SOME SIMPLE MANIPULATIONS OF SOUND USING DIGITAL SIGNAL PROCESSING Richard M.

Stern
18-791 demo
August 31, 2004

Department of Electrical and Computer Engineering and School of Computer Science Carnegie Mellon University Pittsburgh, Pennsylvania 15213

The original sound and its spectrogram

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Downsampling the waveform

Downsampling the waveform by factor of 2:
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0.015 0.01 0.005 0 -0.005 -0.01 -0.015 0 5 10 15 20 25 n 30 35 40 45 50

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Consequences of downsampling
Downsample
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Upsampling the waveform

Upsampling by a factor of 2:
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0.015 0.01 0.005 0 -0.005 -0.01 -0.015 0 20 40 60 80 100 n 120 140 160 180 200

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Consequences of upsampling

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Linear filtering the waveform

x[n]

y[n]

Filter 1: y[n] = 3.6y[n1]+5.0y[n2]3.2y[n3]+.82y[n4] +.013x[n].032x[n1]+.044x[n2].033x[n3]+.013x[n4] Filter 2: y[n] = 2.7y[n1]3.3y[n2]+2.0y[n3.57y[n4] +.35x[n]1.3x[n1]+2.0x[n2]1.3x[n3]+.35x[n4]

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Filter 1 in the time domain

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Output of Filter 1 in the frequency domain

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Filter 2 in the time domain

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Output of Filter 2 in the frequency domain

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The source-filter model of speech

A useful model for representing the generation of speech sounds: Amplitude

Pitch

Pulse train source p[n]

Vocal tract model Noise source

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Separating the vocal-tract excitation from the filter

Original speech: Speech with 75-Hz excitation:

Speech with 150-Hz excitation:

Speech with noise excitation:

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