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|
/* ex: set tabstop=2 shiftwidth=2 expandtab:
* Copyright © 2019 Jan Kelling
*
* This program is made available under an ISC-style license. See the
* accompanying file LICENSE for details.
*/
#include <assert.h>
#include <dlfcn.h>
#include <pthread.h>
#include <errno.h>
#include <time.h>
#include <stdatomic.h>
#include <aaudio/AAudio.h>
#include "cubeb/cubeb.h"
#include "cubeb-internal.h"
#include "cubeb_resampler.h"
#include "cubeb_log.h"
#ifdef DISABLE_LIBAAUDIO_DLOPEN
#define WRAP(x) x
#else
#define WRAP(x) cubeb_##x
#define LIBAAUDIO_API_VISIT(X) \
X(AAudio_convertResultToText) \
X(AAudio_convertStreamStateToText) \
X(AAudio_createStreamBuilder) \
X(AAudioStreamBuilder_openStream) \
X(AAudioStreamBuilder_setChannelCount) \
X(AAudioStreamBuilder_setBufferCapacityInFrames)\
X(AAudioStreamBuilder_setDirection) \
X(AAudioStreamBuilder_setFormat) \
X(AAudioStreamBuilder_setSharingMode) \
X(AAudioStreamBuilder_setPerformanceMode) \
X(AAudioStreamBuilder_setSampleRate) \
X(AAudioStreamBuilder_delete) \
X(AAudioStreamBuilder_setDataCallback) \
X(AAudioStreamBuilder_setErrorCallback) \
X(AAudioStream_close) \
X(AAudioStream_read) \
X(AAudioStream_requestStart) \
X(AAudioStream_requestPause) \
X(AAudioStream_setBufferSizeInFrames) \
X(AAudioStream_getTimestamp) \
X(AAudioStream_requestFlush) \
X(AAudioStream_requestStop) \
X(AAudioStream_getPerformanceMode) \
X(AAudioStream_getSharingMode) \
X(AAudioStream_getBufferSizeInFrames) \
X(AAudioStream_getBufferCapacityInFrames) \
X(AAudioStream_getSampleRate) \
X(AAudioStream_waitForStateChange) \
X(AAudioStream_getFramesRead) \
X(AAudioStream_getState) \
// not needed or added later on
// X(AAudioStreamBuilder_setFramesPerDataCallback) \
// X(AAudioStreamBuilder_setDeviceId) \
// X(AAudioStreamBuilder_setSamplesPerFrame) \
// X(AAudioStream_getSamplesPerFrame) \
// X(AAudioStream_getDeviceId) \
// X(AAudioStream_write) \
// X(AAudioStream_getChannelCount) \
// X(AAudioStream_getFormat) \
// X(AAudioStream_getFramesPerBurst) \
// X(AAudioStream_getFramesWritten) \
// X(AAudioStream_getXRunCount) \
// X(AAudioStream_isMMapUsed) \
// X(AAudioStreamBuilder_setUsage) \
// X(AAudioStreamBuilder_setContentType) \
// X(AAudioStreamBuilder_setInputPreset) \
// X(AAudioStreamBuilder_setSessionId) \
// X(AAudioStream_getUsage) \
// X(AAudioStream_getContentType) \
// X(AAudioStream_getInputPreset) \
// X(AAudioStream_getSessionId) \
//
#define MAKE_TYPEDEF(x) static typeof(x) * cubeb_##x;
LIBAAUDIO_API_VISIT(MAKE_TYPEDEF)
#undef MAKE_TYPEDEF
#endif
#define MAX_STREAMS 16
static const struct cubeb_ops aaudio_ops;
enum stream_state {
STREAM_STATE_INIT = 0,
STREAM_STATE_STOPPED,
STREAM_STATE_STOPPING,
STREAM_STATE_STARTED,
STREAM_STATE_STARTING,
STREAM_STATE_DRAINING,
STREAM_STATE_ERROR,
STREAM_STATE_SHUTDOWN,
};
struct cubeb_stream {
/* Note: Must match cubeb_stream layout in cubeb.c. */
cubeb * context;
void * user_ptr;
/**/
_Atomic bool in_use;
_Atomic enum stream_state state;
AAudioStream * ostream;
AAudioStream * istream;
cubeb_data_callback data_callback;
cubeb_state_callback state_callback;
cubeb_resampler * resampler;
// mutex synchronizes access to the stream from the state thread
// and user-called functions. Everything that is accessed in the
// aaudio data (or error) callback is synchronized only via atomics.
pthread_mutex_t mutex;
void * in_buf;
unsigned in_frame_size;
cubeb_sample_format out_format;
_Atomic float volume;
unsigned out_channels;
unsigned out_frame_size;
};
struct cubeb {
struct cubeb_ops const * ops;
void * libaaudio;
struct {
// The state thread: it waits for state changes and stops
// drained streams.
pthread_t thread;
pthread_t notifier;
pthread_mutex_t mutex;
pthread_cond_t cond;
_Atomic bool join;
_Atomic bool waiting;
} state;
// streams[i].in_use signals whether a stream is used
struct cubeb_stream streams[MAX_STREAMS];
};
// Only allowed from state thread, while mutex on stm is locked
static void shutdown(cubeb_stream * stm)
{
if (stm->istream) {
WRAP(AAudioStream_requestStop)(stm->istream);
}
if (stm->ostream) {
WRAP(AAudioStream_requestStop)(stm->ostream);
}
stm->state_callback(stm, stm->user_ptr, CUBEB_STATE_ERROR);
atomic_store(&stm->state, STREAM_STATE_SHUTDOWN);
}
// Returns whether the given state is one in which we wait for
// an asynchronous change
static bool waiting_state(enum stream_state state)
{
switch (state) {
case STREAM_STATE_DRAINING:
case STREAM_STATE_STARTING:
case STREAM_STATE_STOPPING:
return true;
default:
return false;
}
}
// Returns whether this stream is still waiting for a state change
static void update_state(cubeb_stream * stm)
{
// fast path for streams that don't wait for state change or are invalid
enum stream_state old_state = atomic_load(&stm->state);
if (old_state == STREAM_STATE_INIT ||
old_state == STREAM_STATE_STARTED ||
old_state == STREAM_STATE_STOPPED ||
old_state == STREAM_STATE_SHUTDOWN) {
return;
}
// If the main thread currently operates on this thread, we don't
// have to wait for it
int err = pthread_mutex_trylock(&stm->mutex);
if (err != 0) {
if (err != EBUSY) {
LOG("pthread_mutex_trylock: %s", strerror(err));
}
return;
}
// check again: if this is true now, the stream was destroyed or
// changed between our fast path check and locking the mutex
old_state = atomic_load(&stm->state);
if (old_state == STREAM_STATE_INIT ||
old_state == STREAM_STATE_STARTED ||
old_state == STREAM_STATE_STOPPED ||
old_state == STREAM_STATE_SHUTDOWN) {
pthread_mutex_unlock(&stm->mutex);
return;
}
// We compute the new state the stream has and then compare_exchange it
// if it has changed. This way we will never just overwrite state
// changes that were set from the audio thread in the meantime,
// such as a draining or error state.
enum stream_state new_state;
do {
if (old_state == STREAM_STATE_SHUTDOWN) {
pthread_mutex_unlock(&stm->mutex);
return;
}
if (old_state == STREAM_STATE_ERROR) {
shutdown(stm);
pthread_mutex_unlock(&stm->mutex);
return;
}
new_state = old_state;
aaudio_stream_state_t istate = 0;
aaudio_stream_state_t ostate = 0;
aaudio_result_t res;
if (stm->istream) {
res = WRAP(AAudioStream_waitForStateChange)(stm->istream,
AAUDIO_STREAM_STATE_UNKNOWN, &istate, 0);
if (res != AAUDIO_OK) {
pthread_mutex_unlock(&stm->mutex);
LOG("AAudioStream_waitForStateChanged: %s", WRAP(AAudio_convertResultToText)(res));
return;
}
assert(istate);
}
if (stm->ostream) {
res = WRAP(AAudioStream_waitForStateChange)(stm->ostream,
AAUDIO_STREAM_STATE_UNKNOWN, &ostate, 0);
if (res != AAUDIO_OK) {
pthread_mutex_unlock(&stm->mutex);
LOG("AAudioStream_waitForStateChanged: %s", WRAP(AAudio_convertResultToText)(res));
return;
}
assert(ostate);
}
// handle invalid stream states
if (istate == AAUDIO_STREAM_STATE_PAUSING ||
istate == AAUDIO_STREAM_STATE_PAUSED ||
istate == AAUDIO_STREAM_STATE_FLUSHING ||
istate == AAUDIO_STREAM_STATE_FLUSHED ||
istate == AAUDIO_STREAM_STATE_UNKNOWN ||
istate == AAUDIO_STREAM_STATE_DISCONNECTED) {
const char * name = WRAP(AAudio_convertStreamStateToText)(istate);
LOG("Invalid android input stream state %s", name);
shutdown(stm);
pthread_mutex_unlock(&stm->mutex);
return;
}
if (ostate == AAUDIO_STREAM_STATE_PAUSING ||
ostate == AAUDIO_STREAM_STATE_PAUSED ||
ostate == AAUDIO_STREAM_STATE_FLUSHING ||
ostate == AAUDIO_STREAM_STATE_FLUSHED ||
ostate == AAUDIO_STREAM_STATE_UNKNOWN ||
ostate == AAUDIO_STREAM_STATE_DISCONNECTED) {
const char * name = WRAP(AAudio_convertStreamStateToText)(istate);
LOG("Invalid android output stream state %s", name);
shutdown(stm);
pthread_mutex_unlock(&stm->mutex);
return;
}
switch (old_state) {
case STREAM_STATE_STARTING:
if ((!istate || istate == AAUDIO_STREAM_STATE_STARTED) &&
(!ostate || ostate == AAUDIO_STREAM_STATE_STARTED)) {
stm->state_callback(stm, stm->user_ptr, CUBEB_STATE_STARTED);
new_state = STREAM_STATE_STARTED;
}
break;
case STREAM_STATE_DRAINING:
// The DRAINING state means that we want to stop the streams but
// may not have done so yet.
// The aaudio docs state that returning STOP from the callback isn't
// enough, the stream has to be stopped from another thread
// afterwards.
// No callbacks are triggered anymore when requestStop returns.
// That is important as we otherwise might read from a closed istream
// for a duplex stream.
// Therefor it is important to close ostream first.
if (ostate &&
ostate != AAUDIO_STREAM_STATE_STOPPING &&
ostate != AAUDIO_STREAM_STATE_STOPPED) {
res = WRAP(AAudioStream_requestStop)(stm->ostream);
if (res != AAUDIO_OK) {
LOG("AAudioStream_requestStop: %s", WRAP(AAudio_convertResultToText)(res));
return;
}
}
if (istate &&
istate != AAUDIO_STREAM_STATE_STOPPING &&
istate != AAUDIO_STREAM_STATE_STOPPED) {
res = WRAP(AAudioStream_requestStop)(stm->istream);
if (res != AAUDIO_OK) {
LOG("AAudioStream_requestStop: %s", WRAP(AAudio_convertResultToText)(res));
return;
}
}
// we always wait until both streams are stopped until we
// send STATE_DRAINED. Then we can directly transition
// our logical state to STREAM_STATE_STOPPED, not triggering
// an additional STATE_STOPPED callback
if ((!ostate || ostate == AAUDIO_STREAM_STATE_STOPPED) &&
(!istate || istate == AAUDIO_STREAM_STATE_STOPPED)) {
new_state = STREAM_STATE_STOPPED;
stm->state_callback(stm, stm->user_ptr, CUBEB_STATE_DRAINED);
}
break;
case STREAM_STATE_STOPPING:
assert(!istate ||
istate == AAUDIO_STREAM_STATE_STOPPING ||
istate == AAUDIO_STREAM_STATE_STOPPED);
assert(!ostate ||
ostate == AAUDIO_STREAM_STATE_STOPPING ||
ostate == AAUDIO_STREAM_STATE_STOPPED);
if ((!istate || istate == AAUDIO_STREAM_STATE_STOPPED) &&
(!ostate || ostate == AAUDIO_STREAM_STATE_STOPPED)) {
stm->state_callback(stm, stm->user_ptr, CUBEB_STATE_STOPPED);
new_state = STREAM_STATE_STOPPED;
}
break;
default:
assert(false && "Unreachable: invalid state");
}
} while (old_state != new_state &&
!atomic_compare_exchange_strong(&stm->state, &old_state, new_state));
pthread_mutex_unlock(&stm->mutex);
}
static void * notifier_thread(void * user_ptr)
{
cubeb * ctx = (cubeb*) user_ptr;
pthread_mutex_lock(&ctx->state.mutex);
while (!atomic_load(&ctx->state.join)) {
pthread_cond_wait(&ctx->state.cond, &ctx->state.mutex);
if (atomic_load(&ctx->state.waiting)) {
pthread_cond_signal(&ctx->state.cond);
}
}
// make sure other thread joins as well
pthread_cond_signal(&ctx->state.cond);
pthread_mutex_unlock(&ctx->state.mutex);
LOG("Exiting notifier thread");
return NULL;
}
static void * state_thread(void * user_ptr)
{
cubeb * ctx = (cubeb*) user_ptr;
pthread_mutex_lock(&ctx->state.mutex);
bool waiting = false;
while (!atomic_load(&ctx->state.join)) {
waiting |= atomic_load(&ctx->state.waiting);
if (waiting) {
atomic_store(&ctx->state.waiting, false);
waiting = false;
for (unsigned i = 0u; i < MAX_STREAMS; ++i) {
cubeb_stream * stm = &ctx->streams[i];
update_state(stm);
waiting |= waiting_state(atomic_load(&stm->state));
}
// state changed from another thread, update again immediately
if (atomic_load(&ctx->state.waiting)) {
waiting = true;
continue;
}
// Not waiting for any change anymore: we can wait on the
// condition variable without timeout
if (!waiting) {
continue;
}
// while any stream is waiting for state change we sleep with regular
// timeouts. But we wake up immediately if signaled.
// This might seem like a poor man's implementation of state change
// waiting but (as of december 2019), the implementation of
// AAudioStream_waitForStateChange is pretty much the same:
// https://android.googlesource.com/platform/frameworks/av/+/refs/heads/master/media/libaaudio/src/core/AudioStream.cpp#277
struct timespec timeout;
// pthread_condattr_setclock isn't available before android api v21.
// pthread condition variables use the realtime clock by default, which
// means that system time changes can interfer with wakeup timeouts.
#if __ANDROID_API__ >= 21
clock_gettime(CLOCK_MONOTONIC, &timeout);
#else
clock_gettime(CLOCK_REALTIME, &timeout);
#endif
timeout.tv_nsec += 5 * 1000 * 1000; // wait 5ms
pthread_cond_timedwait(&ctx->state.cond, &ctx->state.mutex, &timeout);
} else {
pthread_cond_wait(&ctx->state.cond, &ctx->state.mutex);
}
}
// make sure other thread joins as well
pthread_cond_signal(&ctx->state.cond);
pthread_mutex_unlock(&ctx->state.mutex);
LOG("Exiting state thread");
return NULL;
}
static char const *
aaudio_get_backend_id(cubeb * ctx)
{
(void)ctx;
return "aaudio";
}
static int
aaudio_get_max_channel_count(cubeb * ctx, uint32_t * max_channels)
{
assert(ctx && max_channels);
// NOTE: we might get more, AAudio docs don't specify anything.
*max_channels = 2;
return CUBEB_OK;
}
static void
aaudio_destroy(cubeb * ctx)
{
#ifndef NDEBUG
// make sure all streams were destroyed
for (unsigned i = 0u; i < MAX_STREAMS; ++i) {
assert(!atomic_load(&ctx->streams[i].in_use));
}
#endif
// broadcast joining to both threads
// they will additionally signal each other before joining
atomic_store(&ctx->state.join, true);
pthread_cond_broadcast(&ctx->state.cond);
int err;
err = pthread_join(ctx->state.thread, NULL);
if (err != 0) {
LOG("pthread_join: %s", strerror(err));
}
err = pthread_join(ctx->state.notifier, NULL);
if (err != 0) {
LOG("pthread_join: %s", strerror(err));
}
pthread_cond_destroy(&ctx->state.cond);
pthread_mutex_destroy(&ctx->state.mutex);
for (unsigned i = 0u; i < MAX_STREAMS; ++i) {
pthread_mutex_destroy(&ctx->streams[i].mutex);
}
if (ctx->libaaudio) {
dlclose(ctx->libaaudio);
}
free(ctx);
}
/*static*/ int
aaudio_init(cubeb ** context, char const * context_name) {
(void) context_name;
int err;
// load api
void * libaaudio = NULL;
#ifndef DISABLE_LIBAAUDIO_DLOPEN
libaaudio = dlopen("libaaudio.so", RTLD_NOW);
if (!libaaudio) {
return CUBEB_ERROR;
}
#define LOAD(x) { \
WRAP(x) = (typeof(WRAP(x))) (dlsym(libaaudio, #x)); \
if (!WRAP(x)) { \
LOG("AAudio: Failed to load %s", #x); \
dlclose(libaaudio); \
return CUBEB_ERROR; \
} \
}
LIBAAUDIO_API_VISIT(LOAD);
#undef LOAD
#endif
cubeb * ctx = (cubeb*) calloc(1, sizeof(*ctx));
ctx->ops = &aaudio_ops;
ctx->libaaudio = libaaudio;
atomic_init(&ctx->state.join, false);
atomic_init(&ctx->state.waiting, false);
pthread_mutex_init(&ctx->state.mutex, NULL);
pthread_condattr_t cond_attr;
pthread_condattr_init(&cond_attr);
// pthread_condttr_setclock isn't available before android api v21.
// pthread condition variables use the realtime clock by default, which
// means that system time changes can interfer with wakeup timeouts.
#if __ANDROID_API__ >= 21
pthread_condattr_setclock(&cond_attr, CLOCK_MONOTONIC);
#endif
err = pthread_cond_init(&ctx->state.cond, &cond_attr);
pthread_condattr_destroy(&cond_attr);
if (err) {
LOG("pthread_cond_init: %s", strerror(err));
aaudio_destroy(ctx);
return CUBEB_ERROR;
}
// The stream mutexes are not bound to the lifetimes of the
// streams since we need them to synchronize the streams with
// the state thread.
for (unsigned i = 0u; i < MAX_STREAMS; ++i) {
atomic_init(&ctx->streams[i].in_use, false);
atomic_init(&ctx->streams[i].state, STREAM_STATE_INIT);
pthread_mutex_init(&ctx->streams[i].mutex, NULL);
}
err = pthread_create(&ctx->state.thread, NULL, state_thread, ctx);
if (err != 0) {
LOG("pthread_create: %s", strerror(err));
aaudio_destroy(ctx);
return CUBEB_ERROR;
}
// TODO: we could set the priority of the notifier thread lower than
// the priority of the state thread. This way, it's more likely
// that the state thread will be woken up by the condition variable signal
// when both are currently waiting
err = pthread_create(&ctx->state.notifier, NULL, notifier_thread, ctx);
if (err != 0) {
LOG("pthread_create: %s", strerror(err));
aaudio_destroy(ctx);
return CUBEB_ERROR;
}
*context = ctx;
return CUBEB_OK;
}
static void
apply_volume(cubeb_stream * stm, void * audio_data, uint32_t num_frames) {
// optimization: we don't have to change anything in this case
float volume = atomic_load(&stm->volume);
if (volume == 1.f) {
return;
}
switch (stm->out_format) {
case CUBEB_SAMPLE_S16NE:
for (uint32_t i = 0u; i < num_frames * stm->out_channels; ++i) {
((int16_t*)audio_data)[i] *= volume;
}
break;
case CUBEB_SAMPLE_FLOAT32NE:
for (uint32_t i = 0u; i < num_frames * stm->out_channels; ++i) {
((float*)audio_data)[i] *= volume;
}
break;
default:
assert(false && "Unreachable: invalid stream out_format");
}
}
// Returning AAUDIO_CALLBACK_RESULT_STOP seems to put the stream in
// an invalid state. Seems like an AAudio bug/bad documentation.
// We therefore only return it on error.
static aaudio_data_callback_result_t
aaudio_duplex_data_cb(AAudioStream * astream, void * user_data,
void * audio_data, int32_t num_frames)
{
cubeb_stream * stm = (cubeb_stream*) user_data;
assert(stm->ostream == astream);
assert(stm->istream);
assert(num_frames >= 0);
enum stream_state state = atomic_load(&stm->state);
// int istate = WRAP(AAudioStream_getState)(stm->istream);
// int ostate = WRAP(AAudioStream_getState)(stm->ostream);
// ALOGV("aaudio duplex data cb on stream %p: state %ld (in: %d, out: %d), num_frames: %ld",
// (void*) stm, state, istate, ostate, num_frames);
// all other states may happen since the callback might be called
// from within requestStart
assert(state != STREAM_STATE_SHUTDOWN);
// This might happen when we started draining but not yet actually
// stopped the stream from the state thread.
if (state == STREAM_STATE_DRAINING) {
memset(audio_data, 0x0, num_frames * stm->out_frame_size);
return AAUDIO_CALLBACK_RESULT_CONTINUE;
}
long in_num_frames = WRAP(AAudioStream_read)(stm->istream,
stm->in_buf, num_frames, 0);
if (in_num_frames < 0) { // error
atomic_store(&stm->state, STREAM_STATE_ERROR);
LOG("AAudioStream_read: %s", WRAP(AAudio_convertResultToText)(in_num_frames));
return AAUDIO_CALLBACK_RESULT_STOP;
}
// This can happen shortly after starting the stream. AAudio might immediately
// begin to buffer output but not have any input ready yet. We could
// block AAudioStream_read (passing a timeout > 0) but that leads to issues
// since blocking in this callback is a bad idea in general and it might break
// the stream when it is stopped by another thread shortly after being started.
// We therefore simply send silent input to the application.
if (in_num_frames < num_frames) {
// ALOGV("AAudioStream_read returned not enough frames: %ld instead of %d",
// in_num_frames, num_frames);
unsigned left = num_frames - in_num_frames;
char * buf = ((char*) stm->in_buf) + in_num_frames * stm->in_frame_size;
memset(buf, 0x0, left * stm->in_frame_size);
in_num_frames = num_frames;
}
long done_frames = cubeb_resampler_fill(stm->resampler, stm->in_buf,
&in_num_frames, audio_data, num_frames);
if (done_frames < 0 || done_frames > num_frames) {
LOG("Error in data callback or resampler: %ld", done_frames);
atomic_store(&stm->state, STREAM_STATE_ERROR);
return AAUDIO_CALLBACK_RESULT_STOP;
} else if (done_frames < num_frames) {
atomic_store(&stm->state, STREAM_STATE_DRAINING);
atomic_store(&stm->context->state.waiting, true);
pthread_cond_signal(&stm->context->state.cond);
char * begin = ((char*)audio_data) + done_frames * stm->out_frame_size;
memset(begin, 0x0, (num_frames - done_frames) * stm->out_frame_size);
}
apply_volume(stm, audio_data, done_frames);
return AAUDIO_CALLBACK_RESULT_CONTINUE;
}
static aaudio_data_callback_result_t
aaudio_output_data_cb(AAudioStream * astream, void * user_data,
void * audio_data, int32_t num_frames)
{
cubeb_stream * stm = (cubeb_stream*) user_data;
assert(stm->ostream == astream);
assert(!stm->istream);
assert(num_frames >= 0);
enum stream_state state = atomic_load(&stm->state);
// int ostate = WRAP(AAudioStream_getState)(stm->ostream);
// ALOGV("aaudio output data cb on stream %p: state %ld (%d), num_frames: %ld",
// (void*) stm, state, ostate, num_frames);
// all other states may happen since the callback might be called
// from within requestStart
assert(state != STREAM_STATE_SHUTDOWN);
// This might happen when we started draining but not yet actually
// stopped the stream from the state thread.
if (state == STREAM_STATE_DRAINING) {
memset(audio_data, 0x0, num_frames * stm->out_frame_size);
return AAUDIO_CALLBACK_RESULT_CONTINUE;
}
long done_frames = cubeb_resampler_fill(stm->resampler, NULL, NULL,
audio_data, num_frames);
if (done_frames < 0 || done_frames > num_frames) {
LOG("Error in data callback or resampler: %ld", done_frames);
atomic_store(&stm->state, STREAM_STATE_ERROR);
return AAUDIO_CALLBACK_RESULT_STOP;
} else if (done_frames < num_frames) {
atomic_store(&stm->state, STREAM_STATE_DRAINING);
atomic_store(&stm->context->state.waiting, true);
pthread_cond_signal(&stm->context->state.cond);
char * begin = ((char*)audio_data) + done_frames * stm->out_frame_size;
memset(begin, 0x0, (num_frames - done_frames) * stm->out_frame_size);
}
apply_volume(stm, audio_data, done_frames);
return AAUDIO_CALLBACK_RESULT_CONTINUE;
}
static aaudio_data_callback_result_t
aaudio_input_data_cb(AAudioStream * astream, void * user_data,
void * audio_data, int32_t num_frames)
{
cubeb_stream * stm = (cubeb_stream*) user_data;
assert(stm->istream == astream);
assert(!stm->ostream);
assert(num_frames >= 0);
enum stream_state state = atomic_load(&stm->state);
// int istate = WRAP(AAudioStream_getState)(stm->istream);
// ALOGV("aaudio input data cb on stream %p: state %ld (%d), num_frames: %ld",
// (void*) stm, state, istate, num_frames);
// all other states may happen since the callback might be called
// from within requestStart
assert(state != STREAM_STATE_SHUTDOWN);
// This might happen when we started draining but not yet actually
// stopped the stream from the state thread.
if (state == STREAM_STATE_DRAINING) {
return AAUDIO_CALLBACK_RESULT_CONTINUE;
}
long input_frame_count = num_frames;
long done_frames = cubeb_resampler_fill(stm->resampler,
audio_data, &input_frame_count, NULL, 0);
if (done_frames < 0 || done_frames > num_frames) {
LOG("Error in data callback or resampler: %ld", done_frames);
atomic_store(&stm->state, STREAM_STATE_ERROR);
return AAUDIO_CALLBACK_RESULT_STOP;
} else if (done_frames < input_frame_count) {
// we don't really drain an input stream, just have to
// stop it from the state thread. That is signaled via the
// DRAINING state.
atomic_store(&stm->state, STREAM_STATE_DRAINING);
atomic_store(&stm->context->state.waiting, true);
pthread_cond_signal(&stm->context->state.cond);
}
return AAUDIO_CALLBACK_RESULT_CONTINUE;
}
static void
aaudio_error_cb(AAudioStream * astream, void * user_data, aaudio_result_t error)
{
cubeb_stream * stm = (cubeb_stream*) user_data;
assert(stm->ostream == astream || stm->istream == astream);
LOG("AAudio error callback: %s", WRAP(AAudio_convertResultToText)(error));
atomic_store(&stm->state, STREAM_STATE_ERROR);
}
static int
realize_stream(AAudioStreamBuilder * sb, const cubeb_stream_params * params,
AAudioStream ** stream, unsigned * frame_size)
{
aaudio_result_t res;
assert(params->rate);
assert(params->channels);
WRAP(AAudioStreamBuilder_setSampleRate)(sb, params->rate);
WRAP(AAudioStreamBuilder_setChannelCount)(sb, params->channels);
aaudio_format_t fmt;
switch (params->format) {
case CUBEB_SAMPLE_S16NE:
fmt = AAUDIO_FORMAT_PCM_I16;
*frame_size = 2 * params->channels;
break;
case CUBEB_SAMPLE_FLOAT32NE:
fmt = AAUDIO_FORMAT_PCM_FLOAT;
*frame_size = 4 * params->channels;
break;
default:
return CUBEB_ERROR_INVALID_FORMAT;
}
WRAP(AAudioStreamBuilder_setFormat)(sb, fmt);
res = WRAP(AAudioStreamBuilder_openStream)(sb, stream);
if (res == AAUDIO_ERROR_INVALID_FORMAT) {
LOG("AAudio device doesn't support output format %d", fmt);
return CUBEB_ERROR_INVALID_FORMAT;
} else if (params->rate && res == AAUDIO_ERROR_INVALID_RATE) {
// The requested rate is not supported.
// Just try again with default rate, we create a resampler anyways
WRAP(AAudioStreamBuilder_setSampleRate)(sb, AAUDIO_UNSPECIFIED);
res = WRAP(AAudioStreamBuilder_openStream)(sb, stream);
}
// When the app has no permission to record audio (android.permission.RECORD_AUDIO)
// but requested and input stream, this will return INVALID_ARGUMENT.
if (res != AAUDIO_OK) {
LOG("AAudioStreamBuilder_openStream: %s", WRAP(AAudio_convertResultToText)(res));
return CUBEB_ERROR;
}
return CUBEB_OK;
}
static void
aaudio_stream_destroy(cubeb_stream * stm)
{
pthread_mutex_lock(&stm->mutex);
assert(stm->state == STREAM_STATE_STOPPED ||
stm->state == STREAM_STATE_STOPPING ||
stm->state == STREAM_STATE_INIT ||
stm->state == STREAM_STATE_DRAINING ||
stm->state == STREAM_STATE_ERROR ||
stm->state == STREAM_STATE_SHUTDOWN);
aaudio_result_t res;
// No callbacks are triggered anymore when requestStop returns.
// That is important as we otherwise might read from a closed istream
// for a duplex stream.
if (stm->ostream) {
if (stm->state != STREAM_STATE_STOPPED &&
stm->state != STREAM_STATE_STOPPING &&
stm->state != STREAM_STATE_SHUTDOWN) {
res = WRAP(AAudioStream_requestStop)(stm->ostream);
if (res != AAUDIO_OK) {
LOG("AAudioStreamBuilder_requestStop: %s", WRAP(AAudio_convertResultToText)(res));
}
}
WRAP(AAudioStream_close)(stm->ostream);
stm->ostream = NULL;
}
if (stm->istream) {
if (stm->state != STREAM_STATE_STOPPED &&
stm->state != STREAM_STATE_STOPPING &&
stm->state != STREAM_STATE_SHUTDOWN) {
res = WRAP(AAudioStream_requestStop)(stm->istream);
if (res != AAUDIO_OK) {
LOG("AAudioStreamBuilder_requestStop: %s", WRAP(AAudio_convertResultToText)(res));
}
}
WRAP(AAudioStream_close)(stm->istream);
stm->istream = NULL;
}
if (stm->resampler) {
cubeb_resampler_destroy(stm->resampler);
}
if (stm->in_buf) {
free(stm->in_buf);
}
atomic_store(&stm->state, STREAM_STATE_INIT);
pthread_mutex_unlock(&stm->mutex);
atomic_store(&stm->in_use, false);
}
static int
aaudio_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)
{
assert(!input_device);
assert(!output_device);
(void) stream_name;
aaudio_result_t res;
AAudioStreamBuilder * sb;
res = WRAP(AAudio_createStreamBuilder)(&sb);
if (res != AAUDIO_OK) {
LOG("AAudio_createStreamBuilder: %s", WRAP(AAudio_convertResultToText)(res));
return CUBEB_ERROR;
}
// atomically find a free stream.
cubeb_stream * stm = NULL;
for (unsigned i = 0u; i < MAX_STREAMS; ++i) {
// This check is only an optimization, we don't strictly need it
// since we check again after locking the mutex.
if (atomic_load(&ctx->streams[i].in_use)) {
continue;
}
// if this fails with EBUSY, another thread initialized this stream
// between our check of in_use and this.
int err = pthread_mutex_trylock(&ctx->streams[i].mutex);
if (err != 0 || atomic_load(&ctx->streams[i].in_use)) {
if (err && err != EBUSY) {
LOG("pthread_mutex_trylock: %s", strerror(err));
}
continue;
}
stm = &ctx->streams[i];
break;
}
if (!stm) {
LOG("Error: maximum number of streams reached");
return CUBEB_ERROR;
}
assert(atomic_load(&stm->state) == STREAM_STATE_INIT);
atomic_store(&stm->in_use, true);
unsigned res_err = CUBEB_ERROR;
stm->user_ptr = user_ptr;
stm->data_callback = data_callback;
stm->state_callback = state_callback;
stm->context = ctx;
WRAP(AAudioStreamBuilder_setErrorCallback)(sb, aaudio_error_cb, stm);
WRAP(AAudioStreamBuilder_setBufferCapacityInFrames)(sb, latency_frames);
AAudioStream_dataCallback in_data_callback;
AAudioStream_dataCallback out_data_callback;
if (output_stream_params && input_stream_params) {
out_data_callback = aaudio_duplex_data_cb;
in_data_callback = NULL;
} else if (input_stream_params) {
in_data_callback = aaudio_input_data_cb;
} else if (output_stream_params) {
out_data_callback = aaudio_output_data_cb;
} else {
LOG("Tried to open stream without input or output parameters");
goto error;
}
#ifdef AAUDIO_EXCLUSIVE
WRAP(AAudioStreamBuilder_setSharingMode)(sb, AAUDIO_SHARING_MODE_EXCLUSIVE);
#endif
#ifdef AAUDIO_LOW_LATENCY
WRAP(AAudioStreamBuilder_setPerformanceMode)(sb, AAUDIO_PERFORMANCE_MODE_LOW_LATENCY);
#elif defined(AAUDIO_POWER_SAVING)
WRAP(AAudioStreamBuilder_setPerformanceMode)(sb, AAUDIO_PERFORMANCE_MODE_POWER_SAVING);
#endif
unsigned frame_size;
// initialize streams
// output
uint32_t target_sample_rate = 0;
cubeb_stream_params out_params;
if (output_stream_params) {
WRAP(AAudioStreamBuilder_setDirection)(sb, AAUDIO_DIRECTION_OUTPUT);
WRAP(AAudioStreamBuilder_setDataCallback)(sb, out_data_callback, stm);
res_err = realize_stream(sb, output_stream_params, &stm->ostream, &frame_size);
if (res_err) {
goto error;
}
// output debug information
aaudio_sharing_mode_t sm = WRAP(AAudioStream_getSharingMode)(stm->ostream);
aaudio_performance_mode_t pm = WRAP(AAudioStream_getPerformanceMode)(stm->ostream);
int bcap = WRAP(AAudioStream_getBufferCapacityInFrames)(stm->ostream);
int bsize = WRAP(AAudioStream_getBufferSizeInFrames)(stm->ostream);
int rate = WRAP(AAudioStream_getSampleRate)(stm->ostream);
LOG("AAudio output stream sharing mode: %d", sm);
LOG("AAudio output stream performance mode: %d", pm);
LOG("AAudio output stream buffer capacity: %d", bcap);
LOG("AAudio output stream buffer size: %d", bsize);
LOG("AAudio output stream buffer rate: %d", rate);
target_sample_rate = output_stream_params->rate;
out_params = *output_stream_params;
out_params.rate = rate;
stm->out_channels = output_stream_params->channels;
stm->out_format = output_stream_params->format;
stm->out_frame_size = frame_size;
atomic_store(&stm->volume, 1.f);
}
// input
cubeb_stream_params in_params;
if (input_stream_params) {
WRAP(AAudioStreamBuilder_setDirection)(sb, AAUDIO_DIRECTION_INPUT);
WRAP(AAudioStreamBuilder_setDataCallback)(sb, in_data_callback, stm);
res_err = realize_stream(sb, input_stream_params, &stm->istream, &frame_size);
if (res_err) {
goto error;
}
// output debug information
aaudio_sharing_mode_t sm = WRAP(AAudioStream_getSharingMode)(stm->istream);
aaudio_performance_mode_t pm = WRAP(AAudioStream_getPerformanceMode)(stm->istream);
int bcap = WRAP(AAudioStream_getBufferCapacityInFrames)(stm->istream);
int bsize = WRAP(AAudioStream_getBufferSizeInFrames)(stm->istream);
int rate = WRAP(AAudioStream_getSampleRate)(stm->istream);
LOG("AAudio input stream sharing mode: %d", sm);
LOG("AAudio input stream performance mode: %d", pm);
LOG("AAudio input stream buffer capacity: %d", bcap);
LOG("AAudio input stream buffer size: %d", bsize);
LOG("AAudio input stream buffer rate: %d", rate);
stm->in_buf = malloc(bcap * frame_size);
assert(!target_sample_rate || target_sample_rate == input_stream_params->rate);
target_sample_rate = input_stream_params->rate;
in_params = *input_stream_params;
in_params.rate = rate;
stm->in_frame_size = frame_size;
}
// initialize resampler
stm->resampler = cubeb_resampler_create(stm,
input_stream_params ? &in_params : NULL,
output_stream_params ? &out_params : NULL,
target_sample_rate,
data_callback,
user_ptr,
CUBEB_RESAMPLER_QUALITY_DEFAULT);
if (!stm->resampler) {
LOG("Failed to create resampler");
goto error;
}
// the stream isn't started initially. We don't need to differntiate
// between a stream that was just initialized and one that played
// already but was stopped
atomic_store(&stm->state, STREAM_STATE_STOPPED);
LOG("Cubeb stream (%p) init success", (void*) stm);
pthread_mutex_unlock(&stm->mutex);
WRAP(AAudioStreamBuilder_delete)(sb);
*stream = stm;
return CUBEB_OK;
error:
WRAP(AAudioStreamBuilder_delete)(sb);
pthread_mutex_unlock(&stm->mutex);
aaudio_stream_destroy(stm);
return res_err;
}
static int
aaudio_stream_start(cubeb_stream * stm)
{
pthread_mutex_lock(&stm->mutex);
enum stream_state state = atomic_load(&stm->state);
int istate = stm->istream ? WRAP(AAudioStream_getState)(stm->istream) : 0;
int ostate = stm->ostream ? WRAP(AAudioStream_getState)(stm->ostream) : 0;
LOGV("starting stream %p: %d (%d %d)", (void*) stm, state, istate, ostate);
switch (state) {
case STREAM_STATE_STARTED:
case STREAM_STATE_STARTING:
pthread_mutex_unlock(&stm->mutex);
LOG("cubeb stream %p already starting/started", (void*) stm);
return CUBEB_OK;
case STREAM_STATE_ERROR:
case STREAM_STATE_SHUTDOWN:
return CUBEB_ERROR;
case STREAM_STATE_INIT:
pthread_mutex_unlock(&stm->mutex);
assert(false && "Invalid stream");
return CUBEB_ERROR;
case STREAM_STATE_STOPPED:
case STREAM_STATE_STOPPING:
case STREAM_STATE_DRAINING:
break;
}
aaudio_result_t res;
// NOTE: aaudio docs don't state explicitly if we have to do this or
// if we are allowed to call requestStart while the stream is
// in the transient STOPPING state. Works without in testing though
// if (ostate == AAUDIO_STREAM_STATE_STOPPING) {
// res = WRAP(AAudioStream_waitForStateChange)(stm->ostream,
// AAUDIO_STREAM_STATE_STOPPING, &ostate, INT64_MAX);
// if (res != AAUDIO_OK) {
// LOG("AAudioStream_waitForStateChanged: %s", WRAP(AAudio_convertResultToText)(res));
// }
// }
// if (istate == AAUDIO_STREAM_STATE_STOPPING) {
// res = WRAP(AAudioStream_waitForStateChange)(stm->istream,
// AAUDIO_STREAM_STATE_STOPPING, &istate, INT64_MAX);
// if (res != AAUDIO_OK) {
// LOG("AAudioStream_waitForStateChanged: %s", WRAP(AAudio_convertResultToText)(res));
// }
// }
// Important to start istream before ostream.
// As soon as we start ostream, the callbacks might be triggered an we
// might read from istream (on duplex). If istream wasn't started yet
// this is a problem.
if (stm->istream) {
res = WRAP(AAudioStream_requestStart)(stm->istream);
if (res != AAUDIO_OK) {
LOG("AAudioStream_requestStart (istream): %s", WRAP(AAudio_convertResultToText)(res));
atomic_store(&stm->state, STREAM_STATE_ERROR);
pthread_mutex_unlock(&stm->mutex);
return CUBEB_ERROR;
}
}
if (stm->ostream) {
res = WRAP(AAudioStream_requestStart)(stm->ostream);
if (res != AAUDIO_OK) {
LOG("AAudioStream_requestStart (ostream): %s", WRAP(AAudio_convertResultToText)(res));
atomic_store(&stm->state, STREAM_STATE_ERROR);
pthread_mutex_unlock(&stm->mutex);
return CUBEB_ERROR;
}
}
int ret = CUBEB_OK;
bool success;
while (!(success = atomic_compare_exchange_strong(&stm->state, &state, STREAM_STATE_STARTING))) {
// we land here only if the state has changed in the meantime
switch (state) {
// If an error ocurred in the meantime, we can't change that.
// The stream will be stopped when shut down.
case STREAM_STATE_ERROR:
ret = CUBEB_ERROR;
break;
// The only situation in which the state could have switched to draining
// is if the callback was already fired and requested draining. Don't
// overwrite that. It's not an error either though.
case STREAM_STATE_DRAINING:
break;
// If the state switched [draining -> stopping] or [draining/stopping -> stopped]
// in the meantime, we can simply overwrite that since we restarted the stream.
case STREAM_STATE_STOPPING:
case STREAM_STATE_STOPPED:
continue;
// There is no situation in which the state could have been valid before
// but now in shutdown mode, since we hold the streams mutex.
// There is also no way that it switched *into* starting or
// started mode.
default:
assert(false && "Invalid state change");
ret = CUBEB_ERROR;
break;
}
break;
}
if (success) {
atomic_store(&stm->context->state.waiting, true);
pthread_cond_signal(&stm->context->state.cond);
}
pthread_mutex_unlock(&stm->mutex);
return ret;
}
static int
aaudio_stream_stop(cubeb_stream * stm)
{
pthread_mutex_lock(&stm->mutex);
enum stream_state state = atomic_load(&stm->state);
int istate = stm->istream ? WRAP(AAudioStream_getState)(stm->istream) : 0;
int ostate = stm->ostream ? WRAP(AAudioStream_getState)(stm->ostream) : 0;
LOGV("stopping stream %p: %d (%d %d)", (void*) stm, state, istate, ostate);
switch (state) {
case STREAM_STATE_STOPPED:
case STREAM_STATE_STOPPING:
case STREAM_STATE_DRAINING:
pthread_mutex_unlock(&stm->mutex);
LOG("cubeb stream %p already stopping/stopped", (void*) stm);
return CUBEB_OK;
case STREAM_STATE_ERROR:
case STREAM_STATE_SHUTDOWN:
pthread_mutex_unlock(&stm->mutex);
return CUBEB_ERROR;
case STREAM_STATE_INIT:
pthread_mutex_unlock(&stm->mutex);
assert(false && "Invalid stream");
return CUBEB_ERROR;
case STREAM_STATE_STARTED:
case STREAM_STATE_STARTING:
break;
}
aaudio_result_t res;
// NOTE: aaudio docs don't state explicitly if we have to do this or
// if we are allowed to call requestStop while the stream is
// in the transient STARTING state. Works without in testing though.
// if (ostate == AAUDIO_STREAM_STATE_STARTING) {
// res = WRAP(AAudioStream_waitForStateChange)(stm->ostream,
// AAUDIO_STREAM_STATE_STARTING, &ostate, INT64_MAX);
// if (res != AAUDIO_OK) {
// LOG("AAudioStream_waitForStateChanged: %s", WRAP(AAudio_convertResultToText)(res));
// }
// }
// if (istate == AAUDIO_STREAM_STATE_STARTING) {
// res = WRAP(AAudioStream_waitForStateChange)(stm->istream,
// AAUDIO_STREAM_STATE_STARTING, &istate, INT64_MAX);
// if (res != AAUDIO_OK) {
// LOG("AAudioStream_waitForStateChanged: %s", WRAP(AAudio_convertResultToText)(res));
// }
// }
// No callbacks are triggered anymore when requestStop returns.
// That is important as we otherwise might read from a closed istream
// for a duplex stream.
// Therefor it is important to close ostream first.
if (stm->ostream) {
// Could use pause + flush here as well, the public cubeb interface
// doesn't state behavior.
res = WRAP(AAudioStream_requestStop)(stm->ostream);
if (res != AAUDIO_OK) {
LOG("AAudioStream_requestStop (ostream): %s", WRAP(AAudio_convertResultToText)(res));
atomic_store(&stm->state, STREAM_STATE_ERROR);
pthread_mutex_unlock(&stm->mutex);
return CUBEB_ERROR;
}
}
if (stm->istream) {
res = WRAP(AAudioStream_requestStop)(stm->istream);
if (res != AAUDIO_OK) {
LOG("AAudioStream_requestStop (istream): %s", WRAP(AAudio_convertResultToText)(res));
atomic_store(&stm->state, STREAM_STATE_ERROR);
pthread_mutex_unlock(&stm->mutex);
return CUBEB_ERROR;
}
}
int ret = CUBEB_OK;
bool success;
while (!(success = atomic_compare_exchange_strong(&stm->state, &state, STREAM_STATE_STOPPING))) {
// we land here only if the state has changed in the meantime
switch (state) {
// If an error ocurred in the meantime, we can't change that.
// The stream will be stopped when shut down.
case STREAM_STATE_ERROR:
ret = CUBEB_ERROR;
break;
// If it was switched to draining in the meantime, it was or
// will be stopped soon anyways. We don't interfere with
// the draining process, no matter in which state.
// Not an error
case STREAM_STATE_DRAINING:
case STREAM_STATE_STOPPING:
case STREAM_STATE_STOPPED:
break;
// If the state switched from starting to started in the meantime
// we can simply overwrite that since we just stopped it.
case STREAM_STATE_STARTED:
continue;
// There is no situation in which the state could have been valid before
// but now in shutdown mode, since we hold the streams mutex.
// There is also no way that it switched *into* starting mode.
default:
assert(false && "Invalid state change");
ret = CUBEB_ERROR;
break;
}
break;
}
if (success) {
atomic_store(&stm->context->state.waiting, true);
pthread_cond_signal(&stm->context->state.cond);
}
pthread_mutex_unlock(&stm->mutex);
return ret;
}
static int
aaudio_stream_get_position(cubeb_stream * stm, uint64_t * position)
{
pthread_mutex_lock(&stm->mutex);
enum stream_state state = atomic_load(&stm->state);
AAudioStream * stream = stm->ostream ? stm->ostream : stm->istream;
switch (state) {
case STREAM_STATE_ERROR:
case STREAM_STATE_SHUTDOWN:
pthread_mutex_unlock(&stm->mutex);
return CUBEB_ERROR;
case STREAM_STATE_STOPPED:
case STREAM_STATE_DRAINING:
case STREAM_STATE_STOPPING:
// getTimestamp is only valid when the stream is playing.
// Simply return the number of frames passed to aaudio
*position = WRAP(AAudioStream_getFramesRead)(stream);
pthread_mutex_unlock(&stm->mutex);
return CUBEB_OK;
case STREAM_STATE_INIT:
assert(false && "Invalid stream");
pthread_mutex_unlock(&stm->mutex);
return CUBEB_ERROR;
case STREAM_STATE_STARTED:
case STREAM_STATE_STARTING:
break;
}
int64_t pos;
int64_t ns;
aaudio_result_t res;
res = WRAP(AAudioStream_getTimestamp)(stream, CLOCK_MONOTONIC, &pos, &ns);
if (res != AAUDIO_OK) {
// when we are in 'starting' state we try it and hope that the stream
// has internally started and gives us a valid timestamp.
// If that is not the case (invalid_state is returned) we simply
// fall back to the method we use for non-playing streams.
if (res == AAUDIO_ERROR_INVALID_STATE && state == STREAM_STATE_STARTING) {
*position = WRAP(AAudioStream_getFramesRead)(stream);
pthread_mutex_unlock(&stm->mutex);
return CUBEB_OK;
}
LOG("AAudioStream_getTimestamp: %s", WRAP(AAudio_convertResultToText)(res));
pthread_mutex_unlock(&stm->mutex);
return CUBEB_ERROR;
}
pthread_mutex_unlock(&stm->mutex);
*position = pos;
return CUBEB_OK;
}
static int
aaudio_stream_set_volume(cubeb_stream * stm, float volume)
{
assert(stm && stm->ostream);
atomic_store(&stm->volume, volume);
return CUBEB_OK;
}
static const struct cubeb_ops aaudio_ops = {
.init = aaudio_init,
.get_backend_id = aaudio_get_backend_id,
.get_max_channel_count = aaudio_get_max_channel_count,
// NOTE: i guess we could support min_latency and preferred sample
// rate via guessing, i.e. creating a dummy stream and check
// its settings.
.get_min_latency = NULL,
.get_preferred_sample_rate = NULL,
.enumerate_devices = NULL,
.device_collection_destroy = NULL,
.destroy = aaudio_destroy,
.stream_init = aaudio_stream_init,
.stream_destroy = aaudio_stream_destroy,
.stream_start = aaudio_stream_start,
.stream_stop = aaudio_stream_stop,
.stream_reset_default_device = NULL,
.stream_get_position = aaudio_stream_get_position,
// NOTE: this could be implemented via means comparable to the
// OpenSLES backend
.stream_get_latency = NULL,
.stream_set_volume = aaudio_stream_set_volume,
.stream_get_current_device = NULL,
.stream_device_destroy = NULL,
.stream_register_device_changed_callback = NULL,
.register_device_collection_changed = NULL
};
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