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This document discusses simple digital signal processing manipulations of sound, including downsampling and upsampling a waveform, applying linear filters, and separating vocal tract excitation from filtering. It presents the original sound and effects of downsampling on its spectrogram. Filters are designed and their effects shown in the time and frequency domains. A source-filter model of speech production is described.
This document discusses simple digital signal processing manipulations of sound, including downsampling and upsampling a waveform, applying linear filters, and separating vocal tract excitation from filtering. It presents the original sound and effects of downsampling on its spectrogram. Filters are designed and their effects shown in the time and frequency domains. A source-filter model of speech production is described.
This document discusses simple digital signal processing manipulations of sound, including downsampling and upsampling a waveform, applying linear filters, and separating vocal tract excitation from filtering. It presents the original sound and effects of downsampling on its spectrogram. Filters are designed and their effects shown in the time and frequency domains. A source-filter model of speech production is described.
Department of Electrical and Computer Engineering and School of Computer Science Carnegie Mellon University Pittsburgh, Pennsylvania 15213
Carnegie Mellon Slide 2 18-791 Digital Signal Processing I The original sound and its spectrogram Time 0 0.2 0.4 0.6 0.8 1 1.2 0 1000 2000 3000 4000 5000
Carnegie Mellon Slide 3 18-791 Digital Signal Processing I Downsampling the waveform Downsampling the waveform by factor of 2: 0 10 20 30 40 50 60 70 80 90 100 -0.015 -0.01 -0.005 0 0.005 0.01 0.015 n 0 5 10 15 20 25 30 35 40 45 50 -0.015 -0.01 -0.005 0 0.005 0.01 0.015 n
Carnegie Mellon Slide 4 18-791 Digital Signal Processing I Consequences of downsampling Time 0 0.1 0.2 0.3 0.4 0.5 0.6 0 1000 2000 3000 4000 5000 Original: Downsample Downsampled:
Carnegie Mellon Slide 5 18-791 Digital Signal Processing I Upsampling the waveform Upsampling by a factor of 2:
Carnegie Mellon Slide 8 18-791 Digital Signal Processing I Filter 1 in the time domain 0 20 40 60 80 100 120 -0.015 -0.01 -0.005 0 0.005 0.01 0.015 n 0 20 40 60 80 100 120 -2 0 2 4 6 8 x 10 -3 n
Carnegie Mellon Slide 9 18-791 Digital Signal Processing I Output of Filter 1 in the frequency domain Time 0 0.2 0.4 0.6 0.8 1 1.2 0 1000 2000 3000 4000 5000 Original: Lowpass:
Carnegie Mellon Slide 10 18-791 Digital Signal Processing I Filter 2 in the time domain 0 20 40 60 80 100 120 -0.015 -0.01 -0.005 0 0.005 0.01 0.015 n 0 20 40 60 80 100 120 -0.01 -0.005 0 0.005 0.01 n
Carnegie Mellon Slide 11 18-791 Digital Signal Processing I Output of Filter 2 in the frequency domain Time 0 0.2 0.4 0.6 0.8 1 1.2 0 1000 2000 3000 4000 5000 Original: Highpass:
Carnegie Mellon Slide 12 18-791 Digital Signal Processing I The source-filter model of speech A useful model for representing the generation of speech sounds: Pitch Pulse train source Noise source Vocal tract model Amplitude p[n]
Carnegie Mellon Slide 13 18-791 Digital Signal Processing I Separating the vocal-tract excitation from the filter