Mirror of roytam1's UXP fork just in case Moonchild and Tobin decide to go after him
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/*
* Copyright © 2012 Mozilla Foundation
*
* This program is made available under an ISC-style license. See the
* accompanying file LICENSE for details.
*/
#undef NDEBUG
#include <assert.h>
#include <dlfcn.h>
#include <stdlib.h>
#include <pthread.h>
#include <SLES/OpenSLES.h>
#include <math.h>
#include <time.h>
#if defined(__ANDROID__)
#include <dlfcn.h>
#include <sys/system_properties.h>
#include "android/sles_definitions.h"
#include <SLES/OpenSLES_Android.h>
#include <android/log.h>
#include <android/api-level.h>
#define LOG(args...) __android_log_print(ANDROID_LOG_INFO, "Cubeb_OpenSL" , ## args)
#define ANDROID_VERSION_GINGERBREAD_MR1 10
#define ANDROID_VERSION_LOLLIPOP 21
#define ANDROID_VERSION_MARSHMALLOW 23
#endif
#include "cubeb/cubeb.h"
#include "cubeb-internal.h"
#include "cubeb_resampler.h"
#include "cubeb-sles.h"
static struct cubeb_ops const opensl_ops;
struct cubeb {
struct cubeb_ops const * ops;
void * lib;
void * libmedia;
int32_t (* get_output_latency)(uint32_t * latency, int stream_type);
SLInterfaceID SL_IID_BUFFERQUEUE;
SLInterfaceID SL_IID_PLAY;
#if defined(__ANDROID__)
SLInterfaceID SL_IID_ANDROIDCONFIGURATION;
#endif
SLInterfaceID SL_IID_VOLUME;
SLObjectItf engObj;
SLEngineItf eng;
SLObjectItf outmixObj;
};
#define NELEMS(A) (sizeof(A) / sizeof A[0])
#define NBUFS 4
#define AUDIO_STREAM_TYPE_MUSIC 3
struct cubeb_stream {
cubeb * context;
pthread_mutex_t mutex;
SLObjectItf playerObj;
SLPlayItf play;
SLBufferQueueItf bufq;
SLVolumeItf volume;
uint8_t *queuebuf[NBUFS];
int queuebuf_idx;
long queuebuf_len;
long bytespersec;
long framesize;
long written;
int draining;
cubeb_stream_type stream_type;
cubeb_data_callback data_callback;
cubeb_state_callback state_callback;
void * user_ptr;
cubeb_resampler * resampler;
unsigned int inputrate;
unsigned int outputrate;
unsigned int latency;
int64_t lastPosition;
int64_t lastPositionTimeStamp;
int64_t lastCompensativePosition;
};
static void
play_callback(SLPlayItf caller, void * user_ptr, SLuint32 event)
{
cubeb_stream * stm = user_ptr;
int draining;
assert(stm);
switch (event) {
case SL_PLAYEVENT_HEADATMARKER:
pthread_mutex_lock(&stm->mutex);
draining = stm->draining;
pthread_mutex_unlock(&stm->mutex);
if (draining) {
stm->state_callback(stm, stm->user_ptr, CUBEB_STATE_DRAINED);
(*stm->play)->SetPlayState(stm->play, SL_PLAYSTATE_PAUSED);
}
break;
default:
break;
}
}
static void
bufferqueue_callback(SLBufferQueueItf caller, void * user_ptr)
{
cubeb_stream * stm = user_ptr;
assert(stm);
SLBufferQueueState state;
SLresult res;
res = (*stm->bufq)->GetState(stm->bufq, &state);
assert(res == SL_RESULT_SUCCESS);
if (state.count > 1)
return;
SLuint32 i;
for (i = state.count; i < NBUFS; i++) {
uint8_t *buf = stm->queuebuf[stm->queuebuf_idx];
long written = 0;
pthread_mutex_lock(&stm->mutex);
int draining = stm->draining;
pthread_mutex_unlock(&stm->mutex);
if (!draining) {
written = cubeb_resampler_fill(stm->resampler,
NULL, NULL,
buf, stm->queuebuf_len / stm->framesize);
if (written < 0 || written * stm->framesize > stm->queuebuf_len) {
(*stm->play)->SetPlayState(stm->play, SL_PLAYSTATE_PAUSED);
return;
}
}
// Keep sending silent data even in draining mode to prevent the audio
// back-end from being stopped automatically by OpenSL/ES.
memset(buf + written * stm->framesize, 0, stm->queuebuf_len - written * stm->framesize);
res = (*stm->bufq)->Enqueue(stm->bufq, buf, stm->queuebuf_len);
assert(res == SL_RESULT_SUCCESS);
stm->queuebuf_idx = (stm->queuebuf_idx + 1) % NBUFS;
if (written > 0) {
pthread_mutex_lock(&stm->mutex);
stm->written += written;
pthread_mutex_unlock(&stm->mutex);
}
if (!draining && written * stm->framesize < stm->queuebuf_len) {
pthread_mutex_lock(&stm->mutex);
int64_t written_duration = INT64_C(1000) * stm->written * stm->framesize / stm->bytespersec;
stm->draining = 1;
pthread_mutex_unlock(&stm->mutex);
// Use SL_PLAYEVENT_HEADATMARKER event from slPlayCallback of SLPlayItf
// to make sure all the data has been processed.
(*stm->play)->SetMarkerPosition(stm->play, (SLmillisecond)written_duration);
return;
}
}
}
#if defined(__ANDROID__)
static SLuint32
convert_stream_type_to_sl_stream(cubeb_stream_type stream_type)
{
switch(stream_type) {
case CUBEB_STREAM_TYPE_SYSTEM:
return SL_ANDROID_STREAM_SYSTEM;
case CUBEB_STREAM_TYPE_MUSIC:
return SL_ANDROID_STREAM_MEDIA;
case CUBEB_STREAM_TYPE_NOTIFICATION:
return SL_ANDROID_STREAM_NOTIFICATION;
case CUBEB_STREAM_TYPE_ALARM:
return SL_ANDROID_STREAM_ALARM;
case CUBEB_STREAM_TYPE_VOICE_CALL:
return SL_ANDROID_STREAM_VOICE;
case CUBEB_STREAM_TYPE_RING:
return SL_ANDROID_STREAM_RING;
case CUBEB_STREAM_TYPE_SYSTEM_ENFORCED:
return SL_ANDROID_STREAM_SYSTEM_ENFORCED;
default:
return 0xFFFFFFFF;
}
}
#endif
static void opensl_destroy(cubeb * ctx);
#if defined(__ANDROID__)
#if (__ANDROID_API__ >= ANDROID_VERSION_LOLLIPOP)
typedef int (system_property_get)(const char*, char*);
static int
__system_property_get(const char* name, char* value)
{
void* libc = dlopen("libc.so", RTLD_LAZY);
if (!libc) {
LOG("Failed to open libc.so");
return -1;
}
system_property_get* func = (system_property_get*)
dlsym(libc, "__system_property_get");
int ret = -1;
if (func) {
ret = func(name, value);
}
dlclose(libc);
return ret;
}
#endif
static int
get_android_version(void)
{
char version_string[PROP_VALUE_MAX];
memset(version_string, 0, PROP_VALUE_MAX);
int len = __system_property_get("ro.build.version.sdk", version_string);
if (len <= 0) {
LOG("Failed to get Android version!\n");
return len;
}
int version = (int)strtol(version_string, NULL, 10);
LOG("%d", version);
return version;
}
#endif
/*static*/ int
opensl_init(cubeb ** context, char const * context_name)
{
cubeb * ctx;
#if defined(__ANDROID__)
int android_version = get_android_version();
if (android_version > 0 && android_version <= ANDROID_VERSION_GINGERBREAD_MR1) {
// Don't even attempt to run on Gingerbread and lower
return CUBEB_ERROR;
}
#endif
*context = NULL;
ctx = calloc(1, sizeof(*ctx));
assert(ctx);
ctx->ops = &opensl_ops;
ctx->lib = dlopen("libOpenSLES.so", RTLD_LAZY);
ctx->libmedia = dlopen("libmedia.so", RTLD_LAZY);
if (!ctx->lib || !ctx->libmedia) {
free(ctx);
return CUBEB_ERROR;
}
/* Get the latency, in ms, from AudioFlinger */
/* status_t AudioSystem::getOutputLatency(uint32_t* latency,
* audio_stream_type_t streamType) */
/* First, try the most recent signature. */
ctx->get_output_latency =
dlsym(ctx->libmedia, "_ZN7android11AudioSystem16getOutputLatencyEPj19audio_stream_type_t");
if (!ctx->get_output_latency) {
/* in case of failure, try the legacy version. */
/* status_t AudioSystem::getOutputLatency(uint32_t* latency,
* int streamType) */
ctx->get_output_latency =
dlsym(ctx->libmedia, "_ZN7android11AudioSystem16getOutputLatencyEPji");
if (!ctx->get_output_latency) {
opensl_destroy(ctx);
return CUBEB_ERROR;
}
}
typedef SLresult (*slCreateEngine_t)(SLObjectItf *,
SLuint32,
const SLEngineOption *,
SLuint32,
const SLInterfaceID *,
const SLboolean *);
slCreateEngine_t f_slCreateEngine =
(slCreateEngine_t)dlsym(ctx->lib, "slCreateEngine");
SLInterfaceID SL_IID_ENGINE = *(SLInterfaceID *)dlsym(ctx->lib, "SL_IID_ENGINE");
SLInterfaceID SL_IID_OUTPUTMIX = *(SLInterfaceID *)dlsym(ctx->lib, "SL_IID_OUTPUTMIX");
ctx->SL_IID_VOLUME = *(SLInterfaceID *)dlsym(ctx->lib, "SL_IID_VOLUME");
ctx->SL_IID_BUFFERQUEUE = *(SLInterfaceID *)dlsym(ctx->lib, "SL_IID_BUFFERQUEUE");
#if defined(__ANDROID__)
ctx->SL_IID_ANDROIDCONFIGURATION = *(SLInterfaceID *)dlsym(ctx->lib, "SL_IID_ANDROIDCONFIGURATION");
#endif
ctx->SL_IID_PLAY = *(SLInterfaceID *)dlsym(ctx->lib, "SL_IID_PLAY");
if (!f_slCreateEngine ||
!SL_IID_ENGINE ||
!SL_IID_OUTPUTMIX ||
!ctx->SL_IID_BUFFERQUEUE ||
#if defined(__ANDROID__)
!ctx->SL_IID_ANDROIDCONFIGURATION ||
#endif
!ctx->SL_IID_PLAY) {
opensl_destroy(ctx);
return CUBEB_ERROR;
}
const SLEngineOption opt[] = {{SL_ENGINEOPTION_THREADSAFE, SL_BOOLEAN_TRUE}};
SLresult res;
res = cubeb_get_sles_engine(&ctx->engObj, 1, opt, 0, NULL, NULL);
if (res != SL_RESULT_SUCCESS) {
opensl_destroy(ctx);
return CUBEB_ERROR;
}
res = cubeb_realize_sles_engine(ctx->engObj);
if (res != SL_RESULT_SUCCESS) {
opensl_destroy(ctx);
return CUBEB_ERROR;
}
res = (*ctx->engObj)->GetInterface(ctx->engObj, SL_IID_ENGINE, &ctx->eng);
if (res != SL_RESULT_SUCCESS) {
opensl_destroy(ctx);
return CUBEB_ERROR;
}
const SLInterfaceID idsom[] = {SL_IID_OUTPUTMIX};
const SLboolean reqom[] = {SL_BOOLEAN_TRUE};
res = (*ctx->eng)->CreateOutputMix(ctx->eng, &ctx->outmixObj, 1, idsom, reqom);
if (res != SL_RESULT_SUCCESS) {
opensl_destroy(ctx);
return CUBEB_ERROR;
}
res = (*ctx->outmixObj)->Realize(ctx->outmixObj, SL_BOOLEAN_FALSE);
if (res != SL_RESULT_SUCCESS) {
opensl_destroy(ctx);
return CUBEB_ERROR;
}
*context = ctx;
return CUBEB_OK;
}
static char const *
opensl_get_backend_id(cubeb * ctx)
{
return "opensl";
}
static int
opensl_get_max_channel_count(cubeb * ctx, uint32_t * max_channels)
{
assert(ctx && max_channels);
/* The android mixer handles up to two channels, see
http://androidxref.com/4.2.2_r1/xref/frameworks/av/services/audioflinger/AudioFlinger.h#67 */
*max_channels = 2;
return CUBEB_OK;
}
static int
opensl_get_preferred_sample_rate(cubeb * ctx, uint32_t * rate)
{
/* https://android.googlesource.com/platform/ndk.git/+/master/docs/opensles/index.html
* We don't want to deal with JNI here (and we don't have Java on b2g anyways),
* so we just dlopen the library and get the two symbols we need. */
int r;
void * libmedia;
uint32_t (*get_primary_output_samplingrate)();
uint32_t (*get_output_samplingrate)(int * samplingRate, int streamType);
libmedia = dlopen("libmedia.so", RTLD_LAZY);
if (!libmedia) {
return CUBEB_ERROR;
}
/* uint32_t AudioSystem::getPrimaryOutputSamplingRate(void) */
get_primary_output_samplingrate =
dlsym(libmedia, "_ZN7android11AudioSystem28getPrimaryOutputSamplingRateEv");
if (!get_primary_output_samplingrate) {
/* fallback to
* status_t AudioSystem::getOutputSamplingRate(int* samplingRate, int streamType)
* if we cannot find getPrimaryOutputSamplingRate. */
get_output_samplingrate =
dlsym(libmedia, "_ZN7android11AudioSystem21getOutputSamplingRateEPj19audio_stream_type_t");
if (!get_output_samplingrate) {
/* Another signature exists, with a int instead of an audio_stream_type_t */
get_output_samplingrate =
dlsym(libmedia, "_ZN7android11AudioSystem21getOutputSamplingRateEPii");
if (!get_output_samplingrate) {
dlclose(libmedia);
return CUBEB_ERROR;
}
}
}
if (get_primary_output_samplingrate) {
*rate = get_primary_output_samplingrate();
} else {
/* We don't really know about the type, here, so we just pass music. */
r = get_output_samplingrate((int *) rate, AUDIO_STREAM_TYPE_MUSIC);
if (r) {
dlclose(libmedia);
return CUBEB_ERROR;
}
}
dlclose(libmedia);
/* Depending on which method we called above, we can get a zero back, yet have
* a non-error return value, especially if the audio system is not
* ready/shutting down (i.e. when we can't get our hand on the AudioFlinger
* thread). */
if (*rate == 0) {
return CUBEB_ERROR;
}
return CUBEB_OK;
}
static int
opensl_get_min_latency(cubeb * ctx, cubeb_stream_params params, uint32_t * latency_frames)
{
/* https://android.googlesource.com/platform/ndk.git/+/master/docs/opensles/index.html
* We don't want to deal with JNI here (and we don't have Java on b2g anyways),
* so we just dlopen the library and get the two symbols we need. */
int r;
void * libmedia;
size_t (*get_primary_output_frame_count)(void);
int (*get_output_frame_count)(size_t * frameCount, int streamType);
uint32_t primary_sampling_rate;
size_t primary_buffer_size;
r = opensl_get_preferred_sample_rate(ctx, &primary_sampling_rate);
if (r) {
return CUBEB_ERROR;
}
libmedia = dlopen("libmedia.so", RTLD_LAZY);
if (!libmedia) {
return CUBEB_ERROR;
}
/* JB variant */
/* size_t AudioSystem::getPrimaryOutputFrameCount(void) */
get_primary_output_frame_count =
dlsym(libmedia, "_ZN7android11AudioSystem26getPrimaryOutputFrameCountEv");
if (!get_primary_output_frame_count) {
/* ICS variant */
/* status_t AudioSystem::getOutputFrameCount(int* frameCount, int streamType) */
get_output_frame_count =
dlsym(libmedia, "_ZN7android11AudioSystem19getOutputFrameCountEPii");
if (!get_output_frame_count) {
dlclose(libmedia);
return CUBEB_ERROR;
}
}
if (get_primary_output_frame_count) {
primary_buffer_size = get_primary_output_frame_count();
} else {
if (get_output_frame_count(&primary_buffer_size, params.stream_type) != 0) {
return CUBEB_ERROR;
}
}
/* To get a fast track in Android's mixer, we need to be at the native
* samplerate, which is device dependant. Some devices might be able to
* resample when playing a fast track, but it's pretty rare. */
*latency_frames = NBUFS * primary_buffer_size;
dlclose(libmedia);
return CUBEB_OK;
}
static void
opensl_destroy(cubeb * ctx)
{
if (ctx->outmixObj)
(*ctx->outmixObj)->Destroy(ctx->outmixObj);
if (ctx->engObj)
cubeb_destroy_sles_engine(&ctx->engObj);
dlclose(ctx->lib);
dlclose(ctx->libmedia);
free(ctx);
}
static void opensl_stream_destroy(cubeb_stream * stm);
static int
opensl_stream_init(cubeb * ctx, cubeb_stream ** stream, char const * stream_name,
cubeb_devid input_device,
cubeb_stream_params * input_stream_params,
cubeb_devid output_device,
cubeb_stream_params * output_stream_params,
unsigned int latency_frames,
cubeb_data_callback data_callback, cubeb_state_callback state_callback,
void * user_ptr)
{
cubeb_stream * stm;
assert(ctx);
assert(!input_stream_params && "not supported");
if (input_device || output_device) {
/* Device selection not yet implemented. */
return CUBEB_ERROR_DEVICE_UNAVAILABLE;
}
*stream = NULL;
SLDataFormat_PCM format;
format.formatType = SL_DATAFORMAT_PCM;
format.numChannels = output_stream_params->channels;
// samplesPerSec is in milliHertz
format.samplesPerSec = output_stream_params->rate * 1000;
format.bitsPerSample = SL_PCMSAMPLEFORMAT_FIXED_16;
format.containerSize = SL_PCMSAMPLEFORMAT_FIXED_16;
format.channelMask = output_stream_params->channels == 1 ?
SL_SPEAKER_FRONT_CENTER :
SL_SPEAKER_FRONT_LEFT | SL_SPEAKER_FRONT_RIGHT;
switch (output_stream_params->format) {
case CUBEB_SAMPLE_S16LE:
format.endianness = SL_BYTEORDER_LITTLEENDIAN;
break;
case CUBEB_SAMPLE_S16BE:
format.endianness = SL_BYTEORDER_BIGENDIAN;
break;
default:
return CUBEB_ERROR_INVALID_FORMAT;
}
stm = calloc(1, sizeof(*stm));
assert(stm);
stm->context = ctx;
stm->data_callback = data_callback;
stm->state_callback = state_callback;
stm->user_ptr = user_ptr;
stm->inputrate = output_stream_params->rate;
stm->latency = latency_frames;
stm->stream_type = output_stream_params->stream_type;
stm->framesize = output_stream_params->channels * sizeof(int16_t);
stm->lastPosition = -1;
stm->lastPositionTimeStamp = 0;
stm->lastCompensativePosition = -1;
int r = pthread_mutex_init(&stm->mutex, NULL);
assert(r == 0);
SLDataLocator_BufferQueue loc_bufq;
loc_bufq.locatorType = SL_DATALOCATOR_BUFFERQUEUE;
loc_bufq.numBuffers = NBUFS;
SLDataSource source;
source.pLocator = &loc_bufq;
source.pFormat = &format;
SLDataLocator_OutputMix loc_outmix;
loc_outmix.locatorType = SL_DATALOCATOR_OUTPUTMIX;
loc_outmix.outputMix = ctx->outmixObj;
SLDataSink sink;
sink.pLocator = &loc_outmix;
sink.pFormat = NULL;
#if defined(__ANDROID__)
const SLInterfaceID ids[] = {ctx->SL_IID_BUFFERQUEUE,
ctx->SL_IID_VOLUME,
ctx->SL_IID_ANDROIDCONFIGURATION};
const SLboolean req[] = {SL_BOOLEAN_TRUE, SL_BOOLEAN_TRUE, SL_BOOLEAN_TRUE};
#else
const SLInterfaceID ids[] = {ctx->SL_IID_BUFFERQUEUE, ctx->SL_IID_VOLUME};
const SLboolean req[] = {SL_BOOLEAN_TRUE, SL_BOOLEAN_TRUE};
#endif
assert(NELEMS(ids) == NELEMS(req));
uint32_t preferred_sampling_rate = stm->inputrate;
#if defined(__ANDROID__)
if (get_android_version() >= ANDROID_VERSION_MARSHMALLOW) {
// Reset preferred samping rate to trigger fallback to native sampling rate.
preferred_sampling_rate = 0;
if (opensl_get_min_latency(ctx, *output_stream_params, &latency_frames) != CUBEB_OK) {
// Default to AudioFlinger's advertised fast track latency of 10ms.
latency_frames = 440;
}
stm->latency = latency_frames;
}
#endif
SLresult res = SL_RESULT_CONTENT_UNSUPPORTED;
if (preferred_sampling_rate) {
res = (*ctx->eng)->CreateAudioPlayer(ctx->eng, &stm->playerObj, &source,
&sink, NELEMS(ids), ids, req);
}
// Sample rate not supported? Try again with primary sample rate!
if (res == SL_RESULT_CONTENT_UNSUPPORTED) {
if (opensl_get_preferred_sample_rate(ctx, &preferred_sampling_rate)) {
opensl_stream_destroy(stm);
return CUBEB_ERROR;
}
format.samplesPerSec = preferred_sampling_rate * 1000;
res = (*ctx->eng)->CreateAudioPlayer(ctx->eng, &stm->playerObj,
&source, &sink, NELEMS(ids), ids, req);
}
if (res != SL_RESULT_SUCCESS) {
opensl_stream_destroy(stm);
return CUBEB_ERROR;
}
stm->outputrate = preferred_sampling_rate;
stm->bytespersec = stm->outputrate * stm->framesize;
stm->queuebuf_len = stm->framesize * latency_frames / NBUFS;
// round up to the next multiple of stm->framesize, if needed.
if (stm->queuebuf_len % stm->framesize) {
stm->queuebuf_len += stm->framesize - (stm->queuebuf_len % stm->framesize);
}
cubeb_stream_params params = *output_stream_params;
params.rate = preferred_sampling_rate;
stm->resampler = cubeb_resampler_create(stm, NULL, &params,
output_stream_params->rate,
data_callback,
user_ptr,
CUBEB_RESAMPLER_QUALITY_DEFAULT);
if (!stm->resampler) {
opensl_stream_destroy(stm);
return CUBEB_ERROR;
}
int i;
for (i = 0; i < NBUFS; i++) {
stm->queuebuf[i] = malloc(stm->queuebuf_len);
assert(stm->queuebuf[i]);
}
#if defined(__ANDROID__)
SLuint32 stream_type = convert_stream_type_to_sl_stream(output_stream_params->stream_type);
if (stream_type != 0xFFFFFFFF) {
SLAndroidConfigurationItf playerConfig;
res = (*stm->playerObj)->GetInterface(stm->playerObj,
ctx->SL_IID_ANDROIDCONFIGURATION, &playerConfig);
res = (*playerConfig)->SetConfiguration(playerConfig,
SL_ANDROID_KEY_STREAM_TYPE, &stream_type, sizeof(SLint32));
if (res != SL_RESULT_SUCCESS) {
opensl_stream_destroy(stm);
return CUBEB_ERROR;
}
}
#endif
res = (*stm->playerObj)->Realize(stm->playerObj, SL_BOOLEAN_FALSE);
if (res != SL_RESULT_SUCCESS) {
opensl_stream_destroy(stm);
return CUBEB_ERROR;
}
res = (*stm->playerObj)->GetInterface(stm->playerObj, ctx->SL_IID_PLAY, &stm->play);
if (res != SL_RESULT_SUCCESS) {
opensl_stream_destroy(stm);
return CUBEB_ERROR;
}
res = (*stm->playerObj)->GetInterface(stm->playerObj, ctx->SL_IID_BUFFERQUEUE,
&stm->bufq);
if (res != SL_RESULT_SUCCESS) {
opensl_stream_destroy(stm);
return CUBEB_ERROR;
}
res = (*stm->playerObj)->GetInterface(stm->playerObj, ctx->SL_IID_VOLUME,
&stm->volume);
if (res != SL_RESULT_SUCCESS) {
opensl_stream_destroy(stm);
return CUBEB_ERROR;
}
res = (*stm->play)->RegisterCallback(stm->play, play_callback, stm);
if (res != SL_RESULT_SUCCESS) {
opensl_stream_destroy(stm);
return CUBEB_ERROR;
}
// Work around wilhelm/AudioTrack badness, bug 1221228
(*stm->play)->SetMarkerPosition(stm->play, (SLmillisecond)0);
res = (*stm->play)->SetCallbackEventsMask(stm->play, (SLuint32)SL_PLAYEVENT_HEADATMARKER);
if (res != SL_RESULT_SUCCESS) {
opensl_stream_destroy(stm);
return CUBEB_ERROR;
}
res = (*stm->bufq)->RegisterCallback(stm->bufq, bufferqueue_callback, stm);
if (res != SL_RESULT_SUCCESS) {
opensl_stream_destroy(stm);
return CUBEB_ERROR;
}
{
// Enqueue a silent frame so once the player becomes playing, the frame
// will be consumed and kick off the buffer queue callback.
// Note the duration of a single frame is less than 1ms. We don't bother
// adjusting the playback position.
uint8_t *buf = stm->queuebuf[stm->queuebuf_idx++];
memset(buf, 0, stm->framesize);
res = (*stm->bufq)->Enqueue(stm->bufq, buf, stm->framesize);
assert(res == SL_RESULT_SUCCESS);
}
*stream = stm;
return CUBEB_OK;
}
static void
opensl_stream_destroy(cubeb_stream * stm)
{
if (stm->playerObj)
(*stm->playerObj)->Destroy(stm->playerObj);
int i;
for (i = 0; i < NBUFS; i++) {
free(stm->queuebuf[i]);
}
pthread_mutex_destroy(&stm->mutex);
cubeb_resampler_destroy(stm->resampler);
free(stm);
}
static int
opensl_stream_start(cubeb_stream * stm)
{
SLresult res = (*stm->play)->SetPlayState(stm->play, SL_PLAYSTATE_PLAYING);
if (res != SL_RESULT_SUCCESS)
return CUBEB_ERROR;
stm->state_callback(stm, stm->user_ptr, CUBEB_STATE_STARTED);
return CUBEB_OK;
}
static int
opensl_stream_stop(cubeb_stream * stm)
{
SLresult res = (*stm->play)->SetPlayState(stm->play, SL_PLAYSTATE_PAUSED);
if (res != SL_RESULT_SUCCESS)
return CUBEB_ERROR;
stm->state_callback(stm, stm->user_ptr, CUBEB_STATE_STOPPED);
return CUBEB_OK;
}
static int
opensl_stream_get_position(cubeb_stream * stm, uint64_t * position)
{
SLmillisecond msec;
uint64_t samplerate;
SLresult res;
int r;
uint32_t mixer_latency;
uint32_t compensation_msec = 0;
res = (*stm->play)->GetPosition(stm->play, &msec);
if (res != SL_RESULT_SUCCESS)
return CUBEB_ERROR;
struct timespec t;
clock_gettime(CLOCK_MONOTONIC, &t);
if(stm->lastPosition == msec) {
compensation_msec =
(t.tv_sec*1000000000LL + t.tv_nsec - stm->lastPositionTimeStamp) / 1000000;
} else {
stm->lastPositionTimeStamp = t.tv_sec*1000000000LL + t.tv_nsec;
stm->lastPosition = msec;
}
samplerate = stm->inputrate;
r = stm->context->get_output_latency(&mixer_latency, stm->stream_type);
if (r) {
return CUBEB_ERROR;
}
pthread_mutex_lock(&stm->mutex);
int64_t maximum_position = stm->written * (int64_t)stm->inputrate / stm->outputrate;
pthread_mutex_unlock(&stm->mutex);
assert(maximum_position >= 0);
if (msec > mixer_latency) {
int64_t unadjusted_position;
if (stm->lastCompensativePosition > msec + compensation_msec) {
// Over compensation, use lastCompensativePosition.
unadjusted_position =
samplerate * (stm->lastCompensativePosition - mixer_latency) / 1000;
} else {
unadjusted_position =
samplerate * (msec - mixer_latency + compensation_msec) / 1000;
stm->lastCompensativePosition = msec + compensation_msec;
}
*position = unadjusted_position < maximum_position ?
unadjusted_position : maximum_position;
} else {
*position = 0;
}
return CUBEB_OK;
}
int
opensl_stream_get_latency(cubeb_stream * stm, uint32_t * latency)
{
int r;
uint32_t mixer_latency; // The latency returned by AudioFlinger is in ms.
/* audio_stream_type_t is an int, so this is okay. */
r = stm->context->get_output_latency(&mixer_latency, stm->stream_type);
if (r) {
return CUBEB_ERROR;
}
*latency = stm->latency * stm->inputrate / 1000 + // OpenSL latency
mixer_latency * stm->inputrate / 1000; // AudioFlinger latency
return CUBEB_OK;
}
int
opensl_stream_set_volume(cubeb_stream * stm, float volume)
{
SLresult res;
SLmillibel max_level, millibels;
float unclamped_millibels;
res = (*stm->volume)->GetMaxVolumeLevel(stm->volume, &max_level);
if (res != SL_RESULT_SUCCESS) {
return CUBEB_ERROR;
}
/* millibels are 100*dB, so the conversion from the volume's linear amplitude
* is 100 * 20 * log(volume). However we clamp the resulting value before
* passing it to lroundf() in order to prevent it from silently returning an
* erroneous value when the unclamped value exceeds the size of a long. */
unclamped_millibels = 100.0f * 20.0f * log10f(fmaxf(volume, 0.0f));
unclamped_millibels = fmaxf(unclamped_millibels, SL_MILLIBEL_MIN);
unclamped_millibels = fminf(unclamped_millibels, max_level);
millibels = lroundf(unclamped_millibels);
res = (*stm->volume)->SetVolumeLevel(stm->volume, millibels);
if (res != SL_RESULT_SUCCESS) {
return CUBEB_ERROR;
}
return CUBEB_OK;
}
static struct cubeb_ops const opensl_ops = {
.init = opensl_init,
.get_backend_id = opensl_get_backend_id,
.get_max_channel_count = opensl_get_max_channel_count,
.get_min_latency = opensl_get_min_latency,
.get_preferred_sample_rate = opensl_get_preferred_sample_rate,
.enumerate_devices = NULL,
.destroy = opensl_destroy,
.stream_init = opensl_stream_init,
.stream_destroy = opensl_stream_destroy,
.stream_start = opensl_stream_start,
.stream_stop = opensl_stream_stop,
.stream_get_position = opensl_stream_get_position,
.stream_get_latency = opensl_stream_get_latency,
.stream_set_volume = opensl_stream_set_volume,
.stream_set_panning = NULL,
.stream_get_current_device = NULL,
.stream_device_destroy = NULL,
.stream_register_device_changed_callback = NULL,
.register_device_collection_changed = NULL
};