This is a python based project !!!!
In this project, you will be filling the shoes of an audio technician and explore the world of digital sound. You will be learning about what sound is, how it can be represented in the computer, and how to transform and modify the digital representation of sound waves in order to create your own audio files (in .wav format).
Use and manipulate arrays/lists with numeric data
Practice different loop structures
Learn about what sound is and how it is represented and managed in the .wav format
Become familiar with basic principles of digital sound
Background: Sound and Digital Encoding
In the physical world, sound exists as fluctuations of pressure in the air around [url removed, login to view] fluctuations are continuous fluctuations over time that take on a continuous range of values, which is referred to as an analog signal. Any periodic analog signal can be decomposed into (and is thus the sum of) a (possibly infinite) number of sinusoidal waves. The simplest pure sound is a single sinusoidal wave, defined by the sine function (see Sine function)
Computers, however, cannot easily represent continuous functions because they have a finite amount of memory. Instead, an approximate representation is used, by taking sample values of the continuous function at discrete time intervals. Thus, although we don't have the original wave, we have samples of where the wave was for a number of points in time. This is referred to as a digital signal. A digital-to-analogue converter can reconstruct the original wave from samples, which can in turn be played on speakers.
The rate at which samples are taken is known as the sampling frequency. It is measured in Hertz (Hz), or samples per second. The sample size is the number of digits used in the digital representation of each sample. Higher sampling frequency and higher sample size lead to higher quality sound.
An analogue-to-digital converter (ADC) takes an analog signal as an input and produces a digital signal as output. Such a device is used in a microphone. As you speak or play into a microphone, your voice causes a membrane to vibrate. It is recorded how far the membrane is displaced in either direction, many times per second. These samples may be saved easily for playback at a later time. A digital-to-analog converter (DAC) performs a transformation that is the the inverse of the one performed by the ADC. Namely, a DAC takes discrete samples in time and constructs a continuous wave from them. You may find such a device in your sound card or in your CD player. Whenever you listen to a sound on your computer, these discrete samples are converted to a continuous signal that can be fed to your speakers.
It is possible to exactly reconstruct an analogue signal from digital samples subject to a few conditions. When recording an analog signal it is important to know the highest frequency that will be captured. This is known as the Nyquist frequency. The Nyquist-Shannon theorem states that it is possible to exactly reconstruct a wave consisting of frequencies up to the Nyquist frequency if the sampling frequency is at least twice as large.
The most common sampling frequencies found on audio hardware are currently 11025 Hz, 22050 Hz, and 44100 Hz. Since the average human ear can only hear a maximum frequency of up to 20000 Hz, we should set our sampling rate to at least 40000Hz. Sampling at a higher frequency only results in more samples that can capture sounds that are inaudible to humans.
.wav File Format