jkxr/Projects/Android/jni/OpenJK/code/client/snd_local.h
Simon 4597b03873 Initial Commit
Opens in Android Studio but haven't even tried to build it yet (it won't.. I know that much!)
2022-09-18 16:37:21 +01:00

243 lines
7.1 KiB
C

/*
===========================================================================
Copyright (C) 1999 - 2005, Id Software, Inc.
Copyright (C) 2000 - 2013, Raven Software, Inc.
Copyright (C) 2001 - 2013, Activision, Inc.
Copyright (C) 2013 - 2015, OpenJK contributors
This file is part of the OpenJK source code.
OpenJK is free software; you can redistribute it and/or modify it
under the terms of the GNU General Public License version 2 as
published by the Free Software Foundation.
This program 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 this program; if not, see <http://www.gnu.org/licenses/>.
===========================================================================
*/
// snd_local.h -- private sound definations
#ifndef SND_LOCAL_H
#define SND_LOCAL_H
#include "../qcommon/q_shared.h"
#include "../qcommon/qcommon.h"
#include "snd_public.h"
#include "../mp3code/mp3struct.h"
#if defined(_MSC_VER) && !defined(WIN64)
#define USE_OPENAL
#endif
// Open AL Specific
#ifdef USE_OPENAL
#include "OpenAL/al.h"
#include "OpenAL/alc.h"
#include "eax/eax.h"
#include "eax/EaxMan.h"
/*#elif defined MACOS_X
#include <OpenAL/al.h>
#include <OpenAL/alc.h>
#else
#include <AL/al.h>
#include <AL/alc.h>*/
#endif
// Added for Open AL to know when to mute all sounds (e.g when app. loses focus)
void S_AL_MuteAllSounds(qboolean bMute);
//from SND_AMBIENT
extern void AS_Init( void );
extern void AS_Free( void );
#define PAINTBUFFER_SIZE 1024
// !!! if this is changed, the asm code must change !!!
typedef struct {
int left; // the final values will be clamped to +/- 0x00ffff00 and shifted down
int right;
} portable_samplepair_t;
// keep this enum in sync with the table "sSoundCompressionMethodStrings" -ste
//
typedef enum
{
ct_16 = 0, // formerly ct_NONE in EF1, now indicates 16-bit samples (the default)
ct_MP3,
//
ct_NUMBEROF // used only for array sizing
} SoundCompressionMethod_t;
typedef struct sfx_s {
short *pSoundData;
bool bDefaultSound; // couldn't be loaded, so use buzz
bool bInMemory; // not in Memory, set qtrue when loaded, and qfalse when its buffers are freed up because of being old, so can be reloaded
short iLastLevelUsedOn; // used for cacheing purposes
SoundCompressionMethod_t eSoundCompressionMethod;
MP3STREAM *pMP3StreamHeader; // NULL ptr unless this sfx_t is an MP3. Use Z_Malloc and Z_Free
int iSoundLengthInSamples; // length in samples, always kept as 16bit now so this is #shorts (watch for stereo later for music?)
char sSoundName[MAX_QPATH];
int iLastTimeUsed;
float fVolRange; // used to set the highest volume this sample has at load time - used for lipsynching
// Open AL
#ifdef USE_OPENAL
ALuint Buffer;
#endif
char *lipSyncData;
struct sfx_s *next; // only used because of hash table when registering
} sfx_t;
typedef struct {
int channels;
int samples; // mono samples in buffer
int submission_chunk; // don't mix less than this #
int samplebits;
int speed;
byte *buffer;
} dma_t;
#define START_SAMPLE_IMMEDIATE 0x7fffffff
// Open AL specific
#ifdef USE_OPENAL
typedef struct
{
ALuint BufferID;
ALuint Status;
char *Data;
} STREAMINGBUFFER;
#endif
#define NUM_STREAMING_BUFFERS 4
#define STREAMING_BUFFER_SIZE 4608 // 4 decoded MP3 frames
#define QUEUED 1
#define UNQUEUED 2
typedef struct
{
// back-indented fields new in TA codebase, will re-format when MP3 code finished -ste
// note: field missing in TA: qboolean loopSound; // from an S_AddLoopSound call, cleared each frame
//
int startSample; // START_SAMPLE_IMMEDIATE = set immediately on next mix
int entnum; // to allow overriding a specific sound
soundChannel_t entchannel; // to allow overriding a specific sound
int leftvol; // 0-255 volume after spatialization
int rightvol; // 0-255 volume after spatialization
int master_vol; // 0-255 volume before spatialization
vec3_t origin; // only use if fixed_origin is set
qboolean fixed_origin; // use origin instead of fetching entnum's origin
sfx_t *thesfx; // sfx structure
qboolean loopSound; // from an S_AddLoopSound call, cleared each frame
//
MP3STREAM MP3StreamHeader;
byte MP3SlidingDecodeBuffer[50000/*12000*/]; // typical back-request = -3072, so roughly double is 6000 (safety), then doubled again so the 6K pos is in the middle of the buffer)
int iMP3SlidingDecodeWritePos;
int iMP3SlidingDecodeWindowPos;
// Open AL specific
bool bLooping; // Signifies if this channel / source is playing a looping sound
// bool bAmbient; // Signifies if this channel / source is playing a looping ambient sound
bool bProcessed; // Signifies if this channel / source has been processed
bool bStreaming; // Set to true if the data needs to be streamed (MP3 or dialogue)
#ifdef USE_OPENAL
STREAMINGBUFFER buffers[NUM_STREAMING_BUFFERS]; // AL Buffers for streaming
ALuint alSource; // Open AL Source
#endif
bool bPlaying; // Set to true when a sound is playing on this channel / source
int iStartTime; // Time playback of Source begins
int lSlotID; // ID of Slot rendering Source's environment (enables a send to this FXSlot)
} channel_t;
#define WAV_FORMAT_PCM 1
#define WAV_FORMAT_ADPCM 2 // not actually implemented, but is the value that you get in a header
#define WAV_FORMAT_MP3 3 // not actually used this way, but just ensures we don't match one of the legit formats
typedef struct {
int format;
int rate;
int width;
int channels;
int samples;
int dataofs; // chunk starts this many bytes from file start
} wavinfo_t;
//====================================================================
#define MAX_CHANNELS 32
extern channel_t s_channels[MAX_CHANNELS];
extern int s_paintedtime;
extern int s_rawend;
extern vec3_t listener_origin;
extern dma_t dma;
#define MAX_RAW_SAMPLES 16384
extern portable_samplepair_t s_rawsamples[MAX_RAW_SAMPLES];
portable_samplepair_t *S_GetRawSamplePointer(); // TA added this, but it just returns the s_rawsamples[] array above. Oh well...
extern cvar_t *s_allowDynamicMusic;
extern cvar_t *s_initsound;
extern cvar_t *s_khz;
extern cvar_t *s_mixahead;
extern cvar_t *s_nosound;
extern cvar_t *s_separation;
extern cvar_t *s_show;
extern cvar_t *s_testsound;
extern cvar_t *s_volume;
extern cvar_t *s_volumeVoice;
wavinfo_t GetWavinfo (const char *name, byte *wav, int wavlength);
qboolean S_LoadSound( sfx_t *sfx );
void S_PaintChannels(int endtime);
// picks a channel based on priorities, empty slots, number of channels
channel_t *S_PickChannel(int entnum, int entchannel);
// spatializes a channel
void S_Spatialize(channel_t *ch);
//////////////////////////////////
//
// new stuff from TA codebase
byte *SND_malloc(int iSize, sfx_t *sfx);
void SND_setup();
int SND_FreeOldestSound(sfx_t *pButNotThisOne = NULL);
void SND_TouchSFX(sfx_t *sfx);
void S_DisplayFreeMemory(void);
void S_memoryLoad(sfx_t *sfx);
//
//////////////////////////////////
#include "cl_mp3.h"
#endif // #ifndef SND_LOCAL_H