mirror of
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e6f3713169
OpenAL devices can disconnect, and with some luck they're back after a few seconds. This especially seems to happen with Intels Windows GPU driver and display-audio when switching the resolution or enabling fullscreen, see #209 Now a disconnect is detected and we try to reset the device for 20 seconds, hoping it comes back. This needs at least openal-soft 1.17.0 to build and 1.20.0 or newer to actually work. Also added missing stub functions in openal_stub.cpp (used by dedicated server so it doesn't have to link libopenal)
901 lines
28 KiB
C++
901 lines
28 KiB
C++
/*
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===========================================================================
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Doom 3 GPL Source Code
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Copyright (C) 1999-2011 id Software LLC, a ZeniMax Media company.
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This file is part of the Doom 3 GPL Source Code ("Doom 3 Source Code").
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Doom 3 Source Code is free software: you can redistribute it and/or modify
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it under the terms of the GNU General Public License as published by
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the Free Software Foundation, either version 3 of the License, or
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(at your option) any later version.
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Doom 3 Source Code is distributed in the hope that it will be useful,
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but WITHOUT ANY WARRANTY; without even the implied warranty of
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MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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GNU General Public License for more details.
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You should have received a copy of the GNU General Public License
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along with Doom 3 Source Code. If not, see <http://www.gnu.org/licenses/>.
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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.
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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.
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===========================================================================
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*/
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#ifndef __SND_LOCAL_H__
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#define __SND_LOCAL_H__
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#ifdef ID_DEDICATED
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// stub-only mode: AL_API and ALC_API shouldn't refer to any dll-stuff
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// because the implemenations are in openal_stub.cpp
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// this is ensured by defining AL_LIBTYPE_STATIC before including the AL headers
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#define AL_LIBTYPE_STATIC
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#endif
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#include <AL/al.h>
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#include <AL/alc.h>
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#include <AL/alext.h>
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#include "framework/UsercmdGen.h"
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#include "sound/efxlib.h"
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#include "sound/sound.h"
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// demo sound commands
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typedef enum {
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SCMD_STATE, // followed by a load game state
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SCMD_PLACE_LISTENER,
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SCMD_ALLOC_EMITTER,
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SCMD_FREE,
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SCMD_UPDATE,
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SCMD_START,
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SCMD_MODIFY,
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SCMD_STOP,
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SCMD_FADE
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} soundDemoCommand_t;
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const int SOUND_MAX_CHANNELS = 8;
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const int SOUND_DECODER_FREE_DELAY = 1000 * MIXBUFFER_SAMPLES / USERCMD_MSEC; // four seconds
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const int PRIMARYFREQ = 44100; // samples per second
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const float SND_EPSILON = 1.0f / 32768.0f; // if volume is below this, it will always multiply to zero
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const int ROOM_SLICES_IN_BUFFER = 10;
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class idAudioBuffer;
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class idWaveFile;
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class idSoundCache;
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class idSoundSample;
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class idSampleDecoder;
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class idSoundWorldLocal;
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/*
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===================================================================================
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General extended waveform format structure.
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Use this for all NON PCM formats.
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===================================================================================
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*/
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#ifdef WIN32
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#pragma pack(1)
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#endif
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struct waveformatex_s {
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word wFormatTag; /* format type */
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word nChannels; /* number of channels (i.e. mono, stereo...) */
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dword nSamplesPerSec; /* sample rate */
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dword nAvgBytesPerSec; /* for buffer estimation */
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word nBlockAlign; /* block size of data */
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word wBitsPerSample; /* Number of bits per sample of mono data */
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word cbSize; /* The count in bytes of the size of
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extra information (after cbSize) */
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} PACKED;
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typedef waveformatex_s waveformatex_t;
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/* OLD general waveform format structure (information common to all formats) */
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struct waveformat_s {
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word wFormatTag; /* format type */
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word nChannels; /* number of channels (i.e. mono, stereo, etc.) */
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dword nSamplesPerSec; /* sample rate */
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dword nAvgBytesPerSec; /* for buffer estimation */
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word nBlockAlign; /* block size of data */
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} PACKED;
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typedef waveformat_s waveformat_t;
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/* flags for wFormatTag field of WAVEFORMAT */
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enum {
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WAVE_FORMAT_TAG_PCM = 1,
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WAVE_FORMAT_TAG_OGG = 2
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};
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/* specific waveform format structure for PCM data */
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struct pcmwaveformat_s {
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waveformat_t wf;
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word wBitsPerSample;
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} PACKED;
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typedef pcmwaveformat_s pcmwaveformat_t;
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#ifndef mmioFOURCC
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#define mmioFOURCC( ch0, ch1, ch2, ch3 ) \
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( (dword)(byte)(ch0) | ( (dword)(byte)(ch1) << 8 ) | \
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( (dword)(byte)(ch2) << 16 ) | ( (dword)(byte)(ch3) << 24 ) )
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#endif
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#define fourcc_riff mmioFOURCC('R', 'I', 'F', 'F')
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struct waveformatextensible_s {
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waveformatex_t Format;
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union {
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word wValidBitsPerSample; /* bits of precision */
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word wSamplesPerBlock; /* valid if wBitsPerSample==0*/
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word wReserved; /* If neither applies, set to zero*/
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} Samples;
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dword dwChannelMask; /* which channels are */
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/* present in stream */
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int SubFormat;
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} PACKED;
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typedef waveformatextensible_s waveformatextensible_t;
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typedef dword fourcc;
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/* RIFF chunk information data structure */
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struct mminfo_s {
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fourcc ckid; /* chunk ID */
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dword cksize; /* chunk size */
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fourcc fccType; /* form type or list type */
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dword dwDataOffset; /* offset of data portion of chunk */
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} PACKED;
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typedef mminfo_s mminfo_t;
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#ifdef WIN32
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#pragma pack()
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#endif
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/*
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===================================================================================
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idWaveFile
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===================================================================================
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*/
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class idWaveFile {
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public:
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idWaveFile( void );
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~idWaveFile( void );
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int Open( const char* strFileName, waveformatex_t* pwfx = NULL );
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int OpenFromMemory( short* pbData, int ulDataSize, waveformatextensible_t* pwfx );
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int Read( byte* pBuffer, int dwSizeToRead, int *pdwSizeRead );
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int Seek( int offset );
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int Close( void );
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int ResetFile( void );
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int GetOutputSize( void ) { return mdwSize; }
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int GetMemorySize( void ) { return mMemSize; }
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waveformatextensible_t mpwfx; // Pointer to waveformatex structure
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private:
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idFile * mhmmio; // I/O handle for the WAVE
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mminfo_t mck; // Multimedia RIFF chunk
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mminfo_t mckRiff; // used when opening a WAVE file
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dword mdwSize; // size in samples
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dword mMemSize; // size of the wave data in memory
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dword mseekBase;
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ID_TIME_T mfileTime;
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bool mbIsReadingFromMemory;
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short * mpbData;
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short * mpbDataCur;
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dword mulDataSize;
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void * ogg; // only !NULL when !s_realTimeDecoding
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bool isOgg;
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private:
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int ReadMMIO( void );
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int OpenOGG( const char* strFileName, waveformatex_t* pwfx = NULL );
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int ReadOGG( byte* pBuffer, int dwSizeToRead, int *pdwSizeRead );
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int CloseOGG( void );
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};
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/*
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===================================================================================
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Encapsulates functionality of a DirectSound buffer.
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===================================================================================
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*/
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class idAudioBuffer {
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public:
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virtual int Play( dword dwPriority=0, dword dwFlags=0 ) = 0;
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virtual int Stop( void ) = 0;
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virtual int Reset( void ) = 0;
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virtual bool IsSoundPlaying( void ) = 0;
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virtual void SetVolume( float x ) = 0;
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};
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/*
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===================================================================================
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idSoundEmitterLocal
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===================================================================================
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*/
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typedef enum {
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REMOVE_STATUS_INVALID = -1,
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REMOVE_STATUS_ALIVE = 0,
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REMOVE_STATUS_WAITSAMPLEFINISHED = 1,
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REMOVE_STATUS_SAMPLEFINISHED = 2
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} removeStatus_t;
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class idSoundFade {
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public:
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int fadeStart44kHz;
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int fadeEnd44kHz;
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float fadeStartVolume; // in dB
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float fadeEndVolume; // in dB
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void Clear();
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float FadeDbAt44kHz( int current44kHz );
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};
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class SoundFX {
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protected:
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bool initialized;
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int channel;
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int maxlen;
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float* buffer;
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float continuitySamples[4];
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float param;
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public:
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SoundFX() { channel = 0; buffer = NULL; initialized = false; maxlen = 0; memset( continuitySamples, 0, sizeof( float ) * 4 ); };
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virtual ~SoundFX() { if ( buffer ) delete buffer; };
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virtual void Initialize() { };
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virtual void ProcessSample( float* in, float* out ) = 0;
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void SetChannel( int chan ) { channel = chan; };
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int GetChannel() { return channel; };
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void SetContinuitySamples( float in1, float in2, float out1, float out2 ) { continuitySamples[0] = in1; continuitySamples[1] = in2; continuitySamples[2] = out1; continuitySamples[3] = out2; }; // FIXME?
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void GetContinuitySamples( float& in1, float& in2, float& out1, float& out2 ) { in1 = continuitySamples[0]; in2 = continuitySamples[1]; out1 = continuitySamples[2]; out2 = continuitySamples[3]; };
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void SetParameter( float val ) { param = val; };
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};
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class SoundFX_Lowpass : public SoundFX {
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public:
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virtual void ProcessSample( float* in, float* out );
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};
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class SoundFX_LowpassFast : public SoundFX {
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float freq;
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float res;
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float a1, a2, a3;
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float b1, b2;
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public:
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virtual void ProcessSample( float* in, float* out );
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void SetParms( float p1 = 0, float p2 = 0, float p3 = 0 );
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};
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class SoundFX_Comb : public SoundFX {
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int currentTime;
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public:
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virtual void Initialize();
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virtual void ProcessSample( float* in, float* out );
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};
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class FracTime {
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public:
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int time;
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float frac;
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void Set( int val ) { time = val; frac = 0; };
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void Increment( float val ) { frac += val; while ( frac >= 1.f ) { time++; frac--; } };
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};
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enum {
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PLAYBACK_RESET,
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PLAYBACK_ADVANCING
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};
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class idSoundChannel;
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class idSlowChannel {
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bool active;
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const idSoundChannel* chan;
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int playbackState;
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int triggerOffset;
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FracTime newPosition;
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int newSampleOffset;
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FracTime curPosition;
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int curSampleOffset;
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SoundFX_LowpassFast lowpass;
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// functions
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void GenerateSlowChannel( FracTime& playPos, int sampleCount44k, float* finalBuffer );
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float GetSlowmoSpeed();
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public:
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void AttachSoundChannel( const idSoundChannel *chan );
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void Reset();
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void GatherChannelSamples( int sampleOffset44k, int sampleCount44k, float *dest );
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bool IsActive() { return active; };
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FracTime GetCurrentPosition() { return curPosition; };
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};
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class idSoundChannel {
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public:
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idSoundChannel( void );
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~idSoundChannel( void );
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void Clear( void );
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void Start( void );
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void Stop( void );
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void GatherChannelSamples( int sampleOffset44k, int sampleCount44k, float *dest ) const;
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void ALStop( void ); // free OpenAL resources if any
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bool triggerState;
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int trigger44kHzTime; // hardware time sample the channel started
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int triggerGame44kHzTime; // game time sample time the channel started
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soundShaderParms_t parms; // combines the shader parms and the per-channel overrides
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idSoundSample * leadinSample; // if not looped, this is the only sample
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s_channelType triggerChannel;
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const idSoundShader *soundShader;
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idSampleDecoder * decoder;
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float diversity;
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float lastVolume; // last calculated volume based on distance
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float lastV[6]; // last calculated volume for each speaker, so we can smoothly fade
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idSoundFade channelFade;
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bool triggered;
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ALuint openalSource;
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ALuint openalStreamingOffset;
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ALuint openalStreamingBuffer[3];
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ALuint lastopenalStreamingBuffer[3];
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bool stopped;
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bool disallowSlow;
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};
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class idSoundEmitterLocal : public idSoundEmitter {
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public:
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idSoundEmitterLocal( void );
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virtual ~idSoundEmitterLocal( void );
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//----------------------------------------------
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// the "time" parameters should be game time in msec, which is used to make queries
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// return deterministic values regardless of async buffer scheduling
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// a non-immediate free will let all currently playing sounds complete
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virtual void Free( bool immediate );
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// the parms specified will be the default overrides for all sounds started on this emitter.
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// NULL is acceptable for parms
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virtual void UpdateEmitter( const idVec3 &origin, int listenerId, const soundShaderParms_t *parms );
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// returns the length of the started sound in msec
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virtual int StartSound( const idSoundShader *shader, const s_channelType channel, float diversity = 0, int shaderFlags = 0, bool allowSlow = true /* D3XP */ );
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// can pass SCHANNEL_ANY
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virtual void ModifySound( const s_channelType channel, const soundShaderParms_t *parms );
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virtual void StopSound( const s_channelType channel );
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virtual void FadeSound( const s_channelType channel, float to, float over );
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virtual bool CurrentlyPlaying( void ) const;
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// can pass SCHANNEL_ANY
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virtual float CurrentAmplitude( void );
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// used for save games
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virtual int Index( void ) const;
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//----------------------------------------------
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void Clear( void );
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void OverrideParms( const soundShaderParms_t *base, const soundShaderParms_t *over, soundShaderParms_t *out );
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void CheckForCompletion( int current44kHzTime );
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void Spatialize( idVec3 listenerPos, int listenerArea, idRenderWorld *rw );
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idSoundWorldLocal * soundWorld; // the world that holds this emitter
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int index; // in world emitter list
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removeStatus_t removeStatus;
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idVec3 origin;
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int listenerId;
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soundShaderParms_t parms; // default overrides for all channels
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// the following are calculated in UpdateEmitter, and don't need to be archived
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float maxDistance; // greatest of all playing channel distances
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int lastValidPortalArea; // so an emitter that slides out of the world continues playing
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bool playing; // if false, no channel is active
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bool hasShakes;
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idVec3 spatializedOrigin; // the virtual sound origin, either the real sound origin,
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// or a point through a portal chain
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float realDistance; // in meters
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float distance; // in meters, this may be the straight-line distance, or
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// it may go through a chain of portals. If there
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// is not an open-portal path, distance will be > maxDistance
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// a single soundEmitter can have many channels playing from the same point
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idSoundChannel channels[SOUND_MAX_CHANNELS];
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idSlowChannel slowChannels[SOUND_MAX_CHANNELS];
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idSlowChannel GetSlowChannel( const idSoundChannel *chan );
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void SetSlowChannel( const idSoundChannel *chan, idSlowChannel slow );
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void ResetSlowChannel( const idSoundChannel *chan );
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// this is just used for feedback to the game or rendering system:
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// flashing lights and screen shakes. Because the material expression
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// evaluation doesn't do common subexpression removal, we cache the
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// last generated value
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int ampTime;
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float amplitude;
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};
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/*
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===================================================================================
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idSoundWorldLocal
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===================================================================================
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*/
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class s_stats {
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public:
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s_stats( void ) {
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rinuse = 0;
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runs = 1;
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timeinprocess = 0;
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missedWindow = 0;
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missedUpdateWindow = 0;
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activeSounds = 0;
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}
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int rinuse;
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int runs;
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int timeinprocess;
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int missedWindow;
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int missedUpdateWindow;
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int activeSounds;
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};
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typedef struct soundPortalTrace_s {
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int portalArea;
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const struct soundPortalTrace_s *prevStack;
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} soundPortalTrace_t;
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class idSoundWorldLocal : public idSoundWorld {
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public:
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virtual ~idSoundWorldLocal( void );
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// call at each map start
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virtual void ClearAllSoundEmitters( void );
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virtual void StopAllSounds( void );
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// get a new emitter that can play sounds in this world
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virtual idSoundEmitter *AllocSoundEmitter( void );
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// for load games
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virtual idSoundEmitter *EmitterForIndex( int index );
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// query data from all emitters in the world
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virtual float CurrentShakeAmplitudeForPosition( const int time, const idVec3 &listererPosition );
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// where is the camera/microphone
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// listenerId allows listener-private sounds to be added
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virtual void PlaceListener( const idVec3 &origin, const idMat3 &axis, const int listenerId, const int gameTime, const idStr& areaName );
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// fade all sounds in the world with a given shader soundClass
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// to is in Db (sigh), over is in seconds
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virtual void FadeSoundClasses( const int soundClass, const float to, const float over );
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// dumps the current state and begins archiving commands
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virtual void StartWritingDemo( idDemoFile *demo );
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virtual void StopWritingDemo( void );
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// read a sound command from a demo file
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virtual void ProcessDemoCommand( idDemoFile *readDemo );
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// background music
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virtual void PlayShaderDirectly( const char *name, int channel = -1 );
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// pause and unpause the sound world
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virtual void Pause( void );
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virtual void UnPause( void );
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virtual bool IsPaused( void );
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// avidump
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virtual void AVIOpen( const char *path, const char *name );
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virtual void AVIClose( void );
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// SaveGame Support
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virtual void WriteToSaveGame( idFile *savefile );
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virtual void ReadFromSaveGame( idFile *savefile );
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virtual void ReadFromSaveGameSoundChannel( idFile *saveGame, idSoundChannel *ch );
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virtual void ReadFromSaveGameSoundShaderParams( idFile *saveGame, soundShaderParms_t *params );
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virtual void WriteToSaveGameSoundChannel( idFile *saveGame, idSoundChannel *ch );
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virtual void WriteToSaveGameSoundShaderParams( idFile *saveGame, soundShaderParms_t *params );
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virtual void SetSlowmo( bool active );
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virtual void SetSlowmoSpeed( float speed );
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virtual void SetEnviroSuit( bool active );
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//=======================================
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idSoundWorldLocal( void );
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void Shutdown( void );
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void Init( idRenderWorld *rw );
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// update
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void ForegroundUpdate( int currentTime );
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void OffsetSoundTime( int offset44kHz );
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idSoundEmitterLocal * AllocLocalSoundEmitter();
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void CalcEars( int numSpeakers, idVec3 realOrigin, idVec3 listenerPos, idMat3 listenerAxis, float ears[6], float spatialize );
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void AddChannelContribution( idSoundEmitterLocal *sound, idSoundChannel *chan,
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int current44kHz, int numSpeakers, float *finalMixBuffer );
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void MixLoop( int current44kHz, int numSpeakers, float *finalMixBuffer );
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void AVIUpdate( void );
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void ResolveOrigin( const int stackDepth, const soundPortalTrace_t *prevStack, const int soundArea, const float dist, const idVec3& soundOrigin, idSoundEmitterLocal *def );
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float FindAmplitude( idSoundEmitterLocal *sound, const int localTime, const idVec3 *listenerPosition, const s_channelType channel, bool shakesOnly );
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//============================================
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idRenderWorld * rw; // for portals and debug drawing
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idDemoFile * writeDemo; // if not NULL, archive commands here
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idMat3 listenerAxis;
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idVec3 listenerPos; // position in meters
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int listenerPrivateId;
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idVec3 listenerQU; // position in "quake units"
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int listenerArea;
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idStr listenerAreaName;
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ALuint listenerEffect;
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ALuint listenerSlot;
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ALuint listenerFilter;
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int gameMsec;
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int game44kHz;
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int pause44kHz;
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int lastAVI44kHz; // determine when we need to mix and write another block
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idList<idSoundEmitterLocal *>emitters;
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idSoundFade soundClassFade[SOUND_MAX_CLASSES]; // for global sound fading
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// avi stuff
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idFile * fpa[6];
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idStr aviDemoPath;
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idStr aviDemoName;
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idSoundEmitterLocal * localSound; // just for playShaderDirectly()
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bool slowmoActive;
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float slowmoSpeed;
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bool enviroSuitActive;
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};
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/*
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===================================================================================
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idSoundSystemLocal
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===================================================================================
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*/
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typedef struct {
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ALuint handle;
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int startTime;
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idSoundChannel *chan;
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bool inUse;
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bool looping;
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bool stereo;
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} openalSource_t;
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class idSoundSystemLocal : public idSoundSystem {
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public:
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idSoundSystemLocal( ) {
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isInitialized = false;
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}
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// all non-hardware initialization
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virtual void Init( void );
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// shutdown routine
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virtual void Shutdown( void );
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// sound is attached to the window, and must be recreated when the window is changed
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virtual bool ShutdownHW( void );
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virtual bool InitHW( void );
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// async loop, called at 60Hz
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virtual int AsyncUpdate( int time );
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// async loop, when the sound driver uses a write strategy
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virtual int AsyncUpdateWrite( int time );
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// direct mixing called from the sound driver thread for OSes that support it
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virtual int AsyncMix( int soundTime, float *mixBuffer );
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virtual void SetMute( bool mute );
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virtual cinData_t ImageForTime( const int milliseconds, const bool waveform );
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int GetSoundDecoderInfo( int index, soundDecoderInfo_t &decoderInfo );
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// if rw == NULL, no portal occlusion or rendered debugging is available
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virtual idSoundWorld *AllocSoundWorld( idRenderWorld *rw );
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// specifying NULL will cause silence to be played
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virtual void SetPlayingSoundWorld( idSoundWorld *soundWorld );
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// some tools, like the sound dialog, may be used in both the game and the editor
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// This can return NULL, so check!
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virtual idSoundWorld *GetPlayingSoundWorld( void );
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virtual void BeginLevelLoad( void );
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virtual void EndLevelLoad( const char *mapString );
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virtual void PrintMemInfo( MemInfo_t *mi );
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virtual int IsEFXAvailable( void );
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|
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//-------------------------
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|
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int GetCurrent44kHzTime( void ) const;
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float dB2Scale( const float val ) const;
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int SamplesToMilliseconds( int samples ) const;
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int MillisecondsToSamples( int ms ) const;
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void DoEnviroSuit( float* samples, int numSamples, int numSpeakers );
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|
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ALuint AllocOpenALSource( idSoundChannel *chan, bool looping, bool stereo );
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|
void FreeOpenALSource( ALuint handle );
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// returns true if openalDevice is still available,
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// otherwise it will try to recover the device and return false while it's gone
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// (display audio sound devices sometimes disappear for a few seconds when switching resolution)
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bool CheckDeviceAndRecoverIfNeeded();
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idSoundCache * soundCache;
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idSoundWorldLocal * currentSoundWorld; // the one to mix each async tic
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|
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int olddwCurrentWritePos; // statistics
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int buffers; // statistics
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int CurrentSoundTime; // set by the async thread and only used by the main thread
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|
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unsigned int nextWriteBlock;
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|
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float realAccum[6*MIXBUFFER_SAMPLES+16];
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float * finalMixBuffer; // points inside realAccum at a 16 byte aligned boundary
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|
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bool isInitialized;
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bool muted;
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bool shutdown;
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|
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s_stats soundStats; // NOTE: updated throughout the code, not displayed anywhere
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|
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int meterTops[256];
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int meterTopsTime[256];
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|
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dword * graph;
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|
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float volumesDB[1200]; // dB to float volume conversion
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|
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idList<SoundFX*> fxList;
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|
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ALCdevice *openalDevice;
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ALCcontext *openalContext;
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ALsizei openalSourceCount;
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openalSource_t openalSources[256];
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|
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LPALGENEFFECTS alGenEffects;
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|
LPALDELETEEFFECTS alDeleteEffects;
|
|
LPALISEFFECT alIsEffect;
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|
LPALEFFECTI alEffecti;
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LPALEFFECTF alEffectf;
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LPALEFFECTFV alEffectfv;
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LPALGENFILTERS alGenFilters;
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LPALDELETEFILTERS alDeleteFilters;
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LPALISFILTER alIsFilter;
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LPALFILTERI alFilteri;
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LPALFILTERF alFilterf;
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LPALGENAUXILIARYEFFECTSLOTS alGenAuxiliaryEffectSlots;
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LPALDELETEAUXILIARYEFFECTSLOTS alDeleteAuxiliaryEffectSlots;
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LPALISAUXILIARYEFFECTSLOT alIsAuxiliaryEffectSlot;
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LPALAUXILIARYEFFECTSLOTI alAuxiliaryEffectSloti;
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|
|
idEFXFile EFXDatabase;
|
|
bool efxloaded;
|
|
// latches
|
|
static bool useEFXReverb;
|
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// mark available during initialization, or through an explicit test
|
|
static int EFXAvailable;
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|
|
|
// DG: for CheckDeviceAndRecoverIfNeeded()
|
|
LPALCRESETDEVICESOFT alcResetDeviceSOFT; // needs ALC_SOFT_HRTF extension
|
|
int resetRetryCount;
|
|
unsigned int lastCheckTime;
|
|
|
|
static idCVar s_noSound;
|
|
static idCVar s_device;
|
|
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_useEAXReverb;
|
|
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__ */
|