Frequency modulation (FM) is the encoding of information in a carrier wave by varying the instantaneous frequency of the wave. The technology is used in telecommunications, radio broadcasting, signal processing, and computing.
In electronics and telecommunications, modulation is the process of varying one or more properties of a periodic waveform, called the carrier signal, with a separate signal called the modulation signal that typically contains information to be transmitted. For example, the modulation signal might be an audio signal representing sound from a microphone, a video signal representing moving images from a video camera, or a digital signal representing a sequence of binary digits, a bitstream from a computer.
In electronics, an analog-to-digital converter is a system that converts an analog signal, such as a sound picked up by a microphone or light entering a digital camera, into a digital signal. An ADC may also provide an isolated measurement such as an electronic device that converts an analog input voltage or current to a digital number representing the magnitude of the voltage or current. Typically the digital output is a two's complement binary number that is proportional to the input, but there are other possibilities.
Delta modulation is an analog-to-digital and digital-to-analog signal conversion technique used for transmission of voice information where quality is not of primary importance. DM is the simplest form of differential pulse-code modulation (DPCM) where the difference between successive samples is encoded into n-bit data streams. In delta modulation, the transmitted data are reduced to a 1-bit data stream representing either up (↗) or down (↘). Its main features are:
In signal processing, distortion is the alteration of the original shape of a signal. In communications and electronics it means the alteration of the waveform of an information-bearing signal, such as an audio signal representing sound or a video signal representing images, in an electronic device or communication channel.
Signal-to-noise ratio is a measure used in science and engineering that compares the level of a desired signal to the level of background noise. SNR is defined as the ratio of signal power to noise power, often expressed in decibels. A ratio higher than 1:1 indicates more signal than noise.
In wireless communications, fading refers to the variation of signal attenuation over variables like time, geographical position, and radio frequency. Fading is often modeled as a random process. In wireless systems, fading may either be due to multipath propagation, referred to as multipath-induced fading, weather, or shadowing from obstacles affecting the wave propagation, sometimes referred to as shadow fading.
A Dolby noise-reduction system, or Dolby NR, is one of a series of noise reduction systems developed by Dolby Laboratories for use in analog audio tape recording. The first was Dolby A, a professional broadband noise reduction system for recording studios in 1965, but the best-known is Dolby B, a sliding band system for the consumer market, which helped make high fidelity practical on cassette tapes, which used a relatively noisy tape size and speed. It is common on high-fidelity stereo tape players and recorders to the present day, although Dolby has as of 2016 ceased licensing the technology for new cassette decks. Of the noise reduction systems, Dolby A and Dolby SR were developed for professional use. Dolby B, C, and S were designed for the consumer market. Aside from Dolby HX, all the Dolby variants work by companding: compressing the dynamic range of the sound during recording, and expanding it during playback.
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.
In signal processing, sampling is the reduction of a continuous-time signal to a discrete-time signal. A common example is the conversion of a sound wave to a sequence of "samples". A sample is a value of the signal at a point in time and/or space; this definition differs from the term's usage in statistics, which refers to a set of such values.
This is an index of articles relating to electronics and electricity or natural electricity and things that run on electricity and things that use or conduct electricity.
Audio system measurements are a means of quantifying system performance. These measurements are made for several purposes. Designers take measurements so that they can specify the performance of a piece of equipment. Maintenance engineers make them to ensure equipment is still working to specification, or to ensure that the cumulative defects of an audio path are within limits considered acceptable. Audio system measurements often accommodate psychoacoustic principles to measure the system in a way that relates to human hearing.
Dither is an intentionally applied form of noise used to randomize quantization error, preventing large-scale patterns such as color banding in images. Dither is routinely used in processing of both digital audio and video data, and is often one of the last stages of mastering audio to a CD.
Noise shaping is a technique typically used in digital audio, image, and video processing, usually in combination with dithering, as part of the process of quantization or bit-depth reduction of a signal. Its purpose is to increase the apparent signal-to-noise ratio of the resultant signal. It does this by altering the spectral shape of the error that is introduced by dithering and quantization; such that the noise power is at a lower level in frequency bands at which noise is considered to be less desirable and at a correspondingly higher level in bands where it is considered to be more desirable. A popular noise shaping algorithm used in image processing is known as ‘Floyd Steinberg dithering’; and many noise shaping algorithms used in audio processing are based on an ‘Absolute threshold of hearing’ model.
FM broadcasting is a method of radio broadcasting that uses frequency modulation (FM) of the radio broadcast carrier wave. Invented in 1933 by American engineer Edwin Armstrong, wide-band FM is used worldwide to transmit high-fidelity sound over broadcast radio. FM broadcasting offers higher fidelity—more accurate reproduction of the original program sound—than other broadcasting techniques, such as AM broadcasting. It is also less susceptible to common forms of interference, having less static and popping sounds than are often heard on AM. Therefore, FM is used for most broadcasts of music and general audio. FM radio stations use the very high frequency range of radio frequencies.
Delta-sigma modulation is an oversampling method for encoding signals into low bit depth digital signals at a very high sample-frequency as part of the process of delta-sigma analog-to-digital converters (ADCs) and digital-to-analog converters (DACs). Delta-sigma modulation achieves high quality by utilizing a negative feedback loop during quantization to the lower bit depth that continuously corrects quantization errors and moves quantization noise to higher frequencies well above the original signal's bandwidth. Subsequent low-pass filtering for demodulation easily removes this high frequency noise and time averages to achieve high accuracy in amplitude which can be ultimately encoded as pulse-code modulation (PCM).
RIAA equalization is a specification for the recording and playback of phonograph records, established by the Recording Industry Association of America (RIAA). The purposes of the equalization are to permit greater recording times, to improve sound quality, and to reduce the groove damage that would otherwise arise during playback.
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.
Equalization, or simply EQ, in sound recording and reproduction is the process of adjusting the volume of different frequency bands within an audio signal. The circuit or equipment used to achieve this is called an equalizer.
Audio compact cassettes use magnetic tape of three major types which differ in fundamental magnetic properties, the level of bias applied during recording, and the optimal time constant of replay equalization. Specifications of each type were set in 1979 by the International Electrotechnical Commission (IEC): Type I, Type II, Type III, and Type IV. 'Type 0' was a non-standard designation for early compact cassettes that did not conform to IEC specification.
