Sound is produced by variations in air pressure that form waves. The frequency of a sound wave is measured in Hertz (Hz) and determines its pitch, while the amplitude determines its perceived volume. Digital sound works by sampling the amplitude of sound waves thousands of times per second and storing each value as a number. For accurate reproduction, the sampling rate must be at least twice the highest frequency per the Nyquist theorem. Computer sound files store long arrays of sample values that can be manipulated programmatically.
Sound is produced by variations in air pressure that form waves. The frequency of a sound wave is measured in Hertz (Hz) and determines its pitch, while the amplitude determines its perceived volume. Digital sound works by sampling the amplitude of sound waves thousands of times per second and storing each value as a number. For accurate reproduction, the sampling rate must be at least twice the highest frequency per the Nyquist theorem. Computer sound files store long arrays of sample values that can be manipulated programmatically.
Sound is produced by variations in air pressure that form waves. The frequency of a sound wave is measured in Hertz (Hz) and determines its pitch, while the amplitude determines its perceived volume. Digital sound works by sampling the amplitude of sound waves thousands of times per second and storing each value as a number. For accurate reproduction, the sampling rate must be at least twice the highest frequency per the Nyquist theorem. Computer sound files store long arrays of sample values that can be manipulated programmatically.
Sound is produced by variations in air pressure that form waves. The frequency of a sound wave is measured in Hertz (Hz) and determines its pitch, while the amplitude determines its perceived volume. Digital sound works by sampling the amplitude of sound waves thousands of times per second and storing each value as a number. For accurate reproduction, the sampling rate must be at least twice the highest frequency per the Nyquist theorem. Computer sound files store long arrays of sample values that can be manipulated programmatically.
Acoustics, the physics of sound Sounds are waves of air pressure Sound comes in cycles The frequency of a wave is the number of cycles per second (cps), or Hertz (Complex sounds have more than one frequency in them.) The amplitude is the maximum height of the wave
Volume and pitch:
Psychoacoustics, the psychology of sound Our perception of volume is related (logarithmically) to changes in amplitude If the amplitude doubles, its about a 3 decibel (dB) change Our perception of pitch is related (logarithmically) to changes in frequency Higher frequencies are perceived as higher pitches We can hear between 5 Hz and 20,000 Hz (20 kHz) A above middle C is 440 Hz
Lectures Page 1 Logarithmically? Its strange, but our hearing works on ratios not differences, e.g., for pitch. We hear the difference between 200 Hz and 400 Hz, as the same as 500 Hz and 1000 Hz Similarly, 200 Hz to 600 Hz, and 1000 Hz to 3000 Hz
Intensity (volume) is measured as watts per meter
squared A change from 0.1W/m2 to 0.01 W/m2, sounds the same to us as 0.001W/m2 to 0.0001W/m2
Decibel is a logarithmic measure A decibel is a ratio between two intensities: 10 * log10(I1/I2) As an absolute measure, its in comparison to threshold of audibility 0 dB cant be heard. Normal speech is 60 dB. A shout is about 80 dB
Digitizing Sound: How do we
get that into numbers? Remember in calculus, estimating the curve by creating rectangles? We can do the same to estimate the sound curve Analog-to-digital conversion (ADC) will give us the amplitude at an instant as a number: a sample
Lectures Page 2 us the amplitude at an instant as a number: a sample How many samples do we need?
Nyquist Theorem We need twice as many samples as the maximum frequency in order to represent (and recreate, later) the original sound. The number of samples recorded per second is the sampling rate If we capture 8000 samples per second, the highest frequency we can capture is 4000 Hz Thats how phones work If we capture more than 44,000 samples per second, we capture everything that we can hear (max 22,000 Hz) CD quality is 44,100 samples per second
Digitizing sound in the computer
Each sample is stored as a number (two bytes)
Whats the range of available combinations? 16 bits, 216 = 65,536 But we want both positive and negative values To indicate compressions and rarefactions. What if we use one bit to indicate positive (0) or negative (1)? That leaves us with 15 bits 15 bits, 215 = 32,768 One of those combinations will stand for zero Well use a positive one, so thats one less pattern for positives
Lectures Page 3 +/- 32K Each sample can be between -32,768 and 32,767
Why such a bizarre number?
Because 32,768 + 32,767 + 1 = 216
<0 >0 0 i.e. 16 bits, or 2 bytes
Compare this to 0..255 for light intensity
(i.e. 8 bits or 1 byte)
Sounds as arrays Samples are just stored one right after the other in the computers memory (Like pixels in a picture) Thats called an array Its an especially efficient (quickly accessed) memory structure
Manipulating Sounds
1. Get a filename of a WAV file and make an object
from it. 2. To work with samples, we next get the samples of
Lectures Page 4 2. the sound as an array. The array is of object class SoundSample. Analagous to the Pixel array result from getPixels() on a Picture. 3. Perform the desired manipulations on the sample(s) 4. Perhaps write the sound back out to a new file.
