Companding

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A signal before (top) and after m-law compression (bottom) Demo of mu-law compression.svg
A signal before (top) and after μ-law compression (bottom)

In telecommunications and signal processing, companding (occasionally called compansion) is a method of mitigating the detrimental effects of a channel with limited dynamic range. The name is a portmanteau of the words compressing and expanding, which are the functions of a compander at the transmitting and receiving ends, respectively. The use of companding allows signals with a large dynamic range to be transmitted over facilities that have a smaller dynamic range capability. Companding is employed in telephony and other audio applications such as professional wireless microphones and analog recording.

Contents

How it works

The dynamic range of a signal is compressed before transmission and is expanded to the original value at the receiver. The electronic circuit that does this is called a compander and works by compressing or expanding the dynamic range of an analog electronic signal such as sound recorded by a microphone. One variety is a triplet of amplifiers: a logarithmic amplifier, followed by a variable-gain linear amplifier, and ending with an exponential amplifier. Such a triplet has the property that its output voltage is proportional to the input voltage raised to an adjustable power.

Companded quantization is the combination of three functional building blocks – namely, a (continuous-domain) signal dynamic range compressor, a limited-range uniform quantizer, and a (continuous-domain) signal dynamic range expander that inverts the compressor function. This type of quantization is frequently used in telephony systems. [1] [2]

In practice, companders are designed to operate according to relatively simple dynamic range compressor functions that are suitable for implementation as simple analog electronic circuits. The two most popular compander functions used for telecommunications are the A-law and μ-law functions.

Applications

Companding is used in digital telephony systems, compressing before input to an analog-to-digital converter, and then expanding after a digital-to-analog converter. This is equivalent to using a non-linear ADC as in a T-carrier telephone system that implements A-law or μ-law companding. This method is also used in digital file formats for better signal-to-noise ratio (SNR) at lower bit depths. For example, a linearly encoded 16-bit PCM signal can be converted to an 8-bit WAV or AU file while maintaining a decent SNR by compressing before the transition to 8-bit and expanding after conversion back to 16-bit. This is effectively a form of lossy audio data compression.

Professional wireless microphones do this since the dynamic range of the microphone audio signal itself is larger than the dynamic range provided by radio transmission. Companding also reduces the noise and crosstalk levels at the receiver. [3]

Companders are used in concert audio systems and in some noise reduction schemes.

History

The use of companding in an analog picture transmission system was patented by A. B. Clark of AT&T in 1928 (filed in 1925): [4]

In the transmission of pictures by electric currents, the method which consists in sending currents varied in a non-linear relation to the light values of the successive elements of the picture to be transmitted, and at the receiving end exposing corresponding elements of a sensitive surface to light varied in inverse non-linear relation to the received current.

— A. B. Clark patent

In 1942, Clark and his team completed the SIGSALY secure voice transmission system that included the first use of companding in a PCM (digital) system. [5]

In 1953, B. Smith showed that a nonlinear DAC could be complemented by the inverse nonlinearity in a successive-approximation ADC configuration, simplifying the design of digital companding systems. [6]

In 1970, H. Kaneko developed the uniform description of segment (piecewise linear) companding laws that had by then been adopted in digital telephony. [7]

In the 1980s and 1990s, many of the music equipment manufacturers (Roland, Yamaha, Korg) used companding when compressing the library waveform data in their digital synthesizers. However, exact algorithms are unknown, neither if any of the manufacturers ever used the Companding scheme which is described in this article. The only known thing is that manufacturers did use data compression [8] in the mentioned time period and that some people refer to it as "companding" while in reality it might mean something else, for example data compression and expansion. [9] This dates back to the late '80s when memory chips were often one of the most costly components in the instrument. Manufacturers usually quoted the amount of memory in its compressed form: i.e. 24 MB of physical waveform ROM in a Korg Trinity is actually 48 MB when uncompressed. Similarly, Roland SR-JV expansion boards were usually advertised as 8 MB boards with '16 MB-equivalent content'. Careless copying of this technical information, omitting the "equivalence" reference, can often cause confusion.

Related Research Articles

An analog signal or analogue signal is any continuous-time signal representing some other quantity, i.e., analogous to another quantity. For example, in an analog audio signal, the instantaneous signal voltage varies continuously with the pressure of the sound waves.

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.

<span class="mw-page-title-main">A-law algorithm</span> Algorithm Global 64-bit PCM

An A-law algorithm is a standard companding algorithm, used in European 8-bit PCM digital communications systems to optimize, i.e. modify, the dynamic range of an analog signal for digitizing. It is one of the two companding algorithms in the G.711 standard from ITU-T, the other being the similar μ-law, used in North America and Japan.

<span class="mw-page-title-main">Delta modulation</span> Signal conversion technique

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:

μ-law algorithm Audio companding algorithm

The μ-law algorithm is a companding algorithm, primarily used in 8-bit PCM digital telecommunications systems in North America and Japan. It is one of the two companding algorithms in the G.711 standard from ITU-T, the other being the similar A-law. A-law is used in regions where digital telecommunication signals are carried on E-1 circuits, e.g. Europe.

<span class="mw-page-title-main">Repeater</span> Relay station

In telecommunications, a repeater is an electronic device that receives a signal and retransmits it. Repeaters are used to extend transmissions so that the signal can cover longer distances or be received on the other side of an obstruction. Some types of repeaters broadcast an identical signal, but alter its method of transmission, for example, on another frequency or baud rate.

<span class="mw-page-title-main">Time-division multiplexing</span> Multiplexing technique for digital signals

Time-division multiplexing (TDM) is a method of transmitting and receiving independent signals over a common signal path by means of synchronized switches at each end of the transmission line so that each signal appears on the line only a fraction of time according to agreed rules, e.g. with each transmitter working in turn. It can be used when the bit rate of the transmission medium exceeds that of the signal to be transmitted. This form of signal multiplexing was developed in telecommunications for telegraphy systems in the late 19th century but found its most common application in digital telephony in the second half of the 20th century.

<span class="mw-page-title-main">Digital audio</span> Technology that records, stores, and reproduces sound

Digital audio is a representation of sound recorded in, or converted into, digital form. In digital audio, the sound wave of the audio signal is typically encoded as numerical samples in a continuous sequence. For example, in CD audio, samples are taken 44,100 times per second, each with 16-bit resolution. Digital audio is also the name for the entire technology of sound recording and reproduction using audio signals that have been encoded in digital form. Following significant advances in digital audio technology during the 1970s and 1980s, it gradually replaced analog audio technology in many areas of audio engineering, record production and telecommunications in the 1990s and 2000s.

<span class="mw-page-title-main">Digital-to-analog converter</span> Device that converts a digital signal into an analog signal

In electronics, a digital-to-analog converter is a system that converts a digital signal into an analog signal. An analog-to-digital converter (ADC) performs the reverse function.

<span class="mw-page-title-main">G.711</span> ITU-T recommendation

G.711 is a narrowband audio codec originally designed for use in telephony that provides toll-quality audio at 64 kbit/s. It is an ITU-T standard (Recommendation) for audio encoding, titled Pulse code modulation (PCM) of voice frequencies released for use in 1972.

<span class="mw-page-title-main">Dynamic range compression</span> Audio signal processing operation

Dynamic range compression (DRC) or simply compression is an audio signal processing operation that reduces the volume of loud sounds or amplifies quiet sounds, thus reducing or compressing an audio signal's dynamic range. Compression is commonly used in sound recording and reproduction, broadcasting, live sound reinforcement and some instrument amplifiers.

<span class="mw-page-title-main">Sound quality</span> Assessment of the audio output from an electronic device

Sound quality is typically an assessment of the accuracy, fidelity, or intelligibility of audio output from an electronic device. Quality can be measured objectively, such as when tools are used to gauge the accuracy with which the device reproduces an original sound; or it can be measured subjectively, such as when human listeners respond to the sound or gauge its perceived similarity to another sound.

dbx (noise reduction) Family of noise reduction systems

dbx is a family of noise reduction systems developed by the company of the same name. The most common implementations are dbx Type I and dbx Type II for analog tape recording and, less commonly, vinyl LPs. A separate implementation, known as dbx-TV, is part of the MTS system used to provide stereo sound to North American and certain other TV systems. The company, dbx, Inc., was also involved with Dynamic Noise Reduction (DNR) systems.

Near Instantaneous Companded Audio Multiplex (NICAM) is an early form of lossy compression for digital audio. It was originally developed in the early 1970s for point-to-point links within broadcasting networks. In the 1980s, broadcasters began to use NICAM compression for transmissions of stereo TV sound to the public.

Continuously variable slope delta modulation is a voice coding method. It is a delta modulation with variable step size, first proposed by Greefkes and Riemens in 1970.

<span class="mw-page-title-main">Delta-sigma modulation</span> Method for converting signals between digital and analog

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).

<span class="mw-page-title-main">Wireless microphone</span> Microphone without a physical cable

A wireless microphone, or cordless microphone, is a microphone without a physical cable connecting it directly to the sound recording or amplifying equipment with which it is associated. Also known as a radio microphone, it has a small, battery-powered radio transmitter in the microphone body, which transmits the audio signal from the microphone by radio waves to a nearby receiver unit, which recovers the audio. The other audio equipment is connected to the receiver unit by cable. In one type the transmitter is contained within the handheld microphone body. In another type the transmitter is contained within a separate unit called a "bodypack", usually clipped to the user's belt or concealed under their clothes. The bodypack is connected by wire to a "lavalier microphone" or "lav", a headset or earset microphone, or another wired microphone. Most bodypack designs also support a wired instrument connection. Wireless microphones are widely used in the entertainment industry, television broadcasting, and public speaking to allow public speakers, interviewers, performers, and entertainers to move about freely while using a microphone without requiring a cable attached to the microphone.

<span class="mw-page-title-main">Audio bit depth</span> Number of bits of information recorded for each digital audio sample

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.

Adaptive differential pulse-code modulation (ADPCM) is a variant of differential pulse-code modulation (DPCM) that varies the size of the quantization step, to allow further reduction of the required data bandwidth for a given signal-to-noise ratio.

Pulse-code modulation (PCM) is a method used to digitally represent 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. Alec Reeves, Claude Shannon, Barney Oliver and John R. Pierce are credited with its invention.

References

  1. ↑ W. R. Bennett, "Spectra of Quantized Signals", Bell System Technical Journal , Vol. 27, pp. 446–472, July 1948.
  2. ↑ Robert M. Gray and David L. Neuhoff, "Quantization", IEEE Transactions on Information Theory , Vol. IT-44, No. 6, pp. 2325–2383, Oct. 1998. doi : 10.1109/18.720541
  3. ↑ A description of companding in wireless microphones
  4. ↑ USpatent,A. B. Clark,"Electrical picture-transmitting system",issued 1928-11-13, assigned to AT&T
  5. ↑ Randall K. Nichols and Panos C. Lekkas (2002). Wireless Security: Models, Threats, and Solutions . McGraw-Hill Professional. p.  256. ISBN   0-07-138038-8. companding a-b-clark pcm.
  6. ↑ B. Smith, "Instantaneous Companding of Quantized Signals," Bell System Technical Journal, Vol. 36, May 1957, pp. 653–709.
  7. ↑ H. Kaneko, "A Unified Formulation of Segment Companding Laws and Synthesis of Codecs and Digital Compandors," Bell System Technical Journal, Vol. 49, September 1970, pp. 1555–1558.
  8. ↑ "Gearspace - View Single Post - Why is My Hardware Sampler Sounding Better Than My Software Sampler!? Same Samples!." gearspace.com. Retrieved 25 October 2024.
  9. ↑ "Gearspace - View Single Post - Roland JV-1080 vst plugin". gearspace.com. Retrieved 25 October 2024.
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