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CSOUND

PANNING AND SPATIALIZATION

Simple Stereo Panning 

Csound provides a large number of opcodes designed to assist in the distribution of sound amongst two or more speakers. These range from opcodes that merely balance a sound between two channels to one that include algorithms to simulate the doppler shift which occurs when sound moves, algorithms that simulate the filtering and inter-aural delay that occurs as sound reaches both our ears and algorithms that simulate distance in an acoustic space.

First we will look at some methods of panning a sound between two speakers based on first principles.

The simplest method that is typically encountered is to multiply one channel of audio (aSig) by a panning variable (kPan) and to multiply the other side by 1 minus the same variable like this:

aSigL  =  aSig * kPan
aSigR  =  aSig * (1 – kPan)
          outs aSigL, aSigR

kPan should be a value within the range zero and 1. If kPan is 1 all of the signal will be in the left channel, if it is zero, all of the signal will be in the right channel and if it is 0.5 there will be signal of equal amplitude in both the left and the right channels. This way the signal can be continuously panned between the left and right channels.

The problem with this method is that the overall power drops as the sound is panned to the middle.

One possible solution to this problem is to take the square root of the panning variable for each channel before multiplying it to the audio signal like this:

aSigL  =     aSig * sqrt(kPan)
aSigR  =     aSig * sqrt((1 – kPan))
       outs  aSigL, aSigR

By doing this, the straight line function of the input panning variable becomes a convex curve, so that less power is lost as the sound is panned centrally.

Using 90º sections of a sine wave for the mapping produces a more convex curve and a less immediate drop in power as the sound is panned away from the extremities. This can be implemented using the code shown below.

aSigL  =     aSig * sin(kPan*$M_PI_2)
aSigR  =     aSig * cos(kPan*$M_PI_2)
       outs  aSigL, aSigR

(Note that '$M_PI_2' is one of Csound's built in macros and is equivalent to pi/2.)

A fourth method, devised by Michael Gogins, places the point of maximum power for each channel slightly before the panning variable reaches its extremity. The result of this is that when the sound is panned dynamically it appears to move beyond the point of the speaker it is addressing. This method is an elaboration of the previous one and makes use of a different 90 degree section of a sine wave. It is implemented using the following code:

aSigL  =     aSig * sin((kPan + 0.5) * $M_PI_2)
aSigR  =     aSig * cos((kPan + 0.5) * $M_PI_2)
       outs  aSigL, aSigR

The following example demonstrates all three methods one after the other for comparison. Panning movement is controlled by a slow moving LFO. The input sound is filtered pink noise.

 

   EXAMPLE 05B01_Pan_stereo.csd

<CsoundSynthesizer>

<CsOptions>
-odac ; activates real time sound output
</CsOptions>

<CsInstruments>
sr = 44100
ksmps = 10
nchnls = 2
0dbfs = 1

  instr 1
imethod  =         p4 ; read panning method variable from score (p4)

;---------------- generate a source sound -------------------
a1       pinkish   0.3            ; pink noise
a1       reson     a1, 500, 30, 1 ; bandpass filtered
aPan     lfo       0.5, 1, 1      ; panning controlled by an lfo
aPan     =         aPan + 0.5     ; offset shifted +0.5
;------------------------------------------------------------

 if imethod=1 then
;------------------------ method 1 --------------------------
aPanL    =         aPan
aPanR    =         1 - aPan
;------------------------------------------------------------
 endif

 if imethod=2 then
;------------------------ method 2 --------------------------
aPanL    =       sqrt(aPan)
aPanR    =       sqrt(1 - aPan)
;------------------------------------------------------------
 endif

 if imethod=3 then
;------------------------ method 3 --------------------------
aPanL    =       sin(aPan*$M_PI_2)
aPanR    =       cos(aPan*$M_PI_2)
;------------------------------------------------------------
 endif

 if imethod=4 then
;------------------------ method 4 --------------------------
aPanL   =  sin((aPan + 0.5) * $M_PI_2)
aPanR   =  cos((aPan + 0.5) * $M_PI_2)
;------------------------------------------------------------
 endif

         outs    a1*aPanL, a1*aPanR ; audio sent to outputs
  endin

</CsInstruments>

<CsScore>
; 4 notes one after the other to demonstrate 4 different methods of panning
; p1 p2  p3   p4(method)
i 1  0   4.5  1
i 1  5   4.5  2
i 1  10  4.5  3
i 1  15  4.5  4
e
</CsScore>
</CsoundSynthesizer>

 

An opcode called pan2 exists which makes it slightly easier for us to implement various methods of panning. The following example demonstrates the three methods that this opcode offers one after the other. The first is the 'equal power' method, the second 'square root' and the third is simple linear. The Csound Manual describes a fourth method but this one does not seem to function currently.

 

   EXAMPLE 05B02_pan2.csd

<CsoundSynthesizer>
<CsOptions>
-odac ; activates real time sound output
</CsOptions>

<CsInstruments>
sr = 44100
ksmps = 10
nchnls = 2
0dbfs = 1

  instr 1
imethod        =         p4 ; read panning method variable from score (p4)
;----------------------- generate a source sound ------------------------
aSig           pinkish   0.5              ; pink noise
aSig           reson     aSig, 500, 30, 1 ; bandpass filtered
;------------------------------------------------------------------------

;---------------------------- pan the signal ----------------------------
aPan           lfo       0.5, 1, 1        ; panning controlled by an lfo
aPan           =         aPan + 0.5       ; DC shifted + 0.5
aSigL, aSigR   pan2      aSig, aPan, imethod; create stereo panned output
;------------------------------------------------------------------------

               outs      aSigL, aSigR     ; audio sent to outputs
  endin

</CsInstruments>

<CsScore>
; 3 notes one after the other to demonstrate 3 methods used by pan2
;p1 p2  p3   p4
i 1  0  4.5   0 ; equal power (harmonic)
i 1  5  4.5   1 ; square root method
i 1 10  4.5   2 ; linear
e
</CsScore>
</CsoundSynthesizer> 

In the next example we will generate some sounds as the primary signal. We apply some delay and reverb to this signal to produce a secondary signal. A random function will pan the primary signal between the channels, but the secondary signal remains panned in the middle all the time.

   EXAMPLE 05B03_Different_pan_layers.csd

<CsoundSynthesizer>
<CsOptions>
-o dac -d
</CsOptions>

<CsInstruments>
; Example by Bjorn Houdorf, March 2013

sr = 44100
ksmps = 32
nchnls = 2
0dbfs = 1
           seed       0

instr 1
ktrig      metro      0.8; Trigger frequency, instr. 2
           scoreline  "i 2 0 4", ktrig
endin

instr 2
ital       random     60, 72; random notes
ifrq       =          cpsmidinn(ital)
knumpart1  oscili     4, 0.1, 1
knumpart2  oscili     5, 0.11, 1
; Generate primary signal.....
asig       buzz       0.1, ifrq, knumpart1*knumpart2+1, 1
ipan       random     0, 1; ....make random function...
asigL, asigR pan2     asig, ipan, 1; ...pan it...
           outs       asigL, asigR ;.... and output it..
kran1      randomi    0,4,3
kran2      randomi    0,4,3
asigdel1   delay      asig, 0.1+i(kran1)
asigdel2   delay      asig, 0.1+i(kran2)
; Make secondary signal...
aL, aR     reverbsc   asig+asigdel1, asig+asigdel2, 0.9, 15000
           outs       aL, aR; ...and output it
endin
</CsInstruments>

<CsScore>
f1 0 8192 10 1
i1 0 60
</CsScore>
</CsoundSynthesizer>

3D Binaural Encoding 

3D binaural encoding is available through a number of opcodes that make use of spectral data files that provide information about the filtering and inter-aural delay effects of the human head. The oldest one of these is hrtfer. Newer ones are hrtfmove, hrtfmove2 and hrtfstat. The main parameters for control of the opcodes are azimuth (the horizontal direction of the source expressed as an angle formed from the direction in which we are facing) and elevation (the angle by which the sound deviates from this horizontal plane, either above or below). Both these parameters are defined in degrees. 'Binaural' infers that the stereo output of this opcode should be listened to using headphones so that no mixing in the air of the two channels occurs before they reach our ears (although a degree of effect is still audible through speakers).

The following example take a monophonic source sound of noise impulses and processes it using the hrtfmove2 opcode. First of all the sound is rotated around us in the horizontal plane then it is raised above our head then dropped below us and finally returned to be level and directly in front of us. For this example to work you will need to download the files hrtf-44100-left.dat and hrtf-44100-right.dat (a place to download them may be here) and place them in your SADIR (see setting environment variables) or in the same directory as the .csd.

 

   EXAMPLE 05B04_hrtfmove.csd

<CsoundSynthesizer>
<CsOptions>
-odac ; activates real time sound output
</CsOptions>

<CsInstruments>
; Example by Iain McCurdy

sr = 44100
ksmps = 10
nchnls = 2
0dbfs = 1

giSine         ftgen       0, 0, 2^12, 10, 1             ; sine wave
giLFOShape     ftgen       0, 0, 131072, 19, 0.5,1,180,1 ; U-shape parabola

  instr 1
; create an audio signal (noise impulses)
krate          oscil       30,0.2,giLFOShape            ; rate of impulses
; amplitude envelope: a repeating pulse
kEnv           loopseg     krate+3,0, 0,1, 0.05,0, 0.95,0,0
aSig           pinkish     kEnv                             ; noise pulses

; -- apply binaural 3d processing --
; azimuth (direction in the horizontal plane)
kAz            linseg      0, 8, 360
; elevation (held horizontal for 8 seconds then up, then down, then horizontal
kElev          linseg      0, 8,   0, 4, 90, 8, -40, 4, 0
; apply hrtfmove2 opcode to audio source - create stereo ouput
aLeft, aRight  hrtfmove2   aSig, kAz, kElev, \
                               "hrtf-44100-left.dat","hrtf-44100-right.dat"
               outs        aLeft, aRight                 ; audio to outputs
endin

</CsInstruments>

<CsScore>
i 1 0 24 ; instr 1 plays a note for 24 seconds
e
</CsScore>
</CsoundSynthesizer>

Going Multichannel

So far we have only considered working in 2-channels/stereo, but Csound is extremely flexible at working in more that 2 channels. By changing nchnls in the orchestra header we can specify any number of channels but we also need to ensure that we choose an audio hardware device using -odac that can handle multichannel audio. Audio channels sent from Csound, that do not address hardware channels, will simply not be reproduced. There may be some need to make adjustments to the software settings of your soundcard using its own software or the operating system's software, but due to the variety of sound hardware options available, it would be impossible to offer further specific advice here.

Sending Multichannel Sound to the Loudspeakers

In order to send multichannel audio we must use opcodes designed for that task. So far we have used outs to send stereo sound to a pair of loudspeakers. (The 's' actually stands for 'stereo'). Correspondingly there exist opcodes for quadophonic (outq), hexaphonic (outh), octophonic (outo), 16-channel sound (outx) and 32-channel sound (out32).

For example:

 outq  a1, a2, a3, a4

sends four independent audio streams to four hardware channels. Any unrequired channels still have to be given an audio signal. A typical workaround would be to give them 'silence'. For example if only 5 channels were required:

nchnls   =  6

; --snip--

aSilence =    0
         outh a1, a2, a3, a4, a5, aSilence

These opcodes only address very specific loudspeaker arrangements (although workarounds are possible) and have been superseded, to a large extent, by newer opcodes that allow greater flexibility in the number and routing of audio to a multichannel output.

outc allows us to address any number of output audio channels, but they still need to be addressed sequentially. For example our 5-channel audio could be design as follows:

nchnls   =  5

; --snip--

    outc a1, a2, a3, a4, a5

outch allows us to direct audio to a specific channel or list of channels and takes the form:

outch kchan1, asig1 [, kchan2] [, asig2] [...]

For example, our 5-channel audio system could be designed using outch as follows:

nchnls   =  5

; --snip--

    outch 1,a1, 2,a2, 3,a3, 4,a4, 5,a5

Note that channel numbers can be changed at k-rate thereby opening the possibility of changing the speaker configuration dynamically during performance. Channel numbers do not need to be sequential and unrequired channels can be left out completely. This can make life much easier when working with complex systems employing many channels.

Flexibly Moving Between Stereo and Multichannel

It may be useful to be able to move between working in multichannel (beyond stereo) and then moving back to stereo (when, for example, a multichannel setup is not available). It won't be sufficient to simply change nchnls = 2. It will also be necessary to change all outq, outo, outch etc to outs. In complex orchestras this could laboursome and particularly so if it is required to go back to a multichannel configuration later on. In this situation conditional outputs based on the nchnls value are useful. For example:

 if nchnls==4 then
     outq  a1,a2,a3,a4
 elseif nchnls==2 then
     outs  a1+a3, a2+a4
 endif

Using this method, it will only be required to change nchnls = ... in the orchestra header. In stereo mode, if nchnls = 2, at least all audio streams will be monitored, even if the results do not reflect the four channel spatial arrangement.

Rendering Multichannel Audio Streams as Sound Files

So far we have referred to outs, outo etc. as a means to send audio to the speakers but strictly speaking they are only sending audio to Csound's output (as specified by nchnls) and the final destination will be defined using a command line flag in <CsOptions></CsOptions>. -odac will indeed instruct Csound to send audio to the audio hardware and then onto the speakers but we can alternatively send audio to a sound file using -oSoundFile.wav. Provided a file type that supports multichannel interleaved data is chosen (".wav" will work), a multichannel file will be created that can be used in some other audio applications or can be re-read by Csound later on by using, for example, diskin2. This method is useful for rendering audio that is too complex to be monitored in real-time. Only single interleaved sound files can be created, separate mono files cannot be created using this method. Simultaneously monitoring the audio generated by Csound whilst rendering will not be possible when using this method; we must choose one or the other.

An alternative method of rendering audio in Csound, and one that will allow simulatenous monitoring in real-time, is to use the fout opcode. For example:

fout  "FileName.wav", 8, a1, a2, a3, a4
outq  a1, a2, a3, a4

 

will render an interleaved, 24-bit, 4-channel sound file whilst simultaneously sending the quadrophonic audio to the loudspeakers.

If we wanted to de-interleave an interleaved sound file into multiple mono sound files we could use the code:

a1, a2, a3, a4   soundin   "4ChannelSoundFile.wav"
 fout      "Channel1.wav", 8, a1
 fout      "Channel2.wav", 8, a2
 fout      "Channel3.wav", 8, a3
 fout      "Channel4.wav", 8, a4 

VBAP

Vector Base Amplitude Panning1  can be described as a method which extends stereo panning to more than two speakers. The number of speakers is, in general, arbitrary. You can configure for standard layouts such as quadrophonic, octophonic or 5.1 configuration, but in fact any number of speakers can be positioned even in irregular distances from each other. If you are fortunate enough to have speakers arranged at different heights, you can even configure VBAP for three dimensions.

Basic Steps

First you must tell VBAP where your loudspeakers are positioned. Let us assume you have seven speakers in the positions and numberings outlined below (M = middle/centre):


The opcode vbaplsinit, which is usually placed in the header of a Csound orchestra, defines these positions as follows:

vbaplsinit 2, 7, -40, 40, 70, 140, 180, -110, -70

The first number determines the number of dimensions (here 2). The second number states the overall number of speakers, then followed by the positions in degrees (clockwise).

All that is required now is to provide vbap with a monophonic sound source to be distributed amongst the speakers according to information given about the position. Horizontal position (azimuth) is expressed in degrees clockwise just as the initial locations of the speakers were. The following would be the Csound code to play the sound file "ClassGuit.wav" once while moving it counterclockwise:

 

   EXAMPLE 05B05_VBAP_circle.csd

 

<CsoundSynthesizer>
<CsOptions>
-odac -d ;for the next line, change to your folder
--env:SSDIR+=/home/jh/Joachim/Csound/FLOSS/audio
</CsOptions>
<CsInstruments>
sr = 44100
ksmps = 32	
0dbfs = 1
nchnls = 7

vbaplsinit 2, 7, -40, 40, 70, 140, 180, -110, -70

  instr 1
Sfile      =          "ClassGuit.wav"
iFilLen    filelen    Sfile
p3         =          iFilLen
aSnd, a0   soundin    Sfile
kAzim      line       0, p3, -360 ;counterclockwise
a1, a2, a3, a4, a5, a6, a7, a8 vbap8 aSnd, kAzim
outch 1, a1, 2, a2, 3, a3, 4, a4, 5, a5, 6, a6, 7, a7
  endin
</CsInstruments>
<CsScore>
i 1 0 1
</CsScore>
</CsoundSynthesizer>
;example by joachim heintz

In the CsOptions tag, you see the option --env:SSDIR+= ... as a possibility to add a folder to the path in which Csound usually looks for your samples (SSDIR = Sound Sample Directory) if you call them only by name, without the full path. To play the full length of the sound file (without prior knowledge of its duration) the filelen opcode is used to derive this duration, and then the duration of this instrument (p3) is set to this value. The p3 given in the score section (here 1) is overwritten by this value.

The circular movement is a simple k-rate line signal, from 0 to -360 across the duration of the sound file (in this case the same as p3). Note that we have to use the opcode vbap8 here, as there is no vbap7. Just give the eighth channel a variable name (a8) and thereafter ignore it.

The Spread Parameter

As VBAP derives from a panning paradigm, it has one problem which becomes more serious as the number of speakers increases. Panning between two speakers in a stereo configuration means that all speakers are active. Panning between two speakers in a quadro configuration means that half of the speakers are active. Panning between two speakers in an octo configuration means that only a quarter of the speakers are active and so on; so that the actual perceived extent of the sound source becomes unintentionally smaller and smaller.

To alleviate this tendency, Ville Pulkki has introduced an additional parameter, called 'spread', which has a range of zero to hundred percent.2  The 'ascetic' form of VBAP we have seen in the previous example, means: no spread (0%). A spread of 100% means that all speakers are active, and the information about where the sound comes from is nearly lost.

As the kspread input to the vbap8 opcode is the second of two optional parameters, we first have to provide the first one. kelev defines the elevation of the sound - it is always zero for two dimensions, as in the speaker configuration in our example. The next example adds a spread movement to the previous one. The spread starts at zero percent, then increases to hundred percent, and then decreases back down to zero.

 

   EXAMPLE 05B06_VBAP_spread.csd

<CsoundSynthesizer>
<CsOptions>
-odac -d ;for the next line, change to your folder
--env:SSDIR+=/home/jh/Joachim/Csound/FLOSS/audio
</CsOptions>
<CsInstruments>
sr = 44100
ksmps = 32	
0dbfs = 1
nchnls = 7

vbaplsinit 2, 7, -40, 40, 70, 140, 180, -110, -70

  instr 1
Sfile      =          "ClassGuit.wav"
iFilLen    filelen    Sfile
p3         =          iFilLen
aSnd, a0   soundin    Sfile
kAzim      line       0, p3, -360
kSpread    linseg     0, p3/2, 100, p3/2, 0
a1, a2, a3, a4, a5, a6, a7, a8 vbap8 aSnd, kAzim, 0, kSpread
outch 1, a1, 2, a2, 3, a3, 4, a4, 5, a5, 6, a6, 7, a7
  endin
</CsInstruments>
<CsScore>
i 1 0 1
</CsScore>
</CsoundSynthesizer>
;example by joachim heintz

New VBAP Opcodes

As a response to a number of requests, John fFitch has written new VBAP opcodes in 2012 whose main goal is to allow more than one loudspeaker configuration within a single orchestra (so that you can switch between them during performance) and to provide more flexibility in the number of output channels used. Here is an example for three different configurations which are called in three different instruments:

 

   EXAMPLE 05B07_VBAP_new.csd

<CsoundSynthesizer>
<CsOptions>
-odac -d ;for the next line, change to your folder
--env:SSDIR+=/home/jh/Joachim/Csound/FLOSS/audio
</CsOptions>
<CsInstruments>
sr = 44100
ksmps = 32	
0dbfs = 1
nchnls = 7

vbaplsinit 2.01, 7, -40, 40, 70, 140, 180, -110, -70
vbaplsinit 2.02, 2, -40, 40
vbaplsinit 2.03, 3, -70, 180, 70

  instr 1
aSnd, a0   soundin    "ClassGuit.wav"
kAzim      line       0, p3, -360
a1, a2, a3, a4, a5, a6, a7 vbap aSnd, kAzim, 0, 0, 1
outch 1, a1, 2, a2, 3, a3, 4, a4, 5, a5, 6, a6, 7, a7
  endin

  instr 2
aSnd, a0   soundin    "ClassGuit.wav"
kAzim      line       0, p3, -360
a1, a2     vbap       aSnd, kAzim, 0, 0, 2
           outch      1, a1, 2, a2
  endin

  instr 3
aSnd, a0   soundin    "ClassGuit.wav"
kAzim      line       0, p3, -360
a1, a2, a3 vbap       aSnd, kAzim, 0, 0, 3
           outch      7, a1, 3, a2, 5, a3
  endin

</CsInstruments>
<CsScore>
i 1 0 6
i 2 6 6
i 3 12 6
</CsScore>
</CsoundSynthesizer>
;example by joachim heintz

 

Instead of just one loudspeaker configuration, as in the previous examples, there are now three configurations:

 

vbaplsinit 2.01, 7, -40, 40, 70, 140, 180, -110, -70
vbaplsinit 2.02, 2, -40, 40
vbaplsinit 2.03, 3, -70, 180, 70

The first parameter (the number of dimensions) now has an additional fractional part, with a range from .01 to .99, specifying the number of the speaker layout. So 2.01 means: two dimensions, layout number one, 2.02 is layout number two, and 2.03 is layout number three. The new vbap opcode has now these parameters:

 ar1[, ar2...] vbap asig, kazim [, kelev] [, kspread] [, ilayout]

The last parameter ilayout refers to the speaker layout number. In the example above, instrument 1 uses layout 1, instrument 2 uses layout 2, and instrument 3 uses layout 3. Even if you do not have more than two speakers you should see in Csound's output that instrument 1 goes to all seven speakers, instrument 2 only to the first two, and instrument 3 goes to speaker 3, 5, and 7.

In addition to the new vbap opcode, vbapg has been written. The idea is to have an opcode which returns the gains (amplitudes) of the speakers instead of the audio signal:

k1[, k2...] vbapg kazim [,kelev] [, kspread] [, ilayout]

Ambisonics

Ambisonics is another technique to distribute a virtual sound source in space.

There are excellent sources for the discussion of Ambisonics online3 and the following chapter will give a step by step introduction. We will focus just on the basic practicalities of using the Ambisonics opcodes of Csound, without going into too much detail of the concepts behind them. 

Ambisonics works using two basic steps. In the first step you encode the sound and the spatial information (its localisation) of a virtual sound source in a so-called B-format. In the second step you decode the B-format to match your loudspeaker setup.

It is possible to save the B-format as its own audio file, to preserve the spatial information or you can immediately do the decoding after the encoding thereby dealing directly only with audio signals instead of Ambisonic files. The next example takes the latter approach by implementing a transformation of the VBAP circle example to Ambisonics.

 

   EXAMPLE 05B08_Ambi_circle.csd

<CsoundSynthesizer>
<CsOptions>
-odac -d ;for the next line, change to your folder
--env:SSDIR+=/home/jh/Joachim/Csound/FLOSS/Release01/Csound_Floss_Release01/audio
</CsOptions>
<CsInstruments>
sr = 44100
ksmps = 32	
0dbfs = 1
nchnls = 8

  instr 1
Sfile      =          "ClassGuit.wav"
iFilLen    filelen    Sfile
p3         =          iFilLen
aSnd, a0   soundin    Sfile
kAzim      line       0, p3, 360 ;counterclockwise (!)
iSetup     =          4 ;octogon
aw, ax, ay, az bformenc1 aSnd, kAzim, 0
a1, a2, a3, a4, a5, a6, a7, a8 bformdec1 iSetup, aw, ax, ay, az
outch 1, a1, 2, a2, 3, a3, 4, a4, 5, a5, 6, a6, 7, a7, 8, a8
  endin
</CsInstruments>
<CsScore>
i 1 0 1
</CsScore>
</CsoundSynthesizer>
;example by joachim heintz

The first thing to note is that for a counterclockwise circle, the azimuth now has the line 0 -> 360, instead of 0 -> -360 as was used in the VBAP example. This is because Ambisonics usually reads the angle in a mathematical way: a positive angle is counterclockwise. Next, the encoding process is carried out in the line:

aw, ax, ay, az bformenc1 aSnd, kAzim, 0

Input arguments are the monophonic sound source aSnd, the xy-angle kAzim, and the elevation angle which is set to zero. Output signals are the spatial information in x-, y- and z- direction (ax, ay, az), and also an omnidirectional signal called aw

Decoding is performed by the line:

a1, a2, a3, a4, a5, a6, a7, a8 bformdec1 iSetup, aw, ax, ay, az

The inputs for the decoder are the same aw, ax, ay, az, which were the results of the encoding process, and an additional iSetup parameter. Currently the Csound decoder only works with some standard setups for the speaker: iSetup = 4 refers to an octogon.4 So the final eight audio signals a1, ..., a8 are being produced using this decoder, and are then sent to the speakers in the same way using the outch opcode.

Different Orders

What we have seen in this example is called 'first order' ambisonics. This means that the encoding process leads to the four basic dimensions w, x, y, z as described above.5 In "second order" ambisonics, there are additional "directions" called r, s, t, u, v. And in "third order" ambisonics again the additional k, l, m, n, o, p, q. The final example in this section shows the three orders, each of them in one instrument. If you have eight speakers in octophonic setup, you can compare the results.

 

   EXAMPLE 05B09_Ambi_orders.csd

<CsoundSynthesizer>
<CsOptions>
-odac -d ;for the next line, change to your folder
--env:SSDIR+=/home/jh/Joachim/Csound/FLOSS/Release01/Csound_Floss_Release01/audio
</CsOptions>
<CsInstruments>
sr = 44100
ksmps = 32	
0dbfs = 1
nchnls = 8

  instr 1 ;first order
aSnd, a0   soundin    "ClassGuit.wav"
kAzim      line       0, p3, 360
iSetup     =          4 ;octogon
aw, ax, ay, az bformenc1 aSnd, kAzim, 0
a1, a2, a3, a4, a5, a6, a7, a8 bformdec1 iSetup, aw, ax, ay, az
outch 1, a1, 2, a2, 3, a3, 4, a4, 5, a5, 6, a6, 7, a7, 8, a8
  endin

  instr 2 ;second order
aSnd, a0   soundin    "ClassGuit.wav"
kAzim      line       0, p3, 360
iSetup     =          4 ;octogon
aw, ax, ay, az, ar, as, at, au, av bformenc1 aSnd, kAzim, 0
a1, a2, a3, a4, a5, a6, a7, a8 bformdec1 iSetup, aw, ax, ay, az, ar, as, at, au, av
outch 1, a1, 2, a2, 3, a3, 4, a4, 5, a5, 6, a6, 7, a7, 8, a8
  endin

  instr 3 ;third order
aSnd, a0   soundin    "ClassGuit.wav"
kAzim      line       0, p3, 360
iSetup     =          4 ;octogon
aw, ax, ay, az, ar, as, at, au, av, ak, al, am, an, ao, ap, aq bformenc1 aSnd, kAzim, 0
a1, a2, a3, a4, a5, a6, a7, a8 bformdec1 iSetup, aw, ax, ay, az, ar, as, at, au, av, ak, al, am, an, ao, ap, aq
outch 1, a1, 2, a2, 3, a3, 4, a4, 5, a5, 6, a6, 7, a7, 8, a8
  endin
</CsInstruments>
<CsScore>
i 1 0 6
i 2 6 6
i 3 12 6
</CsScore>
</CsoundSynthesizer>
;example by joachim heintz

 

In theory, first-order ambisonics need at least 4 speakers to be projected correctly. Second-order ambisonics needs at least 6 speakers (9, if 3 dimensions are employed). Third-order ambisonics need at least 8 speakers (or 16 for 3d). So, although higher order should in general lead to a better result in space, you cannot expect it to work unless you have a sufficient number of speakers. Of course practice over theory may prove to be a better judge in many cases.

Ambisonics UDOs

Usage of the ambisonics UDOs

This chapter gives an overview of the UDOs explained below.

The channels of the B-format are stored in a zak space. Call zakinit only once and put it outside of any instrument definition in the orchestra file after the header. zacl clears the za space and is called after decoding. The B format of order n can be decoded in any order <= n. 

The text files "ambisonics_udos.txt", "ambisonics2D_udos.txt", "AEP_udos.txt" and "utilities.txt" must be located in the same folder as the csd files or included with full path.

These files can be downloaded together with the entire examples (some of them for CsoundQt) from here (as of September 2015).

 

zakinit isizea, isizek    (isizea = (order + 1)^2 in ambisonics (3D); isizea = 2·order + 1 in ambi2D; isizek = 1)

#include "ambisonics_udos.txt"	(order <= 8)
  	ambi_encode	asnd, iorder, kazimuth, kelevation (azimuth, elevation in degrees)
  	ambi_enc_dist asnd, iorder, kazimuth, kelevation, kdistance	
a1 [, a2] ... [, a8]	ambi_decode	iorder, ifn	
a1 [, a2] ... [, a8]	ambi_dec_inph	iorder, ifn	
f ifn  0  n  -2 p1 az1 el1 az2 el2 ... (n is a power of 2 greater than 3·number_of_spekers + 1) (p1 is not used)
  	ambi_write_B	"name", iorder, ifile_format	(ifile_format see fout in the csound help)	
  	ambi_read_B	"name", iorder (only <= 5)
kaz, kel, kdist	xyz_to_aed	kx, ky, kz

;#include "ambisonics2D_udos.txt"	
  	ambi2D_encode	asnd, iorder, kazimuth	(any order) (azimuth in degrees)
  	ambi2D_enc_dist	asnd, iorder, kazimuth, kdistance	
a1 [, a2] ... [, a8]	ambi2D_decode	iorder, iaz1 [, iaz2] ...	[, iaz8]	
a1 [, a2] ... [, a8]	ambi2D_dec_inph	iorder, iaz1 [, iaz2] ...	[, iaz8]	(order <= 12)
  	ambi2D_write_B	"name", iorder, ifile_format
  	ambi2D_read_B	"name", iorder	(order <= 19)
kaz, kdist	xy_to_ad	kx, ky	

#include "AEP_udos.txt"	(any order integer or fractional)
a1 [, a2] ... [, a16] AEP_xyz	asnd, korder, ifn, kx, ky, kz, kdistance
f ifn  0  64  -2  max_speaker_distance x1 y1 z1 x2 y2 z2 ...
a1 [, a2] ... [, a8] AEP	asnd, korder, ifn, kazimuth, kelevation, kdistance (azimuth, elevation in degrees)
f ifn  0  64  -2  max_speaker_distance az1 el1 dist1 az2 el2 dist2 ...  (azimuth, elevation in degrees)

;#include "ambi_utilities.txt"
kdist	dist	kx, ky
kdist	dist	kx, ky, kz
ares	Doppler asnd, kdistance
ares	absorb	asnd, kdistance
kx, ky, kz	aed_to_xyz	kazimuth, kelevation, kdistance
ix, iy, iz	aed_to_xyz	iazimuth, ielevation, idistance
a1 [, a2] ... [, a16]	dist_corr	a1 [, a2] ... [, a16], ifn
f ifn  0  32  -2  max_speaker_distance dist1, dist2, ... (distances in m)
irad	radiani	idegree	
krad	radian	kdegree
arad	radian	adegree
idegree	degreei	irad
kdegree	degree	krad
adegree	degree	arad

Introduction

In the following introduction we will explain the principles of ambisonics step by step and write an opcode for every step. The opcodes above combine all of the functionality described. Since the two-dimensional analogy to Ambisonics is easier to understand and to implement with a simple equipment, we shall fully explain it first.

Ambisonics is a technique of three-dimensional sound projection. The information about the recorded or synthesized sound field is encoded and stored in several channels, taking no account of the arrangement of the loudspeakers for reproduction. The encoding of a signal's spatial information can be more or less precise, depending on the so-called order of the algorithm used. Order zero corresponds to the monophonic signal and requires only one channel for storage and reproduction. In first-order Ambisonics, three further channels are used to encode the portions of the sound field in the three orthogonal directions x, y and z. These four channels constitute the so-called first-order B-format. When Ambisonics is used for artificial spatialisation of recorded or synthesized sound, the encoding can be of an arbitrarily high order. The higher orders cannot be interpreted as easily as orders zero and one. 

In a two-dimensional analogy to Ambisonics (called Ambisonics2D in what follows), only sound waves in the horizontal plane are encoded.

The loudspeaker feeds are obtained by decoding the B-format signal. The resulting panning is amplitude panning, and only the direction to the sound source is taken into account.

The illustration below shows the principle of Ambisonics. First a sound is generated and its position determined. The amplitude and spectrum are adjusted to simulate distance, the latter using a low-pass filter. Then the Ambisonic encoding is computed using the sound's coordinates. Encoding mth order B-format requires n = (m+1)^2 channels (n = 2m + 1 channels in Ambisonics2D). By decoding the B-format, one can obtain the signals for any number (>= n) of loudspeakers in any arrangement. Best results are achieved with symmetrical speaker arrangements. 

If the B-format does not need to be recorded the speaker signals can be calculated at low cost and arbitrary order using so-called ambisonics equivalent panning (AEP). 

 

Ambisonics2D 

Introduction

We will first explain the encoding process in Ambisonics2D. The position of a sound source in the horizontal plane is given by two coordinates. In Cartesian coordinates (x, y) the listener is at the origin of the coordinate system (0, 0), and the x-coordinate points to the front, the y-coordinate to the left. The position of the sound source can also be given in polar coordinates by the angle ψ between the line of vision of the listener (front) and the direction to the sound source, and by their distance r. Cartesian coordinates can be converted to polar coordinates by the formulae: 

  r =    and  ψ = arctan(x, y), 

polar to Cartesian coordinates by 

  x = r·cos(ψ) and y = r·sin(ψ).  

 

 

 

The 0th order B-Format of a signal S of a sound source on the unit circle is just the mono signal: W0 = W = S. The first order B-Format contains two additional channels: W1,1 = X = S·cos(ψ) = S·x and W1,2 = Y = S·sin(ψ) = S·y, i.e. the product of the Signal S with the sine and the cosine of the direction ψ of the sound source. The B-Format higher order contains two additional channels per order m: Wm, 1 = S·cos(mψ) and Wm, 2 = S·sin(mψ).

 

 W0 = S

 W1,1 = X = S·cos(ψ) = S·x W1,2 = Y = S·sin(ψ) = S·y

 W2,1 = S·cos(2ψ) W2,2 = S·sin(2ψ)

 ...

 Wm,1 = S·cos(mψ)    Wm,2 = S·sin(mψ) 

 

From the n = 2m + 1 B-Format channels the loudspeaker signals pi of n loudspeakers which are set up symmetrically on a circle (with angle ϕi) are:

   pi = 1/n(W0 + 2W1,1cos(ϕi) + 2W1,2sin(ϕi) + 2W2,1cos(2ϕi) + 2W2,2sin(2ϕi) + ...)

  = 2/n(1/2 W0 + W1,1cos(ϕi) + W1,2sin(ϕi) + W2,1cos(2ϕi) + W2,2sin(2ϕi) + ...)

(If more than n speakers are used, we can use the same formula)

In the Csound example udo_ambisonics2D_1.csd the opcode ambi2D_encode_1a produces the 3 channels W, X and Y (a0, a11, a12) from an input sound and the angle ψ (azmuth kaz), the opcode ambi2D_decode_1_8 decodes them to 8 speaker signals a1, a2, ..., a8. The inputs of the decoder are the 3 channels a0, a11, a12 and the 8 angles of the speakers. 

  EXAMPLE 05B10_udo_ambisonics2D_1.csd

<CsoundSynthesizer>
<CsInstruments>
sr      =  44100
ksmps   =  32
nchnls  =  8
0dbfs 	 = 1

; ambisonics2D first order without distance encoding
; decoding for 8 speakers symmetrically positioned on a circle

; produces the 3 channels 1st order; input: asound, kazimuth
opcode	ambi2D_encode_1a, aaa, ak	
asnd,kaz	xin
kaz = $M_PI*kaz/180
a0	=	asnd
a11	=	cos(kaz)*asnd
a12	=	sin(kaz)*asnd
		xout		a0,a11,a12
endop

; decodes 1st order to a setup of 8 speakers at angles i1, i2, ...
opcode	ambi2D_decode_1_8, aaaaaaaa, aaaiiiiiiii		
a0,a11,a12,i1,i2,i3,i4,i5,i6,i7,i8	xin
i1 = $M_PI*i1/180
i2 = $M_PI*i2/180
i3 = $M_PI*i3/180
i4 = $M_PI*i4/180
i5 = $M_PI*i5/180
i6 = $M_PI*i6/180
i7 = $M_PI*i7/180
i8 = $M_PI*i8/180
a1	=	(.5*a0 + cos(i1)*a11 + sin(i1)*a12)*2/3			
a2	=	(.5*a0 + cos(i2)*a11 + sin(i2)*a12)*2/3	
a3	=	(.5*a0 + cos(i3)*a11 + sin(i3)*a12)*2/3	
a4	=	(.5*a0 + cos(i4)*a11 + sin(i4)*a12)*2/3	
a5	=	(.5*a0 + cos(i5)*a11 + sin(i5)*a12)*2/3	
a6	=	(.5*a0 + cos(i6)*a11 + sin(i6)*a12)*2/3	
a7	=	(.5*a0 + cos(i7)*a11 + sin(i7)*a12)*2/3	
a8	=	(.5*a0 + cos(i8)*a11 + sin(i8)*a12)*2/3				
		xout			a1,a2,a3,a4,a5,a6,a7,a8
endop

instr 1
asnd	rand	.05
kaz   	line	0,p3,3*360 ;turns around 3 times in p3 seconds
a0,a11,a12 ambi2D_encode_1a asnd,kaz
a1,a2,a3,a4,a5,a6,a7,a8 \
        ambi2D_decode_1_8  a0,a11,a12,
                           0,45,90,135,180,225,270,315
        outc    a1,a2,a3,a4,a5,a6,a7,a8
endin

</CsInstruments>
<CsScore>
i1 0 40
</CsScore>
</CsoundSynthesizer>
;example by martin neukom

 

The B-format of all events of all instruments can be summed before decoding. Thus in the example udo_ambisonics2D_2.csd we create a zak space with 21 channels (zakinit 21, 1) for the 2D B-format up to 10th order where the encoded signals are accumulated. The opcode ambi2D_encode_3 shows how to produce the 7 B-format channels a0, a11, a12, ..., a32 for third order. The opcode ambi2D_encode_n produces the 2(n+1) channels a0, a11, a12, ..., a32 for any order n (needs zakinit 2(n+1), 1). The opcode ambi2D_decode_basic is an overloaded function i.e. it decodes to n speaker signals depending on the number of in- and outputs given (in this example only for 1 or 2 speakers). Any number of instruments can play arbitrary often. Instrument 10 decodes for the first 4 speakers of an 18 speaker setup. 

  EXAMPLE 05B11_udo_ambisonics2D_2.csd 

<CsoundSynthesizer>
<CsInstruments>

sr      =  44100
ksmps   =  32
nchnls  =  4
0dbfs 	 = 1

; ambisonics2D encoding fifth order
; decoding for 8 speakers symmetrically positioned on a circle
; all instruments write the B-format into a buffer (zak space)
; instr 10 decodes

; zak space with the 21 channels of the B-format up to 10th order
zakinit 21, 1	

;explicit encoding third order
opcode	ambi2D_encode_3, 0, ak	
asnd,kaz	xin	

kaz = $M_PI*kaz/180

		zawm		asnd,0
		zawm		cos(kaz)*asnd,1		;a11
		zawm		sin(kaz)*asnd,2		;a12
		zawm		cos(2*kaz)*asnd,3	;a21
		zawm		sin(2*kaz)*asnd,4	;a22
		zawm		cos(3*kaz)*asnd,5	;a31
		zawm		sin(3*kaz)*asnd,6	;a32
		
endop

; encoding arbitrary order n(zakinit 2*n+1, 1)
opcode	ambi2D_encode_n, 0, aik		
asnd,iorder,kaz	xin
kaz = $M_PI*kaz/180
kk =	iorder
c1:
   	zawm	cos(kk*kaz)*asnd,2*kk-1
   	zawm	sin(kk*kaz)*asnd,2*kk
kk =		kk-1

if	kk > 0 goto c1
	zawm	asnd,0	
endop

; basic decoding for arbitrary order n for 1 speaker
opcode	ambi2D_decode_basic, a, ii		
iorder,iaz	xin
iaz = $M_PI*iaz/180
igain	=	2/(2*iorder+1)
kk =	iorder
a1	=	.5*zar(0)
c1:
a1 +=	cos(kk*iaz)*zar(2*kk-1)
a1 +=	sin(kk*iaz)*zar(2*kk)
kk =		kk-1
if	kk > 0 goto c1
		xout			igain*a1
endop

; decoding for 2 speakers
opcode	ambi2D_decode_basic, aa, iii	
iorder,iaz1,iaz2	xin
iaz1 = $M_PI*iaz1/180
iaz2 = $M_PI*iaz2/180
igain	=	2/(2*iorder+1)
kk =	iorder
a1	=	.5*zar(0)
c1:
a1 +=	cos(kk*iaz1)*zar(2*kk-1)
a1 +=	sin(kk*iaz1)*zar(2*kk)
kk =		kk-1
if	kk > 0 goto c1

kk =	iorder
a2	=	.5*zar(0)
c2:
a2 +=	cos(kk*iaz2)*zar(2*kk-1)
a2 +=	sin(kk*iaz2)*zar(2*kk)
kk =		kk-1
if	kk > 0 goto c2
		xout			igain*a1,igain*a2
endop

instr 1
asnd	rand		p4
ares 	reson		asnd,p5,p6,1
kaz   	line		0,p3,p7*360		;turns around p7 times in p3 seconds
 		ambi2D_encode_n	asnd,10,kaz
endin

instr 2
asnd	oscil		p4,p5,1
kaz   	line		0,p3,p7*360		;turns around p7 times in p3 seconds
		ambi2D_encode_n	asnd,10,kaz
endin

instr 10	;decode all insruments (the first 4 speakers of a 18 speaker setup)
a1,a2		ambi2D_decode_basic 	10,0,20
a3,a4		ambi2D_decode_basic 	10,40,60
		outc	a1,a2,a3,a4			
		zacl 	0,20		; clear the za variables
endin


</CsInstruments>
<CsScore>
f1 0 32768 10 1
;			amp	 cf 	bw		turns
i1 0 3 	.7 	 1500 	12 		1
i1 2 18 	.1  2234 	34 		-8
;			amp		fr	0	turns
i2 0 3   .1	 	440	0	2
i10 0 3
</CsScore>
</CsoundSynthesizer>
;example by martin neukom

 

 

In-phase Decoding

The left figure below shows a symmetrical arrangement of 7 loudspeakers. If the virtual sound source is precisely in the direction of a loudspeaker, only this loudspeaker gets a signal (center figure). If the virtual sound source is between two loudspeakers, these loudspeakers receive the strongest signals; all other loudspeakers have weaker signals, some with negative amplitude, that is, reversed phase (right figure).

To avoid having loudspeaker sounds that are far away from the virtual sound source and to ensure that negative amplitudes (inverted phase) do not arise, the B-format channels can be weighted before being decoded. The weighting factors depend on the highest order used (M) and the order of the particular channel being decoded (m). 

 gm =  (M!)^2/((M + m)!·(M - m)!) 

 

The decoded signal can be normalised with the factor gnorm(M) = (2M + 1) !/(4^M (M!)^2)  

The illustration below shows a third-order B-format signal decoded to 13 loudspeakers first uncorrected (so-called basic decoding, left), then corrected by weighting (so-called in-phase decoding, right).

Example udo_ambisonics2D_3.csd shows in-phase decoding. The weights and norms up to 12th order are saved in the arrays iWeight2D[][] and iNorm2D[] respectively. Instrument 11 decodes third order for 4 speakers in a square.

  EXAMPLE 05B12_udo_ambisonics2D_3.csd 

<CsoundSynthesizer>
<CsInstruments>

sr      =  44100
ksmps   =  32
nchnls  =  4
0dbfs 	 = 1

opcode	ambi2D_encode_n, 0, aik		
asnd,iorder,kaz	xin
kaz = $M_PI*kaz/180
kk =	iorder
c1:
   	zawm	cos(kk*kaz)*asnd,2*kk-1
   	zawm	sin(kk*kaz)*asnd,2*kk
kk =		kk-1

if	kk > 0 goto c1
	zawm	asnd,0	

endop

;in-phase-decoding
opcode	ambi2D_dec_inph, a, ii	
; weights and norms up to 12th order
iNorm2D[] array 1,0.75,0.625,0.546875,0.492188,0.451172,0.418945,
					0.392761,0.370941,0.352394,0.336376,0.322360
iWeight2D[][] init   12,12
iWeight2D     array  0.5,0,0,0,0,0,0,0,0,0,0,0,
	0.666667,0.166667,0,0,0,0,0,0,0,0,0,0,
	0.75,0.3,0.05,0,0,0,0,0,0,0,0,0,
	0.8,0.4,0.114286,0.0142857,0,0,0,0,0,0,0,0,
	0.833333,0.47619,0.178571,0.0396825,0.00396825,0,0,0,0,0,0,0,
	0.857143,0.535714,0.238095,0.0714286,0.012987,0.00108225,0,0,0,0,0,0,
	0.875,0.583333,0.291667,0.1060601,0.0265152,0.00407925,0.000291375,0,0,0,0,0,
	0.888889,0.622222,0.339394,0.141414,0.043512,0.009324,0.0012432,
	0.0000777,0,0,0,0,
	0.9,0.654545,0.381818,0.176224,0.0629371,0.0167832,0.00314685,
	0.000370218,0.0000205677,0,0,0,
	0.909091,0.681818,0.41958,0.20979,0.0839161,0.0262238,0.0061703,
	0.00102838,0.000108251,0.00000541254,0,0,
	0.916667,0.705128,0.453297,0.241758,0.105769,0.0373303,0.0103695,
	0.00218306,0.000327459,0.0000311866,0.00000141757,0,
	0.923077,0.725275,0.483516,0.271978,0.12799,0.0497738,0.015718,
	0.00392951,0.000748478,0.000102065,0.00000887523,0.000000369801

iorder,iaz1	xin
iaz1 = $M_PI*iaz1/180
kk =	iorder
a1	=	.5*zar(0)
c1:
a1 +=	cos(kk*iaz1)*iWeight2D[iorder-1][kk-1]*zar(2*kk-1)
a1 +=	sin(kk*iaz1)*iWeight2D[iorder-1][kk-1]*zar(2*kk)
kk =		kk-1
if	kk > 0 goto c1
		xout			iNorm2D[iorder-1]*a1
endop

zakinit 7, 1		

instr 1
asnd	rand		p4
ares 	reson		asnd,p5,p6,1
kaz   	line		0,p3,p7*360		;turns around p7 times in p3 seconds
 		ambi2D_encode_n		asnd,3,kaz
endin

instr 11		

a1 		ambi2D_dec_inph 	3,0
a2 		ambi2D_dec_inph 	3,90
a3 		ambi2D_dec_inph 	3,180
a4 		ambi2D_dec_inph 	3,270
		outc	a1,a2,a3,a4
		zacl 	0,6		; clear the za variables
endin

</CsInstruments>
<CsScore>
;			amp	 cf 	bw		turns
i1 0 3 	.1 	 1500 	12 		1
i11 0 3
</CsScore>
</CsoundSynthesizer>
;example by martin neukom

 

Distance

In order to simulate distances and movements of sound sources, the signals have to be treated before being encoded. The main perceptual cues for the distance of a sound source are reduction of the amplitude, filtering due to the absorbtion of the air and the relation between direct and indirect sound. We will implement the first two of these cues. The amplitude arriving at a listener is inversely proportional to the distance of the sound source. If the distance is larger than the unit circle (not necessarily the radius of the speaker setup, which does not need to be known when encoding sounds) we can simply divide the sound by the distance. With this calculation inside the unit circle the amplitude is amplified and becomes infinite when the distance becomes zero. Another problem arises when a virtual sound source passes the origin. The amplitude of the speaker signal in the direction of the movement suddenly becomes maximal and the signal of the opposite speaker suddenly becomes zero. A simple solution for these problems is to limit the gain of the channel W inside the unit circle to 1 (f1 in the figure below) and to fade out all other channels (f2). By fading out all channels except channel W the information about the direction of the sound source is lost and all speaker signals are the same and the sum of the speaker signals reaches its maximum when the distance is 0. 

 

 

Now, we are looking for gain functions that are smoother at d = 1. The functions should be differentiable and the slope of f1 at distance d = 0 should be 0. For distances greater than 1 the functions should be approximately 1/d. In addition the function f1 should continuously grow with decreasing distance and reach its maximum at d = 0. The maximal gain must be 1. The function atan(d·π/2)/(d·π/2) fulfills these constraints. We create a function f2 for the fading out of the other channels by multiplying f1 by the factor (1 – E^(-d)).

 

In example udo_ambisonics2D_4 the UDO ambi2D_enc_dist_n encodes a sound at any order with distance correction. The inputs of the UDO are asnd, iorder, kazimuth and kdistance. If the distance becomes negative the azimuth angle is turned to its opposite (kaz += π) and the distance taken positive. 

EXAMPLE 05B13_udo_ambisonics2D_4.csd 

<CsoundSynthesizer>
<CsInstruments>

sr      =  44100
ksmps   =  32
nchnls  =  8
0dbfs 	 = 1

#include "ambisonics2D_udos.txt"

; distance encoding
; with any distance (includes zero and negative distance)

opcode	ambi2D_enc_dist_n, 0, aikk		
asnd,iorder,kaz,kdist	xin
kaz = $M_PI*kaz/180
kaz	=			(kdist < 0 ? kaz + $M_PI : kaz)
kdist =		abs(kdist)+0.0001
kgainW	=		taninv(kdist*1.5707963) / (kdist*1.5708)		;pi/2
kgainHO =	(1 - exp(-kdist))*kgainW
kk =	iorder
asndW	=	kgainW*asnd
asndHO	=	kgainHO*asndW
c1:
   	zawm	cos(kk*kaz)*asndHO,2*kk-1
   	zawm	sin(kk*kaz)*asndHO,2*kk
kk =		kk-1

if	kk > 0 goto c1
	zawm	asndW,0	
	
endop

zakinit 17, 1		

instr 1
asnd	rand		p4
;asnd	soundin	"/Users/user/csound/ambisonic/violine.aiff"
kaz   	line		0,p3,p5*360		;turns around p5 times in p3 seconds
kdist	line		p6,p3,p7			
        ambi2D_enc_dist_n asnd,8,kaz,kdist
endin

instr 10		
a1,a2,a3,a4,
a5,a6,a7,a8 		ambi2D_decode		8,0,45,90,135,180,225,270,315
		outc	a1,a2,a3,a4,a5,a6,a7,a8
		zacl 	0,16		
endin

</CsInstruments>
<CsScore>
f1 0 32768 10 1
;        amp turns dist1 dist2
i1 0 4   1   0     2     -2
;i1 0 4  1   1     1     1
i10 0 4
</CsScore>
</CsoundSynthesizer>
;example by martin neukom

 

In order to simulate the absorption of the air we introduce a very simple lowpass filter with a distance depending cutoff frequency. We produce a Doppler-shift with a distance dependent delay of the sound. Now, we have to determine our unit since the delay of the sound wave is calculated as distance divided by sound velocity. In our example udo_ambisonics2D_5.csd we set the unit to 1 metre. These procedures are performed before the encoding. In instrument 1 the movement of the sound source is defined in Cartesian coordinates. The UDO xy_to_ad transforms them into polar coordinates. The B-format channels can be written to a sound file with the opcode fout. The UDO write_ambi2D_2 writes the channels up to second order into a sound file. 

  EXAMPLE 05B14_udo_ambisonics2D_5.csd  

<CsoundSynthesizer>
<CsInstruments>
sr      =  44100
ksmps   =  32
nchnls  =  8
0dbfs 	 = 1

#include "ambisonics2D_udos.txt"
#include "ambisonics_utilities.txt" ;opcodes Absorb and Doppler

/* these opcodes are included in "ambisonics2D_udos.txt"
opcode xy_to_ad, kk, kk
kx,ky		xin
kdist =	sqrt(kx*kx+ky*ky)
kaz 		taninv2	ky,kx
			xout		180*kaz/$M_PI, kdist
endop

opcode Absorb, a, ak
asnd,kdist	xin
aabs 		tone 		5*asnd,20000*exp(-.1*kdist)	
			xout 		aabs
endop

opcode Doppler, a, ak
asnd,kdist	xin
abuf		delayr 	.5
adop		deltapi	interp(kdist)*0.0029137529 + .01 ; 1/343.2
			delayw 	asnd 	
			xout		adop
endop
*/
opcode	write_ambi2D_2, 0,	S		
Sname			xin
fout 	Sname,12,zar(0),zar(1),zar(2),zar(3),zar(4)
endop

zakinit 17, 1		; zak space with the 17 channels of the B-format

instr 1
asnd    buzz     p4,p5,50,1
;asnd   soundin  "/Users/user/csound/ambisonic/violine.aiff"
kx      line     p7,p3,p8		
ky      line     p9,p3,p10		
kaz,kdist xy_to_ad kx,ky
aabs    absorb   asnd,kdist
adop    Doppler  .2*aabs,kdist
        ambi2D_enc_dist adop,5,kaz,kdist
endin

instr 10		;decode all insruments
a1,a2,a3,a4,
a5,a6,a7,a8     ambi2D_dec_inph 5,0,45,90,135,180,225,270,315
                outc            a1,a2,a3,a4,a5,a6,a7,a8
;               fout "B_format2D.wav",12,zar(0),zar(1),zar(2),zar(3),zar(4),
;                                zar(5),zar(6),zar(7),zar(8),zar(9),zar(10)
                write_ambi2D_2  "ambi_ex5.wav"	
                zacl            0,16 ; clear the za variables
endin

</CsInstruments>
<CsScore>
f1 0 32768 10 1
;			amp	 	f 		0		x1	x2	y1	y2
i1 0 5 	.8  200 		0 		40	-20	1	.1
i10 0 5
</CsScore>
</CsoundSynthesizer>
;example by martin neukom

The position of a point in space can be given by its Cartesian coordinates x, y and z or by its spherical coordinates the radial distance r from the origin of the coordinate system, the elevation δ (which lies between –π and π) and the azimuth angle θ.

The formulae for transforming coordinates are as follows:

 

 

The channels of the Ambisonic B-format are computed as the product of the sounds themselves and the so-called spherical harmonics representing the direction to the virtual sound sources. The spherical harmonics can be normalised in various ways. We shall use the so-called semi-normalised spherical harmonics. The following table shows the encoding functions up to the third order as function of azimuth and elevation Ymn(θ,δ) and as function of x, y and z Ymn(x,y,z) for sound sources on the unit sphere. The decoding formulae for symmetrical speaker setups are the same.

 

 

In the first 3 of the following examples we will not produce sound but display in number boxes (for example using CsoundQt widgets) the amplitude of 3 speakers at positions (1, 0, 0), (0, 1, 0) and (0, 0, 1) in Cartesian coordinates. The position of the sound source can be changed with the two scroll numbers. The example udo_ambisonics_1.csd shows encoding up to second order. The decoding is done in two steps. First we decode the B-format for one speaker. In the second step, we create a overloaded opcode for n speakers. The number of output signals determines which version of the opcode is used. The opcodes ambi_encode and ambi_decode up to 8th order are saved in the text file "ambisonics_udos.txt".

  EXAMPLE 05B15_udo_ambisonics_1.csd  

<CsoundSynthesizer>
<CsInstruments>
sr      =  44100
ksmps   =  32
nchnls  =  1
0dbfs 	 = 1

zakinit 9, 1	; zak space with the 9 channel B-format second order

opcode	ambi_encode, 0, aikk		
asnd,iorder,kaz,kel	xin
kaz = $M_PI*kaz/180
kel = $M_PI*kel/180
kcos_el = cos(kel)
ksin_el = sin(kel)
kcos_az = cos(kaz)
ksin_az = sin(kaz)

	zawm	asnd,0							; W
	zawm	kcos_el*ksin_az*asnd,1		; Y	 = Y(1,-1)
	zawm	ksin_el*asnd,2 				; Z	 = Y(1,0)
	zawm	kcos_el*kcos_az*asnd,3		; X	 = Y(1,1)

	if		iorder < 2 goto	end

i2	= sqrt(3)/2
kcos_el_p2 = kcos_el*kcos_el
ksin_el_p2 = ksin_el*ksin_el
kcos_2az = cos(2*kaz)
ksin_2az = sin(2*kaz)
kcos_2el = cos(2*kel)
ksin_2el = sin(2*kel)

	zawm i2*kcos_el_p2*ksin_2az*asnd,4	; V = Y(2,-2)
	zawm i2*ksin_2el*ksin_az*asnd,5		; S = Y(2,-1)
	zawm .5*(3*ksin_el_p2 - 1)*asnd,6		; R = Y(2,0)
	zawm i2*ksin_2el*kcos_az*asnd,7		; S = Y(2,1)
	zawm i2*kcos_el_p2*kcos_2az*asnd,8	; U = Y(2,2)
end:

endop

; decoding of order iorder for 1 speaker at position iaz,iel,idist
opcode	ambi_decode1, a, iii		
iorder,iaz,iel	xin
iaz = $M_PI*iaz/180
iel = $M_PI*iel/180
a0=zar(0)
	if	iorder > 0 goto c0
aout = a0
	goto	end
c0:
a1=zar(1)
a2=zar(2)
a3=zar(3)
icos_el = cos(iel)
isin_el = sin(iel)
icos_az = cos(iaz)
isin_az = sin(iaz)
i1	=	icos_el*isin_az			; Y	 = Y(1,-1)
i2	=	isin_el					; Z	 = Y(1,0)
i3	=	icos_el*icos_az			; X	 = Y(1,1)
	if iorder > 1 goto c1
aout	=	(1/2)*(a0 + i1*a1 + i2*a2 + i3*a3)
	goto end
c1:
a4=zar(4)
a5=zar(5)
a6=zar(6)
a7=zar(7)
a8=zar(8)

ic2	= sqrt(3)/2

icos_el_p2 = icos_el*icos_el
isin_el_p2 = isin_el*isin_el
icos_2az = cos(2*iaz)
isin_2az = sin(2*iaz)
icos_2el = cos(2*iel)
isin_2el = sin(2*iel)


i4 = ic2*icos_el_p2*isin_2az	; V = Y(2,-2)
i5	= ic2*isin_2el*isin_az		; S = Y(2,-1)
i6 = .5*(3*isin_el_p2 - 1)		; R = Y(2,0)
i7 = ic2*isin_2el*icos_az		; S = Y(2,1)
i8 = ic2*icos_el_p2*icos_2az	; U = Y(2,2)
	
aout	=	(1/9)*(a0 + 3*i1*a1 + 3*i2*a2 + 3*i3*a3 + 5*i4*a4 + 5*i5*a5 + 5*i6*a6 + 5*i7*a7 + 5*i8*a8)

end:
		xout			aout
endop

; overloaded opcode for decoding of order iorder
; speaker positions in function table ifn
opcode	ambi_decode,	a,ii
iorder,ifn xin
		xout		ambi_decode1(iorder,table(1,ifn),table(2,ifn))
endop
opcode	ambi_decode,	aa,ii
iorder,ifn xin
		xout				ambi_decode1(iorder,table(1,ifn),table(2,ifn)),
		ambi_decode1(iorder,table(3,ifn),table(4,ifn))
endop
opcode	ambi_decode,	aaa,ii
iorder,ifn xin
		xout		ambi_decode1(iorder,table(1,ifn),table(2,ifn)),
		ambi_decode1(iorder,table(3,ifn),table(4,ifn)),
		ambi_decode1(iorder,table(5,ifn),table(6,ifn))
endop

instr 1
asnd	init		1
;kdist	init		1
kaz		invalue	"az"
kel		invalue	"el"

 	    ambi_encode asnd,2,kaz,kel

ao1,ao2,ao3 	ambi_decode	2,17
		outvalue "sp1", downsamp(ao1)
		outvalue "sp2", downsamp(ao2)	
		outvalue "sp3", downsamp(ao3)	
		zacl 	0,8
endin


</CsInstruments>
<CsScore>
;f1 0 1024 10 1
f17 0 64 -2 0  0 0   90 0   0 90   0 0  0 0  0 0
i1 0 100
</CsScore>
</CsoundSynthesizer>
;example by martin neukom

Example udo_ambisonics_2.csd shows in-phase decoding. The weights up to 8th order are stored in the arrays iWeight3D[][]. 

  EXAMPLE 05B16_udo_ambisonics_2.csd 

<CsoundSynthesizer>
<CsInstruments>
sr      =  44100
ksmps   =  32
nchnls  =  1
0dbfs 	 = 1

zakinit 81, 1 ; zak space for up to 81 channels of the 8th order B-format

; the opcodes used below are safed in "ambisonics_udos.txt"
#include "ambisonics_udos.txt"

; in-phase decoding up to third order for one speaker
opcode	ambi_dec1_inph3, a, iii		
; weights up to 8th order
iWeight3D[][] init   8,8
iWeight3D     array  0.333333,0,0,0,0,0,0,0,
	0.5,0.1,0,0,0,0,0,0,
	0.6,0.2,0.0285714,0,0,0,0,0,
	0.666667,0.285714,0.0714286,0.0079365,0,0,0,0,
	0.714286,0.357143,0.119048,0.0238095,0.0021645,0,0,0,
	0.75,0.416667,0.166667,0.0454545,0.00757576,0.00058275,0,0,
	0.777778,0.466667,0.212121,0.0707071,0.016317,0.002331,0.0001554,0,
  	0.8,0.509091,0.254545,0.0979021,0.027972,0.0055944,0.0006993,0.00004114

iorder,iaz,iel	xin
iaz = $M_PI*iaz/180
iel = $M_PI*iel/180
a0=zar(0)
	if	iorder > 0 goto c0
aout = a0
	goto	end
c0:
a1=iWeight3D[iorder-1][0]*zar(1)
a2=iWeight3D[iorder-1][0]*zar(2)
a3=iWeight3D[iorder-1][0]*zar(3)
icos_el = cos(iel)
isin_el = sin(iel)
icos_az = cos(iaz)
isin_az = sin(iaz)
i1	=	icos_el*isin_az			; Y	 = Y(1,-1)
i2	=	isin_el					; Z	 = Y(1,0)
i3	=	icos_el*icos_az			; X	 = Y(1,1)
	if iorder > 1 goto c1
aout	=	(3/4)*(a0 + i1*a1 + i2*a2 + i3*a3)
	goto end
c1:
a4=iWeight3D[iorder-1][1]*zar(4)
a5=iWeight3D[iorder-1][1]*zar(5)
a6=iWeight3D[iorder-1][1]*zar(6)
a7=iWeight3D[iorder-1][1]*zar(7)
a8=iWeight3D[iorder-1][1]*zar(8)

ic2	= sqrt(3)/2

icos_el_p2 = icos_el*icos_el
isin_el_p2 = isin_el*isin_el
icos_2az = cos(2*iaz)
isin_2az = sin(2*iaz)
icos_2el = cos(2*iel)
isin_2el = sin(2*iel)


i4 = ic2*icos_el_p2*isin_2az	; V = Y(2,-2)
i5	= ic2*isin_2el*isin_az		; S = Y(2,-1)
i6 = .5*(3*isin_el_p2 - 1)		; R = Y(2,0)
i7 = ic2*isin_2el*icos_az		; S = Y(2,1)
i8 = ic2*icos_el_p2*icos_2az	; U = Y(2,2)
aout	=	(1/3)*(a0 + 3*i1*a1 + 3*i2*a2 + 3*i3*a3 + 5*i4*a4 + 5*i5*a5 + 5*i6*a6 + 5*i7*a7 + 5*i8*a8)

end:
		xout			aout
endop

; overloaded opcode for decoding for 1 or 2 speakers
; speaker positions in function table ifn
opcode	ambi_dec2_inph,	a,ii
iorder,ifn xin
		xout		ambi_dec1_inph(iorder,table(1,ifn),table(2,ifn))
endop
opcode	ambi_dec2_inph,	aa,ii
iorder,ifn xin
		xout		ambi_dec1_inph(iorder,table(1,ifn),table(2,ifn)),
		ambi_dec1_inph(iorder,table(3,ifn),table(4,ifn))
endop
opcode	ambi_dec2_inph,	aaa,ii
iorder,ifn xin
		xout		ambi_dec1_inph(iorder,table(1,ifn),table(2,ifn)),
		ambi_dec1_inph(iorder,table(3,ifn),table(4,ifn)),
		ambi_dec1_inph(iorder,table(5,ifn),table(6,ifn))
endop

instr 1
asnd    init       1
kdist   init       1
kaz     invalue    "az"
kel     invalue    "el"

        ambi_encode asnd,8,kaz,kel
ao1,ao2,ao3 ambi_dec_inph 8,17
        outvalue   "sp1", downsamp(ao1)
        outvalue   "sp2", downsamp(ao2)
        outvalue   "sp3", downsamp(ao3)
        zacl       0,80
endin

</CsInstruments>
<CsScore>
f1 0 1024 10 1
f17 0 64 -2 0  0 0   90 0   0 90  0 0  0 0  0 0  0 0  0 0
i1 0 100
</CsScore>
</CsoundSynthesizer>
;example by martin neukom

 

The weighting factors for in-phase decoding of Ambisonics (3D) are:

Example udo_ambisonics_3.csd shows distance encoding. 

  EXAMPLE 05B17_udo_ambisonics_3.csd 

<CsoundSynthesizer>
<CsInstruments>
sr      =  44100
ksmps   =  32
nchnls  =  2
0dbfs 	 = 1

zakinit 81, 1		; zak space with the 11 channels of the B-format

#include "ambisonics_udos.txt"

opcode	ambi3D_enc_dist1, 0, aikkk		
asnd,iorder,kaz,kel,kdist	xin
kaz = $M_PI*kaz/180
kel = $M_PI*kel/180
kaz	=		(kdist < 0 ? kaz + $M_PI : kaz)
kel	=		(kdist < 0 ? -kel : kel)
kdist =	abs(kdist)+0.00001
kgainW	=	taninv(kdist*1.5708) / (kdist*1.5708)		
kgainHO =	(1 - exp(-kdist)) ;*kgainW
	outvalue "kgainHO", kgainHO
	outvalue "kgainW", kgainW
kcos_el = cos(kel)
ksin_el = sin(kel)
kcos_az = cos(kaz)
ksin_az = sin(kaz)
asnd =		kgainW*asnd
	zawm	asnd,0							; W
asnd = 	kgainHO*asnd
	zawm	kcos_el*ksin_az*asnd,1		; Y	 = Y(1,-1)
	zawm	ksin_el*asnd,2 				; Z	 = Y(1,0)
	zawm	kcos_el*kcos_az*asnd,3		; X	 = Y(1,1)
	if		iorder < 2 goto	end
/*
...
*/
end:

endop

instr 1
asnd    init      1
kaz     invalue "az"
kel     invalue "el"
kdist   invalue "dist"
        ambi_enc_dist asnd,5,kaz,kel,kdist
ao1,ao2,ao3,ao4 ambi_decode 5,17
        outvalue "sp1", downsamp(ao1)
        outvalue "sp2", downsamp(ao2)
        outvalue "sp3", downsamp(ao3)
        outvalue "sp4", downsamp(ao4)
        outc      0*ao1,0*ao2;,2*ao3,2*ao4
        zacl      0,80
endin
</CsInstruments>
<CsScore>
f17 0 64 -2 0  0 0  90 0   180 0 	 0 90  0 0	0 0
i1 0 100
</CsScore>
</CsoundSynthesizer>
;example by martin neukom

In example udo_ambisonics_4.csd a buzzer with the three-dimensional trajectory shown below is encoded in third order and decoded for a speaker setup in a cube (f17).

  EXAMPLE 05B18_udo_ambisonics_4.csd  

<CsoundSynthesizer>
<CsInstruments>
sr      =  44100
ksmps   =  32
nchnls  =  8
0dbfs 	 = 1

zakinit 16, 1	

#include "ambisonics_udos.txt"
#include "ambisonics_utilities.txt"

instr 1
asnd    buzz    p4,p5,p6,1
kt      line    0,p3,p3
kaz,kel,kdist xyz_to_aed 10*sin(kt),10*sin(.78*kt),10*sin(.43*kt)
adop Doppler asnd,kdist
        ambi_enc_dist adop,3,kaz,kel,kdist
a1,a2,a3,a4,a5,a6,a7,a8 ambi_decode 3,17
;k0		ambi_write_B	"B_form.wav",8,14
        outc    a1,a2,a3,a4,a5,a6,a7,a8
        zacl    0,15
endin

</CsInstruments>
<CsScore>
f1 0 32768 10 1
f17 0 64 -2 0 -45 35.2644  45 35.2644  135 35.2644  225 35.2644  -45 -35.2644  .7854 -35.2644  135 -35.2644  225 -35.2644
i1 0 40 .5 300 40
</CsScore>
</CsoundSynthesizer>
;example by martin neukom

Ambisonics Equivalent Panning (AEP)  

If we combine encoding and in-phase decoding, we obtain the following panning function (a gain function for a speaker depending on its distance to a virtual sound source):

  P(γ, m) = (1/2+ 1/2 cos γ)^m 

where γ denotes the angle between a sound source and a speaker and m denotes the order. If the speakers are positioned on a unit sphere the cosine of the angle γ is calculated as the scalar product of the vector to the sound source (x, y, z) and the vector to the speaker (xs, ys, zs). 

In contrast to Ambisonics the order indicated in the function does not have to be an integer. This means that the order can be continuously varied during decoding. The function can be used in both Ambisonics and Ambisonics2D.

This system of panning is called Ambisonics Equivalent Panning. It has the disadvantage of not producing a B-format representation, but its implementation is straightforward and the computation time is short and independent of the Ambisonics order simulated. Hence it is particularly useful for real-time applications, for panning in connection with sequencer programs and for experimentation with high and non-integral Ambisonic orders.

The opcode AEP1 in the example udo_AEP.csd shows the calculation of ambisonics equivalent panning for one speaker. The opcode AEP then uses AEP1 to produce the signals for several speakers. In the text file "AEP_udos.txt" AEP ist implemented for up to 16 speakers. The position of the speakers must be written in a function table. As the first parameter in the function table the maximal speaker distance must be given.

  EXAMPLE 05B19_udo_AEP.csd   

<CsoundSynthesizer>
<CsOptions>
</CsOptions>
<CsInstruments>
sr      =  44100
ksmps   =  32
nchnls  =  4
0dbfs 	 = 1

;#include "ambisonics_udos.txt"

; opcode AEP1 is the same as in udo_AEP_xyz.csd

opcode	AEP1, a, akiiiikkkkkk ; soundin, order, ixs, iys, izs, idsmax, kx, ky, kz
ain,korder,ixs,iys,izs,idsmax,kx,ky,kz,kdist,kfade,kgain	xin
idists =		sqrt(ixs*ixs+iys*iys+izs*izs)
kpan =			kgain*((1-kfade+kfade*(kx*ixs+ky*iys+kz*izs)/(kdist*idists))^korder)
		xout	ain*kpan*idists/idsmax
endop

; opcode AEP calculates ambisonics equivalent panning for n speaker
; the number n of output channels defines the number of speakers (overloaded function)
; inputs: sound ain, order korder (any real number >= 1)
; ifn = number of the function containing the speaker positions
; position and distance of the sound source kaz,kel,kdist in degrees

opcode AEP, aaaa, akikkk
ain,korder,ifn,kaz,kel,kdist	xin
kaz = $M_PI*kaz/180
kel = $M_PI*kel/180
kx = kdist*cos(kel)*cos(kaz)
ky = kdist*cos(kel)*sin(kaz)
kz = kdist*sin(kel)
ispeaker[] array 0,
  table(3,ifn)*cos(($M_PI/180)*table(2,ifn))*cos(($M_PI/180)*table(1,ifn)),
  table(3,ifn)*cos(($M_PI/180)*table(2,ifn))*sin(($M_PI/180)*table(1,ifn)),
  table(3,ifn)*sin(($M_PI/180)*table(2,ifn)),
  table(6,ifn)*cos(($M_PI/180)*table(5,ifn))*cos(($M_PI/180)*table(4,ifn)),
  table(6,ifn)*cos(($M_PI/180)*table(5,ifn))*sin(($M_PI/180)*table(4,ifn)),
  table(6,ifn)*sin(($M_PI/180)*table(5,ifn)),
  table(9,ifn)*cos(($M_PI/180)*table(8,ifn))*cos(($M_PI/180)*table(7,ifn)),
  table(9,ifn)*cos(($M_PI/180)*table(8,ifn))*sin(($M_PI/180)*table(7,ifn)),
  table(9,ifn)*sin(($M_PI/180)*table(8,ifn)),
  table(12,ifn)*cos(($M_PI/180)*table(11,ifn))*cos(($M_PI/180)*table(10,ifn)),
  table(12,ifn)*cos(($M_PI/180)*table(11,ifn))*sin(($M_PI/180)*table(10,ifn)),
  table(12,ifn)*sin(($M_PI/180)*table(11,ifn))

idsmax   table   0,ifn
kdist    =       kdist+0.000001
kfade    =       .5*(1 - exp(-abs(kdist)))
kgain    =       taninv(kdist*1.5708)/(kdist*1.5708)

a1       AEP1    ain,korder,ispeaker[1],ispeaker[2],ispeaker[3],
                   idsmax,kx,ky,kz,kdist,kfade,kgain
a2       AEP1    ain,korder,ispeaker[4],ispeaker[5],ispeaker[6],
                   idsmax,kx,ky,kz,kdist,kfade,kgain
a3       AEP1    ain,korder,ispeaker[7],ispeaker[8],ispeaker[9],
                   idsmax,kx,ky,kz,kdist,kfade,kgain
a4       AEP1    ain,korder,ispeaker[10],ispeaker[11],ispeaker[12],
                   idsmax,kx,ky,kz,kdist,kfade,kgain	
         xout    a1,a2,a3,a4
endop

instr 1
ain      rand    1
;ain		soundin	"/Users/user/csound/ambisonic/violine.aiff"
kt       line    0,p3,360
korder   init    24
;kdist 	Dist kx, ky, kz	
a1,a2,a3,a4 AEP  ain,korder,17,kt,0,1
         outc    a1,a2,a3,a4
endin

</CsInstruments>
<CsScore>

;fuction for speaker positions
; GEN -2, parameters: max_speaker_distance, xs1,ys1,zs1,xs2,ys2,zs2,...
;octahedron
;f17 0 32 -2 1 1 0 0  -1 0 0  0 1 0  0 -1 0  0 0 1  0 0 -1
;cube
;f17 0 32 -2 1,732 1 1 1  1 1 -1  1 -1 1  -1 1 1
;octagon
;f17 0 32 -2 1 0.924 -0.383 0 0.924 0.383 0 0.383 0.924 0 -0.383 0.924 0 -0.924 0.383 0 -0.924 -0.383 0 -0.383 -0.924 0 0.383 -0.924 0
;f17 0 32 -2 1  0 0 1  45 0 1  90 0 1  135 0 1  180 0 1  225 0 1  270 0 1  315 0 1
;f17 0 32 -2 1  0 -90 1  0 -70 1  0 -50 1  0 -30 1  0 -10 1  0 10 1  0 30 1  0 50 1
f17 0 32 -2 1   -45 0 1   45 0 1   135 0 1  225 0 1
i1 0 2

</CsScore>
</CsoundSynthesizer>
;example by martin neukom

 

 

Utilities 

The file utilities.txt contains the following opcodes:

dist computes the distance from the origin (0, 0) or (0, 0, 0) to a point (x, y) or (x, y, z)

kdist dist kx, ky

kdist dist kx, ky, kz

 

Doppler simulates the Doppler-shift

ares Doppler  asnd, kdistance  

 

absorb is a very simple simulation of the frequency dependent absorption

ares absorb asnd, kdistance

 

aed_to_xyz converts polar coordinates to Cartesian coordinates

kx, ky, kz aed_to_xyz kazimuth, kelevation, kdistance

ix, iy, iz aed_to_xyz iazimuth, ielevation, idistance

 

dist_corr induces a delay and reduction of the speaker signals relative to the most distant speaker.

a1 [, a2] ... [, a16] dist_corr a1 [, a2] ... [, a16], ifn

 f ifn  0  32  -2  max_speaker_distance dist1, dist2, ... ;distances in m

 

radian (radiani) converts degrees to radians.

irad radiani idegree 

krad radian kdegree

arad radian adegree

degree (degreei) converts radian to degrees

idegree degreei irad

kdegree degree krad

adegree degree arad 

VBAP or Ambisonics?

Csound offers a simple and reliable way to access two standard methods for multi-channel spatialisation. Both have different qualities and follow different aesthetics. VBAP can perhaps be described as clear, rational and direct. It combines simplicity with flexibility. It gives a reliable sound projection even for rather asymmetric speaker setups. Ambisonics on the other hand offers a very soft sound image, in which the single speaker becomes part of a coherent sound field. The B-format offers the possibility to store the spatial information independently from any particular speaker configuration. 

The composer, or spatial interpreter, can choose one or the other technique depending on the music and the context. Or (s)he can design a personal approach to spatialisation by combining the different techniques described in this chapter.

 

  1. First described by Ville Pulkki in 1997: Ville Pulkki, Virtual source positioning using vector base amplitude panning, in: Journal of the Audio Engeneering Society, 45(6), 456-466^
  2. Ville Pulkki, Uniform spreading of amplitude panned virtual sources, in: Proceedings of the 1999 IEEE Workshop on Applications of Signal Processing to Audio and Acoustics, Mohonk Montain House, New Paltz^
  3. For instance www.ambisonic.net or www.icst.net/research/projects/ambisonics-theory^
  4. See www.csounds.com/manual/html/bformdec1.html for more details.^
  5. Which in turn then are taken by the decoder as input.^

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