7.95 kbits/s. There is no apparent advantage in selecting a specific coding mode based on the channel error condition.
The second focus of this thesis is DSP implementation. We implement the Viterbi decoder on the Texas Instrument (TI) TMS320C6701 digital signal processor (DSP).
The trade-off between code size and speed is our focused point. After a brief review of the DSP hardware architecture and its software tools, we will show how we optimize the source program step by step and finally achieve a significant improvement on speed and/or code size. Then, we analyze the feedback from the DSP compiler and examine the flow of the algorithm executed on the DSP. At the end, the optimized version runs 23 times faster than the un-optimized one.

Content
1 Introduction 1
2 Adaptive Multi-Rate of Speech Coding 3
2.1 The Overview of AMR.................................................................................... 3
2.2 Principles of the Encoder ................................................................................. 5
2.2.1 Pre-processing................................................................................... 6
2.2.2 Linear Prediction .............................................................................. 6
2.2.2.1 Windowing and auto-correlation ....................................... 6
2.2.2.2 Levinson-Durbin algorithm ............................................... 9
2.2.2.3 LP to LSP Conversion........................................................ 9
2.2.2.4 Monitoring resonance in the LPC spectrum .....................11
2.2.3 Open-loop pitch analysis ................................................................ 12
2.2.4 Impulse response computation (all modes) .................................... 12
2.2.5 Target signal computation (all modes) ........................................... 13
2.2.6 Adaptive codebook ......................................................................... 13
2.2.6.1 Adaptive codebook search............................................... 13
2.2.6.2 Adaptive codebook gain control (all modes) ................... 15
2.2.7 Algebraic codebook ........................................................................ 15
2.2.7.1 Algebraic codebook structure .......................................... 16
2.2.7.2 Algebraic codebook search.............................................. 16
2.2.8 Quantization of adaptive and fixed codebook gains ....................... 18
2.2.8.1 Adaptive codebook gain limitation.................................. 18
2.2.8.2 Quantization of codebook gains ...................................... 18
2.2.9 Memory update (all modes) ............................................................ 20
2.3 Functional description of the decoder............................................................ 21
2.3.1 Decoding and speech synthesis ...................................................... 21
2.3.2 Post-processing ............................................................................... 24
2.3.2.1 Adaptive post- filtering (all modes).................................. 24
2.3.2.2 High-pass filtering and up-scaling................................... 25
2.4 Bit Allocation................................................................................................. 26
3 Adaptive Multi-Rate of Channel Coding 28
3.1 Introduction.................................................................................................... 28
3.1.1 Unequal error protection................................................................. 29
3.1.2 Recursive systematic convolutional codes ..................................... 30
3.2 Principle of channel Encoder ......................................................................... 31
3.2.1 Speech Channel at Full Rate (TCH/AFS)....................................... 32
3.2.1.1 Coding of the in-band data .............................................. 32
3.2.1.2 Ordering for subjective importance ................................. 32
3.2.1.3 Parity for speech frames .................................................. 33
3.2.1.4 Convolutional encoder ..................................................... 33
3.2.2 Speech Channel at Half Rate (TCH/AHS) ..................................... 36
3.2.2.1 Coding of the in-band data .............................................. 36
3.2.2.2 Ordering for subjective importance ................................. 36
3.2.2.3 Parity for speech frames .................................................. 37
3.2.2.4 Convolutional encoder ..................................................... 38
3.3 Principle of Channel Decoder........................................................................ 40
3.3.1 Decoding with a “standard” NSC decoder ..................................... 40
3.3.2 Viterbi decoding algorithm............................................................. 41
3.4 Principle of Interleaver .................................................................................. 43
4 Codec Robustness against Channel Error 47
4.1 SQA methodology ......................................................................................... 48
4.1.1 Spectral Distance ............................................................................ 49
4.2 Channel Error Models.................................................................................... 49
4.2.1 AWGN Channel Model .................................................................. 49
4.2.2 Gilbert Channel Model ................................................................... 51
4.3 Simulations .................................................................................................... 52
4.3.1 Error Correction Capability ............................................................ 53
4.3.2 CRC Error Detection ...................................................................... 60
4.3.3 Measurement of Spectral Distance ................................................. 65
4.3.3.1 Bit Sensitivity .................................................................. 67
4.3.3.2 Measurement of Spectral Distance (Other Cases) ........... 71
4.4 Performance Analysis .................................................................................... 75
5 Optimization of Viterbi decoder on TI TMSC6201 DSP 78
5.1 TMS320C6000 Environment......................................................................... 78
5.1.1 TMS320C6000 Architecture .......................................................... 78
5.1.2 TMS320C6000 Pipeline ................................................................. 79
5.1.3 Code Development Flow................................................................ 81
5.2 Optimization Method ..................................................................................... 81
5.2.1 Four Basic Optimization Methods .................................................. 82
5.2.2 Software Pipelining ........................................................................ 83
5.2.2.1 Qualification of Loop Pipelining ..................................... 85
5.2.2.2 Compiler’s Feedback ....................................................... 86
5.2.2.3 Software Pipeline the Loop ............................................. 87
5.3 Optimization for Viterbi Decoder.................................................................. 85
5.3.1 Structure analysis and simplification.............................................. 85
5.3.1.1 State Initialization............................................................ 89
5.3.1.2 State correlation............................................................... 90
5.3.2 Refine C code on TMSC6201 DSP ................................................ 91
5.4 Performance Analysis .................................................................................. 102
6 Conclusions and Future Work 104
6.1 Conclusion................................................................................................... 104
6.2 Future Work ................................................................................................. 106
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