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Amplitude modulation (AM) is a modulation technique used in electronic communication, most commonly for transmitting messages with a radio wave. In amplitude modulation, the amplitude of the wave is varied in proportion to that of the message signal, such as an audio signal. This technique contrasts with angle modulation, in which either the frequency of the carrier wave is varied, as in frequency modulation, or its phase, as in phase modulation.
The decibel is a relative unit of measurement equal to one tenth of a bel (B). It expresses the ratio of two values of a power or root-power quantity on a logarithmic scale. Two signals whose levels differ by one decibel have a power ratio of 101/10 or root-power ratio of 101⁄20.
The amplitude of a periodic variable is a measure of its change in a single period. The amplitude of a non-periodic signal is its magnitude compared with a reference value. There are various definitions of amplitude, which are all functions of the magnitude of the differences between the variable's extreme values. In older texts, the phase of a periodic function is sometimes called the amplitude.
Sound can be recorded and stored and played using either digital or analog techniques. Both techniques introduce errors and distortions in the sound, and these methods can be systematically compared. Musicians and listeners have argued over the superiority of digital versus analog sound recordings. Arguments for analog systems include the absence of fundamental error mechanisms which are present in digital audio systems, including aliasing and associated anti-aliasing filter implementation, jitter and quantization noise. Advocates of digital point to the high levels of performance possible with digital audio, including excellent linearity in the audible band and low levels of noise and distortion.
Audio normalization is the application of a constant amount of gain to an audio recording to bring the amplitude to a target level. Because the same amount of gain is applied across the entire recording, the signal-to-noise ratio and relative dynamics are unchanged. Normalization is one of the functions commonly provided by a digital audio workstation.
In digital and analog audio, headroom refers to the amount by which the signal-handling capabilities of an audio system can exceed a designated nominal level. Headroom can be thought of as a safety zone allowing transient audio peaks to exceed the nominal level without damaging the system or the audio signal, e.g., via clipping. Standards bodies differ in their recommendations for nominal level and headroom.
A class-D amplifier or switching amplifier is an electronic amplifier in which the amplifying devices operate as electronic switches, and not as linear gain devices as in other amplifiers. They operate by rapidly switching back and forth between the supply rails, using pulse-width modulation, pulse-density modulation, or related techniques to produce a pulse train output. This passes through a simple low-pass filter which blocks the high-frequency pulses and provides analog output current and voltage. Because they are always either in fully on or fully off modes, little energy is dissipated in the transistors and efficiency can exceed 90%.
Decibels relative to full scale is a unit of measurement for amplitude levels in digital systems, such as pulse-code modulation (PCM), which have a defined maximum peak level. The unit is similar to the units dBov and decibels relative to overload (dBO).
A peak programme meter (PPM) is an instrument used in professional audio that indicates the level of an audio signal.
Clipping is a form of waveform distortion that occurs when an amplifier is overdriven and attempts to deliver an output voltage or current beyond its maximum capability. Driving an amplifier into clipping may cause it to output power in excess of its power rating.
The loudness war is a trend of increasing audio levels in recorded music, which reduces audio fidelity and—according to many critics—listener enjoyment. Increasing loudness was first reported as early as the 1940s, with respect to mastering practices for 7-inch singles. The maximum peak level of analog recordings such as these is limited by varying specifications of electronic equipment along the chain from source to listener, including vinyl and Compact Cassette players. The issue garnered renewed attention starting in the 1990s with the introduction of digital signal processing capable of producing further loudness increases.
dBm0 is an abbreviation for the power in dBm measured at a zero transmission level point (ZLP).
Audio noise measurement is a process carried out to assess the quality of audio equipment, such as the kind used in recording studios, broadcast engineering, and in-home high fidelity.
The alignment level in an audio signal chain or on an audio recording is a defined anchor point that represents a reasonable or typical level.
In digital audio using pulse-code modulation (PCM), bit depth is the number of bits of information in each sample, and it directly corresponds to the resolution of each sample. Examples of bit depth include Compact Disc Digital Audio, which uses 16 bits per sample, and DVD-Audio and Blu-ray Disc which can support up to 24 bits per sample.
Clipping is a form of distortion that limits a signal once it exceeds a threshold. Clipping may occur when a signal is recorded by a sensor that has constraints on the range of data it can measure, it can occur when a signal is digitized, or it can occur any other time an analog or digital signal is transformed, particularly in the presence of gain or overshoot and undershoot.
In signal processing, control theory, electronics, and mathematics, overshoot is the occurrence of a signal or function exceeding its target. Undershoot is the same phenomenon in the opposite direction. It arises especially in the step response of bandlimited systems such as low-pass filters. It is often followed by ringing, and at times conflated with the latter.
Pulse-code modulation (PCM) is a method used to digitally represent sampled analog signals. It is the standard form of digital audio in computers, compact discs, digital telephony and other digital audio applications. In a PCM stream, the amplitude of the analog signal is sampled at uniform intervals, and each sample is quantized to the nearest value within a range of digital steps.
A white clipper is a circuit in professional video products that limits the maximum amplitude of the luminance part of the analogue video signal to 1 volt. It is essential for both analogue recording and transmission of video material.
Pulse-density modulation, or PDM, is a form of modulation used to represent an analog signal with a binary signal. In a PDM signal, specific amplitude values are not encoded into codewords of pulses of different weight as they would be in pulse-code modulation (PCM); rather, the relative density of the pulses corresponds to the analog signal's amplitude. The output of a 1-bit DAC is the same as the PDM encoding of the signal.
References
- ↑ "AES Standard » AES17-2015: AES standard method for digital audio engineering - Measurement of digital audio equipment". www.aes.org. Retrieved 2016-04-29.
amplitude of a 997-Hz sine wave whose positive peak value reaches the positive digital full scale, leaving the negative maximum code unused. NOTE In 2's-complement representation, the negative peak is 1 LSB away from the negative maximum code.
- ↑ "IEC 61606-3:2008 Audio and audiovisual equipment - Digital audio parts - Basic measurement methods of audio characteristics - Part 3: Professional use". International Electrotechnical Commission. 2008. Retrieved 2018-07-27.
amplitude of a 997 Hz sinusoid whose peak positive sample just reaches positive digital full-scale (in 2's-complement a binary value of 0111…1111 to make up the word length) and whose peak negative sample just reaches a value one away from negative digital full-scale (1000…0001 to make up the word length) leaving the maximum negative code (1000…0000) unused
- ↑ "Wave File Specifications". McGill University Telecommunications & Signal Processing Laboratory. Retrieved 2020-05-03.
For float data, full scale is 1.
- ↑ "Adobe Audition User Guide" (PDF).
32-bit 0.24 normalized float (type 3 – 32-bit) is the standard floating point format for type 3 .wav files. Values are normalized to the range +/-1.0
- ↑ Nielsen, Søren H.; Lund, Thomas. "0dBFS+ Levels in Digital Mastering" (PDF). TC Electronic A/S. Denmark.