Reverb is the effect a room or space has on a sound where the sound we perceive is a mixture of the direct sound and the dense overlapping echoes of that sound reflecting off walls and objects within the space.
Csound's earliest reverb opcodes are reverb and nreverb. By today's standards these sound rather crude and as a consequence modern Csound users tend to prefer the more recent opcodes freeverb and reverbsc.
The typical way to use a reverb is to run as a effect throughout the entire Csound performance and to send it audio from other instruments to which it adds reverb. This is more efficient than initiating a new reverb effect for every note that is played. This arrangement is a reflection of how a reverb effect would be used with a mixing desk in a conventional studio. There are several methods of sending audio from sound producing instruments to the reverb instrument, three of which will be introduced in the coming examples
The first method uses Csound's global variables so that an audio variable created in one instrument and be read in another instrument. There are several points to highlight here. First the global audio variable that is use to send audio the reverb instrument is initialized to zero (silence) in the header area of the orchestra.
This is done so that if no sound generating instruments are playing at the beginning of the performance this variable still exists and has a value. An error would result otherwise and Csound would not run. When audio is written into this variable in the sound generating instrument it is added to the current value of the global variable.
This is done in order to permit polyphony and so that the state of this variable created by other sound producing instruments is not overwritten. Finally it is important that the global variable is cleared (assigned a value of zero) when it is finished with at the end of the reverb instrument. If this were not done then the variable would quickly 'explode' (get astronomically high) as all previous instruments are merely adding values to it rather that redeclaring it. Clearing could be done simply by setting to zero but the clear opcode might prove useful in the future as it provides us with the opportunity to clear many variables simultaneously.
This example uses the freeverb opcode and is based on a plugin of the same name. Freeverb has a smooth reverberant tail and is perhaps similar in sound to a plate reverb. It provides us with two main parameters of control: 'room size' which is essentially a control of the amount of internal feedback and therefore reverb time, and 'high frequency damping' which controls the amount of attenuation of high frequencies. Both there parameters should be set within the range 0 to 1. For room size a value of zero results in a very short reverb and a value of 1 results in a very long reverb. For high frequency damping a value of zero provides minimum damping of higher frequencies giving the impression of a space with hard walls, a value of 1 provides maximum high frequency damping thereby giving the impression of a space with soft surfaces such as thick carpets and heavy curtains.
<CsoundSynthesizer> <CsOptions> -odac ; activates real time sound output </CsOptions> <CsInstruments> ;Example by Iain McCurdy sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gaRvbSend init 0 ; global audio variable initialized to zero instr 1 ; sound generating instrument (sparse noise bursts) kEnv loopseg 0.5,0,0,1,0.003,1,0.0001,0,0.9969,0,0; amp. env. aSig pinkish kEnv ; noise pulses outs aSig, aSig ; audio to outs iRvbSendAmt = 0.8 ; reverb send amount (0 - 1) ; add some of the audio from this instrument to the global reverb send variable gaRvbSend = gaRvbSend + (aSig * iRvbSendAmt) endin instr 5 ; reverb - always on kroomsize init 0.85 ; room size (range 0 to 1) kHFDamp init 0.5 ; high freq. damping (range 0 to 1) ; create reverberated version of input signal (note stereo input and output) aRvbL,aRvbR freeverb gaRvbSend, gaRvbSend,kroomsize,kHFDamp outs aRvbL, aRvbR ; send audio to outputs clear gaRvbSend ; clear global audio variable endin </CsInstruments> <CsScore> i 1 0 300 ; noise pulses (input sound) i 5 0 300 ; start reverb e </CsScore> </CsoundSynthesizer>
The next example uses Csound's zak patching system to send audio from one instrument to another. The zak system is a little like a patch bay you might find in a recording studio. Zak channels can be a, k or i-rate. These channels will be addressed using numbers so it will be important to keep track of what each numbered channel is used for. Our example will be very simple in that we will only be using one zak audio channel. Before using any of the zak opcodes for reading and writing data we must initialize zak storage space. This is done in the orchestra header area using the zakinit opcode. This opcode initializes both a and k rate channels; we must intialize at least one of each even if we don't require both.
zakinit 1, 1
The audio from the sound generating instrument is mixed into a zak audio channel the zawm opcode like this:
zawm aSig * iRvbSendAmt, 1
This channel is read from in the reverb instrument using the zar opcode like this:
aInSig zar 1
Because audio is begin mixed into our zak channel but it is never redefined (only mixed into) it needs to be cleared after we have finished with it. This is accomplished at the bottom of the reverb instrument using the zacl opcode like this:
zacl 0, 1
This example uses the reverbsc opcode. It too has a stereo input and output. The arguments that define its character are feedback level and cutoff frequency. Feedback level should be in the range zero to 1 and controls reverb time. Cutoff frequency should be within the range of human hearing (20Hz -20kHz) and less than the Nyqvist frequency (sr/2) - it controls the cutoff frequencies of low pass filters within the algorithm.
<CsoundSynthesizer> <CsOptions> -odac ; activates real time sound output </CsOptions> <CsInstruments> ; Example by Iain McCurdy sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 ; initialize zak space - one a-rate and one k-rate variable. ; We will only be using the a-rate variable. zakinit 1, 1 instr 1 ; sound generating instrument - sparse noise bursts kEnv loopseg 0.5,0, 0,1,0.003,1,0.0001,0,0.9969,0,0; amp. env. aSig pinkish kEnv ; pink noise pulses outs aSig, aSig ; send audio to outputs iRvbSendAmt = 0.8 ; reverb send amount (0 - 1) ; write to zak audio channel 1 with mixing zawm aSig*iRvbSendAmt, 1 endin instr 5 ; reverb - always on aInSig zar 1 ; read first zak audio channel kFblvl init 0.88 ; feedback level - i.e. reverb time kFco init 8000 ; cutoff freq. of a filter within the reverb ; create reverberated version of input signal (note stereo input and output) aRvbL,aRvbR reverbsc aInSig, aInSig, kFblvl, kFco outs aRvbL, aRvbR ; send audio to outputs zacl 0, 1 ; clear zak audio channels endin </CsInstruments> <CsScore> i 1 0 10 ; noise pulses (input sound) i 5 0 12 ; start reverb e </CsScore> </CsoundSynthesizer>reverbsc contains a mechanism to modulate delay times internally which has the effect of harmonically blurring sounds the longer they are reverberated. This contrasts with freeverb's rather static reverberant tail. On the other hand screverb's tail is not as smooth as that of freeverb, inidividual echoes are sometimes discernible so it may not be as well suited to the reverberation of percussive sounds. Also be aware that as well as reducing the reverb time, the feedback level parameter reduces the overall amplitude of the effect to the point where a setting of 1 will result in silence from the opcode.
A more recent option for sending sound from instrument to instrument in Csound is to use the chn... opcodes. These opcodes can also be used to allow Csound to interface with external programs using the software bus and the Csound API.
<CsoundSynthesizer> <CsOptions> -odac ; activates real time sound output </CsOptions> <CsInstruments> ; Example by Iain McCurdy sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; sound generating instrument - sparse noise bursts kEnv loopseg 0.5,0, 0,1,0.003,1,0.0001,0,0.9969,0,0 ; amp. envelope aSig pinkish kEnv ; noise pulses outs aSig, aSig ; audio to outs iRvbSendAmt = 0.4 ; reverb send amount (0 - 1) ;write audio into the named software channel: chnmix aSig*iRvbSendAmt, "ReverbSend" endin instr 5 ; reverb (always on) aInSig chnget "ReverbSend" ; read audio from the named channel kTime init 4 ; reverb time kHDif init 0.5 ; 'high frequency diffusion' (0 - 1) aRvb nreverb aInSig, kTime, kHDif ; create reverb signal outs aRvb, aRvb ; send audio to outputs chnclear "ReverbSend" ; clear the named channel endin </CsInstruments> <CsScore> i 1 0 10 ; noise pulses (input sound) i 5 0 12 ; start reverb e </CsScore> </CsoundSynthesizer>
Many reverb algorithms including Csound's freeverb, reverb and reverbn are based on what is known as the Schroeder reverb design. This was a design proposed in the early 1960s by the physicist Manfred Schroeder. In the Schroeder reverb a signal is passed into four parallel comb filters the outputs of which are summed and then passed through two allpass filters as shown in the diagram below. Essentially the comb filters provide the body of the reverb effect and the allpass filters smear their resultant sound to reduce ringing artefacts the comb filters might produce. More modern designs might extent the number of filters used in an attempt to create smoother results. The freeverb opcode employs eight parallel comb filters followed by four series allpass filters on each channel. The two main indicators of poor implementations of the Schoeder reverb are individual echoes being excessively apparent and ringing artefacts. The results produced by the freeverb opcode are very smooth but a criticism might be that it is lacking in character and is more suggestive of a plate reverb than of a real room.
The next example implements the basic Schroeder reverb with four parallel comb filters followed by three series allpass filters. This also proves a useful exercise in routing audio signals within Csound. Perhaps the most crucial element of the Schroeder reverb is the choice of loop times for the comb and allpass filters – careful choices here should obviate the undesirable artefacts mentioned in the previous paragraph. If loop times are too long individual echoes will become apparent, if they are too short the characteristic ringing of comb filters will become apparent. If loop times between filters differ too much the outputs from the various filters will not fuse. It is also important that the loop times are prime numbers so that echoes between different filters do not reinforce each other. It may also be necessary to adjust loop times when implementing very short reverbs or very long reverbs. The duration of the reverb is effectively determined by the reverb times for the comb filters. There is certainly scope for experimentation with the design of this example and exploration of settings other than the ones suggested here.
This example consists of five instruments. The fifth instrument implements the reverb algorithm described above. The first four instruments act as a kind of generative drum machine to provide source material for the reverb. Generally sharp percussive sounds provide the sternest test of a reverb effect. Instrument 1 triggers the various synthesized drum sounds (bass drum, snare and closed hi-hat) produced by instruments 2 to 4.
<CsoundSynthesizer> <CsOptions> -odac -m0 ; activate real time sound output and suppress note printing </CsOptions> <CsInstruments> ;Example by Iain McCurdy sr = 44100 ksmps = 1 nchnls = 2 0dbfs = 1 giSine ftgen 0, 0, 2^12, 10, 1 ; a sine wave gaRvbSend init 0 ; global audio variable initialized giRvbSendAmt init 0.4 ; reverb send amount (range 0 - 1) instr 1 ; trigger drum hits ktrigger metro 5 ; rate of drum strikes kdrum random 2, 4.999 ; randomly choose which drum to hit schedkwhen ktrigger, 0, 0, kdrum, 0, 0.1 ; strike a drum endin instr 2 ; sound 1 - bass drum iamp random 0, 0.5 ; amplitude randomly chosen p3 = 0.2 ; define duration for this sound aenv line 1,p3,0.001 ; amplitude envelope (percussive) icps exprand 30 ; cycles-per-second offset kcps expon icps+120,p3,20 ; pitch glissando aSig oscil aenv*0.5*iamp,kcps,giSine ; oscillator outs aSig, aSig ; send audio to outputs gaRvbSend = gaRvbSend + (aSig * giRvbSendAmt) ; add to send endin instr 3 ; sound 3 - snare iAmp random 0, 0.5 ; amplitude randomly chosen p3 = 0.3 ; define duration aEnv expon 1, p3, 0.001 ; amp. envelope (percussive) aNse noise 1, 0 ; create noise component iCps exprand 20 ; cps offset kCps expon 250 + iCps, p3, 200+iCps ; create tone component gliss. aJit randomi 0.2, 1.8, 10000 ; jitter on freq. aTne oscil aEnv, kCps*aJit, giSine ; create tone component aSig sum aNse*0.1, aTne ; mix noise and tone components aRes comb aSig, 0.02, 0.0035 ; comb creates a 'ring' aSig = aRes * aEnv * iAmp ; apply env. and amp. factor outs aSig, aSig ; send audio to outputs gaRvbSend = gaRvbSend + (aSig * giRvbSendAmt); add to send endin instr 4 ; sound 4 - closed hi-hat iAmp random 0, 1.5 ; amplitude randomly chosen p3 = 0.1 ; define duration for this sound aEnv expon 1,p3,0.001 ; amplitude envelope (percussive) aSig noise aEnv, 0 ; create sound for closed hi-hat aSig buthp aSig*0.5*iAmp, 12000 ; highpass filter sound aSig buthp aSig, 12000 ; -and again to sharpen cutoff outs aSig, aSig ; send audio to outputs gaRvbSend = gaRvbSend + (aSig * giRvbSendAmt) ; add to send endin instr 5 ; schroeder reverb - always on ; read in variables from the score kRvt = p4 kMix = p5 ; print some information about current settings gleaned from the score prints "Type:" prints p6 prints "\\nReverb Time:%2.1f\\nDry/Wet Mix:%2.1f\\n\\n",p4,p5 ; four parallel comb filters a1 comb gaRvbSend, kRvt, 0.0297; comb filter 1 a2 comb gaRvbSend, kRvt, 0.0371; comb filter 2 a3 comb gaRvbSend, kRvt, 0.0411; comb filter 3 a4 comb gaRvbSend, kRvt, 0.0437; comb filter 4 asum sum a1,a2,a3,a4 ; sum (mix) the outputs of all comb filters ; two allpass filters in series a5 alpass asum, 0.1, 0.005 ; send mix through first allpass filter aOut alpass a5, 0.1, 0.02291 ; send 1st allpass through 2nd allpass amix ntrpol gaRvbSend, aOut, kMix ; create a dry/wet mix outs amix, amix ; send audio to outputs clear gaRvbSend ; clear global audio variable endin </CsInstruments> <CsScore> ; room reverb i 1 0 10 ; start drum machine trigger instr i 5 0 11 1 0.5 "Room Reverb" ; start reverb ; tight ambience i 1 11 10 ; start drum machine trigger instr i 5 11 11 0.3 0.9 "Tight Ambience" ; start reverb ; long reverb (low in the mix) i 1 22 10 ; start drum machine i 5 22 15 5 0.1 "Long Reverb (Low In the Mix)" ; start reverb ; very long reverb (high in the mix) i 1 37 10 ; start drum machine i 5 37 25 8 0.9 "Very Long Reverb (High in the Mix)" ; start reverb e </CsScore> </CsoundSynthesizer>
This chapter has introduced some of the more recent Csound opcodes for delay-line based reverb algorithms which in most situations can be used to provide high quality and efficient reverberation. Convolution offers a whole new approach for the creation of realistic reverbs that imitate actual spaces - this technique is demonstrated in the Convolution chapter.
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