mirror of
https://github.com/DrBeef/JKXR.git
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4597b03873
Opens in Android Studio but haven't even tried to build it yet (it won't.. I know that much!)
243 lines
7.1 KiB
C
243 lines
7.1 KiB
C
/*
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===========================================================================
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Copyright (C) 1999 - 2005, Id Software, Inc.
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Copyright (C) 2000 - 2013, Raven Software, Inc.
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Copyright (C) 2001 - 2013, Activision, Inc.
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Copyright (C) 2013 - 2015, OpenJK contributors
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This file is part of the OpenJK source code.
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OpenJK is free software; you can redistribute it and/or modify it
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under the terms of the GNU General Public License version 2 as
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published by the Free Software Foundation.
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This program 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 this program; if not, see <http://www.gnu.org/licenses/>.
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===========================================================================
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*/
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// snd_local.h -- private sound definations
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#ifndef SND_LOCAL_H
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#define SND_LOCAL_H
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#include "../qcommon/q_shared.h"
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#include "../qcommon/qcommon.h"
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#include "snd_public.h"
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#include "../mp3code/mp3struct.h"
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#if defined(_MSC_VER) && !defined(WIN64)
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#define USE_OPENAL
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#endif
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// Open AL Specific
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#ifdef USE_OPENAL
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#include "OpenAL/al.h"
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#include "OpenAL/alc.h"
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#include "eax/eax.h"
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#include "eax/EaxMan.h"
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/*#elif defined MACOS_X
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#include <OpenAL/al.h>
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#include <OpenAL/alc.h>
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#else
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#include <AL/al.h>
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#include <AL/alc.h>*/
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#endif
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// Added for Open AL to know when to mute all sounds (e.g when app. loses focus)
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void S_AL_MuteAllSounds(qboolean bMute);
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//from SND_AMBIENT
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extern void AS_Init( void );
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extern void AS_Free( void );
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#define PAINTBUFFER_SIZE 1024
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// !!! if this is changed, the asm code must change !!!
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typedef struct {
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int left; // the final values will be clamped to +/- 0x00ffff00 and shifted down
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int right;
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} portable_samplepair_t;
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// keep this enum in sync with the table "sSoundCompressionMethodStrings" -ste
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//
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typedef enum
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{
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ct_16 = 0, // formerly ct_NONE in EF1, now indicates 16-bit samples (the default)
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ct_MP3,
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//
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ct_NUMBEROF // used only for array sizing
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} SoundCompressionMethod_t;
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typedef struct sfx_s {
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short *pSoundData;
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bool bDefaultSound; // couldn't be loaded, so use buzz
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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
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short iLastLevelUsedOn; // used for cacheing purposes
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SoundCompressionMethod_t eSoundCompressionMethod;
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MP3STREAM *pMP3StreamHeader; // NULL ptr unless this sfx_t is an MP3. Use Z_Malloc and Z_Free
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int iSoundLengthInSamples; // length in samples, always kept as 16bit now so this is #shorts (watch for stereo later for music?)
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char sSoundName[MAX_QPATH];
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int iLastTimeUsed;
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float fVolRange; // used to set the highest volume this sample has at load time - used for lipsynching
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// Open AL
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#ifdef USE_OPENAL
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ALuint Buffer;
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#endif
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char *lipSyncData;
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struct sfx_s *next; // only used because of hash table when registering
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} sfx_t;
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typedef struct {
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int channels;
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int samples; // mono samples in buffer
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int submission_chunk; // don't mix less than this #
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int samplebits;
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int speed;
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byte *buffer;
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} dma_t;
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#define START_SAMPLE_IMMEDIATE 0x7fffffff
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// Open AL specific
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#ifdef USE_OPENAL
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typedef struct
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{
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ALuint BufferID;
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ALuint Status;
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char *Data;
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} STREAMINGBUFFER;
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#endif
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#define NUM_STREAMING_BUFFERS 4
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#define STREAMING_BUFFER_SIZE 4608 // 4 decoded MP3 frames
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#define QUEUED 1
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#define UNQUEUED 2
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typedef struct
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{
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// back-indented fields new in TA codebase, will re-format when MP3 code finished -ste
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// note: field missing in TA: qboolean loopSound; // from an S_AddLoopSound call, cleared each frame
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//
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int startSample; // START_SAMPLE_IMMEDIATE = set immediately on next mix
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int entnum; // to allow overriding a specific sound
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soundChannel_t entchannel; // to allow overriding a specific sound
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int leftvol; // 0-255 volume after spatialization
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int rightvol; // 0-255 volume after spatialization
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int master_vol; // 0-255 volume before spatialization
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vec3_t origin; // only use if fixed_origin is set
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qboolean fixed_origin; // use origin instead of fetching entnum's origin
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sfx_t *thesfx; // sfx structure
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qboolean loopSound; // from an S_AddLoopSound call, cleared each frame
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//
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MP3STREAM MP3StreamHeader;
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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)
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int iMP3SlidingDecodeWritePos;
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int iMP3SlidingDecodeWindowPos;
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// Open AL specific
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bool bLooping; // Signifies if this channel / source is playing a looping sound
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// bool bAmbient; // Signifies if this channel / source is playing a looping ambient sound
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bool bProcessed; // Signifies if this channel / source has been processed
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bool bStreaming; // Set to true if the data needs to be streamed (MP3 or dialogue)
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#ifdef USE_OPENAL
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STREAMINGBUFFER buffers[NUM_STREAMING_BUFFERS]; // AL Buffers for streaming
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ALuint alSource; // Open AL Source
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#endif
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bool bPlaying; // Set to true when a sound is playing on this channel / source
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int iStartTime; // Time playback of Source begins
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int lSlotID; // ID of Slot rendering Source's environment (enables a send to this FXSlot)
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} channel_t;
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#define WAV_FORMAT_PCM 1
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#define WAV_FORMAT_ADPCM 2 // not actually implemented, but is the value that you get in a header
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#define WAV_FORMAT_MP3 3 // not actually used this way, but just ensures we don't match one of the legit formats
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typedef struct {
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int format;
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int rate;
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int width;
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int channels;
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int samples;
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int dataofs; // chunk starts this many bytes from file start
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} wavinfo_t;
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//====================================================================
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#define MAX_CHANNELS 32
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extern channel_t s_channels[MAX_CHANNELS];
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extern int s_paintedtime;
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extern int s_rawend;
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extern vec3_t listener_origin;
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extern dma_t dma;
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#define MAX_RAW_SAMPLES 16384
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extern portable_samplepair_t s_rawsamples[MAX_RAW_SAMPLES];
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portable_samplepair_t *S_GetRawSamplePointer(); // TA added this, but it just returns the s_rawsamples[] array above. Oh well...
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extern cvar_t *s_allowDynamicMusic;
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extern cvar_t *s_initsound;
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extern cvar_t *s_khz;
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extern cvar_t *s_mixahead;
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extern cvar_t *s_nosound;
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extern cvar_t *s_separation;
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extern cvar_t *s_show;
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extern cvar_t *s_testsound;
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extern cvar_t *s_volume;
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extern cvar_t *s_volumeVoice;
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wavinfo_t GetWavinfo (const char *name, byte *wav, int wavlength);
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qboolean S_LoadSound( sfx_t *sfx );
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void S_PaintChannels(int endtime);
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// picks a channel based on priorities, empty slots, number of channels
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channel_t *S_PickChannel(int entnum, int entchannel);
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// spatializes a channel
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void S_Spatialize(channel_t *ch);
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//////////////////////////////////
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//
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// new stuff from TA codebase
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byte *SND_malloc(int iSize, sfx_t *sfx);
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void SND_setup();
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int SND_FreeOldestSound(sfx_t *pButNotThisOne = NULL);
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void SND_TouchSFX(sfx_t *sfx);
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void S_DisplayFreeMemory(void);
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void S_memoryLoad(sfx_t *sfx);
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//
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//////////////////////////////////
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#include "cl_mp3.h"
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#endif // #ifndef SND_LOCAL_H
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