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doom3/neo/sound/snd_local.h

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/*
===========================================================================
Doom 3 GPL Source Code
Copyright (C) 1999-2011 id Software LLC, a ZeniMax Media company.
This file is part of the Doom 3 GPL Source Code (?Doom 3 Source Code?).
Doom 3 Source Code is free software: you can redistribute it and/or modify
it under the terms of the GNU General Public License as published by
the Free Software Foundation, either version 3 of the License, or
(at your option) any later version.
Doom 3 Source Code is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
GNU General Public License for more details.
You should have received a copy of the GNU General Public License
along with Doom 3 Source Code. If not, see <http://www.gnu.org/licenses/>.
In addition, the Doom 3 Source Code is also subject to certain additional terms. You should have received a copy of these additional terms immediately following the terms and conditions of the GNU General Public License which accompanied the Doom 3 Source Code. If not, please request a copy in writing from id Software at the address below.
If you have questions concerning this license or the applicable additional terms, you may contact in writing id Software LLC, c/o ZeniMax Media Inc., Suite 120, Rockville, Maryland 20850 USA.
===========================================================================
*/
#ifndef __SND_LOCAL_H__
#define __SND_LOCAL_H__
// you need the OpenAL headers for build, even if AL is not enabled - http://www.openal.org/
#ifdef _WIN32
#include "../openal/include/al.h"
#include "../openal/include/alc.h"
#include "../openal/idal.h"
// broken OpenAL SDK ?
#define ID_ALCHAR (ALubyte *)
#elif defined( MACOS_X )
#include <OpenAL/al.h>
#include <OpenAL/alc.h>
#define ID_ALCHAR
#else
#include <AL/al.h>
#include <AL/alc.h>
#define ID_ALCHAR
#endif
#include "../openal/include/efxlib.h"
// demo sound commands
typedef enum {
SCMD_STATE, // followed by a load game state
SCMD_PLACE_LISTENER,
SCMD_ALLOC_EMITTER,
SCMD_FREE,
SCMD_UPDATE,
SCMD_START,
SCMD_MODIFY,
SCMD_STOP,
SCMD_FADE
} soundDemoCommand_t;
const int SOUND_MAX_CHANNELS = 8;
const int SOUND_DECODER_FREE_DELAY = 1000 * MIXBUFFER_SAMPLES / USERCMD_MSEC; // four seconds
const int PRIMARYFREQ = 44100; // samples per second
const float SND_EPSILON = 1.0f / 32768.0f; // if volume is below this, it will always multiply to zero
const int ROOM_SLICES_IN_BUFFER = 10;
class idAudioHardware;
class idAudioBuffer;
class idWaveFile;
class idSoundCache;
class idSoundSample;
class idSampleDecoder;
class idSoundWorldLocal;
/*
===================================================================================
General extended waveform format structure.
Use this for all NON PCM formats.
===================================================================================
*/
#ifdef WIN32
#pragma pack(1)
#endif
#ifdef __MWERKS__
#pragma pack (push, 1)
#endif
struct waveformatex_s {
word wFormatTag; /* format type */
word nChannels; /* number of channels (i.e. mono, stereo...) */
dword nSamplesPerSec; /* sample rate */
dword nAvgBytesPerSec; /* for buffer estimation */
word nBlockAlign; /* block size of data */
word wBitsPerSample; /* Number of bits per sample of mono data */
word cbSize; /* The count in bytes of the size of
extra information (after cbSize) */
} PACKED;
typedef waveformatex_s waveformatex_t;
/* OLD general waveform format structure (information common to all formats) */
struct waveformat_s {
word wFormatTag; /* format type */
word nChannels; /* number of channels (i.e. mono, stereo, etc.) */
dword nSamplesPerSec; /* sample rate */
dword nAvgBytesPerSec; /* for buffer estimation */
word nBlockAlign; /* block size of data */
} PACKED;
typedef waveformat_s waveformat_t;
/* flags for wFormatTag field of WAVEFORMAT */
enum {
WAVE_FORMAT_TAG_PCM = 1,
WAVE_FORMAT_TAG_OGG = 2
};
/* specific waveform format structure for PCM data */
struct pcmwaveformat_s {
waveformat_t wf;
word wBitsPerSample;
} PACKED;
typedef pcmwaveformat_s pcmwaveformat_t;
#ifndef mmioFOURCC
#define mmioFOURCC( ch0, ch1, ch2, ch3 ) \
( (dword)(byte)(ch0) | ( (dword)(byte)(ch1) << 8 ) | \
( (dword)(byte)(ch2) << 16 ) | ( (dword)(byte)(ch3) << 24 ) )
#endif
#define fourcc_riff mmioFOURCC('R', 'I', 'F', 'F')
struct waveformatextensible_s {
waveformatex_t Format;
union {
word wValidBitsPerSample; /* bits of precision */
word wSamplesPerBlock; /* valid if wBitsPerSample==0*/
word wReserved; /* If neither applies, set to zero*/
} Samples;
dword dwChannelMask; /* which channels are */
/* present in stream */
int SubFormat;
} PACKED;
typedef waveformatextensible_s waveformatextensible_t;
typedef dword fourcc;
/* RIFF chunk information data structure */
struct mminfo_s {
fourcc ckid; /* chunk ID */
dword cksize; /* chunk size */
fourcc fccType; /* form type or list type */
dword dwDataOffset; /* offset of data portion of chunk */
} PACKED;
typedef mminfo_s mminfo_t;
#ifdef WIN32
#pragma pack()
#endif
#ifdef __MWERKS__
#pragma pack (pop)
#endif
/*
===================================================================================
idWaveFile
===================================================================================
*/
class idWaveFile {
public:
idWaveFile( void );
~idWaveFile( void );
int Open( const char* strFileName, waveformatex_t* pwfx = NULL );
int OpenFromMemory( short* pbData, int ulDataSize, waveformatextensible_t* pwfx );
int Read( byte* pBuffer, int dwSizeToRead, int *pdwSizeRead );
int Seek( int offset );
int Close( void );
int ResetFile( void );
int GetOutputSize( void ) { return mdwSize; }
int GetMemorySize( void ) { return mMemSize; }
waveformatextensible_t mpwfx; // Pointer to waveformatex structure
private:
idFile * mhmmio; // I/O handle for the WAVE
mminfo_t mck; // Multimedia RIFF chunk
mminfo_t mckRiff; // used when opening a WAVE file
dword mdwSize; // size in samples
dword mMemSize; // size of the wave data in memory
dword mseekBase;
ID_TIME_T mfileTime;
bool mbIsReadingFromMemory;
short * mpbData;
short * mpbDataCur;
dword mulDataSize;
void * ogg; // only !NULL when !s_realTimeDecoding
bool isOgg;
private:
int ReadMMIO( void );
int OpenOGG( const char* strFileName, waveformatex_t* pwfx = NULL );
int ReadOGG( byte* pBuffer, int dwSizeToRead, int *pdwSizeRead );
int CloseOGG( void );
};
/*
===================================================================================
idAudioHardware
===================================================================================
*/
class idAudioHardware {
public:
static idAudioHardware *Alloc();
virtual ~idAudioHardware();
virtual bool Initialize( ) = 0;
virtual bool Lock( void **pDSLockedBuffer, ulong *dwDSLockedBufferSize ) = 0;
virtual bool Unlock( void *pDSLockedBuffer, dword dwDSLockedBufferSize ) = 0;
virtual bool GetCurrentPosition( ulong *pdwCurrentWriteCursor ) = 0;
// try to write as many sound samples to the device as possible without blocking and prepare for a possible new mixing call
// returns wether there is *some* space for writing available
virtual bool Flush( void ) = 0;
virtual void Write( bool flushing ) = 0;
virtual int GetNumberOfSpeakers( void )= 0;
virtual int GetMixBufferSize( void ) = 0;
virtual short* GetMixBuffer( void ) = 0;
};
/*
===================================================================================
Encapsulates functionality of a DirectSound buffer.
===================================================================================
*/
class idAudioBuffer {
public:
virtual int Play( dword dwPriority=0, dword dwFlags=0 ) = 0;
virtual int Stop( void ) = 0;
virtual int Reset( void ) = 0;
virtual bool IsSoundPlaying( void ) = 0;
virtual void SetVolume( float x ) = 0;
};
/*
===================================================================================
idSoundEmitterLocal
===================================================================================
*/
typedef enum {
REMOVE_STATUS_INVALID = -1,
REMOVE_STATUS_ALIVE = 0,
REMOVE_STATUS_WAITSAMPLEFINISHED = 1,
REMOVE_STATUS_SAMPLEFINISHED = 2
} removeStatus_t;
class idSoundFade {
public:
int fadeStart44kHz;
int fadeEnd44kHz;
float fadeStartVolume; // in dB
float fadeEndVolume; // in dB
void Clear();
float FadeDbAt44kHz( int current44kHz );
};
class SoundFX {
protected:
bool initialized;
int channel;
int maxlen;
float* buffer;
float continuitySamples[4];
float param;
public:
SoundFX() { channel = 0; buffer = NULL; initialized = false; maxlen = 0; memset( continuitySamples, 0, sizeof( float ) * 4 ); };
virtual ~SoundFX() { if ( buffer ) delete buffer; };
virtual void Initialize() { };
virtual void ProcessSample( float* in, float* out ) = 0;
void SetChannel( int chan ) { channel = chan; };
int GetChannel() { return channel; };
void SetContinuitySamples( float in1, float in2, float out1, float out2 ) { continuitySamples[0] = in1; continuitySamples[1] = in2; continuitySamples[2] = out1; continuitySamples[3] = out2; }; // FIXME?
void GetContinuitySamples( float& in1, float& in2, float& out1, float& out2 ) { in1 = continuitySamples[0]; in2 = continuitySamples[1]; out1 = continuitySamples[2]; out2 = continuitySamples[3]; };
void SetParameter( float val ) { param = val; };
};
class SoundFX_Lowpass : public SoundFX {
public:
virtual void ProcessSample( float* in, float* out );
};
class SoundFX_LowpassFast : public SoundFX {
float freq;
float res;
float a1, a2, a3;
float b1, b2;
public:
virtual void ProcessSample( float* in, float* out );
void SetParms( float p1 = 0, float p2 = 0, float p3 = 0 );
};
class SoundFX_Comb : public SoundFX {
int currentTime;
public:
virtual void Initialize();
virtual void ProcessSample( float* in, float* out );
};
class FracTime {
public:
int time;
float frac;
void Set( int val ) { time = val; frac = 0; };
void Increment( float val ) { frac += val; while ( frac >= 1.f ) { time++; frac--; } };
};
enum {
PLAYBACK_RESET,
PLAYBACK_ADVANCING
};
class idSoundChannel;
class idSlowChannel {
bool active;
const idSoundChannel* chan;
int playbackState;
int triggerOffset;
FracTime newPosition;
int newSampleOffset;
FracTime curPosition;
int curSampleOffset;
SoundFX_LowpassFast lowpass;
// functions
void GenerateSlowChannel( FracTime& playPos, int sampleCount44k, float* finalBuffer );
float GetSlowmoSpeed();
public:
void AttachSoundChannel( const idSoundChannel *chan );
void Reset();
void GatherChannelSamples( int sampleOffset44k, int sampleCount44k, float *dest );
bool IsActive() { return active; };
FracTime GetCurrentPosition() { return curPosition; };
};
class idSoundChannel {
public:
idSoundChannel( void );
~idSoundChannel( void );
void Clear( void );
void Start( void );
void Stop( void );
void GatherChannelSamples( int sampleOffset44k, int sampleCount44k, float *dest ) const;
void ALStop( void ); // free OpenAL resources if any
bool triggerState;
int trigger44kHzTime; // hardware time sample the channel started
int triggerGame44kHzTime; // game time sample time the channel started
soundShaderParms_t parms; // combines the shader parms and the per-channel overrides
idSoundSample * leadinSample; // if not looped, this is the only sample
s_channelType triggerChannel;
const idSoundShader *soundShader;
idSampleDecoder * decoder;
float diversity;
float lastVolume; // last calculated volume based on distance
float lastV[6]; // last calculated volume for each speaker, so we can smoothly fade
idSoundFade channelFade;
bool triggered;
ALuint openalSource;
ALuint openalStreamingOffset;
ALuint openalStreamingBuffer[3];
ALuint lastopenalStreamingBuffer[3];
bool disallowSlow;
};
class idSoundEmitterLocal : public idSoundEmitter {
public:
idSoundEmitterLocal( void );
virtual ~idSoundEmitterLocal( void );
//----------------------------------------------
// the "time" parameters should be game time in msec, which is used to make queries
// return deterministic values regardless of async buffer scheduling
// a non-immediate free will let all currently playing sounds complete
virtual void Free( bool immediate );
// the parms specified will be the default overrides for all sounds started on this emitter.
// NULL is acceptable for parms
virtual void UpdateEmitter( const idVec3 &origin, int listenerId, const soundShaderParms_t *parms );
// returns the length of the started sound in msec
virtual int StartSound( const idSoundShader *shader, const s_channelType channel, float diversity = 0, int shaderFlags = 0, bool allowSlow = true /* D3XP */ );
// can pass SCHANNEL_ANY
virtual void ModifySound( const s_channelType channel, const soundShaderParms_t *parms );
virtual void StopSound( const s_channelType channel );
virtual void FadeSound( const s_channelType channel, float to, float over );
virtual bool CurrentlyPlaying( void ) const;
// can pass SCHANNEL_ANY
virtual float CurrentAmplitude( void );
// used for save games
virtual int Index( void ) const;
//----------------------------------------------
void Clear( void );
void OverrideParms( const soundShaderParms_t *base, const soundShaderParms_t *over, soundShaderParms_t *out );
void CheckForCompletion( int current44kHzTime );
void Spatialize( idVec3 listenerPos, int listenerArea, idRenderWorld *rw );
idSoundWorldLocal * soundWorld; // the world that holds this emitter
int index; // in world emitter list
removeStatus_t removeStatus;
idVec3 origin;
int listenerId;
soundShaderParms_t parms; // default overrides for all channels
// the following are calculated in UpdateEmitter, and don't need to be archived
float maxDistance; // greatest of all playing channel distances
int lastValidPortalArea; // so an emitter that slides out of the world continues playing
bool playing; // if false, no channel is active
bool hasShakes;
idVec3 spatializedOrigin; // the virtual sound origin, either the real sound origin,
// or a point through a portal chain
float realDistance; // in meters
float distance; // in meters, this may be the straight-line distance, or
// it may go through a chain of portals. If there
// is not an open-portal path, distance will be > maxDistance
// a single soundEmitter can have many channels playing from the same point
idSoundChannel channels[SOUND_MAX_CHANNELS];
idSlowChannel slowChannels[SOUND_MAX_CHANNELS];
idSlowChannel GetSlowChannel( const idSoundChannel *chan );
void SetSlowChannel( const idSoundChannel *chan, idSlowChannel slow );
void ResetSlowChannel( const idSoundChannel *chan );
// this is just used for feedback to the game or rendering system:
// flashing lights and screen shakes. Because the material expression
// evaluation doesn't do common subexpression removal, we cache the
// last generated value
int ampTime;
float amplitude;
};
/*
===================================================================================
idSoundWorldLocal
===================================================================================
*/
class s_stats {
public:
s_stats( void ) {
rinuse = 0;
runs = 1;
timeinprocess = 0;
missedWindow = 0;
missedUpdateWindow = 0;
activeSounds = 0;
}
int rinuse;
int runs;
int timeinprocess;
int missedWindow;
int missedUpdateWindow;
int activeSounds;
};
typedef struct soundPortalTrace_s {
int portalArea;
const struct soundPortalTrace_s *prevStack;
} soundPortalTrace_t;
class idSoundWorldLocal : public idSoundWorld {
public:
virtual ~idSoundWorldLocal( void );
// call at each map start
virtual void ClearAllSoundEmitters( void );
virtual void StopAllSounds( void );
// get a new emitter that can play sounds in this world
virtual idSoundEmitter *AllocSoundEmitter( void );
// for load games
virtual idSoundEmitter *EmitterForIndex( int index );
// query data from all emitters in the world
virtual float CurrentShakeAmplitudeForPosition( const int time, const idVec3 &listererPosition );
// where is the camera/microphone
// listenerId allows listener-private sounds to be added
virtual void PlaceListener( const idVec3 &origin, const idMat3 &axis, const int listenerId, const int gameTime, const idStr& areaName );
// fade all sounds in the world with a given shader soundClass
// to is in Db (sigh), over is in seconds
virtual void FadeSoundClasses( const int soundClass, const float to, const float over );
// dumps the current state and begins archiving commands
virtual void StartWritingDemo( idDemoFile *demo );
virtual void StopWritingDemo( void );
// read a sound command from a demo file
virtual void ProcessDemoCommand( idDemoFile *readDemo );
// background music
virtual void PlayShaderDirectly( const char *name, int channel = -1 );
// pause and unpause the sound world
virtual void Pause( void );
virtual void UnPause( void );
virtual bool IsPaused( void );
// avidump
virtual void AVIOpen( const char *path, const char *name );
virtual void AVIClose( void );
// SaveGame Support
virtual void WriteToSaveGame( idFile *savefile );
virtual void ReadFromSaveGame( idFile *savefile );
virtual void ReadFromSaveGameSoundChannel( idFile *saveGame, idSoundChannel *ch );
virtual void ReadFromSaveGameSoundShaderParams( idFile *saveGame, soundShaderParms_t *params );
virtual void WriteToSaveGameSoundChannel( idFile *saveGame, idSoundChannel *ch );
virtual void WriteToSaveGameSoundShaderParams( idFile *saveGame, soundShaderParms_t *params );
virtual void SetSlowmo( bool active );
virtual void SetSlowmoSpeed( float speed );
virtual void SetEnviroSuit( bool active );
//=======================================
idSoundWorldLocal( void );
void Shutdown( void );
void Init( idRenderWorld *rw );
void ClearBuffer( void );
// update
void ForegroundUpdate( int currentTime );
void OffsetSoundTime( int offset44kHz );
idSoundEmitterLocal * AllocLocalSoundEmitter();
void CalcEars( int numSpeakers, idVec3 realOrigin, idVec3 listenerPos, idMat3 listenerAxis, float ears[6], float spatialize );
void AddChannelContribution( idSoundEmitterLocal *sound, idSoundChannel *chan,
int current44kHz, int numSpeakers, float *finalMixBuffer );
void MixLoop( int current44kHz, int numSpeakers, float *finalMixBuffer );
void AVIUpdate( void );
void ResolveOrigin( const int stackDepth, const soundPortalTrace_t *prevStack, const int soundArea, const float dist, const idVec3& soundOrigin, idSoundEmitterLocal *def );
float FindAmplitude( idSoundEmitterLocal *sound, const int localTime, const idVec3 *listenerPosition, const s_channelType channel, bool shakesOnly );
//============================================
idRenderWorld * rw; // for portals and debug drawing
idDemoFile * writeDemo; // if not NULL, archive commands here
idMat3 listenerAxis;
idVec3 listenerPos; // position in meters
int listenerPrivateId;
idVec3 listenerQU; // position in "quake units"
int listenerArea;
idStr listenerAreaName;
int listenerEnvironmentID;
int gameMsec;
int game44kHz;
int pause44kHz;
int lastAVI44kHz; // determine when we need to mix and write another block
idList<idSoundEmitterLocal *>emitters;
idSoundFade soundClassFade[SOUND_MAX_CLASSES]; // for global sound fading
// avi stuff
idFile * fpa[6];
idStr aviDemoPath;
idStr aviDemoName;
idSoundEmitterLocal * localSound; // just for playShaderDirectly()
bool slowmoActive;
float slowmoSpeed;
bool enviroSuitActive;
};
/*
===================================================================================
idSoundSystemLocal
===================================================================================
*/
typedef struct {
ALuint handle;
int startTime;
idSoundChannel *chan;
bool inUse;
bool looping;
bool stereo;
} openalSource_t;
class idSoundSystemLocal : public idSoundSystem {
public:
idSoundSystemLocal( ) {
isInitialized = false;
}
// all non-hardware initialization
virtual void Init( void );
// shutdown routine
virtual void Shutdown( void );
virtual void ClearBuffer( void );
// sound is attached to the window, and must be recreated when the window is changed
virtual bool ShutdownHW( void );
virtual bool InitHW( void );
// async loop, called at 60Hz
virtual int AsyncUpdate( int time );
// async loop, when the sound driver uses a write strategy
virtual int AsyncUpdateWrite( int time );
// direct mixing called from the sound driver thread for OSes that support it
virtual int AsyncMix( int soundTime, float *mixBuffer );
virtual void SetMute( bool mute );
virtual cinData_t ImageForTime( const int milliseconds, const bool waveform );
int GetSoundDecoderInfo( int index, soundDecoderInfo_t &decoderInfo );
// if rw == NULL, no portal occlusion or rendered debugging is available
virtual idSoundWorld *AllocSoundWorld( idRenderWorld *rw );
// specifying NULL will cause silence to be played
virtual void SetPlayingSoundWorld( idSoundWorld *soundWorld );
// some tools, like the sound dialog, may be used in both the game and the editor
// This can return NULL, so check!
virtual idSoundWorld *GetPlayingSoundWorld( void );
virtual void BeginLevelLoad( void );
virtual void EndLevelLoad( const char *mapString );
virtual void PrintMemInfo( MemInfo_t *mi );
virtual int IsEAXAvailable( void );
//-------------------------
int GetCurrent44kHzTime( void ) const;
float dB2Scale( const float val ) const;
int SamplesToMilliseconds( int samples ) const;
int MillisecondsToSamples( int ms ) const;
void DoEnviroSuit( float* samples, int numSamples, int numSpeakers );
ALuint AllocOpenALSource( idSoundChannel *chan, bool looping, bool stereo );
void FreeOpenALSource( ALuint handle );
idAudioHardware * snd_audio_hw;
idSoundCache * soundCache;
idSoundWorldLocal * currentSoundWorld; // the one to mix each async tic
int olddwCurrentWritePos; // statistics
int buffers; // statistics
int CurrentSoundTime; // set by the async thread and only used by the main thread
unsigned int nextWriteBlock;
float realAccum[6*MIXBUFFER_SAMPLES+16];
float * finalMixBuffer; // points inside realAccum at a 16 byte aligned boundary
bool isInitialized;
bool muted;
bool shutdown;
s_stats soundStats; // NOTE: updated throughout the code, not displayed anywhere
int meterTops[256];
int meterTopsTime[256];
dword * graph;
float volumesDB[1200]; // dB to float volume conversion
idList<SoundFX*> fxList;
ALCdevice *openalDevice;
ALCcontext *openalContext;
ALsizei openalSourceCount;
openalSource_t openalSources[256];
EAXSet alEAXSet;
EAXGet alEAXGet;
EAXSetBufferMode alEAXSetBufferMode;
EAXGetBufferMode alEAXGetBufferMode;
idEFXFile EFXDatabase;
bool efxloaded;
// latches
static bool useOpenAL;
static bool useEAXReverb;
// mark available during initialization, or through an explicit test
static int EAXAvailable;
static idCVar s_noSound;
static idCVar s_quadraticFalloff;
static idCVar s_drawSounds;
static idCVar s_minVolume6;
static idCVar s_dotbias6;
static idCVar s_minVolume2;
static idCVar s_dotbias2;
static idCVar s_spatializationDecay;
static idCVar s_showStartSound;
static idCVar s_maxSoundsPerShader;
static idCVar s_reverse;
static idCVar s_showLevelMeter;
static idCVar s_meterTopTime;
static idCVar s_volume;
static idCVar s_constantAmplitude;
static idCVar s_playDefaultSound;
static idCVar s_useOcclusion;
static idCVar s_subFraction;
static idCVar s_globalFraction;
static idCVar s_doorDistanceAdd;
static idCVar s_singleEmitter;
static idCVar s_numberOfSpeakers;
static idCVar s_force22kHz;
static idCVar s_clipVolumes;
static idCVar s_realTimeDecoding;
static idCVar s_libOpenAL;
static idCVar s_useOpenAL;
static idCVar s_useEAXReverb;
static idCVar s_muteEAXReverb;
static idCVar s_decompressionLimit;
static idCVar s_slowAttenuate;
static idCVar s_enviroSuitCutoffFreq;
static idCVar s_enviroSuitCutoffQ;
static idCVar s_enviroSuitSkipLowpass;
static idCVar s_enviroSuitSkipReverb;
static idCVar s_reverbTime;
static idCVar s_reverbFeedback;
static idCVar s_enviroSuitVolumeScale;
static idCVar s_skipHelltimeFX;
};
extern idSoundSystemLocal soundSystemLocal;
/*
===================================================================================
This class holds the actual wavefile bitmap, size, and info.
===================================================================================
*/
const int SCACHE_SIZE = MIXBUFFER_SAMPLES*20; // 1/2 of a second (aroundabout)
class idSoundSample {
public:
idSoundSample();
~idSoundSample();
idStr name; // name of the sample file
ID_TIME_T timestamp; // the most recent of all images used in creation, for reloadImages command
waveformatex_t objectInfo; // what are we caching
int objectSize; // size of waveform in samples, excludes the header
int objectMemSize; // object size in memory
byte * nonCacheData; // if it's not cached
byte * amplitudeData; // precomputed min,max amplitude pairs
ALuint openalBuffer; // openal buffer
bool hardwareBuffer;
bool defaultSound;
bool onDemand;
bool purged;
bool levelLoadReferenced; // so we can tell which samples aren't needed any more
int LengthIn44kHzSamples() const;
ID_TIME_T GetNewTimeStamp( void ) const;
void MakeDefault(); // turns it into a beep
void Load(); // loads the current sound based on name
void Reload( bool force ); // reloads if timestamp has changed, or always if force
void PurgeSoundSample(); // frees all data
void CheckForDownSample(); // down sample if required
bool FetchFromCache( int offset, const byte **output, int *position, int *size, const bool allowIO );
};
/*
===================================================================================
Sound sample decoder.
===================================================================================
*/
class idSampleDecoder {
public:
static void Init( void );
static void Shutdown( void );
static idSampleDecoder *Alloc( void );
static void Free( idSampleDecoder *decoder );
static int GetNumUsedBlocks( void );
static int GetUsedBlockMemory( void );
virtual ~idSampleDecoder( void ) {}
virtual void Decode( idSoundSample *sample, int sampleOffset44k, int sampleCount44k, float *dest ) = 0;
virtual void ClearDecoder( void ) = 0;
virtual idSoundSample * GetSample( void ) const = 0;
virtual int GetLastDecodeTime( void ) const = 0;
};
/*
===================================================================================
The actual sound cache.
===================================================================================
*/
class idSoundCache {
public:
idSoundCache();
~idSoundCache();
idSoundSample * FindSound( const idStr &fname, bool loadOnDemandOnly );
const int GetNumObjects( void ) { return listCache.Num(); }
const idSoundSample * GetObject( const int index ) const;
void ReloadSounds( bool force );
void BeginLevelLoad();
void EndLevelLoad();
void PrintMemInfo( MemInfo_t *mi );
private:
bool insideLevelLoad;
idList<idSoundSample*> listCache;
};
#endif /* !__SND_LOCAL_H__ */
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