Turntables of life

Sound Systems: Mono versus Stereo

2015-01-18T13:00:00.000-08:00

It's a common question in many church sound system projects, "Will our system be mono or stereo?" What usually follows is a lengthy discussion about the applicability of mono sound systems versus stereo systems, the difference between two channel sound systems and stereo sound systems, the benefits and limitations of each, the architectural constraints and program requirements that will affect the decision, and the cost implications of each.

It is very apparent that everyone has their own interpretation of the terms 'mono' and 'stereo', influenced by their own experiences and expectations. Translating one's experience with home audio systems or project studios into a large venue like a church or a theatre often takes an adjustment in conceptual thinking, so we always have to provide a reference point for the discussion of mono, stereo and two channel sound systems. Let's start off with mono.

MouseOver - Single channel versus two channel loudspeaker coverage

Compare the single channel centre
cluster coverage to the following:

1. Single centre cluster
2. Both left & right speakers
3. Left channel speakers only
4. Left channel with left "stinger"
5. Left channel with right "stinger"
6. Both channels with "stingers"

Mono

Mono or monophonic describes a system where all the audio signals are mixed together and routed through a single audio channel. Mono systems can have multiple loudspeakers, and even multiple widely separated loudspeakers. The key is that the signal contains no level and arrival time/phase information that would replicate or simulate directional cues. Common types of mono systems include single channel centre clusters, mono split cluster systems, and distributed loudspeaker systems with and without architectural delays. Mono systems can still be full-bandwidth and full-fidelity and are able to reinforce both voice and music effectively. The big advantage to mono is that everyone hears the very same signal, and, in properly designed systems, all listeners would hear the system at essentially the same sound level. This makes well-designed mono systems very well suited for speech reinforcement as they can provide excellent speech intelligibility.

Stereo

True stereophonic sound systems have two independent audio signal channels, and the signals that are reproduced have a specific level and phase relationship to each other so that when played back through a suitable reproduction system, there will be an apparent image of the original sound source. Stereo would be a requirement if there is a need to replicate the aural perspective and localization of instruments on a stage or platform, a very common requirement in performing arts centres.

This also means that a mono signal that is panned somewhere between the two channels does not have the requisite phase information to be a true stereophonic signal, although there can be a level difference between the two channels that simulates a position difference, this is a simulation only. That's a discussion that could warrant a couple of web pages all by itself.

An additional requirement of the stereo playback system is that the entire listening area must have equal coverage of both the left and right channels, at essentially equal levels. This is why your home stereo system has a "sweet spot" between the two loudspeakers, where the level differences and arrival time differences are small enough that the stereo image and localization are both maintained. This sweet spot is limited to a fairly small area between the two loudspeakers and when a listener is outside that area, the image collapses and only one or the other channel is heard. Living with this sweet spot in your living room may be OK, since you can put your couch there, but in a larger venue, like a church sanctuary or theatre auditorium, that sweet spot might only include 1/3 the audience, leaving 2/3 of the audience wondering why they only hear half the program.

In addition a stereo playback system must have the correct absolute phase response input to output for both channels. This means that a signal with a positive pressure waveform at the input to the system must have the same positive pressure waveform at the output of the system. So a drum, for instance, when struck produces a positive pressure waveform at the microphone and should produce a positive pressure waveform in the listening room. If you don't believe that this makes a tremendous difference, try reversing the polarity of both your hifi loudspeakers some day and listening to a source that has a strong centre sound image like a solo voice. When the absolute polarity is flipped the wrong way, you won't find a stable centre channel image, it will wander around away from the centre, localizing out at both the loudspeakers.

Two Channel

This is what many people mistake for stereo sound systems, because there are two channels and a "stereo" console is connected to the front of the system, and stereo amplifiers and equalizers are used throughout the system. What is missing from most of these systems is uniform coverage of the entire listening area, and a minimal level and phase response difference for each channel's coverage of the listening area. To achieve proper loudspeaker coverage to replicate a stereo image in a large venue, it is necessary to have a loudspeaker system for each channel that can provide uniform coverage of the entire listening area while maintaining the directional cues. This is a very expensive, and sometimes impossible proposition.

Two channel systems usually suffer from having half the people in the listening area only hear half the audio program, which makes two channel systems a poor choice for music reinforcement. A large portion of the listeners hear a completely different music mix from other listeners. This is an all-too-common oversight in venues that are intended for music and entertainment, even high profile venues where they deserve better system designs. It tends to be a common misconception brought forward by people with a background in portable or live sound systems.

When a two channel system is used to reinforce a mono voice microphone, the seats either side of the room centreline, exactly between the two loudspeakers, also experience substantial variations in frequency response and uniformity of coverage due to interference and signal cancellation when identical signals arrive from the two channels at different times. This makes two channel systems especially ineffectual for speech reinforcement applications.

Left/Centre/Right

There are specialized applications for sound systems described as Left/Centre/Right configurations. This must combine the best of both worlds, right? Well it can, but the loudspeaker system must also be designed to the highest common denominator not the lowest. It is also important with an LCR system that the mix engineer understand which signals must be routed to which loudspeakers, and which signal routings will create problems. LCR systems also are not suitable for use in all room shapes and configurations.

LCR systems are common in drama theatres and large churches where there is a requirement for both mono speech reinforcement and music or sound effects cues to be localized or mixed with a particular perspective, with a stereo or stereo-like imaging. The three loudspeaker systems must each provide coverage of the entire seating area while maintaining level and directional cues, just like the mono and stereo systems described above. There are some clever "cheats" a system designer can use to achieve the extended listening area for stereo coverage, and that invloves using "stingers" or infill loudspeakers.

Stingers

As shown above in the MouseOver demo, a stinger is a speaker positioned to provide coverage of a floor area that may be inaccessible from the left or right cluster mounting position.

In the example above, the left and right stingers are separate from the centre cluster, and would be fed from a signal delay so that for people in the right side of the room, the signal from the left stinger would arrive at the same time as the sound from the left cluster would have if it could have reached that location. For people sitting near the front right, they would still hear the program from the left channel to maintain the integrity of the program content, even though the stereo image would be skewed to one side compared to the image perceived by the people sitting in the middle of the room. The speaker selection, as well as the level and delay setting of the stingers are all quite critical to their successful integration, if they are too loud, too late or too early they will detract from the image the rest of the audience hears. If you're going to try this at home study up on Haas Effect, setting levels and delays for precedence, as well as time domain measurement systems.

It is also possible to use some of the components of the centre cluster as the stingers, especially with the advent of lower cost DSP systems that allow a matrix of signal delays to be developed for each input to the unit. Even a few years ago this would have been exceedingly expensive and complicated (and nearly impossible to explain), now it can be programmed quickly on a computer screen and be made to work quickly and easily. This approach works best when the centre cluster has similar sounding loudspeaker components to the left/right clusters.

Which one's the best?

As with many questions about sound systems, there is no one right answer. A well designed mono system will satisfy more people than a poorly designed or implemented two channel sound system. The important thing to keep in mind is that the best loudspeaker design for any facility is the one that will work effectively within the, programmatic, architectural and acoustical constraints of the room, and that means (to paraphrase the Rolling Stones) "You can't always get the system that you want, but you find some times that you get the system that you need." If the facility design (or budget) won't support an effective stereo playback or reinforcement system, then it is important that the sound system be designed to be as effective as possible, even if that means giving up a desirable program requirement like stereo.

Digital and Analog Mixers

2015-01-18T10:12:00.001-08:00

Digital and Analog Mixers

The mixing console or "mixer" is a central component of most sound systems. In fact, the mixer used will have a large influence on the operability and efficiency of the entire system. One of the first decisions that will have to be made when designing a new sound system is whether a digital or analog mixer will be used.
Digital Mixers are Rapidly Gaining Ground
Described in the simplest terms, the difference between analog and digital mixers is whether audio signals are internally processed in their original analog form or converted to and processed in digital form. Digital mixers are rapidly gaining popularity for their convenience, expandability, and resistance to noise (see "The Merits of Digital Sound" in Part 1). But there are still situations in which analog is preferred. Let's take a look at the features of each.

Digital Mixer Features

Five of the most important features of digital mixers are listed below. All of these features are related to the benefits described inThe Merits of Digital Sound" in Part 1. It might be easier to grasp the "big picture" if you go back and review that information as well.
Settings can be pre-programmed and recalled when necessary
Most digital mixers feature some kind of memory into which settings can be stored and then instantly recalled whenever they are needed. This can be a tremendous advantage in a banquet hall facility for example, where the required settings might change frequently for different types of events and different room configurations. Even if some fine tuning is required, the ability to simply recall a complete set of basic parameters that are close to what is needed can dramatically reduce the time and effort required to set up for an event. And if a mistake is made, it's easy to revert to the basic settings.

Mixing and processing features that only digital can provide

Digital technology has enabled the development of a number of mixing functions that were simply not available in analog systems. Automatic feedback suppression is one example. Automatic "ducking" that decreases the BGM level while an announcement is being made is another. More details will be provided in the "Automated Mix Functions" section.
Many digital mixers also feature built in signal processing functions such as effects that are designed to suit the mixer's intended applications. This makes it unnecessary to purchase extra external signal processing devices, thus significantly reducing overall system cost as well as installation space. This is a radical departure from analog systems for which additional equipment had to be purchased and installed to provide the signal processing functions required by each individual application. There are analog mixers that include some built-in processing functions, but unlike digital mixers in which all of the necessary processing can be implemented through software, additional processing capability has a direct influence on the cost and physical size of an analog mixer. For the same mixing and processing capabilities, a digital mixer will almost always offer superior economy.

Expansion and external devices

Expandable digital mixers make it possible to connect to a wide variety of external devices. Touch panel controllers, for example, can make day-to-day operation of a specified set of frequently used parameters easy for even inexperienced users, while keeping parameters that should not be changed hidden. More details will be provided in the "Peripheral Equipment" section.

Noise-resistant digital transmission

A sound system that is based on a digital mixer that uses digital transmission for audio signals will be highly resistant to induced noise, as described in the "The Merits of Digital Sound" section in Part 1. Since noise becomes more of a problem with longer transmission distances, the benefits of digital transmission in an audio installation increase as the size of the facility increases.

Multiple functions in small spaces

Although analog mixers usually have one control per function, in a digital mixer it is possible to assign numerous functions to a single control, with function switching either via physical controls or virtual controls on a display. This makes it possible to pack a large number of functions into a relatively small space. If you compare analog and digital mixers that offer the same number of channels, you'll see that the digital mixers tend to be significantly smaller.

Analog Mixer Features

Analog mixers have two main advantages.

* Lower cost for a limited set of features

If only a few channels with a basic set of mixing features are required, a simple analog mixer may be a more economical choice than a digital mixer.

*Easy operation for first-time users

As mentioned above, analog mixers usually have one control per function, all of which are visible and directly accessible via the control panel. The control layout logically follows the mixer's signal flow and is therefore relatively easy to understand. This type of logical, easy operation can be an advantage in public halls and schools, for example, where a variety of people, some having little or no previous experience, may need to operate the system. In such situations it may be necessary to protect controls and functions that should not be changed with security covers.
That covers the basic differences between digital and analog mixers. Depending on the application, the choice of a digital or analog mixer can make a large difference in the operability and overall economy of the system. Next we'll look at another important consideration for mixer selection: configuration.

Whats the Difference Between a DJ Mixer and a Live Mixer?

2015-01-18T09:58:00.001-08:00

For a disc jockey or an audio technician, combining signals or sounds is both an art and a craft. These people know how to effectively direct and mix sounds so the audience can hear a harmonious output. Knowing how to wield a mixer, the tool of the trade, is the expertise of a professional disc jockey or audio technician. But even aspiring DJs or regular music lovers can learn how to use a mixer to create great music mixes.
There is a variety of mixers out on the market, and it is important to know what type of mixer is suitable for individual needs. A studio mixer, for instance, is a great tool for mixing sounds and recording a final output. Other types of mixers are a DJ mixer and a live mixer. Is there difference between the two mixers? Buyers must know the difference, as well as the different parts and functions of a mixer, in order to make a wise judgement when choosing the right type of mixer to use for either work or leisure.

Difference Between a DJ Mixer and a Live Mixer

All mixers have the same basic function: to combine signals. Mixers only vary because of their intended applications. Essentially, a mixer can either be a "live mixer" when it used in a live setting or a "studio mixer" when used for recording mixes and performances in a studio. A live mixer combines signals from different inputs and then sends it to a finite number of speakers, such as the main speakers, balcony speakers, and monitors. A studio mixer, on the other hand, combines signals to create a recorded master output.
Furthermore, a DJ mixer, which is mostly used in a live function, is in essence a live mixer. Usually, the two mixers both have the basic features, such as stereo channels, an input selector, a gain control, equalisers, and a microphone input. But what sets apart a DJ mixer from a typical live mixer is its ability to route a non-playing source to headphones, as well as the crossfader function that allows easier transition from one source to another.
Basic Parts of DJ and Live Mixer
Both live and DJ mixers have these basic parts: stereo channels, input selector, gain control, equaliser, cue switch, and microphone input. Although they are basic parts, they are important in the process of directing and mixing music during a live function, such as a party or a concert. A live or DJ mixer has two to six stereo channels for connecting source devices, such as a CD player, and sending the signals for mixing.
A DJ mixer has a phono input with RIAA equalisation, which is an essential specification if a DJ wants to attach a turntable to the mixer. Each channel is arranged vertically in both live and DJ mixers, and each channel has a knob or switch for selecting inputs. An equaliser is also present in any type of mixer so a DJ can fade in or fade out the different parts of a track. In mixing two tracks, one common technique is to mute the bass of one track and use only the other. This way, the bass lines of the two tracks do not clash.
There are mixers that feature separate knobs for the low, mid, and high frequency ranges, and controls like a balance knob, aux-sends for external effects unit, and built-in sound effects. A cue switch is important in a DJ mixer because it sends a signal to headphones, allowing a DJ to listen to that signal without it affecting the master output. At the end of a channel strip is a fader that is useful in varying the volume in the final mix. There are also microphone inputs in both a DJ mixer and a live mixer. Larger live mixing consoles have inputs for microphones and source devices to support a band setup.

Analogue Mixers vs. Digital Mixers

In the quest of finding the right DJ or live mixer, another question can come to mind: Should the mixer be analogue or digital? To have a better judgement in this matter, it is necessary to discuss the common desirable qualities in a mixer, such as fast signal processing, ease of use and great sound quality, as well as how an analogue or a digital mixer rate in these qualities.
Ease of Use
In terms of knob-, fader-, or button-function ratio, an analogue mixer wins over a digital mixing console. An analogue mixer has one knob, fader, or button per function, which makes it is easier and faster for a user to perform tasks on the console. A digital mixer, on the other hand, has fewer knobs, buttons, and faders; it has a simpler and clean look that does not intimidate a user.
Most digital mixers make up for the lack of physical controls with virtual pages or layers that change the fader banks into separate controls. The controls are useful in varying equalisation; however, this feature can be quite confusing for a beginning user. It is also harder to make a hardware patch change on a digital mixing console because of internal reassignment, making convenient groupings of inputs appear near each other at the fader bank. Modern digital mixers allow users to store and edit live mixes in a computer, which makes it easier for a DJ to create a remix.
Sound Quality
Both analogue and digital mixers make use of an analogue preamplifier that increases the strength of a signal at line level, so when it comes to sound quality, neither mixer beats the other. For overloading or clipping concerns, both mixers have a solution, with the digital mixer using a digital converter to break down an audio stream into small but measurable increments. The faint hissing sound to the main outputs can be corrected through good gain stage management in an analogue mixer and by low-level gating in a digital mixer.
Signal Processing
It is no secret that a digital mixer delays the release of a signal from the speakers because of the conversion of analogue signals to digital ones. The delay, depending on a digital mixer model and functions that are currently engaged in the process, ranges from 1.5 to 10 milliseconds. Although the delay is not a problem to someone hearing the output from loudspeakers, it is quite confusing and disorienting to an artist wearing in-ear monitors.
Remote Control Capability
Although analogue mixers has had the option of using a wired remote control since the 90s, it is the digital mixer that offers flexibility when it comes to remote controlling. Digital mixing platforms allow the use of both wired and wireless remote controls using a laptop or a tablet computer, especially if there is no need for quick changes in the mixing process during a live performance. There are computer networks linking different elements of a digital system to make it possible to manipulate devices even when they are far from the disc jockey or technician.
Buying a DJ or a Live Mixer on eBay
eBay is a great website to buy DJ mixers and live mixers because you have the option to choose from new, refurbished, and used mixers depending on your budget. Also, there are so many brand options for a DJ mixer or a live mixer on eBay, and you can easily filter your search to a certain brand that you fancy.
Searching on eBay is fast and easy. All you need to do is type the keywords "DJ mixers" or "live mixers" in the search bar to produce a listing of relevant results. You can also do a search based on features, such as an equaliser, USB connection, integrated effects unit, loop function, and more. Always remember to read the product description carefully to make sure that the specifications of a certain mixer is what you are really looking for. Also, check the shipping details of a product before purchasing to know when to expect it to arrive.

Conclusion

A quality mixer is any aspiring disc jockey’s dream. A live mixer or a DJ mixer, as opposed to a studio mixer, is usually a fixture in live events. It is such a versatile device; it allows not only mixing of tracks to produce a desirable output but also controlling of sounds that come out of the speakers to make an event more organized. Both a live mixer and a DJ mixer have the same basic parts, including the stereo channels, input selector, gain control, equaliser, cue switch, microphone input, and headphone input. Knowledge of the similarities and differences of a DJ mixer and a live mixer is essential in understanding their functions.
Moreover, it is also helpful to know the advantages and disadvantages of a digital DJ or live mixer and an analogue DJ or live mixer in order to know the best option for a particular purpose. Knowing a mixer inside and out may take years of experience, but knowing the different types of mixers and the basic parts are a first step to understanding how to maximise the functions of the device.

What Are the Differences Between All Those Audio Formats?

2012-03-06T02:31:00.014-08:00

Digital audio has been around a very long time so there’s bound to be a plethora of audio formats out there. Here are some of the more common ones, what differentiates them, and what to use them for.

Before we talk about everyday audio formats, it’s important you understand the basics, and that means understanding PCM. After that, we’ll tackle compressed formats.

PCM Audio: Where It All Starts

Pulse-Code Modulation was created back in 1937 and is the closest approximation of analog audio. That is, an analog waveform is approximated in regular intervals. PCM is characterized by two properties: sample rate and bit depth. Sample rate measures how often (in times per second) the amplitude of the waveform is taken, and the bit depth measures the possible digital values. In terms of audio formats, this is pretty much the foundation.

True sound, in the real world, is continuous. In the digital world, it’s not. Somehow this is more confusing with audio than with video, so let’s look at video as a point of comparison. What we interpret to be “motion” or think of as “fluid” and constantly-moving is, in actuality, a series of still pictures. In that same way, the amplitude of sound waves in a digital format isn’t “fluid” or constantly changing. It’s changing based on certain criteria at pre-defined intervals.

I know there’s a lot here that may not be second-nature unless you’re an engineer, physicist, or an audiophile, so let’s pare it down further with an analogy.

Let’s say that the water flowing from an open faucet is your “analog” audio source. The temperature of the water we can compare to the amplitude of an audio wave; it’s a property that needs to be measured so you can enjoy it properly. Sampling is the number of times per second you dip your finger into the flowing water. The more often you dip your finger into it, the more “continuous” the temperature changes become. If you stick your finger into the running water 44,100 times per second, it’s almost like keeping your finger under there the whole time, right? That’s the basic idea behind sampling.

Bit depth is a little trickier. Instead of using your finger, let’s say you used a really crapper thermometer. It basically said “Hot” for anything above room temperature and “Cold” for anything below. Regardless of how many times you dipped it into the water, it wouldn’t really give you much useful information. Now, if instead of just 2 options, let’s say the thermometer had 16 possible values which you could use to gauge the water temperature. More useful, right? Bit depth works the same way, in that higher values allow more dynamic changes in sound amplitude to be accurately portrayed.

As previously mentioned, PCM is the foundation for digital audio, along with its variants. PCM attempts to model a waveform, in as much of its uncompressed glory as possible. It’s special, it’s ready to be stuck in a digital signal processor, and it’s more or less universally playable. Most other formats manipulate audio via algorithms, so they need to be decoded while playing. PCM audio is considered “lossless,” it is uncompressed, and therefore, takes up a lot of hard drive space.

The Uncompressed Bunch: WAV, AIFF

Both WAV and AIFF are lossless audio container formats based on PCM, with some minor changes in data storage. PCM audio, for most people, comes in these formats, depending on whether you use Windows or OS X, and they can be converted to and from each other without degradation of quality. They are both also considered “lossless,” are uncompressed, and a stereo (2-channel) PCM audio file, sampled at 44.1 kHz (or 44100 times per second) at 16 bits (“CD quality”) amounts to roughly 10 MB per minute. If you’re recording at home for the purposes of mixing, this is what you want to use because it’s full quality.

Lossless Formats: FLAC, ALAC, APE

The Free Lossless Audio Codec, Apple Lossless Audio Codec, and Monkey’s Audio are all formats which compress audio, much in the same fashion that anything is compressed in digital world: using algorithms. The difference between zipped files and FLAC files is that FLAC is designed specifically for audio, and so has better compression rates without any loss of data. Typically, you’re seeing about half the size of WAVs. That is, a FLAC file for stereo audio at “CD quality” runs roughly 5 MB per minute.

The up-side is that if you want to do audio manipulation, you can convert back to a WAV without any loss of quality. If you’re an audiophile and listen to a lot of music with dynamic ranges, these formats are for you. If you’ve got a great set of speakers, cans, or earbuds, these formats will bring out the tones to showcase them.

Lossy Formats: MP3, AAC, WMA, Vorbis

Most of the formats you see in day-to-day use are “lossy”; some degree of audio quality is sacrificed in exchange for a significant gain in file size. An average “CD quality” MP3 runs about 1 MB per minute. Big difference compared to PCM, no? This is called compression, but unlike with lossless formats, you can’t really get that quality back once you strip it in lossy formats. Different lossy formats use different algorithms to store data, and so they typically vary in file size for comparable quality. Lossy formats also use bitrate to refer to audio quality, which usually looks like “192 kbit/s” or “192 kbps.” Higher numbers means that more data is being pumped out, so there’s more preservation of detail. Here are some details for the more popular formats.

MP3: MPEG 1 Audio Layer 3, the most common lossy audio codec today. Despite a heap of patent issues, it’s still incredibly popular. Who doesn’t have MP3s lying around?
Vorbis: A free and open-source lossy format used more often in PC games such as Unreal Tournament 3. FOSS fans, such as many Linux users, are bound to see plenty of this format.
AAC: Advanced Audio Coding, a standardized format now used with MPEG4 video. It’s heavily supported because of its compatibility with DRM (e.g. Apple’s FairPlay), its improvements over mp3, and because no license is needed in order to stream or distribute content in this format. Apple fans will probably have plenty in AAC.
WMA: Windows Media Audio, Microsoft’s lossy audio format. It was developed and used to avoid licensing issues with the MP3 format, but because of major improvements and DRM compatibility, as well as a lossless implementation, it’s still around. It was really popular before iTunes became champion of DRMed music.

Lossy formats are what you use for all of the stuff you listen to and store. They’re designed to be an economy of hard drive space. Which format you choose depends on what digital audio player you use, how much space you have, how big of a quality nitpicker you are, and a bunch of over variables. Nowadays, computers will play anything, most audio players (except Apple’s, of course) will do multiple lossy formats, and more and more do FLAC and APE. Apple sticks to MP3, ALAC, and AAC.

Isn’t Audio Quality Subjective?

Absolutely, it is. Ultimately, it’s your ears that are consuming most of this stuff, but that’s more reason to think of quality seriously. When I first started creating my digital music collection, I couldn’t really tell the difference between 128kbit MP3s and audio CDs. To my ears, there was no noticeable difference. Over time, however, I noticed that 256 kbit sounded much better, and after I got a really nice (and expensive!) set of headphones, I went back to audio CDs full time! It also depends on the genre of music.

There are a LOT of variables here, folks, make no mistake about that. It took a while before I settled on using FLAC for some music and 320kbps MP3 for the rest. The point I’m trying to make is that you should experiment to see what works best for you and your music, but be aware that as your tastes change, your perceptions, your equipment, and the importance of quality will, too.

And all of this stuff get even trickier when you’re not just talking about music, but about voice tracks, sound effects, white and brown noise, etc. There’s a whole world of sound out there, so don’t get discouraged! By learning what you can and listening for yourself, you can use this info to your advantage in your future audio projects. I’ll leave you with some of the best advice I’ve ever gotten: “do what just plain sounds good.”

For every (EQ) action there is an equal and opposite (EQ) reaction

2012-03-05T13:18:00.007-08:00

For example, if you add too much to the 2 kHz EQ band, eventually anything will sound thin and harsh. If you compensate by adding some 100 Hz to warm it up, you’ll end up with “scooped mids” and the sound will be thin and lack body. So you add some 500 and suddenly you’re back where you started, but it all sounds a bit processed and un-natural.

So I’ll finish with a final rule of thumb for you:

Less is more !

- and an outstanding resource, to an interactive frequency chart with even more rules of thumb and suggestions for the best EQ band to use eachinstrument. I don’t agree with all of them, but as Joe said in his video, there are no rules in audio – use your ears !

Instrument Frequency ranges and some crucial EQ bands and what they sound like

2012-03-05T12:38:00.004-08:00

50-60 Hz

-Thump in a kick drum
-Boom in a bassline
-Essential in dub, dubstep and reggae !
-Too much and you’ll have flapping speakers and a flabby mix
-Too little, and the mix will never have enough weight or depth

100-200 Hz

-This EQ band adds punch in a snare
-Gives richness or “bloom” to almost anything
-Too much makes things boomy or woolly
-Too little sounds thin and cold

200-500 Hz

-Crucial for warmth and weight in guitars, piano and vocals
-Too much makes things sound muddy or congested
-Too little makes them thin and weak

500-1000 Hz

-One of the trickiest areas
-Gives body and tone to many instruments
-Too much sounds hollow, nasal or honky
-Too little sounds thin and harsh

2 kHz

-Gives edge and bite to guitars and vocals
-Adds aggression and clarity
-Too much is painful!
-Too little will sound soft or muted

5-10 kHz

-Adds clarity, open-ness and life
-Important for the top end of drums, especially snare
-Too much sounds gritty or scratchy
-Too little will lack presence and energy

16 kHz

-Can add air, space or sparkle
-Almost too high to hear
-Too much will sound artificial, hyped or fizzy
-Too little will sound dull and stifled

What is Audio mastering ?

2012-03-05T12:07:00.003-08:00

Mastering, a form of audio post-production, is the process of preparing and transferring recorded audio from a source containing the final mix to a data storage device (the master); the source from which all copies will be produced (via methods such as pressing, duplication or replication). Recently, the format choice includes using digital masters although analog masters, such as audio tapes, are still being used by the manufacturing industry and by a few engineers who have chosen to specialize in analog mastering.

In order to make a deterministic process, mastering requires critical listening; therefore, it cannot be achieved without the presence of a mastering engineer. There are software mastering tools available to facilitate this last step, but results still depend upon the accuracy of speaker monitors. In addition, "music mastering" engineers may also need to apply corrective equalization and dynamic enhancement in order to improve upon sound translation on all playback systems.

>The studio<

The music mastering studio is very different from a normal audio recording studio. In fact, all the equipment and gear found in most recording and mixing studios can actually hinder the acoustics of a room to accurately monitor sound. Thus, the correct room acoustics and arrangement of the equipment inside a mastering studio is an important factor since the mastering engineer (ME) needs to hear each mix in detail. This room design should be non-environmental or with a minimum room interference. By working with an experienced mastering engineer, the recording artist is also open to alternative opinions and technical advice.

>Process<

The source material, ideally at the original resolution, is processed using equalization, compression, limiting, noise reduction and other processes. More tasks, such as editing, pre-gapping, leveling, fading in and out, noise reduction and other signal restoration and enhancement processes can be applied as part of the mastering stage. This step prepares the music for either digital or analog, e.g. vinyl, replication. The source material is put in the proper order, commonly referred to as assembly (or 'track') sequencing.

If the material is destined for vinyl release, additional processing, such as dynamic range reduction, frequency dependent stereo–to–mono fold-down and equalization, may be applied to compensate for the limitations of that medium. Finally, for compact disc release, Start of Track, End of Track, and Indexes are defined for disc navigation. Subsequently, it is rendered either to a physical medium, such as a CD-R or DVD-R, or to a DDP file set, the standard method of secure delivery for CD and DVD replication masters. The specific medium varies, depending on the intended release format of the final product. For digital audio releases, there is more than one possible master medium, chosen based on replication factory requirements or record label security concerns. Regardless of what delivery method is chosen, the replicator will transfer the audio to a glass master that will generate metal stampers for replication.

The process of audio mastering varies depending on the specific needs of the audio to be processed. Mastering engineers need to examine the types of input media, the expectations of the source producer or recipient, the limitations of the end medium and process the subject accordingly. General rules of thumb can rarely be applied.

Steps of the process typically include but are not limited to the following:

Transferring the recorded audio tracks into the Digital Audio Workstation (DAW) (optional).
Sequence the separate songs or tracks (the spaces in between) as they will appear on the final release.
Process or "sweeten" audio to maximize the sound quality for its particular medium (e.g. applying specific EQ for vinyl)
Transfer the audio to the final master format (i.e., CD-ROM, half-inch reel tape, PCM 1630 U-matic tape, etc.).

Examples of possible actions taken during mastering:

Editing minor flaws
Applying noise reduction to eliminate clicks, dropouts, hum and hiss
Adjusting stereo width
Adding ambience
Equalize audio across tracks
Adjust volume
Dynamic range expansion or compression
Peak limit

A Club Track's Frequency Map

2012-03-05T11:55:00.002-08:00

Use it as your wallpaper so you'll always have refrence when balancing up those club bangers!

The difference between mixing and mastering

2012-01-30T09:23:00.000-08:00

Are you still confused about the difference between mixing and mastering?

Do you think you’re mastering when you use a limiter on your mix bus? Do you avoid the term ‘mastering’ because you’re doing it yourself? Do you have trouble explaining what you do when you’re finished mixing? The difference between mixing and mastering is becoming more and more blurred.

In simple terms: Mixing is what you’re doing when you’re balancing the individual elements of a song. Mixing starts with several individual tracks (usually one track per instrument or sound), and finishes with one (usually) stereo track – the ‘mixdown’ – that sounds like a combination of all the individual tracks.

You know you’re mixing when you’re working with the individual sounds within a song.

Mastering, on the other hand, is what you’re doing when you’re preparing a mixdown for duplication or publication. This is often within the context of a ‘release’ – a collection of several songs that are published as a package. Mastering starts with a mixdown for each song, and finishes with audio that is ready for the world to hear.

You know you’re mastering when you’re making the final adjustments to a mixdown before the audio is considered ‘final’ and ready for your audience.

Notice that I didn’t mention tools. The tools you use don’t define the process – you’re not mastering just because you’re using a ‘mastering’ limiter. You’re not mixing just because you’re loading plugins into a mixer window. Instead, the process defines the tools – it’s the intent of what you’re trying to achieve the counts. Choose whatever tools are necessary to get you there, regardless of how they’re labelled.

For example:

Sometimes a single is ready for publication as it comes out of the mixer (perhaps with the help of a limiter on the mix bus ). It doesn’t matter that the whole process happens within the mixing environment – mastering is choosing the start and end points and applying the mix bus processing. Hopefully it also included a reference to other comparable commercial releases.
Sometimes a song requires some compression on the mix bus. This can make the individual tracks gel together, and even produce a distinctive pumping sound. It doesn’t matter that this is achieved by processing the mix bus (or even the stereo mixdown) – mixing is focussing on the sound of the instruments to make them blend well and express the creative direction of the artist and producer.
Sometimes a song needs to be converted to MP3 (or other lossy format) for online distribution. The processing that happens as part of the conversion is a part (hopefully the last!) of mastering because its goal is to prepare the audio for distribution. It doesn’t matter whether this is achieved by an external program program or your built into your software’s export function.

If you confuse mixing and mastering, you’ll lose focus of what you’re trying to achieve. If you try to master when your mix isn’t finished, you’ll waste trying to solve problems on the mix bus that are best addressed in balancing individual elements within the mix. You’ll also be waste time fine-tuning a treatment (processing chain) for a mix that will later change. On the other hand, if you try to adjust the balance of instruments when you’re mastering, you’ll find it difficult to do it effectively without unintentionally changing other parts of the mix. If you try to ‘mix into’ a mastering chain, you’ll easily find yourself chasing your tail in an infinite loop of adjusting the mix, adjusting the mastering chain, adjusting the mix again to compensate, adjusting the mastering chain again…

It’s important to separate the two processes – not necessarily by using different tools (although that helps), but by being clear in your own mind where mixing ends and mastering begins. It will help focus your workflow goals and ultimately make your work more efficient, effective… and more fun!

Couple Of Guidelines On Mixing Electronic Music

2012-01-30T08:55:00.000-08:00

Mixing isn’t something that can be learned by overnight. I’ve been making electronic music over 10 years now and even though I can make decent sounding tracks, I still feel that I have a lot to learn. However it’s not rocket science either and anyone CAN learn to mix. You just need a tad of a patience and decent monitors or headphones. Practice makes perfect like they say.

In this article I give you a couple of tips on mixing that I’ve personally found helpful in my own music production. I hope you find them useful as well!

Preparing To Mix

I start the mixing process by dropping the master volume level to around -5.0dB. This is because I wan’t to leave some headroom to avoid clipping. When adding more sounds and instruments to the mix – especially those which have loud spikes – there’s a risk of clipping. If you export your song to a WAV and it’s clipping, it’s impossible to fix it afterwards. Therefore, decreasing the master volume at this point will leave room for those spikes and the clipping issue is avoided.

At this point it doesn’t matter if the mix doesn’t sound loud enough. Just turn up the volume of your speakers. And when you export (mixdown) your song to WAV, you can normalize it afterwards (that means increasing the amplitude of the WAV so that the loudest peak is at maximum possible level) and via limiter and compressor you can really make it loud. But this is done in the mastering process.

However, you might wan’t to consider using 32 bit floating point format when you export your song to WAV. That way you don’t have to worry about the clipping issue.

Okay. Before going to the actual mixing process, I usually have some sort of rough mix in my hands created already in the composing stage. But after that I usually do this: in the mixer tracks I drop every instrument level to INF except the kick drum. And the mixing process starts here.

Drums

I usually mix everything after the kick drum and I always start with the drums. So basically I start by setting the kick drum mixer track volume level to 0.0dB. My opinion, in most of the electronic music, the kick drum should be the dominant force. Then I mix the other instruments so that kick drum stands out of the most.

Also, I never pan the kick drum. I keep it always centered.

Few words about panning: Panning is a great and simple practice to widen the stereo image of a mix. With panning you can place different sounds around the stereo panorama. It’s also a good way to “separate” sounds that are on the same frequency range to avoid them messing with each other. This is very handy if you have a “busy” (lot of sounds and instruments) mix. I really recommend to practice the use of panning.

Sometimes – very rarely though – I layer two or more kick drums if I’m not satisified with the one I’m currently using. At this point, to make the kicks fit together, I need to edit the samples quite a lot by adjusting the attack and relase via the volume envelope and add some equalizing. Tuning may be needed as well. (Tuning is adjusting the pitch). But that is what I let my ears to decide.

There’s compression as well. Compression is basically automated gain control that reduces the dynamic range of sounds and it’s useful for many things such as making the whole mix sound louder. With compression you can modify the volume shape of single sounds as well – works well with kick drum. For example, you can add more “thump” or “snap” to your kick and control how the tale of the kick (oomph) behaves.

I have to say that I rarely compress kick drums. I have a huge collection of kick drum samples and most of the times I can find a sample that suits my needs as is. So if I’m not satisfied with a kick sample, I usually don’t start playing around with the compression – or layering or tuning in that matter – instead I change it to another kick drum sample.

Ok. Now that the kick drum is at level 0.0dB and the rest are INF, I start to increase the levels. First, the claps (or snare). I usually set it few dB’s below the kick drum level, but this is really depending on the amplitude of the snare or clap sample. Also, depending on the sample, I may do a slight high shelf boost via eq around the 7-8kHz area to give it a bit more clarity.

I may also add a very gentle reverb to the claps or snare to create a bit of a sense of space.

For claps and / or snare sounds, I use layering more often than for kick drums, but still – quite rarely.

Next, I mix in the hihats. Usually the closed hihats first. Again, setting the level a few dB’s below the kick drum depending on the amplitude of the sample. I often pan the closed hihats a little bit left or right. Equalizing may be needed as well as hihats may sometimes compete with snare or claps. In this case I usually either slightly boost or cut the high frequencies.

Gentle reverb may be suitable for closed hihats as well.

For open hihats I pretty much give the same treatment as with closed hihats.

Panning and equalizing helps to separate closed and open hihats from each other as they are often in a same frequency range.

Crash cymbal may be sometimes quite challenging to mix. It may mess with the hihats or even clap / snare. Depending on the amplitude of the crash cymbal sample, I set the levels to several dB’s lower than the other drums just to make sure it’s not overtaking the drums.

Then comes the percussion sounds. I set the levels few dB’s below the kick drum. If I’m using low tom -type of percussion sounds, they may compete with the kick drum so I use a high pass eq to cut around 100-150 Hz and below.

If I’m using a drumloop to spice up the drums, it usually needs some pretty heavy equalizing to avoid messing up with the rest of the drums.

Sometimes, claps, hihats and percussion sounds may need to be tuned as well to make them work together.

As a general rule of thumb, try to find a spot for each sound in the frequency range and stereo image and do not use too much sounds to avoid the drums sounding messy. Keep it simple.

Allright, now that the drums are pretty much ready, comes the hardest part: mixing the bass with the drums.

Bass

In electronic music, the bass is at least as important as the kick drum. Bass is a sort of rhythmical instrument as well so it’s a bit challenging to mix it with the kick drum so that they work nicely together and won’t fight.

I’ve found there’s two things that affects most to how hard or easy it is to mix the bass with the kick drum: the sound and the sequence. In addition to the bass sound, it makes a difference what kind is your bass sequence: is it in arpeggio style or other staccato type of composition or looooong notes or do the notes hit a same time with the kick drum, what notes are used etc. Sometimes I end up changing the bass sequence just to make it work better with the kick drum and I find I need to do less equalizing.

I usually layer two bass sounds together. First one has a lot of low end and it’s a soft sub-bass type sine wave sound, while the second sound is much more sharper saw or square wave typish. I use both bass sounds to play the same sequence, but I transpose the other (that sharper one) an octave higher. This will make the bassline to stand out in the mix better even if it’s listened through a smaller speaker systems.

However, eq is usually needed here: I cut the low frequencies out of that second, sharper bass sound to avoid it messing the low end. After cutting off frequencies such as low end I need to increase the sound volume level a bit, usually around +1-3dB’s.

Panning I set to center. Just like the kick drum, I always leave the bass to the center.

I use it quite a lot sidechain compression when mixing kick drum and bass. It’s a saviour. With sidechain compression you can make almost anykind of bass sound and kick work nicely together. It’s also a trademark in a modern electronic music – ducking or pumping sound.

So now that the drums are ok with bassline, I start to mix the lead synth.

Lead Synth

If I create uplifting trance music, I let the lead synth to really stand out in the mix. Depending on the sound and the lead synth sequence, I set the volume level only little below the kick drum.

Next step is to eq the lead synth. If there’s a lot of low end in it, it may compete with the bassline. The sequence makes a difference as well: if the lead melody goes really low and it is a type of sound that has a fat low end, then the low cut filter is most likely needed.

I often use a little high shelf boost to add some brightness to the lead. Sometimes I use peaking eq to slightly boost frequencies around 2kHz with a quite narrow bandwidth to make the lead synth to be more audible through the mix. But this depends on the other sounds in the mix.

Layering different lead sounds is what I do a lot. I like to use three different leads and pan them like this: lead 1 to 20-40% left, lead 2 to center and lead 3 to 20-40% right. This creates a nice, wide stereo image.

Delay effect with pingpong effect panning from left to right and reverb is what I use to make them sound really huge.

Just like with the bass, I may sidechain compress the lead synth with kick drum. Works well though I usually use a little more gentle settings than with the bass – a little less ducking that is.

Allright. Now there’s drums, bassline and lead synth in place. Usually, pads are next.

Pads

Pads I like to keep in the background so I set the levels several dB’s lower than the other instruments. Equalizing is needed as well to cut off those low frequencies and maybe some highs too to not let it mess with the lead synth. But as always this depends on the pad sound and other sounds used in the mix.

Sound Effects

Noise sweeps, explosions, synth thwirls and things like that are very important in electronic music. These kind of sounds may be on whatever frequency range so I just say that I try to make them sit nicely in the mix, not too loud, not too quiet.

Compression

Compression might be a good deal to use on individual instruments for more tighter, fuller and present sound. Suits very well for bass. Personally, I use compression on single instrument tracks quite rarely though, but it’s a handy tool for evening out the most extensive changes in the dynamic range: there might be some areas that are a tad too loud or quiet. Compressor can balance these “gaps”. This way, the individual instrument tracks sit better in the mix – the quieter parts can be heard more clearly through the mix and the louder parts won’t dominate as much.

Single band compressor like Fruity Compressor should work pretty well. Maybe start with something gentle like threshold: -23, ratio: 2.0:1, attack: 15ms, release: 200ms, Gain: 5dB. Remember though, with compressor it’s easy to kill all the dynamics so avoid overusing it as things will start to sound unnatural and stressed. Unless you wan’t that as in electronic music that kind of sound may be sometimes desirable.

Listen Through Several Speaker Systems

It’s always a good thing to test out your mix through a several different speaker systems: from very small ones to HUGE and on all possible volume levels. By doing that you can hear what needs to be tweaked. Your main goal should be to make your mix sound as balanced as possible on all sound systems.

More?

Few more words about the delay effect: subtle delay works great with closed hihats and basslines. It’s also easy way to add etheric quality and dense to pads.

Also, very gentle flanger is a great way to add a little life to otherwise monotonous hihat or clap patterns.

Okay, it’s actually kinda hard to give even general guidelines on what settings to use in mixing as so much depends on the sounds, melodies and what kind of elements you wan’t to stand out in the mix. There’s just so much you can do to make your mixes sound great.

General rule I’ve been personally following is I try to make the drums and bass sound as good and powerful as possible and try to keep the mix simple by not adding too much stuff.

I hope this helped.

How To Master A Song In FL Studio

2012-01-30T07:52:00.000-08:00

Mastering is the final step in music production. It’s a process of improving the sound quality (if possible) of your song using a set of such tools as equalizer, compressor and limiter. It’s also preparing a song for the proper medium. After mastering, your song should be ready to go public.

Keep in mind that mastering goes hand-in-hand with mixing. The truth is, if your song is mixed poorly, most likely amateur level mastering isn’t going to do any good for it – in fact, it propably makes it even worse.

So before starting to experiment with mastering, you need to learn to mix. Your song needs to sound as good and balanced as possible before going to the mastering stage.

The thing is, professional mastering makes great sounding mix to sound outstanding.

Lot of producers and professionals say that mastering should be left to mastering engineers and that you shouldn’t master your songs by yourself if you don’t know what you’re doing. I agree, because mastering engineers have the skills, professional equipment, right kind of listening environment (which is ESSENTIAL), trained ears and usually a years of experience so they pretty much know what can be done (if any) to enhance the quality of your mix (check the bottom of this article for links to good quality and affordable online-mastering/mastering services).

You should also know that I’m NOT a professional – I’m just an average hobbyist so what I’m writing here may NOT be the best practice. If you are serious about mastering your song, I STRONGLY advice you to contact a professional mastering engineer.

However, the aforementioned shouldn’t stop you to TRY to experiment with different plugins and settings to see if there’s anything you can do by yourself to make your mixes sound better.

Mastering Tools

There are a wide range of VST plugins that can be used for mastering: brickwall limiters, single band compressors, multiband compressors, equalizers, stereo enhancers etc. and there’s even a bunch of presets to get you started. I have to say though that there really isn’t a “one button solution” or “setting that fits for all” -quick fix for mastering. Each mix is different and that’s why the mastering tools needs to be chosen and tweaked according the situation – everything depends on the audio material you have and what you wan’t to achieve with the mastering. That defines the tools and settings you need to use. Sometimes, you may just need a little help of a limiter in the master channel to do the job for you and if you can make your mix sound good with it, it’s ok. The thing is, you just need to learn to use your ears.

With all that said, I am now going to introduce to you a one possible method and set of plugins in FL Studio that can be used to master a song. The settings and plugins I’m using here is tweaked to work with my example song, so the exact same settings (or plugins) won’t propably work straightly with your mix so you just need to adjust them.

At the end of this tutorial I’m also sharing the FL Studio Mixer State file where I’ve set the plugin FX chain and parameters so you can start to play with it right away and even though the settings are adjusted to work with my example song, you can start with it and tweak the parameters according to your own needs.

I should also emphasize that this is NOT a definitive guide for mastering as there is a lot more to it than just throwing a bunch of plugins to the master channel – rather this is like a starting point and introduction to a method and set of tools that CAN be used for this final processing – just to get you going and it might come useful to you as you start to experiment and learn what the different plugins and settings do to your sound to get the results you are after for.

Okay, let’s begin!

Preparing The Song For Mastering And Getting Down To Business

There’s actually two ways to master your songs in FL Studio: 1.) You can master the song while you’re mixing it or 2.) you can export the song to a WAV file, create a new empty project and import it back to FL Studio and then master it.

I personally like to keep mixing and mastering as a separate projects so I’m using the method 2. and this tutorial will focus on it.

So here’s the step-by-step process:

1. Export your mix to WAV.

Before you export your mix, it is recommended to drop the master mixer track volume level to around -6.0dB – -8.0dB. This is to avoid the appearance of clipping from the exported WAW.
HOWEVER, if you decide to use 32 bit floating point format, you don’t have to worry about the clipping.

Also, it’s best to not to use any compressor or limiter in the master mixer track when you are exporting.

Use the highest possibly quality for rendering. Here’s the settings I’m usually using:

2. Create a new empty FL Studio project, and open your song in Edison. Cut the unnecessary silence at the end or beginning of the song (if any). Check the pic below:

3. Create a very short fade out at the end of the song (if needed). That’s just to avoid the song to be cut abruptly as there might be a tail of reverb or delay effect still playing after the actual song has ended. Check below:

4. Normalize the WAV file.
Normalizing increases the amplitude of the waveform so that the loudest peak is at 100% maximum possible level. This maximizes the volume without clipping. Save the changes to a WAV file.

5. Create another empty FL Studio project and import the WAV to a Playlist.

6. Load the mastering plugins.

Open the Mixer and add these plugins in a following order to the Master mixer tracks FX slots:

Fruity Parametric EQ 2 for cutting unwanted frequencies.
Fruity Compressor to make sure that the dynamic range of the whole mix is balanced so that there isn’t too big gaps between the loudest and quietest parts.
Fruity Multiband Compressor for fine tuning the volume level in areas that are still too loud or quiet in the mix (if any).
Fruity Parametric EQ 2 for boosting frequencies.
Fruity Limiter for maximizing the volume of the whole mix.

Basically, this set of plugins should be enough to master your song satisfyingly.

Now, few words about the order of the plugins in the FX chain: it might be a good idea to put the eq that you use for cutting, before the compression and eq you use for boosting, after the compression. Why? Because the eq settings affects a bit how the compression behaves. For example, if you boost some low fequencies before the compression, you raise their peak level and compressor will most likely rea-act to these peaks by trying to attenuate them.

However, like always in audio production, there shouldn’t be too much do’s and dont’s: in some situations the aforementioned can be exactly the effect that the you are after for, but in a normal mastering situation, cutting frequencies might be best to be done before and boosting after the compression..

Below is a pic of the mastering fx chain:

7. Fruity Parametric EQ 2 for CUTTING.

This is the eq unit that is BEFORE the compressors. Use it for cutting the low frequencies around 30Hz using high pass filter with steep filter slope and also cut around 20kHz with low pass filter. Those are frequencies that can’t be heard through most of the speaker systems, but they just add unnecessary energy to your mix and makes the mix sound louder (not in a good way) than it really is. It’s like leftover noise that needs to be cleaned. Check below:

Note about the low frequency cut: some producers recommend to cut the low frequencies starting from 20Hz instead of 30Hz. This is because if the song is played in the clubs and large PA systems, with 30Hz cut you might loose too much of the low end response.

However, you should know that not nearly all of the large PA systems are able to go down to 20Hz, most will actually roll off the 30-40Hz, so if your mix has a lot of energy in the 20Hz region, it will just uselessly waste headroom. With that in mind, setting the cut to 30Hz will give you a chance to use the larger headroom to tighten up the mix and make it louder.

Also, remember this: if the PA system has been setup poorly, there’s always a risk that sub-bass heavy tracks may blow up the system – even though that’s not your fault, it’s still worth considering.

I have to say that I’ve personally found that cutting the 30Hz range removes the unnecessary rumble clearing up the mix nicely and gives quite a lot more headroom, so that’s why I’m recommending it. But it’s really up to you and what you’re planning to do with your song. You may actually end up to not use the cutting eq at all.

Note about the high frequency cut: if you feel like it you may also like to try start rolling off the high frequencies already from 16kHz to eliminate more high end transients and distortion. Cutting the high frequencies at 16k hz will help to reduce the occurance of ear aching high end on large PA systems.

Now, to the compression.

8. Fruity Compressor.

Fruity Compressor is a single band compressor and it’s useful in mastering fx chain for evening out the dynamic-range variations in your mix. Compressor is great for “geling” different sounds in the mix together – to tighten up the mix.

Let me explain the idea of compression a bit more:

Basically, compression is a process of making sure that there isn’t too much difference between the loudest and quietest parts. It’s like controlling the volume automatically so that the quiet parts are louder and the loud parts won’t cause clipping. It reduces the gain of the peaks and you can set the level where this reduce is happening and how. It is much like automatically tweaking a volume controller so that the volume level is kept within a defined minimum and maximum range. This is to make sure your mix sounds dynamically balanced.

Here’s the most common compressor controllers explained:

Threshold sets the signal level (in dB’s) at where the compressor starts to kick in.
Ratio sets the amount of compression that will be applied.
Attack controls the time how quickly the compression starts to affect.
Release sets how quickly the compression effect will stop after the signal drops below the defined threshold level.
Knee sets at what rate the full amount of compression is applied. Decreasing the knee value lets the full compression to be applied more rapidly – as soon as the attack allows. This is the harder knee. Increasing the value makes the knee softer letting the compression kick in more gently.
Gain (make-up gain) is used to boost the compressed signal because the compression reduces the level of the signal quite a lot.

Couple of tips using single band compressor in mastering situation: use low ratio, long attack and release times for more transparent sounding results (transparent in this context means uncolored sound – keeping the audio as natural as possible). Transparency is needed for keeping the original feel of your mix.

However, every song is different so it’s impossible to give any exact settings, but you can start experimenting with these:

Threshold: -25dB
Ratio: 1.5:1 – 2:1
Attack: 50ms
Release: 150ms
Gain: 2-4 dB

If your mix starts to sound too squashed, or unnatural, try decreasing the ratio and experiment with different rhreshold/ratio combinations.

As a rule of thumb, use the compressor sparingly in a mastering situation to avoid squashing.

9. Fruity Multiband Compressor.

Multiband compressor is a great tool in mastering fx chain for fine tuning the volume levels even further.

Let me explain the multiband compressor a bit: normal single band compressor (like Fruity Compressor) affects to the whole frequency band, but multiband compressor lets you apply the compression to a chosen frequency bands – in this case, three different bands (low, mid and high). For each band you can set the filter cutoff point. That means you can define what frequency areas the band compressors will affect.

Multiband compression is very handy tool in mastering because, if you wan’t to compress (control the volume behavior) of just the low frequency range, you can do that without affecting too much to the the mids and highs and vice versa. For example, if you have loud peaks somehwere in the low frequency area, you can attenuate these without reducing the gain of the mids and/or highs.

Basically you can do this with an eq as well, but using multiband compressor here will give you a bit more transparent and organic results than eq. Of course, this depends on the plugins – some are more transparent than others. Also, what kind of sound you’re after for defines what tool should to used. But with the plugins I’m using in this tutorial, multiband compressor might be a better choise for fixing the gain behavior.

Here’s a quick introduction to different parameters in Fruity Multiband Compressor. Check the pic below (click to see it bigger):

What settings do I recommend to use with Fruity Multiband Compressor? It’s almost impossible to give any recommendations as it all depends what kind of mix you are mastering. You can START with these though, but the rest is up to you and your ears:

Limiter: 0ff
Filter: FIR

LOW BAND:

Threshold: -25.0dB
Ratio: 1.2:1
Knee: 0%
Attack: 3.3ms
Release: 62ms
Gain: -1.6dB

MID BAND:

Threshold: -9.0dB
Ratio: 1.8:1
Knee: 76%
Attack: 31ms
Release: 56ms
Gain: 1.6dB

HIGH BAND:

Threshold: -11.0dB
Ratio: 2.0:1
Knee: 76%
Attack: 3.6ms
Release: 48ms
Gain: 2.7dB

Now, to the boosting eq.

10. Fruity Parametric EQ 2 for BOOSTING.

This is the eq unit that comes AFTER the compression. What I like to do at this point is to slightly boost frequencies around 8-15kHz using peaking filter with a semi-broad bandwidth. This will add a bit of brightness to the whole mix.

The boosting could’ve been done with multiband compressor gain controls as well, but the eq is here for even more precise control over the frequencies. Like I mentioned in the multiband compression section above, it will give a slight different results than eq when dealing with the frequency adjustements.

Okay, check the pic below:

11. Fruity Limiter.

Fruity Limiter is what you use for maximizing the volume of the whole mix without distortion. The purpose of a limiter in a mastering situation is to limit the output level to a defined maximum level (usually o 0.0dB) to avoid clipping when you’re increasing the gain.

I’ll try to explain a bit how the limiter works:

Limiter is a close relative to compressor. It restrains the signal from going above a threshold value, just like compressor. If the peaks are trying to go above the threshold value, a gain reduction will occur.

However, in compressor, the gain reduction is applied in a more gentle way and it also affects to the quieter parts of the signal by making them louder, whereas limiter (at least brick-wall limiter) squashes the peaks that are trying to go above the threshold value and leave the quieter peaks alone. Limiters gain reduction is absolut and more aprubt than compressors. That’s the main difference.

In a limiter, you set the level above which the signal will be limited by using the output Ceiling. Usually it’s set to 0.0dB. Now when you start to increase the volume by using input Gain, the limiter will squash all the signal peaks that are trying to go above the 0.0dB. So the more you are trying to maximize the volume of your mix, the more squashed it will sound. Try to be gentle here by not destroying your mix :)

Some limiters like Fruity Limiter has the attack and release controls as well. Attack controls how quickly the limiter responds to the signal peaks. The Release controls how quickly the signal will “recover” from the limiting.

Here’s some of the most important Fruity Limiter settings introduced:

What settings do I recommend to use with Fruity Multiband Compressor? It’s almost impossible to give any recommendations as it all depends what kind of mix you are mastering. You can START with these though, but the rest is up to you and your ears:

Limiter: 0ff
Filter: FIR

LOW BAND:

Threshold: -25.0dB
Ratio: 1.2:1
Knee: 0%
Attack: 3.3ms
Release: 62ms
Gain: -1.6dB

MID BAND:

Threshold: -9.0dB
Ratio: 1.8:1
Knee: 76%
Attack: 31ms
Release: 56ms
Gain: 1.6dB

HIGH BAND:

Threshold: -11.0dB
Ratio: 2.0:1
Knee: 76%
Attack: 3.6ms
Release: 48ms
Gain: 2.7dB

Now, to the boosting eq.
Fruity Limiter is what you use for maximizing the volume of the whole mix without distortion. The purpose of a limiter in a mastering situation is to limit the output level to a defined maximum level (usually o 0.0dB) to avoid clipping when you’re increasing the gain.

I’ll try to explain a bit how the limiter works:

Limiter is a close relative to compressor. It restrains the signal from going above a threshold value, just like compressor. If the peaks are trying to go above the threshold value, a gain reduction will occur.

However, in compressor, the gain reduction is applied in a more gentle way and it also affects to the quieter parts of the signal by making them louder, whereas limiter (at least brick-wall limiter) squashes the peaks that are trying to go above the threshold value and leave the quieter peaks alone. Limiters gain reduction is absolut and more aprubt than compressors. That’s the main difference.

In a limiter, you set the level above which the signal will be limited by using the output Ceiling. Usually it’s set to 0.0dB. Now when you start to increase the volume by using input Gain, the limiter will squash all the signal peaks that are trying to go above the 0.0dB. So the more you are trying to maximize the volume of your mix, the more squashed it will sound. Try to be gentle here by not destroying your mix :)

Some limiters like Fruity Limiter has the attack and release controls as well. Attack controls how quickly the limiter responds to the signal peaks. The Release controls how quickly the signal will “recover” from the limiting.
Here’s some of the most important Fruity Limiter settings introduced:

For recommended settings – it all depends on your mix again – but you could start with these:

Ceiling: 0.0dB
Gain: 4.0dB-7.0dB
Saturation: 0.0dB
Attack: 4ms
Release: 250ms
Attack Curve: 3
Release Curve: 3
Ahead: 9ms

Experiment with the input gain, but be gentle with it, to avoid squashing your mix. You know you’re putting too much gain when the song is starting to sound like stressed or like it’s in great pressure. Try to find a balance between maximum possible loudness and not letting your mix to sound overly squashed.

11. Export the song once again to WAV using highest quality settings (expect set the WAV depth to 16 bit).

That’s it. With a set of tools like these it is possible to master a song in FL Studio.

I admit, this whole mastering thing is not the easiest task. Personally, I’m always struggling with it as well. It takes quite a lot patience and bit of a trial and error. But experimenting is the best way to learn.

Sometimes, in the mastering stage, you may realize that you can’t fix something with the mastering tools. Then you might need to go back to the mixing stage and do the necessary adjustements there.

Remember also this: using compression in mastering isn’t necessity. If your song sounds good without compression, then don’t use it. It’s not something that is required for making a mix sound great. It’s just there to fix the dynamic range. Actually, EVERYTHING – be it eq, compressor or limiter – depends on the mix and what kind of sound you’re after for. Ask yourself: what do I want – and start from there.
I hope this tutorial helped you to start mastering a song in FL Studio.