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What is Pulse Code Modulation? – Unlocking Sound’s Digital Journey

by Stuart Charles Black
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Stuart Charles here, HomeStudioBasics.com helping YOU make sound decisions, so…

If you’re into audio, buy gear, or read spec sheets, oftentimes you’ll see the term “PCM” thrown around to indicate the types of files that an Amp/DAC can handle.

If you’re anything like me, you probably wondered what the heck it all means, basil!

Well, I’ll tell you what it means for the low low price of JUST KIDDING.

Stick around and let’s talk about it.

What is Pulse Code Modulation?

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K5 Pro playing back a standard PCM file.

Pulse Code Modulation (PCM) is a digital representation technique used in telecommunications and digital audio applications to convert analog signals into digital form.

It involves the sampling and quantization of an analog signal to create a discrete, binary representation that can be easily processed, transmitted, and stored using digital technology.

Here’s a breakdown of the key components of Pulse Code Modulation:


The continuous analog signal is sampled at regular intervals to capture its amplitude at specific points in time.

These samples are taken at a rate known as the sampling frequency.

The Nyquist-Shannon sampling theorem states that to accurately reconstruct an analog signal from its samples, the sampling frequency should be at least twice the highest frequency present in the original signal.

For instance, if you have a 24-bit/96kHz file, the actual sample rate is only 48kHz.


Each sample’s amplitude is assigned a digital value using quantization.

Quantization involves dividing the range of possible amplitude values into a finite number of discrete levels.

The number of quantization levels determines the bit depth of the digital representation.

Common bit depths include 8-bit, 16-bit, and 24-bit.

A higher bit depth allows for a more accurate representation of the original analog signal.


If you’re having trouble picturing quantization, think of it like sorting candies into different jars.

Imagine you have a bunch of candies with different sweetness levels, from not sweet at all to super-duper sweet.

Quantization is like deciding how many jars you want to use and putting candies in them based on their sweetness.

You choose a certain number of jars, let’s say 8 (for 8-bit), and each jar represents a different level of sweetness.

The trick is that you can only put each candy in one jar, and you have to pick the closest jar that matches how sweet the candy is.

So, if you have a not-so-sweet candy, you put it in the jar for “Not Very Sweet.” If you have a really, really sweet candy, you put it in the “Super-Duper Sweet” jar.

In the same way,

quantization is about grouping the different sound levels (like loud or quiet) into specific “jars” or levels.

It’s like making categories for sound strengths and assigning each sound to the closest category.

This helps turn the continuous range of sound levels into distinct, manageable groups that computers can understand.


The quantized values are then encoded into binary code (remember: computers only understand in binary), creating a series of binary numbers that represent the amplitudes of the sampled signal.

These binary values are typically stored in digital formats like WAV (Waveform Audio File Format) or AIFF (Audio Interchange File Format) for audio applications.


To play back the digital audio signal, the binary values are decoded.

The process involves converting the binary values back into quantized amplitudes and then reconstructing the continuous analog signal using these amplitudes.

This reconstructed signal can be played through speakers or headphones to reproduce the original sound.

PCM is a fundamental technique in digital audio processing and transmission.

It provides a way to accurately capture and reproduce audio signals using digital technology, making it suitable for various applications such as voice communication, music production, digital audio recording, and broadcasting.

It forms the basis for many audio file formats and is an essential component in the design of audio codecs and digital audio interfaces.


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K7 playing back a DSD file (Green)

By now you may be wondering what the difference is.

DSD (Direct Stream Digital) and PCM (Pulse Code Modulation) are two distinct methods for encoding and representing audio signals in the digital domain.

They have different underlying principles and characteristics, which lead to notable differences in terms of sound quality, storage requirements, and usage.

Here’s a comparison of DSD and PCM:

Encoding Principle


DSD uses a one-bit sigma-delta modulation technique.

Instead of representing the audio signal with a series of discrete amplitude levels (jar analogy as in PCM), DSD encodes the signal’s information as a sequence of very high-frequency pulses.

The density of these pulses represents the analog signal’s amplitude variations.

So the density of the DSD pulses represents either 1 or 0.

  1. When the analog signal goes up, it’s like saying “1,” and DSD sends a pulse.
  2. When the analog signal goes down, it’s like saying “0,” and DSD doesn’t send a pulse.

So, by carefully timing these pulses based on whether the signal goes up or down, DSD is able to encode the audio information as a sequence of “1”s and “0”s.

This binary sequence represents the changes in the sound’s amplitude over time.

Remember, DSD is unique in this approach compared to other methods like PCM, where the amplitude is directly quantized into specific numerical values.

DSD’s method allows it to capture the audio signal’s details using very high-frequency pulses and binary codes.


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Direct Stream Digital (DSD) operates akin to a painter using pointillism.

Instead of blending colors on a palette, DSD captures audio by representing sound as a series of discrete points, much like individual dots in a painting.

Each dot, or sample, holds precise information about the sound at that exact moment, forming a sequence that, when combined, creates a detailed audio image.

Like a pointillist artwork revealing depth and texture through meticulously placed dots, DSD reproduces sound by focusing on the subtleties of each individual sample rather than blending them like traditional audio formats.


PCM samples the analog signal at regular intervals and quantizes each sample’s amplitude into a binary value using a fixed bit depth (such as 16-bit or 24-bit).

The resulting binary values are then encoded as digital data.

Bit Depth and Sampling Rate



DSD typically operates at a high sampling rate (e.g., DSD64 at 2.8224 MHz, DSD128 at 5.6448 MHz) but uses only a single bit per sample.

This means that the quantization levels are limited to just two values: 0 and 1.


PCM can have various bit depths (e.g., 16-bit, 24-bit) and a range of sampling rates.

Higher bit depths allow for finer quantization levels and potentially greater dynamic range.

Sound Quality


DSD is often praised for its natural and detailed sound, particularly in higher frequencies.

Its simple encoding process can lead to less quantization noise, potentially resulting in a smoother and more analog-like sound.

But does it sound better than high PCM? Well, that’s up for debate.


PCM can also provide excellent sound quality, especially when using higher bit depths and sampling rates.

The quantization noise in PCM is typically lower with higher bit depths, leading to improved dynamic range and signal-to-noise ratio.

Editing and Processing


DSD signals are challenging to edit and process directly due to their one-bit nature.

Converting DSD signals to PCM for editing and then back to DSD for playback is a common approach.


PCM signals are easily editable and can undergo various digital processing techniques without requiring complex conversions.

Storage and Bandwidth


DSD files can be large due to their high sampling rates.

They require significant storage space and can strain playback systems with limited processing capabilities.


PCM files’ size depends on the bit depth and sampling rate but can be more manageable than DSD files for equivalent audio quality.


    • DSD: DSD support may be less widespread in consumer audio equipment and software than PCM.
    • PCM: PCM is the standard format for most digital audio content, making it highly compatible with various devices and software applications.

The Bottom Line

In summary, both DSD and PCM have their strengths and weaknesses.

DSD is praised for its potential to capture intricate musical nuances, especially in high-frequency content, while PCM offers wider compatibility, editing flexibility, and efficient storage.

The choice between DSD and PCM often depends on personal preferences, playback equipment, and specific use cases.

Well, that’s about it for today folks! I hope you’ve enjoyed this article on What is Pulse Code Modulation? and gained some valuable insight.

Questions? Comments? Requests? Did I miss the mark on something? Please let me know down below or Contact me!!

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All the best and God bless,





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