By default, the synthesizer outputs a single stereo signal. Using this option, the synthesizer can output multichannel audio. Sets the number of stereo channel pairs. So 1 is actually 2 channels (a stereo pair).</desc>
When set to 1 (TRUE) the chorus effects module is activated. Otherwise, no chorus will be added to the output signal. Note that the amount of signal sent to the chorus module depends on the "chorus send" generator defined in the SoundFont.</desc>
</setting>
<setting>
<name>cpu-cores</name>
<type>int</type>
<def>1</def>
<min>1</min>
<max>256</max>
<desc>
(Experimental) Sets the number of synthesis CPU cores. If set to a value greater than 1, then additional synthesis threads will be created to take advantage of a multi CPU or CPU core system. This has the affect of utilizing more of the total CPU for voices or decreasing render times when synthesizing audio to a file.</desc>
</setting>
<setting>
<name>device-id</name>
<type>int</type>
<def>0</def>
<min>0</min>
<max>126</max>
<desc>
Device identifier used for SYSEX commands, such as MIDI Tuning Standard commands. Only those SYSEX commands destined for this ID or to all devices will be acted upon.</desc>
When set to "yes" the LADSPA subsystem will be enabled. This subsystem allows to load and interconnect LADSPA plug-ins. The output of the synthesizer is processed by the LADSPA subsystem. Note that the synthesizer has to be compiled with LADSPA support. More information about the LADSPA subsystem later.</desc>
</setting>
<setting>
<name>midi-channels</name>
<type>int</type>
<def>16</def>
<min>16</min>
<max>256</max>
<desc>
This setting defines the number of MIDI channels of the synthesizer. The MIDI standard defines 16 channels, so MIDI hardware is limited to this number. Internally FluidSynth can use more channels which can be mapped to different MIDI sources.</desc>
Sets the minimum note duration in milliseconds. This ensures that really short duration note events, such as percussion notes, have a better chance of sounding as intended. Set to 0 to disable this feature.</desc>
This is the low-latency setting. If on, you're allowed to call fluid_synth_write_s16, fluid_synth_write_float,
fluid_synth_nwrite_float or fluid_synth_process in parallel with the
rest of the calls, and it won't be blocked by time intensive calls to
the synth. Turn it off if throughput is more important than latency, e g
in rendering-to-file scenarios where underruns is not an issue.
<br/><br/><strong>Deprecated:</strong><br/>As of 1.1.7 this option is deprecated. This option enforces thread safety for rvoice_mixer, which causes rvoice_events to be queued internally. The current implementation relies on the fact that this option is set to TRUE to correctly render any amount of requested audio. Also calling fluid_synth_write_* in parallel is not considered to be a use-case. It would cause undefined audio output, as it would be unpredictable for the user which rvoice_events specifically would be dispatched to which fluid_synth_write_* call.
The polyphony defines how many voices can be played in parallel. A note event produces one or more voices. Its good to set this to a value which the system can handle and will thus limit FluidSynth's CPU usage. When FluidSynth runs out of voices it will begin terminating lower priority voices for new note events.</desc>
</setting>
<setting>
<name>reverb.active</name>
<type>bool</type>
<def>1 (TRUE)</def>
<desc>
When set to 1 (TRUE) the reverb effects module is activated. Otherwise, no reverb will be added to the output signal. Note that the amount of signal sent to the reverb module depends on the "reverb send" generator defined in the SoundFont.
</desc>
</setting>
<setting>
<name>sample-rate</name>
<type>num</type>
<def>44100.0</def>
<min>22050.0</min>
<max>96000.0</max>
<desc>
The sample rate of the audio generated by the synthesizer.
</desc>
</setting>
<setting>
<name>threadsafe-api</name>
<type>bool</type>
<def>1 (TRUE)</def>
<desc>
Controls whether the synth's public API is protected by a mutex or not. Default is on, turn it off for slightly better performance if you know you're only accessing the synth from one thread only, this could be the case in many embedded use cases for example. Note that libfluidsynth can use many threads by itself (shell is one, midi driver is one, midi player is one etc) so you should usually leave it on. Also see synth.parallel-render.
</desc>
</setting>
<setting>
<name>verbose</name>
<type>bool</type>
<def>0 (FALSE)</def>
<desc>
When set to 1 (TRUE) the synthesizer will print out information about the received MIDI events to the stdout. This can be helpful for debugging. This setting cannot be changed after the synthesizer has started.
<desc>The gain is applied to the final or master output of the synthesizer. It is set to a low value by default to avoid the saturation of the output when many notes are played.</desc>
The number of the audio buffers used by the driver. This number of buffers, multiplied by the buffer size (see setting audio.period-size), determines the maximum latency of the audio driver.
</desc>
</setting>
<setting>
<name>period-size</name>
<type>int</type>
<def>64 (Linux, Mac OS X),<br/>
512 (Windows)
</def>
<min>64</min>
<max>8192</max>
<desc>
The size of the audio buffers (in frames).
</desc>
</setting>
<setting>
<name>realtime-prio</name>
<type>int</type>
<def>60</def>
<min>0</min>
<max>99</max>
<desc>
Sets the realtime scheduling priority of the audio synthesis thread (0 disables high priority scheduling). Linux is the only platform which currently makes use of different priority levels. Drivers which use this option: alsa, oss and pulseaudio
</desc>
</setting>
<setting>
<name>sample-format</name>
<type>str</type>
<def>16bits</def>
<vals>16bits, float</vals>
<desc>
The format of the audio samples. This is currently only an indication; the audio driver may ignore this setting if it can't handle the specified format.
<vals>ALSA device string, such as: "hw:0", "plughw:1", etc.</vals>
<desc>
Selects the ALSA audio device to use.
</desc>
</setting>
</audio.alsa>
<audio.coreaudio>
<setting>
<name>device</name>
<type>str</type>
<def>default</def>
<desc>
Selects the CoreAudio device to use.
</desc>
</setting>
</audio.coreaudio>
<audio.dart>
<setting>
<name>device</name>
<type>str</type>
<def>default</def>
<desc>
Selects the Dart (OS/2 driver) device to use.
</desc>
</setting>
</audio.dart>
<audio.dsound>
<setting>
<name>device</name>
<type>str</type>
<def>default</def>
<desc>
Selects the DirectSound (Windows) device to use.
</desc>
</setting>
</audio.dsound>
<audio.file>
<setting>
<name>endian</name>
<type>str</type>
<def>'auto' if libsndfile support is built in,<br/>
'cpu' otherwise.</def>
<vals>auto, big, cpu, little ('cpu' is all that is supported if libsndfile support is not built in)</vals>
<desc>
Defines the byte order when using the 'file' driver or file renderer to store audio to a file. 'auto' uses the default for the given file type, 'cpu' uses the CPU byte order, 'big' uses big endian byte order and 'little' uses little endian byte order.
</desc>
</setting>
<setting>
<name>format</name>
<type>str</type>
<def>s16</def>
<vals>double, float, s16, s24, s32, s8, u8 ('s16' is all that is supported if libsndfile support not built in)</vals>
<desc>
Defines the audio format when rendering audio to a file.
<ul>
<li>'double' is 64 bit floating point,</li>
<li>'float' is 32 bit floating point,</li>
<li>'s16' is 16 bit signed PCM,</li>
<li>'s24' is 24 bit signed PCM,</li>
<li>'s32' is 32 bit signed PCM,</li>
<li>'s8' is 8 bit signed PCM and</li>
<li>'u8' is 8 bit unsigned PCM.</li>
</ul>
</desc>
</setting>
<setting>
<name>name</name>
<type>str</type>
<def>'fluidsynth.wav' if libsndfile support is built in,<br/>
'fluidsynth.raw' otherwise.</def>
<desc>
Specifies the file name to store the audio to, when rendering audio to a file.
</desc>
</setting>
<setting>
<name>type</name>
<type>str</type>
<def>'auto' if libsndfile support is built in,<br/>
'raw' otherwise.</def>
<vals>aiff, au, auto, avr, caf, flac, htk, iff, mat, oga, paf, pvf, raw, sd2, sds, sf, voc, w64, wav, xi (actual list of types may vary and depends on the libsndfile library used, 'raw' is the only type available if no libsndfile support is built in).</vals>
<desc>
Sets the file type of the file which the audio will be stored to. 'auto' attempts to determine the file type from the audio.file.name file extension and falls back to 'wav' if the extension doesn't match any types.
</desc>
</setting>
</audio.file>
<audio.jack>
<setting>
<name>autoconnect</name>
<type>bool</type>
<def>0 (FALSE)</def>
<desc>
If 1 (TRUE), then FluidSynth output is automatically connected to jack system audio output.
</desc>
</setting>
<setting>
<name>id</name>
<type>str</type>
<def>fluidsynth</def>
<desc>
ID used when creating Jack client connection.
</desc>
</setting>
<setting>
<name>multi</name>
<type>bool</type>
<def>0 (FALSE)</def>
<desc>
If 1 (TRUE), then multi-channel Jack output will be enabled if synth.audio-channels is greater than 1.
</desc>
</setting>
<setting>
<name>server</name>
<type>str</type>
<def></def>
<desc>
Jack server to connect to. Defaults to an empty string, which uses default Jack server.
</desc>
</setting>
</audio.jack>
<audio.oss>
<setting>
<name>device</name>
<type>str</type>
<def>/dev/dsp</def>
<desc>
Device to use for OSS audio output.
</desc>
</setting>
</audio.oss>
<audio.portaudio>
<setting>
<name>device</name>
<type>str</type>
<def>PortAudio Default</def>
<desc>
Device to use for PortAudio driver output. Note that 'PortAudio Default' is a special value which outputs to the default PortAudio device.
<desc>Sets the realtime scheduling priority of the MIDI thread (0 disables high priority scheduling). Linux is the only platform which currently makes use of different priority levels. Drivers which use this option: alsa_raw, alsa_seq, oss</desc>
</setting>
<setting>
<name>portname</name>
<type>str</type>
<def></def>
<desc>Used by coremidi and alsa_seq drivers for the portnames registered with the MIDI subsystem.</desc>
<desc>If true, reset the synth before starting a new MIDI song, so the state of a previous song can't affect the new song. Turn it off for seamless looping of a song.</desc>
</setting>
<setting>
<name>timing-source</name>
<type>str</type>
<def>sample</def>
<vals>sample, system</vals>
<desc>Determines the timing source of the player sequencer. 'sample' uses the sample clock (how much audio has been output) to sequence events, in which case audio is synchronized with MIDI events. 'system' uses the system clock, audio and MIDI are not synchronized exactly.</desc>
<desc>In dump mode we set the prompt to "". The ui cannot easily handle lines, which don't end with cr. Changing the prompt cannot be done through a command, because the current shell does not handle empty arguments.</desc>
</setting>
<setting>
<name>port</name>
<type>num</type>
<def>9800</def>
<min>1</min>
<max>65535</max>
<desc>The shell can be used in a client/server mode. This setting controls what TCP/IP port the server uses.</desc>