Digital Sound |
![]() ![]() ![]() |
This page explains some of the concepts associated with the digital reproduction of sound. Amplitude The height of the wave. This value determines the loudness of the sound. Higher amplitude waves are represented by higher output voltages from the sound card. Frequency (or pitch) Frequency is the number of complete waves that pass a point each second. The unit of measurement for frequency is the Hertz (Hz). One Hertz equals one wave cycle per second. 1kHz is equal to 1000Hz. The range of human hearing is between approximately 20 to 20,000 Hz.
Wavelength The distance before a wave repeats itself. The wavelength is inversely proportional to the frequency. The higher the frequency the shorter the wavelength. The wavelength is also determined by the velocity of the wave. Sample rate In order to store a waveform in a digital format, the waveform must be converted into a series of numbers. This is done via a process called sampling. The amplitude of the sound is measured thousands of times per second in order to digitize the sound. The higher the sample rate the more accurate the digital representation of the sound will be. Higher sample rates also mean that higher frequencies can be digitized. This is due to the Nyquist theory. Nyquist theory states that a waveform must be sampled twice in order to get a true representation. Thus a regular CD sampled at 44.1kHz is theoretically capable of reproducing frequencies up to 22kHz. It is therefore important to take into consideration the highest frequency of the audio material to be recorded. If a frequency of 14,000 Hz is to be recorded, a sample rate of 44.1 kHz would be the logical choice to use. 14,000 Hz falls within the Nyquist range for a 44.1 kHz sample rate, which is 0 Hz to 22.05 kHz. While a higher sample rate is desirable, the amount of RAM required to store the sample will be larger and the ability of a computer to process the signal in real time is decreased. High sample rates are better at capturing high frequency waveforms, but if you are sampling lower frequency sounds, such as drums, bass, etc., you might consider sampling at a lower rate to save space. Bits (resolution) The more bits used to describe sound or video, the better the clarity and fidelity. An 8 bit sample contains 256 steps, while a 16 bit sample contains 65,536 steps. Obviously a 16 bit sample will have much greater definition. Each of these numbers represents a different analog signal voltage. The bit resolution of a system defines the theoretical dynamic range of the system. 6dB is gained for every bit. For example, 8 bits equals 256 states = 48 dB, 16 bits equals 65,536 states = 96 dB. In practice the background noise level can limit the real dynamic range. Decibels (dB) A logarithm scale of measurement, used to measure relative loudness levels for sound. Stereo / Mono (channels) Almost all sound cards have the ability to record and playback two streams of data at the same time. (Stereo). If you are using a mono microphone, there is no point selecting stereo as the input will be the same for both of the channels. Some newer sound cards can record and playback even more channels than the standard 2. These additional channels are not currently supported by SoundCheck. PCM Coding Pulse Code Modulation (PCM) is the standard used internally to transfer sound data from the sound card to the main memory of the computer. PCM is effectively the raw data stream of samples, broken up into channels, without any compression having been applied. Spectrum A spectrum is a continuous range of sound frequencies. |