How To Record A Whole Band In Your Home Studio

Home Recording

Before powerful home computers were commonplace, music recording required a considerable investment in hardware. Commercial studios were one of the only ways to access sophisticated recording equipment.

While it can seem daunting at first, producing quality recordings at home is a perfectly achievable goal. All it takes is some recording/mixing knowledge, and some key pieces of equipment:

– An audio interface
– Recording software (DAW)
– Studio monitors
– Basic room acoustic treatment
– Suitable microphone(s)

Selecting an audio interface

In order to get started, you’ll need a way of getting high quality audio in and out of your computer. In the early days this was achieved using a ‘Sound Card’. Some people still apply this term to external audio interfaces. Most audio interfaces for home studio use connect to the computer via USB and have inputs that allow you to connect an electronic instrument such as a keyboard, an electric guitar or bass, or a microphone. Condenser microphones require what’s known as phantom power, a 48 volt power supply provided by the device into which they are plugged. You’ll need an interface that can provide phantom power if you plan to use a capacitor microphone or active DI (Direct Injection) box.

The simplest audio interfaces have just one or two audio inputs, as well as a pair of audio outputs to drive your monitors. They will probably also include a headphone output, which is essential if you want hear what your mixes really sound like without the room acoustics misleading you. If you wish to use a MIDI controller keyboard to access the software instruments that come with your recording software (or that can be bought as third-party plug-in add-ons), your choice of interface will also need to include MIDI In and Out ports, usually provided as 5-pin DIN sockets so that you can plug your MIDI cable straight in. Some controller keyboards can connect directly via USB but having standard MIDI ports is always useful and makes it easy to connect external sound modules or synthesizers.

If you are recording several musicians at the same time, then you’ll need a more sophisticated audio interface and it’s here that the Roland Studio Capture ticks all above boxes and more. The Studio Capture can record up to 16 inputs at the same time, 12 of which can be microphones. It can also provide up to 10 outputs at the same time in addition to the main monitor outs feeding your speakers, and though you might not have an immediate use for all of them, it means you’ll be ready to explore surround and immersive sound when the time comes and will also give you more freedom in connecting up hardware studio processors should you wish to. Those 16 inputs come in handy too as even with a small band, you’ll probably want to use quite a lot of microphones around the drum kit. In other words, this isn’t an audio interface that you are going to grow out of for a very long time. Indeed, if you do find yourself needing more inputs you can hook up two of them to double your number of inputs. Another practical point is that the meters on the front panel are easily visible making it easy to keep an eye on your signal levels.

However, there’s even more musician-friendly technology inside the Studio Capture to make recording even easier. With any recording device you have to set the input gain level so that you record a healthy signal level. Too high and you run into distortion, but then you don’t want it to be too quiet either. Studio Capture is equipped with Auto-Sens that automatically sets the input gain level for you. Leave it on while you rehearse the part you are about to record and it will learn from what you play and set the input gain for you based on the loudest parts it ‘hears’.

Recording Techniques

Now to introduce a word that you may have heard mentioned in the context of home recording — latency. Computers are very fast but there is still a slight delay between what you feed into them and what comes out. The degree of latency depends mainly on the ‘buffer size’. This is a small reservoir of data used by the computer to make sure that there are no breaks in the audio data stream when it turns its attention to something else such as accessing a hard drive or updating the screen graphics. The trick is to use the smallest buffer size at which your system will still run reliably, without any breaks or glitches in the audio output. Even then, some plug-in effects and instruments increase the basic latency so are best switched off unit you come to do your final mix. Latency has no effect when mixing, only when overdubbing new parts, so you can use all your favourite plug-ins when mixing and set the buffer size as high as you need.

Again Roland’s engineers have made advances using their VS Streaming Driver technology to maintain reliability and keeping latency as low as possible. There’s also a direct monitoring control that allow the actual input to be heard in the headphone or speaker output (or a mix of the direct input and any signal coming back from the computer) when recording in which case you hear no latency at all. Of course vocalists often perform better if they have a bit of reverb in their headphones so Roland have included a DSP reverb as part of the UA1610 that can be applied to the direct monitor signal. And, if different performers need different headphone mixes, no problem as the Control Panel software allows you to set up four different headphone mixes — just connect the corresponding line outputs to your own headphone amplifiers. Why four different headphone mixes? Different musicians usually want to hear a different balance of the other performers. For example, vocalists need to hear the harmonic content and other vocalists clearly. The drums and bass, however, may not be as high of a priority. Bass players need more of the drummer. Taking time to set the correct headphone mix for each performer can lead to a much better performance.

Playing Guitar

Another trick professionals often use is to apply some compression to the signal being recorded to even out its level. A compressor can be thought of as increasing the level of quieter sounds or reducing the level of loud peaks. The UA1610 has built-in compressors so that you can achieve the same results without expensive studio hardware.

The software driver (available for both Windows and macOS) comes on a CD ROM. If your computer doesn’t have a CD ROM drive, however, you can download the latest version from the Roland web site. The software also includes an on-screen control panel to provide easy access to any parameters you may need to adjust.

Sound quality is related to both the quality of the analogue electronics, such as microphone preamps, and the digital audio recording format. Today any serious audio gear needs to record at 24-bit resolution, and though the sample rate of a standard CD is 44.kHz, those who are very serious about sound quality may choose to record at double or even four time this sample rate. Many interfaces support sample rates up to 96kHz but the Studio Capture can run at up the 192kHz if you need it to. Even at the basic 44.1kHz sample rate, the audio recording quality is significantly better than that of even the most costly tape machines.

Selecting recording software (DAW)

There’s a wide choice of recording software for both macOS and Windows, ranging from relatively simple ‘lite’ packages to extremely sophisticated pro products. We term these DAWs or Digital Audio Workstations, and most will work seamlessly with the Roland Studio Capture. Whenever somebody asks me which one they should buy, I usually tell them to see what their friends use as that way they get free ‘tech support’ while they are still finding their way.

Other than that, if you plan to add vocals or acoustic instruments, you’ll need at least one microphone. In order to hear what you’ve recorded, you’ll also need some accurate headphones and a pair of studio monitor speakers. “Active” studio monitors are the most common choice as all the necessary electronics, including the amplifiers, is built right into the speaker cabinet. This makes setup very easy.

Studio Monitors and Room Acoustics

If your studio is set up in a small domestic room, don’t be tempted to buy speakers with a hugely impressive bass end as they tend to produce misleading results in small spaces. For rooms up to around 3 metres long I’d suggest monitors with between 4 and 6 inch diameter bass drivers. In rooms up to 5 metres long, a monitor with a 6 to 8 inch bass driver would be appropriate. Systems with large subwoofers are only really suitable for large studio spaces.

You should set up your speaker system as symmetrically as possible. Avoid placing the monitors in corners as that can cause problems with bass frequencies. Monitor stands can improve the sound where the monitors are standing on a desk and they also help to get the speakers at the right height for your ears. Ideally the tweeters should be aimed towards your forehead when you are leaning back in your chair. This can often be achieved by angling the speaker supports. Avoid wobbly speaker stands as monitor speakers needed to be mounted on something stable to deliver their best performance. Most monitors come with useful advice on positioning so always read the manual as there may be rear panel controls to adjust according to your room type and speaker position.

The way you arrange your room can make a big difference to the accuracy of your monitors. In a small to medium sized domestic room, always arrange the monitors to aim down the longest axis of the room. Square rooms tend to introduce more unevenness in the bass response of monitors, especially if the width of the room is similar to its height. If you have to work in a small square room, always use headphones to confirm what the bass end of your mixes really sounds like and be aware that if you sit near the center of a small, square room, the amount of bass you hear from your speakers is likely to be less than the real level of bass in your mixes.

recording

While there’s little you can do to even out the bass response of speakers set up in small rooms, controlling the mid and high frequencies is much easier and well worth doing. The simplest solution is to use acoustic foam panels of between 50 and 100mm thick. You only need a few of these to make a big difference and the best place to put them is at the so-called mirror points. If you get a friend to hold a mirror flat against the wall while you sit in your usual mixing position, any place you can see a reflection of either speaker is where an acoustic panel needs to go. Usually this means having a panel either side of you at head height and slightly forward of your position with another panel or two behind the speakers. There might also be a benefit in putting a couple on the back wall if it isn’t already occupied with shelves or furniture. In rooms with low ceilings, an extra panel on the ceiling above you, again at a mirror point can help.

While you can fix foam panels to the wall using spray adhesive, taking them down again can be a messy job. One solution is to glue a couple of old CDs to the back of the panel close to the top edge, then hang the panel on a couple of pins or screws positioned to line up with the center holes of the CDs.

Recording and Mixing Vocals

Recording vocals is actually not difficult as long as you take a few precautions to reduce sound reflections from around the room as these can make your recordings sound boxy. Always use a pop screen in front of the mic, otherwise blasts of air when enunciating P and B sounds can cause loud popping that is almost impossible to remove at a later stage.

In terms of selecting a microphone for recording vocals, please see the “Microphone Basics” section below.

You may also have seen commercial curved screens that go behind the mic, designed to help dry up the room acoustics. They can help, but in my experience, it’s the reflections from the wall behind you that pass over your shoulders that cause the biggest problems. An easy solution is to set up a spare tall boom mic stand in a T shape behind the singer and hang a duvet over it. This will really help to clean up the sound, which is important as adding compression further down the line will only emphasizes poor acoustics. You can still put a screen (or another duvet) behind the mic too as every little bit helps.

For recording acoustic instruments such as guitars, you’ll find lots of advice on-line and in magazines. I’d suggest you experiment further to find the best place for the microphone as that will vary depending on the instrument itself and on room acoustics. If you listen on headphones as you move the mic around, you’ll soon find that magic ‘sweet spot’. The more effort you put into getting the sound right at source, the less you’ll have to rely on plug-ins to try and rescue a poor recording.

While there are hundreds of plug-ins that do all kinds of wondrous things, the four you need to concentrate on are Equalisation (EQ), Compression, Delay/Echo and Reverb. If you’ve taken care with your recording, you shouldn’t need a lot of EQ to make things sound good. It can be helpful, however, to use low-cut EQ to remove unwanted low frequency sounds such as vocal breaths, passing traffic or floor vibrations. The easy way to set this up is to gradually increase the frequency of your low cut filter until you can just hear it affecting the voice or instrument, then back it off just slightly. It is a simple trick but it can make your mixes sound a lot cleaner.

Recording microphone

Reverb and echo can make a vocal sit more naturally with the backing instruments but be aware that too much reverb can take away the sense of presence and closeness in a vocal part so use it sparingly. A common technique is to combine a very subtle delay/echo treatment with a hint of reverb so take some time to experiment. Always make your final adjustments to effects and EQ with the whole track playing as the way something sounds in the final mix depends on what else is playing at the same time. What sounds like too much effect in isolation may sound like too little when everything is playing.

Compression is used to even out the level of sound such as vocals, acoustic guitars and basses. You can think of a compressor as something that reduces the level difference between quiet and loud sounds. This helps create a more even sound but it can also bring up the level of room ambience and background noise. For that reason don’t overdo compression — use you DAW’s mix automation to take out the worst of the level changes, then add compression to give your track a final polish.

A typical compressor has a threshold control to set at what level compression starts to take place. Below it, no gain changes are applied. Once the signal is above the threshold, the ratio controls sets how much gain reduction is applied. Basically the higher the ratio, the stronger the compression effect. The attack and release controls set how quickly the compressor responds to changes in input level but you’ll often find presets or suggested settings for different applications. In that case the only control you still need to adjust is threshold; the lower the threshold the more gain reduction is applied as more of the signal is above the threshold. There’s usually a make-up gain control to restore any overall level lost by compression.

While you are gaining experience in mixing, the main thing to concentrate on is the balance between the parts. To avoid escalating levels, it is usually best to turn down the levels of parts that are too loud rather than keep turning up the things that seem too quiet. It can help to start out with just the bass and drums to create a solid foundation for your mix, then add the other parts as needed. I favor bringing in the vocals fairly early on too so that as the other instruments are introduced, they are not allowed to obscure the vocals. One tip that the pros often pass on is to listen to the overall balance from outside the room with the door open as anything that is too loud or too quiet suddenly becomes very obvious. This really works.

Finally, always check out your mixes a day or two later on as many speaker systems as possible, including the car stereo. This will show up any balance issues that might be down to your studio acoustics allowing you to go back and tweak your mixes as necessary.

Microphone Basics

There are three basic types of microphone: moving coil dynamic, capacitor (also called condenser) and ribbon; they all do the job of converting sound into an electrical signal but work on different principles and have different strengths and weaknesses.

Most common are moving coil dynamic microphones, especially popular in live sound because of their affordability, robust construction and the fact that they require no electrical power to operate. Most moving coil microphones start to become inefficient at frequencies above 16kHz or so because of the inertia of their moving parts. Even so they are popular for recording drums, electric guitars, brass instruments and some vocalists choose them for studio work simply because they like the sound.

Capacitor microphones are very popular for studio recording as they are able to respond to high audio frequencies much better than moving coil microphones. Electronic circuitry is required to generate an audio signal from the diaphragm’s movement, which is why capacitor microphones need electrical power to operate. Most mic preamps, mixing consoles and serious audio interfaces with mic inputs provide the necessary 48V phantom power that is fed via a standard (balanced) XLR mic cable. Note that valve or tube microphones invariably have their own external high voltage power supply so phantom power is not required.

As a very general rule, small diaphragm capacitor microphones are used for instrument recording where a higher degree of accuracy is required while large diaphragm models tend to have more of an inbuilt tonal ‘character’ and are often chosen for vocal recording.

Ribbon mics are less popular than capacitor or moving coil microphones as they produce a very low electrical output and are less robust. Where they do score is that they produce a smoother-sounding high end than most capacitor microphones making them popular for some vocal styles and for use as drum overheads where they can produce a less brash cymbal sound than capacitor mics.

Polar Patterns

How a microphone responds to sound coming from different directions is defined by what we term its polar pattern. There are three main polar patters: Omnidirection (or omni), cardioid (or unidirectional) and figure-of-eight. Some capacitor microphones have switch-selectable polar patterns.

Omnidirectional microphones pick up sound equally well from any direction and tend to have a very natural sound.

Unidirectional mics pick up sound mainly from one direction. Several widths of the cardioid pattern are available where narrower settings are usually termed hyper-cardioid or super-cardioid. Most studio vocals are recorded using large diaphragm cardioid pattern, capacitor microphones. The majority of moving coil dynamic mics have a unidirectional polar pattern.

The figure-of-eight pattern mic is equally sensitive both front and back, but completely insensitive at 90 degree off-axis. Figure-of-eight mics are often used for specialist applications but can also be very helpful where sound spill from a specific direction needs to be avoided. Essentially you just need to aim the ‘deaf’ 90 degree axis towards the sound you don’t want to pick up. Most ribbon mics have a figure-of-eight polar pattern.

Proximity Effect

All conventional directional mics exhibit the so-called proximity effect, which results in a significant bass boost when the microphone is used with the sound source closer than 150mm or so. Only omni pattern mics and specialised cardioid mics incorporating sophisticated design innovations are free from the proximity effect.

Presence Peak

The frequency response of a perfectly accurate microphone would be perfectly flat across the human hearing range and some of the more sophisticated measurement mics get very close to this ideal. However, for music recording and vocals in particular, microphones are often designed to emphasise those parts of the audio spectrum that help make vocals sound brighter or more airy. The term ‘Presence Peak’ is often used to describe the region where frequencies are boosted. There is no right or wrong voicing for a vocal microphone — the most important thing is to choose a microphone that suits the voice being recorded on an artistic level.

band recording

Conclusion

Recording an entire band at home can seem like a daunting task. With a bit of guidance and some strategic pieces of equipment, however, you’ll quickly find that it’s a very achievable goal. If you’ve followed the advice in this guide, you’ll be well on your way to recording your next demo track yourself!

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