0ad/source/lib/res/snd.cpp

// OpenAL sound engine
//
// Copyright (c) 2004-5 Jan Wassenberg
//
// This program is free software; you can redistribute it and/or
// modify it under the terms of the GNU General Public License as
// published by the Free Software Foundation; either version 2 of the
// License, or (at your option) any later version.
//
// 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.
//
// Contact info:
//   Jan.Wassenberg@stud.uni-karlsruhe.de
//   http://www.stud.uni-karlsruhe.de/~urkt/

#include "precompiled.h"

#include "res/res.h"
#include "res/snd.h"

#include <sstream>	// to extract snd_open's definition file contents
#include <string>
#include <vector>
#include <algorithm>
#include <math.h>

#ifdef __APPLE__
# include <OpenAL/alut.h>
#else
# include <AL/al.h>
# include <AL/alc.h>
# include <AL/alut.h>	// alutLoadWAVMemory
#endif

// for AL_FORMAT_VORBIS_EXT decl on Linux
#ifdef OS_LINUX
# include <AL/alexttypes.h>
#endif

// for DLL-load hack in alc_init
#ifdef _WIN32
# include "sysdep/win/win_internal.h"
#endif


#define OGG_HACK
#include "ogghack.h"

#ifdef _MSC_VER
# pragma comment(lib, "openal32.lib")
# pragma comment(lib, "alut.lib")	// alutLoadWAVMemory
#endif

// components:
// - alc*: OpenAL context
//   readies OpenAL for use; allows specifying the device.
// - al_listener*: OpenAL listener
//   owns position/orientation and master gain.
// - al_buf*: OpenAL buffer suballocator
//   for convenience; also makes sure all have been freed at exit.
// - al_src*: OpenAL source suballocator
//   avoids high source alloc cost. also enforces user-set source limit.
// - al_init*: OpenAL startup mechanism
//   allows deferred init (speeding up start time) and runtime reset.
// - snd_dev*: device enumeration
//   lists names of all available devices (for sound options screen).
// - io_buf*: buffer suballocator
//   avoids frequent large alloc/frees and therefore heap fragmentation.
// - stream*: streaming
//   passes chunks of data (read via async I/O) to snd_data on request.
// - snd_data*: sound data provider
//   holds audio data (clip or stream) and returns OpenAL buffers on request.
// - hsd_list*: list of SndData instances
//   ensures all are freed when desired (despite being cached).
// - list*: list of active sounds.
//   sorts by priority for voice management, and has each VSrc update itself.
// - vsrc*: audio source
//   owns source properties and queue, references SndData.
// - vm*: voice management
//   grants the currently most 'important' sounds a hardware voice.


static bool al_initialized = false;
	// indicates OpenAL is ready for use. checked by other components
	// when deciding if they can pass settings changes to OpenAL directly,
	// or whether they need to be saved until init.


// used by snd_dev_set to reset OpenAL after device has been changed.
static int al_reinit();

// used by VSrc_reload to init on demand.
static int snd_init();
	
// used by al_shutdown to free all VSrc and SndData objects, respectively,
// so that they release their OpenAL sources and buffers.
static int list_free_all();
static void hsd_list_free_all();


// check if OpenAL indicates an error has occurred. it can only report
// 1 error at a time, so this is called after every OpenAL request.
static void al_check(const char* caller = "(unknown)")
{
	debug_assert(al_initialized);

	ALenum err = alGetError();
	if(err == AL_NO_ERROR)
		return;

	const char* str = (const char*)alGetString(err);
	debug_printf("openal error: %s; called from %s\n", str, caller);
	debug_warn("OpenAL error");
}


///////////////////////////////////////////////////////////////////////////////
//
// OpenAL context: readies OpenAL for use; allows specifying the device,
// in case there are problems with OpenAL's default choice.
//
///////////////////////////////////////////////////////////////////////////////

static const char* alc_dev_name = 0;
	// default (0): use OpenAL default device.
	// note: that's why this needs to be a pointer instead of an array.


// tell OpenAL to use the specified device in future.
// name = 0 reverts to OpenAL's default choice, which will also
// be used if this routine is never called.
//
// the device name is typically taken from a config file at init-time;
// the snd_dev* enumeration routines below are used to present a list
// of choices to the user in the options screen.
//
// if OpenAL hasn't yet been initialized (i.e. no sounds have been opened),
//   this just stores the device name for use when init does occur.
//   note: we can't check now if it's invalid (if so, init will fail).
// otherwise, we shut OpenAL down (thereby stopping all sounds) and
// re-initialize with the new device. that's fairly time-consuming,
// so preferably call this routine before sounds are loaded.
//
// return 0 on success, or the status returned by OpenAL re-init.
int snd_dev_set(const char* alc_new_dev_name)
{
	// requesting a specific device
	if(alc_new_dev_name)
	{
		// already using that device - done. (don't re-init)
		if(alc_dev_name && !strcmp(alc_dev_name, alc_new_dev_name))
			return 0;

		// store name (need to copy it, since alc_init is called later,
		// and it must then still be valid)
		static char buf[32];
		strcpy_s(buf, sizeof(buf), alc_new_dev_name);
		alc_dev_name = buf;
	}
	// requesting default device
	else
	{
		// already using default device - done. (don't re-init)
		if(alc_dev_name == 0)
			return 0;

		alc_dev_name = 0;
	}

	return al_reinit();
		// no-op if not initialized yet, otherwise re-init.
}


static ALCcontext* alc_ctx = 0;
static ALCdevice* alc_dev = 0;


static void alc_shutdown()
{
	if(alc_ctx)
	{
		alcMakeContextCurrent(0);
		alcDestroyContext(alc_ctx);
	}

	if(alc_dev)
		alcCloseDevice(alc_dev);
}


static int alc_init()
{
	int ret = 0;

	// HACK: OpenAL loads and unloads these DLLs several times on Windows.
	// we hold a reference to prevent the actual unload,
	// thus speeding up startup by 100..400 ms. everything works ATM;
	// hopefully, OpenAL doesn't rely on them actually being unloaded.

#ifdef _WIN32
	HMODULE dlls[3];
	dlls[0] = LoadLibrary("wrap_oal.dll");
	dlls[1] = LoadLibrary("setupapi.dll");
	dlls[2] = LoadLibrary("wdmaud.drv");
#endif

	// for reasons unknown, the NV native OpenAL implementation
	// causes an "invalid handle" exception internally when loaded
	// (it's not caused by the DLL load hack above). everything works and
	// we can continue normally; we just need to catch it to prevent the
	// unhandled exception filter from reporting it.
#ifdef _WIN32
	__try
	{
		alc_dev = alcOpenDevice((ALubyte*)alc_dev_name);
	}
	// if invalid handle, handle it; otherwise, continue handler search.
	__except(GetExceptionCode() == EXCEPTION_INVALID_HANDLE)
	{
		// ignore
	}
#else
	alc_dev = alcOpenDevice((ALubyte*)alc_dev_name);
#endif

	if(alc_dev)
	{
		alc_ctx = alcCreateContext(alc_dev, 0);	// no attrlist needed
		if(alc_ctx)
			alcMakeContextCurrent(alc_ctx);
	}

	// check if init succeeded.
	// some OpenAL implementations don't indicate failure here correctly;
	// we need to check if the device and context pointers are actually valid.
	ALCenum err = alcGetError(alc_dev);
	if(err != ALC_NO_ERROR || !alc_dev || !alc_ctx)
	{
		debug_printf("alc_init failed. alc_dev=%p alc_ctx=%p alc_dev_name=%s err=%d\n", alc_dev, alc_ctx, alc_dev_name, err);
		ret = -1;
	}

	// release DLL references, so BoundsChecker doesn't complain at exit.
#ifdef _WIN32
	for(int i = 0; i < ARRAY_SIZE(dlls); i++)
		if(dlls[i] != INVALID_HANDLE_VALUE)
			FreeLibrary(dlls[i]);
#endif

	return ret;
}


///////////////////////////////////////////////////////////////////////////////
//
// listener: owns position/orientation and master gain.
// if they're set before al_initialized, we pass the saved values to
// OpenAL immediately after init (instead of waiting until next update).
//
///////////////////////////////////////////////////////////////////////////////

static float al_listener_gain = 1.0;
static float al_listener_pos[3];
static float al_listener_orientation[6];
	// float view_direction[3], up_vector[3]; passed directly to OpenAL


// also called from al_init.
static void al_listener_latch()
{
	if(al_initialized)
	{
		alListenerf(AL_GAIN, al_listener_gain);
		alListenerfv(AL_POSITION, al_listener_pos);
		alListenerfv(AL_ORIENTATION, al_listener_orientation);
		al_check("al_listener_latch");
	}
}


// set amplitude modifier, which is effectively applied to all sounds.
// must be non-negative; 1 -> unattenuated, 0.5 -> -6 dB, 0 -> silence.
int snd_set_master_gain(float gain)
{
	if(gain < 0)
	{
		debug_warn("snd_set_master_gain: gain < 0");
		return ERR_INVALID_PARAM;
	}

	al_listener_gain = gain;

	al_listener_latch();
		// position will get sent too.
		// this isn't called often, so we don't care.

	return 0;
}


static void al_listener_set_pos(const float pos[3], const float dir[3], const float up[3])
{
	int i;
	for(i = 0; i < 3; i++)
		al_listener_pos[i] = pos[i];
	for(i = 0; i < 3; i++)
		al_listener_orientation[i] = dir[i];
	for(i = 0; i < 3; i++)
		al_listener_orientation[3+i] = up[i];

	al_listener_latch();
}


// return euclidean distance squared between listener and point.
// used to determine sound priority.
static float al_listener_dist_2(const float point[3])
{
	const float dx = al_listener_pos[0] - point[0];
	const float dy = al_listener_pos[1] - point[1];
	const float dz = al_listener_pos[2] - point[2];
	return dx*dx + dy*dy + dz*dz;
}


///////////////////////////////////////////////////////////////////////////////
//
// AL buffer suballocator: allocates buffers as needed (alGenBuffers is fast).
// this interface is a bit more convenient than the OpenAL routines, and we
// verify that all buffers have been freed at exit.
//
///////////////////////////////////////////////////////////////////////////////

static int al_bufs_outstanding;

// allocate a new buffer, and fill it with the specified data.
static ALuint al_buf_alloc(ALvoid* data, ALsizei size, ALenum al_fmt, ALsizei al_freq)
{
	ALuint al_buf;
	alGenBuffers(1, &al_buf);
	alBufferData(al_buf, al_fmt, data, size, al_freq);
	al_check("al_buf_alloc");

	al_bufs_outstanding++;
	return al_buf;
}


static void al_buf_free(ALuint al_buf)
{
	// no-op if 0 (needed in case SndData_reload fails -
	// sd->al_buf will not have been set)
	if(!al_buf)
		return;

	debug_assert(alIsBuffer(al_buf));
	alDeleteBuffers(1, &al_buf);
	al_check("al_buf_free");

	al_bufs_outstanding--;
}


// make sure all buffers have been returned to us via al_buf_free.
// called from al_shutdown.
static void al_buf_shutdown()
{
	if(al_bufs_outstanding != 0)
		debug_warn("al_buf_shutdown: not all buffers freed!");
}


///////////////////////////////////////////////////////////////////////////////
//
// AL source suballocator: allocate all available sources up-front and
// pass them out as needed (alGenSources is quite slow, taking 3..5 ms per
// source returned). also responsible for enforcing user-specified limit
// on total number of sources (to reduce mixing cost on low-end systems).
//
///////////////////////////////////////////////////////////////////////////////

static const uint AL_SRC_MAX = 64;
	// regardless of sound card caps, we won't use more than this ("enough").
	// necessary in case OpenAL doesn't limit #sources (e.g. if SW mixing).
static ALuint al_srcs[AL_SRC_MAX];
	// stack of sources (first allocated is [0])
static uint al_src_allocated;
	// number of valid sources in al_srcs[] (set by al_src_init)
static uint al_src_used = 0;
	// number of sources currently in use
static uint al_src_cap = AL_SRC_MAX;
	// user-set limit on how many sources may be used


// called from al_init.
static void al_src_init()
{
	// grab as many sources as possible and count how many we get.
	for(uint i = 0; i < AL_SRC_MAX; i++)
	{
		alGenSources(1, &al_srcs[i]);
		// we've reached the limit, no more are available.
		if(alGetError() != AL_NO_ERROR)
			break;
		debug_assert(alIsSource(al_srcs[i]));
		al_src_allocated++;
	}

	// limit user's cap to what we actually got.
	// (in case snd_set_max_src was called before this)
	if(al_src_cap > al_src_allocated)
		al_src_cap = al_src_allocated;

	// make sure we got the minimum guaranteed by OpenAL.
	debug_assert(al_src_allocated >= 16);
}


// release all sources on freelist (currently stack).
// all sources must have been returned to us via al_src_free.
// called from al_shutdown.
static void al_src_shutdown()
{
	debug_assert(al_src_used == 0);
	alDeleteSources(al_src_allocated, al_srcs);
	al_src_allocated = 0;
	al_check("al_src_shutdown");
}


static ALuint al_src_alloc()
{
	// no more to give
	if(al_src_used >= al_src_cap)
		return 0;
	ALuint al_src = al_srcs[al_src_used++];
	return al_src;
}


static void al_src_free(ALuint al_src)
{
	debug_assert(alIsSource(al_src));
	al_srcs[--al_src_used] = al_src;
	debug_assert(al_src_used < al_src_allocated);
		// don't compare against cap - it might have been
		// decreased to less than were in use.
}


// set maximum number of voices to play simultaneously,
// to reduce mixing cost on low-end systems.
// return 0 on success, or 1 if limit was ignored
// (e.g. if higher than an implementation-defined limit anyway).
int snd_set_max_voices(uint limit)
{
	// valid if cap is legit (less than what we allocated in al_src_init),
	// or if al_src_init hasn't been called yet. note: we accept anything
	// in the second case, as al_src_init will sanity-check al_src_cap.
	if(!al_src_allocated || limit < al_src_allocated)
	{
		al_src_cap = limit;
		return 0;
	}
	// user is requesting a cap higher than what we actually allocated.
	// that's fine (not an error), but we won't set the cap, since it
	// determines how many sources may be returned.
	else
		return 1;
}


///////////////////////////////////////////////////////////////////////////////
//
// OpenAL startup mechanism: allows deferring init until sounds are actually
// played, therefore speeding up perceived game start time.
// also resets OpenAL when settings (e.g. device) are changed at runtime.
//
///////////////////////////////////////////////////////////////////////////////

// called from each snd_open; no harm if called more than once.
static int al_init()
{
	// only take action on first call, OR when re-initializing.
	if(al_initialized)
		return 0;

	CHECK_ERR(alc_init());

	al_initialized = true;

	// these can't fail:
	al_src_init();
	al_listener_latch();

	return 0;
}


static void al_shutdown()
{
	// was never initialized - nothing to do.
	if(!al_initialized)
		return;

	// somewhat tricky: go through gyrations to free OpenAL resources.

	// .. free all active sounds so that they release their source.
	//    the SndData reference is also removed,
	//    but these remain open, since they are cached.
	list_free_all();

	// .. actually free all (still cached) SndData instances.
	hsd_list_free_all();

	// .. all sources and buffers have been returned to their suballocators.
	//    now free them all.
	al_src_shutdown();
	al_buf_shutdown();

	alc_shutdown();

	al_initialized = false;
}


// called from snd_dev_set (no other settings require re-init ATM).
static int al_reinit()
{
	// not yet initialized. settings have been saved, and will be
	// applied by the component init routines called from al_init.
	if(!al_initialized)
		return 0;

	// re-init (stops all currently playing sounds)
	al_shutdown();
	return al_init();
}


///////////////////////////////////////////////////////////////////////////////
//
// device enumeration: list all devices and allow the user to choose one,
// in case the default device has problems.
//
///////////////////////////////////////////////////////////////////////////////

static const char* devs;
	// set by snd_dev_prepare_enum; used by snd_dev_next.
	// consists of back-to-back C strings, terminated by an extra '\0'.
	// (this is taken straight from OpenAL; dox say this format may change).

// prepare to enumerate all device names (this resets the list returned by
// snd_dev_next). return 0 on success, otherwise -1 (only if the requisite
// OpenAL extension isn't available). on failure, a "cannot enum device"
// message should be presented to the user, and snd_dev_set need not be
// called; OpenAL will use its default device.
// may be called each time the device list is needed.
int snd_dev_prepare_enum()
{
	if(alcIsExtensionPresent(0, (ALubyte*)"ALC_ENUMERATION_EXT") != AL_TRUE)
		return ERR_NO_SYS;

	devs = (const char*)alcGetString(0, ALC_DEVICE_SPECIFIER);
	return 0;
}


// return the next device name, or 0 if all have been returned.
// do not call unless snd_dev_prepare_enum succeeded!
// not thread-safe! (static data from snd_dev_prepare_enum is used)
const char* snd_dev_next()
{
	if(!*devs)
		return 0;
	const char* dev = devs;
	devs += strlen(dev)+1;
	return dev;
}


///////////////////////////////////////////////////////////////////////////////
//
// stream: passes chunks of data (read via async I/O) to snd_data on request.
//
///////////////////////////////////////////////////////////////////////////////

// one stream apiece for music and voiceover (narration during tutorial).
// allowing more is possible, but would be inefficent due to seek overhead.
// set this limit to catch questionable usage (e.g. streaming normal sounds).
static const int MAX_STREAMS = 2;

// maximum IOs queued per stream.
static const int MAX_IOS = 4;

static const size_t STREAM_BUF_SIZE = 32*KiB;


//
// I/O buffer suballocator: avoids frequent large alloc/frees
// when streaming, and therefore heap fragmentation.
//
///////////////////////////////////////////////////////////////////////////////

static const int TOTAL_IOS = MAX_STREAMS * MAX_IOS;
static const size_t TOTAL_BUF_SIZE = TOTAL_IOS * STREAM_BUF_SIZE;

static void* io_bufs;
	// one large allocation for all buffers

static void* io_buf_freelist;
	// list of free buffers. start of buffer holds pointer to next in list.


static void io_buf_free(void* p)
{
	debug_assert(io_bufs <= p && p <= (char*)io_bufs+TOTAL_BUF_SIZE);
	*(void**)p = io_buf_freelist;
	io_buf_freelist = p;
}


// called from first io_buf_alloc.
static void io_buf_init()
{
	// allocate 1 big aligned block for all buffers.
	io_bufs = mem_alloc(TOTAL_BUF_SIZE, 4*KiB);
	// .. failed; io_buf_alloc calls will return 0
	if(!io_bufs)
		return;

	// build freelist.
	char* p = (char*)io_bufs;
	for(int i = 0; i < TOTAL_IOS; i++)
	{
		io_buf_free(p);
		p += STREAM_BUF_SIZE;
	}
}


static void* io_buf_alloc()
{
	ONCE(io_buf_init());

	void* buf = io_buf_freelist;
	// note: we have to bail now; can't update io_buf_freelist.
	if(!buf)
	{
		if(!io_bufs)
			debug_warn("io_buf_alloc: not enough memory to allocate buffer pool");
		else
			debug_warn("io_buf_alloc: max #streams exceeded");
			// can't happen (tm) because stream_open enforces MAX_STREAMS.
		return 0;
	}

	io_buf_freelist = *(void**)io_buf_freelist;
	return buf;
}


// no-op if io_buf_alloc was never called.
// called by snd_shutdown.
static void io_buf_shutdown()
{
	mem_free(io_bufs);
}


//
// stream: owns queue and buffers; uses VFS async I/O
//
///////////////////////////////////////////////////////////////////////////////

// rationale: no need for a centralized queue - we have a suballocator,
// so reallocs aren't a problem; central scheduling isn't necessary,
// because we'll only have <= 2 streams active at a time.

struct Stream
{
	Handle hf;
	Handle ios[MAX_IOS];
	uint active_ios;
	void* last_buf;
		// set by stream_buf_get, used by stream_buf_discard to free buf.
};

// called from SndData_reload and snd_data_buf_get.
static int stream_issue(Stream* s)
{
	if(s->active_ios >= MAX_IOS)
		return 0;

	void* buf = io_buf_alloc();
	if(!buf)
		return ERR_NO_MEM;

	Handle h = vfs_start_io(s->hf, STREAM_BUF_SIZE, buf);
	CHECK_ERR(h);
	s->ios[s->active_ios++] = h;
	return 0;
}


// if the first pending IO hasn't completed, return ERR_AGAIN;
// otherwise, return a negative error code or 0 on success,
// if the pending IO's buffer is returned.
static int stream_buf_get(Stream* s, void*& data, size_t& size)
{
	if(s->active_ios == 0)
		return ERR_EOF;
	Handle hio = s->ios[0];

	// has it finished? if not, bail.
	int is_complete = vfs_io_complete(hio);
	CHECK_ERR(is_complete);
	if(is_complete == 0)
		return ERR_AGAIN;

	// get its buffer.
	CHECK_ERR(vfs_wait_io(hio, data, size));
		// no delay, since vfs_io_complete == 1

	s->last_buf = data;
		// (next stream_buf_discard will free this buffer)
	return 0;
}


// free the buffer that was last returned by stream_buf_get,
// and remove its IO slot from our queue.
// must be called exactly once after every successful stream_buf_get;
// call before calling any other stream_* functions!
static int stream_buf_discard(Stream* s)
{
	Handle hio = s->ios[0];

	int ret = vfs_discard_io(hio);

	// we implement the required 'circular queue' as a stack;
	// have to shift all items after this one down.
	s->active_ios--;
	for(uint i = 0; i < s->active_ios; i++)
		s->ios[i] = s->ios[i+1];

	io_buf_free(s->last_buf);	// can't fail
	s->last_buf = 0;

	return ret;
}


static uint active_streams;


// open a stream and begin reading from disk.
static int stream_open(Stream* s, const char* fn)
{
	if(active_streams >= MAX_STREAMS)
	{
		debug_warn("stream_open: MAX_STREAMS exceeded - why?");
		return -1;
			// fail, because we wouldn't have enough IO buffers for all
	}
	active_streams++;

	s->hf = vfs_open(fn);
	CHECK_ERR(s->hf);

	for(int i = 0; i < MAX_IOS; i++)
		CHECK_ERR(stream_issue(s));
	
	return 0;
}


// close a stream, which may currently be active.
// returns the first error that occurred while waiting for IOs / closing file.
static int stream_close(Stream* s)
{
	int ret = 0;
	int err;

	// for each pending IO:
	for(uint i = 0; i < s->active_ios; i++)
	{
		// .. wait until complete,
		void* data; size_t size;	// unused
		do
			err = stream_buf_get(s, data, size);
		while(err == ERR_AGAIN);
		if(err < 0 && ret == 0)
			ret = err;

		// .. and discard.
		err = stream_buf_discard(s);
		if(err < 0 && ret == 0)
			ret = err;
	}

	err = vfs_close(s->hf);
	if(err < 0 && ret == 0)
		ret = err;

	active_streams--;

	return ret;
}


///////////////////////////////////////////////////////////////////////////////
//
// sound data provider: holds audio data (clip or stream) and returns
// OpenAL buffers on request.
//
///////////////////////////////////////////////////////////////////////////////

// rationale for separate VSrc (instance) and SndData resources:
// - need to be able to fade out and cancel loops.
//   => VSrc isn't fire and forget; need to access sounds at runtime.
// - allowing access via direct pointer is unsafe
//   => need Handle-based access
// - don't want to reload sound data on every play()
//   => need either a separate caching mechanism or one central data resource.
// - want to support reloading (for consistency if not necessity)
//   => can't hack via h_find / setting fn_key to 0; need a separate instance.

// rationale for integrating IO logic into SndData:
// if the IO layer were completely separate, we'd need a callback to pass
// the buffer to OpenAL, or mark processed buffers as "discardable" for
// next time. both are ugly; we instead integrate IO into the SndData code.
// IOs are passed to OpenAL and then discarded immediately.

struct SndData
{
	// stream
	Stream s;
	ALenum al_fmt;
	ALsizei al_freq;

	// clip
	ALuint al_buf;

	bool is_stream;

#ifdef OGG_HACK
// pointer to Ogg instance
void* o;
#endif
};

H_TYPE_DEFINE(SndData);


//
// SndData instance list: ensures all allocated since last al_shutdown
// are freed when desired (they are cached => extra work needed).
//
///////////////////////////////////////////////////////////////////////////////

// rationale: all SndData objects (actually, their OpenAL buffers) must be
// freed during al_shutdown, to prevent leaks. we can't rely on list*
// to free all VSrc, and thereby their associated SndData objects -
// completed sounds are no longer in the list.
//
// nor can we use the h_mgr_shutdown automatic leaked resource cleanup:
// we need to be able to al_shutdown at runtime
// (when resetting OpenAL, after e.g. device change).
//
// h_mgr support is required to forcibly close SndData objects,
// since they are cached (kept open). it requires that this come after
// H_TYPE_DEFINE(SndData), since h_force_free needs the handle type.
//
// we never need to delete single entries: hsd_list_free_all
// (called by al_shutdown) frees each entry and clears the entire list.

typedef std::vector<Handle> Handles;
static Handles hsd_list;

// called from SndData_reload; will later be removed via hsd_list_free_all.
static void hsd_list_add(Handle hsd)
{
	hsd_list.push_back(hsd);
}


// called by al_shutdown (at exit, or when reinitializing OpenAL).
static void hsd_list_free_all()
{
	for(Handles::iterator it = hsd_list.begin(); it != hsd_list.end(); ++it)
	{
		Handle& hsd = *it;

		h_force_free(hsd, H_SndData);
		// ignore errors - if hsd was a stream, and its associated source
		// was active when al_shutdown was called, it will already have been
		// freed (list_free_all would free the source; it then releases
		// its SndData reference, which closes the instance because it's
		// RES_UNIQUE).
	}

	// leave its memory intact, so we don't have to reallocate it later
	// if we are now reinitializing OpenAL (not exiting).
	hsd_list.resize(0);
}


///////////////////////////////////////////////////////////////////////////////


static void SndData_init(SndData* sd, va_list args)
{
	// olsner: pass as int instead of bool for GCC compat.
	sd->is_stream = va_arg(args, int) != 0;
}

static void SndData_dtor(SndData* sd)
{
	if(sd->is_stream)
		stream_close(&sd->s);
	else
		al_buf_free(sd->al_buf);
#ifdef OGG_HACK
if(sd->o) ogg_release(sd->o);
#endif
}


static int SndData_reload(SndData* sd, const char* fn, Handle hsd)
{
	hsd_list_add(hsd);

	//
	// detect sound format by checking file extension
	//

	enum FileType
	{
		FT_WAV,
		FT_OGG
	}
	file_type;

	const char* ext = strrchr(fn, '.');
	// .. OGG (data will be passed directly to OpenAL)
	if(ext && !stricmp(ext, ".ogg"))
	{
#ifdef OGG_HACK
#else
		// 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)
			return ERR_NO_SYS;

		sd->al_fmt  = AL_FORMAT_VORBIS_EXT;
		sd->al_freq = 0;
#endif

		file_type = FT_OGG;
	}
	// .. WAV
	else if(ext && !stricmp(ext, ".wav"))
		file_type = FT_WAV;
	// .. unknown extension
	else
		return ERR_UNKNOWN_FORMAT;


	if(sd->is_stream)
	{
		// refuse to stream anything that cannot be passed directly to OpenAL -
		// we'd have to extract the audio data ourselves (not worth it).
		if(file_type != FT_OGG)
			return -1;

		return stream_open(&sd->s, fn);
	}

	// else: clip

	void* file;
	size_t file_size;
	RETURN_ERR(vfs_load(fn, file, file_size));

	ALvoid* al_data = file;
	ALsizei al_size = (ALsizei)file_size;

	if(file_type == FT_WAV)
	{
		ALbyte* memory = (ALbyte*)file;
		ALboolean al_loop;	// unused
		alutLoadWAVMemory(memory, &sd->al_fmt, &al_data, &al_size, &sd->al_freq, &al_loop);
	}

#ifdef OGG_HACK
std::vector<u8> data;
data.reserve(500000);
if(file_type == FT_OGG)
{
 sd->o = ogg_create();
 ogg_give_raw(sd->o, file, file_size);
 ogg_open(sd->o, sd->al_fmt, sd->al_freq);
 size_t datasize=0;
 size_t bytes_read;
 do
 {
  const size_t bufsize = 32*KiB;
  char buf[bufsize];
  bytes_read = ogg_read(sd->o, buf, bufsize);
  data.insert(data.end(), &buf[0], &buf[bytes_read]);
  datasize += bytes_read;
 }
 while(bytes_read > 0);
 al_data = &data[0];
 al_size = (ALsizei)datasize;
}
else
{
 sd->o = NULL;
}
#endif

	sd->al_buf = al_buf_alloc(al_data, al_size, sd->al_fmt, sd->al_freq);

	mem_free(file);

	return 0;
}


// open and return a handle to a sound file's data.
static Handle snd_data_load(const char* fn, bool is_stream)
{
	// don't allow reloading stream objects
	// (both references would read from the same file handle).
	const uint flags = is_stream? RES_UNIQUE : 0;

	return h_alloc(H_SndData, fn, flags, (int)is_stream);
		// (int) rationale: see SndData_init
}


// close the sound file data <hsd> and set hsd to 0.
static int snd_data_free(Handle& hsd)
{
	return h_free(hsd, H_SndData);
}


// (need to convert ERR_EOF and ERR_AGAIN to legitimate return values -
// for the caller, those aren't errors.)
enum BufRet
{
	// buffer has been returned; barring errors, more will be available.
	BUF_OK = 0,

	// this was the last buffer we will return (end of file reached).
	BUF_EOF = 1,

	// no buffer returned - still streaming in ATM. call again later.
	BUF_AGAIN = 2,

	// anything else: negative error code
};

static int snd_data_buf_get(Handle hsd, ALuint& al_buf)
{
	int err = 0;

	// in case H_DEREF fails
	al_buf = 0;

	H_DEREF(hsd, SndData, sd);

	// clip: just return buffer (which was created in snd_data_load)
	if(!sd->is_stream)
	{
		al_buf = sd->al_buf;
		return BUF_EOF;
	}

	// stream:

	// .. check if IO finished.
	void* data;
	size_t size;
	err = stream_buf_get(&sd->s, data, size);
	if(err == ERR_AGAIN)
		return BUF_AGAIN;
	CHECK_ERR(err);

	// .. yes: pass to OpenAL and discard IO buffer.
	al_buf = al_buf_alloc(data, (ALsizei)size, sd->al_fmt, sd->al_freq);
	stream_buf_discard(&sd->s);

	// .. try to issue the next IO.
	// if EOF reached, indicate al_buf is the last that will be returned.
	err = stream_issue(&sd->s);
	if(err == ERR_EOF)
		return BUF_EOF;
	CHECK_ERR(err);

	// al_buf valid and next IO issued successfully.
	return BUF_OK;
}


static int snd_data_buf_free(Handle hsd, ALuint al_buf)
{
	H_DEREF(hsd, SndData, sd);

	// clip: no-op (caller will later release hsd reference;
	// when hsd actually unloads, sd->al_buf will be freed).
	if(!sd->is_stream)
		return 0;

	// stream: we had allocated an additional buffer, so free it now.
	al_buf_free(al_buf);
	return 0;
}


///////////////////////////////////////////////////////////////////////////////
//
// virtual sound source: a sound the user wants played.
// owns source properties, buffer queue, and references SndData.
//
///////////////////////////////////////////////////////////////////////////////

// rationale: combine Src and VSrc - best interface, due to needing hsd,
// buffer queue (# processed) in update

enum VSrcFlags
{
	// SndData has reported EOF. will close down after last buffer completes.
	VS_EOF       = 1,

	// tell SndData to open the sound as a stream.
	// require caller to pass this explicitly, so they're aware
	// of the limitation that there can only be 1 instance of those.
	VS_IS_STREAM = 2,

	// this VSrc was added via list_add and needs to be removed with
	// list_remove in VSrc_dtor.
	// not set if load fails somehow (avoids list_remove "not found" error).
	VS_IN_LIST = 4
};

struct VSrc
{
	ALuint al_src;

	// handle to this VSrc, so that it can close itself.
	Handle hvs;

	// associated sound data
	Handle hsd;

	// controls vsrc_update behavior
	uint flags;

	// AL source properties (set via snd_set*)
	ALfloat pos[3];
	ALfloat gain;	// [0,1]
	ALfloat pitch;	// (0,1]
	ALboolean loop;
	ALboolean relative;

	// priority for voice management
	float static_pri;	// as given by snd_play
	float cur_pri;		// holds newly calculated value
};

H_TYPE_DEFINE(VSrc);


//
// list of active sounds. used by voice management component,
// and to have each VSrc update itself (queue new buffers).
//
///////////////////////////////////////////////////////////////////////////////

// VSrc fields are used -> must come after struct VSrc

// sorted in descending order of current priority
// (we sometimes remove low pri items, which requires moving down
// everything that comes after them, so we want those to come last).
//
// don't use list, to avoid lots of allocs (expect thousands of VSrcs).
typedef std::vector<VSrc*> VSrcs;
typedef VSrcs::iterator It;
static VSrcs vsrcs;

// don't need to sort now - caller will list_sort() during update.
static void list_add(VSrc* vs)
{
	vsrcs.push_back(vs);
}


// skip past <skip> entries; if end_idx != 0, stop before that entry.
static void list_foreach(void(*cb)(VSrc*), uint skip = 0, uint end_idx = 0)
{
	It begin = vsrcs.begin() + skip;
	It end = vsrcs.end();
	if(end_idx)
		end = vsrcs.begin()+end_idx;

	// can't use std::for_each: some entries may have been deleted
	// (i.e. set to 0) since last update.
	for(It it = begin; it != end; ++it)
	{
		VSrc* vs = *it;
		if(vs)
			cb(vs);
	}
}


static bool is_greater(const VSrc* vs1, const VSrc* vs2)
{
	return vs1->cur_pri > vs2->cur_pri;
}

static void list_sort()
{
	std::sort(vsrcs.begin(), vsrcs.end(), is_greater);
}


// O(N)!
static void list_remove(VSrc* vs)
{
	for(size_t i = 0; i < vsrcs.size(); i++)
		if(vsrcs[i] == vs)
		{
			// found it; several ways we could remove:
			// - shift everything else down (slow) -> no
			// - fill the hole with e.g. the last element
			//   (vsrcs would no longer be sorted by priority) -> no
			// - replace with 0 (will require prune_removed and
			//   more work in foreach) -> best alternative
			vsrcs[i] = 0;
			return;
		}

		debug_warn("list_remove: VSrc not found");
}


static bool is_null(VSrc* vs)
{
	return vs == 0;
}

static void list_prune_removed()
{
	// prune removed entries (value 0), so code below can grant the first
	// al_src_cap entries a soure. see rationale in list_remove.
	It new_end = remove_if(vsrcs.begin(), vsrcs.end(), is_null);
	vsrcs.erase(new_end, vsrcs.end());
}

static void free_vs(VSrc* vs)
{
	snd_free(vs->hvs);
}

static int list_free_all()
{
	list_foreach(free_vs);
	return 0;
}


///////////////////////////////////////////////////////////////////////////////


// send the properties to OpenAL (when we actually have a source).
// called by snd_set* and vsrc_grant.
static void vsrc_latch(VSrc* vs)
{
	if(!vs->al_src)
		return;

	alSourcefv(vs->al_src, AL_POSITION,        vs->pos);
	alSourcei (vs->al_src, AL_SOURCE_RELATIVE, vs->relative);
	alSourcef (vs->al_src, AL_GAIN,            vs->gain);
	alSourcef (vs->al_src, AL_PITCH,           vs->pitch);
	alSourcei (vs->al_src, AL_LOOPING,         vs->loop);
	al_check("vsrc_latch");
}


// return number of entries that were removed.
static int vsrc_deque_finished_bufs(VSrc* vs)
{
	int num_processed;
	alGetSourcei(vs->al_src, AL_BUFFERS_PROCESSED, &num_processed);

	for(int i = 0; i < num_processed; i++)
	{
		ALuint al_buf;
		alSourceUnqueueBuffers(vs->al_src, 1, &al_buf);
		snd_data_buf_free(vs->hsd, al_buf);
	}

	al_check("vsrc_deque_finished_bufs");
	return num_processed;
}


static int vsrc_update(VSrc* vs)
{
	if(!vs->al_src)
		return 0;

	int num_queued;
	alGetSourcei(vs->al_src, AL_BUFFERS_QUEUED, &num_queued);
	al_check("vsrc_update alGetSourcei");

	int num_processed = vsrc_deque_finished_bufs(vs);

	if(vs->flags & VS_EOF)
	{
		// no more buffers left, and EOF reached - done playing.
		if(num_queued == 0)
		{
			snd_free(vs->hvs);
			return 0;
		}
	}
	// can still read from SndData
	else
	{
		// decide how many buffers we are going to request
		// (start off with MAX_IOS; after that, replace finished buffers).
		int to_fill = MAX_IOS;
		if(num_queued > 0)
			to_fill = num_processed;

		// request and queue each buffer.
		int ret;
		do
		{
			ALuint al_buf;
			ret = snd_data_buf_get(vs->hsd, al_buf);
			CHECK_ERR(ret);

			alSourceQueueBuffers(vs->al_src, 1, &al_buf);
			al_check("vsrc_update alSourceQueueBuffers");
		}
		while(to_fill-- && ret == BUF_OK);

		// SndData has reported that no further buffers are available.
		if(ret == BUF_EOF)
			vs->flags |= VS_EOF;
	}

	return 0;
}


// attempt to (re)start playing. fails if a source cannot be allocated
// (see below). called by snd_play and voice management.
static int vsrc_grant(VSrc* vs)
{
	// already playing - bail
	if(vs->al_src)
		return 0;

	// try to alloc source. if none are available, bail -
	// we get called in that hope that one is available by snd_play.
	vs->al_src = al_src_alloc();
	if(!vs->al_src)
		return -1;

	// 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("vsrc_grant");

	// set remaining (user-specifiable) properties.
	vsrc_latch(vs);

	// queue up some buffers (enough to start playing, at least).
	vsrc_update(vs);

	alSourcePlay(vs->al_src);
	al_check("vsrc_grant alSourcePlay");
	return 0;
}


// stop playback, and reclaim the OpenAL source.
// called when closing the VSrc, or when voice management decides
// this VSrc must yield to others of higher priority.
static int vsrc_reclaim(VSrc* vs)
{
	// don't own a source - bail.
	if(!vs->al_src)
		return -1;

	alSourceStop(vs->al_src);
	al_check("src_stop");
		// (note: all queued buffers are now considered 'processed')

	// deque and free remaining buffers (if sound was closed abruptly).
	vsrc_deque_finished_bufs(vs);

	al_src_free(vs->al_src);
	return 0;
}


///////////////////////////////////////////////////////////////////////////////


static void VSrc_init(VSrc* vs, va_list args)
{
	vs->flags = va_arg(args, uint);
}


static void VSrc_dtor(VSrc* vs)
{
	// only remove if added (not the case if load failed)
	if(vs->flags & VS_IN_LIST)
	{
		list_remove(vs);
		vs->flags &= ~VS_IN_LIST;
	}

	vsrc_reclaim(vs);
	snd_data_free(vs->hsd);
}


static int VSrc_reload(VSrc* vs, const char* fn, Handle hvs)
{
	// cannot wait till play(), need to init here:
	// must load OpenAL so that snd_data_load can check for OGG extension.
	int err = snd_init();
	// .. don't complain if sound is disabled; fail silently.
	if(err == ERR_AGAIN)
		return err;
	// .. catch genuine errors during init.
	CHECK_ERR(err);

	//
	// if extension is .txt, fn is a definition file containing the
	// sound file name and its gain; otherwise, read directly from fn
	// and assume default gain (1.0).
	//

	const char* snd_fn;		// actual sound file name
	std::string snd_fn_s;
		// extracted from stringstream;
		// declare here so that it doesn't go out of scope below.

	const char* ext = strchr(fn, '.');
	if(ext && !strcmp(ext, ".txt"))
	{
		void* def_file;
		size_t def_size;
		RETURN_ERR(vfs_load(fn, def_file, def_size));
		std::istringstream def(std::string((char*)def_file, (int)def_size));
		mem_free(def_file);

		float gain_percent;
		def >> snd_fn_s;
		def >> gain_percent;

		snd_fn = snd_fn_s.c_str();
		vs->gain = gain_percent / 100.0f;
	}
	else
	{
		snd_fn = fn;
		vs->gain = 1.0f;
	}

	// note: vs->gain can legitimately be > 1.0 - don't clamp.

	vs->pitch = 1.0f;

	vs->hvs = hvs;
		// needed so we can snd_free ourselves when done playing.

	bool is_stream = (vs->flags & VS_IS_STREAM) != 0;
	vs->hsd = snd_data_load(snd_fn, is_stream);
	CHECK_ERR(vs->hsd);

	return 0;
}


// open and return a handle to a sound instance.
//
// if <snd_fn> is a text file (extension ".txt"), it is assumed
// to be a definition file containing the sound file name and
// its gain (0.0 .. 1.0).
// otherwise, <snd_fn> is taken to be the sound file name and
// gain is set to the default of 1.0 (no attenuation).
//
// is_stream (default false) forces the sound to be opened as a stream:
// opening is faster, it won't be kept in memory, but only one instance
// can be open at a time.
//
// note: RES_UNIQUE forces each instance to get a new resource
// (which is of course what we want).
Handle snd_open(const char* snd_fn, bool is_stream)
{
	uint flags = 0;
	if(is_stream)
		flags |= VS_IS_STREAM;
	return h_alloc(H_VSrc, snd_fn, RES_UNIQUE, flags);
}


// close the sound <hvs> and set hvs to 0. if it was playing,
// it will be stopped. sounds are closed automatically when done
// playing; this is provided for completeness only.
int snd_free(Handle& hvs)
{
	return h_free(hvs, H_VSrc);
}


// request the sound <hs> be played. once done playing, the sound is
// automatically closed (allows fire-and-forget play code).
// if no hardware voice is available, this sound may not be played at all,
// or in the case of looped sounds, start later.
// priority (min 0 .. max 1, default 0) indicates which sounds are
// considered more important; this is attenuated by distance to the
// listener (see snd_update).
int snd_play(Handle hs, float static_pri)
{
	H_DEREF(hs, VSrc, vs);

	// note: vs->hsd is valid, otherwise snd_open would have failed
	// and returned an invalid handle (caught above).

	vs->static_pri = static_pri;
	list_add(vs);
	vs->flags |= VS_IN_LIST;

	// 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(vs);
	return 0;
}


// change 3d position of the sound source.
// if relative (default false), (x,y,z) is treated as relative to the
// listener; otherwise, it is the position in world coordinates.
// may be called at any time; fails with invalid handle return if
// the sound has already been closed (e.g. it never played).
int snd_set_pos(Handle hvs, float x, float y, float z, bool relative)
{
	H_DEREF(hvs, VSrc, vs);

	vs->pos[0] = x; vs->pos[1] = y; vs->pos[2] = z;
	vs->relative = relative;

	vsrc_latch(vs);
	return 0;
}


// change gain (amplitude modifier) of the sound source.
// must be non-negative; 1 -> unattenuated, 0.5 -> -6 dB, 0 -> silence.
// may be called at any time; fails with invalid handle return if
// the sound has already been closed (e.g. it never played).
int snd_set_gain(Handle hvs, float gain)
{
	H_DEREF(hvs, VSrc, vs);

	if(!(0.0f <= gain && gain <= 1.0f))
		return -1;

	vs->gain = gain;

	vsrc_latch(vs);
	return 0;
}


// change pitch shift of the sound source.
// 1.0 means no change; each reduction by 50% equals a pitch shift of
// -12 semitones (one octave). zero is invalid.
// may be called at any time; fails with invalid handle return if
// the sound has already been closed (e.g. it never played).
int snd_set_pitch(Handle hvs, float pitch)
{
	H_DEREF(hvs, VSrc, vs);

	if(!(0.0f < pitch && pitch <= 1.0f))
		return -1;

	vs->pitch = pitch;

	vsrc_latch(vs);
	return 0;
}


// enable/disable looping on the sound source.
// used to implement variable-length sounds (e.g. while building).
// may be called at any time; fails with invalid handle return if
// the sound has already been closed (e.g. it never played).
//
// notes:
// - looping sounds are not discarded if they cannot be played for lack of
//   a hardware voice at the moment play was requested.
// - once looping is again disabled and the sound has reached its end,
//   the sound instance is freed automatically (as if never looped).
int snd_set_loop(Handle hvs, bool loop)
{
	H_DEREF(hvs, VSrc, vs);

	vs->loop = loop;

	vsrc_latch(vs);
	return 0;
}


///////////////////////////////////////////////////////////////////////////////
//
// voice management: grants the currently most 'important' sounds
// a hardware voice.
//
///////////////////////////////////////////////////////////////////////////////


static float magnitude_2(const float v[3])
{
	return v[0]*v[0] + v[1]*v[1] + v[2]*v[2];
}


// list_foreach callback
static void calc_cur_pri(VSrc* vs)
{
	const float MAX_DIST_2 = 1000.0f;
	const float falloff = 10.0f;

	float d_2;	// euclidean distance to listener (squared)
	if(vs->relative)
		d_2 = magnitude_2(vs->pos);
	else
		d_2 = al_listener_dist_2(vs->pos);

	// scale priority down exponentially
	float e = d_2 / MAX_DIST_2;	// 0.0f (close) .. 1.0f (far)

	// assume farther away than OpenAL cutoff - no sound contribution
	float cur_pri = -1.0f;
	if(e < 1.0f)
		cur_pri = vs->static_pri / pow(falloff, e);
	vs->cur_pri = cur_pri;
}


static void reclaim(VSrc* vs)
{
	vsrc_reclaim(vs);

	if(!vs->loop)
		snd_free(vs->hvs);
}


static void grant(VSrc* vs)
{
	vsrc_grant(vs);
}


// no-op if OpenAL not yet initialized.
static int vm_update()
{
	list_prune_removed();

	// update current priorities (a function of static priority and distance).
	list_foreach(calc_cur_pri);

	// sort by descending current priority.
	list_sort();

	// partition list; the first al_src_cap will be granted a source
	// (if they don't have one already), after reclaiming all sources from
	// the remainder of the VSrc list entries.
	uint first_unimportant = MIN((uint)vsrcs.size(), al_src_cap); 
	list_foreach(reclaim, first_unimportant, 0);
	list_foreach(grant, 0, first_unimportant);

	// add / remove buffers for each source.
	list_foreach((void(*)(VSrc*))vsrc_update);

	return 0;
}


///////////////////////////////////////////////////////////////////////////////


// perform housekeeping (e.g. streaming); call once a frame.
//
// additionally, if any parameter is non-NULL, we set the listener
// position, look direction, and up vector (in world coordinates).
// (allow any and all of them to be 0 in case world isn't initialized yet).
int snd_update(const float* pos, const float* dir, const float* up)
{
	// there's no sense in updating anything if we weren't initialized
	// yet (most notably, if sound is disabled). we check for this to
	// avoid confusing the code below. the caller should complain if
	// this fails, so report success here (everything will work once
	// sound is re-enabled).
	if(!al_initialized)
		return 0;

	if(pos || dir || up)
		al_listener_set_pos(pos, dir, up);

	vm_update();

	return 0;
}


// prevent OpenAL from being initialized when snd_init is called.
static bool snd_disabled = false;

// extra layer on top of al_init that allows 'disabling' sound.
// called from each snd_open. returns ERR_AGAIN if sound disabled,
// otherwise the status returned by al_init.
static inline int snd_init()
{
	// (note: each VSrc_reload and therefore snd_open will fail)
	if(snd_disabled)
		return ERR_AGAIN;

	return al_init();
}


// (temporarily) disable all sound output. because it causes future snd_open
// calls to immediately abort before they demand-initialize OpenAL,
// startup is sped up considerably (500..1000ms). therefore, this must be
// called before the first snd_open to have any effect; otherwise, the
// cat will already be out of the bag and we debug_warn of it.
//
// rationale: this is a quick'n dirty way of speeding up startup during
// development without having to change the game's sound code.
//
// can later be called to reactivate sound; all settings ever changed
// will be applied and subsequent sound load / play requests will work.
int snd_disable(bool disabled)
{
	snd_disabled = disabled;

	if(snd_disabled)
	{
		if(al_initialized)
			debug_warn("snd_disable: already initialized => disable is pointless");
		return 0;
	}
	else
		return snd_init();
			// note: won't return ERR_AGAIN, since snd_disabled == false
}


// free all resources and shut down the sound system.
// call before h_mgr_shutdown.
void snd_shutdown()
{
	io_buf_shutdown();

	al_shutdown();
		// calls list_free_all
}