====== Simulating Analog Audio Circuits Part II ======

With minor modifications the [[start|ngSpice-sound]] libsndfile/libsamplerate wrapper source can just be compiled with other Software. 
Here's an experiment with [[sf>qUCS]]((QUCS: quite universal circuit simulator)).



===== Patch =====


You need to get the build-dependencies for QUCS which includes [[sf>mot-adms|admsXml'']]. You can start-off by patching the debian source which is easier to accomplish as it includes those as prebuild maintainer-files: ''sudo apt-get build-dep qucs'', ''apt-get source qucs'' and continue to patch in ''./qucs-0.0.11/qucs-core''
Note: the patch is just a quick hack and experiment: The one to be thrown away.
  * {{qucs-0.0.11-core-snd-v1.1.diff}}

  cd /tmp
  wget http://downloads.sourceforge.net/qucs/qucs-0.0.11.tar.gz
  wget http://gareus.org/_media/oss/spicesound/qucs-0.0.11-core-snd-v1.1.diff
  tar xzf qucs-0.0.11.tar.gz
  cd qucs-0.0.11/qucs-core
  patch -p1 < ../../qucs-0.0.11-core-snd-v1.1.diff
  cd ..
  ./autogen.sh
  #CXXFLAGS="-Wall -g -O3" CFLAGS="-g -O2 -Wall" ./configure '--host=i486-linux-gnu' \
   '--build=i486-linux-gnu' '--prefix=/usr' '--mandir=/usr/share/man' \
   'build_alias=i486-linux-gnu' 'host_alias=i486-linux-gnu' && make
  ./configure && make
  sudo make install

===== Syntax =====

The Audio-Source is represented by a //voltage source// in the circuit and the Audio-Sink is implemented as //transient analysis// command.

**Audio Source Syntax:**
  Vsnd:V1 _net0 gnd U="1 V" TimeOffset="0.0" Samplerate="48000" Channel="0" Oversampling="64" File="test.wav"


''Vsnd'' is a //spin-off// from ''Vac'' and has similar syntax in the netlist. The parameters are:

  * **U** (default 1V) - Voltage multiplier when converting the audio-floating point value to Volts
  * **UOffset** (default 0.0V) - Voltage offset
  * **TimeOffset** (default 0s) - offset in seconds added to the simulation-time when seeking the source-file.
  * **Samplerate** (default 48000) - Samplerate to use when converting time to samples
  * **Oversampling** (default 64) - upsampling factor
  * **Channel** (default 0) - audio-channel of the source file to use
  * **File** (default "in.wav") - audio file to use

Note: 
  * qUCS syntax is very restrictive and does only allow the following characters ''[A-Z][a-z][0-9]_.'' as options. This limits the //path// to input/output files to the current working directory!
  * Only the qUCS parser has been patched to allow audio-IO; qUCS can import and parse other netlist formats but audio-IO is not available for them. 


**Audio Sink:**

the ''TR'' transient analysis command has been re-used. ''TSOUND'' is equivalent to ''TR'' but has a few extra options and writes an audio-file instead of XML.

  .TSOUND:TR1 Type="lin" Start="100ms" Stop="200ms" Points="76802" IntegrationMethod="Trapezoidal" Order="2" InitialStep="1 ns" MinStep="1e-8" MaxIter="150" reltol="0.001" abstol="1 pA" vntol="1 uV" Temp="26.85" LTEreltol="1e-3" LTEabstol="1e-6" LTEfactor="1" Solver="CroutLU" relaxTSR="no" initialDC="yes" MaxStep="0" Oversampling="16" File="test_out2.wav"

Parameters:

  * **File** (default ''out.wav'') - Output-File to write 
  * **Samplerate** (default 48000) - Samplerate of the file to write
  * **Umult** (default 1.0 [1/V]) - multiplier when converting Volts into audio samples
  * **Uoffset** (default 0.0 V) - zero Volts offset
  * **Format** (default "wav24") - Audio format to write. numeric of text shortcut
    * -1:default (wav 24 bit/sample little endian)
    * 0: aliki .ald data file
    * >0 : //libsndfile// [[http://www.mega-nerd.com/libsndfile/api.html|numeric format]] SF_FORMAT_XXX - example: (SF_FORMAT_WAV | SF_FORMAT_PCM_24) = 0x010000 | 0x003 = 65539
    * currently the following txt shortcuts are defined: ''wav16'', ''wav24'', ''wav32'', ''aiff'' and ''aliki''. =
  * **Oversampling** (default 64) - downsample output by this factor


Note: set ''TSOUND''s ''Points'' parameter to ''((Stop-Start)*Samplerate*Oversampling)+2'' 



===== Examples =====

  # Qucs 0.0.11 
  .TSOUND:TR1 Type="lin" Start="100ms" Stop="200ms" Points="76802" IntegrationMethod="Trapezoidal" Order="2" InitialStep="1 ns" MinStep="1e-8" MaxIter="150" reltol="0.001" abstol="1 pA" vntol="1 uV" Temp="26.85" LTEreltol="1e-3" LTEabstol="1e-6" LTEfactor="1" Solver="CroutLU" relaxTSR="no" initialDC="yes" MaxStep="0" Oversampling="16" File="sw4.wav"
  #.TR:TR1 Type="lin" Start="100ms" Stop="200ms" Points="76802" IntegrationMethod="Trapezoidal" Order="2" InitialStep="1 ns" MinStep="1e-8" MaxIter="150" reltol="0.001" abstol="1 pA" vntol="1 uV" Temp="26.85" LTEreltol="1e-3" LTEabstol="1e-6" LTEfactor="1" Solver="CroutLU" relaxTSR="no" initialDC="yes" MaxStep="0"
  #.AC:AC1 Type="lin" Start="20Hz" Stop="20 kHz" Points="100" Noise="no"
  Vsnd:V1 _net0 gnd U="1 V" TimeOffset="0.0" Samplerate="48000" Channel="0" Oversampling="64" File="slin5s48k_F.wav"
  R:R3 _net0 _net1 R="200 Ohm" Temp="26.85" Tc1="0.0" Tc2="0.0" Tnom="26.85"
  C:C2 gnd _net1 C="1 uF" V=""
  C:C1 _net1 sound C=".1 uF" V=""
  R:R4 gnd sound R="10 kOhm" Temp="26.85" Tc1="0.0" Tc2="0.0" Tnom="26.85"

Save the stanza above as ''bandpass.txt'' copy an audio-file to ''slin5s48k_F.wav'' in the same folder. Executing ''qucsator -i bandpass.txt'' will write ''sw4.wav'': a 3 channel audio-file corresponding to the voltages on ''_net0'' ''_net1'' and ''sound''

====== References ======