apple2ix/src/audio/soundcore-openal.c

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/*
* Apple // emulator for *nix
*
* This software package is subject to the GNU General Public License
* version 2 or later (your choice) as published by the Free Software
* Foundation.
*
* THERE ARE NO WARRANTIES WHATSOEVER.
*
*/
// HACK NOTE: DO NOT REFACTOR (yet) ...
// OK this is more hackish than it needs to be because it's attempting to mimic a DirectSound backend ...Oh God, Why?...
// Here I must confess that because of general ignorance of the mockingboard and other soundcard code at this time,
// there is a need to track any changes/fixes implemented in AppleWin...
#include "common.h"
#include "audio/soundcore-openal.h"
#include "audio/alhelpers.h"
static long OpenALCreateSoundBuffer(AudioParams_s *params, ALSoundBufferStruct **soundbuf_struct, void *extra_data);
static long OpenALDestroySoundBuffer(ALSoundBufferStruct **soundbuf_struct);
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typedef struct ALVoices {
const ALuint source;
ALVoice *voice;
UT_hash_handle hh;
} ALVoices;
static ALVoices *voices = NULL;
// ----------------------------------------------------------------------------
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// uthash of OpenAL buffers
static ALPlayBuf *PlaylistEnqueue(ALVoice *voice, ALuint bytes)
{
ALPlayBuf *node = voice->avail_buffers;
if (node == NULL)
{
ERRLOG("OOPS, sound playback overflow!");
return NULL;
}
//LOG("Really enqueing OpenAL buffer %u", node->bufid);
ALPlayBuf *node2 = NULL;
HASH_FIND_INT(voice->queued_buffers, &node->bufid, node2);
if (node2 != NULL)
{
ERRLOG("OOPS, confused ... ALPlayBuf already added!");
return NULL;
}
// remove from list / place on hash
voice->avail_buffers = node->_avail_next;
node->_avail_next = NULL;
HASH_ADD_INT(voice->queued_buffers, bufid, node);
node->bytes = bytes;
voice->_queued_total_bytes += node->bytes;
voice->index = 0;
assert(voice->_queued_total_bytes > 0);
#if 0
ALPlayBuf *tmp = voice->queued_buffers;
unsigned int count = HASH_COUNT(voice->queued_buffers);
LOG("\t(numqueued: %d)", count);
for (unsigned int i = 0; i < count; i++, tmp = tmp->hh.next)
{
LOG("\t(bufid : %u)", tmp->bufid);
}
#endif
return node;
}
static ALPlayBuf *PlaylistGet(ALVoice *voice, ALuint bufid)
{
ALPlayBuf *node = NULL;
HASH_FIND_INT(voice->queued_buffers, &bufid, node);
return node;
}
static void PlaylistDequeue(ALVoice *voice, ALPlayBuf *node)
{
//LOG("Dequeing OpenAL buffer %u", node->bufid);
// remove from hash / place on list
HASH_DEL(voice->queued_buffers, node); // remove from hash
node->_avail_next = voice->avail_buffers;
voice->avail_buffers = node;
voice->_queued_total_bytes -= node->bytes;
assert(voice->_queued_total_bytes >= 0);
node->bytes = 0;
#if 0
ALPlayBuf *tmp = voice->queued_buffers;
unsigned int count = HASH_COUNT(voice->queued_buffers);
LOG("\t(numqueued: %d)", count);
for (unsigned int i = 0; i < count; i++, tmp = tmp->hh.next)
{
LOG("\t(bufid : %u)", tmp->bufid);
}
#endif
}
// ----------------------------------------------------------------------------
long SoundSystemCreate(const char *sound_device, SoundSystemStruct **sound_struct)
{
assert(*sound_struct == NULL);
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assert(voices == NULL);
ALCcontext *ctx = NULL;
do {
if ((ctx = InitAL()) == NULL)
{
ERRLOG("OOPS, OpenAL initialize failed");
break;
}
if (alIsExtensionPresent("AL_SOFT_buffer_samples"))
{
LOG("AL_SOFT_buffer_samples supported, good!");
}
else
{
LOG("WARNING - AL_SOFT_buffer_samples extension not supported... Proceeding anyway...");
}
if ((*sound_struct = malloc(sizeof(SoundSystemStruct))) == NULL)
{
ERRLOG("OOPS, Not enough memory");
break;
}
(*sound_struct)->implementation_specific = ctx;
(*sound_struct)->CreateSoundBuffer = (int (*)(AudioParams_s *, LPDIRECTSOUNDBUFFER *, void *))OpenALCreateSoundBuffer;
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(*sound_struct)->DestroySoundBuffer = (int (*)(LPDIRECTSOUNDBUFFER *))OpenALDestroySoundBuffer;
return 0;
} while(0);
// ERRQUIT
if (*sound_struct)
{
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FREE(*sound_struct);
}
return -1;
}
long SoundSystemDestroy(SoundSystemStruct **sound_struct)
{
// ugly assumption : this sets the extern g_lpDS ...
assert(*sound_struct != NULL);
ALCcontext *ctx = (ALCcontext*) (*sound_struct)->implementation_specific;
assert(ctx != NULL);
(*sound_struct)->implementation_specific = NULL;
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FREE(*sound_struct);
CloseAL();
return 0;
}
long SoundSystemEnumerate(char ***device_list, const int limit)
{
assert(*device_list == NULL);
*device_list = malloc(sizeof(char*)*2);
(*device_list)[0] = strdup("unused-by-OpenAL");
unsigned int num_devices = 1;
(*device_list)[num_devices] = NULL; // sentinel
return num_devices;
}
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// pause all audio
long SoundSystemPause()
{
ALVoices *vnode = NULL;
ALVoices *tmp = NULL;
int err = 0;
HASH_ITER(hh, voices, vnode, tmp) {
alSourcePause(vnode->source);
err = alGetError();
if (err != AL_NO_ERROR)
{
ERRLOG("OOPS, Failed to pause source : 0x%08x", err);
}
}
return 0;
}
long SoundSystemUnpause()
{
ALVoices *vnode = NULL;
ALVoices *tmp = NULL;
int err = 0;
HASH_ITER(hh, voices, vnode, tmp) {
alSourcePlay(vnode->source);
err = alGetError();
if (err != AL_NO_ERROR)
{
ERRLOG("OOPS, Failed to pause source : 0x%08x", err);
}
}
return 0;
}
// ----------------------------------------------------------------------------
/* Destroys a voice object, deleting the source and buffers. No error handling
* since these calls shouldn't fail with a properly-made voice object. */
static void DeleteVoice(ALVoice *voice)
{
alDeleteSources(1, &voice->source);
if (alGetError() != AL_NO_ERROR)
{
ERRLOG("OOPS, Failed to delete source");
}
if (voice->data)
{
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FREE(voice->data);
}
ALPlayBuf *node = NULL;
ALPlayBuf *tmp = NULL;
HASH_ITER(hh, voice->queued_buffers, node, tmp) {
PlaylistDequeue(voice, node);
}
while (voice->avail_buffers)
{
node = voice->avail_buffers;
alDeleteBuffers(1, &node->bufid);
if (alGetError() != AL_NO_ERROR)
{
ERRLOG("OOPS, Failed to delete object IDs");
}
voice->avail_buffers = node->_avail_next;
FREE(node);
}
memset(voice, 0, sizeof(*voice));
FREE(voice);
}
/* Creates a new voice object, and allocates the needed OpenAL source and
* buffer objects. Error checking is simplified for the purposes of this
* example, and will cause an abort if needed. */
static ALVoice *NewVoice(AudioParams_s *params)
{
ALVoice *voice = NULL;
do {
voice = calloc(1, sizeof(*voice));
if (voice == NULL)
{
ERRLOG("OOPS, Out of memory!");
break;
}
ALuint buffers[OPENAL_NUM_BUFFERS];
alGenBuffers(OPENAL_NUM_BUFFERS, buffers);
if (alGetError() != AL_NO_ERROR)
{
ERRLOG("OOPS, Could not create buffers");
break;
}
alGenSources(1, &voice->source);
if (alGetError() != AL_NO_ERROR)
{
ERRLOG("OOPS, Could not create source");
break;
}
// Set parameters so mono sources play out the front-center speaker and won't distance attenuate.
alSource3i(voice->source, AL_POSITION, 0, 0, -1);
if (alGetError() != AL_NO_ERROR)
{
ERRLOG("OOPS, Could not set AL_POSITION source parameter");
break;
}
alSourcei(voice->source, AL_SOURCE_RELATIVE, AL_TRUE);
if (alGetError() != AL_NO_ERROR)
{
ERRLOG("OOPS, Could not set AL_SOURCE_RELATIVE source parameter");
break;
}
alSourcei(voice->source, AL_ROLLOFF_FACTOR, 0);
if (alGetError() != AL_NO_ERROR)
{
ERRLOG("OOPS, Could not set AL_ROLLOFF_FACTOR source parameter");
break;
}
#if 0
alSourcei(voice->source, AL_STREAMING, AL_TRUE);
if (alGetError() != AL_NO_ERROR)
{
ERRLOG("OOPS, Could not set AL_STREAMING source parameter");
break;
}
#endif
voice->avail_buffers = NULL;
for (unsigned int i=0; i<OPENAL_NUM_BUFFERS; i++)
{
ALPlayBuf immutableNode = { .bufid = buffers[i] };
ALPlayBuf *node = calloc(1, sizeof(ALPlayBuf));
if (!node)
{
ERRLOG("OOPS, Not enough memory");
break;
}
memcpy(node, &immutableNode, sizeof(ALPlayBuf));
node->_avail_next = voice->avail_buffers;
voice->avail_buffers = node;
}
voice->rate = (ALuint)params->nSamplesPerSec;
// Emulator supports only mono and stereo
if (params->nChannels == 2)
{
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voice->format = AL_FORMAT_STEREO16;
}
else
{
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voice->format = AL_FORMAT_MONO16;
}
/* Allocate enough space for the temp buffer, given the format */
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voice->buffersize = (ALsizei)params->dwBufferBytes;
voice->data = malloc(voice->buffersize);
if (voice->data == NULL)
{
ERRLOG("OOPS, Error allocating %d bytes", voice->buffersize);
break;
}
LOG("\tRate : 0x%08x", voice->rate);
LOG("\tFormat : 0x%08x", voice->format);
LOG("\tbuffersize : %d", voice->buffersize);
return voice;
} while(0);
// ERR
if (voice)
{
DeleteVoice(voice);
}
return NULL;
}
// ----------------------------------------------------------------------------
static long ALGetVolume(void *_this, long *volume)
{
LOG("ALGetVolume ...");
if (volume)
{
*volume = 0;
}
return 0;
}
static long ALSetVolume(void *_this, long volume)
{
LOG("ALSetVolume ...");
return 0;
}
static long ALStop(void *_this)
{
LOG("ALStop ...");
return 0;
}
static long ALRestore(void *_this)
{
LOG("ALRestore ...");
return 0;
}
static long ALPlay(void *_this, unsigned long reserved1, unsigned long reserved2, unsigned long flags)
{
LOG("ALPlay ...");
return 0;
}
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static int _ALProcessPlayBuffers(ALVoice *voice, ALuint *bytes_queued)
{
ALint processed = 0;
int err = 0;
*bytes_queued = 0;
alGetSourcei(voice->source, AL_BUFFERS_PROCESSED, &processed);
if ((err = alGetError()) != AL_NO_ERROR)
{
ERRLOG("OOPS, error in checking processed buffers : 0x%08x", err);
return err;
}
if ((processed == 0) && (HASH_COUNT(voice->queued_buffers) >= OPENAL_NUM_BUFFERS))
{
//LOG("All audio buffers processing...");
}
while (processed > 0)
{
--processed;
ALuint bufid = 0;
alSourceUnqueueBuffers(voice->source, 1, &bufid);
if ((err = alGetError()) != AL_NO_ERROR)
{
ERRLOG("OOPS, OpenAL error dequeuing buffer : 0x%08x", err);
return err;
}
//LOG("Attempting to dequeue %u ...", bufid);
ALPlayBuf *node = PlaylistGet(voice, bufid);
if (!node)
{
ERRLOG("OOPS, OpenAL bufid %u not found in playlist...", bufid);
ALPlayBuf *tmp = voice->queued_buffers;
unsigned int count = HASH_COUNT(voice->queued_buffers);
LOG("\t(numqueued: %d)", count);
for (unsigned int i = 0; i < count; i++, tmp = tmp->hh.next)
{
LOG("\t(bufid : %u)", tmp->bufid);
}
continue;
}
PlaylistDequeue(voice, node);
}
ALint play_offset = 0;
alGetSourcei(voice->source, AL_BYTE_OFFSET, &play_offset);
if ((err = alGetError()) != AL_NO_ERROR)
{
ERRLOG("OOPS, alGetSourcei AL_BYTE_OFFSET : 0x%08x", err);
return err;
}
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assert((play_offset >= 0)/* && (play_offset < voice->buffersize)*/);
long q = voice->_queued_total_bytes/* + voice->index*/ - play_offset;
if (q >= 0) {
*bytes_queued = (ALuint)q;
}
return 0;
}
// returns queued+working sound buffer size in bytes
static long ALGetPosition(void *_this, unsigned long *bytes_queued, unsigned long *unused_write_cursor)
{
ALVoice *voice = (ALVoice*)_this;
*bytes_queued = 0;
Refactor speaker system to prevent audio glitches and to support CPU automatic speed switching Squashed commit of the following: REFACTOR : fix comments, logging, and rename some variables REFACTOR : fix up some commentary, clean deadc0de REFACTOR : mostly fix all the audio glitches - amplitudes of samples are gradually shifted to zero when speaker has fallen silent - simplifies speaker state machine - fullspeed mode only enqueues quiet samples REFACTOR : fix up a number of other functions and comments REFACTOR : clean up code to submit normal speed wave buffer to OpenAL Move some initializations to the cpu_thread() REFACTOR : properly reset the speaker cycles access counter so we don't underflow and assert REFACTOR : should never get a split buffer from our soundcore implementation Fix warning from gcc ... static array size needs to be computed from integer values REFACTOR : gcc (but not clang) complains about these, so just make them preprocessor defines REFACTOR : use unsigned long long because we don't actually care that this counter is 64bit REFACTOR : remainder_buffer and miscellaneous tweaks - Adds implementation commentary to document remainder_buffer purpose - Also adds sample average for square wave boundary in case where remainder_buffer not used (whole-sample boundary) - Variable renaming and code shuffling REFACTOR : do not dynamically alloc remainder buffer - Never attribute to cache-coherancy bugs what is a simple thread race =P REFACTOR : comments and whitespace REFACTOR : rename public speaker API functions REFACTOR : clean up public speaker API REFACTOR : tabs to spaces REFACTOR : moar deadc0de clean up and renaming REFACTOR : remove deadc0de paths from soundcore REFACTOR : fully excise soundtype stuff now that we only support soundcard output Move a file static to function scope REFACTOR : rename more variables and remove deadc0de REFACTOR : samples_buffer naming and change to explict int16_t REFACTOR : removed deadc0de and shuffled code locations REFACTOR : remainder buffer naming and clarify type REFACTOR : move joystick timing to VM module and remove header visibility REFACTOR : clarify speaker variable name REFACTOR : clarify cycle counting codepaths REFACTOR : VBL/timing interfaces - eliminates passing around a common global REFACTOR : names and comments HACK around volume issue REFACTOR : rename speaker feedback variable REFACTOR : rename global total cycle count REFACTOR : rename a constant Fix test builds REFACTOR: rename to is_fullspeed REFACTOR : local variable naming changes REFACTOR : migrate cycle timing variables to correct location and remove header visibility Allow fullspeed codepath to update speaker REFACTOR : remove deadc0de paths in prep for cleanup REFACTOR : speaker now manages its own VM entry point
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if (unused_write_cursor)
{
*unused_write_cursor = 0;
}
ALuint queued = 0;
int err = _ALProcessPlayBuffers(voice, &queued);
if (err)
{
return err;
}
static int last_queued = 0;
if (queued != last_queued)
{
last_queued = queued;
//LOG("OpenAL bytes queued : %u", queued);
}
*bytes_queued = queued + voice->index;
return 0;
}
// DS->Lock()
static long ALBegin(void *_this, unsigned long unused, unsigned long write_bytes, void **audio_ptr1, unsigned long *audio_bytes1, void **unused_audio_ptr2, unsigned long *unused_audio_bytes2, unsigned long flags_unused)
{
ALVoice *voice = (ALVoice*)_this;
if (unused_audio_ptr2)
{
*unused_audio_ptr2 = NULL;
}
if (write_bytes == 0)
{
write_bytes = voice->buffersize;
}
ALuint bytes_queued = 0;
int err = _ALProcessPlayBuffers(voice, &bytes_queued);
if (err)
{
return err;
}
if ((bytes_queued == 0) && (voice->index == 0))
{
LOG("Buffer underrun ... queuing quiet samples ...");
int quiet_size = voice->buffersize>>2/* 1/4 buffer */;
memset(voice->data, 0x0, quiet_size);
voice->index += quiet_size;
}
else if (bytes_queued + voice->index < (voice->buffersize>>3)/* 1/8 buffer */)
{
LOG("Potential underrun ...");
}
ALsizei remaining = voice->buffersize - voice->index;
if (write_bytes > remaining)
{
write_bytes = remaining;
}
*audio_ptr1 = voice->data+voice->index;
*audio_bytes1 = write_bytes;
return 0;
}
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static int _ALSubmitBufferToOpenAL(ALVoice *voice)
{
Refactor speaker system to prevent audio glitches and to support CPU automatic speed switching Squashed commit of the following: REFACTOR : fix comments, logging, and rename some variables REFACTOR : fix up some commentary, clean deadc0de REFACTOR : mostly fix all the audio glitches - amplitudes of samples are gradually shifted to zero when speaker has fallen silent - simplifies speaker state machine - fullspeed mode only enqueues quiet samples REFACTOR : fix up a number of other functions and comments REFACTOR : clean up code to submit normal speed wave buffer to OpenAL Move some initializations to the cpu_thread() REFACTOR : properly reset the speaker cycles access counter so we don't underflow and assert REFACTOR : should never get a split buffer from our soundcore implementation Fix warning from gcc ... static array size needs to be computed from integer values REFACTOR : gcc (but not clang) complains about these, so just make them preprocessor defines REFACTOR : use unsigned long long because we don't actually care that this counter is 64bit REFACTOR : remainder_buffer and miscellaneous tweaks - Adds implementation commentary to document remainder_buffer purpose - Also adds sample average for square wave boundary in case where remainder_buffer not used (whole-sample boundary) - Variable renaming and code shuffling REFACTOR : do not dynamically alloc remainder buffer - Never attribute to cache-coherancy bugs what is a simple thread race =P REFACTOR : comments and whitespace REFACTOR : rename public speaker API functions REFACTOR : clean up public speaker API REFACTOR : tabs to spaces REFACTOR : moar deadc0de clean up and renaming REFACTOR : remove deadc0de paths from soundcore REFACTOR : fully excise soundtype stuff now that we only support soundcard output Move a file static to function scope REFACTOR : rename more variables and remove deadc0de REFACTOR : samples_buffer naming and change to explict int16_t REFACTOR : removed deadc0de and shuffled code locations REFACTOR : remainder buffer naming and clarify type REFACTOR : move joystick timing to VM module and remove header visibility REFACTOR : clarify speaker variable name REFACTOR : clarify cycle counting codepaths REFACTOR : VBL/timing interfaces - eliminates passing around a common global REFACTOR : names and comments HACK around volume issue REFACTOR : rename speaker feedback variable REFACTOR : rename global total cycle count REFACTOR : rename a constant Fix test builds REFACTOR: rename to is_fullspeed REFACTOR : local variable naming changes REFACTOR : migrate cycle timing variables to correct location and remove header visibility Allow fullspeed codepath to update speaker REFACTOR : remove deadc0de paths in prep for cleanup REFACTOR : speaker now manages its own VM entry point
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int err = 0;
ALPlayBuf *node = PlaylistEnqueue(voice, voice->index);
if (!node)
{
return -1;
}
//LOG("Enqueing OpenAL buffer %u (%u bytes)", node->bufid, node->bytes);
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alBufferData(node->bufid, voice->format, voice->data, node->bytes, voice->rate);
if ((err = alGetError()) != AL_NO_ERROR)
{
PlaylistDequeue(voice, node);
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ERRLOG("OOPS, Error alBufferData : 0x%08x", err);
return err;
}
alSourceQueueBuffers(voice->source, 1, &node->bufid);
if ((err = alGetError()) != AL_NO_ERROR)
{
PlaylistDequeue(voice, node);
ERRLOG("OOPS, Error buffering data : 0x%08x", err);
return err;
}
ALint state = 0;
alGetSourcei(voice->source, AL_SOURCE_STATE, &state);
if ((err = alGetError()) != AL_NO_ERROR)
{
ERRLOG("OOPS, Error checking source state : 0x%08x", err);
return err;
}
if ((state != AL_PLAYING) && (state != AL_PAUSED))
{
// 2013/11/17 NOTE : alSourcePlay() is expensive and causes audio artifacts, only invoke if needed
LOG("Restarting playback (was 0x%08x) ...", state);
alSourcePlay(voice->source);
if ((err = alGetError()) != AL_NO_ERROR)
{
LOG("Error starting playback : 0x%08x", err);
return err;
}
}
return 0;
}
// DS->Unlock()
static long ALCommit(void *_this, void *unused_audio_ptr1, unsigned long audio_bytes1, void *unused_audio_ptr2, unsigned long unused_audio_bytes2)
{
ALVoice *voice = (ALVoice*)_this;
int err = 0;
ALuint bytes_queued = 0;
err = _ALProcessPlayBuffers(voice, &bytes_queued);
if (err)
{
return err;
}
voice->index += audio_bytes1;
while (voice->index > voice->buffersize)
{
// hopefully this is DEADC0DE or we've overwritten voice->data buffer ...
ERRLOG("OOPS, overflow in queued sound data");
assert(false);
}
if (bytes_queued >= (voice->buffersize>>2)/*quarter buffersize*/)
{
// keep accumulating data into working buffer
return 0;
}
if (HASH_COUNT(voice->queued_buffers) >= (OPENAL_NUM_BUFFERS))
{
//LOG("no free audio buffers"); // keep accumulating ...
return 0;
}
// ---------------------------
// Submit working buffer to OpenAL
err = _ALSubmitBufferToOpenAL(voice);
if (err)
{
return err;
}
return 0;
}
// HACK Part I : done once for mockingboard that has semiauto repeating phonemes ...
static long ALCommitStaticBuffer(void *_this, unsigned long audio_bytes1)
{
ALVoice *voice = (ALVoice*)_this;
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voice->replay_index = (ALsizei)audio_bytes1;
return 0;
}
// HACK Part II : replay mockingboard phoneme ...
static long ALReplay(void *_this)
{
ALVoice *voice = (ALVoice*)_this;
voice->index = voice->replay_index;
int err = 0;
err = _ALSubmitBufferToOpenAL(voice);
if (err)
{
return err;
}
return 0;
}
static long ALGetStatus(void *_this, unsigned long *status)
{
ALVoice* voice = (ALVoice*)_this;
int err = 0;
ALint state = 0;
alGetSourcei(voice->source, AL_SOURCE_STATE, &state);
if ((err = alGetError()) != AL_NO_ERROR)
{
ERRLOG("OOPS, Error checking source state : 0x%08x", err);
return err;
}
if ((state == AL_PLAYING) || (state == AL_PAUSED))
{
*status = DSBSTATUS_PLAYING;
}
else
{
*status = _DSBSTATUS_NOTPLAYING;
}
return 0;
}
static long OpenALCreateSoundBuffer(AudioParams_s *params, ALSoundBufferStruct **soundbuf_struct, void *extra_data)
{
LOG("OpenALCreateSoundBuffer ...");
assert(*soundbuf_struct == NULL);
const SoundSystemStruct *sound_struct = (SoundSystemStruct*)extra_data;
ALCcontext *ctx = (ALCcontext*)(sound_struct->implementation_specific);
assert(ctx != NULL);
ALVoice *voice = NULL;
do {
if ((voice = NewVoice(params)) == NULL)
{
ERRLOG("OOPS, Cannot create new voice");
break;
}
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ALVoices immutableNode = { .source = voice->source };
ALVoices *vnode = calloc(1, sizeof(ALVoices));
if (!vnode)
{
ERRLOG("OOPS, Not enough memory");
break;
}
memcpy(vnode, &immutableNode, sizeof(ALVoices));
vnode->voice = voice;
HASH_ADD_INT(voices, source, vnode);
if ((*soundbuf_struct = malloc(sizeof(ALSoundBufferStruct))) == NULL)
{
ERRLOG("OOPS, Not enough memory");
break;
}
(*soundbuf_struct)->_this = voice;
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(*soundbuf_struct)->SetVolume = (int (*)(void *, long))ALSetVolume;
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(*soundbuf_struct)->GetVolume = (int (*)(void *, long *))ALGetVolume;
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(*soundbuf_struct)->GetCurrentPosition = (int (*)(void *, unsigned long *, unsigned long *))ALGetPosition;
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(*soundbuf_struct)->Stop = (int (*)(void *))ALStop;
(*soundbuf_struct)->Restore = (int (*)(void *))ALRestore;
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(*soundbuf_struct)->Play = (int (*)(void *, unsigned long, unsigned long, unsigned long))ALPlay;
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(*soundbuf_struct)->Lock = (int (*)(void *, unsigned long, unsigned long, void **, unsigned long *, void **, unsigned long *, unsigned long))ALBegin;
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(*soundbuf_struct)->Unlock = (int (*)(void *, void *, unsigned long, void *, unsigned long))ALCommit;
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(*soundbuf_struct)->GetStatus = (int (*)(void *, unsigned long *))ALGetStatus;
// mockingboard-specific hacks
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(*soundbuf_struct)->UnlockStaticBuffer = (int (*)(void *, unsigned long))ALCommitStaticBuffer;
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(*soundbuf_struct)->Replay = (int (*)(void *))ALReplay;
return 0;
} while(0);
if (*soundbuf_struct)
{
OpenALDestroySoundBuffer(soundbuf_struct);
}
else if (voice)
{
DeleteVoice(voice);
}
return -1;
}
static long OpenALDestroySoundBuffer(ALSoundBufferStruct **soundbuf_struct)
{
if (!*soundbuf_struct) {
// already dealloced
return 0;
}
LOG("OpenALDestroySoundBuffer ...");
ALVoice *voice = (*soundbuf_struct)->_this;
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ALint source = voice->source;
DeleteVoice(voice);
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ALVoices *vnode = NULL;
HASH_FIND_INT(voices, &source, vnode);
if (vnode) {
HASH_DEL(voices, vnode);
FREE(vnode);
}
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FREE(*soundbuf_struct);
return 0;
}