First cut at Android OpenSLES backend

This commit is contained in:
Aaron Culliney 2015-06-25 21:32:37 -07:00
parent c4397123cb
commit bb8d51817b
7 changed files with 871 additions and 15 deletions

View File

@ -13,7 +13,7 @@ include $(COMMON_SOURCES_MK)
LOCAL_MODULE := libapple2ix
LOCAL_SRC_FILES := jnihooks.c androidkeys.c
LOCAL_CFLAGS := $(APPLE2_BASE_CFLAGS) -DHEADLESS=0
LOCAL_LDLIBS := -llog -landroid -lGLESv2 -lz
LOCAL_LDLIBS := -llog -landroid -lGLESv2 -lz -lOpenSLES
# Add assembly files first ... mostly for the benefit of the ARM assembler ...
ifeq ($(TARGET_ARCH_ABI),x86)
@ -22,7 +22,7 @@ else
LOCAL_SRC_FILES += $(APPLE2_ARM_SRC)
endif
LOCAL_SRC_FILES += $(APPLE2_MAIN_SRC) $(APPLE2_META_SRC) $(APPLE2_VIDEO_SRC)
LOCAL_SRC_FILES += $(APPLE2_MAIN_SRC) $(APPLE2_META_SRC) $(APPLE2_VIDEO_SRC) $(APPLE2_AUDIO_SRC)
# Build a shared library and let Java/Dalvik drive
include $(BUILD_SHARED_LIBRARY)

View File

@ -2,7 +2,7 @@
package_id="org.deadc0de.apple2ix"
apple2_src_path=apple2ix-src
glue_srcs="$apple2_src_path/disk.c $apple2_src_path/misc.c $apple2_src_path/display.c $apple2_src_path/vm.c $apple2_src_path/cpu-supp.c"
glue_srcs="$apple2_src_path/disk.c $apple2_src_path/misc.c $apple2_src_path/display.c $apple2_src_path/vm.c $apple2_src_path/cpu-supp.c $apple2_src_path/audio/speaker.c $apple2_src_path/audio/mockingboard.c"
do_load=0
do_debug=0
@ -90,7 +90,7 @@ else
ln -s apple2ix.mk Android.mk
fi
ndk-build V=1 NDK_DEBUG=1
ndk-build V=1 NDK_DEBUG=1 #NDK_TOOLCHAIN_VERSION=clang
#ndk-build V=1 NDK_DEBUG=1 && \
# ant -f ../build.xml debug

View File

@ -23,8 +23,8 @@ APPLE2_VIDEO_SRC = \
$(APPLE2_SRC_PATH)/video_util/vectorUtil.c
APPLE2_AUDIO_SRC = \
$(APPLE2_SRC_PATH)/audio/soundcore.c $(APPLE2_SRC_PATH)/audio/soundcore-openal.c $(APPLE2_SRC_PATH)/audio/speaker.c \
$(APPLE2_SRC_PATH)/audio/alhelpers.c $(APPLE2_SRC_PATH)/audio/mockingboard.c $(APPLE2_SRC_PATH)/audio/AY8910.c
$(APPLE2_SRC_PATH)/audio/soundcore.c $(APPLE2_SRC_PATH)/audio/soundcore-opensles.c $(APPLE2_SRC_PATH)/audio/speaker.c \
$(APPLE2_SRC_PATH)/audio/playqueue.c $(APPLE2_SRC_PATH)/audio/mockingboard.c $(APPLE2_SRC_PATH)/audio/AY8910.c
APPLE2_META_SRC = \
$(APPLE2_SRC_PATH)/meta/debug.c $(APPLE2_SRC_PATH)/meta/debugger.c $(APPLE2_SRC_PATH)/meta/opcodes.c \
@ -35,5 +35,5 @@ APPLE2_MAIN_SRC = \
$(APPLE2_SRC_PATH)/timing.c $(APPLE2_SRC_PATH)/zlib-helpers.c $(APPLE2_SRC_PATH)/joystick.c $(APPLE2_SRC_PATH)/keys.c \
$(APPLE2_SRC_PATH)/interface.c $(APPLE2_SRC_PATH)/disk.c $(APPLE2_SRC_PATH)/cpu-supp.c
APPLE2_BASE_CFLAGS := -DAPPLE2IX=1 -DINTERFACE_TOUCH=1 -DMOBILE_DEVICE=1 -DVIDEO_OPENGL=1 -DDEBUGGER=1 -std=gnu11 -I$(APPLE2_SRC_PATH)
APPLE2_BASE_CFLAGS := -DAPPLE2IX=1 -DINTERFACE_TOUCH=1 -DMOBILE_DEVICE=1 -DVIDEO_OPENGL=1 -DDEBUGGER=1 -DAUDIO_ENABLED=1 -std=gnu11 -I$(APPLE2_SRC_PATH)

View File

@ -87,7 +87,7 @@ Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
#include "common.h"
#ifdef APPLE2IX
# ifdef __linux
# if defined(__linux) && !defined(ANDROID)
# include <sys/io.h>
# endif
@ -302,7 +302,7 @@ pthread_t CreateThread(void* unused_lpThreadAttributes, int unused_dwStackSize,
bool SetThreadPriority(pthread_t thread, int unused_nPriority)
{
// assuming time critical ...
#if defined(__APPLE__)
#if defined(__APPLE__) || defined(ANDROID)
#warning possible FIXME possible TODO : set thread priority in Darwin/Mach ?
#else
int policy = sched_getscheduler(getpid());
@ -327,7 +327,7 @@ bool SetThreadPriority(pthread_t thread, int unused_nPriority)
bool GetExitCodeThread(pthread_t thread, unsigned long *lpExitCode)
{
#if defined(__APPLE__)
#if defined(__APPLE__) || defined(ANDROID)
int err = 0;
if ( (err = pthread_join(thread, NULL)) ) {
ERRLOG("OOPS pthread_join");
@ -893,6 +893,10 @@ static void Votrax_Write(uint8_t nDevice, uint8_t nValue)
static void MB_Update()
{
#ifdef APPLE2IX
if (!audio_isAvailable) {
return;
}
static int nNumSamplesError = 0;
if (!MockingboardVoice->bActive || !g_bMB_Active)
{

View File

@ -0,0 +1,836 @@
/*
* 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.
*
*/
// soundcore OpenSLES backend -- streaming audio
#include "common.h"
#include <SLES/OpenSLES.h>
#if defined(ANDROID)
# include <SLES/OpenSLES_Android.h>
#else
# error FIXME TODO this currently uses Android BufferQueue extensions...
#endif
#include "playqueue.h"
#include "uthash.h"
#define DEBUG_OPENSL 0
#if DEBUG_OPENSL
# define OPENSL_LOG(...) LOG(__VA_ARGS__)
#else
# define OPENSL_LOG(...)
#endif
#define OPENSL_NUM_BUFFERS 4
typedef struct SLVoice {
unsigned int voiceId;
SLObjectItf bqPlayerObject;
SLPlayItf bqPlayerPlay;
SLAndroidSimpleBufferQueueItf bqPlayerBufferQueue;
SLMuteSoloItf bqPlayerMuteSolo;
SLVolumeItf bqPlayerVolume;
// OpenSLES buffer queue management
PlayQueue_s *playq;
pthread_mutex_t bqThreadLock;
long currentNodeId;
unsigned long currentNumBytes;
SLmillisecond startingPosition;
SLmillisecond currentBufferDuration;
long queuedTotalBytes; // a maximum estimate -- actual value depends on query
bool bufferIsPlaying;
// working data buffer
uint8_t *data;
size_t index; // working buffer byte index
size_t buffersize; // working buffer size
size_t replay_index;
// misc
uint16_t nChannels;
unsigned long nSamplesPerSec;
} SLVoice;
typedef struct SLVoices {
unsigned int voiceId;
SLVoice *voice;
UT_hash_handle hh;
} SLVoices;
typedef struct EngineContext_s {
SLObjectItf engineObject;
SLEngineItf engineEngine;
SLObjectItf outputMixObject;
} EngineContext_s;
static SLVoices *voices = NULL;
static AudioBackend_s opensles_audio_backend = { 0 };
// ----------------------------------------------------------------------------
// AudioBuffer_s processing routines
static long _SLGetCurrentQueuedBytes(SLVoice *voice, unsigned int *bytes_queued) {
*bytes_queued = 0;
long err = 0;
do {
SLmillisecond position = 0;
long play_offset = 0;
// ------------------------------ LOCK
pthread_mutex_lock(&(voice->bqThreadLock));
SLresult result = (*(voice->bqPlayerPlay))->GetPosition(voice->bqPlayerPlay, &position);
if (result != SL_RESULT_SUCCESS) {
ERRLOG("Could not get position of current sample");
}
assert(position >= 0);
assert(voice->startingPosition >= 0);
SLmillisecond positionInCurrentBuffer = 0;
if (position < voice->startingPosition) {
OPENSL_LOG("OpenSLES wrapping position!");
} else {
positionInCurrentBuffer = position - voice->startingPosition;
}
if (voice->currentBufferDuration) {
float scale = positionInCurrentBuffer/(float)(voice->currentBufferDuration);
if (scale > 1.f) {
//OPENSL_LOG("OOPS scale > 1.f!");
play_offset = voice->currentNumBytes;
} else {
play_offset = (long)(voice->currentNumBytes * scale);
}
//OPENSL_LOG("totalQueuedBytes:%ld currentNumBytes:%ld startingPosition:%ld position:%ld inCurrentBuff:%ld scale:%f (play_offset:%ld)", voice->queuedTotalBytes, voice->currentNumBytes, voice->startingPosition, position, positionInCurrentBuffer, scale, play_offset);
} else {
//OPENSL_LOG("totalQueuedBytes:%ld currentNumBytes:%ld startingPosition:%ld position:%ld inCurrentBuff:%ld (play_offset:%ld)", voice->queuedTotalBytes, voice->currentNumBytes, voice->startingPosition, position, positionInCurrentBuffer, play_offset);
}
long q = voice->queuedTotalBytes - play_offset;
pthread_mutex_unlock(&(voice->bqThreadLock));
// ---------------------------- UNLOCK
if (q > 0) {
*bytes_queued = q;
}
} while (0);
return err;
}
// returns queued+working sound buffer size in bytes
static long SLGetPosition(AudioBuffer_s *_this, OUTPARM unsigned long *bytes_queued) {
*bytes_queued = 0;
long err = 0;
do {
SLVoice *voice = (SLVoice*)_this->_internal;
unsigned int queued = 0;
long err = _SLGetCurrentQueuedBytes(voice, &queued);
if (err) {
break;
}
*bytes_queued = queued + voice->index;
} while (0);
return err;
}
static long SLLockBuffer(AudioBuffer_s *_this, unsigned long write_bytes, INOUT int16_t **audio_ptr, OUTPARM unsigned long *audio_bytes) {
*audio_bytes = 0;
*audio_ptr = NULL;
long err = 0;
//OPENSL_LOG("SLLockBuffer request for %ld bytes", write_bytes);
do {
SLVoice *voice = (SLVoice*)_this->_internal;
if (write_bytes == 0) {
write_bytes = voice->buffersize;
}
unsigned int bytes_queued = 0;
err = _SLGetCurrentQueuedBytes(voice, &bytes_queued);
if (err) {
break;
}
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;
}
#if 0
else if (bytes_queued + voice->index < (voice->buffersize>>3)/* 1/8 buffer */)
{
LOG("Potential underrun ...");
}
#endif
unsigned int remaining = voice->buffersize - voice->index;
if (write_bytes > remaining) {
write_bytes = remaining;
}
*audio_ptr = (int16_t *)(voice->data+voice->index);
*audio_bytes = write_bytes;
} while (0);
return err;
}
static SLresult _send_buffer_to_opensles(SLVoice *voice, PlayNode_s *node) {
voice->currentNodeId = node->nodeId;
voice->currentNumBytes = node->numBytes;
// calculate the new buffer duration
unsigned long numSamples = (node->numBytes>>1);
if (voice->nChannels == 2) {
unsigned long numSamplesStereo = numSamples;
numSamples = (numSamplesStereo>>1);
} else if (voice->nChannels !=1) {
assert(false && "only mono or stereo supported");
}
voice->currentBufferDuration = ( numSamples / (voice->nSamplesPerSec/1000.f) );
OPENSL_LOG("Enqueing OpenSL buffer %ld (%lu bytes, %lu millis)", node->nodeId, node->numBytes, voice->currentBufferDuration, node->bytes);
// enqueue buffer to OpenSLES
SLresult result = (*(voice->bqPlayerBufferQueue))->Enqueue(voice->bqPlayerBufferQueue, node->bytes, node->numBytes);
if (result != SL_RESULT_SUCCESS) {
ERRLOG("OOPS ... buffer queue callback enqueue reports %ld", result); // the most likely other result is SL_RESULT_BUFFER_INSUFFICIENT
}
return result;
}
static long _SLSubmitBuffer(SLVoice *voice) {
SLresult err = 0;
do {
// Micro-manage play queue locally to understand the total bytes-in-play
PlayNode_s playNode = {
.nodeId = INVALID_NODE_ID,
.numBytes = voice->index,
.bytes = (uint8_t *)(voice->data),
};
//OPENSL_LOG("_SLSubmitBuffer : %ld bytes", voice->index);
bool isCurrentlyPlaying = false;
// ------------------------------ LOCK
pthread_mutex_lock(&(voice->bqThreadLock));
err = voice->playq->Enqueue(voice->playq, &playNode);
if (err) {
pthread_mutex_unlock(&(voice->bqThreadLock));
break;
}
voice->queuedTotalBytes += voice->index;
voice->index = 0;
assert(voice->queuedTotalBytes > 0);
isCurrentlyPlaying = voice->bufferIsPlaying;
err = SL_RESULT_UNKNOWN_ERROR;
if (!isCurrentlyPlaying) {
err = _send_buffer_to_opensles(voice, &playNode);
if (err != SL_RESULT_SUCCESS) {
ERRLOG("OOPS ... buffer queue callback enqueue reports %ld", err); // the most likely other result is SL_RESULT_BUFFER_INSUFFICIENT
pthread_mutex_unlock(&(voice->bqThreadLock));
break;
}
}
voice->bufferIsPlaying = true;
pthread_mutex_unlock(&(voice->bqThreadLock));
// ---------------------------- UNLOCK
SLuint32 state = 0;
err = (*(voice->bqPlayerPlay))->GetPlayState(voice->bqPlayerPlay, &state);
if (err != SL_RESULT_SUCCESS) {
ERRLOG("OOPS, could not get source state : %lu", err);
break;
}
if ((state != SL_PLAYSTATE_PLAYING) && (state != SL_PLAYSTATE_PAUSED)) {
LOG("Restarting playback (was %lu) ...", state);
err = (*(voice->bqPlayerPlay))->SetPlayState(voice->bqPlayerPlay, SL_PLAYSTATE_PLAYING);
if (err != SL_RESULT_SUCCESS) {
ERRLOG("OOPS, Failed to pause source : %lu", err);
break;
}
}
} while (0);
return err;
}
static long SLUnlockBuffer(AudioBuffer_s *_this, unsigned long audio_bytes) {
long err = 0;
do {
SLVoice *voice = (SLVoice*)_this->_internal;
unsigned int bytes_queued = 0;
err = _SLGetCurrentQueuedBytes(voice, &bytes_queued);
if (err) {
break;
}
voice->index += audio_bytes;
assert((voice->index < voice->buffersize) && "OOPS, overflow in queued sound data");
if (bytes_queued >= (voice->buffersize>>2)/*quarter buffersize*/) {
// keep accumulating data into working buffer
//OPENSL_LOG("accumulating more data %lu -> (queued:%u/buffersize:%u) prequeued:%u", audio_bytes, bytes_queued, voice->buffersize, bytes_queued+voice->index);
break;
} else {
//OPENSL_LOG("possibly submit %lu -> (queued:%u/buffersize:%u)", audio_bytes, bytes_queued, voice->buffersize);
}
if (! (voice->playq->CanEnqueue(voice->playq)) ) {
//LOG("no free audio buffers"); // keep accumulating ...
break;
}
// Submit working buffer
err = _SLSubmitBuffer(voice);
} while (0);
return err;
}
// HACK Part I : done once for mockingboard that has semiauto repeating phonemes ...
static long SLUnlockStaticBuffer(AudioBuffer_s *_this, unsigned long audio_bytes) {
SLVoice *voice = (SLVoice*)_this->_internal;
voice->replay_index = audio_bytes;
return 0;
}
// HACK Part II : replay mockingboard phoneme ...
static long SLReplay(AudioBuffer_s *_this) {
SLVoice *voice = (SLVoice*)_this->_internal;
voice->index = voice->replay_index;
long err = _SLSubmitBuffer(voice);
return err;
}
static long SLGetStatus(AudioBuffer_s *_this, OUTPARM unsigned long *status) {
*status = -1;
SLresult result = SL_RESULT_UNKNOWN_ERROR;
do {
SLVoice* voice = (SLVoice*)_this->_internal;
SLuint32 state = 0;
result = (*(voice->bqPlayerPlay))->GetPlayState(voice->bqPlayerPlay, &state);
if (result != SL_RESULT_SUCCESS) {
ERRLOG("OOPS, could not get source state : %lu", result);
break;
}
if ((state == SL_PLAYSTATE_PLAYING) || (state == SL_PLAYSTATE_PAUSED)) {
*status = AUDIO_STATUS_PLAYING;
} else {
*status = AUDIO_STATUS_NOTPLAYING;
}
} while (0);
return (long)result;
}
// ----------------------------------------------------------------------------
// SLVoice is the AudioBuffer_s->_internal
// this callback handler is called every time a buffer finishes playing
static void bqPlayerCallback(SLAndroidSimpleBufferQueueItf bq, void *context) {
SLVoice *voice = (SLVoice *)context;
SLresult result = SL_RESULT_UNKNOWN_ERROR;
assert(voice != NULL);
assert(bq == voice->bqPlayerBufferQueue);
assert(pthread_self() != cpu_thread_id);
// ------------------------------ LOCK
pthread_mutex_lock(&(voice->bqThreadLock));
do {
// reset starting position to current best estimate
result = (*(voice->bqPlayerPlay))->GetPosition(voice->bqPlayerPlay, &(voice->startingPosition));
if (result != SL_RESULT_SUCCESS) {
ERRLOG("Could not get position of current sample");
voice->startingPosition = 0;
}
// dequeue finished buffer and reset in-play stats
PlayNode_s head = { 0 };
voice->playq->Dequeue(voice->playq, &head);
assert(voice->currentNodeId == head.nodeId);
voice->queuedTotalBytes -= head.numBytes;
voice->currentNodeId = -1;
voice->currentNumBytes = 0;
voice->currentBufferDuration = 0;
// get current queue head
long err = voice->playq->GetHead(voice->playq, &head);
if (err) {
RELEASE_ERRLOG("Could not get head and size of queue!");
voice->bufferIsPlaying = false;
break;
}
result = _send_buffer_to_opensles(voice, &head);
if (result != SL_RESULT_SUCCESS) {
RELEASE_ERRLOG("Could not submit buffer to OpenSLES!");
voice->bufferIsPlaying = false;
break;
}
} while (0);
pthread_mutex_unlock(&(voice->bqThreadLock));
// ---------------------------- UNLOCK
}
static void _opensl_destroyVoice(SLVoice *voice) {
// destroy buffer queue audio player object, and invalidate all associated interfaces
if (voice->bqPlayerObject != NULL) {
(*(voice->bqPlayerObject))->Destroy(voice->bqPlayerObject);
voice->bqPlayerObject = NULL;
voice->bqPlayerPlay = NULL;
voice->bqPlayerBufferQueue = NULL;
voice->bqPlayerMuteSolo = NULL;
voice->bqPlayerVolume = NULL;
}
if (voice->data) {
FREE(voice->data);
}
playq_destroyPlayQueue(&(voice->playq));
memset(voice, 0, sizeof(*voice));
FREE(voice);
}
static SLVoice *_opensl_createVoice(const AudioParams_s *params, const EngineContext_s *ctx) {
SLVoice *voice = NULL;
do {
//
// General buffer memory management
//
voice = calloc(1, sizeof(*voice));
if (voice == NULL) {
ERRLOG("OOPS, Out of memory!");
break;
}
long longBuffers[OPENSL_NUM_BUFFERS];
for (unsigned int i=0; i<OPENSL_NUM_BUFFERS; i++) {
longBuffers[i] = i+1;
}
voice->playq = playq_createPlayQueue(longBuffers, OPENSL_NUM_BUFFERS);
if (!voice->playq) {
ERRLOG("OOPS, Not enough memory for PlayQueue");
break;
}
assert(params->nSamplesPerSec == SPKR_SAMPLE_RATE);
assert(params->nChannels == 1 || params->nChannels == 2);
voice->nChannels = params->nChannels;
voice->nSamplesPerSec = params->nSamplesPerSec;
SLuint32 channelMask = 0;
if (voice->nChannels == 2) {
channelMask = SL_SPEAKER_FRONT_LEFT | SL_SPEAKER_FRONT_RIGHT;
} else {
channelMask = SL_SPEAKER_FRONT_CENTER;
}
// Allocate enough space for the temp buffer
voice->buffersize = params->dwBufferBytes;
voice->data = malloc(voice->buffersize);
if (voice->data == NULL) {
ERRLOG("OOPS, Error allocating %d bytes", voice->buffersize);
break;
}
//
// OpenSLES buffer queue player setup
//
SLresult result = SL_RESULT_UNKNOWN_ERROR;
// configure audio source
SLDataLocator_AndroidSimpleBufferQueue loc_bufq = {
.locatorType = SL_DATALOCATOR_ANDROIDSIMPLEBUFFERQUEUE,
.numBuffers = 2,
#warning FIXME TODO ... verify 2 numBuffers is best
};
SLDataFormat_PCM format_pcm = {
.formatType = SL_DATAFORMAT_PCM,
.numChannels = params->nChannels,
.samplesPerSec = SL_SAMPLINGRATE_22_05,
.bitsPerSample = SL_PCMSAMPLEFORMAT_FIXED_16,
.containerSize = SL_PCMSAMPLEFORMAT_FIXED_16,
.channelMask = channelMask,
.endianness = SL_BYTEORDER_LITTLEENDIAN,
};
SLDataSource audioSrc = {
.pLocator = &loc_bufq,
.pFormat = &format_pcm,
};
// configure audio sink
SLDataLocator_OutputMix loc_outmix = {
.locatorType = SL_DATALOCATOR_OUTPUTMIX,
.outputMix = ctx->outputMixObject,
};
SLDataSink audioSnk = {
.pLocator = &loc_outmix,
.pFormat = NULL,
};
// create audio player
#define _NUM_INTERFACES 3
const SLInterfaceID ids[_NUM_INTERFACES] = {
SL_IID_BUFFERQUEUE,
SL_IID_EFFECTSEND,
//SL_IID_MUTESOLO,
SL_IID_VOLUME,
};
const SLboolean req[_NUM_INTERFACES] = {
SL_BOOLEAN_TRUE,
SL_BOOLEAN_TRUE,
//params->nChannels == 1 ? SL_BOOLEAN_FALSE : SL_BOOLEAN_TRUE,
SL_BOOLEAN_TRUE,
};
result = (*(ctx->engineEngine))->CreateAudioPlayer(ctx->engineEngine, &(voice->bqPlayerObject), &audioSrc, &audioSnk, _NUM_INTERFACES, ids, req);
if (result != SL_RESULT_SUCCESS) {
ERRLOG("OOPS, could not create the BufferQueue player object : %lu", result);
break;
}
// realize the player
result = (*(voice->bqPlayerObject))->Realize(voice->bqPlayerObject, /*asynchronous_realization:*/SL_BOOLEAN_FALSE);
if (result != SL_RESULT_SUCCESS) {
ERRLOG("OOPS, could not realize the BufferQueue player object : %lu", result);
break;
}
// get the play interface
result = (*(voice->bqPlayerObject))->GetInterface(voice->bqPlayerObject, SL_IID_PLAY, &(voice->bqPlayerPlay));
if (result != SL_RESULT_SUCCESS) {
ERRLOG("OOPS, could not get the play interface : %lu", result);
break;
}
// get the buffer queue interface
result = (*(voice->bqPlayerObject))->GetInterface(voice->bqPlayerObject, SL_IID_BUFFERQUEUE, &(voice->bqPlayerBufferQueue));
if (result != SL_RESULT_SUCCESS) {
ERRLOG("OOPS, could not get the BufferQueue play interface : %lu", result);
break;
}
// register callback on the buffer queue
result = (*(voice->bqPlayerBufferQueue))->RegisterCallback(voice->bqPlayerBufferQueue, bqPlayerCallback, voice);
if (result != SL_RESULT_SUCCESS) {
ERRLOG("OOPS, could not register BufferQueue callback : %lu", result);
break;
}
#if 0 // mute/solo is not supported for sources that are known to be mono, as this is
// get the mute/solo interface
result = (*(voice->bqPlayerObject))->GetInterface(voice->bqPlayerObject, SL_IID_MUTESOLO, &(voice->bqPlayerMuteSolo));
assert(SL_RESULT_SUCCESS == result);
(void)result;
#endif
// get the volume interface
result = (*(voice->bqPlayerObject))->GetInterface(voice->bqPlayerObject, SL_IID_VOLUME, &(voice->bqPlayerVolume));
if (result != SL_RESULT_SUCCESS) {
ERRLOG("OOPS, could not get the BufferQueue volume interface : %lu", result);
break;
}
return voice;
} while(0);
// ERR
if (voice) {
_opensl_destroyVoice(voice);
}
return NULL;
}
// ----------------------------------------------------------------------------
static long opensl_destroySoundBuffer(INOUT AudioBuffer_s **soundbuf_struct) {
if (!*soundbuf_struct) {
return 0;
}
LOG("opensl_destroySoundBuffer ...");
SLVoice *voice = (SLVoice *)((*soundbuf_struct)->_internal);
unsigned int voiceId = voice->voiceId;
_opensl_destroyVoice(voice);
SLVoices *vnode = NULL;
HASH_FIND_INT(voices, &voiceId, vnode);
if (vnode) {
HASH_DEL(voices, vnode);
FREE(vnode);
}
FREE(*soundbuf_struct);
return 0;
}
static long opensl_createSoundBuffer(const AudioParams_s *params, INOUT AudioBuffer_s **soundbuf_struct, const AudioContext_s *audio_context) {
LOG("opensl_createSoundBuffer ...");
assert(*soundbuf_struct == NULL);
SLVoice *voice = NULL;
do {
EngineContext_s *ctx = (EngineContext_s *)(audio_context->_internal);
assert(ctx != NULL);
if ((voice = _opensl_createVoice(params, ctx)) == NULL)
{
ERRLOG("OOPS, Cannot create new voice");
break;
}
SLVoices *vnode = calloc(1, sizeof(SLVoices));
if (!vnode) {
ERRLOG("OOPS, Not enough memory");
break;
}
static unsigned int counter = 0;
vnode->voiceId = __sync_add_and_fetch(&counter, 1);
voice->voiceId = vnode->voiceId;
vnode->voice = voice;
HASH_ADD_INT(voices, voice->voiceId, vnode);
if ((*soundbuf_struct = malloc(sizeof(AudioBuffer_s))) == NULL) {
ERRLOG("OOPS, Not enough memory");
break;
}
(*soundbuf_struct)->_internal = voice;
(*soundbuf_struct)->GetCurrentPosition = &SLGetPosition;
(*soundbuf_struct)->Lock = &SLLockBuffer;
(*soundbuf_struct)->Unlock = &SLUnlockBuffer;
(*soundbuf_struct)->GetStatus = &SLGetStatus;
// mockingboard-specific hacks
(*soundbuf_struct)->UnlockStaticBuffer = &SLUnlockStaticBuffer;
(*soundbuf_struct)->Replay = &SLReplay;
return 0;
} while(0);
if (*soundbuf_struct) {
opensl_destroySoundBuffer(soundbuf_struct);
} else if (voice) {
_opensl_destroyVoice(voice);
}
return -1;
}
// ----------------------------------------------------------------------------
static long opensles_systemShutdown(AudioContext_s **audio_context) {
assert(*audio_context != NULL);
EngineContext_s *ctx = (EngineContext_s *)((*audio_context)->_internal);
assert(ctx != NULL);
// destroy output mix object, and invalidate all associated interfaces
if (ctx->outputMixObject != NULL) {
(*(ctx->outputMixObject))->Destroy(ctx->outputMixObject);
ctx->outputMixObject = NULL;
}
// destroy engine object, and invalidate all associated interfaces
if (ctx->engineObject != NULL) {
(*(ctx->engineObject))->Destroy(ctx->engineObject);
ctx->engineObject = NULL;
ctx->engineEngine = NULL;
}
FREE(ctx);
FREE(*audio_context);
return 0;
}
static long opensles_systemSetup(INOUT AudioContext_s **audio_context) {
assert(*audio_context == NULL);
assert(voices == NULL);
EngineContext_s *ctx = NULL;
SLresult result = -1;
do {
//
// Engine creation ...
//
ctx = calloc(1, sizeof(EngineContext_s));
if (!ctx) {
result = -1;
break;
}
// create basic engine
result = slCreateEngine(&(ctx->engineObject), 0, NULL, /*engineMixIIDCount:*/0, /*engineMixIIDs:*/NULL, /*engineMixReqs:*/NULL);
if (result != SL_RESULT_SUCCESS) {
ERRLOG("Could not create OpenSLES Engine : %lu", result);
break;
}
// realize the engine
result = (*(ctx->engineObject))->Realize(ctx->engineObject, /*asynchronous_realization:*/SL_BOOLEAN_FALSE);
if (result != SL_RESULT_SUCCESS) {
ERRLOG("Could not realize the OpenSLES Engine : %lu", result);
break;
}
// get the actual engine interface
result = (*(ctx->engineObject))->GetInterface(ctx->engineObject, SL_IID_ENGINE, &(ctx->engineEngine));
if (result != SL_RESULT_SUCCESS) {
ERRLOG("Could not get the OpenSLES Engine : %lu", result);
break;
}
//
// Output Mix ...
//
result = (*(ctx->engineEngine))->CreateOutputMix(ctx->engineEngine, &(ctx->outputMixObject), 0, NULL, NULL);
if (result != SL_RESULT_SUCCESS) {
ERRLOG("Could not create output mix : %lu", result);
break;
}
// realize the output mix
result = (*(ctx->outputMixObject))->Realize(ctx->outputMixObject, SL_BOOLEAN_FALSE);
if (result != SL_RESULT_SUCCESS) {
ERRLOG("Could not realize the output mix : %lu", result);
break;
}
// create soundcore API wrapper
if ((*audio_context = malloc(sizeof(AudioContext_s))) == NULL) {
result = -1;
ERRLOG("OOPS, Not enough memory");
break;
}
(*audio_context)->_internal = ctx;
(*audio_context)->CreateSoundBuffer = &opensl_createSoundBuffer;
(*audio_context)->DestroySoundBuffer = &opensl_destroySoundBuffer;
} while (0);
if (result != SL_RESULT_SUCCESS) {
if (ctx) {
AudioContext_s *ctxPtr = malloc(sizeof(AudioContext_s));
ctxPtr->_internal = ctx;
opensles_systemShutdown(&ctxPtr);
}
assert(*audio_context == NULL);
LOG("OpenSLES sound output disabled");
}
return result;
}
// pause all audio
static long opensles_systemPause(void) {
SLVoices *vnode = NULL;
SLVoices *tmp = NULL;
HASH_ITER(hh, voices, vnode, tmp) {
SLVoice *voice = vnode->voice;
SLresult result = (*(voice->bqPlayerPlay))->SetPlayState(voice->bqPlayerPlay, SL_PLAYSTATE_PAUSED);
if (result != SL_RESULT_SUCCESS) {
ERRLOG("OOPS, Failed to pause source : %lu", result);
}
}
return 0;
}
static long opensles_systemResume(void) {
SLVoices *vnode = NULL;
SLVoices *tmp = NULL;
int err = 0;
HASH_ITER(hh, voices, vnode, tmp) {
SLVoice *voice = vnode->voice;
SLresult result = (*(voice->bqPlayerPlay))->SetPlayState(voice->bqPlayerPlay, SL_PLAYSTATE_PLAYING);
if (result != SL_RESULT_SUCCESS) {
ERRLOG("OOPS, Failed to resume source : %lu", result);
}
}
return 0;
}
__attribute__((constructor(CTOR_PRIORITY_EARLY)))
static void _init_opensl(void) {
LOG("Initializing OpenSLES sound system");
assert(audio_backend == NULL && "there can only be one!");
opensles_audio_backend.setup = &opensles_systemSetup;
opensles_audio_backend.shutdown = &opensles_systemShutdown;
opensles_audio_backend.pause = &opensles_systemPause;
opensles_audio_backend.resume = &opensles_systemResume;
audio_backend = &opensles_audio_backend;
}

View File

@ -40,6 +40,11 @@ long audio_createSoundBuffer(INOUT AudioBuffer_s **pVoice, unsigned long dwBuffe
long err = 0;
do {
if (!audioContext) {
ERRLOG("Cannot create sound buffer, no context");
err = -1;
break;
}
err = audioContext->CreateSoundBuffer(&params, pVoice, audioContext);
if (err) {
break;
@ -50,7 +55,9 @@ long audio_createSoundBuffer(INOUT AudioBuffer_s **pVoice, unsigned long dwBuffe
}
void audio_destroySoundBuffer(INOUT AudioBuffer_s **audioBuffer) {
audioContext->DestroySoundBuffer(audioBuffer);
if (audioContext) {
audioContext->DestroySoundBuffer(audioBuffer);
}
}
bool audio_init(void) {

View File

@ -308,15 +308,22 @@ void speaker_destroy(void) {
void speaker_init(void) {
long err = audio_createSoundBuffer(&speakerBuffer, SOUNDCORE_BUFFER_SIZE, SPKR_SAMPLE_RATE, 1);
assert(!err);
_speaker_init_timing();
if (!err) {
_speaker_init_timing();
}
}
void speaker_reset(void) {
_speaker_init_timing();
if (audio_isAvailable) {
_speaker_init_timing();
}
}
void speaker_flush(void) {
if (!audio_isAvailable) {
return;
}
assert(pthread_self() == cpu_thread_id);
if (is_fullspeed) {
@ -398,7 +405,9 @@ GLUE_C_READ(speaker_toggle)
is_fullspeed = false;
}
_speaker_update(/*toggled:true*/);
if (audio_isAvailable) {
_speaker_update(/*toggled:true*/);
}
if (!is_fullspeed) {
if (speaker_data) {