What is the difference between analog and digital audio

What should be understood, however, is that most recording studios today use bit recordings, and that a high-quality digital recording fed through a lower-quality processor will have its quality reduced.

At the end of the day, what separates an analog from a digital headset is where the audio processor is located. All audio, whether from an analog or a digital source, must be converted into an analog vibration to produce the sound. Analog headphones connect via a traditional headphone jack to an audio processor built into the computer.

Digital headphones connect either into a digital audio out jack or a USB port and receive digital information to an internally-installed audio processor. There are various upsides to using digital or analog. Digital , however, refers to representing these variable quantities in terms of actual numbers, or digits.

If you consider the numbers 1 and 2 on a number line, there are actually an infinite number of points between 1 and 2. This is what analog represents—the infinite number of possibilities between 1 and 2.

Can you see the difference? As another example, think of analog vs. Not to be confused with digital video vs. This kind of imprint was able to store a much wider frequency range than ever before, and it soon became a standard all over the world. Further developments allowed multiple tracks of sound to be recorded on the same tape. That is where we are at this date in regards to analog recording. Technology moved forward, and that took us into the digital era.

The development and availability of computer technology was a prerequisite for digital sound. The critical difference is that, in digital, you need to convert those sound waves into a series of 1s and 0s to store them.

So, how do you do that? Just as in analog sound recording, a signal is being picked up by the microphone. The standard method to do this is called PCM. It stands for Pulse Code Modulation. It works by creating a sound wave model as a series of ones and zeroes by recording values at specific points along the wave and turning them into binary code. These groups of binary bits are called samples, and an analog-to-digital converter does this process. To playback the music, the digital-to-analog converter reverses this process.

Then the electric signal goes to the amplifier and the speakers. Professional recording equipment uses 24 bits at, up to more than 96 kHz. This kind of sound recording offers good quality but presents a new problem — file size. A couple of minutes of recording at top quality would take up tens, if not hundreds of megabytes of storage space. And that is not very practical on the larger scale. This issue is being addressed by using one of many compression standards.

One of the most popular ones we use today is MP3. It allows a significant reduction in space the file takes while retaining good enough quality for most users. Therefore, the sample rate of an audio file determines the highest frequency that can be sampled with no loss. In theory, an audio file with a 48kHz sample rate is capable of perfectly recording and reproducing frequencies up to just below 24kHz — half of the sample rate.

Given that the range of human hearing and low pass filters within microphones and other audio equipment generally top out around 20kHz, we can safely say that a 48kHz sample rate provides adequate frequency range. Taking a close look at the samples within the DAW, we can see that they are represented as a series of dots connected by straight lines. This representation can be misleading…. The truth is that the waveforms played back through the digital-to-analog converter will be identical to the original waveforms recorded by the analog-to digital converter.

There is another drawback to digital audio that is important to mention, called aliasing. It also has to do with the Nyquist Theorem. If the analog-to-digital converter attempts to sample a frequency that exceeds the Nyquist frequency, it will result in aliasing. On a basic level, aliasing describes a situation where the A-to-D converter confuses a high frequency for a much lower frequency due to sample rate restrictions.

In an effort to mitigate this problem, engineers will apply low pass filters to the signal chain which prevent frequencies that exceed the Nyquist frequency from being quantized by the system. This provides space for the low pass filter to remove extraneous frequencies without negatively affecting the audible frequency range.

OK — aside from the noise, the signal fidelity of analog audio vs digital audio seems pretty close. It can probably be boiled down to the distortion and noise that comes with using analog equipment. As we already discussed, each time you pass a signal from one device to another, the inherent noise from each device will become a part of the signal. Overdriving an analog system or saturating analog tape can sound awesome! Analog gear tends to sound more musical and organic when overdriven.

As you approach the limits of analog circuitry or tape, the quality of the signal will start to change. This nonlinear response to signals is something that can be modeled by digital systems, but is built into analog systems. Digital systems will behave the same no matter where the signal level is in relation to the limitations until the signal actually exceeds those limitations.

On the other hand, the tones you can achieve with skillful gain structure in the analog realm can also be great. We already discussed the general process of analog and digital recording. In analog — the signals are stored to magnetic tape and played back using a tape machine. In digital — the signals are stored to a hard drive and played back using a computer. Aside from the basic recording and playback process, there are a lot of differences between analog and digital music production.

One of the biggest differences is the editing process. In a digital audio workstation or a DAW , we can make cuts and adjustments to the length of a clip without any worry of making an irreversible mistake.

Analog edits have to be performed with a razor blade and adhesive tape to physically cut the tape and reassemble it. Unlike using a DAW, an analog tape editor has to rely solely on their ears to know where the cut should be made, and they have to create diagonal cuts to achieve a crossfade between the two clips. Anyone who has worked with analog tape has probably found themselves in the unfortunate situation where the tape is unravelled from the reel, and the next hour is spent meticulously rewinding the tape.

Not only that, but there are virtually no limitations to the number of tracks that can be stacked in a digital production, while most multitrack analog tape tops out at 24 tracks. These advancements in technology have completely revolutionized not only music production, but music itself! Think about it — the ability to make precise edits to multitrack recordings and record as many takes as you want with no negative consequences.

These benefits have opened the doors to much more elaborate and polished recordings using digital audio. Once the production process is complete, we still need a way of distributing that production to millions of listeners.

This is easily the biggest difference between analog and digital audio. The best method for distributing analog audio is vinyl. Tape is bulky, in addition to being more difficult to reproduce and maintain. However, vinyl has its share of disadvantages too. Firstly, vinyl is a physical medium which means each copy is costly to produce and needs to be physically shipped to listeners. Secondly, vinyl changes over time. Not only is there a quality loss when the master tape is printed to vinyl, but everytime you listen to the vinyl record the quality degrades even further.

Today, digital audio can be streamed over the internet with no loss. You can instantly distribute infinite copies across the world with no quality loss.