FAQs

Q?

What is the Loudness War ?

A.

The Loudness War is a sonic “arms race” where every artist and label feel they need to crush their music onto CD at the highest possible level, for fear of not being “competitive” – and in the process removing all the contrast, all the light, shade and depth – ruining the sound.

This short video demonstrates the effect of the Loudness War – and how pointless it is – using real-world examples.

In order to achieve these super-high levels, the music has to be squashed up against the digital maximum level “ceiling” – reducing the difference between the peak and average levels in the music. In the process, the contrast between loud and soft moments (often referred to as the dynamic range) is dramatically reduced.

(Strictly speaking, this terminology isn’t quite correct – the “Loudness War Sound” suffers from limited crest factor, low RMS variability and in the worst cases distortion. But “limited dynamic range” is an intuitive way to describe all this for Dynamic Range Day. For a more rigorous technical analysis, click here.)

What is Dynamic Range Day ?

The “Loudness War” is built on the idea that “louder is better”. However this concept is fatally flawed. The goal of Dynamic Range Day is to reveal this flaw and spread an alternative message:

 

Dynamic Music Sounds Better

You don’t need to compete in the “Loudness War”. In fact in the 21st Century your music can gain a competitive advantage – not by being “louder”, but more dynamic. To hear this simple secret in action, click here.

The fatal flaw of the “Loudness War” sound

In a nutshell: it doesn’t sound good.

So – “loud” music on CD has no benefit on the radio, online, on an mp3 player, or in your CD player. That’s why I call it a legend – the “Loudness War” makes no sense, in the 21st Century.

And yet almost every new release is crushed to within an inch of it’s life.

The effect is now so extreme that we have reached a bizarre situation where Justin Bieber’s new CD is louder than Motorhead, AC/DC and The Sex Pistols !

So why do people do it ?

Great question ! The answer is here: The Loudness War’s dirty little secret

 

 

Q?

Two compressors in one Channel?

A.

Here are two compressor tricks to try for your next mixdown:

1. Parallel compression – split the signal so that on one side/channel it passes directly through unprocessed while the other side goes through the compressor. Combine the “dry” and processed signal to taste – the compressed signal adds stability and punch while the uncompressed signal keeps the dynamics intact. Parallel compression is popular on drums and bass, but can be used on any type of signal.

2. Use two or more compressors in series. Set each compressor so that it is gently compressing the peaks. None of the compressors will be working hard; the result can be a signal that is compressed, but that doesn’t sound squashed.

Q?

What is a Line Array loudspeaker system

A.

A line array is a loudspeaker system that is made up of a number of usually identical loudspeaker elements mounted in a line and fed in phase, to create a near-line source of sound. The distance between adjacent drivers is close enough that they constructively interfere with each other to send sound waves farther than traditional horn-loaded loudspeakers, and with a more evenly distributed sound output pattern.

Line arrays can be oriented in any direction, but their primary use in public address is in vertical arrays which provide a very narrow vertical output pattern useful for focusing sound at audiences without wasting output energy on ceilings or empty air above the audience. A vertical line array displays a normally wide horizontal pattern useful for supplying sound to the majority of a concert audience. Horizontal line arrays, by contrast, have a very narrow horizontal output pattern and a tall vertical pattern. A row of subwoofers along the front edge of a concert stage can behave as a horizontal line array unless the signal supplied to them is adjusted (delayed, polarized, equalized) to shape the pattern otherwise. Loudspeakers can be designed to be arrayed horizontally without behaving as a horizontal line source.[1]

Modern line arrays use separate drivers for high-, mid- and low-frequency passbands. For the line source to work, the drivers in each passband need to be in a line. Therefore, each enclosure must be designed to rig together closely to form columns composed of high-, mid- and low-frequency speaker drivers. Increasing the number of drivers in each enclosure increases the frequency range and maximum sound pressure level, while adding additional boxes to the array will also lower the frequency in which the array achieves a directional dispersion pattern.

The large format line array has become the standard for large concert venues and outdoor festivals, where such systems can be flown (rigged, suspended) from a structural beam, ground support tower[2] or off a tall A-frame truss tower.[3] Since the enclosures rig together and hang from a single point, they are more convenient to assemble and cable than other methods of arraying loudspeakers. The lower portion of the line array is generally curved backward to increase dispersion at the bottom of the array and allow sound to reach more audience members. Typically, cabinets used in line arrays are trapezoidal, connected by specialized rigging hardware.[4]

Q?

Understanding the differences between Line Arrays and Point Source speaker configurations and which one to choose from.

A.

There are several distinct differences between line arrays and point source loudspeakers. Sometimes, point source loudspeakers are comprised of simply full-range drivers. Many times, they include multiple drivers including combinations of horns and direct radiating cones that reinforce specific frequency bands. In either case, point source loudspeaker cabinets are generally built so that all the drivers function as one source. An example of point source type loudspeakers are EAW KF series boxes (KF650,750,850). Much like the behavior of a true acoustical point source, these systems usually have a fairly short near-field coverage; as the distance from the source increases, their SPL decreases according to the inverse square law. A line array, on the other hand, is typically very large in one dimension (usually vertical), compared to the wavelength of frequencies it radiates. This gives it superior directivity control for frequencies with wavelengths greater than twice the length of the line. Put another way, the length of a line array should be equal to or greater than one-half the wavelength of the lowest frequency over which directivity control is desired. At frequencies much higher than determined by L = A/2 (A = wavelength), a line array can have a very small coverage angle. Typically line arrays are oriented vertically. This enables a small vertical coverage angle, or opening angle as it is sometimes called, to be used to great advantage in reducing reflections and keeping sound off ceilings and other surfaces. This can be very beneficial in highly reverberant spaces. In the near-field of a line array, the SPL falls off at -3 dB per doubling of distance, instead of -6 dB as dictated by inverse square law. Line array systems tend to have a much greater near-field coverage distance than point source systems. This enables a line array to potentially offer higher SPL at a given distance than a point source system. However, there is a limit to the distance at which the line array can maintain this -3 dB SPL decrease per doubling of distance. It can only do this in the near-field of the line array. Beyond the near-field, the SPL from a line array will decrease at -6 dB per doubling of distance. The near-field can extend very far from a line array, and is dependant on the length of the line array as well as the given frequency. This means that for a fixed-length line array the near-field distance will change as a function of frequency. Thus the frequency response of a line array may change depending on the distance away from it. Another important point to consider is the difference in the size of the origin of radiation for these two types of loudspeakers. As the name implies, a point source has its origin at a single point. (While this is overly simplistic, it will help to illustrate this difference.) The sound from a line array, on the other hand, does not originate from a single point, but from a line. While a point is infinitesimally small and has no physical height, a line does. This difference can play a large role in understanding the radiation and coverage from a line array. As an example, consider a point source with a 5 degree vertical coverage angle. At a distance of 30 feet (9.15 m), this point source will cover a vertical height of approximately 2.6 feet (0.80 m). By comparison, a 3-foot (0.91 m) line array with the same 5 degree vertical angle will cover a vertical height of approximately 5.6 feet (1.71 m). This sets the origin for its radiation and will have a bearing on the overall coverage. (This is an extreme example but it does help to demonstrate this fundamental difference.)

When to use Line Arrays

A line array’s high degree of directivity control makes it particularly well suited for use in highly reverberant spaces, where it is imperative that sound be directed to the audience areas only and not excite other highly reflective surfaces. This will help to maximize speech intelligibility in these difficult spaces. Line arrays also work well in rooms with relatively low ceilings. When the ceiling is low in relation to the depth of the room, it may not be possible to position a point source system located at the front of the room sufficiently high enough to provide consistent coverage from the front to the rear-most point in the room. Additional point source loudspeakers would be required farther back in the room on a separate “delay” feed. Using a line array system at the front of the room can make it possible to achieve a consistent SPL distribution from the front to the back of the room, without the need for delay loudspeakers. The directivity control of a line array is generally only within the plane of the line. That is, if the drivers in a line array are arranged vertically (which is most often the case), the directivity control will be in the vertical plane. In the horizontal plane, directivity will be fairly broad. For this reason, line arrays cannot be rotated on there side and still maintain high vertical directivity.

Q?

Sound System Design

A.

There are two main types of sound system designs that have been prominent in the market, consisting of single point source or multiple point source concepts. Multi point source arose from the requirements for very high output power. The idea satisfied that criteria, but with the increasing number of sound sources came an overall reduction in the quality of the sound. The two big disadvantages of multipoint source systems were the suppression of the high frequency output and the physically time-shifted outputs from the individual speakers. Adding a number of time-shifted outputs from individual speakers together causes poor system impulse response. The first types of multipoint sources were simply a large pile of cabinets, stacked together like building blocks and intended to array on all axis. A major improvement in the next generation of systems was the introduction of multipoint, one-axis systems that provided better frequency response and increased definition than previous multi axes systems.

Unfortunately, whilst a step forward, the frequency response and impulse responses were still not ideal and the coverage was often inconsistent. A typical representation of the one axis multipoint source sound system used commonly today is a line array system. Line array does reduce the effect of multipoint sources interfering with each other like the systems of twenty-five years ago, but it is still a long way from the superior results achievable with single point sources. A single point source sound system offers the highest possible definition and dynamic range available today. High intelligibility is a by-product of this, but is only guaranteed by maintaining this high definition and high dynamics through the use of fast and accurate electronics, with low distortion transducers. A line array’s natural frequency response before processing shows a continual roll off of high frequencies from 2 kHz upwards due to cancellation caused by the proximity of the numerous high frequency drivers. This requires large amounts of equalisation to be added to the top end to correct this phenomena. This huge boost in gain on the highs, lowers the system’s overall headroom, on average a line array requires ten times the power to drive the top end compared to a single point source cabinet. Hence high power is not necessarily a requirement for large-scale coverage but quite often a result of a system’s inefficiencies.

Q?

Power compression-Thermal compression in Speaker Drivers

A.

Speaker voice coils are made of copper or aluminum. As these voice coils increase in temperature during normal operation, their resistance increases. Greater voice coil resistance means less power transfer from the amplifier. As a result, the speaker will not play as loud when it’s “warmed up” as it did when it was “cold”. Some speakers may exhibit 3 to 6 dB of power compression. A mere 3 dB of power compression is equivalent to cutting the available wattage of your power amps in half. Speaker manufacturers who develop systems for use in demanding applications such as concerts or nightclubs spend a great deal of their research and development energy working on ways to keep speaker voice coils cool while in operation.

By Sweetwater

Q?

Should I boost or cut EQ?

A.

The rule of thumb that many experts espouse — and we’ve said it here at inSync — is that cutting with an EQ is better than boosting. The typical reasons given for this is that cutting doesn’t impact the headroom available on the track, and that some EQ designs have more phase shift when boosting than when cutting.

But, as you have observed, in practice, many engineers are quite happy to crank up the gain on an EQ band — and obviously EQs are designed to both boost and cut. Here’s what we’ve observed (both watching ourselves and watching other engineers): Boosting seems to be most common when shaping a tone. Think of it this way; despite the “rule of thumb,” when most of us hear a track that needs more low end, we don’t turn down the highs and the mids then turn up the volume. Instead, we naturally reach to turn up the gain on the bass band of the EQ. It feels natural to do so.

However, when dealing with “corrective” EQ — removing problem resonances, filtering out rumble and bleed, cutting a spiky frequency, most engineers cut the gain on a particular frequency band.

Our observation is, then, that in the “real world,” the way in which most engineers default to working with EQ is to generally boost when shaping a tone that lacks a band of frequencies, but to cut when correcting problems. In practice, this works fine; you just have to make sure you manage levels to prevent clipping when boosting large bands of frequencies by a significant amount.

Q?

Sidechain Tuning with EQ

A.

Want to get more out of your compressor? Combine its sidechain input with an equalizer! (Some compressors have filtering and/or EQ available as built-in processing on their sidechain inputs.)

Running the signal coming into a sidechain input allows you to make the compressor more sensitive to certain frequencies or a range of frequencies. For example, a compressor can be turned into a de-esser by running the sidechain input signal through an EQ and boosting the high freqencies. Likewise, the compressor can be used to tighten up the bass range by boosting the low frequencies of the sidechain input signal.

Q?

Compression Before or After EQ?

A.

It depends on what you want; either order can work fine, depending on the results you are after. There is one caveat: if the EQ is in front of the compressor and you adjust the EQ, the compressor will become more sensitive to any frequencies that are boosted — that’s how a de-esser works, for example.

Our advice is to experiment and see which you like best and which performs best to give you the results you want. You may find you put the EQ first in the chain some times and the compressor first other times.