What is Pulse Code Modulation (PCM)?

Pulse Code Modulation

Modulation is a process which involves changing one or more signal carrier parameters based on the current values ​​of the message signal.

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A message signal is a signal which is transmitted for communication, and a carrier signal is a high-frequency signal which has no data but which is used for long distance transmission.

There are several modulation techniques which are classified according to the type of modulation used. Among all these, the digital modulation technique is Pulse Code Modulation (PCM).

Pulse Code Modulation (PCM) is a digital display of an analogue signal that samples the amplitude of an analogue signal at regular intervals. The sampled analogue data is modified and then represented by binary data.

PCM requires an exact clock. The number of samples per second, which ranges from 8,000 to 192,000, is typically several times the maximum frequency of the analogue waveform in Hertz (Hz), or cycles per second, ranging from 8 to 192 kHz.

The word pulse refers to the pulses located on transmission lines, which are a natural consequence of two other analogue methods developed almost simultaneously: pulse width modulation and pulse position modulation, where each uses pulses of a discrete signal of different widths or positions.

Otherwise, PCM bears little resemblance to these other forms of signal coding. These methodologies were introduced in the United States in the early 1960s when telephone companies began converting voice to digital signals to enhance transmission between cities.

Each pattern in PCM is quantized, approximating an extensive set of possible values ​​to a relatively small set of values, which can be integers or even discrete symbols. No matter how complex, all analogue data can be digitized. This includes analogue data such as general videos, sound, telemetry, and virtual reality.

PCM data is raw digital audio samples. Audio files in formats like MP3 and AAC are first converted to PCM data. The PCM data is then converted to analogue speaker signals.

Post-processing of digital signal processors can create many data streams. These streams, in turn, can be multiplexed into more massive data streams which are transmitted very quickly over long distances by processes such as time-division multiplexing, frequency division multiplexing, etc. TDM is used more because of its natural compatibility with digital communication and lower bandwidth requirements.

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Once these data streams reach their destination, they are demultiplexed, redistributed into individual data streams, and demodulated, with the modulation procedure being applied in reverse to recreate the original binary numbers.

They are then processed to restore the original analogue waveform. In the process of changing from one sampling period to another, the signal receives significant high-frequency energy.

Analogue filters are used to equalize the signal and remove these unwanted frequencies, called alias frequencies. Depending on the requirement for accurate output signals, these analogue filters may or may not be required.

How PCM Works?

Pulse Code Modulation (PCM) devices receive analogue signals with continuously varying voltages and quantize these signals into respective voltages sampled at regular time intervals, typically 8000 times per second, each sample having 8 bits.

This allows a total transfer rate of 64 Kbps, as in digital telephone communication with an Integrated Digital Network Service (ISDN). The result of this quantization procedure is a series of discrete voltages over time.

The voltage levels correspond to powers of 2 and represent a series of binary numbers, so the output of the PCM device is essentially a binary number.

A typical PCM device consists of a sample-and-hold circuit that samples an analogue voltage signal and holds it long enough for the analogue-to-digital converter to convert it to a digital (binary) format.

A single device with associated software that can also perform analogue to digital conversion and vice versa is known as a codec or encoder/decoder.

Processing and Coding

Some forms of PCM integrate signal processing with coding. Previous versions of these systems applied analogue domain processing as part of the analogue-to-digital process; the newest implementations do so in the digital domain.

These simple techniques are largely obsolete by modern transform-based sound compression techniques, such as Modified Discrete Cosine Transform (MDCT) coding.

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  • Linear PCM (LPCM) is a linear quantization PCM.
  • Differential PCM (DPCM) encodes PCM values ​​as the difference between the current value and the expected value. The algorithm predicts the next sample based on the previous samples, and the encoder only stores the differentiation between this projection and the actual value. If the projection is reasonable, less bits can be used to represent the same information. For audio, this type of encoding decreases the number of bits required per sample by about 25% compared to PCM.
  • Adaptive DPCM (ADPCM) is a variation of DPCM that changes the size of the quantization steps to allow further reduction in the bandwidth required for a given signal-to-noise ratio.
  • Delta modulation is a form of DPCM which uses one bit per sample to indicate whether the signal is increasing or decreasing from the previous sample.

In telephony, the standard audio signal for a telephone call is encoded at 8,000 samples per second, 8 bits each, resulting in a 64 kbit / s digital signal called DS0.

The default signal compression encoding on DS0 is PCM μ-law (mu-law) (North America and Japan) or A-law PCM (Europe and most other countries in the world).

These are logarithmic compression systems in which a 12-bit or 13-bit linear PCM sample number is mapped to an 8-bit value. The international standard G.711 describes this system.

When circuit costs are high, and the loss of voice quality is acceptable, it sometimes makes sense to compress the voice signal further. The ADPCM algorithm is used to map a series of 8-bit μ-law or PCM A-law samples into a series of 4-bit ADPCM samples.

In this manner, the capacity of the line is doubled. The technique is described in detail in the G.726 standard.

Audio coding formats and audio codecs have been developed to achieve higher compression. Some of these techniques are standardized and patented.

Advanced compression techniques, such as MDCT and Linear Predictive Coding (LPC), are widely used today in mobile phones, voice over IP (VoIP) and broadcast media.


The word pulse in the term Pulse Code Modulation (PCM) refers to “pulses” located on a transmission line.

This may be a reasonable consequence of the fact that this technique was developed in conjunction with two analogue methods, pulse width modulation and pulse position modulation, in which the encoded data is represented by discrete signal pulses of variable width or position, respectively.

In this regard, PCM bears little resemblance to these other forms of signal coding, except that they can all be used in time-division multiplexing, and PCM code numbers are represented as electrical pulses.

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Pulse Code Modulation (PCM) is a method used to represent sampled analogue signals digitally. It is a standard form of digital audio in computers, compact discs, digital telephony, and other digital audio applications.

In the PCM stream, the amplitude of the analogue signal is regularly sampled at uniform intervals, and each sample is quantized to the closest value in the range of digital steps.

The PCM sequence has two basic properties that determine the fidelity of the chain to the original analogue signal: the sample rate, which is the number of times per second of sampling; and bit depth, which ascertain the number of possible digital values ​​that can be used to represent each sample.

Fadehan Emmanuel
Moore is an Associate—Editor of TheXplorion. Got a NEWS TIP related to this story — or to anything else in the world of big tech? Please e-mail him: mooreplug[at]gmail.com. You can also connect with him via the connections below the box.