Print This Page

7. Level issues

All analogue and acoustic inputs and outputs - or terminals - of an audio device can accept a certain maximum level input signal - above this level the device’s electronic circuit or mechanical construction connected to the terminal will clip. Also, the connected electronic circuits possess a noise floor that is independent from the signal level flowing through them.

As clipping of terminals cause unintended distortion, and noise floors limit the audio signal in the level dimension, system designers and sound engineers spend a considerable amount of their working time to manage the levels in audio systems to produce the highest dynamic range, and to make sure the signal never causes a terminal to clip. Also, signal chain levels must be constantly monitored and controlled to prevent clipping in case the SPL of the sound source or the output level of one of the system’s processes is higher than anticipated.

7.1 0dBFS

An audio system comprising of two or more audio devices possesses four or more terminals, each with their own clip level and noise floor. A signal in such a system passes all terminals on its route from the system input to the system output - risking to clip each terminal, and also picking up all noise levels. The ratio of the system’s lowest clip level and the accumulated noise level constitutes the system’s dynamic range or clip to noise ratio. There are three basic methods to route a signal through the successive devices in an audio system: random level, matched noise floors and matched clip levels - or 0dBFS - where FS stands for Full Scale: the highest level a terminal can handle without clipping. Figure 702 presents the three methods for a system of two devices, displaying the dynamic range of each terminal as a gray bar with the top representing the terminal’s clip level and the bottom representing the terminal’s noise floor. The resulting dynamic range is the ratio between the lowest clip level in the signal path and the accumulated noise floor at the last terminal.

Figure 702 shows that when a random level alignment is used to route a signal through the audio system (which is any configuration other than matched noise floors or 0dBFS), the system’s dynamic range is the smallest because the dynamic range of individual terminals always overlaps with others. Matching noise floors minimises overlapping - resulting in a higher dynamic range, but it has the risk that the dynamic range of the last two devices in the signal path - the power amplifier and the speaker - are not always optimally used, causing higher costs. The 0dBFS method has the highest dynamic range because the noise floors are kept at the lowest level at each terminal - resulting in the lowest accumulated noise floor. Also, the 0dBFS method offers the lowest risk of clipping because all terminal clipping levels are aligned to the same level. Only the first terminal, which is normally the microphone’s acoustic input, can clip as a result of excessive input SPL - all other terminals can never clip (at unity gain). The conclusion is that the 0dBFS method is the optimal strategy to make the most use of a system’s dynamic range with the lowest risk of clipping and the lowest cost.

Assuming that the digital part of a networked audio system is always referenced to 0dBFS, there are five main parameters to align the system’s acoustical and analogue terminals: the selection of the microphones, power amplifiers and speakers, the setting of the analogue gain of the head amp, and the voltage gain of the power amplifier.

Microphone selection

The microphones must be suited to handle the SPL of the sound source - with appropriate directional characteristics to best fit the sound source and environment characteristics, and of course the preferred Response (‘sound’).

Speaker selection

At the other end of the system, the loudspeakers must be suited to generate the required amount of SPL at the listener’s position - and of course to have the preferred Response. To support electro-acoustic design, loudspeaker manufacturers specify the sensitivity of the speaker - being the SPL delivered to a listening position of 1 meter with an input of 1W at nominal impedance at 1kHz, allowing calculation of SPL at any distance using software such as EASE. Also, a maximum SPL is specified - often using the AES2-1984 (R2003) recommended practice for the specification of loudspeaker components(*7A). Over-specification of loudspeakers leads to unused power and thus high costs, while under-specification of loudspeakers leads to unintended harmonic distortion that can not be classified as Response anymore - and of course the potential failure of the speaker.

power amplifier selection

The selection of amplifiers must support the power requirements of the selected speakers. Similar to the selection criteria for speakers, over-specification of amplifiers lead to unused power and high noise floors, while underspecification of amplifiers lead to unintended distortion that can not be classified as Response - and of course the potential failure of the amplifier or speaker.

HA gain setting

The head amp gain allows the microphone’s peak output voltage - generated by the sound source peak SPL - to match the A/D converter’s peak input voltage. This sets the microphone input SPL and the microphone output voltage to match the system’s 0dBFS reference - creating the maximum dynamic range and eliminating the possibility of the A/D converter’s analogue input to clip.

Power amplifier gain setting

The power amplifier gain allows the speaker’s maximum voltage input - matched to the speaker’s maximum SPL output by the sensitivity and peak SPL specifications - to be aligned with the D/A converter’s peak output at the system’s 0dBFS reference - creating the maximum dynamic range and eliminating the possibility of the power amplifier and speaker to clip.

>>7.2 Head amps

Return to Top