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Category: Reverb

An improved Reverb design in Synthedit .

Using a single Delay2 for a very basic reverb effect is all well and good, but has its limitations, one of which is that at the short delay times we are using there can be noticeable peaks and troughs in the frequency response, as what we are setting up is basically a comb filter, so we can get some ringing effects on transients at certain frequencies, and also a “drainpipe” like effect on other sounds. Not good. So what’s the answer?

The Schroeder Model of Reverb.

If we put a sharp transient sound such as a handclap into a delay line then
exponentially decaying impulses follow the first impulse. Though this
is similar to an exponentially decaying reverb, in the frequency
response peaks occur at equally spaced frequencies like the teeth of a
comb, hence the name comb filter.

How sound decays (fades away) in a room or hall.

This will result in a ringing metallic sound, especially once feedback is added into the mix. Digital audio pioneer Manfred Schroeder proposed using parallel delays with differing delay times, in combination with all-pass filters for digital reverb. His idea proposed a structure with four parallel delays and two serial all-pass filters. The delays create the reflections, and the all-pass filters “smear” any transients thus making the reverb more diffuse and reducing any resonances. Schroeder’s design is shown below in block form. Each comb is a Delay2 module.

Block diagram of the Schroeder type reverb
The complete Schroeder emulation in SynthEdit

6 x Delay container:
A set of six Delay2 modules pre-set to differing delay times within the range of 10 to 60 mS which is the ideal range for reverb, any longer and we start to get an echo effect, and shorter and there’s no noticeable reverb effect.
All the outputs from the Delay2 modules are fed into the Input 1 of a Divide module, this is because if we just feed all the signals into a normal module they will all be added together, more than likely exceeding the input range of the module causing clipping. Input 2 of the divide module is set to 6 (the number of delay modules) to restore the normal signal range. There have been various calculations made for the “ideal” delay times, but in practice a natural reverb depends on the shape and size of the area, and the things in that area.
There has been a calculation made of the “ideal” delay times which are
50, 53, 61, 68, 72, and 78 ms

The structure of the 6 X Delay container

All Pass Container:
The All Pass container uses two more Delay two modules in series one set to 17mS, and another to 74mS these two delays give an added “blurring” effect to any transients, along with a 1pole Low Pass filter for variable HF damping on the reverb “tail”

The structure of the all-pass container

Feedback Container:
The feedback container has a Feedback – Volts module to allow feedback. A combination of the inherent time delay in a feedback loop in SynthEdit, and the two All Pass filter modules adds further frequency dependant phase shifting to further reduce ringing and metallic sounding transients. Again the frequencies of the All Pass filters are fairly random, but spaced out to try and give maximum effectiveness in reducing unwanted audio artefacts.

The structure of the Feedback container

Creating a very basic Reverb effect in Synthedit with the Delay2 Module.

BBD Reverbs.
First there were plate and spring line reverbs, then along came an electronic “chip” called a Bucket Brigade Delay (BBD). This used high impedance CMOS devices and capacitors to pass the electric charge (audio) along the line, each stage in the line adding a small delay. These worked well, and eliminated the problems of external noise entering the signal chain. However the way they worked introduced their own artefacts such as aliasing at high frequencies, filtering was used filtering to restrict this, and they also introduced a certain amount of distortion, which gave them their own unique low-fi sound. By using multiple chips with different delay times you could get some quite complex reverbs.

Delay2.

Creates a delay (echo or reverb) effect on audio signals very similar to BBD chips.
Note: If you intend to modulate the delay time, you should enable interpolation, this reduces clicks and “zipper” noise.

The SynthEdit Delay2 module

Plugs.
Left Hand Side:
Signal In:- (Voltage) Input Signal
Modulation:- (Voltage) Varies the delay time dynamically (0 to 10V)
Feedback:- (Voltage) Controls the amount of feedback of the delayed signal.

Right Hand Side:
Signal Out: – (Voltage) Audio output signal
Parameters:
Delay Time (secs): – Maximum delay time in Seconds. The delay time is limited to a maximum of 10 seconds.
Interpolate Output: – Provides smoother modulation of delay time, but with an increase in CPU load.

Basis BBD style, single delay reverb effect.
We can emulate this in a SynthEdit module, which takes the audio input, and much like the old BBD chips, slices up the audio into samples and passes them down the delay line.
Shown below is our very basic reverb effect.
Delay Time:- Set the delay time to 0.001 Seconds (10mS) in properties
Delay control:– Set the maximum voltage for the delay slider to 0.06 volts, this gives us a delay time range of 10mS to 70mS.
Feedback:– Set the maximum voltage for the Feedback slider to 8 volts, as we don’t want the absolute maximum feedback level as it will just give uncontrolled oscillation.
Wet/Dry:- This control will default to a range of -5 volts to +5 volts. Leave this as it is to give the full range of wet to dry effect.

Very basic reverb structure

There’s one important point to note, if this reverb is being included as part of a Synthesizer VST we need to think about polyphony. It really isn’t needed for a reverb effect so we need to containerise our reverb, put a Voice Combiner module in place in the input- the voice combiner forces then the Modules in this container to stay Monophonic. Otherwise every time you play more than one note on your Synth plug in it will create an un-necessary clone of the reverb module wasting CPU and memory.

Optimising CPU with your reverb

Making the Reverb Stereo.

Stereo Reverb

Converting to Stereo is easy, we just create two identical Reverb chains, one for the left channel, and one for the right channel. Keep the two delay and feedback levels linked otherwise some very peculiar (and unwanted) phase and stereo imaging effects can be introduced.

Limitations of basic Delay reverbs.

This is structure is OK for a very basic reverb effect, but has its limitations, one of which is that at the short delay times we are using there can be noticeable peaks and troughs in the frequency response, as what we are setting up is basically a comb filter, so we can get some ringing effects on transients at certain frequencies, and also a “drainpipe” like effect on other sounds. Not so good. So what’s the answer?
Fortunately there is one that was devised by Manfred Schroeder, known as the Schroeder Model of reverb.

Reverb. What exactly is it, and how can we create it in SynthEdit?-The basics.

Reverberation or reverb, in music is a persistence of sound, or a very short echo after a sound is produced. Reverb is created when a sound or signal is made inside an enclosed or semi enclosed area, the reflected sounds from the enclosing walls causing complex sound reflections to build up and then decay as the sound is absorbed by the objects in the space – which could include furniture, people, walls etc. This is most noticeable when the sound source stops and the reverberations continue, their volume decreasing until zero is reached. Reverb is pitch dependent, as sounds of different pitches will arrive back to the listener at slightly different times, and unless the room is very carefully designed it will have a resonant frequency at which sounds will be emphasized. 
In comparison to a distinct echo, which usually 50 to 100 ms after the previous sound, reverb is made up of reflections of sounds that arrive in less than about 50 ms. As time passes, the level of the reflections gradually reduces to non-noticeable levels. Anywhere there are surfaces to reflect sounds from you get reverb. The more complex the shape of the room and the more objects in it the more complex the reverb is. Certain frequencies can be boosted, and others cut due to the wavelength of the sounds interacting with the size of the room (resonance). It may be created through physical means, such as echo chambers, or electronically through audio signal processing. There are various means of achieving a reverb effect listed below.

Echo chambers
The first reverb effects, introduced in the 1930s, were created by playing recordings through loudspeakers in medium to large spaces (various spaces such as empty rooms, bathrooms and even stairwells have been used) and mixing the sound with the original using strategically placed microphones.
The American producer Bill Putnam is credited for the first artistic use of artificial reverb in music, on the 1947 song “Peg o’ My Heart” by the Harmonicats. Putnam placed a microphone and loudspeaker in the studio bathroom to create an echo chamber, adding an “eerie dimension”.
The first two examples of reverb (Spring and Plate) are included just to give some historical background as we are going to be using digital sound processing to achieve our reverb effects.

Plate reverb
A plate reverb system uses an electromechanical transducer, similar to the driver in a loudspeaker, to create vibrations in a large plate of sheet metal. The plate’s motion is picked up by one or more contact microphones. The audio signal from these is then mixed with the original “dry” signal. Plate reverb was introduced in the late 1950s by Elektromesstechnik with their EMT 140 design. The greatest problem with plate reverb units is their size and weight, which limits their use to recording studios.

A spring reverb.
Spring reverbs, were introduced by the company Bell Labs, using a set of springs mounted inside a box. They work in a similar way to plate reverb, with a transducer at one end of the spring, and a pickup placed at the far end of the spring, to reproduce more realistic reverb different length springs could be mixed together (longer spring = longer delay time) .
They can have a very distinctive “twangy” sound with loud percussive sounds due to the springs. One major drawback is sensitivity to outside vibration.
They were popular in the 1960s, and were first used by the Hammond company to add reverb to Hammond organs.
They became popular with guitarists, including surf musicians such as the Beach Boys, and spring reverb could easily be built into guitar amplifiers.
They were also used by dub reggae musicians such as King Tubby. The American engineer Laurens Hammond of the Hammond company was granted a patent on a spring reverb system in 1939.

Digital reverb
Digital reverb units simulate reverb by using multiple delay lines that have different delay times and variable feedback, giving the impression of sound bouncing off of multiple surfaces. Some digital effects allow users to independently adjust early and late reflections. Digital reverb was introduced in 1976 by EMT with the EMT 250, and became increasingly popular with many groups and studios in the 1980s.

Gated reverb
Gated reverb combines reverb with a noise gate, creating a “large” reverb sound with a short tail (the tail is cut short by the noise gate).
It was pioneered by the English recording engineer Hugh Padgham and the drummer Phil Collins, and became a staple of 1980s pop music.

Convolution reverb.
Convolution uses impulse responses to record the reverberation of physical spaces and recreate them digitally. The first real-time convolution reverb processor, the DRE S777, was announced by Sony in 1999.
Convolution reverb is often used in film production, with sound engineers recording impulse responses of sets and locations so sounds can be added in post-production with realistic sounding reverberation.
Basically, a convolution reverb takes an input signal (the sound to be reverberated) and processes it with the sound of an actual or virtual acoustic space to create the illusion that the input was recorded in that space. The sound of the acoustic space is captured in what is called an impulse response (IR), which often starts as a recording of a short, sharp sound, such as the firing of a starter pistol or the bursting of an inflated balloon (the impulse), in the acoustic space in question. As you can imagine, such a sound excites the reverberation (the response) in the space, and so the impulse response (or at least its initial recording) sounds like an explosion followed by the reverberating reflections created by the recording space.
Once you have the IR for the space as a file, you then load it into the convolution reverb and input your sound to be processed. At that point, the software convolves the two digital audio signals together to create the output. Convolution itself is a mathematical process that has many applications including statistics, image processing, and electrical engineering as well as audio processing (It’s a very complex subject, understanding the precise details of how it works is not easy!). If you like, you can think of convolution as a kind of multiplication of each sample of the input with each sample of the IR, with the result being that the audio input takes on the sonic characteristics of the space in which the original IR was recorded.

Shimmer Reverb.
Shimmer reverb alters the pitch of the reverberated sound up or down in frequency by placing a pitch shifter in the delay lines feedback loop.
Once we apply any feedback in the reverb the pitch of the reverberated sound will steadily change up or down in pitch depending on the amount of pitch change and the feedback level. Shimmer reverb is often used in ambient music. Brian Eno has been one of the earliest adopters of this sound.