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📄 ale.mdl

📁 用dsp解压mp3程序的算法
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      ShowName		      off
      SourceBlock	      "dspsigops/Integer Delay"
      SourceType	      "Integer Delay"
      delay		      "1"
      ic		      "0"
      reset_popup	      "None"
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      SourceBlock	      "dspadpt3/LMS\nAdaptive Filter"
      SourceType	      "LMS Adaptive Filter"
      n			      "16"
      mu		      "0.01"
      ic		      "0.0"
      leakage		      "1.0"
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"noisy signal and enhances the sinewave.\nThe adaptive filter output contains "
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      Phase		      "0"
      SampleMode	      "Discrete"
      OutComplex	      "Real"
      CompMethod	      "Trigonometric fcn"
      TableSize		      "Speed"
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      Inputs		      "|+-"
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	Name			"Input signal"
	TestPoint		off
	LinearAnalysisOutput	off
	LinearAnalysisInput	off
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	DataLoggingNameMode	"SignalName"
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	DataLoggingDecimation	"1"
	DataLoggingLimitDataPoints off
	DataLoggingMaxPoints	"5000"
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    }
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      DstBlock		      "Flip"
      DstPort		      1
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      SrcBlock		      "Integer Delay"
      SrcPort		      1
      DstBlock		      "LMS\nAdaptive Filter"
      DstPort		      1
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      SrcPort		      1
      Points		      [45, 0]
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      SrcBlock		      "LMS\nAdaptive Filter"
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	Labels			[1, 0]
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	DstBlock		"LMS\nAdaptive Filter"
	DstPort			2
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      SrcBlock		      "Flip"
      SrcPort		      1
      DstBlock		      "Predictor\nCoefficients"
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    Annotation {
      Name		      "LMS Adaptive Linear Prediction"
      Position		      [291, 21]
      FontName		      "MS Sans Serif"
      FontSize		      14
      FontWeight	      "bold"
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}
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