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Diffstat (limited to 'Runtime/Audio/ScriptBindings')
| -rw-r--r-- | Runtime/Audio/ScriptBindings/AudioBindings.txt | 996 |
1 files changed, 996 insertions, 0 deletions
diff --git a/Runtime/Audio/ScriptBindings/AudioBindings.txt b/Runtime/Audio/ScriptBindings/AudioBindings.txt new file mode 100644 index 0000000..57ed241 --- /dev/null +++ b/Runtime/Audio/ScriptBindings/AudioBindings.txt @@ -0,0 +1,996 @@ +C++RAW + + +#include "UnityPrefix.h" +#include "Configuration/UnityConfigure.h" +#include "Runtime/Mono/MonoManager.h" +#include "Runtime/Graphics/Transform.h" +#include "Runtime/Utilities/PathNameUtility.h" +#include "Runtime/Profiler/ProfilerHistory.h" +#include "Runtime/Allocator/MemoryManager.h" +#include "Runtime/Audio/AudioClip.h" +#if ENABLE_AUDIO +#include "Runtime/Audio/AudioSource.h" +#include "Runtime/Audio/AudioListener.h" +#include "Runtime/Audio/AudioManager.h" +#include "Runtime/Audio/AudioReverbZone.h" +#include "Runtime/Audio/AudioReverbFilter.h" +#include "Runtime/Audio/AudioHighPassFilter.h" +#include "Runtime/Audio/AudioLowPassFilter.h" +#include "Runtime/Audio/AudioChorusFilter.h" +#include "Runtime/Audio/AudioDistortionFilter.h" +#include "Runtime/Audio/AudioEchoFilter.h" +#endif +#include "Runtime/Animation/Animation.h" +#include "Runtime/Utilities/PlayerPrefs.h" +#include "Runtime/Misc/BuildSettings.h" +#include "Runtime/Scripting/ScriptingUtility.h" +#include "Runtime/Scripting/ScriptingExportUtility.h" +#include "Runtime/Scripting/GetComponent.h" +#include "Runtime/Scripting/Backend/ScriptingBackendApi.h" +#include "Runtime/Scripting/Backend/ScriptingTypeRegistry.h" +#include "Runtime/Scripting/Scripting.h" + +using namespace Unity; + +/* + Mono defines a bool as either 1 or 2 bytes. + On windows a bool on the C++ side needs to be 2 bytes. + We use the typemap to map bool's to short's. + When using the C++ keyword and you want to export a bool value + to mono you have to use a short on the C++ side. +*/ + + +void PauseEditor (); +using namespace std; + +CSRAW +using System; +using System.Runtime.CompilerServices; +using System.Runtime.InteropServices; +using System.Collections; +using System.Collections.Generic; +using UnityEngineInternal; + +namespace UnityEngine +{ + + + +// These are speaker types defined for use with [[AudioSettings.speakerMode]]. +CONDITIONAL ENABLE_AUDIO_FMOD +ENUM AudioSpeakerMode + // Channel count is unaffected. + Raw = 0, + // Channel count is set to 1. The speakers are monaural. + Mono = 1, + // Channel count is set to 2. The speakers are stereo. This is the editor default. + Stereo = 2, + // Channel count is set to 4. 4 speaker setup. This includes front left, front right, rear left, rear right. + Quad = 3, + // Channel count is set to 5. 5 speaker setup. This includes front left, front right, center, rear left, rear right. + Surround = 4, + // Channel count is set to 6. 5.1 speaker setup. This includes front left, front right, center, rear left, rear right and a subwoofer. + Mode5point1 = 5, + // Channel count is set to 8. 7.1 speaker setup. This includes front left, front right, center, rear left, rear right, side left, side right and a subwoofer. + Mode7point1 = 6, + // Channel count is set to 2. Stereo output, but data is encoded in a way that is picked up by a Prologic/Prologic2 decoder and split into a 5.1 speaker setup. + Prologic = 7 +END + +// Controls the global audio settings from script. +CONDITIONAL ENABLE_AUDIO_FMOD +CLASS AudioSettings + // Returns the speaker mode capability of the current audio driver. (RO) + CUSTOM_PROP static AudioSpeakerMode driverCaps + { + return GetAudioManager().GetSpeakerModeCaps(); + } + + // Sets or gets the current speaker mode. Default is 2 channel stereo. + CUSTOM_PROP static AudioSpeakerMode speakerMode + { + return GetAudioManager().GetSpeakerMode(); + } + { + GetAudioManager().SetSpeakerMode(value); + } + + // Returns the current time of the audio system. This is based on the number of samples the audio system processes and is therefore more exact than the time obtained via the Time.time property. + // It is constant while Unity is paused. + THREAD_SAFE CUSTOM_PROP static double dspTime + { + return GetAudioManager().GetDSPTime(); + } + + // Get and set the mixer's current output rate. + CUSTOM_PROP static int outputSampleRate + { + int sampleRate; + GetAudioManager().GetFMODSystem()->getSoftwareFormat( + &sampleRate, + NULL, + NULL, + NULL, + NULL, + NULL + ); + return sampleRate; + } + { + int currentSampleRate = AudioSettings_Get_Custom_PropOutputSampleRate(); + if (currentSampleRate != value) + { + GetAudioManager().CloseFMOD(); + FMOD_RESULT result = GetAudioManager().GetFMODSystem()->setSoftwareFormat( + value, + FMOD_SOUND_FORMAT_PCM16, + 0, + 8, + FMOD_DSP_RESAMPLER_LINEAR + ); + if (result != FMOD_OK) + { + ErrorString(Format("%dHz is an invalid output samplerate for this platform", value)); + } + + GetAudioManager().ReloadFMODSounds(); + } + } + + // Get or set the mixer's buffer size in samples. + CUSTOM static void SetDSPBufferSize(int bufferLength, int numBuffers) + { + bufferLength = clamp(bufferLength, 64, 4096); + + GetAudioManager().CloseFMOD(); + FMOD_RESULT result = GetAudioManager().GetFMODSystem()->setDSPBufferSize(bufferLength, numBuffers); + if (result != FMOD_OK) + { + ErrorString(Format("DSP ringbuffer of %d samples (x %d) is invalid for this platform", bufferLength, numBuffers)); + } + GetAudioManager().ReloadFMODSounds(); + } + + // Get or set the mixer's buffer size in samples. + CUSTOM static void GetDSPBufferSize(out int bufferLength, out int numBuffers) + { + FMOD_RESULT result = GetAudioManager().GetFMODSystem()->getDSPBufferSize((unsigned int*)bufferLength, numBuffers); + FMOD_ASSERT( result ); + } + + +END + +// Type of the imported(native) data +ENUM public AudioType + // 3rd party / unknown plugin format. + UNKNOWN = 0, + //acc - not supported + ACC = 1, /* [Unity] Not supported/used. But kept here to keep the order of the enum in sync. */ + //aiff + AIFF = 2, +// ASF = 3, /* Microsoft Advanced Systems Format (ie WMA/ASF/WMV). */ +// AT3 = 4, /* Sony ATRAC 3 format */ +// CDDA = 5, /* Digital CD audio. */ +// DLS = 6, /* Sound font / downloadable sound bank. */ +// FLAC = 7, /* FLAC lossless codec. */ +// FSB = 8, /* FMOD Sample Bank. */ + //game cube ADPCM + GCADPCM = 9, + //impulse tracker + IT = 10, +// MIDI = 11, /* MIDI. */ + //Protracker / Fasttracker MOD. + MOD = 12, + //MP2/MP3 MPEG. + MPEG = 13, + //ogg vorbis + OGGVORBIS = 14, +// PLAYLIST = 15, /* Information only from ASX/PLS/M3U/WAX playlists */ +// RAW = 16, /* Raw PCM data. */ + // ScreamTracker 3. + S3M = 17, +// SF2 = 18, /* Sound font 2 format. */ +// USER = 19, /* User created sound. */ + //Microsoft WAV. + WAV = 20, + // FastTracker 2 XM. + XM = 21, + // Xbox360 XMA + XMA = 22, +// VAG = 23, /* PlayStation 2 / PlayStation Portable adpcm VAG format. */ + //iPhone hardware decoder, supports AAC, ALAC and MP3. Extracodecdata is a pointer to an FMOD_AUDIOQUEUE_EXTRACODECDATA structure. + AUDIOQUEUE = 24, +// XWMA = 25, /* Xbox360 XWMA */ +// BCWAV = 26, /* 3DS BCWAV container format for DSP ADPCM and PCM */ +// AT9 = 27, /* NGP ATRAC 9 format */ +END + +// A container for audio data. +CONDITIONAL ENABLE_AUDIO +CLASS AudioClip : Object + + // Check if reading of the audioclip is allowed by crossdomain security and throw if not + C++RAW + static void CheckReadAllowedAndThrow(AudioClip *clip) + { +#if ENABLE_MONO && ENABLE_SECURITY && ENABLE_WWW + if ( clip&&!clip->GetReadAllowed() ) + Scripting::RaiseSecurityException("No read access to the audioclip data: %s", clip->GetName()); +#endif + } + + // The length of the audio clip in seconds (RO) + AUTO_PROP float length GetLengthSec + + // The length of the audio clip in samples (RO) + // Prints how many samples the attached audio source has + // + AUTO_PROP int samples GetSampleCount + + // Channels in audio clip (RO) + AUTO_PROP int channels GetChannelCount + + // Sample frequency (RO) + AUTO_PROP int frequency GetFrequency + + // Is a streamed audio clip ready to play? (RO) + AUTO_PROP bool isReadyToPlay ReadyToPlay + + CONDITIONAL ENABLE_AUDIO_FMOD + // Fills an array with sample data from the clip. The samples are floats ranging from -1.0f to 1.0f. The sample count is determined by the length of the float array. + CUSTOM void GetData(float[] data, int offsetSamples) + { + CheckReadAllowedAndThrow(self); + self->GetData(&Scripting::GetScriptingArrayElement<float>(data, 0), GetScriptingArraySize (data) / self->GetChannelCount(), offsetSamples); + } + + CONDITIONAL ENABLE_AUDIO_FMOD + // Set sample data in a clip. The samples should be floats ranging from 0.0f to 1.0f (exceeding these limits will lead to artifacts and undefined behaviour). + CUSTOM void SetData(float[] data, int offsetSamples) + { + self->SetData(&Scripting::GetScriptingArrayElement<float>(data, 0), GetScriptingArraySize (data) / self->GetChannelCount(), offsetSamples); + } + + CONDITIONAL ENABLE_AUDIO_FMOD + /// *listonly* + CSRAW public static AudioClip Create(string name, int lengthSamples, int channels, int frequency, bool _3D, bool stream) + { + AudioClip clip = Create (name, lengthSamples, channels, frequency, _3D, stream, null, null); + return clip; + } + + CONDITIONAL ENABLE_AUDIO_FMOD + /// *listonly* + CSRAW public static AudioClip Create(string name, int lengthSamples, int channels, int frequency, bool _3D, bool stream, PCMReaderCallback pcmreadercallback) + { + AudioClip clip = Create (name, lengthSamples, channels, frequency, _3D, stream, pcmreadercallback, null); + return clip; + } + + CONDITIONAL ENABLE_AUDIO_FMOD + // Creates a user AudioClip with a name and with the given length in samples, channels and frequency. + CSRAW public static AudioClip Create(string name, int lengthSamples, int channels, int frequency, bool _3D, bool stream, PCMReaderCallback pcmreadercallback, PCMSetPositionCallback pcmsetpositioncallback) + { + if(name == null) throw new NullReferenceException(); + + AudioClip clip = Construct_Internal(); + if ( pcmreadercallback != null) + clip.m_PCMReaderCallback += pcmreadercallback; + if ( pcmsetpositioncallback != null) + clip.m_PCMSetPositionCallback += pcmsetpositioncallback; + + clip.Init_Internal( name, lengthSamples, channels, frequency, _3D, stream ); + + return clip; + } + + /// *listonly* + CONDITIONAL ENABLE_AUDIO_FMOD + CSRAW public delegate void PCMReaderCallback(float[] data); + CONDITIONAL ENABLE_AUDIO_FMOD + CSRAW private event PCMReaderCallback m_PCMReaderCallback = null; + /// *listonly* + CONDITIONAL ENABLE_AUDIO_FMOD + CSRAW public delegate void PCMSetPositionCallback(int position); + CONDITIONAL ENABLE_AUDIO_FMOD + CSRAW private event PCMSetPositionCallback m_PCMSetPositionCallback = null; + + CONDITIONAL ENABLE_AUDIO_FMOD + CSRAW private void InvokePCMReaderCallback_Internal(float[] data) + { + if (m_PCMReaderCallback != null) + m_PCMReaderCallback( data ); + } + + CONDITIONAL ENABLE_AUDIO_FMOD + CSRAW private void InvokePCMSetPositionCallback_Internal(int position) + { + if (m_PCMSetPositionCallback != null) + m_PCMSetPositionCallback( position ); + } + + CONDITIONAL ENABLE_AUDIO_FMOD + CUSTOM private static AudioClip Construct_Internal() + { + AudioClip* clip = NEW_OBJECT(AudioClip); + return Scripting::ScriptingWrapperFor ( clip ); + } + + CONDITIONAL ENABLE_AUDIO_FMOD + CUSTOM private void Init_Internal(string name, int lengthSamples, int channels, int frequency, bool _3D, bool stream) + { + self->CreateUserSound( name, lengthSamples, channels, frequency, _3D, stream ); + } + +END + +// Describes when an [[AudioSource]] or [[AudioListener]] is updated. +ENUM AudioVelocityUpdateMode + // Updates the source or listener in the fixed update loop if it is attached to a [[Rigidbody]], dynamic otherwise. + Auto = 0, + // Updates the source or listener in the fixed update loop. + Fixed = 1, + // Updates the source or listener in the dynamic update loop. + Dynamic = 2, +END + + +// Representation of a listener in 3D space. +CONDITIONAL ENABLE_AUDIO +CLASS AudioListener : Behaviour + // Controls the game sound volume (0.0 to 1.0) + CUSTOM_PROP static float volume { return GetAudioManager ().GetVolume (); } { GetAudioManager ().SetVolume (value); } + + // The paused state of the audio. If set to True, the listener will not generate sound. + CUSTOM_PROP static bool pause { return GetAudioManager ().GetPause (); } { GetAudioManager ().SetPause (value); } + + // This lets you set whether the Audio Listener should be updated in the fixed or dynamic update. + + AUTO_PROP AudioVelocityUpdateMode velocityUpdateMode GetVelocityUpdateMode SetVelocityUpdateMode + + CONDITIONAL ENABLE_AUDIO_FMOD + CUSTOM static private void GetOutputDataHelper(float[] samples, int channel) + { + FMOD::ChannelGroup* channelGroup; + FMOD_RESULT result = GetAudioManager().GetFMODSystem()->getMasterChannelGroup(&channelGroup); + + if (result == FMOD_OK && channelGroup) + { + channelGroup->getWaveData(Scripting::GetScriptingArrayStart<float>(samples), GetScriptingArraySize(samples), channel); + } + } + + CONDITIONAL ENABLE_AUDIO_FMOD + CUSTOM static private void GetSpectrumDataHelper(float[] samples, int channel, FFTWindow window) + { + FMOD::ChannelGroup* channelGroup; + FMOD_RESULT result = GetAudioManager().GetFMODSystem()->getMasterChannelGroup(&channelGroup); + + if (result == FMOD_OK && channelGroup) + { + channelGroup->getSpectrum(Scripting::GetScriptingArrayStart<float>(samples), GetScriptingArraySize(samples), channel, (FMOD_DSP_FFT_WINDOW)window); + } + } + + // Returns a block of the listener (master)'s output data + CSRAW + CONDITIONAL ENABLE_AUDIO_FMOD + OBSOLETE warning GetOutputData returning a float[] is deprecated, use GetOutputData and pass a pre allocated array instead. + public static float[] GetOutputData(int numSamples, int channel) + { + float[] samples = new float[numSamples]; + GetOutputDataHelper(samples, channel); + return samples; + } + + // Returns a block of the listener (master)'s output data + CONDITIONAL ENABLE_AUDIO_FMOD + CSRAW static public void GetOutputData(float[] samples, int channel) + { + GetOutputDataHelper(samples, channel); + } + + // Returns a block of the listener (master)'s spectrum data + CSRAW + CONDITIONAL ENABLE_AUDIO_FMOD + OBSOLETE warning GetSpectrumData returning a float[] is deprecated, use GetOutputData and pass a pre allocated array instead. + public static float[] GetSpectrumData(int numSamples, int channel, FFTWindow window) + { + float[] samples = new float[numSamples]; + GetSpectrumDataHelper(samples, channel, window); + return samples; + } + + // Returns a block of the listener (master)'s spectrum data + CONDITIONAL ENABLE_AUDIO_FMOD + CSRAW static public void GetSpectrumData(float[] samples, int channel, FFTWindow window) + { + GetSpectrumDataHelper(samples, channel, window); + } + +END + + +// Spectrum analysis windowing types +CONDITIONAL ENABLE_AUDIO_FMOD +ENUM FFTWindow + // w[n] = 1.0 + Rectangular = 0, + // w[n] = TRI(2n/N) + Triangle = 1, + // w[n] = 0.54 - (0.46 * COS(n/N) ) + Hamming = 2, + // w[n] = 0.5 * (1.0 - COS(n/N) ) + Hanning = 3, + // w[n] = 0.42 - (0.5 * COS(n/N) ) + (0.08 * COS(2.0 * n/N) ) + Blackman = 4, + // 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)) + BlackmanHarris = 5 +END + +// Rolloff modes that a 3D sound can have in an audio source. +CONDITIONAL ENABLE_AUDIO +ENUM AudioRolloffMode + // Use this mode when you want a real-world rolloff. + Logarithmic = 0, + + + // Use this mode when you want to lower the volume of your sound over the distance + Linear = 1, + + + // Use this when you want to use a custom rolloff. + Custom = 2 +END + +// A representation of audio sources in 3D. +CONDITIONAL ENABLE_AUDIO +CLASS AudioSource : Behaviour + // The volume of the audio source (0.0 to 1.0) + AUTO_PROP float volume GetVolume SetVolume + + // The pitch of the audio source. + CUSTOM_PROP float pitch + { + return self->GetPitch(); + } + { + if(!IsFinite(value)) + { + WarningStringObject("Attempt to set pitch to infinite value from script ignored!", self); + return; + } + if(IsNAN(value)) + { + WarningStringObject("Attempt to set pitch to NaN value from script ignored!", self); + return; + } + self->SetPitch(value); + } + + // Playback position in seconds. + CONDITIONAL ENABLE_AUDIO + AUTO_PROP float time GetSecPosition SetSecPosition + + // Playback position in PCM samples. + CONDITIONAL ENABLE_AUDIO_FMOD + THREAD_SAFE AUTO_PROP int timeSamples GetSamplePosition SetSamplePosition + + // The default [[AudioClip]] to play + AUTO_PTR_PROP AudioClip clip GetAudioClip SetAudioClip + + // Plays the ::ref::clip with a certain delay (the optional delay argument is deprecated since 4.1a3) and the functionality has been replated by PlayDelayed. + + CUSTOM void Play (UInt64 delay=0) + { + if (delay > 0 && IS_CONTENT_NEWER_OR_SAME (kUnityVersion4_1_a3)) + { + WarningStringObject("Delayed playback via the optional argument of Play is deprecated. Use PlayDelayed instead!", self); + } + self->Play((double)delay * (const double)(-1.0 / 44100.0)); + } + + // Plays the ::ref::clip with a delay specified in seconds. Users are advised to use this function instead of the old Play(delay) function that took a delay specified in samples relative to a reference rate of 44.1 kHz as an argument. + CUSTOM void PlayDelayed (float delay) + { + self->Play((delay < 0.0f) ? 0.0 : -(double)delay); + } + + // Schedules the ::ref::clip to play at the specified absolute time. This is the preferred way to stitch AudioClips in music players because it is independent of the frame rate and gives the audio system enough time to prepare the playback of the sound to fetch it from media where the opening and buffering takes a lot of time (streams) without causing sudden performance peaks. + CUSTOM void PlayScheduled (double time) + { + self->Play((time < 0.0) ? 0.0 : time); + } + + // Changes the time at which a sound that has already been scheduled to play will start. Notice that depending on the timing not all rescheduling requests can be fulfilled. + CUSTOM void SetScheduledStartTime (double time) + { + self->SetScheduledStartTime(time); + } + + // Changes the time at which a sound that has already been scheduled to play will end. Notice that depending on the timing not all rescheduling requests can be fulfilled. + CUSTOM void SetScheduledEndTime (double time) + { + self->SetScheduledEndTime(time); + } + + // Stops playing the ::ref::clip. + CUSTOM void Stop() + { + self->Stop(true); + } + + // Pauses playing the ::ref::clip. + AUTO void Pause (); + + // Is the ::ref::clip playing right now (RO)? + AUTO_PROP bool isPlaying IsPlayingScripting + + // Plays an [[AudioClip]], and scales the [[AudioSource]] volume by volumeScale. + CONDITIONAL ENABLE_AUDIO + CUSTOM void PlayOneShot (AudioClip clip, float volumeScale = 1.0F) { if (clip) self->PlayOneShot (*clip, volumeScale); } + + // Plays the clip at position. Automatically cleans up the audio source after it has finished playing. + + CONDITIONAL ENABLE_AUDIO + CSRAW public static void PlayClipAtPoint (AudioClip clip, Vector3 position, float volume = 1.0F) + { + GameObject go = new GameObject ("One shot audio"); + go.transform.position = position; + AudioSource source = (AudioSource)go.AddComponent (typeof(AudioSource)); + source.clip = clip; + source.volume = volume; + source.Play (); + // Note: timeScale > 1 means that game time is accelerated. However, the sounds play at their normal speed, so we need to postpone the point in time, when the sound is stopped. + Destroy (go, clip.length * Time.timeScale); + } + + // Is the audio clip looping? + AUTO_PROP bool loop GetLoop SetLoop + + // This makes the audio source not take into account the volume of the audio listener. + CUSTOM_PROP bool ignoreListenerVolume + { + return self->GetIgnoreListenerVolume(); + } + { + self->SetIgnoreListenerVolume(value); + } + + // If set to true, the audio source will automatically start playing on awake + AUTO_PROP bool playOnAwake GetPlayOnAwake SetPlayOnAwake + + // If set to true, the audio source will be playable while the AudioListener is paused + AUTO_PROP bool ignoreListenerPause GetIgnoreListenerPause SetIgnoreListenerPause + + // Whether the Audio Source should be updated in the fixed or dynamic update. + AUTO_PROP AudioVelocityUpdateMode velocityUpdateMode GetVelocityUpdateMode SetVelocityUpdateMode + + + // Sets how much the 3d engine has an effect on the channel. + AUTO_PROP float panLevel GetPanLevel SetPanLevel + + // Bypass effects + CONDITIONAL ENABLE_AUDIO_FMOD + AUTO_PROP bool bypassEffects GetBypassEffects SetBypassEffects + + // Bypass listener effects + CONDITIONAL ENABLE_AUDIO_FMOD + AUTO_PROP bool bypassListenerEffects GetBypassListenerEffects SetBypassListenerEffects + // Bypass reverb zones + CONDITIONAL ENABLE_AUDIO_FMOD + AUTO_PROP bool bypassReverbZones GetBypassReverbZones SetBypassReverbZones + + // Sets the Doppler scale for this AudioSource + AUTO_PROP float dopplerLevel GetDopplerLevel SetDopplerLevel + + // Sets the spread angle a 3d stereo or multichannel sound in speaker space. + AUTO_PROP float spread GetSpread SetSpread + + // Sets the priority of the [[AudioSource]] + CONDITIONAL ENABLE_AUDIO_FMOD + AUTO_PROP int priority GetPriority SetPriority + + // Un- / Mutes the AudioSource. Mute sets the volume=0, Un-Mute restore the original volume. + AUTO_PROP bool mute GetMute SetMute + + // Within the Min distance the AudioSource will cease to grow louder in volume. + AUTO_PROP float minDistance GetMinDistance SetMinDistance + + // (Logarithmic rolloff) MaxDistance is the distance a sound stops attenuating at. + AUTO_PROP float maxDistance GetMaxDistance SetMaxDistance + + // Sets a channels pan position linearly. Only works for 2D clips. + AUTO_PROP float pan GetPan SetPan + + // Sets/Gets how the AudioSource attenuates over distance + + AUTO_PROP AudioRolloffMode rolloffMode GetRolloffMode SetRolloffMode + + CUSTOM private void GetOutputDataHelper(float[] samples, int channel) + { +#if ENABLE_AUDIO_FMOD + self->GetOutputData(Scripting::GetScriptingArrayStart<float>(samples), GetScriptingArraySize(samples), channel); +#endif + } + + // Returns a block of the currently playing source's output data + CSRAW + CONDITIONAL ENABLE_AUDIO_FMOD + OBSOLETE warning GetOutputData return a float[] is deprecated, use GetOutputData passing a pre allocated array instead. + public float[] GetOutputData(int numSamples, int channel) + { + float[] samples = new float[numSamples]; + GetOutputDataHelper(samples, channel); + return samples; + } + + // Returns a block of the currently playing source's output data + CONDITIONAL ENABLE_AUDIO_FMOD + CSRAW public void GetOutputData(float[] samples, int channel) + { + GetOutputDataHelper(samples, channel); + } + + CONDITIONAL ENABLE_AUDIO_FMOD + CUSTOM private void GetSpectrumDataHelper(float[] samples, int channel, FFTWindow window) + { + self->GetSpectrumData(Scripting::GetScriptingArrayStart<float>(samples), GetScriptingArraySize(samples), channel, (FMOD_DSP_FFT_WINDOW) window); + } + + // Returns a block of the currently playing source's spectrum data + CSRAW + CONDITIONAL ENABLE_AUDIO_FMOD + OBSOLETE warning GetSpectrumData returning a float[] is deprecated, use GetSpectrumData passing a pre allocated array instead. + public float[] GetSpectrumData(int numSamples, int channel, FFTWindow window) + { + float[] samples = new float[numSamples]; + GetSpectrumDataHelper(samples, channel, window); + return samples; + } + + // Returns a block of the currently playing source's spectrum data + CONDITIONAL ENABLE_AUDIO_FMOD + CSRAW public void GetSpectrumData(float[] samples, int channel, FFTWindow window) + { + GetSpectrumDataHelper(samples, channel, window); + } + + FLUSHCONDITIONS + + OBSOLETE error minVolume is not supported anymore. Use min-, maxDistance and rolloffMode instead. + CUSTOM_PROP float minVolume + { + ErrorString("minVolume is not supported anymore. Use min-, maxDistance and rolloffMode instead."); + return 0.0f; + } + { + ErrorString("minVolume is not supported anymore. Use min-, maxDistance and rolloffMode instead."); + } + OBSOLETE error maxVolume is not supported anymore. Use min-, maxDistance and rolloffMode instead. + CUSTOM_PROP float maxVolume + { + ErrorString("maxVolume is not supported anymore. Use min-, maxDistance and rolloffMode instead."); + return 0.0f; + } + { + ErrorString("maxVolume is not supported anymore. Use min-, maxDistance and rolloffMode instead."); + } + OBSOLETE error rolloffFactor is not supported anymore. Use min-, maxDistance and rolloffMode instead. + CUSTOM_PROP float rolloffFactor + { + ErrorString("rolloffFactor is not supported anymore. Use min-, maxDistance and rolloffMode instead."); + return 0.0f; + } + { + ErrorString("rolloffFactor is not supported anymore. Use min-, maxDistance and rolloffMode instead."); + } +END +// Reverb presets used by the Reverb Zone class and the audio reverb filter +ENUM AudioReverbPreset + // No reverb preset selected + Off = 0, + + // Generic preset. + Generic = 1, + + // Padded cell preset. + PaddedCell = 2, + + // Room preset. + Room = 3, + + // Bathroom preset. + Bathroom = 4, + + // Livingroom preset + Livingroom = 5, + + // Stoneroom preset + Stoneroom = 6, + + // Auditorium preset. + Auditorium = 7, + + // Concert hall preset. + Concerthall = 8, + + // Cave preset. + Cave = 9, + + // Arena preset. + Arena = 10, + + // Hangar preset. + Hangar = 11, + + // Carpeted hallway preset. + CarpetedHallway = 12, + + // Hallway preset. + Hallway = 13, + + // Stone corridor preset. + StoneCorridor = 14, + + // Alley preset. + Alley = 15, + + // Forest preset. + Forest = 16, + + // City preset. + City = 17, + + // Mountains preset. + Mountains = 18, + + // Quarry preset. + Quarry = 19, + + // Plain preset. + Plain = 20, + + // Parking Lot preset + ParkingLot = 21, + + // Sewer pipe preset. + SewerPipe = 22, + + // Underwater presset + Underwater = 23, + + // Drugged preset + Drugged = 24, + + // Dizzy preset. + Dizzy = 25, + + // Psychotic preset. + Psychotic = 26, + + // User defined preset. + User = 27 +END + +// Reverb Zones are used when you want to gradually change from a point +CONDITIONAL ENABLE_AUDIO_FMOD +CLASS AudioReverbZone : Behaviour + // The distance from the centerpoint that the reverb will have full effect at. Default = 10.0. + AUTO_PROP float minDistance GetMinDistance SetMinDistance + + // The distance from the centerpoint that the reverb will not have any effect. Default = 15.0. + AUTO_PROP float maxDistance GetMaxDistance SetMaxDistance + + // Set/Get reverb preset properties + AUTO_PROP AudioReverbPreset reverbPreset GetReverbPreset SetReverbPreset + + // room effect level (at mid frequencies) + AUTO_PROP int room GetRoom SetRoom + // relative room effect level at high frequencies + AUTO_PROP int roomHF GetRoomHF SetRoomHF + // relative room effect level at low frequencies + AUTO_PROP int roomLF GetRoomLF SetRoomLF + // reverberation decay time at mid frequencies + AUTO_PROP float decayTime GetDecayTime SetDecayTime + // high-frequency to mid-frequency decay time ratio + AUTO_PROP float decayHFRatio GetDecayHFRatio SetDecayHFRatio + // early reflections level relative to room effect + AUTO_PROP int reflections GetReflections SetReflections + // initial reflection delay time + AUTO_PROP float reflectionsDelay GetReflectionsDelay SetReflectionsDelay + // late reverberation level relative to room effect + AUTO_PROP int reverb GetReverb SetReverb + // late reverberation delay time relative to initial reflection + AUTO_PROP float reverbDelay GetReverbDelay SetReverbDelay + // reference high frequency (hz) + AUTO_PROP float HFReference GetHFReference SetHFReference + // reference low frequency (hz) + AUTO_PROP float LFReference GetLFReference SetLFReference + // like rolloffscale in global settings, but for reverb room size effect + AUTO_PROP float roomRolloffFactor GetRoomRolloffFactor SetRoomRolloffFactor + // Value that controls the echo density in the late reverberation decay + AUTO_PROP float diffusion GetDiffusion SetDiffusion + // Value that controls the modal density in the late reverberation decay + AUTO_PROP float density GetDensity SetDensity +END + + +// The Audio Low Pass Filter filter passes low frequencies of an +CONDITIONAL ENABLE_AUDIO_FMOD +CLASS AudioLowPassFilter : Behaviour + // Lowpass cutoff frequency in hz. 10.0 to 22000.0. Default = 5000.0. + AUTO_PROP float cutoffFrequency GetCutoffFrequency SetCutoffFrequency + + // Determines how much the filter's self-resonance is dampened. + AUTO_PROP float lowpassResonaceQ GetLowpassResonanceQ SetLowpassResonanceQ +END +// The Audio High Pass Filter passes high frequencies of an AudioSource and +CONDITIONAL ENABLE_AUDIO_FMOD +CLASS AudioHighPassFilter : Behaviour + // Highpass cutoff frequency in hz. 10.0 to 22000.0. Default = 5000.0. + AUTO_PROP float cutoffFrequency GetCutoffFrequency SetCutoffFrequency + + + // Determines how much the filter's self-resonance isdampened. + AUTO_PROP float highpassResonaceQ GetHighpassResonanceQ SetHighpassResonanceQ +END +// The Audio Distortion Filter distorts the sound from an AudioSource or +CONDITIONAL ENABLE_AUDIO_FMOD +CLASS AudioDistortionFilter : Behaviour + // Distortion value. 0.0 to 1.0. Default = 0.5. + AUTO_PROP float distortionLevel GetDistortionLevel SetDistortionLevel +END + +// The Audio Echo Filter repeats a sound after a given Delay, attenuating +CONDITIONAL ENABLE_AUDIO_FMOD +CLASS AudioEchoFilter : Behaviour + // Echo delay in ms. 10 to 5000. Default = 500. + AUTO_PROP float delay GetDelay SetDelay + + + // Echo decay per delay. 0 to 1. 1.0 = No decay, 0.0 = total decay (i.e. simple 1 line delay). Default = 0.5. + AUTO_PROP float decayRatio GetDecayRatio SetDecayRatio + + + // Volume of original signal to pass to output. 0.0 to 1.0. Default = 1.0. + AUTO_PROP float dryMix GetDryMix SetDryMix + + + // Volume of echo signal to pass to output. 0.0 to 1.0. Default = 1.0. + AUTO_PROP float wetMix GetWetMix SetWetMix +END + +// The Audio Chorus Filter takes an Audio Clip and processes it creating a chorus effect. +CONDITIONAL ENABLE_AUDIO_FMOD +CLASS AudioChorusFilter : Behaviour + // Volume of original signal to pass to output. 0.0 to 1.0. Default = 0.5. + AUTO_PROP float dryMix GetDryMix SetDryMix + + + // Volume of 1st chorus tap. 0.0 to 1.0. Default = 0.5. + AUTO_PROP float wetMix1 GetWetMix1 SetWetMix1 + + + // 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. + AUTO_PROP float wetMix2 GetWetMix2 SetWetMix2 + + + // 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. + AUTO_PROP float wetMix3 GetWetMix3 SetWetMix3 + + + // Chorus delay in ms. 0.1 to 100.0. Default = 40.0 ms. + AUTO_PROP float delay GetDelay SetDelay + + + // Chorus modulation rate in hz. 0.0 to 20.0. Default = 0.8 hz. + AUTO_PROP float rate GetRate SetRate + + + // Chorus modulation depth. 0.0 to 1.0. Default = 0.03. + AUTO_PROP float depth GetDepth SetDepth + + /// Chorus feedback. Controls how much of the wet signal gets fed back into the chorus buffer. 0.0 to 1.0. Default = 0.0. + OBSOLETE warning feedback is deprecated, this property does nothing. + CUSTOM_PROP float feedback + { + return 0.0f; + } + {} + +END +// The Audio Reverb Filter takes an Audio Clip and distortionates it in a +CONDITIONAL ENABLE_AUDIO_FMOD +CLASS AudioReverbFilter : Behaviour + // Set/Get reverb preset properties + AUTO_PROP AudioReverbPreset reverbPreset GetReverbPreset SetReverbPreset + // Mix level of dry signal in output in mB. Ranges from -10000.0 to 0.0. Default is 0. + AUTO_PROP float dryLevel GetDryLevel SetDryLevel + // Room effect level at low frequencies in mB. Ranges from -10000.0 to 0.0. Default is 0.0. + AUTO_PROP float room GetRoom SetRoom + // Room effect high-frequency level re. low frequency level in mB. Ranges from -10000.0 to 0.0. Default is 0.0. + AUTO_PROP float roomHF GetRoomHF SetRoomHF + // Rolloff factor for room effect. Ranges from 0.0 to 10.0. Default is 10.0 + AUTO_PROP float roomRolloff GetRoomRolloff SetRoomRolloff + // Reverberation decay time at low-frequencies in seconds. Ranges from 0.1 to 20.0. Default is 1.0. + AUTO_PROP float decayTime GetDecayTime SetDecayTime + // Decay HF Ratio : High-frequency to low-frequency decay time ratio. Ranges from 0.1 to 2.0. Default is 0.5. + AUTO_PROP float decayHFRatio GetDecayHFRatio SetDecayHFRatio + // Early reflections level relative to room effect in mB. Ranges from -10000.0 to 1000.0. Default is -10000.0. + AUTO_PROP float reflectionsLevel GetReflectionsLevel SetReflectionsLevel + // Late reverberation level relative to room effect in mB. Ranges from -10000.0 to 2000.0. Default is 0.0. + AUTO_PROP float reflectionsDelay GetReflectionsDelay SetReflectionsDelay + // Late reverberation level relative to room effect in mB. Ranges from -10000.0 to 2000.0. Default is 0.0. + AUTO_PROP float reverbLevel GetReverbLevel SetReverbLevel + // Late reverberation delay time relative to first reflection in seconds. Ranges from 0.0 to 0.1. Default is 0.04. + AUTO_PROP float reverbDelay GetReverbDelay SetReverbDelay + // Reverberation diffusion (echo density) in percent. Ranges from 0.0 to 100.0. Default is 100.0. + AUTO_PROP float diffusion GetDiffusion SetDiffusion + // Reverberation density (modal density) in percent. Ranges from 0.0 to 100.0. Default is 100.0. + AUTO_PROP float density GetDensity SetDensity + // Reference high frequency in Hz. Ranges from 20.0 to 20000.0. Default is 5000.0. + AUTO_PROP float hfReference GetHFReference SetHFReference + // Room effect low-frequency level in mB. Ranges from -10000.0 to 0.0. Default is 0.0. + AUTO_PROP float roomLF GetRoomLF SetRoomLF + // Reference low-frequency in Hz. Ranges from 20.0 to 1000.0. Default is 250.0. + AUTO_PROP float lFReference GetLFReference SetLFReference +END + + + +CONDITIONAL ENABLE_MICROPHONE +// Use this class to record to an [[AudioClip|audio clip]] using a connected microphone. +CLASS Microphone + // Start Recording with device + + CUSTOM static AudioClip Start(string deviceName, bool loop, int lengthSec, int frequency) + { + return Scripting::ScriptingWrapperFor( GetAudioManager().StartRecord( GetAudioManager().GetMicrophoneDeviceIDFromName ( deviceName ) , loop, lengthSec, frequency ) ); + } + + // Stops recording + CUSTOM static void End(string deviceName) + { + GetAudioManager().EndRecord( GetAudioManager().GetMicrophoneDeviceIDFromName ( deviceName ) ); + } + + // Gives you a list microphone devices, identified by name. + CUSTOM_PROP static string[] devices + { + std::vector<std::string> names; + names = GetAudioManager().GetRecordDevices(); + + ScriptingArrayPtr array = CreateScriptingArray<ScriptingStringPtr> (MONO_COMMON.string, names.size ()); + for (int i=0;i<names.size ();i++) + Scripting::SetScriptingArrayElement (array, i, scripting_string_new ( (const char*)names[i].c_str() )); + + return array; + } + + // Query if a device is currently recording. + CUSTOM static bool IsRecording(string deviceName) + { + return GetAudioManager().IsRecording( GetAudioManager().GetMicrophoneDeviceIDFromName ( deviceName ) ); + } + + // Get the position in samples of the recording. + THREAD_SAFE CUSTOM static int GetPosition(string deviceName) + { + return GetAudioManager().GetRecordPosition( GetAudioManager().GetMicrophoneDeviceIDFromName ( deviceName ) ); + } + + // Get the frequency capabilities of a device. + CUSTOM static void GetDeviceCaps(string deviceName, out int minFreq, out int maxFreq) + { + GetAudioManager().GetDeviceCaps( GetAudioManager().GetMicrophoneDeviceIDFromName ( deviceName ), minFreq, maxFreq ); + } + +END + + +CSRAW } + |