The analog vs digital audio debate is very polarizing. I have noticed that most people defend one side or the other, allowing little room for nuance. The truth is, both analog and digital audio have drawbacks and advantages. I hope to offer an unbiased explanation of the differences in this post.
The difference between analog and digital audio is found in the way audio information is stored. Sound waves are a series of vibrations through a medium. Analog audio recording technology stores this information by creating a series of magnetic charges along a reel of magnetic tape. Digital audio technology stores audio information as a series of numeric values on a hard drive.
In this post, you will learn the pros and cons of each recording method along with the difference between analog and digital audio technologies for live sound applications.
The difference between analog and digital audio is found in the way audio information is stored. Analog audio recording technology stores this information by creating a series of magnetic charges along a reel of magnetic tape. Digital audio technology stores audio information as a series of numeric values on a hard drive.
The information in this post was written to be as easy-to-understand as possible. Some concepts in this article will make more sense if you have a basic understanding of how sound works. If you find any of the following sections confusing, feel free to reference this post I wrote on audio basics.
Analog vs Digital Audio: Recording and Playback
Before I begin discussing the differences between digital and analog audio systems, I think it’s important to mention that all digital audio systems include some analog audio technology. Microphones are analog audio devices that transduce acoustical energy into an analog electrical signal. Preamplifiers, power amplifiers, and loudspeakers are all analog devices, as well. The primary focus of this section will be to highlight the key differences in analog and digital recording technology.
Magnetic tape is the best analog audio method for recording and playback. Tape machines operate on the following principle: when an electric current is sent through a wire, a magnetic field forms around the wire, and vice versa.
To record audio, a tape machine sends electrical audio signals through a coiled wire surrounding a magnet which is held in close proximity to magnetic tape. This coil of wire surrounding the magnet is called the record head. As the tape passes through the magnetic field created by the record head, the particles along the tape are magnetically charged. The pattern of the magnetic charges along the tape resembles the audio signal sent through the coil of wire. The amplitude of the audio signal correlates with the magnitude of the magnetic charges created on the tape.
To playback the audio, the process is reversed. The magnetized tape creates an electric current on the play head, which connects to an amplifier to be played through speakers.
There are various types of tape and tape machines which affect the quality of the audio recorded. The two main variables are tape speed and tape width.
The rate at which the tape passes the record head affects the quality of the recording. A faster tape speed produces a recording with greater frequency response, less hiss, and shorter dropouts. Tape machine speed is measured in inches per second (ips). Common tape machine speeds are 7-½ ips, 15 ips, and 30 ips. The standard for professional recording is 15 ips.
The width of the tape also affects the quality of the recording. Wider tape allows for a higher-quality recording. However, tape width can be utilized to record more tracks rather than improving the audio quality of a single track. This allows several sources to be recorded and played back independently.
Vinyl records are the standard consumer medium for analog audio recordings. They are easier to maintain, store, and distribute. Compared to tape, vinyl records are less vulnerable to the elements. Whereas tape can be destroyed by magnetic exposure, vinyl records are immune to magnetic fields because they use a different means of audio storage. Rather than magnetic charge, the textured grooves on the surface of vinyl records store the audio information.
As a vinyl record spins at a specific rate, a stylus travels through the grooves on its surface. As the stylus moves back and forth with the grooves, it creates an electric current in a wire which connects to an amplifier to be played through speakers. The amplitude of the audio signal is correlated with the intensity in the movement of the stylus.
You can see an animation of how a vinyl record works by Animagraffs. Animagraffs is a website that creates amazing animations of various technologies.
Vinyl records are used only for playback in the modern world. Analog recordings are made with magnetic tape. The tapes are used to create casts for pressing the information to vinyl discs.
PCM (Pulse Code Modulation)
PCM, or Pulse Code Modulation, is the standard method for encoding audio signals into binary information. In analog audio recording, a model of the sound waves is created using magnetic charge. However, PCM creates a model of the sound waves by storing a sequence of numerical values that represent the amplitude at various points along a wave.
These values are represented by groups of binary bits, called samples. Each sample represents a numerical value within a predetermined range of possible values. This process is called quantization, and is performed by an analog-to-digital converter (A-to-D converter).
During playback of a digital recording, the samples are converted back to electrical signals and sent to speakers. This process is performed by a digital to analog converter (D-to-A converter or DAC).
Here is a simplified illustration of how audio waves are stored using digital samples:
Each sample represents a value within a range of possible values. The range of possible values is determined by the bit depth. Bit depth is the term that describes how many bits are included in each sample.
Each bit can represent two possible values. Samples which utilize more bits can represent a larger range of values, and therefore can store more precise information about the amplitude of a sound wave. Each time a bit is added, the number of possible values is doubled. Whereas one bit can represent two values, two bits can represent four values, three bits can represent eight values, and so on.
|Bit Depth||Possible Values|
|16-bit (CD Standard)||65,536|
|24-bit (Professional Standard)||16,777,216|
The standard bit depth for CDs is 16-bit, allowing for 64,536 possible amplitude values. The professional standard is a bit depth of 24-bit, which allows 16,777,216 possible amplitude values! However, most studios record and mix using 32-bit floating point, which will be discussed in a different post.
The sample rate determines how many samples are taken of a sound wave per second. Sample rate is measured in Hertz (Hz). Recording at a higher sample rate allows higher frequencies to be recorded.
The Nyquist Theorem states that digital sampling can only faithfully represent frequencies less than half of the sampling rate. This means that if you want to capture 20kHz, the highest frequency audible to humans, you must use a sample rate greater than 40kHz.
For this reason, 44.1kHz is the standard sample rate for CDs. Professional audio for video utilizes a standard of 48kHz. Many recordings greatly exceed these standards, with sample rates of 96kHz and beyond!
While the benefit of higher sample rates is often understood to be an extension of recorded frequency range, this isn’t the main benefit. I won’t get too deep into it in this post, but it has more to do with the type of anti-aliasing filter that can be used to filter out higher frequencies with fewer artifacts. The resulting bandwidth of a 44.1 kHz recording and a 96 kHz recording are virtually the same in the end.
Digital Audio Data Compression Formats
The audio files produced by recording studios are very large, due to the amount of information they contain. If a 3-minute song is recorded with a bit depth of 24-bit and a sample rate of 96kHz, the file size will be approximately 52MB. This file is too large for consumer applications, such as streaming. For this reason, data compression formats are used. Data compression is a method of reducing the size of a file. There are two main categories of data compression formats, lossy and lossless.
Lossy Data Compression Formats (MP3 & Streaming)
If information is lost through the process of compressing data, the compression format being used is lossy. Unfortunately, the most widely used data compression formats in consumer audio are lossy. This means that, although special algorithms are used to reduce negative effects, data is lost during the process of compressing the file. Once data is lost, it can never be restored.
The most common lossy audio data compression formats are MP3, AAC, and Ogg Vorbis. These formats are used for storing many files with limited hard drive space or streaming content over limited-bandwidth internet connections.
The proprietary algorithms behind these formats aim to prioritize content based on models of human perception of sound and destroy the low-priority content.
Lossless Data Compression Formats (FLAC & AIFF)
If no information is lost through the process of compressing data, the compression format being used is lossless. Some streaming services, such as Tidal, utilize lossless compression. Using these formats, information can be encoded into a smaller file and later decoded, ultimately restoring the original PCM information as a WAV file. Although these formats do save some space compared to uncompressed files, they are nowhere close to the efficiency of lossy formats.
Key Differences Between Analog and Digital Audio: Recording and Playback
As you can see, analog and digital audio recording technologies share a common goal – to create a model of acoustic waveforms that can be played back as accurately as possible. Each technology accomplishes this goal quite well. The audio quality achieved using one method is not necessarily better than the other, just different. The unique qualities of each method will be explored in this section.
Frequency Range (Bandwidth)
As mentioned above, the frequency range of a digital signal is limited to frequencies below the Nyquist Frequency. In theory, the upper limits of analog recording media is much greater than the human range of hearing.
This difference is not as significant as you might think. First of all, any benefit of an extended bandwidth beyond a digital recording at 44.1 kHz sample rate would be beyond the range of human perception – not to mention the extended frequency ranges made possible by higher sample rates.
Secondly, most audio equipment (microphones, speakers, etc.) has built-in band limiting filters. These are effectively low pass filters that prevent the capture or reproduction of frequencies beyond the human range of hearing. Thus, there is a technical difference in the frequency range between analog and digital audio, but not a practical difference.
In fact, the primary benefit of higher sample rates in digital audio isn’t actually a greater frequency range for the listener, but the ability to use different anti-aliasing filters. This doesn’t effectively extend the frequency range, but instead reduces the artifacts caused by sampling. I’ll go deeper into this concept in a later post.
The major drawback to analog audio recording technology is that it has a significantly higher noise floor compared to digital technology.
Even the most high-quality analog tape contains magnetic noise. This is the cause of hiss in analog recordings. The theoretical noise floor of a 24-bit digital recording is -144dB – effectively infinite.
Remember, the noise floor of any system is only as low as the combined noise floor of all of its components. This means that even digital systems will be noisy if the signal chain contains noisy electronic elements.
Vulnerability & Longevity
Not only do analog media, such as tape and vinyl, contain inherent noise, but they are also extremely vulnerable to degradation over time. Digital media, such as hard drives and CDs, are far more resilient.
All physical media, both analog and digital, degrade over time. The first time a recording is played back is the best that recording will ever sound. Listen to old vinyl records, and this becomes obvious.
Analog tape must be preserved in very specific conditions to prevent loss of quality over time. Vinyl records are damaged each time they are played. Digital media can also be damaged, but the degradation is much easier to prevent.
A digital recording is a series of numbers that can be reproduced infinite times with perfect precision, whereas each reproduction of analog audio adds to the total noise of the recording. For example, if you transfer one tape recording to another reel of tape, you will have recorded the noise from the first reel to the second reel.
Portability and Reproducibility
Finally, digital audio media are drastically more portable and reproducible than analog media. Not only are hard drives and flash storage much smaller and lighter than reels of tape and vinyl records, but the digital information saved on them can be sent across the planet in seconds using the Internet. The reproduction of digital information comes at virtually no cost compared to reproduction of analog media.
Analog vs Digital Audio: Reinforcement and Distribution
In this section, rather than recording systems, I will highlight the differences between analog and digital audio reinforcement and distribution systems. These are the systems used in public address and live concert applications.
Analog audio systems for sound reinforcement and distribution require no recording technology.
An acoustical signal is converted to electricity using a microphone. The electrical audio signal is sent to a microphone preamplifier, then to analog audio effects and mixers, and finally to an amplifier. The amplified audio signal is converted back to acoustical energy by a loudspeaker.
From the beginning to the end of any analog signal chain, the audio signal is either in the form of acoustical or electrical energy. There is no need to store the signal. Everything happens in real time at the speed of electricity in a wire: about 75% the speed of light.
Digital audio systems for sound reinforcement and distribution do require recording technology.
The electrical audio signal is converted, or quantized, into PCM (Pulse Code Modulation). Any time a signal is converted from analog-to-digital or digital-to-analog, this quantization occurs. That means that every signal sent to and from a digital effect using analog cables is converted to PCM inside the unit, processed, and then converted back to electrical energy. All digital audio processors, mixers, and amplifiers create brief recordings to process audio signals.
Key Differences Between Analog and Digital Audio: Reinforcement and Distribution
Although the speed at which these digital quantizations are processed is extremely fast, they are still much slower than electricity moving through a wire. This characteristic of digital audio systems has the negative effect of adding latency to the signal. Latency is the delay of a signal caused by processing.
All digital audio systems add latency to the signal chain. However, the effects of latency have been drastically reduced as technology continues to improve. One of the primary drawbacks to adding latency to a system is the risk of destructive phase interference. If a signal takes two paths, each adding latency to the signal differently, the signals will be out of phase, and might cause comb filtering or echo. Latency can also make for an unnatural monitoring experience for musicians and other talent. If a signal is delayed, the person speaking or playing an instrument may be confused as they monitor themselves in headphones. For this reason, it is usually best to monitor directly through an analog signal chain if the digital system adds too much latency to a signal.
The primary drawback of analog systems is their weight and size. Modern digital audio mixers contain within them equalizers, compressors, gates, and other effects for every channel. Analog systems with the same processing capabilities would require several racks and thousands of pounds of gear.
It is much simpler to configure digital effects on the fly within a digital console, with no need to add analog cables for patching. If a mix engineer would like to try a different effect mid-show, they must simply press a few buttons with a digital system. This change might require repatching an analog system.
Whereas analog equipment contains the heavy electrical components that make up equalizers, compressors, and reverb effects, digital signal processing chips offer similar tools at a fraction of the space and weight.
As you chain together more and more analog effects, the electronic noise from each device sums together. Using more digital effects adds no noise to the signal because the signal never leaves the digital signal processor. Only the inherent noise of a single device is present, rather than the inherent noise of many devices.
The Debate Continues
The truth is that both analog and digital audio systems have value in the modern world. The debate over which is better and which is worse will never end, because there is not a clear answer.
There are a million applications for audio technology, and each one calls for a unique set of equipment. As an audio engineer, musician, or listener, we must each decide on a set of audio equipment that caters to the needs of each unique situation.