Analog and digital sound recordings
As you learned in the previous section, sound waves change air pressure. These changes are the basis for recording sound with analogue or digital methods. Both methods require a microphone that translates the changes in air pressure into electrical signals.
In a dynamic microphone like the one shown above, sound waves enters through the microphone head, and the changing air pressure forces the diaphragm and metal coil to move back and forth past the magnet. This creates a varying voltage in the coil: an electrical signal. Analogue and digital recording methods process this signal in different ways.
Analogue recording through the ages
Analogue recording methods store the electrical signal directly in some form of physical medium.
Analogue recording dates back to 1877 when Thomas Edison, creator of the lightbulb, invented a mechanical machine called the phonograph, which produced and recorded sound.
Ten years later, Emile Berliner created the gramophone to improve upon the storage capability of the phonograph, which wore out its storage media very quickly. The storage medium Berliner invented was similar to a vinyl record!
These inventions could store sound recordings, but because storage media were physically changed during the recording process, the recordings were nearly impossible to edit. In addition, playing back sound also changed the storage media, so each storage medium was viable for only a limited number of playbacks before it was too damaged to read.
In the 20th century, Fritz Pfleumer’s invention, the magnetophone, overcame these limitations by using magnetic tape as a storage medium. The magnetophone encoded the electrical signals from the microphone via a process of magnetisation; reading the magnetic tape to play back sound did not damage the tape. In addition, tapes could be edited through the process of splicing: a metal blade cut the tape, and sections of tape are joined together.
An analogue sound recording is stored safely on a mechanical device, disk, or magnetic tape, and we can easily convert the analogue recording back into sound. Computers, however, don’t understand analogue and need another way to process sound.
Digital sound recording
In the electrical signal from a microphone, the voltage changes depending upon changes in air pressure (i.e. the sound waves). A computer converts these changes in voltage into a digital signal using an analogue-to-digital converter or ADC. And the digital signal is stored in the form of binary numbers.
Possible classroom project: you can easily create an ADC by adding a small chip to a Raspberry Pi.
The resulting binary numbers are available for the computer to:
- Edit: no need to splice sections of tape anymore: the binary numbers can be directly manipulated (a variety of software programs let you do this very easily, e.g. Audacity)
- Play back: sending the binary numbers directly to the loudspeaker would be like sending a stream of ‘loudspeaker on’ and ‘loudspeaker off’ commands — this would not be nice to listen to! So when you want to play sound back, the computer converts the binary numbers back to an analogue signal using a digital-to-analogue converter (DAC) before sending it to the loudspeaker.
You may have heard the term ‘high fidelity’ in the context of stereos or CD players, where it’s often abbreviated as hi-fi. The term refers to the quality at which sound is recorded or played back. To produce enough data for a high-fidelity digital sound recording, the computer needs to check the electrical signal coming from the microphone many thousands of times per second. This process of checking is known as sampling, and you will learn about it in the next step, where we’ll look more closely at how digital sound recording works.