var searchIndex = {};
searchIndex["rfmod"] = {"doc":"rust-fmod\n=========","items":[[3,"Channel","rfmod","Channel Object",null,null],[3,"SpeakerMixOptions","","Structure which contains data for [`Channel::set_speaker_mix`](struct.Channel.html#method.set_speaker_mix) and [`Channel::get_speaker_mix`](struct.Channel.html#method.get_speaker_mix)",null,null],[12,"front_left","","",0,null],[12,"front_right","","",0,null],[12,"center","","",0,null],[12,"lfe","","",0,null],[12,"back_left","","",0,null],[12,"back_right","","",0,null],[12,"side_left","","",0,null],[12,"side_right","","",0,null],[3,"ReverbChannelProperties","","Structure defining the properties for a reverb source, related to a FMOD channel.",null,null],[12,"direct","","[r/w] MIN: -10000 MAX: 1000 DEFAULT: 0 - Direct path level",1,null],[12,"room","","[r/w] MIN: -10000 MAX: 1000 DEFAULT: 0 - Room effect level",1,null],[12,"flags","","[r/w] FMOD_REVERB_CHANNELFLAGS         - modifies the behavior of properties",1,null],[12,"connection_point","","[r/w] See remarks.                    - DSP network location to connect reverb for this channel.",1,null],[3,"Sys","","FMOD System Object",null,null],[3,"Guid","","Structure describing a globally unique identifier.",null,null],[12,"data1","","Specifies the first 8 hexadecimal digits of the GUID",2,null],[12,"data2","","Specifies the first group of 4 hexadecimal digits.",2,null],[12,"data3","","Specifies the second group of 4 hexadecimal digits.",2,null],[12,"data4","","Array of 8 bytes. The first 2 bytes contain the third group of 4 hexadecimal digits. The remaining 6 bytes contain the final\n12 hexadecimal digits.",2,null],[3,"SoftwareFormat","","Wrapper for arguments of [`Sys::set_software_format`](struct.Sys.html#method.set_software_format) and\n[`Sys::get_software_format`](struct.Sys.html#method.get_software_format)",null,null],[12,"sample_rate","","",3,null],[12,"format","","",3,null],[12,"num_output_channels","","",3,null],[12,"max_input_channels","","",3,null],[12,"resample_method","","",3,null],[12,"bits","","",3,null],[3,"AdvancedSettings","","Settings for advanced features like configuring memory and cpu usage for the FMOD_CREATECOMPRESSEDSAMPLE feature.",null,null],[12,"max_MPEG_codecs","","[r/w] Optional. Specify 0 to ignore. For use with FMOD_CREATECOMPRESSEDSAMPLE only. Mpeg codecs consume 21,684 bytes per\ninstance and this number will determine how many mpeg channels can be played simultaneously. Default = 32.",4,null],[12,"max_ADPCM_codecs","","[r/w] Optional. Specify 0 to ignore. For use with FMOD_CREATECOMPRESSEDSAMPLE only. ADPCM codecs consume 2,136 bytes per\ninstance and this number will determine how many ADPCM channels can be played simultaneously. Default = 32.",4,null],[12,"max_XMA_codecs","","[r/w] Optional. Specify 0 to ignore. For use with FMOD_CREATECOMPRESSEDSAMPLE only. XMA codecs consume 14,836 bytes per instance\nand this number will determine how many XMA channels can be played simultaneously. Default = 32.",4,null],[12,"max_CELT_codecs","","[r/w] Optional. Specify 0 to ignore. For use with FMOD_CREATECOMPRESSEDSAMPLE only. CELT codecs consume 11,500 bytes per\ninstance and this number will determine how many CELT channels can be played simultaneously. Default = 32.",4,null],[12,"max_VORBIS_codecs","","[r/w] Optional. Specify 0 to ignore. For use with FMOD_CREATECOMPRESSEDSAMPLE only. Vorbis codecs consume 12,000 bytes per\ninstance and this number will determine how many Vorbis channels can be played simultaneously. Default = 32.",4,null],[12,"max_AT9_codecs","","[r/w] Optional. Specify 0 to ignore. For use with FMOD_CREATECOMPRESSEDSAMPLE only. AT9 codecs consume 8,720 bytes per instance\nand this number will determine how many AT9 channels can be played simultaneously. Default = 32.",4,null],[12,"max_PCM_codecs","","[r/w] Optional. Specify 0 to ignore. For use with PS3 only. PCM codecs consume 12,672 bytes per instance and this number will\ndetermine how many streams and PCM voices can be played simultaneously. Default = 16.",4,null],[12,"ASIO_num_channels","","[r/w] Optional. Specify 0 to ignore. Number of channels available on the ASIO device.",4,null],[12,"ASIO_channel_list","","[r/w] Optional. Specify 0 to ignore. Pointer to an array of strings (number of entries defined by ASIONumChannels) with ASIO\nchannel names.",4,null],[12,"ASIO_speaker_list","","[r/w] Optional. Specify 0 to ignore. Pointer to a list of speakers that the ASIO channels map to. This can be called after\n[`Sys::init`](doc/rfmod/struct.Sys.html#method.init) to remap ASIO output.",4,null],[12,"max_3D_reverb_DSPs","","[r/w] Optional. Specify 0 to ignore. The max number of 3d reverb DSP's in the system. (NOTE: CURRENTLY DISABLED / UNUSED)",4,null],[12,"HRTF_min_angle","","[r/w] Optional. For use with FMOD_INIT_HRTF_LOWPASS. The angle range (0-360) of a 3D sound in relation to the listener, at\nwhich the HRTF function begins to have an effect. 0 = in front of the listener. 180 = from 90 degrees to the left of the\nlistener to 90 degrees to the right. 360 = behind the listener. Default = 180.0.",4,null],[12,"HRTF_max_angle","","[r/w] Optional. For use with FMOD_INIT_HRTF_LOWPASS. The angle range (0-360) of a 3D sound in relation to the listener, at\nwhich the HRTF function has maximum effect. 0 = front of the listener. 180 = from 90 degrees to the left of the listener to\n90 degrees to the right. 360 = behind the listener. Default = 360.0.",4,null],[12,"HRTF_freq","","[r/w] Optional. Specify 0 to ignore. For use with FMOD_INIT_HRTF_LOWPASS. The cutoff frequency of the HRTF's lowpass filter\nfunction when at maximum effect. (i.e. at HRTFMaxAngle). Default = 4000.0.",4,null],[12,"vol0_virtual_vol","","[r/w] Optional. Specify 0 to ignore. For use with FMOD_INIT_VOL0_BECOMES_VIRTUAL. If this flag is used, and the volume is 0.0,\nthen the sound will become virtual. Use this value to raise the threshold to a different point where a sound goes virtual.",4,null],[12,"event_queue_size","","[r/w] Optional. Specify 0 to ignore. For use with FMOD Event system only. Specifies the number of slots available for\nsimultaneous non blocking loads, across all threads. Default = 32.",4,null],[12,"default_decode_buffer_size","","[r/w] Optional. Specify 0 to ignore. For streams. This determines the default size of the double buffer (in milliseconds) that\na stream uses. Default = 400ms",4,null],[12,"debug_log_filename","","[r/w] Optional. Specify 0 to ignore. Gives fmod's logging system a path/filename. Normally the log is placed in the same\ndirectory as the executable and called fmod.log. When using\n[`Sys::get_advanced_settings`](doc/rfmod/struct.Sys.html#method.get_advanced_settings), provide at least 256 bytes of\nmemory to copy into.",4,null],[12,"profile_port","","[r/w] Optional. Specify 0 to ignore. For use with FMOD_INIT_ENABLE_PROFILE. Specify the port to listen on for connections by the\nprofiler application.",4,null],[12,"geometry_max_fade_time","","[r/w] Optional. Specify 0 to ignore. The maximum time in miliseconds it takes for a channel to fade to the new level when its\nocclusion changes.",4,null],[12,"max_spectrum_wave_data_buffers","","[r/w] Optional. Specify 0 to ignore. Tells [`Sys::init`](doc/rfmod/struct.Sys.html#method.init) to allocate a pool of\nwavedata/spectrum buffers to prevent memory fragmentation, any additional buffers will be allocated normally.",4,null],[12,"music_system_cache_delay","","[r/w] Optional. Specify 0 to ignore. The delay the music system should allow for loading a sample from disk (in\nmilliseconds). Default = 400 ms.",4,null],[12,"distance_filter_center_freq","","[r/w] Optional. Specify 0 to ignore. For use with FMOD_INIT_DISTANCE_FILTERING. The default center frequency in Hz for the\ndistance filtering effect. Default = 1500.0.",4,null],[12,"stack_size_stream","","[r/w] Optional. Specify 0 to ignore. Specify the stack size for the FMOD Stream thread in bytes. Useful for custom codecs that\nuse excess stack. Default 49,152 (48kb)",4,null],[12,"stack_size_non_blocking","","[r/w] Optional. Specify 0 to ignore. Specify the stack size for the FMOD_NONBLOCKING loading thread. Useful for custom codecs\nthat use excess stack. Default 65,536 (64kb)",4,null],[12,"stack_size_mixer","","[r/w] Optional. Specify 0 to ignore. Specify the stack size for the FMOD mixer thread. Useful for custom dsps that use excess\nstack. Default 49,152 (48kb)",4,null],[3,"OutputHandle","","Wrapper for OutputHandle",null,null],[3,"CreateSoundexInfo","","Use this structure with [`Sys::create_sound`](struct.Sys.html#method.create_sound) when more control is needed over loading.\nThe possible reasons to use this with [`Sys::create_sound`](struct.Sys.html#method.create_sound) are:",null,null],[12,"length","","[w] Optional. Specify 0 to ignore. Size in bytes of file to load, or sound to create (in this case only if FMOD_OPENUSER is\nused). Required if loading from memory. If 0 is specified, then it will use the size of the file (unless loading from memory\nthen an error will be returned).",5,null],[12,"file_offset","","[w] Optional. Specify 0 to ignore. Offset from start of the file to start loading from. This is useful for loading files from\ninside big data files.",5,null],[12,"num_channels","","[w] Optional. Specify 0 to ignore. Number of channels in a sound mandatory if FMOD_OPENUSER or FMOD_OPENRAW is used.",5,null],[12,"default_frequency","","[w] Optional. Specify 0 to ignore. Default frequency of sound in a sound mandatory if FMOD_OPENUSER or FMOD_OPENRAW is used.\nOther formats use the frequency determined by the file format.",5,null],[12,"format","","[w] Optional. Specify 0 or ::SoundFormatNone to ignore. Format of the sound mandatory if FMOD_OPENUSER or FMOD_OPENRAW is used.\nOther formats use the format determined by the file format.",5,null],[12,"decode_buffer_size","","[w] Optional. Specify 0 to ignore. For streams. This determines the size of the double buffer (in PCM samples) that a stream\nuses. Use this for user created streams if you want to determine the size of the callback buffer passed to you. Specify 0\nto use FMOD's default size which is currently equivalent to 400ms of the sound format created/loaded.",5,null],[12,"initial_subsound","","[w] Optional. Specify 0 to ignore. In a multi-sample file format such as .FSB/.DLS/.SF2, specify the initial subsound to seek\nto, only if FMOD_CREATESTREAM is used.",5,null],[12,"num_subsounds","","[w] Optional. Specify 0 to ignore or have no subsounds. In a sound created with FMOD_OPENUSER, specify the number of subsounds\nthat are accessable with [`Sound::get_sub_sound`](doc/rfmod/struct.Sound.html#method.get_sub_sound). If not created with\nFMOD_OPENUSER, this will limit the number of subsounds loaded within a multi-subsound file. If using FSB, then if\nFMOD_CREATESOUNDEXINFO::inclusionlist is used, this will shuffle subsounds down so that there are not any gaps. It will mean\nthat the indices of the sounds will be different.",5,null],[12,"inclusion_list","","[w] Optional. Specify 0 to ignore. In a multi-sample format such as .FSB/.DLS/.SF2 it may be desirable to specify only a subset\nof sounds to be loaded out of the whole file. This is an array of subsound indices to load into memory when created.",5,null],[12,"pcm_read_callback","","[w] Optional. Specify 0 to ignore. Callback to 'piggyback' on FMOD's read functions and accept or even write PCM data while\nFMOD is opening the sound. Used for user sounds created with FMOD_OPENUSER or for capturing decoded data as FMOD reads it.",5,null],[12,"pcm_set_pos_callback","","[w] Optional. Specify 0 to ignore. Callback for when the user calls a seeking function such as [`Channel::set_time`](doc/rfmod/struct.Channel.html#method.set_time)\nor [`Channel::set_position`](doc/rfmod/struct.Channel.html#method.set_position) within a multi-sample sound, and for when it is opened.",5,null],[12,"non_block_callback","","[w] Optional. Specify 0 to ignore. Callback for successful completion, or error while loading a sound that used the\nFMOD_NONBLOCKING flag. Also called duing seeking, when setPosition is called or a stream is restarted.",5,null],[12,"dls_name","","[w] Optional. Specify 0 to ignore. Filename for a DLS or SF2 sample set when loading a MIDI file. If not specified, on Windows\nit will attempt to open /windows/system32/drivers/gm.dls or /windows/system32/drivers/etc/gm.dls, on Mac it will attempt to\nload /System/Library/Components/CoreAudio.component/Contents/Resources/gs_instruments.dls, otherwise the MIDI will fail to\nopen. Current DLS support is for level 1 of the specification.",5,null],[12,"encryption_key","","[w] Optional. Specify 0 to ignore. Key for encrypted FSB file. Without this key an encrypted FSB file will not load.",5,null],[12,"max_polyphony","","[w] Optional. Specify 0 to ignore. For sequenced formats with dynamic channel allocation such as .MID and .IT, this specifies\nthe maximum voice count allowed while playing. .IT defaults to 64. .MID defaults to 32.",5,null],[12,"suggested_sound_type","","[w] Optional. Specify 0 or SoundTypeUnknown to ignore. Instead of scanning all codec types, use this to speed up loading by\nmaking it jump straight to this codec.",5,null],[12,"speaker_map","","[w] Optional. Specify 0 to ignore. Use this to differ the way fmod maps multichannel sounds to speakers. See SpeakerMapType\nfor more.",5,null],[12,"initial_sound_group","","[w] Optional. Specify 0 to ignore. Specify a sound group if required, to put sound in as it is created.",5,null],[12,"initial_seek_position","","[w] Optional. Specify 0 to ignore. For streams. Specify an initial position to seek the stream to.",5,null],[12,"initial_seek_pos_type","","[w] Optional. Specify 0 to ignore. For streams. Specify the time unit for the position set in initialseekposition.",5,null],[12,"ignore_set_file_system","","[w] Optional. Specify true to ignore. Set to false to use fmod's built in file system. Ignores setFileSystem callbacks and also\nFMOD_CREATESOUNEXINFO file callbacks. Useful for specific cases where you don't want to use your own file system but want to\nuse fmod's file system (ie net streaming).",5,null],[12,"cdda_force_aspi","","[w] Optional. Specify 0 to ignore. For CDDA sounds only - if non-zero use ASPI instead of NTSCSI to access the specified CD/DVD\ndevice.",5,null],[12,"audio_queue_policy","","[w] Optional. Specify 0 or FMOD_AUDIOQUEUE_CODECPOLICY_DEFAULT to ignore. Policy used to determine whether hardware or software\nis used for decoding, see FMOD_AUDIOQUEUE_CODECPOLICY for options (iOS >= 3.0 required, otherwise only hardware is available)",5,null],[12,"min_midi_granularity","","[w] Optional. Specify 0 to ignore. Allows you to set a minimum desired MIDI mixer granularity. Values smaller than 512 give\ngreater than default accuracy at the cost of more CPU and vice versa. Specify 0 for default (512 samples).",5,null],[12,"non_block_thread_id","","[w] Optional. Specify 0 to ignore. Specifies a thread index to execute non blocking load on. Allows for up to 5 threads to be\nused for loading at once. This is to avoid one load blocking another. Maximum value = 4.",5,null],[3,"MemoryUsageDetails","","Structure to be filled with detailed memory usage information of a FMOD object",null,null],[12,"other","","[out] Memory not accounted for by other types",6,null],[12,"string","","[out] String data",6,null],[12,"system","","[out] System object and various internals",6,null],[12,"plugins","","[out] Plugin objects and internals",6,null],[12,"output","","[out] Output module object and internals",6,null],[12,"channel","","[out] Channel related memory",6,null],[12,"channel_group","","[out] ChannelGroup objects and internals",6,null],[12,"codec","","[out] Codecs allocated for streaming",6,null],[12,"file","","[out] File buffers and structures",6,null],[12,"sound","","[out] Sound objects and internals",6,null],[12,"secondary_ram","","[out] Sound data stored in secondary RAM",6,null],[12,"sound_group","","[out] SoundGroup objects and internals",6,null],[12,"stream_buffer","","[out] Stream buffer memory",6,null],[12,"dsp_connection","","[out] DSPConnection objects and internals",6,null],[12,"dsp","","[out] DSP implementation objects",6,null],[12,"dsp_codec","","[out] Realtime file format decoding DSP objects",6,null],[12,"profile","","[out] Profiler memory footprint.",6,null],[12,"record_buffer","","[out] Buffer used to store recorded data from microphone",6,null],[12,"reverb","","[out] Reverb implementation objects",6,null],[12,"reverb_channel_props","","[out] Reverb channel properties structs",6,null],[12,"geometry","","[out] Geometry objects and internals",6,null],[12,"sync_point","","[out] Sync point memory.",6,null],[12,"event_system","","[out] EventSystem and various internals",6,null],[12,"music_system","","[out] MusicSystem and various internals",6,null],[12,"fev","","[out] Definition of objects contained in all loaded projects e.g. events, groups, categories",6,null],[12,"memory_fsb","","[out] Data loaded with preloadFSB",6,null],[12,"event_project","","[out] EventProject objects and internals",6,null],[12,"event_group_i","","[out] EventGroup objects and internals",6,null],[12,"sound_bank_class","","[out] Objects used to manage wave banks",6,null],[12,"sound_bank_list","","[out] Data used to manage lists of wave bank usage",6,null],[12,"stream_instance","","[out] Stream objects and internals",6,null],[12,"sound_def_class","","[out] Sound definition objects",6,null],[12,"sound_def_def_class","","[out] Sound definition static data objects",6,null],[12,"sound_def_pool","","[out] Sound definition pool data",6,null],[12,"reverb_def","","[out] Reverb definition objects",6,null],[12,"event_reverb","","[out] Reverb objects",6,null],[12,"user_property","","[out] User property objects",6,null],[12,"event_instance","","[out] Event instance base objects",6,null],[12,"event_instance_complex","","[out] Complex event instance objects",6,null],[12,"event_instance_simple","","[out] Simple event instance objects",6,null],[12,"event_instance_layer","","[out] Event layer instance objects",6,null],[12,"event_instance_sound","","[out] Event sound instance objects",6,null],[12,"event_envelope","","[out] Event envelope objects",6,null],[12,"event_envelope_def","","[out] Event envelope definition objects",6,null],[12,"event_parameter","","[out] Event parameter objects",6,null],[12,"event_category","","[out] Event category objects",6,null],[12,"event_envelope_point","","[out] Event envelope point objects",6,null],[12,"event_instance_pool","","[out] Event instance pool memory",6,null],[3,"UserData","","Structure used to store user data for file callback",null,null],[3,"Sound","","Sound object",null,null],[3,"FmodTag","","Structure describing a piece of tag data.",null,null],[12,"_type","","[r] The type of this tag.",7,null],[12,"data_type","","[r] The type of data that this tag contains",7,null],[12,"name","","[r] The name of this tag i.e. "TITLE", "ARTIST" etc.",7,null],[12,"updated","","[r] True if this tag has been updated since last being accessed with [`Sound::get_tag`](struct.Sound.html#method.get_tag)",7,null],[3,"FmodSyncPoint","","Wrapper for SyncPoint object",null,null],[3,"ChannelGroup","","ChannelGroup object",null,null],[3,"SoundGroup","","SoundGroup object",null,null],[3,"Dsp","","Dsp object",null,null],[3,"DspParameterDesc","","Structure to define a parameter for a DSP unit.",null,null],[12,"min","","[w] Minimum value of the parameter (ie 100.0)",8,null],[12,"max","","[w] Maximum value of the parameter (ie 22050.0)",8,null],[12,"default_val","","[w] Default value of parameter",8,null],[12,"name","","[w] Name of the parameter to be displayed (ie "Cutoff frequency")",8,null],[12,"label","","[w] Short string to be put next to value to denote the unit type (ie "hz")",8,null],[12,"description","","[w] Description of the parameter to be displayed as a help item / tooltip for this parameter",8,null],[3,"DspDescription","","When creating a DSP unit, declare one of these and provide the relevant callbacks and name for FMOD to use when it creates and uses a DSP unit of this type.",null,null],[12,"name","","[w] Name of the unit to be displayed in the network.",9,null],[12,"version","","[w] Plugin writer's version number.",9,null],[12,"channels","","[w] Number of channels. Use 0 to process whatever number of channels is currently in the network. > 0 would be mostly used if the unit is a unit that only generates sound.",9,null],[12,"create","","[w] Create callback. This is called when DSP unit is created. Can be null.",9,null],[12,"release","","[w] Release callback. This is called just before the unit is freed so the user can do any cleanup needed for the unit. Can be null.",9,null],[12,"reset","","[w] Reset callback. This is called by the user to reset any history buffers that may need resetting for a filter, when it is to be used or re-used for the first time to its initial clean state. Use to avoid clicks or artifacts.",9,null],[12,"read","","[w] Read callback. Processing is done here. Can be null.",9,null],[12,"set_position","","[w] Set position callback. This is called if the unit wants to update its position info but not process data, or reset a cursor position internally if it is reading data from a certain source. Can be null.",9,null],[12,"num_parameters","","[w] Number of parameters used in this filter. The user finds this with DSP::getNumParameters",9,null],[12,"param_desc","","[w] Variable number of parameter structures.",9,null],[12,"set_parameter","","[w] This is called when the user calls DSP::setParameter. Can be null.",9,null],[12,"get_parameter","","[w] This is called when the user calls DSP::getParameter. Can be null.",9,null],[12,"config_width","","[w] Width of config dialog graphic if there is one. 0 otherwise.",9,null],[12,"config_height","","[w] Height of config dialog graphic if there is one. 0 otherwise.",9,null],[3,"DspState","","DSP plugin structure that is passed into each callback.",null,null],[12,"instance","","[r] Handle to the DSP hand the user created. Not to be modified. C++ users cast toDSP to use.",10,null],[12,"speaker_mask","","[w] Specifies which speakers the DSP effect is active on",10,null],[3,"DspConnection","","DspConnection object",null,null],[3,"Reverb","","Reverb object",null,null],[3,"ReverbProperties","","Structure defining a reverb environment.",null,null],[12,"instance","","[w]   Min: 0 - Max: 3 - Default: 0 - Environment Instance. (SUPPORTED:SFX(4 instances) and Wii (3 instances))",11,null],[12,"environment","","[r/w] Min: -1 - Max: 25 - Default: -1 - Sets all listener properties. -1 = OFF. (SUPPORTED:SFX(-1 only)/PSP)",11,null],[12,"env_diffusion","","[r/w] Min: 0.0 - Max: 1.0 - Default: 1.0 - Environment diffusion  (SUPPORTED:WII)",11,null],[12,"room","","[r/w] Min: -10000 - Max: 0 - Default: -1000 - Room effect level (at mid frequencies)  (SUPPORTED:SFX/WII/PSP)",11,null],[12,"room_HF","","[r/w] Min: -10000 - Max: 0 - Default: -100 - Relative room effect level at high frequencies  (SUPPORTED:SFX)",11,null],[12,"room_LF","","[r/w] Min: -10000 - Max: 0 Default: 0 - Relative room effect level at low frequencies  (SUPPORTED:SFX)",11,null],[12,"decay_time","","[r/w] Min: 0.1 - Max: 20.0 - Default: 1.49 - Reverberation decay time at mid frequencies  (SUPPORTED:SFX/WII)",11,null],[12,"decay_HF_ratio","","[r/w] Min: 0.1 - Max: 2.0 - Default: 0.83 - High-frequency to mid-frequency decay time ratio  (SUPPORTED:SFX)",11,null],[12,"decay_LF_ratio","","[r/w] Min: 0.1 - Max: 2.0 - Default: 1.0 - Low-frequency to mid-frequency decay time ratio  (SUPPORTED:---)",11,null],[12,"reflections","","[r/w] Min: -10000 - Max: 1000 - Default: -2602 - Early reflections level relative to room effect  (SUPPORTED:SFX/WII)",11,null],[12,"reflections_delay","","[r/w] Min: 0.0 - Max: 0.3 - Default: 0.007 - Initial reflection delay time  (SUPPORTED:SFX)",11,null],[12,"reverb","","[r/w] Min: -10000 - Max: 2000 - Default: 200 - Late reverberation level relative to room effect  (SUPPORTED:SFX)",11,null],[12,"reverb_delay","","[r/w] Min: 0.0 - Max: 0.1 - Default: 0.011 - Late reverberation delay time relative to initial reflection  (SUPPORTED:SFX/WII)",11,null],[12,"modulation_time","","[r/w] Min: 0.04 - Max: 4.0 - Default: 0.25 - Modulation time  (SUPPORTED:---)",11,null],[12,"modulation_depth","","[r/w] Min: 0.0 - Max: 1.0 - Default: 0.0 - Modulation depth  (SUPPORTED:WII)",11,null],[12,"HF_reference","","[r/w] Min: 20.0 - Max: 20000.0 - Default: 5000.0 - Reference high frequency (hz)  (SUPPORTED:SFX)",11,null],[12,"LF_reference","","[r/w] Min: 20.0 - Max: 1000.0 - Default: 250.0 - Reference low frequency (hz)  (SUPPORTED:SFX)",11,null],[12,"diffusion","","[r/w] Min: 0.0 - Max: 100.0 - Default: 100.0 - Value that controls the echo density in the late reverberation decay. (SUPPORTED:SFX)",11,null],[12,"density","","[r/w] Min: 0.0 - Max: 100.0 - Default: 100.0 - Value that controls the modal density in the late reverberation decay  (SUPPORTED:SFX)",11,null],[12,"flags","","[r/w] FMOD_REVERB_FLAGS - modifies the behavior of above properties  (SUPPORTED:WII)",11,null],[3,"Vector","","Structure describing a point in 3D space.",null,null],[12,"x","","X co-ordinate in 3D space.",12,null],[12,"y","","Y co-ordinate in 3D space.",12,null],[12,"z","","Z co-ordinate in 3D space.",12,null],[3,"Geometry","","Geometry object",null,null],[3,"FmodFile","","A little struct to wrap C files. I'll try to improve this or to replace it by File",null,null],[4,"SeekStyle","","",null,null],[13,"SeekSet","","Seek from the beginning of the stream",13,null],[13,"SeekEnd","","Seek from the end of the stream",13,null],[13,"SeekCur","","Seek from the current position",13,null],[4,"Result","","Error codes. Returned from every function.",null,null],[13,"Ok","","No errors.",14,null],[13,"AlreadyLocked","","Tried to call lock a second time before unlock was called.",14,null],[13,"BadCommand","","Tried to call a function on a data type that does not allow this type of functionality (ie calling Sound::lock on a streaming sound).",14,null],[13,"CDDADrivers","","Neither NTSCSI nor ASPI could be initialised.",14,null],[13,"CDDAInit","","An error occurred while initialising the CDDA subsystem.",14,null],[13,"CDDAInvalidDevice","","Couldn't find the specified device.",14,null],[13,"CDDANoAudio","","No audio tracks on the specified disc.",14,null],[13,"CDDANoDevices","","No CD/DVD devices were found.",14,null],[13,"CDDANoDisc","","No disc present in the specified drive.",14,null],[13,"CDDARead","","A CDDA read error occurred.",14,null],[13,"ChannelAlloc","","Error trying to allocate a channel.",14,null],[13,"ChannelStolen","","The specified channel has been reused to play another sound.",14,null],[13,"COM","","A Win32 COM related error occured. COM failed to initialize or a QueryInterface failed meaning a Windows codec or driver was not installed properly.",14,null],[13,"DMA","","DMA Failure. See debug output for more information.",14,null],[13,"DSPConnection","","DSP connection error. Connection possibly caused a cyclic dependancy. Or tried to connect a tree too many units deep (more than 128).",14,null],[13,"DSPFormat","","DSP Format error. A DSP unit may have attempted to connect to this network with the wrong format.",14,null],[13,"DSPNotFound","","DSP connection error. Couldn't find the DSP unit specified.",14,null],[13,"DSPRunning","","DSP error. Cannot perform this operation while the network is in the middle of running. This will most likely happen if a connection or disconnection is attempted in a DSP callback.",14,null],[13,"DSPTooManyConnections","","DSP connection error. The unit being connected to or disconnected should only have 1 input or output.",14,null],[13,"FileBad","","Error loading file.",14,null],[13,"FileCouldNotSeek","","Couldn't perform seek operation. This is a limitation of the medium (ie netstreams) or the file format.",14,null],[13,"FileDiskEjected","","Media was ejected while reading.",14,null],[13,"FileEOF","","End of file unexpectedly reached while trying to read essential data (truncated data ?).",14,null],[13,"FileNotFound","","File not found.",14,null],[13,"FileUnwanted","","Unwanted file access occured.",14,null],[13,"Format","","Unsupported file or audio format.",14,null],[13,"HTTP","","A HTTP error occurred. This is a catch-all for HTTP errors not listed elsewhere.",14,null],[13,"HTTPAccess","","The specified resource requires authentication or is forbidden.",14,null],[13,"HTTPProxyAuth","","Proxy authentication is required to access the specified resource.",14,null],[13,"HTTPServerError","","A HTTP server error occurred.",14,null],[13,"HTTPTimeout","","The HTTP request timed out.",14,null],[13,"Initialization","","FMOD was not initialized correctly to support this function.",14,null],[13,"Initialized","","Cannot call this command after System::init.",14,null],[13,"Internal","","An error occured that wasn't supposed to. Contact support.",14,null],[13,"InvalidAddress","","On Xbox 360, this memory address passed to FMOD must be physical, (ie allocated with XPhysicalAlloc.)",14,null],[13,"InvalidFloat","","Value passed in was a NaN, Inf or denormalized float.",14,null],[13,"InvalidHandle","","An invalid object handle was used.",14,null],[13,"InvalidParam","","An invalid parameter was passed to this function.",14,null],[13,"InvalidPosition","","An invalid seek position was passed to this function.",14,null],[13,"InvalidSpeaker","","An invalid speaker was passed to this function based on the current speaker mode.",14,null],[13,"InvalidSyncPoint","","The syncpoint did not come from this sound handle.",14,null],[13,"InvalidVector","","The vectors passed in are not unit length, or perpendicular.",14,null],[13,"MaxAudible","","Reached maximum audible playback count for this sound's soundgroup.",14,null],[13,"Memory","","Not enough memory or resources.",14,null],[13,"MemoryCantPoint","","Can't use FMOD_OPENMEMORY_POINT on non PCM source data, or non mp3/xma/adpcm data if FMOD_CREATECOMPRESSEDSAMPLE was used.",14,null],[13,"MemorySRAM","","Not enough memory or resources on console sound ram.",14,null],[13,"Needs2D","","Tried to call a command on a 3d sound when the command was meant for 2d sound.",14,null],[13,"Needs3D","","Tried to call a command on a 2d sound when the command was meant for 3d sound.",14,null],[13,"NeedsHardware","","Tried to use a feature that requires hardware support. (ie trying to play a GCADPCM compressed sound in software on Wii).",14,null],[13,"NeedsSoftware","","Tried to use a feature that requires the software engine. Software engine has either been turned off, or command was executed on a hardware channel which does not support this feature.",14,null],[13,"NetConnect","","Couldn't connect to the specified host.",14,null],[13,"NetSocketError","","A socket error occurred. This is a catch-all for socket-related errors not listed elsewhere.",14,null],[13,"NetURL","","The specified URL couldn't be resolved.",14,null],[13,"NetWouldBlock","","Operation on a non-blocking socket could not complete immediately.",14,null],[13,"NotReady","","Operation could not be performed because specified sound/DSP connection is not ready.",14,null],[13,"OutputAllocated","","Error initializing output device, but more specifically, the output device is already in use and cannot be reused.",14,null],[13,"OutputCreateBuffer","","Error creating hardware sound buffer.",14,null],[13,"OutputDriverCall","","A call to a standard soundcard driver failed, which could possibly mean a bug in the driver or resources were missing or exhausted.",14,null],[13,"OutputEnumeration","","Error enumerating the available driver list. List may be inconsistent due to a recent device addition or removal.",14,null],[13,"OutputFormat","","Soundcard does not support the minimum features needed for this soundsystem (16bit stereo output).",14,null],[13,"OutputInit","","Error initializing output device.",14,null],[13,"OutputNoHardware","","FMOD_HARDWARE was specified but the sound card does not have the resources necessary to play it.",14,null],[13,"OutputNoSoftware","","Attempted to create a software sound but no software channels were specified in System::init.",14,null],[13,"Pan","","Panning only works with mono or stereo sound sources.",14,null],[13,"Plugin","","An unspecified error has been returned from a 3rd party plugin.",14,null],[13,"PluginInstances","","The number of allowed instances of a plugin has been exceeded.",14,null],[13,"PluginMissing","","A requested output, dsp unit type or codec was not available.",14,null],[13,"PluginResource","","A resource that the plugin requires cannot be found. (ie the DLS file for MIDI playback or other DLLs that it needs to load)",14,null],[13,"Preloaded","","The specified sound is still in use by the event system, call EventSystem::unloadFSB before trying to release it.",14,null],[13,"ProgrammerSound","","The specified sound is still in use by the event system, wait for the event which is using it finish with it.",14,null],[13,"Record","","An error occured trying to initialize the recording device.",14,null],[13,"ReverbInstance","","Specified instance in FMOD_REVERB_PROPERTIES couldn't be set. Most likely because it is an invalid instance number or the reverb doesnt exist.",14,null],[13,"SubsoundAllocated","","This subsound is already being used by another sound, you cannot have more than one parent to a sound. Null out the other parent's entry first.",14,null],[13,"SubsoundCantMove","","Shared subsounds cannot be replaced or moved from their parent stream, such as when the parent stream is an FSB file.",14,null],[13,"SubsoundMode","","The subsound's mode bits do not match with the parent sound's mode bits. See documentation for function that it was called with.",14,null],[13,"Subsounds","","The error occured because the sound referenced contains subsounds when it shouldn't have, or it doesn't contain subsounds when it should have. The operation may also not be able to be performed on a parent sound, or a parent sound was played without setting up a sentence first.",14,null],[13,"TagNotFound","","The specified tag could not be found or there are no tags.",14,null],[13,"TooManyChannels","","The sound created exceeds the allowable input channel count. This can be increased using the maxinputchannels parameter in System::setSoftwareFormat.",14,null],[13,"Unimplemented","","Something in FMOD hasn't been implemented when it should be ! contact support !",14,null],[13,"Uninitialized","","This command failed because System::init or System::setDriver was not called.",14,null],[13,"Unsupported","","A command issued was not supported by this object. Possibly a plugin without certain callbacks specified.",14,null],[13,"Update","","An error caused by System::update occured.",14,null],[13,"Version","","The version number of this file format is not supported.",14,null],[13,"EventFailed","","An Event failed to be retrieved, most likely due to 'just fail' being specified as the max playbacks behavior.",14,null],[13,"EventInfoOnly","","Can't execute this command on an EVENT_INFOONLY event.",14,null],[13,"EventInternal","","An error occured that wasn't supposed to. See debug log for reason.",14,null],[13,"EventMaxStreams","","Event failed because 'Max streams' was hit when FMOD_EVENT_INIT_FAIL_ON_MAXSTREAMS was specified.",14,null],[13,"EventMismatch","","FSB mismatches the FEV it was compiled with, the stream/sample mode it was meant to be created with was different, or the FEV was built for a different platform.",14,null],[13,"EventNameConflict","","A category with the same name already exists.",14,null],[13,"EventNotFound","","The requested event, event group, event category or event property could not be found.",14,null],[13,"EventNeedsSimple","","Tried to call a function on a complex event that's only supported by simple events.",14,null],[13,"EventGuidConflict","","An event with the same GUID already exists.",14,null],[13,"EventAlreadyLoaded","","The specified project or bank has already been loaded. Having multiple copies of the same project loaded simultaneously is forbidden.",14,null],[13,"MusicUninitialized","","Music system is not initialized probably because no music data is loaded.",14,null],[13,"MusicNotFound","","The requested music entity could not be found.",14,null],[13,"MusicNoCallback","","The music callback is required, but it has not been set.",14,null],[13,"ResultForceInt","","Makes sure this enum is signed 32bit.",14,null],[4,"SpeakerMapType","","When creating a multichannel sound, FMOD will pan them to their default speaker locations:\n* For example a 6 channel sound will default to one channel per 5.1 output speaker.\n* Another example is a stereo sound. It will default to left = front left, right = front right.\n* This is for sounds that are not 'default'. For example you might have a sound that is 6 channels but actually made up of 3 stereo pairs, that should all be located in front left, front right only.",null,null],[13,"Default","","This is the default, and just means FMOD decides which speakers it puts the source channels.",15,null],[13,"AllMono","","This means the sound is made up of all mono sounds. All voices will be panned to the front center by default in this case.",15,null],[13,"AllStereo","","This means the sound is made up of all stereo sounds. All voices will be panned to front left and front right alternating every second channel.",15,null],[13,"_51ProTools","","Map a 5.1 sound to use protools L C R Ls Rs LFE mapping. Will return an error if not a 6 channel sound.",15,null],[4,"SoundFormat","","These definitions describe the native format of the hardware or software buffer that will be used.",null,null],[13,"None","","Uninitialized / unknown.",16,null],[13,"PCM8","","8bit integer PCM data.",16,null],[13,"PCM16","","16bit integer PCM data.",16,null],[13,"PCM24","","24bit integer PCM data.",16,null],[13,"PCM32","","32bit integer PCM data.",16,null],[13,"PCMFloat","","32bit floating point PCM data.",16,null],[13,"GCADPCM","","Compressed Nintendo 3DS/Wii DSP data.",16,null],[13,"IMAADPCM","","Compressed IMA ADPCM data.",16,null],[13,"VAG","","Compressed PlayStation Portable ADPCM data.",16,null],[13,"HEVAG","","Compressed PSVita ADPCM data.",16,null],[13,"XMA","","Compressed Xbox360 XMA data.",16,null],[13,"MPEG","","Compressed MPEG layer 2 or 3 data.",16,null],[13,"CELT","","Compressed CELT data.",16,null],[13,"AT9","","Compressed PSVita ATRAC9 data.",16,null],[13,"XWMA","","Compressed Xbox360 xWMA data.",16,null],[13,"VORBIS","","Compressed Vorbis data.",16,null],[13,"Max","","Maximum number of sound formats supported.",16,null],[13,"ForceInt","","Makes sure this enum is signed 32bit.",16,null],[4,"SoundType","","These definitions describe the type of song being played.",null,null],[13,"Unknown","","3rd party / unknown plugin format.",17,null],[13,"AIFF","","AIFF.",17,null],[13,"ASF","","Microsoft Advanced Systems Format (ie WMA/ASF/WMV).",17,null],[13,"AT3","","Sony ATRAC 3 format",17,null],[13,"CDDA","","Digital CD audio.",17,null],[13,"DLS","","Sound font / downloadable sound bank.",17,null],[13,"FLAC","","FLAC lossless codec.",17,null],[13,"FSB","","FMOD Sample Bank.",17,null],[13,"GCADPCM","","Nintendo GameCube/Wii ADPCM",17,null],[13,"IT","","Impulse Tracker.",17,null],[13,"MIDI","","MIDI. extracodecdata is a pointer to an FMOD_MIDI_EXTRACODECDATA structure.",17,null],[13,"MOD","","Protracker / Fasttracker MOD.",17,null],[13,"MPEG","","MP2/MP3 MPEG.",17,null],[13,"OGGVORBIS","","Ogg vorbis.",17,null],[13,"Playlist","","Information only from ASX/PLS/M3U/WAX playlists",17,null],[13,"Raw","","Raw PCM data.",17,null],[13,"S3M","","ScreamTracker 3.",17,null],[13,"SF2","","Sound font 2 format.",17,null],[13,"User","","User created sound.",17,null],[13,"WAV","","Microsoft WAV.",17,null],[13,"XM","","FastTracker 2 XM.",17,null],[13,"XMA","","Xbox360 XMA",17,null],[13,"VAG","","PlayStation Portable ADPCM VAG format.",17,null],[13,"AudioQueue","","iPhone hardware decoder, supports AAC, ALAC and MP3. extracodecdata is a pointer to an FMOD_AUDIOQUEUE_EXTRACODECDATA structure.",17,null],[13,"XWMA","","Xbox360 XWMA",17,null],[13,"BCWAV","","3DS BCWAV container format for DSP ADPCM and PCM",17,null],[13,"AT9","","NGP ATRAC 9 format",17,null],[13,"VORBIS","","Raw vorbis",17,null],[13,"MediaFoundation","","Microsoft Media Foundation wrappers, supports ASF/WMA",17,null],[13,"Max","","Maximum number of sound types supported.",17,null],[13,"ForceInt","","Makes sure this enum is signed 32bit.",17,null],[4,"TagType","","List of tag types that could be stored within a sound. These include id3 tags, metadata from netstreams and vorbis/asf data.",null,null],[13,"Unknown","","",18,null],[13,"ID3V1","","",18,null],[13,"ID3V2","","",18,null],[13,"VORBISComment","","",18,null],[13,"ShoutCast","","",18,null],[13,"IceCast","","",18,null],[13,"ASF","","",18,null],[13,"MIDI","","",18,null],[13,"Playlist","","",18,null],[13,"Fmod","","",18,null],[13,"User","","",18,null],[13,"Max","","Maximum number of tag types supported.",18,null],[13,"ForceInt","","Makes sure this enum is signed 32bit.",18,null],[4,"TagDataType","","List of data types that can be returned by [`Sound::get_tag`](../../struct.Sound.html#method.get_tag)",null,null],[13,"Binary","","",19,null],[13,"Int","","",19,null],[13,"Float","","",19,null],[13,"String","","",19,null],[13,"StringUTF16","","",19,null],[13,"StringUTF16BE","","",19,null],[13,"StringUTF8","","",19,null],[13,"CDTOC","","",19,null],[13,"Max","","Maximum number of tag datatypes supported.",19,null],[13,"ForceInt","","Makes sure this enum is signed 32bit.",19,null],[4,"ChannelIndex","","Special channel index values for FMOD functions.",null,null],[13,"Free","","For a channel index, FMOD chooses a free voice using the priority system.",20,null],[13,"ReUse","","For a channel index, re-use the channel handle that was passed in.",20,null],[4,"DspFftWindow","","List of windowing methods used in spectrum analysis to reduce leakage / transient signals intefering with the analysis.\nThis is a problem with analysis of continuous signals that only have a small portion of the signal sample (the fft window size).\nWindowing the signal with a curve or triangle tapers the sides of the fft window to help alleviate this problem.",null,null],[13,"Rect","","w[n] = 1.0",21,null],[13,"Triangle","","w[n] = TRI(2n/N)",21,null],[13,"Hamming","","w[n] = 0.54 - (0.46 * COS(n/N) )",21,null],[13,"Hanning","","w[n] = 0.5 *  (1.0  - COS(n/N) )",21,null],[13,"BlackMan","","w[n] = 0.42 - (0.5  * COS(n/N) ) + (0.08 * COS(2.0 * n/N) )",21,null],[13,"BlackManHarris","","w[n] = 0.35875 - (0.48829 * COS(1.0 * n/N)) + (0.14128 * COS(2.0 * n/N)) - (0.01168 * COS(3.0 * n/N))",21,null],[13,"Max","","Maximum number of FFT window types supported.",21,null],[13,"ForceInt","","Makes sure this enum is signed 32bit.",21,null],[4,"DelayType","","Types of delay that can be used with [`Channel::set_delay`](../../struct.Channel.html#method.set_delay) / [`Channel::get_delay`](../../struct.Channel.html#method.get_delay).",null,null],[13,"EndMS","","Delay at the end of the sound in milliseconds. Use delayhi only. [`Channel::is_playing`](../../struct.Channel.html#method.is_playing) will remain true until this delay has passed even though the sound itself has stopped playing.",22,null],[13,"DSPClockStart","","Time the sound started if [`Channel::get_delay`](../../struct.Channel.html#method.get_delay) is used, or if [`Channel::set_delay`](../../struct.Channel.html#method.set_delay) is used, the sound will delay playing until this exact tick.",22,null],[13,"DSPClockEnd","","Time the sound should end. If this is non-zero, the channel will go silent at this exact tick.",22,null],[13,"DSPClockPause","","Time the sound should pause. If this is non-zero, the channel will pause at this exact tick.",22,null],[13,"Max","","Maximum number of tag datatypes supported.",22,null],[13,"ForceInt","","Makes sure this enum is signed 32bit.",22,null],[4,"OutputType","","These output types are used with [`Sys::set_output`](../../struct.Sys.html#method.set_output) / [`Sys::get_output`](../../struct.Sys.html#method.get_output), to choose which output method to use.",null,null],[13,"AutoDetect","","Picks the best output mode for the platform. This is the default.",23,null],[13,"Unknown","","All             - 3rd party plugin, unknown. This is for use with [`Sys::get_output`](../../struct.Sys.html#method.get_output) only.",23,null],[13,"NoSound","","All             - All calls in this mode succeed but make no sound.",23,null],[13,"WAVWriter","","All             - Writes output to fmodoutput.wav by default. Use the 'extradriverdata' parameter in [`Sys::init`](../../struct.Sys.html#method.init), by simply passing the filename as a string, to set the wav filename.",23,null],[13,"NoSoundNRT","","All             - Non-realtime version of FMOD_OUTPUTYPE_NOSOUND. User can drive mixer with [`Sys::update`](../../struct.Sys.html#method.update) at whatever rate they want.",23,null],[13,"WAVWriterNRT","","All             - Non-realtime version of FMOD_OUTPUTYPE_WAVWRITER. User can drive mixer with [`Sys::update`](../../struct.Sys.html#method.update) at whatever rate they want.",23,null],[13,"DSound","","Win32/Win64     - DirectSound output.                     (Default on Windows XP and below)",23,null],[13,"WinMM","","Win32/Win64     - Windows Multimedia output.",23,null],[13,"WASAPI","","Win32           - Windows Audio Session API.              (Default on Windows Vista and above)",23,null],[13,"ASIO","","Win32           - Low latency ASIO 2.0 driver.",23,null],[13,"OSS","","Linux/Linux64   - Open Sound System output.               (Default on Linux, third preference)",23,null],[13,"ALSA","","Linux/Linux64   - Advanced Linux Sound Architecture output. (Default on Linux, second preference if available)",23,null],[13,"ESD","","Linux/Linux64   - Enlightment Sound Daemon output.",23,null],[13,"PulseAudio","","Linux/Linux64   - PulseAudio output.                      (Default on Linux, first preference if available)",23,null],[13,"CoreAudio","","Mac             - Macintosh CoreAudio output.             (Default on Mac)",23,null],[13,"Xbox360","","Xbox 360        - Native Xbox360 output.                  (Default on Xbox 360)",23,null],[13,"PSP","","PSP             - Native PSP output.                      (Default on PSP)",23,null],[13,"PS3","","PS3             - Native PS3 output.                      (Default on PS3)",23,null],[13,"NGP","","NGP             - Native NGP output.                      (Default on NGP)",23,null],[13,"Wii","","Wii             - Native Wii output.                      (Default on Wii)",23,null],[13,"_3DS","","3DS             - Native 3DS output                       (Default on 3DS)",23,null],[13,"AudioTrack","","Android         - Java Audio Track output.                (Default on Android 2.2 and below)",23,null],[13,"OpenSL","","Android         - OpenSL ES output.                       (Default on Android 2.3 and above)",23,null],[13,"NACL","","Native Client   - Native Client output.                   (Default on Native Client)",23,null],[13,"WiiU","","Wii U           - Native Wii U output.                    (Default on Wii U)",23,null],[13,"ASound","","BlackBerry      - Native BlackBerry asound output.        (Default on BlackBerry)",23,null],[13,"AudioOut","","Orbis           - Audio Out output.                       (Default on Orbis)",23,null],[13,"XAudio","","Durango         - XAudio2 output.",23,null],[13,"Max","","Maximum number of output types supported.",23,null],[13,"ForceInt","","Makes sure this enum is signed 32bit.",23,null],[4,"Speaker","","These are speaker types defined for use with the [`Channel::set_speaker_level`](../../struct.Channel.html#method.set_speaker_level) command.\nIt can also be used for speaker placement in the [`Sys::set_3D_speaker_position`](../../struct.Sys.html#method.set_3D_speaker_position) command.",null,null],[13,"FrontLeft","","",24,null],[13,"FrontRight","","",24,null],[13,"FrontCenter","","",24,null],[13,"LowFrequency","","",24,null],[13,"BackLeft","","",24,null],[13,"BackRight","","",24,null],[13,"SideLeft","","",24,null],[13,"SideRight","","",24,null],[13,"Max","","Maximum number of speaker types supported.",24,null],[13,"Null","","A non speaker. Use this with ASIO mapping to ignore a speaker.",24,null],[13,"ForceInt","","Makes sure this enum is signed 32bit.",24,null],[4,"SpeakerMode","","These are speaker types defined for use with the [`Sys::set_speaker_mode`](../../struct.Sys.html#method.set_speaker_mode) or [`Sys::get_speaker_mode`](../../struct.Sys.html#method.get_speaker_mode) command.",null,null],[13,"Raw","","There is no specific . Sound channels are mapped in order of input to output. Use [`Sys::set_software_format`](../../struct.Sys.html#method.set_software_format) to specify speaker count. See remarks for more information.",25,null],[13,"Mono","","The speakers are monaural.",25,null],[13,"Stereo","","The speakers are stereo (DEFAULT).",25,null],[13,"Quad","","4 speaker setup. This includes front left, front right, rear left, rear right.",25,null],[13,"Surround","","5 speaker setup. This includes front left, front right, center, rear left, rear right.",25,null],[13,"_5Point1","","5.1 speaker setup. This includes front left, front right, center, rear left, rear right and a subwoofer.",25,null],[13,"_7Point1","","7.1 speaker setup. This includes front left, front right, center, rear left, rear right, side left, side right and a subwoofer.",25,null],[13,"SRS5_1_Matrix","","Stereo compatible output, embedded with surround information. SRS 5.1/Prologic/Prologic2 decoders will split the signal into a 5.1 speaker set-up or SRS virtual surround will decode into a 2-speaker/headphone setup. See remarks about limitations.",25,null],[13,"DOLBY5_1_Matrix","","Stereo compatible output, embedded with surround information. Dolby Pro Logic II decoders will split the signal into a 5.1 speaker set-up.",25,null],[13,"MYears","","Stereo output, but data is encoded using personalized HRTF algorithms. See myears.net.au",25,null],[13,"Max","","Maximum number of speaker modes supported.",25,null],[13,"ForceInt","","Makes sure this enum is signed 32bit.",25,null],[4,"DspResampler","","List of interpolation types that the FMOD Ex software mixer supports.",null,null],[13,"NoInterp","","No interpolation. High frequency aliasing hiss will be audible depending on the sample rate of the sound.",26,null],[13,"Linear","","Linear interpolation (default method). Fast and good quality, causes very slight lowpass effect on low frequency sounds.",26,null],[13,"Cubic","","Cubic interpolation. Slower than linear interpolation but better quality.",26,null],[13,"Spline","","5 point spline interpolation. Slowest resampling method but best quality.",26,null],[13,"Max","","Maximum number of resample methods supported.",26,null],[13,"ForceInt","","Makes sure this enum is signed 32bit.",26,null],[4,"PluginType","","These are plugin types defined for use with the [`Sys::get_num_plugins`](../../struct.Sys.html#method.get_num_plugins), [`Sys::get_plugin_info`](../../struct.Sys.html#method.get_plugin_info) and [`Sys::unload_plugin`](../../struct.Sys.html#method.unload_plugin) functions.",null,null],[13,"Output","","The plugin type is an output module. FMOD mixed audio will play through one of these devices",27,null],[13,"Codec","","The plugin type is a file format codec. FMOD will use these codecs to load file formats for playback.",27,null],[13,"DSP","","The plugin type is a DSP unit. FMOD will use these plugins as part of its DSP network to apply effects to output or generate sound in realtime.",27,null],[13,"Max","","Maximum number of plugin types supported.",27,null],[13,"ForceInt","","Makes sure this enum is signed 32bit.",27,null],[4,"OpenState","","These values describe what state a sound is in after FMOD_NONBLOCKING has been used to open it.",null,null],[13,"Ready","","Opened and ready to play.",28,null],[13,"Loading","","Initial load in progress.",28,null],[13,"Error","","Failed to open - file not found, out of memory etc. See return value of [`Sound::get_open_state`](../../struct.Sound.html#method.get_open_state) for what happened.",28,null],[13,"Connecting","","Connecting to remote host (internet sounds only).",28,null],[13,"Buffering","","Buffering data.",28,null],[13,"Seeking","","Seeking to subsound and re-flushing stream buffer.",28,null],[13,"Playing","","Ready and playing, but not possible to release at this time without stalling the main thread.",28,null],[13,"SetPosition","","Seeking within a stream to a different position.",28,null],[13,"Max","","Maximum number of open state types.",28,null],[13,"ForceInt","","Makes sure this enum is signed 32bit.",28,null],[4,"SystemCallbackType","","These callback types are used with [`Channel::set_callback`](../../struct.Channel.html#method.set_callback).",null,null],[13,"DeviceListChanged","","Called from [`Sys::update`](../../struct.Sys.html#method.update) when the enumerated list of devices has changed.",29,null],[13,"DeviceLost","","Called from [`Sys::update`](../../struct.Sys.html#method.update) when an output device has been lost due to control panel parameter changes and FMOD cannot automatically recover.",29,null],[13,"MemoryAllocationFailed","","Called directly when a memory allocation fails somewhere in FMOD. (NOTE - 'system' will be NULL in this callback type.)",29,null],[13,"ThreadCreated","","Called directly when a thread is created.",29,null],[13,"BadDSPConnection","","Called when a bad connection was made with [`Dsp::add_input`](../../struct.Dsp.html#method.add_input). Usually called from mixer thread because that is where the connections are made.",29,null],[13,"BadDSPLevel","","Called when too many effects were added exceeding the maximum tree depth of 128. This is most likely caused by accidentally adding too many DSP effects. Usually called from mixer thread because that is where the connections are made.",29,null],[13,"ThreadDestroyed","","Called directly when a thread is destroyed.",29,null],[13,"Max","","Maximum number of callback types supported.",29,null],[13,"ForceInt","","Makes sure this enum is signed 32bit.",29,null],[4,"SoundGroupBehavior","","These flags are used with [`SoundGroup::set_max_audible_behavior`](../../struct.SoundGroup.html#method.set_max_audible_behavior) to determine what happens when more sounds are played than are specified with [`SoundGroup::set_max_audible`](../../struct.SoundGroup.html#method.set_max_audible).",null,null],[13,"Fail","","Any sound played that puts the sound count over the [`SoundGroup::set_max_audible`](../../struct.SoundGroup.html#method.set_max_audible) setting, will simply fail during [`Sound::play`](../../struct.Sound.html#method.play).",30,null],[13,"Mute","","Any sound played that puts the sound count over the [`SoundGroup::set_max_audible`](../../struct.SoundGroup.html#method.set_max_audible) setting, will be silent, then if another sound in the group stops the sound that was silent before becomes audible again.",30,null],[13,"StealLowest","","Any sound played that puts the sound count over the [`SoundGroup::set_max_audible`](../../struct.SoundGroup.html#method.set_max_audible) setting, will steal the quietest / least important sound playing in the group.",30,null],[13,"Max","","Maximum number of open state types.",30,null],[13,"ForceInt","","Makes sure this enum is signed 32bit.",30,null],[4,"DspType","","These definitions can be used for creating FMOD defined special effects or DSP units.\nUsed with [`Dsp::set_parameter`](../struct.Dsp.html#method.set_parameter) and [`Dsp::get_parameter`](../struct.Dsp.html#method.get_parameter)",null,null],[13,"Unknown","","This unit was created via a non FMOD plugin so has an unknown purpose.",31,null],[13,"Mixer","","This unit does nothing but take inputs and mix them together then feed the result to the soundcard unit.",31,null],[13,"Oscillator","","This unit generates sine/square/saw/triangle or noise tones.",31,null],[13,"LowPass","","This unit filters sound using a high quality, resonant lowpass filter algorithm but consumes more CPU time.",31,null],[13,"ITLowPass","","This unit filters sound using a resonant lowpass filter algorithm that is used in Impulse Tracker, but with limited Cutoff range (0 to 8060hz).",31,null],[13,"HighPass","","This unit filters sound using a resonant highpass filter algorithm.",31,null],[13,"Echo","","This unit produces an echo on the sound and fades out at the desired rate.",31,null],[13,"Flange","","This unit produces a flange effect on the sound.",31,null],[13,"Distortion","","This unit distorts the sound.",31,null],[13,"Normalize","","This unit normalizes or amplifies the sound to a certain level.",31,null],[13,"Parameq","","This unit attenuates or amplifies a selected frequency range.",31,null],[13,"PitchShift","","This unit bends the pitch of a sound without changing the speed of playback.",31,null],[13,"Chorus","","This unit produces a chorus effect on the sound.",31,null],[13,"VSTPlugin","","This unit allows the use of Steinberg VST plugins",31,null],[13,"WinampPlugin","","This unit allows the use of Nullsoft Winamp plugins",31,null],[13,"ITEcho","","This unit produces an echo on the sound and fades out at the desired rate as is used in Impulse Tracker.",31,null],[13,"Compressor","","This unit implements dynamic compression (linked multichannel, wideband)",31,null],[13,"SFXReverb","","This unit implements SFX reverb",31,null],[13,"LowPassSimple","","This unit filters sound using a simple lowpass with no resonance, but has flexible Cutoff and is fast.",31,null],[13,"Delay","","This unit produces different delays on individual channels of the sound.",31,null],[13,"Tremolo","","This unit produces a tremolo / chopper effect on the sound.",31,null],[13,"LADSPAPlugin","","This unit allows the use of LADSPA standard plugins.",31,null],[13,"HighPassSimple","","This unit filters sound using a simple highpass with no resonance, but has flexible Cutoff and is fast.",31,null],[13,"Hardware","","Offset that platform specific FMOD_HARDWARE DSPs will start at.",31,null],[13,"ForceInt","","Makes sure this enum is signed 32bit.",31,null],[4,"DspOscillator","","Parameter types for the FMOD_DSP_TYPE_OSCILLATOR filter.\nUsed with [`Dsp::set_parameter`](../struct.Dsp.html#method.set_parameter) and [`Dsp::get_parameter`](../struct.Dsp.html#method.get_parameter)",null,null],[13,"Type","","Waveform type. 0 = sine. 1 = square. 2 = sawup. 3 = sawdown. 4 = triangle. 5 = noise.",32,null],[13,"Rate","","Frequency of the sinewave in hz. 1.0 to 22000.0. Default = 220.0.",32,null],[4,"DspLowPass","","Parameter types for the FMOD_DSP_TYPE_LOWPASS filter.\nUsed with [`Dsp::set_parameter`](../struct.Dsp.html#method.set_parameter) and [`Dsp::get_parameter`](../struct.Dsp.html#method.get_parameter)",null,null],[13,"Cutoff","","Lowpass Cutoff frequency in hz. 10.0 to 22000.0. Default = 5000.0.",33,null],[13,"Resonance","","Lowpass resonance Q value. 1.0 to 10.0. Default = 1.0.",33,null],[4,"DspITLowPass","","Parameter types for the FMOD_DSP_TYPE_ITLOWPASS filter.\nThis is different to the default FMOD_DSP_TYPE_ITLOWPASS filter in that it uses a different quality algorithm and is \nthe filter used to produce the correct sounding playback in .IT files.\nFMOD Ex's .IT playback uses this filter.\nUsed with [`Dsp::set_parameter`](../struct.Dsp.html#method.set_parameter) and [`Dsp::get_parameter`](../struct.Dsp.html#method.get_parameter)",null,null],[13,"Cutoff","","Lowpass Cutoff frequency in hz. 1.0 to 22000.0. Default = 5000.0",34,null],[13,"Resonance","","Lowpass resonance Q value. 0.0 to 127.0. Default = 1.0.",34,null],[4,"DspHighPass","","Parameter types for the FMOD_DSP_TYPE_HIGHPASS filter.\nUsed with [`Dsp::set_parameter`](../struct.Dsp.html#method.set_parameter) and [`Dsp::get_parameter`](../struct.Dsp.html#method.get_parameter)",null,null],[13,"Cutoff","","Highpass Cutoff frequency in hz. 1.0 to output 22000.0. Default = 5000.0.",35,null],[13,"Resonance","","Highpass resonance Q value. 1.0 to 10.0. Default = 1.0.",35,null],[4,"DspTypeEcho","","Parameter types for the DspTypeEcho filter.\nUsed with [`Dsp::set_parameter`](../struct.Dsp.html#method.set_parameter) and [`Dsp::get_parameter`](../struct.Dsp.html#method.get_parameter)",null,null],[13,"Delay","","Echo delay in ms. 10 to 5000. Default = 500.",36,null],[13,"DecayRatio","","Echo decay per delay. 0 to 1. 1.0 = No decay, 0.0 = total decay (ie simple 1 line delay). Default = 0.5.",36,null],[13,"MaxChannels","","Maximum channels supported. 0 to 16. 0 = same as fmod's default output polyphony, 1 = mono, 2 = stereo etc. See remarks for more. Default = 0. It is suggested to leave at 0!",36,null],[13,"DryMix","","Volume of original signal to pass to output. 0.0 to 1.0. Default = 1.0.",36,null],[13,"WetMix","","Volume of echo signal to pass to output. 0.0 to 1.0. Default = 1.0.",36,null],[4,"DspDelay","","Parameter types for the FMOD_DSP_TYPE_DELAY filter.\nUsed with [`Dsp::set_parameter`](../struct.Dsp.html#method.set_parameter) and [`Dsp::get_parameter`](../struct.Dsp.html#method.get_parameter)",null,null],[13,"CH0","","Channel #0 Delay in ms. 0  to 10000. Default = 0.",37,null],[13,"CH1","","Channel #1 Delay in ms. 0  to 10000. Default = 0.",37,null],[13,"CH2","","Channel #2 Delay in ms. 0  to 10000. Default = 0.",37,null],[13,"CH3","","Channel #3 Delay in ms. 0  to 10000. Default = 0.",37,null],[13,"CH4","","Channel #4 Delay in ms. 0  to 10000. Default = 0.",37,null],[13,"CH5","","Channel #5 Delay in ms. 0  to 10000. Default = 0.",37,null],[13,"CH6","","Channel #6 Delay in ms. 0  to 10000. Default = 0.",37,null],[13,"CH7","","Channel #7 Delay in ms. 0  to 10000. Default = 0.",37,null],[13,"CH8","","Channel #8 Delay in ms. 0  to 10000. Default = 0.",37,null],[13,"CH9","","Channel #9 Delay in ms. 0  to 10000. Default = 0.",37,null],[13,"CH10","","Channel #10 Delay in ms. 0  to 10000. Default = 0.",37,null],[13,"CH11","","Channel #11 Delay in ms. 0  to 10000. Default = 0.",37,null],[13,"CH12","","Channel #12 Delay in ms. 0  to 10000. Default = 0.",37,null],[13,"CH13","","Channel #13 Delay in ms. 0  to 10000. Default = 0.",37,null],[13,"CH14","","Channel #14 Delay in ms. 0  to 10000. Default = 0.",37,null],[13,"CH15","","Channel #15 Delay in ms. 0  to 10000. Default = 0.",37,null],[13,"MaxDelay","","Maximum delay in ms. 0  to 10000. Default = 10.",37,null],[4,"DspFlange","","Parameter types for the FMOD_DSP_TYPE_FLANGE filter.\nUsed with [`Dsp::set_parameter`](../struct.Dsp.html#method.set_parameter) and [`Dsp::get_parameter`](../struct.Dsp.html#method.get_parameter)",null,null],[13,"DryMix","","Volume of original signal to pass to output. 0.0 to 1.0. Default = 0.45.",38,null],[13,"WetMix","","Volume of flange signal to pass to output. 0.0 to 1.0. Default = 0.55.",38,null],[13,"Depth","","Flange depth (percentage of 40ms delay). 0.01 to 1.0. Default = 1.0.",38,null],[13,"Rate","","Flange speed in hz. 0.0 to 20.0. Default = 0.1.",38,null],[4,"DspTremolo","","Parameter types for the FMOD_DSP_TYPE_TREMOLO filter.\nUsed with [`Dsp::set_parameter`](../struct.Dsp.html#method.set_parameter) and [`Dsp::get_parameter`](../struct.Dsp.html#method.get_parameter)",null,null],[13,"Frequency","","LFO frequency in Hz. 0.1 to 20. Default = 4.",39,null],[13,"Depth","","Tremolo depth. 0 to 1. Default = 0.",39,null],[13,"Shape","","LFO shape morph between triangle and sine. 0 to 1. Default = 0.",39,null],[13,"Skew","","Time-skewing of LFO cycle. -1 to 1. Default = 0.",39,null],[13,"Duty","","LFO on-time. 0 to 1. Default = 0.5.",39,null],[13,"Square","","Flatness of the LFO shape. 0 to 1. Default = 0.",39,null],[13,"Phase","","Instantaneous LFO phase. 0 to 1. Default = 0.",39,null],[13,"Spread","","Rotation / auto-pan effect. -1 to 1. Default = 0.",39,null],[4,"DspDistortion","","Parameter types for the FMOD_DSP_TYPE_DISTORTION filter.\nUsed with [`Dsp::set_parameter`](../struct.Dsp.html#method.set_parameter) and [`Dsp::get_parameter`](../struct.Dsp.html#method.get_parameter)",null,null],[13,"Level","","Distortion value. 0.0 to 1.0. Default = 0.5.",40,null],[13,"Unused","","Useless enum",40,null],[4,"DspNormalize","","Parameter types for the FMOD_DSP_TYPE_NORMALIZE filter.\nUsed with [`Dsp::set_parameter`](../struct.Dsp.html#method.set_parameter) and [`Dsp::get_parameter`](../struct.Dsp.html#method.get_parameter)",null,null],[13,"FadeTime","","Time to ramp the silence to full in ms. 0.0 to 20000.0. Default = 5000.0.",41,null],[13,"Threshold","","Lower volume range threshold to ignore. 0.0 to 1.0. Default = 0.1. Raise higher to stop amplification of very quiet signals.",41,null],[13,"MaxAmp","","Maximum amplification allowed. 1.0 to 100000.0. Default = 20.0. 1.0 = no amplifaction, higher values allow more boost.",41,null],[4,"DspTypeParameq","","Parameter types for the DspTypeParameq filter.\nUsed with [`Dsp::set_parameter`](../struct.Dsp.html#method.set_parameter) and [`Dsp::get_parameter`](../struct.Dsp.html#method.get_parameter)",null,null],[13,"Center","","Frequency center. 20.0 to 22000.0. Default = 8000.0.",42,null],[13,"Bandwidth","","Octave range around the center frequency to filter. 0.2 to 5.0. Default = 1.0.",42,null],[13,"Gain","","Frequency Gain. 0.05 to 3.0. Default = 1.0.",42,null],[4,"DspPitchShift","","Parameter types for the FMOD_DSP_TYPE_PITCHSHIFT filter.\nUsed with [`Dsp::set_parameter`](../struct.Dsp.html#method.set_parameter) and [`Dsp::get_parameter`](../struct.Dsp.html#method.get_parameter)",null,null],[13,"Pitch","","Pitch value. 0.5 to 2.0. Default = 1.0. 0.5 = one octave down, 2.0 = one octave up. 1.0 does not change the pitch.",43,null],[13,"FFTSize","","FFT window size. 256, 512, 1024, 2048, 4096. Default = 1024. Increase this to reduce 'smearing'. This effect is a warbling sound similar to when an mp3 is encoded at very low bitrates.",43,null],[13,"OverLap","","Removed. Do not use. FMOD now uses 4 overlaps and cannot be changed.",43,null],[13,"MaxChannels","","Maximum channels supported. 0 to 16. 0 = same as fmod's default output polyphony, 1 = mono, 2 = stereo etc. See remarks for more. Default = 0. It is suggested to leave at 0!",43,null],[4,"DspChorus","","Parameter types for the FMOD_DSP_TYPE_CHORUS filter.\nUsed with [`Dsp::set_parameter`](../struct.Dsp.html#method.set_parameter) and [`Dsp::get_parameter`](../struct.Dsp.html#method.get_parameter)",null,null],[13,"DryMix","","Volume of original signal to pass to output. 0.0 to 1.0. Default = 0.5.",44,null],[13,"WetMix1","","Volume of 1st chorus tap. 0.0 to 1.0. Default = 0.5.",44,null],[13,"WetMix2","","Volume of 2nd chorus tap. This tap is 90 degrees out of phase of the first tap. 0.0 to 1.0. Default = 0.5.",44,null],[13,"WetMix3","","Volume of 3rd chorus tap. This tap is 90 degrees out of phase of the second tap. 0.0 to 1.0. Default = 0.5.",44,null],[13,"Delay","","Chorus delay in ms. 0.1 to 100.0. Default = 40.0 ms.",44,null],[13,"Rate","","Chorus modulation rate in hz. 0.0 to 20.0. Default = 0.8 hz.",44,null],[13,"Depth","","Chorus modulation depth. 0.0 to 1.0. Default = 0.03.",44,null],[4,"DspITEcho","","Parameter types for the FMOD_DSP_TYPE_ITECHO filter.\nThis is effectively a software based echo filter that emulates the DirectX DMO echo effect. Impulse tracker files can support this, and FMOD will produce the effect on ANY platform, not just those that support DirectX effects!\nUsed with [`Dsp::set_parameter`](../struct.Dsp.html#method.set_parameter) and [`Dsp::get_parameter`](../struct.Dsp.html#method.get_parameter)",null,null],[13,"WetDryMix","","Ratio of wet (processed) signal to dry (unprocessed) signal. Must be in the range from 0.0 through 100.0 (all wet). The default value is 50.",45,null],[13,"FeedBack","","Percentage of output fed back into input, in the range from 0.0 through 100.0. The default value is 50.",45,null],[13,"LeftDelay","","Delay for left channel, in milliseconds, in the range from 1.0 through 2000.0. The default value is 500 ms.",45,null],[13,"RightDelay","","Delay for right channel, in milliseconds, in the range from 1.0 through 2000.0. The default value is 500 ms.",45,null],[13,"PanDelay","","Value that specifies whether to swap left and right delays with each successive echo. The default value is zero, meaning no swap. Possible values are defined as 0.0 (equivalent to FALSE) and 1.0 (equivalent to TRUE). CURRENTLY NOT SUPPORTED.",45,null],[4,"DspCompressor","","Parameter types for the FMOD_DSP_TYPE_COMPRESSOR unit.\nThis is a simple linked multichannel software limiter that is uniform across the whole spectrum.\nUsed with [`Dsp::set_parameter`](../struct.Dsp.html#method.set_parameter) and [`Dsp::get_parameter`](../struct.Dsp.html#method.get_parameter)",null,null],[13,"Threshold","","Threshold level (dB) in the range from -60 through 0. The default value is 0.",46,null],[13,"Attack","","Gain reduction attack time (milliseconds), in the range from 10 through 200. The default value is 50.",46,null],[13,"Release","","Gain reduction release time (milliseconds), in the range from 20 through 1000. The default value is 50.",46,null],[13,"GainMakeup","","Make-up gain (dB) applied after limiting, in the range from 0 through 30. The default value is 0.",46,null],[4,"DspSfxReverb","","Parameter types for the FMOD_DSP_TYPE_SFXREVERB unit.\nUsed with [`Dsp::set_parameter`](../struct.Dsp.html#method.set_parameter) and [`Dsp::get_parameter`](../struct.Dsp.html#method.get_parameter)",null,null],[13,"DryLevel","","Dry Level      : Mix level of dry signal in output in mB. Ranges from -10000.0 to 0.0. Default is 0.",47,null],[13,"Room","","Room           : Room effect level at low frequencies in mB. Ranges from -10000.0 to 0.0. Default is -10000.0.",47,null],[13,"RoomHF","","Room HF        : Room effect high-frequency level re. low frequency level in mB. Ranges from -10000.0 to 0.0. Default is 0.0.",47,null],[13,"DecayTime","","Decay Time     : Reverberation decay time at low-frequencies in seconds. Ranges from 0.1 to 20.0. Default is 1.0.",47,null],[13,"DecayHFRatio","","Decay HF Ratio : High-frequency to low-frequency decay time ratio. Ranges from 0.1 to 2.0. Default is 0.5.",47,null],[13,"ReflectionsLevel","","Reflections    : Early reflections level relative to room effect in mB. Ranges from -10000.0 to 1000.0. Default is -10000.0.",47,null],[13,"ReflectionsDelay","","Reflect Delay  : Delay time of first reflection in seconds. Ranges from 0.0 to 0.3. Default is 0.02.",47,null],[13,"ReverbLevel","","Reverb         : Late reverberation level relative to room effect in mB. Ranges from -10000.0 to 2000.0. Default is 0.0.",47,null],[13,"ReverbDelay","","Reverb Delay   : Late reverberation delay time relative to first reflection in seconds. Ranges from 0.0 to 0.1. Default is 0.04.",47,null],[13,"Diffusion","","Diffusion      : Reverberation diffusion (echo density) in percent. Ranges from 0.0 to 100.0. Default is 100.0.",47,null],[13,"Density","","Density        : Reverberation density (modal density) in percent. Ranges from 0.0 to 100.0. Default is 100.0.",47,null],[13,"HFReference","","HF Reference   : Reference high frequency in Hz. Ranges from 20.0 to 20000.0. Default is 5000.0.",47,null],[13,"RoomLF","","Room LF        : Room effect low-frequency level in mB. Ranges from -10000.0 to 0.0. Default is 0.0.",47,null],[13,"LFReference","","LF Reference   : Reference low-frequency in Hz. Ranges from 20.0 to 1000.0. Default is 250.0.",47,null],[4,"DspLowPassSimple","","Parameter types for the FMOD_DSP_TYPE_LOWPASS_SIMPLE filter.\nThis is a very simple low pass filter, based on two single-pole RC time-constant modules.\nThe emphasis is on speed rather than accuracy, so this should not be used for task requiring critical filtering.\nUsed with [`Dsp::set_parameter`](../struct.Dsp.html#method.set_parameter) and [`Dsp::get_parameter`](../struct.Dsp.html#method.get_parameter)",null,null],[13,"Cutoff","","Lowpass Cutoff frequency in hz. 10.0 to 22000.0. Default = 5000.0",48,null],[13,"Unused","","Useless enum",48,null],[4,"DspHighPassSimple","","Parameter types for the FMOD_DSP_TYPE_HIGHPASS_SIMPLE filter.\nThis is a very simple single-order high pass filter.\nThe emphasis is on speed rather than accuracy, so this should not be used for task requiring critical filtering.\nUsed with [`Dsp::set_parameter`](../struct.Dsp.html#method.set_parameter) and [`Dsp::get_parameter`](../struct.Dsp.html#method.get_parameter)",null,null],[13,"Cutoff","","Highpass cutoff frequency in hz. 10.0 to 22000.0. Default = 1000.0",49,null],[13,"Unused","","Useless enum",49,null],[11,"default","","",7,{"inputs":[],"output":{"name":"fmodtag"}}],[11,"drop","","",50,null],[11,"get_system_object","","",50,null],[11,"release","","",50,null],[11,"play","","",50,null],[11,"play_with_parameters","","",50,null],[11,"play_to_the_end","","",50,null],[11,"set_defaults","","",50,null],[11,"get_defaults","","",50,null],[11,"set_variations","","",50,null],[11,"get_variations","","",50,null],[11,"set_3D_min_max_distance","","",50,null],[11,"get_3D_min_max_distance","","",50,null],[11,"set_3D_cone_settings","","",50,null],[11,"get_3D_cone_settings","","",50,null],[11,"set_3D_custom_rolloff","","",50,null],[11,"get_3D_custom_rolloff","","",50,null],[11,"set_sub_sound","","",50,null],[11,"get_sub_sound","","",50,null],[11,"get_name","","",50,null],[11,"get_length","","",50,null],[11,"get_format","","",50,null],[11,"get_num_sub_sounds","","",50,null],[11,"get_num_tags","","",50,null],[11,"get_tag","","",50,null],[11,"get_open_state","","",50,null],[11,"set_sound_group","","",50,null],[11,"get_sound_group","","",50,null],[11,"get_num_sync_points","","",50,null],[11,"get_sync_point","","",50,null],[11,"get_sync_point_info","","",50,null],[11,"add_sync_point","","",50,null],[11,"delete_sync_point","","",50,null],[11,"set_mode","","",50,null],[11,"get_mode","","",50,null],[11,"set_loop_count","","",50,null],[11,"get_loop_count","","",50,null],[11,"set_loop_points","","",50,null],[11,"get_loop_points","","",50,null],[11,"get_num_channels","","",50,null],[11,"set_music_channel_volume","","",50,null],[11,"get_music_channel_volume","","",50,null],[11,"set_music_speed","","",50,null],[11,"get_music_speed","","",50,null],[11,"set_sub_sound_sentence","","",50,null],[11,"seek_data","","",50,null],[11,"get_memory_info","","",50,null],[11,"lock","","",50,null],[11,"unlock","","",50,null],[11,"set_user_data","","",50,null],[11,"get_user_data","","",50,null],[11,"save_to_wav","","",50,null],[11,"clone","","",0,null],[11,"partial_cmp","","",0,null],[11,"lt","","",0,null],[11,"le","","",0,null],[11,"gt","","",0,null],[11,"ge","","",0,null],[11,"eq","","",0,null],[11,"ne","","",0,null],[11,"fmt","","",0,null],[11,"default","","",0,{"inputs":[],"output":{"name":"speakermixoptions"}}],[11,"drop","","",51,null],[11,"new","","",51,{"inputs":[],"output":{"name":"channel"}}],[11,"release","","",51,null],[11,"get_system_object","","",51,null],[11,"stop","","",51,null],[11,"get_spectrum","","channel_offset:  0/1 -> left channel/right 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Use this for FFT, DSP, compressed sample support, 2D multi-speaker support and other software related features.",null,null],[17,"CREATESTREAM","","Decompress at runtime, streaming from the source provided (ie from disk). Overrides CREATESAMPLE and CREATECOMPRESSEDSAMPLE. Note a stream can only be played once at a time due to a stream only having 1 stream buffer and file handle. Open multiple streams to have them play concurrently.",null,null],[17,"CREATESAMPLE","","Decompress at loadtime, decompressing or decoding whole file into memory as the target sample format (ie PCM). Fastest for SOFTWARE based playback and most flexible.",null,null],[17,"CREATECOMPRESSEDSAMPLE","","Load MP2/MP3/IMAADPCM/CELT/Vorbis/AT9 or XMA into memory and leave it compressed. CELT/Vorbis/AT9 encoding only supported in the FSB file format. During playback the FMOD software mixer will decode it in realtime as a 'compressed sample'. Can only be used in combination with SOFTWARE. Overrides CREATESAMPLE. If the sound data is not one of the supported formats, it will behave as if it was created with CREATESAMPLE and decode the sound into PCM.",null,null],[17,"OPENUSER","","Opens a user created static sample or stream. Use CREATESOUNDEXINFO to specify format and/or read callbacks. If a user created 'sample' is created with no read callback, the sample will be empty. Use [`Sound::lock`](../struct.Sound.html#method.lock) and [`Sound::unlock`](../struct.Sound.html#method.unlock) to place sound data into the sound if this is the case.",null,null],[17,"OPENMEMORY","",""name_or_data" will be interpreted as a pointer to memory instead of filename for creating sounds. Use CREATESOUNDEXINFO to specify length. If used with CREATESAMPLE or CREATECOMPRESSEDSAMPLE, FMOD duplicates the memory into its own buffers. Your own buffer can be freed after open. If used with CREATESTREAM, FMOD will stream out of the buffer whose pointer you passed in. In this case, your own buffer should not be freed until you have finished with and released the stream.",null,null],[17,"OPENMEMORY_POINT","",""name_or_data" will be interpreted as a pointer to memory instead of filename for creating sounds. Use CREATESOUNDEXINFO to specify length. This differs to OPENMEMORY in that it uses the memory as is, without duplicating the memory into its own buffers. For Wii/PSP HARDWARE supports this flag for the GCADPCM/VAG formats. On other platforms SOFTWARE must be used, as sound hardware on the other platforms (ie PC) cannot access main ram. Cannot be freed after open, only after [`Sound::release`](../struct.Sound.html#method.release). Will not work if the data is compressed and CREATECOMPRESSEDSAMPLE is not used.",null,null],[17,"OPENRAW","","Will ignore file format and treat as raw pcm. Use CREATESOUNDEXINFO to specify format. Requires at least defaultfrequency, numchannels and format to be specified before it will open. Must be little endian data.",null,null],[17,"OPENONLY","","Just open the file, dont prebuffer or read. Good for fast opens for info, or when sound::readData is to be used.",null,null],[17,"ACCURATETIME","","For [`Sys::create_sound`](../struct.Sys.html#method.create_sound) - for accurate [`Sound::get_length`](../struct.Sound.html#method.get_length) / [`Channel::set_position`](../struct.Channel.html#method.set_position) on VBR MP3, and MOD/S3M/XM/IT/MIDI files. Scans file first, so takes longer to open. OPENONLY does not affect this.",null,null],[17,"MPEGSEARCH","","For corrupted / bad MP3 files. This will search all the way through the file until it hits a valid MPEG header. Normally only searches for 4k.",null,null],[17,"NONBLOCKING","","For opening sounds and getting streamed subsounds (seeking) asyncronously. Use [`Sound::get_open_state`](../struct.Sound.html#method.get_open_state) to poll the state of the sound as it opens or retrieves the subsound in the background.",null,null],[17,"UNIQUE","","Unique sound, can only be played one at a time",null,null],[17,"_3D_HEADRELATIVE","","Make the sound's position, velocity and orientation relative to the listener.",null,null],[17,"_3D_WORLDRELATIVE","","Make the sound's position, velocity and orientation absolute (relative to the world). (DEFAULT)",null,null],[17,"_3D_INVERSEROLLOFF","","This sound will follow the inverse rolloff model where mindistance = full volume, maxdistance = where sound stops attenuating, and rolloff is fixed according to the global rolloff factor. (DEFAULT)",null,null],[17,"_3D_LINEARROLLOFF","","This sound will follow a linear rolloff model where mindistance = full volume, maxdistance = silence. Rolloffscale is ignored.",null,null],[17,"_3D_LINEARSQUAREROLLOFF","","This sound will follow a linear-square rolloff model where mindistance = full volume, maxdistance = silence. Rolloffscale is ignored.",null,null],[17,"_3D_CUSTOMROLLOFF","","This sound will follow a rolloff model defined by [`Sound::set_3D_custom_rolloff`](../struct.Sound.html#method.set_3D_custom_rolloff) / [`Channel::set_3D_custom_rolloff`](../struct.Channel.html#method.set_3D_custom_rolloff).",null,null],[17,"_3D_IGNOREGEOMETRY","","Is not affect by geometry occlusion. If not specified in [`Sound::set_mode`](../struct.Sound.html#method.set_mode), or [`Channel::set_mode`](../struct.Channel.html#method.set_mode), the flag is cleared and it is affected by geometry again.",null,null],[17,"UNICODE","","Filename is double-byte unicode.",null,null],[17,"IGNORETAGS","","Skips id3v2/asf/etc tag checks when opening a sound, to reduce seek/read overhead when opening files (helps with CD performance).",null,null],[17,"LOWMEM","","Removes some features from samples to give a lower memory overhead, like [`Sound::get_name`](../struct.Sound.html#method.get_name). See remarks.",null,null],[17,"LOADSECONDARYRAM","","Load sound into the secondary RAM of supported platform. On PS3, sounds will be loaded into RSX/VRAM.",null,null],[17,"VIRTUAL_PLAYFROMSTART","","For sounds that start virtual (due to being quiet or low importance), instead of swapping back to audible, and playing at the correct offset according to time, this flag makes the sound play from the start.",null,null],[17,"INIT_NORMAL","","All platforms - Initialize normally",null,null],[17,"INIT_STREAM_FROM_UPDATE","","All platforms - No stream thread is created internally. Streams are driven from [`Sys::update`](../struct.Sys.html#method.update). Mainly used with non-realtime outputs.",null,null],[17,"INIT_3D_RIGHTHANDED","","All platforms - FMOD will treat +X as right, +Y as up and +Z as backwards (towards you).",null,null],[17,"INIT_SOFTWARE_DISABLE","","All platforms - Disable software mixer to save memory. Anything created with SOFTWARE will fail and DSP will not work.",null,null],[17,"INIT_OCCLUSION_LOWPASS","","All platforms - All SOFTWARE (and HARDWARE on 3DS and NGP) with 3D based voices will add a software lowpass filter effect into the DSP chain which is automatically used when [`Channel::set_3D_occlusion`](../struct.Channel.html#method.set_3D_occlusion) is used or the geometry API.",null,null],[17,"INIT_HRTF_LOWPASS","","All platforms - All SOFTWARE (and HARDWARE on 3DS and NGP) with 3D based voices will add a software lowpass filter effect into the DSP chain which causes sounds to sound duller when the sound goes behind the listener. Use [`Sys::set_advanced_settings`](../struct.Sys.html#method.set_advanced_settings) to adjust Cutoff frequency.",null,null],[17,"INIT_DISTANCE_FILTERING","","All platforms - All SOFTWARE with 3D based voices will add a software lowpass and highpass filter effect into the DSP chain which will act as a distance-automated bandpass filter. Use [`Sys::set_advanced_settings`](../struct.Sys.html#method.set_advanced_settings) to adjust the center frequency.",null,null],[17,"INIT_REVERB_PREALLOCBUFFERS","","All platforms - FMOD Software reverb will preallocate enough buffers for reverb per channel, rather than allocating them and freeing them at runtime.",null,null],[17,"INIT_ENABLE_PROFILE","","All platforms - Enable TCP/IP based host which allows FMOD Designer or FMOD Profiler to connect to it, and view memory, CPU and the DSP network graph in real-time.",null,null],[17,"INIT_VOL0_BECOMES_VIRTUAL","","All platforms - Any sounds that are 0 volume will go virtual and not be processed except for having their positions updated virtually. Use [`Sys::set_advanced_settings`](../struct.Sys.html#method.set_advanced_settings) to adjust what volume besides zero to switch to virtual at.",null,null],[17,"INIT_WASAPI_EXCLUSIVE","","Win32 Vista only - for WASAPI output - Enable exclusive access to hardware, lower latency at the expense of excluding other applications from accessing the audio hardware.",null,null],[17,"INIT_PS3_PREFERDTS","","PS3 only - Prefer DTS over Dolby Digital if both are supported. Note: 8 and 6 channel LPCM is always preferred over both DTS and Dolby Digital.",null,null],[17,"INIT_PS3_FORCE2CHLPCM","","PS3 only - Force PS3 system output mode to 2 channel LPCM.",null,null],[17,"INIT_DISABLEDOLBY","","Wii / 3DS - Disable Dolby Pro Logic surround.  will be set to STEREO even if user has selected surround in the system settings.",null,null],[17,"INIT_SYSTEM_MUSICMUTENOTPAUSE","","Xbox 360 / PS3 - The "music" channelgroup which by default pauses when custom 360 dashboard / PS3 BGM music is played, can be changed to mute (therefore continues playing) instead of pausing, by using this flag.",null,null],[17,"INIT_SYNCMIXERWITHUPDATE","","Win32/Wii/PS3/Xbox/Xbox 360 - FMOD Mixer thread is woken up to do a mix when [`Sys::update`](../struct.Sys.html#method.update) is called rather than waking periodically on its own timer.",null,null],[17,"INIT_GEOMETRY_USECLOSEST","","All platforms - With the geometry engine, only process the closest polygon rather than accumulating all polygons the sound to listener line intersects.",null,null],[17,"INIT_DISABLE_MYEARS_AUTODETECT","","Win32 - Disables automatic setting of of _STEREO to _MYEARS if the MyEars profile exists on the PC. MyEars is HRTF 7.1 downmixing through headphones.",null,null],[17,"INIT_PS3_DISABLEDTS","","PS3 only - Disable DTS output mode selection",null,null],[17,"INIT_PS3_DISABLEDOLBYDIGITAL","","PS3 only - Disable Dolby Digital output mode selection",null,null],[17,"INIT_7POINT1_DOLBYMAPPING","","PS3/PS4 only - FMOD uses the WAVEFORMATEX Microsoft 7.1 speaker mapping where the last 2 pairs of speakers are 'rears' then 'sides', but on PS3/PS4 these are mapped to 'surrounds' and 'backs'. Use this flag to swap fmod's last 2 pair of speakers on PS3/PS4 to avoid needing to do a special case for these platforms.",null,null],[17,"TIMEUNIT_MS","","Milliseconds.",null,null],[17,"TIMEUNIT_PCM","","PCM samples, related to milliseconds * samplerate / 1000.",null,null],[17,"TIMEUNIT_PCMBYTES","","Bytes, related to PCM samples * channels * datawidth (ie 16bit = 2 bytes).",null,null],[17,"TIMEUNIT_RAWBYTES","","Raw file bytes of (compressed) sound data (does not include headers). Only used by [`Sound::get_length`](../struct.Sound.html#method.get_length) and [`Channel::get_position`](../struct.Channel.html#method.get_position).",null,null],[17,"TIMEUNIT_PCMFRACTION","","Fractions of 1 PCM sample. Unsigned int range 0 to 0xFFFFFFFF. Used for sub-sample granularity for DSP purposes.",null,null],[17,"TIMEUNIT_MODORDER","","MOD/S3M/XM/IT. Order in a sequenced module format. Use [`Sound::get_format`](../struct.Sound.html#method.get_format) to determine the PCM format being decoded to.",null,null],[17,"TIMEUNIT_MODROW","","MOD/S3M/XM/IT. Current row in a sequenced module format. [`Sound::get_length`](../struct.Sound.html#method.get_length) will return the number of rows in the currently playing or seeked to pattern.",null,null],[17,"TIMEUNIT_MODPATTERN","","MOD/S3M/XM/IT. Current pattern in a sequenced module format. [`Sound::get_length`](../struct.Sound.html#method.get_length) will return the number of patterns in the song and [`Channel::get_position`](../struct.Channel.html#method.get_position) will return the currently playing pattern.",null,null],[17,"TIMEUNIT_SENTENCE_MS","","Currently playing subsound in a sentence time in milliseconds.",null,null],[17,"TIMEUNIT_SENTENCE_PCM","","Currently playing subsound in a sentence time in PCM Samples, related to milliseconds * samplerate / 1000.",null,null],[17,"TIMEUNIT_SENTENCE_PCMBYTES","","Currently playing subsound in a sentence time in bytes, related to PCM samples * channels * datawidth (ie 16bit = 2 bytes).",null,null],[17,"TIMEUNIT_SENTENCE","","Currently playing sentence index according to the channel.",null,null],[17,"TIMEUNIT_SENTENCE_SUBSOUND","","Currently playing subsound index in a sentence.",null,null],[17,"TIMEUNIT_BUFFERED","","Time value as seen by buffered stream. This is always ahead of audible time, and is only used for processing.",null,null],[17,"MEMBITS_OTHER","","Memory not accounted for by other types",null,null],[17,"MEMBITS_STRING","","String data",null,null],[17,"MEMBITS_SYSTEM","","[`Sys`](../struct.Sys.html) object and various internals",null,null],[17,"MEMBITS_PLUGINS","","Plugin objects and internals",null,null],[17,"MEMBITS_OUTPUT","","Output module object and internals",null,null],[17,"MEMBITS_CHANNEL","","[`Channel`](../struct.Channel.html) related memory",null,null],[17,"MEMBITS_CHANNELGROUP","","[`ChannelGroup`](../struct.ChannelGroup.html) objects and internals",null,null],[17,"MEMBITS_CODEC","","Codecs allocated for streaming",null,null],[17,"MEMBITS_FILE","","Codecs allocated for streaming",null,null],[17,"MEMBITS_SOUND","","[`Sound`](../struct.Sound.html) objects and internals",null,null],[17,"MEMBITS_SOUND_SECONDARYRAM","","Sound data stored in secondary RAM",null,null],[17,"MEMBITS_SOUNDGROUP","","[`SoundGroup`](../struct.SoundGroup.html) objects and internals",null,null],[17,"MEMBITS_STREAMBUFFER","","Stream buffer memory",null,null],[17,"MEMBITS_DSPCONNECTION","","[`DspConnection`](../struct.DspConnection.html) objects and internals",null,null],[17,"MEMBITS_DSP","","[`Dsp`](../struct.Dsp.html) implementation objects",null,null],[17,"MEMBITS_DSPCODEC","","Realtime file format decoding [`Dsp`](../struct.Dsp.html) objects",null,null],[17,"MEMBITS_PROFILE","","Profiler memory footprint.",null,null],[17,"MEMBITS_RECORDBUFFER","","Buffer used to store recorded data from microphone",null,null],[17,"MEMBITS_REVERB","","[`Reverb`](../struct.Reverb.html) implementation objects",null,null],[17,"MEMBITS_REVERBCHANNELPROPS","","Reverb channel properties structs",null,null],[17,"MEMBITS_GEOMETRY","","[`Geometry`](../struct.Geometry.html) objects and internals",null,null],[17,"MEMBITS_SYNCPOINT","","Sync point memory.",null,null],[17,"MEMBITS_ALL","","All memory used by FMOD Ex",null,null],[17,"EVENT_MEMBITS_EVENTSYSTEM","","EventSystem and various internals",null,null],[17,"EVENT_MEMBITS_MUSICSYSTEM","","MusicSystem and various internals",null,null],[17,"EVENT_MEMBITS_FEV","","Definition of objects contained in all loaded projects e.g. events, groups, categories",null,null],[17,"EVENT_MEMBITS_MEMORYFSB","","Data loaded with preloadFSB",null,null],[17,"EVENT_MEMBITS_EVENTPROJECT","","EventProject objects and internals",null,null],[17,"EVENT_MEMBITS_EVENTGROUPI","","EventGroup objects and internals",null,null],[17,"EVENT_MEMBITS_SOUNDBANKCLASS","","Objects used to manage wave banks",null,null],[17,"EVENT_MEMBITS_SOUNDBANKLIST","","Data used to manage lists of wave bank usage",null,null],[17,"EVENT_MEMBITS_STREAMINSTANCE","","Stream objects and internals",null,null],[17,"EVENT_MEMBITS_SOUNDDEFCLASS","","Sound definition objects",null,null],[17,"EVENT_MEMBITS_SOUNDDEFDEFCLASS","","Sound definition static data objects",null,null],[17,"EVENT_MEMBITS_SOUNDDEFPOOL","","Sound definition pool data",null,null],[17,"EVENT_MEMBITS_REVERBDEF","","Reverb definition objects",null,null],[17,"EVENT_MEMBITS_EVENTREVERB","","Reverb objects",null,null],[17,"EVENT_MEMBITS_USERPROPERTY","","User property objects",null,null],[17,"EVENT_MEMBITS_EVENTINSTANCE","","Event instance base objects",null,null],[17,"EVENT_MEMBITS_EVENTINSTANCE_COMPLEX","","Complex event instance objects",null,null],[17,"EVENT_MEMBITS_EVENTINSTANCE_SIMPLE","","Simple event instance objects",null,null],[17,"EVENT_MEMBITS_EVENTINSTANCE_LAYER","","Event layer instance objects",null,null],[17,"EVENT_MEMBITS_EVENTINSTANCE_SOUND","","Event sound instance objects",null,null],[17,"EVENT_MEMBITS_EVENTENVELOPE","","Event envelope objects",null,null],[17,"EVENT_MEMBITS_EVENTENVELOPEDEF","","Event envelope definition objects",null,null],[17,"EVENT_MEMBITS_EVENTPARAMETER","","Event parameter objects",null,null],[17,"EVENT_MEMBITS_EVENTCATEGORY","","Event category objects",null,null],[17,"EVENT_MEMBITS_EVENTENVELOPEPOINT","","Event envelope point objects",null,null],[17,"EVENT_MEMBITS_EVENTINSTANCEPOOL","","Event instance pool data",null,null],[17,"EVENT_MEMBITS_ALL","","All memory used by FMOD Event System",null,null],[17,"EVENT_MEMBITS_EVENTINSTANCE_GROUP","","All event instance memory",null,null],[17,"EVENT_MEMBITS_SOUNDDEF_GROUP","","All sound definition memory",null,null]],"paths":[[3,"SpeakerMixOptions"],[3,"ReverbChannelProperties"],[3,"Guid"],[3,"SoftwareFormat"],[3,"AdvancedSettings"],[3,"CreateSoundexInfo"],[3,"MemoryUsageDetails"],[3,"FmodTag"],[3,"DspParameterDesc"],[3,"DspDescription"],[3,"DspState"],[3,"ReverbProperties"],[3,"Vector"],[4,"SeekStyle"],[4,"Result"],[4,"SpeakerMapType"],[4,"SoundFormat"],[4,"SoundType"],[4,"TagType"],[4,"TagDataType"],[4,"ChannelIndex"],[4,"DspFftWindow"],[4,"DelayType"],[4,"OutputType"],[4,"Speaker"],[4,"SpeakerMode"],[4,"DspResampler"],[4,"PluginType"],[4,"OpenState"],[4,"SystemCallbackType"],[4,"SoundGroupBehavior"],[4,"DspType"],[4,"DspOscillator"],[4,"DspLowPass"],[4,"DspITLowPass"],[4,"DspHighPass"],[4,"DspTypeEcho"],[4,"DspDelay"],[4,"DspFlange"],[4,"DspTremolo"],[4,"DspDistortion"],[4,"DspNormalize"],[4,"DspTypeParameq"],[4,"DspPitchShift"],[4,"DspChorus"],[4,"DspITEcho"],[4,"DspCompressor"],[4,"DspSfxReverb"],[4,"DspLowPassSimple"],[4,"DspHighPassSimple"],[3,"Sound"],[3,"Channel"],[3,"ChannelGroup"],[3,"SoundGroup"],[3,"UserData"],[3,"Sys"],[3,"Dsp"],[3,"DspConnection"],[3,"Geometry"],[3,"Reverb"],[3,"FmodFile"],[3,"Mode"],[3,"TimeUnit"],[3,"FmodCaps"],[3,"PluginHandle"],[3,"InitFlag"],[3,"MemoryBits"],[3,"EventMemoryBits"]]};
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