Learning Materials For The Aspiring Music Producer & Engineer

Creating Dimension In Mixing

With the diverse productions in the pop-rock genres these days, one can almost say that many of the mixes sound very “one” dimensional, rather than sounding multi-layered with a sense of distance and depth. A lot of mixes have the lead vocal up front with little or no reverb and then the band’s perspective is placed behind the vocal, often mixed with using one reverb preset. The perspective of depth of a one-dimensional mix is mainly shaped by equalization and level differences between the various instruments and the singer. Add in mastering equalization and compression and you have a lot of today’s music. Very flat and in your face sounding. Mind you, this can present an effect as being high energy, but after repeated listening, it eventually wears your hearing down. At the opposite end of the various pop/rock genres, you have classical music, which sounds amazingly dimensional, especially if the music is a “concerto” which features a solo instrument like a piano or violin.

The reason for a realistic sense of dimension in classical music is that it is recorded using multiple microphones positioned at various distances from the instruments. You have close microphones very near to the instruments and room microphones placed at the rear of the concert hall to record the reverberation of the hall (enclosed environment). If one analyzes the various microphone pick up positions and there related audio capture, you would discover radical differences in the overall sound of the microphones pickups in the concert hall.

The question is? What are the different sounding characteristics of the various microphone position pickups, placed in different locations in the concert hall? How is dimension created? How can one use like minded strategies to create mixes with a sense of depth in the pop- rock genres?

If one fully understands how dimension and perspective work in an enclosed environment, they will then possess the knowledge to re-create characteristics of dimension in the mixing stage for the various pop/rock genres. One basic conclusion you will discover is; there are various differences in the levels and frequency content of the direct sound, the early reflections and the highly diffused reverb. It is the mixture of all these elements that contribute to a sense of dimension in the overall listening experience.

In this age of digital technology, artificial reverberation is not only more affordable than ever before, but can also be stunningly realistic and exceptionally controllable. With a decent understanding of the physics of natural reverberation and the fundamental operational principles of reverb processors, it is possible to easily create the impression of any acoustic environment one can possibly imagine.

In this lab, we will look at all the properties that constitute reverberation in creating a sense of depth and dimension.


Reverberation is very persuasive in enhancing realistic dimensional sounding characteristics in an enclosed acoustic environment. It is basically made up of the regeneration of numerous reflections emanating from the originating source sound that bounce off all the surfaces in the enclosed environment.

The reason reverb is so appreciated in a mix is because the many individual performances in music mixes can be connected with a cohesive sense of space and dimension using reverb, thus elevating the quality of the listening experience. This is because most of the instruments and vocals were recorded in various recording spaces that differ in size and ambience, which often are very dry sounding and unappealing to the ear.

Fig: 1 Reverberation in an Enclosed Environment

This illustration shows how sound disperses in an enclosed environment.

The various performances sound fairly displaced and detached from other instruments when left unaided and dry sounding and adding early reflections/reverb glues the instruments together. Vocals and instruments are mainly recorded with individual close microphones in dead sounding spaces, as to reduce room ambience as much as possible. They will often need reverb/delays in order to enhance a sense of dimension in the final mix. Space the instruments and vocals to appear closely connected dimensionally and holistically. Instruments recorded with Di’s (direct box) have little sense of dimension and often need reverb/delay.

EMT Reverb 250 Plug-in

EMT Reverb 140 Plug-in

As an ascetic production value, reverb works incredibly well in enhancing a performance by extending the time duration of a melodic idea, which is considered realistic and appealing to the listening ear. Adding reverb to a melody of a lead vocalist or instrumental solo is often very desirable for the listener, for it extends the duration of a melodic idea accompanied by harmonic support.

Recording Ambience

Despite the obvious advantage of keeping the recordings dry sounding, most engineers acknowledge that recording natural ambience with the addition of ambient microphones has a lot to offer in a production in terms of creating a sense of establishing natural realism and dimension for the mix. The tracking engineer at times may record instruments with both close and ambient microphones and use some of the recorded natural ambience in the final mix to enhance a sense of dimension. As a result, the practice of recording additional ambient microphones to separate tracks during recording has become popular, for it gives many options for the engineer to use to create dimension in mixing. I often record the room sound for drums or a rock guitar solo, where I’ll walk around the room and clap my hands and position the microphones where I think the ambience sounds the best. Quite often, I’ll insert a small baffle between the ambient microphones and the instrument, for I only need the reflections and ambience and not the direct sound path of the instrument going into the ambient microphones.

In Classical music, the sense of dimension is almost always generated through additional ambient microphones, flank microphones with a Decca-tree configuration. Seldom is artificial reverb added and when it is, it is used modestly to extend the decay time of the enclosed environment.

The best sounding mixes contain a sense of dimension and distance where you can actually visualize depth and distance of the instruments/vocals in a performance. In order to achieve this, you need to understand how direct sound, early reflections and reverb and their inherent EQ, time durations and levels interact collectively in creating a sense of distance and dimension.

Direct Path Sound

If you were to suspend two individuals ten meters above the ground and three meters apart from each other in an open field, you would be able to set up a situation where they could have a conversation with each other where the only audio heard is via the direct path route. There would be no floor, ceiling, or walls to reflect the audio signal. The two individuals would describe the audio characteristics as being totally dry sounding without the ambience one would normally hear in an enclosed environment. When the distance between the two individuals increases, the amplitude and high frequency response would decrease, due to the “inverse law of sound” and the absorptive atmospheric conditions.

In an enclosed environment the direct path sound is always the loudest portion of the overall audio experience, where the early reflections and reverb are always lesser in amplitude and in high frequency content.

In an enclosed environment like a concert hall, the listener will always look on axis at the sound source (singer); therefore the image of the performer is heard as being dead center no matter what the acoustics of environment are or where the listener is sitting.

Direct Path Sound

Fig 2: Amplitudes of Original Sound, Early Reflections and Reverb

Early Reflections

Sounds radiating from reflecting surfaces in an enclosed environment are known as “early reflections” (coming from the walls, floor, and ceiling). These reflections contribute to enhancing a sense of dimension in an enclosed environment. Highly sophisticated mathematics and physics are used by studio designers, in their efforts to build excellent sounding recording studios, mixing rooms and live concert venues.

The frequency response of the early reflections suggests the acoustic properties of the reflective materials of the enclosed sound environment (the walls). If the walls are hard like concrete, the early reflections will contain a lot of mid-range and high frequency content. If the walls are made of wood, the surfaces will absorb the high and mid-range frequencies and suggest an overall warmer sound. Many musicians prefer older concert halls because of their warm acoustical properties that tend to just reflect the musical tonal content of the instruments and vocals (200hz-2khz). This is the main reason why folk and alternative artists like Gordon Lightfoot and Blue Rodeo perform concerts at Massey Hall (warm reverb) rather than Roy Thompson Hall (bright reverb).

In a mix situation, one must follow the laws of physics in order to recreate early reflections that are realistic and believable to the listener. Simply by using and adjusting a stereo digital delay, one can easily accomplish this task.

Early reflections typically arrive quickly after the arrival of the direct path sound but are always lesser in amplitude and high frequency content. The length of time difference between the arrival of the direct sound and the arrival of the early reflections will influence the amplitude, with amplitude decreasing as the difference in time lengthens due to the reflective surfaces being further away from the listener. (large concert hall)

Effective sounding reflections that enhance a listening experience need to arrive to the listener’s ear within 15msec-80msec of the arrival time of the direct path and must sound lower in amplitude with less high frequency content. The length of time between the direct sound and the early reflections arriving to the listener influences the amplitude of the reflections. Therefore the distance of the reflective surfaces from the listening position effect their amplitude. Reflections arriving to the listener between 15msec (left) and 30msec (right) will be louder than reflections arriving between 60msec (left) and 75msec (right). (Fig:3)

Left and right early reflections arriving less than 15msec between each other will produce a flanging effect and/or image location difficulties with the original sound source. This flanging effect can easily be reproduced if one claps their hands and listens for a flutter echo flange in a square room, which is caused by two or more reflections arriving less than 15msec apart from each other. Once the first and early reflections pass the 80msec mark, they begin to sound detached and distinct from the direct path signal and no longer contribute in influencing a sense of distance and dimension for the overall sound experience. If the sound source is transient sounding in nature, the early reflections are easier to hear and distinguish from each other and might prove to be distracting in a listening experience. So as a rule, we stay away from creating reflections that are less than 15ms and longer than 80ms. Early reflections continue to multiply over time until there are so many of them they will eventually be perceived as reverb.

NB: The transient nature of the original signal will influence the 15msec to 80msec range for replicated transient reflections. A transient snare drum may begin to sound discrete in the 50msec-80msec range, where a smoother sounding instrument, such as a cello or flute, will not generate reflections that will begin to sound discrete from the original sound source until at least approx. 100msec.

Fig:3 Early Reflections


How are reverb times determined? A reverb time or “RT-60” is the time it takes for a sound burst to decay 60db from its original level in an enclosed environment. Since 20db of ambient room noise is usually common, I’d like to state, “it’s the time it takes for a sound burst to decay from 80dB to 20db”. With modern day concert hall design and large studios, time consuming and costly tests are performed to see exactly what frequencies are reverberating over a fixed time duration, whereby the acoustic designer and architect can accurately predict the RT-60 of a soon to be finished acoustic environment (concert hall). An important test is when the acoustic designer analyzes the length of the reverb to its relationship of high frequency content during the decay of the reverb in his RT-60 calculations. Most good concert halls have the high frequencies diminishing more as the sound decays, giving the hall a warm reverb sound. Aesthetically speaking, short reverb times may leave the mix a bit lacking in melodic and harmonic continuity, resulting in occasional undesirable dead spots in the listening experience. Short reverb times are good for re- creating a small room ambience.

At the other end, extreme long reverb may overwhelm the rhythmic and musical details in the mix and produce a very confused rhythm and a muddled harmonic and melodic sound. It is like a quick-tempo piano performance with the sustain peddle pressed down all the time. (Fig:4) At the opposite end, where the tempo is very slow and melodic and no sustain peddle is used, the overall sound will have unnatural dead spots. Therefore the goal in determining a good reverb time will factor in how clear the direct path sound is with the proper length of reverb time and its frequency content and overall level relationships.

Fig:4 Washed out Reverberation (Red = dry signal, green = reverb

Fig:5 Good Clean Reverb (Red = dry signal, green = reverb)

The above illustration displays the original sound in red and the reverb in green, As shown, the reverb never over powers the dry signal, whereby the clarity of the original sound is excellent.

High frequency and low frequency content are not much use when using reverb to create dimension. Mid-range to high frequency content tends to make the reverb too audible and excessively noisy (sibilant) and takes away the sense of presence from an original sound (lead vocal/solo). Low frequency content in reverb reduces definition at the low end of the frequency spectrum where low-end clarity may be essential. In most pop-rock productions the frequency bandwidth of reverb will be between 200hz-4khz. Rhythmic reverb can contain frequency content all the way up to 4khz. Melodic and harmonic reverb works very well with all the frequency content under 3khz. Short reverb times of approximately one second are used mainly to create a sense of distance and dimension for a cohesive small-medium room sound. Longer reverb times may be used to enhance a melodic performance in a larger room.

Longer reverb times also work better in sparser productions where the quality of the reverb is highlighted and is a significant production component of the mix. It is like using the sustain peddle on a piano when the right had is playing a melodic line. When listening to a good ballad song from a pop artist, the sound will usually contain a long warm reverb time. (Think of a love song by Adele, Bruno Mars…..) With pop-rock music genres I always remove the low frequencies with a high-pass filter set somewhere in the 100Hz-200Hz range, simply because it allows me to keep the needed punch of the drums and bass guitar uncompromised.

Imagine what the low end of a mix would sound like if the piano player performed in the lower register with their foot on the sustain peddle all the time; very muddy and incoherent.

I also cut the high frequencies as well with a low-pass filter. This is because it helps make the reverb less audible as an added effect and keeps the reverb form sounding too noisy especially in its response to vocal consonants and percussive transients. If there are a lot of the “S” sound coming form the vocal, any reverb will elongate the “S” sound, which is basically adding a lot of unnecessary noise to the mix. Long reverb times on lead vocals work very well when they are warm sounding, leaving the presence range for the dry vocal only. “De-Essing” the reverb send on vocals is essential for my production work.

Fig:4 Stock “de-esser” to remove sibilance

Instruments that tend to create a steady blanket of sound will rarely require any reverb. Adding reverb will seldom be noticeable, for the performance needs to stop on a regular basis, so one can hear the actual reverb. Reverb on these performances, often generate disagreeable harmonic dissonance with chord changes.

So if you have a synth pad or B3 organ playing non-stop in your mix and you want to create some for of dimension, just try removing some presence from the sound or simply turn the level down.

I tend to put short reverbs (1.0sec-1.5sec) on instruments where the main role is rhythmic such as a strumming and/or “chicken-pickin” guitar, drums, percussion. In this situation, the reverb will most likely be short enough to maintain clarity of the rhythmic performance.

Fig 6: Relationship of High Frequency Content to Reverb Over Time

High Frequency Content in Reverberation

In a good listening environment, as the reverb decays, its related high frequency content will decay exponentially over the length of the reverb time (see Fig 6). This is a fact for all enclosed environments. Every time sound bounces off a surface it loses high frequency content. If the surfaces are wood rather than concrete, then the high frequency content will decay faster. Over time, the early reflections and reverb get duller sounding. If you sampled and analyzed the reverb at the 1 second point and also at the 2 second point, the reverb will be much duller sounding at the 2 second point. The reverb will also always sound duller than the early reflections.

Remember that reverb can also be treated in enhancing a musical idea. It can elongate the duration of beautiful melodies and it can create more resonance to drums and add perspective to various harmonic/melodic instruments in the mix.

In the process of creating the illusion of sound reflecting off walls in your mix, you will need to create early reflections with a stereo digital delay. The controllable high frequency content in the delay settings will suggest the idea of what materials the reflective surfaces (walls) are made of. This can be set up in your mix plug-ins.

Fig 7: Three different seating-listening positions in a Concert Hall

Creating Dimension Through Digital Delays and Reverb Plug-ins

In Fig: 7, you will notice the setting of a concert hall with three different seating positions situated at fixed distances from the concert stage. For the purpose here, I will have position “A” situated in the fifth row, dead center from the stage (Approx. 4 meters from the sound source). Position “B” will be situated in the first row of balcony (Approx. 12 meters from the sound source). Position “C” will be in the center of the tenth row of the upper balcony (Approx. 24 meters from the sound source). These three different listening positions sound very different to the listener. In creating dimension in pop-rock music it works best to exaggerate the difference in the type of sound of the different listening positions. What are exaggerated are the time variants, frequency content and levels of the early reflections and reverb in relationship to the direct path signal.

If we vision a lead singer with an additional guitar, piano, bass and drums on a stage we will notice that the singer is usually much closer to all listening positions than the drums, which are usually the furthest away from the listener. In this presentation we allow ourselves to enhance the sense of dimension of the three listening positions (A & B & C) between the singer (closest), then the guitar and piano (behind the singer) and furthest back, will be the drums.

Because low frequencies do not create a sense of dimension in a mix, we can rely on reducing the high frequency content on the bass guitar to create the illusion of it appearing further away from the listener if needed.

In relation to Fig: 7, what we do in the mix is have the lead vocal sounding like the listener is in the “A” position, the guitar and piano sounding like the listener is in the “B” position and the drums sounding like the listener is in the “C” position. It is like the listener is moving his sitting position from position A to position B and to position C, to hear an exaggerated perspective from the singer and the band. The challenge in a mix situation is we can’t move the listener from a fixed position to another. So what we are attempting to do in this exercise, is have the listener (mixer) in one stationary position, but maintain the same distance between all three listening positions and their different distances from the sound source. What we are effectively doing is keeping the singer in one stationary position and moving the singer and the musicians further away. Through the use of digital delays and reverb, we can create the A, B and C positions in the mix. First, we must analyze the levels and characteristics of 1) the direct path sound, 2) the early reflections and 3) the reverb.

The “A” Position

The direct path sound will be the loudest and contain the widest frequency spectrum content in the “A” position. The early reflections will not be audible because the listener is sitting too close to the sound source, where the side walls are too far away from the listening position to have any early reflection influence in the overall sound experience. Therefore no early reflections will be needed to be created for the “A” listening position. 100% of the total listening experience will be direct path sound (approx. 85%-90%) and the remainder, reverb (approx.10%-15%).

The reverb will be significantly delayed and highly diffused (smooth sounding) when it arrives back to the “A” listening position. This is because the audio from the singer needs to bounce repeatedly off the walls to create reverb before it eventually arrives back to the “A” listening position and this takes a bit of time. The actual time of this delay will suggest the distance from the original sound source to the walls and then back to the ear of the listener (70ms–100ms pre-delay setting on reverb). The reverb will mostly contain frequency content below 3khz and will be warm and slightly dull sounding for in a real life situation, the high frequency content of the reverb is eliminated by the absorption of the walls over time. Every time sound bounces off a wall it loose more high frequency content. Rolling off frequencies below 200hz will keep also the reverb from sounding muddy.

When dealing with reverb in the “A” position, you want the listener to feel that the sound source (singer) is very close to them while they are sitting in a large and warm sounding concert hall. Another goal is to enhance a melodic idea by extending the vocal performance in time duration with the addition of a warm sounding reverb with a long pre-delay (between 70ms & 200ms).

In an enclosed environment such as a concert hall, a singer will perform at the front of the stage, projecting sound into the hall. If the listener is sitting in the “A” position, they will hear the sound as follows; the direct path sound from the singer and then, the highly defused reverberation. By the time the reverb arrives to the listener, it is highly diffused and exceptionally smooth sounding.

The pre-delay time of the onset of the reverb will indicate how far the walls are from the listener. The longer the pre-delay will suggest that the walls in the listening environment are further away from the listener and original sound source.

The length of time of the reverb (RT-60) will indicate how reflective the environment is. If the environment has plenty of bright reflective surfaces, the reverb time can be very long (up to 3.0sec). To conclude, the amount of reverb pre-delay defines the distance between the sound source via the walls and the listener. The length of the reverb decay time defines how reflective the enclosed environment is (concert hall).

N.B. The high frequency content is determined by the types of surfaces of the walls and ceiling (absorption coefficient). It is important to note that the reverb decay time will be almost identical for any listening position situated in the hall.

To create the “A” listening position in mixing, you will need to use a reverb that rolls off high frequency content on the decay of the reverb. This means, as a reverb gets longer it keeps getting duller. For example at the reverb one-second point, the high frequency content will contain frequencies up to 3khz. At the 2-second point, the high frequency content will contain frequencies only up to 2.5khz. (Fig: 3) A pre-delay of 70ms- 100ms is needed to create the effect that the singer/soloist is situated close to the listener and in a pleasing reflective sounding environment. If you notice that there is an “early reflection” control on the reverb plugin you are using, turn it off when using the reverb for creating the “A” position.

An additional idea that would enhance the “A” listening position especially if the source sound is an intimate lead vocal singing verses, is to hype the top end in the 10khz–12khz region by a couple of db. This will create the idea that the lead vocalist is singing directly on axis to the listener’s ear, instead of standing directly in front of the listener. I will often boost between the 10khz-12khz range, with a bell curve anywhere from +2db to +3db.

Watch out for sibilance in the reverb. Sibilance is just noise and will effect the clarity in the mix. The way to get rid of this is to heavily “de-ess” the reverb send from approx. 4khz and up (Fig: 4) Also roll off the low frequency area (all frequency content below 200hz), leaving only high-quality musical tonality within the reverb. As stated in an earlier lecture, most musical tonal content resides in the 200hz to 2khz range.

Lead Vocals In A Ballad (“A” Position)

If using a long reverb time between 2.0sec to 3.0sec, it might be a beneficial idea to continually enhance the reverb, especially if it’s a lead vocal performing a ballad with sparse production, where the instrumentation is minimal. Long reverb times manufactured by digital plug-ins unfortunately tend to thin out over time and lose a sense of appropriate diffusion (reverb tail). I often add in a regenerating delay based on the tempo of the song to re-stimulate the vocal reverb. I do not use the dry signal of the delay in the channel return and only use the delay as a reverb send to re-excite the reverb. If the song has a rhythm of 100bpm, a quarter note will equal 600msec. It is important that when you add in a delay to your reverb, that it be a fundamental of the rhythm tempo, for the landing of the beginning of the delay will usually land on an instrument playing on the same beat and any transient of the delay will be masked. This will allow you to increase the level of the delay to your reverb without noticing it as a discrete delay. Obviously, if the delay is 500ms or 700ms, you would hear the delay sounding detached and rhythmically out of time, for it is landing in awkward places in the rhythm of the song.

Remember when adding a delay to reverb, it should be used to re-excite the music tonality to the reverb being used. Make sure to “de-ess” the reverb send and also the delay send. Also insert an EQ filter on the “High Frequency Roll Off” on the delay plugin, so when it regenerates, it sounds less bright on each additional delay therefore more believable to the listener, for this is what truly transpires in a reflective acoustic environment. To do this, you will need to use the feedback setting on the actual digital delay plug-in. (My average setting for this function is 25%- 35% of feedback control within the plugin33).

In a mix situation you will have one channel (e.g. Lead Vocal) and another channel for the reverb return. And if you are using a quarter note delay to re-stimulate the reverb, you will need another channel for the delay return to send to the reverb.

The “B” Position

In listening position “B”, the direct path sound is approximately 75% of the total listening experience. The frequency range above 10khz remains flat with no boost or cut. The original sound source will not have a slight high-end boost above 10khz, like it may be used in the “A” position. Due to the increased distance between the sound source and the listener in comparison to position “A”, we leave the high frequency range unaltered (no high end boost). The listening position is still close enough to not have any impact on high frequencies.

After the direct path sound reaches the listening position, the next thing we hear are the early reflections coming from the left and right walls. The early reflections (approximately 15%) will arrive between 40ms-60ms to the listening position. In an enclosed environment, the early reflections will be arriving mainly from the let and right walls (because most ceilings are too high and the floor is usually covered in carpet, we don’t consider these reflections as contributing to the overall listening experience).

N.B. Early Reflections work best in the 15msec-80msec ranges. If they are less than 15msec, it will create phasing and image problems with the direct path audio. If they are greater than 80msec, they will start to sound like discreet delays and will not contribute in creating dimension.

The difference in time between the arrival of the early reflections and the original direct sound path indicates the distance between the listening position and the reflective walls, and the walls to the original sound source. The sound leaves the stage (sound source) and hits the left and right walls and then continues on to the “B” listening position. The sound then continues on and keeps bouncing off the walls, creating more early reflections where eventually the reflections become so dense and diffused it becomes and is now known as reverb. The further the walls are, the longer the early reflection times are. The early reflections stimulate the psycho-aural response by informing the listener that they are in an acoustic environment with at least two reflective walls (left and right). The reverb will arrive to the “B” listening position sooner than in the “A” listening position. (See Fig: 7) This is because in the “B” position, the distance between the sound source and the listener is greater than in the “A” position.

Therefore in the “B” listening position, the time difference between the arrival of the direct sound and the arrival of the reverb is shorter than in the “A” listening position. (Reverb pre-delay between 40msec-60msec)

The reverb frequency response will sound slightly brighter than the “A” reverb. This is due to the fact that since the listener is experiencing the reverb sooner, the amount of high frequency content in the reverb signal is higher. Remember that reverb loses more high frequency content, as the decay time gets longer. I usually set the reverb high frequency roll off at approx. 4khz.

The level difference between the direct sound and the reverb will be smaller in the “B” position than in the “A” position. So the listener will hear more reverb compared to the original path signal in the “B” position than in the “A” position. With creating early reflections using a stereo digital delay, you need to have the high frequency roll off on the delay plug-in set between 4khz- 5khz. Remember that the early reflections will always sound brighter than the reverb. You will also need to use the feedback on the digital delay to simulate that the early reflections are still occurring, which actually happens in a real situation. The amount of feedback (25%-35%) suggests how live sounding the enclosed environment is. The higher the feedback percentage is, the more reflective the environment is.

Use the feedback as a fader send in the “pre-fade” mode from the actual delay return channel. Do not use the “FEEDBACK” control on the actual plugin. Leave the, “GAIN, DEPTH and RATE” controls at 0. Use the LPF to use as your high frequency roll off. (See Fig: 9)

Fig: 9 Digital Delay for Early Reflections

With the B reverb, make sure the pre-delay is set between 40ms-60ms, but never have the pre-delay shorter than the early reflection times. For example; if your early reflections are set to 40ms (left) and 55ms (right) than the pre-delay has to be longer than 55ms. I would most likely set the pre-delay to 60ms. If you wanted the walls of the listening closer to the listener than you would have to make the all times smaller but always have a difference of 15ms between the two delays. And you would also have to shorten the reverb pre-delay by the same amount. As a rule never have the earliest reflection sooner than 35ms and the latest no longer than 70ms. There should always be a 15ms difference between the two delays.

Be careful that a long reverb time does not corrupt the harmonic content of the original sound source. Most instruments that require this effect will be playing in a harmonic content rather than a melodic content, like a guitar or piano. A good rule of thumb is to make sure that the reverb time of an instrument playing harmonically is not too long where the mix becomes harmonically confusing.

My starting of settings for the “B” position are, early reflections set to 40ms-left and 55ms-right. The high frequency roll off is set to 4.5khz. The amount of feedback is set by the fader in the -15 to -20 range (if using pro-tools). The feedback send on the digital delay is in the “pre- fade” setting. The reverb is set to a pre-delay of 60ms. The high frequency roll off is set to 4khz.

The “C” Position

In the “C” listening position of the “Balcony” the direct path sound will arrive to the listening position at a lower level than in positions “A” and “B”. The high frequency response of the direct path sound will be less than listening positions “A” and “B”, because in a real world situation, the atmosphere eats up high end. This is the listening position where the level differences between the direct sound (55%), early reflections (25%) and reverb (20%) are the least between them. In the “C” listening position, the original sound arrives to the listening position later than all the other positions for it has farther to travel to get to the listener and the early reflections/reverb will follow very quickly. Because the direct path signal is the furthest between the 3sound source and the listener, the high frequency content over 10khz will slightly suffer.

In creating dimension, you should not go out of your way to deteriorate the sonic quality of the direct sound. It is more like “Do not put too much effort into making it sound full”, especially in the top end, and you should try to make the presence work instead in the mid-range (4khz-8khz). It should contain realistic low end and clarity in keeping with the character of the instrument. The reverb time will be shorter than the other two listening positions.

This “total sound event length”, from the first sign of audio to that last sound of reverb is the shortest in time duration in the “C” position than in the A & B positions with the “A” position being the longest. The high frequency difference between the direct path sound and the early reflections/reverb will also be the smallest in the “C” position. The reverb will also contribute more to the overall sound and its pre-delay for creating early reflections will be even shorter (15ms-30ms) than the other positions. With the onset of early reflections and reverb occurring almost immediately after the original sound source arrival, the difference between the reverb/early reflections arriving to the listening position and the original sound will be between. 15ms–30ms. Therefore the arrival times between the direct sound and the reverb will be the shortest of all three positions. The C listening position is almost at the back of the hall, where the left and right walls are now converging inwards to the listening position. (see Fig: 10)

Fig: 10 View of the center upper balcony (C position)

With the reverb arriving very shortly after the arrival of the early reflections and direct path, the high frequency content will be the highest of all the listening positions. As previously stated, the high frequencies decay more over as the reverb time decays into nothing. The level difference between the direct sound and the early reflections/reverb will be lesser in the “C” position than in the “A & B” position.

For setting up the “C” position, I assign the instruments that I want to be sounding the furthest back in the mix, and in most cases, they are the drums. Of all the different parts of a drum kit I usually favour the snare, the toms and a little bit if the kick drum. I usually stay away from creating dimension with the cymbals and hi-hat for they are mainly sounds that contain no resonance and mostly noise characteristics.

My post fader sends from the snare and toms channels are set to the same output level with the kick drum approx. -5db lower. I then set up the stereo digital delay to 15ms-left and 30ms right. I roll off the high frequencies at 5.5khz. The pre-fader send from the digital delay channel is set to about -15db to -10db on the send fader (if using pro tools). The reverb is set to approx. 1sec and the high frequency roll off on the reverb is set to 5khz. One thing that is different with the “C” position, than the “B” position is that I send the reverb from the digital delay channel only (early reflections). I do not send to the reverb from the individual drum channels. This allows for the reverb to be stimulated more often because of the repeated delays coming from the digital delay channel (early reflections). Therefore there is no pre-delay (0ms) needed for the reverb plugin, for it is only being fed from the digital delay channel (early reflections). The reverb send is done in the “pre-fade” mode. So to conclude (1) I have the basic drum channels. The snare, toms and a little kick drum are sent (post fade) to the digital delay channel (early reflections). The digital delay channel (2) is fed back into the digital delay channel in a pre-fade mode to create more reflections. This feedback keeps the delays spinning and the higher the send the more duration in reflections you will get. Then I have the Reverb plugin channel. So you have three channels elements, the drums, the digital delay (early reflections) and the reverb.

Since the direct path sound takes longer to arrive, the reverb arrives the earliest of all positions for “C”, and will be the brightest. The mixer needs to set the high frequency roll off point between 4khz-6khz, which is higher than the “A & B” positions. Therefore, the reverb will be louder, have a smaller pre-delay and sound slightly brighter than the reverb in the “A & B” listening position. I usually set my high frequency roll off on the reverb plugin to 5khz.

Overall as we move further back from the sound source the frequency response of the original sound source gets smaller (less top end) with the early reflections and reverb adding more level to the overall sound experience. As you move further away from the sound source the reverb/early reflections increase in level to the overall sound. The distance in time between the original sound source and the early reflections and reverb will decrease. The overall sound source should always be louder than the early reflections/reverb for this is a fundamental rule in creating dimension in your mix.