forked from 0ad/0ad
1148 lines
24 KiB
C++
Executable File
1148 lines
24 KiB
C++
Executable File
#include "precompiled.h"
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#include "res/res.h"
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#include "res/snd.h"
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#include <sstream>
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#ifdef __APPLE__
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# include <OpenAL/alut.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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# include <AL/alut.h>
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#endif
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// Linux OpenAL puts the Ogg Vorbis extension enums in alexttypes.h
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#ifdef OS_LINUX
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# include <AL/alexttypes.h>
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#endif
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// for DLL-load hack in alc_init
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#ifdef _WIN32
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#include "sysdep/win/win_internal.h"
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#endif
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#ifdef _MSC_VER
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#pragma comment(lib, "openal32.lib")
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#pragma comment(lib, "alut.lib")
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#endif
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static const char* alc_dev_name = 0;
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// default: use OpenAL default device.
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static float listener_pos[3];
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// rationale for "buffer" handle and separate handle for each sound instance:
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// - need access to sound instances to fade them out / loop for an unknown period
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// - access via handle for safety
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// - don't want to reload sound data every time => need one central instance
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// that owns the data
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// - want to support normal reload mechanism (for consistency if not necessity)
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// - could hack something via h_find / if so, create new handle with fn_key = 0,
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// but that would break reloading and is dodgy. we will create a new handle
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// type instead.
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static void al_check()
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{
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const char* str = 0;
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switch(alGetError())
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{
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case AL_NO_ERROR:
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return;
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#define ERR(name) case name: str = #name; break;
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ERR(AL_INVALID_NAME);
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ERR(AL_INVALID_ENUM);
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ERR(AL_INVALID_VALUE);
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ERR(AL_INVALID_OPERATION);
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ERR(AL_OUT_OF_MEMORY);
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}
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debug_out("openal error: %s", str);
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debug_warn("OpenAL error");
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}
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// convenience function (this is called from 2 places)
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// buffer allocation overhead is very low -> no suballocator needed
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static ALuint al_create_buffer(ALvoid* data, ALsizei size, ALenum al_fmt, ALsizei al_freq)
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{
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ALuint al_buf;
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alGenBuffers(1, &al_buf);
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alBufferData(al_buf, al_fmt, data, size, al_freq);
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al_check();
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return al_buf;
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}
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///////////////////////////////////////////////////////////////////////////////
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//
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// AL source suballocator: allocate all available sources up-front and
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// pass them out as needed (alGenSources is quite slow, taking 3..5 ms per
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// source returned). also responsible for enforcing user-specified limit
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// on total number of sources (to reduce mixing cost on low-end systems).
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//
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///////////////////////////////////////////////////////////////////////////////
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// stack
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static const int AL_SRC_MAX = 64;
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// regardless of sound card caps, we won't use more than this
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// (64 is just overkill).
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static ALuint al_srcs[AL_SRC_MAX];
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static int al_src_used = 0;
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static int al_src_cap = AL_SRC_MAX;
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// user-set limit on how many sources may be used
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static int al_src_allocated;
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// called from al_init.
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static void al_src_init()
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{
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// grab as many sources as possible and count how many we get.
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for(int i = 0; i < AL_SRC_MAX; i++)
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{
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alGenSources(1, &al_srcs[i]);
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// we've reached the limit, no more are available.
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if(alGetError() != AL_NO_ERROR)
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break;
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assert(alIsSource(al_srcs[i]));
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al_src_allocated++;
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debug_out("init %p\n", al_srcs[i]);
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}
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// limit user's cap to what we actually got.
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if(al_src_cap > al_src_allocated)
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al_src_cap = al_src_allocated;
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// make sure we got the minimum guaranteed by OpenAL.
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assert(al_src_allocated >= 16);
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}
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// release all sources on freelist (currently stack).
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// all sources must have been returned to us via al_src_free.
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// called from al_shutdown.
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static void al_src_shutdown()
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{
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assert(al_src_used == 0);
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alDeleteSources(al_src_allocated, al_srcs);
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}
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static ALuint al_src_alloc()
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{
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// no more to give
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if(al_src_used >= al_src_cap)
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return 0;
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ALuint al_src = al_srcs[al_src_used++];
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assert(alIsSource(al_src));
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return al_src;
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}
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static void al_src_free(ALuint al_src)
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{
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al_srcs[--al_src_used] = al_src;
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assert(0 <= al_src_used && al_src_used < al_src_cap);
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}
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int snd_set_max_src(int cap)
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{
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// non-positive - bogus.
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if(cap <= 0)
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{
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debug_warn("snd_set_max_src: cap <= 0");
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return -1;
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}
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// cap must be at least as much as we're currently using.
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if(cap < al_src_used)
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cap = al_src_used;
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// either cap is legit (less than what we allocated in al_src_init),
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// or al_src_init wasn't called yet. note: we accept anything in the
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// second case, as al_src_init will sanity-check al_src_cap.
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if(!al_src_allocated || cap < al_src_allocated)
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{
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al_src_cap = cap;
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return 0;
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}
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// user is requesting a cap higher than what we actually allocated.
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// that's fine (not an error), but we won't set the cap, since it
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// determines how many sources may be returned.
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else
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return -1;
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}
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///////////////////////////////////////////////////////////////////////////////
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//
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// OpenAL init / shutdown
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//
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///////////////////////////////////////////////////////////////////////////////
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// called as late as possible, i.e. the first time sound/music is played
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// (either from module init there, or from the play routine itself).
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// this delays library load, leading to faster perceived app startup.
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// registers an atexit routine for cleanup.
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// no harm if called more than once.
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static ALCcontext* alc_ctx;
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static ALCdevice* alc_dev;
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static void alc_shutdown()
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{
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alcMakeContextCurrent(0);
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alcDestroyContext(alc_ctx);
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alcCloseDevice(alc_dev);
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}
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static int alc_init()
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{
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int ret = 0;
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// HACK: OpenAL loads and unloads these DLLs several times on Windows.
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// we hold a reference to prevent the actual unload, thus speeding up
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// sound startup by 100..400 ms. everything works ATM;
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// hopefully, OpenAL doesn't rely on them actually being unloaded.
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#ifdef _WIN32
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HMODULE dlls[3];
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dlls[0] = LoadLibrary("wrap_oal.dll");
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dlls[1] = LoadLibrary("setupapi.dll");
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dlls[2] = LoadLibrary("wdmaud.drv");
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#endif
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alc_dev = alcOpenDevice((ALubyte*)alc_dev_name);
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if(alc_dev)
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{
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alc_ctx = alcCreateContext(alc_dev, 0); // no attrlist needed
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if(alc_ctx)
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alcMakeContextCurrent(alc_ctx);
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}
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ALCenum err = alcGetError(alc_dev);
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if(err != ALC_NO_ERROR || !alc_dev || !alc_ctx)
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{
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debug_out("alc_init failed. alc_dev=%p alc_ctx=%p err=%d", alc_dev, alc_ctx, err);
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ret = -1;
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}
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// release DLL references, so BoundsChecker doesn't complain at exit
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#ifdef _WIN32
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for(int i = 0; i < ARRAY_SIZE(dlls); i++)
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if(dlls[i] != INVALID_HANDLE_VALUE)
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FreeLibrary(dlls[i]);
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#endif
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return ret;
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}
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static bool al_initialized = false;
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// called from each sound_open, and from snd_dev_set
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static int al_init()
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{
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// only take action on first call, OR after snd_dev_set calls us again
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if(al_initialized)
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return 0;
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al_initialized = true;
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ONCE(atexit(alc_shutdown));
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// we might init OpenAL several times, but must register
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// the atexit function only once!
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CHECK_ERR(alc_init());
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al_src_init(); // can't fail
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return 0;
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}
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static void al_shutdown()
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{
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al_initialized = false;
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alc_shutdown();
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}
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// if OpenAL hasn't been initialized yet, we only remember the device
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// name, which will be set when alc_init is later called; otherwise,
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// OpenAL is reinitialized to use the desired device.
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// (this is to speed up the common case of retrieving a device name from
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// config files and setting it; OpenAL doesn't have to be loaded until
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// sounds are actually played).
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// return 0 to indicate success, or the status returned while initializing
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// OpenAL.
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static int al_reinit()
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{
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if(!al_initialized)
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return 0;
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// was already using another device; now re-init
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// (stops all currently playing sounds)
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return al_init();
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}
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///////////////////////////////////////////////////////////////////////////////
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//
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// device enumeration: list all devices and allow the user to choose one,
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// in case the default device has problems.
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//
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///////////////////////////////////////////////////////////////////////////////
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static const char* devs;
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// set by snd_dev_prepare_enum; used by snd_dev_next.
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// consists of back-to-back C strings, terminated by an extra '\0'.
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// (this is taken straight from OpenAL; dox say this format may change).
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// prepare to enumerate all device names (this resets the list returned by
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// snd_dev_next). return 0 on success, otherwise -1 (only if the requisite
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// OpenAL extension isn't available). on failure, a "cannot change device"
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// message should be presented to the user, and snd_dev_set need not be
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// called; OpenAL will use its default device.
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// may be called each time the device list is needed.
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int snd_dev_prepare_enum()
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{
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if(alcIsExtensionPresent(0, (ALubyte*)"ALC_ENUMERATION_EXT") != AL_TRUE)
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return -1;
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devs = (const char*)alcGetString(0, ALC_DEVICE_SPECIFIER);
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return 0;
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}
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// return the next device name, or 0 if all have been returned.
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// do not call unless snd_dev_prepare_enum succeeded!
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// not thread-safe! (static data from snd_dev_prepare_enum is used)
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const char* snd_dev_next()
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{
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if(!*devs)
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return 0;
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const char* dev = devs;
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devs += strlen(dev)+1;
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return dev;
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}
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// tell OpenAL to use the specified device (0 for default) in future.
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//
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// if OpenAL hasn't been initialized yet, we only remember the device
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// name, which will be set when snd_init is later called; otherwise,
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// OpenAL is reinitialized to use the desired device (thus stopping all
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// active sounds). we go to this trouble to speed up perceived load times:
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// OpenAL doesn't need to be loaded until sounds are actually played.
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//
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// return 0 on success, or the status returned while re-initializing OpenAL.
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int snd_dev_set(const char* new_alc_dev_name)
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{
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// requesting a specific device
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if(new_alc_dev_name)
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{
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// already using that device - done
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if(alc_dev_name && !strcmp(alc_dev_name, new_alc_dev_name))
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return 0;
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// store name (need to copy it, since we snd_init later,
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// and it must then still be valid)
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static char buf[32];
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strncpy(buf, new_alc_dev_name, 32-1);
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alc_dev_name = buf;
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}
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// requesting default device
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else
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{
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// already using default device - done
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if(alc_dev_name == 0)
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return 0;
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alc_dev_name = 0;
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}
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return al_reinit();
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}
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///////////////////////////////////////////////////////////////////////////////
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//
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// IO for streams
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//
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///////////////////////////////////////////////////////////////////////////////
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// one stream apiece for music and voiceover (narration during tutorial).
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// allowing more is possible, but would be inefficent due to seek overhead.
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// set this limit to catch questionable usage (e.g. streaming normal sounds).
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static const int MAX_STREAMS = 2;
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static const int MIN_BUFFERS = 2;
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static const int MAX_IOS = 4;
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static const int TOTAL_IOS = MAX_STREAMS * MAX_IOS;
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static const size_t RAW_BUF_SIZE = 32*KB;
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static void* raw_buf;
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struct VSrc;
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struct IO
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{
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Handle hio;
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void* raw_buf;
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//
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void* vs;
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// so we can issue the next
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};
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// advantage over per-stream queue: we don't have to realloc a buffer
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// for every IO issue.
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static IO ios[TOTAL_IOS];
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static uint next_issue;
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static uint next_complete;
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static uint available_ios = TOTAL_IOS;
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static int io_init_ios()
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{
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const size_t total_size = RAW_BUF_SIZE * TOTAL_IOS;
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raw_buf = mem_alloc(total_size, 4096);
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if(!raw_buf)
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return ERR_NO_MEM;
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char* p = (char*)raw_buf;
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for(int i = 0; i < TOTAL_IOS; i++)
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{
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ios[i].raw_buf = p;
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p += RAW_BUF_SIZE;
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}
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}
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static int io_free_ios()
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{
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// !wait for all IOs to terminate; do not issue new ones!
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mem_free(raw_buf);
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memset(ios, 0, sizeof(ios));
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}
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static int io_issue(VSrc* vs, Handle hf)
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{
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if(!available_ios)
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return 0;
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IO* io = &ios[next_issue];
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// slot->hio = vfs_stream(hf, RAW_BUF_SIZE, slot->raw_buf);
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CHECK_ERR(io->hio);
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next_issue = (next_issue + 1) % TOTAL_IOS;
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available_ios--;
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return 0;
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}
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/*
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static int sound_add_buffer(VSrc* vs, void* buf, size_t size);
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static int sound_update(VSrc* vs);
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*/
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// error return value checked by snd_update
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static int io_check_complete()
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{
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for(;;)
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{
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// ring buffer is empty; nothing to wait on
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if(next_complete == next_issue)
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break;
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IO* slot = &ios[next_complete];
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// check if first IO is finished; if not, bail.
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int is_complete = vfs_io_complete(slot->hio);
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CHECK_ERR(is_complete);
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if(is_complete == 0)
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break;
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next_complete = (next_complete+1) % TOTAL_IOS;
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void* buf;
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size_t size;
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CHECK_ERR(vfs_wait_io(slot->hio, buf, size));
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// returns immediately
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// rationale: could issue right after we determine an IO is complete -
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// it might take a while for OpenAL to copy the buffer, and we could
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// start transferring in that time. however, there will be enough IOs
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// in-flight to cover long intervals between calls, so this isn't a
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// problem. it would also require more slots, since we wouldn't have
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// discarded the IO slot yet. finally, we want to limit issues to the
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// actual rate of data consumption - the IO shouldn't run ahead.
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// therefore, issue from sound_update for each finished buffer.
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// CHECK_ERR(sound_add_buffer(slot->s, buf, size));
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CHECK_ERR(vfs_discard_io(slot->hio));
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}
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return 0;
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}
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///////////////////////////////////////////////////////////////////////////////
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//
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// sound data provider
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//
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///////////////////////////////////////////////////////////////////////////////
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struct SndData
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{
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bool stream;
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// clip
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ALuint al_buf;
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// stream
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Handle hf;
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ALenum al_fmt;
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ALsizei al_freq;
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int queued_buffers;
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};
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H_TYPE_DEFINE(SndData);
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static void SndData_init(SndData* sd, va_list args)
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{
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sd->stream = va_arg(args, bool);
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}
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static void SndData_dtor(SndData* sd)
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{
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alDeleteBuffers(1, &sd->al_buf);
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al_check();
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vfs_close(sd->hf);
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}
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static int SndData_reload(SndData* sd, const char* fn, Handle)
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{
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int ret = -1;
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if(sd->stream)
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{
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sd->hf = vfs_open(fn);
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CHECK_ERR(sd->hf);
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// for(int i = 0; i < MAX_IOS; i++)
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// CHECK_ERR(io_issue(s, vs->hf));
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}
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else
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{
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}
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// freed in bailout ladder
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void* file = 0;
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{
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size_t file_size;
|
|
CHECK_ERR(vfs_load(fn, file, file_size));
|
|
|
|
|
|
//
|
|
// detect sound format by checking file extension
|
|
//
|
|
|
|
ALvoid* al_data;
|
|
ALsizei al_size;
|
|
|
|
const char* ext = strrchr(fn, '.');
|
|
// .. OGG (data will be passed directly to OpenAL)
|
|
if(ext && !stricmp(ext, ".ogg"))
|
|
{
|
|
// first use of OGG: check if OpenAL extension is available.
|
|
// note: this is required! OpenAL does its init here.
|
|
static int ogg_supported = -1;
|
|
if(ogg_supported == -1)
|
|
ogg_supported = alIsExtensionPresent((ALubyte*)"AL_EXT_vorbis")? 1 : 0;
|
|
if(!ogg_supported)
|
|
{
|
|
ret = -1;
|
|
goto fail;
|
|
}
|
|
|
|
al_data = file;
|
|
al_size = (ALsizei)file_size;
|
|
sd->al_fmt = AL_FORMAT_VORBIS_EXT;
|
|
sd->al_freq = 0;
|
|
}
|
|
// .. WAV
|
|
else if(ext && !stricmp(ext, ".wav"))
|
|
{
|
|
ALbyte* memory = (ALbyte*)file;
|
|
ALboolean al_loop; // unused
|
|
alutLoadWAVMemory(memory, &sd->al_fmt, &al_data, &al_size, &sd->al_freq, &al_loop);
|
|
}
|
|
// .. unknown extension
|
|
else
|
|
{
|
|
ret = -1;
|
|
goto fail;
|
|
}
|
|
|
|
sd->al_buf = al_create_buffer(al_data, al_size, sd->al_fmt, sd->al_freq);
|
|
|
|
ret = 0;
|
|
fail:
|
|
mem_free(file);
|
|
|
|
}
|
|
|
|
return ret;
|
|
}
|
|
|
|
|
|
// open and return a handle to a sound file's data
|
|
static Handle snd_data_load(const char* const fn, const bool stream)
|
|
{
|
|
//
|
|
//
|
|
//
|
|
//
|
|
// TODO: unique when stream - no reload
|
|
//
|
|
//
|
|
//
|
|
//
|
|
uint flags = 0;
|
|
return h_alloc(H_SndData, fn, flags, stream);
|
|
}
|
|
|
|
|
|
// close the xxx <y> and set y to 0.
|
|
static int snd_data_free(Handle& hsd)
|
|
{
|
|
return h_free(hsd, H_SndData);
|
|
}
|
|
|
|
|
|
// snd_data_get_buf return value
|
|
enum BufRet
|
|
{
|
|
BUF_OK = 0,
|
|
BUF_EOF = 1,
|
|
// otherwise, a negative error code.
|
|
};
|
|
|
|
static int snd_data_get_buf(Handle hsd, ALuint& al_buf)
|
|
{
|
|
H_DEREF(hsd, SndData, sd);
|
|
|
|
// clip: just return buffer (which was created in snd_data_load)
|
|
if(sd->al_buf)
|
|
{
|
|
al_buf = sd->al_buf;
|
|
return BUF_OK;
|
|
}
|
|
|
|
return -1;
|
|
}
|
|
|
|
|
|
///////////////////////////////////////////////////////////////////////////////
|
|
//
|
|
// sound instance
|
|
//
|
|
///////////////////////////////////////////////////////////////////////////////
|
|
|
|
struct VSrc
|
|
{
|
|
ALuint al_src;
|
|
|
|
// handle to this VSrc, so that it can close itself
|
|
Handle hvs;
|
|
|
|
// associated sound data
|
|
Handle hsd;
|
|
|
|
// - can't have 2 active instances of a streamed sound, so make sure
|
|
// caller is aware of the limitation by requiring them to set this.
|
|
// - data layer can't know if src is asking for a clip's buffer the first
|
|
// time, or whether "EOF" (clip done). src must not ask for second
|
|
// buffer if stream = true
|
|
//
|
|
// alternatives:
|
|
// - pass source info down to snd_data. increased coupling, bad interface
|
|
// - if src gets same buffer returned as last call by snd_data_get_buf,
|
|
// assume clip EOF. not watertight! buffer may change during OpenAL
|
|
// reinit (e.g. if changing provider)
|
|
bool stream;
|
|
|
|
ALfloat pos[3];
|
|
ALfloat gain;
|
|
ALboolean loop;
|
|
ALboolean relative;
|
|
|
|
float static_pri;
|
|
float cur_pri;
|
|
};
|
|
|
|
H_TYPE_DEFINE(VSrc);
|
|
|
|
|
|
|
|
|
|
|
|
|
|
static int vsrc_grant_src(VSrc* vs)
|
|
{
|
|
// already playing - bail
|
|
if(vs->al_src)
|
|
return 0;
|
|
|
|
// try to alloc source
|
|
vs->al_src = al_src_alloc();
|
|
// called from 2 places: sound_play can't know if a source is available,
|
|
// so this isn't an error
|
|
if(!vs->al_src)
|
|
return -1;
|
|
|
|
debug_out("got %p\n", vs->al_src);
|
|
|
|
// OpenAL docs don't specify default values, so initialize everything
|
|
// ourselves to be sure. note: alSourcefv param is not const.
|
|
float zero3[3] = { 0.0f, 0.0f, 0.0f };
|
|
alSourcefv(vs->al_src, AL_VELOCITY, zero3);
|
|
alSourcefv(vs->al_src, AL_DIRECTION, zero3);
|
|
alSourcef(vs->al_src, AL_ROLLOFF_FACTOR, 0.0f);
|
|
alSourcei(vs->al_src, AL_SOURCE_RELATIVE, AL_TRUE);
|
|
al_check();
|
|
|
|
// we only now got a source, so latch previous settings
|
|
// don't use snd_set*; this way is easiest
|
|
alSourcef(vs->al_src, AL_GAIN, vs->gain);
|
|
alSourcefv(vs->al_src, AL_POSITION, vs->pos);
|
|
alSourcei(vs->al_src, AL_LOOPING, vs->loop);
|
|
|
|
ALuint al_buf;
|
|
int ret = snd_data_get_buf(vs->hsd, al_buf);
|
|
if(ret == BUF_OK)
|
|
alSourceQueueBuffers(vs->al_src, 1, &al_buf);
|
|
else
|
|
debug_warn("snd_data_get_buf failed");
|
|
al_check();
|
|
|
|
alSourcePlay(vs->al_src);
|
|
al_check();
|
|
return 0;
|
|
}
|
|
|
|
|
|
static int vsrc_reclaim_src(VSrc* vs)
|
|
{
|
|
// not playing - bail
|
|
if(!vs->al_src)
|
|
return 0;
|
|
|
|
alSourceStop(vs->al_src);
|
|
|
|
// free all buffers
|
|
if(vs->stream)
|
|
{
|
|
int num_bufs;
|
|
alGetSourcei(vs->al_src, AL_BUFFERS_PROCESSED, &num_bufs);
|
|
// all are considered processed, as the source has been stopped
|
|
for(int i = 0; i < num_bufs; i++)
|
|
{
|
|
ALuint al_buf;
|
|
alSourceUnqueueBuffers(vs->al_src, 1, &al_buf);
|
|
alDeleteBuffers(1, &al_buf);
|
|
}
|
|
}
|
|
|
|
al_check();
|
|
|
|
debug_out("free %p\n", vs->al_src);
|
|
al_src_free(vs->al_src);
|
|
return 0;
|
|
}
|
|
|
|
|
|
static void vsrc_update(VSrc* vs)
|
|
{
|
|
ALint num_finished_buffers;
|
|
alGetSourcei(vs->al_src, AL_BUFFERS_PROCESSED, &num_finished_buffers);
|
|
al_check();
|
|
|
|
|
|
|
|
if(!vs->stream)
|
|
{
|
|
if(num_finished_buffers == 1)
|
|
snd_free(vs->hvs);
|
|
}
|
|
/*
|
|
|
|
if(vs->flags & SF_STREAMING)
|
|
for(int i = 0; i < num_finished_buffers; i++)
|
|
{
|
|
ALuint al_buf;
|
|
alSourceUnqueueBuffers(vs->al_src, 1, &al_buf);
|
|
alDeleteBuffers(1, &al_buf);
|
|
al_check();
|
|
|
|
CHECK_ERR(io_issue(s, vs->hf));
|
|
}
|
|
*/
|
|
}
|
|
|
|
|
|
|
|
|
|
|
|
static void list_remove(VSrc*);
|
|
|
|
|
|
static void VSrc_init(VSrc* vs, va_list args)
|
|
{
|
|
vs->stream = va_arg(args, bool);
|
|
}
|
|
|
|
|
|
static void VSrc_dtor(VSrc* vs)
|
|
{
|
|
list_remove(vs);
|
|
vsrc_reclaim_src(vs);
|
|
snd_data_free(vs->hsd);
|
|
}
|
|
|
|
|
|
static int VSrc_reload(VSrc* vs, const char* def_fn, Handle hvs)
|
|
{
|
|
// cannot wait till play(), need to init here:
|
|
// must load OpenAL so that snd_data_load can check for OGG extension.
|
|
al_init();
|
|
|
|
/*
|
|
void* def_file;
|
|
size_t def_size;
|
|
CHECK_ERR(vfs_load(def_fn, def_file, def_size));
|
|
std::istringstream def(std::string((char*)def_file, (int)def_size));
|
|
mem_free(def_file);
|
|
|
|
std::string snd_data_fn;
|
|
float gain_percent;
|
|
def >> snd_data_fn;
|
|
def >> gain_percent;*/
|
|
|
|
float gain_percent = 100.0;
|
|
std::string snd_data_fn = def_fn;
|
|
|
|
vs->gain = gain_percent / 100.0f;
|
|
|
|
// store so we can shut ourselves down via sound_free when done playing
|
|
vs->hvs = hvs;
|
|
|
|
vs->hsd = snd_data_load(snd_data_fn.c_str(), vs->stream);
|
|
|
|
return 0;
|
|
}
|
|
|
|
|
|
// open and return a handle to the sound <fn>.
|
|
// stream: default false
|
|
Handle snd_open(const char* const fn, const bool stream)
|
|
{
|
|
return h_alloc(H_VSrc, fn, RES_UNIQUE, stream);
|
|
}
|
|
|
|
|
|
// close the sound <hs> and set hs to 0.
|
|
int snd_free(Handle& hvs)
|
|
{
|
|
return h_free(hvs, H_VSrc);
|
|
}
|
|
|
|
|
|
int snd_set_pos(Handle hvs, float x, float y, float z, bool relative /* = false */)
|
|
{
|
|
H_DEREF(hvs, VSrc, vs);
|
|
|
|
vs->pos[0] = x; vs->pos[1] = y; vs->pos[2] = z;
|
|
vs->relative = relative;
|
|
|
|
if(vs->al_src)
|
|
{
|
|
alSourcefv(vs->al_src, AL_POSITION, vs->pos);
|
|
alSourcei(vs->al_src, AL_SOURCE_RELATIVE, vs->relative);
|
|
al_check();
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
|
|
int snd_set_gain(Handle hvs, float gain)
|
|
{
|
|
H_DEREF(hvs, VSrc, vs);
|
|
|
|
vs->gain = gain;
|
|
|
|
if(vs->al_src)
|
|
{
|
|
alSourcef(vs->al_src, AL_GAIN, vs->gain);
|
|
al_check();
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
|
|
int snd_set_loop(Handle hvs, bool loop)
|
|
{
|
|
H_DEREF(hvs, VSrc, vs);
|
|
|
|
vs->loop = loop;
|
|
|
|
if(vs->al_src)
|
|
{
|
|
alSourcei(vs->al_src, AL_LOOPING, vs->loop);
|
|
al_check();
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
static float norm(const float* v)
|
|
{
|
|
return v[0]*v[0] + v[1]*v[1] + v[2]*v[2];
|
|
}
|
|
|
|
static float dist3d_sqr(const float* v1, const float* v2)
|
|
{
|
|
const float dx = v1[0] - v2[0];
|
|
const float dy = v1[1] - v2[1];
|
|
const float dz = v1[2] - v2[2];
|
|
return dx*dx + dy*dy + dz*dz;
|
|
}
|
|
|
|
|
|
const float MAX_DIST2 = 1000.0f;
|
|
|
|
static void vsrc_calc_cur_pri(VSrc* vs)
|
|
{
|
|
float d2; // euclidean distance to listener (squared)
|
|
if(vs->relative)
|
|
d2 = norm(vs->pos);
|
|
else
|
|
d2 = dist3d_sqr(vs->pos, listener_pos);
|
|
|
|
// farther away than OpenAL cutoff - no sound contribution
|
|
if(d2 > MAX_DIST2)
|
|
{
|
|
vs->cur_pri = -1.0f;
|
|
return;
|
|
|
|
// TODO: make sure these never play, even if no sounds active
|
|
}
|
|
|
|
// scale priority down exponentially
|
|
float e = d2 / MAX_DIST2; // 0.0f (close) .. 1.0f (far)
|
|
const float falloff = 10.0f;
|
|
vs->cur_pri = vs->static_pri * pow(falloff, e);
|
|
}
|
|
|
|
|
|
static bool vsrc_greater(const VSrc* const s1, const VSrc* const s2)
|
|
{
|
|
return s1->cur_pri > s2->cur_pri;
|
|
}
|
|
|
|
|
|
|
|
|
|
|
|
///////////////////////////////////////////////////////////////////////////////
|
|
//
|
|
// list of sounds
|
|
//
|
|
///////////////////////////////////////////////////////////////////////////////
|
|
|
|
// currently active sounds - needed to update them, i.e. remove old buffers
|
|
// and enqueue just finished async buffers. this can't happen from
|
|
// io_check_complete alone - see dox there.
|
|
|
|
|
|
// sorted in ascending order of current priority
|
|
// (we remove low pri items, and have to move down everything after them,
|
|
// so they should come last)
|
|
typedef std::vector<VSrc*> VSources;
|
|
typedef VSources::iterator It;
|
|
static VSources vsources;
|
|
|
|
// don't need to sort - that's done during full update
|
|
static void list_add(VSrc* vs)
|
|
{
|
|
vsources.push_back(vs);
|
|
}
|
|
|
|
static void list_foreach(void(*cb)(VSrc*))
|
|
{
|
|
std::for_each(vsources.begin(), vsources.end(), cb);
|
|
}
|
|
|
|
|
|
// O(N)!
|
|
static void list_remove(VSrc* vs)
|
|
{
|
|
for(size_t i = 0; i < vsources.size(); i++)
|
|
if(vsources[i] == vs)
|
|
{
|
|
vsources[i] = 0;
|
|
return;
|
|
}
|
|
|
|
debug_warn("list_remove: VSrc not found");
|
|
}
|
|
|
|
|
|
static int list_update()
|
|
{
|
|
// prune NULL-entries, so code below doesn't have to check if non-NULL
|
|
// (these were removed, but we didn't shuffle everything down to save time)
|
|
It src = vsources.begin(), dst = vsources.begin();
|
|
size_t valid_entries = vsources.size();
|
|
for(;;)
|
|
{
|
|
if(src == vsources.end())
|
|
break;
|
|
if(*src == 0)
|
|
{
|
|
++src;
|
|
valid_entries--;
|
|
continue;
|
|
}
|
|
*dst = *src;
|
|
++dst;
|
|
++src;
|
|
}
|
|
vsources.resize(valid_entries);
|
|
|
|
// update current priorities (a function of static priority and distance)
|
|
std::for_each(vsources.begin(), vsources.end(), vsrc_calc_cur_pri);
|
|
|
|
// sort by descending current priority
|
|
std::sort(vsources.begin(), vsources.end(), vsrc_greater);
|
|
|
|
It it;
|
|
It first_unimportant = vsources.begin() + min((int)vsources.size(), al_src_cap);
|
|
|
|
// reclaim source from the less important vsources
|
|
for(it = first_unimportant; it != vsources.end(); ++it)
|
|
{
|
|
VSrc* vs = *it;
|
|
if(vs->al_src)
|
|
vsrc_reclaim_src(vs);
|
|
|
|
if(!vs->loop)
|
|
snd_free(vs->hvs);
|
|
}
|
|
|
|
// grant each of the most important vsources a source
|
|
for(it = vsources.begin(); it != first_unimportant; ++it)
|
|
{
|
|
VSrc* vs = *it;
|
|
if(!vs->al_src)
|
|
vsrc_grant_src(vs);
|
|
}
|
|
|
|
|
|
std::for_each(vsources.begin(), vsources.end(), vsrc_update);
|
|
|
|
return 0;
|
|
}
|
|
|
|
|
|
int snd_play(Handle hs)
|
|
{
|
|
H_DEREF(hs, VSrc, vs);
|
|
|
|
list_add(vs);
|
|
|
|
// optimization (don't want to do full update here - too slow)
|
|
// either we get a source and playing begins immediately, or it'll be
|
|
// taken care of on next update
|
|
vsrc_grant_src(vs);
|
|
return 0;
|
|
}
|
|
|
|
|
|
|
|
|
|
|
|
///////////////////////////////////////////////////////////////////////////////
|
|
//
|
|
// sound engine
|
|
//
|
|
///////////////////////////////////////////////////////////////////////////////
|
|
|
|
|
|
|
|
|
|
|
|
int snd_update(float lx, float ly, float lz)
|
|
{
|
|
float* p = listener_pos;
|
|
p[0] = lx; p[1] = ly; p[2] = lz;
|
|
|
|
list_update();
|
|
|
|
CHECK_ERR(io_check_complete());
|
|
|
|
return 0;
|
|
}
|
|
|
|
|
|
int snd_shutdown()
|
|
{
|
|
// io_shutdown();
|
|
return 0;
|
|
}
|