Category

Audio

Category

Of all of the tools in your musical toolbox, compression seems to have an unshakeable mystique.

This might be because it is so versatile, with both super-subtle and extreme applications being common in music production.

There is also the fact that compression is quite a transparent effect. It’s not like stomping on a distortion pedal where the result is immediately obvious!

In fact, well-used compression can be completely unnoticeable, though if it wasn’t there you’d definitely miss it.

So far, so mysterious! How can you unlock the magic box that is compression?

What Does Compression Mean

The first step is to understand what compression actually does, and then to get to know how it can be applied to create the results that you want, whether you’re using a studio hardware compressor, a plugin, or a stompbox.

That sounds like a big task, but we’re here to help with a simple guide to what compression means.

You’ll quickly discover that compression isn’t as mystical as it appears, and then you’ll be able to use it in your music with no worries.

Soundwaves

The first step in understanding compression is to understand sound.

Every sound is expressed as a wave that has louder and quieter parts. Think about when you play a note on a guitar or piano.

It starts out loud, then dies away over time. Alternatively, sing a song.

You’ll notice that your voice isn’t the same volume the whole way through. As you wind up into a big chorus, you’ll get louder and more passionate.

That’s how good performances should be, full of dynamics and feel.

However, this can present a problem from a recording and mixing point of view.

Sounds that suddenly jump out at the listener aren’t generally what you’re looking for, but the sonic characteristics of those louder sounds are important for bringing your performance to life.

Squashing The Peaks

At its very simplest, compression reduces the volume of the loudest parts of a sound. Say you’ve got a nice guitar part, and in some bits, you dig in slightly harder.

Adding some compression will make those parts where you play harder less different in volume from the parts where you play softly.

When you apply a compressor, you are setting a point at which the volume of your sound is reduced.

You also set how much it is reduced by, how fast this happens, and how long the reduction lasts.

This is why it’s important to think about the character of the sound that you’re working with. Next up, we’ll talk about how you go about squashing those peaks to suit your sound.

Threshold And Ratio

Threshold And Ratio

Two of the most important controls on any compressor, be it hardware or software, are the threshold and ratio controls.

These set when the compression effect kicks in, and how much of it there is.

The threshold sets the level that your compressor starts working at. Say you have a piano part that normally sits at around -12db, but occasionally reaches -10db.

If you want to bring that in a little while preserving the dynamics, you can set your compressor at around -11db. That way, any signal that rises above 11db will be affected by the compressor.

The ratio control tells your compressor how much to reduce the volume once the signal goes over the threshold you have set.

It is expressed as a ratio. A compressor set at 2:1 lets 1db through for every 2db the signal exceeds the threshold. An 8:1 ratio means that the compressor lets 1db through for every 8db a signal exceeds the threshold, a much greater reduction.

The higher the first number in the ratio, the more extreme the compression will be.

Attack And Release

The two other important functions of your compressor are attack and release.

These control how quickly the compressor kicks in and how long the effect lasts. These two tools are powerful and make a massive difference in how your compressor works, giving you the ability to fine-tune your compression.

Attack governs how quickly the compressor acts when it detects a signal above the threshold you’ve set.

A super-short attack will jump right on that peak and instantly squash it, whereas a longer attack will let the peak happen but suppress the volume of the signal afterward.

Too short an attack can sound unnatural, so even with sounds like drums you probably don’t want an instantaneous attack.

The best way to dial attack in is to set it to a relatively conservative length, and then make it shorter or longer until your ears tell you it’s musically right. Longer attacks generally sound more musical than short ones, but there is a place for both.

Release sets how long the signal’s volume stays suppressed after the compressor kicks in. Setting your release is really a matter of using your ears because what sounds right is so dependent on the specific sound that you’re working with.

Make sure you adjust your attack and release settings in the context of the whole track rather than with it solo.

That way you’ll be able to see if you’re creating an unnatural feel much easier.

Gain

The final key part of a compressor is the gain control. Naturally, when you’re compressing a sound you are reducing its volume. You’re going to need to add a little of what’s called make-up gain to bring the track back up to the appropriate level for your mix.

Rather than watching the meters, try to bring your track back into place by ear. Adding gain back in isn’t just a matter of noting your peak reduction and raising it to compensate!

A Note On Sidechains

One powerful effect you can achieve with compression is ducking. Say you have a kick drum and a bass part that occupy a similar frequency range. You can make your kick drum punchier if you can get that bass out of the way!

Setting up a compressor on your bass track but feeding it the kick drum signal via the side chain means that the bass part gets compressed when the kick drum plays.

By playing with the compressor controls you can make this effect subtle or very prominent, whatever suits your sound.

Key Takeaways

Hopefully, you now have a better understanding of what compression is and how it works. It’s a super-powerful tool that is essential in getting your mixes to sit together properly and for shaping individual vocal and instrument sounds.

The best thing you can do is to load up a compressor and experiment with it to see what you can make it do!

You might have seen the words bass, treble, or mid, while setting on your speaker system, guitar amplifier, television, or mobile phone.

You might know what it does, but not exactly what it means.

And whether you are wondering what treble, bass, and mid is, or how to set them to get the best sound, you have arrived at the right place.

In this simple guide we go through what these words means and what they’re is used for, as well as how to set bass, mid, and treble for the best sound possible.

We also provide a simple explanation of audio frequencies and the audio spectrum.

So, without further ado, let’s dive in.

What Is Treble

What Is Treble On A Speaker?

Treble refers to the higher frequencies of sound, collectively.

More specifically: the frequency range from 4000 hertz to 20,000 hertz. (Don’t worry about hertz for now, as we explain these later.)

Treble is a setting common on sound systems, guitar and bass amplifiers, and other music hardware, in addition to televisions, mobile phones, and music software.

If you have fiddled around with treble before, you will have noticed that it increases or decreases the “brightness” of the sound being played.

By reducing treble numerically, or turning a treble knob anti-clockwise, you are decreasing a sound’s brightness.

By increasing treble numerically, or turning a treble knob clockwise, you are increasing a sound’s brightness.

What Is Bass?

Unlike treble, bass is a pretty common word. We all know what bass is, which collectively refers to the lower frequencies of sound.

Specifically, bass is attributed to sound frequencies between 20 hertz and 250 hertz.

Similar to treble, bass is a common setting on speaker systems and amplifiers, and can also be found as a setting on televisions, mobile phones, and certain music applications.

This goes without saying: by reducing bass numerically, or by turning a bass knob anti-clockwise, you are decreasing the bass of a sound or speaker in volume.

By increasing bass numerically, or turning a bass knob clockwise, you are increasing the level of bass.

What Is Mid?

Now we know what bass and treble are, it is obvious what mid is.

Mid is short for midrange and refers to sound, or audio frequencies, that are not bass and not treble. In other words: the frequencies that fit in between treble and bass.

Specifically, mid refers to the audio frequencies between 250 hertz and 4000 hertz.

If you played around with the mid settings on your speaker system or television, you will have noticed that it can make the music sound more “boxy”, or more “empty”.

Midrange frequencies are typically associated with the sound of the human voice, as well as guitars, saxophones, tom-tom drums, and so on.

Mid, as a setting, can be used to reduce a “boxy-sounding” sound system or blend the balance of treble and bass in music.

What Should Bass, Mid, And Treble Be Set At?

What Should Bass, Mid, And Treble Be Set At

What is the best bass, mid and treble setting?

The simple answer to this question is that it depends; it depends on the music or sounds being played, as well as the speakers from which the sound is coming.

No song or piece of music is the same, and neither are all speaker systems the same.

Music is also entirely different from the sound production of a movie.

At the same time, everyone has different preferences for how bass, mid, and treble should be set.

Some people prefer more bass when listening to music, while others prefer more “brightness” that comes with increased treble.

The best bass, mid, and treble setting is the one that pleases the listener, improves the sound of the speakers, and complements the room (acoustics) where the sound is being played.

What Is An Audio Frequency?

Audio frequency is a specific measure of pitch caused by vibration.

Audio frequencies are measured on an audio spectrum, or frequency spectrum, in hertz (Hz).

We know what treble, mid, and bass are, but frequencies – specifically frequency bands – are a more accurate way of processing sound tonality, as well as understanding sound in terms of its pitch on a scale from low to high.

The audio spectrum can be split up into seven main frequency bands: sub-bass, bass, low midrange, midrange, high midrange, presence, and brilliance.

As for their respective frequencies:

  • Sub-bass is frequencies between 20 Hz and 60 Hz.
  • Bass is frequencies between 60 Hz and 250 Hz.
  • Low midrange is frequencies between 250 Hz and 500 Hz.
  • Midrange is frequencies between 500 Hz and 2000 Hz.
  • High midrange is frequencies between 2000 and 4000 Hz.
  • Presence is frequencies between 4000 and 6000 Hz.
  • Brilliance is frequencies between 6000 Hz and 20,000 Hz.

What Is An EQ In Music?

Just like bass, mid, and treble, you might have seen EQ settings on your television or mobile phone, on music hardware, or within a music software program.

EQ stands for equalizer or equalization (both are the same thing).

Equalizers are used to process sound by increasing or decreasing the volume of audio frequency bands – which can be anything from the seven main frequency bands to up to 30 bands.

In other words, EQs are advanced bass, mid, and treble controls, offering greater control over how music, or a sound, sounds in terms of its tone.

EQs are typically used for music production, as well as among audiophiles.

EQs can be music hardware, such as parametric EQs and guitar pedals, as well as software programs used within a digital audio workstation (DAW).

Professional EQs can even offer a live visual display of the sound on the audio spectrum as it is being played in real-time.

What Is The Best Equalizer Setting For Music?

Again, just like the question of what is the best bass, mid, and treble setting, the best equalizer setting for music (casual listening) is subjective.

It will also depend on the speakers on which the sound is being played, the room acoustics, and the music itself.

The best EQ setting will change depending on the song, simply because not all music is the same.

For example, a piece of classical music sounds different to a modern EDM track. Therefore, the same equalizer settings will not complement both tracks.

At the same time, everyone has different tastes when it comes to how much bass and treble sounds ideal.

Room acoustics are also a big factor when it comes to equalizer settings, as large bass frequencies do not fit in small rooms.

Higher frequencies can also be more prominent in rooms with lots of reflective surfaces. As a result, different rooms can require different EQ settings when it comes to music listening.

Conclusion

In short: treble in music refers to the higher frequencies of sound.

Treble is a common setting on speaker systems, guitar and bass amplifiers, televisions, and music software.

By adjusting the treble, you are increasing or decreasing how “bright” music, or a specific sound, sounds.

Treble is attributed to the frequencies between 4000 hertz and 20,000 hertz.

On the other end of the audio spectrum is bass. Mid is the audio range between treble and bass – collectively all sound frequencies between 250 hertz and 4000 hertz.

 

Whether you’re recording the next great true-crime podcast, working on a rock opera in the basement, or trying to get your record player hooked up to the same receiver as your television, it’s essential to know the difference between a line input and a mic input.

With the gadgets and connectivity that modern technology affords us, it can be messy figuring just what cords go where. This confusion is especially relevant for line and mic inputs. Keep reading for a short guide to help you know why and when to use each.

The Difference Between Line in and Mic in

In the simplest terms, a mic input is designed specifically for the signal from microphones. A line input design is for higher-powered signals from mixers and consumer electronics such as Blu-Ray players, MP3 devices, and smartphones.

While they seem similar, you need to know the difference to get the best performance from your equipment while keeping it from damage. To better understand when to use each, let’s take a deeper look at the differences between the two.

Power

The main difference between the inputs is the amount of power they support. Line inputs produce more volume than microphone inputs. A line-level signal is one volt, 1,000 times stronger than a mic-level signal. For this reason, plugging a microphone into a line input will be extremely quiet – if audible at all.

Alternatively, because a mic input handles much lower voltage, using mixers, instruments, or other audio sources into a mic input will overpower the equipment. The resulting sound will be too loud, to the point of distorting or damaging the equipment.

Mono vs. Stereo

Many microphones are directional, meaning their design picks up sound from a specific point while minimizing surrounding sounds. This directional ability comes in handy while recording in spaces where extraneous sounds are audible. Because of this design, mic inputs only use a single channel, making them mono.

Since the mic input is mono, any line-level signal plugged into the mic input will result in mono audio. This difference between mono and stereo is why using the line input is essential for mixers and receivers. Most media utilizes multiple channels for stereo and surround sound. The best movie and music experiences will come from the line input.

Connectors

Often mic inputs and line inputs have different connectors that can be helpful in discerning which to use. Generally, a line input will support three types of connectors: RCA, quarter-inch jack, or eighth-inch jack (also known as a 3.5 mm phone jack).

Mic inputs, particularly for recording purposes on audio interfaces and mixers, will have a female XLR input. On consumer-grade electronics, such as handheld recording devices and laptops, mic input connectors might also be quarter-inch or eighth-inch jacks.

Most products will take great pains to label their inputs properly. Always read the device’s instruction manual to be sure of proper use.

Purpose

It can be helpful to boil down any questions regarding line and mic inputs as a question of purpose. Ask yourself, what is this device supposed to do?

With few exceptions, microphone inputs are strictly for microphones. If there is no mic input on the device, the voltage can be modified for a line input. We will cover these techniques in the next section.

Since most devices need higher voltage and multiple channels, they generally require a line input.

Can a Line in and a Mic in Be Used Interchangeably?

In short, line inputs and mic inputs are not interchangeable. This incompatibility is due to the variance in the amount of power required for each. Resorting to using a mic input for a line-level device could distort your sound, damage your equipment, and cause harm to your hearing.

Using a microphone in a line input will create a barely audible signal. Turning up the volume will not solve the problem – it will amplify a noisy mic that still won’t be loud enough.

Modifying a Device’s Signal to Converting an Input

Fortunately, necessity is the mother of invention. Below are a few ways to modify your signal to use in its opposite input. These are handy workarounds for devices that have limited inputs. These also serve as creative ways to maximize the sound quality from minimum inputs.

Always consult the instruction manual for additional information on input and output specifications.

Using a Mic Input for a Line Input

Below are a few modifications that allow a microphone to work a line input.

Attenuator

Attenuators are electrical components that act as the middle man between your device and the input. They help maintain the integrity of a signal while reducing its overall power or amplitude. Attenuators come in several different shapes and sizes depending on the connector needed.

Direct Injection Box

Working in a similar vein to an attenuator, a Direct Injection box (also known as a DI box) helps connect high-level signals to mic-levels, usually through an XLR connector.

Additionally, these devices are helpful for those that wish to pursue audio engineering. They are common in recording studios and music venues because they minimize the amplitude of signals without additional devices, ensuring the sound is unaffected.

Using a Line Input for a Mic Input

Below are a few modifications that allow a microphone to work through a line input.

Mic Pre-Amp

A microphone pre-amp is the opposite of a DI box or attenuator as it boosts the level of a microphone to a line-level signal. While doing this, it also ensures that hiss and other forms of white noise often gained by cranking a microphone’s signal are kept at a minimum, ensuring a clear and true microphone sound.

A mic pre-amp is also a useful tool for recording artists and podcasters as it allows you to control the microphone tone. This control can help to bolster vocals and add warmth to conversations.

Mixer

A mixer is a very versatile audio engineering tool. It can be used to narrow multiple signals down to one output, which is very helpful when recording with multiple mics while only having a single input. A mixer enables a microphone to run through it and then out as a line input.

Mixers also allow for the use of microphones that require phantom power with line inputs. These microphones, known as condenser mics, are more sensitive to sound and require additional DC voltage to work. Without phantom power, these mics will not yield any sound at all. This makes having a mixer essential for any use of a condenser mic.

Final Thoughts on Line in vs. Mic in

Though they seem to do the same thing — and can even look the same — a line and a mic input are different parts of a device. To achieve the best sound, you should learn their differences to know which to use for your connectivity needs.

Always be sure to read your device’s instruction manual, especially if you’re ever in doubt about which input to use. Remember that sometimes the connector for both inputs can be the same.

After learning the differences between the two inputs, you’re now able to capture your creativity freely – even knowing how to work with input limitations.

Mono and stereo are two different types of sound, both with their own advantages.

Mono, or monaural sound, is an audio system that uses one channel to broadcast signals. Stereo is an audio system with two channels, though it can also be used with more than two.

Stereo has the advantage of offering a richer experience for listeners because it provides depth, while mono offers limited depth.

In some cases, mono is preferred over stereo. For example, when a sound engineer wants to maintain a specific volume level in a mix, they might use mono to make the sound more consistent across all channels. Additionally, mono can be used when there are limited channels, such as in a live setting.

In the end, it comes down to what the listener prefers and what sounds best in the given situation. Mono and stereo are valuable audio options with their own strengths and weaknesses.

Mono vs. Stereo: What’s the Difference?

The difference between mono and stereo sound is how each type of audio system processes sound.

There is only one channel for all sounds to be processed through in a monaural sound system, while in a stereo sound system, there are two channels for sound. The difference in channels can cause certain sounds to be more or less emphasized than others and affect the balance of the sounds in the mix.

In a stereo sound system, individual sounds are processed in two separate channels. When the listener is positioned between the two speakers, they will hear an immersive 3D surround sound effect. This type of sound processing is called stereophonic sound reproduction, or stereo for short.

Both of these sound systems are common today; however, mono sound systems were more common until the 1950s, when stereo became more popular.

What Are the Advantages of Mono?

The advantage of mono is its simplicity. Mono is easy to use with only one channel and can be more efficient because it doesn’t require two speakers. The main benefit of using mono over stereo is bandwidth efficiency, meaning you can transmit more information in less space or time.

For example, if you want to send a voice message across town on your walkie-talkie, the signal will need to be transmitted at 10 watts for both channels (20 watts total).

But if you use just one channel, it only needs 5 watts (10 watts total) to communicate the same message. This means that by switching from stereo to mono transmission mode, you can save energy and space.

Mono is often thought of as an inferior format when it comes to music because it doesn’t offer the same level of sonic detail as stereo. However, mono sound has several advantages that make it a viable format for specific applications.

Mono is great for public announcements or when you need to be heard by many people, such as a football stadium or large room. It’s also particularly well-suited for spoken word content, such as news and podcasts.

What Are the Advantages of Stereo?

There are many advantages to stereo sound. When you listen to music or watch a movie in stereo, you get a more immersive experience. You can hear sounds coming from all directions, making it feel like you’re right in the middle of the action. Stereo sound also provides a more realistic and lifelike experience.

Another advantage of stereo sound is that it makes the audio sound clearer and more distinct. This is especially true if you’re listening to music or watching a movie with surround sound. Surround sound creates an even more immersive experience by providing additional audio channels that make you feel surrounded by sound.

Stereo audio is also an industry-standard format. Whether you’re listening to music on your phone or watching a movie in the theater, the sound is probably stereo. This enables you to easily transfer your home stereo setup to other devices, which means that it’s compatible with most vehicles and smartphones.

Mono vs. Stereo: Which Is Better?

Mono or stereo? Which is better? Which should you choose? This question has been asked since the invention of the radio. People have argued for both ways, trying to find the answer. But the answer is not as clear-cut as you might think.

Let’s start with mono. Mono is a single channel of sound. It is usually the simpler option and is often used for spoken word or music with a basic melody. Mono is perfect for when you want to focus on one thing and don’t want any distractions. It can be used in various situations, from watching TV to listening to music while you work.

Stereo is a two-channel system that provides a more immersive experience, giving you a sense of depth and space. It uses left and right channels to create a stereo image.

The left channel is sent to the left speaker, and the right track is sent to the right. This gives you a sense of spatial awareness, which allows you to hear where instruments are placed in relation to each other (usually across a horizontal plane). Stereo sound adds an extra level of richness and clarity.

It’s easy to focus on one thing with mono, but stereo can be more immersive.

How to Decide Which Is Right for You

So, which is better? Well, it depends on what you need. Mono is standard for most applications because it’s more balanced and requires fewer speakers. However, stereo can provide a more immersive experience, particularly with headphones.

There are a few other factors to consider when deciding between mono and stereo sound.

The first is the type of content that you will be listening to. If you watch a movie or listen to music, the content was likely created with stereo sound in mind. However, if you watch a news program or talk on the phone, mono sound may be a better choice.

The second factor to consider is the size of the space in which you will be listening. In a small room, stereo sound may not be as effective because the sounds will bounce off the walls and create echoes. In this case, mono sound may be a better option.

The third factor is the number of speakers available to use. Mono sound systems consist of a single speaker, while stereo sound systems require at least two.

When deciding between a monaural and a stereo system, keep these factors in mind to choose the best option for your needs.

Mono is the better option if you want to focus on one thing and don’t want any distractions. However, if you want a more full-bodied experience with a sense of depth and space, stereo is the better choice. Stereo is also perfect for watching movies or listening to music, as it gives you a more realistic sound.

The choice is ultimately yours.

Conclusion: When to Use Mono vs. Stereo

Both mono and stereo audio are great for different purposes, so it’s important to understand when to use each. Some favor mono because it is simple, while some prefer stereo for spatial awareness. So, which one should you choose?

It really comes down to preference, but mono is better for situations where you need to focus. Stereo is better if you want a more immersive experience.

If you don’t know anything about home recording, Phantom Power can be confusing. So what is it? Phantom Power is an industry-standard approach of transmitting DC voltage via an audio cable to supply power to professional audio equipment.

The technique was initially developed to allow condenser microphones to operate without batteries. Phantom power is also commonly used to power active DI boxes and other audio devices. The voltage is usually supplied by a mixing console, preamplifier, or other audio devices.

In this article, we’ll provide an in-depth guide in unraveling the mystery of phantom power and its role with microphones.

Phantom Power: Definition

As mentioned above, phantom power is a method of supplying power to microphones. It is provided by audio interfaces, mixing consoles, microphone preamplifiers, and standalone phantom power supplies.

Phantom power gets its name from the phantom or ghost images that appeared on early television screens. These images were caused by a 60 Hz electrical current on all televisions but were not visible to the naked eye.

The same principle applies to phantom power in audio systems. The 48 volts DC is too low for audio monitoring but is strong enough to power active microphones. A single standard connector carries the entire phantom supply.

History Of Phantom Power

Phantom power was initially introduced in the year the 1960s. It used to be a popular way of powering ribbon microphones, as many condenser mics have been designed to work with it from the start.

In 1904, Sir John Ambrose Fleming invented the vacuum tube, and in 1905, the first triode vacuum tube was created by Lee De Forest. It wasn’t until 1928 that the tube condenser was invented. This new design was a significant advancement in microphones, as an external power supply could now power them.

Schoeps produced the first-ever solid-state microphone (CMT20) in 1965. And in 1966, Neumann released the CMV3, the world’s first phantom-powered microphone.

How Does Phantom Power Work?

Now that we know a little history about phantom power let’s look at how it works.

As mentioned earlier, phantom power transmits DC voltage via an audio cable to supply power to professional audio equipment.

The following steps outline how phantom power works:

This power supply is very low voltage, usually around 48 volts, and it cannot be detected electrically by most microphones because of the high impedance of its internal transducer element. Microphones that do not require such a power supply (e.g., dynamic microphones) do not sense this voltage at all.

When the audio devices are plugged in, they transmit DC across the two wires of the balanced output cable. Usually, this is done using a charge pump circuit. This current passes through the microphone’s transducer element (at about 2mA), creating a voltage across it.

The polarity of this voltage is opposite to the voltage created by the audio signal, which is why a transformer is used to step down the voltage and create a common ground for both signals.

Phantom Power Standards

The general standard for phantom power is 12 to 48 volts of DC power transmitted on a balanced line (3 or 3.15 mm TRS connector) along with the audio signal, resulting in a current flow of 2 mA.

The actual voltage can be anywhere from 9 to 52 volts. It is all relative to the connected equipment. The standard for phantom power is 48 volts. However, there are some variations in the voltage, as different devices require different voltages to operate correctly.

For example, most ribbon microphones require a higher voltage of 52 volts to function correctly. And some condenser microphones can operate with as low as 9 volts.

Will Phantom Power Affect The Audio Signal?

No. Phantom power will not affect the quality of your audio signal in any way. It is just a way of transmitting power to microphones and other pro audio equipment.

However, if you use a low-impedance mic with a high-impedance input (like a guitar amp), you may get some noise from the phantom power supply. There is a voltage difference between the two devices, and the amp will “hear” this as noise.

What Is Digital Phantom Power?

Digital phantom power is a term that applies to phantom power in digital audio systems. The Audio Engineering Society (AES) issued AES42, which standardized 10 volts of DC phantom powering in digital audio equipment. If your gear is designed to receive digital phantom power, it will automatically detect it and adjust its internal circuitry accordingly.

Digital phantom power supplies transmit their power through XLD or XLR connectors. They can be found on audio distribution equipment, digital microphones, and audio interfaces. Currently, the standard voltage is 12 volts. However, 10 volts is more widely used.

Phantom Power vs. Battery Power

There are some pros and cons to using phantom power vs. battery power:

Pros of Phantom Power

  • You don’t have to worry about changing batteries
  • It is more reliable than battery power
  • It is less susceptible to interference than battery power

Cons of Phantom Power

  • It can be more expensive than using batteries
  • You need to have a phantom power supply available to use it

Pros of Battery Power

  • Battery power is cheaper than phantom power
  • You don’t need any special equipment to use it
  • It is less susceptible to interference than phantom power

Cons of Battery Power

  • Batteries can run out of juice unexpectedly
  • They can be heavy to carry around

If you don’t want to have to worry about changing batteries, then phantom power is the way to go. But battery power is the more affordable option if you’re on a tight budget. And if you’re worried about interference, battery power is the better choice. It all comes down to what works best for you and your needs.

What Types Of Microphones Require Phantom Power?

Most condenser microphones require phantom power, but they can also be used with dynamic microphones and ribbon microphones. You just need to make sure that your microphone is compatible with phantom power before you try to use it with a phantom power supply.

If you’re not sure if your microphone requires phantom power, consult the manufacturer’s specifications or do some research online. There are plenty of resources available that will help you determine whether or not your microphone needs phantom power to operate.

Should You Turn Off Phantom Power Before Plugging in a Microphone?

Yes, and you need to make sure you also turn off the power before you disconnect the mic.

If you’re using an XLR cable with a 48 volts battery, turn the phantom power off before plugging in your microphone. It will eliminate any loud pops or sudden changes in volume when you connect it.

Final Thoughts

Phantom power is a way of transmitting power to microphones and other pro audio equipment. It is a standard feature on most digital audio systems, and it can also be found on audio distribution equipment, digital microphones, and audio interfaces.

It is essential to understand that there are two types of phantom power: digital phantom power and analog phantom power. Digital phantom power is standardized, whereas analog phantom power isn’t.

There are some pros and cons to using both battery power and phantom power, but if you want the best option for reliability, then go with the standard 12 volts of phantom power. Just make sure your microphone is compatible with phantom power before you try to use it.

MP3 and M4A are both audio file formats within the same family for digital audio formats. MP3 was developed by the Fraunhofer Society in Germany with support from different scientists in the US. M4A was developed by the ISO project.

MP3 stands for MPEG-1/MPEG-2 Audio Layer III. The format is essentially a lossy format for audio files. M4A is the audio codec intended as a successor to MP3. Audio-only MPEG-4 files usually have an M4A extension.

However, there are still several audio files or professional recordings you can find in MP3. Clearly, many still prefer that format. M4A, though technically superior, is perhaps not preferred by certain audiophiles. What’s the truth of the matter?

Let’s find out.

What is the Difference Between MP3 and M4A?

As mentioned above, M4A was meant as a successor to MP3. The latter was originally designed for audio only but wasn’t its own format. It was the third layer in an MPEG-1 or MPEG-2 file.

  • Other differences between the formats include different levels of popularity. In the MP3 vs M4A debate, it’s clear that MP3 is much more popular among audiophiles since it’s been there for longer. MP3 is used by Google Music, now YouTube Music. However, M4A formats are used by Apple for its iTunes and Apple Music library.

NOTE: As for other streaming services, they use very different file formats to enable less lossy conversion of audio. Spotify uses AAC, SoundCloud Go+ uses AAC, and Amazon Music uses FLAC.

  • The extended file format for MP3 is MP2, while the extended form of M4A is QuickTime.

Generally, M4A files are considered better in sound quality to MP3 files, since they have better compression capabilities. However, the specifics of the compression and the processing have to be made more clear.

MP3 and M4A: Which is Better?

While M4A is technically better at compressing digital audio files than MP3, the difference is based on perception. It’s not within human perception to differentiate between two audio formats with nearly equivalent quality of sound.

Advantages of M4A File Format

  • M4A files sound better than MP3 files when encoded at the same bit rate since the compression algorithm is better.
  • M4A files also have a smaller sample block size of 120/128 samples rather than 192 for changing signals. What this means is that more precise details are sampled within music like tones, different frequencies, etc.
  • For stationary signals, M4A files have larger block sizes of 1024 vs MP3 block sizes of 576. This allows for less data to represent a portion of the music which doesn’t have the same complexity.

 

Advantages of MP3 File Format

  • You can share MP3 file formats on all platforms and media players. It’s a nearly universal audio codec which will play on any media player or OS.
  • MP3 allows a higher compression of the file size. Hence, you can store more audio files on your device using MP3 compression.
  • MP3 audio files are less compressed than M4A files. Hence, they may be able to offer better quality audio at higher sampling bit rates (320 KBPS)
  • To Learn More About Sampling Block Sizes and Frequencies Read This Blog

What Should You Choose: MP3 vs M4A?

Hence, it really depends on your audio setup, OS, and your preferences. If you have the latest equipment and prefer the best sound quality with efficient storage, choose M4A. You should also choose M4A if you have MacOS devices.

However, if you’re not worried too much about lossy compression or about impeccable sound quality, MP3 will do well. Remember that it’s a universal format and can play on any media player or platform.

Other Audio Formats Better than MP3 or M4A

While MP3 or M4A may be the most popular audio formats, they’re not the best. Not even close. There are other audio file formats which are considered superior in terms of audio quality. Here are 3 audio file formats which are considered the gold standard for audio quality.

WAV

You may remember WAV files from when you tried to record audio on your computer. It’s one of the first audio file formats ever. It’s considered a staple across all file formats today. They capture and recreate original audio waveforms at the highest quality.

Since WAV files are uncompressed, they don’t get rid of any of the data stored. This provides great versatility when editing or mixing music.

AIFF

While less common and less well known than WAV, AIFF is an incredibly useful audio file format. It’s lossless, of course, and provides studio grade file recording and playback. It offers a sample rate and bit depth options like WAV files.

AIFF was created by Macintosh in 1988 which allowed for full studio quality audio recording on Apple computers. WAV was created by Microsoft and IBM in 1992. Both are great formats which are played well natively on either OS.

FLAC

FLAC is a high-resolution single-bit format which was used for recording on audio CDs. It is a very high-quality codec, so it’s not lossless. However, it can still allow you to play very high-quality audio on larger speakers.

FLAC is also royalty free and is considered a preferred format for downloading and storing music for audiophiles. The downside is that it’s not compatible across all platforms. It’s not supported by Apple products nor is it playable on Apple Music.

MP3 and M4A are both very capable file formats for audio storage and playback. Choosing one over the other really shouldn’t matter unless you’re too picky.

Pin It