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Sound is made when an object vibrates through a medium such as air. When sound reaches your ear, the vibrations in the molecules move the eardrum back and forth, and those vibrations are cinverted into signals that your brain understands as sound.

When we speak, vibrations, called sound waves, are created These sound waves have a recurring pattern called analog wave pattern The wave pattern has two attributes of sound on a computer: volume & frequency

Analog sound in the recording world works by capturing sounds through a device (like microphone and transmitting the sound by electrical means to another device like an amplifier. Digital storing sound in digital form involves converting the signals from analog means to information the computer can understand.

The height of each peak in the sound wave relates to its volume - the higher the peak, the louder the sound The distance between the peaks is the frequency (pitch) - the greater the distance, the lower the sound The amplitude is graphed as the height of the wave, and frequency is seen as the curviness of the lines.

Frequency is measured in hertz (Hz) A pattern that recurs every second is equal to one hertz

The pattern that recurs every second is equal to one hertz If the pattern recurs 1000 times/sec, it would be equal to 1000 Hz or 1 kHz (kilohertz)

For sound to be included in a multimedia application, the sound waves must be converted from analog (sound waves) to digital (bits) form This conversion is done through a process called sampling

Two factors that affect the quality of the digitized sound:

The number of times the sample is taken (sample rate) The amount of information stored about the sample (sample size)

The three most common sample rates:

11.025 kHz 22.05 kHz 44.1 kHz

The higher the sample rate, the more samples that are taken and thus the better the quality of the digitized sound

Two most common sample sizes:

8-bit sample allows for 256 values that are used to describe the sound 16-bit sample allows for 65,536 values

The higher the sample size, the better the quality of the digitized sound and the larger the file size

A sound card and software program are used to digitize sound A sound from an external source is sent to the card The card samples the sound based on the sample rate and bit value, and then produces the digital approximation of the analog signal

Once a sound has been digitized, it can easily be manipulated using a sound-editing program Sound files can be saved in several file formats

Popular audio file formats:

WAV - developed by Microsoft and IBM; used for audio on the Web; plays on both Windows & Mac computers AIFF (Audio Interchange File Format) and AIFC (AIFF Compressed) - developed by Apple; standard for Mac computers

 RealAudio (.ra) - a compression format developed by Real Networks that allows streaming of the sound file over the Web MP3 (MPEG Layer 3) - a format that compresses large audio files (entire songs) for playback on the Web at near stereo quality

 MIDI (Musical Instrument Digital Interface) - a standard format that enables computers and electronic musical instruments to communicate sound information

A primary consideration when using sound on the Web is file size Through a process called file compression, the size of a file can be reduced significantly File compression is automatically performed when audio files are saved using certain file formats. Compression can be lossless or lossy. With lossy compression, some of the sound quality is lost to achieve smaller file sizes. When using lossless compression, no quality in the audio is lost when reproducing the sound.

Sound files delivered via the Web are stored on a server A server is a computer with software that responds to other computers and makes data and programs available to them

Streaming media is a technology that allows audio and video to be played in real time A media player must be installed on a computer to stream sound files through the Web