If the tonality of your mixes does not translate acceptably to other speakers outside of your mixing room, it means your mixing decisions are being influenced by frequency response issues coming from your monitors and/or your mixing room’s acoustics. If your mixes have significantly different levels of reverberation and/or strange panning issues when heard on other speakers outside of your mixing room, it means your mixing decisions are being influenced by your mixing room’s reverberation and first order reflections. If the tonality of some sounds within your mix (particularly the snare) changes significantly when you lean forward or backward from your normal mixing position it means you’ve got comb filtering off your work surface, and you’re wasting your time trying to find the right mixing position because it probably doesn’t exist…

Most contemporary monitor speakers provide acceptable performance within their intended bandwidth, which means the problems described above are not caused by your monitors, and therefore buying new monitors is not the solution (unless you’re living in that acoustic fantasy world where gut-shaking dance-floor subsonics come from shoebox-sized desktop speakers). The smart solution is to seek the advice of an acoustician. Alternatively, you could stop relying on big monitor speakers and the acoustic treatments they require, and switch to mixing on headphones – which is, coincidentally, what this series is all about. So read on…

VARIABLE COMPENSATIONS

Speaker listening introduces a lot of variables that don’t exist with headphone listening. Compensating for those variables with the tiny on-going tweaks and refinements that take place during the course of a mix – in response to cross-referencing with other speakers, changing seating posture, feedback from others inside the room but outside of the sweet spot, returning to the mix after a break, and so on – tends to make our mixes more resilient and thereby improves their translation across numerous playback systems.

A mix made only on speakers will usually need very little tweaking to sound ‘right’ when heard through headphones, even though it might not take advantage of all that headphones have to offer. A mix made only on headphones can take advantage of all that headphones have to offer, but will often need considerable tweaking to sound ‘right’ when heard through speakers.

How can we make ‘market relevant’ mixes that exploit headphone’s strengths without losing the ‘tweaking-for-the-variables’ benefits that speaker mixing introduces? We can start by understanding a) how human hearing works, b) what we hear and feel when listening to speakers, and c) what we don’t hear and feel when listening to headphones…

HOW DOES HUMAN HEARING WORK?

Human beings have two ears, one on either side of the head, to capture two slightly different versions of the same sound. The ear/brain system uses the differences between these two versions of the same sound to determine where that sound is coming from in a process called ‘localisation’.

The illustration below shows a listener receiving sound information from a sound source located to the left of centre. There are three ‘difference’ mechanisms the ear/brain system uses to localise the sound source, and, for this example, they all occur because the right ear is further from the sound source than the left ear.

Firstly, the sound will arrive at the right ear a short time after it arrives at the left ear, creating an Interaural Time Difference (ITD) – which is sometimes referred to as an Interaural Phase Difference (IPD), particularly at lower frequencies where the wavelength is longer than the width of the average human head and therefore the time difference occurs within one cycle.

Secondly, the signal arriving at the right ear has travelled further than the signal arriving at the left ear and will therefore have a lower SPL due to the Inverse Square Law. This creates an Interaural Amplitude Difference (IAD) – which is sometimes referred to as an Interaural Level Difference (ILD).

Thirdly, because the signal at the right ear travels across the listener’s face and enters the right pinna from a different angle than it enters the left pinna, the signal arriving at the right ear will have a different frequency spectrum than the signal arriving at the left ear due to ‘acoustic shadowing’ of the head, hair absorption, skin reflections, diffraction across the face, and the numerous comb filters and cavity resonances introduced by the pinna. All of these result in an Interaural Spectral Difference (ISD).

Collectively, the ITDs, IADs and ISDs are referred to as ‘HRTFs’ (Head Related Transfer Functions), because they represent the changes imposed on the signal as it passes around the listener’s head and into their ears. The ear/brain system uses the differences between the left and right HRTFs to determine where a sound is coming from, i.e. to localise it.

Loudness vs Frequency

An important quirk of human hearing is that its sensitivity to individual frequencies changes with the SPL. At lower SPLs we are less sensitive to low and high frequencies than we are to midrange frequencies, while at higher SPLs our sensitivity to the low and high frequencies increases significantly.

This behaviour is shown in the graph below, which contains a number of ‘Equal Loudness Contours’. Each contour uses a 1kHz tone at a stated SPL as a reference, and shows how much SPL is required for other frequencies to be perceived as being ‘equally as loud’ as the 1kHz reference.

Each contour is labelled with a Phon value, which represents the SPL of the 1kHz reference tone. For example, the 80 Phon contour shows the SPLs required for different frequencies to be perceived as being ‘equally as loud’ as a 1kHz tone that is being reproduced at 80dB SPL. As shown in the graph below, 125Hz will need an SPL of approximately 89dB to be perceived as being ‘equally as loud’ as 1kHz at 80dB SPL. Similarly, 8kHz would need an SPL of approximately 92dB to be perceived as being ‘equally as loud’ as 1kHz at 80dB SPL.

To put it another way, let’s say we had three separate sine wave oscillators: one generating a 1kHz tone, one generating a 125Hz tone and one generating an 8kHz tone. If the 1kHz oscillator’s output was adjusted to provide an SPL of 80dB, the 125Hz oscillator’s output would need to be 9dB higher than the 1kHz oscillator to sound like it is ‘equally as loud’, and the 8kHz oscillator’s output would need be 12dB higher than the 1kHz oscillator to sound like it is ‘equally as loud’. So a 125Hz tone at 89dB SPL, a 1kHz tone at 80dB SPL and an 8kHz tone at 92dB SPL will all have ‘equal loudness’ – but those differences only apply when we’re on the 80 Phons curve (i.e. 1kHz at 80dB SPL). If we change the SPL of the 1kHz tone, the differences required for other frequencies to sound ‘equally as loud’ will also change, as seen by the differing shapes of the Equal Loudness Contours. If they were all the same shape we wouldn’t have to think about how our mix will translate to different playback levels…

The Equal Loudness Contours show us that as the SPL gets higher, the ear becomes more sensitive to low and high frequencies. This means that a mix made at a high monitoring level will sound lacking in low and high frequency energy when heard at a low monitoring level, and a mix made at a low monitoring level will have excessive low and high frequency energy when heard at a higher monitoring level. In other words, the frequency spectrum and tonal balance of our mixes is affected by the monitoring level used when mixing.

The illustration below shows what happens if we mix at a high monitoring level but play back at a low monitoring level. The blue contour represents the balance of frequencies in a mix made at a high monitoring level of 100 Phons, and the red contour represents how much energy is needed for that mix to have the same perceived frequency balance (i.e. sound the same) when heard at a low monitoring level of 40 Phons. Any frequencies on the blue contour that are below the red contour will be quieter than intended in the mix, and any frequencies on the blue contour that are above the red contour will be louder than intended in the mix.

From the graph above we can see that a mix made at a high monitoring level will be seriously lacking in low and high frequency energy if heard at a low monitoring level. As a matter of interest, this is exactly why some hi-fi systems have a ‘loudness’ button: it boosts the low and high frequencies in a way that looks very similar to the differences between the blue and red contours shown above, allowing the music to be heard at a very low level (to avoid waking up the family, for example) while still having a sufficient perceived balance of low and high frequencies.

The same problem occurs the other way around, as shown below. The blue contour represents the balance of frequencies in a mix made at a low monitoring level of 40 Phons, and the red contour represents how much energy is needed for that mix to have the same perceived frequency balance (i.e. sound the same) when replayed at a high monitoring level of 100 Phons. Any frequencies on the blue contour that are above the red contour will be louder than intended in the mix, and any frequencies on the blue contour that are below the red contour will be quieter than intended in the mix. We can see that a mix made at a low monitoring level will have excessive low and high energy if replayed at a high monitoring level.

Neither of the mixes shown above will translate well to different playback levels – even through the monitors they were mixed on – and both will require corrective EQ in mastering or perhaps even a re-mix, meaning more time and more cost. Mixing loud is proud, mixing quiet is polite, but in both cases you’re mixing on credit: it feels good now but you’ll be paying for it later because for everything else there’s mastering card…

It’s also worth noting that over a long day of mixing our hearing mechanisms become tired, or ‘fatigued’, and this causes us to inadvertently turn up the monitoring level so that things continue to sound exciting. Although our hearing mechanisms suffer from fatigue the Equal Loudness contours remain the same, so by increasing the monitoring level we are inadvertently shifting our hearing ‘baseline’ up to a higher Equal Loudness contour – incurring all the problems that come with that. If you were to mix five songs over a 15 hour day in the studio, it would not be surprising to find that during playback the next day (with rested hearing) the first mix made the day before sounds good, but the last mix made the day before is considerably lacking in low and high frequencies. Why? Because the monitoring level was regularly increasing throughout the day that the mixes were made, so that the last mix was made on a very different equal loudness contour than the first mix.

For these reasons, professional audio facilities calibrate all of their monitoring systems to a standard SPL, typically somewhere around 80 Phons (e.g. the monitoring volume control is adjusted so that a 1kHz tone at -20dBFS or 0dB VU on the stereo bus creates an SPL of 80dB at the monitoring position), which helps to maintain spectral consistency from mix to mix and within a range of playback levels from about 60 Phons to 100 Phons.

The illustration above shows how a mix made at a monitoring level of 80 Phons (blue) translates to higher monitoring levels (100 Phons, green) and lower monitoring levels (60 Phons, red). In both cases, the perceived differences above 500Hz are insignificant. Below 500Hz, when the 80 Phons mix is replayed at 100 Phons there will be a gradual increase in perceived low frequency energy (rising to +10dB at 31.5Hz), and when the 80 Phons mix is replayed at 60 Phons there will be a gradual decrease in perceived low frequency energy (falling to -10dB at 31.5Hz). These changes are not ideal, but they’re acceptable considering they occur over a range of 40dB (from 60 Phons to 100 Phons) and retain very high consistency above 500Hz throughout that range.

80 Phons or thereabouts is also a good level for minimising short-term hearing fatigue and long-term hearing damage, and acts as a warning sign: if the calibrated monitoring volume is not feeling loud enough during a long session, it means either a) the engineer’s hearing is becoming fatigued and it’s time to take a break, or b) the metered level of the mix is lower than the chosen calibration level (typically averaging -20dBFS or 0dB VU) and should be adjusted or compensated for accordingly.

Using a calibrated monitoring level streamlines the entire process from recording to mastering, improves ‘mix confidence’ and translation, and removes the dreaded ‘cold light of day’ disappointment, i.e. the mix that sounded amazing at the end of a long day in the studio sounds underwhelming and disappointing when heard the next morning through fresh ears and at a more civilised (i.e. lower) playback level.

We should always be aware of our monitoring levels, regardless of whether we’re using speakers or headphones. More about that in the final instalment of this series…

As a matter of interest, turning the Equal Loudness Contours upside down, as shown below, allows them to be considered as statistically averaged frequency response graphs of the human ear. This makes it easier to see how the frequency sensitivity of human hearing changes with the SPL.

WHAT DO WE HEAR WITH SPEAKERS?

The illustration below shows the correct configuration for stereo reproduction through speakers, where the acoustic centres of the monitor speakers form two points of an equilateral triangle, and the listener is aligned with the third point. The stereo image is therefore capable of extending across 60° in front of the listener (±30° either side of centre).

The left speaker provides the IADs and ITDs that are embedded into the mix for the left ear, and the right speaker provides the IADs and ITDs that are embedded into the mix for the right ear. A listener in the proper monitoring position receives these signals in the correct relationships to re-construct the stereo image(s) contained within the recording.

Phantom Images

Sound sources are easily localised anywhere in the space between the speakers in a process known as phantom imaging. In the example shown below, the sound source is perceived as coming from the left of centre but there is no sound source in that location. The localised sound is, therefore, a phantom image. When you can hear a sound source where you cannot see one in a stereo image (e.g. a vocal directly in the centre of a stereo system), you are hearing a phantom image. The ability to create a phantom image is the very core of creating a stereo soundstage; without it we’d just have sounds coming from hard left and hard right.

[With the right microphone techniques and/or processing, we can make a sound source appear right in front of our noses, a long way back behind the speakers, or even extend out beyond the sides of the speakers. This type of illusion is also easily created with binaural recordings played through headphones, but that only works when heard through headphones.]

Interaural Crosstalk & More…

Because there is no acoustic isolation between the two speakers, some of the sound intended for the left ear will reach the right ear, and vice versa. This creates a form of interaural crosstalk.

Every stereo mix will contain IADs due to the use of the pan pot and/or panning effects, and it will also have IADs if it has any stereo tracks that were recorded by a coincident (e.g. XY) or near-coincident (e.g. ORTF) pair of microphones. Likewise, every stereo mix will contain ITDs due to the use of stereo time-based effects processors (reverb, delay, etc.), and it will also have ITDs if it has any stereo tracks that were recorded by near-coincident (e.g. ORTF) and/or widely spaced microphone pairs (e.g. AB, drum overheads).

Interaural crosstalk allows the left and right channel IADs and ITDs to blend in the air, reducing the audibility of the differences between them and thereby reducing the perceived width of the stereo image. It can also cause perceived comb filtering if the mix itself contains any delays of 25ms (0.025s) duration or less between the channels.

The Speakers & The Room

Another challenge arises because each speaker remains a distinct sound source and therefore creates its own ITDs and IADs, ultimately telling the ear/brain system that there are really only two sound sources, which conflict with the ITDs and IADs embedded into the mix. These side-effects, and the imaging changes caused by interaural crosstalk described above, are inherently compensated for when mixing through speakers because each signal’s level and panning is adjusted as required to sound right.

Listening through speakers also introduces the possibility of first order reflections from nearby surfaces in the listening space that will be superimposed over the playback and ultimately confuse any spatial information from the recording itself – such problems can greatly interfere with panning decisions. Most listening environments will also have some reverberation of their own, which ultimately affects the levels of reverberation we add to the mix (more about that below). Room reflections and reverberation are addressed with acoustic treatment in a studio control room or in a mixing room, but can be a problem with general listening through speakers outside of the studio environment.

If we introduce enough spatial information (reverberation, etc.) into our mixes the sonic presence of the speakers and the room becomes insignificant – assuming the room is acoustically acceptable to begin with. A good mix transcends the speakers and the room, hopefully invoking a ‘willing suspension of disbelief’ – that feeling when a mix somehow transports you to another place, dimension or world where you cannot see the man behind the curtain.

Visceral Impact

The word ‘viscera’ refers to the soft internal organs of the human body: the lungs, heart, digestive organs, reproductive organs and so on. Therefore, ‘visceral impact’ refers to the impact the sound or mix has on the soft internal organs of our bodies; in other words, how we physically ‘feel’ the sound. Low frequency sounds have the longest wavelengths and generally the highest energy of all sounds in a mix, and therefore provide the most visceral impact.

It is often said that low frequencies stimulate the adrenal glands (located above the kidneys), causing them to generate the hormone ‘adrenaline’ which is responsible for making us want to move and dance when listening to music. However, there is little research to substantiate this. If adrenaline due to visceral impact was a factor required for dancing, then silent discos, silent raves and similar events – which are all based on people dancing to music heard through headphones – would not exist.

WHAT DON’T WE HEAR WITH HEADPHONES?

Headphone listening differs to speaker reproduction much more significantly than most people assume. There are no listening room acoustics to alter the frequency response, there is no interaural crosstalk to mess with the stereo imaging and introduce acoustic comb filtering in the space between the speakers, and there is no visceral impact to add an enhanced/exaggerated sense of excitement.

What we hear when mixing with headphones is the stereo mix directly from our DAW, without any external influences other than the frequency response and distortions of the headphones and the amplifier that is driving them. A good pair of headphones can consistently and reliably deliver frequencies that extend considerably above and below the range of human hearing. It therefore seems logical that a pair of low distortion headphones with a perfectly flat frequency response would provide the ultimate audio reference. Right? The answer is ‘yes’ for listening, but ‘no’ for mixing. Why not?

A mix done on speakers contains compensations for those external influences as they existed in the mixing room (frequency response, room acoustics, interaural crosstalk, visceral impact, etc.), and those compensations ultimately make the mix more resilient – giving it better translation through a wider range of playback systems. Headphone mixing does not have those external influences and therefore our headphone mixes do not compensate or allow for them, resulting in less resilient and more ‘headphone specific’ mixes that do not translate as well to reproduction through speakers and sometimes even through other types of headphones.

ADDING RESILIENCE

What can we do to incorporate those valuable ‘speaker mixing’ compensations into our headphone mixing process and thereby make our headphone mixes more resilient? Let’s start by looking at what headphone manufacturers are doing with frequency responses, then we’ll look at trickier ‘hands on’ mixing problems like making sense of panning in headphones, establishing a reverberation reference when there is no mixing room, and anticipating problems that might be introduced by interaural crosstalk that doesn’t occur when monitoring in headphones.

Frequency Response & Voicing

One of the goals of speaker manufacturers, regardless of whether their products are intended for professional or consumer use, is to create speakers with a relatively flat frequency response from 20Hz to 20kHz. Most studio monitors include their frequency response graph in the documentation that comes with them; it’s rarely a perfectly flat line but if the deviations are gradual and remain within about ±2dB throughout the intended bandwidth the monitors are considered to be acceptable and we can learn to work with them. The illustration below shows the theoretical flat response (from 20Hz to 20kHz) that most speaker manufacturers aspire to (dark red), and the ±2dB window of deviation that is generally considered acceptable (light red).

The dominance of speaker listening prior to the ascent of headphone listening means that our impression of what a flat frequency response sounds like has been skewed by the contribution of the listening room acoustics – to the point that, in comparison, a pair of headphones with a flat frequency response will sound excessively bright while also lacking in low frequency energy. In comparison to speakers, the sound from the headphones does not get the high frequency attenuation that occurs as sound passes through the air along with the absorption by soft furnishings in the room (hence the excessive brightness), and it does not get the low frequency enhancement from the listening room’s resonant modes (hence the lack of low frequencies).

Due to these differences, contemporary headphones are not designed to have a flat frequency response. Rather, they’re ‘voiced’ (i.e. their frequency response has been moved away from the theoretical ideal of ‘flat’) so that they sound like speakers with a flat frequency response. Hence we see headphone marketeers using descriptive phrases like ‘neutral tonality’ and ‘voiced to sound natural’, rather than showing frequency response graphs – because such graphs would alarm anybody who expected to see a perfectly straight line.

Here’s the concept: start with a speaker with a flat frequency response, place a measurement microphone in front of it at the distance a typical listener would be, run a frequency sweep through the speaker, and capture it with the microphone. The result is the ‘flat’ frequency response as it is reproduced by the speaker and captured at the listening position. Build that frequency response into the headphones and they should sound like speakers with a flat frequency response.

This seems simple enough, but it raises questions about the kind of room that should be used for such measurements because the room acoustics influence the sound captured by the microphone.

Throughout the 1970s it was standard practice to use an anechoic chamber, thereby creating a ‘free-field’ environment where the only sound to reach the microphone was the direct sound from the speaker with no contribution from the room itself (i.e. no resonances, no reflections and no reverberation) other than the loss of high frequencies over distance through the air. This was known as ‘free-field equalisation’ and, not surprisingly, headphones that use ‘free-field equalisation’ sound rather like listening to a speaker with a flat frequency response placed in an anechoic chamber. It was an improvement over the sound of headphones with the theoretically perfect flat response, but it still did not correlate well with speaker listening because nobody listens to speakers in an anechoic environment. The headphone designers had the right idea, but there was more work to be done…

In an attempt to create something that correlated better with speakers, the 1980s saw the introduction of ‘diffuse-field equalisation’ – a method that is still popular. A ‘point-source’ loudspeaker (i.e. a speaker that radiates frequencies equally well in all directions), with a flat frequency response, is placed in a reverberation chamber rather than an anechoic chamber. A frequency sweep is reproduced by the speaker and captured by a dummy head placed at a sufficient distance to ensure it is in the diffuse field (i.e. where the room’s reverberation is the dominant sound). This measurement provides the frequency response the headphones are voiced to reproduce. Many critically-acclaimed and widely-adopted headphones conform to the diffuse-field equalisation curve.

More recently, tests by Dr Sean Olive and others working for Harman International (parent company of AKG, Crown, dbx, JBL, Lexicon, Soundcraft, Studer et al) replaced the free-field environment and the diffuse-field environment with what was generally considered to be a good sounding listening room. The results were then combined with the results of tests in which numerous listeners were asked to audition and rate their preferences for numerous headphones with different frequency responses. These tests and measurements resulted in the Harman target curve (aka the ‘Harman Curve’), as shown below:

The Harman Curve is hardly ‘flat’, but it significantly closes the tonality gap between headphone and speaker reproduction, and provides very good tonal translation between them. Neumann seem to have taken this approach one step further by using their KH series monitors as the reference speakers for voicing their NDH headphones, resulting in headphones that correlate remarkably well with their monitors and minimise the ‘line-of-best-fit’ compromising that occurs when an instrument in the mix is too loud on one set of monitors but too soft on another set of monitors. This level of correlation between monitor speakers and headphones is something that very few other manufacturers can offer because most headphone manufacturers don’t make studio monitors, and most studio monitor manufacturers don’t make headphones.

Mixing on headphones that are voiced this way (i.e. to sound like speakers with a flat response in a good room) will usually result in better translation to speaker playback in terms of tonality, and solves one of the major differences between speaker mixes and headphone mixes. However, it does not resolve spatial disparities such as panning and reverberation levels, and it doesn’t counter the effects of interaural crosstalk. Solving and/or compensating for those problems requires a more strategic approach…

Panning Compensation

As discussed earlier, speaker listening creates a stereo image that can be up to 60° wide (±30°, with 0° being directly in front of the listener). A sound panned hard left should appear at 30° to the left of centre (i.e. coming directly from the left studio monitor), and a sound panned hard right should appear at 30° to the right of centre (i.e. coming directly from the right studio monitor).

In comparison, headphone listening creates a stereo image that can be up to 180° wide (±90°), depending marginally on the placement of the drivers within the ear cups. A sound that is panned hard left will be 90° to the left of the centre (coming directly from the left ear cup) and a sound that is panned hard right will be 90° to the right of centre (coming directly from the right ear cup).

The difference between the widths of their stereo soundstages can be represented as a ratio of 180:60, or 3:1, meaning the soundstage width of a headphone mix needs to be approximately 3x wider than it is expected to be when heard through speakers. This is an important consideration when mixing on headphones, because a sound that is panned to 45° to the left of centre when heard in headphones will be heard at 45°/3 = 15° to the left when heard through speakers.

Although headphone monitoring exaggerates panning when compared to speaker monitoring, both monitoring systems downplay panning positions when compared to the visual placement indicated by the pan pot – which rotates through a range of 270° (±135°). The panning ratios between the pan pot, headphones and studio monitors are therefore 270:180:60, or 4.5:3:1. From the point of view of a mixing engineer sitting in the stereo sweet spot, a hard left pan will be seen at 135° to the left on the pan pot, but will be heard at 90° to the left on headphones and heard at 30° to the left through studio monitors. (To add the confusion, it will be shown at 45° to the left on a goniometer, but more about that later…)

For any given headphones, it is always good procedure to start a mix by establishing the location of five reference points across the stereo soundstage – hard left, mid-left, centre, mid-right, hard right – and reminding ourselves of where those locations appear on the pan pot. This is especially important if we’re new to headphone mixing and intuitively pan sounds by ear to the same locations we’re used to hearing them when mixing through speakers. This will result in a very narrow soundstage when heard in speakers because speaker playback reduces the width of a headphone mix by a factor of 3:1, as mentioned earlier.

Remember, the stereo soundstage on headphones is approximately 3x wider than it is on speakers. So, if you want a sound to appear 15° to the left (i.e. mid-left) when heard through speakers, you have to pan it 45° to the left (i.e. 3 x 15°) if mixing on headphones.

One interesting aspect of headphone mixing related to panning is the location of ‘centre’. Depending on the headphones and the listener, ‘centre’ often appears to be inside the listener’s head or directly above it. To overcome this problem, many contemporary headphone designs use angled drivers and/or angled ear pads to place the drivers slightly forward of the ear canal. This allows the pinnae to create subtle ISDs that place the soundstage in front of the listener, at the possible expense of a minor reduction in the width of the stereo soundstage.

Reverberation Compensation

Every well-designed mixing room conforms to a reverberation curve that ensures a level of background reverberation representing an idealised real-world listening environment. Among other things, this creates a ‘reverberation reference’ to balance the levels and times of our reverberation effects against, ensuring they are not significantly lower, higher, longer or shorter than intended when the mix is taken out of the room and played in the real world.

It is commonly believed that if we mix in a room that does not have enough reverberation of its own, we will add too much reverberation to our mixes to compensate. The same thinking implies that if we mix in a room that has too much reverberation of its own we won’t add enough reverberation to our mixes. Although this appears to make sense, it ignores the ear/brain’s remarkable ability to distinguish between the reverberation of the mixing room and the reverberation added to the mix. The mixing room’s reverberation is not necessarily heard as part of the mix’s reverberation, but it does provide a masking effect that the reverberation in our mixes needs to overcome. The result is as follows…

If we mix through speakers in a room that has a particularly low reverberation reference, the levels of the reverberation effects we add to the mix might not be high enough because they are easily heard over the room’s reverberation reference. Likewise, the reverberation times we choose might be too short because the room’s low reverberation reference makes it easier to hear the added reverberation tails for longer.

Similarly, if we mix through speakers in a room that has a particularly high reverberation reference, the levels of the reverberation effects we add to our mix might be too high in order to be heard over the room’s high reverberation reference. Likewise, the reverberation times we choose might be too long because the room’s high reverberation reference makes it harder to hear the added reverberation tails for the desired time.

The illustration below illustrates this problem. The upper graph shows reverberation (green) being added to a mix in three different mixing rooms: one with a very low reverberation reference, one with a good reverberation reference, and one with a very high reverberation reference. Each room’s reverberation reference level is shown in grey, and in each case the mix reverberation (green) has been added at an appropriate level and duration to achieve the same perceived level and duration in each room – represented by the green area above the grey areas.

The lower graph shows what happens when each mix is replayed in a room with a good reverberation reference. The first mix’s reverberation is inaudible, resulting in a very ‘dry’ mix. The second mix is consistent with the upper graph. The third mix’s reverberation is too high, resulting in a very ‘wet’ mix.

Headphone listening is not affected by the mixing room’s acoustics and therefore has no reverberation reference, making it harder to judge and set reverberation levels and times in a way that translates well to speakers. A mix made entirely through speakers in an acoustically-designed mixing room translates to headphones with no surprises in reverberation levels or times because the reverberation effects have been balanced against the room’s reverberation reference. However, a mix made entirely through headphones could have surprising changes of reverberation levels when heard through speakers because it has been made with no reverberation reference. What sounds ‘just right’ when mixed in headphones is often too low when heard through speakers.

We can solve the reverberation reference problem when mixing with headphones by using a reference track as a reality check, which we’ll talk more about in the next instalment.

Interaural Crosstalk Compensation

When mixing with headphones we cannot predict what changes will happen to our mix when the left and right signals combine together in the air and at the ears. As mentioned earlier, this is known as ‘interaural crosstalk’ and is an unavoidable part of speaker monitoring: some of the left channel’s signal will enter the right ear, and some of the right channel’s signal will enter the left ear.

Interaural crosstalk can affect the perceived levels and panning of individual instruments in our stereo mix, it can introduce comb filtering, and it can alter the perceived level of reverberation and similar stereo time-based effects.

The easiest way to check for the effects of interaural crosstalk when mixing with headphones is to check the mix in mono. This creates a ‘worst case’ crosstalk scenario (i.e. both channels are completely added together) that will exaggerate any level changes or comb filtering issues that might occur when the mix is heard through speakers. Subtle changes in individual signal levels within the balance are to be expected, but can also be indicators of hidden weaknesses in the mix that are worth addressing and fine-tuning. For example, sounds in the stereo mix that become too loud or too soft when monitored in mono are probably not at the right level in the stereo mix, and should be adjusted accordingly. A headphone mix that sounds acceptable when monitored in stereo and acceptable when monitored in mono stands a good chance of sounding acceptable when heard through speakers, too.

Another useful tool for revealing potential interaural crosstalk problems in a headphone mix is the goniometer, also known as a correlation meter – a popular prop in old science fiction movies. More about that useful tool in the next instalment…

In those days nobody listened to headphones except audio pros, musicians in studios, Dads enjoying their stereograms without waking up the kids, and old men listening to race calls through mono earbuds jacked into pocket radios. Oh, and that weirdo standing in the train door fingering an air guitar while wearing one of those newfangled ‘Walkmans’ (a fad we all knew would never last), completely oblivious to the silent audience hiding behind walls of newspapers and magazines. Headphones were definitely not fashion accessories, and anyone wearing them in public looked ridiculous.

Disillusioned, I abandoned my plan of imposing artificial HRTFs onto my electronic soundscapes to create immersive binaural illusions. Those illusions only worked in headphones, and nobody listened in headphones…

A dozen lifetimes later and, thanks to the proof-of-concept provided by Sony’s Walkman and refined by Apple’s double-whammy iPod/iTunes combo, everybody is listening in headphones. Oh, except for that pencil-clutching weirdo in the train door immersed in the pages of one of those newfangled ‘journals’ (a diary by any other name is still a diary), device-less and oblivious to the rows of headphoned performers spot-lit by tiny screens while silently fingering air guitars, conducting orchestras with finger batons, striking out at knee drums and air cymbals, or navigating app-worlds that have artificial HRTFs imposed onto their electronic soundscapes to create immersive binaural illusions.

The cynical nostalgia evoked by the ‘nobody talks to anybody any more’ memes and tropes would have you believe that in the days before mobile devices, every train carriage, every bus and every waiting room was filled with strangers striking up genial conversations and filling the air with chatter. Atavistic nonsense! Before mobile devices people isolated themselves with newspapers, magazines and window seats, intentionally filling the air with the same lack of chatter as they do now. So slip on your headphones and forget about the negative-calorie small talk that Luddites cling to like Replicants embracing implanted memories – because it never really happened. Air boom, air tish…

Headphones have become fashionable, and therefore headphones have become status symbols. If I had a dollar for every time my Neumann NDH20s have been selfied by passers-by and diners at the night markets of South East Asia, I’d have enough to buy a Lamy clutch for every Yuccie arguing the difference between a journal and a diary.

Headphones have democratised high fidelity; for a few hundred dollars you can get a pair of high-status headphones that shrug “hold my beer” when pitted against thousands of dollars worth of speakers with their obligatory acoustic treatments and tightly-defined ‘sweet spots’. Headphones allow you to sit, stand, lay or move about wherever you like because you are the sweet spot; the room’s acoustics don’t even know you’re there.

Every major shopping mall has at least one store dedicated to mobile audiophilia and ‘head-fi’, i.e. high fidelity audio through headphones. If the stuff they sell seems expensive then you’d better re-assess your priorities because it’s chicken-feed compared to what it costs to get similar performance from speakers and their obligatory acoustic treatments, and you can take it with you anywhere.

Most significantly, headphones have supplanted speakers for the vast majority of music purchasing decisions and active music consumption (i.e. listening with intent rather than plastering sonic wallpaper over the background noise). Simultaneously, there has been a resurgence of interest in binaural recording and the immersive possibilities it offers without requiring a room full of speakers and the hope that the playback system adheres to the same format adopted during recording and mixing. Microphone manufacturers like Sennheiser and DPA have added binaural microphone systems to their product lines, and Neumann’s dummy head (aka ‘Fritz’) has reached new levels of celebrity. Popular music artists are now inserting binaural elements into their multitrack recordings, taking the headphone listener by surprise with sounds and voices appearing from beyond the musical soundstage.

As audio professionals, we would be foolish to underrate the significance of headphones in our mixing and monitoring decisions. Rather, we should be celebrating their popularity and exploiting their advantages. It is little wonder that many of the younger generation of producers are doing most of their mixing work on headphones, and consider the big monitor speakers as being primarily for show. However, the key word in that phrase is ‘most’. Speakers bear little relevance to their world or their market, but they’re still cross-referencing on speakers, and they’ve got mastering engineers downstream ironing out the kinks while listening through big monitor speakers.

What was once ridiculous is now mainstream. What is now ridiculous is placing too much significance on big monitor speakers and their obligatory acoustic treatments and inflexible sweet spots when mixing, because we’re living in a world where most people’s exposure to speaker reproduction is background music in cafés and shopping malls, platform announcements at train stations, and – topping the list – device notifications.

Ding!

I have hate mail…

THE RØDE ECOSYSTEM

Aside from using Røde Central to control your mic, Røde has a growing library of software designed for its USB microphone products. Røde Connect allows multitrack recording on a computer with up to four Rode USB-capable microphones. Unify is a virtual mixing application ideal for gaming and streaming. For iOS users, the Røde Capture app lets you control and record using Rode microphones. Now that PodMic is USB-capable, it plays nice with all these applications. And with all being available as free downloads, this turns into a significant value add.

Essentially, if there’s even a slight chance you might use the PodMic with a computer, smartphone or tablet, you’re better off spending the extra A$150 to acquire a PodMic USB. You’re given a load of digital connectivity and processing options without losing the XLR-equipped purity of the original. Win/win.

A CASUAL ARRANGEMENT

Given that Katy’s voice is the album’s centrepiece, how did the pair tackle the task of capturing, treating, and mixing it? After landing on an overall desired vocal sound, they began looking at how best to present the essence of these parts. Some of this essence was in the effortless, off-the-cuff nature of the guide-tracks. Katy explained how these early takes would influence the song’s development, and often end up as part of the song. She said, “As we began to track vocal demos we’d usually set out thinking, ‘Oh, it’s only a demo, it’s just rough’ — not taking much care with them. We found that this would result in something that felt more spontaneous, and sounded really great. In fact, we wound up using the original demos on ‘LMK’ because they had a kind of quality that we just couldn’t replicate.”

Vocals were recorded at home in their DIY studio which was a flexible, dual purpose space: the spare bedroom. The setup was minimal but effective, and allowed them to work whenever the mood took them. “We had a really good home setup, and it sounded amazing,” recollected Graham. “Most of it was recorded in the bedroom with a Neumann U87. I worked completely in-the-box, so was mainly using UAD’s collection of plugins — generally using the 1176, LA-2A, or 1073 at the start of the chain, and then sometimes adding a bit of EQ at the other end.” He continued, “I did start experimenting with other plugins too — the Manley VoxBox and Avalon VT-737 both worked really well with Katy’s vocals — but her voice was so consistent that I didn’t need to go too hard on that side of things.”

SIGNATURE VOCAL SOUND

The key to Katy’s signature sound was less about plugins and more to do with layering her vocal parts, relying more on doubling and panning to highlight her unique delivery. When replicated in unison, the precision of Katy’s performances results in a gently chorussing movement. Katy commented that she’s also a fan of adding harmonies to create lush vocal beds. “I always double up my vocals, but I love harmonising too. They come to me quite naturally, so we end up having lots to play around with.”

But Graham, who proves to be the patient one, quickly added, “There were sometimes triple and quadruple vocal parts — and then there were the harmonies. We took the songs to a certain point — kind of throwing everything at the wall, so to speak — and then had to scale it all back again to restrain the pallete. I mean, it’s great to have so many vocals to work with, don’t get me wrong, but it was also quite a process to thin them all out.”

With all of the vocal comping now out of the way, both Katy’s main and backing parts have been left relatively intact, only receiving subtle processing to help with focus and placement in the stereo field. “We did use Arturia’s Dimension D chorus and UberMod from Valhalla, and the Eventide H910 plugin,” recalled Graham. “And down the track, when we had a remix of ‘Come See Me’ done in a studio in Perth, they put Katy’s vocals through a physical H3000 unit.” Katy also noted that “We tried the Antares Harmony Engine as well. I didn’t really like what it came up with but Graham definitely vibed on it.”

Through all of this trial and error, the pair refined their process and ended up with a solid approach that allowed them to confidently capture and process vocal takes. More importantly, being a simple home setup meant that they could leave it partially set up. “Now we can have things waiting, ready for an inspired moment from Katy,” said Graham. “And we know we can get something sounding good, straight off the bat. We just need a good mic and preamp. And a good vocalist!”

TRADE TOOLS

As far as instrumentation went, most of it was performed by Katy and Graham on the collection of instruments they’d picked up overseas. “The Juno was doing arpeggiated bass and leads on ‘Feel So Bad’ and ‘Come And See Me’, but unfortunately we’ve just sold that,” shared Graham. “And ‘End Is Near’ has Katy’s SV-2 parts all over it.”

So, in addition to the the Roland Juno 6 and Korg SV-2 hardware synths — and the Casio keyboard — a couple of software options were employed too. These included: Reverb Machine’s RM-20, the virtual instrument that samples sounds from the Yamaha PS-20; the Arturia CMI-V plugin emulation of the Fairlight digital sampler and synthesiser; and the U-He Diva soft synth that models the components and modules of a handful of vintage analogue synthesisers.

Graham gave some further detail: “I was watching a lot of ‘Mix With The Masters’ at the time, which I was finding really inspirational. There was one in particular with Illangelo, and while I’m not a massive fan of the music, his process really resonated with the way Katy and I were approaching the record. I spotted that he was using the Diva in the video so we gave it a shot and ended up using it quite a bit. ‘Feel So Bad’ had a lot of that on it.” He continued, “‘Beside that one, there was also a preset from the Arturia CMI-V plugin that we used on ‘Falling Apart’.”

Katy chimed in, saying, “The Casiotone, the SV-2, and the CMI-V plugin patch all created such a canvas across the album. We also dug our old Casiotone 401, but then found the RM-20 plugin which is essentially the Beach House sound.”

HIRED TUNES

When working on an album by yourself, having only a single creative source can create a homogenous outcome, so while Katy and Graham were the core songwriters and performers, they also enlisted the talent of a few others. These additions contributed different perspectives to the process, which helped to broaden their sound.

Local Perth multi-instrumentalist, Josiah Padmanabham, made an appearance, and ended up playing quite a significant role across the record. Graham explained: “It was a really interesting process. Josiah was such a gun, and really brought a different colour to things. When he sat down in front of a synth you’d have to hit record straight away because he’d do something magical and you’d never get him to recreate it.” Katy also added, “We added a couple of backing singers too — Tara and Solomon came and helped out. They were recorded at Poons Head Studios in Fremantle, and you’ll no doubt hear them throughout the album.”

MIX & MATCH

Of the three mix engineers auditioned to work on the mixdown, Dave Hammer was the one they selected to carry out the final duties. But besides being hired to give the album a general spruce up, and a tying up of loose-ends, he also ended up making some creative decisions that helped reduce the clutter of mixes and sessions. So, after working intimately on something for so long, how did Katy and Graham feel about passing that kind of control on to an outsider?

Katy remarked: “We were still kind of working things out when we got Dave on board, but he really elevated the album. Graham had produced and mixed pretty much all the songs, but then Dave took it all up a notch in his final pass. He was also very patient with us.” Graham continued, saying, “Dave did the final mixes on the majority of the album, but he helped us to strip things back a bit too. We’d send him our sessions, and some of those were pretty big — sometimes with 80-100 tracks… But he came in with fresh ears and a different approach and made such a difference. I knew this all needed to happen, but I wasn’t the right person to do it because I was too attached. So Dave helped us by making things more manageable — but I’m sure he could have gotten rid of a bunch of other stuff without us noticing too!”

LIVE WIRES

Katy and Graham settled on Ableton as their DAW of choice (via Logic and Luna) which proved a good choice when it came time to prep for live performances. And speaking of which, what does their live contingent look like? Well, Katy has been so inspired by the performances that the Korg SV-2 and its RH3 keybed is joining them on tour. Straight to the pool room. She explained that, “The SV-2 has really changed everything — it’s just so beautiful to play… It’s changed how I write and perform, so it’s replacing my Nord on tour. Those old Nord keys feel so flimsy now, but the Korg’s keys are weighted so it plays just like a real piano.” And compared to the Nord, the Korg probably feels like it weighs as much as one too.

But the Korg has played a creative role in more ways than one. Given this preference for it over the Nord, the comparatively smaller library of sounds in the Korg means they’ve had to be more particular about choices in the studio to ensure the sound can be replicated live on stage. Graham shares, “It sounds great straight out the box, you know? It’s not got as many sounds as the Nord which means we have to be more conscious of what we’re adding to songs to ensure the Korg’s live patches can translate them — but it has such a nice feel to it, so it’s all worth it.”

With a 14-date national tour lined up, tour preparations began. Graham offered some insights on this, and stated that, “We had Josiah playing in the band in the studio, but we’ll be stripping it back for the live shows, with just Katy and I performing. A lot of the synths will be on backing tracks, but I’ll be playing some live synth and guitar too. And because we’ve had Ableton running in the studio we’ll be able to transition to our live performances easily.” Katy will be also juggling some roles on tour, and said, “I’ll be playing acoustic and electric guitar, and keyboard. And obviously some singing!”

The pair are stepping it up a notch on stage in other departments too. Katy goes on to say, “Graham is currently doing a tech rehearsal because we’ve just hooked up our new in-ear monitoring rig. Everything will be running through a Behringer x32 Rack so that we can have more consistency from venue to venue.”

THE VIEW FROM HER

Considering Katy’s history in the music industry, it’s interesting to imagine how she might have found this experience compared to her previous album-making endeavours. This one has been different on a few levels: with a new addition to her repertoire being the straddling of parental commitments while being a musician; and then more widely, that there’s been so much social and musical change in the world in the last 20 years. Katy says it’s been an interesting journey, stating, “I’ve kind of taken it all in my stride — being a mother to a 14 month old and 5 year old means that life’s hectic. Time is just so precious to us, so that’s been really hard. And the industry is just so completely different now, it’s kind of an upside-down world. I mean, Kanye West puts out a record, then decides to change the mix after it had been released — he just took it down and changed the files, you know? Crazy.” Katy said that as part of that, she and Graham often discuss the importance of being independent, and not taking that for granted. “It’s amazing to realise that now we can do what we want, because we just can.”

With ‘abundance of choice’ being an underlying theme of this album’s story, and without deadlines in place, the creative luxury of working from a home studio quickly became a hindrance. Having no time constraints made it harder to finalise ideas — and easier to get bogged down in the extra choices that were needing to be made. In regard to having such fluid goal posts, Katy said: “It kept ballooning and getting more adventurous. The finished product is the result of a lot of work, and a lot of experimentation — a lot of hours, a lot of sleepless nights… And sleepless days!”

As the album project wound down there was some reflection on the journey. With confidence built, lessons learned, and new skills developed, Graham shares his thoughts on the final outcome by saying: “There’s definitely lots of things we’d do differently next time, 100 percent. It was pretty stressful towards the end, for sure.” Katy wrapped up by saying, “I’m really proud of what we’ve achieved. I can’t wait to do the next one, and see where we end up. It feels like we’ve just started, and this is just the beginning.”

MEET THE FAM

VTX is comprised of A Series (its array loudspeakers, including the A12, A8 and A6); the B Series (‘B’ for ‘bass’, including the B28, B18 and B15); M Series (‘M’ for stage ‘monitors’); F Series models for fill; and associated software (including the Line Array Calculator prediction software; Performance Monitor, the tuning, amp control and configuration package; and Array Link, a nifty app for crew to have the rigging details on their smart device). Crown I-Tech amps power the system as part of a VTX package. As you’d expect, the range has a full complement of rigging and accessories.

VTX isn’t brand new, but the baby of the range, the A6, is. The launch was the first time the A6 had been sparked up as a system in Australia, but more than that, it was the first time most had heard VTX at all, certainly not a side-by-side comparison of the complete A Series range.

All three mid/high boxes adhere to the same design principles. The A8 has 6dB less output than the A12, and likewise the A6 is 6dB down on the A8, but they share the same sonic signature. You can pair any of the mid/high boxes with any of the subs. The B15 is designed to be flown in a A6-based array, but an A12/B15 rig is entirely possible. 

JBL’S RACE CAR

VTX represents the best of JBL Pro; packing its latest transducer and loudspeaker design technology. Or, as JBL’s Director, Product Management – Line Array Systems, George Georgallis, describes it: “VTX is our race car”.

I’ve pulled out the various standout technologies of VTX in box items but it’s worth dwelling on the HF drive unit, which does more than anything to differentiate VTX from previous JBL line arrays. 

George Georgallis: “The main thing we wanted to improve over the VerTec was the the high frequency character of the system. The majority of a line array’s performance is derived from the high frequency section, both in terms of output, the sound character, and also how well it works as a line array — how well it sums for the long-throw situations. Which is why we invested a lot of R&D time into the high frequency section of this product.”

JBL has the luxury of being a transducer manufacturer and was able to develop a brand new HF unit from scratch. And the first thing you notice is the waveguide is built into the HF unit — it’s all one assembly.

The next big departure: the diaphragm of the compression driver isn’t metallic it’s a composite, proprietary material, losing any characteristic harshness you can get from a titanium- or aluminium-based design.

George Georgallis: “Compression drivers tend to be influenced by the materials because not only do we hear the direct sound, but also the material sound, especially when we start getting to the break-up modes of the materials, and if those materials are metallic, as they’re moving and breaking up, they introduce that character.”

The diaphragm isn’t dome-shaped either. It’s shaped to match the waveguide geometry. George Georgallis: “Compression driver exits are historically circular to mate with a horn, which provides coverage in the horizontal and vertical. That doesn’t make much sense with a line array’s waveguide, in terms of manipulating what’s coming out of it to create a flat wave.”

SHOWING UP

MadisonAV and Harman really showed out and provided VTX with the best possible conditions for Australian JBL fans and sceptics alike to appreciate what the system is capable of. More important than a flashy launch, MadisonAV has invested heavily in demo stock and has appointed a product specialist (Peter Kubow) to connect with the rental market. VTX is already being specified into significant audio installation projects in the region, and is well suited to those kind of applications (performing arts centres, exhibition centres, houses of worship etc) but the question mark is around its appeal to mid- to larger-sized PA rental companies. A VTX package offers good value but is in no way a thrifty choice, and VTX doesn’t yet have the same brand name cachet as the Euro incumbents. But the VTX performance and versatility can’t be questioned. It’s a comprehensive, solid system with equally solid local support behind it.