Ideas On 5.1 Mixing

Surround Sound Mixing Current mood:

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Category: Music Surround Mixing Ideas

Ideas on 5.1 Mixing Making the transition from stereo to surround sound is to some extent like adapting from mono to stereo. The wider latitudes and options are a welcome format that allows one to be even more creative in their search for elevating audio to higher standards that appeal even more to the average listener. When approaching a 5.1 surround project, I try to vision the sound of the final outcome before even getting started, an approach that worked well mixing in the stereo format. How should I record everything now that I'll have a surround sound palette to fill? What will be the focus in the mix and how can I maximize the quality and environment of the surround sound format in the mixing stage? In this document I am going to explore past experiences dealing with surround mixing, analysis of conventional methodology and personal ideas on creativity. This will be a subjective viewpoint based on how I deal with surround sound and I will likely differ with conventional opinions and standardized methods. One method is to record and mix in the effort of trying to emulate the acoustic environment, versus, abandoning unadventurous rules to use a totally creative approach. A third method would be to use ingredients from both schools. With reality mixing, the methodology is one of achieving true realism of the associated acoustic environment. This the most common approach, but implied too literally may prove to be either boring or too distracting to the listener/viewer. For example; if an important character is off-screen and is heard talking from the back left channel, it might cause the listener to turn their heads around in the theatre and remove the focus of what is happening on the screen. It might be realistic, but prove to be too confusing to the movie story. Another situation is when all the audio is coming from the front channels and only the recorded or simulated ambience is coming from the rear-a practice well endorsed today by many engineers and producers who feel that all audio should originate from the front viewing source with the exception of ambient and sound design elements. In most movies today almost all dialogue and music originates from the front channels with music ambience, environmental ambience and some sound design originating in the rear channels. In the movie series of "The Lord of the Rings" great effort was put into achieving a great balance between realism and creativity in the dialogue, sound design and music without ever sacrificing the focus of the story. In this movie the on screen dialogue is panned entirely across the front channels, the ambience is coming in a surround format, occasional sound design elements are situated according to image location-with panning between channels to simulate motion and music coming from the front channels with the acoustic ambience coming from the rear channels. This approach seems to be the most ambitious so far, yet I feel it can even go further. A creative approach abandons conventional mixing styles in order to embellish the listening environment when viewing the film in a theatre or home. When done constructively, this creative approach can add power and emotion to the film's final mix.

Presently most recording and mixing of music for surround films starts with a process of midi elements such as synthesis and percussive sequencing with the addition of harmonic and melodic ideas played from samples to be replaced later. After the music has been approved it is off to the studio to record orchestral elements. The recording process often utilizes surround microphone techniques with close and/or spot microphones. (Decca tree, X/Y, M/S….). The microphones used are high quality condensers with various pickup patterns to allow for environmental flexibility in the mix process. Once the recording is finished the mixer will hopefully have all the ingredients to explore some ideas that will utilize all the listening environment parameters. In the following pages I will discuss some ideas that I have used in 5.1 mixes for movie scores and a song by Andrea Bocelli "Go Where Love Goes". Room for dialogue Phantom centre Extension through delays

Audio Envelope Before getting started with surround sound, there are fundamentals dealing with audio physics that need to be put in plain words. The basic audio envelope structure.

Diagram 1-Audio Waveform

A) This part of the signal is the first part you hear and is known as the attack of the signal. In most audio waveforms, this part of the signal contains a lot of mid-high frequency content. In an instrument like a piano this is where you will hear the sound of the hammers hitting the first part of the strings producing many high overtones. With a drum, it will be when the stick first hits the head. With dialogue, words that begin with hard consonants, there is no tonal content in this part of the waveform and what is present is a signal containing sonic elements that are similar to noise properties. In dialogue with words like 'Time' the 'T' part of the word contains mostly noise. The "ime" part of the word contains tone-the vowel sound with pitch. It is safe to conclude that when editing dialogue you can literally take any word that begins with 'T' and by its sonic character, use it in other places in the dialogue that contain 'T's at the beginning of a word. This is not true for vowel sounds like the 'ime', for it will contain a certain inflection associated with pitch. B) This part of the signal is a mix of the decay of the attack and the onset of resonance and pitch. As the attack part decays, the first sign of pitch begins. The change from A to B and occurs rather quickly and is not noticeable to the average human ear. C) This part of the signal is the sustain part that contains the resonance and pitch of the signal. This is

where vibrato and tremolo would occur and for those who use pitch correction in singers-this is the part that gets processed. D) This part of the signal is the decay part that occurs when the audio stops projecting. Most of the audio content of the decay part is the reflective sound of the environment. It concert halls this can be as long as 4 seconds and shorter than a quarter of a second for an interior room environment.

Recording In a conventional 5.1 film there is often a desire to place all the close microphones to the orchestra in the front-left, centre and right channels and leave the rear channels for the ambient microphones. In some cases the recording engineer will place 3 or 5 microphones (Decca Tree directly over the front of the orchestra near the conductor), to be panned over the front channels and also place 2 or more microphones in the back of the hall, to record the ambience and then position these microphones in the rear channels. (See Diagram 2) This configuration will most likely have all microphones in an omni pickup pattern but some engineers will switch to cardiod-pickup patterns for a closer sound. An array of spot or close microphones will also be set up close to the instrumental sections in case the composer wants to feature a certain instrument in the mix. However, this introduction of the spot microphones to the mix is random and only used when necessary. This process creates a realistic pick up of the orchestra listened from a fixed distance from the orchestra. This works well in theory, but at times it does not translate very well to the optimum listening position in the theatre (sitting approx 1/2 to 2/3 from the screen). One problem with this approach is that there can be a lot simultaneous audio going on in the film from effects, dialogue and sound design, that you barely notice the rear channels due to the masking effect and lack of rhythmic articulation. Even when there is only music in the mix, the engineer will maintain the Decca setup across the front channels, with the ambient microphones in the rear channels, occasionally turning up the volume of the rear channels to create more of a true realistic hall effect. What might occur when this occurs? With a fast tempo, the music sounds harmonically undefined and inarticulate for the ambience starts to mask the beginning of the next music envelope. Relating to the above diagram, the 'C' part of the waveform elongates, gets louder and will overpower the 'A' part of the waveform of the next incoming signal. Considering that the rhythmic characteristics of music come from the 'A' part of the waveform, you can certainly see how the rhythm can sound obscure for the buildup of the resonance, 'C', will now mask the rhythmic interpretation of the composition. If the envelope has a slow attack time the problem will be exacerbated. In some theatres I've noticed harmonic dissonance due to the fact that a lot of the ambience of the hall microphones, added in with the theatre's RT-60, (reverb decay time) contributes to a longer reverb time for mid-low frequencies in the theatre, creating an effect somewhat like a piano player playing at a fast tempo with the sustain peddle down all the time. In a good recording hall you will get at an approximate reverb time of 1-2 seconds that will be captured by the rear microphones. Add that music pickup in with a large reflective theatre you may get a T-60 time of close to 2-2.5 seconds. This might sound great for Adagios but once the tempo is picked up the music will soon sound harmonically confusing once it arrives to the listeners ears. When I mix music for films, I will do split mixes where I have the close mics mix on one stem and the distant mics and reverb mix on another (Glenn Gould-Siegfried Idyll). This allows me to balance between the articulation and the sustain parts of the music waveform in the final mixing

stage. If the tempo is slow, I can add in more of the ambient microphones and elongate the reverb time to fill out the composition with more harmonic sound duration (The 'C' part of the waveform). If the tempo is brisk I will get a balance of the close microphones (The 'A' part of the waveform) and mix them into the final mix at a level where the rhythmic articulation is clearly heard. Remember, all I need is a small amount of level from the spot microphones to emphasize the "A" part of the waveform. If the music's focus is more on its rhythm than harmonic structure and is secondary to the dialogue, mixing in the close microphones will allow for clarity in the music when the music is mixed in at a lower level compared to the dialogue. Creative approaches to surround music are vast and can add emotion and depth to the mix. Some music engineers feel that staying true to the reality of music perspective is vital, even in movies that have a lot of open audio time between rhythmic beats. "We must stay true to acoustic environmental reality" is their anthem. I feel that one of the main goals of a good film is remove you from reality and transcend you to a place of elsewhere. If you can create a musical landscape that adds a sense of wonder and awe to related visuals without corrupting the visual focus of the film, then go for it I say.

With an orchestra you can widen the stereo image from left side all the way over to right side. Why not pan the left-flank microphone to the phantom centre location between the left front and the left rear and the right-flank microphone to the phantom centre location between the right front and the right rear (See Diagram 2). Why not double the strings and mirror the orchestral positioning in the rear channels. Instead of having the listener in a position where they are hearing the music thirty rows back from the orchestra, place them in the front of the orchestra or even surround them with an orchestra in the front and in the back. I just recently mixed a film where if the viewer in the theatre is sitting about two-thirds back from the screen, it sounds like they are sitting in the conductors listening position.

Diagram 2. Decca Tree 5 Microphone Setup

Decca Tree-3 Microphones

Decca-3 Microphone Setup

Decca Tree 5 Microphone Setup Audio Reflections The best sounding mixes in surround have dimension and perspective where one can actually visualize depth

in the music. To achieve this one needs to understand how direct-sound, reflected- sound and reverb work with each other. How to alter these elements when you are mixing to achieve desired dimensional perspective in creating dimension. Dimension is simply a combination of multiple delays (reflections) and original sound. Once reflections get dense enough, that you can no longer distinguish them as separate individual sounds, they turn into diffused reverb. To use depth effectively one needs to look at music sounding 3-dimensional rather than a 2–dimensional. With creative use of these elements, level, frequency response and time duration you will have the basic knowledge on how to create dimension in mixing. However there are fundamental laws of physic that need to be adhered to when trying to create believable dimension. If you are into creating dimensional landscapes one should posses a basic understanding in how human hearing relates to audio and how to manipulate the various elements. As they say; "If you want to break the rules, you need to know the rules you are breaking". In this age of digital technology, artificial reverberations such as convolution reverb algorithms* are not only more affordable than ever before, but can be easily manipulated in creating believable realism. With a good understanding of the physics of natural acoustic environments, and the fundamental operational principles of reverb processors, it is possible to quickly create the illusion of any acoustic environment you can imagine. First, one needs to know how sound arrives to the ear in certain listening positions in a concert hall and how to re-create this listening position if you want realism in your mix.

Breakdown of an Audio Signal in a Closed Space When it comes to creating the impression of a believable reverb environment, what are the factors that contribute to achieving this? An audio signal takes three different paths while listening in an enclosed environment. The direct signal from its originating source to the listening position, the first and early reflections coming off the walls, ceiling and floor from the source to the listening position and the many diffused reflections know as 'reverb' arriving to the listening position. The individual level and frequency response of these signals determine the size and the quality of the listening environment. The unobstructed direct signal is always the loudest and most defined in its frequency response. The time it takes for the signal to travel from the source to the ear is determined by the speed of sound (approx 1 meter/sec). If the audio source moves slightly to the left, the ear will distinguish this movement for the audio signal will arrive to the left ear slightly sooner than the right ear. This ability to detect the location of the sound source is based of the survival mechanism of human hearing to be able to detect where the source of potential danger is coming from. If for some reason the volume of the source is slightly louder in the right ear, the left ear, which hears the signal first will still inform the brain that the audio source is coming from the left. (Hass Effect). This fundamental principle of hearing will allow the ear to detect the location of the sound source as long as the difference in arrival time to both ears is less than 15 ms. In the mix studio, if a delay arrives within 15ms of the original it will create imaging problems for locating the

position of the original signal. For example, if you have a sound panned centre (direct) and a delay of the original sound source 1-15ms on the right side, what you will hear is the image in the centre shifting to the left. Again, this is caused by the innate characteristics of psycho-aural system of hearing in its relationship to localization. The ear perceives localization because a sound wave will arrive at one ear slightly earlier than the other ear. This is an innate survival mechanism for human behavior. If a delay of 1-15ms is brought back and panned to the same position as the original it will create phasing effects. In a stereo setup, if a reflected signal arrives later than 15ms but before 100ms (approx) from the original and at a lower level, it will create dimension, especially if it arrives to the ear from a different location other than the original sound source. In most listening environments there will be 2 delays called the first reflection, coming from the left and right walls. In most circumstances the delay times will be slightly different from each other but distinct from the direct signal. For example: If you are sitting 18 meters from the sound source between two walls that 30 meters apart, the direct sound will arrive in 18ms and the reflections from the walls will arrive in approx 36 ms. A difference of 18ms. The reflections will be lower in level and contain less high frequencies, for the walls will be absorbing some of the sound. If the left and right delays are exactly 36 ms, than the listener at this position should will determine that the direct sound and the delays will be coming from the same place and that it will sound mono, which is unrealistic in a natural listening environment for it is impossible to have the left and right delays arriving at the same exact time with the same amplitude and frequency response. So to create dimension in an environment like this, one needs to take liberties with the delay settings. First the goal is to create distance behind a localized sound image. To do this, there has to be a minimum delay of at least 15 ms between the direct sound and the delays and 15 ms difference between the right and left delay to prevent image problems caused by the Hass effect. To replicate the above settings, offset the left and right delays from the actual distance delay you are trying to create. If the direct sound takes 18 ms to arrive have the 36 ms delays offset by at least 8 ms each. Have the left delay at 28 ms and the right delay at 44 ms but slightly lower in level and in high frequency content. This will create dimension to the direct signal with the slight appearance that the listener is sitting slightly closer to the left wall for the left reflection is arriving sooner than the right. If the want the listening position to appear further back, have the left delay at 75ms and the right delay at 60 ms, both at a lower level and even less high frequency content. Even though there is a difference in arrival times of the left and right delays, In reality the effect of dimension will greatly over ride the slightly off centre listening position if the direct sound is panned in the middle. It is through this extra delay and altered frequency response that contributes depth to the direct sound. We must also note that the type of envelope, a percussive attack or a slow attack will determine the delay time as sounding dimensional or discrete. The frequency response of the delay will determine the absorption coefficients of the reflective surfaces. What occurs, the psycho-aural response is alerted, which tells you that you are listening to the sound at a distance in a reflective environment. Where as if you just heard the original sound only without reflections, the psycho-aural response would suggest you are listening to a signal while standing elevated in the middle of a field. If you had a signal panned centered and duller sounding reflections of 40ms (left) and 60ms (right) it would sound like you were sitting at a distance, slightly left of center to the left, for the left reflected delay is slightly closer, brighter and louder in relation to the right reflection.

If these reflected signals are very dull sounding, it will imply that the reflective surfaces are absorbing the high frequency content and placing you in an environment of wooden walls rather than glass. When a signal bounces off a surface it will always sound duller than the original for any type of surface absorbs some

sound. The duller the reflection, the higher the absorption co-efficient of the reflective surface. If a reflection is heard after 100-150ms (approx), you will perceive it as a separate form of sound energy and as a discrete delay. When the delay is discrete, it will be easy to localize in the stereo image and might prove to be distracting. So if you have a reflection coming in at 200ms, and it's panned to the left side, you will hear it coming directly from the left and will not prove to be beneficial in creating depth for it is detached from the original signal. For example if you had a percussive instrument like a snare drum and you wanted to add depth to the sound, the delays will have to be in the vicinity of 15-60ms. If the delay is any longer it will sound discrete, for you now hear the difference between the transient of the original drum and the transient of the delay resulting in a confusing sound. A good rule is to remember for adding dimension with percussive elements, the faster the attack of the sound envelope, the shorter the delay will have to be to prevent a discrete delay from appearing. If you would like to create a slap effect coming from the rear to simulate a canyon, then go ahead and add in discrete delays but make sure they are at a lesser volume, duller and not at a time setting that is also a rhythmic factor in the tempo of the piece of music, for the delay will most likely land on a half, quarter, eight or sixteenth note of the tempo and will be masked and hard to hear as a dimensional contribution to the sound. If the instrument happens to be a piano or guitar playing with even dynamics, the delays can be approximatley15-100ms. If the instrument is a violin, the delays can be 70-120ms. The slower the attack, the longer the delay can be in achieving dimension. In a surround setting, if you add in additional longer non-discrete delays to the rear channels you will create an even more realistic listening environment. The delays in the rear channels will have to be longer than the 2 delays in the front left and right channels, yet short enough that they don't sound discrete in the rear channels. Another good rule is when adding in longer delay times, dampen the high frequency content of the delay as the time gets longer. This will create the illusion that the signal is losing fidelity because it is traveling over a longer distance than the original and also indicates that the reflective surface is further away from the listener. Another situation to factor in is that the duller the delay is, the higher the reverb co-efficient of the reflective surface. A delay's frequency response can therefore dictate the reflective properties and distances of the reflective surfaces of the listening environment. It is up to the engineer's discretion on how they want to manipulate the sound of the delays to simulate a realistic listening situation. This creative manipulation of delays works very well with audio that have slow to medium attack times with harmonic content, ambient sound and effects. Generally, any reflections arriving between approx15-100ms in the front channels and 50-150 ms in the rear channels will not affect clarity when equalized and mixed in accordingly. Adding reverb with these delays will create a natural sounding acoustic environment. With additional reverb and correct pre-delay settings, one can create a more realistic environment. In figure 1, you will see the layout of a concert hall with different sound location sources situated at fixed distances from the optimum visual and listening position. The goal here is to figure out what elements contribute to the overall sound from the 3 different listening positions-seated centre but at differing distances from the sound source. If one can figure out what is involved in what is happening to the audio signal at these 3 different listening positions, then it would make sense that if we reverse the scenario where the listening position is stationary and the sound sources can be placed at different distances, than one should be able to manipulate the audio elements to create dimension at the listening position in the theatre. Instead of the listener having to physically move to hear 3 different perspectives, the sound source can be moved to different positions and depth.

Listening Position ' A'

The listener will hear audio in the following relationships, 80% Direct sound from the source to the listener 15% Early reflections 5% Reverb Listening Position ' B' The listener will hear audio in the following relationships, 60% Direct sound from the source to the listener 20% Early reflections 20% Reverb Listening Position ' C' The listener will hear audio in the following relationships, 40% Direct sound from the source to the listener 30% Early reflections 30% Reverb NB: These ratios are approximate and are used to distinguish the different levels of the 3 elements to demonstrate dimension

Listening Placements

Placement "A" In placement "A" the original direct sound (80%) will be almost full-frequency response and arrive to the listening position in approx 5ms (15 feet). The early reflections from the front and surfaces of the hall will be low in level compared to the direct sound because the listener is sitting very close to the original sound source and far from the walls therefore the level of the direct sound will be substantially louder than the reflections. The difference in time between the arrival of the reflections (1st) and the direct time of the source will dictate how far the listener is sitting from the sound source. How much mid-high frequency content in the reflections will dictate what type of reflective surface it is (Absorption coefficient). The diffused reverb will also be low in level in relation to the direct sound and will arrive to the listening position even more delayed than the early reflections. This is because the signal from the sound source continually bounces off the halls surfaces to generate reverb and then takes time to arrive back to the listening position. The farther the reflective surfaces are and the absorption coefficient of these surfaces will dictate the size of the environment and the type of reflective surfaces there is. If you wanted to add in dimension, add short delays to the front channels and longer delays to the rear channel at a lower level and less mid-high frequency content than the front delays. The delays coming from the walls will have to be slightly duller in sound, which is what happens to sound when it bounces off reflective surfaces that absorb mid-high frequencies. The delays coming from the rear will need to be longer and lower in level than the delays coming from the front channels. When adding in delays they must be at least 15ms apart to prevent the Hass effect, which will compromise your imaging. There are two ways of creating dimension with early reflections that gives an option of creating depth and imaging across the front channels. One way is to have both front-left and front-right delays the same time value and mirror the image panning with your bussreverb sends. This will further enhance your imaging with the original sound source coming from anywhere

in the front image, with the delay appearing directly behind it. For example if you wanted the Cello section to appear like it is arriving from centre-right than make sure that the right delay is the same as the left but getting more level from the stereo buss send that is following your panning. This will give you a sound where the Cello section and its first reflection is coming from the same place. This approach will give you a clean image with accurate localization but might not create a more natural sounding depth perspective. The other approach is to have the front-left and front-right channels delays different by at least 15 ms in order to prevent the Hass effect from contributing inaccurate imaging and potential phasing problems. For example: if the Cellos are panned centre-right and you wanted to add depth, then use a delay setting of 20 ms left and 35 ms right. Make sure both delays are at equal volume. This will create a natural sounding dimension rather than reflection imaging. In addition you could also have the right delay shorter than the left delay and louder which will create the idea that the listener is sitting close to the sound source but also close to the right wall. The delay contribution to the mix should be low, for you do not want the delays to corrupt the presence of the direct signal. In position "A" the reverb will be delayed when it arrives back to the listening position for it takes a significant amount of time for the audio signal to go to the walls and diffuse itself into reverb before it arrives back to the listeners ear. The frequency response of the reverb will show that the mid-high frequency content has been rolled off. The actual time difference between the onset of the reverb signal and the original signal and will be approximately between 100-150ms. The longer the pre-delay the duller the reverb should be. Remember what you are trying to create is a full frequency sounding direct signal and reflections sounding duller and the reverb sounding even duller. Most good listening environments that have great acoustics feature halls that roll off more mid-high frequency content over a time period. To achieve a believable position 'A', you will need the contrast of mid-high frequencies between the original direct sound and the delays and reverb. If the reverb is bright it will tell the ear that the walls are reflective and give the impression that the reflective surfaces are hard, somewhat like concrete and unpalatable to the listening experience.

. Here is an example of the delay settings for position 'A' Front Left 20m (less high freq)(9db lower in level) Front Right 35ms (less high freq) (9db lower in level) Rear Left 85ms (lesser high freq)(12db lower in level) Rear right 100ms (lesser high freq)(12db lower in level) Reverb Pre-delay 100 ms (RT=2.5 sec)(High Pass roll-off) In position 'A' you are trying to place a sound source 15 feet in front of you with the goal of the source sounding close and intimate. The original signal needs to reflect off the walls for a while to create reverb and then make its way back to the ear, yet sound distinct from the early reflections. The time for the reverb to arrive at the listening position has to be greater than the time of the latest early reflection for it to make sense and sound believable. The frequency response of the reverb will depend on the reflective properties of the walls. If you wanted the hall environment to sound warm you will have to incorporate a high frequency roll-off on the reverb return. Because the 'A' listening placement is not close to a wall and at a distance to the original sound source you will barely hear any early reflections. Adding in delays to the rear channels the delay of the onset of the reverb will indicate how far the walls are from the listener. The length of the reverb will indicate how live the environment is. The overall sound will be intimate, clear and pleasing to the ear

especially if it is a great singer or soloist performing a ballad. To create this in mixing you will need to add in a reverb that rolls off more high frequency content over the decay of the reverb which means as a reverb gets longer it also gets duller. Roll off the reverb return in the high frequency and low frequency area and maybe slightly boost around 2-2.5K to add a little presence for clarity in the reverb. Watch out for low frequency build up that might clutter the mix. Incorporating a low frequency roll-off in the reverb return around 150hz will help in maintaining articulation in the reverb. In total the original sound will be 80%, early reflections 5% and reverb 15%.

Placement "B" In listening placement "B" (slightly ahead of the exact middle position of the theatre) the original sound source (60%) will have less high end and low end due to the increased distance between the sound source and the listener and will arrive 15ms later to the distance to the listening position. The early reflections (20%) will be arriving from the all the walls. The longer the early reflections are, the greater the distance between the sound source and the listening placement will be. The early reflections inform the psycho-aural response of the listener that they are in an acoustic environment with at least two reflective surfaces. A delay of 40 and 55ms will indicate that the listener is sitting further away from the sound source than with a delay of 20 and 35ms. If you keep the delays relatively close to each other but at least a difference of 15 ms between the left and right channels, you will create the effect that the listening placement is some distance from the original sound source. The levels of the delays not should be the same, where the earliest of the left and right reflections should be slightly louder, for you are trying to create distance instead of imaging across the front channels. It is best to rely on panning the original signal for image positioning and leaving the delays panned hard left and hard right to create the distance effect. If you do desire to create a distant listening placement that is in the centre, the delays from both channels must be the same in time and level. Take note that the even though the delay times are getting longer, the difference between the delays and the original signal is actually getting smaller. Front Left 40m (less high freq) (5db lower in level) Front Right 55ms (less high freq) (5db lower in level) Rear Left 85ms (lesser high freq) (7db lower in level)) Rear right 70ms (lesser high freq) (7db lower in level) Reverb Pre-Delay 75 ms

The reverb will arrive to the "B" listening placement closer to the original signal than in the "A" listening placement. This is because in the "B" placement, the time it takes for the original signal to arrive to the listening placement is 25ms and the onset of reverb begins at approx 100ms. The difference is 75 ms between the arrival of the direct sound and the reverb. In the "A" placement the original signal arrives to the ear in 5 ms and the reverb arrives 100 ms to the "A" placement, a difference of 95 ms. The rule here is that when you want the sound to appear more distant, the time difference between the original direct sound and the early reflections and onset of reverb should get closer in time to each other. Also the reverb frequency response will sound better in relation to the original signal due to the slight degradation of the original sound caused by frequency loss due to the time it takes for sound to arrive to the "B" listening placement from the originating sound source. So to create this dimensional effect make sure that the original sound source does not have an extremely wide frequency response. The depth will be created by two or delays arriving to the listening position between 40-55ms. In the rear channels add in delays of 70-85ms to create the illusion of

rear reflections. Make sure all delays are equal in level have some high frequency roll off so the ear will not confuse the delayed signal with the original signal as being the focus. As previously stated the reverb predelay will have a smaller pre delay time than position "A" and sound slightly brighter to the slight degradation in the "B" original signal. The purpose of the "B" placement allows you to add depth and perspective to audio that needs to situated in a placement that doesn't fight with audio in the "A" placement. Placement "C" Realistic In placement position "C" the original sound, coming from the rear channels (40%), will arrive to the optimum listening placement (25ms) at a lower level than positions "A" and "B" and its frequency response will be even smaller than listening placements "A" and "B". That is not to say you should go out of your way to deteriorate the sonic quality of the sound. It is more like do not go to great efforts to make it sound full. It should contain low end and presence in keeping with the character of the instrument. The early reflections from the rear channels will arrive earlier than the reflections of placement position "B". They will sound less bright than the original sound but will be more prominent in level to the overall sound mix (30%). For the front channels, create additional delays from 70-85ms in to add in the illusion of front reflections. Be careful with transient sounds that might start sounding discrete if the delay times are too long. The reverb will also contribute more to the overall sound and its pre delay time will be even shorter in relation to the direct sound originating in the "C" placement. Because the acoustics of the environment are fixed the reverb decay time should not change dramatically. With Reverb to be believable for placement "C", the pre-delay needs to be small. When the pre-delay gets smaller in time, it indicates to the listening placement that there is distance. Front Left 70m (lesser high freq) (12db lower in level) Front Right 85ms (lesser high freq) (12db lower in level) Rear Left 35ms (less high freq) (3db lower in level)) Rear right 20ms (less high freq) (3db lower in level) Reverb Pre-Delay 75 ms

Placement "C" Creative In placement position "C" the original sound, coming from the rear channels (40%), will arrive to the optimum listening placement (25ms) at a level similar to positions "A". The frequency response will be much lower in the high frequency range, due to the psycho-aural effect of the ear while listening to sound coming from behind the listener. In creating more depth from the rear channels you would need to have a mix that contains a lot of delays between 15-120ms. This will pull the sound back even further and create more distance when added to the front channels. Do not go out of your way to deteriorate the sonic quality of the sound especially in the high-end. It is more like "do not go to great efforts to make it sound full". It should contain low end and presence in keeping with the character of the instrument. The early reflections from the rear channels will be slightly longer and louder than the depth relationship created in listening position "A" or "B". Be careful with transient sounds that might start sounding discrete if the delay times are too long. The reverb will also contribute more to the overall sound and its pre delay time will be even shorter in relation to the direct sound originating in the "C" placement. Because the acoustics of the environment are fixed the reverb decay time should not change dramatically. With reverb to be believable for placement "C", the pre-delay needs to be small. When the pre-delay gets smaller in time, it indicates to the listening

placement that there is distance. Front Left 70m (lesser high freq) (9db lower in level) Front Right 85ms (lesser high freq) (9db lower in level) Rear Left 35ms (less high freq) (3db lower in level)) Rear right 20ms (less high freq) (3db lower in level) Reverb Pre-Delay 20 ms

In most situations the reverb time should be short enough as not to corrupt the harmonic content of the original sound source. If a piano is playing as a harmonic foundation for a melodic idea, oc occasion the harmonic changes will be often and change quickly. If the reverb time is too long audible dissonant effects might prove to be confusing to the articulated harmonic structure of the piece of music. If the BPM is 100 and the playing is sparse a 3 second reverb time will most likely sound fine but if the reverb time gets up to 6 or more seconds, a harmonic mess is almost guaranteed. If one is trying to achieve a very lush reverberant sound it might be a better idea to insert a longer pre-delay time with a short reverb time. In conclusion try to make sure that the reverb time of an instrument playing a harmonic role is not too long where the mix becomes harmonically confusing and dissonant sounding. Overall as we move further back from the sound source the frequency response of the original sound source gets smaller and early reflections and reverb add to the overall sound. As you move further away from the sound source the reflections and the reverb increase in content 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 reflections and the reverb for this is a fundamental rule in creating depth in your mix. If you chose reverb as a pre send the reverb that is generated will still contribute qualities of the original sound source. The frequency response of the reverb dictates the acoustic properties of the reflected materials. If it is hard like concrete the reverb will contain a lot of mid range and high end. If the reflective material is wood it will mainly absorb high and mid range frequencies. Many musicians prefer older concert halls because of their warm acoustical properties that tend to just reflect the musical content of the sound source. Remember that reverb works best when it is treated in a musical context. It can elongate the duration of beautiful melodies; it can create more resonance to drums and add perspective to various harmonic instruments in the mix. If you have a sound source like singing and want the vocal to sound like a ballad performance, you'll find that you can create a recording were the singer sounds close or far away, at the same time. This is a very easy thing to do if you have no 1st reflections, early reflections) and late reflections (highly diffused). You get this by close-miking the original sound and then adding delays, pre-delay to the late energy (reverb), and reverb with hall or a plate setting. If for example you use a reverb setting of say, 3 secs with 100 msec pre delay on the onset of later highly diffused reflections (reverb), you'll find that the vocalist sounds very intimate, a sort of in your face sound but in hall environment. If you like this reverb setup but wish to create more distance between the singer and the listener and don't want to change the reverb decay time, you will have to introduce a series of 1st reflections so you can slightly recess the singer. Just using a standard 3sec reverb setting with no pre-delay and no delays (reflections) will just give you a basic 2 dimensional hall environment without any sense of listener to vocalist distance, no matter how much reverb or length of reverb time you assign to the original signal. By adding the earlier energy and over a wider range we can create what type of a 3 dimensional

sound we desire in our mix. If you add in delays from 50ms-250msec you might create problems with the sound remaining intelligible and clear. However you might want to utilize this for creating a slap back type of effect for lead vocalist. If used effectively it will create distance between the original sound source and the listener. It is also important to decrease the high frequency response to keep the original vocal more present and clear and also allowing you to use more of the delay signal. If using a long reverb time that tends to thin out over time add in a delays based on the rhythmic value of the song. If the song has bpm of 120 a quarter note will equal 500ms. It is important that when you add in a delay to your reverb that it be a fundamental of the rhythm for the landing of the beginning of the delay will also land on an instrument playing on the same beat. This will allow you to increase the delay to your reverb without noticing it as a discrete delay. Obviously if the delay was 400 or 600ms you would hear the delay sounding discrete for it is landing in awkward places in the rhythm of the song. If you add in a deessed slightly regenerated quarter note delay to your reverb sound you will add musical body to the sound of the reverb. Instead of adding a mono 500ms delay add a stereo delay of 490ms (left) and 510ms (right). Make sure the 2 delays are at least 15ms apart to prevent the Haas effect. This stereo delay will enhance the effect of the reverb and still sound in time with the song.

Also insert a low pass filter on the delay so when it regenerates it sounds less bright on each additional delay and more believable to the listener for this is what truly happens in a natural acoustic environment. With the use of digital delays the mixer can create interesting effects that maximize and expand the boundaries of the listening environment and create listening perspectives beyond the walls of the theatre.